IRC log for #asterisk on 20110605

00:01.05p3nguinAnyone here use Whistle and know how to register a second account?  I can't figure out how to create a second account using the same client.
00:01.49p3nguinI don't need to use both accounts on the single client at the same time, but I do need to create the account.
00:11.01*** join/#asterisk Moe__ (~MJames@c-71-62-116-235.hsd1.va.comcast.net)
00:13.48*** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf)
00:30.58p3nguinTerms of Service does not permit it, so nevermind.
01:11.22*** join/#asterisk tomaw_ (tom@freenode/staff/tomaw)
01:30.28WIMPyWasn't there a dahdiras or something to terminate data calls? But I guess that's only for X.75 or something.
01:52.16g_r_eekcan i ask a question about freepbx in here? i try to use the endpoint conf. mannager, i set up yealink phones inside there but i see my tftp directory is empty , is there another script that i need to do to populate it with the conf files?
02:02.59WIMPyThat's definitely something for #freepbx
02:03.15*** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista)
02:19.44showmekage:  I don't really have any idea who might want to do this, in theory it sounds like a pretty simple piece of software, but I'm just not a programmer, and to find someone willing to write it, would probably be someone that also has the need for it.
02:21.05WIMPyNot at all. It's a very complex thing to do a modem. That's why you hardly find any non-commercial attempts.
02:22.43coppicethat depends on whether you want to keep it legal, and whether you really want a 64 bit vesion
02:22.57coppicealthough doing a SIP anything is a PITA
02:24.51showmeIf you're going to support v.92 and everything, sure, I'd agree with you, but for a v.32 (9600) it should be fairly simple.  Back 10 years ago, most of the pci modems you bought were really just a sound card interfaced to the phone line and the "modem" was pure software, a "sip modem" should be as simple as just the software piece of that and a bit of network code, no hardware needed.
02:25.56coppiceshowme: now describe just how simple a V.32 modem is
02:26.26WIMPyThst has been commecial. And I have no idea, how well they worked.
02:26.29showmeI really wonder if some source code for those older modems could be found, if it wouldn't be fairly easy to adapt.
02:26.54showmev.32 is very well documented, and unlike newer stuff like v.90 and v.92, its not tied up with patents and legal junk.
02:26.55WIMPyshowme: Do it and become famous :-)
02:27.23showmeif I were a programmer, I would, I'm strictly a networking guy.
02:27.51showmeNot sure how you become famous with a piece of software that only about 15 people in the world would use...
02:28.35coppiceif you were a programmer you wouldn't have a clue where to start on a V.32 modem, unless you were also a DSP expert
02:28.57WIMPyYou should get an old NAS, if you're luck you find one for a tenner on ebay.
02:29.16WIMPy+y
02:29.38*** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com)
02:29.40showmeI might give you that, but if you could find the source code for one of the old pci modems, most of that code would be there, just add networking code and you're done..
02:30.06showmealthough I'm not sure if there were a bunch of v.32 software modems, they didn't really seem to get too popular until the 14.4 or even the 28.8 days..
02:30.37coppiceV.32 was the 14.4k modem
02:31.21showmeand for everything faster, software modems dominated the pci (internal) market, but that would be a bunch of code that probably wouldn't work.
02:31.37showmeactually, v.32bis was 14.4, v.32 never addrressed anything faster than 9600
02:31.48coppiceThe V.17 modem in spandsp is basically half a V.32 modem. You would need to add an echo canceller and an elaborate startup procedure.
02:32.11coppiceV.32bis is just the updated V.32
02:32.44coppiceand V.17 is the 14.4 variant
02:33.21showmemore technically, v.32bis was an extention to v.32, not just an "update", that would infer any 9600 modem could be updated with new software, definately not the case.
02:34.22coppiceif you look at the softmodem packages for Linux, like slmodem, you can actually fudge those to work with things like Asterisk and Freeswitch with a little DSP shim in the way
02:34.44showmev.17 is a fax standard for 14.4, faxing is a half duplex activity
02:35.01coppiceshowme: so you believe version 1 of anything can do 100% of what version 2 can?
02:35.17coppiceas I just said, V.17 is *half* a V.32bis modem
02:35.54showmeno, but in the case of v.32 and v.32bis, they really are different things, I argued 20 years ago that I didn't think it was appropriate to use the term v.32bis..
02:36.27coppiceV.32bis is just a V.32 modem stretched
02:36.32showmefaxing and modem communication, although similar, really are different beasts..
02:37.13florzshowme: coppice is the author of spandsp, you know ...
02:39.37showmethen he'd be the ideal guy to write a v.32 sip modem..
02:40.58coppiceV.32 just isn't interesting. V.34 half duplex for FAX will be interesting in 3 or 4 years when the patents expire
02:42.14showmeI don't have a need for faxing, but in my search for a sip modem, I did run across a couple sip fax modems...  I thought one I saw was open source, but I'd have to dig again to find it..
02:44.34showmev.32 has a very practical application though, tons of equipment (cisco for example) that has a serial console port that at default runs at 9600, thats one of the only reasons I keep an analog line in my home office, so the 3 times a year I need to dialup to something like that, I still can.  If I could do that over my voip, that would be a killer good thing for me, save me about $300/year, and just make life simpler.
02:45.36coppicewouldn't a TCP/IP<->serial gateway be more useful for that?
02:46.33showmethe reason I'd dialup to a router like that is if the outside IP link is down, so no, once IP dies, so does my gateway.
02:46.57coppicein that case you still need the analogue line
02:47.05showmeI'm getting in to check the T3 or whatever, becaues I can't get IP to the unit.
02:47.32showmeYeah, every where I've deployed such a unit, I require the customer to keep an analog line for that very purpose...
02:47.45showmeI want the sip modem for my end, not theirs
02:49.13showmeplus every now and then I come across an old bbs or something that doesn't seem to have telnet or ssh..
02:50.03showmeour city government even has a dialup (bbs style) for getting some city info, although "most" of that has finally migrated to the web.
02:50.30WIMPyshowme: Where in the world do you live?
02:50.56showmeI'm in the USA, a town in Missouri
02:53.02WIMPyThey knew BBSs? Cool.
02:53.09showmethe biggest thing on the city dialup thats not on the web is a complete phone listing for every city employee
02:53.26showmeYeah, the whole bbs thing was up in the early 80's before internet was even a thought
02:53.34showmeits just never been shut down.
02:54.05showmeits probably hitting 40 years old
02:54.16xxiaoplan to buy an ATA then setup asterisk at home, any recommendation on voip service provider? ooma(?), vonage?
02:55.07showmevonage doesn't allow you to connect directly to them via sip, only preprgrammed devices,
02:55.07xxiaoboth provides ATA i think, but i'd like to use vendor-agnostic ATA with asterisk instead
02:55.07*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
02:55.29coppicethe first BBSes appeared around 1980, so its more like 30 than 40 years
02:55.48xxiaoi.e. a voip service provide that has PSTN gateway worldwide and i can register/use without getting devices from them
02:56.09showmexxiao: it really depends on your usage, how much time do you use the phone, a lot of long distance or international?
02:56.37showmecoppice: your right, bad math on my end, but like I said, early 80's..
02:56.41xxiaoshowme: about 200-300 minutes international call each month, same for national long distance
02:57.18xxiaoi need it to block calls at certain time to avoid the ring waking my kids at nap/sleep
02:57.32WIMPyMany countries or mainly one?
02:57.49coppicethe word "community" is thrown around wildly these days, but early BBSes had a real sense of community
02:58.05xxiaoi'm using onesuite for international calls
02:58.14xxiaoWIMPy: half europe, half asia
02:58.26coppicelike my kids :-)
02:58.37WIMPyThen get an ITSP in europe and one in asia.
02:58.57showmeI used to have a 4 line bbs, it was definitely a community, we had a couple picnics a year and everything.
02:59.15WIMPyThose were the days...
03:00.10xxiaoshowme: nowadays it's twitter/facebook/sms and people hate to see each other
03:01.08xxiaoWIMPy: thanks. the "popular" voip service providers are all supplying their equipments while i don't want to be locked in by that
03:01.39xxiaoguess i'm just looking for a company has lots of PSTN gateway and SIP proxy/registra servers
03:02.05WIMPyI don;t know any that sell VOIP but aren't open.
03:02.42WIMPyThere are lots that sell phone lines that actuially are voip and only work with their IADs, but they don't sell (it as) VOIP.
03:03.17showmexxiao: I don't do much international, but your best bet may be to (your in the usa, right?) get one "main" provider, then look at providers in asia and europe that you can do low volume there too, its easy to have a ton of providers all on one single asterisk box.
03:03.18xxiaoWIMPy: ? you normally got a device(ATA) from them with one or two year contract, i want to use my own ATA instead
03:03.40showmeif you're doing your own asterisk box, you don't need an ATA at all
03:04.01WIMPyxxiao: Not here (EU).
03:04.33showmeyour asterisk box talks directly to your voip provider, no hardware inbetween (that is unless you're using a voip provider that locks down like vonage).
03:04.42xxiaojust curious, does EU's celluar contract work like US, i.e. one or two year contract?
03:05.01xxiaoI now in Asia they don't lock your cellphone to operators/contracts at all
03:05.05xxiaos/now/know
03:05.06WIMPyYou can get anything and more :-)
03:05.56xxiaoshowme: cool, that's what I'm trying to set up, want to get openwrt+asterisk on my little wireless router for family use
03:06.20xxiaoonly need buy a good ATA, best, with Wifi
03:07.23WIMPyYou don't want to do voip over wifi, usually.
03:07.57showmeunless you're wanting to use traditional land line type phones, you don't need any ata at all, your asterisk box talks to your (multiple if wanted) and you use sip phones in your house and they talk to your asterisk box.
03:09.05xxiaoi have 3 analog phones now that would like to continue to use...besides SIP phones are expensive still
03:09.33showmeoops:  your asterisk box talks to your voip provider (multiple if wanted) and you use sip phones (as many as you want) in your house and they talk to your asterisk box.
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03:10.49xxiaoif i use ATA who service the ring tone? asterisk, the voip-provider, or the ATA?
03:10.51showmethey've come way down, depending on features you need, you can actually find simple phones in the $30 range, you don't need those $400 cisco ip phones
03:11.25xxiaoshowme: let me google for a good sip phone then, but then i need run ethernet cable in the house if wifi is not optimal for echo etc
03:11.32showmethe voip signals the asterisk box, the asterisk box signals the ata, and the ata generates dial tone, ring voltage, etc.
03:12.32showmeanother option (not cheap, but not really expensive either), is panasonic makes a multi-line multi-handset sip cordless (dect) system.
03:13.49WIMPyYes, those things are an option. But I'd make sure it's CAT-iq.
03:14.16showmeyou can get the base as either just the dect box (needs power and ethernet, then dect to the phones), or you can get a base that includes one traditinal phone built in then add as many cordless handsets as you want (I think it maxes out at 8?)
03:14.55WIMPyI think the most bases support 6 handsets.
03:16.39showmeyour right, I just looked, it supports 6 handsets (plus the base phone if you get that model), they support adding up to 8 sip accounts to them..
03:17.17xxiaoit's possible i can hook ATA directly to the network(or SIP home) without installing asterisk, that should also work at the start
03:17.20*** join/#asterisk cesar_CR (~cesar@celord.ice.co.cr)
03:17.33xxiaos/SIP home/SIP phone/
03:17.41WIMPysure
03:18.05xxiaothat's how vonage/ooma etc works i think
03:18.26showmethat the ONLY way vonage works, they don't allow direct sip access
03:18.46*** join/#asterisk cesar_CR (~cesar@celord.ice.co.cr)
03:19.15showmeso if you wanted  vonage on your asterisk box, you have to get an FXO gateway to hook up to the vonage ATA
03:19.49*** join/#asterisk celord (~celord@celord.ice.co.cr)
03:19.57xxiaohas a classmate producing lots of DECT using DSP-groups solution for years....
03:20.02xxiaos/has/have/
03:20.08xxiaocan't type today
03:21.25xxiaoshowme: why? i thought vonage's ATA has ethernet and FXS, that's about it?
03:21.34xxiaowhy do I need a FXO gateway
03:21.39WIMPyIf you want to go voip, don't use the old dect, use cat-iq.
03:22.14showmeYeah, but vonage views it as closed, so there is no way to directly hook an asterisk box up to vonage.
03:22.38xxiaoshowme: no wonder vonage can't fly high
03:22.39showmethey won't give you the sip info, and the ATA is locked down to where you can't even log into it for any real purpose
03:22.47xxiaono vonage for me
03:22.58coppiceThe DSP-Group is a company with a weird history
03:23.06xxiaocoppice: why?
03:23.18showmeThe panasonic phones I'm thinking of are cat-iq
03:23.29xxiaothey're doing cortex-A8 with DECT block in it, will possibly run android as told
03:23.48xxiaoit will be a tablet + CAT-iq i think, 7" LCD
03:23.52xxiaoor even bigger
03:24.17showmepanasonic kx-tgp500 (pure cordless) and kx-tgp550 (cordless with one corded phone)
03:25.21coppicexxiao: they sort of became audiocodes, without really becoming audiocodes, and then reappeared as DSP Group again as a separate company
03:25.43xxiaocoppice: i feel they're doing well
03:25.57coppicexxiao: and much of what was the DSP Group is now called CEVA
03:26.15coppiceCEVA does well
03:26.47coppiceA8 + DECT sounds like an odd mix
03:26.55WIMPyWhy?
03:27.14coppicea high performance core + a low performance comms channel
03:27.35xxiaojust revisit dsp-group's site, saw "DECT Android Phone" there
03:27.52xxiaoCEVA, not sure how it is doing though. DSP-group seems booming
03:28.16WIMPyWith todays software I wouldn't call an A8 high performance.
03:28.55coppiceceva gets a cut from a large percentage of the cell phones sold
03:29.24xxiaocoppice: what does that mean?
03:29.38coppiceceva is the arm of DSP cores
03:30.42coppicefor some reason "there's an arm in every cellphone" gets a lot more publicity than "there's a ceva in a high percentage of cellphones"
03:31.19xxiaoyou mean ceva owns some DSP IP in cellphone, or inside ARM chip?
03:31.35xxiaosay, the NEON, or Mali(which is not really a DSP per se)
03:32.43coppiceceva licences DSP cores in teh same way ARM licences general purpose CPU cores, and they are very popular
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03:34.05xxiaoi c, though i don't what exactly ceva owns inside arm(or anything), only know TI DSP, freescale DSP, and ARM's Neon DSP
03:34.32coppiceceva owns nothing inside arm.
03:35.21xxiaothen where
03:36.23xxiaoon the A8+DECT, the A8 is also acting as a tablet to some extent, so people can touch the screen to do more things
03:36.25coppiceI just said. they licence DSP cores. If you see a chip that mentions it has an oak or teak core, it uses a core licenced from ceva. most cellular chips either have a TI core or a ceva one
03:36.39xxiaoi c
03:36.52xxiaogood to know something new
03:42.29stramhi, i installed asterisk 1.8.4.2 from source on a linux machine, it seems to be segfaulting out of the box: http://pastebin.com/xKmu46BM
03:42.42stramany tips? i can't seem to get friggin GDB installed to take a look at the core dump
03:45.06coppicexxiao: some of those DSP Group chips have a TeakLite core in them
03:49.13coppicexxiao: that Cortex + DECT shipset is really Cortex + 802.11 + DECT. That makes a lot more sense
04:04.01xxiaocoppice: right
04:04.21xxiaohope they can have that A8 in production soon
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04:37.42WIMPystram: Without gdb you kann only search bu hand. Disable autoload in modules.conf and load the modules one by one until you find the one that crashes.
04:38.03WIMPyYou can use a shell script however.
04:39.38coppicexxiao: there is an amazing flood of A8 and A9 chips appearing from people you've never heard of
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04:57.04WiretapSevenstram, make sure you're only loading ONE voicemail module
04:57.40xxiaocoppice: for A8 yes, for A9 not so
04:58.07xxiaoso far only tegra2 is selling in volume
04:59.35xxiaoi have a pandaboard and is less interested in it as you can not get the chips,anyway, all devices should have a light-arsterisk built-in
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05:26.44coppicexxiao: there must be at least 10 people shipping dual A9s now
05:27.30coppicexxiao: what do you mean when you say you can't get the chips in the pandaboard?
05:28.52stramWiretapSeven: thisi s completely out of the box, but ok i'll try that
05:29.18stramwait, WiretapSeven, it crashes even when i rm rf'd /etc/asterisk
05:29.35WiretapSevenstram, ensure your RAM is good, and ensure you compiled fro the right architecture
05:30.57stramWiretapSeven: its a xen domU, think that's causing issues?
05:31.53WiretapSevenvirtualisation and asterisk don't get along so well, very dependant on the type of virtualisation though
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05:49.29xxiaocoppice: well many claim they have A9 but only Tegra is selling them in  volume. omap4 is still quite hard to get unless you are a key customer to TI
05:49.59xxiaoAMlogic is the one I know is shipping in volume, single core A9 though
05:50.21xxiaohope i missed some major A9 players, I'd happy to find that out
05:56.16*** part/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net)
06:00.24coppicexxiao: the snapdragon, samsung's own dual A9, TI and others ship in volume. I don't know any big sellers with the Tegra 2. Its in things like the LG dual core phone, but that seems to be failing
06:00.48coppiceTI are on backlog for the dual A9s ships, because they *are* selling
06:02.04coppiceFreescale seem to be selling quite a few A9s into Chinese tablet makers, and they have single, dual and quad core A9s available
06:02.51WIMPyo.O   Quad A9, available?
06:04.18coppiceI somehow doubt the quad one will be a big seller, but someone told me he was actually playing with them
06:05.15WIMPyWould be nice for eco servers.
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09:37.37PoWeRKiLLhello
09:38.13PoWeRKiLLsomeone know what version of asterisk with wich version of spandsp is the more reliable for fax receiving ?
09:39.40irrootdont use 1.4 its not easy to get set up
09:39.58irroot1.6+ and spandsp 0.0.6
09:40.13irrootuse 1.8 if you looking at T.38 gateway
09:40.58PoWeRKiLLT.38 gateway mean asterisk receive the fax from sip to tiff ?
09:42.04irrootnope to/from a T.38 endpoint to a TDM endpoing
09:42.07irrootpoint
09:42.51PoWeRKiLLso with 1.6 and spandsp 0.0.6 asterisk can act as a T38 endpoint
09:42.53PoWeRKiLL?
09:44.14irrootyes endpoint only
09:44.42irrootthe gateway allows 2 different endpoints to send/recieve  [T.30/T.38]
09:45.52PoWeRKiLLI'm using 1.6.2.18 and spandsp-0.0.6pre18 and I have problem with receiving fax more than 1 pages
09:46.13coppicewhat happens with more than one page?
09:46.39PoWeRKiLLfax are getting cut
09:47.07coppicecut in what way? betwen pages? in the middle of a page?
09:47.09irrootmake sure timing is right faxes require timing to be critical
09:53.55*** join/#asterisk wonderworld (~ww@port-92-201-31-72.dynamic.qsc.de)
09:58.49PoWeRKiLLtiming is right
09:59.00PoWeRKiLLi have a digium card inside
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09:59.22coppicewhat makes you think the timing is right?
10:00.00*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
10:00.07PoWeRKiLLi use the conference and other stuff
10:00.11irroottiming set in asterisk.conf and you are using res_timing_dahdi.so ?? just having a card does not make timing right
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10:05.45PoWeRKiLLIs there a utility to run to test timing ?
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10:21.27*** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt)
10:21.32[sr]howdy friends
10:22.19[sr]asterisk CDR doesn't have the ability of showing me the info: time between rings and the answer? and also if a call was not answered?
10:27.15PoWeRKiLLcoppice I still get WARNING[19623]: app_fax.c:173 span_message: WARNING T.30 Page did not end cleanly
10:27.43PoWeRKiLLsr yes it's the diff between duration and billsec
10:28.02PoWeRKiLLfor answered and not answered you have the column disposition
10:50.20coppicePoWeRKiLL: what kind of digium card are you using? if its an E1 or T1 what is your clock source configuration?
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11:07.07[sr]PoWeRKiLL: hum, ok this is not really asterisk related but freepbx
11:07.10PoWeRKiLLcoppice it's E1 card but I don't have any pri on it
11:07.15[sr]my call logs doesn't show me nothing :(
11:07.32PoWeRKiLLsr I'm not using freepbx
11:07.55[sr]PoWeRKiLL: what do you use to have that information?
11:08.18PoWeRKiLLwich one WARNING[19623]: app_fax.c:173 span_message: WARNING T.30 Page did not end cleanly ???
11:09.20irrootwhere is the fax endpoint on DAHDI ?? or sip or the like
11:10.15PoWeRKiLLthe scenario is like this TELES GW -> Opensips -> Asterisk 1.6 Fax Endpoint (with Dahdi and digium hardware on it)
11:13.19irrootyou using T.30 on a IP link this is bound to be a problem
11:13.30irrootT30 SHOULD be used on TDM
11:13.42irrootT.38 on NON TDM
11:14.16irrooteven if the timing starts out fine UDP / IP / .... does not facilitate it stays that way
11:14.50PoWeRKiLLso what is the solution ?
11:20.44irrootuse T.38
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11:21.19irroota ATA on a local ethernet with A/U LAW can be reliable to a point
11:21.36irrootread the spandsp home page re faxing over IP
11:31.49irroothttp://www.soft-switch.org/spandsp_faq/ar01s03.html
11:31.59irrootCan I FAX reliably through a VoIP channel?
11:33.24irroothttp://www.soft-switch.org/foip.html
11:33.32irrootFaxing over IP networks
11:42.11*** join/#asterisk hetii (~hetii@87.99.51.172)
11:42.15hetiiHello :)
11:43.51hetiiI have access to sip provider that support t38 protocol to sending fax. I use asterisk 1.8 + freepbx. What i need else to be able to send/recive faxes ?
11:44.23[sr]PoWeRKiLL: i see, i have some problems with cdr with unexisting files
11:45.17hetiiwhat i can image is t38modem --> hylafax --> iaxmodem --> asterisk  but i`m not sure if i need all of that stuff ? Is there some other more easy solution ?
11:45.24irroothetti if it stays all sip a good ATA that does T.38 hooked up to asterisk
11:46.04irroothylafax <-> t38modem <-> gnugk <-> ooh323 <-> asterisk <-> SIP
11:46.11irrootis the setup i use
11:46.40irrootthen i also use T.38 GW to replace the last sip with T38<->T.30<->DAHDI
11:47.05hetiiok thats a bit twisted :)
11:49.44irrootit works well i promise :P
11:50.15hetiihmm but why you use gnugk and ooh323 when hylafax can comunicate over iaxmodem with *  i thought that t38modem is only required on this case by sip provider that support t38?
11:53.19coppicesadly, if you need to use H.323 and T.38 for FAXing, we haven't provided great options :-)
11:54.05coppicebut if you only use T.38, T.38modem + hylafax is the best current option
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11:54.16hetiion my case H.323 is not required
11:54.54coppicethen why does you chain look so weird? what's wrong with just asterisk?
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11:58.11hetiii don`t know how its look now on * 1.8, on the past when i want have fax sending/reciving solution i needed hylafax +iaxmodem+* and some pstn line But know  when i know that my sip provider support t.38 i want to use it. So the question is what i need to run it. As i see on freepbx there are option T38 Pass-Through so something else are required as well
11:59.40coppiceinstall spandsp. install asterisk. set the config files. FAX away
12:01.44PoWeRKiLLasterisk 1.6 with spandsp can act as a T38 endpoint or not ?
12:02.08PoWeRKiLLcoppice can you help to check why I got this error WARNING[19623]: app_fax.c:173 span_message: WARNING T.30 Page did not end cleanly ???
12:02.12hetiiI have isdn card on my * box but i don`t plan to use it to sending fax over it
12:02.15coppiceyes it can. only gateway is missing from 1.6
12:05.21hetiibut spandsp is only a librarie, as i suppose i need also something that will handle faxes like hylafax or whatever else that will allow send faxes to mail etc...
12:07.04coppicespandsp does everything hylafax does. you just need fax to mail and mail to fax scripts. various versions of which are around. keep a low profile doing email to fax or fax to email, or you may get lawyer's letters from J2
12:08.05hetiiJ2 ?
12:08.21PoWeRKiLLhetii J2 = eFax
12:08.38PoWeRKiLLcoppice how can I debug the page truncate problem ?
12:12.33hetiiok i need to read more about spandsp but all off that is strange a bit, for example i cannot understand why i should letters form J2 when i use spandsp to send few fax :/
12:12.37hetiior recive ;/
12:14.12coppiceif you only send a few there won't be enough in it for them to bother. if you send or receive a lot you may get letters demanding patent royalties, or that you get out the business
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12:29.16*** join/#asterisk gavimobile (~user@bzq-84-108-104-165.cablep.bezeqint.net)
12:33.52gavimobileI get this error when trying to install asterisk configure: error: *** termcap support not found (on modern systems, this typically means the ncurses development package is missing)
12:37.59tzafrir_laptopgavimobile, what distro is it?
12:42.20gavimobiletzafrir_laptop: centos5 new install baby
12:42.44tzafrir_laptopyum install ncurses-devel
12:43.19gavimobiletzafrir_laptop: and then try to ./configure again
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12:46.36*** mode/#asterisk [+o russellb] by ChanServ
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12:59.17[sr]PoWeRKiLL: problem solved, i didn't have the -addons installed after the upgrade from 16 to 18 :
12:59.18[sr]:)
13:00.54khussein78problem with connecting asterisk to cisco 1700v router
13:01.01khussein78logs is here http://pastebin.com/4UgCYAjY
13:01.32khussein78how can i debug this
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13:19.33*** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt)
13:19.37[sr]howdy again :)
13:19.58[sr]what's the best option to block an extension from going to the outside?
13:20.34[sr]sample, if it dials 234XXXXXX it'll go to the default outgoing route and goes outside
13:24.03*** join/#asterisk boazb (~boaz@bzq-82-80-219-90.red.bezeqint.net)
13:25.31hetiiyou can check callerid and setsome other route eg. to hangup
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13:46.16[sr]hetii: hum
13:47.16hetiiyou write your own configs or use freebpx ?
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13:49.05anonymouz666good morning!
13:49.41hetiior afternoon :)
13:52.38anonymouz666hetii: you are too fast :)
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13:55.35LemensTSIm using the hpec echo canceller, in /etc/dahdi/system.conf it has echocanceller=mg2,1   ..... is mg2 hpec?
13:56.53anonymouz666nops
13:56.56anonymouz666mg2 is mg2.
13:57.31anonymouz666it's the default dahdi echocan I guess.
13:57.45anonymouz666In the beggining I was using mg2 but I switched to oslec.
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13:57.56LemensTSIs oslec better then hpec?
13:57.58irrootoslec rox
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13:58.07anonymouz666hello irroot
13:58.16anonymouz666are you the master of t.38 gateway? :)
13:58.24anonymouz666LemensTS: I don't think so.
13:58.45irrootlo anon master mmm not sure but i have been getting working
13:58.52irrootmastered me perhaps it has
13:59.40irrootyou using it ??
13:59.49anonymouz666not yet, but I will soon
14:00.11irrootcool please get hold of me its timing critical as all T.30
14:00.32irrootIMHO T.30 must be on TDM
14:01.00irrootthe OOH323 updates allow it to work with t38modem / hylafax / gnugk
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14:01.53anonymouz666good to know, I have been using asterisk since 2004 never had to touch in h323 channels :-) I am lucky
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14:03.17irrootused all 3 versions of h323
14:03.34irrootthe chan_h323 is worst ooh323 best FYI
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14:11.16anonymouz666nice to see a setup with 8 e1's in the same machine and asterisk running rock-solid without a restart.
14:11.49tzafrir_laptopirroot, whis is the third?
14:12.06irrootthe old chan_oh323
14:12.21irrootit was run by i belive greek company in access
14:12.49irrootstarted with asterisk 1.[01]
14:13.03irrooti kept it in my distro till 1.4
14:13.31irrootswitched to patched chan_h323 @ 1.6 to support h323plus
14:13.48irrootbut could not figgure out why it would load unload and crash on reload
14:14.11irrootthen used ooh323
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14:20.53LemensTShttp://pastebin.com/20mQRHvz   <---anyone have an idea on this hpec error digium support is not open today
14:24.26PoWeRKiLLhow to go from t30 to t38 ?
14:25.31irrootpowerkill many ATA's do T.38 and your ITSP needs to support it too
14:26.37irrootlemensts you change network card ??  i believe the licences are bound to the hardware
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14:38.19LemensTSirroot: nope haven't changed it
14:38.51irrootjust checking when you signed up you needed the ethernet detials right ??
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14:39.27i_eat_childrengood morning
14:40.27i_eat_childrenim getting this error in console, i thought it was bug in 1.6.1 but it seems to be even more severe (causing higher CPU usage) in 1.6.2
14:40.28i_eat_childrenhttp://pastebin.com/vxGM2n5j
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14:42.26LemensTSirroot: i do not remmeber it had been a long time. i will try to reregister maybe that wil fix it
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14:44.34t_dot_zillaim getting this error in console, i thought it was bug in 1.6.1 but it seems to be even more severe (causing higher CPU usage) in 1.6.2
14:44.38t_dot_zillahttp://pastebin.com/vxGM2n5j
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14:48.46LemensTSechocancel=1024 will do more than 256 right, it just uses more cpu?
14:50.26t_dot_zillai dont know... i dont think anyone is here...
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15:25.38t_dot_zillahttp://nerdvittles.com/?p=743#comments
15:26.00t_dot_zillaor rather http://nerdvittles.com/?p=743
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15:51.45xxiaogosh, i had a sip2sip account that worked out of  the box, but ekiga.net account never worked
15:51.57nex_necishi, i need to add /quit
15:52.13xxiaotry to make calls between these two accounts, guess i should use two sip2sip instead
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16:01.16xxiaotwo calls between sip2sip worked well, now replacing sip2sip with my own asterisk setup
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16:53.55admin0hi .. does asterisk send all AAA records to radius or just auth only ?
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17:16.09admin0found my answer . thanks for listening
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19:06.13gavimobileremind me please where I need to put the language pack
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19:43.37xxiaoset up arsterisknow on a local IP, set up 2 extensions, but softphone can not register with it
19:43.56xxiaosoftpohne registered with public sip proxy just fine though
19:45.09xxiaois there a way for me to turn on  some debug on asterisk and see what's wrong?
19:46.27xxiaosoftphone reports "no response timeout"
19:49.31seraphiexxiao: on the CLI "sip set debug on" may help
19:56.43xxiaocan u please define "CLI" in this case, a command line at where?
19:56.52xxiaodo I need restart asterisk?
19:57.03xxiaonew to asterisk, started yesterday
19:57.31seraphiego to commandline and type "asterisk -r"
19:57.40seraphiethis will connect you to asterisk without restarting asterisk
19:57.45seraphieand give you an asterisk command prompt
19:58.02seraphiethis is called the CLI "commandline interface"
19:58.36carrartry #asterisknow
19:58.39xxiaoseraphie: thanks, just did it
20:05.07xxiaosame procedure tried on trixbox and it worked, strange
20:08.56gavimobiledoes asterisk need something special to receive faxes to a regular incoming trunk
20:10.22xxiaoalright, did 'service asterisk restart" and it now works with asterisknow
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21:23.21gavimobileam I running asterisk as root? http://pastebin.com/sBH19Eh7
21:24.07carrardoesn't appear so
21:24.23carrarRunning as user 'asterisk'
21:30.59gavimobilecarrar: ok, thanks
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21:46.01xxiaounder asterisk, i need use extension@my-IP to login then make calls, under freeswitch, i need
21:46.25xxiaomy-identity@my-ip to login, then use my-extension@my-ip to make calls,
21:47.36xxiaolooks like freeswtich allows one identity to have multiple extensions, i only saw "extension" under asterisk
21:48.03xxiaodoes asterisk carry the same concept as "identity" or "username", e.g. username@sip2sip.info
21:48.35xxiaoi thought they're identical, i.e. username _is_ extension, just the latter is digits and the former is easier to remember?
21:51.36WIMPyExtensions can lead to peers. Peers can have user names.
21:53.04leifmadsenextensions are not associated with a particular endpoint -- an extension is simply something that gets matched in the dialplan
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22:13.18xxiaoweekend project of asterisk/freeswitch is done, both are great stuff, look forward to dive in
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