IRC log for #asterisk on 20071117

00:00.29MatBoyare there companies providing area based telphony using Asterisk ?
00:00.46Strom_M"area based telephony:?
00:00.48BBHossarea based?
00:00.56Corydon76-digYou mean CLECs?
00:00.56[TK]D-Fender.... nvm :)
00:01.01Strom_Mcatsex?
00:01.13[TK]D-Fenderasciipr0n
00:01.51MatBoyStrom_M, no that customers will still have their normal telephone number that they used to have at home
00:02.00Corydon76-digI know at least one CLEC is using Asterisk to route their customers' long distance calls... considering I wrote their routing logic
00:02.10BBHossyou mean porting?
00:02.21Strom_MMatBoy: so what you're asking, really, is whether there are ITSPs that support number portability
00:02.30Strom_Mand yes, there are plenty
00:03.02Corydon76-digSo yes, there's a traditional CLEC using Asterisk in their POTS service plan
00:03.10MatBoyStrom_M, no the real question is... how stable it is comparing to PortaOne for an example. I think it's very stable, but also good on load ?
00:03.13Corydon76-digat least one
00:03.36Corydon76-digPortaOne is a billing platform
00:04.14Corydon76-digAFAIK, they don't even have an interface to the PSTN
00:04.37MatBoyCorydon76-dig, no, it's a full platform, indeed with billing, that supports Cisco
00:04.51MatBoyCorydon76-dig, Portaone has sip-servers
00:05.08Corydon76-digMatBoy: in other words, they're using a 3rd party vendor to support PSTN connections
00:05.08*** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca)
00:05.29Corydon76-digMatBoy: Asterisk has native interfaces to both sides
00:05.30MatBoyCorydon76-dig, true, and I was looking if Asterisk can do the same
00:06.04Corydon76-digMatBoy: You're comparing apples and oranges.  Asterisk is a telephony platform.  PortaOne is a billing platform.
00:06.16Corydon76-digYou certainly can interconnect the two
00:07.04MatBoyyeah, OK, let's say can I build with Asterisk, also on billing, what I can do with Portaone ?
00:07.21Corydon76-digI dunno, ask their sales dept
00:07.43Corydon76-digI'm sure they can set you up with some very expensive equipment
00:07.47BBHossMatBoy: you ought to rephrase that question
00:08.27Corydon76-digYou can build a billing platform on top of Asterisk, but Asterisk does not have a native billing app
00:08.35MatBoyCorydon76-dig, PortaOne costs about 70K without HW
00:08.41MatBoyfull SMP license
00:08.53Corydon76-digAsterisk is free without HW
00:09.08MatBoyCorydon76-dig, indeed, but you need billing for calling :)
00:09.09BBHossand the hardware is cheap
00:09.15Corydon76-dig$70k is more than I make in a year
00:09.16MatBoyyeah indeed
00:09.27BBHossyou can use a2billing, although ive not used it personally
00:09.59MatBoyNice to investigate this, I have to check in how far I can clutser Asterisk :)
00:10.03MatBoywould be awsome to do
00:10.13BBHossand you have CDR, so really all it takes is a program that can parse the files
00:10.19BBHossor pull from mysql
00:10.37*** join/#asterisk PaulAviles (n=salinas9@dsl-7-36.cofs.net)
00:10.52PaulAvilesany cisco 79xx users?
00:10.53MatBoyBBHoss, indeed, that's all it takes
00:11.07BBHosslooks like a2billing has grown a lot
00:11.12BBHosshttp://trac.asterisk2billing.org/cgi-bin/trac.cgi
00:12.36MatBoyBBHoss, I was just looking at it, looks nice
00:13.29MatBoyBBHoss, I think you can start a good business with it
00:13.34BBHossyeah
00:13.56BBHossnot sure how clustering works exactly, havent ever needed that big of a system before
00:14.41MatBoyBBHoss, ok, but I know platforms that are doing at least 1 million minutes per month
00:14.41MatBoyso
00:14.51MatBoyclustering might be nice I think
00:15.05BBHosswhy do you want to cluster, for failover or for performance
00:15.39MatBoyBBHoss, both actually, but if a system can hold it... failover
00:16.32*** join/#asterisk anthm (n=anthm@mbe0736d0.tmodns.net)
00:16.32*** mode/#asterisk [+o anthm] by ChanServ
00:16.56MatBoyBBHoss, keep quiet, an op arrived :P
00:17.25BBHosslooks like there is something out there called biocluster, but then again i've never used it
00:17.46MatBoyBBHoss, but your other unused idea was also quite nice, so let me look :)
00:17.57BBHossheh
00:19.22MatBoyBBHoss, and again... seems to be nice
00:23.35BBHossMatBoy: a2billing has a demo you can try out
00:24.02MatBoyBBHoss, yep I was already in, looks perfect
00:29.55*** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca)
00:30.17PaulAvilesany cisco 79xx users?
00:32.25*** join/#asterisk nny (n=Scott@64.20.149.250.dyn-e-pool22.pool.hargray.net)
00:32.28*** join/#asterisk captiancrash (n=captianc@c-68-53-165-155.hsd1.ky.comcast.net)
00:32.59*** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net)
00:33.12nnyso.. what if someone wanted to tear trixbox out of a system, and use the real mc coy. Is there a howto for excorsing the beast?
00:33.22[TK]D-Fendernny, format.
00:33.23nnyexorcising*
00:33.27nny[TK]D-Fender: lol yeah
00:33.34[TK]D-Fendernny, grab some sane distro and wipe it clean
00:33.35nny[TK]D-Fender: we are/were except one is in Panama
00:33.43nny[TK]D-Fender: yeah thats the plan
00:34.01nny[TK]D-Fender: just spent the better part of the day discovering who did what wrong
00:34.06nnylong story
00:34.09[TK]D-Fendernny, Or you could just stop freePBX from loading.  Thats the only really evil part.
00:34.25nnyoriginal plan was to wipe and re setup
00:34.37nnyyeah how doe sfreepbx load? not in /etc/init.d/
00:34.42nnyi noticed this is centos
00:35.17nnycause that would save me having to walk customer through installing the initial OS
00:37.00nnyand kill it dead
00:39.23*** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp)
00:39.55nnyhttp://www.freepbx.org/2006/09/28/un-trixbox-your-trixbox/
00:58.48*** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com)
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01:02.19*** part/#asterisk nny (n=Scott@64.20.149.250.dyn-e-pool22.pool.hargray.net)
01:02.20BBHossnny: i read somewhere i guide to changing an OS wholly over ssh
01:03.30*** join/#asterisk ManxPower (n=manxpowe@209.16.72.135)
01:08.37*** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net)
01:08.53*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
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01:09.19*** mode/#asterisk [+o mog] by ChanServ
01:13.38*** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net)
01:16.57*** join/#asterisk cybrside (n=cybrside@ppp-70-253-88-208.dsl.austtx.swbell.net)
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01:24.14*** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net)
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01:29.19*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)
01:37.28MackesWhats that?
01:39.00BBHosswhat?
01:39.01MatBoyBBHoss, ow ?
01:39.06MatBoywhere di you read it
01:39.16BBHossoh
01:39.21BBHosslemme find it
01:39.28MatBoyBBHoss, not a VM ?
01:39.47BBHossit was like for dedicated hosting providers that dont let you use ubuntu, only debian and such, hang on
01:39.50BBHossbut yes, no vm
01:40.04MatBoyow kewl
01:40.06MatBoyhehe
01:40.24MatBoybut what if they don't provide you ssh ? make a good hack script than :P
01:41.01BBHossyeah you could do it through perl, but you really need root access
01:41.34BBHosshttp://www.goudkov.com/public/articles/changing_distro.jsp
01:42.18MatBoyBBHoss, ah as I expected... you actually "move" it
01:42.57BBHossit would have solved his problem though
01:43.11MatBoykewl
01:43.19MatBoylet us try it in a vm :D
01:43.36BBHossOR you could install it on your own box or a vm, then make a dd image, then apply that dd image to the server
01:44.33MatBoyyep indeed
01:44.38MatBoyok, now I really need to sleep
01:44.43MatBoy3 am here almost
01:45.46MatBoysleep well
01:45.49MatBoyBBHoss, thanks !
01:45.52MatBoyfor everything
01:45.59MatBoybbl
01:46.00BBHossnp, good luck
01:46.08BBHossreport back on that clustering stuff
01:46.19*** join/#asterisk ming_zym (n=ming_zym@124.14.234.227)
01:46.19MatBoyyep will be done next year I think
01:46.25BBHossheh
01:46.30MatBoymaybe next month
01:46.33MatBoydunno yet
01:58.45*** join/#asterisk jero (n=jerome@modemcable169.212-70-69.mc.videotron.ca)
01:59.52jerohi
02:01.24BBHosssup
02:02.01jeroi'm looking for a way to trap a phone registration in chan_sip.. To run a program at the time a phone registers
02:02.05ManxPowerIf clustering was easy there would be many people doing it.
02:02.22ManxPowerjero: be prepared to do extensive coding in chan_sip.c
02:02.37*** join/#asterisk PepOSX (n=pepOSX@190.72.149.231)
02:02.37jeroManxPower: extensive ?
02:02.40mostyis there a way to forward a call between two asterisk servers that will preserve channel variables?
02:02.51ManxPowerjero: extensive = much = many
02:03.09mostyie so channel variables on the first server are available on the second server?
02:03.12jeroManxPower: you think id better parse logs in realtime ?
02:03.55MackesI dont think so.
02:05.07jeroi maybe should use SER ? i'm implementing a distributed architecture where a wireless phone can register in multiple offices on the same SSID / using the same credentials .. and have its calls re-routed to the correct site.
02:05.35BBHossoh shit! here we go again!!!
02:06.09MackesJero, I might be able to help... what are you using... what is your plan?
02:06.11BBHossjero: are you trying to do this for a single project?
02:07.57jeroi'm using nokia cellular+wifi phones.. + anything required. the only missing part is catching register. if the feature is not in chan_sip, we'll either add it, or use SER if more appropriated
02:08.15BBHossbecause with SIP-DECT, you can carry a handset across the world, and have it ring the same extension
02:08.27BBHossbut since your using cell, that could be a problem
02:08.29MackesIf you are interested, I would be happy to tell you about our setup
02:08.49jeroBBHoss: its a sip + cell phone. it does both
02:09.20jeroMackes: of course, any info will be helpful
02:09.51*** join/#asterisk GuyOCanada (i=GuyOCana@75.155.220.205)
02:09.54GuyOCanadaHello
02:09.59BBHosssup
02:10.01MackesWe have one Asterisk server that handles all calls across 21 locations
02:10.07Mackes31 locations
02:10.17[TK]D-FenderBBHoss, ;)
02:10.23MackesThey are all tied together with T1's
02:10.25BBHossheh
02:10.31jerooh okay
02:10.33Mackeswith me so far
02:10.35jerobrb
02:10.46GuyOCanadais there a rpm package for asterisk-1.4.14
02:10.54Mackesdo you have something similar?
02:11.00BBHossGuyOCanada: its best to build from source with *
02:11.16BBHossthere may be something in yum, but its probably old as dirt
02:11.28MackesJero?
02:11.33MackesOh ok
02:11.35*** join/#asterisk mihinomenest (n=argh@66.255.220.17)
02:11.55BBHossGuyOCanada: http://www.voip-info.org/tiki-index.php?page=Asterisk%20RPM may help
02:11.55*** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net)
02:12.18GuyOCanadaBBHoss: I want to install asterisk asterisk-1.4.14 on one of my servers which will only handle voip calls no hardware do i still need to install the packages (libpri zaptel etc.) or would it be ok if i just install asterisk-1.4.14 from source?
02:12.34BBHossGuyOCanada: depends on what you want to do with asterisk
02:12.49BBHossGuyOCanada: some * features require the zaptel dummy module (ztdummy)
02:13.18GuyOCanadahow do you compile the ztdummy module?
02:13.20BBHossMeetMe is the biggest one, i think there are a few more
02:13.26BBHosscompile zaptel
02:14.18bkw_no comment
02:14.26[TK]D-FenderGuyOCanada, first go download THE BOOK.  It will tell you how to do all of this.
02:14.27[TK]D-Fender~book
02:14.28jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
02:14.29[TK]D-Fender^^^^^^^^^^^^^^^
02:14.38bkw_I use that book for a monitor stand
02:14.40bkw_its all its good for
02:14.48bkw_poor rain forest
02:14.50BBHossshit, here we go
02:15.07MrTelephonetoiletpaper.pdf
02:15.12bkw_MOVING ON
02:15.14bkw_NEXT!!!
02:15.32MackesYOu dont like the book?
02:15.38[TK]D-Fenderbkw_, No, its much thicker than rev 1..... it is now a suitable tactical defense solution!
02:15.39MackesWho doesnt like that book
02:15.46bkw_my name is in the book
02:15.54Mackesits THE book
02:15.54BBHosssomone ought to go through voip-info.org take all the good info, clean it up, and put it in its own wiki
02:16.08bkw_Mackes: I don't.. it gives false hope to some
02:16.19Mackeshow is that?
02:16.19GuyOCanadaok another question
02:16.30GuyOCanadaanyone running asterisk on a box with plesk control pannel installed?
02:16.41JTBBHoss: you mean sort of like copyright infringement?
02:16.50*** join/#asterisk Qb3rt (n=eric@modemcable156.182-80-70.mc.videotron.ca)
02:17.00bkw_yah copyright infringement?
02:17.20BBHossdoubt it
02:17.20BBHossoh is it copyrighted?
02:17.23JT....
02:17.28JTyou've got to be joking
02:17.36JTit's a PUBLISHED O'REILY BOOK
02:17.37bkw_oh lord help us
02:17.39JTyes it has copyright
02:17.50BBHossno im talking about the voip-info wiki, not the book
02:17.56bkw_it too is copyrite
02:17.58bkw_er right
02:18.05bkw_lord help ya
02:18.17BBHossmany wikis use GNU
02:18.27JTvoip-info, commpartners owns the copyright to everyone's work there, apparently
02:18.39bkw_GNU is still a copyright
02:18.49MackesWell, I think its a great book.
02:18.51bkw_someone needs to get the clue stick out
02:18.58bkw_Mackes: I don't
02:19.00Qb3rtwhat is the minimum cpu and memory needed if i want to setup one server with 20 lines (3 talkings and the rest wainting with music on hold) All calls recorded... And a PRI on it??
02:19.05*** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net)
02:20.00MackesIs there a better book, I would love to read another? BTW, who has it given false hope, and why?
02:20.07[TK]D-FenderGuyOCanada, I've never heard of it in my years here, and I can't see any reference to an Asterisk module for it anywhere.
02:20.36[TK]D-FenderQb3rt, decent P4 + 1 gig should be fine.
02:20.41MackesPLesk is a web management pannel.... kinda like Webmin
02:20.43GuyOCanada[TK]D-Fender: plesk is a produfct of swsoft www.swsoft.com its a control panel for hosting environments
02:20.46MackesClosed Source
02:20.53JTyes i think we all know what plesk is
02:20.53BBHossHow does GNU FDL keep you from copying and modifying it, as long as you apply the license?
02:20.54[TK]D-FenderGuyOCanada, I know, I'm on their site
02:21.13JTBBHoss: it's still a goddamn type of copyright
02:21.23GuyOCanadabut it forks linux installations
02:21.31Qb3rtAlready got that and its not going well... not enought memory and the cpu is really loaded
02:21.32BBHossok, but you follow the license, you can use it, correct?
02:22.03JTBBHoss: everything on voip-info is property of commpartners
02:22.14BBHossyeah i was speaking in a general sense
02:22.15JTthere is no gnu or anything
02:22.19GuyOCanadafor some reason freepbx or other web gui's do not work on a plesk installed server
02:22.21[TK]D-FenderQb3rt, well you haven't mentioned anything like what card, what codecs, etc.
02:22.51[TK]D-FenderGuyOCanada, Ask THEM.  GUI's are not supported here.
02:23.03ManxPowerhttp://www.voip-info.org/terms_of_service.html
02:23.07MackesCheck the port number it uses
02:23.19BBHossManxPower: yes i saw that, too bad
02:23.20GuyOCanada[TK]D-Fender: i am not asking for support of a GUI i just asked if someone was using it
02:23.21*** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60)
02:23.21Mackesit might be attempting to use a port that is in use
02:23.32Qb3rti dont have these details right now... but ill get back here when i have them... thanks for you help :D
02:23.39ManxPowerBBHoss: if you don't like it, don't use it.
02:23.51BBHossManxPower: what are you talking about
02:23.55ManxPowerGuyOCanada: nobody here uses GUIs
02:23.58*** join/#asterisk CrashHD (n=crashhd@67-107-9-130.starstream.net)
02:24.04BBHossManxPower: i never said i didnt like it
02:24.13MackesI think Webmin is very helpful
02:24.16[TK]D-FenderGuyOCanada, Well I've answered that.  There has been no mention of Plesk here.
02:24.25JThelpful if you can't use ssh
02:24.34BBHossMackes: webmin has a tendency to trash installations
02:24.36MackesYep
02:24.40ManxPowerSorry, I assumed that when you said "ManxPower: yes i saw that, too bad" you were expressing a dislike for the policy.
02:24.44MackesReally? How so?
02:24.52GuyOCanadaManxPower: sorry
02:25.23BBHossManxPower: no, it just means that i can't  use it :)
02:25.23MackesI guess if you have made custom changes and Webmin rewrites your configs
02:25.39BBHossManxPower: however it is odd that a user-created page somehow becomes their copyright
02:25.57BBHosscompared to other wikis ive seen
02:25.59Corydon76-digI suspect it means something different
02:26.20Corydon76-digThey don't want you to put up pages that are specific to any one company
02:26.35Corydon76-digrather, general information that is useful to everybody
02:26.43BBHossMackes: well if it gets the version wrong and writes a config file wrong, or changes it someway
02:27.06Corydon76-digYes, they have to word it strongly to ensure that they can go after people who abuse the service
02:27.17MackesYeah.. That would be a bumper.
02:27.34BBHossCorydon76-dig: so you think they just dont want people profiting off of it?
02:27.38MackesHowever I have installed it on Serveral Distros without an issue.
02:28.04MackesHas anyone worked with openVZ
02:28.54Corydon76-digBBHoss: I think they just don't want to be an involuntary host to information that promotes a single company
02:29.08Corydon76-digBBHoss: that's always a danger when running a Wiki
02:29.22BBHossyeah, its probably best for me to email them
02:29.39MackesWhy the discussion about the Wiki... Do you have an issue with them?
02:29.51JTas if anyone ever gets a reply from commpartners
02:29.56JTMackes: clearly there are issues
02:30.07BBHosswell, the design is ugly
02:30.17Corydon76-digYou can very easily set up multiple disparate "sites" on a single Wiki... It's all in how the pages are linked together
02:30.18BBHossthe pages don't work
02:30.24BBHossthe examples/info are bad
02:30.30rob0And its mother dresses it funny.
02:30.38BBHossrob0: indeed
02:30.43JTthe info being bad is not a fault of the sponsor
02:31.07BBHossyeah i know, but if im going to clean up all the info, id rather have all the good docs in one place
02:31.22MackesHmmm... Wow, tonight we have knocked the best book on Asterisk and the most detailed Wiki on Asterisk....
02:31.34Mackesfantastix
02:31.42JTMackes: yes, such is the real world
02:31.58Corydon76-digWhat's wrong with TFOT2?
02:31.59BBHossi didnt trash the book, but a lot of the stuff on the wiki sucks
02:32.01JTMackes: and it is a VERY WELL KNOWN fact that the wiki has tonnes of incorrect or outdate info on it
02:32.05BBHossand its very unorganized
02:32.28JTunorganised isn't a word
02:33.55MrTelephonewhat the hell is TFOT2
02:34.00BBHossthe book
02:34.02jeroMackes: back
02:34.05[TK]D-FenderJT : Apparently it IS... http://dictionary.reference.com/browse/unorganised
02:34.06JTthe second edition of the book
02:34.24MrTelephonei want to get back into online gaming
02:34.31MackesI am sooo thankful to the community at large. Asterisk and the folks who contribute to Wikis help me do my job, which in turn feeds my family--- and ---- and---- its free. Microsofts, Cisco, Alcatel, Dell, HP...... All of there Tech support Docs sections are difficult. Why knock the one that is a direct result of our community... especially here!
02:35.04JT[TK]D-Fender: only americans would add such an incorrect word to the dictionary ;)
02:35.04MrTelephoneasterisk is a good project but its becoming a little bloated
02:35.05BBHosswanting to do something better isnt knocking it
02:35.30BBHossJT: what would u use?
02:35.42JTMackes: things will never improve if everyone pretends everything is perfect
02:35.48JTBBHoss: the word is "disorganised"
02:36.06BBHossJT: ok, that word is not used often in america
02:36.25JTi guess that's because they speak american, and not english
02:36.34BBHossJT: yep
02:36.42JTit's a horrible abomination :P
02:37.34MackesBut bitching about it doesn't help? Lots of Wikis are available and they are all a mess. Why, because they are community driven. Does anyone have a BETTER site, that is free, and current?
02:37.53BBHossMackes: wikipedia is the way a wiki should be
02:38.04JTeven wikipedia has issues :P
02:38.11BBHossnoone is perfect
02:38.22MrTelephoneJT is perfect
02:38.53Mackesman
02:39.03JTthe asterisk book is pretty good
02:39.14MackesI think so as well
02:39.15JTalthough i don't like how it hypes stuff up
02:39.29JTespecially the whole asterisk being the future thing
02:39.30*** join/#asterisk [N00B] (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net)
02:39.36JTclearly very biased in that respect ;)
02:39.42MackesI have all of the Cisco and Alcatel/Lucent books on Voip
02:39.55Mackesand they are very difficult to read and ue
02:39.56Mackesuse
02:40.32MackesThe * book is direct, and a great reference
02:40.38asdxi agree
02:40.56MrTelephoneall the problems I have in asterisk are all packet loss issues
02:41.00[TK]D-FenderJT : If they called the "suggestions of physics" nobody would be following them and we'd all be flying around!  Its called MARKETING!
02:41.04MrTelephoneand its not even asterisks' fault
02:41.06*** join/#asterisk iamamoron (n=t@202.137.121.84)
02:41.11iamamoronhi there
02:41.31JTasterisk is lacking in a lot of areas, but is fine in others
02:41.34iamamoroni have an existing pabx panasonic kx-td816
02:41.39JTit's certainly no be all and and all
02:41.50iamamoronand there is an existing telephone line connected to it
02:42.17iamamoronwhat i want is the asterisk all cater all incoming calls and pass it to the kx-td816
02:42.22iamamoronwould it be possible?
02:42.30BBHossiamamoron: yes
02:42.32iamamoronby using digium tmd400p
02:42.43BBHossiamamoron: just get 1 fxo and 1 fxs
02:43.04BBHossiamamoron: but you will be limited in what you can do with asterisk
02:43.42Mackesis the  kx-td816 use digital handsets?
02:44.01Mackesif so, and FXS/FXO isnt going to work
02:44.03[TK]D-Fenderiamamoron, What exactly do you want * to do for you?
02:44.22Mackesyou might need to create a tie line using a PRI
02:44.38BBHossif you just want to play a message then pass the call to the panasonic, then it will do it
02:44.50JTMackes: he wants to do pbx intercept, it is irrelevant if the pbx has digital handsets or not
02:44.58ManxPowerMackes: you are assuming that the Panasonic supports T-1/PRI
02:45.13iamamoronfrom pstn asterisk will get all calls
02:45.18iamamoronand outbound calls
02:45.26MackesYep
02:45.37iamamoronnow an IVR is in asterisk
02:45.49JTMackes: there is no reason it can't be done over FXO/FXS ports
02:45.53BBHossyeah its possible, but it aint gonna be easy
02:45.57JTnot as nice as digital obviously
02:46.01MackesOK
02:46.03MackesSo
02:46.17iamamoronsay if user at pstn press 1 it will go to extension 222 at panasonicj kx-td816
02:46.18MackesCall comes in, Asterisk Grabs it, says a few things
02:46.27JTi do pbx intercept for BRI lines
02:46.29*** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au)
02:46.34Mackesand then passes it to the  kx-td816
02:46.38JTexcept i changed the telco lines to PRI after a bit
02:46.49iamamoronany ideas?
02:46.57MackesIt will do that if you only have POTS with FXO/FXS
02:47.06MackesHowever, that is all you will be able to do
02:47.11iamamoronyes i have
02:47.21iamamoronwhat should i do?
02:47.32MackesRemove the  kx-td816
02:47.34iamamoroni have a x100p here
02:47.36JTyou can make asterisk give voip calling capabilities to the panasonic
02:47.39BBHosscan you plug a fax machine into that PBX?
02:47.43JTor recording capabilities
02:47.45BBHosshehehehe
02:47.46MackesHow many extentions do you need?
02:48.07iamamoronyou said all i need is 1 fxo and 1 fxs
02:48.14iamamoroni have a x100p here
02:48.20iamamoronwhat should i do now
02:48.21MackesThat will give you 1 channel
02:48.25Mackesjust one
02:48.25JTthe x100p is junk
02:48.30Mackesone caller at a time
02:48.30JTand you need an fxs port
02:48.31BBHossiamamoron: get a better card first
02:48.38Mackesis that what you want
02:48.38JTperhaps get a linksys SPA-3102
02:48.48iamamoronfor trial only
02:48.53MackesI mean, why would you have a  kx-td816
02:48.55iamamoroni know x100p sucks
02:49.00Mackesif you only handled one call
02:49.01JTyou still need an fxs port for that trial, iamamoron
02:49.03iamamoronbut i want to test it first before i buy
02:49.07Mackesahhh
02:49.10BBHossiamamoron: avoid the x100p like the plague
02:49.18MackesAre you using POTS on your  kx-td816
02:49.26iamamoronkx-td816 is obsolete
02:49.31GuyOCanadalinksys SPA-3102 can you use that remotely?
02:49.33MrTelephonei bought a linksys 8-port sipura
02:49.34iamamoronMackes; yes
02:49.36*** join/#asterisk BiG^DoG (n=BiG^DoG@c-67-162-233-20.hsd1.de.comcast.net)
02:49.38iamamoronPOTS
02:49.45JTGuyOCanada: sure
02:49.50MrTelephoneavoid anything analogue if you can
02:50.07iamamoronno i cant coz i need to interop
02:50.14MackesThe  kx-td816 has BRI interface
02:50.16iamamoronMackes?
02:50.19Mackesyep
02:50.26BiG^DoGI'm going crazy... Is there any way to get call waiting to work with my SPA3102 and asterisk?
02:50.26iamamoroncan i simulate using x100p?
02:50.33Mackesok
02:50.40JTBiG^DoG: what do you mean call waiting?
02:50.41BBHossiamamoron: you would be better off trashing the kx-td816 and getting a new system with polycom or another brand ip phone
02:50.44GuyOCanadaso I can have spa-3102 at my home and connect it using IAX2 or SIP to my asterisk box at work and answer calls from another sip location or call using my home phone from another sip location?
02:50.48JTiamamoron: NO, you need an FXO AND and FXS port
02:50.50BBHossthen get a tdm400p or tdm800p to do POTS
02:50.52MackesHow many stations does your  kx-td816 support
02:51.08iamamoronJT: yes i have
02:51.12obnauticusBBHoss, do you know how to setup a Cisco 7960?
02:51.15iamamoronmy x100P has fxo and fxs
02:51.19BiG^DoGJT: I have call waiting on my PSTN line and i want to be able to pick up that second call
02:51.22JTTelco POTS <---> Asterisk <---> Panasonic
02:51.26JTiamamoron: rubbish
02:51.30iamamoronyes
02:51.31JTiamamoron: NONE have FXS
02:51.32BBHossobnauticus: sorry, never played with ci$co before
02:51.32JTEVER
02:51.36iamamoronthats the setup i want
02:51.37Mackeshttp://www.prodcat.panasonic.com/shop/NewDesign/ModelTemplate.asp?ModelId=16638&show_all=false&product_exists=True&active=1&ModelNo=KX-TD816&CategoryId=
02:51.48JTiamamoron: let me repeat ths to be crystal clear
02:51.57JTiamamoron: the X100P does NOT HAVE AN FXS PORT
02:51.59rob0The other RJ11 is a passthrough port.
02:52.05MackesIt looks as if oour KX has 8 stations.
02:52.06JTit is physically IMPOSSIBLE
02:52.07iamamoronMackes: fxs --> panasonic dco?
02:52.20BBHossbut dont you set it up as fxsks is the zapata config :)
02:52.25BBHoss:}
02:52.29MackesI would replace them all with Asterisk and new phones... its the best way
02:52.34iamamoronJT: mine has
02:52.35JTthe X100P has NO 90VAC @ 20Hz ringing generator, NO -48VDC line battery feeding
02:52.37BBHossMackes: indeed
02:52.37iamamoronfxs
02:52.42JTiamamoron: absolute garbage
02:52.51JTiamamoron: there is no way it is an X100P
02:52.57Mackesfxs --> panasonic dco? ?????
02:52.58JTif it has FXS
02:53.05BBHossthe panasonic system will just cripple your system
02:53.12rob0Mine has 2 RJ11 jacks as well. The second one passes through to another phone/device.
02:53.27MackesReally, I know you want to test first, but BBHoss is right, the Panasonic will just hold you back
02:53.28iamamoroncdo is where i connect my pstn going to panasonic
02:53.38iamamoronsome kind of fxs
02:53.46JTiamamoron: why won't you listen to us?
02:53.46BBHosssweet, i'm right!
02:53.58iamamoronJT:
02:53.59iamamoronok
02:54.03iamamoroni will listen to you
02:54.06JTiamamoron: the X100P DOES NOT, absolutely NOT have an FXS port EVER
02:54.08MackesHere is the specs on the Pan:
02:54.08iamamoronwhat should be done
02:54.09rob0Who's going to pick up on the straight line first?
02:54.12Mackeshttp://www.prodcat.panasonic.com/shop/NewDesign/ModelTemplate.asp?ModelId=16638&show_all=false&product_exists=True&active=1&ModelNo=KX-TD816&CategoryId=
02:54.20rob0Get something with FXS. :)
02:54.29BBHossthe CO ports on the panasonic system are FXS devices, the ports at your phone company are FXO ports
02:54.32JTiamamoron: the cheapest option is to buy an SPA-3102
02:54.39BBHossits really damn confusing
02:54.43*** join/#asterisk ltd (n=z@pat.transact.net.au)
02:54.47iamamoronand then
02:54.52MackesYep. Get a SPA 3102
02:54.55iamamoronhow can i integrate that to my panasonic?
02:55.08BBHossiamamoron: trash the panoshit
02:55.12iamamoroni want to preserve all the telephone
02:55.18BBHossiamamoron: or pass it off on ebay
02:55.21JTiamamoron: the 3102's FXS ports plugs into the Panasonic's FXO port. the 3102's FXO port connect to the telco
02:55.34BBHossiamamoron: how many stations/phones are you  using?
02:55.43iamamoronnow 16
02:55.57iamamoronquite many i cant just throw it away
02:55.59Mackessorry man
02:56.05MackesI understand
02:56.21MackesIs this your companies system?
02:56.29iamamoronyeap
02:56.33BBHoss$1400 for new IP hones (IP 320 Polycom)
02:56.33MackesOk
02:56.37MackesWell,
02:56.44ManxPoweriamamoron: Generally if you can't replace your PBX, don't bother trying to try tieing Asterisk into the existing PBX
02:56.49MackesBuild yourself an Asterisk System on the side.
02:56.49iamamoroni am planning to have asterisk to be my IVR
02:56.58Mackesdont tie it in to your pan
02:57.03JTManxPower: i disagree, there are times when PBX intercept is useful
02:57.05ManxPoweriamamoron: you'll still have issues when you use analog
02:57.16JTiamamoron: forget about an IVR on analogye
02:57.19iamamoronJT?
02:57.20BBHossJT: this is a rather small system though
02:57.22ManxPowerJT:  only if you have PRI or E&M./Wink 8-)
02:57.23MackesWhat will the Intercept do for you?
02:57.46JTiamamoron: analogue has terrible call status signalling
02:57.46Mackesit only will have ONE FXS to route to
02:57.55JTiamamoron: IVRs need good signalling
02:58.06BBHossanybody ever buy from ip phone warehous? they have the 320s for $83.99
02:58.12MackesI have
02:58.15MackesVery goo
02:58.15JTiamamoron: you might be able to get away with polarity reversal from the telco
02:58.18Mackesd place
02:58.23BBHossdamn, they are cheap
02:58.37MackesJust got a Snom 370 and an hitachi Wifi from them
02:58.57iamamoronso if i buy SPA 3102?
02:59.24BBHossiamamoron: please dont punish us by trying to do what you're trying to do
02:59.37iamamoronJT?
02:59.41iamamoroni am talking to JT
02:59.58BBHossthis is a chat room
03:00.15obnauticusDoes anyone here have knowledge of how to update firmware on a Cisco 79xx (preferably 7960) IP Phone?
03:00.19ManxPoweriamamoron: expect to have lines stuck offhook if you insist on using analog
03:00.44iamamoronbecause i am PSTN here
03:00.53iamamoronBBHoss: dont use us
03:00.56BBHossiamamoron: JT even said ivr was a bad idea
03:00.58iamamoronyes it is a chat room
03:01.05iamamoronbut i am talking to JT
03:01.13ManxPowerobnauticus: that would be on the Cisco web site.
03:01.14iamamoronyou are not JT
03:01.21BBHossmoron :)
03:02.06JTiamamoron: then you would link the 3102 to asterisk via sip
03:02.29JTiamamoron: you need polarity reversal on disconnect from your telco to run an ivr
03:02.31obnauticusManxPower, I understand thatr but they have a bug in their newest SIP firmware apperentally.
03:02.42BBHossobnauticus: as always :)
03:02.53*** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60)
03:02.53ManxPowerobnauticus: The release notes and firmware upgrade instrucitons have a bug?
03:02.53obnauticusyeh
03:02.59obnauticuslol
03:03.10obnauticushttp://www.voip-info.org/wiki/index.php?page=Firmware+issues+on+7940+-+7960 <-- that is what I'm reading
03:03.15obnauticusand it looks like I'm getting that issue
03:03.17ManxPoweryou asked "how to update", well "how to update" would be answered "on cisco web site"
03:03.18obnauticusor something like that.
03:03.28obnauticusMy bad :/
03:04.16*** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au)
03:04.26ManxPowerobnauticus: here you go http://www.cisco.com/en/US/products/hw/phones/ps379/prod_installation_guides_list.html
03:04.34BiG^DoGso no help on my spa3102 call waiting problem? :(
03:05.12JTBiG^DoG: best to forget about call waiting
03:05.41ManxPowerBiG^DoG: What is your "call waiting problem"?
03:05.53[TK]D-FenderManxPower, he wants to flash the FXO port
03:05.59BiG^DoGyes
03:06.01ManxPowerobnauticus: Actually, a link to the correct upgrade/release notes will be included with the firmware you have.
03:06.09BiG^DoGor find a suitable alternative
03:06.11BBHossiamamoron: use your asterisk box to recieve calls from the telephone company
03:06.14ManxPowerBiG^DoG: It is not possible to use the telco call waiting with SIPura products
03:06.21[TK]D-FenderBiG^DoG, Which I don't know if *'s SIP has a way of passing, nor does an app exist for that channel
03:06.25BiG^DoGworkarounds/alternatives?
03:06.34obnauticusManxPower, no it's not
03:06.42[TK]D-FenderBiG^DoG, you CAN do this on a zaptel interface.
03:06.45obnauticusIt's the SIP firmware :/ open to the public.
03:06.45ManxPowerBiG^DoG: If there were workarounds/anternatives then it would not be "impossible"
03:07.14ManxPowerobnauticus: I have no idea what you mean.  The firmware should include release notes,.  Are you saying that the firmware does NOT include release notes?
03:07.44obnauticusno i'll find the,m
03:07.45BiG^DoGcould I ditch the spa3102 and get another ATA?
03:07.56MackesCisco
03:07.58JTBiG^DoG: no, a physical card
03:08.02MackesWhat do you want to know
03:08.04ManxPowerBiG^DoG: I am not aware of ANY ATA that supports what you want to do
03:08.10JTBiG^DoG: can't signa; hookflash over SIP
03:08.24ManxPowerobnauticus: that is the "m"
03:08.37ManxPowerobnauticus: odd.
03:08.59[TK]D-FenderBiG^DoG, * cann't tell any SIP device to "flash".  Your only current option is to get a Zaptel interface.
03:09.48Mackesobnauticus, did you find what you wanted?
03:10.00MrTelephoneis that the guy who ignored me yesterday
03:10.02MrTelephonefruitcake
03:10.16BiG^DoGI thought the most recent sipura firmware allowed a double hook flash to pass through to pstn
03:10.18obnauticusStill looking this is interesting :/
03:10.23JTwhich fruitcake?
03:10.35MrTelephoneobnauticus
03:10.42MrTelephonechristmas fruitcake :P
03:10.43[TK]D-FenderBiG^DoG, * can't SEND one in the first place.
03:10.43JTBiG^DoG: maybe from the fxs port
03:10.56BBHossi hate fruitcakes
03:11.15BiG^DoGcould it be done from the fxs port?
03:11.24MackesHere you go
03:11.27[TK]D-FenderJT : Shouldn't.  The FXS & FXO ports are INDEPENDANT.  So I guess that'd only apply if those 2 ports are bridged and KNOW IT.
03:11.27Mackesftp://ftp.cisco.com/pub/voice/ip-phone/sip-7960/
03:11.41Mackesand
03:11.44Mackeshttp://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP
03:11.57obnauticushttp://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/sip/relnote/phnrn82s.htm#wp1149999
03:12.16JTMackes: how about the firmware for a Cisco 12SP+? :)
03:12.19GuyOCanadawhat cisco phone would you recommend me
03:12.24obnauticusnone.
03:12.26BBHossJT: heh
03:12.26obnauticusPolycom
03:12.37MackesDo you have  a Cisco account.
03:12.42JTno
03:12.44obnauticusNo.
03:12.50MackesBecause if not, you are going to need my link
03:12.52ManxPowerBiG^DoG: no matter how many times you ask that question, the answer will be the same.
03:13.01JTthe 12SP+ fills a niche that no other ip phones does
03:13.02Mackesftp://ftp.cisco.com/pub/voice/ip-phone/sip-7960/
03:13.03MrTelephonemackes do you have a cisco account?
03:13.05BiG^DoGit was a different question
03:13.09Mackesyep
03:13.15MackesBut you dont need it
03:13.17Mackesftp://ftp.cisco.com/pub/voice/ip-phone/sip-7960/
03:13.22GuyOCanadaok which Polycom do you recommend
03:13.28JTthe 12SP+ are indestructable
03:13.28MackesThat is the magic man
03:13.37MrTelephonecan you download me the newest ubr7200 image :P
03:13.43JTwe deploy a dozen of them in the middle of the bush every year
03:13.53Mackes12SP+ ??? What is that?
03:14.09JTbeige coloured ip phone
03:14.13JTdoes H.323 or SCCP
03:14.22Mackesohhh. No SIP?
03:14.25JTno
03:14.45ManxPowerMrTelephone: Sure!  You want a copy of MS Office and CD Key as well?
03:14.45BBHossGuyOCanada: depends on application
03:14.55MackesI am looking for an indestructible Sip Phone
03:14.58ManxPowerI can give you the latest pirated music too!
03:14.58BBHossGuyOCanada: the IP330s work good for general purpose
03:15.17MackesThe Cisco SIP Image is availible to the public right now
03:15.22JTindestructable H.323 phone, good enough ;)
03:15.27GuyOCanadaI have only one place I need a real ip phone for the use with asterisk
03:15.34obnauticusManxPower, it's weird because it as the ip 10.0.0.95, then when it continues on to try and get the CM list it changes to 10.0.1.71
03:15.36JTMackes: can you get firmware or documentation for the 12SP+ series?
03:15.40obnauticusAnd it starts timing out
03:15.41GuyOCanadaall my stuff is routed over voip and there are no hard phones everyplace is using softphones
03:15.54Mackesthey are Ciscos?
03:15.59obnauticusThen it says opening 10.0.0.110 and in my TFTP window it's getting a TIMEOUT waiting for ACK Blcok #1
03:16.10JTMackes: yes, i already mentioned that :)
03:16.25JTthey're pre 7000 series
03:17.17MackesThat is an Old Ass looking phone
03:17.35JTthey're actual strong, unlike the latest ip telephones
03:17.37MackesSorry, my login info is at the office
03:17.41Qwell12SP+ == relabeled Selsius
03:17.49Qwell(Selsius Phone - get it?)
03:18.02JTnever heard of selsius
03:18.06[TK]D-FenderQwell, freezes solid randomly :p
03:18.19Qwellthe 12SP+?  heh
03:18.25QwellI can lock mine up pretty easily..
03:18.37MackesYou have one?
03:18.41Qwellactually, the 30VIP is *easy* to lock up..  just try to set 30 speeddials
03:18.42JTi have one
03:18.47JTi haven't used it yet
03:18.56MackesHey Qwell.... how did you get Mod status
03:19.05MackesThat would be helpful in this group
03:19.08JTMackes: work for digium?
03:19.11QwellMackes: /whois me
03:19.23*** join/#asterisk wothinn (n=Allfathe@vs1.svartalfheim.net)
03:19.36MackesHaaa Cool
03:20.03MackesQwell, How do you feel about Cisco Sip, Polycom, and Snom and Aastra
03:20.09Qwellgoogle images has failed me
03:20.13Qwell~phones
03:20.13jbotextra, extra, read all about it, phones is http://bani.anime.net/phones/.  While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows:  Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else.  Do not consider Grandstream ...
03:20.27MackesWe have had a long disagrement about such things
03:20.35Mackesyep ,phones
03:22.08GuyOCanadaI have a question i know its not asterisk specific but when i run menuconfig to select what i want to install with asterisk on the codecs list it says speex and when i am on it it gives me XXX (Depends on speex(E)) what does that E mean?
03:24.51*** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net)
03:26.44Nuggetit means you need to install the speex libraries on your machine before you can compile that support in to asterisk.
03:27.08Nuggethttp://www.speex.org/downloads/
03:27.38GuyOCanadaNugget: I already installed it using yum install speex and libspeex
03:28.04Nuggetyou probably need speex-devel or something along those lines
03:28.45GuyOCanadaspeex-devel is also installed
03:28.55NuggetI dunno then
03:29.31Nuggethave you re-run ./configure since doing that?
03:29.43GuyOCanadayes
03:29.53GuyOCanadamake clean ; ./configure
03:30.27*** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com)
03:31.55*** join/#asterisk implicit (n=implicit@ip68-101-122-30.oc.oc.cox.net)
03:34.47*** join/#asterisk dkwiebe (n=chatzill@h66-112-187-16.mcsnet.ca)
03:35.03dkwiebegood evening.
03:35.50dkwiebeI was on here a while back looking for help setting up a couple of analog DID trunks.  I received help but I was unable to get it to work.  Anyways, I'm just following up.
03:36.08dkwiebeWe got it working by using a couple of Audiocodes Analog DID Trunk gateways.
03:36.17dkwiebeThey were priced reasonably and work well.
03:36.36*** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk)
03:38.15*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
03:42.24[TK]D-Fenderdkwiebe, How is it signalled?  Flash? Immediate DTMF?
03:43.03*** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com)
03:43.25dkwiebeThey called it a "wink" which I believe is the same as a flash.
03:44.12jameswf-homewink is often a revpol
03:44.42*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
03:45.12dkwiebeSomebody here, was it you [TK]D-Fender? suggested some dialplan code.  I tried it but Telus was not very helpful and they insisted that our side was marking the lines as "busy"
03:45.14[TK]D-Fenderdkwiebe, So you answer, issue the wink, they pass fixed length dtmf and then hand off the channel?
03:45.18coppicea wink is in the opposite direction from a flash, so one is a pulse of reversal and the other a pulse of open loop, the intended effect is similar
03:45.54dkwiebethat is the theory.  They're acting exactly like an extension of our pbx except that they want a wink when they pick up.
03:46.27[TK]D-Fenderdkwiebe, nifty
03:46.41[TK]D-Fendercoppice, thanks for that little tidbit.  Something to remember.
03:46.48dkwiebecoppice:  Thanks, I'll remember that.
03:47.09dkwiebeDo you know if it's possible to get asterisk to "wink" from the dialplan?
03:47.38jerowink ?
03:47.47jameswf-homeno wink is signalling
03:47.59[TK]D-Fenderdkwiebe, not to send a signal to the Audiocodes to do that...
03:48.04jameswf-homethe hardware does it
03:48.37dkwiebeIf I thought we'd ever need it again I'd open a bounty. :-)
03:48.44dkwiebemaybe I should anyways...
03:49.09dkwiebethe audiocodes takes care of it all
03:49.16dkwiebeok.
03:49.36dkwiebeThey're actually pretty cool little boxes, trivial to configure and they just work.
03:49.57*** join/#asterisk izaak (n=izaak@modemcable132.248-130-66.mc.videotron.ca)
03:51.36*** part/#asterisk jero (n=jerome@modemcable169.212-70-69.mc.videotron.ca)
03:51.37[TK]D-Fenderdkwiebe, "trivial" is a term I can't say I've ever heard applied to AudioCodes configuration.  Cryptic, convoluted, obscene are more common :)
03:51.49dkwiebeI'm sorry
03:52.00dkwiebemy mistake.
03:52.06izaakAnyone wanna shout their recommendation for a fan-less system to drive a tdm400?  i'm thinking of the intel D201GLY2.  the wrap alix 1c also looks interesting, but i'm unsure if an amd lx 500mhz processor could handle it.
03:52.10dkwiebenot audiocodes.  They're brutal to configure
03:52.39dkwiebeI'm going to have to look up these boxes as my mind has gone blank.
03:53.05dkwiebelol, that's better.  "Multitech Analog DID Trunk Adapter"
03:53.22dkwiebe"trivial" isn't a term I'd use after spending the last 2 days fighting with them.
03:53.27dkwiebethem = audiocodes
03:53.39*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
03:54.04[TK]D-Fenderdkwiebe, For a second I was thinking you must be from Bizarro-World from whence they came ;)
03:54.25dkwiebelol, yeah I bet.
03:56.20MrTelephoneobnauticus has me on ignore from yesterday and I could help him
03:56.39MrTelephone<obnauticus> Okay, I have a cisco 7960 IP Phone, I am trying to upgrade the firmware to the SIP firmware, and now it's in some weird loop.
03:56.40[TK]D-FenderMrTelephone, some people you just can't reach!
03:56.43MrTelephonehaha
03:56.49MrTelephoneno i said something offensive to him yesterday
03:57.42MrTelephoneknocked up was a pretty good movie
04:00.15QwellMrTelephone: ripped it last night :p
04:00.20Qwellerm, yeah
04:00.24Qwellmade a backup copy
04:00.34Qwell...in case hollywood video ever needs it
04:01.56MrTelephonehahha
04:02.12MrTelephonehey i paid for a pack of dvds and some royalties went to hollywood
04:02.19MrTelephoneso basically i own a percentage of anything they make
04:03.06MrTelephoneqwell, that problem i was having with the sip clients had something to do with packet loss.. but it may happen again :(
04:08.17coppiceIts a sad day on IRC when at least one idiot doesn't add you to their ignore list or call you a retard
04:09.19MrTelephonecoppice your mean sometimes
04:09.39MrTelephone:(
04:09.44MrTelephonebut i didn't ignore you
04:09.58MrTelephonei havn't been ctcp flooded in 12 years
04:10.53MrTelephoneqwell is working on skinny?
04:10.59QwellLIES
04:11.13MrTelephoneget those wirelss voip phones working already will ya
04:11.13coppicehe's just mildly overweight
04:11.21[TK]D-FenderQwell, it's not just a river in Egypt! ;P
04:11.24Qwellmildly, hell
04:11.31QwellMrTelephone: they work just fine
04:11.37QwellMrTelephone: send me one, and I'll keep it updated
04:13.31*** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net)
04:13.37UserReg_CLhi, helpme please...
04:13.46UserReg_CL(alguien habla espaol?)
04:14.01coppice不是
04:15.59MrTelephonedo you know how to timestamp graphs made by rrdtool?
04:17.11UserReg_CLcall to extension fail for one context (number internal)
04:17.19GuyOCanadawhat does the module embedding options do?
04:18.07*** join/#asterisk hawky (n=geoff@c-71-231-188-226.hsd1.or.comcast.net)
04:18.36MrTelephonedo you guys get ftp bruteforce attacks alot?
04:18.44*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
04:19.25coppiceno, our FTPs get lured into a false sense of security by the Sirens
04:19.48asdxUserReg_CL: si, pero soy un newbie con asterisk
04:20.16UserReg_CLasdx: tengo un problemita... quiza puedas darme una mano...
04:20.23asdxUserReg_CL: ok
04:20.49UserReg_CLasdx: no puedo efectuar una llamada a una extension :(
04:20.54asdxs/newbie/principiante
04:22.30MrTelephonethe guy keeps trying to login as Administrator
04:22.56UserReg_CL~book
04:22.56jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
04:22.59asdxUserReg_CL: postea tus configuraciones en pastebin y copia/pega aca, estoy seguro que te van a poder ayudar los que tienen mas experiencia, yo soy muy nuevo con asterisk
04:23.00MrTelephoneNov 16 23:22:37 bonnie proftpd[4413]: localhost.localdomain (::ffff:88.191.14.100[::ffff:88.191.14.100]) - no such user 'Administrator'
04:23.05MrTelephoneover and over
04:23.23UserReg_CLasdx: gracias
04:23.30asdxUserReg_CL: de nada
04:24.21asdxUserReg_CL: yo estoy probando conectarme a un ITSP
04:24.21coppiceMrTelephone: sounds like its a windows attack
04:25.58UserReg_CLasdx: que es eso ?
04:26.04[TK]D-FenderMrTelephone, nothing says "I Love You" like an iptables drop rule ;)
04:26.23MrTelephoneyou read my mind
04:26.41coppiceI know spanish is the third most spoken language, but I wonder why almost anything on here that is not in English is in Spanish?
04:26.45[TK]D-FenderMrTelephone, Get off my channel... this is a licensed frequency!
04:26.51MrTelephonehehe
04:26.55asdxUserReg_CL: proveedor de voip, para hacer llamadas locales/internacionales
04:27.08asdxUserReg_CL: a PSTN
04:27.22[TK]D-Fendercoppice, there is a nearly derogatory sounding socio/political explanations for that....
04:27.29UserReg_CLasdx: yo tengo una targeta Digium conectada a una trama E1 por la que salgo a pstn
04:27.41asdxUserReg_CL: que bueno
04:27.41MrTelephonei added a 5minute crontab folder in /etc/ and added the appropriate line to /etc/crontab to run-parts in the new folder.. restarted cron using /etc/init.d/cron restart.. doesn't work :(
04:27.51[TK]D-FenderUserReg_CL / asdx : Hey
04:27.55[TK]D-Fender~asteriskspanish
04:27.56jbot[~asteriskspanish] Asterisk Community in Spanish, just visit http://www.asterisk-la.org -=- IRC channel #asterisk-es
04:27.57[TK]D-Fender^^^^^^^^^^
04:28.09asdxlol sorry
04:28.13[TK]D-Fender:p
04:28.36[TK]D-FenderIf you want help you're more able to understand, why not use the resources that are out there...
04:28.48UserReg_CLsorry..
04:29.15[TK]D-FenderUserReg_CL, I'm not actually annoyed or anything, but we all feel like we're missing out :)
04:30.08asdx:P
04:30.09coppicethe americans here just feel anyone not writing in english must be terrorists plotting against them. :-)
04:31.36[TK]D-FenderAl-aqueba, jihad! Jihad!
04:31.44[TK]D-Fender.... I mean.... pass the mayo please!
04:31.51[TK]D-Fender*cough*
04:33.19UserReg_CLI don't know how to call from extensions to other extension, have configuration: http://pastebin.com/m6a05de80
04:34.15[TK]D-FenderUserReg_CL, exten => 9XX,1,Dial(SIP/${EXTEN}) <- tis doesn't work because you are missing the "_" in front of your PATTERN <---
04:34.30[TK]D-FenderUserReg_CL, should look like THIS : exten => _9XX,1,Dial(SIP/${EXTEN})
04:34.36[TK]D-FenderUserReg_CL, Now go fix the rest
04:34.51[TK]D-FenderUserReg_CL, and remove the quotes from : include => "internos"
04:35.02[TK]D-FenderNEXT@!@
04:35.06[TK]D-Fender(c) BKW
04:35.07UserReg_CLThank, sorry
04:35.14[TK]D-FenderUserReg_CL, You're welcome
04:35.17*** join/#asterisk n7okn (n=n7okn@ip68-109-169-42.ph.ph.cox.net)
04:37.25UserReg_CLnow can call to 951 from 950 ?
04:38.09obnauticusF*cking hell this Cisco 7960 is almost Too good of a learning experience (bad) lol.
04:38.58n7oknI have a question. I'm a newbie and I found an alternative conference app for Asterisk I'd like to try... mainly for the experience of installing an Asterisk app. It has a .c extension. I tried the make and configure way but no good. What is it, and how do I compile it?
04:40.01*** join/#asterisk ManxPower (n=manxpowe@55.sub-70-197-188.myvzw.com)
04:40.41[TK]D-FenderUserReg_CL, go try
04:40.41n7oknI tried to configure a 7960 and had to really scratch my head. I think I'll stick with Linksys LOL
04:40.47*** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
04:40.52UserReg_CLyes
04:40.58grandpapadotAnyone see why this won't evaluate? exten => _8XX,n,ExecIf($[${LEN(${DNID})}!=3]|Set|GROUP()=${CONTEXT})
04:41.06obnauticusn7okin what about change the uhh
04:41.09obnauticusFirmware on the thing?
04:41.14grandpapadotIf I just do "=3" it works, but "!=3" doesn't work.
04:42.47n7okn???
04:43.24n7oknu mean on the 7960? not my phone.
04:43.54UserReg_CLI return... reboot machine :)
04:46.45n7oknstill looking for .c script install path.
04:50.29*** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com)
04:50.38obnauticusOh n7okn im doing it right now
04:50.40obnauticusi think I got it
04:50.48obnauticusi have to do an incremental upgrade from their first release.
04:51.01*** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net)
04:51.05UserReg_CLhi :)
04:51.16UserReg_CLone softphone know ?
04:51.31UserReg_CL~software
04:51.32jbotfrom memory, software is doing something that hardware should
04:51.38obnauticus~softphone
04:51.39jbotsomething that should be drug out into the street and shot
04:51.46obnauticus~phone
04:51.46jbotextra, extra, read all about it, phone is warbling while I'm updating the flash from blob...it's amusing now, it's like it knows I'm erasing its brain. Mwa ha ha ha.
04:51.54obnauticusgod damn it
04:51.55obnauticus:|
04:51.59GuyOCanadacan you recommend me a good softphone
04:52.12JTnone of them are good
04:52.15JTsome are ok though
04:52.32UserReg_CLyes need one softphone for windows
04:52.40GuyOCanadai need one too
04:52.45coppicewhat they fail to tell you is all IP phones are soft, and none are very good :-)
04:52.46obnauticusXlite is decent.
04:52.48GuyOCanadauntill i have finished my testing and decide which phone to buy
04:53.06JTlol, decent, sort of ok is more my take on slite
04:53.08JTxlite
04:53.12*** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi)
04:53.37GuyOCanadaI used SJphone before but not so good
04:53.38*** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net)
04:53.39asdxdoes xlite uses alsa on linux?
04:53.52obnauticusprobably
04:54.17UserReg_CLneed one small
04:54.43dlynes!seen flauto
04:54.49[TK]D-FenderZoiper is pretty decent.  One of the few with Transfer.  Or go try Ekiga.
04:55.41UserReg_CLthank
04:56.39ManxPowerXlite is like one of those Japanese gadgets that do a million things, but the user interface is so cluttered, complicate, and confusing that you can't actually do much with it.
04:58.01*** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com)
04:58.26UserReg_CLmmm
04:58.37GuyOCanadaxlite has no spywares in it does itr
04:59.15ManxPowerGuyOCanada: no, it does not as far as I know
05:01.29*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
05:01.51[TK]D-Fendereek
05:02.30dlynes~seen flauto
05:02.33jbotflauto <n=zhao@71.194.141.225> was last seen on IRC in channel #asterisk, 13d 11h 25m 35s ago, saying: 'and email'.
05:02.45UserReg_CLGrr not connect Zoiper
05:03.29obnauticusanyone here with Cisco Phone Experience: `Phone Unprovisioned' recent firmware upgrade to SIP 3.4
05:06.39NuggetI get that whenever the phones aren't getting the tftp server address from my dhcp server
05:07.50UserReg_CLbearercapability notauth
05:08.37obnauticusNugget how do I tell my DHCP server what the TFTP is
05:08.41*** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com)
05:08.44*** join/#asterisk mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com)
05:08.53mackeshey
05:10.38*** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net)
05:11.32[TK]D-Fenderobnauticus, depends which dhcp server you're using.
05:11.42obnauticusWhatever pfsense default is :/
05:12.05[TK]D-Fenderobnauticus, Odds are if you don't even know what you have, you shouldn't be using it :)
05:12.47De_Monobnauticus read the documentation of said DHCP server, duh.
05:13.10obnauticusim using my primary monitor right now otherwise i would
05:13.29BBHossobnauticus: u have to set option 66
05:13.39coppice[TK]D-Fender: that sounds like a line from a bad country and western song
05:13.44UserReg_CLnot work zoiper :(
05:14.21[TK]D-Fendercoppice, you say that... as though there were any other kind ;)
05:14.26coppiceit was only for emphasis
05:15.27asdx[Nov 17 05:14:25] NOTICE[6407]: chan_iax2.c:5258 register_verify: Peer 'teliax' is not dynamic (from 190.52.158.12)
05:15.30asdxwhat does that means
05:15.37BBHossobnauticus: you can edit the dhcpd file to includ this
05:15.52obnauticusk
05:15.53BBHossoption tftp-server-name"YOURIP";
05:15.57BBHossthen
05:16.16BBHossput a tab after tftp-server-name
05:16.38BBHosshttp://cvstrac.pfsense.com/tktview?tn=1026
05:17.46coppicetesting T.38 with 2M bytes FAX pages allows so much time for tea and chatting on IRC :-)
05:17.58obnauticusBBHoss where does it say this file is
05:18.04UserReg_CLneed 2 xlite in same windows but not work
05:18.09BBHossobnauticus: i think that was wrong
05:18.18*** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60)
05:18.38BBHossobnauticus: it just uses dhcpd though, you could look up how to set it in there
05:19.49obnauticusfound it BBHoss
05:20.07BBHossim pretty sure its just freebsd's version of dhcpd
05:20.24[TK]D-FenderUserReg_CL, you'll have to configure one to use a different port.  And even then it'll fight over sound card resources.
05:20.29Mavviefreebsd doesn't have their own dhcpd, it's the ISC DHCPD you are using.
05:20.30GuyOCanadaI successfully connect to asterisk but i can not hear any response
05:20.49UserReg_CLthank
05:21.14mackesneaaaaaaaaaaaattttttoooo
05:21.32UserReg_CLthank friends..
05:21.37UserReg_CLgood night
05:22.11*** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-207-168.rgv.res.rr.com)
05:22.17asdxUserReg_CL: night
05:22.23GuyOCanadahttp://pastebin.com/m1e0937e5
05:22.44UserReg_CLasdx: Buenas noches ...
05:22.55TrentCreekI know Fender knows this one...reject calls from PayPhones
05:23.00TrentCreekBUENOS NACHOS
05:23.13asdxUserReg_CL: igualmente
05:23.19TrentCreeknop
05:23.19asdxTrentCreek: lol
05:24.08TrentCreekdiesperto
05:24.41TrentCreekeveryone wake up
05:24.45mackeshey
05:25.04mackesDoes anything ever happen in the other asterisk channels?
05:25.07TrentCreekgood another victim
05:25.15TrentCreekduring the day in the US
05:25.15mackesOh no
05:25.26mackesIm not new to this happy bunch
05:25.29TrentCreekI needs to know how to reject payphone calls
05:27.50GuyOCanadawhat is the default codec that is assigned?
05:29.32TrentCreekthat would be in extensions.conf
05:30.49[TK]D-FenderGuyOCanada, default is whatever YOU set in your channel drivers conf file.
05:32.43TrentCreekoh thereis meester geetar
05:33.26MrTelephoneIF ${CLIENTID}=PAYPHONE Hangup()
05:33.32coppicehis real name is probably Les Paul or Mr Fernendes
05:33.34MrTelephonei wish
05:33.47TrentCreekhey!! thanks Mr Obvious!
05:34.03TrentCreekor Esteban!
05:34.10MrTelephoneyour welcome skywalker
05:34.28GuyOCanada150 ms delay thats too much for voip isnt it?
05:34.45MrTelephonetrentcreek, see if your telco delivers ANI2 codes
05:34.58MrTelephoneim 150ms and it sounds good
05:35.15MrTelephonepacketloss and jitter are the killers
05:35.28MrTelephonesat phones are like 300-600ms
05:35.47GuyOCanadawell i hear my sound like 1 sec after i talk on the echo test
05:35.49TrentCreek"You know thousand of youngers long to express theior hearts with the gift of Music!
05:35.57GuyOCanadabut the ping latency is 150 ms
05:36.10MrTelephonebut the echo test is already delayed
05:36.28GuyOCanadait is?
05:36.37MrTelephonedid you try it on your local network?
05:36.47ectospasmasterisk-gui needs work... I couldn't figure out how to get it to go to my voice menu (instead of the demo) without hand editing extensions.conf...
05:36.50GuyOCanadano i work remote
05:37.03MrTelephone150ms should be good
05:37.34GuyOCanadabut the lady talking when i call stops talking for a sec or two sometimes
05:37.37TrentCreekoh Mr telephone...
05:37.57MrTelephoneyou get used to it
05:38.15MrTelephoneits even worse if your not tied into the pstn at your server
05:38.15TrentCreekI realise I need to reject before it hangs up..I dont want to be wracking up 50 cents payphoen carges
05:38.35TrentCreeki mean reject before it answers
05:38.53MrTelephonei read one article where if you disconnect before 10 seconds its not considered a call
05:38.57MrTelephonethat was for collect though
05:39.08GuyOCanadagrr i think its a codec problem but not sure
05:39.42MrTelephonetheres 1000ms in a second
05:39.45TrentCreekI wonder if I can put that statement before the asnwer statement
05:39.53mackesDo any of you use FWD or Sipphone?
05:40.09MrTelephonetrentcreek, you don't have to answer() incoming on a t1
05:40.18MrTelephone?
05:40.46MrTelephoneGuyOcanada, why do you think that?
05:40.54GuyOCanadaMrTelephone yes there is but still it is annoying when the auto attendant drops for a second
05:41.08TrentCreeknot operating on T1
05:41.10MrTelephoneat the beginning of the call or in the middle of the call?
05:41.28GuyOCanadacon grat ula tions , con figu rate d
05:41.32GuyOCanadaits talking like that
05:41.41[TK]D-FenderGuyOCanada, 150 can be just fine.  What codec are you using, and what is the bandwidth of your connection?
05:41.50GuyOCanadaif you need more te c hni cal info rma
05:41.52MrTelephonethats scray
05:41.57MrTelephonethats packetloss or jitter
05:42.05GuyOCanadawell
05:42.11GuyOCanadathe server runs on a 100mbit connection
05:42.32GuyOCanadathe clients runs on a 1.5 mbit dsl connection
05:42.38*** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
05:42.38*** mode/#asterisk [+o russellb_] by ChanServ
05:42.41GuyOCanadai have not changed any config files so its the default codec
05:43.28MrTelephoneif you access your voicemail on it, does it sound shitty?
05:43.30[TK]D-FenderGuyOCanada, Never say "default".  Go look in your configs and find out what you're doing.
05:43.33MrTelephoneor just when you phone another client?
05:43.47MrTelephonenever say default. hah thats a new one
05:43.49MrTelephonehaha
05:43.52GuyOCanadalet me check
05:44.07MrTelephone./nick MrDefault
05:44.26GuyOCanadaquality 3
05:44.32GuyOCanadacomplexity 2
05:44.42GuyOCanadaenhancement true
05:44.43MrTelephoneis that iax?
05:44.45[TK]D-FenderGuyOCanada, in ASTERISK.
05:44.55[TK]D-FenderGuyOCanada, That was meaningless
05:44.59MrTelephonei never seen those figures before
05:45.06MrTelephonei guess i have to start drinking
05:45.21GuyOCanada:) i was looking in to the codecs.conf
05:45.26[TK]D-FenderMrTelephone, Necrophilliac?
05:45.44[TK]D-FenderGuyOCanada, You are clealy clueless.  Go stop and read THE BOOK.
05:45.45[TK]D-Fender~book
05:45.46jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
05:45.50MrTelephoneNecrophilliac=ulaw
05:46.13coppiceGo stop - what kind of assistance is that?
05:46.34GuyOCanadaI already have the book open
05:46.35MrTelephonei hope my grandmother doesn't fall down in front of that guy
05:46.49GuyOCanadaSetting Up the Dialplan for Some Test Calls thats where i am
05:47.17MrTelephonetest play your site opz without bitttorent open
05:47.34[TK]D-FenderGuyOCanada, you set codecs in your channel driver's conf file.  if thats SIP, then look in sip.conf
05:48.06MrTelephonewhy can't asterisk configure itself? who the hell made this program.. who didn't think of that
05:48.13MrTelephonei should be on the board
05:48.22MrTelephone:P
05:49.25GuyOCanadabittorent and voip please not in the same building
05:49.50MrTelephonethat breaking up tho
05:49.54MrTelephonei have that on my cable network
05:50.02[TK]D-FenderMrTelephone, You asked for it...
05:50.29MrTelephonei went out today and changed a bunch of connectors on the poles and got my errors down to %0.001
05:50.39MrTelephoneand things are clear again
05:56.40*** join/#asterisk sergey (n=sergey@91.189.233.71)
05:58.43russellb_the release of res_telepathy is expected tomorrow
05:59.18*** topic/#asterisk by russellb_ -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.4.14 (2007/11/16), *-Addons 1.4.4 (2007/10/16), Zaptel 1.4.6 (2007/10/18), Libpri 1.4.2 (2007/10/16) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn -=- #freepbx (freepbx.org) or #trixbox for trixbox (trixbox.org) support
05:59.38TrentCreek"You know thousand of youngers long to express their hearts with the gift of Music"
05:59.56TrentCreek"Hello, this is Fender and I have important news"
06:00.41TrentCreek"I just saved a bundle by switching my Geetar Insurance to GEICO"
06:00.51*** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com)
06:01.18[TK]D-Fenderrussellb_ : But then again... we knew that already ;)
06:01.43De_MonI didn't know that, you must be testing before release
06:01.46[TK]D-FenderTrentCreek, and my nick isn't named for the guitar maker...
06:01.53MrTelephonei rootshelled a nasa.gov computer once
06:01.57MrTelephonethey were running IRIX
06:02.05TrentCreekoh....
06:02.13MrTelephoneis that another name for MAC/OS
06:02.20coppice"Hello, this is Fender and I have important news. That guy in the movie Robots was an imposter"
06:02.26TrentCreekI just saves a bund by switching my Classic Car insurance to GEICO
06:02.43De_Monwhere does your name come from [TK]D-Fender ?
06:03.04TrentCreekfrom Les Paul as Mr Obvious said
06:03.40MrTelephonewhose mr obvious
06:03.46De_Monwhos Les Paul?
06:03.56De_Mons/s/se/
06:04.02MrTelephonehe was the guy who played kermit the frog on sesame street
06:04.13coppiceDe_Mon: he plays a Fender, and has so far only learned to play in D
06:04.29De_Monokay...
06:04.34[TK]D-Fendercoppice, how ASStute :p
06:04.36ectospasmI'm confused as well
06:04.57De_MonMuppet humor I suppose
06:05.13[TK]D-FenderDe_Mon, its from my preffered role while playing tribes 1 CTF years ago.  It stuck with me through my AHL that followed
06:05.13MrTelephonei had a friend who burned out on guitar hardcore
06:05.24MrTelephonehe downloaded the song wipeout and memorized it in 7 minutes
06:05.25TrentCreekis an American jazz guitarist and inventor. He is a pioneer in the development of the solid-body electric guitar which "made the sound of rock and roll possible."[1] His many recording innovations include overdubbing, delay effects such as "sound on sound" and tape delay, phasing effects and multitrack recording.
06:05.37De_MonThat, makes much more sense
06:05.51[TK]D-FenderAHL clan*
06:05.53MrTelephoneyou thought it was a lesbian didn't you?
06:06.21TrentCreekBurnt out on a geetar? Its kept Les Paul going for 92 years
06:07.20De_Mondont let D`mon getchoo down fight back with our black powwah
06:07.47MrTelephoneI had to look up IIRC
06:08.04TrentCreekHere's Mr Obvious
06:08.05TrentCreekhttp://riz.vox.com/library/audio/6a00cd96fcbade4cd500d4142b13ea6a47.html
06:08.21coppiceI don't think the Les Paul guitar exactly made rock and roll possible. This guitar http://www.stevehowe.com is older than the Les Paul
06:08.26MrTelephoneanother 3 years and we'll all be writing in 4 and 5 letter words capitalized
06:13.13De_MonIDTS
06:13.28MrTelephoneovern cleaner?
06:13.46De_Monthey will be acronyms in that case
06:14.00MrTelephoneDMN R PPL GUNA USE ABREVS ALL THE TIME
06:14.04MrTelephoneheheh
06:14.15MrTelephonewhose listening to mr obvious?
06:14.24*** join/#asterisk bintut (n=chatzill@cm246.gamma178.maxonline.com.sg)
06:14.28De_Monoh, that
06:14.30TrentCreekeveryone!
06:14.41TrentCreekHe helps me a lot
06:14.47De_MonMrTelephone have you watched Californication?
06:14.58MrTelephonehe thinks oven cleaner is made for cleaning out a vagina
06:15.06MrTelephonei don't think so
06:15.13MrTelephonei watched the californian fires
06:15.21TrentCreekDon't forget that EasyOff for your wife!
06:15.28bintuthello all.. i'm wondering why the output of "sip show peers" inside the asterisk shell tells me that one of my users is using port 10064? any idea? i'm running asterisk-1.4.13 here
06:15.36De_Mondavid ducovney(sp) is a writer and he bitches about that cyber talk in one episode
06:15.46MrTelephonebintut, because thats what they are using
06:15.59MrTelephonebintue, you should phone MrObvious
06:16.03bintutMrTelephone: you mean, on the sip softphone side?
06:16.05MrTelephoneyeah
06:16.11bintuti see..
06:16.18MrTelephonethe outbound/return port of the client
06:16.29bintuti have to inform the user then to use 5060
06:16.37MrTelephonehow can you
06:16.43MrTelephonemaybe they have a router and its randomly picking a port
06:17.06TrentCreekYes..Mr Obvious is giving excellent advice! http://riz.vox.com/library/audio/6a00cd96fcbade4cd500d4142d73793c7f.html
06:18.32De_MonI've got peers behind nat on all sorts of random ports
06:18.48MrTelephonebintut, that is not the ports on your server
06:18.54MrTelephoneits on the clients side
06:18.58MrTelephonewhy does it matter
06:19.02MrTelephoneme too
06:19.10bintutMrTelephone: yeah. thanks.. :)
06:19.30De_Monmaybe you should share the problem instead of jump to the wrong solution
06:20.03GuyOCanadahow do you setup a DNS SRV record for sip?
06:22.05MrTelephonei even have a customer on port 65536 for pete sakes
06:22.23MrTelephonegoogle dns srv asterisk
06:25.02MrTelephoneIIRC = Interactive Illinois Report Card
06:29.26bintutgtg now.. thanks all.. thanks MrTelephone.. :)
06:30.58MrTelephonei love the linux convos about mounting
06:31.21[TK]D-Fender~sex
06:31.22jbotupdatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/
06:34.33MrTelephonehahaha
06:34.35MrTelephonefunny
06:35.45TrentCreekMy command: exten => _X.,3,Dial(SIP/phone1@phone1,10) is not calling the device..darn it
06:36.08MrTelephone<edman007> Jester_, i just mounted an FS inside itself, and it works :) and its ext3
06:36.08MrTelephonenerd
06:36.20GuyOCanadawhich is the least bandwidth using codec?
06:36.21TrentCreekapp_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
06:36.27MrTelephoneguy0canada, g729
06:36.38MrTelephonetrent, show sip peers
06:36.46MrTelephonetrent, are they registered?
06:36.51[TK]D-FenderG.723
06:36.54[TK]D-Fender^
06:37.14MrTelephoneis g723 supported by most phones?
06:37.20MrTelephonemight as well use gsm
06:37.29TrentCreekooops..no such command
06:37.51TrentCreekohhh
06:37.58MrTelephonesip show peers
06:39.00TrentCreekoh...I see now...thanks..I go the wrong name
06:39.34MrTelephonewhen is asterisk going to add background noise i wonder
06:39.49TrentCreeknext version
06:39.59MrTelephone1.6?
06:40.42TrentCreekalreayd near 1.6? I am still on 1.4.11
06:40.49MrTelephoneim using 1.2
06:40.52GuyOCanadabackground noise?
06:41.03*** join/#asterisk Defraz (n=t0tal@63.228.246.245)
06:41.24MrTelephoneso you don't hear complete silence when someone stops talking
06:41.26TrentCreeklast I saw it was at 1.4.13
06:41.34TrentCreekmaybe a month ago?
06:43.19MrTelephoneyeah i think so
06:43.24MrTelephoneim too scared to try it
06:45.17[TK]D-Fender1.4.14
06:46.06asdxi need a phone number of a pstn line to try this
06:46.45MrTelephonewhere the hell is sqrt on wincalc
06:46.52MrTelephoneis it 1/x^2
06:47.15[TK]D-FenderMrTelephone, install something better :)
06:49.02MrTelephonepiece of crap
06:49.10MrTelephone1/x^-2
06:49.15MrTelephone?
06:51.35[TK]D-FenderMrTelephone, stop whining and REPLACE IT already...
06:52.47GuyOCanada; a call in the case of a phone disappearing from the net,
06:52.47GuyOCanada; like a powerloss or grandma tripping over a cable.
06:52.54GuyOCanadagrandma tripping over a cable (LOL)
06:53.13MrTelephoneoh its X^.5
06:53.14TrentCreekstaple them in the wall
06:53.17MrTelephoneto get square root
06:53.24TrentCreekanyone seen the goodie ads for next week?
06:54.11MrTelephoneno but neo beat the machines
06:54.37TrentCreekwww.bfads.net
06:54.42TrentCreeknot for long
06:55.19asdxi already hear tone
06:55.32asdxdoes that means that i'm connected to pstn?
06:55.36asdxwhen i dial
06:56.10TrentCreekare you uising a SIP device?
06:56.23asdxIAX2
06:56.51TrentCreeki think it may provide the dial tone for you then it merely dials out on that line
06:57.05TrentCreekafter you enter the number
06:57.22asdxok
06:59.22TrentCreekbecause you may dial an extension rather than a phone number so it needs to know what you want to do before it starts doing it
07:01.46asdxi see
07:05.27[TK]D-Fender<PROTECTED>
07:07.40asdxcool
07:13.10asdxwhen i'm making the call, do i have to use Answer() or just Dial()?
07:14.16[TK]D-Fenderasdx, Just Dial
07:14.25asdx[TK]D-Fender: ok
07:15.32[TK]D-Fenderok, bed time.  Later all
07:24.02asdxi'm dialing some numbers but i don't get answer
07:24.16asdxi hear tone though
07:28.14*** join/#asterisk BeeBuu (n=chatzill@218.13.66.237)
07:30.14BeeBuui had installed asterisk 1.2.13,can i play wav files without install addons?
07:36.57GuyOCanadahow can you authenticate a user?
07:39.24TrentCreekwith SIP?
07:40.22GuyOCanadayes
07:40.33GuyOCanadabut what i want to do is
07:40.42GuyOCanadaI have an incoming line which is a tollfree number
07:41.07GuyOCanadawhen i call to that number from a payphone i want to be able to call someone using an outbound connection
07:41.19GuyOCanadaso i need a method to authenticate the caller
07:44.30BBHosslook up DISA
07:49.51GuyOCanadaso if i do "exten => 555,1,DISA(134|payphone-dialout)" and call my tollfree number and enter 555 when im asked for the extension and then right after it enter 134 i will be redirected to the payphone-dilaout context?
07:51.54GuyOCanadais it possible to send video using asterisk (a media file that is saved on the system?)
07:52.42tzafrir_homeBeeBuu, sure. As long as those files are 8000Hz, mono, 16 bits per sample
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08:02.13asdxi can dial and everything but i never get an answer from the other side
08:02.18asdxwhy could that be?
08:03.28*** join/#asterisk BeeBuu (n=chatzill@218.13.97.55)
08:03.47BeeBuuany one help me please
08:04.04BeeBuui can't heard any music when set music on
08:04.48BeeBuuthere is "-- Started music on hold,class 'default' on channel 'sip/998-09d3aa80' "
08:04.51DarkDlx<tzafrir_home> BeeBuu, sure. As long as those files are 8000Hz, mono, 16 bits per sample
08:04.53BeeBuuin CLI>
08:04.57tzafrir_homeBeeBuu, where exactly do you try to hear it?
08:05.05GuyOCanadaalarm receiver?
08:05.14tzafrir_homeIs there anything else you can hear there?
08:05.20BeeBuunothing...
08:05.24BeeBuusilent....
08:05.36tzafrir_homeBeeBuu, please help us help you
08:05.55BeeBuutzafrir_home: so what i need to do now?
08:06.04tzafrir_homeah, a SIP phone
08:06.27tzafrir_homecan you hear a simple sound file in it?
08:06.37tzafrir_homesomething with Playback?
08:06.45BeeBuui had set mode=files and directory=/var/lib/asterisk/mohmp3
08:06.57tzafrir_homeCan you hear anything from Asterisk?
08:06.58BeeBuuno,just silent....
08:07.05BeeBuunothing at all.
08:07.43BeeBuuthere is "-- Started music on hold,class 'default' on channel 'sip/998-09d3aa80' " in CLI>
08:08.09BeeBuuso ,that's mean music playing,right?
08:08.44*** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net)
08:11.14tzafrir_homeBeeBuu, not from on-hold music. Can you call some test extension? echo test?
08:11.23tzafrir_homeplay a file?
08:12.02BeeBuuexten=>_99X,n,SetMusicOnHold(default)
08:12.29BeeBuuexten=>_99X,n,waitMusicOnHold(20)
08:12.44BeeBuuthat's all about musiconhold
08:13.39BeeBuumy headphone is OK
08:18.34tzafrir_homeBeeBuu, how can you tell that?
08:19.16tzafrir_homelet's see....
08:19.28BeeBuui can hear the dial sound
08:19.51tzafrir_homethe dialtone comes from the handset itself, not from Asterisk
08:20.23BeeBuui can hear anything play with other application..
08:20.43tzafrir_homefine
08:21.06tzafrir_homeso, what is the output of:  'moh show classes'  in the CLI?
08:21.16tzafrir_homeplease pastebin it if it is longer than 3 lines
08:21.57BeeBuuNo such command
08:26.10BeeBuu"moh classes show " show this http://pastebin.comd1f99926
08:27.20BeeBuuo,god,it's OK now
08:27.31BeeBuuthanks tzafrir_home
08:27.55*** join/#asterisk porche (n=porche@81.215.122.108)
08:28.01porchehi everybody
08:28.53porchei am looking for some reasonably prices + reliable voip termination service, like gafachi (but cannot use gafachi currently), which one do you suggest?
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08:36.20tzafrir_homeBeeBuu, something is wrong with that URL
08:36.33porchetzafrir hi
08:36.57tzafrir_homeah, problem fixed, I see. So was this a matter of reload?
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08:38.56BeeBuutzafrir_home: you are right  :-P
08:39.00BeeBuumust reload
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09:34.41JimmyDeemorning everyone, I have a problem, my http.conf is set to 8080 and my manager.conf 8081 and neither port is open on nmap, what gives?
09:38.54GuyOCanadaJimmyDee
09:39.05GuyOCanadawhat os?
09:39.56DarKnesS_WolFany idea about a decent IAX client opensource ?
09:40.30GuyOCanadaJimmyDee: service iptables stop and try again
09:48.15JimmyDeeiptables is stopped, no joy
09:55.14CBU[^_^]M``my SPA3102 wont register :(
10:07.24GuyOCanadaAnyone using Windows Mobile 6 SIP client with asterisk?
10:09.36tzafrir_homeDarkDlx, basically two to look at: kiax and iaxcomm
10:10.00tzafrir_homeDarKnesS_WolF, that is.
10:15.05*** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
10:16.34DarKnesS_WolFtzafrir_home: hehe long time didn't see u too ;)
10:16.49DarKnesS_WolFtzafrir_home: i tryped zoiper but sometime the asterisk server loses teh DTMF
10:17.02DarKnesS_WolFkiax nahhh i'm not big fan of qt and kde in gernal will try iaxcomm it is ugly but works
10:17.08tzafrir_homeLosing DTMFs? on IAX?
10:17.21DarKnesS_WolFtzafrir_home: yes
10:17.26DarKnesS_WolFi'm using a2billing
10:17.30tzafrir_homein what direction?
10:17.38DarKnesS_WolFso when i enter teh cardnumber sometime it loses
10:17.54DarKnesS_WolFzoiper ---> IAX --->
10:17.55DarKnesS_WolF*
10:18.11obnauticusCan someone here tell me what is wrong with this:
10:18.11obnauticusexten => *5,1,Dial(sip/*5)
10:18.18tzafrir_homevery strange.
10:18.19obnauticus(retorical question)
10:18.37obnauticusWhen I call *5 from an other line it's not ringing it.
10:18.44obnauticusan other device*
10:19.18tzafrir_homeTry a more robust test:    exten => _123.,1,SayDigits(${EXTEN:3})
10:19.52tzafrir_home"123" is an arbitrary prefix to allow you to combine this in your dialplan
10:20.12tzafrir_homejust dial a number and listen to what Asterisk thinks it has recieved
10:20.42obnauticus<--- SIP read from 10.0.0.110:51546 --->
10:20.42obnauticusACK sip:123*5@10.0.0.109 SIP/2.0
10:20.54obnauticusIt still doesn't ring SIP/*5
10:20.58obnauticusbut i can Origionate a call.
10:21.03obnauticusoriginate SIP/*2 application Dial sip/*5
10:21.07obnauticusThat works :|
10:25.30*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
10:26.12obnauticusK i got it tzafrir_home
10:27.52tzafrir_homeobnauticus, I agree that using '*5' as a sip peer name is quite strange
10:28.12obnauticusI'm trying to orgonize peers with a * in their name :/
10:28.19obnauticusor.... ya
10:28.21obnauticusyou know what I mean
10:28.28obnauticusservices with ##<serv_#>##
10:28.32obnauticusand etc.
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10:49.29porchehi all
10:49.51porchecan someone suggest me a good termination service?
10:57.06*** join/#asterisk Shaun2222 (n=shaun@ip68-4-127-67.oc.oc.cox.net)
10:57.28Shaun2222how's asterisk work on 64bit linux distros, any problems?
10:58.36DarKnesS_WolFShaun2222: i don't think so i never did but i don't think there is should be a problem
10:59.36tzafrir_homeShaun2222, it is regularily used
11:05.40*** join/#asterisk _feqma (n=paul@pool-72-65-41-105.bflony.east.verizon.net)
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11:17.22SiyaAnyone here with any hints why a Linksys ADSL router might prevent me from connecting to my * server via SIP?
11:17.51SiyaOther places/routers/links with NAT work fine
11:18.02SiyaSo I'm suspecting the Linksys
11:18.10Siyaor the ISP blocking SIP
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12:44.43feqma\quit
12:45.41CBU[^_^]M``hello
12:45.44CBU[^_^]M``anyone here?
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13:06.59ussrbackHi all
13:07.07ussrbackI have error [Nov 17 16:48:16] NOTICE[3580]: src/chan_h323.c:1885 reload_config: Unable to load config ooh323.conf, OOH323 disabled
13:07.08ussrbackLoaded chan_ooh323 => (Objective Systems H323 Channel)
13:07.20ussrbackhow can i fix it
13:07.27ussrbacki have installec chan_ooh323
13:14.57*** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca)
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13:49.38rob0I'm back in the USSR ... you don't know how lucky you are, BOY.
13:50.12*** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
13:50.44grandpapadotHi all.  Using Asterisk Static-Realtime in 1.2.x, can I have my extensions.conf in a db, then still call external files via #include?
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14:10.46grandpapadotIs there any way to set context specific "global" variables?
14:11.42deeperrorCONTEXT_VAR?
14:19.22*** join/#asterisk BiG^DoG (n=BiG^DoG@c-67-162-233-20.hsd1.de.comcast.net)
14:20.05BiG^DoGis there a list of good voip providers?  I'm considering switching my analog phone number to voip
14:20.25*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
14:20.37deeperrorBiG^DoG: I use VoicePulse
14:23.11*** join/#asterisk Trionnis (i=lordkuri@s233-51-251.nap.wideopenwest.com)
14:23.23BiG^DoGsatisfied with them?
14:23.57deeperrorused them for a while
14:24.03deeperrornot had many issues
14:24.26deeperroralso ran 10 agents in a callcenter on them for a while
14:24.40*** join/#asterisk disa-help (n=phobosd@shell.intarwebnetorg.com)
14:24.51deeperroralso used callcentric not too bad
14:25.06BiG^DoGmy big question is what happens if my asterisk box dies and I'm away for a day or two?
14:25.22deeperrorhaha
14:25.29deeperroryour * box doesn't die
14:25.38TrionnisI would think the answer to that would be kinda obvious :)
14:25.53BiG^DoGI didn't know if they offered some kind of voicemail service
14:25.59BiG^DoGas an add on
14:26.23deeperrorthey might if you signup for regular service but not on connect
14:26.38disa-helpanyone know much about this error msg?
14:26.40disa-help-- Executing NoOp("SIP/5553-08201b48", "Dial failed due to CONGESTION - failing through to other trunks") in new stack
14:26.55disa-helpinbound calls are getting all circuits busy too -- but i'm not sure if it's an asterisk tag
14:27.16deeperrorfirewall?
14:27.28disa-helpnegative
14:27.35deeperroriptables?
14:27.42disa-helpnope :-/
14:27.46disa-helpzaptel config the same
14:27.51disa-helpPRI provider claiming it's us
14:27.55disa-helpyou familiar with asterisk tags?
14:28.03disa-helpmaybe you can call to check? hehe
14:28.09disa-helpi honestly can't remember what they sound like..
14:28.23[TK]D-Fenderdisa-help, means you're running a GUI that isn't supported here.
14:28.40disa-helpGAH
14:28.42disa-helpALWAYS GETTIN ME
14:28.43disa-helplol
14:28.46Trionnisuh oh, now you have Andrew after you
14:28.50BiG^DoGdeeperror: so I don't sign up for the $24.99 plan?  I get the IAX termination, right?
14:29.02disa-help[TK]D-Fender: that, or it's somtehing on the provider side
14:29.09deeperrorfor asterisk yea
14:29.15BiG^DoGk
14:29.18deeperrorthey call it connect
14:29.26deeperroryou can shop around though
14:29.30deeperrorthere might be a better deal out there
14:29.47Trionnishow many minutes are you talking about?
14:29.49BiG^DoGthat's what I was looking for... I know if I'm looking for a webhosting company, I can go to a website that ranks the top hosting companies
14:29.52Trionnisrough guess
14:29.54BiG^DoGis there a simliar thing for IAX termination?
14:30.04[TK]D-Fenderdisa-help, that message doesn't actually mean ANYTHING.  we don't see your dial, link status, or configs.
14:30.39disa-helptrue, but i find it hard to believe it's my pbx's problem if nothing has changed
14:30.50Trionnishow many minutes a month?
14:31.05BiG^DoGinbound or outbound?
14:31.10Trionniseither or both
14:31.12BiG^DoGit's a residential thing so typical "home use"
14:31.15Trionnisah
14:31.16BiG^DoGI haven't actually measured
14:31.17[TK]D-Fenderdisa-help, pastebin the full CLI output of your failed call at verbose 10 and do "pri debug" first so we can see whats going on
14:31.17Trionnisn/m
14:31.28TrionnisI thought it was for a call center... must have misread
14:31.49deeperrori was just stating that i have ran some cc activity thru VP with no issues
14:32.01Trionnisyou can get really good pricing from voipjet if you push enough minutes through them :)
14:32.37deeperrorhttp://www.digium.com/en/ecosystem/partners/partners.php
14:34.03deeperrormy old man put 100 bucks on a pre pay account...told him he will never use that much on voip that would be like 5 years of calls haha
14:34.48BiG^DoGso $50 for startup would last a while?
14:35.03deeperrorif you get did will cost 11/month
14:35.09deeperrorhe just has it for outbound calls
14:35.42deeperrorbut you would probably want did
14:35.52deeperrorcould even port your number to them
14:36.27BiG^DoGand is there a per minute charge on DID?
14:36.50deeperrordid per month per minute outbound
14:37.51Trionnissome places do per minute, some do flat rate
14:37.55Trionnismost are flat rate
14:38.17BiG^DoGper minute outbound and there's no local area?  every outbound call has a per minute?
14:38.37Trionnismost I've found are like that
14:38.54deeperrordepends on how many minutes you run.  if your not using the flat rate worth of minutes better to not pay if your making a lot of calls flat rate
14:40.05BiG^DoGgot it... if I'm making more than $25 worth of outbound a month, go with the flat rate... Otherwise, stick with per minute
14:40.42Trionnis$25/mo if calls...... wow...
14:40.54TrionnisI don't think I've ever seen one of our phone bills that cheap
14:40.55Trionnis;)
14:41.13BiG^DoGis $25 a month a lot?
14:41.27BiG^DoGremember, I'm talking residential, not commercial
14:41.33TrionnisI know
14:41.54Trionnisdepending on the per minute, that can be a good amount of calls
14:42.06Trionnisat home one of the DID providers I use is sipmedia
14:42.07[TK]D-Fenderother calculation is that if you need multiple simultaneous channels, the per-min option often lets you have more than 2 calls unlike "residential" style fixed channel services
14:42.17Trionnis$5/mo for a DID with 500 minutes outbound
14:42.29TrionnisI've never gone over
14:42.55Trionnismy bad, the 500 outbound is $9.95/mo, the $5 plan is with 100 outbound
14:42.58BiG^DoGI was just looking over the voicepulse website and it says they don't offer callerid name on did... is that fairly common?
14:43.10Trionnismost voip providers don't offer that
14:43.25Trionnisthey don't have the CNAM equipment to do the lookups
14:43.34deeperrorI setup a script and database and make custom names for callers
14:43.39Trionnisthat works too
14:43.42BiG^DoGgotcha
14:43.46BiG^DoGthat was what I was going to say
14:43.56BiG^DoG* can take the cid and do a database lookup and insert it
14:44.20deeperroreven wrote a windows service that will pop open web browser to edit that db info on inbound calls
14:44.46Trionniswow
14:44.50deeperrorbuttons for callback, edit info
14:44.56Trionnisthat's a lot more effort than I'd put into it ;)
14:45.15BiG^DoGthanks for the good information... the dead end last night on call waiting put the final nail in my analog coffin I think
14:45.19deeperrorwell i play around to learn new things and sometimes toys evolve haha
14:46.02Trionnisspeaking of playing around... anyone have about 200-250 channels sitting on the PSTN they want to let me tie up for about 30 minutes? ;)
14:46.11Trionnisneed to do some capacity testing :)
14:51.35*** join/#asterisk cypherdelic (n=cypher@p5B27EA05.dip.t-dialin.net)
14:54.30*** join/#asterisk UserReg_CL (n=dede@200.113.130.111)
14:54.36UserReg_CLhi, good day !!!
14:54.55UserReg_CLhi TK
14:56.28UserReg_CLone question... where is defined Trunks ?
14:59.46BiG^DoGis broadvoice any good?
15:00.11*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
15:02.20Greek-Boywho has tried Five 9's?
15:07.39*** join/#asterisk irule (n=irule@200.53.61.4)
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15:14.28mackesGood Morning everyone?
15:15.30brookshirehi
15:15.50mackesHey
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15:26.12disa-helpfixed
15:26.17disa-helphad to unplug/replug the damn thing ;p
15:26.25disa-helpzaptel cards are pissy sometimes i guess
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15:26.25*** mode/#asterisk [+o russellb_] by ChanServ
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15:55.10jameswf-homejbot: you suck
15:55.11jbotand very well I might add
15:55.26Greek-Boyjbot: stuff you
15:55.27jbotACTION grab's you, fills them up with stuffing, and sticks them in the oven
15:55.45Greek-Boyjbot: u not clever
15:55.48jameswf-homejbot: drob table;
15:56.03Greek-Boyjbot: you idiot
15:56.04*** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net)
15:56.25unixdogwhat is asterisk and how will it better my life ?
15:56.26jameswf-homejbot: dropdatabase;
15:56.26jbotSo you think you could kill me, did you. You thought you could come in here, in my own home, and take me down without a fight. I would have given you a chance, really. But now, you have unleashed hell. May god have mercy on your soul.
15:56.31unixdoghow will it make me rich
15:56.32jameswf-homelol
15:56.57coppiceunixdog: its not about wealth. its about sex
15:56.58jameswf-homeit wont if you wanna get rich go to M$
15:57.35riddleboxlol
15:58.13jameswf-homeI get Laid every other friday thankls to asterisk, but seriously its more likely the paycheck that makes my wife give it up
15:58.52riddleboxlol, jameswf-home so you sell and install asterisk servers?
15:59.03Greek-Boyyeah women just love money!
15:59.20jameswf-homeno I develop and support for an asterisk hardware mfg
15:59.36Greek-Boyjameswf: which one?
15:59.42*** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com)
15:59.43jameswf-homenot in china not sangoma
15:59.56riddleboxjameswf-home, cool, I wish I could work on linux/asterisk full time
16:00.13Greek-Boyu lucky man
16:00.15jameswf-homemove to az i will try to get you a job lol
16:01.27riddleboxjameswf-home, I did my Avaya IP Office training in AZ, I would love to live there
16:01.56Greek-Boyriddlebox: isn't it too hot? where do you live now?
16:02.04riddleboxGreek-Boy, Illinois
16:02.10jameswf-homeIn our office you can use windows but people will make fun of you so most have some flavor of linux or mac
16:02.56riddleboxI would love an environment like that, I always complain how we work on Avaya systems like Communication Manager which runs redhat but the admin software in windows only
16:03.27jameswf-homeWhen I started in telephony I installed toshibas... it was ass
16:04.23riddleboxI work on a Toshiba Perception e --it was installed in 1983, I was 3 haha
16:04.33unixdogMS is for loosers
16:04.42unixdogwho have to real kill sets
16:05.15unixdogwho are appliance users
16:05.33unixdogand I was making a bad joke
16:05.41unixdog< ==== formerly darwin 35
16:05.51jameswf-homewell all money hungry r tards should use MS as they have the same goals
16:05.56unixdog<==== he who ported asterisk to freebsd
16:06.25jameswf-homedidnt you tell me your not a programmer unixdog
16:06.35unixdogMS has one goal make users stupid so they have to pay more money for support
16:06.55jameswf-home*cough Tricboc *cough
16:07.01unixdogI am not but back whan I ported it . it was not hard
16:07.15riddleboxlol
16:07.15unixdog9trashbox
16:08.11jameswf-homejbot kerry
16:08.12jbotit has been said that kerry is something you eat
16:08.42coppicejbot pussy
16:08.43jbotRead: coppice
16:08.58riddleboxhaha
16:08.59jameswf-homejbot coppice
16:09.00jbotwell, coppice is rather like underwood, only different, or the faxing master
16:09.13riddleboxjbot riddlebox
16:09.23jameswf-homejbot faxing
16:09.23jbotwell, faxing is 8% knowledge, 5% skill, 11% luck, and 76% voodoo
16:09.29riddleboxdang
16:09.43coppicejbot voodoo
16:09.44jbot[voodoo] black magic, stay away from it. It is also a chipset, ask me about 3dfx.
16:09.54Qwell~3dfx
16:09.54jbot3dfx is probably the name of the company who makes the legendary voodoo 3D acceleration cards (see http://www.3dfx.com). There is also /dev/3dfx which is needed for glide support under X <=3.3.6 (see device3dfx-source package). It is not needed for X4, use tdfx instead. now owned by nVida.... time to start buying Matrox  :), or at #3dfx
16:09.54coppicejbot T.38
16:09.55jbothmm... t.38 is see t38
16:10.05jameswf-homejbot sex
16:10.06jbotupdatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/
16:10.07riddleboxis there any sip phones that will allow you to just start pressing buttons to dial without hitting speaker or picking up the handset?
16:10.11coppicejbot t38
16:10.11jbotit has been said that t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon
16:10.58coppicejbot foip
16:10.59jbotfrom memory, foip is Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject
16:11.53jameswf-homejbot: you
16:11.54jbotjbot is probably a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or  [TK]D-Fender's b*tch
16:12.20coppicejbot [TK]D-Fender
16:12.20jbot[TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy.
16:13.04jameswf-homejbot: wtf sex
16:13.14Qwell~botabuse
16:13.14jboti guess botabuse is fun
16:13.24coppicejbot virgin
16:13.24jbotI'm sexless
16:13.29jameswf-homejbot: botsnack
16:13.29jbotaw, gee, jameswf-home
16:13.31*** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt)
16:14.33coppice100 page FAX tests are boriiiiinnnngggg zzzzzzzzz.......
16:15.16jameswf-hometry 200
16:15.40coppicenah. I'm using 100..... but over and over all day
16:16.01coppiceand a 2M byte single FAX page
16:16.26Greek-Boyanyone here use astpp?
16:16.46riddleboxcoppice, you are testing faxing over IP?
16:16.56coppiceyes
16:17.32riddleboxthrough a provider? or just through asterisk?
16:17.32Greek-Boycoppice: Have you tested T.37 and if yes which solution/package do you recommend?
16:19.22coppiceI don't think anyone has done a precise implementation of T.37 for use with free faxing - e.g. hylafax, spandsp, etc - but there are several email-to-fax solutions that come pretty close
16:19.54Greek-Boywell
16:19.57tzangercoppice: sounds like what I do to listen to radio stations on satellite
16:20.05Greek-BoyI just want to be able to send faxed and receive them from asterisk via e-mail
16:20.15tzangerthe receiver wants a video signal so I have to fake one by sending the same video frame over and over and precise intervals
16:20.42coppicedon't let Catch Curve catch you doing that :-)
16:20.54tzangercatch curve?
16:21.04coppiceJ2 by another name
16:21.10jameswf-homejbot: poo
16:21.10jbotpoo is smelly
16:21.26coppicejbot J2
16:21.55coppicehe doesn't know any of the interesting stuff
16:22.03jameswf-homejbot: f
16:22.03jbotACTION gives jameswf-home a big [kiss,hug]
16:22.10jameswf-homelol
16:22.33riddleboxis there a way to convert the voicemail email to ogg or something else besides wav? before it is emailed?
16:23.52coppicejbot J2 is the Devil's appointment reaper of FAX users.
16:23.53jbotcoppice: okay
16:25.28jameswf-homejbot: j2
16:25.28jbot[j2] j squared
16:25.44coppicejbot J2
16:25.45jbotj2 is probably j squared
16:25.59coppicejbot J2 is the Devil's appointment reaper of FAX users.
16:25.59jbot...but j2 is already something else...
16:26.17coppicejbot J2 is also the Devil's appointment reaper of FAX users.
16:26.17jbotcoppice: okay
16:26.37coppicejbot J2 is also -1
16:26.38jbotcoppice: okay
16:26.47coppicejbot J2
16:26.48jboti heard j2 is -1
16:26.57endrei heard u like mudkips
16:27.01coppicejbot J2
16:27.12coppicejbot J2
16:27.26endrejobd raid
16:27.29endrejbod
16:27.35coppicenow he's sulking
16:27.45jameswf-home<PROTECTED>
16:27.46jbotjameswf-home: okay
16:27.50jameswf-homejbot: j2
16:27.51jboti guess j2 is the Devil's appointment reaper of FAX users.
16:29.01coppicejbot Catch Curve is see leech
16:29.01jbotACTION lures Curve is see leech into the crab motel -- Curve is see leech checks in, but won't be checking out...
16:29.36jameswf-homejbot: wiki sangoma
16:30.19jameswf-homejbot: wiki bamf
16:30.32coppicejbot wiki j2
16:30.50jameswf-homejbot: bamf
16:30.51jbotextra, extra, read all about it, bamf is to disappear with a poof.
16:30.53*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
16:31.20coppicejbot wiki catch curve
16:31.31jameswf-homejbot: bamf is also bad a$$ mother fu**er
16:31.31jbotokay, jameswf-home
16:31.32De_Monwhat is th purpose of the Verbose app?
16:31.47tzangerDe_Mon: to display information only if verbose level is > some value
16:31.57riddleboxI guess I may as well upgrade my system this morning
16:32.00tzangerDe_Mon: think of a conditional noop
16:32.06De_Monyeah, but on the CLI I see the message regardless, do I have something else turned on thats show that to me?
16:32.29jameswf-homeadjust logger.conf
16:32.33*** join/#asterisk mamep (i=fallen@helios.edu.uoc.gr)
16:32.42De_Mondoes the verbose stuff goto a log file somewhere that the execution of the priority doesn't go to
16:32.53coppice100 pages in a FAX TIFF file
16:32.54coppiceYo ho ho, and a flavour of rum
16:32.59mamephello, i'm trying to connect to cisco callmanager using ooh323 but i can't find a guide how to do it..anyone can help?
16:33.18De_Monjameswf-home is it debug thats show the priority execution, or...?
16:33.21*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
16:33.22blitzragemamep: you'll have to ask specific questions
16:33.48mamepblitzrage : i need a little bit assistance with ooh323
16:34.02mamepis it possible to route my calls through cisco's callmanager?
16:34.16blitzragemamep: right -- but you should ask specific questions with examples of what is not working -- if you need someone to guide you through the whole install, that is more of a job for a consultant
16:34.32*** join/#asterisk kotyagin (n=knkbox@ppp85-140-239-38.pppoe.mtu-net.ru)
16:34.36mamepblitzrage : ok let me ask then..
16:34.53mamepfirst of all ooh323 module is loaded automatically in asterisk?
16:36.05disa-helphey guys..
16:36.07disa-helpshow queues
16:36.08disa-help(dynamic) (In use)
16:36.11disa-helpfor an extension
16:36.15disa-helpwhat does 'in use' mean?
16:36.35disa-helpi'm assuming it means that the extension is busy?
16:37.02kotyaginHi, all !!! Is there any programmers who able to discuss about VAD/CNG ?
16:38.01tzafrir_homejbot, tell kotyagin about ask
16:43.04nestArjbot is a hata
16:43.09mamephow can i add ooh323 peer with username and pass?
16:43.12kotyaginOk. I'll be more exact... Is asterisk really generates CNG frames with ast_rtp_sendcng function from rtp.c ???
16:43.14nestAr:)
16:48.52mamephow can i add ooh323 peer with username and pass??
16:50.20jameswf-home<butthead> hu hu he said ooh323 peer hu hu </butthead>
16:53.25jameswf-homejbot: ping
16:53.25jbotpong
16:54.08jameswf-homejbot: pong
16:54.09jbotPING!
16:54.20jameswf-homejbot: porn
16:54.21jbotPorn remains one of the largest problems with Open Source Software. Often causing development delays, flooded links and, in extreme cases, disabling programmers ability to type.
16:57.19riddleboxlol
16:57.59*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
17:00.26endrelol
17:06.09riddleboxwhen I make a call out of my TDM card, it takes like 5 seconds or so before the call starts to ring
17:06.23riddleboxis there a way to speed that up?
17:08.03[TK]D-Fenderriddlebox, Not really.  Don't forget, its pulling the line , dialing, and then waiting as well.  in effect you waited in dialing a number to #, and wait again as it send it out your analog line
17:08.23endreriddlebox: switch off echocancel training
17:08.32tzafrir_homeriddlebox, callerid settings, I guess
17:08.46endretzafrir_home: he said calling OUT
17:09.11tzafrir_homedialplan, I guess
17:09.23tzafrir_homeecho training is not something visible to Asterisk
17:09.34tzafrir_homeand not 5 seconds
17:10.57mamepWARNING[4451]: channel.c:3393 ast_channel_make_compatible: No path to translate from OOH323/ucnet-99f8(256) to SIP/51030-b6f01c20(4)
17:11.48mamepwhat's this?
17:13.41DarkRiftIs there a macro variable that returns the caller extension ?
17:14.01DarkRiftOr somewhere I can find help on the available macro's variable
17:15.50De_MonDarkRift your question relveals much about your lack of knowledge, seek the book and learn from it
17:15.58De_Mon~book
17:15.58jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
17:16.18DarkRiftthx
17:16.32*** join/#asterisk cesar_CR (n=cesar@201.195.35.62)
17:17.57*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
17:17.58[TK]D-Fendermamep, means you don't have any free G.729 licences and * can't transcode
17:18.11mamepyeah changed to gsm now
17:18.15mamepbut having some other problems
17:18.55mamephttp://pastebin.ca/778505
17:19.32jameswf-homesomeone should work on adding aix to gaim (pidgin)
17:20.00PaulAvilesanyone using 79xx phones?
17:20.30PaulAvileshow can you customize the buttons on the bottom?
17:20.51jameswf-homea/aix/iax
17:24.31*** join/#asterisk snazm (n=snazm@89.243.184.171)
17:25.14snazmHi folks
17:25.37[TK]D-Fendermamep, looks like all your sound files are missing or can't be accessed due to auth issues
17:27.44PaulAvilesall the files may be missing
17:27.59PaulAvilesare you ussing h323? if not don't load the driver
17:28.20snazmI'm new to VoIP but have been consuming information on Asterisk almost solid for the last few weeks (and wow, what a system!)  My boss has asked me to look at getting VoIP for our office (about 80 seats) and quotes from some companies are ridiculously high so thought I'd look at doing it myself, and I have a few questions the howto's can't answer :)
17:28.38PaulAvilesshoot..
17:30.23*** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net)
17:30.30snazmI think I can do it myself, but there's a few consideratios.  Firstly all the seats are going to get thin clients, and I'm wondering the best way to integrate VoIP with this.  Is it better to have PoE telephones using the CAT5e cabling and run it as a separate system, ose use USB VoIP phones from the thin clients themselves, or simply having a headset and using the thin clients audio in/out?
17:30.31jameswf-homethe cost of asterisk no matter how high usualy kicks the crap out of normal PBX's
17:31.00jameswf-homedepends on the thinclient
17:31.13snazmLinux based
17:31.16PaulAvilesyes, having a separe network for voip is always better and if need poe and can afford it even better
17:31.40[TK]D-FenderPoE costs very little depending how you do it.
17:31.50[TK]D-Fendersnazm, Where are you located?
17:32.00PaulAvilesa phone IMHO is better than any pc emulation
17:32.08jameswf-homethe term "giant collision domain" gets me excited
17:32.17mamephow can add ooh323 peer with user and pass?
17:32.24snazm[TK]D-Fender: UK
17:32.24[TK]D-FenderAnd soft-phone + usb phone = not much cheaper and works like shit
17:33.12[TK]D-Fendersnazm, Theneconomically speaking, look at linksys SPA-922/942's on Linksys/D-Link PoE switches.
17:33.17[TK]D-Fendersnazm, rather affordable
17:33.20PaulAvilessome soft phones are kinda nice, portsip is free and still has g729 support natively
17:33.44snazmThe benefit of having the thin client based phone (whether through direct audio or a USB deskphone) is that the users extention can follow their hotdesking
17:33.50snazm(should have mentioned hotdesking)
17:33.54jameswf-homeI know a school that uses linksys (due toi cost) they seem happy
17:34.21PaulAvilesor you can have a phone and log in /log out and have the same
17:34.25snazmThe benefit of having a separate PoE phone system is it will work even if the power goes out, with a single UPS
17:34.42PaulAvileswith a BIG.. ups..
17:34.52snazmPaulAviles: You mean login to the thin client AND the VoIP phone?
17:35.01jameswf-homewhat can brown do for you lol
17:35.05snazmPaulAviles: I was thinking if they were integrated, it could be automagic
17:35.22PaulAvilesno.. you can have say a 7940 and have multiple profiles depending on who is sitting using the phone
17:35.48jameswf-homecould also use agents
17:36.32snazmA 7940?
17:37.02jameswf-homeagents happen in asterisk reguardless of the channels type
17:37.05PaulAviles7940 or 7960
17:38.23snazmOK, so you're suggesting that I can configure the thin client to communicate with Asterisk to tell it to route numbers to the port matching the thin clients?
17:38.29snazmOr am I getting a bit lost here?
17:38.39QwellYou guys ever seen transcoding to/from gsm just sound horrid?
17:38.51Qwelllike...just...terrible
17:39.19PaulAvilesmamep, have you tried register=userID:pass@host
17:39.37PaulAvilesqwell, not really good quality
17:39.54QwellI mean, gsm doesn't sound great to begin with - but this is worse than lpc10
17:40.06PaulAvilesI mean, good quality.. sorry
17:40.19[TK]D-FenderQwell, Whats on each end?
17:40.21PaulAvilesi did some testing and had no issues
17:40.36Qwellcalling in on a PRI, playing a gsm prompt with Playback
17:40.53PaulAvilescan you emulate the same internal with 7777?
17:40.58Qwellswitching to ulaw or wav prompts sound great
17:40.59PaulAvileswith the same results?
17:41.05[TK]D-FenderQwell, EEK.  Ok, I recall only 1 version of * where something went horribly wrong with the GSM codec, but thats it
17:41.13PaulAvilesdo you have the proper files for gsm?
17:41.22QwellPaulAviles: it builds gsm
17:41.54Qwellit sounded like the gsm transcode is...horribly broken
17:42.28Qwellimagine the gain being set to around 50, and trying to play a file
17:42.33PaulAvilesso under /var/lib/asterisk/sounds/digits you have all the .gsm files too /
17:42.34Qwellthat's about how it sounds
17:42.34PaulAviles?
17:43.10unixdogwhat ver of gasterisk
17:43.18Qwell1.4.13
17:43.26unixdoghmmm
17:43.36unixdogbeen runing it with no problem
17:43.42[TK]D-FenderQwell, how does it sound on G.711 outside your PRI?
17:43.47unixdogand I have it on bsd
17:44.02Qwell[TK]D-Fender: they tested with softphones, and it was just as bad, I think
17:44.13Qwelleverything else on the PRI is fine - it's just gsm
17:44.13[TK]D-FenderQwell, ouch
17:44.27unixdogtry recompileing
17:44.41Qwelldid
17:44.45PaulAvilescan you create an external extension to test?
17:44.51[TK]D-FenderQwell, if it sucks on hardphones as well I'd say look at codecs.conf next
17:45.05Qwellcodecs.conf? O.o
17:45.19Qwellahh, nothing in there for gsm
17:45.21[TK]D-FenderQwell, And then check your base sound files... maybe someone globall sox-d them to death
17:45.29QwellI didn't know that conf existed :P
17:45.41unixdogwhat type of system is this on
17:45.45Qwellyeah, I reinstalled the prompts.  recorded prompts sound like ass too
17:45.49Qwellxeon
17:46.08Qwellno k6opts
17:46.12unixdogwhat flavor of *nix
17:46.18Qwelllinux, of course
17:46.24unixdogwhat flavor
17:46.28[TK]D-FenderSOUR :p
17:46.29Corydon76-digQwell: what version of gcc?
17:46.36Qwell4.2.3
17:46.40unixdogcent/suse
17:46.45Corydon76-digDowngrade to gcc 4.1
17:46.47DarkRiftHow can I permanently associate a mailbox witha SIP user, I mean when I call the VoicemailMain I want it to by default authenticate to his associated voicemail box, is that possible in a simple macro, or I need to build a SQL Database to make that possible ?
17:46.47Qwellwell...prerelease
17:47.05Corydon76-digThere's a performance regression in gcc 4.2
17:47.13DarkRiftThe voicemail box being a number, but the caller is a SIP user, which is not a number itself
17:47.18QwellCorydon76-dig: oh?
17:47.39Corydon76-digQwell: trust me
17:47.48Qwelland that could make gsm sound like that?
17:47.59Corydon76-digRecompile with gcc 4.1 and all will be well.  It's not just gsm
17:47.59[TK]D-FenderDarkRift, use SetVar in your sip.conf entry for use by Voicemailmain to know which box to use
17:48.04Qwellinteresting
17:48.59Qwellpoking him now
17:49.08DarkRiftWell, I'd need to use setvar for each users ? Let's say when you call 100 it calls me (darkrift), if darkrift call voicemailmain I want it to call the voicemail 100 automatically, using a setvar in each users entry would do that ?
17:49.27QwellCorydon76-dig: what else have you seen this affect?
17:49.46DarkRiftGot an example of what you mean ?
17:49.51Corydon76-digQwell: Asterisk is the only app I use where the performance is critical
17:50.00QwellI mean, besides gsm
17:50.31Corydon76-digI think someone said alaw sucked, as well
17:50.40Qwelldidn't try alaw - wav was alright though
17:50.55Qwellso, basically, transcoding in general
17:51.17Qwellis there a way to tell what version of gcc a module was built with?
17:51.35Corydon76-digstrings, maybe?
17:51.59Corydon76-digNope, that didn't work
17:54.43Qwelloh well - I think that answers another problem I heard the other day too
17:58.55DarkRiftAlright ! Thanks [TK]D-Fender
17:59.51[TK]D-FenderDarkRift, You're welcome
17:59.53[TK]D-FenderBBIAB
18:02.07*** join/#asterisk alephcom (n=chatzill@h66-112-187-16.mcsnet.ca)
18:10.40*** join/#asterisk selsky_ (n=selsky@12.sub-70-216-142.myvzw.com)
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18:12.10asdxhi
18:12.47asdxwhat is register => for
18:13.46mvanbaakin sip.conf ?
18:13.56asdxiax.conf
18:14.13mvanbaakto make your asterisk register to an IAX2 machine
18:14.26mvanbaakthe other side has: host = dynamic
18:14.35asdxis that the same in sip.conf too?
18:14.40mvanbaakyup
18:14.54asdxoh i see
18:15.03asdxso if i set host=dynamic in my [user]
18:15.17mvanbaakthe other box will need a 'register =
18:15.19asdxi will be able to connect to that user from my local client?
18:15.55mvanbaakuhhuh
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18:16.59mackeswhat irc clients are you all using?
18:17.02asdxirssi
18:17.21jameswf-homeKonversatioin and Xchat
18:17.30DarkRiftmIRC
18:17.50bobkareirssi, runs in screen
18:18.20mackesneat
18:18.24mackesMirc
18:18.32asdxthe host dynamic confuses me a bit
18:18.32mackesWindows based?
18:18.54jameswf-homescreem is overrated. people want to see what I am doing like it will help them understand...
18:18.56DarkRiftmIRC is windows yeah
18:19.01jameswf-home*screen
18:19.12mvanbaakmackes: irssi
18:19.33jameswf-homeI think xchat can be used in windoze not sure
18:19.44snazmPidgin rocks
18:19.51snazmWorks in windows too
18:19.52mvanbaakirssi can be used in winblows too
18:19.53mackesThats two votes for irssi.. How did you ding me mvanbaak
18:20.03asdxit is the same if i use register => foo:bar@foo.org and if i use [user] host=foo.org username=foo secret=bar ?
18:20.21mvanbaakmackes: simply by putting mackes: as the first word in my line
18:20.23mackesCool
18:20.26jameswf-homei need to get my oil changed
18:20.33mackesmvanbaak: Neat
18:20.35mvanbaakI need more beer
18:20.39bobkarejameswf-home: are you thinking about screen with multiple connected clients or something?
18:21.02jameswf-homelinux program screen,, crap
18:21.17jameswf-homewell not crap just useless
18:21.40mackesdigium has helped me using screen
18:21.48mackesits the only time I have needed it
18:21.56jameswf-homeand did you learn anything?
18:22.21mackesHmmmm I little because I was able to ask questions while they worked
18:22.22bobkareit's genius. lets me have programs running that I can get at no matter what computer i'm sitting at
18:22.28mackesand I could see what was happing
18:22.45mvanbaakjameswf-home: screen is convenient
18:22.50mackesthere is a VNC for terminal sessions now as well
18:23.01jameswf-homewell typicaly if you can learn from a screen session the tech is not fast enough
18:23.45mackeshahah
18:23.51mackesI ask they slow down
18:24.15mackesIf someone is in my corp network, I want them to tell me exactly what they are doing
18:24.29jameswf-homeUsusaly its like i will fix it you go buy a book I will leave comments
18:24.30mackesThey could be installing a rootkit
18:24.43mackesyeah
18:24.55mackesTech support is hard
18:25.00mackesI feel for those guys
18:25.13jameswf-homewell digium and us are not fonality we dont need rootkits. lol
18:25.46mackesI have only had trouble first setting up a 4 port PRI
18:25.56mackesBut once I had it, I was set
18:26.10mackesso I was very happy to have the free support
18:26.58jameswf-homeI was readintg the tb terms of support, they require you to install software and keep it up and if you shut it off your out of luck and $$$ verry windowsish
18:27.25rob0But what's useless about screen? For me it's essential.
18:27.59mvanbaakI think jameswf-home just likes to have tons of xterms open
18:28.05mackesWhich software is that?
18:28.15jameswf-homegood 4 you, some people cant live without webmin doesnt mean it belongs
18:28.24*** join/#asterisk Maan (n=Maan@155.48.255.23)
18:28.25mackesHey.. I do have a suggestion for Digium
18:28.36mackesWe use and love RPath Poundkey
18:28.51mackesBut I dont want the Web Interface in the newer versions
18:28.57asdxcan some of you please take a look at this: http://pastebin.com/m57964d71  <-- does this should be enough for making a call?
18:29.06asdxi hear tone when i dial test@default
18:29.12asdxbut my phone doesn't ring
18:29.32mackesIt would be great if they released Poundkey without the Web Interface but has the updates and 1.4
18:30.19jameswf-homepeople who cant administer a system without a gui should stick to M$
18:30.25rob0Comparing screen and Webmin ... wow.
18:30.27snazmIn Asterisk, is it possible to have it redirect a call if the end point goes dead without clearing the call down properly?
18:30.45jameswf-homesadly it is faster to admin a windows system through the cli as well
18:30.54mackesYep
18:31.18bobkareadmining windows without touching the command-line isn't possible
18:31.24mackesAnd the Web Interfaces really limit the admins to predefined setups
18:31.32jameswf-homescreen and webmin are both useless so comparisons can be made
18:31.47mackesWebmin is a Very Good tool.
18:31.58asdxanyone?
18:32.08mackesIf you have 20 Linux boxes ... It helps
18:32.34jameswf-homewebmin is good for a jr admin fresh out of college, but he shouldnt be making system changes so nix that
18:32.40bobkarejikes, I haven't seen anything break configurations so completly as webmin since that horrid config util for redhat 6.something
18:32.48mackesIt would really help this group if no one spoke in absolutes
18:33.12mackesman.. those are short sighted comments
18:33.32tzafrir_homeasdx, you need to have a secret / md5secret in the [voipjet entry as well. I suppose you have not pasted your real md5secret there
18:34.11jameswf-homeyou know what webmin does is it lowers wages by giving people who should be flipping burgers the power to do rermedial tasks...
18:34.12mackesJames, you must be an absolute genius. You can setup every server in linux direct from the command line from memory!
18:34.19asdxtzafrir_home: that account is free ;-)
18:35.05jameswf-homeI can complete most tasks from memory, and if not there are these neat things called books
18:35.08tzafrir_homemackes, there's another aspect: the ammount of data you need to pass over the phone to support
18:35.27mackesJames- I would love to know more about your background, where did you cut your teeth?
18:35.38jameswf-homepeople these days are too afraid to RTFM
18:35.45tzafrir_homeA phone line is a communication channel with a high rate of errors
18:36.10jameswf-homemackes: your question is to vague
18:36.11bobkareif you want a really cool utility for administering many *nix boxes have a look at cssh
18:36.12tzafrir_homeHence you need to provide there information with high redundancy and reduce the ammount of information passed there
18:36.35snazmSorry if mine was a stupid question but I'm a little baffled :$
18:36.50tzafrir_homeWith proper use of command-line you can minimize the ammount of data that needs to be passed
18:36.55asdxwhat i was wondering is: if i have register => username:password@somehost under [general], do i still need username/password entries in my [user] entry?
18:37.21jameswf-homemissed snazm's wuestion due to venting.. Im back :)
18:37.33tzafrir_homeOr better: get them to use an IM, and serve as a sort of remote terminal for you
18:38.20tzafrir_homeasdx, yes, you do
18:38.28snazmjameswf-home: lol :)
18:38.32snazmIn Asterisk, is it possible to have it redirect a call if the end point goes dead without clearing the call down properly?
18:38.59jameswf-homesnazm: probably with a creative dialplan
18:39.56snazmlol @ creative
18:40.02snazmYou mean much hackery?
18:40.12mackesJust so I am clear james, Can you configure Apache, Sendmail, Postfix, MySQL, et all from the command line, quickly, so much so that you think Webmin is a waste of time?
18:40.18jameswf-homemackes: the generic answer is I was born with it..... I popped out attached to a mainframe(my mom was pissed)
18:40.23mvanbaakmackes: I can
18:40.29jameswf-homeyes
18:40.46*** join/#asterisk alephcom_ (n=chatzill@h66-112-187-16.mcsnet.ca)
18:40.54jameswf-homeif its a mundane task I use a thumbdrive and a bash script
18:40.56*** join/#asterisk lemanal (n=lemanal@Paawc.gbis.com)
18:41.03mvanbaakI use vim
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18:41.15tzafrir_homemackes, yes. You go and read the docs
18:41.22jameswf-homesome people still write their own scripts
18:41.38mackeswait, wait.. I'm taking from scratch, not sitting down with Docs and scripting it first...
18:41.47mvanbaakfrom scratch
18:41.53jameswf-homeor steal and rewrite tzafrir's
18:42.03mvanbaakjust last week I did setup 2 new postfix boxen and 5 Mysql boxen
18:42.07mvanbaakI never use webmin
18:42.21mackesNew Install, open vi, and edit the confs, and start the service....
18:42.26mvanbaakyup
18:42.44tzafrir_homeactually apache used to be a nightmare, but now distros have tamed it quite nicely
18:42.46mackesWhat ever.. I know what you guys are saying, and its not apples to apples
18:43.10mackesyou spend a morning working on it, or you use a premade script for rollout.
18:43.11mvanbaaktzafrir_home: most distro's corrupt it
18:43.13tzafrir_homePostfix has always had saner defaults
18:43.15mackesThat is your webmin
18:43.26jameswf-homelol  mackes spend a few months purelly in the black window it wont seem so insane
18:43.26mvanbaakmackes: no, ssh != webmin
18:44.48tzafrir_homewebmin badly lacks scriptability
18:45.01mvanbaakand security
18:45.14jameswf-homewebmin is like a woman al nice and sweet but one wrong move you lose half your stuff
18:45.14tzafrir_homefor instance: how do you automate the simple task of allowing a certain IP to control it?
18:45.28tzafrir_homeOr allowing every IP?
18:45.33asdx[Nov 17 18:44:49] NOTICE[7886]: chan_iax2.c:7241 socket_process: Rejected connect attempt from 190.52.149.212, who was trying to reach 'test@'
18:45.34mvanbaaktzafrir_home: you open the conf in vi ;)
18:45.36asdxi get that now
18:45.51asdxbut i get AUTHENTICATED
18:45.55asdxwhen i connect
18:46.09*** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur)
18:46.46tzafrir_homemvanbaak, I did it once. But finding exactly the file to edit and what to put there is not trivial
18:46.55tzafrir_homenot to mention not documented anywhere
18:47.08mvanbaakI hate webmin
18:47.23mvanbaakit's a memory hog, it's insecure, and it's a mess without documentation
18:47.28jameswf-homewe have a consensus on webmin lol
18:47.37tzafrir_homeIf they designed it with a nice command-line interface to configure everything it might have helped
18:47.59jameswf-hometzafrir_home:  at that rate use ssh
18:48.04mvanbaaktzafrir_home: no. I dont like some setuid perl giant to be reachable on port 80
18:49.50jameswf-homeI wonder if i can get googles mobile platform to load on my blackberry without it crapping out and voiding my warranty
18:50.24jameswf-homemy boss would be pissed lol
18:50.50jameswf-homehe would say this is why you cant give geeks nice things
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18:53.39*** mode/#asterisk [+o mog] by ChanServ
18:56.07jameswf-homehttp://code.google.com/android/ << neat
18:56.25*** join/#asterisk gardo (n=gardo@121.97.110.119)
18:58.26jameswf-hometime to learn java
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19:00.07unixdogif you build it they will come
19:03.19jameswf-homeunixdog: the apperant answer to your question about getting rich on asterisk is build a hybrid hosted solution
19:03.37tzafrir_homejameswf-home, it smells too much like "almost free software"
19:03.49tzafrir_homeread their usage license carefully
19:04.00asdxthis is my configuration now: http://pastebin.com/m27b9f8a3 seems like i can authenticate but i get this: [Nov 17 19:02:25] NOTICE[8034]: chan_iax2.c:7241 socket_process: Rejected connect attempt from 190.52.149.212, who was trying to reach 'test@'
19:04.09tzafrir_homethe neo1973 has a nicer license and is closer to reality
19:05.11jameswf-homeyou should hear the prices to get your stuff added to trixbox pro
19:05.14jameswf-homeeek
19:05.15tzafrir_homeJust a small reality check: how can cellular providers really support an open handset device?
19:05.35jameswf-homethey can verry easily... they wont
19:06.13tzafrir_homeofficially some of them are behind this Android
19:06.26unixdoglol
19:06.32bobkarewhy on earth should cell providers care what software a users handset is running?
19:06.55jameswf-homebobkare: because they make a fortune by locking it down
19:07.44bobkarewhat country do you live in? (just so I know where not to move)
19:07.56jameswf-homeyou know howmuch extra "blackberry services" are, it doesnt cost them a penny but they charge for everything
19:08.30bobkareno such thing here in norway
19:08.43tzafrir_homebobkare, http://gizmo5.com/
19:09.08tzafrir_homeNot that I recommend them or anything.
19:09.37tzafrir_homeJust as an example of how this gives you a way to pay less to your mobile provider
19:09.43bobkarehere they sell you a sim card, or optionally lease you a phone as well (which is a real ripoff)
19:10.15tzafrir_homeBut a phone they don't approve will have a hard time connecting to the network
19:10.40unixdogI wish I could find a softphone for the blackberry
19:10.45tzafrir_homeThe neo1973 devices will find all sorts of difficulties connecting to some providers
19:10.46unixdog7520
19:10.57tzafrir_homelet alone getting support for connectivity problems
19:11.51*** join/#asterisk mcab (n=mb@mostly-harmless.ca)
19:11.55bobkareluckily that kind of behaviour will get them lots of unwanted attention from the government here
19:13.03jameswf-homethe us goverment is owned by big business.....
19:13.14*** join/#asterisk mcab (n=mb@mostly-harmless.ca)
19:14.07jameswf-homemicrosoft temporarily stopped paying their "protection money" and bam anti trust.. they are back on trac
19:14.59jameswf-homeMa Dell has started catching back up on payments and you notice that they are reforming ma bell
19:15.23jameswf-home2/Dell/Bell
19:15.25bobkaremy condolances to anyone living with a government like that
19:16.15*** part/#asterisk PaulAviles (n=salinas9@dsl-7-36.cofs.net)
19:16.26jameswf-homeits all good cause we can carry guns and kill our kids without much hassle
19:17.12jameswf-home7% of our bill of rights still applies in 4 states lol
19:18.35jameswf-homereally if you goy rid of the people america would be perfect
19:18.41jameswf-home*got
19:21.54*** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net)
19:23.17dlynesAnyone know how buffer overruns, framing errors, collisions, or carrier errors can affect voice quality?
19:24.05mamepsomeone can help me with ooh323 channel?
19:27.08rfxrhow do you prevent "Auto fallthrough"?  I tried setting Set(TIMEOUT(response)=7) but as soon as the Background(enter-ext-of-person), it falls through and hangs up :(
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19:27.50rfxras soon as it completes the playback that is
19:28.55tzafrir_homejbot, tell mamep about ask
19:29.22tzafrir_homerfxr, you provide a different dialplan
19:29.25mamephow can i add peer with username and password to ooh323 channel?
19:29.38tzafrir_homePlease pastebin the relevant dialplan context
19:31.05rfxrhttp://pastebin.com/m14b1bae3
19:31.40rfxrthis is from the docs except the timout which I found elsewhere trying it out
19:31.49dlynesrfxr: in your [general] section, set autofallthrough=no
19:31.55rfxrthanks
19:33.31rfxrawesome, thanks
19:33.35mamephttp://pastebin.ca/778608
19:33.39mamepcan someone help me with this..
19:33.44rfxrhope that makes it to the version 2 docs ;)
19:34.28dlynesrfxr: version 2 docs?
19:34.36dlynesrfxr: version 2 of what?
19:34.52asdxdamn i cant make my pstn phone ring
19:34.58rfxrthe AsteriskTFOT.pdf
19:35.16dlynestfot?
19:35.27rfxrthe future of telephoney
19:35.32dlynesoh
19:35.33rfxrphony :P
19:35.53dlynesrfxr: look in your sample extensions.conf file that comes with asterisk 1.4
19:35.59dlynesrfxr: it's documented in there
19:36.09rfxrI saw that there, but the docs say to start fresh
19:36.18rfxrand it was omitted
19:37.09mamepsomeone?
19:38.25dlynesmamep: well, for one, you don't have your ulaw sound files installed
19:38.43dlynesmamep: try installing them first, and then rerunning your issue
19:38.49mamephmm
19:38.52tzafrir_homemamep, I don't really know ooh323c, but I figure you better pastebin also the relevant parts of extensions.conf and of the ooh323c conf file (with passwords and such obfuscated, of course)
19:38.52mamepyeah beside that
19:38.56dlynesmamep: there'll be less spam then, and maybe the error will be more apparent
19:39.12tzafrir_homedlynes, by why ulaw?
19:39.15mameptzafrir_home : i don't know how to add username pass to peer in ooh323
19:39.18tzafrir_homeit can be any format, right?
19:39.26mamepdlynes : let me check
19:39.40tzafrir_homemamep, isn't there a sample ooh323c config file?
19:39.48dlynestzafrir_home: perhaps he's only got the g729 sound files installed and no g729 codec installed, or perhaps he doesn't have any sound files installed
19:39.55mamepyeah but i can't find user pass
19:41.14dlynesmamep: you also have a sip peer defined with a 'mailbox=' entry, but no corresponding entry in your voicemail.conf file
19:41.58dlynesmamep: but anyways, it seems the number you're calling doesn't exist
19:42.23mamepi mean h323 gateway is working?
19:42.45dlynesmamep: seems to be, but I would clear up the other spam, and then up your verbosity level
19:42.53mamepk
19:42.55mameptrying
19:42.55dlynesmamep: you might find that there's a dialplan bug
19:43.00mamep?
19:43.32dlynesmamep: i.e. you're trying to dial an extension from your phone, for which asterisk doesn't currently handle
19:43.56dlynesmamep: the reason I'm getting at that, is that you're executing the following lines from your dialplan:
19:44.05*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
19:44.15mamepwell i actually want to forward my calls through cisco's callmanager
19:44.21dlynesmamep: exten => _X.,n,Playback(the-number-u-dialed)
19:44.29dlynesmamep: exten => _X.,n,Playback(is-currently)
19:44.34dlynesmamep: exten => _X.,n,Playback(unavailable)
19:44.43*** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net)
19:45.18dlynesmamep: please note that the extension and priorities I've used do not necessarily match what is in your dialplan
19:45.28mamepyeah ok sure..
19:46.13rfxrmy playback doesn't quite play from the beginning and cuts off the first half second or so... is there a way to adjust that?
19:46.27mamepdlynes : http://pastebin.ca/778624
19:46.36mamepcheck my outgoing using ooh323
19:48.56dlynesrfxr: pastebin your entire extensions.conf file
19:49.13*** join/#asterisk CVirus (n=GoD@196.205.192.118)
19:50.51*** join/#asterisk Greek-Boy (n=Greek-Bo@41.221.58.2)
19:51.58rfxrdlynes, http://pastebin.com/d29bfb1b9
19:52.06dlynesmamep: try the following, and dump the output you get to pastebin:  http://pastebin.ca/778631
19:52.18mamepwhich one d3wayne ?
19:52.24mamepah
19:52.40mamepone sec
19:52.57dlynesrfxr: which line number is it where the playback is cutting off after the first half or so?
19:53.13dlynesrfxr: oh...nvm
19:53.14rfxrit cuts off the beginning
19:53.16dlynesrfxr: i see it
19:53.38rfxrit cuts off the beginning of line 13 background
19:54.04dlynesrfxr: try the following:  http://pastebin.com/m3a12e5
19:54.14mamepdlynes : http://pastebin.ca/778635
19:54.43mamepbtw i have the file in gsm format
19:54.48dlynesmamep: now, do a dialplan reload
19:54.55dlynesmamep: and then redo it, and repastebin it
19:55.06dlynesmamep: you forgot to reload your dialplan before trying the new dialplan
19:55.32dlynesmamep: either that, or you never updated your dialplan, or the dialplan code you gave me, is not where it's failing
19:55.49dlynesmamep: which file?
19:56.16mamephttp://pastebin.ca/778639
19:56.21mamepthe-number-
19:57.12rfxrdlynes, it doesn't cut of the end of the playback.  For example, "Please enter the extension of the person you are trying to reach" plays back as "...ter the extension..."
19:57.26dlynesrfxr: ah
19:57.28rfxrcutting off the beginning
19:57.35rfxrthe rest is fine
19:57.46dlynesrfxr: Right after the 'Answer()', add in exten => _XXXXXX.,n,Wait(1)
19:58.06dlynesrfxr: replace the _XXXXX. with whatever your extension pattern was
19:58.15rfxrok, thank you
19:58.18rfxragain ;)
19:58.24rfxrthis stuff is great!
19:58.27linageedoes anyone know why a voip phone won't work? :(
19:58.36dlyneslinagee: it's not plugged in?
19:58.41linageei can ping it, i haven't changed the config. they've rebooted it... i get the gui... wtf
19:58.41rfxr:)
19:58.44linageepolycom 320
19:58.52linagees/gui/web gui/
19:58.58linageeit worked before
19:59.16dlyneslinagee: I would start with a sip debug, if nothing is showing up in verbose mode
19:59.17linageeand i'm using the same config scripts. i'm like, wtf? (i tried rebooting mine too. works just fine)
19:59.24dlyneslinagee: also check your syslog for indications of dhcp issues
19:59.38linageedlynes: exactly. and nothing shows up in sip debug! hah. i looked at the polycom log file and it looks normal.
19:59.39dlyneslinagee: it might not be grabbing the latest tftp scripts
19:59.54alephcom_lol, I like the "not plugged in one".  We had a customer upset because their phone didn't work recently.  Umm, well, it was unplugged from the network.
19:59.54linageedlynes: i'm using ftp and it puts the logs up just fine
19:59.58dlyneslinagee: i'm guessing htat's how polycom does autoconfig, anyways
20:00.01rfxrdlynes, works like a charm now, thanks ;)
20:00.09linageedlynes: do you know if there's a way to do SIP to SIP calls with polycom?
20:00.29dlyneslinagee: no idea...I use aastra, not polycom
20:00.54dlyneslinagee: polycom's become quite anal lately...they'll only deal with you, if you're certified by one of their channel partners
20:00.59linageeusers at other remote sites work just fine
20:01.05linagees/users/family/
20:01.12mamepdlynes : did u check it?
20:01.13linageejbot: stop being annoying
20:01.13jbotACTION leaps to his feet and stops being annoying
20:01.49dlynesmamep: yeah...you'll need to pastebin the entire extensions.conf file
20:01.57mamepk
20:02.00linageedlynes: nothing at a sip debug is pretty weird when nothing has changed (that i know of)
20:02.07dlynesmamep: you're not showing me the correct area of your dialplan, or you're not adding my changes to your dialplan
20:02.11dlynesmamep: i'm not sure which it is
20:02.59dlyneslinagee: have you monitored your /var/log/xferlog to make sure the phone is grabbing the latest configs?
20:03.12linageedlynes: yes
20:03.18mamephere you go http://pastebin.ca/778647
20:03.27linageedlynes: is there a way to see what sip phones are currently registered? (in asterisk)
20:04.02dlyneslinagee: have you double checked the contents of those config files to see if someone hasn't inadvertently changed them on you, without your knowledge?
20:04.05dlyneslinagee: sip show users
20:04.30dlyneslinagee: also sip show peers will show you the qualifies, and whether they're dynamic or not
20:06.20linageedlynes: just checked them. did a diff on the regular <MAC HERE>.cfg and overrides/<MAC HERE>.cfg
20:06.36linageedlynes: both are the same on the first one, the overrides has only the extension as the difference.
20:06.48linagee(as it should be)
20:09.11linageedlynes: weird. all three are registered using sip show users....
20:09.18linagee(and i still go straight to voicemail)
20:10.09linageedlynes: here is the actual error: "Everyone is busy/congested at this time (1:0/0/1)"  (in logs)
20:10.13mamepdlynes : any chance?
20:11.33linageedlynes: wtf? sip show peers lists it differently.
20:11.38linageehostname = unknown
20:12.44linageewtf this is strange
20:13.14linagee(yes i have tried reloading asterisk. i have also tried restarting and stopping/starting asterisk)
20:16.45dlynesmamep: http://pastebin.ca/778655
20:17.07*** join/#asterisk CrashHD (n=crashhd@67-107-9-130.starstream.net)
20:17.12dlynesmamep: you'll find it's completely reworked...I eliminated all the guesswork by using a macro...it also compressed your dialplan down to half the size in the process
20:17.50dlyneslinagee: hostname=unknown means it hasn't registered, or it had an error registering
20:18.02dlyneslinagee: set core verbose=100
20:18.36dlyneslinagee: then reboot the phone in question, and monitor the registration in asterisk
20:18.36dlyneslinagee: you'll probably find the username and/or password doesn't match for the registration
20:18.55dlyneslinagee: double check by comparing your ftp config files to your sip.conf file for the phone
20:19.06dlyneslinagee: also do a sip reload to be on the safe side
20:19.37dlyneslinagee: perhaps some changes are made to the sip.conf file, that are not reflected in the running asterisk
20:19.59linageedlynes: set core verbose?
20:20.06mamepdeclined
20:20.06dlyneslinagee: yeah
20:20.10dlynesmamep: ?
20:20.24mamephttp://pastebin.ca/778660
20:20.26mamepcall declined
20:20.43dlynesmamep: oops...my mistake
20:20.49mamepwhat?
20:20.50linageedlynes: i've restarted and reloaded asterisk to heck already. and diffed the phone config files. :(
20:21.06linageedlynes: maybe i can  packet dump just from that phone ID and see if it's even attempting to register
20:21.21linagees / ID / IP
20:21.30linageehah! i outsmarted jbot
20:21.37dlynesmamep: replace the macro code with this:  http://pastebin.ca/778661
20:22.26*** join/#asterisk jetlagmk2 (i=jetlag@pool-70-106-89-254.hag.east.verizon.net)
20:22.47dlynesmamep: actually you can get rid of the EXTEN = ${EXTEN} part of the noop in the macro
20:22.51dlynesmamep: it's not useful anymore
20:23.01mamepk
20:23.04mamephttp://pastebin.ca/778664
20:25.16dlynesmamep: did you do a dialplan reload before trying the call again?
20:25.55*** join/#asterisk Toerkeium (i=oo@201.216.206.221)
20:26.20mamepyeah
20:26.58mamepfirst of all how can i get rid of message about sound files?
20:28.13dlynesmamep: Are you in Europe?
20:29.07dlynesmamep: do a make menuselect in your asterisk source directory, go to the section for sound files, and make sure you install all the sound files, all the extra sound files, and all the music on hold files
20:29.29mamepi have only in gsm format
20:29.31mamep[Nov 17 22:29:05] NOTICE[4682]: chan_sip.c:5335 process_sdp: No compatible codecs, not accepting this offer!
20:29.33mamepi get this one
20:29.36mamepif i use only gsm
20:30.13dlynesmamep: nod...just install all the sound files
20:30.44dlynesmamep: I suspect your issue is that you're not using autoload=yes in your modules.conf file, and that there's certain modules you're not loading, or loading in the wrong order
20:31.05mamephmm
20:31.30mamepinstalling sounds
20:31.50dlynesmamep: replace your modules.conf file with the following lines:
20:31.52dlynesmamep: [modules]
20:31.56dlynesmamep: autoload=yes
20:31.58dlynesand nothing else
20:32.07linageedlynes: wow that's strange. i had the remote person try to call and saw nothing on a packet dump from that IP. hah
20:32.18linageedlynes: but yet it fetches it's config. weird
20:32.37mamepdlynes : http://pastebin.ca/778670
20:32.40mamepmy modules.conf
20:32.42dlyneslinagee: perhaps the config is telling it that the sip proxy and sip registrar are something other than your asterisk server
20:33.11dlynesmamep: yeah, your modules.conf file is fine
20:33.22dlynesmamep: I suspect you don't have certain modules built
20:33.32mamepwhich ones?
20:33.34linageedlynes: it could be something weird like that, but all the other phones would be acting up. (i did a diff and rebooted the other phones. hrm)
20:33.50dlynesmamep: some of the format_... and codec_... modules
20:34.06mamepjust a sec let me install first sounds
20:34.06[TK]D-Fendergo prove is. "show modules like codec" <----
20:34.08[TK]D-Fenderit*
20:34.12dlyneslinagee: I don't know about the polycoms, but on the aastras
20:34.19[TK]D-FenderGeez... stop guessing and start SHOWING.
20:34.40linageedlynes: i was about to say maybe it's the cable provider, but i remembered this is over an encrypted VPN. :)
20:35.01mamephttp://pastebin.ca/778673
20:35.05dlyneslinagee: you can override the web settings with dialpad settings
20:35.19dlyneslinagee: get them to factory default the phone
20:35.32linageedlynes: on polycoms its pretty much the same way. when you do a dialpad setting, it gets uploaded to the overrides dir
20:35.35*** join/#asterisk Squeeb (n=squirt@87-194-8-66.bethere.co.uk)
20:35.39dlyneslinagee: also, does the polycom allow you to get a dump of the current config used by the phone?
20:35.41SqueebHello.
20:35.51SqueebDoes anybody know when the asterisk docs page will be back up?
20:35.53linageedlynes: i haven't heard of that if it does exist
20:35.57[TK]D-Fendermamep, Good, you have GSM, now pastebin your phone's entries, the CLI output of the failed call at verbose 10 and with channel debug enabled
20:36.01dlyneslinagee: you might want to do that, and compare it with your ftp config files
20:36.07[TK]D-Fenderdlynes, No
20:36.31dlynes[TK]D-Fender: ah..pretty crappy
20:36.53dlynes[TK]D-Fender: it really helps with debugging issues on the aastra phones...i'm sure it would help on the polycoms, too
20:37.02[TK]D-Fenderdlynes, only if you royally screwed stuff up.  Then again, anyone capable of that kind of damage won't be saved by an export either
20:37.24[TK]D-Fenderdlynes, Wouldn't need debugging if you didn't spend so much time BUGGING them :p
20:37.47dlynes[TK]D-Fender: well, if their phones worked properly, i wouldn't spend so much time bugging them
20:37.59[TK]D-FenderPEBKAC <-
20:38.04dlynespebkac?
20:38.12[TK]D-Fenderdlynes, feel free to point fingers at charis :)
20:38.14mamephttp://pastebin.ca/778678
20:38.18dlynescharis?
20:38.23[TK]D-Fenderchairs*
20:38.43[TK]D-Fenderdlynes, http://en.wikipedia.org/wiki/PEBKAC
20:39.23dlynes[TK]D-Fender: nothing to do with me, or my users
20:39.32dlynes[TK]D-Fender: their phones aren't terribly well tested
20:39.44[TK]D-Fendermamep,  ooh323_request - data ucnet format 0x8 (alaw) <--- why is this only asking for **ALAW**?
20:39.45dlynes[TK]D-Fender: i have one phone locking up like clockwork every 1/2 hour
20:39.54[TK]D-Fendermamep, fix your endpoint.
20:40.11mamepyou mean i need to change it to gsm?
20:40.22[TK]D-Fendermamep, means it'd better bloody well match
20:40.51[TK]D-Fendermamep, don't have it asking for apples when all you have is ORANGES
20:41.47*** join/#asterisk ghento (n=ghento@75.155.241.7)
20:43.08mamep?
20:46.07*** part/#asterisk deeperror (n=deeperro@d14-69-9-250.try.wideopenwest.com)
20:47.04mamepcdr.c:434 ast_cdr_free: CDR on channel 'OOH323/ucnet-73f9' not posted
20:47.24snazmAfter a bit of investigation in to the subject further and considering all the options, I've decided to go for a USB VoIP desk phone (with headset capabilities) and use Linux on the desktop to manage the VoIP connections and the phone hardware
20:48.01snazmDo people have preferences for the best type of USB desk phone to use and the best Linux software to enable it all to happen?
20:48.23[TK]D-Fendersnazm, I highly recommend against that...
20:49.26snazmI have considered all of the options though and it seems to be the most practical solution
20:49.28[TK]D-Fendersnazm, soft-phones are kludgy, low performing & usually low on features.  This is not something you'd use for a business and I couldn't even tell you a USB handset that would work for Linux for that purpose.
20:49.46*** join/#asterisk obnauticus (n=obnautic@c-71-236-181-11.hsd1.or.comcast.net)
20:49.51[TK]D-Fendersnazm, That'd mean you'd have to KNOW all the options.  What have you thought about so far?
20:50.23snazmWell all it would be is a phone-shaped microphone and speaker essentially, just like a normal headset, but with a keypad and hopefully a screen also
20:50.40snazmWhy is that so wrong or different if softphones through headsets have been used so successfully for so long?
20:51.31[TK]D-Fendersnazm, For one its a question of making the dialpad of any use.  What will TELL the soft-phone to use it?  And then the display... that too.
20:51.39obnauticusHow do I have multiple phones ring on 1 extension?
20:51.52[TK]D-FenderSoftphones should only be used by people like those on laptops
20:52.05[TK]D-Fenderobnauticus, "core show application dial"
20:52.26obnauticusI know that silly
20:52.29snazmI am hopefully going to find a softphone which will allow the keypad to be read just as if it were another input device (I've seen that mentioned a few times)
20:52.40obnauticusDo i put an other line with the same priority?
20:52.50snazm[TK]D-Fender: I don't think it would be too hard to tweak the softphone driver to pick this up and act on it
20:53.12[TK]D-Fendersnazm, Ok well if your set on your path, best of luck with that....
20:53.48[TK]D-Fenderobnauticus, No, its 1 line.  Read the instructions.
20:53.50snazm[TK]D-Fender: Well it seems to be the best option with regards on price sensitivity, features provided and upgradability
20:53.58obnauticusk
20:54.27[TK]D-Fendersnazm, Which soft-phone do you have in mind?
20:54.49*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)
20:54.50*** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca)
20:55.16snazm[TK]D-Fender: I don't yet, I am now looking in to this.  All I've decided so far is that according to all the criteria, it seems to be the best option to go for, that's why I was asking what the recommendations were
20:56.16[TK]D-Fendersnazm, You're expecting normal users to go through this for business purposes.  there will be a very noticable performance hit, lack of quality, redundancy, etc.
20:56.53[TK]D-Fendersnazm, And puts more things into the picture that can go wrong.
20:57.22snazm[TK]D-Fender: Performance hit where?  How can it be lack of quality also if the computer and softphone is doing exactly what a dedicated phone does anyway?
20:58.11*** join/#asterisk deeperror (n=deeperro@d14-69-9-250.try.wideopenwest.com)
20:58.13snazm[TK]D-Fender: I also think it simplifies things, a dedicated VoIP system (with expensive phones :S) would double the ethernet connections, more cabling, more switches and loads of other stuff
20:58.19[TK]D-Fendersnazm, soundcard quality and control sucks.  if they touch the mixer, etc.  USB handsets are usually junk quality speaker & mic themselves
20:58.42[TK]D-Fendersnazm, Or you just buy phones with a pass-through port
20:59.09snazm[TK]D-Fender: Well yes I do know this, I have a softphone handset which I've been playing around with, but it's a cheapo.  I'm sure that a professional desk phone would have a much superior audio quality, some have had good reviews
21:00.02obnauticus<PROTECTED>
21:00.05obnauticusErr, what does that mean
21:00.10[TK]D-Fendersnazm, well you've heard my recommendation. Hope you find a way to make it work and be worth the difference
21:00.17[TK]D-Fenderobnauticus, means a phone is forwarded
21:00.26obnauticusHuh?
21:00.33obnauticusOh I see
21:00.34obnauticusnevermind
21:00.38snazm[TK]D-Fender: Yes I know what you're recommending is the dogs bollocks of VoIP.  But I have to justify everything I do including cost
21:01.11obnauticuslol
21:01.14[TK]D-Fendersnazm, I don't know a manager out there who'd sacrifice so much for so little savings.
21:01.24obnauticusI got a lot of stuff done thanks to you [TK]D-Fender
21:01.28snazm[TK]D-Fender: You don't know many managers then :D
21:01.40snazm[TK]D-Fender: Don't get me wrong, I would love it
21:01.58snazm[TK]D-Fender: But unfortunately for 80 seats, the added costs soon add up :(
21:02.21[TK]D-Fendersnazm, I know plenty and have sold several projects.  People want a phone they can hold in their hands and feels natural.  Something of quality that is intuitive.
21:02.40[TK]D-Fendersnazm, Whats your budget for this?
21:02.54snazmThat's what I mean by a USB VoIP desk phone, it looks like a normal office phone with a USB plug
21:03.04[TK]D-Fendersnazm, And what USB phone & spftphone are you looking at?
21:03.17snazmrob0: Always :D
21:03.27snazmI've been looking at loads
21:03.40[TK]D-Fendersnazm, price?  and which soft-phone?
21:03.52snazmThere isn't a budget as such but some quotes have been in the region of £20k/$40k
21:04.09snazmThat was rejected quite spectacularly
21:04.24[TK]D-Fendersnazm, 40$k!? lol
21:04.32snazm:-o?
21:04.41[TK]D-Fendersnazm, Dunno where you're getting pricing..
21:04.49snazmThe suppliers :-|
21:04.59snazmThat's why I'm of the DIY mentatlity now
21:05.36Nukemizeris there a core command list, I am trying to fing  like {core:userlogon} but for acd login ?
21:05.46snazmI tell you what then, give me your recommendation of a good dedicated desk phone and I will look in to it again
21:06.00[TK]D-Fendersnazm, 80 x $100USD (Linksys SPA-941) = $8000 for phones.  How much is a kludgy USB setup going to cost you?
21:06.19snazm[TK]D-Fender: I am of course still very much open to ideas, but as I said, as far as I know, that would be the most cost effective and practical option
21:06.26[TK]D-FenderNukemizer, "show applications"
21:06.38Nukemizerthank you
21:07.21[TK]D-Fendersnazm, So at $8000 for phones, 1000$ for a PRI card, and $2000 for a decent RAID server, you're looking at $11,000, not $40k
21:08.23snazmOK you're making it sound quite attractive, but the only thing is a softphone approach would still be cheaper than $11l
21:08.49snazmAnd that still seems like quite a bit for what it is (there is already a working POTS PABX in place, so replacing it needs to be as justifiable as possible)
21:09.11[TK]D-Fendersnazm, if you have a pots PBX, then just get FXS gateways!
21:09.58[TK]D-Fendersnazm, and reuse your phones
21:10.08snazmIt's a crap PABX, it's even stoped working - some users can't even transfer calls!
21:10.11[TK]D-Fendersnazm, Still much better than soft--phones.
21:10.30snazmIt's also not very flexible at all, has no auto-login or follow-me features
21:10.35[TK]D-Fendersnazm, if its POTS (for the stations), then its the PBX's fault, not the phones.
21:10.42snazmI know
21:10.51snazmThe whole system is flaky
21:10.59[TK]D-Fendersnazm, So keep the phones & wiring, and jsut dropin in * in place o fthe PBX itself
21:11.02snazmLooks very old and worn
21:11.10snazmAll the wiring is being ripped out too lol
21:11.28snazmTo explain, the building was build in the 1800's and the current wiring was done about 20 years ago
21:11.35snazmNo Cat5e also
21:12.53snazmThis whole project is very big, and even a slight saving per seat on VoIP would reduce the total cost of it by a lot.  Sorry for being more sensitive to cost than technical excellence (although I wouldn't like to be of course), but I have to present options and reasons with the prices before anything is agreed
21:13.39snazmThe SPA941 looks nice though
21:13.42[TK]D-Fendersnazm, Well it sounds like you won't be PRESENTING anything but what you find as the lowest costs.
21:14.09[TK]D-Fendersnazm, and that will cost you for a usage & support performace hit.
21:14.29snazmI will offer a few options, but judging by the response to the previous quotes, I don't want it thrown back in my face
21:14.37[TK]D-Fendersnazm, You will spend a asignificant amount of time trying to tweak your plan into something only second-rate
21:15.14[TK]D-Fendersnazm, I jsut showed you $11K ($8K of phones).  That already undercuts the other buy 2/3
21:15.31snazmWell the price did include setup and consultancy
21:15.37snazmWhich still needs to happen
21:16.18snazmI'm still not an expert by any means, but I feel as though I could manage a lot of what they wanted, and maybe with one expert on board for a week we could get it done which would be cheaper
21:16.30[TK]D-Fendersnazm, Keep your old analog phones and you can drop that amout from $8000 to $2100
21:16.48*** join/#asterisk dlynes_home (n=dlynes@d154-20-9-152.bchsia.telus.net)
21:16.59[TK]D-Fendersnazm, using 10x SPA-8000
21:17.09snazmNone of the POTS wiring will remain though, it's a mega mess (there are even exposed live mains wires in places lol)
21:17.23snazmThe whole thing needs to be ripped out and done properly and safely
21:17.53snazmSPA 8000
21:18.12Qwell8000... 8 port?
21:19.04snazmHow easy would it be to implement follow-me and auto-logon on analogue POTS phones with this device?
21:20.34snazmI don't even know how that would work TBH
21:21.41[TK]D-FenderQwell, yup, 8 ports at $200. almost impossible to beat
21:22.08snazmIt really effs me off when you order diet coke and they bring you normal coke 8-)
21:22.17snazmHow hard is it for the freaks to understand it's not the same thing
21:23.45De_Monsnazm the diet stuff is actually worse for you dontcha know
21:23.56snazmAspartame ftw :D
21:24.06snazmI know but I'm low-carbing at the moment
21:24.13*** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
21:24.17*** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net)
21:24.19Qwellwater > diet anything
21:24.42snazmNot in the UK lol, in my home county they add fluoride to everything :S
21:24.52Qwelland?
21:24.58*** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com)
21:25.05snazmDo you know how nasty fluoride is? :-O
21:25.12Qwellfluoride is...tasteless
21:25.26snazmIt's also a neurotoxin that doesn't get flushed out of your system, you accumulate it for life
21:25.38snazmAnd it's more poisonous than a lot of other chemicals including lead and arsenic
21:25.53Qwellbesides - you think they don't use the same water in coke that you drink?
21:26.10snazmI doubt it, most of this coke comes from abroad lol
21:26.18lirakismy asterisk server "crashed" this morning.  I rebooted it and now i can not get it to start. 'asterisk -c' shows 'ERROR[13931]: chan_zap.c:7053 mkintf: Unable to open channel 1: No such device or address'
21:27.18*** join/#asterisk |dennis| (n=dennis@200.32.236.10)
21:27.31lirakishere is a full pastebin of the asterisk -c http://pastebin.ca/778728
21:27.51lirakis<10 lines FYI
21:29.11lirakisi have a sangoma A101 card
21:33.41asdxi registered two users and it has too much noise/echo when i call the other user
21:33.57asdx300 ms of latency
21:34.07asdxgsm codec
21:35.15*** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60)
21:41.07[TK]D-Fenderlirakis, if it bombs on your first channnel odd are zaptel didn't load.  Either for not being called, or for wanpipe not having been ready
21:42.39*** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
21:43.16lirakis[TK]D-Fender: ive been working with the modules... i have manually modprobed zaptel.. and still got nothing.  I backed up zapata.conf and wiped it out to a empty file, then mod probed ztdummy and asterisk started.
21:43.54[TK]D-Fenderlirakis, you aren't supposed to be using ztdummy.
21:44.08[TK]D-Fenderlirakis, and go verify wanpipe.  Then do "ztcfg -vvvv"
21:44.09lirakis[TK]D-Fender: honestly.. im unsure of what wanpipe is... i didnt setup the card on this box.  It seems odd.. that when i do lspci .. it shows up as a "Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card"
21:44.25[TK]D-Fenderlirakis, then you have NO slue what the hell you're doing ! :p
21:44.37lirakis[TK]D-Fender:yeah i know.. (about ztdummy) i just did it to make sure asterisk would start.
21:44.40[TK]D-Fenderlirakis, Wanpipe is Sangoma's drive which needs to be loaded first
21:45.21*** join/#asterisk dlynes_ (n=dlynes@d154-20-9-152.bchsia.telus.net)
21:45.25lirakis[TK]D-Fender: okay.. i wasnt sure if it enabled some "special features" or some thing.. or if it was needed.. i tried to do a quick look on their website.. but it wasnt immediately apparent.
21:45.33lirakis[TK]D-Fender: i will check out wanpipe
21:45.53[TK]D-Fenderlirakis, "wanrouter status" <- PB it
21:49.01lirakis<PROTECTED>
21:49.16[TK]D-Fenderlirakis, yup, that'd kill it
21:49.55[TK]D-Fenderlirakis, when that happens on my server I lose internet as I use an S518 for ADSL :)
21:51.00lirakis<PROTECTED>
21:51.18*** part/#asterisk PepOSX (n=pepOSX@190.72.149.231)
21:51.19[TK]D-Fenderlirakis, saves me a lot of wiring though... I love it
21:51.51lirakis[TK]D-Fender: im sure.. and one less appliance in the network
21:53.43[TK]D-Fenderlirakis, 1 ls appliance, 1 less power brick.
21:56.29lirakis[TK]D-Fender: okay .. its up now with a recompile and reloading of the driver
21:56.38lirakis[TK]D-Fender: thanks.. i
21:56.41lirakis* -i
21:59.24[TK]D-Fenderlirakis, np
22:00.14*** join/#asterisk anthm (n=anthm@adsl-70-226-55-121.dsl.milwwi.ameritech.net)
22:00.14*** mode/#asterisk [+o anthm] by ChanServ
22:00.25*** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net)
22:07.21*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
22:24.35*** join/#asterisk cyberpass2 (n=mataz@ppp-64-219-79-16.dsl.hstntx.swbell.net)
22:24.49cyberpass2which modems can you reprogram to make them into phone adapters?
22:24.57*** join/#asterisk jozu (n=torrent@84.120.184.91.dyn.user.ono.com)
22:25.02jozuhello to all
22:25.08BBHosscyberpass2: none of them
22:25.37cyberpass2<BBHoss> hrmm..r u sure?
22:26.04cyberpass2<BBHoss> can you suggest a cheap phone adapter that works well with asterisk?
22:26.08BBHosscyberpass2: there may be some out there but you don't want to use them
22:26.27BBHosscyberpass2: what kind of ports do you want
22:26.35BBHossfxo, fxs or both?
22:26.56cyberpass2regular US phone jack...whats the diff between fxs and fxo?
22:27.10BBHossheh
22:27.31BBHosshttp://www.patton.com/technotes/fxs_fxo.pdf read that and come back
22:27.50BBHossor ~book
22:27.56BBHoss~book
22:27.57jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
22:30.28cyberpass2oh ok
22:31.05cyberpass2well..what I want is to have multiple FXS's so I can have a number of phone lines to one box
22:31.19cyberpass2maybe one FXO just to test
22:31.36cyberpass2or do i have it backwards?
22:32.23BBHossyou need an FXO to connect to the phone company
22:32.35BBHossand you need an fxs to connect to one of your phones
22:32.37*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
22:32.49BBHossor you can ditch FXS and go for IP Phones
22:33.08BBHossor SIP/IAX2 SoftPhones, which you install on your computer
22:33.33cyberpass2well i want to set up my voip service
22:33.50BBHosswhat kind of voip service?
22:34.25BBHossare you paying for a voip service from an ITSP
22:34.25cyberpass2so have multiple internet users connect to the box using their own voip gateways...
22:34.34cyberpass2no im not...i dont intend to
22:35.05cyberpass2then have multiple phone lines(POTS) connected to my asterisk box
22:35.13BBHossso you want other people to be able to connect their asterisk boxes to you, or do you just want them to be able to link their phones to you
22:35.33cyberpass2i want them to connect to me...
22:35.49BBHoss1st or second?
22:35.50cyberpass2i was going to give them those cheap linksys PAP2s
22:36.26cyberpass2i want them to connect to my askterisk box which is connected to my telephone company using regular phone lines
22:36.54cyberpass2so, they could make calls
22:37.07cyberpass2maybe not recieve them...but make outgoing calls for sure
22:37.12BBHossyeah you can use PAP2
22:37.18BBHossthey could make and recieve
22:37.27BBHossi would use the digium IAXy though
22:37.31BBHossit works better with NAT
22:37.35[TK]D-Fenderiaxy yuck....
22:38.01BBHossbut a pap2 will work too
22:38.22cyberpass2can i set asterisk so that any incoming calls from the POTS phone lines get automatically disconnected?
22:38.33BBHossyeah
22:38.42BBHossor you could just never answer them
22:38.51rob0:) Not plug in the FXO.
22:39.09cyberpass2<BBHoss> but if i never answer them, the phone lines wont be availble for outgoing calls
22:39.20BBHosscyberpass2: yeah they will
22:39.38cyberpass2damn...i really gotta read up more on this...
22:39.40BBHosscyberpass2: only the incoming calls wouldnt be answered
22:39.41BBHossyeah
22:39.43BBHoss~book
22:39.44jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
22:39.50rob0I use a menu for that.
22:39.51BBHossFREE!!!^^^^^^^^^^^^^^^^^^^^
22:40.35cyberpass2so..i could still set up a call, even if there is an incoming call on that line?
22:40.46BBHossno
22:40.52BBHossi see what youre saying now
22:41.06BBHossif someone called it and never hung up
22:41.24cyberpass2anyways...i can call up POTS provider and tell them to make it outgoing phone line only
22:41.31cyberpass2hopefully they will agree
22:41.33rob0there's a thought
22:41.34BBHossyeah
22:41.46BBHossits called a "something" terminate
22:42.18cyberpass2now...when all this is set up and running, would asterisk also allow for internal phone to phone routing?
22:42.37rob0if you set it up that way :)
22:42.37cyberpass2ie i give each one an internal number so they can call each other
22:42.41BBHossYES READ THE BOOK
22:42.56BBHossITS A FREE DOWNLOAD
22:42.59cyberpass2i wil i wil
22:43.12BBHossyou can do anything with asterisk
22:43.21De_Moncyberpass2 all your wet dreams will come tru, just read the book and come back if you still have questions
22:43.41De_Monmy * box orders pizza automatically I just dial 2, its really cool...
22:44.04cyberpass2so for what ive describe...how much am i looking for interms of hardware for the FXO/FXS?
22:44.09BBHosswhat do you call the company and give them a voip menu?
22:44.33BBHosswell how many phone lines from the telco to you want?
22:44.40cyberpass2<De_Mon> 3
22:45.00cyberpass2<De_Mon> do you automatically play a sound file when the pizza store answers?
22:45.12BBHossso you need 3 lines?
22:45.12cyberpass2<BBHoss> 3
22:45.14De_Monno, it waits 5 seconds and places the order, its always the same so :)
22:45.15cyberpass2yea
22:45.25cyberpass2oh ok
22:45.39De_Monthen it adds a if you have any questions press 0 for operator and will call me :)
22:46.02BBHosscyberpass2: i would get a TDM400p with 3 FXO modules
22:46.06De_MonI like smiley faces :)
22:46.33*** join/#asterisk weazahl (n=revwease@adsl-68-93-176-137.dsl.ksc2mo.swbell.net)
22:47.08*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)
22:47.24jozui have and error installing 1.4.14, chan_iax2.o error
22:47.26jozu???
22:47.40De_Monthe actuall error will help
22:47.47cyberpass2<De_Mon> thats sick!...so you pick up ur phone, dial 2, shut ur phone...then asterisk calls the pizza store, dials in the DTMF codes for your pizza and play a sound file after?
22:48.47De_Monno.. it plays a recording of me saying "Hi id like 2 large peparonie pizzas <dramatic pause> that will be all. <long pause> If you have any questions press 0
22:49.22De_Monoiy I butchered pepperoni that time
22:49.32De_Moncrap I forgot what I was doing!
22:51.19weazahlok, i have 2 boxes that use vitelity for in/out trunks.  one has been running for almost a year without reboot and no problems.  the other, on a regualar basis will be able to accept calls but not be able to make outbounds until i restart asterisk. it restarts at 3am daily.  any ideas on why?
22:51.46De_Monweazahl are you restarting asterisk at 3am daily?
22:52.02De_Monweazahl lets see the actual errors when outbound calls fail instead of us guessing
22:52.15weazahlwell, for a while it was just asterisk.  now i reboot since it dint help
22:52.30De_Monsounds like your hardware is hosed
22:53.05weazahlinfortunatly i dont have any right now.  laptop got wiped clean and havent had it happen in a few days
22:53.54De_Monmaybe you should come back when you actually have some useful info
22:54.00weazahlits a new poweredge core duo, 2 gigs of ram (overkill i know) and any memtest or other test i throw at it works fine
22:54.08jozusomeone get an error installing 1.4.14?
22:54.24De_Monjozu you did, duno what it was though
22:54.36De_Monweazahl telephony hardware
22:54.58weazahlDe_Mon: no analouge all VOIP
22:55.05De_Monyou know.. the things you are trying to call out on and cant?
22:55.36De_Monweazahl okay... there is more than one VOIP protocol supported by asterisk do they both stop working?
22:55.36weazahllike i stated, vitelity.net no local hardware
22:56.16weazahlDe_Mon: thank you, i will switch it to IAX in my window tonight
22:56.25weazahli will see what happens then
22:56.40De_Monif you can actually get an ERROR we could tell you why its happening (the error probably does that already, but hey)
22:58.09weazahlthe sip debug is hard to use with 32 stations chattering constanly.  and it is infreuqent.  just a pain.
22:58.36De_Monso, it stops some outgoing calls not all of them?
22:59.04weazahlonly calls going out on that trunk.
22:59.16De_Monbut all calls going out on that trunk, not "some"
22:59.24weazahlhang on a minute. i think the logs havent rotated out yet.
22:59.31weazahlcorrect all
23:00.29De_MonIts hard to believe restarting asterisk doesn't solve problems with a SIP/IAX provider
23:00.40De_Monand that rebooting the box does
23:04.00weazahlrestarting does solve it.  the problem is, i restart at 3am. and sometimes the problem apears  at 4am, sometimes at 22pm
23:04.24weazahl11pm rather
23:05.05weazahlreal problem when i get woke up at 5
23:05.16De_Monif restartign asterisk fixes the problem, why are you rebooting the box?
23:05.37De_MonSVM support vector machines? Never heard of one of those before
23:06.30*** part/#asterisk deeperror (n=deeperro@d14-69-9-250.try.wideopenwest.com)
23:06.34weazahlaccually no.  and the reboot is really just something i was trying since a 3am restart didnt really help anything
23:07.10weazahlit is a RANDOM and INTERMITTENT problem
23:07.45weazahllog is taking a while to load. 600+mb
23:17.09weazahlDe_Mon:  Nov 14 09:38:21 NOTICE[5667] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
23:17.13*** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net)
23:17.40weazahlit is pingable at that time though, and is still registered and accepting inbound
23:21.43[TK]D-Fenderweazahl, that measn nothing.
23:22.17[TK]D-Fenderweazahl, next time enable SIP DEBUG, pand pastebin the full output.  Also describe in detail the networking path between * and your provider
23:27.37weazahl[TK]D-Fender:  ok.  i'll do so.  network path is quite simple, Static DSL to KC MO, Dallas TX, Denver CO terminating at coloc on XO.NET
23:27.59weazahlsame as another machine 3 blocks away that NEVER has a problem
23:29.42weazahli have replaced the DSL router because the forst one cooked in the summer while there was no AC during renovations
23:29.45[TK]D-Fenderweazahl, So you * server has a public fixed IP?
23:30.10weazahlsure does.  is there any other way? :D
23:30.22*** join/#asterisk yannj_fr (n=yannj_fr@tvn95-3-82-237-158-147.fbx.proxad.net)
23:30.31[TK]D-Fenderweazahl, Sure... behind behind a NAT router with a dynamic ip, duh.....
23:31.10weazahli know that.  but i WAY perfer the static route.
23:31.23weazahlmakes maintnence much easier
23:32.39*** join/#asterisk _matt (i=matt@2001:770:168:1:20b:cdff:fe04:843a)
23:33.06*** join/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net)
23:33.26katsuodoeveryone hello
23:34.28katsuodoDoes anyone know how many calls does dsl support using sip and asterisk
23:34.48[TK]D-Fenderkatsuodo, Depends on what codec, and how much bandwidth
23:35.21katsuodoHalo [Tk]D-Fender
23:37.40rob042
23:38.01[TK]D-Fender6 * 7?
23:38.27weazahlwith 6016/768 and ulaw, at 5 you will have signifigant jitter.  with GSM, dont know cant get that many channels going.  i figure about 20 though
23:38.38rob0The answer!
23:39.10weazahlrob0:  took me a second but, what is the question to the answer
23:40.17Corydon76-digHow many roads must a man walk down before they call him a man?
23:40.35weazahlNULL SET
23:40.44[TK]D-FenderCorydon76-dig, Just one.... the one with the red light... she'll make a man outta you ;)
23:41.10Corydon76-dig[TK]D-Fender: no, it's rumored that is the question, to which the answer is 42
23:44.42katsuodo[TK]D-Fender G.729 not sure of bandwidth?  Is dsl idea for office?
23:45.40[TK]D-Fenderkatsuodo, you don't know what your bandwidth is, and you didn't describe your needs including any suitabe description of your "office".  Just how psychic do you think we are?
23:46.14katsuodopardon
23:48.56katsuodotwo company, one asterisk pbx, (4) polycom 301 ip phone, (4) RCA digital phone, one user needs the extension to connect to other extension in other company, (8) pots only use (5), of (5) one for fax, dsl (is all I am told), no have bandwidth number yet
23:49.45katsuodoSix extension in both company total 12 extension
23:52.05[TK]D-Fenderkatsuodo, what are you planning on doing over the internet?
23:53.28katsuodothere was discussion of iax to other office
23:53.58katsuodoso really three office.  other office (3) ip phones connect to asterisk
23:54.15[TK]D-Fenderkatsuodo, you made it sound like there were 2 companies sharing 1 PBX localluy.  Perhaps you should split your description up and try again
23:55.01*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)
23:56.09katsuodoyes, (2) company in one office sharing asterisk; other company different location with asterisk pbx connect iax (maybe!) no not yet decide
23:56.32*** join/#asterisk jero (n=jerome@modemcable169.212-70-69.mc.videotron.ca)
23:56.36jerohi all
23:56.42[TK]D-Fenderkatsuodo, how many channels do you want to have between the 2 systems?
23:56.44katsuodoone office two company internet no important for phone
23:57.04katsuodounless connect to other asterisk pbx in other office
23:57.36*** join/#asterisk asdx (n=diego@adsl-149-212.click.com.py)
23:57.40asdxhi
23:57.41katsuodoin one office two company two channels, no?
23:57.44jeroI'm experiencing problems with asterisk 1.4 and polycom 601 expansion module: can't get the module to switch its light to ON for outgoing calls (asterisk does not notifies an event when a phone makes an outgoing call)
23:57.44[TK]D-Fenderkatsuodo, thats what I'm talking about.  how many simultaneous calls do you figure you'll pass between these 2 boxes if your expansion happens?
23:57.55asdxi have a little echo in my pure voip calls
23:58.05asdxhow can i remove echo
23:58.08asdxjitterbuffer?
23:58.11[TK]D-Fenderasdx, what hardware?
23:58.29*** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net)
23:58.29asdx[TK]D-Fender: asterisk is running on a vps
23:58.37[TK]D-Fenderasdx, I mean the ENDPOINTS
23:58.47katsuodoguess maybe 20 in whole day?
23:58.52katsuodoif expansion
23:59.01[TK]D-Fenderkatsuodo, I'm talking simultaneous.
23:59.27asdx[TK]D-Fender: this is pure voip, no special hardware
23:59.31katsuodoinbound / outbound you speak of, yes?
23:59.46[TK]D-Fenderasdx, what kind of PHONES........ what is on each end of the conversation?!
23:59.58asdx[TK]D-Fender: softphones (zoiper)

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