irclog2html for #asterisk on 20051105

00:00.18cripitomanx i think he have a prob in the kernel hearder or source
00:00.34justinuanyone know snom?
00:00.35cripitoinclude/asm/processor.h:79: error: array type has incomplete element type
00:00.48cripitothe error is in 1 include from the kernel header
00:00.58justinutrying to get the retreive message button to dial the VM extension
00:01.07*** join/#asterisk fugitivo (n=ajf@209.13.242.109)
00:01.52justinui found the page on the wiki, set vmexten
00:02.11*** join/#asterisk freat (n=freat@h-69-3-229-184.chcgilgm.covad.net)
00:02.31justinufor some reason the phone still won't dial it... it shows up in the display as 2999@
00:02.34deezedHey I installe cvs head, can I do a make, make install of a stable and will it replace everything?
00:03.10freatdeezed: I don't think that's recommended. you can delete all the stuff though and then do it
00:03.38deezeduh oh.. there are like 5 dirs in different folders.. better get busy
00:03.44ManxPower<PROTECTED>
00:03.47freatthat's how I've done it anyways... search for all directories w/ asterisk (/etc/asterisk /var/lib/asterisk) and then just delete them
00:04.00freatdelete the src too
00:04.07deezedoh ok.. thats a better idea
00:04.09*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
00:05.49justinusip reload
00:05.52justinuwhops
00:06.09delmarmint. distinctive ring detection all done.. phone and fax calls are handled accordingly.
00:06.24cripitou don't need to delete /etc/asterisk ;)
00:09.43BigJoe1justinu, if you setup an extension for your voicemail as exten => asterisk,1,VoiceMailMain() it will go to the id from the snom phone to voicemail
00:10.31*** join/#asterisk FuriousGeorge (n=ads@pool-151-198-127-128.nwrk.east.verizon.net)
00:10.53FuriousGeorgehi all, i got an easy question for you
00:11.04*** join/#asterisk riksta (n=rick@62.6.163.90)
00:11.12FuriousGeorgeit appears my zap channel start dialing too fast so the first digit (the one) is getting cut off
00:11.13BigJoe1hello FuriousGeorge, there is never an easy question. But go ahead
00:11.26FuriousGeorgeinst there an option in zapata.conf that could fix that?
00:11.26BigJoe1put a www in the dial string
00:11.53FuriousGeorgedial(tech/www${EXTEN})
00:12.07FuriousGeorgelike that?
00:12.17BigJoe1FuriousGeorge, Zap/g0/www${EXTEN}
00:12.54FuriousGeorgeinst there a way i can, one time, in zapata.com to tell it to slow tf down
00:13.01FuriousGeorgerather than edit all my dials
00:13.04BigJoe1depending on how long you need to wait you put a either take w off or add htem.
00:13.13FuriousGeorgew=one second?
00:13.21BigJoe1no .25 ms
00:13.25BigJoe1something like that
00:13.31FuriousGeorge.25s?
00:13.35BigJoe1but no not that I know of in zapata.conf
00:13.49FuriousGeorgeBigJoe1:  thanks
00:13.56FuriousGeorgeill be back in a bit, later
00:13.57BigJoe1FuriousGeorge, yes or .5 don't remember
00:14.07FuriousGeorgek
00:14.08FuriousGeorgelater
00:14.10*** part/#asterisk FuriousGeorge (n=ads@pool-151-198-127-128.nwrk.east.verizon.net)
00:14.21*** join/#asterisk hellop (n=hellop@cpe-70-95-165-136.hawaii.res.rr.com)
00:16.34hellopQuestion about nat=no/yes in sip.conf.   If both phone and * are behind a router do I put yes?
00:16.51BigJoe1nat=yes
00:17.12BigJoe1if there natted but if there on the same lan then it's nat=no
00:18.19hellopgotcha
00:18.35*** join/#asterisk alexns (n=ibtek04@pa-bethlprk-cad2-grp2a-1-167.pittpa.adelphia.net)
00:18.42hellopOnly put yes if there is NAT translation between seperate networks.
00:18.57BigJoe1yes
00:19.49delmari wish ipv4 would just piss off, then all this NAT nonsense would be gone for good.
00:19.51BigJoe1wow been on hold with Cingular for over 20 minutes now.... Argh there slow.
00:20.11BigJoe1delmar, no that is not going to happen.  IP
00:20.14fugitivodelmar: nat is a good thing
00:20.22BigJoe1V4 or IPV6 will not change things
00:20.23alexnsi need some help with qos...
00:20.23delmarnat is a pain in the arse
00:20.26*** join/#asterisk marcus2 (i=marcus@pompeii.outer.org)
00:20.36fugitivodelmar: you don't want your users with public ip
00:20.50*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
00:20.51delmarnat has its uses.. and is good right now where we are still forced to use IPv4 in which address space is limited due to the abuse of the US DoD.
00:21.12delmarfugitivo, im a user and im forced to have 1 public IP and everything else on 1918
00:21.13marcus2so if zttool is unreliable about telling you what a pri span's clock source is, what is a reliable way to tell?
00:21.15delmarthat is bull shit
00:21.49fugitivodelmar: well, i don't want my users or employees to have public ip addresses
00:22.20delmaryou have no idea how much of a problem NAT has been and how much stress and pain in the ass coding has had to be done to get around users wanting to do cool stuff but.. are NATted.
00:22.27justinunat is always a PITA
00:22.37fugitivoi don't have idea? are you sure? :)
00:22.40delmarfugitivo, why?
00:22.50*** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it)
00:22.59delmarfugitivo, if you have a firewall .. as everyone does.. it doesnt matter.
00:23.06BigJoe1delmar, IPV6 has the same nat problems as IPV4 so what is the difference
00:23.14delmarscreened segment topology is no protection.
00:23.18fugitivoBigJoe1: he wants public ip for everyone
00:23.22delmarbut it helps
00:23.27alexnswhat is everyone doing for qos?
00:23.30delmarBigJoe1, yes and no...
00:23.33BigJoe1never going to happen
00:23.53fugitivoalexns: at what level?
00:24.13delmarif IPV6 was in play right now and IPv4 was in the bin... there would be much more address space to hand out and having a real IP on a LAN would be viable
00:24.24alexnson the asterisk box, say sharing voice & a few users across t1 using iax trunk
00:24.39justinudo any sip phones support ipv6?
00:24.42alexnshow can it be implemented in linux
00:24.50delmarpeople are so used to this 1918 address space behind their router/firewall etc... its rubbish.
00:25.01fugitivojustinu: check voip-info, there are nice examples
00:25.02BigJoe1justinu, no
00:25.04delmarjustinu, now that is also a problem
00:25.25justinuanyone know snom360?
00:25.32delmarIPv6 is never going to happen with lazy people like... PA168X software devs.. not adding IPv6 support.
00:25.43BigJoe1justinu, what do you want to know about the 360
00:25.46delmarSnom are wankers.
00:25.46justinuhaving a tough time getting the message retreival button to work...
00:25.52justinui read the instructions on voip-info
00:26.04justinuwhen I press it, snom tries to dial 2999@
00:26.07BigJoe1justinu, I already posted what you need to do
00:26.12justinuoh
00:26.15BigJoe1exten => asterisk,1,VoiceMailMain()
00:26.19justinuoh, i did that.
00:26.24justinuhere's the problem.
00:26.32justinusnom tries to dial asterisk@
00:26.34BigJoe1if it's in the correct context it will work
00:26.36justinusnom says 404 not found
00:26.42iCEBrkrMy hump, my hump... my hump my hump my hump
00:26.43justinubut snom isn't even sending the invite.
00:26.58justinui checked with sip debug, and the snom sip trace.
00:27.02marcus2sigh.  so i just plugged in my first TE410P
00:27.04BigJoe1you need to set the realm as asterisk on the phone
00:27.14marcus2and it doesnt seem to want to sync with our MCI PRI
00:27.15justinuwhere/how?
00:27.35delmarI pulled apart an Snom phone.. found the JTAG port... sent them pics and asked them to confirm that it had JTAG access.. to repair it.. they claimed No.. it was an RS232 that would only be up if the phone was booted... I sent them pics and a diagram of the alteraMax chip they were using.. and told them it WAS a Jtag.. they told me oh.. you are right.. hey.. we will jsut send you a new board free... they sent me a package alright.. a stack of 10 Snom prod
00:27.47BigJoe1justinu, I don't have one here but look at there web site. They have a pdf for setting up asterisk voicemail acces
00:28.02justinui'll take a look
00:28.09delmarso Snom are fucktards.
00:28.33BigJoe1marcus2, what is the error your getting?
00:29.35alexnsfugitivo: qos on fc4 iptables iax & data on t1
00:29.46marcus2bigjoe1; a "RED" error, according to zttool and /proc/zap/1
00:30.21hellopI'm trying to figure out transfering calls to a Zyxel P2000W Wireless phone.  It works sometimes.   Here is CLI output: http://pastebin.ca/27664   For some reason, the P2000W always switches to "UNREACHABLE".  Can someone splain this to me?
00:30.41Starmakerdoes anyone here use a linksys pap2?
00:30.52marcus2i use a pap2 at home
00:31.04marcus2i've had it for about a week, seems to work great... better than the cisco 2600 i had been using
00:31.23hellopIs it correct that a phone will go UNREACHABLE in sip show peers, if there is no communication from/to it greater then the time specified in sip.conf: qualify=x?
00:31.30Starmakerany idea why it doesn't work with certain numbers using the star key
00:31.42marcus2nope, havent tried
00:31.46Starmakercrap
00:31.56marcus2i mean, the only number i've dialed with a * in it was ****
00:32.01Starmakeri'm guessing it's some sort of odd feature in it
00:32.15delmarhellop, qualify=200 for example.. means that if the phone exceeds that in terms of latency... it will go unreachable
00:32.20hellopIf so, it makes sense that a wireless phone would not always want to be sending/receiving.  Is there a way to make * not "qualify" I think is the term.  And just assume the line is working?
00:32.23delmaras in 200ms
00:33.01Starmakermarcus2, im setting up odd services on my asterisk, and i want to use the star key for that :)
00:33.24justinuit's weird, any time I try to dial a URI with snom360, it says "not found", but it's not even sending the invite out.
00:33.27Starmakerlike a queue that I can transfer telemarketers to
00:33.52Starmakerwhich, of course, there will never be any agents on
00:34.01Starmakerand their number is auto-black-listed
00:34.13hellopdelmar, yes that seems to be the case..  The phone always works when I try to dial.  As long as asterisk doesn't think it is "Unreachable" then I can transfer to it.  What I want to do it force it to always say "Reachable"
00:34.35alexnsne1 using <iax> <-> <iax> both trunked no transcoding how many calls over t1 approximately ?
00:34.46justinuhellop: take out the qualify statement in sip.conf
00:35.11hellopjustinu: then sip show peers will always say UNREACHABLE and I can't transfer to it..
00:35.25justinuit should say "unmonitored"
00:35.28justinunot unreachable
00:35.34hellopoh yes
00:35.37sahafeezwhen you reload from the console it drops all the calls right?
00:36.19hellopjustinu,  yeah, that just breaks it.  Can only call out without qualify...
00:36.34justinuum
00:36.36justinuthat's odd
00:36.46justinuqualify just means it sends NOTIFY to "ping" the phone
00:36.48hellopMaximum retries exceeded on call
00:37.22hellopI feel that I've narrowed it down.  The phone works when I get an OK in sip show peers
00:37.50alexnshas anyone come up with a good way to combine multiple *machines for simple call termination
00:37.54*** part/#asterisk BigJoe1 (n=BigJoe1@dsl-20-177.cofs.net)
00:38.14hellopOtherwise, with qualify removed, I'll get behaviour like:  Dial the extension, or transfer, Get Maximum Retry Error, hangup, 10 seconds later the phone rings.
00:38.49justinusounds like a networking problem somehow
00:38.51hellopmaybe Zyxel just needs like qualify =100000
00:39.05justinucan you ping your phone's ip ok?
00:39.12hellopjustinu, good question
00:39.22marcus2so, is anyone else here using PRI cards with * ?
00:39.27tzangeryep
00:39.30hellopjustinu, one thing, I can ALWAYS dial out from the phone.
00:39.31Dr_Ray~yes
00:39.32jbotYou don't say!
00:39.37Dr_Rayer, yes
00:39.39justinuhellop: nat/firewall?
00:40.02hellopjustinu, just a regular local network
00:40.27hellopyou can see the CLI output at: http://pastebin.ca/27664
00:41.07delmaranyone know what the story is with distinctive ring? i thought i had it going good but it seems the detection isn't so great... was detecting 337,0,0 now it got 307,0,0 .. can u specify a range ?
00:41.24justinuhellop: what "maximum retries exceeded means" is that * tried to send your phone a SIP message, and your phone never responded.
00:41.39justinu* resends a number of times (8 times?) and then gives up
00:42.13hellopjustinu, yes, that seems to be the problem.  lik for some reason, my phone stops receiving after a certain time, until I go dial out from it.
00:42.27justinuit's almost like the phone crashes
00:42.56hellopBut it's weird, that I've never not been able to dial out.
00:42.58justinuaccording to your paste, your phone comes back online about 25 seconds after it stops
00:43.08justinui'm wondering if zyxel has problems with SIP Refer
00:43.18*** join/#asterisk yaboo (n=jsirucka@220-245-131-131.static.tpgi.com.au)
00:43.18justinui've got one on the way
00:43.29justinubut haven't had a chance to mess with it yet
00:44.26hellopOne tip, one their Site,  when getting the firmware, don't choose Prestige P2000W,   scroll up to P2000W_V2
00:44.26hellopone=on
00:44.28justinuok
00:45.54hellopyeah, the phone stop being pingable until I dial out from it. Maybe its a feature
00:46.01justinuno, it's crashing :)
00:46.20justinusomehow, you dialing out fixes the IP stack
00:46.33sahafeez5pm. cut time. (saying a prayer)
00:46.38justinugood luck
00:46.44justinudamn snom is making me angry now
00:46.46hellopmaybe I should RMA it..
00:47.18justinuso as soon as you transfer a call to the zyxel, it stops responding to ICMP pings, right?
00:47.31marcus2hrm.  i just plugged an old CSU into the MCI PRI
00:47.34marcus2and its showing a red error too
00:48.03marcus2damn mci
00:48.24hellopno,  if it's working, everything works, ping, transfer...   Then it stops responding on it's own, with no discernable time/event cause.    Sometimes it becomes Reachable on its own.  It always fixes when I dial out.
00:49.20hellopI'll just get a series of UNREACHABLE then REACHABLE msgs in CLI..
00:49.42hellopthe fix is to take out Qualify and don't ever transfer to it..
00:49.43justinuoh, i see
00:49.53justinubut transfer isn't the cause of the problem?
00:50.04hellopjustinu, no
00:50.06justinumaybe the wifi signal just isn't strong enough?
00:50.09hellopit's voodoo
00:50.11deezedAughey: If I use G.726, how do I specify bitrate to use?
00:50.12justinui've noticed that wifi sucks ass
00:50.13marcus2i wish i knew jack about telephony stuff, heh
00:50.26hellopjustinu, no  it's strong.. works great dialing out..
00:50.29*** join/#asterisk XTR (n=xtr@staff-nat.netnation.com)
00:50.31file[laptop]Asterisk only supports G726-32
00:50.33justinuodd.
00:50.36deezedoops Aughey - my nick completer went off
00:50.39file[laptop]it does not support 16, 24, or 40
00:50.42deezedi have cvs head
00:50.53justinuhellop: i'll be experimenting with one myself... i'll be hanging around on this channel too
00:50.54deezedand wiki said it supports the others
00:51.03hellopk
00:51.26deezedso do i say allow=G.726 or allow=G.726-32
00:51.35hellopOne annoying thing, after you dial a number,  the phone will not send any more keypresses, so it can't be used with say, a calling card.
00:51.44file[laptop]deezed: g726
00:51.48deezedthanks file
00:51.56file[laptop]and it only supports what I said...
00:53.04justinuhellop: really? that's pretty stupid.
00:55.04test34I have an X100P card and it look likes asterisk don't detect when I answer the phone, the voicemail starts anyways.. anyway I can fix that
01:04.37marv[work]mahna mahna
01:11.11shido6where is twisted
01:12.30justinusnom 360 sounds nowhere near as good as polycom
01:12.37tzangerjustinu: no?
01:12.45justinuseems more tinny
01:12.56justinupolycom must do something to the audio, because it sounds a lot nicer
01:13.03tzangerpolycom is known for good audio
01:13.35justinueveryone here hates the snom
01:13.50justinumust be a german thing, or something
01:14.16DrukenHMEwho's got an rt31p2 that they can overnight to me??
01:18.04tzangerjustinu: haha
01:19.41sahafeezand it all works!!! yes! thank you everyone that helped!
01:20.02justinunoyce
01:20.21justinuso what do I have to do to get emails with voicemail attachments going out?
01:20.27justinui set attach=yes in voicemail.conf
01:21.12ManxPowerjustinu, did you sacrafice a chicken?
01:21.20justinulol
01:21.21ManxPowerI think that's required,
01:21.32justinui usually stay away from voodoo
01:22.08sahafeezattach=yes
01:22.42sahafeez4121 => 4121,Bawb Bitchen,bawb@helpmepls.com
01:22.49justinudid that.
01:23.03sahafeezhave a mail server that the box can route to
01:23.40justinui suppose i'll have to set up sendmail
01:23.42justinujeeze
01:24.03sahafeezi have sendmail on the localbox. it just worked after setting the stuff in the file
01:24.13*** join/#asterisk wolfson (n=ggggg@usr-kdh-208-6-58-26.beachlink.com)
01:24.15justinui'd think this would have it too
01:24.43justinushouldn't I see something in the full log about it calling sendmail?
01:24.47sahafeezjustinu: i never thought about it. it worked and i assumed it was using the local install of sendmail
01:24.52sahafeezhum. dont know.
01:24.58ManxPowerit does use the local sendmail
01:25.03sahafeezlet me make a all and see
01:25.07justinuoh, ok
01:25.11justinuit is calling sendmail
01:25.13justinunm
01:25.17justinuprobably just a sendmail error
01:25.46snitti've found a bug on digium.com
01:25.48justinuah yes, all the mail is sitting int the queue
01:25.51snitta self referencing 404 error
01:28.20marvmahna mahna
01:30.14justinuyeah, it was working
01:30.22justinubut sendmail isn't set up to automatically send for whatever reason
01:30.23*** part/#asterisk justinu (n=j2@72.18.13.48)
01:30.48sahafeezhey, anyone know how to make the msg light on a ploycom work
01:32.55*** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net)
01:33.59*** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net)
01:34.57snitti went into an infinite loop in a make install of 1.2b2
01:36.37*** join/#asterisk azzie (i=az@cpe-24-168-17-173.si.res.rr.com)
01:40.53*** join/#asterisk fugitivo (n=ajf@209.13.244.213)
01:46.56file[laptop]it's quiet
01:46.58file[laptop]...too quiet
01:47.06fugitivoyeah, mysql is evil...
01:47.16snittmaybe i've found a typo in Makefile
01:47.17file[laptop]linux sucks...
01:47.32QwellSo does file[laptop] :D
01:47.33fugitivoyes
01:47.42file[laptop]Qwell: oh no you didn't!
01:47.54fugitivois codec_ilbc working?
01:47.59fugitivoor should i post a bug?
01:50.41*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
01:50.44sahafeezhttp://www.voip-info.org/wiki-Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk
01:51.04sahafeezcan someone tell me what this part means..
01:51.04sahafeezYou may have to add @context to the mailbox entry. This seems to fix things for many users. Note that this context is the context specified in voicemail.conf for the extension, not the context specified in sip.conf
01:51.15*** join/#asterisk SwK (i=ldtlek@12-219-144-126.client.mchsi.com)
01:51.19tzangersahafeez: easy
01:51.25tzangersahafeez: just as you have dialplan contexts
01:51.29tzangeryou also have voicemail contexts
01:51.33tzangerthe two are *totally* separate
01:51.40tzangerthink of voicemail contexts as companies
01:51.42sahafeezok
01:51.45tzangerextension 500@companyA
01:51.52tzangertotally different from extension 500@companyb
01:51.55tzangerso when you dump someone to voicemail
01:52.00sahafeezso which file are they talking about?
01:52.03tzangeryou need to say which context their voicemail belongs in
01:52.05sahafeezin which file
01:52.06tzangervoicemail.conf
01:52.19sahafeezthey are in the [default]
01:52.34tzangerthen say VoiceMail(${EXTEN}@default)
01:52.40tzangeror however you want it
01:52.45tzangerbut have the @default there
01:52.54sahafeez4205 => 4205,Jennifer Gibson
01:52.57sahafeezfor this line?
01:53.20tzangerno
01:53.23tzangerthat will be
01:53.25tzanger[default]
01:53.31tzanger4205 => 4205,Jennifer Gibson
01:53.32tzangerright?
01:53.35sahafeezyes
01:53.46tzangerso Jennifer GIbson is voicemail box 4205 in the default context
01:53.50tzangerto pint VOicemail to her
01:53.51sahafeezyes
01:53.58tzangeryou say VoiceMail(4205@default)
01:54.00tzangerin the dialplan
01:54.09sahafeezso in extension.conf
01:54.38sahafeezi current have a cache all
01:54.39sahafeezexten => _4XXX,2,Voicemail,u${EXTEN}
01:54.42sahafeezcatch even
01:54.53kuku5is broadvoice donw ?
01:55.21tzangersahafeez: what are you missing there
01:55.23sahafeezso exten=>_4XXX,2Voicemail,u${EXTEN}@default
01:55.25tzangercorrect
01:55.32sahafeezthanks so much.
01:55.35tzangerno prob
01:56.18sahafeezso times your read the doc and it is not clear what they are talking about. that line should read in extension.conf you must add..blablalba
01:56.22sahafeezs/so/some
01:59.27sahafeezis there a reload gracefully command
02:01.58sahafeezwell after all that it did not work.
02:01.59sahafeezdamn
02:02.16tzangerno
02:02.23tzangersome changes require restart
02:02.28tzangeror total shutdown and restart
02:02.31tzangerI forget if voicemail's one of 'em
02:02.43tzangeryo can just extensions reload though and it does not disrupt anything if all you want to do is reload the dialplan
02:06.58QwellDoes a reload of a module stop calls in that module?  ie; chan_sip.so
02:07.15sahafeezno. got. it .needed to setup mailbox=4205@default in sip.conf
02:07.33sahafeezlight when on as soon as i hit reload
02:08.47sahafeezthanks tzanger for you help!
02:09.01*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
02:09.21*** join/#asterisk docelm0 (n=docelmo@71.251.95.2)
02:09.35docelm0whadup?
02:09.46Qwellhttp://www.linuxworld.com.au/index.php/id;754084996;fp;2;fpid;1  nifty
02:11.52sahafeezi know those guys. they said the wifi was so bad at MS when the went to look that MS had marked seats for having the fastest access. seems they deployed their network just like their code, half assed and without a good plan
02:12.28*** join/#asterisk Inv_arp (i=junya@adsl-144-17-25.mia.bellsouth.net)
02:13.56Lostfroglol.. Microsoft IT.. run by Linux.
02:14.43sahafeezheck it was all bayan vines untill nt 3.51
02:14.53sahafeezand billing was on AS/400s untill 98
02:15.23*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) [NETSPLIT VICTIM]
02:15.23*** join/#asterisk Nivex (i=kjotte@user-0c8hq5r.cable.mindspring.com) [NETSPLIT VICTIM]
02:15.23*** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) [NETSPLIT VICTIM]
02:15.59*** part/#asterisk mog_work (n=mogorman@gateway.digium.com)
02:20.07*** join/#asterisk RouterMaN (n=diego@201.240.65.151)
02:20.49*** part/#asterisk RouterMaN (n=diego@201.240.65.151)
02:30.54docelm0say anyone know if you can set accountcode by using the set variable command?
02:31.26Qwelllike Set(ACCOUNTCODE=1234) ?
02:32.12delmarlol my Asterisk compile (HEAD) looped out.. so i do a make clean.. and thats looping also.. beh
02:33.30delmarblow it away and start again. (sigh)
02:33.33docelm0ya.. they have issues
02:33.43docelm0Qwell does SetAccount still work in head?
02:34.04snittdelmar: well
02:34.13snitt[0234] < snitt> i went into an infinite loop in a make install of 1.2b2
02:34.16snitt[0247] < snitt> maybe i've found a typo in Makefile
02:34.29delmarsnitt, lol snap :P
02:34.35delmarsnitt, did u spot it?
02:34.37snittyeah
02:34.49delmarok sec. ill bring it up.
02:34.54snittAsterisk CVS-NHEAD-11/05/05-02:46:49
02:35.06delmardamn
02:35.11Qwelldocelm0: got me...
02:35.13delmarso where was the problem?
02:35.29snittmoment, i'll make a diff on the makefile
02:35.51QwellAren't you supposed to do a make update, instead of cvs up?  Aren't there changes that cause such breakage occasionally?
02:36.18delmardamn it i gotta fix this damn fan control.. cpu gets way to hot and fan takes way too long to get the rpm's going... wierdness.
02:36.41delmarsnitt, throw me a pastebin of the diff or some crap.
02:37.03snitthttp://urbnet.hu/astmake.diff
02:37.14snittapply it when you are in the loop :DD
02:37.25snitti dont know if it works from the clean start
02:38.04snittand yeah
02:38.13snittcustomize back those settings
02:38.14delmari will try it from start.. cant hurt
02:38.20snittlike this: -OPTIMIZE+=-O2 -march=athlon-xp -fomit-frame-pointer
02:38.57snitttry first to replace '.depend: include/asterisk/version.h' with only '.depend:'
02:39.23snittand apply the patch after it when it fails
02:42.39delmarim just lookin thru that diff....
02:43.06delmarim a bit tired but... i just cant see what u are changing that is effecting the loop issue....
02:43.22delmari see the customizations etc.. i dont need all that ...
02:43.27snittit looks like that .depend: line caused all these
02:44.01*** join/#asterisk yeawsing (i=yeawsing@218.50.182.187)
02:44.02delmarah right.. wasnt going down far enuf...
02:44.12delmarbottum of the file...
02:44.30delmaryeah it was failing on the depend if i recall
02:44.50snittmkdep
02:45.01snittand some version.h rm -rf
02:45.05snittand if cmp stuff
02:46.54delmarmight have been fixed already...
02:48.41delmarchecking against the HEAD i just got via CVS and the diff portion at the end there looks the same...
02:48.52snittmmmh
02:49.29*** join/#asterisk BleedingMe (n=Bleeding@adsl-69-227-209-65.dsl.scrm01.pacbell.net)
02:49.56snittMakefile 1.218 2 days
02:50.19delmaryeah snap.. i just looked it up and see that...
02:50.28yeawsingCan I have a comma in context.  Eg context=abc,efg
02:50.48delmaru dont do it like that...
02:50.55Qwellyeawsing: in what context? (pun intended)
02:50.59*** join/#asterisk jmjones (n=jmjones@adsl-61-114-214.sdf.bellsouth.net)
02:51.01yeawsingin sip.conf
02:51.11Qwellno...  What do you want it to do?
02:51.15delmarcontext=onecontext   then in extensions.conf u can include => othercontext
02:51.20*** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net)
02:51.49yeawsingI just try to put a incoming and outcoming for different user.
02:51.57Qwellshavedgoats.net?  mishehu knows how to party
02:52.21delmarie context=siphones   then under [sipphones] have include => othercontext1 and include => othercontext2 etc.
02:53.01snittgn8
02:53.26yeawsingtq
02:54.08delmarsnitt, i just dont see whats different about that last change in the diff.. take a look at the original Makefile ....tell me what u see?
02:56.31delmaroh ok im reading it wrong.. its changing depend: include/asterisk/build.h include/asterisk/version.h .depend defaults.h  to just .depend: include/asterisk/version.h ?
02:57.04delmarie changing line 829 to be like line 832 ?
03:01.04Connoranyone have a script that can read sip.conf and convert it into sql ?
03:02.59delmarwouldnt that be nice :P
03:06.52*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) [NETSPLIT VICTIM]
03:06.52*** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) [NETSPLIT VICTIM]
03:08.22*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
03:09.18*** join/#asterisk Flauto (n=zhao@71.194.38.168)
03:10.09FuriousGeorgewhich one of these zapata.conf options will cause my zap channel to wait a given amount of time after pickup:  start=>, wink=>, txwink=>
03:10.36Qwellcallerid=yes will wait to answer, if you don't have cid on the line
03:10.59FuriousGeorgei do have callerid=yes on the line and i have callerid
03:11.01Qwellor was that not the question?
03:11.24FuriousGeorgemy problem is the thing starts dialing to soon so the first digit is missed
03:11.31delmarFuriousGeorge, on an incomming call this is handled in extensions.conf .. using the Wait command.
03:11.49delmardialing out.. insert w the more w's the longer.. ie wwww
03:11.51delmar:P
03:12.11*** join/#asterisk scuba-steve (n=steve@cpe-071-065-215-219.nc.res.rr.com)
03:12.19scuba-steveDangit, some dingleberry took my nick!
03:12.31Qwellscuba-steve: msg nickserv and recover it
03:12.35FuriousGeorgedelmar:  im only talking about calling out over pots, and id rather not do it to every dial command when im pretty sure i can do it with one line in zapata.conf.  isnt that what start=> does?
03:12.35*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
03:12.41scuba-steveQwell?
03:12.50Qwellscuba-steve: /msg nickserv help
03:12.58Qwellassuming you own the nick
03:12.58scuba-steveSweet :)
03:13.25delmarFuriousGeorge, no this is done in extensions.conf
03:13.48delmarFuriousGeorge, you want a pause on every call going our your pots correct?
03:14.20scubasteveCool.
03:14.27scubasteveI'm back, baby!
03:14.33scubasteveQwell.. thanks :)
03:14.43Qwell~thanks
03:14.43jbotbitte, Qwell
03:14.50Qwellwrong one, heh
03:18.15scubasteveAnyone in here use toll free DID's?
03:18.31scubasteveI've got a question about billing, should be universal...
03:18.35FuriousGeorgedelmar: i want it to pause before it dials.  i know i can do this in extensions but i have four servers which pass variables to macros for outbound calling, which means either i change it in the macro and have it pause even when its a sip or iax call, or find every time i pass zap/g1 and change it to w,zap/g1 or sometheing
03:18.43*** join/#asterisk Blake0PS (n=blake@c-67-190-231-69.hsd1.mn.comcast.net)
03:18.50Blake0PShowdy
03:18.57scubasteveI was under the impression that toll-free account holders paid some sort of fee to telephone operators, or at least pay phone operators..
03:19.09Blake0PShas anyone had it where asterisk can't send a voicemail via email? I have everything setup in voicemail.conf
03:19.09scubasteve... on a per-connection basis.
03:19.24Flautodoes anyone here would help me with setting up my linux machine as a router?
03:19.30Qwellscubasteve: generally, you get charged per minute on incoming calls
03:19.33Flautoi use mandriva 10.2
03:19.34scubasteveBlakeOPS... do you have sendmail or another MTA running on your machine?
03:19.45FuriousGeorgescubasteve: the way it works with mabell is you pay for the call as if you had called them from the "main line" you associate to the toll free
03:19.46Blake0PSscubasteve : I have sendmail on the machine
03:20.13scubasteveQwell, so when they say 2c/min .. it's really 2c/min, no other fees... usually?
03:20.20delmarFuriousGeorge, ie
03:20.29Blake0PSscubasteve : /usr/sbin/sendmail
03:20.30FuriousGeorgeit turns out these wink,start,txwink is only for pri :(
03:20.31delmarexten => _9.,1,Dial(ZAP/g1/www${EXTEN:1},30,T)
03:20.32scubasteveBlakeOPS, let's start by looking at /var/log/maillog or where ever your MTA logs..
03:20.36Qwellscubasteve: unless somebody calls from like a payphone
03:20.45scubasteveQwell, ok.. that's what I thought.
03:20.51FuriousGeorgei.e.?
03:20.52*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
03:20.55scubasteveQwell, then there's a whopper fee.
03:21.10Qwellbut, you can block calls based on ANI2
03:21.18scubasteveSweet :)
03:21.24konfuzedya hoo this 2 x 512MB OCZ DDR 400 is smokin
03:21.27Qwellif they pass it
03:21.45Blake0PSscubasteve : interesting, it shows Connection refused by [127.0.0.1]
03:21.46delmarFuriousGeorge, a line like this for example exten => _9.,1,Dial(ZAP/g1/www${EXTEN:1},30,T)
03:22.04scubasteveBlakeOPS, in linux:   ps axww|grep sendmail
03:22.12scubasteveBlakeOPS.. let me know what you see.
03:22.32Blake0PSscubasteve :  1206 pts/0    S+     0:00 grep sendmail
03:22.39FuriousGeorgedelmar: or in my case, since i pass the zap/g1 in one ARG and the ${EXTEN} in another, i would make that second ARG w${EXTEN}
03:22.43scubasteveBlakeOPS, you don't have sendmail running.
03:22.53scubasteveBlakeOPS, you running linux?
03:22.57Blake0PSit can't just call it as it needs it?
03:23.02delmarFuriousGeorge, adjust to suit your needs :P
03:23.04scubasteveBlakeOPS... if so... /etc/init.d/sendmail start
03:23.24Blake0PSscubasteve : okay let me give that a shot and i'll see what happens, thanks
03:23.27scubasteveBlakeOPS, Asterisk is trying to talk to Sendmail through 127.0.0.1:25 (network socket)... nobody is home.
03:23.28FuriousGeorgewhat is the default?  .5?  i believe every w =,5s no?  or .25?  was talking to someone before about it, cant remember what he said
03:23.42*** join/#asterisk diazonin (i=diazonin@114.222.8.67.cfl.res.rr.com)
03:23.48QwellFuriousGeorge: I think 250ms
03:23.54scubasteveBlakeOPS... If all else fails (from memory here, I don't use sendmail)  /path/to/sendmail -bd -q15m
03:23.59delmarFuriousGeorge, yeah .25secs
03:24.14FuriousGeorgewhats the default?  (no ws)  .25s?
03:24.23QwellFuriousGeorge: no wait
03:24.27scubasteveBlakeOPS ... -bd means become a daemon (detach from consult) -q15m means process the queue every 15 minutes.
03:24.43QwellFuriousGeorge: no w = no wait
03:24.48Qwell1 w = 1 wait
03:24.52Qwellwhere wait == 250ms
03:24.58Blake0PSscubasteve : okay cool. i am on slackware though
03:25.04FuriousGeorgeQwell: thanks
03:25.27scubasteveBlakeOPS... ok... well the sendmail -bd -q15m should work.. don't know anything about how to start it "properly" on your distribution.
03:25.40scubasteveYou can test to see if it's running by typing:
03:25.41delmarFuriousGeorge, whats the cause of the problem anyway... is there something old/slow hanging off the fxo port like an old phone system extension or some device ? most telco's can handle it with no wait at all.
03:25.46scubastevetelnet 127.0.0.1 25
03:25.50scubasteveif you get something like
03:26.02scubasteve250 - Welcome to sendmail, the world's crappiest MTA..
03:26.05scubastevejust type QUIT
03:26.18scubasteveYou'll get connection refused if it's not running.
03:26.36Blake0PSyeah, i get something like that. thanks again scubasteve :)
03:26.41FuriousGeorgedelmar: when dialing out i always mungle the exten to the full 10 digits, so normally, when i dial and it's to quick to get the initial 1 in there, its no problem, but when its an interstate, or 800 call, it doesnt go through
03:26.46scubasteveBlakeOPS.. no problem.. good luck.
03:26.57FuriousGeorgeand i get the "you must dial the full 10 digit number" from the operator
03:27.35delmarsnitt, u around?
03:28.15FuriousGeorgedelmar: now i see what you were asking.  nothing old and slow, just a jack to ma bell -> fxo
03:28.38delmarFuriousGeorge, wierd. dont often hear of ppl having issues like that
03:29.27FuriousGeorgedelmar: at first i thought it was something with the number i was calling the i put an extra 1 in front of the toll free rule in my dialpattern and it went through
03:29.35FuriousGeorgethe i =then i
03:29.39scubasteveHey, is SIP/IAX Tollfree ... $2 setup, $1/mo and 2c/min (presumably 6sec interval) a decent rate?
03:29.48scubasteveWith no commit..
03:30.12FuriousGeorgenufone.org is cheaper if thats what your asking
03:30.19FuriousGeorgenufone.net
03:30.22QwellFuriousGeorge: .net
03:30.25FuriousGeorgegotta run
03:30.33delmardoesnt FWD do free 1800's ?
03:30.41Qwelldelmar: outgoing to 800...
03:30.48delmarah ok
03:30.53delmaru want incomming
03:31.08*** join/#asterisk Icemaann (n=flimpy@204.228.197.139)
03:32.31scubasteveFuriousGeorge .. NuFone is 2c/min .. something about .. if you wish to choose the number $25 fee...?!  Damn.
03:33.04Qwellscubasteve: $25 one time is not bad at all for a vanity DID
03:34.16scubasteveQwell, "vanity DID" ... pick one from a list?
03:34.57Qwella "vanity DID" can be any available number
03:35.08Dr_Raywhy would you pay for that?
03:35.12scubasteveQwell, ah cool.
03:35.17Qwellie; 800-4-la-times  (damn LA times, stealing my freaking number...)
03:35.25scubasteveQwell, hahahaha.
03:35.40Dr_Rayeven though letter phone numbers are harder to dia?
03:35.45Dr_Rayer, dial
03:35.51QwellDr_Ray: not really
03:36.05scubasteveEh, I guess NuFone would save me $1/mo (fee for having a tf #)... I'm pretty damn happy with my provider, so I'll give em the business.
03:36.31scubasteveI've been with them for > 6 months and haven't had a second of outage... ( sellvoip.net )
03:36.45QwellI'm actually still sore about losing that DID...  It just so happened to be in the list of asterlink tollfree numbers, and I happened to spot it...
03:37.04scubasteveQwell, how did they take it from you?
03:37.08Qwellscubasteve: got me
03:37.17Dr_Raywell, they are the la times
03:37.20Qwellbut, it never worked...so, they stripped it from asterlink
03:37.21scubasteveQwell, well.. that's not very nice of them.
03:37.39scubasteveQwell.. I used to live in LA.. LA Times sucks anyway.
03:37.52Qwellscubasteve: eh...it was just the novelty of it
03:37.53file[laptop]our upstream provider gives us a list, and sometimes they give us numbers that are in use (how I want to kill them sometimes)
03:38.00Dr_Rayis LA a 1 or 2 newspaper town?
03:38.02Qwell^
03:38.06QwellThat's what happened.  heh
03:38.20scubasteveSounds like what the idiots from SixTel did to me.
03:38.29scubasteveI had a DID from them and one day it stopped working...
03:38.31Qwellscubasteve: It wasn't asterlink's fault at all.
03:38.32file[laptop]the only way we know is to actually call a toll-free, type the toll-free in, and get the resporg
03:38.43scubasteveEnded up going to some other person's busted-ass AGI app.
03:39.10scubasteveOf course, I had the damn # plastered to all of my paperwork... and of course I was getting billed for the DID even though they routed it elsewhere..
03:39.26Qwellscubasteve: my first nufone did went to another nufone customer...so, they gave me another one.  Then one day, out of nowhere, nufone tried sending a call to that old DID to me...and it's been mine since
03:39.32*** join/#asterisk Sp3ciaL_K (n=alex@d141-139-99.home.cgocable.net)
03:39.41Sp3ciaL_Kw00t got a new powerbook!
03:39.49scubasteveLast I checked, about 6 mos ago.. the support ticket about it was still open and nobody has responded yet... suffice to say I am no longer a customer either..
03:39.59QwellSp3ciaL_K: new powerbook implies you'll be throwing your old powerbook out? ;]
03:40.18Sp3ciaL_Knah i'll keep it
03:40.39Qwellfile[laptop]: I like my new DID from asterlink btw.  I can tell people my number in 6 digits
03:40.56file[laptop]:)
03:41.43*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) [NETSPLIT VICTIM]
03:41.43*** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) [NETSPLIT VICTIM]
03:42.51Qwellactually, I could probably shorten it to 4 digits...
03:43.04scubasteveHeh, I have had an interesting couple of days with my cable modem.. and had * solve it... at least, until the cable modem died..
03:43.18scubasteveEvery hour or two my network would die, resetting cable modem would solve it..
03:43.52scubasteveWrote a cron job that tries to ping a bunch of places.. if it can't get out, would announce network is down on the house intercom and then send some X10 magic to the cable modem ... and reset it.
03:44.00scubasteveTonight... it announced network down and the network stayed down.
03:44.08Lostfroglol
03:44.08scubasteveThe cable modem power supply bit it.
03:44.14Lostfrogowww
03:44.26scubasteveNot too big of a deal, it was 12v 1 amp brick..
03:44.44scubasteveDug around in the magical closet and found my 25 amp 12v power supply I use for ham radio junk.
03:44.54Lostfrog25AMP??
03:44.57LostfrogGeezus.
03:44.58scubasteveEverything is happy again, cable modem isn't going bonkers..
03:45.06scubasteveyeah, it's a honker (about a cubic foot in size)
03:45.12scubasteveBut it will get me through the night...
03:45.27Qwellwake up in the morning with a fried modem
03:45.40Sp3ciaL_Khehe thats cool
03:45.41scubasteveI'm assuming the old power supply was doing bad things to the cable modem in it's final throwes of death..
03:46.12scubasteveI think I'm going to reclaim the monster UPS and use it on something else.
03:46.20scubasteveThe only computer gear in that room is the dorky cable modem.
03:46.28delmaranyone else got HEAD to compile and install ?
03:46.35scubasteveI'm thinking 12v gel cell and a small battery trickle charger to run the cable modem.
03:46.57delmarmake works fine.. make install loops and i can't fix it.
03:47.17scubasteveLostFrog... 25 amps... 100 amps.. no difference to the cable modem.. it only needs 1amp and that's all it will draw.... just will make really big sparks when you short stuff :)
03:47.43*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
03:49.01*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
03:49.30*** join/#asterisk zamsler (n=zamsler@71.194.76.156)
03:49.51jmjoneshow do i record calls made to a certain extension?  i've got this for a dialplan for the extension that I want to record: http://pastebin.com/417881
03:50.09jmjonesbut it never forwards the call on
03:50.57jmjonesand when i have the "Dial()" first, it doesn't start recording...
03:52.23*** join/#asterisk marc324 (n=marc3234@206-248-157-247.dsl.teksavvy.com)
03:52.33marc324does ser/acc module work with mysql?
03:53.12jmjonesand I get an error like this: Nov  4 22:44:59 ERROR[4164]: cdr_sqlite.c:136 sqlite_log: cdr_sqlite: library routine called out of sequence
03:53.52zamslerdoes anyone know of dtmf issues on ubuntu?
03:54.20Qwellzamsler: something like that isn't distro specific
03:54.43zamsleri know, but it works everywhere else
03:54.45zamsler:(
03:55.33zamslerI am baffeled.
03:55.52zamslerso I thought it may be one of the depend packages from ubuntu
03:56.05QwellWhat is the problem?
03:56.07scubastevezamsler... are you having trouble getting * to recognize DTMF?
03:56.11Qwelland don't say "dtmf is broken"
03:56.24zamslerdtmf is not borken.
03:56.26scubastevezamsler... is the channel from PSTN or IP?
03:56.29zamslerit works on my other server.
03:56.31scubasteveb0rken
03:56.32zamslerip
03:56.36zamslerIAX
03:57.06scubastevezamsler... do you have a protocol mismatch between * and the phone?  ie phone sends rfc2833 and * is looking for inband??
03:57.16zamslernope server -> server
03:57.23Qwellisn't all iax dtmf out of band?
03:57.24scubastevezamsler .. interesting.
03:57.25zamslerer phone -> server -> server
03:57.35Qwellzamsler: iax phone?
03:57.38scubasteveQwell, donno.. don't do IAX between servers :)
03:57.42zamslernope sip and sccp
03:57.46delmarmeh. app_rxfax.so undefined symbol blah blah... fax_set_phase_d_handler.. blah blah. Spandsp b0rked :(
03:57.51QwellSo then it isn't an iax problem
03:58.19zamslerI don't think so.
03:58.21scubasteveQwell, I think zamsler has 2 * machines running IAX .. but is using sip/sccp on telephones.
03:58.22Qwellzamsler: check the dtmfmode on the phones, vs that of the asterisk box
03:58.25scubasteveMaybe...?
03:58.36Qwellscubasteve: yeah...would be irrelevant
03:58.56zamslerrfc2833
03:59.04Qwellzamsler: check the phones and the * box
03:59.06scubastevezamsler.. you shouldn't have any problems if everyone is using rfc2833 (phones and *)
03:59.29zamsler:(
04:00.52*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
04:03.37*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
04:05.18zamslerhmm
04:06.13zamslerok.. it is not picking up the # key
04:06.34scubastevezamsler, just the # key??
04:06.38zamsleryeah
04:06.48scubastevezamsler, whoa.. that's interesting.
04:06.55scubastevezamsler, what phone is this??
04:06.55zamslerso far it is everywhere.
04:07.09zamslerI have to wait for the timeout for the input to be taken
04:07.25zamslerhmm
04:07.52scubastevezamsler, what kind of phone are you using?
04:08.06zamslerspa-841 and cisco 7960
04:08.09scubastevezamsler, do you have any other phones ??  (am wondering if the phone is whacky)
04:08.16zamsleryeah 14 of em
04:08.21zamslersame on 6
04:08.27scubastevezamsler, uck on the spa-841..
04:08.30scubasteve7960 running sccp?
04:08.31zamsleri know.
04:08.35zamsleryup
04:08.40zamslervery well
04:08.54scubastevezamsler:  http://www.miselconsulting.com/?page=841
04:08.56Qwellzamsler: chan_sccp?
04:09.10zamsleryeah
04:09.15zamslerthe 11/2 build
04:09.23Qwellever use sccp system message?
04:09.33scubastevezamsler: the 7960's *and* the spa both have the same problem?
04:09.46zamslerQwell, yeah it says get 2 work
04:09.47zamslerlol
04:09.57zamslerscubasteve, problem?
04:09.59scubasteveQwell, is rfc2833 debug info available from the console in * ?
04:10.01Qwellzamsler: When you do that, does it update the phones automatically?
04:10.01scubastevezamsler: dtmf
04:10.06Qwellscubasteve: rtp
04:10.09zamsleryeah
04:10.17scubasteveQwell, that's what I thought.
04:10.34Qwellzamsler: sorry, not automatically...immediately
04:10.49scubastevezamsler, does "rtp debug ip xxx.xxx.xxx.xxx" show any rfc2833 for the "bad" tones?
04:10.50QwellI had to write a patch for it to update all the phones right away
04:10.51zamsleron the next keep alive
04:10.56mog_homeqwell!
04:11.01Qwellmog_home!
04:11.05mog_homehow goes it
04:11.26Qwellzamsler: any idea if it's supposed to wait like that?  I'm considering posting my patch to them (if my boss lets me...)
04:11.30Qwellmog_home: alright
04:11.37Qwellmog_home: how things going with you?
04:11.49mog_homeive seen better days
04:11.56Qwellfair enough
04:11.57mog_homebut it is the end of the day which is good
04:12.03Qwellend of the week too...
04:12.11Qwell(unless you're a sucker, and work Saturdays. ;) )
04:12.13zamslerQwell, I think so as it is treated like MWI, but it will say it after you get off a call also
04:12.25Qwellzamsler: eh, thats lame
04:12.31mog_home28 hours a day
04:12.36mog_home400 days a year
04:12.41mog_homeand lovin it....
04:12.42Qwellzamsler: with my patch, we can do realtime information, like a stock ticker...
04:12.45Qwellmog_home: yeah, I bet
04:13.02Qwellzamsler: but, we haven't tested it with more than one phone registered, heh
04:13.13zamslerlol
04:13.28zamslerok, rtp debug shows the dtmf on the first server, but not on the second.
04:13.40scubastevezamsler, arg
04:13.41Qwellzamsler: both servers are using same dtmfmode?
04:13.46zamsleryeah
04:13.57zamslerit gets the digits and * just not #
04:14.02zamslerthast is wierd.
04:14.10scubastevehm
04:14.15zamslerall my transfer cmds ate #X
04:14.21scubasteveThat sucks.
04:14.24Qwellzamsler: press # twice.  See what happens
04:14.52mog_homeman i need to do something, i am bored, but so tired
04:14.52*** join/#asterisk Nivex (i=kjotte@user-0c8hq5r.cable.mindspring.com)
04:14.55zamslernope
04:14.57zamslerok..
04:15.03*** join/#asterisk sahafeez (n=sahafeez@ip68-7-184-38.sd.sd.cox.net)
04:15.13zamslerhmm
04:15.17zamslerI had an idea.
04:15.19zamslerbut
04:15.41zamsleryeah that is it.
04:15.45zamslerdamn it
04:15.53zamsleri guess it is not that big of a deal
04:16.26Qwellmog_home: you have any idea why chan_sccp isn't a part of *?  Is it because they won't disclaim, or digium doesn't want it?  Combination of both?
04:16.39mog_homei think it is legal
04:16.50mog_homechan_sccp is on shaky legal grand was my theory
04:16.54Qwellahh
04:17.00mog_homeground*
04:17.12mog_homeat least what i heard
04:17.15mog_homeone time
04:17.18mog_homelate at night
04:17.20mog_homein a bar
04:17.22Qwellfrom a guy in a dark alley
04:17.22Qwellheh
04:17.23mog_homeetc
04:17.25mog_homeyeah
04:17.28file[laptop]^.^
04:17.35Qwellhappened to be selling rolax watches
04:17.35mog_homeso is chan_sccp any good?
04:17.38zamslerlol
04:17.40QwellI'm liking it...a lot
04:17.41mog_homeand new a guy
04:17.42zamslercisco got mad again
04:17.45mog_homewho could get me a liver
04:17.55file[laptop]mog_home: Mattttttt how are you
04:18.10mog_hometired and lazy
04:18.19scubasteve<-- fat and happy :)
04:18.21file[laptop]good state to be in
04:18.40mog_homeyup scubasteve
04:19.19*** join/#asterisk desktophero (n=chatzill@ip24-56-30-250.ph.ph.cox.net)
04:19.36desktopherohello all
04:19.51mog_homehi
04:20.11QwellSo, what do people usually do to get the latest cvs when they're behind a restrictive proxy/firewall?
04:20.27mog_homescp
04:20.29scubasteveHey, has anyone played around with an ATA and a fax machine?  I am all ulaw on the network and to the provider.. and call quality is solid and clean... worth trying??
04:20.41Qwellmog_home: only thing that can get out of my work is http, heh
04:20.49mog_homeput ssh on port 80
04:20.53Qwelldoesn't work
04:20.58mog_homei can host it on a webserver for you
04:20.59desktopheroi've been pounding my head against the wall dealing with HDLC errors on my d-channels on a PRI circuit...I have tried EVERYTHING but can not get these errors to go away...anyone overcome these?
04:21.02mog_homeif you ask real nice
04:21.16Qwellmog_home: I've got a server, but...eh, too much effort that way
04:21.17desktopheroI have checked hdparm, IRQs, recompiled Zaptel drivers
04:21.26desktopherough...I am going nuts!
04:21.31mog_homewell if you only have 80
04:21.37mog_homeand ssh doesnt work
04:21.40mog_homethats the way to be
04:21.48mog_home1.2beta 2 is available on web too...
04:21.58Qwellyeah, I know..
04:22.20mog_homei know you know i know you know you know, you know?
04:22.22Qwellmaybe I'll just make a nightly snapshot or something
04:22.31mog_homethat isnt a bad idea
04:22.42*** join/#asterisk ComputerWarm (n=dan@rddrpx22-port-72.dial.telus.net)
04:22.50QwellThat's what I've been doing, but when you've got patches applied...makes it difficult
04:22.55ComputerWarmHello all question i am trying to install h323 the one thats included with asterisk and i keep getting a error with it chan_h323.h:31: warning: `sockaddr_in bindaddr' defined but not used ar cr libchanh323.a ast_h323.o  anyidea what i might be doing wrong?
04:23.10Qwelloh well, heh
04:23.22mog_homeyes
04:23.53shido6:)
04:24.03mog_homeman i really need to work on my jabber just cant bring myself to think
04:24.19Qwellon the plus side - a VERY large mortgage bank might start using Asterisk soon
04:24.33mog_homeheh oh really
04:25.04ComputerWarmsome Sandman Very Large motels in Canada already do
04:35.40delmarscubasteve, re fax over ulaw on LAN, i hear it can be made to work. also look into t.38 etc.
04:35.57scubastevedelmar, i hear t38 is a train wreck
04:36.42delmarscubasteve, still in development really
04:37.01scubastevedetmar, am sending a fax now.. or at least trying...
04:37.32delmarsent at 9600 rather than anything higher that is likely to cause problems.
04:37.36delmarsend*
04:37.53docelm0Anyone know if I can pull the variables DURATION and BILLABLE times out of the dialplan?
04:38.01scubasteveI have sent the same page 2x... will go check email and see if it showed up in my efax
04:39.00delmarcheck http://www.voip-info.org/wiki/view/Asterisk+fax anyway..lots of info
04:40.39delmarjust got spandsp002pre21b working with latest HEAD
04:41.05Qwelldocelm0: cdr vars?
04:41.11QwellCDR(Duration), and such?
04:41.13docelm0yes
04:41.18docelm0I need billsec
04:41.50docelm0I mean I could do a deadagi and pull it that way.. but well thats just too much work.. :)
04:42.27docelm0I havent seen it on asterisk variables wiki.. So I thought I would ask
04:42.28Qwell${CDR(billsec)}
04:42.36Qwelldocs/README.variables
04:42.37docelm0Your shitting me.. NICE!
04:44.34Qwelloh shit, I almost fell backwards in my chair
04:47.48*** join/#asterisk trelane (i=trelane@66.93.203.199)
04:50.09sahafeezSIP is port 5060 only right?
04:50.36wunderkin5060/udp, plus the rtp ports/udp that you defined
04:51.06Qwellwell...technically, I guess you could say yes, but you'd get no audio. :D
04:52.45sahafeezhum. is the rtp ports on the asterisk side?
04:53.43*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
04:57.00wunderkinya it should  be in sip.conf i believe
04:57.15Qwellrtp.conf
04:57.18wunderkinah
04:57.31wunderkini thought so, weird..
04:57.56wunderkinwonder why it wasnt put in sip if its not used for anything else
04:58.25sahafeezok. trying to figure out what to forward thru a firewall to connect to an asterisk box from a sip client on the outside
04:58.37Qwellgenerally 10000-20000
04:59.20sahafeezhum. that is way to much.
04:59.30sahafeezscary number of ports to have open forwarded
04:59.39sahafeezi have like, 2 right now
04:59.48*** join/#asterisk bmg505 (n=leon@rndf-146-51-95.telkomadsl.co.za)
05:01.57Qwellsahafeez: You can change it in rtp.conf
05:02.19sahafeezdoes narrowing the range have any effects on sip
05:03.04Qwellfor the love of god, don't quote me on this, because I'm probably wrong.  With that said...heh.  I think that you can only have 1 channel per port
05:07.19pauldyfor each remote connection to a give ip you can only have one per outbound port but udp is stange because it does not maintain a stateful connection so you port forward just in case the picked port isn't managed properly by your nat device
05:07.48delmarhmmm
05:08.07delmaranyone here know how i can set the station_id and such when sending a fax with spandsp?
05:09.17*** join/#asterisk Prowler1 (i=Prowler@d142-59-42-93.abhsia.telus.net)
05:10.51sahafeezpauldy: yah, thanks. using openbsd. pf. pretty cool firewall
05:11.13pauldynp you may find for your setup you don't need to port forward 0k-20k
05:11.18pauldyerr 10k-20k
05:11.37pauldyand if your registering to a sip provider you may not even need port 5060 forwarded
05:13.16sahafeezi am just trying to get it working from home to the new office asterisk server. i am thinking i will just skip the holes and setup a vpn from my home obsd to work obsd. thanks for the info. it will help with the security planning
05:13.34pauldydon't forget IAX
05:13.58pauldyyou may find it to be even more reliable
05:14.48sahafeezhum. to be honest i do not even know what it is. my 1st asterisk box
05:15.15pauldylook into it on voip-info.org its aix sorry
05:15.36pauldyno sorry IAX
05:15.40sahafeezneeded to go from 4 line analog phones at the family biz and i spent the last 10 years doing tech so looked, sip phones and asterisk.
05:15.46pauldybeen a long day here second guessing myself
05:15.49sahafeezAIX is unix from ibm ;
05:16.14pauldyto many tlas makes for long days
05:16.53sahafeezis AIX = IAX
05:17.35pauldyinter asterisk exchange
05:17.52sahafeezvoip wiki search for AIX goes to IAX
05:17.55pauldynot advanced IBM unix or whatever the other stands for
05:18.27hellopOk, I got outgoing call recording.  Can anyone give any hints on incoming call recording?\
05:18.34sahafeezAdvanced Interactive eXecutive
05:18.52sahafeezworked at I've Been Moved
05:19.51hellopoh, maybe I can just put the monitor before the incomin dial command.  Hopefully monitor will break if it gets transfered to voicemail.
05:20.31pauldysahafeez, they are not equal to my knowlege but I it may be a common mistake
05:20.50moralew
05:22.15sahafeezha!
05:23.29pauldywhatcha haing
05:24.07sahafeezthe wrong window ;)
05:25.05morale22:14 < sahafeez> voip wiki search for AIX goes to IAX
05:25.09moralewoops
05:25.20moralegotta watch that trigger finger
05:25.27sahafeez?
05:27.46hellophmm  Putting monitor command before Dial(SIP...) did nothing.  Is there another easy way to record incoming calls? Like with a flag to dial?
05:28.20pauldyshow application dial?
05:28.54Qwellhellop: You could run a macro from dial
05:29.36hellopshow application   Well, that's helpful.
05:30.03Qwellhellop: I hope you aren't being sarcastic...because, really, it is quite helpful
05:30.05pauldyor you could allow monitoring f the channel created
05:30.19hellopno  I didn' t know that command..
05:30.21sahafeezhellop: asterisk is great but the docs, as far as a nice clean, read this book in order of this are lacking
05:30.31Qwellhellop: k, sounded a bit sarcastic...had to check
05:30.49hellopI thought about that after I typed.. sorry
05:31.29sahafeezhellop: i went nuts trying to figure out how to setup the box. understand it now more so but it was trial and error and asking alot of questions of i want to do this and having someone tell me to seach for XYZ
05:31.29hellopIs ther an ${EXTEN} var on the incoming context?
05:31.51pauldyyea but it is usually just set to s
05:32.00pauldyI believe
05:33.21hellopWell, the Monitor feature in the wiki is slick.  thanks for all the suggestions.  I'll keep playing.
05:34.56pauldynp
05:40.48delmaranyone know anything about the fxotune utility ?
05:41.23sahafeezits a utility to tune fxo lines?
05:41.30sahafeezsorry. in smartass mode
05:41.55delmarsupposedly.. creates a file .. /etc/fxotune.conf
05:42.08delmarso in the file I have 1=11,0,0,0,0,0,0,0,0
05:42.12sahafeezi dont know much but there are only 4 pages of google hits
05:43.31delmari just wanna know what those values are all about is all
05:43.32sahafeezhttp://www.voip-info.org/wiki-Asterisk+echo+cancellation
05:44.39*** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net)
05:45.56delmarso SetVar can now just be Set ?
05:46.18sahafeezhum, dont know. did not read the page. sorry
05:46.33sahafeezi have PRI so..
05:46.45delmarhuh?
05:46.59sahafeezI have PRI so I do not use the fxo cards
05:47.08sahafeezI have a..sorry
05:47.20newmemberno
05:47.42delmarim referring to setting variables... SetVar appears to have been depreciated according to messages im getting... use Set instead...
05:47.46newmemberuse t1 card
05:47.56delmarim using latest HEAD of course...
05:48.06delmarso must be new in the later versions
05:48.24sahafeezah, ok. i have not run across that yet
05:50.42sahafeezsetting up a vpn to work. should be interesting to see how my sparc5 handles the ipsec keys..
05:51.33delmarsparc5.. thats going back a bit
05:51.44sahafeezhave 2 of em
05:51.49delmarsahafeez, i still have a couple of old space IPC's in the shed :P
05:52.00delmarmight even have a sparc1 that works somewhere.
05:52.04sahafeezone is an openbsd firewall the other is just a spare in the closet
05:52.19QwellI love my ss5
05:52.52sahafeezhad a big freebsd box to but to nosey. just bought a terra station 1TB. nice and quiet
05:53.28sahafeezhave an ultra60 2x450. dont do much with it.
05:53.39sahafeez99% of the time i just use the powerbook.
05:57.15Qwellmog_home: you still around?
06:00.19mog_homeyes
06:00.56Qwellanything I can use that does xmpp(xmmp?), and just like...click a name to initiate a SIP call?
06:02.09Qwellgonna try to convince my friends to ditch skype, but need an alternative
06:03.22mog_homeasterisk-im
06:03.30Qwellthat run on Linux?
06:03.47mog_homeyeah
06:03.52Qwellnifty
06:03.52moralewhat version of windows runs linux?
06:04.35Qwellmorale: umm...is that a trick question?
06:05.06moraleim kidding :)
06:05.38Qwellmorale: on a related note...did you see the article link I pasted earlier?
06:05.46moralenope.
06:05.51Qwellheh
06:05.56Qwellhttp://www.linuxworld.com.au/index.php/id;754084996;fp;2;fpid;1
06:06.05*** join/#asterisk matrix05 (i=naderian@217.14.80.215)
06:06.10moraleinterrupting my /. reading
06:06.11matrix05Hiz
06:06.21matrix05mmmmm
06:06.24Qwellvery surprised that isn't on /. yet
06:06.34matrix05does anyone know how to disable VAD in asterisk 1.0.9
06:06.43drumkillaQwell: !!!!!!!!!!
06:06.52Qwellmatrix05: asterisk doesn't support VAD.  You can't "disable it".
06:07.08Qwellmatrix05: You'll need to disable it on the other end
06:07.15Qwelldrumkilla: (delayed reaction)!!!
06:07.19matrix05ouch
06:08.16matrix05am using vonage and it is impossible to disable it , am getthing this error Dropping extra frame of G.729 since we already have a VAD frame at the end
06:08.26moraleqwell, thats nifty - although wireless support in linux is trash still.. well i shouldn't say that, it is improving but still lacks lots of functionality.
06:08.38Qwellmorale: not if you have a good card
06:08.49moraleyeah a prism2, aironet or hermes card.
06:09.17QwellI imagine if a company is dedicating themselves to making linux based wireless devices...they'd do it right
06:09.55matrix05what about this in sip.conf Transmit Silence=YES
06:09.59moraleyeah. linux hardly supports WPA yet
06:10.05moralewep is pointless.
06:12.10matrix05guys what about this error when i use G711 codec
06:12.12matrix05rtp.c:298 process_rfc3389: RFC3389 support incomplete.  Turn off on client if possible
06:12.36Qwellmatrix05: means you need to turn off VAD on the client
06:13.00matrix05:'(
06:14.01matrix05Quintum(H323) -----Asterisk (G711)--- Vonage (SIP)
06:14.04matrix05this is my path
06:14.13matrix05Quintum and vonage must have VAD off?
06:14.18matrix05or just quintum /
06:14.28Qwellvonage likely uses VAD, and likely won't turn it off
06:14.31marc324ne1 know SER ACC here?
06:15.01matrix05yes but in order to make it work . shall both sides have VAD off ?
06:15.06matrix05or just the quintum ?
06:18.31matrix05Qwell what about the silence suppression feature in oss.conf
06:18.33matrix05does it help ?
06:26.08Netgeeksanyone here fairly comfortable with realtime as it stands in head right now?
06:29.27*** join/#asterisk wolfson` (n=ggggg@usr-kdh-208-6-58-26.beachlink.com)
06:30.09*** join/#asterisk Yellow_Fuzzy (n=yellowfu@c211-30-3-75.wavrl1.nsw.optusnet.com.au)
06:30.11Yellow_Fuzzyhi
06:30.36Yellow_Fuzzywould anyone here know of a way to backup the configuration settings on a Linksys PAP2?
06:37.04*** join/#asterisk Tili (n=Tili@211.147.234.5)
06:43.40moraleYellow_Fuzzy: im still trying to figure out how to gain remote access to mine
06:43.49moraleit doesn't support snmp, web-based config..
06:44.33Beirdotry a hammer
06:50.50*** join/#asterisk santiago (n=santiago@208.195.215.124)
06:56.34Yellow_Fuzzymorale: what is yours a Vonage or other provider locked version?
07:01.15Math`some ATAs are only accepting snmp from a specific host
07:08.47moraleYellow_Fuzzy: yeah vonage.
07:10.24moralehttp://www.voipuser.org/forum_topic_1931.html
07:16.03*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
07:22.31*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
07:25.40*** join/#asterisk matrix05 (i=naderian@217.14.80.215)
07:26.44matrix05guys
07:26.56matrix05have u heard of x10 soft phone for sip /
07:33.33*** join/#asterisk colinm_ (n=colol@VDSL-130-13-9-157.PHNX.QWEST.NET)
07:37.22*** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram)
07:37.22*** mode/#asterisk [+o kram] by ChanServ
07:41.48*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
07:45.04*** join/#asterisk rubes (n=scrappy@ip68-4-142-119.oc.oc.cox.net)
07:46.33rubesI deleted all of my Outbound Routes.  How do I add one back in?
07:47.01moraleif you deleted all of them how are we seeing your irc messages?
07:47.18moralejust use the route command, or if you use dhcp just restart dhcp
07:47.38rubesSorry thought this was the Asterisk channel
08:03.10*** join/#asterisk grimse (n=grimse@p5481C37F.dip.t-dialin.net)
08:14.44*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
08:24.41BoRiS-iiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiik]
08:28.10*** join/#asterisk coppice (n=chatzill@125.196.17.210.dyn.pacific.net.hk)
08:37.12SupaplexI think BoRiS fell asleep on his keyboard
08:37.43BoRiS=--fgtr27trjizztjvbtq2222222222222222222222222b3;
08:37.43BoRiS'
08:39.22Supaplexlol
08:45.46*** part/#asterisk yeawsing (i=yeawsing@218.50.182.187)
08:46.06*** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com)
08:46.36jeffikanybody using H323?
08:48.47*** join/#asterisk giesen (i=giesen@CPE0040f404c559-CM00122546417a.cpe.net.cable.rogers.com)
08:49.40jeffikgisen: you in to?
09:00.31*** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net)
09:12.40*** join/#asterisk skyphyr (n=alan@80-195-238-25.cable.ubr01.hari.blueyonder.co.uk)
09:13.00skyphyrZT_CHANCONFIG failed on channel 1: No such device or address (6)
09:13.08Qwellskyphyr: need to config it
09:13.08skyphyroops sorry - meant to type the message first :-)
09:13.18skyphyrahhh thanks Qwell
09:13.33skyphyrI've done zaptel.conf I think
09:13.52skyphyrI just put in fxoks=1 and the zones to uk
09:13.55Qwellztcfg -v, I believe will give more detail to why it's failing
09:14.10skyphyrgreat :-) thanks
09:14.31Qwellmaybe two v's will give more info...dunno
09:14.45skyphyrI'm reading through "Asterisk: The future of telephony" but it goes no suggestions for that error :-)
09:14.55Qwellreally?  That error is quite common
09:15.14skyphyrChannel map:
09:15.14skyphyrChannel 01: FXO Kewlstart (Default) (Slaves: 01)
09:15.14skyphyr1 channels configured.
09:15.23*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
09:15.40QwellWhen did you get that error?  On load of modules?
09:15.52skyphyryeah - the book only mentions Invalid argument
09:16.01skyphyrI got it when running ztcfg
09:16.03skyphyrthe modules load ok
09:16.18Qwellbut it doesn't give that error when you do -v?
09:17.04skyphyryeah - sorry I meant with either -v or -vv
09:17.34QwellYou get the ZT_CHANCONFIG error when you run without -v, but you don't get it with -v or -vv?
09:17.50QwellI'm tired...you'll have to be explicit. :p
09:18.24skyphyrhehe ok sorry :-) the command ztcfg -vv outputs
09:18.25skyphyrZaptel Configuration
09:18.25skyphyr======================
09:18.25skyphyrChannel map:
09:18.25skyphyrChannel 01: FXO Kewlstart (Default) (Slaves: 01)
09:18.25skyphyr1 channels configured.
09:18.27skyphyrZT_CHANCONFIG failed on channel 1: No such device or address (6)
09:18.31Qwellok
09:18.41QwellWhat type of module is it?
09:19.05skyphyrfxs
09:19.29Qwelland you've got "fxoks=1" in /etc/zaptel.conf?
09:19.40skyphyryes :-)
09:20.04skyphyrand the fxs module is in tel1 on the card
09:20.15Qwelltdm400p?
09:20.22skyphyryes
09:20.26Qwellred or green?
09:20.34skyphyrgreen
09:20.48Qwellpastebin.com your zaptel.conf
09:20.59skyphyrok - thanks :-)
09:22.07skyphyrhttp://pastebin.com/418022
09:22.30Qwellyeah, that looks fine
09:23.23QwellWhat happens if you rmmod wctdm and zaptel, then modprobe wctdm again, and run ztcfg?
09:23.42Qwell(or wcfxs if running 1.0.x stable)
09:24.24skyphyrI've removed and reloaded a couple of times already - same result
09:24.32*** join/#asterisk ]data[ (n=data@69.56.182.146)
09:24.35skyphyrI just realised I'm running 1.0.9 - will upgrade to 1.2
09:25.44Qwellcan't hurt
09:26.05Qwellgonna head to bed...good luck with getting that working.
09:26.35skyphyrthanks for your help :-)
09:29.33pooh_skyphyr: change the card's PCI slot
09:29.55pooh_skyphyr: check IRQ (lspci)
09:31.00skyphyris this the one?
09:31.00skyphyr0000:03:11.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
09:32.15pooh_Yup
09:38.40skyphyrgoing for a reboot - brb
09:38.50*** part/#asterisk skyphyr (n=alan@80-195-238-25.cable.ubr01.hari.blueyonder.co.uk)
09:47.04*** join/#asterisk skyphyr (n=alan@80-195-238-25.cable.ubr01.hari.blueyonder.co.uk)
09:49.04skyphyrhi all - I've got an error with my tdm card with a single fxs port - tried on both 1.0.9 and 1.2
09:49.54pooh_skyphyr: did you change PCI slot ?
09:49.58skyphyrmodules load without a problem, but I get this from ztcfg -vv
09:49.58skyphyrChannel map:
09:49.58skyphyrChannel 01: FXO Kewlstart (Default) (Slaves: 01)
09:49.58skyphyr1 channels configured.
09:49.58skyphyrZT_CHANCONFIG failed on channel 1: No such device or address (6)
09:50.01skyphyryou mean physically?
09:50.04pooh_yes
09:50.13skyphyroh - no will take another look at that
09:50.14pooh_in what slotnumber is it now ?
09:50.27skyphyrthe bottom one, but I don't think I can shift it
09:50.48pooh_skyphyr: I encoutered your erro several times
09:51.00skyphyrshifting slot fixed it?
09:51.13pooh_skyphyr: always neede to change PCI slot or resolve a IRQ problem by disabling USB at all
09:51.24pooh_yes, shifting PCI fixed it most of the times
09:51.45skyphyrahhh ok - will try find a way - though I think my graphics card overlaps the only other slot by too much. :-) back soon
09:51.49skyphyrthanks for your help
09:51.52pooh_np
09:52.14*** join/#asterisk fulgas (n=fulgas@209.8.233.169)
09:52.16*** part/#asterisk fulgas (n=fulgas@209.8.233.169)
09:57.12*** join/#asterisk fulgas (n=fulgas@209.8.233.169)
10:00.46*** join/#asterisk potsboy (n=chrisg@c1-113-5.rrba.isadsl.co.za)
10:07.23*** join/#asterisk loick (n=loic@APuteaux-151-1-20-154.w82-124.abo.wanadoo.fr)
10:17.30*** join/#asterisk skyphyr (n=alan@80-195-238-25.cable.ubr01.hari.blueyonder.co.uk)
10:17.41skyphyrno luck with shifting pci slots :-(
10:18.45pooh_hmmm, same error ?
10:20.15skyphyryeah - I just found the error in a mailing list archive
10:20.23skyphyrsomeone asked about plugging power into the card
10:20.31pooh_yeah, it needs power
10:20.31skyphyrI've not plugged any power in
10:20.47skyphyrthat'd be the problem :-)
10:20.52pooh_maybe...
10:20.56skyphyroff to shut down again :-)
10:20.56moraleis it an fxs card?
10:20.59skyphyryes
10:21.12moralethe power just provides the ringtone/ extra 48v
10:21.16moraleit will work without it
10:21.27skyphyroh :-(
10:21.36moraleis it a tdm400p?
10:21.40skyphyryes
10:21.49pooh_skyphyr: try disabling USB being loaded by renaming the modules to .backup and reboot to see what happens
10:21.58moraleare you using cvs head?
10:22.05skyphyrusing beta1
10:22.07moraleand have the 'zaptel' and 'wctdm' modules loaded?
10:22.16skyphyryeah - have both those modules loaded
10:22.42moraleare you using devfsd?
10:23.14skyphyrdevfsd?
10:23.20skyphyrI'm running udev
10:23.24skyphyris that related?
10:23.39moraledid it create all of the device nodes in the /dev/zap directory correctly?
10:23.41moraleit could be.
10:24.09skyphyrI just rmmod ohci_hcd, uhci_hcd and usbhid
10:24.12pooh_skyphyr: what distro are you using ?
10:24.14skyphyrgentoo
10:24.21pooh_kernel ?
10:24.28skyphyrgentoo-sources
10:24.33skyphyr2.6.13 I think
10:24.37moraleps aux | grep -E '(devfsd|udev)' | grep -v grep - should report nothing
10:24.39skyphyryeah - 2.6.13
10:24.52moralei have a tdm400p in this machine and it works fine
10:25.26*** join/#asterisk PakiPenguin (n=pingu@linuxpakistan/admin/pakipenguin)
10:25.30skyphyrudevd
10:25.31PakiPenguinmorning
10:26.07*** join/#asterisk oej (n=Olle@apollo.webway.se)
10:26.14skyphyrshows up when I run that command, morale
10:26.14moraleskyphyr: are you sure your fxs module is on channel 1?
10:26.25skyphyrthat's when it's plugged into tel1 right?
10:26.31*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
10:27.04moralethere should be 4 ports on the pcb board, the one closest to the rj11 ports is channel 1.
10:27.10*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)
10:27.15moraleit could just be your zaptel.conf
10:27.35skyphyryeah - pretty sure that's where it is
10:27.58moraletry changing your /etc/zaptel.conf, add the line fxoks=4
10:27.59skyphyrmy zaptel.conf is all comments except for fxoks=1 and the zone info
10:28.08skyphyralrighty - will do thanks
10:28.12moralethen run ztcfg -vvv
10:28.27moralejesus i need some coffee.
10:29.11skyphyronly has one error message :-)
10:29.26skyphyrthat means 4 is correct right?
10:29.39moraleeh?
10:29.50skyphyrI have both the 1 and 4 line
10:30.13skyphyroh wait
10:30.14moraleoh.. yeah take the fxoks=1 out then
10:30.14skyphyrnope
10:30.20skyphyrremoves it so it only had channel 4
10:30.21skyphyrsame error
10:30.37moralenext thing i guess is try plugging power into the card.
10:30.47moralemaybe the modules aren't getting any power
10:30.47PakiPenguinlol
10:31.12skyphyrthere's no power plugged into the card yet
10:31.18skyphyroh wait
10:31.21skyphyryou mean through pci
10:31.27moraleyes.
10:31.30voipjoyanybody can advice why i can't compile zaptel driver http://pastebin.ca/27578 ?
10:31.35moraleit has a 5.5v plug on it
10:31.56skyphyrhmmm ok - I'll shutdown and check the card
10:32.05skyphyrthanks for your help :-) back soon
10:32.11moralevoipjoy: use a newer kernel possibly.. 2.6.14 is the latest kernel.
10:32.25moralealso i haven't seen anyone using amd64 with asterisk yet.
10:32.40skyphyrI'm amd64 :-)
10:32.48skyphyrthough your statement still holds true
10:32.49PakiPenguinlol
10:32.54moraleah.. well we will see when skyphyr comes back.
10:33.07voipjoymorale: i have one in the office dual amd 2.6.13.4 #2 SMP Thu Oct 20 22:01:23 IST 2005 x86_64 GNU/Linux
10:33.26skyphyrbrb
10:33.37moraleyeah my workstation is an amd64.. i would never run linux on it, all the hardware is too unsupported yet.
10:35.22Madkisshm.
10:35.44MadkissIs some asterisk-developer now available? I need to talk to one urgently.
10:36.05moraleabout what?
10:36.49MadkissI may not talk about that.
10:37.12PakiPenguinrofl
10:37.54MadkissYou're laughing; I'm sure enough you would not laugh anymore once you hear what I am up to.
10:38.53PakiPenguinMadkiss: share it up!
10:39.09MadkissYou're all so funny, really.
10:39.12coppicewhat's the point of finding a developer if your problem is a secret? :-\
10:39.22MadkissI don't *have* a problem
10:39.31PakiPenguinhaha
10:39.33pooh_you're making one ?
10:39.33PakiPenguin:)
10:39.35MadkissI just said that I urgently need to talk to a developer, and that phrase is still valid.
10:39.47voipjoyinstalling 2.6.12 apt-get install linux-headers-2.6.12-1-amd64-generic linux-image-2.6.12-1-amd64-generic
10:39.48coppiceso talk
10:39.50MadkissNot everytime somebody needs to talk to an asterisk developer, he has a problem
10:40.10pooh_Madkiss: security related ?
10:40.23MadkissNot at all
10:40.29MadkissIt's *not* *a* *problem*
10:40.44coppiceso talk
10:41.27PakiPenguin:) lol
10:41.58moraleugh.. 4 more hours to go then its home time
10:42.00coppiceso, you are looking for a developer so you can call them rude names? :-\
10:42.23pooh_morale: nightshift ?
10:42.28PakiPenguinmorale : what work allows irc?
10:42.30PakiPenguin:p
10:42.31moralepooh_: yep.
10:42.37pooh_morale: ewwww
10:42.43moralei work at a datacenter.
10:42.55PakiPenguinoh :)
10:43.27coppicewhat is a datacentre? is that like the middle few bits of each word? :-)
10:43.32moraleno pengiuns here, just billy gates dolls.
10:44.03PakiPenguinoh
10:44.04*** join/#asterisk skyphyr (n=alan@80-195-238-25.cable.ubr01.hari.blueyonder.co.uk)
10:44.11skyphyrthanks all :-)
10:44.38PakiPenguinmorale: windows :o :o /me thinks morale might need hair transplant soon
10:44.42moraleim scared to use our coffee maker, last week we bleached it because it was growing mold.
10:44.47skyphyrhehe I think I owe the capuccino
10:44.57pooh_skyphyr: speak up :-)
10:45.04PakiPenguinskyphyr : power connector was it ?
10:45.06skyphyrI'm not sure whether it was from plugging in the power
10:45.17skyphyror because I removed and reconnected the fxs module
10:45.21skyphyrbut I've got no more errors
10:45.26skyphyrand it shows up in zttool
10:45.29pooh_skyphyr: no config changes ?
10:45.38skyphyrno config changes
10:46.18pooh_skyphyr: unplug the power and try again, then we know for sure
10:46.27pooh_the power to the card that is :-)
10:46.31skyphyryeah - I really should shouldn't I
10:46.42skyphyrhehe alrighty - it's the least I can do to say thank you :-)
10:46.44skyphyrback soon
10:46.47pooh_:-)
10:47.09moraleim still trying to find out how to dialout with my fxs card.. whenever i dial a number i get static.
10:47.15moralei can recieve calls fine though
10:47.24*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
10:47.27moralestory of my life.
10:48.04pooh_morale: I have the same witha setup of a friend of mine
10:48.10moralemaybe he went to #openpbx and wants to find a developer.
10:48.15pooh_morale: inbound fine, outbound nothing
10:48.26moraledo you get loud static?
10:48.42moralei think it is the slinear protocol it is using
10:48.44pooh_morale: nope, nothing, the CLI says ringing but nothing....
10:48.57Madkisspooh_: No, I just found an asterisk-developer :)
10:49.07pooh_Madkiss: Good for you!
10:49.26pooh_Madkiss: and thanks for sharing ;-)
10:49.39MadkissYou will all get to know about it soonishly anyway.
10:49.49MadkissIt's just that I can't say more right now
10:50.03*** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net)
10:50.12pooh_Madkiss: ok, I guess you have your reasons
10:50.28moraletopsekret nasa stuff
10:50.57Madkissabsolutely t0psekr1t, yes.
10:51.06pooh_morale: this setup I was talking about lives in .aus any specifics I overlooked for that reason you think ?
10:51.07Madkiss70ps3kr1t, so to say.
10:51.07PakiPenguinoh no , ;)!
10:51.33pooh_Madkiss: nadastrovja komrad
10:51.33moralepooh_: when making an outbound call do you see it make the outbound call?
10:51.40pooh_morale: yes
10:51.46Madkisspooh_: Nasdorowje?
10:51.56pooh_Madkiss: even more secret ;-)
10:52.33moralepooh_: aus uses dtmf right?
10:52.38pooh_yes
10:53.35pooh_the CLI shows 'ringing zap/etc' and show channels show the channel up
10:53.52pooh_but nothing really happens to the called party
10:54.03Madkisspooh_: You know Pilzener?
10:54.05moralepooh_: i would suggest turning off all of the call features in /etc/asterisk/zapata.conf and enabling them one by one
10:54.27moralepooh_: can you make outbound calls correctly with a SIP phone?
10:54.38pooh_Madkiss: Nope, but I do know Pilsner in good old Tjech..
10:54.59moralei want a kangaroo
10:55.05pooh_morale: none ot the outbound zap channels will go, all other trunks and internal calls work fine
10:55.37pooh_morale: both SIP and IAX2
10:56.00moraleand you are using cvs head?
10:56.01pooh_morale: yeah, need to debug some more, but the lag to .aus is not inviting
10:56.05pooh_beta2
10:56.12pooh_oh no... 1.09 over there
10:56.54moralehmm.. you should try upgrading to cvs head. i have found it works pretty well, if someone breaks too it usually can be fixed easy enough.
10:57.11pooh_Yeah, maybe I'll do just that
10:58.36*** join/#asterisk skyphyr (n=alan@80-195-238-25.cable.ubr01.hari.blueyonder.co.uk)
10:58.45skyphyrit was the power
10:58.52pooh_skyphyr: good, thx
10:59.21skyphyrno worries :-) is there a wiki or something I should log this away in?
10:59.33Madkisspooh_: That's the same
10:59.44coppicepooh_ only parts of .eu use bri extensively
11:00.17pooh_Madkiss: ah well, the product it produces has *many* names now :-)
11:00.27pooh_coppice: correct
11:00.52coppicei bet all the places that heavily investe in BRI must be kicking themselves now. it had a pretty limited life by telco standards
11:01.28pooh_coppice: 99% of businesses use BRI/PRI over here in western Europe
11:01.42coppicePRI yes. BRI no
11:01.52pooh_and... PRI costs about 3K just to have it connected!!
11:01.55PakiPenguincoppice: here in PK , PRI == YES and BRI == partly yes  :)
11:01.56*** join/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au)
11:02.23PakiPenguinpooh_: 3k what? here its pretty cheap , but they only give pri to Service Providers
11:02.33coppicein most of asia a BRI installation would be the first the installer has ever done
11:02.34pooh_3K USD
11:02.54pooh_then monthly costs about 250 USD
11:02.56PakiPenguin:o
11:03.02PakiPenguinno not that much here
11:03.16PakiPenguincoppice: i am on a bri right now , 128k :D
11:03.56PakiPenguinhttp://ptcl.com.pk/isdn_bri.html
11:04.32PakiPenguinhttp://ptcl.com.pk/customer_care/forms/isdn_pri_policy_new.pdf
11:04.39coppice.pk isn't exactly the majority of asia :-)
11:05.00PakiPenguinoh well coppice :) hehe
11:05.14PakiPenguinindia has bri+pri too
11:05.50coppicethere is little BRI in india. there is more R2 than PRI as well, although PRI is fairly widespread
11:06.13*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
11:06.23coppiceeverywhere has BRI, but in most places nobody uses. often because the fees are set to prevent it
11:07.00skyphyrdumb question (going through my zapata.conf at the moment) I've just got an fxs so I'd assume switchtype is irrelevant to me and can be commented out?
11:07.17RoyK.de did it The Right Way, ditching analog phones when ISDN was rolled out, and when was this? the eightees?
11:07.42PakiPenguincoppice: here the govt. really pushed bri , they even gave NTs ( with isdn modems free ) to customers
11:07.43coppiceRoyK: BRi didn't get much momentum until the early 90s
11:08.10RoyKk
11:08.12PakiPenguincoppice: we've had it for ages too , but they just didnt push it till now
11:08.37moraleit is really hard setting up a linux machine to jumpstart solaris installs
11:08.51moraleit doesn't support the sun partition stuff properly
11:08.51coppicehere an analogue line is HK$90 a month and no call charges. a BRI is HK$750 a month and substantial call charges. guess how many people use BRI? :-)
11:09.09coppicepushing BRI now is crazy. It is essentially obsolete
11:09.31RoyKcoppice: you mean analog is better?
11:09.31PakiPenguinwoah
11:10.00coppiceI mean ADSL has replaced it for data, and voice is moving to VoIP
11:10.25RoyKperhaps
11:10.28PakiPenguinhere switching from analog to bri costs ( 800rs ,comparing to getting another analog line which costs 1800rs. ) and the line rent is 260 rs. / month , analog linerent is around 180rs. / month , and call rates are same for both! so i guess its pretty much the same deal
11:10.42pooh_how much is 1 rs ?
11:10.45*** join/#asterisk oej (n=Olle@apollo.webway.se)
11:10.53PakiPenguin1Us$ = 60rs.
11:10.56RoyKwhat _is_ an rs
11:11.10PakiPenguinrs. = Pakistani Rupee
11:11.21pooh_coppice: what is the costs for a HK DID in and outbound ?
11:12.10PakiPenguinand VoIP is banned here! 5060 & 1720 and some other port banned on the internet exchange
11:12.24PakiPenguinofcourse they dont know about iax :p
11:12.27pooh_PakiPenguin: use 4569 :-)
11:12.28moraleuse IAX2 :)
11:12.32pooh_LOL
11:12.49coppiceHK DID means using PRI or RBS. Those are about HK$3500/month, before discounting. 1 E1 only gives you 200 or 300 DID numbers, unless you can work a deal.
11:12.49PakiPenguini know pooh_ :) or move sip to some other port
11:12.52PakiPenguin:p
11:13.29coppiceCanute Telecom - we stop your VoIP :-)
11:13.45pooh_coppice: ok... how about 1 DID for personal usage IAX2 terminated somewhere ?
11:14.37coppice1DID would be an ordinary line, wouldn't it? that's HK$90 a month
11:15.22pooh_coppice: how about connecting and using it through IAX2 ?
11:15.33*** join/#asterisk zotz (n=zotz@24.231.47.168)
11:15.42coppiceThe Philipinnes bans VoIP, and they back up port blocking with such awful internet performance VoIP cannot work :-)
11:16.05coppicepooh_: do you mean putting your own kit here to do it?
11:23.24pooh_coppice: nah , more like a HK based provider
11:23.54coppicedunno. nobody here uses those. VoIP is too expensive for IDD use, and local calls are free anyway
11:24.30pooh_coppice: yeah, so I noticed ;-)
11:25.06PakiPenguincoppice: local calls are free :o
11:25.16pooh_coppice: making calls from the star ferry back in the 90' s was fun!
11:25.44coppicethey really fleece tourists at the start ferry
11:25.59pooh_:-)
11:26.42pooh_coppice: stayed a few weeks in Whampoa Fa Yu. Had good look around, especially Lang Kwai fung area ;-)
11:27.42coppicethat would be Fa Yuen, but most people use the english equivalent - garden
11:27.43coppiceand it is Lan Kwai Fong
11:28.19pooh_coppice: :-) long time ago
11:29.21coppiceWhampoa Garden made an awful lot of money for Hutchison. I guess it helped to finance all their 3G screwups :-)
11:29.28pooh_:-)
11:30.03coppiceshanghai isn't the nest place to learn mandarin either. the locals seldom speak it
11:31.36coppiceI don't really like shanghai, but I can never really work out why
11:31.56gstdoes the new sip-domains support also allow to users with the same name (but different domains) to register to asterisk (e.g. user@foo and user@bar). i only found a method to utilize the different domains in the extensions.conf but not for the registration of the users.
11:31.58pooh_coppice: too busy, different way of doing things
11:32.05gsts/to/two/
11:32.18coppiceI like HK, so too busy isn't an issue :-)
11:32.36coppiceBeen to HangZhou? That's a nice city
11:32.54*** join/#asterisk skyphyr (n=alan@80-195-238-25.cable.ubr01.hari.blueyonder.co.uk)
11:32.57pooh_coppice: nope, I really need to go over to HK and China area again
11:33.24pooh_coppice: and play golf!!
11:33.34skyphyrhi all - sorry to be a pain (again) - was wondering if I could grab some help
11:33.49coppiceHangZhou is only 2 hours away from shanghai on the motorway. a very different atmosphere, though
11:34.08pooh_coppice: Yup, so I heard from some friends in HK
11:34.43skyphyrfollowing the instructions in the Asterisk book, but it seems asterisk isn't starting.
11:35.04pooh_skyphyr: /var/log/asterisk/messages ?
11:35.42skyphyrahhh thanks - lots in there :-)
11:35.50pooh_skyphyr: yup
11:36.30*** join/#asterisk MatsK (n=root@55.80-203-80.nextgentel.com)
11:38.14pooh_Did anyone play with Xen and * yet ?
11:39.07*** join/#asterisk Tili (n=Tili@211.147.234.5)
11:39.56skyphyrhmmm - googling on the error, well one of the errors, isn't helping much
11:40.16pooh_skyphyr: what is the error
11:40.18skyphyrapparently it's getting permission denied on /dev/zap/channel
11:40.37pooh_skyphyr: *exact* output pls
11:40.57skyphyrNov  5 11:28:28 NOTICE[6178] cdr.c: CDR simple logging enabled.
11:40.57skyphyrNov  5 11:28:28 WARNING[6178] res_musiconhold.c: Unable to open pseudo channel for timing...  Sound may be choppy.
11:40.58skyphyrNov  5 11:28:29 WARNING[6178] chan_iax2.c: Unable to open IAX timing interface: Permission denied
11:40.58skyphyrNov  5 11:28:29 WARNING[6178] chan_zap.c: Unable to open '/dev/zap/channel': Permission denied
11:40.58skyphyrNov  5 11:28:29 ERROR[6178] chan_zap.c: Unable to open channel 1: Permission denied
11:40.59skyphyrhere = 0, tmp->channel = 1, channel = 1
11:41.01skyphyrNov  5 11:28:29 ERROR[6178] chan_zap.c: Unable to register channel '1'
11:41.03skyphyrNov  5 11:28:29 WARNING[6178] loader.c: chan_zap.so: load_module failed, returning -1
11:41.05skyphyrNov  5 11:28:29 WARNING[6178] loader.c: Loading module chan_zap.so failed!
11:41.11skyphyrthat's one lump
11:41.24skyphyrsimilar thing later on
11:41.32pooh_skyphyr: did the udev entries ?
11:41.56skyphyrI haven't create any, but all the stuff in /dev/zap is owned by the group dialout
11:42.38moraleskyphyr: put asterisk in the dialout group.
11:42.59pooh_skyphyr: do you have /etc/udev/rules.d/50-udev.rules ?
11:43.07moraleor else just change the permissions of /dev/zap
11:43.26skyphyryeah - gentoo has one standard
11:43.34skyphyrI add my own bits to 10-udev.rules
11:43.53*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
11:43.59pooh_skyphyr: dunno for sure, but /dev/zap will NOT be used but udev instead
11:44.41skyphyrudev just creates /dev entries right?
11:45.02coppicepooh_: HK is a bad place to move to if you are in telecoms
11:45.05pooh_skyphyr: see the rules I just haded over
11:45.11skyphyryep :-) thanks
11:45.19pooh_coppice: yeah, but a *LOVELY* place to be
11:45.50skyphyrseems I already have them in 10-zaptel.rules
11:46.00skyphyrthat's the one which sets the group to dialout
11:46.01pooh_skyphyr: ok
11:46.20coppiceif used to be a great place to be if you are in telecoms. deregulation made the industry a train wreck. now everyone wants to copy the HK deregulation model :-(
11:46.50PakiPenguincoppice: we'd be in the people copying too:p
11:47.18pooh_coppice: hmmm, you're a native telecom person ?
11:47.22pooh_skyphyr: http://lists.digium.com/pipermail/asterisk-users/2005-May/108724.html
11:47.25coppicelots of countries explicitly say they are following the HK model.
11:47.45coppicepooh_: a native of telecoms? :-\
11:48.25PakiPenguincoppice: engineer?
11:48.39pooh_coppice: telecom pro from the start? or evolved into voip?
11:49.20coppiceI've worked in various things, but I was doing nicely from telecoms in HK for a while
11:49.50coppicewe have like 50 IDD providers. Not a single one can make money. what kind of deregualtion model is that?
11:50.01pooh_coppice: bad
11:50.08PakiPenguincoppice : very bad !
11:50.30coppiceweird thing is people keep applying for IDD licences
11:50.51pooh_coppice: I guess voiping into China mainland isn't easy (regulated) too?
11:51.03PakiPenguinhow much is it for a lisc?
11:52.28coppicenot much. basically just admin costs. why does anyone want a licence, though
11:53.05PakiPenguinpooh_ : look at starpbx @ sf.net
11:53.10coppicepooh_ you mean a latency causer? hotels are where things dwell :-)
11:53.39PakiPenguincoppice : here the voip lisc. costs just way too much
11:53.51PakiPenguinso much that only telecos can afford it :)
11:55.20PakiPenguinpooh_: holdon :) lemme link you
11:55.23pooh_k
11:56.40skyphyrnot well yet :-(
11:56.53skyphyrstill haven't figured out why this error is there
11:57.16pooh_skyphyr: read the mailinglist thread ?
11:57.23skyphyrwhich one?
11:57.32pooh_skyphyr: http://lists.digium.com/pipermail/asterisk-users/2005-May/108724.html
11:58.39pooh_skyphyr: subject referred to by morale (tm) ;-)
11:59.25skyphyrthanks :-) I'm going to reboot just to make sure everything has updated that should have
11:59.37skyphyrthough I think it has, but ideas are running short :-)
11:59.41skyphyrbrb
12:02.42*** join/#asterisk skyphyr (n=alan@80-195-238-25.cable.ubr01.hari.blueyonder.co.uk)
12:03.10skyphyrthis is what I get when I try start and then connect
12:03.10skyphyr# asterisk
12:03.10skyphyr# asterisk -r
12:03.10skyphyrUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
12:03.20skyphyrand to answer it's question - yes it idoes
12:03.42azzieskyphyr, does it start at all ? :)
12:04.16skyphyrseems not
12:05.03skyphyrhttp://pastebin.com/418102
12:06.44moralethat means your zapata.conf is wrong
12:06.56moraleor you don't have the zaptel/wctdm modules loaded.
12:07.07skyphyrahhh ok - conf is wrong then :-) thanks
12:07.14moralehold on a second.
12:08.05moralehttp://deadbolt.ca/zapata.conf.txt and http://deadbolt.ca/zaptel.conf.txt for reference.. i use bell202 signalling here thats all that is different
12:08.42skyphyrthanks very much :-D
12:11.16pooh_PakiPenguin: Ah, I know of http://sourceforge.net/projects/starshop/
12:11.41pooh_PakiPenguin: I was mor referring to virtualization of the hardware
12:12.05PakiPenguinoh
12:12.07PakiPenguin:)
12:12.28pooh_PakiPenguin: e.g. multiple, secure and isolated * instances on 1 box
12:12.46PakiPenguini see
12:13.12pooh_PakiPenguin: sorta VMWare but on OS/kernel level with very limited overhead
12:13.42pooh_PakiPenguin: sharing host resources
12:13.49PakiPenguin:)
12:13.56PakiPenguini see
12:14.06PakiPenguinsomething like linnode
12:14.08PakiPenguinmaybe
12:14.23pooh_PakiPenguin: moving over instances from 1 hardware server to another in a couple of milisecs
12:14.43PakiPenguinumm i see
12:14.45*** join/#asterisk rigid (n=The@port-212-202-73-82.dynamic.qsc.de)
12:14.51rigidre
12:14.53rigidHow can I prevent the caller from hearing "nothing" when my line is busy and getting the error "Call from user 'testuser' rejected due to usage limit of 1"?
12:15.25skyphyrI cut it back to this and it still doesn't work http://pastebin.com/418108
12:15.29pooh_rigid: who is producing that specific output?
12:15.45skyphyrbut if I remove the channel from it then it starts fine
12:16.07pooh_skyphyr: ks_fxo is correct?
12:16.24pooh_fxo_ks I mean
12:16.34*** join/#asterisk underbill (n=bill@dsl092-234-029.phl1.dsl.speakeasy.net)
12:16.38rigidpooh_, asterisk... it registers with a sip account and then Dial(SIP/testuser) to forward incoming calls on that SIP... if i already use my phone, i get this error and the new caller hears nothing until the call times out
12:17.03pooh_rigid: I wonder who sets the limit thing ?
12:17.24rigidpooh_, i do... i want the new caller to hear a busy-signal...
12:17.46rigidpooh_, so that she knows, my line is busy... :)
12:17.50pooh_rigid: send it to voicemail to test if your dialplan is ok
12:17.53rigidi got exten => <sipnumber>,1,Ringing
12:18.03skyphyrI thought so
12:18.06rigidexten=> <sipnumber>,2,Dial(SIP/testuser)
12:18.11skyphyrI have an fxs card
12:18.17pooh_skyphyr: ok
12:18.20rigidexten => <sipnumber>,3,Hangup
12:18.54pooh_rigid: you need a <sipnumber>,103 prio to handle the call
12:18.55rigidhow can i insert a command in my dialplan that only executes if Dial(SIP/testuser) fails?
12:19.16rigidpooh_, will that be called if Dial() fails?
12:19.21pooh_rigid: yes
12:19.40rigidpooh_, tnx... where is that documented? is it always 10x ?
12:19.49pooh_prio+101
12:20.01pooh_so 2 fails it will jump to 103
12:20.11rigidpooh_, i guess it has to be AFTER the failing command (with prio)
12:20.22pooh_rigid: dialplan basics over at voip-info.org
12:20.30pooh_yup
12:20.34rigidpooh_, tnx alot
12:20.38pooh_np
12:21.46rigidpooh_, one more q: does the following command (in my case Hangup()) need to be prio+1 or prio+2 ?
12:22.55rigidpooh_, so <sipnumber>,2,Dial(SIP/testuser) ; <sipnumber>,103,Busy() ; <sipnumber>,3,Hangup() ?
12:22.56pooh_rigid: prio 104 will be hangup() too
12:23.34pooh_rigid:  better leave the original sequnce in tact, so 1,2,3 and BELOW it order the prio+101 for readability
12:23.48rigidpooh_, ok tnx
12:25.43*** join/#asterisk deezed (i=none@adsl-065-006-189-182.sip.bct.bellsouth.net)
12:27.37skyphyranyone know how to check the pci spec on a motherboard?
12:27.55pooh_skyphyr: add fxoks=1
12:29.00skyphyrto which file?
12:29.20pooh_zaptel.conf, remove channel =
12:30.12skyphyrI already have that in my zaptel.conf - that pastebin was my zapata.conf
12:30.15pooh_rmmod zaptel etc and reload them
12:30.32pooh_nope, not there
12:30.47pooh_fxoks=1
12:31.31pooh_then do ztcfg -vvv, then dmesg
12:31.42*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
12:32.57skyphyrhttp://pastebin.com/418124
12:33.49pooh_skyphyr: I was talking about /etc/zaptel.conf not about /etc/asterisk/zapata.conf
12:34.12skyphyryeah - my zaptel.conf already has the fxoks=1 in it
12:34.24pooh_so you posted zapata.conf ?
12:34.28skyphyryeah
12:35.31pooh_skyphyr: pastebin your /etc/zaptel.conf
12:35.35skyphyrthere's nothing wrong with these lines right?
12:35.36skyphyr<PROTECTED>
12:35.36skyphyr<PROTECTED>
12:35.36skyphyr<PROTECTED>
12:35.51pooh_skyphyr: first /etc/zaptel.conf
12:36.10pooh_low level start ;-)
12:37.29skyphyrzaptel.conf http://pastebin.com/418130
12:37.29skyphyrzapata.conf http://pastebin.com/418129
12:37.42snitthey
12:37.56snittdelmar: yeah
12:38.35pooh_skyphyr: ok, zaptel.conf looks ok
12:39.47rigidis it possible for asterisk to recognize an incoming fax without answering the call?
12:40.43pooh_skyphyr: yu may want to add group=1 to zapata.conf below context=blah
12:41.07pooh_rigid: No, unless you know for sure it is a dedicated fax extension :-)
12:41.43rigidpooh_, :( i thought maybe the network somehow transmits some information that asterisk can recognize :(
12:42.18pooh_rigid: what is your extact wish
12:43.17rigidpooh_, i want to send/receive fax via voip... but if possible on the same number like my voice-phone
12:43.18skyphyrok :-) thanks
12:43.30rigidpooh_, voice-phone (voip-phone for voice)
12:43.49pooh_rigid: study the asnwer() command ;-)
12:43.59rigidpooh_, i will
12:44.03skyphyryou're the man pooh!
12:44.10skyphyrwell you may be
12:44.10pooh_skyphyr: works ?
12:44.17skyphyryou're really gender agnostic to me at the moment :-)
12:44.20skyphyrwell it started
12:44.25skyphyrso I think it's getting there
12:44.57pooh_skyphyr: pastebin any suspicious output
12:45.22pooh_rigid: answer() command, type earlier
12:45.35skyphyrhow do you exit asterisk cli? exit and quit both come up as no such command
12:45.45pooh_exit
12:46.05pooh_how did you enter the CLI?
12:46.10rigidpooh_, i guess i'll try T.38
12:46.21pooh_rigid: good luck!!!!!!! :-)
12:46.27skyphyrasterisk -c
12:46.39pooh_skyphyr: kill asterisk and try asterisk -r
12:47.32pooh_skyphyr: how do yu start asterisk ?
12:57.15pooh_hmmm, all gone
13:00.36skyphyrI'm here again now :-)
13:00.36pooh_ahhhhhh, somebody alive :-)
13:00.36skyphyrdragged away from the biggest fan of yours I know
13:00.36skyphyrfrom here by the biggest fan
13:00.39skyphyrthough he does tend to beat you I'm afraid
13:00.58skyphyrmy son has a toy winnie the pooh
13:01.36coppiceI thought all kids saw pooh as male.
13:02.38pooh_coppice: dunno, look at me at any pic and you will notice my fine red shirt, but not wearing anything else, so you would be able to tell
13:03.29coppicewell pooh is supposed to be a bear of little brain, so he might be a male of little dick too
13:03.54pooh_coppice: To you a question, for me a knowledge
13:03.59skyphyrhehe
13:04.18*** join/#asterisk lehel (n=mey@86.125.98.100)
13:04.27lehelhello
13:06.58lehelcheers pooh_ ;)
13:06.58coppiceas soon as the pooh movies appeared he was obviously male, because no woman could have that voice
13:07.12skyphyrhave you seen who sings the songs on the movies?
13:07.56lehelu ppl saw skeleton key?
13:07.59coppicepooh_ I must say your ride really sucks as the new HK Disneyland
13:08.04pooh_skyphyr: nope, different native language
13:08.35pooh_coppice: I know, it was designed for kids ;-)
13:08.50coppicethe kids think it sucks too :-)
13:09.07pooh_coppice: ask them again when they CAN speak ;-)
13:09.09coppiceyet, oddly, it has by far the longest queues
13:09.39pooh_coppice: not odd, look at how many people use *@home and how many are disappointed ;-)
13:10.01coppiceHK Disneyland has very few good things.
13:10.13RoyKrotfl. someone reported an error: Customer receives FM 98.8 on his IP telephone
13:10.20pooh_LOL
13:10.33*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net)
13:10.44pooh_RoyK: he must have a GREAT gateway somewhere
13:10.54RoyK:)
13:19.33*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
13:21.32skyphyranyone have recommendations for services I should hook up to asterisk?
13:22.06InfraRedsex line
13:22.44RoyKmusic^Wporn on hold
13:23.34skyphyrwhere do I get those?
13:23.42skyphyr:-P
13:31.39skyphyrwhich log do I check to see if I got my SIP conf happening correctly?
13:33.10RoyK~openpbx?
13:33.13jbotextra, extra, read all about it, openpbx is an asterisk fork without asterisk's limitations of using other GPLed code. see http://openpbx.org/ for more info, or join #openpbx
13:33.36*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
13:41.15*** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
13:56.55skyphyranyone here using sipgate.co.uk?
13:59.46skyphyrwhen I do a sip show registry it says I've registered
13:59.53skyphyrbut when I check on the website it says device offline
14:08.46*** join/#asterisk marten_ (n=marten@ua-83-227-150-77.cust.bredbandsbolaget.se)
14:09.50marten_Can anyone help with the 'callerid', I have a few problems setting and displaying
14:10.11*** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
14:10.27Drukenanyone around?
14:10.41tzafrir_laptopno
14:10.56Druken:)
14:11.08Drukenis there a cdruserfield variable?
14:11.22tzafrir_laptopmarten_, with what channel type? (sip/zap/iax)?
14:11.34marten_Sip
14:11.52tzafrir_laptopwhat exactly are the problems?
14:12.46*** join/#asterisk cjames (n=james@203.177.21.164)
14:13.40marten_I'm trying to do this : exten => 1000/Unknown,7,Set(CALLERID(all)="secret <${Number}>") ; and I get an error that set is not a command !!!
14:14.11marten_Im catching the Number with the read command.
14:14.16Drukenuhmm... don't think you can do that....
14:14.48ManxPowermarten_, What version if Asterisk?
14:15.06moralethe function is SetCallerID('My Name' <1231231234>)
14:15.25ManxPowermorale, not in CVS-HEAD or 1.2bbetax
14:15.33ManxPowerAnd you don't use quotes
14:16.32marten_Give me a hint of how to check version please
14:16.46Drukenshow version ?
14:16.49ManxPowermarten_, "show version"
14:17.34marten_Obviously ... so it shows! 1.0.8
14:18.04DrukenManxPower: cdruserfield variable?
14:18.06ManxPowermarten_, Set is only in CVS-HEAD or 1.2beta
14:18.35marten_But SetCallerID should work ?
14:18.47ManxPowermarten_, for 1.0.x you would use SetCallerID(Robert Dobbs <666>)
14:18.51ManxPowermarten_, correct
14:19.03marten_I'll try, will be back.
14:19.06marten_Thanks
14:22.07marten_Hmmm. I got an error ! "bx.c:1948 ast_pbx_run: Timeout, but no rule '"
14:24.39skyphyrgetting this from sip show registry should mean it worked right?
14:24.39Drukendoes ${CDRuserField} exsist?
14:24.40skyphyrsipgate.co.uk:5060              1993240            105 Registered
14:25.06ManxPowerDruken, No idea.
14:25.14*** join/#asterisk oej (n=Olle@apollo.webway.se)
14:26.49*** join/#asterisk fugitivo (n=ajf@209.13.245.29)
14:28.21Drukendamn,.... i don't think it does....
14:29.11ManxPowerREADME.variables would be where you would look.
14:31.33Drukenwel... this sucks ass...
14:33.21ManxPowerOf course if you are looking for an application, perhaps "show applications like CDR"
14:35.04Drukenno... i was using the userfield to know what calls, i'm not replacing the CID on...
14:35.11Drukenbut apparently it doesn't exsist...
14:35.23Drukennor does userfield...
14:35.33ManxPower<PROTECTED>
14:35.48Drukensets the field... but how do you READ it.. :)
14:36.05RoyKSELECT userfield FROM cdr ...
14:36.06RoyK:P
14:36.27Drukenbad RoyK BAD! :)
14:36.30ManxPowerDruken, One might assume that if you set it, you know what it is already.
14:36.30morale${CDR(userfield)}               The channels uses specified field.
14:36.47RoyKDruken: if it's in the same call, use a variable, then set cdr userfield to that variable, then read that variable further on
14:37.09RoyKmorale: i beleive that's only for custom cdr. i don't think you can use that in the dialplan......
14:37.25ManxPowermorale, you are talkking about all 1.2/CVS-HEAD stuff.
14:37.31ManxPowermorale, where are you getting your information?
14:37.40moraleManxPower: yeah. the asterisk/doc directory.
14:37.44moralei run cvs-head
14:38.06ManxPowermorale, Ah.  Poor thing.
14:38.34Drukenwell, reguardless... that's what i was lookin for :)
14:38.39Drukenthanks a bunch morale
14:39.14RoyKs/morale/low morale/?
14:46.57*** join/#asterisk jvictorfc (n=jvictorf@201009015204.user.veloxzone.com.br)
15:00.46*** join/#asterisk azzie_ (i=az@cpe-24-168-17-173.si.res.rr.com)
15:17.06fugitivois codec_ilbc fixcd?
15:17.08fugitivofixed
15:18.18RoyKyes. all bugs are fixed
15:18.21RoyKfixated
15:18.26RoyKstabilised
15:19.41fugitivoshould i get a tdm2400 with only fxs modules with echocan or without?
15:20.22file[laptop]hardware echo cancellers are a good thing
15:20.23fugitivoi'll use fxo with hardware echocan
15:20.42fugitivoso i don't know if it's necessary the fxs modules with echocan
15:20.45RoyKcan't really see the point of having double echocan
15:21.00fugitivoRoyK: that's my question
15:21.14file[laptop]I don't think you can, there's only one slot for the echo canceller
15:21.19RoyKfugitivo: that's my answer
15:21.56fugitivofile[laptop]: what do you mean with one slot?
15:21.57coppicefile: good echo cans are a good thing. a lot of hardware ones are not that good
15:22.12file[laptop]fugitivo: there's a slot for the echo canceller, at the bottom
15:22.16file[laptop]it's separate
15:22.58file[laptop]http://www.digium.com/downloads/product_sheets/tdm2400p.pdf -> look at the bottom of the card, it's going horizontal... it's purple, has an Asterisk on it
15:23.08file[laptop]that's the echo canceller
15:23.29fugitivook, but i'll use a separate card for fxo
15:23.36fugitivothat card will have echocan
15:23.45fugitivoand a tdm2400 with only fxs modules
15:24.29ManxPowerI would get a TE110P and a Channel Bank
15:24.42fugitivoManxPower: how much?
15:25.10ManxPowerfugitivo, Channel bank is about $300 on eBay and a TE110P is about $1,000
15:25.22ManxPowersorry, $500 for the TE110P
15:25.29ManxPowerso assume a total of $1,000
15:25.58fugitivoi have to check what is the price here
15:26.25ManxPowerDigium does not have a good track record when it comes to analog cards.
15:27.22fugitivodid you try channel banks?
15:27.41ManxPowerfugitivo, We have something like 4 channel banks that we use with Digium cards
15:28.11fugitivoany problem with channel banks?
15:28.16ManxPowernot for us
15:28.28fugitivowhat brand/model?
15:28.41ManxPowerAdtran Total Access 750 or 850
15:30.07fugitivothe TE110P is $687 here, i don't know about the channel banks
15:31.15fugitivoif you don't buy it on ebay how much is the adtran?
15:32.30fugitivono resellers in my country
15:36.50*** join/#asterisk svenna_ (n=svenna@p548D38A4.dip0.t-ipconnect.de)
15:56.08*** join/#asterisk dalfry (n=dalfry@ool-435285b1.dyn.optonline.net)
16:01.49*** join/#asterisk mmmToop (n=chatzill@mtngprs7.mtn.co.za)
16:13.53*** join/#asterisk fulgas (n=FuLg0r3@a81-84-116-219.cpe.netcabo.pt)
16:13.53*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
16:14.19pun`hi, anybody know of any WIFI IAX phone
16:15.03RoyKpun`: don't think such a thing exists
16:15.09pun`wow
16:15.17pun`how come, that being such a good solution
16:16.19*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
16:17.47RoyKpun`: not many of the large producers have turned towards iax yet
16:17.56pun`ok
16:18.25pun`i found a log of this channel where someone was talking about having converted a wifi sip phone to iax
16:18.46pun`NewSole2 ya we brought 10 WiFi IAX Phones to Bingo Hall..... gave out free calls... lol
16:18.53pun`NewSole2 they are sip phones but we re flshed them they use the 8086 chips
16:18.53pun`we put IAX and g729 on them
16:18.53pun`we have a buch... but we ordered from china
16:19.28pun`that guy newsole around?
16:19.46*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
16:19.46*** mode/#asterisk [+o anthm] by ChanServ
16:21.10marcus2is there any sort of timeline for 1.2.0 being released?
16:22.40mog_homeyes marcus when its done ^_^
16:22.51marcus2uh huh :)
16:22.51skyphyranyone know how to get callerID on UK phones? I get an error didn't finish callerID spill
16:25.56tzafrir_laptopskyphyr, which card do you have?
16:26.34skyphyrTDM400P
16:26.54skyphyrwith a single FXS module and no FXO ones
16:27.45tzafrir_laptopin zapata.conf:
16:28.01tzafrir_laptopcidsignalling=v23
16:28.08tzafrir_laptopcidstart=polarity
16:30.00skyphyrthanks :-) had the signalling right, but hadn't set the start
16:31.30skyphyr:-( same error
16:31.30RoyKhi
16:31.36RoyKi don't have any debug info about this
16:31.37RoyKbut
16:32.06RoyKit has happened that channels get 'connected' with asterisk 1.0.7
16:32.11RoyKacross calls
16:32.19RoyKanyone seen that?
16:34.54*** join/#asterisk freat (n=freat@h-69-3-229-184.chcgilgm.covad.net)
16:36.26*** join/#asterisk freat (n=freat@h-69-3-229-184.chcgilgm.covad.net)
16:44.43*** join/#asterisk vitaminmoo (n=vitaminm@70.58.177.107)
16:44.53*** join/#asterisk Assid (n=assid@203.115.64.59)
16:49.57Inv_arp~seen ariel_
16:49.58jbotariel_ <n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net> was last seen on IRC in channel #asterisk, 8d 2h 52m 31s ago, saying: 'morning'.
16:50.09*** join/#asterisk mrmags (n=stryfe@dsl254-076-201.nyc1.dsl.speakeasy.net)
16:50.23Inv_arpguess he still has no power
16:58.02*** join/#asterisk tclark_ (n=TC@S0106001310ead739.gv.shawcable.net)
16:58.04fugitivo8 days without power_
16:58.22fugitivoi kill myself if that happens
16:59.34skyphyrtzafrir - is it possible that the type of phone makes a difference? the phone I've got connected is an ntl one
16:59.50*** join/#asterisk santiago (n=santiago@208.195.215.124)
17:00.55Inv_arpjust got mine yesterday...
17:01.00Inv_arpdamn hurricanes
17:01.16fugitivoInv_arp: where are you located?
17:01.24Inv_arpmiami, fl
17:01.41fugitivomiami beach?
17:02.00Inv_arpnorth miami , bout 6 miles from the beach tho
17:02.17fugitivoit didn't hit that way in the beach
17:02.17*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
17:02.30*** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
17:03.33pun`~seen newsole2
17:03.36jbotnewsole2 <n=dave@d38-53-48.commercial1.cgocable.net> was last seen on IRC in channel #asterisk, 14d 20h 3m 39s ago, saying: 'harryvv... any ideas'.
17:03.44*** join/#asterisk santiago (n=santiago@208.195.215.124)
17:04.09pun`Im in Miami
17:04.15pun`still a lot of people without power
17:04.29Inv_arpyep
17:05.02pun`thx god I never lost power
17:05.16fugitivoweird, on the beach it was only 1 or 2 days without  power
17:05.33Inv_arppun`: where are u?
17:05.39pun`yeah normally is much more and they even keep people out of their appartments
17:05.42Inv_arpfugitivo: and aventura
17:05.50snittthe most ROTFL debug message in * is in app_voicemail.c line 3790!!
17:05.58snittcvs head
17:06.00fugitivoInv_arp: well, aventura is near the beach
17:07.13fugitivomaybe they worked hard in that area because it's a turist area
17:07.28*** join/#asterisk cybertank (n=todd@CPE000dbd0f269c-CM00111ae6ff9c.cpe.net.cable.rogers.com)
17:11.08skyphyris there any voip provider that particularly cheap to call Indian PSTN?
17:11.18skyphyrthat is rather..
17:11.35filecheap isn't all it's cracked up to be
17:11.39tecnicohttp://www.voip-list.com/voip_rate_exchange/india_voip_route.html ??
17:12.37tecnicoI was looking at the ones for Nepal just now ...
17:12.42skyphyrthanks tecnico.
17:12.42skyphyrfile - quality suffers?
17:13.05Assiddamn.. these guys are complaining that whenever they want to make a call.. ti takes them 1-2 tries to actually connect
17:13.30fileskyphyr: it can, yes
17:13.43af_I am trying to install in the same pc an hcf (bristuff) and a tdm400 with 3fxs and 1 fxo. Can be accomplished this setup?
17:14.09filedon't you all wonder, "how can they make this so cheap..." either they're a good company doing lots of business, or a horribly small company with horrible cheap routes
17:14.10file:)
17:14.19tecnicoDoes anyone have any comments on Teliax ? A friend of mine just got an Iaxy and is trying to use it straight to Teliax (no asterisk on his side). He is getting a bad echo and the Teliax people havent replied to him for a week...
17:15.19*** join/#asterisk tclark_ (n=TC@S0106001310ead739.gv.shawcable.net)
17:15.26skyphyrhehe good point :-) not really bound to them too much editing sip.conf is pretty convenient :-)
17:15.28xhelioxtecnico: Their customer service in regards to tech support has a lot to be desired. I use them with Asterisk and rarely have problems. But when I do, getting them to deal with it is like pulling teeth.
17:15.48tecnicotnx ..
17:19.35*** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net)
17:26.25*** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net)
17:26.51*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
17:30.33fiber0ptican you kill channels from the cli?
17:31.17mmmToopstop now ;  )
17:32.07fiber0ptiheh. thanks :P
17:40.01*** part/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl)
17:45.36*** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com)
17:48.04*** join/#asterisk tclark_ (n=TC@S0106001310ead739.gv.shawcable.net)
17:55.39*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
18:04.41*** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com)
18:04.53*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
18:06.12harryvvmorning
18:08.21e-miliohello everybody
18:08.29e-milioI have a question
18:08.38e-miliohaving lots of probs with span dsp
18:08.47e-milioframe slips, etc
18:09.16e-milioanybody knows a board that ONLY does de dsp on HW
18:09.20e-milioand can be used by *
18:09.49e-milioto send faxes through a digital circuit ?
18:09.52*** join/#asterisk zeedo (n=zeedo@80.68.92.188)
18:11.20harryvvsending faxes though asterisk always been a pain. I have not used it yet.
18:12.55e-miliowe have just HAD to move to hylafax
18:14.20e-miliosteve U. [spandsp] says it has to work
18:14.37e-miliobut we just were not able
18:14.57e-milioAnybody any ideas to use a hw dsp on * ??
18:15.58JonR800yes, remove * from the equation.
18:16.54vitaminmooAnyone know why the three DID's that were analog lines before we switched to asterisk stop working (they don't even cause anything to appear in the full log when you call them) upon upgrading from asterisk 1.0.6 to 1.2beta2?
18:16.55alephcome-milio: I think everybody here will say to forget about using * to fax.
18:17.27*** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
18:17.52harryvvvitaminmoo might want to reword that so it makes a little more sence.
18:18.13vitaminmooOh, I forgot the stopped working part
18:18.37e-milioalephcom: unfortunatly seems that way
18:18.37vitaminmooThe three original DID's cease working when I upgrade, the other like 5 DIDs I have keep working fine
18:19.27vitaminmooOn 1.0.9 with the exact same configs, they all work fine
18:19.36harryvvvitaminmoo did you save all the previos configuration files then move them back to the upgraded system?
18:19.59vitaminmooharryvv: I upgraded on the system (after hours), so the configs were all there
18:20.07vitaminmooIs that not safe?
18:20.25vitaminmooI ended up blowing it all away and reinstalling 1.0.9 with the configs I had backed up
18:21.20harryvvI would have personally built a new system and installed asterisk then loaded the old configs. If it did not work, then take out the new system and put back the old system.
18:21.42harryvvThat way there is no disruption of the network.
18:22.06*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
18:22.08vitaminmooharryvv: Yeah, I'm probably going to go in and re-try shortly, and I want to take a cleaner approach
18:22.42vitaminmooWish I had an extra box and zaptel card just for testing
18:23.04harryvvor just another hard drive.
18:23.18vitaminmooYeah
18:24.04vitaminmooAlright, well if there's no known issues like that, then I guess it's pretty damn likely it's a non-clean upgrading problem
18:24.11vitaminmooOff to try again in a bit
18:24.22vitaminmooThanks
18:24.55*** join/#asterisk santiago (n=santiago@208.195.215.124)
18:25.38harryvvyes, try a older version of asterisk
18:27.16vitaminmooThe older version works fine, but doesn't have the SUBSCRIBE support that I need for the Snom 320 extension monitoring feature to work, which I /really/ need
18:27.25vitaminmooIf I understand correctly
18:27.56vitaminmooWhen 1.2beta2 was up, the main numbers didn't work,b ut I did get lots of debug output involving SUBSCRIBE and PUBLISH, which is exactly what I was after
18:27.57vitaminmooHehe
18:28.04vitaminmooOur main phone number just didn't work
18:28.07vitaminmoo:)
18:28.57alephcomvitaminmoo: The upside is your customers can't "bother" you then. :-)
18:29.07vitaminmooHahahah
18:31.39*** join/#asterisk docelm0 (n=docelmo@static-71-251-95-2.tampfl.fios.verizon.net)
18:32.17*** join/#asterisk mickeyc (n=mike@rabies.blubbernet.com)
18:32.21docelm0Does anyone know when cdr_addon_mysql cut its CDR? Is it when the DIAL command is complete OR when the channel is completely closed?
18:33.07NetgeeksAny big realtime users here?
18:34.39docelm0yes?
18:35.19vitaminmooWhat's the name of that company that made phones that Linksys just bought?
18:35.27vitaminmooOh, nvm, sipura
18:35.28Netgeeksdoce: not sure
18:36.26vitaminmooAnyone know why sipura 841's hate dialing * unless it's *XX (ie, *XXX for an extension's voicemail just makes the phone freak out)
18:36.48vitaminmooThey also refuse to do XXX* and XXX** for hopping into and out of queues
18:38.22*** join/#asterisk yxa (n=diablo@cm121.gamma228.maxonline.com.sg)
18:40.59yxaif i have 2 external lines, whats the best way to dialout 2 numbers and pull them into a conference w/o any manual intervention?
18:41.56*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
18:44.44*** join/#asterisk riksta (n=rick@62.6.163.90)
18:45.20vitaminmooHmm, I suspect it's the dialplan
18:45.54yxavitaminmoo is that for me?
18:46.34*** join/#asterisk jd86 (n=jim@ip68-9-96-155.ri.ri.cox.net)
18:49.51*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
18:50.59vitaminmooyxa: no
18:56.41*** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net)
19:09.24*** join/#asterisk santiago (n=santiago@208.195.215.124)
19:09.47*** join/#asterisk SkramX (n=skramy@vistech.org)
19:11.21*** join/#asterisk fulgas (n=fulgas@a81-84-117-79.cpe.netcabo.pt)
19:19.44*** join/#asterisk marc324 (n=marc3234@206-248-128-112.dsl.teksavvy.com)
19:20.00*** join/#asterisk lenne_dk (n=mirc@83.72.129.7.ip.tele2adsl.dk)
19:20.33jd86i've begun to do some.. research i guess you could call it. It seems like there  is limited (very limited) support for some modems to work as (ptsn?) interfaces? (i'm very very new to this whole thing) is such a thing possible? and if so is it just the 3 chips that are the "clones" of x100p?
19:21.21lenne_dkI need help making asterisk accept incoming calls via enum.
19:22.56lenne_dkI have setup records at e164.org for both direct calls to my asterisk and via my voip-provider.
19:23.35lenne_dkThe testcalls via my voip work, but not the calls directly to my asterisk.
19:24.07*** join/#asterisk klasstek (n=nunyobiz@c-24-9-139-210.hsd1.co.comcast.net)
19:24.09lenne_dkI can see the direct calls vith trafshow, but nothing in asterisk console.
19:24.23lenne_dkAnybody tried that?
19:26.41*** join/#asterisk PakiPenguin (n=pingu@linuxpakistan/admin/pakipenguin)
19:27.15*** join/#asterisk PBXtech (i=nik@71-37-115-78.slkc.qwest.net)
19:27.49PBXtechwho else does a bring your own device plan beside broadvoic?
19:29.53*** join/#asterisk javo (n=nemitzj@68-187-84-135.dhcp.eucl.wi.charter.com)
19:31.11*** join/#asterisk enemy^x (n=null@morpheus.dataguard.no)
19:32.52sylehow many calls can an asterisk server handle
19:33.10alephcomstyle:  That is a wike open question:  1 - 3000.
19:33.13alephcoms/b wide
19:34.01filecome with me now... now we are going my way
19:34.06fileI wanna show you baby the world you never seen
19:34.26sylejoin #lyrics hehe
19:34.54sylefile
19:35.04filehiiiiii
19:35.15syleis there a max on how many threads can be mutex locked at once?
19:35.25fileyou can only lock a mutex once :)
19:35.34filethe others will sit there waiting for it to be unlocked
19:35.36sylei mean a max on how many clients will block
19:35.49filehow many threads will block? mmm dunno, never looked that up
19:36.30fileI don't think there is
19:37.21jd86I don't know where else to ask, but all of those 'vonage' ata's at like compusa and best buy (such as: http://www.compusa.com/products/product_info.asp?product_code=314460&pfp=SEARCH )  do they work with other services or just vonage?
19:37.57Augheyyou can "unlock" pap2's.  I have one and it works great
19:38.29sylejust vonage, they'll lock you out of the ata on purpose so you can only use them, make sure you know you can unlock it first or buy one unlocked
19:38.31jd86how hard is it to 'unlock' ? and do you have one which you would suggest? (i also need to figure out how to get around ordering vonage when i go there)
19:38.41jd86what about packet8 ones
19:38.58Augheywell if you're ordering vonage, then you don't need to unlock it.
19:39.10sylewhy not use asterisk
19:39.19alephcomAughey:  Where did you find info on unlocking the PAP-2s?
19:39.34Augheylemme find it
19:39.40jd86i dont want vonage..
19:39.50jd86espeically with the 90 day commitment thing
19:40.03sylepay as you go is much better
19:40.40jd86well i'md oing more of a fooling around with voip and figure out how it works.. just to know it and to see if there is any real advantage
19:41.24*** join/#asterisk L|NUX (i=asad@ility.net)
19:41.26AugheyI can't find the url right now.  If you search for linksys pap2 unlock, on google, you'll probably find it
19:41.48alephcomTks,  I will look again.  I haven't looked for a while.
19:42.44jd86Aughey: i found one, but they dont have the firmware because its 'illegal' you have it?
19:43.10alephcomjd86: I think you can download that from Linksys
19:43.51jd86This test uses specific firmwares.. These are not for any other use, so if you are looking for upgrade firmware for your PAP2, DO NOT assume that this will be them.. These are for unlocking your PAP2 unit.
19:43.52sylei don't
19:44.24jd86anyway... this pap2 , is it decent?
19:44.31alephcomOops, I guess not.  Linksys does not have PAP2 firmware available for download anymore.
19:44.43file[montreal]syle[winnipeg]: I'm GOING to Montreal :P
19:45.02syle[winnipeg]where are you now?
19:45.06file[montreal]New Brunswick
19:45.22syle[winnipeg]damn i hear alot of woman to men ratio in maritimes
19:45.30syle[winnipeg]more women than men that is
19:49.29jd86so i'm guessing you cant unlock them as easily any more..
19:54.25*** join/#asterisk zotz (n=zotz@24.231.47.168)
19:56.09*** join/#asterisk delmar (i=delmar@203.114.178.231)
19:56.46*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
20:00.45lenne_dke164 anyone?
20:01.35*** join/#asterisk kb1_kanobe (n=jsmith@h24-207-96-50.cst.dccnet.com)
20:01.37kb1_kanobeg'day all.
20:02.49kb1_kanobeI just came in to the office to get my daily fix of asterisk-users, and it looks like we stopped getting mailing list traffic around 5pm PST yesterday... ether that or 1.2 has been released and everyones problems have dissapeared... any ideas?
20:03.17delmardunno about that but.. yeah im on latest HEAD
20:03.21delmarseems nice so far
20:03.42delmarwas a little hickup getting it to install... make works.. make install and then make clean will loop.
20:03.49delmarbut its not hard to fix
20:04.06kb1_kanobeLots of good changes in 1.2
20:04.25delmarits wierd.. make clean works before u do a make... after u make... things will loop.
20:04.50delmaryeah i only updated to it yesterday.. and haven't had a chance to get to know much of it :P
20:05.07kb1_kanobeno listserv traffic... I'm starting to get a little twitchy...
20:05.25delmartested your own mta etc
20:05.26delmar?
20:05.45delmarim not on the list.. yet.. i will sort that out in the next day or so
20:05.47kb1_kanobeYeah, just digium traffic has stopped. I had another 1500 messages or so that were fine.
20:05.52delmarim sure there are many interesting things onthe list :P
20:05.59delmarah ok
20:06.36kb1_kanobeThey're quite busy lists. Between -users, -dev and -biz I've had about 14,700 messages since mid august.
20:06.47wunderkinmy last message received was 7 minutes ago
20:07.08kb1_kanobeOk, I will search further. Perhaps one of the backup mtas is hogging everything.
20:07.14kb1_kanobethanks wunderkin
20:07.45delmarwell thats something u dont see too often.. vmware processes hanging all over the place. wierdness.
20:12.27delmarmust just be windows itself crashing all over da place :)
20:12.49*** join/#asterisk jroysdon (n=jroysdon@c-67-182-64-213.hsd1.ca.comcast.net)
20:12.50delmarhappy is a windows server running in a vmware session on a linux box.
20:13.03jroysdonhar, har ;-)
20:13.37jroysdonI don't think a windows server is ever truely happy.  Unless perhaps if it is getting re-formatted for a Fedora install
20:13.40delmarhell of alot better than running raw winblows on the box.
20:13.57delmarkill -9 windowsprocess
20:13.58delmar:P
20:14.26*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-7-60.dsl.sfldmi.ameritech.net)
20:14.31jroysdonopenh323 question... well, actually, it's a pwlib compile question.  I can't seem to get pwlib to compile (which I need for openh323).
20:14.42delmaroh and when the server needs a hardware upgrade.. how much easier is THAT to content with when the windows virtual server has no clue :P
20:14.48*** join/#asterisk FuriousGeorge (n=bri@pool-70-111-110-204.nwrk.east.verizon.net)
20:14.50FuriousGeorgehey all
20:14.54FuriousGeorgei had a question:
20:14.54jroysdonSeems fairly simple, and I've got the required bison and flex installed (and gcc, gcc-c++).  Running Fedora Core 3
20:15.22FuriousGeorgeit appears the 5 REN provided by my FXS is not enough voltage to power both the doorphone and the device that toggles the door strike
20:15.23jroysdonAnyone recall problems with pwlib... or have h323 working with Asterisk?
20:15.38jroysdon(that or SIP trunks working with Cisco CallManager 4.1)
20:15.40delmarjroysdon pwlib = portable windows library etc?
20:15.50FuriousGeorgethe manufacturer suggests putting a 9 volt battery on the line to verify its a talk back problem:  iwo, if that solves it, there you go
20:15.57jroysdondelmar - uhm, it's a prereq for openh323
20:16.21jroysdonhttp://www.openh323.org/code.html
20:16.33delmarjroysdon, yeah pwlib is the portable windows library thing right?
20:16.34FuriousGeorgeis there a way to either have the fxs provide more voltage, if not, could i wire the line to the jack and fashion a more permanent solution
20:16.36jroysdonI dunno what it does, actually, I just follow the instructions that it is required ;-p
20:16.49delmarjroysdon, what linux distro u have?
20:16.54delmarRH?
20:17.05jroysdonRH FC3
20:17.49delmarwell on debian that lib and dev files etc.. are all packages... im sure there must be an installable package for RH too...rather than needing to compile your own.. unless u really want to... have u tried the distro package first?
20:18.01delmaron debian the package is called libpt
20:18.09jroysdondistro package was pretty old
20:18.16delmarah i see
20:18.39delmarok throw me some details on where i can source the .. source.. and ill see if i can compile it here
20:18.40jroysdonI'd much rather roll-my-own to get the latest anyway
20:18.56jroysdonPWLib source is here: http://www.openh323.org/code.html
20:19.08delmarnever done h323 support so .. be good to check it out :P
20:19.09jroysdonwget http://www.openh323.org/bin/pwlib_1.5.2.tar.gz
20:19.10delmarok sec
20:19.20delmareven better.. cheers
20:19.41delmarok so while im doing that.. tell me whats happening when u try to compile...
20:20.11jroysdonI'm running it one more time... I, uhm, had pwlib and openh323 installed from RPMs (which re-reading the instructions, it says to track down all versions and destroy)
20:20.45delmarah right
20:21.02jroysdon(although I don't see how having those installed show affect compiling pwlib)
20:21.21jroysdonbut I could see how it would mess up openh323 (which needs pwlib) and asterisk (which needs openh323 for h323 support)
20:21.54delmardamn it... can't be good having spu temp at 80c
20:22.00jroysdonEver done any integration with Cisco CallManager (any protocol) or Unity (SCCP)?
20:22.29delmarnone, but it cant be hard
20:22.34jroysdonI wouldn't think
20:22.55jroysdonWhat I'd done in the past (oh, 3 years ago) was use a router to talk between the two... it had h323 to the CCM and SIP to *
20:23.19jroysdonBut that's silly now that CCM has (some) SIP support
20:23.24delmarwhats so great about h323 vs other methods?
20:23.34jroysdonI can't get SIP to work at all
20:23.42delmaroh
20:23.49delmarbetween where/what etc?
20:23.52jroysdonI don't mind troubleshooting that either, but I really want h323 support as well
20:23.57jroysdonBetween * and CCM
20:24.12delmarCCM is running on what/where?
20:24.24delmarno nothing about it so .. :)
20:24.32delmarknow*
20:24.48jroysdonCisco CallManager (CCM) is its own proprietary system... it's on a box at my office and I have a VPN tunnel there
20:25.08Supaplexcisco call mangler? oh joy
20:25.16jroysdonI've got a 7960 at my desk here running a SCCP (Skinny image, not SIP) talking to the CCM.  I want the CCM to have a SIP trunk to * so I could call into * and check my VM
20:25.44delmarright..
20:25.44jroysdonThat, and I've got a PRI at the office I'd love to take personal calls into * with (incoming calls are free)
20:25.48delmarok just a sec
20:26.07delmarok pwlib bombed for me...
20:26.10delmarmake[3]: *** [/usr/src/pwlib/lib/obj_linux_x86_d/asner.o] Error 1 and so on
20:26.10jroysdonI know I can make the 7960 work with SIP and *, but I need it talking to the office.  So I need the office CCM to talk to *
20:26.11Supaplexthere is some support (beta) for SCCP
20:26.25jroysdondelmar - you have bison, flex, gcc, gcc-c++ ?
20:26.28Supaplexwe use 79xx's at my work
20:26.34jroysdonSupaplex - SCCP is for end nodes, not trunking between systems
20:26.52delmarjroysdon, yep on the later, unsure on the others, ill see later.
20:26.54jroysdonI can't make a 7960 register with two SCCP servers
20:27.00FuriousGeorgeis there any way to augmnt the voltage of talk battery provided by an fxo?  mine isnt providing enough to a doorphone i have on the line.  someone suggested a nine volt battery to help out, but i was looking for a more permanent solution
20:27.00jroysdondelmar - ok *wave*
20:27.05delmarok so the wiki has a couple of pages im sure u have looked at.. http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration
20:27.08FuriousGeorgefxo=fxs
20:27.16jroysdonCME != CCM
20:27.27delmarah sec
20:27.28jroysdonCME (CallManager Express) runs on Cisco IOS routers
20:27.33delmarhttp://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration
20:27.34delmar:P
20:27.39*** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
20:27.39*** mode/#asterisk [+o twisted] by ChanServ
20:27.41jroysdondelmar - yeah, tried it, it's lacking some details
20:27.47delmarhrm ok...
20:27.51delmarlet me read thru it a little
20:27.56jroysdon(ie, I did all the steps and it doesn't just work, or I don't know * well enough yet)
20:28.10jroysdonthe CCM SIP Trunk is /really/ easy, so I don't think I could have that wrong
20:28.23delmaranything coming up on the * console?
20:28.30jroysdonnot for inbound calls
20:28.41jroysdonmakes me think it either isn't trying to call, or * isn't seeing it...
20:28.52jroysdonooooh, duh
20:28.54jroysdonnevermind
20:29.00delmarjroysdon, what version ov CCM ?
20:29.12jroysdonI'm feeling stupid.  I have my PIX locked way down so that only my phone and my laptop can talk to the office, and no other home PCs
20:29.19jroysdonCCM 4.1(3)sr1 (latest shipping)
20:29.30jroysdonLemme open up my firewall a bit
20:29.32delmar#  Using H.323: In CCM Asterisk appears as a H.323 Gateway.
20:29.32delmar# Using SIP (only in CCM 4.0):
20:29.43*** join/#asterisk lehel (n=mey@86.125.98.100)
20:29.50jroysdondelmar - yeah, I've read it, but I don't mind walking through with ya
20:29.51lehelhello
20:29.52delmarnot sure if that could mean 4.0+ or what
20:29.55jroysdonlehel - hey
20:30.05delmarbut it does say 4.0
20:30.07jroysdondelmar - yeah, CCM 4.1 has the same SIP trunks
20:30.23delmarhey so u have a firewall .. is that by any chance effecting the coms between ccm and * ?
20:30.31jroysdonBut h323 is more... generic, standardized, and implimented, which is why I was going down that route
20:31.01jroysdondelmar - yeah, not from my laptop (where I tried it first) but from this PC.  Gonna open it up a bit so my PC here can talk to the office more
20:31.13delmarok so the steps 1-10 are simple enuf for ya im sure....
20:31.34*** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net)
20:32.06delmardamn it. i gotta do something about this fan control.
20:32.17jroysdonheh, yeah, FW is the problem.  Hah.  I can hit the public IP of the CCM, but not the private (tunneled) due to how I have it locked down
20:32.39delmari dont see why it just doesnt go hard.. noise doesnt bother me... q-fan control is off but its as if linux is running q-fan control... grr.
20:33.08delmarjroysdon, ok, play with the firewall a bit.
20:33.18delmarmy first thought would be.. u will see at least some crap on the * console
20:33.21delmarand can debug from there
20:33.31delmarif u are not seeing anything... there is blockage :P
20:34.40jroysdonfirewall is unlocked now
20:35.22jroysdon/usr/bin/ld: ./obj_linux_x86_d/asn_grammar.o(.gnu.linkonce.t._ZN10PCharArrayD2Ev+0x30): unresolvable relocation against symbol `PCharArray::operator delete(void*)'
20:35.22jroysdon/usr/bin/ld: final link failed: Nonrepresentable section on output
20:35.22jroysdoncollect2: ld returned 1 exit status
20:35.22jroysdonmake[3]: *** [obj_linux_x86_d/asnparser] Error 1
20:35.22jroysdonmake[3]: Leaving directory `/opt/download/asterisk/pwlib/tools/asnparser'
20:35.24jroysdonmake[2]: *** [debug] Error 2
20:35.26jroysdonmake[2]: Leaving directory `/opt/download/asterisk/pwlib'
20:35.30jroysdonjust a nasty mess at the end of 'make' for pwlib
20:35.41jroysdonGoing back to SIP for a sec
20:37.14delmarbison+flex are/were installed btw.
20:37.41jroysdonI have all the stuff it says it wants, no luck.  I'll come back to h323 later and try sip again with some * help from ya'all
20:37.54jroysdonOk, in my /etc/asterisk/extensions.conf I have
20:37.55jroysdon[nlsccm41]
20:37.55jroysdon; nlsccm41
20:37.55jroysdonexten => 5751,1,Dial(SIP/5751@cm01,30,rT)
20:37.55jroysdonexten => _57XX,1,Dial(SIP/_57XX@cm01,30,rT)
20:37.56jroysdonexten => _17XX,1,Dial(SIP/_17XX@cm01,30,rT)
20:38.14jroysdon(my desk CCM 7960 is 5751 and has 5757 on it as well)
20:38.46jroysdonmy /etc/asterisk/sip.conf has:
20:38.46jroysdon[cm01]
20:38.47jroysdontype=friend
20:38.47jroysdon;host=cm01
20:38.47jroysdonhost=10.12.254.11
20:38.47jroysdonport=5060
20:39.08jroysdonoh, lemme switch that back to use the name not IP
20:39.48jroysdonmy /etc/asterisk/sip.conf has:
20:39.50jroysdon; cm01.nlsys.biz
20:39.50jroysdon[cm01]
20:39.51jroysdontype=friend
20:39.51jroysdonhost=cm01
20:39.51jroysdon;host=10.12.254.11
20:39.51jroysdonport=5060
20:39.53jroysdon;port=5061
20:40.01jroysdonin /etc/hosts I have:
20:40.02jroysdon10.12.254.11    cm01    cm01.nlsys.biz
20:40.08SkramXWTF
20:40.29delmaruse IP address where possible if it's static
20:40.42jroysdon* rejects the call if I define it by IP
20:40.45delmarso go back to using the IP and think nothing more of it
20:40.52delmarreally?
20:41.08jroysdonyeah, but this is better at least:
20:41.09jroysdon*CLI> dial 5757
20:41.09jroysdon*CLI>     -- Executing Dial("OSS/dsp", "SIP/_57XX@cm01|30|rT") in new stack
20:41.09jroysdon<PROTECTED>
20:41.09jroysdon<PROTECTED>
20:41.11jroysdon<PROTECTED>
20:41.19jroysdonAt least now it may be talking
20:41.26delmaryup
20:41.32jroysdonWOOOT, it's ringing on 5751
20:41.44delmaryay :P
20:41.48jroysdonstupid firewall ;-p
20:42.10delmarfirewalls are good but can be tedious if u forgett about them when u are doing things. :P
20:42.29jroysdonAre these two lines both right?
20:42.29jroysdonjroysdon exten => 5751,1,Dial(SIP/5751@cm01,30,rT)
20:42.30jroysdonjroysdon exten => _57XX,1,Dial(SIP/_57XX@cm01,30,rT)
20:42.43jroysdon(I mean, I need the _ if I have wildcards with X, right?)
20:43.31delmarwell...
20:43.41delmarif there can only possibly be 2 more digits.. then XX is fine...
20:43.52delmarotherwise.. XX.,
20:43.58jroysdonYeah, 5700 - 5799
20:44.08delmarall good then
20:44.36jroysdonok, now it range, but when I pick up, no connect... probably a codec thing.  Going to force g711u on the CCM side of things
20:44.57delmarmight not be...
20:45.05delmartell me what ports you opened up?
20:45.32wunderkinexten => _57XX,1,Dial(SIP/${EXTEN}@cm01,30,T)
20:45.42delmarno audio or audio in one direction is what can happen when there are nat and/or firewall issues in the way.
20:46.02delmarwhat ports did you relax on your pix?
20:46.16jroysdonI opened up all traffic to/from my PC here to my office
20:46.25jroysdon(IP to NET, no ports, just all)
20:46.55jroysdon*CLI> dial 5751
20:46.55jroysdon<PROTECTED>
20:46.55jroysdon<PROTECTED>
20:46.55jroysdon<PROTECTED>
20:46.55jroysdon<PROTECTED>
20:46.56jroysdon<PROTECTED>
20:46.59jroysdonWOOT, codec issue
20:47.10jroysdon* doesn't have "free" g729a
20:47.21jroysdonCan you "talk" from a console call?
20:47.25delmarah i can help with that :P
20:49.11tessierdelmar: You can help get * a free g729 codec? I don't think that is possible.
20:49.25delmaroh really?
20:49.30jroysdonwoohoo, SIP call from my CCM 7960 to VM
20:49.31tessierdelmar: Yep.
20:49.35delmarwell im running g729 here qyite nicely thanks
20:49.37delmarand its free
20:50.57tessierdelmar: It's not pirated? How can you have it for free when the many patents in g729 require licensing?
20:51.23delmarnot my problem
20:51.31tessierThen it's not free. :P
20:51.33tessierIt's pirated.
20:51.41tessierAnd therefore of little use to most people.
20:51.50*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
20:51.58delmarfree as in cost me nothing .. sure its pirated.
20:52.28delmarits also a case of.. if u are making money.. u can afford to spend money on a legit g729 license for sure.. and its not too bad so i would consider it
20:52.42delmarbut from my testing.. nowhere near as thin as they advertise
20:52.47jroysdonHow much is a legit g729 license?
20:52.57delmarabout us$10 per channel
20:53.01mog_home10 dollar
20:53.06delmarbut  u dont need one
20:53.19mog_homewell you need one to be legal in the civilized world
20:53.23jroysdonWell, if I want to take VM from my office CCM via g729, I do
20:53.28mog_homei guess if you lived in botswana go right a head
20:53.29delmarjroysdon, u are doing pass through are u not?
20:53.43jroysdonI want to store personal VMs on * from my CCM/PRI
20:53.46mog_homeif you ever need conversion you need 729
20:54.30delmarjroysdon, well maybe u do need one but gsm or ilbc are quite good. use g729 for low bandwidth good quality audio.. ie. u dont use gsm/ilbc running a voip business im sure :P
20:54.31jroysdonRight.  What about GSM?
20:55.00jroysdonThis is just for my personal home use.. I don't have a land line anymore, but my cell reception sucks at home.
20:55.00*** join/#asterisk ahole (n=i@spirit.segfault.net)
20:55.10jroysdonInbound PRI calls to my office are free though
20:55.18*** join/#asterisk rat1101 (n=rat1101@ip68-100-158-77.dc.dc.cox.net)
20:55.35delmari yak with a guy here in NZ all the time sip to * via iax to his box using GSM on * to his phone... sounds great.
20:55.59rat1101does anyone use voipjet here?  I can't connect right now.  just wondering if anyone is also having problems
20:56.07mog_homegsm rocks
20:56.13mog_homei use it anywhere i can
20:56.17lehelthis why: Nov 5 23:33:20 DEBUG[25577] chan_iax2.c: Peer xxx.xxx.xxx.xxx lag measured as 2098ms - it is possible that i can't answer the call?
20:56.18delmarjroysdon, so if u want to take calls that sound good and professional g729 might be it but i reckon gsm sounds great too
20:56.22mog_homebut people are lame and only do ulaw and 729 sometimes
20:56.26mog_homegsm rocks
20:56.28delmarlehel, yep
20:57.05leheldelmar, it's a huge lag isn't?
20:57.09jroysdonWell, I know CCM supports GSM now, so I want to try them all
20:57.22jroysdonPlus, inbound g711u will be fine on my cablemodem (256k up, 3mbit down)
20:57.33lehelNov 5 23:32:59 DEBUG[25577] chan_iax2.c: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13)
20:57.33lehelNov 5 23:32:59 DEBUG[25577] chan_iax2.c: Acking anyway
20:57.36delmarmog_home, i still think many of these codecs are big bloat vs what people say they will do. i mean.. g729 12Kbit/sec my ass. i clocked it way higher than that...
20:57.37jroysdonReally, I'm doing this for testing and learning and experience.
20:57.42lehel"Acking"< wht's this?
20:57.52mog_homeumm it is at 9k
20:57.53delmarlehel, yes thats over 2 Seconds. way too much lag
20:57.54jroysdong729a with overhead is 24k
20:57.58mog_homebut you have to ad in over head of ip
20:58.11mog_homeiax trunking with enough calls you can get closer
20:58.12jroysdon(that's what we provision with Cisco QoS)
20:58.17delmarjroysdon, yeah.. 12Kbit/sec each leg... but its not
20:58.25jroysdonwhat is the default/demo VM account extension and password?
20:58.30tzangerdelmar: g729 *is* 8kbps
20:58.37mog_homeit is 8k
20:58.42tessierThe audio encoding rate is 8kbps
20:58.44mog_homeyou have to add over head of iax or sip
20:58.46tzangerdelmar: you need to take into account that you are not sending one packet with 1s of audio
20:58.47tessierActual bandwidth used is more like 30kbps
20:58.47mog_homeand ip
20:58.50tzangeryou're sending 1 packet with 20ms of audio
20:58.52mog_homeexactly
20:58.58mog_homewith 100 iax trunked calls
20:59.01mog_homebandwith per call
20:59.04tzangerand 20ms of g729 audio is almost dwarfed by the size of the UDP packet header
20:59.07mog_homeprobably closer to 9-10
20:59.14jroysdontzanger - exactly
20:59.22*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-73-7-60.dsl.sfldmi.ameritech.net)
20:59.40mog_homeand thats why we have trunking ^_^
20:59.43jroysdonAnyone know what a default VM account X and PW is?
20:59.45delmartzanger, depends, I have seen 8 or 12kbps mentioned.
20:59.48jroysdon(or where to check it)
20:59.53mog_home8 is the answe
20:59.53mog_homer
20:59.56mog_home13 is gsm
21:00.11tzangermog_home: exactly.  iax2 trunking is teh shit
21:00.16mog_homei know
21:00.22mog_homewish more people realized that
21:00.25delmarmog_home, well in that case, its even bigger BS. like i say, I have clocked it using WAY more bandwidth than this
21:00.27tzangerit's even better with stevek's patches for in-trunk timestamps and such
21:00.27mog_homeover 100 calls on a t1
21:00.28mog_homedamn
21:00.36tzangerdelmar: yes, remember
21:00.38tzanger20ms packets
21:00.39mog_homethen you have nubbed it somewhere delmar
21:00.42tzangerUDP headers
21:00.50mog_homecodecs dont lie
21:01.05tzangerI find a single g729 to be 2.4kBps, gsm to be 3.4kBps
21:01.14tzangernot trunked, IAX2.
21:01.26mog_homeat 20 ms?
21:01.48delmartzanger, right so it's not a truthful rate... sure it might be using 8Kbit/sec, but add all the other crap up and its not.. and you would only see benefit in trunking with 4+ calls concurrently anyway.
21:02.19mog_homeduh
21:02.22tzangerdelmar: it's truthful
21:02.27tzangerg729 is 8kbps
21:02.27mog_homethere is a simple calculator for all of that
21:02.36mog_homeits on asterisk guru site
21:02.45mog_homeit will tell you the answers you seek
21:02.50mog_homethere is no hocus pocus in voip
21:02.50delmartzanger, depends on how u want to look at it... payload only.. or entire packet :P
21:03.01mog_homewell when we say 8k
21:03.07tzangerif I made a funky-ass DS1 that just stuffed g729 frames into a pipe I could achieve 8kbps
21:03.09mog_homewe are talking about payload only
21:03.14*** join/#asterisk MikeJ__ (n=ircatjer@adsl-67-38-9-250.dsl.sfldmi.ameritech.net)
21:03.19tzangerdelmar: you have to talk just raw codec bandwidth
21:03.26delmartzanger, 2.4kBps both directions total? ie 1.2kBps each way?
21:03.29tzangeradding in the other stuff makes comparisons too hard
21:03.35tzangerdelmar: no.  2.4kBps both ways
21:03.52mog_homewell not to hard but more difficult to compare all fairly
21:03.58mog_homebut its not hard go to asteriskgurus
21:03.59delmartzanger, so 4.8 full duplex?
21:04.00mog_homeuse there calc
21:04.08tzangerdelmar: you can't add them together like that
21:04.17tzangeryou take 1-way bandwidth
21:04.19tzangernot 2-way
21:04.26delmartzanger, well just so i can understand..please do for the moment
21:04.37tzangerbecause downloading at 800kbps does not impede your ability or cause any change in uplink bandwidth
21:04.41tzangerno
21:04.55tzangerunderstand it the right way or you will spend huge amounts of time and energy re-learning it and trying to UNlearn the wrong way
21:05.16delmarok well 2.4kBps = 19,200bits/sec or were u meanint 2.4kbits/sec ? not kBytes ?
21:05.33tzangerI said kBps not kbpos
21:05.34tzangerer kbps
21:05.46delmaryes i saw the B not b hence why i asked
21:06.09*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-67-38-9-250.dsl.sfldmi.ameritech.net)
21:06.16delmarso the total usage.. paylad.. and packet and all.. that you see is 2.4kBps aka 19,200bit/sec?
21:06.46tzangerabout that
21:06.51tzangerit's been a while since I've done g729
21:07.03tzangeras mog_home has tried to point out several times there are bandwidth calculators on the internet
21:07.06tzangergoogle is your friend
21:07.12mog_homeone sec im getting the good one
21:07.29*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-67-38-9-250.dsl.sfldmi.ameritech.net)
21:08.17mog_homehttp://www.asteriskguru.com/tools/bandwidth_calculator.php
21:08.18mog_homebam
21:08.24mog_homeall your answers can be answered
21:09.20delmarcheers. ill book mark that :P
21:10.25mog_homeyup
21:11.22*** part/#asterisk skyphyr (n=alan@80-195-238-25.cable.ubr01.hari.blueyonder.co.uk)
21:12.33MikeJ[Laptop]sigh..
21:13.06mog_homesigh?
21:15.59tzangerfuck
21:16.06tzangernovell's standardizing on gnome
21:16.08delmari had one of those... just last night
21:16.11tzangerso much for suse as my next destko[p
21:16.12*** join/#asterisk warthog (n=nvadekar@216.249.38.88.ppp.northrock.bm)
21:16.24mog_homelol
21:16.27mog_homegnome rocks
21:16.29enderhah.
21:16.30delmarnovell still exists?
21:16.34delmar:P
21:16.36tessierI'm sure they will still ship kde.
21:16.38tzangerno gnome blows :-(
21:16.42tessierAll this means it that gnome is the default.
21:16.45enderlets see them try and re-write yast in gtk
21:16.59mog_homegnome rules
21:17.02tessierI really don't notice much what desktop I am using. I hardly ever interact with it.
21:17.14tessierThe window manager gives you little borders to drag around your apps with. That's it.
21:17.27enderand a file manager, if you so choose to use it.
21:17.38tessierI don't know many who do.
21:17.42tessierMy file manager is bash
21:18.17tessierAs Linux gets more desktop acceptance and more mere mortals use the gui file manager it will become more important though.
21:18.24endertzanger: did you see a news post about that?
21:18.33endertessier: 'course.
21:18.46warthogexten => 511,7,GotoIf($[${RESULT} = '0']?20:30)   , can someone help me figure out how to read a return code from an AGI, I tried this to no avail.
21:18.55tzangerender: yeah on /.
21:18.59tzangerhaven't read it yet
21:19.25warthoghowz it going ender
21:19.26tzangerI just dislike gnome's "we hate C++ so we'll bring OO to C with eighteen bazillion macros encapsulating everything, and then draw in millions of interdependent libraries"
21:19.29tzangerit makes me bonkers
21:19.40tzangerC++ is not my favourite but it's certainly the right tool for the job
21:19.53tessiertzanger: Do you plan on doing gnome programming or something? I don't see how the end user would ever know what lang it is written in.
21:20.02tzangertessier: the end user no
21:20.12*** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar)
21:20.21tzangerbut as someone who does interact pretty heavily with the desktop yes, I do try to fix bugs and change things here and there to make them better
21:21.44warthogafter running an agi ascript, how do you read the return code in a gotoif statement?
21:21.59tzangeryou don't
21:22.03tzangeryou can't get return codes
21:22.22tzangerthe app either exits normally, which is n+1, abnormally which is n+101 or hangs up (jump to h)
21:22.22endergod, are they just blatently trying to copy Red Hat?
21:22.26enderwarthog: not bad.
21:22.30tzangerwhich is totally fucking wrong in my opinion
21:22.46tessierThe whole way extensions are handled in * is rather weak.
21:23.28tessierIf it were implemented in some sort of programming lang like python or something it would be a lot more flexible while still remaining easy if it shipped with some examples.
21:23.38tzangertessier: yeah I suppose...  :-)
21:23.41tzangerI like python, just not very good at it
21:23.42tessierThere is python module for using it in the dialplan but you still run into problems a lot.
21:24.03tzangerI don't mind the way it's done now...v ery linear.. .but not being able to see app results or having ot rely on app-specific return vars is very very poor
21:24.12*** part/#asterisk ahole (n=i@spirit.segfault.net)
21:24.31warthogtzanger, I guess I could set a variable in the agi perl script itself though and read that?
21:25.06tzangerprobably yes
21:25.41*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
21:25.46warthogtzanger, stupid question, but normal is 0 or 1 as an exit code
21:26.27tzangeroffhand I don't recall :-)
21:26.58warthoganyone try one of the new sun ultra 20 workstations?  nice price eh!
21:27.29enderI played Unreal Tourney 2004 on one of them at Linux World San Fran.
21:27.40enderwon the round.
21:27.44warthogcool
21:27.53enderone frag aay from getting to the finals.
21:28.12warthogI just bought one yesterday
21:28.20endercool
21:28.40warthogno ps2 keyboard
21:29.19enderyeah just usb
21:29.42warthoglooking forward to trying the new 24xx from digium
21:30.04enderFree Developer Workstation. Sign up for three years of Sun services subscription at $29.95 per month and get a free Sun Ultra 20 Workstation.
21:30.11enderdamn.
21:30.30mog_homeyay!
21:31.26warthogwhat do you get for the service subscription, that works out to pretty much the price of their low end ultra20 with 2 disks in it
21:32.29enderwarthog: no idea.
21:32.41enderI wouldn't buy a sun though.  No reason to pay the extra $$ for the pretty case.
21:33.07*** join/#asterisk file[ontrain] (n=jcolp@207.231.238.3)
21:33.13warthogwhere can you get a better pc for less?
21:33.52warthogI think they are asking for 799 for their entry level opteron
21:34.12enderyou can buy the parts yourself.
21:34.27SkramXwhy do you NEED a sun?
21:34.43warthogdid that for years, find that I spend more time fiddling that working when I do that
21:34.52enderfile[ontrain]: when does the other train leave and when do you met?
21:34.56endermeet that is.
21:35.09harryvvanyone running the calling card option of asterisk
21:35.18enderwarthog: pfft.  Buying the wrong stuff then (:  I spend 1/2 a day putting it together, 2 years using it.
21:35.20file[ontrain]it left here at 5:30PM
21:35.25file[ontrain]I'll arrive at 8:15AM Montreal time
21:36.04enderwarthog: it's $895 for the bottom line Ultra 20, you still have to select a video card.
21:36.29warthogyou get a 2d video included, only extra if you get a 3d board
21:37.10*** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net)
21:37.15enderwarthog: yeah, what are you going to do w/ a 2d desktop?
21:37.53warthogI don't really ever play games, never really got bothered by 3d
21:38.35enderI could get a custom system from Pogo Linux that has 3 year warranty too for same price range.
21:38.45warthogI could not find a reasonable priced server with 12volt and ataraid so I am going to use the ultra20 for my mid range servers for viop
21:38.58*** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
21:39.01enderfor $1800 you can get a dual opteron workstation, WITH a 3d card.
21:39.19Drukenanyone know what causes     -- Got SIP response 302 "Moved Temporarily" back from  ??
21:39.31harryvvender, who wants a opteron system?
21:39.56warthogin 32 mode it it still VERY fast for the buck
21:39.56enderharryvv: we're discussing the Sun Ultra20 workstation which is a single opteron.
21:40.15enderwarthog: can you rackmount the sun ultra20 chassis?
21:40.45alephcom[Lloydmiharyvv:  As in ASTCC?
21:40.53harryvvI built my opteron system.
21:40.59*** join/#asterisk tsume (n=tsume@zanshin.tsumelabs.com)
21:41.01harryvvalephcom yes, astcc
21:41.10tsumeoi
21:41.50alephcom[Lloydmiharryvv: I have customers using it.
21:42.09tsumehey, curious. I just added more zaptel cards. The DTMF detection was working. These are all incoming lines. Now the DTMF detection just doesn't work :) I can sit here and induce tones and nothing happens. Any clues?
21:42.41warthogender, I have looked for a rack mount and have not found one yet, I am talking to a sun engineer about it though, they are thinking about it for me.
21:43.36tsumedamn :) all I did was add 2 more modules :)
21:43.38enderwarthog: I custom order mine through Pogo.  SuperMicro makes a workstation chassis that turns on it's side and has rails to rackmount.  Hotswap sata disks, triple redundant power supply optional, lots of room for pci slots and extra power leads.
21:43.39tsumethis bites
21:43.55tzangerender: I have the SCSI version of that
21:44.00tzangerit's huge and it's fucking heavy
21:44.01tsumenfc what could even cause DTMF problems
21:44.04enderwarthog: you can fit opteron or xeon chassis in them.  We use dual opteron chassis.
21:44.07harryvvalephcom[Lloydmi how long you been using it and also is it generating a profit?
21:44.11tzangerwhat kind of DTMF problems
21:44.12endertzanger: but it racks real nice.
21:44.17tzangeryes, yes it does
21:44.27enderand can easily host a few zaptel cards.
21:44.47*** join/#asterisk santiago (n=santiago@63.245.87.62)
21:44.48tsumetzanger: I added 2 FXO modules, and now none of the lines work properly. No DTMF tones are working at the call tree
21:45.40warthogender, I will check with POGO, but being in Bermuda, our support options are very limited.
21:45.43tsumefunny I can call out and call in though. I thought if the cards would stop worknig, they would just break, not be ghosty
21:45.54tzangertsume: have you played with the gains?
21:45.57enderwarthog: does Sun give you on-site ?
21:46.20tsumetzanger: no, but they seemed fine
21:46.21warthogyup, and for not too bad a price either, we have a sun dealer local
21:46.28endernifty
21:46.41tzangertsume: try rxgain=3.0 (and restart asterisk) to see if that cleans up the receive DTMF issue
21:46.48tzangerthat doubles the volume on whatever comes in off the line
21:47.21tsumetzanger: someone turned it previously to 8.2
21:47.31tzangertsume: ??
21:47.39tsumetzanger: rxgain=8.2 :)
21:47.58ender*blink*
21:48.41*** join/#asterisk flashnet (n=flashnet@62.79.123.79.adsl.hvi.tiscali.dk)
21:48.41warthogender, it took me weeks to make up my mind on the ultra20, I guess it was the local support and price, I ran autodesk viz render farm on opteron 2.4 single vs intel 3.6 ht, the intel only just beat it out by a fraction on a 20 minute render job, the opteron cost WAY less and can be upgraded to support 64 bit os, so I was pretty happy with them all around.
21:49.09enderwarthog: yeah, opteron is great for price.
21:49.32tsumetzanger: heh, I've never really pondered this, but how high can rxgain go?
21:49.35enderwarthog: your support situation means Sun is a far better option if you can get local support.
21:49.55warthogender. do you think  your guys could rig up a rack mount kit for my ultra20?
21:50.06alephcom[Lloydmiharryvv:  Go ahead and email me @ darren@aleph-com.net if you have questions on astcc.  I may be able to help.
21:50.11enderwarthog: probably not.
21:50.17tsumetzanger: having it set to 8.2 does seem.. :) odd. I wonder why it was set like that in first place
21:50.19enderwarthog: they're not a machine shop.
21:50.55enderwarthog: I'd buy a rack shelf, and then figure out a way to mount some ears on the sparc chassis.
21:52.12warthogender, I am having trouble getting the agi script at http://www.voip-info.org/wiki/view/Asterisk+authenticate+using+voicemail+passwords working, any chance you can take a look at it?  It is the first one writen in perl.
21:52.21harryvvwarthog do you do alot of 3d rendering?
21:52.41enderI would, but A) I don't know AGI, and B) I don't really know perl.  I'm a python guy (:
21:52.55warthogharryvv, I have one client, a good one, that renders all the time
21:53.07tzangertsume: set tx/rxgain to 0.0 and restart *
21:53.26harryvvworthog, I see so you build the equipment for him?
21:53.31tsumetzanger: the clients must use horns to hear if its set to 8.2, but it still didn't work at 3.0 nor at 16
21:53.46tsumetzanger: I'll try it. Maybe it is a context problem though?
21:54.08tzangertsume: I am assuming that the call is answered and you're getting the right prompts, just not hearing their dtmf
21:54.09warthogharryvv, nothing to special, I just setup the autodesk software and setup the render manager and farm and let them go hogwild on it
21:54.28tsumetzanger: correct. *sob* I've never had this happen
21:54.42tzangertsume: then it's likely a gain issue
21:54.42tsumeI've several boxes. I hate ghosts! :)
21:54.44*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-68-21-46-215.dsl.sfldmi.ameritech.net)
21:54.44tzangerstart at 0.0
21:55.18harryvvI see
21:55.33harryvvI have done my share of 3d modeling over the years.
21:56.16warthoghaarryvv, I just hate autodesk for ONLY supporting windows
21:56.25harryvvI see
21:56.51tzangerharryvv: what do you use?
21:56.57tzangerI'd like to make some simple cases
21:57.03tzangerlike www.mixdown.ca/~andrew/dump/case1.jpg
21:57.10*** join/#asterisk techie (n=gus@70.86.57.50)
21:57.14tzangerblender3d's a little much I think
21:57.41Drukenanyone know what would cause a 302 Moved Temporarily?
21:57.47harryvvtz, rhino3d
21:58.32tzangerahh win32
21:58.33harryvvbeen using it for years.
21:58.59tsumetzanger: :( seems to be just channel 12. heh
21:59.07tzanger"channel 12" ??
21:59.10tzangerwhat hardware is this
21:59.31tsumetzanger: is a zap channel
21:59.43tsumethis server is loaded with zap cards
21:59.43harryvvgive me some concept to model and I will do it.
21:59.48tsumethey really need a t1
22:00.50tzangertsume: what hardware
22:01.15*** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net)
22:01.40tsumetzanger: 100P cards I believe
22:01.51tsumetzanger: minute, debugging 12
22:01.51tzangerT1?
22:01.58tzangeror twelve X100P
22:02.23tsume4 X100P cards
22:02.52tzanger?
22:02.59tzangerno 4 TDM400P cards
22:03.01tzangerif there are twelve channels
22:03.02tsumeone is empty, but that shouldn;t matter
22:03.05tzangerand that's crazy
22:03.15tzangerwow that's a lot of PCI load, it's working fine except for the last channel?
22:03.21tsumetzanger: correct
22:03.30tsumetzanger: I'm using older driversx, there might have been changes
22:03.30tzangertsume: take the phone line connected to #12 and put it in #11
22:03.34tzangerdoes the problem go away or stay?
22:04.39tzangerharryvv: unfortuantely US$500 is too rich for my blood
22:04.55tzangerwe have inventor at the office but I'm not gonna try to learn something that heavy :-)
22:05.10harryvvyes
22:05.10harryvv:)
22:05.20harryvvlet me show you one thing I created recently
22:05.56harryvvmembers.shaw.ca/glyfx3d/18floorsper.jpg
22:07.09harryvvwant real looking metal ? members.shaw.ca/glyfx3d/bearing3.jpg
22:07.13harryvvanyway
22:07.30fugitivonice, it looks like an elf modern building
22:07.51harryvvelf modern building?
22:07.58fugitivoyes
22:08.03harryvvWhat do you mean
22:08.07fugitivothe design
22:08.33harryvvexplain elf
22:09.17tsumetzanger: bad module or card
22:09.20tsumetzanger: *sigh*
22:09.28tsumetzanger: I switched 12 to 11, and now it works
22:09.30tsume:(
22:09.56tzangerharryvv: wow, nifty
22:10.03tzangertsume: honeslty why the hell didn't you try that first?
22:10.15tzangerharryvv: so you do a lot of 3d modeling work?
22:12.42harryvvno. Need real render software which is costly.
22:12.51harryvvBut my modeling is very good.
22:13.08harryvvI have one of my images on rhino3d gallery.
22:13.17*** join/#asterisk clive- (n=pirch@ndn-165-147-172.telkomadsl.co.za)
22:13.26tzangerwow awesome
22:13.38tzangeras I get this case design inputted mind if I bounce some questions off you?
22:14.08harryvvsure
22:14.15harryvvwhat kind of case?
22:14.18tzangerI haven't even started yet admittedly
22:14.24tzangerharryvv: http://www.mixdown.ca/~andrew/dump/case1.jpg
22:15.07clive-tzanger whats t for?
22:15.10clive-it
22:15.16tzangerclive-: industrial protocol gateway
22:15.26tzangermodbus <--> anything CAN- or Ethernet-based
22:15.47tzanger(so DeviceNet/EthernetIP, ProfiBus-DP, CANOpen and Modbus/TCP)
22:15.51clive-tzanger sounds like fun
22:16.02tzangeryeah except the fucking CAN controller's giving me issues
22:16.06tsumetzanger: I *thought* it would have worked fine :)
22:16.06tzangerCan4Linux kind of sucks :-(
22:17.07tsumetzanger: thanks though. You helped me think while I was looking at the screen with verbose on
22:17.54tzanger:-)
22:18.04tzangertypically if I'm having trouble with one port and others are working I swap ports to see
22:18.16tsumeI'll have to log my journey down to a site
22:18.20*** part/#asterisk warthog (n=nvadekar@216.249.38.88.ppp.northrock.bm)
22:18.47tsumejourneyman status for phone technician still. I've only worked with Nortel and Avaya for 6 months, training on the job.
22:18.52tsumeasterisk for 3
22:19.44tsumeasterisk is a dreamboat compared to nortel equiptment. ESP for prices, ATA converters are just too 300USD too expensive for me :)
22:20.56fugitivotsume: nortel is cheaper?
22:21.22*** join/#asterisk file[ontrain] (n=jcolp@207.231.239.119)
22:22.54fugitivothat's cool, irc on a train
22:23.11file[ontrain]yeah
22:23.16file[ontrain]when one has a digital signal :)
22:23.27tzangertsume: do you know much about the internal workings of Nortel's SL1 protocl?
22:23.28file[ontrain]I think we're in the middle of nowhere
22:23.33clive-file watch out for the tunnel...:)
22:23.38fugitivowhat kind of connection is that?
22:23.49file[ontrain]bluetooth to my cellphone
22:23.53file[ontrain]CDMA 1XRTT to my provider
22:24.08fugitivonice, is that expensive there?
22:24.18file[ontrain]I found a flaw in my provider so it's free
22:24.22fugitivolol
22:24.31lunkhaha
22:24.35fugitivoshare it with us
22:24.37*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
22:24.49file[ontrain]they don't block their WAP browser account
22:24.56file[ontrain]so I have full internet access, unfiltered
22:24.56tzangerfile[ontrain]: whereabouts are ya
22:25.03file[ontrain]sweet
22:25.05file[ontrain]full digital now
22:25.12file[ontrain]tzanger: somewhere between Moncton, NB and Montreal
22:25.15fugitivoand the wap browser account is free?
22:25.17file[ontrain]where exactly? not a clue.
22:25.19file[ontrain]fugitivo: yeah
22:25.28lunkwhat provider?
22:25.31file[ontrain]they bill per page through their proxy
22:25.37tzangerfile[ontrain]: smart ass
22:25.41FuriousGeorgeis there any way to augment the talk bat on an fxs?  i got a doorphone thats a real REN hog when i turn the speakerphone volume up
22:25.52fugitivobut you don't visit any page, lol
22:25.52file[ontrain]tzanger: well I honestly don't know
22:25.59FuriousGeorgeif i turn it up too high the connection drops
22:26.02file[ontrain]fugitivo: yeah, never goes through their proxy
22:26.02tzangerFuriousGeorge: yes.  boostringer=yes
22:26.09tzangerFuriousGeorge: when you load the wctdm module
22:26.35file[ontrain]AHA
22:26.38tzangerfile[ontrain]: go ask the engineer :-)
22:26.39file[ontrain]we're in Rogersville, New Brunswick
22:26.43fugitivotzanger: what's that for?
22:26.50tzangerfugitivo: sorry that was for FuriousGeorge
22:27.24fugitivotzanger: i know, i'm just curious :)
22:27.32tzangeroh
22:27.38tzangerit brings the ringing voltage up on the wctdm card
22:27.44FuriousGeorgetzanger: so in my case /etc/modules.autoload/kernel-2.6 gets a line "wctdm boostringer=yes"
22:27.48tzangeryou can also break into the driver source and do funky things :-)
22:27.51tzangerthe SLIC's pretty cool
22:28.02FuriousGeorgemaybe you can :)
22:28.59tzangerFuriousGeorge: yes that's what you do
22:29.04clive-can someone please remind me what is a "VNAK"?
22:29.07tzangerand check dmesg to make sure you got it right
22:29.23tzangerclive-: sounds like what you do when you taste something awful
22:29.29tzangerspit it out with a "VNAK" kind of noise
22:29.31tsumeFuriousGeorge: nortel is certainly not cheaper
22:29.34FuriousGeorgetzanger: im also gonna run new wiring, thicker guage, to see if i can decrease the resistance
22:29.42tzangerFuriousGeorge: I wouldn't worry about that
22:30.03tzangerFuriousGeorge: the ohm-per-foot difference between like 26 and 22AWG is not really significant
22:30.06tsumetzanger: I don't know of the internals. my friend Roger Legg knows more on the internals. I think he was a programmer for them.
22:30.10tzangerunless you're talking hundreds or thousands of feet
22:30.13clive-tzanger..lol....I am getting lots of VNAK messages on my asterisk command line...wondering what they mean
22:30.33tsumetzanger: he had me step through a section of the BIOs one tmie looking for a password reset
22:30.37FuriousGeorgetzanger: but right now i got such a nasty hodgepodge of cat 1 and cat 5 running audio around the building
22:30.41fugitivoclive-: what did you asterisk eat?
22:30.59tsumetzanger: I gave him the hex values, which were encrypted, and he decrypted them for me :D
22:31.36tzangerclive-: odd
22:31.37tsumeI don't understand why one would need to encrypt information in a system.. like whose really going to hack it :D
22:31.47clive-its sending VNAK's...not receiving...wonder hwat its about
22:31.53file[ontrain]I think I'm going to lose digital service again in a bit
22:32.05tzangertsume: :-)  well I'm looking for more of the protocol spec on the fiber cards or even the protocol that runs on the DNs themselves.  (FUMP I think it's called)
22:32.16tzangerFuriousGeorge: yeah step one si to clean it up
22:33.00tsumetzanger: I can ask him if he has them
22:33.21tsumetzanger: he lets me use his freebies he gets. He is a reseller and tech
22:33.27tzangertsume: if you could...  I'd owe you a lot of ${YOUR_FAVOURITE_ALCOHOL} :-)
22:33.41tzangerbasically protocol manuals or documentation
22:34.03tzangerSL1 is the official name of the protocol I think but they "export it" to something called MCDN or NAPN, that's what they call the key codes anyway
22:51.44*** join/#asterisk file[ontrain] (n=jcolp@207.231.238.87)
22:51.47*** part/#asterisk MatsK (n=root@55.80-203-80.nextgentel.com)
22:55.22Corydon76-homeEvening, kram
22:55.33tzangerwerd to the kram
22:55.39kramevening corydon & tzanger :)
22:56.01tzangerI think he's come to beat me up over that dialstatus bug I reopened... :-)
22:57.06*** join/#asterisk nagl (n=nagl@213.235.241.6)
23:00.24*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)
23:04.57*** join/#asterisk zotz (n=zotz@24.231.47.168)
23:05.19*** join/#asterisk JunK-Y (n=junky@69.156.218.158)
23:16.28*** join/#asterisk GXTi (n=omgwtfbb@freenode/developer/GXTi)
23:18.39*** join/#asterisk bmg505 (n=leon@rndf-146-51-95.telkomadsl.co.za)
23:29.48*** join/#asterisk emp (n=emp@70.57.239.38)
23:31.58delmarok, i have some signal level problems and need to adjust them on the TDM card, anyone here know much about that?
23:33.08*** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv)
23:34.39*** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net)
23:50.48*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
23:53.16*** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
23:53.22delmarLooking for some help with fxotune and /etc/fxotune.conf etc.
23:54.17*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
23:58.34*** join/#asterisk nexis (n=nexis@12-219-60-253.client.mchsi.com)

Generated by irclog2html.pl by Jeff Waugh - find it at freshmeat.net! Modified by Tim Riker to work with blootbot logs, split per channel, etc.