irclog2html for #asterisk on 20050928

00:00.22justinuwhy oh why can I not get the PBX to send me any RTP when I make sip calls?
00:00.22hardwirehttp://pastebin.ca/24075
00:00.23PeggerFuriousGeorge ok i will owrk on it
00:00.24hardwirethere
00:00.32justinuever since upgrade to 1.2, it's broken
00:00.42hardwiregood quick and dirty way of doing IVR for real quick autoattendants and different DID's
00:00.44justinuiax calls work
00:01.00FuriousGeorgejustinu: what ports are you forewarding how much nat is going on
00:01.23justinua fair bit of nat, but it doesn't seem to stop it from working if I route the inbound calls into an IVR
00:01.43justinubut if I try to route the inbound call into a Dial(SIP/exten), no rtp at all
00:01.52FuriousGeorgeIVRs are outside of my scope, what ports are you forwarding
00:02.07FuriousGeorgeare both the client and server behind nat
00:02.20justinuyes
00:02.35justinui'm not explicitly forwarding any udp ports for media
00:03.49FuriousGeorgejustinu: you need to set externip and localnet in sip.conf
00:03.55justinudone already
00:04.12FuriousGeorgeyou also need to foreward (iirc) 10000-20000 to server and client
00:04.16FuriousGeorgeudp
00:04.23FuriousGeorgeand 5060 tcp and udp
00:04.51justinui'll try it... wondering why it worked on the older pbx version without any changes tho
00:06.07FuriousGeorgejustinu: also 8000 udp and maybe 5061 udp and tcp not sure about that last one.  basically there is no elegant way to do it for many clients w/o 3rd part software or taking * out from behind nat
00:06.43*** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net)
00:07.04JerJerFuriousGeorge: do you even know what you are saying?
00:07.44hardwireweird
00:07.47FuriousGeorgeJerJer: yeah, i actually got itworking, the prot numbers are straight from voip-info, and someone here told me about externip and localnet, and it didnt work till i did it
00:07.48shido6insert filter
00:07.50hardwiremy snom phones are recieving hints
00:07.54hardwirebut only once they boot
00:08.04hardwireso I see all the extensions in use on my network
00:08.07hardwireonly on the bootup
00:08.11hardwirethe LED's never change
00:08.15JerJerFuriousGeorge: and you followed this info?
00:08.31JerJerfirst off asterisk only ever deals with UDP on sip, so port forwarding tcp is useless
00:08.42JerJersecond, whatever ports are specified in rtp.conf would need to be forwarded
00:09.01JerJerand sip has nothing to do with udp port 8000, unless you very specifically set that port in sip.conf
00:09.24justinuwell, riddle me this... i have no ports forwarded for media, but I can see the media packets going in and out when I point an inbound call at a Playback statement.
00:09.29FuriousGeorgefirst i forewarded the ports to client and server as per voip-info then when that didnt work i set localnet and externip and voila
00:09.40justinui'm looking at tcpdump on the * host.
00:10.02justinuthey're on exactly the right ports that the SDP says they should be on.
00:10.18JerJerso then turn off the firewall
00:10.30justinuBUT, if I change the exten => command, and point the inbound DID at a Dial(SIP/exten) command, no more RTP comes at all.
00:10.31FuriousGeorgeJerJer: i said i wasnt sure about 5061, and i prefaced the port numbers with an "iirc" and you gotta admit i nailed the default udp ports in rtp.conf ;)
00:10.59justinuwhy do I need to turn off the firewall if it's obvious RTP is making it thru?
00:11.15JerJeris it?
00:11.19justinuyes
00:11.22JerJerthen why don't you hear audio?
00:11.24justinui do
00:11.28FuriousGeorgeu said it wasnt a second ago
00:11.39justinuI said that if I point the inbound DID to a playback statement, I get RTP, and hear audio.
00:12.18justinubut if I point the inbound DID to a Dial(SIP/whatever), there is no RTP at all (to either the phone, or the sip server outside my network)
00:12.19justinuand no audio
00:12.45justinusignalling works fine, the phone rings, answers, hangs up, all that is fine.
00:13.00JerJerthen something is blocking it
00:13.06JerJeror it is not getting routed correctly
00:13.29justinuwhat would be blocking outbound RTP from the * server?
00:13.45JerJerfirewall
00:13.47justinui'm watching tcpdump running on the same box.
00:13.50JerJeror incorrect SDP
00:13.53justinuthe firewall has nothing to do with it.
00:14.38justinuis * doing any media processing at this point, or is it just redirecting rtp?
00:15.06JerJeryou tell us
00:15.22FuriousGeorgeim off
00:16.19justinuyou're obviously the expert
00:16.24justinuwhy don't you tell me?
00:19.25*** join/#asterisk Snake-Eyes (n=blog@203.201.97.178)
00:24.41Corydon-wjustinu: is your firewall wide open?
00:25.32justinuno, and it's also natting
00:25.37Corydon-wIf not, you better try that first, after JerJer's excellent suggestion
00:25.43justinui'm just befuddled, because this was working fine with 1.0.7
00:25.52justinuupgrade to 1.2 and broken.
00:26.17Corydon-w1.2 isn't out yet
00:26.20JerJerdid you read UPGRADE.txt ?
00:26.35justinuno, i'll take a look at that right now.
00:28.22phpboyI seriously need some help with ISDN support in BSD :<
00:28.36justinui know PRI well, but not bsd
00:29.22phpboyshit :<
00:31.39phpboylooks like I'm gonna have to install fedora :/
00:31.50hardwirefile: wheres your pickup app?
00:31.59hardwireI have a shitload oof whiney employees now
00:33.24justinuif * places an outbound call to a SIP peer, and gets a 302 Moved Temporarily back, is there any way to get it to automatically follow that new SIP URI and send the call to it?
00:34.37Ariel_any one knows how to forward a channel to another one?  on freenode here?
00:35.00*** join/#asterisk l-fy (n=diana@yate/developer/l-fy)
00:35.02l-fyhello
00:35.10Ariel_phpboy, use CentOS instead of fedora
00:35.10l-fydid someone ever used a sangoma a104 ?
00:35.20Ariel_l-fy, hello
00:35.29l-fyhi Ariel_
00:38.05*** join/#asterisk Snake-Eyes (n=blog@203.201.97.178)
00:39.34xhelioxReally stupid question, if I want to bring in a T1 to support inbound and outbound calls, do I need to put a CSU/DSU in front of it before I terminate it to a 410P?
00:39.57justinuwhat's a 410p?
00:40.13xhelioxTE410P? The quad T1 card from Digium
00:40.15phpboyAriel_: perhaps I should just install asterisk@home str8?
00:40.18JerJerTE410P
00:40.31justinuusually you only need a CSU if you're running data
00:40.43JerJerxheliox:  no the TE410P has an intergrated CSU/DSU
00:40.56xhelioxJerJer: Is that right? Very nice.
00:41.13JerJerjust come right off your smart jack or mux (whatever you gots)
00:41.28xhelioxHrm. That seems just too easy. :)
00:42.49Ariel_phpboy, that is your choice
00:42.54*** join/#asterisk Moc (n=mochouin@modemcable173.101-70-69.mc.videotron.ca)
00:43.39phpboyI think it'll prolly be best
00:43.49Mochi all
00:43.55JerJerphpboy:  then don't expect people to support you in here
00:44.20phpboyok
00:44.25phpboyCentOS it is then
00:44.28phpboyI'll get that
00:44.38Ariel_phpboy, you can use #amportal
00:44.46Ariel_and there are some that will help you here
00:44.47Moccentos is working good for me
00:45.05phpboyI've got one of the quad Junghanns cards... need it installed and working by the weekend...
00:45.09Ariel_I have switch all my servers over to centOS and I have not had any problems
00:45.11phpboyCentOS will be fine with this?
00:45.23JerJerphpboy:  then asterisk@home is the worst thing you can do
00:45.25phpboywhere's the best place to get CentOS?
00:45.30phpboyah, ok
00:45.33Mocphpboy: www.centos.org
00:45.44Mocthere is multiple mirror... find the fastest for you
00:45.51phpboycool, I should obviously get the newest one?
00:46.08JerJerphpboy: last i knew his cards required patches and what not
00:46.11phpboyam I gonna have to do kernel reconfigs... etc? to get the card going?
00:46.16JerJermost likely
00:46.21JerJeri don't know
00:46.23phpboyfuck :/
00:47.17JerJerhire a consultant (not me)
00:47.25Ariel_phpboy, there are two different ones for them. One based on 2.4 kernel and the otherone as 2.6 kernel
00:47.38phpboyI see...
00:47.46phpboywhere's the best place to get docs for that?
00:52.25AgiNamuanyone here know a bit about 5350? I'd like to hire you for a bit
00:53.15AgiNamuwhat kinda switch type is most probably for a GC PRI? 5ess right?
00:55.43Ariel_5ess
00:57.37*** join/#asterisk shido6 (n=shido@d221-68-210.commercial.cgocable.net)
01:00.38hardwirewhy the f*ck does m,ake clean on CVS HEAD result in a large CPU intense loop?
01:02.29X-RobI'm fucking GOOD
01:02.31X-RobFUCKING GOOD.
01:02.35hardwireyou fuck good
01:02.41hardwirebah
01:02.51hardwireapp_paging.c ?
01:02.53X-RobWorks with GXP2000's, Snoms, Cisco and Polycoms
01:03.02X-Robnah. it's an agi.
01:03.09hardwireah
01:03.12hardwirelets write an app
01:03.18X-Robdrags 'em into meetme.
01:03.30hardwireI wrote an agi that does it
01:03.39hardwireworked really damn well
01:03.42X-Robhardwire - wouldn't take much. Take app_conference, plug it into Dial, and add the Call-Info header.
01:03.43hardwirebut its such a damn hack
01:04.14X-Robhardwire - how is it a hack?
01:04.26hardwireno offense to your fine work
01:04.38hardwirelets talk about this in 48 horus again if you are still upset that I called it a hack :)
01:04.41*** join/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net)
01:05.10hardwirehttp://bugs.digium.com/bug_view_page.php?bug_id=0003644
01:05.16hardwireI really wanted that to just work
01:05.17hardwirebut no
01:05.23hardwireit seems like most of it is in CVS HEAD anyways
01:05.29hardwireso I am off to use CVS HEAD from now on
01:05.32X-RobYeah
01:05.34*** part/#asterisk l-fy (n=diana@yate/developer/l-fy)
01:05.35X-Robthis only works with HEAD
01:05.39X-Robyou need SipAddHeader
01:05.44hardwireI use SipAddHeader
01:06.18X-RobIf you've written an agi, and have found problems, tell me, coz I want to avoid them 8)
01:09.11*** join/#asterisk anthm (n=anthm@h4608ac83.area4.spcsdns.net)
01:09.11*** mode/#asterisk [+o anthm] by ChanServ
01:10.10*** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr)
01:10.59jskcrhy all
01:12.05hardwirewow
01:12.07hardwireCVS HEAD is just
01:12.10hardwirelooping
01:12.14hardwireover and over and over and over
01:12.27hardwireany ideas?
01:12.29dougheckaits olympic!
01:12.35hardwirequite!
01:12.40hardwireman I want a beer
01:12.45*** join/#asterisk Sk3tCh (n=wrickout@86.127.41.114)
01:12.46Sk3tChhi
01:13.03jskcrlooping?
01:13.15jskcrplease define looping a little better.
01:13.52hardwirehttp://pastebin.ca/24080
01:13.55hardwiregood enough?
01:14.51Sk3tChhow can i make a call forward?
01:15.00Sk3tChmy iptelephon has FWD button
01:15.05hardwireneat
01:15.11hardwiremaybe its for Free World Dialup :)
01:15.14hardwirenow Forward :)
01:15.19hardwireor maybe its for transferring :)
01:15.27jskcrwow thats a new one
01:15.31X-Robit's a broken build
01:15.38X-Robrm -rf /usr/src/asterisk
01:15.41Sk3tChyes but i must set something in asterisk?
01:15.41hardwireyeh
01:15.43X-Robcheck it out again
01:15.43hardwireresnagging
01:15.52hardwiretakes so long to snag
01:15.53hardwiresoo
01:15.56X-Robsubversion DOESN'T HAVE THESE PROBLEMS
01:16.03hardwireanybody here use the 686 optimizations
01:16.05X-Robhardwire - use '-z9' on your cvs co
01:16.06drumkillait's not a cvs issue
01:16.08drumkillado make clean
01:16.11hardwireor the MMX optimizations to GSM?
01:16.15*** join/#asterisk JunK-Y (n=junky@Toronto-HSE-ppp3742731.sympatico.ca)
01:16.20hardwirewhats -z9 btw?
01:16.21*** join/#asterisk anthm (n=anthm@h4608ac83.area4.spcsdns.net)
01:16.21*** mode/#asterisk [+o anthm] by ChanServ
01:16.26X-Robdrumkilla - I've had that and make clean didn't fix it.
01:16.34jskcrI think I see the problem
01:16.35X-Robhardwire - compresses the data with gzip -z9
01:16.41hardwireohyeh
01:16.59hardwiresome 1999 files in here
01:17.10jskcrremove then echo; else \
01:17.25jskcrthen the  \ at the end of the next line
01:17.26hardwireX-Rob: haha.. make clean was doing that too
01:17.35hardwireit keeps moving a vresion file where it wants it
01:17.37hardwirethen removing it
01:17.40hardwireand moving in another
01:17.42hardwirethen removing it
01:18.10jskcrtry that hardwire
01:18.36hardwirehold
01:18.43hardwirelets see if it even wants to compile
01:18.49hardwireI just rechecked it all out
01:19.27hardwireit wants to compile now.
01:19.30Sk3tChi use actos. there is a program better like this?
01:19.33*** join/#asterisk Snake-Eyes (n=blog@203.201.97.178)
01:19.42hardwireSk3tCh: probably
01:19.54Sk3tChwhat is the name?
01:19.56Sk3tCh:D
01:19.59hardwiremake clean worked
01:19.59hardwireyay
01:20.18hardwireSk3tCh: other.
01:20.22Sk3tChpff
01:20.55Sk3tChwhat do u use for config? (simple edit conf?)
01:21.07jskcrvi
01:21.09Sk3tChvi
01:21.13Sk3tChi thought
01:21.25hardwireSk3tCh: I use jed :)
01:21.45Sk3tChi think jed is like vi
01:21.50hardwireX-Rob: know much difference inbetween CVS and 1.2.0-beta ?
01:21.52Sk3tChi am newbie
01:22.03hardwireSk3tCh: no friggin way
01:22.18X-Robhardwire - bug fixes mainly. I don't think theres any new functionality.
01:22.25hardwireor config changes I hope
01:23.44FG-awayis there some way to control the volume of an mp3 in musiconhold.conf?
01:23.55hardwirequietmp3
01:23.57hardwirevs mp3
01:24.04hardwireor.. resample the mp3 :)
01:24.08hardwirewith quieter volume
01:24.15FG-awayhardwire: thats what im using.  its beyond loud.  im using the default file
01:24.24hardwireuse an mp3 that says "to lower the volume.. move handset away from head"
01:24.31*** join/#asterisk santiago (n=santiago@63.245.86.245)
01:24.31FG-awayill just reencode my own tracks
01:24.43hardwireor change the code for app_musiconhold
01:24.44hardwire:)
01:24.45hardwireor
01:24.58FG-away*hopes
01:24.58hardwireyou could probably use custom
01:25.05hardwireand just doethe mpg123 line yourself
01:25.23hardwirethen all is happy in FuriousGeorge land
01:25.27hardwireI will be here all week
01:25.30hardwireI will
01:25.34hardwireand all night
01:25.37hardwireall day
01:25.39hardwireall night
01:25.56FuriousGeorgei hope you at least take a break to copulate with a humanoid
01:26.08hardwire######### More GSM codec optimization
01:26.09hardwire######### Uncomment to enable MMXTM optimizations for x86 architecture CPU's
01:26.11hardwire######### which support MMX instructions.  This should be newer pentiums,
01:26.11hardwire######### ppro's, etc, as well as the AMD K6 and K7.
01:26.11hardwireK6OPT  = -DK6OPT
01:26.11hardwire]
01:26.11hardwirekinda neat
01:26.19hardwireI copulated earlier
01:26.21hardwirethanky ou very much
01:26.28FuriousGeorgefornicator
01:26.38hardwireI guess
01:27.01hardwireI hope to high heaven the notify devstate crap works somewhat in CVS
01:27.08hardwirecause I am gonna freak out if it doesnt
01:27.54FuriousGeorgeits just wierd.  some installs its at a nice level, and on some its quite loud.  its the default MoH so the only variable is the actual hardware in the box
01:28.22FuriousGeorgehey is it ilegal to just rip my own mp3s for cds and use them for MoH
01:28.31FuriousGeorgefrom cds*
01:28.51wolfsonyes
01:28.56FuriousGeorgenuts
01:29.05wolfsonits called rebroadcasting
01:29.32FuriousGeorgebut its legal to play those same cds in my waiting room (if i had one)
01:29.35hardwireFuriousGeorge: resample.. normalized and 50% volume
01:29.40hardwireand use like.. rawplayer
01:29.44hardwiresince you are going to do all that crap
01:29.52wolfsonfuriousgeorge: actually that can get you in trouble as well
01:30.24wolfsonthey come around and bug restaurants all the time for royalties
01:30.59FuriousGeorgehardwire: apparently im gonna be using public domain classical pieces
01:31.19hardwireas in
01:31.22hardwireyou are being forced to?
01:31.36hardwireI would still normalize and make native or use rawplayer
01:32.05FuriousGeorgehardwire: well i dont wanna do anything ileagal, and apparently its ilegal to rebroadcast my cds as mp3 for MoH
01:32.18FuriousGeorgei got a link thats supposed to be encoded for * use
01:32.19jskcrFuriousGeorge:  umm can you find anything better after a few minutes on hold customers may commit suicide
01:32.34FuriousGeorgejskcr: whats wrong with mozart
01:32.39X-RobFuriousGeorge - get the MOH MP3s from AussieVoip. They're PD and you have permission to rebroadcast.
01:33.01FuriousGeorgeX-Rob: facinating, ill look into that
01:33.04*** join/#asterisk spackle (n=spackle@209.234.83.19)
01:33.33hardwiresee
01:33.37hardwirewhere do you go to get permissions
01:33.43hardwirethere needs to be a moh agency
01:33.50Hogieriaa
01:33.51Hogie:P
01:33.55hardwireno
01:33.57hardwiresomebody else
01:33.58hardwire:)
01:34.05spackleAASCAP
01:34.06hardwiresweet.. upgradded to CVS
01:34.11hardwireits like 500 times more chatty
01:34.15spackleor BMI
01:34.16hardwireon single verbose
01:34.21hardwireyeh
01:34.32hardwirebut why would I.. the little guy.. wanna contact them
01:34.32FuriousGeorgehavent the grateful dead "public domained" any of their concerts or something :)
01:34.36hardwirewhy not get a broker to do it
01:35.04spacklehardwire, because if they catch, you they can fine you.
01:35.36Hogiejust make your own music
01:35.42HogieI sing on my moh
01:35.48FuriousGeorgespackle: he's trying to do it legally, and i think the reason is simply b/c the royalties to "Miss You" by the "Rolling Stones" is gonna be more $ than its worth
01:37.14X-RobMan that was a funky video clip
01:37.18X-RobI'm sure I've got it somewhere, too
01:37.27spacklehardwire, it really isn't an unreasonable license fee, as far as unreasonable license fees go.
01:37.50jskcrFuriousGeorge:  hmmm thats true any of the the greatful dead bootlegs are public domain
01:37.54*** join/#asterisk NDT (n=me@cpe-24-195-219-245.nycap.res.rr.com)
01:37.57hardwirewell
01:37.58X-Robah. I've got it on DVD. How inconvenient.
01:38.00hardwireI hate all things
01:38.06hardwirehows that .. RIAA!
01:38.15FuriousGeorgespackle: but if you wanted say ten songs from each the beatles, the rolling stones, etc. till you had a hundred tracks for MoH how much would that be
01:38.28Hmm-homeshe's an 8 she's a 9 she's a 10 i know, shes got really big tits blonde hair blue eyes and i'm about to kiss my heart goodbye
01:38.43spackleit's based on how many times it gets heard daily, IIRC
01:38.46hardwireSep 27 17:38:27 NOTICE[25279]: chan_sip.c:10484 handle_request_subscribe: Failed to authenticate user <sip:1135@voip001.corp.anc.tdxnet.com>;tag=5zqf610ps0 for SUBSCRIBE
01:38.48hardwireSep 27 17:38:27 NOTICE[25279]: chan_sip.c:10484 handle_request_subscribe: Failed to authenticate user <sip:1135@voip001.corp.anc.tdxnet.com>;tag=0od1tlhxhm for SUBSCRIBE
01:38.50hardwireSep 27 17:38:27 NOTICE[25279]: chan_sip.c:10484 handle_request_subscribe: Failed to authenticate user <sip:1135@voip001.corp.anc.tdxnet.com>;tag=wxqf871jyi for SUBSCRIBE
01:38.50hardwiregod
01:38.52FuriousGeorgejskcr: i was tempted for a second but i think dead bootlegs arent quite professional enough for MoH
01:38.53hardwireI am so tired of this notice
01:38.57spackleso you can fudge your numbers a little.
01:38.59hardwireI really think this should be at a higher verbose setting
01:39.06hardwirecause it kills my console at v 1
01:39.06Supaplexwell quit fscking pasting it multiple times :P
01:39.14hardwireits different each time
01:39.15hardwire20 times
01:39.16hardwireper phone
01:39.19hardwiretimes 25 phones
01:39.27hardwireevery few minutes
01:39.51hardwiredon't you dare.
01:39.54hardwireI love my enter key.
01:40.03spacklelooks like you can find out everything you want here: http://www.ascap.com/weblicense/webintro.html
01:40.03jskcrFuriousGeorge:  there are some really good greatful dead bootlegs and remember the quality of a phone line is alot less
01:40.23hardwireass cap?
01:40.25jskcrhttp://www.iq451.com/music/sites/grateful-dead-web.htm
01:40.28X-RobFuriousGeorge - You've got me listening to Talking Heads now.
01:41.00FuriousGeorgejskcr: but arent they bootleg of concerts with their 20 min incoherent to the non-stoned jams, and cowd noise
01:41.05spackleThis is not my beasutiful asterisk this is not my beautiful sip phone
01:41.08*** join/#asterisk probonic (i=probonic@host-84-9-223-181.bulldogdsl.com)
01:41.14spackleletting the calls go by...
01:41.17hardwirebah.. it may or may not be worth something to just become a company that deals with liscencing music for Music on Hold
01:41.18hardwireheh
01:41.25hardwirebut how do DJ's at say.. a party do it?
01:41.36spackleillegally in many instances
01:41.41FuriousGeorgeX-Rob: people sleep, sleep in the daytime, uf they want to, if they want to
01:42.41*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
01:42.54X-Robw00t fear.
01:43.16hardwirefile: !
01:43.30file[laptop]oh god what
01:43.36X-Roblookout.
01:43.42hardwirewheres your pickup app?
01:43.53file[laptop]in CVS.
01:43.56hardwiremo to the fo
01:44.01X-Robapp_pickup?
01:44.07file[laptop]app_directed_pickup
01:44.12spacklefile: were you watching all the fireworks today?
01:44.13hardwireshow application picjup
01:44.14hardwiredamn
01:44.15hardwireit worked
01:44.17hardwirepickup
01:44.33jskcrI think phish also lets you make booty's at there concerts too
01:44.43X-Robfile - yeah. That was me that said 'It's fucking broken, cocksucker'
01:44.55X-Rob..but slightly nicer than that.
01:45.06hardwirecan it pickup a queue?
01:45.09file[laptop]it was broken on commit
01:45.12X-Robyeah I know
01:45.20file[laptop]I don't use it :P
01:45.23X-Robheh
01:45.30X-RobObviously no-one had used it until I tried.
01:45.36file[laptop]yeah
01:45.37file[laptop]fixed now!
01:45.39FuriousGeorgejskcr: yeah but phish has the same problem, can you imagine the average joe sitting through a half hour divided sky jam
01:45.53FuriousGeorgeor worse, 5 minutes of it while on hold
01:46.06FuriousGeorgethe radio on the other hand is legal
01:46.14jskcrFuriousGeorge:  check out http://www.archive.org/audio/etreelisting-browse.php?collection=etree&cat=Grateful%20Dead
01:46.15X-RobFuriousGeorge - Not in .au it's not.
01:46.56FuriousGeorgeX-Rob: but in the us i think its ok, or atleast everyone does it, does * support interfacing with an fm tuner card
01:47.14jskcrFuriousGeorge:  theres open source audio on that link too
01:47.36hardwirefile[laptop]: yay.
01:47.37hardwireit worked
01:47.40hardwireonly ringing right?
01:47.44file[laptop]correct
01:47.47hardwiregood
01:47.57hardwireand if I get dropped calls.. I have your current addy right?
01:48.00hardwirefor the mail bomb
01:48.02file[laptop]unfortunately :P
01:48.14FuriousGeorgejskcr: i bet if i look through that i can find an hour of acceptable music
01:48.22hardwireit worked so fine
01:48.29hardwirenow get devstate workign
01:48.39FuriousGeorgechop chop
01:48.49jskcryup theres some really good GD tracks on there too
01:49.09spackleFuriousGeorge, what do you mean the radio is legal?
01:49.47FuriousGeorgespackle: either its legal to play radio for MoH or everyone just does it <--  the essence of what i said
01:50.23spackleI think everyone just does it.  Like putting a radio behind the counter at a store.
01:50.38spackleYou can still get spanked if they catch you.
01:51.30FuriousGeorgeall classical music is public domain though right?
01:51.35hardwiregreat
01:51.36dougheckano
01:51.38jskcrno
01:51.41hardwireX-Rob: chanisavail I use for paging
01:51.42FuriousGeorged'oh
01:51.44hardwireand its funky in CVS HEAD
01:51.44spackleno, the music is but the performances are not
01:51.53dougheckaexactly
01:52.01dougheckaspackle: ever get your problem resolved?
01:52.27spackledoughecka, yeah they have me pennacillan.  Oh, you mean THAT problem
01:52.35dougheckayea
01:52.48spackleI made the changes but haven't checke on it yet.
01:53.03dougheckalol, cool :)
01:53.35spacklethe project you were interested in is on sourceforge  acrophobia.sourceforge.net
01:53.38FuriousGeorgespackle: what about covers?  can a cover band put out a covers album and have it be public domain
01:54.14spackleFurousGeorge, I'm not a lawyer, don't even play one on TV, but I work for some.  So this is speculation.
01:54.26spackleBut I think they would have to get a license to do it legally.
01:54.54JamesDotCom"all classical music is public domain though right?"
01:54.55JamesDotComhahaha
01:54.56JamesDotComwtf?
01:55.42spackle"all classical music" -> I assumed he meant mozart, etc, not recent material.
01:55.52FuriousGeorgeu assumed right
01:55.58JamesDotComahh fair enough
01:56.30spackleFurouseGeorge - Just like software licensing, music is a freakin' minefield to do legally.
01:56.40dougheckathx
01:56.50spacklemustpeople just go to ascap and ask "what to I pay"
01:57.13spackleor muzak probably has special rates.
01:57.16*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
01:57.27*** join/#asterisk znoG (n=gs@200.115.218.81)
02:01.09*** join/#asterisk TheCops (n=dump@206.248.136.146)
02:01.11TheCopsHi
02:01.14hardwirewow
02:01.16hardwiresendtext worked
02:01.19hardwirewow
02:01.39spacklehardwire - on a hard phone?
02:01.45hardwiremy snom 360
02:01.47*** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net)
02:01.52hardwireexten => 664,1,Answer()
02:01.52hardwireexten => 664,n,SendText("Monkey")
02:01.52hardwireexten => 664,n,MusicOnHold(default)
02:01.55hardwireI got off the line
02:01.59hardwireand I had an SMS icon
02:01.59TheCopsSomeone know if with SPA-2002 adapter you can configure 2 different SIP users for each FXS port
02:02.04digimeWhat have people done with Asterisk? I'm particularly interested in DIY projects and things that can be done on a small/home office (or even hobbiest's) budget. If you have clever hacks or creative functionality you've implemented, I'd love to hear what a few people have come up with
02:02.22hardwirethe MWI blinked even while I dialed 664
02:02.24hardwireand I could press it
02:03.06TheCopshardwire, ho, nice! I guess it is working with snom 320
02:03.18hardwireweird he?
02:03.23hardwireI heard it did
02:03.26hardwirejust hadn't cared
02:03.33hardwirelet me do something really nasty
02:03.54hardwireexten => 665,1,SendText("MonkeyLove")
02:03.55hardwiremohaha
02:03.59TheCops:P
02:04.06hardwirenow to place a call file
02:04.19hardwireyou must answer first
02:05.26hardwirewell
02:05.30hardwirethats officially annoyingh
02:05.40hardwiresipsak is much easier
02:05.53X-RobDidn't work for me.
02:05.59hardwirewell
02:06.04hardwireI am using the 4.3 firmware
02:06.06X-Rob<PROTECTED>
02:06.06X-Rob<PROTECTED>
02:06.06X-Rob<PROTECTED>
02:06.08hardwireand I have all sorts of things turned off
02:06.10X-RobAh, I'm using 4.1
02:06.13hardwiremost of the things needed for intercom
02:06.13JamesDotCom4.3 :|
02:06.15JamesDotComnew firmware every week
02:06.16JamesDotComhaha
02:06.21hardwire4.3 has the worst changelog
02:06.25hardwirefixed this
02:06.26hardwirefixed that
02:06.29hardwirenyeh
02:06.29JamesDotComworst as in best?
02:06.32JamesDotComcool
02:06.34hardwireno
02:06.39hardwireits a very lame changelog
02:06.53hardwireI can't wait to see "Fixed shanes speakerphone woes"
02:07.06hardwire"Fixed for option to use multi line CID"
02:07.07TheCopshardwire, intercom, do you mean a direct paging system ?
02:07.11hardwireremoved need for animated bell
02:07.24hardwireTheCops: I have an intercom extension
02:07.26hardwire*nxxx
02:07.36hardwireand a paging extension to page certain areas
02:07.38TheCopsand you speak on all phone directly ?
02:07.43hardwireyes
02:07.43TheCopsho god
02:07.48spackleanyone have sendtext work with Polycom or is it only supported on certain phones?
02:07.50TheCopsthis is fucking nice
02:07.54hardwirewhat
02:07.57TheCopsI was searching how to do that
02:07.58hardwirehearing your voice down the hall?
02:08.00TheCopslol
02:08.04hardwireyes
02:08.07Hogiehardwire: we have an outside speaker also for our pageall...
02:08.14hardwireI made every phone.. including the one down the street with a page all
02:08.19hardwireI want to get a paging application made
02:08.20TheCopsI'm using asterisk for my business, and this is very useful :)
02:08.21hardwirefor zone paging
02:08.27hardwireas well as intercomming
02:08.30hardwireyeh
02:08.31hardwirewell
02:08.34hardwirethey won't shut up about it here
02:08.39hardwireso I got their damn paging working
02:08.44hardwireand now they annoy everybody
02:09.02hardwireSupaplex: hows my enter key now bitch!
02:09.05Hogietoday, I silently transfered someone to the outside pager...  talk about funny, since they were at their desk singing with the radio
02:09.05hardwire:)
02:09.15TheCopslol
02:09.16hardwireHogie: thats mean yo
02:09.27spacklegood clean fun
02:09.38hardwireTheCops: talk to X-Rob  about his handy dandy AGI :)
02:09.41Supaplexmeh
02:09.52HogieI built in my own silent intercom numbers, the normal ones give a ding when they autoanswer, my other ones dont:)
02:09.52hardwireSupaplex: sorry I scroll your terminal :)
02:09.55TheCopsFor intercom ?
02:09.57hardwireits all the hacking I can do.
02:10.14hardwireI am gonna DOS your scrollback buffer!
02:11.23SupaplexDOS is my friend
02:11.47spacklethat's just sad
02:11.52TheCopshardwire, do you have some docs about intercom and paging ?
02:12.00hardwireno
02:12.09hardwireand you cannot get them no matter how much you ask :)
02:12.16hardwirewhat phones are you using anyways?
02:12.41TheCopssnom 320
02:13.12hardwireyeh
02:13.16hardwirewhat versino of * ?
02:13.19*** join/#asterisk cp5 (n=samy@adsl-69-232-108-255.dsl.irvnca.pacbell.net)
02:13.22jskcranyone use the zip4x5?
02:13.25TheCops1.0.9
02:13.31hardwireTheCops: can't help yah :)
02:13.38hardwirewell
02:13.40hardwireI sorta can
02:13.41cp5anyone ever have the agent module (with persistence) deadlock on them?
02:13.41TheCopsyou are using 1.2.0 ?
02:13.44hardwirelet me get you something
02:13.57cp5or know of issues with chan_agent.so deadlocking in previous CVS versions?
02:13.59TheCopsI'm reading about chaing the SIP header, you can make the phone to answer
02:14.01hardwireexten => s,1,SetVar(VXML_URL=intercom=true)
02:14.14HogieTheCops: what type of phone are you using?
02:14.15hardwireexten => s,2,ChanIsAvail(Sip/${ARG1}|s)
02:14.18*** join/#asterisk dev2008 (n=dev2003@222.33.36.205)
02:14.19*** join/#asterisk Heng (n=hengff@61.6.65.226)
02:14.21hardwireTheCops: use that setvar
02:14.25hardwireand use chanisavail
02:14.30hardwireand then you should be fine
02:14.32TheCopsHogie, snom 320
02:14.33hardwirethe setvar sets the intercom
02:14.41hardwireand chanisavail checks to see if its in use
02:14.43TheCopshardwire, and after, I need to dial the phone ?
02:14.44hardwireotherwise you hose the 320
02:14.51hardwirejust dial the phone after that
02:14.57TheCopsnice
02:14.58hardwireexten => s,3,Dial(Sip/${ARG1},12,TtA(beep))
02:14.59Hogiedamn, that's cool
02:15.03hardwireif you were using 1.2.0
02:15.09HogieI wish our c7960's would do that
02:15.16hardwireexten => _*[123]xxx,1,SipAddHeader(Call-Info: 209.112.194.40\; answer-after=0)
02:15.16hardwireexten => _*[123]xxx,n,ChanIsAvail(Sip/${EXTEN:1}|s)
02:15.18hardwireexten => _*[123]xxx,n,NoOp(${AVAILSTATUS})
02:15.19hardwireexten => _*[123]xxx,n,Cut(ICHANNEL=AVAILCHAN,,1)
02:15.19hardwireexten => _*[123]xxx,n,Dial(${ICHANNEL},12,Tt)
02:15.19hardwireexten => _*[123]xxx,n,Hangup()
02:15.29hardwirethat does it the snom compliant way
02:15.32hardwirewhere the snom beeps
02:15.36hardwirenot your * box
02:15.44hardwiredon't dos my IP
02:15.48TheCops:)
02:15.58hardwireCall-Info: can be 0.0.0.0 for the IP
02:16.04dev2008E400P port2 and port 4 can not work
02:16.04hardwirebut thats freaking verbatim
02:16.12dev2008http://lists.digium.com/pipermail/asterisk-dev/2005-September/015681.html
02:16.15hardwireHogie: you can do that really easily with c7960's
02:17.45hardwireAFAIK it works with SipAddHeader(Call-Info: \; answer-after=0
02:18.30hardwirebut if not just add a new line to it
02:18.30hardwire1 + the first line extension
02:18.30hardwireand make new sip peers for it
02:18.30hardwireand make that second line auto answer
02:18.30Hogiethat's how I have it now
02:18.30Hogieand its on 3rd line
02:18.31Hogieintercom_XXX where XXX is the extension
02:18.34Hogieis the peer
02:19.11hardwireyeh
02:19.11TheCopsIncoming call: Got SIP response 489 "Bad Event" back from 192.168.50.107 <--- did you already seen this error hardwire with snom ?
02:19.17hardwiretry  SipAddHeader(Call-Info: \; answer-after=0)
02:19.21hardwireon line 1
02:19.23hardwiresee if that works
02:19.27hardwireTheCops: yeh
02:19.33hardwireyou sorta also have to do a lot of crap tot he snom
02:19.37*** join/#asterisk thiboxnk (n=thiboxnk@66-188-89-49.dhcp.mdsn.wi.charter.com)
02:19.42TheCopsAfter a reload
02:19.46TheCopsI see this
02:19.52hardwireSet "Filter Packets from Registrar" to OFF
02:19.52Hogiehardwire: is SipAddHeader something newly added?
02:19.57hardwire1.2.0 and CVS
02:20.23Hogieah, not upthat far
02:20.29hardwireas well as under line 1.. sip.. support broken registrar on and long-sip-contact off
02:20.33hardwiregot that TheCops
02:20.34Hogieand...  if I upgrade, its gonna break half my dial plan
02:20.47TheCopsthanks
02:20.47hardwireHogie: yeh of course
02:20.49thiboxnkneed help.. having dropped calls during call waiting ne1 have problems with call waiting?
02:21.26HogieI dont have any problems right now, and I dont really have time to upgrade our 2 * boxes
02:21.43TheCopshardwire, the setvar command dont work, the phone still ring
02:21.44TheCopshehe
02:21.46bkw_OH thats rich... callerid on em_w is broken now
02:23.27*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
02:23.30hardwireTheCops: its a pain
02:24.00Hogiethanks for the info hardwire
02:24.30hardwireI just wish devstate was working correctly in CVS
02:24.35hardwireso my snom leds would be happy
02:24.53TheCopsin 1.2.0 ?
02:25.02hardwireCVS > 1.2.0
02:25.20hardwireand ChanIsAvail is broken in CVS :(
02:25.25TheCopshardwire, duh
02:25.35TheCopsdevstate come by default in CVS?
02:25.40hardwirenot at all
02:25.45hardwirehence it no workie
02:26.41bkw_chan_zap.c:4561 zt_read: DTMF digit: * on Zap/73-1
02:26.42*** join/#asterisk uma1 (n=sudhir@pool-71-114-84-37.washdc.dsl-w.verizon.net)
02:26.43bkw_haha
02:26.44bkw_thats rich
02:27.33hardwirebkw_: get out more I guess
02:27.47hardwireheh
02:28.00hardwirethere should be a "CVS Reliability" extension via IAXtel
02:28.06hardwirewhere it accepts 1 through 9
02:28.24hardwirethen we just plot it on a chart :)
02:30.15dev2008who is familiar with tormenta2.vhd?
02:30.56thiboxnkwow all this * knowledge and nobody has had a problem with call waiting.. maybe everyone in the world ids using ip phones not ata's or channel banks so they dont have these problems
02:31.11TheCopsExecuting SIPAddHeader("SIP/105-5f44", "Call-Info: 0.0.0.0; answer-after=0") in new stack
02:31.12TheCopsHo yeah
02:31.14TheCopshardwire
02:31.17TheCopsall is working proprely
02:31.18TheCopsnice!
02:31.20Saaibumhh.. why is that /var/tmp is hardcoded into channels/chan_iax2.c instead of using env(TMP) or env(TMPDIR) ?
02:31.34hardwireoh
02:31.39hardwirewhen did you get SipAddHeader?
02:31.53TheCopsWhat you mean ?
02:31.57hardwireSaaib: to piss you off
02:32.01hardwireTheCops: I thought you were 1.0.9
02:32.05TheCopsyeah I am
02:32.09hardwire...
02:32.12hardwireok
02:32.16TheCopsHo god
02:32.17hardwireits heavily patched then IMHO
02:32.17TheCopsno
02:32.21TheCopsyou are right
02:32.26TheCopsgod it is my friend
02:32.31TheCopsHa god
02:32.33hardwire?
02:32.37hardwireare you on the drugs?
02:32.46Saaibhardwire:  lol :P  i have a very restrictive system and just found out that asterisk wont work with /var/tmp
02:32.54TheCopsyeah, but it's not that, it's a friend of mine who have changed the version
02:32.57Saaibguess i'll replace that
02:33.11TheCopsthat's why my led dont work at all
02:33.29TheCopsgod he have changed a production version
02:33.35TheCopsstupid guys
02:34.18TheCopshardwire, but, you told me before ChanIsAvail is not working on CVS
02:34.28TheCopsand I'm using it in my config and it's working perfectly
02:35.00thiboxnkok what has everyone been doing for MTU/MDU apartments ?
02:36.07hardwireTheCops: as of an hour ago?
02:36.13hardwireyou are using the CVS?
02:36.18TheCopsyeah
02:36.33hardwireits not working worth a damn here
02:36.55TheCopswhen I'm using chanisavail(ZAP/3&ZAP/3)
02:37.02TheCopsit do what it is suposed to do
02:37.35TheCopsho...but I have problem with it fail (I mean, when these to channel are in use) it dont use the n+101 priority
02:37.46*** join/#asterisk flenders (n=fserto@61.8.29.101)
02:37.50TheCopsbut the major feature is ok
02:38.38hardwireTheCops: I use the |s option
02:38.39hardwireat the end
02:38.40hardwireadd that
02:38.46hardwirechanisavail(ZAP/3&ZAP/3|s)
02:38.48hardwiresee
02:38.51TheCopsWhat is the S options ?
02:39.11hardwireIf the option 's' is specified (state), will consider channel unavailable
02:39.11hardwirewhen the channel is in use at all, even if it can take another call.
02:39.20hardwiresee
02:39.27hardwireif you intercom a snom
02:39.35hardwireand the person is on the line.. eall waiting or not
02:39.37hardwirethe intercom takes over
02:39.39hardwireand drops the call
02:39.49hardwirethats why its imperative to use chanisavail
02:39.57TheCopsLOL
02:39.57hardwireso for now paging is off in my office
02:40.03hardwireotherwise.. if you do a mass page
02:40.07TheCopsyou're right
02:40.09TheCopsjust tried
02:40.09hardwireyou do a mass call drop
02:40.15hardwireso add the s option
02:40.16TheCopsthe secretary was on phone!
02:40.16hardwireand try again
02:40.17TheCopslol
02:40.57file[laptop]anyone who is using my pickup app owes me a beer
02:41.02file[laptop]because I say so.
02:41.07hardwirefile[laptop]: I can't voip you a damn beer
02:41.10hardwireotherwise I would
02:41.12hardwireand it works fine
02:41.13bkw_better watch it
02:41.14hardwireits too simple to break
02:41.22hardwirebkw_: it cannot be done
02:41.42bkw_I faxed 26 pages over IAX which kinda shocked me
02:41.48hardwirenice
02:41.52FuriousGeorgei know this is sooo 1/2 hour ago, but i used to work for a lawfirm that played a cd with "o fortuna" on it (the song that the movies play when the devil's about)  anyone got a link to a public domain cover of that one :)
02:41.53hardwirenone of them was a beer
02:41.58thiboxnkseems similar to what im experiencing .. im in a caal .. caller calls in on call waiting.. if i ignore it will disconnect all calls.. seems  broken?
02:42.14hardwireFuriousGeorge: I have a midi om fortuna
02:42.19hardwireo, fortuna :)
02:42.30JohnsieRather appropriate for a law firm.
02:42.32hardwirejust use timidity to convert it :)
02:42.46hardwirewhat about rebroadcasting FM radio
02:42.50hardwireeverybody and their dog does it
02:42.56hardwirewhy is nobodie or their dog getting busted?
02:43.03hardwireman my spelling could use some work
02:43.09FuriousGeorgehardwire: we collected debt :)
02:43.19hardwireFuriousGeorge: you got pinned?
02:43.20*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
02:43.27spacklehardwire, they do, they just get slapped with a fine, not carries away to jail like the RIAA would like
02:43.33hardwireyou know another thing you can di is explicit permission of the artist
02:43.40hardwirewe have that for the native music we use
02:43.43FuriousGeorgehardwire: i wonder the same thing, but then i start thinking:  i dont see that as much as i used to
02:43.47*** join/#asterisk AgiNamu (n=Michael@dsl081-096-215.den1.dsl.speakeasy.net)
02:44.06FuriousGeorgehardwire: i always wanted to write george clinton a letter
02:44.07AgiNamuwtf is ISDN FAS?
02:44.09wolfsonrebroadcasting fm radio via MOH is just as bad
02:44.14FuriousGeorgehe didnt od yet did he?
02:44.31FuriousGeorgeo.d.*
02:44.58hardwireFuriousGeorge: what brings that up?
02:45.08hardwirealso because its MoH for a company .. its illegal?
02:45.21thiboxnkAgiNamu isdn Nfas is usually when 2 pri's are controlled by the same d channels
02:45.27hardwirefor instance my Rammstein MoH for my house isn't illegal because I don't plan to turn a profit with whoever is calling me?
02:45.37FuriousGeorgehardwire: i brought that up b/c u said i could get explicit permission from the artist
02:45.42hardwirethiboxnk: that sounds fun
02:45.57hardwirethiboxnk: how used up is the d-channel for most t1s?
02:46.06hardwireFuriousGeorge: oh
02:46.07hardwireGeorge
02:46.09hardwireFunk
02:46.16hardwirejesus.. I thought Bill
02:46.20hardwirethat made 0 sense
02:46.27hardwireunless you mean you have sax music
02:46.30dev2008who use E400P?
02:46.31hardwireor sex music
02:46.33hardwireeither way
02:46.34thiboxnkactually it is from the old dialin modem days for multiple PRI's doing dialin with a hunt
02:46.42*** join/#asterisk ManxPower (i=eric@110.sub-70-197-211.myvzw.com)
02:46.43AgiNamuthiboxnk, but there's no N. my GC cutsheet says "Trunk type: ISDN FAS"
02:46.48FuriousGeorgeDear George, Can i get it on the good foot?  Ha Hah! (feet don't fail me now)
02:46.53hardwirethiboxnk: to get that one extra channel?
02:47.02hardwirehmm
02:47.04thiboxnkso you dont waste a channel for the pri signalling on multiple PRI's
02:47.13*** join/#asterisk hellagony (n=hellagon@c-24-130-45-125.hsd1.ca.comcast.net)
02:47.20hardwireyou sorta got your artists mixed up :)
02:47.32FuriousGeorgeno i didnt
02:47.37hardwireJames brown != George Clinton
02:47.38thiboxnkis this a BRI or PRI
02:47.46mmlj4oh, the irony... i'm working with phones now, yet i loathe talking on them
02:47.49AgiNamuPRI
02:47.52FuriousGeorgehardwire: that song is...  im thinking
02:48.00AgiNamuthiboxnk, you have any exp. with a 5350?
02:48.05thiboxnkSINGLE OR MULTIPLE
02:48.11spacklemmlj4, I feel exactly the same way
02:48.18hardwireFuriousGeorge: thinking or googling
02:48.28hardwirehttp://letrasdecanciones.tomamusica.com/i/Ice-Cube-F-George-Clinton/BopGunOneNation_20339.htm
02:48.30AgiNamuim willing to hire someone that understands either PRI or 5350 :P
02:48.34FuriousGeorgei think its atomic dog
02:48.34AgiNamuits a single PRI
02:48.43FuriousGeorgeim thinking, i know the lyrics i dont know the titile
02:49.00thiboxnkk what hardware are you using and what country/ switch type/ etc..
02:49.08*** join/#asterisk azzie (i=az@cpe-24-168-17-173.si.res.rr.com)
02:49.21FuriousGeorgenow im googling
02:49.29hardwireFuriousGeorge: I already posted it
02:49.31azzieanybody can suggest a descent cisco sip phone 79xx ?
02:49.40hardwireazzie: decent..
02:49.41hardwirehmm
02:49.45hardwireGranstream :)
02:49.49hardwireGrandstream
02:49.50hardwirePolycom
02:49.58FuriousGeorgenot atomic dog
02:50.05hardwireBop Gun.
02:50.09bkw_can i have someone try to fax 866 536 7405
02:50.13FuriousGeorgethats Ice Cube
02:50.14mmlj4someone wanna call me and tell me how horrible my setup is?
02:50.15azziehardwire: so, which cisco would that be? :)
02:50.17*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
02:50.25hardwireazzie: are you shitting with me?
02:50.32azziehardwire?
02:50.45hardwirebkw_: uno momento
02:50.57bkw_not sure I have all my clocking correct
02:50.59FuriousGeorgeone nation under one groove
02:51.48cp5anyone ever experience agent/queue deadlocks with cvs?
02:52.04azziehardwire, can't be they are worth than GS :))
02:52.37mmlj4polycoms are cool, grandstreams are poop
02:53.31hardwirebkw_: you have been faxed
02:53.46bkw_you wanna make a bet :P
02:53.49bkw_I don't htink it went thru
02:53.49bkw_haha
02:53.53hardwiremo fo
02:53.58mog_homefaxing so old school
02:53.59mog_home....
02:54.00hardwireI better go get the tx results from my fax
02:54.03hardwireits naughty
02:54.05hardwiredon't wanna get fired
02:54.36azziemmlj4, worth than GS is only windows messenger :-)
02:54.43mmlj4heh
02:55.06mmlj4so someone call me and gripe about my call quality: 504-272-2060
02:55.47azziecsim start 15042722060 :)
02:56.18mmlj4no workie?
02:57.11mmlj4hey manx
02:57.19mmlj4so call me, that's my teliax number
02:57.46spacklebkw_ you gettin too many faxes now?
02:57.54mmlj4i can mostly call out, except this lousy grandstream refuses to think it's authorized to place calls
02:58.05bkw_I don't think any of those worked
02:58.09bkw_now that pisses me off
02:58.24bkw_I think the clocking is off
02:58.28bkw_let me try something else
02:58.43ManxPowerUm, I left Texarkana today went thru Alexandria, Lafayette, Baton Rouge, to Hammond, met John in Hammond, then went back thru Baton Route to Lafayette where I'm staying.  Getting up at 6am in the morning to go back to Covington
02:58.57mmlj4ugh, what a road trip
02:59.03ManxPowermmlj4, Yeah.
02:59.11mmlj4so, has UMC gotten a server ordered yet?
02:59.11ManxPowerLeft at 6am
02:59.23ManxPowermmlj4, no idea, they have not talked to me about it.
02:59.28ManxPowerand they have my cell phone number.
02:59.30bkw_ok changed the clock priority
02:59.35hardwirebkw_: busy
02:59.40ManxPowerLast time I talked to them was about 1.5 weeks ago when I got tranks from them
02:59.40hardwireyou gave me a bad number
02:59.44mmlj4they keep thinking that one's been ordered
02:59.45bkw_doubt ful
02:59.47bkw_I just bounced the PRI
02:59.58bkw_8665367405
03:00.06ManxPowermmlj4, The telco won't be able to even deal with converting the lines for at least a month
03:00.15hardwirebkw_: I am in alaska
03:00.16mmlj4heh, nice
03:00.19hardwirethats why it isn't working
03:00.24mmlj4and they think they're opening in a week
03:00.25hardwireduh
03:00.33ManxPowermmlj4, That's a optimistic estimate.
03:00.41ManxPowerbrb smoke break
03:00.47mmlj4i smell a link to covington and a lot of transferred calls
03:01.02bkw_ok this puzzles me to no end now
03:01.36TheCopshardwire, in 1.0.9 with setvar, intercom is not working
03:01.36hardwirepuzzle muzzle guzzle fuzzle
03:01.36bkw_MUHAHA
03:01.37bkw_that was it
03:01.43hardwireTheCops: it happens
03:01.44bkw_the clock priority in the TNT was wrong
03:01.47hardwireits a tricky pickle
03:01.54hardwireand you need to patch 1.0.9
03:01.56hardwireI forgot :)
03:01.59hardwireand move a %s around
03:02.06TheCopsho
03:02.06TheCopslol
03:02.10TheCopsI'm not a coder hehe
03:02.18TheCopsI'll wait 1.2.0 release
03:02.23hardwireI haven't had papa johns in forever
03:02.45mmlj4hardwire: not long enough, in my book
03:02.46bkw_XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX
03:02.51hardwireI curse on ChanIsAvail
03:02.51X-RobTheCops - what sorta phone?
03:02.52bkw_Love that one
03:03.02TheCopsX-Rob snom 320
03:03.03hardwiremmlj4: been goign to this place down the street for so long
03:03.09X-Robhttp://www.voip-info.org/tiki-index.php?page=Asterisk+Paging+and+Intercom
03:03.11hardwirepapa johns heavy carbs is probably going to kill me
03:03.12X-RobSee 'allpage.agi'
03:03.15X-RobI've _just_ put it there.
03:03.19hardwireX-Rob: oh have you now
03:03.25hardwireI think I am going to one up you X-Rob
03:03.30hardwireif you can deal with the heat
03:03.33X-Robhardwire - woot.
03:03.36hardwirewoot
03:03.37X-Robyou go!
03:03.41hardwireI need to contribute to asterisk damnit
03:03.50hardwireso app_paging_system_of_doom.c will have to exist
03:04.01X-Robwell.
03:04.06X-Robjoin #openpbx
03:04.36Johnsiehahaha
03:07.16BrianR___Anyone know if palm PDA's can play wav49 encoded voicemail attachments?
03:08.03*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
03:08.06FuriousGeorgecan anyone think of a good reason to use call parking when your client supports 5 conversations?
03:08.27X-Robpark call. page all systems 'Fred. Call Parked on 72. Fred. Call parked on 72'
03:08.40FuriousGeorgethanks x-rob
03:08.48BrianR___FuriousGeorge: Moving a support call from your desk to the server room
03:08.51ManxPowermmlj4, on the bright side I *might* have broadband by saturday.
03:08.57mmlj4cool
03:09.05MikeJ[Laptop]and monkeys
03:09.20BrianR___FuriousGeorge: You can't just use transfer in that case because you won't be there yet when the phone starts ringing.
03:09.31ManxPowertranks = good
03:09.39FuriousGeorgeBrianR___: yeah, i figured that was the advantage
03:09.51mmlj4the only PC parts wholesaler in NOLA can't get DSL nor cable, they're hoping for wireless
03:10.23mmlj4ManxPower: so call me, i need you to laugh at how horrible my call quality is: 504-272-2060
03:11.11FuriousGeorgethe only reserved numerical extension is 0, right?  technically i could set up the parked calls to be on 1-20, for instance?  the reason i ask is b/c these people are used to an old shcool switched phone system where they had a line 7 which corresponds to an actual telco did
03:11.43MikeJ[Laptop]key systems are great for that
03:11.46MikeJ[Laptop]:P
03:11.50MikeJ[Laptop]night all.
03:11.53FuriousGeorgenight
03:12.06spacklesleep tight, don't let bed bugs bite
03:12.18FuriousGeorgethough in a sense that is good b/c everyone gets a headsup status as to where there is a call.  if its blinking its on hold, etc
03:12.26MikeJ[Laptop]the only bug in my bed is my wife ;)
03:12.30spackleif they do report them to mantis.digium.com
03:12.42MikeJ[Laptop]bugs.digium.com
03:12.44*** join/#asterisk websae (i=websae@207-118-147-212.dyn.centurytel.net)
03:13.15*** join/#asterisk littleball (n=littleba@bb219-75-114-161.singnet.com.sg)
03:13.25FuriousGeorgeis there any softphone which supports mapping status lights to * parked call slots?  does asterisk support that?
03:13.57websaeanyone here using MySql with Asterisk?
03:14.00websaehaving some issues...
03:14.54*** part/#asterisk spackle (n=spackle@209.234.83.19)
03:15.02hardwireFuriousGeorge: yes and no
03:15.16hardwireI don't know any soft phones that do that
03:15.23hardwireother than the snom 360 softphone
03:15.27hardwirewhich you can download
03:15.33hardwireand set destination functino keys
03:15.36hardwirefunction keys
03:15.40*** join/#asterisk argos73 (n=argos@65-85-207-125.client.dsl.net)
03:15.43hardwirehowever.. getting that to work with asterisk is a headache
03:15.56blitzrageevening all
03:15.57hardwireI can only get the snom to subscribe once and get a report on all extensions
03:16.01hardwireand then the leds' just stic
03:16.02hardwirek
03:16.07hardwireblitzrage: mornin
03:16.27filehey blitzrage
03:16.37hardwirewell I was all set to fax bkw some porn
03:16.44blitzragehardware, file.
03:16.44hardwirebut his toll-free inbound hates AK
03:16.48blitzragehow goes?
03:16.49*** join/#asterisk spackle (n=spackle@209.234.83.19)
03:17.53blitzragegood to hear - just playing with my new printer/scanner which I got last week (finally got a USB cable)
03:18.15blitzrageactually printing and scanning and signing all these NDAs I had but couldn't do anything with since I didn't have a printer :)
03:18.17fileUSB is overrated
03:18.18ManxPowermost tollfrees hate AK
03:18.22filebluetooth is where it's at
03:18.55blitzrageI need a better scanning program... Windows doesn't handle multiple pages :)
03:22.38FuriousGeorgewhats the snazziest iaxphone?
03:22.47FuriousGeorgeim looking for an opinion :)
03:22.49thiboxnkstill ne help with call waitng call drops will be appreciated..
03:23.07spackleFuriousGeorge: hardphone?
03:23.14FuriousGeorgespackle: softphone
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03:24.08*** join/#asterisk kfuq (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
03:24.09pauldyanyone have any suggestions on the cheapest way to send out bulk fax
03:24.50FuriousGeorgespackle: ok hardphone
03:24.53blitzrageanyone who asks to send out bulk faxes is sketchy in my books :)
03:24.56spacklepauldy, have a service do it.
03:25.06pauldyblitzrage, it isn't spam
03:25.16blitzragepauldy: haha, cool :D
03:25.26pauldyits contracts multiple pages maybe 10-20 a day
03:25.31blitzragepauldy: I'm just being a pain really... continue about your business :)
03:25.44blitzragepauldy: yah, that'd be a lot of faxes
03:26.14pauldyjfax killed their account this month
03:26.26ManxPowerI got an authentic looking e-mail from "paypal" tonight.  Was the standard notification of a purchase made for an expensive watch, shipped to beverly hills.  Even had a nice "dispute" link, which sent you to a site that was NOT paypal.
03:26.29pauldyI looked at what they were paying I know there has to be a cheaper solution
03:26.34ManxPowerdamn fscking scammers
03:27.05ManxPowersleep now.
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03:37.25drbrownany opinons on the TE110P used in conjunction with a rhino channel bank???? specifically echo????
03:38.46bjohnsonpauldy: local or long distance
03:39.28websaemy company provides termination at $.017/minute
03:40.11pauldybjohnson, mostly local but a bit of it might be inside the metro area but long distance
03:40.32bjohnsonand they use jfax for local faxing?
03:40.39pauldyyup
03:40.42bjohnsoncan't they just use a fax machine?
03:41.02pauldytrying to find a service they can be happy with for reliability or just build a hylafax server
03:41.31pauldybjohnson, they could
03:41.32bjohnsonwhy not a regular fax machine on a pstn?
03:41.45pauldybecause that doesn't cut down on the paper
03:41.51BrianR___Is there any way to answer calls coming in over PRI with hylafax on a box also running asterisk?
03:41.53bjohnsonfor $30/mo for pstn and $150 for a fax machine with memory
03:41.58*** join/#asterisk da_monumental_1 (n=da_monum@rrcs-24-172-102-198.midsouth.biz.rr.com)
03:42.02pauldynever been in a big office waiting on someones 40 page contract have you
03:42.04bjohnsonpauldy: use hylafax for incoming
03:42.22bjohnsonpauldy: more lines?
03:42.33*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
03:42.42pauldylisten now your just second guessing my advice
03:42.43bjohnsonactually we use mgetty for incoming
03:43.09pauldythe idea is a service that requires little effort on my part to switch them over to
03:43.22pauldynext best thing is roll my own
03:43.36pauldyI was hiopping someone else has been down this road before in here
03:44.04bjohnsonI don't see how a fax machine is such a problem
03:44.52pauldywell I understand why they don't like it but I guess that comes from experience
03:45.05BrianR___having only one fax machine in a busy office sucks hardcore...
03:45.17bjohnsonso get more than one
03:45.24bjohnsonuse it for outgoing only
03:45.38*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
03:45.54bjohnsonany other option requires original docs in electronic format .. not something that is acceptable in all offices
03:45.59BrianR___fax machines are best reserved for outgoing only...
03:46.02bjohnson(or easy)
03:46.49bjohnsonwe use a fax machine for outgoing and a linux box for incoming
03:46.54bjohnsonon a shared line
03:47.14bjohnsonbut maybe we don't do as much volume
03:47.16BrianR___We're using multifunction copiers at my office... Take up two analog lines, but they're decent if someone sends a huge fax. Can also do outbound faxes via an email gateway - would like to get that set up..
03:47.42BrianR___I have a pri terminated on my asterisk box, would like to give every employee a direct dial fax->email for inbound... But app rxfax doesn't work so well.
03:47.44bjohnsona simple fact is that faxes take time, if one is too slow, the easiest solution is to use two
03:48.34BrianR___bjohnson: Well.. If you're sending the faxes via an email gateway on a box with a PRI, you can run dozens of faxes at once...
03:49.10BrianR___ditto for big office fax machines that can take multiple analog lines
03:49.18bjohnsonwhat hardware sends faxes over PRI?
03:49.18pauldynot only that but the reciept comes to your e-mail and all sorts of nice things that just make it easier
03:49.25websaeanyone know of a good DID provider?
03:49.33pauldyThe copier thing might be a solution I had not thought of that
03:49.45bjohnsonpauldy: if you have to scan in documents before faxing them, email gateways are not easier
03:49.55pauldyif they can get it on a lease and replace the copier they have it might work out to be worthwhile
03:49.55BrianR___bjohnson: The combination of a multifunction copier with fax to email and a hylafax email to fax gateway...
03:50.18bjohnsonand fax to PRI?
03:50.24bjohnsonmodems?
03:50.30*** join/#asterisk mkl1525 (n=daniel@84.19.197.144)
03:50.39BrianR___The toshiba copiers we're playing with can tack an email suffix onto any unqualified address the user enters, so it's nice and transparent for the end user.
03:50.55BrianR___bjohnson: softmodem running on the asterisk box - either spandsp or t38modem
03:51.04pauldyBrianR___, do you happen to have the model info?
03:51.08bjohnsonhow stable are those now?
03:51.21BrianR___pauldy: estudio 350, I think...
03:51.27pauldyty
03:51.30BrianR___bjohnson: My testing with spandsp so far shows it sucks
03:51.41BrianR___About to do some testing with t38modem - word is it's much more mature.
03:51.44bjohnsonso not really a solution then
03:51.44*** join/#asterisk bmg505 (n=leon@rndf-146-17-28.telkomadsl.co.za)
03:52.00bjohnsonmight be at some point .. but not now
03:53.37FuriousGeorgeis cable generally a better voip solution than dsl?
03:53.44pauldyBrianR___, the copier you have used does it do email replies for the fax transmission results  etc..
03:53.57FuriousGeorgecan we generalize about latencies and bandwidth in the US48
03:54.25blitzrageOT: anyone know of some good software for linux that I can use to monitor traffic in a network? Ideally I'd like to break it down based on IP address (internal)
03:54.27pauldyFuriousGeorge, my guess is it would be move of  a debate over pppoe vs dhcp
03:54.37blitzrageby monitor traffic I mean usage
03:54.45blitzragepretty graphs would be nice too :)
03:54.51spackleblitzrage: ntop?
03:54.57bjohnsonblitzrage: iptables with mrtg
03:55.02BrianR___pauldy: haven't played with it enough - looks like the user has to enter their email address if they want a receipt.
03:55.18FuriousGeorgepauldy: is that what its about?  i dont really care about authentication, im more worried about latencies and whatnot.  let me do a little experiment
03:55.19bjohnsonntop doesn't do bandwidth accounting which I think he wants
03:55.26blitzragebjohnson: ahhh yes, I used MRTG once... thanks
03:55.36bjohnsonmrtg just graphs
03:55.47bjohnsonyou needs data for it to make the graphs from
03:55.48blitzrageyah.. bandwidth accounting is what I want - need to watch how much the room mates are downloading
03:56.03bjohnsonmost ipaccounting stuff on linux uses iptables as the back end
03:56.13pauldypppoe inherently introduces more latency as packets must be filtered
03:56.20blitzragebjohnson: hrmmm... that might be a bit of a problem since I"m already running gShield...
03:56.34bjohnsonblitzrage: if you have an extra box, ipcop them into a separate subnet
03:56.36pauldyeven if they use a method similar to the cutthrough method on switches it is still that extra milisecond
03:57.03blitzragebjohnson: unfortunately no extra box - just a single exit point - needs to do all :(
03:57.13FuriousGeorgei got this one dsl connection.  i got a download on an ftp going on @ 40kBps and the pings to google are around 1 second
03:57.32bjohnsonblitzrage: if the exit point is a linux box, you're laughing
03:57.43blitzragebjohnson: it is
03:57.44bjohnsonotherwise you're out of luck
03:58.18websaeanyone know of a good DID provider....?
03:58.35bjohnsonmake iptables rules so that data to/from their IPs go to custom tables, the tables will log the data transfered through them
03:58.53blitzragewebsae: www.mixnetworks.com (yes, I work for them)
03:59.13bjohnsonthen you use mrtg to track the differences in the kB count over time increments
03:59.25bjohnsonand display them
03:59.37BrianR___it looks like t38modem is a softmodem which exposes a h323 interface on one side and a pty on the other end, so you can use hylafax even if asterisk is controlling your PRI. Very slick.
04:00.19blitzragebjohnson: hrmmm... seems like more work than I'm able to spend right now :(
04:01.03pauldyblitzrage, see if ntop pushes your buttons
04:01.22pauldyit does do some abdnwidth accounting
04:01.40pauldynot per minute but cumulative
04:01.43*** join/#asterisk cio (n=na@adsl-068-209-198-242.sip.bhm.bellsouth.net)
04:02.07*** join/#asterisk pashah (n=pashah@ns.itconnection.ru)
04:02.13pashahhello
04:02.24*** join/#asterisk ikey (i=ikey@220.226.28.95)
04:02.34cioHi all.  I'm hearing some chirping on the line in my * system, it just seemed to start a few days ago.  Any ideas what may be causing it?  I'm using Polycom IP phones.  This is a localnet phone to asterisk to 1fb call out.
04:02.44pashahcould anyone please tell me if te205p has a jumper to toggle t1/e1
04:02.56JunK-Yunlock the PAP2-CA wohooo.
04:03.09pauldycio, did you make sure you don't have any rats in your * box
04:03.19spackleor crickets
04:03.26JamesDotCompashah: stop being so fucking lazy, http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE205P
04:03.30cioNo, nothing like that ...
04:03.33cioheh
04:03.40*** join/#asterisk argos73 (n=mike@65-85-207-101.client.dsl.net)
04:03.40hypa7iablitzrage: TAUG!
04:03.48BrianR___I suppose I could use hylafax normally if I put a two port ISDN card in the box and used a crossover cable...
04:03.59blitzragepauldy: thats fine... what I'm looking for is just to track how much bandwidth each user on the network is using so I can blame someone for the 60GB of b/w used so far this month :)
04:04.03cioAny other suggestions?
04:04.05pashahjamesDotCom: it does not say if it has a jumper
04:04.10blitzragehypa7ia: w00t!  I found two people to carpool with tomorrow!
04:04.17hypa7iablitzrage: nice!
04:04.23pashahJamesDotCom: have you seen the card?
04:04.27hypa7iai hope we move it next month
04:04.32hypa7iai'm missing TASK
04:04.35JamesDotCom"The TE205P supports both E1 and T1/J1 environments and is selectable on a per-card or per-port basis"
04:04.48hypa7iaand i forgot to call Mix today but i will tomorrow :-)
04:05.00cioOn a completely different note, is there anyway to improve the quality of the standard asterisk voice prompts?  If I record prompts on my own, the sound is crisp and clear as I record at 16k, but the stock recordings are only 8k I think.
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04:05.07blitzragehypa7ia: I doubt it'll get moved yet again... :(
04:05.15ronaldl79G'day, room.
04:05.21pashahJamesDotCom: so you have not seen it? but it should have jumper similar to te110p i presume
04:05.33blitzragehypa7ia: we actually just moved it this month :)
04:05.37JamesDotCompashah: i see the jumpers i'm sure they are on the picture on that page
04:05.46blitzragehypa7ia: bad idea to keep moving it each month... I would assume
04:05.50ronaldl79Q: Will the sound drivers for Gnome or KDE override the audio output of chan_oss??
04:05.52JamesDotComso yes
04:05.58pashahJamesDotCom: thanks
04:05.59argos73pashah: just installed one of those cards today..
04:06.03ronaldl79...for the console
04:06.04hypa7iablitzrage: i know... it's like my career debate but in user group for,  security or telecom?!? arrgh!
04:06.29JamesDotCompashah: np, sorry for the anger in the initial response
04:06.29JamesDotComahah
04:06.43pashahJamesDotCom: np
04:07.58pauldyhey blitzrage I would just find the user with the empty lotion bottle and blame it on that one, pr0n
04:08.26blitzragepauldy: LOL
04:08.42joelsolankiHi jamesdotcom
04:09.47JamesDotComhi joelsolanki
04:09.52*** join/#asterisk santiago (n=santiago@63.245.86.245)
04:09.56JamesDotComhow did you go yesterady? i had to leave sorry
04:10.10Cresl1nping
04:10.13joelsolankiYes i waited for u long but u didnt came :)
04:10.16cioargh... chirping seems to be related to something else running on that linux box.. I stopped everything but * and the chirping goes away.. What sucks is I have to go one at a time and test...
04:10.54joelsolankii too had to move out of city so just came today only. no progress on it. here is the pastebin for the ser+asterisk on same box. http://pastebin.ca/24028
04:14.16drumkillahypa7ia: !
04:14.35hypa7iadrumkilla!
04:14.39blitzragedrumkilla: !
04:14.45blitzrageCresl1n: !
04:14.45drumkillablitzrage: !
04:14.53blitzragelol
04:14.58blitzragethat never gets old for me
04:15.00Cresl1nblitzrage!!!!
04:15.05joelsolankiJamesDotCom: got any clue ?
04:15.09drumkillaCresl1n: !!!1!
04:15.22Cresl1ndrumkilla!!!!
04:15.24blitzragewasn't me!
04:15.55hardwirefile: reload chan_sip.so.. reloads the lib?
04:16.06fileit calls the reload function
04:16.07hardwirelike.. new apps and so forth tada?
04:16.10argos73there something funky I need to do to get incoming caller id to work with a channelized T1?  works when it's plugged into the channel bank, but doesn't when I send those channels to asterisk...
04:16.11hardwireoh
04:16.14hardwiredamn the reload function
04:16.51hypa7iaeeek!
04:17.47joelsolankiJamesDotCom: U there?
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04:28.33philmI'm having trouble calling the console. I dial->get music on hold->and have to hangup and answer before I can talk. Why the music on hold?
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04:30.59zapahi all
04:31.25philmhi
04:31.45philmZapa ever use console/dsp?
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04:32.09zapano philm sorry
04:33.44ronaldl79Can anyone recommend a BYOD provider comparable to BroadVoice?
04:34.50zapai am having troubles with a PRI30 telrad, to Digium Card when i call from telrad pbx to the trunk i only hear dial tone but asterisk answer the ivr, but when i call from asterisk to telrad i don't have any problem, any idea?
04:35.33websaeronald179--i have a suggestion, i messaged you
04:37.38*** join/#asterisk justinnnnn (n=justinnn@61.95.68.85)
04:37.39justinnnnnhey
04:37.42justinnnnndoes anyone no if voipjet is down ?
04:37.50justinnnnni havnt been able to call through em for like 24 hours now...
04:37.51justinnnnn???
04:38.14file[laptop]have you e-mailed their fastsupport?
04:38.23justinnnnnnah they usualy just ignore it
04:39.35justinnnnndo u use them ??
04:40.11ronaldl79Anyone have any ideas as to why a virtual number won't pass DTMF to Asterisk?
04:41.53justinnnnnare there any other good international places ?
04:42.54FuriousGeorgesomeone should start a bsd orchestra
04:44.07FuriousGeorgerelease public domain performances for music on hold
04:44.36*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
04:45.28FuriousGeorgeronaldl79: ur codec?  passing it inband when u shouldnt?  "lose" dtmf?
04:45.36FuriousGeorgeur client?
04:45.52FuriousGeorgea combination?
04:47.22justinnnnninternational providers ???
04:47.23justinnnnnanyone ?
04:47.36ronaldl79furious: 711
04:48.06FuriousGeorgehuh?
04:48.24ronaldl79furious: I had my mom and one other person test a virtual DID in Detroit .. which rings here in Denver...neither could navigate the IVR. However, all is fine on the primary DID.
04:48.42FuriousGeorgeronaldl79: im kinda new myself
04:48.44ronaldl79furious: u/a law
04:48.45FuriousGeorge~IVR
04:48.46jbothmm... ivr is Interactive Voice Response
04:48.54FuriousGeorgethey have that for asterisk?
04:49.08FuriousGeorgei use festival :)
04:49.12ronaldl79Furious: Yes...and it's simple to build. Asterisk has everything.
04:49.43ronaldl79So, the DTMF isn't being passed from the virtual DID to navigate the IVR menus. As for festival, I haven't really tried to get that working yet.
04:49.47FuriousGeorgesnazzy..  i use ulaw and iax mostly with sip clients almost always behind nat with my * server.   i never have problems
04:50.21FuriousGeorgeeyebeam is my client but ive used xlite too
04:50.58ronaldl79I have a Cisco ATA 186 I'm using.
04:51.00*** join/#asterisk Knight_DKN (n=knight_d@61.95.68.85)
04:51.14ronaldl79Just got it unlocked from Vonage a few days ago ... I'll only use a soft phone on a PDA or laptop now.
04:51.21ronaldl79The call quality is superb.
04:51.39FuriousGeorgeronaldl79: i can tell u my clients dtmf settings
04:52.11ronaldl79I just built another * PBX tonight for a non-profit ... installing it tomorrow to demo and test....they'll be quite pleased. I also just discovered the console/intercom feature ... * has so much sh*t ... it's hard to keep up.
04:52.35FuriousGeorgeconsole intercom?
04:52.43ronaldl79Yes ....
04:52.46mog_homeuse dial
04:52.55FuriousGeorgeno sound card
04:52.55mog_homeapp_intercom is dead
04:53.00ronaldl79The 'console' can be used a pager/intercom system.
04:53.17FuriousGeorgei was just thinking about that functionality
04:53.17ronaldl79right, Dial(console/dsp)
04:53.18Cresl1ndingdong
04:53.24Cresl1napp_intercom is dead
04:53.35FuriousGeorgeand all your channels ring?
04:53.38ronaldl79I wasn't speaking of app_intercom to begin with...dingdong.
04:54.07FuriousGeorgeronaldl79: have you messed with the relaxed dtmf setting?
04:54.13ronaldl79I haven't, furious.
04:54.28ronaldl79* is just a big toy of toys ... lol .... I love it.
04:54.32FuriousGeorgeniether have i :)
04:55.23FuriousGeorgewhat dtmf related settings does your sipphone have?
04:56.12ronaldl79I can't recall from memory .. but it's not me.... it's those calling from the PSTN ... to the virtual DID ... which isn't passing DTMF..
04:56.34flendershi, I'm just setting up asterisk now, but I'm having some weird problems while testing: I can make and receive calls just fine, but though, voice is not going both ways. When ring from the PSTN line to my DID, I can hear voice coming from PSTN->VoIP, but not from VoIP->PSTN. and When calling from VoIP to PSTN (or mobile), I can't hear anything on any direction. any hints?
04:56.38ronaldl79If they call the primary DID directly, DTMF passes just fine....so I'm wondering what the virtual DID is cuasing....
04:57.00spackle~NAT
04:57.02jbotmethinks nat is Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
04:57.05ronaldl79Flenders: Nat
04:57.29heath__flenders: which protocol?
04:57.41FuriousGeorgeronaldl79: i would assume it has to do with the codec and/or how your client is passing the dtmf.  you can hear eachother over the virtual did right
04:57.58ronaldl79Yes, furious.
04:58.06flendersheath__ on my sip.conf: disallow=all
04:58.07flendersallow=alaw
04:58.08flendersallow=ulaw
04:58.17ronaldl79Remember, it's the PSTN passing DTMF ... it's not me...
04:58.43FuriousGeorgeronaldl79: i know what ur saying, but you cant change settings on the PSTN so...
04:58.44kramn
04:58.49*** join/#asterisk mydssmojo (n=bib@S0106001217475911.cg.shawcable.net)
04:59.07ronaldl79Furious -- Everything works well on the primary DID ... DTMF is passed, etc...it's the virtual DID where it's not happening.
04:59.27*** join/#asterisk jmg (n=cartel@shinobi.thoughtcrime.org.nz)
04:59.29jmghey all
04:59.34ronaldl79So, I'm wondering if something's screwed up on BroadVoice's end...
04:59.36justinnnnnanyone using voipjet ?
04:59.42ronaldl79I do, justinnn.
04:59.42jmgis there a howto for asterisk on sarge?
04:59.43Knight_DKNI am
04:59.46FuriousGeorgeronaldl79: whats the primary did?
04:59.50FuriousGeorgepots?
05:00.00justinnnnnronald is it actualy working for u ?
05:00.11mydssmojoanyone using AMP?
05:00.11ronaldl79No, it's VoIP via Global Crossing, Furious.
05:00.14ronaldl79Yes, justinn.
05:00.17Knight_DKNYep I'm using voipjet all day
05:00.19ronaldl79I have, mydssmojo.
05:00.30jmgim trying to install now
05:00.33FuriousGeorgeronaldl79: my bad, i was misunderstanding.  both use the same codec then?
05:00.36jmgwhats the canonical install guide?
05:00.41ronaldl79Knight -- I can't imagine how much money you spend if you're using it all day.
05:00.47ronaldl79Yes, furious.
05:01.30flendersronaldl79 , do I have to open port 5060 on my firewall if I'm using a SIP account?
05:01.42FuriousGeorgeflenders: yes and many more
05:01.43mydssmojoIs there a current installation instructions for AMP on fedora? Thanks
05:02.00flenderssomeone in here told me me the other day I didn't need it
05:02.03ronaldl79mydssmojo ... I don't believe so, but you can search on Google for some reference points.
05:02.10ronaldl79Someone lied to you, flenders.
05:02.17flendersFuriousGeorge , do you think that may be the reason for my problems?
05:02.32blitzragekram: evening
05:02.33justinnnnni like to ride around on the leckie bus
05:02.34mydssmojothanks ronald179!
05:02.41Cresl1nkram: hey :-)
05:02.43FuriousGeorgeflenders: your just trying to log asterisk into a voip provider?
05:02.50flendersyes
05:02.50ronaldl79yw, mydss
05:02.54*** join/#asterisk santiago (n=santiago@63.245.86.245)
05:02.57FuriousGeorgeflenders: udp 5060
05:03.00Knight_DKNYeah the leckie bus is a pretty good ride
05:03.08FuriousGeorgeflenders: voip is sip right?
05:03.18flendersFuriousGeorge , yes
05:03.24FuriousGeorgeflenders: and you have a nat or firewall
05:03.37flendersFuriousGeorge : yes
05:03.57FuriousGeorgeand your sip client is on the same network as *
05:04.07flendersFuriousGeorge , yes
05:04.23FuriousGeorgejust udp 5060 --> asterisk, i think
05:04.30FuriousGeorgecan someone verify
05:04.33flenders* is behind the firewall on an 192.168.x.x network
05:04.47FuriousGeorgewhats *'s ip on your network
05:05.01flenders192.168.10.20
05:05.08flendersclient is 192.168.10.185
05:05.25FuriousGeorgetry it
05:06.32FuriousGeorgeronaldl79: the only thing that comes to mind is that when the call goes PSTN --->  BroadVoice's Server, it gets compressed to much or something
05:07.27flendersDNAT       udp  --  0.0.0.0/0            203.x.x.x     udp dpt:5060 to:192.168.10.20:5060
05:07.30flendersright?
05:07.47flendersjust added that rule on the firewall
05:07.57FuriousGeorgeronaldl79: i know you can play a recording of DTMFs into a telephone on the PSTN and it will call so i dont think theres much too it on that end
05:08.15FuriousGeorgeflenders: dont ask me, i used ipcop's gui to fwrd ports
05:08.32FuriousGeorgeit /looks/ ok...  whats with the 203.X.X.X
05:08.48flendersthat's my firewall/external ip
05:09.24FuriousGeorgeoh i get it, youre censoring it :)
05:09.34FuriousGeorgeright? :|
05:09.47flendersyup
05:09.58flendersjust to avoid shit loads of probes on my firewall
05:11.13*** join/#asterisk cio (n=na@adsl-068-209-198-242.sip.bhm.bellsouth.net)
05:11.32FuriousGeorge(dnt foget to restart asterisk)  so?
05:11.42flendersjust trying now
05:11.44cioAnyone ever see chirping with 1.0.7 on a debian system?  I'm runnning samba, dhcp3, bind9, mysql, apach2, dovecot on that box w/asterisk.
05:12.07FuriousGeorgechirp=jitter?  ive had that
05:12.20cioFuriousGeorge: What did you do to solve it?
05:12.21FuriousGeorgenot on debian tho :)
05:12.35flendersSIPclient->PSTN not working
05:12.49cioAbout every four seconds theres a 'chirp'.
05:13.00cioSounds like somebody taking a corless phone off the hook.
05:13.05cioI'm using polycom sip phones.
05:13.06FuriousGeorgeused settings like echocancel and jitterbuffer in client ans server.  used ulaw or gsm
05:13.10cioIt started about three days ago.
05:13.18cioI'm runing g729, think that would do it?
05:14.05FuriousGeorgecio:  how much bandwidth does it use?
05:14.30cioThe phones and * are on a gig lan, barely utilized.
05:14.43FuriousGeorgebut what about your isp
05:15.01FuriousGeorgeyour talking internally it chirps?
05:15.06cioNo ISP in the loop, the phone is connected to a switch, switch conneted to *.
05:15.16FuriousGeorgeahh
05:15.25flendersFuriousGeorge : not working yet mate.
05:15.42flendersronaldl79 : what did you say about NAT before?
05:16.15FuriousGeorgecio: you running X on debian box?
05:16.46cioNope.  It just started a few days ago.  I did install SQUID in that time frame, but I've stopped it and no changes.
05:17.31cioI did put a single ip tables rule in there to get port 80 traffic redirected to 3213 or something like that.
05:19.02FuriousGeorgecio: i dunno, you tried changing codecs, clients, seeing if anything makes it go away
05:19.12FuriousGeorge?
05:19.20cioYea, pretty much everything.
05:19.23joelsolankiJerJer: Hello
05:19.43FuriousGeorgeflenders: did you try setting externip or localnet in sip.conf
05:19.43cioRebooting my phone, haven't done that ...
05:20.04cioYea, same here.  Thanks, though!
05:20.54flendersFuriousGeorge : yup
05:21.08argos73cio: could be one of those other jobs doing frequent (probably large) disk accesses..  some motherboards get a little goofy...
05:21.10flendersFuriousGeorge : and I have nat=yes on sip.conf
05:21.26cioI'm going to kill everything again and see if it stops...
05:21.36argos73cio:  try killing mysql first
05:21.45argos73then apache
05:21.51cioI will.  You have problems there?
05:22.08argos73doubt it's dhcp or bind - they're pretty lightweight
05:22.11flendersFuriousGeorge: I think I got it
05:22.22flendersFuriousGeorge: netmask on localnet
05:22.29argos73oh, i've battled my share of these problems...
05:22.55cioNadda.... hrm...
05:23.20FuriousGeorgeflenders: lemme know how that goes
05:24.01argos73hmm - dovecot's a mail handler?  those can be pretty rough on a system.
05:24.20cioWhat's the least problematic codec?  ulaw?
05:24.49argos73if bandwidth isn't an issue, probably.
05:24.55argos73less cpu-intensive
05:25.23cioWasn't that.
05:25.38argos73syslogd?
05:25.48*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
05:26.05cioWell, at this point I need to make a IP to IP call to make sure it's not my dual X100P's.
05:26.20flendersFuriousGeorge: all good now mate! thanks for your help
05:26.22argos73you've checked for interrupt sharing, right?
05:26.26cioI have a couple phones that will be in use tomorrow.
05:26.35FuriousGeorgeflenders: it was all you ;)
05:26.52cioargos73: This just started.. if it was interrupt, I would think it would have been around since the server was built, right?
05:27.20argos73most likely, but stranger things have happened...
05:27.32cioYea, I agree.
05:27.40FuriousGeorgecio: i had this box that had a slight echo despite all the usual settings, and it turned out i'd built * w/o mmx support
05:28.01cioHrm... Well, I'm just using the standard 1.0.7 debian packaged...
05:28.28cioIt's been great until now.
05:29.05argos73FuriousGeorge: second that - made a world of difference when I found that commented-out mmx line
05:29.31cioI do have an echo with my X100P's, but I just contributed that to crappy wiring as my phone-to-phone calls, even remote, were great with no echoing.
05:29.50FuriousGeorgecio: how come u dont upgrade to 1.0.9
05:30.24argos73hmm - my puppy is insane... :)
05:30.39cioThe debian package is just so easy to install and it's stable.  Plus I've heard (read?) issues about 1.0.9.
05:30.40FuriousGeorgecio: i find that jitter is most often caused by a) bandwidth or b) the sip/client's computer being too busy
05:31.16*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:31.17FuriousGeorgeim amazed that opening IE will audibly affect call quality on a barton 2800 with a gig of ram running eyebeam on xp
05:31.35cioThe nature of x86 computing ...
05:31.45argos73heh - just starting xp has the same effect... :)
05:31.45*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
05:32.17cioI just got a dual-core p4 box, it's lightning fast, but XP kills it.  I'm tempted to install windows 3.1 just to feel like I have a fast computer.
05:32.37argos73i have a copy of windows 2 if you really want to feel special
05:33.30cioheh..
05:33.38ciolater all -
05:33.39ciotire
05:33.40ciod
05:33.45argos73good luck - night
05:33.54cio:)
05:36.28FuriousGeorgehere's another one.  when i use eyebeam's xfer feature to send a call from the pstn back out to the pstn the sound quality is terrible
05:38.45websaeanyone here know why the heck in MySQL when I try to start it...it starts........then turns off....
05:38.46websaeStarting mysqld daemon with databases from /var/lib/mysql
05:38.46websaeSTOPPING server from pid file /var/lib/mysql/localhost.localdomain.pid
05:38.50websaeany ideas...anyone?
05:38.59websaetrying to get MySQL to use with asterisk
05:39.06websaewould appreciaate any insight/help
05:40.03FuriousGeorgewebsae: is that the output from an initscript
05:40.23websaefrom trying to run the daemon
05:40.39*** join/#asterisk lehel (n=asd@82.79.20.17)
05:40.41lehelhello
05:41.29lehelwhy is that?: manager.c:468 authenticate: xxx.xxx.xxx.xxx failed to pass IP ACL as 'admin'
05:44.13Vcowell....tdmoe is cool....
05:44.21Vcoit's an absolute bandwidth whoar
05:44.26Vcobut it's cool
05:44.38Vcoerr..
05:44.44Vcowhore
05:44.46Vcohmm.
05:46.15blitzragewhore!
05:47.52*** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net)
05:48.00jayk-i keep getting this messages on my asterisk console
05:48.00jayk-<PROTECTED>
05:48.09jayk-they happen periodically..is this normal?
05:48.26wunderkinomg its going to blow
05:48.36Vcostop playing with it
05:49.16jayk-haven't touched it
05:49.25websaeanyone here good with MySQL?
05:49.42websaei am curious..trying to get it to work with asterisk...let alone get it to start (as in a service)
05:50.53wunderkinjayk-: yes..
05:51.07jayk-how come the channels restart? will it cause hangups?
05:51.43mydssmojoCan you guys recommend GUIs?
05:52.54*** join/#asterisk normal1 (i=IipsjLhN@ip70-181-165-140.sd.sd.cox.net)
05:53.52blitzragejayk-: from my experience, no, it doesn't drop channels
05:54.03Juggiejayk-, it happens peridiocally yes
05:54.06Juggieno it doesnt drop calls
05:54.16blitzragejayk-: not knowing for sure (no T1's here) I think it just happens periodically... my guess is to resync the channels?
05:55.11zapahi alll i alredy resolve my trouble with e1 telrad the troueble was outgoing tone provisoning
05:55.35zapathanks for all
05:55.56zapain telrad pri30 card
05:57.20jayk-okie doke
05:58.39*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
06:01.19*** join/#asterisk alkalineX (n=alkaline@CN-ESR1-69-61-204-172.fuse.net)
06:03.08blitzrageanyone know what the format for tcpIpApp.sntp.gmtOffset in the Polycom config file is?
06:04.28Peggerwebsae can you get mysql to start
06:04.30*** join/#asterisk wasim_ (n=wasim@pdpc/supporter/active/wasim)
06:04.31blitzragehrmmmm... going to just try -5, but I keep seeing examples of -21500 or something :)
06:06.12*** join/#asterisk joelsolanki (i=joelsola@202.160.161.93)
06:06.49joelsolankiHello JerJer
06:09.01djin_ibI have a question regarding zaptel modules. I'm trying to setup an as minimal as possible CentOS config and having problems modprobing zaptel.
06:09.23djin_ib/lib/modules/2.6.9-11.EL/misc/zaptel.ko is there but "modprobe zaptel" gives "FATAL: Module zaptel not found."
06:09.43djin_ibAm i overlooking something (hopefully simpel :)
06:10.00*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
06:10.03djin_ibI mean simple (simpel = Dutch :)
06:10.31pashahdjun_ib: depmod -a ?
06:11.20djin_ibThat doesn't say or do anything?
06:11.38*** join/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net)
06:11.55Dr_Rayfxs modules from digium can take the rxgain/txgain argument in zapata.conf right?
06:12.12blitzragepretty sure
06:12.32blitzrageman... I really gotta stop building GUIs for Asterisk and start using Asterisk again
06:12.38djin_ibMmm, perhaps I have to edit modules.conf manually?
06:12.43pashahdkin_ib: does not say, but generates modules.dep
06:13.14djin_ibMmm, here is doesn't.
06:13.51djin_iboh, wait.
06:14.31djin_ibI'm sorry, it does :)
06:18.18djin_ibNo zaptel, though :(
06:19.39mydssmojoanybody here from city of Calgary?
06:20.43flendersdoes anyone know a good VoIP provider in the UK?
06:21.14pashahflenders: gradwell
06:22.08flenderspashah: cheers :o)
06:28.27pashahlater all
06:32.13*** join/#asterisk clive- (n=pirch@ndn-165-131-217.telkomadsl.co.za)
06:32.53*** join/#asterisk KaBewM (n=kabewm@66-215-7-137.dhcp.psdn.ca.charter.com)
06:33.36*** join/#asterisk gordonjcp (n=gordonjc@cpc3-broo2-5-0-cust232.renf.cable.ntl.com)
06:34.44*** join/#asterisk zedkatuf (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk)
06:38.25*** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net)
06:40.00joelsolankiHi shido6
06:43.57shido6hello
06:44.31*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)
06:49.18wasim_mmmToop always reminds me of that hansen song ...
06:49.48mmmToopthanks guys ; )
06:51.19joelsolankiShido6: i m still unable to setup asterisk+ser on same machine
06:51.36clive-howzit toop
06:51.51joelsolankiShido6: can u look in to my config ?
06:51.52mmmToopall good & u?
06:52.44shido6whats up?
06:53.29clive-lekker thanks
06:53.52clive-waiting for telkom to install some lines...for a change
06:54.02mmmToopus to...
06:54.34mmmToopwhere do you purchase your Digium cards etc.?
06:54.57clive-i bought one from digium and one from jerjer
06:55.17*** join/#asterisk ikey (i=ikey@202.54.37.184)
06:55.33mmmToopoh...we have been buying stuff from miro & get the feeling that they are overpriced
06:55.45clive-you can get them from stee in cape town
06:55.51clive-steve
06:55.59clive-connection-telecom
06:56.41mmmToopah....on their site now
06:57.13mmmToopu got any PRI cards lying around that we can use/ buy?
06:57.22clive-i do actually
06:57.26clive-e100p
06:58.14mmmToophas that guy got echo cancelation/
06:58.15mmmToop?
06:58.40*** join/#asterisk lenne_dk (n=mirc@83.72.129.7.ip.tele2adsl.dk)
06:58.51*** join/#asterisk _omer (i=p@203.215.180.250)
06:59.07clive-i decided to do bri's instead
06:59.10_omerhow to allow the specific IP Addresses in the asterisk?
06:59.28clive-omer host=
06:59.33*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
06:59.52_omerin the sip.conf [general]
06:59.55_omer??
07:00.18Hengomer: In each client.
07:00.24Hengcontext
07:00.39_omerbut my client doesnt register to my asterisk...he sends call directly....
07:00.59Hengomer: oh. I use firewall for that
07:02.08_omerI dont have firewall :(
07:02.24Hengpls look at Shorewall
07:03.46_omerokey
07:03.49Dr_Rayor even smoothwall
07:04.22lenne_dkI want to announce incoming calls over the speaker. I can make calls to the console/dsp, and it autoanswers. But when a call comes in, I don't want to connect it to the speaker, I just want to send a soundfile to the speaker, and the real phone extention should keep ringing
07:05.05_omerDr_Ray: cant I do it in asterisk? allow IPs ..etc
07:05.28shido6ok
07:05.28shido6back
07:05.30shido6what sup?
07:05.34*** join/#asterisk heka (n=heka@80.80.174.140)
07:06.04Dr_Rayyou could also use firestarter
07:06.25Dr_Rayfirestarter would run on an asterisk box
07:06.28shido6hrmm
07:06.35shido6so use festival :)
07:06.37shido6lenne_dk:
07:07.02wasim_lenne_dk: chan_local
07:07.37_omerDr_Ray: ok..thnx
07:07.42lenne_dkA litle more clue, wasim
07:13.35lenne_dkwasim, what do you mean? if I do a exten => 111,1,Dial(${CONSOLE}), the incoming call is connected to the console.
07:13.42flendersguys, if I have a bunch of DIDs, each of them beeing used to direct dial an extension, is there a way to configure it so when the extension dials out, the corresponding DID is shown on the caller ID on the callee's device?
07:17.33flendersalso, just to let you know, I'm not forwarding port UDP/5060 to my * server (I'm using a SIP account)
07:17.37*** join/#asterisk akrall (i=user@201.144.58.10)
07:18.37akrallGuys, whats the function of ztdummy and timing source for meetme, moh, etc? if you use x100p clones, do you still need ztdummy or by loading the wcfco driver are you covered?
07:19.53*** join/#asterisk newl (n=newlook@203-59-91-217.dyn.iinet.net.au)
07:20.07ronaldl79I figured out why my virtual DID wasn't passing DTMF to * -- The dtmf was set to 'inband,' instead of 'rfc2833'. Works like a charm now!
07:20.12djin_ibakrall, you covered by teh wcfo driver.
07:24.34*** join/#asterisk shanky (n=shanky@85.137.127.142)
07:24.37*** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it)
07:25.07shankyhi, good morning
07:25.48lenne_dkso how to use the ztdummy.ko under FreeBSD?
07:26.56lenne_dk-bash-2.05b# kldload ztdummy.ko
07:26.56lenne_dkkldload: can't load ztdummy.ko: No such file or directory
07:26.56lenne_dk-
07:27.06zzzirkanyone have any tips on getting a reasonable understanding of the various codecs available?
07:27.17shankydo you if is there any specific channel for asteriskt@home ?
07:28.14lenne_dkdo what, zzzirk
07:28.17lenne_dk?
07:28.37lenne_dkSorry, not zzzirk, shanky
07:28.38zzzirkI just want to understand the differences, strenghts, weaknesses of the various codecs
07:28.41zzzirkah, okay
07:28.42zzzirkheh
07:28.42shankyhttp://www.voip-info.org/tiki-index.php?page=Asterisk%20codecs
07:29.25shankythat's for zzzirk
07:31.12wasim_lenne_dk: you can fork in * dialplan using chan_local ...    Dial(Zap/1&LOCAL/s@AlsoDoThis&LOCAL/s@AndThisToo)
07:31.53*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
07:33.19shankywell, I have install an asteriskt@home 1.5, I have add 2 extensions and I can call between them
07:33.50wasim_shanky: welcome to hell
07:34.02shankybut now I'm trying to setup a trunk to my voip provider
07:34.11ronaldl79Does anyone know the link to an asterisk alarm app???
07:34.15*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
07:34.23zzzirkthanks for the pointer shanky
07:34.45shankyI have already added the trunk, and reconfigure an outbound routing
07:35.14lenne_dkronald: I have my app drop a .call file, it starts an agi, telling me the temp of the fridge
07:35.44ronaldl79Pretty wild, lenne
07:35.53ronaldl79* might even be able to turn on the oven for ya
07:35.55ronaldl79lol
07:35.59ronaldl79It's so damn extensible, it's wild.
07:36.10ronaldl79I wonder if anyone's tied in any X10 stuff...
07:37.56lenne_dkI'm using the 1-wire bus for temperature sensors, rrdtools for storing the data, and big brother for monitoring and generating alarms.
07:38.24lenne_dkSo if the fridge is left open for too long, I get a call.
07:38.29shankylenne_dk: nice job
07:40.53shankywell, after I setup the trunk and the outbound routing, I try to do a external call an I get this:
07:41.11shanky09:38 < lenne_dk> I'm using the 1-wire bus for temperature sensors, rrdtools for storing the data, and big
07:41.13shanky<PROTECTED>
07:41.15shankyups
07:41.18shanky09:38 < lenne_dk> So if the fridge is left open for too long, I get a call.
07:41.28shankysorry
07:41.32Dr_Raydallas semiconductor?
07:41.42shankyCapabilities: us - 0x8 (alaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x8 (alaw)
07:41.46shankyNon-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
07:41.49shankyLooking for 952486557 in from-internal
07:41.51shankyReliably Transmitting (NAT):
07:41.54shankySIP/2.0 484 Address Incomplete
07:42.06*** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com)
07:42.08lenne_dkDr_Ray: Yes
07:42.15Dr_Raykewl
07:44.07lenne_dk+4536949142 extension 79 or 36949142@musimi.dk extension 79
07:44.43shankyany idea what I'm doing wrong?
07:45.37shankywhat I don't understand is why asterisk is looking for 952486557 in the context "from-internal"
07:46.57shido6yeah
07:47.11shido6from-internal doesnt have that many digits or the amount of digits is off
07:47.14shido6for the context, shanky
07:48.55shankyyeah, I have added "952486557" in the Dial Patterns and now it works
07:48.56_omerhow to do "SIP Proxy Authentication" in * ?
07:51.20lenne_dkIs it possible to ignore that a module can not be loaded? My soundcard sometimes(?) need a powerdown to be detected after boot, a software reset is not enough.
07:51.41lenne_dkSo chan_oss sometimes can't be loaded and asterisk will not start.
07:58.06lenne_dkone solution is to put * in a wrapper script. First start *, then have 'if [ -r /dev/dsp ] then asterisk -r -x "load chan_oss.so" fi'
08:02.20akralldjin_ib: thx!
08:02.25*** join/#asterisk tobiasWolf (n=konversa@195.162.255.10)
08:02.44akrallanybody using a sipura spa841?
08:06.24*** join/#asterisk Ilya (n=Ilya@mail.tex.kiev.ua)
08:06.43Ilyahi
08:07.39*** part/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com)
08:08.04*** join/#asterisk jeh_work (n=jeh@ext122.almare.com)
08:08.12jeh_workgood morning
08:08.26*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
08:08.28IlyaSipClient: Receiving message...
08:08.32IlyaSipClient: Received: 11:09:04.018
08:08.36Ilya---------------------------------
08:08.41IlyaSIP/2.0 401 Unauthorized
08:08.44IlyaVia: SIP/2.0/UDP 10.0.55.1:5062;branch=z9hG4bK214FDD9B
08:08.48IlyaFrom: "ilya" <sip:200@ilya>
08:08.52IlyaTo: "ilya" <sip:200@ilya>;tag=as63decd7a
08:08.57IlyaCall-ID: 1700781529@10.0.55.1
08:09.00IlyaCSeq: 6163 REGISTER
08:09.05IlyaUser-Agent: Asterisk PBX
08:09.08IlyaAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
08:09.09IlyaContact: <sip:200@10.0.55.1>
08:09.09IlyaWWW-Authenticate: Digest realm="ilya", nonce="388437cf"
08:09.09IlyaContent-Length: 0
08:09.10Ilyai'm trying to connect to newly installed asterisk with Kphone.. i get this
08:09.27*** join/#asterisk rat1101 (n=vinay@ip68-100-31-133.dc.dc.cox.net)
08:10.57jeh_workcan anyone point me to some docs as to what the difference would be when i try to redirect a call using AMI (Redirect action) and doing it on a real plastic phone using #xxxx?
08:11.13jeh_workthe former never works while the latter works
08:11.14Ilya[general]
08:11.18Ilyacontext=default
08:11.22Ilyarealm=ilya
08:11.26Ilyaport=5060
08:11.30Ilyabindaddr=10.0.55.1
08:11.34Ilya[200]
08:11.37*** join/#asterisk gordonjcp (n=gordonjc@lodge.glasgownet.com)
08:11.38Ilyatype=friend
08:11.42Ilyausername=200
08:11.46Ilyasecret=secret
08:11.50Ilyanat=no
08:11.50Ilyacontext=office
08:11.50Ilyacallerid="ilya" <222>
08:11.51Ilyahost=dynamic
08:11.51Ilyain sip.conf i have:
08:13.36jeh_workdoing the redirect using the real phone gives a voice message asking for the new extension to transfer to. so apparently the "#" puts asterisk in some kind of state of mind that the Redirect action in code can't do
08:13.58*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
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08:15.20KeX_WorXhi
08:15.33KeX_WorXany1 users srtp in asterisk ?
08:15.41KeX_WorXor knows anything bout it ?
08:18.27*** join/#asterisk musical_Duck (n=kvirc@wblv-146-224-229.telkomadsl.co.za)
08:20.10musical_DuckCan any1 help me with iax2?  I can't dail my clients register initial but then I get lost of failed md5 auth messages
08:22.04musical_DuckMy clients register initially but then I get lots of failed md5 auth messages, soz late night
08:23.46*** part/#asterisk musical_Duck (n=kvirc@wblv-146-224-229.telkomadsl.co.za)
08:25.37*** join/#asterisk musical_Duck (n=kvirc@wblv-146-224-229.telkomadsl.co.za)
08:26.09musical_DuckAny1 here?
08:27.37*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
08:31.11*** join/#asterisk oej (n=Olle@apollo.webway.se)
08:34.29*** join/#asterisk Thoran (n=Thoran@p54A5A6A9.dip0.t-ipconnect.de)
08:35.21ThoranMorning every1!
08:36.18avizionmorning Thoran
08:36.21avizion;)
08:36.56musical_Ducklo
08:37.30musical_DuckSay any of you guys use iax clients?
08:37.58pauldydamb getting ready for bed here sun already up on that side of the ppond huh
08:38.16avizionnot much... but once in a while I use FireFly
08:38.51musical_DuckMy clients register initialy but I get lots of failed md5 auth messages
08:39.25musical_DuckAnd I can't seem to dial them with Dial IAX2/clientbob wich used to word
08:39.31musical_Duckwhich rather
08:39.40musical_Duckand work
08:39.41musical_Duck:)
08:43.53*** join/#asterisk ErMeS`wrk (n=ermsewrk@217.220.121.62)
08:46.27*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
08:48.52*** part/#asterisk akrall (i=user@201.144.58.10)
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09:09.51*** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
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09:22.47*** part/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
09:25.16jeh_worki see asterisk logging this: "dialparties.agi: Dial return value was -1 and dialstring was SIP/1002|120|tr"
09:25.24*** join/#asterisk ful|work (n=fulgas@213.58.130.46)
09:25.32jeh_workhow can i figure out why the dialing failed?
09:26.04JamesDotComby debugging the agi script
09:26.07jeh_workthis is then done using AMI redirect. when done using a plastic ugly phone the return value is 0 and the redirect proceeds
09:26.34jeh_workugh, it seems to be some kind of line noise
09:28.16*** join/#asterisk RoyK (n=roy@80.239.107.80)
09:28.40*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
09:29.26jeh_workah, i had agi debuggin disabled. i hope it prints me some more clues when it's enabled
09:33.40X-Robjeh_work - you probably don't have Asterisk::AGI installed.
09:34.00jeh_workX-Rob: i should as it works for hardware phones
09:34.01X-Robat your shell prompt, type /var/lib/asterisk/agi-bin/dialparties.agi
09:34.05X-Robit'll tell you what's not working
09:34.33jeh_workit just freezes
09:34.39X-Robpush enter a few times.
09:34.40jeh_workwaiting for args maybe?
09:34.48X-Robbut, ok, that means it's not a dialparties problem
09:34.58X-Rob..also, I just read what you said about it working sometimes.
09:35.22jeh_workit always works when used with # from a real phone
09:35.43jeh_workbut when  try to do the same using AMI it never works (a Redirect action)
09:36.13jeh_workso i guess pressing # on the phone does something that Redirect can't mimic
09:37.01jeh_workjust to answer a call Redirect works ok, but if i want to further Redirect that answered call it fails. but works when done with #
09:37.27X-Robok
09:37.29X-Robthe word is 'transfer'
09:37.40X-Robor is it not?
09:40.38jeh_workyes, but in AMI it's down with Redirect?
09:41.03X-Robjbot: ami?
09:41.08X-Rob~ami
09:41.30jeh_workasterisk management interface
09:41.49X-Robok. So what does that have to do with transfers?
09:42.21RoyKjeh_work: management interface? as in the so-called Manager API?
09:42.28jeh_workyes
09:43.01jeh_worki use that interface to connect to asterisk and then in with my application control it
09:43.02RoyKhas someone finally renamed it?
09:43.20RoyKjeh_work: what app was that?
09:43.36jeh_workRoyK: an inhouse app
09:43.56jeh_workRoyK: built in java using the asterisk-java bindings
09:44.03RoyKok
09:44.26jeh_worki get access to all events i need, all channels, queues etc
09:44.58jeh_worki can answer calls, origiate calls, but a transfer/redirect of an answered call doesn't work
09:45.09Ilyado i need a sound card on a server to run asterisk?
09:45.27Dr_Rayno, unless you want sound on the console (probably not)
09:46.28Ilyais p-pro200(no MXX)/192mb RAM enough to run asterisk?
09:46.57jeh_worklunch time
09:47.19Ilya)
09:49.38X-RobIlya - it -might- be. But I wouldn't put money on it. Jump onto ebay with $50 and buy a Piii500 or something.
09:49.47*** join/#asterisk jaxkz (n=cyrieldo@tbnb-165-193-112.telkomadsl.co.za)
09:49.54jaxkz'n morning
09:50.50IlyaX-Rob: i just have this machine running freebsd
09:51.12jaxkzDoes anyone use the Eurorings Network for voip Termination?
09:51.40X-RobIlya - so? Get a Piii 500, run Centos 4.1 on it.
09:51.55X-Robyou won't have _any_ problems.
09:53.53*** join/#asterisk Jzalae (n=sk@216-220-248-44.midmaine.com)
09:55.09ennuyeux72i have a question about RTP timing in asterisk
09:55.34ennuyeux72am i correct in saying that for every RTP packet that asterisk receives it will send one back in response
09:56.13ennuyeux72i am getting a scenario where tethereal trace shows multiple RTP packets being received by asterisk from a phone
09:56.29ennuyeux72without anything being sent back
09:56.36ennuyeux72till a later time
09:56.46ennuyeux72where multiple RTP packets are sent back
09:57.07ennuyeux72i was under the impression that there was some sort of one to one deal going on with the RTP  packets
10:13.55*** join/#asterisk cjk (n=cjk@80.92.64.103)
10:15.37X-Robennuyeux72 - that is incorrect.
10:15.39*** join/#asterisk bsd3 (n=bsd@203.134.194.176)
10:15.55X-Robvoicemail is a prime example of this. When asterisk is recording a voicemail message, it is _only_ receiving, not sending RTP at all.
10:21.17*** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it)
10:21.31bsd3hi, friends!
10:23.42bsd3is the refresh in iax2 and sip is like a keep-alive?
10:24.06*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
10:24.36FABRIZIOxxxhello all .. i set up an asterisk box with 2 tdm04b (8 fsx lines) .. the problem is that sometimes from the pstn network people occasionally hear a choppy sound.. how can i solve this?
10:25.18lehelX-Rob, i gotta a prob with recorded vm messages, it is recorded well, but when i want to play/download from the web-interface cannot becouse the owner of the *.wav is asterisk, not www-data
10:27.18bsd3lehel: i have not tested it, but you need to use an agi script to do that and NOT a cgi
10:29.02bsd3how do i change refresh times for an IAX and, or a SIP channel?
10:29.03jaxkzDo the wildcards support ATM?
10:31.03*** join/#asterisk N4SH (n=Bernardo@c-67-180-105-69.hsd1.ca.comcast.net)
10:31.14N4SHi got a question
10:31.33X-RobHow can I help you N4SH 8)
10:31.54lehelbsd3: do you know an agi script for it?
10:31.58N4SHx-rob tnx. is asterisk capable of doing IP telephony/VOIP?
10:32.04X-RobN4SH - heh.
10:32.07X-RobYes. Yes it is
10:32.07lehel;p
10:32.27bsd3lehel: sorry, no dear
10:32.38N4SHx-rob oh wow. i don tknow how to use asteisk is there a websiet for dummmies?
10:32.58*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
10:32.59X-RobN4SH - try Asterisk@Home
10:33.01X-Robgoogle for it.
10:33.03lehelbsd3: deaR?!:P
10:33.10N4SHok thanks..
10:33.56N4SHwhere is it best to rn asterisk?
10:34.02N4SHrun
10:34.08X-Rob...on a linux machine?
10:34.13N4SHmepis?
10:34.33X-RobCentOS 4.1/WhiteBox/RHEL is the 'best' distro, because that's what all the devels use.
10:34.43X-Robbut, really, there's not that much difference. Use a 2.6 kernel.
10:35.16N4SHoh cool... tnx. let me read the @home stuff
10:35.28X-RobN4SH -
10:35.29X-Rob~docs
10:35.31jbotmethinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
10:35.38X-Rob^^^ ses them.
10:35.40X-Rob^^^ see them
10:35.41X-Robeven
10:35.49N4SHok tnx
10:36.09X-Robjbot: wiizard is an anal wart
10:36.11jbotX-Rob: okay
10:36.18X-Rob*snigger*
10:37.15frenzyhello
10:37.24frenzyHow do I enable transcoding ?
10:38.03jaxkzthat will be done automaticly i think
10:38.12bsd3N4SH: if you look at the asterisk source, building tools like Makefile's etc does not show any such dependency
10:39.17frenzyI'm getting a very jittery connection
10:39.44frenzyWhen connecting dirclty to the provider using ilbc I get a clear line
10:39.55bsd3N4SH: i'm running 1.2.0-beta1 on a Knoppix machine kernel version 2.6.12
10:40.20*** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk)
10:40.25frenzywhen setting * to use ulaw and my ata on ilbc the audio is jittery
10:40.26N4SHtnx for the advices... =)
10:40.47*** join/#asterisk ^X-works (n=r0x0r@host34-3.pool871.interbusiness.it)
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11:00.00geinIm having problems with my queues, calls keeps getting placed in the queue even if there aren't any agents logged in, and I
11:00.04gein<PROTECTED>
11:00.32Cresl1ntoo early
11:00.37geinrunning v1.0.9 stable
11:01.46geinif the queue doesn't have any members, then I can't join.. seems like strict option doesn't really work in my case?
11:04.40*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
11:07.10*** part/#asterisk bsd3 (n=bsd@203.134.194.176)
11:07.11*** join/#asterisk [jedi] (n=hhgds4@213.162.200.226)
11:14.22[jedi]I get 'chan_iax2.c:3052 iax2_trunk_queue: Maximum trunk data space exceeded to MY_IP:4659' when calling throught IAX between two asterisk servers
11:14.27[jedi]SIP works great
11:15.56*** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
11:16.03geinno one who uses joinempty=strict in queues.conf and that it is actually working?
11:18.42RoyKwhat does that do?
11:19.27[jedi]what does that mean?
11:19.50[jedi]iax2 is saying continuously these messages when calling between IAX servers
11:20.10[jedi]and also voice only goes from calling to caller
11:20.14[jedi]but not from caller to calling
11:20.27clive-jedi I got that too, sop I swicthed off trunking
11:20.48*** join/#asterisk jeh_work (n=jeh@ext122.almare.com)
11:20.53[jedi]trunking?
11:21.01[jedi]should be activated, shouldn't
11:21.01[jedi]?
11:21.16geinRoyK: do you mean joinempty=strict?
11:21.16clive-i think its a bug
11:21.29jeh_workstupid asterisk can't log why it thinks my Dial() command must fail
11:21.40RoyKgein: yes
11:21.56geinit make calls not queued if there aren't any agents logged in which is a member to that queue
11:22.09jeh_workit's so nice to just be given a "-1" and nothing in the log and no error message anywhere
11:22.31jeh_worknot even with verbosity and debugging at 100 and agi debug on
11:23.21shankyhey, I'm getting this error:     -- Got SIP response 488 "Not Acceptable Here" back from (my voip provider)
11:23.49shankyit's suppose to be a codec problem, but I have try with allow=all and it doesn't work
11:24.08geinI just want the feature that if no member of a queue has logged in, then calls should not be placed in that queue, also, already queued calls should exit the queue
11:25.52*** join/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au)
11:26.03jeh_workdebugging perl in itself is a horrible fate, debugging perl without any meaningful logging is worse
11:28.08jeh_workwhat is the purpose of AGI and the external perl stuff anyway?
11:28.17mostyis it possible to allow particular codecs only when both src and dest of a call are SIP channels?
11:28.43jeh_workdialparties.agi even has to do manager connections back to asterisk to get some info. that's not too smart
11:30.16[jedi]jeh_work: perl isn't the only AGI choice you have
11:30.18[jedi]jeh_work
11:30.43jeh_work[jedi]: sure, but i'm not writing new stuff, i'm debugging why my transfers die
11:31.23jeh_work[jedi]: and that means diving into the code that does the dialing
11:33.18*** join/#asterisk knobo (n=knobo@217.77.34.124)
11:36.55frenzySep 28 07:25:58 NOTICE[20598]: chan_sip.c:6704 handle_response: Peer '7777777' is now TOO LAGGED!
11:36.56frenzySep 28 07:26:09 NOTICE[20598]: chan_sip.c:6698 handle_response: Peer '7777777' is now REACHABLE!
11:36.56frenzySep 28 07:27:13 NOTICE[20598]: chan_sip.c:8085 sip_poke_noanswer: Peer '7777777' is now UNREACHABLE!
11:36.56frenzySep 28 07:27:52 NOTICE[20598]: chan_sip.c:6698 handle_response: Peer '7777777' is now REACHABLE!
11:37.11frenzyhow do I go about tackling this ?
11:37.48mostyfrenzy: reduce the load on the network between the client and the server?
11:38.22frenzythe server is remotly located (US)
11:38.31*** join/#asterisk phpboy (n=shane@c1-114-12.tbnb.isadsl.co.za)
11:38.44frenzywhile the ATA is behind a satellite connection
11:38.51frenzyin a different country
11:38.56phpboyto install a working CentOS for asetrisk... do I have to download all 4 CD's?
11:39.06mostyfrenzy, 2-way satellite?
11:39.11frenzyyap
11:39.54mostyfrenzy: perhaps satellite just has too high a latency. are you able to make calls, and if so how much delay is there?
11:40.13frenzyyes
11:40.21frenzyabout 800 - 1000ms
11:40.33mostyoh yuck :)
11:41.02frenzyconsidering the ATA is in Africa
11:41.09frenzyI wouldnt say the quality is bad
11:41.51phpboyfrenzy: where in africa?
11:42.20frenzysouth
11:42.32phpboycool, I'm from South Africa!
11:42.33phpboyaskhd
11:42.35Drukenfrenzy: just take the qualify off
11:42.51frenzymy bad the 800 - 1000ms is when doing an echo
11:42.59shankyphpboy: you can try wit asterisk@home
11:43.07frenzyso most probably termination routes would be shorter
11:43.42frenzyremoving qualify wont cause the ATA to get dissconnected ?
11:43.56Drukenuhmm.... no...
11:44.07frenzyI do face that at times... it disconnects and then after couple seconds reconnects
11:44.48phpboyshanky: I need to get a Junghanns Quad  ISDN card going on it... will it be possible?
11:45.04frenzyjust for your info my qualify=2000 :)
11:45.50Drukenqualify=0
11:46.09shankyphpboy: take a look at asteriskathome.sourceforge.net
11:48.11frenzyalso get Sep 28 07:48:06 WARNING[20598]: chan_sip.c:701 retrans_pkt: Maximum retries exceeded on call 66aa1986c5024e4e@192.168.1.101 for seqno 29700 (Non-critical Response)
11:48.13frenzysometimes...
11:48.14phpboyok, I'm prolly gonna need to build it in myself
11:48.29phpboydoes anybody have the Junghanns Quad ISDN card?
11:49.56Drukenya know it's funny, all those commercials talk about the poorness in africa, yet we get people in here doing voip, and looking for massive routes and shit...
11:49.58Drukenwhich is it?
11:50.43*** join/#asterisk iDunno (n=brettp@mike.catnip.org.uk)
11:51.02wasimDruken: both, its the .0001% that make it here
11:51.23Drukenoh..
11:51.51iDunnocan someone point me at some documentation that tells me about attended transfers?
11:51.59wasimiDunno: features.conf
11:52.08Druken~voipinfo
11:52.19Druken~voip-info
11:52.21jbot[voip-info] the Voice Over IP wiki.  It is a community resource which will answer all of your questions, from Asterisk to ZTDummy.  You can find it over at http://www.voip-info.org - well worth bookmarking
11:52.22iDunnobeen there, I appear to be missing something obvious.
11:53.31iDunnoassume that I've got kphone as the client, and that the call has come in via sip, I want to do an attended transfer to another sip phone, and using DTMF appears to be failing me, and using the transfer button also appears to be failing me.
11:56.10phpboyguys, most of africa is a fuckup
11:56.24phpboybut South Africa is pretty well off
11:58.39geincan someone tell me how exactly joinempty in queues.conf works or where I can read about it?
11:59.49Drukeni'm not sure what it means, i've always assumed it means people can join the queue even without any agents to answer their call
12:01.18geinand what is meant by "without any agents to answer their call"? I can't figure out if they mean that there are no agents assigned as members in queues.conf or if they mean agents who's not logged in
12:02.41geinwhen joinempty=strict that is
12:02.55*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
12:03.01dudes_gein - I'm not sure what he means, nor do I know what it means. But I've always found the asterisk source code to be helpful with certain questions
12:03.58geinI'm not that good at intepreting source code
12:04.20geinI'm experiencing the same problem as the author of this feature-request: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=3754
12:04.28mostycan you allow/disallow codecs from the dialplan?
12:04.57dudes_mosty - I don't think so.
12:05.15geinAnd they closed that ticket, saying: 'not an issue'...huh?
12:05.30ThoranAny1 willing to help a newbie debug his asterisk conf. I can't make calls from my sip phones to the PSTN
12:05.39mostydudes: is there a way i can allow certain codecs for sip->sip calls, but not for sip->zap calls?
12:05.58mostythoran: what connection do you have to the pstn?
12:05.59dudes_sip.conf and zapata.conf ?
12:06.23RoyKcansandstring.conf
12:06.53mostydudes: i want to allow ulaw for sip->sip calls, but if i put allow=ulaw in sip.conf then won't that allow ulaw for sip->zap calls?
12:07.03wasimmosty: SetGroup
12:07.26Thoranmosty: ISDN
12:07.27*** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net)
12:07.34dudes_gein - Bug Reported before option added.
12:08.09geinyeah, but then this should have been solved? And I can't even get it to work with CVS-HEAD
12:08.46Drukenis it me, or is more shit broken in stable then in head?
12:09.00dudes_Are you using head?
12:09.15geinI've tried both stable and head
12:09.29geinno luck at all... joinempty=strict seems to work as joinempty=no
12:09.34dudes_Druken - A lot of docs are written about head, not stable.
12:10.08mostywasim, i don't understand what setgroup does
12:11.27mostyThoran, and you've setup your dialplan to dial out on a zap channel for pstn calls?
12:11.45Drukenmostly: just don't allow ulaw on the zap channels
12:12.02Drukenwhy you don't want ulaw to your zaps i don't know... but that should fix it
12:12.04dudes_mosty - what are you trying to do
12:12.38*** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com)
12:12.40mostydruken: but won't that just make asterisk transcode ulaw to some other codec? i don't want to allow ulaw for sip calls unless it's sip->sip with asterisk just doing the signalling
12:13.13mostydudes: i want to allow ulaw for sip->sip calls (where asterisk is just doing the signalling), but not allow ulaw for any other situation
12:13.14dudes_mosty - zap uses slin which is similiar to ulaw in the fact that it's raw data
12:13.14RoyKanyone here that knows how I can use an analog link to send SMS?
12:13.18Drukenmosty: probably... i highly doubt you can do what you want
12:13.21RoyKcan i do that with asterisk?
12:13.25*** join/#asterisk klictel (n=klictel@207.107.208.140)
12:13.38klictelmorning all
12:14.16Drukenw00t! costers!
12:14.24Drukener.. coasters
12:14.29mostydudes: i don't want incoming network data to use anything but compressed codecs, say gsm or g729, but i don't mind if sip->sip calls use other codecs since asterisk isn't in the media path
12:14.59Thoran@mosty yes
12:15.18Thoranexten => _8.,1,Dial(Zap/g1/${EXTEN:1}/,60)
12:15.18Thoranexten => _8.,2,Congestion
12:15.18Thoranexten => _8.,3,Busy
12:15.18Thoranexten => _8.,4,Hangup
12:15.29mostythoran: and what error do you get in the asterisk logs when you make such a call?
12:15.46dudes_Then allow g729/gsm on all context that are outgoing
12:15.49Thoransometimes I get cannot forward voice
12:15.50dudes_or incoming
12:16.06Thoranlike some codec problem
12:16.15dudes_But that's only the first few bytes normally
12:16.23Thorancan I paste my logfile here ?
12:16.30mostythoran: no, use a paste site
12:16.37*** join/#asterisk gambolputty (n=gambolpu@72.240.241.108)
12:17.54mostydudes: a context in the dialplan?
12:18.43dudes_Set your codecs you want to use in the contexts
12:19.10dudes_ie, sip.conf context for your incoming/outbound providers to disallow=all allow=g729
12:19.17Thoranjust a sec
12:19.23dudes_then allow your sip phones to allow=all
12:19.24Thoranasterisk acting up atm
12:20.50mostydudes: what is an incoming sip provider? is that the same as a phone?
12:20.54*** join/#asterisk djin_ib (n=djin_ib@217-19-19-229.dsl.cambrium.nl)
12:26.41RoyKanyone here that knows how I can send SMS over an analog line with a modem or perhaps a digium board?
12:27.24mostyroyk: i don't think it's possible over POTS lines
12:27.36RoyKmosty: i beleive it is...
12:28.33mostyroyk: i'm pretty sure in australia you can't, i don't know about the rest of the world
12:30.28dudes_iDunno - enable it and try it out
12:31.26iDunnodudes_: I, erm, did. if I hit *2 with the DTMF pad, it appears to just throw the DTMF through to the other end, if I use the transfer button on kphone, it appears to do a blind transfer :/
12:34.46dudes_so you see how it works
12:37.14iDunno... erm, not at all?! ;)
12:37.51*** join/#asterisk siggi (n=siggi@wlan-gw.denic.de)
12:39.55dudes_it allows you to answer the call
12:40.12*** join/#asterisk docE (n=docE@66.237.242.41.ptr.us.xo.net)
12:40.22dudes_then if you hangup before the called party hangs up it's connected to the party ou were talking to
12:40.49iDunnoerm, that's how I expect it to work, that's not what's happening...
12:41.12docEAnyone in here work on the h323 addons module for asterisk?    I am having issues with it crashing and taking down asterisk
12:41.14dudes_so you hit *2 ... hit the extensions
12:41.29iDunnodudes_: that appears to be the bit that doesn't work.
12:41.36NuggetI think I'll call him tonight at 03:30 netherlands time.
12:41.55dudes_So what does it do
12:42.07*** join/#asterisk arnis (n=arnizzz@82.148.188.56)
12:42.43arnisSo have anyone had success with cmd Conference on Gentoo and asterisk 1.0.9?
12:42.51iDunnodudes_: the *2 just gets sent to the other phone, it doesn't get picked up by asterisk or anything else
12:42.58iDunnodudes_: as real tones.
12:43.30dudes_Are you the called party or the calling party
12:43.46iDunnoboth, in this case. but for the *2 I'm the called.
12:43.50arnisMy asterisk just hangs when a second caller tried to enter the conference.
12:44.00arnistries*
12:44.12iDunnoand the Dial has got a tr in it.
12:44.20dudes_arnis - never had that.  Get some debug info
12:44.49dudes_Maybe your DTMF's are getting send right
12:45.41iDunnooh bother.
12:46.05*** join/#asterisk djin (n=djin_ib@84-245-25-231.dsl.cambrium.nl)
12:47.28iDunnonope, that still wasn't quite right.
12:49.41arnisWhat do you mean by Debug information?
12:50.00arnisthe information i get in asterisk -vvvvvgr?
12:50.49arnisI get loads of notices and a few warnings on unable to translate from unkown to unkown/alaw
12:52.04dudes_iDunno - erm, I don't know then.  What does your sip context look like?
12:52.26dudes_arnis - probably an unknow codec that can't be translated
12:52.47dudes_show translation on the CLI will show you what you can translate from to
12:55.27arnisk. thanks. My can't translate anything from g723 or g729. Guess i have to set my phones to g726 then?
12:55.57dudesarnis - that'd be about the case.  Or buy the g729 license
12:56.29dudesOr if your provide supports g729 and your phone you can passthru
12:57.32arnisI can call and recieve just find with the current codecsettings so it's wierd if that should be the problem. Don't you think?
12:59.08Thoranone of the guys who wanted to help me still here ? I found out what a paste site is :)
12:59.37arnisok. but thanks anyway. i have to log off to reach my phones. (stupid wireless think).
12:59.55*** join/#asterisk cuco (n=elcuco@local.xorcom.com)
13:00.06iDunnodudes: http://share.runtime-collective.com/~brettp/sip-context.conf
13:00.09cucodoes anyone has a FWD account?
13:00.10mostythoran: like http://pastebin.ca/
13:00.11iDunnoit looks a little like that
13:00.21tzafrirhi, anybody here with a FWD account (preferebly IAX)?
13:00.34cucotzafrir: me :)
13:00.42mostythoran: or http://rafb.net/paste/
13:00.45tzafriroops, ignore (note IPs of cuco and tzafrir)
13:02.44KeX_WorXanyone knows if there is a deb package for asterisk-1.2-beta1 ?
13:03.14cucoKeX_WorX: not yet, that I know of
13:04.54*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
13:05.18KeX_WorXcuco, k. u know if ther is one in dev or planing ?
13:06.33*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
13:07.51*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:07.57*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
13:07.59tzafrirI hope that this is a matter of less than a week
13:09.04RoyKcan someone help me understand how SMS is transported over isdn/analog line? using data/modem/spandsp, or how does this work?
13:09.30*** join/#asterisk arnis (n=arnizzz@82.148.188.56)
13:09.39docEAnyone in here work on the asterisk-ooh323 addons package for asterisk?   I am having some issues
13:09.47arnisnope. didn't change anything.
13:10.20arnisalso tested with  G711a 64k  on my phone..
13:11.00*** join/#asterisk pussfeller (n=todd@12.150.129.171)
13:13.14Thoranmosty: http://pastebin.com/376763
13:13.35KeX_WorXtzafrir, u mean the deb package, with less than a week ?
13:14.30klictelyou have 8 Xs?
13:16.31klictelThoran: the number showing in dial has 8 digits?
13:17.23*** join/#asterisk coppice (n=chatzill@246.204.17.210.dyn.pacific.net.hk)
13:17.28*** join/#asterisk psycodad (n=obiwan@2001:4060:4419:b1:0:0:0:2)
13:18.04*** join/#asterisk ^X-works (n=r0x0r@81-208-62-98.ip.fastwebnet.it)
13:18.23*** join/#asterisk Abbas (n=Abbas@203.81.196.92)
13:18.54Thorani am not sure what you mean
13:19.18ThoranI dial 8
13:19.21Thoranthen the number
13:19.39ThoranThe XXXXs are the number i dialed minus the 8
13:19.56Thoranin this case the number has 8 digits yes
13:22.01*** join/#asterisk riemensc (i=riemensc@83-169-155-130-dynip.superkabel.de)
13:22.09riemenschello
13:24.41arnisvoip*CLI> show conferences
13:24.41arnisDeprecated! Please use 'meetme' instead.
13:24.48arnisbah
13:26.59Kattyyay, morningstarfarms has 'chicken' and 'steak' strips!
13:27.10kajtzuarnis: yes, you want "meetme list"
13:27.52tzangerhow do you get a vegan *chicken* enchilada?
13:28.25coppicemost chickens are vegans
13:28.32iDunnovegan!=vegetarian
13:28.47kajtzu"A vegan diet is one which excludes any animal products."
13:28.49*** join/#asterisk Mike9 (n=sturdee@ireland.pathwaynet.com)
13:28.56kajtzuiDunno: try using google define:vegan
13:29.00sivanaA vegan diet is one which excludes any animal products.
13:29.10sivanaheh
13:29.20*** join/#asterisk pussfeller (n=todd@12.150.129.171)
13:29.20MikeJ[Laptop]what about sproutists
13:29.20iDunnoah.
13:29.23iDunnohmm
13:29.26sivanaas long as they pet the chicken first, before they kill it, than it's ok
13:29.40gordonjcpyeah
13:29.50coppicethat means they can't east veggies, because those are made from animal droppings :-\
13:29.52kajtzu;-)
13:29.54sivanahehe
13:29.54gordonjcpI wondered about that, surely having chicken in precludes it being vegan
13:30.00kajtzugordonjcp: me too ;)
13:30.02Kattyyou know, if you're going to tease people
13:30.09Kattyfind something worth teasing them about (=
13:30.13sivana:)
13:30.15kajtzu.. make sure they get it first? ;)
13:30.23Kattykajtzu: grow up ;)
13:30.31gordonjcpKatty: not teasing, just wondering how you can have a vegan chicken *anything*
13:30.32Kattytzanger: it's soy, silly rabbit.
13:30.37kajtzuKatty: nah
13:30.45Kattygordonjcp: it's not /real/ chicken. it's a soy based substitute.
13:30.51kajtzuKatty: growing up is for people iwth no imagination ;)
13:30.52Kattygordonjcp: duh :P
13:30.52gordonjcpkajtzu: surely that would be a tofu enchilada then?
13:31.02Thoran@mosty: Do you happen to have an idea, what I did wrong ?
13:31.04tzangerKatty: ok but if it's soy then what makes it chicken?  Or do you mean chicken-flavoured?
13:31.07*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@corail-gw.clients.easynet.fr)
13:31.17coppiceI think all those Buddist fak meat dishes are deeply against the spirit of being a vegitarian
13:31.18gordonjcphang on, why would you make the tofu chicken-flavoured?
13:31.19Kattytzanger: it looks (and simulates) a non breaded chicken strip.
13:31.24gordonjcpcoppice: indeed
13:31.26PoWeRKiLLhi coppice
13:31.27tzangerKatty: ah okay
13:31.37PoWeRKiLLI have a core file to give you
13:31.44gordonjcpKatty: why, though?  surely if you don't want to eat meat, you wouldn't want to eat meat-flavoured things?
13:31.53Kattygordonjcp: there IS NO MEAT IN IT
13:31.56Kattygordonjcp: THE PRODUCT IS VEGAN
13:31.58Kattygordonjcp: kthx.
13:32.04gordonjcpKatty: yes, but it's meat-flavoured
13:32.06coppicePoWeRKiLL any core files are due to someone else's code
13:32.09Kattygordonjcp: it is not.
13:32.17Kattygordonjcp: it would not be vegan if there was meat flavoring.
13:32.18gordonjcpKatty: it's chicken flavoured
13:32.25Kattygordonjcp: what part of no meat don't you get? :P
13:32.37gordonjcpKatty: what part of my question don't you get?
13:32.42Kattygordonjcp: apparently all of it.
13:32.43gordonjcpit tastes like chicken, right?
13:32.48Kattyno
13:32.51Kattyit simply simulates it
13:32.57PoWeRKiLLcoppice when a specific fax machine send me a fax it's crash asterisk
13:33.19Kattygordonjcp: vegans are againts the unethical treatment of animals. not the taste.
13:33.24MikeJ[Laptop]it tastes like chicken, not tastes of chicken
13:33.25Thoranok I found it
13:33.26Kattywhat a moron ;)
13:33.29sivananow we have engineered chicken meat
13:33.38Kattysivana: they've had it for years.
13:33.41sivanahrm.. cancer not on the rise
13:33.45Thoranthere was a / at a place where there shouldn't be one
13:33.52coppicePoWeRKiLL: can't you just look at it with GDB and tell me where it died?
13:33.59Kattyanyway!
13:34.00gordonjcpKatty: mmm, that sounds a bit spurious to me
13:34.02sivanahehe
13:34.06MikeJ[Laptop]mmmm gotta love that soy protine :D
13:34.23Kattyi'm plotting using these chicken soy strips to make enchiladas
13:34.29gordonjcpif I was a vegan, I wouldn't eat stuff that tasted like meat products
13:34.36MikeJ[Laptop]they are good... and I eat chicken
13:34.53Kattygordonjcp: that's your call then.
13:35.00gordonjcpKatty: yes, true
13:35.04Kattygordonjcp: i have no personal problem with it. they taste good. they're not harming animals in the process.
13:35.08MikeJ[Laptop]if your vegan, you need protine from somewhere.. simulated meat products are a very easy way to do it
13:35.11gordonjcpI am also against the unethical treatment of animals
13:35.20coppicethe fake meat in the Buddist monestaries can be a pretty good fake
13:35.22MikeJ[Laptop]except chickens :D
13:35.37MikeJ[Laptop]cuz they are mean anyways
13:35.45MikeJ[Laptop]I'm not
13:35.52sivaname either :)
13:35.52MikeJ[Laptop]I am being fecious...
13:35.56gordonjcpKatty: I don't know about anyone else, but I'm not picking on you for being vegan
13:36.01coppiceI'm against the unethical treatment of vegetables
13:36.01tzangerfecious?  haha
13:36.02Kattyexcellent.
13:36.04MikeJ[Laptop]I have a veggy wife
13:36.05Kattythen stop teasing me :P
13:36.07tzangerfacetious maybe?
13:36.13gordonjcpI really just don't get why people who don't want to eat animals want to eat things made up to taste like animals
13:36.14MikeJ[Laptop]sure
13:36.17PoWeRKiLLcoppice yes it's die in spandsp
13:36.20MikeJ[Laptop]I need a spell checker
13:36.25PoWeRKiLLgordonjcp where are you from ?
13:36.29coppiceMikeJ: severe brain damage? :=)
13:36.41Kattygordonjcp: if you dont' want it. don't eat it.
13:36.50gordonjcpPoWeRKiLL: Scotland
13:36.51Kattygordonjcp: doesn't mean you have to go harping on people who do
13:36.53Kattygordonjcp: don't be a snob
13:36.57coppicePoWeRKiLL: can you be a little more specific? like the file and line number?
13:37.09gordonjcpKatty: I only eat animals that I know personally
13:37.21Kattyi think we should just move on (=
13:37.23KattyNEXT
13:37.31kajtzuwhat I'm not getting is how a vegan anything can be chicken anything no matter what
13:37.47gordonjcpkajtzu: it tastes a bit like chicken, apparently
13:37.49Kattyi give up.
13:37.59kajtzugordonjcp: how can it, not being chicken ;)
13:38.04coppicei'm over 80% vegetarian. its only my mouth which eats meat
13:38.07kajtzugordonjcp: at the very least it is deceptive advertising ;)
13:38.09gordonjcpcoppice: lol
13:38.09sivanaok.. ok.. stop.. if vegans want to eat fake, man-engineered meat, then who cares
13:38.13*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
13:38.19KattyMikeJ[Laptop]: does your wife have any really good recipes?
13:38.27kajtzuanyway... vegatarian stuff is what food eats >:)
13:38.30sivanaoops.. not politically correct
13:38.35sivanahuman-engineered
13:38.41[jedi]coppice: I'm trying to find out a bit more about the problem I'm having with spandsp + txfax... what would you need from my bug report so that it's useful?
13:38.49*** join/#asterisk mutilator (n=animenod@65.111.201.79)
13:38.59sivana[jedi]: what problem you having?
13:39.07coppicejedi: a patch which fixes the problem :-)
13:39.16[jedi]sivana: txfax gets executed but does nothing
13:39.17PoWeRKiLLcoppice #0  0x4227b54b in process_baud () from /usr/lib/libspandsp.so.0
13:39.19KattyMikeJ[Laptop]: because if she does, you must commander them.
13:39.30[jedi]coppice: I wish I was able to do that :D
13:39.45PoWeRKiLL#9  0x0807be5e in pbx_extension_helper (c=0x8e5bbe0,
13:39.45PoWeRKiLL<PROTECTED>
13:39.45PoWeRKiLL<PROTECTED>
13:39.45PoWeRKiLL<PROTECTED>
13:39.45PoWeRKiLL#10 0x08075faa in ast_pbx_run (c=0x42253bb4) at pbx.c:1769
13:39.46PoWeRKiLL#11 0x0807c521 in pbx_thread (data=0x0) at pbx.c:1992
13:39.48PoWeRKiLL#12 0x4002aa21 in pthread_start_thread () from /lib/i686/libpthread.so.0
13:40.10coppicePoWeRKiLL: which version of spandsp?
13:40.19KattyMikeJ[Laptop]: i mean commandeer.
13:40.39coppicejedi: I suppose you did use the "caller" parameter?
13:40.52[jedi]I'm using 0.0.2pre20, with a CVS asterisk from july 13+-
13:41.04[jedi]coppice: of course
13:41.13*** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
13:41.15[jedi]coppice: in fact, when I dial my own phone, I can hear the "beep"
13:41.37[jedi]but when I dial a real fax machine it doesn't do anything, nor it shows anything in any log or in the screen, even with debug parameter
13:42.08gordonjcpKatty: I've got some semi-decent recipes, which I will chuck in your direction as soon as I get time to write them up
13:42.14[jedi]I've tried three different fax machines to discard a problem with a concrete fax
13:42.27coppicePoWeRKiLL: what happened to backtrace entries 1 to 8?
13:42.36Kattygordonjcp: i see.
13:42.40Kattygordonjcp: k
13:42.49gordonjcpKatty: you'd like the miso soup one
13:42.54*** join/#asterisk Moc (n=mochouin@modemcable173.101-70-69.mc.videotron.ca)
13:43.01coppiceeveryone likes miso
13:43.02Kattygordonjcp: i'm not a big soup person, but sure.
13:43.03Mocmorning
13:43.05[jedi]what's miso?
13:43.09Kattymister Moc (=
13:43.27coppicemiso == key staple of the japanese diet
13:43.33[jedi]oh
13:43.39Kattyrice?
13:44.37gordonjcp[jedi]: it's a kind of bean curd paste
13:44.38*** join/#asterisk kahuna_ (n=booger@209-254-56-194.ip.mcleodusa.net)
13:44.52kahuna_Hi
13:44.55[jedi]ok
13:45.10*** join/#asterisk spackle (n=spackle@209.234.83.19)
13:46.19[jedi]coppice: have you ever heard of a problem like mine with txfax? or i'm the only one ?
13:47.02coppicetxfax generally works, although people have been complaining lately of a high corrupt pages issue
13:47.03Connor_When you guys setup you linux box for VoIP applications, how do you all setup the partitions..
13:47.23[jedi]I wish I had page corruption, at least
13:47.27[jedi]:)
13:47.44spackle"Cheese Grommit."
13:47.47*** join/#asterisk Corydon76-home (i=mauve@pdpc/supporter/sustaining/Corydon76-home)
13:49.18Kattygordonjcp: k'then
13:49.46*** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr)
13:49.49jhiverhi all
13:50.13jhiverI've *finally* finished 'frenchizing' and 'cleaning' up astcc, wohoo :)
13:50.22jhiverit was like, a big day's work :)
13:50.23*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
13:50.31[jedi]jhiver: I gave up on that months ago
13:50.35zoid99what kind of "cleanup" did you do?
13:50.37[jedi]jhiver: developed my own calling card app
13:50.43jhiveruse strict; :)
13:51.08zoid99cool
13:51.23jhiverI've also done quite a few mods so that it tells the max. lenght of call
13:51.32jhiverand it speak like, french, now :)
13:51.47djinYou do not appear to have the sources for the 2.6.9-11.ELsmp kernel installed (while compiling zaptel). I installed the kernel sources (2.6.9-11.EL) and made the links /usr/src/linux and /usr/src/linux-2.6 pointing to this source. What am I overlooking?
13:51.55jhiveraudiocity came in handy but I have to do more work because my mike is shit and my voice ain't any better
13:51.59zoid99one step forward.. 2 steps backwards :)
13:52.05kajtzudjin: missing kernel-devel-smp
13:52.10kajtzu(or something)
13:52.32kajtzukernel-smp-devel-2.6.9-11.EL.x86_64
13:52.39kajtzu(or whatever your arch is)
13:53.09djinI have kernel-devel, but no kernel-smp-devel.
13:53.12jhiverbut it's cool! I don't know if it's faster starting from scratch, but considering it took one day to install Xorcom + AstCC + play around with it, and another day to customize it, i'd say 'no'
13:53.25[jedi]zaptel works well with x86_64 machines?
13:53.27kajtzudjin: you have a smp kernel, you need the kernel-smp-devel package
13:53.28djinkajtzu, but it's downloading ;)
13:53.35kajtzu[jedi]: yes
13:53.37kajtzudjin: check ;)
13:53.38[jedi]great
13:53.47[jedi]jhiver: astcc is quite limited
13:53.59kajtzudjin:  you don't need kernel-devel thjougu
13:54.00kajtzuthough
13:54.08jhiveryeah I realize that
13:54.12jhiverbut hey, it's a start
13:55.04*** join/#asterisk Laerte (n=io@195.47.232.200)
13:55.05Laertehy
13:55.07[jedi]most calling-card providers do many tricks with the pricing of the cards, which can't be done with astcc
13:55.22djinkajtzu, very cool. thanks :)
13:55.28kajtzudjin: np
13:55.28jhiveryeah sure like disconnect charges and other horrible stuff ;)
13:55.48[jedi]hehe
13:55.49[jedi]yes
13:55.58[jedi]I had to develop a bunch of these tricks in my app
13:56.17[jedi]I really hated it but not my choice
13:56.30jhiverI don't quite get what 'brand' is all about with AstCC
13:56.58jhiverThere's a few things I'm not familiar with
13:57.02[jedi]brand is a product
13:57.11[jedi]you may have different products, different cards
13:57.12*** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com)
13:57.38jhivermhhh
13:57.50[jedi]you may apply different tariffs to each product
13:57.56jhiverso 'brand name' I get that...
13:58.04jhiverthen 'language' that seems fair enough...
13:58.13jhiver'published number' and 'did' WTF?
13:58.22jhiverInc = Increment I would guess
13:58.30jhiverService Fee = ???
13:58.33[jedi]I don't really know what uses these numbers astcc for
13:58.38jhiverService Fee Days = ??? again
13:58.42[jedi]but all these are tricks
13:58.49jhiverand markup = ??? markup of what?
13:58.53[jedi]markup on pricing
13:59.02[jedi]you put a pricing for each route
13:59.14[jedi]then you put a markup on these pricing
13:59.24[jedi]astcc tariffication model is too simple
13:59.30jhiverOK so this is for some kind of reseller malarki
13:59.40[jedi]or at least it was last time I checked
13:59.56[jedi]most calling card distributors will want many features astcc hasn't
14:00.03[jedi]most of them want a clone of digitalk :D
14:00.15jhiveris this your product?
14:00.22[jedi]digitalk ? no hehe
14:00.46[jedi]digitalk is almost a 'standard' platform for calling cards
14:00.51jhiverah ok
14:00.58[jedi]is a switching platform which has calling card facilities
14:01.11mutilatorholy fscking nuts
14:01.15coppicewhere did that PoWeRKiLL fellow go with the other half of his bug report? :-\
14:01.30jhiverok and I guess that costs a mint as well doesn't it
14:01.32mutilatormy brother just called my cellphone from his cell, i said hello, then all the sudden it got cut off and i was talking to someone else
14:01.33clive-I have digitalk, its a great product
14:01.44clive-I have used digitalk, i mean
14:01.49mutilatorso i called my brother back, and he said the same, he got cut off and someone else was on the line
14:01.52[jedi]it's quite cryptic
14:01.55[jedi]but it's powerful, yes
14:02.03mutilatorlike the telco just crossed the lines or some weird thing
14:02.07mutilatorwas weird
14:02.21jhiverclive-, what's so cool about it?
14:02.52clive-jhiver it just has every feature in the callingcard book
14:02.52mutilatorwonder how that happened
14:03.06clive-i probably costs a fortune as well
14:03.11clive-it*
14:03.12[jedi]jhiver: it can do anything. It's just quite difficult to manage
14:03.17clive-jeez my typing today
14:03.21ThoranAbbas ?
14:03.29wasimmutilator: its very common with mobilink in pk
14:03.48jhiverok, so it's basically what Oracle is to RDBMS, some monster elephant that does it all but actually works
14:04.03mutilatorhe uses cingular and i have verizon wireless
14:04.11mutilatornever even heard of that happenin before
14:04.14[jedi]jhiver: almost hehe
14:04.24[jedi]jhiver: oracle is not as pricey as digitalk
14:04.37mutilatorhe got connected to someone looking for parts at a store, and i got someone trying to reach some number down in detroit
14:05.37jhiveryou guys have tried other CC apps?
14:05.47jhiverother than AstCC that is?
14:06.41spackle~seen bkw_
14:06.45jbotbkw_ is currently on #asterisk (19h 13m 32s).  Has said a total of 85 messages.  Is idling for 11h 3m 53s
14:06.47clive-jhiver astcc is just a perl script, you can easly just add in any feature you want
14:07.02coppice~seen redder86
14:07.03jbotredder86 <n=lee@gateway.howardsilvan.com> was last seen on IRC in channel #asterisk, 22h 48m 23s ago, saying: 'c'mon down!'.
14:07.06jhiverclive-, I know i've done that today :)
14:07.06bkw_yes
14:08.42jhiverclive-, I've done a few mods to AstCC it's quite easy to work with although I would have preferred a nice OO perl code :)
14:09.01*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
14:09.04[jedi]mine is done on java
14:09.16jhivermhhh interesting
14:09.31coppicejedi: cheap indonesian labour?
14:09.38jhiveris there a page somewhere about it or it kept inhouse?
14:09.42[jedi]coppice: cheap my-hands labour
14:10.01zoid99jedi:  Did you use the asterisk-java suite?
14:10.04[jedi]zoid99: yes
14:10.17skyflexdudes: vad gör du då?
14:10.19zoid99very nice framework
14:10.21skyflexwho, wrong chan
14:10.23[jedi]yes, very powerful
14:10.24*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:10.24*** mode/#asterisk [+o anthm] by ChanServ
14:10.32*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfldr.dialup.mindspring.com)
14:10.36*** part/#asterisk Moc (n=mochouin@modemcable173.101-70-69.mc.videotron.ca)
14:10.51zoid99we've used Asterisk-Java on a few projects
14:11.09[jedi]I tried JAGI but its non-threaded approach was not as easy to manage as asterisk-java's
14:11.13zoid99only on the manager side.. not agi
14:11.34RoyKyagi antenna
14:11.35RoyK:P
14:11.45[jedi]FastAGI + Hibernate + Spring == powerful robust well-architected Agi in a breeze
14:11.45kahuna_I'm getting this: PRI Error: We think we're the CPE, but they think they're the CPE too.
14:12.05*** join/#asterisk wsuff (n=wsuff@pcp04243496pcs.eatntn01.nj.comcast.net)
14:12.08zoid99What is spring?
14:12.09kahuna_When I switch signalling to pri_net I get the same except switch thinks theyre net too...
14:12.17[jedi]zoid99: spring framework... java stuff
14:12.18kahuna_How do I fix that?
14:12.40[jedi]zoid99: www.springframework.org
14:12.44zoid99Hybernate is DB persistence... is Spring for UI?
14:12.48*** part/#asterisk wsuff (n=wsuff@pcp04243496pcs.eatntn01.nj.comcast.net)
14:12.48[jedi]no
14:13.28[jedi]spring is a framework which adds Aspect-oriented features, Inversion of Control, declarative transactions, all without messing with your own code
14:14.03[jedi]it does also transparent remoting throught many protocols and many other things, but I don't use these features right now
14:14.44zoid99nice..
14:14.59*** join/#asterisk g__ (n=goakham@itd01fw-fibre.itdepartment.com)
14:15.43kahuna_maybe my provider is running an interface loop?
14:17.11*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
14:18.16L|NUXevening all
14:22.04*** join/#asterisk zobia (i=zobia@222.212.71.126)
14:22.51zobiahello everyone , i want to wait () how many sec equal to 4 rings ? i want to let it ring 4 times and transfer a inbound call
14:23.12cpatryzobia: take a watch and seee
14:24.24zobiacpatry. do u know what's difference between wait() and ring?
14:25.07cpatryno i dont. are they both animals?
14:26.48RoyKanyone using zaphfc with pci-s?
14:27.36*** join/#asterisk file (n=jcolp@mctnnbsa31w-142166116178.nb.aliant.net)
14:27.36*** join/#asterisk santiago (n=santiago@63.245.86.245)
14:27.36*** join/#asterisk alerios (n=alerios@200.24.109.199)
14:28.59ThoranI am searching for SIP Phones which support VLAN Tags. I have found Grandstream GXP 2000, some cisco phones, and snom 320 and 360. Does any1 know other phones which support VLAN ? Preferably for a price under 200 Eu.
14:29.06*** join/#asterisk lubomier (i=lubomier@217.118.109.179)
14:29.15*** join/#asterisk azzie (i=az@cpe-24-168-17-173.si.res.rr.com)
14:29.59lubomierhi, it's here anybody who configuers avm fritz!pci 2.0 in NT mode? or point-to-point?
14:30.08_m_Thoran: the snom 190 supports VLAN, too
14:30.24*** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it)
14:30.39Thoranok thx
14:30.39*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
14:31.11FABRIZIOxxxhello .. i was just wondering which is the best echo cancellation method for the zaptel driver ..? mar, mark2, mark, steve or steve2 .. anyone have some experience??
14:31.54*** join/#asterisk Mc_Tr (n=Mc_Tr@bacterio.knet.es)
14:32.04Mc_Trhi!
14:32.05jake1932FABRIZIOxxx: kb1 is the current
14:32.11Mc_Trhas anybody running asterisk?
14:32.19Mc_Tri need to do one question.
14:32.22jake1932no - try the asterisk channel
14:32.31FABRIZIOxxxoh ok .. so do you mean i have to use the zaptel 1,2 driver?
14:32.52jake1932FABRIZIOxxx: the newest echo can is kb1
14:32.58Mc_Trknow how i do a Phone Book in asterisk?
14:32.58jaxkzMc_Tr: just as the question
14:33.34Mc_Trjaxkz: do you know i can have a centralized phone book?
14:33.35jake1932http://lists.digium.com/pipermail/asterisk-users/2005-August/122944.html
14:34.12jaxkzMc_Tr: i do not
14:34.26Mc_Trok, thanks jaxkz
14:34.28jaxkzI think you need an ldap implementation for that
14:35.08Mc_Tri'm searching in voip-info.org and google, but nothing happens :(
14:35.42*** part/#asterisk santiago (n=santiago@63.245.86.245)
14:38.16*** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg)
14:38.54*** join/#asterisk e3g (i=p@203.215.180.250)
14:38.56e3ghi
14:39.09*** join/#asterisk folsson (n=filip@h82n1fls32o985.telia.com)
14:39.13e3gI have a question about Asterisk Manager.
14:39.53*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@corail-gw.clients.easynet.fr)
14:39.59e3gI want my manager to receive only specific Events, currently he receives all the events.
14:40.05PoWeRKiLLcoppice still here ?
14:40.13PoWeRKiLLcoppice sorry i got a internet disconnection
14:40.25PoWeRKiLLcoppice you want a bt full by mail ?
14:41.06coppiceyep
14:41.19*** join/#asterisk DagMoller (n=DagMolle@2001:5c0:8fff:ffff:0:0:0:41)
14:42.23jake1932e3g:  All connected terminals will receive all "Events" that happen on the Asterisk box
14:42.29DagMolleralguem fala portugues?
14:42.42RoyKjeg snakker ikke portugisik....
14:42.53ful|workDagMoller: diz
14:43.38*** join/#asterisk oej (n=Olle@apollo.webway.se)
14:44.47RoyKtelco switch monkey says "you cannot send SMS on a PRI"
14:44.55RoyKoej: god afton
14:45.06oejGod afton
14:45.10mostyroyk: snakk du spansk?
14:45.39RoyKnei
14:46.04mostyi just think "spansk" sounds funny, heh
14:46.04jaxkzAnyone know a reliable voip network?
14:46.11*** part/#asterisk lubomier (i=lubomier@217.118.109.179)
14:48.09kahuna_how do I turn off musiconhold. I keep getting unable to spawn mp3player and  Found no files in '/usr/share/asterisk/mohmp3'
14:48.35*** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
14:48.45mostykahuna: just put some mp3 files in that directory
14:50.40iDunnomosty: that's not how to turn it off! that's how to stop the second message ;)
14:50.49geinno one into queues, who can help me out with my configuration?
14:51.38mostyif you want to turn music on hold off completely, i think you can turn off that module in the modules.conf file
14:51.49*** join/#asterisk Moc (n=mochouin@modemcable111.229-203-24.mc.videotron.ca)
14:53.08iDunnonoload => res_musiconhold.so
14:53.12iDunno(at a guess!)
14:53.24iDunno(that could be wildly inaccurate and eat your cat, or something)
14:54.42tzangerha
14:54.59kahuna_I already ate my cat!
14:55.47*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
14:56.14jake1932was it tasty?
14:57.19*** join/#asterisk fugitivo (n=ajf@201.255.106.137)
14:57.25spackletastes like chicken?
14:57.26fugitivohello
14:57.35geinhej hej
14:57.39Kattyhmm.
14:57.46kahuna_it's the other, other white meat!
14:58.29Ariel_Hope your morning is going well
14:58.37Kattyyes. new vegan food products :>
14:58.43Ariel_nice
14:59.09Kattyi'm plotting something like chicken enchiladas (=
15:00.01geinmorning? bah!
15:01.04azziewho knows which cisco 79xx phones support SIP and work well ?
15:03.09Nugget7940 and 7960 both use the exact same firmware.  the only functional difference is the number of lines.
15:03.55Nuggetthe sip firmware for the cisco phones is not as nice as the call manager firmware, but it's functional.  it's also a total pain in the ass to buy.
15:04.25jake1932azzie: but if you get the phone with SIP firmware already loaded - life is much easier
15:04.25spackleand install it would seem
15:04.29*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
15:04.39bendy24would probably be better with a polycom phone
15:04.43redder86coppice: hello
15:05.07bendy24i threw my cisco 7905g in the trash
15:05.11bendy24piece of crap
15:05.21coppiceredder86: did you see my e-mail?
15:05.26NuggetI like my 7960s, but I wouldn't recommend them.
15:05.38NuggetThey're expensive and a hassle to work with
15:05.44RoyKanyone using zaphfc with pci-s? please shout out loud
15:05.52bendy24Nugget: yep
15:05.53*** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au)
15:06.00redder86coppice: yes, I did.  What kind of modem do you have that does not echo "ATH0E0H0" when that same command is given to it and echo is on initially?
15:06.07*** part/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au)
15:06.32redder86coppice: I've tested USR Sportster, MultiTech MT5634, Conexant K56... all echo the full command
15:07.09*** join/#asterisk Chotaire (i=chotaire@chotaire.net)
15:07.21Chotairegood morning vietnam.
15:07.33RoyKanyone here that can help me sort out how an incoming SMS looks on a PRI (with PRI DEBUG)?
15:08.44*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
15:08.47filestabby stabby stabby
15:09.20MikeJ[Laptop]e-mail again?
15:09.27Mocmorning again
15:09.29MikeJ[Laptop]or Sip?
15:09.32tzangerI'd love to send the proper MWI message to a cell phone (i.e. my asterisk voicemail lights up a Telus cellphone MWI indicator with the right callback)
15:09.52MikeJ[Laptop]tzanger, oh would you now?
15:09.53MikeJ[Laptop]:P
15:09.58jake1932bendy24: did you really throw away your 7905? i'd buy it off you
15:10.14tzangerthere are companies who say they can do it but they want like $0.17/SMS sent
15:10.28jake1932bendy24: unless it smells like garbage now
15:10.38bendy24jake1932: naw, just kidding, i'm frustrated as hell with it
15:10.48bendy24i *will* get it to work
15:10.57redder86coppice: I don't think that the command interpretation is supposed to occur until the end-of-line, yet the fact that echo is on will require that the full command be echoed.  As for case sensitivity, there's no problem toupper()ing everything internally.  I'll send you an updated patch.  As for invalidating At and aT... well, that seems rather silly no matter what the spec says.  The modems I've tested don't care about case... not sure why we sho
15:11.04kahuna_Nothing better than a 7905 covered in ranch dressing
15:11.11Beirdobendy24: smashy smashy?
15:11.11bendy24how true
15:11.16*** join/#asterisk }btorch{ (n=kvirc@208.63.19.172)
15:11.17bendy24Beirdo: please go ahead
15:11.20Beirdohehe
15:11.23bendy24:)
15:11.28BeirdoI'll bring over a baseball bat
15:11.29azzieNugget, why are they hussle to work with?
15:11.43}btorch{anyone here uses the call park (feature.conf) with X-Lite ?
15:11.48kahuna_I like how you have to "license" the firmware :(
15:11.51spackleredder86 - Old modems used to be upper-case.
15:12.07Nuggetloading the firmware is a pain, obtaining the firmware is an even larger pain.
15:12.17}btorch{I configured my extension.conf file as the instructions mention on voip-info.org but doesn't seem to work
15:12.18coppiceredder86: I just tried it with a PCtel and it seems to do what you said. I remember putting in complexity specifically to make it stop echoing at the E0 to match the modems I played with at that point
15:12.21redder86spackle: some older modems that I've tested converted everything to upper case, yes
15:12.30Nuggetand the documentation is poor
15:12.30azzieNugget, and besides loading firmware?
15:12.31kahuna_People should put the firmware up on bittorrent sites
15:12.33}btorch{any ideas ?
15:12.40brad_msswHmmhesays: around?
15:12.43Beirdokahuna_: hardly legal :)
15:12.51Kattybrad_mssw: i can call him if you want
15:12.59redder86coppice: what modems were those that acted differently?  They'd not be doing immediate echo, I don't think.
15:12.59NuggetIt took me two months to find someone who would sell me the support contract to obtain the sip firmware.
15:13.00kahuna_Yea.
15:13.01Beirdomorning, Katty
15:13.10kahuna_Firmware nazis are hardly ethical though.
15:13.18KattyBeirdo: hi (=
15:13.20coppiceredder86: the reason I specifically look for AT and at is to be compliant. not a common concept in modems, i know :-)
15:13.29Nuggetdon't be such an OSShole, kahuna_.
15:13.37}btorch{I have added the include =>parkedcalls right below my [defaults] and also right below mi [macro-stdexten]
15:13.42kahuna_lol
15:13.44Nuggetit's cisco's kit, they get to set the rules.
15:13.56Nuggetif you don't like it, buy a polycom.
15:13.58Beirdoeven when they suck greatly :)
15:14.08Hmmhesaysyeah
15:14.09Hmmhesayswhat up?
15:14.12kahuna_Butt nugget - I was just voicing my opinion.
15:14.19Nuggetsure, but it's a boneheaded one.  :)
15:14.20kahuna_Err, s/butt/but/
15:14.23jake1932maybe one day someone will write Open79XX?
15:14.24Beirdohehehe
15:14.35brad_msswHmmhesays: vonage says asterisk doesn't work with their business plus service
15:14.36azzieNugget, what would you recommend to have as a office phone ?
15:14.39Beirdojake1932: that's a lovely idea.  get me specs :)
15:14.40Beirdohehe
15:14.47brad_msswHmmhesays: they won't switch my company to it
15:14.47Kattybrad_mssw: then maybe it doesn't?
15:14.49jake1932i wish
15:14.55Hmmhesaysbrad_mssw, they lie
15:15.00NuggetPeople in here seem to like the polycoms and really hate the grandstreams, but I really couldn't say.
15:15.04brad_msswHmmhesays: I know ... its frustrating
15:15.08zobiaI got the answer wait(5) equal to 1 ring
15:15.11kahuna_I'd rather have something IAX based anyway. Pee on Vonage :)
15:15.11NuggetI haven't used enough of them to have solid opinions on the alternatives
15:15.15brad_msswHmmhesays: I have some lady on the phone that doesn't even know what SIP is
15:15.18azzieNugget, I agree that GS sucks :)
15:15.24Hmmhesaysbrad_mssw, give me a call and we'll hook you up
15:15.27brad_msswKatty: Hmmhesays uses the vonage business plus
15:15.36kahuna_I have a few grandstreams for testing. I would never give them to the users though.
15:15.42azzieNugget, thanks a lot
15:15.45brad_msswHmmhesays: do you know if I can switch phone numbers yet ?? we're currently with vonage
15:15.49kahuna_I'd spend waaay too much time supporting them
15:15.52[jedi]uhmmm
15:15.56Hmmhesaysyeah they are really dragging their ass on that
15:15.57brad_msswHmmhesays: and we'd need to keep the same numbers
15:16.02[jedi]TDMoE is able to carry a FAX signal ?
15:16.06redder86coppice: I guess one could say that modems that accept At and aT are broken, but my take on it is that the spec is broken.  :-)
15:16.08*** join/#asterisk wolfson (n=hehe@65.174.122.198)
15:16.08*** join/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca)
15:16.08Hmmhesaysthe only thing we can do right now is call forward the old numbers
15:16.08[jedi]or just voice?
15:16.20Hmmhesaysthey are really pissing me off in that area
15:16.21TripleFFF2sdfLOL bkw.. just saw astricon performance..
15:16.24[jedi](IAX2 with A-Law can't carry a FAX signal, right?)
15:16.26KattyHmmhesays: you sure are popular of late.
15:16.29kahuna_tdm doesn't get compressed so why not?
15:16.39Hmmhesaysme? why for?
15:16.51brad_msswHmmhesays: well there's only one number that needs to be forwarded since it's in a call hunt
15:16.54KattyHmmhesays: because people are all talking to you, etc.
15:17.16KattyHmmhesays: and drunk people following you around all night.
15:17.35Hmmhesaysbrad_mssw  yeah you'd have to keep one plan and call forward the number until portablility is completely in place
15:17.37tzangerKatty: becaue they smell tofu that smells like chicken?  <ducks>
15:17.38kfuqwould anyone happen to know what the MTU is on ATM ?
15:17.38HmmhesaysKatty: haha
15:17.40redder86coppice: I think that there's an "intent" or "spirit" behind the specification, and if the actual printing of that specification doesn't adequately describe the "intent" or "spirit", well, then the spec is broken.  That said, one must be able to accept behavior per the specification, however, invalidating other behaviors seems... well... pedantic.
15:17.55[jedi]kfuq: same as on IP I think...
15:17.57Hmmhesaysmaybe people find it odd I don't talk much tech in here
15:18.03Kattytzanger: down boy.
15:18.06brad_msswHmmhesays: pvt msging you ..
15:18.14kahuna_Tech is boring.
15:18.24[jedi]kfuq: or maybe not, provided ATM is circuit-based... don't know :)
15:18.28Kattydespite that fact, it's very popular, kahuna_
15:18.38kahuna_lol
15:18.52NuggetKatty mostly hugs in here.
15:18.56kahuna_I'd rather talk about whats for dinner.
15:18.59kfuq[jedi]: heh.. me either.. ima n00b with atm lol
15:19.00Kattyand demands recipes, Nugget
15:19.09KattyNugget: GIVE ME YOUR RECIPES
15:19.15Beirdohehe
15:19.29Nuggetwe made some amazing cheese and pepper stuffed pork tenderloins the other night.
15:19.38Kattyexcellent, post it.
15:19.44*** join/#asterisk royth1 (n=royth1@200.121.129.178)
15:20.35Kattyspeaking of such, i require a recipe that uses little strips of steak. sorta like fajita steak strips.
15:21.24coppiceredder86: what about my other points?
15:21.51bendy24Katty: i prefer bacon
15:21.54HmmhesaysKatty I sang karaoke last night
15:22.08Hmmhesaysit was fun
15:22.11KattyHmmhesays: did you sing a recipe?
15:22.13KattyHmmhesays: using steak strips?
15:22.19Hmmhesaysnope
15:22.23Kattysad.
15:22.24}btorch{is voip-info.org down or is it just me ?
15:22.29HmmhesaysI sang "she thinks my tractors sexy" and "bad touch"
15:22.34Katty..!
15:22.37KattyHmmhesays: what is wrong with you?!
15:22.48Hmmhesays?
15:22.54Kattycountry music >.<
15:22.55*** join/#asterisk toddf (n=toddf@net-66-210-104-252.theshop.net)
15:23.05Kattydon't make me smack you with this polycom handset!
15:23.19HmmhesaysKatty: i gotta sing like 6 country tunes in this band
15:23.30KattyHmmhesays: eww.
15:23.31bendy24chattahoochi!
15:23.31KattyHmmhesays: that's dirty.
15:23.35zobia}btorch{ i also can not access voip-info now
15:23.57Hmmhesaysyou'd have fun
15:24.01zobiaoh. now it's okay
15:24.01}btorch{ist back
15:24.08redder86coppice: the echo should be in the same case that it was supplied, I think.  However internally the command can be converted to uppercase.  No problem.  I'll send a new patch.  As for At and aT, I think that the modem needs to accept them as valid.  As for the OK response to AT, I think we're in agreement there.  Were there other points that I missed?
15:24.11KattyHmmhesays: fun?
15:24.13zobiayes. it's back now
15:24.22KattyHmmhesays: somehow, that doesn't seem to be my thing.
15:24.26Hmmhesayswatching me play country
15:24.33Kattywell, that might be amusing.
15:24.36Hmmhesaysi like to stand on tables
15:24.44Kattyyou would.
15:24.53Hmmhesaysyes, I would
15:24.58coppiceredder86: I think that is it
15:25.34KattyHmmhesays: you should record songs and post them like i do.
15:25.36anthmI think a pluggable logger would be nice too
15:25.44Kattyanthm: morning (=
15:25.50[jedi]FAX can be carried with TDMoE?
15:25.52anthmso instead of stderr you can hook up a func pointer
15:25.59anthmhi
15:26.01*** join/#asterisk mutilator (n=animenod@65.111.201.79)
15:26.09Kattyanthm: how is doggy?
15:26.18anthmannoying as ever
15:26.26Kattyannoying as wife?
15:26.37gordonjcpHmmhesays: you know what happens if you play country music backwards, don't you?
15:26.49Hmmhesaysyou get your dog back your girl back and your house back
15:26.56Kattyheh, that reminds me of the JWs and back tracking or whatever.
15:27.07Kattythey said it was DEMONIC
15:27.10gordonjcp... you get out of jail, you get your truck back...
15:27.11SwK[Work]Hmmhesays: you left out sobering up and gettingyour job back
15:27.14gordonjcpyeah, you heard it
15:27.19Kattythere were hidden demonic messages in these songs.
15:27.24Kattyand it was Very Bad to listen to such things
15:27.25anthmhe eats EVERYTHING
15:27.26Kattybad bad bad!
15:27.32Kattyanthm: gosh.
15:27.32Hmmhesaysi dont' play any twangy country, ain't going down by garth, she thinks my tractors sexy, the race is on by sawyer brown
15:27.34gordonjcpKatty: I played some Iron Maiden records backwards
15:27.42Hmmhesayscadillac ranch
15:27.51sivanagordonjcp: did you get a secret msg?
15:27.56SwK[Work]SATAN!
15:27.59anthmoh besides jalipinos *snicker*
15:28.02gordonjcpthey said "shlub yub veeerp yaab vip"
15:28.10redder86coppice: something also needs to be done about Caller*ID and DID information.  For example, the DTE needs to be able send a command to get the DCE to repeat that information.  If that information is not stored internally in spandsp, then spandsp will need to interact with the spandsp-using application much more (for nearly every AT command) to allow the application to provide that kind of information.
15:28.23gordonjcpthen the stylus broke and I had to go shopping for a new one
15:28.35NuggetKatty: http://www.marga.org/food/blog/archives/001738.html
15:28.46NuggetI suggest using two or three times the amount of cheese, though
15:29.14coppiceredder86: I think I handled caller ID in my channel driver. take a look at that
15:29.31gordonjcpNugget: that may not really be to Katty's taste
15:30.32redder86coppice: yeah, I'm handling it in IAXmodem itself, too.  But getting it to *repeat* that information by way of an AT-command is not possible without either putting that information into spandsp and without the application being able to participate in the AT command-response stuff.
15:31.16FABRIZIOxxxis it normal that asterisk does not answer pstn after a couple of hours that its working but only sip to sip works??
15:31.27redder86coppice: so my question is whether you would prefer to see this stuff internalized into the library or whether you would prefer to see interaction between spandsp and the application on every AT command?
15:31.38Kattygordonjcp: i told him to post it.
15:32.30*** join/#asterisk convey (n=test@66.55.43.2)
15:32.34gordonjcpKatty: ah, nm then
15:33.16coppiceredder86: I can't remember what I did. I'll take a look and get back to you. which AT commands do you expect to work for this? Most modems don't seem to support the standards
15:35.04filewhy won't you build statically
15:35.28redder86coppice: okay, thanks.  Yeah, every modem seems to have it's own set of additional commands... but the reason that is - is because the standard set of commands is inadequate.  For example, I'd like a command such as AT+VRID=n where it would enable and disable the displaying of Caller*ID and DID information and would also (depending on n) cause the DCE to repeat the received Caller*ID and DID for that call.
15:35.29bendy24file: i'd blame your gcc inculdes
15:35.40filekurg blah
15:36.33*** join/#asterisk makhtar (n=ageller@mail.bulletinnews.com)
15:38.40coppiceredder86: I think there is a caller ID command, but its somewhere obscure. you may have noticed I went through all the ITU and 3GPP specs, and put all the commands in the interpreter :-)
15:39.04kahuna_how can I disconnect from the asterisk console once I connect via asterisk -r
15:39.05redder86coppice: yeah, I saw a lot of commands there.  I haven't gone through them all.
15:39.33redder86coppice: I'll have another look and if I find something I'll work with that and send a patch.
15:40.08*** join/#asterisk pashah (n=pashah@ns.itconnection.ru)
15:40.16pashahhi all
15:40.20KattyNugget: margarita's stuff is all posh
15:40.32*** join/#asterisk Firestorm-voip (n=Firestor@mail.mysoft.se)
15:42.07pashahhas anyone an example of zaptel.conf to use te110p and tdm with 4 FXO in one * box,
15:42.22hardwirepashah: not that hard
15:42.47hardwiredefine the t1 span
15:42.48coppiceredder86: there is some truly horrible buffering in t31.c which defintely needs improving before it can be called complete
15:42.52hardwirethen define the fxo span
15:43.31redder86coppice: oh, I've been meaning to talk to you about buffers.  Can you elaborate?
15:43.31wunderkinand make sure the t1 is loaded first
15:43.34}btorch{do I need to create a parkedbcalls extenstion on extention.conf ?
15:43.43hardwireyup
15:43.45}btorch{to get thing to work ?
15:43.50pashahthanks
15:44.08coppiceredder86: It buffers a whole page, which was a very temporary measure
15:44.08hardwire}btorch{: sorry.. you need to include => parkedcalls
15:44.14hardwirein whatever context the phone you are using is in
15:44.20redder86coppice: ah, that may explain some things
15:44.35}btorch{hardwire: noprobs ... I have but that still didn't work
15:44.49hardwiremake sure its enabled in features.conf
15:44.53redder86coppice: can the modems support getting little bits of data at a time or do they need the full page?
15:45.15}btorch{where should I include it though ? right below [general] on extension.conf ?
15:45.23*** join/#asterisk [ViRii] (n=virii@68-186-170-246.dhcp.smrt.tn.charter.com)
15:45.28hardwireKatty: did you get your steak information?
15:45.28royth1hello people
15:45.30redder86coppice: I implemented flow-control in IAXmodem (xon/xoff)... it seems to send immediately.
15:45.30}btorch{hardwire: it's enabl;ed
15:45.40*** join/#asterisk syllogism (n=syllogis@adsl-69-152-41-249.dsl.ltrkar.swbell.net)
15:45.44hardwiredial 700 from your phone
15:45.48royth1someone uses asthome ?
15:45.58redder86coppice: but I need a way to "look in" and see how much "buffer" is left.
15:45.59*** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se)
15:46.08[ViRii]hey the intercom function for asterisk, how do i do this? i imagine that an overhead speaker is registered as an extension?
15:46.17Kattyhardwire: no
15:46.28hardwireKatty: you just have too much steak ?
15:46.39redder86coppice: right now I do it by way of calculation ... using the DCE-DCE bitrate and the time elapsed to know when to feed more data to the DSP.
15:46.52redder86coppice: but that can be grossly inaccurate.
15:47.07*** part/#asterisk syllogism (n=syllogis@adsl-69-152-41-249.dsl.ltrkar.swbell.net)
15:47.31pashahlaters
15:47.36Kattyhardwire: i'm a vegan. (=
15:47.45hardwireKatty: you require odd things then.
15:47.49}btorch{hardwire: it works when I dial 700 and it gives me 701 as the parked extension
15:47.51Kattyhardwire: and, thusly, i'm adapting reicpes.
15:47.59hardwireKatty: ah.. use TVP strips
15:48.04R3DB0xwhat is better.....the polycom ip501 or the cisco 7970G
15:48.09Kattyhardwire: morningstar farms has just come out with a 'steak' strip simulant.
15:48.15hardwireso TVP
15:48.22hardwirebut it won't soak..
15:48.23hardwirehmm
15:48.25Kattyit has more flavoring than tvp
15:48.27hardwireand you are a vegan
15:48.29hardwireso no sour cream?
15:48.31Kattytvp tastes like crap
15:48.34Kattytofutti
15:48.45Kattytofutti makes vegan sour cream and cream cheese.
15:48.48coppiceusually people just pump data into the modem, and let flow control take care of things. with non-blocking reads you have to keep accepting stuff. I should probably dynamic accept and ignore the device in the select calls, and control things that way
15:48.51hardwireeww
15:48.51arp2tvp plain certainly does
15:49.00royth1hello
15:49.01}btorch{hardwire: but If I call another phone and pick the call on that other phone and try to park the call with #700 it doesn't work
15:49.04hardwireKatty: can I try to remind you that Soy is a muscle development inhibitor?
15:49.05arp2tvp is great in veg chili
15:49.07hardwirebut none the less
15:49.12hardwiresoudns like you could go make some stroganoff
15:49.18hardwirewith some nice wide noodles
15:49.19arp2or veg tacos or whatnot
15:49.27Kattyhardwire: yep, that's what i'm planning
15:49.29coppiceeh? soy is packed with protein
15:49.30}btorch{hmm I like stroganoff
15:49.47royth1who knows the asthome
15:49.51royth1?
15:49.54kahuna_I like doing that too
15:50.00Kattycoppice: everyone likes to bitch aobut something.
15:50.00hardwirecoppice: the soy protein blocks muscle development ;)
15:50.06Kattycoppice: quite frankly, i'm tired of all the bitching :)
15:50.08hardwireI do like to bitch
15:50.17FABRIZIOxxxif i use the tdm04b 4 fxo modules do i need to plug the power supply into the cards?
15:50.18arp2soy is also packed with chemicals that could potentiall interfere with one's hormones (for some people)
15:50.20gordonjcphardwire: does it?
15:50.24arp2(I'm one of those people, so I avoid soy now)
15:50.28*** part/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca)
15:50.29hardwirethis all came about for me after a partner of mine did a study on the effects of a soy oil spill in the aleutians
15:50.33*** join/#asterisk mydssmojo (n=smi23le@d209-89-197-205.abhsia.telus.net)
15:50.33kahuna_I think the PS is for FXS
15:50.36redder86coppice: I actually prefer the flow-control to come from the IAXmodem-side of things.  But I do need to be able to peer-in to the buffer.
15:50.51Kattyhardwire: and by the way, you better stop breathing too.
15:50.54hardwireto find out what effect it would have on the wildlife and kept running acrossed the results that we will basically just have smaller halibut and sea life in that area for a few years
15:51.01Kattyhardwire: cause all that stuff is you breathe is positively disgusting.
15:51.01kahuna_Oh, stroganoff is a food?
15:51.13*** part/#asterisk mydssmojo (n=smi23le@d209-89-197-205.abhsia.telus.net)
15:51.17arp2stroganoff is a recipe
15:51.18redder86coppice: do you know right-off what state variable I'm looking for that tells me how big the buffer is at that moment?
15:51.23spacklechan_stroganoff
15:51.26kahuna_lol
15:51.30hardwireKatty: just pointing out something I found out.
15:51.32*** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com)
15:51.35arp2usually a rich cream-based sauce
15:51.35hardwirenot insulting how you do things
15:51.38kahuna_I thought it was a reference to masturbation.
15:51.51redder86coppice: i.e. are they modem state variables or T.31 state variable?
15:52.13arp2<hardwire> Katty: can I try to remind you that Soy is a muscle development
15:52.13arp2<PROTECTED>
15:52.14coppicethe relevant bit is
15:52.16coppice<PROTECTED>
15:52.18coppice<PROTECTED>
15:52.19arp2... in certain people
15:52.20coppice<PROTECTED>
15:52.20arp2not all
15:52.28kahuna_that's why all vegetarians are wimps!
15:52.37Kattykahuna_: i'm no wimp.
15:52.39redder86coppice: thank you
15:52.44Kattykahuna_: i'd beat you any day.
15:52.51arp2I have a thyroid condition, so I have to stay away from soy
15:52.54kahuna_I bet you would ;)
15:53.14kahuna_But I was only joking, as I was a vegan for 3 years myself.
15:53.16hardwirearp2: the same with my g/f
15:53.32redder86coppice: is it technically possible to transmit (say, V.17) and listen simultaneously throughout the entire transmission and pick up any signals that the other end may be transmitting?
15:53.48sivanalol
15:53.49redder86coppice: or is the communication only half-duplex?
15:53.50sivanatoo funny
15:53.56Kattyanyone have a stroganoff recipe?
15:54.08arp2www.google.com I'm sure
15:54.09tzangernot without beef
15:54.15kahuna_Soy wasn't my thing though. I got most of my protein via legumes
15:54.17Kattytzanger: just post the recipe
15:54.18*** join/#asterisk mydssmojo (n=smi23le@d209-89-197-205.abhsia.telus.net)
15:54.20gordonjcpuse something else insted of beef
15:54.24gordonjcpmushrooms maybe
15:54.28Kattyblah
15:54.35Kattytzanger: i'll redo it how i want it.
15:54.49coppiceV.17 is one half of V.32. V.32 is full duplex through echo cancellation. V.17 is not
15:55.02kahuna_What about chicken stroganoff? Would that be doable? I'm not into beef all that much either.
15:55.08Kattykahuna_: sure, post that.
15:55.09arp2yes
15:55.20arp2its just meat in sauce
15:55.33kahuna_I don't have the algo for that. I was wondering if it would tase good.
15:55.36arp2I'm sure theres a chicken stroganoff recipe on the internets
15:56.05mydssmojoCan anyone recommand any good GUI for Asterisk, I know of AMP; but any others?
15:56.50royth1mydssmojo, i install asthome
15:56.59*** join/#asterisk ManxPower (n=eric@stirprop-s4-0-0-21.ndcr2.datasync.net)
15:57.03[ViRii]i have a fax machine, how can i send a fax through asterisk?
15:57.36redder86ViRii: plug it in to a T1 channel bank
15:57.42spackleViRii: using an ATA and a hope and a a prayer
15:57.51[ViRii]lol
15:57.54[ViRii]thanks
15:58.02spacklemake sure G711
15:58.37mydssmojoroyth1 > I will be using it more on a commercial bases that's why I don't want to use ast@home
15:58.38royth1but i can not raise my graphical environment
15:58.46spackleRedder and Coppice are discussing fax implementation right now.
16:00.07ronaldl79G'Day, room.
16:00.33royth1mydssmojo, of that it forms i raise the web
16:00.54*** part/#asterisk spackle (n=spackle@209.234.83.19)
16:01.29[ViRii]ok fax to email requires email accnts to be setup on the asterisk server?
16:01.45redder86spackle: yes, but we're talking about a softmodem... in ViRii's case the DSP is in the faxmodem, so the data is already modulated audio... in which case we're not really discussing a solution for that.  The "ideal" solution would be for him to use a channel bank or a T.38 ATA (however, Asterisk does not support T.38 at the moment, so channel bank is the only choice that will be reliable).
16:03.09coppiceAh. T.38. I suppose you'll want a T.31 interface to that next :-)
16:03.15olivier_i use FAX-->PAP2--SIP(g711)-->*---ISDN Provider work well in a local network
16:03.31mydssmojoanybody using Sangoma cards?
16:03.50[ViRii]ouch
16:03.56coppiceolivier: it depends on luck and the particular ATA you use
16:04.12redder86coppice: we already have a T.31 interface to T.38 in OpenH323's t38modem
16:04.21*** join/#asterisk bsd3 (n=bsd@203.134.194.176)
16:04.35coppicebut that doesn't integrate with  *
16:04.36royth1mydssmojo, help me to enter the environment AMP, please
16:04.39redder86coppice: can't use it with Asterisk though because of the lack of T.38 support and because the H.323 support is poor
16:05.06bsd3hi, friends!
16:05.08}btorch{man now I have no voice going on when I make a call... I can record calls to voicemail
16:05.17coppiceI ain't even trying to get T.38 with H.323 working. H.323 is obsolete :-)
16:05.18bkw_who wants to receive faxes inside of atserisk?
16:05.18redder86coppice: in my case I really wanted access to the actual DSPs.
16:05.21bkw_thats just not right
16:05.30}btorch{has anyone had this problem .... not been able to hear any sound
16:05.31bkw_t38modem is a nice way to pipe that stuff into hylafax
16:05.35mydssmojoroyth1 > sorry i don't understand, you want to know where you can download AMP?
16:05.38royth1now i am in the way commands
16:05.53coppicet38modem looks really clunky
16:06.01bkw_not really
16:06.07sivanabkw_: most places what to work with an actual fax machine harware
16:06.12redder86bkw: yeah, but a T.38 gateway still requires that a large portion of the fax protocol stuff be on the gateway
16:06.13sivanas/what/want
16:06.15*** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
16:06.19bkw_sivana, then they need to get with the program
16:06.35file[laptop]fax fight!
16:06.38sivanabkw_: what's the program? :)
16:06.42redder86so with T.38 you've got fax protocol happening on the sender, the gateway, and the receiver... not just the sender and receiver.
16:06.46coppiceredder86: yep. that's what I am working on
16:06.57bsd3i have question on refresh time for IAX and SIP
16:07.05redder86coppice: yep, I know.  :-)
16:07.25sivanabkw_: with hylafax, how do you take a piece of paper and fax it?
16:07.38bkw_I have had better luck faxing over IP using IAX one on one than I have over Zap at the moment
16:07.40royth1mydssmojo, a already have it installed in ,my laptop but not know like entering to the way AMP
16:07.41redder86sivana: scanner?
16:07.51sivanaredder86: not practical
16:08.06bkw_yes it is
16:08.17redder86sivana: you can also hook up a fax machine to send it's faxes through HylaFAX.
16:08.19bkw_you just get one of those ones that will take a stack of paper and scan it
16:08.30sivanaya, try to explain that process to a pencil pusher
16:08.37Kattyhmm.
16:08.39bkw_its really a one button process
16:08.42Kattymy office is going through a vegan recipe fad.
16:08.42redder86sivana: many new fax machines support a verison (or a corruption) of T.37.  You can get that into HylaFAX.
16:09.02bkw_redder86, you're right now which one does that?
16:09.02Hmmhesaysbeverly hills, thats where I want to be
16:09.03file[laptop]yay, corruption
16:09.14redder86sivana: but if you want to be really expensive you can plug the fax machine into a channel bank, send the fax to the HylaFAX server and have HylaFAX relay it.
16:09.15*** join/#asterisk spackle (n=spackle@209.234.83.19)
16:09.17coppiceredder86: how many fax machines support t.38?
16:09.21redder86bkw_: the Panafax machines do that
16:09.34redder86coppice: I don't know of any that support T.38 at the moment... just T.37.
16:09.36sivanainteresting
16:09.46bkw_so see their are ways to get around this :P
16:09.50sivanaI have a channel bank, I have a spare server for hylaFax
16:09.51[ViRii]sms messaging = instant messaging?
16:09.58bkw_[ViRii], in a sick world sure
16:10.00coppiceredder86: really? that was the whole point of the TCP mode in T.38
16:10.18Hmmhesayshey file
16:10.26[ViRii]how do i get ip to resolv to a name like Kenny
16:10.27sivanaOR we can fix * to actually handle timing properly to support faxes
16:10.30coppicemaybe the fax makers actually caught on to real time being dumb :-)
16:10.33kahuna_wht do u mean bk? sms rulz!
16:10.34[ViRii]<--- entire newb.
16:10.35blitzragedoes the 7970G have a SIP firmware image yet?
16:10.39bkw_sivana, you want a lot
16:10.41*** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox)
16:10.41redder86coppice: Yeah, most fax machines I've seen that have an ethernet jack in them only support a fax-to-email facility... sometimes T.37, sometimes a corruption of it.
16:10.42sivanahehe
16:10.43blitzragemorning all btw :)
16:10.44sivanaI guess I do
16:10.59mydssmojoroyth1 > sorry i don't understand your english, but i think your asking how to get into AMP?  http://<your amp ip address>/admin
16:11.16bkw_still fax to email is something you can warp into hylafax :P
16:11.17file[laptop]hi blitzrage
16:11.32redder86coppice: I worry about reliablity in a sender -> gateway, gateway -> receiver scenario.  That's two sets of fax protocol happening instead of just one.  Double the trouble for doulble the fun.
16:11.36sivanabkw_: I'm going to play with hylafax.. if I can plug it into a channel bank
16:11.40coppicesivana: the only timing problem in * affecting fax is the TDM400P card problem
16:11.57sivanacoppice: nope, I can't get past 8 pages on 405P
16:11.58coppiceredder86: well that is what T.38 is all about
16:11.59bkw_coppice, well thats not totally true either
16:12.13coppicesivana: then fix your installation
16:12.14redder86coppice: end-to-end T.38 would be wonderful.
16:12.52coppiceredder86: I still think real time is dumb. e-mail queues. it retries. it does all sorts of nice things
16:12.53sivanacoppice: how much you charge per hour? :)
16:13.11blitzragefile[laptop]: ahoi hoi
16:13.48redder86coppice: HylaFAX queues ;-)
16:13.51coppicesivana: I have seem literally hundreds of people complaining about timing when it is their fault. I have a mass of e-mailed audio files from spandsp with frame slips in them
16:14.24coppiceredder86: yeah, kind of, but multiple concurrent deliveries beats that hands down
16:14.38sivanacoppice: maybe so, but 8 pages in?
16:14.59kahuna_that is the TDM400P problem (curious)
16:15.13kahuna_err, s/that/what
16:15.18coppicesivana: whatever. some people want to fix their problems. others just want to moan
16:15.21redder86coppice: with multiple lines out and multiple lines in on the other end you can do that with HylaFAX, too.  The only difference is the communication medium... really.
16:15.41sivanacoppice: just be easy money for you then
16:15.59royth1mydssmojo, i understand, i was thinking that of the same pc, i  can enter to environmente AMP, but it is done from another pc
16:16.04coppice"multiple lines" is a concept more for the 20th century than the 21st
16:16.11*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166116178.nb.aliant.net)
16:16.20redder86coppice: multiple channels?
16:16.47redder86coppice: multiple ports?
16:16.54*** join/#asterisk shmaltz (n=chatzill@69.28.255.210)
16:16.58redder86choose the word that makes you feel most astute
16:17.01coppicee-mail is really good, you know. its *very* hard to beat as a reliable delivery mechanism
16:17.28shmaltzcan I use an analog ADSI phone (like the Aastra PT-480e) with an Adit 600 and get all the nice features?
16:17.34bkw_email isn't something you can rely on if you don't provide the services
16:17.42kahuna_I agree too. Isn't email supposed to replace fax in most instances?
16:17.46bkw_for example AOL blocks all our faxes we send to customers with AOL accounts
16:17.47royth1mydssmojo, thanks
16:17.56bkw_they end up in the spam folder for no good reason
16:18.10kahuna_AOL sucks :(
16:18.29bkw_<PROTECTED>
16:18.29bkw_<PROTECTED>
16:18.29bkw_<PROTECTED>
16:18.32redder86hehe... well, we've had this debate.  I don't have any problem with e-mail.  The difference between e-mailing an image file and faxing an image file, though, is that the data format type is negotiated in the fax protocol while it is not negotiated in e-mail.  Meaning I can send you a JBIG image to you in an e-mail attachment, and you won't be able to view it.  But with fax if I try to send you a JBIG image file and your fax machine doesn't suppor
16:18.34bkw_haha
16:18.44kahuna_I get the same thing when I email my customers. AOL also tends to mangle the messages.
16:19.05*** join/#asterisk \Grooby\ (n=Grooby@66.160.105.186)
16:19.12spacklestop using HTML email
16:19.13kahuna_redder86: looks like your fax got cut off :)
16:19.18spackle<grin>
16:19.29shmaltzcan I use an analog ADSI phone (like the Aastra PT-480e) with an Adit 600 and get all the nice features?
16:19.35kahuna_spackle: I send in both HTML and plantext. AOL refuses to display them both at times.
16:19.51spackleyou said it best - AOL sucks
16:20.01[ViRii]just a question, where did you all learn so much about asterisk =P
16:20.10coppicebkw_: lots of things start with "e-mail has flaws" and end with a huge failed attempt to do better :-)
16:20.12blitzrage[ViRii]: using it for 3 years will do it
16:20.15kahuna_[ViRii]: My mom taught me.
16:20.20[ViRii]lol
16:20.38\Grooby\hey guys.....I have 3 sip phones in Taiwan that register to my * here in DC and every night those 3 phones looses registration
16:20.39blitzrage[ViRii]: sitting in a dark room for 16 hours a day for 4 months talking to bkw_ taught me most of what I know :)
16:20.47file[laptop]blitzrage: LOL
16:20.48kahuna_Really I just talked my boss into buying a quad span T1 card and some voip phones.
16:20.52kahuna_In the name of R+D
16:20.53\Grooby\as of now, i can see the phone register and then lose registration in about 3 minutes
16:20.59spackleMy first job was programming binary load lifters, very similar to our moisture vaporators
16:21.16[ViRii]blitzrage: thats probably what ill have to do :P
16:21.18\Grooby\where can I start debugging this?  I don't think it's network problem cause my brother's there and he's online and I am chatting w/ him fine
16:21.31[ViRii]ive recently been employed to setup an office and i dont know jack about it
16:21.32[ViRii]:>
16:21.39anthmhey this r2 unit has a bad motivator!
16:21.40*** join/#asterisk Moc (n=mochouin@modemcable111.229-203-24.mc.videotron.ca)
16:21.52kahuna_\Grooby\: But it still may be a network problem. Get out ethereal or something.
16:21.53spackleschmaltz, it used to be there were Aastra phones sold by Digium to work with ADSI
16:22.00shmaltzViRii, in the john
16:22.28spackleanthm: great moments in movie whining "But I was going in to Toshi station to pick up some power converters"
16:22.31shmaltzViRii, neither did I a year ago
16:22.37*** join/#asterisk argos73 (n=mike@adsl-70-228-109-5.dsl.akrnoh.ameritech.net)
16:22.53redder86coppice: just sent you an updated patch
16:22.54kahuna_I spent way to much time shooting womprats in beggars canyon
16:23.22\Grooby\hmm ok
16:23.22sigtermlol
16:23.23anthmreviews say anakin was a whiner but if so his son was a chip off the ol' block =>
16:23.36[ViRii]what kinda phones do you use shmaltz?
16:23.37*** part/#asterisk Firestorm-voip (n=Firestor@mail.mysoft.se)
16:23.40*** join/#asterisk thuper (n=thuper@gateway.digium.com)
16:23.59ronaldl79pbx.c:1294 pbx_extension_helper: No application 'Playback ' for extension (default, *90, 4) -- Why is * prompting this when attempting to playback a recording?
16:24.10shmaltzViRii, depends on the job
16:24.24cpatrycause u dont have app_playback.so loaded?
16:24.28cpatryshow modules like playback
16:24.35[ViRii]polycoms?
16:25.07file[laptop]bah, bah hum bug!
16:25.33ronaldl79Okay, I do not see app_playback.so in the list -- shouldn't this be a given?
16:25.36kahuna_Anything but Grandstream
16:25.51spackleschmaltz, you should call voipsupply and ask them about ADSI, asterisk, and channel banks.
16:25.59cpatryronaldl79: load app_playback.so
16:26.23*** join/#asterisk Firestorm-voip (n=Firestor@ua-83-227-140-131.cust.bredbandsbolaget.se)
16:26.37*** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
16:26.47ronaldl79Okay, cpatry, I just tried that and asterisk says it already exists.
16:26.55Drukenanyone here have experince with wan networking?
16:27.01cpatryu just said it was not there.
16:27.05file[laptop]ronaldl79: you have a space... in the name of the application
16:27.13file[laptop]at the end of it
16:27.35ronaldl79doh
16:27.45[ViRii]anyone using polycom phones with asterisk?
16:27.48jake1932good catch file
16:27.54jontowvirii; lots of people
16:28.05[ViRii]jontow: do you recomend that?
16:28.07*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
16:28.07spacklevirii, they are sweet phones
16:28.12*** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net)
16:28.13kahuna_I can't think of a reason why ASDI would not work with a channel bank.
16:28.34kahuna_It's all sent in-band
16:28.51ronaldl79* can be picky, it seems!
16:29.05*** part/#asterisk bsd3 (n=bsd@203.134.194.176)
16:29.44ronaldl79Thanks for that catch, file ... that did the trick...
16:29.58ronaldl79Would it be safe to say not to include ANY spaces in the dialplan? :)
16:30.02ronaldl79I normally don't...
16:30.37jake1932ronaldl79: there are places you need spaces
16:31.03ronaldl79arrrggg
16:31.45*** join/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca)
16:31.50TripleFFF2sdf!seen damin
16:31.58anthmuse parens and you wont have that issue
16:32.06TripleFFF2sdfhttp://www.astertest.com/forum/viewtopic.php?t=8
16:32.08anthmPlayback(woohoo)
16:32.11TripleFFF2sdfanyone have that file
16:32.17TripleFFF2sdfttp://astricon.asterisk.pl/2004-09-recordings/astricon_developer_conference_breakout_failover_09-24-2004.wav
16:33.11jake1932ronaldl79: read this (if you haven't already regarding spaces): http://www.voip-info.org/wiki-Asterisk+Expressions
16:33.25drumkillathat's a totally different issue
16:33.31drumkillaand, isn't relevant anymore :)
16:33.46Abbasany one used asterisk-oh323?
16:33.53*** join/#asterisk svenna (n=svenna@p548D37C2.dip0.t-ipconnect.de)
16:33.56ronaldl79I haven't, jake -- I'm discovering new things about * everyday. Thanks for the URL.
16:34.13jake1932np
16:34.35fileugh I'm hungry
16:34.49spackleHi hungry, I'm spackle
16:35.03filenice to meet you!
16:35.16cpatrynice to EAT you!
16:35.27*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
16:35.27filenoooooo
16:35.55cpatrydont be scared boi!
16:37.57cpatrywith chan_fax?
16:38.46sivanaheh
16:39.34*** join/#asterisk simbulu_ (n=chatzill@80.77.133.83)
16:40.07tzangerugh
16:40.20tzangerF = (a*b)/(a+b) ... is there any way to figure out b?
16:40.27tzangerI can't get b alone on this
16:41.16bendy24b=a
16:41.53cpatry=a(a*b)+b(a*b)
16:43.56simbulu_does anyone know how to use the redhat/asterisk.spec with rpmbuild ?
16:44.53drumkillatzanger: well, my calculator gave me b as a function of F
16:44.57drumkillaif you want that :)
16:45.05tzangerdrumkilla: sure, but does it look pretty?
16:45.57drumkillab = (-6 * F) / ( sqrt(3) * F - 6 )
16:46.04tzangerwhat the fuck
16:46.17tzangerwhere did -6 and sqrt(3) get into there
16:46.32Juggiehah
16:46.33Juggiewhat a joke
16:46.50Juggiei call up digium to tell them i want to do a quote we got a while ago
16:46.57Juggieand they say "oh, our rates have doubled"
16:47.06tzangerhow long is "a while ago" ?
16:47.18ManxPowerJuggie, Did the quote specify an expiry date?
16:47.23Juggieyah, it is expired
16:47.34Juggiebecause in typical goverment fassion my boss dragged his ass for months
16:47.45Juggienone the less, i find it funny they decided to double their rates
16:47.50jake1932i got b = F(a+b)/b
16:47.59ManxPowerAnother example of why I won't work for a govt agency.
16:48.24Juggiei think they are still going to get it done
16:48.24jake1932can you take it more than that?
16:48.43Juggiebut i dont understand why digium would double the rates
16:48.53coppiceredder86: I implemented a solution for your 5 flags requirement that should be generic. I'll pass it over when I have tidied up some other stuff. Do you intend to implement "A/"?
16:49.13spackleto make more money?  They can charge what the market will bear.
16:49.33TripleFFF2sdf[12:47] jake1932: i got b = F(a+b)/b
16:49.43tzangerjake1932: yeah that's what I ended up with but it doesn't help since b's on both sides :-)
16:49.43TripleFFF2sdfi got e=mc2
16:49.46*** join/#asterisk santiago (n=santiago@63.245.86.245)
16:49.47TripleFFF2sdfand e=ri
16:50.40spackleand F=M*V
16:51.14spackleD=R*T
16:52.16redder86coppice: I hadn't gotten around to "A/" yet... but I was planning on getting to everything.
16:52.48*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
16:53.29*** join/#asterisk kahuna_ (n=booger@209-254-56-194.ip.mcleodusa.net)
16:54.43sivanatzanger: b = a / ( a/F - 1)
16:56.19kahuna_I'm getting lots of messages like these for my channel bank channels: zt hook failed: Device or resource busy
16:56.23kahuna_How can I fix it?
16:56.34*** join/#asterisk BartAleph (i=ircap751@pc-64-208-120-200.cm.vtr.net)
16:56.49BartAlephhello
16:57.53*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
16:57.54coppiceredder86: I gave the hdlc_rx_init() call a new parameter to set the number of flags before "OK" is declared
16:58.25kahuna_would flash and rxflash timings possible solve that problem?
16:58.33MikeJ[Laptop]I geb b=a/((a/F)-1))
16:58.37MikeJ[Laptop]do I win a prize
16:58.47MikeJ[Laptop]damn... sivana
16:58.48MikeJ[Laptop]:(
16:59.04MikeJ[Laptop]at least we both got the same answer...
16:59.05MikeJ[Laptop]heh
16:59.27BartAlephi´m lookin for information about use and configure of asterisk like a call center...anybody could help me? (please, excuse my poor english)
16:59.58kahuna_BartAleph: outbound dialling?
17:01.24BartAlephkahuna: just like a call center, you know, with options to the caller, voice-mail, forwarding calls, etc
17:02.22BartAlephkahuna: could you say me where i can find information?
17:02.45kahuna_Well you can start with asteriskdocs.org and read the stuff there,
17:02.56kahuna_That seems like a really basic asterisk config.
17:02.57BartAlephinternet address, forums, anything
17:03.20BartAlephok..i´m taking notes
17:03.50*** join/#asterisk Veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
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17:04.28kahuna_What does the term "debounce" mean when it comes to analog phones?
17:04.54BartAlephanyother useful address
17:04.58[ViRii]for overhead paging i need an fxo port? this port is on ,.... a soundcard?
17:05.06kahuna_www.google.com
17:05.30ronaldl79Anyone care to share why they prefer PRI over VoIP connectivity?
17:05.38BartAlephok...i used it :)...but a need some guide to look...i'm new with asterisk
17:05.45kahuna_[ViRii]: If I remember right you can just Dial() the console and the output will go out thru the soundcard
17:07.26*** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
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17:07.52obiyodaHello. I am runnig A@H  I have a queue but when all agents are on the phone it starts to beep at them when people are in the queue. I have turned off call waiting on the agents phone but still get the beep any ideas on what i can do?
17:07.59*** join/#asterisk Cresl1n (n=matt@m415e36d0.tmodns.net)
17:09.18*** join/#asterisk Beave (n=beave@vistech.org)
17:09.19*** join/#asterisk syle (n=blag@unaffiliated/syle)
17:11.57kahuna_what's the heck is kewlstart? I've never heard of this before I configured my * box.
17:12.18blitzragekahuna_: its loopstart with far end disconnect supervision
17:12.39kahuna_Ok
17:13.17kahuna_So if I have lines going into a channel bank to be multiplexed into an * t1 port then ks is the right signalling?
17:13.53blitzragekahuna_: kewlstart is usually the first thing to try - if it doesn't work, then its either groundstart or loopstart plain
17:14.03kahuna_I see.
17:14.20*** join/#asterisk ikey (i=ikey@220.226.37.186)
17:14.45kahuna_I'm having trouble with users not pressing the hook button down long enough and instead of hanging up it actually initiates a transfer.
17:15.12*** join/#asterisk Derkommissar (n=alberto@66.64.215.6.nw.nuvox.net)
17:15.14DerkommissarHalo
17:15.25kahuna_There was no trouble at all when using loop start on the channel bank.
17:15.37Derkommissarquestion... i started to see a lot of this lately,,,,, Sep 28 13:09:11 WARNING[30442]: channel.c:1768 ast_indicate: Unable to handle indication 3 for 'SIP/5060-f539a738'
17:15.37Derkommissar<PROTECTED>
17:16.33*** part/#asterisk \Grooby\ (n=Grooby@66.160.105.186)
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17:19.44jarrodcan someone please give me an example mgcp.conf that works with Adit600
17:20.36}btorch{hmm .. whenever I enable the parkedcalls and add the Dial(...., rtT)  ... if I call aanother phone there is no sound
17:21.26*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
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17:24.13kahuna_If I change something in my zapata.conf and reload will it kill any of my channels that are live?
17:24.47*** join/#asterisk loick (n=loick@per92-7-82-236-197-96.fbx.proxad.net)
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17:25.16TheCopsHi
17:27.44Syncrosmoo
17:27.59Syncrosbon matin cops
17:27.59ManxPower}btorch{ don't use "r" and don't use "t" or "T" unless you HAVE TO
17:28.20ManxPowerkahuna_, 1.0.x or CVS-HEAD/1.2beta?
17:28.50*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
17:29.08kahuna_1.0.6-2 ubuntu .deb
17:29.27*** join/#asterisk jief- (n=jief@modemcable163.182-80-70.mc.videotron.ca)
17:29.40ManxPowerkahuna_, it will not kill any calls, but it will not update the settings either
17:29.59kahuna_Ok. Best to wait until everybody is off the phone then :)
17:30.03ManxPowerin 1.0.x you must unload chan_zap.so and then load chan_zap.so or stop and start asterisk for changes to take effect.
17:30.09jief-hello. i have to deploy a PBX with asterisk that will support up to 4000 phones. i was wondering if 1. anyone ever deployed such a large system 2. what kind of hardware should we use?
17:30.26ManxPowerjief-, that is a question for the mailing lists
17:30.35*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
17:30.51jief-ManxPower: just out of curiosity, why isn't this an irc question?
17:31.37R3DB0xdoes anyone have any suggestions bout the cisco 7920 phone....my customer wants a cordless setup and i heard that cordless VoIP stuff sucks and its better to use an analog phone with an IAXY converter
17:31.43Drukenbecause 4000 phones is a major company, and major companies don't irc :)
17:33.48JerJerand that question leaves a lot open for discussion
17:34.08JerJerirc is only for verbally hating on those that annoy you
17:34.26g__(All the more reason big companies should be on irc.)
17:34.28jief-JerJer: i would think its more because no one here knows the answer
17:34.51JerJerno - i have two entire school districts that i manage, using Asterisk
17:35.00JerJerthey've got about 25,000 phones
17:35.02[ViRii]jerjer: what kinda phones?
17:35.20JerJer7905s, 7960s and a few Sipura ATAs
17:35.38[ViRii]how do you get an overhead speaker to work with asterisk jerjer
17:35.49JerJerone of the sound card channels
17:36.31JerJerdon't pm me
17:36.35[ViRii]sry
17:36.38JerJerbkw says i lie about people pm-ing me
17:37.07[ViRii]jerjer, a school has a pa system how does it work do you dial an extension and it activates?
17:37.23JerJerchan_oss or chan_alsa - depending on what sound system you use
17:37.28JerJeruse/need - duno
17:37.39jarroddoes anyone have an example mgcp.conf that works with the Adit 600?
17:37.44[ViRii]whats the difference between the two?
17:37.45jief-JerJer: since you've done such a large install, do you think one 2 way box can support up to 4000 phones? only SIP, no zap.
17:37.57JerJerdepends on what's running on that box
17:38.08[ViRii]onboard sound.
17:38.08JerJerSER, certianly
17:38.08JerJerAsterisk, doubtful
17:38.17jief-JerJer: yeah would only be *
17:38.23JerJerplus you don't want just one media gateway / application server
17:38.43JerJeri would rather have 2 or more so i know not one outage will bring down the entire system
17:38.51jief-JerJer: well, * will makes calls through a Quintum Tenor CMS softswitch
17:38.56JerJeri'm sorry
17:39.06jief-JerJer: oh, its planned, there'll be a hotstandby
17:39.16kahuna_[ViRii]: So you hook a line from your seaker out into the overhead pagind and dial the console in your dialplan.
17:39.32kahuna_[ViRii]: Google it, there's examples all over.
17:39.55jief-JerJer: how would you split the load then? half the extensions on one box, and the other half on another? connected with iax2?
17:39.59JerJerno
17:40.09[ViRii]thanks
17:40.11bjohnsondidn't someone have a way to use vonage economically?  deal on at future shop and best buy this week .. $15 in your pocket after credits
17:40.17JerJerdon't use vonage
17:40.35kahuna_Vonage is * unfriendly
17:40.46Juggiewhos goin to astricon?
17:40.47JerJerunless you are stupid enough to pay for the softphone account
17:40.47[ViRii]oh no
17:40.51bjohnsonheh .. I got flack from some nitwits who I told shouldn't use skype
17:40.52JerJerJuggie:  not me
17:40.53hypa7iathere are sooo many better canadian voip providers
17:40.58[ViRii]vonage =! asterisk?
17:41.03filejief-: if you're just doing SIP on a scale that large, just use SER routing calls to your Quintum and Asterisk for features
17:41.13JerJerfile: word
17:41.45Drukenone of these days i'll have to play with ser....
17:41.58fugitivowhere is documentation about ser_
17:42.08azziejief-, are you already doing SIP on Quintum CMS or just planning to?
17:42.13JerJerdocumentation?   that's funny
17:42.17*** join/#asterisk chiardon1 (n=chiardon@200.71.58.39)
17:42.25}btorch{how come asterisks keeps giving me problems when I connect from the same IP as different users ?
17:42.33JerJeryeah i would avoid Qunitum like the plauge
17:42.39bjohnsonthe simplicity of vonage is very attractive to home users
17:42.49jief-azzie: its alraedy in use with 32 PRIs
17:43.01}btorch{I change the SIP Proxy config on X-Lite and than I keep getting this Awaiting proxy login information
17:43.02JerJerbjohnson: if said home user wants to waste money, sure
17:43.04azziejeif: 32 pri's and sip ?
17:43.08*** join/#asterisk [hC] (n=hardcore@8.10.2.53)
17:43.10jief-azzie: yes
17:43.10spacklebjohnson: plug'n'play like AOL, brainless.  But you pay for it.
17:43.11[hC]morning fellas.
17:43.32azziejief-, impressive, so quintum managed to make SIP more-or-less stable?
17:43.40JerJerless
17:43.57jief-azzie: well, last i was there 2 days ago, and asterisk didnt seem to be able to talk with it. i was always getting a 503 error code
17:44.03bjohnsonJerJer: I've run into numerous ones who insist that anything else is too hard
17:44.05*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-202.rockynet.com)
17:44.05[hC]I've upgraded to cvs head as of last night (from cvs head august 15th) and I'm still having odd IAX frames show up on an iax link of mine. Ive included output in a pastebin, incase anyone can take a look and offer a suggestion
17:44.10[hC]http://pastebin.ca/24128
17:44.10jief-azzie: going back there tomorrow, hoping to get something done
17:44.41azziejief: emm... so do you have standing sip calls on your CMS or you're just planning to have them in the future?
17:44.51kahuna_You know who's the worst voip provider though?
17:45.02kahuna_Nufone!
17:45.03*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
17:45.22jief-azzie: the client used Ondo (from bekeke.com) for a while with the CMS. so SIP is fine
17:45.38JerJeryeah the cock suckers at NuFone will never return my calls
17:45.44kahuna_lol
17:45.53bjohnsonJerJer: I even met one bonehead who (talking about skype) said if it's easy now, who cares about ease of upgrading in the future. I had mentioned that they could get a standard SIP fxs and config that easily
17:45.57jief-azzie: just doesnt seem like it likes *. if i use a phone, and set the CMS' IP as the outbound gateway, i can make outgoing calls, but not if  i route them thru * first
17:46.11JerJerbjohnson:  yeah really
17:46.39JerJerjief-:  Qunitum very much so does their own thing
17:46.40hardwiregrr
17:46.41azziejief-, i see... my company has a pile of quintums (all kinds) and I was just wondering if it's the time for SIP in this department :)
17:46.44hardwireI updated to CVS :)
17:46.45kahuna_skype sucks unless you're stuck with dialup. period.
17:46.46hardwireand now the t1 doesn't work
17:46.48hardwireturns out we had an outage at the ILEC right when I changed
17:46.50JerJerwe were forced to drop their bullshit from Interop testing
17:46.59hardwirekahuna_: skype also can use a wideband codec
17:47.00ronaldl79Guys: Please refresh my memory ... but in the diaplan, how is the ampersand used to ring two SIP devices at once?
17:47.05hardwiremore hz for the buck
17:47.10hardwirefor skype to skype calls
17:47.20JerJerDial,SIP/bob&SIP/dillan
17:47.20kahuna_yes it can, but you can do standard voip then...
17:47.30jief-azzie: might have been my * config that was wrong though. in a macro, i had exten => s,1,Dial(SIP/1.2.3.4). and that's how i was getting that 503 error
17:47.32hardwirekahuna_: really?
17:47.33azziejer: dropped Q-m's?
17:47.35ronaldl79Yeah, that's what I did, JerJer...hmmm...thanks
17:47.42hardwiredo you have some voip software that does wideband codecs?
17:47.45}btorch{why does asterisk keeps showing me 71488/71488 192.168.120.212 D 255.255.255.255 5060 Unreachable ?
17:47.50JerJerQ-m's ?
17:48.11[hC]hardwire: haha. doh. I spent 20 minutes trying to debug why wcfxo wasnt inserting last night only to discover the card had been removed from the box i was upgrading earlier to debug. :)
17:48.11azziejer: dropped quintums from interop testing?
17:48.14JerJerazzie: yes
17:48.17azziejer: wow
17:48.19JerJerthey suck
17:48.20hardwire[hC]: your a fucking moron.
17:48.23hardwire:)
17:48.29hardwireatleast I bet thats how you feel
17:48.37*** join/#asterisk brookshire (n=matt@gateway.digium.com)
17:48.40azziejer: true
17:48.46azziejer: any particular reason?
17:48.46[hC]hardwire: yes. its worse though, cause the box is remote, and i couldnt actually check for myself.
17:48.51hardwireI am a fucking moron.. because I spent all night trying to add a new app to chan_sip.c
17:48.58hardwirejust to find out there was a dialplan functino that does what I want
17:49.04JerJernot even the guy Qutium sent could configure the bastards right
17:49.07jief-ok, how do you send SIP calls, to another SIP device (like a softswitch)? Dial(SIP/1.2.3.4) ?
17:49.07hardwiresame code.. same everything
17:49.08[hC]hardwire: haha. what were you trying to add?
17:49.10kahuna_Well there are proprietary wideband codecs, so I retract my statement about "standard" voip/
17:49.18hardwireI awnted to get sip peer information
17:49.20hardwirewanted
17:49.30JerJerjief-:  use a type=peer, so you have more control
17:49.33JerJer~docs
17:49.34jbotit has been said that docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
17:49.34hardwirelike the UserAgent
17:49.41[hC]hardwire: ahh.
17:49.48*** join/#asterisk chiardon (n=chiardon@200.71.58.39)
17:49.48hardwireso I added SipGetPeerData
17:49.51JerJerhardware sip show peer bob
17:50.04hardwirejust to find out ${SIPPEER(1153:useragent)} exists
17:50.15[hC]hardwire: yeah. you are a fucking moron too, then.
17:50.18JerJerRTFM
17:50.18hardwireyup
17:50.20[hC]:)
17:50.21kahuna_hehe
17:50.32hardwireJerJer: for use in another app
17:50.36jief-JerJer: yeah, i tried adding an entry in sip.conf for the softswitch, but with type=peer, but i always get a 503 error
17:50.36hardwirenot from the CLI
17:50.38azzieJerJer, oh... their SIP is screwed in particular or you are talking in general?
17:50.50[hC]hey, has anyone bought that oreilly book on *? amazon has it for 25 bucks, i was thinking of picking it up for fun.  i suspect it will be way too basic, though.
17:50.56kahuna_you should go work for Microsoft then, trying to re-invent the wheel/
17:51.02JerJerazzie:  all of the above - they don't follow the reco
17:51.09azzie[hC], I did, but they still did not ship it :(
17:51.12hardwirekahuna_: ?
17:51.13JerJerthey do their own thing quite a whole lot
17:51.44hardwiremy span is freaking red alarm
17:51.45hardwireI hate this
17:51.57ronaldl79JerJer: Sip/XXX&Sip/XXX -- is that the best way to ring two devices at once? Does the same apply to IAX, etc. too?
17:51.58*** join/#asterisk AgiNamu (n=Michael@dsl081-096-215.den1.dsl.speakeasy.net)
17:52.13JerJerronaldl79:  you can even mix channel types
17:52.19ronaldl79Good deal.
17:52.26ronaldl79Gotta love *...
17:52.35JerJerother than using a queue setup, thats really the only way i can think of
17:52.45azzieJerJer, I see
17:52.55azzieJerJer, thank you for the info
17:53.03JerJerthank you, drive-thru
17:53.08azzie:)
17:53.33JerJeriax2 is male
17:53.34kahuna_ouch
17:53.45kahuna_A male with ovaries?! Good God!
17:53.45spacklejust undecended?
17:53.51JerJerperhaps transgender, we don't really know
17:53.53[hC]JerJer: really... it seems like a bitch to me. :|
17:53.59bendy24o0
17:54.02[hC]pre-op or post-op?
17:54.04chiardonovaries is a so deeply layer . .isn't it?
17:54.05AgiNamuAnyone here decent with configuring PRIs (non asterisk, but I'll pay for help)
17:54.17hardwirehmm
17:54.21hardwireis zaptel in CVS dead?
17:54.22azzieAgiNamu, what equipment
17:54.27AgiNamuazzie, cisco as5350
17:54.28}btorch{great .... asterisk is pissing me off
17:54.30*** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net)
17:54.36JerJerAgiNamu:  try #cisco
17:54.42azzieAgiNamu, what could be simpler then
17:54.44AgiNamuoh.. duh :P
17:54.47[hC]}btorch{: get in line, sunshine
17:54.52}btorch{hehe
17:55.09ronaldl79JerJer: Is it possible to ring more than two sip devices, e.g, Sip/XXX&Sip/XXX&Sip/XXX?
17:55.24azzieAgiNamu, set up t1/e1, isdn type and that's all you need... :)
17:55.27*** part/#asterisk santiago (n=santiago@63.245.86.245)
17:55.33*** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
17:55.34AgiNamuazzie, yea, I did... that was easy
17:55.39mutilatorronaldl79 use a comma..
17:55.40AgiNamubut it just stays at "TEI_ASSIGNED" :P
17:56.12*** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
17:57.03azzieAgiNamu, show contr t1 and make sure it's up and no errors; then debug isdn q921 and debug isdn q931
17:57.21ronaldl79How are you guys addressing instant termination in your dialplan? There's generally a delay unless you press "#".
17:57.35AgiNamuazzie, yep, T1 is up fine. I'm debugging q931 and 921
17:57.48fileronaldl79: if it's a SIP device that's on the SIP device's dialplan, not asterisk
17:57.56AgiNamuI just see a bunch of "sending SABME", User TX.... and on 931 I see "Ux_DLRelInd: DL_REL_IND received from L2"
17:57.59AgiNamuthat's it
17:58.01ronaldl79What about IAX, file?
17:58.08filean IAX2 phone?
17:58.15kahuna_I just configured my 1st PRI on * today.
17:58.15ronaldl79Yes, file...hard or soft.
17:58.26fileit's all the phone/ata...
17:58.30azzieAgiNamu, your side hopefully is user-side (not network) ?
17:58.35ronaldl79Cool, file...thanks.
17:58.36AgiNamuright, userside
17:58.39chiardonkahuna_ clap . .clap . . .clap . . .
17:58.44fileit doesn't communicate with Asterisk when you're dialing, it waits for you to finish before it goes
17:59.30JerJerronaldl79:  last i knew you could string together 254 devices to ring
17:59.43JerJerand its not a comma its an ampersand
17:59.49kahuna_I'm sure you guys have done hundreds of them but you probably still remember the satisfaction of your first time.
17:59.51ManxPowerWhat?!?!?
17:59.57ronaldl79Whew, JerJer, that's amazing....
18:00.04ManxPowerTech/device&tech/device
18:00.34JerJerronaldl79:  at some point a queue becomes more efficient
18:01.07ManxPowerQueues are EVIL.  However, there always has to be some evil in the world.
18:01.28*** join/#asterisk fiber0pti (n=johndoe@207.114.199.98)
18:01.53*** join/#asterisk extremis (i=extremis@equinox.alluvium.com)
18:02.04fiber0ptiI'm trying to get some questions answered about Polycom 500/501 phones and asterisk. Is it possible to have these 3 line phones utilize more than 3 lines?
18:02.13extremisIs there a way on a 7960 to transfer an incoming call to voicemail imediately without having to pick up the phone?
18:02.28JerJerextremis: have a direct to voicemail extension
18:02.55extremisJerJer: as the user receiving a call, I want to bounce the incoming call to voicemail after it has been transfered to me
18:02.55JerJerlike  exten => 6XXX,1,Voicemail,u${EXTEN:1}
18:02.55[ViRii]jerjer: oss or alsa , got an onboard soundcard how to get it to recognize in linux/asterisk
18:03.17JerJer[ViRii]: duno - i have never had luck with either
18:03.22[ViRii]lol
18:03.24[ViRii]thanks :>
18:03.43JerJerextremis: where ${EXTEN} is their three digit mailbox, of course
18:03.50extremisfor example, someone dials me from the main menu
18:03.52ronaldl79Now, this is strange...I just built a Suse Linux box last night with Asterisk 1.0.9 ... I was just attempting to modify some things, and Pico is telling me I can't ... because it's a 'read only filesystem.' Damn, I hope this box isn't fucking up...I'm demoing some stuff today for a client!!!!
18:03.54extremisand I dont want to talk to them
18:03.58extremisand I don't want it to ring
18:04.07JerJerput your phone on DND
18:04.09extremisso I evaluate based on caller id and decide, send that person to voicemail now
18:04.15JerJerthen do that
18:04.28extremisso I can put it in DnD in the middle of an incoming call, and it will get redirected?
18:04.30JerJeranti ex-girlfriend logic style
18:04.34JerJerno
18:04.35[ViRii]ronald: ack
18:04.41JerJerwell doubtfull - test it
18:04.43extremiswell, a client wants it, yes anti-exgirl logic
18:04.50JerJerthen use it
18:04.52extremisthey said their existing pbx does it
18:05.02extremiswell, dnd is there, but does it work when the call is already coming?
18:05.04JerJercall routing based on Caller*id ?
18:05.06lathos42extremis: The Polycom phones have a "Reject" feature.. You should see if the 7960 has a similar feature
18:05.27extremisagain, the call is ringing at my phone, and at that exact moment, I want to bounce them to voicemail
18:05.32extremisthe next call I may want to receive
18:05.58bjohnsontransfer?
18:05.58JerJerthats' not anti ex-girlfriend logic then
18:06.14bjohnsonbut you would have to answer the call first
18:06.16extremiswell, that is what they are used to
18:06.29[hC]hardwire: huh.. looking further, when i had thought that frame_control type 3 was video.... its actually a "remote end is ringing" indication.
18:06.30bjohnsonyou mean they want to see the callerid and then decide whether to answer or not?
18:06.31extremisthey have a pbx that you can hit a button on and it dumps the current incoming call to voic email
18:06.44bjohnsonwhy not just wait for the time out
18:06.46extremisbjohnson: yes, but if they don't want to answer, they want to send to voicemail imediately
18:06.47JerJersee if cisco's can do that
18:07.04bjohnsoneg ring for 10 seconds and then send to vm
18:07.07JerJersee lathos42's comment
18:07.07extremisthey don't want to wait for a timeout
18:07.10*** join/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net)
18:07.14bjohnsonif they don't want to answer, turn off the ringer
18:07.22bjohnsonstrange request
18:07.27extremiswell, they are used to the feature
18:07.29ManxPowerall that stuff is controlled BY THE DAMN PHONE
18:07.29bzbw<PROTECTED>
18:07.30extremisthat is why they want it
18:07.31bjohnsonthat they would act that quickly
18:07.38extremisI can change their behavior, or I can find a way to do it
18:07.44extremisit doesn't look like I can find a way to do it
18:07.44bjohnsonno I don't want to wait 10 seconds to not talk to this person
18:07.44ManxPowerbzbw, yes, yes
18:07.45JerJerAsterisk v1.2 is not out yet
18:07.55JerJerextremis:  just show them DnD feature
18:08.04extremisJerJer: any way to put it on the softmenu?
18:08.10bzbwI mean beta for 1.2
18:08.19JerJerduno - read the TFTP configs for cisco
18:08.21JerJerperhaps
18:08.24bjohnsonbut dnd will not ring their phone correct?
18:08.27bzbwManxPower, what is the driver name?
18:08.31ManxPowerextremis, the 7905 supports rejecting the calls.
18:08.41lathos42extremis: Cisco would have to support it in the phone.. Polycom has a feature in the Softmenu that does exactly what you want to do.. call is coming in, hit reject, it bounces immediately to voicemail
18:08.44extremisManxPower: really?
18:08.45ManxPowerbzbw, read the zaptel README file, it lists each driver and what board it is for
18:08.47hardwire[hC]: I freaking told you
18:08.50ManxPowerextremis, yes.
18:08.53hardwirewhat did I freaking tell you
18:08.57hardwirethat it was ringing
18:08.59hardwireI told you
18:09.00bzbwthx
18:09.00bjohnsoneasier to just turn off the ringer if it has a ringer switch
18:09.02hardwireI told you
18:09.07ManxPowerextremis, But I don't think the other 79XX SIP phones support it
18:09.12JerJerlathos42: well if the call agent is setup to send them to voicemail   :)
18:09.26extremisManxPower: they have some 7912s... do those support it?
18:09.33ManxPowerextremis, I would assume so.
18:09.41lathos42JerJer: True :)  I think i'm going to make poeple have to work to get to my voicemail box
18:09.45JerJerwhy wouldn't 7960 have it?   its just a software function
18:10.00ManxPowerWhen I had a 7905 for testing I had to update my dialplan to make the call go to the right place when you rejected the call
18:10.11bjohnsonlathos42: guess a number between one to ten?
18:10.16ManxPowerI think it's not called "REJECT" I think it's a softbutten labeled "toVM" or something like that
18:10.25lathos42bjohnson: That's a good idea..  That might be step 5 :)
18:10.28bjohnsonlathos42: all wrong answers they hear "that's not it.  try again"
18:10.30JerJerlathos42:  i can see it now..."Please dial Pi out to the 30th digit to leave VM for bob"
18:10.58lathos42JerJer: Yep, and its their problem you can't dial .
18:11.04geincan someone help me configuring queues? (having problems with the joinempty option)
18:11.19geinconfigure*
18:11.28JerJercan we?
18:12.05geinmay
18:12.14JerJerdon't ask to ask, just ask
18:12.23lathos42bjohnson: It would be even better if with every wrong answer, they have to start the menu over again from the top
18:12.46geinok, sorry, seems like joinempty=strict has no effect at all... calls are placed in the queue even if there's no agents logged in
18:12.58ManxPowerJerJer, The 7905/7912 use their own firmware, not the same at the 7940/7960, there are some differences between the two firmwares
18:13.22JerJeryea i believe the 7905/12 are based on the komoto ata shit
18:13.32ronaldl79I know this is only *, but why the hell is this Suse box not loading anything -- everything is read-only now? If the timing couldn't be better!!!! Is something corrupted?
18:14.01JerJerpress ctrl-alt-delete
18:14.02}btorch{how do you do a database del in the CLI ?
18:14.03ronaldl79I've never experienced this before, and here I am all excited about demoing this box ... and now I've got to troubleshoot before the 1:30 PM meeting!
18:14.04JerJersee if that fixes anything
18:14.11ronaldl79I already did that, JerJer, it's still the same thing.
18:14.21JerJerdont' run suse
18:14.23spackledid you mount the volume read-only
18:14.30ronaldl79No, spackle.
18:14.33ManxPowerronaldl79, NEVER EVER demo a box on short notice.
18:14.34JerJerdrop a netinstall of debian on there - worst of both worlds
18:14.38ronaldl79Everything was fine last night and this morning...
18:14.50bjohnson}btorch{: check the chan that supports your db .. ie #mysql for mysql
18:14.55mutilatorshort notice = ?
18:14.55jarrodhmm.. adit600 not registering with asterisk mgcp
18:14.56ronaldl79I've always used Suse....
18:14.59mutilatorappropriate time = ?
18:15.31bjohnson}btorch{: there are typically commands to do that.  If you got the db as part of a package, you could look at the files the package contains that go into some /bin dir
18:15.32}btorch{bjohnson: I'm not using mysql though but when I do a database show it shows me to phones in the registry
18:15.36JerJerlathos42: hell yeah
18:16.00[hC]hardwire: stfu
18:16.01[hC]:)
18:16.10}btorch{bjohnson: the strange thing is one phone is connect  but the other one is not I closed the app and still shows as register
18:16.12JerJerBUFU
18:16.14ManxPowerjarrod, I thought Asterisk's MGCP support was client only, not server.
18:16.27jhiverHi All
18:16.38jarrodthats incorrect
18:16.56fileit's server only
18:17.17jhiverAny body has some info about that GSM Asterisk card that has been advertised on the ML?
18:17.51JerJerits vapor until i see one
18:17.53ManxPowerjarrod, must have changed in the past year then
18:18.10JerJerManxPower:  yeah i think someone contributed a serious amount to mcgp
18:18.12JerJermgcp
18:18.16ManxPowerI have all my projects on hold until I can get the Digium DS-3 card!
18:18.17ManxPowerNOT!
18:18.23jhiverSounds good though, I hope they manage to get some stuff working
18:18.53jhiverit would be good to have sub €500 / port GSM card
18:18.56JerJerDid atacom ever release their bullshit?
18:20.17bjohnsonI don't know, but bullshit is generally freely available from multiple sources
18:20.36bjohnsonI've got some here if you want it
18:20.42[hC]hardwire: i wonder then if this is even my FREAKING problem after all.
18:20.56hardwireprobably not
18:21.01hardwirehave yup upped to 1.2.0 or CVS yet?
18:21.08hardwireyup/you
18:21.21[hC]i was running cvs head 8/15, and cvsup'ed to last night's cvs... last night...
18:21.25bjohnson[hC]: if you have an underling, then it can quickly become somebody else's problem
18:21.35[hC]bjohnson: oh god, do i ever wish.
18:21.40blitzragein stable, is it possible to strip a group of digits from the "centre" of the ${EXTEN} ?
18:21.41[ViRii]anyone got paging and intercom to work with polycom600's?
18:21.49spackle"underling" heh
18:21.51hardwirebjohnson: hahahaha
18:21.56blitzrageie.  111222444 <- strip 222
18:22.03blitzrageresulting string 111444
18:22.10hardwireblah
18:22.12[hC]we call those ones 'boah' around here... as in, fetch me my coffee, boah.
18:22.14hardwirebbl
18:22.16pauldyronaldl79, did you try a little mount -o remount,rw /
18:22.21bjohnsonunderling .. what do they call the buddies of the bad guys in the old movies?  henchmen?
18:22.37BeirdoI like "peon" or "minion"
18:22.38hardwireboard of animal health?
18:23.06bjohnsonserf
18:23.15spacklegrunt
18:23.39Beirdoif it's a guy (and only then): "bitch"
18:23.40spacklelathos42: when you get promoted you can be the IT mangler
18:23.54JerJerperhaps ${EXTEN:9:-3}
18:23.55filehave a little faith in me!
18:24.11JerJerwomen are bitches though
18:24.14Beirdosome are
18:24.22lathos42spackle: I was reading Slashdot yesterday and someone mentioned in a comment that he was able to make up his own job title once, so he chose "Master of the Devices"
18:24.25Beirdobut it's not polite to call them that in a work situation
18:24.27pauldyholy mother flooder
18:24.34pauldythis server must be lagged
18:24.58spacklepauldy - why did you send your message 20 minutes ago?
18:25.00Beirdoyou can get slapped with sexual harassment SOOO fast
18:25.24pauldyspackle, see previous statement
18:25.52*** join/#asterisk Mw3 (i=mw3@spool12-14.gatesgroup.hu)
18:25.55Beirdowoohoo, go linode
18:26.07[hC]<3 linode
18:26.30Beirdoit just bounced my IRC connections :)
18:26.50Beirdoat least I'm gonna blame that
18:26.51Beirdoheh
18:27.25}btorch{is there a place to change some timeout value for when a user becomes unreachable
18:27.35*** join/#asterisk MnxPower (n=eric@stirprop-s4-0-0-21.ndcr2.datasync.net)
18:27.35*** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net)
18:27.41pauldymaybe that will help
18:29.47bjohnson}btorch{: you're gonna have to be way more specific
18:29.47g__jerjer, followup question from yesterday: why are "agents" important to queues.. why doesn't it work properly otherwise?
18:29.47JerJerg__:  i use queues without agents all of the time
18:29.52JerJermember  = > SIP/bob
18:30.07JerJerbut SIP/bob has to have call waiting turned off
18:30.09}btorch{bjohnson: for some reason I had an  sip phone registration that was still showing up if you did a database show on the asterisk CLI ...
18:31.00}btorch{bjohnson: it took around 10min for the registration to go way ... it keept the port open and status as unreachable ...
18:31.02*** join/#asterisk bockx (i=bock@128-5-112.adsl.terra.cl)
18:31.41g__..and if SIP/bob doesn't, he gets calls even when he's busy?  Ok, then if SIP/bob doesn't pick up the phone, will it go to someone else?
18:32.04}btorch{bjohnson: using the same PC  with a new X-Lite SIP proxy configuration would register the new phone and bring the old one back online
18:32.13JerJerthen SIP/bob need to unregister or set DND
18:32.33JerJerand yes, if SIP/bob does not pick up the call will get sent to another member of the queue
18:33.05g__I've found if SIP/bob sets the phone to DND or forgets to unregister, calls get stuck on his phone.
18:33.13JerJerstuck?
18:33.13malverian[work]Hmm.. asterisk keeps getting killed by kernel.. I need to figure out why :-/
18:33.25JerJerby kernel?
18:33.28g__Ie, they don't pass on to another extension
18:33.31dsfrhah
18:33.43JerJerg__: then you are doing something wrong
18:33.45malverian[work]JerJer, Almost positive.. It's saying "Killed" and then closing.. might be an OOM condition.
18:34.00malverian[work]Though I'm not seeing anything in dmesg
18:34.01JerJerno someone is doing kill <pid of asterisk>
18:34.54malverian[work]JerJer, I'm the only person with access to the machine.
18:34.59malverian[work]Some_THING_ is doing it.
18:35.03malverian[work]And it's happening quite frequently.
18:35.15g__Under some conditions, the kernel does pick arbitrary processes to kill.. but those conditions are usually "low memory" or something equally bad.
18:35.25pauldywhile test true; do kill -9 $RANDOM;done
18:35.30malverian[work]g__, Yeah, that's what I was saying.. OOMkilller
18:35.31pauldythats a fun one to run
18:35.54JerJerhow does $RANDOM get set?
18:35.56bjohnson}btorch{: I believe SIP device registration timeouts are controlled on each device
18:35.58g__oh right, sorry.
18:36.02JerJeror is that a special shell var?
18:36.02malverian[work]pauldy, Except 99% of the time, you won't get a valid pid ;)
18:36.09*** part/#asterisk simbulu_ (n=chatzill@80.77.133.83)
18:36.11malverian[work]JerJer, Yeah
18:36.13malverian[work]JerJer, Bash var.
18:36.15JerJerahh ok
18:36.21pauldyJerJer, shell var
18:36.34pauldyyea malverian[work] but its like playing rusian roulet
18:36.39malverian[work]pauldy, Heh :)
18:36.45malverian[work]Anyone know how to disable OOM killer?
18:36.46JerJerclick click click click click
18:36.49JerJerboom
18:36.51malverian[work]Is it sysctl?
18:36.52}btorch{bjohnson: are you talking about the timeout for trying to register or when you close the application ?
18:36.54pauldyand running it like that is bound to kill something
18:36.56pauldymaybe itself
18:37.05g__I'll stick the Doom systems administration mode.
18:37.14g__s/stick/stick with
18:37.30*** join/#asterisk ManxPower (n=eric@stirprop-s4-0-0-21.ndcr2.datasync.net)
18:37.32*** join/#asterisk psypete (n=psypete@pix-nat.makosurgical.com)
18:38.29*** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net)
18:38.45}btorch{bjohnson: do you use x-lite ?
18:38.50g__Ok, so the point of 'Agents' is to let people move around a bit more?
18:39.04[hC]this might sound stupid, but is it possible that a frame request for 'remote ringing' on an IAX channel, could cause other calls via that IAX link to cease audio rx/tx?
18:39.14*** join/#asterisk Inkubot (n=pancho@200.75.4.7)
18:39.37pauldyg__, resource allocation and call management
18:39.47pauldyotherwise just use a ringgroup and be done with it
18:40.13pauldyplus I think they only way you cna have hold music in asterisk is in a queue, is that correct?
18:40.17Inkuboti can't dial trough a sip proxy to the pstn
18:40.29Inkuboti don't know what it's wrong
18:40.30jake1932just saw this start - my screen repaints the CLI output every so often to the term window
18:40.45jake1932was this added in HEAD recently?
18:40.45conveyI need help.. lol
18:40.47g__pauldy: no, you can put someone on hold as well
18:40.54malverian[work]g__, Hmm.. maybe it could be the program hitting a stack limitation?
18:40.57g__Or just play them hold music for the fun of it.
18:40.59ManxPowerpauldy, see the "m" option to "show application dial"
18:40.59malverian[work]g__, Eg. ulimit -s
18:41.00pauldyI mean while you ring the ext
18:41.14pauldyoh yea forgot all about that thanks MP
18:41.25blitzragebeing on hold r0x
18:41.56jake1932blitzrage: that all depends on the music
18:42.11g__malverian: anything's possible.. for starters, what kernel are we talking about?  Linux, FreeBSD, Darwin.. and what version?
18:42.13blitzragejake1932: yah... I should have added an </sarcasm> tag :)
18:42.19pauldyI like calling in via 7777 dialing my own extension then putting myself on hold
18:42.21pauldyit soothes me
18:42.29jake1932being on hold with metallica r0x
18:42.30blitzrageyou're all sick, you know that right? :)
18:42.41blitzragejake1932: I hate metallica (mostly for political reasons)
18:42.44ManxPowerjake1932, only if you are using an uncompresed codec
18:42.49jake1932lol
18:43.17AgiNamumetallica on ipc10 is awesoem
18:43.28JerJerLPC10
18:43.36AgiNamuyea yea thats what I meant
18:43.47AgiNamuwhats even cooler is that with LPC10 is that me just talking sounds like metallica
18:44.12JerJerstrangely enough we do have a few ppl that regularly use LPC10
18:44.20AgiNamus/whatever I wrote/what I meant to write
18:44.33*** join/#asterisk tdonahue (n=tdonahue@64.201.13.50)
18:44.42malverian[work]pauldy, # for a in `seq 1 10000`; do ps ax | cut -d' ' -f1 | grep $RANDOM | grep -v grep; done | sort | uniq -u | wc -l
18:44.42malverian[work]4
18:44.45malverian[work]pauldy, ;)
18:44.46*** part/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca)
18:46.07*** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net)
18:46.11*** join/#asterisk denon (i=denon@synapse.subneural.net)
18:46.11*** mode/#asterisk [+o denon] by ChanServ
18:46.27pauldywhats the point of that
18:46.40tdonahuehi all, i just got asked a question that I have no idea how to answer.  how can I have a receptionist transfer directly to someone's voicemail box
18:46.41*** join/#asterisk abcd100 (n=abcd@61.16.172.254)
18:47.16brookshireyeah
18:47.18pauldy*<ext>
18:47.20arp2malverian, you dont do shell scripting for a living do you? :)
18:47.37pauldytdonahue, for most close to default setups it will be just * plus the ext
18:47.44Kattywhat do you call someone who develops dialplans?
18:47.46JerJertdonahue:  have a direct to voicemail extension
18:47.59arp2katty, communications specialist?
18:48.04JerJerexten => 6XXX,1,Voicemail,u${EXTEN:1}
18:48.06Kattyhmmmmmmmmm.
18:48.08CoffeeIV_tdonahue: you can (or may already -- it's in the sample configs) have the extension *123 be the voicemail for *123
18:48.08pauldykajtzu, a dial planner
18:48.09[hC]Katty:  a sucker?
18:48.09JerJeris there an echo in here?
18:48.21brookshireahaha.. a dial planner
18:48.25tdonahuethanks guys
18:48.28brookshirethat's great
18:48.31fugitivolol
18:48.32CoffeeIV_tdonahue: so then she just transferes to *xxx
18:48.39*** join/#asterisk dabigshiznizzle (n=dabigshi@gwsecure.sctelcom.com)
18:48.54JerJerwhere XXX is their three digit mailbox
18:49.02JerJermodify accordingly, of course
18:49.32}btorch{bjohnson: I though asterisk is the one that takes care of changing SIP registration status from OK -> UNREACHABLE -> UNKOWN
18:49.43kajtzupauldy: ?
18:51.10pauldykajtzu, can I get an ahmen
18:51.52netnameusanyone have any opinions on the Grandsteam Budgetone 101 IP phones?  Good or bad
18:52.08JerJeryou mean BarbieTones
18:52.16netnameuslol
18:52.20denonnetnameus: the good is that they're cheap
18:52.21spacklenetnameus: they should be the fisher-price barbietone
18:52.22pauldybetter than a hot poker in your eye but don[t get your hopes up
18:52.23denonthe bad is that they're cheap
18:52.44kajtzupauldy: anytime :)
18:52.47netnameusthat's what i thought... just wanted to make sure
18:52.51denonpersonally I've had pretty decent luck with them, but I wouldnt say that's the norm
18:53.01denonand I wouldnt trust them to anything important
18:53.03spacklenetnameus: they are link training wheels for Asterisk
18:53.07pauldyI spent a little more for the gxp-2000 still not happy with how it works
18:53.09spacklenetnameus: they are like training wheels for Asterisk
18:53.10denontheir speakerphone isnt even usable
18:53.54netnameusha, well i guess i'll just use an spa-1001 like i had originaly planned
18:53.55dabigshiznizzleI have been using the Snom 190s with good luck...If you don't bother the handset falling off the phone from time to time
18:54.30rkingwhat symptoms are there for using a software zaptel emulator for iax conferences (via the meetme app)?  i don't have a card, and i'm curious what buying one might do.
18:54.53pauldynetnameus, for me it was more important I had something that looked imrpessive sitting on my desk in my office than the actual functionality
18:54.55pauldyYMMV
18:55.06spacklerking  - ztdummy is the no-card substitute
18:55.20cpatryis there any cheap 2 FXO ports ?
18:55.24cpatryATA
18:55.48rkingspackle: right - and i'm just wondering what benefit i'll get from an actual card.
18:55.55pauldycpatry, for a phone or phone line to plug into
18:56.32spacklecpatry, It all depends on what you want to do.
18:56.59cpatryfor a line
18:57.16cpatrypauldy: if its a phone, that would be FXS.
18:57.20pauldyhaven't seen anyhing under the 100 dollar mark yet
18:57.36cpatrypauldy: which 2 ports FXO is at 100$?
18:57.37pauldycpatry, understood making sure you did also
18:57.51bjohnsoncpatry: 2 spa 3000's are about as cheap as fxo get for small installs
18:57.54pauldyno it was like 130
18:58.07pauldycheck froogle on google thats how I found it
18:58.20cpatryi dont need any FXS, just FXO, and spa3000 is 1 o , 1 s
18:58.21pauldyvoip fxo
18:58.34bjohnsoncpatry: 2 port fxo digium card is competitive, but not ATA as you specified
18:58.48cpatrypauldy: which model did u get?
18:58.52bjohnsoncpatry: yes.  you would need to buy 2 spa 3000
18:59.04cpatrybjohnson: i would prefer another solution.
18:59.15bjohnsoncpatry: then you will pay more
18:59.20cpatrypauldy: i dont want asterisk there if possible.
18:59.30cpatryfile: u know any solution?
18:59.31*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
18:59.55file[laptop]not really
18:59.58file[laptop]:(
19:00.00cpatryi saw 4 fxo, but its like 800$
19:00.04bjohnsonexactly
19:00.06blitzrageanyone know how I can debug DTMF? For some reason I'm getting duplicate digits when calling into VoicemailMain() using RFC2833
19:00.16cpatrybjohnson: u know something cheaper?
19:00.18spackle....and pulls out nothing.
19:00.32spackle8-(
19:00.32bjohnsoncpatry: yes.  as said, cheapest is 2 spa 3000
19:00.36azzieblitzrage, tcpdump is your friend
19:00.49blitzragetopology is:  PRI --> Asterisk 1.0.x --> billing box --> Asterisk 1.0.x (#2) --> VoicemailMain()
19:00.53cpatrybjohnson: but 4 fxo would requires 4 spa3k.
19:00.57pauldycpatry, I didn't purchase but I came across it
19:01.07bjohnsoncpatry: and would still be cheaper than $800
19:01.12blitzrageazzie: yah - I figured I should have seen it in a SIP debug, but no go - RFC2833 go on a different port?
19:01.16bjohnsoncpatry: plus .. you said you only needed 2
19:01.24cpatrybjohnson: true.
19:01.27cpatryive different sites.
19:01.34azzieblitzrage, rfc2833 goes inside RTP packets. You need to watch for RTP
19:01.43cpatrythat would be great seeing an ATA at 2 fxo.
19:01.45blitzrageazzie: ahhh yes... I'm so dumb today
19:01.52pauldyyou would be cheaper just buying two atas
19:01.58blitzrageazzie: much obliged :)
19:02.14azzieblitzrage, and get ethereal - it shows RTP nicely, you'll be able to spot digits from codec packets
19:02.21cpatrypauldy: i never saw a 2 ports FXO ata
19:02.23bjohnsoncpatry: you can get multiple fxo units
19:02.33spacklefour ports is common
19:02.33bjohnsoncpatry: no, they are not cheaper than 2 spa 3000
19:02.44blitzrageazzie: yah, I have both of those - guess I need to dump tcpdump into a file, then download that from the server (console only) and analyze it in ethereal
19:02.57blitzrageprobably my best bet
19:02.58malverian[work]arp2, Of course not :-P Why do you ask?
19:03.05azzieblitzrage, would work... there's text ethereeal too btw :)
19:03.24blitzrageazzie: yah... seems to be a bit harder to watch though :)
19:03.25*** join/#asterisk advorak (n=advorak@12-211-14-43.client.insightbb.com)
19:03.37blitzrageazzie: the gui version is niiiiiiice
19:04.10cpatrybjohnson: if if see some ATA at 2 fxos, let me know please.
19:04.26azzieblitzrage, so, you'll see RTP type 101 many times for each digit, and RTP 101 with "(end)" marker (usuall 3 times).
19:04.53blitzrageazzie: right - thanks much
19:05.13pauldycpatry, don't forget you can also get fxs to fxo converters
19:06.01*** join/#asterisk advorak (n=advorak@12-211-14-43.client.insightbb.com)
19:06.27malverian[work]pauldy, Just that in 10,000 loops through it found 4 valid running processes using $RANDOM ;)
19:06.34bjohnson2 fxo and 2 "bonus" fxs
19:06.58spacklebut wait, there's more!  Buy now and get double power supplies
19:07.04bjohnsonpauldy: I thought they only came in fxo to fxs versions
19:07.21*** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
19:07.48*** part/#asterisk delink (n=delink@ziegchen.delink.net)
19:07.51bjohnsonand 2 cat 5 cable
19:07.58pauldybjohnson, http://worldcall.brinkster.net/pcphoneline/fxsfxo.htm
19:08.13spackleand a coupon for 10,000 free packets.
19:08.24cpatrypauldy: ya true. thx for hints.
19:08.33pauldymalverian[work], aah well in an infinite loop it will likely kil the box pretty quick
19:08.50*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
19:08.53spackleuse the xml lathos42
19:09.23g__Um, interesting.  It appears that queues still accept callers, even if none of the agents are logged in.
19:09.28lathos42spackle: I tried setting it once without success, but I didnt try very hard at that point
19:09.50bjohnsonpauldy: not bad for $39
19:10.02bjohnsonpauldy: kind of like a dumb 2 port fxo device
19:10.32cpatryany suggestion on good fxs->fxs converter?
19:10.38pauldycan't say I would ever use it but there it is for someone who absolutly must have it
19:10.53bjohnsoncpatry: phone line extension cord from dollar store
19:10.55*** join/#asterisk mkrufky (n=mk@68.160.103.77)
19:10.59bjohnsoncpatry: today only $5
19:11.05bjohnson(plus shipping and handling)
19:11.18cpatryhuh!
19:11.42bjohnsonfxs to fxs converter?
19:12.01cpatryfxs->fxo converter, sorry boi :)
19:12.02tzangerbjohnson: called a piece of telephone wire :-)
19:12.55bjohnsontzanger: with a lump of plastic attached for "marketing" purposes
19:13.26tzanger:-)
19:13.39bjohnsonpatented plastic of course
19:13.39pauldyto many acronyms tla here tla there I got an fla for that that describes little or nothing of what the actual functionality is
19:13.58bjohnsondon't need any competition to come between me and recovering my R&D costs
19:14.51*** join/#asterisk bumblefsck (n=bumblefs@69-160-158-193.ontrca.adelphia.net)
19:15.25*** join/#asterisk mrlovatt (n=fgd@62-30-33-29.cable.ubr01.pres.blueyonder.co.uk)
19:15.27*** join/#asterisk hellagony (n=egutierr@200.202.206.247)
19:15.29empwhat would cause overlap in a call recording between two parties?
19:16.43cpatrybjohnson: huh? dollar-store?
19:17.20empcpatry, a store where everything they sell is $1
19:17.55Beirdoor $2
19:18.02Beirdo(these days)
19:18.31spackleBeirdo - It's getting to be you can't afford to change your mind anymore
19:18.36Beirdoyeah
19:18.37Drukendollar stores rule!
19:18.47*** join/#asterisk ClubBarf (n=spamme@host-87-74-0-72.bulldogdsl.com)
19:18.53Beirdodollar stores are great ways to buy CHHEEEEEEP crap
19:18.53ClubBarfHey ppl!
19:18.54Beirdo:)
19:19.06mutilatoryea.. crap
19:19.19DrukenBeirdo: dollar stores are a great place to buy shit you don't care if it gets broken
19:19.28Beirdobingo
19:19.44mutilatorheh i care if my dollar store items are broken
19:19.47mutilatormeans i have to spend more money
19:19.48ClubBarfAnyone here know anything about connecting INDeX telephones to *?
19:19.50Drukeni buy all my glasses and the like there
19:20.00Druken3 cups for a buck.. how can ya go wrong?
19:20.12mutilatorbreak all 3 cups
19:21.34mutilatoromg
19:21.35ClubBarfI guess that's a no then...
19:21.36ClubBarf:p
19:21.44mutilatori just got a job offer for a 75k/yr job outta the blue today
19:21.54mutilatorso i emailed the guy back askin where he got my info
19:21.57mutilatorand i get the reply..
19:22.04mutilatoryou had been in touch with us in '04 about a prospect with Pfizer...but this was sent in error...there is a candidate with a similar name in the NYC area...It was meant for him and his skill set...I apologize...
19:22.04mutilator<PROTECTED>
19:22.05mrlovattany1 know a good cheap voip provider in the uk.. or how to use skype as a channel on a astersik pbx?
19:22.34spacklemutilator: look on the bright side:
19:22.50*** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca)
19:22.51ClubBarfmrlovatt - I use pipecall.  They do nice, cheap 0845 numbers and have unlimited call accounts.
19:22.52spacklehe didn't read about you in "ass-whores daily"
19:23.06Drukenmutilator: ya stupid ass, i would have just said yes :)
19:23.09mutilatoryea, i use my other email address for that ;)
19:23.15spacklelol
19:23.21mutilatorDruken: yea. i also mentioned that in the email too
19:23.32ClubBarfAnd you can't use skype as a channel on asterisk, skype won't connect to any other VoIP systems.
19:23.32mrlovattthanks i'll look into them
19:23.34ClubBarfYet.
19:23.45mutilatorsorry about the confusion..thanks for bringing it to my attention...
19:23.45mutilatorRegards,
19:23.45mutilatorJoe
19:23.48Drukenmutilator: oh... well that sucks... i'd be pissed :)
19:23.51}btorch{are there any other free Voip phones better than X-Lite ?
19:23.57mrlovattno1's created a cheeky little hack yet :P?
19:23.58ClubBarfOh, and I hear Dixons is going to do a supercheap VoIP service too, but I don't know any more about that.
19:24.00mutilatori think this "guy with a similar name" in nyc owes me big time
19:24.08mutilatori coulda just got him a nice job
19:24.32Drukeni say ya go on a manhun for him...
19:24.39ClubBarfmanhun?
19:24.40mutilatorheh
19:24.40Drukenlets say 20g a year?
19:24.43ClubBarf:p
19:24.48Drukenmanhunt
19:24.56ClubBarfI liked manhun better.
19:24.58mutilatori could do with an extra 5g a year
19:25.05*** join/#asterisk hellagony (n=egutierr@224-123.81-161.gts.tkb.net.pl)
19:25.13Drukenwell, then he's still getting 70 :)
19:25.20mutilatorgives ya more of a comfort level over my 15k i currently make
19:25.23ClubBarfKinda like a german WW1 pilot's name.
19:25.31*** part/#asterisk trig (n=jb@xob.neospire.net)
19:25.42Drukenmutilator: i claimed... 11.5k last year
19:26.02mutilatoryeh i had almost 13
19:26.17mutilatoryea buddy
19:26.24bjohnsonmrlovatt: don't count on it.  skype isn't interested in connecting to other systems and other systems aren't interested in connecting to them
19:26.24ClubBarf"Und zis is ze ace pilot Manhun, unt he vill be szchuutink down ze englander pigdogs"
19:26.47mutilatormanthunt england pipedogs?
19:27.17spackleManhun von rickitybritches
19:27.25mutilator.. um just by the off chance..
19:27.34mutilatoranyone in here ever built an accounting system to do financing?
19:27.44ClubBarfja, das ist him, ja!
19:27.45mutilatorand billing
19:28.04Drukendoubtfull
19:28.24ClubBarfAnyone here connected INDeX systems to *?
19:28.43mutilatorwell i gotta integrate financing into my current system, and well it doesn't blend together very well
19:28.49spackleanyone here ever go outside during the day in summertime?
19:28.51mutilatorwondering how others might have built theirs
19:29.14Drukenspackle: you mean, enter the bright light?
19:29.24bumblefsckmutilator, define billing/financing.  are you talking double entry?
19:29.24spackleit has chosen us.
19:29.34spacklethe big blue room
19:29.42cpatrybjohnson: and whats the relation of the dollard-store with the fxs->fxo converter?
19:30.03bjohnsonno idea
19:30.03spackledullard store - I like that.
19:30.10bjohnsonbut they have lots of fxs-fxs converters
19:30.31bjohnsonAND fxo-fxo converters
19:30.31cpatryfxs-fxs :P
19:30.40ClubBarfWhat's an ISDN FXS called?  Is it still an FXS?
19:30.45cpatryi need fxo->fxs
19:30.47mutilatorwell we sell the customer their hardware at say $900, then they pay $20.50/month for 18 months or something until it's paid off
19:30.50*** join/#asterisk eziman (i=superop@64.116.231.226)
19:31.11ezimanasterisk:/lib/modules/2.4.31-1-686/misc# modprobe ztdummy
19:31.11eziman. /lib/modules/2.4.31-1-686/misc/zaptel.o: /lib/modules/2.4.31-1-686/misc/zaptel.o: unresolved symbol __stack_smash_handler
19:31.23mutilatorproblem i run into, that initial 900 is already billed into our AR and i can't add the 20.50 that shows on their bill to the AR each month, only the interest thats incorporated into the 20.50 can be added to AR
19:31.24ezimanideas for solving that ?
19:31.40mutilatorso i need to invoice for the interest but still show they need to pay their 20.50
19:31.53bjohnsonahhh .. the stack'em and smash'em handler
19:32.22bjohnsonI have a son that does that
19:32.44bjohnsonalso a mailman that does that
19:33.31ezimanso? any fix for that ?
19:34.09*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
19:34.15bendy24duct tape or some rope may help
19:34.22Kattymew.
19:34.27psypeterawr
19:34.32Katty...
19:34.37psypeteso my company has an old shitty key system
19:34.41psypetewith lotsa phones
19:34.46Kattytwisted[asteria]: wake up!
19:34.48bendy24bjohnson: although your son may squirm his way out of that
19:34.50psypeteis there a place i can sell my phone system to?
19:34.52Kattytwisted[asteria]: no napping!
19:35.04bjohnsonpsypete: ebay?
19:35.07psypeteso i can then afford the PRI cards for Asterisk
19:35.14psypetebjohnson: well besides them
19:35.30bjohnsonyou might want to set up the new system before you remove and sell the old one
19:35.36psypeteand i'd like to make a decent amount for it
19:35.39rkingbjohnson++ # thinking like a pro
19:35.42psypeteyeah i was kinda thinking the same thing bjohnson
19:36.04psypetemy coworkers might have a word or two with me if the phones are gone for a month
19:36.12Beirdootherwise.... print resume first
19:36.13bjohnsonpsypete: no decent amount to be had for used key systems
19:36.16Beirdothen sell phone system
19:36.19psypete"where did the phones go pete?"
19:36.27psypetebjohnson: frowny-face
19:36.35rkingemotitext?
19:36.36bjohnsonor in other-words .. you can only get what they're worth
19:36.53psypetewell think 50 phones
19:37.04spackleor what someone thinks they are worth.
19:37.07psypeteand i guess i could sell the patch panels too...
19:37.13bjohnsonpsypete: many people are interfacing * to their key systems
19:37.16*** join/#asterisk Nix (n=Nix@81.213.125.220)
19:37.33psypetebjohnson: our key system is a bit limited in functionality
19:37.34bjohnsonthen use the same phones and existing wiring
19:37.37spacklepsypete so what is the existing system?  Brands types and models?
19:37.37mutilatorbumblefsck ideas?
19:37.46*** part/#asterisk pigpen (n=mark@fw.seamans.cc)
19:37.54psypetespackle: NEC EliteSomething 192
19:37.54bumblefsckmutilator, honestly, I think you'd be better off writing code to interface with an existing accounting package.
19:38.07bumblefsckwhat do you use to do your accounting now?
19:38.20mutilatormy code
19:38.25mutilatorbeen using it 2 yrs or so
19:38.34bjohnsonmutilator: accounting packages are complicated.  make an interface to an existing system
19:38.38bjohnsonlook at sql-ledger
19:38.56psypeteand BTW, a SIP phone, does it give you the more advanced options of other phones, like menus and stuff?
19:39.11bjohnsonpsypete: if you pay enough
19:39.14mutilatorcurrent system has been workin great
19:39.17psypete:|
19:39.17ClubBarfpsypete - depends on the phone.
19:39.19*** part/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
19:39.20mutilatorjust financing doesn't fit well into it
19:39.25mutilatori guess i can check out options
19:39.31psypetewell like
19:39.35ClubBarfI have a basic SIP phone that is nothing more than a very basic phone.
19:39.49psypetei want a cordless phone i can take multiple calls on and transfer to different extensions
19:39.55Katty<coworker> zomgpolycomsarecrapcausethere'sechoonthesecalls
19:39.56ClubBarfI also have a Polycom IP300 on my desk that'll do everything bar the washing up.
19:40.00psypeteand i want it to work with asterisk
19:40.10bjohnsonKatty: shout in their ear and point out the "echo"
19:40.12Katty<coworker> zomgciscophonesatOTHER$companydon'tdothat. lolzkthxyousuck
19:40.22spackleand smell pretty in the springtime
19:40.27ClubBarfPsypete - a multi-line capable SIP phone that uses 802.11b?
19:40.46psypeteKatty: i think you messed up your variable
19:40.46bjohnsonpsypete: multiple calls at the same time, or one at a time?
19:41.03*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com)
19:41.17Kattypsypete: go away.
19:41.31psypetedon't hate me because you suck at coding :)
19:41.33spackleI hear an echo.
19:41.41Kattypsypete: don't bitch at me because you don't get the language i'm using.
19:41.49*** part/#asterisk eziman (i=superop@64.116.231.226)
19:41.51psypetebjohnson: well like... for a receptionist
19:41.54*** join/#asterisk mitcheloc (n=mitchelo@69-169-28-46.anhmca.adelphia.net)
19:42.11bjohnsonpsypete: just get an ATA and use a regular cordless phone
19:42.12Kattywhy does everyone presume i'm dumb?
19:42.15psypetewhy language differentiates between $company in a string "$companydon'tdothat" ?
19:42.16Kattythat really pisses me off sometimes.
19:42.23ClubBarfWhy would you want a cordless phone for a receptionist?
19:42.33psypetei don't presume you're dumb, you just did a dumb thing
19:42.38bjohnsonpsypete: or a SIP phone that can take a headset (BT or otherwise)
19:42.43makhtaryou know what they say about presumptions
19:42.50spackleClubBarf - she can fly so she's away from the phone a lot
19:43.08ClubBarfAh, he's got supergirl for a receptionist?
19:43.11psypetebjohnson: but i need to like, be able to park calls and call up certain lines again, access voice mail via the buttons like on our existing phones, etc
19:43.15bjohnsonme wonders wtf psypete is talking about
19:43.26Kattypsypete: i'm trying VERY hard not to be rude to you
19:43.30bjohnsonoops .. forgot the slash
19:43.31psypetetheres a particular provider which basically provides this all-in-one managed service i've been looking at
19:43.43psypeteit's expensive but its cool
19:43.48Kattypsypete: i'm going to POLITELY suggest you drop the topic and move on.
19:44.03ClubBarfKatty - I think he has...
19:44.11Kattyexcellent.
19:44.14KattyNEXT
19:44.23ClubBarfpsypete - Why do you want the phone to be cordless?
19:44.24psypetefor instance you take your phone to your house, plug it into your router and you're connected to the office pbx
19:44.39bjohnsonpsypete: ANY phone can do all that .. phone model justs control whether there is a special button for it or not
19:44.43psypetei want it to be cordless because our dumb receptionist keeps leaving her desk and we never get any calls
19:44.52bjohnsonpsypete: and a cordless phone won't have ALL the buttons
19:44.59mitcheloclol
19:45.11psypetebjohnson: right, i'd rather there be special buttons for certain features, and have those buttons supported with asterisk or whatever
19:45.17bjohnsonpsypete: get a BT headset
19:45.37ClubBarfI think you're looking at a Cisco 7920.
19:45.39psypeteerr... once i get her the headset, what's it gonna connect to?
19:45.51bjohnsonpsypete: a decent desk phone
19:45.52spacklehmmm, technology to fix a personnel issue.
19:46.04psypetea bluetooth headset?
19:46.10bjohnsonspackle: it's been done since the beginning of history
19:46.12psypetethat won't reach to her desk when she leaves it
19:46.21spacklepsypete, where is all the moeny for this coming from again?
19:46.24bjohnsonpsypete: how far is she going?
19:46.42bjohnsonpsypete: maybe you need a cellular network
19:46.42psypetebjohnson: say 100-200 feet
19:47.03psypeteBT typically only reaches like a couple meters
19:47.16bjohnsonpsypete: how many button are you picturing on this wifi cordless phone?
19:47.17psypetenot to mention has no special buttons
19:47.28mitchelocCan someone help me figure this out....I need to collaborate with an associate in Russia and his ISP blocks VOIP calls.....is there anyway to change up the ports or something?
19:47.29psypetebjohnson: 14?
19:47.35ClubBarfpsypete Get an 802.11b/g access point and a Cisco 7920.
19:47.40spacklehow about a leash?  That's a kid of technology, isn't it?
19:47.56bjohnsonspackle: I was thinking a club
19:48.01ClubBarfSecretary on a leash?  Spackle, how S&M of you.
19:48.14mutilatorany other systems you recommend i should look at other than sql-ledger?
19:48.25CoffeeIV_get a cordless phone with a belt clip and headset, and teach her to park/transfer without specail buttons
19:48.26spackleI thought it sounded practical.  Maybe hobbles instead?
19:48.30bjohnsonmutilator: there are thousands of acounting systems out there
19:48.37mutilatorexactly
19:48.45mutilator"you recommend
19:48.46mutilator"
19:48.53ClubBarfSpackle, maybe just spank her alot.
19:48.57bjohnsontotally depends on your needs
19:49.06bjohnsonspankle?
19:49.20ClubBarfYup, that'll do.
19:50.05[ViRii]whats this feature for asterisk E911?
19:50.22psypetewell if i'm going to use a standard cordless phone, is the calling party gonna hear the receptionist typing in the transfer code or whatever? i don't want it to be like a hack job or something if possible
19:50.40ClubBarfI still think 802.11x and a wifi voip phone would suit psypete's needs best.
19:50.58bjohnsonpsypete: compared to the existing system where the receptionist doesn't answer the phone?
19:51.21psypetei think my boss would rather a clean solution or none at all
19:51.29bjohnsonpsypete: replace her with an ivr
19:51.32psypeteso far it's just me pissed that my calls aren't getting transferred in
19:51.35psypeteeww
19:51.43dabigshiznizzleHas anyone had any luck getting the vmail.cgi to work?  I can get my and listen to my messages, but if I try and delete or move them I get a Software error:  Invalid Context....Any suggestions?
19:51.54bjohnsonClubBarf: depends on his budget
19:51.58psypeteClubBarf i've already responded to them
19:52.03hypa7iabjohnson: "leave me alone or i will replace you with a very simple IVR"
19:52.26[ViRii]whats the difference between alsa and oss?
19:52.31bjohnsonan engineering buddy once told a technician that she could be replaced with a spreadsheet
19:52.33ClubBarfNo you havn't.
19:52.36psypeteone sucks, the other blows, [ViRii]
19:52.50[ViRii]psypete: haha
19:53.16psypeteone is new the other is old
19:53.20Kattywhy do incoming analog calls have echo?
19:53.21psypeteone is complicated one is simple
19:53.27Kattyand not outgoing analog calls?
19:53.38Kattylike that 2 second echo bit
19:53.41psypetethe incoming calls are making fun of you
19:53.41Kattybefore asterisk chops it off
19:53.51KattyHmmhesays: explain this to me.
19:53.58ClubBarfAnyone here know anything about INDeX systems?
19:54.04*** join/#asterisk SplasPood (n=jwb@dsl081-201-143.nyc2.dsl.speakeasy.net)
19:54.22bjohnsonKatty: do you have echotraining on?
19:54.29Kattybjohnson: this is NOT an issue
19:54.30mutilatori'll have to use that sometime bjohnson
19:54.31Kattybjohnson: i want to know why
19:54.40psypetehmm... this cisco 7920 cordless phone says it supports XML applications to the display? is this something Asterisk supports?
19:54.52ClubBarfNo idea.
19:54.58ClubBarfDo you think you'd be using that?
19:54.58bjohnsonKatty: well IF you have echotraining on, it takes a few seconds to process it
19:55.03[ViRii]my polycom 600 supports xml
19:55.07Kattybjohnson: i don't care about processing it
19:55.11Kattybjohnson: i want to know /why/ there is echo
19:55.13mitchelocNo, it's something you program yourself ;).
19:55.14mutilatorthe support/sales team "watch out, i can replace you with a simple IVR"
19:55.23mitchelocJust point it to a webserver and start serving up xml instead of html.
19:55.27bjohnsonKatty: to much volume from the remote end
19:55.32mitcheloc(to psypete)
19:55.33bjohnsontoo
19:55.41psypeteClubBarf: the XML applications could do the more advanced feature crap i want though, right?
19:55.49bjohnsonyour signal is bouncing back
19:56.04ClubBarfViRii - I've been sent a Polycom IP300.  Dear god, have you ever come across a config file quite that complex?
19:56.05spackleKatty: there are many causes of echo.
19:56.09Kattyspackle: elaborate.
19:56.20bjohnsontypically playing with gain will help
19:56.22ClubBarfPsypete - I don't think XML is the solution you're looking for, no.
19:56.25psypetemitcheloc: oh it's HTTP based?
19:56.33bjohnsonand turn echotraining off
19:56.33Kattyspackle: from what i understand, there should be no echo
19:56.38spackleKatty: high latency can cause echo, conversion from analog to digital can cause echo.
19:56.42Kattyspackle: if pots lines terminate at the 21x before they go to a client.......
19:56.53Kattyspackle: then why is there 2 seconds of echo?
19:57.02ClubBarfpsypete - I think you're looking for a phone that can handle more than 1 line, which isn't a terrably hard thing to do.
19:57.19psypeteyeah but not multiple analog lines
19:57.19bjohnsonpsypete: and typically not needed
19:57.37ClubBarfI think you're looking for the Cisco 7920.
19:57.44ClubBarfI would email cisco about it.
19:57.49spacklekatty, what end is the echo on?
19:57.52psypetewell with our existing key system we have four buttons at the top of an LCD display which act as an interactive menu, which Cisco phones support iirc
19:58.00Kattyspackle: when we get a call INTO the building
19:58.08bjohnsonpsypete: a one line phone can easily have asterisk pick the first line that is available = multi line phone
19:58.09ClubBarfzyxel also do a wifi sip phone, but I don't know if it supports multiple lines.
19:58.11psypetei'd like to emulate that sort of function so people don't have to listen to options
19:58.25spackleand the echo is on your end or the other end?
19:58.34Kattyspackle: our end
19:58.47ClubBarfAh, well Cisco phones let you remap buttons, but I don't know if you have enough spare buttons on a 7920...
19:58.48Kattyspackle: echo cancelation kicks in in about 2 seconds and it's fine
19:58.54Kattyspackle: i just want to know /why/ it takes 2 seconds.
19:59.05spacklekatty, it has to train.
19:59.06bjohnsonthat's the "training" part
19:59.08psypetemaybe i could build a wifi phone with Gumstix and some 802.11b chippy thingy...
19:59.10ClubBarfHow many people here use a soundpoint phone?
19:59.11Kattyspackle: train?
19:59.13Hmmhesays<PROTECTED>
19:59.16KattyHmmhesays: me
19:59.20bjohnsonlearning
19:59.23Hmmhesayswhats up?
19:59.23spackleit samples the echo and begins to correct it and tunes until it is canceled.
19:59.32KattyHmmhesays: i'm attempting to understadn why i have echo on these lines
19:59.35KattyHmmhesays: 2 seconds.
19:59.39ClubBarfYo!  Who's using a Polycom Soundpoint IPxxx phone?
19:59.56KattyHmmhesays: and i don't want the answer, oh enable echo cancelation
20:00.02KattyHmmhesays: there is a logical reason for echo
20:00.03Hmmhesaysi was just reading the speech the edonkey guy gave in a senate testimony.... he blasted them
20:00.09ClubBarflathos42 - do you know if there's any apps out there that can front-end the config files?
20:00.09bjohnsonDISABLE
20:00.29bjohnsonKatty: it is your signal bouncing off the far end
20:00.34psypeteKatty: why not google for reasons that echo is introduced, as i'm sure many people have had this problem before and there must be technical explinations of how the stuff works
20:00.35ClubBarfI've been sent an IP300 for review, and the config (at first glance) is a fkn NIGHTMARE.
20:00.40bjohnsonturn off echotraining and adjust the gain until the echo is gone
20:01.02*** join/#asterisk darkskiez (n=darkskie@host86-132-168-185.range86-132.btcentralplus.com)
20:01.04spackleClubBarf: it really makes a lot of sense once you get used to it.  I used to feel the same way.
20:01.06psypetebjohnson: she doesn't want an answer, she wants a reason
20:01.11*** join/#asterisk TheCops (n=dump@206-248-136-187.dsl.teksavvy.com)
20:01.15bjohnsonshe's had it
20:01.16lathos42ClubBarf: I vaguely recall a script that will configure the phone, but I can't remember where its at
20:01.20bjohnsonand the solution
20:01.22TheCopsX-Rob, are you busy ?
20:01.26psypetewell then she just can't be satisfied
20:01.28lathos42spackle:  Have you ever tried to remap one of the keys?
20:01.29Kattypsypete: indeed. and bjohnson gave me plenty of reasons.
20:01.33psypeteLIKE MOST WOMEN ROFL
20:01.37*** join/#asterisk Raceman (n=bla@cust-02-E169.adsl.scarlet.nl)
20:01.46Kattypsypete: wow, you're incredibly annoying (=
20:01.49ClubBarfspackle - No, I get why they did it that way - if I ever have to config 300 phones, they'll be sounpoints.  But for a single phone, it's a damn nightmare.
20:01.54psypeteThanks!
20:01.59Kattymy pleasure.
20:02.04ClubBarfsoundpoints, even...
20:02.14Hmmhesaysecho is usually caused by external impedance imbalances
20:02.19Hmmhesaysnow there is a lot of things that can cause that
20:02.19*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166116178.nb.aliant.net)
20:02.19spacklelathos42: no, I messed with the directory some, haven't hit on the need yet.
20:02.30KattyHmmhesays: how much echo is normal?
20:02.30spackledefine external impedance imbalances
20:02.33KattyHmmhesays: 2 seconds?
20:02.39KattyHmmhesays: .1 seconds?
20:02.41psypetehumidity?
20:02.43Hmmhesaysi don't think there is a "normal"
20:02.55ClubBarfdifferent is normal.
20:02.56bjohnsonyou're not talking about an echo 2 seconds later right .. just for te first 2 seconds of the call
20:02.58lathos42spackle:  Ahh.. I'm trying to change the useless "services" button into a call pickup button, but the phone isnt cooperating
20:03.07Kattybjohnson: indeed
20:03.44*** part/#asterisk dabigshiznizzle (n=dabigshi@gwsecure.sctelcom.com)
20:03.57Hmmhesayscheck out the wikipedia for the definition of impedance and it will make sense
20:03.58*** join/#asterisk dabigshiznizzle (n=dabigshi@gwsecure.sctelcom.com)
20:04.00bjohnsonor 6
20:04.03ClubBarfI just wish Polycom would use FTPS, SCP or something equally secure for EVERYTHING, not just config files on the 301, 501 etc.
20:04.13Kattyi'm fairly happy with the answer i got.
20:04.24Hmmhesaysk
20:04.31spacklekatty: summarize please
20:04.33Kattyit's just lolzenableechocancelation i don't want.
20:04.39Kattyspackle: shan't.
20:04.48bjohnsonDISABLE
20:04.53bjohnsonnotenable
20:04.54psypeteomgkattythatsprettyannoyingafterthe10thtimemmkay
20:04.57Hmmhesaysimpedance is fun to read about, it makes my mind warp
20:05.05Kattypsypete: omgwtfbbqlolzkthxbi
20:05.13fugitivoHmmhesays: docs
20:05.17psypetegay
20:05.19spackleHmmhesays: isn't there a pill for that?
20:05.23Hmmhesaysfugitivo what?
20:05.32Hmmhesaysspackle, LSD?
20:05.35bjohnsonwe've been learning the alphabet at my house using the abc song.  Don't make me sing it here
20:05.46spackleHmmhesays - no that's for BSD
20:05.53HmmhesaysX?
20:05.59bjohnsonscared him away
20:06.23psypetehappy belated birthday
20:06.28ClubBarfOr so some people seem to think...
20:06.49psypetenow on to my next question
20:06.51Hmmhesays"i was born at night but not last night baby" - Kid rock
20:06.53psypeteyou guys will love this one
20:07.05psypeteHow do I spoof Caller ID?
20:07.08lathos42ClubBarf: I've got some land in Antartica to sell you..  Beautiful Sunny beaches as far as the eye can see
20:07.22bjohnsonHmmhesays: I don't think he originated that
20:07.23Hmmhesayswww.engrish.com <-- very funny today
20:07.42Hmmhesaysbjohnson no, but he did make redneck music popular
20:08.14bjohnsonpsypete: use a PRI or a voip provider and set the CID info in your extensions.conf
20:08.24psypeteHmmhesays: kid rock popularized Country?
20:08.31spacklelathos42: there is probably some other silly switch or option you have to set to yes or no before it will all take effect.
20:08.34bjohnsonno, just redneck
20:08.42Hmmhesayshe created a whole new genre
20:08.51psypetewow... i didn't think i would get a serious answer from that, bjohnson
20:08.58spackleI though ashton Kutcher did that.
20:08.59psypeteway to kill the joke :(
20:09.09Hmmhesaysband practice tonight
20:09.12Hmmhesayswoot
20:09.31Hmmhesaysi get to bust out mas tequila
20:09.44spackleand inna gada da vita?
20:10.01lathos42spackle: I'll have to read the guide again..  I'd contact Polycom, but I have this feeling they wouldnt be a whole lot of help
20:10.46spacklelathos42 - I dunno, they have an agreement with Digium now.  Besides, its a phone feature not a pbx feature.
20:11.20spacklelathos42 - Probably have to get support from the reseller
20:12.05Hmmhesaysi need to quit smoking
20:12.10Hmmhesayslike today
20:12.25lathos42spackle:  I guess the worst they could tell me is No
20:12.39bjohnsonHmmhesays: tomorrow is likely soon enough
20:12.51spackleis it from tritechcoa or somewhere else?
20:12.53Hmmhesaysbjohnson: bad
20:13.26spackleHmmhesays: why quit, each one takes off the worst three minutes of your life.
20:13.32bjohnsonClubBarf: not directly
20:13.39hypa7iaClubBarf: you need some third-party hardware to do that
20:13.43Hmmhesaysspackle cause its making me sick and giving me lung problems
20:13.45MikeJ[Laptop]ClubBarf, yes, via bri cards
20:14.00MikeJ[Laptop]although I do not know specifically with those phones
20:14.03lathos42spackle: This one is from tritechcoa, so i'm not sure what kind of support they have
20:14.13ClubBarfMikeJ - do you know of any BRI cards that can handle more than 1 phone per card?
20:14.21MikeJ[Laptop]ummm
20:14.24MikeJ[Laptop]there are several
20:14.29ClubBarfOtherwise it would be cheaper to replace all the phones with sip phones.
20:14.35MikeJ[Laptop]I think there are quad cards supported by midsn
20:14.42MikeJ[Laptop]and then there are the other ones.. hmmm
20:14.45MikeJ[Laptop]let me find them
20:14.53bjohnsonClubBarf: are they already connected to a pbx or key system?  that's where most people try to connect *
20:14.57ClubBarfI was thinking a backplane server filled with quad ISDN cards, but I can't find any quad ISDN cards that are supported...
20:15.15ClubBarfbjohnson - yeah, they're live atm.
20:15.18*** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV)
20:15.33*** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net)
20:15.37ClubBarfThey're connected to an Avaya INDeX PBX at the college I just started working for.
20:15.51bjohnsonClubBarf: consider planting * between the pbx and the telco
20:16.02MikeJ[Laptop]http://www.voip-info.org/tiki-index.php?page=Asterisk+Hardware
20:16.05MikeJ[Laptop]there is a list there
20:16.05spackleMikeJ, earilier sombody was asking if asterisk supported ADSI through an Adit channel bank for Aastra phones.  Any idea?
20:16.34MikeJ[Laptop]ummm
20:16.34bjohnsonI don't think so
20:16.38ClubBarfWell, I'm trying to find out if the hardware the phones plug into could be plugged into an * box directly.
20:16.47MikeJ[Laptop]spackle, adsi should work through a channel bank
20:16.54ClubBarfBut the more I look into it, the more it seems that getting sip phones would be cheaper.
20:16.57MikeJ[Laptop]but I have never used one of those personally
20:17.08bjohnsonreally?  for those analog AASTRAs with the big screens?
20:17.19spackleMikeJ, digium used to sell special Aastra phones, but I don't see that they do any more.
20:17.19MikeJ[Laptop]there are 8 port bri cards too
20:17.24bjohnsonClubBarf: depends on how much wiring is needed
20:17.42MikeJ[Laptop]spackle, not sure about digium selling them, but they are still out there
20:18.00bjohnsonAASTRA also make real VOIP phones
20:18.03hypa7iai've seen some channel-bank like things that suport old nortel and vaya isdn phones
20:18.11hypa7iabut i can't find them  :-(
20:18.31bjohnsonhypa7ia: yeah .. but they're like $100 per port.
20:18.46TheCopsSomeone is using allpage.agi script by X-Rob ?
20:19.05*** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk)
20:19.06*** join/#asterisk raceman (n=bla@cust-02-E169.adsl.scarlet.nl)
20:19.17hypa7iabjohnson: true
20:19.43ClubBarfbjohnson - that's the problem, I can't find the technical docs on the kit we have in place atm.
20:19.46bjohnsonusually better to buy new phones
20:19.53ClubBarfI think that's probably true.
20:20.12ClubBarfWhich is why Polycom have sent me an IP300.
20:20.19hypa7ianice :-)
20:20.22bjohnsonClubBarf: yeah .. typical telco problem.  Sell you the hardware then sell you partial docs .. nobody knows real specs
20:20.39ClubBarfAnd Aastra, Grandstream etc are sending me samples too.
20:20.41hypa7iamy boss was going to throw it out!
20:21.04*** join/#asterisk ^X-works (n=r0x0r@host34-3.pool871.interbusiness.it)
20:21.06ClubBarfWell, I like the idea of going VoIP all the way, with POE etc.
20:21.32*** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com)
20:21.37hypa7iaPoE is nice
20:21.44darkskiezClubBarf: what did you say to get samples?
20:21.46bjohnsonhypa7ia: I thought they weren't very standards compiant and hard to get working
20:21.58hypa7iabjohnson: that would be exactly correct
20:22.03bjohnson"darkskiez: money is no object"
20:22.11*** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
20:22.16hypa7iaClubBarf: look at the Adtran PoE switches.  they are snazzy.
20:22.18ClubBarfI just asked a reseller to send me samples, because we're likely to need about 300 phones soon.
20:22.36darkskiezAh, I may need to do a deployment with 1500
20:22.37ClubBarfThey're not letting me keep them, but it's nice to compare phones side by side.
20:23.05ClubBarfdarkskiez - tell a local reseller that.  They'll bend over backwards for ya.
20:23.14darkskiezto begin with
20:23.26ClubBarfJust don't promise them anything other than to return the phones in good working order.
20:23.53racemanhi all. Anyone experience with the SendUrl() command in extensions.conf ? I tryed it with idefisk iax client, but did'nt received an url
20:24.22bjohnsonnone are voip though
20:24.31*** part/#asterisk dabigshiznizzle (n=dabigshi@gwsecure.sctelcom.com)
20:24.39bjohnsonsince we're providing inventories
20:24.45darkskiezI've read the asterisk dimensioning page, but i'm not sure what you need to run a TE4xxp with 120 channels and 1500 clients.
20:24.45ClubBarfI have 2 voip phones (1 is mine, the other's a loan) and 3 softphone machines.
20:25.03ClubBarfIt's cheaper to buy old PC's to use as softphones than to buy a hardware phone...
20:25.21hypa7iaClubBarf: not when you look at the power bills :-/
20:25.27syleis it
20:25.28ClubBarfMeh.
20:25.30spackledarkskiez>, that sounds more like multiple machines.
20:25.37sylecheckout your electricity bill
20:25.38ClubBarfThose desks needed PC's anyway.
20:25.44racemannobody?
20:25.50ClubBarfFor excell, word etc.
20:26.14ClubBarfNothing fancy, which is why a P2 350 with 256MB ram does the trick nicely.
20:26.24darkskiezspackle: but you can only put a TE4xxp in one machine :/
20:26.25jake1932hmm.  is there a softphone out there for windows that consistantly works?
20:26.28spackleThe Microspendaplenty office menagerie
20:26.45darkskiezspackle: and there will be no transcoding
20:26.50ClubBarfjake1932 - I'm beginning to think... No.
20:27.31spackledarkskiez>, maybe some machines to register to and then another machine to connect to the PSTN, or a two port cad instead of a four port card.
20:27.59darkskiezspackle: could SER handle registrations and run on same machine as asterisk?
20:27.59ClubBarfAnyhew, nice chatting with you fellas.  I'm having a hot bath and going to bed.
20:28.49spackledarkskiez>: file could probably answer that better.  But that's a lot of folk, better to not put all eggs in one basket unless you have a killer support contract on the machine and spare cards.
20:28.54darkskiezspackle: there is a chance i'll need 10 Pri lines with failover machines.
20:29.31spackleClubBarf: TMI
20:30.13darkskiezwas looking at http://www.junghanns.net/en/ISDNguard_produkt.html that, anyone used it?
20:30.17file[laptop]you can run SER on the same machine as Asterisk... just need to bind to different ports
20:30.25hardwiregrr
20:30.29spackledarkskiez>: some kind of a call-center or just the whole company?
20:30.34darkskieza hotel.
20:30.36hardwirethe debian contrib init script exports to LD that we are 2.4.x
20:30.43hardwireand splits asterisk into a milion tiny little pids
20:30.47hardwireon my 2.6.x system
20:31.04spackledarkskiez>: in vegas?
20:31.10darkskiezin london
20:31.23spackledarkskiez>: sip phones in the rooms?
20:31.25darkskiezone there already has 16 PRI lines
20:31.42darkskiezspackle: u want to go and steal them?
20:31.53spackledarkskiez>: not me, but somebody
20:31.55darkskiezThat concerns me too.
20:32.00darkskiezits not a done deal.
20:32.13spackledarkskiez>: we had this discussion a couple months ago here.
20:33.40darkskiezwonder if we can get someone to make a custom firmware that wont upgrade to generic and will be subtly incompatible.
20:33.49darkskiezNeeds a custom sip header or something to work.
20:34.05spackledarkskiez>: but you don't find that out until the phone is gone.
20:34.14bjohnsondarkskiez: SIP ATAs attached to wall with analog deskphones as a better option?
20:34.29darkskiezis there any SIP ATAs that take POE ?
20:34.44bjohnsondarkskiez: that only helps if someone has already stolen them and word has gotten out that they are not usable
20:34.56spackledarkskiez>: Is the thought to deliver phone and IP to the room without extalines
20:34.57bjohnsondarkskiez: make your own POE adapters
20:35.26darkskiezspackle: without extra plug sockets and lines etc.
20:35.34darkskiezATA's are easy to nick too.
20:35.35bjohnsonall it is doing is sending DC over 2 or more of the cat5 lines
20:35.49spackledarkskiez>: what kind of room phones have you spec'd?
20:36.01darkskiezbjohnson: there is a POE protocol to stop damaging non POE hardware, so its not that simple.
20:36.02bjohnsonput in a cat5 wall box and wire out the power lines to a plug into the ATA
20:36.34bjohnsondarkskiez: you would remove the power before it got to the ATA
20:36.35spackleyeah, like those 3com boxes?
20:36.40darkskiezbjohnson: oh. duh. true.
20:36.55darkskiez'those 3com boxes' ?
20:37.18spacklethey take POE, and are a switch or hub that fits in a standard wall-jack box.
20:37.28spackleoffer 3 or 4 lines out
20:37.38darkskiezOh, aye, them.
20:37.51darkskiezbut with an ATA built in.. drool.
20:38.11*** join/#asterisk shido6 (n=shido@d221-68-210.commercial.cgocable.net)
20:38.13spacklethat would be neat.
20:38.35hardwirestop when convenient
20:38.35hardwirewell
20:38.37spacklerun, don't walk to the patent office.
20:38.39hardwireits convenient for the callers here
20:38.41hardwirebut not for me
20:39.28darkskiezI've been making to make a patch to *. stop after apology
20:39.32spackledarkskiez> I bet some of the phone manufacturers would produce a special firmware in a volume like that.
20:39.38darkskiezplays a sample to all active channels and then stops.
20:39.44bjohnsondarkskiez: http://www.nycwireless.net/poe/
20:39.50bjohnsonI was thinking something like that
20:40.07bjohnsonthen mount the ATA right besdie it on the wall
20:40.31darkskiezbjohnson: Except the POE switch wont deliver the power until after a handshake.
20:40.41*** join/#asterisk Firestorm-voip (n=Firestor@ua-83-227-140-131.cust.bredbandsbolaget.se)
20:40.48darkskiezbjohnson: and I dont want to sit down with a dremel and soldering iron and make 1000 of them.
20:41.03spacklecall taiwan
20:41.32bjohnsonexactly
20:42.15darkskiez00886 and dial random numbers till someone picks up?
20:42.34bjohnsonor put the works, including the ATA into a larger box and just have the cat5 hookups showing
20:43.19bjohnsondarkskiez: there's got to be something similar that is provided by POE switch providers
20:43.36darkskiezi think there is a cisco config command to force poe to be on etc.
20:43.47bjohnsonI like this one http://www.nycwireless.net/poe/lap1000.jpg
20:44.06bjohnsonexcept picture an ATA where the wifi AP is and both mounted to a wall
20:44.07spackledarkskiez: market is about ready for a "disposable" IP phone.  oh wait, what about budgetones?
20:44.27bjohnsondarkskiez: what phones currently in use?
20:44.31racemanAnyone experience with the SendUrl() command in extensions.conf ? I tryed it with idefisk iax client, but did'nt received an url
20:44.42darkskiezbjohnson: this particular spec is for a hotel that isnt built yet.
20:45.12bjohnsonip to rooms for data too?
20:45.14darkskiezso i have to spec hardware and a range of phones etc
20:45.19darkskiezbjohnson: and TV :)
20:45.20spackledarkskiez> - so you can spec a phone that hasn't been made yet?
20:45.32bjohnsonso you would want a hub or switch in each room too
20:45.41darkskiezif the phone can be built in time
20:45.53bjohnsonunlikely
20:46.17bjohnsonor at least .. I wouldn't count on it
20:46.19darkskiezbjohnson: well, a phone with hub built in is whats being looked at.
20:46.21Drukenbjohnson: not a hub or switch, i'd do two seperate networks, one voice, one data
20:46.35bjohnsonwhy?
20:46.42darkskiezdruken: the hub is only at the room level, not a high contention.
20:46.48Drukenless chance of interference
20:46.52bjohnsondo one cat5 to each room and split the voice/data at the switch
20:47.18bjohnsonyou can even limit the data bandwidth
20:47.47spackleyou could do that with CoS, Qos and VLANs too couldn't you?
20:48.05spackledarkskiez>: is there an IP TV?
20:48.07bjohnsondarkskiez: would a hotel desk analog phone be any cheaper than a voip phone anyway?
20:48.32bjohnsonoh yeah .. ip TV might suck up the bandwidth
20:48.34spacklebjohnson: probbaly not  at least by a lot.
20:48.37jake1932some of the cable networks are using "digital" - might be ip based?
20:48.56bjohnsonbut not cat5 that I've seen
20:49.05darkskieza phone timing out could raise a theft alarm
20:49.14*** join/#asterisk ian_k (n=ian@gateway.digium.com)
20:49.20AgiNamuWhen someone says to loop the T1 at our end
20:49.24AgiNamuwhat do we physically do?
20:49.25bjohnsonyou'd need a box to convert the cat5 video stream to something the tv could use
20:49.32Dr_Rayok, how do I write up the e-ink dev kit as something that bossman should buy me
20:49.37spacklemythtv only transmits the "channel" or show you are watching, it could easily share the pipe with a phone call and data given good QoS
20:49.47*** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net)
20:49.54darkskiezthe TV takes up about 4Mbits
20:50.13bjohnsonspackle: control for each TV would be killer with such a configuration
20:50.22spackleDr_Ray call it critical developmental info fodder for future development
20:50.24bjohnsondarkskiez: what hardware does that?
20:50.40darkskiezbjohnson: ours, i work for an iptv company
20:51.02bjohnsoncrap, roll the hub, POE, and ATA into THAT!
20:51.07spackledarkskiez, why don't you make the switchbox into your TVs or settop box then?
20:51.13darkskiezbjohnson: i'd like to but we dont make the hardware
20:51.14bjohnsonyeah yeah
20:51.15bjohnsondo it
20:51.24spacklesomebody does
20:51.29darkskiezbjohnson: amino do
20:51.56darkskiezwe vape their software stack and have our own
20:52.10spacklebummer
20:53.06*** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com)
20:53.50rayvdAgiNamu: make a loopback cable and plug it into your CSU/DSU card on your router
20:53.51darkskiezthey screw under tables and hide away nicely: http://www.aminocom.com/products/ipstb/aminet110h.html
20:54.06AgiNamurayvd oh ok
20:54.30rayvdusually you cross a couple of wires and voila
20:56.26sylewtf is diff between xp pro and xp home
20:56.50mitchelocrunning iis on it and connecting to a domain
20:57.01mitcheloc...well the two features that are important to me
20:57.25syleyou can still share shit right on home? and connect to samba shares right?
20:57.55syleapache > iis
20:58.10hardwireSep 28 12:56:47 WARNING[10117]: chan_zap.c:8288 pri_dchannel: Ring requested on channel 0/1 already in use on span 1.  Hanging up owner.
20:58.10hardwireSep 28 12:57:05 WARNING[10117]: chan_zap.c:8288 pri_dchannel: Ring requested on channel 0/1 already in use on span 1.  Hanging up owner.
20:58.10hardwiredamnit
20:58.31jake1932syle: http://www.microsoft.com/windowsxp/pro/howtobuy/choosing2.mspx
20:58.52jake1932bunch of marketing material mixed in with facts
20:59.34Dr_RayI need to get better at having a slushfund
20:59.37Dr_Rayat work
20:59.39*** join/#asterisk websae (i=websae@207-118-145-168.dyn.centurytel.net)
20:59.52darkskiezhardwire: show channels?
20:59.59hardwirethey all died
21:00.10darkskiezwhats show channels say
21:00.21mitchelocsyle: you can't give up remote desktop either
21:00.24hardwiredarkskiez: nothing
21:00.25hardwirethey all died
21:00.29hardwirea few seconds later
21:00.31hardwireSep 28 12:57:17 WARNING[10117]: channel.c:741 channel_find_locked: Avoided initial deadlock for '0x8730470', 10 retries!
21:00.44darkskiezcvs head? whend u build it
21:00.45hardwiresame thing a little before
21:00.52hardwireupped from today
21:00.55darkskiezdepressing
21:00.56darkskiezhttp://www.voip-info.org/tiki-index.php?page=Ring+requested+on+channel
21:00.57hardwirethis morning about an hour ago
21:01.04*** join/#asterisk DannyF (n=dannyf@c-794fe353.24-0099-74657210.cust.bredbandsbolaget.se)
21:01.04darkskiezIve got a potential fix for it
21:01.15darkskiezhardwire: did u rebuild libpri too/
21:01.20hardwireyeh
21:01.26darkskiezfirst and reiinstalled?
21:01.36hardwirebuild libpri first
21:01.38hardwireinstalled
21:01.40hardwirebuilt zaptel
21:01.41hardwireinstalled
21:01.42darkskiezi described a potential fix there, but i dont know the code well enough
21:01.45hardwireuseing wct1xxp
21:01.52mitchelochow sad that you have to run that script...
21:02.04hardwiremitcheloc: it just started happening today
21:02.05mitcheloci've never need to
21:02.17darkskiezi dont get it very often
21:02.18hardwireafter the ILEC broke our T1
21:02.23darkskiezdont run the script.
21:02.25syledamn
21:02.29hardwirethey unhooked our pairs
21:02.34darkskiezbut I added a comment
21:02.34hardwirethen hooked them back up
21:02.36sylewell wouldn;t mind getting my hands on media center
21:02.43frenzyhello
21:02.44mitchelocwhy would the code freeze anyways? it shouldn't stop asterisk because of something like that
21:02.50mitchelocso why need to restart it..
21:02.55hardwiredarkskiez: thanks for that
21:03.13frenzywhats the simplest way to create an extension and point to to a specific sip ?
21:03.16darkskiezhardwire: I dont run the script...but it might be good for you
21:03.21hardwireno
21:03.25hardwireI think the ILEC can fix it
21:03.37hardwirebut it totally kills asterisk
21:03.40hardwirewhich I dislike
21:03.41darkskiezI think its a bug in asterisk too
21:03.48darkskiezi think it should be handled better
21:03.52darkskiezas i've described
21:04.01frenzyexten => 2000,1,Dial(SIP/2000) ?
21:04.01hardwireyeh
21:04.16hardwirefrenzy: looks fine
21:04.38hardwireI need to get some graphiing going
21:04.44hardwireabout how often our PRI is in use
21:04.48hardwirestuff like that
21:05.13*** join/#asterisk hypa7ia (i=hypatia@silenceisdefeat.org)
21:05.32syleanyone know if windows media center is more like pro or home?
21:05.40hypa7iapro
21:05.41hypa7iaish
21:05.53frenzyhow about if the user is SIP ?
21:06.01frenzywhoops
21:06.05frenzymeaning characters
21:06.14darkskiezasterisk doesnt handle chars so well
21:06.57darkskiezit can though
21:07.02darkskiezbut there are caveats
21:07.06frenzyI see...
21:07.08empcan someone explain how exten => 1234,5,dial(${TRUNK}c/9871234321,20,r) works?  especially the c/ part
21:07.16frenzywhat about priorities ?
21:07.53darkskiezwhat about zebras?
21:08.20spackle"Cheese Grommit."
21:08.24shido6?
21:08.24hardwireewwww
21:08.25frenzy:P
21:08.26*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
21:08.31PupenoLhello.
21:08.39hardwirespackle: did you see the new movie?
21:08.44frenzydoes it really matter setting it to 1 ?
21:09.00spacklehardwire, not yet.  Any good?
21:09.00darkskiezfrenzy: your talking disjointed jibberish
21:09.04hardwiredunno
21:09.30*** part/#asterisk makhtar (n=ageller@mail.bulletinnews.com)
21:09.34spacklehardwire, It's lost it luster a little.
21:09.40hardwire:(
21:09.42frenzyheh
21:10.01spackletwo day to Serenity
21:10.07spackledays that is
21:10.31*** join/#asterisk sudhir492 (n=sudhir@pool-71-114-84-37.washdc.dsl-w.verizon.net)
21:10.32hardwirethat looks interesting
21:10.37*** join/#asterisk rat1101 (n=vinay@ip68-100-31-133.dc.dc.cox.net)
21:10.38hardwirethe captain guy looks like he has a really broken nose
21:10.48darkskiezHhahahaha love the slashdot story about a phone survey to find out if people were getting unwanted calls. Genious.
21:11.13sudhir492On my Asterisk, I see this message quite frequently: stale nonce received from ....
21:11.15spacklehardwire: never saw the TV show?
21:11.20hardwirefirefox?
21:11.22hardwiresomething like that
21:11.26spacklefirefly
21:11.29hardwireyeh
21:11.30hardwirenever saw it
21:12.00darkskiezthere is a firefly torrent doing the rounds
21:12.04spackleI "found" the DVDs ripped onto my hard drive one time and watched them all.  Grew on me really fast.
21:12.04frenzyI get stale nonce all the time too
21:12.10sudhir492I am using Polycom phones,
21:12.19darkskiezahh 90% woo
21:12.23hardwirefrenzy: it happens
21:12.24Dr_Rayqwest - called me a few years ago asking if I wanted so subscribe to a service that stopped unwanted phone calls, "You mean like this one?".  The guy on the phone was a good sport about it. Said yes, I told him I already had a mechanism for dealing with those kind of calls, thanked him and hung up.
21:12.25*** join/#asterisk expressfone1 (n=expressf@62-15-97-163.inversas.jazztel.es)
21:12.29expressfone1Hi
21:12.29sudhir492what causes that, and what does that mean?
21:12.39sivanais there a PRI channel usage grapher?
21:13.20*** part/#asterisk raceman (n=bla@cust-02-E169.adsl.scarlet.nl)
21:13.30*** join/#asterisk Brixius (n=Brixius@162.96.15.48)
21:13.47expressfone1any one can helpme to boot Astlinux on net4801 (CF), getting VFS: Cannot open root device "hda1" or unknown-block(0,0)
21:16.07hardwireits hde
21:16.34frenzysay, any good music on asteirks I can use to test playback ?
21:17.01expressfone1thanks hardwire, let me test it
21:17.54*** join/#asterisk KranZ (n=user@sme.bestline.net)
21:19.08expressfone1hardwire> same error :-S
21:19.15hardwiresorry
21:19.42expressfone1any one running astlinux on soekris net4801 ????
21:20.41spackleexpressfone1: I've had it working in the past.
21:21.12spackleexpressfone1: are you running it off CD or a hard disk?
21:21.18expressfone1CF
21:21.29spacklewhat brand of CF?
21:21.35Connor_CVS-HEAD or 1.2 tar file ?
21:21.42Connor_which one should I use now..
21:21.42expressfone1scandisk 128M
21:22.28spacklethere are bios settings and I think a jumper for boot device settings, have you fiddled with either?
21:23.43frenzySep 28 17:19:21 WARNING[20598]: file.c:475 ast_openstream: File fpm-calm-river does not exist in any format
21:23.44frenzySep 28 17:19:21 WARNING[20598]: file.c:787 ast_streamfile: Unable to open fpm-calm-river (format ilbc): Resource temporarily unavailable
21:23.44frenzySep 28 17:19:21 WARNING[20598]: app_playback.c:83 playback_exec: ast_streamfile failed on SIP/7777777-dcde for fpm-calm-river
21:23.44expressfone1i´m install monowall and run ok
21:23.54expressfone1boot ok from cf
21:23.56frenzyis there a way I can play mp3 on ilbc ?
21:24.03expressfone1my problem is boot astlinux
21:24.39spackleI guess so.  How are you flashing it?  using Windows or linux?
21:25.02expressfone1xp
21:25.44Uberbotshido6?
21:26.20spackleI had trouble last time I tried to flash from XP too.  That's why I don't have it working right now.  Maybe check with the astlinux listserv?  Kristian is really good at getting back to people.
21:26.21CleanerXanyone familiar with pri signalling
21:26.45expressfone1ok spackle
21:26.52CleanerXi've figured out a problem with q932 protocoll
21:26.55CleanerX-l
21:27.01jalsothi
21:27.47jalsotI have a problem with iax2 softphone and ringback tone [hearing local and remote RBT :( ]
21:28.21jalsotdoes anybody have an idea what can be wrong? even if I dial out with 'r' option, I hear 2 ringback tones
21:32.15sudhir492from time to time I cannot make call from my Polycom phone. PAP2-NA does not have the same problem
21:33.50*** join/#asterisk SplasPood (n=jwb@dsl081-201-143.nyc2.dsl.speakeasy.net)
21:33.58KranZany way to force a progress indicator so asterisk generates the ringback when the calling party doesnt supply a progress indicator?
21:34.08KranZhappens with cell phones
21:36.39*** join/#asterisk kshumard (n=root@gateway.digium.com)
21:37.53*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
21:38.06generalhanwhats going on everyone ?
21:39.22websaeanyone here familiar with AstBILL?
21:39.29websaeor has anyone tried it?
21:41.17bkw_frenzy, if you go get format_mp3 from asterisk-addons
21:41.46*** join/#asterisk afrosheen (n=test@txprotoa2.august.net)
21:42.03*** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net)
21:42.36*** join/#asterisk Gerriall (n=NonYa@209.42.198.18)
21:42.36*** part/#asterisk Brixius (n=Brixius@162.96.15.48)
21:43.53generalhanIs there a feature in Asterisk that will allow me to dial a number to listen into someones extension? for like training purposes ??
21:45.04darkskiezgeneralhan: yes
21:45.19bkw_chan_spy
21:45.21generalhandarkskies: can you link me to some info
21:45.26hardwireI have yet to play with chan_spy
21:45.37mitcheloci thought chan_spy was *discontinued*?
21:45.46bkw_its in CVS
21:45.52afrosheenused to use zapbarge but I think it only works on zap trunks
21:45.59generalhanhmm that makes me a bit nervous when people that i see in here ALL THE TIME havent even tried it. makes a little concerned that i might not be able to use it at all ! lol
21:46.03mitchelocoh ok, so its just *undocumented* ;)
21:46.46malverian[work]generalhan, I use app_chanspy
21:46.52malverian[work]generalhan, It works well and is in CVS HEAD now.
21:47.02malverian[work]generalhan, It works on all channels.
21:47.13generalhanhow does that work in comparison to chan_spy ?
21:47.14malverian[work]generalhan, It is "chan_spy" .. app_chanspy.so
21:47.24afrosheenI think chan_spy is the new implementation, zapbarge is old school
21:47.42malverian[work]It's not really "chan_spy" because it is not a channel.. it is an application.
21:47.46generalhananyone have a link that might explain to me how to set it up ? or is it in the wiki pages ?
21:47.53malverian[work]show application chanspy
21:47.57malverian[work]generalhan, ^ from cli
21:48.01generalhanok
21:48.02*** join/#asterisk goatmilk (n=goatmilk@130-127-45-11.chouse.resnet.clemson.edu)
21:48.05*** part/#asterisk mkrufky-not-here (n=mk@68.160.103.77)
21:48.50mitchelocdamnit...i'm running asterisk 1.0.0, bah, i wanted to try chan_spy, it's probably newer then that time period anyways
21:50.10*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
21:50.56generalhanmalverian: i get this when i try the show application cmd : PhoneBox*CLI> show application chanspy
21:50.57generalhanYour application(s) is (are) not registered
21:51.36mitchelocwhat is your "show version"
21:51.42generalhan1.0.9
21:51.58mitchelocthey said it's only in cvs head, that doesn't sound like it's cvs head
21:52.01sudhir492what WiFi SIP phone works well with Asterisk?
21:52.09sudhir492Any inexpensive ones?
21:52.29afrosheenwifi sip? ohhhh...
21:52.38generalhanmitcheloc: who is "they"
21:52.52afrosheenare you talking something like a standard portable phone wifi or an actual cellphone-ish sip phone
21:52.55*** join/#asterisk Moc (n=mochouin@modemcable111.229-203-24.mc.videotron.ca)
21:52.56mitchelocmitcheloc: i thought chan_spy was *discontinued*?
21:52.56mitchelocbkw_: its in CVS
21:53.19wunderkinim getting ready to use it now
21:53.22mitchelocso basically it's not in the release versions, probably for some sort of liability reasons
21:53.30generalhan<malverian[work]> generalhan, It works well and is in CVS HEAD now.
21:53.57generalhanso when i read that i thought he meant that it was working AND is working in CVS HEAD
21:54.27mitchelocwell it wasn't very clear ;), anyhow you'll need cvs head
21:55.01generalhanwell i really dont want to change anything, damnit
21:55.13mitchelocsame here, so i can't use it either lol
21:55.42*** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net)
21:55.48AlexCTIHi..
21:56.21AlexCTICan some one help me to set up the music on hold?
21:57.58generalhan~jbot Dict crack
21:58.13generalhan~jbot Dict crack pipe
21:58.16*** join/#asterisk srt (n=nobody@gw0-cgn.reucon.net)
21:58.19generalhanlol
21:58.23mitchelocis jbot in the dictionary?
21:58.31mitcheloc~jbot dict dict crack
21:58.42generalhan~jbot dict jbot
21:58.57RoyKlol
21:59.08mitchelocit should have errored when it ran into an unknown defition for itself...
21:59.11mitchelocwhat crappy programming
21:59.17RoyK~lart mitcheloc
21:59.50generalhanlol
21:59.51generalhanwhat other kind of commands can he do ?
22:01.08RoyKgeneralhan: people from the .us are evil and deserves nothingg but harrasment
22:01.22FuriousGeorgeit seems the only way to "lower the volume" of my MoH is to resample the tracks and lower the gain(?).  is there any way to tell what the gain is on these tracks now, or do i just shoot in the dark?
22:01.36FuriousGeorgeit is gain, isnt it?
22:03.05afrosheenFuriousGeorge: yeah there's no volume tweaking for MOH, you have to normalize your tracks all by yourself
22:03.16FuriousGeorgei got that part
22:03.27FuriousGeorgeim gonna use sox and everything
22:03.31afrosheenif you're handling mp3's, just decode them all to .wav, normalize it in your encoder
22:03.33afrosheenoh ok
22:04.08generalhanRoyK: whats wrong with people from the .us ?
22:04.14FuriousGeorgeim just wondering what "value" i need to be lowering, whats it called.  surely its not called "colume"
22:04.18FuriousGeorge*volume
22:04.53fugitivoFuriousGeorge: normalize
22:04.54FuriousGeorgegeneralhan: we're far enough from europe to not really care
22:05.16FuriousGeorgefugitivo: thanks
22:05.26fugitivoFuriousGeorge: that´s the name "normalize", some programs have preset values for speech, music, etc
22:06.10fugitivoFuriousGeorge: for speech i think it's -20db
22:06.35FuriousGeorgefugitivo:  its mozart, the stuff you linked me to the other day actually
22:06.40afrosheenyeah, we'll invade in ford taurus' equipped with rocket launchers lol
22:06.43*** join/#asterisk BasketCase (n=BasktCas@asylum.sanitarium.net)
22:06.45RoyKgeneralhan: quite a lot
22:06.50RoyKgeneralhan: sorry
22:06.52Connor_OMFG someone fix tab-complete with the "like" operator for iax2/sip show peers
22:06.56fugitivoFuriousGeorge: what stuff? i don't remember :)
22:07.10FuriousGeorgefugitivo: exactly
22:07.26gordonjcpafrosheen: American tanks that can go about 50 miles on a tank of fuel, assuming they don't just go on fire
22:07.48gordonjcpand then if there is anything like a bend in the road, or a bump, they're stuffed
22:07.51afrosheenwhat is this, a ww2 history lesson of the Shermans?
22:08.21FuriousGeorgeRoyK: generalhan:  i view nationalism like a family, i can make fun of my family, and other people in my family can make fun of us, but outsiders cant
22:08.31gordonjcpafrosheen: it's just extrapolating from the American cars I deal with
22:08.40FuriousGeorgewell, they can, but i reserve the right to not care
22:08.52fugitivoFuriousGeorge: ?
22:08.57gordonjcpI've never seen a car that could set itself on fire by leaking water
22:09.03afrosheengordonjcp: most americans don't even drive american cars much if they can afford not to...and the tanks...well, they're world famous for blowing other things up
22:09.13gordonjcpuntil I started fixing yank tanks
22:09.29FuriousGeorgefugitivo: scroll up for the context
22:09.29*** part/#asterisk BasketCase (n=BasktCas@asylum.sanitarium.net)
22:09.38gordonjcpafrosheen: I'm not surprised, they're just plain dangerous
22:09.49R3DB0xcan anyone tell me anything good or bad about the cisco 7920 phones
22:09.49generalhanRoyK: "quite a lot"??? like what ?
22:09.50afrosheenha, nearly got me
22:10.01fugitivooh well, i only have 10 lines for buffer
22:10.02gordonjcpno brakes, no lights, no suspension
22:10.17gordonjcpslow too
22:10.39FuriousGeorgeone true thing:  they're all automatic
22:11.08gordonjcpyeah
22:11.20*** part/#asterisk alerios (n=alerios@200.24.109.199)
22:11.22gordonjcpthe worst I've ever driven was a Z28
22:11.30gordonjcpquick, I'll give it that
22:11.35gordonjcp0-60 in about 5 seconds
22:11.37RoyKgeneralhan: so far I've met ONE person from the US being reflective
22:11.38fugitivoyou didn't drive a tracker eh?
22:11.46gordonjcp... but that was it
22:11.50gordonjcptop speed 90mph
22:11.51generalhaninteresting
22:12.01generalhanand where are you ?
22:12.08RoyK.no
22:12.32gordonjcp8mpg and getting passed by everything on the road
22:12.36generalhan.no ?
22:12.44KranZnorway?
22:12.52KranZ.yes
22:12.55generalhan.lol
22:12.59r0d3ntSo since when does Asterisk and Digium's hardware have a known problem with PRI echo ????????????????????????
22:13.05KranZ.yesrway
22:13.07afrosheensince awhile
22:13.16gordonjcpit does?
22:13.17r0d3nti loaded zaptel cvs head, no change.
22:13.44r0d3ntafrosheen, no one has reported this until today.
22:13.58r0d3ntthe only pri asterisk installs i have trouble with echo are on telepacific.
22:14.25Connor_msg bkw_ never mind
22:16.10r0d3ntso Chris from Digium said the magical fix for echo on PRI's was in the CVS HEAD of zaptel....
22:16.12r0d3nti loaded it...
22:16.14r0d3ntno change...
22:16.16r0d3ntNONE.
22:16.25drumkillait has a new echo cancellation module
22:16.26r0d3ntand since he told telepacific that this is a known problem....
22:16.28drumkillaer, not module
22:16.35r0d3ntguess what ?!?!?! THEY JUST TOLD ME TO FUCK OFF.
22:16.37drumkillanew software echo can, though
22:17.14KranZdrumkilla: do u have to edit zconfig.h to enable that?
22:17.15r0d3ntwhere is this new software ???
22:17.35r0d3nthe said and I quote " do a cvs check out of zaptel and make install and you're done "
22:17.51r0d3ntand that is supposed to enable this new magical echo cancellation code... ???
22:17.59KranZdid you reload the module?
22:18.02r0d3ntUH YES>
22:18.05sivanar0d3nt: the new code is on by default
22:18.10r0d3ntsivana, ok....
22:18.22KranZdid you click your heels and count to 3?
22:18.36drumkillar0d3nt: calm down, son :)
22:18.39r0d3ntKranZ, i've done all the voodoo magic i know.
22:18.43drumkillait has fixed the problems for most people
22:18.43KranZheh
22:18.44*** join/#asterisk zotz (n=zotz@24.231.36.100)
22:18.44afrosheendid you utter a short anti-american rant just to be sure?
22:18.46r0d3ntdrumkilla, .......
22:18.54sivanar0d3nt: but.. did you say abra
22:18.54r0d3ntafrosheen, absolutely not.
22:19.01drumkillaif it's that bad, maybe you need to invest in a hardware echo can board
22:19.02afrosheenwell there's your problem
22:19.49r0d3ntdrumkilla, all our tests show that it's not within asterisk and it's not within the PRI and it's not within Telepacific, but that it happens when they hand off to Sprint or MPower or anothe CLEC....
22:19.50KranZor a separate pri echo can hardware to insert in the pri
22:20.48KranZyou might need to raise the echocancel taps
22:20.50r0d3ntdrumkilla, so how would that effect the issue ?? onboard quad PRI echo canceller card would only help it if there is echo on the PRI no ???
22:21.07r0d3ntKranZ, I've done all of the above...
22:21.28KranZis it on inbound or outbound calls?
22:21.31r0d3nt32, 64, 128, 256, ON OFF  bridged, YES NO, jitterbuffer up and down...
22:21.35r0d3ntKranZ, BOTH.
22:21.45r0d3ntduplicatble with certain phone numbers 100% of the time
22:21.49r0d3ntmostly mpower customers
22:21.53FuriousGeorgesox looks like fun.  ive got 4 * servers and only one has bareable moh
22:22.06r0d3nttelepacific says " we ran tests between TPAC and mpower and we show no echo "
22:22.22FuriousGeorgethe other too are far too loud.  the only variable i can find is that the good one has an installed sound card
22:22.38KranZr0d3nt: you're not getting slips or having timing issues are you?
22:22.43r0d3ntnone.
22:22.55r0d3ntno bit errors, no timing slips, no alarms.
22:23.21*** part/#asterisk extremis (i=extremis@equinox.alluvium.com)
22:23.58KranZare you calling from voip lines?
22:24.25r0d3ntcisco 7940g's
22:24.50r0d3ntcisco <-> sip <-> asterisk <-> PRI <-> telepacific ---------------
22:24.59KranZ---- pooop
22:25.03afrosheenhaha
22:25.11r0d3ntyes it is poop.
22:25.13KranZthere's the problem, too much poop
22:25.16KranZhmm...
22:25.23r0d3ntKranZ, funny.
22:25.30r0d3nti'm glad you all can joke and laugh @ me.
22:25.32KranZheh
22:25.39KranZim honestly trying to think up solutions
22:25.46KranZbut everynow and then....
22:25.59r0d3ntwell this is my job.. and i'm about ready to quit.
22:26.09KranZyou try a different box?
22:26.13r0d3ntuh
22:26.15r0d3nti have 3.
22:26.20KranZall echo?
22:26.34afrosheenis it local or far end echo
22:26.36r0d3ntdell sc420, dell 2850, no-name with dual piii 750's Asus momboard...
22:26.47sivanar0d3nt: what are you using for interface with PRI?
22:26.49fugitivodo you have echo on all your calls?
22:26.49sivana405?
22:26.54r0d3ntafrosheen, far end echo.. only I/we can hear it.. the other person on the PSTN cannot hear it...
22:27.02afrosheenthat's near end
22:27.09r0d3ntsivana, the single span PRi card, the T101P
22:27.19fugitivor0d3nt: do you have echo on all your calls?
22:27.21sivanaso get the one with onboard echo can
22:27.23r0d3ntafrosheen, telepacific says that is far end...
22:27.29sivanawe have the same issue, software isn't going to solve it
22:27.34KranZfrom their end
22:27.46r0d3ntfugitivo, only on certain calls.. reproducable to certain phone #'s 100% of the time... usually mpower customers....
22:28.00r0d3ntsivana $2500 out of my pocket....
22:28.02afrosheenr0d3nt: well from their point of view it is I think...not sure, but generally if you can hear echo in your ear but the person you're calling can't, that's considered 'near end'
22:28.15sivanar0d3nt: $2500 no echo, $0 echo
22:28.16sivanapick
22:28.20KranZyou got a diff voip device?
22:28.28r0d3ntsivana, i didn't even get paid $2500 for this job.
22:28.29KranZto test
22:28.34afrosheenthat was my next suggestion, something besides a cisco to try
22:28.45r0d3ntKranZ, yes.. same effect...
22:28.56fugitivoKranZ: echo is not generated on his side
22:28.57r0d3ntgrandsteam, cisco, softphone.
22:29.01afrosheenso everything regardless of sip hardware is echoing
22:29.13KranZfugitivo, but couldnt it get cancelled if he's hearing it
22:29.16r0d3ntonly certain phone calls
22:29.22r0d3ntmostly to mpower customers
22:29.34r0d3ntwhich go from Telepacific, to Sprint, over the tandem to mpower
22:29.43KranZfugitivo, cancelled at the voip device...
22:29.46sivanar0d3nt: we have the same issue, software isn't going to solve it
22:30.54r0d3ntsivana, i agree.. which is why I want to kill digium
22:30.54sivanawhy
22:30.54KranZr0d3nt: have you tried messing with the tx gains?
22:30.55afrosheenwill those ciscos train any of it out on their own?
22:30.55r0d3ntsince they just told telepacific that it would.
22:30.55KranZon the voip device
22:30.55r0d3ntKranZ, YES.......
22:30.55KranZturning them all the way down
22:30.55r0d3ntKranZ, there is setting for the voip device.
22:30.55fugitivor0d3nt: did you set gains to 0?
22:30.55r0d3ntfugitivo, yes
22:30.56r0d3ntdefault is 0 / 0
22:30.56*** join/#asterisk Umaro (n=umaro@209.140.74.64)
22:30.56KranZis there a -xx?
22:30.56r0d3ntwe are currently @ 0 and -6 on tx
22:30.56KranZhm
22:30.59*** join/#asterisk TrainConductor (n=jdenton@208.255.225.67)
22:31.11r0d3ntwe tried -3 -3 per TPACs instructions.
22:31.15afrosheen-6 on tx is dangerous sometimes, we had issues with a low tx and everyone cried and cried about how they were too quiet to callers
22:31.24UmaroHey guys.. trying to get asterisk going on my new athlon 64 box, but having some troubles getting mpg123 compiled.. anyone here that has done it before on x86_64 and can give me some advice?
22:31.37fugitivoUmaro: use gentoo
22:31.38KranZemerge mpg123
22:31.38tzangerwell -6 is 1/4 normal volume
22:31.42r0d3ntafrosheen, ya.. well telepacific lines are so fucking loud.. it wasn't even noticable.
22:31.47afrosheenlol
22:31.51Umarofugitivo: ew ;)
22:32.07afrosheentzanger: in an office setting people talk at 1/4 normal volume anyway
22:32.14tzangeryup
22:32.23r0d3ntwe ran everything @ -9 and -6 to balance the audio on the meters.....
22:32.24tzangerI would really like if people spoke quieter even
22:32.25afrosheenso that's 1/2 as loud as normal overall :)
22:32.28KranZr0d3nt: what cisco's are u usin?
22:32.36fugitivoUmaro: honestly, i use madplay, not mpg123
22:32.37FuriousGeorgeUmaro: google the output of the compile where it fails
22:32.42r0d3ntKranZ, 7940g's and ATA 186's
22:32.50r0d3ntsip flash 7.5
22:32.53r0d3nton the 79xx's
22:32.54FuriousGeorgeyou gotta be more specific
22:33.03FuriousGeorgenot that i know the problem :)
22:33.23afrosheenr0d3nt: how long have you been working on this
22:33.36UmaroFuriousGeorge: yeah, I tried that.. :/
22:34.06r0d3ntafrosheen, since july.
22:34.12*** join/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net)
22:34.20r0d3ntafrosheen, i'm about to quit my job and put bullet in my head.
22:34.24afrosheenr0d3nt: 3 months for less than $2500? dude I'd leave right now if it was me
22:34.57r0d3ntya well
22:35.02r0d3nti just sold another system
22:35.05r0d3ntfor over 50 handsets
22:35.07r0d3ntand guess what
22:35.10r0d3ntthey have mpower.
22:35.25r0d3nti'm going to tell them to go buy a panasonic hybrid system and to never call me again.
22:35.26afrosheenso you gotta learn this one way or another huh
22:35.27obsidian-studiosgreetings, I got some funkyness going on. I have a Cisco 7960 with 6 sip channels configured for it, 1001-1006, Inbound works great, if I ring SIP/1001 I get the first line on the phone, 1006, the sixth. However no matter what button is pressed for dialing out it uses SIP/1006?
22:35.27fugitivoexperience has a cost
22:36.09afrosheenobsidian-studios: is * registering it as only 1006?
22:36.16*** part/#asterisk TrainConductor (n=jdenton@208.255.225.67)
22:36.31obsidian-studiosafrosheen: * sees all 6 registered as peers? I am thinking maybe it
22:37.02afrosheenso you get 1001-1006 on sip show peers?
22:37.06r0d3ntso is this card really all that ????
22:37.07*** join/#asterisk TheCops (n=dump@206.248.136.146)
22:37.11r0d3ntit'll handle echo from the entire work
22:37.13r0d3ntworld*
22:37.28obsidian-studiosI am thinking the phone might be keeping a memory of the # dialed? I know if a call comes in and you return the call, the phone will use the same line in came in on. I know once you dial a #, it will show that # if you have dialed it before, but not sure if it uses the same line as before or what
22:37.41RoyK<PROTECTED>
22:37.41KranZr0d3nt: is this an in office box, so no latency from the handsets?
22:37.54r0d3nt2-3ms latency to the handsets from the server.
22:37.59obsidian-studiosafrosheen: yes, sip show peers shows all 1001-1006, all six. Also inbound works fine, if a call comes in, I can map it to any line, and it will ring that line
22:38.05fugitivothe problem is not on his side
22:38.16afrosheenhe's saying it's mainly mpower customers so there's a trunking issue there, sounds like an echocan card would fix it but it's anybody's guess where the problem lies
22:38.54afrosheenthe problem is probably somewhere between telepacific and mpower like he said earlier
22:39.56afrosheenobsidian-studios: so * sees it as 6 devices..but does it see itself as 6? there's probably some config voodoo for outbound you may have missed
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22:41.04afrosheenI know the polycom multiline phones have 2 sections for each line, inbound and outbound registry
22:42.51obsidian-studiosafrosheen: I can call on various lines that are directly mapped to various sip channels, and make 1001-1006 ring on the phone by dialing SIP/1001-1006
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22:43.43afrosheenright, but your problem is with outbound right, you say it only dials out via 1006
22:43.43obsidian-studiosafrosheen: however from the phone, no matter what button is pressed for the corresponding line, it all uses SIP/1006 acting like a 1 line phone, it's not allowing outbound stuff to use the other lines
22:43.52obsidian-studiosafrosheen: yes
22:43.53afrosheengotcha
22:44.33obsidian-studiosafrosheen: I know if a call comes in on line 1004, and you miss it, or receive it, navigate the phones menus for missed or received calls, and place the call again, it goes out the line it came in
22:45.33obsidian-studiosI also know that if I dial 555-1212 once, as I start to dial 555, the phone will show the rest so I do not have to dial a # in memory. I am wondering that if I dial a # out of 1001, and then pick up 1006, if it uses 1001 again, since the previous call was placed out that line?
22:46.30conveyCan someone tell me how to create an extension that forwards to a sip address for example exten => s,3,Dial(sip/happy@10.0.0.1)?
22:47.00r0d3ntwe've decided to try and talk with digium some more, being that it was their recommendation for the cvs head of zaptel, perhaps they have more ideas, before we go and buy a card. But I told everyone that, that is our last viable option is to use the quad echo cancelling card....
22:47.02obsidian-studiosanother issue, in the SIP*.cnf file sent to the phone, I reversed the order or line1 and line2, to where line1=1002, and line2=1001, and they still showed up in order on the phone?
22:47.09afrosheenobsidian-studios: you're going over my head now, I'm not sure how the feature set works on that model
22:47.26obsidian-studiosafrosheen: I am wondering if it's firmware version or something
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22:47.43obsidian-studiosafrosheen: pretty sure it's he newest release, and it's very weird
22:47.45r0d3ntalso, in talking with my other technician helping on the new pbx sale, it was decided a few days ago to only sell the quad echo cancel cards and to inform the client of the additional cost before they sign a contract and give deposit.
22:48.30afrosheenobsidian-studios: I'd consider rolling it back a version
22:48.32r0d3ntmy execution has been delayed once again.
22:48.48obsidian-studiosafrosheen: might have to, getting the firmware was a pain, got to get a smartnet contract so I can get them directly
22:49.05wunderkinr0d3nt: what card are you  using now?
22:49.12afrosheenobsidian-studios: you'll run into fellow cisco users here that may email you images, I've seen cd's for sale on ebay also
22:49.14obsidian-studiosafrosheen: pretty sure it's someone to do more so with the phone, than config, config is fairly straight forward
22:49.17r0d3ntsingle span T/E/1/PRI card
22:49.30r0d3ntx401p ?
22:49.51wunderkinr0d3nt: oh ok, i was going to say.. if you  had the 405 or 410 you could just add the  module.. oh well
22:50.01r0d3ntdamn
22:50.02r0d3ntnope
22:50.10r0d3ntand i sold all my quad pri cards i had...
22:50.15r0d3ntof which i've purchased 3.
22:50.24wunderkinok
22:50.38obsidian-studiosafrosheen: it's also weird that the phone displays a different order than I tell it to in the cnf file, I am going to play with that a bit and see as well
22:50.43r0d3nti didn't know it was a module like that.. it looked like a daughter board of sorts tho'
22:50.54KranZwunderkin: add the module?
22:51.05obsidian-studiosafrosheen: should show 1002, 1001, 1003-6 on the phone, but they are in order?
22:51.19afrosheenobsidian-studios: if what you're mapping and what it's displaying are different, that's definitely a problem
22:51.35websaeany AstBILL users here at all?
22:51.40r0d3nti would imagine that it would require firmware updates....
22:51.45afrosheenthen again maybe cisco thinks they're smarter than you and 'adjust' that automagically
22:51.45wunderkinKranZ: yes, you can just add the echo can module.. its just an addon card
22:51.56KranZlike a daughterboard?
22:52.00wunderkinyes
22:52.03KranZlink?
22:52.52obsidian-studiosafrosheen: check this out http://pastebin.ca/24149
22:54.20obsidian-studiosafrosheen: makes no sense there to me, I assume the phone must want to put them in order, but at the same time it's mapping all outbound to 1006? Maybe a bug in the firmware?
22:54.49*** join/#asterisk buddah (n=djbrianc@67.110.253.129)
22:55.06wunderkinKranZ: not sure, it doesnt really say on the website
22:55.07buddahanyone know if there is a way to use call forwarding with polycom ip 500/501s if i'm using g711?
22:55.22buddahi get an error message spamming in * and it screeches on the line of the caller when it does
22:55.43wunderkinKranZ: you would need to ask digium sales
22:56.13KranZhmm
22:56.31KranZanyone already know the price on the echo can module??
22:57.06drumkillait is an addon card, but it is not a user servicable part
22:57.22drumkillait requires a change of firmware.  however, i believe it can be added to an existing card if you ship it back
22:57.42KranZic
22:58.35KranZthey'd shit if a tool and firmware were leaked, potential lost revenue
22:58.45*** part/#asterisk shido6 (n=shido@d221-68-210.commercial.cgocable.net)
22:58.45websaeanyone here used ASTBILL?
22:58.47KranZguess i wait till i get a spare card
22:59.13KranZbut its nice to know its not obsolete quite yet
22:59.16ronaldl79Which TOS setting do you all prefer for SIP?
22:59.22*** join/#asterisk hound (n=MrHound@tor/session/x-430f00bf6e96a805)
22:59.23jayk-how do i check out -head using cvs?
22:59.30ronaldl79Currently, I am using 'lowdelay' -- which is best?
22:59.37houndSo is the backdoor in asterisk underway for US law enforcement ?
22:59.41wunderkinonce your card is upgraded to 2nd gen, you can upgrade the firmware yourself.. although ive never seen anywhere to download it.. or how to do that
22:59.47ronaldl79Speaking of CVS, is 'CVS HEAD' a daily build of Asterisk? I never bothered to find out.
22:59.56houndor is it already there with monitor! good planning
23:00.00jayk-ronaldl79: i thought there were nightly builds somewhere, but they wern't being updated
23:02.26ronaldl79BTW, guys, I demo'd an * box today for a non-profit -- The director was THRILLED! We covered voicemail, transfer and parking, calling, and IVR.
23:02.41afrosheenbuddah: on our system with the polycoms we just do a *71 and enter a number, boom, call forwarding is done
23:03.08AgiNamuhound, or ChanSpy?
23:03.41*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
23:03.45websaeanyone here use MySQL and ASTERISK together...the CDRs into a MySQL database...?
23:04.16afrosheenbuddah: when the call hits the phone the phone tells asterisk to call the forwarded number I believe
23:04.31glm2kwebsae: i do.
23:04.39generalhanwhy doesnt Chan_Spy work with 1.0.9 ? this sux
23:05.20*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
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23:05.31KranZ6pm
23:05.34file[laptop]generalhan: backport it if you want it in stable
23:05.59generalhanfile: what do you mean ?>
23:06.01*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
23:06.23file[laptop]well chanspy uses stuff that's in the core of asterisk... that has been put there for chanspy to use
23:06.27*** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net)
23:06.37file[laptop]you'd have to backport that stuff, plus the application...
23:07.38generalhanawww, i wouldnt know how to do that. My boss just came to me today and said that we wanted to be able to listen in to our new guys for training purposes and i told him that i couldnt do it. he seemed to just accept it !
23:07.53AgiNamuhmm, if I have 2 calls in a Polycom, do I press "Join" or "Conference" to connect em both in a 3way?
23:07.58*** join/#asterisk Rez (i=lorez@freenode/staff/lorez)
23:09.23JerJertry both
23:09.33JerJersee what works better for you
23:09.36AgiNamujoin does it whew
23:09.43AgiNamuyea, I have 2 live calls :)
23:09.57AgiNamuwhere you at VON?
23:10.05AgiNamus/where/were
23:10.08JerJerno - too damned expensive
23:10.40AgiNamuget digium to sponsor you? ;)
23:11.11darkskiezgeneralhan: if u are using zap, try zapscan
23:11.36generalhanhmmm
23:15.26[hC]Anyone have an idea why IAX indication requests and the indications themselves appear to be happening about 10 seconds apart, according to my logs?
23:15.28[hC]http://pastebin.ca/24147
23:15.33[hC]<PROTECTED>
23:15.51bzbwanyone tried to set up * with line sharing?  It is based on draft-anil-sipping-bla-02.txt for SIP client.
23:16.21file[laptop][hC]: is there a problem?
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23:17.40bzbwI talked to Grandstream, they said their next release will support line sharing.
23:18.23afrosheenthe latest release of AMP has some cool stuff in it like separating users from devices
23:18.52afrosheenyou sit down at any phone, dial *11 and your extension and you can take/make calls from your normal ext number
23:19.34bzbwwhat is AMP?
23:19.49websaeanyone here use asterisk and MySQL?? for like CDRs, etc...??? looking for some help getting it configured right
23:20.30*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
23:20.56JohnsieAMP = Asterisk Management Portal
23:21.11redder86What's the difference between "congestion" and "busy"?
23:21.11JohnsieBasically a web suite to control Asterisk, automate configurations to an extent, etc.
23:21.23JohnsieCongestion = fast busy/reorder tone
23:21.27[hC]file[laptop]: well yeah but im trying to figure out what could be causing it. kind of a wild goose chase at this point. im getting one way audio drop outs on some calls. cant figure out the cause yet, so im trying to eliminate everything that shows up in my logs just incase.
23:21.31JohnsieBusy = slow busy/standard busy
23:21.33fugitivowebsae: mysql is evil
23:21.36[hC]file[laptop]: usually lasts 5-10 seconds, one way
23:22.51redder86Johnsie: so congestion will cause the dialplan to increase by 101 steps, right?  And not busy?
23:22.59bzbwJohnsie: so you can configure extensions and dial plan without using editor to edit either "extensions.conf" and other files?
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23:26.30ronaldl79What are the recommended TOS/QOS bit settings for *? I'm trying 0x18 now (low delay and throughput)
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23:28.06Kattymew.
23:28.36afrosheenbzbw: basically, google amp asterisk and you'll see what it can do
23:29.40mrfrenzyare there any debian packages for amp?
23:29.49ariel_mrfrenzy, you can ask on #amportal
23:29.56ariel_I think they were working on one
23:30.09Kattyariel_: :>
23:30.09afrosheenlol yeah right, that's a KOA kampground in there
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23:30.31ariel_Katty, hello
23:31.26ariel_hope your day is going well.
23:31.26mrfrenzythx ariel_ will do
23:31.27Kattyariel_: mostly (=
23:31.27ariel_mine has been pretty shitty
23:31.27Katty:<
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23:31.27ariel_too much to do too little time to do it in.
23:31.48Dr_Raycan, ick, have you tried the pouches?
23:31.53Kattyariel_: yeh, i know that feeling :<
23:32.02afrosheenewww, I don't eat ANYTHING from pouches
23:32.18Dr_Raythe tuna in the pouch is nice
23:32.33Kattyariel_: :<<<<<
23:32.37afrosheenI was just joking..the only tuna I'll eat is yellow fin on sushi
23:32.38Johnsieredder86: I am not sure if congestion will do that or not, I don't believe so...it's just a means of error indication.
23:33.01AgiNamuif a T1 is delivered to a smartjack in the wall , do I use a T1 crossover or a straighthru rj45
23:33.06Johnsiebzbw: I believe so, yes... if you do a Google search, you can see AMP in action on their SourceForge site.
23:33.10ariel_Katty, family is not here. But thanks. (There at my Mom's)  I am stuck working.
23:33.16JohnsieSorry for the delay, I have about 20 windows going on here.
23:35.18ariel_argh do people really belive it when they get an email saying.  Be a travel agent in 1 week make lots of money just send us 1495 for cource.
23:35.36redder86Johnsie: turns out that with busy the call goes to the 101th step after the current one.  With congestion it will go to the next step in the dialplan.
23:36.08Johnsieredder86: Oh okay...my bad. Thank you for correcting me, though.
23:36.39[hC]wow, cool netsplit.
23:36.41azzieAgiNamu, whatever brings the T1 up
23:36.47[hC]file[laptop]: does it make sense that the timestamps would be skewed like that?
23:38.24*** join/#asterisk Tiveron (n=someone@66.146.140.5)
23:38.52Tiveronhas anybody got a couple minutes to answer a couple questions?
23:39.12AgiNamuazzie, ok :)....
23:39.17file[laptop]I don't see where you say the timestamps are being skewed...
23:39.24ariel_Tiveron, hello and maybe
23:39.44azzieAgiNamu, the other cable would keep T1 in "loss of signal" state...
23:39.51Tiveronariel: that's nice and definite :P
23:39.53file[laptop][hC]: iax2 debug is a good thing
23:40.54AgiNamuazzie, yea, that's what I'm seeing with a straighthru. T1 crossover brings layer1 up
23:40.54*** join/#asterisk roulduke (i=j5zw1bxx@p508D1BF1.dip0.t-ipconnect.de)
23:40.57[hC]oh boy, i cant iax debug for a particular peer!
23:40.57[hC]:)
23:41.13afrosheensure you can, that's what grep is for
23:41.15afrosheen;)
23:41.23[hC]haha :)
23:41.35file[laptop]I have an evil patch that may not apply that does iax2 debug peer and iax2 debug ip...
23:41.47[hC]im running cvs head last night
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23:42.56[hC]...from. last night.
23:42.56[hC]gotta love freenode.
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23:43.01Tiveroncan anyone tell me if there's any way to propegate a Message Waiting Indicator originating from a Zap FXO channel to the rest of my FXS/SIP clients?
23:43.19[hC]file: I could try incorporating the patch, if you wanna send it?
23:43.54file[laptop]gotta find it
23:44.18file[laptop]http://neutrino.file-radio.com/asterisk/iax2_specific_debug.diff
23:44.36[hC]heh. damned sure that lilo has nothing better to do than to sit at irc waiting for the next reason to wall.
23:44.46[hC]file[laptop]: thanks. i'll give it a go
23:46.07*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
23:46.07*** join/#asterisk arguile (i=user224@66.38.201.234) [NETSPLIT VICTIM]
23:46.07*** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com) [NETSPLIT VICTIM]
23:46.08*** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) [NETSPLIT VICTIM]
23:48.57Tiveronhow do I enable call waiting on asterisk?
23:50.15file[laptop]on a zaptel channel?
23:50.40file[laptop]what technology/protocol... you have to be specific
23:51.21Tiveronon any channel. I've got a Zap FXO channel as my trunk, and a number of SIP and Zap FXS extensions
23:51.42file[laptop]it's not up to Asterisk to do call waiting on stuff like SIP
23:51.48file[laptop]it's up to the ATA or Phone
23:52.29file[laptop]on zaptel stuff it's configured in zapata.conf
23:52.40*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
23:53.39Tiveronok... I'll look into that... what about if I get a call waiting signal through my PSTN trunk? I can hear the tone on my SIP hardphone (GXP-2000), but can't flash to it.
23:53.58hardwiremy mwi is blinking
23:54.03hardwireit doesn't need to be
23:54.06hardwiredamnit
23:54.06file[laptop]you have to setup like an extension that flashes it
23:54.25TiveronI don't follow...
23:54.41file[laptop]there's an application, Flash, that flashes zaptel channels
23:54.46file[laptop]that's how you have to do it
23:55.01Tiveronoh. an asterisk addon you mean?
23:55.09file[laptop]no... it's an included application...
23:55.21file[laptop]you can't flash from your phone, because you're not actually flashing the line
23:55.45file[laptop]you have to setup another extension that goes to the flash extension that will flash the zaptel line, then go back to your original call to the zaptel line
23:56.13file[laptop]and this is why analog through Asterisk to a SIP phone is bad, mmmk?
23:56.25[hC]file[laptop]: hate to sound ignorant here, but do you have any suggestions on what to look for in an iax2 debug, to notice errors?
23:56.37Tiveronheh. understood.
23:56.42file[laptop][hC]: not really, I haven't done an iax2 debug in a very very long time
23:56.46Tiveronone more question...
23:57.05[hC]file[laptop]: heh.. yeah.. and because there's so much of it, going thru every line is a pita. :)
23:57.17file[laptop]I'm much more at home with SIP
23:57.28[hC]me too.
23:57.35Tiveronmy PSTN trunk comes from my ILEC, and can generate a MWI signal. Is there anyway to recognize that with asterisk and generate an internal MWI signal?
23:57.44[hC]might just try switching from iax2 to sip, and see if the problem goes away first.
23:58.03hardwireheh
23:58.08[hC]ive spent almost 2 weeks on this now, and still havent nailed down the cause.
23:58.10[hC]hardwire: shut. up.
23:58.13hardwireall these subscriptions for SIP to my phone make my phone wiggle
23:58.13[hC]:)
23:58.22file[laptop]Tiveron: not really, no...
23:58.26[hC]wiggle??
23:58.30hardwirejitter
23:58.31hardwirepoop
23:58.37[hC]oh
23:58.57[hC]i envisioned a phone wiggling around your desk.. remided me of pager races, back in the day,..
23:58.59azzieTiveron, give up your fxo line and port it to some SIP provider...
23:59.17TiveronReccomendations for NA proviers?

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