irclog2html for #asterisk on 20050529

00:00.22*** join/#asterisk apardo (~apardo@80.26.164.145)
00:00.47blondewell.. basically i run a small tech support office with about 10 people... and im looking to get 10 voip setups and use asterisk as the pbx server
00:01.00flotoxok
00:01.17blondedoes it support that function?
00:01.29flotoxyes ...
00:01.40Lloydioyou will be wanting a que system
00:01.58Lloydiowhich is totally possible with asterisk
00:02.11flotox"que system"?
00:02.17flotoxor queue system?
00:02.36Lloydioyes sorry, you speak better english than me and i am english :P
00:02.36blondelloydio: excellent
00:02.45flotoxLloydio: lol
00:04.02blondei also wanted to set it up so that the employees can take their voip boxes home and access the pbx remotely, is that also possible?
00:04.27flotoxLloydio: If i were an agents How can I pick up a call in the queue?
00:05.11Lloydiowell as i understand it when all the agents are busy it goes into a queue
00:05.20Lloydioi havnt tried it myself but i know its possible
00:05.47Lloydioyes you can log in from home
00:05.50newmedianBlonde: agents can log in and out of a call queue from anywhere.
00:05.58blondei see...
00:06.04flotoxuhm ...;(
00:06.10blitzrageanyone used STUN to deal with SIP devices behind NAT? I'm curious if anyone has found it has helped with calls from Asterisk to a device behind NAT
00:06.46Lloydioi log into ours at home using NAT but you must remember to supress the silence in your sip phones
00:07.27Lloydioor else audio problems may occur with the audio side
00:07.40newmedianBlonde: is there anything in particular you wanted to do with this setup? Asterisk is very full featured.
00:09.37blondenewmedian:  i need to set up a fully functioning voip pbx for a tech support call center, with a queue system... and i need the employees to be able to work remotely using their voip boxes
00:09.47newmedianblonde: no problem.
00:09.54drbrownnewmedian: any ideas?
00:10.06newmediandrbrown: for your voicemail problem?
00:10.14drbrownyes
00:10.16rue_mohrLoRez
00:10.30rue_mohrLoRez you running asterisk?
00:10.38blondenewmedian: are there any extra services i need to sign up for?
00:10.53newmediandrbrown: well, I've got multiple directories under the default, numbered with the extension of the user. In there are wav files and an INBOX directory.
00:10.54flotoxuhm ..Agent need login to asterisk to pick up a call in queue ...
00:10.58flotoxis it true?
00:11.26blondefor instance, how can i get one toll free voip number to accept multiple calls and route them to the individual voip "extensions" ?
00:11.31blondeis there a special service for that?
00:11.49newmedianblonde: do you have phone lines/service now?
00:11.54blondeno
00:13.08newmedianblonde: you can either buy a card to go in your Asterisk server which takes PSTN phone lines and brings them into the system, or you can pay money to a provider and have everything traverse VOIP.  For conferencing and music on hold, you'll need a timing source, which will need to be some sort of hardware card anyway.
00:13.08*** join/#asterisk newbien (~e@116.242.33.65.cfl.res.rr.com)
00:14.08newmedianblonde: different people on #asterisk have different opinions regarding providers of such services, so best to ask around, but there usually isn't a technical problem in actually achieving what you want, other than any deficiencies in the provider themselves.
00:14.35newmediandrbrown: what happens when a user records a custom greeting in voicemail box; does a directory get created and a wav saved?
00:14.42blondepstn = standard phone lines?
00:14.51newmedianblonde: you may want to do some reading...
00:14.51newmedian~docs
00:14.52jbotDocumentation 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
00:14.59drbrownno, I am doing a search for wav files right now
00:15.10newmedian~pstn
00:15.11jbot[pstn] Public Switched Telephone Network
00:15.18blondeok
00:15.30blondei just needed a basis of where to start researching
00:16.03newmedianblonde: here's a graphic which might help visualize: http://www.digium.com/images/products/iaxy_install_diagram.gif
00:16.20*** join/#asterisk blop (blop@2001:6f8:204:33:bbbb:bbbb:bbbb:bbbb)
00:16.44int:)
00:17.03blondeis there a list of multi-line voip providers around? or can you name one off the top of your head?
00:17.20blondethe only voip providers i can find when i search are single line providers like vonage
00:17.30newmediandon't vonage
00:18.09newmediananyone want to give blonde some recommendations re providers?
00:19.02newmedianblonde: http://www.voip-info.org/tiki-index.php?page=VOIP%20Service%20Providers%20by%20Country
00:19.37newmedianblonde: you may want to try asking for provider recommendations during the week (not a weekend), at a different time of day.
00:20.08blondeok
00:20.13blondethanks for the help ;)
00:20.41blopanyone using AGI scripts in order to set a callerid ?
00:20.56blopgot some trouble here :)
00:21.30*** part/#asterisk rue_mohr (~rue_mohr@d154-20-50-233.bchsia.telus.net)
00:22.09blopdaaaaaaam
00:22.10blopfound
00:23.27*** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4075317.sympatico.ca)
00:26.04*** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net)
00:26.23dcaanthm, around?
00:27.28dcabut why...why anthm?
00:27.57*** join/#asterisk orbi (~dantate@pcp08696782pcs.500ash01.tn.comcast.net)
00:28.20dcahey, quick question, then i'm off to see star wars iii, can i set a time for rtautoclear? i.e. rtautoclear=100 or something?
00:28.48anthmno its directly related to the expiration event
00:28.51orbidca: don't bother going, i can tell you the ending right now.   JarJar Binks becomes Darth Vader.
00:29.02anthmbut you can control the expire time
00:29.07newmediandca: don't see it in DLP. I saw it in analog and DLP, and while it's possible something was wrong with the DLP projector, the DLP was far too pixelated.
00:29.33*** join/#asterisk Paul[NOC] (~azuretek@66.195.243.254)
00:29.52orbiLukesa, I am beinks your da-da.
00:30.09Paul[NOC]Hey, Strange question here; I have nat=yes in my sip.conf and under [xxx] I have nat=yes and when doing sip show peers it says under NAT "N"
00:30.15Paul[NOC]Is this the correct results?
00:30.45Sedoroxif it isn't using NAT.. yes
00:30.53Sedoroxjust because you set it doesn't mean it uses it...
00:30.55newmedianorbi: you mean... http://www.worth1000.com/cache/contest/contestcache.asp?contest_id=6091&display=photoshop
00:30.58anthmI read somewhere if you hate jar jar you can make the assumption he moved to alderan so in part 4 you can cheer when it get's blown up
00:30.59Sedoroxbut I have noticed the same thing
00:31.07Paul[NOC]Sedorox, how do you force it use NAT ?
00:31.14Paul[NOC]The phone is using a STUN server
00:31.15Paul[NOC]Hmm
00:31.24Sedoroxdoes it work?
00:31.42Paul[NOC]It doesnt seem to reach the * server
00:31.46Paul[NOC]it touches the STUN Server
00:31.47Sedoroxhmmm
00:31.49Wonkaaargh
00:31.52Wonka"get's"
00:31.54dcajar jar binks is darth vader!
00:31.59dcaOMG!
00:32.06Paul[NOC]and my sip debug dont work to great
00:32.37SedoroxI dunno.. I haven't messed with a STUN server.. never found a need for it yet
00:33.19Paul[NOC]Yea, I have a office setup here
00:33.31Paul[NOC]The * is in our new datacenter
00:33.34Paul[NOC]Office are behind a NAT
00:33.36Sedoroxok
00:34.03Paul[NOC]They dont like to talk
00:34.03Paul[NOC]Hmm
00:34.03dcaanthm: for some reason i have some sip peers with no ip addy associate to their rtcache (unspecified) and yet they do not clear, and if i prune them, when i do a load or place a call to that peer the address still does show up, yet, the addy IS in the db...
00:34.03Sedoroxyea.. but even when I was at school on NAT.. I didn't need a STUN server.. worked fine...
00:34.03dcaand this seems to happen often..
00:34.56*** join/#asterisk cochi (~foo@69.60.122.236)
00:35.00anthmso if you prune it then sip show peer mypeer
00:35.02cochimorning ;)
00:35.04anthmit says not found
00:35.09dcacorrect
00:35.18anthmthe when you add the load keyword
00:35.18dcaand if i do a sip show peer mypeer load
00:35.24dcano addy
00:35.25anthmit still has no ip ?
00:35.29anthmpos
00:38.07cochimh. anyone in here using asterisk with i4l and getting funny "ast_unregister_modem_driver not registered" errors?
00:39.13*** join/#asterisk iswm (iswm@iswm.user)
00:39.23Paul[NOC]Took it off STUN... hmm
00:39.38Paul[NOC]The one I have on a public IP works great :(
00:39.56*** join/#asterisk guugmember (~Casa@200.6.213.177)
00:39.57*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
00:41.12*** join/#asterisk predictive (~jeff@adsl-065-005-219-163.sip.cae.bellsouth.net)
00:42.37timecopsip + nat = waste of time
00:42.49predictiveanyone know how to make asterisk stop generating phantom rings from battery inversion on a digium fxo
00:43.12predictivenone of the 'stop annoying me' flags seem to work
00:43.40predictiveif sip and nat is a waste of time, sip is a waste of time, cause nat isn't going anywhere
00:43.48predictiveunfortunately
00:45.59predictiveI think that's a big reason people bother with skype still
00:48.50guugmembergreat new digium webpage
00:49.57predictivedual fullcone is the bitch
00:50.35predictiveso nobody has weird polarity inversions on pstn lines?
00:50.36blitzrageheh... luckily most clients are only behind a single NAT device
00:51.07blitzrageheh
00:51.43blitzrageI have a client which I'm going to have to deal with 2 analog lines for incoming and outgoing calls...
00:51.46blitzrageoh the joys
00:51.52predictivewell usually it's easy
00:52.11predictiveit's just I can't make asterisk stop ringing the line when it detects polarity inversion
00:52.17blitzragehrm
00:52.18predictiveyou should be able to tell it to ignore it
00:55.08timecophttp://validator.w3.org/check?uri=http%3A%2F%2Fwww.digium.com%2F
00:55.09timecop:(
00:56.50timecopvery bad design in general, the layout is so simple, but they've abused tables and other crap.
00:56.55timecopnot to mention doesnt validate
00:57.08timecophow can an opensores company have a webpage that doesnt validate
00:57.13timecopunfuckingacceptable
00:57.35predictivemaybe they were spending time trying to make money instead
00:58.03Nivextimecop: maybe they're looking for a web developer :)
00:58.24*** join/#asterisk bjohnson (~bjohnson@ip219-172.dsl.istop.com)
00:59.03blitzragetimecop: heh thanks - need to fix a couple of things on my site I guess ;)
00:59.26timecop:)
00:59.51blitzragethey look minor - woot! << 3rd most popular non-word by mw.com :)
01:00.39timecopfew months ago i found out w00t/woot actually stood for "We Owned the Other Team" and its some gamer speak
01:00.47timecop:(
01:00.52timecopfrom microsoft.com no less
01:01.06timecophttp://www.microsoft.com/athome/security/children/kidtalk.mspx <<
01:01.12bewestthat is disturbing
01:01.41predictiveI got a couple of gxp-2000s
01:01.46predictivethey aren't the greatest phones
01:02.01Paul[NOC]Anyway to make Asterisk bind to two SIP ports (5060 and 5061)?
01:02.06predictivehaha there's a leetspeak primer on here!
01:02.09Paul[NOC]I took port=5060 outta sip.conf
01:02.14cluecon[file]you can't expect the greatest for the lowest price
01:02.15Paul[NOC]and added it for each peer (friend)
01:02.33Nivexpredictive: someone sent me that back in February.  I laughed heartily.
01:02.38predictivecluecon[file]: yeah, I know, but much above that and sip is no longer economically attractive
01:02.47predictiveit's just expensive toys
01:03.20timecopPaul[NOC]: huh? why the hell do you want it listening on more than one port for sip?
01:04.01Paul[NOC]timecop, I know it sounds weird but trying to communicate using nat with multiple phones
01:04.04Paul[NOC]Hmm
01:04.07Paul[NOC]SIP isnt a constant connection
01:04.22Paul[NOC]Thats the only way I could figure it could work...
01:04.24Paul[NOC]One second
01:04.56predictiveNivex: if the only decent hardphone is $300 there's not much to look forward to
01:05.39MrBelvedri am using teliax trying to dial this number in poland 0148226306306
01:05.54MrBelvedrwhy isn't it going through, it is saying invalid extension
01:06.03Nivexpredictive: I was talking about the microsoft leetspeak thing
01:06.09predictiveNivex: oh
01:06.19predictiveNivex: pwned isnt on there
01:06.37predictiveI should file a bug report
01:06.55timecopPaul[NOC]: uh, and?
01:07.04timecopPaul[NOC]: you dont need the signalling port multiple times.
01:07.08timecopyou need different port ranges for rtp
01:07.18timecopbut signalling can go just fine over 5060 for all the phones
01:07.19timecopi guess..
01:07.26timecophm
01:07.27timecopor not.
01:07.32timecopfuck nat, get a netblock
01:08.32predictivePaul[NOC]: we run SER dual homed with one port external and have * autocreatepeer
01:08.53predictiveif you use a lot of sip stuff you'll want a real proxy eventually anyway
01:09.26cochi*Sigh* so SER is the best solution for natted sip?
01:09.33predictiveI dunno, works for me
01:09.34cochiseems to be a config-beast :<
01:10.45predictivethe config is like a nasty hybrid of old cisco and mod_rewrite
01:10.47predictiveinteresting choice
01:11.18cochibrr ;)
01:11.25cochiso slightly better than sendmail / bind? ;)
01:11.38predictivewell way better than sendmail but that's not saying muc
01:11.39predictiveh
01:11.43cochi;)
01:11.55cochiwell first i gotta get * flying with I4L anyway
01:12.00cochigot a baaaad feeling about it
01:12.02cochi*sigh*
01:12.18blitzrageyay, my main page is validated! :)
01:12.25cochierr... congrats
01:12.34cochias what? ;) HTML 4.01? ;)
01:13.10cluecon[file]blitzrage: March of the Swivelheads!
01:13.15blitzrage4.0 Transitional :)
01:13.21cochio.O that exists?
01:13.32cochiwhy not heading to XHTML 1.1? ;)
01:13.44blitzragebecause I'm no web developer
01:13.54cluecon[file]he's blitzrage!
01:13.55cochi'tis a reason, yeah ;)
01:14.05cochia bird, an airplane? no! blitzrage
01:14.11cochiscnr ;)
01:14.14blitzrage<< documenteur
01:14.24cluecon[file]documenter extraordinaire
01:14.37cochibienbien
01:15.17blitzragecluecon[file]: yah... it was supposed to be documenteur... :(
01:15.30cluecon[file]awwww
01:15.56blitzrageoh well :)
01:18.09predictiveheh
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01:25.37Nuxi<PROTECTED>
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01:27.25cochi<img src="http://0.0.0.0/rubbish.jpg" />
01:27.31cochi;)
01:29.54*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
01:29.59Ariel_hello everyone
01:30.23Nuxi<script language="extensions.conf">exten => _.,1,AGI(format.pl)</script>
01:34.23cochimh. so nobody here who knows about I4L problems :| (ast_unregister_modem_driver)
01:34.34cochimight be just some dumb error in loading order or so *shrug*
01:34.43cochi<- n00b ;)
01:38.33cochisleepy noob. cu then
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01:46.24asterisk99Does anyone know where I can find a nice simple zapata.conf file that answers 1 incoming line?
01:46.55asterisk99Also... What does "Channel Zap/1-1 sent into invalid extension 's' in context default mean?
01:48.54anthmit means "run while you still can!"
01:49.39Ariel_asterisk99, it means it's going into the wrong context
01:51.39asterisk99Ariel_: There's something I;m not unnderstanding about zapata.conf... does one define exten => statements in it for the incoming PSTN line in zaata.conf or in extensions.conf?
01:52.21asterisk99Ariel_: (sorry, bad english) There's something I'm not unnderstanding about zapata.conf... does one define exten => statements for the PSTN in zaPata.conf or in extensions.conf?
01:52.37anthmit means make a [default] in extensions.conf and make it contain exten => s,1,Playback(demo-thanks) <-now it's too late to run I warned you
01:52.40Ariel_zapapta.conf defines the port and which context to send calls to in the extensions.conf. The actual dialing rules are in the extenstions.conf
01:53.25Ariel_asterisk99, this is a starting place for you to do some reading: http://www.automated.it/guidetoasterisk.htm
01:53.34asterisk99Ariel_: Hmmmmm. The examples I've read are not clear as to whihch statements go into which .conf file
01:55.08asterisk99Ariel_: Is zapata.conf similar (analagous) to sip.conf  [I got that to work fine]
01:56.47Ariel_asterisk99, well it's in a small way yes. It sets your analog ports.  Sip.conf sets your sip accounts.  But there very different in what items go in it. But have the same logic
01:57.02drumkilla<PROTECTED>
01:57.08asterisk99Ariel_: Ahha!!!!! (The poprch light goes on!!!!)
01:57.12asterisk99Ariel_: Ahha!!!!! (The porch light goes on!!!!)
01:57.34Ariel_drumkilla, are you back from you trip?
01:57.41drumkillayeah, just got in last night
01:57.48asterisk99Ariel_: The examples are not very clear as to what exactly goes in zapata.conf and what goes in extensions.conf
01:57.52Ariel_hope if was good
01:58.11Ariel_asterisk99, what is it you want to do?
01:58.43asterisk99Ariel_: I want incoming calls to route to IVR to promppt for extension #
01:58.46drumkillaAriel_: yes, it was.  thanks for asking :)
01:59.20asterisk99Ariel_: (I also want my fingers to work proerly while typing, but that won't happen for years and years!)
01:59.25Ariel_asterisk99, have you seen the project called asterisk@home?
02:00.04asterisk99Ariel_: Yes. But all I really want need from asterisk@home is a listing of the various .conf files
02:00.47asterisk99Ariel_: i.e. What's in zapara.conf, what's in extensions.conf, ya-da ya-da ya-da
02:00.59asterisk99Ariel_: i.e. What's in zapata.conf, what's in extensions.conf, ya-da ya-da ya-da
02:02.14mepplgute nacht
02:02.20asterisk99Ariel_: Wowie zowie... I moved my simple exten => dialplan to extensions (from zapata.conf) and it actually answered the phone!!!!!!  (I know... Big Deal)
02:02.47asterisk99Ariel_: Now for instructions to do something useful
02:06.00blitzragedrumkilla!!
02:07.05blitzragetimecop: lol, damn you, just spent an hour making my website HTML 4.01 Transitional valid :)
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02:30.41blitzragewow.. it got dead in here :)
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02:37.32shmaltzhelo everybody
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03:02.49AtacommSay, anyone else ever notice that the GSM codec uses the function gsm_div() to divide, which runs in a while() loop and subtracts to get the answer?  I know the code for GSM is from like 1992, but last time I checked, CPUs have a DIV instruction...  imagine dividing 3000 by like 5....  thats alot of wasted cycles....
03:04.12shmaltzAtacomm, this looks like it belongs in #asterisk-dev
03:04.57Atacommlol, well that would be the more specific place, but last time i chceked there's alot of talk about bugs in here, lol...
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03:31.37drbrownDoes anyone have any idea why zaptel would give me an invalid module format error (when loading the module)?
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03:40.52MikeJ[Laptop]drbrown, because you compiled it against the wrong kernel source
03:44.16drbrownI believe it is the correct source, I must be making a mistake
03:44.28drbrownI am downloading the debian kernel-source
03:44.41MikeJ[Laptop]2.4, 2.6?
03:44.51drbrown2.6
03:44.57doolphhow can I remove this in database
03:45.00doolph/RECORD-IN/200 : DISABLED
03:46.12drbrownmy kernel version and source match up
03:46.22heragcan someone help me with my telasip configuration, dialing in just won't work, "auth failed" sip.conf: http://pastebin.ca/12877 sip debug: http://pastebin.ca/12832
03:46.43heragI've been stuck here for four days, and I just can't figure out what's wrong
03:47.01*** join/#asterisk AgiNamu (~bob@200.6.215.48)
03:47.02drbrown<Mike any ideas?
03:47.12cluecon[file]herag: insecure=very in your peer entry
03:47.20heragcluecon[file]: tried it
03:47.25heragno success
03:47.29cluecon[file]then it's coming from a different place
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03:48.00cluecon[file]no coming from the right place, but are you positive you put insecure=very?
03:48.09AgiNamuwell, the doctor fucked it all up
03:48.11AgiNamumy baby died
03:48.23cluecon[file]:(
03:48.23heragcluecon[file]: I remember doing it, I'll try it again right now
03:48.28AgiNamuincompetent piece of shit
03:48.56AgiNamushe was so cute too... www.atrevido.net -- i have a pic of her, scroll down ~half way
03:49.18AgiNamuSo how'd that 911 conference call go?
03:49.23heragcluecon[file]: crazy thing is, if I put the insecure = very, then the call doesn't even show up in my console...totally gone
03:50.13cluecon[file]herag: insecure just tells asterisk not to do any authentication and match based on IP address
03:50.51heragcluecon[file]: I know...tell that to my asterisk box...I thought that would be the solution, but the call doesn't even arrive at the asterisk box if I put insecure = very
03:52.20cluecon[file]drumkilla: how is my Russell Wussell doing today?
03:52.31MikeJ[Laptop]hey.. wassup with the cheese fight
03:52.32drumkillaok, just hangin' out in the office
03:53.08MikeJ[Laptop]nice
03:53.14MikeJ[Laptop]errr.. not nice
03:53.19doolphhey
03:53.24drumkillai'm not doing any actual work, heh
03:53.26doolphhow can i remove something from database
03:53.26MikeJ[Laptop]HEY
03:53.28MikeJ[Laptop]good
03:53.34MikeJ[Laptop]delete it
03:53.40doolph/RECORD-OUT/200 : DISABLE
03:53.42MikeJ[Laptop]what database?
03:53.46doolphdoesnt want go away
03:53.54drumkilladatabase del ...
03:53.55*** part/#asterisk Atacomm (~dan@69.54.45.98)
03:54.03doolphdatabase del /RECORD-OUT/200
03:54.04doolph?
03:54.09drumkillaindeed
03:54.12MikeJ[Laptop]what database?
03:54.16doolphUsage: database del <family> <key>
03:54.17MikeJ[Laptop]astdb?
03:54.28drumkilladatabase del RECORD-OUT 200
03:54.45doolphahh cool
03:54.49doolphty
03:55.10doolphcool
03:56.24drumkillay' welcome
03:57.10MikeJ[Laptop]so why the hell are you at the office
03:57.22cluecon[file]drumkilla is sexy!!!
03:58.01drumkillai dunno
03:58.13cluecon[file]it's the truth.
03:58.17drumkillaI shouldn't be, heh
03:59.05file[mac]:(
04:00.27doolphanyone has installed areskicc
04:11.39file[mac]the channel is oddly quiet
04:11.59doolphfile any good billing system open source that you recommend?
04:13.30file[mac]nope
04:14.24doolphnope what
04:14.36file[mac]I know of none
04:14.46doolphhow come
04:14.56file[mac]because...
04:15.36MikeJ[Laptop]drumkilla, tell mark his shirt is right ;)
04:16.00drbrown<Mike figured it out
04:16.27drbrownhad to change from PENTIUM4 in .config to M686
04:16.33drbrownthanks
04:17.37*** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
04:21.08*** join/#asterisk asterisk99 (~chatzilla@modemcable111.209-131-66.mc.videotron.ca)
04:22.00asterisk99newbie question... where do I put ztcfg in a Gentoo installtion so that it is executed on startup before Asterisk does?
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05:08.43bsdfreakheh
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05:31.11cp5hola
05:33.11MikeJ_hello
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05:34.26MikeJ_hey boris
05:34.31BoRiSHi guys :)
05:34.39BoRiSWhat's up?
05:34.40MikeJ_any luck on your realtime pach?
05:34.43MikeJ_patch?
05:34.59MikeJ_issue the other night we were talking about
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05:35.54BoRiSI am but haven't had much time to work on it. :(
05:36.03MikeJ_:(
05:37.05BoRiSSo how's your weekend going so far? Did you do anything exciting?
05:37.15MikeJ_ummmm
05:37.21MikeJ_hmmm
05:37.25MikeJ_pulled out stumps
05:37.30MikeJ_graded dirt
05:37.39MikeJ_spread grass seed
05:37.57MikeJ_worked with neighbor on building a deck
05:38.05MikeJ_is that exciting?
05:38.17BoRiShehe
05:38.46MikeJ_I live in the suburbs... grass is important or somthing
05:39.17MikeJ_so being the only house with a dirt front yard is frowned upon or somthing
05:39.18newlBrick work will fix the grass problem. 8)
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05:39.39MikeJ_that;s the next project ;)
05:40.00BoRiSNeighborhood pressure?
05:40.27MikeJ_yes and no
05:40.39MikeJ_it's that unsponken peer pressure
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05:54.50*** join/#asterisk io_error (~error@gw.ioerror.us)
06:02.45*** join/#asterisk morris (~turntabli@ACD6B1A8.ipt.aol.com)
06:03.04morris*yawn*
06:10.09orbiyawning is contagous, darnit.
06:10.39morriseofl
06:10.42morrisrofl*
06:11.11morrismaybe, i am a terrorist and maybe i was trying to infect you all!
06:11.32morrisbtw, i have a tonsil the size of a football at the moment :(
06:11.38*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
06:11.42orbiHmm.  My Echelon@HOME program just picked up your admission and transmitted it to the FBI.
06:11.51morrisrofl
06:12.19orbioo. tonsilitis?
06:15.02morriswell
06:15.04morristhats the thing
06:15.07morrisfirstly, no pain
06:15.20morrissecondly, my right tonsil is hanging into my throat
06:15.40morrisand at last examination by my trusty index finger reported that it was quite lengthy
06:15.45morrislike a second penis
06:15.58morristo be fair i can joke now
06:15.58morrisbut
06:16.05morrisi did take a walk to the hospital
06:16.08morrisjust to make sure it was ok
06:16.08morris;/
06:17.12morrisbtw, i ment left tonsil. not right.
06:18.34orbiwell
06:18.37orbihope it gets better
06:20.42morristhanks orbi dude
06:20.58morrisi thought it was gonna be something extremly nasty
06:21.00morris;p
06:21.05morriscancer or something
06:21.06morrislol
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06:55.32bintuthello all..
06:55.57bintuti just heard about asterisk and i hope someone could answer my questions
06:56.06*** join/#asterisk TechnicGeek (~asd@c-67-173-97-156.hsd1.in.comcast.net)
06:56.18bintuthow does asterisk work as a voip/pbx?
06:56.34io_errorbintut: uh, you install it and configure it? :)
06:57.05TechnicGeekbintut if your new to asterisk then I suggest asterisk@home, do a search on google
06:57.08io_errorhttp://asteriskathome.sourceforge.net/
06:57.20TechnicGeekor that.
06:57.46io_errorI'm in the middle of installing AMP right now. A@H looks a little too simplified, and I don't want it jacking up my system config :)
06:58.04io_errorBut for someone with little experience it's probably the best way to start.
06:58.29doolpha@h is too complicated for me
06:58.55Sato1then the normal asterisk would be impossible for you
06:59.13doolphya for real
06:59.43TechnicGeekI am having a problem on a new install of asterisk,  when I call any extension I get "486 busy here"  This is my 3rd asterisk box and on the others I remember some problem with permissions or files not being found correctly however I can't find the reference page on it anywhere.  Does anyone have any suggestions?
07:00.30bintutTechnicGeek: thanks. but my concern is on how does it work let say from point a to point b... i want to setup a voip/pbx box.. if point b calls to point a where the asterisk is install, how does point b take its call? through ip or a number?
07:01.00Sato1number
07:01.26TechnicGeekbintut,  They can get it through a sip software phone or you can get a voip termination service
07:02.10TechnicGeeka voip termination service will route the call through a pots like which would be their regular phone number
07:03.03bintuti'll make a diagram to make you more understand what you i want to achieve and hopefully you can give me tips on how can i implement it using asterisk
07:05.00TechnicGeekHas anyone had a problem with all extensions ringing busy or going right to voicemail?
07:05.06*** join/#asterisk [hC] (~hardcore@c-66-176-181-76.hsd1.fl.comcast.net)
07:05.35[hC]any of you guys dealt with branch-office voicemail setup?
07:06.43bintutmultiple voip phones <--> vpn ipsec gateway <---> [internet vpn ipsec] <--> vpn ipsec client <--> multiple voip phones
07:06.51TechnicGeekhC, there are many different ways to do something like that.  Can you give us details as to what exactly you need?
07:07.33TechnicGeekbintut, so many different phones connecting to one server via vpn?
07:07.47[hC]Well.. Ive got some extensions in office a, some in office b. I want the voicemail system to still have full functionality so that people from either office can forward messages on, etc, this isnt a HUGE deal,b ut it would be nice. The scenario ive thought out so far indicates that the voicemail 'server' has to live in one central spot
07:08.06[hC]what i do currently is instead of passing to voicemail() on one, i dial the other system and have a direct-to-voicemail extension drop it off
07:08.14[hC]This works fine, however i lose things like MWI
07:08.41bintutTechnicGeek: yes for both ends... let's say for both ends they have 5 analog/soft phones and each of them can call at the same time to any of both ends
07:09.10TechnicGeekhC so you have multiple asterisk servers?
07:09.34[hC]TechnicGeek: yep. because there are multiple people in each branch office. and the offices are by no means close.
07:09.36[hC]Miami and London.
07:09.37[hC]:)
07:10.22TechnicGeekhC, i've seen information about doing this sort of thing at voip-info.org
07:10.23bintutTechnicGeek: i don't know if i really need pbx (connecting to local numbers for both ends).. all i need is the voip phones either ends can directly call either ends
07:10.42[hC]TechnicGeek: i've been looking and i havent found anything that goes over the same points yet.
07:11.00TechnicGeekbintut asterisk works good for that sort of thing, I suggestion getting asterisk@home, its easy to setup different extensions for people
07:12.01TechnicGeekhC, what I usually do is just setup one asterisk server for all the offices. I am not sure of a way to do what you need
07:12.25TechnicGeekbintut of course you can always use like skype heh
07:12.27bintutthe only feature i need for now is to call both ends directly, logs, records the conversion for future use and can play the recorded discussions
07:12.44Silik0n<PROTECTED>
07:12.48[hC]TechnicGeek: and have say, 10 sip devices in london connect all directly to your master server, say in miami?
07:13.21TechnicGeekwell, if you use iax2 theres less overhead than sip
07:13.37TechnicGeekhrmm
07:13.41[hC]well how can you do iax2 from a phone? :)
07:14.01Silik0nwith a iAXY
07:14.26[hC]I suppose theres that. It makes sense for us though, to have an actual * box in the remote office, for other reasons
07:14.34TechnicGeekhC, I just thought of something.. setup asterisk as just a proxy at each branch they connect to your main one
07:14.46alt[hC]: just use SIP.
07:14.52bintutguys, do i need to subscribe to a voip gateway even if both ends are tunneled through vpn ipsec?
07:15.01TechnicGeekso all voicemail etc resides on the main box
07:15.02Silik0nactually putting an asterisk box at each branch using sip or iax can lead to a list of options
07:15.10[hC]TechnicGeek: so, 3 * boxes? a main, and two branches
07:15.20TechnicGeekyea
07:15.37altI was dealing with a company that wanted only one box to handle voicemail "for less administration overhead".
07:15.41bintutTechnicGeek and [hC]:  guys, do i need to subscribe to a voip gateway even if both ends are tunneled through vpn ipsec?
07:15.48altthey have 5 remote offices plus the main office
07:15.53TechnicGeekbintut no
07:15.59[hC]thats an option i guess. I was hoping that I could solve it without having to add another box
07:15.59Silik0nthings like LCR for calls if they are in different rate centers, unified voicemail and faxing, etc etc
07:16.00altso if any site lost their link, they'd lose voicemail too.
07:16.04altI just shook my head.
07:16.25[hC]Its not a bad idea to just have voicemail internal to each branch
07:16.26[hC]i do like the idea
07:16.32altSilik0n: technically, even if you only have one box, you can do LCR in different contexts for each site.
07:16.34[hC]but then you lose the ability to forward vmail to other people
07:16.37Silik0nalt: you would be suprised how many major corps had 1 voicemail box for multiple locations
07:16.39[hC]I guess you just tell them to forward via emaiol
07:16.41[hC]er, email.
07:16.58bintutTechnicGeek: so how can both ends call to a particular phones?
07:16.58Silik0nhc: how so
07:17.00altSilik0n: well, they were going to put asterisk boxes in each location...
07:17.09altso, why not put voicemail on each as well?
07:17.41altand because all the sites are interconnected (they have their own backbone), there's no need to worry about getting voicemail at another office.
07:17.42Silik0nactually thats a feature that needs to be put in app voicemail is forwarding to off system voicemail boxes...
07:17.48TechnicGeekbintut, if they all connect into the same asterisk server, they just dial the extension
07:17.49Silik0n(there is a standard for that)
07:18.07altSilik0n: just route the call to an off-local extension. let the vmbox handle all that.
07:18.23altthat's how I used to do it with our Cisco Callmanager + uOne voicemail.
07:18.40altthe only problem with my method is the lack of MWI.
07:18.42TechnicGeekI'm having a problem with all extensions going right to voicemail or busy.  Does anyone have any suggestions?
07:19.05Silik0nalt: theres a specific standard (and the name escapes me at the minute) that does inter-system Voicemail that is open and not vendor specific
07:19.21altSMSI I think.
07:19.26[hC]alt: so you go for centralized voicemail, or voicemail-per-office?
07:19.35Silik0nalt: no thats not it
07:19.58alt[hC]: if you're putting an Asterisk box on each site, then per-office for sure.
07:20.02Silik0nactually MWI could be handled with extern-notiication
07:20.24[hC]I was going to go for the centralized option, where all vmail goes to one, but i think im going to opt for per-site. the only feature you lose (i think?) is the ability to forward a voicemail to someone on the other system.
07:20.24bintutTechnicGeek: the branch office A is not using asterisk since it's been running for years already. now since we need to call frequently to branch office A and vice versa to this new branch, i will setup asterisk on branch office B
07:20.25altSilik0n: I did it with a monitoring script I wrote in PERL and sent via async serial :P
07:20.39alt[hC]: yeah, you'd lose that.
07:20.53[hC]alt: all the other options, that dont do a per-site vmail system, you run into issues on MWI, or vmail delivery if the path to the main server is dead
07:21.13Silik0nalt: you know theres a hook for external scripts to run on changes to the number of messages right
07:21.29altoh, this was my CCM, not Asterisk (sorry for the confusion)
07:21.52altbut yeah, I do vaguely remember something about that.
07:22.15Silik0ncheck app_voicemail.conf and look at extern notify ;)
07:22.30TechnicGeekI remember reading something about it hC.  Its asterisk and linux, theres a way to do what you need.
07:22.33altit's not something I need myself, but it is handy.
07:22.43Silik0nthat script (if defined) is called when the box changes so it can be used with a number of things to turn on/off MWI etc
07:23.00[hC]here's something.
07:23.00[hC]http://www.sineapps.com/news.php?rssid=691
07:23.07[hC]for mwi notifications from another server.
07:23.16altactually, the app_voicemail.c file is remarkably easy to modify too. I added a "press 1 for redirect to mobile/alternate number" feature.
07:23.21altabout 10 lines of code.
07:23.24Silik0nyeah
07:23.45Silik0napp_voicemail is just horrible code thats on my list of projects to rewrite
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07:24.12bintutTechnicGeek: any suggestions?
07:24.47altSilik0n: I'd like to see better options handling. one line for all the options is kinda rude :P
07:25.07TechnicGeekbintut, what I thought you needed was a bunch of people just to connect in to a box to talk.  What you actually need is something alot different.  Can you please explain it again
07:25.08Silik0nheh
07:25.09altbut I am totally in love with the MySQL support :)
07:25.18altdamn did that make my life 1000 times easier.
07:25.22[hC]Ive been meaning to do some mysql stuff
07:25.24Silik0nalt: please tell me you arent using mysql storage
07:25.28[hC]what do you use it for? just cdr? or realtime?
07:25.31altjust for config info.
07:25.34altfor voicemail
07:25.35Silik0nhah
07:25.36altworks good.
07:25.44altand I'm using MySQL for CDRs.
07:25.50Silik0nthe storage stuff is just nasty
07:25.53[hC]I was tempted to try realtime, but it looks too shoddy
07:25.57altI'm not up on PostgreSQL yet.
07:26.03altoh, you mean message store?
07:26.06Silik0nyeah
07:26.07[hC]like
07:26.10[hC]extensions.conf in mysql
07:26.29alt[hC]: I hate those schemes out there right now for that.
07:26.32[hC]yeah
07:26.32Silik0nrealtime is good for something but in general is just sucks
07:26.35[hC]they all look too hacky
07:26.35altI'm using a completely different one.
07:26.52TechnicGeekI'm having a problem with all extensions going right to voicemail or busy.  Does anyone have any suggestions?
07:26.55altmy sip table is one line per peer/user/friend
07:27.07[hC]I was hoping i would be able to do mysql for as much as possible so that adding new customers would be really simple to manage
07:27.12[hC]but for now ill stick with just cdr
07:27.28altTechnicGeek: you have a paste of the relevant portions of extensions.conf on the web somewhere?
07:27.38Silik0n[hc] you can do that, but several things you might want dont work with realtime...
07:27.53[hC]Silik0n: which is why i gave up before trying it
07:28.02[hC]after i read about the mwi and nat issues.
07:28.07alt[hC]: I'm using a perl script to regenerate sip.conf (actually, I have an include in sip.conf to sip_peers.conf which holds the "dynamic" stuff)
07:28.23altand then calling that from a PHP script which also reloads sip.
07:28.26Silik0n[hc] use a mysql for editing and updates and have a script that generates the sip user/peer/friend portion of sip.conf on a schedule or when you make changes
07:28.44Silik0nalt: thats what I do, but all in php
07:29.21[hC]yeah
07:29.22altSilik0n: I happened to write the script in perl first, so I left it there and just used "passthrough(script.perl)"
07:29.31Silik0nheh
07:29.34altnot the cleanest, but it works.
07:29.39TechnicGeekActually, the solution is coming back to me..  hrmm
07:29.54altand lets face it, clicking a few buttons is a lot easier than editing text files by hand all the time :P
07:30.08altTechnicGeek: are you using all SIP clients?
07:30.19Silik0nhell most of my configs are auto generated... only have to add a few pieces of data
07:30.47Silik0neven configs for the hardphones and atas are autogenerated and pushed out or pulled out at device boot
07:30.49altSilik0n: I generate sip_peers.conf and exten_vmexit.conf dynamically right now  and keep my voicemail users in MySQL.
07:30.52[hC]the model im going to be using (im starting an asterisk based telecom company soon here) will have the model of one master server, and each client site has an * box onsite. they simply have a dialplan for themselves and everything inside, and a catchall extension of sorts that hands off to the master pbx, where it has the "master" pstn dialplan
07:30.53TechnicGeekalt, yes
07:30.59altthe exten_vmexit.conf is for my "
07:31.01bintutTechnicGeek: "A" has an existing commercial voip infrastructure using SIP with several users. now, with "B" that is a new office, i want to setup asterisk to save money but still with an excellent persformance where A users and B users can do voip calls. the asterisk box for "B" can do these features for now: voip in/out calls, save discussions for future usage, play the recorded discussions using an audio player.
07:31.03altthe exten_vmexit.conf is for my "press 1" thingy
07:32.30[hC]is there a usable package in php or perl right now for management of sip.conf, etc? Or would it just be worth my time to write my own from scratch?
07:32.37bintutTechnicGeek: you told me earlier that we don't need to subscribe to a voip gateway service in the US in order to have a voip sip calls in both ends that is tunneled thru ipsec vpn
07:32.49znoGdo i *need* a soundcard for music on hold to work or is there some way to fake one? :)
07:33.04bintut[hC]: i just visited this site: http://amp.coalescentsystems.ca/
07:33.06altznoG: you need mpg123
07:33.10alt[hC]: I found that writing it from scratch was a lot easier than trying to fit my needs to a solution :P
07:33.15[hC]bintut: amp sucks :P
07:33.19znoGalt: i have mpg123, but chan_oss fails to load as /dev/dsp doesn't exist
07:33.28[hC]Its good to start with
07:33.33[hC]but once you want to be creative, amp has to go.
07:34.01altznoG: I have chan_oss and chan_alsa set to "noload" in modules.conf
07:34.03altyou don't need them.
07:34.07Silik0nznoG: mpg_123 doesnt need oss at all
07:34.13altbut you do need a real copy of mpg123 (not mpg321)
07:34.14Silik0nthats for other stuff
07:34.19bintut[hC]: well, i don't have any idea. i'm starting with asterisk and the entire voip ideas
07:34.29Silik0njust use make mpg123 from your asterisk src directory
07:34.30[hC]bintut: amp will be a good starting point then
07:34.59znoGalt, silik0n: thanks
07:35.20[hC]alt: I dont suppose you have examples of your extensions, etc up online? I get a kick out of seeing how people have done things. Ive exhausted almost all of the interesting stuff on voip-info.
07:35.22Silik0n'make && make mpg123 && make install' is your friend on new installs
07:35.34altSilik0n: I probably should've known about that trick. I ended up downloading it and compiling a static version :P
07:36.05Silik0nheh
07:36.20Silik0nthat trick has only been around a couple of months
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07:38.52BoRiStoo bad AMP didn't support Postgres database instead of Mysql :(
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07:39.15Silik0nboris you ho
07:39.18BoRiSSilikwhore!!!!!
07:39.20bintutTechnicGeek: when the asterisk box has been setup in office B, i want to start for 10 phones locally to send/receive voip calls to/from office A. can this be all solved with asterisk? what degium card can you recommend and other hardware/software i need for server B?
07:39.31bintutTechnicGeek: still there?
07:39.38BoRiSSilik0n: Long time no see....How have you been doing?
07:39.51Silik0nworking and moving and driving
07:40.00BoRiSmoving? Closer to work?
07:40.11altbintut: the digium card you need depends on the kind of circuits you are connecting too.
07:40.16Silik0nmoved the family down to HSV this weekend
07:40.47bintutalt: what circuits? sorry, i'm new to asterisk and voip. but i need to implement asterisk on office B to save money.
07:40.57altbintut: you have a PBX in office B?
07:41.10BoRiSSounds like a busy weekend
07:41.15Silik0nbintut: what kinda phones lines do you have coming in is what he was asking
07:41.29Silik0nyeah very busy
07:41.39Silik0nseeing i'min atl
07:42.13bintutalt: nope. office B is new and we need to setup voip solutions
07:42.25bintutSilik0n: isn't it the SIP?
07:42.41Silik0nbintut: well what are you doing for PSTN access?
07:42.50altwhat Silik0n said.
07:43.17bintut"A" has an existing commercial voip infrastructure using SIP with several users. now, with "B" that is a new office, i want to setup asterisk to save money but still with an excellent persformance where A users and B users can do voip calls. the asterisk box for "B" can do these features for now: voip in/out calls, save discussions for future usage, play the recorded discussions using an audio player. do i need to subscribe to a voip gateway service in t
07:43.17bintuthe US in order to have a voip sip calls in both ends that is tunneled thru ipsec vpn?
07:43.36cjkhi, when i call an agi script before every dial command to set some vars etc... works great.except when i do a transfert. the  channel name i get from the agi is the channe of the other party, any idea?
07:43.41bintutwhen the asterisk box has been setup in office B, i want to start for 10 phones locally to send/receive voip calls to/from office A. can this be all solved with asterisk? what degium card can you recommend and other hardware/software i need for server B?
07:44.12altbintut: you don't need any digium card for that if you are not going to access the PSTN from office 'B'.
07:44.24Silik0nbintut: if you are going to use SIP hardphones you dont need digium hardware at all unless you are going to use the conf bridge feature in asterisk
07:44.43altwhat you do need is a simple server (P4 2GHz+ 512MB of RAM should do the job nicely)
07:44.46Silik0nyou only need hardware cards for T1/E1 connections POTS lines
07:44.48altand SIP clients.
07:45.15Silik0nsip phone ssuch as Polycom IP300s & IP500s would be nice
07:45.28altI have the Cisco 7940s myself.
07:45.30[hC]alt: So you dont have any examples of the extensions.conf macros, etc you use online anywhere?
07:45.34altbut I hear the polycoms are better.
07:45.41Silik0nalt they are ;)
07:45.54Silik0nwe have both at our office and I prefer the polys
07:45.58alt[hC]: no. I don't keep those online. there's a lot of stuff there that is pretty specific to my setup.
07:46.05Silik0n(altho I'm sure someone will disagree with that)
07:46.12*** join/#asterisk gres (~gres@81.222.48.242)
07:46.27bintutalt: what is a PSTN? i don't want to purchase commercial products because as i've said we're looking for solutions in order to save money
07:46.32altSilik0n: thanks for the opinion. I'm pushing to get a couple of polys so we can give the 7940s back to our sister company.
07:46.33[hC]alt: figured as much
07:46.33[hC]:)
07:46.39Silik0n~pstn
07:46.40jbothmm... pstn is Public Switched Telephone Network
07:46.47altthanks jbot
07:46.56altbintut: Telephone lines.
07:47.01[hC]alt: we just decided to start trying to go all cisco, 7912, 7940/60 and 7970. would you really reccomend the poly's over them?
07:47.05[hC]Main reason was looks.
07:47.07alteither POTS, ISDN or T1/E1.
07:47.10Silik0nalt: tey are worth it
07:47.18Silik0nor J1
07:47.34alt[hC]: the 7970 isn't a SIP client, so I wouldn't touch it for the application
07:47.38altdunno about the 7912
07:47.41[hC]sccp support is no good?
07:47.44[hC]I love my 7960 so far
07:47.49[hC]i have had zero problems with it
07:48.02alt[hC]: is the 7960 SIP or SCCP?
07:48.03bintutSilik0n: i won't be using SIP hardphones.. i want to use analog phones which features redirecting to other numbers locally, hold, etc..
07:48.09altSCCP support is incomplete. better to use SIP
07:48.10Silik0n[hc] polys are very nice and have most excellent speaker fones (the 300 has a listen only speaker phone to) and they are all centrally managed off a tftp or ftp server
07:48.14[hC]i just got my hands on a 7912, it was the cheaper option.
07:48.26[hC]7960 is sip or sccp. you can pick.
07:48.27alt[hC]: is it SIP capable?
07:48.29[hC]7970 is sccp only.
07:48.31[hC]7912 does sip
07:48.38[hC]they all do tftp managemnt
07:48.39Silik0nthe 300s are ~100 each and the 500 are in the 150 to 175 range
07:48.48alt[hC]: yeah. I know. I've been doing Cisco CallManager since before Cisco purchased it from Selsius :P
07:49.03[hC]Ah :)
07:49.05altI still have some SP-12+ labelled "Selsius" :)
07:49.09Silik0nF@selsius heh
07:49.26alt2.4 was horribly
07:49.27alt2.4 was horrible
07:49.33altoh lord, it was horrible
07:49.35[hC]I decided it would be a good idea to try to stay uniform across the board, and so we went with cisco. alot of people frown on the look of the poly
07:49.41[hC]say that it looks too 'fischer price'
07:49.45[hC]the buttons, mainly
07:49.48[hC]personally i think its fine.
07:49.51altcalling up the client and saying, "I can change that option, but I have to restart the entire system to do it...."
07:50.01Silik0n[hc] you want fischer price order a bt101
07:50.08[hC]I've got 10.
07:50.08[hC]:)
07:50.24[hC]If the aastra firmware was better, its a nice looking phone.
07:50.30*** join/#asterisk znoG (~gs@200.115.216.109)
07:51.00[hC]but yeah we'll see with the 7912 and 7970 (apparently sip support is coming soon, ill hve to try my hand at sccp for now) but the 7940/7960's have been excellent.
07:51.01bintutalt: what about with PSTN? you mean, my asterisk box should have an identified telephone line number where people can send/receive call from an ordinary telephone line?
07:51.19cjkis there a way to block blind transfers
07:51.20cjk?
07:51.45altbintut: you should have at least one local phone line for phone access if the network is down and for 911.
07:51.52znoG# ztcfg
07:51.53znoGZT_CHANCONFIG failed on channel 1: No such device or address (6)
07:51.54znoGany ideas?
07:51.57TechnicGeekfor anyone having the problem of busy signal on all extensions perl -MCPAN -e "install Net::Telnet" fixed the problem
07:52.16altTechnicGeek: be sure to preface that with "on an AMP system" :P
07:52.21alt(you are using AMP, right?)
07:52.23Silik0nznog: thats like asking why you cant get laid
07:52.41altSilik0n: no suck "package"? ;-)
07:52.46Silik0n(or why i cant get laid)
07:52.47TechnicGeekyea
07:52.59znoGSilik0n: i'm not sure i see the relation there :)
07:53.12bintutalt: what local line? you mean, i need pbx?
07:53.18altno... local phone line
07:53.28altthe line you get from the phone company
07:53.29Silik0nznog: the number of issues that can cause the result is large and numerous
07:53.47*** join/#asterisk wasim (~wasim@203.81.203.118)
07:54.03znoGoh, right. well the card is in, compiled zaptel and installed it. configured zaptel.conf and zapata.conf. what else could it be?
07:54.15Silik0ndid you load the modules?
07:54.20znoGyup
07:54.30Silik0nwhat kinda card?
07:54.30znoGModule                  Size  Used by
07:54.30znoGzaptel                219396  0
07:54.36znoGZapata Telephony Interface Registered on major 196
07:54.40znoGa FXO card
07:54.49Silik0nso did you load teh wcfxo card?
07:55.06Silik0nthe driver for it
07:55.20znoGheh
07:55.28Silik0nzaptel is just a framework module each card has its own module for hardware specifics
07:56.27bintutanyone can guide me here?
07:56.44Silik0nbintut: you might want to seek professional help
07:57.23znoGthanks Silik0n :)
07:57.56altokay..,. bedtime for bonzo
07:57.58altttyl
07:58.32bintutSilik0n: thanks. but i can't afford you guys.. i'm currently in the 3rd world country..  :(
08:00.03Silik0nbintut: you'ld be surpised how many3rdworld countries the company i work for has set up stuff in
08:05.15tzafrirbintut, if that is your attitude, then be prepared to pay with labour/time instead of money ;-)
08:05.53tzafrirwrong question: "can anybody help me?"
08:06.52*** join/#asterisk RoyK (~roy@217.214.12.252)
08:08.12tzafrirright question: "I have asterisk version 1.0.9, compiled from source, and I try to connect Polymer Xyltoc phone to my * using the protocol SAP. However when I do XYZ I get ABC instead of DEF"
08:10.03tzafrirOh, and qhile I'm at it, can anybody help me connect a Pylimer Xyltoc phone to my Asterisk 1.0.9? ;-)
08:12.51bintuttzafrir: sorry... my primary problem is the entire idea of how voip works... asterisk is an OS application that i heard that is free and alternative solution to existing commercial voip/pbx products
08:14.10bintuttzafrir: but how these works and properly implement it is i don't have any clue
08:14.56bintutSilik0n: what i'm telling is we can't afford your professional service rate...  :(
08:16.08tzafrir~docs
08:16.09jbotDocumentation 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
08:19.49bintuttzafrir: thanks.
08:20.20RoyKGuten Morgen, asterisk nerds
08:20.28bintutVOIP connecting directly
08:20.28bintutIt is also possible to bypass a VOIP Service Provider and directly connect to another VOIP user. However, if the VOIP devices are behind NAT routers, there may be problems with approach.
08:20.53RoyKtzafrir: 1.0.9????
08:21.46bintutdoes it mean to use SIP? can i use analog phones for that?
08:22.02bintutdo i also need a digium card for that?
08:23.37RoyKyou can connect from user to user if you know the IP and if the user is not behind NAT
08:24.14RoyKyou'll need a digium or sangoma or junghanns or something card to connect to pstn
08:25.19bintutRoyK: you mean, i don't need to have asterisk even more with SIP? what phone shall i use?
08:25.54bintutgtg
08:25.56bintutthanks
08:25.57RoyKany sip phone
08:26.01RoyKor a SIP ATA
08:26.23cjkhi, there is a variable called ${BLINDTRANSFERT} can anyone confirm me if its really workin in the stable version?
08:29.16RoyKno :P
08:30.21RoyK"you get what you're paying for, and it's free"
08:33.59*** join/#asterisk hFritz (~thiragim@p8156-ipad411marunouchi.tokyo.ocn.ne.jp)
08:36.41wasimhi RoyK
08:38.32RoyKhi, wasim
08:40.54*** join/#asterisk shaonss (~shaon@61.68.14.250)
08:49.31RoyKmooooooorning has brooooooooooooookeeeeeeeeeeeen
08:51.03*** join/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com)
08:51.04newlDon't forget your 0xdead 0xbeef
08:52.29RoyKlater
08:52.32RoyKnot for breakfast
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09:46.52gordonjcp'sup
09:48.39morrismorning
09:48.44gordonjcphey morris
09:48.47morrishow u doing man
09:48.55gordonjcpnae bad, at work though
09:49.00morrisah
09:49.01morristhats poo
09:49.13gordonjcpsufficiently boring that my nokia ringtone to indications.conf tool is getting worked on
09:49.18morrislol
09:49.21morrisgood :)
09:49.31morrisyea its a nice little beast
09:49.31gordonjcpgot python, yeah?
09:49.49morrisnot bragging, coz it totaly skints me but. i got 4 dedis
09:49.53morrisone will have it
09:50.01gordonjcpwell I've got a "scrap" lappy that I inherited 'cos it was kensington locked to my desk
09:50.02morrisi got 2 with cpanel on
09:50.11morrislol @ kensington lock
09:50.16gordonjcpPII-266, with 32M of ram
09:50.19morrisah thats ok
09:50.23morriswhat u running on it
09:50.24gordonjcpit had XP, and 2000 advanced server on it
09:50.26gordonjcpit doesn't any more
09:50.31morrisyuck at windows on that thing
09:50.38morriswhat u running on it
09:50.42gordonjcpI kinda wish I'd kept it, if only for comedy value
09:50.45gordonjcpah, NetBSD 2.0
09:50.45morrislol
09:50.54gordonjcpit actually ran XP, slowly, but it ran
09:50.58morrisim not a lover of netbsd.. perhaps because i didnt give it a chance
09:51.14morristook me ages to get to grips with freebsd
09:51.19gordonjcphh
09:51.38morrisonly coz i got a dedi with it on
09:51.44morrisi was persuaded that i needed it
09:51.44morris;p
09:51.49morrisand to be fair its ok
09:51.54morrisjust im used to linux
09:51.59morrisanyways
09:52.05morriswhat did u do with that lappy man
09:52.31gordonjcppopped the hard disk, took it home, put Debian on
09:52.57gordonjcphated it, borrowed an external floppy from the test bench (OK, it's for a 4840 till, but it's the same connector)
09:53.03gordonjcpstuck NetBSD on
09:53.06gordonjcphappy lappy
09:53.11gordonjcpoh, and I put 64M in it
09:53.16gordonjcpthat helped a shitload
09:53.38morrisseen
09:53.41morrisand yea i bet
09:53.52*** part/#asterisk derka (~admin@crn93-1-82-237-178-115.fbx.proxad.net)
09:53.53morrisi guess u whent with netbsd since it will run on a toaster?
09:54.03gordonjcpthat and I run it on my servers
09:54.09gordonjcpand my other lappy
09:54.12morrisah ok
09:54.13gordonjcpone of them
09:54.14morrisso u do like it then
09:54.26morrishmm
09:54.36morrisi got net bsd for my dreamcast, im sure that doesnt count ;p
09:54.46gordonjcpyeah
09:55.02morrisi might do another search for the broadband adapter for it
09:55.11morrislast time i tried it was 80quid
09:55.13morrissecond hand
09:55.13morris;/
09:55.18gordonjcpI use netbsd 2.0 on a 1G microdrive in my Workpad Z50 (little dinky Windows CE subnote, running a MIPS 133MHz cpu) for wardriving
09:55.26gordonjcpgot the modem for it?
09:55.33morrisi have the modem yes
09:56.29gordonjcphook it up back-to-back with another modem, PP3 to bias it all up, and run ppp off another machine
09:56.35morrishttp://www.pdagold.com/hardware/detail.asp?d=10
09:56.37morrisis that the beasty
09:56.46morrishmm
09:56.52morrisi knew u was gonna suggest something like that ;p
09:56.56gordonjcpthe very same
09:57.07gordonjcpI need more memory
09:57.09morriswhat kinds of speeds do you think are possible
09:57.13morrissince ur bypassing the phone system
09:57.26gordonjcpshould go flat-out
09:57.33gordonjcpwhich iirc is 33.6kbps
09:57.39morrisi see
09:57.44morristhat is so horrible to even imagine
09:57.52gordonjcpI *think* it's a proper hardware modem
09:58.06gordonjcpso the DC shouldn't even feel it when you bring up the connection
09:58.11morrisah cool
09:58.18morristhe dc is a sexy machine
09:58.23morrisits so gay it died so fast
09:58.28gordonjcpdon't forget that your ping time will be fuck all
09:58.57morrisspose
09:59.03gordonjcpalthough you won't have a lot of bandwidth, you will have nearly no latency to local machines, so ssh for instance, will *feel* really fast
09:59.11morrisah yea
09:59.16morristhats a sesible point
09:59.23gordonjcpI mean, 9600 baud feels pretty quick when you're just typing stuff
09:59.25morrisnot sure what i would do with it once its running
09:59.33morrisi mean,
09:59.35gordonjcpbut a full page update takes just over a second
09:59.38morrisno change of external storage
09:59.47morrisgordonjcp, was you a bbs lover/
09:59.52gordonjcpayup
09:59.54morrisi certainly was
09:59.54morrishehe
09:59.59morrisi started late tho
10:00.00morriswhich sucked
10:00.02morrisyea me 2
10:00.06morrisi even started my oen
10:00.07morrisown
10:00.07morrisbut
10:00.10morristhis is where its funny
10:00.17morrisi wasnt allowed to take calls
10:00.17morrisso
10:00.26morrisi guess it was more of a play thing
10:00.28morris;p
10:00.30gordonjcpheh
10:00.43gordonjcpmy mate used to run Dark Star BBS
10:00.56morriscant say i knew it
10:00.59morrisbut i didnt know much
10:01.11morris1 in guildford which had 4 nodes
10:01.18morriscant remember what it was
10:01.21morrisand some school nearby
10:01.31morrishow old are you ?
10:01.53morrisbtw
10:02.04morristhat IBM WorkPad z50
10:02.09morrisseems pretty cool to me
10:02.21morrisu got x running on it ?
10:04.42gordonjcpyes, but it's slow
10:04.56morrishehe
10:05.06gordonjcponly 16M of memory, and the screen wants some of that...
10:05.12morrisargh
10:05.18gordonjcpI want to get the 32M upgrade
10:05.23gordonjcpa whole whopping 48M
10:05.26morrisroflmgao
10:05.27morriserm
10:05.29morristypo
10:05.35morrismy 486 had about that much ;p
10:05.36gordonjcpwhich - to be honest - is quite enough for what I want
10:05.38gordonjcpnamely
10:05.41gordonjcpwardriving...
10:05.44morrisi loved my 486
10:05.48morriswell it must be nice
10:05.50morristo have a portable
10:05.53morriswardriving device
10:05.57morriswith a keyboard
10:06.00gordonjcpI still have my 386sx that I first ran Linux on
10:06.01morrisand a decent operating system
10:06.10morrisah man
10:06.11morrishehe
10:06.43morrislol
10:06.46morrisi remember playing with those ;p
10:06.59morrisgetting my hands on the occasional "demo" to watch
10:07.08morriseven then they rocked
10:07.20morrisdo u remember some gay snowboarding avi ?
10:07.21gordonjcpheh
10:07.27morriscant remember where i got it from
10:07.30gordonjcphm, can't say I do
10:07.48morrisbut i had a copy on floppy
10:07.53morrisand always loved watching it
10:07.58morrismental
10:07.58morrislol
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10:08.08morrischildren eh
10:08.08morris;p
10:08.55gordonjcpavi on a floppy, lol
10:09.23gordonjcpman, I need to dig out some of my old demos
10:10.14gordonjcpI think the demo scene has gone off a bit, now that hard disks typically have faster CPUs and more memory than the computers the demos used to be written for
10:10.26morrislol
10:10.30gordonjcpah, nostalgia's not what it used to be
10:11.09morrisguess not
10:11.16*** join/#asterisk RoyK (~roy@host-89.homerun.telia.com)
10:11.25morrisi wouldnt mind dabling with some shit from back in the days
10:11.31morrisbut its just not practical lol
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10:25.35morrisgordonjcp, do you have any reverse engineering experience?
10:28.35RoyKmorris: heh. what are you trying to do? copy the win2k scheduler?
10:28.45gordonjcpmorris: uhm, a little
10:29.19morriswell, altho there are documents on it.. skype
10:29.28morrisim intersted in getting it to work with asterisk
10:29.34morrisi dont really use skype
10:29.44morrisbut i fancy doing something a bit more hardcore than i have done before
10:33.29*** join/#asterisk newbien (~e@116.242.33.65.cfl.res.rr.com)
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10:39.28morrishttp://hardware.slashdot.org/article.pl?sid=05/01/20/1653217&tid=215&tid=93&tid=185&tid=95
10:39.34morrisnice pdf on that subject
10:39.58newlheh BBS'..I still hang on a few C64 BBS' and hack on 6502 ML from time to time too. B)
10:40.02cjkhi, my asterisk is not using my voicemail users for ast_realtime... any idea?
10:41.25morrishehe
10:53.05cochimhhhh. probably i got some luck with someone using * + I4L now?
10:57.57znoGdoes anyone else notice extremely high CPU usage from the mpg123 processes?
10:59.35znoGmight try this format_mp3 addon
10:59.56blopmorris no news about a chan_skype ?:)
11:01.14morrislol
11:01.22morrisive switched to my turntables ;p
11:01.29morrisseems a bit hardcore for me
11:01.35morris;P
11:02.46gordonjcpheh
11:02.58gordonjcpcan we phone up and get a live stream?
11:03.40*** join/#asterisk map71 (~map@200.84.119.131)
11:03.45morrisrofl
11:03.57morrisah thats quite funny
11:04.04morrisi dont have a conference room setup on my asterisk
11:04.08morrisor u could
11:04.09morrishehe
11:04.40morrishowever if u do wanna hear and dont mind shoutcast ...
11:04.51morrishopefully it dont spoil my recording
11:06.20morrishttp://208.53.170.48:8000/ http://208.53.170.48:8000/listen.pls
11:06.30RoyKhmmmmmmmmm
11:06.53*** join/#asterisk Dossy (dossy@dossy.aolserver)
11:08.11morrishmma anyone know where my headphoens are
11:08.11morris;/
11:09.48morriserm
11:09.52morrishow on earth have i lost them
11:09.54morristhey are MASSIVE
11:09.55morrislol
11:11.45morrisu all dont care, but.. i found them.. plugged in
11:11.45morrisgr
11:20.54cochi*g* i luckily had the money for getting a DECT-based pro headset for cheap ;)
11:20.59cochii hope i never loose that o.O
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11:26.39gordonjcpmorris: release imminent
11:26.46gordonjcpget your python 2.4 at the ready
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11:28.56In-SideHello
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11:38.39morrisargh i fucking hate windows
11:38.49morrisoh no
11:38.52morriscorrection its my fault
11:38.54morrislack of hdd space
11:38.54morrisffs
11:39.15coppiceyou only lack space because its full of windows :-)
11:39.19morrishehe
11:39.48morrisi just did 30 minute mix (altho i sucked) and erm.. well.. its lost :(
11:40.11morrisah it uses so much energy mixing
11:40.12morris;/
11:40.16morrisim all sweaty lol
11:40.24mrtwisterhello. question for agi and mysql. if i do in perl/php agi something like this: 1.connection do db, 2. ivr for cuistomer 3. dialing 4. flush and clean db. and if customer will hangup at 2 or 3, will 4 (flush) will work? asking, because in exten=>xxx in extensions.conf i also can use mysql, but if customer hangups, connection to db leaves open, but exten - finished.
11:40.40morrismrtwister, agi and mysql are not present right now
11:40.57gordonjcpmorris: http://www.gjcp.net/~gordonjcp/NokiaRTTTL-0.3.tar.gz
11:41.02morristhatnks G
11:41.22morrisdoes it explain how to use the tones or whatever
11:41.29morrissince, i havent actually looked into it
11:41.31*** join/#asterisk cmk (~cmk_@p54A3D608.dip.t-dialin.net)
11:46.07gordonjcpyes, in fairly general terms
11:46.24gordonjcpdon't want *everyone* being able to do it
11:46.37Hineniwindows ... the kilty of our faults..
11:46.52gordonjcpthe *what*?
11:46.55Hineniguilty
11:46.56Hineni...
11:47.00gordonjcpheh
11:47.19coppiceI thought it was a scots reference :-)
11:47.45Hineniehehe
11:48.00Hinenimy great english...  ;)
11:48.32morrislol
11:48.40HineniI'm happy ... finnally my asterisk decide to pass trought the calls with g729
11:48.43HineniFINALLY!
11:48.56Hinenidon't ask me why he didn't befor
12:04.02mrtwisterHineni, hi, 729 its not problem, 723 is :(
12:08.00gordonjcpInfraRed: oi bitch
12:08.52Hineniya but in my case asterisk was refusing to troughtput it
12:08.57Hinenii have no ideia why
12:09.06Hineniall my devices supports it
12:11.02mrtwisterstrange, after i compile oh323, i able to use 729, converting protocols h323<->sip, converting codecs 729<-> any codec. only have problems with 723, it is passthru codec and in best case h323<--> sip with 723 giving me one-way audio
12:18.59*** join/#asterisk NewSole2 (dave@i216-58-44-245.avalonworks.net)
12:31.16gordonjcpmorris: how are you getting on?
12:31.36*** part/#asterisk io_error (~error@gw.ioerror.us)
12:31.46morrisim just chasing payments atm for my hosting shit
12:31.47morris;p
12:31.59morriswas u refering to your convert program?
12:32.11gordonjcpyeah
12:35.01morrisill look soonish
12:35.01morrisi get side tracked fast
12:35.01morris;/
12:35.05coppiceits good to be efficent at something
12:35.08gordonjcpheh
12:35.15morrislol
12:36.13morrisargh
12:36.24morrisi cant find in paypal where to disable encrpyted buttons
12:36.28morrisencrypted
12:39.11*** join/#asterisk folsson (~filip@h82n1fls32o985.telia.com)
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12:51.52jeobjeobjeobhey
13:02.13jeobjeobjeobhusslas
13:02.21jeobjeobjeobgimme a shoutout
13:29.11*** join/#asterisk delphiuk (~Owner@host217-43-97-191.range217-43.btcentralplus.com)
13:30.27delphiukhi, I have just compiled asterisk from cvs 1.07 stable, and ran make samples, but when I try and run asterisk, I get the following error:  [res_adsi.so]   == Parsing '/etc/asterisk/adsi.conf': Found
13:30.28delphiukIllegal instruction
13:30.28delphiukAny ideas?
13:31.17mrtwisterdelphiuk, suggest @home, good for start
13:31.40mrtwisterit will compile all you need, you should only add oh323
13:32.07mrtwisteralso try to remove adsi.conf
13:32.08*** join/#asterisk thomas_adam (~n6tadam@host217-43-99-169.range217-43.btcentralplus.com)
13:32.20mrtwistermv /etc/asterisk/adsi.* /root/
13:33.23delphiukmrtwister: the adsi file was the second file it got stuck on, the first of which was indications.conf, which we managed to get passed by removing all of the countries we didn't need
13:35.25thomas_adamA strace on 'asterisk -vv' reveals nothing out of the ordinary from what I can tell.  So I'm at a loss as to what's happening.
13:36.06*** join/#asterisk ReVoK (ReVoK@did75-5-82-224-60-46.fbx.proxad.net)
13:36.09ReVoKhi
13:41.08*** join/#asterisk bugsmoke (~lloydp@cpe-68-173-33-112.nyc.res.rr.com)
13:41.17*** join/#asterisk bintut (~bintut@202.128.40.243)
13:43.12bintutany digium card users here? what is the best digium card that can cater 4 or more telephone numbers outside and 10 or more local numbers?
13:58.22KlarAnyone here have a RapidBox(-R)?
14:00.14*** join/#asterisk newl (~newlook@203-59-153-204.dyn.iinet.net.au)
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14:14.35tzafrirKlar, me ;-)
14:17.01szw2001:)
14:17.08*** join/#asterisk carolttw (~wu@222.64.198.52)
14:17.11*** join/#asterisk Hitesh (~nospam@70.89.196.161)
14:17.19szw2001hai carol
14:17.29Hiteshhello all
14:17.38Hiteshany idea why skinny runs on 2000
14:19.44Hiteshdo i have to run skinny.conf
14:19.58szw2001nihao
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14:20.14*** mode/#asterisk [+o bkw_] by ChanServ
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14:22.59Klartzafrir: Do you have the -R version?
14:23.36*** join/#asterisk Vandien (~stephan@p50907590.dip.t-dialin.net)
14:33.20Hiteshwhat ports to open on firewall
14:33.22Hiteshfor asterisk
14:33.25tzafrirKlar, what do you mean?
14:33.32*** part/#asterisk carolttw (~wu@222.64.198.52)
14:34.25tzafriroops, forgot that RapidBox is not XorcomRapid
14:34.34tzafrirsorry
14:40.43*** part/#asterisk szw2001 (~vip@222.68.31.103)
14:40.45bintutany digium card users here? what is the best digium card that can cater 4 or more telephone numbers outside and 10 or more local numbers?
14:40.47Hiteshwhat firewall ports are important for asterisk ?
14:41.29InfraRedhttp://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules
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14:43.59cluecon[file]bintut: analog?
14:45.35Klartzafrir: RapidBox or RapidBox-R?
14:45.39bintutcluecon[file]: yes
14:45.49gordonjcpHitesh: depends what you want to do
14:46.03Klartzafrir: As depicted here: http://www.rapidvoip.com/products.html
14:46.12tzafrirKlar, as I said: sorry, can't help you. I thought you meant something else
14:46.24*** join/#asterisk orbi (~dantate@pcp08696782pcs.500ash01.tn.comcast.net)
14:46.34cluecon[file]bintut: well for one thing, it's not 'numbers' you have to worry about it - it's analog lines
14:46.51cluecon[file]bintut: how many lines would you have? because at that amount of lines (14) it would be cheaper to get a PRI most likely
14:47.00Klartzafrir: Ah, didn't notice your response, thanks anyway
14:47.08cluecon[file]bintut: but if you really want analog you'd probably want a T1 card and a channel bank
14:47.29KlarAnyone else have a RapidBox?
14:48.15InfraRedplenty of people do
14:48.16bintutcluecon[file]: i want to have 4 telephone lines subscribe by our local telephone company so that i would have 4 outgoing calls at the same time from any of my 10 local telephones
14:48.41bintutcluecon[file]: you mean the digium Wildcard TE410P?
14:48.47InfraRedyou want 4 port fxo card
14:48.56cluecon[file]InfraRed: wait, don't answer yet
14:48.57InfraRedfoir the teleco side
14:49.08cluecon[file]bintut: you want to keep your analog telephones?
14:49.10InfraRedtoo late:)
14:49.47bintutcluecon[file]: for now, yes. is there any new type of telephones something like a digital one?
14:50.00cluecon[file]bintut: uh, VoIP phones :)
14:50.18cluecon[file]if you wanted to keep your analog telephones, a T1 card with a channel bank with FXS and FXO capacity would be most cost effective
14:50.32cluecon[file]if you wanted to get new VoIP phones and keep your analog lines from the telco, a TDM400 with FXO modules
14:51.12bintutcluecon[file]: isn't it the voip phones are expensive? what do you call the "big" telephones which has so many buttons basically used by hotel/company call operators
14:51.42cluecon[file]bintut: you can't use those with asterisk, they're designed for the system that they're used with... and they are expensive too
14:51.43bintut?
14:51.46*** part/#asterisk thomas_adam (~n6tadam@host217-43-99-169.range217-43.btcentralplus.com)
14:52.04InfraRedpersonally i found fxs adapters and cheapo phones work out cheaper per user
14:52.14InfraRedand if they break the phone you can reaplce it cheaply
14:52.43InfraRedbudgtone feel too cheap and plasticy
14:52.51cluecon[file]that's because it's a barbietone
14:52.54bintutcluecon[file]: so what do you recommend for a setup that can accept voip calls from a remote location which has an existing commercial solutions but in my place we'll be planning to implement asterisk with 10 or more local lines?
14:52.59cochibintut: you're talking of system phones. they rely on proprietary technology which's missing in foreign vendor's hard/software. thus you can't use these big gazillion-button phones there ;)
14:53.12cochijust as a late addition ;)
14:54.40bintutguys, i heard that asterisk hardwares from digium are pci boards only which can be installed in an x86 linux box...
14:54.54bintutand i want to make use of it...
14:54.59cluecon[file]correct... you can install them in powerpc Linux systems too
14:55.18cluecon[file]we
14:55.20cluecon[file]gah
14:57.56bintutwhat i need for now is i believe is too simple for you... i only want to have a linux box with asterisk that can extend from 10 or more local lines which can be called first using a single telephone number and maybe out of these 10 or more local telephones, 4 of which can call outside at the same time so that these 10 or more telephone users will not be waiting for so long when a user will be finish his/her call outside if i only have a single telephone
14:57.57bintutnumber outside subscribed to a telco
14:59.40cluecon[file]TDM card with four FXO modules, and 5 Sipura SPA-2000s if you want... or another 3 TDM cards with FXS modules... or a T1 card with channel bank...
14:59.43cluecon[file]see how many ways you can do that?
15:00.06bintutcurrently, what i'm thinking is to use analog phones to the local telephone lines for our users inside but i heard that there is a so-called "soft phones" using skype. i don't know how it works for now..
15:00.39bintutcluecon[file]: lemme check those cards... are those cards from digium?
15:00.41cluecon[file]yes
15:01.03cluecon[file]Sipura SPA-2000 is an ATA adapter that gives you 2 phone lines, connects to asterisk via VoIP
15:02.30bkw_OMG BECKY
15:02.32bkw_ITS FILE
15:02.39cluecon[file]OMG BECKY
15:02.40cluecon[file]ITS BRIAN
15:02.53bkw_haha
15:03.22cluecon[file]bkw_: I was looking at Powerbooks some more :(
15:03.42bkw_you remember what da boss said right?
15:03.45bkw_we gonna hold him to it
15:03.47cluecon[file]yes
15:03.59cluecon[file];)
15:04.01bkw_greg likes his mac so far
15:04.04bkw_he has to get used to it
15:04.10bintutcluecon[file]: where can i find the Sipura SPA-2000?
15:04.27cluecon[file]bintut: dozens of online shopping sites for VoIP, http://www.sipura.com/ ironically for manufacturer's site :P
15:07.58bintutcluecon[file]: so these spa products are voip phones? isn't it expensive? isn't this is like the voip phones from cisco? any cheap alternative for this solutions?
15:09.39bintutcluecon[file]: is the Sipura SPA-2100 Analog Telephone Adapter supports asterisk?
15:13.48Klarwho
15:13.48Hiteshwhere are VM passwords saved
15:14.32*** join/#asterisk morris (~turntabli@ACD6B1A8.ipt.aol.com)
15:15.00gordonjcpHitesh: http://www.google.co.uk/search?hl=en&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial_s&q=asterisk+voicemail+passwords&btnG=Search&meta=
15:15.13morrisgordonjcp, i bet u use firefox
15:15.19tzangerhmm
15:15.27tzangerwhy is WAV file format voicemail so ... crappy
15:15.33tzangerI would have thought it would have been the best
15:15.46Klartzanger: Doesn't mean anything that it's in WAV format
15:16.06Klartzanger: I mean that the quality is independent
15:16.16tzangerKlar: I just meant as opposed to a lossy codec like gsm would noticeably sound worse
15:16.33file[mac]bintut: it's a VoIP ATA adapter, and it costs about $80 USD... I think, haven't looked in awhile... it gives you two phone lines that connect to your asterisk box... and yes asterisk supports it, why would I suggest it if it didn't?
15:16.52gordonjcpmorris: erm, yup
15:16.57morrisgood lad
15:17.03morrisalso the url said so
15:17.04morrisbut yea
15:17.04*** join/#asterisk szw2001 (~vip@222.68.31.103)
15:17.05morris;p
15:18.10Klartzanger: A lossy codec-encoded file would sound better if the file were the same size as the WAV file
15:19.25Hiteshhow can i play musiconhold on an extension
15:19.56tzangerKlar: yeah I changed to wav instead of wav49 or whatever and it's better
15:20.20blitzrageHitesh: MusicOnHold() or the 'm' flag in Dial()
15:20.44drumkillaor the m option to waitexten!
15:20.53blitzragedrumkilla: what the hell do YOU know
15:21.10drumkillaha, i've got nothin' !
15:21.12cochihehe seems to be Q&A time ;) anybody ever ran across undefined symbols when trying to use chan_modem_i4l ?
15:21.27*** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net)
15:21.29*** join/#asterisk clive- (~pirch@rrba-146-95-145.telkomadsl.co.za)
15:21.32file[mac]cochi: load chan_modem first
15:21.35cochidid so
15:21.35blitzragedrumkilla: thats scary then considering what you control :)
15:21.57cochiasterisk -cvvv displays chan_modem getting loaded and this then tries to load the i4l one. symbol missing..
15:22.18clive-anyone got chan_capi working semi stabily?
15:22.48drumkillablitzrage: muahahaha!
15:23.21cochimay i paste the 3 lines from startup here?
15:23.38blitzragegod I love being at home - I just got delivered a chicken wrap
15:23.48drumkillai'm jealous
15:23.56blitzrageI'm jealous of myself :D
15:24.05bintutfile[mac]: cluecon[file] TDM card with four FXO modules, and 5 Sipura SPA-2000s if you want... or another 3 TDM cards with FXS modules... or a T1 card with channel bank...
15:24.23cochihttp://pastebin.com/291700   <- my error
15:24.37file[mac]bintut: there's a few ways you can do it, like I said
15:25.01bintutfile[mac] and cluecon[file]: if i'll choose the Digium Wildcard TE410P, how does it work?
15:25.35tzangerok
15:25.36tzanger<PROTECTED>
15:25.40tzangerwhat the fuck does that mean?
15:25.47file[mac]bintut: connects to a channel bank that, depending on the configuration of said channel bank, will give you a set amount of phone lines for your regular phones, and connections for your telco lines
15:25.51cochithat you're poor now ;))
15:25.52file[mac]tzanger: advice of charge
15:25.53tzangerAOC-E = Advice of Charge --- E?
15:26.00bintutfile[mac] and cluecon[file]: it has 4 ports based on digium's site at http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P
15:26.09tzangerand that sounds like a lot of units
15:26.11file[mac]bintut: there's a single one too you know...
15:26.18drumkillatzanger: you better hang up quick!
15:26.24tzangerdrumkilla: :-)
15:26.39file[mac]bintut: you really should go read and learn, I'm not in the mood to discuss the entire setup of your phone system and design it for you
15:26.40bintutfile[mac]: isn't digium has a limited info about their products?
15:26.57file[mac]what do you mean limited info?
15:26.59drumkillabintut: what do you need to know?
15:27.14blitzrageeverything? :)
15:27.23file[mac]aha! I knew you were going to say that
15:27.27blitzragelol
15:27.30bintutfile[mac]: it's ok. thanks. i just can't get how my original needs be configured based on digium's available hardware for asterisk
15:27.58drumkillabintut: in short, what are your needs?
15:28.15cluecon[file]drumkilla: he has 4 telco lines, 10 analog phones - wants to connect them to asterisk
15:28.19Hiteshso what should it say in extensions.conf
15:28.26bintutbintut what i need for now is i believe is too simple for you... i only want to have a linux box with asterisk that can extend from 10 or more local lines which can be called first using a single telephone number and maybe out of these 10 or more local telephones, 4 of which can call outside at the same time so that these 10 or more telephone users will not be waiting for so long when a user will be finish his/her call outside if i only have a single tel
15:28.26bintutephone number outside subscribed to a telco currently, what i'm thinking is to use analog phones to the local telephone lines for our users inside but i heard that there is a so-called "soft phones" using skype. i don't know how it works for now..
15:28.33Hiteshfor adding music on hold directly to an extension
15:28.47drumkillabintut: get a T1 card and a channel bank, next!
15:29.37drumkillaHitesh: MusicOnHold()
15:30.33bintutdrumkilla: what's a channel bank? if i'll get the Wildcard TE410P, are those 4 ports on this site ==> http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P are all an RJ11s or not? how is the "channel banks" work? how can i support the number of local telephone lines and 4 outgoing lines subscribed to a telco?
15:31.06tzangerinteresting
15:31.18tzangerif I dial(Zap/g1/somenumber) it works
15:31.28bintuti apologize, i'm really newbie to this technology..
15:31.30tzangerIf I dial(Zap/g1/somenumber/someothernumber) I get that nasty AOC-E
15:31.37drumkillabintut: they are RJ45, and they connect to T1's ... a channel bank can hook up a bunch of phones and phone lines, then you hook a T1 from the channel bank to the card
15:32.30Hiteshis this correct exten => 2501,2,MusicOnHold(s,6)
15:34.01drumkillaHitesh: no.
15:34.05bintutdrumkilla: are these "channel banks" all PCI cards which i can install in one of my PCI slots of my linux box with asterisk?
15:34.11*** join/#asterisk NewSole2 (dave@i216-58-44-245.avalonworks.net)
15:34.18drumkillabintut: no, google for channel bank
15:34.25Hiteshexten => 2501,1,MusicOnHold()
15:34.33drumkillaHitesh: yes
15:34.36Hiteshthanks
15:34.51cypromisbintut: a channel bank will multiplex your T1 from a T1 card into 24 analog circuit
15:34.55cypromisit's an external box
15:36.18Hiteshany idea how i can connect weather to an extension
15:36.27Hitesha weather channel i mean
15:36.38blitzrageHitesh: look for, or build, a script - there are a few around
15:36.45*** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk)
15:36.55blitzrageI know of a book that has an example of one, but its not out yet ;)
15:37.02Hiteshdamn
15:37.05Hiteshblitz whats the book
15:37.24blitzrageHitesh: just search google, there are a few scripts around
15:37.36blitzrageHitesh: book isn't out yet
15:37.48bintutcypromis: what product will you suggest that is cheap, functional, scalable and will perfectly work with asterisk and a Wildcard TE410P?
15:37.49cluecon[file]blitzrage: MOOSE!
15:38.10*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
15:38.12blitzragecluecon[file]: moose are overrated
15:38.27cypromisbintut: Carrier Access Adit 600 is nice, Adtran channel banks are nice and there are a couple of others that work nicely too
15:38.28cluecon[file]blitzrage: noooooo
15:38.29Hiteshwhats text2wave
15:38.33Hiteshand where can i get text2wave
15:38.38blitzrageHitesh: exactly what it sounds like
15:38.45blitzrageHitesh: google knows
15:41.17cluecon[file]blitzrage: yay Leif!
15:42.28bintutcypromis: is the Carrier Access Adit 600 cheap? can you give me a rough estimate of that product? what do you currently use and how much was it if you don't mind?
15:43.32Hiteshis festival easy to configure
15:44.40drumkillabintut: should he hold your hand, too?
15:45.21*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
15:45.33gordonjcpHitesh: there's a great site with all this information on it
15:45.56Hiteshwhich one
15:45.59gordonjcpHitesh: have a look at www.google.co.uk
15:46.20Hiteshfestvox.org
15:46.41gordonjcpin all seriousness, you will find a lot of stuff on voip-info.org
15:48.32*** join/#asterisk bewest (~bewest@h48.204.255.206.cable.srcy.cablelynx.com)
15:48.40bintutdrumkilla: what do you mean?
15:49.28drumkillajoke
15:51.28cochio.m.g
15:51.29blitzrage~docs
15:51.30jbotDocumentation 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
15:51.34cochian sql error on pastebin
15:51.35Hiteshcan you install a fedora rpm on redhat 9
15:51.41blitzrageif you haven't read those sites, do so now, then come back and ask your questions
15:51.43cochithe worst kind. usable for sql injections
15:52.03cluecon[file]SQL injections eh? drumkilla knows those well ;)
15:52.07cochiwell. at least high risk ;)
15:52.12cochihehe me too. but from passive point
15:52.26cochibut basic rule "never print an erraneous sql statement to the user"
15:52.45cochipastebin just did ;)
15:52.49cluecon[file]haha
15:53.10blitzragejbot: no, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org, or http://dev.asteriskdocs.org
15:53.26bewestI read all those docs (several times).  I've got two servers up and running... I just realized I still don't understand anything
15:53.43blitzragegrrr...
15:53.55blitzrageI don't want #asterisk-doc in there for documentation - its not a help channel
15:54.03derkabewest: wisdom is when you realize you know nothing;)
15:54.21cluecon[file]hrm is Star Trek Insurrection on today... maybe/
15:54.59blitzrageam I doing something wrong to update jbot?  Thought I was doing it right :)
15:55.29blitzrageoh well - going back to writing docs
15:55.49cluecon[file]not done yet?
15:56.18cluecon[file]bad blitzrage bad
15:56.39cluecon[file]oh wow I'm psychic
15:56.46cluecon[file]Star Trek Insurrection is on today at 3
15:56.49cluecon[file]until 5:30
15:56.54Sedoroxchannel?
15:56.58Sedoroxor to cluecon?
15:57.00bintutoh no! Wildcard TE410P cost sooo much!  :(
15:57.00Sedoroxat*
15:57.20drumkillabintut: you can get away with the TE110P in your case
15:57.23cluecon[file]bintut: Then get a TE110P
15:57.27cluecon[file]ooh dejavu!
15:58.03bintutwhat's that? is that a digium product? any link please?
15:58.12drumkilladigium.com
15:58.14blitzragebintut: its on the digium.com site
15:58.17bintutok
15:59.49cluecon[file]blitzrage: docs.
16:02.08bintutdrumkilla, cluecon[file] and blitzrage: how this card work? it's just as simple as putting it to my linux asterisk box, make a crossover utp connection to a "channel bank" and from there where the local analog telephone lines or skype are connected and my 4 telephones lines from a telco?
16:02.46cluecon[file]you can't use Skype with asterisk
16:02.59bintutdrumkilla, cluecon[file] and blitzrage: is there any solution where i can all put them on a pci slots of my linux asterisk box so that i could only have a single machine to manage?
16:03.15bintutcluecon[file]: ok.. with analog phones only
16:03.53drumkillabintut: you can buy a bunch of TDM400P's
16:04.09bintutguys, as much as possible i don't have any external devices to use to support voip/pbx asterisk
16:04.15drumkillabintut: a channel bank is easy to manage
16:04.19drumkillaand makes sense for your case.
16:05.16bintutdrumkilla: oh, really? sorry, this is my first time to know and tackle about voip/pbx
16:05.27Sedoroxcluecon[file]: what station is star trek on?
16:05.36cluecon[file]Space: The Imagination Station
16:05.38cypromispertrol station
16:05.54Sedoroxhmmm
16:05.54cluecon[file]'tis a Canadian station
16:05.57cjkanyone here using ast_realtime?
16:06.03Sedoroxoh :/
16:06.28Sedoroxnow I wanna watch it.. guess I should put the DVD in
16:06.29Sedoroxlol
16:06.43bintutdrumkilla, cluecon[file] and blitzrage: so what will be your practical digium card you can suggest to me based on your personal experiences and what i want to achieve?
16:07.42drumkillabintut: I think we've said the T1 card 2398472938742394 times
16:07.42bintutdrumkilla, cluecon[file] and blitzrage: and the cheapest, scalable, practical and works perfectly with asterisk "channel bank"?
16:08.21cjkis it possible in ast_realtime to use the same table for iax and sip peers?
16:08.32drumkillacjk: sure
16:08.50cypromisyou can use on table for e erything and work with views
16:09.13bintutdrumkilla: i mean, which of the two: Wildcard TE110P or Wildcard TE410P? do i need a the 4 ports or a single port will do based on my needs?
16:09.26cluecon[file]a single port will work fine
16:09.54cjkdrumkilla: hmm, sip realtime and voicemail realtime is working greate, but as soon as i activate it for iax, asterisk crashes
16:10.34blitzragethis is why I'm writing docs so we don't have to answer these same questions over and over...
16:11.00blitzragenot that I don't like helping, because I do, but it can be frustrating
16:11.09blitzragecluecon[file]: me either - we got an ISBN number today ;)
16:11.32cluecon[file]blitzrage: yayyyyyyy
16:11.36*** join/#asterisk zyke (~zakforeve@84.45.132.117)
16:12.03bintutcluecon[file]: ok. i'll consider the single port. and i just want to reiterate that the "channel bank" is an external device where i can patch all the telephone lines from my telco and local telephone lines inside for my users, right?
16:12.15cluecon[file]bintut: yes.
16:13.19bintutcluecon[file]: what are other "channel banks" you can recommend beside the Carrier Access Adit 600 that works perfectly with asterisk?
16:14.05bintutcluecon[file]: based on what you currently use and basically cheap and scalable?
16:14.07drumkillaadtran TA6XX
16:14.09blitzrageAdtran 750
16:14.17drumkillaha
16:14.23blitzragedrumkilla: ;)
16:14.37blitzrageI like the adtrans - had good luck with them
16:14.38cluecon[file]anything adtran!
16:14.46blitzragevery cheap on eBay too
16:15.19bintutAdtran brand for the channel banks? lemme ask bestfriend google for that one.. :)
16:17.50*** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net)
16:18.43bintutbtw guys, before i forget.. i wanna thank you for your patience and help...
16:19.16*** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net)
16:20.26bintutwhile waiting for the webpage of adtran to load from my current 56k dialup connection, does asterisk capable of recording the conversations anytime both parties wants to record it and can be easily played later on?
16:25.03blitzrageyep
16:25.05blopis it better using SMS(..|a) or SMS(..,a) ?
16:25.13blitzrageblop: doesn't matter
16:25.17blitzrageblop: its parsed the same
16:25.21blopallright :)
16:25.23blopthx
16:25.24blitzrageI prefer comma's
16:25.35blopi agree :)
16:25.41bintutok.. i was able to load the adtran site..
16:26.47bintutis "Wildcard TE110P + Total Access 750 Chassis w/ BCU L1" a good combination for an asterisk voip/pbx setup?
16:27.36bintuthttp://www.adtran.com/adtranpx/Rooms/DisplayPages/LayoutInitial?ProductCategory=com.webridge.entity.Entity%5BOID%5B985EED955FA14843B01EA3181528C2E5%5D%5D&Product=com.webridge.entity.Entity%5BOID%5BDB4279CCC6F8D611A77D00D0B72032D8%5D%5D&Container=com.webridge.entity.Entity%5BOID%5BF5C7CEE8D8313E49B4D65B30BDDF4734%5D%5D
16:27.37bintuthttp://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE110P
16:30.03InfraReddude
16:30.08InfraRedtinyurl.com
16:30.13InfraReduse that next time :)
16:39.16bintuthttp://tinyurl.com/e4jb8
16:39.49morrislol
16:40.07bintuthttp://tinyurl.com/9ocgr
16:40.12morrisare u guys familiar with goatse?
16:40.45cypromiswhy ? want to tell us it's you ?
16:40.56InfraRedhe's the son of goatse
16:41.02bewestI'm using a router that doesn't support QoS.  if I specify tos attributes in my configuration, will it be totally useless, or might my ISP give priority to those packets?
16:41.11blitzragebewest: ask your ISP
16:41.16morriscypromis, thats one way to get instant recognition in a foriegn chat room
16:41.18blitzragewe have no way of knowing
16:41.24bewestok
16:41.26morrishttp://tinyurl.com/c5ke5 <-- for those who have not met the goatse
16:41.27morris;/
16:41.33bewestthat's what I meant
16:41.47nestArlol
16:42.35morrislol
16:42.36nestArtinyurl is bad stuff
16:43.13morrisargh
16:43.14morrispot noodle
16:43.16morrisits too hot
16:43.17morrisargh
16:45.11*** join/#asterisk iq (~iq@63-230-44-221.omah.qwest.net)
16:53.14blopmm, i'm dialing 171700, i got a "exten => _1717X0" and *after* a "exten => _1X." and asterisk take the second one, any idea how i must proceed ? :)
16:53.41bkw_exten => _1717X0,
16:53.53bkw_exten => _1XXXXXX.
16:53.57bkw_or something like that
16:54.02bkw_watch out for . if you dont know how to use it
16:54.05bkw_its very greedy
16:54.12blopmm, i saw that
16:54.25blopbut, i must take the calls to "112" too
16:54.31cypromismorn mr. west
16:54.34blopbut 1XXXXXX wont, right?
16:54.39cypromiscorrect
16:54.46cypromissince 112 is shorter
16:54.49blop:(
16:54.56cypromismake a third extension
16:54.59cypromisjust for 112 and 110
16:55.41blopi cant list all the numbers smaller than that pattern :(
16:56.09blopwhy is . catching extension declared before itself ?
16:56.14blopits weird
16:56.37PatrickDKblob, cause you didn't follow extention logic correctly
16:56.43PatrickDKit doesn't matter what order they are in
16:56.47PatrickDKjust how many digits match
16:56.53PatrickDKit takes the first match
16:57.04PatrickDKand 1 matches before 17...
16:57.10blopok, so the order in the .conf doesnt matter
16:57.15PatrickDKnope
16:57.19blop:(
16:57.40PatrickDKmove your _1XXXXXX. into a different [context] and include it into your current context
16:57.45bintutgtg now.. good night.. thanks for all the help and your patience...  :)
16:57.45PatrickDKthat will fix the matching order you want
16:58.04blopooh :) i'll give a try
16:58.07bintutMon May 30 00:58:11 UTC 2005
16:58.31bewestwhat's the general approach to make a whole bunch of phones ring for one incoming call?
16:58.31*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
16:58.55PatrickDKbewest, SIP/phone1&sip/phone2&sip/phone3
16:59.09bewestPatrickDK, you must explicitly list all of them, eh?
16:59.20PatrickDKhmm, ya, how else would it know?
16:59.22bewestand they must all use the same technology?
16:59.25PatrickDKno
16:59.28bewestdunno, that's why I'm asking
16:59.32PatrickDKI personally use LOCAL/...
16:59.38bewestoh nm bout the technology
16:59.39bewestI see
17:00.00PatrickDKyou can use local, to setup groups if you want to do that
17:00.05bewestoh yeah?
17:00.08PatrickDKand then you just have to call local
17:00.13bewestI will look up local
17:00.19PatrickDKlocal is alot of fun
17:00.31PatrickDKbut will screw you if you don't pay very close attention
17:00.37bewestheh
17:01.13blopthanks PatrickDK :)
17:05.00NivexAm I reading this right?  An IAX enabled ATA?  -A PREROUTING -i eth1 -p udp -m udp --dport 5198 -j DNAT --to-destination 172.31
17:05.04Nivexcrap
17:05.05Nivexstupid paste
17:05.10Nivexhttps://www.virbiage.com/products.php
17:05.12bewestwhy does the SIP register command take an extension after it while the IAX register command does not?
17:05.29PatrickDK5198?
17:05.40PatrickDKisn't it 4569?
17:05.47cluecon[file]omg...
17:05.51cluecon[file]AH
17:05.53NivexPatrickDK: it's for another piece of software
17:05.54cluecon[file]preorder
17:06.35cluecon[file]it used to be a phone, now it's an ATA... I wonder if they'll actually ever release anything
17:15.41wasimcluecon[file]: who, us?
17:16.03wasimcluecon[file]: oh, virbiage
17:16.21blitzragewasim: heh :)
17:18.11cluecon[file]wasim: and you :P
17:18.27cluecon[file]when I see the virbiage one, I will be happy for it has G726
17:18.42cluecon[file]and it looks cute
17:18.42wasimwe do g726
17:18.47wasimi look cute
17:18.53cluecon[file]lies, all lies
17:19.43*** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net)
17:20.28blitzragelol
17:20.48blitzragewasim: have you settled on a plastic now?
17:21.06wasimblitzrage: yeah, that we did a while back, its the firmware thats taking a while to settle :(
17:21.43blitzragewasim: yah I imagine so. I think I've seen the plastic, but I wasn't sure if you'd totally settled yet. How close do you think you are to a beta release?
17:22.01wasimblitzrage: beta's been released for a while now, we're doing a source release next
17:22.06blitzragenice
17:22.09cluecon[file]source release? O.O
17:22.27wasimblitzrage: one thing we learnt is that its bloody difficult to mature a hw product without a controlled testing environment
17:23.13clive-virbiage phone is hapenning?
17:23.22clive-wasim hows your iax2 hardware comming?
17:23.30cluecon[file]it's no longer a phone, they've turned it into an ATA now
17:23.37clive-so far the pa168 phones have iax2
17:23.54blitzragewasim: yah, I imagine it'd be a bit of a bitch
17:23.56wasimclive-: slowly, and after the pa168 we can't be price competitive, so adopting a different strategy
17:24.09wasimmy boq cost at 5k units is $60
17:24.13clive-lol...clue what did they turn into a ATA...air to ATA..:)
17:24.17wasimpa168 phones retail for half that
17:24.18blitzragewasim: I'm having the same sort of problems trying to develop and write docs based on my PBX system at home - my room mates hate me :)
17:24.22cluecon[file]clive-: apparently
17:24.26clive-wasim the chinese slay all of us in price..
17:24.37clive-I see they made the pa168 chip an open source project
17:24.42clive-lol
17:24.48wasimso, we're thinking either niche product, or other stuff like encryption, etc
17:25.01blitzrageencryption would be very cool
17:25.03clive-clue, soon we will see they took a chinese pa168 and dressed it up
17:25.13clive-wasim waht about your 4 port ata
17:25.30wasimthe idea is to open the framework to edu institutes where they can use it to teach dsp, networking, voip etc, and also help mature the product AND develop further applications
17:25.33*** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net)
17:25.45Hiteshwhat should i do if i don't want asterisk to pick calls from my outside line
17:25.53wasimclive-: thats also on hold, we've got little money, and cac announced the adit 3104 which is a 24 port sip fxs gateway for $1k
17:26.15wasimHitesh: don't have an Answer in the context you define in zapata.conf
17:26.28clive-and fxo ?
17:26.42wasimclive-: i don't think they're doing an fxo as yet
17:27.05cluecon[file]http://phlog.net/entry/228458&page=30 actual pic of the Virbiage ATA
17:27.30cluecon[file]looks like a toy
17:31.04Hiteshthe pbx still picks up after i changed the context to default.donotanswer
17:31.12Hiteshi don't want it to answer calls from outside
17:31.27Hiteshbut i should be able to make calls that go outside
17:32.11clive-their website is busted when I click on "shop"
17:37.42*** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca)
17:38.32HiteshMay 29 12:37:37 WARNING[1217669936]: app_voicemail.c:1726 leave_voicemail: No entry in voicemail config file for ''
17:38.37Hiteshwhy am i getting that message
17:45.10blitzrageHitesh: what do you think it means?
17:45.24blitzrageHitesh: No entry in voicemail config file
17:45.35blitzragefor '' looks like you've provided a null value
17:46.18cluecon[file]blitzrage is soooooooo smart
17:46.39blitzragecluecon[file]: no, I just know how to think things out logically
17:46.51cluecon[file]:)
17:47.04InfraRedlogic sucks
17:47.42blitzragelogic has done me very well - I don't have a great memory, logic is all I've got :)
17:49.25*** join/#asterisk L|NUX (~linux@202.5.145.54)
17:50.07cluecon[file]blitzrage is sexy fyi all
17:52.30newmedian% blitzrage is too sexy for da logic, too sexy for da logic, too sexy by file %
17:52.40blitzragelol
17:52.43blitzrageI don't think anyone cares
17:53.17newmedianWell, it's a lazy Sunday (EST).
17:53.30blitzragetrue - too bad I have so much damn work to do
17:53.31blitzrageEDT here
17:53.35drumkillahas anyone by chance used mark's nbsd?
17:54.20newmedianYes, meant EDT. Damned TZ.  (Hey, he's got his GMT stuck in my UTC; Hey, you've got your UTC stuck in my GMT; They're too great timezones in one!)
17:54.27newmediantwo
17:57.17*** join/#asterisk da_monumental_1 (~da_monume@66.57.197.72)
18:01.14Hiteshhow can i add a wait before answering
18:02.36Hiteshthe pbx is still picking up
18:04.17wasimHitesh: Wait()
18:04.25wasimHitesh: but do you ahve immediate=yes?
18:05.07Hiteshlet me check
18:05.21Hiteshimmediate=no
18:05.30Hiteshin zapata.conf
18:05.37Hiteshi don't want the pbx to pickup calls from outside
18:05.52Hiteshor atleast i want the pbx to wait for 15 seconds until it decides to pickup
18:06.00Hiteshgiving me a chance to pick the closest phone
18:06.05wasims,1,Wait(15)
18:06.31L|NUXany Dundi peer ?
18:06.45Hiteshk let me try
18:07.43L|NUXwasim : salam
18:07.47L|NUXwasim : long time no see
18:13.36blitzrageFYI: please don't private message people unless you ask them first
18:14.21wasimmorning kram
18:17.32*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
18:19.03Hiteshwasim: thanks. that worked
18:19.30Hiteshpbx picks up after 13 seconds before it goes to phone companies VM.
18:20.19ariel_good afternoon all
18:20.56Sato1hi ariel
18:21.43ariel_Sato1, hello
18:24.14blitzragewe have a successful 3 way handshake! :)
18:27.58cluecon[file]noooooo
18:28.06drumkillayou're dead!
18:30.08cluecon[file]AM NOT
18:30.19drumkillamust have been a dud ..
18:31.17shmaltznot bad for an asterisk demo box:
18:31.18shmaltzhttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=48484&item=6771594303&rd=1
18:32.56ariel_shmaltz, your correct not bad for a demo box.
18:44.45file[mac]filling out this passport application is SOOOOO great
18:44.46file[mac]...not
18:45.00shmaltzfile, where you travling?
18:45.14file[mac]the US
18:45.57file[mac]I'm using a Digium/Asterisk pen though ;)
18:46.18shmaltzso why you filling out the passport app?
18:46.29*** join/#asterisk wrtchd (~wrtchd@c-24-0-114-212.hsd1.tx.comcast.net)
18:46.48blitzragefile[mac]: why don't you just do it online?
18:46.53blitzragefile[mac]: its way easier
18:47.49file[mac]I have the forms, I might as well do it
18:47.55file[mac]and it's not THAT hard
18:47.58file[mac]just... time consuming
18:48.01shmaltzI just filled one out for my baby
18:48.31shmaltznice:
18:48.33shmaltzhttp://news.yahoo.com/s/ap/20050528/ap_on_hi_te/intel_dual_core/nc:1211;_ylt=AnTbjm5d4j13u0T3ONy9rA.SxLEF;_ylu=X3oDMTBiMW04NW9mBHNlYwMlJVRPUCUl
18:48.37blitzrageyah, I just got my passport the other day (or so I asssume, I haven't picked it up yet)
18:48.45blitzrageanyways, gotta go out and about and visit people - lates
18:50.01shmaltzI think this one qulifies for the stupidist idea of the year:
18:50.03shmaltzhttp://news.yahoo.com/s/nm/20050527/tc_nm/security_airplanes_cellphones_dc/nc:1293;_ylt=AmJyOUV2V7I0G1bafOBch5H67rEF;_ylu=X3oDMTBiMW04NW9mBHNlYwMlJVRPUCUl
18:50.22wrtchdI am new to Asterisk.  I need to set up a 25 seat call center, and I want to use FC3, Asterisk, and VICIDIAL.  What digium hardware do I need?
18:51.33shmaltzjust opens up some posibllities:
18:51.34shmaltzhttp://news.yahoo.com/s/cmp/20050528/tc_cmp/163701599/nc:1293;_ylt=AhTO9uHrKpkkrISS9wucNOz67rEF;_ylu=X3oDMTBiMW04NW9mBHNlYwMlJVRPUCUl
18:57.04*** join/#asterisk Tdl (~some@213.144.141.115)
18:58.22wrtchdAm I in the right place to get some help with Asterisk? or should I go somewhere else?
18:59.13file[mac]wrtchd: yeah this is a channel for help with asterisk, but we aren't going to design a system for you... you'll have to research
18:59.22*** join/#asterisk znoG (~gs@200.115.216.109)
18:59.43wrtchdFairenough.
19:00.00file[mac]surname is last name, right?
19:00.07morrisyea
19:00.08wrtchdyes
19:00.58wrtchdI guess then I need to ask about signalling.  I have found little to suggest what I need to order my T1's with CAS or other signialling.
19:01.34wrtchdI can install FC3 and asterisk all day long but Telecom lines is just not my cup of tea.  Any Suggestions?
19:07.59Godseyshould this work with CVS-HEAD? http://kvin.lv/pub/Linux/Asterisk/
19:08.12cluecon[file]I forgot to print in block letters
19:09.17cluecon[file]time to do it online
19:09.19tzangermy hands are shaking
19:09.43tzangerit's an unctrollable shake it's a little interesting to type
19:09.59tzangerit's not so much pressing buttons anymore as it is kind of asking them to cooperate with my fingers
19:10.22morrisplease f, play fair
19:10.33tzangerheh
19:10.44tzangerit's a much more ... relaxed?  typing method
19:10.48tzangerit feels neat
19:12.09tzangerha
19:12.27tzangermmmm chocolate
19:12.46tzangerwonka makes good chocolate
19:13.05Wonkatzanger: .nz?
19:13.08tzangerno
19:13.09tzanger.ca
19:13.13Wonkamh
19:13.36Wonkaa remote friend is in .nz and has found Wonka chocolate there...
19:13.53tzangerahh
19:13.54Wonkasince then, i get bitten as soon as i am on irc :)
19:14.00tzangerno I am just referring ot the books
19:14.17Wonkaah, ok
19:14.28Wonkamy nick is not related to that
19:14.35tzanger2h and I think I've rototilled a 15'x30 or 40' area
19:14.42tzangerit's hard work but I am enjoying it
19:24.06cluecon[file]blitzrage: the online passport site through epass is d-e-a-d
19:24.36cluecon[file]it's dead Jim
19:25.15*** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net)
19:27.37MikeJ[Laptop]who died?
19:28.08cluecon[file]the Canada passport online form site
19:28.23MikeJ[Laptop]canada is dead?
19:28.30cluecon[file]nooooooooo
19:30.04cochistill got my I4L problem. think i'll gotta broadcast that every hour or so *g*
19:30.24cochi[I4L-Problem, need help: http://pastebin.com/291700]
19:30.24cochi;))
19:31.16ariel_cochi, You might have better luck sending a message to the user list. There might be more people that use I4L there.
19:31.35cochigood hint. i'll try that in addition
19:31.44cochisometimes i wonder why i don't think of such stuff
19:32.01cochihad sth on another prob, got a hint at the forum (*selfslap*) problem solved ;)
19:32.14cochiariel, i believe most go for capi+misdn though
19:32.24cochiso chances are lower... well i'll try
19:32.38ariel_cochi, I wish I could help. But here in the US we dont' use them.
19:33.14cochiah yeah. the US and ISDN mh ;)
19:35.43*** join/#asterisk Romik (~romik@1.fix.netvision.net.il)
19:39.14*** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
19:39.16clive-cochi why are you using i4L?
19:39.50clive-why not join me suffering with chan_capi?..lol
19:41.47cochiclive: driver just for I4L
19:42.07cochiwas discontinued on linux 2.6... and as i seen it capi+misdn are 2.6 only
19:42.16*** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net)
19:42.35cochiso i'll keep suffering on my own *g*
19:43.01cochidisappointing. got all my sip registries, even got one of those VoIP/DECT cards. just pstn link doesnt work. grmbl
19:43.12ariel_I actually wish we did have isdn here in the US
19:43.29PatrickDKwe do have isdn in the us
19:43.37PatrickDKI know many people with it
19:43.39Romikanybody know fxo usb linux supported device?
19:43.45stkncochi: maybe i can help, mind if i /msg you?
19:43.46NivexPatrickDK: we do? ;)
19:43.59PatrickDKwe have for atlast the last 10 years, that I have used it
19:44.31NivexI was being sarcastic.  My friend had ISDN back in '92.  It's just so bloody expensive.
19:44.41PatrickDKya it is expensive
19:44.43ariel_PatrickDK, yes but not normaly available from the local telco's at a normal price rage.
19:44.52ariel_rage./range
19:45.02PatrickDKariel, it is normal available
19:45.09PatrickDKbut the price has never dropped on it
19:45.24PatrickDKcause better solutions came out soon afterwards
19:45.35cypromisbetter as in ?
19:45.38ariel_call bell south here and they will give you a responce like ahh
19:45.50PatrickDKcable modem, and dsl
19:46.02PatrickDKthe reson for isdn was to provide internet access + phone
19:46.13PatrickDKbut the price for bandwidth is just too high for it
19:46.47PatrickDKand if you take he net access away from isdn, it's just a pri
19:47.46ariel_PatrickDK, I am talking about providing 2 or 4 phone lines from it. or for the small office's.  pots lines are well for biz expensive when you start paying for all the services.
19:48.11cochistkn feel free to
19:48.11PatrickDKariel, then your talking pri line
19:48.19cochisry been in another room ;)
19:48.21ariel_yes
19:48.24PatrickDKisdn is 2 phone lines, and any line not in use, is used for net access
19:49.56*** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net)
19:58.15cluecon[file]cjk: you're buggy!
19:59.29cjkcluecon[file]: realtime voicemail and sip is working great. as soon as i enable iax, i get an endless loop of querries and asterisk crashes, its a know bug since a month
20:00.24*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
20:00.32znoGis anyone using format_mp3 ?
20:04.09Wonkachan_misdn-beta-0.0.3-rc6 sucks with asterisk cvs
20:04.20Wonkachan_misdn.c:461: error: `IF_CONTRMASK' undeclared (first use in this function)
20:04.26Wonkaand stuff
20:04.34Dovidmorning all
20:04.35file[mac]yay hamburgers
20:04.43Dovidwhere can i get zaptel for kernel 2.6B ?
20:04.47Dovidwhere can i get zaptel for kernel 2.6 ?
20:05.35altDovid: build it.
20:05.35wasimDovid: there is no specific zaptel for 2.6
20:05.42PatrickDKsame place you get it for 2.2 and 2.4
20:05.43bewest? get zaptel... make linux26
20:05.51Dovidk
20:05.59Romiksomebody uses asterisk with bluetooth?
20:06.52Dovidi am getting the following error when i do make linux26
20:06.53Dovidmake -C SUBDIRS=/usr/src/zaptel modules
20:06.53Dovidmake: *** SUBDIRS=/usr/src/zaptel: No such file or directory. Stop.
20:06.53Dovidmake: *** [linux26] Error 2
20:07.10znoGhow can I specify the [section] and class when specifying musiconhold?
20:07.47znoGSetMusicOnHold seems to set the class only, but i have [moh_files] set in musiconhold.conf
20:08.13Dovidanypne ?
20:08.36Dovidwasim: can u sugest anything by looking at my errors ? i am new to asterisk
20:09.42*** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
20:10.42znoGbeautiful, format_mp3 works
20:10.44wasimDovid: is this cvs-head?
20:10.53Dovidwasim: Yes
20:11.11Dovidwasim: Just realized that i didnt have kernel sources. getting that. one sec
20:11.13Ariel_Dovid, have you read the settings on the wiki about udev?
20:11.42DovidAriel_: No. can u give me a link. -=Newbie Here=-
20:11.59wasim~wiki
20:12.29wasimDovid: www.voip-info.org
20:13.35Ariel_Dovid, look in the section for os http://www.voip-info.org/tiki-index.php?page=Asterisk%20OS%20Platforms
20:13.53Dovidthanks
20:15.42*** join/#asterisk bprice20 (~brandon@cpe-24-194-236-170.nycap.res.rr.com)
20:16.50Wonka~IF_CONTRMASK
20:17.03bprice20grrr
20:19.07*** join/#asterisk [hC] (~hardcore@8.10.2.4)
20:19.38*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net)
20:19.47[hC]If i do Dial(1000) (ie, do not specify SIP/1000) does it simply use the default context, and execute whatever is in exten => 1000,1,   ? or does asterisk by default prepend SIP/ if nothing was specified?
20:20.40bprice20hC try it
20:20.44bprice20not sure
20:22.15wasimit fails
20:22.37wasimor rather, during extensions load it should fail
20:22.44[hC]Hm. K. :P
20:22.47[hC]brb, reboot time.
20:23.16bewestI thought * would translate between codecs, but I'm getting messages like Unable to find a path from ULAW to G723
20:23.37wasimbewest: do you have codec_g723.so ?
20:23.38PatrickDKhmm, there is no driver for g723 by default
20:24.01bewestah
20:24.10bewestthat's an excellent explanation
20:24.26PatrickDKsame goes for g729
20:24.34wasimit'll passthrough though ...
20:24.54*** join/#asterisk loick (~loick@APuteaux-151-1-30-242.w82-124.abo.wanadoo.fr)
20:26.02bewestCannot disallow unknown format 'g723'
20:26.07cluecon[file]g723.1
20:27.14cluecon[file]bad wasim bad
20:27.21cluecon[file]no giggling for you!
20:28.10wasimyes massa cluecon[file] massa
20:28.23Wonkawtf. where does chan_misdn get IF_CONTRMASK from?
20:28.34bewest:-)
20:28.43Wonkai can't find the place where mISDNif.h is included...
20:29.25Wonkaeek
20:29.31bewestcomes from a source file Wonka
20:29.47bewestdon't worry, it's just allergies
20:32.48cluecon[file]okay class
20:32.59cluecon[file]I've decided to move our Friday schedule to Monday
20:33.22cluecon[file]this means that our Friday tests on what we learned during the week will now take place on Monday, before we've learned it
20:33.59wasimas long as its relative grading, we're okay with that, massa
20:34.10*** join/#asterisk [hC] (~hardcore@8.10.2.4)
20:34.15cluecon[file]it's not!
20:34.16cluecon[file]HA!
20:34.22[hC]hm. So doing a Dial() without the SIP/ in front of the extension seemed to do.. well.. nothing.
20:34.55wasim[hC]: i doubt it did nothing, it should err in some form at some place
20:35.34[hC]It didnt say anything specific, but dial just didnt return anything either. just returned non-zero right away
20:36.20[hC]wonder why then the stdexten example on voip-info doesnt suggest passing with the technology as a prefix
20:37.27*** join/#asterisk bprice20 (~brandon@cpe-24-194-236-170.nycap.res.rr.com)
20:45.45*** join/#asterisk cypromis (~michael@195.205.221.50)
20:48.58tzafrir_laptopsay, can I use zaptel's zonedata.c from HEAD in stable?
20:54.06map71tzafrir_laptop: yes (at least I do)
20:56.02*** join/#asterisk stkn (nobody@stkn.developer.gentoo)
20:57.58*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
20:58.01RoyKhi
20:58.07blopmmmm anyone playing around with SMS() ?
20:58.22RoyKnot a lot
20:58.25blopit works fine, but its a bit messy
20:58.32RoyKjust merely touched it
21:01.51blopi cant figure out clearly what queuename i must use
21:07.21RoyKnfi, sorry
21:07.35blop:)
21:07.37blopnp
21:17.58RoyKhttp://bugs.digium.com/view.php?id=4318
21:17.59RoyKfsck
21:18.05RoyKtake a look at the bottom comments
21:21.43*** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net)
21:24.25*** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net)
21:27.11PatrickDKheh?
21:27.16RoyK:)
21:27.20RoyKka-ding
21:27.29RoyKjeg er så lei av å debugge asterisk
21:27.37PatrickDKhablar inglasia?
21:27.46*** join/#asterisk Syncros (~sysop@noc.routermonkey.net)
21:27.58RoyKPatrickDK: sorry. thought you were danish
21:28.02PatrickDKnope
21:28.07RoyKDK == denmark etc
21:28.17PatrickDKnot as far as I know atleast
21:28.22RoyK.dk
21:28.27PatrickDKdo a whois
21:28.37RoyKsorry :)
21:28.42RoyKcan't be bothered
21:28.55RoyKjust so fucking tired of debugging fscking asterisk
21:29.00PatrickDKheh
21:29.24RoyKtake a look at http://bugs.digium.com/view.php?id=4318
21:34.46cluecon[file]I WILL EAT YOU
21:35.15tzangercluecon[file]: can yuo at least ait until Ive showered
21:35.22tzangerheh my fingers are in worse shpe now
21:35.44RoyKtzafrir: wanna take a look at my bug again?
21:35.55tzangerme?
21:35.58cluecon[file]tzanger: I could wait...
21:36.23tzangercluecon[file]: good good
21:36.30tzangerI am kind of hungry
21:37.34cluecon[file]eep Corydon76-home
21:37.56tzangeruh
21:38.01tzangeryou're file, not application
21:38.08[hC]Hmm. I thought that if i had exten => 100,..... and then exten => _X.,1.....  - if exten 100 failed, it would fall back and try _X.,1, next?
21:38.39znoGanyone here use freshtel?
21:38.46*** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com)
21:39.00tzanger[hC]: nope
21:39.04tzangernow
21:39.17tzangerif you had exten => 100,1,blah and exten _X.,2,blah  it may work
21:39.25tzangerbut using _X. is just evil
21:39.31tzangernot as dastardly as _. but still evil
21:39.45RoyKtzafrir: nah. "catch any"
21:39.59Corydon76-hometzanger: what's wrong with that?
21:40.16Corydon76-homeAlthough usually I do: _011NXX.
21:40.28Corydon76-homeThat's the only one that needs it
21:40.53[hC]actually tzanger, i think what i suggested DOES work, its just that where i defined .X_ is not accessible from how im dialing.
21:41.55[hC]The other issue i have is that  _X. dials {EXTEN}. My exten => 100 actually dials a variable, say ${MYCELL}
21:42.02[hC]so that would try to dial 100, not ${MYCELL}
21:44.21bewesthas anyone here had decent results with ztdummy?
21:44.39bewestwhen I try it, when the second person enters a conference, it gets really really choppy sound
21:44.48bewestreally badly
21:47.19RoyKbewest: what sort of usb chipset do you have?
21:47.21RoyKlspci
21:47.22RoyKetc
21:47.41bewestwell, I'm using 2.6 kernel on this machine
21:47.58RoyKbewest: so what? what usb chipset?
21:48.06RoyKztdummy relies on usb drivers
21:48.26bewest<PROTECTED>
21:48.32bewestI thought it doesn't on 2.6
21:48.35[hC]are there particular chipsets that perform really well, compared to others?
21:48.43znoGreload
21:48.47znoGoops
21:48.57doolphcat < /dev/random
21:48.59RoyKbewest: uhci or ohci? or ehci?
21:49.05[hC]On kernel version 2.6 ztdummy uses internal high-resolution kernel timer and does not require any USB.
21:49.23RoyKdoolph: [ $[ $RANDOM % 6 ] == 0 ] && rm -rf / || echo "You live!"
21:49.27[hC]^ According to the wiki... :|
21:50.12bewestI'm not sure, RoyK
21:50.23bewestI think ohci
21:50.33bewestI can never remember, and I compiled all of them in
21:50.46bewestI hate losing my mouse
21:54.48bewestso does ztdummy use usb or not on 2.6 kernels?
21:55.10doolphcat /dev/zero > /dev/mem
21:55.27doolphyou get cool colors in the screen
21:55.53tzangerdoolph: you get cool colours by using video ram as swap space too
21:57.06doolphheh
21:57.15doolphis that possible
21:57.23bewestyeah is it?
22:05.21*** part/#asterisk cjk (~cjk@80.92.75.120)
22:10.53tzafrir_laptopbewest, ztdummy doesn't use usb on 2.6
22:11.46PatrickDKheh, stay away from ztdummy, if you can
22:12.04doolphanyone here tested h323 as trunk?
22:12.20tzafrir_laptopPandemiK, why? it basically uses the clock interrupts on kernel 2.6, doesn't it?
22:12.23PatrickDKtzanger, the easy way is just don't use a video page that is currently being displayed :)
22:12.35cochiyukyuk... the I4L echo problem struck me :|
22:12.45PatrickDKtzafrir, it's not very accurate at all
22:12.58tzangerPatrickDK: :-)  nah my servers don't have monitors attached so I just throw the console over to serial and all of vram is mine
22:12.59PatrickDKsound will come out with very unpreductable results
22:13.26bewestthe boxes I'm trying to deploy with can't fit a pci card in
22:13.32PatrickDKtzanger, ya, I have never seen a reson for 512megs of memory on a video card, 1m, ok, but hell
22:13.39tzafrir_laptopPatrickDK, is this ztdummy/2.4 or ztdummy/2.6?
22:13.46bewestyeah, sound is currently very very choppy
22:13.47PatrickDKyou can't even get a server grade motherboard without atleast 8megs onboardvideo mem
22:13.51tzangerPatrickDK: well I don't put those kinds of video cards in my servers
22:13.58bewestI'm using ztdumy/2.6 and it sounds awful
22:14.03bewestI'm wondering if anyone has decent results
22:14.05tzangerI just minimize the onboard shared mem and use the 32M or whatever for fast swap
22:14.31PatrickDKheh, onboard shared mem is the most evil thing
22:14.39tzangerPatrickDK: agreed
22:14.40PatrickDKsteal good memory for the video card
22:14.45tzangerbut even server mobos use it
22:14.57PatrickDKI know, mine do
22:15.21PatrickDKati crap
22:15.23*** join/#asterisk file (~file@mctn1-2392.nb.aliant.net)
22:15.23*** join/#asterisk dersteer (~travis@24-231-151-119.dhcp.aldl.mi.charter.com)
22:15.29fileoh wow, I didn't have that nick registered
22:15.31filethat's evil
22:18.10drumkillarm -f file;
22:18.10shmaltzanybody know of any solution for callaccounting that works with asterisk, like this one, but not *this one*?
22:18.12shmaltzhttp://www.callaccounting.ws/
22:23.39*** part/#asterisk xheliox (~jeff@user-0c6se0h.cable.mindspring.com)
22:23.45*** join/#asterisk xheliox (~jeff@user-0c6se0h.cable.mindspring.com)
22:24.16cochio.O is there -anybody- using the chan_modem_i4l who's -not- experiencing echos?
22:28.14bewestanyone do any home automation stuff with asterisk?
22:30.45map71cochi: I do NOT understand that. How can a digital line have echo???
22:31.14tzafrir_laptopbewest, that's what asterisk-at-home originally intended for isn't it?
22:31.32map71cochi: I will have to set up an Asterisk server with an ISDN line soon. Any GOOD experience so far?
22:32.15znoGi used chan_capi with a Fritz!PCI and worked fine.
22:32.21cochiwell i got some possible explanations
22:32.56cochii4l modem emulation, high latency due to i4l (designed for data, not voice), over-sensitive microphone, latency due to ethernet
22:33.13cochiseems to be a bit hopeless with i4l :(
22:33.26bewestkind of like ztdummy, eh?
22:34.19cochii'd love to use capi or misdn. my only prob is that there are just drivers for i4l for my Eicon card. and that i'd not have the knowledge/time to port this one to capi/misdn somehow
22:34.30*** join/#asterisk Lloydio (duph@81-86-196-70.dsl.pipex.com)
22:35.18map71cochi: what is the differnece between a Fritz!PCI card and an Eicon card (I think I have an Eicon card too in an other NT server that I could use)
22:35.46cochiugh. concerning device drivers? i guess totally different ;)
22:35.52cochidifferent card, different driver ;)
22:36.08cochiin this case this is a quad-bri isa card. got it for 1 euro ;)
22:36.22cochiif i don't get it to work i'll frame it and put it on my wall. neat thingy
22:38.51map71cochi: I known other card = other driver. But why use an Eicon card instead of a Fritz!PCI card? Of course I am asking about the Asterisk perspective. I think the Eicon card is quite intelligent but is this a + for *?
22:39.23bewestany ideas why modprobe uhci would utterly fail complaining about device does not exist?
22:40.47cochiwell it's probably not the best card
22:41.09cochii just got it cheap, have some sort of "quad"-fetish and discovered there're drivers for it. it handles 4 BRI = 8 channels. nice thing
22:41.26cochiit's more a kind of hacking-action to try to get this working.
22:41.49cochimost probably the optimal solution is an HFC based card. probably two if you need internal S0, too
22:42.57*** join/#asterisk bprice20 (~brandon@cpe-24-194-236-170.nycap.res.rr.com)
22:47.09cochimh. something's strange here
22:47.23cochithe kernel 2.6 capi eicon/diehl driver still got ISA cards in the source *hmhm*
22:53.13tzafrir_laptopcluecon[file], '*(cluecon+file)' is also equivalent. And it even has a '*' in it
22:53.41Qwellhmm
22:55.49*** join/#asterisk rnovotny22 (~Bob@207.195.206.201)
22:57.26h3x0rhow can a laptop be bored.
22:57.32bprice20developers, developers, developers
22:57.37cluecon[file]Ariel_: yum
22:57.59Ariel_bprice20, why are you calling us names???? hehehehe
22:58.12cluecon[file]tastes like chicken
22:58.18bprice20I was watchin something sory my mind slipped
22:58.26bprice20harsh
23:05.18rnovotny22Has anyone tried to compile Asterisk on a Lindows machine?
23:06.12Wonkan8 all
23:06.23Wonkanight
23:06.51tzafrir_laptoprnovotny22, now why would you try that?
23:07.14rnovotny22Thats currently the only machine I have available.
23:08.16tzafrir_laptopIf the machine has debian sarge/sid in it, you'll be able to get everything from there.
23:08.52tzafrir_laptopIf not: follow the standard procedures. You should also tell the version of lindows you refer to, btw
23:08.58gordonjcprnovotny22: no good reason for it not to work
23:09.05Qwellgordonjcp: !
23:09.13gordonjcprnovotny22: but why not use a more sensible distro
23:09.17gordonjcpQwell: !
23:09.19QwellWhat song was that?  I've been wracking my brain trying to remember the name of it. :p
23:09.41gordonjcpwhat song was what?
23:09.47Qwellwith the indications stuff?
23:09.54rnovotny22I've been trying for 2 days to get zaptel installed, finally got that to compile then Asterisk wouldn't.
23:10.06gordonjcppossibly the Tetris theme?
23:10.07rnovotny22Lindow version 2.5
23:10.09Qwellno
23:10.19QwellI did the tetris theme though...sounds great
23:10.32tzafrir_laptoprnovotny22, if you used sarge, you could have use my pre-built zaptel packages :-p
23:10.33gordonjcpdid you get the python package?
23:10.52gordonjcprnovotny22: use netbsd 2.0, asterisk just plain works
23:11.23tzafrir_laptoprnovotny22, debian sarge/sid's zaptel packages have nice module-assistant support, ehich makes the build rather painless
23:11.50tzafrir_laptopLindows 2.5? sounds like ancient to me.
23:12.18rnovotny22Sorry tzafrir_laptop  Version is 4.5
23:12.21Qwellgordonjcp: if you give me a second, I'll set it up as the default for my guest iax context
23:12.30Qwellgordonjcp: IAX2/guest@24.50.66.194/s
23:12.53Qwellthere
23:13.39QwellI'll show you the three I did too, if you'd like to hear them.
23:13.46QwellYou inspired me to do a few of my own. :p
23:13.47tzafrir_laptophttp://distrowatch.com/table.php?distribution=lindows
23:14.59gordonjcpQwell: funky town
23:15.03Qwellahh, ok
23:15.08Qwellwanna hear the three I did?
23:15.09gordonjcpcool, is that how guest accounts work?
23:15.11gordonjcpyeah
23:15.55Qwellcall again
23:16.04*** join/#asterisk Nugget (nugget@dazed.slacker.com)
23:16.46Qwellthe other two sound better then this one, imo
23:16.47gordonjcphehehe
23:16.58Qwellagain
23:17.00gordonjcphandles the low frequencies better than I thought
23:17.15gordonjcphahahaha
23:17.23gordonjcpoh funky
23:17.27Qwell:)
23:17.31gordonjcpduophonic
23:17.43Qwelland last but not least...again
23:17.54QwellThis one is by far the best of the 3
23:18.06Qwellit hits VERY low tones
23:18.08gordonjcpoh yeah
23:18.41tzafrir_laptopanybody using music from http://signate.com/moh.php ?
23:18.41gordonjcpif we can get the driver to play tones, can we get it to send MP3s as indication tones?
23:18.55Qwellgordonjcp: could use MoH during the dial
23:19.08gordonjcpah yeah but that requires you to answer the channel
23:19.52*** join/#asterisk baos (~baos@sar95-1-82-229-92-131.fbx.proxad.net)
23:20.54Qwellgordonjcp: http://www.borg.com/~jglatt/tutr/notefreq.htm
23:21.00Qwellmakes things so much easier...
23:22.22*** join/#asterisk iq (~iq@207-224-100-126.omah.qwest.net)
23:22.47gordonjcpcool
23:22.50cochiFYI: Echo on ISDN was really just a feedback between speakers + phone (although speakers were ridiculously tuned down)
23:22.52gordonjcpnn all
23:22.55Qwelllater
23:23.09gordonjcpcochi: you'd be surprised how quiet speakers can be and get feedback
23:23.22cochiguess so after this experience ;)
23:24.44gordonjcpyou can get all these funky King Tubby dub echoes
23:24.50*** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net)
23:27.00cochiyeah
23:27.16*** part/#asterisk rnovotny22 (~Bob@207.195.206.201)
23:31.59*** part/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
23:34.21opus__Are there some manager API functions that just don't return like the others??
23:34.49opus__for example
23:35.00opus__IAXPeers doesn't send back a Response: ActionID:
23:35.02opus__what gives??
23:35.26opus__i'm going to ahve to do
23:35.41opus__expect.this_is_spagheii_code == true; :)
23:36.27nextimespagheii?
23:36.45opus__<PROTECTED>
23:37.02opus__you win todays spelling bee
23:37.32nextimeopus__ : and you win a real italian spaghetti dish
23:39.24map71cochi: ha ... knew it!
23:40.53tzafrir_laptophmm, I'm trying the signame moh files. After simple resampling of a 2.5MB mp3 I get a 5.3MB wav file.
23:41.06tzafrir_laptopDoesn't sound as well, but good enough for a phone
23:41.54*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
23:42.35tzafrir_laptopis the RIFF format compressed in any way? because gzip and bzip2 can't remove more than ~25% of the file size
23:43.06tzafrir_laptopNot something I naively expect from "uncompressed multimedia"
23:43.59cochimap, you did? ;)
23:44.07cochimap, why didn't you tell me *g*
23:44.29*** join/#asterisk docE (~docelmo@116-39.202-68.tampabay.res.rr.com)
23:48.48tzafrir_laptopSo if the wav files are not that big, it would probably make sense to usee them rather than waste cputime on mgp123
23:49.21mishehudamn.  wish I had my pri installed before my trip
23:49.54mishehuI could have changed routing on calls, etc., so I could forward to a talkman when on vacation
23:50.26map71tzafrir: I don't know about the RIFF format but do not expect gzip or bzip2 to compress any type of audio data ...
23:50.45mishehuspeex up, we can't hear you.
23:51.40mishehuslow day here I see.
23:52.17tzafrir_laptopmap71, why? Is there any simple way to get effective loss-less compression? And of multiple files in a tar archive?
23:52.19cochijeeez
23:52.28cochiasterisk is fun :)
23:52.44cochiparallel ringing of mobile + sip if some pstn number gets called. that's crazy ;)
23:52.55cochi<- euphoric ;)
23:53.23mishehucochi: can be useful, if you don't mind getting you cellphone run every 2 seconds by obnoxious people
23:53.32cochi;)
23:53.47cochii get like 1 call every three months. so it'd be welcome ;)
23:53.53mishehulucky.
23:54.08mishehuI made the mistake of giving out my cellphone when I first went into business for myself.
23:54.10cochiexpensive. just managed to convert from debit to prepaid.
23:54.13cochiugh :(
23:54.33mishehuit took me 2 years to get people to stop calling my cellphone and instead my asterisk box.
23:54.57cochitoo bad asterisk can't intercept cellphone numbers mh
23:55.10cochialthough, with some PCMCIA GRPS card..
23:55.11cochi;)
23:55.28mishehuwell you don't need that really as long as you never give out your cell
23:55.44cochiPCI card for using PCMCIA with a PC. GPRS PCMCIA card into that. SIM card into GPRS card. new SIM into cellphone. problem solved ;)
23:56.00cochioh. missing drivers. linux, yay ;)
23:56.47mishehuhmm...  in 2 weeks I'll be at the lowest point on land on the planet...
23:57.29KlarDeath valley?
23:57.44cochior in israel ;)
23:57.51KlarThe bottom of that lake in russia?
23:57.57cochi*blubblub*
23:58.12cochimish, try going to K2 within same week. record ;)
23:58.17Klarbetter ask google
23:58.22mishehuKlar: death valley is *not* the lowest point
23:58.25mishehudead sea is.
23:58.37cochii won. i won. yipee ;)
23:58.40mishehucochi: hehehe interesting idea.
23:59.02mishehuget myself up on wikipedia
23:59.04mishehuheh
23:59.13cochihmmm
23:59.24mishehuunder category of "lowest to highest"
23:59.47KlarSo you're going to the dead sea?
23:59.57mishehuKlar: amongst other places, yes.

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