irclog2html for #asterisk on 20040928

00:01.43visik7how can I solve this : http://pastebin.ca/1140
00:03.24file2old code is bad mmmk
00:04.47*** join/#asterisk JRB (~jbenson@genpubad.gotadsl.co.uk)
00:04.49visik7so it's better to leave pbx_*console.so away ?
00:05.10Mocis there a cli command to register and unregister from a sip connection
00:05.14MocÉ
00:05.16Moc?
00:05.59visik7is the client that register a sip connection to a server
00:06.13AgiNamucan someone call a 866 number for me?
00:06.18visik7not the opposite
00:06.20AgiNamuPM me if so :)
00:06.23Mocwell * also connect to other server
00:09.33visik7Moc understand
00:10.32bkw_la la la la
00:11.24*** part/#asterisk TEKjacob_ (~tekjacob@c7.e3bccf.client.atlantech.net)
00:11.51brc_lala?
00:12.05brc_znoG, no.
00:12.08*** part/#asterisk ahampton78759 (~ahampton@rrcs-24-173-78-2.sw.biz.rr.com)
00:12.33*** join/#asterisk m-00kie (~r3l34s3d@pcp09300018pcs.arlngt01.va.comcast.net)
00:14.07ariel_Lets see who might be able to help me configure a Cisco 7960 that is behind a Nat to talk to an Non natted Asterisk?
00:15.31habakuksurvey. anyone be interested in DID's from the San francisco bay area? I can provide IAX or sip
00:15.50paulcariel_ I have that config working
00:17.03ariel_Can you give me a peak at your setup files like the one from the Cicso phone Please?
00:17.55pfnhabakuk is that so, pricing?
00:18.07paulcariel_ I can email you my SIPDefault and SIPxxxx.cnf if you like?
00:18.10pfnand TOS?
00:18.29ariel_paulc, sure arielb27@gmail.com thank you very much.
00:18.40habakukpfn 5$ / DID
00:19.03*** join/#asterisk imcdona (~imcdona@sbi-24-177-181-60.mtv.al.charter.com)
00:19.22lokoAnyone here offering the 412 (Pittsburgh) DID
00:19.39pfnhabakuk and what are the terms of that?  unlimited incoming, etc. etc. ?
00:19.45pfnpayment?  website?  quantity?
00:19.46pfnhmm
00:19.54pfnwebsite as in what's your site
00:21.13habakukpfn : email sales@fonomni.com, website: www.fonomni.com, and yes unlimited incoming
00:22.18pfnwhat codecs do you have?
00:23.44slePPanyone doing spa-2000 provisioning?
00:23.46slePPvia https/tftp
00:23.52habakukpfn: supporting all at this point. working on g.729
00:24.05paulcariel_ sent! although outlook was being wank about .cnf files so I renamed to .txt
00:24.59ariel_thanks I got them.
00:30.45file2mmm Enya
00:31.12paulcEnya? Really?
00:31.15*** join/#asterisk WifiFred (~wififred@198.231.65.5)
00:31.18file2good music.
00:31.28paulcSail away sail away sail away...
00:31.46paulcor "c'est le way, c'est le way, c'est le way".. depending on if you're ripping the piss or not..
00:31.59paulccurrent favourite: Blinded By The Lights by The Streets
00:32.50file2it's soothing music
00:32.53file2helps me think and clear my mind
00:32.54*** join/#asterisk EarlGrey (~EarlGrey@earlgrey.user)
00:34.36*** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc)
00:34.36*** mode/#asterisk [+o twisted] by ChanServ
00:35.13icebalm"ripping the piss"
00:35.19icebalminteresting expression
00:35.31twisteduhm, sounds extremely painful... like... an std even.
00:35.56icebalmor like a kidney stone
00:36.25EarlGreyripping the piss = uk slang = more extream version of taking the piss
00:36.29ariel_kidney stone's argh
00:36.37brc_ahhhhhhhhh
00:36.44icebalmtaking the piss?
00:37.06icebalmwhy don't you just say pissing
00:37.29tzangerhaha
00:37.31gambolputtyYou don't take shit you leave it
00:38.13icebalmyeah I don't understand this whole line of expressions, and I'm a native english speaker
00:38.20file2Flora's secret!
00:38.57*** join/#asterisk AgiNamu (~zzzs@4.79.150.34)
00:39.08ariel_Funny when I was in the service we would say we need to drain the vain....
00:39.36icebalmvein
00:39.57twistedi usually say i'm gonna go take a shit
00:39.59PhilMI need to drain my lizard
00:40.02twistedbut i must start correcting myself
00:40.07twistedturn some heads
00:40.30twistedOOOOH
00:40.39twistedI need that samuel L jackson soundbyte for a 7960 ringer
00:40.56twisted"If you're going to shoot me, go ahead and shoot me!  But  i HAVE TO ANSWER THAT PHONE!"
00:41.09icebalmlol
00:41.20paulcLOL.. that's porno cheesey but kinda funny
00:41.24twistedpaulc
00:41.24twistedyeah
00:41.34twistedit's a line from die hard with a vengance
00:42.05*** join/#asterisk isamar (isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp)
00:42.22paulcevening all :-)
00:42.25paulcwhat's shaking twisted?
00:42.54file2Enya!
00:43.10file2Enya Enya Enya.
00:43.43Mucklanyone in here using asterisk to register => a SIP account from sipgate.de?
00:43.50paulcshe's almost as bad as Celine Dione
00:43.50icebalm"If you're going to shoot me then you go ahead and you shoot me, but I have to answer this phone."
00:43.57*** join/#asterisk algorithmn (~none@ool-182f915a.dyn.optonline.net)
00:43.57paulcslePP - chuck me a Sipura and I'll let you know ;-)
00:44.14file2no sight for you!
00:44.30paulcomg you're one evil little mofo sometimes!
00:44.31ariel_oh no the radio is playing Shakira... got to get another station on.
00:44.45slePPpaulc: heh.
00:44.59file2SO who wants to see the latest tracking on my PAP2-NA?
00:45.10*** join/#asterisk isamar (isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp)
00:45.11ariel_slePP, I guess no one is doing the mass setups right now.
00:45.17isamarHi folks
00:45.18*** join/#asterisk lancey (Shady@support.net1.cc)
00:45.19algorithmnwhy isn't $agi -> exec(dial,UN@HOST\${NUM}); working??
00:45.21slePPariel_: i wanna tftp boot my new pap2-na :>
00:45.22fileit's not good tracking
00:45.23isamaranyboyding usingnufone here?
00:45.32slePPisamar: what's nufone?
00:45.43mlh407I am
00:45.47lanceyguys
00:45.55lanceyi seem to be having trouble with H323
00:45.56fileSep 27 2004, 1:00PM. CONCORD, ON, CA: THE PACKAGE IS DELAYED DUE TO EMERGENCY CONDITIONS BEYOND UPS' CONTROL
00:45.57isamarmlh407...
00:46.01lanceycould i call someone of you?
00:46.05lanceyto test it?
00:46.08isamarslePP: www.nufonet.net
00:46.10isamarops
00:46.14isamarwww.nufone.net
00:46.18slePPoh.
00:46.18mlh407lancey: I am lost
00:46.18slePPneat.
00:46.27isamarmlh407... trying to call prefix 380
00:46.28slePPfile: nice.
00:46.32isamarukraine...
00:46.38file2isn't it just peachy?
00:46.45isamarnot getting... which dial plan do you use for other countries but US?
00:47.00mlh407I am only using it for incomming class
00:47.00slePPemergency conditions
00:47.02mlh407*calls
00:47.03slePPie: it's in florida. sorry
00:47.08paulcfile: what's that all about then? emergency conditions? like.. a lack of donuts for airport freight workers? or..?
00:47.10lancey<mlh407> lancey: I am lost?
00:47.12isamarok
00:47.16lancey* ariel_ does not use h323
00:47.22lanceyariel_ i don't need someone with h323
00:47.25lanceyi just need a phone number
00:47.27lanceyPOTS
00:47.27file2algorithmn: $AGI->exec('Dial',UN@HOST\${NUM});
00:47.38lanceyanyone willing to help me?
00:47.38ariel_lancey, where?
00:47.40file2paulc: nobody knows!
00:47.45isamarlancery; wasup?
00:47.46lanceyariel_ doesn't matter
00:47.49file2I think it's a massive conspiracy
00:47.50lanceyanywhere but bulgaria
00:48.13paulclancey: talking yellow pages: +1 604 299 9000
00:48.18AgiNamufile2, dont you need to specify IAX or SIP?
00:48.18twistedpaulc, not that
00:48.31lanceypaulc: i love you :)
00:48.33file2AgiNamu: it's his AGI, I'll let him figure it out
00:48.37paulctwisted: not?
00:48.43AgiNamuheh
00:48.43paulclancey: I'm flattered ;)
00:48.45twistedpaulc, nah.
00:48.46lancey:))
00:48.51paulcuh.. not what?
00:48.56twisted[19:42] <paulc> what's shaking twisted?
00:49.02paulcah
00:49.02*** join/#asterisk hastur (~hasturr@sobek.7g0.net)
00:49.04ariel_lancey, get your self one from: http://www.kallfree.com/ it's free
00:49.08paulclagged reply ;-)
00:49.09file2people can't learn by being given things, they have to discover them on their own!
00:49.15twistedpaulc, yeah, mentally lagged.
00:49.52paulcI get like that..
00:49.58paulcI got to write a report but it's kind of a headfuck
00:50.21file2you IRC like you talk.
00:50.31paulcI always type like I talk :-p
00:50.32twistedlol
00:50.43paulcemail, MSN, IRC... s'how I am :)
00:50.56Mucklwhats the IVR number again? 1800 555 tell?
00:51.02algorithmni've tried a few ways
00:51.12*** join/#asterisk TEKjacob_ (~tekjacob@c7.e3bccf.client.atlantech.net)
00:51.29algorithmnthinking about ditching perl and using c
00:51.57fileexxxxxxxxxxxxcellent
00:52.03algorithmnit is creating errors in logging
00:52.18AgiNamuYES :D
00:52.22AgiNamuUse the C API :)
00:52.34algorithmni was thinking it was a perl api specific glitch
00:52.38twistedsing
00:52.39WifiFredsong
00:52.40twistedsing a song
00:52.43twistedmake it loud
00:52.45twistedmake it strong
00:52.46twistedsing
00:52.47WifiFredsong
00:52.48twistedof good things
00:52.48twistednot bad
00:53.01TEKjacob_Hey all... I have been messing around with the 7960 Cisco phone... Got it working (thanks to you all) Although I am having three problems. 1. Cant telnet (.cfg file seems to be set right) 2. won't display custom logo 3. wont pickup extra ringers
00:53.05TEKjacob_any ideas?
00:53.10*** join/#asterisk Vco (~Vco@S0106080020aa7650.wp.shawcable.net)
00:53.21VcoHello all....
00:53.29twistedTEKjacob_, sounds likt the tftp isn't working right
00:53.36fileVco: Hello.
00:53.45TEKjacob_yeah, but I can't find any logs...
00:54.54twistedof course not
00:55.00algorithmnagi: have you ran across this problem if i may ask?
00:55.04VcoAny */mpg123/alsa/musiconhold efficianados around?
00:55.22lanceyweee
00:55.24fileVco: verify you have version 0.59r if you're having MOH probs
00:55.25lanceyit's working
00:55.25lancey:)
00:55.40lanceyanyone knows if chan_h323 has been modified lately?
00:55.45lanceyi updated 1 hour ago
00:55.51lanceyand now everything seems to run just fine
00:56.20Vcoyea..0.59r is on the box..
00:57.08fileh323 was last updated: September 21st
00:57.16lanceythank you, file!
00:57.21lanceyJerJer's done a nice work!
00:57.24lanceyi love him too :)
00:57.35lanceybkw_ what's up?
00:57.38Vcoshows as running on the * console when connecting to music on hold, audio card is working under alsa...grandstream is behind a nat, voice and everything works fine...just zero audio output from moh
00:57.40filebkw_: you and your silly screaming
00:57.42paulcpull yourself together!
00:57.56Vcobeen scouring wiki and google and whatever else i can find.
00:57.59paulche screams like a gurrrrl eh? ;-)
00:58.03Vcolosing my mind at this point..
00:58.05fileVco: you verified there's no rogue copies of an old mpg123 on the system?
00:58.34*** join/#asterisk Vulture- (~Vulture@247.131.vbnet.net)
00:58.38Vulture-urg damn internet
00:58.44lanceyVulture-
00:58.48Vcoi'll doublecheck...clear and do another make isntall to be sure..
00:59.00fileVco: well your distro may have installed mpg123 in another location
00:59.08algorithmnbkw: i've been thinking of an idea for you
00:59.12bkw_or mpg321 with a synlink to mpg123
00:59.19bkw_algorithmn and that is?
00:59.22Vcono..it is mpg123
00:59.23fileindeed
00:59.40lanceyguys
00:59.42algorithmnmind if i talk in... ahhum, private?
00:59.45lanceyi need to test H323 2-way
01:00.04lanceyanyone willing to speak with me for just 10 seconds
01:00.24fileI don't do H323.
01:00.27lanceyno
01:00.31bkw_I dont either
01:00.33AgiNamuIS it possible to send 2 updates in Mysql at once (via real_query)?
01:00.36lanceyi'll call you on a POTS
01:00.38bkw_even though its a gay protocol.. I wont do it!
01:00.43lanceyneed to test the gateway
01:00.45fileI don't have a DID at this time anyway
01:00.48Vulture-I don't do fatchicks... I mean H323
01:00.58lanceyyeah yeah i know H323 is crappy
01:01.04lanceybut one of my providers uses it
01:01.09lanceyi can't do anything about that
01:01.10lancey:(
01:01.29slePPbkw_: do you have the sipura provisioning compiler? :>
01:01.45bkw_slePP go talk to sipura they will give it to ya
01:01.47bkw_if you ASK for it
01:01.53bkw_if not let me know i'll smack em around
01:02.03bkw_how long ago?
01:02.07slePP30 seconds
01:02.09lancey:)
01:02.16bkw_god damn impatient fucker
01:02.18bkw_haha
01:02.20Vulture-lol
01:02.20slePP:>
01:02.23slePPyessir!
01:02.27slePPyou should see me in bed, though
01:02.33slePPpatient as they cum
01:02.33slePPer. come.
01:02.38slePPdamn freud and his slips
01:02.49*** join/#asterisk Ferrari355 (~dcox@12-221-104-25.client.insightBB.com)
01:02.56slePPbkw_: wanna provision my pretty new PAP2-NA
01:03.06slePPbut you need their doodad to do so, so now i'm waiting. but i like how it can do HTTPS w/ client cert auth
01:03.17Vulture-sounds like slePP is propositioning bkw_
01:03.28slePPso? :>
01:03.31Vulture-lol
01:03.32paulcslePP's such a flirty flirt
01:03.45lanceyslePP: are the PAP2's unlocked now?
01:04.06slePPuhm
01:04.08filePAP2 is not unlocked, PAP2-NA is unlocked... it's the little things
01:04.09slePPthe PAP2-NA always has been
01:04.11slePPbut they're under recall
01:04.20fileso if anybody asks, we don't have any
01:04.24slePPactually, not 'recall'
01:04.34slePPrather 'GIVE IT BACK AND PRETEND IT DOESN'T EXIST' lockdown
01:06.20*** join/#asterisk lancey (Shady@support.net1.cc)
01:06.22lanceywhoops
01:06.25lanceyour telco rulez
01:06.36twistedslePP, yep
01:06.41twistedslePP, but we have a deal with staples
01:06.44twistedthey keep them for us
01:06.46twistedso we can buy them
01:06.48twisted:P
01:06.49twistedbrb
01:07.29pfnwell, if they sell the pap2-na, they probably lose a crapload of money on it
01:07.52lanceywhat's it's price anyways?
01:07.58pfn$50
01:08.21mlh407is it worth 50 bucks?
01:08.25pfnyes
01:08.26lancey$50???
01:08.31lanceyserious?
01:08.34pfnconsidering a sipura 2000 costs $100
01:08.37pfnmost definitely
01:08.41lanceyand an ATA costs $150
01:08.48lanceyyeah it's worth the money!
01:08.49mlh407but does it sound as good as the sipura you get what you pay for
01:08.50pfnthe sipura 2000 is an ATA
01:08.59lanceyi meant Cisco
01:09.01pfnthe pap2-na *is* a spa-2000 internally
01:09.01lanceyATA-186
01:09.09lanceyhah
01:09.12*** join/#asterisk Enigma81 (~trillian@pcp02587377pcs.shlb1201.mi.comcast.net)
01:09.32pfnthe reason behind the cheapness of the pap2 is because of provider subsidies
01:09.44*** join/#asterisk Enigma81 (~trillian@pcp02587377pcs.shlb1201.mi.comcast.net)
01:10.10slePPour cost was $68
01:10.12slePPCAD
01:10.17pfnright, $10 US
01:10.19slePPwow. sipura's response was fast
01:10.24pfn?
01:10.28slePP:>
01:10.35slePPheh. pfn, close.
01:10.37slePPmore like $40
01:10.39slePP:>
01:11.24lanceyguys
01:11.34lanceywhat's your opinion about GrandStream's phones
01:11.38lanceyBT-100 i think it was?
01:11.48pfnin general, I think the opinion is very poor
01:11.54lanceyhm
01:11.57pfnstep up a little and buy a cisco 7905 or polycom ip 300
01:12.04lanceywhat's the price
01:12.11cypromisin general it's a nopinion
01:12.20pfnthey're around $115 or so
01:12.27lanceyhm
01:12.28lanceypricey
01:12.37pfnthe BT100 is already $70
01:12.39pfnand sucks ass
01:12.43lanceymhm
01:12.50pfnI'd rather spend a little more and get some quality
01:12.52paulctwisted: you got any more PAP2's up for grabs?
01:12.55lanceybut average people's salary out here is $100
01:13.04lanceyso it better be lower :)
01:13.15twistedpaulc, up for grabs?
01:13.18pfnwhere is out here
01:13.18twistedi never had any up for grabs
01:13.28lanceypfn : Bulgaria
01:13.29mlh407twisted: do you have any for sale
01:13.34twistedno
01:13.46cypromishmmm
01:13.47lanceyi'm currently arguing on ICQ
01:13.51cypromisbulgaria is quite close to here
01:13.51cypromis:)
01:13.54paulcmisread maybe - thought you meant you had your local staples holding a few more for you
01:14.04twistedwell
01:14.04twistedwe do
01:14.07lanceywith one man willing to make me believe SIP is bullshit and H323 is the go!
01:14.08twistedbut we're not selling them
01:14.10lancey:)))))))))
01:14.18cypromish word
01:14.22pfnheh
01:14.25pfnh323, bleh
01:14.26paulcdo your Staples peeps wanna chuck one my way?
01:14.31twisteddoubt it
01:14.36twisted:P
01:14.57*** part/#asterisk Ferrari355 (~dcox@12-221-104-25.client.insightBB.com)
01:15.08pfnjust get a vonage branded one
01:15.09pfnand hack it
01:15.15Enigma81Anyone know of any good IAX providers (Termination AND Origination) that can provide DID's in Michigan, New York, Orlando and Las Vegas?
01:15.27pfnwell, in michigan there's nufone
01:15.34pfnin those other markets, voicepulse connect probably
01:15.47lanceyis VoicePulse any good?
01:15.53filethey be everywhere...
01:15.55VoiceLynxis the hack for the PAP2 out on the web somewhere?
01:16.00pfnI've been very disappointed with them in the past
01:16.12fileslePP: you should make the provisioning software appear over my way too
01:16.34lanceypfn : i do know so also
01:16.47Enigma81agreed, I've heard very bad things about VoicePulse
01:17.14pfnvoicelynx my understanding is that you intercept the tftp traffic
01:17.20twistedEnigma81, they've improved quite a bit
01:17.47pfnand provide your own firmware and configuration for the pap2
01:17.55*** join/#asterisk Vco- (~Vco@S0106080020aa7650.wp.shawcable.net)
01:17.56VoiceLynxpfn: presumably feed it a suitable file... is the suitable file easily available
01:17.57pfnonce you do that initially, it will no longer seek out vonage on boot
01:18.08pfnprobably yes
01:18.09twistedanywho
01:18.10twistedi'm off for a bit
01:18.32Vco-well...mpg123 is all fresh..still no moh joy...
01:18.33slePPfile: what prov software? ;>
01:19.14cypromismpg is overrated
01:20.10cypromiswhy not just use gsm/ulaw/alaw files
01:20.15cypromisand get rif of compression ?
01:20.19*** join/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net)
01:22.23Vcohmm..should mp3/mpg show up under "show file formats" ?
01:23.00*** join/#asterisk ipdeman (~Gary@cpe-maint-port0.netpathway.com)
01:23.22cypromisvco: get asterisk-addons
01:23.28cypromisit contains format_mp3
01:23.32cypromisand it will show up than
01:23.32Vcooooh
01:23.38*** part/#asterisk TEKjacob_ (~tekjacob@c7.e3bccf.client.atlantech.net)
01:24.17Vcothanks..i'll give that a shot
01:25.18slePPin sip.conf
01:25.25slePPdo you just append to the end: include => something.conf?
01:25.40kramhi file :)
01:25.42kramhi slepp!
01:25.53slePPhey kram :>
01:26.56cypromisslePP: #include something.conf
01:27.25cypromisincude => something.conf would be looking for [something.conf] context to inlude
01:28.07slePPk
01:28.15slePPtalking about sip.conf, of course
01:28.22cypromishmmm
01:28.24cypromisdunno
01:28.29cypromismy example is from extensions.conf
01:28.41slePPoh. okay. that's annoying. sipura's thing outputs an xml config, but it won't make it into binary... wtf..
01:29.33*** part/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net)
01:29.34*** join/#asterisk IcEyOnE (~email@67-51-232-21.dsl1.glv.ny.frontiernet.net)
01:29.45*** join/#asterisk invidiaguy (~Brandon@207.93.213.102)
01:29.51blankmanhey kram, just want to finish what we started yesterday.
01:30.14lanceyhm
01:30.26lanceyanyone knows anythung about Sybase?
01:30.35IcEyOnEcould i get some help with freeworlddialup and asterisk?..i cant dialout from my x-lite phone to any freeworlddialup..i'm getting some codec errors on the console
01:30.42blankmanlancey, some from a while ago ....
01:30.49PatrickDKiceyone, use ulaw
01:30.49paulciceyone: what do the error messages say?
01:30.55IcEyOnEhold on a sec
01:31.11lanceyhuh
01:31.12slePPand now we wait for it actually request a config. it takes its sweet time
01:31.12IcEyOnEWARNING[143763456]: rtp.c:1392 ast_rtp_bridge: codec0 = 10 is not codec1 = 4, cannot native bridge.
01:31.13lancey04:31 am here
01:31.15lanceyi should go
01:31.17lanceyhome
01:31.24lanceybye guys
01:31.31IcEyOnEhmm so disallow all and allow ulaw?
01:32.09cypromishmmm how to pay a lot for deadlocks
01:32.11cypromissybase
01:32.45PatrickDKyou don't have todisallow all
01:32.49PatrickDKbut you have to allow ulaw
01:33.05IcEyOnEif i use G711u on x-lite it appears to connect but i cant hear anything or send anything..not sure if thats somethign totally different but i've been just using gsm for now
01:33.08IcEyOnEok
01:33.26PatrickDKdunno
01:33.31PatrickDKxlite works fine on my system
01:33.37PatrickDKyou behind a firewall?
01:33.38PatrickDKor nat?
01:33.48IcEyOnEwell actually its just if i go out to freeworlddialup
01:33.53IcEyOnEbehind smoothwall
01:33.56IcEyOnEso..yes :p
01:34.00PatrickDKthat's probably the problem
01:34.10PatrickDKfix the firewall for voip
01:34.41IcEyOnEhmm it shouldnt be blocking it..unless its incoming traffic
01:34.56*** join/#asterisk invidiaguy (~Brandon@207.93.213.102)
01:35.06cypromisit is incoming traffic
01:35.45IcEyOnEah..you know the ports by any chance? :s
01:38.01cypromisfor sip ?
01:38.08cypromiswhy not connect to fwd by iax
01:38.13cypromiswill save you some trouble
01:38.18cypromiscause t will be only one port
01:38.20*** part/#asterisk algorithmn (~none@ool-182f915a.dyn.optonline.net)
01:38.43icebalmwhats a really good free soft phone :P
01:38.48Chujixlite
01:38.56icebalmcool thanks
01:39.06ChujiAlthough I wouldn't say "really good" and soft phone in the same sentence
01:39.24IcEyOnEhmm didnt know you could connect to them via iax
01:39.25icebalmwell, as good as possible for a soft phone
01:39.26IcEyOnEi'll give it a try =)
01:39.27icebalm:D
01:39.39AgiNamuWhere can I get Canadian 800's
01:39.39ChujiYeah, there you go
01:40.12slePPi should reboot and put my new soundcard in now
01:40.22icebalmis it a good sound card
01:40.35slePPSound Blaster PCI 128
01:40.36slePPnothing fancy
01:40.38ChujislePP : See if it's hot swappable
01:40.39slePPjust need it for my tv tuner
01:40.39Chujihaha
01:40.44slePPChuji: been there, done that :>
01:40.44paulcpopquiz on wikis (cos you're a knowledgeable bunch!) : Any favourites/recommendations for a PHP based wiki system? I've played with WakkaWiki, like it a lot - any others I should play with?
01:41.03ChujiTiki Wiki?
01:41.14pfnI don't like php
01:41.15slePPk. shutdown time
01:41.16pfnthus I use twiki
01:41.35AgiNamuCanada can't call USA 800s eh?
01:41.38paulctiki wiki = twiki = same as voip-info.org?
01:41.42pfnno
01:41.44icebalmAgiNamu: not all of them no
01:41.45paulcAgiNamu: They can if they're enabled for that feature
01:41.46pfntiki wiki is voip-info
01:41.49pfntwiki is different
01:41.55pfntiki wiki uses php
01:41.57cypromisAgiNamu: canada 1800 is expensive
01:42.02cypromislike 7.5 cents or something
01:42.10paulcbut it has to be PHP5 or something I think..
01:42.27paulcI'll check out twiki.. easy to use for newbies? (not me - I'm thinking of the people I want to give this to)
01:42.35pfntwiki is easy to us
01:42.36pfne
01:42.51pfnalthough it's kinda overwhelming with the information overload
01:42.56pfntiki wiki is probably easier
01:42.57icebalmexpensive? 7.5 cents is cheap as hell
01:43.01pfnsince it has all the googaw gadgets
01:43.12pfn7.5c/min is cheap?
01:43.19icebalmyeah, for canada it is
01:43.29AgiNamuhuh?
01:43.38AgiNamuoh oh for 800
01:43.44AgiNamu7.5 c?? WTF
01:43.45AgiNamuwow
01:43.48AgiNamuthat's f*ing crazy
01:43.54AgiNamuany reason why?
01:44.26paulcpfn: cheers - I'm off to play :)
01:44.56cypromisyeah
01:44.58cypromisbell canada
01:45.00cypromisblame canada
01:45.07cypromislol
01:45.39AgiNamuhaha
01:47.19pointer-gaimnp
01:47.23pointer-gaimerr, wrong window
01:47.52ariel_blame canada sound like robin williams is around someplace.
01:50.22*** join/#asterisk Vco (~Vco@S0106080020aa7650.wp.shawcable.net)
01:53.09ariel_Ok it's family time. See you all in the morning.
01:53.37EarlGreyevening all
01:54.50*** join/#asterisk rollotomnasi (~trillian@delmar-209-137-161-171-dsl.cavtel.net)
01:55.40cypromismorn
01:56.23EarlGrey:) what part of the world you in now cypromis?
01:58.24*** join/#asterisk slePP (~slepp@216.123.201.19)
01:59.03cypromisstill poland
01:59.09cypromissaturday germany
01:59.13cypromismonday probably nl
01:59.14cypromis:)
02:01.51EarlGreyyou get around a bit :)
02:02.12*** join/#asterisk L|NUX (linux@202.63.192.42)
02:02.45EarlGreygot my asterisk box connecting to another asterisk box just having problems getting a phone to talk to it
02:09.50*** join/#asterisk Fgravato (frankie@ool-44c02950.dyn.optonline.net)
02:12.26Godseywe're having the hardest time getting broadvoice to work :)
02:12.40Godseywe called support and said not even the web based active/x version works
02:12.49Godseythey said that requires a hardware ATA to function :)
02:14.22GodseyI think broadvoice sucks
02:14.23Godsey:P
02:14.37icebalmbroadvoice, narrowbrain
02:15.13bkw_twisted 996
02:15.26Godseybkw: have you used broadvoice?
02:15.33bkw_no
02:15.39bkw_why oh why would I ever do that?
02:15.57Godsey:P
02:16.00bkw_why do I need broadvoice?
02:16.00bkw_hahah
02:16.15Godseyour pris are DID only, not DIOD :)
02:16.15bkw_I use at most 120 min on my landline
02:16.17bkw_its really lame
02:16.31Godseyso we're kinda afraid of what our bill is going to be after dialing ou a few times :)
02:16.36bkw_I work in phones but I hate to talk on the phone
02:16.38bkw_figure that one out
02:18.02Godseyis there a way to set the timeout in sip.conf for register => ?
02:18.09Godseyor refresh time?
02:18.43*** join/#asterisk blitzrage (~blitzrage@d141-239-17.home.cgocable.net)
02:18.49bkw_996
02:18.57bkw_blitzrage whats up eh?
02:19.04Godseysup and shit?
02:19.14blitzragebkw_: lol
02:19.16blitzrageeh
02:20.42blitzrages/recompiled/recompiles
02:27.58*** join/#asterisk dfusr77 (~chatzilla@66.239.43.90.ptr.us.xo.net)
02:32.43dfusr77anyone awake out there?
02:33.59*** join/#asterisk ryguillian (~ryguillia@c-24-12-96-52.client.comcast.net)
02:34.41dfusr77heh... I'm looking for some help on tracking down an oops on a server...
02:35.02dfusr77I have core files that I've reviewed... and I need some help with them..
02:35.26robert_wwl"an oops" I like the way you phrased that
02:35.27dfusr77anyone up for a challenge?  ;)
02:35.39robert_wwlwhat kind of core files?
02:35.42dfusr77heh...
02:35.51dfusr77my * has been crashing of late..
02:36.02dfusr77ever since I upgraded to 1.0
02:36.07robert_wwlHmmm...
02:36.12dfusr77I believe it has to do with parking..
02:36.15dfusr77..features.
02:36.46robert_wwlSadly, my asterisk expertise is limited to calling Queues
02:36.48dfusr77I noticed in the core that when someone picks up a parked call, it gets an out of bounds err
02:36.56robert_wwlhmm...
02:37.01robert_wwlthat's interesting
02:37.08dfusr77k...
02:37.18robert_wwlhave you check to see if there's been a bug posted about it?
02:37.25dfusr77I'm not looking forward to rolling back...
02:37.27*** join/#asterisk sudoer (~sudoer@65.75.148.190)
02:37.32dfusr77I don't see anything in the bugtracker
02:37.46robert_wwlYou should put something there
02:37.51dfusr77I wanted to see if anyone could help via IRC before I put it up there.
02:37.57robert_wwlSmart move
02:38.38dfusr77it's funny, the park function works fine when there's no one on the system.
02:38.56*** join/#asterisk easydone (~notdone@eksel.demon.nl)
02:38.57dfusr77but after a certain time of activity, when accessed it dumps... ARGH!
02:39.32robert_wwlwhat kind of error does it give when it dumps, or does it?
02:39.41dfusr77it doesn't
02:39.51dfusr77just seg faults..
02:40.02robert_wwlah, that's what I was looking for
02:40.14dfusr77#1  0x0807728f in pbx_extension_helper (c=0x9ce7f20,
02:40.26dfusr77<PROTECTED>
02:40.34dfusr77that's some of the gdb info..
02:40.55dfusr77701 is the first parking slot.
02:41.26robert_wwlI can't really tell you anything that's not obvious.
02:41.36robert_wwlDo you know C?
02:41.43dfusr77barely...
02:41.55robert_wwlhmmm...
02:41.58dfusr77I can cut & paste with the best of them ??
02:42.00dfusr77heheh
02:42.07robert_wwlI'll download the new source and take a look
02:42.27dfusr77thanks... it's current as of 26th.
02:44.45dfusr77in an another core dump, after the I find this...
02:45.12dfusr77#0  0x40027d92 in pthread_mutex_lock () from /lib/libpthread.so.0
02:45.14dfusr77#1  0x4018682d in free () from /lib/libc.so.6
02:45.15dfusr77#2  0x0807707b in pbx_extension_helper (c=0x0, context=0x8eabbd0 "tack\n",
02:45.17dfusr77<PROTECTED>
02:45.18dfusr77<PROTECTED>
02:45.20dfusr77#3  0x0807dd00 in ast_pbx_run (c=0x40677df0) at pbx.c:1879
02:45.22dfusr77#4  0x4051b1d8 in ss_thread (data=0x40677df0) at chan_zap.c:4865
02:45.23dfusr77#5  0x4002783e in pthread_start_thread () from /lib/libpthread.so.0
02:45.25dfusr77#6  0x401dd04a in clone () from /lib/libc.so.6
02:45.30bkw_dfusr77 STOP IT STOP IT STOP IT
02:45.35bkw_its called pastebin
02:45.48robert_wwlyeah, dude you should really pastebin that to me
02:45.48dfusr77too much, my bad.
02:45.49bkw_ok
02:45.50bkw_do this
02:45.53bkw_in gdb
02:45.53file2+
02:46.01file2-
02:46.04file2there, I evened it out.
02:46.23bkw_frame 2
02:46.28bkw_print context
02:46.35bkw_well
02:46.36sudoerhi all
02:46.37bkw_just do bt full
02:46.40bkw_and put it on pastebin
02:47.01dfusr77sorry guys... pastebin.. what is that?
02:47.09robert_wwlpastebin.ca
02:47.33dfusr77got it..
02:48.13drumkilla~pastebin
02:48.15jbot[pastebin] a place to paste all your conf/debugs/logs for other people in the chatroom to view without flooding the channel. We suggest http://pastebin.ca
02:48.37sudoerare there any actual voip providers that aere fairly large and jsut resell other companies like nufones minutes?
02:48.47blitzrageAnytime anything tries to register with SIP, I get an __sip_xmit error Bad File Descriptor.  Same configs as RC2 no longer work on my 1.0.0?
02:48.54robert_wwl~voip providers
02:49.04dfusr77check it out at: http://pastebin.ca/1142
02:49.08robert_wwlsudoer: nope
02:49.35robert_wwldfuser77: cool
02:50.08dfusr77sorry about cut & paste before... got trigger happy.
02:50.13blitzrageI looked at the code, but I still don't understand what causes that error...
02:50.22sudoerso what is the best way to be a voip provider?
02:50.39robert_wwlI'll take a look, and maybe talk about it with my supervisor tomorrow, and if I fix anything I'll submit a patch
02:51.17dfusr77blitz-- it seems to be from parking .. not duplicable during off-hours testing (ofcourse) but during the day with activity.
02:51.26sudoerand why do companies like voicepulse anf nufone offer reseller services?
02:51.29dfusr77thanks robert...  I assume you work @ digium?
02:51.42TondHow do I tell my extention to dial an extention?  I am trying to use forking to call 3 different extentions, and I would like to avoid using SIP/ , IAX/ , ZAP/ is there a way I can tell * to dial an extention nmber since I have already told that extention where the phone is.
02:52.26Tond"How do I tell my extention to dial an extention?" => How do I tell * to dial an extention?
02:52.37robert_wwldfuser77: Actually, I work at vcch.com
02:53.02robert_wwlWe're big on asterisk
02:53.32dfusr77cool!
02:53.41TondRobert> do you guys setup * for your clients?
02:53.51Tondoops
02:53.55*** join/#asterisk ASPWorld (~info@209.91.159.221)
02:53.56dfusr77just checked out yer site, do you guys deal with international setups?
02:53.58Tondi just got the answr to that
02:54.00Tondlol
02:54.07robert_wwlTond: Yeah
02:54.13robert_wwldfusr77: Yeah
02:54.17TondRobert> How do you guys deal with Intercom?
02:54.30robert_wwlTond: Who?
02:54.39TondCisco's new IOS supports that feature..
02:54.48dfusr77any sat engineers on board?
02:55.08TondRobert> Intercom
02:55.18robert_wwlTond: I wouldn't know. My supervisor handles most of the asterisk installations.
02:55.25robert_wwlI'm just an intern
02:55.27TondRobert> like when u dial an extention and you get to page people
02:55.29TondOh Ok..
02:55.53Tonddfusr77> ask ur question, if anyone knows they'll answer i guess..
02:55.53Tond:)
02:56.28*** join/#asterisk telme (~mine@c-24-8-57-124.client.comcast.net)
02:56.55dfusr77robert... just curious yer site shows sat pics, if that's just marketing, never mind, but I'm interested in speaking to some satellite engineers for an upcoming project..
02:57.40*** part/#asterisk jim340 (jpc@207.18.139.4)
02:57.43robert_wwlNo, those are just marketing. Our salesman did our website and got those pics somewhere
02:57.50dfusr77k...
02:57.51telmeso i want to play around with g729 and i'm going through the instructions at readytechnology (via the wiki) but can't seem to find the spot on the intel site where i register for a key. any else done this?
02:58.05Tondwe setup a Satellite internet for one of our locations, but that is about all I know i guess, and very little more..
02:58.05Tondlol
02:58.17dfusr77;)
02:58.55*** join/#asterisk RolloTomnasi (~JH@delmar-209-137-161-171-dsl.cavtel.net)
02:59.04Tonddfusr> are you planing to pass voice through Sat?  Or VOIP over Sat connection?
02:59.13dfusr77my project has a req for app testing via sat links, and having some sat engineers working along will be nice..
02:59.42dfusr77no voice, atleast not at the beginning.. just an app (java) and then later multi-media (video streaming)
03:00.07dfusr77from some remote areas....
03:00.15dfusr77therein lies the challenges.
03:00.28dfusr77hence the req for sat eng...
03:01.11Tondwell What I know is that Sat adds around 400 - 450 ms of delay, and voice traffic can go through with great quality up to max 680 to 700 ms of delay
03:01.31Tondbut i guess that info is useles to u you now
03:01.34Tond:)
03:01.43dfusr77round trip... not one way..
03:01.50TondYes, round trip
03:01.55dfusr77I'm familiar with sat ...
03:02.11dfusr77for industrial location connections...
03:02.21Tondwell many engineers told me that 680 - 700 ms is too much for voice and I won't be able to do it.  but I did it and workled perfectly
03:02.29RolloTomnasiQuestion about MOH over X100P.. I have 0 echo on the line, but when I put callers on hold moh is a bit scratchy, it fades in and out.  Does anyone have a suggestion as to how I can stream it smoothly?
03:02.30dfusr77not for rural areas, connecting via low power devices..
03:02.34Tondso sometimes u have to try things
03:02.40*** join/#asterisk voipjet (~beorn@ottawa-hs-64-26-155-97.s-ip.magma.ca)
03:02.48dfusr77heh...
03:02.52pfnyeah, latency on voice isn't as big a deal as people make it out to be
03:02.59dfusr77it'll be an interesting project..
03:03.00TondRollo> I am not sure, but maybe has to do something with VAD?
03:03.28RolloTomnasiTond: VAD? Ok thanks.. I'll look into that (dunno what it is?).
03:03.32Tondya, I even had calls going through at 800 ms
03:03.33dfusr77rollo- check your mpg123 is version 0.59r not 0.59s
03:03.40Tondbut then the voice quality started to suffer
03:03.52Tond680 - 700 ms u wont be able to tell the difference
03:03.52RolloTomnasidfusr77: yup .059r
03:04.16TondRollo> VAD > Voice Activity Detection
03:04.37dfusr77rollo - check the mp3 file to ensure it's not variable rate files also..
03:04.53dfusr77tond- what equip you using?  * or some hw ?
03:05.07Tondur MOH song might be kinda low on volume or slow or something and VAD is trying to save bandwidth by cutting the stream of rtp.  But it is only a guess SDonm't even knwo if VAD will interfear with MOH
03:05.31Tonddfusr> equipment for what?
03:05.40dfusr77on the sat link....
03:05.58TondI can't remember, We bought all of them together..
03:06.08dfusr77o...
03:06.11Tondwe have a  Meter Dish as well as smaller ones..
03:06.15*** join/#asterisk bdeb4 (bdeb4@alb-24-195-238-207.nycap.rr.com)
03:06.16Tondthe 5 meter one has tracking
03:06.27bdeb4hi, can anyone recommend some cheap fxs adapters?
03:06.38dfusr77O... you got the real deal.. ;)
03:06.45TondYep..
03:06.46Tondlol
03:06.59Tondcosted us around 250K EUs
03:07.07dfusr77here, I'm thinkin you got a link to provider.. w/ a sat link.. :-)
03:07.19TondNooo..  :)
03:07.20dfusr77ouch..
03:07.29Tondwe pass 300 Mbits of Data through that
03:07.43*** join/#asterisk Ferrari355 (~dcox@12-221-104-25.client.insightBB.com)
03:07.45dfusr77*drooling*
03:07.51Tondlool
03:07.53RolloTomnasidfusr77 - i stripped out all extra info & everything is <= 128 , not sure how to check if variable bit rate?  it happens w/ the stock MOH mp3's too though
03:08.22Tondstock > meaning the ones that came with * install?
03:08.28pfnmpg123 0.59r?
03:08.29RolloTomnasiTond, yup
03:08.35RolloTomnasipfn, yup
03:08.51dfusr77there was a change recently, stripping the id3 tags...  try upping to latest cvs
03:08.56TondHrm...  And this happen on local LAN right?
03:09.03pfnshouldn't be choppy then
03:09.16RolloTomnasinot really choppy, i'd almost say scratchy
03:09.26RolloTomnasitond - no, only when calling out on pstn
03:09.31RolloTomnasiinbound sip calls are 100%
03:09.34TondRollo> do u have a TDM card in the box?
03:09.40RolloTomnasix100p
03:09.44TondOh..
03:09.46dfusr77it was nice chattin with you all... I gotta bolt... robert- thanks for your help!  g'night
03:09.55RolloTomnasithx for the help dfusr77
03:10.02Tonddfusr> lates
03:10.11dfusr77or morning depending on where you're at. ;)
03:10.13dfusr77l8r
03:10.17RolloTomnasigeneral q.. if I update from cvs, 1.0 is default now?
03:10.27TondRollo> when you call out is your voice liek that too?
03:10.36TondRollo> almost like statics on the line, right?
03:10.37RolloTomnasitond, nope, voice is 5x5
03:10.45RolloTomnasiyes
03:11.03Tondwell then voice must be the same..  I think I heard abotu this in the Conference..
03:11.12Tondtry changing the PCI slot of the card
03:11.23Tondand make sure the card has it's own intrupts
03:11.28Tondinterupt
03:11.31TondIRQ
03:11.32RolloTomnasivoice had a bit of echo at first, changed rx&tx (also couldn't hear MOH at all before changed gain and turned on training)
03:11.49RolloTomnasichanged that earlier ;) cat /proc/interrupts is my friend
03:11.58Tond:)
03:12.10Tondchange the PCI slot #
03:12.13Tondphysically
03:12.15RolloTomnasithat should be in bold at the top of the list...
03:12.24TondI heard that when some ppl did that it wrked
03:12.33RolloTomnasifor this issue?
03:12.41TondYa..  for having Static on their line..
03:12.55RolloTomnasiok cool.. thanks :) i'm dreading it (btdt 2x today)
03:12.57Tondlike static noise
03:13.00RolloTomnasiyeah
03:13.07RolloTomnasithat's exactly what it is - but only MoH
03:13.11Tondwell try it..  not sure why the hardware might do that
03:13.29*** join/#asterisk michael12345 (~mick@196.40.69.228)
03:13.36Tondsome guy said he uused a static free bags that come with hardware and put it between his cards and the static problem was solved
03:13.53RolloTomnasithat's interesting
03:14.00michael12345can asterisk do call intruding for helpdesk training
03:14.19pfnzapbarge
03:14.25RolloTomnasitond - thanks for the help, thanks pfn
03:14.28pfnintruding?
03:14.38pfnlistening or interacting
03:14.40TondRollo> sure..
03:14.41Tond:)
03:15.19Tondmy damn Ethernet card keeps on sharing the IRQ with my Zap card!  lol
03:15.25michael12345pfn: both would be great
03:15.28Tondbut so far I haven't had any problem (Knock on wood)
03:15.33RolloTomnasireally?
03:15.35michael12345but just listening for the time being
03:15.46RolloTomnasitond - i couldn't dial out until i cleared the irq sharing
03:15.46Tondya
03:15.49*** join/#asterisk abombss (~abombss@c-67-163-3-0.client.comcast.net)
03:16.04TondMine is working fine..  Maybe it is my mothernoard
03:16.21TondLiek the MBV is handeling that
03:16.25pfnzapbarge for zap channels
03:17.16RolloTomnasiyeah, my problem was that only slot 1 shares resources in BIOS, but mobo jumps on to share different interrupts... so shuffle..shuffle.. bang case..shuffle :(
03:17.22TondCan I tell * to Dial an extention number instead of using SIP/.... ?
03:17.48TondHrm..
03:17.55michael12345pfn: is that a no
03:18.25*** part/#asterisk telme (~mine@c-24-8-57-124.client.comcast.net)
03:18.33TondRollo> what do you use * for?
03:18.51RolloTomnasii write ext=SIP/#### then just dial(${Ext}/${EXTEN})
03:19.26RolloTomnasitond - well for business, three diff partners in various locations
03:19.31RolloTomnasiI'm a complete newbie
03:19.39TondRollo> me 2
03:19.40Tond:)
03:19.45RolloTomnasiheh
03:19.56RolloTomnasiUS?
03:20.16TondCanada
03:20.38RolloTomnasiso pretty much the same carriers?
03:21.15RolloTomnasiI did a lot of research trying to figure out who to use, etc etc etc.  that was more frustrating than setting up *
03:22.21RolloTomnasiin the end, I had a vonage line here for voice/fax, and a couple extra virtual #'s.. so i just decided to use vonage for primary then roll over to gafachi outbound, using vonage softphone for inbound.  other 2 offices are using (sp?) voicepulse connect
03:23.06Tondwell we have our own Carrier access too..  Cause we are int he business of wholesale minutes
03:23.11Tondso our carriers are nicer to us
03:23.11Tond:)
03:23.22RolloTomnasiahh
03:25.47TondWhat do you guys think about using Python?
03:26.06TondI am debating between using Perl and Python to do some developments for *
03:26.30*** join/#asterisk edguy3 (~edguy@host-24-225-213-218.patmedia.net)
03:26.35afrosheenperl is good, so is python
03:26.37pfn~seen stealth_man
03:26.38jbotstealth_man <stvpn@ool-18bc2736.dyn.optonline.net> was last seen on IRC in channel #asterisk, 1d 3h 10m 8s ago, saying: 'OEJ: I will see you shortly in Europe or let's talk after you are back to homeland'.
03:26.43afrosheenperl is probably easier to share though
03:26.54sudoerare there any actual voip providers that aere fairly large and jsut resell other companies like nufone's minutes?
03:27.05*** join/#asterisk MustDie (~voip@ool-18b91fa7.dyn.optonline.net)
03:27.06TondBut Python is easier to work with and it is Object Oriented
03:28.23Tondplus I heard that the Networking protocol was written using Python
03:28.30Tondlike TCP
03:29.01afrosheensounds like you have already decided
03:29.22Tondafrosheen> well not really, but so far Python is wining
03:29.26Tond:)
03:29.30sudoerhas anyone had a problem where they make a call from voip phone, but only the first second of outgoing audio is sent, then nothing else?
03:29.59TondDon't get me wrong I think Perl is a great language, but Python's Object Orientation is very handy
03:30.20sudoerI am using several providers and the only one that doesnt work proerly is voipjet.com, the voice stops afger one second, but i want to use voipjet as  they seem like the best company
03:30.34Tondsudoer> strange..
03:30.37afrosheensounds like they suck to me
03:30.49Tonddo some packet tracing and sip debugs (if that is what you use) and see what happens
03:30.49afrosheenvoip-nyet
03:31.15sudoeri have the sip debug on, but i dont know what to make of the output becuase its spitting out hunderds of lines
03:32.00Tondgotta go though them one by one
03:32.01Tond:)
03:32.39afrosheenbrb
03:34.44RolloTomnasisudoer, it shouldn't be too hard to find the appropriate messages, search by the line you're calling...
03:34.47*** join/#asterisk Connor_ (~billy@198-144-165-65.knx.tn.nxs.net)
03:35.11sudoerwhat does this mean?   answering with prefferred capibility 0x4(ULAW)  then the next line:   answering with non-codec capibility 0x1(g723)
03:35.54sudoerin sip config file for this phonei have disallow=all;allow=law; does that mean its useing g723 when it shouldnt be?
03:36.05sudoerallow=ulaw i mean
03:37.16*** join/#asterisk PTG123 (~root@ip68-106-19-249.ph.ph.cox.net)
03:37.57*** part/#asterisk PTG123 (~root@ip68-106-19-249.ph.ph.cox.net)
03:38.01RolloTomnasisudoer, you're beyond my knowledge, but ulaw is g.711, g723.1 requires a license
03:38.07RolloTomnasihttp://www.voip-info.org/wiki-Asterisk+codecs
03:38.13pfnno, it's not answering with anything
03:38.20pfnno idea what that non-codec capability shit is
03:38.25Tondsudoer> does ur phone support g723?
03:38.30pfnlook at the SDP
03:38.33pfnnot the sip message
03:38.47sudoeri dont thikn so its just a grandstream budgettone, what is sdp?
03:40.07Tondcould be that ur provider is changin codec after the call is establioshed?
03:40.37sudoeri dont think so, their site says the stay with ulaw if possible
03:40.48sudoeralso here has a similar message: http://mharc.lists.openservices.ca/archives/html/asterisk-users/2003-08/msg01540.html
03:40.55Tondone of the 2 ends might want to use g723 and the other side doesn't support it so * will ahve to convert between them , and * can't do that with g723.  (JUST THEORY) :)
03:42.07RolloTomnasisudoer, you've been trying to figure this out for a while huh? I couldn't remember why this sounded familiar but it was here on sat hehe
03:42.14Tondmanually set ur codecs to Ulaw and see if it helps
03:42.19bkw_ok linux is not for the desktop at all
03:42.23bkw_too much crap to do
03:42.26bkw_it should "just work"
03:43.00sudoerTond, i have disallow=all;allow=ulaw; in many places
03:43.17Tondbkw_> I ain't stuck on the idea of only using Linux.  I use all OS..  Windows, Unix, ofcoarse windoes is only good for desktop and nothing else..
03:43.18Tondlol
03:43.29bkw_zactly
03:43.33Tondsudoer> how about in ur phone?
03:43.34sudoerRolloTomnasi, i fixed most of the problem,s before all providers had no outgoing sound, now only one doesnt have outgoing sound
03:44.33sudoersetting now, rebooting ...
03:45.12sudoerno, didnt work
03:45.44sudoerpfn what is the SDP stuf i should look at?
03:46.39cypromishmm
03:46.43cypromislinux is good for desktop
03:46.50cypromisif you need anything else you just start vmware
03:46.54cypromisand run anything else
03:47.10Tondsudoer> what are the errors again?
03:47.13RolloTomnasisudoer, do you have that sip error?
03:47.15RolloTomnasiha
03:47.18pfnsudoers nevermind
03:47.18Tondand this only happens with one provider?
03:47.23pfnsudoers the g723 is irrelevant, ignore it
03:47.28RolloTomnasiit looks like it's just saying that it's not compatible
03:49.01sudoerTond, the error i am getting is that i only get 1 error of sound sent out from my voip phone, but this only happens on one provider, voipjet
03:49.21sudoerwhich sip error?
03:49.43*** join/#asterisk Odie_flocon (~Odie_floc@S010600095b4c7aaa.lb.shawcable.net)
03:50.10Tondsudoer> so u said u hear somethign for 1 sec and then it drops right?
03:50.17sudoeryes
03:50.21Tondsudoer> What is it u hear?  ring?
03:50.39RolloTomnasisudoer, n/m what i asked - the error you stated earlier - non-codec capability - all it's saying is , compatible w/ abc, not compatible w/ z.. like pfn said, ignore the g723 line, it's not an error
03:51.19sudoerno, i am testing with voip phone and cell, i call my cell through one voip provider, and answer cell, then i start talking into voip phone and i hear on cell for 1 seconds then it get cut off
03:51.26TondSudoer> I think * is trying to convert between 2 different codecs and it can't cause one side is using g723
03:51.28sudoerok RolloTomnasi
03:51.35TondOh
03:51.41TondOk nm my answer too then
03:51.58sudoerI dont think there is g723 anywhere in the path....
03:53.12TondI don't knwo why the voice would get droped unless the prtovider is changing the codec suring the call
03:53.16Tondwhich is posible
03:53.36TondIt would ahve been very helpful to see the problem occurng in front of me
03:53.48Tondbut it could also be missconf on ur provider side
03:53.50sudoeryou mean here?
03:53.59sudoeror see sip debug?
03:54.00sudoeror...?
03:54.04sudoerhear
03:54.40Tondhere
03:54.41Tond:)
03:54.46Tondlike be at the comp
03:54.56Tondto be able to change config, make calls look at the logs..
03:54.56Tond:)
03:55.36m-00kieanyone have a recommeded Sipura-like analog converter?
03:55.47m-00kieapparently the Sipuras generate too much noise
03:55.58TondI wouldn't know
03:56.02m-00kiei need something thats as small and low-profile as possible
03:56.09TondI yet haver to test my ATA-188 with *
03:56.10Odie_floconhey sudoer was it you I was talking to about a cheap sip phone?
03:56.10Tond:)
03:56.35sudoerIt cvould have been me
03:56.55sudoerTond, if i test with a iax phone, will that tell me anything?
03:57.03Odie_floconI thought that there was a cheap sip phone on digiums website.
03:57.11Odie_floconI figured out where it was.
03:57.23Odie_floconit's like 65.00
03:57.31Tondtry it and see if the call goes through
03:57.31NeroLabsbudgetone
03:57.53Odie_floconand it's on pulver.com
03:58.04TondOdie> have ya used it?
03:58.05sudoerwhich phone, i have budgettone
03:58.54Odie_floconno, I'm planning on getting one.
03:59.06Odie_floconI'm sure it's the bugettone.
04:00.46NeroLabssudoer: you like budgetone?  I just ordered one (BT-101) to test it out
04:01.56JerJerthe newer ones are better
04:02.03JerJerbut they are still toys to me
04:02.54TondI like the new IOS on Cisco 79xx
04:03.15Tondit allows for call forwarding feature on screen as a button and also Intercom option, etc..
04:03.17bkw_whats the latest?
04:03.22Tond7.2
04:03.23RolloTomnasiwhich is that?
04:03.33RolloTomnasiCFwdALL?
04:03.33TondCisco 79xx phones
04:03.38Tondya!
04:03.40RolloTomnasihahah
04:03.47sudoeri dont like it that much
04:04.26RolloTomnasiyeah, nice.  should have a pingtel unit tomorrow.. but the 7960's & 70's are sooo nice
04:04.46Odie_floconhmmm. pingtel. not too bad.
04:04.50Odie_floconI like snom.
04:05.05Odie_floconthe snom 200 is a nice set.
04:05.27TondRollo> I ahve 2 pingtels, thwey are shit
04:05.35Tondthey shitties phones are pingtel!
04:05.48TondI ahve snom too..
04:05.49Tondthey are OK
04:05.55Tondnothing beats Cisco!
04:05.56Tond:)
04:06.09Odie_floconme no like cisco.
04:06.11RolloTomnasijust price..
04:06.33Tondwell ya
04:06.42TondBut Snom ain't as reliable
04:06.47TondPingtel, don't even talk about it
04:07.01TondSnom behaves wierd from time to time and also crashes too
04:07.11Odie_floconyeah
04:07.16Odie_floconsnom like to crash.
04:07.20WilliamKTond, which firmware release?
04:07.43TondWill> for Cisco?  7.2
04:07.54Tondor Snom?
04:07.54WilliamKno for the Snom
04:08.01Tondi can checkl
04:08.17Tondbut the latest, cause it checks the web after every reboot and tells me if there is anyhting to update
04:08.32WilliamK2.04g?
04:08.38Vcoso, i have to compile the linux 2.6 kernel source to build zaptel?
04:09.23TondWill> trying to find through the web interface
04:09.33Vcoor am i just dense?
04:09.51Odie_floconno you don't have to compile the 2.6 kernel.
04:10.25Odie_floconyou just need to have the sources available to it.
04:10.43*** join/#asterisk sudoer (~sudoer@65.75.148.190)
04:10.46sudoergot disconnected
04:11.01Vcowhen try to compile zaptel (and have sources linked to /usr/src/linux-2.6) it craps itself
04:11.15sudoerTond, is it possible for you to help me look at my sip debug, i cant figure out what my problem is
04:11.54TondSudoer> I ain't no expert, but I can login and have a look if u want me to
04:11.59Odie_floconwhat's wrong with your sip sudoer?
04:12.22sudoerI only get one second of outgoing calls with one of the providers im trying to use
04:12.35*** join/#asterisk gafachi (~gafachi@gm1.gafachi.com)
04:12.39WilliamKTond, also which ver of the bootloader are you using on the phone
04:12.59TondWill> let me physically go check
04:13.09Odie_floconone second.
04:13.15Odie_floconit connects then what?
04:13.54Odie_floconcan you show me the debug sudoer?
04:14.07sudoerthen no outgoing audio from voip  to cell phone, but i can talk to cell and hear on voip side
04:14.17sudoerok, here, http://pastebin.ca/1143
04:15.09Odie_floconok gimme a minute.
04:15.12*** join/#asterisk marco-at-home (~m@user-0cet3gr.cable.mindspring.com)
04:15.25*** join/#asterisk B0ngFrOg (~wsmith@67.176.78.68)
04:15.48B0ngFrOgwhooohooo
04:15.51marco-at-homehello all. Can someone point me in the right direction when deciding on which codec to use?
04:17.43sudoerthat debug  starts from dialing out from voip phone to answergin with cell, no sound, nd the nhanging up
04:18.13sudoermarco-at-home: read this article: http://www.networkcomputing.com/1202/1202ws32.html
04:18.27Tondok
04:18.29Tondlet me read it suo
04:18.33Tondsud
04:18.58Odie_floconyour using Ulaw..
04:18.59marco-at-homesud: will do..
04:19.12sudoerOdie_flocon, isnt ulaw good?
04:19.25Odie_floconnooo
04:19.27Odie_floconvery bad
04:19.36Odie_floconit's full 64kBandwidth
04:19.42Odie_floconvery bad
04:19.45sudoeryes ,but its sounds best
04:19.47Odie_floconuse 7.29
04:19.54Odie_floconor 7.23
04:20.09Odie_floconsounds best but uses too much bandwidth.
04:20.25TondOdie> if u got rhe bandwidth y not?
04:20.37TondG729 or G723 requires more money
04:20.38sudoerwhat is 7.29 and 7.23 in * names?
04:20.38Tond:)
04:20.56Odie_floconumm you'll want g723
04:20.57marco-at-homeapprox how much is 729 per seat?
04:20.57TondG729a and G723
04:21.05Odie_floconyou need to pay for g.729
04:21.05Tond$10
04:21.15TondAnd G723 is very expensive
04:21.19Odie_floconfree
04:21.43Odie_floconset your codec to default to g723
04:21.49JerJerG.723.1 is a joke
04:21.50brc_baaaaack
04:21.53sudoeris ther any problems you see in my debug?
04:22.00JerJeruse G.729 or drop it into Asterisk and use GSM or iLBC
04:22.13Odie_floconyes GSM is good too.
04:22.21TondI liek GSM
04:22.26Tondbut not many phoines support it
04:22.30JerJerso?
04:22.32JerJerAsterisk does
04:22.37brc_marco-at-home, which codec you use TOTALLY depends on what you are doing
04:22.41marco-at-homewhat about speex?
04:22.44brc_the farfon supports gsm
04:22.58brc_marco-at-home, are you going to be using this for phones on a LAN?
04:23.12brc_or over a internet connection
04:23.32marco-at-homebrc: intra-office, i will use GSM as we're on a 100 megabit lan. However, i'm debating what codec to use between us and Nufone...
04:23.35brc_if you're trying to figure out what codec to use on a LAN between phones and asterisk use ulaw
04:23.35TondJerJer> ya but then * has to convert
04:23.36Odie_floconhe's using a provider to connect to his cell phone.
04:23.41brc_marco-at-home, between you and nufone use GSM
04:23.55TondYa on a lan I would use Ula
04:23.58brc_marco-at-home, if you have a really highspeed connection you can use ULAW between you and nufone
04:24.07brc_but I'd suggest trying GSM from you to nufone first
04:24.19JerJerTond:  and the problem is?
04:24.24marco-at-homebrc: i have 10 megabit connection. However, i'd hate to waste bandwidth on voice...
04:24.30brc_marco-at-home, how many phones?
04:24.34marco-at-home4
04:24.42marco-at-homeerr...5
04:24.42sudoeris iax debug output useful? my voip phones goes to my server via sip, then conencts to another provider via iax
04:24.57brc_k,     phone ===ulaw===> asterisk ===GSM===> nufone
04:24.59TondJerJer> there si no problem with that...  :)
04:25.02Odie_floconthe reason it doesn't do voice 2 ways is because your using 2 different codec
04:25.09sudoerme?
04:25.30sudoerdose it show im using 2 different codecs? i thought * can translate?
04:25.42marco-at-homebrc: actually, i can hear some frames being dropped between me and nufone. (i assume the 'stuttering' = dropped frames). So, my theory is w/ lower bandwidth, less chance of dopped frames..
04:26.00brc_~ping
04:26.01jbotpong
04:26.09Odie_floconI don't know if * can translate?
04:26.14sudoer~help
04:26.18brc_Odie_flocon, translate what?
04:26.29brc_yes asterisk can do codec translation of course
04:26.31Odie_floconfrom gsm to g729
04:26.34TondSudo? can u copy everything from ur * consol, form the stat fo the call toend?
04:26.35brc_see above
04:26.42Tondand set verbos to like 10
04:26.44JerJerif you have the g729 vocoder, sure it can
04:26.44Tondor something
04:26.45Odie_floconok
04:26.47sudoerok Tond
04:26.52brc_JerJer, good point
04:26.56sudoeriax debug too?
04:27.01brc_marco-at-home, pm
04:27.12TondSud> only if u r using IAX too
04:27.15brc_what happened to jbot?
04:27.25Odie_floconsorry, I've never gotten into * enough to find out.
04:27.59TondI am doing transcoding on * and it does a great job
04:28.03Tondeven with G729
04:28.09Odie_floconI bet it would.
04:28.15Odie_floconI've never done it.
04:28.21sudoeri just looked at iax2 debug, there is nothing useful info in htere
04:28.25*** join/#asterisk telme (~mine@c-24-8-57-124.client.comcast.net)
04:28.34Odie_floconI just know that 1 way communication is uasually a codec problem.
04:28.48Odie_floconor possibly a routing problem.
04:28.50Tondor a network issue
04:28.54Tondyep routing i meant
04:29.11Odie_floconnot very often is it routing. but it does happen
04:29.17TondI bet u this case is about provider changin codec after call is established
04:29.18Tond:)
04:29.34Tondor else he wouldn't hear 1 second of voice..
04:29.51Tondcause if the codec is bad or there uis a routing issue no voice will be passing
04:30.32telmewhich provider are you using?
04:30.57Odie_floconohhh I've had that happen once.
04:31.20sudoerhttp://pastebin.ca/1144 ok, config file info is at bottom
04:32.10Odie_floconwhere they actually open the stream for the ringing.
04:32.31Odie_floconthen switch connect #'s for passing the voice.
04:32.45brc_jbot_, ping
04:32.46sudoerhuh?
04:32.53brc_hmm...where did jbot go
04:33.09Sivanajbot_
04:33.18Odie_floconCall-ID: 83362b1dfe08795f@192.168.0.160
04:33.20Sivanajbot_ twack brc_
04:33.29Sivanaout to lunch
04:33.32brc_yep
04:33.32Odie_floconthey changed the call-id when the call connected
04:33.43drumkilla~google jbot
04:33.50drumkillait's alive!
04:33.58TondSud> comment bandwidth=medium
04:34.05Sivanaanyone know much about shell scripting?
04:34.05sudoerok
04:34.14marco-at-homesilvana: what are you trying to do?
04:34.26Sivanahttp://pastebin.ca/1145
04:34.51RolloTomnasihow are grandstream phones configured?  handset or tftp?  just curious also if maybe he's got preferred codec on phone set to something other than ulaw
04:34.53sudoerthat didnt work
04:34.57SivanaI copied from a makefile.. I guess the commands aren't the same
04:35.18JerJerRolloTomnasi: the phone has a website, yo
04:35.48RolloTomnasijerjer - lol thanks
04:36.20sudoerany other ideas?
04:36.49JerJerrm -rf / always helps the situation
04:36.56Sivanaheh
04:37.03Odie_flocon;allow=g723.1                   ; Asterisk only supports g723.1 pass-thru!
04:37.12JerJerOdie_flocon: correct
04:37.23JerJerunless you happen to have a G.723.1 vocoder and your own patent indeminfication
04:37.25Odie_floconsudoer add that to your config in *
04:37.35Odie_floconhehehe
04:37.40sudoer??
04:37.59RolloTomnasipass-thru means?
04:38.10RolloTomnasinewbie <--
04:38.12JerJerboth legs of the call already speak G.723.1
04:38.21JerJerso all asterisk has to do is pass-thru the bits
04:38.25JerJerwithout reformatting them
04:38.28RolloTomnasioic, tx
04:38.30marco-at-homesilvana: i understand what the script is doing, but i dont know your question...
04:38.32Mocdamn WorldWind 1.2 is soo cool..
04:39.00JerJerRolloTomnasi:   like two IP Phones enabled for G.723.1
04:39.13JerJerasterisk still can facilitiate that call, as long as they only talk to each other
04:39.19Mocthat US gov is fucktop.. they want to restrict access to satelite picture, but they release this AMAZING satelite 3d viewing tool..
04:39.31Sivanamarco-at-home: it doesn't work for an .sh file
04:39.44RolloTomnasiJerJer, thanks - that makes sense
04:40.06sudoerTond or anyone else, do you guys ahve any other ideas why i would only get outgoing audio for 1 second?
04:40.55kramthanks brc
04:41.06drumkillakram!
04:41.15kramdrumkilla!
04:42.53Mocwe should make a nice table of codec on the wiki...
04:43.25JerJerMoc: why?   type show codecs in your asterisk CLI
04:43.29TondSud> sorry I still ahven't got a chnace to look at ur debug
04:43.36Tondbeen doing something ehere
04:43.38Tond:(
04:43.47JerJerMoc: and/or  show translations
04:43.48brc_Moc, already exists
04:44.07RolloTomnasihttp://voip-info.org/wiki-Codecs
04:44.57MocJerJer, well that show only the cpu load it take
04:45.35JerJerwhat mo do yo kneed?
04:45.37MocRolloTomnasi, I mean a page that rate each codec, like to compare g723 with g729, and g726 and speex
04:46.17RolloTomnasiah
04:46.44Mochow does speex sound like ? better, less good than g729 ?
04:47.07Odie_floconhmm dunno.
04:49.25Mocwell the audio sample does sound good...
04:49.32Mochow does it work with * ? good ?
04:50.30brc_kram, update THE PLAN
04:50.36kramhehe
04:50.48brc_</dramatic music>
04:50.53bkw_lalala
04:50.59brc_lala?
04:51.05bkw_hrm
04:52.10brc_mrh!
04:52.15brc_Mr. T!!!
04:52.48brc_this should be fun
04:53.52Moccan we get g723.1 codec for * somewhere ?
04:54.14brc_nop
04:54.15brc_e
04:54.18JerJerMoc: do you have your own patent indeminfication?
04:54.39Mocim canadian ;)
04:54.47bkw_screw patent's
04:54.48brc_so?
04:54.49bkw_its stupid
04:55.01brc_bkw_, yes, fine but you still must either a. follow the law
04:55.08bkw_the law is stupid
04:55.12brc_or b. pay the judgement after you are sued and the patent holders win
04:55.20bkw_like they gonna sue me
04:55.26bkw_hahahaha
04:55.27brc_yes
04:55.29RolloTomnasirtp audio format 4 = g723?
04:55.36WilliamKonly thing I personally like is trademark/service marks, and that's just so that there's only "1" freaking company with that name
04:55.41JerJerRolloTomnasi: show codecs
04:55.47JerJerin your asterisk CLI
04:55.48brc_not likely if you're using it in a small install of course
04:55.49WilliamKnot 100 companies with the same name running around that aren't affiliated
04:56.11bkw_the whole world is like "Only one person can think of an idea or design and its theirs even when others think of it on their own"
04:56.19Mocthey will have a hardtime sue me for a speech codec parent in canada for home use
04:56.20bkw_stupid shit
04:56.20RolloTomnasiJerJer- thanks again
04:56.31JerJerThe bill is already in the mail
04:57.39RolloTomnasioh boy
04:58.05TondMoc> is that so?
04:58.08*** join/#asterisk manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com)
04:58.09Tondcause me in Canada too
04:58.10Tond:)
04:58.16Sivaname three
04:58.20manipuraCanada good eh?
04:58.38Odie_floconme 4
04:58.42TondLets hots people's Servers in Canada and put G723 on it
04:58.43Tondlool
04:58.46Vcothe beer's ok..other than that it's so-so
04:58.49Moclet say you are more protected in canada
04:58.51Sivanasee.. the US don't know it.. but we taking over :)
04:59.11Mocbecause the gov doesn't change the law as the buisness wishes
04:59.15Sivanashh.. don't tell em either
04:59.26manipuraThe US is our big brother
04:59.26Sivanahail Martin!
04:59.29Tondthe best is I guess to go to a country that doesn't respect patent laws and install a server there
04:59.29Tond:)
04:59.31manipuraThey look out for us
04:59.48bkw_hrm
05:00.00TondI think that will work no?
05:00.05Odie_floconhey canada is the only country in the world that was able to burn down the whitehouse.
05:00.19Tondthe gambeling guys do that, to void paying tax
05:00.26Tondand be able to do business
05:01.06Tondso we got us 3 Canadians here EY?
05:01.06Tondlol
05:01.29MocTond, there is alot of canadian * user
05:01.40Vcocuz we're all cheap?
05:01.44Vco;)
05:01.45Sivanaheh
05:01.46Mocthe ratio population/* user is impresive..
05:01.51TondYa Canada is really up to speed with technology which is great
05:02.11TondWe are really good with Wireless technology!
05:02.23Vcojeez where are you living?
05:02.26Mocwell europe is better... but we are for sure better than the US ;)
05:02.42Vcoany companies that are providers out here are useless tools
05:02.52Tondwell actually Canada is very advanced in wireless, even better than EU..
05:02.59TondI am talking about wireless networks, but phones..
05:03.13Mochhaaa... hehe
05:03.34Odie_flocon4 Tond
05:03.38TondMoc, where ya from?
05:03.38Odie_floconI'm canadian 2
05:03.55TondOdie, Sivana, where ya from?
05:04.24Tonddid any of u guys attend the conference in Atlanta?
05:04.31Odie_floconI'm in Lethbridge Ab. right now.
05:04.33Odie_floconno
05:04.35Odie_floconI was working.
05:04.38SivanaON
05:04.41Vcoleth?
05:04.45TondI am from Toronto
05:04.50Vcoi'm heading down there in a few days for work..
05:04.53Sivanacool. I'm in North Bay
05:04.54Odie_floconjust south of Calgary.
05:04.56Vconice place...
05:04.59Vcofor a week
05:05.08Odie_floconbeen to T.O. and to North Bay.
05:05.10Tondcool
05:05.11Odie_floconhey Sivana.
05:05.14Odie_floconhow goes it?
05:05.15Sivanahey dude :)
05:05.15Vconot sure if i could handle it much longer than that
05:05.17Sivanapretty good
05:05.26Sivanayou?
05:05.31MocMontreal here
05:05.38Odie_floconBeen there too Moc.
05:05.48TondMoc> Oh Montreal!  the best party place..!
05:05.49TondL)
05:05.49Odie_floconI'm doing good.
05:06.05Odie_floconI hear they have the Best Strip clubs there.
05:06.25TondMontreals got some hard core party animals and easy beautifull women
05:06.26Tondlol
05:06.27manipuraHey Odie, I'm IN Calgary
05:06.34Odie_floconreally
05:06.38Odie_floconI'm in Lethbridge.
05:06.44manipuraI see that
05:06.51Odie_floconI grew up in Forest lawn.
05:07.01manipuraare you proud of that?
05:07.19Odie_floconyup
05:07.21Odie_flocon:D
05:07.30Odie_floconit wasn't that bad when I was there.
05:07.37Odie_floconit was a nice area.
05:07.41TondOdie> do u setup * in Calg?
05:07.44manipuraYeah, you must be old :)
05:07.45Odie_floconbefore all the condos.
05:07.53Odie_floconyes.
05:08.02Odie_floconI've just started a company there.
05:08.24manipura~seen dbruce
05:08.29jbotmanipura: i haven't seen 'dbruce'
05:08.40manipura~seen cybermage
05:08.40jbotcybermage <~CyberMage@user-24-236-84-79.knology.net> was last seen on IRC in channel #asterisk, 170d 12h 37m 59s ago, saying: 'biot: Thanks'.
05:09.04manipura170 days, thats a while...
05:09.51RolloTomnasidoes it make sense that while sudoer's dropping voice after one second he can still hear tones both ways afterwards?
05:10.38RolloTomnasiguess it does
05:11.22*** join/#asterisk yooyo (~l14aa2@c-67-160-217-46.client.comcast.net)
05:11.30*** join/#asterisk foxdevel (~chatzilla@rrcs-24-227-76-106.se.biz.rr.com)
05:16.17*** join/#asterisk af_ (~af@62.94.148.227)
05:16.36DaminAlright..
05:16.39DaminYou want to be famous?
05:16.54DaminI need some example configurations for my presentation.. :)
05:17.28DaminSee the asterisk-users mailing list for details..
05:18.31brc_Damin, such as?
05:18.33brc_oh
05:18.34brc_looking
05:19.32Kumbangwhere can i find x-lite for pocket pc
05:20.14*** join/#asterisk brettnem (~brettnem@66.102.167.245)
05:21.31*** join/#asterisk Moc (~mochouina@modemcable021.49-80-70.mc.videotron.ca)
05:22.19tessierI know it's probably a foolish question but has anyone ever written any documentation for end useres of * phone systems?
05:22.32Mocoh oh... 109 open bug ...
05:23.18Moctessier, well there is a open Asterisk Documentation project
05:23.28Mocbut there is a book someone made also.. no idea if it good or not
05:24.43brc_does AddQueueMember return 0 or -1 on sucess?
05:25.20paulcI heard the book was dodgy
05:25.26paulcbut that was on the mailing list I think
05:25.28*** join/#asterisk serdiehard (~serdiehar@202.65.128.18)
05:25.40JerJerOtherwise it will return an error
05:25.40JerJerReturns -1 if there is an error.
05:25.40JerJerExample: AddQueueMember(techsupport|SIP/3000)
05:25.50JerJer*CLI> show application AddQueueMember
05:26.18serdiehardpaulc:the astcc solution is working finely
05:26.25brc_JerJer, yes, it returns -1 if there's an error....doesn't mention what it returns if there's no error
05:26.29paulcnice one! shouldn't you be in bed now though? ;-)
05:26.44brc_I'd assume 0 but assuming is a dangerous thing to do with asterisk
05:27.02serdiehardiam at my work its 11:07 in india
05:27.06serdiehardam
05:27.08*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:27.33manipuraIts 11:07 in canada
05:27.37manipuraPM
05:27.40Odie_floconhey sivana you still there?
05:27.42JerJerbrc_: the source says 0
05:27.46manipurawell, where I am in canada
05:27.51Odie_floconit's more liek 11:27
05:28.04brc_~jerjer++
05:28.11brc_~karma jerjer
05:28.11jbotjerjer has karma of 1
05:28.16JerJerline 1493 of 2418 in app_queue.c
05:28.26manipuraYeah, I gotta slow clock, It's acutaly 11:11
05:28.43paulc10:28 in BC.. 11:28 in Alberta.. and I'm sure MOC's up past his bed time too
05:29.01Odie_floconsivana?
05:30.10JerJerlook at the switch statement above
05:30.36JerJer<PROTECTED>
05:30.39brc_aha
05:30.46JerJerleme update
05:31.48JerJerok line 1505 on current cvs
05:32.57JerJerin apps/app_queue.c ?
05:33.02brc_yes
05:33.04brc_odd
05:33.16brc_I do see the switch stuff though...
05:33.23brc_doesn't really matter
05:33.25JerJeri just cvs updated this box
05:33.30brc_ahh
05:33.36brc_I'm on a couple weeks ago
05:33.37brc_doh!
05:33.39brc_sorry
05:33.42Odie_floconhey tond u still here????
05:33.47brc_~thwack brc_
05:33.50jbotACTION hits brc_ on the head with a Holy Bible
05:33.57brc_~kick brc_
05:34.00jbotbugger off sod!
05:34.14ard~help me!
05:34.22ard~help
05:35.34brc_ard you'll have to be more specific
05:35.51*** join/#asterisk Exomorph (~Exomorph@216.251.134.2)
05:35.55ardbrc_ : Well, I got a complete overview /msgd...
05:36.25ardIt says: I learn mainly by observing declarative statements such as "x is at http://www.xxx.com", and then reply when people ask things like "where can i
05:36.29ardSo:
05:36.34ardwhere can I find sex
05:36.42ardHmmm, no reply :-)
05:36.47brc_you dind't ask jobt
05:36.52ardAh...
05:37.20manipuraThats a good site
05:37.23ardjbot_ where can I find sex
05:37.37Daminmanipura: Are you the long lost spanish brother of Sippura?
05:38.01manipuraSorry, but no
05:38.15manipuraI don't speak spanish
05:38.18brc_Damin, where did you send your message?
05:38.23brc_I don't see it on -lusers
05:38.32ManxPowerDamin: Naw.  Some customer of his got REALLY angry with his SIPura and shoved it...well...you get the idea.
05:38.40manipurabrc_ you're a luser
05:38.56brc_YAY!
05:39.04manipuraoh wait.. your an op.... Uuhhh... your the greatest man alive
05:39.13ardLOL
05:39.14manipuraI worship you brc_
05:39.17brc_I thought so
05:39.24ardUntil he gets op...
05:40.25brc_bifocal contacts?
05:40.29brc_I thought those didn't really work
05:40.56brc_dude
05:40.57brc_bwhhaahahhaha
05:41.50EarlGreyanyone awake?
05:42.21ardThen my contract ends...
05:42.22denonsomeone give me some good thoughts on what's cool in cordless phones these days
05:42.24EarlGreybeen trying to get a phone to connect to my * pc but it just dosn't want to know
05:42.32paulcDenon: VTech have always done me proud
05:42.32denonI dont really need range, so engenius is probably overkill
05:42.43denonpaulc: man, vtech always look like junky toys to me
05:42.56denonI like my current Siemens Gigasets, but they're getting a bit dated
05:43.02arddenon : No power cords ... (My "cordless" setup needs a power supply and a notebook with wifi :-) )
05:43.05paulcDunno what model mine is but I'm dead picky about shit like that.. it's nice, it's silver, compact base unit, no frills, it just works
05:43.05EarlGreycan you have a quick look through my sip.conf and see if you can see any thing that could be wrong
05:43.08Daminbrc_: Haha.. that's because I sent it to digium.org instead of .com. Take a look in a few minutes. ;)
05:43.13brc_hahaha
05:43.19paulcEarlGrey: www.pastebin.ca and I'll take a look for you
05:43.22brc_tat no wek so gud
05:43.26Daminbrc_: Subject: [Asterisk-Users] Peer Review - Linuxfest Presentation Outline
05:43.29brc_k
05:43.37brc_dere it is
05:43.41EarlGreywww.nomorepasting.com/pater.php?pasteID=21455
05:43.47brc_sheesh
05:43.49brc_loooong
05:43.53EarlGreysorry url wrong
05:44.15EarlGreywww.nomorepasting.com/paste.php?pasteID=21455
05:44.28denonpaulc: vtech generally have a sep flash button?
05:44.54Daminbrc_: Yeah.. lots of details that will get glossed over. I.E. the Channels may only be 2 slides, and simply be a quick run-down of what is available..
05:45.01EarlGreythe account i'm trying to connect on is the 2000 one for the grandstream
05:45.05brc_ya
05:45.10brc_reedeng
05:45.19EarlGreymost of it is the stock sip.conf that is there from fresh
05:45.20Daminbrc_: With a summary of "Asterisk can pretty much connect anything to anything else."
05:45.28brc_right
05:45.38denonwow, vtech only has one 5.8GHz model
05:45.55DaminYou know..
05:46.11brc_nope
05:46.12DaminNever mind..
05:46.13brettnemDamin: but remember, it's the "open source PBX" so not quite anything, right?
05:46.45brc_brettnem, uh...that doesn't really make sense (to me)
05:46.52Daminbrettnem: Bahh. :) Symantics. With res_perl it can be anything to anything! :)
05:47.06paulcEarlGrey: Give me 2 mins
05:47.16paulcDenon: Nah, the flash is the same as the "off hook" button on my VTech
05:47.20brettnembrc_: there was some discussion that that tagline suggests * is limited. ;)
05:47.31brc_mmm
05:47.32EarlGreyk
05:47.36brc_at astricon you mean?
05:47.41brettnemyeah
05:47.44brc_ahh
05:48.03DaminToday's res_perl trick of the day; Write an extension that Emails you the CallerID whenever someone uses it.
05:48.10Daminasstrickon
05:48.14brettnemheh
05:48.35Odie_floconhey how do I start a Priv chat?
05:48.53brc_asstrickOFF!
05:48.55DaminOdie_flocon: Pick up the phone and call the person you want to chat with?
05:48.57brc_Odie_flocon, /query
05:49.00DaminAlright..
05:49.05DaminI need to get some sleep.
05:49.09DaminBedtime for Damin..
05:49.10brc_likewise
05:49.21cypromiswho needs perl for that ?
05:49.30denonpaulc: yeah .. I dunno, people get confused when I tell em to use the same button .. I think I'd rather have a separate button
05:49.30cypromisthere is an app submittd by scaredycat for that
05:49.31cypromis:)
05:49.49Odie_floconwhat you mean brc?
05:50.00*** join/#asterisk Mike (~mike@201.135.48.52)
05:50.22ardYeah.... It is 7:50 am... I need to shower and get the bus...
05:50.56brc_<allison>AGENT Logged Off!...Goodbye!
05:51.54EarlGreypaulc  to fill you in on the symptoms i get no dial if i do dial a number it just sits there then i get a 4 on the screen
05:52.04brc_thankyou for calling 800 free-asterisk-support, is there anything else I can help you with? toobad *click*
05:52.07DaminAlright must go to bed..
05:52.15*** join/#asterisk jmhunter (~jacob@wire2-150.razzolink.com)
05:52.15*** mode/#asterisk [+o jmhunter] by ChanServ
05:52.26EarlGreyis there anyway to see from asterisk logon attempts from a phone?
05:52.49paulcyeah - start the console with "asterisk -vvvvr" and see what's going on
05:52.53paulcis it registered correctly?
05:52.57ManxPowerYou usually can't see anything with a phone.
05:53.05ManxPowerUse your eyes, they are best at that sort of stuff.
05:53.17WilliamKall varies per manufacturer
05:53.22ManxPowerI *said* "usually"
05:53.22brc_~thwap WilliamK
05:53.24jbotACTION pees on WilliamK and does them dry
05:53.48WilliamK=)
05:54.16denonpaulc: you just have a single vtech base?
05:54.35denonim curious if you can register the phones with multiple bases
05:54.44paulcdenon: yeah
05:54.54paulcand I think you're right..
05:54.57paulcone handset --> one base
05:55.34*** part/#asterisk RolloTomnasi (~JH@delmar-209-137-161-171-dsl.cavtel.net)
05:55.59denonpaulc: im kinda looking for something new at the office, so we can just have cordless phones that roam between base stations
05:56.13denonbut im not ready to spring for 802.11
05:56.26paulcwhat was I looking at the other day.. some kind of DECT system that roams between multiple base units.. hmm..
05:56.29paulccan't remember
05:56.49paulcmight have been panasonic actually, part of their PBX range..
05:57.13denonpaulc: yeah .. most times the pbx ones do it, for a price
05:57.48paulcdon't reckon you'll find roaming handsets without a PBX in the back end
05:57.56EarlGreypaulc what command can i use to see once i'm at the asterisk cli?
05:57.58yooyowat up denon
05:58.19paulcEarlGrey: type "set verbose 3" then try and make a call from your soft phone
05:58.28denonuh, same old .. do I know you?
05:58.34yooyonaw
05:58.39yooyoi like the nick denon
05:58.51denonic
05:58.59yooyo802.11 with WiSip?  anyone got that working?
05:59.02denonseeing as thought I've used it for going on a decade now ..
05:59.06denonI guess I do too
05:59.26EarlGreynothing
05:59.49EarlGreyoh you said softphone would a hard phone make much diffrence?
05:59.59*** join/#asterisk implicit (~implic1t@ip68-5-148-1.oc.oc.cox.net)
06:00.00paulcif you do "sip show peers" do you see your softphone registered with asterisk?
06:00.15paulchow's the hard phone connected?
06:00.28implicitwhat the hell is going on?
06:00.47denonpaulc: http://catalog2.panasonic.com/webapp/wcs/stores/servlet/ModelDetail?displayTab=F&storeId=11251&catalogId=11005&itemId=65209&catGroupId=12973&modelNo=KX-TD7690&surfModel=KX-TD7690&ignoreRedirect=1 ?
06:00.47EarlGreyhavent configured the softphone yet i thought the hard phone would be more direct
06:01.43denonpaulc: those are 300ea :\
06:01.47*** join/#asterisk dnc (~duncan@213.244.225.42)
06:01.50denonmay as well go 802.11 then
06:01.52paulcHmm.. nice.. but how'd they work then?
06:02.31paulcEarlGrey: So what kind of hard phone are yuo using and how is it connected? Might be worth playing with your softphone too
06:02.35EarlGreywww.nomorepasting.com/paste.php?pasteID=21457
06:03.04EarlGreyGS budge tone 101
06:03.51paulcok. Your soft phone hasn't registered with your asterisk box..
06:03.55paulchow's the hard phone setup?
06:05.21*** join/#asterisk BaDBuG (~brnbug@h000094b662bb.ne.client2.attbi.com)
06:05.31BaDBuGHello everyone
06:05.45BaDBuGhow is everyone?
06:06.18BaDBuGjust wanna introduce my self! this is my frst time to join this chat room
06:06.29BaDBuGI played around witg asterisk and i love it !
06:06.40BaDBuGkeep up the good work guyz!
06:07.53ardpaulc : wifi can roam perfectly... You just need "good" ap...
06:07.58implicitok
06:08.30ardpaulc : Take a bunch of linksys, and give them the same essid...
06:08.49Odie_floconhmmm.
06:09.25BaDBuGhey guyz I had asterisk setup properly I also got the demo working! I am full linux noob but got it up and running
06:09.31denonwhat we need, is cheap 802.11 phones..
06:09.33BaDBuGthe VI is kind of hard to use
06:09.39implicitdenon: or expensive ones
06:09.46paulcard: yeah that'll work.. but wifi phones ain't cheap
06:09.51denonimplicit: from what Ive heard, the ciscos arent bad
06:10.10BaDBuGI just want to know how hard is it to setup a pHP to edit those sip.conf and extention.conf?
06:10.14denonthese engenius phones are just too damn pricey
06:10.26denonfor what you really get, anyway
06:10.32Odie_floconhey what's iaxprov.conf for?
06:10.33denonI dont need range, really
06:10.44paulcBadBug: Check the wiki - there's some apps that'll do what you want on there
06:10.53ardpaulc : Correct... :-( Expensive and probably insecure :-(
06:11.11denonOdie_flocon: IAX privisioning, of course
06:11.27Odie_floconthanx denon.
06:11.28BaDBuGyeah i saw that i also tried setting it up but the intructions are kind of hard
06:11.51citatsdenon: how much are the engenius phones?
06:12.14denoncitats: varies, but like 2-300 bucks for a single phone/base
06:12.27Odie_floconso do I use that when using Diax?
06:12.28denonand kinda pricey per phone after that, if your base supports it
06:13.15ardgot to go to work...
06:13.21Odie_floconit's for the Iaxy provisioning right?
06:13.21yooyothat's 100 cheaper than my lucent 4065D phone
06:14.07BaDBuGanyone here from boston?
06:14.21BaDBuGI am looking if there is any kind of training for asterisk
06:14.40BaDBuGI whould really appreciate it!
06:18.03BaDBuGHello?
06:18.31Odie_floconsorry not from boston?
06:18.58blankmanFor those on the wifi question ... the dlink ethernet2wifi bridge is a good bet ;-) We are testing them with the IAXy's.
06:18.59BaDBuGOdie: thank you
06:19.52blankmanMake the price come in at around 185 for the package ... plus you can split them up and use the e2w bridge to stream movies.
06:20.24blankmanYou have to have the g setup right though for the movies to work well.
06:21.13*** mode/#asterisk [+o brc_] by jmhunter
06:21.59denonman, I dont understand why 5.8Ghz isnt more popular
06:22.03BaDBuGokay i have an ipcop machine as a router + firewall do you guyz recomend placing the asterisk box on the orange card?
06:22.04denonI mean, 2.4 is so saturated
06:22.34denon(for cordless phones I mean, obviously not for networking/802.11a)
06:25.56Odie_floconso denon what is the purpose of the alarm reciever application?
06:27.17paulcisn't it for burglar alarm systems that dial out to monitoring stations and use FSK modem stuff to indicate the cause of the alarm, location etc? it acts as a destination that can be called to receive that stuff?
06:27.17denon[Synopsis]:
06:27.17denonProvide support for receving alarm reports from a burglar or fire alarm panel
06:27.17denon[Description]:
06:27.17denonAlarm receiver application for Asterisk. Only 1 signalling format is supported at this time:
06:27.17denonAdemco Contact ID. This application should be called whenever there is an alarm panel calling in
06:27.17denonto dump its events. The application will handshake with the alarm panel, and receive events,
06:27.19denonvalidate them, handshake them, and store them until the panel hangs up. Once the panel hangs up,
06:27.21denonthe application will run the command line specified by the eventcmd setting in alarmreceiver.conf
06:27.23denonand pipe the events to the standard input of the application. Alarmreceiver.conf also contains settings
06:27.25denonfor DTMF timing, and for the loudness of the acknowledgement tones.
06:28.37Odie_floconhmm neat:D
06:29.05brettnemhey guys, I just had idledial thrash my pri. Now I can't recieve any calls on it.. anyone have any ideas how to restore it? I've tried rmmoding the drivers.. no avail..
06:29.17brettnemI'm about to reboot
06:29.34denonpull the plug?
06:29.46brettnemon the pri?
06:29.53Odie_flocondid you pull out the pri?
06:30.11brettnemno.. the pri is 160 miles away
06:30.24denonreboot then . .the time it'll take to figure it out will just be more downtime
06:30.49citatsbrettnem: what kind of state is it in now?
06:30.50WilliamKhope you have a remote control power bar =)
06:30.53brettnemI've been toying with idledial.. and it setup a whole t1 of calls at once; which has been causing havoc on my d-channel..
06:31.11Odie_floconwhat is idledial?
06:31.13Odie_flocon:D
06:31.16brettnemcitats: well * thinks everything is ok, but I get congestion dialing in
06:31.19Odie_floconmore questions hey denon
06:31.34brettnemOdie_flocon: idledial instructs * to dial out and connect to applications on idle channels on a pri
06:32.05citatsbrettnem: if you flip on 'pri intense debug span 1' is it just going nuts with messages?
06:32.25citatsor whatever span it is
06:32.26brettnemOdie_flocon: usefull for networking 2 * systems together for IP connectivity over the PSTN
06:32.36brettnemcitats: haven't tried intense.. let me see.
06:32.52denonman! all these cordless phones .. support up to like 10 handsets, but one pstn
06:32.56citatsbrettnem: also check 'pri show span 1'
06:33.00denonwho on earth would want 10 handsets on a single line
06:33.12brettnemthis is a bug.. I should document it..it's repeatable everytime.. gr
06:33.24Odie_floconI have wondered that myself denon.
06:33.25denonbrettnem: bugs.digiumc.om
06:33.27WilliamKdenon, someone who wants a phone in every room of their mansion
06:33.28WilliamK=)
06:33.41brettnemdenon: yes, I know.. thx
06:33.43Odie_floconyeah but how do you charge them all?
06:33.55WilliamKeach has an individual charger usually
06:33.58citatsbrettnem: if you tell me how to reproduce it i'll see if i can
06:34.03Odie_floconohh ok.
06:34.11WilliamKbedroom, office, and kitchen
06:34.13WilliamK=)
06:34.14denonWilliamK: yeah, but if I have 10 rooms with phones, I need a few lines :)
06:34.19paulcif you had a mansion surely you'd have a small PBX?
06:34.24denonor if I have 10 rooms, I probably need expansion bases
06:34.29brettnemcitats: I'm just setting up idledial; mostly like in the wiki.. for zapras
06:34.35Odie_floconnow it would be nice if you could incorperate them into a PBX...
06:34.35denonI dont see that many of these companies let you roam between bases
06:34.36citatsbah, each of my cordless phones has its own extension :)
06:34.43WilliamKpaulc, you'd be surprised how CHEAP people with mansions are
06:34.44WilliamK=)
06:34.51denoncitats: each with its own base?
06:34.54paulcs'how they can afford 'em in the first place.. cheap cunts innit
06:35.00Odie_floconone handset, with 13 phones connected to pbx lines.
06:35.04brettnemcitats: the pri looks ok.. doing the intense debug now..
06:35.08citatsdenon: yeah, just random cordless phones
06:35.09denonright now I have a 2-line siemens .. works ok .. but I kinda want 5.8
06:35.15Odie_floconone base set I mean.
06:35.22denonIve got like 5 phones on those 2 lines
06:35.31denonthen 7960s for everything else
06:35.34yooyoi use my cell phone if I have ten rooms
06:35.38WilliamKworks well, cept I've dropped my handsets too many times
06:35.53brettnemcitats: no excessive messages on the intense debug
06:36.08*** part/#asterisk Kumbang (~kumbang@167.205.22.54)
06:36.13yooyoi use vonage now
06:36.14Odie_floconwould be nice to have a wireless * handset.
06:36.23*** join/#asterisk Kumbang (~kumbang@167.205.22.54)
06:36.34brettnemcitats: know of a way to really reset that pri without rebooting?? seems like a rmmod ought to do it; but it didn't..
06:36.45citatsbrettnem: when you stop asterisk the pri is basically reset
06:36.47denondunno, I think I'll just give up on finding the ideal cordless phone
06:36.50WilliamKthe guy at vonage dang near cried and gave me 3 months free
06:36.50brettnemcitats: btw, had the same probelm on both sides of the ras connection
06:36.54yooyoi want to use vonage in addition to asterisk
06:37.06brettnemcitats: I had to do a full reboot to get the near end back up
06:37.09denonWilliamK: really..
06:37.27WilliamKdenon, yeah...vonage has lost ALOT of customers in the last 2 months
06:37.32brettnemcitats: fuser /dev/zap/* is even clear.. and it still isn't resetting the pri..
06:37.33WilliamK3-6 outages so far
06:37.35Odie_floconvonage??
06:37.39*** join/#asterisk kiel (~kiel@zen.via.ecp.fr)
06:37.41yooyo3-6 outages!
06:37.43kielyouhou !
06:37.50yooyooh no, i need to cancel
06:37.57WilliamKyeah in the last 2 months
06:37.59citatsbrettnem: any more info you can provide?  any log messages or anything?  what q931 do you see when you try a call?
06:38.01WilliamKbroadvoice is also cheaper
06:38.02yooyomy phone has never had an outage EVER
06:38.11WilliamKmine has
06:38.22WilliamKI regullarly drop calls, and loose DTMF
06:38.27brettnemcitats: I don't see anything when trying to call into the box..
06:38.29WilliamKand SBC doesn't want to fix it
06:38.34Odie_floconno wonder.
06:38.46brettnemcitats: ie: no debug output
06:38.47yooyomy ATT cell phone always drop calls
06:38.47Odie_floconlook at the price for vonage.
06:38.59WilliamK#1 reason why SBC is loosing my business here in 3 days
06:39.11brettnemcitats: had the same problem on the near end until I rebooted.
06:39.12Odie_floconsbc?
06:39.14yooyohey broadvoice is cheaper!
06:39.20Odie_floconyeah way cheaper
06:39.20WilliamKjust waiting on them to port my # to the PRI at the clec
06:39.25Odie_floconalthough it's US.
06:39.44Odie_floconwilliam where u from again?
06:39.53citatsbrettnem: well reboot to fix it then :) but i'd like to figure out whats causing the problem (if its something on asterisks side)... any interesting messages right before tre problem started?
06:40.15brettnemcitats: all sorts of interesting messages while the idledial is going on..
06:40.24citatsbrettnem: can you paste them somewhere?
06:40.29brettnemcitats: let me see if I can pull something out of the messages log
06:40.43brettnemhmm.. why does my office smell like paint thinner..
06:40.45Odie_floconman vonage is soo expensive.
06:41.11WilliamKmy SBC line with less features than vonage is 117.00 a month
06:41.43WilliamKand yeah that's BASIC service with a few features added so it's compareable to vonage
06:41.50WilliamKno voicemail either
06:41.55citatsbrettnem: also, do you get receive any messages from the other side with intense debugging?  check /proc/interrupts to see if your card is still taking interrupts
06:42.26brettnemcitats: let me see what my exact cause code is on the outgoing call.. that should be useful.
06:43.35brettnemhmm network congestion.. it really looks like a network problem, but it isn't.
06:44.03WilliamKoh by the way... yooyo, when you goto cancel vonage you will have to send the ATA back to them, or pay a 39.95 cancelation fee
06:44.08Odie_floconI'm looking into Broadvoice it's like 19.95
06:44.30brettnemcitats: do you think this is a problem: 10: 1791404387          XT-PIC  usb-uhci, usb-uhci, eth0, tor2
06:44.43brettnem(from /proc/interrupts)
06:45.01citatsbrettnem: bad to have stuff sharing interrupts... but is the number incrementing a 1000/sec?
06:45.15RevKbkw_, "RevK you want want want want hahaa", what was that in relation to?
06:45.38Odie_floconyeah I will
06:45.58brettnemcitats: that seems about right
06:47.46*** join/#asterisk serdiehard (~serdiehar@202.65.128.18)
06:47.58*** join/#asterisk isamar (isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp)
06:48.02isamarhi folks
06:48.14isamarI wanna do:
06:48.25serdiehardwhat is the ARQ means in H323?
06:48.30isamarSystem(/usr/bin/callout ${TEST})
06:48.47isamarbut the variable is not being passed out to the shell script :-(
06:48.53isamarwhat should be the problem
06:48.56serdiehardwith respect to openh323gk
06:49.01Odie_floconno IRQ in h323
06:49.25serdiehardARQ
06:50.42isamaranybody knows this prob??
06:54.24*** part/#asterisk jmhunter (~jacob@wire2-150.razzolink.com)
06:54.38*** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
06:56.25brettnembbiab
07:02.28*** join/#asterisk moonwick (~moonwick@core.dump.net)
07:03.18moonwickhm, is ${HANGUPSTATUS} not set for IAX2 calls?
07:04.01*** join/#asterisk dome_c (~dome@61.19.211.250)
07:04.02moonwickactually, I suppose DIALSTATUS will also tell me if a Dial() returns busy.
07:04.14dome_chi Thailand here
07:06.49*** join/#asterisk echo_bg (~asdasdasd@212.116.151.30)
07:06.53*** part/#asterisk implicit (~implic1t@ip68-5-148-1.oc.oc.cox.net)
07:06.56echo_bghi all how are you people
07:07.51Odie_flocongoof
07:07.54Odie_flocongood
07:07.56Odie_floconI mean
07:08.59echo_bgOdie_flocon :)
07:09.27echo_bgguys can you tell me a codec ( exept g729) that uses less than 64kbps ?
07:09.39nextimespeex
07:10.03echo_bgis it working OK ?
07:10.14echo_bgand what about ilbc ?
07:10.44dome_cgsm
07:11.47echo_bgdome_c - tryed - quality was terrible
07:12.07dome_ci use in my comercial. it's ok
07:12.51echo_bgheh - will ahev to try
07:13.06dome_cWhat's client ?
07:13.12dome_csoftware ot ip phone ?
07:13.18echo_bga xlite agent and sj phone agent connecting to eachother and the quality was not same as with ulaw
07:13.22echo_bgor alaw
07:13.53dome_ctry again. it's work in my solution :)
07:14.11echo_bgdome_c - will do for sure :)
07:14.24echo_bgany experiense with hardware phones -
07:14.35echo_bgwhitch is working best with *
07:14.45dome_cI use ip phone from china
07:14.53dome_c80 Usd
07:15.20echo_bggrandstream ?
07:15.27dome_cin SIP mode. (IAX on develop)
07:15.29dome_catcom.
07:15.47dome_cand another brand. they use PA1688 chipset
07:17.28yooyodoes anyone know if asterisk can do caller ID?
07:17.45yooyoi want to hook asterisk into my contact management program
07:17.58moonwickheh.  so, does anyone happen to have a standard busy tone in GSM?  I don't see one in the regular collection of sounds, or in the addon package.
07:18.14yooyowhen a customer calls, the caller ID is picked up and a search query is initiated and popup the contact info of the customer
07:18.14moonwickyooyo: it can certainly handle it
07:18.29moonwickwhether or not it'll have access to it depends on how calls are coming in
07:18.54yooyois there a demo of asterisk in the Bay Area?
07:18.59yooyoi would like to attend one
07:21.48*** join/#asterisk grant_a (~none@ca-stmnca-cuda3-blade5b-55.stmnca.adelphia.net)
07:22.13grant_aWHy does asterisk no allow my firefly sip phone to connect it gives me this error msg any ideas? --> Sep 28 00:08:53 NOTICE[-235017296]: chan_iax2.c:5183 socket_read: Rejected connect attempt from 192.168.1.100
07:22.58brc_because you are using iax2 not sip it seems
07:23.26grant_aOk i meant Soft-phone not sip phone
07:23.28*** join/#asterisk Silik0n (~krice@cpe-066-061-042-120.midsouth.rr.com)
07:23.29grant_aFirefly is set to IAX
07:23.47Silik0nanyone played with the netweb-301
07:24.01grant_abrc?
07:24.08brc_no clue
07:26.44grant_aIf I send you MoneyGram® of $20.00 USD will you help me brc
07:26.59*** join/#asterisk GoRK (GoRK@ip68-109-58-244.lu.dl.cox.net)
07:29.20GoRKhello -- im testing out some 'follow me' features and trying to figure out how best to implement callerid stuff.. if i have callerid=8885551212 or similar in iax.conf for a peer, the callerid will always be that no matter what.. i cant override it for when i relay an incoming call back out
07:29.38GoRKbut if i unset it, how is the best way to set a 'default' callerid for the outgoing calls?
07:29.41nextimeanyone using app_meetme2 with meetme control web interface?
07:30.15grant_aSep 28 00:26:02 NOTICE[-235414608]: chan_iax2.c:3700 register_verify: Peer 'NuFone' is not dynamic (from 192.168.1.100)
07:30.17grant_aWhat about that?
07:30.36maruzgrant_a: have setupped firefly to use sip and then u changed it to use iax?
07:31.24grant_aYea
07:32.52isamaranybody already tried to callin through a FXO and callout through the other???
07:33.13isamarI have been tried that but I get too much noise even with aggressive echo suppression... :-((((
07:33.41Odie_floconnot with the web control interface yet
07:34.00GoRKisamar: are these both zaptel devices? you should use zaptel bridging instead.. no echo
07:34.40isamarGork.. zaptel
07:35.04isamarI am using bridging... but they are X100Ps...
07:35.12isamarnot IRQ conflict though...
07:35.18GoRKit works with x100p i am pretty sure.. dont see why not
07:35.23isamarand they work perfectly for dialout and dialin
07:35.32GoRKis echocancelwhenbridged=no in zapata.conf?
07:35.57brc_isamar, sometimes echo is actually CAUSED by the echo can
07:36.07brc_try turning it off and see what happenes
07:36.13brc_happens
07:36.18isamarechocancelwhenbridged=yes
07:36.23GoRKyeah set that to no
07:36.31isamarhmmm
07:36.38isamarok.. I will give some tries
07:36.44maruzgrant_a: try to detect something using database show commend
07:36.59GoRKyou should not have to echo cancel bridged zap channels unless you have a dog slow system :)
07:37.13isamarok
07:37.53grant_aEverydayIsLikeSn (12:36:16 AM): p 28 00:32:32 NOTICE[-234890320]: chan_iax2.c:5488 socket_read: Rejected connect attempt from 192.168.1.100, request '1000@default' does not exist
07:38.15*** join/#asterisk r1 (~erwan@www.thiscow.com)
07:38.18grant_aApparently firefly does connect i added the context [12345] in iax.conf and it connects to that.. i duno i cant get much further though
07:38.35GoRKi thought there was an argument to app dial to suggest native bridging (or native 64k bridging -- ie data call quality)
07:38.42GoRKthat would probably help also :)
07:40.03echo_bgbrc_ wnat me ?
07:40.37brc_echo_bg, uh...no
07:40.41brc_echo_bg, why?...
07:41.06echo_bg<brc_> isamar, sometimes echo is actually CAUSED by the echo can
07:41.20brc_uhm...uhm
07:41.24brc_haha?
07:41.41isamarsome fun for MSN users:
07:41.43isamarhttp://www.msnskins.be/emoticons/erotiek/
07:42.04isamarour life is not only work :-)
07:42.56brc_~seen ptg
07:42.58jbotptg <~PTG@ip68-106-19-249.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 3d 4h 5m 51s ago, saying: 'actually you could use xargs and cut to do it :)'.
07:43.03ardecho_bg : Hmmm... that means that the filter is out of sync...
07:43.33ardecho_bg : When everything keeps in sync and the adaption is ok (detects double talk and such) it should work more or less
07:43.36echo_bg:)) sorry guys - just a joke - I know :)
07:43.38ardecho_bg : But you are right..
07:44.13EarlGreycan anyone think of a reason why astreisk refuses to alow phones to connect to it it's on fedora core 2 with all the updates and a 3 day old version of asterisk?
07:44.25echo_bgard - can it be used when a calling site is a softphone with speakers and mic but not with headset ?
07:44.33EarlGreycould there be anything missing or present in fedora?
07:44.47echo_bgEarlGrey - havent you change the OS ?
07:45.07EarlGreyto what?
07:45.16ardecho_bg : iek... cancelling remote acoustic echo is a pain... And it really doesn't matter if it is a speakerphone or a sensitive handset...
07:45.44EarlGreyecho_bg to which os?
07:45.47echo_bgahm
07:45.53ardecho_bg : For acoustic cancel you should put it in the softphone...
07:46.00echo_bgEarlGrey i have switched from FreeBSD to Slackware
07:46.18echo_bgthere is no a package 1.0 ver for FBSD
07:46.30echo_bgso slack is OK for linux
07:46.38echo_bgclose to BSD style OS
07:47.17ardisamar : echo can be caused by the remote site. In 40% of the calls we make from the call-center land-terminated phones are sending back a loud echo (acoustic or electric...)
07:48.08moonwickasterisk 1.0 compiles with a minimum amout of work under freebsd
07:48.38grant_aAnyone here using firefly and asterisk with nufone by chance?
07:49.13echo_bgmoonwick - maybe - but when i installed it from ports collection - i didnt compile the cdr_mysql.so and other modules for work with database
07:50.07brc_ewww
07:50.18brc_is 1.0 already in ports?
07:50.23brc_are you _SURE_ it's 1.0?
07:50.57brc_some distros stupidly labeled the 1.0 branch 1.0 in their package systems a long time ago (noteably debian)
07:51.18brc_nevermind
07:52.10echo_bgbrc_ -
07:52.14echo_bgno it is not
07:52.15brc_ya doh
07:52.25brc_echo_bg, btw I've heard gentoo is closer to *bsd then slack
07:52.27echo_bgi used 0.9.0_1 or something like that
07:52.32brc_w portage etc
07:52.38brc_oh no no no no no
07:52.42brc_update right now
07:52.48brc_0.9 is _very_ old
07:52.51echo_bgbrc_ - dont delieve - slack has same rc.* starting system
07:52.57brc_k
07:53.04echo_bgand working over ports and packages in FBSD style
07:53.27echo_bgbrc_ i am runing it on linux now -
07:53.43brc_ahh
07:53.57echo_bgi am planing to use digium HW and it gives best performace with linux
07:54.15*** join/#asterisk mixi (mixi@pD954592C.dip.t-dialin.net)
07:54.24echo_bgfbsd is a perfect platform - unfortunately still not for asterisk
07:55.01maruzwhy?
07:55.16Silik0ndrivers
07:55.27echo_bgyep -drvrs
07:56.08maruz:(
07:56.10Silik0nwould be nice if someone would put some serious work into them
07:56.16echo_bgi read that if I am planing to use digium HW i need LINUX
07:56.33*** join/#asterisk clive- (~pirch@myw-stp-66-18-85-242.sentechsa.net)
07:56.34Silik0nthats where all the driver dev happens
07:57.14echo_bg:))
07:57.17echo_bgmdam
07:57.34Silik0nbrc_ we could start the bsd/linux wars... those are better then distro wars heh
07:57.49echo_bgso regarding codecs - share you experiense - what is the best codec to use for low rate BW
07:57.59echo_bgSilik0n - no need to
07:58.02Silik0nheh
07:58.07ardSilik0n : Yep. That's true. Everody knows that debian is better than fedora, so that's a done deal :-)
07:58.08echo_bgit doesnot matterr the platform
07:58.25echo_bgit does matter who's behind the KBD :)
07:58.39Silik0nard: and that bsd is better then linux ;)
08:00.39brc_gsm
08:00.43brc_linux
08:00.45brc_debian
08:01.33ardDebian!
08:01.38ard;-)
08:01.51echo_bgbrc_ - thx
08:02.06echo_bgany comment over ilbx ?
08:02.15brc_there is no ilbx
08:02.48brc_gsm > ilbc
08:02.51echo_bguff - yes
08:02.58echo_bgsyntax again :))
08:03.07brc_that's not syntax
08:03.09brc_:)
08:03.12echo_bg:))
08:03.18*** join/#asterisk RoyK (~roy@221.80-202-161.nextgentel.com)
08:03.19echo_bgokay :)
08:03.19RoyKhm
08:03.25RoyKcypromis?
08:03.32echo_bgbecause it is supported by grandstream phones
08:03.33brc_greets roy!
08:03.44brc_nothing wrong with ilbc
08:03.51RoyKexcept the license?
08:03.51brc_if you are on a lan use ULAW of course
08:03.55brc_right
08:04.13RoyKhas anyone managed to tune down the speex cpu overhead?
08:04.16RoyKor how is it?
08:04.23RoyKbrc_, s/ULAW/ALAW/g
08:04.34brc_ULAW! > ALAW!
08:04.36brc_=D
08:04.41echo_bgbrc - yes - thank you
08:04.55RoyKULAW == idiocy. ALAW == sanity.
08:04.59brc_bwhahaha
08:05.02brc_it's ONE BIT difference
08:05.09RoyKthe wrong bit :)
08:05.09brc_and alaw has the evil bit set
08:05.12RoyK:)
08:05.17echo_bghahahahah
08:05.25brc_honestly
08:05.28brc_there is a evil bit
08:05.29echo_bgyou guys are gonna laugh me to dead :)
08:05.31brc_search slashdot
08:05.35brc_they did a article on it
08:05.44brc_it was a tcp/ip thing iirc
08:06.01brc_the idea was to ask virus writers to kindly set the evil bit on their packets
08:06.16brc_then we could set QoS on evil bit to ReallyReallySlow(TM)
08:06.24RoyK/. is american. americans hava ULAW and americans always beleive everything an american is doing is a Good Thing
08:06.42brc_that is A Good Thing (TM) thankyouverymuch
08:07.26brc_uh
08:07.32brc_don't you mean a beowulf cluster?
08:07.44RoyK:)
08:07.46RoyKbeofart
08:08.33brc_wasim, !!!
08:09.55RoyKbrc_, from nothern america, right?
08:09.55*** join/#asterisk heka (~heka@82.114.68.126)
08:10.00brc_yes
08:10.10hekaany one using app_callingcard?
08:10.48RoyKthat in cvs yet?
08:10.54RoyKbrc_, whaddaya vote?
08:11.00brc_vote?
08:11.03RoyKah
08:11.19echo_bgheka - I am planing to
08:11.24echo_bgwhat about it ?
08:11.29hekaIm getting an error
08:11.34echo_bghuh
08:11.41hekaSep 28 10:11:15 ERROR[1076191872]: app_callingcard.c:881 load_module: app_callingcard: cannot connect to database server loca¨B.  Calls will not be logged
08:12.00hekaI have the correct configuration of callingcard.conf
08:12.03brc_wasim, wasim wasim
08:12.04echo_bgcheck your cdr_mysql.conf
08:12.05hekabut it wont connect
08:12.25hekacdr_mysql.conf
08:12.33hekais it needed by callingcard.conf?
08:12.42echo_bgnot sure
08:12.47RoyKbrc_, he's prolly asleep. it's like 13:12 over there now
08:12.54brc_no he isn't
08:13.00echo_bgi think that if you are usig database for cdr loging - you have to do it
08:13.05brc_yeah...it's 13:12!
08:13.14brc_he mesged me a sec ago
08:13.15hekawell Im not sure
08:13.28echo_bgme too - just suggestion
08:13.29hekahow can I stop cdr loging
08:13.29echo_bg:(
08:13.29heka?
08:13.46echo_bgwhy you need that ?
08:14.16hekaprepaid?
08:14.32echo_bgso anyway you will need a cdr records -
08:14.57echo_bghow your customer will check his call history
08:16.14*** join/#asterisk bigfoot (~simon@80.88.192.113)
08:19.05*** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com)
08:21.46echo_bgfxo is a module that allows me to attach normal phone to the device - right ?
08:22.10echo_bgand fxs allows me to attach a pots or pstn line to a device - right ?
08:23.35Jackejust the other way arround.
08:24.01echo_bg:))
08:24.08echo_bgLAME user I am :))
08:24.15Jackekinda.
08:24.37echo_bgso - fxo is for pstn/pots and fxs is for phones ?
08:24.42brc_~fxo
08:24.44jbotsomebody said fxo was foreign exchange office - or the type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx
08:24.44brc_~fxs
08:24.45jbotfrom memory, fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx
08:25.11echo_bgbrc_
08:25.18brc_echo_bg
08:25.22echo_bghow can I get the full help for jbot
08:25.26brc_can't
08:25.28echo_bg!help
08:25.29WifiFredhello, i am a bot.  my home page is http://www.bcwireless.net/wififred
08:25.30WifiFredi listen for commands prefixed with a !
08:25.36brc_~tell echo_bg about help
08:25.42*** join/#asterisk libpcp (libpcp@210.16.20.5)
08:25.45brc_okaya
08:25.45libpcpHello everyone
08:25.47echo_bgaa got it
08:25.50brc_who is running the rouge bot
08:26.19libpcpanyone got a problem with PTHREAD_MUTEX_RECURSIVE
08:26.28brc_~owner
08:26.29jbotTimRiker (or BZFlag) is my owner
08:26.36brc_!owner
08:27.08echo_bgso if i have a phoneline I have to connect it to asterisk with fxo module - right ?
08:27.10brc_Matthew_I,
08:27.13brc_~fxo
08:27.14jbothmm... fxo is foreign exchange office - or the type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx
08:32.32Jas_Williams!help
08:32.33WifiFredhello, i am a bot.  my home page is http://www.bcwireless.net/wififred
08:32.34WifiFredi listen for commands prefixed with a !
08:33.19echo_bg~cdr
08:33.20jboti heard cdr is Compact Disc Recordable, see cdrw, or a copy & crunch set of programs for ripping cds, or in telecommunications, a call-detail-record
08:34.25echo_bg~tell echo_bg about call-detail-record
08:34.35echo_bg~tell echo_bg about cdr
08:34.41wasim~no cdr is Call Detail Record, a listing of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also
08:34.42jbotokay, wasim
08:35.05wasim~no cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also
08:35.08jbotokay, wasim
08:35.08brc_wasim, fyi jbot is shared between here and #debian and others
08:35.30echo_bgis there any other billing method but cdr ?
08:35.49wasim~no cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw
08:35.50jbotokay, wasim
08:45.42wasimwhy do we share jbot with #debian?
08:45.51brc_because
08:45.56brc_I'm working on a python bot just for us
08:46.07brc_it will be ready RealSoonNow(TM)
08:46.28paulcI haven't seen RealSoonNow(tm) for ages.. well, from someone else at least.. I'm sure I typed it last week..
08:51.35manipuraanyone know what it would take to get my USB phone to work with X-lite?
08:53.35*** join/#asterisk dome_c (~dome@61.19.211.250)
08:54.12dome_chi all. i have problem with sipfriend  from database
08:57.07BaDBuGhey can anyone help me configure sip.conf? and extentions.conf?
08:57.48BaDBuGit will have 4 clients only
08:57.52dome_cwhat problem ?
08:58.07BaDBuGi am not really sure how to configure them ?
08:58.22BaDBuGi am not fimilliar with vi at all
08:58.22dome_cmake samples
08:58.33dome_cit's wll genreate you default config
08:58.53BaDBuGyeah did that but this plus vi scares me lOl
08:59.10BaDBuGjust not knowing where to start
09:00.11clive-badbug, just play with it...i for insert, then you can just edit
09:00.39clive-the esc, shift wq! to exit and save
09:01.27brc_! is bad
09:01.32dome_cWhat's best solution for 1000 user ?  ast_data Or mysql default ?
09:01.55dome_cor  default sip.conf text file ?
09:01.59clive-brx howcome?
09:02.01clive-brc
09:02.14BaDBuGi just need to have fwd configured for incoming calls and rings @ my extention in "U.S." with the ata-286. and my family have the pcphoneline usb
09:02.16clive-I get no points for all my typos:)
09:02.30BaDBuGand my bro will kall me with xten and my sis same
09:02.52BaDBuGand we all kall each other with extentions
09:03.02BaDBuGis it too bad?
09:03.11BaDBuGi mean lots of work?
09:04.50paulcbadbug should be pretty easy
09:04.57BaDBuGwanna use the 729 codec. cause it faaaar away
09:05.05BaDBuGokay i will give it a shot
09:05.06RoyKargh
09:05.10RoyKif I use sipfriends
09:05.16BaDBuGI wish man i can finish with it now
09:05.18RoyKhow can I tell * to use nat = yes?
09:05.23BaDBuGi have been reading soooo much !
09:05.42BaDBuGtrying to learn more bout linux
09:06.13dome_cin genreal config you can set nat=yes
09:06.14BaDBuGnow i am stuck trying install slakeware on my 2X raptors in raid ICH5r
09:06.20dome_croyk
09:06.47RoyKdome_c, will that work with sipfriends from mysql?
09:06.53RoyKit doesn't look so
09:07.00BaDBuGokay i have the ipcop machine as a firewall will it be better if i added a third card and have the * in the orange interface?
09:07.04dome_cme too.
09:07.19dome_cI can enable NAT in general config
09:07.30dome_cbut problem is sipaddr , port
09:07.34RoyKand it works with all sorts of Nated clients?
09:07.38RoyKrait
09:08.05BaDBuGroy you asking me?
09:08.05dome_cwhen i connect from softphone sip chan update mysql with wrong information
09:08.35RoyKI extract the userid from SIP/userid@..
09:08.56RoyKand then set the callerid according to that (to allow for independant callerid/sip name)
09:09.05dome_cNow i'm looking for new solution may be  retreive config from mysql to sip.conf
09:09.17dome_cOr ast_data from pgsql. i test it before.
09:09.31RoyKbut after a while, NATed clients tell me "SIP/sip-server-host-name@...
09:09.37RoyKinstead of SIP/username
09:09.42dome_cUhm..
09:09.49dome_cWe got same problem.
09:09.50RoyKthis is _only_ NATed clients
09:09.52RoyKok?
09:09.57RoyKthat's good, though
09:10.20dome_cI 'm working on it. i'll let's you know if i fixed it.
09:11.28RoyKdome_c, have you enabled SIP_USERS in chan_sip.c?
09:11.41dome_cNo only MYSQL_FRIENDS
09:11.54dome_cNeed to enable ?
09:12.39RoyKchannels/chan_sip.c:61
09:12.40RoyK#ifdef SIP_MYSQL_FRIENDS
09:12.40RoyK#define MYSQL_FRIENDS
09:12.40RoyK#define MYSQL_USERS
09:12.40RoyK#include <mysql/mysql.h>
09:12.40RoyK#endif
09:13.01RoyKI had that enabled but I'm not sure if it made it into the last (1.0.0) build
09:13.02dome_cI enable from Makefile
09:13.09RoyKwhatever
09:15.48*** join/#asterisk ckruetze (~ckruetze@host217-42-237-183.range217-42.btcentralplus.com)
09:17.18echo_bgguys - there is a link that is not working in voip-info-org
09:17.30echo_bgconcerning xlite and asterisk sip.conf
09:17.34echo_bgI have the PDF
09:17.43echo_bgwould you please add it to the page ?
09:18.13maruzwhich link? which page?
09:18.17echo_bgmo
09:18.18echo_bgmom
09:18.44echo_bghttp://www.voip-info.org/www.astmasters.net/stuff/X-Lite-and-Asterisk.pdf
09:20.59RoyKdome_c, works for me (tm)
09:21.35echo_bgmaruz - got it ?
09:22.14ckruetzeecho_bg there is a link o that pdf on http://www.voip-info.org/tiki-index.php?page=xten and it works
09:22.19maruzok
09:24.04maruzecho_bg: this does work: Getting started with X-Lite: A step by step guide (PDF)
09:24.28dome_cRoyk.. i'm patching ast_data
09:26.20echo_bgckruetze - thx
09:26.59*** join/#asterisk vs_ (vs@univac.spamcheck.net)
09:27.02vs_morning
09:27.27vs_asterisk -r aint able to STOP NOW
09:27.52vs_Asterisk CVS-HEAD-09/28/04-05:06:49,
09:30.52BaDBuGsorry guyz for all the question s but what do you recomned for changing sip.cof and all from the web
09:31.03vs_if remote started with -fc it does work
09:31.05BaDBuGi took a look at wiki
09:31.12vs_even w/o -f
09:31.13BaDBuGbut you guyz will know better
09:31.25BaDBuGeven if there is any GUI
09:33.40RoyKdome_c, ast_data???
09:34.46dome_chttp://svn.asteriskdocs.org/res_data/
09:40.59dome_cRoyK. it's work :)
09:41.12ScaredyCatdestar  is a GUI
09:43.43wasimand ScaredyCat is a FELINE
09:44.20wasimPrincess SmellyCat Bananhammock
09:44.33maruzthere is no link to Obelisk, or i'm blind...
09:44.42RoyKdome_c, what does it do_ the READMEs don't tell
09:45.53dome_cREADME is  Script for patch
09:46.12dome_cGive me email i'll send you asterisk-1.0.0+astdaya
09:46.25dome_cI patch already you can compile.
09:46.34dome_cbut need postgresql-devel , mysql-devel
09:49.04maruzdome_c: what have you patch in ast_data?
09:50.01dome_cnew sip-firend and iax-freind table
09:50.07dome_cstructure
09:51.48maruzdo u prefer send it by email than to give us an url where to download it?
09:52.33dome_cftp://203.146.102.2/asterisk/
09:52.59maruzi dive into it :)
09:53.01dome_cvery happy with ast_data..   :)
09:53.25dome_cbecause i plan to integreate asterisk to my billing for long time...
09:57.01dome_chave someone service DID in Japan and Sigapore here ?
10:00.50manipuraanyone know how do can make asterisk start when the computer boots?
10:01.04dome_cwhat's linux distro ?
10:01.10dome_credhat or debian ?
10:01.13manipuraRH9
10:01.31dome_ccreate /etc/init.d/asterisk
10:01.59manipuraand put what in it?
10:03.23dome_cftp://203.146.102.2/asterisk/init.d/
10:03.41dome_ccopy asteriskd start script
10:03.50dome_cand then chkconf --add asteriskd
10:04.15dncany ideas about this message:
10:04.16dncSep 28 18:10:16 NOTICE[245775]: chan_zap.c:7358 pri_dchannel: PRI got event: 8 on Primary D-channel of span 2
10:04.16dncSep 28 18:10:16 NOTICE[245775]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 2
10:04.16dncSep 28 18:10:16 NOTICE[229390]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1
10:04.17dncSep 28 18:10:16 NOTICE[229390]: chan_zap.c:7358 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1
10:04.37arddnc : zaphfc?
10:04.50manipuraAwesome, thanks dome_c
10:05.36dnchmmm i think it was a ccs vs cas problem
10:05.46dncjust checking the configs again
10:06.59dome_cbecarefull manipura it's virus he he
10:07.01dome_c:)
10:07.53echo_bgdome_c what is virus ?
10:08.04ScaredyCatdnc: check you aren't getting irq misses
10:08.10dome_csorry  it's joke.
10:08.17dome_c<dome_c> ftp://203.146.102.2/asterisk/init.d/
10:08.17dnchey ScaredyCat :)
10:08.24ScaredyCatlo dnc
10:09.59ScaredyCatI bet it's a 1u box isn't it dnc
10:10.25dncerr actually its a tower case
10:10.32ScaredyCatoh..
10:10.59ScaredyCattry accessing the disk while in a call... see if that causes the messages to appear
10:11.12Jas_Williamsdnc: Looks like a config issue, is this euroISDN or T1 ?
10:11.21ScaredyCatseen it b4, irq misses caused by card shared with disk
10:11.21dncE1 but not euro isdn
10:11.23ScaredyCatirq
10:11.27dnci think it was the ccs, cas thing that did it
10:11.49dncits ok now ive changed over to cas
10:11.56ScaredyCathmmm...
10:11.58dncjust making some test calls now to be sure
10:12.27Jas_Williamsdnc: As far as I know we do not support CAS over E1 currently so span needs to be css not cas
10:12.37Jas_Williamsoops ccs
10:12.48dncreally?
10:12.51dnchmmm
10:12.57dncwell lets see how the test calls go
10:13.29*** join/#asterisk lseror (~lseror@217.174.204.42)
10:13.47lserorHi
10:14.01lserorAnyone know if we can filter incoming call with asterisk ?
10:15.26Jas_Williamslseror: Yes what do you need to do ?
10:17.48manipuradome_c that script didn't work, is there something I'm missing?
10:18.05dome_cwhat's error ?
10:18.35dome_c[ -x /usr/sbin/asterisk ] || exit 0
10:18.35dome_c[ -d /etc/asterisk ] || exit 0
10:19.00dome_cIs you asterisk  install in /usr/sbin ?
10:19.02manipurano error, it just wasn't started when I rebooted
10:19.20dome_ccan you test /etc/init.d/asteriskd
10:19.31dome_cby manual first
10:19.35manipurak, gimme a sec, its rebooting
10:19.41dome_ck
10:20.04*** join/#asterisk Ngy_Dave (~davy@213.219.141.105.adslpower.by.edpnet.be)
10:21.18manipurayeah, I can
10:21.38dome_cadd to default by
10:21.58dome_cchkconfig --add asteriskd
10:22.19dome_cOr setup -> system service
10:23.21manipuraYahoo, its 4:20AM
10:25.28RoyK~seen cy
10:25.29jbotcy <~darknovae@63.225.225.227> was last seen on IRC in channel #utah, 115d 17h 29m 20s ago, saying: 'night all'.
10:25.30RoyK~seen cypromis
10:25.30jbotcypromis is currently on #asterisk.  Has said a total of 43 messages.  Is idling for 4h 35m 59s
10:27.09manipuraawesome, got it working, thanks dome_c, I'll remember that chkconfig command for sure!
10:27.37dome_c:)
10:28.46lserorJas_Williams : I want to blacklist someone who is sending a fax every hour to my phone :)
10:29.39Jas_Williamslseror: Based on cli you can send call to busy or other location
10:30.26lserorJas_Williams: what document should I read ?
10:32.04*** join/#asterisk inspired (mikael@a217-118-63-4.bluecom.no)
10:34.44*** join/#asterisk glLoadIdentity (~tuyan@dsl81-214-9283.adsl.ttnet.net.tr)
10:37.30lserorIn fact what's i'm looking for is some kind of PBX Firewall, to filter incoming call by example
10:39.17salimfadhleyAnybody know of some high-quality asterisk logo artwork available?
10:43.14wasimmamboserver.com
10:43.16wasim:P
10:44.08salimfadhleymambo... :-(
10:44.11salimfadhleyzope :-)
10:44.40wasimyeah, but mamboserver logo passes for a neat * logo
10:45.07salimfadhleyoh... yes... well I kind of like the real one. I just want to find something thats nice enough to make a 16x16 asterisk icon in my zope application.
10:45.25salimfadhleystill... it might fool the masses
10:45.26salimfadhley:-)
10:47.01fileI'm so comfy in bed, I don't want to go to school
10:48.45*** join/#asterisk sob0l (~devil@80.51.169.19)
10:50.33manipurafile, you have a computer in your bed?
10:50.35manipuraI want one
10:50.55nextimemanipura : laptop&wifi :)
10:51.29filenextime: indeed
10:51.40manipurafile, send me yours
10:51.42ScaredyCat<PROTECTED>
10:51.43fileand now I must depart, thankfully I have a substitute for english... so yah!
10:51.46filemanipura: ha
10:51.53echo_bgguys - gotta go -
10:51.54manipurafile, hope she's hot!
10:51.56echo_bgc ya soon
10:52.04*** part/#asterisk echo_bg (~asdasdasd@212.116.151.30)
10:52.10Jas_Williamslseror: Can you post the relevant section from your extensions.conf for the incoming fax call to pastebin.ca and then I could come up with some sort of blocking config
10:53.22*** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl)
10:53.26PETER22quick q. - if i have an incoming zap pstn call, can i make it ring on combination of zap fxo and sip channels ?
10:55.16*** join/#asterisk postel (~postel@host217-42-116-153.range217-42.btcentralplus.com)
10:55.47Jas_WilliamsPETER22: Yes exten => s,1,Dial(SIP/1234&Zap/1) but the zap channel would always except the call unless you are using ISDN/T1/E1
10:56.30Jas_WilliamsPETER22: An fxo Zap call is considered answered as soon as dialled.
10:56.42PETER22thanks jas_williams
10:57.27PETER22i think i have to make it ring sip first, then after timeout go dialing zap
10:58.05Jas_WilliamsPETER22: That will work.
10:58.53lserorJas_Williams : What is pastebin.ca ?
10:59.19lserorJas_Williams : I found something about Call Blocking on Caller ID
10:59.25ardpastebin
10:59.35PETER22anyway to keep orginal persons caller id on a call accepted by *, then forwarded out zap fxo ?
10:59.36lserorJas_Williams : It says that I should use the dialplan
10:59.40clive-iseror look at ex girlfreind logic
10:59.42ardjbot : what is pastebin
10:59.47jbotfrom memory, pastebin is a place to paste all your conf/debugs/logs for other people in the chatroom to view without flooding the channel. We suggest http://pastebin.ca
11:00.43lserorok I found pastebin
11:03.51Jas_WilliamsPETER22: No you cannot maintain CalerID the caller ID displayed will be that of the fxo.
11:04.06*** part/#asterisk dome_c (~dome@61.19.211.250)
11:04.09manipurajbot : what is a dead hooker
11:04.10jbotI think you lost me on that one, manipura
11:05.14manipurais there a list of what jbot knows?
11:05.24lserorJas_Williams, I pastebin it
11:05.52manipuraIseror, http://pastebin.ca/1147
11:07.01manipurajbot what is asterisk
11:07.02jbota PBX (Private Brance eXchange) and telephony toolkit. http://www.asterisk.org, or very cool, or awesome
11:07.18lseroryes this is my post, 1147
11:08.22manipuraHey EVYERONE http://pastebin.ca/1146
11:08.24Jas_Williamslseror: How are the calls delivered to *
11:10.35PETER22i have heard ISDN can bounce incoming calls to a diversion without answering ?
11:10.38lserorJas_Williams : They are directly send to Zap
11:11.47PatrickDKwhat username are you using in sip.conf?
11:11.50PETER22I have a main business pstn line - i use my carrier "divert all" feature and i get the original persons caller id (important)
11:12.42PETER22i would like these calls to be handled by * in some way (if i'm in the office)
11:17.54lserorJas_Williams : it seems that there is a blacklist database implemented in Asterisk
11:18.34_dwjbot_: forget asterisk
11:18.54_dwjbot_: asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org, or very cool, or awesome
11:19.00_dwjbot_: what is asterisk?
11:19.19_dwif they are not the correct commands, can someone fix the spelling mistake. its bloody shameful
11:19.27_dwjbot what is asterisk
11:19.28jbota PBX (Private Brance eXchange) and telephony toolkit. http://www.asterisk.org, or very cool, or awesome
11:19.42manipurajbot can you die?
11:19.49_dwjbot asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org, or very cool, or awesome
11:19.50jbot...but asterisk is already something else...
11:19.56_dwjbot forget asterisk
11:19.56jboti forgot asterisk, _dw
11:20.00_dwjbot asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org
11:20.01jbotokay, _dw
11:20.06_dwjbot what is asterisk?
11:20.09jbotwell, asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org
11:20.09manipurajbot what is google
11:20.11jbotrumour has it, google is a search engine found at http://www.google.com/
11:20.29manipurajbot what is jbot
11:20.31jbotjbot is, like, ibot's stupid cousin
11:22.17*** join/#asterisk ckruetze (~ckruetze@host217-42-237-183.range217-42.btcentralplus.com)
11:25.37BuzzBudAny reason not to noload => chan_skinny.so ?
11:29.22lserorOK i think I find how to do it
11:29.23*** join/#asterisk sid (~sid@gethsemane.odgers.id.au)
11:29.26sidlo guys
11:29.27lserorit is working
11:29.39lserorbrb
11:30.08sidis there a way you can tell asterisk to provide dialtone to the caller when a particular extension is called?
11:30.14sidlike a redialler?
11:30.45sidi've got iaxtel's free 1-800 stuff set up on my asterisk box, and want to give a couple people access thereto via pots
11:31.07miller7sid: I think that there is, let me check my notes, I have this from a long time now
11:31.17sidmiller7: ta :)
11:36.06*** join/#asterisk dabba (~d@matrix.lgw.ip6net.net)
11:40.17manipura~seen dabba
11:41.34jbotdabba is currently on #asterisk (5m 28s)
11:41.34sid'lo manipura :)
11:41.59manipurai've always wanted my own robot
11:44.32ckruetzedabba, here in Cambridge it isn't cold, but a hooverbot would be cool
11:45.09dabba$249 and it just wandered around hoovering, my front room is too small though :-(
11:45.10sidit's cold here in melbourne.
11:45.11sidvery very cold.
11:46.48manipuraWe are getting great weather in calgary
11:47.01ckruetzesid, then move up north
11:47.34sidnah, no work up that way
11:47.58Dibbler_Cold in Melbourne, that sounds wierd to us Brits, who generally always think of Oz as being sunny and warm
11:50.57sidabout 12 and torrential today :/
11:51.42*** join/#asterisk fool (~dwinter@67.106.194.90.ptr.us.xo.net)
11:52.25Dibbler_12 isn't cold, hehe
11:53.03lserorAnyone want to know how you blacklist a caller-id ?
11:53.24sidset up an extension in the correct context which specifies it, pointing somewhere dumb?
12:00.55*** join/#asterisk autobus (~autobus@80.172.12.87)
12:02.05*** join/#asterisk |Vulture| (~Vulture@247.131.vbnet.net)
12:03.05autobusas it is that I can copy the filing-cabinet automatically a filing-cabinet of /var/spool/asterisk/callback.call for = "/var/spool/asterisk/outgoing/
12:03.49autobuswhen to cal for a extension
12:04.27autobus?