IRC log for #asterisk on 20070606

00:02.57seanwganyone on here using voiceme up?
00:03.11seanwgi signed up - whenever i make calls, i get the nuber you are trying to reach is out of service
00:03.11*** join/#asterisk doseyg (n=doseyg@h249.221.89.75.ip.alltel.net)
00:03.14seanwgand no inbound calls
00:05.59doseygInternally I have 2 extensions which I would like to both be able to dial out via a single SIP connection. Is this possible ?
00:10.33baprilAre there any known issues (1.2) recording (with monitor) a 2 user Dynamic MeetMe between 2 Zap callers? I only get one channel recorded with 2 zap channels. Works file Sip-Zap.
00:13.35*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
00:13.35*** mode/#asterisk [+o anthm] by ChanServ
00:14.05*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
00:15.49*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
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00:21.19aravaHi guys , Can someone please help me configure my openvox a1200p card. i got the module up and the leds are glowing, Will i get any dial tone on my phone ?
00:23.08aravaAny one ?
00:33.09*** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au)
00:33.15*** join/#asterisk pabs3 (i=daemon@60-242-186-48.tpgi.com.au)
00:34.18ManxPowerarava: unfortunately OpenVox is not a popular card, so there are few people that can help you.  Check the mailing list archives.
00:34.18*** part/#asterisk saftsack (n=saftsack@pD9E0493F.dip.t-dialin.net)
00:34.23ManxPower~mailinglist
00:34.30jbotSearch Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm
00:35.07*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
00:37.30*** join/#asterisk i3inary (i=i3inary@ip72-207-116-178.sd.sd.cox.net)
00:38.54i3inaryhi guys i hate to ask such a stupid question but i cant find my answer anywhere...is it possible to accept dtmf input while you are using WaitMusicOnHold() ?
00:39.39i3inaryor even MusicOnHold?   i realize i could use a long prompt but i assumed that musiconhold might be more efficient...?
00:40.46doseygDo I need to configure a Trunk to accomplish what I want ?
00:41.40pabs3I've a whole bunch of rules in extensions.conf for different types of outgoing PSTN calls, is there an easy way to set the callerid for all of them based on the sip account that is calling out, so that the outgoing callerid is the same as the indial number to reach the sip account that is calling out?
00:44.32[TK]D-Fenderi3inary, Nope.
00:44.54mockerHmm.
00:44.58i3inaryD-Fender, thanks
00:44.59mockerMy plans might be filed.
00:45.14[TK]D-Fenderpabs3, thers some tweaking to be done, but its not rocket-science
00:45.26mockerI just shared a voicemail directory via NFS and now when I call to check for new messages there is a definite 'pause' as it checks..
00:45.39mockerer, foiled not filed.
00:46.20mockerGreat, that time it was like 15 seconds.
00:46.21mocker:(
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00:50.13*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
00:51.37ptiggerdineis dwmw2 working getting zaptel into the kenrel?
00:52.12*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
00:54.34crimethinkerit isn't in the kernel yet?
00:55.04*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
00:59.34frocos11292anyone know any click2dial tool?
00:59.46[TK]D-Fenderfrocos11292, Apache :)
00:59.56QwellI would've said mouse
01:00.30[TK]D-Fenderqwell : "This web-site is best viewed with... YOUR EYES!"
01:02.05frocos11292<[TK]D-Fender> : apache!?
01:02.56[TK]D-Fenderfrocos11292, Yes, it serves up web pages.  So make a scripted page that uses AMI to Originate a call :)
01:02.58[TK]D-Fender~ami
01:03.01jbothmm... ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API
01:03.36frocos11292<[TK]D-Fender> : it doesn't work
01:03.50frocos11292<[TK]D-Fender> : it's for a website so anyone can click and talk to me
01:04.25[TK]D-Fenderfrocos11292, .... and how is that not EXACTLY what I said?!
01:04.54frocos11292<[TK]D-Fender> well.. what about audio?
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01:06.16frocos11292<[TK]D-Fender> it works for callback, but for click2dial i don't see how...
01:07.22[TK]D-Fenderfrocos11292, I guess if you're expecting them to use their own speaker & mic you'd have to use some sort of Java soft-phone or the like.
01:08.38frocos11292<[TK]D-Fender> yep. i know iaxlclient can do it but i'm having problems using it cause there isn't much information on the ner
01:08.40frocos11292net
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01:11.42*** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net)
01:12.09RyanWHello, I'm having troubles getting MWI working after updating the firmware in my GXP2000
01:14.35*** join/#asterisk komradebob (n=komradeb@164.55.254.106)
01:14.49RyanWcan someone tell me why MWI doesn't work in this case please. http://pastebin.ca/543252
01:16.01komradebobany clue why exten => s,n,Set(ENUM=${ENUMLOOKUP(+${ARG1},ALL,1,mydomain.com)}) is still trying to look up in e164.arpa?
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01:28.32sevardRyanW: How is Victoria these days?
01:34.54RyanWsevard, I wouldn't know I'm in Gosford NSW
01:36.03JTRyanW: wow, totally can't tell from your hostname either....
01:36.04JTus host?
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01:48.31joe[TK]D-Fender: have you played w/ the c100?
01:49.07ManxPowerC100?  And here I am stuck on a C64
01:49.16joepolycom c100
01:49.23ManxPowerwhat is it?
01:49.38joehttp://www.polycom.com/usa/en/products/voice/mobile/communicator_c100.html
01:50.04*** join/#asterisk bkunyiha (i=Billk@66-113-79-5.rev.ibsinc.com)
01:50.27ManxPowerOh.  Those.
01:50.39bkunyihaI have an AGI question
01:50.39bkunyihaI have the following in extensions.conf
01:50.39bkunyihaexten => 105,1,Set(this_location=${DUNDILOOKUP(105|priv)})
01:51.22ManxPowerJoe: chances are it works like most any other softphone (which is what it really is)
01:52.02bkunyihaand would like to do the same in AGI  but
01:52.02bkunyiha$this_location=$AGI->get_variable("DUNDILOOKUP(105|priv)");
01:52.02bkunyihadoes not seam to work.
01:52.35ManxPowerIn fact it does not even come with SOFTWARE.
01:53.13ManxPowerbkunyiha: DUNDILOOOKUP is a function, not a variable.
01:53.45bkunyihaI know
01:53.57bkunyihaWhat sould be the return type
01:53.58ManxPowerJoe: x-lite and eyebeam support is in beta
01:54.22ManxPowerbkunyiha: I have no idea.  Have you check the mailinglist archive?
01:54.23ManxPower~mailinglist
01:54.26jbotSearch Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm
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01:59.36joeManxPower: Yeah, I know it's a softphone... would you have an url about the beta status?
02:01.57joenm google to the rescue :)
02:02.51joeManxPower: thanks for the info... now to read about x-lite and eyebeam
02:02.56SomethingISODDanyone here ever used the wireless lan access point?
02:05.18*** join/#asterisk btsteve (n=btsteve@75.131.197.123)
02:06.04btstevehas anyone had a problem compiling zaptel drivers under centos 5?
02:06.49Juggieyeah, run make menuselect and disable the module that errors while compiling
02:06.55Juggietheres a good chance you do not need it
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02:09.17btsteveok i was just informed that running svn and compiling fixed that problem but no we are having a modprob problem on our dell server.
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02:13.25joeManxPower: any gpl good softphones out there?
02:14.16joeI know there are quite a few out there... just not very good so far
02:14.52joeI've been waiting for something like the c100 w/ noise canceling and a good softphone
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02:22.33pabs3is it possible to have a call queue where agents don't need to login to be called?
02:23.47*** join/#asterisk abrdeco (n=root@c9113931.rjo.virtua.com.br)
02:24.32JTjoe: pretty much no such thing as a good softphone
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02:40.36abrdecohi all, can anybody helps  me with queues issue ? when the call stills into the queue the caller listen MOH normally, but when answered by extension no audio at all.
02:42.07*** part/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net)
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02:45.11tzafrir_laptopabrdeco, I don't really think I can help you, but maybe a trace from the CLI can help. pastebin it
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02:50.30abrdecoCalls directed from incoming route directly to the extension works fine, but when the incom,ing call enter in the queue and directed to extension i've got no audio.
02:50.56abrdecoIt was working fine yesterday.
02:51.27abrdecoWhat king of trace do you suggest ? dddvvvg ?
02:52.51tzafrir_laptopset verbose 4
02:53.31mockerSo for a dundi configuration you list all the servers in the dundi network in the dundi.conf, correct?
02:53.36*** join/#asterisk fbffff (n=fbffff@c-67-167-98-42.hsd1.il.comcast.net)
02:53.38tzafrir_laptopand see wht you have on the CLI (or log the same that goes to "console" to a log file)
02:53.43tzafrir_laptopin logger.conf
02:54.05tzafrir_laptopmocker, right
02:54.29abrdecoOk, i will do that. I think it is an queue issue because all works fine without the queue and before the call be answered by extension.
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03:02.48burt75hello to all
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03:04.13burt75guys anyone knows how to send automatically a # as a End-of-Dial to a Zap?
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03:11.33tzafrir_laptopburt75, use a proper dialplan?
03:13.08tzafrir_laptopwell, that is: avoid using something of the sort of '_XX.' (the dot) in your dialplan
03:13.31tzafrir_laptopok. "proper" may not be the right word here
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03:20.46burt75tzafrir but any idea if exten => _XXXXXXXXXX,1,Dial(ZAP/g0/01${EXTEN}#,55)
03:20.48burt75might work?
03:21.14*** join/#asterisk CuriosCat (i=stian@mack.bigrig.org)
03:21.19CuriosCathey all.
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03:34.58tzafrir_laptopah, you actually need the extra #?
03:35.13tzafrir_laptopthen sure you can add it. I thought you wanted to strip it
03:35.24tzafrir_laptop(which is not a real problem)
03:35.44tzafrir_laptopand to stop the dialtone when it is pressed (which is a bit more tricky)
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03:55.55*** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits, #astridevcon -=- Join #freepbx for freepbx/#trixbox for trixbox support.
03:56.52mostyburt75, can you use Read followed by a Dial?
03:57.13tzafrir_laptopburt75, is this PRI or analog?
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04:02.00mockerWow, dundi pretty much means rewriting my entire dialplan.
04:04.46mostymocker, do you know of a way to add qualify data to dundi responses? i only want to return a valid result if the peer at the specific number is reachable
04:06.38mockermosty: Nope, tonight's really my first night digging in.
04:06.45mockermosty: Lots of coffee, etc...
04:07.18mostywell i would be interested to know how you find it
04:07.27mockermosty: But you're just advertising extensions, I don't know if it really cares about if your user has registered or not.
04:08.12mostymocker, i need to know if the user has registered, trying to avoid writing a multihost peer cache DB myself
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04:14.42RyanWin the output from the command "show queues" what does W:0, C:4, A:2, SL:0.0% within 0s represent ?
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04:15.24mostysomething like waiting/calls/answered/service level
04:16.57CuriosCatHrm
04:17.18CuriosCatSeveral places talk about running "zttest" to verify that my TDM400P FXO is working correctly
04:17.28CuriosCathowever, none seems to tell me what arguments to supply to zttest :P
04:17.48JTno arguments
04:17.51JTyou just run it
04:18.09CuriosCatUnable to open zap interface: No such device or address
04:18.14CuriosCatso obviously, something's amiss
04:18.20JTzaptel is not setup
04:18.29CuriosCatwhat should this card look like in lspci, to just start with the basics?
04:18.34_VoiceMeUp_COMand /dev/z* there
04:18.41CuriosCatJT: Correct. It's just recently installed.
04:19.48Trevor_bCommunication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
04:20.10Trevor_bThats how mine looks, but i usually go to /proc to ensure how its loaded into the system
04:20.29JTTrevor_b: zaptel needs to be configured and running too
04:20.37Trevor_bcat /proc/zaptel/1 and /proc/zaptel/2
04:20.39Trevor_bright
04:20.39CuriosCatah.
04:20.44CuriosCatwctdm
04:20.51CuriosCathad to modprobe wctdm
04:20.56Trevor_bbut you start out with the basics, make sure its loaded, then u can worry about udev and init.d issues.
04:21.13Trevor_bwell basics
04:21.14Trevor_bnot really
04:21.17JTyou need to run ztcfg too
04:21.18Trevor_bbackwards
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04:21.29JTyou need to read the documentation for zaptel, CuriosCat :)
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04:21.56Trevor_bi think of it in logical working components, not from the other way, had too many issues with weird systems not loading zap hardware (old p3 boards and whatnot).
04:22.02CuriosCatJT: Yeah, you're right. I was just eager to verify the card detected correctly, at least
04:22.25Trevor_bthat and trixcrap causing udev issues before that ;)
04:23.45CuriosCatBest: 99.926758 -- Worst: 99.914551 -- Average: 99.923706
04:23.53CuriosCatok, so zttest seems to think the card is working well now.
04:24.15CuriosCatNow I need to figure out /etc/zaptel.conf. That looks like a nightmare of a config file.
04:24.24wunderkinumm
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04:24.31wunderkinthat zttest is pretty craptastic
04:25.31russellbCuriosCat: what card do you have?
04:25.57CuriosCatrussellb: TDM400P with a single FXO module
04:26.05russellbCuriosCat: what country are you in?
04:26.09CuriosCatUS
04:26.30russellbok, zaptel.conf is this ..... 3 lines ...... loadzone=us ... defaultzone=us ... fxsks=1
04:26.59russellbthen run "ztcfg -vv" as root, and you should see that the channel gets successfully configured
04:27.39CuriosCatseems to have worked, russell
04:27.40CuriosCatthanks
04:27.44russellbyou're welcome
04:28.04CuriosCatnow...if the zttest is craptastic, how do I improve it? (I started reading something about ensuring the card has a dedicated IRQ)
04:28.17russellbthe IRQ thing is no longer an issue
04:28.23russellband i think those values look fine
04:28.58wunderkin99.92?
04:29.02CuriosCatNow...I do have a regular POTS line plugged into that FXO module. Is there a way to actually verify dialtone?
04:29.26russellbfor /etc/asterisk/zapata.conf, add a "signalling=fxs_ks", "callerid=asreceived", and "channel => 1" to the end
04:29.45russellbthen when you run asterisk, you should be able to call the line and see it show up on the Asterisk CLI
04:31.08CuriosCatasterisk
04:31.27russellbyes, that is what it is called :-p
04:33.14*** part/#asterisk Defraz (n=t0tal@fw.fuzecore.com)
04:33.19CuriosCaterr. Confusing my IRC client with my ssh session is beyond embarassing. :P
04:33.46JTrussellb: 99.92 is fine?
04:34.22JTirqs are no longer an issue, since what release of zaptel/asterisk?
04:34.31russellb1.2.13 or something like that
04:34.49mostyJT, you're an aussie right?
04:34.53JTyes
04:34.58russellbthere was a driver error that caused problems with shared interrupts, but it has been fixed
04:35.10JTwhat about poor zttest scores like 99.92?
04:35.51russellbi have seen plenty lower than that on systems that work just fine
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04:35.59mostyJT, do you run any large asterisk servers? we are seeing lots of mass unreachable sip peers every few hours, since yesterday. wondering if it's to do with the telstra internet backbone problem
04:36.01russellbi'd rather wait for an actual problem
04:36.08JTmy experience is it all goes to shit when it goes below 99.97%
04:36.16JTat least on digital
04:36.25russellbhmm..
04:36.38JTmosty: telstra was stuffed last night, yes
04:36.40russellbwell, you may be right.  I don't work in tech support so I don't look at these things every day
04:37.19CuriosCatJT: So...how can I improve that score?
04:37.27CuriosCatand keep in mind this is an analog line
04:37.49JTCuriosCat: try not loading unnecessary drivers and removing unnecessary hardware, and swapping pci slots
04:38.40mostyJT, we're still having problems today though
04:38.52JTi guess they must be having a variety of issues
04:39.43CuriosCatJT: I'll look into that. This is a Dell PowerEdge server, they have a bit of quirky hardware.
04:39.52CuriosCatincluding more PCI-PCI bridges than you can shake a stick at.
04:40.02JThmm
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04:47.26Trevor_bHas anyone ever seen a good iax hardphone?  Like actually nice, not polycom level mind you, but something?
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04:49.33JTapparently a couple exist, but they're very rare
04:49.36JTas for good, who knows
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04:49.57mostythe few i've seen have been crappy
04:50.02[TK]D-FenderTrevor_b, Nope... and crao
04:50.07[TK]D-Fenderall crap*
04:50.32Trevor_byeah, i have one, its junky, if it only looked as nice as the budgettone then it wouldnt be so bad, but otherwise the web page sucks and manual setup sucks.  Once it works its nice, but it stinks for config.
04:51.11Trevor_bthing looks like a throwback to the 90's or something
04:53.13Trevor_bSo for SIP other than Polycom and Cisco, is anything worth while?  Budgets are junk, snoms i hear are junk although some swear by them, anything else out there worth thinking about?
04:53.49CuriosCatHrm. I called the number, but "console answer" states "noone is calling us"
04:53.56mostyTrevor_b, snom are ok, just don't use beta firmware
04:54.06JTaastra is probably the only other common one worth considering
04:54.10mostyTrevor_b, some people like aastra and linksys 9XX series
04:55.06Trevor_blinksys is a copy of the cisco made cheap (or offered cheap since it wont run MGCP and SCCP)
04:55.23mostythe only major issue with snom phones is the headsets suck. but then so do polycom headsets, from listening in here
04:55.31Trevor_bi like the look, but if its configs like cisco, then i prefer poly, WAY easier to deploy.
04:55.40JTpolycom don't make headsets
04:55.44JTyou buy whatever you want
04:55.47JTlike plantronics
04:55.48Trevor_bplantraonics makes the headset (usually)
04:55.50Trevor_byeah
04:55.54JTsnom is ugly too
04:56.07jqlwhich is a damn shame, since I'd really like a word with Polycom about their phones and my headsets...
04:56.19Trevor_bthe port for the headset could be junk, the headset, the battery.  In general Poly has the best audio quality (speaker at least, i think in general though).
04:57.16mockerWell crap.
04:57.30mockerdundi lookup is replying w/ an extensiuon at two locations..
04:57.35mockerbut it's only at one.
04:59.39[TK]D-FenderMy call center runs on IP 600's w/ Plantronics M11 Amps, and H261 binaural headsets.  They love them.
05:00.33JT[TK]D-Fender: PoE?
05:00.55[TK]D-FenderJT : Naturally
05:01.50[TK]D-FenderJT : I got them before the 430 came out because the price difference between a PoE'd 501 and 600 was so close and the 600 had the better scrre, microbrowser at the time, etc...
05:02.14[TK]D-FenderJT : then the 430 became my category killer until the 320 came along...
05:02.21JTright
05:02.33[TK]D-FenderJT : 320 is really slick as it supports both headset std's at the same time.
05:02.35JTat least you're not using headset + power supplies
05:03.25[TK]D-FenderJT : Best part : M12 Amps don't come with the power brick.  I use my IP 600's w/ PoE and *ITS* power brick is compatible ;)
05:03.49[TK]D-FenderJT : Makes a winner out of the IP 430.
05:04.11[TK]D-FenderJT : because the IP 430 is still a technically superior phone to the IP 320/330
05:04.27JT[TK]D-Fender: you can't use the included 430 power brick and also use a headset on the same phone
05:04.59Trevor_bYeah right now were running the Plantronics DSP-400's because we have a cheap client.  We use the 501's and 601's ourselves.
05:05.11Trevor_bAlthough we were about to setup all the Call Stations to be using NIS+ and allow roaming for the agents.  I really want to get off the soft phones, but the roaming agents are 'really' useful with an extension that follows them.  Although once they get more then 25 agents creating the users on all of them
05:05.15jqlnice
05:05.29[TK]D-FenderJT : I wasn't saying SHARE the brink, I mean power the phone off PoE and use the brick to power the amp (since it DOESN'T come included and WOULD cost extra)
05:05.38*** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp)
05:05.48JT[TK]D-Fender: not tried, but i'd suspect that would suck as well
05:05.50[TK]D-Fenderbrick*
05:05.57[TK]D-FenderJT : Nope.
05:06.07JTtried?
05:06.10[TK]D-FenderJT : yup
05:06.16[TK]D-FenderJT : all works 10%.
05:06.21[TK]D-Fender100% lol
05:06.22JTthe power brick makes a horrible hum on headsets when plugged into phone
05:06.55*** join/#asterisk stridernzl (n=neville@125-237-111-143.jetstream.xtra.co.nz)
05:07.07mockercrap.
05:07.16Trevor_bTK: 320 supports RJ and DIN?
05:07.19mockerdoes dundi keep it's cache after an asterisk shutdown?
05:08.50Trevor_b[TK]D-Fender: You running Vici?  Do your agents ever switch desks?
05:09.21[TK]D-FenderTrevor_b, RJ & 2.5mm
05:09.28Trevor_bnice
05:09.30[TK]D-FenderTrevor_b, Never touched vici
05:09.44Trevor_bwrote your own code for call center?
05:09.50mockerwow, it looks like it does.
05:10.13mockerAnd it will return a cached bad result to other servers and they will cache them as well.
05:10.31mocker...coffee...
05:11.13russellbthe cache is stored in astdb, so yes, it does persist shutdown
05:11.19russellbyou can clear it using "dundi flush"
05:11.21Trevor_b[TK]D-Fender: You have roaming agents or no?  Trying to decide to change over this 'mini' call center to hardphones, but have to decide a few things first.
05:11.51jqlahh, roaming agents. that's a fun one
05:12.01russellband note that the dundi cache has a default expiry of an hour ...
05:12.27mockerrussellb: dundi pwned me. :(
05:12.39mockerI kept thinking that restarting asterisk was flushing my cache.;
05:12.42Trevor_byeah, the NIS+ is an easy one for softphones and roaming agents.  Its just the hardphones and whatnot.  Plus getting this damn diskless client worked out properly.
05:12.53[TK]D-FenderTrevor_b, no roaming and only INBOUND
05:13.02Trevor_bAhhh
05:13.04Trevor_byeah
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05:13.30[TK]D-Fenderok, bed time *yawn*
05:13.35Trevor_bgunna have to price up those 320's again with your suggested headsets. Thanks.
05:13.35[TK]D-Fenderoutta here, back later.
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05:25.30Trevor_bi thought plantronics had built in amps?
05:25.37flendershey, does anyone know if it's normal to get these messages on the PRI every hour:
05:25.40flenders-- B-channel 0/1 successfully restarted on span 1
05:25.44JTno, they have amp boxes, Trevor_b
05:25.49flendersI get that on all 10 channels
05:26.03Trevor_bamp box? how do you mean JT
05:26.10JTan amplifier box
05:26.15JTcontains an amplifier
05:26.18JTinput
05:26.19JToutput
05:26.24Trevor_blike an m11 or m22 from plantronics?
05:26.25JTbatteries/power in
05:26.27JTyes
05:26.29Trevor_bugh
05:26.45Trevor_bmaybe it was the 601 series had a built in AMP?  I am sure i read somewhere about poly's having them...
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05:44.42flewidgooday
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06:07.59lsodihi
06:09.44lsodican any one reccomend asterisk GUI for voip providers (free or commercial) where provider can add users(extensions) and user groups?
06:11.19mostytried trixbox and asterisknow? i haven't used either, but they seem popular
06:11.35JTi don't think they'd be suitable for providers
06:11.40mockerOr vim? </sarcasm>
06:11.41mocker:)
06:11.42JTproviders should just write their own system
06:11.52mostyprovider of what?
06:12.14JTvoip
06:12.37mockermosty: So, after my adventure w/ dundi tonight I learned a valuable lesson.  Restarting asterisk does not clear your dundi lookup cache.
06:12.40mocker;)
06:13.29mostyJT, obviously, 99% of installed trixbox/asterisknow boxes probably do that
06:13.34lsodivopi provider, yes I have tryed trixbox and asterisk now, but user creation and management isnt that great and you cant make user groups,
06:13.49mostylsodi, you probably need to write your own
06:14.10jqlyeah, I had to write my own
06:16.54JTmosty: voip provider out of the box? wow, trix/astnow does that?
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06:22.33mostyJT, all that means is you accept voip calls? i assume that's easy to do with both those setups
06:22.39Aces1Upwhats the standard way of writing code to interface with asterisk? is there like a api for php or something?
06:22.42kenaedahello, im interested in flash based audio which connects to asterisk server. i found an example with openfire and red5gateway, but does anyone know additional possibilities?
06:22.59mostyAces1Up, AGI and AMI
06:23.21Aces1Upmostly thanks.
06:23.57JTmosty: i assume he means more like a service provider, with user accounts and what not
06:24.27mostyit's unclear without more info
06:25.28tzafrir_laptopAces1Up, but a common mistake is to do in AGI something that can already be implemented in plain dialplan
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06:39.07TimTFHello all
06:39.23jmlsmorning
06:39.26TimTFis there someone here familiour with TrixBox and multiple peer registrations?
06:40.25TimTFI'm trying to register two sipaccounts on the same provider. But asterisk seems to be unable to keep the two accounts apart, generating authentication errors
06:49.00Aces1Upis there asr and ACD measurements for DID's?
06:49.44n0n4m3umm.. in case i have a client using iax defined as a friend in iax.conf, let's say [1234] ... and have the "exten => 1234,1,Dial(IAX/1234)" in extensions.conf that should work, right?
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06:55.54mostyn0n4m3, you also need calls to get to that context, but yes
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07:03.31Aces1Upanyone know the rss feed for asterisk-biz and asterisk-users mailing list?
07:08.03cy303Aces1Up: is there one?  that'd be really nice, I'd unsub from the list heh
07:09.58Aces1Updunno saw a link on this page for it.. on voip-info ...   http://www.voip-info.org/wiki-Asterisk+Mailing+Lists
07:10.27Aces1Upcy303, weird, i'm trying to subscribe to the lists, but am not retting a confirmation e-mail.
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07:15.31manopulushello all, I have a question. can I use two feature-maps in features.conf, let say, someone can do blind transfer with # and another customer with ##. ?
07:15.54manopulusif so, how I can select proper map in dialplan?
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07:21.06TimTFthis is driving me insane. If I configure one inbound route, it works. If I add another trunk the first stops working and the second one works
07:21.48TimTFif I remove the second one, the first one works again
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07:25.58cy303Aces1Up: hell yeah man
07:26.03cy303thanks for pointing that out heh
07:27.31FlatFootmorning all
07:27.42angryusermorning
07:27.54FlatFootgot any preferred motherboards to use with a TE210P ?
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07:43.36mostyFlatFoot, have you had a problem with a specific combination?
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07:46.38Project2501ahi guys. any success stories compiling asterisk on AIX 5?
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07:48.05Project2501agot an old H70 at work siting on it's behind
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07:59.00mostyi imagine zaptel would be your biggest issue
08:00.08n0n4m3mosty what do you mean by calls to get to that context...
08:00.17cy303hmm
08:00.37mostyn0n4m3, do you know what a dialplan context is?
08:00.45cy303anyone recommend a good online voip-stuff vendor?  phones, cards, etc..
08:01.36*** join/#asterisk n0rus (n=n0rus@210.85-200-239.bkkb.no)
08:01.57n0n4m3umm
08:02.06*** join/#asterisk zdrulio (n=krlozano@82.119.72.130)
08:02.07n0n4m3the [something-incoming] part...
08:02.11n0n4m3the context of it?
08:02.16zdruliohello all
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08:03.29mostyn0n4m3, you should probably read the section of the book about the dialplan
08:03.35mosty~book
08:03.55jboti heard book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
08:03.56n0rusAnyone using Munin to graph Asterisk data here?
08:03.56n0rusI need help with this plugin: http://pastebin.ca/544024
08:03.56mosty~afot
08:04.04n0n4m3mosty i belive i have an okay dialplan
08:04.34mostyn0n4m3, then try it out
08:05.12FlatFootmosty: sorry had to get my cup 'o' tea , no specific probs just wondered if there was a preffered mboard as i am building a new "Carlos Fandago" system
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08:08.18mostyFlatFoot, i've only ever had problems with ztdummy on some motherboards- ie when you don't have a digium card
08:10.38angryuseri have read about queues , but in dialplan how to redirect a person int it? what action/command? exten => 2020,1,which command to join queue support?
08:10.48mostyangryuser, Queue
08:10.58angryuserok
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08:12.20angryuserhow do you handle login of agents for queue? i hae a simple zap/fxs phones
08:12.22n0n4m3mosty http://rula.net/19
08:14.16n0n4m3calling from iax to sip works okay (4126 > 4125), but vice versa doesn't... (sip/4125 > iax/4126)
08:15.01n0n4m3here's the error
08:15.02n0n4m3http://rula.net/20
08:15.29TimTFI finally fixed my issue, (multiple users on same provider); asterisk matched the wrong account failing authentication failing to setup the inbound calls. The sollution is adding "insecure=very" to the peers definition
08:17.07angryuserok i have found the answer
08:17.46mostyn0n4m3, you need to use IAX2 in the dial command, not IAX
08:17.48zdrulioi have 2 asterisk connected by IAX.  if user from A1 want to dail user from A2 dial 9xxx and realy dial xxx. IT`s OK, but all calls from A1 to A2 pass from user yyy and in A2 softphone come call form yyy. can i make to view real number how dial. i mean A1 user(xxx) dial A2 user(xxx2) and in softphone of A2user view xxx number ?
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08:18.32n0n4m3mosty argh... i hate this kind of typos :D
08:18.41n0n4m3thanks a million
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08:21.48mostyzdrulio, add or remove the leading digit used in the Dial command as required
08:22.51zdruliomosty: i don`t understand u. exlain plz
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08:23.47mostyzdrulio, you specify the extension you want to call in Dial command, so just modify the dial command so it dials the extension you want
08:24.20n0n4m3zdrulio something like... exten => _91.,3,Dial(IAX2/${FWDCIDNUM}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:2},60,r)
08:24.43n0n4m3the ${EXTEN:2} means that it 'cuts' the first two digits when dialing
08:25.31n0n4m3so if you dial 91612 in that context it means asterisk dials 612 to iax2.fwdnet.net
08:25.47zdrulioyes
08:26.17zdruliobut A1 call to A2 pass "YYY" user.
08:26.45zdruliocall is OK have a connection but in display of A2 view calling YYY
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08:29.07n0n4m3umm
08:29.20n0n4m3exten => ${FWDCIDNUM},1,Set(CALLERID(num)=91${CALLERID(num)})
08:29.24n0n4m3something like that
08:29.34n0n4m3i mean.. you have to play with this command
08:31.23zdrulioin the moment i have this  exten => _14.,1,Dial(IAX2/EXTnumber:password@IP:PORT/${EXTEN:2},30,r)
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08:34.53n0n4m3zdrulio look at this one...
08:34.54n0n4m3http://rula.net/21
08:38.13zdruliothx i will test it
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08:40.50kavitwhich phones are the most asterisk compliant??
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08:44.21tengulrehi,all
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08:49.54n0n4m3kavit umm what phones do you mean... softphones or hardware phones?
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08:50.23n0n4m3kavit check out this page http://www.voip-info.org/wiki-VOIP+Phones
08:51.50kavitn0n4m3: i meant hardphones... I have used asterisk for a long time with various hardphones
08:51.59kaviti just wanted the general communities opinion
08:52.07kavitwith stuff like BLF and SLA
08:52.40kavitn0n4m3: I am looking to change stock phones for deployments... from snoms to something else
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08:54.07mostykavit, step up to polycom
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08:55.22kavitmosty: last time i used a polycom i had to modify a lot of .cfg files and mwi had issues
08:56.21kavitmosty: also NAT on polycom phones made me cry :(
08:56.57kavitbut this was a year ago
09:00.18JTnat worked fine on polycom a year ago, it's just a matter of knowing what to configure...
09:02.34kavitJT: i set the registration time out to a lower value
09:02.38kavitor something like that
09:03.44JTi didn't need to fiddle with that at all personally
09:04.37n0n4m3anyone of you guys uses belco bcip-300?
09:04.47kavitJT: would you recommend polycom? especially for stuff like metermaid and sla?
09:05.06JTi would recommend polycom, i have no idea what metermaid does
09:05.31kaviti  dont understand why people cant let go of stupid key system features
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09:05.47zapp-braniganhi, there is any way to detect answer supervision in a sipura ?
09:05.59zapp-branigan<PROTECTED>
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09:07.11mostywhat do you mean by answer supervision?
09:07.18kavitJT: metermaid = lets you hint park:701@parkinglotcontext
09:07.28JThrm ok
09:07.40kavitblinking lamps for a parking lot
09:07.50JTright
09:10.18zapp-branigansipura never dettect when a call is started, from the first second the call has been answered
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09:10.47zapp-braniganincluding the time of ring
09:11.40JTzapp-branigan: sounds normal
09:11.58zapp-branigani'm looking for a double-talk detector for asterisk
09:12.11zapp-braniganthere is some?
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09:13.01JTdouble talk, isn't that something out of the novel 1984?
09:13.10kippihey
09:13.22kippiis there away to recored internal phone calls?
09:13.56zapp-braniganusing a sipura who can i use to detect answer calls
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09:15.04mostykippi, lookup one touch recording
09:15.29mostykippi, and the Monitor command
09:17.40kippiok, using the Monitor command at the mo to recorde incoming calls,
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09:52.38pabs3anyone heard of a problem with asterisk 1.2.18 and cisco 7960 phones not being able to register (401 unauthorised)? They worked on our 1.2.7 testing install
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10:19.00canberkhello
10:19.12canberkhow can I make asterisk read the caller id from my fxo gateway
10:19.18canberki see that my fxo gateway sends ******** Line : 2, Start Inviting ********
10:19.18canberkstrDes To:sip:1001@192.168.2.196, strOri From:"05333585867"<sip:3003@192.168.2.196>
10:19.32canberkthere is the From tag there, which has the caller id
10:19.39canberkbut asterisk is not reading that i think
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10:22.36eeoshi everybody
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10:31.41CBU[^_^]M``weeeeeee
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10:34.38doolittleworksomeones happy
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10:39.10doolittleworkhi zeedo
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10:44.58zeeeshhi
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11:01.40IntertweakerDoes somebody know how to use the IM ans presence functions in asterisk, combined with the GUI "ScopServ"?
11:03.04cy303Does anyone here have any experience with got-name.com ?
11:03.54*** join/#asterisk Intertweaker (n=I@a213-84-148-202.adsl.xs4all.nl)
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11:14.34zdruliohm in fedora how can i load zaptel, ztdummy before asterisk ?
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11:21.37eeosI have some problem with the documentation: I am trying to find a description of all parameters allowed within exten => but cannot find it
11:21.43eeoscan anyoe point me out to it?
11:21.50Kumbanghello guys, i just curious what's the meaning of part number in digium product, say TE120 1 for what 2 for what and 0 what's that
11:22.16angryuser<eeos> core show applications in CLI
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11:22.32angryuser<eeos> "core show applications" in CLI
11:23.11angryuser<eeos> "core show functions" in CLI
11:23.16angryuseralso
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11:24.31eeosangryuser: thanks. but what do you mean?
11:24.33eeos:)
11:25.08angryuser<eeos> start asterisk, type "asterisk -vvvvvvvvvvvr"
11:25.17angryusertype 2 commands above
11:25.39eeoswait
11:26.03eeosI have to start the server
11:26.15eeosI was hoping to find something on thebook
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11:26.20eeos:(
11:26.55angryuser<eeos> if you want more data on any function/application type "core show function NameOfFunction" or "core show application NameOfApplication"
11:27.08angryuser<eeos> you got everything on wiki
11:27.15angryuser~wiki
11:27.48angryuser~book
11:28.00jboti guess book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
11:29.31eeosangryuser: that is the book I mean when I said the book
11:30.04angryuser<eeos> you want to but one?
11:30.07angryuser*buy
11:30.27eeosangryuser: i have it ad read it
11:30.49angryuser<eeos> and?
11:30.56eeosangryuser: but cannot find a reference to exten with all paremeters, as you would for a programming language
11:31.14angryuser<eeos> do what i described above
11:31.17eeoslike a nusthell? you canot go back to the main chapter every time
11:31.43mostyeeos: look it up on voip-info.org
11:33.18eeosangryuser: I did core show function exten
11:33.30eeosangryuser: but it says no fucntion with that name
11:33.34mostyexten isn't a function
11:33.38angryuser<eeos> no core show functions/applications
11:34.29*** part/#asterisk zdrulio (n=krlozano@82.119.72.130)
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11:34.57angryuserhave to go, moste please reexplait how to get function/application info in cli/web to <eeos> ;) thx
11:35.01angryuser*mosty
11:35.12eeosis not under functions and is not under applications
11:35.21mostyexten is neither a function nor an application
11:35.24eeosangryuser: I got it, but exten is not there
11:35.43eeosmosty: yes, so where do i find it?
11:36.02mostyeeos, in any document that describes the asterisk dialplan setup
11:36.22eeosmosty: yes, I did read them
11:36.41mostythen what's the problem?
11:37.06eeosmosty: my question is: I need a nutshell for exten parameters, so I do not have to go back to thebook every time. Is there anyting similar?
11:37.25eeosmosty: or do you remeber al parameters by heart? because I am not able to do it.
11:37.50eeosmosty: and cannot reread chapter 4 and 5 every time I eed a parameter.
11:38.01mostyexten has only 3 parameters
11:38.12mostyextension, priority and command
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11:39.47eeosyes, so we would need a nutshell of: 1) all possible estension parameters 2) priorities is relatively easy 3) all possible commands, with syntax and possible values
11:40.14eeosI do not think it is so strage to ask, it does exist for most of programmig languages.
11:41.08*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
11:41.10mostyeeos, extensions are any number or string...
11:41.40mostynot sure about punctuation- stick to alphanumeric and - and _
11:42.22mostyeeos, and google for "documentation of dialplan commands site:voip-info.org"
11:44.23eeosmosty: I am actually on voip-info, but the information I am looking for is not there
11:44.31eeosmosty: anyway, it does not matter
11:44.50*** join/#asterisk Strom_M (n=strom@208.47.199.4)
11:45.45mostyeeos, yes it is, the page is called something like "asterisk documentation of dialplan commands"
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11:47.47drrthi. any 1.4.4 users is here?
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11:51.04eeosmosty: thanks! I got it finally. the search did not return anything before. :(
11:51.08eeosmosty: thanks a lot.
11:51.10Zeeekzaptel 1.4.2.1 make problem on centos 5
11:51.12Zeeekmake[3]: Entering directory `/usr/src/asterisk-1.4/zaptel-1.4.2.1/menuselect/mxml'  autoconf: no input file
11:51.13*** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.144)
11:51.24Zeeekasterisk make ok
11:51.53*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
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11:56.41Zeeekok, a yet more idiotic question: zaptel will make without hardware installed, will it not? I'm pretty sure it will but I haven't plugged anything in yet
11:57.19*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
11:57.42Uatecone way to find out
11:58.00*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
11:58.28ZeeekI'm pretty sure it doesn't matter
11:58.36Zeeekbut since make isn't working
11:58.41*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
11:59.05Zeeekah. autoconf: no input file
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12:00.50JTseanwg: can you stop that already?
12:01.00Uatecthat's not a hardware problem though
12:01.27*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:01.32Zeeekwhat is the latest recommended zaptel for use with 1.4?
12:01.41ZeeekI expected it to be 1.4-current
12:01.57Uatecsomebody ban seanwg briefly so he stops trying to rejoin
12:02.08*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:02.41zagyhi
12:02.57zagywe have an asterisk running with a 4 channel isdn card
12:03.02drrtJT, hey jt. which branch do you use ?
12:03.13zagyis it possible to use this card to connect to isdn services from another linux box?
12:03.32*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:04.13*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:04.50Uateczagy, what services? telephone, internet or other?
12:04.55*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:05.17zagyUatec: well, actually it'll be PPP.
12:05.39*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:06.26*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:06.47Uatecwell if you just want to connect to the internet there doesn't seem any point to do it via your asterisk box, just connect your asterisk box to the interweb and route it
12:06.54Uatecor am i mis understanding you?
12:07.10*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:07.11zagyUatec: no no, it's not internet
12:07.13zagyinternet we have :)
12:07.31zagyi need to dial in to a client's network via PPP/isdn
12:07.38zagyso far we've just used an old ISDN modem to connect to the isdn card which then calls outside... but the modem just died :)
12:07.56*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:08.23Uatecoh, well a tcp network anyway
12:08.24JTdrrt: 1.2
12:08.26*** join/#asterisk Pilko (n=pirch@213.80.169.119)
12:08.35Uatecwhat isdn card do you use?
12:08.45*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:09.24*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:10.05zagyUatec: sec
12:10.12*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:10.47*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-118-241.ph.ph.cox.net)
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12:11.58zagyUatec: junghanns quadbri
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12:13.34*** part/#asterisk Pilko (n=pirch@213.80.169.119)
12:13.34zagyUatec: but that shouldn't really matter i guess :)  of course i might be able to start the ppp on the asterisk box, but that's not really what I want. but anyway, i might just as well go for a cheap isdn card and put it in some workstation
12:13.51*** join/#asterisk Corydon76-home (i=silver@pdpc/supporter/sustaining/Corydon76-home)
12:13.51*** mode/#asterisk [+o Corydon76-home] by ChanServ
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12:14.10waKKumorning folks ;)
12:14.41Uatecsounds like the best plan
12:14.43*** join/#asterisk Pilko (n=pirch@213.80.169.119)
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12:14.57zagyUatec: ok, thanks :)
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12:24.17*** mode/#asterisk [+b seanwg!*@*] by Corydon76-home
12:24.55Uatec*sigh*
12:27.17*** join/#asterisk The_LightSide (n=dialt@wbs-41-208-238-122.wbs.co.za)
12:27.54*** join/#asterisk psk (n=psk@golia.caltanet.it)
12:28.09The_LightSidehi all, has anyone gotton called user id working wiht asterisk 1.2.17 and the snom 320/360?
12:28.48Uatecs/gotton/gotten
12:28.50Uatecs/gotten/got
12:29.03[TK]D-FenderCorydon76-home: tHANKS
12:29.17*** mode/#asterisk [-b seanwg!*@*] by Corydon76-home
12:29.30mostyThe_LightSide, you mean dialed number id?
12:29.47Corydon76-homeThat should have been long enough to stop the bouncing client
12:29.53*** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm)
12:29.56Corydon76-homeNope
12:30.08The_LightSidemosty yes
12:30.17*** mode/#asterisk [+b seanwg!*@*] by Corydon76-home
12:30.29[TK]D-FenderCorydon76-home: You should wait for the 2nd join.....
12:30.32mostyThe_LightSide, i mangle it into the callerid
12:30.46Corydon76-homeEh?
12:31.06The_LightSidemosty, how?
12:31.19[TK]D-FenderCorydon76-home: Sure he came back, but you banned him again before he even bombed out and rejoined.
12:31.49Corydon76-home[TK]D-Fender: he already bombed out
12:32.08[TK]D-FenderCorydon76-I'm talking about an IN/OUT/IN since unbanning.
12:32.53mostyThe_LightSide, well voip-info.org seems dead at the moment, but see http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid
12:34.11*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
12:34.35The_LightSideok, thanks mosty
12:35.23The_LightSidebtw, are there any ports other than 6667 that i can use to connect to freenode?
12:35.35Uatectry any of the surrounding ports
12:35.36Uatec6666
12:35.38Uatec6668 6669
12:36.47The_LightSideim behind a seriously port restricted proxy.. nothing in that area seems to be working
12:37.16The_LightSideim vnc'ed into my home server
12:37.58[TK]D-FenderThe_LightSide: got web access?
12:38.12The_LightSidei do, yes
12:38.14Uateclol
12:38.25Uateci'm vnc'ed in to my home PC from my mobile for google talk
12:39.20The_LightSidelol
12:39.30[TK]D-FenderThe_LightSide: There are a few web-irc clients out there, not sure if FreeNode has one on their page...
12:39.51*** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com)
12:39.59The_LightSideill have a look, thanks
12:40.12VJFROMGTanyone know to something little to run asterisk on, eg a router?
12:40.31[TK]D-FenderVJFROMGT: Linksys WRT's
12:40.39[TK]D-FenderVJFROMGT: Go look it up on the WIKI
12:41.20VJFROMGT54g?
12:41.24angryuserwhich function  i have to use to check if Agent1 is logged in?
12:43.04UatecThe_LightSide, google for cgiirc
12:43.08Uatecthat's a good web based client
12:43.13Uatecthere are many instances of it
12:43.22Uatecyou'll just have to try them to find one that can connnect to freenode
12:44.04*** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) [NETSPLIT VICTIM]
12:44.12jacqdo you guys know if asterisk manager triggers an event if it cent register SIP?
12:47.00angryuserthere is no any? for example i have a queue 1, i have 2 agents, one is called, but he went to toilet;) i need to do 2 checks 1: if agent2 s present 2: if he is, that forward call to him
12:47.14angryuserif he is not, forward somewhere els
12:47.20angryuser*else
12:47.53angryusersecond check if agent 2 is present and busy, forward to another queue
12:48.14[TK]D-Fenderjacq: No.
12:48.26*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
12:48.28drrtangryuser, you can configure the schema using asterisk only
12:49.18[TK]D-Fenderangryuser: Sounds like Dial #1, Dial #2, Do something else.  This isn't a queue.
12:49.24drrtangryuser, do you know about queue penaltys?
12:49.35angryuser<drrt> i need queues for position status, and i need to give theagents the possibility to login/logout
12:49.38[TK]D-Fenderangryuser: It's 2 Dials + "Something else"
12:49.46*** join/#asterisk bcnx (n=root@d54C0E110.access.telenet.be)
12:50.20angryuser<drrt> no dont know much about penaltys
12:50.20[TK]D-Fenderangryuser: You can do even a rudimentary login/out all in dialplan.  Queues don't just give 1 chance.
12:50.33bcnxHi all. About to do an Asterisk install and the hardware provided forces me to use Centos, but not sure if I should use 64-bit distro or not. Any ideas?
12:50.38[TK]D-Fenderangryuser: So forget about them.  You're going to have to do it yourself
12:51.04[TK]D-Fenderbcnx: No, just use 32-bit.  And how does HARDWARE force your distro choice?
12:51.06angryuser<[TK]D-Fender> why, queues are bad?
12:51.27[TK]D-Fenderangryuser: Queues were not designed to work they way you want them to.
12:51.30bcnxD-Fender: simple: drivers for things like RAID controllers
12:51.40drrt[TK]D-Fender, disagree
12:51.51bcnxonly available for SLES 9 and RHEL 4
12:51.53[TK]D-Fenderbcnx: typically they are all jsut kernel based, but whatever,....
12:52.10angryuser<[TK]D-Fender> i know that i can do it using dialpan, but the word queue sounds good;)
12:52.11drrtangryuser, which asterisk branch do u use ?
12:52.27bcnxD-Fender: I don't like using Centos, because I hear ISDN support is a bit off
12:52.29angryuser<drrt> i am using the latest stable releases
12:52.42angryuser<drrt> 1.4.4 ect
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12:52.51*** part/#asterisk manopulus (n=manopulu@213.197.181.154)
12:52.55[TK]D-Fenderbcnx: No, it is quite supported I'm sure.
12:52.56bcnxbut if the driver situation forces me ...
12:53.09[TK]D-Fenderbcnx: Either way, CentOS is a fine choixce
12:53.13drrtangryuser, then move to queues.conf and point your second agent with penalty 1, 1st agent should has penalty 1
12:53.30bcnxD-Fender: OK, thank you for your advise. Centos 4.5 32-bit it is ...
12:54.12*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
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12:55.01drrtangryuser, like this: member => agent/bla-bla2,1 member  => agent/bla-bla1,2
12:55.25angryuser<drrt>  i got a simple ivr, "to joinf agent1 press 1, agent 2 press 2" after the person need to be directe to a queue, and if no answer from agent 2 or 2 (1 min) i need to do a check if agent 2 or1 os here, if not go somewhere else
12:55.40puzzledhi
12:55.59angryuser<drrt> difficult to explain :)
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12:58.02angryuserjust tell me if i cant do acheck if Agent1 is logged in? i can do this with db * functions but just to be sure that there is no anf function programmed for this
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12:58.34angryuser<drrt> and what is panalty for?
12:58.49drrtangryuser, you can put queue into different context and put Goto function to anywhere you need after Queue()
12:58.51angryuser*penalty (damn keyboard
12:59.17*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
12:59.42drrtangryuser, when the call gets into a queue less penalty agent will answer in 1st order
13:00.41drrtangryuser, when less pen agent's line is busy or not connected higher pen agent will take a call
13:01.45*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
13:02.06angryuser<drrt> ok, dont really need that, we have only 2 peoples in support, and i got a small ivr to choose their agent, i just need to integrate the wait time for a secon person which calls the same agent
13:03.05angryuser<drrt> if time out, cal the first agent, but brfore check if he is logged in, if not goto(elewhere)
13:03.35angryusermaybe i should do it by dialplan really
13:03.43[TK]D-Fenderangryuser: Again this is not a job for queues.  Each agent gets at most 1 shot.....
13:04.14drrtangryuser, and  back to front right ?
13:04.26angryuseryea
13:05.13angryuserok sticking to dialplan
13:07.41*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
13:08.07Kattymew.
13:08.16blitzragemew urns.
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13:10.22Kattyman, the chick at the best western in chicago is bitchy this morning
13:10.55*** join/#asterisk msetim (n=msetim@200.195.161.164)
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13:11.28d-nrw-marchi folks
13:12.18*** join/#asterisk badcfe (i=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr)
13:12.28d-nrw-marcanyone there?
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13:12.32*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
13:13.03badcfei see "Call on SIP/xxxxx-08618008 left from hold" in CLI, what does it mean?
13:14.09[TK]D-FenderKatty: Mew.
13:14.09waKKuA number
13:15.06Katty[TK]D-Fender: who do i call to get my tax stuff updated after i move? is that the court house or the assessor's office?
13:15.24badcfei see it a lot and i dont intend to do any "hold" function, so i guess it has nothing to do with "hold" in practice -- thats wht i wander why i keep seeing it in the CLI for almost each of the calls.
13:15.25[TK]D-FenderKatty: .......
13:15.29[TK]D-Fender<- Canadian
13:15.31JT[TK]D-Fender's tax advice
13:15.44Katty[TK]D-Fender: oh, right, i forgot lol
13:15.54[TK]D-Fender;)
13:15.56Kattydear americans in channel, who do i call to update my tax records after i move :P
13:16.11Kattyi should know this by now.
13:16.21JTi don't know, but could you teach me australian taxation law, Katty ?
13:17.07KattyJT: sure!
13:17.13KattyJT: it's a sliding scale rate.
13:17.13JTsweet
13:17.25Dr-Linuxis there any data variable in asterisk? date without time
13:17.40KattyJT: when icome is earned in the form of capital gains, on your assets, only half the gain is essessable for the CGT....provided it's ben held for at least 12 months
13:18.07JTand gst?
13:18.11KattyJT: if it's owned for less than 12 months, then all of it is accessible for CGT purposes.... of course, your residence is exempt from it.
13:18.12[TK]D-FenderDr-Linux: "show functions"
13:18.38KattyJT: GST is a multi-stage tax of 10% on most stuffs
13:18.52JTmulti stage?
13:18.55KattyJT: yes
13:19.02KattyJT: as in corporate is different
13:19.07KattyJT: being 30%
13:19.13KattyJT: however!
13:19.28KattyJT: some things are GST free...like basic food, medical, edjimication, and exports
13:19.51KattyJT: and the revenue from the tax is distributed to the states.
13:19.59JTright, i'm more interested in gst input tax credits with regards to businesses
13:20.07JTKatty: are you from australia?
13:20.15KattyJT: hence why state government doesn't have any sales taxe...even tho they do have a stamp thing
13:20.23KattyJT: no, i just want to move there
13:20.32JTclearly :o
13:20.36Katty^_^
13:20.39JThave you ever worked there?
13:20.48Kattylol, nope. never stepped foot in aussieland
13:20.50Dr-Linux[TK]D-Fender: can't see any for date only
13:20.51Kattyi just want out of america.
13:21.03cpmWhere to?
13:21.04JTonly americans say aussieland <_<
13:21.17cpmyou mean Oz?
13:21.19Kattycpm: canada or australia
13:21.22JTAustralia
13:21.33[TK]D-FenderDr-Linux: There is one where you can specify the return format....
13:21.33Kattyaustralia! home to everything deadly.
13:21.38cpmKatty, umm, so? why are you still here?
13:21.42JTexcepd WMDs
13:21.46JTexcept
13:21.56Dr-Linux[TK]D-Fender: can you tell me the name for that please?
13:22.06JTKatty: i've been researching gst input tax credits recently
13:22.10[TK]D-FenderDr-Linux: "show functions"
13:22.28cpmKatty, I've spent a bit of time in Canada, and were I younger, I'd be gone
13:22.31Dr-Linux[TK]D-Fender: yeah there are alot, so coudn't find
13:22.58*** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net)
13:23.13[TK]D-FenderDr-Linux: No.  Go look.
13:23.13*** join/#asterisk Dovid (n=Dovid@bzq-88-155-87-253.red.bezeqint.net)
13:23.27Dovidis there any way to set a STUN server in sip.conf /
13:23.30Dovid?*
13:23.30JTno
13:23.38KattyJT: i've been researching investiment things.
13:23.40[TK]D-FenderDovid: No.
13:23.46KattyJT: i've determined that ingdirect.com is awesomeness.
13:24.00JTit is, i have an account with them
13:24.17d-nrw-marcpbx_spool.c:269 attempt_thread: Call failed to go through, reason 3
13:24.18DovidTK: I have a server behind NAT with externhost set and for some reason the provider is getting the private IP of the box and not the public I{
13:24.19Kattykind of a pain having 3 bank acconts tho :<
13:24.27d-nrw-marchas anyone an idea?
13:24.34Kattyi have lots of ideas.
13:24.40Dovidd-nrw-marc: paste the CLI on pastebin
13:24.41Kattybut theyre not pertaining to your problem.
13:24.43*** join/#asterisk key2 (n=Ritual@193.33.36.22)
13:24.44key2hey
13:24.49Kattyhello.
13:24.55Dovidhello
13:25.02Dovidanyone else wana say hello ?
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13:25.10key2someone can tell me if when connected to the telco with mISDN, it's a NT or a TE ?
13:25.16JTTE
13:25.26JTTerminal Equipment
13:25.28key2JT: thx
13:25.47JTof course i don't recommend misdn, but that's another story
13:25.49key2JT: and whats the diff btw ptp and ptpm
13:26.00JTthe S-bus network topology
13:26.11JTpoint to point only has a device at each end
13:26.19JTpoint to multipoint can have multiple devices
13:26.26JTit's a setting set at the telco
13:26.50d-nrw-marcpastebins database is damaged ;-)
13:27.02key2JT: mmhh ok, but when it's RJ45, how is it called ?
13:27.09key2like european
13:27.11Qwelld-nrw-marc: uuoc.com
13:27.17JTkey2: ?
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13:27.38blitzragehello
13:27.43DovidQwell: whats uuoc ?
13:28.02Dovidnm
13:28.02QwellGo there and find out
13:28.09Dovidi went. took me a sec
13:28.15d-nrw-marchttp://uuoc.com/1785
13:28.18*** part/#asterisk zagy (n=zagy@195.62.106.83)
13:28.19blitzragebreakfast time!!!
13:28.38Dr-LinuxSTRFTIME              STRFTIME([<epoch>][,[timezone][,for  Returns the current date/time in a specified format.
13:28.44JTuuoc traditionally stands for Useless Uses Of Cat
13:28.48*** join/#asterisk eeos (n=eeos@86.53.50.16)
13:28.55DovidCat being ?
13:29.04Dovidseems to be a kinda pb site
13:29.04JTa *nix utility
13:29.17Dovidlike cat /foo/bar ?
13:29.20Uateccat being http://www.kaplink.net/wrong/meow.jpg
13:29.21JTyes
13:29.37Dovidah ok. know what that is
13:29.46JTlike cat blah|program
13:29.50JTis a useless use of cat
13:29.53JTinstead of doing
13:29.57JTprogram < blah
13:30.03eeoshi everybody
13:30.08d-nrw-marcDovid: nice
13:30.11DovidUatec: remake of (damn forgot the movie name)
13:30.20Qwellcat file | grep text
13:30.27Qwellthat's a more traditional uuoc
13:30.57d-nrw-marchttp://uuoc.com/1785
13:31.00Dovidok. so cat - gets data from file
13:31.11JT...
13:32.08d-nrw-marcthats right, cat prints all contents to STDOUT
13:32.14Dovidd-nrw-marc: its 1.4 (which i didnt have time to play with yet) but what do you have for dial ?
13:32.21DovidDial(SIP/ ?
13:32.32d-nrw-marcjep
13:32.55d-nrw-marcvia outgoing queue
13:33.11Dovidah. never used that. TK or Qwell can maybe help u
13:33.18Dovidso ur doing Dial(SIP/jep)
13:33.19Dovid?
13:33.51d-nrw-marci'm pasting my sip.conf & extensions.conf
13:33.55d-nrw-marcmom
13:35.14VJFROMGTanyone else got problems accessing voip-info.org?
13:35.40johann8384yeah
13:36.13*** join/#asterisk boch (n=fran@190.48.209.159)
13:36.31*** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk)
13:36.36VJFROMGTdoes anyone know a good forum for mini asterisk devices such as wrt?
13:37.32*** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com)
13:38.05[TK]D-FenderVJFROMGT: Wait for the WIKI and check its list
13:38.45d-nrw-marchttp://uuoc.com/1786 here is my config
13:39.27angryuserhow to check in realtime if person is busy? and when is is not anymore, call him
13:39.49*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
13:39.49*** mode/#asterisk [+o anthm] by ChanServ
13:40.44d-nrw-marcDovid: gateway asterisk # cp sample.call outgoing/
13:41.04d-nrw-marcsry, wrong window :)
13:41.19d-nrw-marcDovid: http://uuoc.com/1786
13:41.53anthmhey =D did you get all your issues straightened out?
13:42.02d-nrw-marccan anyone help me?
13:42.35DovidDial(SIP/${EXTEN}@49180512578xxx
13:42.40Dovidwhat is the sting of numbers after that ?
13:42.58Kattyanthm: pfft. i /am/ an issue.
13:43.04Dovidnormally what follows the @ is the providers IP or the name that u have for it in sip.conf
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13:43.13Kattyanthm: also, no. i've still got 50 billion people to call and tell i've moved :<
13:43.25*** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
13:43.37JTget asterisk to call them for you ;)
13:43.41d-nrw-marcthe numbers are in sip.conf defined
13:44.18anthmdid you move far ?
13:44.39*** join/#asterisk sci_05 (n=peter@waterfall.bestserversllc.net)
13:45.16Kattyanthm: eh. 5 miles.
13:46.08d-nrw-marcDovid: any idea?
13:46.44Dovidshow me ur sip.conf
13:46.50Dovidtake out the passwords
13:47.26d-nrw-marchttp://uuoc.com/1786 on the top of the post ;-)
13:48.13Dovidthe xxx is to not show us ur number or u r actually using xxx ?
13:48.36d-nrw-marcthe xxx hides the complete number
13:48.41Dovidokies
13:50.33*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:51.52Dovidfor ur first line in ur .call file try
13:52.03Dovidhannel: SIP/02103612xx@49180512578xxx
13:52.16Dovidinstead of Channel: SIP/49180512578xxx/02103612xx
13:53.14d-nrw-marcmy file is correct, must be lost on copy'n'paste
13:53.20*** join/#asterisk doolittlework (i=doolittl@196.211.34.2)
13:53.26Dovidcorrect as in ?
13:54.16d-nrw-marcin my sample.call is the right row
13:54.38*** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal)
13:55.01d-nrw-marcthe asterisk is calling out - my phone rings one time
13:55.18d-nrw-marcthen i get: NOTICE[24735]: pbx_spool.c:269 attempt_thread: Call failed to go through, reason 3
13:55.32[TK]D-Fenderd-nrw-marc: Turn on sip debug and retry
13:55.35d-nrw-marcmy firewall is disabled
13:55.41d-nrw-marcokay
13:55.58[TK]D-Fenderd-nrw-marc: And your * box has multiple IP's?
13:56.13[TK]D-Fenderd-nrw-marc: At least one of which is public?
13:58.31d-nrw-marcyes, it's my linux-dsl-router
13:58.45d-nrw-marchttp://uuoc.com/1787  this is the debug msg
14:00.51*** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru)
14:00.58*** join/#asterisk eeos_ (n=eeos@86.53.50.16)
14:02.37[TK]D-Fenderd-nrw-marc: Looks like ti doesn't like the "RINGING" for some reason.  Don't know enough about SIP to figure WHY, but that might help someone else figure it out.
14:03.13*** join/#asterisk gammah (n=gammah@cpe-70-113-113-209.austin.res.rr.com)
14:05.41d-nrw-marcdamn
14:06.07*** join/#asterisk waptaxi (n=cahe@stat-5-160.e-sky.ru)
14:07.55*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)
14:08.48*** join/#asterisk yannj_fr (n=yannj@APuteaux-152-1-75-147.w86-205.abo.wanadoo.fr)
14:08.51yannj_fr-dev
14:09.13angryuser<PROTECTED>
14:09.13Mercestesyannj_fr, sorry, just us users here.
14:09.46[TK]D-Fenderangryuser: "show application agentcallbacklogin"
14:09.52angryuseri want exten => 11111,1,WhateverToMake,theAgentLogout?
14:10.37angryuser<[TK]D-Fender> log out ;)
14:10.53[TK]D-Fenderangryuser: SAME COMMAND <-
14:11.04angryuser<[TK]D-Fender> ok ;)
14:11.22angryusersometimes it is too simple
14:13.13*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
14:13.17eeos_I have configured my asterisk to connect to an external provider. Every time we try to open a call to an external number through the external provider using a sip softphone, asterisk return an error 603
14:13.44eeosany clue? I used the JTodd configuration files as starting point
14:14.47*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)
14:14.50angryuserAgentCallbackLogin([AgentNo][|[options][|[exten]@context]]) what is exten option for?
14:14.52eeosI am confused because everything seems to work properly, and the configuration seems correct
14:15.57angryuser<eeos> pastabin something
14:15.59mosty603 means declined. are you trying to dial a phonesex line or something? ;)
14:16.12[TK]D-Fenderangryuser: It dials the agent via the dialplan.  use your imagination.
14:16.43angryuserokay
14:19.52eeosangryuser: http://phpfi.com/239824
14:19.53eeosand
14:21.01angryuserpeer options also
14:21.09[TK]D-Fendereeos: That is thoroughly USELESS
14:21.39angryuserwhy do you make hangup?
14:21.58eeoshttp://phpfi.com/239826 that is sip.conf
14:22.05eeos[TK]D-Fender: why????
14:22.13eeos[TK]D-Fender: internal calls work properly.
14:22.15angryuserstrange, and playfile, when g option is not actiated
14:22.30eeos[TK]D-Fender: I do not understand.
14:22.50*** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com)
14:23.28*** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com)
14:24.00angryuseri got to work, fender please ;)
14:24.21eeoswe get the error 603 when we try to get out on the provider.
14:24.29eeosdialing 6 as first to get the line
14:24.55d-nrw-marci have solved my problem
14:25.23d-nrw-marcin my sample.call WaitTime: 30 works great.
14:25.43eeos[TK]D-Fender: why is it thoroughly USELESS?
14:25.43ManxPower603 Declined
14:26.33eeosangryuser: I make hangup when the call's finished. Is that wrong?
14:26.53Uateco2's hold music is weird
14:27.02Uateci feel like i'm walking through a plane surrounded by bubbles
14:27.08Uatecnot an aeroplane
14:27.37angryuser<eeos> you dont need that, anyway the dialplan is not executed after your call
14:27.46ManxPowereeos: your provider is refusing the call
14:27.51eeosangryuser: why?
14:28.04mostyeeos, ask them
14:28.24eeosso the problem is not in my dialplan, but on the provider side?
14:28.31angryuser<eeos> core show application Dial
14:28.31eeosso the dialpla is good?
14:28.36ManxPowereeos: The 603 is a config error
14:29.15ManxPowerThe dialplan will not continue to execute until Dial has exited (end of the call), since the call is already hungup, using Hangup is useless most of the time.
14:29.40eeosyes, but it does not interfere, does it?
14:29.50eeosit is not the cause of the 603 error
14:29.57*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
14:30.01ManxPowernot is not the cause of the 603 error.
14:30.04eeosor is it?
14:30.09ManxPowerThe cause of the 603 error is in sip.conf
14:30.15ManxPowerOr your Dial line.
14:30.16eeosah! where?
14:30.38ManxPowereeos: 6t03 usually means "bad username or password"
14:30.46eeosManxPower: they both seems to be ok! I have checked against the documentation.
14:30.59ManxPowereeos: The 603 says they are not.
14:31.33eeosManxPower: does fromuser and username require the ""?
14:31.43eeosbecause they are not in the documentation
14:32.36ManxPowerno quotes
14:32.46ManxPoweryou never use quotes unless you really know what you are doing
14:33.09eeosthanks. So the problem is in the data we send, but the structure of the dialplan itself is ok. is that right?
14:33.27ManxPowereeos: I can't tell you that until I see the Dial line you are using.
14:33.37[TK]D-Fendereeos: We see no sip debug, no sip.conf to match those generic looking sip peers you are calling to, etc.
14:33.48ManxPowereeos: I guess 603 could be a codec issue as well.
14:33.56[TK]D-Fendereeos: And you are using depreicated syntax
14:33.57eeosah!
14:34.10ManxPowerBut it is usually a auth problem
14:34.11[TK]D-FenderManxPower: Usually a 488, but yay :)
14:34.24eeoswhat is a sip debug? I posted the sip conf!
14:34.35eeosI cannot post login and password.
14:34.44ManxPowereeos: what is the link to the sip.conf and extensions.conf?
14:35.21eeosextensions.conf  http://phpfi.com/239824 sip.conf  http://phpfi.com/239826
14:36.15*** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
14:36.45ManxPowerDo you really need all that crap in the [whatever] section?
14:36.47casimireeos, in your cli type sip set debug
14:36.58casimirfollowed by the peer that you want to debug
14:37.04eeosManxPower: what crap?
14:37.15ManxPowerfromdomain, proxy, realm
14:37.23ManxPowerI've never used those in 5 years of using Aserisk
14:37.34eeosManxPower: so can asterisk guess?
14:37.36De_Monvoip-info down?
14:37.42ManxPowercontect= as well, since that only applies to incoming calls from the provider.
14:37.53ManxPower..er.. context
14:38.17eeosManxPower: what do you mean? do i not have to declare the context?
14:38.25eeoscasimir: thanks!!!!
14:38.29[TK]D-Fendereeos: It is almost never appropriate to use "nat=yes" with a normal ITSP.
14:39.04eeos[TK]D-Fender: because we are going through the company router that does not
14:39.10eeoss/not/nat
14:39.26eeosso should I remove the line entirely?
14:39.37[TK]D-Fendereeos: No, that mangles the DESTINATION's IP.  This is not how you configure * to deal with YOUR NAT.
14:39.41ManxPowereeos: nat=yes is for REMOTE devices that are behind NAT
14:39.58[TK]D-Fendereeos: For which your pastebin tells me you have not done at all.
14:40.12ManxPowereeos: http://phpfi.com/239834
14:40.22eeos[TK]D-Fender: what I have not done?
14:40.29*** part/#asterisk d-nrw-marc (n=mr@p5085f040.dip.t-dialin.net)
14:40.39[TK]D-Fendereeos: Set up * to deal with your being being behind NAT
14:40.48eeosManxPower: would that apply to the connection to the SIP provider?
14:41.06eeos[TK]D-Fender: ?
14:41.15ManxPowereeos: You need to forward ports 5060/UDP and 10000-20000/UDP fro the router to the Asterisk box, you do not have the localnet= and externip options, but none of this should cause the 603
14:41.21eeos[TK]D-Fender: is that not done through dialplan?
14:41.46[TK]D-Fendereeos: How was that not clear?  You say your * is behind NAT.  Your sip.conf is clearly lacking all of the settings required to make that work.
14:41.49ManxPowereeos: ALL THAT IS DONE IN THE DIALPLAN IS CALL ROUTING, NOT AUTH, ACCOUNT, OR PROVIDER SETTINGS
14:41.51eeosManxPower: I have forwarded udp 5060 and port 8000 to 8005
14:42.11eeosbecause these where the ports used by the provider
14:42.21ManxPowereeos: then you are not forwarding the right ports, the ports are listed in rtp.conf
14:42.22eeosi think I messed it up.
14:42.39ManxPowereeos: your provider obviously doen't know what they are doing or you would not be here.
14:42.41[TK]D-Fendergetting COLDER.....
14:42.55casimireeos, your provider might use a different port range for rtp
14:43.07eeosManxPower: you are right, I have been working on that for 4 days now :(
14:43.22eeoscasimir: they told me they use 8000 - 8005
14:43.34eeosso I forwarded those
14:43.34ManxPowercasimir: so waht, you only have to forward the ports that asterisk uses.  the provider ports should be natted correctly without port forwarding
14:44.04ManxPowerforward 10000-20000 as well
14:44.13eeosManxPower: so I have to reconfigure the router? instead of ports 8000 - 8005, I have to foward 10000 - 20000?
14:44.20eeosok
14:44.35[TK]D-Fendereeos: ...
14:44.36[TK]D-Fender~sipnat
14:45.15jbotsipnat is, like, for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
14:45.16ManxPowereeos: you set whatever ports are listed in /etc/asterisk/rtp.conf  If there is no rtp.conf or you are using the default rtp.conf, then the ports that must be forwarded are 10000-20000/UDP
14:45.19yannj_frhello, i have a silly question, can I ask it anyway?
14:45.40eeosvoip-info is down :(
14:45.42angryuser<[TK]D-Fender> i am unablr to logout my agent ;)) used this tutorial http://www.orderlyq.com/asteriskqueues.html used same logic
14:46.00angryuserbut it asks just to ligin each time
14:46.04angryuser*login
14:46.16yannj_frangryuser tu utilises callbacklogin?
14:46.25angryuseroui
14:46.29ManxPower[TK]D-Fender: I've found that page to be almost totally useless.  It gives you all the technical details without even once giving any real world examples.
14:46.31yannj_fravec # ca fait quoi
14:46.34ManxPowerUhgh, the Wiki seems to be down
14:46.36eeosManxPower: yes, you are right it is using the 10000 -20000 in the rtp.conf
14:46.51angryuser<yannj_fr> # pour option?
14:47.14yannj_frtu tape le numero dagent le mot de passe et quand il demande le nouveau poste tu fais #
14:47.20angryuserok
14:47.22[TK]D-FenderManxPower: Is this the poitn where i should sarcastically tell you to go re-write it? :)
14:47.40*** join/#asterisk dlynes (n=dlynes@d207-216-161-56.bchsia.telus.net)
14:47.41ManxPower[TK]D-Fender: I would have to check the scrtipt, but I think it is.
14:48.05[TK]D-FenderManxPower: I do agree it should be re-written.  Perhaps I'll do it myself and post it.
14:48.07ManxPowerI think my response is supposed to be "there is so little usable information it would require just starting from scratch"
14:48.29angryuser<yannj_fr> ok working thanks
14:48.58yannj_frdoes anybody now how to force a phone to unregister and register changing account?
14:49.15[TK]D-Fenderyannj_fr: Depends on the phone
14:49.22eeosManxPower: opened and fowarded the ports, same problem.
14:49.28[TK]D-Fenderyannj_fr: go read its manual :)
14:49.44ManxPowereeos: did you try what I pasted for you?
14:49.49yannj_fr[TK]D-Fender : I am trying with thomson, but the manual is really poor
14:49.53[TK]D-Fendereeos: You still haven't added the necessary options to sip.conf..... forwarding won't do you much good until then.
14:50.02[TK]D-Fenderyannj_fr: So stop buying them! ;)
14:50.23yannj_fris a method to use sip notify?
14:50.26eeosManxPower: for the provider?
14:50.34eeos[TK]D-Fender: what is missing?
14:50.40ManxPowereeos: correct.
14:50.58eeosManxPower: I will try immediately
14:51.07ManxPower(09:41:08) ManxPower: eeos: You need to forward ports 5060/UDP and 10000-20000/UDP fro the router to the Asterisk box, you do not have the localnet= and externip options, but none of this should cause the 603
14:51.17ManxPowereeos: now you have done ONE of the three required things
14:51.24[TK]D-Fenderyannj_fr: Again, this is entirely up to the phone.  I don't know Thompson.
14:52.04*** join/#asterisk denke (n=denke@mehess.adsl.datanet.hu)
14:52.18Strom_MWhen you go off hook on a CAS T1, what are the ABCD RBS bits supposed to be set to on the TX side?
14:52.30eeosManxPower: but your file did not have this information ....
14:52.37eeosis that essential?
14:52.47eeosI did not see it anywhere ....
14:53.02mogjer, your around
14:53.38ManxPowereeos: my file was ONLY the provider section
14:53.44anonymouz666mog: is there any issue related to jabber with a lots of google talk contacts?
14:54.02mognot that i am aware of
14:54.07mogbut like i said before
14:54.11mogi dont talk to strangers....
14:54.38ManxPowermog: you talk to me and I'm pretty strange.
14:54.57mogbut identifiable
14:55.09moganonymouz666 is a quandary to me
14:55.26anonymouz666ast 1.4 is seg fault lots of times with heavy jabber use
14:55.33anonymouz666lots of contacts
14:55.43eeosManxPower: so I have to set localhost and externip
14:56.03mogif you are using enc. it was fixed recently, in regards to library we use
14:56.56De_Monif I call app queue() without a timeout, it should use the defaults in queues.conf riiight?
14:57.06De_Monhttp://phpfi.com/239841
14:57.09anonymouz666sorry... enc?
14:57.15mogencryption
14:57.28[TK]D-FenderDe_Mon: No.
14:57.52*** part/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net)
14:57.52anonymouz666yes I am using enc.
14:58.11eeosManxPower: externip: sould I point it to our public internet address? we only have one.
14:58.18anonymouz666mog: is there anything written about this bug? mantis?
14:58.19De_Mon[TK]D-Fender why shouldn't it?
14:58.38anonymouz666what version is the bug in the library... what version is fixed etc
14:58.39mogwell what version of 1.4 are you using
14:58.43[TK]D-FenderDe_Mon: because there is no global timout in queues.conf.  Only ring-timeout.
14:58.44anonymouz6661.4.4
14:58.48mogit wasnt fixed in their library
14:58.49eeosManxPower: documentation says it should be a valid routable address, but the only routable address we have is the fix ip address we offer to the internet
14:58.52mogwe fixed it in our use
14:58.52*** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue)
14:59.00mogtry doing latest svn checkout of 1.4
14:59.06ManxPowereeos: that is the address you use in extern=ip
14:59.08*** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-fe74fe70049e094f)
14:59.30neverbluemorning
14:59.40De_Monso that timeout is for the ring strategy? ala, trying the next agent or whatever
15:00.12anonymouz666mog: Ok, I will try that. If does not work can I report to you or I am stranger to do that?
15:00.13[TK]D-FenderDe_Mon: Yup
15:00.27*** join/#asterisk adr3nalin3 (n=Foobar@65-123-189-218.dia.static.qwest.net)
15:00.27mogif you sign on with a real nick sure ^_^
15:00.28eeosManxPower: I added the externip=ip to our sip.conf
15:00.39De_Mon[TK]D-Fender so, what does reset do if not determine the number of times the "timeout" is reached
15:00.42anonymouz666it's real! registered!
15:00.46eeosManxPower: but cannot find any reference to localhost in the documentation
15:00.46De_Monnot reset, retry
15:01.17*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)
15:01.44eeosManxPower: or did you mean localnet?
15:01.45De_Monwith timeout=20 and retry = 1 or 0, it rings for 40 seconds and times out
15:02.56eeosManxPower: I think you meant localnet, to translate the externip into local addresses
15:03.05[TK]D-Fender; How long do we wait before trying all the members again? ; retry = 5
15:03.47[TK]D-FenderDe_Mon: Before describing the results of little bits of things you've tried perhaps you could show us teh BIG picture...
15:04.25De_Monthats about it, I expected retry to be the number of times a ring strategy is used, and it's not behaving the way I expected
15:05.02[TK]D-FenderDe_Mon: Well its clearly not what you expected, and is quite likely exacly what is documented...
15:05.37[TK]D-FenderDe_Mon: If it was doing what you expected, we wouldn't be hearing about it ;)
15:07.39eeos[TK]D-Fender: ManxPower that eejit of the provider sent the wrong domain name, can you believe?
15:08.02[TK]D-Fendereeos: I can believe anything.  ESPECIALLY stupidity
15:08.33[TK]D-Fendereeos: Ignorance is hig on that list too
15:09.57De_Monits a pretty terse description...
15:10.53[TK]D-FenderDe_Mon: Sorry... no dice :0
15:10.58De_Monthe term 'long' could refer to time or tries. 'all members' could just as easily be interpted as "the queue"
15:11.03[TK]D-FenderDe_Mon: Pretty damn BLATANT :)
15:11.07eeosOK, now i have made all modifications in the sip.conf, and I am sip debug on
15:11.12[TK]D-Fender<- Zen master of the blatantly obvious
15:11.12eeosbut still error 603
15:12.20[TK]D-FenderDe_Mon: ring member(s) for timeout X, wait without ringing for retry Y
15:12.35[TK]D-FenderDe_Mon: wash, rinse, repeat :)
15:12.54*** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net)
15:13.22Kattyhmm.
15:13.43eeosI get "Registration for 'whoever@whatever.net' timed out, trying again (Attempt #10)" in sip debug [TK]D-Fender ManxPower
15:13.47eeosfrom CLI
15:14.02[TK]D-Fendereeos: That seriously means your NAT setup is screwed
15:14.53eeos[TK]D-Fender: in what sense? I opened all the proper ports, set up the proper addresses ....
15:15.01eeosscrewed in what sense?
15:15.15[TK]D-Fendereeos: timing out means the conversaion isn't even coming back
15:15.31eeos[TK]D-Fender: what can cause that?
15:15.32[TK]D-Fendereeos: You're not being refused, you're just no getting an ANSWER
15:15.54eeosah! :( that sounds bad
15:16.20*** part/#asterisk zozo75 (n=cyrilz@dra38-1-82-230-120-110.fbx.proxad.net)
15:16.30eeoshow do you connect it to the nat?
15:16.43eeoscould be that they use a different pot for regsitration?
15:17.04eeosso I have opened 10000 - 20000 and 8000 - 8005 to be on the safe side
15:17.07*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
15:17.10anonymouz666mog: svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 ?
15:17.20eeosbut they are using a different port for registration [TK]D-Fender
15:17.44*** join/#asterisk andyd (n=andyd@212.183.134.209)
15:18.17mogyeah
15:20.41*** join/#asterisk bbryant (i=brett@nat/digium/x-77c80fba2e64d6ef)
15:21.24De_Monwhat would cause a queue to do an 'exiting on time-out cycle' if no timeout is passed to the Queue()?
15:22.36eeos[TK]D-Fender: I get an answer if I register the softphones directly with the provider, so registration works
15:23.01*** part/#asterisk andyd (n=andyd@212.183.134.209)
15:24.59eeoscasimir: is there a way to save debug information?
15:27.15[TK]D-Fendereeos: Softphones are smarter about NAT than *
15:27.22jkiffIs there documentation somewhere that'll tell me which versions of spandsp and asterisk go together?  I haven't been able to find anything on the wiki or google yet.
15:28.06eeos[TK]D-Fender: but what is wrong with my NAT? I revised everything.
15:28.26[TK]D-Fendereeos: I don't know what you missed.  Pastebin it.
15:28.41[TK]D-Fendereeos: I'm only psychic on TUESDAYS
15:28.54dlynesjkiff: 0.0.2 with asterisk 1.2.9.1
15:29.21eeos[TK]D-Fender: I opened ports 10000/20000, 8000/8005
15:29.30De_Monhmmmmmm
15:29.31eeos[TK]D-Fender: externip=our pub lic ip address
15:29.35dlynesjkiff: you won't be able to use anything newer than 1.2.9.1 reliably, because anything newer is not officially supported by spandsp
15:29.37[TK]D-Fendereeos: What about *SIP*?
15:29.47eeos[TK]D-Fender: well, that of course
15:30.12[TK]D-Fendereeos: And still please pastebin your sip.conf.
15:30.12eeos[TK]D-Fender: port 5060 is open
15:30.27[TK]D-Fendereeos: And while you're at it pastebin some backup for your port forwarding.
15:31.13*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
15:32.14jkiffdlynes: Hmm, I see.  I believe you, but is there a page somewhere that says that.  Either google is failing me or I'm failing at google since I can't find it.  :-P
15:32.44dlynesjkiff: not that I'm aware...that's the official word I've gotten from the author of spandsp by talking to him
15:32.51eeos[TK]D-Fender: sip.conf http://phpfi.com/239849 dlynes :)
15:33.09dlynes[TK]D-Fender: ah...guess you didn't need a bat, after all :)
15:33.17jkiffdlynes: Ah, I gotcha.  Thanks!
15:33.22De_Mon[TK]D-Fender ahh, see.  option n tells Queue to exit on timeout. but the only "timeout" it looks at is the one passed by App Queue(). So, I didn't specify one. It's using a default value of something from somewhere
15:33.49Kattyif i have s,1,Dial(SIP/110,30,t)
15:33.53Kattywhat does the 30,t mean?
15:33.56Kattytimeout is 30?
15:34.01[TK]D-Fendereeos: You also need "nat=yes", "canreinvite=no" under [general].  Doublecheck your IP and that localnet range feels funny, but I'll trust you that that its exactly right
15:34.02dlynesKatty: Yes...30 seconds
15:34.13Kattydlynes: so the t is in reference to the 30?
15:34.25*** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue)
15:34.26De_Mon[TK]D-Fender From what you've told me, app_queue.so should rais an error if option n is used and no timeout is specified in the command.
15:34.27dlynesKatty: After which you can catch the value of ${DIALSTATUS}
15:34.28Kattydlynes: t = timeout
15:34.36dlynesKatty: No, 't' means transfer, as does 'T'
15:34.44Kattydlynes: transfer?
15:34.52De_MonKatty its a feature
15:34.55[TK]D-FenderKatty: So phones can use DTMF to transfer the calls.
15:35.00Kattydlynes: what purpose does transfer serve in this senario?
15:35.01eeos[TK]D-Fender: I opened all the ip addresses on our net excluding the routers and printers
15:35.07[TK]D-FenderKatty: thoroughly useless with your polycoms :)
15:35.08Kattydlynes: what can i do for me.
15:35.12eeos[TK]D-Fender:is that wrong
15:35.15dlynesKatty: Check voip-info.org for which is which...I think it's 't' for caller can transfer and 'T' for callee to transfer, but I can't remember for sure...I never use that feature
15:35.25Kattydlynes: k
15:35.30eeos[TK]D-Fender: sorry, s/I opened/I put
15:35.39[TK]D-Fendereeos: eeos>[TK]D-Fender: I opened all the ip addresses on our net excluding the routers and printer <---------- HUH!?!?!
15:35.45dlynesKatty: It's predominantly for zaptel channels, I think
15:35.54[TK]D-Fendereeos: open the whole subnet.
15:36.02eeos[TK]D-Fender: I listed all the ip addresses on our net excluding the routers and printers
15:36.04eeos[TK]D-Fender: ok
15:36.12[TK]D-Fendereeos: If your printer REALLY wants to try placing a call through *, LET IT
15:36.13eeos1/255?
15:36.40[TK]D-Fendereeos:  192.168.1.0/255.255.255.0
15:37.46eeos[TK]D-Fender: but the syntax on the documentation is different :(
15:38.15[TK]D-Fendereeos: Both are valid.
15:38.15[TK]D-Fendereeos: you can use /24 if you like
15:38.35eeos[TK]D-Fender: no I did exactly as you said
15:39.13eeosput aso the nat = yes + canreinvite=no in general
15:39.27eeoswow! there is not timeout anymore!
15:40.55eeos[TK]D-Fender: thanks a lot .... I have not understood where the mistake was
15:40.57eeos:)
15:41.26[TK]D-Fendereeos: /240 masks 240 BITS <---------
15:41.34eeos[TK]D-Fender: but I wtill get the 603 declined
15:41.43dlynes[TK]D-Fender: heh
15:41.46[TK]D-Fendereeos: this is a process.
15:42.04eeos[TK]D-Fender: sorry, but I *still* get the 603 declined
15:42.07[TK]D-Fendereeos: So lets see where you've gotten yourself to then.
15:42.30eeosnow our asterisk clearly registers with the provider because we do not have an error anymore
15:42.37[TK]D-Fendereeos: Pastebin the CLI output with SIP debug and your sip.conf
15:42.56*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)
15:43.15*** join/#asterisk Dio_ (n=dima@83.143.239.85)
15:44.14eeos[TK]D-Fender: there are all the details of the company's connections in the debug :(
15:44.18Dio_hello, I need to reopen 9658 bug. Can anybody help me with that?
15:45.17fileDio_: open a new bug with all available info
15:45.56[TK]D-Fendereeos: Let me be blunt : we don't really give a shit about your company, or your provider.  We are jsut trying to help.
15:45.57*** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net)
15:46.21Dio_file: provided fix for 9658 break some stuff. You think it should be a new bug report?
15:46.29filewhat does it break?
15:46.37bcnxHi all: if you had the choice: would you install Astersik on Centos 4 or Centos 5?
15:46.50bcnxAsterisk, sorry
15:47.35[TK]D-Fenderbcnx: 4.  its better documented.
15:47.35bcnxD-Fender, thanks again
15:47.35[TK]D-Fenderbcnx: as far as what problems may arise
15:47.36tzangerhow do you copy/paste on wx?
15:47.36tzangeroops
15:47.37*** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue)
15:47.39dlyneswxWindows?
15:48.15bcnxunfortunately my RAID controller is not supported in Centos 4 (fakeraid), so I guess I'm looking at software raid
15:48.29bcnxI hope that works fine with Asterisk
15:48.30eeoshttp://phpfi.com/239851
15:48.37eeos[TK]D-Fender: I masked the sensitive data
15:48.38fileDio_: but anyway... if you file a bug with all the information it will get looked at asap
15:50.07[TK]D-Fendereeos: you are declining to your PHONE.
15:50.12[TK]D-Fender<PROTECTED>
15:50.14[TK]D-Fender[Jun  6 16:44:23] WARNING[9003]: pbx.c:1779 pbx_extension_helper: No application 'SetCIDName' for extension (internal-post, 601535648555, 2)
15:50.15[TK]D-Fender<PROTECTED>
15:50.17*** join/#asterisk geoaxis (n=geoaxis@unaffiliated/geoaxis)
15:50.23geoaxishello people
15:50.26[TK]D-Fendereeos: That application is deprecated.
15:50.27_VoiceMeUp_COMSetCID(Name) ?
15:50.31eeos[TK]D-Fender: declining my phone?
15:50.38geoaxisI am looking for a guide of some sort for asterisk svn compile
15:50.46SplasPoodSet(CID(name)=Your Mom)
15:50.47_VoiceMeUp_COMshow application SetCallerID
15:50.49[TK]D-Fendereeos: * is refusing your UA <0
15:51.00_VoiceMeUp_COM1.2 or 1.4
15:51.11eeosrefusing my ua < 0. what does that mean?
15:51.18[TK]D-Fendereeos: * is refusing your UA <-
15:51.27SplasPood_VoiceMeUp_COM: if that was meant for me, both.
15:51.44geoaxisi get this msg on default ./configure
15:51.46eeoswhat is my ua?
15:51.49[TK]D-FenderSplasPood>Set(CID(name)=Your Mom) <- WRONG
15:51.56Dio_file: I wrote you description in private messages
15:52.06dlyneseeos: user agent string
15:52.06[TK]D-Fendereeos: that other account in your sip.conf.  a soft-phone or something, no?
15:52.14[TK]D-Fenderdlynes: NO
15:54.32eeos[TK]D-Fender: you mean already using the connection?
15:54.32fileDio_: I only accept private messages from people I know and I'm soon leaving, but if you file a report with the info - someone will look at it asap
15:54.32[TK]D-Fenderdlynes: He named an entry in sip.conf that...
15:54.33*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
15:54.33dlynes[TK]D-Fender: ah
15:54.33geoaxisThe existing menuselect.makeopts file did not specify  that 'cdr_adaptive_odbc' should not be included.  However, either some dependencies for this module were not found or a  conflict exists.
15:54.33angryuserdo i need tu put any option to make dialplan continue to execute after i do AgentCallBackLogin ?
15:54.34[TK]D-Fendereeos: I mean that PHONE entry you made in si.conf
15:54.34dlynes[TK]D-Fender: thought it was a generic question
15:54.34SplasPood[TK]D-Fender: whats wrong..  Need quotes?
15:54.34Dio_file: oops... I'll copy paste it in bug report
15:54.35eeos[TK]D-Fender: you mean another user is using the ua entry?
15:54.35*** join/#asterisk philippel (n=p_lindhe@c-24-17-254-189.hsd1.wa.comcast.net)
15:54.35[TK]D-FenderSplasPood: "CID" is NOT a function.
15:54.36dlynes[TK]D-Fender: yeah...i see now...he's got a peer called 'ua'
15:54.36eeosanother user on the internal network?
15:54.36SplasPoodoh shit
15:54.36SplasPoodCALLERID
15:54.36SplasPoodsorry, seeing SetCIDName through me off
15:54.36eeosusing the ua peer?
15:54.36SplasPoodthrew, even.
15:54.36[TK]D-Fendereeos: You seriously ahve no idea what you've setup * for at all do you?
15:54.37eeosno I know, but did not understand your sentence about ua ....
15:54.49*** part/#asterisk philippel (n=p_lindhe@c-24-17-254-189.hsd1.wa.comcast.net)
15:54.50eeosand do not understand how it is connected with setcallerid being deprecated ....
15:54.56eeosI am seriously confused now
15:55.10[TK]D-Fendereeos:Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKxmevtmmp;received=192.168.1.40;rport=5060
15:55.19[TK]D-Fendereeos: From: "ua" <sip:ua@192.168.1.36>;tag=welud
15:55.27De_Moneeos they are not related
15:55.28[TK]D-Fendereeos: What the hell are at these 2 IP's?
15:55.59[TK]D-Fendereeos: The application fails so the call is TRASHED.
15:56.16eeosgosh, it looks like someone else's soft phone is logged in the same user
15:56.48eeoswait
15:56.58[TK]D-Fendereeos: Ok, you need to start over completely from scratch with *.  Go read the BOOK for a while, do some INTERNAL calls and mess around with stuff, THEN deal with your ITSP
15:57.12[TK]D-Fender~book
15:57.25jbotwell, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
15:57.29eeosah here it is! there was a user playing with her soft phone.
15:57.43eeos[TK]D-Fender: I have read that thing several time over
15:57.56eeos[TK]D-Fender: made her log out
15:58.05eeosthe internal phoes work properly
15:58.26eeosI went on another machine today and left the ua test user logged in by mistake
15:59.41eeos603 has definitively nothing to do with the other user logged in, she is now logged out, we have exactly the same problem.
16:00.33[TK]D-Fendereeos: YOUR phone is being declined.  You are not even CALLING your provider!
16:00.52[TK]D-Fendereeos: Fix your DIALPLAN.  That app being deprecated is the problem.
16:01.10Kattyis there a command to see which zap channels are In Use?
16:01.14Kattylike sip show channels
16:01.17Kattybut for zap
16:01.29[TK]D-Fendereeos: 601535648555@internal-post:1] SetCallerID("SIP/ua-081c6b28", "user") in new stack <- bad
16:01.30mostyzap show channels
16:01.32Cresl1nzap show channels?
16:01.36Cresl1nor show channels
16:01.38Kattythat just shows me al of them
16:01.43*** join/#asterisk dracosilv (n=draco@CPE-65-29-47-173.wi.res.rr.com)
16:01.44Kattyah ha!
16:01.46Kattythanks
16:01.58eeos[TK]D-Fender: what application being deprecated???? I used the applications from the book!!!!
16:02.09eeos[TK]D-Fender: core show applications
16:02.10[TK]D-Fendereeos: exten => 601535648555,1,Set(CALLERID(name)=use") in new stack <- bad
16:02.14RyushinI'm trying to set up two asterisk server to talk to each other via iax.  But I keep getting this message when the other server tries to connect: Peer 'testiax' is not dynamic (from 172.16.2.1)
16:02.36eeosSetCallerID: Set CallerID, that is shown by my core show application in cli!
16:02.38[TK]D-Fendereeos: the book was NOT writting for * 1.4 and you have clearly not read the dos or read the blatant error that clearly it doesn't exist anymore.
16:03.02angryuser<Ryushin> set host=dynamic
16:03.03[TK]D-Fender[Jun  6 16:44:23] WARNING[9003]: pbx.c:1779 pbx_extension_helper: No application 'SetCIDName' for extension (internal-post, 601535648555, 2)
16:03.04eeosI am actually using core show applications on the CLI
16:03.14[TK]D-Fender^^^^^^ This is NOT LYING.
16:03.26angryuser<Ryushin> in iax.conf for peers/friends
16:03.37eeos[TK]D-Fender: I know, but why is core show applications on my server showing this application????
16:03.46[TK]D-Fendereeos: SHOW ME
16:03.53RyushinI was trying to hard set it so I didn't have to use dynmaic.  But I'll try that.  Thanks angryuser.
16:04.29[TK]D-Fendereeos: And before you waste your time, realize that SetCallerID is NOT SetCIDName, nor is EITHER appropriate
16:04.46eeoshttp://phpfi.com/239856
16:04.50eeoshere it is
16:04.57[TK]D-Fendereeos: See above.
16:05.00angryusermahahaha
16:05.09[TK]D-Fender*sigh*
16:05.40angryuseri would no last as long as fender
16:06.04mostyRyushin, if you set host=<some ip address or hostname> then the client doesnt need to register
16:06.11eeosI actually have not setcidname?
16:06.21[TK]D-Fendereeos: ..............
16:06.50angryusermhahaha
16:07.07[TK]D-Fendereeos: I'd ask if if I have to spell it out for you.... but I already DID
16:07.34Dio_file: if you want you may take a look bug # 9900.
16:07.49[TK]D-FenderBRB, lunch for 15 mins
16:07.54Ryushinmosty:  Yea, that's what I did, and it didn't like it.  Oh well, I'll work on that part later.
16:07.58RyushinThanks though.
16:08.29*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
16:09.20eeos[TK]D-Fender: I have removed any call to all functions trasmitting the callerid or the name.
16:10.04*** join/#asterisk davidcsi (n=davidcsi@82.158.35.53.static.user.ono.com)
16:10.43davidcsihello all, i have an * box with REALTIME, which is reporting a lot of "auto_congest:  Auto-congesting SIP" and many calls are failing... any ideas?
16:11.01eeos[TK]D-Fender: now I would just like to understand what documentation shall I follow, i order to have the correct syntax, because I have gone through all the documentation that all of you pointed my out, but there is always a mistake her eor there, and I get flamed. that is not correct at all!!!! f the documentation is rog it should say it!
16:11.18eeosanyway ....
16:11.27angryuser<PROTECTED>
16:12.04*** part/#asterisk Dio_ (n=dima@83.143.239.85)
16:12.34angryuser<eeos> to get a correct answer, ask a well formulated question
16:13.16eeoswhat is the offical documentation for the 1.4.1 version of asterisk, without syntax errors?
16:15.13bcnxHi again: has anybody got any experience with using software RAID for Asterisk? I hear it can burden the CPU quite a bit while rebuilding stuff
16:15.21[TK]D-Fendereeos: just comment out that line entirely (It looks like a crap action to take REGARDLESS of syntax)
16:15.28[TK]D-Fendereeos: And renumber around it.
16:15.34*** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net)
16:16.11[TK]D-FenderDamn short lunch... pre-cooled soup helps...
16:16.26angryuser<bcnx> i am using it ith raid 1
16:17.17angryuser<bcnx> no big problems till now
16:17.37eeos[TK]D-Fender: ok, thanks a lot!
16:17.48[TK]D-Fendereeos: :)
16:18.28eeos[TK]D-Fender: I think I need some fresh air now .... then I will do some more testing!
16:18.30[TK]D-Fenderdavidcsi: Try some NyQuil ;)
16:18.39bcnxangryuser: have you had abnormal shutdowns? I hear it does a rebuild then, stressing the cpu quite a bit
16:18.39eeos[TK]D-Fender: thanks!
16:20.31Uatecabunai
16:21.04angryuser<bcnx> it does a rebuild only once, and in case of HDD failure, take some time, but not stress much cpu in my case (2xamd 3000 core venice 1gb ecc 2xscsi barracuda)
16:22.48Trevor_bTK know if the m11,12 and 22 use the same disconnect interface?  Noticed some notes when looking them up that said there were more then one disconnect style?
16:24.07bcnxangryuser: thx!
16:27.28*** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net)
16:28.00s0ckangryuser: that nick completer is quite hard to read :P
16:28.13[TK]D-FenderTrevor_b: The amp is one part.  There is an RJ connector on the headset side for which you can get a variety of connection methods.  Most common is their Polaris quick-disconnect.
16:29.03angryuser<s0ck> at least it is working;)
16:30.11*** join/#asterisk Shoeb (n=chatzill@64.34.69.9)
16:30.12s0ck:)
16:30.21s0ckit looks like you've pasted the text
16:30.26davidcsianyone knows what's the defference between "SIP SHOW CHANNELS" and "SHOW CHANNELS"?? I'm getting 1200 channels with SIP SHOW CHANNELS but 20 with SHOW CHANNELS..???
16:30.28*** join/#asterisk oej (n=olle@AMontsouris-152-1-5-176.w82-123.abo.wanadoo.fr)
16:30.32s0ckmirc default wraps the < > around the nick init
16:31.06ShoebHello, getting an error when someone calls in, and it doesn't even come through.
16:31.07Shoebchan_sip.c:10654 handle_request_invite: Failed to authenticate user "TORONTO      ON" <sip:9055555555@207.174.202.4>;tag=as5c5ff47c
16:31.08[TK]D-Fenderdavidcsi: "sip show channels" shows things like register attempts, lingering conenctions that are scheduled to die, etc. "show channels" shows whats really in progress.
16:31.59[TK]D-FenderShoeb: They do not have a proper account set up in sip.conf or if you're intending on allow un-auth'd calls, you didn't do so properly
16:32.05davidcsiFender, thanks. and how do i kill all thouse lingering channels??
16:32.13Trevor_b[TK]D-Fender: Yeah between plantronics and voipsupply im having no fun getting a good list of headsets to use on an m22.  Non amp hooked up headset isnt really worth it even on poly's is it?
16:32.16davidcsii htink it is affecting the system performance
16:32.40[TK]D-FenderTrevor_b: Its WEAK.  I really advise against it.
16:33.30[TK]D-Fenderdavidcsi: not sure really.... could be an * garbage collection issue... this issue is "out of my league"
16:34.18Shoeb[TK]D-Fender: Hmm, weird. Let me double check, it isn't there either.
16:35.52*** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com)
16:36.38*** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com)
16:37.52IOscanneranyone know if there is a way to monitor SIP calls.
16:39.53Trevor_byeah, hmm.  Guess ill have to look around, $82 seems decent on the AMP.  Although the S11 (if the amp doesnt suck) for $90 from Plan themselves seems a good price. Not sure if the amp is decent though.
16:41.29[TK]D-FenderTrevor_b: M12/22's have done well here.
16:42.41angryuser<IOscanne> monitor in which way?
16:42.44cpmAny clues on some oddball hardware that can be used to make BRIs out of a PRI like my current pbx does?
16:42.54angryuser<IOscanner> what info do you need exactly
16:43.08IOscannerEither to be able to talk to both parties
16:43.19IOscanneror  whisper mode
16:43.27IOscanneror just listen
16:43.46*** join/#asterisk [hC] (n=hardcore@190.10.13.145)
16:44.18angryuser<IOscanner> not interested in that, i just set *8 to record in mixmonitor
16:45.31IOscannerThis needs to be one from a manager not from the person on the phone
16:45.52[TK]D-Fendercpm: You mean you effectively want a PRI>BRI channel-bank?
16:47.26*** join/#asterisk BSD_Tech[laptop] (n=UnixDog@adsl-69-230-169-41.dsl.irvnca.pacbell.net)
16:47.32BSD_Tech[laptop]ok major issue here
16:47.34cpm[TK]D-Fender, yeah, just that
16:48.00BSD_Tech[laptop]calling from a iax extension to a sip extension on the same box no audio
16:48.26Trevor_b[TK]D-Fender: So your setup (sans phone) runs about $200 per station?
16:48.31[TK]D-Fendercpm: Ah, that falls squarely in the realm of "Crazy-Euro" gear...... go ask some EU telecom guys... this is not stuff typically seen along-side VoIP gear
16:49.04[TK]D-FenderTrevor_b: A good chunk over actually... I got the H261 Binaural... REALLY nice headsets.
16:49.17[TK]D-FenderTrevor_b: At least $250 total
16:49.36[TK]D-FenderTrevor_b: I consider it an investment in customer/rep satisfaction.
16:49.40cpm[TK]D-Fender, yeah, it's pretty wierd. My current PBX has a 8 port BRI card, and it's configured to strip 6 BRIs from one of my PRIs, which we use for our 2 VTC suites
16:49.59cpmIt's a major issue for moving to * here at work.
16:50.16[TK]D-Fendercpm: If you slapped a bunch of BRI cards into a box you could ahve * do it.  Ugly of course and stability.. heh, whats that? ;)
16:50.26Trevor_bYeah retail on plans site is 120 for the m22 and 103 or 120 depending on the N series.  Currently the call centers we handle are using the Plantronics DSP-400.  Amazing audio on a softphone setup.  But I am tired of soft phone issues.
16:50.32[TK]D-Fendercpm: Ditch BRI.... losing deal
16:50.35anonymouz666Can I use Hangup with ExecIf?
16:50.41anonymouz666What would be the last parameter?
16:50.48cpmNot an option I'm afraid, VTC is still deep in BRI land
16:51.07[TK]D-Fendercpm: Using BRI for phones?
16:51.16cpmVideo Teleconference
16:51.17[TK]D-Fenderanonymouz666: Sure
16:51.20cpmcodecs
16:51.43[TK]D-Fendercpm: EEK... ok, you are in Specialty-Land.  Go talk to a pro.
16:51.50anonymouz666sure?
16:51.56cpmThe pros so "Uhh, nope"
16:52.00cpms/so/say
16:52.02cpm:)
16:52.10anonymouz666exten => 12,1,ExecIf($["${CALLERIDNUM}" = "300"]|Hangup|)
16:52.11[TK]D-Fendercpm: Go get some K-Y ;)
16:52.17[TK]D-Fenderanonymouz666: no other parms
16:52.28anonymouz666exten => 12,1,ExecIf($["${CALLERIDNUM}" = "300"]|Hangup)
16:52.30cpmalready swallowed gallons. (Coral IPx 500 PBX)
16:52.53[TK]D-Fendercpm:  Thought something smelled fishy ;)
16:53.13cpmWe finish writing that horror off this year, and I really want to be done with it.
16:53.46cpmbut I really don't want to go back to BRIs from 'the phone company' and their per minute charges
16:53.58cpmper minute per channel
16:54.38cpmmaybe there is a magic box i can get that will split BRIs off BEFORE the the PRI card, , ,hrmm
16:54.49Trevor_banyone looked into Polycom's Video conferencing package?  I never have personally.
16:54.57cpmjust means less available B channels, but so what
16:55.09cpmTrevor_b, yeah, we sell 'em. Nice boxes
16:56.11BSD_Tech[laptop][Jun  6 09:54:42] WARNING[9521]: channel.c:718 ast_best_codec: Don't know any of 0x0 formats
16:56.11BSD_Tech[laptop][Jun  6 09:54:42] WARNING[9521]: chan_sip.c:2885 sip_call: No audio format found to offer. Cancelling call to sip:123
16:56.27BSD_Tech[laptop]but the codecs are set to ulaw
16:56.44BSD_Tech[laptop]so it seems we have a issue with sip to iax transcoding
16:57.01[TK]D-FenderBSD_Tech : pastbin the call with SIP debug & your sip.conf
16:57.55*** join/#asterisk `pariah (n=josh@unaffiliated/pariah)
16:58.22`pariahThe asterisk console is telling me about a bad file descriptor coming from one of my SIP clients
16:58.32`pariahhere is the output from the console http://pastebin.ca/5448683
16:58.46BSD_Tech[laptop]I am uing a iax client to call a external sip exten I have makked in my extensions .conf
16:58.55rob0Got my zaptel issue fixed, probably just a FUBAR zapata.conf.
16:59.02`pariahsorry its http://pastebin.ca/544868
16:59.05apturayup
16:59.08BSD_Tech[laptop]and I am using the asterisk gui so sip.conf wnt help
16:59.11BSD_Tech[laptop]brb
16:59.27`pariahBSD_Tech[laptop]: there is a channel for ast gui
17:00.25*** join/#asterisk alrs (n=lars@pozug.com)
17:01.08angryuserbye bye
17:01.09BSD_Tech[laptop]yes but its mainly for dev right now
17:02.15Shoeb[TK]D-Fender: I just checked, sip.conf does have the correct username/password and stuff.
17:03.06*** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net)
17:03.15[TK]D-FenderShoeb: they aren't using whatever account you set up properly then
17:03.54[TK]D-FenderBSD_Tech : Fine, just the debug info then
17:03.57Shoeb[TK]D-Fender: Who is 'they'?
17:04.13[TK]D-FenderShoeb: This person calling into your server
17:04.41Shoeb[TK]D-Fender: It's not just this person, I get that error for all inbound calls.
17:05.00*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
17:05.39[TK]D-FenderShoeb: pastebin up the failure and the entry you believe should support the call.
17:06.44*** join/#asterisk toma (n=toma@84-53-90-221.wxdsl.nl)
17:07.14tomaanyone who uses voiceone ?
17:07.15*** join/#asterisk BATHORY (n=kleber@200.245.215.130)
17:07.33BATHORYwhat best gui for asterisk?
17:08.40`pariahanyone know what this means? [Jun  6 05:08:10] WARNING[3133]: chan_sip.c:1754 __sip_xmit: sip_xmit of 0x9d87d48 (len 516) to 10.0.1.100:5060 returned -1: Bad file descriptor
17:08.45[TK]D-FenderBATHORY: vi
17:09.30BATHORY[TK]D-Fender good answer
17:09.35Shoeb[TK]D-Fender: PM pasted.
17:10.12BATHORY[TK]D-Fender I need web interface, do you know?
17:10.22toma`pariah: i saw that when some other app was using the socket
17:10.35[TK]D-FenderBATHORY: Then I'll answer in context : www.scopserv.com
17:10.49`pariahBATHORY: look at AsteriskNOW but dont ask questions about it here
17:11.46`pariahtoma: im looking in netstat, i dont see anything on 5060
17:12.11toma`pariah: ok, then i dont know
17:12.55toma[TK]D-Fender: what does it cost?
17:12.58*** join/#asterisk n0n4m3 (n=NoName@noname.rula.net)
17:13.02n0n4m3evenin
17:13.05[TK]D-Fendertoma: Go ask.
17:13.13tomai am ;-)
17:13.22[TK]D-Fendertoma: THEM YOU LAZY ASS :)
17:13.31toma;-)
17:14.18tomai'm not going to try that, what knowing what it will cost
17:14.26`pariah[TK]D-Fender: have you ever saw that error i posted?
17:14.44BATHORY`pariah oks, i get asteriskNOW, but it's beta for while.
17:14.53tomaBATHORY: i'm trying voiceone, but i;ve no luck in making calls actually
17:14.58[TK]D-Fender`pariah: No idea about it (by itself)
17:15.18BSD_Tech[laptop]http://pastebin.ca/544912
17:15.43BATHORYtoma bug?
17:15.45`pariahit just keeps coming, i can't see anything listening on 5060 :(
17:16.03[TK]D-FenderBSD_Tech : I need the sip debug for that..
17:16.14tomaBATHORY: no idea, their forums state it should simply work (of course)
17:16.17BSD_Tech[laptop]ok brb
17:23.12geoaxishello people
17:24.48geoaxiswhen i run the default asterisk svn compile, it shows me that h232 has an accepting port on  my system
17:25.53Qwell[]geoaxis: yeah, it does that..  either remove the module and disable it in menuselect, or noload it in /etc/asterisk/modules.conf
17:26.45BSD_Tech[laptop]brb tdk have to run the roomie to work sorry
17:27.55geoaxisQwell, would it be chan_h323.so
17:28.00Qwell[]yes
17:29.39*** join/#asterisk Skarmeth (n=Skarmeth@201009036240.user.veloxzone.com.br)
17:30.25geoaxiswell i also get callbook,
17:30.47geoaxisfor now i am only interested in SIP, and that too to the extent of monitoring it via SNMP
17:31.23BSD_Tech[laptop]http://pastebin.ca/544957
17:31.30BSD_Tech[laptop]there is a sip debug tk
17:32.20*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
17:32.50[TK]D-FenderBSD_Tech : pastebin your sip.conf
17:33.49davidcsidoes anyone knows what this means?: sipsock_read: We could NOT get the channel lock for SIP
17:36.00geoaxiswhat is callbook which runs on port 2000 (any thing compareble to SIP)?
17:36.07*** join/#asterisk codazoda (n=Joel@ip69-223.konnections.com)
17:36.26Qwell[]geoaxis: it's skinny
17:36.30Qwell[]chan_skinny, rather
17:37.04codazodaI typically use analog cards, but I'm currently building a system that will never have analog hardware in it.  Do I need zaptel for anything (like zt_dummy) on VOIP only box?
17:37.16Hmmhesaysno
17:37.21Qwell[]codazoda: only if you want to use iax2 trunking or meetme
17:37.26Qwell[](most people do use the latter)
17:37.57davidcsidoes anyone knows what this means?: sipsock_read: We could NOT get the channel lock for SIP???
17:38.26*** join/#asterisk steliosk (n=Stelios@62.169.217.209)
17:38.37geoaxisQwell, i have module_sip loaded but its not bound to any known ports
17:38.42Qwell[]5060 udp
17:38.47Qwell[]netstat doesn't show udp by default I don't think
17:39.09Qwell[]I might just be making that up...
17:39.24BSD_Tech[laptop]http://pastebin.ca/544978
17:39.37*** join/#asterisk irule (n=irule@189.164.43.19)
17:39.50geoaxisQwell, yes well udp doesnt show up
17:39.52Kattyweeeeeeee!!!
17:40.02geoaxisbut it should also be bound to tcp5060
17:40.05[TK]D-FenderBSD_Tech : allow = ulaw,gsm,g729
17:40.06geoaxisfor reference
17:40.07Qwell[]geoaxis: no, just udp
17:40.10NuggetSIP is udp, not tcp.
17:40.12Qwell[]we don't support tcp
17:40.16[TK]D-FenderBSD_Tech : Last I'm aware yuo cannot issue this on ONE LINE.
17:40.22[TK]D-FenderBSD_Tech : Break them up.
17:40.30*** join/#asterisk awk (n=radio@vc-196-207-45-253.3g.vodacom.co.za)
17:40.53[TK]D-FenderBSD_Tech : and rather than the huge disallow, set it to "all" and before your "allow"
17:41.29BSD_Tech[laptop]in 1.4 you can
17:41.44BSD_Tech[laptop]I found the issue
17:41.53geoaxiswhat is this callbook for btw?
17:41.56BSD_Tech[laptop]and its a asterisk + gui issue
17:41.58Qwell[]geoaxis: callbook?
17:42.09geoaxiscahn_skinny , as you called it
17:42.17BSD_Tech[laptop]if you click disallow all in the gui it locks out all the codecs
17:42.21BSD_Tech[laptop]crap
17:42.22Qwell[]skinny is a Cisco proprietary voip codec
17:42.26BSD_Tech[laptop]this is a issue
17:42.31Qwell[]erm, not codec
17:42.33Qwell[]protocol
17:42.59geoaxisit comes as port 2000 call book (probably by wrong name as 2000 tcp might be for some hitng esle
17:43.18Qwell[]geoaxis: yeah, there are a few things that claim they own port 2000, but I'm fairly certain Cisco "owns" it
17:43.31Qwell[]who is it that assigns ports, IANA?
17:43.46Trevor_bsieve           2000/tcp                        # Sieve Mail Filter Daemon
17:43.46Trevor_bsieve           2000/udp                        # Sieve Mail Filter Daemon
17:43.51Qwell[]yeah, see?
17:43.55Qwell[]there is at least one other thing too
17:44.02*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)
17:44.35Trevor_buse /etc/services to pick unused ports (or at least a good guess at them)
17:45.26geoaxishmm, or you stick your port in and hope your product is so popular that it becomes assocaited with it
17:45.34geoaxis(thats what
17:45.41geoaxisCisco probably did)
17:45.42Qwell[]or you submit a formal request to IANA and get an official assignment
17:45.57Qwell[]or whoever owns port assignments
17:45.58Trevor_bhehehe, well sticking it in will only get you your own recognition ;), have to submit up the chain and start using the unused port....
17:46.10Trevor_byup ;)
17:46.16Qwell[]actually
17:46.28Trevor_bless owned and more like RFC style
17:46.29Qwell[]I think IANA only assigns in the 1-1024 range
17:46.33Trevor_byeah
17:46.48Qwell[]so yeah, pick a port and hope your product gets more popular :P
17:48.19geoaxisyeah, rest is first come first serve for procssess
17:48.20[TK]D-FenderQwell[]: Even better.... equally popular and by the same kinds of users so they fight over yourserver ;)
17:48.20Qwell[]or be like bittorrent and use completely arbitrary ports
17:48.20Trevor_byeah its its not quite owns, but close.... "Well-known port numbers        specified by the IANA are normally located  in  this  root        only space. "
17:48.33geoaxisQwell, well they do use a tracker port dont they
17:48.34geoaxisbittorent clients
17:48.34Qwell[]geoaxis: like 80
17:48.34Qwell[]but yeah
17:48.44Qwell[]reuse an existing protocol for that initial communication ;)
17:51.01*** join/#asterisk anzen (n=adam@207.200.28.175)
17:51.09anzenHey , quick question
17:51.18*** join/#asterisk oej (n=olle@AMontsouris-152-1-5-176.w82-123.abo.wanadoo.fr)
17:51.20anzenI'm setting up a Asterisk based test box
17:51.50anzenand I'm looking for a cheap VOIP provider for just 1 month
17:52.00Strom_Mare you in the US?
17:52.03anzenyes
17:52.07Strom_Mtry teliax.com
17:52.13anzenTELIAX! thank you
17:52.28anzenI was trying to remember what it was (I used to do phone systems, been out for aobut a year)
17:52.32anzenI thought Teliax went under?
17:52.58Strom_Mnews to me
17:53.12codazodaI use via:talk.  Pricing looks similar.
17:53.22anzencool
17:54.33codazodavia:talk gives you a seperate trunk for every line of service, however, so I'm thinking of a switch to someone who gives me a single trunk that allows multiple calls.
17:55.11codazodaOr, I might just get over my lazyness and write a macro to pick an available trunk.
17:55.49codazodaSo lazy, in fact, I didn't bother to fix my spelling error.
17:56.33Hmmhesayseh, that would be easy enough
17:56.57*** join/#asterisk doolittlework (i=doolittl@196.211.34.2)
17:57.35codazodaI was lazy at first, however, and used a gui.  Now I have to re-do everything manually (which I should have done in the first place) just to write my macro.  :-)
17:58.16Hmmhesayswhy?
17:58.58codazodaBecause I bought a crappy non-free GUI.  Can't edit the conf files.
17:59.36Hmmhesaysyou can't edit the config files?
17:59.55*** join/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net)
18:00.03*** part/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net)
18:00.06codazodaNo, because they have it setup not to use them.  Their code is all encrypted, so I can't read it to figure out what they are doing.  MySQL tables are involved, but...
18:00.53waKKuhey dudes.. very cool sound techique: http://www.gmoura.com/blog/2007/05/vai-um-corte-de-cabelo.html - in "english": increase volume, put ur headphones, clik play and close your eyes... 3d sound
18:01.19codazodaMaybe they are using some standard tool that lets you store your config in mysql, and I could probably go find out what that is, but I should just scratch the crappy gui and do it right.
18:02.57*** join/#asterisk Assid (n=assid@59.165.14.35)
18:03.03awkdoes anyone here in this channel carry out the updates for bristuff ?
18:03.24*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
18:03.25awkjust wondering why it took this long to update to 1.2.18 and had all its users velnerable to BoF and DoS attacks?
18:04.29awki had to manually update my zaptel and asterisk and patch it myself
18:06.47*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
18:07.44[TK]D-Fenderawk: Yes its as bad as you think, and they're all out to get you.  No please remain seated till the men witht he white-coats come to pick you up....
18:07.46justdaveJun  6 11:04:53 NOTICE[18323]: chan_iax2.c:3171 iax2_read: I should never be called!
18:08.03justdavethat makes me feel really warm and fuzzy :)
18:08.33awk[TK]D-Fender: the wouldn't get me, i have enough sense to keep my system updated!
18:08.41*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-118-241.ph.ph.cox.net)
18:08.46[TK]D-Fenderawk: so stop whining! ;)
18:09.04awki just have a feeling that asterisk 1.2.18 has a few bugs that the developers don't know about, and their is an 0day floating around!
18:09.12awkbut hey, thats just my thought!
18:09.16justdavealways happens right after a user coming in on IAX joins a meetme conference
18:09.51adorahSET(TIMEOUT(absolute)=value) can I use that command in the general section of the dial plan or should I add it to each extension script?
18:10.02krdian_<PROTECTED>
18:11.20tzafrir_laptopIt's a channel setting
18:12.00*** join/#asterisk catpants (n=catling@12-214-191-244.client.mchsi.com)
18:12.09*** join/#asterisk toerkeium (i=oo@201.216.206.221)
18:16.55*** join/#asterisk keulin (n=cray@AMontpellier-152-1-10-145.w81-251.abo.wanadoo.fr)
18:19.41IOscanneranyone use chanspy?
18:20.23IOscannerI can't find any docs on it, since voip-info.org is not working.
18:20.28*** part/#asterisk codazoda (n=Joel@ip69-223.konnections.com)
18:20.44[TK]D-FenderIOscanner: "show application chanspy"
18:21.59sci_05wow what happend to voip-info this time?
18:24.53anonymouz666404 not found from voip-info
18:25.59BSD_Tech[laptop]res_config_mysql.c:624: error: unknown field `store_func' specified in initializer
18:25.59BSD_Tech[laptop]res_config_mysql.c:624: warning: initialization from incompatible pointer type
18:25.59BSD_Tech[laptop]res_config_mysql.c:625: error: unknown field `destroy_func' specified in initializer
18:25.59BSD_Tech[laptop]res_config_mysql.c:625: warning: initialization from incompatible pointer type
18:25.59BSD_Tech[laptop]make[1]: *** [res_config_mysql.o] Error 1
18:26.03BSD_Tech[laptop]sorry
18:26.04*** join/#asterisk CVirus (n=GoD@196.202.18.145)
18:26.09BSD_Tech[laptop]damn wheel
18:26.24BSD_Tech[laptop]asterisk-addons is not compiling
18:27.28awkBSD_Tech you using gnumake ?
18:27.40BSD_Tech[laptop]no this is a linux box
18:27.52BSD_Tech[laptop]this is not on bsd
18:28.19BSD_Tech[laptop]I just type make
18:28.22BSD_Tech[laptop]lol
18:28.53*** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi)
18:28.56awkhaha
18:28.57awkfreak !
18:29.08BSD_Tech[laptop]its a test box
18:29.16BSD_Tech[laptop]its slackware 11
18:29.38awkis it a 1.4  or 1.2 addons?
18:29.44BSD_Tech[laptop]I am not a freak
18:29.48BSD_Tech[laptop]1.4
18:30.20BSD_Tech[laptop]mysql 5
18:30.47awkits maybe missing some include
18:31.04*** join/#asterisk galeras (n=root@201.244.240.115)
18:31.22BSD_Tech[laptop]I installed everything asterisk called for
18:31.29BSD_Tech[laptop]grrr
18:31.57BSD_Tech[laptop]I wish it would tell you missing headers
18:32.05awktry doing a make distclean
18:32.13BSD_Tech[laptop]and more details about the make erros
18:32.49awkI got a feeling you missing mysql dev pack
18:33.00BSD_Tech[laptop]ok and rerun make and then rerun make again
18:33.11BSD_Tech[laptop]ok
18:33.49awkdoes slackware have a "ports tree" these days? :D
18:34.05awkkidding i know its finally got that far :D
18:34.14BSD_Tech[laptop]I believe it does but I do everything by hand
18:34.30awkfor debian the package is libmysqlclient15-dev
18:34.34[TK]D-FenderBSD_Tech : ./configure ; make clean ; make  ; make install
18:35.07awk[TK]D-Fender: it wont compile without mysql libries
18:35.09awklibraries
18:35.13[TK]D-FenderBSD_Tech : Slackware does packaging just fine
18:35.16BSD_Tech[laptop]make in 1.4 in addons runs the .configure script for you
18:35.46BSD_Tech[laptop]I have to see if the dev paks are installed brb
18:35.50[TK]D-FenderI miss Slackware now and again...
18:36.13tzangerI wub slackware
18:36.16*** join/#asterisk jcmoore (n=Curt@unaffiliated/tgrman)
18:36.17tzangerbut kubuntu pulls at me
18:36.25awki dont :D its been years since ive used it, think 1997 or somewhere around there, when grsec first released some of their kernel patches :D
18:36.41[TK]D-Fendertzanger: I hear many prefer to install KDE on regular Ubuntu
18:36.52awkI heard people who use ubuntu don't know how to configure debian :D
18:36.56tzangerisn't kubuntu == ubuntu+kde installed?
18:37.03awkyes
18:37.04Qwell[]tzanger: pretty much
18:37.06tzangerawk: I don't know how to configure debian :-)
18:37.10tzangerpretty much, or exactly?
18:37.11[TK]D-Fendertzanger: A few other differences, but I couldn't go into detail.
18:37.15Qwell[]pretty much exactly
18:37.18*** join/#asterisk ccesario (n=ccesario@200-158-227-58.dsl.telesp.net.br)
18:37.18tzangerhahaha
18:37.20awktzafrir: with some tweaks
18:37.32Qwell[]I'm sure there are some other small things, yeah
18:37.37Qwell[]no idea what though
18:37.38awkextra graphics drivers, etc
18:37.41awkalot of nvidia stuff
18:37.43Qwell[]really?
18:37.48awkso i've heard
18:37.49BSD_Tech[laptop]looks like it installs mysql but not mysqlclient
18:37.54BSD_Tech[laptop]thats stupid
18:37.56*** join/#asterisk [hC] (n=hardcore@190.10.12.97)
18:38.03awkBSD_Tech and mysql libries, its devel pack
18:38.06awklibraries!!!!!!!!!
18:38.20awkyou doing development here so you need the dev pack!
18:38.25BSD_Tech[laptop]yeah I have to get and build them
18:38.26awk*wink*
18:38.27BSD_Tech[laptop]brb
18:38.35*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
18:39.02awkBSD_Tech: if I was you and using slackware i'd pretty much use all my base stuff through a package system
18:39.29awkas for you to monitor the progress and changelog, security mailing lists, etc just to know what package is vulnerable, etc
18:39.42awki believe its similar to that of gentoo but i could be way off :)
18:39.55tzangermeh
18:39.58awknot sure if its headers of files that install from sourced packages with patches, etc or its pre-compiled packages
18:40.00tzangermy slack systems are pretty good
18:40.04tzangerno rootings in 10 years
18:41.21awktzafrir: i know of a 2.6 0day that would root most kernels :D even with PaX security
18:41.21awkBoF isn't the only way to exploit something
18:41.21awkluckly its only local
18:41.21tzanger?
18:41.21tzangerremote root?
18:41.22coppicetzanger: don't you think that after 10 years its time to put down roots?
18:41.22awkno, local
18:41.22tzangerahh
18:41.22tzangersee that's teh main difference, I have no local users :-)
18:41.30tzangeror at least no local users I can't beat the shit out of if they pull a trick like that :-)
18:41.39awkheh, make sure your system accounts are locked or you might just have somebody log on locally :D
18:41.48tzangerphysical access to users is just as important to an admin as physical access to servers is to exploiters :-)
18:42.01tzangerawk: default slack install has no users able to log in except root
18:42.08tzangerand I turn that off and use DSA keys
18:43.29*** join/#asterisk tecnico (n=tecnico@24.96.146.69)
18:45.03BSD_Tech[laptop]bbiab getting and installing pkgs
18:45.15awktzafrir: but i can asure you, your web server is vulnerable to a http trace attack!
18:45.27tzangerI haven't heard of that one
18:45.43tzangerahh XSS
18:46.18tzangerit looks like a RewriteEngine rule can tackle that one
18:46.34jkiffSo is voip-info hosed, or what?
18:46.43tzangerthat's the general consensus
18:46.53awkwant to test? telnet your box port 80 then type, TRACE / HTTP/1.0 new line type A: b    new line C: d   new line  Host: vulnerable
18:46.53Nuggettelnet is eeeeeeevil!
18:47.04awkthen hit enter twice and tell me if that echo's it back to you
18:47.24awktzafrir: yes, it can! but nobody has it set by default
18:47.35tzangeryes it does
18:48.00awkit echo's it back to you? hehe, thought so
18:48.00tzangerthat can lead to information leaking about cookies and such
18:48.00tzangercorrect?
18:48.02*** join/#asterisk mrdigital (n=mrdigita@207-172-228-21.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com)
18:48.13tzangeri.e. ability to steal forum cookies, etc.?
18:48.17mrdigitalhey guys anyone know how to email a pdf to asterisk and have it fax it out
18:48.24Qwell[]awk: <p>Attempt to use unimplemented / unsupported method</p>
18:48.25Qwell[];)
18:48.36tzangerQwell[]: that's what the asterisk miniserver says
18:48.37Qwell[]asterisk webserver is secure!
18:48.39Qwell[]:p
18:48.58awkA web page can send Trace requests to a Web server that has ths Trace method. Its possible for the active HTML content to make external requests to arb* Web servers beyond the hosting Web server..... the Web server then echoes back the client request unfiltered, the response also includes cookie-based or Web-based (if logged on) authentication credentials that the browser automatically sent to the specified Web ap on the web server
18:49.01tzangerQwell[]: did you see my post in -dev?  I'm -->||<-- that close to having rudimentary CGI support (POST too) in Asterisk
18:49.05Qwell[]yeah
18:49.17awkerr this & ap/app
18:49.55*** join/#asterisk seele_ (n=seele@200.30.85.186)
18:50.04tzangerawk: interesting
18:50.07seele_please help, when I apply  the changes from freepbx the asterisk hangs, this is my full log http://www.pastebin.ca/545192
18:50.17seele_how can I solve this??
18:50.22tzangernot something I'm immediately vulnerable to (basic web serving only) but still something that needs to be patched
18:50.23*** join/#asterisk FF|Ell (n=elliot@host81-155-72-43.range81-155.btcentralplus.com)
18:50.42BSD_Tech[laptop]ok I am not finding the mysql-dev pkg
18:50.51BSD_Tech[laptop]mysql client is installed
18:51.00awkhttp://paste.linux-vserver.org/2260
18:51.01BSD_Tech[laptop]so i just need the dev pkg
18:51.05awktzafrir: there is a fix for you
18:51.10awkadd that to your apache conf
18:51.20tzafrirawk, actually it's for tzanger
18:51.36awkhehe, you guys confuze me with the nicks
18:51.40awknever knew there where 2 of you!
18:51.57mrdigitalhey awk?
18:52.29[TK]D-Fenderseele_: FreePBX has blown up your dialplan, etc.  You what's coming...
18:52.31[TK]D-Fender~freepbx
18:52.43jbothmm... freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
18:52.44[TK]D-Fenderknow*
18:53.31mrdigitalhey [TK]D-Fender: i moved to asterisk,  whats the best route to email a message to asterisk and have it Fax it out
18:54.47tzafrirBSD_Tech, what distro?
18:54.48[TK]D-Fendermrdigital: don't use * for this.  Hylafax + modem on its own line
18:55.17LeddyHMmmmm hylafax
18:55.39mrdigitalcan i use hylafax on the same box?
18:55.46[TK]D-Fendermrdigital: Sure
18:55.47LeddyHMsame box as?
18:56.10awkBSD_Tech: the best way to get that info is to find out how to use the ports package for slack and a make search name= or something
18:56.10mrdigitalsame box as asterisk+hylafa
18:56.15*** join/#asterisk Waverly360 (n=Waverly3@209.12.249.243)
18:56.18[TK]D-Fenderawk: "ports" does not exist in linux except for Gentoo.
18:56.23LeddyHMfo sho, as previously mentioned
18:56.31[TK]D-FenderBSD_Tech : www.linuxpackages.net
18:57.05LeddyHMwe use hylafax on gentoo, soon to be moving to centos
18:57.30*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:57.56LeddyHMhttp://www.hylafax.org/content/Email_to_Fax_Gateway
18:59.15awk[TK]D-Fender: yes it does, BSD uses ports and gentoo got its name from their and changed it to portage
18:59.36[TK]D-Fenderawk: .... read what I said again.
19:00.07awkok i should have said learn how to use your package manager!
19:00.10awkhappy ? :D
19:00.24Strom_Mi propose forking portage and naming it portavia
19:00.25[TK]D-Fenderawk: No... but at least vindicated ;)
19:01.16*** join/#asterisk jicallero (n=jcallero@rev-200-40-119-222.netgate.com.uy)
19:02.20jicalleroHI! some one is using * with High Availability ? or some know a good documentation/example ? THX
19:03.48*** join/#asterisk jeffik (n=Valued@206-248-161-46.dsl.teksavvy.com)
19:04.43*** join/#asterisk FF|Ell (n=elliot@host81-155-72-43.range81-155.btcentralplus.com)
19:04.55FF|Ellcan someone help with from user from domain and contact?
19:04.57*** join/#asterisk jebba (n=jebba@220-179-89-200.fibertel.com.ar)
19:05.11awk[TK]D-Fender, sorry, *takes notes*
19:06.06FF|Ellim abit stuck getting it to work
19:07.58*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
19:08.05tzafrir_laptopFF|Ell, noone can help you if you don't provide information...
19:08.36*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
19:08.38tzafrir_laptoppatebin the relevant details
19:09.53*** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net)
19:10.59FF|Ellsorry. I have many numbers with one provider. i want to have multiple trunk connections for this provider using the contact and from user and from domain. i can get the contact to register with the provider, so when the call comes through its been sent to the correct trunk. though it fails to conect as the from user and from domain are incorrect but im finding it difficult to see what the provider is sending so i can set th
19:11.06BZBWanyone know that * will only accept the preferred codec (order) in its sip.conf file? If not, it will drop the call:(
19:11.07Aces1Upin sip.conf, do you typically put the externip in the [general] section?
19:11.31Aces1Upand can i use my dynamic dns entry as the externip?
19:11.49Aces1Upsuch as:   externip = kingdom.dnsdojo.com
19:12.05BZBWit's in 1.4.4, I thought this has been fixed:(
19:12.15*** join/#asterisk tectoris (n=bass@71-34-222-55.spkn.qwest.net)
19:12.58Mercestes<PROTECTED>
19:13.21*** join/#asterisk toerkeium (i=oo@201.216.206.221)
19:14.08BZBWhere is how it is:   SIP(INVITE from *) with G729,G711u,G726 --> my SIP client
19:14.14tectorisis AsteriskNow OT in here?
19:14.50BZBWThen:   SIP(200 OK from my sip client) with G711u --> *
19:14.56BZBW* will drop the call:(
19:15.42Hmmhesaysgetting office workers to plug in a pri, bad
19:16.55*** join/#asterisk znoG (n=gs@235-180-235-201.fibertel.com.ar)
19:17.46tectorisIs AsteriskNow considered OFF TOPIC on the Asterisk Channel?
19:18.07Hmmhesaysyou want a kick in the sack?
19:18.38[TK]D-Fendertectoris: Typically yes
19:18.52Hmmhesaysoh that was a hearty laugh
19:18.58MercestesWe should rename our channel to #asteriskThen
19:19.03[TK]D-FenderHmmhesays: You just want him IN THE SACK
19:19.25Hmmhesayswell the girlfriend hasn't been putting out lately so....
19:19.26HmmhesaysNO
19:19.28[TK]D-FenderMercestes: and #asterisk-dev to #asterisk-soon
19:19.29Aces1Uptkd if i am using a did provider, and they want a sip URI address why can't i just put the public IP of my asaterisk box, it doesn't want to take that as an address.
19:19.43Mercestes[TK]D-Fender, Yes!
19:19.45[TK]D-FenderMercestes: or #asterisk-whenitsreadygoddamnit
19:19.50Aces1Upi know i need to use sip:user:password@host:port;uri-parameters?headers
19:19.50Mercesteslol
19:20.11Hmmhesayssip:username@myhost
19:20.24Aces1Upwhere do i set the username?
19:20.35Aces1Upfor incoming calls? in the sip.conf for that channel?
19:20.50Hmmhesayswhat a fantastically vague question
19:20.53Aces1Uplol
19:20.56Aces1Upsorry.
19:21.12tectorisFender - It seems pretty dead on the AsteriskNow Channel. Is any particular time better than others to catch the *Now folks in?
19:21.13[TK]D-FenderAces1Up: [user] type=user       secret=pass
19:21.29jicalleroMercestes: THX
19:21.33[TK]D-Fendertectoris: SOON!
19:21.42HmmhesaysNOW!
19:21.53MercestesThere is no time like the present.
19:22.02Mercestes...There is no present like time.
19:22.21*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)
19:22.39Aces1Upok if i have a did provider, and they are sending me calls and they want a uri in the forwarding address.  for sip.conf i would do a type=user, username=xxxx, secret=xxxx  this would be enough?
19:22.56tectorisThen I shall presently return to said channel. Thanks All.
19:23.04*** part/#asterisk tectoris (n=bass@71-34-222-55.spkn.qwest.net)
19:23.08MercestesAces1Up, Host=IP Address and type=peer I believe.
19:25.56*** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose)
19:26.57[TK]D-FenderMercestes: So when will then be now sir?  SOON!
19:26.59Waverly360Hey guys, any idea why I wouldn't be able to pull the caller id name from asterisk version 1.2.12 using the CALLERID(all) or CALLERID(name) variable?  The number is set, but name isn't.
19:27.08[TK]D-FenderMercestes: Comb the desert!
19:27.24Mercestes:(
19:27.51[TK]D-FenderWaverly360: My guess is... the name isn't SET
19:28.01Aces1Upok just to make sure all my ducks are in a row....  at the did provider i use the following uri:  sip:richard:abel9939@68.227.41.148   then in sip.conf i use type=friend, username=richard, secret=abel9939  is this correct?
19:28.33[TK]D-FenderAces1Up: [richard] and remove the "username"
19:28.58*** join/#asterisk ljd (n=ljd@nelug/coreteam/luisjose)
19:29.00[TK]D-FenderAces1Up: and add "host=dynamic"
19:30.15Aces1Upi have specific ip's the did provider is giving me for the hosts.
19:30.21Waverly360[TK]D-Fender: Well, when I call a phone, the callerid name shows up on the polycom...but if no one answers, it dials an external phone number.  If that person doesn't answer, then it sends it to local voicemail in the system, and if that person has voicemail sent to email, it's supposed to record their callerid name and number in the subject of the email.
19:30.52Waverly360[TK]D-Fender: Well, the name isn't making it across.  All of this is happening within an agi script.
19:31.37[TK]D-FenderWaverly360: you'd have to show your script and multiple debug points to test the value.
19:31.39Waverly360[TK]D-Fender:  So I know the name is making it into the pbx somewhere...but I can't seem to pull the callerid name from asterisk using the CALLERID(name) varuable
19:31.48[TK]D-FenderAces1Up: Your call
19:32.04[TK]D-FenderWaverly360: Show us how you're calling it (CLI out.  COMPLETE)
19:32.27Waverly360[TK]D-Fender: one sec.
19:33.04FF|Ellfor the from domain and from user which part of this would i use Goto("SIP/voip.co.uk-08252c78", "default|1001|1")
19:35.19Aces1Uptkd could you please check out http://www.pastebin.ca/545322   and tell me what ya think?
19:38.33*** join/#asterisk gardo (n=gardo@121.97.193.142)
19:39.46*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
19:41.11Waverly360Where do you guys prefer pastes?
19:41.11*** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net)
19:42.10[TK]D-Fenderpastebin.ca
19:42.32bcnldoes anyone know if there is a version of Answer Confirmation for SIP/IAX channels that works like the dial feater 'c' for ZAP channels?
19:42.33Waverly360http://www.pastebin.ca/545362
19:42.46[TK]D-FenderAces1Up: NO.  I just told you [richard]
19:43.28[TK]D-FenderAces1Up: [somthingthatlookslikeaniphereisjustWRONGok?]
19:44.24[TK]D-FenderWaverly360: SET VARIABLE CALLERID(all) " "" <6152438798>  < 6152438798 >"
19:44.34[TK]D-FenderWaverly360: I'm not sure thats valid.
19:44.55Waverly360[TK]D-Fender: which part?
19:45.00[TK]D-FenderWaverly360: And your quotes look funny
19:45.10Waverly360[TK]D-Fender: Oh..hmm.
19:45.18[TK]D-FenderWaverly360: that isn't a variable.  its a function and I'm not sure how AGI will parse it.
19:46.07Waverly360[TK]D-Fender: well...it wouldn't be an issue if asterisk would just send the name to agi in the first place.  See where asterisk sends agi certain variables at the beginning?
19:46.17Waverly360[TK]D-Fender: callerid name is always unknown.
19:46.35Waverly360[TK]D-Fender: but when it dials SIP/243|15|rtT the name does show up.
19:46.37*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
19:46.38[TK]D-FenderWaverly360: I'm guessing its coming in from source as such
19:47.09Waverly360[TK]D-Fender: Do you know of any other way I can gain access to the callerid name in asterisk from agi?
19:47.15[TK]D-FenderWaverly360: And you're not helping by NoOping it the second you're able to.
19:48.04Waverly360[TK]D-Fender: To be honest I'm not sure what that does..I didn't write this.
19:48.37[TK]D-FenderWaverly360: I'm talking in the dialplan before you call the AGI even.  And if you don't have a clue.... You'll have to hire me to develop one for you ;)
19:49.08Waverly360[TK]D-Fender: It's been one of those "figure it out as you need to" situations.
19:49.29Waverly360[TK]D-Fender: I'm not asking you to develop anything for me.  I just need to learn to use agi better.
19:50.38*** join/#asterisk asdx (n=diego@adsl-148-227.click.com.py)
19:51.01Waverly360[TK]D-Fender: So basically you're saying that there's no way I can get access to the callerid name then?
19:51.04[TK]D-FenderWaverly360: Goooood..... the first step on the road to rehab!
19:51.49[TK]D-FenderWaverly360: I didn't say that... I don't know personally, but I DO know that 1.2 changed a number of thigs for people on this and it's come up numerous times here before.
19:52.12Waverly360[TK]D-Fender: I was afraid of that.  Upgrading asterisk would be a pain in the ass right now.
19:54.34FF|Ellfor the from domain and from user which part of this would i use Goto("SIP/voip.co.uk-08252c78", "default|1001|1")
19:55.30*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-91-140.red.bezeqint.net)
19:55.51[TK]D-Fenderffthat does not add up at all...
19:56.56FF|Elldo you know the from user from domain and contact bit im talking about for the providers?
19:56.59*** join/#asterisk Greek-Boy (n=g@196.45.144.42)
19:58.02[TK]D-FenderFF|Ell: "show function SIP_HEADER"
19:58.32[TK]D-FenderFF|Ell: That goto doesn't mean ANYTHING conscerning determining the nature of the call you have received.
19:58.38FF|Elland do i use this when i have an incoming call to find out what i need to use?
19:59.05[TK]D-FenderFF|Ell: I gave you the function.  Go read.  Go TEST.
19:59.39FF|Ellok, will give it a look
20:00.03Waverly360[TK]D-Fender: Thanks for the help. :)
20:03.47*** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca)
20:06.44*** join/#asterisk doolittlework (i=doolittl@196.211.34.2)
20:07.20doolittleworkhi there people
20:07.27J4k3!madusb
20:07.46J4k3somebody acted like they had something coming for USB the other day.
20:08.05*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
20:08.42J4k3ever had a PTP T1 randomly lag/loss every 58-64 seconds (it was every 63.8 seconds last night, this afternoon is 58.9 seconds)
20:08.46J4k3?
20:08.55J4k3and generate no CRC errors? :)
20:09.22FF|Ell[TK]D-Fender i have looked at the function and tbh ive not got a clue what im doing.
20:09.56*** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com)
20:10.18*** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com)
20:10.24FarrisGIs anyone else having bad problems with repeated DTMF digits on inbound sip trunks? This has been an issue for a long time, and I've reached the point where I'm willing to pay someone to fix it, but I'm not sure who might be able to help me.
20:10.37[TK]D-FenderFF|Ell: go turn on SIP debug and look at the headers that come in.  Then try to get their values in the dialplan with that function an NoOp them.
20:11.13[TK]D-FenderFarrisG: Jitter with inband would do it.  If no sounds like a provider issue
20:11.30FarrisG[TK]D-Fender: Jitter?
20:11.55[TK]D-FenderFarrisG: or packet-loss
20:13.36*** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com)
20:16.15FarrisG[TK]D-Fender: I'm pretty sure we've eliminated the possibility of it being packet loss. There are some outside lines that will always produce this problem, even in an isolated environment. I'm just not sure how to go about correcting it
20:16.35FarrisGLooks very similar to several unresolved bugs in 1.2.x: http://bugs.digium.com/view.php?id=9382
20:22.43*** join/#asterisk drrt (n=junior@ppp-static2-140.tis-dialog.ru)
20:22.49Greek-Boyanyone know some good software for asterisk for voip route gateway? I want to work out ASR, ACD and PDD.
20:23.54doolittleworkwhy not use iax Greek_Boy?
20:24.14Greek-Boyi will use IAX
20:24.26Greek-Boybut i am talking about the management aspect of running a route
20:24.51*** join/#asterisk CVirus (n=GoD@196.218.136.114)
20:27.59Kattyanthm: )=
20:28.03Kattyanthm: please kill the sales reps.
20:28.13Kattyanthm: they did it AGAIN
20:28.19anthmdid what ?
20:28.28Kattyscheduled me for a network install the friday before i leave
20:28.34Kattywhich i know i won't be able to get done friday
20:28.40*** join/#asterisk sunsmasher (n=Beamer@66.251.47.154)
20:29.00sunsmasherwhats the difference between 1.2.18 and 1.4.4?
20:30.16MercestesKatty, oo!  Hire me!  hire me!
20:30.47Kattyanthm: and that means, heh.
20:30.52Kattyanthm: looks like i won't be going to chicago
20:32.19anthm=(
20:32.20Mercestes:(
20:32.27anthmtell them to make it sooner!
20:32.27MercestesThis is UNACCETPABLE.
20:32.34Mercestesunacceptable too
20:32.55*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
20:32.57Kattyanthm: can't :/
20:32.59FF|Ell[TK]D-Fender i have been playing around with debug on. When i set a contact and not a from user and from domain, asterisk registers with the provider with the contact but when i ring the number i get cannot auth with proxy. its been registering using Re-registration for 785213@proxy.voip.co.uk
20:33.04Kattyanthm: apparently they've been planing this date for a month.
20:33.13Kattyanthm: whihc no one bothered to tell me about, heh
20:33.37Kattyanthm: until after the office manager, sends out mass email about how angie's going be gone blahblahblah, and the sales reps call me in a panic :P
20:34.17Mercestes:(
20:34.24Kattyanthm: so yes :<
20:34.31sunsmasherwhats the difference between 1.2.18 and 1.4.4?  Is it big.. why do they maintain two completely different versions with similar numbers?
20:34.35Kattyanthm: i guess i won't need an escort to the hotel.
20:35.00anthmdoh
20:35.04Kattyanthm: and i guess you better tell leah not to bill us too
20:35.11*** join/#asterisk thevoke (i=michiel@missioncontrol.eng.gbxs.net)
20:35.29thevokeHey Folks, whats the replacement for SetVar in asterisk 1.4 ?
20:35.44Kattyso much for a vacation :<
20:36.11anonymouz666Set(duh=blah) ?
20:36.55Kattyset(anonymouz666=hugged)
20:37.00anonymouz666:)
20:37.48anonymouz666lol
20:38.37FF|Ell:(
20:38.46J4k3(with 27 cameras... orgy!)
20:39.20Kattymeep.
20:40.11*** join/#asterisk skyhawker (n=info@a62-216-22-13.adsl.cistron.nl)
20:44.27KattyMercestes: so, in theory...
20:44.33KattyMercestes: if i have an incoming call...
20:44.38KattyMercestes: and it times out at say, 30
20:45.07KattyMercestes: can i somehow play an audio file to that user, and at the same time, do some sort of an auto-answer and play a lil message that there's a call waiting to come in...and then ring a blast group?
20:45.12KattyMercestes: or is that entirely too complicated
20:48.13hijackedI'm creating an IVR menu on my asterisk server, it's a simple, "press 1 for support, or dial by extension..." setup, but I want to make it so that if there's no answer at the extension they dialed, asterisk drops to another menu giving the caller the option to go to voicemail, the support queue, or dial someone else's extension.
20:48.45hijackedI'm thinking I'll use a macro so I don't have to re-write everything for all th extensions, but I need some way to pass the original extension to this second menu I'm creating.
20:48.53Greek-Boywhat is the best web management package for asterisk?
20:49.24hijackedI don't suppose I could do it like, "2,1,macro(menu2|$variable)" ?
20:57.15Nate9939what are the possible causes if when i look at me inbound did records from my DID provider and it says all calls are congested or busy, does that mean it isn't hitting my asterisk box, or is it usually a problem with my provider?
20:57.23*** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr)
20:57.49yannj_frhello
20:58.07MihiNomenEstNate9939: are you looking at the loogs on your asterisk box?
20:58.18yannj_frdid someone managed to use the sip notify with a grandstream phone?
20:58.22Nate9939no logs on my provider.
20:58.34*** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net)
20:58.40MihiNomenEstI don't suppose you've bothered to run a sip debug on your asterisk box?
20:58.55J4k3Nate9939: possibly, or it could be that your * doesn't know where to send those incoming calls
20:59.05Nate9939my DID is getting long beeps when i call it, so just wondering where to start troubleshooting.
20:59.11J4k3or does the call ever make it to your *?
20:59.11MihiNomenEstyannj_fr: yes, I have.  I used the how-to on grandstream's site.
20:59.25yannj_frI didnt saw it?
20:59.25Nate9939j4k3 when calling the did shouldn't i see something pop as incoming to my box on the cli?
20:59.32yannj_frwas it difficult?
20:59.40J4k3Nate9939: with the debugging turned up, yes.
20:59.59Nate9939j4k3 to what level?
21:00.43murdmathNate9939: Just bump it up to 10
21:00.52*** join/#asterisk notoriousrab1982 (n=root@76.195.14.206)
21:00.56J4k3yeah, it shouldn't need to be real high to see events
21:01.42notoriousrab1982anyone know why you would get [Jun  6 13:55:13] NOTICE[8575]: chan_iax2.c:6777 socket_process: Rejected connect attempt from 192.168.2.54, who was trying to reach 'TBD@company2'
21:01.56notoriousrab1982from a simple switch statement, where does the tbd come from?
21:02.43yannj_frMihiNomenEst : are you using sip notify to reboot the phone or do approvisioning?
21:02.44*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
21:04.10MihiNomenEstyannj_fr: no.
21:04.36yannj_frhum, i guess you are using subscriptions
21:04.37MihiNomenEstthis is like BLF stuff.
21:05.18yannj_frok, I need to use notify message in order to reboot my phone
21:05.34yannj_frdoes anyone managed to do it?
21:05.53MihiNomenEstI don't suppose you can do that fromt he webterface.
21:06.46yannj_frI like to do it automaticaly
21:06.46*** join/#asterisk zotz (n=zotz@24.244.163.157)
21:07.44MihiNomenEstwell, you can always set the "phonebook download interval"
21:08.35yannj_frIn fact I would need to triger the provisionning in order to be able to make a phone register to another account on demand
21:08.51*** join/#asterisk ruied_ (n=ruied@bl7-210-89.dsl.telepac.pt)
21:10.18yannj_frlike a user call an extension, type login/password, the phone reboot and log on the account
21:12.22MihiNomenEstI didn't know you could provision a grandstream that way.
21:13.06MihiNomenEsthttp://www.i2k.com/~nwells/extensions.conf - that's the relevant bit of my config.  as you can see in macro-menu2, I can't figure out how to get the extension number that was dialed in macro-agent.
21:13.51*** join/#asterisk dotSlashW (n=HTP@200.80.197.5)
21:15.16yannj_frin fact my idea is : grandstream can look for config file with http, it ask for prefix_MAC.conf, with url rewriting you can redirect to php script with Mac as argument, I have the configuration wanted by each user in a database and I return the one for the user logged on the phone, but i need to trier reboot or trigger approvisionning
21:17.24dotSlashWI've set up an extension, when I dial *cell calls are forwarded to my celphone
21:17.31dotSlashWthen, if I dial *home the forwarding stops
21:17.38dotSlashW*2355, and * 4663, respectively
21:17.47dotSlashWfor example
21:17.49dotSlashWexten => *4663,1,Answer()
21:17.56dotSlashWexten => *4663,3,System(/usr/bin/mysql -u root -pSOMEPASS -e "UPDATE incoming SET destination = \'ext-group\,600\,1\' WHERE channel = 4" asterisk)
21:18.04dotSlashWexten => *4663,4,Hangup()
21:18.12dotSlashWdo I need to reload any configuration so asterisk knows about this update ?
21:18.26MihiNomenEstuh, yes.
21:18.32*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
21:18.38dotSlashWwhat I'm doing is changing the destination of an incoming line
21:18.47dotSlashWwhat do I need to reload ?
21:19.28[TK]D-FenderdotSlashW, Depends on what kind of "line"
21:21.37notoriousrab1982anyone here know anything about switch statements
21:22.33*** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)
21:23.22*** join/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-160-113.ny325.east.verizon.net)
21:23.35dotSlashW[TK]D-Fender: two lines, a zaptel one and a SIP trunk
21:24.20[TK]D-FenderdotSlashW, Well for zaptel, you'll have to reload chan_zap.so or restart * entirely.
21:24.29[TK]D-FenderdotSlashW, The other is converd by a basic reload
21:24.52dotSlashWI'm changing the destination for two incoming routes
21:25.03dotSlashW(I think I was not clear earlier)
21:25.03Nate9939j4k3 got this error when incoming call comes in...  Executing [525546240128@from-voxbone:1] Dial("SIP/81.201.84.29-09752a98", "SIP/softphone") in new stack
21:25.04Nate9939[Jun  6 14:22:35] WARNING[18710]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
21:25.04Nate9939<PROTECTED>
21:25.04Nate9939<PROTECTED>
21:25.20*** part/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-160-113.ny325.east.verizon.net)
21:25.30*** join/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-160-113.ny325.east.verizon.net)
21:25.32dotSlashW[TK]D-Fender: you sure I need to reload * entirely ?
21:25.48[TK]D-FenderdotSlashW, I just told you your 2 options.
21:26.10dotSlashWok, just wanna make sure I was clear
21:26.40dotSlashWhave to change properties on Inbound routes, not the trunks
21:28.11[TK]D-FenderdotSlashW, Your terminology is very misplaced and very telling...
21:29.23anzenHm
21:29.28dotSlashWthat's why I said it again, in other words...
21:35.18dotSlashWTK]D-Fender thanks for your help
21:35.29dotSlashWwould it be the same price with a smile ?
21:35.33[TK]D-FenderdotSlashW, np
21:36.02[TK]D-FenderdotSlashW, No, smiles are in a different pricing structure ;)
21:37.56*** join/#asterisk frocos11292 (n=ask@80.172.186.99)
21:38.58*** join/#asterisk BSD_Tech[laptop] (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net)
21:39.24anzenfor a trixbox setup, when i'm configuring sip.conf, where should i point the context to, if using freePBX to do the inbound routes?
21:39.33*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
21:39.34BSD_Tech[laptop]I was trying to compile a svn head of addons when it should have been branches/1.4
21:39.46[TK]D-Fenderanzen, With FreePBX you don't mess with sip.conf directly.
21:39.58[TK]D-Fenderanzen, You've already sold your soul, and at a bargain no less.
21:40.04BSD_Tech[laptop]anzen you need to go to #freepbx
21:40.13BSD_Tech[laptop]they will help you with a install
21:40.14anzenk
21:40.19anzenthanks BSD_Tech
21:40.43Nate9939sweet, got my first incoming DID working.  ahh the bliss.
21:40.47BSD_Tech[laptop]poor sucker In for a penny infor a pound
21:40.58BSD_Tech[laptop]he will come crawling back
21:41.57MihiNomenEstis it 'Say-Digits(<digit>)' ?
21:42.10BSD_Tech[laptop]SayDigits
21:42.15MihiNomenEstthx
21:42.18BSD_Tech[laptop]no -
21:42.38[TK]D-FenderSaydigits(12345) = "one two three four five"
21:43.10BSD_Tech[laptop]plattones
21:43.17BSD_Tech[laptop]playtones
21:43.19BSD_Tech[laptop]lol
21:44.58*** join/#asterisk ghenry (n=ghenry@212.159.59.85)
21:44.59BSD_Tech[laptop]if your on asterisk 1.4 do a core show applications on 1.2 its show applications
21:44.59BSD_Tech[laptop]it will show you all you need
21:44.59BSD_Tech[laptop]from the astcli
21:45.23MihiNomenEstso, I followed this guide: http://users.pandora.be/Asterisk-PBX/IVR.htm
21:45.23MihiNomenEstbut, when I dial 1,2,3,4,etc.  it doesn't do anything.
21:45.39MihiNomenEstit just drops to my timeout rule.
21:46.12BSD_Tech[laptop]ok wait
21:46.15BSD_Tech[laptop]you have to read the input then play it back
21:46.58MihiNomenEstwhere do I find out how to do that?
21:47.30[TK]D-FenderLOL
21:47.37lesouvageCan somebody explain to me why this isn't working exten => sshd_stop,1,System(/etc/init.d/sshd stop)
21:47.40[TK]D-FenderMihiNomenEst, That pages "guide" is busted up.
21:47.51[TK]D-FenderMihiNomenEst, I've seen it before.. and its ANCIENT too.
21:48.12MihiNomenEst[TK]D-Fender: yes, but it's one of the few I can find.
21:48.19[TK]D-Fenderoh God... TOTALLY broken.
21:48.38[TK]D-FenderExten => 2,2,Dial(SIP/Phone1/20) <- ICK!
21:49.00[TK]D-FenderExten => 40,5,Background(greeting) <- 1.0 era IVR's running off non"s"?!
21:49.03[TK]D-Fender*cough)
21:49.19[TK]D-FenderExten => 40,4,ResponseTimeout,10 <- GONE!
21:49.40[TK]D-FenderVery little that author DIDN'T get wrong
21:50.06MihiNomenEstno doubt, however, this "guide" appears to be identical: http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu
21:51.44[TK]D-FenderMihiNomenEst, no, quite different and though deprecated is at least PROPER for its time
21:52.05[TK]D-FenderMihiNomenEst, At least 1.0.X compliant
21:53.37lesouvageMihiNomenEst: If you think it is usefull I can put a working example on pastebin.
21:54.33[TK]D-Fenderlesouvage, Show us your attempt to use that exten (CLI output).
21:55.32MihiNomenEstBSD_Tech just showed me a 3k+ line example.  I think it'll take me a week to sort through that.  But, thanks.
21:55.51*** join/#asterisk javar (n=javar@69.79.134.24)
21:56.21lesouvageMiHiNomenEst: I have paste a short example. Goto http://www.pastebin.be/2499
21:57.03MihiNomenEstwow, yeah, that is short.
21:57.39*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
21:58.04[TK]D-Fenderlesouvage, ick.... no answer, no timeouts set, line 37 = missing exten.
21:58.27*** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
21:58.39[TK]D-Fenderlesouvage, and ... "r" = evil! ;0
22:01.21lesouvageIt's a part of a presentation "asterisk for beginners" I don't claim that it is perfect but it helps to explain the principles of making menu's. The answer is above this menu before the "welcome message" and sometimes the default timeout works ok.
22:01.41[TK]D-FenderMihiNomenEst, http://www.pastebin.ca/545858
22:03.01*** join/#asterisk guille1983 (n=chatzill@190.73.188.118)
22:03.05guille1983hello
22:03.06[TK]D-FenderMihiNomenEst, 3-strikes you're out coverage for timout/invalids, include's my internal extensions context (which contains only extensions meant tor ing phone endpoints).
22:03.32guille1983is there a design rule to "design" a dial plan ?
22:04.21dotSlashWwhen I execute "asterisk -rx reload" sometimes I get some output, sometimes not. Why is that ?
22:05.37BSD_Tech[laptop]TK why do you play Background twice when you can save the work and do Playback(file&file)
22:06.45BSD_Tech[laptop]bbiab time to move inside
22:07.28[TK]D-Fenderbsd_Tech, I honestly never thought to merge them.
22:07.54[TK]D-Fenderbsd_Tech : is more readable being in 2 though.
22:08.31*** join/#asterisk CoolGuy21 (n=77889789@cpe-76-173-56-41.socal.res.rr.com)
22:08.32CoolGuy21hi
22:08.40CoolGuy21i have a 7940 and have put in new background images in tftp server but its not showing up, the old image is showing instead?
22:09.22[TK]D-Fenderlesouvage, So.. about that System problem.....
22:10.43lesouvageI will paste the cli output. First I thought it was because of the space so I put the "/etc/init.d/sshd stop" in a file and execute the file.
22:10.51lesouvageto pastebin.be
22:12.31lesouvage[TK]D-Fender>: see http://www.pastebin.be/2501
22:13.08*** part/#asterisk sunsmasher (n=Beamer@66.251.47.154)
22:13.21*** join/#asterisk sunsmasher (n=Beamer@66.251.47.154)
22:13.40[TK]D-Fenderlesouvage, Ok, now "ls -l /var/lib/asterisk/agi-bin/sshd_stop"
22:14.04ruied_is there any linux player for gsm codec? xmms or something like that?
22:14.14*** join/#asterisk unspin (n=unspin@24.82.161.85)
22:15.22lesouvage[TK]D-Fender>:  -rwxr-----  1 root root 22 Jun  6 23:43 /var/lib/asterisk/agi-bin/sshd_stop
22:15.43errranyone know if you can rewind a message in voicemail while listening to it? like in the middle of the message hit some key and it backs up a couple seconds or something...
22:15.48Qwell[]ruied_: sox
22:16.16ruied_Qwell, thanks, going to check...
22:16.33[TK]D-Fendererrr, */# <-
22:16.52errr[TK]D-Fender: ahh thanks
22:17.32*** join/#asterisk Ryzer (n=registra@83.101.1.70)
22:18.40[TK]D-Fenderlesouvage, Tried "/bin/bash /var/lib/asterisk/agi-bin/sshd_stop"?
22:21.00*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
22:22.21lesouvage[TK]D-Fender>: sshd is still up and running
22:22.42lesouvageafter I called in and pick sshd_stop from the menu
22:23.54lesouvage[TK]D-Fender>: btw te typos of my paste on pastebin.be were not in the final presentation (happy me)
22:24.28*** join/#asterisk oej (n=olle@AMontsouris-152-1-41-187.w82-123.abo.wanadoo.fr)
22:25.37lesouvage[TK]D-Fender>: the idea of what I'm trying to do is just start the sshd when I need it by calling in and have the extension just available when my own phone number is used.
22:29.34CoolGuy21hey
22:29.47CoolGuy21can i lock each extension to a MAC address?
22:30.50*** join/#asterisk oej (n=olle@AMontsouris-152-1-41-187.w82-123.abo.wanadoo.fr)
22:36.25RyzerCoolGuy21: If I recall correctly, this feature was mentioned in the dev mailling list, but I don't think it has been implented yet.
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22:38.26JTLayer 2 is generally out of asterisk's scope, CoolGuy21
22:38.31RyzerCoolGuy21: By the way, MAC addresses can be cloned/forged so it's not really secure to use it as a security measurement
22:44.32CoolGuy21hummm
22:44.50CoolGuy21ok what do i do if i want to limit an account to only 1 desktop
22:47.11JTwhat's wrong with a username + password?
22:47.40n0n4m3CoolGuy21 you could limit it to an ip
22:47.48*** join/#asterisk infernix (i=nix@unaffiliated/infernix)
22:49.18n0n4m3in case you'd really like to limit the account to one desktop i guess you'd need a server with multiple NICs
22:49.51n0n4m3and firstly limit all the network mumbo jumbo and then configure sip
22:49.55JTn0n4m3: what on earth for?
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22:53.05*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
22:53.55blitzrageanyone know if you can make it so a caller in a Queue() can hit a button their phone (DTMF) and exit out of the Queue() to another part of the dialplan? (i.e. press 1 to exit to voicemail)
22:55.57n0n4m3JT dunno.. he'd really like to limit one extension to one box
22:56.15n0n4m3guess some people like to complicate things
22:56.17JTwhy do you need a network card for each extension then?
22:56.43n0n4m3so noone could spoof other user
22:57.08n0n4m3because you'd be limiting one ip per nic and only allow one user from that ip in sip.conf
22:57.09blitzrageisn't that what a switch does?!
22:57.18n0n4m3nope
22:57.27blitzrageI'm obviously coming in part way
22:57.35n0n4m3at least a level 1 or level2 switch
22:57.36JTblitzrage: you're correct though
22:57.42JTn0n4m3: a network switch would do fine
22:57.47n0n4m3level3 switches are usually smart enough
22:57.59Qwell[]layer, not level...
22:58.03JTn0n4m3: having a card for each extension is madness
22:58.04n0n4m3sry
22:58.07Qwell[]level3 switches are usually pretty dumb
22:58.22n0n4m3Qwell depends who configures them
22:58.23n0n4m3:D
22:58.24Qwell[]s/l/L/
22:58.26Qwell[];)
22:58.29*** part/#asterisk putnopvut (i=putnopvu@nat/digium/x-fe74fe70049e094f)
22:58.51Qwell[]I think you missed the joke.  That's okay.
22:59.08n0n4m3well never mind
22:59.16n0n4m3while we're at it
22:59.39n0n4m3i've got a question...
23:00.13JTn0n4m3: i still don't see the benefit of one NIC per extension
23:01.05*** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com)
23:02.29n0n4m3i'd like asterisk to 'spoof' a sip client (patton dta). unfortunately my sip provider is very restrictive so it only allows to register sip clients that have the correct user agents and from: and to: strings. asterist idenfifies itself as asterisk <asterisk@ip> even though i have fromuser=.... in sip.conf
23:04.20dotSlashW[TK]D-Fender you thre ?
23:04.55n0n4m3JT okay... let's say you have 10 extensions and 10 users... you limit user1 to ip1, user2 to ip2, ... every ip is on it's own subnet and server isn't configured as a bridge or even as a switch/router/whatever. and you also limit each ip to each own nic. so even though users can spoof other users, they don't register because the spoofed ip isn't on that nic
23:05.54*** join/#asterisk infernix (i=nix@unaffiliated/infernix)
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23:09.59JTn0n4m3: if you wanted to go that route, i don't see why you wouldn't be able to do multiple IPs on one NIC?
23:10.37justdaveis there a way to set a timeout on a SIP link so it'll automatically hang up if the link is unreachable for a certain amount of time?
23:10.57JTjustdave: are you talking SIP or RTP?
23:11.10justdaveeither/both I guess...
23:11.22justdavehad a SIP phone get its cable yanked from the wall while it was on the phone...
23:11.41justdavewhen they plugged it back in, it started new connections, and the old one is still "active" even though the device got disconnected
23:11.45Qwell[]justdave: there is an option for that, but I don't recall what it was...
23:11.56Qwell[]rtptimeout maybe?
23:12.06JTmost likely
23:12.14Qwell[]ie; hang up a call if we don't get an rtp in X seconds
23:12.19Qwell[]s/an/any/
23:12.26Qwell[]justdave: check the sample sip.conf
23:12.50justdaveyep, rtptimeout is it.
23:13.19justdave120 seconds ought to be plenty I'd think.  By that point whoever's on the other end will have given up already anyhow :)
23:13.47justdavehmm, muted X-Lite with send-silence disabled would throw a wrench in that I suppose
23:13.51Qwell[]yes
23:14.00Qwell[]it's a decision you have to make :)
23:14.08Qwell[]bbl
23:14.17justdaveseems like we're already telling people to enable send-silence anyway
23:14.37Qwell[]You could also set it to something unreasonable - like an hour
23:15.04Qwell[]If you don't get a packet in an hour, it's probably not VAD ;)
23:15.19[TK]D-FenderdotSlashW, What is it?
23:15.20justdaveyeah, was just thinking that.
23:15.28justdavejust typed 3600 into the file as you said that :)
23:15.36justdavethat'll keep the ghosts from collecting at least
23:16.29JThrm
23:16.54JTsip/rtp needs a less stupid way to deal with mute/VAD/etc
23:16.54JTit needs keepalive packets
23:17.27*** part/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net)
23:17.56*** join/#asterisk friedrich| (n=friedric@e177241229.adsl.alicedsl.de)
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23:19.39n0n4m3JT with packet filtering.. let's say on linux... iptables -i eth4 -s ! 192.168.4.2 -j DROP
23:20.08JTn0n4m3: still fail to see why you need multiple cards, i haven't heard a single argument for it so far
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23:21.23dotSlashWwhat is the command executed by freepbx when I hit the red bar ?
23:21.25[TK]D-Fendern0n4m3, why so friggen paranoid?
23:21.37[TK]D-Fender~freepbx
23:21.47jbotrumour has it, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
23:21.48[TK]D-Fender^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
23:21.57dotSlashWdidn't get that answer there ...
23:22.06n0n4m3[TK]D-Fender dunno... i'm just looking for a solution for CoolGuy21
23:22.07n0n4m3:D
23:22.08[TK]D-Fender~trixbox
23:22.10jbotTrixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org
23:23.27[TK]D-Fendern0n4m3, Just use a single nic with a single switch and run a bloody VPN and STFU :)
23:23.31JTn0n4m3: there is absolutely no requirement for multiple physical cards
23:23.54*** join/#asterisk friedrich| (n=friedric@e177241229.adsl.alicedsl.de)
23:24.24n0n4m3well never mind then...
23:24.25n0n4m3:D
23:24.29JTn0n4m3: you do know you can have at least 255 IPs per ethernet interface in linux?
23:24.54*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
23:24.54JTand presumably most other unicies
23:25.12[TK]D-FenderJT.... You can ararange less ;)
23:25.32*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
23:26.37*** join/#asterisk friedrich| (n=friedric@e177241229.adsl.alicedsl.de)
23:26.47[TK]D-Fendern0n4m3, Or you could just skip the whole IP thing use a proprietary encrypted digital phone and be done with it as well ;)
23:26.55*** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
23:27.10[TK]D-Fendern0n4m3, or exchange data in sandskrit.  Security through obscurity!
23:27.28[TK]D-Fendern0n4m3, no wait... I've got it!
23:28.17[TK]D-Fendern0n4m3, 2 tin cups, and a face-to-face meeting under the "Cone Of Silence" required to connect the strig to each end!
23:31.42*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
23:36.15*** join/#asterisk heh_v_water (n=heh_v_wa@71-210-35-31.hlna.qwest.net)
23:37.07[TK]D-Fendern0n4m3, Now THAT'S "handshaking"!
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23:52.05JTMihiNomenEst: did it mostly involve the inhaling of drugs?
23:53.58[TK]D-Fender"Cone of Silence" .... what you buy your kid as you pass a Dairy Queen to shut them up :)
23:54.59J4k3my cone of silence has "DUNCE" written on the front
23:55.05DrukenHMEanyone got alison saying AND ?
23:55.27JTand then?
23:55.41[TK]D-FenderAND THEN?
23:56.28J4k3http://www.extremetech.com/article2/0,1697,2142228,00.asphttp://www.extremetech.com/article2/0,1697,2142228,00.asp
23:56.52J4k3so does this mean I can soft-overclock to handle more calls at peak?!?! :)
23:57.09J4k3oh wow
23:57.28J4k3I didn't even paste properly
23:58.35JTcan you overclock a T2000 is more the question ;)
23:59.16J4k3I think AT&T is trying to overclock this T1.
23:59.22J4k3and/or, they need to overclock their router.

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