irclog2html for #asterisk on 20070126

00:00.02Corydon-wsame context, same extension, you'd have a label conflict
00:00.40ThoMeCorydon-w: any ideas?
00:00.41De_Monits unique to that extension is what I'm hearing
00:00.59Ritalin2Corydon:  what thome wants is to dial an extension (say 41) with ** prefexed onto the exten number
00:01.14Corydon-wThoMe: ${EXTEN:2}
00:01.15Ritalin2s/prefexed/prefixed/
00:01.35Corydon-wThoMe: please read doc/README.variables
00:01.47ThoMeCorydon-w: yes i make it.
00:01.53De_Monthats weird
00:01.57ThoMeCorydon-w: exten => _**XX,1,Dial(SIP/${EXTEN:2},30)
00:02.00ThoMeis this correkt?
00:02.09Corydon-wThoMe: have you tried it?
00:02.28ThoMeCorydon-w: yes :-( > Jan 26 01:02:25 NOTICE[12612]: chan_iax2.c:7331 socket_read: Rejected connect attempt from 194.231.22.135, request '**50@from-inside' does not exist
00:02.40*** join/#asterisk mphill (n=mphill@h69-129-245-188.69-129.unk.tds.net)
00:02.49Corydon-wThoMe: then you haven't reloaded
00:03.23Ritalin2thome: ich wuste mein deutsch war besser :(
00:03.38ThoMeCorydon-w: i have reload. when i dial: **50 then i have the error  :/ '**50@from-inside' does not exist
00:03.45ThoMeRitalin2: hm? :-9
00:04.18Corydon-wThoMe: please post your entire extensions.conf on pastebin.ca
00:05.17mphillhas anyone written a comparasion of trixbox and asterixnow on the net?  I am looking for a site that has a list each features.
00:05.32matt_does anybody know of a dialplan i can use for a linksys pap2 that will send everything to the asterisk box and still allow me to use # at the end of the number?
00:06.01Strom_Cx.#
00:06.04Strom_Cer
00:06.10Strom_Cx.#|x.T
00:06.17matt_ok cheers :)
00:06.43matt_what is the T
00:06.43*** join/#asterisk awannabe (n=brad@216.161.165.212)
00:06.45matt_timeout?
00:06.48De_Monblarg. I created a bunch of labeled extensions and reloaded extensions and didn't get any warnings. Whereas I can't add extensions with labels at all using the CLI
00:06.48awannabehi all
00:07.16De_Monadd extension foo1,n(foo),NoOp(test) into local
00:07.21De_Monsee anything wrong with that line?
00:07.29Corydon-wYes
00:07.35De_Mon:(
00:07.42Corydon-wfoo1,n(foo),NoOp,test
00:08.00De_Mon() is prettier
00:08.03Corydon-wCLI uses old-style
00:08.10De_Monum
00:08.22*** join/#asterisk Zorix (n=Brandon@6-24.201-68.swfla.res.rr.com)
00:08.24De_Montry again
00:08.34De_Monit works with either
00:08.42awannabewhen you assign a SIP peer a defaultIP, they dont have to register with the server at all then?
00:08.48Corydon-wDe_Mon: "help add extension"
00:08.52*** part/#asterisk mphill (n=mphill@h69-129-245-188.69-129.unk.tds.net)
00:08.54De_MonI bet it doesnt like the n
00:08.54Zorixis it possible to get the GUI from asterisknow on regular asterisk since i plan on loading asterisk on an existing machine with other uses as well
00:09.03Corydon-wprobably not
00:09.20De_Monpbx*CLI> add extension foo1,1,NoOp(test) into local replace
00:09.20De_MonExtension foo1@local (1) replace by 'foo1,1,NoOp,test'
00:09.32matt_Strom_C, ok now my pap2 is submitting a # after the number to asterisk
00:09.33hadsIt's just based on the source in asterisk-gui svn right?
00:09.44hadsSo it should be fine.
00:09.48Strom_Cmatt_: what are you trying to do, exactly?
00:10.04De_Monif I declare the same label twice it just replaces the last one doesn't it...
00:10.14matt_the default setup it fine i can dial numbers and press # to speed up dialing
00:10.15De_Monwell doesn't that suck
00:10.24matt_but now i need to dial a number with a * in it
00:10.38De_MonI wonder if thats a bug or a feature
00:10.47matt_but when i press * it sends the number i think
00:10.52matt_or does some funny stuff
00:10.54De_Monn(loop) do stuff jump set a new n(loop) and have a big party
00:11.06Corydon-wDe_Mon: adding via command line is obsolete
00:11.19*** join/#asterisk neddy (n=js152033@192.18.43.225)
00:11.34matt_if i can make the pap2 take off the # that would be perfect
00:15.06matt_yay it works now :)
00:15.06*** join/#asterisk dseeb_ (n=dcb@58.165.244.192)
00:15.19matt_i used ... (<#:>x.#|x.T)
00:16.16*** join/#asterisk zapx (n=zap@146-115-115-175.c3-0.lex-ubr2.sbo-lex.ma.cable.rcn.com)
00:16.41NeonLevelgood day everyone, i have sip trunk with one provider I have type=user for receiving the call, but when i call it, it wont enter the call, and sip debug says "Found no matching peer or user for xx", HOWEVER when i change this to type=peer then the call is OK, I need to setip up with type=user but won't work, has anyone had a similar problem, am i missing something?
00:18.51matt_its only matching numbers now and not *'s
00:19.13zapxi'm having problems getting asterisk/trixbox to register to sip provider... it seems like the packet never actually leaves the computer? i see it on a tcpdump on the actual computer... but on another computer that sees all network traffic i doesnt see any traffic on 5060... i do see traffic if i run a softphone to the same provider. I set it up for NAT, but it keeps trying to reconnect. Any ideas? I looked at the firewal
00:19.20zapxyesterday when I installed asterisknow, I was able to register to the sip provider, but i had other issues (under vmware).. such as choppy audio
00:20.15Corydon-wchoppy audio under vmware?  No, really?
00:20.38zapxits not choppy using trixbox under vmware
00:20.39zapxso yes, really
00:20.59Corydon-wvmware isn't a supported configuration
00:21.28Corydon-w~trixbox
00:21.29jbotmethinks trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/
00:21.52zapxCorydon-w: this is an asterisk question
00:22.06Corydon-wYou're using trixbox
00:22.14*** join/#asterisk rickead (n=richard@88-96-99-66.dsl.zen.co.uk)
00:22.14zapxCorydon-w: does trixbox not run asterisk?
00:22.34Corydon-wzapx: you're asking about some configuration using trixbox
00:22.40*** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net)
00:23.09zapxCorydon-w: I am asking why a packet that is sent by asterisk wont actually leave the computer
00:23.24Corydon-wMaybe it's not actually getting generated
00:23.41zapxCorydon-w: i see it on tcpdump though
00:24.00*** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk)
00:24.39Corydon-wFirewall, perhaps?
00:24.49zapxi checked iptables, it's empty, everything is set to allow
00:25.18Corydon-wDunno
00:25.44zapxokay, thanks anyways
00:28.35MrYi have asterisk installed on linux just just 1 ethernet port, w/o fxp or any other cards installed... can i use this box with iax softphone... to handle calls between iax softphone clients, how many simultaneous clients can a 2.8 p4 box w/ 2gigs of ram handle?
00:29.13rickeadmry: depends what codecs you will be using
00:29.31rickeadif theres no trascoding on g711 you can probably handle 100 or so
00:29.35*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
00:29.40rickeadmaybe more
00:30.20*** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com)
00:30.56MrYrickead: what codec use the least resource?
00:31.24rickeadg711 as there is no compression
00:31.51rickeadand you would need to make sure your phones were set to use g711 aswell, then your box doesn't have to do any transcoding
00:33.39*** join/#asterisk x86_ (n=x86@p3m/member/x86)
00:34.58cekcwhy are my zap config files getting overwritten on reboot =(
00:35.15Ritalin2ztcfg
00:36.39MrYg711  sound quality? does it require lots of bandwidth?
00:36.42*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
00:36.43*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
00:36.58J4k3g711 is toll quality, and toll bandwidth
00:39.17MrYi have a community website with about 10,000 members.. i want to give each member a voice mailbox where members can communitcate w/ one another...  hope asterisk can do handle it..
00:41.02*** join/#asterisk dasenjo (n=dasenjo@190.5.196.105)
00:42.43cekcmy ztcfg is looking right now, now do I tell it to save it so that it's there when I reboot
00:43.38Ritalin2cekc: what is it changing that's wrong
00:44.13cekcit's resetting the file to default
00:45.36Ritalin2cekc: trixbox ?
00:46.05cekcno it's that new asterisk thing
00:46.15Ritalin2asterisknow ?
00:46.33cekcyeah
00:46.49NeonLevelgood day everyone, i have sip trunk with one provider I have type=user for receiving the call, but when i call it, it wont enter the call, and sip debug says "Found no matching peer or user for xx", HOWEVER when i change this to type=peer then the call is OK, I need to setip up with type=user but won't work, has anyone had a similar problem, am i missing something?
00:47.31rickeadmry: asterisk should be able to handle the calls for that
00:47.43rickeadbut you should look into something like ser to handle the registrations
00:48.04rickeadyou should only really register a few hundred devices per * box
00:49.25putzzcekc: wich file ?
00:49.47cekczapata.conf and zaptel.conf are getting reset by init.d/zaptel
00:50.58putzztry editing the _additional.conf files instead
00:51.04*** join/#asterisk angom (n=angom@red-corp-201.143.59.181.telnor.net)
00:51.54cekcI don't have an additional file yet.  what about things like signaling, wouldn't two entries for that be bad?
00:52.08putzzyes
00:56.17*** join/#asterisk rikstah (n=rick@host81-155-179-213.range81-155.btcentralplus.com)
00:56.29*** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net)
00:56.32*** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch)
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01:01.39*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
01:01.39*** mode/#asterisk [+o mog] by ChanServ
01:02.11*** join/#asterisk fiber0pti (n=John@207.114.199.107)
01:02.30fiber0ptihow can I print some text in the CLI from the dial plan (extensions.conf)?
01:03.00Ritalin2pastebin
01:03.06cekcit's this zapscan program that's resetting my files
01:03.31matt_fiber0pti, NoOp
01:03.40matt_fiber0pti, http://www.voip-info.org/wiki-Asterisk+cmd+NoOp
01:03.50fiber0ptity
01:05.39*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
01:09.35dlynes_laptopHas anyone had issues with mwi indication not working on the latest version of 1.2 series asterisk?
01:10.01dlynes_laptopon sip channels that is?
01:10.13*** part/#asterisk Lokiji (n=Lokiji@ip-89-102-178-195.karneval.cz)
01:10.27nighty-does anyone knows a way to do push to talk with asterisk  ?
01:10.37nighty-and also the appropriate phones :)
01:11.11nighty-I am looking at using wifi phones for this
01:12.02nighty-using SIP
01:12.04dlynes_laptoppush to talk?
01:12.10nighty-yes
01:12.14dlynes_laptopyou mean intercom?
01:12.25nighty-walkie talkie like functions
01:12.40nighty-one to many
01:12.49nighty-half duplex
01:13.00*** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com)
01:13.01nighty-you push to talk , everybody hears
01:13.07nighty-then someone answers
01:13.09nighty-etc....
01:13.29nighty-only the persons that talks (pressing a button) can speak
01:13.33dlynes_laptopso group paging
01:13.44nighty-uhmmm
01:13.52dlynes_laptopor do you want the other end to be able to talk back?
01:14.01nighty-I want the other to talk back
01:14.07nighty-just like a walkie talkie
01:14.19dlynes_laptopah....never heard of anyone getting that to work on asterisk before
01:15.03nighty-uhmmm
01:17.55hardwireanybody here using dacs with zaptel?
01:18.11*** join/#asterisk yxa (n=lonari@58.185.90.101)
01:18.23yxaJT hey
01:19.35*** join/#asterisk rue_mohr (n=rue@h24-207-91-121.cst.dccnet.com)
01:20.38rue_mohrIf I'm connecting a digium pci T1 card to a newbridge mainstreet 3624, I need a crossover T1 cable, dont I?
01:21.30*** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com)
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01:29.30*** part/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net)
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01:36.52*** join/#asterisk sudhir492 (n=sudhir@c-71-63-59-45.hsd1.va.comcast.net)
01:36.55sudhir492Hi All
01:37.07sudhir492Anyone owns a PS3 here ?
01:37.19syzygyBSDanyone going to admit to owning a ps3 here?
01:39.09JTrue_mohr: probably
01:39.14JTyxa: hi
01:39.53J4k3I'd rather admit to a PS3 than a PS/2 ;)
01:40.06yxaJT i just got a reply from my telco: "not sure where you got the info from, but for ISDN-2, you cant subscribe to MSN and Multi-Line Hunting at the same time"
01:40.32*** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net)
01:41.02JTthat's only company policy, not technical capability
01:41.35sudhir492It will be cool to run Asterisk on PS3
01:41.46J4k3heh
01:41.48J4k3not really
01:41.53sudhir492better still, if G729 coded can be ported there too!
01:41.54J4k3Sony seriously overhyped the processor
01:42.10yxaJT how should i reply him? :)
01:42.12J4k3its not *that* good... if it was, the PS3 would actually cost real money.
01:42.32sudhir492J4k3: But it does have quad pwer pc
01:42.41syzygyBSDmeh, the last gen technology console is winning...
01:42.47J4k3sudhir492: I could stick four 405GP's on a PCB too.
01:43.50sudhir492Just the Power PCs do have a lot of processing power for the price of $499/$599
01:44.11*** join/#asterisk Ridgeback (n=jircii@65.127.64.52)
01:44.22J4k3for $499 I can buy a core2duo e6400-based system, or a lowish end dualcore amd64.
01:44.42J4k3and not deal with sony's sketchy hardware, iffy support, etc.
01:45.07Ridgebackhello, anyone here help with IAX switch() statement using AEL v2?
01:45.19sudhir492J4k3: what is e6400?
01:45.32J4k3sudhir492: fast
01:45.33J4k3;)
01:45.46J4k3the next-from-the-slowest core2duo
01:48.17sudhir492I have to build a system that can handle upto 200 simultaneous calls, (all SIP or IAX), no transcoding. Looking for the cheapest solution, yet sufficient enough.
01:48.48J4k3I'd say that'd depend on the level of options you want to offer
01:49.31J4k3because 200 w/o transcoding sounds like... Pentium 166 territory... unless you want to pile on the services.
01:50.04Ridgebacktrying to use switch to push _7XXX toward my other asterisk switch. how would i do that in ael?
01:50.35J4k3if you're buying new hardware... theres a point where its not reallly smart to buy cheaper because you're saving very little and losing a lot.  Where that line is today will handle a LOT of traffic, at least from what I've gathered.
01:50.41*** join/#asterisk hellop (n=hellop@udp115314uds.hawaiiantel.net)
01:50.44sudhir492The Asterisk will be in the audiopath, and it will be doing minimal transcoding for voicemail
01:51.14perdthat's wicked official
01:51.20yxaJT what's the difference between MSN and DDI
01:52.01sudhir492J4k3: I was considering Dell 1950, with 2 Dual Core Xeon processors, about $2500. But is there a better option
01:53.43Ritalin2I just got a new system yesterday with a e6400 in it to replace an athlon 3000+ (ditching agp was a big reason to get rid of it).  day to day use it doesn't appear that much faster than the athlon.  i haven't done any benchmarks though
01:53.57DavoFrom818man i tryed asertisknow and it sucks so bad you cant even imagine
01:54.13Ritalin2Davo: what sucks about it
01:54.37DavoFrom818it doesnt work!
01:54.46sevardDavoFrom818: I work, for money.
01:54.54DavoFrom818it is going to take alot of time before it reaches freepbx
01:55.05Ritalin2it doens't work period?
01:55.08sevardwow, that's one hell of a high bar there babbo
01:55.14DavoFrom81880 not working
01:55.20*** join/#asterisk nighty- (n=nighty-@66-163-28-100.ip.tor.radiant.net)
01:58.56[TK]D-FenderLast I recall, the AsteriskNow GUI doesn't do EVERYTHING for you like FreePBX does, so there's a little more room to exten incompetence into....
01:59.42J4k3sudhir492: that'd be serious overkill really
01:59.53J4k3I'd think one good dualcore chip would be plenty of horsepower.
02:01.45DavoFrom818VoiceOne is nice
02:01.59DavoFrom818i love the clean gui they have
02:02.50dlynes_laptoppersonally, i like vim...does a much better job
02:03.25sudhir492J4k3: For voicemail, there could be a few channels doing transcoding, and FlashOperatorPanel, etc. will take some CPU too. I guess 1950 should still be sufficient
02:04.32Ridgebacktrying to use the switch command to push _7XXX toward my other asterisk switch. how would i do that in ael? The docs are not clear.
02:05.19sudhir492J4k3: What if I go with 2 Dual Core AMD opteron (1.8GHz) with 2GB memory. Will that still be sufficient?
02:05.31[TK]D-FenderRidgeback : Try it the way you think is should be done, and in * CLI do "show dialpla" and see if it parsed properly.
02:06.07dlynes_laptop2GB of RAM I would think is total overkill
02:06.43J4k3sudhir492: that'd still be overkill...  send that box to me, I want it ;)
02:06.56Ridgebacki tried    _7XXX => Switch(IAX/serverB);     but asterisk complains and says Switch() not a valid command.  although aelparse -n  showed no problems
02:07.44[TK]D-FenderRidgeback : Keep in mind that in NORMAL dialplan logic "switch" ISN'T an application.  keep that in mind.
02:07.50[TK]D-Fender*HINT*
02:08.38[TK]D-FenderRidgeback : http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers
02:08.49Ridgebackperhaps _7XXX => (switch/iax/serverb);  ???
02:09.15RidgebackI read through that, but AEL states switch statements are handled differently than old extensions.conf
02:09.42[TK]D-FenderRidgeback : you are clearly not paying attention at ALL as to how it gets called NORMALLY.
02:10.28Ridgebackok then shw me how it's supposed to work in ael.
02:11.07[TK]D-FenderRidgeback : Read "Example 3" in there again.
02:11.30dlynes_laptopRidgeback, switch => ...
02:11.38dlynes_laptopRidgeback, not => switch(...)
02:12.07[TK]D-Fenderdlynes_laptop : SHH!!! He's missing DIFFERENT big-print.  Let him come to it...
02:12.19dlynes_laptopheh
02:12.21Ridgebackin AEL, they state the switch statements have to be in thier own context
02:12.42dlynes_laptopRidgeback, same for extensions.conf
02:13.25Ridgebackhmmmm
02:15.47*** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net)
02:19.41[TK]D-Fender*sigh*
02:21.29perdsie
02:22.29*** join/#asterisk seva (i=seva@66.90.103.12)
02:22.42sevawhat is record_in and record_out config option in iax.conf?
02:22.44sevai can't find any docs for it
02:22.50sevaand my 1.4 source tree doesn't have it once
02:23.04sevawas this something that was removed pre 1.4?
02:23.45perdi dont see those config options
02:23.55perdat least not in the sample.conf
02:24.30sevait's mentioned all over the web in various samples
02:24.33sevabut never explained
02:24.34*** join/#asterisk zakolus (n=zakolus@c-68-52-111-175.hsd1.tn.comcast.net)
02:24.48sevai've seen it set to never or always
02:24.51zakolusHello Ateriskians, I am in a bit of a pickle and need help
02:24.54sevaer never or adhoc
02:25.05perdmaybe a bunch of people just copied an incorrect config :)
02:25.10*** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com)
02:25.18sevamore likely it was in 1.2? :)
02:25.31perdnope
02:25.37perdim looking now
02:25.41zakolusCan anyone help me?
02:25.44perdnot in my sample conf files.. unless it's just a mystical option
02:26.05sevaweird
02:26.15perdnow you've got me wondering though
02:26.17zakolusI guess not XP
02:26.23perdhttp://forums.digium.com/viewtopic.php?t=12990&highlight=recordin ?
02:26.31perdthat's the best i can find :P
02:26.40sevaheh
02:26.52sevathat answer is so money
02:26.55perd[root@localhost asterisk-1.2.14]# grep -R record_in *
02:26.55perd[root@localhost asterisk-1.2.14]#
02:27.10sevadoes * ingore options it doesn't recognize?
02:27.25perdi think so
02:27.27sevalike if i added poo=eat
02:27.30sevahrm
02:27.30perdtry it
02:27.58perdyeah it does
02:28.04perdi just set hugedongs=awesome in sip.conf, no problems
02:28.08sevaheh
02:28.45perdi checked 1.4.0 source tree, no record_in found
02:29.04[TK]D-Fenderperd : careful what you say around hre... you never know who might be lurking.....
02:29.16perdhmm?
02:29.22sevardhahahah here's a link to
02:29.25sevard"file:///C:/Documents%20and%20Settings/Owner/My%20Documents/My%20Webs/old345websites/internship345courseportfolio/Tutorials/howtomakeatemplate.htm"
02:29.36perdapparently they failed the template class
02:29.59[TK]D-Fenderperd : Hey... did you know the word "gullible" isn't in the dictionary?
02:30.04perdam i going to get banned because i said hugedongs?
02:30.05perdhaha
02:30.17perdor because i said the dogs were awesome?
02:30.35perddongs! not dogs! anyawy.. yeah it's written on the ceiling too, fender
02:36.05zakolusAsterisk hates me T_T
02:37.04*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
02:37.04*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
02:37.04zakolusCorydon-w: Why does Asterisk hate me? XD
02:37.24zakolusCorydon76-home: I meant to speak to y
02:37.27zakolusou
02:37.28zakolusXD
02:37.37*** part/#asterisk seva (i=seva@66.90.103.12)
02:37.44*** part/#asterisk zakolus (n=zakolus@c-68-52-111-175.hsd1.tn.comcast.net)
02:38.05*** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)
02:40.28*** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com)
02:45.44dlynes_laptopsounds like zakolus has a pickle in him; doesn't sound like he's in the pickle
02:47.52[TK]D-Fenderdlynes_laptop : Just a guy who doesn't realize that you get help a lot faster when you actually state your problem instead of whining that "nothing" works, and "everybody hates me"..  Oh... and then leaving :)
02:48.34dlynes_laptop[TK]D-Fender, and asking for specific people to help you, instead of anyone?
02:49.04dlynes_laptopperd, admit it...you like huge dongs
02:49.17dlynes_laptopi think your nick should be perv though, not perd :)
02:49.22[TK]D-Fenderdlynes_laptop : Yeah, thats guaranteed to win you Brownie-points for SURE.....
02:49.52[TK]D-Fender..... wow that can be read in just WAY too many ways :)
02:50.05dlynes_laptophahahaha
02:50.15[TK]D-Fender*shudder*
02:50.34SomeOne1sup homies and homierettes
02:50.37dlynes_laptop[TK]D-Fender, are you an asterisk developer?
02:51.44[TK]D-Fenderdlynes_laptop : No.... I can't seem to get the lens-cap off ;)
02:52.21Ritalin2why would the flash modifier in zapata.conf only be for non-PRI based T1's ?
02:52.56[TK]D-FenderRitalin2 : it would be for any ANALOG channel.
02:53.09Ritalin2Fender: ok that makes  alot more sense :^D
02:53.14[TK]D-FenderRitalin2 : Non-PRI T1 doesn't count.
02:53.18Ritalin2the docs just say "non-PRI T1"
02:53.50Ritalin2you'd think they would have written it as "any analog channel" haha
02:54.13[TK]D-FenderRitalin2 : the nature of flash requires a closed loop circuit, and not just a digital link.
02:54.38[TK]D-FenderRitalin2 : You'd think that well... YOU'D think, and the obvious stuff needn't be said :)
02:57.41Ritalin2well hopefully that'll solve my hook flash problem
02:57.53Ritalin2asterisk thinks i want to hang up
02:58.06J4k3I'm amazed.
02:59.28J4k3cdma2000 evrc -> vzw -> g729 -> asterisk (and xlite softphone on gsm) -> g729 -> vzw -> cdma2000 evrc on a 3-way conference...  I can actually understand people end-to-end.  This amazes me.
02:59.37[TK]D-FenderRitalin2 : add "noload => chan_psychic.so" into modules.conf.
02:59.41*** join/#asterisk HushPe (n=HushPe@mail.kamar.co.nz)
02:59.52J4k3evrc is generally awful
03:00.30[TK]D-FenderJ4k3 : "System of Surprisingly Compatible Flaws"
03:00.53J4k3yes, exactly.
03:01.27Ritalin2it's not trying to be psychic... it's trying to do what someone told it to :-p
03:02.36Zorixcan anyone suggest a cheap analog fxs adapter that works fairly well with asterisk just for a test
03:02.48Ritalin2sipura 3k ?
03:02.57Zorixmaybe a 2 port deal for a better test
03:03.03[TK]D-FenderZorix : Linksys SPA-2002
03:03.15[TK]D-Fender<70$USD if you look around
03:04.42Zorixthanks
03:05.21Zorixim more interested in analog modem/fax test for a possible environment that would require those
03:05.37Zorixbasically passing it thru to a fxo out to the telco
03:05.53dlynes_laptopZorix, just for a test?  not for a production environment?
03:06.07Zorixno if it was in production i would use the big digium cards
03:06.16Zorixor one of those massive things
03:06.19dlynes_laptopZorix, try the grandstreams then
03:06.30dlynes_laptopZorix, they're shitty quality, but they'll do what you want
03:06.41dlynes_laptopZorix, and they're probably the lowest price you'll find anywhere
03:07.04Zorixyea im lookin at voip supply
03:07.21Zorixtheir crappyness wont cause any problems for testing codecs right
03:07.27dlynes_laptopnope
03:07.32Zorixsweet
03:07.35dlynes_laptopjust voice quality
03:07.38dlynes_laptopand possibly echo
03:07.39Strom_CZorix: which codecs are you intending to test?
03:07.54Zorixthe 711 seems to be the best for analog modem/fax
03:07.59[TK]D-FenderZorix : Linksys SPA-3102 = 1 FXS + 1 FXO
03:08.11Zorixhmm
03:08.31HushPeZorix: i was under the impression that fax doesn't voip that well?
03:08.56Zorixwell it really isnt going across the internet, just to the machine and back out to the analog phone network
03:09.43Zorixi just wanted to try some real world tests before i can even think about it
03:09.51[TK]D-FenderZorix : If you want to use an analog fax machine you are strongly advised to leave it on its own analog line with NOTHING else touching it.
03:10.05Zorixwhere i work we have a rolm phone system thats not in good shape and i was lookin at voip
03:10.07Qwellnothing, including the rj11 cable from the wall
03:10.19dlynes_laptop[TK]D-Fender, then you take all the fun out of the combination though :(
03:10.52Zorixheh
03:11.02[TK]D-FenderQwell : Even better if its still packed in styrofoam in its box vacu-sealed and everything!
03:11.14Qwell[TK]D-Fender: exactly
03:11.21Zorixsome of the fax would be inter office so it would not help to hook directly to the telco
03:11.24Qwellmaybe use it as a footstool?
03:11.46[TK]D-FenderZorix : do a lot of inter-company faxing?
03:11.56[TK]D-Fender*inter-office
03:12.03Zorixyes
03:12.13dlynes_laptopmn
03:12.23Ritalin2doesn't asterisk capture incoming faxes as pdfs ?
03:12.24dlynes_laptopsounds like a very lazy company :)
03:12.30dlynes_laptopRitalin2, no
03:12.39Zorixno policies require it
03:12.47*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
03:12.54*** join/#asterisk descartes (n=user@ool-4352b247.dyn.optonline.net)
03:13.29Zorixso you are saying its going to be crappy and unreliable
03:13.34Zorixno codec will help
03:13.48JTprobably, unless you use T.38
03:14.06Zorixright but doesnt help for analog modems
03:14.16JTnot so
03:14.18Zorixscary enough some people use those for banking
03:14.23[TK]D-FenderZorix : It is a MISTAKE to let a busy fax run on anything but a dedicated analog line.
03:14.24JToh
03:14.35JTyou didn't say anything about non-fax modeming
03:14.41[TK]D-FenderZorix : How many sites for into-divisional faxing?  LD to each other?
03:14.50*** part/#asterisk wulfy814 (n=wulfy814@c-67-165-37-20.hsd1.pa.comcast.net)
03:14.55Zorixlike 25-30
03:15.24[TK]D-FenderZorix : if you pass that much, get a couple of HP MFP printers w/ their DSS software.
03:15.41Zorixmost are hp laserjet 3000 mfp series
03:15.43*** join/#asterisk cbullock81 (n=cbullock@adsl-227-34-99.jan.bellsouth.net)
03:15.47Zorix3320, 3380, etc
03:16.17HushPeis asterfax anygood?
03:16.30[TK]D-FenderZorix : I believe they have some new lower nodels that support it now.
03:16.44[TK]D-FenderZorix : I run a large bunch of 4345's personally.
03:16.44Zorixso thats used in error correction?
03:16.49HushPewe are thinking of using it in our office as we're more email oriented, but need a fax as people love to fax us crap
03:16.54Zorixweird never seen that model
03:17.04[TK]D-FenderZorix : its very popular.
03:17.06Ritalin2dlynes: what's the deal with faking fax tones with 666 or 7777 or whatever it is?
03:17.12Zorixwe have 4300s and 4350
03:17.16[TK]D-FenderZorix : high-volume workgroup printer on a budget.
03:17.18Zorixand some 4250
03:17.27[TK]D-FenderZorix : 4300/4350 are just dum printers.
03:17.32[TK]D-Fenderdumb*
03:17.34Zorixyea
03:17.38Zorixwe load them with jetdirect
03:17.42QwellJ4k3: can't email legal documents
03:17.54HushPeJ4k3: i deal with schools, they're back in noah's ark, and even noah hates faxes
03:18.00Zorixright legal documents must be faxed
03:18.11Zorixthis is actually a school i work at
03:18.11[TK]D-FenderZorix : They're decent mind you.. but the 4345 MFP does print/copy/scan/fax/ocr/etc....
03:18.21Zorixnice
03:18.48Zorixjetdirect kinda sucks.. they blow up for no reason all over the place
03:18.54HushPeZorix: you'll know the classic "email me a screenshot" to have it arrive on the fax so black you can't read it anyway ;)
03:19.03[TK]D-FenderZorix : I have mine set up for "remote print" so I can basically photo-copy to anywhere in the corporation, including targeting colour lasers,
03:19.04sevardZorix: legaldocuments can be scanned and emailed or scp'd or ftpd
03:19.20Zorixyea well users wont be bothered if thats even possible
03:19.24sevardi don't see the point of faxing anything, it's expensive and archaic.
03:19.26cbullock81Hey. D-Fender. Have you used any linksys phones before? (I dont have any crazy questions about them, just curious about their quality)
03:19.37HushPesevard: too true :)
03:19.41sevardcbullock81: they're pretty good.
03:20.12[TK]D-Fendercbullock81 : they are in 4th place on my preference list. Where are you located?
03:20.25cbullock81d-fender: mississippi, USA
03:20.32Ritalin2d: whats #1 ?
03:20.38HushPesnom?
03:20.41[TK]D-Fendercbullock81 : Screw  Linksys then.  Polycom all the way.
03:20.43Ritalin2I was probably going to buy a couple polycom ones
03:20.46Zorixfigure best way to test modem over voip is to find some old dos machines and get a dilup program/game going and see how much gets dropped lol
03:20.48Qwell[TK]D-Fender: 4th...wtf
03:20.53cbullock81im just looking for a good phone for less than $150
03:20.55Qwell[TK]D-Fender: list?
03:20.58sevard[TK]D-Fender: Aastra THEN polycom
03:21.13Qwellgrandstream > cisco > snom > polycom > linksys
03:21.17[TK]D-FenderQwell : Polycom, Aastra 480i, Cisco 7940+, THEN Linksys.
03:21.30cbullock81I love the polycoms i've setup, but they are too pricey for some customers
03:21.36JTgrandstream > cisco ?!
03:21.39QwellJT: :P
03:21.43Qwell</trolling>
03:21.43JTheh
03:22.02Qwellcisco (skinny) > polycom though
03:22.09HushPehey ya JT i have my asterisk pbx all sorted now :) just need another TDM card
03:22.11sevardQwell: please purchase a GP2X for me.
03:22.18QwellI think even [TK]D-Fender would admit that (assuming a proper skinny implementation, heh)
03:22.24[TK]D-Fendercbullock81 : Who said good VoIP gear came cheap?  the linksys stuff has iffy screen, poor feel, and lack of functionality.
03:22.30JTHushPe: yeah, i can't even remember what issue you were havuing :P
03:22.55cbullock81D-Fender: thats what i wanted to hear :) i was curious about the quality
03:22.59sevard[TK]D-Fender: good VoIP gear _should_ come cheap, I consider all the phones on the market to be highly overpriced at the moment.
03:23.02Zorixthis grandstream dual fxs adapter doesnt mention GSM codec that asterisk really likes.. is it supported?
03:23.14HushPeJT: my crappy irq problem, then a dodgy dial plan (it was a 'get it going' dial plan), i re-wrote it last night, and make some tweaks this morning... now it's been working for me all day :)
03:23.17[TK]D-FenderQwell : There is a possibility I'd agree were I to have real experience with Skinny, and care for the methodology.  However it violates my open standards requirements :)
03:23.29JTHushPe: ah yes, nice
03:23.34Qwell[TK]D-Fender: SIP may be open, but it's retarded
03:23.35[TK]D-FenderZorix : GrandSuck is to be avoided with extreme prejudice.
03:23.38yxahow can i write a script to restart asterisk only when there are no channels in use?
03:23.50Zorixok id do linksys then since its not much more
03:24.01[TK]D-FenderQwell : Sounds like your Commander-in-cheif :)
03:24.12Ritalin2do you find there's a big difference be aastra and polycom?
03:24.17JTyxa: "restart when convenient"
03:24.21sevardyxa: you could cronjob a script that checks for open channels and if none reboots itself
03:24.22JTyxa: that's the command
03:24.41JTrestarts only at nil call volume
03:24.45rudholmwhat about "restart when you feel like it"
03:24.56sevardrudholm: this is not asterisk for windows! silly.
03:25.04rudholmor "restart now unless you're feeling lazy"
03:25.15Qwellrestart ...eh, whenever
03:25.21J4k3hmm...  I think I'll add a bluetooth-capable cellphone as another trunk.
03:25.26[TK]D-FenderRitalin2 : Polycom is better physical, and audio quality, as well as standard suability.  Aastra wins on back-light and cheap XML-services, and included PoE deployments.
03:25.40[TK]D-FenderRitalin2 : Within the mid-range bracket.
03:25.50*** join/#asterisk chiang_sg (i=kodok@bb121-6-179-58.singnet.com.sg)
03:25.55Ritalin2why do people value backlight so much?
03:25.57sevardAastra is way better physically and audio, the aastra handsets FEEL like real handsets.
03:25.58[TK]D-FenderRitalin2 : Also ont he 480i's side is the vast power of their state-based programmable soft-keys.
03:26.07Ritalin2usually offices have the lights on during working hours
03:26.11sevardI <3 the 480i
03:26.16cbullock81D-Fender: how many call appearances will the 480i handle?
03:26.27chiang_sgI got this warning "WARNING[28959]: chan_sip.c:1085 __sip_xmit: sip_xmit of 0x874ee28 (len 468) to 10.10.0.100:5060 returned -1: Host is down" how to fix it ?
03:26.36[TK]D-FenderRitalin2 : Some people work in seriously shitty lightining conditions.  i for one always prefer Polycom over them, but for thos who do, its a decent phone.
03:26.36sevardcbullock81: I believe 9 sym calls, 4 call appearances
03:26.43JTchiang_sg: make the host go back up
03:26.51Qwell~lart JT
03:27.00JTQwell: what? :(
03:27.02cbullock81cool. it seems to be reasonably prices
03:27.04chiang_sgQwell: i have no control over the host :(
03:27.07[TK]D-Fendersevard : BLEH.  I've got a number of major customers who've got a mix of them and say Polycom wins hands down....
03:27.21J4k3oh snap, a LART.
03:27.22sevardBLEH.  I've used both and I still love the Aastra
03:27.30chiang_sgi mean JT, i have no control over the host
03:27.31JTas if i deserved a lart
03:27.42Qwellchiang_sg: Do you also call google when your internet is out?  that'd be pretty much the same thing
03:27.46[TK]D-Fendercbullock81 : A Polycom IP 501 can shuffle abour 24 calls :)  don't even ASK about a 601 :)
03:27.48sevardPolycoms just 'feel' too much like 1980s to me
03:27.59sevard601s are pretty hot
03:28.02Qwell"google, fix my internet please" "yes, I realize that you don't give me internet"
03:28.10chiang_sgi c :)
03:28.21[TK]D-FenderJ4k3 :You mean ONE dollar.  YGWYPF
03:28.26cbullock81D-Fender: with an expansion module?
03:28.59sevardcbullock81: "sidecar"
03:29.05cbullock81:)
03:29.35[TK]D-Fendercbullock81 : Even without, 6 linekeys & 8 calls each = 48 calls......
03:29.42sevardwhich is an expensive word for an expensive piece of hardware that all boils down to "more buttons"
03:30.10*** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap)
03:30.10J4k3"number plz?"
03:30.13[TK]D-Fendersevard : You take all the fun out of tech-toys you know that?
03:30.30sevard[TK]D-Fender: shake me upside down and call me Frenta
03:30.44Strom_Chow do you like that, Frenta?
03:30.51FrentaI LIKE IT DADDY
03:30.55sevwhat just happened
03:30.59J4k3I can call it the "one hundred dollar phone", and make it white and green
03:31.01*** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com)
03:31.05J4k3and act like I'm going to save the world with it
03:31.09[TK]D-Fendersevard : Ok, you lost me completely on that reference... please fee free to enlighten me...
03:31.17J4k3then I can start the One CrapPhone Per Child group ;)
03:31.25sev[TK]D-Fender: that's the best part, it wasn't a reference at all.  :D
03:32.08[TK]D-Fendersev : for a second there I though you might be a repository for at least a little culture. Clearly YOPLAIT still has more :D
03:32.13sevStrom_C: see? we can get along just fine with these crazy canadians
03:32.47sev[TK]D-Fender: what are you talking about, every other word that comes out of my friggen mouth is a reference to something, well, almost.  i just decided to go for full on randomness there.
03:32.59Strom_Clas Canadienses locas?
03:33.10sevNEIN, GO BACK TO DA BORDER.
03:33.21cbullock81d-fender: I'm still trying to handle the concepts behind VOIP phones (ya... laugh at me everyone), but why does polycom advertise the ip501 as a 3 line phone if it has 6 linekeys?
03:33.24Strom_Cdas catsex
03:33.28descartesStrom_C!!!
03:33.35Strom_Chi
03:33.56sevthat' was a very unenthusastic response
03:34.41*** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
03:34.42[TK]D-Fendercbullock81  : You are mixing stuff up, or the site you are referencing is.  the 501 has 3, the 601 has 6.
03:34.48*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
03:35.13cbullock81ah. k
03:35.14[TK]D-Fendercbullock81 : And fromt here you have to understand the way that Polycom treats those "line keys".  They need not represent only a single call.
03:35.30sevholy shit, that just reminded me of today in class, the teacher was categorizing different types of suicide and the girl in front of me wrote on her notes "Assisted Suicide (Youth In Asia)"
03:35.44[TK]D-Fenderlol
03:36.08sevI just about fell out of my chair laughing
03:36.18J4k3sev: she's gonna suck her way to the top!
03:36.21JTi don't get it :P
03:36.24sevhahaha
03:36.28sevJT: you're smrt
03:36.31Strom_CJT: euthanasia sounds like youth in asia
03:36.36cbullock81D-Fender: ok. so if like in my office. we have 4 lines, and any of our 6 employees need to be able to answer a call on any line at a time, could the 501 handle this?
03:36.37JTrofl
03:36.39NivexI had an education major ask me in a literature why I used such big words.  I told her "What's the point of having a large vocabulary if you do not intend to use it?"  I weep for the future generation.
03:36.42*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
03:36.43*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
03:36.53Nivex*in a literature class
03:37.05[TK]D-Fendersev : You know what cracks me up?  "Attempted suicide".  I mean really.... how dumb do you have to be to FAIL?  too dumb to live, too stupid to die.
03:37.19[TK]D-Fendercbullock81 : THERE is another misunderstanding.
03:37.23cbullock81lol
03:37.32rudholmNivex: nothing wrong with a little gratiutous sesquipedalia, I say!
03:37.35J4k3Nivex: yeah, but sometimes you must dumb things down for the general public.  Like "mass mailing" marketing materials.
03:37.35[TK]D-Fendercbullock81 : and I thoght I went through all of this with you likie TWICE already.
03:37.53Nivexrudholm: well played sirrah!
03:37.57rudholm:)
03:38.11Nivex(i had to look that one up)
03:38.26Nivex<3 the 'dict' command
03:38.29[TK]D-Fendercbullock81 : Sip phones can handle multiple CALLS, and can have multiple unique identities.  this has no impac on the fact you may have 300 analog lines floing into your PBX.
03:38.37sev[TK]D-Fender: or as my "philosophy of death and survival" teacher would call it, it's not attempted suicide, it's failed active voluntary suicide
03:38.44[TK]D-Fendercbullock81 : this jsut says how many aclls you can shuffle around at a given time.
03:39.21rudholmNivex: yeah, dict roxors
03:39.39[TK]D-Fendersev : So basically an autonomic reflex prevented you from acheiving something remarkably easy for our species?
03:39.42Nivex[TK]D-Fender: would it be accurate to state that there is not a 1:1 mapping between PSTN lines and SIP lines?
03:39.50cbullock81d-fender: Im trying to get it in my head... i jumped into a lot of new things @ once :)  linux, phones, asterisk
03:39.54*** join/#asterisk Mportnoy (n=test@200.122.158.88)
03:40.12MportnoyAnybody knows how to fix this error ? or what could be the cause ? /msg NickServ IDENTIFY
03:40.14[TK]D-FenderNivex : there is no such thing as a SIP "line", only a SIP "CALL"
03:40.16NivexI believe someone once explained to me the difference between a key system and a pbx.
03:40.17sev[TK]D-Fender: that reminds me of a story.  http://www.tard-blog.com/archives/entries/ndpn/1_the_first_entry_the_tards_may_be_fucked_up_but_so_are_their_parents.phtml
03:40.24MportnoyAnybody knows how to fix this error ? or what could be the cause ? res_musiconhold.c: Only wrote -1 of 640 bytes to pipe
03:40.32sevMportnoy: /msg NickServ IDENTIFY hahahahahahahaha
03:41.49MportnoyAnybody knows how to fix this error ? or what could be the cause ? res_musiconhold.c: Only wrote -1 of 640 bytes to pipe
03:42.46[TK]D-Fendercbullock81 : Calls come in to *.  * passes calls on to your phones.  there is no concept of grabbing a spefic "line" like in what are known as "key systems" like Nortel Northstar.
03:42.52tzangerhmm
03:43.16Strom_C[TK]D-Fender: that would be Norstar
03:43.23Strom_CNorthstar is a type of engine Cadillac uses, IIRC
03:43.35MportnoyI call to the sip trunk, and hte music dont work, and I call from a local phone and it worked
03:43.35MportnoyIm using ztdummy
03:44.42*** join/#asterisk ghost99 (n=neville@222-152-221-34.jetstream.xtra.co.nz)
03:44.47tzangerDial(SIP/${EXTEN}@provider) -- the other side is try to match the callerid of the phone I'm calling from
03:44.51NivexI got spoiled.  Last two places I worked we had a real PBX.  Place I'm at now is key system again.
03:44.52cbullock81d-fender: so the biggest thing is to find out how many calls a phone can handle per registration?
03:44.54NivexPhone rings all damn day.
03:45.03QwellNivex: so switch them to asterisk
03:45.09[TK]D-FenderStrom_C : Yes... I'm late, and my attention divided like U-235....
03:45.22NivexQwell: wish I could.  We don't handle the phones.  University telecom does.
03:45.45Nivexand I've learned you don't go invading other fiefdoms.
03:45.47[TK]D-FenderStrom_C : yes, Northstar engines... very nice.
03:45.51Strom_C[TK]D-Fender: I know, i'm just being needlessly pedantic :)
03:46.00*** join/#asterisk greendisease (n=jack@fedora/greendisease)
03:46.10Qwell"i'm"..."i'm"?!?!
03:46.17Qwell~lart Strom_C
03:46.21Qwellhaha
03:46.26[TK]D-Fenderit's late...
03:46.27Qwellbest lart ever
03:46.28tzanger[TK]D-Fender: any ideas?
03:46.29[TK]D-Fender.dasfdlaskfdhsfdahsklfdasd
03:46.30[TK]D-Fenderfshfdasfdjhhf
03:46.31Nivexi yam what I yam and that's all that I yam.
03:46.34[TK]D-Fender*BLARG*
03:47.11[TK]D-FenderNivex : "warning to all vegetarians/vegans : YOU ARE WHT YOU EAT"
03:47.14sevard[TK]D-Fender: if you want to blarg, you check out Guts by Chuck palahniuk
03:48.30[TK]D-Fendersevard : Loved Fight Club... suppose it'd be worth a try...
03:48.43sevardgfi, sinner.
03:49.24[TK]D-FenderGFI?
03:49.27*** join/#asterisk CrashHD (n=crashhd@c-67-166-156-157.hsd1.ca.comcast.net)
03:49.28sevardgo for it
03:49.51CrashHDstats
03:49.53CrashHDhello
03:49.55CrashHDhow do we show jb for sip channels?
03:50.00sevardhello
03:50.45sevardjitter buffer?
03:50.47tzanger*CLI> [Jan 25 22:52:26] WARNING[1542]: chan_sip.c:8139 check_auth: username mismatch, have <provider>, digest has <s>
03:50.50tzangerwhat does that mean?
03:50.52CrashHDya jitter
03:51.25[TK]D-Fendersevard : And hre I though it was Ground Fault Interrupt :)
03:51.41sevardEheh
03:51.44sevardnot on irc!
03:52.42sevardtzanger: that looks like a username mismatch
03:52.51tzangersevard: yeah but I'm not sure why
03:53.01tzangerI have two * boxes, A and B
03:53.03sevardare your... usernames correct?
03:53.10*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
03:53.13tzangerA has [aqtest] (for B)
03:53.19tzangerand B has [provider] (for A)
03:53.39tzangerB can call A with Dial(SIP/${EXTEN}@provider)
03:53.48tzangerA tries to call B with Dial(SIP/${EXTEN}@aqtest)
03:53.53tzangerthe former works great
03:53.55shmaltzanybody know of a small touchscreen device that will have just an instant messaging capablities?
03:53.57tzangerthe second fails with that message
03:54.12tzangerboth [aqtest] and [provider] are type=friend
03:54.31sevardshmaltz: you could build one simply
03:54.57shmaltzsevard, right I'm checking my schedule when I could do it
03:55.33sevardtzanger: have both the boxes registered fully?
03:55.40tzangerno
03:55.45sevardsip show peers and whatnot shows them asregistered?
03:55.49CrashHDso nobody has any idea on showing sip jitter buffer stats?
03:55.50tzangerB registers to A since it's behind nat and has a static IP
03:55.55tzangerer behind nat and has dynamic IP
03:56.02tzangerA has static IP and is not behind nat
03:56.06tzangerso B registers to A
03:56.06sevarddouble check A's registration line with B's sip definition block
03:56.16tzangerand sip show peers shows correct on both A nad B
03:56.25tzangersevard: A does not register to B
03:56.58sevardI thought both boxes needed to register to each other.
03:57.12tzangerhow can A register to B if B is behind NAT and has dynamic IP?
03:57.31sevardmagic?
03:58.01tzangersip debug on B shows the invites and "No user 'CID of my cellphone' in SIP users list, followed by "Found peer 'provider' for 'my cell caller ID' from A's IP:5060
03:58.08tzangerfollowed by sending a 401 Unauthorized
03:58.27tzangerwhy's it trying ot match caller ID instead of A's username entry for B?
03:58.35sevardsee, i thought you needed to register to both boxes.
03:58.49tzangero wait
03:58.57tzangerA's entry for B does not have a username
03:58.58sevard[TK]D-Fender?
03:59.13sevardthat would be a mistake, heh
03:59.32tzangerno
03:59.38tzangerB does not have a username for it's entry to A
03:59.46sevardAHHHHHHHHHHHH
04:00.00MportnoyAnybody knows how to fix this error ? or what could be the cause ? res_musiconhold.c: Only wrote -1 of 640 bytes to pipe
04:00.58sevard./ sscxssww (it's hard typing with stubs)
04:04.20cbullock81the Aastra 480i looks SWEET (im just not getting to look @ it)
04:04.48CrashHDwe use a lot of 480i's
04:04.49CrashHDwe like them
04:04.57sevardI love them.
04:05.06cbullock81how much do you have to give for them $$?
04:05.28tzangersevard: http://www.pastebin.ca/328888
04:05.33tzanger[TK]D-Fender: ^^ too if you have a moment
04:05.40sevard<cbullock81> how much do you have to give for them $$?
04:05.57sevardcbullock81: thirty cents
04:06.12cbullock81nice.
04:06.16tzangerbasically it matches the right sip.conf entry, but won't authenticate it because it's trying to use the caller ID of my cell
04:07.30J4k3three fiddy
04:07.35descarteshow stable have you haxors found asterisk 1.4? is it production ready?
04:07.45descarteswhat performance improvements are there?
04:07.48[TK]D-Fendercbullock81 : If you aren't using PoE, they will cost more.
04:07.52descarteser, how drastic are they
04:08.40J4k3who needs programming performance when we have quad processor servers for desktop pc prices.
04:08.46sevard[TK]D-Fender: what do you think about tzanger's issue
04:08.47J4k3;)
04:08.51[TK]D-Fendertzafrir : context/users don't match
04:09.28descartesJ4k3: please go reprogram asterisk in java for us
04:09.30tzanger[TK]D-Fender: eh?
04:09.31sevard[TK]D-Fender: that should be an issue... should it?
04:09.46sevardshould == shouldn't
04:10.49sevardNewwwwwww Java Asterisk, Proceses 2 calls one at a time over a one hour period! amazing! runs on anything... that's powerful enough to run it!
04:11.15jqlbut the transcoding will be JIT
04:11.33Qwelljql: oh god
04:11.41sevardhahaha
04:12.04jqlasterisk with garbage collection, though... count me out
04:12.24cbullock81well goodnight everyone. d-fender, once again... I appreciate the help!
04:12.27tzanger[TK]D-Fender: what do you mean, context/users don't match?
04:12.30J4k3asterisk for IBM Cassette Basic for the PC?
04:12.32Qwelljql: yeah, but you get objects
04:12.41[TK]D-Fendertzafrir : [provider] should be [aqtest] .
04:12.59[TK]D-FenderJ4k3 : All the cool kids had basic in BootROM :)
04:13.11*** part/#asterisk cbullock81 (n=cbullock@adsl-227-34-99.jan.bellsouth.net)
04:13.25J4k3[TK]D-Fender: yeah, the IBM PC did... "cassette basic"
04:13.27J4k3;)
04:13.44tzanger[TK]D-Fender: I'm not sure I understand
04:16.36[TK]D-Fendertzanger : Line 17 of your pastebin.
04:17.03tzanger[TK]D-Fender: why does it need to be [aqtest] ?
04:18.23[TK]D-Fendertzanger : Also not how one side has  asecret,and the other one doesnt....
04:18.28tzangeryeah
04:18.30tzangerI fixed that
04:18.32tzangerthat was the issue I think
04:18.47tzangerI use permit/deny on aqtest to authenticate [provider] now
04:18.54tzangerseems to be authenticating, now I hav ea dialplan issue it seems
04:21.09sevarda pirates life for me
04:21.32[TK]D-Fendersevard : Arrrrrrr...... arrrrrr
04:21.44[TK]D-Fendersevard : Essssss!!!!!
04:22.00sevardniiiiiice+geekyasfuck
04:23.14tzangerwtf now I have it working the other way
04:23.29tzangera can call b, but b cannot call a
04:24.09*** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-45.intrastar.net)
04:25.48sevardhaha
04:25.56tzangerthis makes no sense
04:26.02tzangersip authentication is *fucked*
04:26.14tzangerI am saying Dial(SIP/${EXTEN}@provider)
04:26.20tzanger[provider] includes a secret
04:26.38tzangerinstead of using provider for the username and secret as the secret, it's using the CALLER ID as the username?!
04:26.39*** join/#asterisk Geliman (n=scorpio@unaffiliated/drkshdw)
04:26.56*** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-45.intrastar.net)
04:27.23Juggietzanger, you need username=
04:27.34tzangerJuggie: odd, odd, odd
04:28.45*** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw)
04:28.46tzangerJuggie: didn't seem to help
04:29.13tzangerhmm
04:29.16tzangersip.conf examples say this
04:29.18tzanger; If you define a SIP proxy as a peer below, you may call
04:29.18tzanger; SIP/proxyhostname/user or SIP/user@proxyhostname
04:29.18tzanger; where the proxyhostname is defined in a section below
04:29.20tzangerok
04:29.30tzangerif 'provider' is my proxy, and aqtest is my user for that provider
04:29.43tzangerhow the hell do I specify an extension at that proxy, with that username?
04:29.57tzangerSIP/provider/aqtest/${EXTEN}?
04:33.24*** join/#asterisk erickperez (i=erickper@200.124.29.223)
04:33.32erickperezgood evening to all.
04:36.54*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
04:36.54*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
04:38.38CrashHDanyone know how to see sip jitter stats?
04:46.42*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
04:46.42*** mode/#asterisk [+o mog] by ChanServ
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04:47.00rue_mohrIf I'm connecting a digium pci T1 card to a newbridge mainstreet 3624, I need a crossover T1 cable, dont I?
04:47.33JTrue_mohr: like i said before, most probably
04:48.02rue_mohrah, sorry, scolled off the top
04:48.13rue_mohrwhile we fixed the phase converter
04:48.29rue_mohrok
04:48.35*** join/#asterisk astrsk123 (n=mikem@adsl-69-107-3-128.dsl.pltn13.pacbell.net)
04:48.44rue_mohrIhave some bix strips now too
04:49.02rue_mohrso I just need a computer to run askterisk on and I can start playing
04:49.12rue_mohrasterisk is a toy for geeks, right?
04:49.29JTit can be, but it is used for serious applications also
04:49.40rue_mohrsorta like a rubix cube, its as hard as you want to make it?
04:49.47rue_mohr:)
04:49.48De_Monm
04:49.52JTi suppose
04:49.56astrsk123I'm trying to set it up (via trixbox) for my small business
04:49.58De_Monno, rubix cubes are hard no matter what you do
04:50.11rue_mohrheh,
04:50.20smackusI am having issues where my iax peers are loosing connection, and shortly (15-30 seconds later) reconnecting. I have added qualifysmoothing=yes and the frequency of the disconnects has not changed. I am looking for advice on what more to look for and how to resolve this.
04:50.21rue_mohrif you dont mix them up so bad to begin with
04:50.23Qwellastrsk123: There's your problem
04:50.52De_Monrue_mohr erm, your logic is beyond me
04:51.07astrsk123I've missed it... you mean the concrete goal vs. pure geekification ?
04:51.30Qwellastrsk123: You can't get to the finish line if you have a concrete wall (trixbox) in the way
04:51.31De_Monastrsk123 you said trixbox
04:51.48Qwellso, yes, I guess you could call it a "concrete goal"
04:51.57astrsk123ahhhh.... Yes, i've got qualms about that, but figured it was an OK way to get a basic understanding
04:52.10JT~thebook
04:52.11jbot[thebook] a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
04:52.16JTthe book is probably a better way
04:52.21[TK]D-Fender~trixbox
04:52.22jbothmm... trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/
04:52.26smackusastrsk123: its just as easy to build without trixbox
04:52.42smackusthe sample configs are well documented with many examples
04:52.55astrsk123it's more the maintenance and configuration I wonder about, in the long run?
04:53.07De_Monis that a question?
04:53.10rue_mohrI have in a box... 1 old computer, 1 digium T1 card, some cat5e cable, some RJ45 ends, a mainstreet 3624, some 36pin centronics connectors, some bix strips (only sisies need a frame), and a telephone
04:53.17smackusastrsk123: na.. once you get it going, you will have a good understanding of it all.
04:53.20astrsk123adding mailboxes on the fly, etc...  doesn't tweaking configs take some time?
04:53.23rue_mohrI'm trying ot think of the simplest system I can assemble
04:53.24smackusit is simple line by line scripting.
04:53.41smackusno... take a look at some of the examples for macros
04:53.43De_Monrue_mohr simplest? that would be pure voip
04:53.53smackusi have about 100 agents configured with about 8 lines of script
04:53.56smackustotal
04:53.57rue_mohrok, to get asterisk up, yes
04:54.17rue_mohrI want to do something with my house lines (1) and our phones (4)
04:54.24rue_mohreventaully, anyhow
04:54.29astrsk123my Q is pure asterisk though (I think) not trix or freepbx... can't find via google or rtfm...
04:54.40rue_mohrand I'd like to do something about our answering machines (0)
04:54.47[TK]D-Fenderrue_mohr : You bought a T1 card and channel bank for your HOME?!
04:54.52astrsk123Q is: why does it answer immediately if immediate=no and caller id is disabled?
04:55.01rue_mohr[TK]D-Fender, "yes"
04:55.03astrsk123(WOW :-)
04:55.05JT[TK]D-Fender: i have a T1 card and a channel bank in my house
04:55.14JTdidn't cost too much though
04:55.27*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
04:55.27*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
04:55.28rue_mohrOR I bought a house so I could have something to apply asterisk to
04:55.31De_Monwhy? cable/dsl is faster..
04:56.05J4k3the access servers are all collecting dust now.  Too cheap to 3rd party that stuff.
04:56.09JTinternal only T1 for voice
04:56.25JTi have a stack of access servers too, all collecting dust :)
04:56.37rue_mohrI still need to make a remote for hte house, that when pressed outside, makes all the lights in the house flash twice and the house to emit the sound "beep beep"
04:56.38J4k3yeah, I dunno what to do with them
04:56.44De_Monok, i missed the channel banks part
04:56.48J4k3I figure one day I'll pull the DSP cards out or something
04:56.59J4k3the portmasters are just 486DE-66's.
04:57.05rue_mohrI also have two 64ms echo cans
04:57.31rue_mohrbut I dont have edge connectors for them yet, so I wont think to include them
04:57.37J4k3haha
04:57.49smackusquick question... i have an iax provider for home.. .periodically for whatever reason i drop my registration to them. maybe once a week. iax2 reload and all that does not get it to re register successfully. I have to CLI> restart now before it will connect back up again. is there a way to correct this?
04:58.17rue_mohrI dont know
04:58.55*** join/#asterisk ezw` (n=Ez@c66.203.210-59.clta.globetrotter.net)
04:58.59*** join/#asterisk x86 (n=x86@p3m/member/x86)
04:59.11*** join/#asterisk ltd (n=z@202-161-1-26.dyn.iinet.net.au)
04:59.59smackusis there a setting to make that iax registration persistent?
05:00.51J4k3hmm...  I could get my a car an unlimited 1xRTT/EVDO account, then get it to vpn back into the network here so it always has a static IP and it can run services, then do things like web-based remote start and route planning
05:00.52J4k3hmm
05:00.53tzangerFound user 'aqtest' for 'aqtest'
05:00.58tzangerSO FUCKING LET THE CALL THOUGH
05:01.00tzangerer THROUGH
05:01.50astrsk123newbie q: how to set answer delay? Checked immediate=no and CallerID disabled. Any other hints/direction?
05:02.09smackuswait()
05:02.11smackusmaybe
05:02.40astrsk123as first in dialplan?
05:02.54smackusexten => XXX,1,Wait(2)
05:02.54smackusexten => XXX,n,Answer()
05:03.02smackuswould something like that work for you?
05:03.22smackusthat would pause 2 seconds before answering.
05:03.34smackusthen after answer you would do your Dial() command
05:03.56astrsk123I'll try it... so far I've just used the freepbx interface.... so now it's time to edit the dial plan directly, right?
05:04.06smackusyeah... much easier.
05:04.35smackuscheck out the asterisk wiki... you can get all sorts of command help there.
05:04.57astrsk123is there an easy way to strip off the freepbx/trix stuff and start over?
05:05.13astrsk123their default dialplan is like spaghetti...
05:05.17Qwellreinstall
05:05.18[TK]D-Fenderastrsk123 : FreePBX will teach you NOTHING.
05:05.21Qwellfind a new distro
05:05.40astrsk123vanilla asterisk... or Open?
05:05.49[TK]D-Fenderastrsk123 : it is NOT a laughing pad for learning *.  It is so convoluted it will only serve to drastically HINDER your learning.
05:05.52astrsk123(only ones i've heard of)
05:06.03[TK]D-Fenderpaunching*
05:06.05[TK]D-Fenderaklasdkad
05:06.07[TK]D-Fenderlaunching*
05:06.15Qwelllaughing pad works to
05:06.15Qwello
05:06.20astrsk123yeah... I realized that in 5 minutes...
05:06.45astrsk123looks like either commit to limitations of freepbx,or learn to do it directly. Can't really live in both worlds...
05:07.07JTchoose life
05:07.49rue_mohrso I'd like a dial plan, that consists of dialing 1 and getting hold music, thats it
05:07.59rue_mohror or any button and getting hold music
05:08.18smackusin the asterisk install directory there is a directory containing config samples
05:08.30*** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net)
05:08.42rue_mohrso your saying I shoudl install asterisk on something
05:08.44rue_mohrok...
05:08.56rue_mohrI'll get back to you when I'm done
05:09.16rue_mohr:)
05:09.32smackusastrsk123: downoad and install the files from asterisk.org use the readme file to get it installed.
05:09.32rue_mohrI have some help from kb1kanobe too :)
05:10.38[TK]D-FenderJ4k3 : If its good for the clueless, why are so amy people unable to configure it? :)
05:10.55J4k3[TK]D-Fender: damned good question.  Its really straightforward to me.
05:11.03[TK]D-FenderJ4k3 : See the problem with idiot-proofing everything is the fact this world is full of MORONS.
05:11.27J4k3the "install asterisk and play with the conf files" was a bit "big" for me to wrap my head around all at first
05:11.37J4k3I think now, thanks to trixbox, I'd have some idea of whats going on if I was to do that
05:12.16J4k3and I plan to, because trixbox has too much stuff going on and what appears to be an entire lack of security.
05:12.23tzanger[TK]D-Fender: basically I had ot remove the secret
05:12.29tzangerit would *NOT* let me do what I wanted
05:12.30smackussecurity is over rated
05:12.38tzangerso I'm not sure how the hell it's authenticating
05:12.39tzangerbut fuck it
05:13.15astrsk123yeah -- goofing with trix has helped me get big picture & play some... now I'll probably reinstall OS & build asterisk from scratch
05:14.15[TK]D-Fendertzanger : non-secret username.
05:14.18J4k3if I were to use trixbox in a static/public IP situation, I'd stick it on its own ethernet interface on the router (not a switch port, stick it its own /30 on its own card) then firewall everything except iax and sip from all but the exact hosts I wanted accessing it... and then I'd still be nervous about it.
05:14.25tzanger[TK]D-Fender: ?
05:15.26[TK]D-Fendertzanger : Auth on name only.  if it matches, pass it on.
05:15.31tzangerah
05:15.36[TK]D-Fendertzanger : make sure they can't access outbound....
05:16.13tzanger[TK]D-Fender: http://www.pastebin.ca/328941
05:16.21tzangercouldn't get it to auth both ways any other way
05:16.25*** join/#asterisk AndyCap_ (n=aoy@pdpc/supporter/sustaining/AndyCap)
05:16.32*** part/#asterisk ghost99 (n=neville@222-152-221-34.jetstream.xtra.co.nz)
05:16.41tzangerI mean from provider -> aqtest (SIP/${EXTEN}@aqtest) I understand how it works (permit/deny)
05:17.11tzangerbut from aqtest -> provider (SIP/${EXTEN}@provider)... I can't use permit/deny since aqtest is dynamic/nat, but giving a secret causes it to never ever pass
05:17.22tzangerit says it finds the user, but then rejects without explanation
05:17.28tzangerI'd have to meter out the code to try and figure out why
05:20.26tzangersip authentication is simply fucked.
05:29.35*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
05:32.47*** join/#asterisk mike052278 (n=mike@d118-75-206-122.clv.wideopenwest.com)
05:33.34mike052278hey guys quick question - is it possible to use a FXO card to connect an analog phone without an actual POTS line?  I'm trying to go from VOIP to VOIP
05:34.02JTno, you need FXS
05:34.13mike052278k that sucks :D
05:34.34Strom_Cwhy?
05:34.38astrsk123why are fxs cards so much more expensive than fxo??? I guess because people use ATAs instead?
05:34.44JTfxo cards don't have 90VAC ringing generators etc built in
05:34.55mike052278ah
05:34.55mike052278ok
05:35.05mike052278whats an ATA
05:35.12astrsk123two birds with one stone on that answer!
05:35.22mike052278lol
05:35.37astrsk123Analog Telephonen Adapter... plug normal phone into it, then it goes to IP network
05:35.44JTfxo cards also don't have -48VDC dc battery feed
05:35.44mike052278ok
05:36.05astrsk123letting standard phone into the asterisk world...
05:36.17hardwiremeh
05:36.20mike052278lol
05:36.22mike052278its for my house
05:36.30mike052278im sick of paying my cable company 50$/month
05:36.31mike052278:P
05:36.52hardwirefor a phone?
05:36.55mike052278yeah
05:37.02hardwirewhats your alternative?
05:37.06mike052278asterisk
05:37.06mike052278:P
05:37.12hardwireconnected to?
05:37.13mike052278or a normal land line which is about the same
05:37.27mike052278the internet
05:37.35mike052278i have voip through my cable co now
05:37.36hardwireso how much are you planning on saving?
05:37.38mike052278a lot
05:37.42mike052278cuz i barely use my house phone
05:37.47hardwireand yet its $50
05:37.50mike052278im setup through junction
05:37.52mike052278yeah
05:37.55hardwireamazing
05:37.58mike052278i'd rather pay by minutes
05:37.59mike052278i know
05:38.03mike052278its ridiculous
05:38.09hardwiremine is like $9 :)
05:38.10mike052278they need to do something about these damn cable companies
05:38.11hardwireheh
05:38.15mike052278thats what im saying
05:38.15mike052278:P
05:38.20mike052278im estimating like 2-5$/month
05:38.21mike052278lol
05:38.29hardwiredsl needs a phone line
05:38.34astrsk123Yes... but "they" is us... and asterisk is doing it. Free market always wins :-)
05:38.40mike052278yup yup
05:38.52hardwirewow
05:39.00hardwire2-5 a month, you could start a family off that
05:39.04mike052278lol
05:39.06mike052278i have a family
05:39.08mike052278thats the problem!
05:39.09mike052278lol
05:39.20hardwireyou ever thought about being amish?
05:39.29mike052278na
05:39.30*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
05:39.31*** mode/#asterisk [+o mog] by ChanServ
05:39.33hardwirehey mog
05:39.36hardwiresup bitch
05:39.39mike052278im not cheap, i just dont like to waste money
05:39.40mike052278:)
05:39.49hardwiremike052278: quit calling people
05:39.59mike052278LOL
05:40.07hardwiretell people "dude, I'm on IRC"
05:40.53hardwireI always wondered if I should just forward my phone to a recording stating "I cannot accept people calling me with a phone, please use IRC at so and so...."
05:40.59mike052278lol
05:41.01[TK]D-Fender"we don't fight, we all play nice, when you're livin' in Amish paradise..."
05:41.05mike052278whats a good ATA
05:41.09mike052278er decent
05:41.15hardwiremike052278: sell you one for cheap
05:41.19mike052278how much
05:41.25hardwire$35
05:41.28hardwire1fxs/1fxo
05:41.32mike052278sweet
05:41.35hardwireI have a crapload I am getting rid of
05:41.39astrsk123my voicemail says, "can't answer now... I'm reading my email"
05:41.40[TK]D-Fendermike052278 : Linksys SPA-2002 is decent.  Mediatrix is good.
05:41.41mike052278where do you live at
05:41.41JTmodel?
05:41.43mike052278er state
05:41.44hardwireastrsk123: heh
05:41.47hardwiremike052278: alaska
05:41.52[TK]D-FenderDENIAL :)
05:42.07hardwireJT: Sipura SPA3000
05:42.08mike052278hmm will you overnight it? :D
05:42.20JThardwire: why you getting rid of so many?
05:42.22hardwiremike052278: you said you were cheap
05:42.22ezw`is it possible to run softphone cisco 7940 with asterisk ?
05:42.25mike052278lol
05:42.32hardwireJT: I bought a ton for research
05:42.32mike052278no i dont like to waste money!
05:42.33mike052278lol
05:42.39hardwireI have 5 or so to get rid of if I can find them all
05:42.42JThardwire: ah ok
05:42.49mike052278what brand is it
05:42.54hardwireSipura SPA3000
05:42.58mike052278sweet
05:43.01mike052278have no idea what that is
05:43.02mike052278:D
05:43.13hardwireJT: you are too smart to buy a Audiocodes 4 FXO eh?
05:43.19hardwiremy ebay auction isn't so hot
05:43.24JThardwire: hmm?
05:43.32file[TK]D-Fender: ! ! !
05:43.39rue_mohrthe one thing anoying baout the cisco phones is the time they take to "boot"
05:43.43JTwhat about the audiocodes?
05:43.50hardwireI have a 4 port FXO gateway available
05:43.54rue_mohrI think it was a cisco...
05:43.55JToh ok
05:44.07mike052278oh
05:44.10JTi don't need one hardwire, i have a 24 port T1 channel bank
05:44.13hardwirerue_mohr: anything takes time to boot :)
05:44.16mike052278holy shit thats like the linksys one hardwire?
05:44.18hardwireJT: rub it in
05:44.19rue_mohryes
05:44.22hardwiremike052278: no
05:44.22JThardwire: but if it's cheap enough i'll buy it
05:44.24mike052278oh ok
05:44.30hardwireits a Sipura SPA3000 Pre Linksys
05:44.33rue_mohrits nice if a phone is up and ready in about 4 seconds
05:44.37hardwireolder case, same shit.
05:44.46mike052278k
05:44.53mike052278how much total to ship like overnight
05:44.54hardwirehttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&ih=002&sspagename=STRK%3AMESE%3AIT&viewitem=&item=120077545760&rd=1&rd=1
05:44.56mike052278to michigan
05:44.58hardwirewhile you are buying my shit
05:44.58mike052278ah nice
05:45.00hardwirebuy that too
05:45.02mike052278LOL
05:45.21mike052278i dont need that yet
05:45.21mike052278:P
05:45.23rue_mohrJT, what kinda channel bank you have?
05:45.26hardwiresure you do
05:45.30CrashHDhow come when dialing a peer with sip the account code is not flagged?
05:45.31mike052278lol
05:45.31hardwireeverybody needs one
05:45.35hardwireits just that I have so many
05:45.35hardwireheh
05:45.49JTrue_mohr: Zhone Zplex 10
05:45.52CrashHDfor instance DIAL(SIP/${EXTEN}@voip3)
05:45.58CrashHDcdr records do not show any account code
05:46.09rue_mohrJT, darn, I was hoping it might be a newbridge
05:46.14JTheh
05:46.33mike052278hmmm
05:46.40rue_mohris my newbridge gonna be better or worse than what you got?
05:46.40JTrue_mohr: if your newbridge does channelised CAS T1, then you will need a T1 crossover cable
05:46.41mike052278does the cisco ATA work w/ asterisk?
05:46.42hardwiremike052278: email bogomaster@bogomip.com
05:46.44mike052278k
05:46.51JTprobably better, not familiar with it though
05:46.56mike052278whats the total, i can paypal you
05:46.58rue_mohrhmm
05:47.03mike052278overnight to 48313
05:47.07hardwiremike052278:  I don't have a shipping calculator laying around atm :)
05:47.10mike052278lol
05:47.15hardwireand I need to flash it back to factory
05:47.17JTthe zhone is discontinued
05:47.21hardwirelet me work on that for the next 30 or so minutes
05:47.26mike052278k
05:47.33rue_mohrit would make sense to me that I'd need a crossover, because I expect their both built to be plugged into a telsco
05:47.36mike052278does it have instructions
05:47.52JTrue_mohr: you can make cables, right?
05:47.57rue_mohroh yea
05:48.23rue_mohrI have all the RJ, bix, 110, and 66 tools I could ever need
05:49.02rue_mohrI dont mind if the 66 just gathers dust though
05:49.07rue_mohr:)
05:49.20JTwe just use Krone here in australia, pretty much
05:49.32rue_mohryour in australlia!?
05:49.35JTyes
05:49.37JTamazing
05:49.39JT:P
05:49.43rue_mohrcool, what so they call norstar up there?
05:49.46*** part/#asterisk smackus (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net)
05:49.56JThuh?
05:50.01rue_mohr:/
05:50.05rue_mohrerm...
05:50.12rue_mohr"nortel"
05:50.22JTi guess it's still nortel
05:50.29rue_mohrmeridian
05:50.35rue_mohrI was over there
05:50.42rue_mohrthey have another name for it I cant remember
05:50.43JTyeah i'm sure there's a few meridians over here
05:50.50hardwireugh
05:50.53rue_mohrsame system, different name
05:50.54JTwell meridians used to be NT
05:50.57CrashHDif I were using dundi to dial from a gateway to multiple peers how could I set the account code based on the server it was going to?
05:50.57Strom_CMeridian and Norstar CICS are not the same thing
05:51.01JTbefore nortel bought it
05:51.08hardwirethey sure look it
05:51.13Strom_CJT: um, Nortel IS Northern Telecom
05:51.18Strom_CNORthern TELecom
05:51.20rue_mohrbut I think they have a 3rd name for it all in oz
05:51.23Strom_Csame company, different name
05:51.35JTStrom_C: i thought nortel was an amalgamation of Nt and a couple of others
05:51.39Strom_Cnoooooo
05:51.40rue_mohrI swear the phones over there had a different name again on them
05:51.48Strom_CNTI changed its name to Nortel in 1995
05:51.54JTah ok
05:51.55rue_mohrI was in a casino, and noticed
05:52.03JTin either case, i've seen stuff badged from both
05:52.05Strom_Cbought bay networks in 1998, changed its name to Nortel Networks
05:52.08JTNt badged meridian
05:52.13Strom_Cchanged back to just "Nortel" in 2004
05:52.16JTnortel badged DMS-100
05:52.42Strom_Crue_mohr: the telephone sets, or the payphones?
05:53.06QwellStrom_C: where's your "berg"?
05:53.24Strom_CStromberg-Carlson is the switch manufacturer, Strom Carlson is the phone phreak
05:53.27Strom_Cdurhh :)
05:53.39Qwellwhat, you just decided to drop it? :P
05:54.18Strom_Cno, i never had it to begin with
05:55.03Qwellgotcha
05:55.14hardwiremike052278: ok it works
05:55.28*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
05:55.28*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
05:55.30hardwirebtw there is no overnight shipping from alaska heh
05:55.47mike052278k
05:55.55mike052278shit that sucks
05:56.02mike052278damn it
05:56.03mike052278:P
05:56.06hardwireyou get mail on saturdays right?
05:56.10mike052278yeah
05:56.12mike052278usps
05:56.13hardwireerr nm
05:56.13mike052278?
05:56.15mike052278lol
05:56.16hardwiretoday is wed
05:56.18hardwireerr
05:56.19hardwirethu
05:56.21mike052278yeah
05:56.23hardwireI thought we were in the W
05:56.26mike052278lol
05:56.32hardwirezip?
05:56.34mike05227848313
05:56.37Qwellhardwire: Friday most places ;)
05:56.42mike052278but if sat use 48066
05:56.50hardwire3 day.. so monday
05:56.51JTit's friday here
05:56.56hardwireerr.. tuesday
05:56.58hardwire$7
05:57.03mike052278hmmm
05:57.08hardwireexpress mail is $18
05:57.08mike052278ok do you have a kinkos
05:57.21hardwireno but our USPS has an e-Parcel system
05:57.23Qwellmike052278: it's a 2 day dog sled
05:57.28mike052278lol
05:57.30Qwellto the nearest kinkos
05:57.34mike052278lolol
05:57.38mike052278shit
05:57.43mike052278ok how about fedex
05:57.55*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)
05:57.57hardwirebecause ass rape is fun?
05:58.01mike052278lol
05:58.11Strom_CI can haxxor your kinkos
05:58.15JTups are the kings of rape
05:58.21mike052278no if you have fedex, i will have you bill my fedex account overnight
05:58.23mike052278:)
05:58.24QwellStrom_C: You should bring a card next time you come :p
05:58.29Strom_Ci will
05:58.33hardwirethey have a checkmark that is by default unchecked
05:58.38hardwireits "Drop off packages?"
05:58.51mike052278yeah
05:58.57mike052278most fedex couriers do pick up
05:58.58mike052278:)
05:59.11[hC]Any of you guys tried using a USB audio device on astlinux? or a soekris box in general?
05:59.13hardwireovernight is $66
05:59.19mike052278LOL
05:59.23hardwirepriority (3 day basically) is $38
05:59.25Qwell[hC]: should just need alsa
05:59.26[hC]I think it might be a little to cpu taxing.
05:59.45mike052278dang it
05:59.47hardwireah.. 2 day super standard delux is $17
05:59.48rudholmmaybe I should change my name to West Electric
05:59.55[hC]Qwell: Ive standardized on using Astlinux on Soekris hardware for PBX installs, but one customer wants overhead PA paging systems (which you usually connect via sound card + line out)
06:00.02JThaha $66, it cost about USD$130 for priority from germany to australia via fedex
06:00.03hardwireah.. that was priority overnight..
06:00.06[hC]Id use the pci slot, but i think im gonna need that for an analog interface card.
06:00.07JTarrived in 3 days
06:00.13hardwiremike052278: its up to you
06:00.15[hC]Was  just pondering a USB audio device.
06:00.18mike052278yeah i think there's a overnight deliver by 4:30pm\
06:00.20mike052278what is that one
06:00.32JTGold Bullion Service
06:00.33hardwireby 4:30pm the next day?
06:00.34Strom_Crudholm: how about Wes Electric
06:00.36Strom_Cso it sounds like a name
06:00.37mike052278yeha i think so
06:00.46hardwirethats the $38 option
06:00.48mike052278k
06:00.51mike052278whats teh saturday price
06:00.55hardwirenext business day
06:01.06hardwireI think its still $38
06:01.10mike052278gayness
06:01.11mike052278lol
06:01.14hardwireNext-business-day delivery by 10:30 a.m. to most U.S. addresses; by noon, 4:30 p.m. or 5 p.m. in remote areas; by noon or 1:30 p.m. on Saturdays. Check transit times.
06:01.38mike052278hmm is USPS any cheaper
06:01.42mike052278for like 1-2 day
06:02.18hardwireexpress is $18
06:02.22hardwirethats their most expensive
06:02.25mike052278nice
06:02.29mike052278want to use them?
06:02.42hardwiresure
06:02.44mike052278k
06:02.54mike052278whats the total and email
06:02.56hardwireif they don't deliver you can probably hit up the saturday pick up booth that day
06:03.01mike052278right right
06:03.12hardwireotherwise you cry for a whole day
06:03.28mike052278im trying to beat my cable company for billing me on monday and then prorating a whole nother month on my next bill :D
06:03.47hardwiremike052278: hah, they backdate everything, you are screwed reguardless
06:03.51mike052278LOL
06:03.54mike052278f'ers
06:03.57JThaha, you're going to try and configure all that shit in time :P
06:04.16hardwireyeh my install rates are around $110/h
06:04.23mike052278LOL
06:04.23hardwireGotcha!
06:04.25hardwirenaw, you could do it
06:04.31mike052278yeah it shouldn't be too bad
06:04.33hardwirethis is a FXO/FXS
06:04.48mike052278i got a stupid fxo card and had that up in about 2 hrs after a short headache
06:04.56hardwireyou can connect it to both the phone network and a phone at once, and use both sides independantly
06:04.59mike052278then jumped on here and realized why i dont get a dialtone
06:05.00mike052278:P
06:05.06mike052278ok
06:05.08hardwiremike052278: yeh, the O
06:05.11mike052278i am anti-landline
06:05.17hardwireS = dialtone
06:05.22mike052278LOL
06:05.34hardwireI have some IAXy too, you would hate them
06:05.45mike052278i asked my friend, and i was like hey if i get an fxo card will it work w/ my analog phone
06:05.47hardwiresingle port FXS
06:05.52mike052278he must of assumed i had a landline
06:06.04mike052278k
06:06.05hardwireyeh it and your phone can talk on the same landline
06:06.17mike052278is it basically plug and play w/ minor config to asterisk
06:06.28mike052278or do i have to set an ip on it
06:06.33mike052278and configure it as a sip device?
06:14.25*** join/#asterisk mike052278 (n=mike@d118-75-206-122.clv.wideopenwest.com)
06:14.38mike052278man i hate this cable company :P
06:14.58mike052278so where we at hardwire?
06:15.00hardwiremike052278: email me so I have your info
06:15.03mike052278k
06:15.06J4k3"wideopen" like the continuity of your coax?
06:15.06mike052278whats your email again
06:15.09mike052278yes
06:15.10mike052278lol
06:15.14hardwireI will charge you $35 + shipping (bogomaster@bogomip.com)
06:15.17mike052278k
06:15.27hardwireheh, you guys are gonna think this is neat
06:15.42hardwirebut my telco is one of oh,, two.. in the country using this cool DSLAM from Allied Telesyn
06:16.01hardwireit has uber nifty multicast support for DSL via IGMP joins and a hacked residential gateway
06:16.09hardwirethey are offering 300 something mpeg2 channels over it
06:16.18hardwireand my friend that works there is filling me in on the vlans
06:16.30mike052278k hard i sent ya an email
06:16.31hardwireat home he can just turn on vlc and watch a TV stream
06:16.35mike052278we're expecting sat. delivery right
06:16.54hardwirewell I can't make that garantee
06:16.59mike052278right right
06:17.00hardwirebut I am going to ship it before I head off to work
06:17.05mike052278ok
06:17.07hardwiresuper express 5000
06:17.12mike052278lol
06:17.12J4k3TV over landline is such a waste.
06:17.20*** join/#asterisk X-Rob (n=Rob@dsl-202-173-151-24.qld.westnet.com.au)
06:17.25hardwirewhats your maximum shipping amount?
06:17.28J4k3it doesn't take much of a parabolic to get solid sat tv.
06:17.34hardwireif it turns out to be over $20 bucks just say no?
06:17.42mike052278hmm
06:17.44J4k3terrestrial networks should be used for interactive services only.
06:17.49sevardHow hard would it be to build a FXO to FXS converter?
06:17.58hardwiresevard: many have done it with kits
06:18.00mike052278ya 20 is cool
06:18.17hardwiremike052278: ok
06:18.32hardwireuCasterisk eh
06:18.33hardwireweird
06:18.42hardwirehttp://www.rowetel.com/ucasterisk/
06:19.01sevardHas anyone ever wired up their own FXS to FXO converter?
06:19.02*** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net)
06:19.19hardwiresevard: people here aren't cheap bastards. we established that earlier.
06:19.29mike052278hey hard are there instructions with it? :P
06:19.31sevardWasn't watching.
06:19.36hardwiremike052278: online
06:19.39mike052278k
06:19.41hardwireI will send you an email w/ link
06:19.45sevardhardwire: wiwhat are you selling?
06:19.46mike052278sweet
06:19.46mike052278thx
06:19.50hardwiresevard: my soul
06:19.56mike052278lol
06:20.03sevardhardwire: I'll give you 0.23
06:20.10fetcherJ4k3: even more wasteful is TV over VHF/UHF
06:20.12hardwirenaw, I have had a higher bid
06:20.36hardwirefetcher: heh... I want to live in a faraday cage.
06:20.49hardwirejust a few pair of copper in and out
06:21.09hardwirethe good news is I am re-sheetrocking my whole house when I replace the insulation
06:21.14hardwireso my evil plan could work
06:21.34hardwirejust leave a sign at the door that says "leave your cell phones at the door"
06:21.44JTcopper?
06:21.48J4k3hardwire: the only copper I'd have running is electricity, and I'd have an isolation transformer
06:21.49JTi think you meant to say fibre
06:21.51JTsilly boy
06:21.52fetcherhardwire: any particular reason?  Copper wires will carry in RF and re-radiate it to some extent
06:21.53J4k3use fiber for data/voice
06:21.56hardwireJT: a mans got to have data
06:22.00JTFIBRE
06:22.01J4k3fibre!
06:22.02hardwireoh yeh there we go
06:22.03hardwireFIBER!
06:22.04J4k3jt!
06:22.20JTcopper is silly if you want to be secure, it aids tempest
06:22.27hardwireJ4k3: I was gonna have a pure sine wave converter plugged into the GRID
06:22.33J4k3copper will give you aids
06:22.37hardwireand even that lined in so much lead it would kill a baby
06:22.39hardwireyou hear me
06:22.42hardwirea baby
06:23.02J4k3sounds paranoid
06:23.08J4k3... do you have a tin foil hat?
06:23.27mike052278hmm
06:23.28hardwireI just want to block microwave radiation from eatiny my cellies
06:23.32fetcherhardwire: if cost is no object, fill your roof with solar panels and go completely off-grid
06:23.33mike052278anyone got a quick way to remove zapata drivers
06:23.34mike052278?
06:23.44mike052278er zaptel
06:23.44hardwirefetcher: sigh
06:23.44mike052278:P
06:23.57J4k3I dunno, the loudest microwave generator around here is me.
06:24.02fetcherhardwire: plenty of lead in the requisite batteries, then :)
06:24.19J4k3I'm pro-solar
06:24.23hardwireThe main one is its not thermodynamically viable
06:24.25JTthe whole costing bazillions of dollars thng
06:24.27hardwireits a big battery
06:24.28J4k3if it involves a big mirror collector and a sterling enging
06:24.30J4k3er engine
06:24.30hardwirethat works at noon
06:24.47hardwireand the second is it can't produce as much energy as was put into making it
06:24.55hardwirethats why I am pro-wind/hydro
06:25.00hardwirelet mother nature do the work
06:25.25J4k3I personally just want my own onsite nuke.
06:25.40J4k3our wind here is lame.
06:25.40fetcherhardwire: I think the 2nd point has been disproven, at least for modern panels and grid-tie.  Energy payback time is something like 3-5 years?  (economic payback is much longer, or never if you have to borrow money for the PV)
06:25.50J4k3we'll get 2 days of good wind, then 5 days of nothing
06:25.55hardwirefetcher: interest!
06:26.00hardwiremy visa is expensive!
06:26.01hardwirehahha
06:26.08J4k3even at the top of a 150' tower.
06:26.12hardwirefetcher: it may have been
06:26.18hardwirethose are just the most recent factoids I know of
06:26.41hardwirefetcher: I did soem solar arrays for comme quipment in bush alaska.
06:26.46hardwirethey were great for that
06:26.57hardwirebut thats cause there was no other fucking power source anywhere else
06:27.14hardwiresorry, language. .I know
06:27.21fetcheryeah, remote site power is where it really makes sense.  e.g. cabins where utilty wants $100k to extend out the grid
06:27.27hardwireI'm a little curse factory
06:28.04hardwirefetcher: I would rather they sold me a solar panel and let me hook a $25k light bulb up to the nearest electrical tower.
06:28.30fetcherI'd have a few panels up, if not for all the trees around here.  Same problem as C-band satellite
06:28.38J4k3from what I hear you can push about 1 watt over a single strand of multimode up to 1 km
06:28.52hardwirefetcher: whats a matter, hugged one too many trees?
06:28.54hardwirecut em down1
06:29.04hardwirenot like you can't breathe w/o them
06:29.20fetcherJ4k3: 1 watt optical power, or load power after conversion loss on the receiving end?
06:29.21hardwireJ4k3: isn't that goofy
06:29.26J4k3fetcher: load.
06:29.34fetcherwow
06:29.43hardwireI assume thats how they could power switching equipment on long hauls w/ redundancy switches
06:29.51J4k3yep
06:29.56JThardwire: what long hauls?
06:30.02fetcherI've read about the optically-powered amplifiers in undersea fiber cables
06:30.11hardwireJT: run fiber a long ways, power remote switches with light.
06:30.14J4k3transistors are neat
06:30.17JTthey run power cables in undersea cables
06:30.30JTfor the optical fibre amplifiers
06:30.41J4k3JT: that practice may be abandoned.
06:30.45hardwireI assumed so :)
06:30.55hardwireit would be weird to power the amplifiers for your signal via your signal
06:31.08FuriousGeorgeis there a good reason to install a 32 bit linux for asterisk on a 64 bit platform
06:31.11fetchersome doped-erbium (?) style amps work in the optical domain, without the need to convert signals to electrical form and back
06:31.23JThardwire: probably not possible with current technoligy
06:31.25hardwireFuriousGeorge: run two 32 bit OS's :)
06:31.26JTyes, EDFA
06:31.31JTErbium Doped Fibre Amplifier
06:31.44JTyou still need electricity to power the 900nm pump diode
06:31.52JTthe signals go at 1550nm
06:31.54fetcherhardwire: power could be on a separate, unmodulated wavelength
06:31.59FuriousGeorgehardwire: that was only a one-dinger on a scale of one to three dings
06:32.08hardwireis Erbium one of those iums like plutoniums and uraniums.
06:32.28JTthe edfa fibre section allows pure optical amplification
06:32.30fetcherhardwire: or put a "DC bias" on the carrier signal (but most lasers probably couldn't sustain constant-on)
06:32.31hardwirefetcher: PoF is a neat setup :)
06:32.42JTso they only need to regenerate every 1000km or so
06:32.47hardwirefetcher: use more lasers!
06:32.58hardwiremake a 1ghz DC bias
06:33.01FuriousGeorgei actually checked for a second that i wasnt in #physics
06:33.21hardwireI should go pay attention to my sick GF
06:33.41hardwireinstead of talking about bullshit radioactive light based amplifiers
06:33.47fetcherJT: so, the 900nm doesn't carry far enough to just inject it from each endpoint?
06:33.48hardwirethat make my hair fall out
06:33.55JTfetcher: definately not
06:34.15J4k3bah if she's sick
06:34.20J4k3she's just gonna be in a bad mood
06:34.21J4k3ignore her
06:34.23fetcheroh, well.  Nice idea...
06:34.24J4k3;)
06:34.44JTfetcher: regeneration stations still need a lot of electrical power too
06:34.49hardwirenight all
06:34.56hardwiremike052278: I will wrap and ship in the morn
06:35.04JTregeneration means bringing every wavelength back to electrical level and resynching all the timing etc
06:35.23hardwiremike052278: if it doesn't work for your needs I will send you an IAXy
06:35.51fetcherJT: are any of those in mid-span on undersea runs, or only at the landing stations?
06:36.02JTsome are undersea
06:37.24fetcherJT: so, chaining EDFA amps together eventually causes too much wavelength spreading?
06:37.34JTtoo much dispersion
06:37.37JTeventually
06:37.51fetcherya, couldn't think of the proper term
06:37.54JTthe pulses aren't so nice and distinct
06:38.55fetcherSo, no way to increase capacity on such a cable by just putting better DWDM gear at the ends
06:39.34JTprobably not, but there's usually tonnes of spare wavelengths provisioned in
06:39.45JTand fibres
06:40.15fetcherI always wondered why the FLAG cable was only 4 strands (maybe 4 pair).
06:40.26JTflag?
06:40.45JTyou can fit a lot of data in 4 pairs with the right gear
06:41.07fetcherFiber Link Around the Globe, or something like that.  A Europe-Africa-Asia cable chain that went in about 15 years ago
06:41.17JTah ok
06:41.25JTancient history technology now i bet :P
06:41.44fetcherhttp://en.wikipedia.org/wiki/Fiber-Optic_Link_Around_the_Globe
06:42.01*** join/#asterisk HushPe (n=HushPe@202.36.75.196)
06:42.21fetcherdid Worldcom/MCI lay undersea fiber during their big build-out? (pre-bankruptcy)
06:42.24HushPehads: are you around?
06:42.39JTin 1998, "off the shelf" dwdm gear could go 160 wavelengths @ 10Gbit/s on a single pair, 1.6Tbit/s
06:42.53J4k3geez, thats some throughput
06:43.27JTyeah
06:43.46J4k3I mean... thats like 1/100000th of the world population making a g729a quality call simultaniously.
06:43.52fetcherJT: how big was the equipment for that system, physically?  Separate card for each wavelength?
06:44.16JTiirc SOBO (the sydney olympic broadcasting organisation) had around 5Tbit/s of data running around in fibres from ther complex
06:44.29JTduring the olympics
06:44.42JTfetcher: traditionally yeah
06:45.32JTtelco fibre gear can do 40Gbit/s a wavelength now, but i think most are sticking with 10Gbit/s
06:46.05JTdunno if the 40Gbit/s gear has the same range as 10Gbit/s
06:47.10fetcherI've only seen CWDM stuff in person, years ago.  Even that was impressive
06:48.10JTi want to get a cheap fibre splicer, then i could build a lot of fun toys
06:48.16JTbut alas, they are never cheap
06:48.51sevardfibre termination is exxxxppeennsssiivvee
06:49.08JTsplicers are expensive :)
06:49.14J4k3single mode is expensive
06:49.17J4k3multimode is cheap and easy
06:50.18fetcherI was at a site last year that had used all multi-mode (orange) jumpers in their otherwise single-mode plant.  Amazingly, it was all working without any apparent trouble
06:50.22JTmulitmode is also shit :)
06:50.56JTfetcher: for short distances the impact may not be too bad i guess
06:51.28fetcheryeah, their longest run was maybe 3 km
06:51.39JTof multimode?
06:52.37fetcherJT: no, 3km in total, of which only a few meters was multimode
06:52.48JTright
06:52.52*** join/#asterisk shinux__ (n=shinux@196.220.25.145)
06:53.32fetchercarrying just one wavelength of GigE
06:54.29*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
06:54.39J4k3multimode is shit, but its optimal for short runs
06:54.44J4k3I thought mixing was a no-no?
06:55.27*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
06:55.27*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
06:55.40fetcherJ4k3: yeah, me too
06:58.42Strom_Cnew toy new toy :D
06:58.42Strom_Chttp://www.stromcarlson.com/2D2/
06:59.25*** join/#asterisk Bobthehunter (n=Bobthehu@145-27.mc.cite.net)
07:00.02sevardStrom_C: that's like one you'd see at an airport, where'd you get it?
07:00.17*** join/#asterisk cthorner (n=cthorner@209-234-185-130.static.twtelecom.net)
07:00.34rudholmStrom_C: pshaw, what is that, a modern totalizer??  :)
07:01.01rudholmhere's a *proper* totalizer:  http://topanga.ca.us/2C2/2C2-interior.jpg
07:01.02rudholm:)
07:01.49rudholmsevard: yes, that's the kind of phone you'd see in an airport or building lobby
07:01.54rudholmthey're very very nice, IMO
07:02.06sevardi agree, i think their sexier than regular payphones
07:02.11rudholmyeah, totally
07:02.28sevardone of the first payphones i ever saw that made me automatically think "i want one"
07:02.34rudholmI'm installing one of mine (I now have two) into the wall in my living room
07:02.39rudholmyeah, me too
07:02.57sevardsend me the other!
07:03.09sevardI'll give you stale McDonalds
07:03.18sevardthat should cover shipping
07:03.21rudholmheh
07:03.24rudholmwhere are you?
07:03.26Strom_Crudholm: ive actually got two payphones i bought on ebay that should be arriving tomorrow; here's hoping they have real totalizers in 'em
07:03.29sevardMinnesota
07:03.39rudholmoh, that's not too far
07:03.53sevardStrom_C: TWO MORE? Seriously, won't that be like 7 pay phones active in your fucking house?
07:03.58sevardrudholm: where are you?
07:04.00*** join/#asterisk oej (n=olle@apollo.webway.se)
07:04.09rudholmsevard: Los Angeles
07:04.28Strom_Csevard: one of them is for a friend
07:04.33sevardrudholm: interesting, I was just there a couple week sago
07:04.52Strom_Csevard: rudholm is a meatfriend of mine
07:04.52rudholmbusiness, or pleasure?
07:05.00sevardah, that makes sense
07:05.06sevardrudholm: a little bit of both.
07:05.10rudholmI see
07:05.11*** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.198)
07:05.14Kernel_Corehi all
07:05.15cthornerHas anyone had problems with the hints in 1.4.0?
07:05.18rudholmLA is nice when the weather is good
07:05.19*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)
07:05.20sevardrudholm: but mostly, it was allllll pleasure.
07:05.24rudholmheh
07:05.29sevardrudholm: when is the weather... not good?
07:05.34Strom_Csevard stayed at casa carlson
07:05.38rudholmah
07:05.45rudholmwell, the smog can suck
07:05.51sevardI did.  I was accomidated comfortably
07:05.54*** join/#asterisk The_Ball (n=alex@149.135.94.2)
07:05.55rudholmand in the summer it can get kinda hot
07:06.16rudholmbut hot days make for nice balmy nights
07:06.18sevardhot weather blows.
07:06.32sevardi'd rather be in a snow pile than melted on the ground
07:06.46Strom_Csevard is one of those crazy minnesota types
07:06.54Strom_Cthinks freezing is a fun way to spend half the year
07:07.02sevardexcuse me, 'minnesotian' to you, mister.
07:07.17sevardmore like 89% of the year, thank you
07:07.21rudholmStrom_C could barely handle Seattle
07:07.27The_Ballwith an extension as such: exten => _X.,1,Dial(SIP/${EXTEN}@gotalk,30) how can I have the same functionality but have to dial a 9 first for voip calls?
07:07.27Strom_Crudholm: it looks better with real telco instruction cards in it:  http://www.stromcarlson.com/2D2/DSC03434.JPG
07:07.32rudholmhe was shivering the whole time we were there the other day
07:07.46sevardStrom is one of those crazy Californians, who blare the heat with the windows closed when it's 80 degrees outside
07:08.18rudholmStrom_C: yeah, you need to put one of the blue coin plaques in
07:08.22rudholmthat orange just sucks
07:08.25Strom_Cyeah
07:08.32rudholmit ruins the look of the phone
07:08.35Strom_Ci need to befriend a pacbell installer
07:08.43rudholmalthough it would conflict with the Pacific Bell cards
07:08.46Strom_Cactually, here's a plan
07:08.47sevardwhy don't you just make your own plaques
07:08.50Strom_C1. find broken phone
07:08.53Strom_C2. call it in for repair
07:08.54*** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net)
07:08.56Strom_C3. wait
07:08.59rudholmthe plaques are enameled metal
07:09.15rudholmcamp out by a broken payphone?
07:09.30Strom_Cwell, wait there on the day they say the repairman will show up
07:09.42rudholmhey, yours doesn't say U.S. Coins Only
07:09.45sevardStrom_C: I can give you the address of a broken payphone in SF
07:09.58Strom_Csevard: pacific bell telephone co. serves plenty right here in los angeles
07:10.20Strom_Ci'm still amused that the company is legally "Pacific Bell Telephone Company DBA AT&T California"
07:10.42rudholmStrom_C: notice that the newer 2C and 2D phones have the little screw to hold the cards in
07:10.48Strom_Cyup
07:10.48rudholmmy old one doesn't have that
07:11.11Strom_Cthe housing on mine is dated september 1975
07:11.23rudholmmine is early 70s
07:11.31rudholmwell, the one I got this weekend
07:11.37Kernel_Coreguys... when I set french language as channel language , asterisk should read /var/lib/asterisk/sounds/fr ?
07:12.21[TK]D-FenderKernel_Core : if presend, yes
07:12.25[TK]D-Fenderpresent*
07:12.34sevardStrom_C: What's your landlord going to say when you move out and there are all of these huge holes in the wall where your 80 payphones used to be
07:13.03rudholmsevard: I'm actually going to mount a 2C2 where my main hall ends in my living room :)
07:13.12Kernel_Core[TK]D-Fender: I just copied french prompts to this directory , but asterisk doesn't read it ( I mantained astvarlibdir => /var/lib/asteris in my asterisk.conf)
07:13.40*** join/#asterisk hardwire (n=hardwire@rdbck-5429.wasilla.mtaonline.net)
07:14.17Kernel_Coreops ( I mantained astvarlibdir => /var/lib/asterisk in my asterisk.conf)
07:14.21Strom_Csevard: well the surface mount ones only leave five holes each, and a little spackle and paint will fix that
07:14.34Strom_Cim not planning to mount the 2C2 in the wall
07:14.36*** join/#asterisk codefreeze (n=steve_mu@216.166.159.235)
07:14.39sevardheh
07:14.45sevardwhat's the point of it then?
07:14.55rudholmI'm hiring a carpenter to rip my wall apart and put a window box in for the 2C2 :)
07:14.55Strom_Ci can mount it on my desk instead
07:14.57sevardyou HAVE to put it in the wall
07:14.58Strom_Cin fact...
07:15.13sevardrudholm: come on man, you can do that yourself.
07:15.16sevardsilly californians.
07:15.29rudholmthe whole drywall needs to come out for the whole wall
07:15.35rudholmI don't want it to look like a patch
07:15.42rudholmthe texture won't match from the new drywall
07:15.50sevardyou need a level, drywall, paint, sheetrock
07:16.05rudholmyeah, but there are two corners involved
07:16.11sevardand?
07:16.17naftali5patch -p0 < paint
07:16.19rudholmdrywall can get tricky when you are doing entire walls of it
07:16.26sevarddude
07:16.28[TK]D-FenderKernel_Core : pastebin all the configs related to a call that isn't reacting the way you think it should, and the CLI output of that call.
07:16.29[TK]D-Fender~pb
07:16.35jbotfrom memory, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
07:16.38sevardi'm the worst drywaller in the world, and seriously, it's not hard
07:16.40rudholmtricky to make it look *right*
07:16.46rudholmyou're a drywaller?
07:16.50sevardyou can screw up drywall all you want, it all looks great
07:16.52sevardi was for a while
07:16.55rudholmcorners?
07:17.08sevardcorners? pffft, they START you out on corners
07:17.12Kernel_Core[TK]D-Fender: honestly I use A2billing and it doesn't read those french prompts
07:17.18rudholmwhat about the intersection with the ceiling?
07:17.32sevardcorner bead, drywall, knife
07:17.32[TK]D-FenderKernel_Core : Prove to me that its SUPPOSED to be using them....
07:17.39rudholmhmm
07:17.50rudholmI have that stupid acoustic cottage cheese on my ceiling too
07:17.56Strom_Crudholm: wouldnt the 2D2 make a crazy bedside phone? :)
07:18.01rudholmhahaha
07:18.02sevard3 minutes your first corner, 30 seconds your next
07:18.09rudholmStrom_C: it could *be* the nightstand
07:18.14Strom_Cyes!
07:18.22sevardrudholm: you mean knocked down drywall texture
07:18.46rudholmsevard: huh?
07:19.06sevardyou spray mud at a surface, let it dry a bit, then knock it down with a 6 or 12 inch knife
07:19.14Strom_Cwill an iaxy fit /inside/ the 2D?
07:19.27rudholmStrom_C: I'm sure it will
07:19.51*** join/#asterisk Asterisky (n=chatzill@ip70-190-159-144.ph.ph.cox.net)
07:19.57rudholmsevard: I'm an Electrical Engineer, not a carpenter :)
07:20.06sevardwhy do you prefer the iaxy to something like an spa 2000
07:20.08Strom_Cright, but the question is also whether one can route the cat5 and power cables through the case
07:20.15Strom_Csevard: because it's tiny
07:20.21sevardrudholm: i can honestly say, from experience, carpentry is not rocket science.
07:20.24Strom_Cand it does CPC
07:20.28sevardCPC?
07:20.29[TK]D-FenderDammit Jim, I'm a doctor not a telephone repair man!
07:20.41rudholmsevard: and the IAXy accepts pulse and does CPC on both inbound and outbound calls
07:21.00sevardcan you do PoE?
07:21.14rudholmI don't think the IAXy does PoE
07:21.19rudholmbut I don't know, I don't have one yet.
07:21.20Asteriskyhello everyone!! a very quick question to all you bright people, when I modify zonedata.c on the source directory does zaptel automatically recognize the changes after a reload?
07:21.25rudholmStrom_C owes me one, though
07:21.29sevardwhat's CPC now
07:21.31[TK]D-Fendersevard : There, PoE!
07:21.39rudholmCPC is the signal at the end of a call
07:21.43Strom_Csevard: far end disconnect supervision
07:22.00rudholmloop potential drops (or is reversed) briefly
07:22.08sevard... shouldn't an ATA do that?
07:22.09rudholmthings like my doorphone use it to know when to drop the line
07:22.25rudholmthe Sipuras *only* do CPC on received calls, never on placed calls
07:23.18sevardthat's strange.
07:23.18gfraysse<PROTECTED>
07:23.18rudholmthis appears to be the one thing you cannot configure
07:23.18sevardyou can configure those things to high hell
07:23.19rudholmyep
07:23.19rudholmwhich is why it's odd
07:23.19rudholmyou can configure the CPC duration
07:23.19rudholmbut you can't configure it to happen on placed calls
07:23.22rudholmso my door phone is on a TDM400 port
07:23.31rudholmbecause it has no way to hangup otherwise
07:23.42rudholmCPC is the only way (well, I could set it to timed, but that's lame lame lame)
07:24.08The_BallHow can I make numbers dialed with 9 in front go out on voip, what im trying is: exten => 9XXXXXXXXXX.,1,Dial(SIP/${EXTEN:1}@gotalk,30)
07:24.08sevardthe only thing that bothered me about the SPA ATAs was you can't do MWI Lamp on phones that don't do FSK
07:24.26JTThe_Ball: you need an underscore in front
07:24.44rudholmI See
07:24.51rudholmI have no MWI lamps
07:25.08rudholmalthough I have one SPA-2000 that seems to think I have voicemail on one line for no good reason
07:25.14rudholmgives me stutter dialtone
07:25.16sevardi used to have some sweet hotel phones that didn't do FSK with cool lamps and all i could do for MWI was ring splash
07:25.32sevardhaha, did you try to reflash the firmware?
07:25.39AsteriskyHELP! please question to all you bright people, when I modify zonedata.c on the source directory does zaptel automatically recognize the changes after a reload?
07:25.48The_BallJT, what does the underscore mean?
07:25.55JTAsterisky: no, you must recompile zaptel
07:25.58sevardAsterisky: you might want to um, recompile... reinstall...
07:26.13JTThe_Ball: that it's a pattern match, not a fixed number extension
07:26.14Asteriskythank you so much
07:26.26AsteriskyJY ; servard
07:26.31JTnp
07:26.41sevardANNGRYYY MOBBB
07:27.15rudholmsevard: yes, I tried re-image and factory reset
07:27.21rudholmit's not a big deal
07:27.24sevardstrange
07:28.05*** join/#asterisk wannabeaastguru (n=wannabea@bonhomme.ifi.uio.no)
07:28.12The_BallJT, is there a way I can monitor closer what astrisk is trying to do in the dial plan? cause it's still not dialing and giving a beep bepp tone
07:28.32sevardbeep bepp
07:28.41JTThe_Ball: set verbose 10
07:29.44wannabeaastguruhey, is there a way to save/record that's being written here?
07:30.17Strom_C?
07:30.32dendritewannabeaastguru: Look in your irc client manual, look for "logging."
07:30.45wannabeaastguruok, i will
07:30.49wannabeaastguruthanx
07:31.00dendritenp
07:31.27The_BallJT, cheers
07:33.24wannabeaastgurubut the log is empty
07:33.32Chris-NBhi
07:33.38wannabeaastguruhi
07:33.45Chris-NBI've a 8 Port Basic rate ISDN card
07:33.56JTindeed
07:34.01Chris-NB6 of these ports are used and connected to the telco
07:34.14Chris-NBI've put these 6 ports into group 1
07:34.21wannabeaastgurunow its filled with things
07:34.25JTand port 6 doesn't work?
07:34.32wannabeaastgurubeautifull thing
07:34.35Chris-NBJT, port 6 works : D
07:34.43Chris-NBJT, I've managed it to work
07:34.46JTChris-NB: what did you change?
07:34.57Chris-NBI've added crc4 in the span deffinitions
07:35.03JToh ok
07:35.03Chris-NBand now it's working
07:35.10JTsounds a bit strange but ok
07:35.50Strom_Cok, now THAT looks sexy
07:35.53wannabeaastgurudendrite: Thanx Man
07:36.03JTChris-NB: so what's wrong?
07:36.06Strom_Cthe 2D2 is on the little wooden riser just above my cisco phones
07:36.22rudholmStrom_C: pictures?
07:36.27Strom_Cone moment
07:36.33Strom_Cim still plugging in the iaxy
07:36.41JTboom
07:36.47rudholmtoo bad the iaxy doesn't do coin relay control :)
07:36.57Chris-NBwhen span 1 of the group is down, it isn't possible to make calls anymore
07:37.16JTspans are not meant to be down
07:37.20JThow often does that happen?
07:37.32Chris-NBI hope it doesn't happen
07:37.36Chris-NBbut what if?
07:37.45Chris-NBis it possible to make a fallback?
07:37.45JTthen you have a problem
07:38.00Chris-NBwhen it's immediatle busy
07:38.08Chris-NBwhen trying to place a call over group1
07:38.15JTerm, i guess you could make it dial the lines in reverse order if dialling them in forware order fails
07:38.30JTforward, even
07:38.31Strom_Cyeah yeah, you and your real coin line
07:39.04Chris-NBmhm ... I could give it a try
07:39.13Chris-NBor try in random order
07:39.22Chris-NBok, would be possible
07:39.50rudholmStrom_C: yes, that's right, I have a *real* 17Q Coin Service Line
07:39.53rudholmso :-p
07:40.01JTChris-NB:
07:40.14JTDial(Zap/g1/${EXTEN})
07:40.17JTDial(Zap/G1/${EXTEN})
07:40.22JTforward then reverse
07:40.31JTit can also dial in round robin groups
07:40.43JTbut random probably not, unless you set it up to manually dial lines
07:40.51JThttp://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
07:42.05Strom_Crudholm: oh this works beautifully :F
07:42.07Strom_Cer :D
07:42.51rudholmwhat happens when you put money in?
07:42.56rudholm:)
07:43.06Strom_Cthe totalizer beeps
07:43.13rudholmdoes the money come back?
07:43.19rudholmif the call doesn't supervise?
07:43.24sevardmonies!
07:43.32Strom_Conly if i push where it says DO NOT PUSH
07:43.40rudholmor do you just clog up the escrow tray?
07:43.41rudholm:)
07:44.04sevardI love payphones with a broken coin return catch lever, so when your call fails to go through the quarters spit out at you and roll down the street
07:44.28*** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net)
07:44.32rudholmhow do they do that?
07:44.36Strom_Cit's actually highly unlikely i'll be able to get a 17Q in here, since only one pair comes up from the basement
07:44.36rudholmisn't the door in the way?
07:44.54rudholmno ground anywhere?
07:45.08rudholmoh, you're saying you can't have a 17Q and your POTS line
07:45.13Strom_Cright
07:45.20Strom_Cand i dont think they'll put DSL on a 17Q :)
07:45.23rudholmyou could probably use your fire escape as ground or something :)
07:45.28Strom_Chah
07:45.30rudholmyou know, I'm not sure of that
07:45.37sevardno, like in SF I was using this phone and the door was all screwed up, and when my call failed to go through the quarters were flung out in a violent fashion, hit my chest, and rolled down the street
07:45.52Strom_Cum
07:45.55Strom_Cweird
07:46.14J4k3how big was the rock/hammer/brick/etc you hit it with?
07:46.15J4k3;)
07:46.28sevardreally annoying, and the guy behind me had to use the phone aswell, so i sat back and giggled when it happened to him
07:46.44JTand took photos
07:46.49sevardbut i was in tenderloin and afraid that I was going to get shot for smiling, so i ran home.
07:50.42Strom_Crudholm: you should install your other 2[CD] at your office desk
07:50.55Strom_Ci doubt it will go anywhere
07:51.17rudholmyeah, I was thinking that
07:51.23rudholmit'd fit right into the corner of my cubicle
07:51.40Strom_CThank you for using YT&T
07:52.00rudholmit'd be nice to get my coin control working first (I'd have to run Asterisk on my desktop computer there at the office)
07:52.04rudholmheh
07:52.07rudholmwell, we do do voice now
07:52.11rudholmvia Y!M
07:52.15Strom_Cexactly
07:54.12*** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com)
07:54.31Chris-NBJT, thanks. I meant round-robin. just wrote it in a confused way : )
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07:58.07JTChris-NB: round robin doesn't stop you from hitting a dead channel though
07:58.35Chris-NBJT, true
07:58.36JTChris-NB: you probably want at least 2 dial commands
07:58.41JTlike i suggested
07:59.34Chris-NBjep. 1 Dial, check DIALSTATUS, probably dial again
07:59.52JTno, just a second dial line under the first
08:00.09JTit won't make it to priority n+1 if the call was successful
08:00.27JTdialstatus can also prove unreliable
08:02.03*** join/#asterisk lorinc (n=ang@caracas-0524.adsl.interware.hu)
08:03.57*** join/#asterisk phpboy (i=onelove@dsl-242-51-177.telkomadsl.co.za)
08:04.55Strom_Crudholm: ok, i took a photo after some cleanup :)
08:05.02phpboyhey all, how would I could about logining the end of a call through the asterisk manager(API)?
08:05.26phpboye.g. I want to to tell my via the open connection that the call is finished
08:05.42Strom_Crudholm: http://www.stromcarlson.com/2D2/DSC03435.JPG
08:06.48rudholmit's hard to get that brushed stainless clean
08:06.59Strom_Cyeah...i havent wiped it down yet
08:07.06Strom_Cits still all fingerprinty
08:07.14rudholmand you haven't put in the New York Telephone plaque
08:07.19Strom_Cand i need to remove that faircall plaque
08:07.25rudholmyeah
08:07.27rudholmdefinitely
08:07.32Strom_Cbut still...it looks awesome
08:07.33rudholm"SAVE!"
08:07.35rudholmyep
08:07.35rudholmthey do
08:07.41Strom_Cit's monolithic :D
08:07.53Strom_C2007: a phone odysset
08:08.00Strom_Cs/odysset/odyssey/
08:08.19*** join/#asterisk Aurs (n=Aurs@81.191.123.189)
08:08.20rudholm[cue music: Also Sprach Zarathustra]
08:08.34Strom_Cexactly
08:08.53Strom_Cespecially when you xconsider that my eyes are level with the top of the monitor
08:08.58Strom_Cthis thing towers over ms :)
08:09.07rudholmheh
08:09.40Strom_Cme
08:09.43Strom_Cnot ms
08:09.47Strom_Cdamn my fingers tonight
08:14.06*** join/#asterisk erickperez (i=erickper@200.124.29.223)
08:14.09erickperezHi there.
08:14.34Strom_Cgood afternoon
08:14.35erickperezI just checkout asterisk,zaptel and librpi...how do i checkout addons?
08:14.51Strom_Cwhich branch are you checking out?
08:15.58*** join/#asterisk vader-- (n=me@c-71-226-201-15.hsd1.nj.comcast.net)
08:17.10erickperez1.2
08:17.37phpboyhey all, how would I could about logining the end of a call through the asterisk manager(API)?
08:20.51erickperezStrom_C: 1.2
08:22.14Strom_Csvn checkout http://svn.digium.com/svn/asterisk-addons/branches/1.2 asterisk-addons-1.2
08:23.28erickperezdoes checking out from svn include the jitterbuffer patch ?
08:24.19Strom_C*shrug*
08:25.06erickperezhmmm
08:25.08*** join/#asterisk drone1 (n=kova@tech.quentris.be)
08:25.41erickperezI found this, but im not sure how to apply it:
08:25.42erickperezhttp://asterisk-backports.org/wiki/index.php/Rtp-jb-1.2
08:25.48drone1If I would want to limit the number of calls to a sip trunk, how would I go about it?
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08:40.45juansHi
08:40.55juansi have a tiger 320 card
08:41.17juansand i have a problem whit the out bound call
08:41.30juansstill in ringing all time
08:42.08*** join/#asterisk qdk (n=qdk@213.150.62.32)
08:42.16juansplease help me
08:43.57*** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.3.16.revip2.asianet.co.th)
08:44.20Strom_Cjuans: what kind of line interface?
08:45.07BugKhaManyone knows how to set moh to play stream broadcasted in asf format?
08:46.12juansStrom_C, analogue
08:46.18juansStrom_C, FXO
08:46.56BugKhaMit works with shoutcast mp3 streams but not asf
08:47.44Strom_Cjuans: what exactly is the problem
08:48.12juansStrom_C,
08:48.14J4k3BugKhaM: I think the larger issue is asf support...  I thought asf was a microsoft format?
08:48.21juansStrom_C, -- Zap/1-1 is ringing
08:48.21juans<PROTECTED>
08:48.29J4k3*riiing!*
08:48.31juansStrom_C, but not call
08:48.35Strom_Cjuans: don't flood the channel
08:48.38Strom_Cuse pastebin
08:48.41juansStrom_C, sory
08:48.45Strom_Cthis is on inbound calls, right?
08:48.58juansStrom_C, outbound
08:49.05Strom_Cturn off call progress detection
08:49.20juansStrom_C, how?
08:49.27Strom_Cin zapata.conf
08:50.29BugKhaMJ4k3: yeah, I guess so but is there a way to convert using sox or any other tools?
08:50.30juansStrom_C, ok, and then?
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08:50.59Strom_Cjuans: and then what
08:51.27juansStrom_C, when i make a call, the asterisk still in ringing
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08:51.44Strom_Cdid you reload the zapata configuration?
08:51.49juansStrom_C, yes
08:51.55Strom_Ctry restarting asterisk
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08:53.13juansStrom_C, nothing happen
08:53.21juansStrom_C, still in ringing
08:53.39Strom_Cpastebin your zapata.conf file
08:53.48juansStrom_C, ok
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08:55.27*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
08:56.12juansStrom_C, i can receive a call.. and make a call to extension, but no to line
08:56.24Strom_Clike i said already
08:56.25Strom_Cpastebin your zapata.conf file
08:56.30juanshttp://pastebin.com/867728
08:57.53Strom_Cturn off busydetect as well
08:57.58Strom_Cbusydetect=no
08:58.04Strom_Ccallprogress=no
08:59.41Chris-NBwhat could be the possibility for hearing only one way voice in isdn calls?
09:00.22Manfisha deaf user?
09:00.25defsworkChris-NB: phone muted ? ;)
09:00.34juansStrom_C, nothing change
09:00.43Strom_Cjuans: who manufactured the card?
09:00.51Chris-NBdefswork, deff. not : /
09:00.57juansdigium
09:01.30Strom_Cjuans: if it is a digium card, call digium support, which opens in about four hours
09:01.39Strom_CIAX2/guest@misery.digium.com/s
09:02.11juansStrom_C, what is that??
09:02.33Strom_Cthe iax2 address to digium's pbx
09:04.46Chris-NBcould this be a codec problem ?
09:05.00Strom_CChris-NB: what kind of telephone set are you using?
09:05.14*** join/#asterisk shodan (n=shodan@ip072.99-113-216.pppoe4.joliette.intermonde.net)
09:08.40*** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70)
09:09.00yidiyuehanhello everyone
09:09.25Chris-NBStrom_C, I've a alcatel pbx (4200) connected via E1 to my asterisk (Sangoma A101), asterisk is connected to pstn (Beronet BN8S0)
09:09.45yidiyuehanwhich codec does asteirsk support for video calls?
09:10.11Chris-NBwhen I call my mobile I can hear voice on the asterisk voip phone or alcatel phone, but no voice on the mobile
09:10.20ThoMeGuten Morgen
09:10.23ThoMeMorgen Chris
09:10.27ThoMehi people
09:10.56Chris-NBsometimes there is voice in both directions, sometimes only in 1
09:20.05x86moin
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09:23.35Chris-NBStrom_C, now idea what that could be?
09:24.03Strom_C*shrug*
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09:36.49Chris-NBStrom_C, is that a no?
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09:46.19*** join/#asterisk signius (n=IceChat7@dsl-217-155-69-102.zen.co.uk)
09:48.00signiusHi guys
09:48.11*** part/#asterisk E-bola (i=bola@rbii-valhalla.mrseb.co.uk)
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09:49.28*** part/#asterisk s1gny|wrk (n=s1gny@p54914BB5.dip.t-dialin.net)
09:50.13Chris-NBJT, the channel up/down thing occures again : /
09:50.24Chris-NBanyone knows what that means? http://pastebin.ca/329018
09:50.42Chris-NBit's pri intens debug from a isdn basic rate channel going up/down the whole time
09:50.55hwtin the hint stuff: exten => 200,hint,SIP/phone1, which of them sends SUBSCRIBE, and which sends the NOTIFYes?
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10:31.23phearlesscvs.digium.com is dead?
10:34.13phearless<PROTECTED>
10:34.13phearlessHost cvs.digium.com not found: 3(NXDOMAIN)
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10:39.16drraysvn
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11:23.32ManfishChris-NB you still here ?
11:23.40Chris-NBjep
11:23.57*** part/#asterisk s1gny|wrk (n=s1gny@p54914BB5.dip.t-dialin.net)
11:23.58Manfishyour card keeps dropping the d chan?
11:24.05Chris-NBjep : /
11:24.20Chris-NBnot right now, but I think it will happen again
11:24.27ManfishIRQ ?
11:24.45Chris-NBwhen I stop asterisk, unload modules, reload everything and start asterisk, it works fine
11:25.03Chris-NBbut I think only for a certain amount of time (had that behaviour 2 times)
11:25.25Manfishso its not happening all the time?
11:25.45Chris-NBnop
11:26.06Chris-NBirq ... /proc/interrupts looks ok. no errs
11:26.30Chris-NBqozap and wanpipe are on different irqs and on different cores
11:26.44Manfishok
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11:26.55Chris-NBif the span starts going up/down only a complete reload of everything brings it back working ok
11:27.21ManfishI had a te110p that was doing just the same turned out to be a faulty card
11:27.26Chris-NBI've a second problem, sometins calls are completely ok, sometimes there is only one way audio
11:27.42Chris-NBhow do you discovered that?
11:28.01Chris-NBmine is a bn8s0
11:28.18ManfishI called digium the dialed in and ran some tests, said it was fucked and gave me a RMA number
11:28.32Chris-NBok
11:28.44Chris-NBso probably i should dial beronet
11:28.51Manfishits worth a go
11:28.53Chris-NBok
11:30.23Manfishi messed around for two weeks trying to troubleshoot it and digium did some tests and in 2 mins found it was faulty
11:32.47Chris-NBok
11:32.50Chris-NBI'll call them
11:33.00Chris-NBthanks for that
11:35.23*** join/#asterisk alexandrepos (n=alexandr@201.21.143.130)
11:36.34alexandreposhow to I compile my asterisk w/ g729 ?
11:36.53*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
11:39.58*** join/#asterisk naluckcuck (n=nick@dhcp2.zuk40.mot-tools.co.uk)
11:40.47Ahrimanesalexandrepos, you can buy a license and get a binary module for asterisk from digium.com
11:41.01signiusManfish: LOL the moral of the story then is digium know there shit !!!
11:41.50naluckcuckhi all, anybody know how to disable the "someone has logged into the manager" messages on the asterisk console?
11:42.09alexandreposAhrimanes: its works only w/ license ?
11:43.58Ahrimanesalexandrepos, in most countries there are patents that require you to obtain a license to use g729
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11:53.27znoGanyone know how to do a WaitExten for a few seconds and if it times out, just keep going with the next priority?
11:53.42znoGit otherwise goes to the t extension (timeout)
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12:02.03*** join/#asterisk UVSoft (n=UVSoft@80.254.48.58)
12:02.10UVSofthi
12:06.29znoGstrange, i changed my language with Set(LANGUAGE()=es); and when I play a file, it doesn't look it up in /var/lib/asterisk/sounds/$language/$file
12:06.32znoGany ideas why?
12:07.05*** join/#asterisk drako (n=ljd@unaffiliated/luisjose)
12:07.54Ahrimanesany sip experts up for looking at a pcap trace to see while
12:07.58znoGnever mind
12:08.02Ahrimanessee why a sip registration fails
12:09.30UVSoftwhat dialplan should look like to have the following behavior: i pick up the phone (FXS), hear the dial tone, than type '0', zero extension is triggered, than if the line (FXO) is busy i hear busy tone if not i hear dial tone again and type the number to dial throug that FXO device
12:11.34UVSoftis there a command to make asterisk play dialtone again?
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12:21.32Nobbieheya
12:24.47UVSoftit seems everybody is sleeping now
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12:30.29Nobbiei wish i could be doing that now
12:32.36BugKhaMhi there, how do I interrupt the MusicOnhold(class) command?
12:35.19*** join/#asterisk viperdude (i=viperdud@87.127.248.92)
12:36.41NobbieBug: and do what after the MOH ?
12:37.47viperdudeis it possible to get asterisk to send a invite as follows?
12:38.10viperdudeINVITE sip:12345;phone-context=+44@myprovider;user=phone SIP/2.0
12:39.20Nobbieviper: have you tried SIPAddHeader() ?
12:39.27Nobbiei haven't used it, only seen it.
12:39.39viperdudeyes... it adds to to the SIP message but not to the invite
12:40.35viperdudei need phone-context and user to become part of the SIP URI
12:42.23Nobbie<PROTECTED>
12:44.57*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)
12:46.05anonymouz666who have a nokia e61?
12:46.34pollerI have a e60 connected to astersik
12:48.04bcnlare there any known issues with IAX2 calls between asterisk 1.2.x boxes and 1.4.0?
12:53.07anonymouz666poller: have time for a question?
12:53.13pollerSure
12:53.33anonymouz666I got disconnected all the time... and the SIP register don't reach the server...
12:53.41anonymouz666Using wireless
12:53.46anonymouz666lastest firmware
12:53.54pollerTo your own asterisk?
12:53.56anonymouz666but, I can use WEB with no problem
12:54.06anonymouz666yes
12:54.44pollerIs "Registration" set to "Always on"?
12:54.47pollerIn the phone
12:54.56anonymouz666yes... i think it's wireless stuff
12:54.57*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
12:55.18pollerOk, have you tried to change channel on the wlan?
12:55.27*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
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12:56.20anonymouz666nops
12:56.33anonymouz666poller: where I can see this option?
12:56.58pollerDepends on your wifi equipment
12:59.04MrChimpyBackground will just do digits AFAIK
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13:05.24anonymouz666poller: I got the mensage "unable to connect to connecting network"
13:06.24pollerOn the phone?
13:06.37Ahrimanesany digium folks around?
13:06.54anonymouz666poller: yes...
13:07.09pollerFancy..
13:07.15anonymouz666Jan  3 19:45:06 wlan0: A wireless client is associated - 00:13:FD:5B:E4:8F
13:07.15anonymouz666Jan  3 19:45:13 wlan0: A wireless client is disassociated - 00:13:FD:5B:E4:8F
13:07.42pollerUsing any crypto? Mac filter?
13:08.03anonymouz666not that I know
13:08.09anonymouz666where I can see that?
13:09.01pollerWhat kind of access point are you using? Looks like some kind of *nix system.
13:09.11bkruse_homeAhrimanes: whats up?
13:10.08Ahrimanesbkruse_home, it says at http://www.kirktelecom.com/ that * 1.2.11 is supposed to work with their dect products, just wondering if any digium people were involved in these tests, our kirk test system does not behave well..
13:11.42*** join/#asterisk abstr4kt (n=abstr4kt@150.146.16.151)
13:11.45abstr4kthi
13:11.51pollerHi
13:12.05abstr4ktsomebody speak italian?
13:12.15bkruse_homei do not think they are a digium certified reseller, or anything close to. I think they just use asterisk.
13:12.43anonymouz666poller: it's a ZPlus-G120
13:12.47anonymouz666AP
13:13.03*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
13:13.12abstr4ktis it possible to make videoconferencing qhit asterisk?
13:13.20polleranonymouz666: Okey, well, check the settings and make sure it's the same on the phone.
13:13.20abstr4ktwhit*
13:13.32anonymouz666poller: any suggestion about configuration?
13:13.33pollerBut if regular "surfing" works it's probably fine.
13:13.48UVSoftis there a function to test if a device is busy?
13:13.55pollerI'm using wpa2 protection for my e60.
13:14.17pollere60 with the old firmware didn't wanna play with wpa1.
13:14.42Ahrimanesbkruse_home, ok damn.. would looooove to have it working so i can get rid of ata's and single phone dect bases
13:15.03anonymouz666i am using the lastest
13:15.07anonymouz666firmware
13:15.55abstr4ktdo i need other kind of server for menage a video-call?
13:16.44*** join/#asterisk McGee (n=rootgauf@p54957891.dip.t-dialin.net)
13:16.57anonymouz666poller: its open network, there is no auth
13:17.17McGeeHi, where does zttool get its data from?
13:18.38McGeeIf i try to change a span from [NT] to [TE] is zapata.conf the wrong place ?
13:18.50Nobbie<PROTECTED>
13:19.10Nobbieit's config is from /etc/zaptel.conf and /etc/asterisk/zapata.conf
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13:37.58smackuswhat could cause me to have at the end of the day 1 call and 5 active channels. When only one active channel is listed when i do a show channels?
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13:55.00alexandreposwhat a better protocol for pap2 w asterisk ?
13:55.16drraysmackus? is it a zap device and a channel bank?
13:55.27*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
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13:55.41smackusit is a zap device, but no channel bank. all sip out of asterisk to phones
13:56.07drrayso the zap device is being held open?
13:56.08*** join/#asterisk bkruse_home (n=kruz@69.73.127.92)
13:56.10drrayor the sip?
13:56.50smackusif it is, its not showing up in the show channels.
13:57.06smackusjust the one active call showed up
13:57.09smackusand no one was there
13:57.18smackusbut there were 4 active channels.
13:57.48smackusthe one call that was there was a sip channel to a specific extension
13:58.04*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
13:58.26alexandreposwhat a better protocol for pap2 w asterisk ?
13:59.53*** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir)
14:01.51doolphalexandrepos, sip
14:06.02alexandreposdoolph: ok but better codec ?
14:10.16Chris-NBhi
14:10.17*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
14:10.19Chris-NBanyone using mISDN?
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14:43.06hwtdoes BLF work on thomson st2030 without blf patch now?
14:44.19UVSoftCould you help me? My dialplan looks like: exten => 0, 1, Answer() than Playtones(dial) than WaitExten(60), so and I'd like asterisk to stop playing when the first DTMF has come, how to do it?
14:45.23mercestesVVSoft:  ...what are you trying to do?
14:46.28UVSoftso user hears dial tone, than types '0', than wait for another dial tone, what types the number
14:47.27mercestesHave you looked at DISA?
14:47.40UVSofti even don't know what it is)
14:47.50mercestesGoogle asterisk DISA
14:47.51mercestes~disa
14:47.52jbotfrom memory, disa is direct inward system access.  show application disa
14:48.01mercestes~botsnack
14:48.01jbotaw, gee, mercestes
14:49.57UVSoft'0' is a way out
14:50.11UVSofti mean user can do international calls through it
14:50.18mercestesOk.
14:50.52mercestesyou don't *have* to do it that way you know.
14:51.12UVSoftso disa will help me
14:51.38UVSoftwill disa help me?)
14:51.41mercestesShould.
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14:53.56*** mode/#asterisk [+o anthm] by ChanServ
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14:55.48UVSoftfrom show application disa "...allows someone from outside the telephone switch to obtain an internal system deialtone..."
14:56.37*** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net)
14:56.39mercestesYes.
14:57.00mercestesIt plays a false dialtone and allows someone to "dial out" from within the system.
14:57.10mercestesif it seems retarded to use DISA when your already in the system....it is.
14:57.48mercestesbut it does what you want it to do.  Your *in* the system with a dialtone saying "it's ok to dial whatever you want" and you want to require them to dial a "0" to give them this false impressioni that they are on some seperate line or programming or something....which they are not.
14:58.06mercestesand using Disa on "0" will return that secondary false dialtone (that makes no difference at all) and allow them to dial wahtever they want again.
14:58.19*** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue)
14:58.36mercestesIn fact.....if you dont' set up your contexts exactly right......then your users could still dial any number they want without the "0"/
14:58.46mercesteswhat you are doing is entirely aesthetic.
14:59.20mercestesit's like putting racing stripes on a tri-cycle.  It doesn't make it go any faster and definately doesnt' make it any cooler, but to the untrained eye, it might look cooler.
15:00.31mercestesIt's like dialing "9" for an "outside line"    there is no outside line.  It only does that if you program it and give it that illusion.
15:01.03mercestesbut I pattern match _9 and without the 9 for those old people who just cannot understand a system that doesn't have an "outside line"
15:01.16mercestesbut I just strip the 9 supreficially and proceed.
15:03.34UVSoftthanks a lot, I'll try it
15:04.01mercestesGood luck
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15:06.12*** join/#asterisk viler (i=1000@ip-70-228.telesat.com.co)
15:07.43littleballhello, who can help me about SMS receiving issue? i got lots of "- SMS TX 92 01 FF 6E" when i send sms to the fix line number
15:08.23viperdude#/join #ser
15:11.50UVSoftmercestes: it works, thanks
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15:17.14mercestesVVSoft.  NP.
15:17.33*** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net)
15:22.09mercestesVVSoft:  Msg me if you want a paypal address.
15:22.19mercesteshello?  Hellllllo?  *listens to crickets*  dangit!
15:22.24mike052278hey does anyone know if the zoom ata is any good?
15:22.40mercestesmike052278:  Never 'eard of it.  Is it new?
15:23.15mike052278i dont think so
15:23.18mike052278one sec
15:23.20mike052278i'll get the link
15:23.29mike052278http://www.microcenter.com/single_product_results.phtml?product_id=0235685
15:23.34mike052278they sell them locally by me
15:23.35mike052278:)
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15:24.28mercestesit's locked down to a specific carrier.
15:24.33mercesteseither that, or they are flat out liars
15:24.36mike052278lol
15:24.43mercesteseither way, I don't think it's what you want.
15:24.45mike052278i dunno im gonna check it out on my lunch
15:24.49bhrobinsonI would stick to a gradnstream or a sipura
15:24.53viperdudei use zoom ata at home
15:24.54mike052278hmm ok
15:24.59mike052278is it any good?
15:25.05bhrobinsongrandstream
15:25.11smackusI have an iax provider i use on my asterisk box at home. i register to them, but periodically my registration drops and the only way to get it back is to restart asterisk. iax2 reload does not do it... anyone know a fix for this?
15:25.12mike052278its just for my house phone so i dont need top of the line
15:25.14*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)
15:25.14viperdudeits ok, not rebooted it in months
15:25.18*** join/#asterisk jpe-nyc (n=jpe@p77-37.acedsl.com)
15:25.52viperdudei run the voip provider though
15:25.55*** join/#asterisk vlt (n=dm@p54B322AE.dip0.t-ipconnect.de)
15:26.10viperdudeso not had experience using it with any other service
15:26.38bhrobinsonhttp://www.voipsupply.com/product_info.php?products_id=518
15:26.38*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)
15:26.57bhrobinsonI use these exclusively and no issues with even faxing
15:27.52littleballhello, anyone did successfully receiving sms over ZAP channels?
15:28.21mercestesbhrobinson:  You haven't been in business very long, have you?
15:28.50bhrobinsonnot in the voip market. only about 6 months...
15:28.54mercestesbhrobinson:  give it time.  Your world will collapse.
15:29.02bhrobinsonlol.. love you too
15:29.05mercestesbhrobinson:  hehe, that I was able to guess that should give you an idea...;)
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15:31.51phearlesshello asterisk folks !
15:32.02phearlessI got a really easy question, but hard for me to solve
15:32.02phearlesshttp://rafb.net/p/cP7H2e75.html
15:32.15phearlessplease have a quick look at this
15:32.41smackuswhat is your question?
15:32.51phearlesswhen I use WaitExten with "i", I can not dial any extension
15:33.09phearlesswhen I press 4 (for 408), I got "pbx-invalid" played
15:33.37smackusis 408 a valid extension in that context you are calling it from?
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15:35.02phearless408 is an internal extension
15:35.14phearlessI do not know if it is valid in [subenter]
15:35.23phearless[subenter] is just a loop to enter a local extension
15:35.28phearlessto call the extension
15:36.06phearlesshow can I detect if it is a local extension?
15:36.17smackuscan you paste up the whole extensions.conf so i can see how it is all included?
15:36.22*** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk)
15:36.43phearlessyes
15:37.14smackusyou have to be able to reach extension 408 from where you are at in your dial plan... ie if you are actually in your subenter context, you have to either have exten => 408 in it, or include => the context where 408 is found
15:37.28phearlesshttp://rafb.net/p/G2Wszg18.html
15:38.12phearless" or include => the context where 408 is found" hum maybe
15:38.26phearlessbut how can I create a context that detect if this is a valid local extension ?
15:39.26smackusif it is not included, it is not valid. so since 408 would be part of default, you have to make sure that subenter can dial extensions within default
15:40.21smackusyour dialplan determines if the extension is valid... exten => _[45]XX,1,Dial(SIP/${EXTEN},10,tT)
15:40.33smackusif the extension matches that when called from default, it is valid
15:44.56phearlessso I have to include default ?
15:45.40ThoMeHallo
15:45.50ThoMeviperdude: Endlich mal ein deutscher hier? ;)
15:45.51smackuscorrect... I did not look to see how your calls are getting into the subenter context... but try adding include => default at the very end of the subenter context
15:46.00smackusnein
15:46.03smackuskeine
15:46.10ThoMesmackus: :-)
15:46.23ThoMevlt: Du bist doch 'n deutscher.. gell? ;)
15:46.34smackusneil, bin ich nicht
15:46.43ThoMesmackus: DU bist nicht "vlt".
15:46.51ThoMesmackus: Und nun halt die Schnute.
15:47.26phearlessthank you smackus
15:47.35phearlessit seems to works better
15:47.40smackusphearless: good
15:47.46phearlessI wil try to improve this a little bit
15:48.09smackusinclude can also be used to include files outside of the extensions.conf... check out the command on the wiki
15:49.48*** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)
15:51.11phearlessok smackus
15:52.11phearlessand how can I check of the local ext is really valid, for example 401-409 or 412-415 ?
15:52.40phearlessthese are the only avaliable extensions here
15:52.54smackusyou mean to see if the number is real or if it matches your dialplan?
15:53.29phearlessyou mean the physical phone dialplan?
15:53.40phearlessI talk about extensions.conf
15:54.07phearlessthere are people using 401-409 and 412-415, but not any others ext
15:55.27*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
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15:56.00phearlesssmackus: make sense?
15:56.16*** join/#asterisk suma (n=suma@cm207.omega186.maxonline.com.sg)
16:00.59smackusyes, it make sense
16:01.13smackusread the wiki on pattern matching
16:01.30smackusyour dialplan will specify what is valid.
16:01.37phearlessokay
16:05.00phearless_4[01-0912-15]
16:05.01phearless?
16:05.14phearlessthere are weird regex : http://www.voip-info.org/wiki-Asterisk+Dialplan+Patterns
16:05.18phearlessfor 401-409 and 412-415
16:06.03smackusin asterisk cli> do a show dialplan 401@default and so on and see if it worked correctly for you
16:07.34phearlessno it does not work
16:07.39phearlessThere is no existence of 401@default extension
16:07.54*** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com)
16:08.11*** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com)
16:08.37phearlessthey say that 7[04-9] is 70, 74, 75, 76, 77, 78 or 79
16:09.35phearless40[1-9] works for 401-409
16:10.22phearlessbut it would be retarded for me to use 2 lines, one for 401-409, and one for 412-415
16:10.57smackuswhy retarded?
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16:12.07phearlessthere must be a way to define 401-409 and 412-415
16:12.13phearlesslike in any regex
16:12.19smackusyeah... there is.
16:12.45smackusi have not slept since yesterday, so i havent tried to figure it out. gimme a sec
16:14.14smackusohm wait.. you cant, cuz for the 40x, you need 1-9 in place of the x, but you want to exclude x on 41X unless it matches 2-5
16:14.15smackuswont work
16:14.18smackusyou need to lines
16:14.25*** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi)
16:15.08phearlessokay
16:15.12phearlessthank you smackus !
16:15.15phearlessI'll do this
16:15.16smackusdoes that make sense?
16:15.27*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
16:15.37smackusunless anyone can show me how to do it otherwise... i dont know how to do it with one line
16:16.35*** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com)
16:16.51FarrisGWhere should I start in trying to troubleshoot why dtmf is occasionally decoded wrong? About 15% of our calls end up going to the wrong extension. When it was first reported, I just chalked it up to typos, but I've tested myself and it clearly misreads quite a bit.
16:17.13*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)
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16:17.38smackusFarrisG: try relaxdtmf in the zapata.conf
16:17.55smackuswe had the same issue... that fixed it
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16:28.34Qwell[]~lart brimstone
16:29.49jpe-nycDoes anyone know of any chat line apps, the dating service kind?
16:31.45Qwell[]app_meetme with a small dialplan?
16:31.58Qwell[]could probably hack something up in under an hour, all in dialplan
16:32.47viperdudearen't dating,... listen to the msg and then leave your msg for later retrival
16:33.00Qwell[]viperdude: yeah, it's voicemail + meetme, heh
16:33.34Nuggetstill doesn't sound too cumbersome with account information stored in a database via odbc.
16:33.39viperdudehmm never called one lol so not sure about the meetme
16:33.46jpe-nyci found something called match n chat that was just what I was looking for, but it was missing parts
16:33.59Qwell[]Nugget: exactly
16:35.31jpe-nycdirect me to a paste bin and I will put up what i found to give an idea
16:35.53Qwell[]~paste
16:35.56jboti guess paste is http://rafb.net/paste/
16:36.12Qwell[]wtf
16:36.43jpe-nychttp://rafb.net/p/ZYI9r421.html
16:37.16jpe-nycit didnt have any author or contact info in it, i found it in a paste bin doing a google search
16:37.30Nuggeteh
16:37.32Nuggeter, heh.
16:37.49jpe-nycwell the authors name was there but no contact info
16:39.37Qwell[]Did the author give you rights to use it?
16:40.10jpe-nyci cant find him, to inquire,
16:40.50jpe-nycdo you know the author or how I might contact him?
16:44.33*** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
16:47.45FarrisGsmackus: Just read your advice on relaxdtmf... We're using sip trunks, not zap. So I don't think this will take care of it.
16:48.05smackusyu are correct
16:48.16smackusi wonder if there is something similar
16:48.19smackusfor sip
16:48.23mercestesI want to call a dating chatline.  it sounds like fun
16:48.56*** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com)
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16:52.55CrazyTuxexten => _89X.,1,Voicemail(u${ARG1}) anyone know whats wrong with that, it says its deprecated?
16:53.31mercestesCrazyTux:  What v. of asterisk?  And what does it suggest instead?
16:53.32*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)
16:53.45CrazyTuxmercestes, 1.4, it says something about the u
16:54.05CrazyTuxmercestes, I want to forward the ruri as the mailbox number, how can I do that?
16:55.10mercestesCrazyTux:  Check teh changelog
16:55.27*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
16:55.27*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
16:55.33CrazyTuxmercestes, I fixed that part, but it won't pick up the mailbox number automatically?
16:55.42mercestes${Exten}?
16:56.16*** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-60d9499acd437957)
16:56.43jpe-nycwhat did you think of that stuff i pasted about the chat linepeople?
16:56.47CrazyTuxmercestes, how can I strip 2 from the EXTEN?
16:57.07CrazyTuxmercestes, I want to strip 12 from 12XXXXXXXXXX
16:57.15CrazyTuxmercestes, which would be the EXTEN
16:57.17Qwell[]${EXTEN:2}
16:57.46CrazyTuxQwell, thanks.
16:57.56Qwell[]CrazyTux: That'll be $19.95
16:58.12sevardHe's serious.
16:58.24CrazyTuxQwell[], thank you.
17:01.55CrazyTuxHow can I send the user to check his voicemail and pre-supply the mailbox?
17:02.19Qwell[]show application voicemailmain
17:02.59*** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir)
17:03.24mercestesThat'll be antoher 19.95
17:03.34Qwell[]rates went up
17:03.43mercestesdamn inflation
17:03.54CrazyTuxlol
17:04.03CrazyTuxI am using voicemailmain
17:04.10CrazyTuxdosent seem to pickup the username / mailbox
17:04.27CrazyTuxexten => _899X.,1,VoicemailMain(${EXTEN:2})
17:04.32mercestessomeone randomly posted to IRC a "request for bid" on servers from A through X using multiple configurations of OpenSER and Asterisk with faxing, etc.  I bid 2.5 million.  They laughed.  I laughed back.
17:04.33mercestesidiots.
17:04.34*** join/#asterisk fnordus (n=dnall@24.85.128.203)
17:05.06CrazyTuxmercestes, lol
17:05.13mercestesCrazyTux:  Are you sure it's not ${EXTEN:3}?
17:05.20CrazyTuxmercestes, ah shit, thats it.
17:05.23CrazyTuxexcuse my lang.
17:05.28mercestesThat'll be 2.5 millino.
17:05.31CrazyTuxlol
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17:24.02jservehi all
17:25.10CrazyTuxmercestes, so say I did ${EXTEN:-3} would it strip from the end?
17:27.34mercestesCrazyTux:  no.
17:29.31mercestesCrazyTux:  wait.
17:29.35mercestesCrazyTux:  yes.
17:29.39*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)
17:30.20CrazyTuxmercestes, I'm starting to pick this up :)
17:30.36mercesteslol
17:30.43mercesteshttp://www.voip-info.org/wiki-Asterisk+variables
17:30.47mercestesthank link, search for "substrings"
17:30.53mercestess/thank/that
17:31.03mercestess/thank/that/
17:31.08mercestesdamnit
17:35.43*** join/#asterisk kratzers (n=kratzers@martha.pa.net)
17:37.05kratzersis there an easy way to record to a new file when a call is transferred?
17:38.20mercesteskratzers:  Only when a call is transferred?
17:38.48mercesteswait.
17:39.02mercestesI'm going to assume you are already recording before the call is transferred, right?
17:39.08kratzerscorrect
17:39.15*** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net)
17:39.16mercestesand you want to stop recording to file A; and begin recording to file B: after the call is transferred..aye?
17:39.25kratzersyep
17:40.04kratzersand using the Transfer button on the phone, but *9 or another combination
17:40.09kratzersnot*
17:40.51mercestesTrain your users to use a special "transfer" extension to transfer.
17:41.01kratzersheh, I don't think that's a possibility
17:41.04*** join/#asterisk bigpresh (n=bigpresh@supernova.preshweb.co.uk)
17:41.07mercestesso if you have extension 345, train them to transfer to 9345 instead of just 345.
17:41.24mercestesthen you can use ChangeMonitor or Stopmonitor and a new Monitor command.
17:41.49mercestesor you can preceed every "extension" with an implicit "StopMonitor" whether you are recording or not.
17:42.05mercestesand then it will forcefeed a new recording on every call.
17:42.13mercestesbut then every call to that extension will result in a recording
17:42.19mercestesand probably a nice "Warning" in the CLI
17:42.33kratzershmm
17:43.16UVSofthey guys if an extension exited with non-zero status, what does it mean? can i catch and process this case?
17:44.52*** join/#asterisk nextime (n=nextime@unaffiliated/nextime)
17:45.18mercestesVVSoft:  0 is exitting with no error.  non-zero is exitting with some error.
17:45.31mercestesVVSoft:  "What it means" depends entirely on what your doing in your extension.
17:45.32UVSoftso how to deal with it
17:45.41mercestesset verbose 99
17:45.43UVSoftok i'll try to explane
17:45.44mercestesand see what's breaking
17:46.08UVSoftset verbose 99, where?
17:46.20UVSoftdialplan?
17:46.31UVSoftSet(verbose=99) or what?
17:47.34mercestes...
17:47.59mercestesls extensions.*  What files are listed?
17:48.17*** part/#asterisk nextime (n=nextime@unaffiliated/nextime)
17:48.41UVSoftso for example there's a phone connected to FXO through TLE, i pick it up and dial the number, asterisk connects me with FXS device to another phone, i pick up that phone too, than i hang up the second phone and ... non-zero exitting, and what if i'd like to dial another number without redialing to FXO
17:49.06UVSoftael conf
17:49.27UVSoftextensions.ael extensions.conf
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17:52.45UVSoftmercestes: ??
17:54.09UVSoftso u mean asterisk -vvvvvvvvvvvvvvvvvvvvvvc?
17:55.27*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
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18:00.41UVSoftmercestes: could you tell me something please? just to be sure you're alive)
18:03.58*** join/#asterisk ez` (n=Ez@c66.203.210-59.clta.globetrotter.net)
18:04.37*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)
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18:08.59*** join/#asterisk spr1te (n=spr1te@213.227.193.74)
18:13.23NeonLevelgood day everyone, in a sip trunk if I change the type=friend to type=user then the dtmfmode won't work, is this a bug?
18:13.33NeonLevelAsterisk 1.2.14 built by root @ pbx on a i686 running Linux on 2007-01-16 18:06:24 UTC
18:15.37*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
18:15.41*** join/#asterisk _deg_ (n=deg@200.195.161.164)
18:16.11_deg_Is there a way to get the password value when the users changes his passowrd durinf VoiceMailMain?
18:16.26_deg_I need to update a database qith the new value. Usinf Realtim Static...
18:17.57*** join/#asterisk foxxtrot (n=craig@c-67-185-0-172.hsd1.wa.comcast.net)
18:18.35_deg_There is abug on VoiceMailMain when using Realtim Static, that dont allow a user to change his password...
18:19.10_deg_Is there a workaround for that? Something with externpass...
18:22.20xpotanyone know how many concurrent connections asterisk can handle on one server?
18:23.04Jinglesthat seems to depend on the resources available to the system.
18:23.27xpotso the asterisk app itself has no limits?
18:23.40Jinglesright. there's no 'hardcoded limitation'.
18:23.49xpotok, thank you jingles
18:26.36*** join/#asterisk lullabud (n=lullabud@12.24.42.67)
18:28.14slimahey, I need litle help, http://rafb.net/p/fKIciF60.html
18:29.26*** join/#asterisk colinm_ (n=colol@VDSL-130-13-99-91.PHNX.QWEST.NET)
18:29.34Jingleswhy are you using 'friend' and not 'peer'?
18:30.03Jinglesoh, and using 1.4.0 is generally considered 'a bad idea' unless you're just testing it all out.
18:30.17*** join/#asterisk nosbig (n=nosbig@rrcs-70-62-223-2.central.biz.rr.com)
18:31.18nosbigI am curious if the line presence buttons on the Cisco 7941 and 7961 can display line status when used with the SIP firmware and Asterisk?
18:34.00syzygyBSDCuriousity killed the cat
18:34.51Marty-OTThey, where can I get info on T38?
18:35.00slimaJingles: I changed friend to peer but still doesn`t work. (sorry for my english)
18:36.34anonymouz666Jingles: 1.4 should be stable
18:36.36anonymouz666hehe
18:36.49*** join/#asterisk rickb|server (n=admin@cpe-71-66-110-248.neo.res.rr.com)
18:37.16rickb|serverHow would I make 411 point to 1800free411? in a dial pattern?
18:40.40*** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br)
18:40.49*** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca)
18:41.02DeeJayTwohi... we have a mini-itx via c3 800mhz platform to run asterisk
18:41.10De_Monis anyone using the voicechangedial app?
18:41.14DeeJayTwoit works great but when initiating a call we here a click on every open rtp channels...
18:41.24DeeJayTwoand tu cpu cranks up at this same moment
18:41.36DeeJayTwoand the cpu..
18:42.05DeeJayTwowe hear... a click ;)
18:42.38*** join/#asterisk McLazarus (n=mcallist@pool-72-78-49-80.phlapa.east.verizon.net)
18:42.56DeeJayTwoany idea about what could be the problem?
18:43.08*** part/#asterisk lullabud (n=lullabud@12.24.42.67)
18:43.11DeeJayTwoI don't see the asterisk process using much CPU it is quite idle...
18:43.29DeeJayTwobut the "us" cpu summary part of "top" show 90% usage...
18:43.33DeeJayTwoand the idle "id" is 0%
18:43.50syzygyBSDrickb|server: exten => 411,1,Goto(18003733411)
18:44.07syzygyBSDrickb|server: exten => 411,1,Goto(18003733411,1)
18:44.22rickb|server:)
18:44.23rickb|serverthanks
18:44.37tzafrir_laptopDeeJayTwo, if it is a user process, which process is it?
18:45.05tzafrir_laptophint: to sort by CPU usage, if this is not the default, press p
18:45.28DeeJayTwotzafrir_laptop : I can't see any specific process using all the CPU..
18:45.32DeeJayTwothat's quite weird...
18:45.41DeeJayTwook
18:45.58tzafrir_laptopDeeJayTwo, which system is it? Which kernel?
18:46.14DeeJayTwodebian with 2.6.8
18:46.49DeeJayTwoAs I said, everything is fine but this "performance" problem on call initiation...
18:48.18DeeJayTwothat's weird...
18:48.29DeeJayTwothere's no specific process using much cpu at the top..
18:48.33DeeJayTwothe maximum is 0,6%
18:48.47DeeJayTwoand they're sorted descending...
18:48.54DeeJayTwolooks like an IO problem..
18:48.56DeeJayTwocould it be?
18:55.27*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
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18:55.33*** join/#asterisk tkrin (n=tkrin@201.sub-70-217-83.myvzw.com)
18:55.43tkrinhi all
18:55.58tkrinhas anyone compiled asterisk-addons 1.5 on Solaris 10 x86?
18:58.06*** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com)
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18:58.54tzafrir_laptopDeeJayTwo, I encountered the same problem with one server.
18:59.20vooduhalDoes anyone know how to destroy these calls that are no longer up?
18:59.20vooduhalhttp://pastebin.ca/329415
18:59.26DeeJayTwotzafrir_laptop : Did you just change the server or have you found the cause?
18:59.32tzafrir_laptopIt seems that top, ps and just about any program showed the CPU time devided by 100
19:00.23tzafrir_laptopI have not resolved the problem. I had no time to trace it to either procps or the kernel
19:01.08tzafrir_laptopIIRC a small kerne upgrade did not solve things
19:01.26tzafrir_laptopWhat server is it? Some IBM server, by any chance?
19:01.52*** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch)
19:03.36tkrinWhen trying to compile the addons on a Solaris x86 box, I get the error shown at http://pastebin.com/868056
19:03.39hardwiremike052278: shipped.. sent you an email
19:03.42DeeJayTwo~nope..
19:03.45*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
19:03.47DeeJayTwoit's a mini-itx board...
19:03.52DeeJayTwoVIA C3 CPU 800mhz
19:04.08DeeJayTwowith like 512MB SDRAM.
19:04.12tkrinAny ideas?  I'm not a Solaris guru, so am not sure if it's maybe a path or env issue?
19:04.40hardwireuse linux?
19:05.06tkrinhardwire:  not my server. :(  Can't force that change.
19:05.08hardwireor at least something that supports zap :)
19:05.27tkrinit's strictly a voicemail server.
19:05.35syzygyBSDtkrin: if you have root you can force a change... they just wouldn't like you
19:05.53tkrinsyzygyBSD:  Exactly, and then I'd not get paid.  ;)
19:05.59hardwirethere are no binaries?
19:06.13tkrinNot that I could find.
19:06.13syzygyBSDmeh, money is overratted
19:06.24hardwireits pretty boring really
19:06.36tkrinsynthetiq:  :)  lol
19:08.11rickb|serverOk. I don't want to sound like an idiot. I just bought a SPA 2002. I thought the applications for use where interesting. What exactly could I do with it?
19:08.30Qwell[]rudholm: fix email, kthx
19:16.11*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
19:19.03rudholmQwell[]: np
19:19.55rudholmQwell[]: you using the Beta?
19:20.14Qwell[]rudholm: dunno, somebody is complaining that smtp auth on the business email thingie is b0rked
19:21.00[TK]D-Fenderrickb|server : You bought it and you don't even know what you can do with it?
19:21.22rudholmQwell[]: oh
19:22.38*** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net)
19:22.59elriahHi all.  When a SIP regsitration fails so many times and finally gives up, is there a way to tell it to keep trying in the peer definition?
19:24.04bkruseb0rked!
19:26.15*** join/#asterisk olsen (n=diego@200.61.236.33)
19:26.16*** part/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net)
19:27.44Jingleselriah: how many times is 'so many times'?
19:27.50Qwell[]rudholm: fixed yet? :P
19:27.58JinglesI had one that had tried a few hundred times as of this morning, and was still trying.
19:28.08rudholmQwell[]: not my department :)
19:28.12Qwell[]pfft
19:29.06Bobthehunterdoes aterisk biz editon upgraded with realtime yet ?
19:29.07[TK]D-FenderJingles : .... he already left.  Gotta love people who stick around to actually really learn something...
19:29.07Bobthehunteror still old 1.09
19:29.14Qwell[]Bobthehunter: version B is based on 1.2
19:29.17*** join/#asterisk Ng (n=cmsj@mairukipa.tenshu.net)
19:29.27Ngdon't suppose there's anyone around who uses Polycom phones?
19:29.28Bobthehunterok and any warranty on no deadlocks etc..
19:29.34*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)
19:29.35Bobthehunteror its 1$k out the window
19:29.46Bobthehuntermeanig we have a support option realtime with them ?
19:29.51NgI've got an IP430 connecting to an asterisk box and it makes/receives calls ok, but if I type in a SIP URL and call it, it doesn't really do anything
19:31.45*** join/#asterisk Tamarisk (n=adrian@adsl106242.timewarp.co.uk)
19:34.02TamariskI have searched on google to find a way to completly and safely remove 1.2.4.  One links is broken and another does not seem to work
19:34.51Tamariskwas there anything special about 1.2.4 that I need to consider, or can I search for all directories and files with asterisk and zaptel and trash them?
19:36.42wunderkinare you upgrading or something
19:37.18TamariskI would like to look at it but my suse10 is giving a few issues and need to consider removing it all before I try again
19:37.48wunderkin.. umm.. you're using a package? blah, don't
19:38.12TamariskNo I did not use a package, some many months ago I installed from source
19:38.23wunderkinso what's the problem
19:38.43Nghmm, maybe it is working. Are there any good test sip urls?
19:38.49*** join/#asterisk zotz (n=zotz@24.244.163.157)
19:39.11Nga radio station, or a menu system. just something a bit more interesting than an echo service ;)
19:40.24TamariskWhen I did the search I found a site that said download this script install and run, but script no longer there. I am not a very confident linux operator so any simple script or instructions to follow would be wonderful
19:40.24*** join/#asterisk olsen (n=diego@200.61.236.33)
19:40.30wunderkinuh..
19:41.42*** join/#asterisk Assid (i=assid@59.183.45.152)
19:43.59rudholmI have an exten line that only works when it is directly part of a context.  when it is part of an included context, it never matches a dialed number.
19:44.06rudholmwhich seems to me shouldn't happen
19:44.47*** join/#asterisk dasenjo (n=dasenjo@190.5.196.105)
19:45.22[TK]D-Fenderrudholm : perhaps you should try SHOWING us something......
19:45.23[TK]D-Fender~pb
19:45.24jbotwell, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
19:45.25[TK]D-Fender^^^
19:45.29rudholmthe only thing out of the ordinary about the exten line in question is that it uses the regex operator to truncate the dialed EXTEN
19:45.53rudholmit's one line
19:45.56rudholmexten => _1311377XXXX,1,Dial(${phil}/${EXTEN:4:7})
19:46.38[TK]D-Fenderrudholm : Clearly you are not thinking big enough.  Put into the "included" context style as before and pastebin the WHOLE thing.
19:46.53*** part/#asterisk BarnacleBob (n=karl@38.99.18.98)
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19:47.27rudholm[felix-2]
19:47.27rudholminclude => internal
19:47.32[TK]D-Fenderrudholm : And FYI, exten => _1311377XXXX,1,Dial(${phil}/${EXTEN:4}) would do jsut as well.
19:47.32rudholmthat's the whole thing
19:47.37*** join/#asterisk UVSoft (n=UVSoft@c7204-ge2-500.etelecom.ru)
19:47.39rudholmthat's the whole context
19:47.50tumypHi
19:48.00rudholm[felix-2]
19:48.00rudholminclude => internal
19:48.05[TK]D-Fenderrudholm : You said you tried using an INCLUDED context.  put everything back, and PASTBIN it. do NOT copy it here
19:48.36tumypwho has experience in two wire E&M interface and asterisk here ?
19:48.44rudholmthe whole of the included context isn't really relevant
19:48.50tumypi need help
19:49.03rudholmand pb isn't really relevant for one and two lines of text
19:49.38NuggetI'd rather people err on the side of caution when it comes to that decision (to pastebin or paste)
19:50.00[TK]D-Fenderrudholm : I never say you actually prove that is was IN [internal] <------ stop holding back.  Clearly you are doing something wrong and if you want help you should try and cooperate
19:50.01rudholmyeah, I agree, but two lines?
19:50.03tumypdoes anybody ?
19:50.36[TK]D-Fenderrudholm : its at least 4....
19:51.02rudholmwell, it is in internal, and works for devices that are in the internal context
19:51.05[TK]D-Fendertumyp : Ask your SPECIFIC question and you might get a better answer.
19:51.20[TK]D-Fenderrudholm : Stop TALKING about it and just pastbin the damned thing.
19:51.24tumypThanks for the answer :)
19:51.38rudholm[TK]D-Fender: nevermind, I'm not really interested in your help at this point.
19:51.40tumypI have adtran 616 with FXS ports
19:52.16tumypand I need answer supervision
19:52.36tumypthe only way to use it is to enable E&M on FXS
19:52.42[TK]D-Fenderrudholm : Well you really aren't showing the big picture, and includes work for everybody else.
19:52.52tumypto use E&M signalization
19:53.01rudholmyeah, they work for me too
19:53.04tumypbut I can not
19:53.06rudholmjust not in this one specific case
19:53.09rudholmbut dont' worry about it
19:53.21[TK]D-Fenderrudholm : Then how about showing all of it.
19:53.29[TK]D-Fenderrudholm : Whats the problem here?
19:53.38*** part/#asterisk Tamarisk (n=adrian@adsl106242.timewarp.co.uk)
19:53.52tumypwhen I'm changing to em in zaptel.conf - ztcfg gives me an error
19:54.07*** join/#asterisk tkrin (n=tkrin@112.sub-70-217-39.myvzw.com)
19:54.21[TK]D-Fendertumyp : What card are you using?
19:54.28tumypDigium
19:54.38tumypTDM24P or something like
19:54.40[TK]D-Fendertumyp : not jsut the brand please, the model.....
19:54.47tumypjust a minute
19:55.00[TK]D-Fendertumyp : Ok, thats almost a valid model number.
19:55.27*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
19:55.29*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
19:55.34tumypWCTDM2400P
19:55.52tumypany ideas ?
19:56.06tumypI'm never used E&M on asterisk
19:56.11tumyponly on cisco ..
19:56.46[TK]D-Fendertumyp : Umm... the TDM2400P supports FXO/FXS port based ont he modules you have installed.
19:57.01[TK]D-Fendertumyp : I fail to see how this channel bank factors in.
19:57.05tumypyeah .. right ... :)
19:57.25high-rezwhat about E&M on asterisk?  it works fine
19:57.26tumypI've only FXO ports
19:57.34[TK]D-Fendertumyp : and the TDM2400P doesn't do E&M.  For that you'd need a digital line interface card.
19:57.53tumypyou're right, sorry
19:58.06tumypI have to sleep more than now :)
19:58.18tumypthanks
19:58.47[TK]D-Fendertumyp : What you could do is get something like a TE110P and use your channel bank instead, but both are ways to do basically the same thing : bring analog lines into *
19:58.47*** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa)
19:59.08tumypI need only answer supervision
19:59.12*** join/#asterisk darby_t (n=tom@aapu113.neoplus.adsl.tpnet.pl)
19:59.12tumyponly this
20:00.19tumyp[TK]D-Fender: all I need is correct detection of start and end of the conversation
20:00.23[TK]D-Fendertumyp : Ok, jsut to be clear, are you loking to plug in analog LINES, or analog PHONE?
20:00.49tumyp[TK]D-Fender, I have analog lines
20:01.09tumypand they are ;lugged in into TDM2400
20:02.38[TK]D-Fendertumyp : Ok, answer supervision is really not great on analog channels.  *'s support for it on say the TDM cards is pretty weak.  You should really consider getting a PRI.
20:02.56[TK]D-Fendertumyp : Even a partial.
20:03.03tumypI know it :), but I can not
20:03.35tumypIf I'll have PRI - it will be solution of all my troubles
20:04.14tumypbut I have analog lines and .... I need to configure it
20:04.30tumypwhat about two wire E&M ?
20:04.43tumypdo I still need to have digital card to support it ?
20:04.52[TK]D-Fendertumyp : Ok, well look here : http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf for the "busycount", and "callprogress" options.  This will help for what you want, but you risk having calls randomly disconnected.
20:05.19tumypI've tried them, does not work for me, sorry
20:05.42tumypeven answer on polarity switch
20:05.43[TK]D-Fendertumyp : Right now that ADTRAN has no way to be used.
20:05.52*** join/#asterisk s1gny|wrk (n=s1gny@p54915A95.dip.t-dialin.net)
20:06.17tumyp[TK]D-Fender, it's pity :(
20:06.29*** part/#asterisk s1gny (n=s1gny@p54915A95.dip.t-dialin.net)
20:07.51tumyp[TK]D-Fender, I wonder, if it so hard to implement support of analog signalling type like E&M in FXS card ?
20:08.11vooduhalWhen you Hangup() in a macro,  will the macro still return to the calling context?
20:09.18tumyp[TK]D-Fender, thanks a lot, bye
20:12.08*** join/#asterisk Moobius (i=Moobius@159.250.16.17)
20:12.08kallixI can't get res_crypto to get compiled on svn, it silently fails but * runs just fine, without rsa auth though
20:12.27Qwell[]kallix: install the ssl/crypto dev packages
20:12.34kallixopenssl and libssl-dev ARE installed (debian testing)
20:12.43Qwell[]libcrypto-dev?
20:12.59Qwell[]what dep is make menuselect complaining about?
20:13.27[TK]D-Fendervooduhal : No.  Hangup ends everything immediately, no continuing anywhere.
20:13.36vooduhalGood to hear.  Thank you.
20:13.46MoobiusI've moved from 1.2 on Fedora Core 5 to 1.4 on FC6 and now Curl isn't working for me. I've tried to use the Curl function instead of the command but I get "pbx.c:1497 ast_func_read: Function CURL not registered". Any help?
20:13.53kallixQwell[]: thx, giving it a try
20:14.07Qwell[]kallix: and you have to re-run ./configure afterwards
20:14.10[TK]D-Fendervooduhal : It is an implicit "I told you to STOP" function.  enything you wanted to do, you should have done prior :)
20:14.20[TK]D-FenderAnything*
20:14.48[TK]D-FenderMoobius : That command was deprecated, check the changelogs.
20:15.11Corydon-wMoobius: curl is now found with the configure script, and it's not finding the curl-config binary
20:15.33Corydon-wMoobius: so rerun ./configure, specifying the path for curl
20:15.33Moobiusdo I have to specify a path to ./configure?
20:15.43MoobiusCorydon-w: thanks!
20:16.33rudholmMoobius: "./configure" is a specific path
20:17.02Moobiusrudholm: yes. sorry. just using sloppy shorthand.
20:17.24rudholmah
20:17.31*** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net)
20:17.58rudholmMoobius: the answer is yes
20:18.06MoobiusSo it'll look like: /usr/src/asterisk*/configure --with-curl=/usr/bin/curl-config
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20:24.22*** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk)
20:27.20*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)
20:27.51*** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram)
20:27.51*** mode/#asterisk [+o kram] by ChanServ
20:28.43*** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com)
20:29.35anonymouz666who is using a nokia e61 in here?
20:30.45nibbler_dee70 here
20:30.53nibbler_dehow may i help you? :)
20:31.10J4k3nokia?  snow tires? :)
20:31.24*** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com)
20:31.55anonymouz666nibbler_de: wireless stuff... I got associated and after few seconds dissassociated
20:32.11anonymouz666so... e61 does not send SIP register to *
20:32.31*** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com)
20:32.38anonymouz666the strange is: when i am surfing... I got associated all the time....
20:32.41anonymouz666using web
20:33.34anonymouz666don't know what to do.
20:34.04nibbler_dehm, that's strange
20:34.11Assiderr.. anyone got anything for a p990 sony ericsson?
20:34.12nibbler_dewhat does a tcpdump show you?
20:34.24nibbler_deAssid: yup. pitty.
20:35.31anonymouz666tcpdump ?
20:35.38Assidtheres gotta be some sip client for the 990 right?
20:36.35SomeOne1what does ACL or ASR mean in voip terminology?
20:36.57nibbler_deanonymouz666: yup. tcpdump - on your wireless router or on the asterisk.
20:37.10anonymouz666I got disconnect from wireless
20:37.24anonymouz666how the packet can reach the *?
20:37.46shido6http://en.wikipedia.org/wiki/Answer_Seizure_Ratio
20:38.50nibbler_deanonymouz666: are you sure that your phone gets disconnected before it can send any packet?
20:38.54*** join/#asterisk Lurchtoke (i=Lurchtok@adsl-75-37-76-81.dsl.frs2ca.sbcglobal.net)
20:39.00Lurchtokehello peeps
20:39.06anonymouz666yes
20:39.08Lurchtokea question
20:39.21SomeOne1what does ACL or ASR mean in voip terminology?
20:39.34Qwell[]SomeOne1: shido just told you what ASR was
20:39.44SomeOne1sorry
20:39.49SomeOne1what about ACL then?
20:39.52SomeOne1access control list?
20:39.53Qwell[]~acl
20:39.54jbotColorizes log files using advanced parsing capabilities.. URL: http://www.LinuxRC.org/projects/acl/
20:40.01Qwell[]umm
20:40.06*** join/#asterisk kolton (n=mozveren@13.27.103-84.rev.gaoland.net)
20:40.10Qwell[]~asr
20:40.11jboti heard asr is alt.sysadmin.recovery, the newsfroup for recovering system administrators, or the scary devil monastery
20:40.14koltonhello
20:40.25SomeOne1"please email us the offer in USD 1/1, capacity available and the approximate ASRs and ACLs that we should see on retail.
20:40.26SomeOne1"
20:40.33SomeOne1what does he mean by 1/1
20:42.00Lurchtokewhat file lists my current mailbox configuration....and how do i change the sound file associated with a particular option in my phone menu
20:42.38[TK]D-FenderLurchtoke : voicemail.conf, and what do you mean "phone menu"?
20:42.41Lurchtokeive read the help file a little but havent had the time to learn the ins and outs of how to do that
20:42.51Lurchtokethx fender
20:42.53Lurchtokebrb
20:43.09Jinglesif you're talking about the 'phone tree', then you do that by modifying your dialplan in extensions.conf
20:43.16Lurchtokeok
20:43.26Lurchtokeone sec..lemme grab those and read it
20:43.28Jinglespress one to hear me say one again.
20:43.31Lurchtokebrb
20:43.32Jinglesthat sort of thing.
20:47.41koltonhello
20:49.51syzygyBSDhi kolton
20:49.58koltonhi
20:51.58koltonI have questoin about asterisk installations
20:52.07syzygyBSDand the question would be....
20:52.22koltonIs asterisk a success in USA ?
20:52.34koltonhow many installations have been done ?
20:52.37koltonany figures ?
20:52.59syzygyBSDwell, it is an open source one so their isn't anything like registration to quote numbers from
20:53.23koltonbut it used a lot ?
20:53.24syzygyBSDjust off the top of my head I would say 1.8 Trillion +- 100%
20:53.27koltonor not ?
20:53.29Bobthehunternable to determine channel for data w1g1/
20:53.30Nuggetwho cares?  it works for me.
20:53.32Bobthehunterhmm
20:53.49[TK]D-Fenderkolton : Its used by untold thousands and more.
20:53.56syzygyBSDI use it a lot, I don't really care what other people choose
20:54.09koltonbecause i live in france
20:54.14[TK]D-Fenderkolton : Perhaps you should be more specific as to your goals & expectations.
20:54.37koltonand want to start a business around asterisk services
20:54.39syzygyBSDya, does it matter how many people use it if it does what you want it to?
20:55.00[TK]D-Fenderkolton : If you're ready to form a business around it, should you know all this stuff already?
20:55.02*** join/#asterisk litage (n=nick@203.220.55.70)
20:55.05syzygyBSDselling phone service or supporting asterisk?
20:55.15koltonyeah
20:55.17Bobthehunterhow i send a call to my zap ?
20:55.23koltonselling , installation, services
20:55.23Bobthehunterits interface w1g1
20:55.27*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
20:55.27*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
20:55.45syzygyBSDBobthehunter: it doesn't matter what the interface is if it is configured
20:55.51syzygyBSDonce *
20:55.59Bobthehunterok
20:56.04Bobthehuntershow zap channels show ok
20:56.08Bobthehunter1 to 23
20:56.17Bobthehunterits a sangoma 102d.. pri
20:56.22syzygyBSDso Dial(Zap/1/18008675309)
20:56.28koltondo you think "asterisk consultant" can be a success in the next years ?
20:56.43anonymouz666nibbler_de? you around?
20:56.46syzygyBSDkolton: it is successful now if you are good at it
20:56.50[TK]D-Fenderkolton : try adding "load = chan_psychic.so" to modules.conf....
20:56.54*** join/#asterisk zotz (n=zotz@24.244.163.157)
20:56.56Bobthehunterapp_dial.c:1133 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)
20:56.59*** join/#asterisk BitBandit (n=polx@24-241-129-72.static.stgr.ut.charter.com)
20:57.10syzygyBSDlol tk
20:57.21[TK]D-FenderBobthehunter : Did you recompile * after compiling wanpipe & zaptel?
20:57.35koltonnot in france today
20:57.39koltonand europe
20:58.07syzygyBSDI have talked with dozens of people from europe using it that need help, and would have been willing to pay
20:58.09koltonWill asterisk have the same success that apache ?
20:58.09Bobthehunteryeah
20:58.14Bobthehunteri did
20:58.25[TK]D-Fenderkolton : Please see my previous reference.
20:58.30koltonok
20:58.33Bobthehuntershoudnt it work since i see the zap channels and all ok ?
20:58.37Bobthehunterzap show status =ok
20:58.55koltondid you start ztcfg ?
20:59.02BitBanditanyone know if you can do this? say someone calls and they need to check the status of a item. can you have asterisk say "please enter your number" and have it do a check on a mysql DB and then return the contents of that query
20:59.19[TK]D-FenderBitBandit : Yes, this is very doable.
20:59.53BitBandit[TK]D-Fender : do you know of where in the docs or where i could learn how to do so ?
20:59.54Bobthehunteryes
20:59.59Bobthehunterall ok no alarms
21:00.02Bobthehuntercode green ;)
21:00.03MoobiusBitBandit: I use festival to have * read me things from my SQL server...
21:00.15Lurchtokehey fender
21:00.25syzygyBSDBobthehunter: you have your d channel setup right?
21:00.30Bobthehunteryes
21:00.35BobthehunterSPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
21:00.42BobthehunterChannel 24: D-channel (Default) (Slaves: 24) etc
21:00.43Lurchtokewhats the command to d/l the config file to my puter?  get "filename" c:/  etc?
21:00.47BobthehunterChannel 01: Clear channel (Default) (Slaves: 01)
21:00.48BitBanditMoobius : thanks
21:00.56BitBanditi am sure i will be back in here asking more haha
21:00.57Lurchtokein putty
21:01.18Bobthehuntermaybe just no service ?
21:01.31Bobthehunterlike .. lol is there a way to know if dialtone or something
21:01.37Bobthehunterirs digital pri so i guess not
21:01.52koltonWhen is planned the launch of asterisk box ?
21:02.17[TK]D-FenderBitBandit : ...
21:02.18[TK]D-Fender~book
21:02.19jbot[book] a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
21:02.25[TK]D-FenderBitBandit : Time to start learning AGI
21:02.42BobthehunterProvisioned, Down, Active
21:02.43BobthehunterAH
21:02.55BitBanditsweet, i like learning stuffs
21:03.18[TK]D-FenderBobthehunter : pastebin "pri show span 1"
21:03.25koltonbobthehunter : there is any device like /dev/zap/1 ... /dev/zap/24 ???
21:03.49BobthehunterConfiguring interfaces: w1g1
21:03.49[TK]D-FenderBobthehunter : Silly to ask, but you DID compile libpri first, right?
21:03.53BobthehunterOwning /dev/zap
21:03.53BobthehunterPAWNED /dev/zap
21:03.55Bobthehunter;)
21:04.01Bobthehunteryeah
21:04.21Bobthehunterdev zap has 1 to 24  + peusdo +ctl+ channel + timer
21:04.44BobthehunterChanging signalling on channel 24 from Unused to HDLC with FCS check
21:04.46nibbler_deanonymouz666: yup
21:04.51Bobthehunterthat last line of ztcfg -vvv
21:05.02[TK]D-FenderBobthehunter : So please pastebin all of the following "wanrouter status", "ztcfg -vvvv", "cat /proc/interrupts", and in * CLI "pri show span 1".  After pastebin your zaptel.conf, zapata.conf, and zanpipe1.conf
21:05.42*** join/#asterisk drako (n=ljd@unaffiliated/luisjose)
21:05.56anonymouz666nibbler_de: the 401 goes out asterisk interface... but my nokia doesn't seem to care about
21:06.06anonymouz666it does not reply
21:06.42anonymouz666nibbler_de: do you have any suggestion about SIP configuration?
21:06.44Bobthehunterhttp://pastebin.ca/329553
21:06.54Bobthehunternow for the .conf files
21:07.48Bobthehunterits wanpipe
21:08.16syzygyBSDwanpipe1 :)
21:08.20Bobthehunterhttp://pastebin.ca/329558
21:08.26Bobthehuntersangoma A102D
21:08.38Bobthehuntercard layer is ok
21:08.43Bobthehunteri talk to card etc all ok
21:08.46Bobthehunterno errors
21:09.22Bobthehunterhttp://pastebin.ca/329561
21:09.27Bobthehunterhere is card status
21:09.34syzygyBSDjust for shits and giggles, try dial(Zap/g1/18009993355)
21:09.56[TK]D-FendersyzygyBSD : His PRI is DONW.
21:10.12syzygyBSDhe said it was in a green state...
21:10.14[TK]D-FenderBobthehunter : Have you called the telco to make sure the line is actually supposed to be up, and if they see anything?
21:10.19koltonare you sure about T1 parameters ?
21:10.25Bobthehunteryes
21:10.29Bobthehunteri got confirmation
21:10.47Bobthehunterso it maybe just not activated yet
21:10.51koltonlaunch the asterisk console
21:11.00syzygyBSDthey could always not accept the right number of digits for a phone call, I hate when they do that
21:11.08koltonand launch pri debug
21:11.09[TK]D-FendersyzygyBSD : "Status: Provisioned, Down, Active" in his pastebin
21:11.56syzygyBSDmissed that
21:12.34[TK]D-FenderBobthehunter : that aside the rest of your config looks pretty much perfect.
21:12.41koltonYou can test by some samples in libpri
21:12.52koltonyou can compile pritest
21:13.07koltonIts print verbose information
21:14.14Bobthehunterok so then its waiting time
21:14.19koltonor just launch "cat /dev/zap/24"
21:14.22Bobthehunterdarn # i called for TNA is just crap
21:15.02koltonto see if any data comes from Dchannel
21:15.11Bobthehunterdeice busy
21:15.24koltonyes stop asterisk before
21:17.04BitBanditanother question. i have 4 numbers what card wouild i get to use them with * ?
21:17.12Bobthehunterhmm
21:17.28Qwell[]BitBandit: 4 numbers or 4 lines?
21:17.33Qwell[]completely different questions
21:18.17[TK]D-FenderQwell : I'll bet the are "8088" ;)
21:18.30Qwell[]huh?
21:18.41[TK]D-FenderQwell : Think ..... Intel :)
21:18.51*** join/#asterisk digix84 (n=digix@72-48-74-116.dyn.grandenetworks.net)
21:18.56Qwell[]4 lines on an 8088?
21:19.01[TK]D-FenderQwell : :D
21:19.07Qwell[]wtf are you talking about? :P
21:19.37BitBanditQwell: um both
21:19.47[TK]D-FenderQwell : I'm disappointed.. Intel 8088 AKA the original 8-Bit PC :)
21:19.49Qwell[]BitBandit: are they analog lines?
21:19.53BitBandityes
21:20.06Bobthehunterwell Zap/g1/18004633339") in new stack
21:20.06BobthehunterJan 26 16:19:38 NOTICE[3058404272]: app_dial.c:1133 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)
21:20.11[TK]D-FenderQwell : so what card would he need to run on one? :)
21:20.22Bobthehunterdial exec full .. full means it got a busy from all channels..
21:20.26Bobthehunterright ?
21:20.26[TK]D-FenderBobthehunter : the "down" issue on your span is pretty telling.
21:20.32Bobthehunteryeah
21:20.41Bobthehunterbut is it because of me ? or service just not turned on
21:20.42[TK]D-FenderBobthehunter : just means the D-Chan couldn't reserve a B-Chan.
21:21.04Qwell[]BitBandit: Digium sells a 4 port analog card, called a TDM400P (and an 8 port card was just announced as well)
21:21.08[TK]D-FenderBobthehunter : Like I said, your configs look fine.  Assuing your signalling is right, check with your telco
21:21.17digix84anyone here versed in dundi linking?
21:21.27BitBanditQwell[] sweet thanks man
21:22.29Bobthehunterthansk fender
21:23.01koltonI think also all seems ok
21:23.52Bobthehunterhmm
21:23.58Bobthehunteri mean how much is biz edition V B ?
21:24.03koltonQuestion about sip trunk in USA : anyone use a verizon trunk ?
21:24.42koltonor any other SIP trunk ?
21:25.51*** join/#asterisk mrempire (n=trefpunt@mrempire.demon.nl)
21:26.03koltonany use case of SIP trunk over T1 ?
21:26.05BitBanditwhats the differant between FXO and FXS ?
21:26.12Qwell[]~fxofxs
21:26.13jbotrumour has it, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this.  An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this.
21:26.33BitBanditsick
21:26.35BitBanditthanks
21:27.22rudholmQwell[]: do you happen to have a URL for the newly announced Digium 8-port card?
21:27.33Qwell[]I don't know that we have a URL for it yet
21:27.45Qwell[]but, lemme check
21:28.16Qwell[]afaik, it uses the same modules as a TDM2400p, but I don't know for certain
21:28.18*** join/#asterisk punkgode (n=punkgode@rev-200-40-119-222.netgate.com.uy)
21:28.33hadsI haven't seen it announced yet but I do have info from the distribution channel.
21:28.46Qwell[]it was announced like Wednesday
21:29.01punkgodehello, does anyone know if hints priorities are supported un AEL o AEL2?
21:29.03Qwell[]will start selling in Feb I think they said
21:29.06koltonbobthehunter: Just a advice about you sangoma configuration : Use hardware HDCL framing in place of software (zaptel)
21:29.19koltonYour installation will be more stable
21:29.46hadsAccording to my announcement it supports both the single port modules and the 4 port modules.
21:29.59hadsWhich seems odd
21:30.03Qwell[]hads: I was thinking so, but didn't want to say something that might've been untrue
21:30.28hadsLeave that to me :)
21:30.41Qwell[]rudholm: How's this?
21:30.47Qwell[]http://home.businesswire.com/portal/site/google/index.jsp?ndmViewId=news_view&newsId=20070124005220&newsLang=enhttp://home.businesswire.com/portal/site/google/index.jsp?ndmViewId=news_view&newsId=20070124005220&newsLang=en
21:30.59koltonbobthehunter: in wanpipe1.conf change "TDMV_DCHAN      = 0" to "TDMV_DCHAN      = 24"
21:31.34digix84Are there any common reasons why an extension lookup would not work over a DUNDi cloud, even though the peers are up?
21:31.58rudholmQwell[]: ah, thanks.
21:32.02*** join/#asterisk elgreco (n=mat@redirect.getntense.com)
21:32.21codefreezepunkgode: Yes, you do hints in AEL/2
21:32.23elgrecocan anyone help me with a hardware question?
21:32.35punkgodecodefreeze, just version 2?
21:32.38rudholmQwell[]: cool.  I need more FXS ports, so that might work for me.  what I really need, though, is a National ISDN BRI card.  Analog entrance facilities are lame.
21:32.44codefreezeYes.
21:32.52punkgodecodefreeze, ok, thanks
21:32.55*** part/#asterisk punkgode (n=punkgode@rev-200-40-119-222.netgate.com.uy)
21:33.48hadsQwell[]: HPEC is basically a non-free replacement for the EC in zaptel?
21:34.03Qwell[]hads: correct, however, as the article says...
21:34.18Qwell[]hads: if you've got hardware in warranty...  you can *have* licenses for it
21:34.18hadsYup, will be made available to...
21:34.30hadsYep, sweet.
21:36.07*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)
21:39.08BitBanditto make sure i under stand this right, each FXO module will handle 4 lines ?
21:39.38BitBanditor does that just depend on the module
21:40.39*** join/#asterisk af_ (n=af@ip-173-157.sn1.eutelia.it)
21:40.54*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
21:40.54*** mode/#asterisk [+o mog] by ChanServ
21:45.22[TK]D-Fendermercestes : Finally getting some... about time! ;)
21:45.41[TK]D-Fendermercestes : We were beginning to worry...
21:48.20elgrecois the 536ep useless for asterisk?
21:49.05syzygyBSD~mercestes
21:49.06jboti heard mercestes is a total nub
21:49.48[TK]D-Fender:O
21:50.13syzygyBSDI need to find something that intrests me that I can pick up as a hobby...
21:50.29syzygyBSDI was thinking about blowing shit up
21:50.40syzygyBSD"ballon artist" same thing..
21:50.53[TK]D-FendersyzygyBSD : Fun, but expensive risky, and generally frowned upon.
21:51.13*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
21:51.33*** join/#asterisk anthonyl (n=anthonyl@72.146.14.72)
21:51.34sweepermmmm
21:51.41[TK]D-FendersyzygyBSD : Go learn an instrument or something
21:51.50syzygyBSDya, more interested in figuring out a better linear motor or something
21:51.57sweeperI might get drafted to code TDMoIP support for asterisk \o\
21:52.35*** join/#asterisk CrashSys (n=kumba@208.177.233.66.ptr.us.xo.net)
21:52.43*** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc)
21:52.43syzygyBSDcan asterisk handle SMS messages?
21:52.50CrashSysWhat's the option to have voicemail delete the message after e-mailing it?
21:52.59syzygyBSDdelete=1
21:53.07[TK]D-Fendersweeper : how about aiming for something more useful like a REAL SIP stack?  Perhaps better PRI interfaces for 2BCT or something?
21:53.08CrashSyson the user's voicemail line?
21:53.18syzygyBSDya, but it needs to be in the right options area
21:54.59CrashSyssyzygy: so like 8000 => 1234,Blah,name@tld.com,delet=1 ?
21:55.08CrashSyserr delete=1
21:55.27*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
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21:56.24syzygyBSDno, more like 8000 => 1234,blah,email@this.com,smsemail@this.com,|delete=1
21:56.46CrashSysHmmm
21:57.04syzygyBSDlol
21:57.20syzygyBSD8000 => 1234,blah,email@this.com,,|delete=1 would also work
21:57.25CrashSysYeah
21:57.27CrashSyswas about to ask that
21:57.42CrashSyssyzy: quit reading my mind
21:57.51syzygyBSDjust make sure you have the right number of , and |
21:58.10syzygyBSDI don't really know why I have the | at all, but I know it wasn't working without it...
21:58.42CrashSysIf things go wrong i'll tell my boss it was because of some hacker on IRC! :)
21:58.48CrashSyscause right now it's beer-30!
21:58.57sweeper[TK]D-Fender: sorry, my company actually could USE TDMoIP support :)
21:59.00syzygyBSDya, still 1 hour till beer 30 for me
21:59.04syzygyBSDwell, beer hour
21:59.08sweeperwould allow direct interfacing with RAD mux's
21:59.32sweeperand those mux's are awsome CPE's
21:59.52*** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com)
22:00.02syzygyBSDdoesn't rad mean awsome? my 80s lingo is a bit off...
22:00.13CrashSyskawabunga!
22:00.13sweeperno moving parts, 8 FXS, very stable, call trunking
22:00.42syzygyBSDno moving parts?!?! must be electronic...
22:00.52syzygyBSDwait.. don't the electrons move?
22:00.54CrashSysI like hearing the clicks
22:01.03CrashSyslets me know that someone's getting some phone love
22:01.05syzygyBSDso wouldn't it be no mechanical parts?
22:01.06*** join/#asterisk CrashHD (n=crashhd@c-67-166-156-157.hsd1.ca.comcast.net)
22:01.55sweeperhttp://www.rad-direct.com/Product-Voice-Compression-Vmux-110.htm <-- we use lots of these
22:02.11sweeperbut terminating them is spendy
22:04.16syzygyBSDso build support into asterisk !
22:04.25sweeperthat's what I might end up doing
22:04.59syzygyBSDwait.. you said TDM?
22:05.51sweeperright now we go vmux110 -> satellite -> vmux2200 -> * -> ADTRAN Thingy
22:06.02syzygyBSDhttp://www.google.com/patents?vid=USPAT6731649&id=jjUSAAAAEBAJ&dq=TDMoIP
22:06.06syzygyBSDits patented..
22:07.12sweeperhmmm
22:07.31syzygyBSDsee if they will develop one and license it
22:08.03sweepermmmm
22:08.26syzygyBSDwouldn't be free, but neither is g729
22:08.39sweeperwait wait tho
22:08.46sweeperthere was already legal shit about this
22:08.50sweeperwell, similar
22:09.06sweeperwhen people were writing software that would decode other formats
22:09.58syzygyBSDthe fact you still have to buy a license for g729 suggests that it wouldn't fly
22:10.05sweeperyea D:
22:10.28*** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net)
22:10.28*** join/#asterisk Tako-san (n=Tako-san@24.108.162.254)
22:10.48sweeperthat's fucked up. whatever happened to reverse engineering? :/
22:11.10Nuggetis that anything like reverse polish notation?  ;)
22:11.16*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
22:11.16*** mode/#asterisk [+o mog] by ChanServ
22:11.38syzygyBSDhmm, it looks like only the system is patented, not the protocol
22:11.51sweeperisn't that the same thing? :v
22:12.21syzygyBSDna, it is an end to end solution, vs one piece that can communicate with said system
22:12.29*** join/#asterisk Tako-san (n=Tako-san@24.108.162.254)
22:13.01sweepermm
22:13.28sweeperwell, luckily, one of the benefits of being part of a corporation is that I can sic lawyers on it \o\
22:13.55syzygyBSDmeh, ask them if you can, but don't spend your time on it yet
22:14.00sweeperaaand, I seem to have fucked up the sshness of the server I just installed
22:14.21syzygyBSDthere should be enough information in the patent to build the system without much problem though
22:14.40sweepermmhmm
22:14.45syzygyBSDI would still just suggest seeing if they will license something
22:14.55syzygyBSDthen they support it too
22:15.03sweepermebbe
22:15.11syzygyBSDcorporations are willing to pay for support right?
22:15.51sweepersure, but owuld RAD like to lose the hefty profits they get from selling hardware? :D
22:16.01*** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net)
22:16.25syzygyBSDhow can I recieve SMS messages sent to my phone numbers?  do I have to register with some server?
22:16.46syzygyBSDsweeper: you can't sell it very well
22:17.08dlynes_laptopsyzygyBSD: use a WAP gateway such as Kannel
22:17.24syzygyBSDthe whole purpose of this is to get more equipment that works with RAD's excellent hardware
22:17.33dlynes_laptopsyzygyBSD: or you can use an SMS gateway that connects to sip
22:17.38sweeperyea, we'll see
22:17.47sweeperwe buy a lot from them, so it might happen
22:17.50syzygyBSDthere is no need for this unless you already have RAD's hardware
22:17.51dlynes_laptopsyzygyBSD: they require their own sim card, too
22:18.12sweeperor we could tell our orbit people in israel to go threaten the RAD people
22:18.21syzygyBSDdlynes_laptop: I was hoping on a lower level
22:18.57dlynes_laptopsyzygyBSD: well, you could write an agi script to use one of the popular text messaging servers
22:19.14dlynes_laptopsyzygyBSD: or your own dialplan application, for that matter
22:19.45syzygyBSDwell, how would say.. cingular know where to send the text messages going to my phone numbers?
22:20.01dlynes_laptopsyzygyBSD: you don't have to care about that...they do
22:20.13syzygyBSDum.. ?
22:20.17dlynes_laptopsyzygyBSD: when i send a text message to fido, i don't care where it goes...fido just sends it
22:20.28dlynes_laptopsyzygyBSD: and the person receives it
22:20.40dlynes_laptopsyzygyBSD: as long as the other person is in north america, it just works
22:20.52dlynes_laptopsyzygyBSD: i don't know if it works across international boundaries or not...never tried
22:21.06syzygyBSDlol, well the telephone network is a bunch of switches with a direct path and a known destination... I don't see how SMS would know unless I tell someone
22:21.18dlynes_laptopsyzygyBSD: but it certainly works between Canada and the US
22:21.46dlynes_laptopsyzygyBSD: they route it to appropriate network, based on the routing information for that number
22:22.11syzygyBSDso SMS is over phone lines?
22:22.16dlynes_laptopsyzygyBSD: no
22:22.33dlynes_laptopsyzygyBSD: it's all through data
22:22.44syzygyBSDdidn't think so... well how can I update the routing information for my numbers?
22:22.51dlynes_laptopsyzygyBSD: going through gsm/cdma/tdma gateways
22:23.25dlynes_laptopsyzygyBSD: i don't think you can..the cell phone companies would take care of all that
22:23.52dlynes_laptopsyzygyBSD: or are you running your own private gsm network?
22:24.40syzygyBSDno, I just want to say.. have an SMS sent by cingular to my office phone emailed to me
22:25.17dlynes_laptopoh...that's a different story, unless your cingular office phone's sim is plugged into a gsm gateway
22:25.32dlynes_laptopthen you could have it come over sip to your asterisk box
22:25.43dlynes_laptopand then perform the processing to email it to you
22:25.58syzygyBSDk, I have nothing to do with cingular, you can put in any company there
22:26.09*** part/#asterisk elgreco (n=mat@redirect.getntense.com)
22:26.21dlynes_laptoptake company a's sim card, and throw it into the gsm gateway then
22:26.31dlynes_laptopit'll route it over sip to asterisk
22:26.38syzygyBSDIE, SMS sent from anywhere -> goes to my asterisk box
22:26.40dlynes_laptopand then your script in asterisk will convert it into an email
22:26.56dlynes_laptopi.e. you need an sms gateway
22:27.00syzygyBSDk
22:27.08dlynes_laptopand you need one port for every sim card you want to throw in there
22:27.10robin_szsuprised telcos arent offering SMS to email forwarding on the networks
22:27.18Moobiusdoes anyone know if this has been fixed in the SVN?: http://bugs.digium.com/view.php?id=8744
22:27.23dlynes_laptoprobin_sz: they are
22:27.31dlynes_laptoprobin_sz: I get mine emailed to me
22:27.34Qwell[]Moobius: Does it say it's been fixed in svn?
22:27.34robin_sznice
22:27.41dlynes_laptoprobin_sz: well...not quite
22:27.50J4k3robin_sz: yeah, they just charge entirely too much for the service.
22:27.53dlynes_laptoprobin_sz: i get all my fido email sms'd to me
22:27.57robin_szwinkers
22:27.58dlynes_laptoprobin_sz: not the other way around :)
22:28.04robin_szahh
22:28.19robin_szjust get an old phone and a serial lead :)
22:28.24MoobiusQwell[]: it says "give it a try".
22:28.24J4k3somehow the cell companies think that the equiv of 1/30th of 1 seconds worth of phone conversation is as worth as ~2 minutes of prime airtime.
22:28.33robin_szyeah
22:28.35Qwell[]Moobius: did you "give it a try"?
22:28.46dlynes_laptopJ4k3: and it's old technology, to boot
22:28.49robin_szJ4k3, and worse, sms is not even sent on the real time streams
22:28.54dlynes_laptopJ4k3: it's all old pager technology :)
22:30.18dlynes_laptopHas anyone experienced issues on Aastra's with firmware 1.4.1 and Asterisk 1.2.14, when you don't use a registration server, and you use static ips that the voicemail count is not updating on the phones?
22:30.32dlynes_laptopand the mwi light doesn't flash?
22:32.06*** join/#asterisk bitbandit (n=polx@65-103-229-114.slkc.qwest.net)
22:32.35CrashHDyou setting mwi explicit?
22:32.48dlynes_laptopyes
22:32.55CrashHDhmm
22:33.11CrashHDour 480i's are behaving just fine
22:33.23dlynes_laptopHow do you have them configured?
22:33.31CrashHDupdated to the most recent svn on branch 1.4 last night
22:33.40CrashHDnot statically though
22:33.45dlynes_laptopah
22:33.52dlynes_laptopso you're still using the registration server
22:34.02CrashHDya
22:34.19dlynes_laptopI got rid of the registration server and changed everything to static
22:34.26dlynes_laptopTo see if it would make BLF more stable
22:34.42dlynes_laptopthe other way, using BLF was quite unstable
22:35.00russellbi'm having great fun playing with aastra blf right now.
22:35.04CrashHDWe are using BLF on call park extensions and it is extremely stable
22:35.13CrashHDhaven't setup blf on any actual extensions
22:35.20CrashHDyou setup hints for each extension?
22:35.29dlynes_laptoprussellb: buggy as hell for you, too?
22:35.56russellbdlynes_laptop: well, it's working fine for me.  It's my code that is buggy at the moment.  I'm rewriting SLA support ...
22:36.06CrashHDhow the shared line appearance working out?
22:36.07CrashHDhah
22:36.12CrashHDironic timing
22:36.14dlynes_laptoprussellb: ah...so it's the asterisk blf code that's quite buggy then?  not the aastra implementation?
22:36.16russellbheh, yeah
22:36.24russellbdlynes_laptop: no, this is something different
22:36.33russellbto emulate a key system
22:36.38dlynes_laptoprussellb: i just know that polycoms and aastras both get sip/500's, and aastra's additionally get a sip/489
22:36.43CrashHDrussellb hows progress on the sla coming?
22:36.58CrashHDI saw 1.4 had it but was reluctant to test it
22:37.04russellbCrashHD: it's coming well, i guess.  It's just very complicated ...
22:37.15russellbCrashHD: i would hold off until i release this re-write.
22:37.24CrashHDheh
22:37.37CrashHDcurrent sla implementation did not quite meet expectations? lol
22:37.42dlynes_laptopCrashHD: you're putting production systems on 1.4 branch?
22:37.43russellbnot quite :)
22:37.55CrashHDdlynes_laptop, only one currently
22:37.59CrashHDour personal system
22:38.06CrashHDour gateway system
22:38.11CrashHDand all other client systems are still 1.2
22:38.18dlynes_laptopCrashHD: ah...ok...so not crazy enough to put a customer's system on 1.4 yet :)
22:38.26dlynes_laptopCrashHD: much less 1.4 branch :)
22:38.38CrashHDso far I'm happy with the 1.4 branch though lol
22:38.45russellb1.4 branch would be better than 1.4.0 at this point
22:38.51CrashHDthat is what I figured
22:38.53russellba ton of stuff has been fixed
22:38.55dlynes_laptoprussellb: yeah, i would imagine
22:39.30CrashHDsaw some of the svn comments
22:39.34CrashHDhonestly it was just the sip jb that I wnated
22:39.34CrashHD*wanted
22:39.34russellbheh, cool
22:39.38CrashHDheh
22:39.38russellbyeah, that works pretty well
22:39.43CrashHDruss any way to see sip jb stats like iax use to?
22:39.50dlynes_laptopCrashHD: are you using blf on 1.4, or 1.2?
22:39.50russellbyeah
22:40.05russellbCrashHD: there is a jblog option.  It will dump stats to a file
22:40.11CrashHDdlynes_laptop: blf on 1.4 (but only on parking extensions with hint being used)
22:40.11russellbCrashHD: not currently in a CLI command, i don't thin
22:40.28dlynes_laptopCrashHD: ah...yeah...i was using it on monitoring extensions on 1.2
22:40.28CrashHDahh ok, that's the info I was looking for
22:40.41CrashHDsearched high and low yesterday in the cli for a jb stats command
22:40.47dlynes_laptopCrashHD: on that, maybe asterisk is buggy as well, maybe aastra is buggy as hell, or maybe both of them are buggy as hell
22:41.02dlynes_laptopCrashHD: but something is definitely buggy as all hell
22:41.06CrashHDlol
22:41.08Qwell[]asterisk?  bugs?  no such thing
22:41.13*** join/#asterisk malverian (n=malveria@gentoo/developer/malverian)
22:41.14CrashHDasterisk, no way!
22:41.24sevardbullcrap, asterisk is flawless
22:41.27dlynes_laptopQwell[]: well, on this one
22:41.31CrashHDheh
22:41.34dlynes_laptopQwell[]: i think they're both probably to blame
22:41.43dlynes_laptopQwell[]: and polycom for that matter
22:41.44CrashHD*puts his shit wading boots on*
22:41.52russellbor maybe ... the PEBKAC
22:41.58dlynes_laptopPEBKAC?
22:42.03russellb~pebkac
22:42.03jbotSounds like the Problem Exists Between Keyboard And Chair
22:42.07CrashHDlol
22:42.11dlynes_laptopnah
22:42.23dlynes_laptopSIP/500's happen because of something unexpected
22:42.44CrashHDyour getting sip/500's?
22:42.44[TK]D-Fenderdlynes_laptop : Wow.. thats like almost what the verbaim message reads!
22:42.47Aurslots of 500 from polycoms here
22:43.08[TK]D-Fenderdlynes_laptop : And *'s SIP stack couldn't POSSIBLY be in need of some adjustment! ;)
22:43.18CrashHDpinnaple!
22:43.19Aursif I use buddy list on polycom - restart asterisk - polycom starts responding with 500 on subscribe
22:43.32CrashHDif I could spell worth a damn
22:43.36russellbah yes, that is a bug in the polycom firmware
22:43.40Aursreboot polycom - 500 messages gone
22:43.42russellband they have been notified about it ...
22:43.45danpAurs: yeah, they'll do that
22:43.49Aursso I would call it a polycom issue, yes
22:44.15Aursdisable/enable buddy watch also fixes the problem
22:44.36danpif you make and save any kind of change to the directory on the phone it rights itself too
22:44.49dlynes_laptopwell, 500's are from something unexpected, so obviously asterisk is sending something it's not expecting, but at the same time, it should be expecting anything, and respond appropriately
22:44.49Aursrussellb: you have sent a NOTIFY to polycom, and they respond with "500 internal server error" ? ;)
22:44.54CrashHDhah
22:45.00dlynes_laptopbut instead...it sends SIP/500 and starts flipping out
22:45.10CrashHDmore like "666 die for pointing out our bugs"
22:45.20Aursdlynes_laptop: asterisk don't do anything different after a restart
22:45.35dlynes_laptopAurs: nah...we've only experienced this issue for BLF
22:45.35russellbAurs: lol, something like that.  :)  No, they have responded and are supposed to have it fixed here soon (or maybe already in the latest firmware, i don't remember).
22:45.45dlynes_laptopAurs: Aastra?
22:45.46dlynes_laptoperm
22:45.50russellbour polycom phones here at digium were doing it, too, so we talked to them about it.
22:45.52dlynes_laptoprussellb: Aastra?  or polycom?
22:45.56russellbpolycom
22:46.06*** join/#asterisk ccvp (n=axe@user-24-236-120-166.knology.net)
22:46.08*** part/#asterisk ccvp (n=axe@user-24-236-120-166.knology.net)
22:46.13russellbthough i thought we put code in to work around their problem ...
22:46.14Aursrussellb: good.. I think I have the latest fw. 2-something
22:46.44dlynes_laptoprussellb: weird...i thought polycom would've done the same thing as aastra, 'Nobody's reported that problem you're experiencing...it must be a problem on your end.  *click*'
22:46.57CrashHDhah
22:46.59russellbdlynes_laptop: we have a good relationship with them :)
22:46.59Aurshehe
22:47.11CrashHDactually dlynes_laptop I have a guy at aastra that is great for support
22:47.13CrashHDhis name is layne
22:47.24dlynes_laptopI've talked to layne
22:47.28CrashHDgreat guy
22:47.31CrashHDgreat follow up
22:47.33dlynes_laptopHe's as big of an idiot as the rest of them :)
22:47.39CrashHDlol
22:47.55CrashHDhe has the follow up I appreciate from support though
22:48.05dlynes_laptopUnless you can provide them 3 or 4 days worth of syslog and tcpdump, it must be your problem
22:48.11dlynes_laptopYeah...he did follow up
22:48.20CrashHDdoesn't matter what company I call, ultimately it ends up me having to figure out the problem, they just become an inside resource
22:48.20dlynes_laptopbut what he's asking for simply isn't practical
22:48.30CrashHDhah
22:48.36CrashHDone instance he did the same to me
22:48.42CrashHDand I just had him close the ticket
22:48.49*** part/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell)
22:48.49CrashHDso I could figure out the issue myself
22:48.57*** join/#asterisk Ritalin2 (n=dave@74-34-103-241.dsl1.pwll.tn.frontiernet.net)
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22:49.08CrashHDasked for tcpdumps for calls and a bunch of other stuff
22:49.12CrashHDI was just too lazy to provide
22:49.13dlynes_laptopCrashHD: well, and their syslog is so spammy, i don't even find it useful
22:49.16*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
22:49.34CrashHDbut it's better than them not wanting or offering to fix the problem
22:49.38dlynes_laptopCrashHD: you'd think you could adjust the spamminess of it
22:49.43CrashHDhah
22:49.45CrashHDyou would hope
22:50.05CrashHDthey have an engineer dumping data to the syslog
22:50.10CrashHDneeds some csr refinement
22:50.10dlynes_laptopCrashHD: but otoh, they've added a few new options to their 1.4.1 firmware that will allow you to detect if the phone is offhook before trying to send a call to it
22:50.28CrashHDwhat benifit would that have?
22:50.35dlynes_laptopCrashHD: because right now
22:50.49dlynes_laptopCrashHD: if a phone is offhook, i don't know about it, and i still try to send a call to it
22:51.15dlynes_laptopCrashHD: and so if it's an incoming call, they'll get patched to that call right away
22:51.20CrashHDin a normal office scenario?
22:51.23dlynes_laptopCrashHD: even if i'm ringing all phones
22:51.24robin_szI have an Aastra somwhere, but ive lost it
22:51.24CrashHDor acd scenario?
22:51.37dlynes_laptopCrashHD: yes...normal office scenario
22:51.47robin_szits on the network .. i can see it, ut I can;t actually find it :(
22:51.51dlynes_laptopCrashHD: typical keysystem scenario russellb was talking about
22:52.33Aursrobin_sz: traceroute it :P
22:52.45*** part/#asterisk max_______ (i=max__@ts.bestserversllc.net)
22:52.53CrashHDhmm
22:52.55hardwireanybody here using non telco approved DSL hardware on their DSL unknown to the telco?
22:53.09Ritalin2anyone a zapata.conf guru? :)
22:53.12CrashHDI don't see how it would affect anything even if you sent a ringall to the phone
22:53.32dlynes_laptophardwire: apparently with the new ADSL2+, it's all autodetect
22:53.41hardwirethats what I am using
22:53.45hardwireI wanna use the sangoma S518
22:53.51dlynes_laptophardwire: there's somebody in here using the new sangoma s518's
22:53.54hardwiresee my Paradyne modem blocks vlan requests
22:53.58hardwireerr vlans period
22:54.02hardwireand is locked to vlan 900
22:54.10dlynes_laptopthat's moronic
22:54.11hardwirevlan 900 = PPPoE network
22:54.20hardwirevlan 987 = Multicast IGMP Television service
22:54.31dlynes_laptopanyways...the fellow in here doing it, is using PPPoE
22:54.45[TK]D-Fenderhardwire : Where are you located?
22:54.46dlynes_laptopbut you might still have a talk with him
22:54.51hardwire[TK]D-Fender: Alaska
22:54.52dlynes_laptopit's [hC]
22:55.10[TK]D-Fenderhardwire : Well the S518 works jsut great for me, and many other users of it here as well.
22:55.16hardwire[TK]D-Fender: its MTA Solutions that provides this service, they are like one of 5 people using Allied Telesyn 7600 series DSLAMs
22:55.20dlynes_laptopah...didn't know you were using it, too
22:55.25Ritalin2hardwire: what's the weather there like right now?
22:55.27*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
22:55.27*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
22:55.33hardwire[TK]D-Fender: and you are on an ADSL2+ network?
22:55.40robin_szAurs: I know its somwehre hanging off a router in a particualr factory ... but can't find exactly where
22:55.40[TK]D-Fenderhardwire : If you can use a typical DSL modem with it, then I'm sure the S518 will owrk jsut fine.
22:55.45hardwireRitalin2: you ever been in the eye of a huge storm?
22:55.48[TK]D-Fenderhardwire : No, not ADSL2
22:55.48hardwireits super clear
22:55.52hardwirewe are about to die I think
22:56.09hardwire[TK]D-Fender: how did you configure the PCI card for your Telcos network?
22:56.11robin_szhardwire, if its of any reassurance, yes, you probably are
22:56.27hardwirerobin_sz: aww
22:56.35robin_szstill, at least you spent your last few momnents talkign on IRC :)
22:56.53Ritalin2hardware: no.  i've always wanted to visit alaska
22:57.06hardwireRitalin2: whats stopping you, other than Canada of course.
22:57.14hardwire</drumbeat>
22:57.33Ritalin2hardwire: I'd bypass canada.  ;)  just the ole time/money thing
22:57.43hardwiredon't be a puss, move up
22:57.45[TK]D-Fenderhardwire : It automatically adjusted the VPI/whatever other parameters that were needed, and"jus worked"
22:57.54[TK]D-Fenderhardwire : When in doubt, call Sangoma.
22:57.56hardwire[TK]D-Fender: freaky
22:58.01hardwire[TK]D-Fender: I may
22:58.03hardwireits an expensive card
22:58.05Ritalin2hardwire: I'm affraid of commitment.  I'd like to try the cow's milk before buying the cow if you know what i mean
22:58.11robin_szRitalin2, too much time and money?
22:58.12CrashHDyou guys talking about getting a dsl pci card working instead of a dsl modem?
22:58.21CrashHDwhat would be the benifit to that?
22:58.24hardwireRitalin2: well I ain't gonna fly you up here if thats what you are getting at
22:58.34Ritalin2robin: exactly.  not sure where to spend it all
22:58.41robin_szsendit to me
22:58.43[TK]D-Fenderhardwire : Not really... $135USD is not so bad.  asaves on wiring, and gives you better bandwidth control since its a raw interface no being pumped through ethernet.
22:58.57hardwireyou PPPoE as well?
22:58.59[TK]D-FenderCrashHD : Correct
22:59.13[TK]D-Fenderhardwire : Yes, though the card can work in pretty much every standard mode.
22:59.29[TK]D-Fenderhardwire : You use RP_PPPOE jsut like normal.
22:59.40dlynes_laptopCrashHD: no wall wart for one
22:59.42hardwire[TK]D-Fender: on this network they planned on using Allied Telesyn RG's
22:59.51hardwirethe RG can't have PPPoE not on a VLAN
22:59.54dlynes_laptopCrashHD: don't have to worry about the dsl modem screwing up constantly
22:59.57hardwireif you also want to use other VLAN's
23:00.00hardwireits amazingly odd
23:00.10hardwireand they needed to use VLAN for their multicast service
23:00.12[TK]D-Fenderhardwire : It willwork as a raw ATM interface w/o PPPoE.
23:00.13dlynes_laptopCrashHD: one problem we have here, is that the telco's modem needs to be rebooted, periodically
23:00.19[TK]D-Fenderhardwire : CALL :)
23:00.22hardwire[TK]D-Fender: that might not help me much :)
23:00.28hardwire[TK]D-Fender: these aren't sangoma questsions
23:00.35CrashHDohh
23:00.41CrashHDI have never had a problem with a dsl modem
23:00.47hardwiremore like check out how funky this DSL system is
23:01.05dlynes_laptopCrashHD: and also, there's no network cable in question anymore if the dsl card is in the phone system
23:01.06[TK]D-FenderCrashHD : I got mine because I wanted to cut down on the wiring clutter, and have better bandwidth control.
23:01.21CrashHDI could see the bandwidth control
23:01.21hardwireits neat, used smcroute to issue IGMP join requests to a multicast address
23:01.22dlynes_laptopCrashHD: you can be connected directly, and then you can have finer control over the firewall
23:01.24hardwirenever done that before
23:01.25hardwirevery nice
23:01.31dlynes_laptopCrashHD: many reasons to use the card over the modem
23:01.44CrashHDsounds like it
23:01.56CrashHDpvc support I'm guessing?
23:02.14CrashHDwe've been exploring peering with a dsl provider
23:02.20hardwire[TK]D-Fender: so it came up as an 802.3 net device?
23:02.23CrashHDto provide dsl lines to clients
23:02.29CrashHDthat directly connected back to our network
23:02.36CrashHDas a private line sub.
23:03.36[TK]D-Fenderhardwire : something like that.... I don't actually know all the nitty-gritty..
23:03.44hardwireI could always ask Sangoma
23:03.47hardwireright
23:03.49hardwirenudge nudge
23:03.51hardwireheh
23:04.04hardwireAugh, I payed to go to a party tonight and my g/f is sick
23:04.06hardwirebleh.
23:04.41dlynes_laptophardwire: you could always ask sangoma
23:04.47dlynes_laptophardwire: *hint* *hint*
23:05.06dlynes_laptophardwire: or check the specs on their webpage, too
23:06.11hardwirenever!
23:06.33dlynes_laptopomg!
23:06.55hardwireomg yourself!
23:07.05dlynes_laptopRehan Allah Wala is still on asterisk-biz?
23:07.21dlynes_laptopI thought he would have gotten tired of everyone making fun of him by now
23:07.59CrashHDasterisk-biz?
23:08.01CrashHDwhat's that?
23:08.03dlynes_laptopthe mailing list
23:08.23dlynes_laptopfor all business related conversations
23:08.26CrashHDahh
23:08.32CrashHDI stopped checking the mailing lists
23:08.42dlynes_laptopi.e. for finding terminators, originators, ...
23:08.52CrashHDnever was fond of mailing lists
23:08.56CrashHDseemed a hassle to use
23:09.06dlynes_laptophow is it a hassle?
23:09.10dlynes_laptopjust subscribe
23:09.13dlynes_laptopand lean back
23:09.19dlynes_laptopand watch it all flood in
23:09.21CrashHDI dunno, just organizing the incoming content
23:09.30CrashHDso it's understandable
23:09.40dlynes_laptopautofilter it by year and subautofilter it by month
23:09.51CrashHDGuess I'm more of forum guy
23:09.54hadsmmm... sieve
23:10.02dlynes_laptophads: procmail :)
23:10.12hadssieve!
23:10.16hads:)
23:10.24dlynes_laptopsieve is more efficient than procmail?
23:10.29dlynes_laptopand easier to define rules for?
23:10.50hadsNo idea about the efficiency
23:11.02Zorixis there any special things that need to be done for the "make" to work in freebsd 6.2
23:11.05dlynes_laptopbut easier to define rules for?
23:11.27dlynes_laptopI'd love to find a mail filter that's got a nice web page interface for defining filter rules
23:11.32hadsThe rules are pretty easy yeah, let me find an example
23:11.40dlynes_laptopthen i don't have to teach my boss how to the linux command line
23:11.47dlynes_laptops/to the/to use the/
23:11.53CrashHDa nice reliable newsgroup front end for the mailing list is nice
23:11.57CrashHDI remember finding one at one point
23:12.07hadsWell, seive does have managesieve depending on what implementation you use.
23:12.20hadsThere are web interfaces around for it.
23:12.24dlynes_laptophads: is it compatible with maildir?
23:12.29hadsYup
23:12.39dlynes_laptopok, cause that's a must for me
23:12.48dlynes_laptopI use dovecot and drop everything into maildir format
23:12.54hadsI use it with dovecot which doesn't have a mangesieve at the moment.
23:13.06hadsMe too then :)
23:13.09dlynes_laptopDoes it allow you to do post filtering?
23:13.25dlynes_laptopi.e. filter stuff that's already in your inbox?
23:13.45dlynes_laptopnot a must for me
23:13.47dlynes_laptopbut it would be nice
23:13.57dlynes_laptopor that's already in another directory
23:14.16hadsNa, it acts on LDA, you may be able to but I don't know.
23:14.34dlynes_laptopok, so you can throw it into your .forward file?
23:14.49dlynes_laptopand have it process spamassassin as well?
23:15.11hadsif header :contains "Subject" "[asterisk-users]" { fileinto "lists.asterisk"; }
23:15.26dlynes_laptopoh
23:15.35dlynes_laptopSounds like procmail might actually be better then
23:15.40dlynes_laptopprocmail allows me to do regex
23:15.53hadsOh that's pretty basic, there are lots of options etc.
23:16.47dlynes_laptopWell, i'll take a look at it anyways
23:17.01dlynes_laptopbut sounds like i might just stay with procmail
23:17.09hadsYeah, whatever works :)
23:17.12dlynes_laptopI'm just going to have to get around to writing a front end for procmail
23:17.20dlynes_laptopSo that my boss can just use a webpage to set up his filters
23:17.21hadshttp://www.fastmail.fm/docs/sieve/index.html
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23:18.06hadsI like sieve as there's work going on for integrating front ends for it - KMail for example
23:18.30dlynes_laptophads: yeah...I don't really care much about stuff like that
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23:18.39dlynes_laptopI don't run X on the mail server
23:19.05hadsNeither, I'm talking about remote
23:19.32hadsmanagesieve lets you upload rules to the server from different clients
23:21.47*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
23:21.47*** topic/#asterisk is Asterisk: The Open Source PBX -=- Check out the new http://www.asterisk.org !!! -=- Asterisk, Asterisk-addons, Zaptel, and Libpri 1.4.0 released!!! (December 23, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework
23:23.34dlynes_laptop~lart jbot
23:23.44dlynes_laptophahaha
23:24.15mercesteslol
23:25.22dlynes_laptopi guess he doesn't like being larted :)
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23:26.11stubertcan you use "file convert" to convert to g726?
23:26.28*** part/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca)
23:28.57CrashHDdlynes_laptop, what codec are you using with your aastra's?
23:29.10CrashHDI wish they had gsm support
23:29.22hardwireanybody gotten asterisknow to boot in a qemu?
23:29.25CrashHDand I wish I had a better solution for g729
23:29.30CrashHDthan the digium stuff
23:29.55hardwireits stalling right after finding the IDE bus
23:31.04dlynes_laptopCrashHD: the only ones they have
23:31.10dlynes_laptopCrashHD: ulaw, alaw and g729
23:31.28CrashHDI wish I could have a bank of g729 licenses
23:31.32CrashHDfor use across multiple servers
23:31.37CrashHDlike windows ts does
23:31.42dlynes_laptopts?
23:31.46CrashHDterminal services
23:31.50dlynes_laptopoh
23:31.52CrashHDjust that licensing model
23:32.02CrashHDsingle license daemon (or dual)
23:32.03dlynes_laptophardwire: you running qemu in windows, linux, or solaris?
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23:32.12CrashHDbut where multiple servers can utilize the licensing
23:32.21russellbCrashHD: that's an interesting idea ...
23:32.37dlynes_laptopyeah...that would be nice
23:32.39CrashHDwe run multiple installations on a single server
23:32.42CrashHDof asterisk
23:32.54dlynes_laptopor even multiple installations on many different servers
23:33.02dlynes_laptopthat way we can buy a bunch of licenses
23:33.09dlynes_laptopand share them between our customers
23:33.18dlynes_laptopso if one isn't using any
23:33.18CrashHDbasically at any time we may only have 50 g729 channels open
23:33.25CrashHDbut have to have 100 installed licenses
23:33.27dlynes_laptopthat's more that the other customers can use
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23:34.09CrashHDdlynes_laptop: I did not understand what you meant by: "or even multiple installations on many different servers"
23:34.12CrashHDelaborate?
23:34.28dlynes_laptopCrashHD: say we have several customers all running asterisk keysystems
23:34.42dlynes_laptopCrashHD: they all use g729 to connect to our main asterisk softswitch
23:35.08dlynes_laptopCrashHD: share the g729 licenses amongst all those keysystems...i.e. have them access a g729 license pool on the main softswitch
23:35.21CrashHDok, we are on the same page
23:35.45Bobthehunteranyreason why ZAP disconects after 21 seconds any calls ?
23:35.59CrashHDso basically just licensing X number of g729 channels across an infastructure
23:36.10CrashHDwhich is what I would like as well
23:36.21CrashHDalthough
23:36.25Bobthehunterchan_sip.c:1421 retrans_pkt: Hanging up call a2d06599-2838-122a-c991-000039c832f1 - no reply to our critical packet.
23:36.29Bobthehunteri get this then hangup
23:36.35CrashHDthe keysystem should stilll be able to locally terminate the g729 call if need be
23:36.44dlynes_laptopBobthehunter: what does sip have to do with zap?
23:36.57Bobthehunterim calling via sip to zap
23:37.04Bobthehunterand its crashing ever 21 seconds
23:37.09Bobthehunterdropping
23:37.15dlynes_laptopBobthehunter: but it looks like your sip client that's hanging up, not the zap end
23:37.52Bobthehunterhmm
23:38.22dlynes_laptopBobthehunter: but make sure busydetect is set to no, and callprogress is also set to no in your zapata.conf file
23:38.29dlynes_laptopBobthehunter: those can cause randomly disconnected calls
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23:39.26*** part/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net)
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23:39.56Bobthehunterall Ref: len= 2 (reference 8/0x8) (Originator)
23:40.08Bobthehunterah callprogress
23:40.49dlynes_laptopBobthehunter: make sure you do a stop and then a start after setting that
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23:54.33viperdudeukhi
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23:55.30yassinei can not hear the caller nor the caller can hear me :s any ideas please ?
23:56.06*** join/#asterisk docelm0 (n=vircuser@m1a5e36d0.tmodns.net)
23:57.08docelm0[TK]D-Fender you available?
23:58.24[TK]D-Fenderdocelm0 : 'sup?
23:58.42docelm0got a question.. Got any experience with asterisk and hung channels?
23:58.57docelm0Im getting alot of them where they are not dropping when the call is terminated
23:59.29[TK]D-Fenderdocelm0 : Not really....
23:59.50docelm0any idea what could possibly cause that?
23:59.57docelm0Cause I have racked my brain

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