IRC log for #asterisk on 20111129

00:05.58*** join/#asterisk timahvo1 (~rogue@41.81.142.133)
00:06.46*** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com)
00:11.40*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
00:24.12*** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net)
00:25.59SeRip3nguin: ah! ok I understand
00:26.02SeRisorry for the afk
00:26.57p3nguinI think I have like five total translations, so it wasn't hard to narrow it down.
00:28.02SeRilol
00:28.07SeRisorry :P
00:28.16p3nguinEverything headed for 5060 is from 5060.
00:28.46*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
00:28.56SeRiah I see.
00:28.57AdamNdoes asterisk 1.8 still use extensions.conf?
00:29.05p3nguinOnly if you want it to.
00:29.06SeRiAdamN: yes
00:29.38p3nguinBut out of curiosity, what else would you have expected?
00:30.06AdamNI just upgraded from 1.6, and it has lost all configuration, and running "config list" in the cli shows it is not seeing extensions.conf
00:30.36p3nguinIs pbx_config present?
00:31.54AdamNp3nguin: no
00:32.01p3nguinThere's yer problem.
00:32.32AdamNp3nguin: within /etc/asterisk?
00:32.44p3nguinNo
00:33.02p3nguinmodule show like pbx
00:34.02AdamNno, all module show displays is "res_adsi"
00:34.12p3nguinThen you have work to do.
00:34.45p3nguinIn /etc/asterisk, what files do you have present and configured?
00:35.31AdamNwell before I get into that I ran "module load PBX_config"
00:35.38AdamNand it is now loaded
00:36.30p3nguinAnd config list should show that pbx_config is using extensions.conf.
00:36.39*** join/#asterisk [Outcast] (~anonymous@pool-96-252-45-211.bstnma.fios.verizon.net)
00:36.43p3nguinpbx_config           /etc/asterisk/extensions.conf
00:36.53AdamNyes
00:37.11p3nguinPerhaps you have a jacked up modules.conf or something.
00:38.43AdamNwhere is modules.conf reside?
00:38.56p3nguin/etc/asterisk with all the other confs.
00:39.23AdamNthat does not exist
00:39.30AdamNmodules.conf that is
00:39.34p3nguinYOu need one.
00:39.49p3nguinYou'll want to make sure you define autoload to be enabled.
00:40.08p3nguinLook at the sample modules.conf.
00:40.26AdamNI have a backup from before i updated this afternoon
00:40.30p3nguinDid you blow away all your confs from your previous setup?
00:40.57p3nguinI would have thought you'd leave them in place and attempt using them as they were.
00:41.15p3nguinThere will be a few changes, but for the most part, things would have worked.
00:41.23AdamNnope, all the confs from /etc/asterisk were copied and backed up remote, the originals were left on the machine
00:41.40p3nguinSo you're saying you didn't have a modules.conf before?
00:42.09AdamNsorry that got disjointed, I do have my old modules.conf, all the confs were backed up
00:42.38p3nguinI'd copy it into place and restart asterisk to see how it goes.
00:42.46AdamNok
00:43.50p3nguinFor the most part, things are the same in the confs.  There are, however, a few subtle changes in some of them.  I took my original confs and used them, then made the corrections for new values and changed syntax where needed.
00:44.45p3nguinvoicemail.conf has a few changes, sip.conf changed a few parameters as well as added a bunch, some of the dial plan apps' syntax has changed slightly, et cetera.
00:45.17AdamNI notice alot of the cli commands are gone.
00:45.28p3nguinNah, they aren't gone.
00:45.38p3nguinYou're just looking in the wrong places.
00:46.07AdamNhas o'rielly or the like updated for 1.8 yet?
00:46.35p3nguinFor example, originate is now channel originate, soft hangup is now channel request hangup, and so on.  If you take a look at your cli_aliases.conf, you can restore all of the old commands.
00:46.42p3nguin~book
00:46.42infobotrumour has it, thebook is Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or ~buybook
00:47.40AdamNthank you very much, that would have been a loooong process
00:49.45p3nguinI use my cli aliases for when I forget I'm not on 1.4, but I try hard to use all the new commands.
00:51.08AdamNwell it looks like everything but voicemail is back up
00:51.36*** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d)
00:57.03AdamNall my voicemail boxes are stating this user can't accept more messages.
00:58.18p3nguinchown -R asterisk:asterisk /etc/asterisk /var/lib/asterisk /var/log/asterisk /var/spool/asterisk /var/run/asterisk
00:58.35p3nguin(assuming you run asterisk as user asterisk and group asterisk like a sane person)
01:14.34*** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk)
01:15.56*** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com)
01:17.07*** join/#asterisk RiceCracker (~RiceCrack@59.152.236.158)
01:19.52*** join/#asterisk xpot-mobile (~xpot@166-70-100-198.ip.xmission.com)
01:27.58SeRip3nguin: everything ok so far?
01:28.06*** join/#asterisk ariel_ (u3533@pdpc/supporter/active/abatista)
01:28.15ariel_Hello everyone
01:30.22ariel_quick question about queues and agents.  This is on a asterisk 1.4.33.1 system.  The polycom phones are logged in, but from time to time the when a call comes in, it will ring the phone, for one ring then go away, then come back, it sometimes goes to another agent, but most of the time it just waits about 5 sec then rings the same phone.  Only way around this is for us to do a reload or for the ag
01:30.22ariel_ents to log off wait about 30 sec then log back in.  This is using remoteagent log in....
01:31.47*** join/#asterisk neurosys (~neurosys@c-67-191-66-234.hsd1.fl.comcast.net)
01:34.13*** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net)
01:34.52*** join/#asterisk xpot-mobile (~xpot@166-70-100-198.ip.xmission.com)
01:37.42*** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net)
01:44.42*** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net)
02:06.45SeRip3nguin: I have not get my password from freenum.org... should I email them?
02:09.41*** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap)
02:11.04SeRinevermind
02:12.45*** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net)
02:15.24SeRip3nguin: Is there anything special I need to do @ freenum.org? I am loged in in my account
02:17.24p3nguinYou just need to configure some DNS stuff.
02:17.50SeRion my server?
02:19.41p3nguinno
02:19.58SeRiok
02:20.13p3nguinSign in on freenum.org.
02:20.28*** join/#asterisk mintos (~mvaliyav@115.241.41.116)
02:20.30p3nguinIn the DNS settings, create your settings.
02:20.40p3nguin(it's not magical)
02:21.16SeRijust did.
02:21.20SeRi:)
02:21.21p3nguinAfter you do DNS settings, then configure anything in the ITAD settings that you want to configure.
02:21.53*** join/#asterisk master_of_master (~master_of@p57B54457.dip.t-dialin.net)
02:39.51p3nguinHow's that working out for you?
02:40.27SeRiTrying now
02:41.32SeRip3nguin: so with the context you gave me I dial 012623*262?
02:42.01p3nguin012 is the dialing prefix for ISN.
02:42.29p3nguinIf you were trying to dial 623 @ 262, then that would be correct.
02:43.11SeRiin there site they have an echo test number 623*262
02:44.22SeRiI dial 012623*262 and I get a fast bussy tone
02:44.23p3nguinIt isn't echoing.
02:44.28p3nguinYeah, it's broken.
02:44.32SeRiah ok
02:45.08p3nguinI'll give you one to test.
02:45.14SeRiThanks! :)
02:46.14p3nguinSeems to be working.
02:46.15SeRihahahaha!
02:46.21SeRirofl!
02:46.24SeRinice
02:46.53SeRilol cool
02:47.26SeRifor me to recive incoming calls do I just create a normal context?
02:47.54SeRis/recive/receive/
02:48.39p3nguinYou already have a context.
02:48.48p3nguinNow you just need to have extensions.
02:49.01SeRilol :)
02:49.05SeRiindeed
02:49.13SeRi3 months in and still get it confused
02:49.14p3nguinCalls to you via ISN will end up in your misc_calls context.
02:49.24SeRiI see
02:49.38SeRican you share an example?
02:49.40p3nguinI have an incoming extension prefix to be able to determine ISN from regular SIP calls.
02:49.44p3nguinOne moment.
02:49.51SeRiThank You.
02:51.01p3nguinCan you choose an arbitrary prefix for ISN inbound?  This will be on every extension dialed in.
02:51.13p3nguinexample, 123
02:51.22p3nguinor 99
02:51.59SeRisure.
02:52.04SeRi223
02:52.24p3nguinAnd your internal extension number is what?
02:52.31SeRi1003
02:52.40SeRimy office ext^^
02:54.52p3nguinhttp://pastebin.com/ywgYtjYn
02:56.20SeRicool. one sec
02:56.30p3nguinSo if I dial 2231003*yourITAD, it should hit your unauthorized calls context and reroute it to 1003 in your internal context.
02:57.13p3nguinAnd it prepares the caller ID so that all you have to do is press the Dial key on your phone.  It would change my number into 012mynumber.
03:00.00SeRiawesome. :)
03:00.08SeRican we test it?
03:00.47p3nguinYes.
03:00.49p3nguinShall I dial?
03:00.55SeRiPlease :)
03:01.45p3nguinDidn't work.
03:02.22SeRiMhhhhhh
03:02.24SeRione sec
03:03.59p3nguinThe DNS isn't good.
03:04.05SeRiyea I just saw that
03:04.08p3nguinMaybe it hasn't propagated.
03:04.11SeRiit has not propagated.
03:04.16SeRi+1
03:04.31SeRiill wait till tomorrow
03:04.37SeRiThanks any ways :)
03:04.50SeRiBy the way I pass my class!
03:05.14SeRiis going to party!
03:05.18SeRilol
03:05.19SeRi:P
03:05.28SeRiI really thought I was going to fail :/
03:05.42SeRiThat was the hardest class ever!
03:08.35SeRip3nguin: what headset/mic do you use with your computer?
03:09.13p3nguinIt's some $3 crap I got from Hong Kong via ebay.
03:09.57*** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer)
03:10.05SeRinice. I am shopping for one and found a few 3.99 on newegg.
03:11.03p3nguinI also have a $20 Logitech headset that I picked up at CVS on their 75% off clearance rack, but I've never used it.
03:11.26SeRiI see.
03:11.47SeRiwell I was looking at a mic and usem y current head set.....
03:12.05SeRis/usem y/use my/
03:12.16SeRinot sure yet
03:12.35p3nguinYou're planning to use a soft phone instead of your hard phone?
03:13.22SeRifor my laptop down stairs yet
03:13.26p3nguinoh
03:13.30SeRiyes*
03:13.32SeRi:)
03:14.05SeRimy wife is starting to like the idea voip sof phones and such :)
03:14.18SeRidamn that was all fucked up
03:14.49s[X]hey all
03:14.50SeRibut any who yes... she is now using her cell to call my ext on my office.
03:15.03*** join/#asterisk gajini (~root@61.12.17.170)
03:15.05SeRiuising scip
03:15.13SeRis[X]: hola
03:15.53s[X]I wish iphone included Native SIP
03:15.56s[X]Would be so nice
03:16.03SeRidont we all
03:16.09s[X]even JB dont do it
03:16.25JerJeracrobits wont work s[X]?
03:17.07s[X]its not native is it ?
03:17.32p3nguinIt's not native, but it's a SIP phone app.
03:17.45s[X]I like Bria
03:17.48SeRiandroid 2.3 branch has native sip
03:18.00s[X]Nokias have native sip for ages
03:18.21SeRiNokias are like mercedes though
03:18.46SeRiyou have to give your first born child for one of there phones
03:18.53p3nguinI have a Nokia 918.
03:18.59SeRis/tehre/their/
03:19.38SeRilol! keep it around p3nguin they are good phones
03:20.04p3nguinI don't think it's good for anything.
03:20.41SeRiWhen the world sprungs in chaos it would become a nice weapon!
03:21.10SeRiput it inside a sock and wave the sob around!
03:21.17s[X]I had an N95 then an N96
03:21.20s[X]SHouldve kept the N95
03:21.25s[X]made the N96 eat shit
03:21.28SeRismack the firs fucker in the head thatc omes close to you
03:21.52SeRiI bet the phone would still be in one piece
03:22.06SeRis[X]: I had a 770 and a 800
03:22.18SeRiI rape them to no end and sold them.
03:33.42SeRip3nguin: is resolving for me now on this side.
03:33.57SeRithe hostname that is
03:41.43p3nguinStill nothing here.
03:42.17SeRiok thanks :)
03:56.05p3nguinWhere did you come up with the idea that the DNS was able to resolve?
04:01.38p3nguinI'm querying the freenum nameservers directly, and there's no record for you.
04:02.34p3nguinNXDOMAIN
04:02.50SeRiwell I was doing a look up on the domain it self
04:03.01SeRiI guess I was wrong
04:03.23s[X]im ashamed to admit i didnt completely understand linux permissions untill recently
04:03.24SeRiyour doing a look up on freenum.... ok I see
04:03.44s[X]lol
04:04.06*** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d)
04:04.22p3nguinHere's an example:  dig @anyns.pch.net 3.2.1.404.freenum.org. NAPTR
04:04.57p3nguinQuerying one of the freenum ns for ITAD 404
04:05.08p3nguinI get an answer.
04:05.19p3nguinYours, nonexistent.
04:05.26SeRio I see
04:06.14p3nguinSo maybe it's broken.
04:06.23p3nguinOr maybe you didn't fill out the DNS settings.
04:06.27p3nguinI have no idea.
04:06.30SeRiI did....
04:17.44SeRip3nguin: any other ideas?
04:17.58p3nguinContact admin@freenum.org about it.
04:18.22SeRiok
04:18.57p3nguinTell them exactly what you told me about things not working correctly.
04:21.45SeRidone.
04:22.10s[X]Whats the deal with ISN
04:22.18s[X]How does it work
04:23.55p3nguinIt's a SIP URI which is obscured with a fancy DNS lookup.
04:24.47*** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110)
04:25.02p3nguinThe usual SIP URI is something like 123@my.host.com or name@my.host.com, but it's hard to dial those from a standard phone keypad.
04:25.16p3nguinSo we use an ITAD instead of a host name.
04:25.19p3nguin~itad
04:25.20infobotextra, extra, read all about it, itad is an IP Telephony Administrative Domain. Similar in nature to an AS (Autonomous System) number, it is administered by IANA (Internet Assigned Numbers Authority). An ITAD number is part of the TRIP specification in RFC 3219. Although TRIP never took off, ITAD numbers are being used by Freenum.org as part of an ISN (ITAD Subscriber Number). For more information about ISN numbers, see 'isn'.
04:25.28*** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net)
04:25.35p3nguinAnd we use a * instead of an @.
04:25.59p3nguinSo to call 123 on ITAD 404, you'd dial 123*404 from your regular keypad.
04:26.32p3nguinIt makes SIP-SIP calling possible for more people.
04:26.37s[X]ah ok
04:27.27s[X]so you would register a ITAD point it to a subdomain, that points to ur asterisk box and bobs ur uncle ?
04:27.46*** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj)
04:27.52p3nguinSomething like that.
04:28.01s[X]Sounds cool
04:28.06s[X]whats an ISN worth ?
04:28.24hardwireI need to get into freenum
04:28.27p3nguinIt costs nothing, so I don't know how to calculate the value of it.
04:28.43s[X]Priceless ?
04:28.58p3nguinor worthless?
04:29.27p3nguinI'm suddenly reminded of the Family Guy episode where Carter is trying to pay off Peter to leave Lois alone.
04:29.43SeRilol
04:29.53p3nguinShe may be worth a million dollars to you, but to me, she's worthless!
04:30.00s[X]lol
04:32.26p3nguinNo one ever calls me by ISN, but it's still a useful tool to have.  Since it's free, you may as well go ahead and register for an ITAD so you can play with it.
04:32.40s[X]yeah tis what im thinking
04:33.22s[X]Just because im trying to understand it a little better, If i were to dial your iSN from my Soft phone
04:33.35s[X]Does it route direct asterisk box to asterisk box
04:33.42s[X]assuming u had an asterisk box
04:33.49p3nguinyes
04:33.53p3nguinThat is exactly what it does.
04:33.55s[X]Thats pretty cool
04:33.59s[X]i want one
04:34.00s[X]lol
04:34.05p3nguinIt's a SIP URI hidden with fancy DNS.
04:34.12s[X]yeah
04:34.35s[X]goes off to setup a subdomain for his ISN
04:34.59s[X]mmm which domain to pick from
04:35.00s[X]lol
04:35.21SeRipenusinyourface.org?
04:35.29s[X]i dont have that one
04:35.33s[X]i have penus-in-you-face.org
04:35.45s[X]or my-bum-burns-mommy.org
04:35.54SeRilol j/k :) It is avail though! lol
04:36.00s[X]hahah
04:36.25SeRishitshaper.com
04:36.29p3nguinheh
04:36.30SeRidijib: ^^
04:36.48p3nguinKeep in mind that no one dialing by ISN ever sees the domain name.
04:37.05s[X]yeah i know but i figured id use a domain that i likely wont retard
04:38.34SeRifucking weather.com cut off free weather api. fuckers
04:39.19SeRinow I have to go dig a script for conky to work with conkyforecast
04:39.20s[X]Seri...
04:39.24s[X]BOB ?
04:39.51SeRibob?
04:40.14s[X]Seiri, Inc.
04:40.32s[X]There is a ITAD registration with that as the Organization u can see how i jumped to the conclusion lol
04:40.44SeRilol
04:40.52SeRinot me sr :)
04:40.58s[X]lol
04:41.07s[X]I figured as much just thought it was conincidence
04:41.49SeRiI am in there just not as bob :)
04:43.05SeRip3nguin: some fucked up shit just happen :/
04:43.08p3nguinI'm there as well, and also not "bob."
04:43.15p3nguinmodem rebooted?
04:43.25SeRimy record @ freenum record got deleted...
04:43.40SeRimy freenum record*
04:44.40s[X]just because im more curious
04:44.43SeRiI wonder if they are working on it
04:45.27s[X]if ias assigned ITAD 123
04:45.29s[X]was*
04:45.34s[X]and my Extension was 201
04:45.39s[X]would all you need to dial is 201*123
04:45.49p3nguinThat's the idea.
04:45.57s[X]that seems overly simple, why on earth are there so few registrants
04:46.15p3nguinYou have to have dial plan to do it, but that part is very easy as well.
04:46.48s[X]Woot, Submitted
04:46.53s[X]I await there contact
04:47.52s[X]When will telcos allow ISN dialing from land lines ?
04:48.20SeRip3nguin: I sent you a msg with the record
04:51.31SeRip3nguin: the context for inbound goes in misc calls right?
04:59.15p3nguinISN calls will be anonymous calls.  Anonymous SIP calls go to the context you have assigned in the general section of sip.conf.
04:59.40SeRiJust wanted to make sure.
05:02.01*** join/#asterisk ChannelZ (channelz@burner.com)
05:08.50*** join/#asterisk irroot (~gregory@197.110.156.226)
05:11.45s[X]mmm i wonder if i can make outbound calls on my samsung pbx to isn
05:12.15p3nguinIf it does SIP, then you probably can.
05:12.34s[X]yes it does sip
05:12.37s[X]its a 7200s
05:12.58s[X]I will probably have to setup a outbound router for ISN
05:13.11s[X]ill email samsung and get them to do it remotely, i dont want to break anything
05:13.29irroots[X] samsung 7200 is nice but SIP is extra $$$ and does NOT do nat properly
05:13.46irrootmorning folks
05:13.49s[X]its a shitty SIP box imho
05:13.53s[X]morning irroot
05:14.34irroots[x] the 7200 has a identity crisis does not know what it is does legacy PBX + VOIP + Ethernet switch +  ....
05:14.44SeRilol
05:15.00SeRip3nguin: I am wondering if freenum does not like cnames...
05:15.28p3nguinI doubt it matters.
05:16.10s[X]Its very annoying because when i was doing my research I thought it was quite a SIP capable unit
05:16.10*** join/#asterisk kikohnl (~kotis@udp019436uds.hawaiiantel.net)
05:16.25s[X]I got it recommended by 4 seperate companies who quoted me
05:20.19s[X]Samsung are apparently released quite a major overhauled version early next year to acoomodate for Multiple SIP Carriers & More control over codecs
05:20.28s[X]s/released/releasing
05:21.11irroot<PROTECTED>
05:22.34irrootthe way we accomodate these products is pop in a small Atom based micro PC with asterisk on it let asterisk do the real work
05:23.23s[X]So instead of getting SIP trunks for the PBX just get SIP channels for the Asterisk box ?
05:23.49*** join/#asterisk sezuan (bouncer@irc.scheff32.de)
05:23.56irroots[X] and then get the PBX to "trunk" to asterisk
05:23.57SeRis[X]: This is called "What they say and What they mean" http://pastebin.com/raw.php?i=4Pys0PMV
05:24.39s[X]mmm
05:25.02s[X]Because i have 8 SIP trunks for my SIP provider
05:25.06s[X]i could use those on asterisk couldnt i
05:25.18p3nguin~siptrunk
05:25.18infobotsiptrunk is, like, something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk
05:25.35p3nguin~itsp
05:25.35infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.
05:26.11s[X]thanks p3nguin
05:27.03s[X]slides the word channels in
05:28.13irrootp3nguin elephants have trunks and sip through them ??? is that a sip trunk
05:28.28p3nguinhttp://imagebin.org/185866
05:28.30SeRilmao
05:28.52p3nguinasterisk trunk
05:30.34irrootthe term trunk is synonyms with a trunk line its not only in asterisk one of the more vocal ISP's advertise VOIP Trunk to replace existing analogue trunk .... confusing the masses
05:30.38s[X]frantically needs to photoshop a trunk onto the asterisx character
05:31.17irroots[X] you know those Mweb fools here in ZA
05:31.23s[X]yes
05:31.30SeRiheads out to bed. ftw!
05:31.38s[X]later SeRi
05:31.38p3nguinPill?
05:31.42SeRiYes Sr!
05:31.47p3nguin:)
05:31.51SeRilol
05:31.52p3nguinlaytor
05:31.56SeRiIts amazing
05:31.59irrootSeRi cheers /me just woken up
05:32.03SeRil8trs!
05:32.05p3nguinEnjoy the sleeps.
05:32.11SeRiwel g/m and g/n!
05:32.14SeRi:D
05:32.16SeRicya guys
05:32.25irrootcheers
05:33.11SeRi|zzZZzzcya.
05:33.44*** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap)
05:38.09s[X]irroot, im intrigued by setting up a asterisk box between my pbx and ITSP
05:38.48irroots[X] we do it all the time sometimes even use the PRI / BRI interface to the PBX on asteriks
05:39.27s[X]Being that i have 8 licenses for PBX
05:39.30s[X]SIP*
05:39.30irrootwe do a Inline where put a 2 port PRI in asterisk on port to the PBX one to Telco and the ITSP
05:40.09*** join/#asterisk shido6 (~shido6@c-98-234-181-35.hsd1.ca.comcast.net)
05:40.35irrootthen asterisk does the call logging via monitor the ACD via app_Queue and the ivr also add things like call limits by value so say 100R calls per month
05:41.10irrootalso does a good job as TMS
05:41.19s[X]I wrote my own SMDR capturing app in PHP for the 7200s
05:41.36s[X]so I could capture call logs
05:42.01irrootalso allows voip phones on legacy system
05:44.35s[X]Listens in on a port, grabs incomming stream. Cleans it up and stores it in an SQL Database
05:45.41[TK]D-FenderI did something similar in Turbo Pascal for a Vantage 25 system about 17 years ago :)
05:45.54s[X]:P im still learning
05:46.08[TK]D-FenderSMDR logging, feature codes for blacklisting, CID integration, etc
05:46.21[TK]D-FenderI had way too much free time
05:48.16irroot[TK]D-Fender morning there
05:56.20*** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net)
05:58.06*** join/#asterisk irroot (~gregory@197.110.156.226)
06:01.13*** join/#asterisk dhorner_mb (~dhorner_m@184.18.44.244)
06:10.13*** join/#asterisk dlynes (~dlynes@70.50.89.162)
06:14.03*** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com)
06:19.06*** join/#asterisk kayfox (~kayfox@xheotris.zerda.net)
06:19.44*** join/#asterisk mintos (~mvaliyav@117.206.22.8)
06:30.20*** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer)
06:31.51*** join/#asterisk e7e5 (~rudenko@188.134.2.33)
06:46.17*** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt)
07:16.36*** join/#asterisk kleszcz (tick@80.54.23.253)
07:19.07*** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105)
07:21.35*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
07:22.02*** join/#asterisk james_zhu (~Administr@183.16.88.158)
07:25.46*** join/#asterisk micols (~ident@rlogin.dk)
07:26.56*** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190)
07:27.11*** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190)
07:28.37*** join/#asterisk AmirBehzad (~behzad@31.184.187.2)
07:35.50*** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at)
07:35.53schmidtsgood morning
07:38.37james_zhu:-D
07:41.58*** join/#asterisk freeman_u (~freeman@gw.quart-soft.net)
07:43.14wdoekes2morning
07:43.30dymhi
07:46.05*** join/#asterisk bulkorok (~bulkorok@85.183.36.36)
07:53.42*** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247)
07:55.06*** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net)
07:55.41*** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247)
08:00.53kikohnl~itsplist-us
08:00.53infobotHere are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com
08:01.55sawgoodIs there a problem using numbers or symbols in a IAX2 context?
08:03.56sawgoodI keep getting, "Registration Refused" when trying to setup an IAX2 trunk between 3 servers (any tips)
08:04.51*** join/#asterisk phpboy (~shane@blowfish.x86.co.za)
08:04.52kaldemarcheck username/matching peer and secret.
08:05.33phpboyMy asterisk keeps crashing and I can't for the life of me figure out why, looks like a memory related issue
08:05.49phpboyanybody got any ideas what I should put of my checklist of things to check?
08:07.37kaldemar~debug
08:07.37infobotACTION DeBuggers $1
08:07.50kaldemar~backtrace
08:07.50infobotbacktrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt).  See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
08:09.57*** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca)
08:10.35*** part/#asterisk AmirBehzad (~behzad@31.184.187.2)
08:11.10*** join/#asterisk bmg505 (~leon@196-209-123-3.dynamic.isadsl.co.za)
08:11.50phpboywhere would the core files generally be stored?
08:12.02phpboyCentOS 5.5
08:13.25sawgoodsomething else other than username/secret is stopping IAX2 from working (I've been chekcing the context for correctness for 2+ hours)
08:15.19phpboyThis does suck, the daemon doesn't crash, I can still get onto the console, it just stops accpeting calls
08:15.24phpboygoes completely quiet :T
08:15.33ChannelZare you sure they are matching the right peers?
08:15.45kaldemarsawgood: pastebin your config and a CLI output of a registration attempt on the receiving side with iax2 debug and verbosity.
08:15.54sawgoodI will pastebin shortly
08:15.56singlerphpboy: then it probably deadlocks, not crashes, so you will not have core dump
08:16.15singlerin that backtrace link is info about deadlocks too
08:16.19kaldemarphpboy: then see the deadlock section for the same page.
08:16.30kaldemarphpboy: which version are you using?
08:17.13phpboy1.8.7.1
08:17.28phpboyon a 64bit server if that makes any diffs
08:19.03phpboythis looks like it's going to be an issue on a production server :T
08:19.55singlerdeadlocking is already an issue on production, to take debug before killing it is not a big deal
08:20.15phpboylol, good point
08:20.25phpboy"Instead take the 5 mins while everyone is freaking out to attach gdb to the running asterisk process and do"
08:20.29phpboyI'm going to do that
08:20.49singlerit will be like 1 min if you practice before and prepare commands :)
08:20.58singleror even less
08:21.33*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
08:24.40sawgoodIf I've made a stupid rookie style mistake, I am sorry in advance (its been several hours)
08:24.44sawgoodhttp://pastebin.com/AiUwgmcS
08:25.42sawgoodthe devices are on the same Ethernet switch (with iptables off)
08:25.58sawgoodthere might be a 'cheap' router in front of the switch for Internet access
08:28.06singlerI think host must be dynamic for it to register
08:28.22kaldemarsawgood: ^^^
08:28.41sawgoodI am trying to grab a CLI of the failed registration attempt
08:28.50sawgoodty
08:29.25ChannelZyeah if the IPs are static there's no need to register
08:29.40singlersawgood:  also if IP does not change, remove register line
08:29.52sawgoodgone
08:30.08*** join/#asterisk Nasga (~Nasga@110.113.117.78.rev.sfr.net)
08:30.11sawgoodhow come after running iax2 reload (it takes a while for the client to re-register with the server)
08:30.37kaldemarsawgood: and change those credentials ASAP if they were real.
08:31.11singlersawgood: did you set host=dynamic or did you remove register line?
08:31.19zyphlarthat's a pretty bad secret O_o use random.org please
08:31.33sawgoodI removed the register line and changed the host=dynamic
08:31.39ChannelZno
08:31.40*** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e)
08:31.55ChannelZhost=dynamic to be used with register
08:32.01*** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com)
08:32.08sawgoodok register comment going back in
08:32.10ChannelZIf the machine's IPs are static, there's no need to use register, set host= to whatever
08:32.38ChannelZthe only thing register does is tell the remote end "here I am at this IP address"
08:32.49ChannelZIt's totally unrelated to whether or not a call will even work
08:32.58sawgoodcool ... got it
08:33.05ChannelZYou can successfully register but still have bothced your peer configs and nothing will work
08:33.16singlersawgood: I guess you want to use qualify=yes to see if host is online
08:33.35sawgoodI have removed the register line ... and I have host= static IP address
08:34.06*** part/#asterisk james_zhu (~Administr@183.16.88.158)
08:34.09ChannelZcross fingers and make a call over it :)
08:35.11*** join/#asterisk qakhan (~qakhan@182.185.177.66)
08:35.43qakhani want to setup speech to text on asterisk
08:35.57qakhanplz help me how to do that
08:41.01sawgoodWhen I attempt to make a call, I can see on the client machine an error msg (saying NO Authority found) ... and on the server side I can see the call arriving but an error message saying the call cannot be delivered
08:41.23singlersawgood: can you pastebin cli logs?
08:44.33sawgoodhttp://pastebin.com/HeTDfYzC
08:45.27*** join/#asterisk mandla (~quassel@168.167.180.161)
08:46.48kaldemarsawgood: IAX2/173.13.158.17/501 is lacking authentication information. if you dial by ip address, use IAX2/username:secret@173.13.158.17/501. if you have correctly defined peers in iax.conf, use IAX2/IAX-ITSP/501.
08:47.56sawgoodI tried using the IAX-ITSP context name in extensions.conf, but the call would not make it that far (I will try again)
08:48.42sawgoodwhen you saying 'lacking authentication information' (is that the client or server side)?
08:49.37sawgoodexten => 501,1,Dial(IAX2/IAX-ITSP/${EXTEN})
08:49.43sawgoodThis is my statement in extensions.conf
08:50.12phpboywhile I'm waiting patiently for asterisk to stop taking calls
08:50.28phpboycan anybody recommend something I an look into while it's running
08:50.45phpboycalls are cutting and all kinds of weird stuff at the mo :T
08:51.10defsworkanyone any ideas why chanspy would drop the spier after a minute or so ?
08:51.10singlersawgood: what is cli output with that exten config? in pastebined log exten is using IP address
08:51.20*** join/#asterisk like_a_horse (~like_a_ho@firect.saao.ac.za)
08:51.49phpboy640 files in my /fd dir
08:51.50phpboyhmmmm
08:52.39like_a_horsehi all, i have a patton smartnode that doesn't seem to want to push calls through to my asterisk box. This was working on another asterisk box so the config on the smartnode seems to be ok but I still not seeing anything at all coming through in the rasterisk console
08:53.06like_a_horsea tcpdump show traffic to the asterisk box from the smartnode so i know its attempting a connection
08:53.20like_a_horsebut i dont see anything in the rasterisk console
08:53.46like_a_horseis there a log file or something i can enable in the rasterisk console to output all sip attempts and tell me why they failed?
08:56.57defsworkturn on sip debug
08:58.43like_a_horsecore set debug .. ?
09:01.36*** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de)
09:05.14qakhan<PROTECTED>
09:14.15*** join/#asterisk tomarch (~tomarch@fbx1.reseau-concept.net)
09:19.25*** join/#asterisk Wiretap7 (~Wiretap@unaffiliated/wiretap)
09:23.39*** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com)
09:25.26*** join/#asterisk AmirBehzad (~behzad@31.184.187.2)
09:27.51*** join/#asterisk s[X] (~mark@ppp118-208-122-13.lns20.bne4.internode.on.net)
09:41.14*** join/#asterisk Azrael808 (~peter@212.161.9.162)
09:54.55*** join/#asterisk Faustov (~fst@gentoo/user/faustov)
09:54.56*** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de)
09:57.15*** join/#asterisk IsUp (~nocturne@unaffiliated/isup)
09:57.23IsUphi
10:12.26kaldemarlike_a_horse: "core set verbosity 10" and "sip set debug on"
10:16.14like_a_horsekaldemar, thanks..
10:24.26qakhanhi, i want to setup sphinx on asterisk. please help me how to do that
10:33.07like_a_horseqakhan, on a asterisk server?
10:34.13*** join/#asterisk Dovid (42570266@gateway/web/freenode/ip.66.87.2.102)
10:34.50like_a_horseqakhan, logistically its not straight forward. Unless you live in Egypt. You realize setting up a sphinx on your asterisk box will most probably squash the server chassis to bits. Really not worth IMHO
10:43.48*** join/#asterisk timahvo1 (~rogue@197.176.0.49)
10:46.49qakhanlike_a_horse i didnt get u
10:48.55qakhansphinx is speech to text application
10:49.09qakhani want to setup om asterisk
10:51.44*** part/#asterisk AmirBehzad (~behzad@31.184.187.2)
10:58.07*** join/#asterisk Cadey (5a980315@gateway/web/freenode/ip.90.152.3.21)
10:58.37*** join/#asterisk LiuYan1 (~liu.yan@211.154.128.171)
10:58.37CadeyHi guys, is there a current list of SIP related RFC's the latest release of asterisk supports/ad-hears to ?
11:03.59*** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com)
11:10.46*** join/#asterisk s[X] (~mark@ppp118-208-122-13.lns20.bne4.internode.on.net)
11:22.36*** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247)
11:32.43*** join/#asterisk s[X] (~mark@ppp118-208-122-13.lns20.bne4.internode.on.net)
11:39.50*** join/#asterisk LiuYan (~LiuYan@222.125.130.16)
11:44.51*** join/#asterisk AmirBehzad (~behzad@87.248.136.181)
11:44.54*** join/#asterisk gordonjcp (~gordonjcp@aramaki.gjcp.net)
11:44.56gordonjcpmorning
11:45.22gordonjcpis anyone here using Cisco 7910 phones with asterisk, and are you prepared to comment on how good or bad they are?
11:51.22*** join/#asterisk sgimeno (~sgimeno@163.117.206.10)
12:01.05*** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134)
12:02.25*** join/#asterisk sezuan (bouncer@irc.scheff32.de)
12:07.35s[X]gordon
12:07.35s[X]hey
12:08.43s[X]s/gordon/gordonjcp
12:10.50gordonjcps[X]: hi
12:11.11gordonjcpincidentally, it's too late now to tell me that 7910G+SW phones are crap, I just bought two ;-)
12:11.19s[X]lol
12:11.32s[X]They actually dont work
12:11.42s[X]There is a bug with that specific model and asterisk
12:11.45gordonjcpI figured that since I'd got hold of the firmware and configured chan_sccp to the point that I can at least see its settings in the asterisk console, I can't go far wrong
12:11.57s[X]No im just messing with ya
12:12.16gordonjcp:-p
12:12.27gordonjcpfor 20 quid I can't really go wrong
12:12.33s[X]I was actually going to ask how well they work as i wanna get myself a pair of cisco phones
12:12.51s[X]but i was thinking the 7941
12:12.57gordonjcpwell a couple of friends of mine are running some of the fancier cisco phones
12:12.57gordonjcpyeah
12:13.18s[X]I have found a place that has them for $99 AUS
12:13.22s[X]brand new
12:13.29gordonjcpnice
12:13.52gordonjcpthat's insane, the cheapest I've seen them here is around £100
12:14.03qakhanhi, i want to setup sphinx on asterisk. please help me how to do that
12:14.11gordonjcpAU$99 is about £63
12:14.14s[X]yeah
12:14.21s[X]its insanely cheap
12:14.24gordonjcpI spent more than that on a curry
12:14.30s[X]http://www.computeralliance.com.au/parts.aspx?qrySearch=7941
12:15.11s[X]only thing i need to check with them is its a 7941 not a 7940
12:16.43gordonjcpoh no!
12:16.48gordonjcpit doesn't have speakerphone!
12:16.55gordonjcpthat's okay, I bloody *hate* speakerphone
12:16.56s[X]yours ?
12:17.05*** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831)
12:17.15s[X]i never use speakerphone
12:17.21s[X]im in a shared office so i guess thats why
12:18.51gordonjcpI just don't like how they sound
12:26.09s[X]SPA3102 seems to be a deceny way to get a single FXO and FXS into asterisk
12:28.00s[X]and by decent i mean cheap
12:31.05*** join/#asterisk sekil (~sekil@78.24.104.73)
12:34.16*** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18)
12:36.24gordonjcpnice
12:36.29gordonjcpI've got an spa2100
12:37.32gordonjcpdual fxs
12:38.51s[X]no fXO ?
12:39.24gordonjcpno
12:42.00*** join/#asterisk danjekins (~danjenkin@62.254.236.250)
12:43.09danjekinsHi, I wondered if anyone could point me in the right direction to fix the safe_asterisk issue I'm having. asterisk will start fine when just running asterisk, but if i run amportal start or service asterisk start etc then safe_asterisk will give me an error 127
12:43.39gordonjcpcan you still use faxmodem cards as FXOs?
12:45.21*** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br)
12:46.48*** join/#asterisk devmikey (~irc@unaffiliated/devmikey)
12:52.14*** join/#asterisk timahvo1 (~rogue@197.177.152.197)
13:00.02*** join/#asterisk diegocn (~diegocn@unaffiliated/diegocn)
13:00.29diegocnhello ppl... how can i do a 'sip reload' from my linux console and not from asterisk console?
13:01.02danjekinsyou should be able to do asterisk -rx "sip reload"
13:01.24danjekinsi think
13:01.26s[X]yeah
13:01.40s[X]spot on
13:02.02diegocnthanks danjekins
13:02.20danjekinsyoure welcome
13:06.29*** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer)
13:08.34*** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36)
13:13.00*** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162)
13:14.47*** join/#asterisk corretico (~luis@201.201.44.82)
13:16.25*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
13:16.39*** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36)
13:22.12*** join/#asterisk patrickod (~Patrick@lysander.patrickodoherty.com)
13:22.52*** join/#asterisk dom| (~domi@mail.tas.de)
13:23.14*** join/#asterisk mjordan (~mjordan@nat/digium/x-qhvvcnzevmmdbfey)
13:29.00*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
13:33.27patrickodI'm having trouble bridging calls betwen two trunks
13:33.49patrickodI've set up call forwarding, and the call dials outbound on the second trunk yet when they connect there is no audio being passed
13:34.50tompawGuys, I have func_curl selected in menuconfig, it's compiled properly, yet asterisk says no function curl available.
13:34.58tompawDo I have to enable it somehow?
13:35.37tompawAaah, nvm, my bad.
13:37.59*** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de)
13:43.16*** join/#asterisk AdvoWork (~AdvoWork@unaffiliated/advowork)
13:46.21*** join/#asterisk Diffen (~diffen@c-a27ce555.042-17-73746f11.cust.bredbandsbolaget.se)
13:48.34kaldemarpatrickod: what technology are you using? SIP?
13:52.46patrickodkaldemar: yep
13:53.14patrickodI can see the calls progressing through a dialplan, they answer correctly etc
13:53.27DiffenHello all. Is it possible to get my asterisk to look in the From header when receving an invite from the pstn? My invite looks like this: http://pastebin.com/CWdqAHDc
13:53.31patrickodbut when it Packet2Packet bridges them neither side gets any audio
13:53.36*** join/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net)
13:54.54kaldemarpatrickod: that's the fault. set directmedia=no under the peers in sip.conf.
13:55.26patrickodkaldemar: ok.
13:58.21*** join/#asterisk irroot (~gregory@196-210-222-7.dynamic.isadsl.co.za)
13:59.16patrickodkaldemar: I've enabled that option on both peers but yet it's still Packet2Packet bridging the two
13:59.51*** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com)
14:00.48[TK]D-FenderDiffen, "core show function SIP_HEADER"
14:05.53*** join/#asterisk serafie (~erin@nat/digium/x-gooltsrawfmfpnlv)
14:06.04patrickodkaldemar: now it's native bridging but still no audio being received
14:07.09JerJermeep meep
14:07.33Diffen[TK]D-Fender: like this page shows. http://www.the-asterisk-book.com/unstable/funktionen-sip_header.html
14:08.39[TK]D-FenderDiffen, Amazing, isn't it?
14:08.49Diffen:D
14:09.20DiffenIts hard to look for something when you dont know the name of the thing you are looking for :D
14:11.47*** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf)
14:12.29*** join/#asterisk timahvo1 (~rogue@41.90.119.198)
14:12.36*** join/#asterisk FinboySlick (~shark@74.117.40.10)
14:15.22FinboySlickAnyone knows of a business-oriented scanner that just pretends to be a fax machine for dumb people who still expect to punch in a few numbers and hit send in a voip infrastructure?
14:15.42FinboySlick(obviously, it has to be asterisk-friendly)
14:15.45*** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek)
14:21.46tzafrirFinboySlick, mailing the result is good enough?
14:22.06kaldemarDiffen: that's why there are "core show applications" and "core show functions" to list them all.
14:22.12tzafrirBut where do you define the address?
14:22.24*** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek)
14:23.22*** join/#asterisk serafie (~erin@nat/digium/x-uqsgnmojsbzzdxli)
14:24.42*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)
14:27.53*** join/#asterisk timahvo1 (~rogue@41.81.248.67)
14:29.34*** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt)
14:32.50*** join/#asterisk alex_ole (~a.olehnov@86.57.158.78)
14:36.19*** join/#asterisk FinboySlick (~shark@74.117.40.10)
14:36.19*** join/#asterisk corretico (~luis@201.201.44.82)
14:36.19*** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj)
14:36.19*** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net)
14:36.19*** join/#asterisk voipeng (~voipeng45@75-150-128-2-Philadelphia.hfc.comcastbusiness.net)
14:36.19*** join/#asterisk felimwhiteley (~quassel@46.7.101.58)
14:36.19*** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110)
14:38.33*** part/#asterisk AmirBehzad (~behzad@87.248.136.181)
14:40.16*** join/#asterisk neurosys (~neurosys@50.20.73.217)
14:43.42*** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net)
14:46.23*** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4)
14:47.59FinboySlickHmm, netsplit...  Anyway.  Are fxs ATAs good enough with T38 now that one can just assume you plug in the fax machine set things up and it'll work?
14:52.03r0m|uFinboySlick, Thats a vague question..... there is so many ata's out there that one does not know which one supports T.38....
14:53.49*** join/#asterisk razu (~razu@2001:ad0:1:1:202:b3ff:fe36:921c)
14:54.52r0m|uThe one I know that support T.38 out of the box will be the new Cisco SPA-112
14:55.33FinboySlickr0m|u: Well, it was more of a general question.  Client is asking:  "will my fax machine work" and expects a yes/no answer.  Is it fair to assume that having T.38 setup properly, any fax machine will work?
14:56.29*** part/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0)
14:57.25r0m|uShould work* Yes. I would say with a 3% to 10% TX/RX failure. In a heavy used env.
14:57.27irrootFinboySlick yes that is the idea but not all devices work equally the SPA-2102 is a winner
14:57.56FinboySlickirroot: That's lucky, it's exactly what we planned to use.
14:58.05*** join/#asterisk jetlag (jetlag@pool-71-168-248-89.cmdnnj.east.verizon.net)
14:58.49irrootFinboySlick timing is critical so best use dahdi timing source the T38 gateway code was developed with testing on 2102 and HP-6500 multifunctional
14:59.02FinboySlickirroot: Just to be clear, T.38 essentially makes sure the analog portion of the fax transmission doesn't have to move beyond the ATA?
15:00.28FinboySlickapologizes, should really be reading that answer on wiki ;)
15:02.08*** join/#asterisk tris (tristan@2001:1868:a00a::4)
15:02.45*** join/#asterisk cbwest (~cbwest@nat/cisco/x-piaqnledfvzjzfln)
15:02.45p3nguinHey, look!  It's that Cisco guy, cbwest, again.
15:04.29irrootFinboySlick asterisk 1.4 allowed T.38 pass through so if your SIP provider allowed T.38 you could fax directly this was extended in 1.6 to allow T.38 termination of faxing via app_fax/res_fax
15:05.21irrootthis has now been extended to allow "conversion" of audio fax "T30" on TDM lines ie PRI/BRI/FXO to T38 ie SPA-2102
15:05.46*** join/#asterisk krotos (~d3v1l@host211-3-dynamic.5-87-r.retail.telecomitalia.it)
15:05.48FinboySlickirroot: This setup will likely involve an SPA-2102 to a SwitchVox, to a MetaSwitch, to the world.
15:05.53krotoshi all guy
15:05.57krotosand girls
15:07.06irrootT30 faxes do not work on ethernet T38 allows this to work so where you need to have a fax machine on the network [T38] and the fax line is TDM this is now supported in asterisk-10
15:07.06FinboySlickirroot: So basically, the SPA-2102 will convert things to T38 and it would ideally stay T.38 all the way to the MetaSwitch.
15:07.44irrootFinboySlick yes this has worked since 1.4 but better support in 1.6+
15:08.34FinboySlickirroot: I should be safe with SwitchVox then...  It's usually fairly up to date.
15:09.49*** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)
15:09.49*** mode/#asterisk [+o malcolmd] by ChanServ
15:10.11bulkorokHi,... can I override allowguest=no for specific client-ips?
15:11.51[TK]D-Fenderbulkorok, No, but yuo can make peers for them with "insecure=port,invite"
15:12.47bulkorokok... thx
15:12.59krotosmy voip server has two public ip, IP_1 and IP_2. Asterisk is bind on IP_1, but now i need to connect to a nortel pbx using IP_2 for sip (type peer).
15:13.17*** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard)
15:13.19krotoshow can i configure the peer , to use IP_2 for outbound traffic?
15:13.48patrickodI've tried setting canreinvite=no in general in sip.conf yet I'm still seeing Packet2Packet bridging
15:15.47r0m|uman Christmas is killing my pocket.... This new toys the kids want are just insane! "Beyblade" 10.00 dollars a pop! :/ lol
15:16.06[TK]D-Fenderpatrickod, What for of *?
15:16.24bulkorok[TK]D-Fender: insecure=invite is enough in my config :-) thx again!
15:16.29patrickodI'm trying to set up call forwarding by dialing outbound on a sip trunk
15:16.38patrickodthe call is made outbound, it dials and picks up
15:16.48patrickodbut there is no audio pased between either end
15:16.51[TK]D-Fenderbulkorok, You're welcome
15:17.04[TK]D-Fenderpatrickod, What version of *?
15:17.07[TK]D-Fender(oops'd
15:17.26*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
15:17.56patrickod[TK]D-Fender: 1.6.2.9-2+squeeze3
15:18.02[TK]D-Fenderpatrickod, you were already told to use "directmedia=no" for this.
15:18.06*** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek)
15:18.10[TK]D-Fenderpatrickod, the parameter changed
15:19.40*** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc)
15:20.34patrickod[TK]D-Fender: I've set directmedia=no in general and still I'm seing Packet2Packet bridging
15:21.07[TK]D-Fenderpatrickod, pastebin your configs (all related portions) and the complete call attempt with SIP debug enabled
15:21.11filePacket2Packet bridging is not direct media
15:21.31filemedia still flows through Asterisk, it's just optimized internally
15:23.16*** join/#asterisk god-x6 (~poorelanc@funtoo/user/godmachine-x6)
15:24.29*** join/#asterisk bmg505 (~leon@196-209-123-3.dynamic.isadsl.co.za)
15:24.57*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
15:25.21patrickodhttp://pastie.org/private/93wwhuzgrzok1lpgpxw7dg the log and config files
15:26.22*** join/#asterisk carrar (tim@osburn.com)
15:27.55patrickodhttp://pastie.org/private/opmkpjtcm01xe1cvtd0rq last bit of the scrollback from sip debug
15:28.03patrickodincludes the start of packet bridging
15:30.35*** join/#asterisk AdamN (~AdamN@63.230.70.254)
15:31.33patrickodfile: [TK]D-Fender any ideas why I'm not hearing audio ?
15:32.24filehave you done a packet capture to confirm the flowing of audio?
15:32.58*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
15:33.19[TK]D-Fenderpatrickod, <--- Reliably Transmitting (NAT) to 216.115.69.144:5060 ---> <-- flowroute is not behind NAT.  Fix your peer
15:33.19*** join/#asterisk akrohn (~akrohn@38.101.60.42)
15:33.36patrickodok
15:34.28patrickodfile: would sip show channelstats suffice ?
15:34.41patrickodit says nothing received or sent
15:34.56*** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk)
15:35.39patrickod[TK]D-Fender: now I'm not hearing a dialtone ?
15:36.04patrickodand audio still isn't flowing
15:38.02*** join/#asterisk ruied (~ruied@po-217-129-252-25.netvisao.pt)
15:38.03*** join/#asterisk ddickenson (~ddickenso@67-198-0-5.static.grandenetworks.net)
15:38.40[TK]D-Fenderpatrickod, Validate each direction independently first
15:38.49[TK]D-Fenderpatrickod, As well as your forwarding.
15:39.01*** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net)
15:39.23[TK]D-Fenderpatrickod, Then PB a new call with SIP debug.  a complete call, not some half-way measure
15:39.37*** join/#asterisk AmirBehzad (~behzad@31.184.187.2)
15:39.53ddickensoncan someone point me to a person I can talk to offline or one on one about large system architecture and best practices, possibly also help me decide if I need to implement something like OpenSER for the size install I'm looking at?  I've done tons of small installs but I'm looking at a 2500+ endpoint install for a Hospital that has to be up all the time and I'm a bit intimidated.
15:41.18patrickod[TK]D-Fender: Calls were worknig both inbound and outbound when I started
15:41.27patrickodit was only the bridging of the two that was not working.
15:43.09ddickensonanyone?
15:45.14patrickodhttp://privatepaste.com/8a6d30d275 full sip debug for the duration of a call
15:46.42patrickodI'm getting a 407 from flowroute ?
15:47.56*** join/#asterisk X-Rob_ (~Rob@eth2083.qld.adsl.internode.on.net)
15:48.34[TK]D-Fenderpatrickod, Is that .... a question?
15:49.06*** join/#asterisk serafie (~erin@nat/digium/x-wnmsxhnpbyayalfy)
15:49.19patrickodI'm confused as to why it would ask, i'm just executing a dial as per normal in the dialplan
15:50.15[TK]D-FenderCall out : Contact: <sip:+16507017829@66.201.49.164>
15:50.31[TK]D-Fendercall in : INVITE sip:16505215946@50.18.181.199 SIP/2.0
15:50.51[TK]D-FenderNot sure if I've mixed IP's here... but is your WAN IP right on your NAT settings?
15:50.58[TK]D-FenderAnd double check your port forwarding.
15:51.03patrickodyep
15:51.05[TK]D-FenderWhat do you have?
15:51.21patrickodI checked the WAN ip was correct in the NAT settings
15:51.26*** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e)
15:51.32patrickodthe IPs there are the 2 different trunking hosts
15:51.44*** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
15:54.23*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
15:54.40*** part/#asterisk AmirBehzad (~behzad@31.184.187.2)
16:00.39*** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby)
16:00.41wcselbyo/
16:00.48[TK]D-Fender<PROTECTED>
16:02.00patrickod[TK]D-Fender: to what extent could iptabels and FORWARD rules kill this ?
16:02.36patrickodI've noticed here that this box is -A FORWARD -j REJECT
16:03.03p3nguinIt's a router?
16:03.22[TK]D-Fenderpatrickod, Nuke and find out
16:04.28patrickodyep it's port forwarding
16:04.33patrickodnot on FORWARD but on INPUT
16:04.47patrickodnow to figure out what ports need to be opened that wernet' already
16:04.48wdoekes2NAT requires FORWARDing between the two nics
16:04.58p3nguinINPUT doesn't forward.
16:05.05wdoekes2PREROUTING does
16:05.08p3nguinINPUT is for things destined for this system.
16:05.19p3nguinTo route, you preroute and forward.
16:05.19wdoekes2look at -t nat
16:05.51p3nguiniptables -t nat -L PREROUTING -nv
16:05.58*** join/#asterisk irroot (~gregory@197.172.193.39)
16:07.08patrickodprerouting policy is accept
16:07.39p3nguinBut if there are no rules, there is no port forwarding through the NAT.
16:07.59patrickodthis is not a router
16:08.24p3nguinWhy were you talking about port forwarding?
16:09.15p3nguinI just love to be given misinformation when being asked for help.  It keeps me on my toes.
16:11.12p3nguinSo... if it isn't a router (when you said it was, and was doing port forwarding), having a FORWARD policy of REJECT is pretty much a moot point.
16:11.26*** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de)
16:11.59p3nguinIf it isn't a NAT router, then the nat table and PREROUTING chains are not relevant.
16:12.00patrickodthe box has iptables rules which seems to have been stopping calls having audio
16:12.10patrickodthe INPUT chain is at fault it would seem
16:12.24p3nguinYeah, INPUT and OUTPUT control what comes into and goes out of that host.
16:12.25patrickodgiving a black -A INPUT -j ACCEPT has solved the silence problem
16:13.03p3nguinINPUT for things destined for that host, OUTPUT for things originating from that host
16:13.33patrickodI have the call live now and I'm trying to figure out which port its using
16:13.36wdoekes2patrickod: look at rtp.conf and the ports there.. open those selectively with -p udp --dport start:end
16:14.09patrickodcool will do
16:18.18patrickodyep that's solved it
16:18.41patrickodI presume the rules let already established connectinos from these hosts inbound
16:20.47*** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net)
16:22.12p3nguinIf you have a rule for RELATED,ESTABLISHED it will.
16:23.26irrootneed to load the state mod also good idea to do contrack
16:23.35p3nguiniptables -I INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT
16:24.56p3nguinHowever, if your INPUT policy is ACCEPT, any rules that don't have a target of DROP or REJECT may never make a difference.
16:24.57irrootif you use contrac [RELATED] then the RTP will be allowed without any other special ruled with the sip contrack helper
16:26.02voipengany way to isolate why when I go to record an AA greeting using *321 it establishes the call, but it doesnt allow me to record anything?
16:26.27p3nguincore set verbose 3
16:26.35p3nguinMake the call.  Show us what happened.
16:26.39voipengk
16:26.53*** join/#asterisk timahvo1 (~rogue@197.176.25.176)
16:28.47voipenghttp://pastebin.com/hPnns28E
16:29.54leifmadsenpermissions issue in folder potentially
16:29.59leifmadsencan't create the file
16:30.09leifmadsenor the directory doesn't exist you're trying to create the file in
16:30.29leifmadsen<PROTECTED>
16:30.38[TK]D-FenderIIRC It should create a file in the full path if there are permissions up to the point where the path doesn't exist
16:30.57leifmadsen^^^
16:31.01voipengleifmadsen: thanks, so i guess i should manually create the directories?
16:31.06leifmadsenobviously
16:31.10voipengthe problem is im tryin to record
16:31.12voipengand make it
16:31.17voipengnot play an existing file
16:31.20leifmadsenthat's not asterisk's problem to resolve
16:31.23voipengheh
16:31.25[TK]D-Fendergo prove what user owns the foders through that path
16:31.25voipengthanks
16:31.45leifmadsenuse STAT() to check if the directory exists, and if not, then use SHELL(mkdir -p /my/path/)
16:31.46voipengk
16:33.33voipengyea it didnt exist, i guess ill try and work with voiceaxis more to see why its not creating, not sure i want to get into making directories each time someone goes to make a new greeting
16:34.05leifmadsenya that is a sysadmin issue -- not an asterisk issue
16:34.13voipengk
16:34.16voipengthank you
16:40.08*** join/#asterisk kriegerod (~krieger@79.135.222.22)
16:42.23*** join/#asterisk jcook_5xdata (~jcook_5xd@173.162.32.1)
16:42.56jcook_5xdatais there a way to stop dynamic creation of meetme rooms
16:43.44p3nguinYes.  Don't use the option that makes dynamic conferences.
16:45.16[TK]D-FenderDoctor, Doctor!  It hurts when I rai.... erm ... nevermind ...
16:45.45jcook_5xdataI am not sure what you mean. I have no reference to meetme in any dial plans, but somehow ghost are creating a room with two people in there
16:46.14jcook_5xdatais there a option in meetme.conf dynamic= no
16:46.48[TK]D-Fenderjcook_5xdata, Can you show us the problem?
16:48.17jcook_5xdatabe a sec I kick the room. I have to wait till it created again
16:48.30*** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247)
16:48.53p3nguinWhile you're waiting for that, pastebin your "dialplan show" output.
16:49.12voipengi know we established my AA recording was a voiceaxis issue, id like to know if there app_record had permissions I can view or eyeball against a working server...
16:49.59[TK]D-Fendervoipeng, it isn't "record".. its the * user
16:50.29[TK]D-Fendervoipeng, You don't compare to others.. you compare to yourself
16:50.33voipenghah
16:50.37*** join/#asterisk mattchis (~mattchis@75-145-122-77-Colorado.hfc.comcastbusiness.net)
16:50.40p3nguin"ps -C asterisk u"
16:50.46voipengi have other pbx that work
16:51.08[TK]D-Fendervoipeng, Again, they might nt be running as the same user.  You should not be comparing these
16:51.14voipengk
16:51.16p3nguin"namei -m /var/lib/asterisk/sounds/aa/lctcap/"
16:51.33*** join/#asterisk LemensTS (~matthew@70.238.163.254)
16:51.50*** part/#asterisk mattchis (~mattchis@75-145-122-77-Colorado.hfc.comcastbusiness.net)
16:52.29LemensTSIf a telco brings a t1 line into a business, do they just run it into a patch panel and send you a t1 router? What I was wondering is how to run the wire from the patch panel to the t1 router. Or does the patch panel usually have a rj45 port?
16:52.36voipengnamei -m /var/lib/asterisk/sounds/aa/lctcap
16:52.37voipengf: /var/lib/asterisk/sounds/aa/lctcap
16:52.37voipeng<PROTECTED>
16:52.37voipeng<PROTECTED>
16:52.37voipeng<PROTECTED>
16:52.38voipeng<PROTECTED>
16:52.40voipeng<PROTECTED>
16:52.42voipeng<PROTECTED>
16:52.44voipeng<PROTECTED>
16:52.44jcook_5xdatap3nguin, here it is http://pastebin.com/PARB4qbH it is very simple
16:52.58p3nguinHey, [tk]d-fender, my car runs fine... I change the oil regularly, run premium fuel, and even wash it when it isn't raining.  Now, can you tell me what's wrong with my truck?
16:53.37p3nguinvoipeng: Dammit, I did it again.  I meant namei -mo /var/lib/asterisk/sounds/aa/lctcap/
16:54.15voipeng<PROTECTED>
16:54.16voipengnamei: invalid option -- o
16:54.16voipengusage: namei [-mx] pathname [pathname ...]
16:54.16p3nguinThis is the second time this month I have done that.
16:54.24[TK]D-Fendervoipeng, PASTEBIN <-
16:54.25[TK]D-Fender~pb
16:54.26infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
16:54.28*** join/#asterisk r0m|u (~wtf@darkstar.rice.edu)
16:54.34voipengsorry! didnt think 5 lines was alot
16:54.47p3nguinIt isn't... it's "a lot"
16:54.47[TK]D-Fender9 <-
16:54.53voipenghahah
16:54.56*** join/#asterisk celord (~celord@201.195.243.194)
16:54.57voipengk
16:55.16p3nguinhmm, well, namei should be able to show the modes and the owners.  I don't know what kind of namei you have that doesn't work right.
16:55.40p3nguinhttp://pastebin.com/Dx1gmhPM
16:56.03voipenghmm
16:56.12voipengyea that is weird, i did it as sudo as well same output
16:56.34p3nguinRunning it as a different user isn't going to change the options it has.
16:57.14p3nguinI'd like to see -mo or -l to see the modes and owners in one go.  Alternatively, you can use ls -dl on each directory down the path.
16:59.19voipengwell im pretty sure the files arent there, so there wouldnt be an owner
16:59.19*** join/#asterisk r0m|u (~wtf@darkstar.rice.edu)
16:59.47p3nguinWhy aren't you creating any dirs that are required but not present?
17:00.01voipenghttp://pastebin.com/hPnns28E
17:00.04voipengit tries to
17:00.04LemensTSDo you just use a normal ethernet cable from T1 jack from telco to cisco router?
17:00.11voipengbut i get app_record.c:272 record_exec: Could not create file
17:00.28LemensTSnm found a cisco room
17:00.40*** part/#asterisk LemensTS (~matthew@70.238.163.254)
17:00.43voipengrollover/flat cable
17:00.44voipengheh
17:00.49p3nguinapp_record.c:272 record_exec: Could not create file /var/lib/asterisk/sounds/aa/lctcap/1_temp   <-----
17:00.58voipengyea..
17:01.08p3nguinIt can't create the file if ANY OF THE DIRECTORIES is not present.
17:01.13voipengoh
17:01.25p3nguinlctcap is not there.  Why haven't you created it yet?
17:01.29voipengguess i totally misinterrprettted that
17:01.44voipengwe never have to manually create these entries
17:01.50*** join/#asterisk brdude (~brdude@12.155.183.30)
17:02.00p3nguinGreat, so your system works and there is no more question.  Enjoy!
17:02.11voipengthat would be an administrative nightmare if everytime someone went to record we would need to make a directory
17:02.12voipenglol
17:02.24r0m|uwhat a day
17:02.28voipengmmhmm
17:03.28p3nguinIf the directory can be created automatically, it will be done as the user which asterisk runs as.  If the parent directory is not writable by asterisk user, it won't work.
17:03.40p3nguinhence my namei -mo request.
17:04.39r0m|uwaz up p3nguin
17:04.44p3nguinSince your namei sucks, check the ownership of the directories to make sure the user can write there.  To see which user runs asterisk, check "ps -C asterisk u".
17:05.05p3nguin(should be "asterisk" in most cases)
17:06.24jcook_5xdataI dont know maybe when I kick the room the ghost got scared and found anything box to live in
17:08.13[TK]D-Fender<voipeng> that would be an administrative nightmare if everytime someone went to record we would need to make a directory <- that's funny.. because your custom dialplan explicitly names that directory.  I
17:09.11voipenghmm? where
17:10.20leifmadsenwe're still talking about this?
17:10.35voipengim talking to the software provider now
17:10.41voipengsorry
17:10.53leifmadsendon't be sorry
17:11.05leifmadsenI just thought it was already resolved as far as asterisk was concerned
17:12.04voipengit is, i was looking for any additional information to give to the software provider in the interim
17:12.13voipengi can certainly post what was the issue once its resolved
17:13.16*** part/#asterisk patrickod (~Patrick@lysander.patrickodoherty.com)
17:13.56p3nguinr0m|u: I got my lab results... Sorry, but I'm not going to be able to donate my lungs this month.  :)  Results: total cholesterol 167, ldl cholesterol 107.4, triglycerides 83, alt 52 (liver), ast 18 (liver), creat 1.5 (kidney).
17:14.25p3nguinPrognosis:  I'M GOING TO LIVE!
17:14.48carrarBetter start hiking every day
17:15.02p3nguinKidneys are still not too good, but they aren't any worse than they were a year ago.
17:15.13p3nguinHiking?  Why?
17:15.25carrarYou need good excersize
17:15.28carrarYou need good excersize program
17:16.15p3nguinI know i need exercise, but my numbers are all good, with the exception of the kidney being a bit out of whack.
17:16.50carrar"I don't need to wear a seat belt! I haven't been in a accident yet"
17:17.26p3nguinI'm not saying I can just let myself go because I have good results.
17:17.28p3nguinNot saying that at all.
17:17.41carrarOh I think you ARE saying that
17:17.46p3nguinBut you came off as I need to ramp up my routine because my numbers are bad.
17:17.47carrarI heard it LOUD AND CLEAR!! :)
17:18.01carrarOh yeah
17:18.03carrarYou do
17:18.09p3nguinBut the numbers are good.
17:18.14carrarYou need to start walking 40 miles aday
17:18.31p3nguinI don't even have time to walk 40 miles!
17:18.36[TK]D-FenderAND I WOULD WALK 500 MORE!
17:18.44carrarBack in my great great great granpa's day they used to walk 40 miles up hill through snow to get to work!
17:18.59[TK]D-Fendermesses with peoples musical sub-conscious ...
17:19.11carrarheh
17:19.15p3nguinup hill BOTH ways.
17:19.19carrarYEAH!
17:19.28[TK]D-Fenderin show up to our eyeballs!
17:19.56*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
17:19.58p3nguinSo, anyway, I'm going to live for a while longer,
17:20.15carrarYou've live to 29?
17:20.16[TK]D-FenderKiddo you don't how good ja got it... in my day, before electricity we had to watch television by CANDLELIGHT
17:20.19carrar(Logans Run)
17:20.59p3nguinI did live to 29.  But I don't remember ever watching that movie.  :/
17:21.49chuckfI know I watched the movie but have blocked most of it out
17:23.40*** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com)
17:25.59*** join/#asterisk timahvo1 (~rogue@197.178.232.65)
17:26.34*** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net)
17:28.34anonymouz666is there a command that cleans the whole astdb?
17:28.41anonymouz666database deltree all
17:28.44anonymouz666something like that
17:28.52anonymouz666or "rm astdb"
17:28.55p3nguinrm /var/lib/asterisk/astdb
17:29.02*** part/#asterisk mjordan (~mjordan@nat/digium/x-qhvvcnzevmmdbfey)
17:29.15p3nguinThen restart asterisk so it gets recreated fresh.
17:29.38anonymouz666alright, that was what I thought
17:36.30r0m|up3nguin: !!!!!!!!!!!! awesome news man.
17:37.11r0m|ur0m|u: test
17:37.37r0m|uthats very good news p3nguin
17:39.13p3nguinYep, he's going to let me live.
17:39.21r0m|u:)
17:40.25*** join/#asterisk bolkin (~bmint@h174.92.190.173.static.ip.windstream.net)
17:45.08r0m|ur0m|u: test
17:45.53*** join/#asterisk Steel_Reign (~steel@207.239.162.198)
17:47.15*** join/#asterisk kikohnl (~kotis@ext-dip-171.hnl.cdsinc.com)
17:50.27r0m|ur0m|u: test
17:54.04r0m|ur0m|u: test
17:54.40*** join/#asterisk r0m|u (~wtf@darkstar.rice.edu)
17:58.23*** join/#asterisk r0m|u (~wtf@darkstar.rice.edu)
17:58.32*** join/#asterisk zyphlar (~zyphlar@wsip-68-14-229-148.ph.ph.cox.net)
17:59.01r0m|ur0m|u: test
18:00.07WIMPyr0m|u: Have you created an endless loop, replying to your own test?
18:00.21r0m|uWIMPy: lol
18:00.52r0m|uI just finish setting up "notification" :)
18:01.12r0m|uWIMPy: And thank you you prove that it worked
18:01.43WIMPyIf only everything was that easy.
18:02.09r0m|u:)
18:09.19*** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6)
18:16.02wcselbyanyone know of any alternatives to navicat that are free / open source that will work with older versions of mysql?
18:19.35*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
18:23.39*** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)
18:23.39*** mode/#asterisk [+o malcolmd] by ChanServ
18:24.22*** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)
18:24.22*** mode/#asterisk [+o malcolmd] by ChanServ
18:24.59*** join/#asterisk serafie (~erin@nat/digium/x-qjzjboyoqktpwrvq)
18:27.56*** join/#asterisk irroot (~gregory@197.174.83.37)
18:28.16*** join/#asterisk jcook_5xdata (~jcook_5xd@173.162.32.1)
18:28.57r0m|up3nguin: I got a reply from freenum.... They did a large import and by mistake they left my account out. In top of that the admin by accidnet overlooked my account and did not approved it causing the issues that we saw yesterday. He said is fixed and should be in their dns shortly.
18:29.04MrTelephonehow do you get directory() working with odbc voicemail configurations?
18:29.48*** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36)
18:30.07jcook_5xdatap3nguin, hey ghost are back. remember I had the ghost creating meetme rooms
18:30.11p3nguinOver 1500 registrants, and he left yours out.  You believed that?
18:30.26p3nguinjcook_5xdata: core show channels verbose
18:30.37p3nguinmeetme list
18:30.43MrTelephoneWhen I switched to the voicemail_users database I cannot dial by first name anymore :(
18:30.56p3nguinAnd I'm still waiting on the output from "dialplan show"
18:31.46r0m|up3nguin: LOL
18:32.39jcook_5xdatahere is the meetme list http://pastebin.com/prJetrjG
18:33.53[TK]D-FenderAnd a channel list
18:33.58[TK]D-Fender"core show channels"
18:34.02jcook_5xdatathey are gone again so I could not get channels... waiting
18:34.13[TK]D-Fenderget the rest
18:34.21MrTelephoneMy database merge didn't work worth the shit
18:34.23MrTelephonefullname is missing
18:35.38jcook_5xdatahere is the core show http://pastebin.com/HvfgbDv7
18:36.16[TK]D-Fenderno, no meetme there so far
18:36.23[TK]D-FenderWaiting on dialplan...
18:37.02jcook_5xdatafrom me?
18:43.59*** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net)
18:44.27jcook_5xdatahere is my full extension.conf       http://pastebin.com/b6CLCcYx
18:44.46*** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36)
18:44.59MrTelephonehmm dial by name is still not working. Is odbc supported for dial by name in asteirsk 1.8.5?
18:46.26[TK]D-Fenderjcook_5xdata, "core show application meetme" <-
18:47.50gordonjcpto paraphrase the old old internet meme, I've lost a phone somewhere in my house, literally lost it
18:48.00gordonjcpI can ring it but I can't work out where the noise is coming from
18:48.30gordonjcpoh, okay, found it, now this is good
18:48.46gordonjcpI've packed it away in a box with some other stuff - while it was still plugged into the fxs
18:48.49jcook_5xdata[TK]D-Fender, just show a man page
18:48.52gordonjcpo_O
18:49.04[TK]D-FenderPB <-
18:50.13Steel_Reignquestion sorry for the noobness but i am new to asterisk. I have already got a server up and  running. my question is that how many calls can i make on one sip line?
18:50.32[TK]D-Fenderno such thing as a "sip line"
18:50.41Steel_Reignok on one sip then
18:51.09[TK]D-FenderYou send calls over SIP.  Typically with auth info tied to an account.  How many calls you are allowed over that account depends on what service you are being offered
18:51.15[TK]D-FenderAnd we don't know
18:51.31gordonjcpSteel_Reign: how many web pages can you open with one internet connection?
18:51.44[TK]D-FenderIt's probably spelled out pretty clearly in the terms of service for whatever you're paying for.
18:52.01Steel_Reigni have not got one yet
18:52.18[TK]D-Fenderwell it depends on what service you pick
18:52.33[TK]D-FenderHow many slices are there in a pre-sliced loaf of bread?  Depends.
18:52.54Steel_Reignso then i have have 1 sip account and have a office of 50 people make outbound calls on that one sip?
18:53.35[TK]D-FenderYou COULD...
18:53.45[TK]D-Fenderagain depends on what kind of service you pay for
18:53.52Steel_Reignunderstood
18:54.17[TK]D-FenderCould be 1, could be 50.  Could be "abitrarily large up to carrier and chain capacity"
18:55.54[TK]D-FenderSteel_Reign, typically you pay for X # of simultaneous channels (calls), X # of unique inbound phone numbers (DID)  to process calls in the ways you want.  Might only be 1 DID with a provider tha lets you call out 50 times
18:56.47[TK]D-FenderSome providers only offer products that resemble analog line capabilities.  2 calls at a time, 1 DID for that link and account, and if you want more, then they come in as different phone #'s
18:57.33Steel_Reign"call out 50 times" does that mean simultaneous connections of 50 individual calls out?
18:58.02jcook_5xdata[TK]D-Fender, ok here is the core show Channel   http://pastebin.com/XmPzKZND You know what I think it is paging intercom? maybe..
18:59.15p3nguinPage() probably uses MeetMe, then.
18:59.41[TK]D-FenderSteel_Reign, 50 simultaneous calls... any combination of in/out really
18:59.54[TK]D-FenderCorrect
19:00.51Steel_Reignok. is a sip required to make calls to other asterisk servers around the world?
19:02.58[TK]D-FenderSteel_Reign, Ok, lets fix up some basics : SIP is a protocol.  Not a "thing".  Like my sending an e-mail.  that is a protocol, but not a specific service or piece of software.
19:03.32Steel_Reignk
19:03.34[TK]D-FenderSteel_Reign, SIP is a protocol for placing calls (voice/video, etc)
19:04.05[TK]D-FenderSteel_Reign, there are hardware phones that use SIP, and of course * uses it as well.  This means that you can certainly call direct from one * box to another.
19:04.06Steel_Reigngot it
19:05.00[TK]D-FenderThere are other protocols : IAX2 (Inter-Asterisk eXchange), H.323 that can also be used, but SIP is the most popular in general, and IAX used in many cases between 2 * boxes
19:05.14[TK]D-FenderThese are all VoIP protocols.
19:05.41[TK]D-Fender* can also use special hardware to talk to a variety of physical telephony lines.
19:07.42Steel_Reignok so a basic asterisk install of asterisk on a server by itself can call any voip system provided that its configure correctly with no third party service providers right?
19:07.50Steel_Reigncall from box to box
19:08.22MrTelephoneI wonder why there is no app_directory in the asterisk source folder?
19:09.13Steel_Reignsorry for all the ?'s. like i said i am new to this. picked it up little more then a week ago.
19:10.24[TK]D-FenderSteel_Reign, Correct.  SIP is SIP.  Any 2 systems or devices can pretty much talk to each toher as long as configured on each side
19:11.48[TK]D-FenderSteel_Reign, And VoIP providers are just 1 way of reaching the PSTN (real world phone system)
19:11.56voipengany suggestions why a number would give me an error: sent into invalid extension 's'
19:12.24[TK]D-FenderSteel_Reign, What vendor and technology is the best choice for you will depend on a number of factors.
19:12.39[TK]D-Fendervoipeng, It's looking for "s".  It doesn't exist, just like it says
19:13.05voipengd-fender: so i need to see if its defined in extensions.conf ?
19:15.04Steel_Reignthanks D-Fender
19:15.40jcook_5xdatavoipeng, yes look at the error it will state what number or ext it looking for then do something like exten => number it looking for,1,answer()  then exen =
19:15.55voipengthanks
19:16.00jcook_5xdata> #,2, whatever
19:16.41MrTelephoneqwell, you around?
19:19.58*** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk)
19:22.57*** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com)
19:24.24*** join/#asterisk justdave (~dave@unaffiliated/justdave)
19:25.27p3nguinIf it is looking for s, it's probably configured incorrectly in the first place.
19:25.46p3nguinNot to mention extension s isn't there, which is pretty normal.
19:26.54p3nguinWhat's the name of the song by Cross Canadian Ragweed where he sings "I got sober, now it's over, I'm back to drinkin' again"?
19:27.21p3nguinor rather "it's back to drinkin' again"
19:28.30[TK]D-Fender<voipeng> d-fender: so i need to see if its defined in extensions.conf ? <- it clearly isn't.  That's the point.
19:28.52voipeng?
19:28.58voipengi fixed it i did an extensions reload from the cli
19:29.02p3nguinerror: sent into invalid extension 's'   <--- it IS NOT defined.
19:29.05voipengso yea
19:29.10voipengthanks
19:29.27MrTelephoneI had hidefromdir=yes by default :(
19:32.02gordonjcpp3nguin: a quick google suggests it's called "Drinkin' Song"
19:32.21p3nguinIt didn't seem very definitive, though.
19:40.41*** join/#asterisk madsage (~sage@io.ioio.com)
19:43.35*** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36)
19:51.38r0m|up3nguin: It's answering now.
19:52.02p3nguinWhat is?
19:52.17r0m|umy frenum dns
19:52.29r0m|us/frenum/freenum/
19:52.39*** part/#asterisk madsage (~sage@io.ioio.com)
19:53.27p3nguinYep, I see the DNS is updated now.
19:53.38r0m|ucool. :)
19:53.45p3nguinAnd it is propagated to me, even.
19:53.53r0m|ucan you try and dial it?
19:53.56r0m|usweet!
19:54.01p3nguin2231003?
19:54.51p3nguinI guess that's it.  I'll try it.
19:55.02r0m|uYes
19:55.19p3nguinIt's a bunch of numbers to dial, I know that much.
19:55.29p3nguin0122231003*xxxx
19:56.02r0m|uyou should get vmail.....
19:56.09r0m|uthat is.
19:56.13p3nguinYep.  Left a message.
19:56.24r0m|ucool.
19:56.38r0m|uwell I think I am going to dial it down to the two digits
19:56.42r0m|uthat is long as shit
19:57.20r0m|uas soon as my brother allowsme ill put in the company info instead of my info....
19:57.28r0m|us/allowsme/allows me/
20:01.07p3nguinOn the SPA 3102, which setting turns up the heard volume of the handset's mic?  Calls from the phone are super quiet and hard to hear the person on it.
20:10.18*** join/#asterisk irroot (~gregory@197.105.107.161)
20:12.33*** join/#asterisk n3hxs (~ed@63.68.135.4)
20:16.28r0m|up3nguin: Port Output Gain?
20:17.29MrTelephoneFXS_Port_Input_Gain                             "-3" ;
20:17.29MrTelephoneFXS_Port_Output_Gain
20:17.42MrTelephonetoo bad it doesn't have a setting for each port
20:17.51MrTelephonethat's from a pap2t config though
20:18.34r0m|up3nguin: Port output and input gain are under Regional.
20:19.07r0m|u"Miscellaneous"
20:20.17r0m|up3nguin: here are som tips for the PAP2 which should also work with the SPA..... Since they are regional they are pretty much standards http://www.freepbx.org/support/documentation/howtos/how-to-set-up-a-linksys-pap2-or-sipura-spa-2000-for-use-with-freepbx
20:21.22*** join/#asterisk cbwest (~cbwest@nat/cisco/x-pqefvpvtpbjogkla)
20:21.28p3nguinHey, look!  It's that Cisco guy, cbwest, again.
20:25.33*** join/#asterisk As001 (~uros@cable-89-216-191-22.dynamic.sbb.rs)
20:26.44r0m|uAs001: how is your hunt for your new system going?
20:32.28MrTelephonewhat mr poppers penguin?
20:37.52r0m|upupers
20:40.23*** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby)
20:40.26wcselby:(
20:40.34wcselbyI just spilled my entire coke on the floor
20:40.53wcselbyit was a good 38 oz left in the cup too
20:42.17*** join/#asterisk cerberus_za (~coert@8ta-151-32-212.telkomadsl.co.za)
20:43.11drudge`holy crap
20:43.19drudge`no drinks or food in the noc, wcselby
20:43.27wcselbylol
20:43.37wcselbyi'm in my office, but the office happens to be in the owner's house
20:44.09wcselbyit was my fault for trying to work through lunch, nothing ever good comes from that
20:47.00MrTelephonewipe the mustard off your shirt bro
20:48.20*** join/#asterisk cerberus_za (~coert@8ta-151-32-212.telkomadsl.co.za)
20:50.42*** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0)
20:50.42*** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36)
20:50.59*** join/#asterisk Ad-Hoc (~nimbus@athedsl-391127.home.otenet.gr)
20:51.18iprouteth0to record custom IVR annoucements in g.729, does it require a codec license?
20:52.58[TK]D-Fenderno
20:53.08[TK]D-FenderNot if that's the codec of the channel recording
20:53.47*** join/#asterisk timahvo1 (~rogue@197.177.128.196)
20:57.14*** join/#asterisk mjordan (~mjordan@nat/digium/x-oojhtbsuikhgditt)
20:57.28*** join/#asterisk serafie (~erin@nat/digium/x-cofwfnipicfbqbge)
20:59.51p3nguinRecord() won't have to translate it through slin?
21:00.10p3nguinI figured it would, like several other apps do.
21:01.31[TK]D-FenderShouldn't
21:01.34MrTelephonemy dialplan is not even funny anymore
21:01.49[TK]D-FenderYou choose G.729 and are G.729 it should dump the packets straight
21:01.58[TK]D-FenderMrTelephone, It was funny before?
21:02.00As001<PROTECTED>
21:02.05MrTelephoneno
21:02.29MrTelephonei pushed everything into macros and it cleaned a lot of stuff up
21:03.19*** part/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0)
21:04.45r0m|uwcselby: screw the owners floor! The coke! You spilled it! sweet nectar of life!
21:05.07wcselbyr0m|u tell me about it
21:05.18r0m|ulol
21:05.39wcselbyi wasn't happy
21:05.42wcselbyheh
21:05.43r0m|ucoke is my every day coffe.... lol we all have owr poisons :P
21:06.11wcselbyit's funny, because I never eat here in the "office", because it feels weird eating in this dude's house by myself
21:06.23r0m|uThat sucks :( Been there though....
21:06.26wcselbybut today I was working on something and was in the groove so I didn't want to put an hour pause into it
21:06.37r0m|uAnd the day you do..... tan tan taaaan!
21:06.40wcselbywhich is what I ended up pretty much doing anyways
21:07.18r0m|u:(
21:07.40wcselbyah well, that'll learn me
21:07.41wcselby:)
21:07.53r0m|u:P
21:08.53*** part/#asterisk Steel_Reign (~steel@207.239.162.198)
21:09.28p3nguinMy FXS port input gain is 0.  Shall I turn it up to 3?
21:10.06p3nguinThe FXS port output gain is -6.
21:10.46r0m|up3nguin: Yes
21:10.59p3nguinI would think input means from the phone's mic, and output would be to the phone's speaker.
21:11.06p3nguinI should turn them both up.
21:11.11r0m|up3nguin: mine is -3
21:11.28p3nguinI'm going to go to 3 and -3.
21:11.30p3nguininstead of 0 and -6.
21:11.38r0m|ucool
21:12.40p3nguinWe'll see if that helps.
21:13.07r0m|ucool
21:19.15*** join/#asterisk s[X] (~mark@ppp118-208-103-144.lns20.bne4.internode.on.net)
21:21.40*** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com)
21:28.22leifmadsenp3nguin: I suggest turned everything to 11
21:29.40r0m|uleifmadsen: That high?
21:29.46r0m|uis it a us standard?
21:29.50leifmadseno.O
21:30.00r0m|ulol
21:30.34leifmadsensomeone hasn't seen This Is Spinal Tap
21:32.12*** part/#asterisk As001 (~uros@cable-89-216-191-22.dynamic.sbb.rs)
21:33.00r0m|uI am curious as of what settings are Optimal for the "Regional Settings" for the US. I had to change quite a few settings to match the US
21:35.07[TK]D-Fenderleifmadsen, amp capo get yours now...
21:35.18[TK]D-Fendercheckout time, BBIAB
21:35.24leifmadsenr0m|u: 11 probably isn't it
21:35.41leifmadsenand I'm sure the value heavily depends on what network you're on, and where you're located
21:36.26r0m|uleifmadsen: I figure after a quick google of "This Is Spinal Tap"
21:36.31r0m|ulol
21:40.15*** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net)
21:40.38r0m|ujust out of curiosity does most of you guys deny msgs? (Not a trick question)
21:41.12*** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey)
21:46.18Qwellr0m|u: I read them, and then I potentially laugh at the person.
21:46.28QwellA lot of people don't like unsolicited messages.
21:46.36r0m|ulol
21:46.41r0m|uI am sure
21:47.04r0m|uI was asking because as of lately I been getting a lot of msgs and I can understand how it can be annoying quick
21:47.20QwellShame them publicly.
21:47.48r0m|unice. I shall learn from that masters!
21:47.54r0m|ubows
21:47.58r0m|ulol :P
21:48.17leifmadsenafter getting the message, I close it, then come to the channel they messaged me from (usually this one), and say, "SomeCrazyHandle: don't message me directly -- ask publicly so other people can also help you and so that others may learn if I choose to respond"
21:48.48Qwellleifmadsen: exactly
21:48.55Qwell~msg
21:48.55infobot(1) Use private messages to the bots to reduce channel spam, but don't message people on #debian without asking permission first.  Most questions should be asked on channel, so that others can benefit from the question and the answers received.  (2) Always feel free to message freenode network staff.  They're the people with hostnames ending in 'staff.freenode'.  (3) Monosodium glutamate, a food additive (see http://truthinlabeling.org/).
21:49.25r0m|ulmao nice
21:49.31r0m|u#3
21:49.38Qwellthat's nothing
21:49.40Qwell~qwell
21:49.40infobotyou are probably a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. qwellcommunicationsinc, the holding company of telcomjoshleifvoxmartinc
21:50.02r0m|uBAUHAHAHA! LOL!
21:50.07r0m|uROFL
21:50.13tzangerwhat the ...
21:50.28tzangerI'm a little upset there's no zangmegacorp in that name
21:50.28r0m|uI cant stop laughing'1!
21:50.30leifmadsentelcomjoshleifvoxmartinc: you buy now!
21:50.59p3nguin~leif
21:51.03p3nguin~leifmadsen
21:51.03infoboti guess leifmadsen is blitzrage
21:51.13p3nguinYa don't say...
21:51.18r0m|uLOL
21:51.20leifmadsen~blitzrage
21:51.20infobot[blitzrage] a super cool fellow
21:51.24pabelangerinfobot: leifmadsen+
21:51.31Qwell~karma leifmadsen
21:51.31infobotleifmadsen has karma of -1
21:51.33Qwell~karma qwell
21:51.33infobotqwell has karma of 11
21:51.35leifmadseninfobot: pabelanger ++
21:51.38QwellGet. Owned.
21:51.39leifmadseninfobot: pabelanger++
21:51.47pabelangerinfobot: Qwell--
21:51.57r0m|uhahaha I had no clue of this....
21:51.59leifmadsen~karma pabelanger
21:51.59infobotpabelanger has karma of 4
21:52.04pabelangerinfobot: tell Qwell his is lame
21:52.05leifmadsenwtf -1?!
21:52.11Qwell~karma blitzrage
21:52.11infobotblitzrage has karma of 1
21:52.13r0m|ulol!
21:52.16tzangerinfobot: leifmadsen --
21:52.22leifmadsentzanger: failsauce
21:52.29tzangerapparently
21:52.29p3nguin-1 and 1 kind of average out to 0.
21:52.35r0m|uhaha
21:52.42tzangermessages the bot to get leif down to a -50
21:52.46pabelangertell me about leifmadsen
21:52.55pabelangertell me about infobot
21:53.01p3nguinIs it nap time yet?
21:53.05leifmadsendivide by zero
21:53.49p3nguinI have drugs; maybe drugs will make it nap time.
21:53.58pabelangerinfobot: hex carrot
21:53.58infobotcarrot is 63 61 72 72 6F 74
21:54.00r0m|ulmao!!!
21:54.22p3nguinMore Flintstone chewable Morphine, please!
21:54.25pabelangerinfobot: karma Qwell
21:54.25infobotqwell has karma of 10
21:54.59r0m|u~wtf
21:55.00*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
21:55.07*** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net)
21:55.19pabelangerinfobot: wtf iirc
21:55.21r0m|uDesc: Interface to the BSD wtf command
21:56.28*** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za)
21:56.28r0m|uall right.... Time to go home! cya in a bit!
21:56.30*** part/#asterisk FinboySlick (~shark@74.117.40.10)
21:57.24QwellQwell++
21:57.26Qwell~karma qwell
21:57.26infobotqwell has karma of 10
21:57.58leifmadseninfobot: Qwell+10
21:58.22p3nguininfobot: qwell++
21:58.32p3nguin~karma qwell
21:58.32infobotqwell has karma of 11
21:58.53p3nguininfobot: qwell--
21:58.56p3nguin~karma qwell
21:58.56infobotqwell has karma of 10
21:59.00p3nguinBAM
21:59.33leifmadsenweirdos
22:00.08p3nguinsmacks leifmadsen around a bit with a rusty iPhone
22:00.20leifmadsenget a new iPhone?
22:01.16p3nguinI wish.  I'd like to have an iPhone 4 CDMA.
22:01.44p3nguinI guess I could just buy one, huh?
22:01.58p3nguinNot sure why I didn't think of that earlier.
22:04.19leifmadsenI'd prefer to not have an iPhone.
22:04.22leifmadsenMission Accomplished!
22:04.33[TK]D-Fenderabhors fruit-based technology
22:04.52carrarJust get a 1way pager
22:07.14gordonjcpyup
22:07.31gordonjcpget an amateur radio licence and homebrew a POCSAG solution
22:07.34gordonjcpsimple
22:09.15gordonjcptweak the receiver in your pager to work on 2m or 70cm depending on if it started life on 153 or 466MHz, knock up an encoder, and get it on the air
22:13.08*** join/#asterisk jkroon (~jkroon@dsl-241-236-19.telkomadsl.co.za)
22:17.24*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
22:17.37*** join/#asterisk edge (~edge@97-64-216-2.client.mchsi.com)
22:17.43*** join/#asterisk nmjnb (~nmjnb@213.114.116.245)
22:17.48p3nguinWhy would you need an amateur license for either of those frequencies?  Neither is in the ham bands.
22:18.15edgeIs there a way for VoiceMailMain() to grab the extension that is calling it? or somehow get that information to the VoiceMailMain() function?
22:18.33citywok${CALLERID(num)} is a good way
22:18.35p3nguinExtensions don't make calls, phones do.
22:18.44citywokassuming the callerid of the phone is the same as the mailbox number
22:18.55citywokotherwise use a database variable to store and look up that info
22:18.59p3nguinAnd phone have caller ID numbers, which can coincide with mail box numbers.
22:19.45p3nguinYou can also use accountcode.
22:21.32edgep3nguin, accountcode?  I assume there is a some "prefered" method for doing this. I mean the phone has a button with a mailbox on it, i assume that it can be used for jumping right to the sets mailbox
22:22.15p3nguinFor my Messages key, it is programmed to call *86 (*VM)...
22:22.36p3nguinAnd *86 runs VoiceMailMain(${CDR(accountcode)}@default)
22:22.54p3nguinbut it could instead run VoiceMailMain(${CALLERID(num)}@default)
22:22.58edgedo I need to set the accountcode in the sip config?
22:23.02p3nguinYes.
22:23.05p3nguinOr callerid
22:23.11p3nguinDepending on which method you choose.
22:23.29*** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
22:23.30p3nguinAnd the accountcode or callerid number will need to be the same as the voice mail box number.
22:24.31edgep3nguin, I think i will do that, see the 'accountcode' in the sip.conf for the extension, then adjust the dialplan accordingly.
22:24.44p3nguinextensions are not found in sip.conf
22:25.26*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
22:26.54edgep3nguin, err i mean peer devices?
22:27.00p3nguinI'm sure you do.
22:27.25edgep3nguin, Key systems are the only think i've ever known, it has been so tough to not think of phones as ONLY extensions
22:27.41p3nguin*shrug*
22:27.47p3nguinTo me, a phone is a phone.
22:29.02p3nguinI really think it's nap time.
22:29.38edgep3nguin, Thanks again for the help
22:31.25akrohnNAPS FOR EVERYONE YAY
22:32.07citywokyea... i'm on can of coke #2... need energy
22:33.28leifmadsenI think of phones as devices rather than extensions
22:33.41leifmadsenthen you apply the concept of an extension to the device
22:34.35citywokleifmadsen: what naming convention do you use for your devices?
22:34.40leifmadsenyou can then abstract a person (or persons) to an extension
22:34.44leifmadsencitywok: mac address
22:34.58citywoki was guessing that was the answer heh
22:35.07leifmadsenbut what about the softphones?!
22:35.17leifmadsenguess what, they utilize a network card that also has a mac address :)
22:35.35citywok:P
22:35.46citywokdo you have people log in to their phones?
22:35.50*** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36)
22:36.17citywokwe simply use extensions that follow the people, and i do the no-no thing and think of phones as extensions :P
22:36.18p3nguinI would only do that for people who do not have their owns desks.
22:37.01citywokif you move to another PC the softphone knows your extension and that becomes you. not by the book but it works :P
22:39.30p3nguinThe phones do *not* know your extension.
22:39.40p3nguinThey know nothink.  nothink.
22:40.04p3nguinExtensions just dial to devices.  The device does not know what extension was used to reach it.
22:41.33p3nguinIf you move to another PC, the soft phone configuration does not change.  You just change the extension in asterisk and that's the end of the move.  Additionally, most people have some extension information associated with each device in sip.conf settings such as callerid, description, accountcode, mailbox, so you'll want to update those as well.
22:41.47*** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za)
22:42.08p3nguinBut the phones themselves need not be altered in any way.
22:42.48citywokwhen an agent launches the softphone it is launched with their personal extension, and that's what the softphone uses to register.  it has no clue that is the extension, but it is :)
22:42.50*** part/#asterisk cbwest (~cbwest@nat/cisco/x-pqefvpvtpbjogkla)
22:43.13p3nguinAgain, phones do not know extension information.
22:43.21p3nguinThey just don't.
22:43.32citywokyes... i'm aware of that. which is why i said "it has no clue that is the extension, but it is "
22:43.33p3nguinAll they know is the user id and the password.
22:43.56citywokthe phone doesn't know that the userid is the extension, nor does it really care.
22:44.16p3nguinWhy would the user ID match the extension number?  That's just... silly.
22:44.52citywokif it works, how is it silly?
22:45.03p3nguinExtension 762 dials a device by the name of 0123AAAFFFF.  The phone only knows that it is a device by the name of 0123AAAAFFFF, not anything to do with the extension.
22:45.24citywokit works, it's fairly logical, it does the job.
22:45.36p3nguinIt's illogical when you have to made adjustments such as the ones we're discussing.
22:46.06p3nguinYou end up with a phone named 1001 with an extension of 1004, and extension 1004 calls a device named 1007, etc.
22:46.09p3nguinall wrong.
22:46.26citywokah, yea, i don't have that problem at all
22:46.31p3nguin~devicenames
22:46.31infobotDevices, extensions, and people should be entirely abstracted.  Extension numbers are applied to people, and people are applied to devices.  This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address.
22:46.35citywokthat would be _craziness_ to manage
22:47.03citywokin hindsight i would have abstracted it. if i could go back a couple years to when i built this system and teach a couple year younger me what i know now it probably would have been helpful.
22:47.35gordonjcpso where do the extension numbers and device names match up?
22:47.45p3nguinThis is why we must educate the next generation to not use device names which match extension numbers.
22:47.46citywokgordonjcp: most people use astdb to look it up
22:47.53p3nguingordonjcp: extensions.conf or some db
22:48.02p3nguinThe association is made in the Dial().
22:48.40citywokmy people are my extensions, and my extensions are applied to my devices directly rather than through a middleman
22:49.08p3nguinAnd then additionally, people associate extension information to the device by using settings such as callerid, mailbox, and voicemail.
22:49.10citywokso i can't hot-desk a physical device with this setup. it requires reassigning the physical devices.
22:49.16p3nguins/voicemail/mailbox/
22:49.42p3nguinwait...
22:49.45p3nguinThat's not right, either.
22:49.48citywokfortunately the limitations i accidentally gave myself haven't been a problem
22:49.51p3nguincallerid, mailbox, and accountcode.
22:50.16citywokgood thing -- it's too late. lol.
22:50.22p3nguinGood thing you don't care to hot desk.
22:50.23[TK]D-Fender[17:49]citywokso i can't hot-desk a physical device with this setup. it requires reassigning the physical devices. <- How so?
22:50.42[TK]D-FenderDevice name and "extension" aren't inherently tied to each other...
22:50.54gordonjcpp3nguin: I am in a worse place for this than being new to it, because I last used asterisk around 1.4 days
22:50.59citywok[TK]D-Fender: yea, they aren't... unless you do like i did...
22:51.11p3nguinI used 1.4 until a month ago.
22:51.19[TK]D-FenderI fail to see why you can't adapt in-place
22:51.20gordonjcpp3nguin: possibly even earlier than that, I'm just going by some config files kicking around
22:51.24p3nguinThe concepts are still the same.
22:51.36citywok[TK]D-Fender: i probably could if i really needed to, at the moment it's "if it aint broke don't fix it"
22:51.39gordonjcpp3nguin: right, but I haven't done it for a couple of years and lots of stuff *has* changed
22:51.49p3nguinNot too much, really.
22:51.58[TK]D-Fendercitywok: Oh... well "don't really care right now" is another matter.  Carry on then...
22:52.00citywokit only takes a couple seconds to move an extension from one device to another in the GUI we built
22:52.07p3nguinA few settings here and there, a few apps' syntax.
22:52.57citywok[TK]D-Fender: That's pretty much what it comes down to.  Risking breaking something to solve a problem I don't have isn't something I want to do.  I don't want to have to tell my boss i was trying to fix a non-issue when i broke everything and took the call center down.
22:53.04*** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com)
22:53.22p3nguinI'd make the change one device at a time.
22:53.30p3nguinBut that's just how I do things.
22:54.30p3nguinToday, Jim's phone is getting a new device name.  His extension will remain the same, but his phone is going to be reconfigured.
22:54.32citywokI'm honestly not sure what all I would have to change to unwind this problem.  The tools that generate the MAC.cfg files, the scripts that handle agent logins/logouts, the dialers
22:54.38p3nguinTomorrow, I'll choose someone else's phone to change.
22:55.03citywokeverything is 100% autoprovisionsd and network driven so I can't really walk around and change one device at a time
22:55.15*** join/#asterisk pietro (~pietro@88-149-227-118.dynamic.ngi.it)
22:55.21gordonjcpis there a standard set of examples for configuring 1.8?
22:55.25citywokit's definitely doable, it would just take some serious planning
22:55.32gordonjcpeverything I've seen seems to use numbers as the username
22:55.33citywok~thebook
22:55.33infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
22:55.35p3nguingordonjcp: There are sample files.
22:55.45p3nguin~device names
22:55.46gordonjcpp3nguin: yeah, but I find them incomprehensible
22:55.47citywokgordonjcp: see the book
22:55.47p3nguin~devicenames
22:55.47infobotDevices, extensions, and people should be entirely abstracted.  Extension numbers are applied to people, and people are applied to devices.  This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address.
22:55.56gordonjcpcitywok: I've no way of getting that
22:56.03p3nguinYou have no internets?
22:56.04citywokit's in pdf format.
22:56.07gordonjcpcitywok: I'm a very long way from a bookshop
22:56.09p3nguinIt's also in html.
22:56.23citywokor that :)
22:56.32gordonjcpoh, okay
22:57.10citywokor ask russel or leif if they have any copies of it left :p
22:57.57p3nguinThe key to this is to comprehend that there is no relationship between phone and extension until you (the admin) creates one through the use of dial plan apps.
22:58.11p3nguinAnd don't tie a phone to a person.
22:58.22p3nguinGive a person an extension.  It will be his extension for life.
22:58.27citywokgordonjcp: listen to p3nguin he speaks the truf.  and as he says, don't make the mistake i did and tie them together. it's a bitch to unwind later on.
22:58.45p3nguinThen associate any random device of your choosing with the extension.
22:59.56p3nguinMy extension is 762.  It will be mine no matter which office I sit in, which phone I use in said office, and no matter how many phones I break and/or dispose of.
23:00.47p3nguinIf I move around often, I'll hot-desk, and I'll have to login when I arrive at any random phone.
23:01.11p3nguinIf I move around only sometimes, I'll change the device that my extension Dial()s.
23:03.34*** join/#asterisk jkroon (~jkroon@dsl-241-236-19.telkomadsl.co.za)
23:07.47gordonjcphm, I only have one adaptor for my ata
23:09.47gordonjcpinvents
23:17.13*** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net)
23:19.13*** part/#asterisk pietro (~pietro@88-149-227-118.dynamic.ngi.it)
23:35.05*** join/#asterisk zerohalo (~zerohalo@74.60.136.128)
23:36.45*** part/#asterisk mjordan (~mjordan@nat/digium/x-oojhtbsuikhgditt)
23:37.55*** join/#asterisk moos3 (~rgenthner@cpe-72-224-121-41.maine.res.rr.com)
23:43.15*** join/#asterisk fisted (~fisted@unaffiliated/fisted)

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.