IRC log for #asterisk on 20070802

00:01.33Mercestessomethign like that, yea.
00:01.44MercestesFlash hook is an ATA specific feature, your ATA would have to support it.
00:01.58Mercestes*8 for transfer I believe is already canned under features.conf
00:02.04Mercestesjust hit *8 then the extension
00:03.02x86ATA in this case is a rhino CB24-FXS
00:03.26x86I run a T1 from a sangoma card to a rhino channel bank
00:05.33fujinanyone know what this is ?
00:05.34fujinAug  2 12:05:14 WARNING[20465]: app_queue.c:2130 calc_metric: Can't calculate metri
00:05.37fujinc for unknown strategy 7
00:05.55x86sounds like your queue strategy is invalid
00:05.59fujinhttp://rafb.net/p/bFmIlP55.html
00:06.22fujinwas leastcalls not in Asterisk?
00:06.33fujinmaybe it doesn't work cause I only have the one agent logged in
00:07.03x86shouldn't matter
00:08.13*** join/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181)
00:08.36fujinlol, I'm a tard.
00:08.39fujinnevermind all of that ^^
00:10.56riddleboxcan you have asterisk, turn on a message light, for a group of extensions when one extension gets a vm?
00:11.22x86riddlebox: using a SIP phone, yes
00:11.43x86riddlebox: most SIP phones (and even softphones) allow you to specify what number to subscribe to for MWI
00:11.52riddleboxx86, yeah they are all sip phones
00:12.11x86depends on which sip phone
00:12.17x86look at your manual to find out
00:12.38riddleboxok I will look thanks
00:15.13x86no prob
00:18.53*** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home)
00:18.53*** mode/#asterisk [+o Corydon76-work] by ChanServ
00:18.59generalhananyone know how i would go about dialing a phone while doing a playback() ... like a ringback tone ? is there some documentation on this somewhere ?
00:21.10generalhanlol, or that was way more simple than i had thought !
00:21.50*** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home)
00:21.50*** mode/#asterisk [+o Corydon76-work] by ChanServ
00:25.58*** join/#asterisk Innatech (n=it@netblock-68-183-140-137.dslextreme.com)
00:28.37*** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell)
00:28.37*** mode/#asterisk [+o Qwell_] by ChanServ
00:29.16*** join/#asterisk chai_sangeen (n=aaljishi@82-44-68-150.cable.ubr02.nmal.blueyonder.co.uk)
00:29.34*** join/#asterisk sacitec (n=tobi@189.129.223.250)
00:29.58sacitechello
00:30.25sacitecijust a question in IVR
00:30.51chai_sangeenhello im having the exact same problem as this thread http://www.trixbox.org/forums/trixbox-forums/open-discussion/help-no-sound-outside-lan-nat-problem i already did everything they advised but my client keeps registering on port 64042. can anyone please help?
00:31.48sacitechow could i manage an option like "if you know the extension dial now" in IVR ? i need to use a variable to store the dialed extension an then pass it on a new dial action ?
00:32.01x86WaitExten
00:32.04x86or Read
00:32.18InnatechI'd appreciate it if anyone with a knack for reading SIP conversations could have a look at this and give me an idea as to why these inbound calls don't make to the ALL/ANY inbound route.  http://www.pastebin.ca/642477
00:32.56saciteccool, thanks
00:33.26x86most likely you want WaitExten
00:40.59*** part/#asterisk andresmujica (n=andresmu@190.24.227.202)
00:42.35Innatechideas? anything at all? I keep seeing 481 "Call/Transaction Does Not Exist" from the CLI--and this is the sip debug output: http://www.pastebin.ca/642477
00:44.57*** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net)
00:45.10*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
00:46.55*** join/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-860e955f7f98d827)
00:53.41*** join/#asterisk Avero (n=no@24.96.142.67)
00:53.56*** join/#asterisk sakic (n=sakic@cpe-071-075-118-121.carolina.res.rr.com)
00:55.28*** join/#asterisk guillote_GNU (n=guillote@host157.200-117-34.telecom.net.ar)
00:59.39sakicwhoa dead in here at night
01:00.07*** part/#asterisk Innatech (n=it@netblock-68-183-140-137.dslextreme.com)
01:00.25russellbheh
01:00.42*** join/#asterisk anthm (n=anthm@m810f36d0.tmodns.net)
01:00.42*** mode/#asterisk [+o anthm] by ChanServ
01:00.47russellbi used to be on here a lot at night, but not after i started doing it as my day job
01:01.19_DAWi can imagine
01:04.01_DAWCan more than one TC400B be used in the same machine?
01:04.14russellbyes
01:04.58russellbi think there are people using 4 per machine
01:05.02russellbbut don't quote me on ti ...
01:06.05_DAWIm very excited about trying a few.  Going to save me a ton in cpu and licensing.
01:06.13russellbawesome
01:06.17*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
01:06.23russellbusing it for G.729 ?
01:07.29_DAWyes.  we do lots of offshore voice over a vsat platform and bandwidth is a premium.
01:07.39sakicman it is impossible to find people to install this for me
01:07.44sakichaving to look to india
01:07.53MercestesInstall what for you and where?
01:08.28*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
01:08.47russellbi will run "make install" for $200 cheap
01:08.59x86i'll do it for $175
01:09.13russellbowned.
01:09.15MercestesI'll do it for $175 too...but nakid
01:09.16shido6are we bidding?
01:09.20x86indeed
01:09.24sakiclol
01:09.24shido6on what? :)
01:09.26x86i'll do $150
01:09.34Mercestesa satin computer chair, of course.
01:09.34sakicdo I hear $300?
01:09.35_DAW125 + fee home brew
01:09.36x86on sakic's job ;)
01:09.40x86sakic: sold!
01:09.43_DAWerr free
01:09.51shido6whats the job?
01:09.52x86_DAW: SHUT UP
01:10.02_DAW?
01:10.06Mercestesmale-homo-prostitute, of course.
01:10.12wunderkin$5. sucky sucky
01:10.18Mercesteslove you long time.
01:10.20shido6happy ending?
01:10.26_DAWx86: why?
01:10.26Mercestesonly in the end.
01:10.32shido6no mess?
01:10.39x86_DAW: we're bidding on doing a job, dont come in saying free!
01:10.57_DAWI just said free beer if you take my bid of $125
01:11.20Aces1Uponly if you can program the voicemail to advertise your services.
01:12.07*** join/#asterisk pejo_ (n=AB@89.160.93.219)
01:19.50*** join/#asterisk Innatech (n=it@netblock-68-183-140-137.dslextreme.com)
01:23.36*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
01:24.25*** join/#asterisk ManxPower (n=manxpowe@71-8-56-17.dhcp.leds.al.charter.com)
01:24.54MercestesOh, I guess by "find people to install this" he didn't mean to implicate pay
01:25.06MercestesI swear....IRC is like the world's virtual street-corner.
01:25.20Mercestes"Give me free shit."
01:25.53MercestesAnyways.  :D  Goodnight.
01:37.04*** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com)
01:39.03sweeperso true
01:39.33sweeperwhere's a good place to pick up consulting leads? I'm already on linkedin.....
01:40.14ManxPowerI let potential customers beg to be real customers.  That weeds out the wankers.
01:40.32sweeper:P
01:40.41ManxPowerOn the other hand customers stay with me for years and years and years
01:41.01sweepergetting them in the first place is my problem~
01:41.39ManxPowerMost of my customers are from when I was their ISP, or referrals from those customers
01:50.19*** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
01:50.24*** mode/#asterisk [+o mog] by ChanServ
01:50.56*** part/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181)
01:54.50*** join/#asterisk andethemint (n=robert@vcchgate.vcch01.springfield.tn.us.vcch.net)
01:56.16*** join/#asterisk Injen (n=sike@unaffiliated/injen)
01:56.17*** join/#asterisk dijungal (n=kdaniel@64.86.52.254)
01:56.39dijungalhello... how do i make asterisk store it's recorded calls elsewhere
01:56.58dijungalinstead of /var/spool/asterisk/monitor/
01:57.22*** part/#asterisk Injen (n=sike@unaffiliated/injen)
01:59.23ManxPowerdijungal: Does "show application monitor" have any helpful information?
02:00.38*** join/#asterisk tengulre (n=tengulre@222.90.66.10)
02:01.06*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
02:01.10dijungalnot anything that can help me with the above question
02:04.55*** join/#asterisk kikoy (n=kikoy@124.106.83.9)
02:05.35kikoyhi guys, how do i make transcoding work? i have set in my iax.conf allow=all
02:06.04kikoya certain extension uses ilbc and the other is alaw... but they seem not to be able to talk to each other.
02:06.20ManxPowerHow about Monitor(/my/custom/path/fnamebase)
02:06.52kikoy?
02:06.56ManxPowerkikoy: transcoding happens automatically.  However, asterisk CANNOT transcode between G729 or G723.1 and anything else.
02:07.09ManxPower"show translations" will show you what can be translated to what else.
02:07.17ManxPoweryou NEVER want allow=all
02:07.20kikoyoh i see
02:07.24ManxPoweryou want disallow=all and allow=the,codecs,you,want
02:07.46ManxPowerIt is silly to allow both alaw and ulaw, but generally harmless.
02:12.41kikoy:D
02:12.53kikoyit appears that ilbc can't be transcoded.
02:16.24ManxPowerthen the ilibc dev libraries were not installed on your system when you installed asterisk
02:18.43kikoyManxPower is that so? but i can do echo test with ilbc? i can hear my self.
02:26.11*** join/#asterisk weasel00 (n=snowball@pencomsf.com)
02:26.29*** join/#asterisk fnordus (n=dnall@24.84.160.227)
02:26.57pigpenManxPower, would you have time for a quick questions?
02:27.04pigpens/questions/question
02:27.12weasel00where can i information on a asterisk compatible ppc client? or does such a beast exist?
02:27.16ManxPowerpigpen: depends on the question
02:27.25pigpenManager API.
02:27.38ManxPowerkikoy: What protocol are you using?  IAX or SIP?
02:27.43ManxPowerpigpen: I never use Manager
02:28.01ManxPowerweasel00: define "asterisk compatible"
02:28.04*** join/#asterisk mrichmanM (n=richmanm@70.89.184.1)
02:28.18pigpenBummer.  I have it working..via a perl script, but after I run it once, I cannot run it, or any other, again...until I reboot.
02:28.27pigpenThanks however.
02:28.47kikoyManxPower: i use iax.
02:28.54flenderspigpen: connection times out?
02:28.57weasel00ManxPower: looking for client software that would communicate with asterisk that would be run on a ppc
02:29.04ManxPowerkikoy: do the echo test and while you are doing the echo test do an "iax2 show channels" in the Asterisk CLI
02:29.11ManxPowerweasel00: using what protocol?
02:29.21pigpenflenders, no, it runs fine...then connects, seeming like it does nothing.
02:29.33kikoyManxPower: it appeared that ilbc wasn't installed, so i installed ilbc first then rebuilt asterisk.
02:29.35pigpenI have had the same effect on 4 asterisk boxes.
02:29.57*** join/#asterisk koppernet (i=kopperne@116.50.137.250)
02:30.03ManxPowerlook for the codec listed in the outout of iax2 show channels
02:30.24flenderspigpen: I had to keep 'pinging', like refreshing a page, so it would keep working
02:30.34ManxPowerkikoy: after you rebuilt asterisk does "show translations" list ilibc?
02:30.50weasel00ManxPower: any... we are trying to find something to use on our ppc phones for voip communication between a dozen of us while traveling to various clients.. cell reception is not always good but they all have wifi so trying to angle that to keep in communication easier
02:31.09ManxPowerweasel00: so you are looking for a Softphone for PPC.
02:31.34pigpenflenders, http://pastebin.ca/642549
02:31.37weasel00ManxPower: sorry i cant even ask a decent question today... hehe.. yes
02:31.49ManxPowerThat is correct.
02:32.10pigpenI run it as perl page.pl 200   (where 200 is the extension)
02:32.12ManxPowerI don't know of any PPC softphones, but I believe there are some for Mac OSX running on a PPC
02:32.23SplasPoodxten.com
02:32.27SplasPoodx-lite was PPC
02:32.33ManxPoweryou would want to do google searches for that, as we really are not a softphone support channel
02:32.41pigpenManxPower, yeah..there are several.
02:32.47ManxPowerAs far as we are concerned if it talks SIP or IAX then it works with Asterisk
02:33.07pigpenbut xten.com and idefisk I feel are the best for mac's
02:33.10SplasPoodbut will asterisk talk to IT? :)
02:33.30SplasPoodxten is one of the best clients I've found on ANY platform
02:33.34SplasPoodI wish there were more...
02:33.39SplasPooderm, eyebeam/x-lite rather
02:33.48pigpenSplasPood, I agree.
02:34.26*** join/#asterisk jordanb (n=jordanb@adsl-68-20-20-59.dsl.chcgil.ameritech.net)
02:34.30*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
02:34.31pigpenflenders, so I run the perl script, it works great...but never to be run again unless I reboot.  I can even restart asterisk....so maybe it is a perl screw up.
02:34.49pigpenDid I mention I hate perl?
02:34.54jordanbIs there some 555- number that runs echo or does callback?
02:35.57SplasPoodpigpen: thats.... interesting...  maybe some tmp file not gettin removed that's gettin killed on boot?
02:36.23pigpenSplasPood, yeah..I find it kinda interesting.
02:36.50ManxPowerpigpen: what happens if you wait 5 mins, for the closed socket to finish waiting for it's final ack.
02:37.04pigpenI have waited 20 days....no dice.
02:37.18pigpenin fact, the last time I screwed with it was over a month ago.
02:37.24pigpenI rebooted the box today...and now it works.
02:37.30pigpenspooky eh?
02:38.09ManxPowerthat would indicate that is is not a socket timeout issue
02:38.09*** part/#asterisk dijungal (n=kdaniel@64.86.52.254)
02:38.16SplasPoodpigpen: what happens when you try and run it again?
02:38.24pigpennothing in the logs, nothing in dmesg, asterisk cli shows the manager is logging on and off....
02:38.39SplasPoodI'd add some prints into the perl script
02:39.14pigpenSplasPood, well, the only thing I see is in the cli, manager logging on and off....and "Verbose "Doing 200" 0" in the shell where I am running it.
02:40.23SplasPoodright, so I'd throw some print statements into the code to see what it's sending, when
02:40.27SplasPoodnot knowing anything else bout the script thats all I can offer :-/
02:40.27SplasPoodif its not huge you can pastebin it and I'll take a look
02:40.27SplasPoodmaybe something will jump out
02:40.28pigpenin fact, I just checked out the /tmp/input.log   & /tmp/output.log:
02:40.28pigpenhttp://pastebin.ca/642554
02:43.57*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id)
02:45.27pigpenNow...however, if I enter the manager api, and past the same info that is in the output.log...it works fine.
02:46.39pigpenSo maybe I just need to abandon perl.
02:48.47*** join/#asterisk HockeyInJune (n=HockeyIn@pool-141-155-136-189.ny5030.east.verizon.net)
02:50.02ManxPowerpigpen: I believe the asterisk-perl AGI pacakage also has Manager stuff in it
02:54.18pigpenhmm..I may just see if php works.
02:58.29*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
03:02.24*** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net)
03:04.15*** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net)
03:04.23MrTelephonehey guys
03:04.32pigpenOh shit.  It's MrTelephone
03:04.41pigpenJust kidding.
03:04.41pigpen:)
03:04.50MrTelephonewhats happening penpig
03:05.01MrTelephoneopenswan?
03:05.10pigpen:)
03:05.29pigpenI did find out I can do AES256 to china!
03:05.29MrTelephoneis there a good vpn for linux that does ipsec and can use win xp to dial up?
03:05.39tengulrechina?
03:05.48MrTelephonewhat do you need to encrypt that badly?
03:05.58MrTelephonenuclear launch codes?
03:06.09pigpenMrTelephone, yeah...I forget the name...but it is about $130/seat.
03:06.21pigpenneed to encrypt?  I can't say.  It is a secret.
03:06.45pigpenWell, we look at it this way.  We can, so why not.
03:07.08pigpenBut yeah..from the US to China.
03:07.08Juggiethere are 2-3 vpn servers for linux
03:07.19Juggiewell, open source ones
03:07.19jordanbOpenVPN works great but it doesn't do IPSec.
03:07.30jordanbThere's a client for WinXP.
03:07.43MrTelephonewow big bridge collapse
03:07.55MrTelephoneim using pptpd
03:08.07pigpenOpenSwan...crap.
03:08.07pigpenUse StrongSwan.
03:08.19jordanbOpenVPN uses tun/tap interfaces, so it should work over any carrier.
03:08.31MrTelephoneone of them does ipsec doesn't it?
03:08.31pigpenWe manage over 13,000 vpn's on linux for customers.
03:08.52pigpenYes, the comma was at the correct location.
03:08.53pigpenMost to one box...around 4000.
03:09.05MrTelephonevpn clusters?
03:09.10MrTelephonewhy not use cisco if your doing that much?
03:09.19pigpenLinux does it better.
03:09.20J4k3"why not use cisco"
03:09.23J4k3...
03:09.23MrTelephonei setup pptp for guys to video conference without worrying about nat
03:09.26J4k3haha
03:09.40J4k3I can give you a few thousand reasons a year not to do business with cisco.
03:09.44J4k3;)
03:09.49pigpenJ4k3, here here.
03:10.00MrTelephonemostly because people can't afford it i guess
03:10.04pigpenWe had the Pix535's puking at 100 vpns.
03:10.05J4k3heh
03:10.25J4k3some people don't have to pay top dollar for second rate.
03:10.26jordanbWhat kind of hardware do you use for 4k VPNs?
03:10.29pigpenStick linux on a cheap desktop, we pushed over 500 when the customer bought a real server.
03:10.32J4k3some people obviously have stock in cisco, I don't.
03:10.43J4k3I've only been one of their customers...
03:10.53MrTelephonecisco has nice end user vpn software
03:10.59pigpenFor the big ones, we use 64bit dell servers, 4gb ram so so.
03:11.01J4k3"end user"...  uhm
03:11.12J4k3yeah lemmie cram a few thousand vpn connections on an "end user" piece of gear, thx.
03:11.25MrTelephonei mean the software to connect to the vpn
03:11.28jordanbpigpen, Those poweredge 1900s?
03:11.41pigpenNa...2950's / 6850's
03:11.47MrTelephoneyou guys dis cisco and all you buy is dells
03:11.47jordanbOne of my clients just got a few of them. First Dells I've seen that look like real computers.
03:11.48MrTelephone:-
03:12.00pigpenat the remotes, we use mostly Soekris boxes.
03:12.09J4k3people seem to leave them here
03:12.47MrTelephoneit was fun to build a dual xeon
03:13.04MrTelephonei recommend not using anything with riser cards though
03:13.09J4k3now you can build a "dual" box.... and never go past buying cheap consumer gear ;)
03:13.39J4k3I like spares.
03:13.58pigpenYeah..spares are good.
03:14.09J4k3sure, that spare might play a lot of counterstrike and mp3s while its sitting here ;)
03:14.16J4k3(and have a different video card...)
03:14.23J4k3but...  its a spare indeed!
03:14.25J4k3;)
03:15.31MrTelephonefor some reason my 900mhz pentium III's seem to holdup though
03:15.33MrTelephone3 years non stop and only went through a few harddrives
03:15.48MrTelephoneback in the day when that crap was reliable
03:15.56J4k3well
03:16.03J4k3crap has always been reliable, the trick is buying the right crap.
03:16.07MrTelephoneyeah
03:16.11J4k3and that crap changes from day to day it seems
03:16.15MrTelephoneever come across any asus culs mainboard?
03:16.22MrTelephoneworst mainboard ever made
03:16.23MrTelephonehaha
03:16.29J4k3ie - if you're buying a 1985 model car, toyota is a good buy... if you're buying a 2005 car, toyota isn't the way to fly.
03:16.30MrTelephoneCUS L2 or something
03:16.43J4k3Asus tends to make some cra
03:16.44J4k3er crap
03:16.54J4k3they either do great or totally suck
03:17.05MrTelephonewhat do you recommend for xeon boards
03:17.09MrTelephoneand don't say tyan
03:17.11MrTelephone:-/
03:17.15MrTelephonei should have went with tyan
03:17.40JTsupermicro
03:17.56*** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ut.comcast.net)
03:20.11MrTelephoneso this ipsec stuff.. can you have your entire network run on that or is it just for vpns
03:20.35MrTelephonesupermicro.. I see they have a nice seleciton of motherboards
03:20.40pigpenMrTelephone, ipsec "is" vpn's
03:21.04wothinnYou can quite hapily run ipsec between too peers on the same LAN.
03:21.16wothinnThere just aren't many good uses for it and it's quite resource-intensive.
03:21.28pigpenResouce-Intensive?
03:21.39*** join/#asterisk miroesq (n=sam@62.139.252.66)
03:21.43wothinnIt eats CPU doing all that crypto.
03:21.59wothinnUnless, of course, you have a VIA CPU with padlock or a hifn card.
03:22.00pigpenUsing a Soekris 4801, 233MHz Proc, 128 MB Ram, running from flash, we have 80 vpn's.
03:22.12pigpenOh...1 GB CF Flash.
03:22.15JTdoing what?
03:22.16wothinnCongratulations.  What's your packet rate?
03:22.21JTeactly
03:22.23JTpps rate
03:22.24JTbit rate
03:22.26miroesqneed to ask a couple of questions regarding asterisk/tyrixbox if someone can help me out
03:22.30JTencryption algorithm
03:22.43JTbecause i thought those soakris have no floating point
03:23.16pigpenAES 256...it is some sort of inventory management server for a Retail Store.
03:23.30pigpenCustom build, using Gentoo.
03:23.32JTso very low data rate
03:23.36MrTelephonewhats the chances of asterisk being programmed with the secure rtp?
03:23.49*** join/#asterisk mtaht4 (n=m@m815f36d0.tmodns.net)
03:24.25pigpenJT, yeah..sorry, I don't have the data rate stats....but I do know this customer is moving to a DS3...but who knows.
03:24.44pigpenBut i agree, I bet it is on the low side.
03:24.53miroesqcan anyone help with two simple questions?  unfortunately, can't find the answer on any of the forums
03:25.01MrTelephonesucks man, 1st of the month i have to do the billing
03:25.04JTi think the soekris can't push more than 50Mbit/s through ethernet, let alone with any encryption
03:25.23pigpenJT, ah..but that is where the new 5501 comes in.
03:25.39SwKif you have a direct to L3 deal and want to sell some minutes msg me
03:25.59MrTelephonedirect to L3?
03:26.06wothinnIf you're doing any real crypto, Soekris isn't the way to go anymore.  the VIA C7 chip has padlock that blows away even the fastest of the Intel or AMD chips at AES.
03:26.17wothinnAnd it's available on tiny, low-power boards.
03:26.35JTbut it's via
03:26.39pigpencool...link?
03:26.39wothinnSoekris+hifn can't touch it for speed, isn't much cheaper, and draws about the same power.
03:27.03pigpenJT, ha!
03:27.06wothinnJFGI: http://www.google.ca/search?q=via+eden+n
03:27.22*** join/#asterisk smultron (n=lukas@cpe-72-179-47-78.austin.res.rr.com)
03:27.23wothinnJT: VIA used to have a bad name.  They've come a long way.
03:27.33miroesqok, just in the off chance that someone can help out with this, i have a digium tdm400 card installed on my machine and would like to completely remove ztdummy so that it does not conflict with my timing and so i don't get all of the errors during start/stop; that's all.  can someone tell me how. thanks
03:27.44JTthey still have a bad name :)
03:27.51MrTelephonermmod ztdummy
03:27.52wothinnOnly if you haven't used a C7.
03:29.39miroesqthanks mrtelephone.  i tried that, but had ran it from the root directory, but it did not work.  should i have ran it from elsewhere? i also tried doing a yum uninstall ztdummy, but it told me that it could not find the file
03:29.58smultroni'm planning on buying some digium cards and then install an asterisk system in my office. however, i've never done this before. are there any phone support people or just clear tutorials that could guide me through it?
03:30.02MrTelephoneare you sure its loaded?
03:30.10MrTelephonelsmod | grep ztdummy
03:30.27smultronor should I just come back here when i'm installing and ask for help via IRC?
03:31.11MrTelephonesmultron, unless you plan on spending a lot of time you should go with a meridian pbx or something
03:31.13kiscokidI gonna be using a Sangoma A101d card for a PRI.  Now the PRI vendor is asking me what is the "ISDN Standard" and that their default is NI2.  Does anybody know if I should ask for NI2 or something else?
03:31.29J4k3MrTelephone: you're about as helpful as a pile of assholes.
03:31.32*** join/#asterisk osiris (n=osiris@c-71-205-35-230.hsd1.mi.comcast.net)
03:31.36smultronMrTelephone: where would i find that? and what is it?
03:31.39MrTelephonehaha
03:31.41JTkiscokid: asterisk can deal with whatever
03:31.47JTkiscokid: NI2 will be fine
03:31.49MrTelephoneits hard getting analog phones to work without echo
03:31.49J4k3smultron: a pile of money.
03:31.56kiscokidsmultron: buy the Asterisk book from Oreilly
03:31.56JTjust set it to national isdn 2
03:32.01JTMrTelephone: only with rubbish hardware
03:32.10MrTelephonetdm2400 is rubbish then
03:32.11kiscokidJT thanks
03:32.11J4k3MrTelephone: nublet.
03:32.18miroesqit shows that it was loaded ok, but i get errors on the modules diretcly following it. also when i go into freepbx and look at asterisk/config, i see that i have three zaptel drivers. two real ones from my two cards connected to the td400 and what i assume is the ztdummy
03:32.24JTMrTelephone: did you get hardware echo cancellation on the TDM2400P?
03:32.28MrTelephoneyes
03:32.30MrTelephoneit vpm100
03:32.32MrTelephoneit sucks
03:32.40MrTelephoneif there is any buzz on the lines your finished
03:32.41JTyes, the vpm100 is shit
03:32.47wunderkinJ4k3: nubcake!! beefcake!@#
03:32.49smultronkiscokid: so, it reall is a do-it-yourself thing?
03:32.52JTthe new HWEC is much better
03:32.56J4k3wunderkin: meat curtain!
03:32.58MrTelephoneits good?
03:33.10kiscokidJT: one other question.  The telco provider says their default connetor is RJ48.  Do I need to ask for RJ45?
03:33.18MrTelephonewhy does nortel stuff work so good.. is it because its just acting ass a switch?
03:33.46MrTelephonerj48 is the same i think
03:33.52JTMrTelephone: how many FXO lines?
03:33.58MrTelephonerj48 is the same plug as the 45
03:33.59JTkiscokid: the pyhsical connector is identical
03:34.03MrTelephonebut its wired for t1
03:34.06JTthe connector is 8P8C
03:34.12kiscokidsmultron: I guess it is do it yourself if you want to get it done or hire a contractor
03:34.21JTRegistered Jacks specify connector and pinout
03:34.26J4k3smultron: I'd suggest buying some test gear
03:34.32J4k3and learning some about * before going head-first.
03:34.38MrTelephonesmultron, asterisk is a lot of fun actually
03:34.39J4k3and on a buying frenzy
03:34.51MrTelephonesmultron, it will take you like 4 months to get to know it really good
03:35.02smultron4 months!?
03:35.15MrTelephonetheres so many things to learn about voip
03:35.20MrTelephonesip protocol
03:35.31kiscokidthanks JT amd MrT
03:35.36MrTelephonelearning how to use the phones
03:35.39JTMrTelephone: how many FXO lines are connected to this card?
03:35.44miroesq2
03:35.52MrTelephonemy first polycom 501.. i spent 4 hours in my basement getting that thing to boot up, lol
03:36.07MrTelephonejt, i got rid of it and went to a megalink
03:36.14smultronthis isn't sounding good
03:36.18JTyou mean a T1
03:36.20MrTelephoneyeah
03:36.21JTor E1
03:36.23MrTelephonet1
03:36.27JTpri?
03:36.29MrTelephonet1 is awesome
03:36.31MrTelephoneyeah
03:36.33smultronso i spend 4 months reading books and doing test systems to setup a phone system?
03:36.34JTanalogue sucks balls
03:36.37*** join/#asterisk TeeUtada (i=tee@stop.rooting.us)
03:36.44JTsmultron: if you know nothing about the area, then yes
03:36.46MrTelephonesmultron, yeah but the features are endless
03:36.49kiscokidsmultron: it took me 4 months working half time to get familiar enough with Asterisk and the phones to convince my boss we could dump our Norstar crap pbx and switch to Asterisk
03:36.55JTtelecomms knowledge doesn't appear in your brain overnight
03:37.14MrTelephonesmultron, you should have a good linux background too
03:37.25MrTelephoneit will ease your suffering
03:37.40*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:37.52MrTelephonet1 sounds super clean i like it
03:38.02smultronMrTelephone: I'm fairly familiar with Unix systems
03:38.06MrTelephoneso clean that when noone is speaking, you think they hung up
03:38.30kiscokidMrT you need comfort noisr
03:38.33kiscokidnoise
03:38.39MrTelephoneasterisk doesn't support it
03:38.58kiscokidI thought some phones do it
03:39.22J4k3I've got a T1 for data, it can handle a lot of g729, too.
03:39.41kiscokidonly $400 for a PRI here in Palo Alto
03:39.41MrTelephonefinding a good provider is hard
03:39.48JTE1 ftw ;)
03:39.57TeeUtadaHey guys, Random problem I'm having, everything seemed to be working fine, but after going afk for about an hour, coming back and starting up asterisk again, all inbound calls to my DID are getting dropped with "Executing [DIDNUMBERHERE@inbound:1] Answer("IAX2/vit-inbound-2", "") in new stack == Auto fallthrough, channel 'IAX2/vit-inbound-2' status is 'UNKNOWN', anyone have any advice?
03:40.01MrTelephonemaybe you guys can help me out.. how do you failover your sip providers/
03:40.04smultroni mainly just need a beefed up answering machine for the office. is that one of the easier things to set up?
03:40.06sweeperE1 also costs twice as much
03:40.13sweeperso screw that :P
03:40.18JackEStormI've been having a boat load of problems with * and a PRI, and most of it makes no sence.
03:40.21sweeper"zomg we get 8 more channels"
03:40.28MrTelephonesmultron, yes
03:40.28J4k3kiscokid: if I was still colo'd with broadwing (now L3) I'd have T1s for $220/mo and $0.10/did/month... but I gave it up before I realized what I had.
03:40.52J4k3(pretty decent deal considering we were extremely 'small potatos')
03:41.08J4k3but
03:41.12MrTelephonesmultron, grab some hardware, plug it in and come get one of us to get you started ;-)
03:41.34smultronMrTelephone: literally come get someone, or just hop back on IRC?
03:41.36smultron:P
03:41.49MrTelephonei wish someone would have helped me out a little more
03:41.55MrTelephonejust to get the card working the first time
03:42.03kiscokidJ4i3: where are they co-lo'd?  We're going into PAIX in Palo Alto
03:42.14smultronMrTelephone: would you recommend TrixBox?
03:42.19MrTelephonenever tried it
03:42.41smultronbut, digium cards should work without problem, right?
03:42.43*** join/#asterisk skvidal (n=skvidal@fedora/skvidal)
03:42.44sweepersmultron: all signs point to no
03:42.47MrTelephonestick with asterisk it works well
03:42.49sweeperfor trixbox, anyways
03:42.51skvidalhi all
03:42.59MrTelephonebuy sangoma :-/
03:43.11skvidalI was wondering - if I'm hosting a conference on my asterisk server, can I record the conference automatically?
03:43.22MrTelephonesangoma has really good canadian support
03:43.22smultronok, well, i have some more testing to do, it seems
03:43.24skvidalie: is there someway to couple of the Conference() and Record() commands?
03:43.35sweeperMrTelephone: does that include free bacon & weed?
03:43.37MrTelephonei talk to this pakki guy and hes awesome and he phones me back to check if everything is working
03:43.47MrTelephonehaha
03:43.50MrTelephonebacan and weed
03:43.54kiscokidsmultron: I tried Trixbox and I couldn't understand what was going on so I gave up and went with the * sources
03:43.56sweepercanda!
03:44.06MrTelephonedigium was good for support too
03:44.10MrTelephonebut
03:44.15smultronkiscokid: so, just install it ontop of a linux distro?
03:44.16JTsweeper: E1 costs twice as much, what are you talking about?
03:44.20MrTelephonemy wholesaler sells sangoma
03:44.40MrTelephone600 for a t1 here and 24 bucks per pstn channel, 2 bucks a did
03:45.01MrTelephoneits around 1200 for everything
03:45.40kiscokidsmultron: yes and I would use Centos with a 2.6 kernal
03:46.07smultronkiscokid: ok
03:46.36smultronjust seems like spending a lot of money on digium cards not knowing if i can set it up or not.... :/
03:46.51*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
03:46.53MrTelephoneyeah they have a shit resale value too
03:46.59smultrongreat
03:47.03smultron:/
03:47.09MrTelephonehow many lines are you going to run?
03:47.18smultron4
03:47.39MrTelephoneare you in the city or out in the suburbs?
03:47.40J4k3shit resale value is good if you're buying second hand.
03:47.42JTsweeper: ?
03:47.54smultronMrTelephone: small city
03:48.06MrTelephoneyour phone lines sound good when you clear the dial tone?
03:48.07J4k3smultron: do you have broadband available?
03:48.10MrTelephoneno buzzing on the lines?
03:48.13J4k3>100kbit/sec with reasonably low latency?
03:48.28smultronMrTelephone and J4k3: yes
03:48.41pigpenSo, with the manager api, can you have multiple channels per event?  Kinda Like:  Channel:  Local/200*@from-sip&Local/201*@from-sip
03:48.57MrTelephonemy biggest problem is that we were 10miles from the central office so the lines were piss poor and echo cancel wouldn't work
03:49.03JTanyway, that stuff about an E1 costing twice as much as a T1 is utter rubbish
03:49.30MrTelephonee1 will cost the same for the circuit but you would pay a few extra bucks bcause it has more time slots
03:49.36smultronhm, i'm not sure if that's a problem
03:49.42*** join/#asterisk antonrd (n=antonrd@202.151.163.116)
03:49.53JTMrTelephone: you're assuming they even cost the same amount
03:50.01JTMrTelephone: or that you buy the same amount of timeslots
03:50.09JTmore timeslots, even
03:50.11MrTelephoneshouldn't they be close?
03:50.25JTno
03:50.26sweeperJT: it does tho
03:50.28MrTelephonedo telcos offer e1 where they offer t1?
03:50.29kiscokidWhat does E1 give you over T1?  More lines?
03:50.38JTyour T1s in the US vary WILDLY in price
03:50.40*** join/#asterisk osiris (n=osiris@c-71-205-35-230.hsd1.mi.comcast.net)
03:50.44JTthey're pretty constant in .au
03:50.46MrTelephoneE1 has better support for splitting
03:50.49JTfor the one provider
03:50.53sweeperyea, and an e1 is basically 700EU :P
03:51.01JTkiscokid: yes, more channels, too
03:51.10JTsweeper: haha, you get ripped off
03:51.11MrTelephonee1s are sweet if you have a single circuit and you want 2 devices to share the channels
03:51.14sweeperJT: europe
03:51.19sweeperI live in the us~
03:51.28sweeperhow much is an E1 in au?
03:51.28MrTelephoneeverything is a a rip off
03:51.34MrTelephonecollect calls are a rip off too
03:51.34JTa full PRI E1 here is USD$500/mo for 30ch
03:51.39JTUSD$180 for 10ch
03:51.40MrTelephonecell is a rip off
03:51.50sweeperthat's pretty cheap
03:51.57kiscokidMrT: what do you mean by "2 devices to share the channels"?
03:52.02JTit's about 30% more expensive with telstra
03:52.04sweeperstill, T1's can be had for $400 at entry level
03:52.12*** join/#asterisk bmg505 (n=leon@196.209.183.210)
03:52.12JTthose prices i mentioned were optus
03:52.16JTin australia
03:52.28sweeperhere,  I mean
03:52.32MrTelephoneRAD makes some equipemnt wher eyou can over subscribe e1 circuits to other devices.. when i phoned the tech there they said it wouldn't work with t1
03:52.38MrTelephonedon't know much more than that
03:52.39JTso don't assume that just because the max channels is 7 more channels, it will cost more
03:53.01sweeperJT: it generall does tho
03:53.03MrTelephonei was looking for a a device to split my t1 into 2 t1s that share the same timeslots.. supposedly there is no such thing :(
03:53.07weasel00where can i find docs regarding setup of asterisk for softphones?
03:53.08sweeper*generally
03:53.14JTsweeper: maybe in america
03:53.21sweeperJT: we don't have e1's in america :P
03:53.31MrTelephonepoop1's
03:53.31JTthen what's the point of your argument?
03:53.39kiscokidweasel00: Get the Asterisk book from Oreilly.
03:53.56sweeperthat e1 usually costs more per channel :3
03:54.23JTsweeper: not here they don't, and i've seen some ridiculous prices for PRIs in the US, so yeah.
03:54.24sweeperand general angst about everyone not-in-the-us going "ZOMG t1 suck1111?!11"
03:54.37JTsweeper: there was no angst, except from your end
03:54.45sweeperJT: ....that's what I said
03:54.45JTwith crappy wrong costing generalisations
03:54.57sweeperyour generalizations are worse :o
03:55.07sweeperNO U
03:55.13MrTelephonethats awesome we got our 1-800 number today :-/
03:55.15MrTelephoneyippee
03:55.19sweepermuch more satisfying \o\
03:55.22JTwell it was a simple technical point, an E1 is superior to a T1
03:55.39JTnothing to do with cost
03:55.53JTbut you'd generally lose out on cost with a comparison to Australia :)
03:56.10*** join/#asterisk DaveCanoe (n=Dave@adsl-70-235-73-216.dsl.mrdnct.sbcglobal.net)
03:56.18sweepereh. damned euros always waiting a year until the us comes up with a standard, then tacking 2 more features on, and shouting "OURS IS BETTER~"
03:56.44MrTelephoneyeah true
03:56.45JTsweeper: i'm not in fucking europe, okay, give it up.
03:56.56MrTelephonee1 = europe 1
03:57.01MrTelephonedamn germans
03:57.01sweeperJT: then I'm not referring to you, am I?
03:57.09JTMrTelephone: i'm not in europe
03:57.14MrTelephoneim kidding
03:57.15smultroni don't suppose anyone here lives in austin tx and wants to come install the asterisk system for me? :P
03:57.19sweeperwait wait
03:57.23smultronyou've all made me nervous
03:57.23sweeperaustralia isn't in europe?
03:57.29sweeperNO WAI
03:57.35MrTelephonesmultron when you get your card give me some access i'll help you out
03:57.40pigpensmultron, Austin?  shouldn't you be on 6th street?
03:57.59smultroni'm on IRC, what do you think?
03:58.07*** join/#asterisk Rahail (n=rahail@c-68-43-176-199.hsd1.mi.comcast.net)
03:58.08pigpenAh..yes...
03:58.12kiscokidyou can get the Asterisk book for free at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
03:58.12J4k3Austin!
03:58.16J4k3I'm going to roundrock tomorrow
03:58.21pigpenWell, I am about 20 miles NW San Antonio.
03:58.21Rahailany one have any recomendation about DATA center for VOIP
03:58.32sweepersmultron: hey man, buy me a ticket from tampta to austin, and I'll only charge you $300 for a day of consulting :D
03:58.35J4k3pigpen: hrm... comfort?
03:58.41J4k3or is that farther out?
03:58.43weasel00kiscokid : the links on the main site are down for the book
03:58.46pigpenJ4k3, closer....
03:58.53J4k3yeah
03:58.56smultronMrTelephone: what do you mean give you access? like ssh?
03:59.08pigpenJ4k3, The booming town of Boerne, Tx.
03:59.09J4k3I had a friend that lived at canyon lake for a year
03:59.10*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
03:59.11J4k3oh shit
03:59.52sweeperJ4k3: time to change the undies?
04:00.01pigpen....we are waiting for the punch line.
04:00.21J4k3my aunt used to live on sun valley dr (off 46, other side of 281)
04:00.30J4k3haha
04:00.44J4k3well, I guess thats known as Bulverde now.
04:00.47pigpenYeah...nice little speeding trap there.
04:00.51J4k3haha
04:00.51pigpenYep.
04:01.09J4k3I was in a very fast miata, speed traps everywhere.
04:01.12J4k3;)
04:01.41pigpenI am in a big, fat, heavy, dirty, smoking, loud truck.
04:01.53J4k3thats what I drive... I didn't own the miata, I just drove it.
04:01.54pigpenIn fact...it ate a Miata 9 months ago.
04:02.00J4k3haha was it a red 91?
04:02.03J4k3or 90?
04:02.06J4k3I forget which it was
04:02.07MrTelephonesmultron, ssh, its not good to give out to strangers tho
04:02.08pigpenBlue.
04:02.13J4k3ahh, blue deserved to die.
04:02.20pigpenYeah..they almost did.
04:02.37smultronMrTelephone: it'd be on a blank linux install. not much you can do :P
04:02.40pigpenI spilled my coffee.  They hit me head on doing about 50
04:02.46J4k3haha
04:02.55MrTelephonesmultron, just launch a few rockets :-/
04:02.58J4k3I ran over a toyota camry in cedar park
04:03.09J4k3well, they put their camry under my wheel, I just ran it over.
04:03.20smultronMrTelephone: so you're on this channel regularly?
04:03.27pigpenYeah:  Miata vs. Dodge 3500 Dually (9800 lbs), with a front end replacement.  No chance.
04:03.29J4k3I Didn't notice anything until the truck kinda lifted up in the back
04:03.43MrTelephoneyeah at least once a day
04:04.51MrTelephonesmultron, asterisk is fun because you can install eyebeam on your laptop or something and call people whereever you have internet
04:05.00Rahailany one have any recomendation about DATA center for VOIP
04:05.04MrTelephonehaave your messages emailed to you
04:05.13SwKRahail, what kinda volume?
04:05.13pigpenRahail, depends where you want it to live.
04:05.24JT_DATA
04:05.28JT_always needs ALL CAPS
04:05.33smultronMrTelephone: I just need to improve the office VoiceMail system
04:05.38RahailUS
04:05.39Rahailmaby
04:05.44*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
04:05.48JT_"maybe"
04:05.54MrTelephoneim sure you'll be fine smultron
04:06.04MrTelephonehow many station phones are you going to use?
04:06.28MrTelephoneit would be easiest for you to stick with a card that does 4fxo for your outside lines and 8fxs or something for your phones int he office
04:06.34J4k36.5L?!
04:06.38MrTelephoneless dinking around
04:06.45J4k3what happened to the good ol 5.9?!
04:07.01smultronoh, i see what you're saying
04:07.21MrTelephonethen u can use your existing phones and all that
04:07.25pigpenJ4k3, shit...now you got me wondering...
04:07.27J4k3flash flash fun fun fun
04:07.56smultronbut, couldn't i have 1 card with 4fxo and 4fxs daughter cards, then just use phone splitters if i need to connect more FXS phones?
04:08.00miroesqMrTelephone one more question please. i'm doing a fresh install of TrixBox. is there anyway that i can prevent the installation of ztdummy during the install?
04:08.01J4k3I've tried an ata, it made me wanna cry.
04:08.13MrTelephonesmultron, yeah
04:08.37MrTelephone#trixbo
04:08.38MrTelephonex
04:08.46MrTelephonei never used #trixbox
04:08.55miroesqthanks :)
04:08.59pigpenMrTelephone, dont.
04:09.06MrTelephonelet it install and just take ztdummy out of /etc/modules
04:09.08MrTelephoneor something
04:09.15MrTelephonedon't what
04:09.15pigpenBe smart....stay a "Trixbox Virgin"
04:09.19MrTelephonehaha
04:09.23pigpen;)
04:09.27miroesqanything other than trixbox that you can recommend?
04:09.31*** join/#asterisk kikoy (n=kikoy@124.106.83.9)
04:09.35MrTelephonedon't have time to learn anything else really
04:09.48pigpenYeah..me either....
04:09.54pigpen:)
04:10.09MrTelephoneasterisk is really good actually
04:10.36miroesqok, thanks again and have a great night guys
04:10.44MrTelephonei'll pay someone a few hundred bucks here and there to add more functionality to chan_mgcp.c
04:10.47pigpenIf you have a solid deployment, it will do wonders.
04:10.52MrTelephonebut noone has time to do it
04:11.00MrTelephonemiroesq, u too
04:11.04*** part/#asterisk skvidal (n=skvidal@fedora/skvidal)
04:12.41pigpenSo does this sound logical:
04:12.57pigpenI need to "page" a message to around 200 phones.
04:13.14Strom_Cpigpen: step 1: invest in an overhead paging system
04:13.19Strom_Cstep 2: ???
04:13.23Strom_Cstep 3:  profit
04:13.31MrTelephoneuse the dial command to play a wav on pickup
04:13.37MrTelephone?
04:13.58MrTelephonei want asteirsk to phone everyone who is overdue on their internet bill
04:14.07pigpenSo I have a list of exten's, using a phpagi to activate a manager api that make the phone go to speaker which rings an exten that does a playback of a wav file.
04:14.33pigpenOverhead...sorry...building doesn't and won't have it.  ( I checked)
04:14.50pigpenLogical?
04:15.05pigpenPage_app pukes at 108 phones.
04:15.18*** join/#asterisk Won4him (n=Erik@67-132-248-66.dia.static.qwest.net)
04:15.21Strom_Cpigpen: install speakers in the offices
04:15.24Strom_Con the walls or something
04:15.37pigpenToo big.
04:16.06[hC]do them in groups?
04:16.23Won4himwhat u from Phoenix AZ
04:16.26[hC]take your 108 phones and split the list into 2 or 3 groups and do them like that.
04:16.45[hC]Won4him: your sentence didnt make any sense, try again.
04:16.48*** join/#asterisk kiscokid (n=xxx@adsl-216-101-109-187.dsl.snfc21.pacbell.net)
04:16.52Won4himsorry
04:17.00Won4himI was trying to say
04:17.03pigpen[hC], yeah...that would work..and does for that matter.
04:17.08Won4himwhat's up from phx az
04:17.17pigpenBut, it's all about options.
04:17.28[hC]pigpen: how do you see it puke?
04:17.45Won4himI do have a problem that has me guessing though
04:17.46pigpenie: Asterisk crashes.
04:17.55[hC]pigpen: which asterisk version?
04:18.02pigpen1.4.9
04:18.12Won4himbeen looking through all the forums and google and nothing helps
04:18.14[hC]Ah, I am on 1.2, cant speak for 1.4
04:18.22[hC]Won4him: well, out with it.
04:18.24pigpenThe max string length was set to 256, I was told we can bump it up to 8000.
04:18.26pigpenSo we did.
04:18.37Won4himI am trying to get IMAP working in 1.4.9
04:18.44pigpenBefore I could only page 21 phones.  After, 108
04:18.49[hC]Won4him: maybe pigpen can help you, Ive not used 1.4 yet :S
04:18.59pigpenYeah..that is on my todo list.
04:19.01kiscokidIMAP?
04:19.03Won4himthanks for being willing to help HC
04:19.10pigpenI am kinda courious what it will buy me.
04:19.16[hC]pigpen: oh i see. Ive paged about... 50 or 60 so far in 1.2 and didnt have to modify anything
04:19.26Won4himyeah voicemail to IMAP e-mail server kiscokid
04:19.55Won4himpigpen: how do you work voicemail in 1.4
04:20.02pigpen[hC], yeah...that number is easy.  And fast.  But with the allpage.agi (perl) it takes anywhere 20-30 seconds to ramp it up.  page app is great...just pukes.
04:20.02[hC]I think IMAP VM just buys you distributed VM, ie without having to tie it to a specific asterisk box. i dont know if remote mwi works with imap or what the idea is
04:20.06kiscokidWon: you mean you want to have the email go to a server that supports IMAP for clients?
04:20.30Won4himyes like Exchange or Dovecot
04:20.30[hC]pigpen: ahh, I have a Page AGI that just builds the list of extensions and then i pass those to Page()
04:20.46Won4himI heard about MWI
04:20.54pigpen[hC], yeah..with this many phones it is slow.
04:21.03Won4himwilling to try though if I could get it to work
04:21.05[hC]pigpen: how did you speed that up, or have you not got that far yet?
04:21.20kiscokidWon: IMAP is a protocol for receiving email.  Asterisk uses SMTP to send the voicemail to a mail server
04:21.32Won4himwell yes
04:21.33*** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net)
04:21.35pigpenWell, I jacked up the max command length from 256 to 8000, then used the 1.4 page_app.
04:21.35[hC]Won4him: you may want to look on voip-info.org and see if theres anything there.
04:21.47Won4himI will try thanks hC
04:21.49pigpenbut having it puke at 108 phones, tells me the mod may be risky.
04:22.11[hC]pigpen: page_app in 1.4 does things like detects which phones are available, etc? the Page AGI i use simply looks at sip show hints and only pages people that are set to available.
04:22.36kiscokidwon: so what does IMAP have to do with *?
04:22.36[hC]pigpen: so that it doesnt page people who are on the phone, etc.
04:22.37pigpen[hC], yep..it really is very nice...and easy.
04:22.50[hC]pigpen: cool. 1.4 is on my list of things to run in the lab this month
04:23.04pigpenWell, it is running great for us.
04:23.19Won4himKis: IMAP offers the ability to put the VM's in a mailbox store that allows you to check you VM's from either the phone or e-mail client
04:23.24pigpenAfter I have the dam paging behind me, I will turn to realtime psql objects.
04:23.34[hC]IMAP and asterisk go well together so that the sip devices can look to an IMAP box (rather than the local filesystem of the asterisk box) for seeing message status and retrieving your message
04:23.44Won4himthanks hC
04:24.03[hC]there is an inherent problem when you get into spanning multiple asterisk boxes right now where one * box has to store all the voicemail on it, and you have to hack together a remote MWI hack to get it to work
04:24.07Won4himany of you going to astricon?
04:24.22[hC]presumably, with IMAP what happens is all your boxes just query an IMAP account rather than the local filesystem of your asterisk box, so it makes voicemail scale easier.
04:24.24Won4himyeah
04:24.31pigpenWon4him, I want to...but if I go to Phoenix, my Wife wants to go.
04:24.32J4k3Won4him: pft, I'd rather sit at home and enjoy #asterisk :P
04:24.33[hC]I'm going to astricon again this year
04:24.34J4k3and its free.
04:24.35pigpenSo..I won't go.
04:24.49[hC]astricon has been great the last 2 years
04:24.51*** join/#asterisk parolkar (n=abhishek@203.187.254.38)
04:25.05Won4himI went last year and plan on going again this year
04:25.06kiscokidDo the phones know about IMAP or does * do the lookup?
04:25.21J4k3plus, who wants to go to phoenix?  eww.
04:25.23J4k3vegas plz thx.
04:25.29Won4himwell I wish
04:25.32kiscokidwhere is astricon?
04:25.32[hC]kiscokid: * does it.
04:25.36pigpenkiscokid, asterisk must, as I have never seen imap settings in the polycom.
04:25.38Won4himyup *
04:25.42J4k3I mean shit, phoenix might as well be tijuana
04:25.50Won4himHey not that bad
04:25.53J4k3at leat tj is cheaper.
04:25.54Won4himI live here
04:25.57J4k3err least
04:25.59J4k3hah ;)
04:26.17[hC]i have work often in phoenix for the company we just bought there so it works out
04:26.17[hC]:)
04:26.31Won4himyeah right down the street from where I live
04:26.57Won4himlet me ask you all a question
04:27.14Won4himhow well does * handle about 150 phones
04:27.21Won4himon one server
04:27.30Won4himvery well built server though
04:27.43pigpenWon4him, the most phones * ever will handle is 10.
04:27.48pigpen:-P
04:27.53pigpenjust kidding.
04:28.01Won4himDual Xeon with 4 Gig Ram
04:28.10pigpenOnly 4?  Cheap!
04:28.19Won4himgot me worried there pig
04:28.20J4k3Won4him: dual xeon...  how fast? :)
04:28.29J4k3150 should be fine w/o transcoding
04:28.29Won4him2.6 each
04:28.36pigpenWon4him, 64bit?
04:28.40Won4himNo]
04:29.07Won4himwe will not be doing any conferencing
04:29.23[hC]heh you will handle 150 JUST fine.
04:29.24pigpenWell, I have 189 running on a Dell 6850, and it could probably handle about 300-600 phones easy.
04:29.33Won4himnice
04:29.42*** join/#asterisk ber123 (i=brad@neu.cow.org)
04:29.46pigpenPossibly more...
04:29.50[hC]heres one install of mine
04:29.51[hC]219 sip peers [167 online , 52 offline]
04:29.52Won4himI just got my Cisco 7961 to handle SIP on Asterisk
04:29.53pigpenI would love to find the max.
04:29.56[hC]<PROTECTED>
04:30.03Won4himthat took forever
04:30.06[hC]Single p4 2.8ghz
04:30.11lmadsenI can get 500+ simultaneous calls with media on a dual quad-core 2.8 GHz machine at like 0.30 load (no transcoding)
04:30.21ber123If I have two Dial() commands at say 3 and 4 rank in the extensions.conf, if the first dial executes successfully would it still try and do the second dial?
04:30.26Won4himWOW
04:30.26pigpenlmadsen, nice.
04:30.33Won4himnice madsen
04:30.34ber123I would like to have a failover dial in case my primary routes fail
04:30.38lmadsenber123: no
04:30.45ber123ah thats great then
04:30.55lmadsenber123: it'll only continue on if the Dial() fails
04:31.03lmadsenif a bridge happens, that's where dialplan execution stops
04:31.03ber123ok excellent
04:31.09Won4himhey lmadsen have any experience with IMAP and *
04:31.17ber123i heart asterisk
04:31.30lmadsenWon4him: nah... I use ODBC storage
04:31.42lmadsenya... Asterisk is ok I guess :D
04:31.42Won4himfor you VM
04:31.42[hC]Won4him: careful on lmadsen, he's a bit of an asterisk noob himself
04:31.43[hC];)
04:31.51Won4himhah
04:32.04lmadsenya... I don't know much
04:32.19lmadsenI only wrote part of a book... not even the whole thing :D
04:32.23[hC]haha
04:32.25Won4himI met him last year briefly
04:32.33lmadsenI hope I wasn't too much of a dick :)
04:32.34[hC]hows it going leif? i trust youre going to be at astricon this year again
04:32.39Won4himnah
04:32.45Won4himnot after a few
04:32.51Won4himanyway
04:33.12lmadsenhaha
04:33.14lmadsensounds like me
04:33.40pigpenYa gotta have a set of balls to be in this channel.
04:33.55lmadsennot necessarily... we let girls in too
04:34.12Won4himnice
04:34.19pigpenEspecially the pretty ones....   << sounds creapy.
04:34.27lmadsenyes it does :)
04:34.35pigpen...and I can't spell.
04:34.41lmadsen:D
04:34.59Won4himleif: what do you use as a client to check the VM's, only phone?
04:35.14[hC]ARI is nice, for web.
04:35.16lmadsenphone, and we created our own GUI
04:35.20kiscokidlmadsen: when is the new version of *:TFT coming out?
04:35.38lmadsenhas a little bit of flash embedded so we can play the file from the DB directly from the website
04:36.05lmadsenkiscokid: hopefully August. It's nearing the end of production now.
04:36.18Won4himnice
04:36.26kiscokidgreat book, I got the rough cuts version
04:36.37lmadsenawesome
04:36.50lmadsenI especially like the relational database chapter ... :D
04:37.03kiscokidI haven't read that one yet
04:37.15Won4himIs the new book out?
04:37.29lmadsennot yet
04:37.31Won4himah
04:37.48lmadsensoon though
04:37.53lmadsenwithin' a month I hope
04:38.09Won4himgood to hear
04:38.35lmadsenya... I can't wait :)
04:38.50kiscokidAre you going on a book tour?
04:39.01pigpenDoes anyone know if the issue with overloading asterisk with call files has been resolved?
04:39.19pigpenie: moving say, 200 .call files into the outgoing directory at once?
04:39.31lmadsenif you need to use that many call files in a short period of time, use the manager
04:39.31lmadsenthat's what it's for
04:39.41lmadsencall files are for small numbers of one-off calls
04:40.12kiscokidcall files are for doing an outgoing IVR?
04:40.15pigpenyeah...I have the phpagi working...I just need to figure out how I want it to "for loop" through a list of exten's.
04:40.31pigpenlmadsen, sound like I have the idea?
04:40.31lmadsenpretty use While() or a GotoIf()
04:40.49pigpenah...thanks...I suck at php and perl.
04:40.53Won4himleif are you speaking this year
04:41.18lmadsenWon4him: I hope so!
04:41.25lmadsenthinking about doing something on scaling/clustering Asterisk
04:41.32Won4himnice
04:41.34kiscokidcool
04:41.37lmadsenunless people have another topic they would rather me talk about
04:42.07pigpenlmadsen, great...now I'll have to go.
04:42.29Won4himhow * handles voicemail...differnet options
04:43.08pigpenAsterisk:  How to kill a weekend.
04:43.11Won4himdood my company wants to check their voicemail thru phone and e-mail
04:43.14Won4himsuck
04:43.26Won4himexcatly
04:43.33lmadsenpigpen: how to kill 5 years actually... :)
04:43.45Won4himbetter title
04:44.01pigpenlmadsen, yeah..I am on year 2.5
04:44.01Won4himthat should be your next book!
04:45.25Won4himleif: how do your users know hey have VM's when they are stored in DB?
04:45.33kiscokidlmadsen: is the new book going to be open sourced like the first one?
04:46.21lmadsenkiscokid: yep, same license
04:47.04kiscokidcool
04:47.05*** join/#asterisk CunningPike (n=CunningP@204.239.8.149)
04:47.11lmadsenWon4him: I cheat a little bit and use a patch from Juggie which blasts out the notices from another server, but that's because I have a clustered setup with OpenSER as a registration point, but with a single Asterisk, it just works the same way as before
04:47.47Won4himah, that works well
04:47.54sweepercheaters!
04:47.59lmadsenheh
04:48.13lmadsenyou don't need that patch unless you have a strange setup like mine :)
04:49.49Won4himWell if anyone has any ideas this is the CLI error I am getting      app_voicemail.c:2491 inboxcount: Couldn't find mailbox 2992 in context default
04:50.09Won4himversion 1.4.9
04:53.08kiscokidhere's an off the wall question, what's your favorite sip phone?  (I'm getting ready to by 20 for my first deployment)
04:53.30kiscokid*bye
04:53.39*** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net)
04:53.48kiscokid*buy
04:54.10pigpenPolycom anything
04:54.16kiscokid(I had a senior moment)
04:54.16Won4himI have used Polycom alot
04:54.30Won4himbut now I am using Cisco 7961
04:54.36Won4himI like it
04:55.03kiscokidyeah, a lot of people say Polycom.  I will probably go with that
04:58.06lmadsenPolycom is nice
04:59.45pigpenthey mass deploy very nice.
05:00.07kiscokidyeah, I've been reading the doc on how to set that up
05:03.15JTcome on, let's get a cisco hate tirade going, someone mentioned the evil word
05:03.22kiscokidRoR?
05:03.26sweeperrubyonrails
05:03.51pigpensweeper, me too.
05:04.07sweeper:D
05:04.14sweeperusing adhearsion yet?
05:04.41kiscokidis the provisioner available to the public?
05:04.55sweepersure
05:05.00pigpensweeper, luckly, I have a few devs that work with me....I just bitch and moan alot.
05:06.47pigpensweeper, what is "svn"?  :-p
05:06.55sweepersubversion repo?
05:07.02sweeperI think I'll just do a tgz
05:07.03pigpenJoke...really.
05:07.17kiscokidhow does one get it?"
05:07.30pigpenkiscokid, you may want to start with "begging"
05:07.31*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
05:07.37sweeperuh....
05:07.52sweeperkiscokid: do you know what rubyonrails is?
05:08.06sweeperor do you have time and willingness to learn a bit?
05:08.21pigpenPretty rock on a rail?
05:08.25kiscokidI know its a framework for making web apps
05:08.40sweepermkay
05:08.51J4k3stay off the rails, yo
05:08.52pigpenror is pretty dam sweet...
05:08.53sweeperwell, this is a web app written in it
05:08.59kiscokidI'm a C# programmer usually
05:09.09J4k3else you'll end up in rehab, dead, jail, or somewhere in between!
05:09.29pigpenNot a programing lang, not a scripting lang...
05:09.41pigpenbut benefits of both.
05:09.48J4k3its a state of mind.
05:09.51kiscokiddoes it run on Linux? And do I need apache?
05:10.19pigpenDoes it run on Cisco?  hehe
05:10.30pigpenShit..I said that word.
05:11.05kiscokidwhat's wrong with Cisco?  My son and my best friend work there
05:12.15sweeperkiscokid: it runs on linux, if you installation is under 100 users, you could get away with just mongrel
05:12.21sweeperhttp://built-it.net/childhooddevelopment.tgz <-- thar she be
05:12.44sweeperit also has adhearsion in it, but it'll play nice if you don't use it
05:12.47pigpenOh, they are good, as everyone knows, but most people around here prefer to use linux.  We have more control and well, it is cool.
05:13.42kiscokidYou mean use Linux for a router?
05:13.56kiscokidthanks sweeper
05:13.59sweeperno problem
05:14.11pigpenRouter, firewall, vpn concentrator, pbx, mail server, file server, porn server...oops.
05:14.52sweeperI won't be doing many changes to the provisioning side of things, but if you end up interested in the app as a whole, there'll be a nicer version in a couple of days.
05:14.56kiscokidwell once you get over a certain bandwidth cisco routers beat the pants off linux
05:15.00pigpenPrice out a cisco solution that can handle GB ethernet routing for 16 interfaces.  Then price out a Server with linux.
05:15.19sweeperthe cisco will be more reliable :P
05:15.43pigpenWell, I would say "comparable" if linux is done right.
05:15.49kiscokidsweeper I assume your email is somewhere in the tarball?
05:15.51pigpenMostly it comes to the hardware.
05:15.54sweepermmm
05:16.07sweeperI should probably stick that in there, since I'm bandying these things about now
05:16.19sweeperaleks.clark@gmail.com fwiw
05:16.30kiscokidgreat, thanks
05:17.45pigpenOne thing to note:  Just grabbing any linux dist off a shelf does not constitute as the OS I am referring to.
05:18.12kiscokidpigpen: which distros do you like?
05:18.21sweeperpigpen: what, you mean ubuntu isn't the same as gentoo??!?!?!
05:18.39pigpenWe use Gentoo, but I am biased.  My business partner is a Gentoo Kernel dev.
05:19.02pigpensweeper, that was good.
05:19.17kiscokidMy manager likes Centos so we use Centos
05:19.44*** join/#asterisk tuzhila (i=tuzhila@84.47.128.99)
05:19.47tuzhilahi all
05:19.59*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
05:23.56pigpenkiscokid, think of Gentoo being a bit more "hard core."  I have friends who are working in linux 9 hours a day, who are daunted by Gentoo.
05:24.17pigpenBut with this, you get a very stable system...if it is done right.
05:25.06pigpenGentoo's issue is that there is so much "workstation" crap.  They need a portage tree for enterprise server solutions.
05:25.15kiscokidCentos seems pretty stable.  It doesn't crash
05:25.17pigpensweeper, perv.
05:25.47*** join/#asterisk skvidal (n=skvidal@fedora/skvidal)
05:26.03pigpenCentos isn't bad...it is pretty easy..but it is rpm based.
05:26.39pigpenIf I have a 64bit xeon box, I want it to be built -only- for a 64bit xeon box.
05:27.15skvidalI'm looking for a way to have a more secure way of storing passwords for sip or iax users w/asterisk
05:27.19pigpenCisco builds there OS's for the hardware in mind.  Not every option out there.
05:27.34skvidalI'm looking in users.conf for how to add users - which is very straightforward
05:27.57skvidalbut it looks likes the passwords are all stored in plaintext
05:28.31skvidalis there some way to have the passwords there either be 1. crypted or 2. have asterisk auth to pam, kerberos, radius, cyrus?
05:28.32JTkiscokid: cisco treat their customers with disdain
05:28.40JTand their products aren't as good as reputed to be
05:28.58pigpenJT, nicely put.
05:29.16kiscokidyeah, well they are a bit arrogant since they got so big and successful
05:29.28pigpenIf you have ever worked with them on bugs, you find out real quick what they are all about.
05:29.43JTi have a no new cisco policy
05:29.45JT:)
05:30.17kiscokidBut I also had a nightmare working with Juniper on a vpn bug
05:30.36pigpenWhen it comes to VPN's...Linux rocks.
05:30.38kiscokidtheir vpn client software is crap
05:31.19kiscokidcan I make a linux vpn that can be used by the PPTP or L2TP clients in Windows?
05:31.25*** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com)
05:31.42kiscokidthat would be a real winner for me at work
05:32.12pigpenkiscokid, you can use the vpn crap that comes with windows, but it is a pain in the ass.
05:32.35pigpenbest bet is to buy <I forget the name>, but it is $140'sh per seat.
05:32.58pigpenWe actually have setup the cisco vpn client to connect to strongswan on linux.
05:32.58kiscokidwell it has one advantage, its already deployed and paid for in all our clients
05:33.17pigpenbut don't tell cisco.  :)
05:33.32kiscokidstrongswan?
05:34.02pigpenyeah..they spun off of openswan (I think) quite awhile ago.
05:34.29pigpenI am told, openswan is crap compared to strongswan.
05:36.13*** join/#asterisk tengulre (n=tengulre@222.90.66.10)
05:37.29*** join/#asterisk thx2000 (n=the@netblock-208-127-94-59.dslextreme.com)
05:37.39thx2000Anyone have any hints for getting siptapi to work w/ vista?
05:38.07coldstealthx2000: reformat ant put linux
05:38.10*** join/#asterisk hoowa (n=chatzill@210.83.203.100)
05:38.37thx2000...
05:38.50coldstealthx2000: yeah idk
05:39.07*** join/#asterisk fujin (n=fujin@unaffiliated/fujin)
05:39.11fujinhi there, anyone around?
05:39.14fujincould someone take a look at http://rafb.net/p/S8dy7t42.html
05:39.25fujinand tell me if it is asterisk trying to register, or the 192.168.108.210 device?
05:45.25tengulreyour device's sip protocol is different with asteriks , I think.
05:50.35tuzhilafujin: check your settings on device
05:51.24*** join/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181)
05:53.01*** join/#asterisk af_ (n=getsmart@81-174-47-190.dynamic.ngi.it)
05:56.02coldstealcouls someone help me with this http://rafb.net/p/zOJiFH26.html i have all ther erros and conf files in there
05:56.07coldsteal*could
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06:11.27JTbri to pri porting...
06:16.46thx2000coldsteal: what do u get for "sip show registry"
06:21.48*** join/#asterisk lsodi (n=lsodi@84-50-22-19-dsl.kjj.estpak.ee)
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06:35.15lsodigreetings, I'd like to make web interface for users where they can configure call forwading. and now I have come up with 2 ideas how to do it. 1. web interface puts famili and key values directli to asterisk DB
06:36.32lsodi2. php script puts/gets all valuses into mysql and AGI launches php to determine is call forwarding on or off
06:36.38*** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no)
06:38.22lsodior is there 3 way to do it?
06:38.57*** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl)
06:40.51creativxsounds like 2 good ideas
06:41.03thx2000ini file would prolly work, if u're trying to avoid running mysql
06:41.10thx2000version 1 is prolly the easiest though
06:41.21creativxdepends on the number of users
06:41.39lsodiat least 200 users
06:41.55creativxsql
06:42.03creativxso you dont pollute the astdb
06:42.18tj_daside from Asterisk: The Future of Telephony, can anyone recommend some good resources for a noob?
06:42.25*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
06:42.28creativxnow i havent benchmarked an agi->mysql lookup
06:42.46creativxyou might want to take that into consideration lsodi.. although with ringing() you can afford some execution time :)
06:45.53lsodiok thank you, I'll go with php and mysql
06:45.57thx2000200 entries is pretty modest, can't imagine it'd take all that long
06:46.11thx2000specially on a machine strong enough to handle 200 users
06:46.13*** join/#asterisk dominic1 (n=dob@213.221.82.242)
06:48.48lsodiasterisk and sql are on different servers
06:49.53*** join/#asterisk putzz (n=me@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com)
06:50.15putzzhow would I go about connecting 2 calls in progress together without using Meetme?
06:50.42snuff-workmm JT is there a way to make the transfer method by default on a poly 601 'blind' mode?
06:51.03*** part/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181)
06:52.03JTno idea
06:52.16JTjust migrated a multi BRI setup to PRI :)
06:53.42Aurspresence and realtime peers = nok ok?
06:53.50snuff-workwoohoo die die misdn
06:53.55snuff-work:)
06:54.03*** join/#asterisk zeeesh (i=zeeesh@202.166.161.36)
06:54.38JTi was actually using bristuff :)
06:55.12snuff-workcheater ;)
06:56.03JTstill am, but only in NT mode
06:57.39*** part/#asterisk skvidal (n=skvidal@fedora/skvidal)
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07:01.04nbagshey can someone help me with this problem. i have a register statement in my sip.conf, so i can get incoming calls. this works. but if my ip address changes (i have 2 dsl lines in case one goes down) i cannot re-register. i get no response from the sip server, and registration times out. if i switch back to the original ip address, i can reregister instantly. i have observed this behaviour with 2 different sip providers. is there a way i can force it to reregi
07:02.24snuff-workmm.. wouldn't a 'sip reload' do the trick?
07:02.31nbagsno
07:02.38nbagsit just times out
07:02.48nbagsand sip show registry says 'request sent'
07:03.06nbagsi have sniffed the packets and the server simply doesn't respond to the register
07:04.35*** join/#asterisk shayx (n=shay@unaffiliated/shay)
07:05.19snuff-workmm.. not much u can do.. maybe only if u could 'unregister' from the service that went down.. but kinda illogical
07:05.36nbagsyeah, it doesn't make sense!
07:06.25*** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil)
07:06.53shayxHello, I've configured a new SIP peer in my asterisk server, "sip show peers" shows the peer, but when I try to connect remotely with a SIP client, it says "registration failed". On the CLI, using sip debug, doesn't shows nothing.
07:08.49shayxhmm, for some reason the SIP port does not seems to be opened on the server (port scanned localhost) :-/
07:10.16snuff-workwell u could go assuming centos here.. /etc/init.d/iptables stop
07:10.20snuff-workif u have iptables
07:10.34*** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com)
07:10.44shayxno, locally the host isn't firewalled
07:10.50shayxthere's no iptables running
07:11.13shayxis there a way of enabling/disabling the SIP protocol on the Asterisk configuration?
07:14.01*** join/#asterisk saftsack (n=oliver@p54A7EABC.dip.t-dialin.net)
07:14.56shayx$ nc localhost 5060
07:14.56shayxlocalhost [127.0.0.1] 5060 (sip) : Connection refused
07:14.57shayx:/
07:20.06Juggieudp only.
07:23.56shayxfine, 5060 seems to be open on udp
07:25.47shayxbut i still can't connect from a SIP softfone
07:25.49shayxsoftphone
07:30.00JTwhat softphone?
07:30.10dominic1what is currently the most preferred management proxy
07:30.21JTdominic1: "management proxy" ?
07:30.24dominic1I only know astman, but I think this was for version 1.2
07:30.36JTAMI proxy you mean
07:30.50JTi think hardly anyone uses AMI proxies
07:31.00JTshayx: ?
07:31.03creativxastmanproxy
07:31.04dominic1yes, JT Asterisk management Proxy
07:31.18creativxhas there been any changes to the 1.4 ami?
07:31.18JTdominic1: asterisk manager interface proxy
07:31.23shayxJT, Ekiga
07:31.31JTshayx: there's your problem.
07:31.33dominic1is astmanproxy ready for version 1.4, cause I am only using 1.4 versions in my environment
07:31.37JTshayx: run it from a different host.
07:31.41dominic1@JT URL?
07:31.53shayxJT, which softphone should I use?
07:31.56JTor make ekiga bind to 5061 instead of 5060
07:31.59sweeperpeople who have a lot of boxes to managed probably use AMI proxies :P
07:32.04JTshayx: something not on the same machine as asterisk
07:32.08Aursdominic1: it's in svncommunity.digium.com I think
07:32.18shayxJT, no, I'm not on the same machine
07:32.22creativxdominic1: http://svncommunity.digium.com/view/astmanproxy/
07:32.27JTshayx: ekiga binds to port 5060
07:32.39shayxJT, the server's at work, I'm trying to connect from home
07:32.57JTdominic1: url, what
07:33.19dominic1There hasn't been any changes for 10 month, I hope there is no problem with the support of the 1.4 versions
07:33.23dominic1any hints about that?
07:33.36creativxread the 1.4 changelog
07:33.42creativxi doubt the AMI has been changed
07:33.45shayxI'm starting to think that it's the OpenBSD firewall
07:34.19shayxsince I can connect to the 5060 UDP port remotely (nc -vu says: "sip open") but on the CLI with sip debug, i don't get anything
07:34.38shayxso maybe the port is open but there's something wrong with the forwarding
07:34.46dominic1Are there no other proxies, which are 1.4 compatible?
07:35.03creativxare there any other proxies at all?
07:35.35creativxhttp://www.voip-info.org/wiki/view/Asterisk+Manager+Proxy
07:35.38creativxwonders of google
07:40.07shayxYAY
07:40.08shayxworking
07:40.09shayxregistered
07:40.10shayx:)
07:41.09*** join/#asterisk y7n (n=na@office.intercea.co.uk)
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07:42.16y7nWhat would be the best way to limit calls between certain hours of the day? e.g only allow calls to be made from 7pm - 7am
07:42.53creativxgotoiftime
07:42.57creativxperhaps
07:43.10y7nthanks
07:44.10shayxcan someone try a sip call with me?
07:45.36codeyguess so
07:45.52shayxcodey, interested? :)
07:46.11codeyi can try it, not sure if it works
07:46.17codeyjust fscked up the firmware of my snom phone
07:46.18*** join/#asterisk saftsack (n=oliver@p54A7D9A5.dip.t-dialin.net)
07:46.18codey:P
07:46.59*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
07:49.37*** join/#asterisk sadara (n=sadara@office.wildwoodgroup.com.au)
07:49.55sadara.
07:52.58y7nim using voip-info.org to find descriptions of functions, but is there a complete reference manual for asterisk that gives detailed explanation?
07:53.36creativxshow function <function> in CLI
07:53.44creativxalso google
07:53.46y7nok
07:53.52JTshow application <blah>
07:53.55creativxvoip-info has a good referance
07:53.59JTneg
07:54.04JTvoip-info has a reference
07:54.08JTit's rarely good
07:54.10JToften outdated
07:54.11creativxfor 1.0 - 1....2
07:54.12creativx:)
07:54.12yonahw-workalot of the voip-info data is out of date
07:54.17yonahw-workyeah what JT said
07:54.21creativxi wasnt done typing JT damnit :P
07:54.23y7nyeah, i noticed that
07:54.30creativxi saved the sarcasm for the next line.. hehe
07:56.06JT;)
07:56.36JTof course, the stuff on voip-info about zaptel span definitions is very good...
07:57.56y7nlooks like gotoiftime is deprecated
07:58.05y7nin 1.4.9
07:58.28y7nthere is an IFTIME function
08:04.03mepplgood morning
08:04.12sadaragood afternoon
08:04.41mepplgood afternoon sadara
08:07.36*** join/#asterisk saftsack (n=oliver@p54A7DA26.dip.t-dialin.net)
08:10.27*** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com)
08:11.18Juggieare all the doxygen docs 404ing for anyone?
08:12.40*** join/#asterisk nighty^ (n=nighty@p2136-adsau17honb13-acca.tokyo.ocn.ne.jp)
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08:20.06codeyit's possible to start multiple iax2 connections on one port, right?
08:21.14JTiax2 trunking
08:21.18JTrequires zap timing
08:22.03codeyi've got one host on a dynamic ip that connects to our pbx
08:22.10codeybut since yesterday it times out all the time
08:22.47codeybut i can connect with netcat etc. without any problems
08:23.45JTis the pbx on a dynamic ip
08:23.46JT?
08:23.51codeyyep
08:24.27JTwell are you using dyndns?
08:24.29*** join/#asterisk ghantoos (n=ghantoos@m151.net81-65-52.noos.fr)
08:25.07codeyyes
08:25.39JTof course you did restart asterisk when you got a new ip, right?
08:25.56codeyi've restarted it multiple times yesterday and it just keeps timing out
08:26.06JTwhich times out
08:26.11codeythe dynamic one
08:26.14shayxwhat's the difference between a sip server (as in, the asterisk server itself) and a SIP proxy?
08:26.31JTshayx: asterisk is a SIP B2BUA, it doesn't proxy
08:26.56JTcodey: i thought they were both dynamic
08:27.05shayxI'm trying to connect from two different machines, two different clients, from one user configured as a SIP peer to another, and they say that the host cannot be found
08:27.30shayxfor some reason, the CLI debug shows the NATed IPs of the hosts (they're both remotely connected, not on the same network as the PBX)
08:27.48shayxdo I need a SIP proxy or there's a configuration needed on the Asterisk PBX?
08:27.50JTNATed IPs... public or private?
08:27.55shayxprivate
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08:28.10JT~sipnat
08:28.10jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
08:28.17shayxthanks, JT
08:29.00*** join/#asterisk gardo (n=gardo@121.97.195.87)
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08:30.45dominic1I heard there is a new feature instead of macros
08:30.49dominic1how is it called?
08:31.34*** join/#asterisk matsk (n=mk@194.68.102.175)
08:34.52codeyJT: the pbx at my side is static
08:34.56codeyJT: and the other one is dynamic
08:35.01codeythe dynamic one registers at the static
08:35.08JTcodey: there's two?
08:35.09JTi see
08:35.23codeyerr, you need 2 asterisk to do iax2 peering ;)
08:37.25*** join/#asterisk Renacor (n=kvirc@62.157.211.194)
08:37.41Renacoranybody know if the tdm400p pci card is 3.3v or 5v?
08:38.03codeyJT: so, any hints where to look? i've checked firewalls on both boxen and our core router
08:38.07codeynothing gets dropped at all
08:38.21JTnot sure
08:38.42codeymaybe it's because i'm running 2 iax2 peerings on the same port?
08:45.31*** join/#asterisk adeeln (n=adeel@c-67-161-185-121.hsd1.ca.comcast.net)
08:46.33adeelndoes anyone know if a2billing can administer asterisk itself (e.g. trunk creation, dialplans, etc) or do you need to manually do trunk creation & dialplan ?
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09:03.51tuzhilaadeeln: have you already installed a2billing?
09:04.05adeelntuzhila, yes
09:04.32tuzhilaand how? i can't install that!
09:04.49adeelninstalling wasn't that bad
09:04.58adeelnhave you tried following the instructions on the site?
09:05.14tuzhilayes, there are a pdf instruction
09:05.22tuzhilayes?
09:06.16adeelnthe steps are pretty straight forward....you need to create the database, grant the proper permissions
09:06.36adeelncopy over the related agi-bin files to the right directory, (make sure they're executable)
09:06.52adeelncopy over the UI directory to your web root dir
09:07.31adeelnadd a couple lines here and there in /etc/asterisk and your pretty much done
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09:31.55fors1is there a way to make asterisk check the latency on an IAX2 connection before transferring the call, and if the latency is higher than a given value, just proceed to next priority??
09:35.54*** join/#asterisk michael-i (n=michael-@141.41.40.55)
09:37.31michael-ii have a question about using a single context in iax.conf to both send and receive calls. i'm currently getting "unauthenticated" rejections on incoming calls as it is not finding the appropriate context in iax.conf.
09:38.50michael-icontexts are found by matching incoming usernames (as I understand it) but I don't wish to name my contexts with usernames as I may have the same username with two separate iax accounts. sip.conf does not behave like this...what am I missing?
09:40.14*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
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10:01.38xipihi
10:02.32xipii am trying to setup asterisk
10:02.52xipiwhen i type reload, there is a long list of errors.
10:03.36xipione of those is a complaint regarding missing 'ael-dundi-e164-local' context
10:03.53kikoyanyone with idefisk installed in their computer?
10:04.48xipiwhat can i do? any idea?
10:06.56kikoywill someone connect to my asterisk server? username is 'mytest', password is 'mytest', host is delamar-icite.com, iax protocol.
10:06.59kikoyplease...
10:06.59kikoy:D
10:07.01kikoyT_T
10:07.16*** join/#asterisk Jubei (n=nocuser@uranus.noc.tuc.gr)
10:07.24kikoyi just want to verify if it's working from outside
10:08.11Jubeiguys my digium te212p's 1st port is blinking red. What does it mean coz I can't find the manual (digium makes you register for it)
10:08.57Jubeithe funny thing is that it's blinking red regardless of whether there's a cable in it
10:09.03xipifound info in a forum. seems like it's not essential
10:09.06xipithanks
10:09.08xipibye
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10:41.55aikanaro79hi people...I'm quite new to Asterisk and have been reading the O'Reilly book on it...however...I need some help configuring users....can anyone help me?
10:42.19cpmrule #1, don't ask to ask, just ask.
10:42.30*** join/#asterisk IgorG (n=FeedomPa@195.162.32.126)
10:42.55cpmExplaining your problem is half the solution.
10:44.17aikanaro79thanks cpm...I'm configuring asterisk to act as a server in a private network conference call...only SIP channels are allowed...is there a way to configure a "class of users" so that I don't have to configure every possible user?
10:44.51cpmyou could just add a pin
10:45.30aikanaro79how is that? I mean...was that exactly?
10:47.37*** join/#asterisk IgorG (n=FeedomPa@195.162.32.126)
10:49.59aikanaro79I don't know if it's possible or not but what I would like to do is to configure a class of users (for example with pattern matching if it's supported for that end) so that when someone tries to register with asterisk it just checks to see if the caller matches the user class previously defined
10:50.01cpmadd a pin to the conference, so that folks have to enter a numeric code to join the conference
10:50.17cpmthat's the simple way to control access
10:50.47aikanaro79but that's when you already have a conference call established right?
10:51.27aikanaro79this server is supposed to have users logged in...and any user should be able to start a conference with any other users
10:52.28cpmah, I see. Sorry, I don't know how to do that. I do it differently.
10:52.41cpmHowever, others will, so be patient.
10:52.43aikanaro79:)
10:52.49aikanaro79thanks anyway :)
10:53.39aikanaro79if you know of "somewhere" I could look for the information (other than google and o'reilly book on asterisk) that would be of great help also
10:54.19cpmhttp://www.voip-info.org/
10:54.26*** join/#asterisk y7n (n=na@office.intercea.co.uk)
10:55.45y7nis it legal to do this exten => s,3,Hangup then exten => s,4,System(command) or does it HAVE to be the other way round?
10:59.53*** join/#asterisk menil (n=meni@bzq-179-153-168.static.bezeqint.net)
11:04.22cpmy7n, no idea. Have you tried it?
11:04.28cpmyou can just try things ya know.
11:04.36cpmit's your pbx afterall :)
11:06.42*** join/#asterisk saftsack (n=oliver@p54A7C00C.dip.t-dialin.net)
11:06.46*** join/#asterisk liversmudge (n=chatzill@217-14-176-200.as25582.net)
11:06.57liversmudgehow do!
11:07.15liversmudgewho is awake to day then?
11:07.30cpmno one is awake. You loose 2 points for asking
11:07.45liversmudgeso I now have -2 points
11:07.52liversmudgeokeee ... bad move .
11:07.56cpm:)
11:08.08liversmudgewell then who knows anything about echo cancellation
11:08.20liversmudgecoz Im pulling me hair out chaps.
11:08.23tzafrirwho is assleep right now
11:08.43liversmudgeits like this
11:09.36liversmudgeI had an * box with a TDM400 in it with 4 fxo cards
11:09.44liversmudgeand had no problem with it
11:10.10liversmudgejust upgraded to a trixbox iso moved the card to a newer/faster machine and the echo is BAD
11:10.14liversmudgeVery bad
11:10.44liversmudgeso the question was .. has the EC software changed recently or can echo start on different hardware?
11:11.34liversmudgeyou see I dont know if this is a trixbox question or a core asterisk question
11:14.04tzafrirliversmudge, tried fxotune?
11:14.20liversmudgeno , where do I get that from?
11:15.22tzafririn the zaptel source directory
11:15.34liversmudgerealy .. told you I was out of it!!
11:15.35tzafrirHowever, I recommend you to get the one from zaptel 1.4
11:15.37liversmudgeIll have a look
11:15.50tzafriryou use zaptel from source?
11:16.14cpmtzafrir, run away! it's a trixbox user!
11:16.44*** join/#asterisk Uatec_ (n=uatecuk@adsl.ntsols.com)
11:16.45Uatec_hey
11:17.22Jubeiexit
11:18.39Uatec_do any of the junnhgrnennnans BRI cards have onboard echo cancellation or anything?
11:18.47*** part/#asterisk ming_zy1 (n=ming_zym@124.254.54.15)
11:18.55Uatec_cos my boss insists on using a b410p with misdn becuase it has onboard echo cancellation
11:19.27JTaikanaro79: contexts.
11:19.35*** join/#asterisk Tiaro (n=ddd@203.173.234.144)
11:19.37JTnope
11:19.43JTUatec_: hold out for the sangoma A500
11:19.45TiaroCan someone help me with a Cisco 7940
11:19.47JTyou can get HWEC
11:19.48TiaroI have read all the docos
11:19.50Tiaroits a sip version
11:20.00JTand it has included octastic software EC for 8 chs
11:20.24aikanaro79JT: thought of contexts...but they only appear in the dialplan...my problem is a way to define a class of users (for registration purposes)
11:20.33JTno need
11:20.39JTcontexts
11:20.55TiaroPlease msg me if someone can assist with a Cisco 7960 config
11:20.58TiaroSorry to disturb
11:21.25Uatec_sangma A500? what software do you need to run that? (drivers)
11:21.55JTUatec_: it will be using chan_woomera
11:22.04*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
11:22.06MaliutaTiaro: what issue with the cisco
11:22.14Uatec_JT, it's only got 3 lines.. :(
11:22.24JTUatec_: what are you talking about?
11:22.33Uatec_that's all i can see on the picture
11:22.40MaliutaTiaro: I have my 7940 (only difference from 7960 is less lines - 2 as opposed to 4) working fine
11:22.42puzzledhi
11:23.43Tiarooi
11:23.45Tiarook
11:23.48TiaroMy issue is as follows
11:23.50JTUatec_: read specifications, not pictures.
11:24.01JTUatec_: from 2 to 24 BRI ports.
11:24.05TiaroI have installed Asterisknow and I have two extensions working on 7960
11:24.09TiaroHowever, no matter what I do
11:24.14JTUatec_: the most flexible bri card available.
11:24.18Uatec_the flash thing says "expands up to 24 ports and 48 lines" and shows an image of 12 ports
11:24.22Tiarothe config on line 1 never authenticates
11:24.32Tiarosorry it authenicates but never gets an outside line
11:24.36Tiaroit always goes to reorder
11:24.42JTUatec_: it's between 2 and 24 BRIs, read up on how sangoma cards work.
11:24.44Tiarobut if I put a line on line 2-6, no issue
11:24.50*** join/#asterisk yassaccan (n=yassacca@admin131.hgo.se)
11:25.03Uatec_i am
11:25.07Uatec_hmm
11:25.12*** join/#asterisk Badas (n=badas@82.155.75.48)
11:25.13TiaroI can not make internall calls from Line 1
11:25.20Uatec_the price seems to be mounting up as i add stuff
11:25.27Uatec_but i'm sooo goingt o get one (some) of this
11:25.45TiaroIt says username mismatch on the console
11:25.55Tiarobut its identical on the asterisk box
11:26.03Tiaroand It works if I move it to another line other than one
11:26.09JTUatec_: the price is roughly equal to junghanns, with no HWEC, but it also comes with octastic software ec
11:26.10Tiaroit doesnt matter if I rotate exchanges
11:26.21Uatec_octastic?
11:26.21TiaroIf someone can help pls msg me
11:26.34creativxoctagon fantastico
11:26.38JTUatec_: read.
11:26.54JTUatec_: octastics make the HWEC firmware for digium and sangoma
11:26.59JToctastic, even
11:27.14kikoytest my server please... username is 'mytest', password is 'mytest', host is delamar-icite.com, iax protocol. thanks
11:27.21Uatec_oooo
11:32.07TiaroAnyone can help?/
11:34.08tzafrir~ask
11:34.08jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there, just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
11:34.24TiaroThanks jbot
11:34.27TiaroAS I stated
11:34.31Tiarohere is my issue
11:34.40TiaroI have a handful of Cisco 7960's at Home
11:35.11TiaroI wanted to setup an internal IP PBX, so I downloaded ASterisk now. I do have experience with a different switch and I have a fair understanding of Cisco SIP Handsets
11:35.24TiaroI have the TFTP correctly installed with images, tones, configs, etc
11:35.38TiaroI believe I have the correct configuration to the handsets
11:36.01TiaroHowever, I do not appear to be able to register line 1 on a cisco handset
11:36.08TiaroI can register the account on line one
11:36.12Tiarobut unable to make calls
11:36.15Tiaroit can recieve calls
11:36.44Tiaroon the console it states chan_sip.c:XXXX check_auth: username mismatch
11:36.51puzzledTiaro: I'm not sure if AsteriskNow is in a usable state. Maybe try to install Asterisk 1.2.23 or the latest 1.4 and try with that
11:36.51Tiarothen NOTIC
11:37.01Tiarook
11:37.19TiaroIs that as simple to install as Asterisk now?
11:37.23Tiarocause that was a breeze
11:37.37puzzledI don't know AsteriskNow so can't say
11:38.44ashdi have had problems with asteriskNOW and am currently installing asterisk as i think it will actually work
11:39.16Maliutasounds like an issue in the sip.conf,
11:39.28Tiaroyeah it does
11:39.38TiaroBut I triple checked
11:41.05Tiarohandle_request_invite
11:42.06TiaroFailed to Authenticate user
11:43.11*** join/#asterisk Tiaro (n=ddd@203.173.234.144)
11:45.03*** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com)
11:53.40*** join/#asterisk ming_zy1 (n=ming_zym@124.254.55.15)
12:04.59*** join/#asterisk lirakis (n=etamme@65.200.191.253)
12:06.41*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)
12:07.06*** join/#asterisk eran` (n=eran@kiwidsl.bb.netvision.net.il)
12:07.15eran`hey
12:08.02*** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
12:08.16eran`uhm, any of you guys had some echo problems?
12:11.35tzafrirno, no , no, no
12:11.55tzafrirI suggest you be more specific ;-)
12:11.58hi365can i check the amount of new VM messeages form the cli?
12:12.24hi365hello tzafrir
12:13.41hi365ah. gotit: show voicemail users will show new voice mail
12:14.45*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
12:14.56hi365are voicccemail numbers stored a variable anywhere? i.e. if i want to use externnotify= in voicemail.conf, how do i pass it variables?
12:15.22*** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com)
12:15.22tzafrireran`, what device do you have?
12:18.09*** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.3.41.revip2.asianet.co.th)
12:18.54ManxPowerhi365: you would have to create an extension that runs HasVoicemail then Noop to see the value of HASVMSTATUS in the CLI
12:20.22ManxPowerhi365: no you can't pass your own variables to externnotify, but I can look up what variables the system passes to it by default, if you want
12:23.07eran`linksys
12:23.33eran`how much should I set the gain input/output
12:23.49ManxPowereran`: 0 is a good starting place
12:24.05ManxPowerTo follow the saying "no gain, no pain"
12:24.47eran`ok
12:25.28s0ckis there a general option within asterisk to transmit silence or would it be handset specific
12:26.32tzafrireran`, you hear echo on your side? What's on the other side?
12:26.38hi365ManxPower: i would appreciate if you can look it up for me
12:26.40eran`echo too
12:27.34dominic1Is it possible to set up useraccounts with names und how can I set a digit base extension to that account?
12:27.49dominic1I am currently using asterisk realtime
12:27.49*** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no)
12:28.49eran`on both sides tzafrir
12:29.11dominic1And how can I dial names in asterisk?
12:30.25ManxPowerhi365: it passes 2 command line params to the externnotify script  the voicemailbox and the voicecontext
12:31.26*** join/#asterisk guillote_GNU (n=bancaria@host212.201-252-198.telecom.net.ar)
12:31.43ManxPowerdominic1: for the most part you can't dial name extensions in asterisk unless you are using an IP phone, in that case then you must read the docs for your IP phone to learn how that specific make/model allows the user to enter non-number items for dialing
12:32.04HaMYaIhow do we do load balancing on SIP or IAX2?
12:32.24ManxPowers0ck: Asterisk ALWAYS transmits silence.
12:32.36dominic1and how am I able to process this in the dialplan?
12:32.59ManxPowerdominic1: in non-realtime you would do something like exten > rdobbs,1,Dial(SIP/12345)
12:33.04HaMYaImy * box nearly reaches BW limit and I will need to redirect it to another IP
12:33.12ManxPowerthen you would call rdobbs and it would send the call to sip userid 12345
12:33.30ManxPowerHaMYaI: there are many almost-workable solutions on the Wiki
12:33.42hi365ManxPower: hmm, so i cant even tell the script how many new messages there are or who its from...
12:34.15ManxPowerhi365: with the mailbox and context you script can look at it for itself.  the info is stored in msgNNNN.txt in the users INBOX folder.
12:34.19HaMYaIManxPower: like SER?
12:34.28ManxPowerHaMYaI: no idea.
12:34.33hi365ManxPower: thanks
12:34.47dominic1I want to use shortnames with 3 for authentication and prefer the following  XXX => Dial(SIP/${EXTEN}). In my case XXX should stay for 3 characters (my shortnames)
12:34.55hi365ManxPower: thanks, ill look in to it
12:35.14ManxPowerdominic1: doing it that was is far, far more complicated than using digits.
12:35.35ManxPowerhi365: I didn't say it was EASY. 8-)
12:35.56ManxPowerhi365: remember the user could have more than 1 voicemail
12:36.03hi365i relize, i guess thats why the script is there in the first place
12:36.05ManxPowerheck the user could have more than one NEW voicemail
12:36.35hi365true, i would only give him the count and the callerid of the latest (i want to send it via sms)
12:36.42dominic1oka, but I think the following will work: authenticate the phone with the shortname. How will I have to setup the user? Where will I have to store the number?
12:37.11ManxPowerdominic1: all extension <-> device mappings are done in extensions.conf
12:37.22ManxPowerWe use the MAC of the device as it's SIP userid
12:38.42dominic1Okay, then I will add a entry to my mysql database which is known as number and everytime a extension is dialed, it does a lookup on the database via odbc to fetch the right account for that number in my table sippeers
12:41.09Uatec_hey, i hear that ABE 2.2 is out. Is that based on 1.4 or 1.3 again?
12:43.48JTthat thing still exists? ;)
12:45.52ManxPowerJT: Like Jason or Freddie -- it can never die.
12:46.22lirakiswhat is this "thing"
12:46.46ManxPowerlirakis: the commercial version of Asterisk
12:46.56lirakisah
12:47.14lirakisasterisk business edition.. got it
12:47.53Uatec_of course it still exists
12:50.16*** join/#asterisk Fl1p (n=david@port-83-236-208-174.static.qsc.de)
12:50.22JTheh
12:50.39JTUatec_: even thought no-one uses it :)
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12:53.49*** part/#asterisk HaMYaI (n=LAMER@ppp-58.8.3.41.revip2.asianet.co.th)
12:54.17Uatec_i do
12:54.39Uatec_i'm about to buy two more licences
12:55.40Uatec_lol
12:55.52DavieyUatec_: why?
12:56.18*** join/#asterisk kv0s (n=kv0s@p4FD23918.dip0.t-ipconnect.de)
12:56.45Uatec_our customer wants their file server to use external USB hard disks for some reason
12:56.45Uatec_i cannot imagine why
12:56.45kv0sHi!
12:56.45Sci_05morning all
12:56.45Uatec_Daviey, cos i'm selling two asterisk implementations
12:56.47pj_portability.
12:56.55kv0sI've just started to play with music-on-hold.
12:57.17kv0sCreated a queue with moh (default) ... call the queue! Greate, the asterisk default music plays ...
12:57.37DavieyUatec_: what does ABE bring that vanila + asterisk-gui doesn't give?
12:57.44Uatec_support
12:57.44DavieyOther than the nifty folder?
12:57.48kv0s... create a own playlist/mohcategory and added some mp3s .... changed queue to play my new playlist ...
12:58.26kv0s... call the playlist and nothing to hear. no noise at the waiting time ... any ideas? my logs don't produce any error!
12:58.34kv0sor i can't see anything .. ,-)
12:58.54ManxPowerkv0s: stop asterisk, killall -9 mpg123 start asterisk
12:59.05JTUatec_: have you found the support worthwhile?
13:00.10kv0sManxPower: Why? ps -eaf displays no runnign mpg123!
13:01.00Uatec_yes i have
13:01.11Uatec_but only because my boss insists on gettin the b410p
13:01.18JTseems like a waste to me, but okay
13:01.19JTheh
13:01.23[TK]D-Fenderkv0s: pastebin your musiconhold.conf and this place you configured your custom playlist, etc.  Then include an "ls" dump of your MoH files
13:01.25[TK]D-Fender~pb
13:01.26jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
13:01.27Uatec_yes, well it seems like a waste of time to me
13:01.28[TK]D-Fender^^^^^^^^^^^^^^^^^^^^^^^^^
13:01.38Uatec_but we can always put a mark up on the price of ABE
13:03.33ManxPowerkv0s: Then how are you playing mp3 files in Asterisk?
13:04.01[TK]D-FenderManxPower: Trust--
13:04.15[TK]D-FenderManxPower: I'd wait on the evidence...
13:04.53kv0s[TK]D-Fender: http://pastebin.com/m2647e5c5
13:05.10kv0sManxPower: I think with mpg123. But no running process ...
13:05.18kv0s... mpg123 is installed on my machine.
13:05.40*** join/#asterisk myiagy (n=myiagy@201.56.113.74)
13:05.49*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com)
13:05.49[TK]D-Fenderkv0s: And now the * CLI output of "show modules"
13:05.56ManxPowerkv0s: go to your MoH directory, run this command mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 * > /dev/null
13:06.00ManxPowersee if you get any errors
13:06.08[TK]D-FenderFORGET MPG123
13:06.20[TK]D-FenderYour config is using NATIVE, not MPG123
13:06.39ManxPowerAh, OK.
13:06.48ManxPower[TK]D-Fender: his config also looks rather GUIish
13:06.59[TK]D-Fenderkv0s: And I'd like to see if you even have SUPPORT for MP3's
13:07.05kv0sim added it to the end ... http://pastebin.com/d74a36b68
13:07.07zeeeshis it possible to use meetme.conf without using zap ?
13:07.13[TK]D-FenderManxPower: I'll let that appearance slide for the moment.
13:07.17JTzeeesh: no.
13:07.24ManxPowerzeeesh: Without zaptel drivers?  No.  Without a zaptel card?  Yes.
13:07.34JTmeetme = zaptel conferencing
13:07.37kv0sMhm.
13:07.38[TK]D-Fenderzeeesh: If you want to use it to take up a cluster of disk space sure.. that about sums up its calue...
13:07.52JTit actualy conferences at the kernel level, which is a bit silly
13:09.20DavieyAIUI it get's it's timing from the kernel clock.. That's why you need a zaptel card / ztdummy
13:09.35Daviey-- drivers loaded into the kernel
13:09.50JTDaviey: it pulls conferencing up to the kernel level afaik
13:09.57JTit's more than timing
13:10.04DavieySomebody did make an alternative to meetme that doesn't require kernel modules; but i haven't tried it
13:10.16ManxPowermeetme uses the audio mixing features of zaptel
13:10.40[TK]D-Fenderkv0s: You do not have format_mp3.so
13:10.41Davieyi don't know for certain either way.. maybe it's for timing the audio mixing :)
13:10.53[TK]D-Fenderkv0s: You are therefor unable to play MP3's via *.
13:11.03JTDaviey: no, it mixes with zaptel
13:11.04[TK]D-Fenderkv0s: You need to install "asterisk-addons"
13:13.36*** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil)
13:14.25kv0sMhm.
13:14.40kv0s... and i thougt i've read the documentation ...
13:15.06*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:15.08ManxPowerkv0s: for patent and licensing reasons Asterisk does not include mp3 support as part of it's base system
13:15.09kv0sare asterisk-addons a debian package or dl it from asterisk.org?
13:15.27tzafrirkv0s, both...
13:15.51tzafrirhmmm.... actually not really a package in Debian/main yet...
13:16.09*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
13:16.09*** mode/#asterisk [+o lmadsen] by ChanServ
13:16.50kv0sI've installed asterisk-bristuff as debian-package. can i install the asterisk-addons too?
13:16.54ManxPowerI wonder what percentage of people that download packages end up getting the source and building Asterisk anyway
13:17.04[TK]D-Fenderkv0s: if you are working with  a packaged * install, continue that way
13:17.22*** join/#asterisk flujan (n=flujan@201-42-99-5.dsl.telesp.net.br)
13:17.23[TK]D-Fenderkv0s: when/if you compile from source get the matching version that way as well
13:17.46[TK]D-FenderManxPower: He's BRI-stuff'd
13:17.55[TK]D-FenderManxPower: hook, line & sinker
13:18.04ManxPowerPoor sod
13:18.07tzafrirManxPower, the point is that it should not be "download packages". It should be "ask my packaging system to install asterisk".
13:18.25ManxPower[TK]D-Fender: the device state stuff in BRIstuff is the only thing that I want.
13:18.31lirakistzafrir: i disagree
13:18.35tzafrirWorks reasonbly well on Debian, Gentoo and FreeBSD (last two: from what I heard)
13:18.51lirakistzafrir: asterisk is so simple to compile from source..
13:19.12polerinI had to compile it by hand even in debian iirc.  Could be wrong
13:19.18polerincan't remeber XD
13:19.19lirakistzafrir: and you arent forced into whatever your package system has
13:19.21kv0sMhm. Okay, on my next system i give a self compiled asterisk one try .. ,-)
13:19.23ManxPowertzafrir: The person that did the debian package forgot for put the sounds text file in with the sounds package (or forgot to put in /usr/share/docs)
13:19.25tzafrirlirakis, I have my reservasions there. But it is still not as simple as 'apt-get install'
13:19.34polerincouse I am using an old version now but hey whatever
13:19.39ManxPowerIt's the little things like that that make support of packages not easy here.
13:19.43tzafrirlirakis, but what I was saying was different:
13:19.44[TK]D-FenderManxPower: there is a 1.4 patch that far outstrips it.
13:19.55[TK]D-FenderManxPower: You're going to have to convert shortly....
13:20.07ManxPower[TK]D-Fender: I wonder if Digium ever managed to upgrade their corporate PBX to 1.4....
13:20.15tzafrirIf you need to look in some place for some special packages, then maybe they don't really match your distro well (and you can't really know that)
13:20.20[TK]D-FenderManxPower: That was a few WEEKS ago
13:20.29tzafrirAnd you also need to do some extra , unnecessary, work
13:20.33ManxPower[TK]D-Fender: When Digium Corporate PBX and ABE are running 1.4 then I will consider it production quality.
13:20.41ManxPower[TK]D-Fender: time flys when you are working your ass off.
13:20.42[TK]D-FenderManxPower: they ARE on 1.4
13:20.44[TK]D-Fender^^^^^^^
13:20.51ManxPower[TK]D-Fender: BOTH?
13:20.58tzafrirlirakis, and hence, the packages should be part of the distributions, and not "downloaded from he internet"
13:21.09[TK]D-FenderManxPower: ABE I'm unsure of...
13:21.17[TK]D-FenderManxPower: that was NEVER a point of interest for me ;)
13:21.19jcolpthe version of ABE based off of 1.4 hasn't been released yet, but it's being worked on
13:21.20ManxPowerMy users keep saying "our old PBX didn't make us pay a consultant to upgrade it every year"
13:21.41polerinheh
13:21.54ManxPower[TK]D-Fender: me neither, but it occurred to me that if ABE is on 1.4 that is a pretty ringing endorsement of 1.4
13:22.06DavieyManxPower: "but did your old pbx rock" ;)
13:22.08[TK]D-FenderManxPower: and did their old PBX not have some standarrd maintenance charge to do so?
13:22.31ManxPower[TK]D-Fender: just adds/changes/moves and Asterisk needs consultants for adds/changes
13:22.52SuurMyyhi?
13:23.04lirakistzafrir: .. well if you are saying (and it sounds like you are) "i dont know where my distro stores .. lib files... so i should get a preconfigured package that is setup for my specific distro that already sas where libs are" .. i sort of understand your point.. but also believe you should know where your lib's are stored..
13:23.10DavieyManxPower: you charge per task or a retainer?
13:23.18JackEStormManxPower: I think 1.4 is a long ways from being near as stable as 1.2 ....I just had to roll a 1.4 install back to 1.2
13:23.22ManxPowerAnd since ABE goes thru some "real" testing I would assume that the worst bugs would be fixed (and digium says ABE fixes are ported to the open source Asterisk)
13:23.35ManxPowerJackEStorm: Yours is a very common story.
13:23.36lirakistzafrir: also having stuff in bizzare places preconfigured by your distro .. and un known to you.. makes for very difficult support/trouble shooting
13:23.46ManxPowerDaviey: a little of both
13:23.53polerinlirakis: like the sound files :P
13:24.01ManxPowermostly I charge by the hour, just like a high class whore.
13:24.22Davieyhmm.. no 5 min jobs then?
13:24.23tzafrirlirakis, having Asterisk putting stuff in bizzare place unlike other programs on my system make it more difficult to troublshoot as well
13:24.28ManxPowerlirakis: if you don't know your distro you are screwed no matter what you do.
13:24.36kv0sMhm. What does format_mp3.o? Is mpg123 used from format_mp3.o?
13:24.38*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
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13:24.44JackEStormManxPower: my analog system seems to be working fine, but the system with the PRI in it, had too many issues.
13:24.44*** part/#asterisk tj_d (n=tj_d@mail.ninjamaster.com)
13:24.48lirakistzafrir: if you download and install from source.. you .. or some one here will know where it is
13:24.52ManxPowerDaviey: Sure, but they still get billed for at least 30 mins.
13:24.52*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
13:24.52tzafrirHaving Asterisk break the convensions followed by other programs make it bad behaving as well
13:24.57*** join/#asterisk tj_d (n=tj_d@mail.ninjamaster.com)
13:25.06tzafrirlirakis, actually: no.
13:25.10lirakistzafrir: if you install with XYZ distro and ABC package mangager.. no one .. may ever know
13:25.39tzafriryou know how to follow a 'make install' proedure. That still does not mean you know where things went to
13:25.40DavieyManxPower: pm?
13:25.49lirakistzafrir: in what way does asterisk break convention from other programs?
13:25.53ManxPowerDaviey: as long as your are not looking for free personal support, sure.
13:26.02Davieyheh
13:26.10*** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com)
13:26.14tzafrirlirakis: a classic example here is third-party modules in /usr/lib/asterisk/modules
13:26.15lirakistzafrir: my point was .. if you use the source.. you have the same config as people here.. who use the source
13:26.36tzafrirbut I have a different config than the people on #debian
13:26.37Kattymorning lovables!
13:26.44cpmmorn'n
13:26.47kv0sMhm. svn-checkout works, but make clean doesn't ..
13:26.50kv0s... svn checkout http://svn.digium.com/svn/asterisk-addons/branches/1.2 asterisk-addons
13:26.54lirakistzafrir: .. and its much more likely that other people here have compiled from source .. than.. some random distro pm combo
13:26.57tzafrirNot to mention those who do use packages
13:26.59kv0s... need i the asterisk sources too?
13:27.19Kattymy doggy has been tugging when we walk, cause he hasn't learned he needs to walk by me... and it's making a really BAD blister.
13:27.26tzafrir(getting the source for a package: apt-get source asterisk)
13:27.31Kattyany one know how to fix my puppy? :<
13:27.44cpmyup
13:28.10cpmit's called a 'halti'
13:28.12lirakistzafrir: .. does that actually get the tarball from the digium site..? or is it a non-vanilla source that has been setup for your distro?
13:28.25cpmKatty, http://www.allourpets.com/htmls/halticollar.shtml
13:28.27tzafrirno.
13:28.44*** join/#asterisk bacs (n=bacs@flunge.gladserv.com)
13:28.56tzafrirIt should have been the tarball from digium and a patch with the extra files needed to build a package.
13:29.06*** join/#asterisk michael-i (n=michael-@141.41.40.55)
13:29.31tzafrirlirakis, in practice the tarball needs a few fixes to even reside on Debian's servers in the main repository
13:29.50Kattycpm: do you think that works better than a chokechain?
13:30.02Kattycpm: which is what my co-workers are suggesting.
13:30.03cpmKatty, no comparison
13:30.10Kattyhmm.
13:30.20cpma choke chain is a contest of wills, a halti is completely passive
13:30.23lirakisKatty: i wouldnt do choke chain... dogs are a lot like people.. they learn through repetition
13:30.25kv0stzafrir: Mhm. Now i've several asterisk-directories but nothing named asterisk. the svn-checkout seems to need a asterisk directorie.
13:30.32kv0sres_config_mysql.c:44:29: error: asterisk/module.h: Datei oder Verzeichnis nicht gefunden
13:30.36Kattylirakis: okay.
13:30.36kv0sfor example
13:31.03Kattythat's good enough for me (=
13:31.07tzafrirkv0s, what exactly are you doing? building from the pkg-voip svn repository?
13:31.09Kattycpm: what kinda doggies did you handle?
13:31.11lirakisKatty: yeah .. especially if its a puppy... choke chain isnt needed.  If you had a grownup irish wolf hound... maybe then ;)
13:31.23Kattylirakis: welllll he's 8 months.
13:31.36Kattylirakis: so not really a grown up... but kinda.. but still very much puppy (=
13:31.39lirakisKatty: but is it an irish wolf hound?
13:31.41lirakislol
13:31.46Kattylirakis: no, he's a german shephard.
13:31.52lirakisthose things are like horses (wolf hounds)
13:32.02kv0sI'll try to install asterisk-addons ... documented at http://www.voip-info.org/wiki/index.php?page=Asterisk+addon+asterisk-addons
13:32.04Kattyi like big doggies.
13:32.08Kattyif you're gonna have a dog, have a dog ;)
13:32.12Kattywith the exception of yorkies...
13:32.20Kattyyorkies are just too cute.
13:32.27cpmKatty, while on active duty I worked (very closely) with 2 shutzenhund (shepards) and 1 great pyrenees
13:32.35lirakisKatty: seriously.. i live in manhattan.. so many little dogs in strollers and other wierd stuff...
13:32.43lirakisits messed up
13:32.44Kattycpm: oh wow, pyrenees!
13:32.47Kattycpm: those are HUGE
13:32.58Kattylirakis: oh my, strollers eh?
13:33.08Kattylirakis: but still i can understand a small doggy in a city.
13:33.22Kattylirakis: if i end up moving to vegas, i'm going to have to seriously thing about taking jager.
13:33.23lirakisKatty: yeah.. lots of mental cases.. and wierd old ladys with way too much money and no kids
13:33.44cpmsince then, (it's been over 20 years) I work with other folks dogs. I'm currently dog-unemployed :) Close friends are 2 working SAR dogs who are border collies, and my best ole bud, a working stud black lab just died a few months back
13:34.03*** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:34.24Kattycpm: well i'm sure i'll have questions for you, if you don't mind.
13:34.28Kattycpm: this is my first doggy.
13:34.42cpmtreat him like you would any friend, and ya'll be fine.
13:34.48lirakiscpm++
13:34.52Kattycpm: got a shepherd cause they're really smart, and don't need a /whole/ lot of training.
13:35.00cpmdogs are people too
13:35.08Kattyi do treat him like a friend.
13:35.10dominic1anybody know if the Avay Integral T3 IP is using SIP?
13:35.15Kattyexcept when we walk :/
13:35.23Kattycause then i have to keep yanking on his collar. *sigh*
13:35.26datachomperKatty, Did you buy him as a puppy?
13:35.35Kattydatachomper: no, he was adopted out of an animal shelther.
13:35.43Kattydatachomper: at 6 months.
13:35.56lirakiscpm: but at the same time.. they are dogs.  There is no sense in humanizing dogs... they are dogs.. and do dog things.  I think they just need to be socialized and worked with like people... not .. treated in every way like people
13:36.13datachomperKatty, I /highly/ recommend this book http://www.amazon.com/Good-Owners-Great-Brian-Kilcommons/dp/0446516759
13:36.17Kattylirakis: well i don't treat my dog exactly like a person...
13:36.26Kattylirakis: but.. he sits on the couch with me, and naps in bed with me :P
13:36.39Kattylirakis: he's pretty much always right on my heel
13:36.59lirakisKatty: hmm.. well  id make sure you take him to dog parks etc.  see other dogs..
13:37.06Kattylirakis: oh?
13:37.17Kattylirakis: there are dogs around our neighborhood when we take our walks.
13:37.32lirakisKatty: .. i mean.. think about it in reverse.. if you never saw anyone else.. and just followed around one person/animal all day.. wouldnt it make you kinda mental?
13:38.10lirakisright
13:38.11lirakislol
13:38.12Sci_05:-D
13:38.39Kattyyou're right.
13:38.44Kattyjager just sees us... and the ferrets.
13:38.57datachomperActually, the best thing you can do is go to a weekend obedience class. They are tons of fun for an hour, you get to hang out with a bunch of other dog owners and /really/ teach your dog how to behave around other people and dogs.
13:39.04Kattyguess we need to start having friends over and stuff.
13:39.17*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
13:39.22Kattyi wouldn't trust him in an obedience school just yet.
13:39.25Kattyjager has a few problems.
13:39.56datachomperKatty, that's exactly why you should go
13:40.00*** join/#asterisk gzero (n=gzero@81.175.82.2)
13:40.04Kattydon't they take your dog away from you at obedience school?
13:40.18Kattycause jager has abandonment problems.
13:40.24Kattyand... that would just be horrible for him.
13:40.25datachomperKatty, No! You are there with your dog, they train you to train your own dog
13:40.31Kattyoh, okay
13:40.57Kattywe have a place out here called Happy Tails that two of my co-workers take their doggies to.
13:41.41*** join/#asterisk NirS (n=Nir@87.68.170.190)
13:42.06Uatec_dammit
13:42.29cpmlirakis, agreed. However, dogs are more like folks than many dog owners, esp American owners accept. When in doubt, I trust the dog.
13:42.29kv0sMhm. surprise surprise ... compiling asterisk-addons the following error occours: configure: error: termcap support not found
13:42.34Uatec_neither machine i have to put asterisk on have PCI-E so i can't get a sanguma
13:42.44kv0swhere i can get termcap? what is termcap?
13:43.04*** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk)
13:43.13cpmlirakis, example, do *you* like being tied to stake all the time? or left locked in the basement all day? Well neither do they. :)
13:43.32tzafrirgee, doesn't this thing come with a BuildDepends?
13:43.35Kattypeople do that to dogs?!
13:43.41*** join/#asterisk MindTheGap (n=iote@mail.lpj.com.br)
13:43.48Kattythat's horrible :<
13:43.48tzafrirapt-get install libncurses-dev
13:43.50cpmpeople do much much worse
13:44.03kv0sOh.
13:44.08coppiceeat them, and then they won't suffer
13:44.14lirakiskv0s: termcap is an old unix terminal capability description utility/file
13:44.21cpmcoppice, noted.
13:44.23Kattycoppice: i think my doggie would eat you first.
13:44.23cpm:)
13:44.29kv0sthanks.
13:44.33lirakistzafrir loves apt-get .. lol
13:44.35Kattyi was so proud of jager the other night.
13:44.50Kattya group of guys was walking towards the house...
13:45.03Kattyand they didn't exactly look so... well manner, if you get my drift (=
13:45.14MindTheGapcreativx, im sorry i asked you a question yesterday but had to leave urgerntly...
13:45.15Kattyjager perked his ears up, and sat right down, and watched them the whole time.
13:45.38Kattythey were walking really close to the yard, and one of them stepped on the yard and jager barked at them.
13:45.48Kattyso they moved further away, and jager just sat there some more.
13:45.57*** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com)
13:46.10NOT_guruanyone here try any of the "802.11g wireless skype phone" devices with asterisk?
13:46.21lirakishas anyone here done asterisk training .. of anykind.. or just self taught?
13:46.36lirakisjust curious
13:46.38kv0smhm.... svn checkout 1.2 + compiling seems to be really easy ... i don't know why i've used the debian packages ... ???
13:46.38Kattylirakis: [TK]D-Fender and Hmmhesays taught me most of what i know
13:46.54Kattylirakis: and i blog everything (=
13:47.15datachomperI don't actually know anything about asterisk, but my bosses think I do. So it's all good.
13:47.17NOT_guruif I may be so bold... Katty  where do you blog this stuff?
13:47.25lirakisdatachomper: lol
13:47.27NOT_guruand do you mind putting out the link
13:47.36NOT_guruI love resources
13:47.58*** join/#asterisk ashd (n=ashleydr@user-194-248-151-83.e7even.com)
13:48.08KattyNOT_guru: 42ndgeekstreet.blogspot.com
13:48.34Kattydatachomper: yeah, same here :P
13:48.37lirakisKatty: .. yeah ive learned lots by being here too.. and from playing with my own pbx.. but im getting more into working with * for work .. and id like to get a strong foundation and more indepth understanding of all the stuff
13:48.44NOT_guruwhy thankyou Katty.. I do appreciate it
13:48.54KattyNOT_guru: (=
13:49.08KattyNOT_guru: i'm doing a new set of posts for this sangoma t1 card i just ordered.
13:49.32*** join/#asterisk saftsack (n=saftsack@pD9E07E53.dip.t-dialin.net)
13:49.34KattyNOT_guru: starting today, actually :P
13:49.35NOT_guruoo  that could be wonderfull as I have not rolled out a box with anything but analog cards
13:49.35Kattysaftsack: :>
13:49.47KattyNOT_guru: yeah, i'm in the same boat.
13:49.51KattyNOT_guru: this is my first t1 card!
13:49.54mishehuKatty: the sangoma cards you are going to be setting up - do they have the octastic hwec?
13:49.55saftsackhi
13:49.57saftsackwhats going on?
13:50.04NOT_guruand I am trying to convince the company I am with to move off an ancient phone system to a asterisk box
13:50.05Kattymishehu: the /what/?
13:50.10DavieyKatty: I look forward to the sangoma stuff - does that include the echo stuff?
13:50.17mishehuKatty: HWEC == hardware echo canceler
13:50.24KattyDaviey: yes, the card has echo cancelation on it, if that's what you're asking
13:50.37Kattymishehu: yes'r.
13:50.54DavieyKatty: which model? I'm thinking of getting the 4 port one
13:51.03KattyDaviey: ours just has 1 port on it.
13:51.05mishehuKatty: if you plan on doing any faxing over the line, get wanpipe 3.1.3
13:51.10KattyDaviey: this is a fairly small business (=
13:51.14Kattymishehu: no, no faxing.
13:51.18Kattymishehu: yet, anyway (=
13:51.44DavieyKatty: I need 60 channels (and maybe more), so I thought i would get the 4 channel one for future upgrades
13:51.53Daviey*4 port
13:51.58Kattymishehu: my company's still gets all concerned about 911 stuff, so we keep the fax on a pots line just in case.
13:52.04NOT_guruKatty:  you do blog alot =D... you just need a tech or asterisk topic  =P
13:52.12KattyDaviey: yeah, that'd surely take care of you (=
13:52.12NOT_guruerr  I need  LOL
13:52.21*** join/#asterisk RSAMan (n=a@dsl-242-47-211.telkomadsl.co.za)
13:52.21KattyNOT_guru: lol, sec, i'll get the post.
13:52.23RSAManhiya
13:52.39DavieyKatty: you don't get 911 on your PRI stuff?
13:52.46RSAMani have installed a fresh copy of asterisk . 1.4.. and created sip.conf and extensions.conf files
13:52.58RSAManhow do i let asterisk know that these files exist ?
13:52.58Kattyi should probably make that a 'topic' thingy
13:53.29RSAManam i making sense ?
13:53.30DavieyKatty: Can i ask why you chose Sang' over Digium?
13:53.37creativxim gonna keep 2 analog lines here
13:53.38KattyDaviey: http://42ndgeekstreet.blogspot.com/2007/04/asterisk-install.html
13:53.39creativxafter i port our numbers
13:53.42creativxand get rid of the gt's
13:53.45KattyDaviey: we have digium analog cards.
13:53.48creativxand im gonna have 1 fax, and 1 big ass red phone
13:53.50lirakisRSAMan: .. it already knows they exist
13:53.54kv0sGrml ... package x needed by package y and package z1 needed by package z2 and these package needed by asterisk-addons ... whoooaa... i feels like downloading the whole world wide web .. ,-)
13:53.56KattyDaviey: and we got the sangoma cards because everyone kept ranting and raving about them.
13:54.00mishehuKatty: I'm not sure what the difference between 911 on a fax line and 911 on a T1 would be...
13:54.03KattyDaviey: and because Fonality uses them.
13:54.04lirakisRSAMan: to reload your configs do "asterisk -rx reload"
13:54.05DavieyKatty: heh
13:54.07RSAManlirakis: how do i reload them then ?
13:54.11mishehuKatty: as they're both part of the PSTN
13:54.12Kattymishehu: nothing, probably.
13:54.12RSAManta
13:54.17Kattymishehu: but my boss doesn't really get it (=
13:54.47kv0sand compiling need so a long time on my box .. :-(
13:54.57KattyDaviey: that post is kinda messy.
13:55.20DavieyKatty: pinky!
13:55.50Davieywow you heavily document
13:56.42Davieyeeek.. you like emacs!
13:56.45Kattyyes, yes i do.
13:56.46Kattyyes, yes i do.
13:56.53Kattyi document because my memory is awful.
13:57.07Kattyi am /not/ a very well organized person, and my blog is basically where i spill my chaos.
13:57.26Kattyalso it's there in case i move to vegas, and some poor university student gets hired to handle her job.
13:57.31mishehuKatty: here's a disc labeled "memtest86" ...
13:57.38Kattymishehu: :>
13:57.40Kattymishehu: i have one of those :P
13:57.51mishehuKatty: did it find any errors?  ;-)
13:58.18Kattymishehu: it's still running ;)
13:58.33mishehuKatty: good to hear
13:58.59datachomperHoly crap, federal agents raided 30 houses for using xbox mod chips.
13:59.08mishehu[TK]D-Fender: ewww, no core dumping in public
13:59.09mishehuthat's nasty
13:59.47mishehudatachomper: yeah, the USA is the new nazi state it seems.  corporations > citizens
14:00.09datachomperMaybe i'll start a llc for myself :OP
14:00.25KattyDaviey: i also like documenting. i'm weird ^_-
14:01.00Davieytis great.. where would we be without docnuts like you?
14:01.13KattyDaviey: at the milk bar
14:01.16*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)
14:01.47*** join/#asterisk lukketto (n=lukketto@host171-155-dynamic.10-87-r.retail.telecomitalia.it)
14:02.01KattyDaviey: there, i updated my labels at the top with Geekery.
14:02.06Davieyyah
14:02.07KattyDaviey: now you can find all the geeky ones.
14:02.20DavieyKatty: I'll rss sub to that!
14:03.04Katty(=
14:03.53KattyDaviey: the new posts will be mostly next week.
14:04.04KattyDaviey: depending on how brave i'm feeling :/
14:05.35*** join/#asterisk mog (i=mog@nat/digium/x-84dbbc16f5351be4)
14:05.35*** mode/#asterisk [+o mog] by ChanServ
14:07.48DrAk0damn skype and gizmo takes around 2 kB/s in and 2kB/s out for call while asterisk take 10kB/s in and 10 kB/s out
14:08.08[TK]D-FenderDrAk0: Stop using G.711 then
14:08.19DrAk0[TK]D-Fender, which one should i use?
14:08.28[TK]D-FenderDrAk0: GSm is a good start
14:08.38[TK]D-FenderDrAk0: G.729 if you've got licenses
14:08.53DrAk0[TK]D-Fender, G729 is good enough?
14:08.53Daviey(or live in a country where software patents arn't valid) :)
14:09.55DrAk0if it give me the skype/gizmo quality and bandwidth consume i(my boss)'d pay for it.
14:10.34DavieyDrAk0: What country are you in?
14:10.50DrAk0Daviey, Venezuela and Spain
14:11.28MaliutaDrAk0: g729 is great, I have digium licenses at $10US/channel. But there are others out there
14:12.25*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
14:12.47DrAk0Maliuta, and what about phones and atas?
14:12.53MaliutaI wouldn't use it on a LAN (seeing as how the only advanatge over other codecs is bandwidth), but over smaller links it's fine.
14:12.53DrAk0is it transparents?
14:13.00sheldonhanyone know whether 1.4.10 is likely to be released in the next 5 days?  need the fix for http://bugs.digium.com/view.php?id=10289
14:13.21DrAk0Maliuta, we are not on a lan, actually we are having big bandwidth problems
14:13.33MaliutaDrAk0: yeah, I can use it on my cisco handsets, and asterisk transcodes from my tmd400p for ata
14:13.37kv0sGrml. Compiling asterisk-addons doesn't work ... see http://pastebin.com/d5b48d465 ... any ideas?
14:14.05kv0si've used svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2 to get libpri!
14:14.10DrAk0what about this? http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
14:14.26MaliutaDrAk0: most of the hardware phones I have looked at have g729 on them ... you're paying for a license in the phone cost even if you aren't using it
14:14.52DrAk0Maliuta, the license fee is monthly?
14:15.13[TK]D-Fenderkv0s: chan_zap.c:62:2: error: #error "You need newer libpri"
14:15.16*** join/#asterisk bakermd (n=bakermd@204.10.20.30)
14:15.22Sci_05DrAk0: no one time fee
14:15.34[TK]D-Fenderkv0s: You need to get the version of addons that belongs with your * version.
14:15.38sheldonh[TK]D-Fender: that really ought to give a version number :)
14:16.04bakermdI am using SIPDTMFMode rfc2833 then executing extensions reload, however Asterisk is still not listening for OOB signaling... any ideas?
14:16.41bakermdI have a really up-to-date version... see: Asterisk CVS-NHEAD-05/18/05-04:13:59     ;-)
14:17.04DrAk0and you can use g729 for some users and other codecs for others users right?
14:17.30Maliutayeah, thats what allow and disallow are for in sip.conf
14:17.37DrAk0yah
14:17.39Maliutaor iax.conf :)
14:17.43DrAk0well i might give it a try
14:18.24Katty[TK]D-Fender: :<
14:18.34sheldonhmaybe there's an asterisk-dev mailing list archive i could look at for a hint as to release schedule?
14:19.11[TK]D-Fenderkv0s: You shouldn't be using SVN.... that will get you the "latest".  Go check your version
14:19.15[TK]D-FenderKatty: Mew?
14:19.21Katty[TK]D-Fender: where's my hug? :<
14:20.08Kattyya!
14:20.14Kattyk, all better.
14:20.18MaliutaDrAk0: to see if it is really worth it (over g711-ulaw or g711-alaw) go and look at someones online bandwidth calculator
14:20.27*** join/#asterisk menace (n=deknos@unaffiliated/menace)
14:20.31*** part/#asterisk menace (n=deknos@unaffiliated/menace)
14:20.46MaliutaDrAk0: I am sure I looked at one that basically said difference of 1MB for a 1hr conversation
14:21.31[TK]D-FenderDrAk0: And it depends WHO you are communicating with as well.  If this is between 2 * boxes, you should be using IAX2 trunking + G.729
14:21.49Maliutathis is also true
14:22.17Maliutaand you really only need licenses if asterisk stays in the loop or has to transcode
14:22.34ManxPoweryou almost always eventually need to transcode.
14:22.34*** join/#asterisk TheCompWiz (i=user@wsip-68-109-200-102.mc.at.cox.net)
14:22.47TheCompWizanyone got a moment to explain asterisk syntax for a sec?
14:22.50KattyManxPower: (=
14:22.59Maliutaif the clients are able to re-invite and take asterisk out of the loop you may be able to get away with fewer licenses
14:23.01KattyTheCompWiz: just ask your question.
14:23.03DrAk0[TK]D-Fender, the problem is that we have opers spread over the world, making calls through the PBX that is on spain with TDMs, the problem is that sometimes the opers have bw problem and they cannot talk and the have to switch to skype
14:23.04*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)
14:23.07DrAk0that works for them
14:23.10TheCompWiz... trying to figure out what's the meaning of "=>" ... as opposed to "="
14:23.11[TK]D-Fenderand that often requires even MORE licenses in the end.
14:23.12DrAk0for bw issues
14:23.34DrAk0and well, my boss doesn't like the idea of using skype while his paying for a PBX
14:23.36KattyTheCompWiz: now i'm now expert... but i see = when you have a variable.
14:23.37[TK]D-FenderDrAk0: Are these "loose" phones directly connecting over the net to that PBX?
14:23.44KattyTheCompWiz: like an entry in sip conf.
14:23.55KattyTheCompWiz: type=friend, mailbox=1000@context
14:24.01KattyTheCompWiz: and i see the => in a list of commands.
14:24.02TheCompWizKatty... I would agree... however... in the asterisk.conf ... the variables are assigned using =>
14:24.03mockerProblem with voicemail storage using unixODBC.  Having large pauses when I do calls to the remote server to get voicemail... i.e. "Message 1<pause 5 seconds>Start of message"
14:24.10KattyTheCompWiz: 2000 => 1,Dial(mooo)
14:24.18TheCompWizKatty... I know that one.
14:24.21DrAk0[TK]D-Fender, most of them are softphone, and few IP Phone (crappys like allnet)
14:24.26[TK]D-FenderTheCompWiz: "=>" I've only seen used in "register", and "channel" lines.
14:24.30KattyTheCompWiz: ah, i've never dug around in asterisk.conf before.
14:24.34TheCompWizlook at asterisk.conf
14:24.35TheCompWizastetcdir => /etc/asterisk
14:24.37ManxPowerRemember, before 1.4 Asterisk did NOT have a RTP (SIP audio) jitterbuffer!
14:24.50ManxPowerTheCompWiz: in 1.2 and later they are pretty much the same
14:24.52*** join/#asterisk Won4him (n=Erik@ip24-251-157-86.ph.ph.cox.net)
14:25.06[TK]D-FenderDrAk0: What I mean is these phones you're referring to aren't going through 1 PBX to get to your OTHER one right?  They are jsut connecting direct  from say a residence?
14:25.21DrAk0[TK]D-Fender, yes
14:25.31TheCompWizManxPower... "pretty much" ... still implies there is a difference.   are they truely synonymous?
14:25.36[TK]D-FenderDrAk0: then SIP + G.729 it is.
14:25.38MindTheGaphello all, besides all help you guys are providing, i still got "hints" problems... let me describe it...i got different contexts for ppl according to their permissions. I got one context for internal calls, one for local (pstn) one for long distance and so one, one context includes the other and everyone ends up including [interno] wich is our internal context where everybody calls everybody within the company premisses. for better understanding, he
14:25.38MindTheGapres the dialplan ougoing contexts:  http://www.pastebin.ca/643040. Thing is if i include "hints" inside the [interno] context, even tough all other ontexts include it, it wont pass status from ppl landing on different contexts... maybe i need a whole new structure for the dialplan, i dont know... anyone?
14:25.48Kattywhen you're doing a linux distro install, and it asks you if you want to use a mirror...
14:26.03DrAk0[TK]D-Fender, which would be the other case?
14:26.05Kattythe mirror provides a complete list of packages for the distro, or a backup list of packages for just your set of cds?
14:26.13DrAk0[TK]D-Fender, that you was expecting
14:26.47[TK]D-FenderMindTheGap: that pastebin HAS NO HINTS
14:27.03MindTheGapyes, it hasnt...
14:27.18MindTheGapimagine the hints are inside [interno]
14:27.20MindTheGap:)
14:27.52KattyManxPower: when you're doing a distro install off a cd, and it asks if you if you want to use a mirror, do you know if that supplies the install a complete list of packages...or is it just a means of backup for the packages on the cd?
14:28.01TheCompWizso.... does anyone know if "=" is 100% interchangeable with "=>"  ... or is there a reason to use one over the other?
14:28.11*** join/#asterisk javar (n=javar@69.79.134.24)
14:29.02[TK]D-FenderMindTheGap: set "subscribecontext=[thecontext]" in their sip.conf entries and put them there.
14:29.24kv0s[TK]D-Fender: Grml. Okay. I'm followed the instructions for branches 1.2 at http://www.asterisk.org/developers/getting-started, but when i start make install at asterisk-addons i get the following error
14:29.29[TK]D-FenderTheCompWiz: You're thinking way too much.
14:29.50MindTheGapsubscribecontext =! context ?
14:29.55[TK]D-Fenderkv0s: You need the version that matches the rest of your install
14:30.09kv0s./mkdep -fPIC -I../asterisk -D_GNU_SOURCE     `ls *.c`
14:30.09kv0sapp_addon_sql_mysql.c:23:19: error: mysql.h: Datei oder Verzeichnis nicht gefunden
14:30.19[TK]D-FenderMindTheGap: "subscribecontext=thisisthecontextmyhintsarein"
14:30.34kv0sI've installed the debian packages 1.2.16 - so i think i can use branche 1.2? or not?
14:30.36TheCompWiz[TK]D-Fender.... am I?  (working on a dialplan editor... and trying to make sure it can interpret the conf files correctly.)
14:30.50[TK]D-Fenderkv0s: translate please.
14:31.09[TK]D-FenderTheCompWiz: I just told you the only 2 places I've seen "=>".
14:31.23michael-i"file or folder not found", i think the dev files for mysql aren't installed
14:31.25*** join/#asterisk shay|work (n=shay@unaffiliated/shay)
14:31.27shay|workhello folks
14:31.41*** part/#asterisk shtoom (n=shtoom@221-128-190-221.static.exatt.net)
14:31.43TheCompWiz[TK]D-Fender ... I've seen in both... and others.  I'm just trying to figure out which is correct where and why.
14:31.52*** join/#asterisk masus (n=tet@88.248.14.186)
14:31.57shay|workI have problem using the Asterisk to make calls from the SIP softphone to the PSTN
14:31.57JTthere are traditions
14:32.10JTbut = and => are interchangeable from all reports
14:32.15masushallo , kann mir jemand bei der installation der astersk addons helfen hab einige probleme damit
14:32.16shay|workI'm able to make a call, i can hear the other side from the softphone, but the other side of the line can't hear me
14:32.21*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)
14:32.28shay|workit's not a sound card configuration since I'm able to record sound
14:32.30[TK]D-FenderTheCompWiz: if you want to feel consistent just use "="
14:32.32masusich versuche es auf debian 4 zu installieren
14:32.53TheCompWiz[TK]D-Fender... can you also do exten=blah?
14:32.56[TK]D-Fendershay|work: Let me guess.... one end is behind NAT, right?
14:33.00TheCompWizor only exten=>
14:33.03[TK]D-FenderTheCompWiz: yes
14:33.06TheCompWizok.
14:33.10TheCompWizguess that answers that.
14:33.20[TK]D-FenderTheCompWiz: Although yes I tend to keep the "=>" in there too
14:33.20masusaber bei der instalation stand make clean ; make install , aber es geht einfach nicht so leicht
14:33.22shay|work[TK]D-Fender, The workstation with the softfone is NAT, on the same side as the Asterisk Server
14:33.33[TK]D-Fendershay|work: Go read this :
14:33.34[TK]D-Fender~sipnat
14:33.35jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
14:33.35shay|work[TK]D-Fender, so, they're both NAT but in the same network
14:33.35JTmasus: english please
14:33.48masusjes of course
14:33.56kv0sMhm. masus - congratulations. I'll try it since 60 minutes ... not really successful...
14:34.04masus:)
14:34.05TheCompWiz[TK]D-Fender ... that's fine.  seems a bit odd tho.  thanks for the tip.
14:34.14masusi want to install asterisk addons on debian 4
14:34.17shay|work[TK]D-Fender, thanks, i'll check
14:34.26mockerI wish asterisk-users wasn't so high-volume.
14:34.26masusbut i get always errors after make clean; make install
14:34.34kv0s[TK]D-Fender: Translate? Compile? Or try to write a better english? ;-)
14:34.35mockerThey should make another list for dCAP people or something.
14:34.40masussince 1 day i have tried everything
14:34.42x86why would Asterisk be dropping about 1 out of about 30 or so calls
14:34.44masuscan anybody help
14:34.57shay|work[TK]D-Fender, Asterisk as a SIP server behind nat, clients on the inside connecting to Asterisk
14:35.04shay|work5 Works - no NAT in between
14:35.09JTthat's just what we need, 10 thousand more mailing lists
14:35.17kv0smasus: I'll try the same since a few minutes. What do u do? How u installa?
14:35.19kv0smasus: I'll try the same since a few minutes. What do u do? How u install?
14:35.20kv0sit
14:35.30Fl1pI'm interested in using SRTP with Asterisk on SIP-Channels is it possible ?
14:35.32*** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com)
14:35.36JTFl1p: no
14:35.42masuskv0s ihave try some variables
14:35.49masusbut nothing
14:35.59kv0sWhat? Variables?
14:36.25kv0smasus: Have u installed the debian-packages or used the sources from asterisk.org?
14:36.27Fl1pJT : so what's a good Solution for securing my RTP Streams over SIP ?
14:36.43masusthere are no packages for addons
14:36.44JTFl1p: udp vpn i guess
14:36.46[TK]D-Fendershay|work: No audio between phones local to *?
14:36.49masusihave download the tarbal
14:36.53Won4himI'm getting an error when trying to use IMAP with *
14:37.01shay|work[TK]D-Fender, no audio on the softphone going out to PSTN
14:37.13masushave u get it work ?
14:37.15[TK]D-Fendershay|work: And what are you using for PSTN?
14:37.26shay|work[TK]D-Fender, an Astrikban-BRI
14:37.38[TK]D-Fendershay|work: which is what exactly?
14:37.48kv0smasus: See a few lines above ...
14:37.59Fl1pJT : But i use Asterisk and SIP+RTP for connecting various Channels Landline, Mobile and SIP .....
14:38.18JTFl1p: and? why do i care? :)
14:38.19shay|work[TK]D-Fender, a product from Xorcom that connects BRI Lines to the Asterisk PBX via a external box connected throught USB
14:38.27masuskv0s i have download the tar
14:38.29masusand
14:38.31masusone mom
14:38.36shay|work[TK]D-Fender, and it's recognized as a zaptel device
14:38.42[TK]D-Fendershay|work: can the phones talk to each OTHER fine?
14:38.46Fl1pJT : I thought you would know the current status of secure VoIP
14:38.46*** join/#asterisk hank (n=hank@leonardo.netwichtig.de)
14:38.48hankhi
14:38.52JTFl1p: i just told you it
14:38.53Won4himhi hank
14:38.53masushttp://pastebin.ca/643064
14:39.06JTthere is very little support for secured voip anywhere
14:39.16masuskv0s: please look at this http://pastebin.ca/643064
14:39.27shay|work[TK]D-Fender, I don't have other phones, just two SIP users configured and an extension to the outside
14:39.28TheCompWizJT ... the best options currently for secure voip... is using a vpn tunnel.
14:39.29Fl1pthe vpn solution yeah.... which works for near every Packet transport
14:39.47JTTheCompWiz: i just said that
14:39.48[TK]D-Fendershay|work: Can those 2 internal users call each other fine?
14:39.49masuskv0s:asterisk itself works fine
14:39.53kv0smasus: LOL. Same problem at my side.
14:39.57masus:)
14:39.59hankWe are running bristuffed asterisk 1.0 with snom phones. I need to configure one function key to establish a call redirection or forward to two other SIP accounts. any hints on how to do that?
14:40.03masuswhat we do now :D
14:40.11JThank: seriously, upgrade
14:40.26masussomwhere i have read to change sme lines in the makefile
14:40.42shay|work[TK]D-Fender, didn't try it yet
14:40.43masusCFLAGS+=../asterisk
14:40.47hankJT: i don't think we can. its an appliance by some strange company.
14:40.48shay|worki need another machine for that :/
14:40.49masusto the real path
14:41.03*** join/#asterisk [Mr_X] (n=mrx@78-59-18-15.ip.zebra.lt)
14:41.06JThank: you're playing with the appliance, all bets are off
14:41.08masusor mysql header files
14:41.24masusbut im a newbie on linux so i don't no where to get the header files
14:41.30masusfor mysql
14:41.37Fl1pTheCompWiz : VPN works for known Clients but i not when calling customers mobilephone and redirect them outside the vpn
14:41.48hankJT: sorry? what does that mean? if i was the one to make decisions we would not have bought that thing. but i am not. i just have to get this to work :-/
14:42.19[TK]D-Fendershay|work: Go try to isolate the problem then come back
14:42.27masuskv0s : http://www.voip-info.org/wiki/index.php?page=Asterisk+addon+asterisk-addons
14:42.35masuslook at this one maybe u can do it
14:42.40TheCompWizFl1p ... probably because you're doing a sip-to-sip call.... and the end party does not support encryption.
14:42.48JThank: i'm saying you should be able to upgrade if you're configuring it
14:43.16kv0smasus: i think asterisk-addons need the asterisk sources
14:43.27*** join/#asterisk friedrich| (n=friedric@e177254095.adsl.alicedsl.de)
14:43.35hankJT: i'll forward that advice to the one responsible. so it's possible with some version >1.0?
14:43.36*** join/#asterisk codazoda (n=Joel_Dar@207.155.179.56)
14:43.40masusbut i have installed asterisk
14:43.43masusit's working fine
14:44.03masusmaybe we can find a debian package for addons
14:44.11*** join/#asterisk SwK (n=SwK@63.96.55.2)
14:44.16JThank: yes, the latest bristuffs are based on 1.2
14:44.27masusJT: Boss maybe u can help us :)
14:44.27JTand there's a 1.4 alpha, not for production yet
14:45.28hankJT: i know. but you have not said anything useful regarding my question yet... and thats actually everything i really care about. just saying 'it won't work with 1.0. you need at least 1.2' would be enough.
14:46.07JThank: sounds like a snom configuration issue
14:46.13[TK]D-Fenderhank>We are running bristuffed asterisk 1.0 with snom phones. I need to configure one function key to establish a call redirection or forward to two other SIP accounts. any hints on how to do that?
14:46.29[TK]D-Fenderhank: your phones HAVE a forward feature ON THEM.  Go read the maual.
14:46.49[TK]D-Fenderhank: And if you want * to do ti for you, then its your dialplan.  Go code it in.
14:46.50KattySwK: :>
14:46.53Fl1pTheCompWiz : we connecting customers with initiated calls using a public voip provider
14:47.18JTFl1p: what on earth makes you think many will support srtp?
14:48.10hank[TK]D-Fender: They have a forward feature but afaict not for two numbers simultaneously...
14:48.32JTso forward it to an extension that calls 2 phones at once....
14:48.33[TK]D-Fenderhank: then di it in your dialplan.
14:48.53hankJT: yep i just had that idea too...
14:48.57hankthanks
14:49.20*** part/#asterisk codazoda (n=Joel_Dar@207.155.179.56)
14:49.24kv0smasus: download the asterisk sourcecode too ...
14:49.30*** join/#asterisk tecnico (n=tecnico@24.96.146.69)
14:49.40Fl1pJT : We just wanted to make our calls safe as possible
14:49.55JTFl1p: try and bring logic into the equation
14:50.02JTyou are connecting to the pstn
14:50.11JTthere goes a lot of your safety there
14:50.13*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
14:51.54*** join/#asterisk Op3r (n=Op3r@121.97.251.114)
14:52.24masuskv0s i have download it
14:52.31masuskv0s and after ?
14:52.48SwKKatty, :>
14:52.56*** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-5bf224ab3ce41d7b)
14:53.03*** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-234-206.dsl.irvnca.pacbell.net)
14:53.05Fl1pJT : yeah you're right i become a bit confused with all those SIP Signalling, RTP Streams over * or P2P calls over Voip Provider, SIP-SIP and a boss who wants all this told as easy as possible what is possible and what not....
14:53.09BSD_Techok morning
14:53.23SwKif you have a L3 direct connection and want to sell some minutes msg me
14:53.28BSD_Techanyone here using sqlite and asterisk
14:53.44*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)
14:53.46*** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136)
14:53.47JTFl1p: even if your voip provider did support it, which is rare, you're connecting to the pstn, that's not that secure, especially against legal interception
14:54.36kv0smasus: See http://www.asterisk.org/developers/getting-started ... but as written a few lines before, it's not very successfull on my side
14:54.45masusok
14:55.42*** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net)
14:56.26*** part/#asterisk hank (n=hank@leonardo.netwichtig.de)
14:57.01kv0s[TK]D-Fender: Sorry, asking it again. I've downloaded with svn the actual branch via svn (1.2). asterisk, asterisk-addons, zaptel, libpri ... if i start with make install at asterisk-addons, it fails.
14:57.14*** join/#asterisk lukketto (n=lukketto@host171-155-dynamic.10-87-r.retail.telecomitalia.it)
14:57.17kv0serror: ./mkdep -fPIC -I../asterisk-1.2.13~dfsg -D_GNU_SOURCE     `ls *.c`
14:57.17kv0sapp_addon_sql_mysql.c:23:19: error: mysql.h: Datei oder Verzeichnis nicht gefunden
14:57.35[TK]D-Fenderkv0s: You need to recompile libpri, zaptel, asterisk, addons.  In that order
14:57.40masushave u install mysql
14:57.46kv0sMhm.
14:57.59[TK]D-Fenderkv0s: Sort of an obvious requirement there, no?
14:58.23kv0s[TK]D-Fender: Mhm. I think i've made a mistake at the compile-order ... mhmm... grml.
14:59.19kv0s[TK]D-Fender: Because i've a running system, that works! it's enoug to start with "make clean; make;" without install?
14:59.38JTnot if you want it to have any effect
14:59.48JTwhat's the point of compiling then not installing
14:59.53[TK]D-Fenderkv0s: No, install is necessary..
15:00.18kv0s[TK]D-Fender: Mhm. Destroys "make install" my running configuration?
15:00.34NOT_guruI asked earlier, but the channel was still sleepy
15:00.45[TK]D-Fenderkv0s: no, only "make samples"
15:00.48NOT_guruanyone here try any of the "802.11g wireless skype phone" devices with asterisk?
15:01.02kv0sAt the moment i'll install asterisk-addon and not the whole again?!?
15:01.09[TK]D-Fenderkv0s: But if you're using BRI-stuff, that does sorta imply you have BRI equipment, no?
15:01.33kv0s[TK]D-Fender: Yes! I've installed two hfc-s cards in my system.
15:02.04kv0sI'll add mp3-music-on-hold functionallity to my running asterisk ...
15:02.47JTNOT_guru: forget about that junk
15:03.42NOT_guruwell  this would just be a phone for the wife for around the house
15:04.06NOT_guruour corless phone died the other day so I figured I could just look
15:04.19JT1. it's wifi 2. it mentions skype
15:04.21JTbig no
15:04.29NOT_guruI just wasn't sure if these devices are locked to skype connections only or are SIP compatible as well
15:04.35datachomperDoes the agi command "say alpha" work as a text-to-speech engine?
15:04.44JTjust get a cordless phone
15:05.00datachomperOr does it spell out a string in the phonetic alphabet?
15:05.26*** join/#asterisk CM3_1_2_632 (n=CM3_1_2_@cm222-166-6-33.hkcable.com.hk)
15:05.28Qwell[]datachomper: it spells it out
15:06.02NOT_guruwell I do appreciate your input, have you used one personally. and if so which as there are many 802.11g phones out there at this point
15:06.17bakermdHow do you Force asterisk to use g.729 ? I have licenses from Digium registered in the box, and I have disallow=all allow=g729 and dtmfmode=rfc2833 in the sip.conf - yet the calls into the voicemail are negotiating alaw
15:06.30datachomperQwell[] Is there a command for interfacing with festival from an agi? Or do I need to manually route text-to-speech through festival in the agi script?
15:06.49Mercestesbakermd:  That's annoying.
15:06.58Qwell[]datachomper: When I used festival, I cheated, and used a system command to let the festival CLI command generate the file, and then did a Playback
15:07.10JTNOT_guru: look, my eyes are closing, i can't be bothered re-iterating the point again
15:07.18JTbed time
15:07.27NOT_guruI understand
15:07.37ManxPowerbakermd: If you disallow=all and allow g729 then all calls will be G729 *IF* the call matches that sip.conf entry.  If it does not, then the settings in [general] will be used.
15:07.41datachomperQwell[] Great, thanks bro
15:07.42NOT_gurujust trying to get what caused your great distain is all
15:07.52NOT_gurubut I truely do appreciate your answers
15:07.54NOT_guruthank you
15:07.58JTNOT_guru: it's common knowledge that all wifi sip phones are shit
15:08.05JTand skype is shit :)
15:08.07Qwell[]datachomper: You can get really slick, and do some caching based on the md5 of the string...
15:08.12bakermdOkay
15:08.12Qwell[]it worked really, really well
15:08.21*** part/#asterisk TheCompWiz (i=user@wsip-68-109-200-102.mc.at.cox.net)
15:08.23Qwell[]if you're having it say the same text over and over, you might as well cache it
15:08.42Mercesteshttp://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf
15:08.57JT~wifisip
15:08.58jbotWi-Fi SIP phones suck.  All of them.  HARD.  Some only slightly less than others...
15:09.05datachomperHmm, so when you output the wave, make the name the md5 cache, then do future searches, nifty
15:09.11Sci_05lol
15:09.15Qwell[]datachomper: exactly
15:09.31Mercestesbakermd, clicky that link, then look under "format"
15:09.31ManxPowerIf you want to do more than a little TTS, I would suggest Cepstral
15:09.46Qwell[]I was very happy with how it turned out.  The generation took maybe half a second, and if it was cached it was instant.
15:09.59ManxPowerMercestes: Wow!  You thought of something I should have thought of.  You're growing!
15:10.12Mercestes:D
15:10.17ManxPower..er...Growing up that is
15:10.19*** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com)
15:10.22Mercestesyay..
15:10.24Mercestesgrowing too.
15:10.30Mercestesyour praise makes me happy.
15:11.48Kattyhi Mercestes!
15:12.34ManxPowerKatty! You'll get lead poisoning!
15:12.45Qwell[]You don't know where that pencil has been
15:12.55Katty^_-
15:13.02datachomperWhat if it's a plastic bic pencil.
15:13.03ManxPowerQwell[]: Wasn't that the pencil I gave you?
15:13.08Qwell[]ManxPower: it was
15:13.24*** join/#asterisk luke-jr|work (n=luke-jr@adsl-76-194-177-177.dsl.ksc2mo.sbcglobal.net)
15:13.33Kattythe only type of pencil i'll use is a drafting pencil.
15:13.33ManxPowerAh.  Maybe I'd better not say anything else about it.
15:14.11MercestesHi katty!
15:14.26bakermdMercestes: Thanks
15:14.27Kattybout time :P
15:14.34Mercestessorry, I was working.  >.>
15:16.08*** join/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2)
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15:16.26|Rain|has anyone actually had any luck with PRI signalling over TDMoE?
15:17.00MercestesTDM over Ethernet?
15:17.06|Rain|yes
15:17.30ManxPowerMercestes: |Rain| I believe that it should work, but TDMoE is not really maintained before.
15:18.34Mercestes|Rain|:  I have a PRI card run from my asterisk server to a T1fax board running hylafax right now.
15:18.47Mercestesthat is, in all technicallity, TDMoE but it goes all of five feet.
15:18.54Mercestesand that's only because i looped the cable a few times.
15:19.01ManxPowerMercestes: It's not TDMoE unless you are using the TDMoE driver
15:19.07Mercestesoh...
15:19.31ManxPowerMercestes: It encodes the raw data into raw Ethernet packets.  It's not IP.
15:20.17ManxPowerIt is VERY low latency and keeps ALL the info that comes in on the T-1, but you can't route the packets.
15:20.23*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)
15:20.26|Rain|I'd definitely buy that it's not really maintained anymore, it doesn't seem to work terribly...  well.
15:21.10|Rain|but using e.g. e&m signalling seems to work fine while PRI signalling makes asterisk rather irritable
15:21.25*** join/#asterisk DaveCanoe (n=Dave@ool-182c60c9.dyn.optonline.net)
15:22.10Mercestes|Rain|, What are you seeing?
15:22.54*** join/#asterisk Dead-Bum (n=igli@pool-70-108-232-223.washdc.east.verizon.net)
15:22.54*** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) [NETSPLIT VICTIM]
15:22.54*** join/#asterisk HockeyInJune (i=HockeyIn@pool-141-155-136-189.ny5030.east.verizon.net) [NETSPLIT VICTIM]
15:23.02*** join/#asterisk markgreene (n=markgree@209.12.142.2)
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15:23.07|Rain|the pri_cpe side keeps repeating '== Primary D-Channel on span 1 up', while the pri_net side says nothing
15:23.07*** join/#asterisk gzero (n=gzero@81.175.82.2) [NETSPLIT VICTIM]
15:23.07senthoranyone here running asterisk with ultramonkey?
15:23.24Mercestes|Rain|  Hrm.  Describe your hardware setup.
15:23.34markgreeneHey guys. When I am installing asterisk 1.4.9 and the addons 1.4.2 , in which order do i install them in
15:23.35markgreene?
15:24.01masuslibpri --> asterisk --> asterisk-addons
15:24.04*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) [NETSPLIT VICTIM]
15:24.22markgreenemasus: thanks
15:24.22|Rain|when I attempt to place a call over the fake PRI, the pri_net side says 'NOTICE[20809]: chan_zap.c:8466 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2' and the pri_cpe side keeps repeating 'primary d-channel on span 1 up' once/sec
15:24.43MindTheGap<[TK]D-Fender>, I've put the hints inside [interno] context, updated every sip.conf user with subscribecontext=interno but core show hints still shows no action. its supposed to sho me who is on the phone, isnt it? it shows everyone as idle...
15:25.01ManxPower|Rain|: "ifconfig" and see if you see any errors on the interface
15:25.01*** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com)
15:25.38MindTheGap<[TK]D-Fender>, it also shows 0 watchers for every hint...
15:25.52|Rain|I have 2 boxes, one with an eepro100 and one with a natsemi dp83815, connected with a crossover cable
15:25.54|Rain|no interface errors
15:26.13[TK]D-FenderMindTheGap: Might be nice to see your configs and know what version you are running...
15:26.41|Rain|crap, need to run off for a bit
15:26.42*** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-234-206.dsl.irvnca.pacbell.net)
15:26.47[TK]D-FenderMindTheGap: and 0 watchers sorta tells me your phonesa ren't even trying to look at the hints.  you'lll have to reboot them as well
15:27.02|Rain|tbh I don't really trust that natsemi card that much and might try a different one
15:27.03Hmmhesaysugh why does asterisk put failed in the cdr disposition field on a cancelled call
15:27.05BSD_Techmornign what ver of sqlite is supported in asterisk
15:27.10Kattymister fender, are you uber busy?
15:27.10kv0s[TK]D-Fender: I don't understand it, i can use the default moh ... but nothing other... if i made systemrecordings, nothing will played - is that the same issue as playing mp3s?
15:27.30[TK]D-Fendermasus: Close, but you missed something...
15:28.07[TK]D-Fenderkv0s: Show me that you now have MP3 support, and have completely restarted *
15:28.24Katty[TK]D-Fender: i'll take that as yes.
15:28.29Katty[TK]D-Fender: i'll pester you later.
15:28.30[TK]D-Fenderkv0s: And re-pastbin your configs, etc
15:28.38MindTheGapok, im running 1.4x res_ldap trunk but not using it for realtime peers, yet. what config files shall i show you? extensions and sip?
15:28.42[TK]D-FenderKatty: fire away
15:28.57[TK]D-FenderKatty: didn't see the highlight so I missted your question.
15:29.33Hmmhesaysthe more I work with postgresql the more I LOVE IT
15:30.48*** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)
15:30.50[TK]D-FenderMindTheGap: both.
15:30.52brad_msswHmmhesays: until you need replication ....
15:31.02*** join/#asterisk irule (n=irule@189.164.47.106)
15:31.23kv0sMhm.
15:31.34Kattyohoh! look who it is!
15:31.37kv0s[TK]D-Fender: Sorry. It's not so easy for first time ...
15:31.52masusD-Fender : i don't use zaptel
15:31.59kv0s[TK]D-Fender: What is the best way? Completly recompile all things including patching bristuff?
15:32.12[TK]D-Fendermasus: Waht good does libpri do you without zaptel?
15:32.28kv0sMust i really RECOMPILE and REINSTALL my complete ASTERISK installation for only adding "sound / mp3" support?
15:32.31[TK]D-Fenderkv0s: What hardware cardsa re you using?
15:32.35masus:) i don't know :D
15:32.37Hmmhesaysbrad_mssw: what do you use for that?
15:32.44Kattyirule: :>
15:32.51kv0stwo isdn cards with hfc-s (colognechip)
15:32.51*** join/#asterisk shay|work (n=shay@unaffiliated/shay) [NETSPLIT VICTIM]
15:32.54shay|workhmm
15:32.56*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
15:33.07masusso i don't need to install libpri ?
15:33.12[TK]D-Fenderkv0s: Not if you pick the right version of addon's to compile and have the dependencies satisfied
15:33.13brad_msswHmmhesays: mysql has built-in replication which is easy to use ... last time I looked at postgresql's, it was external, and you had to have per-table triggers, couldn't do whole-db replication
15:33.14shay|worknow when I call the outside, other side just hears a noisy long beeeeeep
15:33.18shay|workany ideas what this can be?
15:33.21[TK]D-Fendermasus:  you = brilliant
15:33.25masus:D
15:33.25irulegood morning everyone, hi Katty :D
15:33.35bakermdOkay - so I am getting "No Compatible Codecs" when only g729 is enabled
15:33.35kv0sMy configuration runs well, the only thing i missing is the music-on-hold feature and sysmterecordnigs ..
15:33.35*** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net)
15:33.51bakermdbut if I do a show g729 I get 0/0 encoders/decoders of 49 licensed channels are currently in use
15:34.00[TK]D-Fenderbakermd: SIP debug + pastebin is your friend.
15:34.10[TK]D-Fenderbakermd: And your CONFIGS.
15:34.25bakermdcool
15:34.31bakermdwill do that in a bit then
15:35.05MindTheGap[TK]D-Fender, here it is http://www.pastebin.ca/643141
15:35.16kv0s[TK]D-Fender: Okay - again. Show version at console says: Asterisk 1.2.13-BRIstuffed-0.3.0-PRE-1v - so i need the asterisk-addon sources for build 1.2.13? But where can i find these old sources? *grml
15:35.29*** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) [NETSPLIT VICTIM]
15:36.47kv0sAt http://svn.digium.com/svn/asterisk-addons i can only find 1.2 - nothing 1.2.13 or others?!?
15:37.06kv0sMhm. I think must restart with a clean installation ... :-(
15:37.17kv0sonly for music on hold ... .-(
15:37.39masusi have do it twice today
15:37.42*** join/#asterisk smultron (n=lukas@cpe-66-69-197-171.austin.res.rr.com)
15:37.49[TK]D-FenderMindTheGap: What ver of *?
15:37.49markgreeneCan someone point me to a crash course on dialplans inside mysql?
15:37.53masusfor newbies the best solution is make a clean installation
15:37.55masusÝ)
15:37.57masus;)
15:38.04*** join/#asterisk Lucky7 (n=Adam@207.200.28.175)
15:38.04MindTheGap[TK]D-Fender, 1.4.x
15:38.12smultronasterisk requires mysql?
15:38.16Kattysmultron: no
15:38.18smultronoh
15:38.19Kattysmultron: it will log to a csv file.
15:38.23masusasterisk-addons requires
15:38.27smultronoh, ok
15:38.29Kattysmultron: some people just like it dumping into mysql (=
15:38.35[TK]D-FenderMindTheGap: Add "call-limit=100" and change to "type=peer" for your phones.
15:38.38smultroni guess so
15:40.03MindTheGap[TK]D-Fender, can you tell me what will it archieve?
15:41.23[TK]D-FenderMindTheGap: Required for presence in 1.4.  there were some wierd changes to chan_sip
15:43.05*** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl)
15:43.08*** join/#asterisk mirco (n=mirco@p54B25056.dip.t-dialin.net)
15:43.33*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
15:44.58*** join/#asterisk thansen|laptop (n=thansen@151.155.248.199)
15:45.14*** join/#asterisk kimosabe (n=kimosabe@189.175.44.143)
15:45.51kimosabewhat distribution do you all recomend for easy use and webinterface for configuration any one please
15:46.09smultronthere's a web interface for Asterisk?
15:46.39mockermental note, strace following forks can kill asterisk
15:46.41pigpensmultron, shit, I have been using text all this time?
15:46.42mocker;)
15:46.56smultronhehe
15:47.12kimosabesuposably theres somthing with a guie
15:47.15pigpenAll this time I could have been "point and clicking"
15:47.22shido6its good to use text first.
15:47.27[TK]D-Fenderkimosabe: LOL.  You are in the WRONG PLACE.
15:47.33kimosabeshido yes im familiar with the text also
15:47.41shido6when the gui fails you you can go in and make the features you need
15:47.43kimosabe13 yr experience in unix
15:47.56*** join/#asterisk saftsack (n=saftsack@pD9E07E53.dip.t-dialin.net)
15:48.08kimosabesantacruz operation and bsd
15:48.12datachomperIs there an equivalent "dialplan reload" on asterisk 1.2 ?
15:48.22shido6extensions reload ?
15:48.43*** join/#asterisk ELBunce (n=erik@kde/developer/bunce)
15:49.19*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)
15:49.31BSD_Tech<PROTECTED>
15:49.34Hmmhesaysthat sucks bridge collapsed in msp
15:49.58irulewhere is a good reference for agi speciffic programming?
15:50.06HmmhesaysI35 over the mississippi
15:50.06*** part/#asterisk lukketto (n=lukketto@host171-155-dynamic.10-87-r.retail.telecomitalia.it)
15:50.43pigpenkimosabe, most of the people here feel that one must be able to "handle themselves" in the shell with asterisk before "embracing" a gui.
15:50.56pigpenMany in here end up developing their own gui.
15:51.20pigpenBecause, well, we want it "my way"
15:51.39kimosabetrue
15:51.45*** join/#asterisk bintut (n=bintut@cm179.gamma187.maxonline.com.sg)
15:51.54bintuttzafrir:  still there?
15:51.55pigpenAnd the GUI's out there seem like, well, they were smoking something while developing it.
15:52.00[TK]D-Fenderkimosabe: And you were not explicit in what this "web interface" was being used to manage.  I'd prefer not to guess what anyone else probably would...
15:53.14kimosabein fact i really wanted the web interface for the person im giving the asterisk box to for there use in fact you know basic administration  thanks though man i apreciate it
15:53.27bintuti upgrade my asterisk here from 1.4.5 to 1.4.9 but whenever i check the help command, the "zap" related commands are not shown at the bottom.. why is this so?  zaptel and wctdm modules are loaded.
15:54.19pigpenbintut, I had that too.  Stop asterisk, and restart.
15:54.36pigpenif not, you may not have the zap modules loaded.
15:55.05pigpenin fact, if I jack with * 1.4.9 (mostly zap related stuff) I loose Zap/Sip/IAX
15:55.16kv0sIt is possible to make system recordings without have installed asterisk addons?
15:56.10Hmmhesaysyes
15:56.18Hmmhesaysrecord with your phone
15:56.20dominic1@kimosabe, search for trixbox or have a look on asterisknow
15:56.44dominic1but I never used them
15:57.15Hmmhesaysor just set up an extension to record your sound files
15:58.16*** join/#asterisk _ViperNetworks (n=Nitesh@66.184.39.174)
15:58.25kv0sHmmhesays: Mhm. If i record via System recordings at freepbx (*77) i can record and play the recorded sounds, but if i use these recordings at ivr, announcements or others the line is quiet ...
15:58.43_ViperNetworksHello All...
15:58.47bintutpigpen:  check this out --> http://paste.debian.net/33912
15:58.52kv0s... so quiet (no sound!) quite? mhmm. .. don't know which the right word for ..
15:59.55_ViperNetworksI need help with SAY TIME... everytime I ask to say time... it says wrong time...
16:00.00bintutanyone knows why my zap channels are lost?
16:00.17pigpenbintut, yeah..jump into the cli and do "zap show channels" , see if they show up.
16:00.29*** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org)
16:00.42bintutpigpen: it doesn't show up
16:00.54pigpenbintut, yeah..same thing I was getting.
16:01.04*** join/#asterisk fx0 (n=fx0@commandline.terrorist.net)
16:01.04pigpenHave they ever showed up?
16:01.11MindTheGap[TK]D-Fender, thanks, that did the trick, got InUse, watchers and everything!
16:01.12bintutpigpen: nope
16:01.12_ViperNetworksbintut: try modprobe zaptel and see if u get any errors
16:01.40bintut_ViperNetworks: the drivers are loaded already.  please check this out ==> http://paste.debian.net/33912
16:01.45[TK]D-FenderMindTheGap: You're welcome
16:01.54pigpenyou may not have zap listed into your /etc/asterisk/modules file.
16:02.07pigpenmod_zap  ( I forget...I never jack with it anymore)
16:02.33pigpenyeah, in modules.conf:  load => chan_zap.so
16:02.47pigpennot there, no workie.
16:03.13datachomperIs there a way to monitor SER similarly to asterisk's CLI ?
16:03.55pigpenBeing the php "Zen Master" that I am, I need some php help.
16:03.57pigpenhttp://pastebin.ca/643184
16:04.38pigpenThis php script just grabs a list of exten's and is supposed to do a playback to each, but it is only processing the first exten of the list.
16:04.57pigpenand I -suck- at php.  Could someone point me in the right direction?
16:05.35bintutanyone knows why i don't get my zap channels after upgrading to 1.4.9?
16:06.07pigpenbintut, did you check your modules.conf?  Also, did you upgrade zaptel?
16:06.28polerinpigpen: what what?
16:06.34polerinyou need some php help?
16:06.44pigpenfyi: we upgraded from 1.4.5 to 1.4.9 on 3 systems, no issues.
16:06.54pigpenpolerin, yeah..I suck.
16:07.03pigpenI suck even worse on perl.
16:07.39[TK]D-Fenderbintut: You have to recompile * after zaptel....
16:08.31bintutpigpen: yes, i upgraded zaptel to 1.4.4..
16:08.54bintut[TK]D-Fender: yes, that's what i did
16:09.12polerinpigpen: msg me
16:09.29*** part/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
16:09.40*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
16:11.25*** part/#asterisk dominic1 (n=dob@213.221.82.242)
16:12.58*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
16:16.59*** join/#asterisk seldon75 (n=chatzill@69.77.161.2)
16:17.07seldon75hello fellows
16:17.40*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
16:17.41*** join/#asterisk billybongo (n=rich@82.153.23.79)
16:18.03*** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus)
16:18.12seldon75what's the Zaptel configuration parameter that adjusts the sensitivity with which Asterisk determines whether a call has been terminated?
16:19.08*** join/#asterisk shazaum (n=shazaum@200.175.61.250.static.gvt.net.br)
16:19.13bintutbusydetect = yes
16:19.19bintutis that the one?
16:19.30*** part/#asterisk nayfan (n=nate@193.82.139.227)
16:21.35seldon75hmm im looking more for a paramter that tweaks what [voltage/impedence?] detected on the line triggers a 'Call-terminated' event in Asterisk
16:22.06Strom_Mseldon75: are you in north america?
16:22.37billybongothose of you who use ser|openser on the front of asterisk, do you have users in openser, asterisk or both?
16:22.48[TK]D-Fenderseldon75: By default you need to ask your telco to enable "Call Disconnect Supervision", and * just does its thing.
16:22.53billybongoI'm reading some tutorial where they have both - this doesn't make sense to me
16:23.25Nuggethrm, does asternic FOP not work with 1.4.x or am I just doing something wrong?  I never had a problem getting it going with 1.2 and 1.0
16:23.48billybongohttp://openser.org/dokuwiki/doku.php/asterisk:realtime-integration is where I'm looking
16:23.55Nuggetthe little buttons in the panel don't show any call status or activity
16:24.02billybongoit seems to me if users a registering with openser, they shouldn't need to register with asterisk as well
16:24.28[TK]D-FenderNugget: 1.4 AMI changed a few things IIRC
16:24.30lirakisbillybongo: that makes sense to me.. but i dont know :\ i wish i knew openser.. but its documentation is so sparse and .. not at beginner level
16:24.38Nuggetbummer
16:24.53Nuggetany alternatives that don't totally suck?  :)
16:24.54[TK]D-Fenderbillybongo: The wouldn't need to register with *.
16:24.59billybongolirakis: I've been using asterisk for a while now and I've just got into openser, I think it's ok if you take it slowly
16:25.02seldon75Strom_M: Canada
16:25.07[TK]D-FenderNugget: Depends what you're using it for
16:25.25[TK]D-Fenderseldon75: then do as I jsut adivised
16:25.27Nuggetjust want a way to view status at a glance.  no actual interaction is necessary
16:25.32billybongo[TK]D-Fender: any idea why they set up views for * to read the openser database?
16:25.38Nuggetwhich extensions are on the phone, how many channels in use, that sort of thing'
16:25.39bintuthello all.. anyone here knows why zap related commands are not listed when executing the command 'help' in asterisk.. i upgraded my zaptel and asterisk from 1.4.3 and 1.4.5 to 1.4.4 and 1.4.9 respectively.. # cat /proc/zaptel/1 says my tdm400 is detected.. lsmod says zaptel related modules were loaded..
16:25.47Strom_Mseldon75: when the far end goes on hook, do you get a half-second talk battery drop/
16:25.48Strom_M?
16:25.50seldon75ok, for my boss could you summarise what "Call Disconnect Supervision" does?
16:25.52lirakis[TK]D-Fender: i asked this earlier.. of everyone.. but now specifically of you.. i am curious.. have you taken any formal asterisk training? or is your knowlege self learned?
16:26.03[TK]D-FenderNugget: there are a few receptionist tools out there for this, and its not a big deal to write your own.
16:26.14seldon75we're seeing lines stay open after the call terminates
16:26.26[TK]D-Fenderseldon75: Polarity reversal on the line of a complete cut depending
16:26.41[TK]D-Fenderlirakis: Self taught
16:26.46*** join/#asterisk Hydrant (n=aj@mailwn.dainty.ca)
16:26.56[TK]D-Fenderbillybongo: i have no details to validate that.
16:27.09billybongoI think it's because of this:    rewritehostport("voip_gw.domain.net:5060");
16:27.15HydrantHey all, does anyone have experience with the Budgetone 200 or 2000 ?  I'm looking to buy an entry-level phone for asterisk, and am looking for advice
16:27.24[TK]D-Fenderbintut:  "load chan_zap.so"
16:27.25billybongoso the user registers first at openser then gets sent over to * for some things
16:27.25bintuteven 'ztcfg -vv' says it's there
16:27.35bintut[TK]D-Fender: it also doesn't work
16:27.38billybongowhereas I think it's sensible to allow openser to proxy all the SIP
16:27.44[TK]D-Fenderbintut: waht does it say?
16:28.03lirakis[TK]D-Fender: hmm.. okay.. i ask b/c .. i only know what i get from here.. or from "hobbying" with my * system at home.. but i am getting more into it at work.. and curious if there is a good source to build a solid base on and better understand the underlying concepts of *
16:28.25[TK]D-Fenderbillybongo: OpenSER should be your front end for the phone 100%.  you can choose to run * unauthed if you with if all phones are considered equal.
16:28.27lirakisHydrant: i love the gxp-2000
16:28.27masus"/usr/bin/ld: cannot find -lssl
16:28.28masus"
16:28.39[TK]D-Fenderbillybongo: and you're only looking to use it as an app / termination server
16:28.47[TK]D-FenderHydrant: ...
16:28.49[TK]D-Fender~gs
16:28.50jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
16:28.51Hydrantlirakis: how is your call quality ?
16:28.53lirakisHydrant: never used the bt-200 .. but i have bt-100, 101, and 102 .. the BT is junk
16:28.56[TK]D-Fender~grandstream
16:28.57jbothmm... grandstream is the Yugo of VoIP hardware.  Run.  Run away now.
16:28.58bintut[TK]D-Fender: please check this out ==>  http://paste.debian.net/33915
16:29.14HydrantCan I get a good IP phone for under $100
16:29.21lirakisHydrant: .. the speaker phone is bad
16:29.26[TK]D-Fenderbintut: Doesn't exist... so its not installed.
16:29.38lirakisHydrant: the rest is good.. and for a hobby or home phone .. i like the gxp-200 a lot
16:29.41lirakis*gxp-2000
16:29.44[TK]D-FenderHydrant: Polycom IP 320 = $87.50
16:30.02sweeper330 = 109 :#
16:30.04sweeper:3
16:30.15Hydrant[TK]D-Fender: I was advised that Polycom was very good as well
16:30.16bintut[TK]D-Fender: i just checked that directory and it's not there.. how come?  i already installed zaptel-1.4.4?  :(
16:30.18[TK]D-FenderHydrant: Add a 20$ 9tops) power supply since I'm sure you don't have PoE and you're set.
16:30.21lirakisHydrant: many people will tell you grandstream sucks etc.  I dont think they suck .. they just arent polycoms
16:30.48sweeperlirakis: but they DO suck
16:30.58sweepervlan's get broken every other software release
16:31.10sweeperfeatures come and go the same way
16:31.11lirakisHydrant: I have 5 phones i use .. 3 grandstreams (2 gxp-2000, 1 bt-102, 1 cisco 7940, 1 polycom 301)
16:31.11[TK]D-Fenderbin recompile * (make SURE to redo the menuselect) and INSTALL both
16:31.36Hydrantlirakis: do you prefer to polycom to the grandstream ?
16:31.37lirakisHydrant: .. the only ones i would reccomend you not get are ANY BT 1XX phone .. or the CRAPPY cisco phone
16:31.49bintut[TK]D-Fender: is it on the side of the asterisk and not on zaptel?
16:31.52*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
16:31.56[TK]D-Fenderbintut: Quite possibly.
16:32.08bintut[TK]D-Fender: which one?
16:32.25[TK]D-Fenderbintut: BOTH
16:32.29Hydrantlirakis: I'd rather pay a bit more if I need to, if it ensures that I don't have something that will drive me crazy
16:32.30NOT_guruI loves my cisco phones
16:32.41NOT_gurubut I will be testing linksys phones next
16:32.41[TK]D-Fenderbintut: Redo Zaptel top to bottom, then *
16:32.54lirakisHydrant: .. i just like the gxp-2000 .. id like to get the gxp-2020 to replace the pos cisco 7940.   I think polycoms are better phones.. but .. i like the grandstreams a lot.. they never give me any problems.. they are simple to configure.. and give you a lot of value
16:33.00*** join/#asterisk aikanaro79 (n=chatzill@89-180-44-112.net.novis.pt)
16:33.03[TK]D-FenderHydrant: You won't regret a Polycom
16:33.10NOT_guruI answered this in #trixbox hydrant... but you left as I was typing
16:33.36|Rain|I've used Polycoms and Grandstreams (although I've not used a Grandstream since they revamped their firmware)...  the polycoms are definitely higher quality phones, but the grandstreams are definitely more bang per buck
16:34.09NOT_guruI have had zero issues with my cisco 7940 - 7961g hpones
16:34.22NOT_gurugreat sound
16:34.23lirakisHydrant: .. probably everyone here will tell you grandstream sux!! .. i think they proably used grandstreams when they were first out.. and they really did suck then... but they have gone through a few hardware revisions.. and lots of firmware updates.  I like them a lot.. (shrug)
16:34.24aikanaro79is it possible to get a list of all the users that registered with asterisk?
16:34.35Strom_Mcalling grandstream "bang for the buck" only works if you use the phone...like...once a month
16:34.38NOT_guru~granstream
16:34.47NOT_guru~grandstream
16:34.48jbothmm... grandstream is the Yugo of VoIP hardware.  Run.  Run away now.
16:34.52[TK]D-Fender|Rain|: depends on usage, and only by the smallest of margins, and for that I'd never sacrifice quality
16:35.02*** join/#asterisk ToyMan (n=Stuart@user-12lcquh.cable.mindspring.com)
16:35.07HydrantI can get a Polycom Soundpoint IP 320 for $87, which is about the same price of the Grandstream 2000 one
16:35.11lirakisStrom_M: grandstreams are my primary phones.. i use them several times a day
16:35.28E-bolaIf i can use a WAN ip staticly on my connection but onyl knwo the range subnet and gateway, is there someway i can test which ip works for me automatically?
16:35.30lirakisHydrant: 320's are not good.. get the 301 .. its better
16:35.40lirakisHydrant: the 320 is a cheaped out polycom
16:35.41[TK]D-Fenderlirakis: LOL.
16:35.52[TK]D-Fenderlirakis: IP320 kills the 301.  301 = DEAD
16:35.58|Rain|I hate the 301s
16:36.00aikanaro79another question: is it possible to include other contexts on a conditional basis?
16:36.05|Rain|I've never used a 320, but it looks a lot better than the 301
16:36.20lirakis[TK]D-Fender: .. 320 has more features.. but .. its not as "solid" a phone i dont think..
16:36.26[TK]D-Fender320 = PoE native, pixkel based display , MICROBROWSER, speakerphone, lit indicatorss.....
16:37.19[TK]D-Fender|Rain|: 301 is bad at all... in fact I'd rather have my bed-side IP 301 at my office desk, than the Aastra 57i CT I have...
16:37.35[TK]D-Fenderisn't*
16:37.42NOT_gurulirakis:  how much you want for your "pos cisco 7940"?  =D
16:37.54|Rain|well, it's certainly not the worst phone in the world, but there are a lot of choices I'd go for before it
16:37.55HydrantThe  301 is about $128 with adapter, the 320 is $87
16:37.56lirakisNOT_guru: .. its in my desk drawer now.. lol
16:38.15lirakisHydrant: psst.. the get the gxp-2000 .. lol
16:38.16NOT_guruI love the cisco phones... with the exception of no backlight
16:38.34[TK]D-FenderHydrant: 301 doesn't need an adapter, it comes with whichever you need.  the 320 requires a power brick or PoE
16:38.34NOT_guruwhich is the only reason I will be testing the linksys
16:39.37NOT_gurubacklit display that is
16:39.43NOT_guru~linksys
16:39.43jbotlinksys is, like, a tool of satan
16:39.43lirakisNOT_guru: .. i and my boss have had issues with them having a long audio delay after the phone picks up... this is on totally seperate pbx's that are unrelated to eachother.  I swapped out my 7940 with a bt-101 to test.. and the audio delay was gone... while my boss is like "hello... .. heloo.... helllo!!!"
16:39.52[TK]D-FenderNOT_guru: Linksys works, but their display usage sucks, call handling is second rate, you get DIALTONE after being hung up on.. (ANNOYING!!! its like a stupid ATA + analog phone).  Their speakerphones are "tinny", and too light (slide around on the desk when the handset cord is a little stretched.
16:39.54HydrantI'm not a business, I have 3 other people sharing an apartment with me, and I'm setting up VoIP... ideally I don't want something that hisses and pops like a 1920 phone, bit I don't want to shell out $200 / phone either for a high end one... I don't care about backlights or anything like that...
16:40.18[TK]D-FenderNOT_guru: Not a bad choice, but in North America, Polycom kills all other competition.
16:41.03[TK]D-FenderHydrant: You COULD be cost-conscious and just get ANA's that way people could sue their own phones (cordless and all)
16:41.05lirakisHydrant: .. thats how i started.. def.. the gxp-2000 will make you happy.  I will say the speakerphone is terrible... but i didnt need it.. which is why i bought them anyway... i love them.. never an issue.. and lots of functionality
16:41.06[TK]D-FenderATA*
16:41.35jcolpI use an SPA3102 to bring my analog line in, and also hook it up to my DECT cordless phone
16:41.35lirakisHydrant: .. a polycom could make you happy too though.. (shrug)
16:41.37jcolpworks great
16:41.44lirakisugg.. sipurra
16:41.57*** join/#asterisk Relan (i=X_S@59.94.242.86)
16:42.15*** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net)
16:42.19ZaVoidmorning
16:42.30ZaVoidanyone using pgsql with 1.2.23?
16:42.40ZaVoidi downloaded 1.2.23 and the pg_sql.so file is missing
16:42.56ZaVoidand i tried an older 1.2.x version and it doesn't like it
16:43.03RelanI am absolutely new to this technology.
16:43.05Hydrant[TK]D-Fender: I have been thinking of just using normal phones too
16:43.19*** join/#asterisk zpertee (n=chatzill@cpe-65-189-209-131.neo.res.rr.com)
16:43.19RelanInfact just heard about it and have been asked to prepare it.
16:43.20[TK]D-FenderHydrant: it IS considerably cheaper....
16:43.34[TK]D-FenderRelan: ~book
16:43.37[TK]D-Fender~wikis
16:43.38jbotrumour has it, wikis is http://www.voip-info.org
16:43.38RelanI am a fresh Comp engineer. Just graduated.
16:43.42[TK]D-FenderRelan: www.asterisk.org
16:43.49[TK]D-Fender~book
16:43.50jbotextra, 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
16:43.57ZaVoidthis file wsa missing: res_config_pgsql.so
16:43.58[TK]D-FenderRelan: Go read the book
16:44.06zperteeI just bought this http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=200136132492#ebayphotohosting computer.  Do you think that it will be powerful enough to support 10-15 users?
16:44.31[TK]D-Fenderzpertee: way more than enough
16:44.32lirakisHydrant: .. ata's can be had for like $40 .. they can some times be a bit of a pain to get into config... like they have a private LAN that you have to go into .. and then make the config avail.. kinda wierd.. (shrug)
16:44.45RelanCan i just have a rough idea what it is all about ?
16:45.07lirakisRelan: voice -> data -> travels over internet/ip
16:45.08[TK]D-Fenderlirakis: 3 touch-tone entries to enable the web-config & get the IP, all web config from there. 10 minute job
16:45.09zpertee[TK]D-Fender: ok thanks
16:45.26lirakis[TK]D-Fender: yeah it depends on the ata
16:45.42[TK]D-Fenderlirakis: I'm talking about SPA-2102, 3102, etc
16:45.48Hydrantlirakis: I need 1 ata per line though, right... or one that has 4 ports
16:45.50RelanWell thanks about that. but what does a programmer has to do into it ?
16:45.52[TK]D-Fenderlirakis: The kind we normally suggest in here
16:46.07lirakisHydrant: yes 1 ATA per line..
16:46.13[TK]D-FenderRelan: Stop now.  Go download the book and get reading.
16:46.24lirakisHydrant: or a multi port ata.
16:46.29[TK]D-FenderHydrant: 2 x 2-port ATA's
16:46.50ZaVoidfender you got a copy 1.2.23?
16:46.50HydrantAm I saving money buy buying the ATAs over just getting the phones
16:46.51NOT_guruLOL  I linke the linksys ATA's as well  and HATED the grandstream handytone
16:46.52Relan[TK]D-Fender alright mate
16:46.53Relan:)
16:46.55lirakisHydrant: unless you want a kind of gateway .. i think 2 ports is as big as they make... otherwise you have to get like 8 ports
16:46.59RelanThanks for the information.
16:47.06[TK]D-FenderHydrant: http://www.telephonydepot.com/Linksys_ATA_s/33.htm
16:47.11NOT_guruyes   normal analog phones are cheap
16:47.12[TK]D-FenderHydrant: $66$ for 2 phones.
16:47.16NOT_guruand you have lots of choices
16:47.30lirakisNOT_guru: .. yeah the handytone ata's arent good
16:47.32NOT_guru( the wife can buy whatever phone she wants )
16:47.36russellbguys, i just cracked asterisk 1.4.9 ... msg me for teh warez
16:47.37lirakisNOT_guru: kinda spastic
16:47.38ZaVoidthe grandstreams?
16:47.44ZaVoidgrandstreams work pretty well for me
16:47.49sweeperrussellb: zomg
16:47.53Hydrant[TK]D-Fender: and I still have to buy normal phones... so I'm kinda thinking to just go with VoIP phones... I considered ATA for a while before too
16:47.55lirakisZaVoid: .. the ATA's or the phones?
16:48.03[TK]D-Fenderlirakis: SPA-8000 = 8 prots @ < $300
16:48.05ZaVoidfxo devices and fxs ones
16:48.11Hydrant[TK]D-Fender: Right now I have no telephone equipment to reuse
16:48.19ZaVoidactaully the phones gxp2000 phone i love
16:48.23[TK]D-FenderHydrant: Ah, if you're starting WITHOUT phones, then may IP phones ARE for you.
16:48.26lirakis[TK]D-Fender: yeah thats what i was talking about
16:48.27ZaVoidso anyone useing 1.2.23?
16:48.37lirakisZaVoid: i love them too ! :D
16:48.51[TK]D-Fenderlirakis: GS has that shit-box 4-port model I won't speak of ;)
16:49.08ZaVoidtrying to find out why latest 1.2 doesn't have a res_config_pgsql.so file
16:49.09[TK]D-FenderHydrant: Do you have an EXTRA RJ45 jack for each phone?
16:49.18lirakis[TK]D-Fender: ah..  yeah i saw it at von last year.. never used it though
16:49.32Qwell[]ZaVoid: there never was one for 1.2
16:49.35ZaVoidoh really?
16:49.52ZaVoidwould there be a different module i would load instead then i guess?
16:50.05Qwell[]res_config_odbc, but you really should upgrade to 1.4
16:50.30ZaVoidyeah well i'm downgrading to 1.2 to try to fix a problem
16:50.33*** topic/#asterisk by Qwell[] -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.23 and 1.4.9 released (July 24, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- 1.2 is in security maintenance mode. No non-security related bug fixes will be applied.
16:50.43Hydrant[TK]D-Fender: No, I don't think I have extra jacks in each room, I figure I'll need a router for each room anyways
16:50.45*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
16:50.50ZaVoidwe are getting random times form all of our sip carriers where the asterisk 1.4 isn't processing the BYE's correctly it seems
16:50.54[TK]D-FenderHydrant: The Linksys SPA-941 might be the phone for you then : http://www.telephonydepot.com/product_p/105-054-941.htm
16:51.06lirakisHydrant: many phones have dual jacks + switching ability
16:51.11ZaVoiddoesn't seem to tear down the calls correctly
16:51.17lirakisHydrant: including the gxp-2000 ;)
16:51.19[TK]D-FenderHydrant: Pass-through switch built into the phone, includes its own power supply, easy confi and acceptable quality.
16:51.21ZaVoidwas hoping maybe 1.2 didn't have that problem Qwell[]
16:51.34*** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net)
16:51.52Qwell[]well, if it happens in 1.2, it will never get fixed...
16:52.04RelanCan it run on windows?
16:52.09ZaVoidyeah is a strange transient issue
16:52.11Qwell[]Relan: no...
16:52.49ZaVoidi got a php script that will kill long calls at 75 minutes on 1.4
16:52.49lirakis[TK]D-Fender: Those SPA's gave me wierdo issues with DHCP .. all the lights would start flashing and they would reboot... then they would be okay.. again.. kinda spastic
16:52.49Daviey[TK]D-Fender: "includes it's own powersupply"  do you mean PoE?
16:52.49Qwell[]ZaVoid: You could set an rtp timeout
16:52.49ZaVoidand some calls hit that.. and my carriers claim the call was(example) 12 minutes according to their cdrs and not 75 minutes
16:52.50jcolpfarewell 1.2
16:52.52Hydrant[TK]D-Fender: What's the advantage of the linksys over the polycom ?
16:52.59NOT_guruoh  waht was that monkey scream addon
16:53.05Qwell[]tt-monkeys?
16:53.05DavieyHydrant: price!
16:53.19ZaVoidsomthing like rtptimeout=30 Qwell[] ?
16:53.20lirakisHydrant: i would strongly reccomend the polycom over linksys/sapurra
16:53.31Qwell[]ZaVoid: I think it's in seconds
16:53.33Qwell[]30 is a bit...short
16:53.34ZaVoidthats a [general] setting though right? not specific to certain peers?
16:53.43ZaVoidyeah i'm just saying as an example 30
16:54.01ZaVoidonly problem with that is if the rtp isn't going through the asterisk that doens't help much
16:54.03Hydrantlirakis: I think I'm kinda between Polycom 320 or Grandstream 2000
16:54.13Qwell[]Hydrant: stay far away from grandstream
16:54.17Hydrantlirakis: Are both equally well supported by asterisk
16:54.19lirakisHydrant: either one will be a fine for you
16:54.23lirakisHydrant: yes
16:54.30ZaVoidHydrant: i like the 2000 and its cheap
16:54.35Qwell[]~cheap
16:54.35jbotextra, extra, read all about it, cheap is a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
16:54.35ZaVoid4 seperate config lines too
16:54.41ZaVoidlol
16:54.43*** join/#asterisk gardo (n=gardo@121.97.195.87)
16:54.46*** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br)
16:55.17[TK]D-FenderHydrant: it is a little cheaper, includes the power supply and has the built in 2-port switch
16:55.17[TK]D-FenderHydrant: $99 = end of story
16:55.22lirakisHydrant: .. like i said.. lots of people will bash it .. but i bet you will like it a lot . and it will work well for you... i think many are carrying  ideas from when the phones first came out.
16:55.30ZaVoidactually qwell i do have an rtptimeout set on in my general section of the sip.conf files
16:55.49lirakisHydrant: either of those phones will work well for you
16:56.09DavieyHydrant: Linksys SPA-9xx series actually look like nice phones
16:56.19*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
16:56.23lirakisDaviey: looking nice is different than working .. lol
16:56.29Davieynot an 90's throwback
16:56.32Qwell[]and gs does neither
16:56.44Hydrant[TK]D-Fender: Why is the Linksys better than the Polycom 320
16:56.46lirakisQwell[]: .. say what you want.. i use them everyday without issue
16:57.03*** join/#asterisk andrebarbosa (n=andrebar@62.48.215.167)
16:57.22NOT_guruI personally was looking at the spa 962 I think it is
16:57.27lirakisHydrant: its not..
16:57.32NOT_guruor maybe 942 if the 962 is too pricey
16:57.38*** join/#asterisk [T]ank (n=ckwall@206.71.78.172)
16:57.42[TK]D-FenderHydrant: Polycom is hands down a better phone.  You'd need the 330 not the 320 since you aren't going to wire thems eperately.  So that add up the price.  Then you have to add the power supply.  By the time you're done, Polycom = $150 +/- / phone for you.
16:57.55ZaVoidrtptimeout=30
16:57.55ZaVoidrtpholdtimeout=60
16:57.58ZaVoidgot that already
16:57.59Hydrantlirakis: I've read mixed reviews on the budgetone stuff , where people say some work some don't... could it be that you were just lucky with yours ?
16:58.22lirakisHydrant: i dont like budgetone... the GXP line is not the Budgetone line though.. they are totally different
16:58.23[TK]D-FenderHydrant: But the SPA-941 is an acceptable choice as you're budget conscious with your specific needs
16:58.33lirakisHydrant: the budgetones are cheap cheap cheap
16:58.47Hydrant[TK]D-Fender: So your main concern is that the 320 doesn't come with a power adapter ?
16:58.59ZaVoidwelltech.com stuff is complete cheapo crap :)
16:59.11Hydrantlirakis: Isn't the 2000 a budgetone though ?
16:59.15lirakisHydrant: no
16:59.29lirakisHydrant: the GXP-2000 is not a budgetone
16:59.32Daviey[TK]D-Fender: The linksys doesn't come with a power supply AIUI, relies upon PoE or seperate purchased power supply
16:59.34lirakisHydrant: nor is the GXP-2020
16:59.38[TK]D-FenderHydrant: Correction $138 for Polycom IP 330  + brick
17:00.00[TK]D-FenderDaviey: I said SPA-941, and it ONLY comes with the brick <-----
17:00.08[TK]D-FenderDaviey: Pay attention!
17:00.17DavieyFor a meeting/conference phone i would hands down pick polycom
17:00.19lirakisHydrant: i have heard better things about the BT-200 .. but i cant say as ive not used it.  I have used all BT-10X models.. they are barebones.. and cheap.. its a phone.. thats it..
17:00.30Daviey[TK]D-Fender: brick = PSU?
17:00.49[TK]D-FenderDaviey: yes.  wall-wart, etc.  Pick your favourite term.  its not PoE at all.
17:00.51sweeperwell, if you've got 24 v somewhere in the building, you can do PoE for the polycoms pretty cheap~
17:01.33Daviey[TK]D-Fender: bah.. I've just purchased (like 10 mins ago) a spa-94_2_ and had to buy a seperate power supply
17:01.47[TK]D-FenderDaviey: you = SMRT :)
17:01.50*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
17:02.46ZaVoidso canreinvite=no will force the media to go through my asteirsk and canreinvite=yes will give it the(essentially) the option to not send the media through the asterisk
17:02.50ZaVoidor do i have that reversed?
17:03.28HydrantAlright, I've gotta get back to work... I'm going to download the datasheets and look for reviews on the gtx-2000 vs. the polycom 320 and see what I find... thanks for all your help everyone
17:03.59*** join/#asterisk jordanb (n=jordanb@adsl-68-20-20-59.dsl.chcgil.ameritech.net)
17:04.01[TK]D-FenderHydrant: as I said you need the IP 330, not 320
17:04.43jordanbI'm having hella trouble with my SPA3102.
17:04.43jordanbI can make one outgoing call.
17:04.43jordanbAnd then after that I get a busy.
17:05.20datachomperI get busy with it also
17:05.26jordanbShould line voltage be -49 (V).
17:06.24*** join/#asterisk friedrich| (i=friedric@trem-servers.com)
17:06.30*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
17:07.22jordanbActually I get one ring, then busy.
17:08.27BSD_Techanyone here done asterisk clustering
17:09.53russellbQwell[]: didn't you write an ajax clustering platform?
17:10.02Qwell[]russellb: nah, just javascript
17:10.05Qwell[]but it's linux specific
17:10.07russellboh ..
17:10.11*** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home)
17:10.11*** mode/#asterisk [+o Corydon76-work] by ChanServ
17:10.12russellbnm, then.
17:10.38*** part/#asterisk masus (n=tet@88.248.14.186)
17:10.48*** join/#asterisk levi_home (n=levi@levi.dsl.xmission.com)
17:10.55ZaVoidso i if put all my canreinvite's back to "no" on all my peers... and the bad call tear down processing i'm seeing is still happening.... that would make no sense
17:11.02Kattyallo!
17:11.05ZaVoidmost of the calls are with arbinet.. but not all
17:11.37ZaVoidand if i put canreinvite=no in the [general] peer that will affect everything basically
17:12.11*** join/#asterisk andresmujica (n=andresmu@190.24.227.202)
17:13.50andresmujicaanyone has connected an avaya s8720 with an asterisk???
17:14.21*** join/#asterisk karleeto_lap (n=karl@techwifi.franklincomputer.net)
17:15.04kempistanyboy can recommend another iax provider like teliax? :P Seems like its about to die
17:15.39Davieyeek.. i think i have a few dollars tied up in teliax
17:15.42*** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net)
17:15.56*** join/#asterisk sharp (n=sharp@c-68-81-156-176.hsd1.pa.comcast.net)
17:16.25kempisti got $51 in :/
17:16.36Davieykempist: you'll find that most support SIP as they cam use SER to reduce the overhead placed on asterisk.. Can't be done with IAX
17:16.55brettnembah.. how do I $[${FLOAT_NUMBER} > ${OTHER_FLOAT}] properly? I'm doing it now.. and if ${FLOAT_NUMBER} is 0.00 and ${OTHER_FLOAT} is .3 It's returning true
17:17.08kempistyeah, but sip is a nightmare for nat
17:17.13kempistiax is nice on our network
17:17.14tzangerkempist: not really
17:17.19Davieykempist: nightmare / works
17:17.27brettnemhmm.. math?
17:17.38[TK]D-Fenderkempist: lol, SIP has never been a problem for me or my clients.  Double-NAT and all
17:17.41tzangerkempist: I used to think so, but unless you've got a screwy firewall, Asterisk's SIP is actually not too bad for NAT
17:17.51Davieykempist: Once you have NAT working, you'll find it rock solid (or in my experience)
17:17.53[TK]D-Fendertzanger: 2 words : USER ERROR
17:18.12kempistanyboy can recommend any sip provider then? if it has iax better
17:18.19tzangerMind you, I put my Asterisk boxes *on* the net (they are also the firewall), and the phones are usually hiding behind regular old linksys boxes in people's homes/remote offices
17:18.35tzangerIt's not for everyone, but my installations generally work that way
17:18.47tzanger[TK]D-Fender: heh
17:18.55Davieykempist: depends on you location and level of usage for 'best'
17:18.58*** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey)
17:19.03Daviey'best' is never cheapest either
17:19.10Daviey*fact*
17:19.25kempistlol, just looking for good experiences
17:19.43DavieyAre we alloowed to mention suppliers here?
17:19.47markgreeneHas anyone in here setup sip user account in a mysql db and used realtime to grab them for authentication and registration?
17:19.59kempistdo on private chat Daviey if you want
17:22.29|Rain|so, I'm back to the fact that PRI signalling over TDMoE just doesn't seem to work...
17:22.41|Rain|in spite of my best efforts
17:22.45jordanbWhen I get a busy signal I get a lot of these:  DEBUG[14929]: chan_sip.c:11399 sipsock_read: Failed to grab lock, trying again...
17:23.15*** join/#asterisk kannan (n=kannan@121.246.24.158)
17:23.52sheldonhdoes the asterisk project provide a viewsvn interface?
17:24.14|Rain|http://svn.digium.com
17:24.53sheldonhgreat
17:25.13sheldonhi got origsvn out of the bug db, but that seems borked right now
17:25.28sheldonhtrying to get my hands on r77889
17:26.34russellbsvn log -r 77889 http://svn.digium.com/svn/asterisk/trunk
17:26.40*** join/#asterisk mtaht4 (n=m@m815f36d0.tmodns.net)
17:26.43russellbsvn diff -r 77888:77889 http://svn.digium.com/svn/asterisk/trunk
17:26.55russellbor the web interface will work too :)
17:27.10sheldonhcool
17:27.24sheldonhi didn't realise you could pull deltas on trees you haven't checked out!
17:27.26*** join/#asterisk NirS_ (n=Nir@87.68.147.173)
17:27.48russellbyeah, pretty cool
17:27.49sheldonhrussellb: i think i want branches/1.4, actually
17:28.10*** join/#asterisk irule (n=irule@189.164.47.106)
17:28.24russellbah, that's a different revision
17:28.38jordanbSo am I in the wrong place? Is there an asterisk help channel?
17:28.46russellbsvn log -r 77887 http://svn.digium.com/svn/asterisk/branches/1.4
17:28.54russellbsvn diff -r 77886:77887 http://svn.digium.com/svn/asterisk/branches/1.4
17:28.57Qwell[]russellb: diff -c <3
17:29.07russellbQwell[]: i wasn't assuming svn 1.4 :)
17:31.35KattyDaviey: were you the one subscribing to my feed?
17:31.39KattyDaviey: the asterisk blog one
17:31.44DavieyKatty: yes
17:31.51Davieyare you currently having an epiphany?
17:31.56KattyDaviey: subscribe to the "asterisk" label instead of geekery
17:31.59*** join/#asterisk solar_ant (n=solar@122.164.54.143)
17:32.00sheldonhlooks like my isp's proxy server's wanking up svn over http
17:32.03KattyDaviey: i'm starting my posts.
17:32.07*** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
17:32.11DavieyKatty: erm.. what is the feed url?
17:32.13KattyDaviey: but nothing major yet, just basic distro installation and stuffs.
17:32.16Kattysec
17:32.19Daviey'cause i could only find a catch all feed
17:32.20kempistany other recommendation for a sip/iax provider like teliax?
17:32.50sheldonhrussellb: either you have the revisions reversed (77887 was on trunk and 77889 was on branches/1.4) or i'm seriously confused :)
17:33.21KattyDaviey: hmm, it won't let you subscribe just based on label.
17:33.26KattyDaviey: http://42ndgeekstreet.blogspot.com/feeds/posts/default
17:33.52*** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net)
17:34.00russellbsheldonh: 1.4 came before trunk
17:34.06russellbso you're likely confused :)
17:34.32KattyDaviey: you can bookmark 42ndgeekstreet.blogspot.com/search/label/Asterisk tho
17:34.50sheldonhrussellb: aaaaaaaaaaaaaaah
17:35.29*** join/#asterisk ServerDown (n=ServerCr@unaffiliated/servercrash)
17:36.04DavieyKatty: bah.. guess i'll just have to sub to your whole blog.. keep it interesting!
17:36.48ServerDownhi i am planing on setting up a phone notify system for a local school, which can send a voice message to a select group of parents about the parent meetings, notices etc, what would u suggest, how should i start and move
17:37.45ServerDown* came into my mind to start with, how far do you think * would be the right choice to select and move forward
17:38.17Davieyhmm.. sounds like an autodialer to me
17:38.29ashdmy cli does not have the zap commands - the a
17:38.37russellbServerDown: asterisk would be a perfect platform for that :)
17:38.54ashdTDM400P i have gets all its modules loaded
17:39.03DavieyServerDown: look into 'call' files
17:39.12ashdi have since recompiled *
17:39.15*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
17:39.17ashdbut no zap
17:39.21ServerDownDaviey, a sort of autodialer, but with ability for the user to call in store the voice message and then use some kind of web interface or dail in interface to send the message to a group
17:39.29Davieyjust need to dump a series of text files into a folder, and it'll process them
17:39.49ServerDownrussellb, how should i start on this? can you give me more idea
17:40.09DavieyServerDown: custom IVR and "call" files
17:40.36russellbwell Daviey said basically
17:40.56russellba special dialplan for letting people call in, record a file, and then run a script which generates the right call files
17:41.04DavieyServerDown: look at http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
17:41.14russellband the call files specify who should be called - they say "call this person and play them this file", it's like 3 lines
17:41.31russellb3 line call files that is
17:41.39russellbthe dialplan and script to generate them will be more than that :)
17:41.52Davieynot by much though.. just a mysql dump
17:42.12ServerDownsounds simple, dont know how hard would be it be to setup
17:42.17generalhancan some one explain to me the possible reasons that MoH would start and immediately stop when ever i try to put someone on hold ? the CLI just shows "Started music on hold, class 'default'", "Stopped music on hold on"
17:42.27ServerDownespecially for me who is very new to *
17:42.53DavieyServerDown: play with "call" files, then once you understand them.. the rest will be easy
17:43.00Daviey(well relitively"
17:43.03ServerDowni have start from very basic, like selecting hardwares and then moving up, i am currently trying to climb the step learning curve of *
17:43.27DavieyServerDown: Is this a sole purpose for the box?
17:43.35DavieyOne call at a time, calling parents?
17:43.51*** join/#asterisk Cyon (n=cyon@216.179.31.170)
17:44.51ServerDownya initally one call at a time, then moving upto 10 calls or more
17:45.04ServerDownyes thats the sole purpose of the box
17:45.39DavieyServerDown: This can be low spec
17:46.02ServerDownwhat i am looking for a web app, where by the user can login and create groups of phone numbers to call, then select a group and select the message to be sent as notification
17:46.12Davieyand if the quality isn't mission critical, think about SIP trunking the calls - will be cheaper and easier to have concurrent calls
17:46.45DavieyServerDown: I would initially make a, say, python script that is interactive
17:46.52Davieythen think about a webapp
17:46.54ServerDownDaviey, how many pstn lines would i need ?
17:47.10DavieyServerDown: none - just decent internet if you route it via SIP
17:47.14ServerDownDaviey, hmm thats also a good option to sart with
17:47.21[TK]D-FenderServerDown: how many do you want?
17:47.46ServerDownDaviey, no not routing via sip, its local calls, so i can use the pstn connection to call local numbers
17:48.10DavieyServerDown: it's normally heck of alot cheaper to use internet routing SIP to PSTN, but don't rely upon it for mission critical stuff
17:48.26ServerDown[TK]D-Fender, whats the ratio that needs, is it going to be 1:1 that is one 1 pstn line for 1 call at a time
17:48.56DavieyServerDown: Ahh.. If you use PSTN, a normal phone line, you will need a line for each concurrent call.. Starts to get messy, then you need to consider a T1 or equivialnt.. The  it get's pricy
17:49.00[TK]D-FenderServerDown: clearly.  So how may simultaneous calls do you want to handle?
17:49.04ServerDownDaviey, the institute has free unlimited call plan
17:49.17*** part/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2)
17:49.30ServerDownso probably they would like to stick with pstn line only
17:49.59*** join/#asterisk ManxPower (n=manxpowe@032-437-131.area7.spcsdns.net)
17:50.18DavieyI don't think there are many PCI cards that have multiple PSTN connections.. So you might be limited by the number of slots in your mobo
17:50.56ServerDownthere are those boxes which connect via ethernet port and has 10, 12, 24 ports
17:51.00kannanDaviey, what about Rhino channelbanks
17:51.10Davieykannan: starts to get pricy then
17:51.16ServerDowndont know exacltly if thats what i would work with
17:51.25kannanah ok, i know audiocodes FXO boxes very costly
17:51.26Hmmhesaysoops
17:51.52Hmmhesaysanyone have any idea why 00 prefixes are removed my from dst in cdr-csv?
17:51.58ServerDownschool will for sure not go for much costly equipments, what would be your suggestion
17:52.05ManxPowerIf you think Channel Banks are pricy, try a SIP gateway
17:52.26coppiceServerDown: a string and two cans? :-)
17:52.26DavieyManxPower: how so?
17:52.36kannanIn windows OS, I see ordinary Voice and fax modems that works with dialer software, does not asterisk have device drivers for such?
17:53.35ManxPowerDaviey: price them out.  SIP/PSTN gateways are expensive
17:53.40DavieyServerDown: might be worth looking into xorcom hardware; they are great guys aswell
17:53.41ServerDownManxPower, generally sip gateways has one or two pstn connection, do you know any such gateway which has over 10 simultaneous ports
17:53.42ManxPowerkannan: No.
17:53.45[TK]D-FenderServerDown: You are NOT answering the question....
17:53.56ManxPowerServerDown: AudioCodes
17:53.57[TK]D-FenderServerDown: How many lines do you want?
17:54.17ServerDown[TK]D-Fender, To start with 20 max
17:54.44ServerDownbut need to upscale it in future
17:54.46[TK]D-FenderServerDown: Wow.  that is a lot.your interface cost will be high.
17:54.52ManxPowerAudioCodes or the TDM2400P card
17:54.58[TK]D-FenderServerDown: Even at 20 you should consider a PRI
17:55.02DavieyI suppose $2000-$3000 isn't too bad, if they don't need to worry about any charges
17:55.03errrI have exten => b231,1,Voicemail(sb231@work)  but every time I ring this extension it plays the busy message then still gives the instructions, any idea why?
17:55.17RSAManhi
17:55.28ManxPowererrr: what version of Asterisk?
17:55.37errrManxPower: 1.2.23
17:55.39kannanOne PRI card is about 600 USD, right?
17:55.39RSAManquestion : make samples ... does it only create files in /etc/asterisk ?
17:55.53ServerDownI think i will setup multiple servers with 4 port cards, that should be more cost effective :p
17:56.15DavieyServerDown: I would consider BRI/PRI if i were you.. not sure how that can be priced in..  No doubt the school is supplied via PRI - so maybe look at interfacing with their exisiting kit?
17:56.23[TK]D-Fenderkannan: < $500 for the most basic 1-port, but we highly recoomnd models with hardware echo cancellation and those go between 800-900 for a 1 port model
17:56.44ManxPowerServerDown: Best of luck.
17:56.45ServerDownBRI/PRI ?? :o
17:56.58Hmmhesaysanyone have any idea about my cdr problem?
17:57.05kannan[TK]D-Fender : ok , these witll work on E-1 also?
17:57.29DavieyServerDown: In the UK at least, BRI supports upto 8 and PRI supports up to 30 - per cable!
17:57.30ServerDownManxPower, thanks, just discussing to understand better and checking out various options
17:57.46ZaVoidhey Qwell[] or fender... does the rtptimeout command have to be inside a peer or in the [general] section is fine?
17:57.46[TK]D-Fenderkannan: Yes
17:57.49*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
17:58.04[TK]D-FenderZaVoid: Never messed with this
17:58.05ZaVoidi got it set to 60 seconds and both ends of the call on mute.. but its not killing the call... canreinvite=no as well
17:58.10ZaVoidnah?
17:58.11ServerDownDaviey, can you define me whats bri/pri and what hardware i would need it to connect with *
17:58.24errrManxPower: I do have the syntax correct dont I?
17:59.12ZaVoidanyone else use rpttime out?
17:59.27DavieyServerDown: hit the wiki first
17:59.37ServerDownya there only :D
17:59.47ServerDownwith google as secondary option
17:59.52Davieyyep
18:00.17DavieyServerDown: trouble is, I don't know too much about the US t1 services
18:00.40*** join/#asterisk newbie`` (i=nouser@203.81.223.86)
18:00.48x86what would cause Asterisk (or a Rhino channel bank / Sangoma T1 card) to drop an active call randomly?>
18:00.50ManxPowererrr: you are using the old syntax, of course.
18:00.58ManxPowererrr: "show application voicemail" will tell you.
18:01.01x86about 1 out of 40 calls is dropped mid-conversation
18:01.05*** join/#asterisk |dennis| (n=dennis@200.32.236.20)
18:01.07x86any ideas?
18:01.08ManxPowerx86: busydetect or callprogress set is the classic reason
18:01.16errrManxPower: any idea why it still plays the instructions then?
18:01.20x86ah, busydetect is on...
18:01.31ManxPowerx86: putz
18:01.36ManxPowerYou know better than to use that.
18:01.47x86do i use that on POTS lines?
18:01.50Kattywhat's the current kernel version?
18:01.54x86or just not at all?
18:01.55ManxPowererrr: What does "show application voicemail" show you as the correct syntax?
18:01.57Katty2.6.18?
18:02.05x86Katty: kernel.org tells you
18:02.09Kattyk
18:02.14ManxPowerx86: It should be called randomlydisconnectmycallsonanytechnology=yes|no
18:02.15ServerDownDaviey, no problem , ;) neither do iI
18:02.25x86ManxPower: hahaha
18:02.27x86ok
18:02.47x86thanks :)
18:02.58DavieyKatty: uname -r  && echo "<Grin>"
18:03.40coppicea dozen call progress options, and not one of them the result of engineering :-)
18:03.45errrManxPower: from the way I read it I have it correct
18:04.18errrManxPower: the wiki seems to back this up
18:04.36KattyDaviey: i know what kernel version i have ^_-
18:05.13errrManxPower: You may not specify both u and b flags together. You may, however, combine them with s, giving six possibiities:   sb being 1 of the 6
18:06.07billybongo[TK]D-Fender: what do you mean by "all phones considered equal?" I will need to bill each phone
18:06.27billybongobut essentially they can all be considered in the same asterisk context
18:06.54Davieybillybongo: fancy seeing you here..
18:07.53*** join/#asterisk aikanaro79 (n=chatzill@89-180-218-180.net.novis.pt)
18:08.38billybongoDaviey: hi
18:08.44Davieyo/
18:08.46*** join/#asterisk jets (n=jets@pdpc/supporter/active/jets)
18:09.12errrManxPower: lol sorry to bug, this was all pebkac.. I was reloading the voicemail only and forgetting I was making changes to extensions.conf
18:09.41lirakiscafff.... inneee... must... .. have... caff...ine....
18:10.34*** join/#asterisk ming_zy1 (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com)
18:10.48*** join/#asterisk merkurie (n=merkurie@192.153.163.44)
18:11.20sheldonhrussellb: thanks for the help.  busy uploading updated packages to our private repo as we speak
18:11.25ServerDownDaviey, I got the details, the school has ISDN PRI (30 connection)
18:11.38ServerDownDaviey, is there some sort of ISDN PC card too ?
18:11.59Davieyyep
18:12.08DavieyBut i'd go for echo cancellation
18:12.10ServerDownand would Asterisk would be able to take benefit of this
18:12.15Davieyyes
18:12.34ServerDownso that means now * would be able to make 30 calls simaltaneously
18:12.37Davieysangoma's are the currentl flavour of choice
18:13.23coppiceyou eat them? :-\
18:13.43*** part/#asterisk merkurie (n=merkurie@192.153.163.44)
18:13.53generalhanso, i got my MoH working by using mode=files ... but now they are playing WAY too quitely, and if i change mode=mp3 (to play loud) it stops working again ... how should i remedy this bprolem ?
18:14.07ServerDownDaviey, hmm echo cancellor ...dont know if those ISDN pc card has this...
18:14.35DavieyServerDown: yeah Sangoma do
18:14.36Davieybrb
18:16.57*** part/#asterisk andresmujica (n=andresmu@190.24.227.202)
18:17.47*** join/#asterisk saftsack (n=oliver@p54A7E4F3.dip.t-dialin.net)
18:18.43Aces1Uphello all if i want to compile a new module into my asterisk box that is running 1.4 what are the procedures?  or can i load a module on the fly?  I can read for myself if you have a link, having trouble finding info on this subject, thanks.
18:19.20*** join/#asterisk aikanaro79 (n=noone@89-180-218-180.net.novis.pt)
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18:23.44aikanaro79is it possible to define a dialplan that make's it possible to dial someone on his or hers caller id alone (assuming only SIP channels with the same peer definition)?
18:24.20*** join/#asterisk backblue (n=igor@82.102.1.42)
18:24.27backbluehi, anyone here use bluethoot?
18:26.04[TK]D-Fenderaikanaro79: "show application gotoif" , "show function DB"
18:27.10Kattyso for a sangoma card, do i need libpri and zaptel both?
18:27.19Kattyor do i need something additional?
18:27.52backblueKatty: you need to read wiki.sangoma.com
18:27.56Kattyk
18:28.01backbluethey have everything there
18:29.46lirakisKatty: http://wiki.sangoma.com/wanpipe-linux-asterisk-install
18:29.53codefreezeHmmhesays: only if you passed it somehow thru an $[ expr ]  ....
18:29.53lirakis?
18:30.46*** join/#asterisk andethemint (n=robert@vcchgate.vcch01.springfield.tn.us.vcch.net)
18:31.20[TK]D-FenderKatty: both
18:31.53[TK]D-FenderKatty: libpri, Zaptel, then Wanpipe (this will recompile zaptel, Asterisk, then Add-ons
18:33.14Aces1Uptkd got a quick question for you, if i compile a new module into asterisk, do i just put the .c and .o files in the app directory and ensure all includes link correctly?  and i should compile when i do a make install?
18:33.43Katty[TK]D-Fender: ya, i found the wanpips drives on the wiki page.
18:33.59Katty[TK]D-Fender: thanks for the order tho. was wondering that
18:34.16*** join/#asterisk anonymouz666 (n=anonymou@189.25.49.71)
18:34.42Katty:>
18:34.42markgreeneAfter I have setup mysql addon for asterisk and created and populated a mysql table for sip users and verified that I can search that table using the realtime command in the CLI, how do I tell asterisk to look at it for sip user registration AS WELL AS look in sip.conf?
18:34.54anonymouz666Katty!!!!!!
18:35.07Katty:<
18:35.10Katty<PROTECTED>
18:36.04aikanaro79[TK]D-Fender: what if you only have one peer type defined? can't different callers be identified by callerid? (assuming I'm not using db as I can't know in advance who might log in, this asterisk server will only work in a private LAN)
18:36.43*** join/#asterisk dharrigan (n=dharriga@82-71-62-76.dsl.in-addr.zen.co.uk)
18:36.56russellbsheldonh: ooh, private repos, huh?
18:37.14[TK]D-Fenderaikanaro79: Sorry... "show function CALLERID"
18:37.15russellbsheldonh: what kind of stuff do you have in your private repo, huh?  :)
18:37.28[TK]D-FenderTMI <---------------
18:38.33*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
18:38.53*** part/#asterisk [T]ank (n=ckwall@206.71.78.172)
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18:40.00*** part/#asterisk Rienzilla (i=rien@sinas.rename-it.nl)
18:44.09lirakisI had a place in my dial plan that you could channel spy.. i moved it to a macro .. and its not working now.  I promt for an extension to monitor.. or 0 for scan all .. and it doesnt seem to "register" and it returns to the calling context...
18:44.15lirakishere is a pastebin http://pastebin.com/d337613f9
18:45.07lirakisi authenticate fine.. and then it plays the background .. but it doesnt seem to react to input after that.. until it falls back to the calling context.. then it seems to act on the dialed number
18:45.31aikanaro79can I get the ip address of a caller? if so how?
18:48.37lirakishmm .. i guess were all out to lunch
18:49.11*** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil)
18:50.02*** part/#asterisk lirakis (n=etamme@65.200.191.253)
18:50.07*** join/#asterisk lirakis (n=etamme@65.200.191.253)
18:50.46Hmmhesaysok something seriously goofy is going on with cdr-csv
18:53.18*** join/#asterisk EricL (n=eric@clydesdale.linkexperts.com)
18:54.05EricLDoes anyone have a method (short of a System() command) to convert conference recordings to another file type besides .wav (like .mp3)?
18:54.14markgreeneI would love if someone in the room can explain where I am going wrong with SIP definitions in mysql... Anyone?
18:54.38billybongomarkgreene: got some debugging?
18:56.24markgreenebillybongo: I think I am ONE step away from making this work. I just don't know where I tell asterisk that it needs to LOOK in the mysql table for sip user/peer info AS WELL as looking in sip.conf. I have it right now so that from the CLI I can run "realtime load sipusers ..." and get a result from my database displayed in CLI. But I don't know how to make it look there for sip reg info
18:56.38kannanEricL , i think you can used lame
18:56.50kannanor sox
18:56.55EricLkannan:Right, but that's with a System() command or a cron job.
18:57.01kannanyes
18:57.30EricLAlright...I can do that.
18:57.35*** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk)
18:57.42EricLHow do you manage conference calls and saving them?
18:57.56kannanMixMonitor
18:57.58EricLIf you have a lot of users and they are all saving calls, how does one discern whose is whos?
18:58.56*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)
18:59.00kannanNo idea , lol, but i know vicidial does it fine
18:59.24Hmmhesaysanyone know why cdr-csv would be stripping off my 00?
18:59.33Hmmhesaysfor the dst field?
19:04.05Dan0maN_Workcan anyone recommend a soft phone for linux that works best to test with asterisk?
19:05.15*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:05.56EricLDan0maN_Work: Twinkle works fine for me (SIP based softphone).  Although I don't know if you need an IAX or a SIP softphone.
19:06.11EricLDan0maN_Work: And Twinkle is for Linux btw.
19:06.12tzafrir_laptoptwinkle, kiax
19:06.24Dan0maN_Workk
19:06.37Dan0maN_Workwill check em out
19:06.41Dan0maN_Worksip to start with
19:07.00Dan0maN_Worklearning it all, just wanting to get my hands on it before the polycoms get here ;)
19:07.08tzafrir_laptopDan0maN_Work, or chan_alsa ...
19:07.39kannanxten x-lite at couterpath.com
19:07.47kannancounterpath
19:07.55EricLdanalien: Ah, Polycom, good choice.
19:08.11Kattyyay for polycom!
19:08.15Kattyand caffeine.
19:08.28Kattywho knows about sugarcrm?
19:08.42Kattyi want general opinions and things.
19:08.54EricLI had to write some plugins for it about a year ago.
19:09.13Kattywhat do you think it does the best?
19:09.23EricLI only got to use it right before the AJAX got put into it, so I felt it to be kind of kludgy.
19:09.40Kattyso you prefer ajax then.
19:09.52lirakisKatty: i know about it
19:09.55Kattylirakis: cheers.
19:09.56EricLMy favorite part was being able to tailor it pretty much to match our customer database (adding fields and what not).
19:10.05Kattylirakis: i'm thinking about installing it, but i'm not really sure if i need it or want it.
19:10.05lirakisKatty: what are you using it for? crm .. or project management?
19:10.09Kattylirakis: i've heard good things about it.
19:10.15Kattylirakis: and i've visited the website.
19:10.37Kattylirakis: but, meh, if i knew what it was really REALLY handy for, or what it was really really good at, i think i'd know a bit better.
19:10.55lirakisKatty: FYI Vtiger is the "true" open source CRM .. apparently sugar is quasi opensource.. and thats not good enough for pureists some times
19:11.12lirakisKatty: .. well i was looking at it for project management mostly.. and CRM + ticketing
19:11.13billybongothat depends if you think vtiger's code was stolen from sugar
19:11.35lirakisKatty: Sugar is really geared towards sales management
19:11.37Kattylirakis: please expand project management
19:11.53Kattyhmm.
19:11.57Kattyso sales queue management?
19:11.59billybongoI've used sugar many times and generally it's not ideal for any task, but it's OK for most
19:12.04Kattyand people's calendars and such
19:12.09lirakisKatty: i decided  to use dotproject for project management.. because sugar was too heavy and had more than i wanted.
19:12.12billybongocalendaring on sugar sucks
19:12.24Kattywe have outlook here, and i doubt i'll be able to pry our sales reps away from that.
19:12.34Kattynor do we have any sales queues. we don't have that high of call volume.
19:12.42lirakisKatty: Sugar is like .. a portal for a call center employee.. or for a sales person... with other features stuck on to it
19:12.58billybongoKatty: have you looked into tinyerp?
19:13.04Kattybillybongo: never heard of it.
19:13.09Kattybillybongo: what is it?
19:13.20Kattythe only sort of add-on thing i've ever used is FOP
19:14.18billybongoKatty: it's a proper business management tool
19:14.38billybongoi.e. you raise sales orders on it, purchase orders, manage your helpdesk etc etc
19:14.42lirakisKatty: ERP and CRM are similar  .. but ERP is for managing business .. CRM is for managing clients/customers
19:14.59lirakispastebin: ERP is like.. uhh.. that evil SAP program
19:15.14billybongoyeah, don't use SAP
19:15.25*** join/#asterisk anonymouz666 (n=anonymou@189.25.219.220)
19:15.26lirakis(shivver) .. i had to do that for some business class in college
19:15.31lirakisi felt dirty
19:16.03billybongothere's a lot of crossover between ERP and CRM
19:16.12billybongobut most companies need a proper customer database
19:16.18billybongoand erp is better at that
19:16.32billybongoso whatever gets bolted on at least has all the customer details in there
19:16.39billybongogenerally CRM is a lot more free form
19:16.46lirakisparticularly when you start integrating ticketing.. project managment.. product pipelining .. etc.  then they really become.. one in the same
19:16.56billybongoso you get nasty data duplications where some salesperson has their own contacts
19:17.05billybongoand the customer already told the company they moved office
19:17.23billybongoat least with your proper ERP package that's all taken care of
19:17.38billybongoalso tinyerp has the option of a windows/linux gui client or web based
19:17.39*** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com)
19:18.15seldon75hello, a while ago I asked about asterisk's line detection because call terminations weren't being detected.  What if I have the opposite problem; ie: asterisk is terminating calls in the middle of a conversation?
19:18.44lirakisseldon75: .. sip ? iax ? h323 ? tdm?
19:18.51seldon75sip
19:19.38lirakisseldon75: is it random in termination .. ? or is it regular .. after X seconds?
19:19.42kannanseldon75 : codec?
19:20.13seldon75hmm it does seem to happen usually about one minute into the call
19:20.21Aces1Upi have a funny question, if you have a recording recorded at 16khz but use a compression that utilizes a lower compression like 8khz, will you have better quality if you record the sample at 8khz and then send it over the codec or does it matter really?
19:20.28seldon75i dont know what codec - it would be the default if there is one
19:21.03lirakisseldon75: g711?
19:21.20aikanaro79is it possible to differentiate callers that use the same peer but have different callerid (and ip addresses)?
19:22.45seldon75im not sure about the codec but I can find out if you give me a hint how
19:23.07lirakisAces1Up: .. 8khz sampling is higher compression than 16 khz sampling.. khz is how many times the sound is sample per second to digitize it.  G711 uses companding.. which is essentially adaptive sampling (like VBR mp3s) .. based on volume levels.
19:23.34lirakisAces1Up: .. did you mean kbps instead of khz?
19:24.01lirakisseldon75: look at the cli when the call is going through
19:24.46seldon75ok
19:25.12lirakisAces1Up: other codecs (not ulaq/alaw) use significantly more complex algorithms than simply sampling / companding .. so your question would be hard to answer outside of g711
19:27.43Aces1Upthank lirakis, i will try and give you a better example later.
19:28.18*** join/#asterisk ^majik^ (n=kvirc@68-187-20-73.static.uncty.tn.ken-tennwireless.com)
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19:31.26^majik^any idea why zap<->zap channel transfers change the caller id info to that of the person doing the transferring?  I have useincomingcalleridonzaptransfer=yes in zapata.conf
19:36.07seldon75the codec we're using is 'ulaw'
19:36.46seldon75calls get dropped mid-conversation.  seems to be about 1 minute in usually
19:36.47markgreenewhere is the asterisk debug file?
19:36.56Katty[TK]D-Fender: for this sangoma install i'm doing, i still need ztdummy, but i don't need wcdtm anymore, right?
19:37.03Katty[TK]D-Fender: that was just for the analog cards.
19:37.18seldon75markgreene: /var/logs/asterisk
19:37.25[TK]D-FenderKatty: No, you don't need ztdummy, and not you don't need wctdm either
19:37.40Katty[TK]D-Fender: what kernel modules do i need for this a101d then?
19:37.46Katty[TK]D-Fender: nothing?
19:37.53markgreeneseldon75: looked there. Nothing that helps
19:38.03markgreeneseldon75: I remember seeing somewhere a diff log file that asterisk kept
19:38.06[TK]D-FenderKatty: Exactly
19:38.09Kattyoh.
19:38.11Kattyk'then
19:38.20[TK]D-FenderKatty: Wanpipe deposites frame data DIRECTLY into zaptel.
19:38.30[TK]D-FenderKatty: Kinda "cheating"
19:38.35markgreeneDoes anyone in here know if asterisk writes to a debug file somewhere other than /var/log/asterisk/*
19:38.46Kattyoooh neat.
19:38.58Kattyi'm gonna do ztdummy anyway, just cause it'll be a little bit before i get my cards in
19:39.03Kattyor will that mess up the wanpipe install?
19:39.16seldon75markgreene: you can edit the log conf in the asterisk folder
19:39.20seldon75set logging levels
19:39.41seldon75by default ti's not very verbose
19:39.50kannan<PROTECTED>
19:40.03markgreeneseldon75: I will look there thanks
19:40.19markgreeneJust one more time, for fun, is there anyone in here using mysql with asterisk?
19:40.47seldon75markgreene: logger.conf  is the file you want
19:40.58fordfroganybody came across odbc problem with asterisk 1.2.21.1 with ESCAPE ...?
19:40.59Katty[TK]D-Fender: do i need any of the utilities?
19:41.25seldon75yes, it only happens sporadically
19:41.30seldon75kannan: yes, it only happens sporadically
19:41.59[TK]D-FenderKatty: only thing you need is the base wanpipe drivers & ustils it compiles along-with
19:42.34Kattycheers
19:42.48Kattyi'll deselect all the utilites listed then. thanks [TK]D-Fender (=
19:44.07[TK]D-FenderKatty: Deselect from where?
19:44.22[TK]D-FenderKatty: isntall the ones included during "./Setup install"
19:44.48Katty[TK]D-Fender: the make menuselect utilities.
19:45.00Katty[TK]D-Fender: the zaptel one
19:45.11[TK]D-FenderKatty: No, install those too.  You never know.  You aren't saving yourself anything by cutting corners.
19:45.17Kattyoh, okay
19:45.30Kattyi'll make clean and start over, i thought you meant wanpipe would take care of those for me.
19:45.47Katty[TK]D-Fender: but i for sure don't need any of those kernel modules, right?
19:46.31[TK]D-FenderKatty: Correct, not the modules, but the apps are a good idea to have around
19:47.09Katty[TK]D-Fender: alright then. for the utilities i have been using fxotune, zfcfg, ztmonitor, ztspeed, and zttest.
19:47.23Katty[TK]D-Fender: i presume i don't need the fxotune anymore, but do i need any other ones?
19:47.40[TK]D-Fenderkeep them all
19:47.47Katty[TK]D-Fender: all the utilities in the list?
19:47.54Kattym'kay then
19:48.54datachomperIs there anyway to monitor the sip traffic running through SER ?
19:49.08[TK]D-FenderKatty: Sure
19:49.25[TK]D-Fenderdatachomper: wireshark <-
19:50.45datachomperugh
19:51.26*** join/#asterisk [hC] (n=hardcore@66.119.167.162)
19:55.56Katty[TK]D-Fender: the wanpipe wiki page has the install in a different order..
19:56.15Katty[TK]D-Fender: i wish things were consistent :<
19:56.22[TK]D-FenderKatty: ok, fine, sure. :)
19:57.23Katty>.<
19:59.41*** join/#asterisk bkruse (i=bkruse@nat/digium/x-f65c74dab00288ac)
19:59.57NOT_guruanyone ever use the linksys spa962 phone with the 932 sidecar for a receptionist?  you know for classic transfers and seeing who's on thier phones?
20:00.45*** part/#asterisk bkruse (i=bkruse@nat/digium/x-f65c74dab00288ac)
20:03.41seldon75can someone please explain to me how Asterisk determines that a call is over [call disconnect]  is it by voltage, or impedence?
20:03.53seldon75we are getting calls dropped prematurely
20:03.57Qwell[]seldon75: on an analog line, I assume?
20:04.02seldon75yes
20:04.10*** join/#asterisk bkruse (i=bkruse@nat/digium/x-f65c74dab00288ac)
20:04.45seldon75occasionally we see "Power alarm on module 4" on the console but not sure it this is related
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20:08.00*** join/#asterisk mirco (n=mirco@p54B24E49.dip.t-dialin.net)
20:08.44_DAWseldon75: I know its not the same deal, but i had the same problem with an audiocodes FXO gateway.  In then end turning down teh current disconnect threshold fixed the problem.
20:08.49*** join/#asterisk Grapsus (n=grapsus@86.71.77.93)
20:09.38seldon75sounds good
20:09.40seldon75where can I find that?
20:09.48_DAWnot sure with a digium card.
20:10.19seldon75i have a Digium, TDM2400 - anyone know where to set the current disconnect threshold?
20:10.43*** join/#asterisk obnauticus (n=obnautic@c-71-236-219-178.hsd1.wa.comcast.net)
20:13.37[hC]Has anyone played with Shared Line Appearances in * 1.4?
20:14.06KattyDaviey: i'm doing that wanpipe driver thing right now with mister fender's help (=
20:14.17KattyDaviey: i'll have the whole install blogged here in another 10 minutes i'd bet.
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20:18.40lirakishas anyone here actually used a GXP-2020 ?
20:19.14smultroncan asterisk work with an existing PBX phone system?
20:19.26lirakissmultron: yes
20:19.38lirakissmultron: how depends on your existing pbx
20:19.48smultronhm
20:20.21[TK]D-Fendersmultron: Depends on your definition of "work with".  Could you be any more vague?
20:20.23*** join/#asterisk MrMister2 (n=mrmister@89.181.177.127)
20:21.38smultronwell, could i use an asterisk box just for a voicemail to an existing PBX system?
20:21.54DavieyKatty: You are a star
20:22.03KattyDaviey: huh?
20:22.11KattyDaviey: i swear i was only in one movie.
20:22.15DavieyKatty: I think I'll find the wanpipe stuff especially useful!
20:22.24Kattyoh, that.
20:22.45KattyDaviey: well it's gonna take a little longer now.
20:22.59KattyDaviey: mister fender has gone and told me to do it in a different order :P
20:23.11KattyDaviey: so no i'm making clean my wanpipe thing and redoing my documentation.
20:23.43DavieyKatty: Thought about wiking, rather than being a blogbot? :)
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20:24.27KattyDaviey: my blog is personal.
20:24.31KattyDaviey: it's pink!
20:24.37KattyDaviey: wikis aren't exactly personal.
20:24.37Davieyvery!
20:24.41KattyDaviey: ....or pink
20:25.43Katty[TK]D-Fender: so, uhh, after i do my asterisk and addons, i usually launch asterisk.
20:25.52Katty[TK]D-Fender: should i do wan and then launch?
20:26.09daqqalwhen i install asterisk 1.4, how do i get the web based front end for it, is that seperate? is the web front end called openpbx or trixbox?
20:26.15[TK]D-FenderKatty: forget wanpipe right now you don't even have the card!
20:26.20Kattybut
20:26.21Kattybuttttt
20:26.33GrapsusHello !
20:26.46[TK]D-FenderKatty: "how do I start my car?  Oh it hasn't arrived but I want to start it now!"
20:27.04Davieydaqqal: freepbx or asterisk gui if you do it manually
20:27.11luke-jr|workiConnectHere/DeltaThree sucks
20:27.21GrapsusI'm making a web-based GUI for asterisk (laster will be open-source) and I have a little problem
20:27.22Katty[TK]D-Fender: i can't help it!
20:27.22Davieydaqqal: trixbox installs everything, it's a whole custom distro
20:27.31Katty[TK]D-Fender: i'm being spongey here :P
20:27.54daqqalDaviey: ah ok, so trixbox is also asterisk but packeged by someone else?
20:28.10Davieydaqqal: also includes a whole linux distribution
20:28.26Davieywill wipe your hardrive, but install everything to get started in 30mins or so
20:28.41daqqalDaviey: oh no, don't need that
20:28.42GrapsusI need to get a list in realtime of all the calls passing by my asterisk server, I found a way to do it : "asterisk -rx 'show channels'" but the problem is that the first call with the ID is truncated
20:28.46[TK]D-Fender~trixbox
20:28.46jbotrumour has it, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it.  Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org
20:28.54[TK]D-Fenderdaqqal:  ^^^^^^^^^^^^^^^^^^^^^
20:29.55lirakisok .. time for home
20:29.59lirakisbye everyone
20:30.12[TK]D-Fendersame here, bbiab
20:30.20*** part/#asterisk lirakis (n=etamme@65.200.191.253)
20:30.50Grapsusso is there a way to get a list of all open channels with full id ?
20:30.59KattyDaviey: i can go ahead and blog everything up to the wanpipe install if ya want (=
20:31.13KattyDaviey: or, actually, i can dump the small bit i did at the end...and just update it when my card gets in.
20:31.32KattyDaviey: it's something at least, if you want it.
20:31.58*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
20:32.24tzafrir_laptopGrapsus, through the manager interface?
20:32.44datachompera "+" matches an actual characer in an exten definition right? It's not an asterisk wildcard for anything?
20:33.31Grapsustzafrir: yes, I want the full id to join with a 'show channel XXXX' command
20:33.48GrapsusGrapsus: and as it's cut to fit in the call I can't !
20:34.19Grapsuss/call/col
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20:35.38Grapsustzafrir_laptop: cdr shows the full id, but once the call finished, and I want it in real time xD
20:36.18tzafrir_laptopshow channels concise   ?
20:38.55Grapsustzafrir_laptop: thank you ! I should have read the f******* manual
20:41.51*** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr)
20:43.56aikanaro79exit
20:44.53KattyDaviey: i just blogged what i had for today. i'll keep on going through it tomorrow: http://42ndgeekstreet.blogspot.com/2007/08/asterisk-t1pri-phone-server_8986.html
20:46.29*** join/#asterisk SwK (n=SwK@wsip-68-98-207-182.ks.ok.cox.net)
20:46.40KattySwK: :>
20:47.06*** join/#asterisk Toerkeium (i=oo@201.216.206.221)
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20:51.34SwK:P
20:51.45SwKand its not that bad
20:51.56SwKits actually much better then what I thought I would ever get at an airport
20:54.34DavieyKatty: You are such a keen blogger
20:55.52DavieyKatty: I haven't blogeged since mid-may
20:55.53*** join/#asterisk shareenergy (i=shareene@62.169.115.91.rev.optimus.pt)
20:55.56Trevor_bSwK: s/wiki/wifi/ ???
20:56.03KattyDaviey: well, i only blog useful things.
20:56.11KattyDaviey: none of this emo read about my problems blahblahblah
20:56.17KattyDaviey: oh, and things of note.
20:56.28shareenergyhello, can anybody help me with a small question?
20:56.32KattyDaviey: things that really excite me, like my doggy learning UP :>
20:56.35JT~ask
20:56.36jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there, just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
20:56.36Kattyspeaking of my doggy.
20:56.47KattyAnyone have any suggestions on what to teach the pup next?
20:56.49DavieyYeah.. i don't blog personal stuff.. just stuff relevant to the planets i'm sync'd with
20:57.46*** join/#asterisk jordanb (n=jordanb@adsl-68-20-20-59.dsl.chcgil.ameritech.net)
20:58.14jordanbIf I wanted to sell this POS SPA3102 and buy Zaptel stuff, where would be the best place to do that?
20:58.18shareenergycan anybody help me with 2 billion hfc ?
20:58.51shareenergyi have 2 nt boxes
20:59.06shareenergyshould i use nt mode or te on the billion cards?
20:59.17JTthey're not nt boxes
20:59.21JTthey're NT1 boxes
20:59.26JTTE mode
20:59.47Kattydatachomper: what should i teach my puppy next?
21:00.00Kattydatachomper: he knows, sit, down, and now up
21:00.07shareenergyi have already in te, but sometimes it gets activated and sometimes deactivated
21:00.19shareenergyis something wrong in zaptel.conf or zapata.conf?
21:00.30JTjordanb: zaptel won't necessarily be any better
21:00.47JTshareenergy: are you using bristuff, are you using ptmp or ptp?
21:00.57shareenergyi am using trixbox
21:00.59SwKTrevor_b, yeah wifi not wiki... been working on a wiki all morning heh
21:01.04JT~trixbox
21:01.05jbotmethinks trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it.  Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org
21:01.20Trevor_bhehe, was wondering why a airport would have a free wiki ;)
21:01.24shareenergyi've used install-ZAPHF
21:01.24jordanbI've been fighting with the 3102 for two days now trying to get the FXO port to work.
21:01.40SwKheh
21:01.50JTshareenergy: have no idea what you're using, try in a trixbox channel
21:02.07JTjordanb: "work"?
21:02.40jordanbJT, Occasionally I'll be able to dial out, but 95% of the time it gives me a busy signal after ringing once.
21:03.00JTin band or out of band?
21:03.10jordanbJT, I also get the same behavior from it when I don't have asterisk running and it falls back.
21:03.32jordanbIn band I believe.. I can hear it on the phone. I'm new to asterisk. :<
21:03.36shareenergyJT: it is bristuff
21:04.19JTshareenergy: if you get rid of trixbox, i can help you, otherwise not worth my time trying to decipher it
21:04.46shareenergyno problem, i can do it on shell, without trixbox
21:05.04JTi mean reinstall with no trixbox.
21:05.14JTjordanb: sounds like an SPA config issue
21:05.23jordanbI've been trying to upgrade the firmware but the program is windows only and the only windows I have access to is in qemu. That wouldn't be a problem except the fuckers set it up so the SPA access the windows box on a UDP port, so I can't use my nated TAP, so now I have to figure out how to do a TUN.
21:05.40*** join/#asterisk |dennis| (n=dennis@200.32.236.20)
21:05.50jordanbJust so I can install a firmware that probably won't fix the problem. I've been two hours at it.
21:06.26*** join/#asterisk Here_And_There (n=Here_And@pool-68-238-242-246.phlapa.fios.verizon.net)
21:06.35jordanbJT, Pretty much the only thing I've done to it besides setting up the network was using that auto configurator on Voxilla.
21:06.41jordanbAnd the fallback wasn't working before I did that.
21:07.24*** join/#asterisk l-fy (n=diana@yate/developer/l-fy)
21:08.04jordanbAlso it's intermittent, it'll let one phone call through and then no others. I thought it was the line staying off the hook or something but the network interface and the syslog messages both show the line going back on hook when I hang up.
21:08.16jordanber, web interface.
21:08.33*** part/#asterisk l-fy (n=diana@yate/developer/l-fy)
21:09.14NOT_guru~ATA
21:09.15jbotata is probably Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA
21:09.27NOT_guru~sangoma
21:09.28jboti guess sangoma is a company that makes PRI cards
21:09.40jordanbI think the lesson is to not buy a closed peice of hardware from a group of slimeballs like linksyss when open hardware is avaliable for a little bit more money.
21:09.40NOT_guru~jbot
21:09.41jbotjbot is, like, a hack!, or known to have only said one useful thing.
21:15.52*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
21:15.54Kattyjbot: :>
21:16.08Kattyjbot: :<
21:16.09jbot< is probably redirection of stdin to a program
21:16.19Kattyjbot: you're silly.
21:16.33Kattyjbot: I love you, anyway.
21:16.33jbotYou love you, anyway.?
21:16.39Kattyjbot: yes.
21:16.40jbotYou don't say!
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21:28.20rbdhi guys, I have asterisk interacting with a cat6500 CMM module, talking g711. Problem is that the CMM is trying to use comfort noise and asterisk doesn't seem to support it. I am thinking that we need to disable VAD/CNG on the 6500 at least to the asterisk peer. anyone know of any resources on how to do this?
21:29.59*** join/#asterisk claudiotainen (n=claudiot@ppp-150-178.33-151.iol.it)
21:30.36claudiotainenI'd like to access a remote asterisk server
21:31.02claudiotainencan I do that using asterisk -r? if so, how do I have to use that command ?
21:31.19*** join/#asterisk Innatech (n=it@netblock-68-183-140-137.dslextreme.com)
21:31.29billybongoclaudiotainen: define "access"
21:31.44claudiotainenwhere ?
21:31.56billybongowhen you say you want to access a remote asterisk server
21:31.56claudiotainenoh sorry
21:32.01claudiotainenok yes
21:32.10claudiotainenI want to use CLI form a remote PC
21:32.16billybongossh ?
21:32.33claudiotainencan't I simply using asterisk -r ?
21:32.43billybongoI think that talks to the unix socket
21:32.52billybongoin any case would you want anyone to be able to do that?
21:33.17claudiotainenwell the thing is that the server is not mine
21:33.28claudiotainenI mean I'm working on a univ project
21:33.43billybongoI'm pretty sure asterisk -r connects to the unix socket, which is just a file
21:34.10billybongoor at least it's on the filesystem
21:34.22billybongoso I doubt you're going to be able to use that to connect remotely
21:34.33claudiotainenoh
21:34.36billybongoI don't really see how you can look after your asterisk unless you get ssh access
21:34.59claudiotainenwell ssh must be configured on the remote server
21:35.10billybongoin any case if it did listen on a port I think that would be insecure to open up
21:35.35billybongoif it's just for a university project then run it on your own computer
21:35.41billybongoand upload when you're done
21:35.57claudiotainenno it's a group project
21:36.10claudiotainenand the server's not at my home
21:36.21billybongothen you need ssh
21:36.39claudiotainenI used to access it using an html interface build with php
21:36.45claudiotainenbut now that's gone
21:36.52billybongothis all sounds scary
21:37.02claudiotainen:) why scary ?
21:37.10billybongoI live on ssh
21:37.14billybongocouldn't do without it
21:37.41billybongoall these other ways sound more complicated and less secure
21:37.50billybongoand I would imagine ssh is already installed anyway
21:38.07billybongoas is the case on most servers of my acquaintance
21:38.14claudiotainenno no I really think it isn't
21:38.16claudiotainenthe thing is
21:38.20billybongowhat OS is on it?
21:38.24claudiotainenubuntu
21:38.38billybongoahh well maybe not out of the box
21:38.47billybongonothing a quick apt-get install openssh won't fix though
21:40.30claudiotainenwell thank you anyhow
21:40.48claudiotainenI'll spent the rest of this night trying to contact serv admin ;)
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21:46.52mepplgood night
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21:56.49dioedusomeone can tell me if i have a bug ? i have a queue and when i have a blind transfer in this queue, the originate call stay up until the call on the other queue hang up. If i have a attended transfer, this doesn't happen.
22:10.07JTjordanb: what are you going on about? what open hardware alternatives?
22:10.54*** join/#asterisk Won4him_ (n=chatzill@67-132-248-66.dia.static.qwest.net)
22:12.29jordanbI think I'd be a lot better off with a TDM400P.
22:13.20jordanbAnd I wouldn't have had to buy a mitel box to use my WE telephone because Linksys are too stupid to make their device handle pulses.
22:13.24*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
22:13.32Won4him_can anyone answer a question about * and VM configuration
22:13.45Hmmhesaysoh this is fscking brilliant
22:14.00shido6ask
22:14.01Hmmhesaysa data entry with a comma, in a comma delimited file.
22:14.09shido6heheh
22:14.09Hmmhesays@#$@#
22:14.23Hmmhesaysdefault cdr-csv is retarded
22:14.24Won4him_I get an error  that says
22:14.25Won4him_<PROTECTED>
22:14.29*** join/#asterisk Grapsus (n=grapsus@86.71.77.93)
22:14.46Won4him_but extensions 2992 exists in default context
22:14.57Won4him_version 1.4.9
22:16.01JTjordanb: the TDM400P is not open. it's also not very good. pulses are stupid.
22:16.05Grapsusis there a special option to make asterisk load configuration from ODBC automatically, when I load it 'show dialplan' doesn't contain my extensions, if I do a manual 'reload' it works...
22:16.12JTseriously, get a dtmf phone
22:18.42jordanbI have a dtmf phone, it's a cheap ass peice of junk like every other phone made after about 1985.
22:19.04Won4him_any ideas on the Voiemail error
22:19.27shido6cronjob a asterisk -rx 'reload' or something :)
22:20.39generalhanhey all ... i need to grab the extension of a phone dialing out ... i have setup something from back in the 1.0.9 days and im trying to find a better way to do it... can anyone think of something better than this:   http://generalhan.pastebin.ca/643596
22:21.40Grapsusshido6: yes I know, but that's dirty
22:21.53Hmmhesayswhy the crap would you do this.
22:21.56Hmmhesaysinsane!
22:22.08generalhanHmmhesays: to me ?
22:22.22JTjordanb: ok, if you think so. I suppose only bakelite phones are good enough?
22:22.42Grapsusshido6: I trying to debug that, in fact at start it does an SQL error when trying to fetch extensions.conf in database, but I wonder why it works when I reload...
22:23.02Hmmhesaysno why asterisk cdr-csv would put a data field with a comma, in a comma delimeted file!
22:23.24generalhanwhich field ?
22:23.42jordanbThey quit making things out of bakalite like 70 years before 1985.
22:23.53jordanb1985 was when the WE500s started getting cheap.
22:24.01jordanbMaybe a little before that.
22:25.21JTjordanb: yeah so anyway, use a dtmf phone, unless it's your life goal to be obtuse
22:25.37generalhanlol ! that IS my life's goal !!!!
22:26.30jordanbI think it'd be better off trying to cut my losses on this Linksys junk on ebay or something and buy some real phone hardware. I'm using my DTMF phone now and still have the dialing out problem.
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22:26.48JTjordanb: real non free phone hardware... right...
22:26.55JTjordanb: sounds like a configuration issue
22:27.10JTthere are tens of thousands of linksys ATAs out in production, working fine
22:27.12jordanbWhat is the misconfiguration then?
22:27.34JTso instead of getting angry and making silly accusations, how about working out what the problem is? :)
22:27.41JTdigium hardware is not open hardware
22:27.50jordanbI'm still trying to get a tun bridge setup to qemu so I can try to get linksys's insanely broken firmware update software to work.
22:28.17JTso you're doing something insanely complex and you're wondering why it fails to work
22:28.49jordanbLinksys's software is stupidly misdesigned. I wouldn't have to do this otherwise.
22:29.11jordanbeverything else works fine with a simple natted tap.
22:30.31JTforget about qemu with firmware updates
22:30.33sweeperhey, where can I find the FastAGI documentation? like, the stuff you'd use to write a FastAGI library in x langugae?
22:30.33Innatechyou might consider the possibility you have buggy hardware. I haven't read the scrollback, but I have a WRT with stock FW that I have to reboot every couple hours. Sometimes you get a bad unit.
22:31.12jordanbIt's the only way I can do it.
22:31.18jordanbI don't have windows elsewhere.
22:31.19*** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au)
22:31.33JTthen get windows elsewhere
22:31.42JTit's the simple solution
22:31.45jordanbYeah that's reasonable.
22:32.29jordanbI'll let the windows installer nuke my linux system just so I can get a firmware into a peice of linksys junk that probably wont' fix it anyway.
22:32.54JTget another system, or dual boot, or something
22:33.08JTbut you can't expect much sympathy for using crackpot unsupported configurations
22:33.14jordanbHow is that the simple solution, or even remotly usable?
22:33.27JTbecause it's a known good solution
22:33.29jordanbAsterisk is a crackpot unsupported configuration from linksys's perspective.
22:33.35jordanbMaybe I should just buy vonage and be happy.
22:33.51JTand i'd rather do that than screw around with qemu to perform such a task
22:33.53DavieyInnatech: do you also find that if you use ping through a WRT router, some packets spike to stupidly high amounts?
22:33.55JTmaybe you should
22:33.58Innatech./hereticalpitchforkbrigade -initialize
22:34.06sweeperDaviey: not if you use dd-wrt :)
22:35.40Davieysweeper: i am
22:35.51sweepero.O
22:35.55sweeperwhat version?
22:36.01Davieyerm..wait 1
22:36.25InnatechDaviey - I find when I ping through a WRT, all kinds of ugliness ensues. Still, it's hard to find readily available alternatives w/o getting into DIY with a miniPCI card. In the end, I buy DD/OpenWRT compat. units and flash them. I see no end of problems from the stock FW. And, FYI, I keep a crapola W2K notebook around just to flash them .
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22:36.58sweeperwhy do you need windows? o.O
22:36.58Davieysweeper: Firmware: DD-WRT v23 SP2 (09/15/06) std
22:37.29sweeperDaviey: well, there's a newer version, but I still think it's strange you get high pings
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22:38.07JoseBravoIm doing  make zttool but I get this error: zttool.c:40:18: error: newt.h: No such file or directory
22:38.13JoseBravoAny idea??
22:38.23JoseBravoIts in zaptel.
22:38.28JTinsall lib newt
22:38.31JTinstall
22:42.11JoseBravoJT its installed
22:42.52JTthe dev library?
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22:44.50Davieysweeper: bah.. it'snot doing it now - it's got a slight rise to .300ms from the normal ~.150-.170ms
22:44.53Davieyrtt min/avg/max/mdev = 0.136/0.176/0.328/0.038 ms
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22:45.31sweepercool
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23:02.15rhiliamHello, I am trying to figure something out. I need to make a call on a zap channel, but if that zap channel is in use, then I need to place the call on another specific channel - how would I do this?
23:02.51generalhanrhiliam: but another Dial statement on the next priority
23:02.58ManxPowerrhiliam: is that other channel also a zap channel?
23:03.06ManxPowerNOT ON THE NEXT PRIORITY!!!!!!
23:03.21generalhanrhiliam: dont listen to me ! lol
23:03.41ManxPowerAs the priority after your Dial check the status of DIALSTATUS *then* determine if you need to Goto another Dial statement.
23:03.59generalhanManxPower: sorry .. that makes more sense i guess !
23:04.31ManxPowerAn example of another way is in extensions.conf.sample as [macro-std-exten]
23:05.28rhiliamlet me check the sample. Essentially, to make a LD call, dial zap 1, if zap 1 is busy, then use Zap 2
23:05.41ManxPowerrhiliam: there is a much easier way
23:05.42generalhanManxPower: if you have a minute or two .. can you take a look at this:  http://generalhan.pastebin.ca/643596     i wrote this back in the 1.0.9 days and im hoping there is something better that i can do now .. or even then that i didnt know about ! lol
23:05.59rhiliamccol - all ears
23:06.07ManxPowerrhiliam: see the group= setting in zapata.conf
23:06.23ManxPowerput both channels in a group (group=1, for example) then Dial(Zap/g1/whatever)
23:06.36rhiliamcan zap channels be assigned to more than one group at a time?
23:06.51rhiliamthey are all in group 1 at the moment and this really cant change
23:07.05*** join/#asterisk kimosabe (n=kimosabe@189.175.44.143)
23:07.12ManxPowergeneralhan: that is not 1.0.  1.0 does not use Set or "n" priority.
23:07.15*** join/#asterisk ccesario_ (n=ccesario@201-0-52-108.dsl.telesp.net.br)
23:07.19ManxPowerrhiliam: channels can be in more than 1 group
23:07.37ManxPowergeneralhan: But your pastebin is pretty much how I would do it.
23:07.40kimosabehas anyone got the sipura 3000 to work as a fxo with asterisk if so can you please lead me in the direction of a good how to please
23:07.50ManxPowerchannel => 1
23:07.56ManxPoweractually....
23:07.59rhiliamso is it as simple as group=1,2?
23:08.14generalhanManxPower: lol you're right 1.2.10 ... i dont know why i thought i was still using 1.0.9
23:08.37generalhanso thats how you are doing it too huh ? :( i was really hoping there was something else.
23:08.40ManxPowerrhiliam:  http://generalhan.pastebin.ca/643653
23:08.58ManxPowergeneralhan: well I don't record calls.
23:09.15generalhanManxPower: right ... but to pull the exten out of the SIP string
23:09.36rhiliamexcellent - thanks much
23:09.38ManxPowergeneralhan: See http://www.fnords.org/~eric/macro-std-exten-v2.inc for some really twisted stuff
23:09.45generalhanlol ! k
23:10.20*** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil)
23:10.27ManxPowergeneralhan: The you can't Set(SAVED_EXTEN=${EXTEN}) somewhere before or extract the info from the callerid?
23:11.08rhiliamOne last question - is there a way to detect if a user is on the phone (extension) without dialling the extension? I guess what I am asking is there a way to track prescene?
23:11.24ManxPowerrhiliam: "show applicaion chanisavail"
23:11.53generalhani guess i could ... but then i would have to add a Set(Callerid line to each dial out context ... cause i just set the CID in sip.conf as what it should show on receiving CID machines. so if i set CID as 7010, then i would have to set it to the number i want to show right before the dial statement
23:11.57rhiliamFrom a user perspective. Essentially an admin wants to know if "john' is on the phone, prior to trying to transfer a call to him.
23:12.11rhiliamadmin/secretary type person
23:12.55ManxPowerrhiliam: you would need real presence using HINT and a phone capable of displaying that like a Polycom or use the Flash Operator Panel
23:12.55kimosabeanyone here using the spa3000 for fxo pstn use if so please send me in the direction of good how to would be apreciated thanks ...
23:13.15rhiliamWe are using Polycom Soundpoint 430's. I did read up on HINT, but.....
23:13.49rhiliamto be honest, didn't make alot of sense to me, and I couldn't find an example.
23:14.14generalhanManxPower: i wonder how long it took him to sit there and do all that !! lol. if it was useful to me and i wanted to send him a donation i would have to pay him for like 100 hours of work, casue thats how long it would have taken me to do that on my own ! lol
23:15.08ManxPowergeneralhan: I wrote that in a couple of days.
23:15.22generalhanoh thats yours ?
23:15.40ManxPowerYes.  But it was based on something I wrote a couple of years ago
23:15.54generalhanyea thats nutz !
23:16.03ManxPowerIt is incredibly useful
23:16.21generalhanand im worrying about the best way to pull the extension for recording labels ! lol
23:16.53ManxPowergeneralhan: I've been using asterisk since 2003 I think.
23:17.54generalhanManxPower: well i started in '04 ... but im the "set it and forget it" type. so i set something VERY basic and then moved on to something else ... then on downtime at this startup company i would go back and put in little useful features here and there
23:18.06coldstealhello im having some problems here are my errors and configs http://rafb.net/p/qqw7ba16.html
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23:19.26rhiliamexit
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23:22.37WilliamKwhat's the most common way for users to login to multiple queues on a Cisco 7940/60?  my thought would be utilizing the soft keys, but I've found limited resources so far describing howto
23:23.22Corydon76-workWilliamK: Using SIP or Skinny?
23:23.48WilliamKSIP
23:23.57Corydon76-workYou cannot use the softkeys, then
23:24.09WilliamKoh great :)
23:24.13WilliamKknow any other options?
23:24.19Corydon76-workHence why Cisco came out with the 7961... more memory
23:24.49Corydon76-workBasically they ran out of memory and couldn't code functionality for the buttons
23:25.08Corydon76-workUse an extension and run AddQueueMember
23:25.38WilliamKnice.... so I'm guessing it's almost easier to use SNOM phones to make it look more like a guenuine PBX setup?
23:26.05Corydon76-workMy personal opinion is that Cisco makes overpriced crap
23:26.21Corydon76-workMost people prefer either SNOM or Polycom
23:26.23WilliamKthat's my opinion in alot of ways too right now
23:26.25JTWilliamK: don't go with snom, go polycom
23:26.38Corydon76-workWe use both for our customers
23:27.00[hC]I prefer polycom or aastra
23:27.05WilliamKJT, I just mentioned SNOM because I used to use a 190 prior
23:27.08[hC]for different reasons
23:27.13[hC]they both have their problems
23:27.18JTpolycom and aastra are the top sip phone brands, with polycom being king :)
23:27.48Corydon76-workSNOM 320 is the base model we sell of the series
23:27.52WilliamKmy client has basically 4 queues, that they need to be able to selectively login/logout of queues so I'm trying to get as close as I can to the real pbx functionality
23:28.22JTsnoms are overpriced and poorly styled imho
23:28.48Corydon76-workYes, but the SNOM 360 is the only way you're going to get close to 100 line appearances
23:28.59Hmmhesayspain in the ass
23:29.01JTaastra...
23:29.02WilliamKI remember when poly didn't care about * (long time ago)
23:29.02Corydon76-workThe Polycom maxes out at around 50
23:29.10JTthe aastra can do more
23:29.25Corydon76-workwhich aastra?
23:29.34JT57i i think it is
23:29.50kimosabeim trying to get 2 fxo interfaces to work with my sips location 1 is mexico i have asterisk box there with 2 spa3000 i want to put my pstn trunks in the spa and send them to a rural area where i have internet so that i can recieve calls and make calls i have the sips working i can call my self but i want to be able to use the fxo interfaces can some one help me out please
23:30.14[hC]the 57i can get.. lets see.
23:30.18[hC]the one i have on my desk
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23:30.31[hC]60 buttons per sidecar, 3 sidecars
23:30.42[hC]180
23:31.06[hC]20 physical buttons per side car, and 3 'page' buttons to scroll thru different views
23:31.17[hC]but i have opinions on performance when you load them up, the phone seems to like to barf
23:31.23[hC]maybe just the firmware i have
23:32.01WilliamK8 if I got happy enough to write something more creative
23:32.03WilliamK:)
23:32.04JTkimosabe: that should be 3 or 4 sentences, i can't understand that
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23:32.19hi365_manyone using a celulink gsm-fxs adapter?
23:32.30hi365_mim not reciving caller id from it :(
23:32.44De_Monhrm, has anyone ever seen this before:
23:32.45De_MonWARNING[3796]: translate.c:163 framein: no samples for lintoulaw
23:33.05JThi365_m: those things are dodgy, i wouldn't expect them to work nicely
23:33.23hi365_mjt - youve ued them?
23:33.35hi365_m*used
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23:34.32JTno, but how reliable can they be? GSM --> POTS --> asterisk = not at all optimal
23:37.00hi365_mJT - can you recomend something better?
23:37.36JThow many gsm connections?
23:37.57hi365_m1-2
23:38.30JTthere is a beta bluetooth channel driver chan_cellphone in svn trunk i think
23:38.39JTuse normal bluetooth phone as a channel
23:38.54hi365_mthats a good reson not to use it :)
23:39.02JTwhy?
23:39.14hi365_mright. how about something sip?
23:39.20hi365_mcause its beta :)
23:39.29JTwell it seems decent
23:39.34JTand you have proper signalling
23:39.36JTunlike analogue
23:39.47JTsip, don't know of much for only 1 or 2 ports
23:39.57JTsip gateways are really expensive
23:40.50hi365_mi see. maybe its worth a try.
23:41.32JTmaybe you have the wrong mode set
23:42.03hi365_mgo ahead - throw some settings at me. i think ive tried every setting in the book!
23:42.20JTyou've tried all the callerid types?
23:43.15hi365_mi think - bell, v23,dtmf  - on polarity/ring
23:43.48JTshow us the lines in the dialplan for incoming calls from gsm
23:44.54hi365_mits the same as from regular pstn - which has working callerID.still wana see it?
23:44.59WilliamKso what phone can I use that's simple to setup for agent login/logout from the queues without having to do custom programming on the phones?
23:45.15JTyes, and what is the regular pstn connection?
23:46.10JTare you using an fxs or an fxo port on asterisk to connect to the gateway?
23:48.02hi365_mfxo
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23:48.16hi365_mgive me a few to dif up the configs
23:49.16JTwhat sort of gateway is it, model number, link?
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