irclog2html for #asterisk on 20060325

00:00.39Dr-LinuxSkid: all i mean to ask,  i wanna do same, like i wanna allocate a few numbers to my client, with IP and port
00:00.46Dr-Linuxas my provider did
00:00.57SkidI don't know really..
00:01.01Skidsorry
00:01.07Skidask him? :)
00:01.38Dr-Linuxhe doesn't tell me
00:01.58*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
00:02.23Dr-LinuxSkid: and i was not given any username or password :S
00:02.42Skidi really dont know to tell the truth
00:03.26Dr-Linuxhhm... :S
00:06.06GrizzyI wish either asterisk regular expressions were unix filename or unix or perl regular expressions, OR that I could find a complete spec on them.
00:07.52Skidhttp://www.voip-info.org/wiki/view/Asterisk+Expressions ?
00:07.54Skidno good?
00:08.39GrizzyI don't see it on that page
00:10.46Grizzythere's a bit of a blurb on p 27 of the handbook, but I don't think it's complete (4.1.4)
00:10.56jbalcombi've just set up Asterisk. when i pickup the phone i get a dial tone, i dial 500 for the demo and the line stays up but i don't hear anything. what should i be looking for?
00:13.09GrizzyThanks for hints, Skid.
00:17.17[av]baniwhats this thing cisco calls "ghost digits" ?
00:19.45Jon335I have the same question as lunaphyte: What is considered a good rate for a Toll-Free DID?
00:19.59*** join/#asterisk iq|mobile (n=iq@71-214-5-20.omah.qwest.net)
00:20.15AlexCTIThere someone familiar with pri_net?
00:27.36*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
00:33.23PrimerI simply cannot get this cisco ata to get its sip settings from the file it sets over tftp
00:33.26Primerany of its settings
00:34.47*** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net)
00:35.07*** part/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net)
00:37.21*** join/#asterisk malverian (n=malveria@adsl-065-005-207-210.sip.gnv.bellsouth.net)
00:42.03*** join/#asterisk _Simon (n=IRC@i216-58-40-193.cybersurf.com)
00:42.38*** part/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it)
00:42.57_Simonhey gang. I'm using IAX2 softphone and hardphone. Even using echo test as a test case, I have to call 3-4 times for it to work. is this an IAX2 NAT issue? Why does it randomly pick up but many times not? Anyone know how to resolve this?
00:43.29*** part/#asterisk Priam (i=mike@towerravens.com)
00:46.17*** join/#asterisk rogercharlie (n=rogercha@c-69-181-20-122.hsd1.ca.comcast.net)
00:50.28*** join/#asterisk pigpen2 (n=mark@66.118.8.74)
00:50.28*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
00:51.43Jon335What is considered a good rate for a Toll-Free DID?
00:51.52pigpen2Hi all.  Quick question:  Using asterisk 1.2.4 with Polycom 601's - How can I disable the notification "beep" when a second call comes in?  This is kinda a pain when receiving credit card numbers....
00:52.40pigpen2Jon335, Nuphone was doing it for .02/min....
00:52.48pigpen2Last time I looked.
00:53.28_Simoncan anyone help with my IAX2 issue please? just looking for some ideas. I thought it was my soft phone but the hard phone is doing the same
00:53.35_Simonwould it be the PBX or the network itself?
00:53.53*** join/#asterisk hfern (n=hfern@h-64-105-50-227.dllatx37.dynamic.covad.net)
00:54.13pigpen2I am quite busy, but I might be able to help.
00:54.14Jon335pigpen2: What about a Canadian Toll-Free DID?
00:54.23QwellJon335: about 10x that much
00:54.31pigpen2heh...I have heard Canada is much higher.
00:55.04_Simonpigpen2: were you speaking to me or Jon, sorry just don't wanna get confused lol
00:55.13pigpen2_Simon, well, your resolution may be several things...but a hint of what is going wrong is a good idea....
00:55.19pigpen2you.
00:55.27Jon335pigpen2: Unlimitel is 4c/min, I think that is the best
00:55.32_Simonsure, I mentioned it earlier, basically what is happening is:
00:55.39_SimonI'm using IAX2 softphone and hardphone. Even using echo test as a test case, I have to call 3-4 times for it to work. is this an IAX2 NAT issue? Why does it randomly pick up but many times not? Anyone know how to resolve this?
00:56.00*** join/#asterisk angom_h (n=angom@red-corp-200.79.145.199.telnor.net)
00:56.00*** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal)
00:56.01_Simonmost of the time the call doesn't even register on the pbx console
00:56.01pigpen2well, nat usually doesn't affect iax....
00:56.08_Simonpigpen2: yeah thats what I thought
00:56.17pigpen2having packet loss?
00:56.20Heimidalwhat codec is ideal?
00:56.20_Simonbut if the asterisk console doesn't output anything
00:56.21pigpen2high latency?
00:56.22*** join/#asterisk jahani (n=k@adsl-240-241-192-81.adsl2.iam.net.ma)
00:56.28_Simonshouldn't be high latency
00:56.35pigpen2Heimidal, depends...how much bandwidth.
00:56.49pigpen2well, if it is so high it doesn't get there....
00:57.06HeimidalI'd like to be able to have 4 channels on half a T1
00:57.10_Simonit should get there no prob
00:57.24_Simonits on a server external of my lan, but the server is on the same ISP as me
00:57.27_Simonso should be pretty fast
00:58.28_SimonI also set the IAX2 port as high QoS on the router
00:58.59pigpen2well, like you said, if the asterisk cli never shows a connection...then...well...no call...
00:59.07pigpen2the packets may not be getting there.
00:59.15_Simonpigpen2: when the call *does* work the audio is fine with no interruption
00:59.25_Simonso I would assume the connection is working well if that is the case
00:59.38pigpen2qualify set?
00:59.48pigpen2qualify=yes or qualify=500 ?
00:59.48_Simonumm whats that? lol
00:59.59pigpen2check it out in the wiki...
01:00.04_Simonk
01:00.06pigpen2but just stick it in the iax config.
01:00.28pigpen2this way your server keeps up with the device....with regular check in's  (the way I understand it)
01:00.31*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
01:00.57Skidho hum
01:01.00Skidany dev's around
01:01.05SkidI think i've found a bug in 1.2.5
01:01.06QwellSkid: always
01:01.18pigpen2crap...it is Qwell....
01:02.28*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
01:02.59pigpen2Qwell, can I ask you a quickie?
01:03.03_Simonbandwidth=low
01:03.03_Simontos=lowdelay
01:03.06_Simonare those okay?
01:03.43Qwellpigpen2: just ask, if I know the answer, I'll tell you
01:03.55Qwell(if it isn't too involved, that is)
01:04.07pigpen2<PROTECTED>
01:04.17pigpen2Other than that...everything is great.
01:05.23Skidit seems mixmontior has some issues already too
01:05.25Skiddoh
01:05.32Skidi like that app :)
01:05.38Qwellpigpen: I didn't know the answer 10 minutes ago when you asked, and I still don't
01:05.45QwellIf somebody knows, they'll answer...
01:05.50pigpen2thank...me too...
01:05.57pigpen2sorry..I didn't know you were watching.
01:06.05pigpen2thanks.
01:06.06QwellI'm always watching
01:06.16pigpen2good to know.
01:06.21Grizzythe eye of god. : o )
01:06.46QwellI lurk - it's what I do
01:07.07GrizzyWhat chat client do you use, Qwell?
01:07.18Qwell/ctcp version me
01:07.34Skidho hum, so can anyone see a problem with exten => s,9,MixMonitor(${DATETIME})
01:07.41Skid(basically kills the cLI)
01:07.59Skidcall's been answered and is in a dialplan, basically
01:08.22Skidbefore i go submitting a bug which may not be a bug
01:08.29Skiddoesn't do it if i just mixmonitor(recording.wav) say
01:08.37Skidbut does with the datetime var
01:08.39Grizzyqwell - &*&%^& GAIM, I can't figure out how to CTCP Version.
01:08.54Skid01:09 [freenode] CTCP VERSION reply from Qwell: xchat 2.4.5 Linux 2.6.15-gentoo-r4 [x86_64/2.22GHz/SMP]
01:08.57Skid:)
01:09.05*** join/#asterisk ambriento (n=ambrient@www.cobranet.com.br)
01:10.07GrizzyIt beats 6 open chat processes (to go with my 3 sipphones), but it's pretty incomplete.
01:10.50*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
01:12.40*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
01:16.14Splatif I set a sipura/linksys spa3000 to use asterisk as an outbound proxy server does that mean it will ignore the dialplan settings it has and just go according to asterisk's dialplan?
01:16.34tehdelyeric_hill: pridialplan=unknown isn't helping
01:16.40tehdelyi set the span on intense debug
01:16.44tehdelyand i'mseeing absolutely nothing
01:16.48tehdelywhen an inbound call is coming
01:16.59tehdelysomething must be misconfigured on the switch
01:17.07tehdelyi have a feeling this ball is in the CLEC's court
01:17.13tehdelyunless there's something else i could have overlooked
01:17.30tehdelyhe claims asterisk is returning an error on INVITE but why wouldn't i at least see evidence of the failed INVITE
01:18.13GrizzyI love the phone company so much.  (NOT)
01:18.56*** join/#asterisk hfern_ (n=hfern@h-64-105-50-227.dllatx37.dynamic.covad.net)
01:20.29tehdelydoes anyone see any evidence of a failed incoming call in this debug output: http://pastebin.com/621008
01:20.39tehdelynote that outgoing calls over this span are working perfectly
01:20.44tehdelybut incoming ones seem to go straight to /dev/null
01:20.47tehdely(caller hears a busy signal)
01:27.29*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
01:30.18*** join/#asterisk hfern (n=hfern@h-64-105-50-227.dllatx37.dynamic.covad.net)
01:32.53*** part/#asterisk Utah_Dave (n=boucha@0-1pool149-100.nas31.salt-lake-city1.ut.us.da.qwest.net)
01:33.04*** join/#asterisk juice (n=juice@mo-69-68-106-145.dyn.sprint-hsd.net)
01:38.29rogercharliedoes the flash proxy work for all manager connections or just flash clients?
01:39.23*** join/#asterisk riddlebox (n=james@24-207-158-49.dhcp.stls.mo.charter.com)
01:40.21*** join/#asterisk websae2k (i=websae@CPE-24-167-204-30.wi.res.rr.com)
01:40.37rogercharlieso many fun things use the manager API
01:41.31websae2kanyone need carrier services :)?
01:41.57Strom_Cwebsae2k, please stop spamming the channel
01:42.17websae2kjust trying to help a fellow asterisk user(s)
01:42.39Strom_Cput your ad on the voip-info wiki
01:42.43Strom_Cdon't spam the IRC channel
01:43.21Heimidalfor what reason would 's' not match for a given number? I keep getting "1112223333@demo doesn't exist" when trying to use the demo context
01:43.34Heimidal(replace 1112223333 with my real number, of course
01:43.36QwellHeimidal: When the number isn't s
01:43.36*** join/#asterisk CrashHD (i=CrashHD@c-24-7-168-46.hsd1.ca.comcast.net)
01:43.51HeimidalQwell: isn't s supposed to mean "start"?
01:43.54Qwellyes
01:44.04QwellIf you want to match your number, put in a pattern that matches it
01:44.10Abydos313evening everyone
01:44.15Heimidalwell, I don't want to match my number
01:44.19Heimidalit just keeps saying that
01:44.27CrashHDhey fella's. I have two * boxes setup for trunking, which always fail to natively bridge a call. is there a way to disable this functionality?
01:44.28QwellYou obviously do want to match your number
01:44.29HeimidalI'm just wanting to call in and have it go to the demo
01:44.57*** join/#asterisk Lino` (n=Lino@i577BDE76.versanet.de)
01:45.12Heimidalok, I don't want it to match my number explicityluy
01:45.17Heimidal*explicitly
01:45.24QwellWhat type of line?
01:45.55*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
01:46.13Heimidalit's configured to register with Teliax, if that's what you mean
01:46.17HeimidalI'm a bit new to this :)
01:49.05Lino`~seen Possible
01:49.09jbotpossible <n=Babbel@23.255-136-217.adsl-fix.skynet.be> was last seen on IRC in channel #asterisk, 14d 13h 29m 48s ago, saying: 'I guess not'.
01:49.15Lino`hm
01:49.29Lino`cisco systems is the worst buerocracy ever.
01:50.01Lino`in order to buy a snt contract i need to send in an invoice, the license, the serial number and a photo of the hardware (!) wtf?
01:50.29*** join/#asterisk Strom_C (n=strom@66.159.243.59)
01:54.24CrashHDwhat dial option do you send to disable native bridging?
01:58.26rogercharlieDoes the FOP work as a manager proxy for programs other than FOP?
01:58.55rogercharlieI am having no luck sending things to the 4445 FOP port instead of the 5038 port
02:00.27Lino`hmmm
02:00.39Lino`rogercharlie: if it would work it would be pretty insecure huh?
02:01.51rogercharliewell locking down and configuring would be next
02:03.20Lino`no
02:03.27Lino`i mean like if you could use FOP server as a proxy
02:03.44Lino`then "almost everybody" could play the manager for you ;)
02:03.58rogercharliewell under manager proxies it is listed
02:04.06Lino`"hello i'm another dirty little hacker and i'll be your manager for today *transfers call*"
02:04.26Lino`quite a funny thought *g*
02:04.37rogercharlieyah it would be easier than getting a smarnet contract
02:04.48Lino`oh i hate cisco.
02:05.21Lino`im trying to give money to them. i asked several resellers if they could sell the contract
02:05.46*** join/#asterisk trbldwine (i=trbldwin@c-71-194-161-170.hsd1.il.comcast.net)
02:05.58Lino`result: one berlin based reseller laughs about me, they printed out my request email framed it and put it on the wall
02:06.09rogercharliethe contract is under 10 so no one cares
02:06.10Lino`just because i sent an email 3 oclock in the morning
02:06.26Lino`yeah but i have 50 phones
02:06.26Lino`:D
02:06.45Lino`and i have a customer who has another 100 and no contract
02:07.13Lino`what i did now is i ordered one single contract from an american reseller
02:07.31Lino`6$ for the contract, 66$ for the shipping
02:07.38*** join/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net)
02:07.50Lino`if its the right contract i'll buy the amount needed, if not then i wont
02:07.51Lino`:-P
02:08.13rogercharlieat least the phones are nice
02:08.41rogercharliebesides no deny call button, and a hidden DND button, and no presence, they rock
02:08.56Qwellsorry, no presence?
02:09.00blitzrageQwell: !
02:10.07Lino`yeah
02:10.14Lino`if you dont have 7961 or 7970
02:10.28Lino`the bare 7960 does not have those lit line buttons
02:10.36QwellLino`: nor do they need them
02:10.44Lino`but with my 7970ies and 7961s the hint stuff works
02:11.00Lino`maybe its because the 7960ies are SIP
02:11.07Lino`and the 7961 7970 are sccp
02:11.09Lino`:-P
02:11.23rogercharlieyahsers no SIP love
02:11.37Strom_Cthere's apparently now SIP firmware for the 7970
02:11.41Lino`i prefer sccp, i have several 7961, 7970 and 7920 with it
02:11.54Lino`but sccp is better than sip, why downgrade?
02:12.08Lino`(ok you'll need an extra module because skinny sucks)
02:12.17QwellLino`: I'm working on that
02:12.24*** join/#asterisk ravenpi (n=chatzill@londonderry-cuda3-24-51-50-156.lndnnh.adelphia.net)
02:12.29Lino`on what? @ Qwell
02:12.47Qwellchan_skinny
02:12.58Qwelltoday, I got it to actually like...do stuff :P
02:13.42Strom_Cheh
02:13.43Lino`yeah, i hate it
02:13.43rogercharlieI miss critch on the mailing list
02:13.43Lino`when i try to connect a 7920 to chan_skinny
02:13.43Qwellwell, Sergio is a prick, so...yeah.  Fuck chan_sccp
02:13.43Lino`my asterisk crashes
02:13.43Lino`let him be a prick, let him be something different, as long as the software works ;)
02:13.45QwellLino`: http://bugs.digium.com/view.php?id=6772
02:14.31Lino`thats what a prick has to look like
02:14.39Strom_Cthat would somehow imply that ballmer has style to begin with
02:14.42QwellLino`: That'll get * to not crash.  It still won't like...work, but...
02:15.06QwellLino`: If somebody were to donate a 7920 to the cause, I'd be sure to get them working too. :P
02:15.41Lino`hey ;)
02:15.52Qwelland actually, I am serious...
02:16.15QwellI need to get/borrow various sccp phones, to test them all out.  See if there are any quirks
02:16.23Lino`where are you from?
02:16.28Qwellsouthern CA
02:16.38Lino`kk
02:16.47Strom_CQwell, I'd donate, but all I've got are 7960s :)
02:16.52Lino`too far away to throw a 7920
02:16.55QwellStrom_C: yeah, I've got 7960's to test with
02:17.03QwellThat's all I've got though
02:17.04X-RobI don't!
02:17.15Lino`lol
02:17.20Qwell7941/7961 would be nice
02:17.31Strom_CI forget Qwell - where are you again?  San Diego?
02:17.32Nuggetit's all qwell's fault
02:17.35Qwelland some of the cheap stuff, like 1912
02:17.37QwellNugget: ;)
02:17.40Lino`today i had to make a screenshot of my working asterisk test installation for a new firmware version... but how to make a screenshot of the console?
02:17.51QwellStrom_C: about 15 miles from downtown LA
02:17.51Lino`7912
02:17.57Qwellright
02:18.04*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
02:18.05Strom_CQwell, oh sweet.  we should have lunch sometime ;)
02:18.05*** join/#asterisk Nodren (n=nodren@64.193.95.10)
02:18.09Lino`cheap stuff is the exact word for it
02:18.22Qwelland a 7985...
02:18.26Lino`huh
02:18.29Lino`that conference thingy?
02:18.34QwellIf somebody gives me a 7985, I'll get video working, for sure. :P
02:18.39Lino`no video
02:18.47Qwellyes video
02:18.52Lino`ah that video conference stuff
02:19.01Lino`its ugly
02:19.01Qwellno, just a video phone
02:19.05Strom_CVidephones are the wave of the future!  </1971>
02:19.08Lino`i dont want it.
02:19.40Lino`oh i remember, when I signed a contract with "Deutsche Bundespost" back in 14 years ago for a single ISDN line (BRI)
02:19.53Lino`they were like buy a videophone, in 5 years everybody will have one
02:20.16Nuggetheh
02:20.26Lino`today the ISDN standard has changed, the videophones sold back then are useless and they were fricken expensive.
02:20.45Lino`thank god i didnt buy those ;)
02:21.11Strom_CYou know what's totally mad?  I have three telephones on my desk and I'm considering moving stuff to a different desk so I can put more telephones on my desk
02:21.29QwellStrom_C: You need at least 6
02:21.45Qwellwith no fewer than 2 in use at any given time
02:21.52Strom_CI already have fourteen line appearances within arm's reach
02:22.20*** join/#asterisk trbldwine (i=trbldwin@c-71-194-161-170.hsd1.il.comcast.net)
02:23.49Lino`lol
02:24.26Lino`i'm happy with 2*7960, 1*7970, 1*Ascom Eurit 40 and 2 non working 7920 on my desk
02:24.42Lino`and a cd burning robot which makes crummy noises
02:25.04QwellLino`: send me a 7920, I'll get it working. ;)
02:25.15Qwell..eventually
02:25.21Lino`well, actually
02:25.35Strom_Cive got 2 7960s, a Nortel 9417CW, a turquoise Western Electric 2500, and a bright orange rotary Stromberg-Carlson Slenderet hanging around my desk
02:25.43Lino`i'll try to get a working call manager ;)
02:25.52Lino`nortel
02:25.55Lino`meridian?
02:26.03Strom_CLino`, it's an analog desk set
02:26.21Lino`ah ok
02:26.50Lino`i spent new years eve in a datacenter of a big bank 1000km from here, they had those really huge nortel phones
02:26.54Lino`butt ugly
02:27.04Strom_Cwhich "really huge nortel phones"?
02:27.40Lino`i am looking for the number
02:27.53Lino`http://paragonnt.com/products/MVC-017L.jpg <= those with a display extension right next to it
02:27.56bugzi like my ip601
02:27.59Lino`like i said, butt ugly
02:28.04bugzi get to watch everyone
02:28.33Strom_CLino`, that's not so bad
02:28.34Lino`ip601?
02:28.46bugzwith an expansion module
02:28.50Heimidalcan someone tell me why this causes the error "Mar 24 20:28:23 NOTICE[4193]: chan_iax2.c:7198 socket_read: Rejected connect attempt from 207.174.202.3, request '312546XXXX@incoming' does not exist"? http://rafb.net/paste/results/RceagI47.html
02:29.19Lino`well they are switching to cisco throwing away (or maybe selling) 2500 phones.
02:29.32Lino`every single one is this butt ugly phone with extension
02:29.51bugzhaha i got some nasty phones off a russian oil rig
02:29.56bugzthese things were like 30 years old
02:29.57Lino`huh?
02:30.01Lino`russian phones o_O
02:30.06Lino`2 line interface?
02:30.07bugzyeah, get this
02:30.13Lino`2 wire
02:30.23bugzthey had this box marked KGB on the deck in the electrical closet
02:30.31Lino`lol
02:30.56bugzit was made of this WWII loking metal box with all these steel conduits and stainless steel buttons
02:31.04Lino`http://www.polycom.com/pw_files/SP_IP601_3ExpMod_Left.gif <= you mean this phone?
02:31.11bugzthat thing could survive a direct hit
02:31.17Lino`http://www.lino-helms.com/kiste1.jpg <= boxes like those?
02:31.31Lino`(i use those to store cd / dvd media *lol*)
02:31.35bugzthats the ip601
02:31.40justinubugz, got any pics of those russian phones?
02:31.53bugzman no
02:31.55bugzbut i can get some
02:31.59bugznow that i think about it
02:32.02justinucool
02:32.02bugzthe rig is still docked
02:32.11justinui like funky electronics from soviet russia
02:32.12Lino`go and steal 'em
02:32.23bugzhaha, they were like "hey you dont got in there!"
02:32.34Lino`in soviet russia phone calls you
02:32.35justinubugs, you live in russia?
02:32.37justinuheh
02:32.39bugz"WOW DUDE LOOK AT THIS SHIT!"
02:32.41Heimidalno one?
02:32.43bugzahaha..
02:32.43bugzno
02:32.46bugzi live in Houston
02:33.05Lino`well houston is not that far away from soviet russia ;)
02:33.14Lino`i used to live in san antonio :p
02:33.24bugzyeah, in Houston the Hurricane does YOU!
02:33.34justinulol
02:33.36Lino`lol
02:33.46bugzi put a 2 u server on it to drive a bunch of ip301's
02:33.57justinucool
02:34.02Lino`i went to corpus christi once, twc announced a hurricane
02:34.03bugztheres like this regulation now where those rigs need a phone like every 10 feet
02:34.04justinuhow they like the new phones?
02:34.16bugzthey were tripping
02:34.27bugzthe fax to email thing blew their mind
02:34.29Lino`as soon as i arrived they started the announcement
02:34.33justinuheh
02:34.39Lino`and all those rednecks started boarding up their windows etc.
02:35.01bugzlol
02:35.05bugzsome of them anyway
02:35.11_Simonpigpen: are you still there?
02:35.15bugzthe ones i know down there go fishing and surfing during hurricanes
02:35.35Lino`thats a once-in-a-lifetime experience - i'll never forget that... cheap houses made of wood which will be blown away anyway but lots of boards in front of the windows and duct tape (!!!)
02:35.53_Simonhey gang. I have to redial iax like 3-4 times to get a call to actually connect. even with echotest, this is on iax hard and soft phones. I am getting these when I do iax2 debug: http://pastebin.ca/
02:35.55_Simonwhat does that mean?
02:36.00bugzi like the whole "im not going anywhere" attitude
02:36.05Lino`as if duct tape would protect them from anything...
02:36.11Luke-JrAnyone here using SellVoIP and supporting a guest IAX account?
02:36.32justinu_Simon: that's not the right link for your paste
02:36.39_Simonops
02:36.41_Simonsorry :)
02:36.46bugzwe got a nice embedded system out now...
02:36.51_Simonhttp://pastebin.ca/46901
02:37.02_Simonjustinu: thanks ;) that line ^^
02:37.02bugzit does all this freaky wanpipe stuff out of the box
02:37.02_Simonhehe
02:37.11Lino`i just thought about how it would be like if cisco had stores
02:37.21Lino`like a seven eleven
02:37.24justinunot sure... i'm not an iax expert
02:37.28justinuwhat does INVAL mean?
02:37.34bugzi would get drunk and do a router run
02:37.40_Simonnot sure. its weird. if I redial and redial over and over it eventually works
02:37.42Lino`like in futurama with that conveyor belt on the ground
02:37.54_Simonif I reset asterisk, it works right away for the first 5 or so times, then starts intermittently do that
02:37.59Lino`"you know our policy is, if you are not 100% satisfied..."
02:38.10X-Rob_Simon, sounds like an * bug.
02:38.10justinu_Simon: you using qualify?
02:38.14Lino`"... i hate you. *pushs button* *customers are taken away by belt*"
02:38.33_Simonjustinu: someone mentioned it before. I set qualify=yes in the iax accounts. i thought it fixed it,b ut really just resetting asterisk helped
02:38.40justinu_Simon: is there a nat involved/
02:38.40Lino`hmmm
02:38.42justinu?
02:38.44_Simonafter 10 mins or so, all the problems again
02:38.45_Simonyes
02:38.53_Simonbut IAX is supposed to be nat traversal?
02:38.54justinuis SPI (stateful packet inspection) turned on?
02:39.06_Simonumm not sure, its a linksys
02:39.09justinusome cheapy linksys/dlink/netgears offer "SPI", but it really just fucks things up
02:39.10Lino`looks like your firewall is trying to kill you.
02:39.17justinui bet it's a firewall problem, simon
02:39.20justinufirewall/nat
02:39.34_Simonbut those iax2 debug errors, that means its receiving something?
02:39.47Lino`yeah, at home i have a wrt54g. that little sonofabitch hates my 7960
02:39.50*** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net)
02:39.50bugzi like this asterisk bug: "when i check my voice mail 'the voice' stutters the date.."
02:39.52justinuturn on iax2 debug when you first start *
02:39.54justinulet it run
02:40.03_Simonthats what I did
02:40.16_Simonit works great for 5-6 calls or whatever
02:40.18justinuyour paste didn't seem to have that much
02:40.19_Simonthen all the sudden that junk shows up
02:40.21_Simonlol
02:40.23bugzanyone have any snort rules for sip over udp
02:40.26_Simonno I just pasted the errors
02:40.34*** join/#asterisk ravenpi (n=chatzill@londonderry-cuda3-24-51-50-156.lndnnh.adelphia.net)
02:40.35justinudude, looking thru boring logs is part of this job
02:40.44bugzhahaha
02:40.47_Simonhehe
02:40.50Lino`now seriously who pays money for a 7902 ???
02:41.04Lino`i wouldnt event want it as a gift.
02:41.04ravenpiTrue 'nuff.  And there are a lot of tools out there to help get rid of the "boring" parts of log reading.
02:41.10justinuyeah
02:41.12Lino`like grep
02:41.21Lino`or rm
02:41.23Lino`*gg*
02:41.24bugzhah, how bout some Luisa
02:41.31bugza Luisa/Asterisk box
02:41.32Sedorox<PROTECTED>
02:41.33justinugrep is my tool; i understand there's a lot of neat new fangled dealies that sort stuff, etc.
02:41.35_Simonis qualify=yes okay? thats what I set
02:41.48justinuyeah, that sends a keepalive ever 3 seconds i think
02:41.56justinuthat should be enough for even the most nazi nat
02:42.07Strom_Cnazi nat!
02:42.09FrogzooLino`: swatch & bb
02:42.10_Simonhaha
02:42.21_Simonhmm wonder what the issue is then
02:42.25*** join/#asterisk Weezey (n=ohno@206.210.109.228)
02:42.32justinucheck that SPI stuff
02:42.38justinui bet your iax2 packets are getting mangled
02:42.52*** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net)
02:42.56_Simonya theres not much on a linksys to check though I don't think, its just a consumer router
02:42.57justinusince it's UDP, there's no guaranteed delivery
02:43.05justinuwhich explains your intermittant issue
02:43.13WeezeyI'm getting no compatible codecs error when I try to dial SIP/exten@hostname.com, what do I need to change?
02:43.24Lino`well simon
02:43.27justinuiax2 packet INVAL is probably "your request is broken, try again"
02:43.27Lino`thats not 100% true
02:43.32Lino`linksys has some nasty functions
02:43.36justinutrue
02:43.37Lino`like packet analyzing or qos
02:43.45justinulook very carefully thru all it's settings
02:43.48_SimonI set IAX2 port as high QoS
02:43.50justinui bet you'll find it
02:43.51_Simonwas that smart to do?
02:43.52Lino`which seems to improve things, but in fact it fucks things up
02:44.08_Simonoh so that was a bad thing to set?
02:44.09Lino`throw away your router and get a real one ;)
02:44.14Lino`well
02:44.15bugzminimize delay, not maxmize throughput
02:44.34_Simonwell.. part of my issue is I'm trying to see if users will be able to use it on regular hardware
02:44.38_Simonlinksys super popular in homes so
02:44.53Strom_CWeezey, is the extension you're dialing on your local box?
02:44.54blitzragedid you guys know its Friday night? :)
02:45.02Weezeyno sirt
02:45.03Weezeysir
02:45.04Lino`yeah but to quote clayton bigsby "linksys stinks and i hate it"
02:45.10Strom_Cwhere is it?
02:45.10_Simonlol
02:45.15_Simonso I should turn QoS off for iax port
02:45.15_Simon?
02:45.17Weezeymy other box.
02:45.20Strom_Cok
02:45.23Lino`well try turning off all the fancy stuff
02:45.25Lino`like qos
02:45.27Lino`firewall
02:45.27Lino`etc
02:45.31Strom_Cwhat codecs are you allowing?
02:45.35Lino`(parental control *lol*)
02:45.36Weezeyulaw
02:45.38Weezeyon both
02:45.38_Simonhehe
02:45.47Strom_Care you disallowing everything else?
02:45.47Frogzoo_Simon: QoS should be fine, so long as sip/iax gets priority
02:46.18WeezeyStrom_C: the no codec thing actually might be right.
02:46.23Weezeyone sec..
02:46.39justinuwhat model linksys
02:46.43CoffeeIVI want to confirm my router is passing IAX2 to the right place.  Is there a netcat command that should provoke a response from asterisk's IAX2 port ?
02:46.47WeezeyStrom_C: nop
02:46.48Weezeye
02:47.03bugztime for some solaris action
02:47.06Lino`(just imagine the posts in the forums: "my mommy does not want me to use iax (whew its late i always want to write aix... im kinda ibm poisoned) how to kill linksys parental control"
02:47.08Lino`)
02:47.19Strom_Cweezey, in the appropriate section of your sip configuration, put "disallow=all"  followed by "allow=ulaw"
02:47.32Heimidalcan someone tell me why this causes the error "Mar 24 20:28:23 NOTICE[4193]: chan_iax2.c:7198 socket_read: Rejected connect attempt from 207.174.202.3, request '312546XXXX@incoming' does not exist"? http://rafb.net/paste/results/RceagI47.html
02:47.42WeezeyStrom_C: all better,  I'm using realtime
02:47.48X-RobHeimidal, it means you don't have 315... in [incoming] in extensions.cofn
02:47.54Weezeyand disallow=all comes after allow=ulaw
02:48.01Weezeyso it ignored allow=
02:48.03Strom_CWeezey, well that will fuck it up
02:48.08Weezeyall better
02:48.12bugzHeimidal: look in extensions.conf for 312546XXXX in the [incoming] context
02:48.14Weezeythanks for the inspiration
02:48.15HeimidalX-Rob: what if I want an extension that can take care of two incoming numbers?
02:48.35justinuwhoever had the linksys, what model is it?
02:48.46X-RobHeimidal, then put both numbers in [incoming]
02:49.04Heimidalwith what syntax?
02:49.10bugzexten => 939222XXXX,1,Dial(Sip/100)
02:49.14bugzexten => 939555XXXX,1,Dial(Sip/100)
02:49.19X-Robor use a wildcard -- exten _312X.
02:49.24X-Robbugz, you forgot _
02:49.27Heimidalah
02:49.54X-RobHeimidal, to correct bugz: exten => _312X.,1,Dial(SIP/123)
02:50.02HeimidalI guess I got confused by the asterisk book... it shows exactly what I put in that paste
02:50.02X-Rob_ means 'I have wildcard characters'
02:50.41bugzi have an asterisk 1.0.9 box up for almost a year
02:51.00bugznormally the drives fail long before that
02:51.10bugzsure have run into some crappy hardware
02:51.41*** join/#asterisk maxx4life (n=max4life@71-35-210-12.slkc.qwest.net)
02:52.21HeimidalI find it strange that you can't pass something to a context and have it start with the "s" extension without having to explicitly declare the number
02:52.53justinuthere's a lot of strange things about the asterisk dialplan
02:53.06bugzyeah, like how it works
02:53.26justinuit seems like every pbx was designed by someone from another solar system
02:53.39justinui had this panasonic kxt that had the weirdest CLI
02:54.01Heimidaland if you have to explicitly declare everything, why does "s" even exist?
02:54.23bugzto confuse n00bs
02:54.28justinuheh
02:54.43justinusupposedly s exists when there is no DNIS info
02:54.50justinufor when...
02:54.55Sedorox"I see n00b$... they're everywhere.. and they don't even know they're n00b$"
02:54.57*** join/#asterisk jroysdon (n=jroysdon@c-67-181-65-139.hsd1.ca.comcast.net)
02:55.47bugzsnort keeps throwing these "(portscan) UDP Filtered Portscan" from my providers sip router
02:56.25bugzi scanned one of those boxes once because of this and found irc running on it but with a Cisco IOS fingerprint
02:56.48bugzthat was a little unsettling
02:57.20justinuodd
02:57.34justinudid it actually speak ircd on 6667, or did 6667 just answer
02:58.38bugzwhen i tried to connect it i got connection refused, scanned it again and then it was closed
02:58.56bugzi called them and asked wtf was going on and they had no idea
02:59.08justinuhmm
02:59.21bugzthat particular router was a peering system i believe with alot of latency on it
02:59.22justinuthat is a bit unsettling
03:00.07bugzi see wierd stuff all the time on this network
03:00.30bugzi had the balls to apply for a job on monster.com that said "LINUX GURU NEEDED"
03:00.44justinuhah
03:01.08bugzits cool here, ive had the opportunity to put snort and iptables on every kind of network you can imagine
03:01.08Nuggetno true unix guru considers himself one.
03:01.22bugzNugget: ahh grasshoppaa..
03:01.29justinuwax on, wax off
03:01.55Nugget"linux" guru makes it even more suspicious.
03:02.49bugzthe IT departments i run in to are either a) MCSE/retardd, b) Cisco noobs, or c) pothead hippies who know everything
03:02.59NuggetI interviewed a guy at ud.com who was a self-described linux expert who, upon learning that we also used freebsd, asked if freebsd used "linux commands like 'tar' and 'ls'"
03:03.35*** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net)
03:03.42ravenpiWhich "matches" first: exten => _1600, or exten => 1600 -- in other words, does wildcard have higher, or lower (or same) matching priority.
03:03.45jroysdonAnyone ever used gparted?  I'm using it now to resize a windows/ntfs install on a borrowed work laptop that I want to test fc5 on.
03:03.49bugzNugget: i ran into a guy that told me he used iptraf once and that made him an expert on my firewall configuration
03:03.53ravenpiNugget: man, now THAT'S a Linux guru.
03:03.56justinuknow it all pothead hippies are ok
03:04.15bugzlike my dad lol
03:04.43bugz"dad put down the pipe, the DHS is calling again..."
03:04.45ravenpijroysdon: I've used parted, but never gparted -- and I never tried doing a resize (call me chicken).  How's it working?
03:04.46justinuheh
03:05.15NuggetI don't think I'll ever trust partition resizing.  call me superstitious.
03:05.21*** join/#asterisk _Simon (n=IRC@i216-58-40-193.cybersurf.com)
03:05.25_Simonhmm sorry got disconnected
03:05.29Nuggetespecially on ntfs, which seems like negligent insanity to me
03:05.31jroysdonravenpi, it's running.. it resized the first fat partition, and now it is doing the ntfs/restore partition.
03:05.32_Simonsetting QoS or not setting it isn't making much difference
03:05.41justinuwhat can't it work right?
03:05.43_Simonthere seems to be like 20 asterisk processes when I do a "ps" is that normal?
03:05.58jroysdonPartition Magic 9 does great with resizing... but I don't own a legal copy these days, and I'm trying to stay totally legal/free
03:06.01justinu_Simon: good question
03:06.20_Simontheres literally 20 asterisk -vvvvcd processes when I run asterisk in CLI
03:06.34bugzhttp://pastebin.com/621104
03:06.43bugzhere is a piece of some old QoS
03:06.49justinuif init runs asterisk from the startup script in /etc/init.d, i see a bunch of asterisk processes in ps
03:06.57justinuif I run it from the shell manually, i see one
03:07.11_Simonjustinu: I'm running from shell, and I still see like 20 of them
03:07.50_Simonso is that normal? if not maybe thats my issue
03:08.01bugzhttp://pastebin.com/621108
03:08.29bugz_Simon: i think all the modules make asterisk spawn processes
03:08.32justinui'm not sure if it's normal
03:08.41justinuit doesn't seem to cause any problems, for sure
03:08.48_Simonok, I'll assume its normal
03:09.03Strom_C_Simon, running asterisk in the background, I only have one asterisk process going
03:09.06bugz_Simon: i know when one of our bigger systems goes under a load there are like 50 processes going
03:09.13_Simonah
03:09.17bugz_Simon: when its in trouble there are hundreds
03:09.26_Simonok :)
03:09.29_Simonso now heres another thing
03:09.42_Simonas soon as I run asterisk
03:09.44_SimonI do a
03:09.49_Simoniax2 show channels
03:09.59_Simonand theres some IP connected to my server I don't know that IP
03:10.01justinu_Simon: you're the one with the linksys? what model is it
03:10.10justinucould be from the samples, simon
03:10.13_Simonits *ALWAYS* connected, even if I stop/start its there
03:10.14justinuthey connect to digium, etc.
03:10.19_Simonahh
03:10.22bugzlol
03:10.36_Simonwhere would that be defined?
03:10.46bugzid sure like to get my hands on about 100 of those wrtg's with 32 megs of ram
03:10.47_SimonI just want to start eliminating things that could be causing issues
03:11.02bugzi think its v4.0 of the firmware
03:11.03_Simonjustinu: and to your question. WRT54G
03:11.09justinuok, then it has SPI
03:11.20*** join/#asterisk ravenpi (n=chatzill@londonderry-cuda3-24-51-50-156.lndnnh.adelphia.net)
03:11.42_Simonso where would I find that connection that might be going to digium?
03:11.47_Simonin which conf file?
03:11.48justinuin iax.conf
03:12.26bugzive created a pretty extensive toolbox for using asterisk -rx stuff w/php and bash
03:12.47bugzit seems like everyone wants to build a gui for it but nobody can agree on exactly how
03:12.50_Simonhmm.. don't see anything in iax.conf except users
03:13.05_Simonany other file perhaps?
03:13.17justinuif it's iax, it's in iax.conf
03:13.22justinucheck for includes, etc.
03:13.25_Simonthing thats weird, the IP is a comcast IP
03:13.41justinuhow are they registered to you?
03:13.45justinuiax2 show registry
03:13.45bugzhaha, who did you buy this from again?
03:13.53justinuiax2 show users
03:13.56_Simonk
03:14.03justinuiax2 show peers
03:14.45justinuand plain: show channels
03:15.44Heimidalaccording to the asterisk book, http://rafb.net/paste/results/4hmgMJ38.html should play the sound file and wait for user input. However, I've just tried it, and after playing the file, it immediately hangs up.
03:15.49Heimidalwhat am I doing wrong?
03:15.51_Simonahh ok found who it is
03:15.53_Simonthats cool :)
03:16.00justinuHeimidal: check the error log
03:16.00_Simonits just an iax peer so I doubt its the problem
03:16.03justinuset debug 5
03:16.09justinui had that problem too
03:16.20justinuturns out i wasn't specifiying the file type, or something
03:16.26_Simonhe just left it running over night
03:16.26_Simonlol
03:16.52Heimidal<PROTECTED>
03:16.52Heimidal<PROTECTED>
03:16.56justinudid you check the SPI setting in your router?
03:17.00Heimidalthat's what happens after it plays the file.
03:17.15justinuHeimidal: check /var/log/asterisk/messages
03:17.17_Simonjustinu: I'm not really sure which part of the web config is SPI
03:17.20justinuor whatever it's called
03:17.26justinu_Simon: get the manual from the net, and find it
03:17.36_Simonwell I know theres port forwarding, port triggering
03:17.45justinuanother guy had this similar problem as you, and that was the fix
03:18.29bugz_Simon: put this in a file called "/usr/local/bin/commands"
03:18.31bugzhttp://pastebin.com/621123
03:18.45bugzthen build some aliases in .bashrc
03:19.06bugzi want to rewrite the whole UNIX/Asterisk interface like that
03:19.30_Simonhmm cool I'll take a look at that after :)
03:21.07Heimidaljustinu: there don't seem to be any errors with my current dialplan
03:21.43CoffeeIVmy * seems to listen on port 4569 when I connect from the localhost, but not externally. No iptables or hosts.deny are blocking anything, bindaddr is 0.0.0.0.  Any hints on what it might be ?
03:22.21QwellCoffeeIV: check netstat
03:23.59CoffeeIVQwell: no 4569 in the netstat output, I see the 5038 for the telnet api and few other things
03:24.47Qwellnetstat -lna | grep 4569
03:25.01jroysdonfyi, gparted just worked great - no problems
03:25.48CoffeeIVQwell: udp       0    0 0.0.0.0:4569    0.0.0.0:*
03:26.01Heimidalbleh... this is frustrating
03:26.16_Simonok this is the only thing I can find
03:26.30_SimonFirewall Protection: "enables or disables the SPI firewall"
03:26.31_Simonits enabled
03:26.44justinudisable that
03:27.10_Simondoesn't that take off the firewall?
03:27.13justinuno
03:28.53X-RobHeimidal, re that enter-number thing, the book is wrong
03:28.58X-Robthere needs to be a 'WaitExten' there.
03:28.59*** join/#asterisk fugitivo (n=user@201.255.182.148)
03:29.07fugitivoHi
03:29.11justinuif you use 1.2, yes
03:29.26justinueither that or set autofallthrough=no
03:29.59X-Robwhich is duplicating old behaviour, which again, is wrong.
03:30.02X-Rob8)
03:30.07justinuheh
03:30.28fugitivoThis nokia 770 is awesome
03:30.57_Simonok actually the docs as far as I can see says "Firewall Protection" disabling this means turning off the firewall
03:30.58_Simonlol
03:31.00_Simonthats not good
03:31.16justinuit will not turn off the firewall, don't worry.
03:31.24justinuit turns off SPI only, which is bullshit
03:31.31_Simonso SPI is bad?
03:32.18justinuyes, it mangels IAX apparently
03:34.38_Simonok I'll give it a shot for 5 mins or so and see if it helps :)
03:34.49*** join/#asterisk usam (n=usam@203.156.42.218)
03:35.08usaminband-dtmf, only for alaw+ulaw ?
03:35.19HeimidalX-Rob: thanks!
03:35.29_Simonthat sucks so bad that defaults for a router kills IAX though
03:35.29_Simonlol
03:35.56_Simonjustinu: if the asterisk box is behind a router as well the same type. will it mangle the server too?
03:36.00_Simonor just clients?
03:36.07justinupossible both
03:36.14justinui'd turn it off on both sides
03:36.26_Simonok, ya I'm getting the problem still
03:36.35_Simonso I'll disable on the server end
03:37.19*** join/#asterisk jero (n=jero@modemcable062.46-82-70.mc.videotron.ca)
03:38.24_Simoncurious to see what other things are affected by SPI being off lol
03:39.17_Simonah crap that didn't help
03:39.20_Simonstill having the issues
03:39.27_Simondamn what a pain in the nuts
03:40.09justinu:(
03:40.17justinusorry that's not it...
03:40.37justinumake a tethereal capture and see if you can figure out why the server may not like the stuff you send it
03:40.42X-Rob_Simon, has anyone asked you what version of * you're using at both ends? (What are they?)
03:41.18_SimonX-Rob: just a iax soft phone/iax hard phone on one end and like a week old SVN version of asterisk
03:41.23justinu_Simon: http://lists.digium.com/pipermail/asterisk-dev/2003-June/000938.html
03:41.25X-Rob*BAHAHAHA*
03:41.26Supaplexand do they stay crunchy in milk?
03:41.31X-Rob_Simon, IAX in SVN is _broken_ at the moment
03:41.34justinulol
03:41.35X-RobUse 1.2.5
03:41.37_Simonomg really?
03:41.38_Simonlol
03:41.44justinuwhy do people run SVN?
03:41.51Strom_CX-Rob, how broken is it?
03:42.02Supaplexbecause ages ago we kept pushing cvs head
03:42.03justinuwhat's wrong with the release versions?
03:42.03X-RobStrom_C, I don't know, but it plays up. There's been a couple of reports of it on -dev
03:42.10_Simonif it was public knowledge it would of been a simple answer for me
03:42.11_Simonlol
03:42.25X-Robsvn is for developers, and those with experience
03:42.32X-Robit was different in cvs times
03:42.51*** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net)
03:42.59_Simonok lemme try getting 1.2.5 and test
03:43.13X-Robsimon, grab 1.2 svn
03:43.27X-Robeg instead of /trunk, get /branches/1.2
03:43.39_Simonk
03:43.56X-RobÇommands to get the current snapshot from the release branch of SVN:
03:43.56X-Rob# svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2
03:43.57X-Rob# svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2
03:43.57X-Rob# svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2
03:44.53_Simonis it a problem if I don't use version 1.2 of zaptel?
03:45.00X-Robtrunk of zaptel?
03:45.05_Simonya
03:45.05X-Robtrunk of zaptel is good.
03:45.08X-Robmuch better echo cancellation
03:45.09X-Robworks fine
03:45.11_Simonk :)
03:45.19_Simonso I'll just leech asterisk and compile it
03:45.24X-Robyup
03:45.40_Simonyou know what might be helpful is some sort of asterisk advisory or something
03:45.40_Simonlol
03:45.51X-Rob'don't use SVN unless you know what you're doing'
03:45.52_Simonlike a news page or something lol
03:46.00_Simontrue but even if someone knows what their doing
03:46.07_Simonfor example. I'm mostly using svn because I need realtime support
03:46.17_Simonas realtime is still experimental
03:46.28_Simonbut I've been 6 months working on "experimental"
03:46.41X-Robrealtime is in 1.2?
03:46.57_SimonI don't even think realtime is completed currently in svn
03:47.02_Simonmany CLI commands don't support it etc
03:51.35*** join/#asterisk bmg505 (n=leon@165.146.13.215)
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03:52.04_Simonare they doing work on iax? making it better or something?
03:53.33X-Robfiik
03:53.45_Simonfiik?
03:54.00X-Robfucked if I know.
03:54.04_Simonoh.. haha
03:57.03*** join/#asterisk coppice (n=chatzill@218.203.17.210.dyn.pacific.net.hk)
03:58.48SkramXIf i come back later, could I get some help with configing a TDM card?
03:58.49bugzhaha thats funny.. "Are they going to work on IAX?"
03:58.54bugz_Simon: good question dude
03:58.59SkramXI have done a lot of virtual asterisk stuff, just nothing physical yet
03:59.32bugzi have a new distro out
03:59.35bugzcalled "Ass-Linux"
03:59.39bugzfor asterisk noobs
03:59.51bugzseriously
03:59.55bugzhaha..
04:00.08mitchelocwhy not call it Asterisk Live? that's a good name,
04:00.33SkramXwhat is this you all are naming?
04:00.47mitchelocan asterisk distro, like asterisk@home
04:00.54SkramXoh okay
04:02.26X-RobI prefer 'assman' as the name-of-the-year.
04:02.39GrizzySomeone's code I was auditing collected segfaults and restarted the code in the main loop.  : o )
04:02.53bugzX-Rob: no, 'assman' is the GUI
04:04.55X-Robthat's pretty damn hard to search for on google, youknow.
04:05.09SkramXwhat do you call a TDM 4 port with 4 incoming/FXO ports?
04:05.14SkramXwhats the model #, thatis?
04:05.41X-RobI can never remember which digit's which. I call it 'A TDM 400 with 4 FXO ports, please'
04:06.00Splatdo you need libpri if you don't have isdn?
04:06.11_Simonso far asterisk 1.2 is still working... I think its fixed but will give it another 10 mins
04:06.39SkramXmeh, ill ask for config help when i get the hardware/ssh access :)
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04:08.40tkprojectsdoes anyone know wafro or how to get in touch with him?
04:09.04bugzthis embedded system rocks
04:09.14bugzthey put a gui on it for us to customize
04:09.27X-Rob~seen wafro
04:09.33jbotwafro <i=matt@i216-58-44-251.cybersurf.com> was last seen on IRC in channel #asterisk, 189d 22h 4m 46s ago, saying: 'but i was looking around and people have made some progress with other firmwares by loading a non encrypted XML'.
04:09.41*** join/#asterisk L|NUX (n=linux@202.5.145.58)
04:10.26SkramXlol
04:10.48bugzi bow to no os
04:10.53bugz>=]
04:16.02*** join/#asterisk file (n=joshnet@mctnnbsa24w-142167032200.pppoe-dynamic.nb.aliant.net)
04:16.17bugztime for some ET
04:16.47*** join/#asterisk astra^^ (n=muhajir_@59.145.104.74)
04:19.12*** join/#asterisk viLeR (i=1000@66.128.47.232)
04:19.24astra^^hi all
04:20.12_Simonwicked thanks guys, asterisk 1.2 did the trick :) you guys rock!
04:20.21_Simonlooks like enabling SPI has no probs as well
04:24.14viLeRA little question: it is possible to make two steps at the same time ? for example: exten => 900,3,Dial(IAX2/john) & System("scipt.sh)
04:26.23tsumegames are a waate of time
04:26.27justinuyep
04:26.29tsume*waste
04:26.34justinui think they help out a bit too
04:26.52tsumeI work for fun
04:27.05justinuthe virtual worlds some of these people have created are pretty amazing
04:27.08tsumeearn money for fun, so I can get a couple of akitas
04:27.22_SimonI love C#
04:27.27_SimonI did an iax wrapper in C#
04:27.33tsumejustinu: real friends are better, especially when they protect you too
04:27.44tsume_Simon: yea?
04:27.46justinuuh, ok
04:28.09_Simontsume: yeah I'm working on a jabber client which integrates iax
04:28.18justinui'm an introvert because I like games?
04:28.23justinuthat's an interesting jump of logic
04:28.56tsumejustinu: ususaaly extroverts do something outside of the crackerbox
04:29.08justinui have my real life hobbies too
04:29.14tsumecracerbox == apartment
04:29.20justinui own a house, also :P
04:30.20X-Robbugger apartmens
04:30.32X-Robairoplanes hit them.
04:30.42X-Robthey'd have to be a damn good shot to hit my house.
04:31.58coppicewe have a big hill just to protect our apartment from the planes :-)
04:32.39Nuggetyay planes.
04:32.39justinuheh
04:33.32justinuany instrument rated pilots here?
04:33.49Nuggetjust vfr for me at the moment.
04:33.53Nuggetand I haven't flown in ages
04:33.59justinuwhat's ages?
04:34.00Nuggettoo busy hacking on flightaware  :)
04:34.07NuggetI haven't been PIC in 10 months.
04:34.09justinuflightaware?
04:34.14justinu10 months isn't too bad
04:34.18justinuit's kinda like riding a bike
04:34.33Nuggetyeah
04:34.38tsumestop chit chatting and go code
04:34.47tsumebefore I yank yer ears off and feek them to the dogs
04:34.51justinucut him some slack, it's friday night
04:34.55justinudamn slave drivers
04:35.05tsume:D
04:35.14tsumeI take classes from the Man
04:35.14coppicei just realised how many years it is since I last designed a piece of an aeroplane. its making me feel old :-(
04:35.19Nuggetwe're actually working on airport resources right now, but mostly I'm drinking guinness and complaining about dbaker's code
04:35.45Nuggetwe're getting all the approach plates and stuff sorted out, and adding user-maintained fbo/ground facility stuff.
04:35.57justinuyou have online approach plates?
04:36.00Nuggetyes
04:36.04justinuwhat about enroute charts?
04:36.16Nuggethttp://flightaware.com/resources/airport/KAUS
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04:37.50justinulooks like a nice resource
04:37.57Nuggetit's been a lot of fun to build.
04:38.12justinuwish the gubment would put the enroute charts online already....
04:39.24Nuggetwe experimented with overlaying the flight tracking on top of the sectional charts, but in practice it was really awful.
04:39.31justinui found a very nice Cirrus SR22 add-on for FS2004
04:39.40justinusectionals are too cluttered
04:39.43Nuggetyeah
04:39.49justinuthe enroute's are a lot cleaner
04:40.01justinuevery instrument is faithfully reproduced in this sim
04:40.08justinuthe glass panel, autopilot, gps
04:40.13justinuvery impressive
04:40.16justinugreat for training
04:40.23NuggetI'm working on getting all the DP/STAR coordinates into the database so we can plot the filed route alongside the flight track.
04:40.27Nuggetthat's my current project
04:40.30justinucool
04:41.08justinusr22 is a nice plane, i'd like to buy a share in one
04:41.14justinufast, modern
04:41.22Nuggetyeah, I don't think I'm a good enough pilote for one yet, though.
04:41.27justinuwhy not?
04:41.33NuggetI'm not a very good pilot.  :)
04:41.36justinuoh
04:41.40justinui'm competent
04:41.44NuggetI've never flown anything other than 172s
04:41.51justinui still prefer flying with other pilots
04:41.54justinumakes life easier
04:41.57Nuggetand I'm pretty far from being able to pursue my IFR rating
04:42.14justinuIFR in the sr22 is a whole different world than in a round guage 172
04:42.18Nuggetyeah
04:42.20justinuyou'd be amazeds
04:43.25justinuwe could probably shape you up as a pilot, anyways
04:43.28justinuit's not all that tough
04:43.52justinui got a chance to fly the airbus A320 sim at united headquaters, in denver
04:43.55justinuthat was fun
04:44.06NuggetI flew the 777 simulator at united in denver.  :)
04:44.09justinucool
04:44.16Nuggetone of the d.net guys works there at the ual flight center
04:44.17justinuthe 777 is a pig tho
04:44.32justinuone of my buddies (real person) is a united captain
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04:44.54NuggetI like that the simulator includes the "fasten seat belt" ding.
04:45.01justinuit's amazing
04:45.08justinuthe gfx are kinda primitive
04:45.14justinubut the motion/sound effects is fantastic
04:45.18Nuggetyeah
04:45.46Nuggethttp://cuckoo.com/daniel/pictures/albums/ga-ual/ding.mov
04:46.24justinuthe a320 is a real cinch to fly
04:46.31justinudoes all the tough work for you
04:46.40justinueven tells you when to retard the throttles in the flare
04:47.06justinui had them simulate a v1 cut, and even that was a cinch
04:47.17justinuv1 cut is the trickiest thing multi pilots have to practice
04:47.41justinuone engine fails at v1, which is commited to take off speed
04:48.08justinuback to the topic, i guess
04:49.15opc0dehey what's it called when you have n phone lines and they're all tied to a single number? ie if one line is busy, someone can call the same number and it rolls over to the next available phone line?
04:49.22justinua hunt group
04:50.05opc0dethis is something that must be provided by your telco right?
04:50.10justinuyeah
04:50.19justinuif you don't control the switch that sends you calls
04:50.44opc0deI have 4 lines coming into an FXO interface
04:51.58opc0dewhere do I find out how to configure this in asterisk?
04:52.11justinuyou'll have to talk to the telco, like you said
04:52.47opc0deand the asterisk configuration is just the same as it would be for 4 separate lines on 4 channels?
04:52.53justinubasically
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04:54.50fjeanjustinu, hi, tell me, is it a real pain to configure the iptable for SIP and IAX on an asterisk box ?
04:55.24fjeanby the way I started to install SER and it's going well...
04:59.04justinuheh, hey
05:04.12Strom_CI say....
05:04.15Strom_CDEAD HOOKERS
05:04.33QwellStrom_C: please stay out of my trunk
05:04.52Strom_CI didn't hide them in your trunk
05:04.57Strom_CI hid them in SVN trunk
05:05.03Qwellinteresting
05:05.18Strom_Cchan_deadhookers
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05:09.12QwellStrom_C: gotta be res_deadhookers...every part of * can use it that way
05:09.18joelsolankianybody can tell where to buy quad server?
05:09.27joelsolankiany website plz
05:09.33Qwelljoelsolanki: google.com
05:09.36Qwellnewegg.com
05:09.39Qwelldell.com
05:09.46Qwellmicrosoft.com
05:09.47Heim|awayrackmountmicro.com
05:09.48Qwelloh wait
05:10.03joelsolankihmm let me check out. thanks
05:10.03Qwellqwellsdiscountquadservers.pk
05:10.08Strom_Chahaha
05:10.13joelsolankihmm ok
05:11.10Qwellthat reminds me...
05:11.16QwellStrom_C: You know what I haven't seen in a while?
05:11.22Heim|awayjoelsolanki: I'll admit that was a shameless plug, but we've been around for two years and have an incredibly good track record.
05:11.24Strom_Cwhat?
05:11.25QwellCrazy Gideon commercials
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05:11.34Strom_Cwho is Crazy Gideon?
05:11.38Qwellwtf
05:11.47QwellHow can you possibly live in CA, and ask that question?
05:11.55Strom_Cbecause I don't own a television set?
05:11.58Qwell...
05:12.06QwellHe's Crazy Gideon!
05:12.21Qwellhe like...smashes shit, throws blenders, etc
05:12.25Qwellbecause he's crazy
05:12.29Strom_Cyou mean Gallagher?
05:12.32justinumaybe he's dead
05:12.37Qwellno, Crazy Gideon...it's a retailer, heh
05:12.44Strom_C*shrug*
05:12.48ravenpiHow do I direct a call based on the Zap channel it came in on?  I don't see an obvious variable (${CHANNEL} appears not to do the trick).
05:12.49Qwellhome appliances and such
05:12.56Qwelljustinu: surely you know what I'm talking about?
05:13.01justinuyeah
05:13.36*** join/#asterisk maxx4life (n=maxx4lif@71-35-210-12.slkc.qwest.net)
05:13.39justinuhe seemed kinda high strung
05:13.47Qwellheh, just a tad
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05:14.29ravenpiIn NYC metro area, there used to be Crazy Eddie, doing much the same stuff for much the same store -- he kinda stopped when it turned out the owners of the chain were cooking the books.
05:14.41justinuyeah
05:14.42Qwellnice
05:14.43ravenpi"Our prices are INSANE!"
05:14.59fjeanravenip, you could throw on different context for each channel...or...
05:15.08justinui wonder if anyone from LA remembers "Z Best carpet cleaning"
05:15.11Qwell"I stock themem deep, and sell them cheap!"
05:15.12Qwell:D
05:16.25ravenpifjean: but how do I make that happen?  Can I bind a channel to a context somehow?  In zapata.conf, maybe?  *goes and looks*  Ahhhh... I guess so.  Thanks!
05:18.03litageare there any disadvantages to enabling NAT options for an extension even if the extension isn't NAT'd?
05:19.07Corydon76-homeextensions are natted?
05:19.24QwellCorydon76-home: I have an extension firewall
05:19.46Corydon76-home...with a spoon...
05:19.56Qwellyes, yes
05:20.15*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167032200.pppoe-dynamic.nb.aliant.net)
05:20.22Qwellfile[laptop]: home?!
05:20.33Corydon76-homeIndeed he is
05:20.39Qwellyuck...sorry
05:21.12justinudoes canada suck that much?
05:21.44Qwelljustinu: some parts.
05:21.45file[laptop]home, home on the range
05:21.48Qwellother parts suck
05:21.52justinui've never been
05:22.01Corydon76-homebut especially New Brunswick
05:22.15fjeanhey, can we adjust the volume in some way without any zaptel device (ztdummy) ?
05:22.16justinusounds nice enough
05:22.35Qwellfjean: volume of what?
05:22.48fjeanqwell: sound
05:22.57Corydon76-homeOddly enough, file escaped back to Canada with his innocence still intact.
05:23.23Qwellfjean: yeah, thanks, I'd have never been able to guess THAT
05:23.48fjeanqwell, well that was a pleasure then  ;-)
05:24.25file[laptop]I'm a lil' bit tired so I'll be... unconscious... for probably a day LOL
05:24.46Qwellfile[laptop]: any trouble on the flight(s(s)) home?
05:24.59file[laptop]yes
05:25.26Qwellsounds...fun
05:25.36file[laptop]delays mostly
05:26.04*** part/#asterisk fjean (n=fjean@201009208229.user.veloxzone.com.br)
05:28.04Corydon76-homeDid they lose your luggage again?
05:29.18file[laptop]no
05:30.21Corydon76-homeWell, there's always next time
05:30.37Corydon76-homeThe question is, will I be around next time to lend you a change of clothes?
05:30.39QwellYou should call and complain
05:31.07Qwell"I expect you to lose my luggage, and you can't get that right?  I demand that my luggage get lost on all future flights."
05:31.12Corydon76-homeBTW, I've been scraping that shirt you wore of DNA.  I'm planning to clone you.
05:31.21file[laptop]yay clone
05:31.29Qwelloh boy..
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05:50.08bugzI HATE ASTERISK
05:50.48bugzanyone wanna work on a dial plan with me
05:51.01bugzi will give you temporary root
05:51.09bugz>;]
05:51.31Strom_Cwhat about candy?
05:56.41Corydon76-homeWhat about male strippers?
05:57.29*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
05:57.37mitchelocwhooo hooo! strippers!
05:57.50Strom_Cbugz, I'll see what I can do for you :)
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06:25.15rogercharliecan the FOP proxy be used for other manager connections?
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06:36.08HamYaIanyone using soyo package with asterisk?
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06:39.22konfuzedhey does anyone use thinktel.ca? im curious about service quality or service levels. I suppose Call Quality is most important?
06:46.43litageare there any disadvantages to enabling NAT options for a sip device even if the device isn't NAT'd?
06:51.27*** part/#asterisk awe6 (n=lba@user-12lml5g.cable.mindspring.com)
06:51.52blitzragenope
06:52.07blitzrageno change really in the packetes
06:52.36blitzragemight even be a good idea incase you put the device behind NAT at certain points (i.e. mobile device)
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06:59.29Altair256hello everyone
07:00.38Strom_Chi
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07:13.07Fedoracore6hai all
07:18.36*** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
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07:36.11litagethanks blitzrage
07:36.32Strom_Cdogballs
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07:53.46argos73grrrr....  dog just chewed up the photo album of our trip to NYC last year...  wife's gonna love that when she wakes up...
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08:13.52Mavvieargos73: very simple solution: don't let her wake up.
08:18.49argos73Mavvie: hehe
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08:45.15Fedoracore6hai all i try using agi code for delete but still cannot delete
08:45.16Fedoracore6http://pastebin.com/621384
08:45.24Fedoracore6this my code ...
08:45.44Fedoracore6hemm any code i must add org somethig code i forget
08:55.10Fedoracore6can i delete data in databases.. using delee by field
08:56.05Fedoracore6cos i do this code http://pastebin.com/621384
08:56.43Fedoracore6whole my data in my databases delete
08:56.45Fedoracore6huhuhuhu
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08:58.14Fedoracore6hehhehhee
08:58.20Fedoracore6krisguy
08:58.24Fedoracore6whyyy zzz
09:01.31vatternI am having trouble receiving faxes with spanDSP
09:01.33vatternhttp://pastebin.com/621398
09:01.59vatternany pointers ?
09:11.14tzafrir_laptopSorry, I'm not familiar with sandsp/rxfax
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09:19.48voip_learnercan I install asterisk and just create a dummy enviorment so that I can test SIP based tools? or may be 2-3 compuers in LAN where asterisk is install in one and rest two will communicate.
09:22.28vatternthat should work for sip based comms.
09:24.08voip_learnerhi vattern , I want to test some toos like SiVuS, vomit...
09:24.24voip_learnerand also some sip fuzzing tool
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09:27.44voip_learnerwhat distro are you using? , vattern
09:29.37voip_learnerok
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10:05.52rogercharliedoes the FOP proxy work for other manager connections
10:07.08rogercharlieI am unable to proxy other manager programs though the flash proxy
10:07.17rogercharlieanyone able to do this?
10:07.23*** join/#asterisk sergeus (n=s@195.112.98.13)
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10:22.21jpksodele... FYI: Powerline steht. Netto 28Mbit.
10:22.35jpkNicht überragend, aber ok.
10:22.49jpksorry. wrong window
10:24.51Fedoracore6where i can find code delete field in my database
10:25.10Fedoracore6cos i do alot of code ... still cannot success
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10:52.41lemmyhi
10:53.17Fedoracore6hi
10:53.58lemmydid anybody manage to use ztdummy in domU 2.6.11? each time i load the ztdummy module the domU dies. i tried zaptel-1.2.4 and head.
11:09.14*** join/#asterisk Plnt (n=someone@goodspeed.vscht.cz)
11:09.21lemmyi just need ztdummy for meetme so i tried app_conference too. but it crashes the asterisk on * 1.2.1 and 1.2.4 if two people join a conference room and start speaking.
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11:27.57tzafrir_laptopyou don't need ztdummy for app_conference
11:30.53lemmytzafrir_laptop: i know, but app_conference isn't working for me
11:31.05lemmyso i tried to get meetme running
11:32.17lemmyi would be happy to use app_conference though
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12:12.22tzafrir_laptoplemmy, please provide more details: kernel version, linux distro
12:12.35tzafrir_laptopWhat do you mean by "ztdummy dies"?
12:12.59tzafrir_laptopDo you get timing working properly? try zttest
12:13.20lemmytzafrir_laptop: the xen domain/kernel crashes.
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12:13.58lemmyits debian/etch, kernel 2.6.11 on xen 2.0.7
12:14.23lemmywhat else do you need?
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12:19.50lemmytzafrir_laptop: some extra informations http://pastebin.com/621508
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12:23.48tzafrir_laptoplemmy, do you have any interesting panic statement?
12:30.38lemmynope, the hole domU crashes right after i load ztdummy. zaptel loads fine though.
12:33.51RoyKdomU?
12:34.06lemmyRoyK: a domain inside xen.
12:35.09lemmytzafrir_laptop: i attached the last few lines i get here: http://pastebin.com/621518
12:43.42SplasPoodlemmy: maybe try xen 3.0??
12:44.39lemmySplasPood: you got ztdummy working with xen 3.0?
12:45.31HamYaIanyone using soyo as your SIP provider?
12:45.47HamYaIor ever used
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12:53.18usnhi folx
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12:55.31x86hmm, i have a Grandstream BudgetTone 101 phone that no longer gets caller ID info...
12:56.01x86i thought maybe it was something I did with my asterisk configuration, but my X-Lite softphone still gets caller ID just fine
12:56.13x86anyone know what might be causing this?
12:56.40x86it happened after i moved all of my SIP peers and users into MySQL and started using RealTime
13:00.48x86the display on the grandstream just says "nu"
13:01.28*** join/#asterisk coppice (n=chatzill@169.198.17.210.dyn.pacific.net.hk)
13:05.47x86this is so bizarre
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13:12.12SparFuxI want to send digits on an existing bri channel. Dial(CAPI/ISDNdev/<digits>/o,4) does the job, but I am afraid it tries to open the capi device again. How can I avoid this and just send the <digits> ?
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13:22.07wiseguy_hello
13:22.24wiseguy_somehow musiconhold plays strange sound
13:22.28wiseguy_what can it be?
13:24.41Jon335This is probably the most noobish question ever, but here it goes: I have an outbound call that I want to transfer to another extension; How do I do that?
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13:25.48tzafrir_laptopwiseguy_, edo you use mpg123?
13:26.06wiseguy_tzafrir_laptop: yes
13:26.21tzafrir_laptopwiseguy_, any chance that you really use mpg321?
13:26.46tzafrir_laptopls -l `which mpg123` and see if it is a symlink to mpg321
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13:28.49wiseguy_tzafrir_laptop: really
13:29.59wiseguy_/usr/bin/mpg123 -> /etc/alternatives/mpg123
13:30.21wiseguy_/etc/alternatives/mpg123 -> /usr/bin/mpg321
13:32.41wiseguy_any other ideas?
13:33.16SparFuxHow can I change the path or name asterisk uses for mpg123 in the config file so that I can use a script even streaming ogg vorbis? I don't want to rename mpg123 and replace it with this script.
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13:51.51Gennarohi some one can help me  with an asterisk@home?
13:52.01Gennaroi installed it with an tdm400
13:52.18Gennaroand i cant recive incoming call..
13:52.21Gennarowhy?!?
13:53.40Gennaroi need id service from my provider?!?
13:54.59Gennarosome channel for asterisk @ home ?
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14:03.14Gennarosomeone is connected?!?
14:08.52Gennarosome one can speack?!?
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14:17.23tzafrir_laptopwiseguy_, still here?
14:18.19tzafrir_laptopwiseguy_, mpg321 is indeed your problem. Either replace it with mpg123 from nonfree or whatever, or (better) convert mp3s to wavs offline
14:18.44tzafrir_laptoplook for rawplayer on voip-info
14:19.35FuriousGeorgeis two cores with the expense for asterisk?
14:19.43FuriousGeorge*are two
14:22.28FuriousGeorge*are two cores WORTH the expense, is what i meant to say
14:24.20*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
14:24.21tzafrir_laptopFuriousGeorge, not much different than a dual-cpu computer
14:24.23*** join/#asterisk Djow1 (n=gavioes@200.103.150.81)
14:24.32tzafrir_laptopthat is: you won't get twice the speed
14:25.00FuriousGeorgetzafrir_laptop: sure, i know what you mean
14:25.29tzafrir_laptopMy hunch is that Asterisk is reasonably-well parallelised, due to the nature of its task
14:25.38FuriousGeorgemy main concern is that i have to page() 17 extensions at the same time reliably, and im not sure how big of a cpu i need
14:25.49FuriousGeorgethis big? |-------------|  or bigger?
14:27.14FuriousGeorgeseriously though, im an amd guy so im deciding between bartons or maybe an athlon 64, but then i got to thinking i might need more
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14:35.02TinoWhrm. msgsm.h:573: warning: 'xmc[48]' is used uninitialized in this function
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14:46.08ambrientofuriousgeorge, how good is a 64 processor when your applications isnt 64 bits?
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14:48.11_DAWgood morning all
14:48.49TinoWambriento: it can leave a whole 32 bits address space to your application ;)
14:49.08coppicefor some things dual cores really fly with *. It depends what you are doing. Software echo cancellation occurs in the driver, and it doesn't use the cores well. Codecs, like G.729, do make very good use of the cores, as long as their caches are big enough
14:49.32SplasPoodDoes anyone know how I would get the name of the current sip user from within the dialplan?
14:49.54FuriousGeorgeambriento:  are the fastest intel 32bit chips comparable to the athlon64s?  i dont follow intel much at all
14:50.06FuriousGeorgemy point is just that all the newer chips are 64bits anyway
14:51.56ambrientoindeed they are
14:52.15FuriousGeorgeis intel still making 32bit chips?
14:52.44coppiceyes. all the laptop chips are still 32 bit
14:52.55ambrientoeven the p4 HT has EMT64 instructions, which will allow you to run 64btis OS on it
14:53.42coppicenot all 64 bit chips are created equal. the AMDs really fly running my DSP code on 64 bit FC4. The Intels are considerably slower
14:58.33ambrientohmmmm
14:59.26ambrientocoppice, there is no way to choose what to do with reversal polarity in Unicall, is there?
14:59.48wiseguy_somehow musiconhold plays strange sound?
15:00.00wiseguy_anyone has/had this problem?
15:01.10ambrientowiseguy_,  your not that wise, are you?
15:01.12ambriento:)
15:01.48ambrientoj/k... by strange sound you mean like "too fast" or "too slow" play?
15:02.07wiseguy_yes, really ;-)
15:02.21wiseguy_but i can't find anything about that in google
15:02.28wiseguy_too slow
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15:03.29wiseguy_ambriento: any suggestions?
15:03.49ambrientowiseguy_, I'm try to remember what I did. It happened to me once
15:04.52ambrientobut this has to do with sample rate, like your file has 22K samples and you play it at 8K
15:05.40ambrientoare you using mpg123? which asterisk version?
15:06.00ambrientodid you try resample the file?
15:06.01wiseguy_1.2.5
15:06.14wiseguy_asterisk
15:06.24ambrientogot it
15:07.29ambrientowhich mode in moh.conf? custom?
15:08.04coppiceambriento: currently people only use unicall for R2, and reverse polarity is not a concept which exists there
15:08.20*** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt)
15:08.38wiseguy_ambriento: i haven't tried resampling it
15:10.22wiseguy_any recomendations?
15:10.43tzafrir_laptopwiseguy_, you're using mpg321
15:11.20tzafrir_laptopDid you see my previous comments?
15:11.29Kattysomeone fed me egg today and didn't tell me :<
15:11.38*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
15:12.01coppicei fed me rice today and didn't tell me
15:12.06ambrientocoppice, do you have like "to collect" calls in hk? like I call you but you will be charged for that?
15:12.07wiseguy_tzafrir_laptop: no, please tell me again :(
15:12.23Kattycoppice: vegans don't eat egg :<
15:12.54Kattyand people who know this should know better.
15:13.29coppiceambriento: we only get charged for IDD. I'm not sure if collect calls are supported there, but I have no idea off hand just how they might be supported
15:13.37ambrientokatty, how do you realize that fed you up with egg? Taste?
15:13.52ambrientoIDD?
15:14.02coppiceKatty: carnivores don't eat rice, and people who know this should know better
15:14.07Kattyambriento: i started feeling sick, so i asked them directly what they put in it
15:14.24coppiceambriento: international calls
15:14.26ambrientowho are they? friends?
15:14.42TinoWhm. is asterisk source missing capi support?
15:14.47Kattyrelatives
15:14.51wiseguy_tzafrir_laptop, ambriento any suggestions?
15:14.55coppiceambriento: what are you trying to do in HK?
15:15.31ambrientocoppice, actually, I'm trying to figure out how PSTN works in there
15:15.51ambrientosince its R2, I can know a little more about it
15:16.03coppiceHK doesn't use R2
15:16.18cpmhttp://pastebin.com/621656
15:16.19cpmclues?
15:16.35ambrientoand the "to collect" stuff I was asking you, its cause we have here in Brazil such thing
15:16.37ambrientooops
15:16.55ambrientoreally? Idk why I had that in my mind
15:18.04coppicei wonder why brazil needs to do that? In other countries R2 delivers collect calls with a special code
15:18.06ambrientomay be I got confused with the unicall.conf.sample and your place :) they are near, rite?
15:18.29tzafrir_laptopwiseguy_, mpg321 is indeed your problem. Either replace it with mpg123 from nonfree or whatever, or (better) convert mp3s to wavs offline
15:18.36cpmfound it, http://bugs.digium.com/view.php?id=6696
15:18.37cpmshit!
15:20.20wiseguy_tzafrir_laptop: you mean use sox instead or what?
15:20.49ambrientospecial code, like one of those bits? I'm not a telephony guy tbh.
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15:30.23TinoWvery strange: http://pastebin.com/621681
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15:33.19ambrientobrb
15:33.29ambrientoRoyk is here
15:33.39ambriento:)
15:33.42cpmTinoW, no clue.
15:34.25TinoWnu sto...
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15:36.25RoyKambriento: ?
15:39.50*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfj06.dialup.mindspring.com)
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15:49.40tzafrir_laptopwiseguy_, basically: use the custom: method there with a wrapper script around sox
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15:52.30RoyKi use that as well
15:52.47RoyKbut if i restart asterisk, a sox process remains spinning on 100% cpu
15:56.24HamYaIanyone using soyo as your sip proxy?
15:57.08RoyKsoya :)
15:57.34HamYaII'm having a problem where the system keeps silent while pronouncing "numbers"
15:57.38Fedoracore6air soya ;D
15:58.05HamYaIRoyK: soya source?
16:01.09wiseguy_HamYaI: seeya
16:01.10wiseguy_:)
16:01.50Fedoracore6:)
16:02.07FuriousGeorgeanyone have experience using page or meetme with 17 people?
16:02.16FuriousGeorgemeetme's would have 1 speaker 16 listeners
16:07.07wiseguy_tzafrir_laptop, ambriento thanks, know it works perfect
16:09.00FuriousGeorgeif you have experience with 12-infinity, that will do, also
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17:06.02flashbac1hey guys
17:06.31flashbac1does anybody have any experience with voicemail ODBC storage?
17:07.08flashbac1hello?
17:07.17RoyKCorydon-w: ding
17:07.23*** join/#asterisk salviadud (n=ralfalfa@201.138.132.150)
17:10.18onsiphello every one. I am a newbie here, and also newbie on SIP.  Is there anyone who could give me some sugestion on learning SIP and the archtecture of PBX. I'd be very happy for your kind hearted help.
17:10.59salviadudhave you done your homework buddyboy?
17:11.04*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
17:11.54salviadudi'd suggest www.voip-info.org
17:12.23onsipOk , thank you very much <salviadud>.:-)
17:13.39justinuonsip, i suggest starting with RFC 3261
17:15.10Kattymew.
17:15.20onsipof course, rfc3261 and related rfcs is the best ones for newbies, but it's of course, tooo, boring. I think the useage of such rfcs is best for reference, not for study. I think.
17:15.30justinui disagree
17:15.41justinui think the RFCs are written very well
17:15.50justinumorning katty
17:15.58Kattyhihi
17:16.33justinuit's saturday... what should we be doing?
17:16.38Kattynapping.
17:17.23TinoWthere are even some funny RFCs...
17:17.38RoyKKatty: http://static.flickr.com/19/116762025_b7a35a854a_o.jpg
17:17.46Kattyjustinu: you could meet me for lunch i suppose
17:17.50justinunapping... i just woke up
17:17.55justinui couldn't sleep in this morning for some reason
17:17.59justinulunch sounds good
17:18.10justinulet me get my scramjet warmed up
17:18.38KattyRoyK: :<
17:18.42justinuwe either need to live in a smaller country, or need faster transportation
17:19.00justinulol, that face has to be photoshopped
17:19.04Kattyi'm all for a bullet train
17:19.07Katty300mph.
17:19.21justinuthey've got them up to 800 kph now, which is approaching jet speed
17:19.27Kattymew!
17:19.31Kattyi'll take two, plskthx.
17:19.35justinuditto
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17:42.22Lino`hmmm
17:42.56SparFuxWhat is a "native bridge"?
17:43.13Lino`lol
17:43.33Lino`thats when a connection is being established within a native trunk
17:43.37iDunnoone made of wood.
17:43.39Lino`e.g. within one isdn card
17:43.40iDunnoand string.
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17:43.50Lino`:D lol @ iDunno
17:45.04SparFuxAnd what's a native trunk?
17:45.31SparFuxLino: Yes, that's what I am trying to do. But all of the sudden, the line is hung up!
17:45.40iDunnoone that grows from the ground after a native has planted the seed?
17:46.31SparFuxLook at this: http://pastebin.com/621885
17:46.57SparFuxI try to initiate an isdn conference with asterisk.
17:47.17justinunative bridge just means there's no protocol or codec translating involved, i thought
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17:50.03SparFuxjustinu: sounds logical.
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17:50.09a1faanybody experiencing issues with broadvoice ATM?
17:50.17a1fai am unable to make outgoing phone calls
17:50.24a1fagod damn those stupid mother fuckers
17:50.41salviadudsue them
17:50.45justinuhaha
17:50.46a1fano
17:50.49salviadudthat's what america is all about
17:50.49a1fafuck sue them
17:51.05a1fatheir support doesnt even answer phone
17:51.14a1fai hate their onhold music
17:51.19salviadudwell, call them up, and put mixmonitor on
17:51.24salviadudthe judge might like it
17:51.43salviadudgood evidence imho
17:52.02Dovidyea
17:52.10Dovidthjey have been going downt he shitter for a while
17:55.19SparFuxWell, after attempting this native bridge I get and active hangup immediately. :-(     -- Attempting native bridge of CAPI/ISDN1/27-5a and CAPI/ISDN1/008001071020-5b
17:55.20SparFux<PROTECTED>
17:56.07TinoWSparFux: just curious, did you compile asterisk + capi stuff yourself?
17:56.19SparFuxNo, it's the debian version I use.
17:56.29TinoWI see
17:56.48SparFuxI will compile myself, but my current computer is too slow and it is no fun this way.
17:57.07TinoWSparFux: its quick, but the capi stuff does not compile for me
17:57.20Lino`lol
17:57.22SparFuxTinoW: Well, I compiled chan-capi myself.
17:57.23Lino`well
17:57.33TinoWSparFux: which version?
17:57.41SparFuxTino: actual cvs.
17:57.51a1fai've been on hold for 10 minutes
17:57.53a1fadamn it
17:57.54a1fadude
17:58.01TinoWSparFux: I get http://pastebin.com/621681
17:59.27SparFuxStrange error...
17:59.43TinoWI rhink so
18:00.54a1fa"Thank you for calling Broadvoice" "Your call is important to us"
18:01.01a1faObviously not if I have to wait 15minutes
18:01.13a1fagod damn bastards
18:01.20a1fafreaking zipperheads
18:01.23iDunnommhmm
18:03.06Lino`cisco is worse @ a1fa
18:03.13Fedoracore6hai all i doing code for delete data in tables but when i using this code , DELETE FROM student WHERE kodsubjek1 = '$exten'
18:03.33Fedoracore6still cant delete by field
18:03.36Fedoracore6http://pastebin.com/621899
18:03.46a1faLino` : no cisco is good.. I have an SLA with cisco
18:03.52a1faLino` : they answer call in 2s
18:04.01a1faLino` : i have a direct line to CISCO engineers
18:04.09Fedoracore6its have code that,can delete by field
18:04.42*** join/#asterisk Muecke77 (n=muecke77@p54A9F93D.dip.t-dialin.net)
18:05.21a1faguys, is vonage's support this bad?
18:07.33a1faok
18:07.37a1famy neck is killing me now
18:07.37salviadudguess so...
18:07.39a1fai am suing them
18:07.40a1fafor pain
18:07.40Dovidpeople have mixed reports on vonage
18:07.42a1fain my neck
18:07.42salviadudyeah!
18:07.47salviadudyou sue those bitches
18:07.50salviadudbut like i said
18:07.55salviaduduse mixmonitor if you can
18:08.02salviadudgood evidence, on audio
18:08.09a1fai got a speakerphone on my uniden cordless phone
18:08.13salviadudvery good leverage
18:08.14a1facomes in handy
18:08.26salviadudare you using asterisk to call those bitches?
18:08.34a1fayup
18:08.35salviadudyou see, i got a sipura 3000
18:08.42a1fai got PAP2-NA
18:08.44a1fa:P
18:08.47a1fai love me a PAP2
18:09.04salviadudalright, just monitor the outgoing call
18:09.10salviadudi monitor all my calls...
18:09.13salviadudjust for fun
18:09.21salviadudi think it's illegal
18:09.25salviadudyet, i don't care
18:09.29salviadudi am mexican
18:10.59salviadudviva la revolucion, viva la tarjeta tormenta y emiliano zapata, metanse un dedo por el culo, ARRIBAAAAAAAAAAA!
18:11.32a1fa:)
18:11.47a1famy ast box is at a remote location
18:11.54a1faits at a datacenter pumping 100mbits
18:12.46salviadudmy ast box is an old sony vaio P3
18:12.53salviadudwith slackware 10.2
18:12.56salviadudand custom kernel
18:13.03salviadudi hate sony
18:13.20salviadudyet linux is a miracle worker
18:13.52Dovida1fa: what data center r u in ?
18:14.51websaeanyone here like voipbuster.com?
18:14.57websaecurious what your thoughts are
18:17.20*** join/#asterisk Sponge_bob (i=None@cpe-66-27-171-121.socal.res.rr.com)
18:17.29tehdelywebsae: howdy
18:17.32tehdelydid you get my email?
18:17.53*** join/#asterisk _deg_ (n=deg@200.250.222.8)
18:20.53TinoWaha, when I remove /usr/lib/include/asterisk and put the build dir includes in the Makefile, chan capi 0.3.5 compiles for asterisk 1.0.x but not for 1.2.5
18:23.34*** join/#asterisk lorinc (n=ang@caracas-0638.adsl.interware.hu)
18:30.30*** join/#asterisk SparFux (n=player@e182030071.adsl.alicedsl.de)
18:34.21TinoWhttp://www.junghanns.net/en/download.html  is the right place? I dont find 0.4.0 there...
18:34.33*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
18:39.59SkalTurawhat's ATA & DID?
18:41.19[av]baniwhats the command to hangup channels again?
18:41.37TinoW++ath
18:41.38TinoW;)
18:41.42*** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net)
18:42.09nitramTinoW: go to chan-capi @sourceforge
18:42.34*** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc)
18:43.30Dovid~ata
18:43.33jbotwell, ata is Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info
18:43.34Dovid!ata
18:43.43Dovid~did
18:43.45jbotextra, extra, read all about it, did is Direct Inward Dialing
18:43.50Sponge_bobif you want to setup a phone systems for 50 users, how do you gauge how many pstn lines to get?
18:43.51[av]banino possible way to clear a channel?
18:44.04[av]bani:(
18:44.14DovidSponge_bob: depends on the placee, what they do etc. whats thier current volume ?
18:45.01Sponge_bobDovid: thats hard to say.  i guess i need those stats first eh?
18:45.20[av]banithis sucks. i have stuck channels and no way to clear them?
18:45.29Dovidyes
18:45.37Dovidu can force close them
18:46.23[av]banihow
18:46.30Dovidus the help cpmmand in cli
18:46.31Dovidbrb
18:46.34[av]banii did
18:46.35[av]baninothing
18:47.19Dovidwhen u type help in the cli what do u get ?
18:48.27*** part/#asterisk _deg_ (n=deg@200.250.222.8)
18:48.48*** join/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-99-41.d-ip.magma.ca)
18:49.09Dovid[av]bani: what happens when u type in help in the cli ?
18:50.06justinusoft hangup didn't work?
18:50.09[av]baninope
18:50.14justinusip channels?
18:50.15[av]banisip show channels
18:50.16[av]baniPeer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last Message
18:50.19[av]bani65.100.14.254    1030        1ccd093a5e0  00102/00000  ulaw  No       Tx: ACK
18:50.22[av]bani192.168.42.254   4000        34a37d1c7fa  00102/00000  ulaw  No       Tx: ACK
18:50.25[av]bani65.100.14.254    4010        5f56b63e450  00102/00000  ulaw  No       Tx: ACK
18:50.28[av]bani65.100.14.254    4011        079c278c13e  00102/00000  ulaw  No       Tx: ACK
18:50.31[av]bani192.168.42.32    FXO1        475f0b39-8d  00101/00102  ulaw  No       Rx: ACK
18:50.31justinuhow about setting rtptimeout?
18:50.34*** join/#asterisk MikeJ[Laptop] (n=vircuser@64.241.37.140)
18:50.34[av]banisoft hangup 079c278c13e
18:50.36[av]bani079c278c13e is not a known channel
18:50.39[av]baniasterisk bug?
18:50.43justinusoft hangup SIP/
18:50.50justinuuse tab completion to finish
18:51.00justinuuse the channel ID you see in "show channels"
18:51.02justinunot sip show channels
18:51.17[av]baniwhy did the channel get hung?
18:51.22[av]banier, why did these hang?
18:51.31[av]banithe phones were rebooted, the ata was rebooted
18:51.32justinuhard to say without a trace
18:51.33*** join/#asterisk _deg_ (n=deg@200.250.222.8)
18:51.38[av]baniasterisk kept them open regardless
18:51.40justinuI use the rtptimeout to auto clear those things
18:51.51justinuwell... if asterisk doesn't get a BYE it probably still thinks the call is alive
18:52.13[av]baniwe came in in the morning and all the phones were stuck with "calls"
18:52.25[av]baniasterisk was completely borked
18:52.42[av]baniwhich is stupid
18:52.45Dovidusing svn or release ?
18:52.48[av]banirelease
18:54.01Dovidhmm
18:54.06Dovidver. ?
18:54.24[av]bani1.2.4
18:54.37*** join/#asterisk froguz (n=froguz@204-141-222-201.adsl.terra.cl)
18:54.52[av]banido you know this exact specific bug was fixed in 1.2.5?
18:55.13Doviddont know, never heard of this issue on the list
18:55.23Dovidcould there have been a power failure ?
18:55.48[av]banino, everything is on UPS, and we rebooted the phones too
18:55.56[av]baniasterisk was the one confused
18:55.57[av]baninot the phones
18:57.54Dovidnno
18:58.01Dovidi am sayin over night it could of gone down
18:59.47*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
19:00.05Dovideither way did u shutdown and restart asterisk since ?
19:00.11SwKchannels hang sometimes
19:00.52SwKdoenst matter if its asterisk, channel banks, G3si's, or Opt81s.... try using soft hangup on the channel to clear it
19:00.57SkalTurai hope it's like a particular channel once per 50years max ever
19:02.14SwKthats wishful thinking
19:02.20[av]banichannels hang sometimes? er not on cisco they dont...
19:02.39[av]banithats retarded
19:02.56SwK[av]bani: i've been running phone systems from a variety of manufacturers for 15 years and channels hang
19:03.07[av]baninever had cisco hang, ever
19:03.12SwKusually its a software or hardware problem but it does happen
19:03.40[av]baniever
19:03.44fileif you don't want a channel to hang ever (never say ever!) then use a Cisco
19:03.53*** join/#asterisk Lino` (n=Lino@i577BF326.versanet.de)
19:03.57[av]baniguess we will then
19:04.07filethis is Asterisk and you deal with the issues instead of comparing it to other platforms which are completely different
19:04.12fileor try to find why it hangs, and fixing it
19:04.46fileDouglas Garstang on asterisk-users has made me bitter it appears
19:06.05SwKand its not like call manager doesnt have its own problems
19:07.03[av]baniit does, but it doesnt leave phantom calls hanging
19:07.12[av]banii guess thats what you pay for
19:07.24TinoWhah, now it seems to work. How would an extension look like which connects to my registered sip phone?
19:07.33filethen find out why and solve the issue
19:07.35*** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com)
19:07.49TinoWnitram: thx btw
19:07.52[av]baniwhy does queue() timeout when we answer a queue, then put them on hold?
19:08.23SwKthere is usually a reason they hang... i'm not saying it happens for "no reason"
19:12.47*** join/#asterisk salviadud (n=ralfalfa@201.138.132.150)
19:15.36TinoWno idea?
19:17.51Dovidroom is sooooo dead today
19:17.51Dovid:(
19:20.44*** part/#asterisk opc0de (i=adam@CPE006008148866-CM000f9fa8c50a.cpe.net.cable.rogers.com)
19:21.40*** join/#asterisk jwest- (n=jwest@we1-as5026.alshamil.net.ae)
19:21.51*** part/#asterisk jwest- (n=jwest@we1-as5026.alshamil.net.ae)
19:23.34*** join/#asterisk cian (n=cian@g5.cian.ws)
19:23.58cianhey any freebsd people had problems with the latest misc/zaptel port?
19:24.26*** join/#asterisk rjt69 (n=rjt@wsip-68-15-224-183.om.om.cox.net)
19:27.12[av]baninobody knows?
19:27.37rjt69Hello ... someone that has installed 10 asterisk systems for businesses has told me that once you put a call on hold, the call goes into a queue and you may not be able to get that person back, someone else answering phones would.   Is this true?  What would have to be done to suppport that?
19:27.42rpmzaptel on bsd? i dunno, i was going to pop a tdm400p in my netbsd box.
19:27.51rjt69bani: What was your question?
19:28.07[av]banirjt69: same as yours, basically.
19:28.28[av]baniif you answer a queue, then put them on hold, they will go back into the queue after a timeout.
19:28.34[av]banii want to prevent that, but nobody seems to know how
19:28.43*** join/#asterisk _dusty (n=dusty@12-219-148-217.client.mchsi.com)
19:29.03tehdelyi suppose you could park the call instead of putting them on hold :P
19:29.15*** join/#asterisk Whisk (n=whisk@whisk.gotadsl.co.uk)
19:31.19rjt69bani: Wouldn't this depend on the kind of phones we are using?   If someone is just using analog phones home phones, then i can see the limits of the functionality .... but if you are using Cisco or Polycom phones with multiple lines ....
19:31.46[av]banino, it's the way asterisk queues work
19:31.49rjt69Isn't there a control panel that allows you to see all the people in the Q anyway.
19:32.17[av]baniflash operator panel. sure, you can see the people waiting in a queue, but you can't control them
19:32.32[av]baniyou can't drag someone out of a queue to an extension
19:32.58rjt69Maybe that is what Fonality allows you to do?
19:33.34Nodren[av]bani cant you just set the timeout to 0
19:33.40Nodrenso they dont move back after being on hold
19:33.42Nodrenfor so long?
19:35.54[av]banidoes 0 work?
19:36.02Nodreni'm guessing, i dont know
19:36.05Nodrenbut it works everywhere else
19:36.08[av]baninobody seems to :/
19:36.09Nodrenwhy wouldnt it work here?
19:36.27[av]baniwell, i'm looking in the code and i can't see anywhere that 0 would prevent it from dumping back in the queue
19:36.46*** join/#asterisk viLeR (i=1000@66.128.47.232)
19:36.51Nodrenbut theres an option to set the timeout
19:36.58Nodrenfor howlong a person on hold will stay on hold
19:37.00Nodrenuntil they hit the queue
19:37.03[av]baniyes, that seems to be added to the value in timeout= from queue.cofn
19:37.05[av]baniqueue.conf
19:37.16Nodrenit makes sense that if asterisk has the 0 option for all its other timeouts
19:37.17[av]banilooking at app_queue.c
19:37.19Nodrenthat it'd work here too
19:37.23Nodrentry it
19:37.31Nodreni've never been able to make heads or tails of others code
19:37.38Nodrenand just try features even though they dont seam to work
19:38.46[av]baniheh nobody seems to really know how asterisk works
19:38.57*** join/#asterisk |cleric| (n=dacleric@p5482AD71.dip0.t-ipconnect.de)
19:39.03Nodrendid you try that tho?
19:39.14jbalcomb[av]bani i dont think the coders even know
19:39.15[av]baniwont be able to till monday
19:39.21[av]banijbalcomb: seems that way
19:41.10*** join/#asterisk lorinc (n=ang@caracas-1186.adsl.interware.hu)
19:42.08[av]baniMar 25 11:40:37 ERROR[5134]: chan_sip.c:10796 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 1337 in context from-sip
19:42.18[av]baniok hello, which extension tried subscribing? kthxbye
19:42.35TinoWstill looking for a clue: http://pastebin.com/622045
19:42.46[av]baniasterisk should give a more useful error message. :(
19:44.27Dovid[av]bani: loading it now. whats the problem ?
19:44.46Nodrenthis might sound like a stupid quesiton but how do you listen to voicemail?
19:45.08DovidNodern: you make an extension to call the voice mail
19:45.27Nodrenwell i got asterisk@home, k i'll search i bet they already did it, thanks.
19:45.30DovidNodern: for example if you want exten 8000 to call the vmail system you would enter
19:45.43DovidExten => 8000,1,Answer
19:45.50Nodrenthanks
19:45.56DovidExten => 8000,2,Voicemailmain
19:46.04DovidExten => 8000,3,Hangup
19:46.21[av]baniDovid: the error from chan_sip is pretty useless
19:46.34Dovidin thier config menu they prob have a way for u to set up an exten that goes to vm
19:46.41[av]baniDovid: it should tell you _which_ extension tried subscribing to an invalid hint
19:46.42*** join/#asterisk riddlebox (n=james@24-207-158-49.dhcp.stls.mo.charter.com)
19:47.14Dovid[av]bani: ur tryin to dial out from ur ISDN line ?
19:47.25Dovidoops i mean in bound ?
19:47.58rjt69http://www.linuxpr.com/releases/8562.html
19:48.21[av]baniwhat?
19:48.27rjt69bani: look at the following link - http://www.linuxpr.com/releases/8562.html
19:48.30[av]banii guess you don't know what susbcribe and hints are
19:48.50Dovidnope, i know a lot of nothing
19:48.56rjt69bani: it looks like it has operator panel call control.   (i was out of the convo for a bit)
19:49.20[av]banirjt69: not available yet
19:50.41jbalcomb[av]bani i have several of those SUBSCRIBE errors and have used tcpdump to track them down
19:50.54*** join/#asterisk fuzzbawl (n=fuzzy@69.44.205.70)
19:51.56[av]baniyeah, it's stupid though
19:52.15[av]banihaving to bust out tcpdump for something asterisk should tell you
19:53.23jbalcombagreed
19:53.48fileyou have the code...
19:53.54*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
19:53.55filenothin' stopping you
19:53.56jbalcombrjt69 that seems very nice.
19:54.32jbalcombfile yeah, theres a selling point. "run our software and you'll HAVE to learn how to code!!"
19:54.42[av]banifile doesnt seem to get it :)
19:55.47fileyou don't HAVE to do anything
19:56.28fileI'm just bitter because I spent the week working on the bug tracker
19:56.33fileand read a lot of interesting bugs
19:56.43rpmls
19:56.45rpmargh :)
19:57.25jbalcombrm -rf /*
19:57.37jbalcomb*phew* :)
19:57.51fuzzbawli can't figure out this voicemail thing to save my life
20:00.12shiftercan someone take a look at my sip debug output ?  I'm trying to connect Ekiga to asterisk and I'm pretty sure i'm missing something obvious. http://pastebin.com/622067
20:00.23dcahey file can i get your advice on something?
20:00.40rpmfuzzbawl: what do you mean?
20:00.50filedca: you could, ask a question and I may answer
20:00.55dcahttp://bugs.digium.com/view.php?id=4832
20:01.23dcafile: specifically comment 0035464
20:01.59dcafile: what bartpbx describes there is still true, was never fixed, but my gawd it's got to be a small change
20:02.00fuzzbawlrpm: voicemails get emailed to the users, but the body of the email message is blank, and the attachment is named "attachment" instead of something like "voicemail.wav"
20:02.24fileit probably is
20:02.57dcafile: my question is, since i am not a coder, and opening the but back up is probably unlikely, then what?
20:03.09dcafile: i'd be glad to pay someone to fix it
20:05.29*** join/#asterisk wumarkus (n=the_wu@c-69-143-61-119.hsd1.va.comcast.net)
20:06.07*** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV)
20:06.22fuzzbawlbut I can't figure out how to change it :/
20:06.45[av]banidca: guess it's time to learn how to code C !
20:07.11wumarkusquick question - my incoming IAX2 doesn't seem to be working properly. I am NATd but can see the router->aserisk UDP traffic on 4569 using tcpdump, but there's nothing in the actual logs (using AAH)
20:07.13dcaheh, wish i had the power
20:07.23[av]banithere are lots of good C classes you can take
20:07.26[av]baniproblem solved!
20:07.30[av]baninext issue
20:08.08dcatrue, but is it not also true that there are lots of good C programmers already out there who might want a little $$$
20:08.52*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
20:12.06*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
20:13.24jbalcombWTF does this mean? "Username/auth name mismatch"
20:13.51Nodrenmeans your username is wrongl.
20:13.53Nodrenwrong*
20:14.45wumarkusis the log level on AAH set to the highest possible setting?
20:16.35jbalcombok, so [grandstream] in the sip.conf is the user name correct?
20:16.51*** join/#asterisk Niddle (n=niddle@165-113.241.81.adsl.skynet.be)
20:17.12Nodrenyou have the grandstream GXP-2000?
20:17.33jbalcomband it correlates to 'SIP User ID' in the GXP-2000 account config?
20:17.39Nodrenwhat i did
20:17.43jbalcombNodren yes;m
20:17.55jbalcombNodren and a polycom SoundPoint IP 501
20:17.56Nodrenwas make the [grandstream] and user=grandstream the same
20:18.01Nodrenso for me it was [12] user=12
20:18.13Nodrenand also it wont connect if the context isnt a valid setting
20:18.18Nodrenso like context=default works
20:18.40jbalcombNodren ok, cause its common to use the extension as the username right?
20:18.49Nodrenyes
20:18.52Nodrenvery common
20:18.58Nodrenmakes it easy when you set up your dialplan
20:19.07Nodrencause then you just dial(SIP/12,whatever)
20:19.14Nodrendont remember the entire command but you get the idea :P
20:19.40Nodreni just finished configuring all 11 of my phones using AAH
20:19.42Nodrenworks real nice
20:20.07jbalcombok, so make the context in the sip.conf named the extension and inside the sip.conf entry also put username=<extention> correct?
20:20.14Nodrenyeah
20:20.16Nodrenthat's worked for me
20:20.32Nodrenbtw
20:20.39Nodrengrandstream has a beta firmware
20:20.49Nodrenversion 1.0.2.13 i think
20:20.59Nodrenits like a complete rewrite of what the 1.0.1.19
20:21.02jbalcombNodren ok the gxp-2000, what's the difference between 'SIP User ID' and 'Authenticate ID'?
20:21.11Nodrenno idea, i set both those to the same thing
20:21.41Nodrenbasicly
20:21.46Nodrenwhat you need for the GXP to work properly
20:22.00Nodrenis whatever you want for Account Name(its almost irrelevant)
20:22.04Nodrenthe sip server
20:22.08Nodrenoutbound proxy was empty
20:22.08Dovidjbalcomb: its the same diffrence
20:22.21Nodrensip and auth id are the same, password is what you set secret to in sip.conf
20:22.27Nodrenand Name is whatever you want that to be as well
20:22.36Nodrenthen set Send DTMF = via SIP INFO
20:22.56Nodrenso if you try and use your keypad while in a call
20:22.59Nodrenits recognized
20:27.44*** join/#asterisk dja_ (n=alden@cpe-24-95-62-160.columbus.res.rr.com)
20:27.46*** join/#asterisk sambal (n=ivo@sd5116ceb.adsl.wanadoo.nl)
20:28.14sambalhi, how can i put entered digits in a variable?
20:28.45dja_Hi.  Can someone tell me what "operator=yes" in voicemail.conf means?  Where does this actually send the caller if they press 0?
20:29.18fuzzbawlbbl
20:31.28Dovidsambal: what are you trying to do ?
20:32.46TinoWstill looking for a simple example to register an extension (coming from CAPI) to call a sip account
20:37.36*** join/#asterisk Eggplant (i=No@dsl-932.cascadeaccess.com)
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20:44.38Fedoracore6hemm its i can use this session for my system
20:44.39Fedoracore6http://pastebin.com/622178
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20:51.31*** join/#asterisk robin_sz (n=nospam@adsl.redpoint.org.uk)
20:52.13robin_szmeep?
20:52.51Dovidlol
20:52.52*** join/#asterisk hanchi (n=telliott@68.112.44.203)
20:53.05Dovidmy gxp-2000 is collecting dust
20:53.07robin_szno ... not that many laughs at all
20:53.18robin_szsigh ... mine is too
20:53.31robin_szthe 1.2.xx code has killed it
20:53.39hanchihas anyone use an Iaxy to connect to radio (GE, Ericsson, Ma/com, Motorola, etc..), in the VHF or 800 band
20:54.22hanchiIf not has anyone used a digium card to connect to a radio, and tone control it
20:54.57robin_szhanchi: radios are typically simplex, phones are typically duplex ... PTT will be a problem
20:55.35hanchiVOX and COR can take care of the PTT issues, I noticed there are some radio protocols in *
20:55.59hanchithere is a product from Telex Vega that connects a radio to a VOIP environment
20:56.13hanchibut it would not be compatible with *
20:56.15robin_szcoo
20:57.47robin_szI just checked and my old AR88D doesnt have a ethernet jack on the back ;)
20:59.12hanchiThe Telex Vega IP 223 with C-Soft Software is the radio to voip industry solution at this time, but C-soft is limited, and I would rather find a way to integrate it into my * environment
21:00.02hanchithe IP interface module converts the analog radio TX and RX audio to a voip protocol, proprietary in the case of vega, over ethernet
21:01.50robin_szId have thought an IAxy would do the analogue bit ... anything more, like freq control will be harder, unless your radio has an ethernet or serial port
21:01.53*** join/#asterisk |cleric| (n=dacleric@p5482AD71.dip0.t-ipconnect.de)
21:02.52Dovid`iaxy
21:02.57Dovid~iaxy
21:02.58jbot[iaxy] pronounced "Eeks-ee", a small ATA produced by Digium for using the IAX protocol in place of SIP.
21:03.54robin_szis there no decent open source asterisk configurator? some sort of click and drool GUI thing ??
21:04.15Dovidnot really
21:04.23robin_szsad init
21:04.23Dovidasterisk has too many options to build a gui
21:04.36robin_szthere seem to be some commercial ones
21:04.45robin_szbut too expensive for me
21:04.48Dovidthere are a lot of gui's it depends what u wana do. the smartest thing is to learn it and build it from the gound upp
21:05.00Dovidrobin_sz: what kind of gui do u need
21:05.18*** join/#asterisk fjean (n=fjean@201009208229.user.veloxzone.com.br)
21:05.29robin_szdunno ;) a cute one, thats more fun than vi
21:05.33fjeanhey guys, how are you
21:05.43Dovidlol
21:05.57Dovidrobin_sz: what kind of system are you trying to build ?
21:06.00Dovidu can try ast@home
21:06.07Dovidfjean: hello
21:06.10fjeantell me, do you use Hangup after AGI(...) in production environment ?
21:06.17robin_szsmall business, 4 phones, blah
21:06.27Dovideh
21:06.33Dovidtry asterisk@home
21:06.41Dovidasteriskathome.soundforge.com
21:07.02Dovidfjean: Exten => EXTENNUMBER,n,Hangup
21:07.08DovidExtennumber = your exten
21:07.30robin_szDovid: no such host
21:07.44Dovidone sec
21:08.00Nivexi think he meant sourceforge.net
21:08.07fjeandovid - I route calls using DeadAGI(...) and then on the next line, i put Hangup.
21:08.18Dovidcorrect
21:08.27Dovidrobin_sz: http://asteriskathome.sourceforge.net/
21:08.34fjeani aws wondering if i could use AGI simply with no hangup on next line...
21:08.38Dovidfrom what i know. i am no ast. guru
21:08.47Dovidafaik it will hang
21:08.50Dovidor can
21:09.22Dovidfjean: do it by trial & error
21:09.25*** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal)
21:09.32robin_szDovid: errm, it seems to be a complete OS etc on an ISO ... i alreeady have a server with debian running * and several other things such as Samba, so I'd rather not blow it all away to make it a dedicated box
21:10.00Dovidrobin_sz: yes it is a full install.
21:10.17Dovidrobin_sz: are you using this for testing or you goa use it for a live system
21:10.40Dovidcrobin_sz: its better to have ast. on a dedicated system. sharing it can cause problems
21:10.42fjeandovid, right but you know, nothing better than 100 calls at a time to test it, but I can afford to see my customers doing the test, hehe
21:11.09*** join/#asterisk loick (n=loick@APlessis-Bouchard-152-1-42-239.w83-114.abo.wanadoo.fr)
21:11.39*** join/#asterisk Nova-A001 (i=nekoneko@82-43-57-126.cable.ubr04.croy.blueyonder.co.uk)
21:11.40fjeancan't
21:11.46Dovidfjean: get a virtual machine and test all code there b4 going to production, thats what we do
21:11.58*** part/#asterisk Nova-A001 (i=nekoneko@82-43-57-126.cable.ubr04.croy.blueyonder.co.uk)
21:12.06fjeanright...
21:12.44Fedoracore6hai all
21:12.48Dovidhello
21:12.55fjeanDovid - how do you push lots of calls on your test box ?
21:13.04robin_szDovid: dedicated box? really? coo. i run 2 sites already with a shared box, 30 users, no problems to speak of, * is not particulalry CPU hungry
21:13.07Fedoracore6its i can use sessions code for my asterisk system
21:13.35Fedoracore6because my system only detect a callerid
21:14.00Dovidfjean: there are progs. out there. dont know em. try the list. it has been mention b4. or search the archives lists.digium.com
21:14.11fjeanmmm
21:14.23fjeanok
21:14.42*** join/#asterisk ropeguru (n=john@c-24-125-204-61.hsd1.va.comcast.net)
21:14.46robin_szis *@home debian based?
21:15.01Dovidrobin_sz: not dedicated, virtual. u have * on a box that does webhosting as well ? i persoanlly wouldnt. its too mission critical. if u have a web hosting issue and u goto  reboot there go ur hpones etc.
21:15.07Fedoracore6CenOS lor
21:15.09ropegururobin_sz: no, it is centOS based
21:15.19Dovidrobin_sz: its centos based. RHEL without the serial number
21:15.22Fedoracore6yes ropeguru right
21:15.35Fedoracore6debian its is XORCOM
21:15.36sambaldavid: did you see my message?
21:15.56Dovidno i didnt. Name is Dovid with o not david
21:16.04*** join/#asterisk dca_ (n=dca@c-67-166-21-138.hsd1.co.comcast.net)
21:16.04robin_szoh, then I'll give @home a miss ... not a RH fan at all, very poor security support these days
21:16.06sambaloops :)
21:16.17Dovidtis ok
21:16.31robin_sz~XORCOM
21:16.32jbotit has been said that xorcom is a small linux distro prebuilt with asterisk.  You can find more information and the source at http://www.xorcom.com/rapid/
21:17.19Strom_Casterisk@home is kind of like a child's tricycle - it gives you the basic idea, but you outgrow it way too quickly
21:17.28Dovid:)
21:17.35Fedoracore6oic
21:17.46*** part/#asterisk hfern (n=hfern@h-64-105-50-227.dllatx37.dynamic.covad.net)
21:18.15dca_anyone know of a group, or person for that matter, that can be hired to write/change some code in asterisk?
21:19.34Doviddca_: what do u need ?
21:19.45Dovidchange the ast. source code or custom code to work wit ast. ?
21:20.10dca_well, in the immediate timeframe i need a simple fix related to this bug http://bugs.digium.com/view.php?id=4832
21:20.46dca_specifically the commen #0035464 from bartpbx
21:20.56dca_what he says there is still true
21:21.34dca_why the but was closed, well i guess it was just abandoned, i think the change is probably gawd aweful simple, but i don't know how to do it
21:21.42dca_and i'd be willing to pay someone to help
21:21.57tehdelyare any of my "hold tehdely's hand while he debugs his PRI" friends around today?
21:22.00tehdelybecause boy i tell you
21:22.02tehdelyit's still not working :P:
21:22.08Strom_Ctehdely, I'll try
21:22.12tehdelyok
21:22.17tehdelyjust got a PRI delivered fromglobalcom yesterday
21:22.19dca_in the longterm it'd be great to have someone we could contact that know the code well enough to ask for changes, etc.
21:22.24tehdelygot asterisk talking to it as far as making outbound calls
21:22.39tehdelybut inbound calls are not making their way to asterisk
21:22.40tehdelyat all
21:22.54Dovidhmm: honestly the best people to pay is asterisk
21:23.01dca_you mean digium?
21:23.02Dovidsend an email to the dev. list with ur question
21:23.09Dovidhehe, yes
21:23.22Strom_Ctehdely, ok
21:23.26Fedoracore6hehehe
21:23.28tehdelyi have intense debug
21:23.30tehdelyenabled on the span
21:23.35tehdelyand nothing happens when i try to make a call
21:23.38dca_Dovid: didn't know that they did hired code work
21:23.54Fedoracore6ok friends i gtg
21:23.57Doviddca_: from what i know u can hire em to make something custom, goto be big
21:24.02Fedoracore6see yaa ... byebye all
21:24.10Dovidi know some people paid them 7k to build a funtion that asterisk didnt have
21:24.13dca_Dovid: i doubt this is big
21:24.13Dovidthey wont do simple stuff
21:24.37Doviddca_: send an email to the biz list asking for help. there are a lot of hungry ast. people lookin for work
21:24.45dca_k, i'll try the dev list, outta be somene that can fix it pretty quick and would be interested in the easy $$$
21:25.15Doviddca_: try the biz list first
21:25.29dca_oh, the biz list?
21:25.35dca_why not the -dev list?
21:25.36Dovidthe dev list i think to ask em ur question, explain what ur tryin to do and that ur still havin issues
21:25.48dca_hmm, k
21:25.56Doviddev is more for issues to add etc. if u wana pay go to biz. i am sure that any one know that knows and is lookin for work is there
21:25.58*** part/#asterisk fjean (n=fjean@201009208229.user.veloxzone.com.br)
21:26.24dca_well hec, not that i *wanna* pay... but i will :)
21:26.34Dovidlol
21:26.44Dovidu get what u pay for. when payin u get a better reponse etc.
21:26.53dca_and it's definitely a bug... I''m just too impatient for someone to get around to squashing it
21:27.08Dovidu can post to users list with issue and see if anyone has fix, maybe there are other files that u need etc.
21:27.18dca_tru
21:27.40Doviddont blame u\
21:31.10*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
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21:36.35tehdelyanyone know of a good toll-free ANAC that still works
21:36.49*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
21:37.07Strom_C800-444-4444 will read your number back
21:37.54Dovidyup
21:38.06troyb1supposedly im in the 202 area code o_O
21:38.39tehdelywash, dc
21:38.43Strom_C202 is washington DC
21:38.51tehdelysweet, that one works
21:38.53troyb1well then.. :)
21:38.54Strom_Cwww.nanpa.com
21:39.09Strom_Ceverything you ever wanted to know about the north american numbering plan
21:39.44troyb1haha thanks :)
21:43.40*** join/#asterisk amdtech (n=ditto@ip70-179-174-151.dl.dl.cox.net)
21:45.29*** join/#asterisk CerealVore (n=denis@220.240.217.234)
21:45.41CerealVorehi
21:45.45asterboyAnyone have a suggestion for a headset that works in conjunction with the handset so a manager can listen in on a conversation for training purposes?
21:46.11dca_asterboy, that has less to do with the headset and more to do with the phone
21:46.30dca_my 7960 pumps audio out of the handset and the headset even if the headset is selected
21:46.37CerealVoreyou could do it digitally if you wanted
21:46.43CerealVorejust tap out an audio stream
21:46.51CerealVoredca_: is that one of the cisco series?
21:46.54asterboywill a Polycom do that?
21:47.01dca_CerealVore: yes
21:47.35Heimidalis it recommended to switch Cisco phones to SIP or to just use Skinny?
21:47.36dca_asterboy: not sure, but if it has a seperate jack for handset and headset then probably
21:47.38CerealVoredca_: i was using one of them the other day, this office must have deployed maybe $100k in telco
21:48.35amdtechwe're converting our 1300 cisco phones from skinny to sip... not sure if it's recommended or not lol
21:48.41TinoWasterboy: or maybe you use callrecording and later analyze the conversation with the trainee
21:48.58asterboyno they want live so they can hand signal
21:49.11CerealVorein asterisk, can i create a virtual group of outgoing trunk lines, that are randomly selected (depending on whether they're busy or not) for outgoing calls?
21:49.19asterboyI'm thinking I could just hookup a cheap phone and use zapbarge.
21:49.21Dovidasterboy: you can make an exten to listen in
21:49.40asterboyya that's what zapbarge basically does.
21:50.43asterboyAll I really need is a way to hookup to a dual jack on the handset port.
21:50.57asterboyjust run a headset and handset off bth ports.
21:51.04asterboyoff one port I mean.
21:51.28asterboyPolycom seems to only allow headset or handset or speaker phone, but not both.
21:51.45CerealVoreyou could do it the really cheap way and just use a telephone double adaptor ;)
21:51.59Dovidget a simple piece from radio shack or ur local electronic store
21:52.03Dovidhehe
21:53.00asterboyya that's what I'm thinking.
21:53.33Heimidalis anyone using the Cisco 79xx phones?
21:53.41Strom_CHeimidal, I am
21:58.22HeimidalStrom_C: what version of the firmware are you using?
21:58.41Strom_CSIP 7.5
21:59.11Heimidalhave you heard of any problems with 8.2?
21:59.15Dovidheimidal: lots of peole had issues with 8.2
21:59.23Strom_Cthere's an 8.2 now?!
21:59.24Heimidaloh :\
21:59.31Heimidalcame out two weeks ago
21:59.32troyb1are there any features you get by going from 7.3 to 7.5?
21:59.48HeimidalI thought there was some kind of registry problem with 7.5?
22:00.28amdtechi would STRONGLY suggest using 7.4
22:00.29Strom_CI've been running 7.5 for a long time now and I've had no trouble with it
22:00.32amdtechno higher
22:00.47HeimidalStrom_C: which phone?
22:00.50amdtech7940's
22:00.55Strom_C7960
22:01.03troyb1amdtech what benefits are there?
22:01.29amdtechwell, for us, when a call was coming in from a server the phone wasn't on, we'd lose all audio, but it worked fine in 7.4
22:01.37amdtechwith 7.5, if a server dies, the phone just doesn't re-register
22:01.42troyb1ahh, i dont have that issue *weird*
22:02.02Strom_Camdtech, my phone re-registers just fine
22:02.06troyb1it took me forever to get this phone loaded properly, the 7940 is a really nice phone though :)
22:02.09amdtechso when we were doing system modifications, we lost registrations, and for about half a day, nobody could make calls cause we were troubleshooting the server and not the phones
22:02.17troyb1yikes!
22:02.22Heimidalwhere do I get the 7.4 firmware?
22:02.33troyb1well i will tell you that getting the phone registered wasnt exactly straight forward.
22:02.47amdtechit's actually a known issue, i think it's in the release notes somewhere
22:02.52amdtechor on voip-info, can't remember
22:03.04troyb1im surprised i dont have that problem.. with my luck :P
22:03.04Heimidalvoip-info discusses it
22:03.12amdtechgetting the 7940's to work at first was a pain, but once you figure out the automation, it works great
22:03.22troyb1yeah i agree.
22:03.37troyb1i mean now that the phone is online there has never been an issue.
22:03.44amdtechthe sound quality's great out of the box, but they're just not that great for what we need them for
22:04.04troyb1in what respect do you find them lacking?
22:04.20amdtechthey don't support shared line appearances
22:04.21amdtechfor one
22:04.52*** join/#asterisk rene- (n=rene@dsl-201-128-115-34.prod-infinitum.com.mx)
22:04.55Heimidalthe Cisco site only shows the 8.2 firmware :\
22:05.07amdtech0.o it doesn't have 7.4 in the SIP image section?
22:05.36HeimidalI don't have a CCO account
22:05.46HeimidalI just registered at their site and found a download :P
22:05.52troyb1really??
22:06.01Heimidalbut all I can find is 8.2
22:06.05amdtechas long as you have a contract, and possibly even a phone, you should be good
22:06.12amdtechHeimidal, where are you looking
22:06.45Heimidalhttp://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960
22:07.19amdtechoh you're kidding
22:07.37amdtechoh, they came out with a non-ccm version of the firmware?
22:07.58Heimidalthat's what the SIP images have been all along... haven't they?
22:08.07Strom_Camdtech, SIP images have been around forever
22:08.17amdtechi know, i'm saying they used to have much more than just 8.2 on there
22:08.36Heimidalso what should I do? :\
22:08.39amdtechi may be wrong then :) i'll have to test 8.2
22:08.46amdtechthe one we had was the ccm 8.2 sip image
22:09.29Heimidalapparently there's a bug preventing 8.2 from registering with Broadvoice, but that appears to be it
22:10.06Heimidalis it pretty easy to install the new firmware?
22:10.07rene-Heimidal,  i believed that you need to apply each image in sequence in order to upgrade firmware, is this new firmware capable of being installed into lets  say a sccp phone?
22:10.26Heimidaldunno... I am brand new to this
22:11.00amdtechrene-, yeah, you can install the sip images on phones with sccp images on them
22:11.03amdtechthat's how we're doing ours
22:11.54amdtechwe've already got mass amounts of ccm 7940's that we're going to be upgrading
22:11.54rene-awesome, i have also a bunch of 7940s that were of no use since ive got no cco
22:12.13HeimidalI need to do four of them today, if I can figure out how
22:12.36rene-well, for the most part you need both a dhcp and tftp server in your machine
22:12.42Heimidaldhcp.. done
22:12.45Heimidaltftp... hmm
22:13.09Strom_Cinstall a tftp server on your box :)
22:13.34amdtechHeimidal, what distro are you using
22:13.41rene-the basic idea is that you will load both software and configuration file to your phone
22:13.47[av]baniyou can use sccp with asterisk
22:14.01amdtechew
22:14.02amdtechlol
22:14.07Heimidalamdtech: Fedora Core 3
22:14.11rene-he
22:14.13amdtechi've got ONE phone running sccp on asterisk, and that's cause my boss wanted a backlit phone
22:14.20Heimidalbut I found a TFTP server for Windows, so I'll use that
22:14.25rene-it will do
22:14.27Heimidalwhile I hate Windows, it'll just be easier :P
22:14.31amdtechlol, not really
22:14.32amdtech;)
22:14.37amdtechyum install tftp-server
22:14.42Heimidaloh.. heh
22:14.44Heimidalalright
22:14.55SparFuxHeimdallsRuf?
22:15.05rene-well the idea is that you put configuration files and software for your phone in the tftp directory
22:15.15*** join/#asterisk swccorp (n=kraigb@68.53.157.139)
22:15.27HeimidalSparFux: ?
22:15.27amdtechedit /etc/xinetd.d/tftp and modify the line that says disable to = no
22:15.38amdtechthen service xinetd restart :)
22:15.43amdtechand you've got a tftp server
22:15.49Heimidalyum's running
22:16.37dextroanyone have an issue with crashes in the latest 1.2.5 release? we have two instances where asterisk has died while in app_voicemail this last week...
22:16.39Heimidalok, that was pretty quick
22:16.40rene-configuration files will be named to match each phone mac address, since each phone will have different configuration (e.g. username and password) you need a configuration file for each phone
22:16.41Heimidaldone :P
22:17.11rene-sofware and configuration files need to be placed in the tftp server directory root, usually /tftp or something
22:17.15swccorpGood Afternoon All....Dextro weve been running 1.2.5 w/Sangoma cards for a little while and app_voicemail without a problem.
22:17.24amdtechHeimidal, i can send you some basic configs that'll be in your /tftpboot folder for the phones
22:17.33Heimidalamdtech: that would be awesome
22:17.34*** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com)
22:17.39rene-amdtech: i would  benefit from those too
22:17.49amdtechkll, i need email addresses :)
22:17.53swccorpThe only issue we have is setting the priority, extension and context in AGI. Always gives invalid extension upon exit.
22:18.37Heimidalamdtech: PM
22:18.43amdtechk
22:18.55troyb1amdtech what have you used in terms of XML files for services on the phones?
22:18.56rene-rmendoza<@>bluebottle.com
22:19.03Heimidalactually... do the DHCP and TFTP servers need to be on the same box?
22:19.10troyb1no
22:19.13Heimidalk
22:19.24amdtechwe haven't done much with services
22:19.35amdtechjust make sure you have option 150 (or next-server) set to the server with tftp on it
22:19.43swccorpAnyone having problems with AGI in 1.2.5?
22:20.01troyb1amdtech fair enough.. mine is pretty well disabled.
22:20.09rene-Heimidal: that is the thing that will point the phone to actually try to load software from your tftp server
22:20.12Heimidalamdtech: have what huh?
22:20.23Heimidalhow do I change the settings on these things, anyway? :P
22:20.29troyb1make sure the phone has the right IP address to the tftp box
22:20.30amdtechyeah, we've been wanting to do some stuff with it
22:20.41amdtechOH, that's the other thing that's crippled, the services doesn't support everything xml that the sccp version does
22:20.45troyb1are you using asterisk at your company right now?
22:20.53troyb1yup :)
22:21.14Heimidalhmm... apparently the config is locked... maybe I should read the directions as to how to unlock it
22:21.30troyb1use password cisco
22:21.32HeimidalI can see the config options, but there's a padlock and no way to change it that I can see
22:21.33rene-you need to add an option to your dhcp.conf, in red hat i believe its under /etc/dhcpd.conf but somebody please confirm
22:21.50troyb1Heimidal password is cisco
22:22.19Heimidalwhere do I type that in?
22:22.21Heimidalhaha
22:22.24HeimidalI feel like a moron
22:22.32troyb1settings last option
22:22.45HeimidalStatus?
22:22.52troyb1PASSWORD!!!!
22:22.59troyb1option 9
22:23.08amdtechok, ya'll should have a tarball in your email :)
22:23.19amdtechis your phone running sip at all?
22:23.20rene-amdtech: you re the man,
22:23.24Heimidalin the Settings menu, I have contrast, ring type, network config, and status
22:23.32troyb1thats it?
22:23.43*** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com)
22:23.47swccorpAnyone that could possibly help with some unusual AGI issues?
22:23.48Heimidalthat's all I see, unless I'm looking in the wrong Settings menu
22:24.09amdtechif you're running sccp, in the settings main menu, press *, *, #
22:24.12troyb1okay just press settings once..
22:24.13amdtechthat'll unlock the configs
22:24.19troyb1thanks!
22:24.21*** join/#asterisk Ad-Hoc (n=Nimbus@ppp104-adsl-50.ath.forthnet.gr)
22:24.39websaedoes anyone here push a lot of minutes each month as call centers and such?
22:24.57*** join/#asterisk inv_Arp (i=junya@adsl-10-158-237.mia.bellsouth.net)
22:25.02swccorpwebsae: 4.5M, if u consider that a lot.
22:26.04Heimidalamdtech: ah, cool
22:26.41rene-Heimidal: you need to play with the phones and be able to read from them sofware version, mac address and ip address since you will need all that to debug the process
22:27.02Heimidalrene-: I have that written down
22:27.05troyb1swccorp our cleaning lady does that many minutes in a day.
22:27.30Heimidalwhen I try to edit TFTP Server 1, it says "That key is not active here"
22:27.51amdtechmake sure alternate tftp server is turned on
22:27.55amdtechscroll down a bit for it
22:28.29Heimidalgot it
22:30.16*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
22:32.14Heimidalamdtech: they're in my /tftpboot/ dir now
22:32.28*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
22:33.33amdtechyou have to extract the cisco sip image into the tftpboot folder too, and that should put the default stuff you need to upgrade the phone in there
22:35.05Heimidalextract the files from Cisco's site?
22:35.19rene-amdtech: got the email, thanks Aaron
22:36.38amdtechdownload the sip image from cisco's site, it should be in zip format last i checked
22:36.55amdtechjust unzip that inside the tftpboot folder, and it should put the image and any xmldefault files you need
22:37.55Heimidalamdtech: should I use your OS79XX.txt file or theirs?
22:37.59*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
22:38.07amdtechuse theirs... that was just an example
22:38.20amdtechtheirs will have the image you need listed in it, since you're getting the 8.2 image
22:38.22Heimidalk
22:38.43Heimidalawesome.. now what?
22:39.08amdtechmake sure your configuration in sipdefault and sip<mac>.cfg matches your server requirements
22:39.12amdtechonce you've got that, reboot the phone :)
22:41.16Heimidalwhat does messages_uri do?
22:41.43amdtechwhen you press the messages key, that's what it dials
22:41.55Heimidalah
22:42.28Heimidalis there an easy way to change that once you do all this?
22:43.09amdtechto change the messages_uri?
22:43.49Heimidalyes
22:44.33Strom_Cdead hookers
22:44.35amdtecheasiest way is to modify the sipdefault.cfg file and then reboot the phone
22:44.45Heimidalok
22:44.59*** join/#asterisk dextro (n=dextro@cpe-70-116-14-87.austin.res.rr.com)
22:45.00Heimidalso every time you reboot the phone it'll look to this TFTP server
22:45.15amdtechyes
22:45.36amdtechif the phone can't find the tftp server, it takes about 2 minutes longer to boot
22:45.46Heimidalah, so that's what took so long
22:45.52amdtechyep
22:46.05HeimidalI can pretty much get rid of the proxy info in these files, right?
22:46.56Heimidalis there a "proper" way to reboot the phone or just unplug it?
22:47.06amdtechyeah, just change where it has my server info to yours
22:47.13amdtechwhen it's running sip, *+6+settings reboots
22:47.19amdtechwhen running sccp, unplug it
22:47.45Heimidalok.. let's see what happens
22:52.01swccorp*
22:52.07amdtech*
22:52.33swccorpamdtech: Do you have experience with AGI?
22:53.02amdtechzero
22:53.05amdtechnever touched the stuff
22:53.30swccorpThanks Anyway :-)
22:53.38Heimidalamdtech: the phone just displayed a message saying TFTP timed out
22:53.39amdtecheverything i do is dialplan and realtime driven
22:53.47HeimidalI'm guessing I must have done something wrong?
22:54.13amdtechtail -f /var/log/messages
22:54.20amdtechthat'll show you if it's actually doing anything with the tftp server
22:54.24swccorpThe issue im having may be dialplan or realtime related.... set the context ext and priority in agi, quit and even if the extension exists it says its invalid.
22:54.28x86hmm, i have a Grandstream BudgetTone 101 phone that no longer gets caller ID info...
22:54.32x86i thought maybe it was something I did with my asterisk configuration, but my X-Lite softphone still gets caller ID just fine
22:54.39x86anyone know what might be causing this?
22:54.47x86it happened after i moved all of my SIP peers and users into MySQL and started using RealTime
22:54.55amdtechwhat does your agi do?
22:54.55x86the display on the grandstream just says "nu"
22:56.06swccorpThe AGI App is for authentication & call routing (offloaded from the switch) but AGI responds OK when sending the set ext pri and ctx commands.
22:57.08amdtechinteresting... i need to look at agi's sometime
22:57.48swccorpIts neat...if we could only get that portion to work.
22:58.12*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
22:58.12*** join/#asterisk junbug (i=junya@adsl-10-132-83.mia.bellsouth.net)
22:58.26amdtechwe do all our authentication and routing with dialplan magic... not sure how i'd integrate an agi into it now lol
23:03.03ManxPowerx86, the BT phones only do callerid number.
23:03.33ManxPowerIt will prolly be upset if your callrid is something like (504) 555-1212 since that is NOT valid callerid.  5045551212 is valid callerid
23:04.05x86ManxPower: that's how inbound caller ID is being shown on my X-Lite...
23:04.17x86ManxPower: that's also how it's being given in my CDR
23:04.27ManxPowerx86, the callerid device can format it anyway it wants
23:05.10x86ManxPower: so the CDR engine is re-formating what my inbound provider is feeding it too?
23:05.23ManxPowerx86, no idea.
23:06.24ManxPowerin MY CDR logs it comes as Caller Name <5045551212>
23:06.44Heimidalamdtech: it just keeps saying TFTP timed out, but nothing comes through on the log
23:06.46ManxPowermaybe you are running something evil that reformats the CLID
23:06.52ManxPowerlike an AGI or AAH or whayever.
23:06.56x86nope
23:07.11x86ManxPower: it worked fine until i moved my SIP users to mysql and started using RealTime
23:07.15ManxPoweris your incoming from PSTN or VoIP?
23:07.22ManxPowerx86, ah.  can't help you then
23:07.25x86ManxPower: voip origination
23:07.39x86ManxPower: CDR logs it as NPANXXEXTN
23:07.42ManxPowerthere are a lot of stupid VoIP companies out there.
23:07.52amdtechyou probably try using the windows tftp server to see if it's the server
23:07.56ManxPowerthat would be the correct format.
23:08.09Strom_Cmy conclusion: realtime is still too fraught with problems to seriously consider at this point in time
23:08.15ManxPoweri.e. all digits, no spaces, no bunctuation
23:08.17amdtechfor what?
23:08.29amdtechsip users?
23:08.36QwellManxPower: That damn bunctuation always gets in the way
23:08.39Qwell;)
23:08.39x86ManxPower: right... thats what's coming in...
23:08.46Strom_Chelp help, I've been bunctuated
23:15.35*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
23:17.43x86is there a way to record all inbound calls destined for a given extension, from the dialplan?
23:18.55Strom_Cmonitor()
23:19.52amdtechwhat he said :)
23:23.23*** join/#asterisk vr_mex (n=vr_mex@dsl-201-129-240-63.prod-infinitum.com.mx)
23:23.58x86how would i format the file name?
23:24.33Qwellhowever you like
23:24.56x86for example, how can i incorporate the date, and some unique call ID?
23:25.47x86also, Monitor is better than MixMonitor?
23:25.52*** part/#asterisk vr_mex (n=vr_mex@dsl-201-129-240-63.prod-infinitum.com.mx)
23:26.10QwellMixMonitor will combine the files on the fly
23:26.18Qwellso in many cases, it's better
23:26.32troyb1hey Qwell
23:26.44x86well how do i put in the unique ID and date?
23:26.47x86is that possible?
23:26.57Qwellx86: channel variables
23:28.02*** join/#asterisk vr_mex (n=vr_mex@dsl-201-129-240-63.prod-infinitum.com.mx)
23:28.34vr_mexI need help on setting up shorewall i am behind a dmz router
23:28.42Qwellvr_mex: #shorewall
23:29.03x86Qwell: right, which ones? :)
23:29.09x86Qwell: is there a list somewhere?
23:29.10vr_mexbut I am setting it up for asterisk@home
23:29.25Qwellvr_mex: #asteriskathome
23:29.26troyb1is there any reason to use asterisk@home instead of asterisk?
23:29.39vr_mexthanks
23:29.42Qwelltroyb1: because you're too lazy and/or stupid to configure an OS
23:29.48*** part/#asterisk vr_mex (n=vr_mex@dsl-201-129-240-63.prod-infinitum.com.mx)
23:29.49QwellThat's just my experience anyhow
23:29.59troyb1well im happy to let you know my installs are 'regular' asterisk :P
23:37.07brookshireO RLY?
23:37.16troyb1@_@ rlly :O
23:37.40brookshireSRLY?
23:37.47troyb1nope.. im lying.
23:38.18brookshireorlyowl.com :(
23:38.23brookshirei'm addicted
23:38.27Heimidalthis is torturous
23:47.46*** join/#asterisk jebmpls (n=brandon@rrcs-67-53-20-211.west.biz.rr.com)
23:48.26Strom_Cman
23:48.32Strom_Cthe channel is so dead today
23:50.28*** join/#asterisk Ad-Hoc (n=Nimbus@ppp9-adsl-129.ath.forthnet.gr)
23:52.49brookshiremyspace is slow too
23:52.52brookshire:(
23:53.06*** join/#asterisk Ad-Hoc (n=Nimbus@ppp9-adsl-129.ath.forthnet.gr)
23:54.45tsumebrookshire: no shit sherlock ;)
23:57.31Heimidalugh.. tftp timeout over and over again
23:58.09tsumeftp > tftp for voip work
23:58.22Strom_Ctsume, the cisco phones require tftp
23:58.35tsumeStrom_C: how sad.
23:58.46HeimidalI don't understand what is going wrong
23:59.07tsumeyou mean.. cisco doesn't make perfect products *gasp* :)
23:59.48Strom_CHeimidal, is there anything between the phones and the tftp server?

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