irclog2html for #asterisk on 20060606

00:07.54*** join/#asterisk bjohnson (n=bjohnson@i216-58-91-250.cybersurf.com)
00:12.55*** join/#asterisk Gabriel25 (n=whatever@user-12ld5f7.cable.mindspring.com)
00:13.57*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
00:16.35djacob-wcgshello all, how are you , i have a question, im trying to send calls from eyebeam sip phone --->sip ---> asterisk--->sip --->cisco 5400---isdn -->pstn and then i call out the pstn to a pbx , it does not detect dtmf tones
00:16.50*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
00:16.56djacob-wcgscan anyone point me in the right direction for some reading about this
00:17.15djacob-wcgsor have any ideas, im just learning all this
00:17.18djacob-wcgsthanks in advance :)
00:17.20*** join/#asterisk zotz (n=zotz@24.244.133.115)
00:18.17*** join/#asterisk denon (i=denon@synapse.subneural.net)
00:18.17*** mode/#asterisk [+o denon] by ChanServ
00:19.13djacob-wcgsmy codec is g7.11 also
00:20.41Gabriel25guys IF I have a regular IP phone .... no sip
00:20.54Gabriel25how I can register with my asterisk box ?
00:21.14Gabriel25I have avaya 4624 ip phone
00:21.48Gabriel25R1.8.3 H.323 release for the 4606/4612/4624 I update with this version
00:22.46Gabriel25How i can make this IP phone working with my asterisk box ?
00:23.54*** join/#asterisk fiXXXerMet (n=Kyle@cmu-24-35-80-91.mivlmd.cablespeed.com)
00:25.18[TK]D-FenderGabriel25 : Did you setup H.323 on *?
00:25.52Gabriel25I don`t think so
00:26.10Gabriel25How I can check that
00:28.00*** part/#asterisk Lord_Drachenblut (n=Lord@12.210.100.18)
00:28.04[TK]D-FenderGabriel25 : Well that phone runs on H.323... I think you'd better do a whole lot more reading first.... the phone isn't the problem yet.
00:28.31Gabriel25I know I have to read a lot but I need a start !
00:28.39*** part/#asterisk fiXXXerMet (n=Kyle@cmu-24-35-80-91.mivlmd.cablespeed.com)
00:28.54Gabriel25I don`t know if I asterisk support H.323
00:29.14Gabriel25or if are exist ..... a sip firmware for this phone
00:29.51*** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net)
00:29.59Gabriel25http://support.avaya.com/japple/css/japple?temp.documentID=283920&temp.productID=107755&temp.bucketID=108025&PAGE=Document
00:30.04Gabriel25I read here
00:30.09[TK]D-FenderGabriel25 : Go read up on how to add H.323 to * first.    Right after confirming if you can flash taht phone to SIP.
00:31.09Gabriel25Ok :)
00:31.21Gabriel25let me have some readling and then I`ll be back
00:31.51jeffpc[TK]D-Fender: alright, I loaded the ztdummy module
00:34.33jeffpcI assume I need to load chan_zap.so?
00:34.37[TK]D-Fenderjeffpc : good... ready to go then
00:34.45Gabriel25[TK]D-Fender can you give me a hint ? where I can add H.323
00:34.55[TK]D-Fenderjeffpc : yup
00:35.24jeffpc[TK]D-Fender: I added load => chan_zap.so to modules.conf..
00:35.24[TK]D-FenderGabriel25 : Your hint.... www.voip-info.org
00:35.28jeffpc<PROTECTED>
00:35.28jeffpcJun  5 20:33:19 WARNING[11272]: loader.c:499 load_modules: Loading module chan_zap.so failed!
00:35.59[TK]D-Fenderjeffpc : wipe out your modules folder completely and redo "make install" for zaptel & asterisk
00:36.12jeffpc[TK]D-Fender: this is debian package
00:36.13opus_jeffpc, run ztcfg -vvv
00:36.20[TK]D-Fenderjeffpc : And I really hope you redid zaptel FIRST, and then recompiled asterisk afterwards...
00:36.25jeffpcduh
00:36.29jeffpcno /etc/zaptel.conf
00:36.34[TK]D-Fenderjeffpc : SCrew packaging.. it'll only lead to trouble
00:36.42jeffpc[TK]D-Fender: hehe
00:37.13*** join/#asterisk denon (i=denon@synapse.subneural.net)
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00:37.37opus_jeffpc, there isn't a stable enough version of asterisk to package yet
00:37.57opus_if you look at the code you will be shitfaced:)
00:38.10jeffpcopus_: I heard bad things about the code..
00:38.22jeffpcbut I'm not in the mood to compile it :)
00:38.29opus_jeffpc: it is very bad :(
00:38.40opus_return "";
00:38.45jeffpcO_o
00:38.46opus_goto outerloop:
00:38.56jeffpco_O
00:39.38[TK]D-Fenderjeffpc : Use the Source Luke!
00:40.11[TK]D-Fenderjeffpc : You'll recompile your kernel, do everything else but won't do this one important easy bit?  Ridiculous.
00:40.29Gabriel25openh323 this is good ?
00:40.35Gabriel25I have no idea sorry
00:41.40jeffpc[TK]D-Fender: :)
00:41.48[TK]D-FenderGabriel25 : Ok, go experiment on your own a bit.. this is to Q&A for where you are at.  You haven't read up on this much at all and seem to be expecting everyone else to figure it out for you fron scratch....
00:42.07opus_Gabriel, its crap
00:43.06Gabriel25http://astrecipes.net/?n=102 this how to is good ?
00:43.08opus_Gabriel, if you want a really good solid implementation you will need to hire somebody who is an expert with it
00:43.29Gabriel25opus_ is only for me at home and just for test
00:43.39Gabriel25It`s working fine with soft phone
00:43.52[TK]D-FenderGabriel25 : Get off your but and just try stuff!
00:44.05Gabriel25but know I want to add an avaya 4624
00:44.18Gabriel25[TK]D-Fender i like to learn stuff
00:44.22Gabriel25And I need a start
00:44.33Gabriel25I installed one week ago asterisk
00:44.40Gabriel25And I made it work
00:44.47Gabriel25with soft phone
00:44.53[TK]D-FenderGabriel25 : 99% of * users are smart enough to avoid H.323 like the plague, you work on it for a few HOURS on your own  The odds that someone here can help you at any time is pretty low.
00:45.18[TK]D-FenderGabriel25 : Well... I guess you should have researched more before finding out now * doesn't come with H.323 by default....
00:45.32Gabriel25[TK]D-Fender I don`t want to use H.323 I want to use sip
00:45.43[TK]D-FenderGabriel25 : Well go flash that phone to SIP if you can....
00:45.58Gabriel25this is what I`m trying to find
00:46.14[TK]D-Fenderand go download the manuals, google up settings info.... just don't nag in here knowing that you have hardly tried at all..
00:46.14Gabriel25a firmware for that phone fo can register sip !
00:46.15Gabriel25:)
00:46.34*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
00:46.42Gabriel25[TK]D-Fender you are right :) I need toearn more !
00:46.49Gabriel25learn more
00:46.55Gabriel25then to ask questions
00:47.20Gabriel25on the avaya website they don`t have
00:47.32Gabriel25and now I have to find out if is working
00:47.33Gabriel25:D
00:47.48Gabriel25ftp://ftp.avaya.com/incoming/Up1cku9/tsoweb/ip_telephone/022006/4624_12_06readme_1_8_3.txt
00:51.29*** join/#asterisk coppice (n=chatzill@44.199.17.210.dyn.pacific.net.hk)
00:52.37*** join/#asterisk Twister (n=bob@host79.poky900.ncn.net)
00:53.29Twisterhi all
00:55.08Twisterim lookin for some feedback on the best softphone
00:55.41[TK]D-FenderTwister : They all suck, but eyebeam is the most functional of all.
00:55.42mjh001idefisk
00:55.57dlynes_office[TK]D-Fender: and snom360
00:56.10Twisterfrom x10?
00:56.13[TK]D-FenderTwister : Why are you looking for a soft-phone at all?
00:56.25*** join/#asterisk Lino` (n=Lino@i577BDDCD.versanet.de)
00:56.38dlynes_office[TK]D-Fender: causest the coolest new gadget for your desktop to infest your system tray!
00:56.56Twistermy boss wants a softphone for when he is out of the office and he wants a headset that he can anwser/end calls from
00:57.00[TK]D-FenderTwister : Yes, from XTen
00:57.24Twisterok that wont work then, unfortuinatly, im looking at this headset
00:57.32[TK]D-FenderTwister : Good reason.  If you don't need video and you want the perks, get X-Pro.
00:57.41Twisterthe plantronics CS50-USB
00:57.57Twisterbut  dont have to get that one
00:58.03[TK]D-FenderTwister : It should work fine with that....
00:58.24Twisterthe only requirment is that it be wireless and be able to anwser/end calls from the headset
00:58.28opus_are there any good call center USB headsets that support high quality audio?
00:59.06[TK]D-FenderTwister : Oh... hrm... not sure about that part....
00:59.19Twisterya
00:59.22[TK]D-Fenderopus_ : define high quality audio....
00:59.46Twistersee that headset will do it but wont work with that softphone or so they say just a sec ill get you a link with a list
00:59.59[TK]D-FenderTwister : Ask him if he'd like fries with that....
00:59.59coppicewell, there ain't much point in it sounding better than a phone :-\
01:00.04Twisterhttp://www.plantronics.com/media/downloads/PerSonoCallSoftphoneCompatibility.pdf
01:00.24[TK]D-Fendercoppice : I was getting there... leave a little bait first :)
01:01.11[TK]D-FenderTwister : Good.. works with eyeBeam.. there you go...
01:01.13[TK]D-FenderTwister : Problem solved.
01:01.17Twisteroh
01:01.23Twisteri didnt see that! whooo
01:02.26mitcheloci got that headset
01:02.30mitchelocit's not so good as a wired one
01:03.15Twisterya but unfortuinatly the wired one is not an option
01:05.11mjh001Is it possable to make asterisk dial from an email, ie 1115550000-3030@asteriskbox.net would make it dial out on a voip/pstn trunk to 1115550000 and also rign extenion 3030 internaly? Has anoyone seen anything like this done befor, so fat the closet thing I've been able to find is a hosted srevice that had a web dialer where a site visitor enters a phone number and there system call them and rings another phone numbr.
01:06.18Twistermitcheloc, is there another solution you would reccomend (as far as the hardware goes)
01:06.28mitchelocmjmac: was that hosted service asterisk compatible?
01:06.41mitchelocTwister: not specifically, just saying wired = much better quality
01:07.11mitchelocmjmac: does it *have* to be initiated from an e-mail? or would 3rd party software work for you?
01:07.23Twisteroh i agree
01:07.33*** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg)
01:07.40*** part/#asterisk jeffpc (n=jeffpc@ool-18ba4c2d.dyn.optonline.net)
01:08.35mjh001I do not belive so, the hosted service basicly connects a pstn to a pstn and is invoked from a web page. What I want to do is make my asterisk ring a local ext. and call out to a pstn over voip
01:09.00mitchelocmjmac: 3rd party software okay?
01:09.40mjh001A 3rd party app would work so ling as it can be automated and perhaps receive it direction from an email
01:09.45hadsmjh001: You could just forward a local mail alias to a script and have that script generate a call file.
01:09.54*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-2-21.red.bezeqint.net)
01:10.37mitchelocis this webmail or outlook?
01:12.59mjh001The remote user might be using either, or it might be from a form on a website, it would come in via smtp so the remote client dosent matter, as long as they formated it correctly. hads ->do you know of a script that will do the trick? I've been looking for about two weeks for a way to do this.
01:13.48mitchelocmjmac: it is possible, but you'll likely need to write your own script
01:13.58hadsmjh001: No sorry, I don't know of a pre-written script, it shouldn't be too hard to make though.
01:15.31mjh001okay, thanks anyway... I guess it is time I start learning to program anyway and stop wasteing so much time searching.... Guess this will be a good place to start...
01:15.32vilerSomebody that can help me in the Rhino Channel bank configuration please.
01:16.13mitchelocmjh0001: it should be very simple, if you run an smtpd on the server or apache and php
01:18.26mjh001okay, well I think I just pulled the wrong svn update to update my asterisk... I ended up with "Asterisk SVN-branch-1.2-r32373" off to go and look...
01:20.32mjh001yup, I pulled a branch abd not the trunk... duh!
01:23.31[TK]D-Fenderviler : Thats almost a contradiction in terms.. Rhino's autodetect jsut about anything you through at them
01:24.31*** join/#asterisk denon (n=denon@synapse.subneural.net)
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01:25.01opus_Jun  5 18:24:24 WARNING[5637]: chan_sip.c:2542 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)
01:25.06*** join/#asterisk dasenjo (n=dasenjo@208.195.215.207)
01:25.13opus_anyone seen that one before?
01:26.13dasenjoHi, I'm trying to compile * 1.0.11 and got this error: http://pastebin.ca/62367
01:26.20dasenjocan you help me?
01:28.19russellbdasenjo: i don't see anything wrong with that gcc command
01:28.33russellband 1.0.11 builds fine for me.
01:28.58dasenjorussellb, this is a "dirty" system ..
01:29.04russellbi can tell :-p
01:29.05[TK]D-Fenderrussellb : Well it does say : gcc: cannot specify -o with -c or -S and multiple compilations
01:29.18[TK]D-Fenderrussellb : Bad GCC version maybe?
01:29.24russellbyeah
01:29.31dasenjoI have installed *-1.2 from packages .. a
01:29.36russellbor ... something
01:29.43dasenjoand another * zap and libpri form sources .
01:29.47russellbwell if you are running 1.2, why are you trying to install 1.0.11?
01:29.49fileso why are you downgrading to 1.0.11?
01:30.04russellbfile: jinx.
01:30.07dasenjofxo interaction problems ..
01:30.20dasenjoi could not make work 1.2 ..
01:30.21russellbdowngrading will not likely help you
01:30.25russellbis it a digium card?
01:30.35dasenjo1.0.10 works very fine ..
01:30.40dasenjoTDM400P
01:30.46russellbhave you contacted digium support?
01:31.00dasenjono
01:31.04russellbwell why not?
01:31.23dasenjoI live in ther "third world"
01:31.44russellbi see ... well it looks like your system needs some upgrading :)
01:32.40dasenjoit is a debian stable without problems ..
01:32.52russellbwell you said dirty
01:32.57russellbi figured it was something terribly out of date
01:33.30[TK]D-Fenderdasenjo : Forget 1.0.X and jsut install the latest release from FTP (1.2.9)
01:33.47dasenjono .. so many installations form source .. that's all ..
01:33.57dasenjo[TK]D-Fender, it does not work ..
01:34.03dasenjofor my conf.
01:34.26dasenjoor for my card/line combination ..
01:34.50dasenjojust 1.0.10 with MARK2 and AGRESSIVE_SUPPRESSION works to detect hang up ..
01:37.04dasenjothis is a very strange error .. asterisk 1.0.11 _should_ compile
01:39.38opus_decodeMP3: Junk at the beginning of frame 00000000
01:39.45opus_anyone know of a solution around that problem?
01:39.56*** join/#asterisk NewSole (n=dave@d226-107-112.home.cgocable.net)
01:40.35NewSoleWee... After Months of Work.... we now Have a Working softphone... weeee.....
01:45.42*** join/#asterisk ceeto (i=cio@adsl-072-149-159-016.sip.bhm.bellsouth.net)
01:45.54ceetoHi all.  I have like 15 asterisk processes running, is that normal?
01:46.11b00merI am trying to setup a working screen, someone earlier said I should check in to the privacy settings in app_dial... anyone have a working example?
01:46.11[TK]D-Fenderceeto : not processes, threads
01:46.30[TK]D-Fenderceeto : I know it looks funy.. you see it more on 2.4 kernels I believe....
01:46.40[TK]D-Fenderceeto : but everything is fine
01:46.55ceetoDoes each process take memory?
01:47.08b00merbut the privacymanager doesn't look like what I am looking for
01:47.16[TK]D-Fenderceeto : Clearly... how much I couldn't say
01:47.17russellball of the threads share the same memory
01:47.39russellbit's not the same as multiple processes ...
01:47.51b00meranyone available to help debug / understand my macro issue?
01:48.15[TK]D-Fenderb00mer : pastebin away... everything related to it.
01:48.21b00merhttp://pastebin.com/760744
01:48.33b00merWhat I am trying to accomplish is:
01:48.38*** join/#asterisk Hmmhesays (n=Neg@31-201.69-92-cpe.cableone.net)
01:48.58Hmmhesaysthis embedded sh1at sucks
01:49.02b00merSomeone calls a support number... 4 engineers are called with a short message saying press 1 if you want to take the calll
01:49.08b00merright now it works, but
01:49.20b00merif they pickup and let it time out... fine
01:49.31b00merif they pick up and hangup it hangs up on the caller
01:49.42b00merif they pickup and press 1 they are connected... fine
01:49.56b00merso I am only having issue with the pickup and hangup
01:50.18Hmmhesaysusing a queue?
01:50.22b00merno queues
01:50.36b00merjust using this "screeen" logic
01:50.59[TK]D-Fenderb00mer : What would you have it do on pickup/hangup?
01:52.43b00merthe CONTINUE
01:52.52*** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca)
01:53.09b00merunless they press 1 ... I want it to assume that the call is not wanted and continue on the call list
01:53.27asterisk-dudi'm using a tdm405p t1 card and i get a whistling sound for a dialtone,
01:53.49asterisk-dudanyone know what could be wrong?
01:53.54matthewsimpsonyou bought the bluebox edition tdm405p :(
01:53.58*** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca)
01:54.15asterisk-dudwhat is bluebox edition?
01:54.20russellbthere is no such thing as a tdm405p :)
01:54.28TESTER2any idea about a very old phone (inband alimentation) not riging on a spa-1001?
01:54.58asterisk-dudte405p
01:54.59asterisk-dudsorry
01:54.59[TK]D-Fenderb00mer : add an "h" exten or force the dial to continue on (can't recall the parm for that)
01:55.43asterisk-dudrussellb it's te405p
01:56.03russellbasterisk-dud: gotcha.  it's most likely a problem with zone settings in /etc/zaptel.conf
01:56.35surfduebk
01:56.44asterisk-dudok, i was wondering about that, currently i have nothing for that, but it worked like that before
01:56.47dlynes_office[TK]D-Fender: I found out why sangoma's not getting compiled
01:56.54b00mer[TK]D-Fender: you think a exten => h,1,SetVar(MACRO_RESULT=CONTINUE) will work?
01:56.59[TK]D-Fenderdlynes_office : oh, do tell :)
01:57.02dlynes_office[TK]D-Fender: its dependencies aren't getting satisfied in the kernel build
01:57.12*** join/#asterisk jeffik (n=Jeff@Maroon-103-176.ADSL.NetSurf.Net)
01:57.14dlynes_office[TK]D-Fender: and so i never have the option to select sangoma in the make menuconfig
01:57.14russellbasterisk-dud: well, you need to add those options :)
01:57.21[TK]D-Fenderb00mer : Give it a shot.  I'm not sure if that will get called in the macro, or in the context with your Dial in it...
01:57.50asterisk-dudwould that be y the first hole doesn't go away from a red alarm?
01:57.52[TK]D-Fenderbrb, gotta restart my server.. X went haywaire...
01:58.27russellbasterisk-dud: um, probably not ...
01:58.45asterisk-dudhow would i know if my card is dunked
01:59.01russellbcontact tech support, but it's probably fine ...
01:59.31asterisk-dudshould i add the location line at the end or at the beginning or isn't it a big deal
01:59.55russellbi would put it before the channels, because i can't remember if it matters
02:02.07asterisk-dudrussellb it didn't help
02:02.25russellbi don't believe you :)
02:02.28Hmmhesaysjust what I needed (just what I needed)
02:02.37asterisk-dudthat's terrible
02:02.38russellbdid you reconfigure the card using ztcfg ?
02:02.38asterisk-dudlol
02:02.42asterisk-dudyes
02:03.00Hmmhesaysdid you swear at it?
02:03.20*** part/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca)
02:03.20asterisk-dudloadzone= us
02:03.21asterisk-duddefaultzone= us
02:05.18*** join/#asterisk kernel20 (n=kernel20@203.160.223.26)
02:05.24[TK]D-Fender.
02:05.30[TK]D-FenderOk, not cut off it seems
02:05.32kernel20hi what is MACRO_EXTEN?
02:05.32Hmmhesaysanyone ever used buildroot for ulibc?
02:05.36Hmmhesays*uclibc
02:05.56[TK]D-FenderWow and it jsut caught me up on everything I missed :)
02:06.16Hmmhesayswhat are you babbling about [TK]D-Fender?
02:06.23kernel20most of the manuals on voicemails deals on MACRO_EXTEN
02:06.23russellbasterisk-dud: please contact digium technical support, they will be able to help you
02:07.10kernel20exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN} will anounce user is unavailable, now the problem is the MACRO_EXTEN, how can i relate it in my voicemail.conf?
02:07.30[TK]D-Fenderkernel20 : Go read up on macro's.....
02:07.41dlynes_office[TK]D-Fender: i only told it to compile tdm drivers last time...let's go see what it says when i tell it all drivers
02:07.49dlynes_office[TK]D-Fender: try this one out
02:07.52dlynes_office~kernel20
02:07.56jboti guess kernel20 is an annoying user that is allergic to reading documentation.
02:07.56kernel20i am reading, but got troubles on understanding MACRO_EXTEN
02:07.56[TK]D-Fenderdlynes_home : um.. no
02:08.15kernel20please dont start
02:08.41kernel20anybody else?
02:09.32dlynes_office[TK]D-Fender: ?
02:09.37[TK]D-Fenderkernel20 : this one is a freebie : When you call a macro like "exten => 110,1,Macro(justamacro,1234)" ${MACRO_EXTEN} = 110 and ${ARG1} = 1234.  Got it?
02:09.37kernel20where should i declare macro_exten? and how can i relate it to my voicemail.conf
02:09.53asterisk-dudanyone using channel banks?
02:10.07[TK]D-Fenderkernel20 : You don't declare it.  its set by the line that calls the macro.  It is a concept you are not required to implement.
02:10.20[TK]D-Fenderasterisk-dud : I use a Rhino modular CB.
02:11.01asterisk-dudhave u ever used adtran
02:11.07littleballhello, i am looking for SER/asterisk reference deployment notes. who can help?
02:11.20[TK]D-Fenderasterisk-dud : nope.
02:11.36asterisk-dudr u fxs or fxo D-Fender
02:11.56kernel20[TK]D-Fender: how can i set it
02:11.57kernel20?
02:13.09*** join/#asterisk IBN287-Jo (i=anpu359@pool-72-65-231-46.pitbpa.east.verizon.net)
02:13.25*** part/#asterisk IBN287-Jo (i=anpu359@pool-72-65-231-46.pitbpa.east.verizon.net)
02:13.36[TK]D-Fenderkernel20 : You don't set it.  It gets set by the line that calls the macro.  Look at my sample and open your eyes.
02:13.46[TK]D-Fenderasterisk-dud : I run both
02:14.13kernel20[TK]D-Fender: http://pastebin.com/761306 <- how can i add voicemail on it?
02:14.23asterisk-dudok, for fxs ports, i use fxo_ks signalling correct?
02:15.21mitchelockernel20 came back?
02:15.32kernel20[TK]D-Fender: what did u see?
02:15.34[TK]D-Fenderkernel20 : use your imagination, and when those 5 seconds are up, go read this and pray for inspiration : http://www.voip-info.org/wiki/view/Stdexten+macro
02:15.44mitcheloc~kernel20
02:15.45jbotfrom memory, kernel20 is an annoying user that is allergic to reading documentation.
02:15.49[TK]D-Fenderasterisk-dud : Correct
02:16.21[TK]D-Fendermitcheloc : Oh just put a lid on it.  You aren't part of the solution...
02:16.34kernel20http://pastebin.com/761306 <- how can i add voicemail on it? base on that link
02:16.48*** join/#asterisk `Kevin (n=Kevin@64.243.236.20)
02:16.51[TK]D-Fenderkernel20 : Did you go to the WIKI page I linked?
02:17.12mitcheloc[TK]D-Fender: =P, i'd help, but i've tried before
02:17.24kernel20yeah
02:17.25mitcheloc[TK]D-Fender: i believe kernel20 called me an "ass" when i gave him a suggestion
02:17.38kernel20but really dont get the whole point sorry
02:17.45kernel20mitcheloc: please dont start
02:17.46[TK]D-Fenderkernel20 : Go see how they did VM for it, and just do the same.
02:18.01[TK]D-Fendermitcheloc : Thats nice.  Now put a lid on it.
02:18.10kernel20[TK]D-Fender: http://pastebin.com/761306 <- how can i add voicemail on it? base on that link
02:18.24kernel20please look at the link
02:18.33kernel20the voicemail is at voicemail.conf
02:18.38asterisk-dudD-Fender, if my channel bank is FXS Loop Start, should i use FXS_ks signalling
02:18.59kernel20my problem is how would it able to access the declarations i have done at voicemail.conf
02:19.05[TK]D-Fenderkernel20 : When do you want to go to VM in there?
02:19.18[TK]D-Fenderasterisk-dud : fxo_ks
02:19.24kernel20if NOANSWER
02:19.35[TK]D-Fenderkernel20 : then add that.
02:19.50kernel20yeah but it wont prompts for voicemail
02:20.00[TK]D-Fenderkernel20: DO IT then show me.
02:20.02kernel20please have a look
02:20.33[TK]D-Fenderkernel20 : I did have a look .  CHANGE IT YOURSELF RIGHT NOW.  try something and then see what doesn't work.
02:20.46kernel20http://pastebin.com/761315
02:21.00kernel20[TK]D-Fender: http://pastebin.com/761315
02:21.21[TK]D-Fenderkernel20 : Good start... now tell me... what value do you imagine is in ${ARG1} ?
02:21.38kernel20SIP/11000
02:21.49kernel20i want to direct it to voice mail
02:21.49[TK]D-Fenderkernel20 : is that a valide mailbox name?
02:21.55kernel20no
02:21.59kernel20thats my problem
02:22.09[TK]D-Fenderkernel20 : then you need to apss it ANOTHER argument that holds a useful value
02:22.10kernel20base on that how can i relate it to my voicemail
02:22.22[TK]D-Fender${ARG2}
02:22.26kernel20can u add sample on that link?
02:22.35[TK]D-Fenderkernel20 : pass 2 parameters in your macro.
02:22.49dlynes_office[TK]D-Fender: got something loading now :)
02:22.50kernel20ahhh
02:22.54kernel20hehehe
02:22.56kernel20got it now
02:22.57[TK]D-Fenderkernel20 : No.  I want you to listen to my rather direct advice and do it yourslef, then show me.
02:22.58kernel20too dumb
02:22.59kernel20hahahah
02:23.05kernel20wait
02:23.39*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
02:24.18dlynes_office[TK]D-Fender: for whatever reason, the sangoma stuff isn't showing up in make menuconfig
02:24.56surfduemy software cant connect to the server im not behind a nat nor is my server
02:25.00surfduewhy is this?
02:25.09surfduecan I use a website to test the connection to asterisk?
02:25.10dlynes_officesurfdue: errors?
02:25.15surfduemaybe
02:25.18surfduewhere are the logs again
02:25.19surfduei forget
02:25.21[TK]D-Fenderdlynes_home : What is this menuconfig of which you speak?
02:25.22dlynes_officesurfdue: /var/log/asterisk
02:25.24surfdue/var/log
02:25.25surfdueok
02:25.28dlynes_office[TK]D-Fender: for the linux kernel
02:25.47dlynes_office[TK]D-Fender: you know?  make menuconfig/make config/make xconfig
02:25.49[TK]D-Fenderdlynes_home : Why are you screwing with your kernel?
02:26.15dlynes_office[TK]D-Fender: to enable the options that are necessary for the sangoma
02:26.39*** join/#asterisk rvhi (n=rv@66.175.65.89)
02:27.09[TK]D-Fenderdlynes_home : Ummm.. I never ran into an out of the box setup that needed anything....
02:27.28kernel20[TK]D-Fender: http://pastebin.com/761334
02:27.33surfduedlynes_office, http://host41.com/log.txt
02:27.38kernel20please do have a check
02:27.45dlynes_office[TK]D-Fender: yeah, but if change something in my kernel, i don't want to have to go through the whole sangoma setup again
02:27.46kernel20it doesnt work what i expected
02:27.48surfduewould those cause it to not beable to connect?
02:27.58NewSoleWee... We now Have a Working softphone... weeee.....
02:28.22[TK]D-Fenderkernel20 : You reloaded this new config and simply didn't answer the call?
02:28.24kernel20[TK]D-Fender:?
02:28.37kernel20i reloaded
02:28.45kernel20it that conf workable?.
02:28.47dlynes_officesurfdue: ummmmare you using the sample config file?
02:28.56surfdueya
02:29.00surfduewell no
02:29.09dlynes_officesurfdue: which is it?  yes, or no?
02:29.11surfduethe default ones in /etc/asterisk
02:29.17surfdueafter make samples
02:29.20dlynes_officeyeah...so you're using hte sample config files
02:29.21[TK]D-Fenderkernel20 : pastebin a call attempt
02:29.46dlynes_officesurfdue: lose all the extra crud from sip.conf and extensions.conf that you're not using
02:29.54surfduereally
02:29.55surfdue?
02:29.59surfduelike even the bind stuff
02:30.00surfdueand all that?
02:30.09dlynes_officesurfdue: use your better judgement
02:30.19surfdueyes?
02:30.27dlynes_officesurfdue: you can keep the stuff in general
02:30.34dlynes_officesurfdue: but anything else you're not using, get rid of it
02:30.57dlynes_officesurfdue: or if you want, pastebin it, and pastebin it back with all the crap taken out
02:31.18dlynes_officesurfdue: you'll find it very difficult to solve your problems in asterisk when your config files are a mess
02:31.31dlynes_officesurfdue: especially when they have a bunch of crap in them that you don't understand
02:31.36surfduedone
02:32.00surfduedlynes_ will you do it?
02:32.03surfdueif i pastebin it
02:32.09dlynes_officesurfdue: yes...just pastebin it
02:32.16surfduewith the password?
02:32.21dlynes_officesurfdue: are you using asterlink as your voip provider?
02:32.26dlynes_officesurfdue: no...scrub the passwords
02:32.35surfdueya
02:32.47kernel20[TK]D-Fender: http://pastebin.com/761343
02:32.48dlynes_officesurfdue: i.e. replace the passwords with XXXXX or something similar
02:33.41[TK]D-Fenderkernel20 : Look at your Goto.  theres the problem.
02:33.54*** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen)
02:34.04surfduehttp://host41.com/sip.conf
02:34.32kernel20[TK]D-Fender: what is problem?
02:34.47asterisk-dudwhen i start up my channel bank all 24 lines flash and my phones start ringing
02:35.01dlynes_officecool trick, asterisk-dud
02:35.07[TK]D-Fenderkernel20 : Look at your Goto VERY closely and then again at your dial-lan and you'd beet er see for yourself whats wrong....
02:35.24kernel20what is the problem
02:35.28kernel20i am very close now
02:35.33kernel20please tell me
02:35.46[TK]D-Fenderkernel20 : try using your eyes... where is the Goto "going"?
02:35.48asterisk-duddlynes-office it's pissing me off
02:35.58asterisk-dudi don't think i could if i tried
02:36.19kernel20to my next line
02:36.21kernel20of course
02:36.40dlynes_officeasterisk-dud: i would imagine your span is incorrect or something
02:36.47dlynes_officeasterisk-dud: are you using a four port pri card?
02:37.27*** join/#asterisk tsurk0 (n=tsurko@85.187.160.157)
02:37.32[TK]D-Fenderkernel20 : Read this and WAKE UP!!!!     -- Executing Goto("SIP/11000-a4a9", "s-NOANSWER|1") in new stack
02:37.37dlynes_officesurfdue: http://pastebin.com/761345
02:37.46asterisk-duda te405p
02:38.07dlynes_officeasterisk-dud: you've got the channel bank plugged into the top port, or the bottom port?
02:38.23asterisk-dudtop one,
02:38.35dlynes_officeasterisk-dud: pastebin your zaptel.conf file
02:38.51asterisk-dudwhere do i pastebin?
02:38.56asterisk-dudi'm new at this
02:39.01dlynes_office~pb
02:39.03jbotpb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
02:39.29kernel20?
02:39.44kernel20[TK]D-Fender: please pastebin
02:40.01[TK]D-Fenderkernel20 : Wake up! there is no  "s-NOANSWER|1" in your macro for it to Goto to.
02:40.27dlynes_office[TK]D-Fender: a prime reason why peeps shouldn't use auto fallthrough :)
02:40.42[TK]D-Fenderdlynes_office : Irrelevent to debugging this,.
02:40.50surfduety dlynes
02:40.53asterisk-duddlynes_office: http://pastebin.com/761346
02:40.57[TK]D-Fenderdlynes_office : the problem is BLATANTLY obvious
02:41.08dlynes_office[TK]D-Fender: yeah, but it would help kernel20 understand where his problems lie, easier
02:41.12dlynes_office[TK]D-Fender: yeah, i realize that
02:41.16toppinganyone have favorite IAX-PSTN providers for the US?
02:41.34dlynes_office[TK]D-Fender: that's why i usually have him on /ignore
02:41.37[TK]D-Fenderdlynes_office : you mean those list 2 lines I made weren't obvious enough? *sigh*
02:42.13asterisk-duddlynes_office: did u get that?
02:42.19dlynes_officeasterisk-dud: yeah
02:42.30NewSoleWe need some Beta Testers..... Any takers
02:42.48surfdueMySQL RealTime: Failed to connect database server  on .
02:42.51surfduehow do i fix this?
02:43.25dlynes_officeasterisk-dud: hrm...i'm not sure how to fix your problem...the loadzone and defaultzone and span should all be fine for a channel bank
02:43.46dlynes_officeasterisk-dud: but i'm not sure about the fxoks; i'm not sure if it should be that, or channels specified
02:43.47surfduei think its missing a mysql conf file
02:43.53surfdueJun  5 22:41:24 WARNING[12374] cdr_addon_mysql.c: Unable to load config for mysql CDR's: cdr_mysql.conf
02:43.55surfduewhere do i get this?
02:43.56dlynes_officeasterisk-dud: i've never configured a channel bank
02:44.04asterisk-dudcould there be a prob with my zapata?
02:44.10dlynes_officesurfdue: go into modules.conf and do an unload => cdr_mysql.so
02:44.20surfduebut i want mysql?
02:44.34dlynes_officesurfdue: worry about it later after you get everything working
02:44.44[TK]D-Fenderasterisk-dud : pastebin your zapata.  And your zaptel should be 1,1,0 not 1,0,0
02:44.47dlynes_officesurfdue: you'll need to configure the database, set it up, get it running and everything else
02:45.06surfduehow?
02:45.09surfduei wanna do this now
02:45.09surfdue:P
02:45.12surfduei need help :
02:45.13surfdue:P
02:45.28dlynes_officesurfdue: you're talking to someone that's never set it up before
02:45.37dlynes_officesurfdue: cdr_mysql is in asterisk-addons-1.2.3
02:46.04surfduewhich i did install
02:46.14dlynes_officesurfdue: well, then
02:46.50dlynes_officesurfdue: you need to read up on the documentation for it, so you know what username, password to set up for the database, how to configure where the database is, what tables to create, and how to define them, ...
02:47.03surfduei see it
02:47.13dlynes_officesurfdue: it should all be in asterisk-addons
02:47.59asterisk-duddlynes_office: i fixed
02:48.07asterisk-dudmoved my card to different slot
02:51.25NewSoleWe need some Beta Testers..... Test New SoftPhone this week Any takers
02:51.43mitchelocNewSole: what's it called?
02:51.46dlynes_office[TK]D-Fender: looks like i might need a firmware upgrade to get this working
02:51.59NewSoleSoftPhone.....
02:52.00[TK]D-Fenderdlynes_office : Good idea as well....
02:52.20mitchelocheh, neutral name ;)
02:52.20[TK]D-Fenderdlynes_office : Make sure to get the EC tools as well if you have that module.
02:52.23mitcheloci'll help test if you want
02:52.39dlynes_office[TK]D-Fender: yeah...i've got that for the a200d with the ec
02:52.55dlynes_office[TK]D-Fender: it's probing fine...just can't allocate PIC memory on the card
02:52.55NewSoleits supports SIP/H323/IAX2 and has built in G729/G723/G726/ULAW/ALAW/GSM/iLBC/SPEEX
02:53.15mitchelocwhy would you add h323?
02:53.26dlynes_officemitcheloc: for connecting to Quintums?
02:53.31surfduewere are the modules located for asterisk?
02:53.35mitchelocwhats that?
02:53.40dlynes_officemitcheloc: and for certain providers that only do h323?
02:53.47mitchelocwho?
02:53.59dlynes_officemitcheloc: Sun Telecom is one I can think of off the top of my head
02:54.03NewSolealso has built in call Forwaring and Voice Mail Recording
02:54.06dlynes_officemitcheloc: they do h323 in Canada, SIP in the US
02:54.23dlynes_officeNewSole: Windows or Linux?
02:54.24*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
02:54.26mitcheloci think he is ignoring me
02:54.26mitchelocheh
02:54.30NewSoleWindows
02:54.42dlynes_officeNewSole: Yeah, I know someone that would probably test it then
02:54.45NewSoleno not ignoring u.... PM's
02:55.17mitchelocNewSole: does it have a command line interface for dialing?
02:55.28NewSoleits going to be freeware wile in beta..... but it will be 20$ when done
02:55.38toppingNewSole: I would test on OS-X :-)
02:56.05NewSoleits not built on Mac yet...
02:56.16dlynes_officeNewSole: Yeah...my friend's definitely interested
02:56.30surfdueanyone?
02:56.31dlynes_officeNewSole: Should I msg you his email address?
02:56.41NewSoleRequires Microsoft .Net Framework..... (100% .Net Framework)
02:56.41dlynes_officesurfdue: /usr/lib/asterisk/modules (default)
02:56.53dlynes_officeNewSole: Not an issue...he's a .NET developer
02:57.03mitchelocNewSole: command line?
02:57.09NewSoleno not yet
02:57.32mitcheloci can add support for it in Snap when you are ready with the command line =)
02:57.59NewSoleit also has a built in emailer... to email the recorded voice mails
02:58.03dlynes_officeNewSole: msg you email details, or is there a download page for it?
02:58.39mitcheloclol, i feel so ignored talking to newsole =/
02:59.03NewSolesend emails requests to mworkman@imbroadcasting.net..... Beta's emailed out on Weekend
02:59.09dlynes_officeNewSole: ok, thanks
02:59.29NewSolemitcheloc... I am not ignoring you
03:00.21NewSolejust alot of PM's from Beta testers.... and new build comming out this weekend
03:00.59dlynes_officeNewSole: it is compatible with asterisk, right?
03:00.59mitchelocdlynes_office: tell your friend about snap, i just put out my mozilla plugins ;)
03:01.11dlynes_officemitcheloc: you mean mmc?
03:01.12NewSolenot doing comand line for it.... but it will support VB Scripts when release mode done
03:01.19surfduehttp://host41.com/log.txt
03:01.21mitchelocmmc?
03:01.24surfduecan someone help me fix those errors
03:01.28dlynes_officeMicrosoft Management Console
03:01.36mitchelocNewSole: why no command line?
03:01.44mitchelocdlynes_home: no i mean, mozilla plugins
03:01.47dlynes_officemitcheloc: because it's a windows plugin?
03:01.53dlynes_officeerm application i mean?
03:02.03dlynes_officemitcheloc: so what's snap then?
03:02.09mitchelocmozilla = firefox/thunderbird
03:02.15dlynes_officeis it like snap crackle pop rice crispies?
03:02.19mitchelocit's a dialer, heh check out www.snapanumber.com
03:02.33mitcheloci'm adding support for softphones and stuff, so they can focus on being phones and i can focus on dialing
03:03.02dlynes_officeah, cool
03:03.21dlynes_officeNow if only I could convince him to get rid of Internet Exploder :)
03:03.45dlynes_officeHe's finally starting to get into the whole linux groove thing :)
03:03.51mitchelocworking on IE also, just not done yet
03:03.53dlynes_officeHe does Python .NET development
03:03.59mitchelocwierd
03:04.08dlynes_officeHe loves python and loves .NET
03:04.16dlynes_officebut he's not a big fan of Microsoft :0
03:04.23mitcheloclol, nice
03:05.22asterisk-dudi would like to builg a timer that could be overridden with a passowrd
03:05.41mitcheloctimer?
03:06.12asterisk-dudthat would hang up the line
03:06.17dlynes_officemitcheloc: so they're thunderbird/firefox extensions then, not plugins
03:06.20dlynes_officemitcheloc: even better
03:06.39mitchelocyea, i keep forgetting the difference in the terminology
03:06.48dlynes_officeasterisk-dud: you would probably need to implement that in agi
03:06.49mitcheloci'll update that
03:07.03dlynes_officemitcheloc: firefox does plugins, too....plugins are different
03:07.19dlynes_officemitcheloc: plugins are separate programs that run as a child window of firefox
03:07.25asterisk-dudi've tried with java-agi, but i think i'll have to use a different language
03:07.28dlynes_officemitcheloc: extensions are XUL applications
03:08.06dlynes_officemitcheloc: but yeah...that's a wicked extensions
03:08.15dlynes_officemitcheloc: we were looking for something like that about 2 years ago
03:08.33*** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net)
03:08.35mitchelocheh, well it's an xul extension then, thanks =)
03:08.36dlynes_officemitcheloc: how much are you selling it for?
03:08.54dlynes_officemitcheloc: or you're not the author?
03:08.54trixterhas anyone noticed a threading issue with 1.2.9 that if you do   asterisk -rx 'show channels' it displays the header but nothing else?
03:08.55mitchelocwell they are free, but i'd appreciate if you buy the pro so i keep developing =)
03:09.00dlynes_officetrixter: 1.2.9?
03:09.01mitchelocit's $29.99, yes i'm the author
03:09.18trixteryes 1.2.9 was released today to fix a security problem in chan_iax2 that lets people crash your server
03:09.21trixter1.0.11 as well
03:09.27trixterits in the topic
03:09.36dlynes_officetrixter: ah...cooll..didn't see it in my email
03:09.41dlynes_officeand i never read the topic :)
03:10.26*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
03:10.37dlynes_officeoh yeah...the announcements show up in my home email
03:10.40mitchelocdlynes_office: try it out for a bit, and let me know how it goes for you ;)
03:10.42trixterI just noticed that show channels, sip show channels and iax2 show channels all fails if done from asterisk -rx with > 50 channels in use (it doesnt appear to happen at all with fewer channels) and this worked on a previous 1.2.x version so something in the last few versions broke that somehow
03:10.46trixterprolly a threading mutex issue
03:10.58dlynes_officemitcheloc: does it work in linux and windows?
03:11.13mitchelocdlynes_home: not on linux...yet, will get to that soon
03:11.32dlynes_officemitcheloc: ah...shouldn't require much extra code if any for linux though, right?
03:11.59mitchelocnah, just some time debugging the problem areas in my code
03:12.14dlynes_officetrixter: hrm...i think i remember someone babbling about it in asterisk-dev last night
03:12.15surfduedlynes_ can you help me clean this up http://www.host41.com/extensions.conf
03:12.23surfduei only added 2 lines 2 of the extension lines
03:13.33NewSoleso who wants to be a beta tester for next week.... email me
03:13.58surfdueNewSole, i have a server and wouldnt mind
03:14.01surfduebeta test what?
03:14.17NewSolea new windows Softphone
03:14.38dlynes_officesurfdue: add newsole's new softphone and mitcheloc's dialer app together for a cool combo :)
03:15.04mitchelocdlynes_office: yea haha, use the "Path Connection" in my dialer, you don't even need to configure it for your asterisk server ;)
03:15.08NewSoleIAX/SIP/H323 with G729/G723/G726/ULAW/ALAW/GSM/iLBC/SPEEX
03:15.13mitchelocautoconfigs for skype and vonage and eyebeam users too
03:15.19surfdueNewSole, sure
03:15.28surfduedlynes are you able to clean that fo rme?
03:15.34dlynes_officesurfdue: are you using dundi?
03:15.37NewSolesend emails requests to mworkman@imbroadcasting.net..... Beta's emailed out on Weekend
03:15.50surfduedundi?
03:15.59dlynes_officesurfdue: e164
03:16.00asterisk-dudi added a second channel bank to my t1 card and i don't get a dialtone
03:16.07asterisk-dudboth lights are green
03:16.26surfduedone
03:16.31surfduee14?
03:16.32dlynes_officeasterisk-dud: if it's green then there's probably something screwy in your zapata.conf file
03:16.34surfduewhats that
03:16.34surfduelol
03:16.43surfduewhat page in the book is it?
03:16.51dlynes_officesurfdue: forget it then...i'll comment it out and move it all down to the bottom of the script
03:17.28asterisk-dudcan u check it?
03:17.49dlynes_officei could if you pastebinned it
03:17.52surfdue:P
03:18.08surfduedlynes http://www.host41.com/extensions.conf
03:18.22dlynes_officesurfdue: i'm already working on that
03:18.28asterisk-duddlynes_office: http://pastebin.com/761418
03:19.11surfduek
03:19.53dlynes_officesurfdue: you've only got asterlink for all outbound calls?
03:20.40surfdueya
03:20.52surfdueand inbound i just wnana set it up so im on ext 200
03:20.59surfdueand i can setup a voice menu with hold music and such
03:21.02dlynes_officeasterisk-dud: do you have asterisk up and running?
03:21.03surfduelike i have on my old server
03:21.10surfdueexcept i dont know how to cuase i use to use freepbx, which sucked.
03:21.24surfdueso im trying to move over 'the book' is very LONG :P
03:21.32surfduety for helping though
03:22.30asterisk-dudyes
03:22.50asterisk-duddlynes_office: i get a dialtone from one channelbank and not from the other right now
03:23.42[TK]D-Fenderasterisk-dud : pastebin your zaptel.conf
03:24.08asterisk-dudok, i fixed
03:24.14asterisk-dudsorry to bother u guys
03:24.33blitzrage! ! !
03:24.47dlynes_officeasterisk-dud: you had a misconfig on your channel bank?
03:24.52blitzrage!! !! !!
03:25.03dlynes_office!!! !!! !!!
03:25.08surfdue!
03:25.09surfdue:P
03:25.20[TK]D-Fenderblitzrage : I DON'T WANT RELATIONSHIP!
03:25.33surfdue[TK]D-Fender, i know you :P
03:25.43blitzragelol
03:25.59[TK]D-Fender:|
03:27.14blitzrage|:
03:27.35*** join/#asterisk focks (n=craig@74.130.97.237)
03:29.23*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
03:29.24*** part/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net)
03:32.13dlynes_officesurfdue: back up your existing file, and try this one:  http://pastebin.com/761438
03:32.19dlynes_officeon that note, i'm out for a while
03:32.26surfduek
03:32.27surfduety
03:32.47*** part/#asterisk dlynes_office (n=dlynes@216.251.149.66)
03:32.53*** join/#asterisk dlynes_office (n=dlynes@216.251.149.66)
03:33.02dlynes_officeoops...wrong window :)
03:35.26surfduehey same error
03:35.26surfdue:P
03:38.06mitchelocdlynes_home: hi
03:44.20Eric-xxi have a question , i setup a incoming trunk and linked to a digital recp, everything works well when i call the incoming trunk
03:44.38Eric-xxbut i can't seems to be able to see the incoming callerid/did number
03:45.02Eric-xxall i could see is     -- Executing Set("SIP/651234567-7daa", "FROM_DID=s")
03:45.14*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
03:45.26Eric-xxanyone know's how i could see the callerid of a incomming route
03:45.43[TK]D-FenderEric-xx : Please read the channel topic....
03:46.05Eric-xxk
03:47.03*** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
03:48.24*** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com)
03:51.31*** join/#asterisk bmg505 (n=leon@c1-161-9.rndf.isadsl.co.za)
03:56.06hadsDoes this look like something I should bring up here, or over in asterisk-dev land? http://pastebin.com/761623
04:00.08*** join/#asterisk PrOsHoCk (i=PrOsHoCk@ppp-71-140-3-39.dsl.scrm01.pacbell.net)
04:11.29*** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar)
04:13.08surfdueanyone?
04:15.07litagehow do you determine which modules asterisk has loaded?
04:15.38russellbshow modules
04:16.59*** join/#asterisk terrapen_ (n=cjs@mauritius.island.nu)
04:17.39litagerussellb: ``show modules'' lists all modules, not just the ones that are loaded
04:17.41*** part/#asterisk terrapen_ (n=cjs@mauritius.island.nu)
04:17.51*** join/#asterisk terrapen_ (n=cjs@mauritius.island.nu)
04:17.52russellblitage: it lists all the modules that are loaded.
04:18.01litageah. thanks. didn't notice that part
04:18.02litage:)
04:18.22russellbasterisk doesn't know anything about modules not loaded :)
04:22.48*** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com)
04:23.39asterisk-dudi want to make a dialplan that will take in the EXTEN argument, take that value and find the variable that is equal to EXTEN
04:23.57asterisk-dudso if EXTEN = 2030, and 2030 = ZAP/1
04:24.13asterisk-dudthen asterisk would dial zap/1
04:24.19asterisk-dudif 2030 is dialed
04:24.47asterisk-dudi can't get it to take in the value of variable 2030
04:25.24asterisk-dudexten => _20NX,1,Set(CHANNEL = ${EXTEN})
04:25.24*** join/#asterisk docelm0 (n=docelmo@55-65.126-70.tampabay.res.rr.com)
04:25.46docelm0oi
04:25.53asterisk-dudnow a want to take the value of channel and find the variable equal to it, and retrieve it's value
04:26.07docelm0~seen zoa
04:26.10jbotzoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 11h 52m 12s ago, saying: 'you need a gsm plugin for winamp'.
04:26.13asterisk-dudand then dial that channel
04:26.27docelm0crap just missed em
04:29.57*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:30.55[shodan]is this usable => http://www.i-rocks.com/2004/product/IR-2500.htm ?
04:31.37*** part/#asterisk BenderNZ (n=bender@ip-58-28-96-39.wxnz.net)
04:32.53*** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca)
04:33.07asterisk-duddid anyone get the question i posted?
04:33.15asterisk-dudi get kicked off
04:35.59asterisk-dudany one out there?
04:37.35kernel20hello
04:38.08kernel20would it possible to put password to number that dials that starts on _951X
04:38.09kernel20?
04:38.28kernel20if that number has been pressed it will asked for password
04:38.40kernel20before a call can be stablished
04:38.49asterisk-dudlets say the value of variable CHANNEL is 2333, how can i take that and find variable 2333, and use it's valur
04:40.31asterisk-dudanybody?
04:41.07*** join/#asterisk cryptnix (n=andrew@64.25.198.123)
04:41.37asterisk-dudanyone know a good SIP app for pocket pc
04:41.42asterisk-dudthat will work with asterisk
04:44.15*** join/#asterisk denon (n=denon@synapse.subneural.net)
04:44.15*** mode/#asterisk [+o denon] by ChanServ
04:51.38*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
05:01.46*** join/#asterisk Strom_C (n=strom@gateway.digium.com)
05:02.28Strom_Cyo
05:02.42asterisk-dudyo
05:02.57Strom_Cis there anyone in here who wouldn't mind sending me a test fax?
05:03.20asterisk-dudwhere r u?
05:03.25Strom_CLong Beach, California
05:03.38asterisk-dudlong ways from here
05:04.09Strom_Casterisk-dud: canada counts as a long way away? :)
05:04.18asterisk-dudlol
05:04.22Strom_Cyou're in the same numbering plan space for crying out loud
05:04.24asterisk-dudwhat number, i can try
05:04.41Strom_Csee privmsg
05:05.02nick125if I had a fax machine plugged in, I wouldn't mind sending you one..but, i dont :(
05:05.39surfduenick125, i do
05:05.43surfduebraodvoice.com
05:05.52nick125surfdue: you didn't see the last requirement
05:06.00surfduemoney order?
05:06.02Strom_Cnick125: why the money order requirement?
05:06.12nick125Strom_C: no credit card
05:06.24Strom_Cget one of those prepaid credit cards tehn
05:06.33nick125broadvoice prohibits them
05:07.21Strom_Cwell, if you absolutely insist on not using a credit card, then you could always go back to the 1950s
05:07.31asterisk-dudguys, if I have a variable 2030 = Zap/1, I have a dialplan that needs to take in EXTEN 2030 and return Zap/1 (the value of 2030)
05:07.36asterisk-dudhow can i do that
05:07.46surfduenick125, try lingo
05:07.51surfduenick125, call them up
05:08.08surfduewhats a moneyorder again?
05:10.56*** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net)
05:10.56nick125lol
05:11.15surfduehttp://www.meritcall.com/neworderinfo.html
05:13.13kernel20would it possible to put password to number that dials that starts on _951X
05:13.20kernel20if that number has been pressed it will asked for password
05:13.22kernel20before a call can be stablished
05:13.23kernel20?
05:14.52surfduecya all
05:16.19*** part/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com)
05:18.07Strom_Cwhat do you know - faxing over voip works beautifully when you're only on a LAN :)
05:21.17h3x0ryewah
05:21.27h3x0ri had a customer doing 672 calls on a single pentium 4 of faxing
05:23.33kaldemarkernel20: take a look at application Authenticate.
05:23.59*** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz)
05:24.33iceyphey guys, i just got 2 cisco 7940's with skinny client on them, im trying to update to P003-08-2-00 is it possible to do a dirct update or do i need a lower version first?
05:24.40kernel20ok htanks
05:24.46kernel20another one
05:25.22kernel20i wanted to have all checking of voicemailmain start with #plus_sip_number
05:28.11kernel20ie #11001
05:28.16kernel20it will check for voicemail
05:28.18kernel20any ideas?
05:29.09kernel20exten => #11001,2,Macro(mymail,11001@myvoicemail)
05:29.12kernel20wont work
05:29.18kernel20any ideas?
05:31.54kaldemaris that extension not working or your macro?
05:32.25kernel20yeah
05:33.42X-Robgwg
05:33.43X-Robheh
05:33.52kaldemaryeah?
05:33.58X-Robgotta love either/or questions that are answered by 'yeah'
05:34.46kernel20yes
05:34.55kaldemarthey're the best.
05:36.10kernel20?
05:37.04kaldemar- do you want to take the read pill, or the blue pill? - yeah.
05:37.19kernel20both
05:37.33kernel20anybody?
05:37.36eipithere's anyway to configure the iaxy to register into named server? or only by ip?
05:37.57kernel20i want to have my voicemail checking that users should start #+their nummbert
05:38.10kernel20?
05:38.12kaldemartry sip debug to see if your phone is actually sending # to asterisk. a good way to start.
05:38.19[shodan]is this any good => http://www.i-rocks.com/2004/product/IR-2500.htm ?
05:38.45kernel20i am on asterisk -vvvc now
05:38.52kernel20i can see any logs pertaining it
05:39.11kernel20i can't see any logs pertaining it
05:39.12dlynes_homeCunningPike: almost got one of those blood sangoma cards working now :p
05:39.17dlynes_homes/blood/bloody/
05:39.27CunningPikedlynes_home: Having fun yet?
05:40.11dlynes_homeCunningPike: are you kidding?  ie still can't do png alpha layers?
05:40.16kernel20i want to have my voicemail checking that users should start #+their nummber
05:40.19kernel20any ideas?
05:40.24kernel20dlynes_home:?
05:40.26CunningPikedlynes_home: Nope - not with out a custom filter
05:40.38dlynes_homethat's pretty ghey
05:40.45CunningPikedlynes_home: Ya think???
05:40.55dlynes_homepng's have been a standard for how long now?
05:41.00kernel20is this correct
05:41.01kernel20exten => _#11001,2,Macro(nkymmail,11001@nkymvoicemail,#11001)
05:41.07CunningPikedlynes_home: Since the flood - Noah had some
05:41.08*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
05:41.13kernel20exten => ${ARG2},1,VoiceMailMain(${ARG1})
05:41.24dlynes_homeman
05:41.27dlynes_homei need to buy more dishes
05:41.29kernel20?
05:41.33harryvvdishes?
05:41.39CunningPikedlynes_home: TMI ;)
05:41.40dlynes_homeI keep running out of clean dishes all the time :p
05:41.42kernel20?
05:42.02harryvvohh I thought it was your wife that was PMsing on you and thoughing the dishes at you.
05:42.08kernel20>
05:42.09kernel20?
05:42.13dlynes_homenah...not married, thankfully
05:42.16harryvv:)
05:42.29dlynes_homeotherwise i'd have been pushed off a cliff a long time ago :p
05:42.41harryvvahh yea woman are easy to anger
05:42.44kernel20?
05:42.52kernel20?
05:42.53kernel20?
05:42.58dlynes_homeyeah...i feel sorry for cunningpike
05:43.04CunningPikekernel20: There's something wrong with your keyboard
05:43.06harryvvIve been bussy all day and now im resting with a ache body and a headache.
05:43.12dlynes_homehe's irish, so i would imagine his wife is too :)
05:43.15CunningPikedlynes_home: You've never met my wife ;)
05:43.28CunningPikedlynes_home: 10 years married last week
05:43.29harryvvdlynes your in van right?
05:43.34dlynes_homeCunningPike: ah...your wife's not irish?
05:43.40CunningPikedlynes_home: Yes, she is
05:43.51dlynes_homeSo she doesn't have a nasty temper?
05:44.14dlynes_homeI always hear all this stuff about how irish woman have nasty tempers :)
05:44.19dlynes_homes/woman/women/
05:44.34kernel20?
05:44.35dlynes_homeharryvv: correct
05:44.35kernel20?
05:44.35harryvvI knew a irish male that had a temper
05:44.36kernel20?
05:44.37LoRezWarning: `kernel20' seems to be spamming, please discontinue or kills/klines will be issued.
05:44.37RezWarning: `kernel20' seems to be spamming, please discontinue or kills/klines will be issued.
05:44.37kernel20?
05:44.43kernel20?
05:44.51dlynes_homelol
05:44.53dlynes_homeit's about time :)
05:44.58harryvvher wife danced with a rock hudson type and he steamed. Then the divorse followed.
05:45.03dlynes_homeI had him ignored, so I didn't even notice he was spamming :)
05:45.05harryvvhis wife I mean
05:45.19mitcheloc*sigh* AND D-Fender told me to be nice to him earlier....
05:45.26dlynes_homemitcheloc: lol
05:46.00dlynes_homehe's apparently banned from most of efnet
05:46.18dlynes_homeCunningPike: i didn't even think you were that old yet
05:46.27dlynes_homeCunningPike: ten years married...damn
05:46.31mitcheloclol, if he made it into that many chat rooms you'd have thought he'd pick up something?
05:46.35harryvvI am a little worried about this terrorist capture. Some times I work at key fedeal sites on the side that are likly targets here in BC.
05:47.08CunningPikedlynes_home: Yup - 10 years. I'll be 40 next year
05:47.17*** join/#asterisk rvhi (n=rv@66.175.65.89)
05:47.20dlynes_homeCunningPike: damn...didn't think you were anywhere near that old
05:47.29dlynes_homeCunningPike: thought you were early to mid 30's
05:47.29mitchelocdlynes_office: happy snappin?
05:47.38harryvvanyway night all
05:47.41CunningPikeharryvv: Much more likely to be a) run over in a crosswalk b) shot by a gangsta
05:47.52CunningPikedlynes_home: Well, thanks :D
05:47.53*** join/#asterisk denon (n=denon@synapse.subneural.net)
05:47.53*** mode/#asterisk [+o denon] by ChanServ
05:48.04dlynes_homelol
05:48.16dlynes_homemitcheloc: not until you make it compatible with linux :)
05:48.57dlynes_homemitcheloc: and even then, i'd probably only get limited use out of it
05:49.06dlynes_homemitcheloc: my boss would probably get a lot of use out of it though
05:49.59mitchelocdlynes_home: well hell, get your boss to use it, and i'll get it to work on linux, good deal?
05:50.30dlynes_homeheh
05:50.49dlynes_homemitcheloc: btw, do you have post dial dtmf support?
05:51.20mitchelocit was just added to asterisk managment api
05:51.30mitcheloci'll add it in, i need to figure out a clean interface though
05:51.54mitchelocthere are so many features i have to add, it's going to get stuffy looking =(
05:52.26dlynes_homeso you need to have access to the asterisk manager socket, for that to work?
05:52.36mitchelocyes sir
05:52.53dlynes_homeso if the server is only listening on the localhost, you won't be able to interface
05:53.05mitchelocno sir, you'll have to fix that
05:53.34dlynes_homewell, what i'm saying is that it's a no go on my main server then
05:53.40dlynes_homebut all the office pbxes it would be fine for
05:54.16mitchelocokay, well, a step at a time, i'm working on improving it
05:54.21dlynes_homeare you able to lock the administrator settings on the dialer?
05:54.26*** join/#asterisk postel (n=jp@unaffiliated/postel)
05:54.31dlynes_homei.e. through a password protect?
05:55.36mitchelocwell manager api already requires a username/password
05:55.39mitchelocis that what you mean?
05:55.43mitchelocbut it needs to be on the client
05:55.46mitcheloclike an instant messenger
05:56.09dlynes_homemitcheloc: no...i meant the settings for the client to know which server to connect to, what port, what server for the manager api, what port for the manager api, all that kinda stuff
05:56.31*** part/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
05:56.43dlynes_homemitcheloc: a lot of people running asterisk don't want their customers to know its asterisk
05:58.06*** join/#asterisk astar` (n=astar@ANantes-154-1-48-51.w81-53.abo.wanadoo.fr)
05:58.15mitchelocdlynes_office: are you referring to where it says "Asterisk Management API"?
05:58.26mitcheloci could come up with something to disguise that...
05:59.02dlynes_homemitcheloc: yeah, but also we don't usually like to have the end user monkeying with the settings, either
05:59.20dlynes_homeonce they start doing that then we have to come down and play tech support guys
05:59.31dlynes_homenot fun
05:59.49mitcheloci understand, i've gotten mixed reactions on that, some people like it, others don't, i'm thinking of a hybrid way to load and lock
06:00.06mitchelocsort of, user types in username/password, settings downloaded and locked in, and the user can't see or modify them
06:03.37dlynes_homemitcheloc: or maybe an oem toolkit to allow you to customize it?
06:03.50dlynes_homemitcheloc: including slapping your own logos on it?
06:04.28mitchelocdlynes_home: i work with resellers, all you have to do is let me know what you need and we can work something out
06:04.39mitchelocalso, there is a deploy example i worked on, so you can set up a logon script
06:04.50dlynes_homeah
06:06.28*** join/#asterisk MatsK (i=MatsK@83.233.97.229)
06:07.42mitchelocdlynes_office: can you read this, and try the download at the bottom? http://www.snapanumber.com/Support/Forums/tabid/58/forumid/1/threadid/69/scope/posts/Default.aspx
06:07.45mitcheloclet me know what you think
06:08.37*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
06:09.24dlynes_homemitcheloc: is that a windows extension or is it the linux version?
06:09.35mitchelocwhich extension?
06:09.44dlynes_homedeploysnap.zip
06:09.56mitcheloc.zip is a zip file?
06:10.08mitchelocheh, but i don't think thats your question?
06:10.31dlynes_homedeploysnap.zip is for use on windows or linux?
06:10.33*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
06:10.58mitchelocit's for use on windows
06:11.01dlynes_homeah
06:11.04dlynes_homethe reason i'm asking
06:11.07mitcheloclike on a logon script, it's only an example, you could do it 100+ ways
06:11.14dlynes_homeis because i don't have access to a windows box here
06:11.35dlynes_homeall my machines at home are either linux or solaris
06:11.38mitchelocwell, just open the zip and look inside the logon.bat file, it should explain what you are asking
06:11.52dlynes_homeah...that's what you were pointing me to then
06:14.02*** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it)
06:15.58dlynes_homemitcheloc: yeah...pretty simple stufff
06:16.17dlynes_homemitcheloc: it could be a lot more elegant in perl though, and then you wouldn't need sqlite.exe :)
06:16.18mitchelocit's just an sql database file
06:16.29mitchelocyep, well, so you have part of your answer
06:16.42mitcheloci was thinking of a name/value called locked/yes and hidden/yes
06:16.53mitchelocso that it can't be updated in snap or seen (depending on yes/no)
06:17.08mitchelocof course any scripter can do this, but most corporate users can't...it's a quick fix for now
06:17.29dlynes_homemitcheloc: well, most of our customers are small offices
06:17.48dlynes_homethe chance of one person in the office having a clue enough to go digging into the registry is pretty remote
06:18.02dlynes_homeand even if they did, they probably know enough to know what they're doing with the settings
06:18.52*** join/#asterisk syle (n=blah@unaffiliated/syle)
06:19.22mitchelocdlynes_home: very true, so is there anything missing then to help make this deployable *now*?
06:19.36mitcheloc(btw, there is a very cool auto-update feature in snap, so you don't have to maintain it on your client's computers)
06:20.45dlynes_homecool...have you used amsn?
06:21.09dlynes_homenone of the stuff i mentioned would stop it from being deployable
06:21.35dlynes_homethey were just things that are pretty much close to being a necessity for offices
06:21.35mitchelocamsn? or msn? i use msn heh
06:21.36dlynes_homeamsn
06:21.48dlynes_homeIt's a TK msn messenger client
06:21.57dlynes_homeit runs on linux
06:22.04dlynes_homeit has an autoupdate feature, too
06:22.06mitchelocah, wikipedia entry, yep, no i don't, why?
06:22.13mitchelocah, cool
06:22.14dlynes_home~wiki amsn
06:22.30dlynes_homeyep...same amsn
06:22.32mitcheloci use trillian, i just hover over "amsn" and it tells me ;), i thought it was a typo though
06:23.26dlynes_homeand unlike msn where you have to wait a while before hitting nudge
06:23.35dlynes_homewith amsn you can hit it repeatedly as fast as you like
06:23.41dlynes_homeheh
06:23.45mitcheloclol, nice feature
06:24.15dlynes_homegood for those clowns that start talking to you, want you to help them, and then seem to forget they were talking to you
06:25.55*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
06:26.07mitcheloc;), so you pointed out amsn to tell me it updates? =P
06:31.22*** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr)
06:31.25dlynes_homeyeah :0
06:31.54dlynes_homeKiraly Parcina!
06:33.48mitchelockiraly parcina?
06:34.14dlynes_homehungarian
06:34.47dlynes_homekiraly is hungarian for king :p
06:35.08tparcinadlynes, i'm not from hungary, i'm from croatia! :))
06:35.16dlynes_homeoh yeah :(((
06:35.18mitcheloctake that!
06:35.32dlynes_homeHis tld keeps throwing me off :((
06:35.37tparcinabut in ast one year i have been for three weeks in budapest, so maybe it counts :))
06:36.20tparcinaand i find hungarian girls weary sweet and nice, who nows maybe i'll become hungarian :))
06:36.45dlynes_homehahaha
06:37.04dlynes_homei dunno
06:37.13dlynes_homei've seen the odd croatian chick that's pretty hot, too
06:37.25tparcinahow are you? is it around midnight at your place?
06:37.40*** join/#asterisk parag7732 (n=root@83.110.214.174)
06:37.41tparcinawhat old croatian chick?
06:37.42dlynes_home11:30pm, yeah
06:37.49dlynes_homeodd, not old
06:38.02mitchelocdlynes_office: don't lie, you like old chicks
06:38.26dlynes_homewhat do you consider old, mitch?
06:38.45mitchelocanything over 30 is over the hill
06:39.01dlynes_homeah then yeah...i definitely like old chicks
06:39.02tparcinayes, i like one old cihck alsoo - 22 years old :)
06:39.21dlynes_homei like chicks around mid 30's or so
06:39.29drraywomen are like dog crap, the older they get, the less messy it is to pick them up
06:39.36mitchelocheck, 16-30, thats my limit
06:39.38mitcheloc** 18
06:39.44tparcinaas you go older the border young/olod is moving...
06:40.03mitchelocdrray: nice
06:40.38tparcinayes drray, i have never heard that one before, it's good
06:41.01dlynes_homeyeah..younger chicks have a lot more problems
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06:42.08tparcinayes, they have problems with themself!
06:42.48dlynes_homeyeah...where as old guys just play with themself!
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06:44.27mitcheloctparcina: like this... mv -rf / /dev/null
06:44.27dlynes_homelol
06:44.35mitchelocdlynes_office: ygpm ;)
06:44.42dlynes_homeygpm?
06:44.48mitchelocprivate message
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06:54.13tparcinaMitcheloc, when you know so much about disk formating :)) give me a hand with this one
06:54.25tparcinawhen i installed FC4 i have formated two hdd with disk druid. both disk head two partitions, first one from 1-3916 and another from 3917-4569. then i taked out one disk, remouved all partitions with partition magic and then i have put it back to fedora machine. then i have formated disk with fdisk and i tried to create the same partitions but i have failed. this is what fdisk showes me - http://pastebin.ca/62450 so, how to format disk so it's the sa
06:54.56mitcheloctparcina: it was a joke,  i'm heading to sleep though, sorry
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06:56.33P-NuTHi all, has anyone got an x100p working under ubuntu?
06:57.04tparcinamitcheloc, i know it was a joke, that why i have put :)) - anyway, you can sleep tomorow :))
06:57.31drrayit's already tomorrow here
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07:20.29littleballhello, i don't think it is a good idea to store all sip user configuration in sip.conf file. if there are 10,000 users, the file will be too big.
07:20.36littleballand not easy to maintainence
07:20.44littleballany suggestion?
07:21.43sylemysql realtime
07:26.54littleballsyle, thanks. another questions about the deployment of media relay system. because most of sip phones will be behind firewall, i am planing to use a few asterisk box as media relay server. refer to http://mit.edu/sip/sip.edu/started.shtml
07:27.13littleballArchitecture figure
07:27.59parag7732Can anybody suggest small billing System for a small company ????
07:28.04littleballassuming all sip phones are behind firewall. and a few asterisk media gateway is needed.
07:30.24parag7732Can anybody suggest small billing System for a small company ????
07:30.29littleballwhat is exact mean of  "nat=yes" in sip.conf ?
07:35.54qdk_parag7732: have you looked at voip_info.org?
07:36.15parag7732yes qdk
07:36.17parag7732I tried
07:36.21parag7732a2billing
07:36.23parag7732is there
07:36.30parag7732but that is prepaid and postpaid billing
07:36.34*** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net)
07:36.36parag7732I need simple billing system
07:36.47parag7732for a small company for mobile phone usage
07:37.21qdk_littleball: it means that your SIP-phone is able to traverse the NAT its behind.
07:37.22Shaun2222are there any linksys routers or anything with built in stun servers?
07:38.08qdk_Shaun2222: are the any routers with buildin STUN servers?
07:38.22Shaun2222i dont know
07:38.31qdk_parag7732: what is the besides pre and post billing?
07:38.40qdk_Shaun2222: i would say no.
07:39.13parag7732like if in the company people calls to mobile or landline I need to add Rate field
07:39.35Shaun2222i have one of those zyxel p-2000w phones, it's going to be behind a NAT where the SIP server is on the public network.
07:40.23qdk_parag7732: make a simple script parsing the CDR.csv fil and the dump whatever you wish to bill.
07:40.28Shaun2222it says for nat it supports outbound proxy, stun or manual configured wan/sip add
07:41.50*** join/#asterisk dec (n=tom@ppp169-75.lns3.adl4.internode.on.net)
07:42.27qdk_Shaun2222: A STUN server i located somewhere on the internet, probably on the same network as your the * you register to.
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07:42.49Shaun2222whats the point of that.
07:43.12Shaun2222i need these phones to work inside a home users little crap NAT network
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07:43.26Shaun2222connecting to a * server out on a public ip
07:43.33qdk_Shaun2222: then dont use STUN?
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07:43.46Shaun2222qdk_: i though that was needed for NAT?
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07:43.59ghenryHardware and Asterisk version for 5 9's uptime?
07:44.05ghenryAnyone at that level?
07:44.12qdk_Shaun2222: no, i have a lot of SIPs behind nat without any problems.
07:44.36Shaun2222qdk_: so do i but they are all cisco 7960 phones
07:44.38ghenrybrb
07:44.40qdk_ghenry: oone can tell you... dependS!!!!
07:44.48Dico_is anybody knows the difference between 'hint' and ' device status' ?
07:44.51Shaun2222this little zytel thingy doesnt appear to work
07:45.10Shaun2222i also remember somebody says the cisco phones had some built in stun server or somthing
07:45.11qdk_Shaun2222: ok, i dont have any Cisco equ.
07:45.33Shaun2222i had to tell the cisco phones that they where on a NAT
07:45.49Shaun2222for them to work
07:45.50Dico_oups. sorry, i forget to say 'hello' ^^. My apologize : hello world :)
07:45.58qdk_Shaun2222: i dont know much about STUN, but im pretty sure that it doesnt make sense to have the STUN behind the NAT.
07:46.06Shaun2222ok
07:46.42ghenryHa, I thought so qdk_
07:46.54qdk_Dico_: hehe... in what situation are you talking about?
07:46.58ghenryWas looking at IBM BladeCenter T
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07:47.32Dico_humm for the 'away' ,  'busy' and so on, using subscribe and notify
07:47.35ghenryis it worth getting the enterprise version from digium or just using the OSS one? I would have thought they would be near enought the same
07:48.02*** join/#asterisk RoyK (n=roy@80.239.107.70)
07:48.37qdk_ghenry: well the specific hardware is only a matter of load and how many you have i your cluster to make sure of the 5 9's.
07:48.46Dico_qdk_,  actually, at first i thought hint and device status were the same stuff, but after having a look in manager.c, it appears to be a difference ...
07:49.01ghenryqdk_: yup
07:49.22qdk_Dico_: ok, dont know the difference. sorry.
07:49.31Dico_ok
07:50.03qdk_ghenry: you could also use different cards...
07:50.09Dico_where is my oej ? ^^
07:50.30RoyK-- --- .-. -. .. -. --.
07:50.37RoyKDico_: prolly still sleeping
07:50.47qdk_ghenry: i would say that a strong setup is made up of different hardware and a large "cluster" solution.
07:51.18Dico_RoyK,  ok
07:51.20RoyKfive nines and asterisk is probably like riding a $100 bike in 200Mph
07:52.02qdk_RoyK: in a SPoF setup, yes.
07:52.03ghenryqdk_: Thanks
07:52.20ghenrySPoF?
07:52.32qdk_Single Point of Failure
07:52.55RoyKsure, but wtf will you put in the front to gain five nines?
07:53.06qdk_robl^: infront?
07:53.15ghenrydoh, thanks
07:53.25qdk_RoyK: ups, for you. ;-)
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07:54.40qdk_RoyK: I would probably make a BSD with CARP setup, so i would have 1 IP for a many *'s, providing failover and/or loadbalancing.
07:56.05qdk_my current setup is 2 single and independet frontend servers with automatic failover to and from my peer and to and from my backend servers.
07:56.18RoyKqdk_: I have an idea, no (or almost no) code yet, but take a look: http://sitsotd.org/
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07:58.25zoaroyk, i actually have that
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07:58.28zoarunning in production
07:59.35zoawell kinda anyway
08:00.07BugKhamhow to silently reload asterisk?
08:00.13RoyKzoa: you do?
08:00.16zoayes
08:00.26zoai have such a thing for a year
08:00.33zoaits part of my asterisk cluster
08:00.40zoaneed to go to meeting now
08:00.46zoaand no i will not give it to you :)
08:00.57zoaits part of our isp asterisk cluster solution
08:01.04decI haven't been following development too closely, but how is the PLC going for SIP? Anything stable available yet?
08:01.07zoathat will probably never see the day of light
08:01.14zoadec: its in trunk
08:01.32deczoa - stable enough for production?
08:02.52zoayes
08:02.56zoawell depends
08:02.58zoatry it and see
08:03.04decOK. :)
08:04.27qdk_So sitsotd removes the problem of lost sessions, when a server(in a cluster) goes byebye?
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08:06.38RoyKzoa: btw, it might seem the new patches from slav works
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08:21.16parag7732Failed to execute SQL : SQL CREATE TABLE rate_engine_rate ( rate_id integer primary key auto_increment, route_id integer NOT NULL references rate_engine_egress(route_id), type varchar(3), country varchar(40) NOT NULL, extra varchar(40), prefix varchar(10) NOT NULL, active_date date, expires_date date, firstperiod integer NOT NULL, periods integer NOT NULL, startcost float, periodcost float, trialcost float, ); failed : You have an error in your SQL synta
08:21.43parag7732help please
08:22.01macTijndoes this look like #sql ?
08:22.15parag7732why
08:22.31parag7732got it
08:23.48Greek-Boysince my D-Link DES-1526 is not providing power to my cisco phones (7912 and 7960), can I make my own custom patch panel and provide power to it?
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08:29.14bmg505I do that with wrt54GL routers, 2 wires, just be carefull of the polarity
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08:34.30Greek-Boywhich wires though?
08:35.18RoyKdoesn't matter
08:36.46opus_hello
08:36.50opus_hello
08:36.58opus_is there a built in unix backtrace() command?
08:39.02stephane_re
08:40.22RoyKopus_: gdb asterisk core.xxxx and then 'bt'
08:40.36opus_RoyK I want to do it programmically
08:41.39RoyKwhy?
08:42.34opus_debugging a deadlock in of course a multithreaded app
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08:50.04Greek-Boyso u guys are saying i must put power on any 2 wires to power up the cisco phone?
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08:56.07RoyKopus_: i thought gdb could do that as well
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08:56.46RoyKopus_: thead apply all bt
08:56.48RoyKiirc
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08:59.03Greek-Boywhere can i get the PoE pinouts for cisco 7912?
09:00.36x86stephane_: re
09:02.03tparcinaI have done it without those guys in #fedora or ##linux channel :)) software RAID-1 works for me :)))
09:02.48tparcinaGreek-Boy: as far as i know, cisco 7912 uses prestandard PoE, and if you want to use it you'll have to buy cisco PoE switch
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09:03.25tparcinahave you discuss anything interesting while i was building RAID?
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09:04.33Greek-Boytparcina does that countf or the 7960 too?
09:06.51tparcinaGreek-Boy: I think so. but 7961 uses standard PoE, so it can be used with any poe switch
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09:09.24tparcinaGreek-Boy: to tell you the thrue, i have 1 7905 that works with standard POE and 6 7905/7912 that doesn't. and i have one 7920 that works with standard POE and 4 7940/7960 that doesn't. so, there is (at least) one hardware revision that works with POE, but most of them don't.
09:09.56tparcinaso, if you would like to use 7905, 7912, 7940 and 7960 on POE, buy cisco POE switch.
09:10.09Greek-BoyYeah but I hard dumb terminals power them up
09:10.17tparcinaand if you would like to use cisco phones on any other POE switch, then buy 7941 or 7961
09:10.21Greek-Boyso i thought about building my own supply as a quick solution
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09:11.24tparcinai have counted, and it would cost me more to buy non-cisco POE switch + terminals (or to buy cisco poe switch) then to buy power cubes for phones. so i bouth power cubes
09:11.51tparcinaand power cubes work for sure :))
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09:25.02SheriF_WorKcan anyone point me to a doc about SIP protocol and how it works ?
09:25.33littlejohnSheriF_WorK, read the rfc
09:25.42maagic4th match from google: http://en.wikipedia.org/wiki/Session_Initiation_Protocol
09:25.47Makenshihttp://www.faqs.org/rfcs/rfc3261.html
09:26.05sgnome`SIP Demystified is an excellent book on the topic from Addision Wesley
09:26.08key2SheriF_WorK: SIP RFC
09:26.10SheriF_WorKcool thx guys
09:27.40SheriF_WorKi just need to know exactly how SIP works .. for ex i have an asterisk and 2 users registering to asterisk from 2 countries and they are talking right now ... so that all the traffic is passed throw asterisk ? or asterisk just opend the port and give infromation to both clients and they open some ports between eachothers and working without needing asterisk help anymore after the session is initiated ?
09:30.39RoyKhttp://comics.linuxzealot.net/Dilbert/2006.06.06.gif
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09:42.55tparcinaRoyK, stop joking around, this is serius channel :))
09:43.19tparcinabtw, RoyK, do it, do it, do it :))
09:46.10key2When asteirsk is in realtime mode
09:46.11clive-i hear they just released asterisk business versi0n supporting aculab and dialogic crads on asterisk
09:46.20key2if I do a reaload, I lose all the queue member, is it normal ?
09:46.22Dico_oej, hello;  sorry for annying you. I had a look in manager.c . Can you tell me the difference between 'hint' and 'device status' ? Is it the same thing a different level ?
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10:10.26frk2dudes- my grandstream has the screen blanking problem..... is this supposed to be normal?
10:18.46kmilitzerIs there a way to disallow SIP-REFER-requests in asterisk?
10:26.32Dico_ok, nvm
10:26.34Dico_cu all
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10:33.38clive-mike
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10:44.08mmmmmToopHi Clive..
10:45.32fourcheezehow do I get one asterisk to dial another and take itself out of the loop? I've been looking at Transfer() but that doesn't seem to work for me
10:46.00viperdudefourcheeze: allow reinvites
10:46.11opus_not sure, but in theory you can use chan_sip's reinvite feature with transfer to get out of the loop
10:46.23fourcheezeviperdude: does that mean I have set the second * as a user?
10:46.43fourcheezeATM I'm just trying Transfer(SIP/ext@host)
10:46.49viperdudeerm not sure I think it will also work with peer
10:46.53fourcheezethat just seems to die
10:47.14opus_thats asterisk!
10:47.19opus_why do you think they call it asterisk
10:47.40fourcheezebecause it's what you explete when it drops calls?
10:47.47opus_exactly
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10:48.08fourcheezewell, I'm about 95% happy with it's reliability so far
10:48.22opus_yeah
10:48.35fourcheezeso if I was to put in a normal dial() and allow reinvites on the destination
10:48.42fourcheezethen it should lose the first * ?
10:49.17viperdudeshould do if the calling party excepts reinvites
10:49.39fourcheezeok, I'll give it a go
10:50.00fourcheezenow assuming this works - how many calls per second can a single * handle like this?
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10:50.14opus_4-5
10:50.18fourcheezeor rather how many should I leave it before starting to think of using SER
10:50.34fourcheeze4-5 a second will probably do
10:50.35opus_depending on your application
10:51.15fourcheezesuppose I needed a * to stay in the middle (i.e. the calling party didn't accept reinvites) how many of those could * pass through at once?
10:52.13opus_just passing a call through maybe 100-200
10:52.30opus_doing something with that call like IVR , i wouldn't trust it with more then 80 calls
10:52.31fourcheezeso that's not really an option for scaling
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10:52.55opus_even on the fastest systems. it will use only like 2% cpu, but most of the code is to fucked up and not threadsafe
10:52.58fourcheezeopus_: how did you guess that I'm building an IVR box?
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10:53.17opus_you should hire methen:)
10:53.21opus_i build large scale ivr systems
10:53.24fourcheezeaha
10:53.28zoaroyk, all he did was disable debugging logging
10:53.28fourcheezehow much do you charge?
10:53.35zoaso that might have made the difference for you
10:53.59opus_fourcheeze depending on the type of work of couse
10:54.07opus_fourcheeze and currency ?
10:54.45fourcheezeYen?
10:54.50fourcheezeRupee?
10:54.53fourcheezeRouble?
10:54.58jgooCoffee beans?
10:55.05opus_i try to be just under the best people but there is always some dumb ass that comes up with the $3k bid that I have to compete with.
10:55.16opus_USD
10:55.32fourcheezeto be honest this isn't going to be large-scale particularly
10:55.58jgooI got an email from colognechip regarding BRI cards, if anyone wants a copy
10:56.10jgoolinks, info, drivers, very nice email
10:56.11zoajgoo
10:56.12zoagimme
10:56.26jgoomsg me email, ill forward
10:56.29fourcheezeI just want to divert incoming calls to an IVR box and then back into the main system
10:56.39opus_fourcheeze, simple enough
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10:57.25esethi, are there any really basic web interfaces for creating/appying for sip accounts for *?
10:58.13opus_i gotta go to bed, but i will PM you my email if you want to jump start your project
10:58.24fourcheezeopus_: should be fine as long as I can actually get a transfer to work!
10:58.51opus_canreinvite is also a global setting , even though it isn't documented anywhere :)
10:59.01opus_canreinvite=yes at the top of sip.conf will be respected
10:59.06opus_in my case I wanted canreinvite=no
10:59.09opus_take it easy
10:59.13fourcheezeok, l8rz
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11:19.03MrChimpybugger. when i trigger a setvar from my AGI script it doesn't produce any AMI event - presumably because the one you'd monitor is Newexten and because I'm using AGI to do the SetVar I'm not actually in a dialplan
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11:54.42nessbrettnem: I tried the "sip show user" today - doesn't work (no such command). I managed to do it (more or less) by using voicmemousers, but it is a bit hacky...
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12:00.47key2someone has tryed the queue realtime ? for some reason, after 30sec, the user in queue gets a hangup
12:00.49key2any idea ?
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12:04.43s-ndh-cwhat hardware would i need to route calls between asterisk and my existing pbx?
12:05.07s-ndh-ci guess if i use a single isdn line i can only transfer 2 calls at the same time right?
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12:12.35*** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar)
12:13.15*** join/#asterisk tsurk0 (n=tsurko@85.187.160.157)
12:14.03key2s-ndh-c right
12:17.18s-ndh-cso i would need some special card or something , cause i need atleast 4 calls at the same time between existing pbx and asterisk
12:17.27s-ndh-cbetter more
12:17.30s-ndh-c:)
12:18.23*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:20.20*** join/#asterisk coppice (n=chatzill@44.199.17.210.dyn.pacific.net.hk)
12:21.47eipithere's anyway to make iaxy connect thru a name server to asteriks?
12:23.19*** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com)
12:25.15*** part/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr)
12:25.26*** join/#asterisk ToTo (n=ToTo@81.174.33.2)
12:25.47*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
12:27.55*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6)
12:29.04*** join/#asterisk uwe (n=uwe@dogbert.palnet.com)
12:29.09RoyKhm. how is an e1 loopback plug wired?
12:32.05*** part/#asterisk ness (n=Tom@pppin-4e-b6.pop-kaltenengers.rz-online.NET)
12:32.28I-MODRoyK, pin 1 to pin 4 and pin 2 to pin 5
12:32.59I-MODeipi, iaxys dont do DNS resolution, IP addresses only
12:33.04key2RoyK: the RJ45 ?
12:34.06*** join/#asterisk Kis (i=vlad@p5080D187.dip.t-dialin.net)
12:35.24RoyKyeah
12:35.26RoyKfound out
12:35.40RoyKI-MOD: thanks
12:35.45I-MODnp
12:37.27kdz13|goneso then, SIP fixes all my dtmf issues
12:41.04*** join/#asterisk Ariel_ (n=Ariel@70.46.87.158)
12:41.28key2RoyK: do you know why when I have a user that gets into the queue, if he stays more than 30sec, he gets a hangup
12:41.28key2?
12:42.25RoyKbecause of timeout=30 ?
12:43.34key2RoyK: I have a timeout = 300
12:43.37key2but still
12:43.45key2RoyK: it's in realtime
12:46.03*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
12:46.32RoyKkey2: dunno. set debug = lots
12:47.01MrChimpyis there some existing grouping mechanism or should I be implementing this at the dialplan or AGI level?
12:47.02[TK]D-FenderMrChimpy : Shove the dial lines back to back.
12:47.16MrChimpywhat do you mean tkd?
12:47.24MrChimpyoh, in the dialplan?
12:47.43MatsKRoyK: Its not RJ45 its RJ48, http://www.arcelect.com/RJ48C_and_RJ48S_8_position_jack_.htm
12:47.45[TK]D-FenderMrChimpy : just put the steps back-to-back.  If it connects on the first it'll never hit the 2nd after a hangup, and all is good.  On failure of first, who cares about the reason?  Just dial the 2nd.
12:47.50[TK]D-FenderMrChimpy : Yes.
12:48.28*** join/#asterisk aze (n=aze@ACayenne-101-1-12-98.w81-248.abo.wanadoo.fr)
12:48.29MrChimpyok, cool. that'll do for dialplan apps. I'll do something fancier in AGI.
12:49.42[TK]D-FenderMrChimpy : Its 1 line!  Would you like your elephant gun sir?
12:49.43coppiceRJ48C is just RJ45, with the 5 crossed out, and 8C written in crayon
12:49.44*** join/#asterisk Bert- (n=bert@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr)
12:49.47Bert-hello there
12:50.06key2RoyK: where do you set debug=lost?
12:50.17MrChimpytkd: this is a 1000 line IVR system. elephant gun required.
12:50.31I-MODcept RJ48C can accept 4, 6, and 8 wire connectors while a RJ45 can only accept 8 wire connectors
12:50.35RoyKkey2: at the console, set verbose 9, set debug 9, and try again
12:50.45MrChimpyneeds to be as resilient as I can make it
12:50.51RoyKkey2: make sure you log debug info somewhere (logger.conf)
12:52.45Bert-lol
12:53.05Bert-will Next versions of asterisk support the new cpu instruction 'maybe' ?
12:53.06Bert-huhu :)
12:53.17[TK]D-FenderMrChimpy : Big deal... I had 347 worth of STDEXTEN macro and associated setup contextx for it :)
12:54.02Bert-It will very funny to see malloc() returning 'maybe' :d
12:54.19SheriF_WorK[TK]D-Fender: hey man ;-) ... i have something in mind and i need u to show me the light .. now about the SIP protcol it's self .. the media packets and the all the talking packets is send directly between the to endpoint bypassing the SIP server " asterisk " right ?
12:55.00I-MODSheriF_WorK, only if you allow reinvites
12:55.12[TK]D-FenderSheriF_WorK : Depends.... SIP always passes through the server, RTP is variables (canreinvite)
12:55.17*** join/#asterisk Killa200 (n=killa200@adsl-153-147-238.cha.bellsouth.net)
12:56.02SheriF_WorK[TK]D-Fender: if my softphones in another NAT networks " each end in a NAT " i should use canreinvite = no .. other way it will not work.
12:56.20SheriF_WorKi'm reading in the RFC for like 2 hours and my brain is damged already :D
12:56.36[TK]D-FenderSheriF_WorK : No reinvites "or else" .....
12:57.08SheriF_WorK?
12:57.38[TK]D-FenderSheriF_WorK : describ the full path of the call like : phone -> nat -> internet -> * -> nat -> otherphone
12:58.10SheriF_WorKso canreinvite = yes " the media and the RTP will bypass the asterisk " canreinvite = no the RTP will pass throw the asterisk ?
12:58.14*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
12:58.45[TK]D-FenderSheriF_WorK : Correct.  Only if set on BOTH ends of the call and assuming conditions allow
12:58.55SheriF_WorKoh ok phone -> nat -> internet -> * <-internet <- nat <- otherphone
13:00.00[TK]D-FenderSheriF_WorK : Forget reinvites... thats not going to work.
13:00.30*** join/#asterisk Katty (n=angela@64.82.232.54)
13:00.58SheriF_WorK[TK]D-Fender: i didn't get it sorry :-s what will not going to work ? this setup works with me using canreinvite = no ..
13:01.32[TK]D-FenderSheriF_WorK : Thats what I was saying... it will not work WITH reinvites
13:01.54*** join/#asterisk Shoragan (n=shoragan@134.169.175.72)
13:02.01SheriF_WorKahh yes :-)
13:02.02SheriF_WorKhehe
13:02.03SheriF_WorKsorry
13:02.26SheriF_WorK[TK]D-Fender: so with this way all the trafic and the call trafic will pass throw * ?
13:03.34*** join/#asterisk zotz (n=zotz@24.244.133.115)
13:03.43frk2Grandstream issues again!!! this time my screen blanks and i get no voice... dialout to an extension works though :(
13:04.03frk2Even my Cisco hangs!!!! Cisco 7960G using SIp 8.2
13:04.07kdz13is the voip-info wiki the best source of AGI documentation
13:04.12[TK]D-FenderSheriF_WorK : Yup, everything will flow through *.
13:04.14kdz13or is there another more complete source?
13:04.35[TK]D-Fenderkdz13 : Wiki + The Book
13:04.38[TK]D-Fender~book
13:04.39jboti guess book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
13:05.38kdz13ah sweet
13:05.43kdz13i was not aware of this book.
13:05.47*** join/#asterisk feld_ (n=feld@12.148.212.157)
13:06.06kdz13[TK]D-Fender: thanks
13:06.13[TK]D-Fenderkdz13 : np, good luck
13:06.49*** join/#asterisk epablo (n=epablo@WLL-24-pppoe194.t-net.net.ve)
13:07.04feld_is there a common reason for registration error timeouts with X-Lite?
13:07.12SheriF_WorK[TK]D-Fender: thx ;-)
13:07.30epabloHi people
13:07.43feld_hi epablo
13:07.58[TK]D-Fenderfeld : Usually bad user/pass
13:08.17*** join/#asterisk zeppelin_ (n=zeppelin@201-40-157-135.paemt700.dsl.brasiltelecom.net.br)
13:08.25Kattymorning
13:08.32epabloAnyone with some experience with queues?
13:08.54*** join/#asterisk Arno[Slack] (n=root@66-163-12-60.ip.tor.radiant.net)
13:09.07coppicei'm british. I've had a lifetime of studying queues
13:09.45kdz13coppice: lol
13:09.51zeppelin_:D
13:09.52sevardI've found conflicting articles.  I'm betting that voip-supply is out of date... Does asterisk have T.38 support ?
13:09.53epabloI need to run a command in order tu signal my CRM to open a popup on the agents screen before I pass him a call
13:10.45epablosevard: it supports it, but I have read that it doesn't garantie it
13:10.50feld_aha! nevermind people. somehow my nat=true disappeared from sip.conf
13:10.52feld_silly me
13:10.53feld_:)
13:11.03sevardepablo: is that why the bounty is still 10,750?
13:11.21coppicesevard: some t.38 support is going into 1.4. Right now * doesn't support T.38
13:12.01epabloI have used * to send faxes and most of the times it has worked..
13:12.10epablomaybye it was pure luck  ;)
13:12.15[TK]D-Fendercoppice : Isn't that just T.38 PASS-THROUGH support?
13:12.36feld_anyone have any tips on how to make it so that when an extension is dialed that doesnt exist the caller is notified of it?
13:12.39coppiceyeah, but I have the rest pretty much complete :-)
13:12.53coppiceit won't go into SVN, though
13:13.07[TK]D-Fendercoppice : Community owes you a lot.... and I think that 10G would go a fair ways ;)
13:13.15[TK]D-Fendercoppice : Disclaimer issues?
13:13.28coppiceyes. the T.38 code in in spandsp
13:13.46sevardI have an ATA hooked up to * and I'm sending calls over a PRI, it looks like faxs are going through and come back in but I'm getting "Unknown RTP codec 100 received" which apparently means I'm using T.30 and faxes are not supposed to be working at all
13:13.57coppiceyes, i think 10 giga dollars will do nicely
13:14.22[TK]D-Fendercoppice : Sorry.. typo... "g" ;)
13:14.39kdz13~book
13:14.40jbotbook is probably a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
13:15.05mutuh
13:15.14mutanyone know where to get a cheap prism 2 pcmcia card?
13:15.16coppiceUnknown RTP codec can mean many things. you need to look at the SDP to see what 100 was allocated as
13:15.22mutsearching on ebay is needle in a hat stack style
13:15.59sevardcoppice: I'm not sure what SDP is.
13:16.31coppicesession dementing protocol. it the stuff in the SIP where the codec config was set up
13:16.45*** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com)
13:17.04sevardSo you want me to do a sip debug and look for.. what?
13:17.19sevardor zap debug
13:17.31[TK]D-Fendersevard : SIP debug on the neg
13:17.31feld_im having nat issues apparently. i can register and dial but i cant seem to receive any calls.... :( any tips?
13:17.43[TK]D-Fenderfeld : pastebin your SIP peer entry
13:17.44[TK]D-Fender~pb
13:17.46jbotsomebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
13:18.10feld_[TK]D-Fender, only x-lite phones at the moment
13:18.10*** join/#asterisk dpryo (n=hn@raphael.ondskap.net)
13:18.11sevard[TK]D-Fender: say what brutha
13:18.29sevardfeld_: he's asking for your sip.conf
13:18.48feld_sevard, heh i know :) im pasting... :P
13:18.52sevardcoppice: so... want to explain that a bit more for me?:)
13:19.31feld_http://sh.nu/p/1923
13:19.48coppicedo SIP debug. look at the call setup, and you should see numbers assigned to various codecs. look for what is assigned as 100
13:20.19[TK]D-Fenderfeld : Add "canreinvite=no", and "qualify=yes" to your peer entries.  Is your * behind NAT as well by any chance?
13:20.37sevardAdding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP
13:20.59feld_[TK]D-Fender, no my * is on a real IP
13:21.54[TK]D-Fenderfeld : Then add what I just suggested, do a "sip reload", and restart your X-Lite's and all should be good.
13:22.00sevardcoppice: is that what I'm looking for?>
13:22.22eipihow i can configure my iaxy to register a named server?
13:23.20epabloI need to  signal my CRM to open a popup on the agents screen before I pass him a call.  Is this posible with * queues?
13:24.33*** join/#asterisk feld_ (n=feld@12.148.212.157)
13:24.43feld_crap didnt mean to close that... hehe
13:25.41viperdudehi gus
13:25.45feld_[TK]D-Fender, hrm still no good. i'm dialing from the * console for testing and it can't seem to reach my phone yet... let me paste my little extensions.conf. maybe something is wrong with it. I'm fairly new to *....
13:25.55gandhijeeanyone know if there is a crosstools channel on this server?
13:25.56viperdudeis there a way to set the privacy flag on outgoing SIP calls?
13:26.22feld_http://sh.nu/p/1924
13:26.39techman97_andyhey all, when was the original Asterisk project started?  (year?)
13:26.48feld_techman97_andy, i think 98?
13:26.51techman97_andycool
13:26.53techman97_andythx
13:27.04viperdudethe copyright on the CLI starts at 1999
13:27.07feld_at least i think thats what the oreilley pdf said
13:27.27*** join/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net)
13:28.54*** join/#asterisk miztic (n=gerard@rarcoa.com)
13:29.07[TK]D-Fenderfeld_ : Your dialplan is a nice idea, and would work except that your x-lites have NAMES for their [] entry, and not numbers.
13:29.14astar`hello i want to do shortcuts on numbers : ex : i type 25 on my phone and its dials 0243434343
13:29.45[TK]D-Fenderfeld : You'd need to be able to do like "Dial(SIP/markxlite)", which "x" does NOT catch.....
13:29.49feld_[TK]D-Fender, i borrowed chunks of the dialplan from somewhere else. Thanks for solving it though :)
13:29.54viperdudeaster`: exten =>25,1,Dial(SIP/0243434343@myvoipproider,20,tr)
13:30.23feld_I should have known that. I guess I just didnt piece it together and realize the variable was failing. :)
13:30.35astar`actually i have something like that but its doesn't respect the dial because its dials directly ONe trunk
13:30.46[TK]D-Fenderfeld_ : np, mod it up and let me know.
13:30.55[TK]D-Fenderfeld : 2 second finx if you put your mind to it ;)
13:30.56astar`dial *rules
13:31.21viperdudeaster`: not sure what you mean
13:31.32sevardfeld_: how are you dialing ?
13:31.48s-ndh-cwhat am i doing wrong?
13:31.50*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.142.Dial1.SanJose1.Level3.net)
13:31.54[TK]D-Fendersevard : I've ID'd the problem... its a minor fix I'm sure he's already working on.
13:32.04sevardI'm just curious how he's dialing
13:32.17s-ndh-ci i call myself using xlite i get a missed call where the address is asterisk@myproxyip
13:32.20feld_[TK]D-Fender, hehe yeah thanks. I appreciate your help. I'm definitely sticking around in this channel as I'll be using this for a long time.... at work, selling, implementing, and supporting it. So I'll be around to help others, too :P
13:32.26*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.142.Dial1.SanJose1.Level3.net)
13:32.57[TK]D-Fenderfeld : If that escaped you and is a sample of the kind of work you're doing, you MIGHT want to study a bit more ;)
13:33.05s-ndh-cwhy does it send asterisk@myproxyip as the adress?
13:33.18s-ndh-cnot my username or my extension@myproxyip?
13:33.27[TK]D-Fenders-ndh-c : My guess is you didn't set CallerID for the calling channel.
13:33.31*** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu)
13:33.38s-ndh-c[TK]D-Fender:  where do i do that?
13:34.03feld_sevard, i was dialing from the console. currently my context is local so it works if i just do "dial #" instead of doing "dial #@context"
13:34.11epabloI once used a command called chan_grab, but I don't remember where I downloaded it.  Does anyone know I can find it?
13:34.28sevardfeld_: I don't have the "dial" command
13:34.36feld_sevard, strange
13:34.48sevard[TK]D-Fender?
13:35.24MatsKthe dial command is only present if the sound card support is loaded in asterisk
13:35.34sevardI see.
13:35.40[TK]D-Fenders-ndh-c : in your sip.conf peer entries for all related phones.
13:35.53feld_oh yeah it uses OSS or ALSA (ugh...).
13:36.02sevardwhowhat I love alsa :)
13:36.53feld_sevard, i know some game programmers... epic, ID, etc... and they've told me that the ALSA devs themselves have admitted that their API is horribly broken and that for some devices the frequencies generated by ALSA are way off.
13:37.10*** join/#asterisk mercestes (n=merceste@69.15.174.114)
13:37.23*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
13:37.49feld_ALSA is open source = good. Broken and stubbord devs = bad. I like 4Front's OSS, but it's not open source. It works well, though. Pretty powerful actually. :)
13:38.03*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
13:38.03*** mode/#asterisk [+o anthm] by ChanServ
13:38.20feld_"to each his own" that's the linux way and that's what matters ;)
13:38.21s-ndh-cok
13:39.04*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:40.00feld_[TK]D-Fender, any suggestions how to throw in logic so that if an unknown extension is dialed it will play a diff audio msg instead of my simple straight-forward dialing plan I have now?
13:40.44epabloLast year the was like an asterisk branch.  Does anyone remember the name of that project?
13:41.14MikeJ[Laptop]yep
13:41.23feld_epablo, freeswitch
13:41.34feld_it's pretty alpha though they're claiming otherwise. =S
13:41.52MikeJ[Laptop]freeswitch isn't an asterisk branch
13:41.56epablo<PROTECTED>
13:42.07feld_MikeJ[Laptop], it's not? i swear it is?
13:42.11MikeJ[Laptop]nope
13:42.15MikeJ[Laptop]totally new code
13:42.23epabloIt was a rebelion.. LOL
13:42.32MikeJ[Laptop]that was openpbx
13:42.40MikeJ[Laptop]it;s dead as far as I know.
13:42.49*** join/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au)
13:42.53feld_oh yes you're correct MikeJ[Laptop]
13:42.56[TK]D-Fenderfeld : don't use a generic mask like that.  Hard code th valid ones.
13:42.58feld_my bad :P
13:43.17feld_[TK]D-Fender, ok and then what do I throw in there to pick up the rest of the extensions?
13:43.36MikeJ[Laptop]epablo, what app you looking for/.
13:43.38[TK]D-Fenderfeld : you don't... they'll just 404 like they should
13:43.47feld_[TK]D-Fender, beautiful. thanks.
13:43.52epabloMikeJ[Laptop]:  You are right.. Thanks
13:44.03MikeJ[Laptop]epablo, what app are you looking for?
13:44.45mostyi need to edit /etc/modprobe.d/zaptel to make sure opermode is set to AUSTRALIA, but there is a line at the top of this file that says it's automatically generated and not to edit it. so what generates this file?
13:45.50MikeJ[Laptop]sounds like you are using one of the gui's?
13:47.55feld_My boss and another cow-orker who is not usually available threw this * box together. It does have a Digium TDM400P in it with 4ports but I dont know which are FXO or FXS. Any way via software to find that out?
13:47.56*** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar)
13:47.58epabloMikeJ[Laptop]:  chan_grab
13:48.37wunderkinepablo, it is app_changrab
13:49.00MikeJ[Laptop]epablo, www.pbxfreeware.org is it's home.
13:49.48epabloMikeJ[Laptop]:   Thanks
13:49.50MikeJ[Laptop]it probably needs some updating for 1.4 if your not using it on stable...
13:49.51astar`someone know if there is something to dial a phone  number via egroupware ?
13:50.04MikeJ[Laptop]if so, let me know, I can get the updates posted up there
13:50.49feld_omg
13:50.55MikeJ[Laptop]?
13:50.56feld_music on hold works by default?
13:51.04feld_lol nice somehow it is.....
13:51.05*** join/#asterisk dools (n=iain@125.62.65.184)
13:51.20feld_i called myself, picked up the second line. it essentially put myself on hold and I hear the music.
13:51.30epabloOk.. thanks
13:51.32*** join/#asterisk Creperum (n=ilya@tex.tsua.net)
13:51.35MikeJ[Laptop]did they finally change default moh in the sample configs to use native??
13:51.42doolshi, is there anything in agi like "register termination handler" or something like that so that i can execute some code after everything else has happend?
13:52.24Creperumhey, how can i make FOP to display 100 extentions and 60 trunks????
13:52.24mostyfeld: you could open the case and look, of just try all the combinations for zaptel.conf (there's not that many)
13:52.40Creperumit only displays part of them!
13:53.33feld_mosty, i'd rather open it up. I'll do that. not like this thing can go into production soon anyway. idiots moved this thing live without DNS, i have not been given our T1's DNS info after I've been asking for the last 3 days, and * isnt updated to fix the vulnerability =S
13:54.27mostyfeld: well the two types of modules will most likely be different colors, then you have only two possible options for zaptel.conf
13:54.39stephane_re
13:55.21mostyrun ztcfg and it will spit out errors if you get it wrong
13:57.00doolsokay, perhaps some more background information ... i'm hacking a reseller module into a2billing and i've set up a 'resource allocation' model which means that when a user makes a call, it calculates the total number of minutes possible and 'allocates' those resources from the reseller's credit so that a reseller is not able to have his user's cost us money. i can quite happily allocate the resources at the beginning of the exeuction of a2biling.php, but the s
13:57.35doolsotherwise i don't know where to de-allocate the resources
13:58.09wunderkinfeld_, you can use any dns server
13:58.27*** join/#asterisk C4T3l (n=rcall01@216.54.143.2)
13:59.24feld_wunderkin, not usually. most will block by who they are servicing in the case of ISPs.
13:59.35techman97_andyanyone worked with a specific Cisco SIP phone?  Any recommendations on a standard desk set?
13:59.42feld_occasionally you can find some open ones but I never have any luck doing that :( got any suggestions?
14:00.23doolstechman97_andy: i use cisco ATA, works well, config interface is ugly as hell (ie. i wouldn't want to support it unless i shipped it pre-configured)
14:00.38C4T3ltechman97_andy: we use the Cisco 7960 series... thy're ok phones
14:01.00doolstechman97_andy: cisco ATA allows you to plug in a standard phone + a LAN cable
14:02.23Vorondilhi all; quick question: shouldn't this (http://pastebin.com/762326) play music on hold songs in random order?  it always seems to play them alphabetically. =/
14:03.02techman97_andycool all - thanks!
14:03.04doolsVorondil: maybe you're just really really unlucky
14:03.06mostyvorondil: prefix the filenames with a random X digit number?
14:03.28*** join/#asterisk Ecio (n=eciostar@194.105.59.42)
14:03.34Eciohi
14:03.34*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
14:04.36Eciodoes anybody know if it's possible to use the new IM and Presence features of X-lite 3.0 with asterisk ?
14:05.23vader--hola
14:06.11Vorondillol, mosty: i thought about that, but the point is that no caller gets the same song twice during, say, a week
14:06.24*** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81)
14:06.29Vorondilsince most folks never stay on hold for an entire song anyway
14:06.45C4T3lEcio, I imagine if you use hints in the dial plan it sould work...??
14:07.07doolsVorondil: seriously... who cares?
14:07.19doolsVorondil: it's hold music
14:07.40EcioC4: yes im tryin, i've just read  that publish is not supported so no extended features (like away etc..)
14:08.00Eciobtw Instant Messages doesnt seem to work...
14:08.21Ecioi see in debug:  WARNING[16727]: chan_sip.c:7281 receive_message to X from Y, dropped it...
14:08.30mostyvorondil: just put loads of music in there and don't restart asterisk very often?
14:08.41doolshaha shittest solutions ever
14:09.02Vorondildools: eh, i mean, it's not a big deal, i just expected random=yes to do what i wanted. it didn't. so i asked.
14:09.23doolsVorondil: just submerge your server in jelly and play checkers for 9 hours
14:09.31doolsVorondil: i know, i'm being facetious
14:09.38Vorondilmosty: it seems to stop playing stuff (and rightfully so) when nobody's on hold, so it just starts over from the top again
14:09.52Vorondillol
14:10.02C4T3lEcio: asterisk will support text sent to a phone from the dialplan, but to my knowledge it will not support UA2UA Instant msging
14:10.05Vorondilthe jelly and checkers does sound appealing..
14:10.09doolsVorondil: maybe random=yes means that it randomly plays the hold music for a particular session
14:10.31doolsVorondil: is there are 'persistent hold music' option or something that would keep the same track going for all callers?
14:10.34Ecioc4 i see :(
14:10.36*** join/#asterisk Hmmhesays (i=negative@66.173.103.110)
14:11.05doolsVorondil: what i know about asterisk, by the way, you could fit on the head of a pin plus 9 angels, so don't listen to me
14:11.10*** join/#asterisk viler (i=1000@200.114.70.228)
14:11.10C4T3lEcio: maybe someday it will :D
14:11.15doolsVorondil: i came here to ask a question about a2billing :-)
14:12.01Vorondildools: hehe, s'all good.  i hadn't touched asterisk until about a month ago
14:12.42Vorondildools: but yeah, i thought the same thing about random=yes, so i listened all the way through a song, and it just went to the next alphabetic one
14:12.52Vorondil(assuming "alphabetic" is the right word there...)
14:13.35*** join/#asterisk anonymouz666 (i=anonymou@200.218.196.5)
14:13.44*** join/#asterisk lorinc (n=ang@caracas-2783.adsl.interware.hu)
14:13.51doolsVorondil: i must say, though, that i've spent hours and hours on hold, and the only thing i ever hear is the same cd repeated ad nauseum (in some cases literally) or a local radio station
14:14.05*** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse)
14:14.22*** join/#asterisk b00mer (i=fwuser@blackhole.c5i.com)
14:15.04Vorondildools: indeed, i'd like to keep from that if i can help it
14:15.20doolsVorondil: you're a nobler person than i :-)
14:15.24vader--hehe
14:16.00Vorondilwait, so is "=" interchangeable with "=>" in the conf files? (namely musiconhold)
14:16.17C4T3lEcio: i have an idea... Maybe you could try a canreinvite =yes in sip.conf, maybe you could set up 2 softphones and try it that way?  *shrugs*
14:16.32Ecioc4t: that's what im doin :)
14:16.48*** join/#asterisk aze_ (n=aze@ACayenne-101-1-8-198.w81-248.abo.wanadoo.fr)
14:16.49frk2dudes- i have a question for a fellow grandstream GXP2000 user/sufferrer - anybody?
14:16.51Vorondilvoip-info has musiconhold with =>'s
14:16.51Vorondilhttp://tinyurl.com/bn57o
14:17.25frk2guys- why do hard phones Hang??? In my problematic client's network- even the Cisco 7960G hangs
14:17.38drraypower?
14:17.40frk2GXP 2000 works WAY better if I do POE on it... so points towards power issues.
14:17.58drraymy ciscos don't hang
14:18.04drraywhich sip image?
14:18.06frk2drray- thats what i think too
14:18.07frk28.2
14:18.10frk2what u using?
14:18.17drrayIm stuck in 6
14:18.19drrayer, 7
14:18.25*** join/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net)
14:18.31drrayI did not renew my smartnet contract
14:18.34drray:)
14:18.36frk2maybe 7 is the way to go :)
14:19.05frk2The crap with Cisco is that 802.3af is unknown to them, till recently
14:19.07drraydo you have a power cube?
14:19.27drrayI'd try a phone with a power cube, then blame the injector
14:19.59frk2yup.. i have a power cube- thats what causes the issues
14:20.13drrayoh
14:20.17frk2See my grandstream GXP would hang DAILY/Hourly with local power
14:20.53frk2so I got a 802.3af injector and hooked it up.. now only the screen blanks after 5 days
14:20.53drrayhmmm
14:20.53drraydirty power?
14:20.53frk2so its a huge improvement
14:20.57frk2thats why im guessing
14:21.07drrayrun a phone on a UPS?
14:21.13frk2However, ATcOM or Dlink shit phones work completely okay
14:21.45frk2however
14:21.58drraymaybe a drop to 7.3 (or whatever the highest 7.x is) is the way to go for one or two
14:22.01frk2more interesting is that this 'hanging' paradigm only happens in a particular area in the office
14:22.06Hmmhesaysblahbitty blahblah
14:22.13Hmmhesaysbuildroot for uclibc is pissing me off this morning
14:22.26drraythat's external
14:22.28frk2The cisco and the grandstream are in cubicles next the each other
14:22.46drray50 foot extension cord
14:22.46drray:)
14:24.35*** part/#asterisk dools (n=iain@125.62.65.184)
14:25.38[TK]D-FenderPolycom > All
14:25.48jarrodhey
14:26.00jarrodis there a way to make it so sip reload doesnt send a signal to all my polycoms
14:26.04mutwell
14:26.13drrayI don't (or did not at this time last year) like hoe Polycom's look
14:26.18mutpolycom >= all
14:27.01mutanyon recommend a good prism 2 pcmcia card?
14:27.44tzafrirdonno. I'm using a zd1211-based usb stick
14:28.56muti have one of those, gf is using it in her pc tho
14:29.17Hmmhesaysi was reminded last night why I was a bachelor for so long (speaking of girlfriends)
14:29.28mostyi heard that intel cards are currently the best supported right now
14:29.47mutmosty: looking for something i can use for sniffing
14:30.17*** join/#asterisk swytch (n=ezcall@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr)
14:30.35feld_hey guys my voicemail is failing login because it keeps claiming the context=default. where in the voicemail extension line do I define the context from which the users calling it will be belonging to?
14:31.24swytchquestion about the manager api.  specific events seem to miss from the output, like Event: Dial.  is there a way to enable such events?
14:31.48*** part/#asterisk epablo (n=epablo@WLL-24-pppoe194.t-net.net.ve)
14:31.49jarrodfelds_: VoicemailMain(@context) ?
14:32.26jarrodswytch: i believe you can enable what output you want to see, or a verbosity of the default
14:33.38mostyhow can i prevent ztdummy from being loaded?
14:33.43feld_jarrod, i actually did (_X.@context)
14:33.45swytchjarrod: i tried "read = Dial" in manager.conf, but nothing.  do i have to let asterisk dump a lot just to get the few events i want?
14:33.49feld_just figured it out :)
14:34.06*** join/#asterisk eKo1 (n=bernd@190.4.7.90)
14:34.36*** join/#asterisk Joshaidan (n=icechat5@thunderbay-voip-4.vianet.ca)
14:34.57eKo1I'm getting this NOTICE message on the Asterisk CLI since I connected my second PRI line this morning: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2
14:35.10*** join/#asterisk }btorch{ (n=kvirc@adelphi.geofocus.com)
14:35.22drraydo you have a d channel on span 2?
14:35.29eKo1Yep.
14:35.46eKo1Or atleast, there should be.
14:36.01drrayis it coming out of a channel bank?
14:36.37drrayI had a similar issue, where everything about the PRI worked
14:36.43drrayjust kept getting a bad d channel
14:37.11eKo1The PRI is comming out of a signaling gateway
14:37.24eKo1that does SS7<->PRI conversion.
14:38.25eKo1I've been googling around and, from the info. I've found, it could be an issue with the card.
14:39.58jarrodhave another PRI card tot est with?
14:41.43coppiceeKo1: you probably have a clock sync problem in your config.
14:42.05eKo1OK. I think I know why it is all messed up. I botched my /etc/zaptel.conf
14:42.54eKo1If the span is going to by the clock source, I should have span=1,0,0,ccs,hdb3,crc4 right?
14:43.09coppicenope
14:43.25eKo1should it be span,1,1 then?
14:43.38mostyi have a wctdm card that isn't by digium, do i still need ztdummy?
14:43.41coppiceif you want to take clock from the span it needs to be something like span=1,1,0,ccs,hdb3,crc4
14:44.19eKo1coppice: and if the clock is going to be obtained from the other end, it should be span=1,2 right?
14:44.48coppicenope. the 1 means the first priority source
14:45.35eKo1What does that mean exactly?
14:45.49*** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com)
14:46.13vader--can anyone explain to me what this dialplan will do
14:46.14vader--exten => _NXXXXXX,1,Dial(ZAP/g1/${EXTEN:1})
14:46.14vader--exten => _NXXXXXX,2,Congestion
14:46.14vader--exten => _NXXXXXX,102,Congestion
14:46.15tzangereKo1: it means that if that span is up, the card will sync to that span's clock.
14:46.21*** join/#asterisk Ahrimanes (n=michael@62.61.133.90.generic-hostname.arrownet.dk)
14:46.25fholmesDoes anyone here use Queues?  I am just curious to find out if they have any problems with them I should know about before I spend time trying to implement them...
14:46.26coppiceif you have a line coming in from an SS7 source you certainly want to slave to its timing. use span=1,1,0,ccs,hdb3,crc4
14:46.50tzangerthink of it as priorities.  #1 = my preferred source.  #2 = my next preferred source if #1 is gone, etc.  #0 = fuck you, I will never sync to your clock.
14:47.10drrayyay #0
14:47.10drrayer
14:47.21eKo1OK. If I want the span to supply the clock, then I should use 0?
14:47.44coppiceyep
14:47.49coppicebut you don't
14:47.55brettnemvader--: if it gets a exten that matches NXXXXXX it will attempt a call out g1 on ZAP.. After the call it will end in congestion or if the dial failed it will indicate congestion
14:47.55tzangerdon't think of it as supply and demand.  think of it as "am I going to try and sync to the far end of this span or not?"  1 = yes I will try to sync ot it, 0 = nope, I will do what I want
14:48.44eKo1Because I plan to hook up a dialin server to one of the ports on my quad E1 card and I need to have it (the card) act as the clock source.
14:48.54coppicethink "am I the centre of the universe" then calm your ego down and try that again. if the answer is "no", then don't use 0
14:49.45coppiceeKo1: then use 0 for that port, and make the dialin server slave to you
14:49.58eKo1OK. Thanks.
14:51.26*** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com)
14:52.53frk2Agree with coppice. If you put 0 there you would get a lot of HDLC frame errors
14:53.05frk2and dudes would complain of random call drops
14:53.16eKo1OK. I made the changes and I rebooted the server.
14:53.32frk2dont reboot man.. its Linux not windows :)
14:54.10eKo1I know but I do it just in case.
14:54.32frk2guys... i BADLY neeed random suggestions on my phone lockup issues. ANYTHING. These grandstream/Cisco lockup issues happen only at a particular client
14:54.38eKo1The notice is gone. Great.
14:54.57eKo1Define lockup.
14:55.15frk2Lockup = 1) phone ceases to respond... need to power cycle
14:55.37frk2OR in the grandstreams case sometimes the screen goes blank, no voice but if I dial the other phone rings (no voice again)
14:55.45[TK]D-Fenderfrk2 : I've heard the 8.0 family of Cisco SIP firmware is flakey and people have reverted back.... as for GS, I hear everything is flakey and people just send THEM back :)
14:56.07frk2TKD- I cannot seem to find the 7.x SIP firmware
14:56.13frk2for cisco's
14:56.25drrayI'd never deploy grandstreams but my budgetone has been "serviceable"
14:57.00frk2whats a good mid range phone then?
14:57.19frk2Polycom/Aastra?
14:57.22drrayI like the cisco 79x0's but that is not an asnwer
14:57.23drrayfor you
14:57.36eipihow i can configure my iaxy to register a named server?
14:57.38frk2can somebody tell me where to get the older firmware please?
14:57.38eKo1frk2: I have that happen to me with Grandstream phones and Sipura ATAs.
14:57.41eKo1No biggy.
14:57.56frk2eK01 - the locking up part?
14:57.58drraycisco.com has teh older firmware if you have a smartnet contract
14:58.10frk2I dont have a smartnet contract :)
14:58.10eKo1Yes, the don't work and have to be power-cycled.
14:58.30eKo1s/the don't work/they stop working/
14:58.53eKo1:)
14:59.00frk2Damn... hanging is total loss.
14:59.12frk2My problem with Aastra is they dont make 220v power adaptors!!!!
14:59.22frk2I guess I need to 802.3af' them
14:59.40eKo1uh oh, the HDLC notice is back again.
14:59.42eKo1rats
15:00.57frk2whats the exact notifce?
15:01.12eKo1pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2
15:01.20[TK]D-Fenderfrk2 : Polycom is quality stuff at a great price point.
15:01.40frk2yeah man... will the 300/301 work right out of the box with SIP support?
15:02.14[TK]D-Fenderfrk2 : yup
15:02.23[TK]D-Fenderfrk, though I'd recommend the IP430 in its place.
15:03.04dlynes_homeeKo1: yeah...i've been getting the same notice now for some time...it shows up every once in a while
15:03.25dlynes_homeeKo1: it seems to be related to missing an interrupt here and there
15:03.31frk2there is no IP 430 TKD
15:03.35frk2301/501
15:03.40frk2shit-- 301 is now $129 at telephonyware!!!
15:03.43frk2im getting it
15:03.44dlynes_homefrk2: yeah, there is
15:03.45[TK]D-Fenderfrk2 : http://www.polycom.com/products_services/0,1443,pw-34-182-15672,00.html
15:03.46eKo1but zttool reports no irq misses
15:03.49dlynes_homefrk2: check polycom's website
15:03.57[TK]D-Fenderfrk2 : IP 301 = $115 at Atacomm....
15:04.15dlynes_homeeKo1: Does your patlooptest show a pass on all of your spans?
15:04.47eKo1patlooptest? Is that when I choose loop in zttool on  the span?
15:04.49*** join/#asterisk ToyMan (n=stuq@74-32-67-126.dsl1.mdl.ny.frontiernet.net)
15:05.10dlynes_homeeKo1: no, that's when you type ./patlooptest /dev/zap/1
15:05.43dlynes_homeeKo1: make sure you use the clear option in zaptel.conf for whatever span you're testing
15:05.54frk2why the hell dont they make 220v power supppliessssssssssssssssssss
15:05.57[TK]D-Fenderfrk2 : But do keep in mind you need to get the 301 with its special PoE cable which is a little bulky.  IP 430 has a lot of perks that make up the difference.
15:06.14[TK]D-Fenderfrk2 : Go PoE with IP430's......
15:06.30eKo1dlynes_home: I don't have that tool.
15:06.41dlynes_homeeKo1: yes you do...you just haven't compiled it
15:07.03dlynes_homeeKo1: go into zaptel source directory and type make patlooptest
15:07.27eKo1OK.
15:07.38}btorch{do I need to recompile asterisk to have it run as a regular user ? or can I just change permissions and runas the user that I want it to run as
15:07.52frk2hmm.. see this is the kind of stuff i dont understand.. GXP 2000 is the best seller at atacomm!! :)
15:07.53dlynes_homeeKo1: then put a t1 loop connector into the span you want to test
15:08.29eKo1t1 loop connector?
15:08.34dlynes_homeeKo1: yeah
15:09.16[TK]D-Fenderfrk2 : The sell a lot of CRAP... doesn't make it GOOD :)
15:09.20*** join/#asterisk _4d4m_ (n=adam@62.69.102.99)
15:09.25Hmmhesaysblargh
15:09.34Hmmhesayswhere did buildroot hide the kernel source
15:09.48*** join/#asterisk ivanfm (n=ivanfm@c9068840.virtua.com.br)
15:09.55[TK]D-Fenderfrk2 : and what kind of marketing genius would label it like "Yeah it sucks, but look at the price!"
15:10.00feld_wheee! i have asterisk working now with voicemail. so neat. :) now to get outside dialing working after updating asterisk.... that will be my afternoon goal.
15:10.15[TK]D-Fenderfeld : AFTERNOON?  eek
15:10.20eKo1I don't have one of those.
15:10.26dlynes_homeeKo1: http://66.102.7.104/search?q=cache:9zhgfJEhcR0J:www.effeng.com/vtc/jseries/T1_Hardloop_Pinout.pdf+what+is+a+t1+loop+connector%3F&hl=en&gl=ca&ct=clnk&cd=1
15:10.41dlynes_homeeKo1: make sure you use the A standard (1, 2, 4, and 5)
15:10.44frk2Dude this is my problem:
15:10.49frk2The client doesnt have POE
15:11.09eKo1dlynes_home: I have one of those already.
15:11.13frk2If they have to install a POE switch just for this- its too much..
15:11.20dlynes_homeeKo1: i thought you said you didn't have one?
15:11.22frk2what input voltage does the Polycom 301 have?
15:11.28eKo1I didn't know you called it that.
15:11.35dlynes_homelol
15:11.37eKo1I just call it a self looped e1 cable.
15:11.37[TK]D-Fenderfrk2 : Hrm..... ok, I know Polycom does have international power standard bricks... just not sure on the part #
15:12.03dlynes_homeeKo1: yeah...e1 might be the b standard...I don't know
15:12.47dlynes_homeanyways...i've gotta run
15:13.14dlynes_homeeKo1: if you have any further problems getting the patlooptest set up, you might be able to find someone from digium that can help you with it
15:13.33*** join/#asterisk wunderkin (i=kev@69.26.192.234)
15:13.37frk2do you guys know of a good online / offline VOIP retail store location in Asia/Middle EAsy?
15:13.40frk2East
15:13.43dlynes_homeeKo1: or you can wait until i'm back in the office
15:13.52dlynes_homeI'll be in the office in about 3 hours or so
15:14.18*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
15:14.38*** join/#asterisk nahirean (n=nahirean@unaffiliated/nahirean)
15:14.49s-ndh-chehe
15:15.26*** join/#asterisk rvhi (n=rv@66.175.65.89)
15:16.23eKo1dlynes_home: thanks
15:17.32feld_[TK]D-Fender, shouldnt be that hard. I have the hardware and connections. I have some account information sitting in my mailbox. Just have to plug it all in I guess :)
15:18.53eipianyone: how i can configure my iaxy to register a named server?
15:19.06[TK]D-Fenderfeld : Meant it should take MINUTES.
15:19.24russellbeipi: you can't
15:19.48[TK]D-Fenderrussellb : Still no DNS on that thing huh?
15:20.05russellbnegative
15:20.19russellbwho needs dns, pfft ...
15:20.28[TK]D-Fenderrussellb : If you want it to sell it needs a fair amount more work on the interface & functionality side...
15:21.00[TK]D-Fenderrussellb : Its a hard sell against ATA's like Linksys & co
15:21.09feld_[TK]D-Fender, yeah but I have more than just this going on. I support a ton of businesses and their networks too.
15:21.17russellbgtkiaxyprov, man
15:21.20russellb:D
15:21.49[TK]D-Fenderfeld : Ok, *15 minutes*
15:21.50eipi:D
15:22.08[TK]D-Fenderrussellb : And still no DNS :)
15:22.13frk2does anybody know the input voltage for Polycom 301/501?
15:22.18feld_[TK]D-Fender, i still have to go power the damn thing down open it up and find out which are FXO and FXS lol
15:22.19frk2is it 12v or 48V
15:23.22*** join/#asterisk mr_horsepower (n=igor@82.102.1.42)
15:23.27mr_horsepowerhi
15:23.46[TK]D-Fenderfrk2 : not sure.... I can tell you in about 10 hours.....
15:24.20*** join/#asterisk vechers-away (n=svecher@64.61.117.139)
15:26.18feld_anyone here use * with video? I just got a requst to find out what it takes....
15:28.36key2feld: I do
15:28.39key2h263
15:29.06feld_what do your clients use for phones?
15:29.22mutdamn i shoulda bought a house a long time ago
15:29.31mutcombine car/house insurance
15:29.40key2eyebeam
15:29.46mutbrought my car ins payment down $80
15:29.57mutand home ins pmt is only $40s
15:30.03feld_nice
15:30.33mutya, that freakin awesome, now i can afford to put that water softener in it
15:30.36mut:P
15:30.39feld_my car insurance is through the roof. yay for getting screwed over.
15:31.01muti pay $223/mo right now
15:31.08mutwill be $152
15:31.45mutyear ago i was paying $293/mo
15:32.06frk2TKd- thanks man- you are always a great help
15:32.54[TK]D-Fenderfrk2 : That sarcasm for the time its going to take me to get home?
15:32.58feld_i honestly dont know what mine is atm. i think its near a grand a month =(  i just graduated college, ins being payed by my parents until it comes down in january. so gay. accident which wasnt my fault and getting a ticket for sometihng i never did within like 2 weeks really screwed me over. that was 2 1/2 years ago. i hate this country :(
15:33.09frk2TKD- no no dude... im serious.. you always help me out :)
15:33.19Nuggetyou get a discount for being gay? cool.
15:33.28[TK]D-Fenderfrk2 : Ok, can't be sure sometimes.. esp after a 10hour warning like that ;) np
15:33.34frk2haha
15:33.39mutNugget: well yea, it's being encouraged in society anymore...
15:33.56frk2if you could tell me where to get the Cisco 7.x firmware from somewhere.. without the freaking contract, it would be aewsome
15:34.01techman97_andygay discounts?  wtf?  that's what I get for minimizing this channel
15:34.03frk2are resellers supposed to have this contract?
15:34.13feld_rofl that reminds me the guy on the radio did the Sports section like he was gay. it was hilarious.
15:34.25*** join/#asterisk Symm (n=s@198.87.2.15)
15:34.33Symmwerd
15:34.48fourcheezehow does someone do a sports section like they're gay?
15:34.52Symmasterisk is being sold here in washington DC for beaucoup bucks
15:35.01fourcheezedo you mean "Camp" ?
15:35.10kdz13fourcheeze: "that guy is so hot in his shorts!!!!"
15:35.11Symmits called, playing on the population's ignorance
15:35.15Symmhey no fag talk
15:35.32fourcheezehmm
15:35.45Symmthis is for visionaries only
15:35.58*** join/#asterisk burizaa (n=freeee@cm66.omega101.maxonline.com.sg)
15:35.59Symmprophetic talk only please
15:36.03fourcheezeso a straight guy doing the sports section on a womens sport would be commenting on their bodies too?
15:36.26[TK]D-Fenderfrk2 : Lemme look
15:36.39kdz13fourcheeze: I dunno, i never pay attention to sports
15:36.43kdz13so i'm just guessing
15:37.01kdz13but that's the only logical reason I can see why tenis playing women have to wear skirts
15:37.19drraytradition
15:37.27drrayback when women were not allowed to wear pants
15:37.37fourcheezehehe
15:37.44fourcheezeover hear "pants" are underwear
15:37.46eKo1blame the british
15:37.57mutlooks hot as hell on those courts anyway
15:38.07drrayand they sell tennis outfits
15:38.08muti wouldn't be in pants
15:38.08fourcheezeso wearing a skirt *because* you're not allowed to wear pants is quite an interesting thought
15:38.26drraywell, we did not have berkas
15:38.28drrayback in the day
15:39.10*** join/#asterisk Overworked554 (n=Ken@atlantis.clearshout.com)
15:39.45fourcheezeI'm pretty sure I've seen women playing tennis in shorts
15:40.06Hmmhesayswomen with penii
15:40.23Hmmhesaysthat we generally refer to as men
15:41.09muti think the proper term is hermaphrodite
15:41.53fourcheezehttp://aeltc.wimbledon.org/en_GB/about/history/fashion.html
15:42.30Vorondilhi all, quick question: when using variables in a dial plan, the expression, "$[2${EXTEN:-2}CELL]" would end up as, say, say, "201CELL" (assuming the correct value of ${EXTEN}).  what if "201CELL" is the name of a global variable?  is there a way to get asterisk to stick the value of /that/ variable in there?
15:42.50*** join/#asterisk SplasPood (n=jwb@206.252.198.101)
15:42.51Vorondil(sans the extra "say"..  :-P  )
15:44.13JackEstormugh, has anyone had problems with Agents locking up the queue, and then leading to * hanging?
15:44.19eKo1Vorondil: are you trying to make variables dynamically or something?
15:44.29feld_JackEstorm, no but I feel your pain :(
15:44.59VorondileKo1: yeah, i suppose you could call it that. the global var is already defined though
15:45.17[TK]D-FenderVorondil : ${2${EXTEN:-2}CELL}
15:45.56Vorondil[TK]D-Fender: so you don't need the sqare brackets for concatenation?
15:46.01mostyi have a TDM400P, when i pickup my handset, the * console says Starting simple switch on 'Zap/2-1' but as soon as i dial a number it says Hungup 'Zap/2-1' - it doesn't appear to be traversing my dial plan. what could be wrong?
15:46.16JackEstormfeld_: I can't belive that after all this time Agents is still buggy as hell, but all these problems say it is.
15:46.22[TK]D-FenderVorondil : no. [] is for logical operations only.
15:46.23mr_horsepowergxp-2000 -> wonderfull :D
15:46.48muthow do yuo tell if an incoming call has no callerid?
15:46.49mostymy zapata.conf has context=voicemail , and extensions.conf has a single extension defined in the voicemail context
15:46.50Vorondil[TK]D-Fender: ahh, i gotcha.  thanks  ^_^
15:46.53Hmmhesays<PROTECTED>
15:47.02Hmmhesaysbah
15:48.00mutvia the dialplan ofcourse..
15:48.03frk2did somebody mention gxp 2000? :)
15:48.26*** join/#asterisk oej (n=oej@apollo.webway.se)
15:48.39mr_horsepowerfrk2: wonderfull, everything works nice, i just dont know how do write the dial number with alpha-numeric numbers.
15:48.54coppicethere is a press release about them using a TI DSP in the GXP2000 like its a new model. have they done a revamp of it?
15:49.04frk2Horsepower- are you saying you have no issues with the GXP?
15:49.22mr_horsepowerfrk2: not even 1
15:49.27mr_horsepowereverything works nice.
15:49.36frk2how many you have / how long you been using them / call load on the GXPs
15:49.41frk2when did you buy them?
15:49.59mr_horsepoweri'm just testing this one.
15:50.02asterisk-dudhello everyone
15:50.15mutanyone know if there is a way?
15:50.21mutcause callerid still comes in..
15:50.25mostyno matter, i found the error
15:50.27mutbut the call is passed as private
15:50.30frk2ah
15:50.31mutso no callerid shows
15:50.35frk2wait till you put it in production
15:50.36frk2:)
15:50.37mr_horsepowerwe need good sip phones, and puting cisco away, i havent found any good phones until this one.
15:50.38muton the user end..
15:50.51frk2its feature rich... just not reliable
15:50.53drraywhy would you get rid of cisco phones?
15:51.02mr_horsepowerfrk2: whats about the problem?
15:51.04frk2if it didnt fuck up as much it would be an awesome phone
15:51.10fourcheezemr_horsepower: have you tried the SPA-941 ?
15:51.11*** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de)
15:51.18mr_horsepowerfourcheeze: no.
15:51.20mr_horsepowernot yet
15:51.23fourcheezeworks for me
15:51.28Overworked554im having probs with my 7960. When an inbound call comes in and i pickup the receiver it hangs up on the caller. its running 8.3
15:51.31fourcheezeas a basic but good phone
15:51.46asterisk-dudI would like to make a dialplan: _20XX that would take the extention dialed (I would have another variable with name equal to the exten, this variable would equal the channel to dial for that extension)
15:51.51Overworked554other than that the ciscos work pretty well
15:51.53mr_horsepowerfourcheeze: do you have sip url dialing?
15:51.57mr_horsepowerwith dots?
15:52.03drraymy ciscos have been rock solid
15:52.06}btorch{hey this is wierd , if I change my voicemail password throught the voicemailman options shouldn't that password be saved on my voicemail.conf file ?
15:52.09fourcheezemr_horsepower: no idea
15:52.10[TK]D-Fenderfourcheeze : SPA-941 is decent, but just not worth the money in North America....
15:52.11drraybut I am running 7.x something
15:52.15asterisk-dudi need the dialplan to take the exten and find the channel value for it and dial it
15:52.37mr_horsepowerfourcheeze: well i need to know that. i will try to test that phone. how much you can find that sip phone?
15:52.43asterisk-dudany ideas?
15:53.01fourcheezemr_horsepower: you can assign a sip url to a speed dial
15:53.05fourcheezeand dial it like that
15:53.06mr_horsepowerthe problem with sip phones it's, the good ones, are so much expensive.
15:53.22drrayyou get what you pay for
15:53.31frk2yeah man.. why not get a polycom 501 for the same price as the 941
15:53.32drrayI"ve not had to dick with an expensive cisco phone
15:53.47}btorch{it seems like everytime i reload app_voicemail.so or restart *  the mailboxes are reset
15:53.49mr_horsepowerfourcheeze: yes, can you try a sip url just like "foo.bar(at)domain.com" it works?
15:53.56fourcheezefrk2: how much is a polycom 501?
15:54.08*** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
15:54.15drrayif you value your time at $20/hr that 9 hours I spent screwing with the budgetone the time it decided to lose its mind
15:54.18fourcheezemr_horsepower: I don't have one on my desk right now otherwise I would
15:54.26[TK]D-Fenderfourcheeze : $170USD.  IP430 is a great alternative @ $160 (so far)
15:54.34fourcheezemore expensive in the UK then
15:54.36vader--can any of oyu guys explain to me what this does
15:54.43mr_horsepowerdrray: well, your hour it's cheap. but you are talking about 1 phone.
15:54.45asterisk-dudcan i use variable arrays for asterisk dialplans
15:54.47vader--exten => _NXXXXXX,1,Dial(ZAP/g1/${EXTEN:1})
15:54.47vader--exten => _NXXXXXX,2,Congestion
15:54.47vader--exten => _NXXXXXX,102,Congestion
15:54.53}btorch{am I configuring something wrong ?
15:54.58fourcheezeover here ip501 is about £180
15:55.07fourcheeze~ $350
15:55.14fourcheezeusd
15:55.15[TK]D-Fendervader-- : It was already explained to you earlier
15:55.21vader--i didn't see
15:55.22fourcheezewhereas the spa-941 is half that
15:55.27[TK]D-Fenderfourcheeze : That royally sucks...
15:55.39mr_horsepowerbut what are the problems with gxp-2000?
15:55.46[TK]D-Fenderfourcheeze : Yeah, thats why I said in North America there is no point to Sipura phones :)
15:55.53fourcheezeyeah
15:56.02frk2well two problems ive faced so far:
15:56.07frk21. It HANGS, randomly
15:56.09Chotairehm, since the last kernel update on opensuse 10.0 I have problems running modprobing ztdummy (zaptel-1.2.4)...
15:56.11Chotaire# modprobe ztdummy
15:56.11ChotaireFATAL: Error inserting ztdummy (/lib/modules/2.6.13-15.10-default/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg)
15:56.17frk22. Screen goes blank and voice leaves
15:56.23ChotaireI never had this problem before.. anyone run across this shit yet?
15:56.24[TK]D-Fendervader-- : Scroll up
15:56.25vader--oh ok i see now
15:56.32mr_horsepowerwho the hell uses suse to install a pbx?
15:56.33mr_horsepower:o
15:56.33vader--someone said i should play the congestion sound?
15:56.49sevardmr_horsepower: correction, who uses suse
15:56.50[TK]D-Fendervader-- : Correct
15:56.56Vorondil[TK]D-Fender: hmm, using ${} doesn't seem to work.  i get this (http://pastebin.com/762601) in the console
15:57.00eKo1Chotaire: make clean and recompile
15:57.02Vorondilit ends up as a ""
15:57.02drraysomeone who uses suse for everything else in their shop
15:57.07ChotaireeKo1: that's what I did
15:57.20Chotairelike I do after every kernel update.
15:57.30vader--is it just called conjection?
15:57.32*** join/#asterisk aze (n=aze@ACayenne-101-1-12-4.w81-248.abo.wanadoo.fr)
15:57.33vader--conjestion
15:57.38mr_horsepowerfrk2: the issues i have, was, in conference the phone crashed. firmware upgrade solved all my problems.
15:57.51eKo1Chotaire: What kernel are you using?
15:58.01[TK]D-FenderVorondil : There is another function to nest that... can't recall the name....
15:58.18frk2what firmware you using now?
15:58.22frk2the new stable?
15:58.45ChotaireeKo1: found the bug... old zaptel was still running...
15:58.47Chotairethanks anyway.
15:59.03Chotairermmod zaptel ; modprobe zaptel ; modprobe ztdummy  ..fixed it.
15:59.12Chotaireduh ;)
15:59.39sevardIt'd be sweet to have a soft phone sitting on a zapscan exten listening for fax and spitting them to pdf
15:59.40sevard;)
15:59.40sevardevil
16:00.13Vorondil[TK]D-Fender: ah, okay.  any idea the best place to look up such a function
16:00.56mr_horsepowerfrk2: Program-- 1.1.0.13    Bootloader-- 1.1.0.1
16:00.56[TK]D-FenderVorondil : WIKI time.....
16:01.17frk2yup
16:01.22Vorondil[TK]D-Fender: hehe, kk.   thanks much :)
16:01.26frk2horsepower.. deploy the phone at a user and observe for 1 week
16:01.45frk2Do not reboot, do not power cycle
16:02.49}btorch{does * have a voicemail password memory I don't know about besides the voicemail.conf ?
16:03.37[TK]D-Fender}btorch{ : Nope.
16:03.44sevardSo, is * 1.2.9 considered bleeding edge and 1.0.11 considered stable?
16:04.21fourcheeze1.0.11 is considered old
16:04.24[TK]D-Fendersevard : no.  1.0.11 is for people who haven't done the big jump.. think of it like service packs to old versions... a waste if you ask me...
16:04.25frk2no man.. 1.2.x is stable
16:04.26fourcheeze1.2.9 is considered usable
16:04.28frk21.0.x = legacy
16:04.37frk2nice of asterisk dudes to support old legacy software
16:04.41}btorch{[TK]D-Fender:  then how come everytime I change  the password the password in my #included file isn't changed and everytime I reload the app it thinks I'm new user
16:05.00*** join/#asterisk jg (n=jg@1cc-dhcp-91.media.mit.edu)
16:05.06eipianyone: how i can configure my iaxy to register a named server? can I?
16:05.40zoasevard: what if i tell you that will happen soon ? :)
16:05.44[TK]D-Fender}btorch{ : PASTEBIN
16:09.08vader--hmmm ok something is messed up in my zapata.conf
16:09.21vader--when i dial ZAP/g1/${EXTEN:1}
16:09.45vader--it rings the first available line in my zapata.conf instead of going out over the pri
16:09.47vader--hehe
16:11.41mostyhow do i create a voicemail mailbox? is there a script i can use?
16:11.48swytchquit
16:11.48*** join/#asterisk timscott (n=a@d198-53-23-18.abhsia.telus.net)
16:12.37Corydon-wmosty: voicemailboxes are created automatically, as they are needed
16:12.47*** part/#asterisk maik (n=maik@bfs.cs.uni-sb.de)
16:13.11mostyCorydon-w, i just tried to leave a message, and the asterisk console has errors about directories not existing
16:13.35Corydon-wAre you running as root?
16:13.50MikeJ[Laptop]do the directories exist ;)
16:14.03*** join/#asterisk Niosop (n=Niosop@isd1.lvti.cc.nm.us)
16:14.05mostyCorydon-w, no it's running as the asterisk user
16:14.09MikeJ[Laptop]sounds like permissions...
16:14.30Corydon-wmosty: then you haven't allowed the asterisk user to own the /var/spool/asterisk directory
16:14.33mr_horsepowerfrk2: do you know any way to write alpha-numeric numbers in gxp-2000?
16:14.54NiosopHello, anyone know of a patch that will force asterisk to send notify events to subscribed sip clients when max-calls has not yet been reached?  Let me know if I'm not being clear.
16:15.14mostyCorydon-w, it is already owned by asterisk, and rwx for that user
16:15.35Corydon-wmosty: then something under that directory isn't
16:16.22mostythe console says it can't open this file for writing (no such file or dir): /var/spool/asterisk/voicemail/local/501/INBOX/msg0000.WAV, there is no voicemail dir in /var/spool/asterisk/
16:17.04feld_mosty, make one
16:17.06feld_lol :P
16:17.22feld_sounds like more than just that, though =(
16:17.58mostyafter adding that dir, it let me leave a message. i'm surprised that this directory wasn't created when i installed asterisk (debian package)
16:18.14feld_report-a-bug
16:18.17feld_:)
16:18.38mostyit's a package from backports.org, i'm not sure where to report bugs
16:18.46Corydon-wOh, you're not installing from source?
16:18.55Corydon-wThere's your problem
16:19.17NiosopProblem:  ext 100 subscribes to notify events (gxp-2k using blf), but asterisk doesn't send notify events if max-calls is greater than 1 and all the lines are not in use.  Anyone know of a config option or patch that would force it to always send notifications?
16:19.26Corydon-wThe source package works fine.  It's all these packagers who screw up the system
16:20.09*** join/#asterisk Splat (n=Splat@220-253-105-69.TAS.netspace.net.au)
16:21.50mostyit seems to be half working now. it creates subdirs are needed, writes out the files, but the files disappear when i hang up
16:22.02feld_configure: error: C++ preprocessor "/lib/cpp" fails sanity check
16:22.08feld_whats up with that ? any suggestions?
16:22.16Niosopyou have glibc installed?
16:22.18feld_happened b4 and after i updated my system. CentOS
16:22.34feld_I would prefer gentoo but I never installed this damn box =/
16:23.04Niosopyeah, gentoo is nice.  Using A@H right now on CentOS though for testing.  Production may end up being Gentoo if I have time.
16:23.16sevardSlackware my friend :)
16:23.50feld_Niosop, keep a chroot up to date and periodically burn it to a dvd or something
16:23.50feld_then u always have a gentoo install ready to deploy
16:23.58feld_keep it like i686 or somethin
16:24.20Niosopshrug, going to be moving to xen VT server pretty soon hopefully, then I'll just have images ready to go.
16:25.25feld_that would work too
16:25.33feld_we run that VMWare ESX or whatever
16:25.42feld_monitor all the VM's across multiple servers
16:25.45feld_pretty cool setup
16:25.50Niosopnod.
16:25.54*** join/#asterisk oej (n=oej@apollo.webway.se)
16:25.57feld_though it's based on windows which is sucky.
16:26.04Niosopyup
16:26.12diLLecoeh
16:26.14feld_"yeah we can run linux" "but it's running on windows" "so what's the problem?"
16:26.24NiosopSo nobody knows of any way to force notify events even if all available lines are not in use?
16:27.02feld_sevard, used slack a few times. have a friend that really likes it. it's a nice quick system too.
16:29.27*** join/#asterisk assert_true (n=Sunil@59.176.16.254)
16:29.46mishehuI'm looking in a recent zapata.conf file, and wondering why would zaptel need a jitter buffer
16:29.58[TK]D-FenderYup... Slackware = trouble-free for me....
16:30.10Daminfield: Vmware ESX is NOT based on Windows. It runs a customized RedHat distribution for the Vmkernel.
16:30.17sevardfeld_: I heart slax
16:30.33Daminfield: Vmware "G"SX can run on Windows..
16:31.11Daminfiled: But that has been deprecated in favor of Vmware Server..
16:31.31feld_Damin, well whatever it is we're running it's based on Windows I believe
16:31.40feld_I'll go check out the exact name later. I'm lazy.
16:31.44feld_:P
16:32.23Daminfeld: Well, Vmware Server can run on Linux or Windows..
16:32.23feld_omg how retarded guys
16:32.34feld_error is because centos doesnt come with g++
16:32.46feld_how counter intuitive
16:33.12[TK]D-Fenderfeld : Did a MINIMAL install... I had a client that started the same way....
16:33.13Daminfeld: And if you are trying to run Asterisk on Vmware, you are in for a really crappy experience..
16:33.19feld_here's my next question: how did these guys get asterisk installed from source if g++ has never been on this system?
16:33.33[TK]D-Fenderfeld : I have to load at least a dozen packages to get everything he needed to get up and running...
16:33.34feld_Damin, no i'm on a real server for Asterisk.
16:34.07feld_I know it requires low latency high priority and nothing else eating its resources
16:34.17Niosopfeld_, they compiled it on a different system, or removed the compiler toolchain after compiling?
16:34.55feld_Niosop, no they arent smart enough
16:35.10feld_for example: SSH is still open to passwords.
16:35.21feld_they installed X
16:35.32feld_these arent bright people we're talking about here lol
16:35.45feld_at least they leave me alone though because they trust I know what I'm doing.
16:36.05Hmmhesaysok I have a stupid error now trying to compile an old version of zaptel
16:36.08Hmmhesaysmy paths are farked up
16:37.04[TK]D-Fenderviler : Your PM's don't seem to be working.. you need to be registered on FreeNode for that.
16:38.13*** join/#asterisk docE (n=docelmo@66.237.242.41.ptr.us.xo.net)
16:38.30feld_[TK]D-Fender, u cant get into this chan without being registered
16:38.43[TK]D-Fenderfeld_ : Thats intermittant...
16:38.53[TK]D-Fenderfeld_ : Its deactivated occasionally.
16:40.10feld_ic
16:42.25*** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.9.1 and 1.0.11.1 released, please upgrade immediately (June 6, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx
16:42.38[TK]D-FenderLOL!
16:42.41russellb:D
16:42.42*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
16:42.45[TK]D-Fenderrussellb : Already?
16:42.48russellbyes
16:42.57feld_aww shit
16:43.00russellbthere was a problem with the security fix :(
16:43.02[TK]D-Fenderhaha
16:43.07feld_i just grabbed like 20 mins ago
16:43.12[TK]D-Fenderrussellb : the "fix" is in!
16:43.13feld_am i up to date or not? ROFL
16:43.21*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
16:43.26russellbfeld: not anymore :)
16:43.29mitcheloc*sigh*
16:43.29[TK]D-Fenderfeld_ : Dunno, ask in 20 mins ;)
16:43.35feld_HAHAHA
16:43.38mitchelocrussellb: lol, is this in cvs?
16:43.49sevardhow does one reload the manager.conf ?
16:43.52[TK]D-Fendermitcheloc : SVN & FTP I suspect
16:43.52russellbno more cvs
16:43.55*** part/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au)
16:43.59russellbsvn and ftp, yes
16:44.14mitchelocrussellb: haha, i mean svn, thats how long ago i updated ;)
16:44.29russellbfeld_: you can actually just download a patch against the previous release
16:44.33russellbno need to download it all over again
16:44.42russellbthat's why we release patch files with out releases :)
16:44.51feld_russellb, lol ok thx
16:44.56feld_i'll go do that :)
16:45.01russellbcool
16:45.05[TK]D-Fenderrussellb : How far back does the exploit go?
16:45.14russellb[TK]D-Fender: very far
16:45.19mtaht3gah
16:45.20[TK]D-Fenderrussellb : O
16:45.40mtaht3updating 15 servers now...
16:45.42mitchelocrussellb: is there a mailing list for security alerts?
16:46.04russellbmitcheloc: asterisk-announce gets all release announcements, which would include security alerts
16:46.21mitcheloccool, i'll sign up on it then
16:46.37[TK]D-FenderI just troll in here..... works plenty fine....
16:46.53docEso any other security issues other than the IAX that just showed up?
16:47.54Hmmhesaysok what is a *.a file ?
16:48.09docEa lib
16:48.13docEor library
16:48.21Hmmhesaysdoes it matter what that library was compiled for?
16:48.24docEbut it hasnt been compiled to be shared..  You have to link it.
16:48.33docEIts just a library
16:48.41Hmmhesaysforgive the n00b question
16:48.42Hmmhesaysis that a no?
16:48.53russellbwhat do you mean, what it was compiled for?
16:48.56russellbarchitecture?
16:49.10Hmmhesaysyeah
16:49.10docEDepends on the library
16:49.16russellbyes, it absolutely does matter.
16:49.32Hmmhesaysok
16:49.36sevardhow does one reload the manager.conf ?
16:49.45vader--any of you guys know a good included sound with asterisk that would be good to play if someone tried dialing a 1900 number?
16:49.45docEstop now
16:49.49docEsafe_asterisk
16:49.59sevardthat's the only way? :|
16:50.00mitchelocsevard: asterisk -vvvvr, reload
16:50.21russellbthe reload CLI command will do it ...
16:50.35mitchelocdoes anyone know if you can get stdoutput through the manager api?
16:50.39docEvader-- I can make you one that say's "DAMNIT STOP CALLING PORN!
16:50.40docE"
16:50.46russellbmitcheloc: no
16:50.57mitchelocrussellb: not even with the System command eh? =/
16:51.05kdz13how to turn off sip debug
16:51.10docEsip no debug
16:51.17russellbmitcheloc: what are you trying to do?
16:51.22docEgood lord..  Does no one used the wiki anymore?
16:51.29russellbdocE: i hope not
16:51.44docEWhy?
16:51.47mitchelocrussellb: i want to get some extended functionality through the manager api,, without the requirement of putting a server piece on the asterisk machine to proxy through
16:51.51sevarddocE: I use the wiki when I need to look up something.. but it's a horrible resource.
16:52.00kdz13docE: thanks
16:52.31docEI found it quite informable when I needed it initially..  I dont use it much now cause of my experience level..
16:52.35docEBut the book is good too
16:53.15docEI had one until a guy I worked with gave it to some dude from South Africa..  last time I ever lend ANYTHING to anyone..
16:54.36*** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net)
16:54.48mishehuhmm...   my pri provider called yesterday to check on why my pri was in lockout.  yet other than echo and choppy echo, I'm not actually seeing any issues on the console.  wouldn't a pri in lockout give an alarm on my t110p card?
16:56.53eKo1Is there a way to configure the te401p so that some ports use T1 and some use E1?
16:57.05eKo1Or do they all have to be either T1 or E1?
16:57.38znoGcan't find the DND setup voip-info.org
16:59.00jsharpYou can set them to T1/E1 on a per port basis
17:01.43cybergypsywhat do I have to do to get calls to sip:user@mydomain.com to enter my asterisk ?
17:01.56HogieIf Im getting:  No D-channels available!  Using Primary on channel anyway 24!   and it also says Primary D-Channel on span 1 down, is that a problem with my machine or the circuit?  They are swearing that it is showing 0 errors on the switch side, but im not getting irq misses or anything else on our * box.  This is on an install that's worked for 1+ years
17:04.10NiosopProblem:  ext 100 subscribes to notify events for ext 101 (gxp-2k using blf), but asterisk doesn't send notify events if max-calls is defined for ext 101.  Anyone know of a config option or patch that would force it to always send notifications?
17:04.30*** join/#asterisk austinnichols102 (n=austinni@70.46.69.131)
17:04.41blitzrageNiosop: bugs.digium.com if it exists... else, you'll have to make it and submit it
17:04.55[TK]D-Fenderblitzrage : ! ! !
17:04.56eKo1jsharp: Can I do that programatically or do I have to do it on the card physically
17:05.46Niosopblitzrage, was hoping not to have to  :)  Gonna take me hours or days to sift through chan_sip.c and figure out what's going on.
17:07.27eKo1jsharp: OK, I found something on asteriskguru about this. Thanks.
17:09.00Hmmhesaysok what does this mean stdtime/libtime.a: could not read symbols: Archive has no index; run ranlib to add one
17:11.02b00merWhat does this mean in my * logs? "-- Requested transfer capability: 0x00 - SPEECH"
17:13.23*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
17:14.22blitzrageNiosop: sounds like you need to hire a consultant than
17:14.26muthow do you tell if a call is coming in with CID blocked?
17:14.56Niosopblitzrage, lol, naaa, there's a pickup patch for 1.2 that I can probably look through to give me an idea of what is involved  :)
17:15.17blitzragemut: CID isn't used to negotiate the call setup
17:15.44mutif i want to block calls with no callerid
17:15.46mutit is
17:15.47docelmob00mer it has to do with POTS or ISDN connections
17:15.53*** join/#asterisk noky (n=noky@200.69.211.18)
17:15.54nokyhi
17:16.09*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
17:16.19blitzragemut: what technology delivery?
17:16.20feld_can anyone here decipher this build error junk?
17:16.23feld_http://sh.nu/p/1925
17:16.27*** part/#asterisk Overworked554 (n=Ken@atlantis.clearshout.com)
17:16.28mutpri
17:16.32mutto sip
17:16.35nokymy asterisk logs in /var/log/asterisk/full ... i have two question, the first is how can i change this path and filename to log in other place... and the second question is how can i desactivate the logs ..?
17:16.45nokyrussellb: hi
17:16.57[TK]D-Fenderfeld_ : ...
17:17.00[TK]D-Fender~centosbug
17:17.01jbotmethinks centosbug is a problem with the latest Centos kernel (4.2 and 4.3).  To fix it, edit the file /usr/src/kernels/2.6.9-34.0.1.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. Note that you may be using the 'smp' or 'hugemem' kernels. Change the filename to suit.
17:17.06noky[TK]D-Fender: hi
17:17.26[TK]D-Fendernoky : y0
17:17.26feld_[TK]D-Fender, you my friend are my hero.
17:17.42nokyfeld_: is my hero too...
17:19.13mutblitzrage?
17:19.26blitzragemut: hrmmm... might be a command in zapata.conf to block calls without CallerID.. but my brain is realyl tired, and I'm not 100% sure since I've not done that before
17:19.39mutwell it's kind of a selective block
17:19.46mutso zapata wouldn't work
17:19.50blitzragethere is exten => 1234/5555,1,NoOp() format where 5555 would be callerID to match on
17:19.54muti need to do it in the dial plan
17:20.00muti also can't do that
17:20.06mutbecause i do get the callerid
17:20.13muti's just being passed as a private call
17:20.15mutit's
17:20.21blitzrageis there not a channel variable that has that info?
17:20.33mutmaybe
17:20.35blitzrageREADME.variables?  I'm not really sure if there is one or not
17:20.43blitzragemight be worth a look anyways
17:20.47muti just know of the callerid variable
17:20.51muti'll check
17:21.22b00merWhat does this mean in my * logs? "-- Requested transfer capability: 0x00 - SPEECH"
17:21.43mut${CALLINGPRES} possibly?
17:21.54muttesties
17:22.13blitzragemut: hrmmm... maybe?  See what NoOp(${CALLINGPRES}) gives you... I'm kinda curious
17:22.46blitzrageI'd even try it here, but I'm in training and don't have Asterisk setup on this box
17:23.46mutwell
17:23.51muttried call 3 different ways
17:24.00mut<PROTECTED>
17:24.00mut<PROTECTED>
17:24.00mut<PROTECTED>
17:24.04*** join/#asterisk Vahram (n=Noname@83.139.6.86)
17:24.05blitzrageheh
17:24.15blitzrageuseful...
17:24.16mutdunno what #3 is
17:24.24b00mer~pb
17:24.25jboti heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
17:24.26blitzragewhat were the 3 different ways?
17:24.36blitzrageoh come on... 3 lines is not a big deal to paste
17:24.43b00mer3 lines of junk
17:24.48mutfirst was dialing in via our internal system, pbx -> adtran -> box
17:24.56blitzragejunk? we're trying to debug something in asterisk
17:25.08mutseconds was picking up an outside pots landline then dialing in -> adtran -> box
17:25.18mutthird was same outside pots line with *67
17:25.27blitzragehrmmm...
17:25.32*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.234.37.Dial1.SanJose1.Level3.net)
17:25.39blitzragewonder if the codes are listed in chan_zap...
17:25.43*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.234.37.Dial1.SanJose1.Level3.net)
17:25.56blitzragelet me checkout the code here and take a look -- let me know if you find it before me
17:25.57mutwiki has it
17:25.58muthttp://voip-info.org/wiki/index.php?page=Asterisk+cmd+callingpres
17:26.01*** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon)
17:26.02blitzragewell thast handy
17:26.22mut0 i guess means nothing exists..
17:26.33blitzragemakes sense
17:29.01blitzragehrmmm... was kinda hoping there was a funciton for that which would return a string instead of numeric value
17:29.51tzangerblitzrage: macro it
17:30.03blitzragemut: you could also verify by doing a SetCallerPres() and then loop the call back in and do a NoOp() to see what the value is
17:30.29blitzrageif you wanted to verify the codes
17:30.51*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-2-21.red.bezeqint.net)
17:30.53blitzragetzanger: I just meant a dialplan macro that would convert the Pres number to a string for you
17:31.04blitzrageerrr... not macro - function :)
17:31.16blitzragelike... CALLERID(pres) or something
17:31.21tzangerbitch bitch bitch
17:31.26blitzragetzanger: look who's talkin!
17:31.31mutlaff
17:31.38*** join/#asterisk ian_k (n=ian@gateway.digium.com)
17:31.50blitzragetzanger: you're the bitch master! :)
17:31.53tzangerheh
17:32.24blitzrageI'm just trying to learn from the best
17:32.39tzangeryou just keep that in mind, my friend...
17:32.46blitzrageoh... I will
17:32.49blitzrageand how!
17:33.04mutman i mosquito bit my on the inside of my hand yesterday
17:33.30mutand it's like bruised now, aparently cause it couldn't raise my skin cause it's callused it had to go deep
17:33.50[TK]D-Fender[13:31] <blitzrage> tzanger: you're the bitch master! :) <- Thigh-Master on "squeeze" ? ;)
17:33.58*** part/#asterisk ian_k (n=ian@gateway.digium.com)
17:34.46*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
17:37.10Vorondilhi all, quick question: does anybody know if there's a dialplan function that will take the contents of one variable and use it as the name of another.  roughly: var1=var2; var2=foo; function(var1) returns foo.
17:37.47Vorondileval() doesn't seem to do it for me (unless i'm missing something)
17:37.49JuggieWHAT?
17:38.23*** join/#asterisk dpryo (i=hn@donatello.nesland.net)
17:38.28Vorondillol, does that not make sense?
17:38.50Juggieno, not really.
17:39.36[TK]D-FenderVorondil : Figured it out.
17:39.37Juggiebecause your words and your psudo code are totally different.
17:40.49mtaht3dang it - an hour ago I started rolling out asterisk-svn
17:40.58mtaht3on a bunch of servers
17:41.04[TK]D-FenderVorondil : ${EVAL(2${EXTEN:-2}CELL)}
17:41.28blitzrageVorondil: almost sounds like you need to use _${VAR} (notice the _ )
17:41.43[TK]D-Fenderblitzrage : Nope... EVAL :)
17:41.56blitzragepfffft
17:41.57[TK]D-Fenderhttp://www.voip-info.org/wiki/index.php?page=Asterisk+func+eval
17:42.23mtaht3is the corrected security update in svn head?
17:42.28mtaht3(as of when)
17:42.35blitzrage[TK]D-Fender: I told you the other day to stop being so damn smart
17:42.38mtaht3s/head/trunk
17:42.41blitzragemtaht3: yah
17:42.45Vorondil[TK]D-Fender: well, i found eval().  it sound like it does *exactly* what i want, but i can't seem to get it to work
17:42.53Vorondili must be missing something
17:42.57mtaht3blitzrage - as of?
17:43.19blitzragemtaht3: as of yesterday I believe
17:43.34[TK]D-FenderVorondil : Pastebin your attempt
17:43.36blitzrager32403
17:43.41blitzragewhich is the rev of 1.2.9
17:44.05Vorondil[TK]D-Fender: alrighty, hold on
17:45.24mtaht3blitzrage: thx
17:45.55blitzragenada problemo
17:50.02Vorondil[TK]D-Fender: http://pastebin.com/762878
17:50.03*** join/#asterisk jgoo (n=e4b80e21@athe730f-2169.otenet.gr)
17:50.16Vorondildo i need to wrap another eval() around that?
17:50.27*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
17:51.41*** join/#asterisk chapeaurouge (n=chapeaur@user-85-201-82-146.tvcablenet.be)
17:51.44[TK]D-FenderVorondil : No, another set of ${} I think
17:52.07Vorondil(err, that should be "called 318 from 207 on line 13)
17:52.09Vorondilokay
17:52.18blitzrageyah.... needs to actually be a variable in the format ${VARIABLE} as opposed to just VARIABLE
17:52.19*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
17:53.40Vorondilbah, now i just get " -- Executing Dial("SIP/207-1beb", "IAX2/username@teliax/|60")"
17:53.53Vorondilso it's making an empty string
17:53.54[TK]D-FenderVorondil : pastebin....
17:54.51blitzrageSet(var2=foo)
17:55.00blitzrageSet(var1=${var2})
17:55.15blitzrage${EVAL(${VAR1})} is the format I believe
17:55.28blitzrages/${VAR1}/${var1}
17:55.31Vorondilhttp://pastebin.com/762893  :)
17:55.59*** join/#asterisk JINDAL (n=root@220.226.79.207)
17:56.01[TK]D-FenderVorondil : ok, try nexting the eval
17:56.16Vorondil[TK]D-Fender: nexting?
17:56.22blitzragecan you show the 2 vars you're setting?
17:56.45[TK]D-Fendernesting*
17:57.01[TK]D-Fenderblitzrage : he's trying to evaluate the variable to pick up.....
17:57.37Vorondil[TK]D-Fender: i gotcha
17:57.38blitzrage?
17:57.41blitzrageI'm confused
17:57.44*** join/#asterisk harlequin516 (n=sham@65.39.84.194)
17:57.47blitzragemoreso than usual
17:58.06JINDALhuy guys, am a total newbie for asterisk......... any suggestions where shd i start wid .......... i hav installed asterisk do i also need Zaptel
17:58.06[TK]D-FenderVorondil : like http://pastebin.com/762903
17:58.07JINDALLibpri, Addons, Sounds
17:58.25[TK]D-FenderJINDAL : Depends what you intend on usings
17:58.28blitzrageJINDAL: www.asteriskdocs.org and click on "Read the book online" on the left hand side
17:58.48harlequin516Are cellphones and SIP phones protocol compatible?  I mean codec and protocol?
17:59.04*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
17:59.06blitzrage[TK]D-Fender: I think you're missing a $ are you not?
17:59.07JINDALokey...
17:59.40blitzragein front of the inside EVAL
18:00.02harlequin516Will public PSTN TElco's use e164.org enum?
18:00.08blitzragedoubt it
18:00.10[TK]D-Fenderblitzrage : Stop being so damned smart ;)
18:00.10harlequin516Do any now?
18:00.22blitzrage[TK]D-Fender: lol ... just good at finding syntax errors...
18:00.44[TK]D-Fenderblitzrage : Yes... even *I* make them... I'm just better at displosing of the witnesses :D
18:01.00blitzrage[TK]D-Fender: lol -- YOU make errors? Does not compute.
18:01.07Vorondil[TK]D-Fender: http://pastebin.com/762907  it still doesn't get to the contents of the global.
18:01.16tzangerblitzrage: not much computes with you.  :-p
18:01.23blitzrageVorondil: can I see how you're setting the variables?
18:01.27blitzragetzanger: its true
18:01.35Vorondilindeed, just a sec
18:01.53[TK]D-FenderVorondil : I'm convinced you'll need to do it in 2 stages at least.... go play around with it for a while :)
18:02.40JINDAL[TK]D-Fender, plzz gimme very short info on Zaptel, Libpri, Addons, Sounds
18:02.52harlequin516Anyone have IAX service provider that doesn't send DTMF properly?
18:03.21Vorondil[TK]D-Fender: here's how he globals are defined.
18:03.42[TK]D-FenderJINDAL : I asked you want you intended to DO with *.  why would I waste time describing stuff you may never need?
18:03.47*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
18:03.54Vorondil[TK]D-Fender: so i should try setting the stuff with EXTEN in it to a new var, then eval()ing the new var?
18:04.00znoGanyone know how to define a blank variable in AEL?
18:04.03*** join/#asterisk vinkega_farmer (n=v_farmer@snoopy.xs4all.nl)
18:04.09[TK]D-FenderVorondil : Just a though... use AstDB... much easier for what you're doing....
18:04.14znoGfoo=""; sets the variable as the actual "", I want it to be just BLANK.
18:04.24*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:04.35blitzrageznoG: foo=  ?
18:04.42blitzrageand don't add ; as its not PHP :)
18:05.05znoGthis is AEL
18:05.08blitzrageahhh
18:05.10znoGin AEL there *is* ;
18:05.21blitzrageyah -- you didn't say it was AEL :)
18:05.22JINDAL[TK]D-Fender, okey my need is to setup a voip+ePBX system integrated wid regular PSTN analog/digital line
18:05.30znoG15:05 < znoG> anyone know how to define a blank variable in AEL?
18:05.34Vorondil[TK]D-Fender: AstDB, eh?  i'll look into that
18:05.40blitzrageznoG: oh -- I totally missed that line :)
18:05.45harlequin516What's AEL?
18:05.46znoGfoo=; didn't work btw.
18:05.51blitzrage~ael
18:05.52jbotrumour has it, ael is Asterisk Extension Language - a dialplan language with 'c like' syntax?
18:06.08harlequin516Oooh
18:06.26blitzrageznoG: hrmmm... good question -- might want to email murf and ask him...
18:06.46*** join/#asterisk Arno[Slack] (n=root@66-163-12-60.ip.tor.radiant.net)
18:06.49[TK]D-FenderJINDAL : then you need ASterisk + Zaptel + Librpri
18:07.17[TK]D-FenderJINDAL : Addons gives support for MP3's for MoH, etc... useful, but not always necessary.
18:07.51JINDALokey, thanks [TK]D-Fender
18:08.22JINDALone more query will d pdf frm asteriskdocs.org describe them....... or i need to look somewhere else
18:08.50[TK]D-FenderJINDAL : Go read The Book for a while...
18:08.52[TK]D-Fender~book
18:08.54jbotbook is probably a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
18:09.05JINDALoks, thanks
18:11.18*** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)
18:12.01*** join/#asterisk lars-ut (n=lars-ut@70.103.228.158)
18:12.08rene-mmm
18:12.19rene-has anyone experience using RAMI?
18:13.33rene-and is it dangerous to connect to the manager console to get a sample of manager output in a running system, just reading events no sending and just one connection
18:14.16harlequin516Hmm, OKay I looked at AEL, I'm not convinced of merits above and beyond extentions.conf.  What's the big appeal.  I was hoping for a true touring complete language..  It appears to just have some superficial syntactical similarity with respect to operators.  Having read the hints/warnings, I am afraid of wasting days trying to perfect this hazzzardous art,
18:14.48harlequin516Does AEL just create addenda to the standard dial plans?
18:14.49[TK]D-Fenderharlequin516 : Its cleaner if you do a lot of loops, but adds nothing that you can't do with standard extensions.conf
18:15.11[TK]D-Fenderharlequin516 : The AEL processor compiles it BACK to extensions.conf std dial-plan...
18:15.20harlequin516okay
18:15.20*** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu)
18:15.34[TK]D-Fenderharlequin516 : Basically.. its worthless
18:15.41harlequin516No need to add complexity for me right now, I suppose.
18:16.06harlequin516It didn't give me the impression of being clean/usable.
18:16.10kaz0358out of curiosity, anyone know why asterisk was re-released again today? was the iax2 security vulnerability not in the 1.2.9 release like they said it was?
18:16.31harlequin516THough I do see how it has appeal compared to extensions.conf
18:16.45*** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org)
18:17.36Katty[TK]D-Fender: did i read somewhere that asterisk has a module for reciving a signal from an alarm console?
18:17.58}btorch{hmm vmail.cgi doesn't really allow a user to change much of their setting , righ t?
18:18.41*** join/#asterisk Chris_Stevenson (n=Mif`@209.172.67.146)
18:19.00harlequin516If I forward a Zap Channel to Dial(IAX2/sham@myco.biz.com/1001), will the DTMF coming from the ZAP channel convey to IAX as IAX DTMF frames?
18:19.32Chris_StevensonHello, I am hoping someone in here might be able to help me get in contact with anyone at NuFone. Their support phone numbers appear to be disconnected, and Support/sales@ emails go unanswered
18:20.12blitzragekaz0358:  channels/chan_iax2.c: clean up yesterday's security fix to not
18:20.13blitzrage<PROTECTED>
18:21.08blitzrageand: * callerid.c: Bug 7268 - Callerid leaks memory on error
18:21.16Chris_StevensonIs anyone else using nufone?
18:21.27blitzragekaz0358: I just checked the ChangeLog file -- you should do the same ;)
18:21.31harlequin516Any clues about DTMF conveyance, across disparate channel types?
18:21.53blitzrageI only looked because I didn't know a .1 was released :)
18:21.55[TK]D-FenderKatty : Can you elaborate on your need?
18:23.11Katty[TK]D-Fender: my boss just sent me a message about an alarm reciever thingy... i /think/ he meant alarmreceiver.conf which i'm looking at right now
18:24.04Katty[TK]D-Fender: all he said is that it recieved signals from an alarm panel in your house.
18:27.01*** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
18:27.09feld_what setting plays the pre-recorded message when you are pushed to someone's voicemail box?
18:27.09[TK]D-FenderKatty : news to me... going to look now
18:28.13Katty[TK]D-Fender: okies.
18:28.21Katty[TK]D-Fender: also! http://www.voip-info.org/wiki/index.php?page=Asterisk+config+alarmreceiver.conf
18:30.03[TK]D-FenderKatty : Beat you there :)  Yeah... could be useful for something I guess....
18:30.19[TK]D-FenderKatty : boss wants notifications on alarms?
18:30.32Katty[TK]D-Fender: i don't comprehend how it works. that's my issue
18:30.58Katty[TK]D-Fender: right now we don't have an alarm control panel attached to a wall somewhere.
18:31.14*** join/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net)
18:31.16[TK]D-FenderKatty : It logs a specific proprietary alarm monitoring language and allows you to trigger apps off it.
18:31.35[TK]D-FenderKatty : Oh.. then this is useless to you :)) its for interfacing with things you don't have :)
18:31.42Katty[TK]D-Fender: and even if we did, i can't imagine what sort of thingy we'd have going because of it
18:31.53Katty[TK]D-Fender: yeah, but one of our clients might like it
18:32.52[TK]D-FenderKatty : And your "thingy" would have to speak that specific language as well... basically its warrantee-less partially usable code....*maybe* :)
18:33.07*** join/#asterisk tdi (n=tdi@reykin.pozman.pl)
18:33.39Katty[TK]D-Fender: so only some security panels can 'interface'?
18:34.04*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
18:35.52[TK]D-FenderKatty : Yup... its a specific protocol... and security world is very proprietary...
18:36.05}btorch{is it possible to configure asterisk to keep all user's preferences ov the voicemail system into mysql ?
18:36.43*** join/#asterisk Bert- (n=bert@i05v-87-90-132-119.d4.club-internet.fr)
18:36.49Bert-hello there
18:37.14Bert-does someone knows a good MGCP softphone for Linux plz ?
18:37.19Katty[TK]D-Fender: connects via phone line or cat 5 line?
18:37.41[TK]D-FenderKatty : Analog.  its an FSK based tech like modem/CID
18:38.13AltnTabis there any way to specify in zapata.conf what echo cancelerr to use: MARK2 for example
18:38.14asterisk-dudy can't i get my call transfer to work, zap channel, it does nothing when i dial *2 or even press flash
18:38.31Bert-~softphone
18:38.32jbotsomething that should be drug out into the street and shot
18:38.32[TK]D-FenderAltnTab : No, it is chosen at compile time.
18:38.41tzafrir_laptopAltnTab, no, those are defined at build-time
18:38.47tzafrir_laptop(of zaptel, not of asterisk)
18:38.52Bert-erf
18:38.52AltnTabso i have just to uncomment it ?
18:38.58*** join/#asterisk kink0 (n=k@62.37.205.161)
18:38.58Bert-~MGCP
18:39.00jbotit has been said that mgcp is Media Gateway Control Protocol
18:39.00AltnTabif i uncomment all ?
18:39.02kink0hi
18:39.07*** join/#asterisk rvhi (n=rv@66.175.65.89)
18:39.09AltnTab[TK]D-Fender,
18:39.38MatsKI begin to get an idea who have writen jbot's answers :-)
18:40.00kink0any idea why SS IXC -> Asterisk , the IXC CDR counts the total call duration for all calls and log CDR even if BUSY/Congestion is returned ?
18:40.38AltnTabtzafrir_laptop, hm, i have uncomment MARK2 and set echocancel=yes, echotrainning=yes and still have haevy echo problems
18:40.42Dr-Linuxwhat's new in version 1.2.9 ?
18:41.19tzafrir_laptopa fix to a security issue with IAX
18:41.37[TK]D-FenderAltnTab : Zaptel EC isn't always so great... thats when you should consider getting an EC enabled card....
18:41.41AltnTabtzafrir_laptop, i've recompiled all, zaptel, asterisk, run fxotune, load it, restart asterisk, calls ok in-out but ugly echo
18:41.44Dr-Linuxtzafrir_laptop: ok thanks
18:41.49[TK]D-FenderDr-Linux : and thats 1.2.9.1 BTW :)
18:42.14tzafrir_laptopAltnTab, did you set opermode to the right value?
18:42.28tzafrir_laptop(before messing with fxotune)
18:42.42AltnTabtzafrir_laptop, rxgain, txgain ?
18:43.04tzafrir_laptopopermode is a parameter of the kernel module wctdm
18:43.17AltnTabtzafrir_laptop, no, haven't mess with that
18:43.38AltnTabtzafrir_laptop, where should i read more ?
18:43.43Dr-Linuxwhat's wrong with GSM , i'm getting this very frequantly >> Jun  6 11:08:54 WARNING[4600]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data
18:44.10tzafrir_laptopAltnTab, in what country are you?
18:44.24AltnTabtzafrir_laptop, bulgaria
18:44.34AltnTabtzafrir_laptop, eastern europe
18:44.58*** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)
18:45.00tzafrir_laptopAltnTab, not everybody here are americans ...
18:46.02tzafrir_laptopyou should set opermode=BULGARIA
18:46.11tzafrir_laptopI believe that this is case-sensitive
18:46.35tzafrir_laptopthat is:
18:47.03tzafrir_laptopoptions opermode=BULGARIA
18:47.04lars-utexample of Asterisk TSP with integrated CRM?
18:47.08tzafrir_laptopoptions wctdm opermode=BULGARIA
18:47.35AltnTabtzafrir_laptop, ok, tnx
18:47.40tzafrir_laptop(use the latter) in /etc/modprobe.conf or /etc/modprobe.d/zaptel if you have that file
18:47.49*** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon)
18:48.42brettnemhey anyone else having trouble compiling asterisk svn head?
18:49.22feld_brettnem, u gettin errors?
18:49.30Dr-Linux[TK]D-Fender: any idea why i got this error on CLI almost for hundred of times?
18:49.33Dr-Linuxwhat's wrong with GSM , i'm getting this very frequantly >> Jun  6 11:08:54 WARNING[4600]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data
18:49.45brettnemfeld_ yes, a menuselect error regarding app_osplookup
18:49.45[TK]D-FenderDr-Linux : nope.
18:49.57brettnemit's probably user error....
18:50.07feld_no idea, brettnem, but i hope u get an answer
18:50.15brettnemgreat
18:50.16Dr-Linuxsomeone post the same error on list, but no one answered him
18:50.16brettnemthanks tho
18:50.34techman97_andyDr-Linux - do you have any new SNOM Phones running?
18:51.08Dr-Linuxtechman97_andy: never use SNOM phone. i use SJphone, Cisco phones and eyeBeam
18:51.30techman97_andyDr-Linux:  When did the messages start happening?
18:51.31brettnemyeah, I compiled svn head a few days ago no problem. but today I tried again with today's svn head and it won't compile
18:52.22brettnemany ideas??
18:52.30MikeJ__brettnem, chat with russellb....
18:52.39MikeJ__menuselect is his baby
18:52.45*** join/#asterisk chin1 (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net)
18:52.51techman97_andyDr-Linux: like # of days ago or whatever - I had a lot of codec warning like that when I was running the newest eyeBeam client
18:52.53brettnemrussellb: you around?
18:53.01chin1how come this ata isn't giving me a dial tone at all ?
18:53.04brettnemthanks mike
18:53.13brettnemMikeJ__: can I just disable it?
18:53.14techman97_andyyou have to speak nice to the ata
18:53.16techman97_andy=P
18:53.26chin1im being really really nice
18:53.37techman97_andydo you see the ata register in the CLI?
18:53.42brettnem~seen russellb
18:53.55jbotrussellb is currently on #asterisk (19h 26m 56s). Has said a total of 58 messages. Is idling for 2h 2m 26s, last said: 'docE: i hope not'.
18:53.55MikeJ__dunno.. havn't really looked at menuselect... I think it reads in a config file
18:53.55[TK]D-Fenderchin1 : And it would help if you told us something USEFUL about your scenario
18:53.55MikeJ__brettnem, try #asterisk-dev.
18:54.00nokyJun  6 15:06:37 DEBUG[32567] db.c: Unable to find key '12' in family 'SIP/Registry'
18:54.03nokywhat is that ?
18:54.03blitzragemenuselect allows you to tell Asterisk which modules to compile
18:54.06brettnemMikeJ__: thanks
18:54.13Dr-Linuxtechman97_andy: i'm running eyeBeam client since 5 months, never get this kind of warning
18:54.14chin1techman97_andy: does the ata have to be registered to the server for it to give a dial tone ?
18:54.23nokyi have a gateway sip trying to register with a username 12 and password 12 too...
18:54.26nokywhat is happened?
18:54.27blitzragenoky: means that the key '12' doesn't exist in the family SIP/Registry in the AstDB
18:54.32MikeJ__blitzrage, yeah.. he is actually getting an error in it so it fails to compile
18:54.36chin1[TK]D-Fender: im just setting up a linksys pap2
18:54.54blitzrageMikeJ__: oh yah?  where is the error? (sorry, I missed it the first time around)
18:54.55[TK]D-Fenderchin1 : Yes, it will only give you dial-tone if it is registered to a server.
18:55.13techman97_andyI'm back
18:55.16nokyit exists. :S
18:55.25blitzrageCunningPike: upgrade minor, or upgrade major? (1.0 -> 1.2 for example)
18:55.25[TK]D-FenderCunningPike : Slightly less so if you have to manually merge SpanDSP, but not a big deal.
18:55.27techman97_andychin1:  Fender is correct on that
18:55.32chin1well im looking at my status screen adn it says that my "Registration State: Online"
18:55.57[TK]D-Fenderchin1 : Registered to WHERE is the question....
18:55.59techman97_andyDr-Linux:  That's just my experience with weird codec warnings like that - something new that hit the network that * doesn't know how to deal with properly
18:56.12chin1[TK]D-Fender: i have my sip login to my server on the internet
18:56.21[TK]D-Fenderchin1 : to a "sip show peers" and pastbin it.
18:56.22[TK]D-Fender~pb
18:56.24jbot[pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
18:56.24techman97_andyassuming it's a SIP ATA - do a sip show peers
18:56.32techman97_andyhehehe - Fender, I'll let you take that one
18:56.33techman97_andyyour
18:56.37techman97_andyyou're typing faster than I
18:57.50nokyblitzrage:
18:57.57CunningPikeblitzrage: Very minor: 1.2.8 -> 1.2.9 -> 1.2.9.1 in the space of an hour :D
18:57.59nokyi have realtime with mysql
18:58.07Dr-Linuxtechman97_andy: hhmm.. my understanding is that,  it due to when an extension try to use different codecs.
18:58.16blitzrageCunningPike: doesn't upgrade easily? Seems to just need to do make install -> restart now
18:58.18nokyit exists... this messages appears from a minutes ago
18:58.30blitzragenoky: realtime eh? sorry... don't use it...
18:58.48CunningPikeblitzrage: I was being genuine - it's a snap
18:59.06blitzrageCunningPike: oh... I just saw a :\ and assumed you were sarcastic :D
18:59.09chin1[TK]D-Fender:  i knwo what your gonna say it says "NAT: NO" for methods   http://pastebin.com/763074
18:59.14blitzrageCunningPike: in which case I'd agree with you :)
18:59.25blitzragetook me only 10 mins to upgrade 3 servers last night
18:59.25techman97_andyDr-Linux:  that could very easily be the case as well, but my experience (just sharing it to maybe trigger a thought for someone else) was that my eyeBeam client was using GSM only for about 3 weeks and everything was OK until I went to eyeBeam 3...then it went nuts.  Maybe it's related, maybe not.
18:59.32CunningPikeblitzrage: It's great
18:59.38chin1the netmask seems messed up too ...
18:59.48nokyblitzrage: why ?
18:59.55[TK]D-Fenderchin1 : And that's your ATA?
19:00.06chin1[TK]D-Fender: yes thats the ata connected
19:00.16blitzragenoky: why? because RT adds unneeded complexity in my opinion
19:00.24CunningPikeTada! Connected to Asterisk 1.2.9.1 currently running on dogmatix
19:00.29[TK]D-Fenderchin1 : Keep in mind the PAP2 is a DUAL port.... I only see 1 peer registered.... maybe you plugged it into the wrong jack :)
19:00.29blitzrageok ... going to pay attention to class
19:00.44chin1[TK]D-Fender:  no im on the right jack
19:00.52techman97_andychin1:  Try the other jack
19:00.57chin1omg i wasn't!
19:01.00chin1wtf....
19:01.01*** join/#asterisk __jkj (n=mail@adsl-69-150-161-180.dsl.lgvwtx.swbell.net)
19:01.13chin1i sware i looked like 20 itmes!
19:01.19russellbbrettnem: hey, i'm back
19:01.21russellbwhat's up
19:01.21[TK]D-FenderI geeeeeeeevve you FEEEEEEESHE!
19:01.24chin1no more drugs for me
19:01.25techman97_andyit's always the most basic thing that f*cks you in the end
19:01.36chin1anyway why does the mask say 255.255.255.255 and why does it say nat=n ?
19:01.37[TK]D-Fenderchin1 : No, just BETTER drugs.
19:01.40techman97_andylike forgetting to wear pants to the store.
19:01.45[TK]D-Fenderchin1 : Suivent, NEXT!!!!!!!
19:01.50chin1what ?
19:01.56brettnemrussellb: getting an error on today's svn head
19:02.00chin1omg a dial tone!~!!!
19:02.02[TK]D-Fenderchin1 : Bilingual queue call :)
19:02.09__jkjMy new iAXY works inside my network but not outside.  Any ideas?
19:02.17brettnemrussellb: apps_osplookup
19:02.18*** join/#asterisk themikester60 (n=mikey@cpe-72-181-92-164.houston.res.rr.com)
19:02.21chin1so after i dial 600 do ihave to hit # or just allways wait 10 seconds lol
19:02.22brettnemin menuconfig
19:02.29[TK]D-Fender__jkj : What IP is it looking for?
19:02.33chin1yep!
19:02.38chin1this is so cool
19:02.42*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
19:02.47nokyblitzrage: yes... could be... but i need realtime
19:02.51[TK]D-Fenderchin1 : you can tweak your ATA dialplan to make certain things instant.
19:03.02nokyblitzrage: i trying to start up an ivr with asterisk+realtime + webpage in php
19:03.05__jkjIt is looking for my private ip inside the network and my public ip outside the network.
19:03.09nokyan ivr configuration...
19:03.09*** join/#asterisk watchy (n=watchy@70.238.57.237)
19:03.17blitzragenoky: yah... just not sure if that stuff that is normally in AstDB gets moved to the DB or not...
19:03.21chin1[TK]D-Fender: ok all that stuff is experiment but im still concered why its saying nat=n
19:03.59[TK]D-Fenderchin1 : because you don't have nat=yes for your peer entry.
19:04.02chin1and my netmask is not all 255
19:04.29chin1i have nat settings on both my ata and on my sip.conf
19:04.41[TK]D-Fenderchin1 : the netmask implies taht the IP specified is precise.
19:04.50*** part/#asterisk Chris_Stevenson (n=Mif`@209.172.67.146)
19:04.53[TK]D-Fenderchin1 : Pastebin it.... its not lying you know...
19:05.14techman97_andychin1:  if it works, quit screwing with it
19:05.15techman97_andy:D
19:05.21chin1http://pastebin.com/763100
19:05.40chin1tahts my _sip_users.conf
19:06.02*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
19:06.26feld_what does this mean? : Jun  6 15:05:58 ERROR[2889]: chan_sip.c:10988 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 2001 in context local
19:06.28[TK]D-Fenderchin1 : Dunno.... hmmm.. well if it works I wouldn't worry
19:06.44[TK]D-Fenderchin1 : do "sip show peer methods"
19:06.57chin1what am i looking for ?
19:07.16[TK]D-Fenderchin1 : pastebin :)
19:07.23*** part/#asterisk austinnichols102 (n=austinni@70.46.69.131)
19:07.54chin1it says NAT: Alwasy
19:08.05[TK]D-Fenderchin1 : link please :)
19:08.43chin1http://pastebin.com/763114
19:10.01chin1i dont undersatnd why it needs my internal ip
19:10.08[TK]D-Fenderchin1 : Ok... I'm stumped, but it works... so thts enough :)
19:10.19chin1i had the ata set to dmz before
19:10.21chin1but i changed that
19:10.32chin1maybe if i can de-register some how ?
19:10.41chin1i tried pulling hte power didnt' change teh value
19:11.22[TK]D-Fenderchin1 : ah whatever... it ain't broke.... be happy you're up and running
19:11.37chin1[TK]D-Fender:  hey this is a test for setting up a real system!
19:11.40chin1i cant play no games...
19:11.58chin1why do i allways see these musiconhold messages
19:12.07feld_they want you to listen
19:12.11chin1http://pastebin.com/763126
19:12.17chin1yea but im not even on the phone
19:13.03[TK]D-Fenderchin1 : Means exactly what it says\
19:13.29[TK]D-Fenderchin1 : you've got stuff to clean up.
19:13.43chin1yea its all default files from debian conf package
19:13.55chin1or is that passed down from teh asterisk source ?
19:14.08[TK]D-Fenderchin1 : you should switch to native MoH and not use MPG123, and then find some MP3's to use there and make sure to have compiled the asteriask-addons package
19:14.23[TK]D-Fenderchin1 : SCREW DEBIAN PACKAGES.  Use the Source Luke!
19:14.34chin1[TK]D-Fender: i think debian has it all compiled allready :\
19:14.37chin1hey common its easy!
19:14.46chin1ill setup slack with some built sources on spare time
19:14.55kdz13chin1: as far as I can tell, debian's is quite old
19:15.01chin1right now were paying like 260$ a month for this stupid pstn links
19:15.56chin1im gonna get a polycom 501 for the secretary but these cordless phoens with the ata are fine for the rest of us were never int he office anyway
19:16.04[TK]D-Fenderchin1 : I never said to not use Debian.. just compile * from source.
19:16.45[TK]D-Fenderchin1 : Get her an IP601.  That way she can handle more calls, and maybe expand to the attendant modules,.
19:16.48chin1[TK]D-Fender: yes i know but im more of a slack guy i just used debian because A: they have precompiled binaries, B: my server in denmark is using debian so it was good to test
19:17.10chin1601 ? i think that might be it let me go look
19:17.33nokyi think that is a fucking bug
19:17.34nokyJun  6 15:30:19 DEBUG[1242] db.c: Unable to find key '12' in family 'SIP/Registry'
21:16.46*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
21:16.46*** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.9.1 and 1.0.11.1 released, please upgrade immediately (June 6, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx
21:17.22*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo)
21:19.15*** join/#asterisk Chris1004 (n=chrislro@c-68-49-240-217.hsd1.md.comcast.net)
21:19.28terrapenanyone using ICD?
21:19.41terrapeni'm trying to find the README mentioned on the wiki
21:21.09*** part/#asterisk chin1 (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net)
21:21.11Chris1004what dial plan will make outgoing calls using a T400 card.  Im not sure what to put in the extensions.conf file
21:21.56*** join/#asterisk C4T3l (n=robert@216.54.143.2)
21:22.01C4T3lhello all
21:22.35C4T3lanyone ever install a sangoma a102 before, I'm having trouble
21:22.45nokyhi
21:22.51nokyanybody test the sipp with asterisk ?
21:22.59terrapenthe sipp?
21:23.05nokyyes
21:23.10terrapenwhat the hell is the sipp
21:23.22nokyis a tester for sip
21:23.50Hmmhesayshttp://www.rowetel.com/ucasterisk/ucasterisk.html
21:23.50mercestes*stares at C4T3l*
21:23.51nokysipp generate much calls with your parameters (call per seconds, duration of call, etc)
21:23.56Hmmhesaysthere is decent instrutions there
21:24.00terrapeninteresting
21:24.06nokyi don't know why my sipp say this:
21:24.07noky2006-06-06 18:00:23: (1) No valid Call-ID: header in reply 'SIP/2.0 100 Trying
21:24.10noky:(
21:24.15nokyany ide?
21:24.17nokyidea?
21:24.28nokymy asterisk answer OK, but the sipp quit whit this error...
21:24.34C4T3li guess no one's ever done it??
21:25.13C4T3lThe installation of the wanpipe program dies before it can finish, no output or anything!
21:25.42terrapenno kidding
21:25.55terrapenor learn about strace
21:27.07Juggienoky, ask oej when hes online.
21:28.21*** join/#asterisk chin1 (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net)
21:31.01chin1can i login via sip from multiple places at once ?
21:33.54zoai tested sipp with asterisk before
21:34.23Juggiechin1, no asterisk does not support that.
21:34.24nokyi found the error..
21:34.27Juggiei wish it would, but it wont ;)
21:34.32nokywas compactheaders = yes
21:34.39nokyin sip.conf
21:34.52Juggieyou can login if you like more then once
21:34.59Juggiebut the latest phone to register will receive the calls
21:35.08chin1oh
21:35.16chin1spooky
21:35.22Juggieindeed.
21:35.23mog_workjust give different ids
21:35.29mog_workbut you can tie em together still
21:35.37*** part/#asterisk Arno[Slack] (n=root@66-163-12-60.ip.tor.radiant.net)
21:35.38Juggiei had a good idea for that but no time to code it.
21:35.40chin1my ata for some reason is not causing my cordless phone to ring
21:35.40dlynes_office~seen mitcheloc
21:35.44jbotmitcheloc is currently on #asterisk, last said: 'russellb: i want to get some extended functionality through the manager api,, without the requirement of putting a server piece on the asterisk machine to proxy through'.
21:35.54dlynes_officemitcheloc: ummm...jbot's not on vacation
21:35.58dlynes_officemitcheloc: he just doesn't like you
21:36.03Juggiei would like to see SIP/sipphone be the last phone to register.
21:36.05Juggiebut say
21:36.19JuggieSIP/sipphone[all] be all phones registered w/ that peer name
21:36.28chin1the screen lights up but i dont get any ringing.. and yes the ringer is set to on
21:36.55generalhan~dict Hmmhesays
21:36.59generalhanlol
21:37.07Hmmhesaysheh
21:37.13chin1dam i was gonna say someoen really answered that fast
21:37.18dlynes_officemitcheloc: however, it looks like he hasn't been logging peeps' logins and logouts and channel talk for a while
21:37.19chin1lol
21:37.48generalhandlynes_home: yeah he wasnt answering Hmmhesays a bit ago when he was asking for ~docs and ~thebook
21:37.58dlynes_office~book
21:38.00jbotextra, extra, read all about it, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
21:38.04filehe was gone you dingos :P
21:38.04dlynes_officeThat's why
21:38.05Hmmhesaysi wasn't asking for either of those
21:38.15dlynes_office~thebook
21:38.21Hmmhesaysi had a cross compiling question
21:38.21dlynes_officeThere's no entry for '~thebook'
21:38.34generalhanHmmhesays: no you werent ... i tried to scroll up and correct myself real fast but my scrollback is already past that ! lol
21:38.45Hmmhesaysheh
21:39.06generalhanit was:: <ManxPower> ~docs
21:39.20dlynes_officeyeah...you can only ask jbot about what he knows..not what he doesn't know
21:39.28generalhanyea but ....
21:39.31generalhan~docs
21:39.32jbotsomebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
21:39.40*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
21:39.41dlynes_officeand your point?
21:39.41generalhansee i think he was on a smoke break ! lol
21:39.58dlynes_office<file> he was gone you dingos :P
21:40.09dlynes_officebut i guess generalhan only sees what he wants to see :)
21:40.10generalhanwas he ?
21:40.16generalhandamn straight
21:40.18generalhanlol
21:40.27generalhanwhy would i want to see what i dont want to see ?>
21:41.11generalhanHAHAHA it even says just a bit ago that jbot has joined #asterisk !
21:41.57generalhandlynes_home: ya know .. i have enough issues with my stupid 7960 firmware; without you slapping me
21:42.03dlynes_officelol
21:42.13*** part/#asterisk C4T3l (n=robert@216.54.143.2)
21:42.27dlynes_officeI can't believe cisco has the gall to charge that much for a phone with broken firmware
21:42.31*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)
21:42.34*** join/#asterisk Wowzers10 (n=pbaker@nnat-gw.adeptra.com)
21:42.34generalhanlol
21:42.45*** join/#asterisk kjs3 (n=quux@c-24-98-110-80.hsd1.ga.comcast.net)
21:42.53Wowzers10hello all, I just bought a Digium Wildcard TE410P, and noticed it was a 1 gen card - is it possible to upgrade firmware on these devices?
21:43.00generalhandlynes_home: it sux cause eveytime i upgrade i think its all solved .. a week goes by then BAM back to swuare one
21:43.00dlynes_officebut what's more unbelievable is people pay them that much for a phone with broken firmware
21:43.30dlynes_officeWowzers10: just bought it on ebay?
21:43.35generalhandlynes_home: when i bought these phones we were using CM so they worked GREAT. then when i came over to * i didnt want to buy new phones so i started using them with the SIP firmware
21:43.41*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
21:43.43Dr-Linuxdlynes_home
21:44.02Dr-Linuxplease check this this is really kililng me >> http://pastebin.com/763501
21:44.06dlynes_officegeneralhan: ah
21:44.33vader--any of you guys know a good included sound with asterisk that would be good to play if someone tried dialing a 1900 number?
21:44.46vader--like don't dial this number pervert
21:44.47vader--:)
21:45.06dlynes_officeDr-Linux: what's the two lines before the lines you pastebinned?
21:45.08generalhanhahaha
21:45.08generalhanyea
21:45.09chin1i cant get my phone to ring!
21:45.25generalhanvader--: teletubbie-murder
21:45.36dlynes_officeyeah...that'd be a good sound....1-900-I-CANT-GET-MY-PHONE-2-RING
21:46.09generalhani made all my users record their name for the directory ... so when ever they do something incorrectly i put their recording into the teletubies recording and it works nicely ! lol
21:46.16chin1Enable IP Dialing:
21:46.17chin1what is that ?
21:46.18kjs3vader: someone had a bunch of MP3s of 70s era pr0n movie sound tracks on their web site (Google, I suppose).  Just play that in an endless loop.
21:46.30Dr-Linuxdlynes_home: lemme show you
21:46.39dlynes_officechin1: something yoiu probably dont' want...especially if you don't want your customers to know they're using voip
21:47.01chin1um
21:47.07Dr-Linuxdlynes_home:
21:47.07Dr-Linuxmake -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel-1.2.6 modules
21:47.07Dr-Linuxmake[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686'
21:47.07Dr-Linux<PROTECTED>
21:47.12chin1im trying to get my phone to ring the ata for some reason is not rigning it
21:47.13Dr-Linuxis this gcc library problem
21:47.23dlynes_officechin1: what hardware?
21:47.37dlynes_officeDr-Linux: no, that's normal
21:47.40chin1linksys pap2t-na and a 900mhz cordless
21:47.46*** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com)
21:48.04Dr-Linuxdlynes_home: what could be the problem?
21:48.17JuggieDr-Linux, centos?
21:48.21dlynes_officechin1: the line on the linksys pap2 that you're plugging your cordless into has not registered yet
21:48.42Dr-LinuxJuggie: RHEL AS 4
21:48.43*** join/#asterisk hagler (i=hagler@psychozoo.com)
21:48.44chin1dlynes_office:  yes it has im using it i see the screen light up and i can read the caller id but hte phone just doesn't ring
21:48.49dlynes_officeDr-Linux: i suspect you've got a fubar typedef somewhere
21:48.56JuggieDr-Linux,
21:48.59Juggie~centosbug
21:49.02jbotsomebody said centosbug was a problem with the latest Centos kernel (4.2 and 4.3).  To fix it, edit the file /usr/src/kernels/2.6.9-34.0.1.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. Note that you may be using the 'smp' or 'hugemem' kernels. Change the filename to suit.
21:49.23dlynes_officeJuggie: he's runnning 2.6.9-34.EL-smp-i686
21:49.26dlynes_officeheh
21:49.29Juggieyes
21:49.30Juggieso
21:49.32Juggieuse your brain
21:49.40haglerso anybody know about an issue with spool calls where they fail when the called channel sends any sort of progress
21:49.41dlynes_officeI was...that's why i was laughing :0
21:49.42Juggieand replace the version in the path
21:49.52haglerJun  6 14:39:44 NOTICE[15746]: pbx_spool.c:269 attempt_thread: Call failed to go through, reason 3 == Spawn extension (default, 571, 1) exited non-zero on 'Local/571@default-60cd,2'
21:50.00dlynes_officethat centosbug snippet should have rhel added to it
21:50.09Juggiedrlinux, nano /usr/src/kernels/2.6.9-34.EL-smp-i686/include/linux/spinlock.h
21:50.17Juggieand follow the instructinos a few lines bank.
21:50.18russellbjbot: rhelbug is aka centosbug
21:50.19jbotrussellb: okay
21:50.19Juggie*back
21:51.11Juggiejbot: tell russell to put on pants
21:51.19MikeJ__heh
21:52.51chin1dlynes_office:  you read that ?
21:52.57Juggiehah
21:52.58Juggie[17:51] <jbot> No, juggie, I won't. (target invalid?)
21:53.19dlynes_officechin1: so you can see the incoming call on the phone?
21:53.19Juggiejbot: tell russelb to put on pants
21:53.37chin1dlynes_office:  yes... it works fine i just cant hear a ring
21:53.42dlynes_officeJuggie: two s's, two l's in russellb
21:53.47chin1obviously a ata config right ?
21:54.00dlynes_officechin1: does the phone work in a normal analog jack?
21:54.03chin1Caller Conn Polarity: ?
21:54.03kink0MikeJ__ I read about woomera, but I have a doubt, basically why are you ussing woomera instead the h323 included in asterisk ? at least if you are not ussing it like endpoint or gatekeeper
21:54.10chin1dlynes_office:  yes i tried that
21:54.38dlynes_officechin1: i can't remember...is there a setting for ring voltage on those devices?
21:54.51chin1on the ata or hte phone ?
21:54.54chin1the phone is set to ring
21:54.57dlynes_officeon the ata
21:55.09dlynes_officealso, what brand is the phone?  is it a siemens or a panasonic?
21:55.17chin1Dist Ring Setting?
21:55.23*** join/#asterisk MoutaPT (n=MoutaPT@85.139.196.14)
21:55.24dlynes_officering voltage
21:55.32MoutaPTany one with BRI cards experience?
21:55.37dlynes_officeit will only be called that, and nothing else
21:55.46chin1its an atlinks i think i bought it at radio shack
21:56.23*** join/#asterisk znoG (n=gs@99-211-126-200.fibertel.com.ar)
21:56.29chin1wait
21:56.34dlynes_office?
21:56.35chin1im looking at a whole section on ring settings
21:57.08chin1um
21:57.16dlynes_officethe ring voltage if it's there will probably be on the bottom end of the settings for the line you're trying to use
21:57.26chin1cfwd ring splash len
21:57.32chin1vmwi ring splash len
21:57.35dlynes_officering voltage
21:57.36*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
21:57.39chin1lol
21:57.42chin1no voltage
21:57.47dlynes_officenot ring splash, not ring tone, not ring ..., just ring voltage
21:57.57chin1nada
21:58.08dlynes_officechin1: check on the regional page as well
21:58.16chin1what on the regional page ?
21:58.27dlynes_officechin1: for ring voltage
21:59.04dlynes_officeif you don't see the regional page, it could be that you're logged in as user instead of admin
21:59.36Symmcan you make free international calls with some kind of crazy hookup with asterisk?
21:59.38Dr-Linuxback
21:59.39dlynes_officechin1: you won't see line 1 or line 2 if you're logged in as user, either
21:59.42Dr-LinuxJuggie: what should i changeeeeee?
21:59.49chin1dlynes_office:  i foudn it !
21:59.57dlynes_officeSymm: yeah, if you know the ip address of the person you want to talk to
22:00.00chin1i know that man
22:00.03Symmhmm
22:00.08Symmthanks, any other way?
22:00.11chin1it says 70
22:00.14*** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com)
22:00.20dlynes_officeSymm: own your own telco :)
22:00.29Symmoh i see, so you're saying, you avoid the pstn?
22:00.42Dr-LinuxJuggie: can you tell me what should i change?
22:00.53Symmotherwise, you HAVE to go through a pstn and that costs money no matter waht
22:00.58Symm?
22:00.59dlynes_officeSymm: no, what i'm saying is that for the most part, using asterisk you don't avoid paying long distance, but you reduce the cost of your long distance
22:01.01Symmcorrect?
22:01.06Symmok
22:01.12dlynes_officeSymm: by avoiding pstn
22:01.21dlynes_officechin1: yeah...70's fine
22:01.30dlynes_officechin1: do you have a digital multimeter?
22:01.30chin1dlynes_office:  well thats not working
22:01.40chin1no
22:02.00chin1you think its the cord ?
22:02.15dlynes_officechin1: try taking the pap2 back to the store where you got it from then, and getting them to replace it
22:02.26dlynes_officechin1: you tried the other line port, right?
22:02.32chin1no
22:02.59dlynes_officetry it?
22:05.07Symmso umm how do i find out how much a pstn will charge me for connecting to them?
22:05.20chin1that one doesn't work either
22:05.20dlynes_officeSymm: you mean voip provider?
22:05.22chin1i think its a setting
22:05.27*** join/#asterisk doolph (n=doolph@200.75.204.169)
22:05.31chin1there is no store for generic brand
22:05.41dlynes_officechin1: generic brand?
22:05.57Symmi guess, im still trying to work out the full topography of a US based voip system, from home
22:05.58chin1its not vonage or anything locked in
22:06.02doolphwhat can I do if I get error on compilation --> [chan_zap.o]
22:06.12Symmill just hit the books
22:06.22Symmi bought some asterisk book off amazon i just need to read
22:06.28generalhan~book
22:06.29jbotextra, extra, read all about it, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
22:06.37generalhanyay jbot !!!
22:06.59doolphhello
22:07.04generalhanjbot: generalhan is THE MAN
22:07.05jbotokay, generalhan
22:07.05kink0hmmmmmmmm
22:07.05chin1dlynes_office:  it works!
22:07.06kink0~h323
22:07.08jbotextra, extra, read all about it, h323 is An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is ...
22:07.11generalhan~generalhan
22:07.12jboti guess generalhan is THE MAN
22:07.15generalhanhaha
22:07.18doolphhow can I skip the that chan_zap
22:07.27Dr-LinuxJuggie: awwwwwww you were right
22:07.42dlynes_officechin1: ?
22:07.45chin1dlynes_office:  i switched to trapezoid wave form sinusoid
22:07.49kink0~IXC
22:07.50jbotwell, ixc is an interexchange carrier, also known as a long distance company, that transports calls between LATAs.  Some examples are Sprint, Global Crossing, MCI, and AT&T.
22:07.50dlynes_officechin1: huh?
22:08.05chin1dlynes_office:  it says "Ring Waveform:"
22:08.15chin1and has two options sinusoid and trapezoid
22:08.15dlynes_officechin1: what did you have it set to?
22:08.16generalhan~Hmmhesays
22:08.20chin1trapezoid
22:08.25chin1it rang right away
22:08.30dlynes_officechin1: and now it's set to sinusoid?
22:08.36chin1what ?
22:08.37chin1no
22:08.43chin1<PROTECTED>
22:08.49dlynes_officechin1: it was sinusoid and now it's trapezoid?
22:08.53chin1yes!
22:09.01dlynes_officeok, you've got a weird phone then
22:09.09dlynes_officenormally sinusoid should work just fine
22:09.43Dr-Linuxthere is fucking bug in RHEL 4 :@ :@ :@ :@
22:10.07dlynes_officeDr-Linux: were you raised in the mountains?
22:10.09kink0Dr-Linux, slackware !! heheehe is a block
22:10.46Symmhttp://www.gizmoproject.com/index.html
22:10.46chin1dlynes_office:  i knew i wasn't gonna take it back though you narrowed the answer down to 5 input feilds and you gave me the impression that the voltage was fine so the rest of hte options are really weird talking about frequency's and stuff i figured theyd be fine too the only white nad black answer wass the waveform
22:11.15dlynes_officechin1: yeah...that's just plain weird though...i've never had to change that setting
22:11.16Dr-Linuxdlynes_home: what are you talking about
22:11.22Dr-LinuxJuggie was right
22:11.22dlynes_officechin1: must be the cheap phone
22:11.28chin1lol
22:11.44dlynes_officechin1: I always have expensive keysystems hooked up to them
22:11.56kink0dlynes_office, may be due to A/D D/A conversion affected by bus clocking ?
22:11.58dlynes_officechin1: I've never used anything cheaper than a panasonic cordless hooked up to them
22:12.23*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
22:12.23*** mode/#asterisk [+o russellb] by ChanServ
22:12.30dlynes_officekink0: well, i was thinking maybe the d/a converter in the cordless phone might only be a half wave converter, instead of full wave
22:12.38*** part/#asterisk chin1 (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net)
22:12.47dlynes_officekink0: that usually makes electronics cheaper
22:13.01dlynes_officekink0: but why bother, when you're only saving 40 or 50c?
22:13.54kink0dlynes_office, yes, but that add a lot of armonic distortion, or you will need to reconvert the signal from an asimetric to simetric
22:13.56dlynes_officekink0: besides...i don't know why the phone would be dealing in AC to begin with
22:14.08dlynes_officekink0: the phone line is 70VDC, not 70VAC
22:15.54dlynes_officekink0: oh yeah..one other thing...no bus clockign...the pap2 is a wholly self-contained unit that gets plugged into the wall and the ethernet
22:15.59kink0yes, I see, but signal is modulated like AC and added about 35 volts, so you always has full wave
22:16.13Dr-Linuxkink0: my friend
22:16.14Dr-Linux= Parsing '/etc/asterisk/enum.conf': Found
22:16.14Dr-LinuxAsterisk Ready.
22:16.14Dr-Linux*CLI>
22:16.16Dr-Linux:D
22:16.29kink0Dr-Linux, did you used another gcc ?
22:16.43dlynes_officekink0: he's using redcrap linux
22:16.50Dr-Linuxkink0: nope there was a bug in RHEL , Juggie helped me
22:18.01kink0a question, any idea why IXC takes for her CDR start-time instead asnwer-time when they sends calls to my Asterisk ?
22:18.22dlynes_officewho's her?
22:18.39kink0dlynes_my peer, he used Cisco and IXC for accounting
22:18.42dlynes_officenot to mention they?
22:19.04dlynes_officekink0: ah...I guess you must be Chinese
22:19.25kink0he sends me h323 and the problem is her IXC takes start-time instead answer-time for CDR, so all calls are billed even BUSY or Congestion
22:19.32dlynes_officecause you're mixing him/her/they
22:19.35kink0dlynes_office, hehehe
22:19.58*** join/#asterisk rvhi (n=rv@66.175.65.89)
22:20.07dlynes_officeni shi zhongguo ren ma?
22:20.10kink0ok, ... when they send :)
22:21.12Dr-Linuxkink0: friend i'm going to home, i'll catch you from there
22:21.30dlynes_officeDr-Linux: lazy boy....always going home :)
22:21.39kink0Dr-Linux, ok, take the bus , not the motocycle :)
22:22.22Dr-Linuxdlynes_home: well i suppose to leave my office 5 hour ago .. anyway ..
22:22.26Dr-Linuxbye
22:22.30Dr-Linuxkink0: thanks friend
22:22.31Dr-Linuxbye
22:22.40Dr-Linux/gone
22:24.08dlynes_officekink0: isn't that a good thing?
22:24.12dlynes_officekink0: you get more money then
22:24.13dlynes_office:)
22:24.48kink0dlynes_office, no...no... that is not good !!! my local CDR is ok, my peer CDR is crazy
22:25.01dlynes_officehehe
22:25.37kink0but I have other peers ussing Cisco to my Asterisk, and they has not problem, just this people who uses IXC has this problem
22:25.54dlynes_officesounds like a bug in ixc, whatever that is
22:25.58*** join/#asterisk _4d4m_ (n=adam@62.69.102.99)
22:26.01kink0all calls are computed with duration from start-time
22:26.31kink0yes, that is what I think, I ask him if the problem is just with me, and they confirm they see this problem only with me :(
22:26.35dlynes_office~wiki ixc
22:26.36fholmeshow can I trace down potential incoming IAX communications?
22:26.42dlynes_officefholmes: iax debug
22:26.59kink0I have asked to IXC softswitch ... no answer yet about the issue.
22:27.06*** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com)
22:27.23fholmesOk so how do I enable debugging?
22:27.27dlynes_officekink0: so you mean interexchange carrier, like wikipedia, or is ixc a piece of software?
22:27.34dlynes_officefholmes: iax debug
22:27.45kink0is a piece of software, a softswitch for accounting
22:27.46dlynes_officefholmes: erm iax2 debug i mean
22:27.46fholmesnm.  I read it wrong.  Thanks
22:27.59kink0sorry for confusion about interxchange or so
22:28.00fholmesI figured out the iax2 part atleast.  :-)
22:28.19dlynes_officefholmes: iax2 no debug to turn it off
22:29.05kink0anywise I think there some association or so , due I search for IXC in wiki and I got the site for interxchange who spokes about her softswitch software for voip accounting
22:29.12fholmesOk, so I am registered with my IAX provider and can call out, but the calls are not coming back to me.  I don't see anything come on the screen when I call my DID.  Is there something I am missing?
22:30.06dlynes_officefholmes: is your iax connection defined as a peer, friend, or user?
22:30.29fholmesfriend I believe.
22:30.44dlynes_officefholmes: so you don't know?
22:30.50kink0dlynes_office, http://ixc.ua/index.php?MenuId=22
22:31.18fholmesWell, I am just confused more than anything else.  I have the register => line in there.  However, down below in the iax.conf file I have type=friend
22:32.09fholmesI just noticed my username= was wrong down there.  So maybe that was the issue.  Let me reload real quick...
22:32.15dlynes_officefholmes: hrm...i've never had to do a register for iax
22:33.02dlynes_officekink0: it's incorrectly recording the calls when it's terminating on your end?
22:33.06fholmesSo I don't have to have the register => line in the iax.conf file?
22:33.20dlynes_officefholmes: i'm guessing it would depend on your terminator
22:33.31fholmesregister => account:pass@iax.provider.com
22:33.35kink0dlynes_office, yes, that is the problem.
22:34.09dlynes_officekink0: i would check your end to make sure you're not using an answer anywhere in the dialplan for any calls coming in from them
22:34.11kink0dlynes_office, her records shows all calls with duration ( even refused with Congestion or Busy calls )
22:34.20SkramXfholmes: who are you using for termination?
22:34.25fholmesIn the iax.conf file I don't exactly understand what he potential context= line might be used for?  D
22:34.28kink0dlynes_office, I did, I have not any Answer()
22:34.29fholmesBinefone
22:34.36fholmesBinfone*
22:34.48dlynes_officefholmes: the context= line is so that when they send you a call, that call will go into the context you've defined there, in the dialplan
22:34.52kink0dlynes_office, also I have not this problem with other peers who sent me from similar Cisco
22:34.54SkramXhmm
22:35.11dlynes_officekink0: are they all going into the same incoming context?
22:35.28kink0dlynes_office, yes , the same context for all
22:35.42fholmesdlynes_office:  The call has to be passed to me first though right?  Before the context comes into play.  I do have a context setup for incoming calls in the extensions.conf file.
22:36.05dlynes_officekink0: then maybe their setup is broken...you might need to give them their own incoming context and set it up special somehow
22:36.17dlynes_officekink0: i.e. maybe their sip stack is broken
22:36.30fholmesSkramX:  I can call out.  I see the registration happening in the console.....
22:36.31*** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen)
22:36.35*** join/#asterisk crshman (n=chatzill@hacienda-heights-cuda2-68-71-5-62.lmdaca.adelphia.net)
22:36.39kink0dlynes_office, I had try, but for some reason contexts inside my h323.conf are not working and use the general context
22:37.02dlynes_officefholmes: what's your context= line say in your iax context in your iax.conf file?
22:37.12kink0dlynes_office, if I set a context for a peer in h323, is like ignored, that happens just with h323
22:37.13dlynes_officekink0: oh...this is for h323...forgot
22:37.24kink0yes, they used h323
22:37.34dlynes_officekink0: i have no clue on that one....h323 is barely even supported in asterisk
22:37.55fholmesdlynes_office:  Right now it says context=incoming-IAX
22:37.57crshmanhi all in my logs i have this: "Allocating new SIP dialog for xxxxxxxxxxxxxxxxxxxxxxxxxx@127.0.0.1 - REGISTER (No RTP)" i am using broadvoice and i can't seem to get it to authenticate, any ideas?
22:38.02kink0i know, I have a lot of troubles with h323, but as my peer is ussing, I am forced to support it
22:38.08Symmis there any need to learn RAGI, ruby on rails stuff.. or is there enough open source software available to set up a good voip system?
22:38.10dlynes_officekink0: but maybe they're expecting a specific q921 result code, and you're sending them the wrong one
22:38.39kink0dlynes_office, I sent actually ISDN Cause 17 ( Busy )
22:38.39dlynes_officekink0: you might end up having to hack your own changes into the h323 channel driver code
22:39.23dlynes_officefholmes: now do you have a context in your dialplan called [incoming-IAX]?
22:39.38kink0dlynes_office, is more easy for me, I can change causes in the gateway ( my asterisk is connected to a 2N Stargate )
22:40.15dlynes_officekink0: ah...cool
22:40.36fholmesdlynes_office:  Yes, I do.  Here is what I have in there:  exten => _19995552342,1,Dial(SIP/1234)
22:40.39dlynes_officekink0: so they go into the gateway first, and hten into asterisk?
22:41.03dlynes_officefholmes: set verbose 6 at the cli
22:41.04*** part/#asterisk Egonis (n=Egonis@207.245.14.10)
22:41.06kink0dlynes_office, no, in reverse order, they call to my asterisk and then call is route to the GSM ussing the 2N
22:41.19dlynes_officefholmes: then make a call into your voip line, and see if it drops into asterisk
22:41.44dlynes_officekink0: ah...the 2N is an h323<=>GSM gateway?
22:42.02fholmesdlynes_office:  No it is not going through.
22:42.17dlynes_officefholmes: check your /var/log/full log
22:42.19kink0dlynes_office, is asterisk -> PRI -> GSM , also I am ussing SIP, and IAX is supported, no just h323
22:42.36*** join/#asterisk Talmage (n=Talmage@mychoice-fw.mychoice.cc)
22:42.53dlynes_officekink0: but between the 2N Stargate and Asterisk it's only h323?
22:43.14kink0dlynes_office, nooo, there q931 signalling between them
22:43.29dlynes_officekink0: so it's a pri link then?
22:43.36kink0h323 arrives to the asterisk, and do not pass to the 2N
22:43.39TalmageI have the pap2-na adpaters...and I am trying to use sip notify to restart the adapter...but it requires that the request be authenicated...how do I do so?
22:43.48kink0yes, there a digium TE405 card there
22:44.12dlynes_officekink0: ah, and you've got one span connected to the 2N then, right?
22:44.23kink0dlynes_office, right.
22:44.28dlynes_officeok
22:45.00dlynes_officekink0: and so the result code from the pri (2N) will get passed back via q.931 to the h323 link?
22:45.04fholmesdlynes_office:  I don't see that log file anywhere.  Is there another log file I need to look for?
22:45.19kink0even code/decode is done by soft, I prefered Asterisk compared with gateway manufacturers voIP cards, so I use just a PRI interface from the gateway manufacturer.
22:45.38dlynes_officefholmes: edit your logger.conf file then, and make sure you've got a line:  full => errors,warnings,notice,verbose,debug,dtmf
22:45.49kink0dlynes_office, right !! the cause is passed, I have a translation table, but normally all causes are passed as is
22:45.51dlynes_officefholmes: then do a logger reload from the cli
22:46.06dlynes_officefholmes: then try your inbound call again
22:46.11dlynes_officefholmes: and then check the full log file
22:46.35dlynes_officekink0: find out from your ixc peer what causes they're expecting
22:46.36*** join/#asterisk SmittyHalibut (n=msmith@adsl-69-239-168-105.dsl.snlo01.pacbell.net)
22:46.45dlynes_officekink0: they might be different from the cisco peeps
22:47.35kink0dlynes_office, yes, that is what I have asked him today, but they really don't know, I will need to try with several until find what one will be ok, in the case that the problem is due to a not reconized isdn cause
22:47.47SmittyHalibutQuestion about the ZapTel drivers.  All documentation I've been able to find says to download the ZapTel drivers via cvs.digium.com, but that hostname doesn't appear to resolve for me.  Am I looking in the wrong place?
22:48.05dlynes_officekink0: they've got a complicated piece of hardware/software, and they don't know how to use it?
22:48.12CunningPikeSmittyHalibut: Yes, you are - CVS was replaced by SVN a while ago
22:48.13dlynes_officekink0: that begs the question, why did they buy it?
22:48.26kink0in the other hand, I must take care not cause missunderstanding for the actual well working cisco peers, since I am unable to set a context different for every one peer
22:48.42dlynes_officekink0: oh...why's that?
22:48.52SmittyHalibut/CunningPike:  just svn.digium.com?
22:48.58kink0dlynes_office, there question I dont ask :) I will never used what they are ussing
22:49.06dlynes_officekink0: why can't you set the ixc peer up on their own context?
22:49.24kink0dlynes_office, because if I set differents context for every one peer in h323.conf appears to be completelly ingnored by Asterisk
22:49.24TalmageI have the pap2-na adpaters...and I am trying to use sip notify to restart the adapter...but it requires that the request be authenticated, how do I authenicate the request?
22:49.43CunningPikeSmittyHalibut: If you want to run trunk (unstable), yes
22:49.50dlynes_officeTalmage: go into yoru pap2-na adapter, and tell it not to use authentication
22:49.53fholmesdlynes_office:  Nothing in there.  Man, it has to be something with my provider.
22:50.08CunningPikeSmittyHalibut: If you want stable, just go to ftp.digium.com and get the tarball
22:50.18Talmagedlynes_home Little worried about someone else sending my customer's sip notify messages.
22:50.27dlynes_officefholmes: well, if it was something on your end, i'm sure you'd see something about the call being sent into an invalid context
22:50.50dlynes_officeTalmage: ah...thought this was for a residential setup
22:50.51SmittyHalibutCunningPike:  That's exactly what I wanted to hear.  Thank you.   :)
22:51.04dlynes_officeTalmage: i'm not sure how to do it, personally
22:51.05CunningPikeSmittyHalibut: np
22:51.13fholmesdlynes_office:  Is there anyway it could have anything to do with the codecs involved?  GSM/alaw/ulaw etc?
22:51.13dlynes_officeTalmage: i always use http to do it
22:51.26TalmageWell, some of them are natt'd
22:51.27Talmageetc
22:51.33TalmageThey call and we find a configuration problem
22:51.49dlynes_officefholmes: if that was the case, you'd still see an error in your full log file
22:52.01dlynes_officeTalmage: yeah...all but one of my customers are natted
22:52.11dlynes_officeTalmage: i open up a port mapping on their router, though
22:52.30fholmesdlynes_office:  Thanks for your help.
22:52.44TalmageTakes 10 minutes to tell them how to power cycle their ata
22:52.55dlynes_officeunplug the power, plug it back in
22:52.59dlynes_officehow difficult can it be?
22:53.02*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
22:53.04*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
22:53.10Talmage...they start unplugging things and the lights go out
22:53.28dlynes_officeThat's gotta be an american company :)
22:54.20dlynes_officeeven my dumbest customers aren't that bad
22:55.24TalmageYeah...
22:55.38Talmageand a few of my customers are like 60 year old guys
22:55.43Talmagewho hit on our female csr...
22:55.46dlynes_officemost of mine are that old
22:56.00Talmage(just had to take a call from him)
22:56.08dlynes_officemost of our customers are mining companies
22:56.11*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
22:56.24*** join/#asterisk Trojan_Hors1 (n=root@220.226.22.191)
22:56.49Trojan_Hors1hullo guys
22:56.59dlynes_officeTrojan_Hors1: dood...don't irc as root
22:57.14dlynes_officeTrojan_Hors1: unless you feel like getting hax0red
22:57.47Trojan_Hors1thanks for advice..... i will switch soon
22:58.22dlynes_officeheh...on efnet, they won't even let you join any channels if you're irc'ing as root
22:58.43*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
22:58.44Talmagedlynes_office well thanks, will try asking my question tommorrow
22:59.17Trojan_Hors1i need to know the largest user base a distributed asterisk servers hav been in production use
22:59.20dlynes_officeTalmage: well, i'd read up  on the sip rfc
22:59.29dlynes_officeTalmage: it might help you figure out how to do it
22:59.47russellbTrojan_Hors1: i know of installations with 10's of thousands of users ...
22:59.56SmittyHalibutCunningPike:  You rule.  I've got them, compiled them, and am continuing my day.  Thanks again.  :)
23:00.17CunningPikeSmittyHalibut: Don't thank me - thank Mark Spencer ;)
23:00.32CunningPikeSmittyHalibut: Actually, thank russellb and the crew
23:00.36dlynes_officeTalmage: are you in the US?
23:00.37Trojan_Hors1russellb, wats it ?
23:01.08*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.18.81.Dial1.SanJose1.Level3.net)
23:01.26dlynes_officeor anyone else in the US that wouldn't mind calling a phone number for me?
23:02.11Mw3hm, does linksys pap2 support t.38?
23:02.17dlynes_officeMw3: yes
23:02.42*** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon)
23:02.43dlynes_officeMw3: no idea how to set it up though
23:02.48Mw3any special configuration needed
23:02.49Mw3?
23:02.51Mw3oh, ok :)
23:03.06*** join/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com)
23:03.29dlynes_officeMw3: try checking the sipura and/or linksys users' group on voxilla
23:03.30kink0see you later !! good night guys
23:03.45Mw3dlynes_office: i'll do that. thank you
23:03.46dlynes_officeseeya kinky
23:04.25Trojan_Hors1okey for a userbase of 3000 wats d optimal number of asterisk servers [assuming 30 PSTN lines]
23:04.28*** part/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com)
23:04.41mitcheloc1 server per user
23:04.46dlynes_officeheh
23:04.52Bert-Hmm
23:04.53Trojan_Hors1errr
23:05.04mitcheloc(it's a joke)
23:05.11mitcheloc30 lines is enough for 3000 users?
23:05.34dlynes_officemitcheloc: why not?  most of the time they're not going to be calling out, only calling each other
23:05.35Trojan_Hors1ya..... d main business is on PC to PC
23:05.36Trojan_Hors1softphones
23:05.38Bert-I have a VoIP account which use MGCP protocol. Can asterisk handle it like a simple PSTN line ?
23:06.02dlynes_officeBert-: asterisk has an mgcp channel driver
23:06.09Bert-ok
23:06.25mitchelocTrojan_Hors1: you could do that with ser then, and only like 2-3 boxes i bet
23:06.26dlynes_officeBert-: so you can use the Dial() application for that channel just like any other cahnnel
23:06.39Bert-because I'm unable to find a linux softphone which use MGCP :
23:06.40Bert-:(
23:06.59dlynes_officeBert-: i think you're going to be hard pressed to find an mgcp softphone for any operating system
23:07.10Bert-I4ve found one for win
23:07.20Bert-but I use Linux ...
23:07.33dlynes_officeBert-: qemu?
23:07.47Bert-hmm never tested
23:07.49Bert-let me try
23:08.07dlynes_officeBert-: http://sf.net/projects/qemu/
23:08.26Bert-thx
23:08.43Trojan_Hors1mitcheloc, okey 2-3 boxes arnt a big deal...... but are ye sure i expect an avg of 30-40 concurrent cnxns and a high of 150-200
23:09.01dlynes_officeBert-: erm...hold on...that's not the url
23:09.09Bert-I see ... :)
23:09.26Talmagedlynes_office yes, sorry on tech support call.
23:09.37*** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net)
23:09.53SmittyHalibutCunningPike:  I've updated the Wiki with the new process, FTP instead of CVS.  Thanks again.  And, thanks to Russellb and Mark Spencer!  :)
23:10.04pdavidhi all!  could anyone suggest a good voip provider for a small business in southern US
23:10.05pdavid?
23:10.14dlynes_officeTalmage: yeah...was just looking for someone from the us that'd be able to call one of my dids for me...i've got a line quality issue that only appears on calls from us users
23:10.14*** part/#asterisk SmittyHalibut (n=msmith@adsl-69-239-168-105.dsl.snlo01.pacbell.net)
23:10.16mitchelocTrojan_Hors1: i'm *not* speaking from experience, but what i understand is nothing goes through the ser machine it'self, so it proxies everything, should be pretty easy
23:10.29mitchelocftp = new process???
23:10.29dlynes_officeTalmage: i'm just curious how widespread the problem is
23:10.45Talmageis it intl?
23:10.57dlynes_officeTalmage: not really...Canada
23:11.13mitchelocTrojan_Hors1: i don't use it, suggest you check #ser
23:11.25TalmageI am already getting yelled at for making too many intl calls
23:11.30dlynes_officelol
23:11.31Talmagethen again...I could just empty the cdr db
23:11.31Trojan_Hors1okey thanks for the help
23:11.40dlynes_officehahah
23:11.44dlynes_officedon't worry about it then
23:11.51pdavidi have been considering voicepulse, but wanted some opinions on it if anyone had any...
23:12.25generalhanpdavid: i have used VP in the past ... i dropped them hardcore and am paying more for a PRI T-1 and its sooooo worth the price
23:12.45dlynes_officeBert-: http://fabrice.bellard.free.fr/qemu/
23:12.46generalhanVP had been dropping about 80% of my packets to them and they refused to do any testing on there end to help me figure out why.
23:12.52pdavidgeneralhan: what type of issues caused you to drop them?
23:12.56pdavidahh
23:13.02generalhan80% is A LOT !
23:13.12*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
23:13.14pdavidthat is a lot
23:13.43generalhani understand that CAN happen but i may have stayed with them if they took more time to help me figure out where the routing issue was taking place
23:13.50dlynes_officegeneralhan: i hope you were getting 0.00000001c/min for your calls?
23:14.09generalhanyea right ... iwas credited ANYTHING for my loss
23:14.11nahireangeneral: how can you be sure VP is dropping the packets?  how do you know which router is actually losing them
23:14.19generalhans/iwas/i wasnt
23:14.43pdavidyes, did you ever resolve the packet loss?
23:15.13nahireangeneral: any router along the way could be tossing those packets like a salad.
23:15.17rpmhttp://pastebin.ca/62749 - does this logic make sense to anyone? i want to be able to transfer a caller to an extension and place them on hold via an external source. and when they are taken off hold it call me back
23:15.19generalhannahirean: yea ... i had been doing pingplotter tests for weeks straight .. i knew which server on their end was starting the packet loss but they kept calling me a liar ! lol
23:16.07generalhanand let me clear up something real quick .. i KNOW that its not all their fault .. but the reason that i was soo dissatisied was because of their apathy about my entire situation
23:16.29nahireangeneral: i have an TF number with VP and i kid you not, i capped out my bandwidth using a binary newsreader, and the quality was still awesome.. seriously, ive not had any issues with them
23:16.43pdavidwell, does anyone have any other thoughts/recommendations on a provider?
23:16.44generalhannahirean
23:16.48Talmageteliax
23:17.05pdavidits like a primordial voip ooze sifting through all the providers
23:17.35pdavidi was just looking at teliax
23:17.59generalhannahirean: yea ... i know lots of people that have used them and not had any issures ... but when i call a provider of mine and tell them that i cant use their service because of XYZ i expect them to show some concern and/or sympathy about my issues ... they were just unwilling to except any responsability
23:18.01pdavidyou know, its a pet peeve of mine when someone says "Unlimited***", then the footnote reads: *** Softcap of 2500 minutes
23:18.27pdavid2500!=inf
23:18.36dlynes_officepdavid: vonage swears up and down that they've got unlimited calls
23:18.47pdavidit's just shady practice
23:18.56pdavidwhy not just say: 2500 minutes
23:18.57dlynes_officepdavid: but apparently if you exceed what they deem to be within normal limits, you get flagged
23:19.09*** join/#asterisk JoseBravo (n=jdbravo@200.24.110.91)
23:19.13pdavidyeah, and/or backcharged like some people complaining about broadvoice.com
23:19.26pdavidat least be up front about what you are offering
23:19.36generalhanpdavid: what i would do is get some form of ping plotter to trace 24 hours of data to their server ... if its good then i would go with VP .. i had their wholesale minutes and i had 30k minutes and paid dirt for it
23:20.02pdavidgeneralhan: thanks, i think i will try that, and let it run all night/day tomorrow
23:20.09pdavidsee how it pans out
23:20.12generalhanthey ARE really really cheap with their wholesale minutes ... if i could have kept a constant connection to them i prolly wouldnt have left
23:20.31pdavidi don't mind paying a little extra for the service
23:20.36*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
23:20.47pdavidi was going to use their connect plan, to use my *
23:21.05pdavidoddly, their calls to canada were more expensive than calling australia...
23:21.11generalhanpdavid: 66.234.228.134 is the server they were using when i had it ... you can test on that one .. otherwise just call them and ask ... if you tell them what you need it for they will be happy to give you the ip
23:21.21*** join/#asterisk Drew___ (n=foo@zux221-156-100.adsl.green.ch)
23:21.29dlynes_officepdavid: it's more expensive for me to call the US, than to call China :)
23:21.37dlynes_officepdavid: and that's from Canada
23:21.43pdavidfreakin yikes
23:22.11dlynes_officepdavid: my rates to Canada though are less than half of China
23:22.32mitchelocit's free for me to call anywhere in the US with skype!
23:22.36pdaviddlyines_office: who are you with?
23:22.43dlynes_officepdavid: I buy wholesale
23:22.49pdavidahh
23:23.00pdavidso any other providers i should consider?
23:23.04dlynes_officepdavid: it wouldn't matter who i buy from because they don't sell retail
23:23.16*** join/#asterisk ToTo (n=ToTo@host212-109.pool8258.interbusiness.it)
23:23.28pdavidi was considering BV, but have read many bad reviews of late that have sort of turned me sour
23:24.02Juggiewhats the best place for wholesale iax/sip minutes?
23:24.09*** join/#asterisk P-NuT (n=P-NuT@fw.office.unitedip.net.au)
23:24.19dlynes_officepdavid: try www.calltermination.com
23:24.36dlynes_officeJuggie: you might want to qualify that
23:24.58dlynes_officeJuggie: what do you consider to be 'best'?  cheapest?  best call quality?  or a nice easy medium?
23:25.04pdaviddlynes_office: thanks, checking it out now
23:26.35*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
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23:26.53*** part/#asterisk ChewyNet9 (n=plugci@adsl-61-111-148.sdf.bellsouth.net)
23:27.16Juggiedlynes_home, best of both worlds
23:27.47dlynes_officeJuggie: well, i've had pretty good luck with Five9sNetwork
23:28.09dlynes_officeJuggie: they're relatively stable, call quality is good, and they're cheap
23:28.33dlynes_officeJuggie: the only issue i've had with them, is sometimes I can't make calls to certain destinations
23:28.43dlynes_officeJuggie: usually for my Indian white routes
23:29.21*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
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23:33.35Drew___how would i get asterisk to simply connect a sip phone with a zap channel - i.e. without dialing anything?
23:34.07*** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net)
23:34.08*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
23:34.14JoseBravoHow I can check if my asterisk is connected to astbill Mysql db?
23:34.17*** join/#asterisk mogorman (i=ejabberd@68.62.237.103)
23:34.30Zodiacalanyone setup a cisco 7914 before? with sccp?
23:34.48Zodiacaldo i need to specific two devices in my sccp.conf?
23:35.03Zodiacalone for 7960 and one for 7914?
23:35.13Zodiacaland a line for each?
23:35.55Zodiacalto specific = specificly
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23:46.55UmaroHi.. anyone here from digium that I can talk to about my g729 license order?
23:47.02mogormanwhats up
23:48.14Umaromogorman, do you work for digium?
23:48.23mogormanthat i do
23:54.31brimstoneUmaro, what's up?
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