irclog2html for #asterisk on 20050228

00:00.03hmm-workheh this is driving me nuts, asterisk doesn't destroy a channel created with a manager originate if the call fails until about 30 seconds after it fails
00:01.32*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
00:01.32*** mode/#asterisk [+o bkw_] by ChanServ
00:01.52|Vulture|sup bkw_
00:02.21bkw_not much
00:02.26bkw_just got my GPRS working in OSX
00:02.29bkw_nice nice nice
00:02.37|Vulture|oh no! not more mac stuff lol
00:02.39|Vulture|:P
00:03.09harryvvgprs?
00:03.34*** join/#asterisk deRost (~deRost@054.209-89-66-0.interbaun.com)
00:03.35bkw_my cellphone boi
00:04.06harryvvbkw, come to #hamradio2 some time I talk with somone by the name of pilotmike who is one of the network engineers at apple. Nice guy. He said he would give me a good deal on g4 some time.
00:04.29bkw_kewl
00:04.46bkw_now i'm not stuck in the airport without wifi
00:05.02*** join/#asterisk RoyK (~roy@8.80-203-22.nextgentel.com)
00:05.10bkw_its like i'm back on dialup
00:05.11bkw_:P
00:05.26harryvvwe have things in common so we talk. served in the same branch of the service both licenced ect :) He even bumps into steve jobs once in a while.
00:05.38tzangerericw: you can learn the 5WPM needed for the advanced license
00:05.47tzanger5WPM is almost painful to try and transcribe it's so fucking slow
00:06.01harryvvI toped out at 20
00:06.23|Vulture|bkw_: we missed you in Orlando last night... big party! lol
00:06.35tzangerstepcut: :-)
00:06.41ericwI read that you can setup 56kbps links over packet radio
00:06.44tzangerfuck that, code it out NRZ and pipe it into a serial port
00:06.50stepcutbut then I quite the job, so that was the end of that :-/
00:07.36*** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl)
00:07.42tzangerled flash sequences, time between flashes, hell one time I had a PWM I wasn't using on the chip so I coded different %age outputs for different meanings
00:07.54tzangeradd an RC for a 1st-order filter and display the output on a scope
00:08.09tzangeryou get a little "cityscape" output which gives you your debug information :-)
00:08.29tzangerby far and large though the easiest is NRZ serial output
00:08.48tzangerpipe it directly into your PC with a logic-level to RS232 converter
00:08.48stepcutoften we just hooked a logic analyzer to the address and data lines and just looked at what ram was being accessed -- no caching/piplining, so it was pretty straight forward :)
00:08.54tzangeri.e. a transistor :-)
00:09.03tzangerstepcut: yeah but that's a lot of I/O
00:09.19stepcuttzanger: well, it depends on the problem that was being solved
00:09.24tzangertrue enough
00:09.49stepcuttzanger: like when the processor is buggy :)
00:09.55bkw_klj
00:10.09RoyKbkw_: do you like aefirion?
00:12.12GoshenHey znoG: you here? I figured out how voipuser.org works...they get paid for calls to your UK number by others...
00:12.32bkw_RoyK, its got some good ideas
00:13.07RoyKas in "stability" and "open source"?
00:21.07*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
00:21.07*** mode/#asterisk [+o bkw_] by ChanServ
00:24.53*** join/#asterisk _phate_ (~phate@phate-0002.user)
00:32.26*** join/#asterisk MrEntropy (~entropy@ppp55-252.lns1.adl2.internode.on.net)
00:32.28MrEntropyyo
00:32.31bkw_where is everyone
00:32.34bkw_nobody is talkin
00:32.45MrEntropyi'm here =)
00:32.53Silik0nthey heard you were on gprs and didnt want to overload your bandwidth
00:33.33cypromishiding
00:33.40bkw_haha
00:33.44bkw_overload?
00:33.46bkw_how?
00:34.04zigmanbkw
00:34.09cypromismost countries gprs is payed per mb transfer
00:34.13Silik0nwell we all know gprs has monsterous bandwidth there hah
00:34.20zigmanyou reset my mantis pass the other day
00:34.24harryvvFunny how companys come up with the wierdest domain names that has nothing to do with there companies type of work :) Finding something close to voip been used up for the .com's so far.
00:34.26zigmanbut i did not receive any mail
00:34.30zigmanwith the new pass
00:34.46zigmancypromis are there countries with flat gprs ?
00:34.54zigmanjapan and usa probably
00:35.16zigmanharryvv ?
00:35.37bkw_ULIMITED GPRS baby
00:35.39bkw_19.95/mth
00:35.41bkw_not bad eh?
00:35.46MrEntropy=O
00:36.43cypromiszigman: I pay flat
00:36.46Silik0nbkw_: TMo?
00:36.54cypromisabout 35$/mth
00:37.08harryvvzigman keep comming up with some more or less voip related domain names and everthing for the most part is hosted. voiptel televoip broadtel you name it. Thats the easy list :)
00:37.14Silik0ni have tmo grps... unlimited gprs 20USD
00:38.07harryvvWhat are the advantages of grps?
00:38.30MrEntropygprs here is a kidney/pm
00:38.34Silik0nmobile IP connection one a GSM mobile network
00:38.40zigmancypromis , bkw_ you suck ;)
00:38.47*** join/#asterisk {zombie} (zombie@soulasylum.penguincare.com.au)
00:38.48bonez39I have an ATA, Motorola model VT1005V which came with my service from Vonage....can I hack this model and set it up to work with asterisk?
00:39.53harryvvsilk I see.
00:40.10hermieI love crackpots
00:40.24hermieand CDMA
00:40.37cypromis:P
00:41.09hermieWard Dean has got to be my favorite web crackpot at the moment... although Ted Gunderson is pretty close
00:41.21hermieexcuse me, Dr. Ward Dean
00:41.53harryvvhermie thats the old apple protocol :)
00:42.00harryvvapple talk.
00:42.03*** join/#asterisk JerJer[mobile] (~jj@equinix.ord.scnet.net)
00:42.17harryvvhave not seen that acronym for a while. Glad thay got rid of it.
00:42.30hermiewhat acronym?
00:43.20harryvvCDMA
00:43.40hermiehuh.... CDMA is still very much around
00:43.51hermiethe next version of GSM is... WCDMA
00:43.52harryvvyea but apple mostly supports CSMA
00:43.55harryvverr
00:44.10hermieCDMA=cell phone protocol
00:44.15harryvvyea
00:44.23hermiethe good one :)
00:44.59bonez39they use GSM in europe, and asia, right?
00:45.22cypromisyup
00:45.28hermiekorea and some of japan use CDMA
00:45.30znoGGoshen: yep i worked that out too. They charge for inbound calls by the minute and they use that money to offer free outbound calls. I think it's a great idea :)
00:47.00bonez39znoG, does that model make economic sense? Life haas taught me any more to question how they can cover the costs....I guess if millions of mintes are used and paid for on incoming..then the outbound calls can be free....I
00:48.47znoGbonez39: well it does make sense ao long as they get lots of people calling their UK numbers.
00:49.34*** join/#asterisk sivana (~sivana@165.154.13.35)
00:50.13bonez39oh, is it based in the UK?
00:52.37*** join/#asterisk bidet (~WinNT@adsl-066-156-072-026.sip.asm.bellsouth.net)
00:53.26shmaltz~cdma
00:53.27jbothmm... cdma is code division multiple access
00:53.35shmaltz~gsm
00:53.36jbotrumour has it, gsm is Groupe Spיcial Mobile
00:54.12bidetSorry if this is covered in a faq, but I haven't found it yet - what is the general system overhead for asterisk, in idle moments?
00:55.11hermieSTATE REPTILE Act 281 of 1995 "The painted turtle (Chrysemys picta) is designated as the official reptile of this state."
00:56.43shepherdbkw: how's your distro coming along?
00:57.07pUmkInhEd~pri
00:57.08jbotsomebody said pri was Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc.
00:57.32shepherdJ1 too
00:57.34shepherdhehe
00:57.59harryvvbidet why do you ask.
00:58.31pUmkInhEd~pstn
00:58.32jbotrumour has it, pstn is Public Switched Telephone Network
00:58.42bidetwell its somewhat involved
00:58.43shmaltzdoes digium ~pots
00:58.51shmaltz~pots
00:58.52jbotsomebody said pots was Plain Old Telephone Service as in "Old Analogue Crap"
00:58.54bidetbasically i have a linux box, and if asterisk isn't a memory hog, i want to install it
00:59.02shmaltz~puts
00:59.10shepherdheh
00:59.14shmaltz~putz
00:59.15bideti am a single user on my voip, but my pstn access service only allows one device logged into it
00:59.15shepherdasterisk loves cpu :)
00:59.30pUmkInhEd~isdn
00:59.31jbotisdn is probably (Integrated Services Digital Network) This is a digital line that is often used to connect to the Internet. It generally come in two flavors: one is a 56 Kbps version, which in actuality only uses half of the ISDN line's bandwidth; the other is the 128 Kbps version, which uses both the 56 Kbps channels on the line. However, that's only 112 ...
00:59.35shmaltzanybody using polycom phones with *?
00:59.35bidethence i can only have one device ring when that line gets an incoming call
00:59.43shmaltz~intarweb
00:59.44bidetasterisk would solve that problem
01:00.08shmaltz~interweb
01:00.10bidetbut its not a big enough problem yet to bother setting up a dedicated box just for asterisk
01:00.19shmaltz~internet
01:00.20jbotfrom memory, internet is A worldwide network of computer networks. It is an interconnection of large and small networks around the globe. The Internet began in 1962 as a computer network for the U.S. military and over time has grown into a global communication tool of many thousands of computer networks that share a common addressing scheme.Unlike online services, ...
01:00.37shmaltz~bye
01:00.38jbotl8tr
01:00.46hermie~putz
01:00.47jbot[putz] a person who constantly asks jbot questions in a channel instead of using /msg
01:00.49pUmkInhEdneato
01:01.03shmaltz:)
01:01.09BrianR___Anyone here played with cellsocket?
01:01.10pUmkInhEdhint taken hermie
01:01.14pUmkInhEd:)
01:01.14shepherdbidet: asterisk is actually pretty minimal unless you are doing a bunch of encoding
01:01.21harryvvthe internet was concived on a napkin in a resteruant. I think it was NSF that created it.
01:01.42shmaltzpUmklnjEd, hermie meant me
01:01.48bidetthe maximum user load would be one user, perhaps calls conferenced
01:01.50shmaltzits OK
01:01.51puzzledharryvv: i think that was ethernet on the napkin
01:01.52bideterr 2 calls
01:02.01BrianR___harryvv: No no.. The napkin/resteruant story refers to appletalk...
01:02.12pUmkInhEdnp i been idle for a few days now
01:02.23pUmkInhEdi turn back over here and see whats going on
01:02.47pUmkInhEdi am eager to test some stuff with asterisk @work but don't have the time or energy to concentrate on something like that atm
01:02.56harryvvpuzzled mmm I saw a documentry by one of the founders and it was the idea of a global network was written on a napkin. Ethernet did not come into being untill 1976 and I personally know one of the first engineers that helped develope it.
01:03.11harryvvkinda interesting though :)
01:03.27harryvvGood things come from the cold war.
01:03.44puzzledharryvv: yeah, must make a mental note to lug napkins along in case I "see the light"
01:03.52harryvvhehe
01:04.28harryvvI wish that I was a little older and really experainced the 1960s.
01:04.42shepherdwhy?
01:05.05harryvvAt least I was a tot and did see the eagle land on the moon.
01:05.15shmaltzanybody here using billing with * ?
01:05.28ariel_harryvv, it's over rated.
01:06.18shmaltzMrEntropy, why?
01:06.18harryvvariel what is
01:06.27MrEntropyjust a joke =)
01:06.35shmaltz:)
01:06.55ariel_1960's
01:07.00shmaltzMrEntropy, ask jbot about me
01:07.41MrEntropyoh...you naughty man
01:07.46*** join/#asterisk visik7 (~ciao@visik7.user)
01:07.48shmaltz:)
01:08.02harryvvarial it was a cool time in history :)
01:08.11shmaltzMrEntropy, you from england?
01:08.19harryvvthen there was the semiconductor age in the 1970s.
01:08.20harryvv;)
01:08.45MrEntropyshmaltz: nope, aust
01:08.56shmaltzclose
01:09.05MrEntropynot geographically =P
01:09.15shmaltznope, you down under
01:09.22shmaltzu know how I figured?
01:09.40MrEntropyis it because you've seen me use the phrase 'bollocks'?
01:09.49shmaltznope
01:09.52shmaltznaughty
01:10.06harryvvarial ohh cool must be worth something. Arial, I regret the fact my perents tossed out the early 1970s volume space edition of life magazine books. Those what really facinated me about science and space. Do anything to aquire a set of those again.
01:10.11MrEntropyhow is that typically anglosaxon?
01:10.21shmaltzmy wife is from England, she keeps saying that word
01:10.36harryvvhehe
01:10.46shmaltzin my 25 years in america I would only read this word, and almost never hear it until I met my wife
01:10.56harryvvhehe
01:11.00harryvv:)
01:11.04shmaltz:)
01:11.07MrEntropyodd
01:11.08ariel_last year we through out a box of old mags from that time frame due to they were yellowing and full of bugs.
01:11.33harryvvI see
01:11.39shmaltzShe doens't say it to me, its just that this word in general is not used to much here, but is used in England
01:12.23shmaltzyou ever been to the states? MrEntropy
01:12.25sivanaweird.. she says it to me
01:12.33MrEntropyshmaltz: never
01:12.42MrEntropyshmaltz: it scares me
01:12.47shmaltzwhy?
01:12.54harryvvshmaltz, thats where the angalosaxons orginated from. Thats a part of my ancestory.
01:13.15MrEntropyshmaltz: it seems a very angry place, politically above all
01:13.19shmaltzit's a nice place here
01:13.38shmaltzthat's b/c we have the best freedom system
01:13.51MrEntropyshmaltz: i can't judge though, i've never been...so my view is completely unjustified
01:15.27stepcutThe US is the best because we have great companies like SBC and Verizon...
01:16.30shmaltz~bush
01:16.31jbot[bush] chick plumbing or the current president and potential dictator, or the guy that made stupidity fashionable.
01:16.38shmaltzi think jbot is a libiral
01:16.53shmaltz~iraq
01:16.54jbotrumour has it, iraq is a country
01:17.00goatmilk~yoga
01:17.02jbotyoga is probably a way that apt flexes himself, when there is a netsplit and none is in the channel
01:17.04shmaltz~israel
01:17.05jbotsomebody said israel was a quiet little heaven, not unlike Hawaii, or a very dangerous place right now!
01:17.12MrEntropy~transmogrification
01:17.19MrEntropy...
01:17.21goatmilk~luggage
01:17.28goatmilkwhat a dumb bot.
01:17.31shmaltz~quiet
01:17.32jbotACTION ok, ok, I will be quiet. When you start making sense, that is.
01:17.43shmaltz~dumb
01:17.44jbotACTION thinks that sumone is drunk, idiotic or a robot abuser.
01:17.49MrEntropy~scalp-wax
01:17.57shmaltz~dumbo
01:17.58goatmilk~~
01:18.07shmaltz~jbot
01:18.08jbotrumour has it, jbot is the shipboard computer, but you may call me eddie if it helps you relax
01:18.12shmaltz~dumbo
01:18.16ariel_wow sounds like jbot is us driven.
01:18.19MrEntropy~eddie
01:18.20jbotRobust, clustering, load balancing, high availability, web server tool.. URL: http://www.eddieware.org/
01:18.38shmaltz~brooklyn
01:18.51shmaltzstupid jbot
01:18.55shmaltzdoens't know much
01:19.04shmaltz~sex
01:19.05jbotI'm hermaphroditic
01:19.18shmaltz~hermaphroditic
01:20.01ariel_ROFL Cartoon network has Tom & Jerry.  Brings back old times.
01:20.30shmaltzadj 1: of or relating to monoclinous plants 2: of animal or plant; having both male female reproductive organs
01:20.30harryvvheeh
01:20.45shmaltzhttp://dictionary.reference.com/search?q=hermaphroditic
01:21.01shmaltz~sleep
01:21.02jbothmm... sleep is overrated, and a poor substitute for caffeine
01:21.09shmaltz~good
01:21.11jbotgood is probably a verb
01:21.16MrEntropy~broken
01:21.17jbot[broken] mailto:nothing@machine.cx -> http://machine.cx/debian/  or screen shots are at http://nivda.machine.cx or that's sid for you.
01:21.20shmaltz~laugh
01:21.21jbotACTION rolls around on the floor laughing
01:21.38MrEntropy~assassinate
01:21.41sivana~putz
01:21.42jbotit has been said that putz is a person who constantly asks jbot questions in a channel instead of using /msg
01:21.58shmaltz~sivana
01:22.18shmaltz~sivana
01:22.19jbotwell, sivana is a putz
01:22.25sivanaheh
01:22.41sivana~shmaltz
01:22.42jboti heard shmaltz is annoying the channel by playing with jbot
01:22.54shmaltzbut I'm not a putz
01:23.07shmaltzsivana, you know what the real definition of putz is?
01:23.12sivananope
01:23.24shmaltzhttp://dictionary.reference.com/search?q=putz
01:23.26shmaltz:)
01:23.48*** part/#asterisk phateSC (~phate@c-67-166-69-185.client.comcast.net)
01:24.01shmaltzknowing yiddish fluently gives me these words
01:24.37harryvvWhen a global ends in =foo what does that mean?
01:24.41MrEntropy~yiddish
01:24.46MrEntropy=(
01:25.03shmaltzsivana, the second definition at the above link is the real translation, but in Yiddish it is commanly used as a slang
01:25.38sivanaI see
01:26.30shmaltz~yiddish
01:26.31jbotrumour has it, yiddish is The language historically of Ashkenazic Jews of Central and Eastern Europe, resulting from a fusion of elements derived principally from medieval German dialects and secondarily from Hebrew and Aramaic, various Slavic languages, and Old French and Old Italian.
01:26.33MrEntropy~paganitzu
01:26.49MrEntropyas if it doesn have paganitzu!
01:27.22GoshenCan someone point me to an extensions.conf that adds the basic telco features like call forwarding?  voip wiki wasn't too helpful
01:27.43sivanait's a bit tricky for call fwd
01:27.55shmaltzGoshen, I think you will have to use your imagination, but for startes look at the dbput and dbget commands
01:27.55sivanayour ATA probably does it easier
01:28.11mishehu~hebrew
01:28.12jbothebrew is probably a stupid language that has way too many rules
01:28.19mishehuwhat?
01:28.19MrEntropyhahaha
01:28.26shmaltzjbot, how do you know that much
01:28.28mishehuhebrew is a very simplistic language
01:28.51mishehuI want to know who the hell said that about hebrew, as they obviously never studied english.
01:29.11mishehu~russian
01:29.12jbotrussian is probably troy says you cannot trust a russian, or #debian-russian
01:29.25mishehu~klingon
01:29.26shmaltzmishehu, I know both Hebrew and english, Hebrew is much easier but it has far more rules
01:29.40MrEntropy~paganitzu
01:29.41jbotpaganitzu is probably A classic, age old Apogee game
01:29.46MrEntropyyeah!
01:29.49mishehushmaltz: aside from the gender rules, 99% of hebrew seems to abide by the rules though.
01:30.18shmaltzunless you get involved in nifals poal and the like
01:30.31mishehuas opposed to english, which has at least as many rules, and where at least 1/3 of the language is in contradiction to those rules.
01:30.42mishehushmaltz: nu.
01:31.35mishehuעברית עדיין יותר הגיונית מאשר רוב לועזית
01:32.13mishehu~commanderkeen
01:32.29mishehuaahha, jbot doesn't know commander keen
01:33.48sivana~mishehu
01:34.20*** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com)
01:35.28MrEntropy~commanderkeen
01:35.29jbot[commanderkeen] A game involving the precise manipulation of the adventurous, pseudo-heroic protagonist kid through 'monster' infested levels with a pogo stick. It features several walking vegetables.
01:35.30MrEntropythere
01:35.44mishehuI forgot how to teach the robot
01:37.34hmm-workanyone know what the 'variable' field is for in the manager originate command?
01:38.58*** join/#asterisk visik7 (~ciao@visik7.user)
01:39.44shepherdit's probably for the best
01:41.07hmm-workheh i guess not
01:44.23*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
01:44.27*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
01:44.27*** mode/#asterisk [+o bkw_] by ChanServ
01:47.21hmm-workslow night tonight
01:48.14Zawonce i get my asterisk pbx set up, i'll need a gateway to use for outbound calls. does anyone have a recommendation?
01:48.17*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com)
01:48.34hmm-workZaw, what kind of interface?
01:48.55Zawhmm-work: i plan on doing everything voip, not pci cards, etc.
01:49.33hmm-workZaw, what kind of interface to the pot network
01:50.36Zawhmm-work: i was under the impression that i didn't need any additional hardware other than Ethernet to use voip, and that i only needed a gateway service.
01:51.23hmm-workyou are looking for a voip service provider not a piece of equipment to plug into the plain old telephone system
01:51.34hmm-workcorrect?
01:51.37Zawhmm-work: yes
01:51.50hmm-workthere's a million of them out there
01:52.20Zawhmm-work: pardon my ignorance, i'm still reading through the plethora of docs and howtos and wikis. i've just set up asterisk and got it to startup today.
01:53.05hmm-workhttp://www.voip-info.org/tiki-index.php
01:53.13hmm-workabout halfway down the page
01:53.14*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
01:53.56Zawthanks, looks like i want a Media Gateway
01:54.11hmm-workwhat?
01:54.13ZawPSTN Gateway
01:54.30*** join/#asterisk sivana (~sivana@165.154.13.35)
01:54.34Zawhmm
01:55.19hmm-worka pstn gateway would be considered a single piece of a equipment you would plug into the pstn get a call off/on the voip network to/from the pstn
01:55.26hmm-workto most of us in here anyway
01:56.05Zawok. i was under the imperssion that i could somehow pass outbound calls to a 'gateway' from my asterisk pbx server for outbound calls, as well as have the same 'gateway' route incoming calls to my asterisk pbx.
01:56.10*** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com)
01:57.16Juggieyou can provided it speaks a protocol asterisk understands
01:57.39hmm-workdepends on how you define 'a gateway' seem to be defining it as a single logical entity that will pass calls to/from anywhere on the pstn network
01:57.49hmm-workwhich would be a voip service provider
02:01.13Zawok. it seems that i have my terminology mixed up.
02:01.35Zawi'm looking for a voip service provider, not a gateway :)
02:01.39Zawany recommendations?
02:02.13scrubbZaw: broadvoice.com
02:02.58shepherdvoicepulse.com
02:03.34Zawwith these voip service providers, i can have an asterisk server configured with voip phones/mailboxes, etc. and just use the service provider for incoming/outgoing 'public' calls right?
02:03.52hmm-workZaw: for the most part yes
02:04.07Zawok
02:04.14hmm-worksome are picky about what you use to access their network, most are not
02:05.01Zawso some just allow the use of a voip phone, while others allow you to connect an asterisk server instead of a voip phone?
02:05.27denonor of course, some require their proprietarily-configured hardware
02:05.46Zawhowdy denon :)
02:05.53denonhey zaw
02:06.05scrubba lot of national ones require proprietary stuff.
02:06.10denonjust saw your q .. just checking my email and settling in after a long weekend out of town
02:06.27denonfor obvious reasons, it's much easier to maintain a network of consistent and managed gear ..
02:06.35denonless idiots botching up configs :)
02:06.39Zawhehe
02:06.46scrubbthere are only a few that allow * boxen.
02:07.05Zawwell, i suppose i'll be needing one that allows an asterisk box
02:07.17loudzaw, are you in the west coast ?
02:07.30scrubbZaw: also what protocol and what number coverage do you need.
02:07.40hmm-workanyone really familiar with the asterisk manager interface?
02:07.41scrubbyou need 911?
02:07.45Zawi'm in the east coast, pittsburgh PA
02:07.52loudi see
02:08.10Zawscrubb: i'd like to find one with the 412 area code
02:08.43scrubbzaw, check out www.broadvoice.com and see if they cover it.  or others that folks have mentioned.
02:08.48Zawscrubb: and yeah, 911 would be nice. i was wondering if i could just configure my asterisk server to dial a 412 or 724 local emergency number if 911 was dialed, but i don't know how that works, haven't read that far yet.
02:09.15*** join/#asterisk klasstek (~nunyobiz@c-24-9-148-246.client.comcast.net)
02:09.24Sedoroxexten => _911,1,Dial(ZAP/4125551212)
02:09.32Sedoroxor minus the _
02:10.06ZawSedorox: that's assuming that i have a zaptel card internal to my pbx, though, right?
02:10.44*** join/#asterisk harryvetch (~noyb@S010600055d210201.vs.shawcable.net)
02:10.57Sedoroxwell yea.. you can do Dial(SIP/privder/number)
02:10.57Sedoroxtoo
02:11.08scrubbnot many have 911 figured out.
02:11.09Sedoroxit doesn't matter.. just using it as a example of what youw ould do for 911...
02:11.17Zawok
02:11.23scrubbjust the big ones, and they wont let you do * boxen.
02:11.32Sedoroxand I think you want to set callerid before it dials out.. so they have your nuber
02:12.17*** topic/#asterisk by drumkilla -> Asterisk: The Open Source PBX || 1.0.6 Released || Dev Conf 1PM CST MARCH 3rd -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf June 8-10th more coming soon....
02:13.13*** join/#asterisk SimonR (~SimonR@64.56.237.14)
02:13.20Sedoroxhmmm
02:13.33harryvetchI may have come across a bug. For some reason other machine the xlite would not log in. Checked all the settings looked good. Stoped and started asterisk did not make a difference. changed the username/passowrd in sip.conf/xlite/extensions.conf to something else. Reloaded asterisk. Did not work. I changed profiles in this windows box to wifes profile now the xlite is loging in sucsessfully to the original usename/pass that was just ch
02:13.41Sedoroxand I just redid a box to 1.0.5.. damnb
02:14.02Juggiegrr :P
02:14.09mtqhharryvetch: sip debug?????
02:14.12harryvetchmabey the old usernam/password info is stuck in memory and was not released?
02:14.25Chuji~softphones
02:14.28Juggieanyone have a list of changes for 1.0.6
02:14.33Sedoroxchangelog?
02:14.35Sedoroxhehe
02:14.47Chuji~softphone
02:14.48jbotsomething that should be drug out into the street and shot
02:14.58Chujixlite ^^^^^^^^^^
02:15.01Chuji:)
02:15.07harryvetchchuji, xlite has been farily reliable with no issues..
02:15.20RoyK~lart Chuji
02:15.39harryvetchbrb
02:15.50*** join/#asterisk daved (~daved@c-24-98-109-138.atl.client2.attbi.com)
02:17.41Juggie1.0.6 gives me something to do tomorow, upgrade our two servers
02:18.04davedwhy would you put a conf in the middle of a working day :(
02:18.07Sedoroxare there change logs anywhere?
02:18.44puzzledcheck asterisk-cvs list
02:19.02fileyay 1.0.6
02:19.53mishehuany changes between cvs 2/24/2005 and 1.0.6 ?
02:20.06fileyes.
02:20.09puzzledcheck asterisk-cvs list
02:20.20mishehualright.
02:20.32fileand omg people, I did the 1.0.6 changelog
02:20.35fileand I did it in detail
02:20.38fileso nobody complain
02:20.43fileor else you die
02:20.47Juggiei found it on the ftp its good :)
02:20.48puzzledfile: it looks nice. thanks
02:20.52davedman that changelog sucks
02:21.00filedaved: no asterisk for you!
02:21.10Juggiei just wish ftp.asterisk.org was mirrored on a http server.
02:21.17*** join/#asterisk Gronker (~Gronker2@adsl-217-248-205.ags.bellsouth.net)
02:21.25Juggieno ftp @ work.. would make my life easier.
02:21.28mishehufile: you didn't do it in klingon!  ;-)
02:21.45mishehudaved: come again, next year.
02:21.46filemishehu: oh well
02:21.48ChujiI don't see the changelog on -cvs When was it posted?
02:21.49filenext time
02:22.24JerJer[mobile]just so everybody knows switch-1 will be going down for a couple minutes
02:22.33JerJer[mobile]here in about 30 minutes
02:22.50fileJerJer is replacing it with a toaster running asterisk
02:23.08mishehuthat's almost as good as the curling iron running asterisk
02:23.38JerJer[mobile]i've hacked the kitchen sink to run asterisk
02:24.04Chujiprolly got h323 only too huh?
02:24.42mishehuJerJer[mobile]: hope that sink has a garbage disposal for telemarketer calls
02:24.56JerJer[mobile]nope H.323 isn't compatiable with the dual drainage and disposal system
02:25.03JerJer[mobile]:)
02:25.09mishehuthat's why I don't like h323
02:25.31scrubbevening Mark.  Not glued to the Oscars?
02:25.37mishehuwhere's bkw so I can tell him it's a shame he didn't pick dates for cluecon about a month ago...
02:25.58mishehufirst asterisk event happening in chicago that I know of, and I'll be overseas then.  :-/
02:26.21goatmilk:'(
02:26.42*** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net)
02:26.46mishehugoatmilk: uhm...   *points at nose*  I think you have something hanging there...
02:26.55goatmilkmishehu: that's my tear for you
02:27.20goatmilki'm just so heart broken that you're gone!
02:27.53mishehugoatmilk: are you in chicago area?
02:28.21goatmilkprobably within 12 hours of it
02:28.32ChujiDid they move the Chicago thing to a weekend yet?
02:28.37mishehutwelve hours by foot, car, train, plane, spaceship?
02:28.38mishehuheh
02:28.50goatmilkcar, most likely
02:28.52mishehuChuji: if they could move it to the end of may, I can go...
02:29.03goatmilki could probably get there by plane in an hour :)
02:29.04mishehugoatmilk: that's a hell of a schlep then.
02:30.14harryvvI found the problem. Context issue.
02:31.08scrubbI have a lot of trouble with context.  Most of the time, I'm just out of it.
02:31.48harryvvscrubb yea well this was just a oversite.
02:32.00harryvvgoing to spend some time with the wife and cook din.
02:32.01harryvvsee ya]
02:34.25*** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net)
02:38.16nestArhttp://bbs.zuwharrie.com/content?topic=15545.new;topicseen#new
02:38.23shido6boink
02:38.24nestArhttp://bbs.zuwharrie.com/content?topic=15545.new;topicseen#new
02:38.30nestAroops
02:38.32nestArsorry
02:38.36nestArleaned on the mouse button
02:38.43nestArdamn laptop
02:39.03goatmilkcould have been a worse url.
02:39.53nestArindeed
02:40.03nestArtubgirl comes to mind
02:40.21*** join/#asterisk Rick_Hunter (~rhunter@01-204.008.popsite.net)
02:43.19mishehufichs!
02:43.57mishehutubgirl on a date with mr. goatse.cx
02:44.07mishehuthat's the worst that can come to mind.
02:51.24JerJer[mobile]time for some fun.... here we go
02:57.50Nuggetwhenever I accidently find myself at tubgirl I reflexively hit back so many times that I end up running ncsa mosaic.
02:59.42moonwickheh.
03:00.29goatmilkit's obvious that you've never used a web browser if you think the back button changes what brower you're using.  gosh.
03:00.58*** part/#asterisk Moc (~Moc@modemcable212.49-80-70.mc.videotron.ca)
03:01.04goatmilk:)
03:01.28goatmilkNugget: your joke actually made me laugh out loud, and forced to think for a few moments.
03:02.03*** join/#asterisk Moc (~Moc@modemcable212.49-80-70.mc.videotron.ca)
03:02.35Juggiemosaic was when the internet was cool
03:02.45Juggieback in the days of trumpet winsock :)
03:02.59stepcutheh
03:03.20stepcuti remember trumpet
03:03.21Juggiebefore efnet split, etc.
03:03.42stepcutbefore spam filters
03:03.50Juggiebefore there was spam
03:04.04Juggiewhen not just any idiot was on the internet
03:04.17stepcutjust special kinds of idiots
03:04.21sivanathere's no more EFnet?
03:04.40Juggieno there was efnet
03:04.48Juggiebut it used to be 100+ servers
03:04.52Juggiethen it split into two networks
03:05.07sivanaya.. I always hung out on DALnet
03:05.08stepcutsivana: there used to be *only* efnet
03:05.11sivanayup
03:05.24sivanaI thought maybe you meant they aren't around anymore
03:05.40PatrickDKI remember the says of packet drivers
03:05.49Juggiehttp://www.ircnet.org/History/veggen1.html
03:06.45stepcutremember fido mail (?? is that even what it was called)
03:06.45stepcutinter-bbs mail?
03:06.46PatrickDKheh, I loved fido
03:06.51sivanalol.. I ran a WWIV BBS
03:07.10tzangerWWIV... ewwwwww
03:07.15tzangertelegard baby
03:07.17sivanahehe
03:07.36PatrickDKheh, I just wrote myown, ran in 286 protected mode
03:07.55stepcutI remember it taking several *minutes* to decode girlie jpg's on my 286 :)
03:07.57PatrickDK37k lines of code :(
03:09.00stepcutoh! and zmodem -- simultaneous upload/download
03:09.15stepcutand, of course, tradewars 2002
03:09.28Juggiei ran a bbs for 3 ish years too, was on fidonet et
03:09.31Juggie*etc
03:10.35stepcutI remember when I was running DOS and I was like, 'it would be cool if I could have two monitors and keyboards', and this older cool guy was like "I can do that on my computer -- it runs linux"
03:10.42stepcutbut I only had a crappy 286
03:11.00PatrickDKstepcut, I did that in dos, dual monitor
03:11.00stepcutit blew my mind
03:11.10PatrickDKit was either two pci, or isa + monocrome
03:11.20PatrickDKI did isa + monocrome for programming
03:11.21stepcutPatrickDK: right, but I wanted multitasking
03:11.24sivanaI had an 8086.. Tandy 1000
03:11.25sivanaheh
03:11.25PatrickDKdebug in monocrom
03:11.32PatrickDKstepcut, deskview :)
03:12.19PatrickDKjust think, memory in the 886 was 150ms
03:12.23stepcutI remember knowing exactly how many bytes of RAM I should have free after DOS booted
03:12.25PatrickDKand now is below 6ns
03:12.25sivanahehe
03:12.54sivanaI was happy with my dual 5 1/4" drives.. hehe
03:12.59sivanaI can copy from one to the other :)
03:13.15PatrickDKhell, my 8080 clone (z80) had dual 5 1/4" drives
03:13.22stepcutmine had one 5 1/4" and one 3 1/2"
03:13.30stepcutand a 'turbo' button ;)
03:13.35sivanalol
03:13.47stepcutin case things where running to fast, you could turn turbo off
03:13.50PatrickDKI still have some original 10meg mfm ibm drives, from the 8086
03:16.15*** join/#asterisk dersteer (~travis@24.231.151.204.gha.mi.chartermi.net)
03:19.04techieI still have my comodore64.
03:21.29ManxPowerDon't you guys throw anything away?
03:21.58sivanahehe
03:22.25techieit's a piece of history
03:22.43Goshenis anyone else having problem with voipjet? It is accepting the call then hanging right up
03:23.58techieGoshen: no
03:27.15Goshen-- Hungup 'IAX2/voipjet/1'
03:27.15Goshen<PROTECTED>
03:27.19Goshenright after I place the call
03:27.31*** join/#asterisk JunK-Y (junk-b@Toronto-HSE-ppp3747471.sympatico.ca)
03:27.45JunK-Ylo
03:28.32Goshen1-800 number went ok
03:28.33*** join/#asterisk dersteer (~travis@24.231.151.204.gha.mi.chartermi.net)
03:29.42ariel_Goshen, Voipjet says that they don't support 800 numbers at least if they go through your still charged for the call.
03:29.59Goshenyea, I was just saying it placed the call to 1-800 ok
03:30.06Goshenbut it won't place the call to my landline tonight
03:30.08Goshenworked fine last night
03:31.43file64what's the number?
03:31.48Goshenpm
03:32.59file64mmm nice quality
03:33.28ariel_Goshen, worked fine just now. But there caller ID is still messed up.
03:33.55*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
03:33.55*** mode/#asterisk [+o bkw_] by ChanServ
03:34.11bkw_file with 64bis
03:34.12bkw_er bits
03:34.13bkw_now nice
03:34.15Goshenyea that was really nice quality file64
03:34.16file64indeed
03:34.23Goshenit must be something on my end then
03:34.41file64could be
03:34.47file64bkw_, this workstations r0x0rs my s0x0rs
03:34.52bkw_oh really
03:34.55bkw_I can too ya know
03:35.02file64ooh baby
03:35.05bkw_;)
03:35.07file64touch my asterisk key!
03:35.13bkw_no no
03:35.17file64you rebel you
03:35.18bkw_pound my * key
03:35.23bkw_har har har
03:35.27file64ooh don't wanna take it slow?
03:35.42bkw_haha
03:35.44file64oh well, we're gonna have fun in a week *G*
03:35.49bkw_yep
03:35.53bkw_FUN FUN FUN
03:35.56bkw_I need jor info boi
03:35.57file64FUNNNNNNNNNNNNNNN
03:36.14bkw_msg me jor cell
03:36.15file64my cell won't work in your funky country
03:36.20bkw_yes it will
03:36.24file64no, it won't
03:36.26bkw_its GSM?
03:36.30file64no, it's not
03:36.33bkw_oh
03:36.35bkw_sucks to be jew
03:36.38file64yes
03:36.46file64which is why I asked for yours :p
03:36.54PatrickDKcdma? tdma?
03:37.01bkw_its in canada
03:37.01file64it's CDMA but it's prepaid
03:37.05file64and yes, in Canada
03:37.05bkw_oh that sucks
03:37.07PatrickDKyuk
03:37.36file64haha
03:37.51PatrickDKmay be old, but damn can it get signal :)
03:38.13file64I bet
03:38.14bkw_ya you can't be an adult at 18 in canada
03:38.17bkw_what a fucked up country
03:38.18*** join/#asterisk xeet3 (~joe@gw1.istx.net)
03:38.24file64well, in the provinces around here you can't
03:38.38file64I *could* get setup for prepaid in the US before I leave and program my phone *maybe*
03:38.47xeet3can a zaptel tdm card do distinctive ringing?
03:38.55bkw_the X101p can
03:39.00bkw_not sure about the TDM
03:39.07xeet3I mean for outbound calls, going to analog phones
03:39.18bkw_ya
03:39.20bkw_you need to go read boi
03:39.29xeet3k =)
03:39.48PatrickDKdamn, I'm so bored of programming this website
03:39.51file64okay bkw, what's a CDMA based provider in the US that will allow me to take my own phone and use a package for setup? eh? EH?
03:39.55file64oh wait, I do have a GSM phone
03:40.42Juggiecdma sucks
03:40.55file64okay people, help me!
03:41.55Juggiefile64 where are you going?
03:42.00bkw_bbl
03:42.05file64California
03:42.06elricis it possible to group users so that when a call comes in on a certain line all the extensions in the that group ring?
03:42.26Juggiefile64, why not just roam how long are u there?
03:42.32file64I can't
03:42.39*** join/#asterisk _JSKCR_ (~jskcr_@c-65-34-143-82.se.client2.attbi.com)
03:42.42file64my phone is prepaid because I'm not of legal age in Canada to get on a plan
03:42.44file64so it won't roam
03:42.50Juggieoh
03:42.55Juggiedont u have a gsm phone/
03:42.55JunK-Yelric: just use Dial with many tech/extension
03:43.00file64yes
03:43.06xeet3elric: look at phone queue's too
03:43.08file64I have both a GSM and a CDMA...
03:43.11Juggiewell then just get a gsm card in california
03:43.12*** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com)
03:43.20xeet3elric: you can set it to ringall
03:43.28Juggiewalk into verizon, or t-mobile and ask for a gsm card
03:43.30Juggieits like 20$
03:43.34Juggieand then pick a plan
03:43.38Juggiepre paid, or month to month whatever
03:43.41elricxeet3, i only want certain extensions to ring.
03:43.43_JSKCR_hy all
03:43.48file64yeah month to month even though I'm only there for 6 days
03:43.48file64HA
03:43.56xeet3xeet3:  yes, you can do that, ring all extensions listed as part of a queue
03:43.58file64hrm
03:44.02xeet3er, bleh
03:44.02bjohnsonmight still be cheaper
03:44.03xeet3elric
03:44.09Juggiewell just get aloan of a friends phone
03:44.16file64I have an idea
03:44.22file64but your thoughts are appreciated, thx
03:44.26elricJunK-Y, thats an option but then I have to edit extensions.conf anytime i add a new person to that group.
03:44.33Juggiefile64 why dont u get a real phone plan
03:44.41Juggiei've never heard of being 18 and not being able to get a phone
03:44.41file64where?
03:44.50file64have to be 19 here
03:44.50Juggiewho are you with?
03:44.54xeet3elric:  you can also just do an & between channels you want to dial  ex:  Dial(Zap/1-1&Zap/2-1&Zap/3-1)
03:44.55Juggiewhere?
03:44.57file64I can't get credit card, I can't live on my own, can't do any of that
03:44.58bjohnsonroaming is a real killer anyway
03:45.01file64New Brunswick
03:45.06PatrickDKheh, I got my phone at 16
03:45.07bjohnsonhe's better to get a phone there
03:45.10PatrickDKno credit
03:45.17elricxeet3, ah
03:45.36Juggiefile64 people move away for univ @ 17 years old
03:45.44Juggiei highly doubt u cant live on your own
03:45.49*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
03:45.59file64the whole reason is due to credit and contract
03:46.09file64at my age I can't really sign a contract... thus can't sign a cellular contract
03:46.21file64so they won't sell, unless you put down a huge deposit
03:46.27Juggieyou can legally sign a contract @ 18
03:46.30Juggiei'm fairly sure
03:46.44Juggieanyways, get your parents to buy you one month of rogers
03:46.46Juggieon your gsm phone
03:46.50Juggieand cancle after
03:46.50xeet3yes, you can sign a contract at 18
03:47.03ManxPowerContract law varies by country
03:47.04xeet3you can also get prepaid at like 5 years old if you have the money
03:47.06file64ha too late now, not too important
03:47.18*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
03:47.23xeet3yes, and canada and the us are 18
03:47.31file64parts of Canada are 18
03:47.32Juggiego without your cell for 6 days its not big deal :p
03:47.46PatrickDKor get a cheap throwaway cell here
03:47.51PatrickDKand forward you number from that one
03:47.51file64Juggie, it's not that - I'm meeting up with bkw and twisted, having a cell would make things easier
03:48.03PatrickDKor just use the new number
03:48.05file64but alas
03:48.06*** join/#asterisk NormAst (~NormAst@Ottawa-HSE-ppp4120806.sympatico.ca)
03:48.18neopheror live like a hermit, hehe
03:48.22Juggiefile64, you have a gsm phone.... get a plan with rogers. or buy a pre paid card in cali
03:48.27file64http://canada.justice.gc.ca/en/ps/sup/steps/s2c.html
03:48.30bjohnsonfile64: get someone there to get you a gsm sim card setup for a verison inpulse plan
03:48.35ManxPowerprepaid is a ripoff
03:48.36file64voila, age of majority
03:48.45*** part/#asterisk JunK-Y (junk-b@Toronto-HSE-ppp3747471.sympatico.ca)
03:49.03ManxPowerat 18 the problem isn't usually with signing a contract, it's the lack of credit history
03:49.09elricis it possible to write a script that would parse a text file for extensions and pass them to * to ring?
03:49.18Juggiefile64, just get your damn parents to get you a plan for one month.... or just stop bitching ;)
03:49.30Juggieor call your prepaid and make arangements to roam.
03:49.33file64Juggie, I did stop bitching :)
03:49.37file64I even said, "thanks"
03:49.49bjohnsonroaming would still be costly
03:49.56Juggieits not that bad
03:49.57bjohnsontaking a CDN phone to CA
03:50.03Juggieespically if the phone is only for just in case
03:50.06Juggienot long conversations
03:50.38ManxPowerJust don't use a cell phone when in california.
03:51.07bjohnsonManxPower: that would be the cheapest
03:51.22ManxPowerI could not really use my cell phone when I was in .ca
03:53.06Juggieget your parents to get you a real phone :)
03:53.16file64oh just shutup :p
03:53.20ManxPowerI discovered none of the 3 tollfrees I have from 3 providers would work from canada as well.  LOL!
03:53.44Juggiemy cell plan is $18+system access fee a month for 250weekday minutes, and unlimited weekends
03:53.48Juggiegov plan, gotta love it
03:53.52SedoroxManxPower: what provider?
03:53.54file64shutup shutup shutup
03:53.56*** join/#asterisk ikey (~kirankuma@220.226.24.163)
04:01.04brc_~seen kpfleming
04:01.07jbotkpfleming <~chatzilla@ip68-3-230-141.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 3d 1h 28m 7s ago, saying: 'Grooby: if you are using HEAD, add 'o' to the Dial string to get the CID to go back to the way it's supposed to be'.
04:05.45datareactorIS THERE any fax over ip peering network for asterisk
04:06.01shido6do u want one, datareactor?
04:07.17_JSKCR_Juggie, my cell phone plan is 40 per month unlimited long distance and local
04:08.21pardata reactor faxing images?
04:08.45parfaxing ascii is easy
04:09.00fileQUIET
04:09.03pari should make something to allow it through asterix
04:09.06fileI said quiet, and when I want quiet I will get quiet
04:09.11fileso back to the topic at hand - asterisk
04:09.48PatrickDKhmm?
04:10.18Juggie_JSKCR_, thats 40$ us but not bad :)
04:10.30fileJuggie: shhhhhhhhhhhh
04:10.42Juggiemine works out to like 25$ cdn which is <20$ us
04:10.53fileshhhhhhhhhhhhhh
04:10.59filedon't make me get out the tranquilizer darts
04:11.16PatrickDKyour just upset your not 64bit anymore
04:11.37fileif I went back to my workstation I would be!
04:13.30datareactorshido6 , par how can i do that ?
04:15.24mishehubah.
04:15.47datareactorpar i only want to text faxing
04:16.01pardatareactor, you can use my website..
04:16.10parwww.faxpad.com
04:16.24parit uses the tpc network
04:17.28datareactorthanks i am looking in to it
04:17.32pari would like to maybe create a tpc fax network module for us with asterix
04:18.29filedoes tpc still work?
04:19.05datareactordont know about tpc network ?
04:19.36paryeah tpc still works.
04:19.41*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
04:19.48filelast I checked the local site disappeared
04:20.57parhttp://wol.ra.phy.cam.ac.uk/FaxMail/
04:21.03|Vulture|how big is iax.cc?
04:21.10parfaxmail can be used to interface with asterix
04:21.58parunfortunately you would be required to scan the original document to be faxed.. and convert it into postscript..
04:22.08paryou could automate that entire process though..
04:22.26parespecially if you have one of those all in one scanner printer fax machines.
04:22.49PatrickDKscan in, convert tiff g3 to ps
04:23.51paryes siree ;)
04:24.05*** join/#asterisk viLeR (1000@ip-33-104.telesat.com.co)
04:25.47sivanaPatrickDK: are those special sound files for the CF stuff
04:25.52parany developer willing to code in support for tpc remote printing?
04:27.10datareactorpar how you do fax billing ?
04:27.12PatrickDKspecial sound files?
04:27.19PatrickDKthose aren't special
04:27.25PatrickDKthey are in the asterisk-sounds package
04:27.40PatrickDKor used to be
04:27.45sivanaok
04:27.55pardatareactor: the tpc network is entirely free
04:28.01*** join/#asterisk doolph (doolph@200.46.148.46)
04:28.03doolphhello
04:28.18parit just puts a small local ad blurb at the bottom of the fax.
04:28.40doolphanyone is looking for a-z termination?
04:28.55datareactorpar if my country is not mentioned can i setup my own gateway for that
04:29.11pardatareactor: you would have to contact and be approved by the tpc.
04:29.36datareactorpar OK
04:30.22datareactorpar we want to setup fax over payable services for our customers what do u recommend so we can bill them
04:31.01sivanaPatrickDK: ya, they're there
04:31.03parthere are many services available over the internet if you do not choose to go with a free ad supported system
04:32.57datareactorpar can you  name a few ?
04:33.20*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-11-74.d4.club-internet.fr)
04:33.39parthere is xpedite's service
04:35.48parcastelle's faxpress
04:37.54datareactorpar thanks
04:39.58pardatareactor: also www.integram.com
04:40.30parsupporting signalling processing for fax would really be great for asterisk
04:41.03parbut it would be really server intensive for it.
04:41.23*** join/#asterisk viLeR (1000@ip-33-104.telesat.com.co)
04:42.21pari agree that some kind of really good preprocessor package needs to be created to do the job.
04:43.48pardatareactor: for hardware look into the FaxFree Portal 500 by Tac Systems
04:44.04datareactorpar thanks too much
04:44.08parit connects to a regular fax machine to provide IP telephony.
04:46.00parused to be atfax.com
04:46.06datareactorpar i there any foip faxing card which i put in asterisk so my user can do faxes world wide
04:46.53sivana~voip
04:46.54jbotsomebody said voip was Voice over IP
04:46.57sivana~foip
04:47.15paryeah fax over ip
04:47.22*** part/#asterisk GreyFoxx (greg@out.of.phaze.org)
04:47.25pari dunno of any card..
04:47.45parlast i heard of was the @fax portal
04:49.45parwait there is multitech
04:50.16parmultitech - multivoip voice/fax over ip gateway fxs fxo
04:51.34datareactorpar thanks again
04:51.34parhttp://www.c-source.com/csource/newsite/ttechnote.asp?part_no=MVP200
04:51.37paryou can buy it there
04:54.13parheh from the asterisk list
04:54.14parI try to use IConnect on my MultiTech MVP 200 VoIP Gateway and didn't
04:54.14parwork:-(.
04:54.14parI try thru my asterisk box and everything works fine.
04:54.14parThe MVP200 is behind the Nat and my * is connected directly on Internet
04:54.14parexactly like IConnect.
04:57.33datareactorpar is there any error on MVP 200
04:58.44parit uses T.38 protocol
04:59.56*** join/#asterisk TheEmperor (~mattn@203.121.47.100)
05:00.54parthere is currently a bounty on asterisk for SIP T.38 fax gateway support.
05:01.21pardatareactor: your best bet is to join in on the bounty
05:01.28pardatareactor: http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38%20Bounty
05:03.18pari am still curious if there are any programmers willing to do it
05:04.31pardatareactor: for the temporary solution your best bet is to use the multitech device through the asterix server
05:04.39Juggiei wish someone with the knowhow could write a asterisk module for minet (mitel's protocol)
05:04.41`SauronApparently it's hard enough to where $8250 isn't worth it.
05:04.46datareactorthanks par
05:04.54`SauronT.38, that is
05:04.56Juggieits based of h323, and i think i can get protocol docs.
05:05.23TrepaliumITU docs can be hard to come by, or merely just expensive.
05:05.23*** join/#asterisk NTJOCK (~brian@txshirts.com)
05:05.26NTJOCKhello
05:05.28parhey big money guy :)
05:05.47NTJOCKbig money?
05:05.48NTJOCKwho me?
05:05.57NTJOCKhehe
05:06.13parno.. juggie :-)
05:06.55Juggiepar, no?
05:07.00NTJOCKHey, I've got a FXS port that isn't giving dialtone.... I've checked zaptel and zapata and it should be working....
05:07.15NTJOCKit's in with port 6 and 7 (same context) and they work... but 5 won't give dialtone
05:07.27TheEmperorhello, is it ok to install * on debian base with 2.6 kernal?
05:07.31NTJOCKyes
05:07.39NTJOCKEmperor: it works fine
05:07.54TheEmperorNTJOCK: Ok, am going to do a new installation, so I wanted to use that :)
05:07.57NTJOCKuse the CVS version, avoid packages like the bubonic plague
05:08.10NTJOCKyou'll need some goodies from packages....
05:08.14TheEmperorNTJOCK: Compile from source you mean?
05:08.17Juggiedont use the CVS if you are running production
05:08.19NTJOCKbut zaptel, libpri, and asterisk you should go to CVS
05:08.30Juggieuse a release ver
05:08.35NTJOCKYou can get release from CVS
05:08.48TheEmperorJuggie: what doy ou mean if I am running production?
05:08.48NTJOCKI had hell with packages and mismatched modules
05:08.49Juggieyes but CVS implies latest and greatest.
05:09.00TheEmperorCVS means compile from source?
05:09.10NTJOCKno CVS is a way of getting the source code
05:09.17TrepaliumDownloading a release version from CVS is generally slower than just getting the tarball.
05:09.18NTJOCKpackages are a precompiled set
05:09.26TheEmperoroh..
05:09.32JuggieTheEmperor, if your install is going to be running actual services which need to be depended on
05:09.35Juggiethen run a stable version
05:09.45TheEmperorJuggie: which is a stable version?
05:09.52NTJOCKyour mileage may vary... I wasn't able to call it "working" until I went and fetchd the CVS version
05:09.52Juggie1.0.6 came out today
05:09.57NTJOCKit's been stable ever since
05:10.08Juggieftp.asterisk.org
05:10.15TheEmperorJuggie: 1.0.6 on a debian base with kernel 2.6 is stable?
05:10.26NTJOCKanyhow... any ideas on troubleshooting a FXS port?
05:10.33Juggie1.0.5 on fedora core2/3 with 2.6 is perfect
05:10.39Juggieso debian should be fine as well
05:10.51TheEmperorJuggie: super! :)
05:11.07Juggiefedora core3 required a few tricks to get it running
05:11.11Juggiebecause of SELinux
05:11.21TheEmperorJuggie: want o install debian as I don't like rpm dependencies..
05:11.39JuggieTheEmperor, you can do that just dont install asterisk from RPM
05:11.39TrepaliumI had some problems with FC3 with it's firewall, too.  Messed up my VLANs.
05:11.52*** join/#asterisk herag (herag@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net)
05:11.59TheEmperorJuggie: how to install then? from CVS like you said?
05:12.17NTJOCKAnyone here running a FXS port adapter with a fax machine or credit card machine?
05:12.22JuggieTheEmperor, compile from source..... like i said if u feel edgy, follow instructions on www.asterisk.org to get the latest cvs code
05:12.27Juggieif you want to run a tested release
05:12.30TheEmperorJuggie, ok will do.
05:12.34Juggieftp to ftp.asterisk.org
05:12.37Juggieand download 1.0.6
05:12.54TrepaliumSince Debian stable will not likely have a recent version of asterisk, and unstable is a little too breakage prone for a production server, compiling from source is pretty much the only way.
05:13.18Juggieyou can allways build your own packages :)
05:13.24paris there a changelog for 1.0.6?
05:13.27TheEmperorthanks guys, I'll do that. 1.0.6 and compile from source..
05:13.32Juggiepar, on the ftp
05:13.36parcool thx
05:13.52TheEmperorhow about the other packages/addons?
05:14.06JuggieTheEmperor, u can get them all from the cvs or the ftp
05:14.15TheEmperorJuggie,sweet. Thanks :)
05:14.17Juggiego to www.voip-info.org
05:14.18pari will probably switchover faxpad.com to use astersik as soon as t.38 gateway support is added to it
05:14.24Juggietheres probally an install doc on there
05:14.31Juggieon what to download and how to install etc.
05:14.33TheEmperorok
05:14.43`Sauronpar: I wouldn't hold my breath.
05:15.03TrepaliumFigures...  Just finished building RPM packages for FC3 for 1.0.5, and now 1.0.6 is out.
05:15.11*** join/#asterisk rodizump (~chatzilla@dsl-213-023-218-080.arcor-ip.net)
05:15.24rodizumpHi everyone
05:15.37NTJOCKanyone have any experience with the grandstream handytone FXS adapter or the Sipura SPA-1000?
05:15.44`Sauronwhat about the spa1k?
05:16.03NTJOCKis it bulletproof?
05:16.18rodizumpDoes anyone know where to get SIPGetHeader application for * ? but without using chan_sip2 ? anyone, please
05:16.18`SauronBased on what criteria?
05:16.18NTJOCKI need to put a send-only fax machine at my bookeepers remote office
05:16.32NTJOCKjust want to make sure it works
05:16.45NTJOCKI like the 841 phones we have....
05:16.52`SauronI would strongly recommend against doing fax over voip
05:16.59NTJOCKalso would like to get rid of a cord running around the office for our credit card terminal
05:17.25`SauronShrug. The spa1k works pretty well
05:17.41NTJOCKwhat is your objection on fax and asterisk?
05:17.42parsauron.. if each end supports t.38 it should be ok
05:17.54NTJOCKIt's an internal network call
05:17.59`Sauronpar: I'm saying I wouldn't hold my breath on the * T.38 support
05:18.01neopheranyone know how i can tell if i have libtiff installed and what version it is
05:18.11JuggieTheEmperor, http://www.voip-info.org/wiki-Asterisk+installation+tips
05:18.17EssobiSauron Coppice is already coding it.
05:18.21dazza_par: but faxpad.com just seems to use tpc.int
05:18.39NTJOCKgood bandwidth between both points, using a FXO line at the * box....
05:18.41EssobiT.38 for * is inevitable.
05:18.44parsauron: sure but an end station behind the asterisk connecting to an endpoint with t.38 support should be fine right?
05:18.45NTJOCKjust using VOIP for transport
05:18.45dazza_par: how would t.38 be of use?
05:18.46`SauronEssobi: Coppice?
05:18.56EssobiThe spandsp author.
05:18.56dazza_Steve Underwood
05:19.06pardazza: yeah i'm using tpc now
05:19.08EssobiSteve.  He's one sharp guy.
05:19.13`SauronAh.
05:19.35parbut i would increase the calling area by setting it to be handled through t.38 gateways
05:19.41*** join/#asterisk sudhir492 (~sudhir@4.7.59.232)
05:19.44sudhir492hi all
05:19.49TheEmperorJuggie, some people say kernel 2.4 more stable than 2.6, what do you recommend?
05:19.50dazza_par: Spotted that. Arlington Hewes mean anything to you?
05:19.59sudhir492is cvs.digium.com down?
05:20.08dazza_par: t.38 gateways would not be public ...
05:20.20JuggieTheEmperor, i have had no problems with 2.6 however i have not really tested it under a heavy load
05:20.21pardazza: right
05:20.41TheEmperorJuggie: yeah, cause this will be for production..we might be safer with 2.4..
05:21.21pardazza: arlington hewes is the tpc administrator
05:21.27`SauronHum. * survived a cvs update
05:21.46dazza_par: That's me.
05:21.52parhi hi
05:21.57dazza_ho ho
05:22.04JuggieTheEmperor, i am going production, 2.6 has shown no problems
05:22.14parhow goes the winding serpent :-)
05:22.35TrepaliumArgh..  The * manager API is so... bizarre.  I can't figure out how to get callerID info from it.
05:22.47TheEmperorJuggie: you were mentioning you haven't tested on heavy load? I am worried about that
05:22.53twistedwho the hell said my name in here?
05:23.12twistedi got a * next to this window
05:23.22`SauronDunno. What is your name? :)
05:23.35twisteduhh.. twisted, maybe?
05:23.43`SauronEh, nevermind.
05:24.13*** join/#asterisk zignig (~simon@203.217.15.10)
05:24.15`SauronMaybe someone casually mentioned the word twisted as part of normal conversation
05:24.31`Sauronsuch as, man - that's a twisted thought
05:24.40twistednah
05:24.55twistedpeople know better
05:24.56twisted:P
05:24.56zignighuzzah , got a voip call between wireless laptops going... ;)
05:25.35Juggiebush sure is twisted
05:25.47sudhir492is there a back for cvs.digium.com?
05:25.51sudhir492backup
05:26.05JuggieTheEmperor, if you have problems you can allways rebuild on 2.4 but it will be fine either way..... asterisk doesnt care if your kernel is 2.4 or 2.6
05:26.09twistedsudhir492, cvs.digium.com is a round-robbin dns.. there are multiple servers
05:26.19`Sauroncvs.digium.com isn't down
05:26.22`SauronI just updated from it.
05:26.25TheEmperorJuggie: ok, thanks :)
05:26.41`SauronWell, whichever one I connected to... :)
05:26.51sudhir492twisted: hmm. for some reason, I get the error "connect to cvs.digium.com(66.225.202.81):2401 failed: No route to host"
05:26.53sudhir492today
05:26.53TheEmperorcan I ask, in my sip settings, i don't want an extension to expire, how do i set that?
05:27.16`SauronTry connecting to the other server
05:28.06rodizumpDoes anyone know where to get SIPGetHeader application for * ? but without using chan_sip2 ? anyone, please
05:28.48Juggietwisted, whats it worth money wise for someone to implement minet (h323 variant, mitel ip protocol)
05:29.15Juggiei think through work i can get protocol docs
05:29.20Juggieas we are a high lvl mitel partner
05:30.00Juggiemight not be legal though probally be a NDA so reverse engineering would likely be better.
05:30.27`SauronReverse engineering might be questionable as well
05:30.37`SauronEspecially if you already have a relationship with them
05:31.27sudhir492I can update from cvs.digium.com now
05:32.02Juggieperhaps
05:32.06Juggiewe have a ton of mitel gear
05:32.17Juggiewould be sweet if it could work on asterisk
05:32.23Juggiei wonder how much its different from h323
05:32.47TrepaliumJust enough to make you buy only their stuff, probably.
05:33.03Juggiewell not really true
05:33.07Juggiethey have some changes
05:33.17Juggielike phone book, phone speed dials
05:33.22Juggieall those are stored server side
05:33.24Juggienot on thephone
05:33.31Juggieunlike a sip phone which is smarter phone side
05:34.56zignigany reccomendations for a cheap(ish) SIP desk phone ?
05:35.40TrepaliumDepends on your price range, and feature requirements.
05:35.57NTJOCKZignig: Sipura 841 is a nice phone
05:36.09NTJOCKgood feature set, $84 is cheap.... reasonable quality.
05:36.13NTJOCKeasy to configure
05:36.25*** join/#asterisk pawnbroker (~rstevensj@ca-santaanahead-cuda1-c5a-45.anhmca.adelphia.net)
05:36.28*** join/#asterisk tuxinator_linux (~anonymous@ip68-109-146-168.ph.ph.cox.net)
05:36.33pawnbrokerHello gang
05:37.23zignigNTJOCK: and asterisk compatible I suspect ;)
05:38.02Juggiei've been annoyed at a few hardware sip phones
05:38.07Juggiethey limit conferences to 3 people
05:38.20Juggieat least the mitel 5055 and cisco 7960
05:38.21pawnbrokera quick question form a nb, is it possible to have * use some type of audio input for MOH?
05:38.24`SauronHuh?
05:38.43Juggiepawn like a line in on a soundcard?
05:38.50`SauronWhy would you do conferencing on the phone?
05:38.56`SauronSigh.
05:38.57zignigNTJOCK: $195 (.au) not bad .....
05:39.05Juggiesauron say i call someone, and i conference someone in, i cant add a 4th
05:39.20`SauronHum, right.
05:40.09pawnbrokerjuggie: yes, i have computer generated MOH on a wdoz box and would like to plug into *
05:40.33`SauronI have a 7940 at work, so I don't worry too much about conferencing people in... :)
05:40.33NTJOCKzignig: yes, quite compatible.
05:40.39NTJOCKI have 10 of them
05:40.52NTJOCKI actually like them better then my two polycome 500's
05:40.58NTJOCKthe Poly's are a pain to configure
05:41.39NTJOCKThe 841's just work.  The config isn't scaleable like the polys... so if you have hundreds of seats it may get annoying.
05:41.54NTJOCKBut the 841 is great for a small business where I don't have to worry about people tinkering iwth their phone settings
05:42.20Juggiepawnbroker, if you can run a shoutcast server on windows
05:42.24Juggiei have seen that
05:42.31NTJOCKIs there a good tutorial for building a IVR tree?
05:42.34Juggienot sure on using soundcard line in, researching
05:42.47NTJOCKI need to get my music on hold behaving and my IVR built.
05:43.05JuggieNTJOCK ivrs are easy just think each context is a menu
05:43.11NTJOCKok
05:43.16`Sauronyup
05:43.16NTJOCKis there an example somewhere?
05:43.19NTJOCKI searched the wiki
05:43.29`SauronI always love the telemarketer torture as a demo for building an IVR
05:43.35`Sauronsearch the wiki for "telemarketer torture"
05:43.37NTJOCKI saw that one once... it was great
05:43.37`Sauron;)
05:43.42NTJOCKis there a practical one?
05:43.44pawnbrokerjug: i have considered that option, i was looking for a more direct solution
05:43.46NTJOCKI think that's in the book I have
05:43.59`SauronI had it running here for a while, while I was doing some testing.
05:44.48*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
05:45.18NTJOCKin looking at the telemarketer torture one do you just GOTO the s,1,?
05:45.27Juggiepawnbroker, u need to find some way to generate a 8khz wav stream from your input
05:45.29NTJOCKand then it reads the lines under script?
05:45.32`Sauronyeah
05:45.35NTJOCKoh nice
05:45.38`Saurongoto context,s,1
05:45.40`Sauronor whatever
05:45.44NTJOCKsooooo much easier then my evil Nortel
05:46.33ManxPowerI don't support anyone has the VZAccess Manager software from Verizon?
05:46.39*** join/#asterisk clive- (~pirch@myw-stp-66-18-86-220.sentechsa.net)
05:47.33Juggiemanx http://vzw.smithmicro.com/download/
05:48.32pawnbrokerjuggie, yup i'm going to see if there are any ux apps that I can schmooze
05:48.50ManxPowerJuggie, Yes, but that is not the verison that comes with the cell phone.
05:48.56ManxPowerWhich I seem to have lost the CD for.
05:49.21joaoviannaMy iaxy is loosing his registration... Anyone having the same problem ???
05:49.25ManxPowerThat software wants the original CD.
05:50.01neopherHELP! tring to install spandsp, when running ./configure it is not seeing  tiffiop.h , i have installed libtiff.  Any ideas?
05:50.11Juggiesleep now....
05:50.31`Sauronneopher: do you have libtiff-devel installed?
05:50.38Juggiemanx, thats what verizon links too
05:50.40Juggiehttp://www.verizonwireless.com/b2c/mobileoptions/nationalaccess/index.jsp?cm_re=Home%20Page*Top%20Nav*MobileOpt-NtlAccess
05:51.02neopheri beleive so, how do i tell, i'm not a linux guru
05:51.21Juggieneopher, which linux distro/
05:51.28neopherRH 9
05:51.45Juggiejust get the rpm and install
05:51.53Juggieid suggest getting apt for redhat
05:51.58Juggieand using that as your package manager
05:52.01Juggieanyway sleep for me
05:52.06ManxPowerJuggie, You keep pointing me to the same place
05:52.20JuggieManxPower, i know notice i said "thats what verizon links to"
05:52.38Juggiei know it took you the same place i was just saying that thats what they officially link to
05:52.43neophertnx
05:52.51*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:52.57tuxinator_linuxWho's going to VON or Meet Asterisk?
05:52.57ManxPowerI'll just call them up tomorrow and tell them to send me another CD.  It only take a week or two
05:53.20`SauronI woulda gone to VON, but I'm not near SFO
05:53.22joaoviannatuxinator_linux: When ???
05:53.31tuxinator_linuxit's in two weeks
05:53.44tuxinator_linuxAll the hotels are sold out.  I need to find a roomie.
05:53.54joaoviannatuxinator_linux: Where ? I went to Von in Boston, very good.
05:54.09tuxinator_linuxSan Jose, CA
05:54.14`SauronHum
05:54.20`SauronI coulda gone to von in sjc
05:54.33`SauronHum di dum. I'll pass. :)
05:55.25tuxinator_linuxI want to go to Meet Asterisk, but need a roomie.  Can't really go over without a place to stay, now can I?
05:59.42NTJOCKWhat is the simplest way to record sound files for IVR menus?
05:59.59`SauronI called my voicemail and left a message with the text
06:00.07`SauronThen it's already gsm encoded and everything
06:00.21`Sauronor record to wav, and use sox to convert it
06:00.26NTJOCKk
06:00.30NTJOCKI like the voicemail one
06:01.01NTJOCKit's getting late.... I need to think out the IVR before I start recording it
06:01.02NTJOCK;)
06:01.09NTJOCKmore fun tomorrow
06:01.12`Sauronnah, just startrecording
06:01.15NTJOCKhehe
06:01.15`SauronIt's more fun that way
06:01.38NTJOCKI really need to figure out why one of my FXS ports won't generate a dial tone
06:01.45NTJOCKthe other 2 work fine
06:01.56TrepaliumDefective?
06:02.29NTJOCKmabye
06:02.32NTJOCKit was working the other day
06:02.39NTJOCKand it just suddenly "stoppped"
06:02.46NTJOCKguess I could try calling it  :)
06:02.53NTJOCKhave to set up an extension for that
06:03.05zignighelllo FXS port CAN YOU HEAR ME ?
06:03.10`Sauronhave you tried rebooting 3 times without saving?
06:03.18`Sauron:)
06:03.58TrepaliumYou need to sprinkle holy water on the card to excise the daemons that have taken residence in it.
06:04.17zignigNTJOCK: try unplugging and reseating the board.
06:04.26`Sauronwithout turning off power to the macine
06:04.27`Sauronmachine
06:04.30zignigthose riser boards somtimes lose their contacts.
06:04.41zignig`Sauron: <not>
06:05.03`Sauronzignig: well, duh. :)
06:06.12zignighehe ,
06:06.30zignig`Sauron: I  just got asterisk going last night actually, pretty cool
06:06.48NTJOCKok....
06:06.50NTJOCKthanks zignig
06:06.53NTJOCKI'll try that
06:06.58zignigI did some nortel years ago and man is this easier to configure ( even with so many conf files )
06:07.18zignigNTJOCK: climate changes can affect them.
06:07.25NTJOCKit's in a pretty consistent room
06:07.29NTJOCKactually in a server cabinet
06:07.32NTJOCKwith 3 other servers
06:07.40zignigbut get some meth or some iso-propal and clean the contacts too.
06:07.54NTJOCKit's not a server room, but it's not outside iether
06:07.59NTJOCKI'll try reseating
06:08.01TrepaliumHeating/cooling of the cards can cause the contacts on a daughterboard to lose their connection.
06:08.04NTJOCKthe other 7 ports work well in this system.
06:08.26zignigNTJOCK: :)
06:09.00*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
06:09.40NTJOCKzignig: you familiar with Nortel's Voicecall feature?
06:10.32zignigNTJOCK: this was 10 years ago; so nope
06:10.35zignigsorry
06:10.37NTJOCKok
06:10.43NTJOCKanyone else know what this feature does?
06:10.53NTJOCKI'm wanting to see if it can be replicated in *
06:10.57NTJOCKit's a way useful feature
06:11.09NTJOCKit's sort of like an auto-intercom
06:12.03pari don't suppose anyone has a pass for VON they could lend me?
06:12.13tuxinator_linuxnope
06:12.22tuxinator_linuxlooking for a rommie
06:12.26tuxinator_linuxI am looking
06:12.30parah
06:12.33tuxinator_linuxno rooms left
06:16.10parnot even at the fairmont?
06:16.18tuxinator_linuxlet me check
06:16.34tuxinator_linuxnope, no rooms
06:16.44paryou are kidding.
06:16.45parwow
06:16.50parramada limited?
06:16.55`SauronThere's a Days in in Sunnyvale that I've stayed in several times.
06:17.02`SauronNot far from SJC (< 10 minutes)
06:17.21parnah dude i can get a room for 245 a WEEK next to the ramada limited right now for th econvention
06:17.34parnot joking i just called them today
06:17.38moonwickhow much is it by the hour?
06:17.51parmoonwick: haha pretty close dude :)
06:18.05parthat about sums it up
06:18.11Nuggetheh
06:18.28tuxinator_linuxPar, just check it at the fiarmont site
06:18.33tuxinator_linuxPar, no rooms
06:18.51parjeez
06:19.30parwell i don't even have a pass otherwise i would book a room in the ghetto place next to the ramada ltd
06:19.49tuxinator_linuxI am not familiar with the area
06:20.04bonez39anyone using motorola VT1005V with asterisk?
06:20.28partux: if you have a pass for me you can room with me
06:20.29parhaha
06:20.40tuxinator_linuxI don't have a full
06:20.48tuxinator_linuxI was going to get an exhibit only
06:20.52paroh ok
06:20.59tuxinator_linuxMostly going for Meet Asterisk
06:21.07pari just wanted to see mikey powell
06:21.17tuxinator_linuxWhat is he known for?
06:21.22parFCC
06:21.27tuxinator_linuxohhhhh
06:21.42parwe used to have to get stuff approved when i was acer advanced labs...
06:21.47parwhat a joyous process
06:21.58tuxinator_linuxI bet
06:22.01parwe usually only had to try twice though
06:32.31hardwirewild
06:32.40EssobiAnyone think of some decent softphones for windows that supports the URL on the dial app?
06:36.05NTJOCKwhat is the keysequence to reboot a Polycom phone? I can never remember the darn keys to push on it
06:38.33*** join/#asterisk DaBigMac (~JJ@203-173-48-1.dyn.iinet.net.au)
06:38.44DaBigMachello anyone around?
06:41.21zignigDaBigMac: nope , nobody here ;P
06:42.12DaBigMac:-) in that case I cant ask my question
06:42.39tuxinator_linuxNT, you always have a lot of questions.  Do you live on here?
06:42.45tuxinator_linuxoh he's gone
06:43.07tuxinator_linuxI hate studying.
06:43.35*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
06:43.58*** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de)
06:44.00shmaltz~seen [Outcast]
06:44.01jbot[outcast] <~knoppix@h00045a737929.ne.client2.attbi.com> was last seen on IRC in channel #asterisk, 1d 2h 24m 15s ago, saying: '~xten'.
06:44.29DaBigMacgents I have installed a X100P into my asterisk box and all works ie calls coming in on my land line get switched to my houshold sipphones etc
06:44.58*** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net)
06:45.20DaBigMacissue is (not sure if its a problem) it takes a few rings between asterisk seeing something coming in on the ZAP interface before it switches it to the sipphone
06:45.24doolphwhere can I get a h323 gateway for linux
06:45.49DaBigMacis there a setting to speed this up?
06:45.59Inv_arpDaBigMac: use wait()  at first
06:45.59tuxinator_linuxDaBigMac, I think that is expected
06:46.20tuxinator_linuxDaBigMac, but I don't have experience with it
06:46.25shmaltzDaBigMac, it needs to detect callerID first
06:46.50DaBigMacInv_arp do you mean in extensions.conf under the default incoming context
06:47.06DaBigMacschmaltz I have set the lables to disable callerid
06:47.20DaBigMacin zapata.conf
06:47.27*** join/#asterisk Rick_Hunter (~rhunter@02-037.008.popsite.net)
06:47.29shmaltzDaBigMac, where? you have to set it in zapata.conf
06:47.52shmaltzThen I'm giving up
06:48.11shmaltz~hacking
06:48.12jbotthe art of hacking your ass with a midi tower case
06:48.12mishehubah.
06:48.13DaBigMacbefore you do what flag would I set.....just to be sure
06:48.21mishehu~mu
06:48.22jbotWhen asked by a monk if a dog had Buddha Nature, Joshu said "Mu."
06:48.42shmaltz~zapata
06:48.44jbotmethinks zapata is http://www.zapatatelephony.org
06:48.48mishehuactually, I would have said "The Kow says Mu."
06:49.03mishehu~mu is The Kow says Mu.
06:49.04jbot...but mu is already something else...
06:49.31tuxinator_linux~food
06:49.32jboti guess food is something I huwaked up, or a poor substitute for beer
06:50.01DaBigMacso is usecallerid=no enough?
06:51.26shmaltz~mu
06:51.27jbotWhen asked by a monk if a dog had Buddha Nature, Joshu said "Mu."
06:51.58shmaltz~bread
06:51.59jbotit has been said that bread is a wheat-based food, and a bad Liverpudlian series that doesn't run anymore. a function used inside the linux kernel
06:52.23shmaltz~jesus
06:52.24jbotmethinks jesus is dead so stop crying about him
06:52.34shmaltz~relegion
06:52.48tuxinator_linux~sleep
06:52.49jboti guess sleep is overrated, and a poor substitute for caffeine
06:52.56tuxinator_linux~caffeine
06:52.57jbot[caffeine] the nectar of the gods in concentrated form or the jiuce which runs through our veins
06:52.57shmaltz~bed
06:52.58jbotmethinks bed is a thing programmers have never heard of, ask me about shower
06:53.07tuxinator_linux~shower
06:53.08jbotrumour has it, shower is man using one hand in a very usefull way
06:53.09*** join/#asterisk iceyp (~icepick@firewall.unix.co.nz)
06:53.28iceyphow do i add a meeting room that doesnt require a password, but is simular to the coffee house at FWD
06:53.34tuxinator_linux~chicken
06:53.35jbothmm... chicken is http://kellari.org/other/linux/wanted.jpg , or free-food for all!
06:53.58shmaltz~hen
06:54.05shmaltz~bird
06:54.09shmaltz~no
06:54.10jbotYES
06:54.11shido6ths tuxinator_linux
06:54.16shmaltz~yes
06:54.16jbotYou dont say!
06:54.21DaBigMac~windows
06:54.22jbotwindows is, like, a 32 bit hack on a 16 bit operating system, originally designed for an 8 bit CPU, with a 4 bit system bus, made by a 2 bit company that can't stand 1 bit of competition...
06:54.39shmaltz~user
06:54.40jbot[user] currently detached. Talk to this user upon their return. You will now be ignored. [HackFactor Elite 2.0], or a synonym for moron
06:54.52shmaltz~loser
06:54.53jbotwell, loser is Al Gore
06:55.01shmaltz:)
06:55.07shmaltz~:)
06:55.10jbotwell, :) is honeypea
06:55.21shmaltz~al
06:55.22jbotwell, al is Albania
06:55.28shmaltz~al gore
06:55.29jbothmm... al gore is the inventor of the internet
06:55.29tuxinator_linux~girl
06:55.30jbot"if you dont believe in the existance of evil, you have a lot to learn"
06:55.48shmaltz~sex
06:55.50jbotI'm sexless
06:55.50tuxinator_linux~chick
06:55.51jboti heard chick is beeyotch
06:56.05tuxinator_linuxdinner time, be back in a bit
06:56.06shmaltz~married
06:56.08tuxinator_linux~dinner
06:56.09jbotdinner is, like, http://www.petsorfood.com/
06:56.11tuxinator_linuxI'm married
06:56.22DaBigMac~beeyotch
06:56.36shmaltz~married
06:56.42shmaltz~single
06:56.48shmaltz~status
06:56.48jbotSince Fri Feb 25 06:43:15 2005, there have been 67 modifications, 728 questions, 0 dunnos, 0 morons and 356 commands.  I have been awake for 3d 13m 33s this session, and currently reference 108214 factoids.  I'm using about 17880 kB of memory. With 0 active forks. Process time user/system 5906/621.86 child 415.39/33.73
06:57.15shmaltz~nice
06:57.16jboti heard nice is prime example of SuperJuan, or a good term for GNOME a derogatory term meaning bland, boring, feeble, or just crap. Example: That's a nice haircut. a city in france, or a program that will run a program with a modified scheduling priority (from -20 to 19, where 19 is the lowest).
06:57.22shmaltz~bye
06:57.23jbotbye bye
06:58.40shmaltz~spam
06:58.41jbotACTION sings, Spam, Spam, Spam, Spam, Spam, Wonderfull spam!
06:58.58shmaltz~putz
06:58.59jbothmm... putz is a person who constantly asks jbot questions in a channel instead of using /msg
06:59.13shmaltz~shmuck
06:59.14jbotNot me, you!
07:00.45odie_flocon~nfas
07:00.46jbotnfas is, like, "Non-Facility Associated Signaling" FixMe: saves a D channel on PRI's orsomethingorother
07:01.14odie_flocondam it's pretty good.
07:02.01DaBigMacall another question : if there is an incoming call and I have say three different sip phones rung/dialled......when one of them answers/picks up the call the other sip phones still ring for a bit.....anyway to kill them off once th call is answered?
07:02.21odie_floconnot really
07:02.49odie_floconwhat is happening is they are not responding to their command to stop
07:03.26DaBigMacso asterisk sends them a stop command as soon as it transfers the call to the answering sipphone?
07:03.32odie_floconyou need to capture some signalling between to find out if it's *, or the phones.
07:03.51odie_floconyes * would do that.
07:03.54*** join/#asterisk kks (~kks@203.115.208.140)
07:04.25DaBigMacok thanks odie.......do we know is sipura 2000's are notorious for that
07:04.39odie_floconI don't know.
07:05.04odie_floconI do know that sometimes sip phones differ in their commands
07:05.28odie_floconie the way they respond to a command is different.
07:05.56odie_floconand there are 2 to 3 messages sent when you are cancelling a call.
07:06.10DaBigMacok I assume the only way to view signalling is via debug in console?
07:06.19iceyphow do i add a meeting room that doesnt require a password, simular to the coffee house at FWD?
07:06.19odie_floconwell that's one way.
07:06.24rvhican realtime db use mysql on another server?
07:06.46odie_floconjust don't put a password in the conf file.
07:06.57iceypodie_flocon what is required to open one
07:07.23odie_floconI havn't implemented Realtime DB so I can't say. but if it's mysql I would assume you could use another server.
07:07.42odie_floconwhat do you have running right now iceyp
07:08.02rvhiwhere do you config the server config, user/pass/table for mysql?
07:08.20odie_floconthat I don't know rvhi
07:08.48odie_floconright now I'm in widoze so I can't go check.
07:08.52*** part/#asterisk djin (~djin@gridfox.xs4all.nl)
07:09.55odie_floconhmmmmm.
07:10.26odie_floconiceyp, do you have a digium card in your system right now?
07:12.08odie_floconhey iceyp, are you from New Zealand?
07:15.05odie_floconhello is anybody home?
07:16.56*** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net)
07:16.59iceypodie_flocon sorry
07:17.01iceypwas on the phone
07:17.04iceypyes i'm from NZ
07:17.04odie_floconohh ok.
07:17.07odie_floconcool.
07:17.08iceypand no i dont have a digium card
07:17.14iceypits just plain old voip
07:17.19iceypno pstn network incluided
07:17.20odie_floconohh ok.
07:17.30iceypsimular to FWD
07:17.34*** join/#asterisk tuxinator_linux (~anonymous@ip68-109-146-168.ph.ph.cox.net)
07:17.41odie_floconyou may have to recomiple your * to make it work.
07:17.42iceypI'm trying to setup a NZ based version
07:17.52iceyprunning on BSD
07:17.53odie_floconhmmm.
07:18.04odie_floconand what's wrong with Linux?
07:18.08odie_flocon:D
07:18.13iceypall my servers are BSD
07:18.21odie_floconI lived in Queensland a long time ago.
07:18.27iceypand i know asterisk runs better on linux
07:18.29iceyptrue
07:18.31iceypnice one :)
07:18.33iceypwhere you now?
07:18.37odie_floconCanada.
07:18.40iceypsweet
07:19.01iceypsure you hate the americans?
07:19.03iceyp:P
07:19.18odie_floconyeah. it is. I was supposed to be in the area for 2 years. but I could only stay 10 mths
07:19.22iceypor dislike them is a little less harsh
07:19.39odie_floconsometimes yes.
07:19.42tuxinator_linuxodio, mission?
07:19.47odie_floconyup
07:20.01iceypwere at code orange!!! watch out something might happen in america today people!!! Hold onto your hats and worry!
07:20.10odie_floconheheh
07:20.23DaBigMachey odie, if I switch debug on does it log all of the messages to a file somewhere?
07:20.26iceyppeople living behind fear
07:20.41tuxinator_linuxodie, So you're a MO?
07:20.42odie_floconummm, I believe it does.
07:20.56odie_floconI'm LDS.
07:20.58odie_flocon:D
07:21.29tuxinator_linuxOdio: I am too.  A friend calls us Mo's.
07:21.31odie_floconummm yeah DaBigMac, it should write it too a file.
07:21.39odie_floconheheh.
07:22.04odie_floconthat's funny tux.
07:22.44DaBigMachmmm it isnt in /var/log/messages
07:22.50DaBigMaclooking
07:22.51*** join/#asterisk shaZwaz (~chatzilla@203.81.196.167)
07:24.10odie_flocondon't think so.
07:25.08odie_floconI doubt it's in /var/log/messages
07:26.01odie_floconit would be in the * directory tree.
07:26.08odie_floconmaybe in /var/log
07:26.12odie_floconbut not in messages.
07:26.40zignigDaBigMac: /var/log/asterisk/messages ?
07:27.26odie_floconok sorry iceyp
07:27.43odie_floconyou need to compile the zaptel devices.
07:28.15iceypok
07:28.19iceypi might give it a miss
07:28.23iceypleave the coffee room out of it
07:28.29odie_floconheheh
07:28.31iceypforward users to the FWD coffee room
07:29.07odie_floconand then you need to do an insmod or modprobe of ztdummy
07:30.22tzafrir<PROTECTED>
07:31.04Inv_arpwhen updating an AGI script is there a need to reload?
07:31.12*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
07:32.08odie_floconyou catching this iceyp?
07:33.04odie_floconicey must be on the phone again.
07:33.38iceypw
07:33.40iceypwoops
07:34.00odie_floconok so you are there.
07:34.01iceypyep i see that, just dont know if its the same with bsd
07:34.21iceypPort:   zaptel-0.8_1
07:34.21iceypPath:   /usr/ports/misc/zaptel
07:34.21iceypInfo:   A FreeBSD Driver for Digium X100P/TDM400P Telephony Cards
07:34.22odie_floconyou may need to recompile your *
07:34.26iceypmaybe i need to install them
07:34.32odie_floconyes
07:34.46odie_floconzaptel must be compiled.
07:34.58odie_floconand in the zaptel drivers there is a ztdummy driver.
07:36.12odie_floconif you have the right "USB chipset" the ztdummy driver will work.
07:36.25odie_floconand you will be able to use a confrence bridge.
07:36.33odie_floconif not you need to get some sort of digium hardware.
07:37.20DaBigMacodie thats a good point now that I have X100P do I still need ZTDUMMY....the reason I ask is, is the x100 cut down version
07:37.39odie_floconI dont' think you still need it.
07:38.32odie_floconwhat it needed was a clocking device.
07:38.39iceypk thats now installed
07:38.58DaBigMacbtw iceyp to compile ztdummy you need to edit the make file and uncomment ztdummy reference for it to compile ztdummy......the default distribution has it commented out
07:39.11iceypzaptel drivers installed
07:39.22DaBigMacodie yup, just wondering if some cards have the clock and others dont
07:39.47odie_flocononce you have the ztdummy device working, you need to configure the conf files
07:40.00odie_floconthey should all have the clock.
07:40.47odie_floconbecuase you need clocking to sync your voice etc.
07:42.15*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
07:44.25odie_floconyou still there iceyp?
07:45.07odie_floconanybody?
07:45.15odie_floconit just goes quiet in here all the sudden.
07:45.28iceypargh
07:45.33iceypno wont install
07:45.51odie_floconit's not easy.
07:45.57odie_floconit takes a while.
07:46.08iceypyeah i dont need it
07:46.16iceypi just need this programmer to finnish the frontend
07:46.17iceyp:)
07:46.23odie_floconheheh
07:46.34zignigiceyp: offer him/her more cafffine.
07:47.06iceyplol, he's having dsl latency issues
07:47.11iceypand that seems more important ;/
07:47.18neopheras i understand, redhat 9 is the absolute late RH distro and now fedora is taking it's place.  Is this true?
07:47.21zignigiceyp: Qos ?
07:47.30odie_floconyes
07:47.38odie_floconthat is correct neopher
07:47.39iceypzignig nah its an isp issue
07:48.03neopheranyone try using asterisk with fedura? any issues?
07:48.13zignigiceyp: ah, ISP's bite I've been getting some vairiability , although I don't run voip here ( just at home so far )
07:48.24odie_floconahh.
07:48.35odie_floconas far as I know neopher there aren't any issues.
07:49.00odie_floconI use Mandrake
07:49.37neophertnx, thinking about updating, i have so many packages out of date i'm thiking it's easier to start over
07:50.55odie_flocondang
07:52.03odie_floconAm I still online?
07:52.12zignigodie_flocon: no
07:52.13zignig:P
07:52.28odie_floconok thanks zignig
07:52.29odie_flocon:D
07:57.27*** join/#asterisk Newbie___ (some@60.48.161.253)
07:59.01*** join/#asterisk B4 (~B4@202.69.48.245)
07:59.16*** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se)
07:59.53*** join/#asterisk shanky (~shanky@238.Red-80-33-29.pooles.rima-tde.net)
08:00.16odie_floconhey all I think it's time for me to go to bed.
08:06.09tuxinator_linux~bed
08:06.10jbotfrom memory, bed is a thing programmers have never heard of, ask me about shower
08:07.39Mavviehmm... anybody experience with a Digium PRI card and isdn4linux ?
08:07.40PCadach~shower
08:07.40jbotwell, shower is man using one hand in a very usefull way
08:08.53tuxinator_linuxI don't get it
08:09.17Mavvietuxinator_linux: you're smelly!
08:09.27tuxinator_linux~smelly
08:09.28jboti heard smelly is fungus is kinda smelly, but grows on you
08:10.39tuxinator_linuxWhy am I smelly?
08:10.46tuxinator_linux~Mavvie
08:10.48Mavviewell, if you don't get a shower.
08:11.13Newbie___hi all, i am given the option rfc2833, inband,info to use on * SIP , anyone had any idea which is better ?
08:11.26Mavvieinband isn't.
08:11.44Newbie___then is rfc2833 instead ?
08:12.32*** join/#asterisk DeanH (JTR@209-203-52-3.network.ods.co.za)
08:12.40DeanHhello
08:12.48DeanHcan somebody help me with the following
08:13.04*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
08:13.18*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
08:13.34DeanHastreix serv linked into 8 port n2p device ?
08:14.18*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
08:14.21Mavvielet me ask it different...
08:14.34Mavviehas somebody experiecne with Digium E1/T1 cards with mgetty?
08:15.28neopherquestion on fedora,   whats the diff, between FC3-i386-SRPMS  FC3-i386
08:18.00clive-DeanH n2p ..yuck
08:18.06clive-:)
08:20.14*** join/#asterisk dersteer (~travis@24.231.151.204.gha.mi.chartermi.net)
08:22.15*** join/#asterisk ars_ (~ars@84.204.22.118)
08:22.32ars_hello everyone
08:22.45*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
08:22.54tzafrirBTW: don't call asterisk asterix, or you may get sued: http://tuxmobil.org/mobilix_asterix.html
08:23.28ars_i'm interested in http://www.nufone.net/ , anyone tried?
08:23.32*** join/#asterisk pashah (~pashah@relay.patentica.com)
08:23.33*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-11-74.d4.club-internet.fr)
08:23.41pashahmorning
08:23.43ars_hello pashah.
08:23.52ars_i'm interested in http://www.nufone.net/ , anyone tried?
08:24.02*** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com)
08:24.29pashahars_: nufone is fine
08:24.39tzafrirneopher, an srpm is a package with the source, patches and build instructions ("spec")
08:25.08ars_seems very strange to have no network map and legal notice, consist of one (1) web-page and a link to credit card number entering.
08:25.41tzafrirrpmbuild --rebuild file.src.rpm should rebuild a binary from the SRPM . It should also notify you if there are missing build dependencies, if it was well-written
08:25.58*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
08:26.05ars_pashah: what's your experience?
08:26.23*** join/#asterisk aggelos (~aggelos@egate.eleven.de)
08:27.05tzafrirBTW: the asterisk tgz itself has a spec. So theroretically it should build an rpm using: rpmbuild -tb asterisk-1.0.6.tar.gz
08:27.06*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
08:28.07*** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f)
08:31.23MavvieWise advise: If you have more than one telco provider into your company, don't let them terminate on the same box since the zaptel driver only takes the clock from one of them and the other telcos will get timing issues.
08:32.08neophertnx, and the regular Fedora src iso's are the actual install cd's, right.  Sorry for the dumb questions, it's been a while since i messed with linux distros
08:35.18DeanHis it possible to ingregrate asterisk with net 2 phone gateway ?
08:35.32neopheryes
08:35.59neopheri beleive net2phone uses sip
08:36.11DeanHI am not sure ?
08:36.11neopherso therefor, yes
08:36.26neopherlemme check for you real quick
08:36.48neopherwhy net2phone?
08:36.54DeanHok, so I can use any sip based phone on my network and then terminate into  asterisk box and into net2phone
08:36.58DeanHI need voip provider
08:37.14DeanHI am in South Africa and need someone to terminate call in London for me
08:37.42parfor everyone with questions about faxing
08:37.50parhttp://www.voip-info.org/wiki-Asterisk+fax
08:37.58RestLessGeminiMavvie : Thanks for advise
08:38.05parthats got the best answers i've seen.
08:38.26DeanHdoes anyone know of good service provider I can use to terminate calls ?
08:38.32DeanHbesides net2phone ?
08:38.33*** join/#asterisk {zombie} (zombie@soulasylum.penguincare.com.au)
08:38.39*** join/#asterisk sudhir492 (~sudhir@4.7.59.232)
08:39.09joaoviannaI need to create a AGI that sends a email. Anyone knows how to execute external scripts with * ?
08:39.39neopherjoaovianna: what are you tring to do
08:39.49sudhir492read the wiki AGI
08:40.08sudhir492create a file in /var/lib/asterisk/agi-bin
08:40.31Mavviejoaovianna: if you use perl for the AGI, use MIME::Lite
08:41.34joaoviannaThanks Mavvie, I'm using phpagi, but I can't execute system(....)
08:41.46Mavvieif you are using PHP, use the mail command.
08:42.38ars_i'm interested in experience with NuFone.Net, can anyone describe them?
08:42.49joaoviannaThanks Mavvie...
08:50.17*** join/#asterisk pdracevich (bob@210.54.152.176)
08:53.56pdracevichcall conferencing? any suggetions?
08:54.48*** join/#asterisk h3x (Justino@ip68-108-176-196.lv.lv.cox.net)
08:55.31rvhianyone used realtime db for sip?
08:55.40rvhidoesn't seem to work
08:58.09pdracevichcall conferencing? any suggetions?
08:59.39ars_i'm interested in http://www.nufone.net/ , anyone tried?
09:03.09pdracevichvoice mail?
09:05.06Mavviepdracevich: have milk?
09:08.37tuxinator_linuxTime for some sleep
09:08.40tuxinator_linuxnight guya
09:08.44tuxinator_linuxguys
09:09.15*** join/#asterisk meppl (~mephisto@pD9E68FFC.dip.t-dialin.net)
09:09.39mepplguten morgen
09:12.37*** join/#asterisk l-fy (~diana@diana.null.ro)
09:12.40l-fymmmm
09:12.41l-fymorning
09:12.45l-fysomeone from Rome mbranca ?
09:13.52pdracevichmavvie: huh?
09:14.18RoyKl-fy: morn
09:14.28Chotairehm, can anyone tell me why there are always 2 mpg123 processes?
09:14.30l-fyhey RoyK
09:14.32l-fyso
09:14.41l-fyis there someone from Rome here?
09:16.42*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
09:16.53l-fyhey scanna
09:16.59l-fyscanna > are you from Rome?
09:23.07*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
09:23.25Zeeekecho "hello world\n";
09:23.30mbrancal-fy, nope, milan
09:24.17RoyK10 PRINT "HELLO, ZEEEK"
09:24.53par20 GOTO 10
09:25.13*** join/#asterisk dersteer (~travis@24.231.151.204.gha.mi.chartermi.net)
09:25.23*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
09:25.56Zeeek~LART RoyK while(1)
09:26.31Zeeek~lart par if (1)
09:30.36*** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk)
09:32.48ChotaireI have a question about "MusicOnHold".. I have an option "3" to hear music.. is there a way I can have the user press "#" or something to return to the main menu? it seems MusicOnHold is indefinitely.. does any other option like Playback etc allow shoutcast streams?
09:35.47*** join/#asterisk visik7 (~ciao@visik7.user)
09:36.31l-fyi have a question
09:36.34*** join/#asterisk Delvar (~irc@83.146.53.34)
09:36.44l-fycan you tell me opinion on www.null.ro as design and look?
09:37.01Chotaireomg
09:37.26l-fywhat Chotaire , is something wrong?
09:39.06*** join/#asterisk mady (~root@61.11.24.250)
09:40.43madyhi denon
09:40.56madycan we chat now
09:43.29Chotairefy: want a true opinion?
09:43.34*** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com)
09:44.08Newbie___hi bjohnson
09:45.21*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
09:46.45*** join/#asterisk mady (~mady@61.11.24.250)
09:46.54l-fy?
09:47.14madyhi denon
09:47.26madycan i talk to u now
09:48.17l-fyRoyK > you are from Norway, i'm from a warm full of snow now country
09:48.39*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
09:48.56madyis there any on who can help me on asterisk
09:49.24madyi am mady wants help on asterisk
09:52.04madyhello drumkilla
09:53.46madyhello twisted
09:53.49*** join/#asterisk Mike_TK (~Mike_@213.180.245.62)
09:53.57l-fyRoyK > i was thinking that trolls are in the cold countrys
09:54.07*** join/#asterisk johngalt (~efort@shell3.sea5.speakeasy.net)
09:54.19l-fyRoyK > and what about options, didn't you told me to be open
09:54.22l-fyi am open
09:57.19madyhello Mike_TK
09:57.27*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com)
09:57.32Mike_TKmady: Hello.
09:58.05madyhi Mike how r u
10:00.57*** join/#asterisk Ron-Na (~ronald@203.70.36.126)
10:07.29*** join/#asterisk jedirl (~fdsafasdf@213.162.200.226)
10:07.30jedirlHello :)
10:07.31*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
10:07.50jedirlAnyone has any kind of drivers for old aculab hardware?
10:09.04ChotaireMP3Player(URL) did the trick regarding my last question..
10:09.20Chotaireanyway, I still wonder why * always fires up two mpg123 processes.. I wonder what for.
10:09.50*** join/#asterisk pdracevich (bob@210.54.152.176)
10:09.57pdracevichhelp??!?!?  No entry in voicemail config file for ''
10:10.44Chotairefy: the site looks VERY basic, as if someone did the site who knows nothing about design. that's my true opinion, no offence.
10:11.40l-fyChotaire > no problem, how do you think i can impove it?
10:12.28pdracevichhelp?
10:12.40l-fyyes pdracevich
10:13.27Chotairefy: change fonts to non-default, use a better layout, use better colors, remember that a screen is bigger than 800x600 so why not make more space for the products on screen etc..etc..etc...
10:13.42Chotairealso, the logo looks very "I used a graphics program plugin".
10:14.33l-fyChotaire > not everyone use a huge screen
10:14.54l-fyin fact i did use a graphics program plugin
10:15.04l-fyit dosen't try to be original
10:15.07l-fyjust profesional
10:15.20Chotaireok but it doesn't look professional ;)
10:15.24Chotaireit looks easy.
10:15.35l-fyi see
10:15.36l-fyok
10:15.44l-fywhat can i do for layout?
10:15.50jedirlAnyone here uses old Aculab E1/T1 isa hardware?
10:15.52jedirlor has used it?
10:16.00l-fyaculab rules
10:16.13jedirlyes but they don't provide me old drivers for my old isa hardware
10:16.20l-fyi don't have them
10:16.22l-fyjust ask them
10:16.25jedirlI've asked
10:16.35jedirlthey say they don't have them in their archives...
10:16.47l-fyWOW
10:17.27jedirlincredible, eh?
10:17.32l-fyyes
10:17.38l-fythat's actualy odd
10:17.43jedirlI guess they want me to buy a new PCI prosody
10:17.49l-fyyou can write to dyfet@gnu.org and ask him if him have them
10:18.05jedirlok, thankyou very much :))
10:18.36l-fyhim use to work with aculab and i bet him can kick aculab ass pretty much
10:18.58l-fyIMNHO aculab is going to die anyway
10:19.07jedirlreally?
10:19.07jedirlwhy?
10:19.41l-fybecause they sell something that dosen't worth anymore
10:20.09jedirlwell they are a quite cheap SS7 solution...
10:20.45l-fynot for long :)
10:23.45*** join/#asterisk ScythelX (~basin@pc-24-151-28-122.newm2.ct.charter.com)
10:23.46jedirlI've just send an e-mail to this person, l-fy
10:23.50jedirlthankyou:)))
10:24.03l-fyno problem
10:24.04ScythelXhello all
10:24.06l-fyhis name is David
10:24.23ScythelXanyone own a cisco IP phone with the sip image loaded, if so does anyone know how to reset the password on it or recover
10:24.27l-fyi think him wrote the first application that knows about aculab for linux
10:24.34jedirlwow
10:24.38l-fyScythelX > learn from here
10:24.46jedirlif he could help me it would be really great
10:24.47jedirl:)
10:24.56l-fy**#02**
10:25.02l-fyor something like that
10:25.07l-fydepend on the cisco model
10:25.10ScythelXthe default password cisco isnt working
10:25.14ScythelXits a 7960
10:25.16l-fyi know
10:25.25l-fyScythelX > i've found that info on the internet
10:25.30l-fyi have a 7905
10:25.59ScythelXive only found people asking how to recover with out and valid answers
10:26.21ScythelXand/any
10:26.30l-fyi've found that answer on cisco website
10:26.32l-fyfor sure
10:26.36l-fylet me find
10:28.02l-fytry **#
10:28.11*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
10:28.15ScythelXthats not for the SIP image though
10:28.21Zeeekanyone have a farfon?
10:28.25*** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg)
10:28.29ScythelXthats with the MCGP image loaded
10:28.45l-fydosen;t matter
10:29.26ScythelXdidnt work
10:30.45l-fyyou must press in the same time
10:30.55l-fy*+6+settings
10:31.02ScythelXthat just reboots the phone
10:31.20l-fyreally?
10:31.50ScythelXyeah it just power cycles it without unplugging it
10:31.55ars_i'm interested in http://www.nufone.net/ , anyone tried?
10:32.08soundguyhmm.. I am having a problem with incoming calls. If I direct them to the softphone (xLite) I can hear on both ends, however if I direct them to come to my grandstream phone (BudgeTone) no audio is heard either end. Both phones are set to only allow the ulaw codec and I can ring from the softphone to the grandstream perfectly.
10:32.08l-fyars_ > nufone is evil
10:32.19l-fynufone owner wrote the h323 module from asterisk
10:32.36l-fysoundguy > is a well known problem with bt
10:32.46soundguyHow is that fixable then?
10:32.49l-fyno idea
10:32.59l-fyi'm waiting for a bt this days to investigate
10:33.11soundguyok, so I am not just the only one then?
10:33.19l-fynope :)
10:33.23soundguyWhat codec is ulaw? g729?
10:33.26l-fyi will find out in about 2 weeks
10:33.28l-fyulaw
10:33.30l-fyor alaw
10:33.54soundguyI cant see that codec in my grandstream phone
10:34.27tzafrirlook for "g711" there
10:34.52soundguyPCMU maybe?
10:34.55soundguyno G711
10:35.50tzafririncoming from what?
10:36.20soundguyanything external (not on the voip network) ((from my sip provider)
10:36.44*** join/#asterisk YoYo (~funknugge@pool-151-199-125-240.roa.east.verizon.net)
10:38.58tzafrirsoundguy, any chance that the sip phone tries to connect directly to your provider? Can you run a sniffer there and see where it sends RTP packets to?
10:39.23ScythelXsip debug
10:40.17tzafrirsip debug shows whatever goes through asterisk. But what if the hardphone tries to send data elsewhere?
10:41.19soundguyIt shouldnt though, as it only knows the asterisk details
10:42.30tzafrirSIP should allow the end nodes to use a direct link and not go through the PBX, right?
10:42.31l-fyPCMU is alaw
10:42.35l-fysorry
10:42.37l-fyULAW
10:43.13ScythelXwhat type of phone is it
10:43.28ScythelXi know the cisco phones you can telnet in an do a sip debug
10:44.15soundguyGrandstream BudgeTone
10:44.17soundguycrap phone
10:44.29ScythelXyeah i dunno then
10:45.44soundguywtf, it is working now
10:46.33soundguyoutgoing isnt
10:47.08soundguyall I did was add ,t to the extension for the grandstream
10:47.13*** join/#asterisk w0w0 (~w0w0@80-29-46-175.adsl.nuria.telefonica-data.net)
10:49.42*** join/#asterisk IsMe (some@218.111.157.77)
10:50.56IsMehi, i am having 1 way audio with xten. * is behind a router and IP place on the DMZ zone, trying to connect to * from outside
10:50.59IsMeanyone ?
10:51.04soundguyhmm..now this is fustrating..incoming now works fine, no outgoign from the grandstream does
10:51.39tzafrirsoundguy, what about a sniffer?
10:51.51soundguyoutgoing from softphone doesnt work either
10:52.14l-fysoundguy > that's weird
10:52.21soundguyI agree
10:52.30tzafrirWhat type of connection is to your upstream provider?
10:52.34soundguysip
10:52.45*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-200-175.dsl.scarlet.be)
10:53.02soundguythat is all they provide, they dont actually support asterisk but I am determined to get it working, they are more of a voip provider than a sip provider. They only support their phones (grandstreams)
10:53.32*** join/#asterisk MicH323 (~micosat@host217-44-194-118.range217-44.btcentralplus.com)
10:53.50tzafrirso, do outgoing packets go through Asterisk?
10:54.13Zeeekwho has a farfon here?
10:54.23soundguyyeah, sip debug spits out heaps of info very quickly during the call
10:54.31MicH323Hi all
10:55.32Ron-Nahi
10:55.37MicH323Anyone doing Asterisk on Solaris?
10:55.51Ron-NaIs anybody using astcc?
10:56.09l-fyMicH323 > not with H323 :)
10:56.11l-fyi bet :)
10:56.14libpcpRon-Na yeah why?
10:56.19ars_l-fly, sorry for delay
10:56.26ars_l-fly why is it evil?
10:56.28Ron-NaI got some troubles with the setup
10:56.46l-fyars_ > because the guy who run nufone wrote the h323 module in asterisk
10:56.58MicH323As I am more used to H323 then AsterisK
10:57.01Ron-Nalibpcp: I get "No such context/extension"
10:57.26*** join/#asterisk aggelos (~aggelos@egate.eleven.de)
10:57.36*** join/#asterisk dkoum33 (~dk@adsl210-static-gw1.access.acn.gr)
10:57.38aggelosanyone know anything on nufone.net ?
10:57.41libpcpi think its a route issue
10:57.51ars_nufone is down for 10 hours already
10:57.59l-fyMicH323 > o well, i wish you luck
10:58.05Ron-Nalibpcp: I am sure, but I cannot find it ;-(
10:58.25libpcpcreate a context for your outbound route
10:58.28ars_don't know about IAX network, but www is down
10:58.49aggeloscan't connect to our box hosted with them :(
10:58.49ars_l-fy, you tried their service?
10:59.00Ron-Nalibpcp: I have it, ...
10:59.13l-fyars_ > i've used the h323 module
10:59.13libpcpso use it in your routes
10:59.37ars_aggelos, you use michigan did or what?
10:59.41aggelosars_: do you know why ?
10:59.41Ron-Nalibpcp: that what I tried, ... but maybe it is because it is through Nufone, ...
10:59.49ars_l-fy, i see :-) <ars_> l-fly, sorry for delay
10:59.49ars_<ars_> l-fly why is it evil?
10:59.49ars_<Ron-Na> I got some troubles with the setup
10:59.49ars_<l-fy> ars_ > because the guy who run nufone wrote the h323 module in asterisk
11:00.13aggelosnope, just iax termination, and server hosting
11:00.13ars_oops
11:00.52ars_aggelos, the toll-free service?
11:01.02ars_what's the uptime?
11:01.13aggelosmore than 400 days, service is realy good.
11:01.31aggeloswe have our communication issues but the service is great
11:01.47aggelosuntil now,
11:01.47Ron-Nathanks, ... that was it, .. I used another context and than it worked, ...
11:02.02ars_so the current situation is really uncommon?
11:02.03aggelosars_: you use them too ?
11:02.21ars_we're thinking about. interested in 0,02 per minute of toll free
11:03.02ars_plus they got good rates for international
11:03.39dkoum33hi all. i'm trying compile cdr_addon_mysql on debian stable and it fails. are there any known problems?
11:03.47aggelosbut they are in no major trouble right now, are they ?
11:04.18ars_i don't know, i'm not a customer yet. just wanted an opinion.
11:04.36aggelosok, besides today, I am realy happy.
11:04.57ars_what's your 1-800 number?
11:05.00aggelosagain, communication with them get's little rocky but usuay its fine.
11:05.05*** join/#asterisk eivindtr (~Eivind@193.91.146.34)
11:06.10ars_they offer a virtual server with SSH or just set-up their gateway on requests?
11:08.51ars_ok thank all, thank aggelos.
11:13.45*** join/#asterisk wasim (~wasim@203.81.213.118)
11:14.03l-fyhey wasim
11:15.18aggelosars_: it was working so good that i can not find the support number for pager
11:15.46aggelosI have not had a direct contact with them for months, no need to.  Everything was running smothly
11:16.03wasimhiya sweetie
11:17.20l-fyhow are you dear?
11:17.36wasimhungover
11:17.45l-fyo well
11:19.50IsMehi, i am having 1 way audio with xten. * is behind a router and IP place on the DMZ zone, trying to connect to * from outside
11:19.51IsMeanyone ?
11:23.15\Grooby\nat=yes for the extension config in sip.conf?
11:24.04\Grooby\also under sip's [general] try to add externip=xxx.xxx.xxx.xxx  you can use a dyndns for that also
11:24.14IsMetried both, nat=yes and nat=no
11:26.06wasimtry sip=no
11:26.10*** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar)
11:26.15*** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com)
11:27.19l-fywhat?
11:27.32l-fywho is offending you deaR?
11:27.50IsMedidnt work either
11:28.51\Grooby\in CLI,sip show debug <extnumb>
11:28.59\Grooby\and see what it's doing
11:31.03aggelosanyone has nufone.net contact info besides the main phone number ?
11:31.17aggelossupport email or page number
11:31.31aggelos??
11:31.44IsMe\Grooby\ : no such command sip show debug
11:34.29\Grooby\oops
11:34.31\Grooby\sip debug peer
11:34.53\Grooby\and when you are done, sip no debug
11:36.52IsMecan i use sip debug ip xxx.xxxx.xxxx
11:37.21\Grooby\if it's a valid commond
11:37.23\Grooby\command
11:37.33\Grooby\it's better to debug an extension
11:37.48\Grooby\then you can see what SIP is doing between * and your xlite
11:37.52IsMeok
11:39.31IsMe\Grooby\: what should i look at ?
11:39.32aggelosnefone info ? anyone ?
11:39.38aggelosnufone.net ?
11:39.40\Grooby\look at the IPs
11:39.44\Grooby\see if they are the right IPs
11:41.19IsMedestination IP is right
11:41.25IsMemy IP is not
11:41.35\Grooby\what is the IP?
11:41.42IsMei should have a dynamic IP while * reports a LAN IP
11:42.02\Grooby\did you add the externalip=xxxxx?
11:42.12\Grooby\in the sip.conf's general section?
11:42.28\Grooby\oops
11:42.33\Grooby\it's externip=xxx.xxx.xxx.xxx
11:42.47IsMeyes i did, externip = IP connected to *
11:43.09\Grooby\IP connected to *?
11:43.14\Grooby\it' should be your dynamic IP
11:43.23\Grooby\or is that what you mean?
11:43.56IsMeno, i mean my * server is behind a router, * IP is static and i am connecting to * using dynamic
11:44.40\Grooby\so the asterisk's IP is not natted
11:45.22IsMei had it set nat=no
11:45.32\Grooby\how's your network setup?
11:45.42\Grooby\what kinda router?
11:46.10IsMedlink DI-714P+ wirelss
11:46.15\Grooby\ok
11:46.34\Grooby\you want the externip = the dlink's WAN ip
11:46.56\Grooby\*'s IP is static because its a static internal IP
11:47.12tzafriranybody here applied bristuff to libpri 1.0.6 ?
11:47.14IsMeyes i did that already
11:47.25\Grooby\ok
11:47.28\Grooby\did you reload?
11:47.52IsMei did that before u said to use sip debug
11:47.55tzafrirThat is to say: my current patch does not apply
11:47.57\Grooby\ok
11:48.13\Grooby\pastbin your [general] section for me
11:48.20IsMeok
11:48.48*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
11:48.52puzzledmorning
11:51.06IsMehttp://www.pastebin.com/247310
11:51.08IsMeis messy
11:51.12*** join/#asterisk zotz (~zotz@24.231.32.191)
11:58.42tzafrirNM. The patch does apply
12:00.12*** join/#asterisk GodThor (~ninja@212.110.95.139)
12:00.52GodThorhello , is any win software for testing asterisk server?or any other way for testing
12:03.15*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
12:04.35GodThoranyone?
12:06.23pashahGodThor: get a software phone
12:06.28pashahfor windows
12:06.38GodThorexample?
12:06.54\Grooby\google softphone
12:06.55\Grooby\;)
12:06.56GodThorbut i need for ip not for cards
12:07.11GodThori dont have a cards on my asterisk box
12:07.50GodThora have a quintum before asterisk :)
12:07.50\Grooby\?!?!
12:08.29GodThormy lines goes to quintum and then over ip to asterisk
12:08.56*** join/#asterisk Xander77 (~Alex@exten-halls-243.soton.ac.uk)
12:09.39GodThorohphone is it good?
12:16.10nirshello everybody
12:16.17nirshow is everybody feeling today ?
12:17.07RoyKsick
12:17.13nirsouch,
12:17.17RoyK:)
12:17.17nirsthat is now good man
12:17.19nirsnot good
12:17.26RoyKjut flu
12:17.34nirsoh, a flu comes and goes
12:17.35RoyKI'll probably survive
12:17.49nirsfor a minute I thought it might be something serious
12:17.51l-fyhey nirs
12:17.59nirshey diana
12:18.01nirswassup ?
12:18.29l-fylong time nirs
12:18.38nirsbtw, where in gods name is mark and the guys
12:18.43nirsyes diana, a very long time
12:18.47nirsI've been ultra busy here
12:18.57nirsbuilding a business isn't easy
12:19.24*** join/#asterisk seong (~seong@219.95.130.82)
12:19.31l-fynirs > i know, especialy when you can't based on parteners, which tent to disapear from time to time
12:19.40*** part/#asterisk GodThor (~ninja@212.110.95.139)
12:20.32nirscare to explain diana ?
12:20.40nirsnone of my partners disappeared on me
12:20.53RoyKnirs: mark is sleeping
12:21.33nirsroyk, well, I really could use marks help in solving a really buggy issue
12:21.46RoyKwhat's this?
12:21.55l-fynirs > just for me
12:22.04nirsyou see, I've been investigating along side with bkw and anthm an asterisk crash caused by a SIGTERM that is sent internally from asterisk
12:22.18l-fynirs > but dosen't matter, now our bussines is quite nice.
12:22.36nirsI've also checked the issue, and it appears to happen both on CVS and stable branch
12:22.47nirsthat's good to hear diana
12:23.11*** join/#asterisk emergen (~mh@202.5.145.13)
12:23.27nirsbe right back
12:23.39nirshave to go grab a sandwich or something, I'm going crazy here
12:24.57ariel_morning everyone
12:25.57RoyKwtf?
12:25.57RoyK<PROTECTED>
12:26.00RoyKwhat is that?
12:26.15filecodec.
12:26.48file"Dude, what yo' talkin' bout? I can't speak that there language!"
12:26.49RoyKhuh
12:27.29filea "Not Acceptable Here" usually comes from a codec problem, like trying to use a codec the device doesn't support... some devices spit that back
12:27.33RoyKfile: I had only set allow=gsm,g726,g729,alaw,ulaw,h263,h261 in sip.conf. without h26[13], it worked well
12:27.41RoyKthat's stupid
12:28.27filetrying to do video?
12:28.57RoyKyes
12:29.09filefun
12:29.14RoyKbut normal audio calls didn't work with that enabled in the global section
12:29.18RoyKand that's stooopid
12:29.47filepoor poor you
12:30.12RoyKfsck fsck you
12:39.13MicH323Cisco to Asterisk codec question: What is the apporiate cisco g.723.1 and g.729 codecs on Cisco? anyone now...
12:39.21eipiasterisk 1.0.6
12:39.24MicH323Cisco seems to cater to sooo many variants
12:39.35eipioops
12:40.32RoyKMicH323: asterisk doesn't support niether of them unless you pay....
12:40.39RoyKMicH323: on a LAN?
12:41.15MicH323Ahhh... I just assumes when I do show codecs it shows the codes...
12:41.27RoyKMicH323: are you on a lan
12:41.30RoyK?
12:41.30RoyKdoing h323?
12:41.32RoyKsip?
12:41.33RoyKzap?
12:41.37MicH323SIP
12:41.49MicH323over the internet
12:42.04RoyKhave you purchased g.729?
12:42.14RoyKg.723.1 isn't available for *
12:42.26RoyKthere's a test codec for both, though :)
12:42.27MicH323I am running on Solaris... Is thereanyone who can supply me those codes (PAID)
12:42.34*** join/#asterisk styx2005 (~styx2005@a-line138.supra.net)
12:42.36RoyKno idea
12:42.46RoyKMicH323: G.726 works though, and iLBC
12:43.28MicH323The Asterisk dosent what bps or is all G.726 supported?
12:43.50*** part/#asterisk styx2005 (~styx2005@a-line138.supra.net)
12:43.54*** join/#asterisk teemu-x (~tnurmine@tuomi.oulu.fi)
12:44.01*** join/#asterisk styx2005 (~styx2005@a-line138.supra.net)
12:44.07RoyKMicH323: asterisk stable only supports 32kbps
12:44.16jedirlpaid codecs are only for linux?
12:44.17RoyKcvs head supports more
12:44.22MicH323I am also interconnecting to others using IAX, so will G726 be supported by them?
12:44.24RoyKjedirl: think so, on intel
12:44.27MicH323I downloaded CVS
12:44.40RoyKMicH323: all codecs are supported on all protocols
12:44.40*** part/#asterisk styx2005 (~styx2005@a-line138.supra.net)
12:44.41MicH323mainly FWD and BroadVoice
12:44.45RoyKprotocol is independant of codec
12:45.13MicH323Thans Roy
12:46.21RoyKnp
12:47.21teemu-xjust updated to current stable version of asterisk and when calling using H323 on either end, the call hangs up right after answering - where should I start looking for the reason?
12:51.32*** join/#asterisk styx2005 (~styx2005@a-line138.supra.net)
12:53.13RoyKanyone that knows the diff between basic h263 and h263 cif 190?
12:53.53*** part/#asterisk styx2005 (~styx2005@a-line138.supra.net)
12:54.11DeanHcan some please help
12:54.21DeanHI am running a packteer etc
12:54.27DeanHwhat ports do sip normaly use ?
12:54.37file5060 UDP, and 10000-20000 UDP
12:54.54RoyKsee rtp.conf for control over the latter range
12:56.52DeanHta
12:58.12RoyKFeb 28 13:50:53 NOTICE[13716]: rtp.c:491 ast_rtp_read: Unknown RTP codec 127 received
12:58.14RoyKwtf?
12:58.56*** join/#asterisk sangee (ravi@209.250.129.135)
12:59.05DeanHwhat does the average call use in bandwidth
12:59.22filedepends on codec
12:59.27filecan range from 22Kbps to 80Kbps
12:59.59DeanHg.711
13:00.03RoyKcould even have been lower if someone thought about doing variable apcketization
13:00.14file80Kbps
13:00.20RoyKg.711 is 64kbps + overhead = 80kbps including IP, RTP and UDP
13:00.35DeanHis that kilo bits per second
13:00.39RoyKDeanH: http://www.newport-networks.com/pages/voip-bandwidth-calculator.html
13:00.43RoyKDeanH: yes
13:00.44fileyes
13:01.04DeanHThanks !
13:03.42*** join/#asterisk shanky (~shanky@238.Red-80-33-29.pooles.rima-tde.net)
13:03.49shankyhi to everyone
13:04.51shankyI just want to know waht exactly the realm variable is for
13:06.07shankyand what about externip?
13:06.31sangeeI did install the g929 codec license, but when i use the g927 codec i got "No compatible codec"
13:06.37sangeewhat could be the issue?
13:06.53*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
13:07.08*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
13:08.28shankyI have read http://www.voip-info.org/wiki-Asterisk+config+sip.conf but I couldn't understand it
13:08.40*** join/#asterisk styx2005 (~styx2005@a-line138.supra.net)
13:08.49*** part/#asterisk styx2005 (~styx2005@a-line138.supra.net)
13:09.05eivindtrHi all. Anyone with two boxes with sangoma-cards and a crossover E1 between them? Need tips on zaptel.conf...
13:10.17eivindtrOh thanks  RoyK !
13:13.22sangeei did install G729 codec, but I got this error "No compatible codec", what could be the issue?
13:13.51fileyour other side isn't saying it supports G729, so asterisk is throwing a fit
13:16.34sangeemy ip phone support G729 codec (i am calling from my ip phone)
13:16.51filelemme explain why the above error occurs
13:16.59visik7maybe it's disabled
13:17.07fileasterisk compares it's own codec settings with the codecs that the phone sent
13:17.22fileif they have nothing in common, the above error "No compatible codec" occurs
13:18.02sangeei put this line this line in sip.conf "allow=g729"
13:18.18fileput it in the general section too
13:20.04RoyK[Sim]: ping
13:22.08sangeethx, now working
13:22.45*** join/#asterisk DevilFish (~me@staff211.qtm.net)
13:25.04bjohnsonshanky: is there a question?
13:25.26DevilFishhello all :)
13:26.13DevilFishwe tried to fire up one of our Asterisk boxes on friday and go live and my worst nightmare came true
13:26.18*** join/#asterisk kamran (~kamran@mbl-82-51-9.dsl.net.pk)
13:26.26DevilFishnot sure what to do but I poseted about it somtime ago
13:26.28DevilFishhttp://lists.digium.com/pipermail/asterisk-users/2005-January/083456.html
13:26.38kamranhello
13:26.44DevilFishif anyone has and idea about this please let me know what you think
13:27.36kamranany one used AbsoluteTimeout(seconds) application
13:28.02wasimkamran: most everybody would
13:28.23wasimkamran: its a Good Idea (tm) as a safety net
13:29.31kamrani want to use but it is not working i want to terminate my call after some time
13:30.07kamranbut it is not sending bye after specified time
13:30.28kamran<PROTECTED>
13:30.29kamran| 12 | default | _.          |      101 | Hangup          |              |
13:30.29kamran|  9 | default | _.          |        1 | Answer          |              |
13:30.29kamran| 10 | default | _.          |        2 | AbsoluteTimeout | 30
13:30.32wasimkamran: it won't, AbsoluteTimout will kill the channel
13:30.38wasimugh ... use pastebin
13:30.53kamranok
13:30.59wasimkamran: it won't send anything down the link, for that you need to Hangup()
13:31.12kamranok
13:31.14wasimkamran: think of it like a kill -9
13:31.21kamranok
13:31.22kamranthanks
13:31.31kamranis there any way to do this
13:31.46wasimuse Hangup()
13:32.08kamrani want hangup after specefic time
13:32.24kamranlike hangup after 30 min
13:32.26*** part/#asterisk l-fy (~diana@diana.null.ro)
13:32.54goatmilkfile: do you ever sleep?
13:33.12fileyes
13:33.38RoyKhi
13:33.38RoyKhttp://karlsbakk.net/videotest.log.gz
13:33.44goatmilkyou're always on irc.. what is the secret?!
13:33.45RoyKcan someone take a look at that sip deubg
13:33.51RoyKI can't get video over SIP
13:33.53RoyK:(
13:34.22filegoatmilk: not closing my IRC client.
13:35.05goatmilk:)
13:35.17wasimyay! fresh feta cheese!
13:35.20teemu-xkamran: use S(n)-option of Dial command when calling?
13:35.27goatmilkhey now.  none of that
13:35.50filebah
13:35.56RoyKhm. someone? please? SIP video probs.
13:35.58goatmilkjust kidding
13:36.19kamranis this work
13:36.57*** join/#asterisk [ro]nic3try (~iancu@81.181.199.39)
13:37.07kamrani ll take a look into it
13:37.19fileooh la la goatmilk
13:37.22fileI didn't know you felt that way
13:37.45DevilFishdoes anyone even have a bad idea about this?? I'll take anything at this point
13:37.46DevilFishhttp://lists.digium.com/pipermail/asterisk-users/2005-January/083456.html
13:39.08*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
13:39.23filehi PCadach
13:39.28eipii dont know what i changed that i dont have musiconhold...
13:39.38PCadachHi file
13:49.36DeanHHi, what happens when you increase the jitter on a sip phone ?
13:49.50RoyKsound becomes terrible
13:50.06DeanHok
13:50.37DeanHThanks, do you know what VAD is and in / out band
13:51.32*** join/#asterisk ke4qqq (~savirc@static-cb-68-115-212-156.spa.sc.charter.com)
13:51.42*** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca)
13:52.05lyroyOn the Digium S100I how many analog phones can I put on a single RJ-11 port?
13:52.19RoyK10000000
13:52.26lyroyhehe
13:52.34lyroyserious?
13:52.39RoyKdunno
13:52.40RoyKone
13:52.55RoyKnone, more likely, as the adapter sucks
13:53.16lyroyreally what do you recomend?
13:54.02*** join/#asterisk file (~file@mctnnbsah25-142166091003.nb.aliant.net)
13:54.06*** part/#asterisk file (~file@mctnnbsah25-142166091003.nb.aliant.net)
13:54.12*** join/#asterisk file (~file@mctnnbsah25-142166091003.nb.aliant.net)
13:54.36*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
13:56.26PoWeRKiLL!seen coppice
13:56.46RoyK~seen coppice
13:56.59jbotcoppice <~chatzilla@245.195.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 1d 46m 36s ago, saying: 'Zeeek: an engineer in India probably gets about 1/5 as much as a US one, so 500xcost would be a little expensive :-)'.
13:56.59ariel_lyroy, what would you like to do with adapter.
13:56.59RoyK~lart PoWeRKiLL
13:57.06PoWeRKiLL:)
13:57.18RoyKPoWeRKiLL: did you see his FoIP article?
13:59.37sangeewhat is the advantage to put SER as proxy in Asterisk?
13:59.49lyroyariel_ replacec my actual analog phone line with a voip service
14:00.10lyroyI have like 10 phones so I don't want to buy 10 adapters
14:00.26ariel_Well the iaxy is something you can plug the phone into but one at a time.  You have not gotten any yet?
14:00.56ariel_lyroy, have you setup an asterisk server before?
14:01.03RoyKsangee: perhaps putting a SER proxy in front of asterisk......
14:01.11lyroyit s done ariel
14:01.16ariel_how many incoming phone lines are you needing to connect as well?
14:01.36lyroyonly one
14:02.12ariel_If you need 10 analog ports you will either need to get a channel bank or get some sipura type devices.  The sipura will work better in your own network and cost less then the iaxy.
14:02.29bjohnsonsangee: most people don't need it.  Usually just when * is behind nat
14:02.39lyroywich model do you recommend?
14:02.47*** join/#asterisk didz_ (didz_@200.218.192.52)
14:03.28bjohnsonI have 3 non-powered phones on the fxs of a SPA 3000
14:03.37*** join/#asterisk seong (~seong@218.208.204.215)
14:03.38sangeeyes
14:04.40sangeesome people said it good to put SER as proxy (in front of asterisk)
14:05.24bjohnsononly if you need it .. many people do not.  Depends what you're trying to do
14:05.40ariel_lyroy, you will need to get 5 sipura 2000 if that is the route you want to take unless you need one at each phone desk?
14:05.42Hmmhesaysreal good if you are trying to route eleventy billion calls
14:05.48RoyKsangee: because SER scales a LOT
14:06.04RoyKsangee: and then you can have lots of asterisk boxes behind that to gateway, do app servers etc
14:07.43sangeeaccually i want to this 5000 softphone clients registered to my asterisk box and they can make pc tp phone call
14:08.00sangeedo i need SER?
14:08.15RoyKnot "need" but perhaps "want"
14:08.24RoyKsangee: pc to pstn?
14:08.29sangeeyes
14:08.39RoyKthen read "asterisk at large"
14:08.42RoyKarticle on the wiki
14:08.47RoyK~wikki
14:08.51RoyK~wiki?
14:08.52jbotsomebody said wiki was http://www.voip-info.org
14:08.53sangeeokay
14:09.25sangeethen i don't need ser?
14:10.43bjohnsoninteresting: http://www.eezeephone.com/index_files/pap2.htm
14:11.04filebjohnson: why is that interesting now?
14:12.19bjohnsonhadn't seen NA's for sale at a store before
14:12.56fileoh, they're out there
14:13.04ariel_bjohnson, there providing 1 month service via there voip service.
14:13.20bjohnsonlyroy: I have SPA units but that link to a linksys is supposed to be the same hardware as the SPA 2000
14:13.50bjohnsonariel_: yes .. but no contract .. so that's a GOOD price
14:14.05ariel_bjohnson, yes as long as it's not locked to them.
14:14.10bjohnson2 fxs for $70 including shipping
14:14.55ariel_bjohnson, it's less at voipsupply http://www.voipsupply.com/advanced_search_result.php?search_in_description=1&keywords=Pap2-na&osCsid=0c3434ac98c435eb9c90dad92ef83c6a
14:15.10ariel_in any case I have to go to a customer see you all later.
14:15.59bjohnsondescription includes: This unit is a "NA" unlocked version capable for use with any VoIP provider.
14:16.14Zeeek.
14:16.34*** join/#asterisk mutilator (~animenodv@65.111.201.79)
14:16.46clive-bjohnson...its only 2 fxo if you use g711, if you use g729, its 1 port
14:16.46roamer323bjohnson - quite a deal - and their atcom phone does iax2 now ... hmm
14:17.06clive-I mean fxs
14:17.07bjohnsonclive-: exactly like most atas
14:17.20clive-its a sipura copy
14:17.25eipiwhat could be the reason that musiconholds never starts when i put on hold the communication? (if i test with CMD it works, but in a call, no musiconhold)
14:17.26bjohnsonvoipsupply is $5 cheaper but you have to buy 5 at a time
14:17.32bjohnsonclive-: yes I know
14:17.41roamer323also - no commitment on buying 5 and signing up with linksys - also they're out of the same town as vonage :-)
14:17.50bjohnsonclive-: that is why I expect it to be okay for quality
14:18.03RoyKwtf?
14:18.03RoyKCapabilities: us - 0xc011e (gsm|ulaw|alaw|g726|g729|h261|h263), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x80000 (h263), combined - 0x8010e (gsm|ulaw|alaw|g729|h263)
14:18.16RoyKthis says "YES DO VIDEO" but video doesn't work ;(
14:20.32Darwin35anyone know what happen to max sobbila
14:20.34RoyKhm.....
14:20.43Darwin35he seems to have fallen off the planet
14:21.33*** join/#asterisk file (~file@mctnnbsah25-142166093180.nb.aliant.net)
14:21.49Darwin35the bsd port still has not been update =d to 1.0.5
14:22.09Darwin35its still 1.0.3 and does not work correct with broadvoice
14:22.12fileDarwin35: latest is 1.0.6 :p
14:22.29Darwin35ok so now we are further behind
14:23.04RoyKer
14:23.10RoyKonce more. Capabilities: us - 0xc011e (gsm|ulaw|alaw|g726|g729|h261|h263), peer - audio=0x32e (gsm|ulaw|alaw|adpcm|g729|speex)/video=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729)
14:23.30RoyKis the one sending INVITE the one who supports h263?
14:23.37RoyKor the INVITEd?
14:23.44fileboth ends have to say they support it...
14:23.56fileyour asterisk is saying it supports h263
14:23.59filebut your peer does not
14:24.04fileboth have to support it.
14:24.50RoyKSip read:
14:24.50RoyKINVITE sip:21970071@10.0.0.10;transport=udp SIP/2.0
14:24.50RoyKTo: <sip:21970071@10.0.0.10>
14:24.50RoyKFrom: 1000070<sip:1000070@10.0.0.10>;tag=f119e054
14:25.00filedo not, DO NOT PASTE
14:25.04RoyKso 1000070 does not support it?
14:25.07RoyKonly four lines
14:25.07fileif you must use a pastebin, http://www.pastebin.ca/
14:25.12RoyKonly four lines
14:25.17fileyour peer does not support h263
14:25.30filethe other end.
14:26.50eipiwhat could be the reason that musiconholds never starts when i put on hold the communication? (if i test with CMD it works, but in a call, no musiconhold)
14:28.16pashaheipi: "m" option set?
14:28.28SuPrSluGeipi:mpg123  ver 59r
14:28.33filem option does not need to be set
14:28.48fileeipi: do you see, on the asterisk CLI, the indication that musiconhold has been started on the channel?
14:28.49pashahfile: no?
14:28.59eipipashash option m work
14:29.00eipis
14:29.07filepashah: that's only for Dial... instead of ringing it provides musiconhold
14:29.12eipibut works while caller waits for called
14:29.27eipifile: no
14:29.46fileeipi: then your device didn't send the indication to asterisk to hold the channel, so asterisk isn't providing musiconhold
14:30.05eipiyes
14:30.05pashahfile: thanks
14:30.11eipithe only that i receive is: -- Stopped music on hold on
14:30.16eipi.......
14:30.36filepastebin an exact CLI log when you place the call on hold
14:30.43eipii remember you that works while caller waits
14:30.47eipiok
14:31.04fileyou mean when you Dial your phone, you used the m option and it worked fine?
14:31.10eipiyes
14:31.12filewell that's COMPLETELY different :)
14:31.22eipii know
14:31.53eipibut with that i suppouse that there isnt a problem with mpeg123 or something like that... musiconhold app is registered
14:31.59filecorrect
14:32.12filebut it's up to your device to tell asterisk that the call is on hold so it can play the audio
14:32.21fileif your device isn't doing, for whatever reason, then musiconhold won't work
14:32.55eipifile: Started music on hold, class 'default', on IAX2/
14:33.02fileI told you to pastebin it
14:33.15fileportions don't help me, I need to see a complete call as it goes through
14:33.46Darwin35well I think Maxium Sobila jumped ship on the asterisk project
14:33.57Darwin354 weeks and no responce from him
14:33.58fileDarwin35: Take over
14:34.25Darwin35I have requested acces to the tree but got no responce
14:34.38ZeeekYo wasim
14:34.42eipifile: http://pastebin.ca/6599
14:34.59shankyhi again
14:35.04Darwin35and I am also pissed off at snom now
14:35.16Darwin35they stole my pbx box design
14:35.28Darwin35looks just like mine
14:35.33eipii think that i have to turn on sip debug ;)
14:35.38fileeipi: your device isn't sending notification that the call isn't on hold, so asterisk isn't playing musiconhold
14:35.43fileer that the call is on hold
14:36.26*** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu)
14:37.04shankyif I do a sip show peers, I get some status "Unmonitored" from my sip account. I'm able to call from that line but I'm not able to receive calls, any suggestion?
14:37.20fileshanky: firewall or NAT, am I right?
14:37.42pashahshanky: or budgetone =)
14:38.11shankyfile: I can use that account with a sipura and it works fine, so I think is not NAT's problem
14:38.17shankypashah: budgetone?
14:38.24fileSipuras handle NATs MUCH better
14:38.31fileI love the Sipura NAT support, it's sexy
14:38.54fileshanky: but anyway... turn on qualify=yes for that peer and see if it punches a hole in the NAT... well, keeps it open
14:38.56pashahshanky: get the same with budge tone-100 all the time
14:39.20fileshanky: if it says it's unreachable... then it is your NAT not letting the packets through... get the device to reregister to open another hole and it should stay open...
14:39.33*** join/#asterisk the1` (jh@pppN2L5.evis.net.ph)
14:39.50shankyfile: ok, I'll try with qualify
14:43.12eipifile: how i can identify a hold message in sip messages?
14:43.58fileit'll be another INVITE
14:46.43*** join/#asterisk bjohnson_ (~bjohnson@jecinc.tor.istop.com)
14:46.44Hmmhesaysugh, why did I not see this before.. it was so simple
14:48.44*** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34)
14:50.18*** join/#asterisk sine (~sine@p54BFDDD6.dip.t-dialin.net)
14:51.08*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
14:51.08*** mode/#asterisk [+o bkw_] by ChanServ
14:51.11*** join/#asterisk Yoda-BZH (~yoda-bzh@80.125.121.161)
14:52.30*** join/#asterisk convey (~test@208-216-127-234.cust.gti.net)
14:53.10*** join/#asterisk styx2005 (~styx2005@a-line138.supra.net)
14:53.25conveyHere is a stupid question, how do I figure out what version of * I am running?
14:53.51puzzledshow version
14:54.38conveyIt says CVS-HEAD, how does that relate to a version?
14:54.49puzzledit tells you a date too
14:55.20convey2-23-2005
14:56.13puzzledso you are running cvs head from 2-23-2005. cvs head doesn't have a version. only stable does
14:56.56conveyI was reading that * realtime will be implemented in 1.1.0, I want to put my configs in a database and I am trying to determins the best approach.
14:57.29*** join/#asterisk zoa (~zoa@pirus.securax.be)
14:57.33zoaelloooow
14:57.54puzzledconvey: then wait for the next major stable release or continue to use cvs head
14:57.58puzzledhi zoa
14:58.02jedirlwhat is * realtime and what improvements will it provide?
14:58.10RoyKmiiiiaaaaaooooowwwwww, zoa
14:58.17zoahehe
14:58.20filezoa: explodifying!
14:58.25zoaroyk did you see the sip jitter buffer ? :)
14:58.35*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com)
14:58.39puzzledjedirl: read the docs in cvs head
14:59.02zoahttp://www.astertest.com/downloads/scx-testlab.jpg --> our small testlab :)
14:59.26filewhat the heck are all those
14:59.28jedirlpuzzled: I was just asking for a small overview, thnks anyway
14:59.46filewhat types of machines...
14:59.50zoathose are all asterisk machines :)
14:59.56RoyKzoa: SIP JITTER BUFFER?
14:59.57fileyes, but specs :p
14:59.59RoyKwahooo
15:00.01zoasmall via nemeiah or so
15:00.04RoyKin cvs?
15:00.04zoa800 mhz
15:00.06fileI thought so
15:00.08zoano no
15:00.09zoanot in cvs
15:00.13puzzledzoa: clean up those wires you slacker
15:00.16RoyKzoa: where?
15:00.27zoaits somewhereo n astertest until we find time to do a diff and post it on mantis
15:00.45zoaits ready for testing, but has some known issues
15:00.53zoaand i suspect one of them to be in stevekanns code
15:01.08*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)
15:01.08*** mode/#asterisk [+o anthm] by ChanServ
15:01.12zoaso warning DO NOT TRY IT ON YOUR CUSTOMERS PC!
15:01.18RoyK:)
15:01.22zoait will break
15:01.22RoyKwe have a test server
15:01.24zoabut it works
15:01.40RoyKas long as it's all usermode, it can't panic the box either
15:01.44zoait better does as we already spent a user on it
15:01.46zoaa
15:01.49zoaa fortune io mean
15:02.12RoyKdo you have a url?
15:02.22RoyKis the patch against HEAD?
15:02.31zoaits not a patch, those are full files now
15:02.40zoamade it really quick
15:02.43zoaits not even cleaned up
15:02.52zoaits what i found on the programmers server :)
15:03.12RoyKcan't someone just cvs diff -u?
15:03.12zoahe will make a patch when hes back
15:03.16zoayeah you could
15:03.17zoabut i wont
15:03.21zoatoo much other things to do
15:03.27RoyKbut url?
15:03.35zoawww.astertest.com/downloads/
15:04.00znoGhey, my DTMF tones don't seem to be going through a SIP connection. Any ideas?
15:04.23RoyKznoG: don't use inband dtmf on complex codecs?
15:04.27MikeJ[Jayden]znog, what kind of device, and what dtmf modes
15:04.38jedirlznoG: using inband?
15:04.43MikeJ[Jayden]hehe...
15:04.55*** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl)
15:05.05mutilatorwhats a good linux compatible laptop?
15:05.08zoahttp://www.astertest.com/forum/viewtopic.php?t=13 its also there
15:05.10MikeJ[Jayden]in #asterisk, you don't just get answers, you get beat down with them
15:05.12zoaibm is known to be good
15:05.15zoaas well as dell
15:05.16znoGwhat exactly is inbind DTMF? :)
15:05.16MuppetMasterHello.
15:05.22znoGusing a Sipura SPA-2000
15:05.26znoGinband even
15:05.30MuppetMasterHas anyone managed to get Goto and GotoIf working with Realtime out of a MySQL DB?
15:05.43MikeJ[Jayden]inband means the dtmf is actually in the audio stream
15:05.58znoGoh, right.. and if it's not in the audio stream, where else would it be? :)
15:06.11MuppetMasterznoG:  Out of band in the signaling
15:06.17EssobiznoG ULAW and ALAW are the only two codecs that uspport in-band DTMF properly.
15:06.18jedirlin the signaling "stream" (SIP)
15:06.41znoGah, i am using GSM
15:06.50jedirlgsm is not supported if you use inband, then
15:06.54MuppetMasterznoG:  Need to go with an out of band signaling then.
15:06.55EssobiSo you're doing it in the control channel.
15:07.03znoGMuppetMaster: and how is that configured?
15:07.15EssobiUse OOB as opposed to IB dtmf.
15:07.15jedirlI have problems with out of band signaling and asterisk, while using a teles VoIP gateway
15:07.46EssobiYup.
15:07.46MuppetMasterznoG:  I have Realtime working for the bulk of my extensions.conf/iax friends & peers / SIP friends /etc.
15:07.46EssobiSounds like a problem then.
15:07.46EssobiMuppetMaster Nice.
15:07.46MuppetMasterznoG:  Just when I use it with a Goto in the extensions table, I get an error everytime.
15:07.48jedirl(asterisk ignores my dtmf's if they are out of band)
15:07.50EssobiWhat error?
15:07.55MuppetMasterznoG:  If I use the same Goto in the same logic in a static file, it works fine.
15:08.03MuppetMasterEssobi:  Just a moment and I will re-generate.
15:08.41*** join/#asterisk TheEmperor (TheEmperor@218.111.49.173)
15:08.43EssobiHmm.. Can you (forgive me for asking.. I havn't used RT yet) generate a macro in a static config to do what you need and call it from the realtime landing?
15:08.56znoGi've no idea as I'm trying to setup OOB myself
15:09.12jedirlznoG: go sip.conf and take a look
15:09.17EssobiOOB works fine with my Cisco gateways. :)
15:09.27jedirlEssobi: with my teles gateway it doesn't
15:09.52*** join/#asterisk ^login^ (~avl@star.ukr.net)
15:09.52Essobisorry to hear that.
15:09.56jedirlI'm forced to use alaw
15:09.57^login^hello
15:09.57znoGso RFC2833 is OOB?
15:10.04fileyes
15:10.10fileit's sent in the RTP stream as packets
15:10.24jedirl(which is not a problem, because the gateway and the asterisk machines are side-by-side connected by ethernet)
15:10.27Essobifile WHAT WHAT?
15:10.33^login^is there avalaible the mechanism to detect busy tone over h323 call?
15:10.37fileEssobi: whaht! WHAT!
15:10.46jedirlOOB signals on RTP?
15:11.00MikeJ[Jayden]zong: http://www.voip-info.org/wiki-Asterisk+sip+dtmfmode
15:11.02fileinband sends it as pure audio in the actual audio stream packets
15:11.07Essobiif rfc2833 is OOB it's not in the RTP stream.
15:11.09filerfc2833 sends it as a different type of packet
15:11.17filebut it's still in the RTP
15:11.17RoyKzoa: from which cvs date are these files?
15:11.22fileinfo sends it as a SIP method
15:11.30zoavery recent
15:11.34zoa16th or so
15:11.40MikeJ[Jayden]file, your so good with people... :)
15:11.51EssobiAh, I was under the impression it was always in the control channel as opposed to the RTP stream.
15:12.11jedirlI was under the same impression
15:12.13Essobithat way if you did a re-invite you could still highjack button interception.
15:12.18filenah
15:12.20Essobithat's how SCCP works anyways. ;)
15:12.21jedirlbut seems that file is right
15:12.30*** join/#asterisk eKo1 (~bernd@63.245.57.70)
15:12.37ManxPower~docs
15:12.38jbotDocumentation 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
15:12.47fileare you all surprised?
15:12.56jedirlI am hehe
15:13.01*** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63)
15:13.01MuppetMasterEsobbi:  Something like this:  switch => Realtime/from_pstn@realtime_ext
15:13.17MuppetMasterEssobi:  Where realtime is configured properly via extconfig.conf
15:13.19EssobiI'm not a big SIP guy anyways.. H323, MGCP and SCCP.
15:13.29EssobiMuppetMaster Ouch.. you read what I said above?
15:13.29bkw_file 2833 is still sent via RTP
15:13.38bkw_just not in the same stream as the voice
15:13.41filebkw_: that's what I said, they didn't believe me
15:13.45jedirlEssobi: which H323 do you use?
15:13.45bkw_haha
15:13.47bkw_ya
15:13.49bkw_I see that now
15:13.58bkw_I HATE IT
15:14.00Essobijedirl The ones that curl up and die. ;)
15:14.01MuppetMasterEssobi:  Not sure?
15:14.02bkw_that rfc should be SHOT
15:14.13Essobibkw_ That's is FRICKING RETARDED.
15:14.13jedirlEssobi: sorry?
15:14.18filebkw_: hush hush
15:14.25bkw_Essobi, whats retarded?
15:14.30EssobiH323 needs to be drug out behind the bike shed and shot.
15:14.37jedirlhehee
15:14.58RoyKEssobi: not really. but chan_sip and it's author should....
15:15.03ayzeequestion: who makes media gateways like the smartnode 2400 (for example) from Patton?
15:15.07`SauronEssobi: There are many protocols that should be shot...
15:15.14bkw_RoyK, so you're talkin about mark?
15:15.15Essobibkw_ the fact the OOB dtmf is in the RTP stream instead of the SIP session control.  you can't intercept DTMF when a re-invite occurs, which is lame.
15:15.16znoGi'm not sure what's going on here, but even using ALAW/ULAW the DTMF tones are not working.
15:15.27ayzeeand are there fxo cards that can handle a pri?
15:15.30bkw_Essobi, thats what dtmfmode=info is for
15:15.32RoyKbkw_: no. jerjer.
15:15.35bkw_info is sent as sip messages
15:15.35Essobi:)
15:15.36filechan_Sip needs some tender loving care
15:15.42`Sauronayzee: Yes.
15:15.42jedirlayzee: digium's?
15:15.45bkw_Essobi, did you not know this?
15:15.53jedirlbkw_: info == rfc
15:16.00jedirlbkw_: (I think)
15:16.06fileinfo is not rfc
15:16.08bkw_info != rfc2833
15:16.13ayzee`Sauron : which other companies make acceptable media gateways?
15:16.14filetwo vastly different things
15:16.15jedirlsorry then :)
15:16.17bkw_yep
15:16.17ayzeejedirl : looking on site now
15:16.20Essobibkw_ I'm not a big sip guy, like I said.  Just started using it two weeks ago.
15:16.21*** join/#asterisk Nukemizer (~Nuke@65.103.231.133)
15:16.33bkw_Essobi, you'll see why its like that later on
15:16.35*** part/#asterisk bonez39 (~aint@c-67-166-77-14.client.comcast.net)
15:16.39jedirlwhat's info, then?
15:16.41*** mode/#asterisk [+o file] by bkw_
15:16.44`SauronEssobi: Hehn. And I just got sccp working here 2 weeks ago... :)
15:16.50EssobiCisco hateways and 79XX's use dtmf info?
15:16.57bkw_nope
15:16.58*** join/#asterisk jsolares (~jsolares@200.30.141.85)
15:17.00EssobiDamn.
15:17.01bkw_whats wrong with 2833?
15:17.07EssobiSee that's jacked up. :)
15:17.09jedirlbkw_: my teles does not like it
15:17.17*** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com)
15:17.18fileI HAVE THE POWER!
15:17.22fileNOW YOU WILL ALL FEEL MY WRATH!
15:17.24file...later
15:17.25bkw_jedirl, get an rtp debug and i'll show you why
15:17.30*** part/#asterisk styx2005 (~styx2005@a-line138.supra.net)
15:17.35EssobiCan't provide inline DTMF feature control after a re-invite occurs.
15:17.48jedirlbkw_: I'm using alaw so it's not a problem *at the moment*
15:17.56ManxPower't' and 'T' break reinvites.
15:18.03EssobiYup.
15:18.04filerfc2833 is nice, when it works
15:18.06EssobiAnd that's GHETO.
15:18.11filekey words, 'when it works'
15:18.15ManxPowerWell, more correctly "prevent reinvites from happening"
15:18.19*** join/#asterisk jterrero (~jterrero@mcse-irc.isys-networks.com)
15:18.31Essobit and T are ghetto.
15:18.39jedirlwhich h.323 is more stable, asterisk's or oh323?
15:18.40Essobi:)
15:18.45Essobioh god
15:18.48ManxPowerJearil, Yes.
15:18.56filejedirl: yes.
15:18.57Essobiif you want to have more then 2 concurrent calls, don't touch oh323.
15:18.57bkw_Essobi, take that negative attitude out of here please.. we are accepting patches to fix that if you wish to contrib.
15:18.58ManxPower..er.. jedirl: Yes.
15:19.05ayzeebut will the TE410P handle 24 channels?
15:19.07Essobibkw_ :)
15:19.08MuppetMasterUsing Goto in a Realtime config I get this error:  Feb 28 16:18:48 WARNING[15561]: pbx.c:5817 ast_parseable_goto: Priority 'from_pstn,s,1' must be a number > 0, or valid label
15:19.09ayzeewell 23 channels.. but ya
15:19.17jedirlEssobi: ok
15:19.21mishehubah.
15:19.24EssobiI give up my share of love too..
15:19.33bkw_MuppetMaster, show us the database dump of the column
15:19.49mishehubkw_: any way you can push cluecon back to the end of may?  ;-)
15:19.50MuppetMasterWhich column?
15:20.02mishehuerr I guess that's actually pushing it forward
15:20.55jedirlI was just asking, I didn't want to contribute to any dispute... if there's any developer here of any of the h.323 versions, I'm sorry for asking that
15:21.32ManxPowerayzee, My TE405P handles 96 channels
15:21.33*** join/#asterisk yaout (eric@CPE-65-30-220-56.wi.rr.com)
15:21.41bkw_mishehu, you'll have to talk to anthm
15:21.59bkw_jedirl, patients my child
15:22.14MuppetMasterbkw_ Have a look here:  http://82.92.73.185/temp.txt
15:22.16mishehubkw_: I'll do so.  first * event I  know of in chicago, and I'll be gone for most of june overseas.  heh.
15:22.19bkw_all better soon it will be
15:22.35MuppetMasterbkw_ The goto goes to a valid config section, this is just for testing.
15:22.43bkw_MuppetMaster, don't use that
15:22.46*** join/#asterisk pimpwell (~pimpwell@ool-44c6ab45.dyn.optonline.net)
15:22.49bkw_EVIL
15:22.54bkw_EVIL EVIL EVIL
15:22.56MuppetMasterbkw_ Don't use what?
15:23.01bkw_pbx_realtime is so hackish
15:23.15bkw_dont get me started.. i'll never shut up
15:23.26RoyKbkw_: then what _is_ good
15:23.31`Sauronc'mon, Brian.. tell us... ;)
15:23.37MuppetMasterbkw_ Works fine for me, only the 'Goto' and 'GotoIf' do not.
15:23.40mishehuNOOOOO!!!!
15:24.05bkw_RoyK, you really wanna see what we hashed out so far this weekend?
15:24.10MuppetMasterbkw_  Besides, RT made it into the distro, so somebody must have approved...
15:24.14ayzeeManxPower : ok. any other choice of companies for media gateways, except Patton?
15:24.33Essobijedirl I don't know who worked on oh323, but Jerjer made the one in asterisk, as and far as the design model went a year ago, JerJer's was the only one I could get any power in concurrent use out of.  If this is the same today, I don't know, but the opinion did come from experience.
15:24.39RoyKanyone that knows how I can, from SQL, do a dial-plan-like lookup like "SELECT blah FROM blah WHERE cid LIKE/BLAH dialplan-like-something
15:25.05EssobiRoyK app MySQL
15:25.13Essobiit's on contrib
15:25.24mishehuit's in addons actually
15:25.27Essobior the postgress app
15:25.33RoyKEssobi: er. I meant the actual SQL
15:25.40RoyKEssobi: I'm doing this via AGI
15:25.46MuppetMasterRoyK:  Maybe this?  http://www.voip-info.org/wiki-Asterisk+cmd+MYSQL
15:25.47pimpwellquestion:  does the .call file contain everything needed to control "the flow" of the call when it comes to pre-recorded messages?   example:   " hi you have reached my company, please press 1 for sales, 2 for billing "   <user presses 2>  please wait until the next representitive is available"    so basically: play_intro()  if(user presses 2) { play_billing_wait() }
15:25.51jedirlI do it with AGI, too
15:26.01RoyKEssobi: but I have no idea how to do a match with SQL against something like a dial plan
15:26.15jedirlRoyK: if using MySQL, RLIKE
15:26.29RoyKhm...
15:26.32mishehujedirl: what is RLIKE ?
15:26.41jedirlmishehu: regular expressions like
15:26.43mishehuI've used LIKE often, never RLIKE
15:26.46mishehujedirl: ah.
15:26.53jedirlis the way astcc performs the lookups
15:30.52*** join/#asterisk sangee (ravi@209.250.129.135)
15:32.25RoyKer
15:32.28RoyKrlike doesn't work
15:32.31RoyKgrr
15:33.11zoaroyk, how is the sip jitter buffer ? :)
15:33.25RoyKzoa: need to wait till tomorrow
15:33.54pimpwellsorry for doing this again but it may have not made it, I have been having problems w/ mIRC recently
15:33.56pimpwellquestion:  does the .call file contain everything needed to control "the flow" of the call when it comes to pre-recorded messages?   example:   " hi you have reached my company, please press 1 for sales, 2 for billing "   <user presses 2>  please wait until the next representitive is available"    so basically: play_intro()  if(user presses 2) { play_billing_wait() }
15:34.06bjohnson_wish I could figure out how to make some money or get some hardware out of this article I'm writing
15:34.37RoyKsay 'spain' is '34', spain cellphone is '34686','346' etc and phone no is 34686387036. that means it can match several columns if just matching the first char, but here I only want the match against 34686, see
15:34.42eKo1About what bjohnson_ ?
15:34.59JohnnyCwhat does it mean DID support ?
15:35.16*** join/#asterisk viLeR (1000@ip-33-7.telesat.com.co)
15:35.41bjohnson_eKo1: voip for beginners in Canada .. vonage, primus, etc .. also pushing buying own ata and sign up for prepaid type accounts
15:36.16*** join/#asterisk Mike (~mike@201.135.48.217)
15:36.27pimpwellin the case of electricity going down noone will have a phone w/ voip
15:36.35pimpwellthat's too scary for me
15:36.35bjohnson_pimpwell: not true
15:36.41Mikeasterisk should include in the releases a changelog really
15:36.44pimpwellhow my modem won't turn on
15:36.57bjohnson_there are bigger service interuption threats than power outage
15:37.01bjohnson_pimpwell: UPS
15:37.15pimpwellwhich doesnt last all that long
15:37.26bjohnson_also, do you have a cell phone anyway?
15:37.29pimpwellno
15:37.59bjohnson_so .. that is one point I'm trying to make .. each person has to identify their own needs and not just follow someone else
15:38.10JohnnyCanyone knows what DID means ?
15:38.15*** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com)
15:38.18bjohnson_~DID
15:38.19jbotextra, extra, read all about it, did is Direct Inward Dialing
15:38.19JohnnyCsupport for DID ?
15:38.24MuppetMasterJohnnyC:  Direct Inward Dialing
15:38.35JohnnyCwhats that ?
15:38.46bjohnson_usually refers to a phone number
15:38.52JohnnyCwhan does it mean ? you can directly call a number inside ?
15:39.03bjohnson_eg DID in Toronto is a Toronto phone number
15:39.04MuppetMasterJohnnyC:  Meaning an externably addressable phone number that terminates directly to an extension.  Like a publically addressable IP address that terminates to a machine with port forwarding under NAT.
15:39.22JohnnyChmmm ok
15:39.24MuppetMasterIn very basic terms.
15:39.31JohnnyCso I can MAP a number to an extension
15:39.39JohnnyCthats DID support
15:39.46MuppetMasterJohnnyC:  More or less.
15:39.50bjohnson_it can mean a lot of things depending on what aspect of the system you are looking at .. eg. from * an extension # could be considered a DID
15:40.10JohnnyCI just want to setup a system for our office
15:40.15JohnnyC3 numbers
15:40.16*** join/#asterisk file (~file@mctnnbsah25-142166093180.nb.aliant.net)
15:40.17JohnnyCISDN
15:40.19JohnnyCsimple
15:40.21MuppetMasterJohnnC:  http://en.wikipedia.org/wiki/Directed_inward_dial
15:40.34bjohnson_but most people shopping for DID from voip suppliers are talking about pstn phone numbers
15:40.36JohnnyCcall number A -> extension B -> transfer to extension C
15:40.58jedirlif anyone needs did's in spain, just ask me
15:40.59jedirl:)
15:41.44MuppetMasterjedirl:  Sevilla?
15:41.59MuppetMasterjedirl:  Have one from Libretel in Madrid...but Sevilla would be even better.
15:41.59JohnnyCDID is nice
15:42.02jedirlValencia
15:42.11bjohnson_is 382-5666 available?
15:42.19MuppetMasterjedirl:  Unfortunately, Valencia won't do me much good.
15:42.27jedirlMuppetMaster: ok :)
15:42.45MuppetMasterjedirl:  Valencia is very nice though.
15:42.55jedirlit is :)
15:43.17jedirl=)
15:45.36ManxPowerI thought a DUB-LOON was a crazy irish person
15:49.22*** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com)
15:49.43HitTopanyone using fc3 for asterisk?
15:50.16MuppetMasterHitTop:  I swore of FC3
15:50.58bjohnsonHitTop: I use it at home
15:51.59HitTopoh.. i just had a problem to initialize zaptel on boot time.. but i think i've found the sol'n.. stupid me~_~
15:54.52*** join/#asterisk Matjing (Matjing@62.8.64.33)
15:56.01MatjingI've just Installed asterisk it works great does it have support for messaging
15:56.15Matjingand does anyone have a url that I can use?
15:56.21Matjingto guide me for now
15:58.38*** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net)
15:58.52*** join/#asterisk Yoda-BZH (~yoda-bzh@80.125.121.161)
15:59.34Yoda-BZHre
15:59.45*** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au)
16:00.45eKo1Matjing: What do you mean by 'messaging'?
16:00.51*** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it)
16:00.53*** join/#asterisk boch (~as24@200.59.172.98)
16:01.01mAsH`hi
16:01.11bochhi
16:01.51DeanHthey say that if my latency is higher than 300ms my phone wont register on there servers, is this true?
16:02.08loudno, it is not.
16:02.16loudsatellite links have over 500 ms and they do OK.
16:02.23Juggiedoes anyone have a url for nightly builds in tar form of asterisk from the cvs? i cant use cvs at the office.
16:02.36DeanHok who can I speak to at broadvoice ? who know ?
16:02.38mAsH`anyone used automon in features.conf ?
16:02.38eKo1VoIP over satellit stinks though.
16:02.46DeanHas my phone will no work ?
16:02.46loudsometimes ..
16:02.55loudnot when you have a DA (Dedicate access) though.
16:03.13DeanHmy account and everything is 100% however I dont seem to register on there server ?
16:03.24DeanHI have 512k dedicated access to london
16:03.37loudDeanH, are you in the west coast ?
16:03.45DeanHna, south africaq
16:03.50eKo1loud: DA is mad expensive though.
16:03.56loudit is, yes.
16:04.16eKo1You might as well buy your own satellite.
16:04.25DeanHin South Africa it costs R 50 000 for 512k or USD$ 8000 per month
16:04.26loudbut sometimes there's  no other choice. thats why satellite providers make lots of cash.
16:04.32loudha
16:04.51*** part/#asterisk hans (fugalh@falcon.fugal.net)
16:05.11parwow south africa data rates are quite high
16:05.21DeanHyip
16:05.42parbroadvoice not working for you?
16:06.02DeanHnope
16:06.08eKo1I was always under the impression that South Africa had a well developed telecom. infrastructure.
16:06.15Juggieno
16:06.19Juggiethey are all cell phones.
16:06.22Juggiehardly any copper
16:06.24DeanHanyonework work with the act 160 phone ?
16:06.25*** join/#asterisk BurnedOutGeek (~BurnedOut@216.215.202.4.nw.nuvox.net)
16:06.28eKo1Any fiber?
16:06.31DeanHyip
16:06.37Juggiepeople allways steal the copper
16:06.40DeanHno tmuch
16:06.42Juggieto make jewelery and shit out of
16:06.43Juggieto sell
16:06.44aggelosdoes anyone know status on nufone.net ?
16:06.44DeanHnot much
16:06.47Juggieso they stopped bothering
16:06.55eKo1So everything is mostly wireless then?
16:06.59Juggieyes.
16:07.03DeanHno thats banned !
16:07.13DeanH1 telephone operator !
16:07.26mAsH`anyone used automon in features.conf ?
16:07.32eKo1But how does South Africa connect to the rest of the world?
16:08.16Juggiefiber? :P
16:08.34DeanHfiber
16:08.40DeanHsat 3 cable to london
16:08.45`SauronCarrier Pidgeon
16:08.45aggeloshas anyone status nufone.net ?
16:09.18`Sauronaggelos: Chances are, since nobody has answered since the last time you asked, that we 1) don't know, 2) don't care, or 3) don't want to tell you.
16:09.31par:-)
16:09.55`SauronI'm betting on 1 and 2 myself.
16:10.40aggelosSauron: extreamly helpful of you.  time has gone by and maybe somebody has more info than I.
16:10.51aggelosmy machine is there and it's down.
16:11.05aggelosso I am realy realy realy interested in finding ANYTHING
16:11.16`SauronSeeing as I can see the last 2 times you asked, enough time for something substantial to happen, hasn't gone by.
16:11.32aggeloshe, you are probably fresh here,
16:11.32`SauronCall them up, if it's so important.
16:11.42aggelosasked already 6 times today,
16:11.59aggelosI did, noone answers,
16:12.04`Sauron`Sauron signed on Tue Feb 22 03:28:03 2005
16:12.09`SauronI doubt that qualifies as "fresh here"
16:12.10cbachmanaggelos, your machine is hosted with nufone?
16:12.16aggelosyes,
16:12.31*** join/#asterisk human39 (~human39@chewie.fyi.net)
16:12.53aggeloswell, sauron, maybe you did not get my questions in the first couple of times, and that's ok. but if you don't care to answer, simply DONT
16:13.01cbachmanI'm curious, how does that work?  I haven't seen them offer hosting or  colocation anywhere.
16:13.29aggeloscolocation is perfect. really nice connection (6 - 8 ms pings to Google)
16:13.50aggeloswe had 442 days of uptime,
16:13.54aggelosuntil last night
16:13.56Nuggetsounds like it's less than perfect if they didn't give you a support phone number you can call.
16:14.16`Sauronaggelos: IF you're colo'ed there, you should have OOB way to contact them. If you didn't, that's your loss.
16:14.18loudand your box practially dissapeared.
16:14.19`SauronBetter luck next time.
16:14.29aggelosnobody answers the numbers that I have, I get the nufone voicemails
16:14.30human39morning all:  Question - I brought my phone into work to test it out on the network here.  It is registered with the * server via SIP.  When I make a call I can hear the person just fine but they cannot hear me.  My asterisk box is at home behind a NAT with sip forwarded, again..the phone is registered.  Does this have anything to do with the RTP ports might be misconfigured?
16:14.34cbachmanaggelos, do they colo in chicago or MI?
16:14.45aggeloschicago,
16:15.01aggelos`Sauron: Projected time to restoration of website: Sunday Feburary 27th 9:00PM (+5 GMT) --- according to their website...
16:15.26eKo1human39: You have NAT problems.
16:15.37cbachmanoh! Hmmm.... I Might want to talk with them.  I'm currently in chicago but colo'd at Anet's data center way out in the burbs from where I live.
16:15.54aggelosI hoped someone else is hosting with them and has more news than I
16:16.10human39even though the phone is registered to the * box fine?
16:16.34`Sauronyes
16:17.00human39any suggestions to try?
16:17.02eKo1human39: Do a 'sip debug' and look at the headerss.
16:17.13mishehucbachman: you are doing business with anet?
16:17.18mishehumy condolences.
16:17.20*** join/#asterisk kamran (~kamran@mbl-82-51-9.dsl.net.pk)
16:17.24kamranhello
16:17.41*** join/#asterisk Mike (~mike@201.135.48.217)
16:17.51kamranany developer
16:17.58cbachmanmishehu, yes, I've had a box colod with them for several years now, before they built up their  chicago datacenter.
16:18.17mishehucbachman: before Don sold out to another group of people, and the company went downhill from there
16:18.52mishehuI used to work for a company that was referring business to anet, and had a friend who worked there until about a year ago.
16:19.24cbachmanmishehu, things do appear to have gone downhill.  I don't use them for anything voip related, but i don't seem to get any answers to problems until I start escalating and calling VPs.
16:19.41human39eKo1, chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call 699693bc-1389c7b@XXX.XXX.XXX.XXX for seqno 101 (Non-critical Response)
16:19.48human39thats the only error Im getting
16:19.49bjohnsonhuman39: is the office behind nat too?
16:19.56mishehucbachman: you ever talk to a guy named Gene who was at anet tech support about a year ago?
16:20.21human39bjohnson, no Im on a static IP
16:20.33cbachmanI did talk with them last july or so about voice stuff, but even after a conference call they never got back to me with a quote.  One time they even didn't notice they hadn't cashed a payment check after 3 weeks.  Bizarre place.
16:20.47bjohnsonyou're phone has a public static ip?
16:20.55human39bjohnson, correct.
16:21.01cbachmanmishehu, I don't think so.  Honestly I haven't had much in the way of connectivity problems.  It's possible though
16:21.07bjohnsontry adding nat=yes to sip.conf
16:21.17*** join/#asterisk ayzee (mario@supermario.org)
16:21.57mishehucbachman: you're lucky then.  heh.  he was telling me how they kept changing their  methods of tech support and methods of business, and how the new owners really didn't give a crap like don did, and that don had no power anymore...
16:22.21mishehuI was wondering why people would deal with them, as there's so many other identical companies out there that dont' give a shit
16:22.21mishehuheh
16:23.05cbachmanmishehu, tell me about it.  I finally did get someone on the phone and recommended that they send out escalation lists to all of their current colo customers.  I was floored when my next bill had a copy.
16:23.47human39bjohnson, that didnt work.
16:24.08cbachmanmishehu, if my voip project ever  gains legs, I'll likely look for a different provider.  Leave a box there to forward stuff while DNS propogates and then pull my last box and cancel.
16:25.16MuppetMasterFigured out how to get Goto to work with Realtime.  Need to use '|' instead of ',' in the app_data field of the database table.
16:25.17cbachmanaggelos, btw, if it's useful, switch-1.nufone.net is pingable, so it appears at least part of their network in chicago is up.
16:25.47aggeloscbachman: yep, but the rest is down, thank you
16:26.37Juggiedid anyone compile cvs lately?
16:26.44DeanHok, does anyone have a sip server, that I can try register on
16:26.50DeanHto see if my phone is working ?
16:26.54Juggieits failing in dsp.c for me.
16:26.56DeanHpppllleeease ?
16:27.36bjohnsonhuman39: I've never had any luck connecting into an * server behind NAT
16:28.26bjohnsonDeanH: fwd
16:28.32bjohnsonor sipphone
16:28.47bjohnsonfree SIP servers for you to test with
16:29.14*** join/#asterisk file (~file@mctnnbsah25-142166093180.nb.aliant.net)
16:29.37bjohnsonhuman39: I ended up making 2 fwd accounts and using that as a proxy to use SIP -> FWD -> * via iax
16:29.38kamrani am using cvs it is working
16:30.30*** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
16:30.55DeanHsip phone
16:31.17*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
16:31.17DeanHbjohnson: I am using a sip phone
16:32.01human39hmm..the thing that is confusing me that registering with the server.  and i can make calls and such
16:32.40bjohnsonhuman39: don't rack your brain.  THAT is the problem with SIP + NAT
16:33.15bjohnsonDeanH: I know.  You said so.
16:33.20jterreroany decent asterisk consultant with references, please msg me
16:33.46human39I want to open the specific RTP ports to my * box, but my linksys router at home doesnt link lynx/elinks
16:34.16Juggiejterrero, what are you looking to have done?
16:34.25human39s/link/likes/
16:34.47jterrero30 phone implementation, Cisco 79xx series, 2 sites connected via p2p T1
16:35.03Nuggetwhat is a "p2p t1"?
16:35.08jterreropoint to point
16:35.11pimpwellquestion:  does the .call file contain everything needed to control "the flow" of the call when it comes to pre-recorded messages?   example:   " hi you have reached my company, please press 1 for sales, 2 for billing "   <user presses 2>  please wait until the next representitive is available"    so basically: play_intro()  if(user presses 2) { play_billing_wait() }
16:35.11human39point 2 point
16:35.12Nuggetoh, point to point.
16:35.20NuggetI'm thinking pirate 2 pirate, like kazaa or whatever.  :)
16:35.27Juggieprivate t1 with ip running sip or iax over.
16:35.55jterrerosip
16:35.56mishehucbachman: what type of voip stuff are you working on here in 'cowgo?
16:35.59jterreroiax2 to the pstn
16:36.10Juggiejterrero, thats a pretty simple install..... i run a few 7960's here, they can sometimes be a hassle however.
16:36.11MicH323Beginerquestion: Is the voicemail number (voicemail.conf) same as the users extension number? Or Should I map the extension to a different voicemail number
16:36.32bjohnsonDeanH: I told you 2.  don't pm me asking again
16:36.45jterrerohas anyone here worked with vonage? reliability is of big concern, i need to have minimal downtime, we have had 0  downtime with our current nortel pbx
16:36.50cbachman<PROTECTED>
16:37.19mishehuMicH323: you can have different voicemail boxes from extension numbers, though it can make the configuration more complex
16:37.28bjohnsonMicH323: totally different things.  you can link them if you wish
16:37.42cbachmanmishehu, asterisk seems to be the perfect solution, but the person I talk with is kinda "flaky" and it may never even get off the ground.  If anything, it's something fun to play with at the moment.
16:37.53Juggiejterrero, ip isnt downtime proof, no one is going to guarantee you 5 9's with ip... not unless everything is on site.
16:37.57bjohnsonjterrero: vonage is a pita to work with
16:38.21MicH323so voicemail syntax is 1234 (voicemail box) => 5678 (actual extention no), Name, email_recipiant ?
16:38.25bjohnsoncbachman: totally possible
16:38.50*** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au)
16:38.58mishehucbachman: I have one system setup for homeworkers of a small company I service, cut down on their cellphone bill drasticly...  have 2 or 3 other prospects right now for it.  it's definitely something these companies want, one central # for everyboyd.
16:39.07bjohnsonMicH323: check voicemail.conf for config info .. but looks good to me from memory
16:39.28jterrerothese 30 phones are a test phase, ideally we are going to implement 150 phones across 7 sites, we are going to use vonage for outbound.. PRI for incoming calls, we have had 0 downtime this year with our internet T1 and all of our point to points, we are going to keep everything at main site, my main concernt is how reliable is vonage, or any other carier
16:39.44jterreroand can any of the telco providers provide proof of uptime if asked ?
16:40.11HitTopi just installed web based voice mail.. but when i tried to login in a web browser .. it always gives "Login Incorrect!".. anyone got this similar problem b4?
16:40.18*** join/#asterisk yashax (~yasha_x@69.15.218.218)
16:40.30MicH323jterrerno: I have been using BroadVoice with little or no trouble
16:40.33bjohnsonjterrero: I doubt you'll ever get a good answer.  Everybody has bad vibes with somebody
16:40.35cbachmanmishehu, definitely lots of incentives to cut costs at the moment for places.  I have a supira spa-3k on order to play with just to get more experience with different things
16:40.53bjohnsonMicH323: you would be the only one
16:41.00Juggiejterrero, it depends on your connection to the provider, your isp, etc...
16:41.22DevilFishdoes anyone even have a bad idea about this?? I'll take anything at this point
16:41.23DevilFishhttp://lists.digium.com/pipermail/asterisk-users/2005-January/083456.html
16:41.26jterreroim doing BGP with 2 T1's to my ISP (cogent)
16:41.43MicH323lolz
16:41.50Juggiecogent? :P
16:41.51Juggieeww
16:42.01jterrero???
16:42.04Juggienow they have a bad rep.
16:42.14MicH323I guess I am not too demanding :)
16:42.21jterreroive had 0 downtime with them since i purchased service back in july
16:42.38yashaxGuys, (http://pastebin.ca/6601) Where do I insert the loop command to loop the Auto Attendand greeting and what the command should be? Than kyou.
16:42.45jsolaresjterrero: you might want to have two carriers for voip-pstn, and configure your asterisk box so in case one craps out it tries the other, and so your users see 0 downtime
16:43.30jterreroyeah thats what im planning... want to have 1 vonage account (they offer services specific for asterisk, 1000/month for 24 channels, 50000 minutes)
16:43.35jterreroand a PRI from verizon
16:43.38jterreroor a verizon reseller
16:44.00DevilFishanyone ever see "Failed to grab lock, trying again" type messages in their debug logs?
16:44.18DevilFishmy asterisk just seems to be randomly dropping calls to the PSTN
16:45.47Juggiejterrero, theres no problem with what you are doing, just you arnt going to get PSTN reliability.
16:45.58*** part/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34)
16:45.59Juggieyou need to have backup, such as a few backup pots lines you can dialout on, etc.
16:47.40*** join/#asterisk jtodd (~jtodd@ti.fox-den.com)
16:47.59bjohnsonjterrero: vonage offers service specifically for asterisk?
16:48.39jterreroyes, they also refered me to a asterisk consultant
16:48.58jterrerobut I want to have 2 consultants who are foreign to each other work on this project
16:49.15mishehuheh, I think that nufone's upgrade is taking longer than expected.
16:51.57yashaxGuys, (http://pastebin.ca/6601) Where do I insert the loop command to loop the Auto Attendand greeting and what the command should be? Than kyou.
16:53.20DevilFishyou want somthing like a timeout  maybe? like  exten => t,1,Goto(whatever)
16:54.04yashaxI want the auto attendand message to keep playing until they hang up
16:54.12Juggiethen use timeout
16:54.59*** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com)
16:55.04yashaxcan you please tell what command and where I should insert it?
16:55.25Juggiehas anyone seen, usr/bin/ld: cannot find -lidn when compiling cvs.
16:55.32Juggiei have the latest libidn
16:55.51Beirdomake sure the path to it is in /etc/ld.so.conf
16:57.31Juggiedo u know where it is generally installed?
16:57.38Juggieapt-get tells me i have the latest.
16:58.03Juggieack
16:58.08Juggieit would help if i install -devel :)
16:58.23*** join/#asterisk Casper_UA (~casper@as-2-22.ar43-2x.kharkov.ukrtel.net)
16:59.40Beirdoheh, yeah that too
17:03.20*** join/#asterisk thieumS (~darkmind@nanterre-7-82-229-210-142.fbx.proxad.net)
17:03.57bjohnsonholy moley .. CDN ATA sales places are popping out of the woodwork replying to my email to asterisk-biz
17:04.43bjohnsonand MAN .. they ALL have CRAPPY web sites
17:04.45Beirdoheheh
17:04.52Beirdoany in the GTA with a storefront?
17:05.15bjohnsonuniversal law of voip = all voip service and hardware provider web sites lack significant amounts of information of interest to potential customers
17:05.43MatjingI had meant to ask whether asterisk supports messaging (instant messaging) and what Id need to do to get lets say msn connecting on it
17:05.50Juggiecvs: app_addon_sql_mysql.c:164:36: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given
17:05.52Juggieany ideas?
17:06.29bjohnsonBeirdo: I just got one with an address at Mills & Finch but their web site has no info
17:06.59junky[work]Juggie: u need to pass 4 arguments to that function :)
17:07.22*** join/#asterisk Hmmhesays (negative3k@66.173.103.108)
17:07.50bjohnsoneven voipsupply replied to me
17:09.29Beirdowow
17:11.08Juggiejunky[work], thanks for pointing out the obvious :) not my code... i added an include to the Makefile and it worked... it needed asterisk/include
17:12.14*** join/#asterisk trym (trym@linux.debian.us)
17:13.36HitTopanyone had login problem with webvmail under fc3.. (i don find a problem on wblinux, but on fc3)
17:14.44*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
17:15.49Matjingdoes whether asterisk support messaging (instant messaging) and what Id need to do to get lets say msn connecting on it
17:16.31*** join/#asterisk numbone (~todd@c-24-129-204-233.se.client2.attbi.com)
17:16.32visik7no msn doesn't go on *
17:16.44Matjingahh something like miranda?
17:16.45jedirlold windows messenger used to support SIP
17:16.46HmmhesaysAnyone know why when you originate a call from the asterisk manager if you recieve a busy back, asterisk doesn't destroy the channel for about 30 seconds
17:17.35HmmhesaysI've been scratching my head on that all weekend
17:17.39Matjingactually if you pinted me on what to do on asterisk itself for the messenger *(any that supports sip) to work, I'll work on getting a client
17:18.04*** join/#asterisk Delvar (~irc@83.146.53.34)
17:19.07*** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34)
17:19.13*** part/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34)
17:19.24ManxPowerJuggie, Are you using iaxfriends/sipfriends from asterisk-addons with CVS-HEAD?
17:19.26*** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230)
17:19.40Matjingno
17:19.48AgiNamuHey, I was told that using an IAX type friend is horrible and doesn'
17:19.59AgiNamut work. and that instead, I should create a user and a peer entry
17:20.12AgiNamuanyone having any backing/disproving information?
17:21.03Matjinghow do you do the peer entry?
17:21.15AgiNamuhuh?
17:21.29AgiNamuthis is just for a normal extension
17:21.32AgiNamuincoming/outgoing calls
17:21.42AgiNamuI used to have it type=friend, one [entry] in iax.conf
17:21.52AgiNamuI was told not to use type=friend, since it's screwed up
17:22.18AgiNamuand instead do [entry]type=peer bla bla [entry]type=user bla bla
17:22.23AgiNamuwhich just seems like more work
17:22.38ManxPowertype=friend works just fine for PHONE entries.
17:22.48ManxPowerBut you should not use it for GATEWAY entries for many reasons
17:22.58NuggetManxPower is wise.  Listen to ManxPower.
17:23.22ManxPowerNugget, Thank you, Grasshopper.
17:24.08AgiNamuRevered sir, please tell me the reasons why not to use it for gateways.
17:24.43AgiNamuapart from that my gateways are differnet outgoing and incoming :)
17:27.11BeirdoAll hail ManxPower  :)
17:27.28Beirdoit's good to have knowlegable people around
17:27.42Beirdoand I'm sure I mis-spelled
17:28.55Matjing* We all bow...why not use it for gateways?
17:31.07*** join/#asterisk eivindtr (~Eivind@062016241059.customer.alfanett.no)
17:31.09*** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18)
17:31.12*** join/#asterisk wwalker (~wwalker@wwalker.sustaining.supporter.pdpc)
17:31.42ManxPowerAgiNamu, That is EXACTLY the reason.  Gateways almost always use different username/secret/host for incoming and outgoing.
17:32.36ctooleyIs there any way to control the number of concurrent calls that will happen due to call files being placed in the spool directory?
17:32.50ManxPowertype=friend requires EXACTLY the same stuff (username/secret/host/codec/context/etc) for incoming and outgoing.
17:33.14junky[work]ctooley: just mv ur limit in the spool dir ?
17:33.27ManxPowerctooley, Don't place so many files in the spool directory or see SetGroup/CheckGroup
17:35.08junky[work]~seen paradise
17:35.18jbotparadise <~paradise@n219079205023.netvigator.com> was last seen on IRC in channel #debian, 14d 59m 19s ago, saying: 'takatumi: in xchat'.
17:39.36ctooleySo there's a possibility that if 10,000 files are dropped and all 10,000 will happen concurrently?
17:40.16ManxPowerctooley, Only if you have 10,000 outgoing "lines"
17:40.26rvhii'd like to use realtime extension db, anyone knows when it will be in stable version?
17:40.30junky[work]ctooley: i never been able to with an so higher number.
17:40.53ManxPowerrvhi, NEVER!  NO NEW FEATURES WILL GO INTO 1.0.x STABLE!
17:41.00Hmmhesaysheh
17:41.16ctooleyManxPower, maybe he means, when will 1.1 be released?
17:41.22ManxPowerI need to set up a regex on channel stuff to send that.
17:41.35ManxPowerctooley, You are not on the mailing lists, are you?
17:41.40junky[work]realtime's unstable for now.
17:41.51dalaberaHi, everyone. I like to use the last stable version of *, but first want to check if an update has being applied to a specific file, How can I verify that??
17:41.57ctooleyManxPower, I'm not asking I clarifying his request
17:42.35ctooleyAnd yes, I am on the mailing lists, they go to gmail, get filtered, and I use the search functionality to parse out the things I can stand to read.
17:43.30*** join/#asterisk oej (~oej@apollo.webway.se)
17:44.23ManxPowerI'll refer you to the mailing list archives then, since the relase timeframe for 1.1/1.2 has been discussed.
17:44.24rvhiso realistically, when should i expect the realtime in stable branch?
17:44.27filehail oej
17:44.29*** join/#asterisk Los415 (~los415@ssf-office.corp.race.com)
17:44.48ManxPowerrvhi, 6 - 9 months
17:45.22ManxPowerBut oej might have a better idea when 1.2 (or 1.1, whatever it's called) will be released.
17:45.51jsolaresanyone know of other eagi examples other than the sphinx one?
17:46.08*** part/#asterisk codebreaker (~codebreak@flexserv.de)
17:46.12rvhiso i guess in the meantime, ast_data is my only choice?
17:47.21hardwirehttp://www.amazon.com/exec/obidos/tg/detail/-/B0007LQQUK/qid=1109612631/sr=8-1/ref=sr_8_xs_ap_i1_xgl23/103-0529937-5657415?v=glance&s=electronics&n=507846
17:47.26hardwireso cool
17:47.28hardwireVHFoIP :)
17:47.30hardwireand to think I was going to gut a snom just to make one
17:47.32hardwireout of a much more powerfull radio
17:47.36hardwirebut still
17:48.39jedirlwhat's different between EAGI and common AGI?
17:48.58jsolaresjedirl: you get sound with EAGI
17:49.02jsolaresand not sure what else
17:49.10jedirlsound?
17:49.16jedirlwhat do you mean with sound?
17:49.17jsolaresi need to make a recording application, but one that doesnt record silence
17:49.31jsolaresyou get the sound you'd hear on a file descriptor with EAGI
17:49.38jedirlwow
17:49.49jedirlthat can't be done with standard AGI?
17:49.53jsolaresnope
17:50.03jedirlyou can't use asterisk's recorder with standard AGI?
17:50.25jsolaresthat i can, but that doesnt suppres silence on the output, just stops recording when it reaches the silence
17:51.07junky[work]jsolares: why not using AGI RECORD with s option?
17:51.20jsolaressay what?
17:51.42ManxPowerjedirl, EAGI allows you to read/write audio on file descriptor 3.  AGI does not.  Record uses a different method to get audio and so does not require EAGI.
17:52.02jedirlManxPower: Ok
17:52.40junky[work]ive no idea what,s file descriptor 3.
17:52.58ManxPowerSome poor sod from Digium is ssh'd into my Asterisk server trying to understand my /etc/zaptel.conf.  LOL!.  /etc/zaptel.conf is VERY complex on my system.
17:54.43Hmmhesaysheh
17:54.53jsolaresjunky[work] : where's AGI RECORD? i'm either blind or stupid, probably a mix of both >_>;
17:55.04fileManxPower: awwww cute
17:55.14junky[work]~agi api
17:55.15jbothmm... agi api is at http://home.cogeco.ca/~camstuff/agi.html
17:55.21junky[work]take a look on page6
17:55.43*** join/#asterisk Gronker (~Gronker2@adsl-220-89-19.ags.bellsouth.net)
17:55.45junky[work]and you can add s=<seconds_blank> to that RECORD
17:55.56ManxPowerMy zaptel.conf: http://pastebin.ca/6610
17:56.02junky[work]that API should be re-done.
17:56.08ManxPowerI hate having to deal with legacy stuff
17:56.34Hmmhesaysheh, indeed
17:57.38fileblah
17:58.50jsolaresic
18:00.11*** join/#asterisk mrgoby (~mrgoby@exchange.synergybroadband.com)
18:03.04rvhiis it safe to do "extension reload" at any time?
18:03.23rvhiwhat is a call is being processed against an extension?
18:04.39ManxPowerI don't suppose anyone knows what bit rate Asterisk's Speex uses?
18:07.07*** part/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com)
18:07.29*** part/#asterisk wwalker (~wwalker@wwalker.sustaining.supporter.pdpc)
18:11.05*** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk)
18:17.05shmaltz~seen [Outcast]
18:17.06jbot[outcast] <~knoppix@h00045a737929.ne.client2.attbi.com> was last seen on IRC in channel #asterisk, 1d 13h 57m 20s ago, saying: '~xten'.
18:17.22*** join/#asterisk Inv_arp (junya@adsl-3-247-135.mia.bellsouth.net)
18:18.05shmaltzanybody here using billing with *?
18:18.12mrgobyindeed
18:18.28shmaltzmrgoby, what billing system are you using?
18:18.57mutilatorcustom made
18:18.59mutilatorO_o
18:19.02mrgobybeen using nufone's for a while, though have not tried the new one since they rebuilt it
18:19.24mrgobyit works well
18:19.44Hmmhesaysit's pretty easy to write a custom billing interface for *
18:19.50Hmmhesaysespecially post paid type
18:20.03mrgobyya, prepaid is a little trickier
18:20.05shmaltzmutilator, you using custom made billing?
18:20.15Hmmhesaysnot by much though
18:20.28mutilatoryeh
18:20.43shmaltz~nufone
18:20.44jbotsomebody said nufone was Visit http://www.nufone.net for an excellent, native IAX termination service.
18:21.03mutilatorit's not tricky or anything
18:21.04mrgoby~jbot
18:21.05jbotit has been said that jbot is the shipboard computer, but you may call me eddie if it helps you relax
18:21.06mutilatori just use the mysql cdr to record the basic info's
18:21.12mutilatorand then crunch some numbers
18:21.23shmaltzcheck this out:
18:21.24shmaltzhttp://www.nufone.net/
18:21.29*** join/#asterisk ast_freak (~yircme@hades-out.universalsystems.net)
18:21.38Hmmhesaysand a script in the language of your choice to wave a magic wand
18:21.59shmaltzHmmhesays, do you have anythihng you can share with me?
18:22.00mutilatoryeh
18:22.04*** join/#asterisk Darkar (~Alex@m174.net81-66-29.noos.fr)
18:22.22Hmmhesaysshmaltz are you looking to do a calling card type situation?
18:22.33*** join/#asterisk NirS_HOME (Nir@l192-117-110-178.cable.actcom.net.il)
18:22.34mutilatorbuilding the international rates db was a bit tedious tho :P
18:22.36NirS_HOMEhey all
18:22.40shmaltznope, just post billing
18:23.01Hmmhesaysmutilator: it's easy if you have the rate tables sent to you in a comma delimited format
18:23.13filekrammy boy!
18:23.40mutilatoryea, if only :P
18:23.46mutilatori got a printout fax
18:23.48*** join/#asterisk Meznev (~Elshar@ip205-68.oregonfast.net)
18:24.01Hmmhesayslol, OCR my friend OCR
18:24.02MeznevIs the Urgent Handler message a certain log level or something? :P
18:24.11MeznevI'm about to go digging through the code to turn it off its so annoying :)
18:24.36ManxPowerkram, Have you talked to Neil latley?
18:24.53Hmmhesaysshmaltz: i'm guessing you want a web interface to give you some type of organized bill per user
18:24.58ManxPowerMeznev, Yes.  Only for debug
18:25.12ManxPoweror starting asteirsk as "asterisk -c" which you should never do anyway.
18:25.29shmaltzHmmhesays, not realy, I just want to be able to mail bills to everybody at the end of the month
18:25.43ManxPowerbrb
18:25.44HmmhesaysI really wish the manager returned the uniqueID of the call it originated on success
18:25.48MeznevAaah, I see :)
18:25.53*** part/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
18:25.59Hmmhesaysinstead of just telling me that it successfully sent the command
18:28.24Inv_arpdoes * need reloading if AGI script changes?  does it parse on every use of it?
18:28.47junky[work]Inv_arp: no
18:28.48jedirlit doesn't parse it
18:28.53jedirlit executes it each time it's needed
18:29.21Inv_arpk thnx
18:30.34*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
18:30.49Hmmhesaysanyone know if that would be tough to implement?
18:30.50bjohnson_sheit, even voxilla replied to me
18:30.52*** join/#asterisk SIP_Help (TheJudge@196.46.64.209)
18:31.03SIP_HelpHi,
18:31.25SIP_HelpI am using a ondo sip server, is it possible to connect it to broadvoice.com services
18:31.47Hmmhesayshmmm...  this is #asterisk
18:32.16SIP_Helpsorry I mean a asterisk server
18:32.17bjohnson_SIP_Help: it is likely possible
18:32.17SIP_Helpsorry
18:32.29bjohnson_same with that
18:32.54SIP_Helpwho can explain how these server work ? or a url ?
18:33.47*** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com)
18:34.35*** join/#asterisk Syncros (~sysop@noc.routermonkey.net)
18:34.54*** join/#asterisk ACiDV (~joel@iteckGW.infoteck.qc.ca)
18:35.54ACiDVHi... I have 2 asterisk server, I have create 2 iax entry (type=friend) ... I can receive/send call on both box... but if I try to use trunking using trunk=yes, all call failed... any clue ? I remove trunk=yes and all work
18:36.47PinholeWe have several sip phones that connect to * at the office.  On one of them, when it connects through the pstn, I can hear them, but they cannot hear me.  if it connects to another of our sip phones, we can hear both directions.  How can I fix/diagnose the problem?
18:37.14bjohnson_SIP_Help: the * server?
18:37.16bjohnson_~docs
18:37.16jbotDocumentation 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
18:37.17*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net)
18:37.31bjohnson_ACiDV: lack of a timing source?
18:38.00bjohnson_Pinhole: is it travelling through NAT to connect to pstn?
18:38.06Pinholeyes
18:38.29Pinholewe do have nat=yes and qualify=yes
18:38.52Pinholeports 5060,5061, and 10000-20000 are all redirected to the sip phone.
18:39.15*** join/#asterisk spackle (~spackle@209.234.83.19)
18:39.40ACiDVbjohnson_ I have set my 1st TE405 port to by primary source...
18:39.48shmaltz~billing
18:40.04shmaltz~rating
18:40.22Pinholesip_phone -> NAT -> * -> sip_phone works, but sip_phone -> NAT -> * -> PSTN does not.
18:40.59NirS_HOMEhey all, here's a funky question
18:41.25*** join/#asterisk liquide (~havard@liquide.user)
18:41.39liquideis it possible to send and recieve sms messages with asterisk?
18:41.52NirS_HOMEhas anyone noticed differences in asterisk stability when compiling with various versions of GCC ?
18:42.50PinholeNirS_HOME, we use gcc 3.3.3 and think asterisk is not very stable.  Have not tried other versions of gcc.
18:44.03NirS_HOMEhow about 3.3.2?
18:44.37nestAri'm still using gcc 2.9.5
18:44.43nestArasterisk doesn't crash on me
18:44.44nestAr:)
18:45.18NirS_HOMEnestar, what distro are you using ?
18:45.18Pinholewe were restarting * once a week and now we restart every night at 3:30am.  Otherwise we get some funky stuff going on in some channels.
18:45.46nestArdebian
18:46.15NirS_HOMEahhh
18:46.20NirS_HOMEi hate debian
18:46.27nestAr:shrug:
18:47.04ManxPowerPinhole, What verison of Asterisk?
18:47.09ManxPowerPinhole, on what OS?
18:47.12Pinhole1.05
18:47.15*** join/#asterisk fishboy1669 (proxyuser@62.69.81.129)
18:47.27fishboy1669anyone here feeling in a helpfull mood?
18:47.37PinholeFC2
18:47.39fishboy1669i have a box with 2 x100p fxo cards
18:47.49fishboy1669using linux 2.6 kernal
18:47.52fishboy1669and udev
18:48.07spacklefishboy1669, ughhh udev.
18:48.19ManxPowerI've had very, very few issues with Asterisk stability.
18:48.19spacklefishboy, FC3?
18:48.24fishboy1669i have specified the irq in the bios for the pci slots but the f*******ng pc still put the cards on other irq's
18:48.38fishboy1669what is FC3?
18:48.47spackleFedora Core 3
18:48.54fishboy1669mandrake 10.1
18:49.02spackleOK.
18:49.16spackleDo you have ACPI enabled in the BIOS?
18:49.20fishboy1669its 6:50pm and i want to go home
18:49.21fishboy1669:(
18:49.21bjohnson_liquide: yes
18:49.37fishboy1669i dont know what is acpi?
18:49.39fishboy1669auto power?
18:49.55fishboy1669i think it is
18:50.04fishboy1669do u think turn that off will help
18:50.05fishboy1669?
18:50.37*** part/#asterisk human39 (~human39@chewie.fyi.net)
18:50.38spackleActually, I think setting up the UDEV rules would be more helpful, have you set them up?
18:50.42ManxPowerfishboy1669, ACPI is a way to get MANY MANY IRQs rather than the usual 15.
18:50.45fishboy1669yes
18:50.51*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
18:50.57ariel_Good afternoon all
18:50.57fishboy1669yes udev rules are set up
18:51.07fishboy1669the modprobes and the ztcfg work fine
18:51.07ManxPowerThe issue, of course, is that most BIOSs don't let you set the ACPI IRQs.  Change the slots the cards are in that try to share IRQs
18:51.10fishboy1669and asterisk boots
18:51.16spackleDoes anyone know if ACPI would override the bios settings for the PCI cards?
18:51.19angler_ariel_, hows it going
18:51.29ManxPowerspackle, in my experience, yes.
18:51.33*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
18:51.45fishboy1669but there are funny squeeks on zap calls which  i am gessing is due to irq confilicts
18:51.45spackleFishboy, how old is your computer, at least last 3 years?
18:51.47PinholeNirS_HOME:  in the makefile for *, I found this:   OPTIMIZE+=-O6
18:51.49fishboy1669yes
18:51.53fishboy1669brand new mb
18:51.55*** join/#asterisk Red_6 (~Alex@m174.net81-66-29.noos.fr)
18:51.55ariel_angler_, just fine how are you doing?
18:51.57Pinholeperhaps there is too much optimization
18:52.13angler_ariel_, pretty good, enjoy the show?
18:52.14ariel_angler_, I see you made it back from Miami
18:52.18liquidebjohnson_ how?
18:52.19*** join/#asterisk PMantis (~PrayingMa@66.251.89.34)
18:52.24angler_ariel_, sure did but wish i didn't leave
18:52.26NirS_HOMEhmmmmm
18:52.27NirS_HOMEcould be
18:52.37fishboy1669manx i only have 3 pci and i have 2 cards and putting a 3rd in tomorrow
18:52.48ManxPowerfishboy1669, It sucks to be you.
18:52.53*** join/#asterisk mbranca_home (~matteo@host-84-222-6-8.cust-adsl.tiscali.it)
18:53.07ManxPowerfishboy1669, turn off acpi in Linux.
18:53.17PMantisCan anyone tell me how astGUIClient and VICIDIAL are related?
18:53.18fishboy1669aha how do i do that manx?
18:53.27junky[work]wohoohooo, just got my gdb book!
18:53.37ManxPowerfishboy1669, Using clone cards is not the way to go.
18:53.40fishboy1669etc/somthing
18:53.40ariel_angler_, I am glad that I finally put a face to the person.
18:53.42ManxPowerfishboy1669, ask on #linux
18:53.49*** join/#asterisk Tough_Nuts (~Tough_Nut@204.110.228.254)
18:53.49fishboy1669it is not clone they are x100p
18:53.54ManxPowerI make sure I buy motherboards with lots of slots
18:54.03fishboy1669lol ok
18:54.07ManxPowerfishboy1669, They must be old cards then.
18:54.16ManxPowerSince Digium doesn't even sell the X100Ps anymore.
18:54.18fishboy1669ill blame my boss be he went home ages ago
18:54.32fishboy1669there still x100p so should work
18:54.52ManxPowerfishboy1669, as long as they are on their own IRQs.
18:54.53fishboy1669just stupid bios or linux wont let me assign irq propper
18:55.00PMantisManxPower, Lots of PCI slots? How many typical? 6?
18:55.04spackleFishboy1669, is the Motherboard a VIA motherboard?
18:55.05ManxPowerSince your mother board seems to suck there's not much you can do about it.
18:55.17ManxPowerPMantis, I don't know.  5 or so is what most of our systems have.
18:55.21ManxPowerIntel motherboards.
18:55.52ManxPowerThe real solution is of course, if you want more than 2 FXO ports get a TDM400P w/FXO modules.
18:55.52PMantisManxPower, OK,  thought you may be referring to a specialty board of some sort.
18:56.18fishboy1669sis motherboard i think
18:56.48spacklefishboy, have you had it working with a single card?
18:56.55ManxPowerPMantis, Our solution to "motherboard insists on sharing IRQs" is to make that machine a non-Asterisk machine and try a different motherboard.
18:57.08fishboy1669manx is it my motherboard tell me more about the acpi
18:57.09ManxPowerSometimes you just have to throw money at a problem until it goes away.
18:57.47ManxPowerfishboy1669, ACPI permits something like 255 IRQs rather than the normal 15 IRQs.  THAT IS ALL I KNOW ABOUT IT.
18:57.49spackleMaxpower, the money or the problem ;-)
18:57.54fishboy1669mmm but then if u pay for a tdm400p u go bancrupt
18:58.05fishboy1669ok i will read up cheers manx
18:58.09ManxPowerspackle, Hopefully the problem goes away before the money.
18:58.31PMantis...and you can EBay the leftovers. hah
18:58.32ManxPowera $10 million / year company cna afford to spend a little money to fix a problem.
18:58.44spacklefishboy, ACPI may be an option you can disable in BIOS.
18:59.34fishboy1669its not on i dont think
18:59.36DevilFishdoes anyone even have a bad idea about this?? I'll take anything at this point
18:59.39DevilFishhttp://lists.digium.com/pipermail/asterisk-users/2005-January/083456.html
19:00.24ManxPowerDevilFish, Upgrade to 1.0.5 or 1.0.6
19:00.37ManxPowerDevilFish, make sure you have recent Polycom Firmware
19:01.20spacklefishboy, what do you get what you cat /proc/interrupts?
19:01.24ManxPowerDevilFish, Maybe your SIP provider is hanging up the calls.
19:01.26DevilFishgot the latest PolyCom firmware
19:02.12fishboy1669spackle thats wehre i see there been irq share
19:02.41DevilFishwas there a fix from 1.05 to 1.06 addressing disconnections of some sort?
19:03.17spackleFishboy1669, I see, OK.
19:03.42jsolaresfishboy1669: what irq does it have?
19:04.05fishboy166918 / 19
19:04.12fishboy1669shared with sis adn eth0
19:04.30spackleall three are on the same IRQ?
19:04.40PMantisDoes 1.0.6 allow for "n" and "s" priorities in the dialplan?
19:04.59fishboy1669all 4
19:05.02fishboy16692 fxo
19:05.06jsolaresouch
19:05.33jsolaresmy quad fxo is on irq 209, acpi is gud
19:05.39DevilFishManxPower, do you if there we some sort of disconnection issues that were fixed from 1.0.5 to 1.0.6?
19:05.55spackleFishboy, have you disbled everything else that gets an interrupt - like USB, serial, parallel etc?
19:06.22BrianR___nufone.net's web interface is still busted...
19:06.25fishboy1669yes even 2nd hd lol
19:07.22BrianR___1.0.6 release is good news.
19:08.03spacklefishboy, the others may be right, it may be time to write off that motherboard.
19:08.08fishboy1669ok i give up
19:08.18fishboy1669cheers guys
19:08.49spacklefishboy, is it Intel or AMD?
19:08.49fishboy1669ill leave till tomorrow
19:08.49fishboy1669intel
19:08.49spackleSorry, G'night.
19:09.44loudDoes anyone have John's number (voipjet)
19:09.57DevilFishwhat is the easies way to upgrade asterisk from what I have now "Asterisk CVS-v1-0-01/17/05-11:18:25" to 1.0.6?
19:10.07Beirdoloud: gonna add it to a wardialer?
19:10.48loudNo, but i would like to do that w/ broadvoice :>
19:10.58Beirdohehe
19:11.05loudI have a question, and the domain has an invalid #
19:11.13Beirdowell, voipjet's domain info is shite
19:11.23loudRight.
19:11.25Juggieyay, after some screwing around i got today's cvs head running on a dev box.
19:11.29Beirdoand I have a good mind to contact the registrar about it
19:12.02Beirdothe postal code's not even valid for Ontario
19:12.08loudtrue, thats like a joke.
19:12.17Juggiebeirdo, which domain?
19:12.25Beirdovoipjet.com
19:12.48BeirdoA1A 1A1 if it exists is in Newfoundland
19:13.13BeirdoI think godaddy.com needs some bitchmail
19:15.11loudor .. John needs to put his shit together.
19:15.24Juggiebeirdo, it says "please use email"
19:15.27Juggieso clearly its fake
19:15.42Juggiedoesnt mean they arnt legit, they just didnt want to give out their physical contact info
19:15.45Beirdoyeah, that should not be allowed
19:16.03Beirdosorry, but that's not acceptable use of domain registrars
19:16.24Juggiewell, they should have the info in their and just hide it
19:16.28*** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl)
19:16.28Juggiesuch that you cant see it in the whois
19:16.30Beirdoand the site has no physical address or phone number listed anywhere
19:16.47Beirdoit just SCREAMS scam
19:16.55loudha
19:17.19Juggiethey offer you a test account for free
19:17.21Beirdoand on top of that, their iax server tends to get 2s lag spikes during the day
19:17.21Juggieso try it out
19:17.31BeirdoI have.
19:17.43JuggieBeirdo, no one is forcing you to go with them
19:17.49BeirdoFeb 28 14:16:24 NOTICE[1060]: chan_iax2.c:5673 socket_read: Peer 'voipjet' is now TOO LAGGED (2053 ms)!
19:17.50Juggiego somewhere else...
19:18.07BeirdoFeb 28 14:16:34 NOTICE[1060]: chan_iax2.c:5668 socket_read: Peer 'voipjet' is now REACHABLE!
19:18.13Beirdolike that :)
19:18.24Juggiethats nice, so they are just a shitty provider
19:18.25Juggieno big deal
19:18.27Beirdomeanwhile, my connection to FWD works great
19:18.28Juggiefind someone else.
19:18.45PMantisfwd is cool
19:18.57BeirdoI have found someone else, just waiting for the money to clear the account :)
19:19.20*** join/#asterisk Moc____ (~mochouina@64.235.210.66)
19:19.29PMantisBeirdo, offering 1.3c/min?
19:19.55BeirdoPMantis: close enough
19:20.08BeirdoI'm always willing to pay more for quality service
19:20.22Juggiethere are a ton of providers
19:20.41Beirdoyep
19:20.58Inv_arpi use voipjet in FL werks fine   64 bytes from 217.160.244.18: icmp_seq=3 ttl=48 time=47.9 ms
19:21.24*** join/#asterisk G0shen (~Goshen@70-57-80-147.slkc.qwest.net)
19:21.27BeirdoInv_arp: I'm pretty sure their connectivity has some issues
19:21.30Inv_arpBeirdo: your connection is fscked up
19:21.39Juggiei cant even ping them.
19:21.40Beirdono it isn't
19:21.51Inv_arpBeirdo: use the for outbound small  business ilibx  not one hiccup
19:21.59Inv_arperr ilbc
19:22.07Hmmhesaysanyone know if it would be tough to add the unique id of the call originated to the success message that the asterisk manager generated?
19:22.22BeirdoInv_arp: which means they have good connectivity to your provider
19:22.28HmmhesaysI'm not much of a C programmer
19:22.30Inv_arpbellsouth
19:22.45Juggie64 bytes from 217.160.244.18: icmp_seq=5 ttl=57 time=25.5 ms
19:22.53Beirdoas my provider uses alternet and many other backbones, I'm sure it's not on my end
19:22.57LuhiwuWhere can i find a list of IAX providers? i'm using voipjet now with bad quality, i've tried simpletelecom before with some problems
19:23.07Juggiewww.voip-info.org
19:23.13sudhir492Someone broke into my asterisk box
19:23.15sudhir492:-(
19:23.18Inv_arpLuhiwu: iax.cc teliax.com
19:23.31sudhir492Here is the log I see:
19:23.37LuhiwuInv_arp: thanks
19:23.43sudhir492reverse mapping checking getaddrinfo for 69-56-187-6.theplanet.com failed - POSS
19:23.43sudhir492IBLE BREAKIN ATTEMPT!
19:23.43sudhir492User root not allowed because not listed in AllowUsers
19:23.44sudhir492reverse mapping checking getaddrinfo for 69-56-187-6.theplanet.com failed - POSS
19:23.44sudhir492IBLE BREAKIN ATTEMPT!
19:23.44sudhir492User root not allowed because not listed in AllowUsers
19:23.46sudhir492reverse mapping checking getaddrinfo for 69-56-187-6.theplanet.com failed - POSS
19:23.49sudhir492IBLE BREAKIN ATTEMPT!
19:23.51sudhir492User root not allowed because not listed in AllowUsers
19:23.53sudhir492Illegal user test from 69.56.187.6
19:23.55sudhir492<PROTECTED>
19:24.02Beirdowelcome to the internet
19:24.03Sedorox~pastebun
19:24.06Sedorox~pastebin
19:24.07jbotwell, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
19:24.07Inv_arpheh
19:24.09Beirdohehe
19:24.15Beirdopastebun.  I like that
19:24.23sudhir492thanks for pastebin
19:24.26Sedoroxlol
19:24.49Hmmhesaysthat makes me hungry
19:24.55sudhir492looks like there is a vulnerability in ssh
19:25.11Hmmhesaysyeah especially if you don't use keys
19:25.14Hmmhesayslol
19:25.34sudhir492hmm. I did use a key
19:25.43Beirdothis is a fact of life
19:26.00Beirdoif you have an externally accessible ssh port, it will get constantly probed
19:26.14sudhir492Now what should I do about it?
19:26.16Inv_arpsudhir492: what version of openssh you run?
19:26.23sudhir492let me check
19:26.37Hmmhesaysnothing like a good probing to get the heart going
19:26.41spacklechange SSH to a different port if you use it externally.
19:26.45Inv_arpsudhir492: i just never run ssh on default port no probs
19:27.07sudhir492openssh-3.7.1p2-1
19:27.21loud3.7 is old
19:27.32Inv_arpheh i run 3.6
19:27.33Hmmhesaysuse rand(5000, 25000); that should get you a good port number
19:27.49sudhir492yes, I installed about 7 months ago, in July 2004
19:27.57Inv_arpsudhir492: what distro?
19:28.04Pinholeor, change the port every day just for fun.
19:28.05sudhir492redhat 9.0
19:28.07eKo1there's no point in using a different port because the port will still be there open for an attack.
19:28.13sudhir492then I upgraded ssh
19:28.19Inv_arpsudhir492:  yum install openssh
19:28.24BeirdoeKo1: quit ruining their fun
19:28.30loudredhat does not bring yum.
19:28.38loudyou have to manually add it.
19:28.50eKo1Why are people still running redhat 9?!
19:28.57Hmmhesayswhy are people running redhat
19:28.58Pinholeor, forward port 22 to 207.46.250.119
19:29.08sudhir492because it provided an extremely stable platform
19:29.13BeirdoeKo1: because you don't needlessly upgrade production machines
19:29.13*** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.res.rr.com)
19:29.15loudso is fc3.
19:29.18Inv_arpeKo1:  yea but im no provider  im just a lonely ip on the net ...   unless a certain heacker wants me
19:29.31Beirdowhich they do
19:29.35Inv_arpand my ip is dynamic  changes every couple of hours
19:29.39loudmass scanners exist.
19:29.39Pinholefc3 is problematic for people that don't have time to learn all those new permissions.
19:29.51spackleand udev.
19:29.53|Vulture|Anyone know if PRI providers allow DIDs from different area codes, for example you have a PRI in the 305 areacode and have DIDs in 212, 904 etc.
19:29.55Inv_arpim just worried bout worms  script kiddies  etc...
19:30.02sudhir492right. One of my machine shows on uptime: 15:28:24  up 259 days,  4:30,  1 user,  load average: 0.06, 0.08, 0.08
19:30.11loudi would be, if im on IRC :)
19:30.15|Vulture|Pinhole: you can turn that crap off on install
19:30.25eKo1Beirdo: What are you talking about, production systems are the ones that need the most upgrading.
19:30.27Hmmhesaysi like my debian machines
19:31.01Beirdoproduction machines get upgraded maybe once every 4 years.  I don't know what kind of production you are talking about
19:31.12Beirdoyou do security patches in the mean time
19:31.14eKo1Take * machines for example.
19:31.31Pinholemy msdos 3.2 box hasn't need upgraded for a long time.  It still collects dust just as well as it did 5 years ago.
19:31.31sudhir492what to do after the machine has been hacked? I cannot shut off the machine as it is being used for a client with about 1000 calls per day
19:31.50sudhir492Got to wait until the evening
19:31.59Beirdosudhir492: that didn't look like they hacked in
19:32.07Beirdoit looked like they probed you
19:32.15mrgobyOT: anyone used a good opensource CRM package geared towards ISPs?
19:32.19Beirdopastebin it, and we'll take a look if you wish
19:32.32sudhir492They have not yet hacked, but somehow they have been able to log in
19:32.35Inv_arpmrgoby:  sugarcrm.com   nice
19:32.37Beirdo[root@oban etc]# uptime
19:32.38Beirdo<PROTECTED>
19:32.43Beirdothat's RH7.3
19:32.48G0shensugarcrm = good :)
19:32.53Beirdoand will stay that way until I replace it
19:33.01Inv_arpmrgoby: not sure if geared for isp's what not
19:33.05Hmmhesaysoh yeah... i got you guys beat.... 34 days wahoo
19:33.06G0shennow we just need Asterisk integration with Sugarcrm
19:33.09Hmmhesayslol
19:33.19Inv_arpG0shen: oh yea u use?
19:33.21G0shenif you want a Company Centric open source CRM use XRMS, they have asterisk integration
19:33.32G0shenI use sugar yes to manage my leads
19:33.42G0shenSugar is contact centric
19:33.55Inv_arpG0shen: u pay yearly service our u host on your own?
19:33.58G0shenif I was only dealing with businesses I would use XRMS
19:34.26G0shenI host my own on my Mandrake/Asterisk/Webserver/CRM/Clinic/mailinglist box :)
19:34.37*** join/#asterisk IOscanner (~IOscanner@c-24-0-186-72.client.comcast.net)
19:34.58IOscannerAnyone know if there is a problem with cvs.digium.com?
19:35.01Beirdonow when I finally replace that machine, I haven't yet decided on distro
19:35.05sivanadoes Nufone have echo can, anyone know?
19:35.22IOscannerI can't seem to download the update.
19:35.39Inv_arpG0shen:  nice
19:36.00IOscannerI know Nufone takes your money and doesn't return it when they can't provide the service paid for.
19:36.55spacklesivana, I haven't had trouble with Nuphone echo, at least nothing that wasn't self-inflicted.
19:37.18sivanaspackle: I haven't either.  But a customer of mine reported echo on a LD call, that we routed through Nufone
19:37.30sivanawhich seems odd
19:37.40sivanaI'm just investigating reports of echo on Nufone routed calls
19:38.04*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
19:38.45neopherhello everyone, time for me to ask dumb linux questions again
19:39.01junky[work]neopher: why not go to #linux ?
19:39.07G0shenonly dumb question is one not asked
19:39.09*** join/#asterisk habakuk (~chatzilla@24-116-201-143.cpe.cableone.net)
19:39.12neopheron freenode?
19:39.15junky[work]ya
19:39.16*** part/#asterisk fishboy1669 (proxyuser@62.69.81.129)
19:39.33junky[work]or #debian if you're using debian.
19:39.39neopherwill do, no using fedora
19:39.59Zawwill i need a touch tone phone to test things with? i have a rotary phone i've been using and when i dial 1000 i get a fast busy. this is my first asterisk setup. i'm using a linksys pap2 device to connect the analog rotary phone to my asterisk pbx.
19:40.11neopherwant to pull the drive from one machin and put in another
19:40.22|Vulture|Anyone know if PRI providers allow DIDs from different area codes, for example you have a PRI in the 305 areacode and have DIDs in 212, 904 etc.
19:40.24eKo1neopher: go to #linux-help
19:40.38Inv_arpZaw: isnt linksys  locked to a vendor?
19:40.38MicH323MCI does
19:41.06neopherno, linksys has a router that is open to all venders too
19:41.11Inv_arpneopher: does the linux prob relate to *?
19:41.24ZawInv_arp: yes, if you buy them through places like staples. the firmware i'm using is from cisco and isn't vendor-locked
19:41.35neopherInv_arp, yes and no
19:41.36Inv_arpneopher: read on the wiki that they stopped production on those
19:42.22Inv_arpZaw: what does console on * say when u dial 1000?  pastebin.ca  if needed
19:43.21ariel_|Vulture|, Allegience used to do that before X/O took them over. I have not asked them if they still do.
19:44.21ZawInv_arp: it's not showing anything in the asterisk console, perhaps i need to increase the debugging level somehow?
19:44.29Zawwhen i dial 1000 that is
19:44.38habakukanyone know the web based asterisk config program on sourceforge?
19:45.07|Vulture|ariel_: oky thanx Ive been trying to get a quote from XO and they still haven't called me back
19:45.50Tough_Nutshabakuk, try AMP..
19:46.06*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-11-74.d4.club-internet.fr)
19:46.07ariel_|Vulture|, also nuVox was talking about doing that here as well.
19:46.28|Vulture|ariel_: we use nuvox right now for our T1 Ill ask them about it
19:46.53*** join/#asterisk minitareck (~minitarec@grosso-modo.org)
19:46.55minitareckhi
19:46.59Tough_Nutshabakuk, here is website... http://amp.coalescentsystems.ca/
19:47.08minitareckis there some asterisk developpers here ?
19:47.41Inv_arpminitareck: they stop by time to time
19:47.44neopherAMP rocks
19:47.55Tough_Nutsamp is nice...
19:48.35neopheri'm working on adding functions to it, kinda like an addon pack, call queue and chan_SCCP
19:49.22G0shenAMP is ok if you are running ASterisk on a dedicated machine
19:49.55neophertrue, if it is on a machine that uses regular web services, it's not a good idea
19:50.05minitarecki have to develop something which use an rtc but i can't find how with to say to the modem that he doesn't need to look for a carrier with hayes commands
19:50.08ariel_G0shen, asterisk should be ran as a dedicated machine.
19:50.09*** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au)
19:50.11Tough_Nutstrue.. but you can make it work anywhere.. with not much fuss..
19:50.15|Vulture|Tough_Nuts: that thing looks like a CPanel addon
19:50.34G0shenariel: I take my chances :)
19:51.04G0shenIf I ran all of my servies on decicated machines, I would have a basement full
19:51.14neopherCPanel, heh, i know those guys, they live like 10 min from me
19:51.19Tough_NutsToo bad flash_op_panel isnt web-configued...
19:52.36habakukTough_Nuts, thanks
19:52.40eKo1I see that the AMP call logs and graphs were ripped from the asterisk-stat package.
19:53.30|Vulture|eKo1: yea looks like they give credit though
19:53.47MicH323newbie question: I am trying the 8500 mailbox retrieve mail. Type the extension number then the passwd. It keeps telling me loking incorrect!
19:53.49Tough_Nutsyou think thats cute.. you should what the "* @ home" guys have ripped and done.. too bad it doesnt work as well as just amp..
19:56.46*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
19:57.08eKo1Thank god it's payday.
19:57.17Hmmhesaysheh
19:57.29Hmmhesayspeople pay you?
19:57.38Hmmhesaysjebus i'm getting stiffed
19:57.44eKo1Of course.
19:58.33*** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com)
19:59.12eKo1Anybody running a network with a wireless backbone?
19:59.54|Vulture|for residential yes...
20:00.35loudme too.
20:00.41eKo1What kind of antenae do you use?
20:00.48loudbreezecoms.
20:01.04sivanawoohoo
20:01.11sivana~seen sixTel
20:01.16jbotsixtel <sixtel@sixTel.iax.cc> was last seen on IRC in channel #asterisk, 53d 14h 42m 31s ago, saying: 'no such host, not in sip.conf right'.
20:02.01eKo1loud: Are you using 802.11b?
20:02.21loudYes
20:02.33loudWB-10 wireless bridge.
20:02.48*** join/#asterisk algorithmn (~na@ool-18bce89c.dyn.optonline.net)
20:02.54|Vulture|oh I just use 2 WRT54G routers with Sveasoft firmware as a bridge
20:02.54eKo1What's the farthest distance you've gone with that?
20:02.57*** join/#asterisk Mneumonic (Mnemonic@ool-18ba58b4.dyn.optonline.net)
20:03.06loudWith or without amplifiers ?
20:03.46sivanacanadian toll-free termination... 5c/min
20:03.49eKo1Without
20:03.51Mneumonicanyone know why when i use the directory and i find myself in there, and then hit 1 to connect it would say "There are no more compatible entries in the directory"
20:04.23loud15km w/o
20:04.45xeet2uhm
20:04.54Mneumonichmm... this is the error im getting in the cli
20:04.57MneumonicCan't find extension '100' in context 'default'.  Did you pass the wrong context to Directory?
20:05.06xeet2what kind of antenna did you say you're using loud?  directional right?
20:05.27eKo1loud: at what power?
20:05.44spackleand at what height.
20:06.01xeet2mneumonic:  what context is your extension 100 in?
20:06.11*** part/#asterisk minitareck (~minitarec@grosso-modo.org)
20:07.00Mneumoniccontext=myextensions
20:07.08Mneumonicahhh
20:07.10Mneumonici see the prob....
20:07.33Mneumonicexten => 0,1,Directory(default) has to be exten => 0,1,Directory(myextenstions) right?
20:07.34loudeKo1, with 100mw is more than enough, 24dbi dish.
20:07.35*** join/#asterisk yves_r (~choucrout@adsl-84-226-109-44.adslplus.ch)
20:07.40xeet2you can include the myextensions context in default, or use a different context
20:07.56xeet2loud:  directional, right?  los too
20:08.03loudyes. directional
20:08.19xeet215km seems a little far for 100mw
20:08.50loudvertical polarization from 6 to nine degrees
20:09.03*** join/#asterisk twod (~me@host116.lan.sequoianet.com)
20:09.07loudsounds ok now ?
20:09.12xeet2mmm
20:09.27yves_rHello everyone ... I'm using oh323 (asterisk stable), and incoming calls don't have any ringtone, everything else is working fine. Do you have any idea ? Not found any info on the web. Thanks.
20:09.45*** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34)
20:10.42eKo1loud: How's the signal? Is the signal good enough for voip?
20:11.03liquideis it possible to send and recieve sms messages with asterisk?
20:11.05eKo1Like, what's the packet loss like?
20:11.14loudits ok, i have 1.5 down half up, i can make two, three calls with no problems.
20:11.31loudand a /29 .. really good provider.
20:11.39xeet2loud:  you have some jitter buffering turned on, right?
20:12.08Luhiwuhey, i can get a /24 with 2mbps here :)
20:12.27loudsure you can, how much ?
20:12.45Luhiwuusd 800 per mbps :(
20:13.02xeet2uhm
20:13.03loudsee, i just pay 200.
20:13.06xeet2damn thats expensive
20:13.11Luhiwuwhere are you? i am in Argentina
20:13.22loudque boludo
20:13.56ariel_we have our selfs some sat users.
20:14.24loudmy 512 sat service goes around 550 dlls.
20:15.33loudariel_
20:15.41loudipsat, gilat, comstream ?
20:15.45loudcomtech
20:16.10eKo1Anybody doing +100 km P2P wireless?
20:17.01Himekosome polish guys set up a 61mile link
20:17.08LuhiwueKo1, i have never done so much distance, but there is a web site that allows you to calculate all the factors for a link that long
20:17.15xeet2eko1:  you'd need some skills to get that set up correctly and legally.  you would need an amp authorized for use with your equipment, and be able to aim the dish correctly
20:17.31louda big tall dish ..
20:17.36xeet2hehe, yeah
20:17.42xeet2might want to rent some tower space
20:18.04eKo1Yeah, I was thinking about an 8 ft. parabolic on a 120 ft. antanea
20:18.13xeet2at least in the us, you can't just use any amp with any equipment, has to be authorized by the fcc for use together
20:18.35Luhiwui think you'll have to get a taller antenna, the earth is not plain :)
20:18.44xeet2yeah, needs to be taller
20:18.50xeet2like I said, rent some tower space
20:19.00eKo1Well, the other end point will be on hill so...
20:19.19xeet2and remember, because you're in unlicensed space, your eirp can't be over 4 watts
20:19.33loud120ft .. and pray.
20:19.33eKo1I plan on using at most 1 watt.
20:19.48xeet2eko:  eirp != just what your transmit power is
20:19.59xeet2there's a calulation, takes into account a few things, I forget the formula
20:20.13LuhiwueKo1, look at this scripts to calculate the numbers: http://www.qsl.net/n9zia/wireless/page09.html
20:20.15Himekoefftive isotropic radiated power
20:20.16eKo1Well, I'm not a wireless telecom. engineer so...
20:20.21*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
20:20.26xeet2eko:  might want to hire one =)
20:20.37Himeko1 watt into a dish liek you are sugesting will probably be over
20:20.39eKo1Yeah.
20:20.55eKo1Himeko: I said at most.
20:21.05xeet2yeah, 1 watt on a dish will be over 4 w eirp definitely
20:21.36Himekoother solutions are to use a licenced band
20:21.45xeet2probably with more success too
20:22.01djinDid anyone here went through dCAP certification?
20:22.03xeet2you would be allowed to go far beyond that 4 watt limit, and it might be cheaper in the long run
20:22.04eKo1What I want to do is bypass the NSPs here and connected directly to the undersea fiber AP wirelessly.
20:22.24xeet2eko:  go look at licensed then, seriously
20:22.30PBXtechis there fax software that works with digium hardware?
20:22.39djinspandsp
20:22.50*** join/#asterisk sektor195 (~sektor@64.19.161.194)
20:23.04LuhiwueKo1: with 120ft tall tower, you have this:
20:23.05LuhiwuTransmitter Distance to the Radio Horizon : 24.94 kilometers  (15.50 miles)
20:23.21xeet2hehe
20:23.54eKo1Hmm...
20:24.26sektor195I have a question I am building an asterisk box with TE405P  card and a TDM40B card when you load the modules is there any particular order you have to load the modules in?
20:24.54sektor195also when you do a make install on libpri you should be able to modprobe it right?
20:24.56BrianR___You can get close to the part 15 limit with very dinky transmitter gear.. For example, a 60mw USB WiFi dongle mounted in a 24db gain pyramidal horn is close    to 4w eirp
20:25.12tzafrir_homesektor195: just load the modules with modprobe
20:25.15djinfirst modprobe zaptel
20:25.24djinthen order doesn't matter much.
20:25.31sektor195oh ok
20:25.36sektor195just wondering
20:25.42tzafrir_homedjin: modprobe will insmod zaptel. At least if you ran depmod -a
20:25.43sektor195because I am getting that 22 error
20:25.53BrianR___I use that dongle and horn from time to time for inet access in a pinch..
20:26.06BrianR___It's made out of cardboard / tinfoil, so it folds down flat.
20:26.20xeet2brianr: its great in the city for getting on one of the thousands of wide open ap's
20:26.33djinIs asterisk the proper channel for this wireless stuff?
20:27.20loudno, but there arent many engineers around, so ..
20:27.25Himeko#wireless
20:27.37djintzafrir nice to see you back.
20:27.51*** join/#asterisk heison (~heison@ns.somanetworks.com)
20:28.30djinloud, what type of engineers are you looking for?
20:28.44Nuggettrain engineers!  with little blue and white striped overalls.
20:28.55xeet2#train
20:28.57xeet2=P
20:28.59Nuggethee
20:29.14loudim not looking for one,
20:29.15*** join/#asterisk SIP_Help (TheJudge@196.46.67.89)
20:29.20loudwas speaking for eKo1
20:29.24SIP_Helplo all
20:29.34sektor195this is the error I'm getting when I modprobe my modules and during ztcfg
20:29.38loudi have like 40 eng here .. heh
20:29.42sektor195modprobe wct4xxp
20:29.42sektor195ZT_SPANCONFIG failed on span 1: Invalid argument (22)
20:29.42sektor195FATAL: Error running install command for wct4xxp
20:29.55SIP_Helpdoes any one know of a provider that will accept voice termination from a sip pabx
20:30.13xeet2sip_help:  uhm, plenty
20:30.28xeet2broadvoice, iconnecthere are two I use, and nufone via iax
20:30.36sivanaSIP_Help: yes
20:30.38ManxPowerI can't rememebr the last time I modprobed the drivers.  I just let the scripts installed by "make config" do it for me.
20:30.46SIP_Helpok so broadvoice do it
20:30.52sektor195and I have all the modules loaded
20:31.03SIP_HelpI dont know linux, thats the problem
20:31.07djinManxPower, same here.
20:31.17SIP_Helpso I will need to implement the asterisk for windows
20:31.27nestArugh
20:31.31ManxPowersektor195, The order the modules are loaded in are the order they should be in the config file.
20:31.40Juggiehow can i distinguish in the dialplan between a sip number that doesnt exist, vs one where the phone isnt registered, they both seem to return CHANUNAVAIL
20:31.50ManxPowerSIP_Help, Smarter people that you have tried to run Asterisk on Windows.  THEY HAVE ALL FAILED>
20:32.03djinsektor195, what does your /etc/zaptel.conf say?
20:32.07ManxPowerI think one is now in therapy, and another is in a psych ward somewhere
20:32.16SIP_Helplol
20:32.17liquidedoes it exist a webinterface wich can show you who called where and for how long for asterisk?
20:32.18sektor195stand by
20:32.22Juggiemanx, u have any idea?
20:32.24ManxPowerJuggie, you can't.
20:32.30SIP_Helpis the linux version have a web version ?
20:32.34sektor195the site to paste stuff is http://pastebin.ca right?
20:32.36Juggiegrr... that puts a needle in my haystack
20:32.36SIP_HelpJuggie you from SA ?
20:32.39djinsektor195, yes
20:32.49ManxPowerJuggie, not knowing anything about your config, no.
20:33.01djinSIP_Help, linux webversion?
20:33.11ManxPowerIf, within your dialplan, you know if the number is valid or not then you can handle it that way.
20:33.13xeet2sip_help:  really, learn linux a bit and use * on it, or get a softphone for windows and use that
20:33.23djinPerhaps change name to Linux_HELP,first?
20:33.28Juggiemanx, its not impossible it just means extra work.
20:33.45SIP_Helpdoes asterisk have a web interface ?
20:33.51ManxPowerJuggie, It's impossible for your to determine that info if ALL YOU HAVE is the response from the provider.
20:34.07xeet2sip_help:  it can, have to go grab what you want to use after you have asterisk installed
20:34.08Juggiemanx, the provider is me.
20:34.17SIP_Helpok
20:34.18SIP_Helpcool
20:34.28SIP_Helpwill ppl in here help me set it up ?
20:34.28Juggieasterisk shows a different error on the console, but the error code is the same.
20:34.31djinSIP_HELP, no. You might want to check razor.
20:34.53xeet2sip_help: people here may help you with a few * questions.  people here will not help you with linux questions
20:35.08SIP_Helpfair enough
20:35.26djinIt's a dedicated Asterisk on Debian config with menu's: http://www.xorcom.com/
20:35.30JuggieManx here's the problem, before i had a block of did's for this asterisk box, all the sip extensions were just names, and when ever you dialed 4 digits it just tried to dial it on the pri because we have alot of 4 digit internal numbers.
20:35.34xeet2sip_help: a few * questions does not include "how do I set up" or "how do I install", etc...
20:35.41Juggiebut now i have a block of 4 digit numbers which are sip phones.
20:35.50SIP_Help:)
20:35.59xeet2sip_help: take a look at voip-info.org, lots of helpful docs there
20:36.14SIP_Helpthanks
20:36.21Juggieso i thought i would just try to dial sip first, and if it failed go pri, this works unless the phone exists but is not registered
20:36.49Juggieit then creates a loop because once asterisk tries to call the unregistered number on the pri, it routes the call back into asterisk
20:36.51bjohnson_does voipjet do sip?
20:36.52Juggieand a loop ensues.
20:37.07bjohnson_hmm .. I guess not
20:37.26*** part/#asterisk didz_ (didz_@200.218.192.52)
20:37.45xeet2juggie: uhm, you should be able to avoid that in your dialplan
20:38.05Juggiexeet2, yeah i know, but then the dialplan has to be aware of the block of numbers which are internal to asterisk...
20:38.08bjohnson_I need a suggested pre-paid voip provider that does sip (already have iax.cc and livevoip.com)
20:38.20Juggiebut i guess i have to.
20:38.43sektor195djin you can find my config here http://pastebin.ca/6616
20:38.50xeet2juggie:  not necessarily
20:39.05heison~seen sivana
20:39.06jbotsivana is currently on #asterisk (18h 44m 36s).  Has said a total of 26 messages.  Is idling for 8m 31s
20:39.10SIP_Helpok, has anyone done it on windows yet ?
20:39.34xeet2juggie:  if a call is placed to a sip peer that is not registered, does it drop to the n+101 priority or just n+1?
20:39.52xeet2juggie: and which do you have configured?  or both?
20:40.03sektor195just out of curiosity on the TE405P card aren't the lights suppose to light up when you load the modules for it?
20:40.18Juggiexeet, i have both, and it drops it to +101
20:40.30bjohnson_shido6: you around?
20:40.34*** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net)
20:40.59Juggiei'll just do it another way...
20:41.06xeet2juggie:  ok, so for the n+101, don't have it dial out the pri, just send it to the users voicemail.  while n+1 will go out the pri
20:41.34xeet2dropping to n+1 would happen when the call fails, which would be the case if the destination sip peer does not exist
20:41.34Juggiexeet then it would try a call on the pri after a sip call terminated.
20:41.47sektor195because I know the lights flash in sequence when the module is not loaded yet.
20:41.57*** join/#asterisk zotz (~zotz@24.231.32.191)
20:41.58xeet2juggie: it shouldn't be doing that, have you tried?
20:42.20*** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net)
20:42.27xeet2juggie:  on a sip<>sip call, when a hangup is received it is a complete hangup, * doesn't go to the next priority in the list
20:42.38harryvvMichael Powell Is going to Resign from the FCC in march.
20:42.45xeet2yay
20:42.57sektor195djin: how does my config file look?
20:43.10Juggiexeet2, i need to handle the Extensions on the second asterisk box anyways, so i am doing it differentally not...
20:43.12Juggie*now.
20:43.15harryvvHe is also going to show up at VON next month
20:43.26Juggiehave to make the dialplan smarter thats all...
20:43.35xeet2juggie:  you can still do that, but ok
20:44.02*** join/#asterisk techie (gus@asterisk.horizonte.us)
20:44.05xeet2what is the general view towards him from voip providers?  I know on the clec side he's not well liked
20:44.43harryvvIm getting tired of the complaints from other people of "you sound like you are talking in the back of a room" to "I hear echo on your end" Must be a way to ballance out this x100p
20:45.22xeet2harryw: inline variable resistors
20:45.42harryvvxeet, is that what you are using?
20:45.43xeet2have you tested your line to see how unbalanced it is?
20:45.52harryvvno is there a way to test it?
20:46.18xeet2harryw:  have in the past, until the balance kept changing based on weather, then we got something that did real echo cancellation from multitech
20:46.51xeet2harryw: yeah there is testing equipment available, usually around 3-400 usd will get you a good tester
20:47.00harryvvmmm
20:47.50*** join/#asterisk snewpy (~markl@203-217-67-238.dyn.iinet.net.au)
20:48.44dalaberaHi, would anyone recommend a good router that behave very good with VOIP ?
20:48.51*** join/#asterisk Matt-E- (~Matt-E-@66-224-125-137.atgi.net)
20:49.02Juggiepbx.c:3033 ast_merge_contexts_and_delete: Requested contexts didn't get merged
20:49.05Juggiewhat are common causes for that
20:49.11Juggiei changed something and i cant find whats causing the error
20:50.27PBXtechis the rzfax app in the stable branch?
20:50.32ManxPowerJuggie, You really like pain?
20:50.33PBXtecherr rxfax
20:50.56ManxPowerJuggie, My GUESS would be duplicate context+exten+priority
20:51.02ManxPowerPBXtech, No.
20:51.09PBXtechk
20:51.11PBXtechthx
20:53.24sektor195djin?
20:53.41djinyes?
20:53.55sektor195did you check out my config?
20:54.12djinsorry, no. Was updating a * config.
20:54.17sektor195oh ok
20:54.33djinchecking now
20:54.39sektor195thank you
20:55.14djinwhat's you hardware again?
20:55.20sektor195djin: also do you know if on TE405P the lights are suppose to light up when you load the modules for it?
20:55.51sektor195TE405P and TDM04B
20:56.28djinkinda weird config.
20:57.10sektor195really?
20:57.36djinyou have t1 connections?
20:57.40sektor195yes
20:57.46PBXtechcvs not working?
20:58.04JuggieManxPower, manx, somehow, one of my editors be it PSPad, or pico mangeled the file into not having any CRLF's
20:58.16Juggieit was looking fine on PSPad, but pico showed it all as one line :P
20:58.22sektor195the pots card is for 4 backup pots connections
20:58.43eKo1Juggie: Probably PSPad...
20:59.22djinsektor195, let me check something
20:59.29sektor195ok
21:00.16djinwhy do you start bchannels @ 12?
21:00.33ManxPowerI don't suppose anyone knows of a consultant that has ACTUALLY set up Samba 3.x and OpenLdap?
21:00.37sektor195djin our T's are like this
21:01.19djin12 channels per T1?
21:01.28sektor195right
21:01.33sektor195the rest is data
21:01.34ManxPowermaybe he has data on channels 1 - 11
21:01.46djinok, thought so.
21:01.56sektor195they suggested we split our t's in case one t goes
21:01.58ManxPowerWe have T-1s with FXO channels, E&M channels, and data channels
21:02.28sektor195that's why B's start at 12
21:02.40sektor195the first 11 channels on each T are data
21:03.00sektor195and the last channel on each T is a D channel
21:03.40*** join/#asterisk NirS_HOME (Nir@l192-117-110-178.cable.actcom.net.il)
21:03.44djinI understand. Have only worked with full 30channels E1's so far :)
21:04.20sektor195do you use the TE405P card?
21:04.46djinyes and a TE410P
21:05.26sektor195do the lights on the card light up when you load the modules?, because I know the module isn't loaded the lights blink in sequence
21:06.28djinwow, have to thank about that one. I remember these is some 'knight rider'  action at first, but not sure if that was the TDM.
21:07.09djinand your modprobe zaptel went well?
21:07.27sektor195right the knight rider action is before the module is loaded
21:07.47sektor195well know it kicked back with that error I mentioned earlier
21:07.51sektor195I mean no
21:08.17sektor195but it sees all the channels
21:09.21PBXtechManxPower, i dont see app_rxfax in the cvs head either.. how do i get it?
21:09.38sektor195I get this error when I modprobe
21:09.42djinPBXtech, ftp://ftp.opencall.org/pub
21:09.52*** join/#asterisk clark_ (~bclark@64.171.107.5)
21:10.04PBXtechthat works on cvs stable as well right?
21:10.39djinsektor195, modprobe zaptel or modprobe wct4xxp
21:10.49djinPBXtech, it should
21:11.20PBXtechthx
21:11.39heison~sivana
21:11.40jbotsivana is, like, a putz
21:11.48heison~seen sivana
21:11.50jbotsivana is currently on #asterisk (19h 17m 20s).  Has said a total of 26 messages.  Is idling for 41m 15s
21:11.55sektor195hmmm.... I rmmoded them and mod probed them again and now I didn't get that error
21:12.02sektor195and my T card is lit
21:12.11djincool
21:12.25sektor195but its blinking red
21:12.32djinperhaps incorrect sequence?
21:12.39djinblinking red?
21:12.44sektor195yeah
21:12.47djinare the line connected?
21:12.52sektor195yeah
21:13.36sektor195I unplug them and plug them back in no change
21:14.10djindid you ztcfg -vv ?
21:14.23PBXtechhow do i know which version of TIFF I have?
21:14.30sektor195yeah
21:14.44djinand that loaded all lines properly?
21:15.00djinPBXtech what OS?
21:15.06PBXtechslackware 10
21:15.06*** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net)
21:15.07*** join/#asterisk drvoip (user@S01060050baab8e4b.cg.shawcable.net)
21:15.41PBXtechis there a command to find out?
21:15.43sektor195djin I am going to post the output of ztcfg -vvv so you can double check if that is ok.
21:16.03djinsektor195, ok
21:16.39tzangerwhat are you looking for?
21:17.18sektor195djin: http://pastebin.ca/6617
21:18.05djinlooks ok. Are you sure you plugged them in the right ports? You can also activate the other spans.
21:18.25sektor195it wouldn't have anything to do with the libpri module would it?
21:18.31sektor195yup I'm sure
21:18.31xeet2is there any way to get * to support t.38?
21:18.41tzangerxeet2: yes, contribute to the bounty
21:18.43sektor195they are plugged in the right ports
21:18.45*** join/#asterisk Katty (~angela@68.112.15.110)
21:18.49xeet2tzanger:  if only I could
21:18.51Kattyafternoon (=
21:18.54tzangerxeet2: then wait like the rest of us
21:18.58tzangerafternoon Katty
21:19.02Kattyhihi tzanger (=
21:19.18xeet2can t.38 between two devices that support it pass through * without any issues?
21:19.24xeet2via sip
21:19.27Kattyi'm going to download kernel-headers today :>
21:19.40Sedoroxhmmm
21:19.45Kattyhi Sedorox!
21:19.49Sedoroxhey
21:19.56Kattyhaving fun? (=
21:19.57Sedoroxhow be thee?
21:19.59Sedoroxof course
21:19.59Sedorox:-p
21:20.03Kattyexcellent
21:20.08Kattyi'm all teh bouncy
21:20.12Sedoroxlol
21:20.15tzangerxeet2: unknown
21:20.17Kattyone step closer today :>>>>
21:20.21tzangerKatty: haha
21:20.34tzangerKatty: wait until you get to a full kernel, you'll be ecstatic
21:20.43xeet2hehe
21:20.57Kattyor panicing
21:21.03tzangertaking all frickin day on this p4 2g
21:21.09Kattyeither way, i have you guys to pester
21:21.12*** join/#asterisk cjk (~cjk@80.92.75.91)
21:21.15Sedoroxlol
21:21.19tzangerKatty: oh.. yay.   :-p
21:21.20Kattyhi ariel!
21:21.24Kattytzanger: i know ;)
21:22.01sektor195djin: I think it may have to do with my cables are you using straight cat 5 cables or did you make T1 Xover cables?
21:22.15Kattytzanger: but maybe by the end of the week it will work and i won't have to pester anyone anymore :P
21:22.22tzangersektor195: first rule of t1 telephony
21:22.25tzangerplug in a loopback
21:22.31tzanger:-)
21:22.34djinit depends where they are connected to.
21:22.38Sedoroxa T1 crossover is just a rollover, right?
21:23.04ariel_hello Katty welcome back to our little home away from home.
21:23.17sektor195I think so I don't remember
21:23.24Sedoroxkk
21:23.25tzangersektor195: NO
21:23.30tzangerdamn why do people think that
21:23.30ariel_Sedorox, it depends on what you call a rollover.
21:23.34tzangerT1 uses pairs 1 and 2
21:23.37sektor195Right now I'm using straight cat 5
21:23.39tzangerpins 1,2 and 4,5
21:23.44tzangercrossover T1 is 1->4, 2->5
21:23.53tzangersektor195: no such thing as "straight" cat5
21:23.56tzangerhow are the pairs wired
21:24.10tzangereasy way to remember
21:24.21tzangerpair #1 is blue, and it is pins 4&5 (i.e. the dead center of the cable)
21:24.22Sedorox1 to 8... 8 to 1.... etc...
21:24.32sektor195wo or gw bw g brw br
21:24.37tzangerpair #2 is orange, and it is pins 1&2 (leftmost pair)
21:24.38sektor1951 to 8
21:24.49ariel_Sedorox, http://www.voip-info.org/tiki-index.php?page=crossover%20T1%20cable has some nice pictures.
21:24.53tzangerpair #3 is green and it is pins 3&6 (i.e. it "straddles" pair 1
21:24.55Sedoroxkk
21:25.01tzangerpair #4 is brown and it is the rightmost two
21:25.03tzangerALWAYS
21:25.06tzangerit NEVER changes
21:25.10tzangerethernet, T1, it's all the same
21:25.12tzangernow
21:25.18tzangerethernet uses pairs 2 and 3
21:25.25Sedorox'cept gig crossover :-p
21:25.34tzangerso to crossover ethernet, you cross the orange and green pairs
21:25.39tzangerbut T1 uses pairs 1 and 2
21:25.43tzangerso you cross the blue and orange pairs
21:25.52tzangerSedorox: gigE still uses pairs
21:25.53Sedoroxhmmm
21:25.58tzangerand IIRC they are the exact same order
21:26.01tzangerjust pairs of pairs
21:26.06Sedoroxyes
21:26.17Sedorox'cept the brown does change
21:26.23Sedoroxthats what I was referring to
21:26.24tzangerSedorox: like I said, pairs of pairs
21:26.28Sedoroxyea
21:26.40tzanger1->8 and 8->1 you have split pairs on pairs 1 and 3
21:26.42tzanger=badness
21:26.55Nuggettzanger is giving me flashbacks to when I used to do 25 pair type66 punch down wiring for a living.
21:27.04Sedoroxwell I just know... oh crap.. for like cisco console's and stuff its a rollover
21:27.05tzangerNugget: so you would hvae this memorized :-)
21:27.08Nuggetblue/orange/green/brown is forever etched in my head.
21:27.35*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
21:27.47SedoroxI need to get a good punch tool
21:27.53Kattyariel_: how're the kids? (=
21:28.23jsolaresnot me :)
21:28.24ariel_Katty, just fine. There with the wife at the mall...
21:28.28Sedoroxhmmm
21:28.39Kattyariel_: Oooo! the mall! girly shopping, eh?
21:28.50Kattyjsolares: k (=
21:29.11KattySedorox: busy?
21:29.15ariel_well since I am self employed she takes our young one to the center cort there. they have a playground with lots of neet stuff.
21:29.25ariel_And it's freeeeeeee.
21:29.36Kattyyay! free distractions for wee ones!
21:29.53tzangerNugget: what's the 5th pair colour, slate?
21:29.56Sedoroxnot really
21:30.04tzangeranyway back to meetings, yay
21:30.06KattySedorox: have enough patience to put up with me?
21:30.18Sedoroxdoes anyone? :-p J/K!!!!!!
21:30.19Sedoroxsure
21:30.21Sedorox:p
21:30.21Kattygosh
21:30.23jsolareshaha
21:30.30Sedorox:-p
21:30.30KattySedorox: probably not ;)
21:30.39Kattysomeday i'll know what i'm doing!
21:30.42Kattymaybe :>
21:31.03Sedoroxlol
21:32.16eKo1Man, I really don't like it when people work alone on projects and then expect you to magically figure out their shit.
21:33.06jsolaresso if i force you to work on my project for me, that'll be ok?
21:33.11jsolares:p
21:33.45eKo1The main problem is lack of documentation.
21:34.02eKo1Hence, the 'figure out their shit' part.
21:38.03harryvvWhat would be a reason that would case a delay from the time a number is entered in analog phone attached to a ata untill a number is initiated? Get like a 12 second delay.
21:38.30*** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com)
21:38.52ariel_harryvv, your ata is the one that is delaying for it's dial plan.
21:38.54Luhiwuharryvv: try adding a # at the end of the number, if it works faster, change the timeout
21:39.21ariel_I have gotten used to the # at the end. works fine.
21:39.23eKo1harryvv: You have a SetDigitTimeout(12) somewhere?
21:39.47harryvvno I dont.
21:39.54harryvvthis is for calling out
21:40.33bjohnson_finished the first draft of my article
21:40.37eKo1Check the ATA.
21:40.38harryvvso add # at the end of the ata dialplan you are saoing.
21:41.32ariel_harryvv, no when you dial the number end it with a #
21:41.38sektor195so I need to make a T1 crossover cable for a TE405P to work? it won't work over regular cat 5e
21:42.06bjohnson_tzanger: did you see this ?   www.voncanada.com
21:42.08harryvvarial, wife and future customers would be irritated by doing that.
21:42.10ariel_sektor195, no you need a crossover cable see the wiki it has the spec and a nice color drawing.
21:42.22ariel_harryvv, what ata is it?
21:42.23Sedoroxhttp://www.gcom.com/home/support/t1crossover.html
21:42.30harryvvarial spa 1000
21:43.01ariel_if you go into the web you can change it. But the problem is that if you have slow people dialing it can be a problem.
21:43.10bjohnson_I'd like to play with my sipura dial settings but haven't yet
21:43.18harryvvhow slow
21:43.27harryvvOhh wait yea let me see how slow :)
21:43.50bjohnson_probably a 2 second delay would be ok
21:44.46*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
21:47.14harryvvadding # to the outbound call still takes 10 seconds.
21:47.35Sedoroxanyone use avaya phones with *?
21:47.54PMantisAny have suggestions for hot-failover * boxes/
21:48.12jsolaresSedorox, only the 4602
21:48.14jsolaresin sip mode
21:48.21Sedoroxhow is it?
21:48.28jsolaresgood
21:48.32Sedoroxhmmm
21:48.32eKo1PMantis: heartbeat + mon?
21:48.33*** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com)
21:48.52jsolaresit kicks the budge tone 100's for audio quality
21:49.01Sedoroxhehe
21:49.04jsolaresbut it only supports g729 and g711u (as far as i can tell)
21:49.10Sedoroxthats not too bad tho
21:49.18*** join/#asterisk RoyKa (~roy@83.80-203-29.nextgentel.com)
21:49.22jsolaresyeah, it does depend i guess
21:49.30Sedoroxyea
21:49.32*** part/#asterisk twod (~me@host116.lan.sequoianet.com)
21:49.58jsolaresi thought at first that i needed an tftp for it, but nooo, the phone wanted to get it files off http
21:50.05Sedoroxjust looking around on ebay.. saw one.. I like their desgin
21:50.11Sedoroxlol
21:50.23jsolaresthe 46xx are sip/h323 as far as i can tell
21:50.27PMantiseKo1, Dunno. I'm Googling for an answer, checking wiki... Need one machine to (probably) assume the initernal IP of the failed server, and get the E1 routed to the good server as well.
21:50.28*** join/#asterisk dahunter (~joe@lsanca1-ar8-4-60-068-194.lsanca1.dsl-verizon.net)
21:50.43*** join/#asterisk stonefly (~stonefly@toby.stoneflytech.com)
21:50.44SedoroxI see a 4620 on ebay...
21:50.46Sedoroxaround $100
21:50.58jsolaresthat's a good price
21:51.10Sedoroxhttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=58344&item=5753937846&rd=1
21:51.47jsolaresthat one can do sip
21:51.52*** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com)
21:51.53Sedoroxoh wow.. ldap.. neat
21:52.09eKo1PMantis: You can do it. You just need to think about the setup a little bit.
21:52.19Sedoroxif I only had the money :/
21:52.29PMantiseKo1, I need more experience. :)
21:53.17jsolaresthe keys have great feel
21:53.35*** join/#asterisk rpoppi77 (~rpoppi77@201.24.15.125)
21:53.54jsolaresthe overall weight is good, doesnt seem flimsy, it seems like it can take a beating
21:54.50Sedoroxthats good
21:55.00Sedoroxrecommend them over cisco?
21:55.47jsolaresno experience with cisco's
21:55.55Sedoroxah
21:55.56jsolaressend me one, and i'll let you know :p
21:55.59Sedoroxlol :-p
21:56.01SedoroxI wish
21:56.03harryvvfixed the dial plan on the ata. Now the wife wont gripe :)
21:56.03*** join/#asterisk marc324 (~marc32344@69-28-224-214.dsl.teksavvy.com)
21:58.21liquidewhy does sip phones have to be so expensive?:P
21:58.50harryvvbecause the customer can afford it? Especially since thay are going to save on long distance.
21:59.21nestArthey aren't really that expensive
21:59.24KattyHmmhesays: let me know when you wake up (=
21:59.27nestArin comparision to other office phones.
21:59.29liquide100$+
21:59.42spackleThere will probably be a sip phone for $20 within 5 years.  And they really aren't more than many proprietary sets.
21:59.50nestAri paid more than $100 for my last cordless phone..
21:59.52harryvvnestar, I have not priced out digital ones. What is the price difference typically?
21:59.55nestArjust the way it goes..
21:59.58stoneflyI've searched the wiki and list, but I can't find anything usefull, except a bug cvs in november, about this error "chan_iax2.c:5769 socket_read: Received trunked frame before first full voice frame" I'm running CVS-HEAD-02/24/05
22:00.21nestArharryvv: it's been a long time but we used to pay ~$300 per set for our existing Mitel PBX
22:00.43liquidebecause i'm stuck, i wanna use a cordless phone, but the sip phones are expensive, i could get a adapter from my voip provider, or i could get one of the digium cards.. not quite sure
22:00.44nestArliquide: there's always a ghettone.. err budgetone!
22:00.45nestAr$60
22:00.47Sedoroxanyone knoe if the 3com ip phones work with *.. or are they 3com proprietary
22:00.58liquidenestAr where=?
22:01.07Sedoroxbt100's are nice tho...
22:01.09nestArliquide: get a Sipura ATA.. they're ~60-70
22:01.09Sedoroxbasic set...
22:01.16nestAradd a cordless phone
22:01.17nestArwin!
22:01.30liquidethats a adapter
22:01.32harryvvI have the same here works well once you get it configured right.
22:01.33stoneflyPolycom's aren't bad either...
22:01.37djindid anyone try to install v1.0.6 with spandsp (pre10)?
22:01.46nestAri have polycom 300's and 500's here
22:01.52nestArthe 300 is a solid phone..
22:01.59nestAri, of course, prefer the 500
22:02.07nestArbut cost rules my deployment
22:02.16liquidethe adapter will cost me near 100$ because of postage :p
22:02.37nestArthese are the bends
22:02.37harryvvI paid more up here because of markup but did not want to wait :)
22:02.52djin[app_rxfax.so]Feb 28 23:05:00 WARNING[5019]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler
22:02.52djinFeb 28 23:05:00 WARNING[5019]: loader.c:440 load_modules: Loading module app_rxfax.so failed!
22:02.56liquidei need to get it shipped to norway
22:03.25liquidewe dont have that nice stuff here.. well, my voipprovider does..for a bit over 100$ but they also offer ip phones for 60-70$
22:03.31liquide*160-170
22:03.38ManxPower~docs
22:03.40jbotDocumentation 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
22:04.02djinManxPower, is that for me?
22:04.14dahunterVerizon has this feature that allows you to forward calls by hitting 'flash' and then doing some Magic.  Is there some way to pass that flash signal to the external lines instead of asterisk capturing it?
22:04.44eKo1dahunter: What external lines?
22:05.03dahunterek: The phone lines, FXO's not the FXS'
22:05.12bjohnson_harryvv: what did you set the timer values to?
22:05.31eKo1So you want to press flash and get access to the FXO?
22:05.40marc324ne1 knows which motherboard has 3.3V PCI slots?
22:05.56harryvvI did not. I modified the ata dial plan and now the delay is down to 4 seconds before i hear the end ring. It was like 10-12 seconds.
22:06.00zigmandjin install spandsp
22:06.12bjohnson_dahunter: someone was trying to use line based call waiting through an * server hee and found the answer on the wiki .. sounds like what you want
22:06.25djinzigman, I'm sure I did, but lemme check
22:06.41dahunterek01: Yeah, a customer may call in on one of the Zap channels, and then if we press flash and some numbers without asterisk intercepting it (the outside phone lines) it will detach and be forwaded all inside the verizon system leaving our phone lines free.
22:06.46bjohnson_harryvv: oh.  I was looking at Interdigit_Long_Timer and Interdigit_Short_Timer
22:06.56dahunterbjohn: Do you have a link to that?
22:07.06liquidewhat do you guys think of this http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61841&item=5755106834&rd=1 ?
22:07.11bjohnson_no
22:07.13harryvvbj, what is that for ?
22:07.25bjohnson_Interdigit_Long_Timer is used after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed.
22:07.33bjohnson_The Interdigit_Short_Timer is used after any one digit, if at least one matching sequence is complete as dialed, but more dialed digits would match other as yet incomplete sequences.
22:07.51harryvvinteresting
22:07.58eKo1dahunter: I don't think you can do that.
22:08.45ManxPowerdjin, That is for all newbies
22:08.51*** join/#asterisk _phate_ (~phate@phate-0002.user)
22:08.51*** join/#asterisk buddah (~hnic@208.179.86.5)
22:09.22djinzigman, I make installed spandsp again, but no luck so far. Will try something else.
22:09.45djinManxPower, well. define newbie ;)
22:09.56dahunterek: Drat, I was hoping you wouldn't say that.
22:10.56dahunterI recently purchased a Sipura 841.  I can make outgoing calls fine, but if I tried to dial that extention it immediately comes back as busy.  Has anyone ever had that problem or correctly configured one?
22:11.18filedid you configure the extension correct? what does it say on the CLI?
22:11.45buddahneed help with an extensions pattern. trying to prioritize _0118802. over _011. and having trouble getting it to sort correctly, so that when a _011 is put through it matches if 8802 is there to _0118802 first, if not it goes to _011
22:11.53shido6user and a peer or friend, dahunter ( check your private messages)
22:12.00buddahbut it keeps putting _011. first, i'm assuming because its shorter perhaps
22:12.17dahunterfile: Got SIP response 486 "Busy Here" back from 192.168.1.102
22:12.18shido6buddah no
22:12.30buddahnot because its shorter?
22:12.31*** join/#asterisk rpoppi77 (~rpoppi77@200-140-015-149.bsace7025.dsl.brasiltelecom.net.br)
22:12.35shido60118802 could be first but then use 011XXXX for a later exten line
22:12.40shido6try that
22:12.49shido6err
22:12.52shido6011XXXX.,
22:12.55shido6rather
22:13.00buddahokm
22:13.05shido6so u have one 0118802.,
22:13.09buddahthat should work
22:13.10shido6and u have another using the XXX
22:13.13shido6and reload
22:13.15shido6and make a call
22:13.23bjohnson_shido6: what's the timing on the SW Ontario DIDs?
22:13.24buddahthe xxxx will be part of the phone number though, so thats ok since none will start out as 8802 right?
22:16.47Sedoroxbbl
22:17.05*** join/#asterisk paulc (paulc@S010600062586a0b4.vc.shawcable.net)
22:18.10*** join/#asterisk GDRA (~1054@209.51.68.120)
22:19.44*** join/#asterisk Alric (~nbowyer@ds1a-ai.1stel.com)
22:19.57*** join/#asterisk Matt-E- (~Matt-E-@66-224-125-137.atgi.net)
22:20.03*** join/#asterisk MatsK (~NNSCRIPT@8.80-202-60.nextgentel.com)
22:20.48*** join/#asterisk dadlan{beginner} (~me@user88.bwa.etnet.fo)
22:21.24*** join/#asterisk MatsK (~NNSCRIPT@8.80-202-60.nextgentel.com)
22:22.56*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
22:23.20Zawok. i have my asterisk server up and running, and tested it connecting via iax with the demo to digium. now i need a provider to use for dialing out/in to the PTSN. i have no local zaptel card. are there any preferred vendors to use?
22:23.39*** join/#asterisk MatsK (~NNSCRIPT@8.80-202-60.nextgentel.com)
22:23.59bjohnson_what country?
22:24.11*** part/#asterisk Poincare (~jefffnode@dD5779B07.access.telenet.be)
22:24.13ZawUS
22:24.17Zawpittsburgh, PA
22:24.21bjohnson_I use voipjet and livevoip
22:24.22*** join/#asterisk Poincare (~jefffnode@dD5779B07.access.telenet.be)
22:24.31Zawthanks
22:26.23buddahhow can i setup the dialplan to strip the 011 off an international call, so it just sends 88028921468, rather than 01188028921468 to the gateway?
22:26.42djin${EXTEN:3}
22:26.43file${EXTEN:3}
22:27.09djindamn you type slow ;)
22:27.15Beirdowow, in stereo :)
22:27.23buddahexten => _0118802.,1,Dial(SIP/${EXTEN:3}@202.52.195.217|50|t)
22:27.24buddahlike that?
22:27.30*** part/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net)
22:27.43djinexten => _0118802X.,1,Dial(SIP/${EXTEN:3}@202.52.195.217|50|t)
22:27.54shido6buddahok
22:27.55shido6im back
22:28.01buddahlol
22:28.11buddahgotta have that X in there?
22:28.30djinif you don't want to limit it to 0118802
22:28.40buddahi do
22:28.48buddahjust the 8802 ones need to be stripped of the 011
22:28.56buddahall other int'l calls are fine
22:29.18djinnow only calls to 0118802 get routed to 8802
22:29.23*** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de)
22:29.31buddahok, good, thanks
22:30.21djinwith the X you say that all calls starting with 0118802 get handled accordingly.
22:31.11|Vulture|how is livevoip for network size etc.?
22:32.08Juggieis anyone working on correcting realtime behavior with 'show voicemail users' or 'sip show peers' such that they show the information from realtime?
22:33.12*** join/#asterisk The_Duke (~the_duke@ppp-124-33.adsl.restena.lu)
22:36.53*** join/#asterisk Grooby (~Grooby@66.160.105.186)
22:38.49Groobyif I cvs checkout the v1-0_stable
22:38.55Groobyi should have 1.0.6 right?
22:39.25*** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net)
22:40.28Nuggetthere is no v1-0_stable.  it's just v1-0
22:41.34cjkanyone here who know an iax termination provider in europe?
22:42.04*** join/#asterisk Rick_Hunter (~rhunter@03-024.008.popsite.net)
22:42.25*** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net)
22:42.33Groobyokie..if I get v1-0
22:42.38Groobythat's 1.0.6 right?
22:43.09djinGrooby, yes. v1.0.6 + updates
22:43.14Groobyokie
22:44.12Mikeguys using allow=g726-16 doenst work?
22:44.22*** join/#asterisk amir (~amir@shield.guindehi.ch)
22:44.27Nuggetdoes it work if gals use it?
22:44.43*** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net)
22:44.46ManxPowerUm, Asterisk only supports G726-32
22:45.00ManxPower(called "G726" in the allow= line)
22:45.05|Vulture|I bet the wiki would say...
22:45.08|Vulture|:P
22:45.29MikeManxPower, cvs doesnt support the rest?
22:45.37ManxPowerMike, Who knows.
22:46.00ManxPower"show codecs" will tell you what are the valid codec names
22:46.42Mikei c
22:47.01wazquishm, i'm trying to get a wakeup call running..the AGI applications i found doesn't seem to work. asterisk executes the scripts, but nothing happens..and it returns 0, anyone?
22:47.45harryvvwakeup call? what you trying to do make a setup reminder to call your phone?
22:47.59harryvvThat would be good for a hotel.
22:48.00wazquisharryvv, nah, line those on hotels...
22:48.08wazquisahhr...yes
22:48.09wazquis:)
22:48.19|Vulture|wazquis: that script is wrong, check the comments I think I fixed it in one of the comments
22:48.29harryvvmake a web interface for a hotel user and you just sold a system to them :)
22:48.37harryvvhotel managment I mean :)
22:48.45wazquis|Vulture|, i tried both a php and a perl script...
22:48.59|Vulture|oh I thought you were doing AGI
22:49.00wazquisharryvv, haha ;)
22:49.13wazquis|Vulture|, i am..?
22:49.28harryvvBTW, what do hotels use still the same thay manually call the hotel guest?
22:49.55wazquisharryvv, it works by placing calls manually to be executed at a specific time... would be very easy to write a webinterface to do that
22:50.03|Vulture|just have it run in cron every min, then check a mysql DB, if it matches the time, then put it into the outgoing call spool
22:50.29wazquis|Vulture|, that part is running...i found a AGI script for setting up the call
22:50.47wazquisbut that one doesn't work
22:50.47|Vulture|oh, your trying to get a web interface with it now?
22:50.50wazquisthose doesn't work
22:51.01wazquisno no
22:51.10liquidei cant get the webinterface for voicemail to work, when i login i get Bleh, no /etc/asterisk/voicemail.conf at /usr/lib/cgi-bin/asterisk/vmail.cgi line 96.   whats wrong?
22:51.11wazquisi wanna be able to call "7000" and press the time i wish to recieve a call...
22:51.21|Vulture|oh oky
22:51.33*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-11-74.d4.club-internet.fr)
22:51.49|Vulture|shouldn't be that hard, set a time value and callback # variables, then add it to the dp
22:51.50|Vulture|db
22:52.35tzafrir_home|Vulture|: why not use a sudo command? write a small script that drops whatever needs dropping
22:52.43wazquis|Vulture|, http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20wake-up <- i tried using that...and as far as i can see i've done everything for it to work....
22:53.48tzafrir_homeliquide: what is the exact error you get there?
22:54.06tzafrir_homeliquide: can your apache read your asterisk configuration?
22:54.19liquidegood question
22:54.25tzafrir_home<PROTECTED>
22:54.31liquideit is
22:54.32wazquis|Vulture|, http://lnxbx.dk/~akv/agi.txt <- the output from asterisk console
22:55.04|Vulture|wazquis: yea Id have to look through the code.. it seems involved
22:55.10liquidetzafrir so i would need to play with permissions?
22:55.23tzafrir_homeThe error there is a bit misleading. The line in the code is:
22:55.49tzafrir_home(open ...) || dir ("Bleh, no $filename")
22:55.54*** join/#asterisk sektor195 (~sektor@64.19.161.194)
22:56.00*** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net)
22:56.20wazquis|Vulture|, okay...i would think it works...when it's posted there and so on...
22:56.21tzafrir_homeapache runs as www-data . Asterisk's configuration is by default world non-readable
22:56.36wazquis|Vulture|, must be my config that fails...but i can't seem to figure out where...
22:56.46tzafrir_homeThat's probably because of the passwords spread around there (including in voicemail.conf)
22:57.20tzafrir_homeeither allow "others" to view /etc/asterisk , /etc/asterisk/voicemail.conf or add www-data to the group asterisk
22:57.56tzafrir_homeThe first option is probably better. The minimal required is:
22:58.14sektor195in zapata.conf when setting up a pri card are you suppose to use signalling pri_cpe?
22:58.19tzafrir_homechmod o+x /etc/asterisk; chmod o+r /etc/asterisk/voicemail.conf
22:58.52djinsektor195, yes
22:59.08sektor195funny I get unknown signal
22:59.33liquidetzafrir thanks :) at least it givew me another erro now..login incorrect:P
23:00.32sektor195djin: I get unknown signal type
23:00.48sektor195I did compile libpri
23:01.41tzafrir_home<PROTECTED>
23:01.43sektor195any suggestions?
23:01.56liquidetzafrir rapid system?
23:02.29tzafrir_homeXorcom Rapid. I figure it's not, then
23:02.46tzafrir_homelook at the apache's error log
23:02.57buddahdjin: ${EXTEN:3} gets rid of 011, how can i shed 0118802?
23:02.58tzafrir_home<PROTECTED>
23:03.50liquidenothing relevant
23:03.58djin${EXTEN:7}
23:04.03buddahthx
23:04.45djincount the numbers, dude ;)
23:04.51sektor195?
23:05.11*** part/#asterisk Grooby (~Grooby@66.160.105.186)
23:05.44djinsektor195, parse the error.
23:05.49buddahyeah
23:05.51buddahi just saw that
23:06.18buddahi didnt realize it until i saw 2 of them that were like that, thought maybe it was a code or somethin
23:06.34*** join/#asterisk myloforreal (~mylo@178-151-222-203.rev.techex.net.au)
23:06.54Zawfor the record, loading kernel modules (specifically ztdummy.ko) in the freebsd port of asterisk is evil.
23:06.56tzafrir_homeliquide: your voicemail sits in /etc/asetrisk/voicemail.conf, right?
23:07.07liquide*asterisk yeah
23:07.25sektor195have fun ice skating on the road zaw
23:08.20tzafrir_homeyou could try to put some prints inside the function "check_login". It seems to "fail"
23:08.56marc324whats the advantage of te410p over te405p?
23:09.34*** join/#asterisk myloforreal (~mylo@178-151-222-203.rev.techex.net.au)
23:09.34djinit works on different PCI-busses
23:09.52marc324why chose te410p over te405p?
23:09.55djinTE405P -> 5volt, TE410P -> 3.3volt.
23:11.00djin3.3volt is supported by more recent boards.
23:11.38myloforrealHello all, in a word - (yes or no) - i've played around with * and can make a software SIPphone call another software SIPphone.  I'm about to try and get an AVM C2 card to work.  Is configuring ISDN hardware tricky?
23:12.25Hmmhesayswhat if I want to consume more power?'
23:14.20terrapenugh....the flu sucks
23:14.46|Vulture|terrapen: yea i just got over it... 1 week of that crap
23:15.07Juggiehmm, after a few hours of playing with realtime, its cool, but needs work.
23:15.15Hmmhesaysanyone else using the manager api to originate calls?
23:15.22JuggieHmmhesays, yes, why?
23:15.45terrapeni've never had a bad sore throat from the flu before
23:15.46shmaltzFEDORE FUCKING GARBAGE CORE
23:15.51modulus_yeah
23:15.53terrapenbut my doc said it was not strep throat
23:15.56HmmhesaysI'm curious how hard it would be for originate to return the uniqueid of the call it originated
23:15.57modulus_welcome to redhat
23:16.05Hmmhesaysmy C skillz are dull as a butter knife
23:16.06modulus_RED FUCKING GARBAGE HAT
23:16.12loudhaha
23:16.12stoneflyshmaltz, whats wrong w/  fc?
23:16.22shmaltzfucking stupid OS, wors than Windoz
23:16.24terrapenDebian is about the only linux i can marginally stand
23:16.32|Vulture|shmaltz: hahaha
23:16.39|Vulture|I use FC3 works great for me
23:16.43Juggiethis isnt the bitch about linux channel, please drop the language.
23:16.53shmaltzstonefly, it just jumped on me thinking Im female and wanted to fuck me
23:16.57wazquis|Vulture|, any luck?
23:17.01shmaltzJuggie, you are right
23:17.03shmaltzsorry
23:17.04JuggieFC3 also works fine for me.
23:17.06terrapenjuggie, he just learned a new word
23:17.10tzafrir_homeNot to mention that some content would make your claims more reliable
23:17.21|Vulture|wazquis: nah I didn't install it I will tommorow on my dev box though
23:17.31Juggiei run two asterisk boxes, one core 2 and one core 3
23:17.33shmaltzOk, here is the story of one whole day wasted on FC3
23:17.37stoneflyGentoo nice...
23:17.41Juggiecore 3 was a bit tricker, but only took a few extra steps.
23:18.00shmaltzmod_perl doesn't work, had to remove the install that came with FC3
23:18.03Juggieits really a matter of just reading the wiki.
23:18.06wazquis|Vulture|, okay, if you want to, you can leave me a mail with the result?
23:18.10Hmmhesaysit's been mentioned on the -dev mailing list but i don't think anyone has done it
23:18.27shmaltzthen I get an error that I can't remove apache
23:18.37shmaltzb/c php-pear isn't intalled.
23:18.37tzafrir_homeshmaltz: can you point me to a bug at RH's bugzilla about the perl problem?
23:18.40|Vulture|wazquis: yea just msg me your email
23:18.44terrapeni don't understand why anyone would battle with a linux distro just to get asterisk going
23:18.52terrapeni installed debian and never had a single problem
23:19.04Hmmhesaysdebian is nice
23:19.06shmaltztzafrir_home, I'm not sure and not even interested
23:19.09Juggieconversley, i installed fc3 and other then the udev crap, had no problems.
23:19.12marc324the te410p is supported only by highend mb.
23:19.16*** join/#asterisk rpoppi77 (~rpoppi77@201.24.15.125)
23:19.20shmaltzit's more the rest of the stuff that bothered me
23:19.33JuggieHmmhesays, why do you need the unique id for the call?
23:19.39shmaltztzafrir, ata mipoh? oh miaratz?
23:20.33HmmhesaysJuggie: so you can track a single call progress
23:20.38shmaltz~fc3
23:20.44shmaltz~fedora
23:20.45jbotsomebody said fedora was RedHat's alpha/beta distro made for testing out stuff to be put into RedHat later.
23:21.02Hmmhesays~debian
23:21.03jbothmm... debian is a genuine free distribution, it isn't only a linux distribution, but also a hurd, freebsd and netbsd distribution. Currently, the linux based system is the only one that is considered stable. Another good point is that non-free and free packages aren't mixed, in that way you can know what is and what isn't free software before install it.
23:21.56terrapensuch a nice day to be sick, too
23:22.01terrapenits beautiful outside
23:22.07terrapenfirst nice day in a week or so
23:22.31|Vulture|umm is google down?
23:22.35Mikewe dont have a nice day since 3 months or so
23:22.35HmmhesaysWhen you originate a call to a busy extension with the manager it will return a success and the channel will sit in a down state for about 30 seconds
23:22.37|Vulture|nvm
23:22.44|Vulture|it didn't work for like 10s
23:22.49myloforreali'd say with fedora it's just a matter of having more people on the job (read: volunteers) they can add more stuff in, and whatever works well they'll chuck into RedHat cos they won't have to do too much with it
23:24.23shmaltzok, here is the problem, I'm trying to take out the RMP intall from php-pear and http-selinux, which the system shows as installed, when I do that I get an error telling me that those are not installed. Where are the files that keep track of the installed RPMs located on the system?
23:25.57Kattymm, dinner.
23:26.42tzangerKatty: yup had it already
23:26.45tzangerhomemade chili
23:26.53tzangerI keep my bed warm at night :)
23:27.14Katty(=
23:27.17Kattyi have stirfry!
23:27.23tzangerstir fry is cool
23:27.23*** join/#asterisk Gronker (~Gronker2@adsl-220-89-19.ags.bellsouth.net)
23:27.28tzangerso long as there's no nuts in it
23:27.37Kattypeanuts
23:27.39tzangerewwwwwwwwwwwww
23:27.44Katty;)
23:27.56tzangerhaha
23:27.58tzangernot true
23:28.02tzangerI will pick out the peanuts
23:28.05Katty:<
23:28.12tzangernuts have no place in supper
23:28.15tzangeras a snack yo ubet
23:28.18Kattyi'm a vegan though
23:28.22Kattyit's my source of protein
23:28.31Kattyso there are...uhmm...LOTS of peanuts
23:28.32tzangerone of my favourite snacks is salted peanuts in the shell
23:28.55Kattythese peanuts aren't salted.
23:29.00tzangerdouble gross
23:29.48tzangeryou seem like a cool person and all but I'd never be able to survive as a vegan :-)
23:30.03tzangerhell when I was married my wife put us both on Atkins and I was a miserable SOB without my bread and dairy
23:30.14Kattyi still eat bread
23:30.14tzangerlord knows she tried... special flour and everything but it just wasn't happenning
23:30.16Kattyand soy cheese (=
23:30.44tzangerI love my meat and I love my dairy and even honey in my tea
23:30.52Kattyk'then (=
23:30.53tzangerbut I guess that's because I'm German and Russian.  :-)
23:31.12tzangerSo are you a level IV vegan or do you still eat food that'll cast a shadow?  :-)
23:31.16tzanger(it's a quote from the Simpsons)
23:33.00Kattyuhm
23:33.04Kattyi don't eat any animals
23:33.08tzangerI know
23:33.10Kattyor anything that comes from animals
23:33.19tzangerI know
23:33.20buddahno jello?
23:33.23buddah:(
23:33.24Kattywhich includes honey and gelatin, amongst other things
23:33.30tzangerI knew a (really hot) girl in high school who was like that
23:33.33*** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com)
23:33.51Kattybuddah: that's right, it has gelatin (=
23:33.52buddahwhy not honey?
23:34.00buddahhoney doesnt come from an animal
23:34.02tzangerbuddah: because it's stealing from bees
23:34.05Kattyit's a by product.
23:34.05buddah...
23:34.08buddahno its not
23:34.12buddaha bee is not an animal
23:34.15buddahit is an insect
23:34.18*** join/#asterisk Inv_arp (junya@adsl-3-247-135.mia.bellsouth.net)
23:34.20buddahbug
23:34.21tzangersealing from another living creature
23:34.24buddahok
23:34.28buddahthat i get
23:34.34Kattybuddah: http://www.vegetus.org/honey/honey.htm (=
23:34.56buddahk, reading
23:35.04Kattyk
23:35.22Kattyhmm, freezer calls.
23:36.20*** join/#asterisk Dr-Linux (adnan@202.147.168.142)
23:36.31*** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18)
23:36.35tzangeryeah I completely disagree with that site but that's alright
23:37.17*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
23:39.08buddahyeah
23:39.18buddahmy opinion has been formed, and i will keep it to myself
23:39.24tzangerha
23:39.30tzangereveryone is entitled to my opinion.  :-)
23:39.35Zawi'm currently sponsoring one vegetarian and one vegan. they hate me.
23:39.36buddahas i eat my bacon cheeseburger for lunch
23:39.36buddahyes
23:39.42tzangersponsoring?
23:39.50buddahbut i'm just not expressing it, since there is no port
23:40.02buddahlike Meat Eaters Anonymous
23:40.03Zawtzanger: yeah, i eat three times as much meat as i normally would, so as to compensate for them not eating any
23:40.57tzangerIf it weren't for the "exploitation" of animals you wouldn't be around here to talk about it
23:41.09tzangerZaw: I'm confused
23:41.29Zawtzanger: about what?
23:41.32KattyZaw: i'm not one of those hatehatehate vegans (=
23:42.09tzangeryou eat more than you need to to compensate for others who choose not to eat what you will??
23:42.10ZawKatty: oh, well the two i am currently sponsoring are way out of control. they actually try to force their beliefs upon you, so it was my way of getting them to shut up.
23:42.15Kattythose are just annoying, amongst other things...
23:42.26Kattyeek!
23:42.29Zawtzanger: yes. i eat their share of meat as well as mine.
23:42.30Katty:<<
23:42.48tzangerI'm cool with katty, I'm actually curious about it but not becaues I'm gonna be one just more for the curiosity
23:42.53tzangerZaw: odd
23:43.28*** join/#asterisk d-tech (~dtc@node-423a1ebb.cle.onnet.us.uu.net)
23:43.36tzangerseems like a good way to obesity
23:44.07Zawtzanger: nah, i work it off and it helps build muscle
23:44.23tzangerheh
23:44.39sivana~seen sixtel
23:44.43jbotsixtel <sixtel@sixTel.iax.cc> was last seen on IRC in channel #asterisk, 53d 18h 25m 58s ago, saying: 'no such host, not in sip.conf right'.
23:44.43Inv_arpis this correct syntax to pass vars to agi?  exten => 151,1,agi,callerid_agi.php|${IDCALLER}
23:44.46tzangermy chili'd be vegan-friendly if it weren't for the beef
23:44.54tzangereven the beer's vegan-friendly I think
23:45.37drvoipVoIP is vegan friendly.
23:46.29Kattyit sure is!
23:47.07Inv_arphow does one get callrid info thru agi?
23:47.40Kattywith a stick
23:47.51Kattyand you poke, or something.
23:47.52Kattyyes
23:47.54Kattythat's how it is.
23:48.38tzangerlast time I poked something I became a father
23:50.04bjohnsonwrong stick
23:50.09tzangerhehhe
23:50.13Kattygosh
23:50.49tzangerkram: do you want me to send you that Panasonic phone?
23:51.20Kattykram (=
23:52.07ManxPowerkram doesn't talk much
23:52.22Kattyk
23:52.32kramthanks katty for the smile
23:53.19*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
23:53.19*** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.6 Released || Dev Conf 1PM CST MARCH 3rd -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf June 8-10th more coming soon....
23:53.38djinwow, kram is checking in and out real quick.
23:56.03tzangergod damn
23:56.07tzangerKDE's been compiling ALL DAY
23:56.17tzangerI shouldn't have used -pipe
23:58.13myloforrealhow come my posts are faded (grey) instead of black?
23:59.50Kattyhi buddah!

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