irclog2html for #asterisk on 20050217

00:01.39*** join/#asterisk yashax (~yasha_x@69.15.218.218)
00:02.02yashaxguys, what is the "best" distro to lead asterisk?
00:02.23tessier_yashax: Which is better, vi or emacs?
00:02.45tessier_yashax: Jew or Muslim? Protestant or Catholic? Ford or Chevy? Budweiser or Miller?
00:02.47yashaxhahaha.. I knew someone would say something like this....
00:02.58tessier_Trojan or Durex?
00:03.00tessier_(Trojan)
00:03.04yashax:)
00:03.10tessier_Ultra-thin spermicidal please.
00:03.16yashaxwhat would YOU recommend?
00:03.17iMediaxuse your favorite distro
00:03.21tessier_I like to feel it but I don't want no kids.
00:03.29yashaxRH9, FC1?
00:03.33iMediaxsure
00:04.17yashaxWhat kind of install do I need to do... most likely no X-windows, what else not to install?
00:04.47tessier_I've got two boxes of 36 each in my closet. I'm prepared in case a bus load of hot chicks show up at my house and demand that I service each and every one of them.
00:05.26iMediaxlol
00:05.38yashaxI am happy to hear that humor is in the room, but to get back to reality... :)
00:05.51terrapeni wish i was more mechanically-oriented
00:05.56terrapeni need a pickup for my guitar
00:06.05terrapenand this guy wants to charge me $50 to install it
00:06.11terrapenand i know it will take him 20min tops
00:06.14tzangerterrapen: I just buy a guitar with pickups
00:06.15tzanger:-)
00:06.25terrapeni have a Martin acoustic
00:06.30terrapenthey don't come with pickups
00:06.30tzangerI've got a Fender Mustang and a Ibanez V70CE
00:06.32terrapenwell, most don't
00:06.46yashaxguys, anyone?
00:07.04tzangeryashax: try building
00:07.10tessier_I'm not sure I ever have.
00:07.11tzangerif it doesn't work, you took out too much
00:07.14tzangertessier_: ??
00:07.20terrapeni'm wanting to go to this open mic night tomorrow
00:07.24tzangereveryone I know with a guitar cna play
00:07.26terrapenand i need something that can be plugged in
00:07.30tzangersome more than others
00:07.38tessier_tzanger: Some more than others?
00:07.49tessier_Anyone can make a bunch of squeeling noise with a guitar it seems.
00:07.50tzangertessier_: well there are different levels of playing ability
00:08.09tessier_I'd like to meet someone who can play something that sounds nice.
00:08.28tessier_I wish I could play classical spanish guitar.
00:08.35tessier_I think that would be a real chick getter.
00:08.38tzangertessier_: well yeah but I'm talking something melodic
00:08.46tzangertessier_: "sounds nice" is also relative
00:08.53*** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk)
00:09.03tzangerI play some classic rock and some blues and bluegrass but not much else
00:09.08tzangerwant to learn more blues
00:09.17terrapeni play country, bluegrass, folk, rock
00:09.25tessier_I want to play like El Mariachi
00:09.28tessier_(Gypsy Kings)
00:09.37terrapennot Nashville radio-style country though
00:09.48terrapenthe kind of country i play, you've probably never heard anything like it
00:09.50terrapenunless you live in texas
00:09.51tessier_o/~ Soy un hombre muy honrado, que me gusta lo major. A mujeres no me faltan, ni el dinero ni el amor. o/~
00:09.54terrapenor oklahoma
00:10.00tessier_mejor
00:10.01__Sparks_Can some kind soul tell me what ports beside 5060 and 10000-20000 (UDP) I need to forward to my asterisk box to make sipgate calls work properly! (Currently I cant hear the caller!)
00:10.15yashaxIs there a good doc available somewhere on the installation? Please help!
00:10.31tessier___Sparks_: asterisk is behind nat? Yikes.
00:10.43__Sparks_yea :S
00:10.57__Sparks_surely port forwading can work!?
00:11.35terrapentzanger, where are you located
00:12.17terrapenbwahahah
00:12.17terrapenIn accordance with NETCOM guidance 2004-11 (https://www.us.army.mil/suite/doc/1229431), AKO has begun stripping attachments with the following suffixes:
00:12.17terrapen.b64,.bat,.bhx,.ceo,.ce0,.cpl,.dbx,.dll,.dot,.eml,.exe,.hqx,.lnk,.mim,.nch,.ocx,
00:12.17terrapen.pi,.pif,.scr,.sct,.uue,.uu,.vbe,.vbs,.wsc,.wsf,.wsh,.xxe, and .zip.
00:12.17terrapenSince this is an Army policy, AKO will not be able to grant exceptions - please do not call the AKO help desk on this issue, as they will not be able to help you.
00:12.19terrapenWe regret any inconvenience.
00:12.21terrapenAKO removed an attachment due to NETCOM 2004-11 restrictions.
00:12.30terrapenmy buddy tried to send me something
00:12.32tzangerterrapen: midwestern ontario, canada
00:12.38ariel___Sparks_, the other ports will be for iax2 4569 & 5061 for a 2nd sip account.
00:12.49*** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net)
00:13.14tzangerterrapen: I dunno about that, I know that kind of country
00:13.22tzangerI'm not a nashville country kind of guy
00:13.33__Sparks_ariel_, Thanks, I will try them now :)
00:13.44terrapeni hate nashvegas country
00:13.48terrapenit sucks
00:13.58yashaxGuys, anyone? Is there a good doc available somewhere on the installation? Please help!
00:14.05tzangeryashax: TRY IT
00:14.06tzangerjesus
00:14.09tzangertry a basic install
00:14.22ariel_yashax, what would like to do from scratch install?
00:14.22tzangeryou OBVIOUSLY have not enough experience to judge what is and isn't needed
00:14.30tzangerso get yourself a basic install and GET SOME EXPERIENCE
00:14.36eKo1yashax: make && make install
00:14.37tzanger**THEN** optimize the system
00:14.38yashaxyes, from scratch....
00:14.47tzangerpremature optimization is the mother of all fuckups
00:14.59ariel_yashax, if your starting out get yourself the iso from asterisk@home it will get you started.
00:15.07ariel_Then read up on the wiki and
00:15.13ariel_~doc
00:15.21tzangera 5G HDD will install practically everything you want outside of X, which you already know you don't want
00:15.25yashaxariel: Thank you SO much....
00:15.26tzangerand I dare you to find a brand new 5G HDD
00:15.30ariel_I guess the link is not working.
00:16.07ariel_http://www.voip-info.org/wiki-Asterisk,
00:16.31ariel_http://asteriskathome.sourceforge.net/
00:17.15yashaxTHANK YOU!!!!
00:17.31*** part/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk)
00:17.33*** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk)
00:17.44*** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk)
00:20.08terrapenyeah, tzanger, the artists here don't get up to .ca very often
00:20.27*** join/#asterisk Legend (~legend@24.244.142.133)
00:20.29tzafrir_homeactually, the installation is not too complicated. And the installation is generally easy to automate. It is the configuration that is complicated
00:21.30tzafrir_homeAnd with a 5G you can even have both gnome and kde, to burn precious cpu cycles on
00:23.02*** join/#asterisk Mneumonic (~Mnemonic@206.231.230.230)
00:23.42ariel_tzafrir_home, I have a test system running asterisk with only a 4 gig wd drive.
00:23.54*** join/#asterisk neuro_[rus] (~neuro_[ru@212.176.51.231)
00:23.55tzanger:-)
00:23.56MneumonicAnyone know how to set up overhead paging thru the sound card in *?
00:24.01ariel_it's a p1 233 with 128mg ram.
00:24.06tzangermy * installs are nder 500M and that's not trying ot scrape much out
00:24.10tzangerI could probably get it in 32M
00:24.13ariel_Mneumonic, no don't do it.
00:24.26greg_workariel_: ?
00:24.29Mneumonicariel - why not? is there a better way?
00:25.07greg_workMneumonic: configure alsa, Dial(console/dsp)
00:25.14tzangeryeah, is there a better way?
00:25.28ariel_overhead paging I do it via an fxo port and a pager from viking.
00:25.34Mneumonichow do i configure alsa?
00:25.37tzangeryou'd want a relay on the parallel port or serila to engage a paging relay but anyway
00:25.44fafniris that you johnny?
00:25.50Mneumonicnope
00:25.51fafnirjohnny mnuemonic?
00:25.56fafnirlong time no see!
00:26.01Mneumonic:)
00:26.05fafnirheh
00:26.11fafnirthat movie is on ondemand
00:26.17Mneumoniccool
00:26.21fafnirfor the next 24 hours
00:26.25Mneumonicthat movie was way underbudgeted
00:26.31fafnirit came out aight
00:26.35ariel_great movie
00:26.38fafniri liked the dolphin
00:26.53fafnirheh
00:26.57fafnirgo to joohns
00:27.07fafnirer jones
00:27.33Mneumonicyou gotta hack your own brain.. the damn dolphin cant help u
00:28.17greg_worki bought a few $15 computer speakers (the $10 ones had an internal transfomer..), put them in the ceiling, and then ran a 2-pair cable back to my electrial room for each.. one pair carries power from a transformer in the server room, the other pair is connected to a 1/8" jack that goes into the soundcard. each set is plugged into a different output, so i can control volumes of each zone in software (ie, reception area is front, hall
00:28.17greg_workway is rear L, kitchen is rear R, warehouse is center)
00:28.58Mneumonicand u have that running thru the sound card?
00:30.25rikstaanyone ever implemented a system with a * box that can have like, tele-marketers working from home, that log in and calls are originated to them?
00:30.49tzangernothing official riksta
00:30.54tzangerwouldn't be too tricky though
00:31.04rikstayeah im sat here planning a system out
00:31.07tzangersomeone at the toronto asterisk meetup was talking about that
00:31.13rikstawas wondering if anyone has any input
00:34.39greg_workMneumonic: yes, thats why i said "the other pair is connected to a 1/8" jack that goes into the soundcard."
00:38.18Darwin35what happen to res_sqlite ?
00:38.35*** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk)
00:41.16*** join/#asterisk {zombie} (zombie@soulasylum.penguincare.com.au)
00:42.11*** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk)
00:44.53*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
00:44.55puzzledhello
00:45.26iMediaxtele-marketing is evil
00:45.35CoaxDI hate telemarketers
00:45.40*** join/#asterisk qwerp (~abc@219.95.105.74)
00:45.48qwerpharlo
00:46.40*** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu)
00:48.26stoneflyDoes anyone have any recomendations on t1 echo cans?
00:50.15ariel_t1 echo cans....wow.
00:50.31qwerpanybody can help me on a TE110p ?
00:51.06*** join/#asterisk Weezey (~Weezey@206.210.109.226)
00:51.16Weezeyhow do I make my SPA-3000 reset?
00:52.04puzzledstonefly: think on the list a while back the products from tellabs were mentioned. at the time some where available on eBay
00:53.59*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
00:56.23stoneflypuzzled, thanx I check the list out... I was just wondering if any had actually used one..
00:56.45*** join/#asterisk ReVoK (ReVoK@82.224.60.46)
00:56.47ReVoKhi
00:57.38puzzledstonefly: np. if you hook up directly to a telco with e.g. a t1 and go straight over to their tdm network then they will already have the echo cans in place. why do you need them?
00:58.23stoneflypuzzled, really? I'm still getting some bad echo on some incomming calls on a t1 from tellamerica
00:58.49stoneflyThe T100p isn't sharing irqs either...
00:59.23puzzledstonefly: hmm, what was the thing again: if *you* hear echo, it is caused by the other side and vice versa? not sure of an echo can on your side would fix it if I had it right
01:00.24stoneflyI hear echo when I talk, so it is fixable....
01:00.39tzafrir_homegrep , /proc/interrupts
01:01.06puzzledstonefly: ok, hope you figure it out. voodoo like changing the card around or using another mobo has helped people in the past iirc
01:01.09stoneflyI've got echocancel, and echocancelwhenbridged  set to yes and echotraining=800
01:01.20tzafrir_homethis will show all of those *evil* irq-sharaes
01:01.37stoneflytzafrir_home, everything is on its own irq....
01:02.17stoneflyI'll have to try changing slos, but I don't think changing mb's is possible...
01:02.25stoneflyslos=slots...
01:02.44puzzledstonefly: for voodoo's sake you may want to try sticking the card in another pci slot or sticking it in an entirely different box just to rule out funky stuff
01:02.46stoneflyumm, I can't change slots.. there is only one pci slot...
01:03.04stoneflyits a 1u server...
01:04.04MocYES !!! YES !!!
01:04.38zigmannice
01:04.42puzzledMoc: on the list he said he was working on it
01:04.49ariel_wounder if he knows this yet?
01:05.02puzzledan extra $1000 would motivate me :)
01:05.08Mocpuzzled ? what ?
01:05.17*** join/#asterisk IsMe (~some@219.95.224.115)
01:05.32Darwin35http://www.konceptusa.com/index.php?page=wifi_phone
01:05.33puzzledMoc: your "YES" was about the T.38 support?
01:05.38Mocnope hehe
01:05.42puzzledah ok
01:06.03qwerptzafrir: harlo..
01:06.08redder86can callgroup be set from the dialplan?
01:06.33rikstaanyone know if any VOIP providers in the UK can do TPS screening on outgoing calls routed through SIP ?
01:07.11stoneflyDoes SMP cause problems with echo?
01:07.18stoneflyDoes=do
01:07.33stoneflyI can't type today.... it's time to go home...
01:07.47puzzledstonefly: dunno but can't you test it by booting with a UP kernel?
01:10.32*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
01:10.41puzzledhi ManxPower
01:11.16stoneflypuzzled, yeah... will do...
01:11.57tzangerevening manx
01:12.17ManxPower~docs
01:12.19jbotDocumentation 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
01:13.27*** join/#asterisk MrEntropy (~entropy@ppp55-252.lns1.adl2.internode.on.net)
01:13.28MrEntropyyo
01:13.40tzangeryo
01:13.45NormAstHi all.
01:13.59tzangerwerd norm
01:14.12redder86is there a way to set up a pseudo-extension?  Something that I can Dial() but that will never answer the call?
01:15.43ManxPowerredder86, what are you trying to accomplish?
01:17.09redder86ManxPower: I want to use callgroup/pickupgroup, but I don't like having to key callgroup onto a phone in sip.conf.  I'd love it if callgroup could be set via the dialplan.
01:17.31redder86ManxPower: so I'd like to Dial() an imaginary SIP phone that has callgroup set.
01:18.01ZX81hi NormAst
01:18.06ZX81hi all
01:18.47ManxPowerredder86, So you can #8 it?
01:19.42ZX81how long till 2PM CST Feb17th?
01:19.53ZX81ne1?
01:20.01redder86ManxPower: yes
01:20.12redder86ManxPower: but it's *8
01:20.53ZX81redder86: app_changrap.c
01:20.54ZX81:)
01:21.00ZX81but I dunno where it is
01:21.04ZX81I need it too
01:21.10ZX81so I can do call pickup from manager
01:21.11ZX81:)
01:21.31redder86ManxPower: I have incoming calls to the "main company number" go to a group of phones, but sometimes people are not there to answer (like on a weekend) and someone in one of the nearby offices where the phone isn't ringing would like to pick up the phone, but doesn't want to run into the "floor" area to answer it.
01:22.06ZX81Big News: Asterisk Daily News and Asterisk Documentation Project link up through cross-syndication ( http://www.sineapps.com/news.php )
01:22.11ZX81:)
01:22.25redder86ManxPower: right now I have to "key" off of one of the "floor" phones, but I don't like that because the "floor" phones all have their own extensions, and I don't want someone intercepting a direct call.
01:22.48shmaltz~seen ManxPower
01:22.50jbotmanxpower is currently on #asterisk (12m 18s).  Has said a total of 3 messages.  Is idling for 4m 3s
01:23.04shmaltzHi, ManxPower
01:23.22ZX81/ignore lists?
01:23.23*** part/#asterisk IsMe (~some@219.95.224.115)
01:23.24ZX81:)
01:23.53QwellZX81: About 18.5 hours
01:24.00ZX81ok cool
01:24.01ZX81ta
01:24.10ZX81enough time to sleep and wakeup
01:24.11ZX81:)
01:24.16*** join/#asterisk Guest^DJ (~some@219.95.224.115)
01:24.32QwellI can barely do conversion from PST to CST, so I might be off by an hour, heh
01:24.37ZX81:)
01:25.07ZX81fkn voicepulse won't accept my money
01:25.17ZX81says declined on credit card for $10
01:25.21ManxPowerZX81, That's good news.  We need more consolidation of Asterisk documentation sites.
01:25.25ZX81so I went and used it for $4000
01:25.26Guest^DJhey ZX81
01:25.27ZX81ManxPower: yah
01:25.29ZX81hi
01:25.30ZX81:)
01:25.32Guest^DJhi ManxPower
01:25.36shmaltzManxPower, can I ask u a question?
01:25.39ZX81Guest^DJ: how are you?
01:26.00ManxPowershmaltz, You can always ask me a question, but I'm not much in the mood for providing answers at the moment.
01:26.11shmaltzManxPower, Do you still have the weather script?
01:27.21ManxPowershmaltz, All that stuff was donated to the Asterisk Documentation Project because I got tired of answering questions about my site.
01:27.43ManxPowerIf you smile REALLY nice I might give you a link to a tar.gz of the entire site that I used to have on line.
01:28.05shmaltzManxPower, :):):):):):):):):):):):):):):):):):):)::)
01:28.25ManxPowerhttp://www.fnords.org/~eric/asterisk/wffs.tar.gz I think
01:28.32shmaltz. -)
01:28.34shmaltz. -)
01:30.36ZX81and there is a street race outside in a few days
01:30.37rikstais there a howto on making "agents" that can log into asterisk on the wiki
01:30.42ZX81sure
01:30.43ZX81wiki
01:30.47ZX81~voip-info
01:30.48jbotit has been said that voip-info is the Voice Over IP wiki.  It is a community resource which will answer all of your questions, from Asterisk to ZTDummy.  You can find it over at http://www.voip-info.org - well worth bookmarking
01:30.48rikstanice
01:30.57rikstawhat search terms would i use
01:31.45rikstai'm looking to see if you can have agents, that can log out briefly for after call work, or for lunch...is that possible too?
01:31.46ZX81agents?
01:31.47ZX81maybe
01:31.48ZX81:)
01:31.49modulus_use terms like "stuff" and "thingy"
01:31.56ZX81hehe
01:32.19*** join/#asterisk soulz- (~Soulz-@host-137-132-45-204.imcb.nus.edu.sg)
01:32.23soulz-hi all
01:32.27ZX81high soulz-
01:32.38soulz-sup?
01:32.41ManxPower~google site:lists.digium.com stuff thingy
01:32.41ZX81nm
01:32.42ZX81you
01:32.45ZX81lol
01:33.16ZX81no kaboodle?
01:33.39ManxPowerI guess I need to post to the -users mailing list a message with "stuff" and "thingy" in it.
01:33.44ZX81~google manxpower
01:33.45ZX81lol
01:33.53ZX81ta rue
01:33.58ZX81:)
01:34.13ZX81~google site:lists.digium.com stuff
01:34.17ManxPowerThe geocities link has nothing to do with me.
01:34.18ZX81heh
01:34.43soulz-if i have a TDM04B and TDM40B, what will my /etc/zaptel.conf be? fxsks=1-4?
01:34.55ManxPowerI should sue them for trademanrk infringemnet.
01:35.14ZX81lol
01:35.16ZX81:)
01:35.22ManxPowersoulz-, it depends on the order in which you load the drivers I think
01:35.27ZX81yup
01:35.45soulz-i did a modprobe wctdm, and it says http://pastebin.ca/5990
01:36.12soulz-manxpower: it sees all if i do fxsks=1-4
01:36.18ManxPowersoulz-, Ah!  Yes.  Then it would (I THINK) depend on the order the cards are recognized, usually the closest to the power supply will be the first card.
01:36.19ZX81soulz-: so what is the problem?
01:36.36ZX81he has success
01:36.38ZX81:)
01:36.54tzafrir_homewhen in doubt, consult /proc/zaptel
01:36.56soulz-zx81: how do i reference it say whether its a fxs or fxo?
01:37.06shmaltzhas anybody used Verizon Wireless BroadBand Access with an IAXy or any other VOIP?
01:37.11ZX81soulz-: groups
01:37.13ZX81maybe
01:37.17ZX81~voip-info
01:37.18jbotsomebody said voip-info was the Voice Over IP wiki.  It is a community resource which will answer all of your questions, from Asterisk to ZTDummy.  You can find it over at http://www.voip-info.org - well worth bookmarking
01:37.19ManxPowersoulz-, FXO ports use FXS signalling
01:37.35soulz-http://pastebin.ca/5991
01:37.37ZX81shmaltz: not i
01:38.08ZX81first four connect to telephone lines
01:38.13soulz-ok as per digium website the zaptel conf says fxsks=1-4
01:38.15ZX81second four connect to extensions
01:38.29ZX81the (FCC mode) gives it away
01:38.36ManxPowersoulz-, then your FXO ports are the first ones.
01:38.37rikstaso in the sip.conf under context, how do i refer to a queue
01:39.31soulz-manxpower: sorry for sounding lame, but what about fxs?
01:39.40ManxPowerriksta, You don't.  That's in extensions.conf and queues.con
01:39.51ManxPowersoulz-, FXS ports use FXO signalling.  Yes, it's confuzzleing
01:40.10rikstaso an incoming in extensions.conf is redirected to a queue how? it's a bit unclear on the wiki
01:40.26soulz-manxpower: so what i did was correct by using fxsks=1-4
01:40.29soulz-?
01:41.31ZX81yep
01:42.33ManxPowerfxsks=1-4 and fxoks=5-8
01:43.05ZX81and fkicantbefkd=1
01:43.07ZX81:)
01:43.16soulz-manxpower: thanks dude
01:44.47rikstacan someone show me an example of putting someone in the queue from extensions.conf please
01:45.01shmaltzanybody used the local channesl for dialing multiple mutiline sip phones?
01:45.15neuro_[rus]please advice some linux software phone for use with asterisk?
01:45.37soulz-gnuphone works
01:45.42QwellI like iaxcomm
01:45.56Qwellnever used gnuphone...any good?
01:46.30soulz-used for testing
01:46.35soulz-use a xten for myself
01:48.32neuro_[rus]thanks...
01:48.33*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
01:50.43*** join/#asterisk Zaw (zaw@zaw.subneural.net)
01:52.12rikstaman i can't get this queue to work, the person calls in, and the agent hears the beep, the music on hols is stopped, then the caller gets cut off immediately, and the agent hears on hold music again
01:52.58ManxPowerriksta, Queues are hard to set up, complicated to use, and have several issues which made them unsuitable for me to use.
01:53.06dsmouseriksta: can the extention that answers recieve normal calls? is it AgentLogin or agentlogincallback?
01:53.20rikstaAgentLogin
01:53.28rikstait shows the agent is logged in successfully on the cli
01:53.43dsmousebut if you call the agent's extention directly does it work?
01:54.07rikstayeah
01:54.29dsmousedo you have ackcall defined in agents.conf?
01:54.36rikstano
01:54.40rikstamy agent hears the beep
01:54.43rikstabut then it cuts off the call
01:54.53dsmouseif you don't set that to no, the agent has to press # to ack the call
01:55.04*** join/#asterisk kks (~kks@203.115.210.253)
01:55.13rikstai think that you mean the other way round
01:55.18rikstaif you set it to yea, he has to press #
01:55.33dsmouseer, that says for agentcallbacklogin... but it says default to yes...
01:55.44rikstai have it undefined, which means the call should connect immediately, which is what it tries to connect the call
01:55.46rikstathen it loses it
01:56.07rikstai'm using AgentLogin
01:56.25dsmouseThat was my best guess :/ sorry
01:56.39rikstaok thanks
01:57.51rikstaoh i tihnk i see whats happening
01:57.57rikstait goes     -- agent_call, call to agent '1000' call on 'SIP/1000-93db'
01:58.04rikstabut im already on a call, with the music on hold?
01:59.39*** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net)
02:00.02algorithmnsilence suppression with rtptimeout can cut a (for example) sip call
02:00.44algorithmnhappends when music on hold is on also
02:01.15algorithmnriksta: is the call cut off after a similar duration?
02:01.42rikstathe caller is cut off directly when the call is tried to be transferred from the Queue to agent 1000
02:04.30algorithmnriksta: sip?
02:04.34rikstayes
02:05.03algorithmnin sip.conf do you have rtptimeout set (other then that off the top of my head i wouldn't know)
02:05.23rikstaits undefined
02:05.39algorithmnas in not mentioned?
02:05.46rikstaits commented
02:05.48algorithmnok
02:06.04algorithmnhmm... let me sit here and grab a beer to help the thinking process
02:06.10rikstathanks dude :)
02:06.34*** join/#asterisk PyroSteve (~steve@wsip-70-183-114-254.no.no.cox.net)
02:06.38PyroSteveyO !!
02:06.49PyroSteve<PROTECTED>
02:06.51PyroSteve<PROTECTED>
02:06.54PyroSteveit works great
02:06.59PyroStevebut for the dtmf
02:07.39PyroStevewhen I press button on my analog phone, the other end just hears a very very short blurp of the tone
02:07.41hardwiremy telco provider just told me they have a patented conferencing called MeetMe
02:07.42hardwirehah
02:08.01PyroSteveand for the rest of duration of the tone is something like static
02:08.21PyroSteveany ideas
02:08.30rikstaalgorithmn: any chance i can PM you ?
02:08.36algorithmnsure
02:08.41algorithmndo it
02:09.10algorithmnPyroSteve: gsm?
02:09.32PyroSteveim using .... i think ulaw
02:09.52PyroSteveyeah.... g711u
02:10.13PyroStevedoes spa-2000 support gsm ?
02:10.55PyroStevewhat codec am i supposed to using
02:12.01algorithmngsm has limited dtmf support
02:12.12algorithmnulaw should be straight
02:12.37PyroStevei noticed the problem when I tried navigating a pbx
02:12.41*** join/#asterisk syslod (~sysglod@65.114.0.198)
02:12.52algorithmnare you AGI'in anything?
02:12.52PyroStevethen I called my cellphone and pressed buttons on the analog phone
02:12.57syslodsup
02:13.01PyroSteveno
02:13.12hardwireany providers that can give me a 1-800 incoming w/o a sales rep :)
02:13.21algorithmnok.. i know during beta agi, things tend to hang and zap tends to crash
02:13.28kksCalling party name:  [Tenor Call Relay SP Gateway], h323 incoming call failed with callerid with space. According to another side that they can't change the callerid. Is anything i can do on my asterisk ?
02:14.10algorithmnkks: try a softphone to test caller id... some older hardware boxes don't understand some characters
02:14.30algorithmnquite intermittent between brands, but not by how old the box is
02:15.35syslodAnyone have any channel banks with *?
02:16.20kksi have tried with SJphone, with callerid [Tenor Call Relay SP Gateway], but fail. Callerid [Tenor_Call_Relay_SP_Gateway] will work
02:16.22*** join/#asterisk JamesDotCom (~james@sweep.bur.st)
02:16.50algorithmni've gotten spaces in before...
02:17.33*** join/#asterisk tzafrir (foobar@85-65-203-192.barak.net.il)
02:20.03*** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net)
02:20.43PyroStevecan someone call me 662-796-1413 to hear my dtmf tone problem
02:20.52PyroStevedial ext 103
02:22.07algorithmnwhere is 662?
02:22.29algorithmnUS48?
02:22.36kksthanks algorithmn.
02:22.42syslodPyroSteve.  No ans.
02:23.40algorithmnkks: umm... with i new exactly what the CID prob is...
02:23.45algorithmnwish
02:23.46algorithmnknew
02:25.24*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
02:25.24*** topic/#asterisk is Asterisk: The Open Source PBX || Dev Conf 2PM CST FEB 17th -> IAX2/guest@66.250.68.194/996
02:28.21kksi have tried many callerid with softphone with and wihout space. and also tried to hardcode the callerid at dialplan, but channel h323 will take the original callerid for session initialization.
02:29.34algorithmnyou know... im not sure.. but h323, maybe, doesn't support spaces??  i really have to idea
02:29.45algorithmnno idea
02:29.51algorithmnits possable though
02:31.52tzangerhahahaha
02:31.59tzangerBlinken, I'd like to you meet Achoo.
02:32.01tzangerA Jew?  Here?
02:32.03tzangerhahahaha
02:32.12ZX81lol
02:32.24tzangerthis movie is awesome
02:32.40ZX81whatcha watching
02:32.42algorithmnyah
02:32.47tzangerRobin Hood -- Men In Tights
02:32.51algorithmnahhhh
02:32.59mikegrbTIGHT TIGHTS!
02:33.01tzangerMel Brooks
02:33.10algorithmnDave Chappel?
02:33.38ZX81lol
02:33.44tzangeryup
02:33.59ZX81and henceforth the Asterisk channel dissolves :)
02:34.03tzangerhahaha
02:34.13*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
02:34.13tzanger.... I have a MOLE??
02:34.29algorithmnmmmm... thanks
02:34.34algorithmni needed that
02:34.52algorithmni missed the l-train downtown to see jimmy on the corner of 89th and 12th
02:35.04algorithmnso im a little tweaked
02:35.32JerJerhop a cab
02:36.02algorithmnnah.. i got my alternate reserve under my bed... but i can't seem to remember the combo to the lock
02:36.20algorithmnshould've not drank the handle of svedka
02:36.32algorithmnnow its on...
02:36.39algorithmnbe back in 2min fellas
02:37.18tzangerdrank the handle of svedka?
02:37.55algorithmnmmmmmmmmmmmmm mmmmmm
02:38.14algorithmnafter it was empty.. i did <|;-(
02:38.45PyroSteveahhh
02:38.46algorithmnwasn't born husulus, i was birthin'em
02:38.48PyroStevehaaaa
02:38.49*** join/#asterisk Chuji (Chuji@pcp09929633pcs.tulipgrove.tn.nash.comcast.net)
02:38.57PyroStevewhoever just called me from NC
02:39.00PyroStevethanks !!
02:39.19PyroSteveI forced both my spa-2000 and my sip connection to broadvice
02:39.22PyroSteveto use ulaw
02:39.27algorithmnJerJer: your buddy missed out on 500 spa2k's?
02:39.32sudhir492when I do ztcfg -v I get the error: line 8: Unable to open master device '/dev/zap/ctl'
02:40.00sudhir492any suggestions? I did run make install in zaptel
02:40.03Chujisudhir492 : what modules are you loading?
02:40.05Nivexwha, someone else in here from NC?
02:40.14algorithmnsudhir492: red hat?
02:40.19sudhir492yes, rh9
02:40.48algorithmnyou must make config before asterisk install/reboot
02:40.50sudhir492I intend to load wct1xxp, for E1 card
02:41.00Chujisudhir492 : Do this....
02:41.09algorithmninit.d red hat n * sometimes don't like eachothr
02:41.10sudhir492I did run make config in asterisk
02:41.18algorithmndarn
02:41.50algorithmn?? gave it a shot, i'll keep a mid priority process running in the background for you
02:41.52Chujisudhir492 : modprobe wct1xxp ; rmmod wct1xxp ; modprobe wct1xxp ; modprobe zaptel ; ztcfg -vv
02:42.09Chujisee if it still does it
02:45.17sudhir492Chuji: Still the same error on ztcfg -vv, line 10: Unable to open master device '/dev/zap/ctl'
02:46.24Chujidoes asterisk start?
02:46.54sudhir492since modprobe wct1xxp fails, I did not even try to start asterisk
02:47.00dsmousePyroSteve: did that fix it?
02:48.07Chujitry starting it. It might actually be loaded
02:48.16dsmouseNivex: you're in NC?
02:48.16vaewynAnyone that missed it earlier: http://www.wwwrogue.com/voip/WIP5000.html  my review of the Hitachi Cable WIP-5000... Cool phone!
02:48.17Chujialso, check dmesg
02:48.25sudhir492Chuji, lsmod doesnot show that :-(
02:48.30Weezeycan I dial an extension to connect to the moh?
02:48.41algorithmnyes
02:48.51Weezeywhat's the command?
02:48.52Nivexdsmouse: I am.
02:48.59*** join/#asterisk WizardWlf (~shawn@wrt54g.djernes.org)
02:49.07dsmouseNivex: where about?
02:49.08algorithmnexten => _X.,1,MusicOnHold(random)
02:49.09algorithmnex
02:49.12Nivexdsmouse: Raleigh
02:49.19dsmouse?!
02:49.20Weezeythanks
02:49.24dsmouseAs am I!
02:49.46algorithmnno worries
02:49.52Chujisudhir492 : I get the same error on one of my boxes with a TE410 in it
02:50.00Nivexdsmouse: whoa... freaky
02:50.02Chujisudhir492 : but I can pass right through it
02:50.04sudhir492Yes, mine is TE110 too
02:50.10dsmouseNivex: yea, really
02:50.17WizardWlfanyone know of a script to autogen .call files from a database
02:50.40ChujiWizardWlf : Wouldn't be hard to right one in perl
02:50.50ChujiWizardWlf : Should only take a few minutes
02:51.03WizardWlfbut don't know enought perl to do the db stuff
02:51.16ChujiWizardWlf : What db you storing it in?
02:51.40WizardWlfprob be mysql but astdb would be ok also
02:51.53sudhir492Chuji: I get the errors when I try modprobe wct1xxp
02:51.54Nivexdsmouse: maybe we should start a VUG :-P
02:51.55sudhir492/lib/modules/2.4.20-8smp/misc/wct1xxp.o: init_module: No such device
02:51.55sudhir492Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters.
02:51.55sudhir492<PROTECTED>
02:51.55sudhir492/lib/modules/2.4.20-8smp/misc/wct1xxp.o: insmod /lib/modules/2.4.20-8smp/misc/wct1xxp.o failed
02:51.55sudhir492/lib/modules/2.4.20-8smp/misc/wct1xxp.o: insmod wct1xxp failed
02:52.10dsmouseNivex: *UG?
02:52.17Nivexhttp://www.voip-info.org/wiki-VoIP+User+Groups+USA
02:52.23Chujisudhir492 : cat /proc/interupts
02:52.42Chujisudhir492 : You got any conficts
02:53.28ChujiWizardWlf : I could probably piece you something together if you don't find anything
02:53.35dsmouse<PROTECTED>
02:53.38dsmousebah
02:53.43dsmousewow that's sparce
02:53.50sudhir492Chuji, how do I look for conflicts in cat /prco/interrupts?
02:54.09WizardWlfChuji: ok will keep looking
02:54.11Chujisudhir492 : is your wcxx sharing anything?
02:54.14Nivexdsmouse: do you make it to any of the LUG meetings around here?
02:54.37sudhir492I do not see wcxx in the list at all
02:54.46dsmouseI made a few... 1/2 the people there know me.
02:55.02Chujisudhir492 : Well hell, bios isn't even seeing it then
02:55.02sudhir492This is what I see after cat /proc/interrupts
02:55.13sudhir492<PROTECTED>
02:55.13sudhir492<PROTECTED>
02:55.13sudhir492<PROTECTED>
02:55.13sudhir492<PROTECTED>
02:55.13sudhir492<PROTECTED>
02:55.14sudhir492<PROTECTED>
02:55.16sudhir492<PROTECTED>
02:55.16Chuji~pastebin
02:55.17jbotfrom memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
02:55.18sudhir492<PROTECTED>
02:55.20sudhir492<PROTECTED>
02:55.24sudhir492<PROTECTED>
02:55.24Nivexgrak!
02:55.26sudhir492<PROTECTED>
02:55.28sudhir492<PROTECTED>
02:55.30sudhir492<PROTECTED>
02:55.30dsmousebad sudhir492 !!!!!!!!!!!!!!!!
02:55.32sudhir492NMI:          0          0          0          0
02:55.34sudhir492locksy:   18137484   18137483   18137483   18137483
02:55.36sudhir492ERR:          0
02:55.38sudhir492mishehu:          0
02:55.41Chuji~pastebin
02:55.42jbotfrom memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
02:55.42Nivex/ignore sudhir492
02:55.48sudhir492sorry about that. I did not about ~pastebin
02:55.54sudhir492Thanks for educating
02:56.06sudhir492~pastebin
02:56.07jbotmethinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
02:56.11Nivexdsmouse: #trilug ?
02:56.21SedoroxQuestion for those with Ops.. why don't you put pastebin in the topic?
02:56.27dsmouseonly a few times
02:56.39dsmouseI was more NCSU lug awhile back
02:56.47JerJerbecause people should be smarter than what they are working on
02:56.51Chujisudhir492 : You sure your mobo is a compatible voltage with the T110?
02:56.55dsmouseyou've met Peter then
02:57.03QwellSedorox: If they put everything in the topic, there wouldn't be room. ;]
02:57.18Chuji~rtfw
02:57.19jboti guess rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php
02:57.25ChujiThat should be in the topic
02:57.27Chuji:)
02:58.09ChujiOr my personal fav for JerJer!
02:58.18Sedoroxlol
02:58.21Chuji~h323
02:58.22jboti heard h323 is evil! Don't ask about h323 here. Ask JerJer if you need to bug someone. He says it works, but others don't.
02:58.29Sedoroxwell I was just thinking the pastebin link
02:59.18*** join/#asterisk didz_ (~omg@200.218.193.30)
02:59.52didz_Ouch ... error while writing audio data: : Broken pipe
02:59.52didz_Warning, flexibel rate not heavily tested!
02:59.52didz_Segmentation fault (core dumped)
02:59.55sudhir492Chuji: hmm, On the same mobo another box, I have TE410, no problem. Hence I put TE110 on this one even without checking
03:00.03sudhir4925v PCI
03:00.06didz_anybody with a good heart could help me ? :)
03:00.19Chujisudhir492 : is 5v what it takes? I can't remember
03:00.47didz_twisted?
03:00.52Chujididz_ : what version of mpg123 you running?
03:00.54Weezeyanyone ever hooked a Norstar ATA into a FXO?
03:01.18didz_i've been using the 0.59r, but now i'm using the format_mp3.so from asterisk-addons
03:01.26didz_the same thing happens...
03:01.44didz_already converted the mp3 to 8000 hz mono with lame
03:02.04Chujiwhen you lose the mp3's does that go away?
03:03.20didz_haven't tried... because this is happening in a production environment... if i lose de mp3's the "clients" will hangup, since they will not hear anything =[
03:03.36didz_using with queues etc...
03:03.55bjohnsonBeirdo: signed up and it works
03:03.56Chujiwell, get the ones out of the asterisk source
03:04.03Chujiand see if it goes away
03:04.05bjohnsongreg_work: you there
03:04.16ChujiThey are nice and calming
03:04.21didz_i'm using the ones from asterisk source
03:04.35didz_fpm-calmriver, fpm-worldmix, fpm-sunshine
03:04.36greg_workbjohnson: i am
03:04.42didz_i'm totally lost
03:04.42bjohnsonpm
03:04.52greg_worksure
03:04.57tzangerhehehe
03:04.58sudhir492Chuji: I am just checking digium website.
03:05.00tzangerstrikey has loxed again!
03:05.10Beirdobjohnson: cool
03:05.13Chujididz_ : Yeah, I use those too
03:05.27*** join/#asterisk ayano (~erik_leee@adsl-66-51-208-150.dslextreme.com)
03:06.09*** join/#asterisk lilneon (~tj_r3@cuscon10992.tstt.net.tt)
03:06.16lilneonhi everyone.. and good night
03:06.20didz_could it be a problem of lots of agents logged in ?
03:06.21Chujididz_ : Does it crash you?
03:06.26didz_yes
03:06.34didz_segfault with coredump
03:06.56Chujididz_ : I wouldn't think so, but it's possible.
03:07.01syslodIs there something you have to do different with FXO/FXS vs PRI???
03:07.06ChujiHow many mpg processes get spawned?
03:07.15Chujisyslod : Huh?
03:07.21lilneonhey guys, recently downloaded FC3. would you recommend moving my asterisk box from RH9 to FC3? any one had problems?
03:07.26ChujiPRI  is fxo
03:07.42Chujililneon : Y?
03:07.51didz_no one... i'm not using mpg123 anymore, i'm using now native mp3 support from asterisk (format_mp3.so)
03:07.53dsmouselilneon: I've heard there's a trick to getting udev to work if you're useing local hardware
03:08.02lilneonchuji: don't know.. someone recommend i do it cuz it was better ..???
03:08.22syslodChuji: I've got a bunch of PRI working but the other day I decided to get Channel banks working and can't.  Now I bought a few FXO cards and they don't seem to wanna work either.
03:08.58Chujisyslod : The more digium cards you put in one box you are asking for trouble
03:08.59lilneonso you would recommend i stay with rh9? still haven't gotten festival to wrk on there yet :(
03:09.03Chujithey are interrupt hogs
03:09.17syslodI kicked the box across the room a min ago but that didn't help.  made me feel a little better though.
03:09.26neuro_[rus]I have Celeron-900Mhz PC with 256Mb RAM... Would it work well with _only_ two concurrent calls? I'm using software phones.
03:09.34Chujililneon : Well, It's just preference actually
03:09.42Qwellneuro_[rus]: should
03:09.50Chujililneon: Maybe try capstral too
03:09.53lilneonChuji: so there is really no performance gain?
03:09.56syslodChuji: I have a box with a new T1/E1 card and another box with 2 digium FXO cards 4 FXO/2FXO
03:09.56dsmouselilneon: I have no first hand knoledge about asterisk on either FC3 or RH9, so I'm just repeating rumors.
03:10.13Chujineuro_[rus] : yes, it will work, but soft phones suck
03:10.17lilneondsmouse: so what do u run it on? debian?
03:10.20Chuji~softphone
03:10.21jbotsomething that should be drug out into the street and shot
03:10.27dsmouseFC2 :)
03:10.36syslod~hardphone
03:10.38neuro_[rus]Chuji: why?
03:10.42NivexI'm still waiting for a good inexpensive hard phone.
03:10.54NivexOne of the guys here pre-ordered the SPA-841.  He still hasn't seen it
03:10.55Chujineuro_[rus] : Because they are only as good as the OS they run on
03:11.01Chujiwhat OS are they on?
03:11.08lilneondsmouse:ok ..and your experience with asterisk on there was??? A.painful? B.very painful c.all of the above?
03:11.15neuro_[rus]Chuji: Linux and Windows
03:11.20dsmouseI like to say I've had no problems getting asterisk to work on fc2 :)
03:11.26Chuji:)
03:11.40dsmousewell, a few problems, but they were pure asterisk
03:12.27lilneondsmouse: ok but since u had no prior experience with rh9 and asterisk i guess i can't really ask which u would recommend..  :S sigh
03:12.47dsmouseyea, but I had no prior experience with asterisk at all.
03:12.58dsmousein fact, till sunday, I've only read about it :)
03:13.08dsmousenow I *LOVE* it.
03:13.30Chujidsmouse : Hope you aren't busy for the next couple of months
03:13.31Chujiheh
03:13.40dsmouseChuji: why?
03:13.53Chuji~aa
03:13.54jbottest
03:14.01lilneondsmouse: dont worry.. they only playing
03:14.02ChujiHeh
03:14.29lilneonasterisk is pretty much a tamed beast since the stable 1.0 release
03:14.30dsmouseEh, asterisk is just fun for me. if it didn't work or annoied me I could just go back to a landline
03:14.34lilneonwell in my opinion
03:14.56Chujijbot AA is Asterisk Anonymous. Something we all should join after a few months of Astriholism
03:14.57jbot...but aa is already something else...
03:15.00lilneondsmouse: well i don't even use my landline anymore. .
03:15.16dsmouselilneon: I don't trust my ISP that much yet.
03:15.25Chujijbot erase aa
03:15.40Chujijbot remove aa
03:15.46Chujiblah, stupid bot
03:15.48dsmousejbot eat aa
03:15.50jbotACTION eats aa and falls over dead
03:16.04dsmouseWho knew?
03:16.06lilneondsmouse: well.. my 'clients' use it to make calls here.  yeah and my ISP is my telco so.. they pretty much suck right thru.. but i don't really make that many calls
03:16.11*** join/#asterisk Carp1 (~chatzilla@ip-204-97-151-110.modem.logical.net)
03:16.38dsmouselilneon: the diffrence between asterisk at home vs asterisk at work*
03:16.50dsmouse* no I didn't use "asterisk at home"
03:17.21Chujijbot Astriholics are people that spend every waking hour working with Asterisk. They need a life!
03:17.22jbotChuji: okay
03:17.35tzangerwe're men... we're men in tights...  we roam aroudn the forest looking for fights...
03:18.14lilneondsmouse: well i am the opposite.. i use it @ home.. and use my wrk's telephone @ wrk.. pretty much underpaying us anywayz
03:18.18Carp1Anyone in here use/have Cell Socket?
03:18.34lilneonso wee use the 'facilities' they give us ;)
03:18.40*** join/#asterisk tangotool (~sysglod@65.114.0.198)
03:19.11tessier_Feb 16 19:15:55 WARNING[14177]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end)
03:19.16tessier_Anyone know what causes this?
03:19.26lilneonhey anyone from teliax in here?
03:19.28tessier_I really wish asterisk had better error messages. I would at least like to know what client or ip is causing this.
03:19.40tessier_It has been streaming across my console like mad for the last hour.
03:20.30Chujitangotool : You work for qwest?
03:21.01sudhir492Chuji: You may be right. TE110 may be 3.3v card :-( I do not see any mention of 3.3v for this card at digium site thoug
03:21.15Qwelllilneon: dca I believe
03:21.23sudhir492TE410 is 3.3v, TE405 is 5v
03:21.35tzangersudhir492: correct
03:21.41*** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-110.modem.logical.net)
03:21.42dsmouselilneon: I do
03:21.44Chujisudhir492 : I think the new cards are 3.3
03:21.59sudhir492Darn
03:22.07Carp1AstWind work good?
03:22.20sudhir492for Quad T1/E1 they have both 5v and 3.3v versions
03:22.20dsmouselilneon: right now I'm lukewarm about them
03:23.03Qwelldsmouse: lukewarm?
03:23.32tangotoolChuji: Used to years ago.
03:23.34dsmouseQwell: well, they're pruduct seems ok, but they screwed up my order...
03:23.41lilneonQwell: u know me?
03:24.45dsmouseI love that I can set my own callerid and they honor it... I can have multiple outgoing streems w/IAX2 support, etc
03:25.17dsmousebut I didn't get a incoming line like I expected, and he/she/it//they haven't returned my email
03:25.24Chujitangotool : I just noticed your ip address when you /joined. I thought that /16 was their reserved backbone addressing
03:25.39dsmouseIf I were a buisness, that would be really annoying
03:26.10tangotoolYea.  I'm at work with a friend.  Sitting in a NAP.
03:26.40ChujiCool, throw up some warez!
03:26.42Chuji:)
03:26.43lilneondsmouse:well i am a bit worried bout their new rates.. seems the routes i use most are those that got raised
03:27.28lilneondsmouse: but i email them and get back a response.. most fo the time.. the following day .. two at most
03:27.43dsmousewell, it's only been 24 hours now
03:27.54Qwelllilneon: Do I?
03:27.56lilneondsmouse: but they have been down quite a bit recently.. so i started looking for a secondary itsp
03:28.13Qwelllilneon: Ask dca, he should be able to help you.
03:28.20QwellHe was trying to pimp teliax to me a few weeks ago, heh
03:28.22dsmouseso, it's not like he's utterly pissed me off. also, I'm using the pay-as-you-go for domestic only...
03:29.12lilneonQwell: oh cool.. when u said Dca.. i thought that was short for Dominica... the country i am originally from :D
03:29.20dsmouseI only chose telaix because they report to have 919 area codes... I'd prolly have gone with broadvoice otherwise...
03:29.29dsmousebut, hey, this is only a hobby for me :)
03:29.30*** join/#asterisk puzzled_ (~patrick@puzzled.xs4all.nl)
03:29.52Nivexdsmouse: you try voicepulse?
03:30.03lilneondsmouse: well this is business for me.. i lvoe asterisk.. and i am trying to set up a lil voip biz
03:30.10dsmouseNivex: no 919 :(
03:30.21lilneonNivex: yeah so far frm wat i have read voicepulse seems pretty ok
03:30.26dsmouselilneon: doing what exactly?
03:30.49*** join/#asterisk Nukemizer (Nuke@66.237.85.58.ptr.us.xo.net)
03:30.52lilneondsmouse:reselling minutes..  here still under a monopoly by one telco
03:31.02Nivexdsmouse: say what?!
03:31.02NivexYour official rate center name: RALEIGH
03:31.03Nivex(Might not match your city name)
03:31.03NivexVoicePulse is available in your rate center!
03:31.04sudhir492Chuji: Thanks a lot for your help. I will confirm tomorrow with Digium
03:31.09lilneonso.. ppl welcome paying lower rates to places like US and UK..
03:31.24dsmouse?!
03:31.50Chujisudhir492 : No problem, hopefully you get it resolved
03:32.56dsmouseNivex: hrms, I must have gotten confused with another carrier... do they do iax?
03:33.11*** join/#asterisk iMediax (lklk@00045a809589.click-network.com)
03:33.13dsmouseoh
03:33.14Nivexdsmouse: connect.voicepulse.com does iax, but it's a different rate structure
03:33.24dsmouse919 RALEIGH *ON ORDER*
03:33.42NukemizerI have turned off my PRI while Digium sends me a new card, my "meetme" rom is now borken. DO I need to have Digium gear in the box in order to do conferencing ?
03:33.57Sedoroxno
03:34.03Chuji~ztdummy
03:34.04jbotztdummy is probably zaptel timing source which uses a usb-ohci compatible usb controller as source. (part of zaptel cvs)
03:34.27Chuji~google ztdummy
03:34.40Nivexwhich kinda sucks cuz every controller I've ever had has been uhci
03:34.52dsmousebeets ntegratedconsulting.com. They wanted a ear to avoid a setup charge.
03:35.13dsmouse~rtfw
03:35.14jbotwell, rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php
03:40.01*** join/#asterisk roamer323 (~sing@Toronto-HSE-ppp4172487.sympatico.ca)
03:42.28greg_workis it bad to have   exten => _123,1,Macro(something..)   (pattern matching with no variables) ?
03:44.28*** join/#asterisk doughecka (~Doug@doughecka.user)
03:45.05dougheckaI have 2 2600 routers...
03:45.10dougheckawhat can I do with them
03:45.35dougheckaI have 2 T1 DSU/CSU, 1 56/64 DSU/CSU and 2 ISDN BRI cards
03:45.52dougheckacan I get asterisk to talk to them?
03:46.24*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
03:48.32*** join/#asterisk PakiPenguin (~uppal@202.176.230.225)
03:50.08lilneonok buh bye guys
03:50.13*** part/#asterisk lilneon (~tj_r3@cuscon10992.tstt.net.tt)
03:52.21NukemizerChuji: jbot: thank you for the ztdummy tip !
04:01.14dsmousejbot: eath Nukemizer
04:01.19dsmousejbot: eat Nukemizer
04:01.21jbotACTION eats Nukemizer and falls over dead
04:01.31dsmousethat th is a hard habbith to break
04:03.06*** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net)
04:03.39Legenddoes anyone else have greatly varying latency to nufone?
04:03.58tzangerLegend: you mean jitter?
04:03.58tzangerno
04:04.20tzangermind you I'm 9 hops from them
04:06.21roamer323when I call from * SIP extension to SIP extension, the callerid assigned in sip.conf shows up for xlite, but on an extension originating from the ATA - the ATA's callerid shows up... anyone know why?
04:07.37tzangerhttp://www.savetoby.com/
04:09.33Legendwow, a blank black page
04:09.34Legendhow moving
04:09.41tzangerit's flash
04:09.42tzangerwait for it
04:10.03Legendtzanger: its slow jitter
04:10.12tzangerslow jitter
04:10.14tzangerinteresting
04:10.40Legendshit, i know i would be heckled for not singing nufone's praises
04:11.01tzangerwho's heckling?
04:11.06Legendthe latency fluctuates during the day, sometimes above 200ms, but voicepulse usually stays between 50ms and 100ms
04:11.22tzangerLegend: sounds like you have a better connection to vpc
04:12.20tzangerI recommend nufone when I feel it's right, as of all the provider's I've tried they Just Work.  They've also gone well, well out of their way to help me out with testing the new jitter buffer in -HEAD
04:12.24*** part/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com)
04:12.28tzangerer not in -HEAD but rather in the bugtracker (2532)
04:12.34tzangerM2532
04:12.42tzangernope jbot odesn't do the bug tracker link thing
04:16.20tzangerwow
04:16.30tzanger6 people got organs/tissues from one woman
04:16.35tzangerI think that's amazing
04:16.46Damascenethat must have been some woman
04:16.52tzangermuch better use of a person's body after they're dead than incinerating them or burying them
04:17.04tzangerhttp://www.cbc.ca/stories/2005/02/16/rabies-transplants050216
04:17.10tzanger3 of the 6 have rabies now though
04:17.20ManxPowerDeath Penalty + Transplants = ....
04:17.28tzangerManxPower: hah
04:17.44tzangeryou onbviously have not seen the simpsons episode where Homer gets the dude's scalp for a transplant
04:18.17ManxPowertzanger, no, but I have read too much Larry Niven
04:18.30tzangerheh
04:19.46tzangerNeither the Great Depression, nor two World Wars could prevent the NHL from awarding the Stanley Cup, but with the league and the NHLPA still divided over the issue of a salary cap, Lord Stanley's trophy will not be contested for the first time since 1919, when a Spanish flu epidemic wiped out the finals
04:19.51tzangerwow
04:19.57tzangeroh well...  nobody there's starving, that's for damn sure
04:21.32tzangermaybe losing the entire season will wake some people up to reality
04:24.02implicit:)
04:24.25tzangeranyway
04:24.26tzanger'night
04:24.44tzangerdamn
04:24.54tzangerall the beer and sauerkraut and saussage has left me quite gassy
04:24.59implicitanyone know any good SIP termination providers for US only that do NOT have asterisk in the media stream
04:25.15tzangerthat's a good question
04:25.22impliciti know of a couple
04:25.24tzangerI am guessing dslreports would have something
04:25.27Mocany channel driver code + RTP expert ?
04:25.29implicitbut i dont want to use them
04:25.55implicitjust need a provider that has SIP jitterbuffers
04:25.57implicitand
04:26.07implicitdoesn't restream RTP (thus the need for no asterisk)
04:26.22tzangerzoa's working on getting 2532 into chan_sip
04:26.28tzangernot sure about your #2 though
04:27.05tzanger'night
04:27.10implicit#2 will never happen in * afaict
04:27.59*** join/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com)
04:31.09*** part/#asterisk didz_ (~omg@200.218.193.30)
04:38.09*** join/#asterisk PTG123 (~PTG123@ip68-106-19-249.ph.ph.cox.net)
04:38.30PTG123hey anyone here know a good source for the polycomm phones that is cheap
04:38.38chipighey PTG123
04:39.33PTG123hey chi :)
04:40.40*** join/#asterisk andrew` (~andrew@adsl-67-119-26-96.dsl.snfc21.pacbell.net)
04:41.04bjohnsonPTG123: hang on
04:42.20bjohnsonhttp://www.tritechcoa.com/phone-systems/O1B2.html
04:42.39PTG123bj that the phone you would commen?
04:42.41PTG123or something else
04:42.45*** join/#asterisk yurpls (~yurplsl@65.114.0.198)
04:42.47PTG123need a good reliable cheap phone to demo to  business clients
04:43.00bjohnsonthat's not what you asked for
04:43.07PTG123http://www.tritechcoa.com/product/b-O1B2-80.html
04:43.13PTG123is that one any good, anyone know? :)
04:43.16bjohnsonI don't have any voip phones
04:43.27bjohnsonthe IP500 is very popular here
04:43.48bjohnsonthe Sipura SPA 841 is beginning to be the popular <$100 voip phone
04:43.49PTG123hmm
04:44.13bjohnsona lot of excitment about iax phones becoming available but not many actually in use yet
04:44.35yurplsIs there a list of IAX phones that are avail?
04:44.53PTG123any ide aon agood plae to buy the sipura?
04:45.12bjohnsonyurpls: no .. not a list
04:45.21bjohnsonPTG123: what country are you in?
04:45.25yurplsatacomm has them.
04:45.26PTG123us
04:45.40yurplsThere are a bunch of other places.  Try google.
04:45.45bjohnsonPTG123: voxilla (free shipping) .. give me the voip coupons :)
04:46.11bjohnsonPTG123: also check voipsupply .. but shipping cost usually makes them more $$
04:46.25bjohnsontry atacomm .. but I think they're in about the same range
04:46.57PTG123so many choices, so little time :)
04:47.14bjohnsonPTG123: take 5 minutes and I think you'll order from voxilla
04:47.35PTG123looking at them now
04:47.37bjohnsonoh yeah !! last I checked, using *users at voxilla got you $10 off
04:48.15PTG123voxilla only has one phone for sale :)
04:50.01sudhir492Occasionally, I have seen mention of DS3 cards for Linux/Asterisk. Any news on that front?
04:50.11yurplsNo DS3 from digium.
04:50.33yurplsAt that level you'll need more of a card than a non-dsp version.
04:50.34PTG123i think the polycomm 300 looks the best/cheapest good combo
04:50.47sudhir492I know not from Digium, but is anyone working on other DS3 cards?
04:51.04yurplsPTG123: I'd go with the 500.  We have a pile of 300's here that ppl won't use after using the 500 and 600s
04:51.18PTG123why don't they use the 300s?
04:51.39yurplsMaybe the SS7 folks but there are just a few that are interested in * supporting that level of calls.  Including myself.
04:51.56sudhir492It may not worthwhile for Digium to develop a DS3 cards as demand for those beasts will be very limited, however it will be cool if someone wrote driver for some other vendor's DS3 card
04:52.11|Vulture|IP500s are very nice
04:52.33yurplsPTG123: The phone isn't backlit, none are, but the 500 is better to see and the 600 even sharper.  The 300 has a calculator display where the others actually have pixels.
04:53.01yurplsI think the SBC or maybe saganoma
04:53.09yurplsSBC definately have it.
04:53.28yurplssorry sbs
04:53.40sudhir492I have Asterisk installed at multiple locations with T1 cards. Consolidating at 1 place (even for 5 T1) will save me money.
04:54.22PTG123well since a dual xeon can only encode 30 g729 streams
04:54.26PTG123why would you want more then one t1 per box?
04:54.28yurplsWe using multi 4 port cards with no probs.
04:54.32sudhir492The provider is willing to give me as many T1 on a Ds3 interface.
04:54.41PTG123not to mention redudance, blah blah
04:54.43yurplsPTG123: Why encode. ULAW rules.
04:54.49PTG123sudhir492: so use a mux
04:54.57yurplssudhir492: just get a mux.
04:55.02PTG123yurpls: because when you encode most problems with voip disappear :)
04:55.04PTG123on a busy network
04:55.05yurpls$1200 in us + cable
04:55.05sudhir492Yes, that is what I am thinking
04:56.06yurplsPTG123: We are using it for TDM mostly.  IP is on a 10Gig backbone with QOS and VLAN for voice.
04:56.23yurplsAdtran MX unit or telco edgelink.
04:56.44PTG123yurpls: well its different if you can control the network, but if you are doing pbxing for a business, or whatever
04:56.48PTG123encoding works much better
04:56.57PTG123which reminds me i got to buy some g729 licneses  now that it works on freebsd :)
04:57.08yurplsIf a DS3 card was avaiblle I'd get it in a heartbeat.  I'd love to do * VM on class 5 switching.
04:57.15Qwelloh, g729 works on freebsd now?
04:57.32PTG123yes :)
04:57.40PTG123it took  mark around a year to implement my code ;)
04:57.41PTG123but it works
04:57.42PTG123heh
04:57.46Qwella year?  heh
04:57.52QwellYou did the mac stuff, right?
04:57.52PTG123well a little opver a month :)
04:57.54PTG123yah
04:57.56Qwella few weeks ago?
04:58.01yurplsPTG123:  yea we do businesses but they pay for upgraded networks.  Some are doing GSM and a few have teleworkers that do G729 but mostly ULAW without probs.
04:58.13PTG123gsm i still have issues with
04:58.22PTG123ulaw i have issues with, but don't seem to have any with g729
04:58.23PTG123for some reason
04:58.40PTG123plus it uses 1/10th the bandwidth of ulaw
04:58.59yurplsYea but we have like 100's of users on a box.
04:59.10PTG123so you can put in a good 30 calls plus have plenty left over for bandwidth on a t1
04:59.28yurplsThe box has 2 4 port T1 cards with NFAS PRI.
04:59.30PTG123someone needs to make a hardware based g729 encoding daughter board
04:59.41yurplsThat would be cool.
04:59.44QwellThat would be nice
04:59.53QwellTake alot of the load off the server
04:59.59yurplsI'd like a T1 that had DSP for fax, modem, etc.
05:00.24yurplsWe make a killing on fax servers.  Hylafax for now.
05:01.00PTG123hear that redder :) hold out your hand to yurpls :)
05:01.00PTG123heh
05:01.16PTG123you know redder in here is the primary hylafax developer? :)
05:01.33yurplsHylafax = gooooood
05:01.49QwellWhats hylafax exactly?
05:01.50yurplsBeen using it for years.  Stupid rightfax servers sucked.
05:01.55QwellOne of those fax-email gateways?
05:01.57PTG123are you terminating t1s directly into the hylfax servers, or handing off via ip?
05:02.27yurplsPTG123: I work for the LEC so its about 100 feet from a class 5 tandem with DS1 handoffs.
05:02.51yurplsDialogic cards.
05:03.12PTG123ah gotcha, my next project is to get hylafax or asterisk fax working well with sip handoffs
05:03.21PTG123so i can allocate numbers to my clients anywhere in the country
05:03.44yurplssip -> fax?
05:04.20PTG123ya
05:04.29PTG123if t.38 worked in asterisk it would be very simple
05:04.30yurplswith g729?
05:04.32PTG123ulaw is a little iffy
05:04.34PTG123nah :)
05:04.39yurplsoh I was confused.
05:04.44yurplsYea t.38 would be nice.
05:05.02PTG123if i could find me someone to set me up a test t.38 sip account
05:05.06bjohnsonQwell: hylafax is linux software to send faxes
05:05.10PTG123i'd do the t.38 iface
05:05.17yurplsand receive faxes.
05:05.19PTG123bjohnson: and receieve
05:05.20bjohnsonI believe mgetty is still used to receive them
05:05.38yurplsKinda like sendmail for faxing
05:05.42PTG123bjohnson: hylafax does all the work
05:05.55PTG123it "can" use mgetty to answer the phone
05:06.17yurpls* seems to work ok with spandsp on the receive.
05:07.05bjohnsonmgetty also auto-senses fax vs data and can autostart ppp for modem calls
05:07.52MocI would love a soft modem... that can connect to ppp or something
05:08.09PTG123moc: their is such a thing
05:08.12yurplsDoesn't * do ras?
05:08.16bjohnsonhow about connecting to my boot?
05:08.19PTG123spandsp guy just came out with it
05:08.35bjohnsonoh you mean for *
05:08.45PTG123yah well sort of
05:08.52yurplsIs that pre10?
05:08.58PTG123basically now you can make asterisk emulate a modem to iface with asterisk
05:09.03PTG123came out like a week ago
05:09.08Qwellbjohnson: thanks
05:10.47*** join/#asterisk ScythelX (Fleb@pc-24-181-176-72.sbi.ct.charter.com)
05:13.11*** join/#asterisk ranliv (~ranliv@210.5.98.224)
05:14.04*** join/#asterisk B4 (~B4@202.69.48.245)
05:14.24MocPTG123: he did ?
05:14.24B4~seen zx81
05:14.26jbotzx81 <matt@222-153-114-115.jetstream.xtra.co.nz> was last seen on IRC in channel #asterisk, 2h 40m 27s ago, saying: 'and henceforth the Asterisk channel dissolves :)'.
05:14.30PTG123moc: yah
05:14.37Moche rock...
05:14.59Mocin pre10 ?
05:15.14PTG123i don;'t know what pre10 means :)
05:15.20PTG123ask redder86 about it he knows more
05:15.35yurplspre10 is the current version buts theres no docs.
05:15.37bjohnsonPTG123: Sedorox was looking for that I think
05:16.07Mocok well I check it once I get this UNISTIM channel driver working like I want to..
05:16.34PTG123using asterisk i am very much use to the no docs part :)
05:17.14PTG123anyone here know about clevo/sager notebooks
05:18.45Sedorox`whats that?
05:19.06PTG123a brand of notebooks
05:19.09SedoroxPTG123: I have a Sager
05:19.15PTG123http://pctorque.com/3790.php
05:19.18PTG123thinking about buying that one
05:19.21PTG123Sedorox: which one?
05:19.45SedoroxI have the 8790 (can't get it anymore.. the 9860 or whatever it is replaced it... )
05:19.50Sedoroxthe very high end model...
05:19.56Sedoroxmy friend here got that model your looking at
05:19.56PTG123where did you buy it from
05:20.00Sedoroxhe loves it
05:20.02PTG123i am looking for very small but powerful
05:20.09SedoroxI got it from sagernotebooks.com, but they don't have it listed anymore...
05:20.13Sedoroxyea.. thats why he got it
05:20.15PTG123Sedorox: great thats what i want to hear :) any recommendations from where to buy?
05:20.18Sedoroxmine doesn't have any battery life.. maybe a hr
05:20.20PTG123i need 1920x1200 too
05:20.24Sedoroxhehe
05:20.25PTG123yah this one has 5 hours
05:20.28Sedoroxmine's 1680x1050.. but eh
05:20.30SedoroxYup.. centrino
05:20.33Sedoroxhe loves it..
05:20.35Sedoroxumm
05:20.42PTG123he get it direct from sager too?
05:20.50Sedorox... he used PCTorque first.. then ended up ordering from another site.. let me see if I can find it
05:21.03PTG123k
05:21.07PTG123why did he not use pctorque?
05:21.32Sedoroxhttp://www.powernotebooks.com/product.php?itemId=414
05:21.44Sedoroxhe had some ordering problems with his credit card or something I think
05:21.54Moc:q
05:21.55SedoroxI think thats where he got it from if I remember correctly
05:22.25SedoroxEverything works in Linux, 'cept the camera in the screen, and the TV Tuner that I have in mine
05:22.32PTG123yah powernotebooks is um $50 more :)
05:22.36Sedoroxbut I think they are working on both them soon
05:22.48PTG123yah i need to dual boot so that is good news
05:22.54PTG123how long did it take you to receive them?
05:22.57Sedoroxyea
05:23.00PTG123and do either of you have dead pixels?
05:23.24SedoroxMine I got within a week I think... I used standard shipping.. not sure with him, I think he got it over one of the breaks last semester
05:23.38SedoroxI have a few on mine.. but I don't notice them unless I realllyyyy look for it
05:23.57Sedoroxonly noticed it because it was rebooting.. so it was a black screen, and I happened to be close to the screen
05:24.02PTG123i wish i could find a place with a dead pixel guarantee
05:24.39SedoroxUmmm I think Sager will replace the screens for dead pixels.. I think its covered in the warrenty.. not sure..
05:25.05PTG123no
05:25.08PTG123only if in center
05:25.27Sedoroxhmmm
05:26.40Sedoroxyea.. overall.. bothof us have been extremely happy with it
05:27.24Sedoroxbtw... notebookforums.com is few good for sager stuff.. and also alienware.. dell's.. gateway.. hehe
05:27.40PTG123yah
05:27.44PTG123thats where i found out about th em
05:27.57PTG123i don't like it b/c pctorque kills threads if you ask about places to buy other then them :)
05:28.07PTG123i found a plae that had a dead pixel guarantee
05:28.09PTG123but loist the url
05:28.26Sedoroxinteresting
05:28.29Sedoroxlol
05:29.36SedoroxI found out about it from a friend of mine in HS.. he got it.. I loved it when I saw it.. and I actually managed to talk my parents into getting it when I went to college..
05:29.50*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
05:30.44PTG123heh
05:30.49PTG123well mine notebook just got fried
05:30.55PTG123ribbon cable broke so lcd is useless
05:30.55bjohnsonsedorax: weren't you looking for a soft modem?
05:30.58PTG123and its pretty beaten up
05:31.01PTG123so time to buy a new one
05:31.14bjohnsonPTG123: selling?
05:31.50QwellI need to get me a good used laptop
05:32.40PTG123bj: selling what?
05:32.47PTG123a broken laptop? :)
05:33.07bjohnsonyes
05:33.16PTG123Qwell: i got  a nice p3 1ghz i'll sell cheap :)
05:33.17Sedoroxbjohnson: no.. I don't think so...
05:33.23PTG123bjohnson: for what?
05:33.24QwellPTG123: How cheap we talking?
05:33.30Sedoroxlol
05:33.31bjohnsonSedorox: sorry .. must have been something else
05:33.36Sedoroxand everyone jumps on a broken laptop
05:33.41PTG123Qwell: not sure :) i never use it.. its pretty nice though
05:33.41bjohnsonPTG123: you mean for what use?
05:33.45*** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au)
05:33.53Sedoroxbjohnson: I don't think I was asking about anything today....
05:33.56Sedoroxallwell :-p
05:33.58bjohnsonPTG123: depends on specs .. maybe print server
05:34.05bjohnsonPTG123: maybe router
05:34.09Sedoroxhehe
05:34.11Qwelllaptop router?  heh
05:34.18PTG123bjohnson: heh why not just use a pc? :)
05:34.21Qwellmakes sense...low power
05:34.21bjohnsonipcop or monowall
05:34.31Sedoroxhehe
05:34.33bjohnsonPTG123: laptop is smaller and has built in UPS
05:34.46Qwellhmm, thats actually not a bad idea
05:34.47PTG123bjohnson: well its in pieces now :)
05:34.59PTG123bjohnson: thats basically what i did is made it into a little computer
05:35.01Sedoroxmy old IBM Thinkpad 600 just sits here on my desk out of the way acting as a tunnel endpoint from my house, to allow me access inside my home network, and to give me IPv6 access here at school
05:35.08QwellPTG123: If you quote me a price, I'll discuss with the wife
05:35.14Sedoroxlol
05:35.23PTG123Qwell: make me an offer, feeling generous today :)
05:35.29PTG123let me find the model one sec
05:35.32Sedorox$20!
05:35.35Sedorox:-p
05:35.37bjohnsonPTG123: what the heck is it?
05:35.39PTG123not that generous
05:35.42Sedoroxlol
05:35.47ScythelXdo I need to setup something so SER handles dtmf for asterisk - or configure asterisk in a certain way?
05:35.50PTG123bjohnson: the broken one, p4 2.4ghz or something like that
05:35.52bjohnsonPTG123: specs
05:35.56Sedoroxwow...
05:36.02roamer323on an incoming sip call, how does * find the [???] entity in sip.conf to use? does it seach by username, host?  anyone?
05:36.04bjohnsonjust the lcd is broken right?
05:36.07PTG123it was a 5k notebook 2 years ago, or so
05:36.09bjohnsonram and HD?
05:36.11jbotIRC Client versions for #debian-bots (4): other - 7 (46.7%) ;; unknown/cloak - 6 (40%) ;; eggdrop - 1 (6.7%) ;; irssi - 1 (6.7%).
05:36.27PTG123bjohnson: like i said i gutted it, like 60gig drive or so, 512+128 ram
05:36.49ScythelXRFC2833?
05:36.55bjohnsonPTG123: you can put it back together?
05:37.38PTG123bjohnson: um probably not i kept the bottom intact, i cut out of amettle a custom cpu headsink, etc
05:37.42PTG123and was gonna cut plexiglass for the top
05:37.46bjohnsonwell, whatever .. $100 in pieces plus shipping cost
05:37.47PTG123so it was like a transparent notebook
05:37.49QwellPTG123: yeah, if you could get the model number, would be cool to look at it
05:38.08PTG123its kind of a project, so probably more fun to keep it
05:38.28bjohnsonPTG123: sure .. now that I've given you ideas for what to use it for
05:38.31PTG123besides the ram alone would be worth keeping to put in new notebook :)
05:38.36bjohnsonhow about a laptop * server
05:38.55Sedoroxhehe
05:38.56bjohnsonwould be okay for homw or small business use
05:38.59Sedoroxgood if you travel
05:39.00Sedorox:-p
05:39.16bjohnsonthere's someone out there running * on a P100 with 16M RAM according to the wiki
05:39.20SedoroxPTG123: I doubt it would go in any new laptop...
05:39.22Qwellwpw
05:39.31Qwellwow, rather...
05:39.38SedoroxI was thinking about getting a Soekris and putting asterisk on it
05:39.44Sedoroxits  a 266 Via chip
05:40.18Qwellheh
05:41.00Sedoroxlol
05:42.15PTG123damn trying to figure out model # :)
05:42.19PTG123notebook at office
05:42.21QwellWhat brand?
05:42.42bjohnsonSedorox: read some info about soekris not being great for *
05:42.48PTG123sedorox: its pc2100 or whatever ram for notebook so it will :)
05:43.20PTG123Sedorox: what do you think my other notebook is worth? :)
05:43.37bjohnsonPTG123: keep the 512 .. give me the rest with the 128M
05:44.08PTG123bj: heh
05:44.24PTG123then i got to package it up, etc :)
05:44.37PTG123alright how about $150 for it i suppose
05:44.40Sedoroxbjohnson: any perticular reason why?
05:44.44SedoroxPTG123: hehe, ok :-p
05:44.45B4feeling generous :) donate the notebook to me
05:44.51Sedoroxwhat other notebook.. the 1gig?
05:45.02PTG123sederox: yah if you want a good cheap notebook
05:45.06PTG123its titanium too, and small
05:45.09PTG123which is why i liked it :)
05:45.11PTG123durable
05:45.14QwellPTG123: What brand, etc?
05:45.16Sedoroxlol
05:45.17PTG123got 2 batteries, etc
05:45.20PTG123its an acer.. :)
05:45.30SedoroxI'm not really looking for anything.. I don't have any money right now :-/
05:45.33Qwellhow much/what type of ram?
05:45.41Sedoroxwell I little.. but I got bills and stuff
05:46.13PTG123qwell: hah why you looking for one? :)
05:46.16PTG123welcome to e#ebay :)
05:46.22PTG123bjohnson you still alive?
05:46.25bjohnsonPTG123: including shipping or not for $150 USD?
05:46.30B4#asteriskbay
05:46.35QwellPTG123: Never owned a laptop, but it was really nice when I was able to bring the one from work home.
05:46.38Sedoroxlol
05:46.40bjohnsonB4: abay
05:46.45B4lol
05:46.51PTG123bjohnson: do you care how its shipped?
05:47.04Sedoroxthis is my first _real_ laptop.. the TP600 could never put a battery in it...
05:47.18B4me no ... I can even accept shipments via donkey cart
05:47.20bjohnsonPTG123: as long as I can get it working once recieved
05:47.22PTG123Qwell: well i never use it, i can't use any res less then 1600x1200 so if you are interested
05:47.26PTG123thats my only complaint about it
05:47.30*** join/#asterisk clive- (~pirch@myw-stp-66-18-86-221.sentechsa.net)
05:47.33bjohnsonPTG123: you can ship it camel back for all I care
05:47.50QwellPTG123: 15"?
05:47.55PTG123bjohnson: well its working for me here :)  Well i am beginning to like usps alot for shipping
05:47.59PTG123Qwell: yah
05:48.04Qwellhow much/what type of ram?
05:48.08B4BTW you can get reaaaaal chep lcds from china :)
05:48.11bjohnsonPTG123: usps is good for me
05:48.16PTG123Qwell: its a very nice notebook, um 512
05:48.20bjohnsonPTG123: shall we move to pm?
05:48.27PTG123sure bj
05:48.54QwellPTG123: Are you talking $150 for the 1ghz, or the b0rked one?
05:49.06PTG123qwell: borked one, i got two of em :)
05:49.17PTG123Qwell: i'll take any decient offer for other
05:49.32PTG123i also got a g4 powerbook any takers on that :) the ti one
05:49.38B4other one also broken lcd?
05:49.47PTG123ok 3 books
05:49.58QwellPTG123: Mind another PM?  heh
05:50.03PTG1231. P4 2ghz-2.4ghz fucked lcd, and kind of in pieces
05:50.08PTG1232. p3 1ghz nice shape :)
05:50.13PTG1233. Powerbook TI g4 :)
05:50.18PTG123and sure the more pm the better
05:50.43wasimPTG123: i'll trade you #2&3 for a farfon :)
05:50.46Sedoroxlol
05:50.52Sedoroxfarfon?
05:50.55B4lol wasim
05:50.55PTG123what is a farfon? :)
05:51.11wasim~farfon
05:51.12ScythelXanyone ever have dtmf problems with client phone - > ser - > asterisk
05:51.13B4hey PTG thats a good offer
05:51.25wasimPTG123: there are only 40 of them in the outside world
05:51.35PTG123ScythelX: think you are in the wrongc hannel for asterisk questions
05:52.00B4this is #abay ScythelX
05:52.01wasimfarfon.com ... an iax2 hardphone shipping prerelease
05:52.13clive-wasim, hey, hows the iax,ata comming along?
05:52.16ScythelXbo yeah
05:52.22B4whats the beta price wasim?
05:52.40wasimB4: ranging between EU 350 and 1000 + vat!
05:53.00B4different models?
05:53.18wasimclive-: its taking a little finangling, coz we're missing a couple of components
05:53.34wasimB4: nope, shortage of supply, and levels of support
05:53.44*** join/#asterisk scratchrf (~dirk@65-102-181-251.tukw.qwest.net)
05:53.47wasimB4: you get a 650 EU rebate with the 1000 price tag
05:54.02clive-wasim, soon youull be selling a iax version on the pa168 ata
05:54.25wasimclive-: not us, but others will be for sure
05:54.27B4hmm well shortage of supply should not raise the price at the manufacturer
05:54.47wasimB4: we're not the manufacturerr, we're just the r&d shoppe, these are all hand made prototypes
05:55.00wasimB4: actual vol retail price should be $75
05:55.12B4ah ok ...
05:55.33wasimB4: these units are for core itsps's to test features etc
05:55.45B4outsourced work from some other company?
05:55.54Sedoroxhmmm
05:56.02wasimB4: outsourced the pcb manufacturing and mounting
05:56.07clive-:)
05:56.36B4no I mean you are r&d ing for for some other manufacturer?
05:56.38wasimclive-: ofcourse, both products are for different markets
05:56.39*** join/#asterisk pygmy (~pygmy@141.110.15.200)
05:56.55wasimB4: no, we're licensing out the firmware for other manufacturers
05:57.25wasimclive-: we're aiming for high end, niche phones ... stuff with encryption and advanced features
05:57.37wasimand ofcourse, phones for techies :)
05:57.51wasimwe should be releasing a lot of the firmware at lots.ch
05:58.03*** join/#asterisk letherglov (~letherglo@8036aa5e.resnet.ucsd.edu)
05:58.04wasimas per our comittment to open source
05:58.15B4whats the hardware based on?
05:58.21wasimti c54xcst
06:00.06scratchrfhas anyone used SixTel/iax.cc?
06:00.29hmodeshas ti released an ata work-alike yet?
06:00.30B4seems popular choice so you should have many firmware customers
06:00.49hmodesi thought someone would have been all over that by now
06:00.54wasimhmodes: ti? no, we use silvertels slics
06:01.05letherglovwhy not ti's slcs
06:01.13letherglovand are those the same slics digium uses?
06:01.25hmodesbecause ti is hopelessly behind the curve perhaps? :)
06:02.11wasimi really want to get the uberATA out and testing
06:02.22wasimbut we need a little dough before we can do that
06:02.28hmodesuberata eh?
06:02.36letherglovit's a shame you're not in afghanistan
06:02.41hmodescan it school a sipura 3k? :)
06:02.42letherglovI hear the poppy crop is great this year
06:02.51wasimhmodes: 30 port modular fxs/fxo iax2 ethernet/dsl "channel bank"
06:03.00wasimletherglov: yeah, bumper crop
06:03.01hmodessooo that would be a yes
06:03.02hmodeshow much?
06:03.12B4lol letherglov
06:03.23wasimhmodes: retail 1200 - 1500 us$ ?
06:03.54hmodesnice
06:04.14wasimhmodes: thats what we think to
06:04.17hmodesdoes it have its own dsps or does the iax host have to do the codec work?
06:04.27wasimhmodes: no, it has its own dsps
06:04.33*** join/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net)
06:04.34*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
06:04.50hmodesyeah, i conceed that is hotness
06:04.53*** part/#asterisk djin (~djin@gridfox.xs4all.nl)
06:04.57*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
06:05.07wasimespecially if we can get it to retail around $1k, that would be a nice product indeed
06:05.10clive-wasim so when is the launch date
06:05.11B4channel bank in the works wasim?
06:05.28hmodeshell, i'd buy one for home :)
06:05.35wasimclive-: we're open sourcing the firmware today i think
06:05.41B4yeah book one for me too :)
06:05.52Firestrmanyone know if engenius is selling their wifi phone yet?
06:06.35clive-wasim does that mean the hardware is done?
06:06.43wasimclive-: the hardware has been done for ages now
06:06.57B4want beta testers? heh
06:07.01wasimB4: sure
06:07.08clive-ah, cool,..send me 2
06:07.15clive-does it work?
06:07.16B4send me one :)
06:07.16wasimbut we're really running short of units
06:07.26wasimclive-: ofcourse, its testing all over EU right now
06:07.39wasimclive-: .de, .ch, .pl. .no, .nl
06:07.50B4impressive
06:07.51Firestrmwasim, i joined half way through conversation, what hardware are we talking about?
06:08.12wasimFirestrm: the elusive farfon
06:08.49implicitwasim whats the pricing?
06:09.20wasimmost of the world seems to :)
06:09.34*** join/#asterisk pranav (sameer@202.149.48.200)
06:10.08implicitFirestrm: you need everything it seems
06:10.34Firestrmimplicit, first the earth.. then the universe..
06:10.51Mavvieredder86: still awake?
06:11.01implicitFirestrm: other way around would save time
06:11.08implicitfirst the universe then no need for the earth
06:11.17Firestrmhmmm.... i like it..
06:11.46Firestrmi can obtain the universe, but i cant get my PSTN line working properly..
06:11.48Firestrm:(
06:12.22implicityou dont know if you can obtain the universe until you do it
06:12.34implicityou can't even really know the essence of what it is to obtain
06:12.41implicituntil you obtain every obtainable thing
06:12.52hmodespfft
06:12.59hmodeseverything == nothing
06:13.26implicithmodes: that is not a bad assumption
06:13.28pranavwe are not able to connect to fwd
06:13.32implicitbut you can't prove by negation
06:13.40hmodesdamn!
06:14.02hmodesthen the only option is to prove everything or nothing
06:14.03hmodeswtf?!
06:14.59pranavhello
06:15.01pranavwe are not able to connect to fwd
06:15.02implicithmodes: no
06:15.39*** part/#asterisk B4 (~B4@202.69.48.245)
06:15.44implicitit depends, is empirical evidence enough for you or do you want to actually prove it?
06:15.56Qwellpranav: We saw that the first time.  Does anything happen that might be remotely useful in debugging the problem?
06:16.18hmodeseff empirical evidence, what has it ever done for me?
06:16.50implicithmodes: a lot?
06:16.58implicitlol
06:17.05hmodesdamn!  foiled again!
06:17.51hmodesso one should accept what they're told then?
06:18.18*** join/#asterisk ayano (~erik_leee@adsl-66-51-208-150.dslextreme.com)
06:18.40ayanowhat are the disadvantages of asterisk@home?
06:18.56Qwellthe extra 5 chars you have to type when discussing it
06:19.05wasimayano: you sound like an alien if the connection is made
06:19.31hmodeshaha
06:19.36hmodessuch hate for the noobs
06:19.44hmodesthat's both disturbing and wonderful
06:20.17hmodesalthough i suppose he was smart enough to ask
06:20.37ayanoI just tried it, I have always just used source for installing * but that was easy, there has to be some kind of drawback.
06:21.34hmodesappliance installs are the devil and guis tend to prohibit touching the text configs in any sort of creative manner?
06:21.49ayanoahh, there we go, so it will only let you configure surtain things?
06:21.56ayanocertain
06:21.59ayanooops
06:22.07hmodesvia the gui as last i saw it, yeah
06:22.22hmodesmuch better to just hack it up yourself if you can manage
06:22.58ayanoI see, but to accually do the install from it is it okay?  It only took like 30 minutes on a laptop...
06:24.17*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:25.37hmodesehhh, probably for a dedicated * box
06:25.52hmodestho I'd still be vaguely paranoid about security after the install
06:26.02hmodesi haven't really played with it personally tho
06:26.02*** join/#asterisk terraHome (~cjs@cs662586-139.satx.rr.com)
06:26.32techiehaha
06:26.40hmodesalso, cvs is always preferred over a package in my mind
06:27.02ayanookay, I'm going to try to screw it up, and I'll let you know what happens hmodes
06:27.20hmodesgood luck :)
06:27.47ayanothanks.  your my hero for the night.... :)
06:28.04pranavhello mr.qwall  when i call to fwd number it says it is invalid extension
06:28.38terraHomeugh, why the hell is russia selling syria weapons
06:28.48terraHomei thought russia hated islamic terrorists.
06:29.00QwellterraHome: Because they were getting mad at my making a profit on them
06:29.06terraHomeheh
06:29.13pranavhello mr.qwall  when i call to fwd number it says it is invalid extension
06:29.22wasimterraHome: so does the US, but that doesn't stop them from arms sales to "terrorists"
06:29.25terraHomehah
06:29.41terraHomewe do not sell arms to terrorists.
06:29.43Qwellpranav: again, I saw that the first time.  No need to repeat
06:29.54terraHomenot this administration, anyway
06:29.58pranavok sorry
06:31.01wasimterraHome: hah ... you really for that one
06:31.06wasimterraHome: fall, even
06:31.25terraHomewasim, where are you located?
06:32.01pranavQwell: i have pasted my sip.conf and extentions.conf in pastebin.ca/6001
06:32.03wasimterraHome: pk
06:32.08terraHomehah
06:32.22Qwellpranav: I can't help you tonight.  Sorry
06:32.42QwellIf it were noon, and I was free for the rest of the day, maybe
06:33.20terraHomeanyone have an IAXy?
06:33.22pranavok some one else please tell me whats the mistake
06:33.24terraHomeor play with one?
06:34.19*** join/#asterisk murangd (~nukaidc@pool-162-84-229-224.ny5030.east.verizon.net)
06:34.54terraHomeis it good?
06:35.42*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
06:35.42*** topic/#asterisk is Asterisk: The Open Source PBX || Dev Conf 2PM CST FEB 17th -> IAX2/guest@66.250.68.194/996
06:36.20`SauronI want cheap FXO interfaces
06:36.49Sedoroxhow many?
06:37.03pranavSedorox: currently i am not using iax to make calls, i have 2 sipura phones , from where i want to make fwd calls
06:37.18terraHomeugh, the iaxy needs a power supply (not included)
06:37.59QwellSedorox: Sorry for pawning that off, couldn't deal with it, heh
06:38.04SuperMMananyone see this http://technology.sympatico.msn.ca/Home/ContentPosting.aspx?contentid=0653c17f5bd348bbb4d3405dc821fa12&show=False&number=4&showbyline=False&subtitle=&detect=&abc=abc
06:38.45pranavSedorox:if you don't mind , just have a look at pastebin.ca/6001, i have pasted the sip.conf and the extensions.conf
06:38.52hmodesterraHome: it's like $10 at radiohack
06:38.57SedoroxQwell: 'tis fine :-p
06:39.04Sedoroxpranav: I looked at it....
06:39.09pranavok
06:39.18terraHomei dread going to radioshack
06:39.37pranavso is there anything wrong in that?
06:39.37*** part/#asterisk SuperMMan (~graphic@d209-89-191-155.abhsia.telus.net)
06:39.40hmodesi imagine if you ask nice enough digium can prolly provide powah
06:39.40Sedoroxummmm
06:40.10Sedoroxthe only thing I can think of is your gonna want that format.. the username:password@fwd/${exten} but let me look something up
06:40.57FirestrmterraHome, digikey is your friend ..
06:41.16FirestrmterraHome, better than radioscrap any day of the week
06:41.51pranavwhen i call fwd number(say 7612) the ring comes but then after say 3 to 4 rings it says "extension invalid"
06:41.55FirestrmterraHome, www.digikey.com
06:41.55terraHomedigikey...
06:41.58terraHomeok
06:42.07Firestrmthey have EVERYTHING!
06:42.11Firestrmallmost..
06:43.02hmodesthey have an annoying minimum order fee tho ;p
06:43.06Firestrmthey really like me :)
06:43.11hmodesi keep getting bitten by it when i order flash
06:43.18implicithehehe
06:43.34Firestrmhmodes, never been a problem for me.. all my orders are around 1000.00 ea
06:43.42hmodesthbbbpt
06:43.45hmodeslucky you!
06:43.45`Sauronfirestrm: What you order?
06:43.52*** part/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net)
06:44.18hmodesalthough even the minimum charge is worth it for most of the obscure shit they can source
06:44.39terraHomewhat do you buy from digikey, firestrm
06:44.49terraHomewell, here's the next problem with the IAXy:
06:44.52terraHomewhat phone do i use? :)
06:44.54Firestrm`Sauron, i run a small prototype electonics design/assembly company on the side, i think at some point ive ordered justa bout everthing
06:45.02`Sauronah
06:45.04`Sauronthat's cool
06:45.13hmodesterraHome: i have a pulse dial western electric on mine :)
06:45.15terraHomei was thinking about one of those really old Bell System desk phones from the late 70s/early 80s
06:45.23terraHomeIAXy can handle pulse?
06:45.41hmodesyup
06:45.53hmodesit even converts pulse to dtmf for ivrs
06:46.00terraHomei'll need a wireless bridge too
06:46.01terraHome:/
06:46.19hmodeslinksys is your friend?
06:46.24terraHomethe idea for this phone is as an emergency phoen for home when my cell runs out of batteries
06:46.31terraHomehow much are 802.11b bridges nowdays?
06:46.51hmodesi don't think anyone even sells b bridges anymore ;p
06:46.56hmodesa g bridge is prolly about $40
06:47.10hmodesmulti-mac bridging mebbe $50 or $60
06:47.11*** join/#asterisk shidan (~shidan@CPE000e08eaf90e-CM014280007905.cpe.net.cable.rogers.com)
06:47.16`SauronI have a b/g bridge
06:47.17Firestrm`Sauron, www.vrl.ca
06:47.29`Sauronbecause they wanted $169 for a a/b/g bridge
06:47.30terraHomehttp://www2.panasonic.com/webapp/wcs/stores/servlet/vModelDetail?displayTab=O&storeId=11251&catalogId=11005&itemId=70256&catGroupId=20952&modelNo=KX-TG5200M&surfModel=KX-TG5200M
06:47.30`SauronGrr
06:47.32terraHomeugh
06:47.37Sedoroxhttp://www.newegg.com/app/ViewProductDesc.asp?description=33-156-154&depa=0
06:47.41SedoroxB Bridge
06:47.48hmodeseff a
06:47.50terraHomei need a super-cheap bridge
06:47.52terraHomean IAXy
06:47.56murangdI don't understand why doesn't they make voip using
06:47.58murangdUDP
06:48.00hmodesand if you're getting b, g can't possibly be much more
06:48.00murangdor TCP
06:48.04murangdwhy SIP or IAX?
06:48.16`SauronI'd rather have A than g
06:48.23`SauronG blows
06:48.29terraHomeA does not go through walls as well
06:48.31hmodeswell, in a closed environment
06:48.34SedoroxI think A blows...
06:48.36hmodesi have friends tho ;p
06:48.39terraHomei have a G router
06:48.43terraHomebut only a B card in my mac
06:48.50terraHomeso i'm fine w/ a B bridge
06:48.59SedoroxI have all G stuff here now... save for a B CF for my zaurus
06:49.03terraHomeof course, if i get that phone, i don't need the bridge
06:49.10terraHomei'll just plug the base station into my switch
06:49.12`Sauronthe problem with b/g is if you have a G network, and a B device starts talking, ALL the devices drop to B
06:49.20`Sauronbecause b/g is on the same spectrum
06:49.24terraHomeyep
06:49.27Sedorox`Sauron: I didn't have that problem
06:49.38`SauronSedorox: You do, you just don't notice it.
06:49.42Sedoroxwhen my zaurus is on.. everything still says 54...
06:49.50`SauronYou can't do both B and G at the same time
06:49.52hmodesand that's when I kick the b user in the sack and verbally berate them in front of their peers ;p
06:50.10hmodesand then they upgrade
06:50.13hmodesand everyone is happy
06:50.28Sedoroxlol
06:50.47letherglovhmodes, no no no
06:50.55letherglovfirst, you steal their music from itunes
06:51.00letherglovTHEN you berate them
06:51.04terraHomehmmmmm
06:51.17hmodesbut it would be so slooow
06:51.21hmodesit's almost not even worth it
06:51.22murangdcan someon eanswer my question
06:51.28`SauronFirestrm: Done any FPGA stuff?
06:51.46terraHomeso i'm debating on cordless phone or old Bell phone + 802.11[bg] bridge
06:51.55Firestrm`Sauron, yes, a long time ago..
06:51.55terraHomeim leaning towards the bell phone
06:52.09`Sauronwebpage says you do asic stuff
06:52.28`SauronI'd imagine fpga stuff would be more common, since they were sort of made to take over the asic stuff
06:53.08`Saurons/common/recent
06:53.11letherglovterraHome, you left the ata out of there
06:53.22letherglovoh, and
06:53.29terraHomelether, i'll use an IAXy
06:53.32letherglovyou'd better check if your microwave is going to interfere
06:53.38terraHomeoooo, what about a PAYPHONE
06:53.41Firestrm`Sauron, the asic stuff was when i was doing avionics design.
06:53.43letherglovyou don't want to be making yourself a frozen burrito and loose your girlfriend too
06:53.48terraHomei could put a payphone in my bedroom next to my bed
06:53.56terraHomeand use it as my talking-in-bed-at-night phone
06:54.09terraHomehooked up to the IAXy and wifi bridge
06:54.15letherglovbetter yet
06:54.20letherglovyou can steal  from the x-men
06:54.22letherglovand get..uhhh.shit
06:54.26letherglovwhat's that thing called?
06:54.34`SauronFirestrm: Ah.
06:54.34letherglovmemorino;
06:54.36letherglov;-)
06:58.02*** join/#asterisk c00ljack (c00ljack@202.69.190.247)
06:58.19c00ljackhi
06:58.50terraHomeok, here's a geeky question
06:58.54terraHomeif you take an IAXy
06:58.56c00ljackto whom  can seek assistance re: quiuntum integration?
06:58.58terraHomeand put a pigtail out of it
06:59.14terraHomeand wire that pigtail into your home phone wiring
06:59.20Sedoroxew
06:59.27letherglovuh
06:59.28terraHomeis it enough to handle all of the phones at home (albeit one at a time)
06:59.33letherglovhow many REN you think it puts out?
06:59.44terraHomedunno
06:59.45Sedoroxummm... as long as you make sure you are disconnected from the PSTN
06:59.51terraHomeim not on the PSTN
06:59.53Sedoroxand yea.. it may not ring all the phones...
07:00.04lethergloveither that or it browns out and reboots
07:00.05terraHomei have no PSTN service at home
07:00.09letherglovoh
07:00.16letherglovlive next to ted kazinsky?
07:00.22hmodesehh, as long as it's under 2.5 or 3 i bet it can do it with a beefy power brick
07:00.23terraHomehas anybody ever tried this?
07:00.27terraHomelether, basically
07:00.38letherglovhow do you connect to the internet?
07:00.41terraHomei have a huge 2br apartment with virtually nothing in it
07:00.42SedoroxI know people do it with a cisco ATA.. so dunno
07:00.43terraHomei have:
07:00.56terraHomea couch, a bed, a chest of drawers, and a desk
07:00.57SedoroxI'm sure it'll support 2 phones...
07:00.58terraHomeand that's it
07:01.09hmodesi actually power about 1000sq ft with an iaxy
07:01.11terraHomesed, is the length of the home wiring gonna be an issue?
07:01.16hmodesand prolly about 2ren
07:01.25hmodesmebbe 1.5 or so
07:01.25`SauronI know the spa-1001 can do like 10 ren's
07:01.26terraHomeREN = ???
07:01.27`Sauronor something
07:01.32letherglovwhere's your evil typewriter for your manifesto
07:01.34hmodesringer equivilency or some crap
07:01.37letherglovringer equivalency number
07:01.40SedoroxI don't think so... I think it really just matters on how many phones...
07:01.42letherglovit's the amount of current the phone draws
07:01.43terraHomelether, heh
07:01.44letherglovwhen it rings
07:01.47terraHomeno typewriter
07:01.55`Sauronyou can configure the max. ren load in software
07:02.01terraHomethough i have met a lady who was blown up by the unibomber
07:02.05hmodesi think 1.0 == oldschool bell phone
07:02.12letherglovwhat software?
07:02.22hmodesso 2-3.0 would be WAY more then enough to ring multiple modern powered ringers
07:02.32hardwireany idea why linphone under debian shows no codecs?
07:02.33letherglovwell
07:02.38letherglovif it's an actual ringer
07:02.39terraHomehmodes, im talking about an old ass Western Electric telephone
07:02.43letherglovversus the ringer on a powered phone
07:02.43letherglovsure
07:02.44terraHomebut i will only have one of them
07:02.48hmodesoh, yeah
07:02.51letherglovif it's a cordless thing
07:02.52hmodesi can run one of them :)
07:02.56letherglovthan it's got it's own power supply
07:02.58hmodeshaven't really tried two
07:03.02letherglovso it probably doesn't draw much
07:03.07terraHomethee point of using the home wiring is that i dont have to have the IAXy in the room with the phone
07:03.09hmodesbut one oldschool phone and extension to *.modern seems stable enough
07:03.22letherglovI have one too
07:03.24letherglovold cortelco
07:03.26letherglovI use it at home
07:03.28terraHomewould this work:
07:03.30letherglovand I'm far away
07:03.36letherglovI can wake anyone in the house with it
07:03.38letherglovit's loud as fuck
07:03.49terraHomeIAXY <----rj11---> wall jack <--------> wall jack <---- rj11----> telephone
07:03.51`Sauronterrahome: So you're hooking the iaxy instead of the NT interface box?
07:04.00terraHomeyeah sauron
07:04.09terraHomei want to plug the IAXy into a wall jack in the computer room
07:04.16`SauronI need to find my NT box so I can do that.
07:04.16terraHomeand a regular old phone into the wall jack in the bedroom
07:04.18hmodesterraHome: almost definately, unless you have the shittiest wiring on earth
07:04.26terraHomefucking a.  word.
07:04.54terraHomethat would save me the cost of the wifi bridge
07:05.07terraHomei buy a cheap old western electric phone for the bedroom and an IAXy
07:05.27`Saurondum di dum
07:05.30`Sauronwonder where my box is
07:05.31terraHomehttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=985&item=6153534553&rd=1
07:05.33terraHomethat.
07:06.20hmodesyeah, i have the same thing in rotory
07:06.25hmodesthe iaxy drives it happily
07:06.43terraHomeyou have both plugged into different wall jacks in the same house?
07:06.56terraHomeoh, this is going to be sweet
07:07.02hmodesi think i had a splitter on the iaxy
07:07.02terraHomei just need to pick out the ideal phone now
07:07.07Sedoroxlol
07:07.16hmodesbut i don't have my wall jack -> apartment nid disconnected
07:07.26Sedoroxhmmm
07:07.27hmodeswhich is probably a good 150' of bleeding copper
07:07.32*** join/#asterisk shidan (~shidan@CPE000e08eaf90e-CM014280007905.cpe.net.cable.rogers.com)
07:07.34hmodesand it's happy with it
07:08.13`Sauronhmodes: Just watch, there's a line at the CO that rings everytime you get a call... :)
07:08.39hmodesthat would be highly amusing
07:08.58terraHomei need to make sure to disconnect my apartment from the PSTN
07:09.00hmodesbut i had to find my line when i ran dsl, so i know it goes nowhere
07:09.17`Sauronnod
07:09.26hmodesyeeah, driving the pstn takes rather uber power :)
07:09.27`SauronI'll find my nid and disconnect it
07:09.34hmodesgranted i would love to light that shit up
07:09.34`Sauronor wait for the CO to axe it
07:09.34terraHomehahahah
07:09.43hmodesbut it's cheaper to just run my home
07:09.46terraHome"why is my nufone balance all gone?"
07:11.01terraHomedammit!
07:11.05terraHomei can't find the TNI
07:11.14terraHomethere is a cable junction box for CATV
07:11.20terraHomebut nothing for PSTN
07:11.26`Sauronterra: at a house, or apt?
07:11.32terraHomeolllllld apartment
07:11.39terraHomelike from teh 50s
07:11.42terraHome40s
07:11.47`Sauronbribe one of the maintenance guys with a 6-pack of beer
07:11.49`Sauronhe'll find it for you
07:11.50`Sauron:)
07:12.22Firestrm`Sauron, i find twinkies work best ;)
07:12.29terraHomewill it cause problems to just plug the IAXy into the wall lines?
07:12.34terraHomehaving no idea where they go
07:12.44Firestrmummmm... yes
07:12.51hmodesi wouldn't think it would damage the iaxy
07:12.54SedoroxterraHome: only if they are still connected to the pstn source
07:12.57hmodestho you may want to check with mark
07:13.03Sedoroxotherwise... if you know for a fact that they are
07:13.04`Sauronwell, you have to make sure the CO is disconnected from your wiring
07:13.04Sedoroxit shouldn't be
07:13.10hmodesif it openly doesn't work definately don't leave it plugged in tho
07:13.14`Saurondunno what 108V ring voltage would do to an iaxy
07:13.30Sedoroxlol
07:13.46hmodeswell i would assume you're not expecting the wall jack to ring
07:14.07hmodesif you're plugging one in to a jack that's live, you deserve to have it fried ;p
07:14.18Sedoroxlol
07:14.31terraHomei'm pretty damned sure that the CO is still physically connected
07:14.32c00ljackhey guys any idea how can connect quintum to asterisk?
07:14.38terraHomebecause its an apartment.
07:15.00`SauronOh well
07:15.01`Sauronsleep time
07:15.34terraHomehmmm
07:15.36terraHomemaybe its out back
07:15.40terraHomebehind the building
07:16.02hmodestone generators are your friend
07:16.07hmodesand fairly cheap
07:16.16SedoroxterraHome: talk with whoever you pay for the reent...
07:16.22Sedoroxcheck to see if you can get it d/c
07:16.25Sedoroxas it may be in the basement
07:16.31Sedoroxor some room you don't have access to
07:16.32Sedoroxlol
07:16.48terraHomeknowing them, they will disconnect all the interior jacks from each other, too
07:16.53hardwireok
07:16.59hardwirejunctionnetworks are neat peeps
07:17.00hmodesso glad my building has a giant well marked bell box on the outside :)
07:17.09hardwireanybody here from junction networks?
07:17.19terraHomeoh man
07:17.25Sedoroxlol
07:17.27terraHomeremember those multi-line phones from the 70s
07:17.32terraHomethe ones with the big square buttons
07:17.34terraHomethat would light up
07:17.36hmodes*thunk*
07:17.40terraHome*that* is what i need
07:17.49*** join/#asterisk jdmjamboo (jdmjamboo@202.69.190.233)
07:17.59hmodesi feel a general fondness for them in the same way i feel a general fondness for original ps/2 keyboards
07:18.13hmodesthe 80s was totally the decade of *keys go thunk*
07:18.18Sedoroxlol
07:18.21jdmjamboohi people
07:18.32jdmjamboohows everybodies doing?
07:18.37FirestrmterraHome, one apartment place i lived in, didn't even lock the telco room.. Phone Phreakers paradise ;)
07:18.38terraHomethunk keys ruled
07:18.47terraHomeall mechanical things should work the way those phones did
07:18.53*** join/#asterisk denon (denon@synapse.subneural.net)
07:18.53*** mode/#asterisk [+o denon] by ChanServ
07:19.13terraHomehttp://www.pensive.org/jeff/mrfone/collect/WE2563.htm
07:19.14terraHomepingo
07:19.16hmodesindeed
07:19.17terraHomebingo
07:19.20hmodeslong live the thunk
07:19.31terraHomeRED HOLD BUTTON!!!
07:20.03terraHomehmodes, ever seen an old Fender guitar amp?
07:20.25hmodesmmm, warm nourishing vacuum
07:20.33jdmjambooanybody here had configured Asterisk to Quintum interconnection?
07:20.35jdmjambooanybody here had configured Asterisk to Quintum interconnection?
07:20.39terraHomego away, idiot
07:20.42FirestrmterraHome, one of the local surplus places has a bunch of the really old style phones with the crank thingy on the side.. i was thinking of turning one into an IP phone..
07:20.42Sedorox.....
07:20.53terraHomefirestrm, we have similar ideas
07:20.59terraHomei want to take a phone like this:
07:21.00terraHomehttp://www.pensive.org/jeff/mrfone/collect/WE2563.htm
07:21.01jdmjambooplease
07:21.04jdmjamboohelp
07:21.09terraHomeand make it into a multi-line IP phone
07:21.17terraHomebut everything needs to be integrated, in-case
07:21.21FirestrmterraHome, better yet make it a Wifi phone ;)
07:21.30terraHomeor hidden nearby
07:21.36terraHomeexactly
07:21.54terraHomei want my phone to look and feel like an old clunk-button phone
07:21.59terraHomebut with modern tech
07:22.24jdmjamboohi terraHome.. have you encounter quintum to asterisk interconnection?
07:22.31FirestrmterraHome, i know where there are a pile of the phones picured in the previous link
07:22.33terraHomehow about a Speak-and-Spell <---> LDAP directory interface
07:22.41hmodesmeh, you could pretty easily rewire a 7960 to drive one of them
07:22.48*** join/#asterisk kks (~kks@203.115.210.253)
07:22.55jdmjamboohi terraHome.. have you encounter quintum to asterisk interconnection?
07:23.05terraHomego away, idiot
07:23.11terraHomefire: where?
07:23.14terraHomecan you get me one or two?
07:23.15hmodesthe question is, would you want to scrap a 7960 to make it work ;p
07:23.23jdmjamboohelp im trying to be nice here... don't be rude
07:23.26jdmjamboohelp im trying to be nice here... don't be rude
07:23.34terraHomejdm, pay me
07:23.37terraHomepaypal me
07:23.39terraHomeand i will helpyou
07:23.40Firestrmi saw them in a pile of electronics and telco rubble at capitol iron in victoria
07:23.44jdmjambooact like a professional
07:23.49terraHomei am.  pay me.
07:23.55letherglovhey now
07:24.07letherglovI've got my hat on the ground, flipped over
07:24.11letherglovtoss some in mine too
07:24.17jdmjamboois you just say that before rather than speaking like idiot one
07:24.18terraHomefire: i'd love to get ahold of a few
07:24.39jdmjamboocupal
07:24.43FirestrmterraHome, i will go get a price for you and let you know..
07:24.47SedoroxQuestion... trying to help pranav with fwd.. when I dial him.. and when he dials me.. its busy.. anyone else having problems with fwd being busy? (he's SIP and I'm iax2...)
07:25.14FirestrmterraHome, i need to go there to buy my new gps toy..
07:25.28terraHomewhat toy?
07:25.30terraHomewe sell GPS
07:25.32terraHomegpstools.com
07:25.40Firestrmgarmin 60cs
07:25.44*** join/#asterisk zoa (zoa@82.103.76.147)
07:26.08`Sauronterrahome: got any trimble gps receivers?
07:26.11terraHomehttp://gpstools.com/components/catalog/product.html?pid=506&cat=375
07:26.17terraHomesauron, don't think so
07:26.28`Sauronbah humbug
07:26.28terraHomefree shipping on that GPS, btw
07:26.29Firestrmim about to drop some cash down on it.. they are the cheapest in town.. $450.00
07:26.37`Saurongarmin can eat my shorts, trimble's where it's at
07:26.50*** join/#asterisk cc (~cc@byte.fedora)
07:26.53terraHomeits hard for us to compete w/ the big box stores on garmin
07:26.59`SauronThe funny thing is, y'alls $400 garmin GPS doesn't have more than $150-200 worth of parts inside
07:27.04terraHometheybuy in mega-quantities
07:27.06Firestrm`Sauron, garmin saved my life more times than i can count.. im a garmin man foreer
07:27.18terraHomei like garmin just fine
07:27.21terraHomeworks good enough
07:27.28terraHomemy buddies in Iraq use them every day
07:27.32Firestrmwhen you are here.. www.vrl.ca/ocarc, you want garmin.
07:27.46`Sauronblah blah blah
07:27.48CMikeoh
07:28.01`Sauronbust open your garmin unit
07:28.07CMikenice repeater
07:28.11`SauronI bet the gps receiver inside, is made by trimble :)
07:28.16FirestrmCMike, thanks
07:28.21terraHomenothing wrong with trimble or garmin
07:28.25terraHomeboth are good products
07:28.44`SauronI'm about to order a couple Lassen iQ's from trimble
07:28.46CMikewhat software did that coveragemap?
07:28.50letherglovwhoa
07:28.54letherglovwhat's that huge steel penis?
07:29.04CMikeI  have a few lassens..
07:29.09`Sauron$46 for a full featured GPS receiver, 12 channels, supplies full PVt and everything
07:29.12CMikehttp://www.trimble.com/lassenlp.html  thoose I think
07:29.21Firestrmletherglov, thats what we call it.. except its fiberglass
07:29.33letherglovradome?
07:29.45CMikeFirestrm: you built that repeater ?
07:29.50`SauronCMike: I think they're discontinuing the SQ and LP,  and replacing them with the iQ
07:29.55CMikeoh
07:29.58FirestrmCMike, yes me and one other guy..
07:30.14CMikelooks like a fun project.
07:30.20`Sauronthe iQ has specs overall, including power drain, which is what the LP was made for
07:30.36CMikeHm.. darn.. Now I have to go there and try the repeater.. :P
07:30.41*** join/#asterisk schurig (~schurig@p5080A330.dip0.t-ipconnect.de)
07:30.51FirestrmCMike, till it breaks in the middle of winter, and its the main SAR repeater..
07:31.18CMike<-- waiting for my callsign..
07:31.18CMike:P
07:31.19terraHomefirestrm, you need a weather station from us :)
07:31.19FirestrmCMike, the coverage is AWESOME!!..
07:31.21terraHomeweathertools.com
07:31.23BeirdoFirestrm: that's about where I wanna move some decade
07:31.31Beirdoif I do, I'll use yer repeater :)
07:31.35FirestrmCMike, VE7GEI
07:31.39BeirdoVA3HGJ (with no radios)
07:31.49CMikeI probably get SA0???
07:32.22CMikeSweden just started useing SA as prefix.. SM was apperently full..
07:32.25CMike:)
07:32.40FirestrmBeirdo, enjoy :) its the best repeater for the okanagan valley and beyond, i can pick it up 200km away in nakusp
07:32.46terraHomefirestrm: 866-859-7359
07:33.17FirestrmterraHome, I would LOVE a weather station for the repeater..
07:33.18letherglovhttp://www.garmin.com/products/rino/positionReport.html
07:33.20letherglovnow that's cool
07:33.25terraHomefirstrm: call that number
07:33.28terraHomeits toll free
07:33.33BeirdoMy dream (post lottery win) is to move out there and start a winery
07:33.34terraHomeits my rooftop weather station
07:33.37FirestrmterraHome, now?
07:33.42terraHomeexcuse my horrible texas accent
07:33.43terraHomesure
07:33.45Firestrmahhh.. cool
07:33.53`Saurontext-to-speech?
07:34.02terraHomecall and see :)
07:34.16`SauronI'm scared of y'all texans :)
07:34.33terraHomeyes, i have a gun.  yes, i like country music.
07:34.36terraHomeyes, i love barbecue
07:34.41CMike:)
07:34.46terraHomeand yes, i dip copenhagen snuff.
07:34.54BeirdoYes, your ex-governor's haunting the world
07:34.54letherglovI bet you voted for bush too
07:34.55FirestrmterraHome, cant call it from canada :(
07:35.01terraHomehahah beirdo
07:35.09terraHomecan't call 1866 from .ca eh
07:35.11terraHomehrmmmm
07:35.19terraHomei dont have a local DID
07:35.29terraHomehrmmm
07:35.31terraHomemaybe i need one
07:35.31Beirdonot unless you get US50/CDN toll-free
07:35.34`Sauronyou can hop through FWD
07:35.39Beirdothat one's likely US48
07:35.46terraHomehow hard is it to get FWD up and running?
07:35.48Firestrmi might be able to route it though iconnect..
07:35.49terraHomedoes it take long?
07:35.58`Sauronterra: I had it up in like, 10 minutes
07:36.01terraHomeshit, i need to sign up for a DID, anyway
07:36.07terraHomeFWD url?
07:36.14FirestrmBeirdo, any idea how much a us50/can tollfree costs?
07:36.15terraHomenm
07:36.37Beirdoif you are lucky and can find them, not much
07:36.40`SauronHum
07:36.40FirestrmterraHome, ya.. i can route through FWD.. let me try
07:36.44`Sauronthat's not bad texas accent
07:36.45*** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it)
07:36.50moonwickFirestrm: what are the grey tubes in http://www.vrl.ca/ocarc/images/DSC00129.JPG ?
07:36.54moonwickbatteries?
07:37.05Beirdorumour has it that livevoip.com has them cheap
07:37.07CMikeOh well.. gotta work .. local time 08:37 am
07:37.08CMike...73
07:37.11`SauronI was expecting the full southern drawl
07:37.12`Sauron:)
07:37.23Beirdohe said Texas, not Alabama
07:37.37terraHomei have a texan accent
07:37.43terraHomenot necessarily southern i guess
07:37.45Beirdoor like my bud in Tennessee.  silly redneck :)
07:37.46kkshey guy, it may be stupid question, where i can set the SMSC number?
07:37.48terraHomewe have our own accent
07:38.01Beirdoyes you do
07:38.06terraHomesomebody just got the weather station :)
07:38.09Beirdoand your own history too
07:39.27`SauronI called it
07:39.44hmodes...
07:39.46`SauronI was going to make something similar
07:39.53`Sauron'cept, have it all be text-to-speech
07:39.54terraHomethose weather conditions are updated every 5s
07:42.04FirestrmterraHome, thats a cool system.. you dont have much of an accent, (or my alberta accent cant diferentiate between alberta accent and texas accent)
07:43.00Firestrmalberta/texas... brothers. not much different.. except texans dont carry as much firepower :)
07:43.06terraHomehahah
07:43.07Beirdoheh
07:43.18BeirdoFirestrm: Alberta is Canada's Texass
07:43.23Beirdoer Texas
07:43.28Beirdofreudian slip
07:43.29Firestrmabsolutly
07:43.50Beirdonow if only UAlberta would call me for an interview :)
07:43.58terraHomethink again
07:43.58terraHomehttp://chrissnell.com/my_guns.jpg
07:44.17Firestrmyou know when your alberta when... all the highway deer crossing warning signs are shot to hell..
07:44.37terraHomethe *school bus stop ahead* signs are shot to hell here
07:45.06BeirdoOf course now the Oilers may not exist long enough for me to see a home game SHOULD I get hired out there
07:45.13Beirdoand screw the Flames very much :)
07:45.25FirestrmterraHome, albrta too.. except they try to target the picture of the child's head
07:45.42Beirdocan you tell I'll fit in in Edmonton? :)
07:45.45terraHomethats my favorite one of those handguns
07:45.54terraHomebecause my grandfather carried it in WWII
07:46.12terraHomeits also the most badass of the four
07:46.20Beirdomy grandfather built bombers
07:46.27terraHomei want to sell that Sig
07:46.31terraHomebut it won't fetch enough
07:46.43Sedoroxnight all.. or should I say morning
07:46.44Sedorox:-p
07:46.50Beirdoye olde Lancaster (dam busters)
07:47.19*** part/#asterisk djin (~djin@gridfox.xs4all.nl)
07:48.57FirestrmterraHome, this is my gun http://www.snipercentral.com/pm_sm.htm
07:49.22*** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se)
07:49.29Firestrm$6000.00 and worth every penny
07:49.44terraHome*CLI> Feb 17 01:49:07 NOTICE[14399]: chan_iax2.c:5869 socket_read: Registration of '616306' rejected: Registration Refused
07:49.48terraHomeugh, FWD doesn't like me
07:50.11FirestrmterraHome, i think its FWD.. im getting the same messages lately
07:50.26FirestrmterraHome, it was working yesterday..
07:50.30terraHomenice weapon
07:51.24moonwickyeah, that's a texan accent :)
07:51.25*** join/#asterisk eivindtr (~Eivind@193.91.146.34)
07:51.26FirestrmterraHome, dropped a 4point buck from a treestand this year.. 450 yd shot, took out the heart, dead center where i put the shot
07:52.16FirestrmterraHome, i have the 300 win sm version
07:52.33terraHomedang.
07:52.55terraHomeim about 4/10 on the M16A4 @ 300yds
07:53.00FirestrmterraHome, he was just finishing up with his girlfriend, i waited to he could die happy :)
07:53.39BeirdoFirestrm: take it to Ottawa and see if you can't get rid of some Liberals so the Conservatives can finally rule, eh?
07:54.14FirestrmBeirdo, don't tempt me.. im just bairly able to restrain myself..
07:54.59terraHomedammit FWD
07:55.01terraHomeregister, you bastard
07:55.15Beirdohehe
07:55.18FirestrmterraHome, its broken. im not registering either..
07:55.32terraHomeok
07:57.41FirestrmterraHome, if i buy from gpstools, what courier do they use for shipping?
07:58.01terraHomeoh duh, i forgot to activate my iax2 account
07:58.13terraHomefire, well for your GPS, we'd probably use FedEx Ground
07:58.18*** join/#asterisk pif (ldm@zenon.apartia.fr)
07:58.19terraHomein fact
07:58.21pranavcan anyone help me with the fwd stuff
07:58.21terraHomei know we would
07:58.22FirestrmterraHome, that would also do it.. although i still think they are broken..
07:58.55pranavwhen i call to an fwd number it rings but then there is no response
07:59.02terraHomefire, by default, we don't accept international orders...but if you get ahold of me tomorrow i can hook you up
07:59.06FirestrmterraHome, ground not the best for canada.. i get assraped for brokerage fees.
07:59.11terraHomeyep
07:59.16terraHomeFedEx International we can ship
07:59.20terraHomethough its not free :)
07:59.33terraHomeour FedEx rates are *very* cheap
07:59.39terraHomesee what we charge people for overnight
07:59.42terraHomeUS$7.99
07:59.46terraHomewon't find that anywhere.
08:00.19Firestrmya no kidding..
08:00.43pranavhi terrahome
08:01.58terraHomeyou just paypal the money and we ship the same day
08:02.12terraHomeoops that was for privmsg
08:05.41*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
08:06.56pranavhi Zeeek
08:07.03terraHomefirestrm, the Davis Instruments stuff rules
08:07.08terraHomethere is a nice Perl module
08:07.12terraHomeDevice::WxM2
08:07.16terraHomei contributed code for it
08:07.24terraHomemakes it easy to pull data off the device
08:07.27terraHomeand then you just AGI it
08:07.44Zeeekgood morning gentlemen
08:07.55wasimmorning Zeeek, any packages from .de?
08:07.55terraHomei wrote a little daemon in perl that pulls the data off the serial port and stores it in a Storable
08:08.03terraHomeand my AGI just reads the storable
08:08.13FirestrmterraHome, nice, i will have to pull it into a serial stream, so i can transmit it over the radio
08:08.38*** join/#asterisk djin (~marius@62.58.40.196)
08:08.39Firestrmterracon, does the windspeed go up to 100mph?
08:09.34ZeeekIs there a way to hit a key during a call to interrupt it, hang up and execute a "wait for DMTF" section?
08:09.57Zeeekwasim - been waiting for you heh heh - nothing ATM
08:10.13Zeeekit would be great for today or tomorrow
08:10.38Zeeekbut was something sent? Das ist das Qveschun
08:11.14Zeeekwasim, Das ist das Qveschun
08:11.18terraHomefirestrm, im not sure
08:11.27terraHomeim sure it does
08:11.43wasimZeeek: ok, now its beginning to piss me off to
08:11.58Zeeekwhat did I di ? :)
08:12.00FirestrmterraHome, ive personally been in 60mph winds up there, at that wasnt even with a storm.. clear sky..
08:12.11wasimZeeek: no, no, not you, me
08:12.20Zeeekwasim email me the phone number and I'll call them if you want
08:12.22terraHomehttp://www.weathertools.com/components/catalog/product.html?pid=357&cat=276
08:12.26terraHomethat's a good one for you
08:12.38terraHomehttp://www.weathertools.com/static/products/davis/6150_spec.pdf
08:13.17*** join/#asterisk chaoscon_ (~ph33r@chaoscon.user)
08:14.06Firestrmterracon, im a little suspious about the tiny strip of what appear to be plastic holding the windspeed/direction unit..
08:14.44Zeeekwasim tell me about it!
08:14.56Firestrmwe have had solar panel's ripped off by the wind, that were supported with 1" steel strapping and 5/8 bolts
08:15.08Zeeekwasim I've enabled queries if you wanna msg me
08:16.29Beirdonight
08:17.34terraHomefire, we can get all sorts of long-range radios for the davis stuff too
08:17.34terraHomefirestrm, do you have a computer at your station?
08:17.34terraHomeit would be nice to have a solar-powered Soekris box + Davis station
08:17.34terraHomefirestrm, you'd also want a heated rain bucket ideally
08:17.34terraHomeand a webcam
08:17.35terraHometo check snow depth
08:17.38terraHomewebcam + yardstick
08:18.06moonwickheh
08:18.19terraHomehey, someone with FWD
08:18.21terraHometry this #:
08:18.29terraHome616306
08:19.35terraHomeanyone?
08:19.52Zeeekwhat happens?
08:19.58Zeeeksomething automatic?
08:19.59terraHomeits my weather station
08:20.00terraHomeyes
08:20.02Zeeekok
08:20.10terraHomejust wanna make sure it works
08:20.10Zeeekiax or sip?
08:20.17terraHomeIAX..shouldn't matter
08:20.22terraHomethats a FWD number
08:20.44terraHome*CLI> Feb 17 02:20:04 NOTICE[14399]: chan_iax2.c:5766 socket_read: Rejected connect attempt from 65.39.205.121, requested/capability 0x4/0x4 incompatible  with our capability 0xfa00.
08:20.46terraHomewierd-o
08:20.49terraHomehrmmm
08:20.58Zeeekso far not even a ring on iax
08:21.06terraHomehang on
08:21.26terraHomehrmmm
08:21.27terraHomecodec
08:21.39FirestrmterraHome, all i get is fast busy
08:21.42ZeeekI'm on zaptel btw tried sip
08:21.49Zeeekone ring and then death
08:22.01Zeeeka horrible fiery death ending wioth congestion
08:22.12FirestrmterraHome, 711 only for FWD
08:22.32Firestrmdisallow all
08:22.42Zeeekcorrect
08:23.12Firestrmallow = ulaw, alaw
08:23.58Firestrmallow=729  == puke on call.
08:24.26Zeeekdepending on versions you may not be able to put many codecs on single line, yes?
08:25.27terraHomelemme see
08:25.27terraHomeugh
08:25.27terraHomecable just went out
08:25.27terraHomefirst time...ever.
08:25.34Firestrmwierd, i cant call *393 numbers through my spa-3000 into my * box.. another bug to squish
08:25.58terraHomeok
08:25.59terraHomehang on
08:26.49terraHomeok try again please
08:26.56terraHome616306
08:27.01Firestrmsame result
08:27.28terraHomeok hang on
08:27.39terraHomeok
08:27.47Firestrm*393 goes though with my wisip.. but blocked by spa-3000.. must be a dialing plan thing on the spa
08:28.06Firestrmstill death.
08:28.10Zeeekok works!
08:28.22Zeeekok works!
08:28.34Zeeekooops hung up after the humidity reading
08:28.37terraHomesweet!
08:28.52terraHomeyeah, thats how it works now
08:28.58Zeeeknow, you only need someone who gives a shit about the weather 6000miles away :)
08:29.05terraHomehahahah
08:29.13terraHomeim happy it works tho
08:29.20Zeeekand maybe put a "thanks for calling" or a commercial at the end
08:29.21terraHomeit will make a nice test # for new FWD users
08:29.25terraHomei had one, zeek
08:29.27terraHometook it off
08:29.31terraHomelemme put it back on
08:29.51Zeeekhow about wait(1) playback(goodbye)
08:29.53terraHometry now :)
08:29.57Firestrmhmmm, im still getting fast busy..
08:30.00ZeeekI believe you :)
08:30.08terraHomeit worked for zeeek
08:30.13terraHomewonder why u get that
08:30.18terraHomei dont even see your call coming in firestrm
08:31.01Firestrmits connecting to FWD, i can see it in the logs, but it keeps coming back busy..
08:31.03terraHomeim registered to FWD via IAX..
08:31.06terraHomewierd
08:31.38Firestrmrings though to my other FWD number
08:31.53terraHomelemme restart again...
08:32.13terraHomeok
08:32.32Firestrmsame result... fast busy..
08:32.42terraHomeyeah, i dont even see any console msgs
08:32.45Firestrmone ring, then fast busy
08:33.09terraHomejust tried FWD's CallMe application
08:33.10terraHomeand it works
08:33.24Firestrmit most have somehing to do with fwd-iax.. maybe iax-iax dont work.. let me try my softphone
08:33.35terraHomethat's kind of wierd
08:33.39terraHomeim on fwd-iax
08:33.44terraHomeok, see you now
08:33.59*** join/#asterisk eipi (eipi@153-218-114-200.fibertel.com.ar)
08:34.05terraHomei guess thats you
08:34.07terraHomedunno :)
08:34.08Firestrmthat though my softphone.. ie.. no iax
08:34.32Firestrmmy * should be 23927
08:34.36terraHomemy * connects to FWD via IAX2
08:34.45terraHome<PROTECTED>
08:34.48terraHomethat was u
08:35.17Firestrmthat was fwd gateway.. same ip for me as well.. i think iax to iax over FWD is broken..
08:35.31terraHomereally?
08:35.31Firestrmsip to iax works though..
08:35.53Firestrmwhen i call sip it goes though.. when i call iax, fast busy..
08:35.58terraHomestrange.
08:36.05terraHomeFWD's sip servers?
08:36.44FirestrmterraHome, yes but dont even bother trying to connect to FWD via SIP.. all you'll get is a migrane headache
08:36.54terraHomeok :)
08:36.58Firestrmfrom asterisk that is..
08:37.22terraHomei really need to get us a local DID
08:37.35*** join/#asterisk shaZwaz (~sasda@216-236-205-66.reverse.newskies.net)
08:37.41Firestrmi tried for a week with no luck, intil someone here called me a dumbass for even trying and pointed me to the iax gateway..
08:37.41shaZwazhi room
08:37.43*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
08:37.44terraHomei could get that one through nufone
08:37.46terraHomebuts its in michigan
08:38.08Zeeekget a toll-free from nufone
08:38.14ZeeekI just used IAX and it worked
08:38.21terraHomewe have a toll free from them
08:38.23FirestrmterraHome, i will have Victoria, Kelowna and Nelson DID's soon.
08:38.27terraHomebut it wont work for our foreign customers
08:38.36shaZwazto register to FWD and other Services I have bind SIP to my Public IP ?
08:39.00shaZwazwhy doesn't it bind to 0.0.0.0
08:39.16FirestrmterraHome, ive found an investor that wants to give me $$$ to play with asterisk and DID's
08:39.22terraHomenice.
08:39.47Firestrmi love it when ppl throw money at me.. i go buy toys like gps :)
08:39.56shaZwazimplicit you around ?
08:40.02terraHomeok night all
08:40.10Zeeekhey why not hook up a GPS and people can call and see where they are? :)
08:40.15FirestrmterraHome, ya me too.. gnite..
08:40.41Firestrmzeeek, www.canaprs.com, enter in ve7gei-2
08:41.03Zeeekn ot found
08:41.32Firestrmhttp://www.canaprs.net/locate.php?stn=ve7gei-2
08:42.26MakenshiMorning.. first Galileo satellite is going to be launched soon :>
08:42.26Makenshii think thats the only good thing to have ever come out of the eu
08:43.00Firestrmzeek or http://www.canaprs.net/locate.php?stn=ve7gei-1
08:43.12Firestrmdepends on vehicle im in
08:44.06*** join/#asterisk zoa (~zoa@pirus.securax.be)
08:44.11Firestrmyou can even send me messages on my radio, but im not sure where that site is anymore..
08:44.38*** part/#asterisk jdmjamboo (jdmjamboo@202.69.190.233)
08:45.17*** join/#asterisk DonX (don@tool.sparkhosting.net)
08:45.42DonXHow can I find out what timing device asterisk is using?
08:46.22FirestrmDonX, good Q, i dont know.. but i want to know as well
08:46.55ZeeekFirestrm what is that?
08:47.18FirestrmZeeek, see DonX's question
08:47.29shaZwazwhu do I have to bind sip to my public IP to register to FWD and other providers
08:47.41shaZwazanyone has any idea ?
08:48.10shaZwazthat leaves my internal sip phones not working
08:48.14FirestrmshaZwaz, sip-FWD through NAT.. forget it.. i tried for over a week, use IAX
08:48.34Zeeekno I mean what is the site and the radio messages? I was too busy to look
08:48.39shaZwazbut its not only FWD ..
08:48.48shaZwazits same for voipuser.org
08:48.49ZeeekI looked but too fast since I was on another site filling something out
08:49.19shaZwazunless I bind sip to my static IP my number doesn't register
08:49.21FirestrmZeeek, i cant find the radio message site anymore, server is down :( i guess i will have to set up an aprs message server..
08:49.38Zeeekbut what IS that site?
08:49.53Firestrm<PROTECTED>
08:50.02Zeeekfunny I do SIP FWD through NAT with no prb
08:50.02Firestrm?
08:50.14Zeeekbut not 0.0.0.0 binding
08:50.23shaZwazwhy not ?
08:50.39Firestrmits an APRS tracker site.. it tracks my location realtime, using gps and packet radio
08:50.40shaZwazI need my internal sip phones
08:50.50Zeeekwait maybe I do but I have only one etheenet interface
08:51.09ZeeekFirestorm via your cellphone?
08:51.21shaZwazZeeek thats the problems it doesn't work with 0.0.0.0 binding
08:51.27FirestrmZeeek, via 2m packet radio 144.390 mhz
08:51.37ZeeekI was W0DBJ for years
08:51.44Zeeek... .... .. _
08:51.59shaZwazis there a way I can use my internal clients while binding sip on my public ip ?
08:52.08shaZwazor an other way around ?
08:52.10Zeeeklet my license expire - my First Class Phone too
08:52.12Firestrmmorris code hurts my brain..
08:52.26Zeeekoh that's why you're stuck on 2M
08:52.28FirestrmZeeek, license never expires in canada.. lifetime..
08:52.49Zeeekwhere I live that's how drivers licenses work. Much more practical
08:52.56FirestrmZeeek, untill they open up nocode HF.. which was supposed to be done allready
08:53.05Zeeekexcept for when people get on the road in their 70's
08:53.22Firestrmi cant drive.. im 55
08:53.40Firestrm:)
08:53.53Zeeeklearning code should be mandatory. WHat will you do in STeven Sagal disaster film when the survors are trying to tap out H E L P ?
08:54.19Zeeekyou are 55? SHit you are almost old enough for the dean's title here
08:54.24ZeeekAlmost
08:54.32Firestrmthey should learn ascii or baudot
08:54.51Zeeekbaudot is just an ecoding method like a codec
08:55.01Zeeekor ascii for that matter
08:55.06FirestrmZeeek, no im 30's.. think of the song , i cant drive 55.. change the lyrics..
08:55.15Zeeekok I was worried
08:55.32Zeeekwhy would anyone over 50 waste their precious time here?
08:55.48Zeeeklike I'm doing right now...
08:56.23Firestrmim making sure im extra tired for my charter flight in the morning :)..
08:56.34Zeeekheh
08:57.21Firestrmi love the look the passengers get when the pilot falls asleep halfway thorough the checklist :)
08:57.44Zeeekwhat no alcohol on the breath?
08:58.47FirestrmZeeek, no, im a strict 8 hours bottle to throttle kind of pilot.. fatigue has an antidote.. caffeen, alchol also has an antidote.. time..
08:59.11Zeeekactually it depends on how much you drink
08:59.29Zeeekthere was an excel calculator making the rounds before the holidays
08:59.50Zeeekyou entered what you drank at what time and it traced the curve of BA (blood alc)
09:00.14*** join/#asterisk dstevens (~dstevens@cpc3-ches1-4-0-cust87.lutn.cable.ntl.com)
09:00.18Firestrmim well beyond coffee, ive graduated to sucking on unroasted beans.. %100 caffeen uptake with that method.. just watch the heart rate..
09:01.50Firestrmmaybe thats why i only get about 3 hours of sleep each night ;)
09:03.02Firestrmanyhoo.. must run... TTFN..
09:03.25pranavhello Zeeek
09:03.33Zeeekhello again pranav
09:04.00pranavi am facing a problem in the fwd
09:04.11shaZwazwhy doesn't it bind to 0.0.0.0 dammn
09:04.35pranavwhen i call an fwd number it says"sorry invlid extensio"
09:05.02murangdInorder for asterisk to call SIP to SIP calls
09:05.08murangdmust I signup with a SIP provider
09:05.09*** join/#asterisk welby (~welby@80-192-119-210.cable.ubr04.dund.blueyonder.co.uk)
09:05.13murangdand insert the user/pass in my sip.conf?
09:05.15pranavwhen i call to 612,or 613 it rings once and then i get no response
09:05.21*** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62)
09:05.28murangdpranav: do you have asterisk installed on your own server
09:05.30Zeeekmurangd no you need no provider
09:05.38murangdzeek: ok cool thanks
09:05.52pranavi have pasted my sip.conf and extensions.conf in pastebin.ca/6001
09:05.54pranavyes
09:06.07Zeeekpranav IAX or SIP ? Sometimes FWD IAX is flaky
09:06.17murangdzeeel: I
09:06.17pranavasterisk i sinstalled on my own server
09:06.28pranavsip
09:06.33murangdzeeel: I just installed asterisk on my own server but for some reason I can't make any calls
09:06.40murangdso I am trying to figure out where I went wrong
09:06.47murangdprobagly I configured my .conf incorrectly
09:07.16murangdZeeek: do you have an SIP phone number I can do a test call on and tell me if it RINGS on your side
09:08.13Zeeektry the weather station at 616306
09:08.24murangdok trying that one now
09:08.25pranavok i'll try
09:08.29*** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net)
09:08.39murangdpranav: is your SIp server operational?
09:08.43murangdif so can I do a test call to you
09:08.48murangdyou don't have to s peak just tell me if it rings
09:09.12*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
09:09.21murangdanyone alive I can do a test SIP TO SIP call to
09:09.46Zeeekcall pranav! you are made for each other
09:10.00murangdpranav: give me your SIP number
09:10.12pranavsip number?
09:10.15murangdyeah
09:10.22murangdso I can test call you
09:10.30murangdI just installed asterisk on my server and want to make sure its worknig
09:10.39pranavbut there is no sip number
09:11.00murangdwell any VOIP number
09:11.20pranavno
09:11.27murangdcan anyone provide me with VOIP number so I may do a test call to make sure my asterisk setting is working
09:11.48murangdZeeek: is there a way I can test asterisk without calling anyone to make sure its working?
09:12.02Zeeekcall 613 the echo test
09:12.06murangdthanks
09:12.11Zeeekit's there for that very reason
09:12.33murangdCouldn't Start Call
09:12.42murangdshould I put * before the number
09:12.47pranavi tried calling that 613, it rings once but then i get no response
09:13.06murangdI am getting 'Couldn't Start Call' error
09:13.20pranavmurangd thats for me
09:13.22*** join/#asterisk Starblazer (star@proxy.vfm.extremepcgaming.net)
09:13.50StarblazerHello, I just noticed something weird with the Agents system, specifically AgentCallbackLogin
09:14.13Starblazerif you specify the device directly via queues.conf (EG. SIP/100), you will be able to transfer calls blindly w/o waiting for them to pick up)
09:14.26Starblazerand you get musiconhold for when you put the device on hold
09:14.39Starblazerhowever, if you use an extention from your extentions.conf, it will not give you the hold music
09:14.54Starblazernor will it let you blind-transfer-before-connection
09:15.16pranavZeeek:any guesses
09:15.49Zeeekabout what?
09:15.55pranavif you can check , i have pasted my sip.conf and the extensions.conf in the pastebin.ca/6001
09:16.15pranavi am not able to call any fwd number
09:16.50Zeeekpranav I think I do see something
09:16.57Starblazeryou use IAX for fwd now
09:17.13Starblazernot SIP
09:17.13ZeeekDial(SIP/${EXTEN:1}@fwd.pulver.com)
09:17.26Zeeek[fwd]
09:17.30Zeeekpranav
09:17.32pranavya
09:17.42Zeeekyou see above the two lines do not correspond IMO
09:17.42murangdOk I have a question
09:17.49murangdwhere can I register to get a VOIP phone number
09:18.00Starblazermuranged: you're talking about a regular phone number?
09:18.01*** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net)
09:18.04soulz-hello all
09:18.06Zeeekpranav change [fwd] to [fwd.pulver.com]
09:18.07murangdStarblazer: no a voip number
09:18.22soulz-does anyone know why i get this error message?
09:18.23soulz-channel.c:2173 ast_channel_make_compatible: No path to translate from IAX2
09:18.24Starblazerwhat do you mean, VoIP number, those are assigned by your provider
09:18.24pranavok i'll change it
09:18.29Zeeekor Dial(SIP/${EXTEN:1}@fwd.pulver.com) to @fwd
09:18.46soulz-http://pastebin.ca/6009
09:18.47murangdStarblazer: I've just installed asterik on my server and I want to have a friend do a test calling to me
09:18.55murangdStarblazer: should I signup here https://www.e164.org?
09:18.57Starblazermurangd, via his land-line phone?
09:19.06murangdStarblazer: no he also has a soft phone client
09:19.14murangdStarblazer: he uses broadband
09:19.21Starblazerokay
09:19.37Starblazerif he has a softphone client, then he can just call your system at [startingcontext]@yourip
09:19.49murangdStarblazer: you mean like
09:19.53murangduserid@myip
09:20.05murangdfor example
09:20.07murangd-- Registered SIP '1001' at 162.84.229.224 port 5060 expires 3600
09:20.12murangd1001@myip.com
09:20.19soulz-starblazer: u seemed to be the one with the answers now, can u help my problem when u get a chance?
09:20.48murangdStarblazer: btw I'm a php programmer so if you need any php scripts done.. I can help you in that department free of charge
09:20.55Starblazersoulz-, ugh.
09:21.09soulz-starblazer: thanks dude
09:21.31Starblazermurangd, example.  If you were to call my asterisk server, using a softphone client, you can dial s@asterisk.myip.com
09:21.35Starblazerand my system will pick up
09:21.45Starblazerdoes your friend subscribe to FWD?
09:21.51Starblazeryou can tie your asterisk system into FWD also
09:21.56Starblazersoulz-, lemmie check
09:22.19*** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
09:22.35murangdStarblazer: I've just tried to do a test call to myself.. I've got this error
09:22.37murangdFeb 17 04:34:56 NOTICE[14107]: pbx.c:1358 pbx_extension_helper: Cannot find extension context 'from-sip'
09:22.42Starblazerah
09:22.51Starblazermurangd, calling from where?
09:22.57murangdStarblazer: myself to myself
09:22.59Starblazersoulz-, are you trying to bridge calls?
09:22.59soulz-murangd: looks like ur context is not right
09:23.05soulz-yes i am
09:23.17murangdsoulz-: can you explain a little bit more what you meant about context not being correct
09:23.19Starblazerit looks like the two systems dont have compatable phone codecs
09:23.25Starblazermurangd, in your sip.conf
09:23.28Starblazeryou have a line called
09:23.32Starblazercontext=[whatever]
09:23.39Starblazerunder YOUR device
09:23.46soulz-starblazer: so i can't make a system talking via g729 to talk with a tdm?
09:24.02Starblazersoulz-, I dont know that much about asterisk my friend.  About as much as I've played with is pure data
09:24.15Starblazernot bridging from an actual hardline-type-thing to software
09:24.29soulz-ok dude
09:25.13murangdStarblazer: can I msg you to tell you what my context settings are
09:25.18pranavzeeek:i get nothing when i dial the fwd number
09:25.29Starblazerpranav, why dont you change your fwd number to asterisk?
09:25.32Starblazeriax
09:26.17pranavbut my configuration is correct then y does'nt the call go?
09:26.21murangdStarblazer: I've msg you context details
09:26.34Starblazerpranav, http://www.freeworlddialup.com/content/view/full/1501
09:26.41Starblazerthat's how I have my asterisk system bridging to FWD
09:26.55pranavbut you have through iax or sip
09:27.17Starblazeriax
09:27.23StarblazerI used to have it thru sip
09:27.40pranavok so was it working properly with sip
09:27.48Starblazermonths ago
09:27.57Starblazernow I have it working properly thru IAx
09:27.58StarblazerIAX
09:28.00pranavif you can check , i have pasted my sip.conf and the extensions.conf in the pastebin.ca/6001
09:28.14*** join/#asterisk chaoscon (~ph33r@chaoscon.user)
09:30.30*** join/#asterisk Stonekeeper (~c252e507@server0.expresshosting.net)
09:36.44*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
09:36.56*** join/#asterisk _PiGreco_ (~a@adsl-215-48.38-151.net24.it)
09:40.19*** join/#asterisk Delvar (~irc@83.146.53.34)
09:40.47Delvarmorning all
09:44.42murangdhas anyone ever gotton this error  == Spawn extension (default, s, 5) exited non-zero on 'SIP/2
09:44.50JunK-Cisnt an error.
09:45.02murangdJunK-C: I receive this message when someone tries to call me
09:45.08JunK-C<PROTECTED>
09:45.08JunK-C<PROTECTED>
09:45.13murangdoh
09:45.17JunK-Csee? :)
09:45.20murangdyes
09:45.45JunK-Cisnt when some1 is calling ya, its when a context reach "his end"
09:46.04murangdI had another asterisk user dial my sip number and he got this message Got SIP response 404 "Not Found" back from 65.125.228.1
09:46.14murangdJunK-C: in which configration file do I set the user?
09:46.33murangdI've added myself in the sip.conf file and I am able to log in but for some reason when someone calls me they get 404 error
09:46.37JunK-Cits cause ya dont have that extension in ur context that user is calling.
09:46.44JunK-Csip.conf
09:46.56murangdJunK-C: ok I'm in my sip.conf what should I look for
09:46.59JunK-Cya did a "reload" command after?
09:47.06JunK-Ccontext=blah line
09:47.29*** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au)
09:47.35murangd[1001]
09:47.35murangdtype=friend
09:47.35murangdusername=1001
09:47.35murangdsecret=1001
09:47.35murangdcontext=default
09:47.36murangdhost=dynamic
09:47.40murangdallow=ulaw
09:47.41murangdJunK-C: you mean that line
09:48.23murangdJunK-C: or I am totatly off
09:48.24JunK-Cin ur default context, add a line exten => 1234,1,NoOp(blah);
09:48.29JunK-Cif 1234 is ur sip number
09:48.34JunK-Cand do a reload
09:48.49JunK-Cso 1234 would be 1001
09:49.05murangdJunK-C: you must forgive me.. in my sip.conf file
09:49.11murangdyou want me to put
09:49.11ozJames79hi can anyone help i have setup another * box and its behind a firewall  i have put it on the dmz and ported forwarded just in case everything works except on calls i have no audio  codec is set to ulaw any ideas .....thanks in advance
09:49.21*** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg)
09:49.26murangdJunK-C: context=default exten => 1234,1,NoOp(blah);
09:49.29murangdlike so?
09:49.32JunK-Cno
09:49.50JunK-Cin ur extensions.conf, in ur default context, add exten => 1001,1,NoOp(blah);
09:50.07JunK-Cur dialplan is ur extensions.conf
09:50.20JunK-Cya should read more infos on www.voip-info.org
09:51.11*** join/#asterisk Xander77 (~Alex@exten-halls-243.soton.ac.uk)
09:51.55*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
09:53.38ozJames79hi can anyone help i have setup another * box and its behind a firewall  i have put it on the dmz and ported forwarded just in case everything works except on calls i have no audio  codec is set to ulaw any ideas .....thanks in advance
09:53.56JunK-Cmurangd: so it works?
09:54.04murangdJunK-C: one sec
09:54.37murangd;[context]
09:54.37murangdexten => 1001,1,NoOp(blah);
09:54.49murangdno wait one sec
09:55.33Starblazernah
09:55.49StarblazerI clued him into it, methinks
09:56.05JunK-CStarblazer: he'll be able, let him just few times. :)
09:56.34Starblazeryeah, he's taking the typical 'new person' approach' of just because it's decalred @ sip.conf, doesn't mean that it's declared everywhere yet
09:56.46Starblazerdeclared*
09:56.47murangdStarblazer: ok added the line
09:56.56Starblazerkks, did you restart your server?
09:57.55StarblazerGot SIP response 481 "Call Leg Does Not Exist"
09:58.23murangdStarblazer: yeah I 've restarted
09:58.30murangdit says
09:58.38murangd<PROTECTED>
09:58.39murangd<PROTECTED>
09:59.26Starblazerdo you have your softphone to register with your asterisk box?
09:59.34*** join/#asterisk TeLLuS (~johan@h187n2fls31o858.telia.com)
09:59.45murangdok now for some reason
09:59.47murangdwhen I do
09:59.49murangdshow channels
09:59.54murangdit's not showing me logged in
10:00.01JunK-Cisnt in show channels
10:00.05JunK-Cits sip show peers
10:00.26murangdName/username              Host            Dyn Nat ACL Mask             Port     Status
10:00.26murangd1001/1001                  162.84.229.224   D          255.255.255.255  5060     Unmonitored
10:00.27murangd1 sip peers [1 online , 0 offline]
10:00.29murangdit shows me connected
10:00.46JunK-Cya need 2 sip no?
10:01.27murangdJunK-C: I don't understand what you mean
10:02.25JunK-Ctell us how ya making ur test exactly.
10:02.33Starblazerokay, here's the "test"
10:02.41StarblazerI've got a dial line in my asterisk server
10:02.41murangdJunK-C: I had starblazer call my sip number
10:02.58Starblazerexten => 1236,1,Dial(SIP/1001@his-ip-number)
10:03.18*** join/#asterisk Stonekeeper (~c252e507@server0.expresshosting.net)
10:03.19Starblazerdoing a Dial(SIP/s@his-ip) brings up the "Cool, you've got it installed" message
10:03.41JunK-Cits exten => 1236,1,Dial(SIP/1001@his-HOSTNAME); no?
10:03.43*** join/#asterisk slav_jb (~k@pirus.securax.be)
10:03.46JunK-Coups, IP :)
10:03.52JunK-Cbit tired huh
10:03.59Starblazeryou? lol
10:04.03JunK-Cwhatcha get when dial his 1001 ?
10:04.07JunK-Con ur side.
10:04.09StarblazerI get a 481
10:04.12StarblazerGot SIP response 481 "Call Leg Does Not Exist"
10:04.24JunK-Cmurangd: ya did a reload?
10:04.37JunK-Cmurangd: do a show dialplan 1001@default
10:04.38murangdJunK-C: yes, before starblazer would get a 404 user does not exist
10:05.03murangd[ Context 'default' created by 'pbx_config' ]
10:05.03murangd<PROTECTED>
10:05.03murangd
10:05.04murangd-= 1 extensions (1 priorities) in 1 contexts. =-
10:05.36*** join/#asterisk qwerp (~abc@219.95.105.74)
10:05.46JunK-Ci told ya to put exten => 1001,1,NoOp(blah);
10:06.03StarblazerNoOp?
10:06.23JunK-Cfor a test
10:06.43Starblazeralright
10:06.48StarblazerI see what you mean
10:06.56*** join/#asterisk bowman (~bowman@snert3.tal.de)
10:07.04Starblazerooooo
10:07.09*** join/#asterisk mak_ (~mak@privat.ua-online.net)
10:07.11Starblazernow i get a "wrong password on auth"
10:07.11murangdok done
10:07.16murangdStarblazer: try now
10:07.17mak_hi
10:07.26murangdStarblazer: I've just restart asterisk
10:07.33JunK-Cmurangd: and in ur sip.conf, in ur 1001 class, add a qualify=yes
10:07.34JunK-Cand reload
10:07.47StarblazerForbidden - wrong password on authentication for INVITE to
10:08.08Starblazerokay
10:08.46murangdok just reloaded
10:08.54JunK-Cmurangd: now do a sip show peers
10:09.13murangdName/username              Host            Dyn Nat ACL Mask             Port     Status
10:09.13murangd1001/1001                  162.84.229.224   D          255.255.255.255  5060     UNREACHABLE
10:09.26JunK-Cits UNREACHABLE
10:09.59murangdhow come?
10:10.07Zeeekhello again
10:10.08JunK-Cdunno, its a soft-phone?
10:10.08murangdI mean.. what would cause a 'unreachable' error
10:10.11JunK-Clo Zeeek.
10:10.14*** join/#asterisk ZX81 (matt@222-153-114-115.jetstream.xtra.co.nz)
10:10.17murangdJunK-C: yes a soft-phone "firefly"
10:10.26JunK-Cclose it, and re-open it.
10:10.32*** part/#asterisk slav_jb (~k@pirus.securax.be)
10:10.42Zeeekanyone know anything about WIndows networking? I just added a LAN card so I can be connected to asterisk and the offcie LAN
10:10.45JunK-Cmurangd: past the last line of sip show peers too.
10:10.59murangdok just closed
10:11.01murangdand re-opend it
10:11.03ZeeekI'm trying to figure out how to specify which connection to use for ssh and other clients
10:11.27murangdName/username              Host            Dyn Nat ACL Mask             Port     Status
10:11.27murangd1001/1001                  162.84.229.224   D          255.255.255.255  5060     UNREACHABLE
10:11.37murangdJunK-C: still un-reachable.. what is it suppose to say?
10:11.51JunK-Cits suppose to give ya a "time"
10:12.09JunK-Clike:
10:12.15JunK-Cgate1*CLI> sip show peers
10:12.15JunK-CName/username              Host            Dyn Nat ACL Mask             Port     Status
10:12.15JunK-C102/102                    (Unspecified)    D   N      255.255.255.255  0        UNKNOWN
10:12.15JunK-C101/101                    (Unspecified)    D   N      255.255.255.255  0        UNKNOWN
10:12.15JunK-C100/100                    192.168.1.208    D   N      255.255.255.255  5060     OK (5 ms)
10:12.16JunK-C3 sip peers [1 online , 2 offline]
10:12.17Starblazer100/100          209.103.209.231  D          255.255.255.255  5060     OK (88 ms)
10:12.31JunK-CStarblazer: help him to get his stuff connected.
10:12.42StarblazerI've never used his SIP client before.
10:13.00Starblazerplus it's 4am here, and I should really head off to bed
10:13.13mak_when I putting 1.call to /var/spool/asterisk/outgoing Can I put to 1.call two Application: ?
10:13.16JunK-CStarblazer: its 5:13 here :)
10:13.37JunK-Cmak: i dont think so.
10:13.59modulus_penis
10:14.04Zeeekmoose
10:14.08modulus_oops wrong window!
10:14.11JunK-Cmooo
10:14.12JunK-C:)
10:14.13JunK-Chehehe
10:14.15Zeeekwho knows anything about windows here?
10:14.24mak_JunK-C: but if I need for example SetLanguage and then SayDigit what should I do ? :)
10:14.37JunK-CZeeek: ive just learned how to click here and there, but after all this :)
10:14.49Zeeekno, seriously
10:14.51JunK-Cmak_: priorities!
10:14.58JunK-C1,setlanguage(fr)
10:14.59ZeeekI have this great ASTERISK
10:15.03JunK-C2,SayDigits(123);
10:15.04Zeeekon a linux box
10:15.12modulus_Zeeek, i once did something cool on windows
10:15.32Zeeekbut I have 2 LAN cards on Win box and can't figure out how to tell ssh client to use the connection to talk to asterisk
10:15.35Delvardeltree c:/windows ?
10:15.36mak_JunK-C: tnx :)
10:15.47wasimZeeek: route
10:15.59Zeeekwasim in a cmd line option?
10:16.01modulus_Zeeek, that's more of a networking issue no?
10:16.05wasimZeeek: yep
10:16.06Zeeekyes it is
10:16.08DelvarZeeek: assuming they are on different subnets use route ad bla....
10:16.33Zeeekerrr hmmm both routers are 192.168.1.1
10:16.43Delvarthats the problem then :)
10:16.44Zeeekis that... bad?
10:16.49Delvarthat IS bad
10:16.53Zeeekshit
10:16.58modulus_indeed
10:17.05Zeeekah but wait
10:17.08Delvaryour por windows box wouldnt have a clue where to end teh packet
10:17.12JunK-Cmak_: instead of     print OUTPUT "Application: Wait\n";
10:17.12JunK-C<PROTECTED>
10:17.17JunK-Cjust use
10:17.22*** join/#asterisk qwerp (~abc@219.93.57.58)
10:17.23JunK-C<PROTECTED>
10:17.23JunK-C<PROTECTED>
10:17.23JunK-C<PROTECTED>
10:17.27JunK-Cinside ur perl script.
10:17.38Zeeekit works the other way - asterisk knows to send me stuff on 192.168.1.60
10:17.39JunK-Cwith Priority
10:18.05Delvarif you have 2 nics int eh same computer they HAVE to be on different subnets or weird things happen
10:18.17*** join/#asterisk murangd (~nukaidc@pool-162-83-240-155.ny5030.east.verizon.net)
10:18.21murangdsorry I got disconnected
10:18.22murangd<Starblazer> 100/100          209.103.209.231  D          255.255.255.255  5060     OK (88 ms)
10:18.24murangd*** Disconnected from IRC.freenode.net
10:18.27Zeeekshit
10:18.28murangdthat was the last message I saw
10:18.40ZeeekI hate changing network stuff
10:18.46Zeeekalthough on linux it's easy
10:19.00Delvari know windows can be a bitch
10:19.06Zeeekcan they be one apart like 192.168.2.x ?
10:19.17Zeeekwindows SUCKS!
10:19.17Delvari still havnt found out how to perminantly modify the routing table on my windows box yet...
10:19.32Delvaryes
10:19.40Zeeekyes /8 ?
10:19.49Zeeek(or is that /24)
10:19.54Delvarone on e 192.168.1.x and the other on 192.168.2.x with netmasks of 255.255.255.0
10:19.56murangdJunK-C: still alive
10:20.09Delvari can never rmeber :/
10:20.12JunK-Cmurangd: yes, get ur peers connected with ur sip-phone
10:20.14Delvarremember*
10:20.14Zeeekme neither
10:20.27mak_JunK-C: why two
10:20.28mak_12:17 < JunK-C>     #print OUTPUT "Context: default\n";
10:20.28mak_12:17 < JunK-C>     #print OUTPUT "Context: in\n";
10:20.30Zeeek<PROTECTED>
10:20.35Delvari think its a /24
10:21.04JunK-Cmak: cause i've mal-pasted
10:21.09murangdJunK-C: I am not sure WHY its says UNREACHABLE
10:21.09JunK-C<PROTECTED>
10:21.09JunK-C<PROTECTED>
10:21.09JunK-C<PROTECTED>
10:21.23JunK-Cmurangd: cause ur peers cant "talk" to *
10:21.32mak_:)
10:21.47Zeeekso I change the router to be 192.168.2.1 - chenge that on linux in the rc and everything in one and 42 ?
10:21.50murangdJunK-C: ok so where would that error lie in.. is it one of the .conf files?
10:22.09Zeeek"all is one"
10:22.11JunK-Cnon, in ur firefly configs
10:22.29JunK-Cmak_: sorry its like 5:30am here, a bit tired huh? :)
10:23.09*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
10:23.25DelvarZeeek: so your * box is ont eh 192.168.2.x range now?...
10:24.12Zeeeklong story short: got a new static ip connex, swithed the office LAN stuff to the older dynamic one - the networks were never connected to each other
10:24.20Zeeekbut now I wanna talk directly to asterisk
10:24.32Zeeekinstead of thru the internet to go 10 meters
10:25.12murangdcan someone try to connect to my asterik server
10:25.16murangdto see if everything is working
10:25.19ZeeekI hope there's only one place I need to change the gateway in linux?
10:26.46*** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62)
10:26.57murangddoes anyone know alterantive softphone other than firefly
10:27.56*** join/#asterisk Delvar (~irc@83.146.53.34)
10:28.00*** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
10:28.05Delvarbah mIRC br0ked
10:28.17JunK-Cmurangd: x-lite
10:28.57DelvarZeeek: what did you say last?
10:30.01JunK-Clook at this:
10:30.02JunK-Chttp://pastebin.ca/6019
10:30.05JunK-Creally strange.
10:40.18*** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au)
10:41.26*** join/#asterisk meppl (~mephisto@62.158.37.97)
10:51.39*** join/#asterisk cjk (~cjk@80.92.64.103)
10:52.06*** join/#asterisk [ro]nic3try (~nic3try@p3.pub.ro)
10:52.08cjkhi, was anyone of you able to get a price quotation from pulver.com for their pulver communicator software?
10:52.11[ro]nic3tryre all
10:52.49[ro]nic3tryhas anyone know how to set the default codec wich asterisk use ?
10:53.47*** join/#asterisk fishboy1669 (proxyuser@62.69.81.129)
10:54.07oejCodec for which protocol?
10:54.15fishboy1669morning guys
10:54.21[ro]nic3trysip
10:54.53CMikehiyas oej
10:54.53[ro]nic3trydefaul is g711
10:55.00JunK-C[ro]nic3try: its ulaw by default.
10:55.03JunK-Cright
10:55.10JunK-Cg711u or a
10:55.28[ro]nic3tryand i want to use g729
10:56.04PakiPenguin[ro]nic3try: buy it
10:56.44[ro]nic3try???
10:57.44JunK-CRO: ya need to pay for that codec
10:58.20PakiPenguin[ro]nic3try: if you dont want to pay , use gsm!
10:59.16[ro]nic3trybut if i give : show codecs
10:59.22[ro]nic3tryi see i have g729
10:59.42[ro]nic3tryhow do i set it to be default on all cals to asterisk
11:01.29PakiPenguinshow codecs , just give a detail about codecs
11:02.54fishboy1669has anyone ever tried this
11:02.55fishboy1669v
11:02.56fishboy1669http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20forwarding
11:03.57fishboy1669is anyone awake here?!
11:03.58fishboy1669lol
11:04.12Delvarno
11:04.32Delvarlooks interesting why not give it a go?
11:04.47[ro]nic3tryok, so .. just to understand.. asterisk does not suport g729? or i cann't use it as default codec
11:05.06Delvarasterisk does support g729 you just need to buy licances
11:05.17fishboy1669i am trying to get it working
11:05.29*** join/#asterisk cc (~cc@byte.fedora)
11:05.43Delvarto set as default codec for all calles in your [general] context add disallow=all allow=g729
11:05.44fishboy1669the cli shows stuff happning but when i dial the extention i have put the frowaring on it still dials
11:06.18[ro]nic3trythat i do in sip.conf ?
11:06.48*** join/#asterisk Jnel (~bit-logic@c2-239-1.pta.dial.mweb.co.za)
11:08.14Jnelhi all; I need help with a sip phone config
11:08.36JnelI am trying to get 2 S3020's talking, but..
11:08.57Jnelthe phones ring and disconnect on answer..
11:09.30JnelThe error message is that the frame type requested is type 1 and the native type is 4..
11:10.08JnelFurther i says that there is no patch to convert from type 1 to 4..
11:10.20Jnelany solutions or suggestions ??
11:11.32*** join/#asterisk HjemmeRoyK (~roy@83.80-203-29.nextgentel.com)
11:12.43*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
11:12.49JnelAny ideas on this problem??
11:13.02Zeeekhuman stupidity has no limits - especially my own !
11:13.39Jnellol.. tell me about it..I feel the same way
11:14.44ZeeekSo I changed the gateway address on one router and couldn't ping it. It took all of 20 min torealize that changing the gateway wasn't enough, I had to change the ip itself. INCREDIBLY STUPID!
11:15.36*** join/#asterisk usam (~usam@203.156.37.115)
11:15.52Jnelmmmm...sounds like you're in a mood to ponder things...try and help me with this..
11:15.55usamis it possible to change a codec on the fly when using SIP ?
11:15.57cjkwhich stund do you recommend, the one from vovida or the mystun?
11:16.26JnelI've got 2 S3020's which disconnects on anser...
11:16.30Zeeekstun.e164.org
11:16.43Zeeekmember of the asterisk community - one of US ;)
11:16.44HjemmeRoyKwhen is it you'll really need stun?
11:17.11ZeeekRoyK Hej! just the man for my windows network question
11:17.13murangdcan someone explain to me what is DID
11:17.15HjemmeRoyKwe're running SIP behind all sorts of NAT.....
11:17.21HjemmeRoyKZeeek: hehe
11:17.22murangdin relations to phone numbers
11:17.23JnelThe error says that the frame type requested is 1 and not 4 as the native...
11:17.34ZeeekDID is a number people can call to ring into your * server
11:17.34murangdis DID a kind of phone number?
11:17.42ZeeekDirect Inward Dial
11:17.43murangdzeek: ok cool thanks
11:17.49murangdah ok I see
11:17.55cjkZeeek, you might be right that that stun is great, but i want a stun server where i have the control. if it goes down i know home to blame
11:18.01ZeeekRoy you know how to tell a particular program to use a certain route?
11:18.03Jnelsetting is... and no patch to convert... any solution??
11:18.12murangdZeeek: do you know where I can get some cheap DID numbers.. like a provider that provides DID numbers?
11:18.14HjemmeRoyKZeeek: as in ip route?
11:18.18Zeeekcjk ok, I don't use stun anyway
11:18.38ZeeekRoyK yeah, I have a second card now and want to talk ssh directly to the asterisk network box
11:19.15Zeeekand maybe even connect an IAXY or SIP phone some day (router has a couple of extra slots)
11:19.17murangdZeeek: why are you buying hardware cards to support asterisk? your making outgoing calls?
11:19.39Zeeekmurangd nothoing to do with that - but I do have three Digium cards
11:19.51murangdZeeek: what exactly are the purpose for Digium cards
11:19.59murangdZeeek: they are used ONLY for PSTN dialing right?
11:20.13Zeeekmine are to connect three regular phones (FXS) and to connect two phone lines (FXO)
11:20.13*** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au)
11:20.26Zeeekyes you don't n,eed hardware murangd
11:20.26murangdah I see
11:20.41murangdwell Junk-C help me setup my asterik server
11:20.50murangdand now I can receive VOIP to VOIP calls
11:21.01murangdno I want to do VOIP to PSTN calls
11:21.09murangdso I need to get a VOIP termination provider
11:21.15murangdbut I don't really know any good ones
11:21.25murangdalso I am looking for a place that provides DID numbers
11:21.36Zeeekyes so why not go looking for one? nufone, voicepulse, iconnecthere, voiptalk, voipjet
11:22.10Zeeekjust add .com to any of the above and stir
11:22.20Jnelhi all; I need help with a sip phone config
11:22.29JnelI am trying to get 2 S3020's talking, but..
11:22.37Jnelthe phones ring and disconnect on answer..
11:22.45JnelThe error message is that the frame type requested is type 1 and the native type is 4..
11:22.55Jnelany solutions or suggestions ??
11:23.07*** join/#asterisk libpcp (libpcp@210.16.20.5)
11:23.18Zeeeksounds like the codec curse
11:23.33*** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk)
11:23.36Jnelmeaning?
11:23.52Zeeekmeaning look at your codec definitions
11:24.06libpcpwould it be possible setup ? if (!($row[10] > 6)) { ?>
11:24.06libpcp<PROTECTED>
11:24.06libpcp<?php
11:24.06libpcp}
11:24.11libpcpops sorry
11:24.25libpcpSIP server ---> E1 Channel Bank ---- PSTN
11:24.47*** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl)
11:25.05libpcpsorry guys, i thought i was not on my clipboard
11:28.26Zeeeks'ok RoyK I found it and now everything is cool as in 42 !
11:30.55*** join/#asterisk TheEmperor (TheEmperor@218.111.48.89)
11:31.03*** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net)
11:32.49Zeeekwho was it said something about not being able to make persistant routes? I found that too
11:33.03HjemmeRoyKroute -p add
11:33.05HjemmeRoyKiirc
11:33.31Zeeekyep - and I just looked, it adds the route to the regisrty. But why didn't you answer me whan I was looking ? :)
11:34.07HjemmeRoyKtrying to do some work as well as helping you :)
11:34.20ZeeekI'll accept that this time
11:34.27Mavviewhat is an persistant route?
11:34.28HjemmeRoyK"this time" :)
11:34.30Zeeek~lart RoyK
11:34.45HjemmeRoyKMavvie: it's saved over a reboot
11:34.52Zeeekpersistant means it stays between boot
11:34.59Zeeek~lart RoyK even more
11:35.05jetscreamerone tha tpersists over reboots
11:35.06MavvieHjemmeRoyK: oh. euhm. windows?
11:35.07jetscreameriirc
11:35.11HjemmeRoyKMavvie: yeah
11:35.17Zeeek~lart RoyK leaving nothing but a smalll greasy spot
11:35.20HjemmeRoyKMavvie: I used to be an MCSE on 3.51 and 4.0 :P
11:35.23Mavvieaha. no miracle I wasn't familiar with the term.
11:35.25HjemmeRoyK~kill Zeeek
11:35.27jbotACTION shoots a inverse  meson gun at Zeeek
11:35.28HjemmeRoyK~LART Zeeek
11:35.30jetscreamerroute -a or something in windows i think
11:35.41HjemmeRoyKjetscreamer: route -p, as stated above
11:35.45jetscreamero
11:35.46HjemmeRoyK~lart jetscreamer
11:35.47jetscreamernm
11:35.49jetscreamer:/
11:35.52fishboy1669anyone know of a advance extentions.conf dial plan documnetation i need to understand it propper now
11:35.54Zeeekyou have to understand that most of the world's offices run WIndows
11:36.04Zeeekso a minimum network knowledge is nec
11:36.09fishboy1669doing fancy stuff like call redirect config from phone
11:36.13HjemmeRoyK~docs
11:36.14jbotDocumentation 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
11:36.14fishboy1669hi zeek
11:36.14Zeeekbut now, I rule the world with two LAN cards
11:36.27MavvieZeeek: don't worry about my network knowledge :-P
11:36.28Zeeek~lart docs
11:36.53ZeeekI'm not worried - only my own is worrying
11:36.54fishboy1669wish jbot could be more specific
11:36.59ZeeekStarter tutorial:
11:36.59Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
11:36.59Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
11:36.59Zeeekhttp://www.automated.it/guidetoasterisk.htm
11:36.59ZeeekTHE reference of the moment:
11:36.59Zeeekhttp://www.asteriskdocs.org
11:37.05fishboy1669~docs extentions.conf
11:37.14ZeeekThe dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls.
11:37.14Zeeekhttp://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650
11:37.16Zeeekthat one
11:37.19fishboy1669aha last one is new
11:38.00Zeeekso the problem of CallerID notification on the Windoze boxes is solved
11:38.08fishboy1669cheers zeek
11:38.13fishboy1669~dov
11:38.18Zeeekuntil the next ip change :(
11:38.21fishboy1669~docks
11:38.25fishboy1669~docs
11:38.26jbotDocumentation 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
11:38.31ZeeekThe dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls.
11:38.31Zeeekhttp://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650
11:39.11fishboy1669dont i know it lol
11:42.41*** join/#asterisk Igor-BZ- (~root@62.123.121.61)
11:43.31*** join/#asterisk Luhiwu (~marsosa@200.63.89.209)
11:45.26*** join/#asterisk pranav (sameer@202.149.48.200)
11:45.27Igor-BZ-hi :)
11:45.34pranavhi
11:46.24murangdis this site voipjet.net
11:46.26murangdloading for anyone?
11:46.39*** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl)
11:46.45HjemmeRoyKmurangd: works for me *tm(
11:46.51HjemmeRoyKmurangd: works for me (tm)
11:46.56jerliquedoes anyone have any experience with agent logouts?
11:47.25HjemmeRoyKmurangd: er. no
11:47.30HjemmeRoyKmurangd: it doesn't
11:48.03pranavi am not able to make calls with fwd , i can make calls to pstn, mobile, internlly
11:48.10libpcpfrom what ive experience, voipjet.net has a noise on the voice quality
11:48.40pranavwhen i make calls to fwd numbers it says "sorry its an invalid extension
11:49.13*** join/#asterisk trym (~trym@linux.debian.us)
11:49.26pranavi have pasted my sip.conf and the extensions.conf in the pastebin.ca/6001
11:49.52pranavcan anyone tell me what is the mistake
11:50.22trymI have installed spandsp to have asterisk receive faxes. When a fax call is made to asterisk, asterisk starts whining about RFC3389. I also notice that the volume spandsp/asterisk is communicating with varies.. which is not normal for a fax session. Any suggestions?
11:50.25murangdwhat's the best PSTN termination provider in terms of COST and Quaility
11:50.32HjemmeRoyKpranav: change your passwords :}
11:51.01murangdwhat's the best PSTN termination provider in terms of COST and Quaility
11:51.14pranavbut then i have registered with that password to that number
11:51.31pranavu mean to say shold i register again
11:51.41trymHjemmeRoyK: have you used spandsp with asterisk ?
11:51.49jerliqueCan asterisk do a desktop->fax gateway?
11:51.49HjemmeRoyKnope
11:52.00HjemmeRoyKjerlique: should work, on a LAN
11:52.12HjemmeRoyKotherwice you'll need t.38, which isn't finished
11:52.14Igor-BZ-I'm trying www.wipphone.com I have try with my BGT
11:52.15jerliqueare there some docs?
11:52.27HjemmeRoyKjerlique: see spandsp
11:52.31jerliquethanks
11:52.47HjemmeRoyK~spandsp
11:52.54HjemmeRoyK~fax
11:52.55jbotWell, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically.
11:53.48jerliquehehehe
11:54.03Makenshithe first fax was an interesting mechanical contraption
11:54.21Makenshimaking use of a swiging arm and a counterbalance
11:54.24Igor-BZ-When asterisk call pvt->read in a channel? channel il ANSWERED, and asterik call pvt->write rigth...
11:54.40pranavwhen i make calls to fwd numbers it says "sorry its an invalid extension
11:55.12*** join/#asterisk [Hug] (~ss@195.244.154.200)
11:56.05jerliqueAre ppls here integrators of asterisk systems, or users within their business?
11:58.23*** join/#asterisk hajekd (~hajekd@mail.idoox.com)
11:59.57hajekdWhats up to VoipJet? Their DNS is screwed.
12:00.33pranavcan someone tell me what is the mistake
12:01.46*** join/#asterisk Weezey (Weezey@lan6.LO.iasl.com)
12:02.01Weezeyif I'm connected to asterisk CLI, how do I exit without bringing down the server?
12:03.04tzafrirpranav, is that your reall password in the file?
12:03.21tzafriror have you bothred modifiying it?
12:03.34JunK-CWeezey: exit
12:03.55murangddoes anyone know any Public VOIP Gateway
12:03.57tzafrirOK, already noted
12:04.19WeezeyJunK-C: hmm, that must not have worked before because I wasn't using -r to connect to it.  thanks.
12:04.34JunK-Cif ya've started it with -c
12:04.40JunK-Cthere's no way i know
12:04.47JunK-Cstop now, then safe_asterisk
12:04.52JunK-Cand asterisk -rv
12:04.59JunK-Cthen exit gonna works
12:05.23pranavyes thats what i have registered with
12:05.26Weezeyokay, thanks.
12:05.40pranavno i have never modified it
12:06.28pranavis there a problem of NAT
12:06.32bowmanthanks for giving out your password.
12:07.44pranavbowman:are u telling me
12:07.52WeezeyI have two SPA-3000s, one connected to POST and the other connected to a Norstar ATA, for some reason the norstar ATA just keeps ringing when there's an incoming call and it's configured the same way.  Outgoing calls throught the norstar ATA work fine however.
12:08.16Weezeys/POST/POTS/
12:09.13*** join/#asterisk Specky[W] (~sspecken-@p508EC9F0.dip0.t-ipconnect.de)
12:09.37*** join/#asterisk GodThor (~ninja@212.110.95.139)
12:10.25*** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
12:10.28pranavtzafrir: is that a mistake of NAT
12:10.47*** part/#asterisk Specky[W] (~sspecken-@p508EC9F0.dip0.t-ipconnect.de)
12:10.55GodThorwhen i start asterisk there is not h323 protocols, i have installed pwlib, openh323
12:11.12GodThordo i must install something else?
12:11.51tzafrirpranav, NAT will generally cause issues with the RTP data channels and not with the SIP control channel
12:12.13tzafrirIIUC
12:12.22pranavok fine
12:12.46tzafrirdo you send them the correct number?
12:13.11pranavyes
12:13.24miller7anyone here familiar with zapras and dial in access to * box?
12:13.38tzafrirHow have you verified that?
12:16.22GodThoranyone help with h323?
12:17.48pranavsee i know a few numbers like 613,612,55555 so tried calling them
12:18.08*** join/#asterisk Tornad (~regis@81.56.183.143)
12:18.22Tornadhi
12:18.27pranavwhen i call them it rings once or twice but then i get no further response
12:19.13WeezeyGodThor: don't quote me or anything, but I think you have to change the asterisk source to have the h323 code from the openh232 site, then re-compile.
12:19.24pranavbut when i call to somebody's fwd number it says "invalid extension"
12:20.36Igor-BZ-there is a channel... che chan_h323 or chan_oh323 is a wrapper from openh323 and asterisk...
12:20.54*** part/#asterisk [ro]nic3try (~nic3try@p3.pub.ro)
12:21.37Igor-BZ-U can found all here: http://www.inaccessnetworks.com/projects/asterisk-oh323
12:22.32Igor-BZ-use pwlib and openh323 version ONLY FROM inaccess-network...
12:22.52Igor-BZ-there is howto on this site ^^^^^^
12:23.39GodThorbecause i didnt install them manually it comes with asterisk package /from cvs
12:24.58GodThorok i would try inaccess-network, thanks
12:27.42Igor-BZ-...mmm.... I'm using this driver... :)
12:29.45*** join/#asterisk Banter (Banter@209.119.214.81)
12:30.19GodThorthanks Igor, thats my real name also
12:30.40*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
12:30.44pranavsee when i call  a few fwd  numbers like 613,612,55555 so tried calling them
12:30.52pranavbut when i call to somebody's fwd number it says "invalid extension"
12:31.02pranavsee when i call  a few fwd  numbers like 613,612,55555 so tried calling them
12:31.16pranavwhen i call them it rings once or twice but then i get no further response
12:31.50*** join/#asterisk r1 (~erwan@www.thiscow.com)
12:32.47pranavhi igor
12:34.59Igor-BZ-hi pranav :)
12:35.22*** join/#asterisk mrempire (~user1@h71032.upc-h.chello.nl)
12:35.26Igor-BZ-do U have a correct configuration on extensions.conf?
12:36.25pranavtell me what to so
12:37.04pranavi have pasted my sip.conf and extensions.conf in the pastebin.ca/6001
12:38.24pranavand the calls are going to pstn, mobile and internally only to a fwd number they are not going
12:38.38pranavdo you want to see what comes on the cli screen when i dial the fwd number
12:40.42*** part/#asterisk pranav (sameer@202.149.48.200)
12:40.56*** join/#asterisk pranav (sameer@202.149.48.200)
12:41.20pranavsorry i got disconnected
12:41.30pranavnow i am back
12:42.27Igor-BZ-ok I'm looking...
12:42.38pranavya
12:45.34*** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg)
12:47.13Igor-BZ-there is a problem on your conf....
12:47.20pranavya tell me
12:47.23Igor-BZ-use this "template" http://www.voip-info.org/wiki-Asterisk+FWD+NAT+Config+Example
12:48.42pranavok i'll go through this site
12:49.25pranavbut is there a mistake of NAT
12:49.50Igor-BZ-why?
12:50.08Igor-BZ-can U open port on your router?
12:50.51pranavbcos someone yesterday gave me a command like "traceroute_ _
12:51.32pranavit was a address after this and he told that your asterisk server is behind a NAT
12:52.35murangdIgor-BZ-: have you tried out voipuser.com ?
12:52.48Igor-BZ-never...
12:53.29pranavsee  the wan is connected to the router and from router to the switch and switch to the asterisk pc
12:53.49Zeeekwhat is the NAT problem this time?
12:54.15pranavi don't know whether it is a NAT problem
12:54.22*** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
12:54.28Zeeekwell, are we still on the same FWD stuff after all these hours?
12:54.36pranavya
12:54.51pranavthe calls are not going
12:54.52Zeeekwhat is the problem -I've been disconected a while)
12:55.21Zeeekpranav look at this
12:55.25Zeeekhttp://willypick.mindsay.com/?entry=10
12:55.35pranavsee i am not able to make fwd calls
12:55.35Zeeek^^^^^^^ The asterisk config that dare not speak its name: Double NAT! ^^^^^^^^^^^^
12:55.43*** join/#asterisk amer (~aaa@203.99.60.27)
12:56.26amermy setup is *sip ----- *IAX------*IAX------*SIP
12:56.34Zeeekaha
12:56.43ZeeekI've never had a problem
12:56.44amermy setup is A----*sip ----- *IAX------*IAX------*SIP-----B
12:56.48pranavya i'll gothrough this
12:56.50*** join/#asterisk mcukstorm (~mcukstorm@neo.matrix-lan.net)
12:56.51Zeeekwhat is your router setup on both ends
12:57.03amerwhen I call from A to B, caller ID is messed up
12:57.15amerI am unable to figure out whats going on
12:57.28Zeeekaha
12:57.34mcukstormhi all, does any one know a pinout i can use for connecting an RJ45 from the 400P card (FXO module) to a BT PSTN Line (UK)
12:57.41Zeeekand you are using setcallerid etc in extensions to FWD
12:57.52amernope
12:58.10ameri dont want to change the callerID
12:58.16Zeeekamer speaking for the 12455 people here?
12:58.25*** join/#asterisk didz_ (didz_@200.218.192.52)
12:58.27Zeeekwho is amer?
12:58.32Zeeekthe other pranav?
12:58.41amerno
12:58.44ameramer is amer
12:58.53Zeeekoh got confused with pranav
12:59.02amernp
12:59.13Zeeekso why is callerid messed?
12:59.15Zeeekamer
12:59.16amerso do u see any problems with my setup?
12:59.17pranavwhat happened?
12:59.26ameri just see 00000000
12:59.30Zeeekamer I use that all the time
12:59.43amerinstead of the caller ID of the SIP phone A
12:59.51Zeeekwhat does arterisk CLI see ?
13:00.11Zeeekinsert a NoOp(${CALLERIDNUM}) in the dialplan
13:00.19*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
13:00.27amerhmmmm
13:00.42Zeeekto see what is coming in from your phone, ya see?
13:00.43amerI see the correct caller ID
13:00.56amerok let me try that
13:01.02Zeeekwhich is like Hey Now <2000>
13:01.05Zeeek??
13:05.17*** join/#asterisk h3x (~Justino@adsl-065-013-150-019.sip.msy.bellsouth.net)
13:05.31h3xhi
13:05.42*** join/#asterisk meshugga (philip@loeblich.linuxteam.at)
13:05.56h3xguess who was at a convention i just went to
13:06.00h3xsysmaster
13:06.01meshuggahi
13:06.02h3xhahahahaha
13:06.13h3xi so wanted to give them a bunch of crap
13:06.18Zeeekshiksa?
13:06.24h3x"hi id like to buy an asterisk box from you for $50k"
13:08.21*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
13:09.17Weezeycan I play musiconhold in the background while transferring a call?
13:12.33Zeeekit should do that automagically
13:12.54h3xif the moh is set up right anyway
13:12.55h3xheh
13:12.55Zeeekcall yourself and see!
13:22.48*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
13:28.03*** join/#asterisk sangee (~rkuru@207.188.77.82)
13:28.56murangdZeeek: I've just setup an account at FWD and it registeries correctly
13:29.09murangdZeeek: but I am unable to make a call.. I am using X-lite softphone software
13:29.24murangdZeeek: do I have to dial 394 then the number inorder to make a call?
13:30.03Zeeekno you dial the number if registered directly to FWD
13:30.16Zeeekif not you need a dialplan extension
13:30.20netsurferhttp://www.theregister.co.uk/2005/02/17/spam_gets_vocal_with_voip/ <-- ffs that takes the piss
13:30.21Weezeynope, no ringing, no moh, not nuthin'.  Hrmmm...  On a completely new topcic, how come sometimes I can press buttons when the biatch is talkin' but sometimes it just ignores them?
13:30.22ZeeekThe dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls.
13:30.22Zeeekhttp://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650
13:30.44murangdZeeek: in my extenstion.conf I have this
13:30.44murangdexten => _394.,1,SetCallerId,voip@goldenbucks.biz
13:30.44murangdexten => _394.,2,Dial(IAX2/13569@fwdOUT/${EXTEN:3},60,r)
13:30.47murangdexten => _394.,3,Congestion
13:30.49sangeedoes asterisk supportx g723 and g729 codec?
13:30.49murangdsorry I mean
13:31.25murangdZeeek: is that what you are refering to when you say DIALPLAN?
13:31.25ZeeekSetcalleridnum(YOURFWDNUM)
13:31.26*** join/#asterisk _Brian (brian@unix01.voicenet.com)
13:31.33Zeeekand set the name to whatever
13:32.00ZeeekThe number one answer is FWD site has this info that is what they did that page for
13:32.22ZeeekIntyerested in FWD? FreeWorlDialup?
13:32.23Zeeekhttp://www.freeworlddialup.com/content/view/sitemap/2
13:32.28ZeeekThis is their site map
13:32.39Zeeekhttp://www.freeworlddialup.com/support/configuration_guide
13:32.47ZeeekThis is the CONFIGURATION GUIDE
13:33.19_Briantired of people asking Zeeek?
13:33.22ZeeekHereis all you need to know about FWD IAX:
13:33.22Zeeekhttp://www.freeworlddialup.com/advanced/iax
13:33.32Zeeekno, do I appear tired? :)
13:33.57bjohnsonsangee: I think g723 (is that gsm?) .. g729 is supported if you buy licenses (I think $10 each)
13:33.59_Brianjust a little bit :)
13:34.14murangdhttp://www.fwdout.net = freeworlddialup.com?
13:34.18sangeewhere can i buy the license?
13:34.20bjohnson~docs
13:34.21jbotDocumentation 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
13:34.23jetscreameri could ask what is freeworldialup? :o
13:34.31murangdZeeek: http://www.fwdout.net = freeworlddialup.com?
13:34.35jetscreamerj/k
13:34.45bjohnsonsangee: from digium or whoever owns g729
13:34.58ZeeekSo my new macro, let's see:
13:35.00Zeeekhttp://www.freeworlddialup.com/content/view/sitemap/2
13:35.00Zeeekhttp://www.freeworlddialup.com/support/configuration_guide
13:35.00Zeeekhttp://www.freeworlddialup.com/advanced/iax
13:35.00Zeeekhttp://www.freeworlddialup.com/support/forum
13:35.06sangeethx
13:35.15HjemmeRoyKhttp://hampage.hu/pdp-11/kepek/1103sys.jpg
13:35.23bjohnsonmurangd: all sorts of info on the wiki listed by ~docs
13:35.27Zeeekfwdout is Bellster after the lawsuit threat
13:35.42bjohnsonI thought fwd was included in the samples
13:35.43*** join/#asterisk CpuID (~none@CPE-203-45-152-22.qld.bigpond.net.au)
13:35.47Zeeeknew asterisk box RoyK ?
13:36.03Zeeekit is but now they do IAX and all, why not go to the source
13:36.32HjemmeRoyKZeeek: hehe. with an 18bit cpu
13:36.38Zeeekhttp://www.freeworlddialup.com/advanced/service_numbers
13:36.45HjemmeRoyKPDP-11
13:37.12ZeeekI used to program PDP-11 in RSX and RT11
13:37.17Zeeekunder, not in
13:37.28Zeeekused assembler and some Fortran
13:37.33felipexi have 2 * box connected via iax2 trunk
13:37.39Zeeekso far so good
13:38.10felipexwhen i try to place a call from *1 to *2 i have this messages
13:38.33felipexCall rejected by 192.168.255.23: No authority found
13:38.45felipexthis message in *1
13:38.59jetscreameri had star trek on papertape
13:39.21felipexin *2 i have Rejected connected attempt from 192.168.0.5
13:41.52*** join/#asterisk mjmac (~mjmac@cpe-68-175-244-78.maine.res.rr.com)
13:42.15bjohnsonfelipex: make sure the username and secrets are the same on both boxes .. use the host arg if you have any that are static ip addresses (then you don't need to register on that machine)
13:45.06*** join/#asterisk TheEmperor (TheEmperor@218.111.50.241)
13:50.19ariel_morning all
13:50.44oejMorning
13:50.58oejls -la
13:51.19oejOops, wrong window
13:51.35*** join/#asterisk zeedo (~notroot@www.bsrf.org.uk)
13:52.45amer:)
13:52.56Mocoej hey what up ?
13:54.19Zeeekmadonna naked
13:54.23Zeeekoops wrong window :)
13:54.37amer-- Executing NoOp("IAX2/edge@edge/4", "19790709709777") in new stack
13:55.02tzangerwerd to the goatherd
13:55.29amercallerid is correct but when its passed to a the SIP server from usr is chaged to asterisk
13:55.53amerwho can I make asterisk not to do this and pass on the actual callerID
13:56.00murangdamer: you have your caller id working correctly?
13:56.14murangdamer: when I dial a local number, it says 'Unknown Number'
13:56.51amerZeeek any ideas
13:57.04oejStable or head?
13:57.14oejCallerid is broken in head (personal opinion)
13:57.27junky[work]oej: since when?
13:57.38ariel_actually callerID is also strange on stable.
13:57.53amerCVS-v1-0-02/10/05-15:59:23
13:58.08ariel_I get it fine on the NoOp line but when it's sent to the phones like it shows the name but unknown number.
13:58.45junky[work]amer: isnt head.
13:59.00amerno but my problem is different
13:59.06ariel_I am going to upgrade to the stable release as of today. I have had it for over 1 week and read on the cvs that he removed the new id to the older stuff that was working.
13:59.18amermy setup is A----*sip ----- *IAX------*IAX------*SIP-----B
13:59.40amermy setup is A----*sip ----- *IAX1------*IAX2------*SIP-----B
14:00.20ariel_amer, your still using iax1?
14:00.40bjohnsonwe'll use the candlestick in the library.  Don't tell Colonel Mustard
14:00.42amerIAX2 received the ccorrect callerID but when it passes the call to *SIP it puts from user as asterisk@10.0.9.1
14:00.44bjohnsonoops wrong window
14:01.07amerno thats just to differentiate b/w 2 servers
14:01.17ariel_bjohnson, so funny "NOT"
14:01.37bjohnsonnow you're just being mean
14:02.44`SauronYEah
14:02.45ariel_bjohnson, really now.... I was trying to be funny.
14:02.49`Sauronquit being mean to my buddy...
14:02.53`Sauron;)
14:03.19amerhey guys what about my problem
14:03.36ameroej, you know whats wrong here
14:04.50EssobiHmm. I got like 5 sip peers I use to dial out.  Anyone have a suggestion how to fail them over to each other with outbound dialing?
14:04.52ariel_amer, I don't use head but it's a problem as well in my stable
14:05.51murangdamer: how does your extenstion.conf look
14:05.56murangdamer: mine is exten => _394.,1,SetCallerId,voip
14:05.59murangdamer: what's yours?
14:06.45amerI dont want to set any callerID, I just want asterisk to pass whatever callerID it gets
14:06.55*** join/#asterisk Ubuz (~momo@DSL217-132-49-219.bb.netvision.net.il)
14:07.26greendiseaseUbuz: manyanim
14:08.02Ubuzgreendisease: hakol tov
14:08.16amerenglish please
14:08.27Ubuzok, hello everybody
14:09.10UbuzI have a question about playing non gsm files with agi. I managed to record a vox file, but how can i play it?
14:09.24UbuzOr any other kind of file.
14:09.27junky[work]Ubuz: STREAM ?
14:09.32junky[work]STREAM FILE
14:09.44UbuzSTREAM FILE ignores the file
14:10.00junky[work]huh?
14:10.38mtqhconvert to gsm them stream
14:11.18UbuzWhy should I convert? If it can record, surely it can play.
14:11.31mtqhdid you do a |vox afterward
14:11.36mtqhare you in head or stable?
14:11.39greendiseasehey can someone donate a conference room for some sessions at linuxworld today
14:11.43mtqhbrb
14:12.32UbuzI didn't do a |vox after, because I don't know after what I should do it.
14:12.38WeezeyAll my incoming calls to my set are coming in with the proper callerid, but with asterisk as the number, no the number they're coming from.  SETCIDNUM is set correctly, how do I pass that to the phone?
14:12.51Weezeyerr CALLERIDNUM
14:13.18`Sauron<PROTECTED>
14:13.21greendiseaseJerJer: ping
14:13.31ariel_greendisease, I think bkw_ said he would last night.
14:14.13greendiseasehe did but hes not here now, and we need to get it set up soon
14:14.20TheEmperorwhat's the best softphone to use? iax2 and messenging
14:15.32ariel_TheEmperor, I use xlite for sip and diax for iax in windows.
14:16.05TheEmperorariel_ : where can i get diax from?
14:17.18ariel_TheEmperor, http://www.laser.com/dante/
14:18.00TheEmperorariel: thanks :)
14:18.58murangdariel_: why do you alterante?
14:19.09murangdI mean couldn't you just do SIP -- IAX via your asterisk setup
14:19.14murangdno need to use two different softphones
14:19.36TheEmperorwould it be better to use iax2 rather than sip on the softphone?
14:19.36fishboy1669can anyone shead light on what this does
14:19.40fishboy1669exten=s,2,Dial(Local/${temp}@pbx/n)
14:19.53murangddoes anyone have their called ID working successfully?
14:20.16Zeeekeveryone murangd
14:20.22fishboy1669lol
14:20.24murangdZeeek: well not I
14:20.25murangdlol
14:20.36murangdcan you paste me an example of your settings in extenstion.conf
14:20.42murangdwhere you have your caller id proberly setup
14:20.43Zeeekcallerid from where? a SIP caller to your asterisk?
14:20.47fishboy1669exten=s,2,Dial(Local/${temp}@pbx/n)
14:20.49fishboy1669?
14:20.53ariel_murangd, I use xlite or most of the setups. I use diax due to sometimes I run into a system that blocks my rtp stream.
14:21.16murangdZeeek: Yes a sip caller to asterisk to fWD to PSTN number
14:21.43Zeeekand where are you looking to see the cid?
14:21.55Zeeekon what phone?
14:21.55murangdZeeek: on local PSTN phone caller id's menu
14:21.59TheEmperorhow's xten's eyeBeam?
14:22.05amercan I set fromuser in Extensions.conf file
14:22.07murangdZeeek: is that possible or no?
14:22.20ZeeekFWD to PSTN? How are you doing that?
14:22.50murangdwoah
14:22.55murangdI'm just a newbee
14:23.05murangdZeeek: how do you dial PSTN numbers?
14:23.16bjohnsongreendisease: fwd?
14:23.18Zeeekfrom FWD? You don't usually
14:23.31greendiseasebjohnson: huh?
14:23.33Zeeekexcept at certain holiday promo times
14:23.38bjohnsonfwd has conf rooms
14:23.43murangdok my orginal question
14:23.48greendiseasewhats fwd?
14:23.52Zeeekhttp://www.freeworlddialup.com/content/view/sitemap/2
14:23.52Zeeekhttp://www.freeworlddialup.com/support/configuration_guide
14:23.52Zeeekhttp://www.freeworlddialup.com/advanced/iax
14:23.52Zeeekhttp://www.freeworlddialup.com/support/forum
14:23.52Zeeekhttp://www.freeworlddialup.com/advanced/service_numbers
14:23.53greendiseaseor whose fwd?
14:23.57greendiseaseah fwd
14:24.00greendiseasethat fwd
14:24.39Zeeekmurangd I still don't see how you are calling a PSTN line from FWD
14:24.43murangdexten => _8.,1,SetCallerId,voip
14:24.45murangdis that correct
14:24.46murangdor no
14:24.47Zeeekso I don't know about your prob
14:25.02murangdZeeek: well answer this exten => _8.,1,SetCallerId,voip <-- is that correct for caller id setup?
14:25.05Zeeekshow application setcallerid
14:25.11bjohnsongreendisease: I haven't used the fwd conf rooms .. but they say they are available
14:25.36*** join/#asterisk Tornad (~regis@81.56.183.143)
14:25.36Zeeekmurangd you need to read the docs to see how comannds work - try to wiki there is a complete list
14:25.41bjohnsonmurangd: you're trying to set outgoing callerid right?
14:26.04bjohnsonmurangd: look at the superdial macro on the wiki .. it does a bunch of things you will want
14:26.12Zeeekto talk to fwd you can set callerid to your FWD number
14:27.06GodThori have problem to compile pwlib from inaccess ,any other solution?
14:27.31`SauronYEah, don't use h.323? :)
14:27.54GodThor:)))))))))
14:27.55murangdbjohnson: cool thanks
14:28.21GodThorfor quintum any other protocols?
14:28.49murangdbjohnson: do you happen to have the URL for wiki
14:29.06Zeeekyou mlight want to read this too
14:29.07Zeeekhttp://www.voip-info.org/wiki-Asterisk+cmd+SetCIDNum
14:29.10*** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com)
14:29.44GodThorto change my panasonic 1232 , what you propose protocols in asterisk, i user 10 analog lines
14:29.53murangdZeeek: thanks
14:31.24*** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com)
14:36.04amercan I set "fromuser" in Extensions.conf file
14:37.06bjohnsonwhat's fromuser?
14:37.36ameror better set fromuser in sip.conf to fromuser=${callerIDnum}
14:37.56amersip FromUser
14:38.13Delvar~time
14:38.15jbotmethinks time is 1 dimensional, or everlasting
14:38.51Delvarwhat is jbot anyway?
14:39.44tclarkGodThor: check the cost of 10 analog plus inet data vs fractional t1/data to get get rid of the 10 analog lines then just use asterisk with a t1 interface
14:39.48Delvaran imp with too much time on his hands?
14:40.52bjohnson~jbot
14:40.53jboti heard jbot is the shipboard computer, but you may call me eddie if it helps you relax
14:41.02bjohnsonhi eddie
14:41.34Delvari see
14:42.08*** join/#asterisk Zaw (zaw@zaw.subneural.net)
14:43.28*** join/#asterisk bill522 (~bill522@182-30.201-68.swfla.rr.com)
14:43.31*** join/#asterisk Syrus_ (~pascal@tahiti.mpl.rullier.net)
14:43.56murangdbjohnson: ok I've just read the entire history of caller id
14:44.09murangdthey should really provide working example of how to implement this
14:45.02bjohnsondid you look at the  superdial macro?
14:46.40shaZwazimplicit u there ?
14:46.58murangdbjohnson: reading it now
14:47.40bjohnsonone caveat .. setting name usually doesn't work unless you go out through your own PRI (and it supports it)
14:48.02bjohnsonoften you can set CIDNUM if going out through a voip provider
14:48.06shaZwazanyone has successfully used outbound providers and local LAN phones while binding on 0.0.0.0 ?
14:48.42bjohnson?
14:48.59bjohnsonbinding what on 0.0.0.0?
14:49.05Ubuzanyone knows what happend to xvoip? are there any other forums for asterisk?
14:49.06shaZwazSIP
14:49.37bjohnsonwell what do you know .. I guess I bind to 0.0.0.0
14:50.03bjohnsonI connect to LAN but I think all my internet connections use IAX
14:50.04shaZwazbjohnson: the prob is that if I use 0.0.0.0 my outbound services dont register me
14:50.41shaZwazwell sure it works fine on IAX but not on SIP
14:50.54greendiseasedoes anyone use kphone?
14:51.02shaZwaz0.0.0.0 listens to all
14:51.45bjohnsongreendisease: better try some alternatives .. you're running out of time
14:52.06bjohnsonbtw .. most of the people here will tend to connect to FWD from * .. not softphone clients
14:52.38shaZwazthat what I am doing
14:53.05bjohnsonoops .. forgot one thing when you decided to try iaxcomm .. you need to sign up for a iax account
14:53.17shaZwazbut I dont understand why doesn't it register me when listening on 0.0.0.0
14:53.23^FenrisI have a POTS line and a VOIP line coming into my * box, in the [default] section of extensions.conf, how do I differentiate between them? (they need to be handled differently)
14:54.26murangdbjohnson: I've read the page on super macros but I don't think that will help me
14:54.38bjohnsongreendisease: follow answer 4 here http://www.freeworlddialup.com/content/view/full/1501
14:54.40tzanger^Fenris: first off, please don't use [default]
14:54.45bjohnsonmurangd: works for me
14:54.56tzangerand make your [default] section nothing more than exten => s,1,Hangup
14:55.04^Fenristzanger: alright
14:55.06tzangerit saves you from unintended operation
14:55.18tzangerthen have your POTS line come into its own [POTS] or [fxo] context
14:55.25bjohnson^Fenris: next .. get rid of that symbal from the front of your nick
14:55.31tzangerand yrou VOIP line into whatever the provider name's next
14:55.54murangdbjohnson: can you paste me a line in your extenstion.conf where you have caller id probaberly setup and visible on a PSTN phone
14:56.13bjohnsonI use the superdial macro
14:56.22bjohnsonthat is why I pointed you to it
14:56.23^Fenrisbjohnson: heh, someone else has the decarrotized ver of my nick already registered on this network
14:56.48shaZwazany SIP guru around ?
14:57.15^Fenristzanger: okay, I'll work on that, thanks
14:57.17bjohnsonthe "context" lines in zapata.conf, sip.conf, iax.conf set where incoming calls go to in extensions.conf
14:57.30*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
14:57.50PBXtechwhy am i getting these messages  == Primary D-Channel on span 2 up    PRI DEBUG shows nothing
14:58.11tzanger^Fenris: if you want them both to do the same thing hten just create a common context and include it from those other specific contextsx
14:58.29tzangerit's far far far safer than throwing everything and everyone in [default] since you are CONSCIOUSLY doing it
14:58.41tzangerPBXtech: are you getting it over and over and over?
14:58.46PBXtechyes
14:58.47tzangerit sounds like your D channel is bouncing
14:58.49bjohnson^Fenris: an overview is available on the wiki pages about extensions.conf
14:58.51tzangeryou won't find anything in the debug
14:59.03PBXtechyea, that because of SLIP?
14:59.08tzangerbecuase the PRI isn't actually sending anything since hte D channels' up and down like a bride's nightie
14:59.29murangdanyone have caller ID working
14:59.32murangdon their system
15:00.05bjohnsonyes.  I do
15:00.05PBXtechis there any way to test the prob?
15:00.15bjohnsonboth for incoming AND outgoing
15:00.22murangdheh
15:00.23*** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net)
15:00.33murangdyes I know bjohnson but what you've suggested I don't understand how to implement
15:00.53bjohnsonwhat is the url for the superdial macro?
15:01.02murangdhttp://www.voip-info.org/tiki-index.php?page=Superdial%20macro
15:01.19bjohnsonso you can successfully dial out correct?
15:01.27murangdcorrect
15:01.28*** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
15:01.28ameris there an application like setFromUser?
15:01.45*** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu)
15:01.51bjohnsonmurangd: give me a one of your working outgoing dial command lines
15:02.09miller7anyone here familiar with zapras and pppd? I need some minor help. I have compiled, installed and tested it and I need help with pppd settings
15:02.24*** join/#asterisk cc (~cc@byte.fedora)
15:02.35*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
15:02.53bjohnsonamer: what is that?
15:02.54PBXtech[tzanger]: is there any way to trouble shoot the D channel?
15:03.37amerI want to change the "from field" in SIP header to the callerID
15:04.30amerso what I will do is _X.,1,setFromUser=${callerIDNum}
15:04.37murangdbjohnson: check msg
15:04.57amerthis way actual callerID will be displayed on the called phone
15:06.10bjohnsonamer: look at the superdial macro
15:06.13bjohnsonon the wiki
15:06.16amerok
15:06.19bjohnson~docs
15:06.20jbotDocumentation 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
15:06.33bjohnsonit uses SetCIDNum()
15:08.18*** join/#asterisk yurpls (~yurplsl@65.114.15.70)
15:08.39yurplsHeloo
15:10.28*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
15:11.20murangdis macro support enabled on asterik by default?
15:11.31NormAstPBXTech: What kind of problems you having with your D-Channel
15:11.34sivanayes
15:11.49*** join/#asterisk eKo1 (~bernd@207.42.191.66)
15:12.56yurplsAnyone know what it means when I attach a trunk (analog) to a FXO port on digium card and it stays busy when connected?
15:13.41*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfndi.dialup.mindspring.com)
15:14.41*** join/#asterisk [Latre] (~latre@148.233.19.133)
15:14.43*** part/#asterisk GodThor (~ninja@212.110.95.139)
15:14.49*** part/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4118784.sympatico.ca)
15:16.25bjohnsonreasons why a person might want more than one voip provider account: 1. auto failover if one is not available (same reason why you often would like to failover to a pstn line if voip is not available) 2. even if not auto failover, having an existing account with another provider means you can manually change your outgoing quickly if you need to for some reason 3. provider1 has DID you want but the provider you use for outgoing does not 3.
15:16.49tzangerI made my own superdial macro based on Manxpower's excelent macro
15:17.20dsmousebjohnson: you were cut off at 3
15:18.35bjohnsonthat was a typo
15:18.55bjohnson4. you temporarily get a special deal or freebie
15:19.11vaewyn5. You like to screw around with your setup more than you should
15:19.16vaewyn;P
15:19.24bjohnsontzanger: I loosely based mine on his .. but posted it on the wiki
15:19.33tzanger*nod*
15:19.36murangdbjohnson: do you know any good DID providers
15:19.39dsmouse6. you like testing out other providers to make sure your service is as good as you think it is
15:19.42bjohnsonmurangd: yes
15:19.45bjohnsonhundreds
15:19.56murangdbjohnson: could you paste a few
15:20.08bjohnsonwhat country .. to what country?
15:20.22murangdUSA&Canda to Usa&Canda
15:20.30bjohnsonoutgoing only?
15:20.39tzangerthe only problem wiht IAX is that I can receive CONGESTION or CHANUNAVAIL
15:20.41murangdoutgoing/incoming
15:20.48tzangerI believe I should only receive CONGESTION if the far end TELLS me it's congested
15:20.50bjohnsonlivevoip, voipjet, teliax, iax.cc, nufone
15:20.53*** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
15:20.57bjohnsonaleph-com.net
15:20.57tzangerif it's too busy to take my call I should get CHANUNAVAIL
15:20.59murangdthanks
15:21.24murangdbjohnson: a DID is a number where someone can call you and that number can be hooked up to your VOIP service or am I mistaken?
15:21.25bjohnsonmurangd: incoming DIDs to Canada are hard to find.
15:21.32fishboy1669is there anyone that can tell me what the oej is in this exten=7001,1,Macro(stdexten,7001,SIP/oej)
15:21.33bjohnsonthat's right
15:21.38fishboy1669its doing my head in
15:21.54fishboy1669what is SIP/oej  ????????
15:21.55dsmousefishboy1669: it should match a context in sip.conf
15:21.55bjohnsonfishboy1669: it's an arg to that macro
15:22.21tzangerbjohnson: you got that right
15:22.25fishboy1669ok i understand the arg to to the macro
15:22.31tzangeriax.cc has 'em through Group but as you've seen on the list their service is spotty
15:22.40bjohnsonfishboy1669: if it's the stdexten macro I copied form an example, the number is the voicemail box and the SIP/oej goes to a SIP device
15:22.43fishboy1669but what is SIP/oej to do
15:22.56*** part/#asterisk Banter (Banter@209.119.214.81)
15:23.02fishboy1669yes its stdeexten macro
15:23.11bjohnsonSIP/oej goes to a SIP device
15:23.18dsmousefishboy1669: it'll get passed to dial later, which will look up oej in sip.conf
15:23.31fishboy1669oh
15:23.32fishboy1669mmm
15:23.34fishboy1669ey
15:23.37bjohnsonioej must be a SIP device configured in sip.conf
15:23.39fishboy1669confused
15:23.56fishboy1669the sip.conf is not mentioned in the example
15:24.03fishboy1669no wonder i couldnt figure it
15:24.04bjohnsonmake your own
15:24.10fishboy1669yup
15:24.12fishboy1669cheers guys
15:24.39bjohnsonlikely that arg is used in a dial command
15:24.53*** join/#asterisk brettnem (~brettnem@208.54.232.29)
15:25.05yurplsAnyone have the pinouts on the 400P card with FXO?  Is it 4&5 or 3&4?
15:25.07dsmousefishboy1669: well, it could be replaced by anything, eg "Zap/1" for line one off some zaptel thing, or "IAX2/guest@66.250.68.194/996" to get the conf today at 2
15:25.14fishboy1669u stars
15:25.41fishboy1669i changed it to a current sip phone extention and it seems to be worknig now
15:25.46fishboy1669thanks again guys
15:26.21rikstahas anyone got a system, where their agents can log into like an "after call work" or "lunch" area, and hop back into the queue?
15:27.35HitTophi all
15:27.53HitTopi wonder if there's any Ser user arround?
15:29.58fishboy1669im just starting with it
15:30.40HitTopfishboy1669: do u use mysql with ser?
15:30.52ariel_yurpls, are you asking about the plugs in the TDM400B card it uses the 4&5 for the active pair
15:31.48fishboy1669i have it installed with that but not used them together yet
15:31.50fishboy1669sorry
15:31.56fishboy1669i am using mysql with *
15:32.13*** join/#asterisk TrevorSHarrison (~trevorsha@24.49.36.218)
15:34.52HitTopoic.. ~
15:38.24*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)
15:38.24*** mode/#asterisk [+o anthm] by ChanServ
15:40.57*** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net)
15:41.04*** join/#asterisk slav_jb (~k@pirus.securax.be)
15:41.43*** part/#asterisk slav_jb (~k@pirus.securax.be)
15:49.22*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
15:50.53*** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-18-115-202.buff.east.verizon.net)
15:51.45SuPrSluGhello
15:53.35*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
15:57.39greendiseaseanyone know how to create a conference room with fwd?
15:57.50murangdcan someone explain to me what is Radius protocol
15:58.03greendiseaseremote authentication dial in user service
15:58.15murangdgreendisease: what is it use in relation to VOIP
15:58.34murangdauthentication method?
15:58.44greendiseasemost likely
16:00.07murangdhow can I encoperate radius with my asterik server for user authentication
16:00.45*** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net)
16:01.12ManxPowermurangd, You do a lot of hacking.
16:02.22*** part/#asterisk mcukstorm (~mcukstorm@neo.matrix-lan.net)
16:06.14*** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com)
16:06.36bprice20is anyone else having trouble with extensions.conf mysql db using realtime?
16:07.17*** join/#asterisk brazil (~cleber@200.198.105.37)
16:07.53brazilhello all
16:09.17*** join/#asterisk jsolares (~jsolares@200.30.141.85)
16:09.36jsolaresis it me, or is iaxtel really spotty
16:10.25`Sauroniaxtel blows
16:10.41jsolaresoh ok, i was beginning to think i was screwing up
16:13.12hajekdhow you handle outgoing faxes with asterisk? Getting a card with analog ports?
16:13.26ManxPowerhajekd, yes.
16:13.32*** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com)
16:14.06hajekdManxPower - any recommendation on good card?
16:14.56ManxPowerYou mean like the TDM400P with FXS and FXO interfaces?
16:16.30hajekdif that one is working with fax then yes ;)
16:16.49PatrickDKheh, fax is a different story
16:17.04greendiseasehmm, kphone doesnt have a number pad
16:17.08PatrickDKthey all work with fax, but you need software to understand fax
16:17.45hajekdI can receive fax with asterisk.
16:18.03PTG123hey is their a way to know in extensions.conf which sip account is dialing, and use the first character in it in the dial string?
16:18.06hajekdBut now what to connect a fax machine to an analog port of asterisk...
16:18.11hajekds/what/want
16:18.33*** join/#asterisk Uajal (~icechat5@ool-182e86f3.dyn.optonline.net)
16:20.36ManxPowerhajekd, Then install an analog port in Asterisk
16:20.44yurplsAnyone have a TDM400P and a couple of minutes?
16:21.33UajalI read documentation. Still I didn't understand idea of extensions. I have asterisk installed. And now I configure it for broadvoice. There in register command I should set extension. Have no Idea. I have SIP phone. I now its IP address. Can be this sip phone be mapped to extension?
16:22.05*** part/#asterisk djin (~marius@62.58.40.196)
16:24.27*** join/#asterisk klicTel (~Claude@207.107.208.137)
16:24.34klicTelhi all
16:25.44*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
16:25.53HitTopi want to ask about sliming asterisk
16:25.58PBXtechwhat is the zttest tool used for
16:26.17ManxPowerPBXtech, checking interrupt latency
16:26.25ManxPowerwell at least for zaptel cards
16:26.48*** join/#asterisk clint_ (~clint@snap.helixsystems.com)
16:26.55HitTopif i only use sip and iax chan, can i remove all other chan*.so?
16:26.59klicTelis anyone aware of issues connecting Cisco call manager to * using SIP?
16:27.12PBXtechso has nothing to do with the bandwith, speed of the card? to help trouble shoot echo?
16:27.35clint_Does anyone here understand the difference between busy and congestion in asterisk?
16:27.43HitTopif i only use ulaw, can i rm all other codec_*.so?
16:28.17HitTopbecause right now, codec_ilbc.so and codec_lpc10.so seems to be running very slowly
16:29.11PatrickDKilbc is always slow
16:29.19HitTopwat is it for?
16:29.21PatrickDKyou can rm any you want
16:29.29PatrickDKilbc has error correction and stuff in it
16:29.35PatrickDKit goes slow, but handles packet loss
16:29.59HitTopbut if i've set every channel to use just one codec, asterisk will bridge calls for me right?
16:30.07brazilcan you help me about QOS? I using an appliance that having HTB and SFQ with Asterisk.. My question is if HTB and SFQ can used to QoS with Asterisk?
16:31.33HitTop(because right now, im trying to install a sip router to a linksys, but SER is just too complicated for me.. i'd just stick with asterisk, and try to slim it)
16:31.55PatrickDKif you just want bridging, ser would be better
16:32.33*** join/#asterisk easydone (~notdone@eksel.demon.nl)
16:32.41HitTopi guess i'd have to bridge for calls, because the performance for the router isn't high~
16:32.43JerJerbridging and ser
16:32.46JerJermmkay
16:32.54JerJerdon't you mean proxy?
16:33.17PatrickDKI think he just want to let one phone call another phone
16:33.21PatrickDKnothing else interesting
16:33.28HitTopright patrickdk
16:33.37brazilanybody had an answer?
16:33.47fishboy1669hi
16:33.54fishboy1669anyone know what im doing wrong
16:34.03JerJerso a SIP proxy
16:34.25fishboy1669i have setup a main menu with exten => s,background,welcome
16:34.28fishboy1669but it dont work
16:34.36JerJeru need a priority in there
16:34.40*** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk)
16:34.47fishboy1669sorry thats in there
16:34.53*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
16:34.54fishboy1669s,1,back ....
16:35.03*** join/#asterisk plappy (~asdf@64.56.147.94)
16:35.06fishboy1669its getting something to point at it thats the issue
16:35.27fishboy1669i have sip ext 117 context=main-menu but it dont do anything
16:35.30Weezeycan someone telnet to 206.210.111.28 port 4800
16:36.17WeezeyI just need to know if it connects (not refused)
16:36.24vagwinit connects
16:36.27HitTopit connects
16:36.40fishboy1669IT connects people
16:36.43HitTopConnected to flowers.loit.ca (206.210.111.28).
16:36.50Weezeycool, thanks.
16:37.03vagwinwow. ddostastic.
16:37.14vagwin:P
16:37.31*** join/#asterisk stepcut (~redlion@ip68-107-21-88.sd.sd.cox.net)
16:37.32fishboy1669jerjer any idea how i set something up to point at my exten => s,1,back .....
16:38.07fishboy1669when i dial 117 it is still looking for 117 context in the extentions
16:38.10fishboy1669.conf
16:39.09stepcutHMI am trying to decide whether to get a toll-free number from iax.cc or teliax, any one want to share their experiences?
16:39.12HitTopJerJer: do u use SER? Im reading SER howto, but I think the documentation teaches base on environment of ser with mysql, but I just have SER by itself.  Could you guild me to any website or guildline just to start SER without mysql?
16:39.33Weezeyfish:   exten => t,1,Goto(s,1)  ?
16:40.07fishboy1669weez whats that do?
16:40.16fishboy1669t ?
16:41.12fishboy1669i have exten => t,1,goto,0|1 at the end of the main menu context
16:41.49HitTopfish: u want sip client to dial 117 to go into ur welcome context?
16:41.54fishboy1669yes
16:42.06fishboy1669i dial 117 on sip phone
16:42.09bjohnsonfor everyone asking about how to send a call from one to another or answer a call .. this is for you
16:42.10bjohnson~docs
16:42.11jbotDocumentation 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
16:42.11HitTopfish: within ur client context, add this line
16:42.41HitTopfish: exten => 117,1,Goto(welcomeContext,s,1)
16:43.02fishboy1669aha ok
16:43.11*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
16:43.12fishboy1669mmm
16:43.27bjohnsonstepcutHM: my experience is that both can be tested with an investment of < $20 .. sign up to or both and test them.  Make notes on the wiki so the next person doesn't have to go through the same
16:44.13HitTopfish: read the asterisk tips in wiki, it helps a lot to begin with asterisk~!!
16:44.13stepcutHMbjohnson: yeah, I was thinking of using iax.cc for toll-free (no monthly fee), and teliax from a local DID (half the cost of iax) :)
16:44.16HitTop^_^
16:44.17fishboy1669so whats the exten => s,1, ........ used for
16:44.29HitTopfish: s stands for start i think
16:44.43fishboy1669yes
16:45.00fishboy1669so if its start then how do i get an incommming call to point to it
16:45.02bjohnsonstepcutHM: no problem with that .. I currently have accounts with 4 voip providers due to that type of thing
16:45.09shido6where is the call coming from fishboy1669?
16:45.11HitTopis there any SER user arround? I need some help~_~
16:45.16fishboy1669sip phone
16:45.25bjohnsonreasons why a person might want more than one voip provider account: 1. auto failover if one is not available (same reason why you often would like to failover to a pstn line if voip is not available) 2. even if not auto failover, having an existing account with another provider means you can manually change your outgoing quickly if you need to for some reason 3. provider1 has DID you want but the provider you use for outgoing does not 4. you t
16:45.25bjohnsonemporarily get a special deal or freebie
16:45.38bjohnsonsomeone should put that on the wiki
16:45.47shido6do you want EVERY sip user to contact your box and get this or just a certain group? or only from a specific endpoint?
16:45.52bjohnsonI'll stop pasting it to the chan .. I know it's annoying
16:46.21fishboy1669i got the code out of my * book but it doesnt say how to point all incomming calls to it
16:46.35fishboy1669at moment just from one phone
16:46.38oejHitTop: There is a #ser channel
16:46.46HitTopthere's no one there in that channel
16:47.13shido6fishboy1669 yes, but who do u want incoming calls to come from? one specific sip endpoint, a group  or all?
16:47.14HitTopi wonder if there's that little ppl using ser compare to asterisk?
16:47.22vaewynhmm... anyone from abptech  on here?
16:47.23bjohnsonfishboy1669: what interface do you have incoming on?  find it in the conf files and change the context to point to where you want incoming calls to go.  Put your 's' exten lines there
16:47.37SuPrSluGi have dundi working. but am having nat issues. on one end it changes the port. any way to force a cable/dsl router to not change the prot when port forwarding
16:47.41fishboy1669one specific sip endpoint
16:47.43fishboy1669for testing
16:47.46oejHitTop: So what's your question?
16:48.11shido6then in your sip.conf in the user stanza put the context with your "s" extensions there so for example
16:48.20shido6check ur pm
16:49.15fishboy1669i have pointed that sip extention context at the mainmenu but the extentions.conf dont play with it
16:49.18fishboy1669jsut cuts it off
16:49.28shido6ok
16:49.34shido6pastebin.ca your sip.conf and extensions.conf
16:52.41HitTopoej: i want to know if there's any beginner guide for ser that doesn't require mysql.  I want to set ser as a proxy, so it routes internal calls and external to asterisk server
16:53.42fishboy1669if i change the s to 117 then it works
16:53.50fishboy1669god this is frustrating
16:53.54HitTopoej: and I got using serctl, when i do serctl restart, it gives this error: "Stopping SER : No PID file found! SER problably not running"
16:53.58__Sparks_If I have multiple SIP routes, for example three for Sipgate, and one for Free World Dialup, do I need to specify different ports somwhere in sip.conf?
16:54.04clint_Busy vs Congestion, anyone?
16:54.31vaewynfast vsslow :}
16:54.34clint_How does asterisk decide whether a call gets busy or congestion?
16:54.36vaewynvs slow even
16:54.50oejhittop: There's a good admin guide on iptel.org/ser
16:55.05HjemmeRoyKclint_: the dial plan
16:55.05vaewynbusy == busy...  congestion == deep shy0t is hapenning with the phone system
16:55.06ManxPower~docs
16:55.08jbotDocumentation 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
16:55.18clint_Okay, got that part.
16:55.31clint_But how is it that when dialing a phone directly connected to asterisk box...
16:55.48clint_(asterisk should know that the station is busy, not inexplicably unavailable...)
16:55.54clint_that congestion is the normal response?
16:56.08clint_('m sure this is an I.O. error, but I'm stumped)
16:56.51shido6clint_, what?
16:57.03shido6some phones report a busy
16:57.07shido6when in use
16:57.09shido6or DND
16:57.19shido6are u talking about PRI's?
16:57.32clint_shido6: config: two phones on channel bank, on asterisk.
16:57.36HitTopls
16:57.45shido6through a T1 interface on asterisk
16:57.46shido6?
16:58.02clint_shido6: take #1 offhook, dial it from #2, we get congestion by default, not busy.
16:58.11shido6or is asterisk running on your channel bank?
16:58.20shido6congestion?
16:58.21shido6ok
16:58.24shido6whats in ur dialplan
16:58.26clint_shido6: asterisk box -> channel bank -> phones
16:58.28shido6what does the CLI error say
16:58.28shido6?
16:58.36shido6pastebin.ca your extensions.conf
16:58.41shido6and we'll help ya out
16:59.03*** join/#asterisk Christopher1 (KRS1@68-233-58-6.atlsfl.adelphia.net)
16:59.10clint_exten -> 1,1,Dial(Zap/1), exten-> 1,2,Busy(); likewise for exten2.
16:59.15__Sparks_Hello! - Do i need multiple port = lines for each SIP account in my sip.conf file?
16:59.15Christopher1Hello.
17:00.01clint_(standby)
17:00.27shido6err
17:00.33shido6noooo
17:01.37Zeeekso wasim... and so it goes
17:01.40SuPrSluGanyone else have problems w/ asterisk.xvoip.com . i get a blank page. no errors.
17:02.28*** join/#asterisk zoa (zoa@82.103.76.147)
17:04.26*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
17:05.50Christopher1what digium hardware would you recommend to use asterisk with 4 analog phone lines?
17:06.18ZeeekTDM400 with 4FXO
17:06.22PatrickDKonly 4, I would say tdm400
17:06.39PatrickDKthough, I do like sipura 2000 for that too myself
17:06.42Christopher1thanx
17:07.01Christopher1asterisk is l33t
17:10.14hajekdIs there are EU store for digium cards?
17:10.50loudcountry ?
17:10.55hajekdCzech Republic
17:10.57loudi know there's one in france
17:11.03zoain belgium too
17:11.08*** join/#asterisk mutilator (~animenodv@65.111.201.79)
17:11.20mutilatormpg321 won't work instead will it?
17:11.24hajekdURLs?
17:11.29loudwhat a beautiful country Czech Republic, want the french l ink ?
17:11.38hajekdloud: yes, please
17:11.47Zeeekeikonex.com ?
17:11.48*** join/#asterisk yashax (~yasha_x@69.15.218.218)
17:11.56hajekddon't wanna pay duty when buying at US store...
17:11.59loudhttps://shop.eikonex.net/catalog/default.php
17:12.03zoahey i know that eikonex guy
17:12.29ZeeekI bought twice from eiko,nex and once from digium direct
17:12.40Igor-BZ-I have a little problem with asterisk... some one know when and where core call pvt->read in a channel?
17:13.01hajekdi bought my quadbri from Junghanns ;)
17:14.32__Sparks_Wonder if anyone can help mere here! - If I make a call via my asterisk box, using my SIP phone to PSTN Number, i get a ringing tone on the phone way before the phone the other end actually rings. is this normal, or can it be corrected, as it is annoying!
17:15.37zoaim wondering if i should make ss7 for asterisk
17:15.47brettnemyes, pleaser
17:15.49brettnem-r
17:16.01*** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com)
17:16.28ManxPowerzoa, Someone already has.
17:17.03brettnemManxPower: it doesn't work in the US
17:17.04zoai know but he is so slow :(
17:17.09zoai still dont know his price
17:17.23brettnemno A-Link capability
17:17.37hajekdI hope I need FXS when want to connect fax to 400P :)
17:17.54shido6take out the ||r or the ,r but that may just cut out a few rings
17:17.58*** join/#asterisk tedh (nobody@angry.mob.net)
17:17.59shido6__Sparks_
17:20.41zoaguess it all depends on the price the others will ask
17:20.48__Sparks_shido6, Is this in extensions.conf (Sorry I am new to this!!)
17:20.48zoaif its affordable, i wont do it
17:20.58zoaheya shido
17:21.52tedhHello. I've read this is the place to ask my digium/asterisk questions. Whats the best way to go about it other than blurting out the questions?
17:22.30zoahey kran
17:22.32zoakram
17:23.09wasimtedh: go read the wiki at www.voip-info.org
17:23.10vaewyntedh: blurt out the questions... and use pastebin.ca for large posts of configs/stuff
17:23.26outtolunctedh, make sure they are well thought out questions with enough info someone here can help
17:23.45tedhI've been through much of voip-info already. This is actually a problem I'm having with the zaptel module.
17:24.01*** join/#asterisk damnsure (~damnsure@wbs-146-171-124.telkomadsl.co.za)
17:24.21Zeeek<PROTECTED>
17:24.22tedhHere's the short version of the blurt: I am running Debian with a 2.6 kernel. Is it possible to get the zaptel module to work? I can't get it to recognize my linux source.
17:24.29Zeeekdid you check with DE land?
17:25.30tedhOr should I give up and go to 2.4 kernel. the voip-info says its possible with 2.6.
17:25.40vaewyntedh: yes it is VERY possible.. as in I do it :P I would shy away from the kernel packages and use the kernel.org tarballs
17:26.13vaewynmake sure you have a link from /usr/src/linux-26  to your source tree and voila!
17:26.16tedhCan you tell me whats specifically looking for when it says it can't find the linux source?
17:26.23tedhoh
17:26.30tedhi just have /usr/src/linux.
17:26.33vaewynsorry linux-2.6  not linux-26
17:26.36tedhmaybe thats my problem.
17:26.45vaewynhmm.. that should work also... but
17:26.48vaewyn*shrugs*
17:27.11tedhbrb, going to try that.
17:27.45yashaxhwo do I restart astersik from command line?
17:27.59outtolunci'd remove your symlink
17:28.08outtoluncthen use make clean; make linux26; make install
17:28.32*** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-18-115-202.buff.east.verizon.net)
17:28.49outtoluncit does a uname -r to find your source
17:29.08vaewynyou shouldn't have to 'make linux26'the makefile senses 26 automagically
17:29.38outtoluncwe know how 'magical' the makefile can get
17:30.06tedhNo luck vaewyn. It appears to be looking for /lib/modules/2.6.8-1-386/build on my system. Which isn't there after running the 'make dep' that the instructions instructed me to
17:30.20tedhAgain, this is still using the debian kernel package.
17:30.22__Sparks_okay, now I have another problem with Sipgate - If I call a PSTN number via my Asterisk box then hang up, the phone the other end rings for about 5 seconds longer - is this somthing I can fix!
17:30.27*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
17:30.51damnsurehi there, i'm playing with * for 2 days now and i got 2 voip phones running but i struggel with isdn on fc3, the card itself connects to the internet and i can go surfing but while dialing a number allways lacking dialtone appears?!
17:30.54*** join/#asterisk keith778 (~kobrien@ool-4355f47e.dyn.optonline.net)
17:31.17keith778Hello.  Is anyone here running a Digium T400P card in a Dell 1750 server?
17:31.26yashaxguys, how do I restart astersik from command line?
17:31.28keith778I am having trouble finding a way to power the card
17:31.31damnsurereload
17:31.53*** join/#asterisk PakiPenguin (~uppal@202.176.230.225)
17:31.56*** join/#asterisk jobi (~jobi@lauga.ssvl.kth.se)
17:32.18tedhyashax: you can do an "asterisk -r" and then 'reload'
17:32.23outtoluncpower the card?  sure you aren't talking about the TDM400P
17:32.38keith778yes, sorry the TDM400P
17:32.45outtolunc(note there is a 4 port t1 card that is a T400P)
17:32.52yashaxthank you
17:32.53vaewynyashax:   asterisk -rx 'reload'
17:32.56PakiPenguinhello everyone , how can i have an extension number for my iax client? , i have this exten => 1075,1,Dial(IAX2/meow@meow/1075) , but it always give the userbusy
17:33.16keith778Has anyone figured out a way to power the tdm400p in a dell server?
17:33.21damnsureanyone any idea what "lacking dialtone" means
17:33.22Sedoroxremove the /1075
17:33.34shido6ZzZzzz
17:33.49Sedoroxthe iaxy doesn't have knowledge of the exten.. so just call it.. not pass a extention
17:33.49shido6err
17:33.56shido6u dont even need @meow , really
17:33.59johnnybIs there a way to get the TDM400P to pick up the phone earlier?  It seems to wait two rings until it actually decides to pick up the phone.
17:34.08PakiPenguinSedorox: its a soft iax client
17:34.18vaewynjohnnyb: only if you don't want callerid
17:34.19shido6exten => 1075,1,Dial(IAX2/meow|20|r)
17:34.26SedoroxPakiPenguin: shido6 is currect too... and you still don't need that
17:34.29johnnybvaewyn: I don't.
17:34.32outtolunckeith778: there is a TDM400P install doc on digium.com
17:34.33shido6exten => 1075,2,Voicemail,u1075
17:34.37Sedoroxthe only time you need the pass the extention is when your doing to another server
17:34.42shido6exten => 1075,103,Voicemail,b1075
17:34.45Sedoroxfollow shido6
17:34.50outtolunchttp://www.digium.com/downloads/tdm_inst.pdf
17:34.53PakiPenguingot it
17:34.53klicTelis anyone aware of issues connecting Cisco call manager to * using SIP?
17:34.56Sedoroxis it 103 or 102 for busy...?
17:35.03johnnybvaewyn: how do I get it to answer sooner?
17:35.04vaewynjohnnyb: then set immediate=yes
17:35.08johnnybThanks!
17:35.12vaewynno prob
17:35.16PakiPenguinthanks
17:35.30fishboy1669night
17:35.31*** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org)
17:35.31shido6klicTel, good luck! u can do it
17:35.40vaewynjohnnyb: that is in the zapata.conf btw... forgot that :}
17:35.40*** part/#asterisk fishboy1669 (proxyuser@62.69.81.129)
17:35.42shido6callmanager needs the sip load tho
17:35.58klicTelshido6: does it take luck?
17:36.15keith778outtolunc: yes, I have read that but it doen't talk about how to power the card in a dell server.  As far as I can tell there is no 4 pin power on dell servers
17:36.25keith778unless there is some adapter that I need
17:36.52vaewynkeith778: got to be 4 pin unless you are running diskless or SATA drives
17:37.12PakiPenguin<PROTECTED>
17:37.12PakiPenguin<PROTECTED>
17:37.21jobiI'm using asterisk as a GW from SIP to the PSTN, and would like the GW only to accept SIP calls coming from my SIP proxy (SER)
17:37.25keith778Nope.  The server is hot swap scsi so there is no 4pin power on the hd
17:38.13jobican I have asterisk check for the source IP address of the incoming SIP calls?
17:38.25jobior have SER authenticate itself somehow
17:38.31vaewynkeith778: bummer then... I think you are SOL
17:38.35damnsureanyone any idea what "lacking dialtone" means and how i can fix it, with i4l and a W6692 based card...
17:38.46keith778yeah, thats what I was afraid of
17:38.47Weezeyis there any way to detect a fax machine?
17:38.52shido6keith778
17:38.53shido6what?
17:39.10brettnemwiki!
17:39.25outtolunckeith778: i'll trade you a 600sc for your 1750 <G> {ducks}
17:39.36damnsurei searched the wiki but nothing helped...
17:39.53damnsurei'm struggeling for a day now...
17:40.14keith778shido6: yeah, hot swap scsi drive have a high density power adapter that is wired directly to the motherboard.  Since the power supplies are also hot swap all
17:40.33keith778of the power connectors are proprietary
17:41.15keith778I was hoping someone hacked together a special cable to pull the power off of these connectors.   Seems like I am the one that will be making the cable ;)
17:42.32SuPrSluGanyone using nagios w/ *
17:42.42UajalMy asterisk demo works with broadvoice. Congratulation to me!
17:42.59*** join/#asterisk amer (desikukar@210.56.9.213)
17:43.07brettnemSuPrSluG: I setup asterisk and nagios.. hate the nagios setup..
17:43.20brettnemactually, I prefered argus
17:43.49shido6nagios is a pain
17:43.52SuPrSluGpia for certain. do u use it to call u if * goes down?
17:43.56Uajalyeah, I remember nagios setup. 2 days. But it worked.
17:44.02shido6but so is getting out of bed
17:44.03brettnemcall you with what? heh
17:44.06*** join/#asterisk _Raptor_ (~RaptorBlu@p5480548A.dip.t-dialin.net)
17:44.07_Raptor_hi#
17:44.23yashaxwhich SIP softphone would you recommend to use for asterisk testing....?
17:44.23brettnemI'm using argus and midas.. midas is a nice setup
17:44.28__Sparks_Ok, I seem to be able to use the "Flash" button on my Grandstream BudgeTone to have two calls on the go at the same time. - Is it possible to conferance these two calls?
17:44.32ameris there anyway I can set SIP "fromuser" in extensions.conf?
17:44.33shido6xlite
17:44.33brettnemnice xmls and all
17:44.37SuPrSluGbrettnem:never tried argus.
17:44.44brettnemit's nice
17:44.48brettnemmidas is too
17:45.00SuPrSluGbrettnem:is less bulky
17:45.11brettnemI'm less bulky?
17:45.15brettnemhmm.. thanks
17:45.21Uajalhow to build autodialer from asterisk? can it be done through config files or I should use AGI?
17:45.48yashaxshido6: thank you
17:45.59_Raptor_i have a problem with speex codec: i have compiled speex and asterisk and i can load the modules codec_speex but when i add the line: codec=speex to oh323.conf then i get:
17:46.00_Raptor_<PROTECTED>
17:46.17*** join/#asterisk garyitcom (~Tech@119-114.8-67.tampabay.rr.com)
17:46.21damnsureFeb 17 19:45:22 WARNING[6900]: chan_modem_i4l.c:374 i4l_read: Device '/dev/ttyI1' lacking dialtone    -- Hungup 'Modem[i4l]/ttyI1' what does that mean, couldn't find anything at voip-info
17:46.25PTG123is their a way to force asterisk to transcode all calls, instead of doing a native bridge?
17:47.24SuPrSluGbrettnem:apparently argus is a landfill monitor. argus.org
17:47.42brettnemhaha
17:47.42brettnemI have a landfill I guess
17:48.03SuPrSluGme too. toxic pc parts everywhere
17:48.52brettnemSuPrSluG: http://www.voip-info.org/wiki-Asterisk+monitoring
17:49.10SuPrSluGbrettnem:i'm there thanx
17:49.21PTG123its anyone in here based in california?
17:49.25brettnemchck out midas too..
17:50.37Jlau515hi, anybody got time to help me troubleshoot my dialplan when dialing a zaptel channel?
17:50.46_Raptor_has anyone an idea concerning my speex problem or a link to read about it?
17:51.20Weezeymy PSTN comes in via a FXO on a SPA-3000, goes out the same way, how do I create a zapata channel for that?
17:51.29Jlau515internal sip to sip calling works, when dialing a zaptel channel it fails
17:51.42Jlau515not sure how to troubleshoot zaptel issues
17:52.24ameris there anyway I can set fromuser=callerIDNUM? (sip fromUser)
17:53.34yashaxis there a quick doc that tells me how to set the Xlite with asterisk?
17:53.41Jlau515my extensions.conf can be found here, http://pastebin.ca/6025
17:54.06ameryashax: its just another SIP phone
17:54.12amervery easy to setup
17:54.13*** join/#asterisk Uther_P (~uther_p@66.180.120.83)
17:54.16Weezeyamer: http://www.voip-info.org/wiki-Asterisk+cmd+SETCIDNUM
17:54.29SuPrSluGbrettnem:can u monitor dundi w/ argus?
17:54.49amerthanks weezey, this isn't what I want
17:54.56yashaxamer: can you please tell me how?
17:55.14Weezeyamer: sorry
17:55.15amercreate a sip user in asterisk
17:55.25amerWeezey: np
17:55.28Uther_Pyou guys know where to point me for information about a really bad and a really LOUD echo problem going from SIP to ZAP.   (sometimes I can hear myself VERY loud, and the other party not at all... and sometimes its fine)
17:55.56junky[work]Uther_P: ya've echocancelwhenbridged=yes ?
17:56.07Uther_Pahh, lemmie check
17:56.10Zeeekamer you want many users dynamically possible?
17:56.22yashaxamer: you mean the extension?
17:56.38amerZeeek: yes
17:56.48amerI am running asterisk in proxy mode
17:56.51ZeeekI don't think you can do it
17:56.58amer:(
17:57.15yashaxk... created the extension, now...?
17:57.16junky[work]<PROTECTED>
17:57.20Zeeekbut what do I know?
17:57.22junky[work]yashax: reload :)
17:57.34damnsureanyone experience with * on fedora core 3 and isdn?
17:57.40Uther_Pjunky[work]: thanks... but I don't find any documetation in the wiki :/
17:58.06*** join/#asterisk Secretive (~polarisx@c-67-161-5-149.client.comcast.net)
17:58.08SecretiveHey
17:58.10ameranother IAX or SIP proxy fwds all calls to my asterisk but when asterisk fwds the call its sets the FROM field to asterisk@10.0.0.1
17:58.31ameryashax: did u create a sip user first?
17:58.38junky[work]uther
17:58.38junky[work]echocancel=yes
17:58.39junky[work]echocancelwhenbridged=no
17:58.43junky[work]both both at yes
17:58.54junky[work]and ya'll have to restart (not reload) ur *
17:58.58junky[work]try and gimme feedbacks
17:59.04Uther_Pok, thanks
17:59.14junky[work]amer: SetCallerID
17:59.17yashaxamer: sip user = extension?
17:59.18SecretiveCan anyone make sense of this situation: I can dial out with my VOIP Phones, but when I try to call between phones...the phone will ring (sometimes) but when I answer it, neither party can hear anything.
17:59.27Uther_PI'll try it.... but I was just mentioning that I can't seem to find the documentation for that option in the wikui
17:59.29Zeeekamer you have no luck, we all wanna set your callerid!
17:59.34Uther_Ps/kui$/ki/
17:59.34junky[work]FROM field? ya mean the source?
17:59.36amerhahahaha
17:59.42*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
17:59.47amerits not the callerID
18:00.00Zeeekamer you do C programming?
18:00.01amersip:fromuser is different
18:00.14amera lil bit, I can hack code
18:00.20Jlau515can anybody help me with my dialplan?
18:00.26ameryashax: yes
18:00.29Zeeekamer prolly your only answer
18:00.29HitTopwat is it for for chan_agent.so?
18:00.35yashaxamer: done.... next?
18:00.39junky[work]describe ur FROM field exactly?
18:00.43amernow create an entry in extensions.conf
18:00.51Zeeekmaybe write a tiny app to manipulate that data - if you can find it :)
18:00.56tzangergrrr
18:01.00tzangermy tr08 channel bank won't ring
18:01.05tzangerit worked before
18:01.51HitTopwhere can i find the description for each modules ?
18:01.51Uther_Pjunky[work]:  I have echocancel and echocancelwhenbridged already both set to yes
18:02.08junky[work]which version?
18:02.13stepcutHMhrm, do I have to edit the source code to change the text of the email that is sent when you get voicemail ?
18:02.23junky[work]its only from sip to zap? zap to sip ? or both?
18:02.27jsolaresstepcutHM: no
18:02.36Uther_PI think both
18:02.41SecretiveAnyone know what this means: Feb 17 12:07:14 WARNING[26440]: chan_sip.c:728 retrans_pkt: Maximum retries exceeded on call 40364e1a7270f78d7bdf16ed4ef6514e@66.55.69.242 for seqno 102 (Critical Request)
18:02.41SecretiveFeb 17 12:07:14 WARNING[26440]: chan_sip.c:1168 find_peer: Looking for SIP: 1.301
18:03.06Uther_Pit doesn't happen sip to sip, I'm sure of that much... and I've bridged a call zap to zap, and I didn't notice there
18:03.09jsolaresstepcutHM: serveremail= in voicemail.conf
18:03.17junky[work]try it and msg me after.
18:03.38vaewynanyone from voipsupply on here?
18:03.44stepcutHMjsolares: thanks, I see it now :)
18:03.45Uther_PSecretive: I've been getting those messages for a while now... noone could tell me what they ment either... I just ignoret hem
18:03.47ZeeekSecretive it means it ain't finding to phone
18:04.10*** part/#asterisk tedh (nobody@angry.mob.net)
18:04.12Uther_Pjunky[work]: its inconsistent though
18:04.51damnsureAnyone knows what this means:  WARNING[6900]: chan_modem_i4l.c:374 i4l_read: Device '/dev/ttyI1' lacking dialtone
18:04.52damnsure<PROTECTED>
18:04.53SecretiveZeeek: Why wouldn't it be able to find the phone
18:05.56Zeeektry an experiment. Turn off or unplusg a sip phone that's registered while watching CLI
18:07.02SecretiveZeeek, I have 3 phones setup all on the same network behind a firewall. Could this be the problem?
18:07.30Zeeekwhere is asterisk? Behind he same fw?
18:07.43SecretiveNope, on a dedicated server at a remote location.
18:08.01Zeeekwhat SIP ports in the phones?
18:08.11Zeeekhow are they forwarded?
18:08.41SecretiveZeeek: I'm not quite sure I understand you're question? They go through a Linksys Router.
18:08.52Secretive*your
18:09.00Zeeekand are ports being forwarded to the phones?
18:09.28vaewynanyone have the polycom firmware and bootrom images? @#$@#$ polycom won't let end users download them
18:10.22vaewynand evidently voipsupply doesn't hand them out as the VAR either
18:11.08HitTopwhen asterisk starts, it will load everything under /usr/lib/asterisk/modules (except those that are listed noload in modules.conf) right?
18:11.29HitTopso.. including those app_*.so and format_*.so etc?
18:12.42junky[work]HitTop: right
18:13.10Jlau515having issue with sip to zap configuration please help
18:13.38HitTopis there any place where I can find wat each modules is for? (other than in the actual source code, because the descritpion is just a name in the source)
18:15.09junky[work]HitTop: make the doc.
18:15.10hardwireanybody have a snom 220?
18:15.20hardwirewhy when I put a call on hold.. does it give me dialtone :)
18:15.27ManxPowerUgh!  The "Stealth Asterisk Install" was anything but "stealth"
18:15.52ManxPowerNobody bother to tell us that there were other customers on the CT1, nor that they were running data, nor that they had multiple hunt groups.
18:17.44HitTopJunky: im now trying to make asterisk small as possible so that i will run well in a linksys router
18:17.52vaewynManxPower: I think that qualifies as "ooops!"
18:17.54*** join/#asterisk nwhit (~chatzilla@65.107.59.67.ptr.us.xo.net)
18:18.53nwhithelp....  I loaded the currect cvs and now the voicemail system is putting a B in front of all the passwords, and not accepting them.  Any suggestions?
18:19.34ManxPowervaewyn, Yeah, nut we designed the setup so NO changes were required on the CT1, the channel bank, or the corporate PBX.  So to back out we just had to patch the t-1 cables back to where they were and everything was back the way it was.
18:19.46amerhey manxPower
18:19.54*** join/#asterisk ionix (~ioni@66.38.219.151)
18:19.55ionixHey
18:20.21ionixanyone has a way for asterisk to pickup the name of the caller from the ANI ?
18:20.27ionixlike RBOC database or something
18:20.40ManxPowerionix, um, ANI does not provide NAME.
18:20.44ionixI know
18:20.47ManxPowerCLID provides name and that works just fine.
18:20.48ionixANI provides number
18:20.58ionixbut I want to get the CLID from the ANI
18:21.04ionixtrying to figure a way
18:21.17*** join/#asterisk CybreWulf (cybre@killcybre.org)
18:21.23Uther_Pdoes anyone see a problem arising from having 4 analog POTS running over a Cat-5, each on its own twisted pair?
18:21.32HitTopI want to ask for format_*.so, they're there for converting right? during a call, if both sides are using the same codec (let say ulaw), then asterisk will bridge the call so asterisk would just pass packets over right?
18:21.40shido6think thats been done, Uther_P
18:21.46Uther_PI ask because I'm having an echo problem, and wonder if that  could be a possible cause
18:22.00Uther_PI didn't think it would be a problem, I've done it before
18:22.13Uther_Pjust reaching for the answer... but I think I know its from our provider
18:23.21Uther_Pa SIP to SIP call internally has virtually no delay and no echo... if I hook up a regular analog phone to the lines from our provider and call a cell phone, there is anywhere from 400-800ms delay :/
18:23.39Uther_PI think my dumbass provider is overselling
18:23.54*** join/#asterisk eipi (eipi@153-218-114-200.fibertel.com.ar)
18:24.15ionixwhos the dumbass provider ;)
18:24.46damnsureAnyone knows what this means:  WARNING[6900]: chan_modem_i4l.c:374 i4l_read: Device '/dev/ttyI1' lacking dialtone
18:25.04amerthere is no dialtone
18:25.06nwhitI loaded the currect cvs and now the voicemail system is putting a B in front of all the passwords, and not accepting the passwords.  Any suggestions?
18:25.28damnsurethx :) any idea how to fix it?
18:25.29ManxPowernwhit, Don't use the developement version if Asterisk.  That's my suggestion.
18:25.51damnsurei'm struggeling for 2 days on this, can't find anything
18:25.55ManxPowerdamnsure, almost nobody uses i4l with Asterisk.  They use CAPI or ZapBRI
18:26.21damnsureso i should stick to capi
18:26.27__Sparks_IOf I have more than one SIP Provider registering in sip.conf, i seem to be getting errors like "chan_sip.c:6801 handle_response: Failed to authenticate on REGISTER to '<sip:ACCOUNT@fwd.pulver.com> - Do I need to specify ports?
18:26.36nwhitmanxpower: ok... not the answer i was looking for
18:27.30*** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca)
18:27.35SecretiveOkay guys, here's the problem. I can call extensions remotely through Asterisk and my Hard-phones. But if I try to call a hard phone that is on the same network as me, it doesn't work.
18:28.25SecretiveThey all dial out just fine, and recieve calls from remote networks..
18:28.39Secretive...but if I try to call a phone that is on the same network (behind the same router) as me....it just doesn't work
18:28.41*** join/#asterisk santiago (~santiago@201.245.167.88)
18:28.44Tall-guyGents, I have a need for 8 pstn's and 16 sets......I'm thinking T1 to a channel bank of fxo's and fxs's....whats the best channel bank recommendation?
18:28.55SecretiveJust get this: Feb 17 12:31:05 WARNING[26440]: chan_sip.c:1168 find_peer: Looking for SIP: 1.303
18:28.55Secretive-- Executing Dial("SIP/1.303-cb17", "SIP/1.301") in new stack
18:28.55SecretiveFeb 17 12:31:05 WARNING[26440]: chan_sip.c:1168 find_peer: Looking for SIP: 1.301
18:28.55Secretive-- Called 1.301
18:30.23ManxPowerSecretive, Do you have a [1.301] section in sip.conf?
18:31.56shido6Secretive do you have peers in  your sip.conf or friends?
18:32.11*** join/#asterisk techie (gus@asterisk.horizonte.us)
18:33.05shido6answer Manxpowers ? first :)
18:33.10*** join/#asterisk paulc (~paulc@S010600062586a0b4.vc.shawcable.net)
18:33.15ZeeekSecretive is being ... secretive
18:33.23Zeeekhe won't say!
18:33.41Secretiveshido6: I have friends.
18:33.44amercan I use Variables in sip.conf? like callerIDNum
18:33.49SecretiveManx: Yes, 1.301 is in my sip.conf ;-)
18:33.52Zeeekfriends good.
18:33.56terrapenThose damned blue-collar tweekers
18:34.07SecretiveThese are PolyCom Soundpoint IP 600's
18:35.30SecretiveIs this problem perhaps because of having multiple phones behind the same Linksys router?
18:37.10Tall-guyphone lines??
18:37.49doughecka_try allowing reinvites
18:38.09Tall-guysorry, mis read there....
18:39.26bjohnsonSecretive: the simple fact of having multiple phones on a LAN would not cause you trouble
18:39.57Tall-guyno group hugs, but group smaks are plentiful :)
18:40.25bjohnsonSecretive: did you already do sip show peers?
18:40.36*** join/#asterisk benno2 (~benno2@host40-116.pool80117.interbusiness.it)
18:41.37*** join/#asterisk buddah (~hnic@208.179.86.5)
18:41.41Mw3is there any iax softphone for windows?
18:41.52doughecka_~firefly
18:41.53jbothmm... firefly is http://virbiage.com/firefly/download/firefly-thirdparty.exe
18:41.56doughecka_there ya go
18:42.13Sedoroxis it better then x-lite?
18:42.18doughecka_far better
18:42.20buddahanyone able to take a look at an extensions.conf entry really quick. i'm having trouble getting an entry to show up before a different one in dialplan
18:42.29buddahits set up that way on extensions.conf
18:42.43Zeeekhttp://www.hotsip.com/sip/tutorial/cartoon/index.html
18:42.45buddahbut when i do show dialplan, its after instead of before the other line
18:43.17mikegrbbuddah: the dialplan is sorted so it doesn't stay in the order you specify
18:43.38mikegrbbuddah: if order is important put the two items in different contexts and then include those contexts in the right order
18:43.41buddahmikegrb: how can i get it to change the order, anyway to set priority?
18:43.46buddahhmm
18:43.57shido6iaxcomm
18:44.06shido6is a windows iax softphone
18:44.06Tall-guyzeek: funny
18:44.13buddahmikegrb: they are both international, but one is _0118802, and if no 8802 then its just _011
18:44.15SecretiveIt appears that the problem I'm having is multiple phones being behind the same NAT router with only one public IP.
18:44.30buddahso make like internationala have the _0118802 and include international?
18:44.43ZeeekSecretive that's what I was getting at earlier
18:44.44buddahthat way if no 8802 it goes to international with just _011
18:44.45buddah?
18:44.54NuggetNAT blows goats.
18:45.03Zeeekchange the port on your clients: set client1 to 5060, client2 to 5061 etc
18:45.07mikegrbbuddah: put both in thier own context
18:45.30Zeeekthen forward 5060 to client1, 5061 to client 2, 5062 to client3 etc
18:45.43SecretiveZeeek: Now you have me somewhat confused.
18:45.45ZeeekIt works for me in exactly that situation
18:46.02*** join/#asterisk attack_ (~attack@213-213-141-51.xdsl.is)
18:46.02ZeeekThe clients are behind NAT router, ya? Linksys
18:46.03SecretiveHow can I force my phone to connect to Asterisk using a different port? Doesn't Asterisk only listen on one port?
18:46.14SecretiveYes, a fancy little Linksys/Cisco thing
18:46.19*** join/#asterisk oej (~oej@apollo.webway.se)
18:46.21Tall-guyAre these clients on the OTHER side of the NAT router that Asterisk is on????
18:46.21Zeeekthe phone will call on 5060 but receive the return on 5061
18:46.36Zeeekbelieve me it works and astersik doesn't care
18:46.40Zeeekastersick
18:46.52ZeeekasterRisc (with reduced instriction set)
18:47.08attack_the S100I "IAXy" model from Digium, can that be used to call conventional phones from a computer?
18:47.42Zeeekyes if asterisk server is hooked to the PSTN
18:47.44attack_hmm
18:47.44*** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net)
18:47.48Tall-guynugget: yeah, I'm verbose with the shifted number keys :)
18:47.51SecretiveThe Asterisk server is at a remote location, on a pubic network.
18:47.51Zeeekbut then you won't need a computer
18:48.23ZeeekSecretive: set client2 to 5061 - set linksys to forward 5061 to the ip of the clioent. Thats' all
18:48.24attack_what i'm looking for is a solution that allows me to call, say, someone in sweden over the internet at the local rate there
18:48.27Tall-guysecretive: ah, ok, then listen to Zeeek, he's on the right track
18:48.27SecretiveZeeek: Is this what you're telling me to change: reg.1.server.1.port=""
18:48.42ZeeekI dunno what that is
18:48.50Zeeekattacke read this pls
18:48.51ZeeekStarter tutorial:
18:48.51Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
18:48.51Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
18:48.51Zeeekhttp://www.automated.it/guidetoasterisk.htm
18:48.51ZeeekTHE reference of the moment:
18:48.52Zeeekhttp://www.asteriskdocs.org
18:48.58attack_like, placing a gateway or something that transfers calls from the internet to conventional lines
18:49.01attack_yes
18:49.11attack_i've been reading the voip-wiki for like 2 hours now
18:49.18Zeeekattack_ the first and second articles above have ALL the answers to your next ten questions
18:49.27attack_aight... i'll look at it
18:49.28attack_thanks
18:49.30harryvvonlamp is pretty good.
18:49.31Zeeekforget the wiki these are articles
18:49.43attack_mkay
18:49.58Zeeekwiki is grat once you know 90% of everything :)
18:49.58harryvvI goto it once in a while :)
18:50.22harryvvzeek, you have deb and a zap running?>
18:50.23ZeeekSecretive I need the points for helping solve at least one problem today
18:50.43ZeeekSecretive - help me out - try what I say - it has been working here for a year now
18:51.02Zeeekwith Linksys WRT54g, GT BT102, X-Lite etc
18:51.06ariel_afternoon all
18:51.14Zeeekevening
18:51.44ariel_Zeeek, yes your right for your area. How is the France these days?
18:51.46ManxPowerSecretive, Asterisk may think peer/friend/users with a . in the name is a host.
18:51.55Zeeekwonderful but cold
18:52.03ManxPowerSecretive, If you do a "sip show peers" do you see 1.301?
18:52.20SecretiveManx, Yes.
18:52.28Zeeekhe has multiple SIP clients behind NAT all on 5060 - no way
18:52.59harryvvbtw, I do not have a sip phone or ata install. Wife still uses the phone hooked to the same land line. When my machine goes into ivr she hears it and it does not stop. I am asuming that when a ata or sipphone is picked up * will detect this and stop the ivr?
18:53.07ManxPowerZeeek, Any DECENT nat router will handle that JUST FINE.
18:53.25ManxPowerZeeek, Also the error message he is getting does not indicate a NAT issue.
18:53.28Zeeekthat depends on what you call a decent NAT router
18:53.39harryvvIVM
18:53.47ManxPowerZeeek, Linksys is "decent" for this discussion I think.
18:53.48*** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com)
18:53.59harryvvi have a d-link its been good.
18:54.01ZeeekI agree I like mine a lot
18:54.18harryvvWhat soho routers should be avoided?
18:54.20Zeeekbut I'm glad you limit the scope to this discussion
18:54.46Zeeekotherwise I'd have to call you in later discussions in case you trashed the same router :)
18:55.30*** join/#asterisk Gwemo (~gen@nsc66.147.62-163.newsouth.net)
18:56.49Tall-guyPix 501 makes a neat device.....has SIP fixup protocl/proxy thing too....but then again, its 500 bucks.
18:57.41vaewynwrt54g rock
18:57.43PatrickDKheh, we use the hell out of pix devices
18:57.44ZeeekI guess the reason I'm so hot on forwarding has to do with the fact that my SIP hardphone doesn't register.
18:58.26ManxPowerZeeek, That would do it.
18:58.27Zeeekso sitting behind the router, I was assuming that asterisk would call it to qualify
18:58.51Zeeekwhich it does. Hence, the 5061 for X-Lite (which does reg)
18:58.56ManxPowerZeeek, You can't ever do manual port forwarding with multiple phones behind the same NAT router all using port 5061
18:58.56Tall-guypatrickdk: have you tried "eyebeam" from xten with the pix'es?
18:59.08ManxPowerYou need to rely on the router's port number translations
18:59.27Zeeekno I use 5060, 5061, 5062
18:59.57HitTophi, i want to know how to disable logging for asterisk
19:00.14Zeeekunfortunately you missed me doing one of the alltime stupidest network manoeuvres ever earlier today
19:00.28Tall-guyzeek: 220V into 10 baseT?   :)
19:00.32harryvvzeek what was that
19:00.56HitTopbecause right now. i have this error msg: "Logger Warning: Unable to open log file '/var/log/asterisk/messages': No such file or directory"
19:01.04ZeeekI set the gateway to 192.168.2.1 to have a second LAN card on a different subnet. Then I wonderd why I couldn't ping the new gateway
19:01.04zoatouch that file
19:01.19harryvvLast fortune 500 company I worked at there sprinker guys tested it in the data center during the week and it shut down the data centers power to all 100 servers.
19:01.49ZeeekHaving set the gateway to a new ip and leaving the network interface on the old subnet was making it a big challenge to communicate
19:01.51*** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc)
19:02.00harryvv1 million dollars in transactions were lost for that day.
19:02.01harryvv:)
19:02.26Zeeekso who am I to be telling people to forward ports?
19:02.49greg_workharryvv: why are there sprinklers in the data center??
19:03.10Zeeekto make the girls blouses transparent?
19:03.25vaewynhalon kills rabbits
19:03.27vaewyn:P
19:03.28greg_workyou're supposed to use that gas stuff that doesn't hurt electronics (just kills people.. ;) )
19:03.39NivexHalon
19:03.48mikegrbhalon is illegal
19:04.02mikegrbthere is a new improved enviromentally friendly replacement
19:04.11mikegrbbut still kills humans
19:04.14mikegrbit has a number
19:04.16*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
19:04.32mikegrbI walked past the cylendars 50 times a day for 2 years but I don't remember the numbers
19:04.36mikegrbwas like 4 digits though
19:04.38greg_worki actually saw a quick spot on discovery channel or something the other day about some new one that puts fires out instantly but doesn't hurt people
19:04.58rikstawtf is halon
19:05.09greg_workit was just a promo though (like 4 seconds long), i didnt actually see the segment on it
19:05.19vaewynriksta: liquid/gas that puts out fire by eating all the oxygen in the room
19:05.21Nuggetriksta: we are not a google proxy.
19:05.34rikstalike i can be arsed to search for that?
19:05.46rikstavaewyn: cool
19:05.57moonwickI don't know how a chemical can starve a fire of oxygen without starving a human.
19:06.00Nuggetriksta: *shrug*  it's not our job to ensure that you aren't ignorant.
19:06.02harryvvActually I think thay were testing the firealarm system . I dont know how it all worked but it cut power to the data center and the UPS were to kick in still providing power and it did not. Gradually each server started powering down and everyones pagers started to go off in IT. Everyone was running to the server room
19:06.08RoyK~lart riksta
19:06.15rikstacan't you just fill the room with CO2?
19:06.26RoyKCO is more effective
19:06.31harryvvThere is halon in the data center
19:06.39RoyKnice
19:06.39moonwickriksta: and kill any humans that don't get out fast enough?
19:06.50harryvvRoyK I have worked with halon in the millitary it replaced co2
19:06.50rikstamoonwick: well how can halon not have the same effect
19:06.51mikegrbharryvv: I doubt it
19:06.52vaewynmoonwick: no worse than halon
19:07.03mikegrbharryvv: halon isn't in use any more
19:07.05moonwickvaewyn: exactly.
19:07.08harryvvyea halon can kill it replaces or eats oxygen
19:07.13rikstaa few humans aren't as important as my data :D :D
19:07.18vaewynriksta: is the same effect... halon is just easier to maintain the volume needed
19:07.30mikegrbmy last employer I was there when we had to retrofit the old halon system
19:07.33RoyKonly halon isn't adopted as oxygen by the lungs
19:07.35harryvvhalon is still use for aircraft in the millitary.
19:07.43RoyKnice
19:07.55RoyKaircraft fire goes out, all people die
19:07.55mikegrbharryvv: but you said it was in your datacenter
19:08.00Tall-guy(and over the frier in mcdonalds)
19:08.01harryvvhuge 50 gallon unit I had to wheel around.
19:08.02RoyKaircraft falls down
19:08.02mikegrbharryvv: also, military doesn't use it anymore either
19:08.06greg_workharryvv: finding out your UPS's don't work when you actually need them sucks.. you need to test every once in a while ;)
19:08.20harryvvmike, yes as of 4 years ago when I worked there thay had a halon system
19:08.28mikegrbdecember 2001
19:08.32mikegrbhalon became illegal
19:08.36harryvvgreg, well thay are tested on sundays.
19:08.57mikegrbI worked on retrofitting a military halon system to the replacement gas
19:09.02harryvvmike, you mean new sales of halon? what about brandfauthered systems?
19:09.07Nuggetfm200 is the replacement.
19:09.13riksta"While the production of Halon ceased on January 1, 1994 under the Clean Air Act, it is still legal to purchase and use recycled Halon and Halon fire extinguishers. In fact, the FAA requires all commercial aircraft to exclusively use halon."
19:09.36mikegrbharryvv: grandfathered is not alowed either
19:09.39Sedoroxhmmm
19:09.40harryvvokay
19:09.43harryvvthat sucks
19:09.50mikegrbNugget: yes!
19:09.55harryvvwhat does halon do eat oxygen?
19:09.55mikegrbNugget: that's the ticket
19:09.58Delvarnn all
19:10.05vaewynharryvv: yep... big time
19:10.08harryvvokay
19:10.24mikegrbdecember 2001 is when it was outlawed in most of europe
19:10.30mikegrbjan 94 is usa
19:10.32ariel_military uses 3 systems depending on if it was replaced or not. 1) ClorobromoMethane, 2) Halon 3) FM200/b All of which will kill you so pilots and crew ware o2 masks.
19:10.35harryvvA chemical agent that perhaps binds with oxygen rendering it uless as a oxident.
19:10.41mikegrber new halong
19:10.52*** join/#asterisk _mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
19:10.53mikegrbdunno when all halon was outlawed in us
19:10.58mikegrbI''d guess around 200
19:11.01mikegrb2000 even
19:11.06*** part/#asterisk _mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
19:11.22riksta1994
19:11.28ariel_halon is outlawed in new setups but if it's inplace it's not required to take out unless it's damaged. or used.
19:11.45*** join/#asterisk _mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
19:11.51harryvvSuch thing as foam fireextinguishers?
19:11.51mikegrbariel_: yes, as we had to take out the halon system to avoid huge fines
19:12.02mikegrbariel_: and it hadn't been used or damaged
19:12.05mikegrbharryvv: AFFF
19:12.18mikegrbharryvv: it is water with a foam additive
19:12.24harryvvI see
19:12.30mikegrbthat is what is used at airports
19:12.34Nugget'94 sounds about right.  I was installing a fire-supression system at tobias.com around then and I recall we chose fm200 because it was clear that halon wouldn't be a long-term solution.
19:12.59NuggetI think we could still buy halon then but were warned that we might never be able to recharge it
19:13.13mikegrbariel_: thay may have been in the past but around 2000/2001 it was outlawed completely including existing installations
19:13.20ariel_Nugget, yes your right. just like Freon 12.
19:13.30harryvvyup
19:13.39harryvvr13 replaced it
19:13.51ariel_mikegrb, just last year we still had in the building for the FAA halon system.
19:14.10ariel_harryvv, R1344 i think it was called to replace 12
19:14.11mikegrbdoubtful
19:14.19harryvvI need to goto a live fire extinguishing course some time :) did it in the millitary.
19:14.32ManxPowerWhat is so good about Haylon .vs. just plain nitrogen?
19:14.34mikegrbharryvv: me too
19:14.36harryvvarial, probebly called r13 for short
19:14.43mikegrbharryvv: we got to put burning planes out and such
19:14.45ManxPowerIsn't the goal to just push all the oxygen our of the room?
19:14.46mikegrbwas quite fun
19:14.49mikegrband hot
19:14.58mikegrbManxPower: yes
19:15.21mikegrbManxPower: halon and fm200 chemically "eat" the O2 as well though
19:15.36mikegrbManxPower: which is what makes them more effective then CO2
19:16.00ManxPowermikegrb, Ah!  OK.
19:16.11harryvvmanx, no the goal is to seperate the fuel fumes from the source and seperate the air from the fumes. putting a barrier between them puts the fire out. People say gasoline is flamable and I say no its not its the hydrocabons that leave it thats flamable. Thay look at me strange :)
19:16.26mikegrbindeed
19:16.31mikegrbgas is very much not flammable
19:16.40ariel_mikegrb, sorry look at this site:http://www.reliablefire.com/halon/halon.html
19:16.47ManxPowerharryvv, Like when I tell people "the web" is not "the internet"? 8-)
19:16.51vaewynever tried to light gas when it is 20 below  :}
19:16.53harryvvheheh
19:16.54ariel_Halon is still in use today.
19:16.58mikegrbharryvv: that's what the AFFF does, sits on top of oil/gas/etc
19:17.37dsmouseso
19:17.41ariel_ok lets get back to Asterisk
19:17.51dsmouseis there supposed to be a conf... cst... nm
19:18.08RoyKariel_: ass-per-risk
19:18.44Mw3when will the conf start ?
19:18.49Mw3in 40 minutes ?
19:18.55harryvvHaving been a aircraft tech was pretty interesting. Actually the millitary service. One day I was refuling my aircraft a heavylift helicopter and heard this shhhhhh and fuel supervisor said SHUT THE Fuel off! I said what? looked down and massive stream of fuel was pooring out from underneath. Fifty Gallons spilled out of it in less then one min.
19:18.56dsmouseMw3: looks like it
19:18.56ariel_RoyK, yes but like I said it's still in use. I have seen it and I know of an FAA building with it still on halon 1301
19:19.16harryvvyea
19:19.18harryvv:)
19:19.27ariel_harryvv, what service where you in?
19:19.30Mw3ok, i'm in CET so if 40 minutes left i've managed to convert thist cst time to cet
19:19.31harryvvusaf
19:19.33Mw3:)
19:19.37ariel_same here
19:19.51harryvvOFSCOM heavy lift crew chief
19:19.51dsmousewhat's cet?
19:19.56dsmouseor where?
19:19.59ariel_I was a flight Mech/Crew chief on C-130e/h
19:20.01Mw3central europen time
19:20.03harryvvohhyea
19:20.07Mw3hungary
19:20.11dsmouseah, yea
19:20.15dsmouseso like gmt+2?
19:20.23harryvvarial cool ever work on the spectere's ?
19:20.25Mw3+1 imho :)
19:20.39dsmouse+1, ok
19:21.02ariel_harryvv, I was mainly in the Grey ghost program working EC130e/h special ops.
19:21.09harryvvahh yea
19:21.15harryvvspookey :)
19:21.33ariel_I was in the PI and Kadenia for almost 8 years
19:21.47harryvvPI
19:21.49harryvv:)
19:22.34*** join/#asterisk Moc____ (~mochouina@64.235.210.66)
19:22.42zoai was in kindergarten
19:22.44zoanice to meet you
19:22.44Moc____hi all
19:23.31benno2stupid question: if I have a zaphfc card. does I need to eg use Dial(Zap/g1) in extensions.conf ? I used an i4l card where it was dial(Modem/g1..)
19:24.07RoyKstupid question, yes
19:24.20RoyKchan_zap is Dial(Zap/something
19:24.32RoyKchan_modem is Dial(Modem/something
19:25.29benno2RoyK: thks :)
19:25.36*** part/#asterisk santiago (~santiago@201.245.167.88)
19:28.48RoyKchan_h323 is Dial(H323/asdf
19:28.48RoyKetc
19:28.48RoyKad infinitum
19:28.48jero_SFLphonecan anyone help me in guessing why I'm getting one of 10 caller IDs on a TDM400 ?
19:28.48ZeeekI never could ad infinitum
19:28.48RoyKbecause of quantum
19:28.49RoyKZeeek: no?
19:28.49ZeeekI can easily ad nauseum though
19:28.49RoyK:)
19:28.49RoyKbibo ergo sum
19:28.49Zeeekwanna see?
19:28.49Zeeekcaveat emptor (esp. non-Digium clone cards)
19:28.49vaewynZeeek: I always tell people that tell me that to go ahead... I'll give them a head start :P
19:28.49RoyKstupido ergo americano est
19:28.49vaewyncarpe carp
19:28.49Zeeekprojectile vomiting has become one of those extreme sports they have magazines about
19:28.49RoyKcarpe crap
19:28.49RoyK....
19:28.52zoacarpe canem
19:28.53RoyKcarpe bibum
19:28.53vaewyncarpe smeg
19:28.53vaewyn:}
19:28.53zoano way
19:28.53Zeeekbibendous pendulum
19:28.53zoaveni vidi party bibi dormi bissi trissi parti
19:28.53RoyKbibendus?
19:28.57RoyKwtf is that?
19:28.57zoabibere = drinking
19:29.01zoaso maybe a drinker
19:29.08RoyKbibo == I drink
19:29.09Zeeekbibendum.com
19:29.19Zeeekahem Roy
19:29.28RoyKahem?
19:29.37Zeeeksorry I was ad nauseuming
19:29.41HitTopi wonder if you can limit the log file size for asterisk?
19:29.46vaewynadd nausea
19:29.59Zeeekno there's enough there already
19:30.09Zeeekshaken, not stirred
19:30.15RoyKHitTop: there is. 2 gigs
19:30.34RoyKHitTop: it doesn't, or didn't, open the log file with O_LARGEFILE
19:30.38HitTopRoyK: thx ^^ do you know where to adjust it?
19:30.45RoyKI've seen asterisk servers crash because of that
19:30.45ZeeekHitTop but I'd recommend using a cron job to check it and copy it to be compressed
19:30.47RoyKrtfs
19:31.10RoyKZeeek: you'll need to reload asterisk, perhaps restart, to close the file
19:31.17Zeeekfunny I never do
19:31.18RoyKunless the file stays open and grows on
19:31.27tzafrir_homeHitTop: isn't the standard logrotte enough?
19:31.34ZeeekI cp it then cp /dev/null to it
19:31.52RoyKsetup asterisk to log to /dev/null
19:32.00tzafrir_homeIt only checks daily, but it can check by size, if you want
19:32.05vaewynmv blah.log blah.log.0; cat /dev/null > blah.log    :}
19:32.06SecretiveZeeek, the router we have is a 10/100 8 port VPN router
19:32.21HitToptzafrir_home: i actually wants to limit the size to be smaller.. i just got a linksys router to work with asterisk.. but the log saves in ram, it will crazy as time goes by
19:32.40ZeeekSecretive but now I've been shot down by Manx who pointed out (rightly so) that your problem doesn't appear to be with NAT
19:32.55tzafrir_homeHitTop: log to syslog and use a syslog that has size limitations
19:33.11tzafrir_homeIIRC the syslog that comes with busybox has such an option
19:33.19vaewyntzafrir_home: or to another machine you don't care about :P
19:34.03vaewynok... WIP5000 review up... --now with pictures!--- http://www.wwwrogue.com/voip/WIP5000.html
19:35.14Zeeekfor $320 there should be free drinks too
19:35.22Zeeekjust to look at the pictures
19:35.29vaewynbwahaha... it really is a kickbutt phone
19:35.52*** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
19:36.18HitToptzafrir_home: syslog is an application?
19:36.21Zeeekgood think they left the chinese on the buttons - i'd be lost
19:36.30vaewynI got 3.75 hours talk yesterday on 1 charge... and it have run flawless except for a glitch after I used the web interface
19:36.38HitToptzafrir_home: sorry. im pretty new to linux
19:36.47vaewynZeeek: umm... isn't that japanese?
19:36.51miller7anyone around that can help me set up pppd server with asterisk? I have installed zapras and pppd but I'm doing something wrong with the IPs and routing so caller cannot see anything
19:37.00Zeeekyou see how rusty I am?
19:37.04vaewynhehehe
19:37.13benno2vaewyn: thanks alot for the review ! I was wondering if using non-vlan switches if roaming could be made faster. I tried two SOHO APs and win xp  with Xten softphone and it works quite well, max 1sec call outage when roaming
19:37.19HitToptzafrir_home: anyways, thx. i'd try syslog
19:37.55vaewynbenno2: I am guessing I can turn down the MAC cacheing on the switches and get the same speedup... just havn't done it yet
19:38.04Zeeekso you guys really get off walking around between AP talking on the phone? :)
19:38.13benno2vaewyn: or perhaps you can tune the values "start roaming when signal goes below x dB", please make more experiments and keep your page update. I think its helpful for many when considering purchasing the phone
19:38.22bjohnsoncan you here me now
19:38.27Zeeeknope
19:38.27bjohnsonhear
19:38.35vaewynZeeek: when I work in a building with 6 aps to provide full building coverage I kindof have to do that
19:39.05vaewynbenno2: i will be playing quite a bit so... more to come
19:39.33benno2vaewyn: they ripped off linphone ? :) time to let them GPL the phone code :)
19:39.38Zeeekonlmy because today it has become unacceptable to be "offline" for more than 20 secods
19:39.50bjohnsonvaewyn: <- calls the cute girl in accounting.  Uses "testing" as an excuse
19:39.57Zeeekprecisely
19:40.07*** join/#asterisk inspired (mikael@host-81-191-119-90.bluecom.no)
19:40.14vaewynZeeek: ohh don't worry... I ignore calls longer than that :P
19:40.18Zeeekwell, at least it's built in the 3rd world by people happy to be making $100 a year
19:40.29Luke-JrHow can I determine why my PAP2's registration is being Forbidden?
19:40.37bjohnsontry to change it
19:40.43vaewynbenno2: not sure where linphone got that chunk of code from yet so... checking on it
19:40.51Zeeekand then sold to gull^h^h^h efficient office people
19:40.52bjohnsonis it an unlocked PAP2?
19:41.15benno2vaewyn: btw are you using open APs , WEP or WPA 802.1x ?
19:41.15Luke-Jrbjohnson: yes
19:41.18_Raptor_can anyone help me finding the problem with my codecs: i have set up asterisk with meetme and oh323 but when i say codec=speex in the oh323.conf it tells me Unknown codec 'speex'. what's going wrong there?
19:41.22TrevorSHarrisonquick question: whats the Vonage sip gateway hostname?
19:41.28Luke-Jrbjohnson: * is returning Forbidden to its reg tho
19:41.29bjohnsonLuke-Jr: on same lan as *?
19:41.35vaewynZeeek: heh... we only purchased it now because we know in a year+ it will be in the 200$ range
19:41.37__Sparks_Helo (again!) I have a PSTN number provided by Sipgate pointing to my Asterisk box, is there a way to make the ringing tone the caller hears like a UK tone rather then a US one?
19:41.40vaewynbenno2: wide open
19:41.58Zeeekvaewyn - right! like everything I've ever bought in my life.
19:42.10bjohnsonTrevorSHarrison: lucifer
19:42.20ZeeekMy first laser printer cost $2500
19:42.26bjohnsonTrevorSHarrison: just kidding
19:42.32TrevorSHarrison:)
19:42.36vaewynZeeek: hehehe... well we are evaluating for a VERY large bulk buy of phones so... 300$ now versus $$$$$ later is a good trade
19:42.55Zeeekbeter yet the MX80 Epson was $600 and came without a cable!
19:43.06Zeeekcourse that was in 1870
19:43.10benno2vaewyn: thks. and stability issues went away with the current firmware ?
19:43.57Zeeekstability... muhahaha
19:43.58vaewynbenno2: havn't had a change to load the current yet... stability did come back after I reloaded factory defaults and reset up the phone....  that web interface is sketchy in my book... it really screwed it up
19:44.10*** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
19:44.17bjohnsonLuke-Jr: I hear they are similar to SPA 2000 but I don't know what might cause that other than deny and permit entries in sip.conf
19:44.24vaewynZeeek: This thing has my grandstream and snom beat 100/1 on stability
19:44.43vaewynwhich isn't saying much with grandstream :P
19:44.44Zeeekhey the GS is rock stable. As long as you NEVER update the FW!!!
19:45.12eKo1Zeeek: You can throw it like a rock...
19:45.26vaewynthe WIP5k is on par with my polycom IP500
19:45.29Zeeeknah, it's break upon impact
19:45.43Zeeekit'd
19:45.55eKo1I never commented about the impact.
19:45.58vaewynheheh... mine would make 1000s of pieces... I put lead sheeting in it so it would quit slidding on the counter
19:46.12Zeeek~eKo1cancelwhenbridged=yes
19:46.29eKo1heh
19:46.45Zeeekwell folks, have fun I'm off until the morning shift
19:47.09vaewynZeeek: also... you gotta think... 100$ of the price is just for the pretty blue LEDs so... ;P
19:47.13vaewyncya
19:47.19ZeeekI love blue LEDs
19:47.46bjohnsonPTG123: let me know when you're in
19:47.57SafThmm, i just put an offer in on 23 SP12+'s
19:48.09SafT~8us each, wonder if they will accept
19:48.10bjohnsonSP12?
19:48.18benno2do I need to assign a MSN to the zaphfc card ? with the i4l card I had the "msn=12345678" string in modem.conf . can(must) I put it in zaptel.conf (or zapata) sorry , new to zap* stuff :)
19:48.19SafTcisco/selsius thing
19:51.05SecretiveOkay guys -- so I want something a bit technical.
19:51.27RoyKcelcius?
19:52.17SecretiveI want to be able to dial into my Asterisk through a VOIP DID ... the call will be answered, I will be prompted to enter a passcode. Once the code is entered, if valid, i want to be able to enter a number to dial out to.
19:52.29bjohnsonSecretive: ooooo
19:52.58bjohnsonSecretive: go the the user authentication wiki page in the tips and tricks section
19:53.26bjohnsonyou will red about, voicemail, auth by CID, authenticate, and DISA
19:53.57bjohnsonlong live the wiki
19:54.11bjohnsonseriously .. anybody doing backups of it?
19:54.20*** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com)
19:54.30dca[laptop]hi all
19:54.55dca[laptop]anyone ever notice a delay when doing a 'reload' on asterisk?
19:55.27eKo1eh, yeah.
19:56.03dca[laptop]it's a long enough delay to make you think you need to do another reload but then, wham!, it reloads
19:56.56dca[laptop]quite around here, is everyone getting on that conf call?
19:58.33SecretiveI get no MoH and this error: Feb 17 14:03:08 NOTICE[27169]: res_musiconhold.c:472 monmp3thread: Request to schedule in the past?!?!
19:58.40ariel_dca[laptop], yes it happens allot. But if you do the reload from the manager api it's almost instant.
19:59.38zoawhen is that conference call ?
19:59.45zoais it 2pm there already?
20:00.01CMikehi all
20:00.11*** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com)
20:01.40HitTopHI~^_^
20:01.57SecretiveCan someone please help me with my Music on HOld issue: Feb 17 14:03:08 NOTICE[27169]: res_musiconhold.c:472 monmp3thread: Request to schedule in the past?!?!
20:02.16junky[work]i think its cause ya dont have enuf cpu.
20:02.42HitTopSecretive: do u have mpg321 instaleld?
20:02.46junky[work]im getting it on an old 200MMX
20:02.48bjohnsonzoa: yes
20:02.54junky[work]its mpg123
20:03.05HitTopops^^ hehe
20:03.12MeznevSecretive: I don't know anything about it, but you might want to make sure that all the devices have the correct time set.
20:03.31jero_SFLphonehi junky
20:03.37MeznevThose type of errors are usually because either the server's time is set ahead or the client's is set behind
20:04.24benno2HitTop: I got the same moh error with mpg321, is this because it has an mpg123 emulation script which does not exactly match the cmdline args and therefore * calls the fake mpg123 with the wrong args resulting in mpg321 exiting ?
20:04.29SecretiveInteresting.
20:04.33harryvvAnyone here that run debian and have modules installed? Did you get this error when doing /usr/src/asterisk make config ? look at http://pastebin.ca/6036 and tell me what you did to resolv it.
20:04.35SecretiveMy mpg123 was renamed to mpg123.old
20:04.38SecretiveSo it was mpg123
20:04.41SecretiveI renamed it and it works now.
20:04.50SecretiveVery choppy though
20:05.04HitTopwow~
20:05.14zoaomf this doesnt sound good :)
20:05.15SecretiveTHe CPU is 100% choppy though.
20:05.21RoyK~lart zoa for fun
20:05.36zoawhat is causing this ?
20:05.46zoaBRIAN
20:05.46SecretiveThe CPU is 100% idle, but it's very choppy
20:05.53zoai cant hear you guys
20:06.11HitTopSecretive: is ur cpu too slow? (sorry if im wrong)
20:06.40JerJerdo you have the PROPER version of mpg123 ?
20:06.57SecretiveP4 2.8ghz with 512k Cache and 512mb Ram
20:07.07JerJerdo you have a zaptel timing device?
20:07.09HitTopwow..@@
20:07.16SecretiveJerJer: Me, no.
20:07.26RoyKJerJer: you didn't answer that nice e-mail on the -dev list, did you?
20:07.27HitTopright.. u need to have ztdummy at least
20:07.45JerJerRoyK: ?
20:08.16*** join/#asterisk kiran (~kiran@203.212.254.27)
20:08.32RoyK:)
20:08.33HitTopis it possible to disable logging in asterisk? if so, where do i set it?
20:08.44bjohnsonlogger.conf
20:09.06SecretiveJerJer: What is the proper version?'
20:09.08*** join/#asterisk DsrtZrzmra (~DsrtZrzmr@dsl-200-67-75-232.prod-empresarial.com.mx)
20:09.28RoyK<quote>
20:09.29RoyKDon't worry Michael.
20:09.30RoyKJust ignore Jeremy. It's clear that he doesn't understand the difference
20:09.30RoyKbetween a codec and a transport protocol. (Which may go some way towards
20:09.30RoyKexplaining some of the problems chan_h323 has ;-)
20:09.30HitTopbjohnson: so if i don want logging, i write messages => ? (leaving the right side blank?
20:09.34RoyK</quote>
20:10.30Secretive[root@localhost asterisk]# mpg123 -v
20:10.30SecretiveHigh Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
20:10.30SecretiveVersion 0.59r (1999/Jun/15).
20:10.36DsrtZrzmraCan anybody help me in a simple problem in asterisk setup? I just cant "register" using kphone, but i can make calls
20:11.07*** join/#asterisk jayden (~ircatjerr@65.170.43.34)
20:11.13DsrtZrzmrai get: SIP/2.0 403 Forbidden
20:11.14kiranhi can any one help me in configuring asterisk CDrs to Mysql Cdrs
20:11.22bjohnsonHitTop: don't know .. try it
20:11.33HitTopbjohnson: ok . i'll try then tell u^^
20:12.01SecretiveHow to I turn on good debugging for asterisk cli
20:13.23MeznevAs in not ultra vague sortaerrors? :P
20:13.42SecretiveMeznev: As in I want to see when agents log in/out etc
20:13.48SecretiveAnd is there anyway to see which agents are logged in right now?
20:13.51bjohnsonDsrtZrzmra: sounds like same problem Luke-Jr has
20:14.38bjohnsonshow agents ?
20:15.30yashaxsimple linux question: ifconfig eth0 192.168.0.10 netmask 255.255.255.0 up (where do I type the gateway?)
20:15.49RoyK~lart JerJer
20:15.56*** join/#asterisk ZeroXeal (~zeroxeal@ool-44c166d7.dyn.optonline.net)
20:16.06MEPHiST0yashax: route add default gw <gw>
20:16.14PatrickDKyashax, you don't
20:16.19PatrickDKinterfaces don't have gateways
20:16.42MEPHiST0indeed
20:16.42SecretiveIs there anyway to logout agents from the cil
20:16.48yashaxMEPHiST0: default gw is parameter?
20:16.55yashaxor default
20:16.55bjohnsonno
20:16.57bjohnsontext
20:17.05bjohnson<gw> is the IP
20:17.07MEPHiST0yashax: you have to set the def-gw in the default routing table
20:17.21MEPHiST0that can be set with route add default gw <ip>
20:17.26MEPHiST0simple thing
20:17.57*** join/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com)
20:18.49SecretiveI really dont like these all over the place: Feb 17 14:15:02 WARNING[27169]: chan_sip.c:4873 check_auth: Secret is #Phone1.304:
20:19.09DsrtZrzmraIts easy, y installed asterisk in debian, added an extension (8004), then im using kphone to "register" but it get  SIP/2.0 403 Forbidden. I have host=dynamic.
20:19.12yashaxMEPHiST0: Awesome... thanks.... hwo do you set the DNS?
20:19.35MEPHiST0in the file /etc/resolv.conf
20:19.48Weezeyhow do I dial a pause?
20:19.55yashaxGOT IT!!!!!!  Thanks...
20:20.12MEPHiST0weezey: atdt nnn,nnn
20:20.15Weezeythanks
20:20.18MEPHiST0where , causes 500ms of delay
20:21.50SexyKenHow do I get rid of all of those messages like: Feb 17 14:25:57 WARNING[27169]: chan_sip.c:1168 find_peer: Looking for SIP: 1.303
20:21.50HitTopbjohnson: yes. its right, but log file must exist.. could its not a big deal
20:22.05TrevorSHarrisondoes anyone have an example vonage sip config they could paste?
20:22.07MEPHiST0touch rules
20:25.30*** join/#asterisk zotz (~zotz@24.231.32.191)
20:26.17kiranhi can any one support on installing asterisk addons?
20:27.10*** join/#asterisk buddah (~hnic@67.110.253.129)
20:27.22buddahhow do i turn on progress tones for inbound?
20:27.39buddahnot getting tones when we call any of our phones
20:27.40jero_SFLphoneusing ringing ?
20:27.48buddahyeah, ringing
20:27.56buddahneed it, but dont have it
20:28.16jero_SFLphonewhat happens if your extensions script calls Ringing() at a given time ?
20:28.20jero_SFLphoneyou should get ringtones
20:28.27buddah?
20:28.43buddahdont have Ringing() in anywhere
20:29.07jero_SFLphoneand you want to hear ringtones telling the user that the call is being established ?
20:29.09Weezeyall of my calls appear from ${CALLERIDNAME}|asterisk  instead of ${CALLERIDNAME}|${CALLERIDNUM} to my SPA-841.  Any thoughts?
20:29.11buddahyes
20:29.25buddahwell its when someone from the outside calls in
20:29.25jero_SFLphonebuddah, then try to add Ringing(). What kind of channels is it ?
20:29.27buddahthey get no ringtones
20:29.30jero_SFLphoneoh
20:29.53buddahso if i call from my cell to the phone behind sip, i hear no ringing
20:30.02buddahits just quiet then either they pick up or it does voicemail
20:30.06jero_SFLphonebuddah: at what time do they not get any ringtone? Before the line is answered, or after
20:30.14buddahbefore its answered
20:30.18jero_SFLphonebrb
20:30.31*** part/#asterisk jayden (~ircatjerr@65.170.43.34)
20:32.51_Raptor_[1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.12.2, PWlib v1.5.2
20:32.51_Raptor_Feb 17 21:31:14 ERROR[21843]: chan_oh323.c:4653 load_module: Failed to insert capability 22.
20:33.04_Raptor_can anyone tell me what this means?
20:33.21_Raptor_codec=SPEEXN8K in oh323.conf
20:33.32bjohnsonbuddah: do you answer and/or do ivr and then dial the internal extension
20:33.58buddahi have no idea what ivr is
20:34.03buddahand no
20:34.08buddahi just dial the number
20:34.09buddahthats it
20:34.11bjohnsonlooks like it specifying the speex codec .. but I don't know what N8K is
20:34.53bjohnsonbuddah: so your extensions.conf has an extension s,1,dial(<the internal phone to be called>) ?
20:35.13^Fenrisis there a way to set up asterisk to send faxes via a print queue?
20:35.31bjohnson^Fenris: I believe so
20:35.35_Raptor_bjohnson: yes i want speex, but with codec=speex it says unknown codec
20:35.41bjohnson^Fenris: the easiest answer is don't use * for faxes
20:35.53bjohnson_Raptor_: no idea .. I don't use h323
20:35.54^Fenrisbjohnson: heh
20:35.57_Raptor_so i looked into the src and found SPEEXN8K
20:36.07^Fenrisbjohnson: i have it working pretty well now for recieving them
20:36.07_Raptor_but now this happens
20:36.33^Fenrisbjohnson: if I setup a different machine to send them there could be line collisions (one picks up the phone while the other is sending)
20:37.17^Fenrisreading about Asterisk spandsp, maybe that does what I want...
20:38.43jero_SFLphonebuddah
20:39.29bjohnson^Fenris: you can buy a $20 hardware peice at radioshack to prevent a line in use from being interupted
20:40.47^Fenrisoh
20:41.30greg_work${EXTEN:1} chops of the first character, right? (read: the first character is 0, not 1, ?)
20:41.34buddahyes
20:41.43greg_workok
20:44.11jero_SFLphoneanyone tried SFLphone ?
20:45.44*** join/#asterisk r0d3nt|m (ctxwvp@perverseengineering.org)
20:46.25greg_work| doesn't have any meaning in dial patterns, does it?
20:47.47*** join/#asterisk adjacent (~scott@office.bftwave.com)
20:48.05*** join/#asterisk yurpls (~yurplsl@65.114.15.70)
20:48.49*** part/#asterisk didz_ (didz_@200.218.192.52)
20:49.01Luke-Jrbjohnson: I can register KPhone fine... just not PAP2
20:49.22brettnemhey all, I once saw a high capacilty ISDN PRI / VoIP gateway.. carrier grade hardened box.. anyone know who it may have ben?
20:49.32brettnem+e
20:49.45harryvvanyone here by chance running debian and zap? have a make config issue with asterisk that needs resolving
20:51.04*** join/#asterisk redder86 (~lee@gateway.howardsilvan.com)
20:52.21*** join/#asterisk guugmember (~nachoramo@168.234.226.39)
20:52.35DsrtZrzmraim running asterisk on debian
20:52.46Darwin35I am also
20:52.54Darwin35new build first time
20:52.59guugmemberhello guys I am making an presentation of how Asterisk can work in an Avaya environment, any pictures or suggestions?
20:53.26guugmemberI have a reunion tomorrow with the Avaya Distributor of Central America
20:53.53harryvvDsrtZrzmra and Darwin35 did both of you encounter problems when doing a /usr/src/asterisk make config and if so what did you do to resolve it. ?
20:54.35Darwin35I had no problems but I have aasterisk 1.0.5
20:54.48*** join/#asterisk Jeroen (~jeroen@084-246-048-082.PN.nl)
20:55.27*** join/#asterisk Banter (Banter@209.119.214.81)
20:55.34*** part/#asterisk Jeroen (~jeroen@084-246-048-082.PN.nl)
20:55.51terrapenhey, someone with FWD
20:55.53terrapencall 616306
20:56.03brettnemhmm WARNINGs and sometimes bridging calls gives me squeals in phone calls.. any ideas?? :-/
20:56.09terrapenfor an automated test
20:56.54DsrtZrzmrano, i used apt-get install asterisk
20:57.05brettnemany ideas???
20:57.12brettnemSIP to SIP calls
20:57.20harryvvDS, I see I did the CVS way mabey that was the problem.
20:57.51*** part/#asterisk Uther_P (~uther_p@66.180.120.83)
20:58.03harryvvDsrtZrzmra and it loads the zap modiles on a system reboot?
20:58.41Darwin35I did not I built from src
20:59.01Darwin35and ztdummy is not building
20:59.04Darwin35grrr
21:00.04SecretiveHow do I reload extensions without restarting asterisk
21:00.12Darwin35reload
21:00.32Darwin35but it will not reload till all the channels are clear
21:00.33junky[work]that makes sense huh? :)
21:00.53vaewynSecretive: extensions reload
21:00.53junky[work]no, it will reload even if channels are actives.
21:01.01yurplsAnyone have a TDM400P with FXO?
21:01.02Darwin35they changed it
21:01.07klicTelextensions reload
21:01.29SecretiveWhat is verbose mean
21:01.30johnnybI'm having trouble getting my transfer to work w/ Grandstream phones.  I've got asterisk transfers working using the "tT" option and "#" on the phone, but I wanted to do attended transfers with the phone.
21:01.51johnnybHowever, every time I try transferring it merely hangs up the line on transfer.
21:01.54*** join/#asterisk rvhi (~rv@mail.o-matrix.org)
21:02.13guugmemberwhere can I find Asterisk pictures to make a ppt presentation
21:02.25*** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com)
21:02.28yurplsPics of what of *?
21:03.00*** join/#asterisk fearnor (~alex@66.250.55.66)
21:03.28fearnor!summon atacomm
21:03.42loudwhy dont you google:// asterisk filetype:ppt ?
21:04.23fearnoratacomm has very nice onhold music
21:04.29vaewynhehehe... I always just take screenshots of code and use that as the background :}
21:04.34fearnorlike 30 minutes and didn't repeat itself
21:04.42fearnorspeaking of
21:04.56Darwin35there are no pics
21:04.57fearnoranyone else has a102u cards? ;)
21:05.10yurplswhats a a102u card?
21:05.18fearnorsangoma a102.
21:05.22yurplsT1?
21:05.26fearnoryes.
21:05.53yurplsI have one of there wan cards.  currently can't get the new digium card T1/E1 to work.
21:06.05fearnoryou probably have digiumme
21:06.16fearnori want to try sangoma
21:06.32yurplsI have a sangoma 2 port WAN card.  Worked for years.
21:06.49yurplsReason you are trying sangoma for *?
21:06.58eKo12 port WAN? What carrier system?
21:07.06BanterI'm trying to install and i get a c compiler error??
21:07.18fearnorcause digium cards are a pain in the freaking neck
21:07.27fearnorits 2005, about time cards have something more than 32 byte buffers.
21:07.35fearnortoo interrupt sensitive.
21:07.42yurplsfearnor: Thanks for telling me after I just bought two of them and I now have problems.
21:07.57fearnoryou problems are probably not the problems i'm having.
21:08.00fearnortehy are fine
21:08.21CoaxDfearnor: C'mon, 16 byte FIFOs should be ALL YOU NEED! *rotfl*
21:08.34yurplsThe 4 port T1s work great.  TDM400P and the new E1/T1 is killing me.
21:08.35fearnorcoaxd: 16450 for LIFE
21:08.42CoaxDfearnor: I wrote the first driver for the Exar 17C158 octal PCIset..
21:08.46fearnorcoax: unless you are a fan of 8250
21:08.50CoaxDfearnor: Wasn't THAT a blast..
21:09.11vaewynouttolunc: check groklaw... is a 9 day ultimatum to get their docs up to date
21:09.23^FenrisOk, I'm convinced not to use * for outgoing faxes, so my next question is, can Asterisk co-exist with Hylafax? both will be using the same modem
21:09.29CoaxDfearnor: The main coolness about that chip is that you could do 32 bit fifo reads/writes..  So, you got 4x as much work done in 1 read or write as you could with a 16550
21:09.56outtoluncaww
21:09.57fearnorfuck this noise
21:10.05CoaxDfearnor: heh. sux
21:11.11*** join/#asterisk sneak (~sneak@64.220.234.21.ptr.us.xo.net)
21:11.21yurplsfearnor: What probs u having with sangoma?
21:11.37fearnorwell, first, i gotta find someone who can guarantee they can ship it to me TODAY
21:11.46*** join/#asterisk chaoscon (~ph33r@chaoscon.user)
21:12.07Banterwill asterisk run on mandrake?
21:12.12CoaxDBanter: Um, yes.
21:12.17CoaxDBanter: If it is linux, it'll run
21:12.18Bantercool thank you
21:12.24*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
21:12.35CoaxDWelcome :)
21:12.48Darwin35unical is for uniden sip phones right
21:13.00Darwin35?
21:13.08Darwin35not seen it before
21:13.16yurplsI think sangoma sell direct. Thats where I get WAN cards.
21:13.45Darwin35asterisk runs on fbsd also
21:13.47vaewynholy cow do polycoms have a ton of crud in their XML... wowzers... you can config them to do about anything
21:14.09harryvvwell, so far now one has a answer to my question concerning why /usr/src/asterisk make config generates a missing init.asterik error for debian.
21:14.13^FenrisDarwin35: but there are no drivers for zaptel hardware for fbsd
21:14.27Darwin35yes there is
21:14.39^Fenriswhat's the name of the driver?
21:14.41Darwin35I have 4 tdm40 cards
21:14.46Darwin35zaptel
21:14.55vaewynbut bsd ain't supported so good luck
21:14.56Darwin35in /usr/ports/misc/zaptel
21:15.16Darwin35once its up its up
21:15.18vaewyn(officially supported I should say)
21:15.25Darwin35and there is a support mail list
21:15.36*** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)
21:15.45^FenrisDarwin35: hrmmm, guess I need to upgrade my fbsd box, what version are you using?
21:15.45fearnorok well, atacom isn't getting this order
21:15.53Darwin35and libpri is also there
21:16.06vaewynfearnor: whatcha buying?
21:16.13Darwin35and asterisk is in /usr/ports/net/asterisk
21:16.20Darwin35its ver 1.0.3
21:16.30Darwin35the port should be updated soon
21:16.32^Fenrisof Freebsd
21:16.47Darwin35it works on fbsd 4.10 and 5.x
21:17.14^FenrisI'm running freebsd 5.0-Release, so maybe I just need to update my ports
21:17.58^FenrisI have Asterisk configured with a modem for receiving faxes, can I use the same modem to send faxes? or does Asterisk have a lock on it
21:18.14Darwin35yes
21:18.27Darwin35you can use asterisk to fax
21:18.27MavvieDarwin35: if you sent patches for the 1.0.5, port, sobomax doesn't need to do it on his own.
21:18.49^FenrisDarwin35: so * doesn't lock the modem, so to speak
21:18.57Darwin35he needs to update the port  in the ports tree
21:19.12*** part/#asterisk djin (~djin@gridfox.xs4all.nl)
21:19.14MavvieDarwin35: if you send patches to him for the port, he doesn't need to do all the dirty work himself.
21:19.16^Fenriswas thinking of using Hylafax to send faxes
21:19.41Darwin35I have sent him patches and nothing has happened and got no responce from him
21:20.12Mavviedid you use send-pr ?
21:20.14Darwin35I use hylafax and a ext 56 k modem
21:20.31^Fenrisand the modem is also used by Asterisk?
21:20.42Darwin35I  use spandsp and the faxx add on from opencall.com
21:20.47Darwin35no
21:20.52MavvieDarwin35: did you use send-pr ?
21:20.55Darwin35I can use it to dial my office
21:21.17Darwin35Mavvie i believe so
21:21.26MavvieDarwin35: what are the PR numbers?
21:21.34Darwin35but I also emailed him directly about another issue
21:21.46Darwin35I will have to search
21:21.49*** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca)
21:21.54Darwin35I am on a diff system right npw
21:22.00Darwin35now
21:22.07Darwin35not on the dev system
21:22.32Weezeyi don't understand zapata.conf.  What does it do?  I have my stuff going out my SIP ATA, so do I need a zapata.conf?
21:22.46vaewynWeezey: nope
21:22.49wizhippohas anyone here tried * and Sphinx? what version of sphinx should i use to try the eagi test script?
21:22.49eKo1Weezey: No.
21:22.55Darwin35zapata is used by the digium cards
21:23.01JerJeryou only need zapata.conf if you have a Zaptel device
21:23.03MavvieDarwin35: what is the name you would have used in your email address?
21:23.07Darwin35for configuration
21:23.09Weezeygotcha, thanks.
21:23.14Darwin35bsdtech
21:23.21guugmembercan I call ENUM the DNS for SIP?
21:23.24Weezey(which also means, it can't fix my problems)
21:23.26Mavvieguugmember: yes
21:23.42eKo1guugmember: of course
21:24.00MavvieDarwin35: can't find any PRs from that person.
21:24.20Darwin35ok i got festival 1.95 working with asterisk
21:24.37Mavvieguugmember: within limits of course, only 1 NAPTR record per person (unless you got my patch)
21:25.09MavvieDarwin35: any difficult patches or just one new Lisp command.
21:25.32moonwickyaay zaptel
21:25.50Darwin35needs to be updated from ver 1.0.3 to 1.0.5 and I will have to find all the changes int the existing patches that have to be pulled out
21:26.10Darwin35and we need to get zaptel updated soon also
21:26.19Darwin35I use the current cvs
21:26.38Darwin35from the fbsd zaptel project
21:26.55rvhihi, in voicemail, can i customize the email header for each context?
21:27.56Weezeyaha!  How come my calls are coming from "asterisk" or an unknown number when ${CALLERIDNUM} is defined before the call gets passed to my SPA-841?    http://www.mail-archive.com/asterisk-users@lists.digium.com/msg75788.html
21:28.32greg_workis it bad to do  exten=>_123,1,Macro(...) as opposed to exten=>123,1,Macro(...) ?
21:28.48Darwin35I will work on it again tonight and get a list of patches
21:28.58Darwin35and try to pr them again
21:29.37*** part/#asterisk Banter (Banter@209.119.214.81)
21:30.03benno2I installed a hfc-s with zaphfc on * stable (fedora core 3, kernel 2.6.10). inbound calls work perfectly but outbound calls (from a budgetone via SIP account) give me a 503 error
21:30.36WeezeyI removed my SETCIDNUM command and the caller ID comes in perfectly to my set.
21:31.21benno2any idea ?
21:32.20*** part/#asterisk afrosheen (afrosheen@txprotoa8.august.net)
21:32.25*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Dev Conf 2PM CST FEB 24th -> IAX2/guest@66.250.68.194/996
21:33.20guugmemberwhat is the true value of ENUM? just not to remmeber extensions? and just remember emials for example?
21:35.47eKo1guugmember: ENUM == DNS for VoIP devices.
21:36.17harryvveko1 are you involved with the development of *
21:37.15johnnybFor incoming calls, we are using land lines.  However, for outgoing calls we wanted to use a VoIP provider.  What do you all recommend?
21:37.28eKo1Well, I discover bugs if that is considered helping.
21:37.43eKo1Haven't really bothered to get into the code.
21:37.53eKo1Although I really should.
21:40.01harryvveKo1 okay I do have a issue and it has to deal with /usr/src/asterisk make config. It cannot find a file and think that make config was setup for redhat not debian. My issue is to get the modules loaded with make config so debian can load the moduled automaticly apon a reboot then asterisk can start.
21:40.12harryvvI have to load them manually to make it work.
21:41.12eKo1eh, just add them to modules.conf
21:42.06eKo1Oh wait, I see what you mean.
21:42.13harryvvhttp://pastebin.ca/6039
21:42.24eKo1You're talking about zap modules?
21:42.27harryvvyes
21:42.34eKo1Or wcfxo or whatever they're called.
21:43.25guugmembereKo1, so what is DUNDI all about?
21:43.44harryvvwcfxs
21:43.49harryvverr
21:43.50harryvvyea
21:43.54bjohnsongreg_work: I don't think there is really a difference unless you're dealing with LARGE call volumes
21:44.22harryvvek01, do you think my issue should be brought up in #asterisk_dev?
21:46.19eKo1harryvv: http://pastebin.ca/6040 <--- That's what I do.
21:46.22*** part/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca)
21:46.54eKo1guugmember: With DUNDI, there is no central server incharge of lookups.
21:47.45harryvvek, which distro are you running?
21:48.06eKo1FC2.
21:48.20harryvvk. people in debian say /etc/modules.conf
21:48.38harryvvBut then..the info is there and uncommented and it should be loading
21:48.56harryvvek, let me get a copy of it and send the pastbin to you
21:49.05eKo1Well, you could do it that way too. Whatever tickles your pickle.
21:49.26*** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net)
21:50.21harryvvhttp://pastebin.ca/6041
21:50.27harryvvtake a look at that
21:50.49*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
21:51.34eKo1Is there an /etc/modprobe.conf in Debian?
21:52.16harryvvyes
21:52.19harryvverr
21:52.22harryvvyes
21:53.16eKo1Do you see anything with wcfxs and ztcfg?
21:54.24harryvvyes
21:54.29harryvvbut thay are uncommented
21:55.02*** join/#asterisk Sesq (~Sesq@gate.us.cyberscience.com)
21:55.35eKo1OK, get rid of the stuff in modules.conf and uncomment the stuff in modprobe.conf.
21:56.10vaewynok... wth if it is all using ulaw can I hear my polycom on SIP/IAX clients... but not on the zaptel?
21:56.25tzangerwhy are we using ulaw anyway?
21:56.33tzangeris it ot save the server?
21:56.36vaewynum... why not?
21:56.41*** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk)
21:56.42harryvvokay
21:56.42tzangerbecause it's bandwidth-wasteful
21:56.47tzangergsm is light on proc and bandwidth
21:56.49Sesqquick question... anyone know if the voicepulse connect service is allowing more than 4 simultaneous lines yet?
21:56.59eKo1harryvv: umm...did you uncomment an alias line in your modules.conf
21:57.00vaewyntzanger: phones don't all support gsm...
21:57.04vaewynor ilbc... or...
21:57.05tzangervaewyn: true enough
21:57.06bjohnsontzanger: what g standard is gsm?  g723?
21:57.09__Sparks_Hi, I am getting wuite a few "app_dial.c:749 dial_exec: Unable to create channel of type 'SIP" messages from Asterisk - is this bad!?
21:57.10tzangerbut that's what your server's for :-)
21:57.30vaewyntzanger: 2 items... gigabit ethernet... and great call clarity :P
21:57.43tzangervaewyn: yeah well I don't have a gigabit link to the internet
21:57.44tzanger:-)
21:57.48tzangerbjohnson: no idea
21:57.52tzangeroh
21:58.02tzangerwtf am I doing in here, I was supposed to be in asterisk-dev
21:58.03vaewyntzanger: this is internal to campus... off campus gets ilbc/gsm/g729
21:58.03tzangerheh
21:58.08vaewynhahaha
21:58.11junky[work]is there any reason, when im generating .call, i cant go more then 143 active channel(s) ?
21:59.03terrapenwhat do y'all think of Speex?
21:59.12tzangerhaven't used it
21:59.27vaewynterrapen: ok.. but not supported enough... and slightly crunchy on the CPU
21:59.29terrapenKeep it in zer ghetto or train to Auschwitz?
21:59.31bjohnsonis there a way to upgrade a user's groups in linux without logging out and back in again?
21:59.34tzangercrunchy
21:59.34tzangerheh
21:59.40tzangercrunchier than ilbc?
21:59.47terrapen(excuse the Ali G reference)
21:59.49vaewynyep
22:00.24terrapenso it is out of zer balloon.
22:00.50*** join/#asterisk SuperMMan (~graphic@d209-89-191-155.abhsia.telus.net)
22:00.55vaewynIf someone can get a hardware encoder out there it will be a mut point
22:01.01vaewynerr... moot
22:01.03SuperMMananyone know where i can get a copy of the psql cdr table?
22:01.16terrapeni want to find the best all-around codec.
22:01.39harryvveko1 well found it in /etc/modprobe.d/zaptel its been uncommented by default.
22:01.47eKo1SuperMMan: wiki
22:01.52_Raptor_<PROTECTED>
22:02.25SuperMManeKo1,  ok thanx
22:02.34*** join/#asterisk zotz (~zotz@24.231.32.191)
22:03.11*** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com)
22:04.49*** join/#asterisk file[laptop] (~file_lapt@mctn1-142166197096.nb.aliant.net)
22:04.59ManxPower~docs
22:05.01jbotDocumentation 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
22:05.17ManxPower_Raptor_, You need a [default] in extensions.conf
22:05.36__Sparks_Please help me here!! - I keep getting "app_dial.c:749 dial_exec: Unable to create channel of type 'SIP" " And I don't know why! - I have a Grandstream BudgeTone if thet helps!
22:06.14_Raptor_ManxPower: i have too
22:06.18_Raptor_[default]
22:06.18_Raptor_exten => 1,1,Meetme(100)
22:06.53*** join/#asterisk Capouch (501@12.176.248.4)
22:07.16eKo1__Sparks_: That doesn't help.
22:07.26_Raptor_terrapen: ?
22:07.34terrapenthe wiki?
22:07.36terrapen~wiki
22:07.38DsrtZrzmraI have Forbidden problems using kphone, i've tryed using password and passwordless account, kphone wonk register
22:07.41_Raptor_yes
22:07.46_Raptor_terrapen: you mean me?
22:07.55terrapenno, Santa Claus :)
22:07.59_Raptor_:-/
22:08.01_Raptor_sry
22:08.01DsrtZrzmraanybody have the same error?
22:08.06terrapenyes, you. :)
22:08.14__Sparks_eK01: What would!
22:08.22jero_SFLphonedid you try sflphone ?
22:08.45harryvveko1 alias wcfxo zaptel ztcfg ?
22:08.56_Raptor_ok, let me ask one simple question: what is the best way to realize h323 conferencing with speex codec?
22:09.30harryvvwell that did not work
22:10.17JerJer_Raptor_: realize the fact that you no longer need H.323
22:10.28terrapenhhahaha
22:10.29__Sparks_eKo1: It seem to happen when I call the Sipgate PSTN number, it reports it as soon as the SIP phone rings
22:10.40_Raptor_JerJer: why not?
22:10.46_Raptor_JerJer: sip?
22:11.16*** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net)
22:11.27*** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com)
22:12.06Darwin35zaptel head sucks
22:12.40tzangerDarwin35: what's wrong with it
22:12.42Trionniscan't say I've ever gotten head from a zaptel.... wouldn't know
22:12.45Trionnis;)
22:12.53tzangerTrionnis: heh
22:13.09Darwin35it wont compile on debian
22:13.14jero_SFLphoneIm having lots of troubles with zaptel cards
22:13.34tzangerTrionnis: put it this way -- you need a big woman for decent zaptel head because there's just not a lot of brains in the hardware
22:13.42tzangerjero_SFLphone: like what
22:13.45Trionnislol
22:13.51Trionniswell played :)
22:14.09harryvvDarwin35 what is your zaptel alias setup as in /etc/modprobe.d/zaptel
22:14.15jero_SFLphonetzanger, like echo (not the most important) and callerid
22:14.24tzangerjero_SFLphone: where are yo ulocated
22:14.30jero_SFLphonein montreal
22:14.50tzangernorth american callerid works just fine
22:14.58*** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au)
22:15.00jero_SFLphoneyes, in 5% of cases for me
22:15.02tzangerremember that with POTS you need to wait AT LEAST two rings to get it since it is sent between the 1st and 2nd ring
22:15.12jero_SFLphoneah
22:15.31harryvvtzanger get what? i have post on this end
22:15.34Trionnisnever understood that
22:15.39harryvvwait two rings
22:15.45ManxPower__Sparks_, If the IP address of the phone does not show when you do a "sip show peers" then the phone is not registering with Asterisk and you cannot call it.
22:15.46Darwin35thereis no modprobe.d on debian
22:15.46Trionnisusing subband anyway, why not just send it prior to the first
22:15.49tzangerand echo -- well there's the fxotune if you're using wctdm and zapmonitor (I think) to help you with that too
22:15.58jero_SFLphone2 rings is how much delay ?
22:16.00tzangerharryvv: that's just how it's sent here in Canada
22:16.02jero_SFLphone8 secs ?
22:16.09tzangerjero_SFLphone: well no, the system should do it for you
22:16.16tzangeryou can try waiting 4 seconds
22:16.23tzanger2 seconds on, 4 seconds off is the ring cadence
22:16.32tzangerso 2 second ring, that 4 second pause is when the callerid is sent
22:16.36ManxPowerDsrtZrzmra, I don't see your paste of the error message you are getting.
22:16.43jero_SFLphonei most often get checksum errors
22:16.58tzangerTrionnis: I didn't write the telco docs :-)
22:17.07Trionnisnor did I ;)
22:17.10tzangerjero_SFLphone: use ztmontior and get your gains set up right first :-)
22:17.17harryvvtzan, sorry I was not following the both of you. Mabey that has to do with some kind of issue where my ivm picks up the pots line at two rings or four rings.
22:17.19Trionnisjust muttering :)
22:17.22jero_SFLphonetzanger, my gains are well tuned
22:17.22tzangerit could be that the audio is just to quiet or loud to reliably decode the FSK datastream
22:17.24harryvvDarwin35 take a closer look its there.
22:17.38__Sparks_ManxPower, It is showing, and I can make & recieve calls fine, it just irritates me thowing an error when it shouldn't!
22:17.40jero_SFLphonetzanger, I tuned my gains with a milliwatt at the other end
22:18.10tzangerjero_SFLphone: that's great for your txgain, but how did you tune your rxgain (which is what the dsp is using to hear the far end)
22:18.22_Raptor_ok, one more simple question: what is the best way for conferencing with speex (with linux and windows clients)
22:18.33jero_SFLphonetzanger, I called a milliwatt test number
22:18.43tzangerahh okay
22:18.43jero_SFLphonetzanger, with my asterisk
22:18.53tzangerit should be pretty good then
22:19.00tzangerI am not sure what to tell you
22:19.05jero_SFLphoneI think so, echo is not a problem anymore
22:19.10Trionnisanyone point me toward info about the output audio bitrate for ulaw?
22:19.15*** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu)
22:19.31Trionnistrying to use it with FWD and icecast streaming, but it's hosing up the voice audio
22:19.32tzangerI thought output bitrate was 64kbps
22:19.35Trionnismoh is fine
22:19.41TrionnisI don't know :(
22:19.42*** join/#asterisk search_learn2005 (~Miranda@209.68.139.150)
22:19.54Trionnisand can't seem to find it in the docs or the wiki
22:19.57jero_SFLphonetzanger, I even patched chan_zap to have a separate gain setting for the CallerID stuff
22:20.04Trionnisyes, I'm lame... flog me at will
22:20.09Trionnis:)
22:20.19*** join/#asterisk DaLion (~Miranda@Toronto-HSE-ppp3884470.sympatico.ca)
22:20.23tzangerjero_SFLphone: you certainly sound like you know what you're doing -- you might want to post something in -dev and see what kicks up
22:20.27tzangeror join #asterisk-dev and ask there
22:20.40tzangerboth places seem very strange for responses... sometimes rihgt awy, other times as if you were talking to a brick wall :)
22:20.55*** join/#asterisk peted20 (~chatzilla@24-113-67-25.wavecable.com)
22:20.57outtolunchuh?
22:21.02*** join/#asterisk Luke-Jr (~luke-jr@207.192.219.246)
22:21.03jero_SFLphone:) I'll try
22:21.08tzangeranyway
22:21.09tzangergotta get kids
22:21.11tzangerlater all
22:21.16jero_SFLphonebye
22:21.17RoyK<PROTECTED>
22:21.34search_learn2005still trying to decide to use the analog way or to buy budgetstream phones: 7 fxos and ~50 fxs, existing 10/100 network, existing analog phones
22:21.50search_learn2005any suggesstions
22:22.37*** join/#asterisk genie (~test@gate.us.cyberscience.com)
22:23.25*** join/#asterisk didz_ (~omg@200.218.193.30)
22:23.53*** join/#asterisk riksta (~rick@81-178-248-194.dsl.pipex.com)
22:24.38bjohnsonsearch_learn2005: existing cat5 to all proposed phone locations?
22:24.56DaLionanyone can tell me if DB odbc version of iaxfriends and sipfriends can hold  like g279;ulaw;;gsm << - not i have a blank entry.. will that use  #1 else #2 else #4 ? or they need to be in order
22:25.24*** join/#asterisk welby (~welby@solas.plus.com)
22:25.27*** join/#asterisk anti (russ@anti.developer.gentoo)
22:25.41bjohnsonsearch_learn2005: I suggest some of both.  voip phones where cat5 wiring is already run and fxs/analog phones where 2 pair is already run.  Service new locations with whatever seems best for that location
22:25.48antihmm is there anyway to get rid of the like 2 ring delay from when I call in a zap line and have it dial a SIP channel right away?
22:26.13bjohnsonanti: do you have a big wait()?
22:26.29DaLionlol
22:26.36antibjohnson: nope no wait, just immediate Dial(SIP...
22:26.38*** part/#asterisk Sesq (~Sesq@gate.us.cyberscience.com)
22:26.49search_learn2005bjhnson: every room has CAT5 avilable, what would you say for the echo issue? Which alternative has less echo trouble?
22:27.14greg_worksearch_learn2005: might also want to consider sipura 841's
22:27.15antibjohnson: I see "Starting simple switch" immediately, during the very first ring, but then it sits there, doing nothing then finally Executes the Dial..
22:29.39*** part/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com)
22:30.50johnnybWhat do you all think of the iLBC codec?  I'm trying to decide on a codec to use for my new budgetones, and iLBC seems to be a good choice for a low-bitrate codec.  However, G728 looked good, too.  What do you all think?
22:30.57johnnybI thought the G711 sounded good, but using 64kbit/s sounded awfully hoggish.
22:31.12jero_SFLphoneanti: do you use callerid ?
22:31.24johnnybI didn't notice any degredation w/ iLBC, but thought someone with more experience could offer some advice.
22:31.53bjohnsonsearch_learn2005: they potentially all have echo problems to be solved.  I doubt that is a decision maker for you.
22:32.35bjohnsonsearch_learn2005: even if cat 5 in all rooms, it is suitable for additional bandwidth required for multiple phones?  Is it available in suitable locations within the rooms?  Maximize what you've already got
22:33.19guugmemberalguien que hable español aca?
22:33.24bjohnsonanti: I don't know .. haven't seen that myself with Sipuras or X100P that I have
22:33.26ionixanyone has a way for asterisk to pickup the name of the caller from the ANI ? Like from a RBOC database or such ?
22:33.36search_learn2005bjohnson: if I get the VOIP phones for every room then I don' t have to get adtrans 750t which makes installation easier. Then I will only need a card with 7 fxos and that's it. Am I right?
22:34.15bjohnsonionix: I am told you can maintain your own db and go lookups based on CIDnum or subscribe to dbs provided by others
22:35.08DsrtZrzmrayo hablo español
22:35.18DsrtZrzmrapero no se mucho, apenas estoy empezando
22:35.41bjohnsonsearch_learn2005: yes and no.  Yes you would only need the fxo's for the lines.  No it probably won't make the installation any easier .. just different.  Also, have you looked for partial PRIs instead of 7 individual lines?
22:35.59nestArPRI > POTS
22:36.06nestAri <3 my pri's
22:38.33bjohnsonsearch_learn2005: a switch from POTS to PRI would take some planning and organization.  Although the monthly cost would be similar (if a partial PRI is even available), there are features available to PRIs that makes managing them and adopting future changes much easier
22:38.43__Sparks_Another question! - If i make calls via Sipgate tp a PSTN number, the quality is very good, no echo - if however I route the call via my x100p card, theere is a lot of echo on the line for a while (it clear up eventually, but to start with it's really bad - what can be done about this!?
22:39.10bjohnsonplay with the x100p settings
22:39.39nestArwe pay like $425 for pri's.. one pots line at business rates is ~$90 a month..
22:39.57dsmousegah
22:40.03harryvvbjohnson who has a x100
22:40.05dsmouseI can get vonage to work, but only with ulaw
22:40.09dsmousewhich sucks
22:40.18search_learn2005bjohnson: I wouldn't have a say at the 7 fxos at least at the moment. the school already has the 7 fxos and they had them for years. So, if I want to go with the adtrans method, how many 750s wil I need, and will I need the most basic model of 750, because there are 3 versions. Are there any configurations on the adtrans or will everything still be done on the asterisk server. That's why I am a li
22:40.18search_learn2005ttle bit scared with the adtrans method, there is just not do many newbie friendly documents out there about it. And, it is an expensive device.
22:40.22__Sparks_bjohnson, I would if I know what to play with! - can you give me a clue :-)
22:41.24bjohnsonharryvv: __Sparks_ does
22:41.37__Sparks_is that a bad thing to have then!?!
22:41.42*** join/#asterisk denon (denon@synapse.subneural.net)
22:41.42*** mode/#asterisk [+o denon] by ChanServ
22:41.43harryvv__Sparks_ you also run debian?
22:41.48__Sparks_yea
22:41.58harryvvgood!
22:42.03bjohnson__Sparks_: no idea.  I fighting an echo with SPA fxo myself that might be related to gain or line impedence
22:42.14harryvvdoes your load the modules apon reboot and start *?
22:42.53harryvvbj, more then likly line impedence
22:42.58Jlau515in a extensions.conf, a exten => s,1,anwser, would that anwser all incoming calls from a zaptel channel
22:43.00DaLionbkw you alone on conf now ;)
22:43.01bjohnsonsearch_learn2005: check ebay .. used adit600 with 24 fxs often go for < $300 I'm told
22:43.22harryvvbj, good example is when my wife picks up the analog phone on the same line that goes into the asterisk box it upsets the impedence and echo goes up.
22:43.28bjohnsonsearch_learn2005: tzanger is usually helpful with chan bank questions
22:44.12bjohnsonsearch_learn2005: generally, a chan bank system could lower your up front cost to about $30 USD per phone compared to about $80 each for cheap new ones
22:44.40Jlau515i'm trying to play a main menu prompt
22:45.18bjohnsonsearch_learn2005: also some chan banks can provide fxo ports .. so 3 chan banks and a quad T1 card from digium would allow you to do anything you want
22:46.22search_learn2005bjohnson: but how do I hook up a channel bank to a pci card no idea, and I am a visual learner. So any website where I see a channel bank hooked up to a card via T1?
22:46.24bjohnsongreg_work: talk to tzanger .. I think they are usually about $800 CDN / mo for 23 channels
22:47.06bjohnsonsearch_learn2005: probably .. but I don't know where.  Try the usual suspects
22:47.12bjohnson~usual_suspects
22:47.15bjohnsondamn
22:47.24bjohnsonwiki, mailing list archives, google
22:47.26*** join/#asterisk Luke-Jr (~luke-jr@207.192.219.246)
22:47.33bjohnsontime for me to go
22:48.36trymI have installed spandsp to have asterisk receive faxes. When a fax call is made to asterisk, asterisk starts whining about RFC3389. I also notice that the volume spandsp/asterisk is communicating with varies.. which is not normal for a fax session. Any suggestions?
22:50.06Darwin35fricking cvs
22:50.25*** part/#asterisk search_learn2005 (~Miranda@209.68.139.150)
22:50.41Darwin35i need zaptel for timing
22:50.49Darwin35and for when the new card gets here
22:51.23*** join/#asterisk Nukemizer (~Nuke@65.103.231.133)
22:53.19NuggetI was never able to get meetme working in freebsd.
22:53.25NuggetI finally just gave up
22:55.26*** join/#asterisk anthm (anthm@208.254.19.131)
22:55.26*** mode/#asterisk [+o anthm] by ChanServ
22:58.16*** join/#asterisk kippi (fc@cpc4-hatf3-6-0-cust243.lutn.cable.ntl.com)
23:00.00Jlau515can someone help me with an ivr issue?
23:00.12kippihey
23:01.17*** join/#asterisk Legend (~legend@24.244.142.133)
23:01.46Jlau515i have this in my extensions.conf
23:01.52Jlau515exten => s,2,Answer                     ; Answer the line
23:01.53Jlau515exten => s,5,BackGround(demo-congrats)  ; Play a congratulatory message
23:02.29Jlau515i want to know should that be playing if an incoming call came from my zaptel channel
23:04.39ManxPowerJlau515, Not with those priorities it won't.
23:04.43ManxPowerHell, it won't work at all.
23:04.57ManxPowerpriorities must be consecutive (normally)
23:05.06*** join/#asterisk jarnaud (~jarnaud@65.217.47.11)
23:05.08jsolareswhere's s,1 ; s,3 ; s,4?
23:05.11ManxPowerstarting at 1
23:05.30jarnaudSomeone has played with auto dialout?
23:09.01Jlau515i actually have prioritys for 1 2 3 4 5
23:09.19Jlau515was wondering if using 's' means that will answer all incoming calls
23:09.54Jlau515i didnt want to spamm the channel my include all my other s,1 ; s,2 ; ...
23:10.18NukemizerDiguim support is having me update my cvs to get my T1 card to work - i did not get instructions on what I need to do so I was hoping someone could verify what I have found "asterisk-update.sh  update"
23:10.20Jlau515dialing out works for me, but dialing in does not so far
23:12.37jsolaresJlau515: it all depends if that's the context you have defined for all your incoming channels
23:13.18Jlau515its in my [from-sip] content
23:13.24*** join/#asterisk Nix (~Nix@81.213.125.220)
23:14.25jsolaresJlau515: what exactly do you want to do?
23:14.31*** join/#asterisk ennuyeux72 (~ennuyeux7@62.53.79.208)
23:14.55Jlau515just play the demo prompt when a call comes in from my zapata channel
23:15.21jsolareswhat context did yo put in your zapata.conf?
23:15.35Nixhey guys
23:15.47Jlau515i just looked i had default context
23:15.54Jlau515it needs to be from-sip right?
23:16.06tzangerdo not use the default context
23:16.10*** join/#asterisk micrisc2 (~micrisc@mail.techhelpresources.com)
23:16.19*** join/#asterisk doctor_za_ljubav (bkwyg@195.252.86.177)
23:16.19tzangerI'd put zap in something like 'from-zap', from sip in 'from-sip' etc until you understand more
23:16.20jsolaresmhnoyes: or you should make an inbound one in extensions with the promp including what you want
23:16.28jsolares*could
23:16.36Jlau515k
23:16.36jsolaresand prompt
23:16.37jsolaresbleh
23:16.39Jlau515i'll try that
23:16.42doctor_za_ljubavhello
23:16.52jsolaresit'll work ;)
23:17.02jsolareshi
23:17.06ManxPower6 hours on cpnference calls today.  *sob*
23:17.12*** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net)
23:17.16doctor_za_ljubavcan someone tell me where can i download a digium T400P linux driver.....
23:17.28Jlau515hmm still dont work
23:17.37Jlau515i see the external number in zap show channel
23:17.40micrisc2anyone using spandsp fax addon?  i am having problems compileing, /usr/include/tiffiop.h:38:18: port.h: No such file or directory i get this but the file is there
23:17.57*** join/#asterisk yurpls (~yurplsl@65.114.15.70)
23:18.02Jlau515but the demo does not answer
23:18.28Jlau515i tested the demo by asigning a extension for demo playback and that plays
23:18.55*** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com)
23:19.05ManxPowerJlau515, Did you fix your priorities?
23:19.24jsolareshe had priorities, just did not want to spam the channel with all of them
23:20.07*** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)
23:21.22*** join/#asterisk lilneon (~tj_r3@cuscon4828.tstt.net.tt)
23:21.29*** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com)
23:21.33lilneonhi guys an dgood night everyone
23:21.47Nugget"linux is great for supporting things like $7 ebay tape drives.  This is of critical importance for those users who have only $7 worth of data."  <-- heh
23:22.08ennuyeux72anyone know if asterisk deals proprely with a pending response to an reinvite request
23:22.34mikegrbNugget: :D
23:25.06*** join/#asterisk ProAtWork (~procyan@host157.reign.radel.com)
23:25.19ProAtWorkhi, anyone know of a source for cisco 7960 ringtones?
23:25.43ProAtWorkalso, if you have problems converting one of these things to SIP I just spent a bit of time figuring it out
23:25.49RoyKwtf is a 9760?
23:26.09ProAtWork7960 is the big cisco phone with the display
23:29.37Jlau515i am getting the follow error from the cli
23:29.40Jlau515<PROTECTED>
23:29.45antiSo I have a SIP phone and a zap channel, someone calls in the zap channel, rings my sip phone, I pick up. I'm trying to come up with a way I can enable recording then, something where I can press #[something] and have it announce on the line this call is about to be recorded (comply with laws) and then run Record() .. but sadly, I can't figure out how.
23:29.53Jlau5154152484088 is the number i am dialing from
23:30.01Jlau515anybody know how to fix this issue
23:31.10NuggetProAtWork: I've got a bundle of them.  the same bundle that I presume everyone in here has.
23:31.31yurplsAnyone help with a TDM400P card?
23:31.38Nuggetthere's that one guy's config linked from the wiki that we all seem to find and use
23:33.52ProAtWorkNugget hmmm... the voip-info wiki?
23:34.56Nuggetof course
23:35.07ProAtWorkI don't find any links to ringtones on that page
23:35.14ProAtWorkI found a link to a bunch of .gsm files
23:35.28ProAtWorkthese won't work on a 7960 according to the docs
23:35.36Nuggetit's not ringtones specifically, just that guy's config.
23:35.43*** join/#asterisk Ayano (~erik_leee@209.143.187.254)
23:35.48ProAtWorkoh.. I can configure the phone.. not a problem
23:35.57ProAtWorkjust looking for .pcm files
23:36.00Nuggetjesus.
23:36.01Nuggetyes.
23:36.02NuggetI know.
23:36.13Nuggetthat sample config includes a whole shitload of ringtones.
23:36.28Nuggetgoogle for ringlist.dat
23:36.50ProAtWorkahhh.. in the tar file
23:36.57ProAtWorkok I'm catching on now
23:36.58Nuggetright-o
23:37.03Nuggetsorry if I wasn't clear.
23:37.18ProAtWorkthanks for pounding me in the head enough times with the clue-by-4
23:37.45rvhihi, for voicemail, is it possible to customize the header of notification email for each context
23:37.46rvhi?
23:38.13_Raptor_cya
23:38.30*** join/#asterisk nextime (~nextime@ns0.nexlab.net)
23:40.44AyanoWhat are the best choice phone for asterisk, I've worked with budgetone, and some lower ciscos, what do you guys recomend?
23:41.44jsolaresi like the avaya 4602 IP
23:42.01AyanoDo you have to upgrade firmware to use it?
23:42.14d-techuniden looks like a good phone for the price
23:42.19jsolaresif it has h323 firmware then yes
23:42.28jsolaresif it has sip firmware then no
23:42.32*** join/#asterisk eye69 (magnus@ipv6.upcore.net)
23:42.34kippihey, where is the best place to read up on how to create SIP users on asterisk (on a freebsd system)
23:42.47jsolaresthe wiki
23:42.48lilneonhey guys.. anyone know where or can help me with a script to open or forward he relevant ports to allow voip(SIP or IAX) on a proxy with Squids behind that proxy?
23:43.17jsolareslilneon: does it have iptables for firewall?
23:43.22NuggetI'm fond of my cisco 7960, but I dunno if they're worth the cost.
23:43.27AyanoKippi: The asterisk wiki has the easy context to set up sip users
23:43.40NuggetI can say that you absolutely do not want that zyxel/pulver wireless thing.
23:43.42lilneonjsolares: yes..
23:44.03AyanoI've used them, sometimes the firmware is a pain.
23:44.08lilneonjsolares: need a script to by pass or open/forward the relevant ports for voip to work
23:44.30jsolaresiptables -t nat -A PREROUTING -s 0/0 -d <public ip> -d udp -port 5060 -j DNAT --to <internal ip>
23:44.38jsolaresor something like that, let me check
23:45.05kippiok, i'll have a read of it and try and get it working
23:45.09jsolares-p udp
23:45.17jsolaresnot -d udp
23:45.25Ayanokippi: hold on, I'll give you a link.
23:45.27jsolaresport 5060 is for SIP
23:45.32doctor_za_ljubavanyone to help me install t400p card?
23:45.43*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
23:46.13Ayanohttp://www.voip-info.org/wiki-Asterisk+config+sip.conf
23:46.41kippiAyano: thanks
23:46.48lilneonjsolares: would it matter if they have a dynamic ip?
23:47.27jsolaresthe internal or the external?
23:47.37jsolaresand yes it does matter, whenever they change the rule is worthless
23:47.38__Sparks_I have a problem calling X-Lite extentions from my BudgeTone - as soon as I talk, X-Lite crashes - if I call the other way (X-Lite to BudgeTone) there isn't a problem - any ideas why!?
23:47.57lilneonjsolarres: ok cool.. thnx..
23:48.19jsolaresread up on iptables and portforwarding, google is your friend ;)
23:48.36lilneonjsolares:way ahead of u
23:49.05jsolaresi'm glad i have my * server with a public ip, no hassle pita configuration for that
23:50.10Ayanojsolares: I set it up using this once....   http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
23:50.37jsolaresyeah, that's a good read :)
23:50.56AyanoIt causes an echo if you don't have a good router.
23:51.17jsolaresso that's where the echo comes from :o
23:52.09AyanoIf the router doesn't translate fast enough it makes it a pain to get rid of the echo.
23:52.38jsolaresoh well, atleast the callee doesnt hear it
23:53.11jsolaresand since i'm the caller it matters not
23:54.03*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
23:54.13AyanoTrue, I found that it is sparatic.  Sometimes good, sometimes bad.  Sometimes all it takes is to hang up and call again and it goes away...... Strange.  I took out the router and had no problems.
23:54.42AyanoThats why I figured it was a cheap nat router problem.
23:54.46jsolarestoo bad i cant take out the router
23:54.48*** join/#asterisk alerios (~alerios@201.245.167.88)
23:54.55jsolaresmaybe i should upgrade the firmware on my wrt54g
23:55.35AyanoRead the spec on the firmware upgrade and it will tell you what it changes.
23:55.46AyanoBe carefull, sometimes it wipes your settings.
23:55.59AyanoMake a copy of your configs on the router first.
23:56.16lilneonjsolares:i have to run this on their server right?
23:56.21jsolaresi dont really have setting that i could wipe out, i do like sveasoft's firmwares, the alchemy one has qos and support for l7
23:56.27jsolareslilneon: yep
23:56.35Nukemizercan somebody confirm for me that to update my cvs for  zaptel - lipri and asterisk   i would do  a    "cvs checkout  zaptel"   " cvs checkout libpri"   " cvs checkout asterisk"  ?
23:56.45lilneonok coo,l
23:57.11didz_cvs checkout asterisk zaptel libpri
23:57.16Ayanolilneon: did you look at this?    http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
23:57.40lilneonayano:yeah.. believe it or not i do take tghe advice on here seriously
23:57.46Nukemizerlilneon, thanks
23:57.50jsolaresyou would be case 3 and 6
23:57.57Nukemizerdidz_  thanks  :)
23:58.04lilneonalso got this from dca[laptop]:http://www.voip-info.org/tiki-index.php?page=Asterisk%20firewall%20rules
23:58.15lilneonthnx to teliax support.. :)
23:59.27jsolaresthat one is for openning up a firewall, but not for portforwarding

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