irclog2html for #asterisk on 20060627

00:01.05drraythere are lots of reason to curse in here
00:01.14ManxPowerhttp://pastebin.ca/72584
00:02.22smackusok, on my old asterisk system, I successfully had cdr logging to my mysql database. On my new system, I thought I had everything set up correctly, but it is still going to the csv. I have may database up, my cdr_mysql pointed at it. what am i overlooking. I am trying to find in the wiki if there is a file that has to tell it to use the database... I am lost. I must have done it by accident last time. Where am I missing my setting.
00:02.28smackuswow... sorry.
00:02.31smackusi type too much
00:02.52*** join/#asterisk Qb3rt (i=jhgjkgui@216.252.87.8)
00:02.57*** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk)
00:03.37*** join/#asterisk albertito (n=net@host10.201-253-236.telecom.net.ar)
00:03.46*** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)
00:03.53terrapendoes anybody know of a voice other than The Voice?
00:03.59terrapenI'm not too keen on her voice
00:04.08rob0There are voices in my head ...
00:04.18terrapenI need a voice that's a little more relaxed and less uptight
00:04.20terrapenheh rob
00:04.36smackusyou dont like alison? she makes me all tingly
00:04.42drrayno kidding
00:04.54smackuslike the rope in gym class :-D
00:04.58ManxPowerMust.  resist.  fighting.  Asterisk.
00:05.00terrapenshe's a hottie no doubt
00:05.02albertitoHi! I've just connected two asterisk boxes with an E1, and both alarms are cleared, but when I start asterisk I see (on only one of them) lots of "PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1". Channels seem to get up tho, but I haven't tested they work yet. Is this expected? What should I check?
00:05.07terrapen(at least the head shot)
00:05.11rob0Hmmm, in fact I could swear that she was one of the voices!
00:05.30terrapenbut I need somebody more suitable for this bicycle stores
00:05.33terrapenerr these
00:05.34smackusis there a way to verify if asterisk-addons is installed?
00:05.43ManxPoweralbertito, either there are errors on the E1 or you have a device on the system that is locking interrupts and causing dropped data
00:06.28albertitoManxPower: so it could be a faulty cable that causes the HDLC errors?
00:06.30rob0You're talking about hiring someone? Just find an actor/actress/broadcaster. An ad agency might help.
00:06.38ManxPoweralbertito, correct.
00:06.42drrayand you'll need studio time
00:06.53ManxPowerThat means "corrupted or lost data"  It does not tell you where the problem is happening
00:06.56drraywhich depending on where you live could be cheap
00:07.00smackusi think he is looking for something already completed
00:07.08smackusrecordings already done
00:07.40albertitoManxPower: given that the box with errors is a decent Pentium D, without significant load, and that the card doesn't share the interrupt, I guess I'll blame on the cable. Thanks a lot! I'll swap the ends just to confirm (they should appear on the other box)
00:08.00terrapennope
00:08.01Qb3rti am having a very nice problem!!! i have 3 polycom phones behind a firewall and when i try to dial it says url call disabled... can someone tell me what is the problem??
00:08.13terrapeni want to e-mail some prompts to someone with a good voice and get some WAVs back
00:08.22terrapenand i'm looking for pricing similar to The Voice
00:08.52ManxPoweralbertito, The problem can also be caused by GigEthernet, RAID, SATA, etc
00:08.57terrapenhttp://www.intervoice24.com/
00:08.58terrapenniiiice
00:09.11terrapenthank you google adwords
00:09.43*** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6)
00:09.57rob0I googled "professional voice recordings", saw a bunch
00:09.58terrapentheir samples suck tho :)
00:10.03albertitoManxPower: I've switched the ends and the error remains on the pentium D (from now on, "big box", because the other one is a Pentium II, "small box")
00:10.35albertitoManxPower: how can SATA (I don't have the other ones) interfer with the digium card, since there's no disk activity going on?
00:10.52ManxPoweralbertito, there is logging going on isn't there
00:11.10albertitoManxPower: it's logging the errors coming out of asterisk, yes
00:11.33ManxPoweralbertito, that's all it took on one system I installed.  Also voicemail caused disk writes.
00:12.01albertitoManxPower: but vmstat shows only activity once every five seconds, and about 80k each write... that just can't be it
00:13.16albertitoManxPower: I have around 1300 interrupts per second, and given that HZ=1000 here, that's about 300 interrupts per second over the base
00:13.41ManxPoweralbertito, I have given you my diagnoses.
00:13.42albertitoManxPower: OTOH the small box is running at 2000 interrupts per second
00:14.00ManxPoweralbertito, what is the interrupt latency and interrupt jitter?
00:14.11albertitoManxPower: Thanks a lot =) I'm just confused because I can't see it fit, but I'll dig into it
00:14.22albertitoManxPower: I don't know how I find out about those
00:14.35ManxPoweralbertito, Neither do I.
00:14.38terrapenso, what makes for the best on-hold music?
00:14.39*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
00:14.49terrapeni like Americana myself
00:15.28albertitoManxPower: I understand your diagnosis about the irqs, but they're just two unloaded machines, with no calls going on, which is why I find it strange
00:16.46ManxPoweralbertito, Perhaps you do not understand that regardless of how many interrupts are happening per second, if one device or driver, such as a SATA controller locks interrupts for a very long time, the digium card will fail to get data fast enough, since Digium cards do not have onboard buffers.
00:17.43ManxPowerI had a problem with a Dual Xeon motherboard that had the same errors any time there was ANY disk activity.  Replaced the motherboard with a different make/model and everything worked.
00:17.59ManxPowerMANY people have reported similar experiences with PCI RAID, SATA, and GigEthernet
00:19.05albertitoManxPower: mmmmm I don't have RAID or GigEthernet, and SATA is inactive most of the time... but I have one USB device that might be the one, now that I think of it...
00:19.42ManxPoweralbertito, All it takes is ONE disk read or write
00:20.40albertitoManxPower: if there is at least one disk read or write, vmstat "bo" or "bi" column should notice it, and it doesn't (except once every 5 seconds, but these errors appear continuously)
00:20.53albertitoManxPower: and the usb device wasn't either
00:21.02Qb3rtI have 3 polycom phones behind a firewall and when i try to dial it says url call disabled... Is it because of th firewall??? how can i fix that?
00:21.21ManxPowerQb3rt, It is NOT because of the firewall.
00:21.33ManxPowerThe phone is not registered to a server and therefore cannot make calls
00:23.15albertitoManxPower: anyway, thanks a lot for the diagnosis and support, I'll look deeper into it. I'll see if I can get irq latency measures using some kernel instrumentation
00:23.16*** join/#asterisk Samoied (n=Samoied@201.22.215.135.adsl.gvt.net.br)
00:23.49Qb3rtManxPower: ok thanks! and when i will call one of these 3 phones how the router will know to wich one send the call?
00:23.56*** join/#asterisk brc__ (n=brc_@pdpc/supporter/basic/brc)
00:24.19ManxPowerQb3rt, The router modifies the source port of the registration
00:24.38ManxPowerThat is what NAT does.
00:25.09Qb3rthehe yeah!!! that was a pretty stupid question if i think about it!!!
00:27.04Qb3rtManxPower: so basically the polycom phone will register through any kind of firewalling?
00:27.31Qb3rti mean on d-link and linksys
00:27.39ManxPowerQb3rt, Do you understand the difference between NAT and Firewall?
00:27.47Qb3rtyeah for sure
00:28.07Qb3rtbut on a router you can enable a basic firewall system
00:29.00ManxPowersomegeek, no if the firewall is blocking the packets then obviously the phones won't work.  However if the router is just doing NAT everything should work if you have nat=yes and qualify=10000 in sip.conf for each registration
00:33.38mindwarpHi guys, I have a question that I have done my best to detail in this forum post: http://forums.digium.com/viewtopic.php?t=7610 ... Not getting very far so far in the forums, so please take a look if you feel so inclined. Thanks a lot!
00:35.06*** join/#asterisk darkgamer20 (n=chatzill@adsl-71-146-156-227.dsl.pltn13.sbcglobal.net)
00:35.58ManxPowerhttp://pastebin.ca/72615
00:38.49darkgamer20an the TDM400P functions like a analog telephone adapter right?
00:41.50ManxPowerdarkgamer20, In a way.
00:42.39darkgamer20ManxPower: yea so I can connect my phone to the FXS and the phone line to the FXO port and have asterisk handle my calls?
00:42.52ManxPowerdarkgamer20, correct
00:43.04darkgamer20ManxPower: perfect
00:43.48darkgamer20ManxPower: how much dose a TDM400P card cost? also you know if theres any alternatives i can turn to if its too expensive?
00:44.13ManxPowerdarkgamer20, did you go to the Digium web site?  All similar cards cost about the same.
00:44.19darkgamer20oh
00:45.39darkgamer20wooooooo the one i need (and the cheapest one) is 241$$
00:45.52darkgamer20isnt there anything under 100?
00:47.55ManxPowerdarkgamer20, Expect to pay $80 - $120 per port, depending on the number of ports you need.  The fewer the number of ports the more expensive each one is.
00:48.24ManxPowerdarkgamer20, Traditional telcom equipment is closer to $500 per port
00:48.31ManxPowermany times much higher.
00:48.35darkgamer20man thats expensive ManxPower
00:48.36darkgamer20wow
00:49.44darkgamer20ManxPower: can you suggest anything cheaper for a hobby project, cause I am not trying this for an office enivironment
00:50.18ManxPowerdarkgamer20, Well if you want to spend 10x the amount of time working on it I guess you could get a SIPura
00:50.43drrayI would get a govarion 4 port tor2 card, and a $100 zhone channel bank
00:51.18darkgamer20drray: can you tell me the exact name of the product your suggesting?
00:52.42drrayhttp://govarion.com/home.php and zhone channel bank
00:54.08drraythe idea is to get things that you can use later
00:54.09drrayyou'll throw the zhone in the garbage 9 months after you get it
00:54.48darkgamer20drray: sorry but where is the zhone again? i cant find in the page you entered
00:55.29darkgamer20ohhh
00:55.52darkgamer20nevermind they are two different things, i kept thinking that zhone is on govarion
00:55.57drrayhttp://cgi.ebay.com/Channel-Bank-24FXS-Ports_W0QQitemZ9744803968QQihZ008QQcategoryZ51271QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
00:56.09*** join/#asterisk test34 (n=test34@unaffiliated/test34)
00:56.20drrayyou can get 24 fxs ports for $150 or so
00:56.27ManxPoweryou'll have trouble finding FXO ports for channel banks
00:56.29drrayor 8 fxo 16 fxs ports
00:56.38drrayebay
00:56.50ManxPowerDude!  The 4-port card is $700
00:57.04drrayand that corpsys site sells them on the side
00:57.26*** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com)
00:57.34drray4 ports of t1 is a bargain
00:57.54ManxPowerYes, but if he can't afford $241, he can't affford $70
00:57.56ManxPower..er.. $700
00:58.07darkgamer20lol
00:58.16darkgamer20yea ManxPower is right
00:58.34ManxPowerand zhone is crap
00:58.41drrayso is a MTA
00:58.51darkgamer20a Mail Transger Agent?
00:59.03drraythere are only two things the zhone can't do
00:59.07drraypass caller ID
00:59.16drrayand detect remote disconnects
00:59.29darkgamer20well i need caller ID
00:59.31drrayfor $150 they are great
00:59.35ManxPowerUm, so that would be called "crap"
01:00.27darkgamer20ok I dont i have the need for a channel bank, do you guys know if I can get a TDM400P on ebay for less?
01:01.28drrayseeing as you cant/shouldnt run 2 tdm400p in one box
01:01.41Un1xdarkgamer20: yea you could but who knows, if it's going to work or , if it does work for how long will it work, or what will happen with it..
01:02.07drraywith the t1 card, you can upgrade to a PRI as you expand
01:02.27Un1xthey have some but most cards are the same price as digium anyway
01:02.30darkgamer20Unlx: you got a point there, but I what can I do...?
01:02.30Un1xhardly any difference
01:02.34Un1xmaybe 2-5 dollars
01:02.44Un1xdarkgamer20: what do you need it for
01:02.47darkgamer20drray: I am not looking to expand this is a hobby project
01:02.49Un1xhow many phones how many pstn lines
01:03.00RoyKka-ding
01:03.08*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
01:03.18RoyKka-ching! i got my tax refund :D
01:03.23Un1xlmao
01:03.38darkgamer20Unlx: 1 PSTN line and 2 phones (1 phone and a fax machine)
01:03.49RoyK~nickometer Un1x
01:03.54*** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
01:03.58Un1xso youd need, 2FXS and 1 FXO
01:04.11Un1x~fxofxs
01:04.13jbotwell, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this.  An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this.
01:04.14RoyKwebsae: it's wasabi
01:04.25Un1xyea u need 2fxs and 1fxo
01:04.35websaehaha
01:04.42websaewhat's going on?
01:04.52Un1xhttp://cgi.ebay.com/Digium-Wildcard-TDM12B-TDM400P-Asterisk-1FXS-2FXO_W0QQitemZ9744374347QQcategoryZ61839QQssPageNameZWD1VQQrdZ1QQcmdZViewItem
01:05.03Un1xheres one darkgamer20 but no gauruntees on it
01:05.08darkgamer20Unlx: before i buy anything, i can have asterisk have the calls go to the right places right? like have a fax go to the fax machine and a phone call to the phone?
01:05.20Un1xlets check digiums price
01:05.22Un1xi beleive so
01:05.25Un1xask dlynes_home
01:05.29RoyKlots of voip nerds and one pretty drunk norwegian engineer - that's going on
01:06.19Un1xhey
01:06.38Un1xDarkgamer20: this is pretty much what youre looking for http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?category_id=17&product_code=RTDM11B
01:06.44*** join/#asterisk Qwellj2me (n=Qwell@unaffiliated/qwell)
01:06.46Un1xyou could get it for the same price almost from digium
01:06.53Un1xcomes with 5 yr warranty also :)
01:07.02Un1xand it's same price as on ebay :p
01:07.16darkgamer20lol
01:07.32Un1xjust email them and tell them you want one extra FXS port, and theyd do that and change the price to what need be
01:07.36Un1xit should be under 300
01:07.59darkgamer20it will probably be as much as the ebay one huh?
01:08.02Un1xyes
01:08.10Un1xpretty much with gauruntee it wil work
01:08.13Un1xand with warranty
01:08.17Un1xand the cables etc
01:08.26Un1xand the screwdriver and mousepad lmao
01:08.51darkgamer20lol
01:08.57Un1xthats just a bonus
01:09.04darkgamer20might as well get it from digium then
01:09.06Un1xheh at least u get somethings added for same price
01:09.10Un1xi personaly think it's worth it
01:09.30darkgamer20alright whatever i can spend more than a 100$ if theres a benifit
01:09.50*** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net)
01:09.59Un1xnah
01:10.01Un1xnot even 100$
01:10.07Un1xfrom wjat i can see on ebay
01:10.09Un1xfor the same config
01:10.17Un1xonly around 30$ extra max
01:10.31Un1xUS $293.40
01:10.35Un1xis the one on ebay for same config
01:10.41Un1xit should be around the same price from digium
01:11.04darkgamer20oh ok
01:11.15Un1xanyway i gotta go mann ask the other people in here, gnite maybe' i'll come around later.,, see ya all
01:11.16Un1x:)
01:11.17darkgamer20Un1x Thanks soooo much for your help
01:11.21Un1xno problem
01:11.24darkgamer20see ya
01:19.53*** join/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk)
01:20.03sorush20hi guys
01:20.08sorush20vonage anyone?
01:20.40rob0TDM S110M FXS Module US$67.5 ... know of any better deals? Preferred vendors?
01:21.00Strom_Cthat's a pretty damned good deal right ther
01:21.09websaelol
01:21.10Strom_Cassuming, of course, that it works ;)
01:21.11websaevonage
01:21.12websaeha
01:21.18Strom_Calso, lol vonage
01:21.23sorush20!voange
01:21.47websaewhat's the word that vonage always makes me think of....
01:22.02websaeoh yeah it's a phrase
01:22.04mindwarpownage?
01:22.06rob0I found 2 vendors with the same price: voipstore.atacomm.com, www.voiplink.com
01:22.10websaesoon to file for bankruptcy
01:22.20websaewhat are you looking for rob0?
01:22.25rob0Voyage :)
01:22.34drrayvonage to the bottom of the sea
01:22.35rob0I need one FXS and one FXO.
01:23.00rob0in that order ... I have an x101p for FXO.
01:23.30websaehttp://www.pbxeq.com/
01:23.42websaethey are rock solid hardware supplier
01:24.22*** part/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk)
01:24.48websaeoh so they helped me out a lot
01:27.10rob0thanks
01:29.37websaeyeah
01:30.15*** join/#asterisk [TK]D-Fender (n=joe@CPE000d3a2c3061-CM00080d8dba84.cpe.net.cable.rogers.com)
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01:42.03skraelings001Hi
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01:44.19FuriousGeorgewould everyone agree that bristuff is probably safer than trunk for getting device states with parking working in a production enviornment?
01:44.28BrijnGood evening all
01:44.49skraelings001Good eve.
01:44.58w0rmzw3rthIs it possible if I buy 5 voip lines from voicepulse.com that I can add those lines to my asterisk box so that 5 users can use it at the same time and place calls from the box
01:45.00[TK]D-FenderNot if you need PRI which it conflicts with as I'm told
01:45.43file[TK]D-Fender: My my look who it is!
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01:49.21[TK]D-FenderOMGZ!
01:49.32[TK]D-Fenderfile : In Mississauga now :)  Leeching Wi-Fi from... somewhere ;)
01:49.40file[TK]D-Fender: ooh, why are you there?
01:49.48[TK]D-Fenderw0rmzw3rth : Sure
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01:51.22*** part/#asterisk userdefined (n=jross@cpe-24-169-142-23.rochester.res.rr.com)
01:51.52skraelings001does anyone experienced strange behaviour when trying to call to phone numbers that doesn't exit? , i should get a msg from telco but i don't
01:52.13[TK]D-FenderVacation.. you knew :)  get with the program.
01:52.24fileI know lots of things, doesn't mean I remember them
01:52.48TholiaAre most of the people in here telecom people by trade?  I'm kinda a half-assed telecom person, thrown to the wolves on a couple Rolm systems and learning everything from scratch
01:52.58anonymouz666file haha good one
01:53.11fileTholia: I was a high school student when I got involved in VoIP, been with it since!
01:53.41Tholiamy company is in the middle of a VoIP conversion from the Rolm and Aspect switches we have to an Avaya VoIP solution
01:53.43skraelings001i recently got involve with asterisk cause of work
01:54.11anonymouz666file? pthread is hard to learn?
01:54.22fileanonymouz666: it's not too bad to learn
01:54.58TholiaTrying to get into asterisk is like drinking water from a fire hose.  I have a spare box, I can afford a simple FXO card, after that trying to find a VoIP provider seems like the dicey part
01:56.26[TK]D-FenderTholia : IP office?
01:56.28fileTholia: cheat, turn the water level down!
01:57.21[TK]D-FenderTholia : I almost got forced down that path.... So glad I'm not stuck with their toaster :)
01:57.24Tholia:)  I'm up for any recommendations y'all have on hardware or software for the home tinkerer (that has an Windows Active Directory domain infrastructure) :)
01:57.57[TK]D-FenderTholia : Windows is irrelevent as * is basically a *nix platofrm.  For hardware it al depends on what you want.
01:58.20[TK]D-FenderTholia : What kind/number of lines & phones would you like to use?
01:58.39PakiPenguincan someone do some testing with me?
01:58.43PakiPenguincall this number ? 708-547-8653 ?
01:59.12Tholiaright just saying that's my level of home tinkering.  initially I'm OK with using a softphone to call wiht (to test), I'll have a dedicated PC.  Eventually I'd like to have a POTS line (or VoIP line) coming into the PBX, with autoattendant and conferencing and whatnot
01:59.29*** join/#asterisk userdefined (n=jross@cpe-24-169-142-23.rochester.res.rr.com)
01:59.39Tholiafor a few softphone extensions initially, then maybe even an ATA or two or some IP phone hardware
02:00.00PakiPenguinplease
02:00.09websaeTholia: try www.pbxeq.com --- got some nice deals
02:00.38*** join/#asterisk b00gz1 (n=b00gz@d233-124-245.col.wideopenwest.com)
02:00.43Tholialooking at their site now.  I don't really think I need a FXS port quite yet
02:00.50Tholiadon't even think I need an FXO port yet
02:01.01websaeget a VoIP provider and ATA/SIP phone
02:01.15*** join/#asterisk hads|home (n=hads@mail.nice.net.nz)
02:01.34Tholiarecommended VoIP providers?  I don't need any international calling, and probably spend about 2 hours a month on the phone really, we're mostly a cell house
02:03.55[TK]D-FenderTholia : Then get one that bills by the minute.  ATA's are great if you want to reuse your wiring, etc.  Good first model for you would be an SPA-3102.  That'd give you 1 FXS (extensions), and take in your home analog line as well (1 FXO).  All for $90USD
02:04.18justinu3102 now? any improvements?
02:04.52Tholiawhats a good rate for US only bill by the minute plans?
02:05.00*** join/#asterisk Katty (n=Administ@dialup-4.244.123.58.Dial1.StLouis1.Level3.net)
02:05.20Kattyevening.
02:05.43[TK]D-Fenderjustinu : More room for firmware, etc.
02:05.47[TK]D-FenderKatty: Mew.
02:05.48justinuhey katty
02:05.52FuriousGeorge[TK]D-Fender: i dont need pri, sorry for delayed response
02:06.04w0rmzw3rthTKD-Fender so if I'm looking for just a small pbx for people to dial into and make out going calls I can just get 5 phone lines at voicepulse.com and add them in the AIX config file and I will have 5 lines for 5 people to use to make outbound phone calls correct
02:06.18Katty[TK]D-Fender: hey you. how's it goin?
02:06.24Kattyhey justinu (=
02:06.26[TK]D-FenderFuriousGeorge : There is a subset for the "pseudo state" driver that was hoped to be merged for 1.4 last I recall so you don't need to patch
02:06.27FuriousGeorgetill your internet goes down :)
02:06.45justinui bought expensive dsl
02:06.54justinuthey actually sorta care
02:07.20[TK]D-Fenderw0rmzw3rth : Yes, you could do that... all depends on whats most econimical.  Some places you just pay be the minute and they allow multiple simultaneous calls up to a prescribed limit.
02:08.15*** join/#asterisk billy-jo (n=billy-jo@AClermont-Ferrand-251-1-97-202.w86-206.abo.wanadoo.fr)
02:09.00justinu~ecfo
02:09.05jbotEcho Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out.  Some users describe it as "screeching", "feedback", "static", or other useless terms.  If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO.
02:09.20[TK]D-FenderKatty : Getting by.  Life changes to come, personal stuff.  On vacation now just taking a load off
02:09.27w0rmzw3rth?
02:10.19justinuweird... getting some ECFO on L3
02:10.31justinusounds just like it does with the zap software EC
02:11.14w0rmzw3rthI wouldn't need to buy any hardware then correct if I did it that way [TK]D-Fender
02:11.32justinuyou at least want your own ATA
02:11.41FuriousGeorge[TK]D-Fender: i see now...  i had looked into that patch, but i applied it against the wrong version i guess, im gonna try again with 1.2.9.1
02:11.57[TK]D-Fenderw0rmzw3rth : Depends what you want to use for phones....
02:12.05w0rmzw3rthjustinu where you talking to me?
02:12.06FuriousGeorgewe are talking about the meetermaid patch that was backported to 1.2.9.1 right?
02:12.37skraelings001should PRI manage all the signals and messages from telco? like "this number does not exist", "the number has changed, the new number is ..."??
02:12.37*** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net)
02:12.46w0rmzw3rthI don't need any phone I just want someone to call into my pbx have it answer have them login thier mail box and have options to check mail or place a call
02:12.48justinuw0rmzw3rth: yeah
02:12.59justinuok, then you need a pbx
02:13.12justinuor you need someone else to set you up on theirs
02:13.21*** join/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do)
02:13.24*** join/#asterisk chumper2342 (n=cj@cpe-70-112-211-200.austin.res.rr.com)
02:13.26*** part/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do)
02:13.50justinuskraelings001: that's a good question... typically you provide those messages yourself
02:14.02justinubut there are ISDN cause codes you can return to the telco to indicate such things
02:14.45chumper2342I need a little help with my asterisk 1.2.9.1 server. When i sip reload, it says "..Cannot allow unknown format 'g711'.
02:14.47justinuasterisk has all the voice prompts you need to play them yourself, and it sounds quite professional
02:15.00justinuchumper2342: change it to "ulaw"
02:15.05chumper2342in my sip.conf i have disallow=all, allow=g711,
02:15.07chumper2342ok
02:15.13justinuor "alaw" if you're not in USA
02:15.29skraelings001sometimes i got those messages in a MESSAGE ALERT and they pass the audio before desconnect but usually it doesn't happen
02:16.01justinuskraelings001: the way it's supposed to work, is you get a "progress" IE telling you inband info is available
02:16.17[TK]D-Fenderchumper2342 : Its ALAW or ULAW, not entered as G711
02:16.49skraelings001justinu: you mean only in the case this is available?
02:17.17justinuyeah, in theory
02:17.23*** join/#asterisk |dennis| (n=dennis@200.32.215.82)
02:17.29justinuotherwise, they just send DISCO right away, with the appropriate cause code
02:17.55justinuafk
02:18.09chumper2342ok, we that didn't solve my origional problem lol, when i call a local number it goes out, rings my cell phone then: "No path to translate from SIP/mysoftphone to SIP/Provider-outbound"  "Had to drop call because I couldn't make Sip/mysoftphone compantible with SIP/Provider-outbound
02:18.12skraelings001yes, i checked Q.931
02:19.05[TK]D-Fenderchumper2342 : "allow=g711" = no legit codec name
02:20.27skraelings001i'm sure that my telco is passing such info but can't get it
02:23.28PakiPenguinhas anyone used these http://www.junghanns.net/en/GSM-PCI_produkt.html ?
02:23.36chumper2342ok calls to my cell work but I can't hear anything on either ends. Provider says g729,g711 ulaw. sip.conf = disallow=all, allow=ulaw
02:24.07[TK]D-Fenderchumper2342 : pastebin your sip.conf
02:24.16skraelings001chumper2342: probably problems with nat
02:25.28[TK]D-Fenderskraelings001 : Not yet, so far its a codec incompatability issue
02:25.47iqPakiPenguin: what r u upto ;)
02:26.42PakiPenguiniq,  a lot of things :)
02:27.17iqPakiPenguin: i can see that
02:27.28PakiPenguin:)
02:27.40SplasPoodhttp://newyork.craigslist.org/mnh/mar/166423486.html <-- that job rocks my world :P
02:27.41PakiPenguingod i miss wifi :'(
02:27.54SplasPoodor rather, the job posting does
02:28.04[TK]D-Fender~pb
02:28.07jbotmethinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
02:28.11chumper2342[general]
02:28.11chumper2342context=default
02:28.11chumper2342srvlookup=yes
02:28.11chumper2342register => user:pass@*.com/#
02:28.11chumper2342[telasip-outbound]
02:28.12chumper2342type=peer
02:28.14chumper2342host=*.com
02:28.17chumper2342username=
02:28.18chumper2342secret=
02:28.20SplasPoodtooooo laaaaaaaate
02:28.20chumper2342qualify=yes
02:28.22chumper2342disallow=all
02:28.24chumper2342allow=ulaw
02:28.26chumper2342context=outbound
02:28.28chumper2342[linux-softphone]
02:28.29Strom_Cchumper2342: DON'T DO THAT AGAIN
02:28.30chumper2342type=friend
02:28.32chumper2342secret=
02:28.34chumper2342qualify=yes
02:28.36chumper2342nat=no
02:28.36SplasPoodhe's not done :)
02:28.38chumper2342host=dynamic
02:28.40chumper2342canreinvite=no
02:28.41Strom_Coh dear god
02:28.42chumper2342context=internal
02:28.44chumper2342disallow=all
02:28.46chumper2342allow=ulaw
02:28.46iqchumper2342: please stop !
02:28.48chumper2342[windows-softphone]
02:28.50chumper2342type=friend
02:28.52chumper2342secret=
02:28.54chumper2342qualify=yes
02:28.54SplasPoodI think it's too late for him
02:28.56chumper2342nat=no
02:28.58chumper2342host=dynamic
02:29.00chumper2342canreinvite=no
02:29.02chumper2342context=internal
02:29.04chumper2342disallow=all
02:29.06chumper2342allow=ulaw
02:29.08chumper2342very sorry
02:29.15Strom_Cchumper2342: PASTEBIN
02:29.16anonymouz666doomed
02:29.17SplasPoodhaha
02:29.18anonymouz666hehe
02:29.19SplasPoodthat ruled
02:29.20Strom_Cidiot
02:29.20SplasPooddo it again
02:29.30PakiPenguinshit
02:29.36PakiPenguinidiot!
02:29.51Strom_C~pb
02:29.53jboti heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
02:30.50SplasPoodOh come on guys
02:30.55SplasPoodeveryone makes that mistake once
02:31.14Strom_Ceven after being told to use pastebin?
02:31.18Strom_Cpfffffft
02:31.21Strom_CI think not
02:31.29chumper2342i was in a hurry, sorrry......
02:31.39*** join/#asterisk [pyro] (n=pyro@tor/regular/bracketed-pyro)
02:32.03SplasPoodStrom_C: I think paste had been initiated before ~pb responded
02:32.30Strom_CI WIN!
02:32.46Strom_C:)
02:33.45chumper2342http://bzflag.pastebin.ca/72713
02:34.28Strom_Cchumper2342: now what's the problem, exactly?
02:34.36chumper2342my xlite soft phone is using g711 ulaw
02:34.53Strom_Cand that's a problem?
02:35.00chumper2342when I call my cell from my soft phone, i hear no sound on either end
02:35.03BrijnAnyone has a nice set of Visio stencils for ASterisk/VoiP?
02:35.26Strom_Cchumper2342: is the asterisk box and/or the SIP phone behind a NAT?
02:35.28chumper2342it works calling computer from 1 soft phone to another (internal)
02:35.40chumper2342no
02:35.53Strom_Cthe asterisk box has a public IP address?
02:35.57chumper2342yes
02:36.02Strom_Cand the SIP phone also has a public IP address?
02:36.12chumper2342oops
02:36.17chumper2342but i tried nat=yes
02:36.26chumper2342no
02:36.48Strom_Cchumper2342: so theres a NAT between the asterisk box and the SIP phone, right?
02:36.53chumper2342yes
02:37.09Strom_Cis the SIP provider using ulaw?
02:37.41chumper2342yep, their email says exactly: vocoder: g729, g711 ulaw
02:37.46chumper2342does that mean it needs both?
02:37.50Strom_Cno
02:38.14NotJohnDavidanyone used one of the aastra phones and can compare it to the grandstream gxp2000 ?
02:38.28Strom_Ctry putting canreinvite=no into the ITSP's entry
02:39.24BrijnCool: Nokia 770 panel for < 400US$, better then all this UMPC shit
02:39.32[TK]D-FenderNotJohnDavid : What do you want out of a phone?
02:40.52NotJohnDavidtkd: well I have a gxp2000.  it seems to work.  doesn't seem to shout "quality" to me though.
02:41.28chumper2342nope, didn't work
02:41.53Strom_Cchumper2342: you did a sip reload, right?
02:41.58chumper2342yep
02:42.27Strom_Cyou're sure there's no NAT between your asterisk box and the ITSP, right?
02:42.32NotJohnDavidThere'll be 3 lines coming in.  probably 5 extensions.  the gxp2000 would work but the aastra's look nicer from the small pics i've seen.  didn't know if anyone had used both and would compare the quality of construction/how they are to use
02:42.44Strom_Caastra has higher quality construction
02:42.53Strom_Cof course, anything has higher quality construction than grandstream :)
02:42.59NotJohnDavidfigures :)
02:43.01chumper2342sure, it has a public ip, i can access it from anywhere
02:43.27NotJohnDavidi liked how this grandstream didn't even come with an instruction manual.  or any piece of paper for that matter
02:43.46Strom_Cchumper2342: what happens when you set up a call just between the asterisk box and the PSTN?  does the asterisk box play audio?
02:44.06NotJohnDavidi guess i'd be considering between linksys/polycom/aastra
02:44.15[TK]D-FenderNotJohnDavid : Where are you located?
02:44.22NotJohnDavidtkd: eastern tennessee
02:44.27Strom_Cpolycom is going to be the highest quality of those three, though I like teh cisco phones, to be honest
02:44.36[TK]D-FenderNotJohnDavid : Polycom is for you then.  Looking at PoE?
02:44.49NotJohnDavidwhy is polycom for me?  because i'm in the US?
02:44.53skraelings001anyone help with these piece of verbose output,   http://pastebin.com/732477
02:45.14[TK]D-FenderNotJohnDavid : Polycom Or Cisco, but unless you've got a real deal on Cisco Polycom's quality comes at a far better price point.
02:45.18NotJohnDavidwhat should I call you... tk.. tkd....fender? PoE is a consideration but I haven't spoken with the client.  I think PoE would be the way to go
02:45.36[TK]D-FenderNotJohnDavid : Well just that Polycom is a fair bit more expensive outside North America, yes.
02:45.46PakiPenguinnight everyone
02:45.56NotJohnDavidand they have XML displays ?
02:46.19[TK]D-FenderNotJohnDavid : PoE Will shape your choice of models a bit.  I'd suspect IP430's fine for most workers, IP 601 for areceptionist, and 501's for anyone in between.
02:46.36[TK]D-FenderNotJohnDavid : No, only the IP 601 has XML capabilities.
02:47.50*** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
02:48.18TripleFFFFi know this isnt asterisk related.. well could be. .security wise. .whats the app that checks if any file was modified since last checksum
02:48.43hads|homeTripleFFFF: samhain
02:48.52TripleFFFF?
02:48.52NotJohnDavidare they backlit ?
02:49.06*** part/#asterisk userdefined (n=jross@cpe-24-169-142-23.rochester.res.rr.com)
02:49.16chumper2342nope, don't hear anything either
02:49.31TripleFFFFis ther somethig for windows ?
02:50.14justinuhey paki
02:50.31justinuoh, he left
02:50.39NotJohnDavidtkd: it's a small company.  basically a retail shop that does service calls.  currently 3 employees.  usually one person at the store answering calls, two people on the road.  probably could grow to 5 employees fairly soon.
02:51.10skraelings001justinu: can u take a look at it http://pastebin.com/732477  ?
02:51.26*** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
02:51.44NotJohnDavidi've heard good things about polycom.  i remember i went to their site.  it'd have been nice to have a comparison chart/table
02:51.55*** part/#asterisk Meaw (n=debian@213.6.131.96)
02:51.59justinuthat av-bani guy did a really nice chart
02:52.35FuriousGeorgeanyone using metermaid patch
02:52.56chumper2342Strom_C: still don't hear anything when I call them
02:53.29NotJohnDavidtkd: thanks for the suggestions
02:53.30[TK]D-Fender~phones
02:53.31jbot[phones] at http://bani.anime.net/phones/
02:53.59Strom_Cchumper2342: what about when you try a different ITSP?
02:54.13NotJohnDavidi get the impression I asked the same question a lot of people do :P
02:54.14FuriousGeorgeanyone using the metermaid patch on 1.2.9.1?
02:54.16[TK]D-FenderNotJohnDavid : Polycom are great choices at a great value.  Cisco is a great feel, slightly less reliable firmwar, and pricey, but some have great deals.
02:54.26[TK]D-FenderMost of the rest just don't add up to those 2.
02:54.47NotJohnDavidwish there were a local store
02:54.51orlockHmm..
02:54.56orlock7960 has started locking up hard :-\
02:54.58justinuskraelings001: what are we looking for?
02:55.14chumper2342i called my company's main number (asterisk also) different itsp and it says connected but my call is silent
02:55.26justinuskraelings001: "called equipment is non-ISDN"
02:55.33justinupretty self explanatory... you won't get cause codes in that case.
02:56.19Strom_Cchumper2342: also SIP?
02:56.39chumper2342yep
02:57.22NotJohnDavidyou know $100 a phone is cheap... considering this client would want each phone on a different extension.  FXS port is going to cost at least $60-70 right?
02:57.50[TK]D-FenderNotJohnDavid : Nope, $35/port using SPA-2002's
02:57.53justinuyou can do it on the cheap if you can find a good deal on a used channelbank
02:57.54*** join/#asterisk bjohnson (n=bjohnson@i216-58-10-18.cybersurf.com)
02:58.00Strom_Cchumper2342: try an IAX2 ITSP.  I wonder if your ISP is messing with SIP traffic
02:58.10justinuused cb $300, te110p/A101u $350
02:58.21justinu$650/24 ports
02:58.42justinusorry, 450 for that T1 card
02:58.45justinuso 750/24
02:59.03[TK]D-Fenderjustinu : Assuming it doesn't go flakey on you... and $350 for a 1 port PRI?  New?
02:59.12justinumy bad
02:59.43[TK]D-FenderNotJohnDavid : For the sizes you're talking about SPA-2002 would probably be the way to go.
02:59.57[TK]D-FenderNotJohnDavid : But for larger, the CB idea can work.
03:00.08justinuspa's are great
03:00.15justinusome guys said they do not like heat tho
03:00.31NotJohnDavidtkd: ip phones just seem more flexable
03:00.31*** join/#asterisk FoFiN (n=FoFiN@201.153.31.166)
03:00.39chumper2342holy crap!, just as a test I set shorewall policy to accept all to all, and it works, something in there is blocking it
03:00.48NotJohnDavidtkd: not to mention that everywhere a phone is there'll already be a network connection.
03:00.51Strom_CI need to brush up on my polycom configuration skills
03:00.53NotJohnDavid(next to computers)
03:00.53riddleboxhrmm anyone having troubles with broadvoice?
03:00.57*** join/#asterisk Eigh (n=jason@69.63.224.130)
03:01.00[TK]D-FenderNotJohnDavid : They are.  It feels more natural for sure and with PoE saves on wiring.
03:01.05Strom_Cchumper2342: gah, see, why didnt you mention the bloody firewall?
03:01.41skraelings001justinu: u saw? i try to call to those non-existing phone numbers. In the first case i was able to her the message from telco but when i tryed with the other i didn't hear it. I checked E1(connecting other equipment from telco- independent of asterisk line was good and that was receiving those messages
03:01.44[TK]D-FenderNotJohnDavid : Well you'll want to aim for your phones being on their own lan if at all possible...
03:02.07justinuskraelings001: so in 2nd case, you hear nothing?
03:02.14justinu2nd call, that is
03:02.18skraelings001justinu: aha
03:02.25Strom_Cjustinu: you're in the san fernando valley, right?
03:02.29justinuStrom_C: yes
03:02.32chumper2342lol, sorry, do you know what by chance i need to accept? i have it to accept net-fw and loc-fw 5060
03:02.38Strom_Cgot an extra polycom phone you want to sell? :)
03:02.55NotJohnDavidtkd: yeah it'd be possibly but unneeded just yet
03:03.04justinuno, those all sold
03:03.04Strom_Cchumper2342: you also have to open up ports 16384-32768 for RTP transport
03:03.10justinui'm stuck with some gxp2000's :P
03:03.13Strom_Cheh
03:03.39[TK]D-FenderStrom_C : Maybe :) But not at a price you'd probably find worthwhile :)
03:03.47justinuskraelings001: you may have a legit bug.
03:03.48Strom_Ctry me
03:03.50Strom_C;)
03:04.01justinuskraelings001: asterisk's q931 implementation is a bit odd.
03:04.06*** part/#asterisk zwelch (n=chatzill@pdpc/supporter/sustaining/zwelch)
03:04.30justinuskraelings001: what is the time delta between DISCONNECT and RELEASE?
03:04.34justinuin the last call
03:05.02skraelings001justinu: i couldn't tell
03:05.28justinucan you refer to your full log maybe?
03:05.33skraelings001actually is pretty quickly, i tryed with NoOp and Wait
03:05.53riddleboxcan someone help me resolve this problem http://pastebin.ca/72738
03:06.01[TK]D-FenderStrom_C : Located where?
03:06.07Strom_CLos Angeles
03:06.17[TK]D-FenderStrom_C : Shipping would suck from Montreal...
03:06.26justinudamn you canadians
03:06.31justinuand your french speaking city
03:06.45justinufender: did you go to the F1 race?
03:06.49Strom_Chence why I asked justinu if he was near me ;)
03:07.14[TK]D-Fenderjustinu : Nope.. I'm anti-spectator.....
03:07.17justinubooo
03:07.33justinuactually, i wanted to ask you about flying to montreal
03:07.34justinui want to go
03:07.37justinuany tips?
03:08.10Kattytake a snack.
03:09.19[TK]D-Fenderjustinu : Not much to say... plan your transport to where you'll stay and the rest is gravy...
03:09.48FuriousGeorgeanyone using the metermaid patch against 1.2.9,1?
03:09.48NotJohnDavidtkd: you know you'd think linksys phoens would be similar to the cisco ones
03:09.52*** join/#asterisk nohope (i=1000@201-13-87-52.dsl.telesp.net.br)
03:09.56[TK]D-Fenderjustinu : Pretty easy actually...
03:10.44[TK]D-FenderNotJohnDavid : I owned an SPA-941.  Nothing to write home about.  Definately not in Pollycom's class.  They shine overseas where their price point strongly defeats Polycom though.
03:11.27NotJohnDavidwell i had heard wonderful thigns about the polycoms.  now i just show the client and see what they want :^D  with a heavy hand haha
03:11.37[TK]D-FenderAnd the speakphone is tinny, the handset only slightly.  the pverall phone is too light.  It makes horrific use of the LCD (much like th GXP-2000).
03:12.30chumper2342Strom_C: thanks. works now
03:12.32justinufender: any airline recommendations? air canda seems really expensive out of LAX
03:13.02[TK]D-FenderBasically in North America you are either trying to be cheap (and getting what you payed for), or going decent with Polycom.  Its competitors in its price bracket aren't worth it excepts for very small cases.
03:13.22NotJohnDavidthe gxp-2k seems pretty hard to read unless the backlight is on.
03:13.26Brijnjustinu: See if Air Alaska flies to Montreal as well
03:13.34justinui was thinking of flying up to vancouver
03:13.35[TK]D-Fenderjustinu : Travel these days is an exercise in bargain hunting.  hit the internet sky auctions, etc for deals
03:13.43justinuthen going to montreal from there
03:13.51Brijnjustinu: Air Alaska to YVR was cheap I think
03:13.59Strom_Cjustinu: I'm seeing $530ish roundtrip on expedia from LAX to Montreal
03:14.02[TK]D-Fenderjustinu : Avoice Air Canada if you can and go with an express charter.
03:14.20[TK]D-Fender530$sounds like a great deal.
03:14.34[TK]D-Fender(to my limited knowledge)
03:14.36Strom_C$530 on NWA, with a stopover in Detroit
03:14.51justinubah, stopping sucks
03:15.00Strom_Cleave on a Tuesday, return the following Tuesday
03:17.38JackEStormWell, I think it's official, I've given up on Sixtel
03:17.38justinufender: you should have gone to the race, if only to see jaques villeneuve crash out w/ 7 laps to go
03:17.44justinulaping a backmarker of all things!
03:18.14[TK]D-Fenderjustinu : Sorry, not a "spectator".  I pplay sports, not watch.
03:18.28riddleboxis anyone else experiencing problems connecting to broadvoice?
03:19.23justinubah, enjoy the moment with me, at least :P
03:19.32*** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca)
03:20.10[TK]D-Fenderjustinu : Yeah I can "join in", but never on my own...
03:20.30*** part/#asterisk nohope (i=1000@201-13-87-52.dsl.telesp.net.br)
03:22.46*** join/#asterisk albertito (n=net@host178.201-252-23.telecom.net.ar)
03:23.55albertitoHi! I'm having some problem with a zapata card, I was here earlier. ManxPower helped me and told me it was probably caused by interrupt latency
03:24.20TelamonAnyone have an opinion on the GXP-2000 1.1 series of firmware?  Is it fairly stable or should I stick with 1.0.2.13?
03:24.24justinuis it sharing interrupts?
03:24.53albertitoAs this surprised me, I decided to measure it: I'm using Ingo Molnar latest -rt patch, which includes an "interrupt off latency meter"
03:25.23FuriousGeorgeanyone got any idea what this means:  chan_sip.c:9962 handle_response: Notify answer on an owned channel?
03:25.44FuriousGeorgei suspect that has something to do with why my presence isnt working with this patch and parking
03:25.44albertitoIt reports almost no delay in serving irqs, the max in both CPUs (it's a dual core) is 19 and 20, and average is 0
03:26.18justinualbertito: cat /proc/interrupts
03:26.28albertitojustinu: it's not sharing an interrupt
03:26.42albertitojustinu: (sorry I didn't replied, I wasn't sure you were talking to me)
03:27.22justinuso what's the issue you're having?
03:27.25CunningPikealbertito: What does zttest show?
03:27.43justinualbertito: try switching kernels to non-SMP?
03:27.59albertitojustinu: I'm getting a lot of "PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1"
03:28.05albertitoCunningPike: hold on
03:28.16justinualbertito: ok, that's bad
03:28.36justinuotoh, your circuit could be taking errors
03:28.36CunningPikealbertito: Ah - you may have a timing problem on your PRI
03:28.41justinuor timing
03:28.54riddleboxcan someone tell me what all this means? http://pastebin.ca/72759
03:28.59CunningPikealbertito: pastebin your zaptel.conf
03:29.03albertitoCunningPike: lot's of 99.987793% and some 100%
03:29.40CunningPikealbertito: That sounds reasonable - it doesn't appear to be an interrupt problem - are you running hyperthreading or anything like that?
03:29.45albertitoCunningPike: (this is a debug kernel and the rt measuring it's supposed to introduce some overhead)
03:29.54albertitoCunningPike: no, but this is a dual core box
03:30.05justinualbertito: i would advice trying the mainline kernel
03:30.05albertitojustinu: I'll compile one non-smp in a second
03:30.12justinuand also try non-SMP, just for fun
03:30.13CunningPikealbertito: That's OK - ours are too, but we have noht set
03:30.17albertitojustinu: I run the mainline kernel all the time
03:30.31justinuso you had problems on both mainline and -rt?
03:30.49albertitojustinu: I compiled this one because somebody earlier said it was possibly an IRQ latency problem, and I wanted to measure it just to be sure
03:31.12Telamonalbertito: Just an FYI, I've had bad luck with any asterisk feature that relies on PRI timing with SMP based systems.
03:31.40albertitojustinu: I have the same problem with mainline with HZ=250 and no preempt (but low latency), HZ=1000 and preempt, and rt with and without the debug code (both HZ=1000 and preempt)
03:31.55justinuyeah, it's something else
03:31.58albertitoCunningPike: this CPU doesn't have HT
03:32.22albertitoCunningPike: one second and I'll pastebin the ztcfgf
03:32.30CunningPikealbertito: OK
03:32.34riddleboxcan someone tell me what all this means? http://pastebin.ca/72759
03:32.41*** join/#asterisk Dico_ (n=niko@60.51.217.61)
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03:36.15albertitoCunningPike: http://pastebin.ca/72764
03:36.47CunningPikealbertito: So, running E1?
03:36.51albertitoCunningPike: the other end of the cable is a smaller box (Pentium II) with identical config, but span = 1,0,0,ccs,hdb3,crc4  (difference is the "0" in the second column)
03:36.56albertitoCunningPike: yes
03:37.12albertitoCunningPike: it's a TE110P in E1 mode (I'm in Argentina)
03:37.18justinuback to back with another * box, obviously
03:37.32albertitojustinu: yes, it's a testbed
03:37.34justinuyour config looks good
03:37.45CunningPikealbertito: OK - that's all good - and it's a crossover cable?
03:37.56CunningPikealbertito: Any d-channel errors?
03:38.14albertitoCunningPike: yes. The errors only show on the big box, so I switched ends and they keep appearing on the big box
03:38.37albertitoCunningPike: not that I can see, but hold on, I'll grep the logs just in case I missed it
03:39.15CunningPikealbertito: Ah - that's instructive. I would do what justinu suggested - switch to a non-SMP kernel and see if that makes it go away
03:39.33skraelings001albertito: no deber?as usar crc4
03:39.41CunningPikealbertito: Failing that, I wonder is it a motherboard compatibility issue
03:39.51justinui agree with CunningPike
03:39.55justinuif you can't solve it, change mobos
03:40.07justinuor slots
03:40.31albertitoskraelings001: why not? Anyway, while I can get away without using it now, I'll probably have to use it eventually since it's going to be connected to exchanges that use it
03:40.33CunningPikealbertito: It's the simplest of setups, so it should work no problem. I assume you tried another cable......
03:41.10CunningPikealbertito: I don't mean to be patronizing, but your English is excellent
03:41.15albertitoCunningPike, justinu: I'll build an UP kernel. Any suggestions regarding which one? (mainline, -rt, preempt/no preempt, etc.)
03:41.21justinumainline
03:41.31albertitoCunningPike: thanks =)
03:41.32justinuvanilla, or your distro's patches
03:41.40riddleboxcan someone tell me what all this means? http://pastebin.ca/72759
03:41.43CunningPikealbertito: Yes, vanilla
03:42.21cingo molnar's branch
03:42.24*** part/#asterisk c (i=ix@c-24-60-193-83.hsd1.ma.comcast.net)
03:42.34CunningPikeriddlebox: No - if someone here could, they would - no need to keep asking. You're probably better off posting to the mailing list or trying again in ~12 hours when more people are awake
03:42.41skraelings001albertito: i got the same board and configured without crc4, mainly because of compatibility with telco. but i think i also read it somewhere
03:43.34albertitoOk, I'll be back in a minute with the new kernel
03:45.39albertitoskraelings001: that's strange... if you find the place where you read it, please let me know. I run with crc4 because that's a requirement for the exchanges I use
03:46.08skraelings001sure
03:49.25skraelings001albertito: hey sorry, it's optional ..i didn't put it cause of telco.
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03:51.39mrdigitalcan recommend parts for my asterisk box?
03:52.29justinua cpu, some ram, possibly a pstn interface board
03:52.37justinuhard drive might be nice
03:52.38mrdigitalyeah like whats good enough
03:52.59justinuanything made in the last 2-3 years will do splendidly
03:53.06Greek-Boydo macros only work with an s extension?
03:53.13justinuGreek-Boy: yes
03:53.18mrdigitaljustinu: what FXO board do you recommend?
03:53.35justinumrdigital: either the TDM400p or the SPA-3000
03:53.40mrdigitalSPA?
03:53.42mrdigitalerr
03:53.45justinusipura spa-3000
03:53.45mrdigitalSPA-3000 hmmmm
03:54.00justinusangoma A200 card possibly as well
03:54.04[TK]D-Fendermrdigital : depends how many lines, if you want to use SpanDSP, etc.
03:54.09mrdigital1 line,
03:54.11mrdigital1 extenstion
03:54.19[TK]D-FenderAnd yes, the A200 is a great contender
03:54.31Greek-Boyi've got like 100 extensions which have the same priority list. if i change something i have to change it for all??? i thought macro would help me with that?
03:54.32[TK]D-Fendermrdigital : then the SPA-3102 is for you.
03:54.32justinueach has its pros/cons
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03:54.41justinudigium is very well supported here
03:55.02justinuthats my chief reason for recommending it to newbies
03:55.11mrdigital[TK]D-Fender: can you est a price range?
03:55.24[TK]D-Fender$90
03:55.32justinui think the 4 line capable PCI boards run north of 350USD
03:55.44Qwelljustinu: just at, I believe
03:55.52Qwelldepending on fxo vs fxs configs
03:55.55skraelings001[TK]D-Fender: i have some PRI problems ,can you take a look at http://pastebin.com/732477 ?
03:56.07mrdigitaljustinu p3 500mhz with 256mb and a 20gig hdd sound decent?
03:56.13justinuyeah, just fine.
03:56.57CunningPikeGreek-Boy: If an extension matches your pattern, it will follow the dialplan for that match
03:57.03mrdigitalhow do i connect the 3102 to asterisk?
03:57.09justinuethernet
03:57.24[TK]D-Fendermrdigital : Its a SIP device... just networked to it somehow
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03:59.06Greek-BoyCunningPike, so it can only do patterns and s extensions?
04:00.11CunningPikeskraelings001: In your first case, the dialed equipment is non-ISDN. In the second, "Cause: Unallocated (unassigned) number "
04:00.34[TK]D-Fenderskraelings001 : No idea
04:00.55CunningPikeGreek-Boy: Yes - that's enough for most applications - what are you trying to accomplish?
04:01.26mrdigitalso with the 3102 or 3000 i can use a pstn line with a regular analog phone?
04:02.08CunningPikemrdigital: Yes - it has an FXO and an FXS port.
04:02.13mrdigitalAwesome
04:02.31skraelings001CunningPike: I know both transmit messages from telco (this number does not exist) but i'm only capable to hear the first one
04:02.39CunningPikemrdigital: It provides direct-to-PSTN failover in the event of a network or power failure
04:02.55mrdigitalAwesome so if asterisjk system crashes
04:03.04Greek-BoyCunningPike, not trying to accomplish much. Just trying to avoid changing a priority list a 100 times (ie, for every phone)
04:03.04mrdigital... phone is still usable without reconnecting stuff
04:03.14[TK]D-Fendermrdigital : yup
04:03.20justinuthat's one plus of the 3000, but have never tried that failsafe myself
04:03.21[TK]D-Fendermrdigital : useful in so many ways....
04:03.24CunningPikemrdigital: Exactly - we use them for 911 - with a red phone plugged into each one
04:03.34[TK]D-Fenderbbiab
04:03.35justinuthe batphone!
04:03.37mrdigitalCunningPike: explain your setup
04:03.44justinuCunningPike: explain yourself!
04:03.53CunningPikejustinu: lol
04:03.53justinuwhat is your purpose?
04:04.04CunningPikemrdigital: The whole thing?
04:04.25justinuno one does
04:04.33mrdigitalsur
04:04.34mrdigitalsure
04:06.04CunningPikemrdigital: Well, at 50,000ft level, we have redundant asterisk servers with a TE410P in each connected to the PSTN via a PRI and to our legacy Nortel via 2 PRIs. We have Polycom phones, and use SPA-3000 for other things like 911, elevator phone and other stuff like that. Any questions?
04:07.12fileyes, what's the meaning of life?
04:07.36justinu42
04:07.40justinuthat answer is well known
04:08.00CunningPike~meaningoflife
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04:08.24CunningPikejbot! You should be ashamed of yourself
04:08.27mrdigitalcool setup CunningPike
04:08.32mrdigitaldid you do it yourself?
04:08.39CunningPikejbot, meaningoflife is 42
04:08.41jbotCunningPike: okay
04:08.46CunningPikesheesh
04:08.50justinui have a pretty neat pure SIP setup
04:09.00justinuwhen my "coworkers" aren't fuicking it up
04:09.01CunningPikejustinu has SIP envy
04:09.05CunningPikehee hee
04:09.19justinui used to run TDM switches
04:09.23fileSIP... envy... I never thought I'd hear those two words used in the same sentence
04:09.24justinuwe had 160 T1s
04:09.31justinui'm so glad to be done with that crap
04:09.59CunningPikemrdigital: Thanks - yes - it's home grown. We're pretty pleased with it so far. Our top guy has moved onto it now and loves it
04:10.21CunningPikemrdigital: He also loves the money it saves us
04:10.52CunningPikejustinu: Yes - our telecom guy is just waiting to push our Nortel over a cliff
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04:11.22justinuPRI works better on asterisk than it did on a $150k switch
04:11.51justinudunno if asterisk can do 20 span NFAS groups tho
04:12.10CunningPikejustinu: Not yet ;)
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04:15.07justinusomeone just told me the 1337 color screen cisco phone is SCCP only
04:15.10justinutrue or false?
04:16.06albertitoCunningPike, justinu: I'm using vanilla UP, and it didn't go away
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04:16.15justinualbertito: :(
04:16.36CunningPikealbertito: Bummer - I can't recall if you told me you'd tried another cable......
04:16.39albertitoCunningPike, justinu: I'll try to get a different cable tomorrow (this one worked just fine yesterday, but on a different setup...)
04:16.53justinuchances of it being cable are slim
04:17.01justinuunless your crimps suck that much
04:17.28albertitoCunningPike: I didn't because I don't have any at hand (I'm at home), but I'll get one tomorrow
04:17.43justinualbertito: what about different slots?
04:17.51CunningPikealbertito: Maybe skraelings001 was right........ could you try esf,b8zs or whatever it is?
04:18.09albertitojustinu: yeah, or maybe swap the cards to rule out connector problems
04:18.32CunningPikealbertito: I know you need crc4 for your telco, but it might indicate something
04:18.45albertitoCunningPike: sure, it's worth the try
04:19.24justinuesf is for T1 only
04:19.37skraelings001albertito: CunningPike: justinu: not all of telco work with crc4
04:19.39justinuas is sf/f4
04:19.46justinus/f4/d4/
04:19.48CunningPikejustinu: Ah - I didn't know that
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04:21.03albertitoI do get the errors, although less. I maybe not having the crc4 is making some of the errors go unnoticed
04:22.05albertitohhhmmmmmmm now I got a couple of errors on the other box too (although just 3 of them, quite apart)
04:22.22justinualbertito: since your testbed is back to back, both sides should use the same line-coding
04:22.38albertitojustinu: yeah, they're both without crc4
04:22.56skraelings001albertito: lately i have realized i was getting almost the same but with indication 6, but they are too few
04:23.09albertito(s/I maybe/Maybe/ up there)
04:24.06CunningPikealbertito: What is the motherboard anyway?
04:25.12albertitoCunningPike: it's ccs,hdb3 with optional crc4 for E1 PRI, and (IIRC) cas,hdb3 for E1 R2. The "ccs" stands for "common channel signalling" and "cas" is "channel associated signalling". I don't remember what's hdb3 anymore :S
04:25.35albertitoCunningPike: Intel D945G, with a Pentium D 830
04:26.20CunningPikealbertito: Hmmm - that should work. I don't think any Intel motherboards give many compatibility issues, do they?
04:26.42Qwelljustinu: 7970?  false
04:26.57Qwell7985 however, I believe is sccp only
04:27.08justinucool, thx
04:27.10albertitoCunningPike: I've always used Intel motherboards with digium cards and never had an incompatibility problem before... but it's the first one I try this model
04:27.39albertitoCunningPike: I'll swap the cards as justinu suggested, to rule out connector and slot problems. I'll be back in a minute
04:27.50CunningPikealbertito: OK
04:28.17skraelings001night fellows
04:28.54*** part/#asterisk skraelings001 (n=skraelin@201.230.111.182)
04:29.34justinualbertito: this stuff crops up once and a while, strangely enough it's always affecting you south americans and E1s
04:29.43justinu(mobo incompatibility)
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04:34.35w0rmzw3rthWhat can I use for a Cheap Backbone for a WISP of a 50 mile radius for about 100 customers
04:37.45Qwellw0rmzw3rth: umm...that's a bit...out of scope
04:40.08w0rmzw3rthwould a tier 1 cut it?
04:41.13inv_ArpQwell: heh awol would prob know...
04:41.54Qwellinv_Arp: indeed, but it's quite complex :p
04:43.39FuriousGeorgesomeone tell me if the fhis sounds feasable:
04:44.43FuriousGeorgei wanna set a button on my snom to call an extension and set a db_variable PAGE_FLAG for each user that, upon the next call, it setssipheader to page the phone
04:45.08Qwellsure
04:45.09FuriousGeorgeso if they call another snom after setting page_flag (by hitting the corresponding button) it pages that phone
04:45.44FuriousGeorgeQwell: based on the CID i check for the flag, then after hangup i reset it?
04:45.49Qwellsure
04:46.01FuriousGeorgeQwell: thanks
04:46.08Qwellsee above
04:46.33FuriousGeorgeQwell: how far up?
04:47.07FuriousGeorgeQwell: what you mean "see above"
04:47.13CunningPikeFuriousGeorge: I think Qwell is humoring you
04:47.13FuriousGeorgeare you just joshing me?
04:47.34CunningPikeQwell: Is actually just a 'sure' bot
04:47.38FuriousGeorgeQwell: now i dont know if you were serious about the feasability of my proposal
04:47.44QwellCunningPike: something like that
04:48.33QwellFuriousGeorge: You and me both
04:49.33FuriousGeorgeQwell: but you in ernest think ill be able to set a page button as described above on my phone?
04:49.42Qwell...sure
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05:03.14albertitoCunningPike, justinu: I swapped the cards between boxes and the problem swapped with them
05:03.34justinuodd
05:03.39CunningPikealbertito: Interesting - could be a dodgy card then
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05:03.48CunningPikealbertito: Got another one? :)
05:03.52albertitoSo I used a third card I had and it has the same problem, but looks worse (more errors per second)
05:04.02justinuvery odd
05:04.06CunningPikealbertito: Are all 3 cards the same type?
05:04.25justinudo you have a BER tester?
05:04.27drrayI need a better phone than a cisco 7960, for forwarding calls, I have a site that can't forward calls.. any suggestions?
05:04.44albertitoYes, all 3 are exactly the same TE110p model, and were ordered together about a year ago, but weren't tested until now
05:05.10albertitojustinu: I don't know what that is... is that a regular electric tester?
05:05.23justinuno, a bit error rate tester
05:05.34justinuthe ones I use are Sunrise Telecom Sunset T1s
05:05.43albertitojustinu: I'm afraid not, I don't have one around
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05:06.20justinualbertito: i'm out of ideas, very strange indeed
05:06.24albertitojustinu: what could I use it for?
05:06.31justinuto test the framer on the digium cards
05:07.04justinuanyways, bedtime
05:07.12albertitojustinu: maybe they're just faulty... or have some connector problem
05:07.18*** part/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
05:07.19albertitojustinu: thanks a lot for all your help! sleep well
05:07.56CunningPikealbertito: I'm out of ideas, too.
05:08.00CunningPikealbertito: Sorry
05:08.08CunningPikealbertito: Where did you get the cards>
05:08.09CunningPike?
05:08.17albertitoCunningPike: please, you've given me invaluable help!
05:08.27albertitoCunningPike: Imported from Digium USA
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05:08.44CunningPikealbertito: I would phone Digium in the morning........
05:09.05CunningPikealbertito: I think you have a couple of bad cards - not very usual, but it sure sounds like it
05:09.13albertitoCunningPike: I will, and I'll retry both cards just in case
05:09.20CunningPikealbertito: OK - good luck!
05:09.38albertitoCunningPike: thanks! I'm off to bed too
05:09.55CunningPikealbertito: OK - good night
05:09.59albertito(after rebooting to get back my SMP kernel =)
05:10.01albertitonight!
05:10.03CunningPikeBuenos nochas
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05:29.00stephane_jour
05:33.11CunningPikenuit
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05:46.34FuriousGeorgeis there any way nat can mess with presence indications on chan sip and chan local?  i just installed the metermaid patch on 12.9.1 and the LED indications for parking spots work flakely, and ringing indications are not working
05:46.49FuriousGeorgetake that back, its working for ringing
05:47.35FuriousGeorgeits the presence on parking patch that i installed (metermaid) that is flakey.  could it be a nat thing, and i can expect it to work ok from inside the building
05:47.36FuriousGeorge?
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05:57.22X-Rob_yes it could be a nat thing.
05:57.40X-Rob__could_
05:57.44X-Rob_dunno if it is tho 8)
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06:03.05X-Rob_Woo.
06:03.12X-Rob_Call pickup works on the GXP2000's too
06:06.37*** join/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au)
06:07.43[hC]X-Rob_: you mean like, *8?
06:07.54X-Rob_as in BLF-push-the-flashy-light pickup
06:08.10[hC]ahh
06:08.16P-NuTHey all, I need some help with SIP trunking. As in, how to I build one..
06:08.28[hC]on like, a parked call?
06:08.31[hC]what are you picking it up from/
06:09.07X-Rob_no, like xtn 345 is ringing
06:09.11X-Rob_the light for xtn 345 flashes
06:09.17[hC]ohhh i see. :)
06:09.17X-Rob_you push the button next to the light
06:09.20X-Rob_and you pick up the call
06:09.32[hC]gotcha
06:10.12X-Rob_omfg
06:10.17X-Rob_the ligths just went berko
06:10.21X-Rob_they've got a ring-all group
06:10.27X-Rob_watching all the lights flash on the gxp is most amusing
06:10.27X-Rob_8)
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06:15.29smackusi am stuck trying to get cdr_mysql working on my new asterisk server.
06:15.46smackusi have edited the configs, installed asterisk-addons
06:15.48smackuscan anyone help me?
06:17.55smackushas everyone gone to be?
06:17.57smackusbed?
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06:23.01smackusi have run the asterisk-addons install, and expected to see cdr_addon_mysql.so. any idea as to how to resolve this?
06:26.50qdksmackus: recompiile asterisk?
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06:28.14smackusqdk: tried that
06:28.34smackuswhat is the link to the svn version of the asterisk-addons.
06:28.36smackusi will try that
06:28.53qdksmackus: and you have mysql and its libs installed?
06:29.42smackusi think so
06:29.44qdksmackus: svn shouldnt give you any better results.
06:29.44smackusmysql for sure.
06:29.53smackushow can i tell for sure that I have all of the libs
06:29.54qdksmackus: ok, dist?
06:29.54smackus?
06:30.01smackusred hat ent
06:31.29qdksad linux, anyway, the libs might be in a seperate packages as most other, and propper, packagessystems have chosen to do.
06:33.54smackushmm ok
06:34.19*** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)
06:35.31smackusas far as the mysql packages that i have installed...
06:35.56smackusmysql-4.1.20-1.RHEL4.1
06:36.14smackusmysql-devel-4.1.20-1.RHEL4.1
06:36.42smackusmysqlclient10-3.23.58-4.RHEL4.1
06:36.54smackusmysql-server-4.1.20-1.RHEL4.1
06:37.07smackusphp-mysql-4.3.9-3.12
06:37.18smackusam i missing anything important to cdr_mysql?
06:37.29rob0The -devel one is probably it.
06:37.30*** join/#asterisk phalacee (n=Sunforge@202.3.110.65)
06:37.43smackusthe one i am missing, or the one i need?
06:38.41rob0The one you need. Were there errors in the "make"?
06:39.10smackusmake of the asterisk-addons?
06:39.14*** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se)
06:39.14smackusi did not notice any
06:39.33smackusi have mysql-devel installed.
06:40.22qdksmackus: perhaps -lib or something.
06:40.43rob0and "make install" installed it?
06:40.44qdksmackus: anyway, RPM-linux is hell, so good luck.
06:41.27smackusqdk:thanks
06:41.46smackusrob0: is there a way to tell that it is for sure installed?
06:42.01qdksmackus: rpm -qa?
06:42.28tzafrirrpm -q packagename
06:43.14tzafrirAre those the rpm packages from mysql.com?
06:43.56smackusi was asking is there a way to know for sure if the asterisk-addons was installed
06:44.14tzafrirmysqlclient10 is obsolete. Unless you need to communicate with an old mysql 3.23 server
06:44.26rob0Look in the Makefile, see where the file should have been copied, see if it's there.
06:45.18rob0configure it, try to use it :)
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06:53.26locelavihi evrybody
06:53.47smackusok, so i am starting from scratch
06:54.02locelavijust a little question...
06:54.06locelavi...of dead
06:54.15locelaviand with iptables
06:54.20smackusi run "perl -MCPAN -e "install DBD::mysql""
06:54.42smackusI get an error... make test had returned bad status, won't install without force
06:54.49locelavihow can i configure it with sip ?
06:55.25locelavii can open 5060 all right
06:55.27smackus"/usr/bin/make" was ok
06:55.29[hC]so, any of you guys got hudlite going on a non trixbox pbx?
06:55.41smackusis there a way to skip "make test"?
06:55.43*** join/#asterisk codestr0m (n=asura@ns2.netsyncro.com)
06:55.59locelavithe phon s ringinig
06:56.11codestr0msomeone willing or know how to help me debug chan_ss7
06:56.34locelavibut can't ear somthing
06:57.15locelavii don't want to open a rank of port
07:00.03Strom_Clocelavi: the only way to do SIP through a firewall is to either (a) restrict the media ports every single SIP device will ever use (and that can be hairy) or (b) open up ports 16384-32768
07:03.05locelavii thought that when the phones were conected the datagrams didn't pass throught the server ?
07:03.08*** join/#asterisk Johnnie (i=odysseus@pdpc/supporter/active/Johnnie)
07:04.10Strom_Clocelavi: that can be the case, but you can also set up Asterisk to not allow reinvites
07:04.26Strom_Calso, if asterisk has to do transcoding or needs to stay in the media path for any reason, it typically will
07:04.38*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
07:04.43Strom_Cand usually you want asterisk to stay in the media path for proper call detail record generation
07:04.52codestr0mStrom_C: Have you ever poked with chan_ss7?
07:05.10Strom_Ccodestr0m: no, I havent...though I don't see why you'd be having /audio/ issues, since SS7 is just signaling
07:05.13locelavii don t understand "poked". i m french
07:05.15kmilitzercodestr0m: I am using chan_ss7 ...
07:05.54kmilitzercodestr0m: But I just came in, so I don't know what your problem is ;)
07:05.58codestr0m<PROTECTED>
07:06.13Strom_Clocelavi: babelfish tells me "poke" translates to french as "poussé"
07:06.34*** part/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au)
07:06.51locelaviok sorry
07:07.32kmilitzercodestr0m: If I recall right not even developers have a good explanation why this happens from time to time ...
07:07.35locelavican i really open a lot of port on Internet ,
07:07.44locelavii think it s no serious
07:08.05kmilitzercodestr0m: Can you hear these glitches in your calls, or do you just get the messages?
07:08.17codestr0mkmilitzer: what versions are you using and is it working in production pretty well for you.. this is my first round trying to debug chan_ss7
07:09.14kmilitzercodestr0m: I am using the latest version of chan_ss7. It's working now for a small group of users since early this year (starting with chan_ss7-0.2)
07:09.44kmilitzercodestr0m: I am going to put more users on it in the next month ...
07:10.31*** part/#asterisk phonic (i=phonic@antisocial.nu)
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07:11.37codestr0mkmilitzer: I'd really test that before going full on with it...
07:11.50codestr0mand are you using digium?
07:11.55Strom_Cwhy are you going with SS7, kmilitzer?
07:12.07kmilitzercodestr0m: Yes, I am using digium TE205P cards
07:12.08smackusok, so is it just me, or has anyone else using redhat enterprise had issues getting asterisk-addons installed.
07:12.24smackusi have been at this for a couple of days now, and it looks like i cant even get the basics to install
07:12.38kmilitzerStrom_C: Becase there is no other way to get different numbers on one E1-trunk ...
07:12.52Strom_Ckmilitzer: what do you mean "different numbers"
07:12.55smackusi have tried to install the "perl -MCPAN -e "install DBD::mysql"" wont go
07:12.56kmilitzerStrom_C: So if I would stick with an E1, I would not be able to port numbers
07:13.09smackustried to connect to the cvs.digium.com
07:13.14smackusunknown host.
07:13.17smackusi am stumped
07:13.20smackusany help?
07:13.21kmilitzersmackus: cvs is offline
07:13.22Strom_Csmackus: uh, cvs has been deprecated forever
07:13.25kmilitzersmackus: Use svn
07:13.26Strom_Cuse svn
07:13.35Strom_Cecho
07:13.42smackusok... mystery solved on cvs
07:13.52Strom_Ckmilitzer: what do you mean "different numbers"
07:13.54smackusI am just going off of http://www.voip-info.org/wiki/index.php?page=Asterisk+addon+asterisk-addons
07:14.20smackusany idea as to how to get past the perl module for DBD_mysql?
07:14.24kmilitzerStrom_C: As I said. Ported number, different blocks of DIDs (for example 01234-5678xxx and 012345-89012xxxx)
07:14.33codestr0mkmilitzer: any ideas on trying to troubleshoot this?
07:14.40Strom_Ckmilitzer: what, and your telephone company won't do PRI?
07:15.33kmilitzerStrom_C: My TelCo (which is acutally our mother company ;) ) cannot do this on an E1, only on DID block works on E1s
07:15.59Strom_Cthat's pretty dumb
07:16.04smackusi had all of this working just fine on FC5, but digium recommends using RHEL4, so I moved over to it. now i have had issues getting stuff installed.
07:16.25kmilitzerkmilitzer: The audio-problem? No idea. There were some rumors on the asterisk-ss7 mailinglist, that it might only happen when using SIP and/or IAX and you have a slight packet loss
07:16.41smackusall i want is to install asterisk-addons (crying)
07:16.53kmilitzerStrom_C: I thought that this was a problem with every TelCo (at least in germany)
07:17.13Strom_Ckmilitzer: no, in theory, multiple noncontiguous DIDs should be deliverable over PRI
07:17.33Strom_Ckmilitzer: I know I've done installs here in the U.S. where the telco just assigns DIDs all over the place
07:17.40kmilitzerStrom_C: And what about ported numbers?
07:17.48Strom_Cported numbers work just fine
07:18.00kmilitzerStrom_C: Maybe it's only a problem in europe ... or just with the switch of the telco ...
07:18.04nounoursfrmorning all
07:18.16Strom_Ceither that or the telco is lazy and doesnt want to learn to provision their stuff correctly
07:19.03kmilitzerStrom_C: Nevertheless, SS7 scales better, when adding voice slots, because you need only one slot for signaling ...
07:19.23codestr0mkmilitzer: are you using digium hardware?
07:19.24kmilitzerStrom_C: Failover works better and so on ...
07:19.26Strom_Ckmilitzer: and with PRI you can have one D-channel for multiple E1s
07:19.38kmilitzercodestr0m: Yes, I do ...
07:20.14kmilitzerStrom_C: AFAIK you have for every E1 on signaling channel, that cannot be used by other E1s.
07:20.26codestr0mkmilitzer: would you maybe be so kind as to post your ss7.conf to pastebin?
07:20.27kmilitzers/on/one/
07:20.44Strom_Cthere's a framing channel on all E1s, sure, but thats the case regardless of whether its channelized or PRI
07:20.48*** join/#asterisk Gamercjm (n=chris@pool-71-254-175-156.lsanca.fios.verizon.net)
07:21.01GamercjmWhats the best way to use fax with asterisk?
07:21.05Strom_Cwith PRI, say you have 30 B-channels and 1 D-channel on the first E1
07:21.25Strom_Cyou can have that same D-channel handle signaling for 31 B-channels on a second E1
07:21.25*** join/#asterisk _4d4m_ (n=adam@62.69.102.99)
07:21.29Strom_Cand on a third
07:21.30Strom_Cand so on
07:21.47Strom_CGamercjm: keep it all-TDM
07:21.52kmilitzerStrom_C: Are you sure? I think this is only possible for T1 and not for E1
07:22.20Strom_Ckmilitzer: I'm not completely sure, but I believe ITU-T PRI specs allow for it
07:22.27kmilitzerThe only way to use one D-channel for more than one E1 (read: two E1s) is V5.2
07:22.44GamercjmStorm_C: whats TDM?
07:22.51Strom_Ctime division multiplexing
07:23.27Strom_CGamercjm: don't use fax over voip
07:23.33Strom_Ckeep it all time-division
07:24.19Gamercjmits more of to just learn about asterisk/fax, so doesnt need to be as reliable or anything like that
07:24.23Strom_Ckmilitzer: ah, what do I know.  I'm just some idiot American telecom consultant ;)
07:24.33Gamercjmso i was wondering what the best fax method or program would be
07:25.20kmilitzerStrom_C: The real differences between the old and the new world can only be seen at a second look, I guess ;)
07:26.16Strom_Ckmilitzer: I'm actually extremely surprised that the telco provisioned SS7 links to your premises
07:26.24Strom_Cnothing like that would ever happen her
07:26.26Strom_Chere
07:27.17kmilitzerStrom_C: As I said, it is our mother company, so technically it's an "internal" deal ... I don't think it would have been easy to get SS7 as a common customer ;)
07:27.49codestr0mhttp://pastebin.ca/72921 That's a sample of what I'm seeing.. I don't know if it's SS7 causing this or maybe something else.. does it give more information?
07:29.26kmilitzercodestr0m: Post more info about your system, like CPU, shared interrupts, etc. Are you using echo cancelation?
07:29.49codestr0mkmilitzer: the interrupts looked okej to me.. one sec though
07:35.39*** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net)
07:36.29raidenzHello
07:37.05codestr0mkmilitzer: http://pastebin.ca/72927  (I just started working on this box today and tell me if anything seems funny there..) hmm...
07:37.21raidenzDoes anyknow how to get the SIP Response message back from a failed dial. I want to see what the provider has sent back to me [running Latest Asterisk SVN]
07:40.16*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
07:40.28kmilitzercodestr0m: Your config/system setting looks good to me, nothing strange to see.
07:40.52kmilitzercodestr0m: How do you get your calls to the ss7-system? Via SIP?
07:41.27codestr0mit's coming in on zap
07:41.56codestr0mtoll free is delievered via zap and then going out over ss7
07:42.31kmilitzercodestr0m: Hmm, then the audio-losses cannot be related to packet losses ...
07:42.57kmilitzercodestr0m: Try to contact sifira.dk (developer of chan_ss7), maybe they have an ide
07:43.03kmilitzers/ide/idea/
07:43.41codestr0mkmilitzer: is there an irc where sifira.dk dev(s) hang out?
07:43.56*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
07:44.22kmilitzercodestr0m: No, I fear not :( try chan_ss7@sifira.dk, or better look it up at their website
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07:56.26smackushow can i change the location where mixmonitor stores the recorded calls?
07:56.45Strom_Cthats an argument to mixmonitor, IIRC
07:56.56Strom_Cshow application mixmonitor
07:57.45smackusthank you
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08:22.51tengulre11HI,all
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08:28.48h3xasterisk cdrs are so useless
08:29.01Strom_Chow so?
08:29.07h3xthe call durations are all off
08:29.15h3xhow can you have a 0 second ANSWERED call
08:29.42Strom_Ca call where the delta between the answer and hangup messages is less than one second?
08:29.50Strom_Cit can happen, you know
08:29.58h3xright but what carrier charges nothing for an answered, 0 second call
08:30.11h3xnobody, except those that dont know and are using asterisk
08:30.11h3xheh
08:30.33h3xits also usually off by a second versus the carriers records
08:30.50Strom_Cwell, if they've written their billing software correctly, they'll strip out anything less than two seconds
08:30.53Strom_Coooh, a whole second
08:30.54Strom_Cgasp
08:30.57Strom_Cthe horror
08:31.00h3xI think the solution would be to measure call durations in tenths of second precision and round up
08:31.17h3xmultiply that by a few million calls and you got a big chunk of change
08:31.39h3xeh, strip out <2 second calls? who does that
08:31.43h3xit aint a cell phone
08:31.55h3xintercarrier compensation happens
08:32.08Strom_C2-second billing delay has been standard telco practice for decades now :)
08:32.25h3x2 seconds the wrong way! heh
08:36.59h3x<PROTECTED>
08:36.59h3x<PROTECTED>
08:37.30h3xthat figures
08:37.31h3xlook at this
08:37.50h3x<PROTECTED>
08:37.50h3x<PROTECTED>
08:37.50h3x<PROTECTED>
08:37.50h3x<PROTECTED>
08:37.50h3x<PROTECTED>
08:38.18Strom_Cso file a bug report and/or write a patch
08:38.19h3xand heres the dirty code
08:38.30h3x<PROTECTED>
08:38.31h3x<PROTECTED>
08:38.31h3x<PROTECTED>
08:38.31h3x<PROTECTED>
08:39.23h3xso the fix would be to keep all 3 call counters in milliseconds
08:39.31Strom_Cfiling a bug report and/or submitting a patch will do a lot more good than kvetching about it in #asterisk
08:39.34h3xin memory anyway, and subtract those
08:39.45h3xheh i will do that :P
08:39.52h3xbut it dosent hurt to debate the best way to do it
08:39.58Strom_Cor, hell, kvetch about it in #asterisk-dev
08:40.04Strom_Cthey'll care more than me
08:40.21h3xtrue. heh
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08:53.50*** join/#asterisk QuAtRo[NL] (n=QuAtRo_@dsl-083-247-051-039.solcon.nl)
08:54.07QuAtRo[NL]I have some kind of ghost calls
08:55.10*** join/#asterisk razu (n=razu@tln-kontor.norby.ee)
08:55.15QuAtRo[NL]When a call ends unexpectedly, it seem te keep a ghost call open
08:55.46QuAtRo[NL]The phone isn't ringing again in the queue
08:56.14*** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com)
08:56.40|oranjia|i usually get those when i have a softphone making calls
08:56.45|oranjia|like xten
08:56.47|oranjia|or wengo
08:57.14h3xyeah me 3
08:57.59QuAtRo[NL]Is that a bug or what?
08:58.28|oranjia|but me this : http://www.thinkgeek.com/computing/input/8193/zoom/
08:58.38*** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com)
08:58.45QuAtRo[NL]Is there some place where i can change the timeout time before a call is killed by Asterisk
08:59.10|oranjia|buy me that
08:59.20h3xyeah
08:59.24h3xshow application dial
08:59.40h3xand i think theres a dialplan variable you can set
09:00.34codestr0mdoes svn latest release branch of 1.2 have the jitterbuffer in it already or do I have to the patches? or go pull trunk?
09:00.35RoyK<PROTECTED>
09:02.39*** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr)
09:03.09kay2Is there any Framework in C for AGI/FAGI ?
09:11.38*** join/#asterisk op3r (n=op3r@124.107.26.34)
09:11.45op3rhi
09:11.47op3ranyone up?
09:11.53Strom_CI am!
09:11.57op3rhi Strom_C
09:12.01Strom_Chi hi
09:12.15op3rmahy i ask whats this error
09:12.17op3r/usr/bin/ld: cannot find -lssl
09:12.17op3rcollect2: ld returned 1 exit status
09:12.17op3rmake: *** [asterisk] Error 1
09:12.37op3rIm trying to install asterisk and thats the first time i saw that error
09:13.04QuAtRo[NL]Asterisk says:  'Agent/3 is ringing' but in phone isn't ringing...
09:13.06Strom_Cinstall libssl
09:13.13Strom_Clike the instructions tell you to
09:13.15op3rhmm ok
09:13.49op3rI thought there was a problem with openssl because it is already installed
09:13.51RoyKop3r: apt-get install libssl-dev
09:14.05op3rso no more open-ssldev?
09:14.13RoyKldconfig -v |grep libssl
09:15.02op3ri think open-ssldev solved it
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09:17.05kmilitzerh3x: If you are going to implent msecs for CDR-calculation make sure these are also writte into the CDR-record ...
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09:34.38kay2lol
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09:35.31kay2could anyone tell me how I could link two channels from AMI ?
09:35.42darkgamer20buying a X100P PCI card and 2 analog phone adapter is the same as buying a TDM400P with 1 FXO module and 2 FXS modules right? just wanted to confirm before I make my purchase
09:37.28kay2lol
09:37.28kay2no
09:38.07darkgamer20kay2: were you answering my question?
09:39.18stringslayerEffectiv use would be the same -- but quality would be be as good
09:39.53stringslayeroops -- quality would NOT be as good
09:40.12darkgamer20you mean the voice quality would not be as good?
09:40.20stringslayercorrect
09:40.33stringslayeroverall throughput would be less
09:40.48darkgamer20damn
09:40.59darkgamer20would this be ok for a hobby project/home
09:41.01darkgamer20?
09:41.44stringslayerSure- but not for business
09:42.32darkgamer20oh ok
09:42.38darkgamer20thanks stringslayer
09:43.17stringslayernp
09:43.48stringslayerdo you already have your ata's
09:44.01darkgamer20no I am going to buy that now
09:44.25stringslayerSeach ebay for dta 310
09:44.37stringslayerYou can buy them for less than $10 each
09:44.57stringslayerthey are locked with packet8 firmware- so everyone sells them cheap
09:44.59stringslayerbut
09:45.23stringslayergo here and you can unlock it and they work great -- I have 8 of them lol
09:45.39darkgamer20wow
09:45.45darkgamer20big talker eh?
09:45.45stringslayerhttp://www.stromcarlson.com/projects/dta-310
09:46.25stringslayerNah -- I let friends/family use them to call me ..etc..
09:46.39darkgamer20ohh i see
09:46.52*** join/#asterisk RoyK (n=roy@cD9088681.inet.catch.no)
09:46.56darkgamer20through the internet i suppose?
09:46.57stringslayerThey have extensions to my box
09:47.00stringslayeryep
09:47.30darkgamer20oh ok so you configure them so that they can connect them to your box through the internet and talk to your for free
09:47.35darkgamer20right?
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09:48.07stringslayerYep, its a beautiful thing..
09:48.12darkgamer20man thats sweet
09:49.11stringslayerAre you keeping your pots line?
09:49.18darkgamer20this site http://www.digitnetworks.com/store/product_info.php?products_id=28 says that i can call long distance for free, now do i have to subscribe for long distance from my phone company? my phone service is att
09:49.35darkgamer20or can i somehow configure that with asterisk?
09:49.47kay2hey
09:49.50stringslayerAsterisk is the way to go
09:50.11kay2how can I put someone in MusicOnHold() until the callee answers his phone ?
09:50.11stringslayerTry telasip.com or teliax
09:50.13kay2is it possible ?
09:50.47*** join/#asterisk blitz[laptop] (n=blitzrag@217.41.237.104)
09:50.53stringslayerbackground
09:51.15*** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au)
09:51.21blitz[laptop]foreground
09:52.04stringslayerno, use background  in the extension config for that extension
09:52.07*** join/#asterisk fulgas (n=fulgas@82.102.2.30)
09:52.32darkgamer20stringslayer: so in order for me to have long distance i have to sign up for it, either from a voip company or a phone company right?
09:52.47blitz[laptop]yes -- asterisk != free calls
09:52.51stringslayeryea --
09:54.16darkgamer20blitz[laptop]: lol just say not not all of us are programmers
09:54.41stringslayerIf you only have voip-- then you only need to buy one ata
09:55.16QuAtRo[NL]I have a problem with calling
09:55.21P-NuTHey all,
09:55.29P-NuTjust so I have this right in my head,
09:55.54P-NuTif I have an SPA3000 and I want to dialout pf the PSTN port on it,
09:56.07QuAtRo[NL]When i call from one phone to another Asterisk says: Calling 1 \n Agent/1 is ringing
09:56.18QuAtRo[NL]But in fact, the phone isn't ringing at all
09:56.25P-NuTI just create an extension in asterisk for the spa3000, and I then dial that followed by a number?
09:56.30P-NuTkinda?
09:57.27darkgamer20stringslayer: thanks a lot for your help, that dta 310 helped me cut my costs by big margin
09:57.29darkgamer20thanks again
09:57.33darkgamer20see you guys
09:57.51*** join/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl)
10:01.36kay2how can I put someone in MusicOnHold() until the callee answers his phone ?
10:02.19*** join/#asterisk nXOR (n=drade@pdpc/supporter/sustaining/nXOR)
10:02.30nXORhello ppl, i have a problem with my asterisk setup
10:02.43stringslayerwhat say you
10:02.48nXORlocally it works fine, however when i try to make a call outside via isdn
10:02.53*** join/#asterisk denon (i=denon@synapse.subneural.net)
10:02.53*** mode/#asterisk [+o denon] by ChanServ
10:02.58nXORit starts then 503's me
10:03.18nXORi see that visdn interface is being raised in asterisk console
10:03.25sshadowkay2: use dial(SIP/xxx|m)
10:03.26nXORbut then it dies
10:03.27stringslayernever messed with asterisk and isdn,, sorry
10:03.34nXORhm
10:03.51nXORdo you perhaps have any good resources on setting up visdn and asterisk
10:04.03nXORive roamed the wastelands of internet searching for a quality tutorial
10:04.18stringslayertry google search for trixbox without tears
10:05.51*** join/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net)
10:07.31P-NuTif I want to dial out of the SPA300 do I use Dial(SIP/${EXTEN:1},70,Tt)  ?????
10:07.32*** join/#asterisk Gamercjm (n=chris@pool-71-254-175-156.lsanca.fios.verizon.net)
10:07.49P-NuTand do I need to CREATE a channel in sip.conf to dial out of?
10:09.53kay2sshadow: what does the |m does ?
10:10.29sshadowkay2: m([class]) - Provide hold music to the calling party until a requested
10:10.29sshadow<PROTECTED>
10:10.29sshadow<PROTECTED>
10:11.05kay2sshadow: thx
10:11.29sshadowkay2: you're welcome
10:11.38kay2sshadow: so if the callee doesnt pick up
10:11.48kay2I could put in the next extension MusicOnHold()
10:12.01Joe__11can anybody help me to understand how chan_local.c works? what does the function 'ast_channel_masquerade' do?
10:12.20blitz[laptop]Joe__11: thats probably more of a #asterisk-dev question...
10:12.38Joe__11ok
10:12.43blitz[laptop]Joe__11: not sure if anyone in here will be able to answer that -- although none of the main devs seem to be up yet
10:15.46sshadowkay2: no. just do this:  exten=> 100,1,Dial(SIP/mysip||m)
10:16.03sshadowkay2: the m option will start MOH
10:24.00blitz[laptop]jeeebuz -- my ssh tunnel worked first try and I can connect to the DB via pgAdmin :)
10:26.04lilalinuxWhen I place a Wait(15) before an Answer(), the CAPI sends a DISCONNECT_IND after 10 seconds. The Telephones continue ringing, but asterisk doesn't handle it anymore.
10:27.19kay2sshadow: that has a lil probleme
10:27.53kay2let say that I want to dial someone that is in an other asterisk, would it work ?
10:27.53*** join/#asterisk tardisx (n=justin@ppp167-251-29.static.internode.on.net)
10:31.14tardisxhi, can you use + signs in dialplans? My handset is passing numbers like +61555555555 to asterisk, and I don't seem to be able to match it (I really just want to strip the plus off)
10:31.46lilalinuxtardisx: then use ${EXTEN:1}
10:32.03*** join/#asterisk mrtwister (n=ambervoi@107.250.broadband5.iol.cz)
10:32.16tardisxbut other handsets don't do that, so I'd be stripping off the important first digit
10:32.50*** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no)
10:33.15tardisxI tried this: exten => +61.,1,Goto(default,0{EXTEN:3},1)
10:33.23tardisxbut it doesn't seem to ever match
10:33.38lilalinuxyou have to use _ if you want an expression
10:34.08lilalinux:)
10:35.21tardisxthanks muchly, I think that's working now
10:35.23Aurshello
10:36.55*** part/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net)
10:37.00sshadowkay2: i think that would work too. haven't tried it, but it should
10:37.32Aursany expertise on polycom phones here? :)
10:40.00blitz[laptop]Aurs: you should probably just ask an actual question
10:40.53QuAtRo[NL]What might cause that my Asterisk says Phone 3 is ringing but in fact it isn't
10:41.14*** join/#asterisk pa (n=paolo@unaffiliated/pa)
10:42.17AursI wonder what this means:
10:42.19Aurs<PROTECTED>
10:42.41Aurs"persist"... does that mean that the setting will not be reset, or something?
10:42.55lilalinuxhow do i turn off "capi debug"?
10:43.00Aurs(the phones fetch config from ftp)
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10:45.46hads|homeAurs: I've never congigured a polycom but I believe you are correct, that line means the volumes will keep the settings the user chooses across calls, and possibly reboots.
10:47.10op3rwhats the command to the CLI to show you the realtime status?
10:48.09*** join/#asterisk oej (n=olle@82.148.165.13)
10:48.27Aursguess the only way to find out is to try it
10:48.55P-NuThooray for the SPA3000, I have it working! zero echo and great quality. Now I'm going to burn my x100p.
10:49.13bugzis there a way to have a user push a key combo on the phone to initiate a call recording??
10:49.29bugzi could swear i read this somewhere not too long ago
10:50.51*** join/#asterisk denon (i=denon@synapse.subneural.net)
10:50.51*** mode/#asterisk [+o denon] by ChanServ
10:51.12hads|homebugz: Check out automon in features.conf
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10:52.30bugzthanks
10:52.33FuriousGeorgeare there any good softphones that work with windows ce?
10:52.38hads|homenp
10:52.56FuriousGeorgei believe eyebeam has a ce version, doesnt it?
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10:59.32P-NuTHey guys can someone help me with inbound SIP calls?
11:01.06fourcheezeP-NuT: don't ask to ask just ask
11:07.52*** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net)
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11:09.49QuAtRo[NL]Currently i have some problems with my queues
11:09.54QuAtRo[NL]My strategy is ringall
11:10.10QuAtRo[NL]But it seem that Asterisk only rings the least recent
11:11.38bugzqueues are hard to manage after they get big
11:11.47bugzsecurity is a big issue
11:11.54*** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net)
11:12.09bugzlock down your phones, make sure assholes arent able to put their phone on DND or otherwise modify the behavior of the phone
11:12.14bugzlike being able to forward the phone
11:12.48bugzwe had to rebuild an entire dial plan for a gigantic call center
11:13.02bugzwe basically did away with the queue all together and came up with something different
11:14.03*** join/#asterisk Zeeek (n=randulo@pdpc/supporter/active/Zeeek)
11:14.15Zeeekba da boom
11:15.17QuAtRo[NL]bugz: But the phones are _not_ on dnd
11:15.24QuAtRo[NL]So it seem to be a bug in Asterisk
11:20.36QuAtRo[NL]bugz: Or am I wrong?
11:21.01_problem_anybody can tell me what is the significance of local context in extensions.conf..and if i omit that then will there be any problem??
11:21.51bugzQuAtRo[NL]: no idea, ive abandoned all hope for queues at this time
11:23.16Zeeek_problem_: look at the docs on dialplans at http://asteriskdocs.org for the answer to this and many other questions
11:26.31*** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net)
11:26.37_problem_Zeeek: ok thanks
11:27.10Zeeeknp
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11:27.35Zeeekheres an even better overview: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650
11:28.09Zeeeknot anywhere near here I hope :)
11:28.34*** join/#asterisk viking78 (n=aherbert@66-168-98-144.static.jcsn.tn.charter.com)
11:28.48RoyKZeeek: yeah
11:29.01_problem_<PROTECTED>
11:29.01_problem_<PROTECTED>
11:29.01_problem_<PROTECTED>
11:29.39_problem_Zeeek: i m getting these errors in cli dont know why it is happenning..the docs reference u gave me i already saw them before
11:30.01Zeeekare you on @home or AMP or something?
11:30.20_problem_asterisk-1.2.4
11:30.40Zeeekmore info is needed. Pastebin your extension
11:30.45Zeeekb
11:30.48RoyK~pb
11:30.51jboti heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
11:30.55QuAtRo[NL]Is there a value where is determined after how many time a agent is able to handle the next call ?
11:30.59RoyK~lart Zeeek
11:34.20QuAtRo[NL]FOund, wrapuptime
11:36.57lilalinuxis festival 1.4.3 (shipped with debian) still ok, or should I install 1.9.5 manually?
11:37.52_problem_Zeeek: http://pastebin.ca/73042
11:38.36Zeeekdidn't I say paste your extension?
11:38.51Zeeekthe one that begins exten => ....
11:38.56op3rQuAtRo[NL]: are you using agentcallbacklogin?
11:39.15Zeeek_problem_:  paste the extension and the sip.conf peer entry that it uses
11:40.17_problem_Zeeek:ok
11:44.52_problem_Zeeek: http://pastebin.ca/73050
11:47.32Zeeek_problem_: is the phone registered with asterisk?
11:48.02*** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net)
11:48.04Bert-hello there
11:48.13Bert-I've a little problem with music on hold
11:48.24_problem_Zeeek: its a softphone when it goes offline it happens like that..but when it is online then no problem happens
11:49.11Bert-All seems to be fine, excepted a notice complaining about that :"nmp3thread: Request to schedule in the past?!?!". But people placed on hold hear nothing at all
11:49.32_problem_Zeeek: the asterisk cli logs which i posted there are from offline scenario
11:49.33Zeeek_problem_:  when it's offline there's no one there
11:49.46Zeeekyou want it to go to voicemail?
11:50.11_problem_Zeeek: yes i want it to go to voicemail.
11:50.30Zeeektry looking up and using DIALSTATUS and see if that works better
11:50.37_problem_Zeeek: without those bad logs
11:51.16_problem_Zeeek: ok, but what dialstatus do? I never heard of that
11:51.23Zeeeklook it up
11:51.43Zeeekit's important to know what it does and too long to go into on an IRC channel
11:51.44_problem_Zeeek: ok thnx
11:51.56blitz[laptop]${DIALSTATUS} is teh coolest
11:52.00Zeeekit is a variable - find it on wiki or in docs
11:52.00_problem_Zeeek: ok fine
11:52.12Zeeekblitz how's london?
11:52.35Zeeekor how was it?
11:52.59blitz[laptop]Zeeek: London is going very well!  I'm currently in the exhibition hall programming :)
11:53.08Zeeekdon't do that!
11:54.00ZeeekI'm copying and pasting a lot of Japanese text atm
11:55.40Zeeekbrb
11:55.42*** part/#asterisk Zeeek (n=randulo@pdpc/supporter/active/Zeeek)
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11:56.30SplasPoodMorning all, question.. can Voicemail() be setup so rather than hanging up on the caller after they leave the message and hit #, it instead continues on with the dialplan?
11:56.50Zeeek_ah that's better
11:58.07Zeeek_SplasPood doesn't show application voicemail tell you that?
11:59.07SplasPoodZeeek_: Not sure if it does, but I just did a quick test and answered my question
11:59.22SplasPoodfor some reason I was convinced that Voicemail() was doing its own hangup
11:59.33Zeeek_heh best way, sometimes! I think there's a key that allows that yeah, either star or pound
11:59.55SplasPoodwell if you hit # it says "Goodbye" ..  but if there's more dialplan, it continues.. that was news to me
12:00.15Zeeek_isn't there a flag that allows it as well? Can't remember
12:02.06QuAtRo[NL]Damn, ringall seem to just don't work in Asterisk
12:02.19QuAtRo[NL]It always uses leastrecent
12:02.52QuAtRo[NL]Which makes Asterisk unusable for a lot of helpdesks
12:04.53SplasPoodQuAtRo[NL]: hrm, ringall works for me
12:05.35QuAtRo[NL]SplasPood: When i call to the helpdesk (which is a queue) it rings two phones...
12:05.52QuAtRo[NL]I pick up phone 1
12:06.14QuAtRo[NL]Then I hang up and call (5 seconds later) again
12:06.23QuAtRo[NL]And only phone 2 rings
12:06.46SplasPoodmaybe its wrapup time
12:06.50SplasPoodin the queue config
12:06.55QuAtRo[NL]My wrapuptime is 10
12:07.11QuAtRo[NL]And wrapuptime is miliseconds
12:07.18SplasPoodso if you called 5 seconds later phone 1 would still be in wrapup...
12:07.23SplasPoodis it?  I'm pretty sure it's seconds
12:07.45lilalinuxthe voicemailbox has to apps: VoiceMail and VoiceMailMain,   Is it possible to get into VoiceMailMain from within VoiceMail?
12:08.10SplasPoodlilalinux: yes, during the greeting one can hit...  * or #, (i forget) and it'll prompt them to login to that mailbox
12:08.24lilalinuxSplasPood: thx
12:08.30SplasPoodlilalinux: you'll need a exten 'a' and stuff..  check out Voicemail() on voip-info.org
12:09.02QuAtRo[NL]SplasPood: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+agents.conf
12:09.19SplasPoodQuAtRo[NL]: thats agents.conf..
12:09.21*** join/#asterisk jpeeler (n=jpeeler4@host81-149-2-72.in-addr.btopenworld.com)
12:09.43QuAtRo[NL]Ah...
12:09.47QuAtRo[NL]Let me have a look
12:09.48SplasPoodQuAtRo[NL]: there's a wrapup setting in queues.conf as well
12:10.38QuAtRo[NL]I saw ;)
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12:13.04Tilihey where do i set zap to show cli as international to switch
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12:15.46*** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no)
12:16.20lilalinuxHow can I get the originally called number (meanwhile beeing in a different exten)
12:17.29SplasPoodlilalinux: I *believe* one of the variables has it... do a search for variables on voip-info.org
12:17.33RoyKlilalinux: i don't think you can. just Set(ORIGINALNUMBER=${EXTEN})
12:17.43SplasPoodor, what RoyK said
12:17.54bugzJun 27 07:16:39 NOTICE[5597]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)
12:17.56Zeeek_or pass it in a macro
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12:18.03olor1nhello
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12:18.55lilalinuxthx
12:19.14QuAtRo[NL]SplasPood: Thank you! It works fine :)
12:20.14SplasPoodQuAtRo[NL]: No problem :)  Queues pissed me the hell off when i first tried to set them up..   Another thing i found is that if you use roundrobin, but want it to continue if the agent doesn't answer (but is available) you need to have timeoutrestart=yes in queues.conf
12:20.50QuAtRo[NL]SplasPood: Do you have any idea what makes my phones log off after a certain time=
12:21.20RoyKdon't listen to Zeeek_
12:21.27Zeeek_they get tired of hearing the same shit?
12:21.42SplasPoodQuAtRo[NL]: there's an option in agents.conf that logs agents out if they don't answer...
12:22.04toppinganyone good with RealTime?  I've got the tables, login and ODBC set up, but i don't see res_odbc loading in the debug output
12:22.21SplasPoodload res_odbc.so ?
12:22.38op3rdoes anyone know this error? Jun 27 20:22:09 WARNING[23303]: app.c:644 ast_play_and_record_full: No audio available on IAX2/u26450
12:22.43jpeeleranybody know why i would get "Primary D-Channel on span 1 down" when the lines are up?
12:23.10toppingSplasPood: maybe I don't have it compiled in?
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12:23.15SplasPoodits a module
12:23.16RoyK[no]jpeeler: could be akk sirts if reasons
12:23.23SplasPoodif you don't autoload in modules.conf
12:23.31SplasPoodyou'd need to explicitly load it
12:23.38SplasPoodwhat happens if you do load res_odbc.so
12:23.39toppingi just don't see it in usr/lib/asterisk
12:23.44SplasPoodoh
12:23.59RoyK[no]topping: then perhaps you don't have the odbc libs installed. then it'll never be built
12:24.00jpeelerRoyK[no], yeah so where to start?
12:24.01*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
12:24.07SplasPoodwell like RoyK, I don't use it.. but maybe you l...  what RoyK said
12:24.19toppingahhh ok lemme try to rebuild now that i have odbc installed thx
12:25.21op3rdoes anyone know this error? Jun 27 20:22:09 WARNING[23303]: app.c:644 ast_play_and_record_full: No audio available on IAX2/u26450?
12:25.41SplasPoodanyone know if the privacy option to dial can be configure to not offer the 'Torture' option?
12:26.12SplasPoodop3r: I'd assume if one of the 5 people here right now knew the answer someone would have responded by now.
12:26.51SplasPoodop3r: might want more info..  what was going on when you got that error, etc..
12:26.56op3rok hehehheeh
12:27.12op3rSplasPood: thats when Im trying to call an inbound DID number
12:27.21op3rand redirected it to the voicemail
12:27.22op3r:(
12:27.46SplasPoodand you get that error when it.....?
12:28.31NotJohnDavidthe asterisk-user list is busier than i thought it'd be
12:28.59RoyK[no]ops. customs... getting closer to the .se border
12:29.09RoyK[no]they're going through my bag...
12:29.14QuAtRo[NL]SplasPood: They do not only log off from the queue... When you call one of those phones Asterisk says the phone is ringing
12:29.19QuAtRo[NL]But it isn't
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12:29.47SplasPoodhrm...  network issue maybe?  some type of NAT/firewall interference?
12:30.07jpeelercan anybody tell me what to start looking at with all the primary d-channel spans down?
12:30.15QuAtRo[NL]The strange thing is, it happends after a certain time...
12:30.19jpeelerit's not the card or the lines
12:30.43bugzchrist
12:30.53bugzis there a way to remotely troubleshoot a te110p
12:31.29olor1nI try to make something with asterisk but i am not quite sure it can be done so please comment: the idea is when a call is received to put the caller in an interactive menu (choice 1,2 etc.) and in the same time to keep anouncing on the called party telephone that there is an incoming event. Is it possible
12:31.56QuAtRo[NL]SplasPood: And when i use the 'restart' function of my phone.. It all works fine again.. Untill a certain time
12:32.53SplasPoodQuAtRo[NL]: always the same time?
12:33.18QuAtRo[NL]Didn't measure it..
12:34.55SplasPoodroughly?
12:34.59SplasPoodI'd do some packet sniffing
12:35.02SplasPoodand see what was up
12:35.10SplasPoodis there a firewall/NAT involved here anywhere?
12:35.16QuAtRo[NL]Yes, there is
12:35.23SplasPoodthat'd be my first guess
12:35.31QuAtRo[NL]The PBX is in the datacenter and we are on a ADSL line
12:35.56QuAtRo[NL]Behind a Eminent router
12:36.12SplasPoodheh a Pre-Eminent router
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12:39.17toppingSplasPood: OK, I can do DB queries from CLI now, but I don't seem to be seeing the extensions in the dialplan yet, any ideas?
12:39.49SplasPoodtopping: I dunno res_odbc, when you say 'from CLI' what do you mean?
12:40.07topping'realtime load extensions exten 411' gives back the correct data
12:40.11SplasPoodahh
12:40.11toppingfrom the database
12:40.30QuAtRo[NL]SplasPood: I guess it is my NAT keepalive function of my phone
12:40.31toppingbut when i dial that extension, i get a reorder
12:40.36SplasPoodyou have the switch statements and what not in extensions.conf
12:40.51Cresl1nblitzrage: !!!!
12:40.54Cresl1nblitz[laptop]: !!!!!
12:40.54toppingdon't think so, i'll rtfm, thanks!
12:41.04blitz[laptop]Cresl1n: !!!!!!!!
12:41.15*** join/#asterisk TeePOG (n=Ender@dsl-145-132-214.telkomadsl.co.za)
12:41.16blitz[laptop]Cresl1n: back to work :)
12:41.22SplasPoodtopping: Warning, the way asterisk realtime extensions work is highly lame IMHO
12:41.26TeePOGafternoon all
12:41.37Cresl1nafternoon
12:41.48toppingSplasPood: yah, i sensed that, but isn't it still the best way to get to a database?
12:41.57blitz[laptop]Cresl1n: did it die again?
12:41.58SplasPoodI suppose
12:42.00SplasPoodwithout a reload..
12:42.11blitz[laptop]Cresl1n: who's laptop are you using now?
12:42.11*** join/#asterisk acrg (n=aragon@decoder.geek.sh)
12:42.14acrghi
12:42.19acrghaving an odd problem
12:42.23Cresl1nblitz[laptop]: when I jump up and down it does
12:42.30acrgif a caller puts someone on hold
12:42.34acrgand then takes the call back
12:42.44acrgthe other party can not hear them
12:42.52acrgbut he can hear the other party
12:42.58blitz[laptop]Cresl1n: for real?
12:43.04Cresl1nyeah
12:43.06Cresl1nsux0rs
12:43.11acrganyone know what's up ?
12:43.17blitz[laptop]Cresl1n: wow... could the CPU or something be a bit loose?
12:43.23blitz[laptop]or a cable?
12:43.32TeePOGhi guys, do you need any special hardware to run asterisk on a *nix box? I want to use asterisk as a voip-to-phone gateway
12:43.55SplasPoodTeePOG: you wanna use a normal phone line?
12:44.27SplasPooda POTS line..
12:44.36TeePOGyes, but i suppose it's going to be a  bitch getting an internal pci 56k modem to work under linux?
12:44.40Bert-hmm what is the 'cleaner solution', resolving the 'mpg123issue' plz ???
12:44.47Bert-I'm unable to find it on voip-info
12:44.53Bert-~mpg123
12:45.06jbotmethinks mpg123 is Real time MPEG Audio Player for Layer 1,2 and Layer3. URL: http://www.mpg123.de/. ONLY MPG123-R  will work with asterisk. PERIOD. use 'make mpg123' in the asterisk source dir
12:45.08SplasPoodTeePOG: well...  technically the low end digium cards were (are?) modems..  but a specific kind
12:45.08Cresl1nwhy can't I build trunk chan_zap
12:45.08Cresl1nsomebody hit me
12:45.24SplasPoodTeePOG: So you'll either need an external FXO gateway, or you'll need one of the analog line digium cards
12:45.25TeePOGok SplasPood -- so I need a digium card? I thought those were only for ISDN lines
12:45.31TeePOG'ok
12:45.39Bert-hmm what is the 'cleaner solution', resolving the 'mpg123issue' plz ???
12:45.49*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
12:46.13*** join/#asterisk Chotaire (i=chotaire@chotaire.net)
12:46.34NotJohnDavidteepog: the x100p works just fine under linux.  that'll connect a POTS line to the asterisk server then you need a way to connect handsets to asterisk (if you so choose)
12:46.55Bert-grr disconnected again and again :(
12:47.35acrgbert do you mean the leftover mpg123 processes running at 100% cpu?
12:48.25Bert-no
12:48.30Bert-I mean
12:48.39*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
12:48.43Bert-as I can read on website, there is no need to use mpg123 anymore
12:48.45TeePOGwell, we're like 3 shops under the same management, next to each other. would it be better to let their *doze SIP clients connect to my asterisk box as extensions?
12:48.54Bert-asterisk 1.2 and higher provide a new solution
12:49.03Bert-but there is nothing about that on the website :(
12:49.34blitz[laptop]ouch!
12:49.54Bert-"Asterisk 1.2 has solved the "mpg123 issue" and comes with a cleaner solution"
12:50.08Bert-then I'm looking on this cleaner solution
12:50.13acrgBert yes, it can read mp3s natively now with the format_mp3 module
12:50.28Bert-does it works with MusicOnHold ?
12:50.31acrgit is in the asterisk-addons package
12:50.32acrgyes
12:50.43blitz[laptop]although converting your mp3's to ulaw/alaw/gsm/g729/etc... and just using the native format can take a lot of load off the CPU
12:51.02Bert-yep for sure
12:51.14Bert-for now I only wnat to have musiconhold working
12:51.24Bert-whatever the soulution used:)
12:51.37NotJohnDavidwhat can you use to play gsm?  and convert mpg->gsm ?
12:51.48Bert-It is just a test, as I have to make a demo to my boss
12:51.56jpeelerBert-, I followed blitz's advice about converting and got my music on hold working
12:51.57*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
12:51.57*** mode/#asterisk [+o anthm] by ChanServ
12:52.13Bert-you convert it with lame ?? sox ??
12:52.14blitz[laptop]jpeeler: w00t!!!
12:52.17blitz[laptop]sox
12:52.21jpeelercorrect sox
12:52.25Bert-ok let me try
12:52.28Bert-thx guys :)
12:52.47jpeelerblitz[laptop], yeah thanks a lot
12:52.49NotJohnDavidsox is like a swiss army knife
12:53.09blitz[laptop]jpeeler: glad it worked for you
12:53.24mutyea, it's useful for a minute or two until it falls apart
12:53.27jpeelerblitz[laptop], did you see any of my earlier problems today?
12:53.43blitz[laptop]jpeeler: sorry, I did not -- been running a conference :)
12:54.00toppingSplasPood: I think this is the last question... everything is almost working, but I get "macro_exec: No such context 'macro-tenant,1415xxxxxxx' for macro 'tenant,1415xxxxxxx'"
12:54.14jpeelerblitz[laptop], I'm kind of at a loss here. I keep getting these errors: Primary D-Channel on span 1 down
12:54.21toppingmy database is populated with calls to 'Macro'
12:54.23jpeelerblitz[laptop], the lines are up and the card is good
12:54.29toppingcan i do that?
12:54.48QuAtRo[NL]SplasPood: It was the so called: 'NAT keepalive' setting in my phone ;)
12:55.02blitz[laptop]topping: how are you calling the macro app?  It thinks the macro name is 'tenant,1415xxxxxxxx'
12:55.11*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
12:55.19toppinglemme pastebin it
12:55.25blitz[laptop]and obviously you are trying to send Macro an argument
12:55.57toppingyes
12:56.19blitz[laptop]jpeeler: hrmmmm... I'm not really a T1/E1 pro... sorry
12:56.25*** part/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com)
12:56.25*** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com)
12:57.46Drukenmorning peoples
12:58.06toppinghttp://rafb.net/paste/results/cQgsUK62.html
12:58.29toppingthe first half is what i have in extensions.conf, the second half is a record from the db
12:58.41jpeelerblitz[laptop], well thanks anyway. it's hard to get help with problems i don't even know where to start
12:59.35toppingi have a macro in the extensions.conf that i should have listed... http://rafb.net/paste/results/dtH4Jq29.html
12:59.51blitz[laptop]jpeeler: well... the D-channel is used to provide signaling between you and the other end (CO, Channel Bank, etc...) and it sounds like something is happening that is causing the D-channel to drop
13:00.00blitz[laptop]jpeeler: how often does it happen?
13:00.35[TK]D-Fenderblitz[laptop] : ! ! !
13:00.57jpeelerblitz[laptop], when something with the lines change
13:01.13jpeelerblitz[laptop], like when i reboot or unplug a line
13:01.41blitz[laptop][TK]D-Fender: I don't want to meet your mom!
13:02.08[TK]D-FenderUp in Mississauga now.  we should schedule
13:02.13lilalinuxhow do I configure asterisk for _incoming_ calls from sipgate?
13:02.28blitz[laptop][TK]D-Fender: yes!  I will be returning tomorrow from London actually
13:02.38toppingblitz[laptop]: do you think this could be because the macro is not in the database with the calls to the macro?
13:02.41[TK]D-Fenderblitz[laptop] : ON or UK?
13:02.59blitz[laptop][TK]D-Fender: From London, UK to Mississauga, ON
13:03.37blitz[laptop]topping: not too sure -- I don't use realtime, but somehow Asterisk isn't parsing the comma
13:03.39[TK]D-Fenderblitz[laptop] : And you got more than just a lousy T-shirt, right? ;)
13:03.51toppingahh, ic
13:03.59blitz[laptop][TK]D-Fender: yah... I bought a bunch of chocolate I hope to give to a hot asian girl I went on a date with once, lol
13:04.02Drukenthere's a mississauga UK ?
13:04.12blitz[laptop]Druken: don't think so :)
13:04.17blitz[laptop]ok .. I'm going back to programming -- peas
13:04.40[TK]D-FenderDruken : Not to my knowledge
13:04.42toppingthanks for the help blitz[laptop]
13:04.48Druken:)
13:04.53Drukeni was gonna say :)
13:06.44Drukendo i want to know why blitz[laptop] would be meating [TK]D-Fender's mom ?
13:06.59*** join/#asterisk Modcuts (n=bob@lan.proporta.com)
13:09.22acrgHaving an odd problem: if a caller puts someone on hold and then takes the call back, the other party can not hear them.  But he can hear the other party.
13:09.39kay2[TK]D-Fender: do you know why I get that:  NOTICE[20945]: chan_iax2.c:5068 register_verify: Peer 'key2' is not dynamic (from 62.39.9.251)
13:09.50toppingSplasPood blitz[laptop]: turns out if I make the database appdata field have a '|' instead of a comma that it all works.  Looks like there's some problem with parsing the comma, as blitz[laptop] pointed out.  Everything works now, THANKS A BUNCH! :-D
13:11.20blitz[laptop]topping: oh yah!!  I have to remember that because Asterisk will definately parse on the pipe, but for some reason it can't always parse on the comma because in the dialplan it actually converts the comma to a pipe
13:11.55toppingcool stuff!
13:11.56toppinghehe
13:12.15*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.21.Dial1.SanJose1.Level3.net)
13:12.18toppingit was easy for me to change because I am reading the data through a view
13:12.25toppingso i just changed the output of the view
13:12.38toppingpostgresql rocks
13:13.04*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.21.Dial1.SanJose1.Level3.net)
13:13.06kay2anyone knows why I get NOTICE[20945]: "chan_iax2.c:5068 register_verify: Peer 'key2' is not dynamic (from 62.39.9.251)" if in iax.conf I put host=62.39.9.251
13:13.21toppingi have these sentex call boxes outside my apartment building, and people are constantly asking me to reprogram them
13:13.35toppingnow i can put their numbers in a database and let them update the door boxes via the web
13:13.57*** join/#asterisk ToyMan (n=stuq@74-32-78-58.dsl1.mdl.ny.frontiernet.net)
13:15.38*** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
13:18.24Bert-hmm
13:18.26Bert-doesn't work
13:18.39Bert-I use the default musiconhold context
13:18.56*** join/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com)
13:19.33benjamin7062Any of you guys using SCCP and noticing Asterisk 1.2.9.1 crashes if the phone terminates in a queue?
13:19.39Bert-I set it like that :
13:19.40Bert-[default]
13:19.40Bert-mode => quietmp3
13:19.40Bert-directory => /var/lib/asterisk/mohmp3/planetvoip
13:19.40Bert-;application=/usr/bin/sox
13:19.44benjamin7062SCCP phone I mean?
13:19.52*** join/#asterisk BertZ (n=bert@bas33-1-82-66-4-198.fbx.proxad.net)
13:19.54BertZgrr
13:20.15Druken~pastebin
13:20.18jbothmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/
13:21.06benjamin7062~fart
13:21.08jbotACTION farts, releasing large quantities of methane and sulfur dioxide. "Evacuate the channel! GO! *gag* SAVE YOURSELVES *cough* MOVE *choke* MOVE!"
13:23.02*** join/#asterisk creadurx (n=creadure@196.82-134-19.bkkb.no)
13:23.58Zeeek_jbot stop flooding the channel with inane remarks
13:23.59jbotACTION leaps to his feet and stops flooding the channel with inane remarks
13:24.05Zeeek_whew!
13:24.28creadurxinane is better than insane
13:25.08Zeeek_not for the person who is insane
13:25.09TeePOGinane is a boring type of insane
13:25.11Drukenuhg.... god damn light
13:25.23Zeeek_insane is a dangerous form of inanity
13:25.30Drukenif not for light, i'd say fuck today and go back asleep
13:25.30TeePOGsnap
13:25.46Zeeek_sleeping masks with digium logo rock
13:25.47toppingDruken: are you on the west coast too?
13:25.57Drukenontario
13:26.11toppingoh so it's extra-bright there lol
13:26.21toppingpeople still have their headlights on here
13:26.22creadurxhow would i solve the problem of knowing which DID is responsible for a SIP phone ringing when using queue()? one SIP phone is a member in 2 different queues, each set up to two different DIDs
13:26.34Drukenno.. it's a god damn overcast miserable rainy day
13:26.43toppingheh
13:26.48Zeeek_insult to injury!
13:27.18*** join/#asterisk op3r (n=op3r@124.107.26.34)
13:27.31*** join/#asterisk wese103 (n=wschaffe@c069.centercall.com)
13:27.57Drukencreadurx: you want to know what did it came from or what queue?
13:28.06op3rdoes any one know how to remove the message that announce the the caller's queue's position?
13:28.06toppingok i'm going to try sleeping lol nite
13:28.23Drukenbastard!
13:28.44creadurxDruken: both :)
13:29.09creadurxim hacking the shit outta the manager interface and astmanproxy
13:29.16creadurxbut this problem stumbled me
13:29.24benjamin7062So * keeps crashing if I terminate an SCCP phone inside a queue.  With Verbose and Debug all the way up it doesn't spit anything unusual up since it's crashing.  Any way for me to get it to spit more info?
13:31.17*** join/#asterisk [pyro] (i=_pyro_@tor/regular/bracketed-pyro)
13:31.21op3rbenjamin7062: do you know how to remove the announcement of the caller's queue position?
13:31.22Drukencreadurx: well, asterisk can announce what queue the call came from, as for what did... no idea... i guess my question would be, what does it matter?
13:31.44Drukenop3r: did you actually read the queue.conf file?
13:31.51op3rDruken
13:31.55benjamin7062op3r, You mean for my problem?  Or do you have a seperate problem.  The config for that is queue.conf
13:31.59op3rbut it kept on telling the queue position?
13:32.16benjamin7062op3r, asterisk -rx'reload'
13:32.51*** join/#asterisk BertoX (n=bert@bas33-1-82-66-4-198.fbx.proxad.net)
13:32.53BertoX:(
13:32.55*** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net)
13:33.00BertoXbig internet issue from my side :(
13:33.18BertoXHow to see where musiconhold fails plz ?
13:33.21BertoXno log, no error
13:33.25BertoXbut no music ... :(
13:33.55benjamin7062Bertox -- Turn up verbose and debug... it will show you everything... from the console... set verbose 100 and set debug 100
13:34.27blitz[laptop]LOL
13:34.30creadurxDruken: well, I cant even phrase my own problem! i gotta do some more thinking.
13:34.33blitz[laptop]100?  nothing above 4 is useful
13:34.47*** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)
13:35.05BertoXI lol
13:35.07blitz[laptop]debugging information is also controlled in logger.conf
13:35.11benjamin7062Yes, I know... but if you say 100 then when they bump it some day you don't have to re-train.. =)
13:35.12BertoXset level 100 for both
13:35.15BertoXnothing more ...
13:35.38blitz[laptop]benjamin7062: if you need to retrain someone regarding how to use 'set' on the CLI they shouldn't be on the CLI in the first place ;)
13:35.54BertoXwhat exactly mean that please ?
13:35.55BertoXmonmp3thread: Request to schedule in the past?!?!
13:36.02BertoXmaybe my problem comes from here
13:36.04benjamin7062blitz[laptop], I can't argue with that... =)
13:36.12*** join/#asterisk Hmmhesays (i=negative@66.173.103.110)
13:36.21benjamin7062blitz[laptop], but 13 = 100 as far as logging.. =)
13:36.27Hmmhesaysyo ho
13:36.33blitz[laptop]and 4 == 100
13:36.50Zeeek_40 + 2 = ?
13:36.52blitz[laptop]or just use 5 which is an easy number to remember
13:36.55benjamin706245?
13:36.57blitz[laptop]Zeeek_: you nerd :)
13:37.08benjamin7062No, I got it... 46
13:37.17Druken69?
13:37.22benjamin7062Damn
13:37.24benjamin7062I was way off
13:37.34*** join/#asterisk Katty (n=aisaacs@64.82.232.54)
13:37.39kay2why do I get that msg ?  NOTICE[22371]: chan_iax2.c:6802 socket_read: Rejected connect attempt from 62.39.9.251, who was trying to reach '400@ddefault'
13:37.45benjamin7062BertoX, Does it show it 'trying' to play music?
13:37.46kay2why does it get Rejected
13:37.47kay2?
13:37.54Kattymorning
13:37.59BertoXno
13:38.02BertoXbenjamin7062 : no
13:38.16benjamin7062kay2, I have a little experience with rejection.. but it involves women.. not *
13:38.24Kattyhaha
13:38.28*** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net)
13:38.38kay2benjamin7062: how weird, I have the oposite
13:38.48Zeeek_key katty, long time no C++
13:39.11Kattyhey Zeeek (=
13:39.14benjamin7062kay2, Hopefully, you can fix your problem cause there is no code fix that will help me.  =)
13:39.45iDunnothe world is trying to take me over :)
13:40.10benjamin7062BertoX, you have a line in your dialplan to play music but never see it hit that line in the log at all?
13:40.20BertoXbenjamin7062 : just   -- Started music on hold, class 'default', on channel 'SIP/Nextone_OUT-06c9'
13:40.20BertoX<PROTECTED>
13:40.22jbalcombwhy is #perl is haven for unhelpful, underage folks?
13:40.31*** join/#asterisk Winkie (n=urmom@gateway.duclicsic.com)
13:40.42DrukenBertoX: do you have a zap interface?
13:40.48Zeeek_did you type #perljam by accident?
13:41.04*** join/#asterisk mog (i=ejabberd@68.62.237.103)
13:41.13benjamin7062BertoX, Hmm.. Do you have Zap...?
13:41.14KattyZeeek_: you framilier with mounting windows 2003 shares?
13:41.25Zeeek_Ewwwwwww
13:41.33benjamin7062Katty, if you do... don't write to them (if they are NTFS)
13:41.38kay2Katty; i am
13:41.43jbalcombZeeek_: haha.. it's seems almost likely.
13:41.47Kattybenjamin7062: i'll do whatever i please ;)
13:41.58KattyZeeek_: i'll take that as a no, heh.
13:41.59Zeeek_katty you've been warned!
13:42.04Drukentypical women....
13:42.14Kattytwisted[asteria]: you around?
13:42.16Zeeek_yes, ask for advice and then do what they please
13:42.18Kattytwisted[asteria]: i know you're helpful.
13:42.24KattyZeeek_: i wasn't asking for advice.
13:42.29KattyZeeek_: i'm going to do what i'm going to do.
13:42.29benjamin7062Katty, K...  Good luck...  I predict mass corruption...  I need some popcorn to watch
13:42.32Zeeek_no, true, you wouldn't
13:42.37KattyZeeek_: i /simply/ asked if if you'd done it.
13:42.51Kattyhow annoying, geeks being bitter.
13:42.57Kattydo lighten up guys.
13:43.04benjamin7062LOL
13:43.08Zeeek_"for sale one used encylopedia britanica. Reason: no longer needed, married, wife knows everything"
13:43.17Drukenhahaha
13:43.18Kattyexactly ;)
13:43.29Kattyariel_: you around?
13:43.29jbalcombKatty: We have no problemds with our NTFS mounts.
13:43.31Zeeek_I'm confirming it Katty, not denying :)
13:43.46KattyZeeek_: eddie izzard moment?
13:43.53KattyZeeek_: just keep confirming and denying
13:44.22Zeeek_Error 42 -unknown culture specific reference or allusion. Please contact your culture manager
13:44.46benjamin7062Heavy writing to NTFS works fine.  It's reading it from W2K3 again that becomes the problem over time
13:44.49Kattyjbalcomb: is it a 2003 share?
13:44.50Zeeek_wtf is eddie_izzard?
13:44.57KattyZeeek_: consult google, dear.
13:45.05Zeeek_naw
13:45.17Zeeek_well, ok, fer u
13:45.17DrukenKatty: i'm tired.... mind if i sleep with you? :) hehe
13:45.32Zeeek_fools rush in!
13:45.44benjamin7062haha
13:45.45jbalcombKatty: Well, actually I beleive they are 2000. I'm not familiar with there being a difference there though.
13:45.46KattyDruken: i think the significant other would mind.
13:45.50Zeeek_where asterisk-angels fear to tread
13:45.55Kattyjbalcomb: and yes, well there is..
13:45.58Drukenit's a possibility
13:46.01Kattyjbalcomb: thanks anyway (=
13:46.17Zeeek_but asterisk-angels never fall in love. They just go to bed
13:46.17*** join/#asterisk tdonahue (n=tdonahue@207.138.151.58)
13:46.19Kattyjbalcomb: i do appreciate you at least /attempting/ to offer help.
13:46.29Kattyjbalcomb: rather than just saying zomgdon'tdoit
13:46.37Kattyjbalcomb: so go you (=
13:46.37jbalcombKatty: ah, well, yes, thank you. Sorry we can't be of more help on this particular matter. Please consider our assistance with future quandries.
13:47.07Zeeek_We know that you have a choice in IRC channels. Thank you for coming to #asterisk
13:47.08benjamin7062Katty, you didn't say remotely mounting NTFS -- You said mounting.  There's a difference.
13:47.16Kattyoh for goodness sake.
13:47.20Kattyi said mount and windows 2003 share.
13:47.30creadurxDruken: this is why i need the DID.. each of our customers have their own DID that gets routed to either queue1 or queue2. the SIP phone that is a member of queue1 needs to know which DID was called so our inhouse app automatically opens the correct customer database
13:47.37Kattyif you're going to nit pick me to death, just go stand in a corner :P
13:47.52Zeeek_and if he's in a round building?
13:48.00benjamin7062Katty, that works fine...
13:48.10Kattyyes i KNOW it works fine
13:48.14Kattythat's not the question here
13:48.26Kattyit never was in the first place
13:48.32benjamin7062Katty, Have you actually asked a question yet or are you here to bitch?
13:48.36Zeeek_if anyone knows how to talk to the file windows server éà03 system in php I'd be interested
13:48.38benjamin7062women
13:48.39benjamin7062...
13:48.40Kattyhahaha
13:48.44Drukencreadurx: so your doing an answering service then ?
13:48.48Kattybenjamin7062: you're cute.
13:48.54Kattybenjamin7062: you ever read what people ask?
13:49.08Kattybenjamin7062: sounds like you're the bitchy one ;)
13:49.23*** join/#asterisk Ifaistos (n=stelios@dslcustomer169.vivodi.gr)
13:49.25Kattybenjamin7062: don't worry about, k? i'll get twisted[asteria] to help me. i know he knows what he's doing.
13:49.56creadurxDruken: no.. im doing db lookups based on the CID, and also trying to do the same with the DID. this is all client side, via the manager interface
13:50.11benjamin7062Katty, mkay, you do that.  I pitty him
13:50.22Kattybenjamin7062: you shouldn't pitty him.
13:50.32Kattybenjamin7062: he probably has more fun in a day with his job than you do all year ;)
13:51.20serczhmm i'm looking for a very cheap way to connect an analogue phone to an asterisk box- any ideas?
13:51.21benjamin7062Katty, But that's simply not true, there are people like you in the world that make life complete
13:51.34benjamin7062sercz, $60.00 cheap?
13:51.36Kattybenjamin7062: glad to be of service.
13:51.51serczless than 60$ benjamin7062 :)
13:52.04Kattybenjamin7062: despite what you think, i'm not a bitch ;)
13:52.05Zeeek_sercz cheap IAX chinese ATA
13:52.05Drukenmy job isn't fun.... granted, i only work about 3-4 hours a day...
13:52.07serczmy phone was 40$
13:52.12Kattybenjamin7062: i just refuse to deal with idiots that don't listen.
13:52.29Hmmhesaysheh
13:52.32Hmmhesaysthat's all I deal with
13:52.33Kattybenjamin7062: or try to tell me something can't be done, when i know full well that it can be...
13:52.33Zeeek_Katty I resemble that remark!
13:52.51*** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com)
13:52.55lilalinuxdoes kphone support md5 secrets?
13:53.04Zeeek_Druken you should dstop being a streetwalker and get a normal job
13:53.09kay2When does a  iax2 call get rejected for the reason "No Authority Found" ?
13:53.55benjamin7062Katty, sure.. k..
13:54.05DrukenZeeek_, i wish i had the looks to be a street walker :)
13:54.46*** join/#asterisk Zaw (i=zaw@unaffiliated/zaw)
13:55.48*** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg)
13:56.24littleballhello,who can recommend a good sip soft phone (free one:-))
13:56.27Drukenok... well i guess i need to get ready to go... that 15 mins of work needs to be done sooner or later :)
13:56.34op3rlittleball: xlite?
13:56.57littleballi have tried two. xlite and firefly
13:57.00benjamin7062littleball, I like twinkle if you are running unix
13:57.01littleballboth work
13:57.25op3rthen choose q
13:57.25littleballbenjamin7062, i already run sip service now. :-)
13:57.29op3r1 rather
13:57.33littleballq?
13:57.35littleballwhat is it?
13:57.52kay2mog: so why shouldnt it work
13:58.36benjamin7062littleball, sweet.
13:58.57littleballfirefly is goood becasue i got g729 dll file
13:59.19Hmmhesayscare to share?
14:00.00littleballfrom where i can download sip phone q? i try to collect all free soft sip phones and recommend to my customers
14:00.10littleballwhy not? Hmmhesays
14:00.15moglol
14:00.31ManxPowerOddly, I find tariffs interesting
14:00.49littleballhttp://www.mobmeee.com/portal/index.html?ctrl:cmd=render&ctrl:window=default.Forums.ForumsPortletWindow&op=showForum&f=10659
14:01.06littleballhi, from where to download sip phone Q?
14:01.36Hmmhesaysso I got my 32 channel mixer yesterday, I have no idea what to do with it
14:02.07Hmmhesayslots and lots of settings
14:02.07fourcheezewhat kind of mixer?
14:02.07mutjust download all kind of animal sounds and mix em
14:02.09Hmmhesaysbehringer 32 channel
14:02.15mutcall it the rain forest in new york
14:02.23muthorns honking
14:02.25muthun shots
14:02.27mutscreams
14:02.40mutbirds chirping and a nice waterfall
14:02.59mutgun shots too
14:03.01fourcheezeHmmhesays: you got 32 sound sources?
14:03.01muto_O
14:03.10Hmmhesaysno, only about 20
14:03.14fourcheezecool
14:03.16fourcheezewhat sort?
14:03.42*** join/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net)
14:04.07fourcheezeoccasionally playing keyboard from the back of the room as well
14:04.19Hmmhesays2 guitar, 1 bass, 3 voice mic's  and about 10 on the drum set
14:04.24fourcheezenice
14:04.25Hmmhesaysso I guess thats about 16
14:04.31Hmmhesaysbut.. the price was right
14:04.39fourcheezeyou always need more
14:04.49mutMORE COWBELL!
14:04.55Hmmhesaysyeah, room for expansion there,  32 channels and 4 sub groups for monitors
14:05.05fourcheezewell you might want things for effects to go in etc
14:05.06mutI NEED MORE COWBELL!
14:05.21Hmmhesayshttp://www.musiciansfriend.com/product/Behringer-Eurodesk-SL3242FXPRO-Mixer?sku=631246
14:05.27fourcheezeyeah
14:05.29Hmmhesaysits got 20 or so build in effects
14:05.33fourcheezecool
14:05.36fourcheezehow much are those
14:05.59fourcheezewow $600
14:06.00*** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
14:06.18*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
14:06.44fourcheezeHmmhesays: well the 4 on the right are just groups anyway
14:06.46benjamin7062I need to buy 50+ Polycom 601's... USA... Vendor suggestions before I just 'pick' on
14:06.48benjamin7062one
14:07.01Hmmhesaysyeah
14:07.12Hmmhesayswhat I can't figure out is where we put monitors on this thing
14:07.23Hmmhesaysthe old board had monitor outs
14:07.29Hmmhesayslabeled "MONITOR OUT"
14:07.39fourcheezegenerally there's a prefade / postfade out
14:07.57HmmhesaysI'm pretty new to the whole PA thing
14:08.12fourcheezeor you could use a group out
14:08.17Hmmhesayscan  you explain to me whats going to sound like a very n00b question?
14:08.24fourcheezehehe
14:08.55Hmmhesaysok, the 1/4 in's on this say they can use balanced or unbalanced... I googled that a lot and got a whole lot of nothing
14:09.21fourcheezehmm
14:09.22fourcheezeok
14:09.26[TK]D-Fenderbenjamin7062 : 601's?  All heavy volume users, or just a high budget?
14:09.26Zeeek_like Canon connectors
14:09.39HmmhesaysI know all my 1/4 instrument cables are un balanced
14:09.40fourcheezegenerally you would use the Canon XLRs for balanced
14:09.44fourcheezethose big 3 pin things
14:09.49Hmmhesaysyeah...
14:09.57fourcheezeI'm guessing that there may be an option of using what would be a stereo 1/4 jack
14:10.40Hmmhesayslook at the top of that board.... http://img3.musiciansfriend.com/dbase/pics/products/4/9/7/282497.jpg
14:10.51fourcheezeok
14:10.54ManxPowerDoes anyone here know where I can find a list of CLECs that have service in a specific CO (CLLI)?
14:10.54Hmmhesaysyeah there is.. that is what says "balanced or un balanced"
14:11.05SpaceBassHey Hmmhesays long time no see
14:11.09Hmmhesayshey SpaceBass
14:11.26SpaceBassplaying a lot these days?
14:11.42Hmmhesaysyeah.. just about done getting the rest of the pa stuff
14:11.55SpaceBassfun
14:11.56fourcheezeHmmhesays: balanced shold be those 3 pin things
14:11.58Hmmhesaysis there an advantage to using either?
14:12.04fourcheezeyeah, use balanced
14:12.07Hmmhesaysso there is no balanced 1/4 cables?
14:12.10kay2Does anyone here know why asterisk doesnt use my file extension.conf when it loads ? is there any way from the CLI to see what config file it loaded ?
14:12.11*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)
14:12.31fourcheezeHmmhesays: if your mic or whatever terminates in an ordinary 1/4" jack it's unbalanced
14:12.36*** join/#asterisk SHad|Work (n=kvirc@84.255.228.2)
14:12.39fourcheezebalanced needs 3 connections
14:12.43SHad|Workhi
14:12.56Hmmhesayskay2 because extension.conf is not right?
14:13.00fourcheezeand it basically eliminates interference down the line
14:13.11Hmmhesaysok
14:13.20Hmmhesayscause there is a ground
14:13.21SHad|WorkDoes anyone have any experience with configuring an octoBRI and quad GSM cards from Junghanns in the same machine?
14:13.28Hmmhesaysso if I can use balanced  then I should
14:13.30SHad|WorkI can't figure out the zaptel/zapata configs
14:13.36fourcheeze1/4" jack is signal + ground
14:13.47Dr-Linuxhi all
14:13.49fourcheezebalanced XLR is -signal, ground and +signal
14:13.53anthmsomeone msg florin2703 with an example of a basic dialplan ivr
14:14.05Dr-Linuxhi all
14:14.07kay2Hmmhesays: what u mean it's not rght ?
14:14.14Dr-Linuxanyone? i need some info about , that how can we handle dialplan using mysql database?
14:14.32Vorondilheh, i did that too once.  global variables aren't set when they're in the "global" context and not the "globals" one.
14:14.33fourcheezeHmmhesays: so basically what you need is a connector that goes into the end of your mic that terminates in a canon XLR
14:14.37Hmmhesaysaccording to the behringer manual here, there are balanced 1/4's
14:14.44SHad|WorkDr-Linux: compile asterisk-addons and then use the MYSQL command
14:14.46Hmmhesayskay2: extensions.conf
14:14.50fourcheezeHmmhesays: if they are then they must have 3 connections
14:15.01Hmmhesaysand they do
14:15.09fourcheezeI don't see why anyone would use that rather than an XLR
14:15.19fourcheezecanon-style
14:15.26fourcheezebut maybe I'm too trad or something
14:15.29Dr-LinuxSHad|Work: asterisk-addones are already compiled and i'm using Mysql DB for CDR reports ..
14:15.45SHad|WorkDr-Linux: what exactly would you like to do?
14:15.57Dr-LinuxSHad|Work: but i'm not sure how can i handle dialplan using mysql db..
14:16.03KattyHmmhesays: hey hun (=
14:16.08Hmmhesayshey Katty
14:16.10fourcheezeHmmhesays: as far as monitors go
14:16.11Dr-LinuxSHad|Work: what things i need to do?
14:16.20fourcheezeI would use group 3/4
14:16.22SHad|WorkDr-Linux: what exactly do you mean by "handle"?
14:16.32Dr-LinuxSHad|Work: only i need mysql() in dialplan ..
14:16.34Hmmhesaysfourcheeze why is that?
14:16.46Dr-LinuxSHad|Work: wait dude, lemme confirm it from my manager.
14:16.52fourcheezeHmmhesays: because it's there and you can fade them down together with main if required
14:16.57fourcheezeusing 1 hand :-)
14:17.06Hmmhesayshmm ok
14:17.09fourcheezeHmmhesays: what do you want monitor to do?
14:17.24Hmmhesayssend me sound so I can hear what i'm playing
14:17.46Hmmhesaysi want my guitar up above everyone else
14:17.49*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
14:17.49Hmmhesaysin the monitor
14:18.03fourcheezedon't you have an amp for that?
14:18.12fourcheezedoes it have anything called aux out?
14:18.38Hmmhesaysyes however I need the rest of the band in the mix too
14:18.57fourcheezeok, so do you want one set of monitors for the whole band?
14:18.59Hmmhesaysthis was easy on the hold mixer cause everything was labeled for idiots like me
14:19.23Hmmhesaysno we had two channels set up,  1 for the bass and drummer 1 for the guitarists
14:19.27fourcheezeok
14:19.36fourcheezedo you have things like aux1 and aux2 out or send ?
14:19.47fourcheezeprobably 1/4" jacks
14:19.58Hmmhesayson my amp I have 1/4' line out
14:20.18*** join/#asterisk dangerarea (n=kevin@podcast-audio.chrysalis.com)
14:20.21fourcheezeno, on the mixer
14:20.35dangerareahey all
14:20.45fourcheezeI'm guessing that the first 2 pink knobs on each line are aux1/2
14:20.56dangerareaanyone an expert on getting ztdummy to work on debian?
14:20.58Hmmhesaysi'm not sure i haven't looked at it carefully enough I guess
14:21.04Hmmhesaysdangerarea its a no brainer
14:21.04fourcheezeshould send stuff out to a line out in the mixer output section
14:21.08Joe__11Hi all! I trying to understand how chan_local.c (Local channel) works. Anybody knows why it uses two channels (owner and chan) not one?
14:21.25fourcheezeHmmhesays: ahh yes look: http://media.zzounds.com/media/brand,zzounds/sl3242fx-b29baf30e8501cd84ec154272bc6c6df.jpg
14:21.26HmmhesaysI'm going to have to scourer the manual
14:21.28dangerareaHmmhesays in a good or bad way :)
14:21.50op3rdangerarea: how about just use centos?
14:21.51fourcheezeHmmhesays: look top right
14:21.52Hmmhesaysdangerarea good
14:22.05fourcheezefrom the right
14:22.05Hmmhesaysholy hi rez batman
14:22.14dangerareai'm having issues loading it in
14:22.21fourcheezesee where it says "aux send"
14:22.32dangerareacan't get my usb-uhci module to load
14:22.45fourcheezeHmmhesays: pick one for you and one for the drummer
14:23.04fourcheezeHmmhesays: then on each line in you set how much of that line to send to each aux out
14:23.07fourcheezedead easy
14:23.13Hmmhesayswith the pink knobs
14:23.22fourcheezeyeah
14:23.24KattyHmmhesays: you framilier with mounting w2k3 shares?
14:23.34HmmhesaysKatty not really
14:23.39KattyHmmhesays: m'kay
14:24.06benjamin7062Katty, I assume mounting from unix?  What is your problem?
14:24.17Hmmhesaysfourcheeze i'm going to have to play with this tonight
14:25.05SHad|Workanyone here has any exerience with octoBRI or GSM cards?
14:25.15SHad|Workor any zaptel interfaces
14:25.30fourcheezeHmmhesays: yeah, I wish I was :-)
14:25.53HmmhesaysI'm thinking of going with some kustom floor wedges... they are cheap
14:25.56Hmmhesaysand the reviews are decent
14:25.59[TK]D-FenderKatty : "man smb"?  The manpage gives a good quick 1-liner for it IIRC
14:25.59benjamin7062Katty, are you trying to configure samba?  Do you have active directory?  etc?
14:26.53benjamin7062Katty, if you want it to be a local mount point make sure you have the smbfs support for the kernel
14:28.36SplasPoodHrm.. is there any way to handle Background() within a macro?  Ie.. I want to Record() something, then give them a menu asking what they wanna do...
14:28.50*** join/#asterisk Persilon (n=ajolodov@200.123.112.152)
14:28.53benjamin7062Katty, if you are running debian... it's as easy as apt-get install smbfs
14:28.56Katty[TK]D-Fender: i can mount 2k shares fine. it's a 2k3 share that's giving me issues (=
14:28.58PersilonHi
14:29.14PersilonI'm getting: pbx.c:1700 pbx_extension_helper: No application 'MeetMe'
14:29.22Katty[TK]D-Fender: i have details, if you'd like them and are framilier with mounting 2k3 shares.
14:29.56nortexKatty, Are you in a Windows domain or are the share on a domain controller?
14:30.31[TK]D-FenderKatty : No, don't know that much... did jsut do it with Samba and WinXP Pro shares.
14:30.38*** part/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com)
14:30.53Katty[TK]D-Fender: m'kay.
14:30.56nortexKatty, 2K3 introduced a specific security feature that cripples connections from DOS and Linux boxes
14:31.18*** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de)
14:31.18Kattynortex: i'm sure i'll find someone to help me with it.
14:31.18*** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
14:32.09Persilon*CLI> show application MeetMe --> Your application(s) is (are) not registered
14:32.12nortexKatty, Do you have access to the 2K3 box? I have changed the setting on ours and can give you the details as to what we found.
14:32.22Persilonis meetme bundled in stantard asterisk distribution ?
14:32.37Kattynortex: i've got access to every box in the house, bring it on (=
14:32.37NotJohnDavidespresso, oh how I love thee
14:32.48*** part/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net)
14:35.00*** join/#asterisk smackus (n=smackus@63.149.122.94)
14:35.01ManxPowerPersilon, Zaptel MUST be installed when you build Asterisk or MeetMe will not be built.
14:35.01*** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243)
14:35.42smackusi have been trying for about a week now to get music on hold to work. I have reinstalled my asterisk and messed with about every little thing I can think of. When i put a call on hold, I get the following message:
14:35.43PersilonManxPower: can I use meetme over a sip channel ?
14:35.43smackus<PROTECTED>
14:35.43smackus<PROTECTED>
14:35.53smackusit starts and stops instantly.
14:35.56ManxPowerPersilon, Yes, but it requires Zaptel for timing
14:36.00smackuswhat could this be from?
14:36.02dangerareaanyone know how to get round this...
14:36.02dangerareapbx:~# modprobe ztdummy
14:36.02dangerareaHint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters.
14:36.03dangerarea<PROTECTED>
14:36.06PersilonManxPower: ok, thank you
14:36.24dangerareaoops
14:36.56[TK]D-Fendersmackus : Pastebin your musiconhold.conf
14:36.58[TK]D-Fender~pb
14:37.00jbotextra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
14:37.03*** join/#asterisk jj- (n=jj-@82.128.184.75)
14:37.11jbalcombwhy does my syntax highlighting work one server and not the other? I've copied .bashrc and .vimrc.
14:37.32nortexKatty, Okay the setting is changed in the Domain Controller Security Settings > Local Policies > Secutiry Options. Look for Microsoft network server: Digitally sign communications (always) and disable it.
14:40.08*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
14:40.36ManxPowerjbalcomb, does ldd /usr/sbin/asterisk show that libcurses is linked in?
14:41.02ManxPoweror libncurses
14:41.21dangerareahttp://pastebin.com/733145
14:41.25dangerareaany suggestions?
14:42.12jbalcombManxPower: well, sometimes I don't think right until I ask. The working system has vim, the non-working system has nvi.
14:43.09*** join/#asterisk MattH (n=matt@noc-wireless.chilitech.net)
14:43.09smackussorry... dropped my network connection.
14:43.13fourcheezeHmmhesays: what's the little button between the 2 sets of pink knobs ?
14:43.24MattHHi.. does anyone know what the sip registration string should be to get an F3000 phone to connect to asterisk?  I can't seem to get mine online
14:43.30jbalcombSo `apt-get install vim` fixed that problem. Hard to beleive a text editor should be 15.2 MB thoughh.
14:43.30MattHer.. rather it's online but I can't get it to register
14:43.49Hmmhesays"PRE"
14:43.52*** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
14:43.55*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
14:44.10*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
14:44.35smackushere is my musiconhold.conf http://pastebin.ca/73156
14:45.03fourcheezeHmmhesays: ok that's probably a pre-fade listen thing to monitor the line
14:45.11Hmmhesaysyeah I believe so
14:45.21fourcheezein which case the 2 pink knobs below that will be post-fade auxes for effects
14:45.36fourcheezeit's amazing how little mixer design changes
14:46.17*** join/#asterisk SwK[Work] (n=SwK@64.89.118.139)
14:47.07jbalcombGrr.. what kinda bonehead forces an anti-virus definitions update during work hours for 100+ PCs?
14:47.36Hmmhesaysyour average computer science major graduate?
14:47.38*** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
14:48.44jbalcombHmmhesays: perhaps, or gumptionless old admin who doesn't really know how to do anything?
14:49.52jbalcombHmmhesays: As I understand it, this guys main qualification is that he has no life and only lives five minutes from the building.
14:49.53[TK]D-Fendersmackus : Got MP3's in that folder?  Without ID3 tags I hope?  Did you install asterisk-addons to provide MP3 support?
14:50.12smackusi did install asterisk-addons.
14:50.14Hmmhesayscool jbalcomb pefect admin
14:50.36jbalcomb[TK]D-Fender: What, no ID3 tages? You need ID3 tags to feed to festival and announce what your playing!
14:50.48jbalcombHmmhesays: Indeed. =)
14:51.35*** join/#asterisk fugitivo (n=ajf@190.48.166.75)
14:51.37fugitivohello
14:51.42jbalcomb[TK]D-Fender: oh wait, it says 'no ID3 tags' on the wiki.. Asterisk is no fun.
14:51.50smackushere is the content of my directory http://pastebin.ca/73160
14:52.26[TK]D-Fendersmackus : get rid of that "backup" file...
14:52.45fugitivoI need agents in a queue to have a limit of 2 outgoing calls and a limit of 1 incoming call, but I see that in asterisk 1.2.x there's only a call-limit parameter in sip.conf for both incoming and outgoings
14:52.46[TK]D-Fendersmackus :err ..folder.. hmm.. shouldn't matter
14:52.50jbalcomb[TK]D-Fender Maybe you could set up a conference server so people can stream the satelite radio over the phone?
14:53.10[TK]D-Fender...
14:54.19fugitivoany way to do that?
14:54.31Kattynortex: thanks for heads up on the local policy setting, but ours is already disabled.
14:55.05Kattynortex: would you like a pastebin?
14:55.51nortexKatty, Sure
14:56.32Persilonis there anyway of setting permissions on monitor recorded files as it records them ?
14:59.01fugitivoor a way to not hear the call waiting tone on queues
14:59.36Kattynortex: http://pastebin.ca/73165
15:00.03Kattynortex: i have a feeling i know where i need to go with this...
15:00.08Kattynortex: but a second opinion is always handy.
15:01.39*** join/#asterisk MACscr (n=MACscr@66.73.154.70)
15:07.02fugitivoso, isn't that possible?
15:07.12*** join/#asterisk NLinington (n=nfl@82-69-27-212.dsl.in-addr.zen.co.uk)
15:07.23nortexKatty, I did mount -t cifs -o username=siusername,password='si(password' //192.168.0.3/asterisk /mnt/asterisk with one of my 2K3 boxes and it worked.
15:07.56*** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg)
15:08.03Tilihow can i force a callerid from sip.conf
15:08.10Tilii set callerid=whatiwant
15:08.14Tilibut still i dont get that one
15:08.54nortexKatty, I am not sure, but the error about cifs not supported would be my guess as to the problem. But I'm more a windows person then linux. I'm running Centos which may make the difference.
15:10.21EinsteinTaylormorning all
15:10.28EinsteinTaylorsomeone pass the caffeine please
15:11.00*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
15:13.25smackusi have chmod'd and chowned the files and the directory in my mohmp3 directory and I still get:
15:13.25smackus<PROTECTED>
15:13.26smackus<PROTECTED>
15:15.06lilalinuxdoes cdr_pgsql work with unix sockets, too?
15:15.16lilalinuxor do I need to enable tcp/ip in postgresql?
15:15.22Kattynortex: i will look into cifs, like i expected then
15:16.01PersilonI need some help with ChanIsAvail... it doesn't jump n+101
15:16.53MACscris their any type of app or website that can be used to test the number of simultaneus phone calls your internet connection can handle with SIP?
15:17.00smackusany other ideas for the music on hold issue?
15:17.15*** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)
15:17.20pdtmobilesmackus: i suggest native, just convert the stuff to gsm
15:17.34pdtmobilesimple sox command and your done
15:17.40smackusdoes native not do mp3?
15:17.41*** join/#asterisk skraelings001 (n=skraelin@201.230.140.95)
15:17.59anthmif you want it to it can
15:18.02Kattynortex: any thoughts on how to get cifs?
15:18.04smackusok
15:18.22smackusso I am using native, and I have the directory pointed to the mohmp3 directory with 3 mp3 files in it
15:18.28*** part/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg)
15:18.29*** join/#asterisk __undef (i=uxjf@rzstud2.stud.uni-karlsruhe.de)
15:18.30skraelings001Hi everyone
15:18.31anthmI made format_mp3 in asterisk-addons you can install
15:18.34__undefhi
15:18.36fugitivoMACscr: google for voip bandwidth calculator
15:18.47anthmbut you are needlessly decoding it over and over again fyi
15:18.56MACscrthanks Fugitivo
15:19.03pdtmobilethats why I suggest a one time convert to gsm and call it a day
15:19.10pdtmobilelower processor overhead blah blah
15:19.19anthmor convert to slin
15:19.28pdtmobileya
15:19.33anthmor if you are really stingy convert it to every codec
15:19.38pdtmobilehehe
15:19.39Kattynortex: actually, this is a local box.
15:19.40anthmthen it will pick the one that matches your channel
15:19.42coppicejust don't keep converting.
15:19.48Kattynortex: cifs shouldn't even be needed.
15:20.30PersilonI'm getting app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
15:20.35*** join/#asterisk tlow (n=tlowe@bgp.terrorist.net)
15:20.57MACscrcrap, i dont know what algorithm to pick. Is their any particular standard for SIP that is most widely used?
15:20.57smackusso if i am doing native, and i have all the files, and it is still not working, where else can I look?
15:21.34mutg711 ulaw
15:21.48anthmnot working how, I made the original cut of native music but I can't say they didn't change it since then
15:22.01smackus<PROTECTED>
15:22.01smackus<PROTECTED>
15:22.04smackusnever plays anything
15:22.12smackusstarts and stops in an instant
15:22.22anthmis there still the cli command to list the files ?
15:22.37anthmmoh files show
15:22.45smackusok, hang on
15:23.00smackusshows nothing.
15:23.15smackusthat would explain why it does not play
15:23.26smackusbut how do i explain why nothing shows in moh files show?
15:23.52anthmI think they changed the way you config lately
15:23.56anthmdid you have it a while ?
15:24.17smackusI am installing from scratch.
15:24.34anthmyou said they are mp3?
15:24.36smackusyes
15:24.41anthmif you dont have format_mp3 loaded
15:24.50smackusin modules?
15:24.51anthmthey will be skipped since they are invalid
15:24.57__undefis there any way to generate a call on a pri_net span?
15:25.09blitz[laptop]__undef: from where?
15:25.15anthmthere is an initial pass to verify the extensions against the various formats you have loaded
15:25.18__undefblitz[laptop]: from the host itself ;)
15:25.24blitz[laptop]__undef: callfiles
15:25.36__undefblitz[laptop]: okay, thanks...
15:25.40blitz[laptop]np
15:26.17anthmso either install format_mp3 or sox them to raw or cp some of those stock audio files as a test
15:26.20smackusshow modules lists: format_mp3.so
15:26.47anthmif that's the case you must have something configed wrong
15:27.05smackusi would assume so, where can we start looking it over.
15:27.13anthmlook at the console when it loads it
15:27.25anthmif you wnat to make it easier
15:27.37anthmadd noload => res_musiconhold.so to the modules.conf
15:27.47smackushere is my musiconhold.conf http://pastebin.ca/73156
15:27.48anthmthen start with debug log on and lots of v
15:27.59anthmthen load it from the cli so you can see the msgs
15:29.21anthmmy guess is moh loads before mp3 does
15:29.30Hmmhesaysi'm kind of like this FC5 for a Workstation
15:29.32anthmso it doesnt know it's a valid format till it's too late
15:29.35Hmmhesaysit is pretty
15:29.44anthmso you can stick in manual load => lines at the top
15:29.54anthmso you load all your file formats first
15:29.58anthmthen the music
15:30.03Hmmhesaysdo i need to load res_crypto if i'm not using any of the md5 stuff?
15:30.24smackusanthm: ok, hang on
15:30.43*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
15:30.44anthmisnt res_crypto one of those lazy linked ones that other code calls stuff from?
15:30.46smackusi have added the noload => res_musiconhold.so
15:30.52Hmmhesaysi'm really not sure
15:31.03anthmthat will prove it
15:31.04Hmmhesaysits sure been a pain in the ass for me trying to compile for mipsel though
15:31.07anthmso if you start up now
15:31.15anthmand it works by hand loading music
15:31.20Tilii can't seem to override callerid by setting it in iax.conf or sip.conf
15:31.35smackusthis is where you lost me... and it works by hand loading music
15:31.53anthmenter >load res_musiconhold.so
15:31.56anthmat cli
15:32.01anthmaka hand loading it
15:32.09smackusok
15:32.35anthmbecause you are loading it after everything else
15:32.45anthmit will probably work now
15:33.08smackusslk-apbx-01*CLI> load res_musiconhold.so
15:33.08smackusUnable to load module res_musiconhold.so
15:33.08smackusJun 27 09:32:45 WARNING[7801]: loader.c:305 __load_resource: Module 'res_musiconhold.so' already exists
15:33.33anthmthen you didnt noload it right
15:33.41smackusok... hang on,
15:34.00smackusnoload => res_musiconhold.so
15:34.03smackusright?
15:34.04Qwelldid you restart?
15:34.14smackusI reloaded... i must restart?
15:34.17anthmyes
15:34.19smackusok...
15:34.20Qwellwell..yeah
15:34.28smackusthat will take some time. I will get back with you
15:34.37Qwellnot the whole machine
15:34.48smackusright... just have alot of callers on it right now
15:34.54anthmif you cant do that then do moh reload
15:34.57anthmsee if that woeks
15:35.01anthmit may rescan the files
15:35.03smackusdid a restart when convenient
15:35.30CunningPikeHeh - if I did a 'restart when convenient' now, I could wait for 12 hours........
15:35.44anthmtry "moh reload"
15:35.48smackusok
15:36.05anthmyou are just trying to prove it's the load catch-22
15:36.32anthmmost likely you will need to add a bunch of .... load => format_*
15:36.37anthmfor all the formats
15:36.39smackusmust not have reloaded, cuz i got the same error
15:36.48anthmthen load => res_musiconhold
15:36.54NLiningtonHi, has anybody got faxing working using the app_rxfax module?
15:36.57anthmso you are sure you have all the file formats loaded before music
15:37.09smackusno
15:37.26smackushow do i make sure?
15:37.49anthmdid moh reload say it reload
15:37.55anthmthen try moh show files again
15:38.07anthmmoh files show i mean
15:38.39smackusnow there are files.
15:39.07anthmso then it should work now
15:39.11smackusyes it does.
15:39.23smackusso just adding the noload was the fix?
15:39.26anthmno
15:39.33anthmthe fix will be
15:39.39anthmadding actual load lines
15:39.46smackusok
15:39.57smackuswalk me through this, because I am a little confused.
15:40.12smackusso do i need to take out the noload line?
15:40.35anthmya
15:40.37anthmtake it out
15:40.41smackusok
15:40.53smackusdone
15:41.37anthmin a shell
15:41.48anthmcd /usr/lib/asterisk/modules
15:42.00stephane_soir
15:42.03MACscris their managed asterisk hosting out there?
15:42.07anthmexecute /bin/ls -1 format_* | awk '{print "load => "$1}'
15:42.16MACscrusing that particular text, i didnt fine much with google
15:42.18smackusdone
15:42.19anthmand cut and paste the output to the top of modules.conf
15:42.44anthmthen add load => res_musiconhold.so under that
15:42.57anthmthat should do it
15:43.59smackusok, thanks
15:44.04smackustesting now.
15:44.14anthmthis way you load all the formats first
15:44.17anthmthen music
15:44.21mutLMFAO!
15:44.22muthttp://www.youtube.com/watch?v=2S89Y4shxtE&search=Maury%20Show%20phobia
15:44.27anthmso they do not skip any
15:44.39anthmcos say format_mp3 is loaded after music
15:44.51smackusthank you very much
15:44.55anthmthen music will skip all mp3 as invalid cos it's not loaded yet
15:44.57anthmnp
15:44.59*** part/#asterisk mog (i=ejabberd@68.62.237.103)
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15:48.52*** join/#asterisk DarKnesS_WolF (n=wolf@82.201.230.137)
15:52.50*** join/#asterisk monkeyshine (n=countjas@216.64.160.84)
15:52.54monkeyshinei need some help
15:53.43monkeyshinemy sipura boxes are acting as a dhcp server and my internet line has a different gateway
15:53.44*** join/#asterisk DarKnesS_WolF (n=wolf@82.201.219.203)
15:53.58monkeyshinecan i bridge 2 gatgeways
15:54.14*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
15:59.33*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
15:59.50generalhanwhats up all !
16:01.03NLiningtonmy * box is doing a seg fault every time I call app_rxfax, has anybody some idea how to fix this?
16:01.31*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
16:01.38h3xnling: are you doing t.30 with it?
16:02.07NLiningtonh3x nope, I have a single pstn inbound line I have fax detection on
16:02.24h3xI would say thats T.30 :)
16:02.49NLiningtonoops
16:03.05NLiningtonthought you meant fax over voip
16:03.11h3xcan you tell if its really app_rxfax that causes the fault or is it fax detection
16:03.24h3xthat is t.37 or t.38
16:03.39h3xor t.30 over g711 :)
16:04.06{zombie}NLinington: make sure the version of spandsp (libtiff) used to compile app_rxfax is the same as is installed on your system
16:04.10*** join/#asterisk speedwagon (n=Ariel@70.46.87.158)
16:04.16h3xthat too
16:04.17{zombie}oh, and that app_rxfax was compiled for your asterisk
16:04.22NLiningtonthe fax detection works ok as I can see the script running ok and it plays a wav file saying 'fax detected'
16:04.27monkeyshineis there  a way to bridge gateway
16:04.33*** join/#asterisk skirmisha (n=vlado@87.126.55.7)
16:04.41skirmishahi guys
16:04.59NLiningtonI have completly re-compiled asterisk, spansp, all the libe and reinsrtalled from scratch
16:05.11*** part/#asterisk fourcheeze (n=rich@82.153.215.21)
16:05.16skirmishacan someone tell me where the asterisk PBX comes from in the email header under Sender field when i send voicemail msg
16:05.26*** join/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com)
16:05.35sonic69i need a script that will delete voicemail messages folder when i delete an account in php interface... can someone help me with that??
16:05.39{zombie}skirmisha: voicemail.conf
16:06.04skirmishait's not from there
16:06.13skirmishai've been playing 2 days
16:06.14*** join/#asterisk mog (i=ejabberd@68.62.237.103)
16:06.31skirmishaand it comes from somewhere and i don;t know how to rewrite it with mail server
16:06.35{zombie}; Change the From: string
16:06.35{zombie}fromstring=Asterisk PBX
16:06.46*** join/#asterisk SplasPood (n=jwb@206.252.198.101)
16:06.49{zombie}that's where I changed mine
16:07.06{zombie}did you restart asterisk after making the change?
16:07.14skirmishayes i did
16:07.27h3xsonic: I'd hang on to the voicemails for extortion purposes :)
16:07.29skirmishathe problem is that from string is ok , but sender string is not
16:07.52skirmishaSender:  asterisk PBX <voicemail@
16:07.56jbalcombis pastebin broken?
16:08.05skirmishathis is what i get in mail header
16:08.10h3xsonic: actually you should set up a cronjob to delete voicemails on accounts that were deleted 30 days ago or something
16:08.49jbalcombh3x: can you send us yours?
16:09.10h3xmy what?
16:09.17jbalcombyour cronjob
16:09.22h3xi dont have one
16:09.28jbalcombwhy not?
16:09.41h3xcoz i have better customer retention than you do ? hahaha
16:10.09skirmishaanyone who can help me?
16:10.09jbalcombhaha.. yeah, that's funny.
16:10.19h3xactually i dont have many customers that have voicemail
16:10.43_problem_skirmisha: whats ur problem u were been told by {zombie}
16:10.58*** join/#asterisk tRSS (n=tRSS@193.220.221.2)
16:11.11_problem_skirmisha: thats what i did what he says and it works for me also
16:11.21skirmisha{zombie} told me to check in the config file
16:11.27skirmishabut the problem comes from source
16:11.30tRSSdo we have anyone interested in developing a fully functional inbound/outbound predictive dialer on top of asterisk?
16:11.32CunningPikeskirmisha: Did you try sendmail.cf like I suggested?
16:11.41generalhanhey can someone explain to me how to get information from the CDR and keep it in that format?
16:11.44_problem_skirmisha: its all in the voicemail.conf
16:11.46generalhanyea htat sounded real clear ... lol
16:11.48{zombie}skirmisha: I don't get a "Sender:" field, so your MTA must be adding that
16:11.54skirmishaCunningPike yes i did manage to chamge the address
16:12.05_problem_skirmisha: pastebin ur voicemail.conf
16:12.19skirmishabut it does not change related info
16:12.30generalhanLike if i use an "fgrep something" in that file it comes out in just text so its hard to read ... i need to be able to do that but keep it in a .csv format
16:12.30CunningPikeskirmisha: So your Sender: header is now correct?
16:12.38skirmishait comes as Sender:  asterisk PBX <voicemail@your domain>
16:12.53skirmishai want to change this asterisk pbx
16:13.03skirmishai did managed to change the email
16:13.04{zombie}is your MTA grabbing that from /etc/passwd maybe?
16:13.14skirmishai am using exim
16:13.26*** join/#asterisk oej (n=olle@212.17.152.81)
16:13.37{zombie}try "chfn asterisk" or "chfn voicemail"
16:14.52skirmisha{zombie} where should i type this
16:15.07{zombie}from the linux commandline (as root)
16:17.30*** join/#asterisk tRSS (n=tRSS@193.220.221.2)
16:18.00skirmishaahhh here it is
16:18.03skirmishathanks guys
16:18.17*** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
16:18.22TripleFFFF: is there a way to sip debug  a peer to file
16:18.29skirmishadamn mail server
16:18.31TripleFFFFthanks..
16:19.47jbalcombTripleFFFF: maybe asterisk -rc sip debug peer <exten> > <file> ?
16:19.59TripleFFFFoh
16:20.22wunderkinyou are thinking of x not c, and no
16:20.40TripleFFFF?
16:20.56TripleFFFFso that would not work
16:20.56TripleFFFFhmm
16:21.09TripleFFFFwould be frigign usefull to imlpement.. loger.conf.
16:21.18TripleFFFFdebugpeer1=username
16:21.20TripleFFFFro somethign
16:21.27*** join/#asterisk Greek-Boy (n=grb@193.220.93.162)
16:21.47NotJohnDavidWish a decent Socket A motherboard was cheaper than they are currently
16:22.23jbalcombNotJohnDavid: what CPU is Socket A for?
16:22.26*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
16:22.49*** join/#asterisk burnproof (n=burnproo@210.213.199.85)
16:22.55*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
16:23.29NotJohnDavidAMD Athlon
16:24.20jbalcombah, I have several Socket A boards then..
16:24.31TripleFFFFis there a way to put a chanel on hold without a hold button ?
16:24.33TripleFFFFlike *99
16:24.37TripleFFFFbut how to get back ?
16:24.40TripleFFFF*99 again ?
16:25.31[TK]D-FenderTripleFFFF : Depends on the phone
16:25.35TripleFFFFnormal phone
16:25.41TripleFFFFno hold button
16:25.42{zombie}hookflash
16:25.45TripleFFFFsoyeah
16:25.45[TK]D-FenderTripleFFFF : Connected how?
16:25.49TripleFFFFpap2
16:26.07[TK]D-FenderTripleFFFF : PAP2 should have a * code feature built in for that like the SPA series does
16:26.17skirmisha{zombie} thank all is working fine now
16:26.21{zombie}cool
16:26.23[TK]D-FenderTripleFFFF : Download the admin/user guide
16:26.31TripleFFFFlol
16:26.38TripleFFFFlinksys pap2 na too ?
16:27.06{zombie}TripleFFFF: hookflash to put them on hold, hookflash again to unhold. if you hookflash then dial a number you can transfer the call.
16:28.34*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
16:29.04Greek-BoyI'm trying to record all calls/channels on my asterisk box. How do I go about it? I had a look at http://www.voip-info.org/wiki/view/Monitor+stereo-example but it didn't work for me. I need to record all outgoing calls from every extension
16:29.59skirmishathere is config file Greek-Boy
16:30.25Greek-Boywhich conf file?
16:30.45skirmishalet me check it
16:31.17skirmishafutures.conf
16:32.15*** join/#asterisk Monkey13 (n=jcheuvro@69.7.217.140)
16:32.25skirmishaautomon=*1
16:32.42TripleFFFFhey
16:32.47TripleFFFFwhats the no bridge option ?
16:33.08TripleFFFFlevel 1: start=2006-06-27 12:23:10
16:33.08TripleFFFFlevel 1: answer=2006-06-27 12:23:21
16:33.08TripleFFFFlevel 1: end=2006-06-27 12:23:21
16:33.13TripleFFFFand channel still up
16:35.38TripleFFFFso i can
16:35.40TripleFFFFknow
16:36.02lilalinuxdoes anyone use twinkle as a sip client/
16:36.30TripleFFFF<PROTECTED>
16:36.35TripleFFFFso how can i know bridge
16:36.39TripleFFFFun bridge
16:38.31*** join/#asterisk tRSS (n=tRSS@193.220.221.2)
16:41.47*** join/#asterisk dacleric (n=dacleric@p54822D4E.dip0.t-ipconnect.de)
16:44.17dlynes_officetwinkle, twinkle little star...
16:44.32*** join/#asterisk nortex (n=nortex@64.136.65.142)
16:45.46*** join/#asterisk rnovotny22 (n=Bob@198.57.19.126)
16:48.19lilalinuxdlynes_office: do you know if twinkle supports md5 passwords?
16:48.30dlynes_officelilalinux: no idea
16:48.38dlynes_officelilalinux: i don't even know what it is
16:48.48lilalinuxa sip client
16:48.55dlynes_officeah
16:49.11dlynes_officesounds like a sip client for fairies
16:49.45Greek-Boywhat is wrong with this?
16:49.45Greek-Boyexten => 301,1,Dial(SIP/kc-cashdesk01,,r)
16:49.45Greek-Boyexten => 301,2,SetVar(CALLFILENAME=i${CALLERIDNUM}-${TIMESTAMP})
16:49.45Greek-Boyexten => 301,3,Monitor(wav,${CALLFILENAME},m
16:49.53Greek-Boy?
16:50.05Greek-Boyi'm not getting any output files
16:50.27Greek-BoyI also have MONITOR_EXEC=/usr/local/bin/2wav2mp3 under my globals as a custom script
16:50.27*** join/#asterisk mog (i=ejabberd@68.62.237.103)
16:50.39nortexGreek-Boy, No closing ) on the monitor command
16:50.47*** part/#asterisk mog (i=ejabberd@68.62.237.103)
16:51.05*** join/#asterisk Curus (n=Curus@kbhn-vbrg-sr0-vl209-213-185-8-10.perspektivbredband.net)
16:52.04Greek-Boywhat should permissions be on /usr/local/bin/2wav2mp3?
16:52.23nortexWhat are some typical causes of one sided audio between PRI channels and SIP clients?
16:52.33h3xNAT
16:52.40h3xdouble NAT
16:52.43h3xheh
16:53.36darkskiezfirewall
16:54.06darkskiezbroken microphones/earpieces
16:54.16darkskiezvolume settings
16:54.33Greek-BoyI now have:
16:54.34Greek-Boyexten => 301,1,Dial(SIP/kc-cashdesk01,,r)
16:54.34Greek-Boyexten => 301,2,SetVar(CALLFILENAME=i${CALLERIDNUM}-${TIMESTAMP})
16:54.34Greek-Boyexten => 301,3,Monitor(wav,${CALLFILENAME},m)
16:54.44Greek-Boybut still no output files
16:55.00Greek-Boywhat could be wrong?
16:55.12vader--anyone have any issues where their sip phones won't do anything in the features.conf?
16:55.14vader--like *1 for recording and stuff
16:55.20vader--i have cisco 7940G phones
16:55.26nortexh3x, No nat or firewall present between Server and client.
16:55.32darkskiezvader--: make sure the right params enable that in the dial options
16:56.19*** join/#asterisk NeonLevel (i=NeonLeve@200.52.142.184)
16:56.50*** part/#asterisk NeonLevel (i=NeonLeve@200.52.142.184)
16:57.12*** join/#asterisk heath__ (n=heath__@71-87-34-39.dhcp.stcd.mn.charter.com)
16:57.31heath__how new does one's version have to be for atxfer to work in features.conf?
16:57.47MACscrwhy are so many voip providers calling Auto Attendants, IVRs
16:57.55*** part/#asterisk skraelings001 (n=skraelin@201.230.140.95)
16:57.55h3xheath lives
16:58.02MACscrit doesnt make sense
16:58.36vader--darkskiez you mean in the extensions.conf?
17:01.06rnovotny22Can incoming PSTN and SIP calls be routed to the same context?
17:01.08TripleFFFFso ..
17:01.19TripleFFFFif you have a normal 1 sip phone pap2
17:01.30TripleFFFFand got 4 lines on sip account.. how can you have callwaiting ?
17:05.01*** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-145-155.gdrpmi.dsl-w.verizon.net)
17:07.10*** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net)
17:07.34variable_officemusiconhold immediatly stops for me, but reports no errors, what could be causing this?
17:08.45*** join/#asterisk mog (i=ejabberd@68.62.237.103)
17:08.51*** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net)
17:08.52dlynes_officevariable_office: your organ grinder monkey died
17:09.19variable_officeic
17:09.56variable_officereally though, i have asterisk-addons and sounds installed, and i am just trying to use the default class that comes with the sample conf file
17:11.05dlynes_officevariable_office: you don't believe me?
17:11.09dlynes_officevariable_office: i'm hurt :((
17:11.28coppiceI thought it was the RIAA who stopped it
17:11.36*** part/#asterisk codestr0m (n=asura@ns2.netsyncro.com)
17:11.45dlynes_officecoppice: yeah...probably...found out he had metallica mp3s on his moh
17:12.11coppicethe system itself should stop that to reduce unneccessary suffering
17:12.45nortexcoppice, Some would say that about the default mp3s
17:13.03*** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com)
17:13.11Stephniehi
17:13.24variable_officeno ideas then i take it eh?
17:13.31nortexStephnie, hello
17:14.07TripleFFFFwonders how a girl naick gets attention ;)
17:14.46nortexTripleFFFF, I could answer her question :)
17:15.28*** join/#asterisk pengyong (n=lala@218.93.154.125)
17:15.31TripleFFFF;)
17:15.44TripleFFFFi could answer her needs';)
17:15.48TripleFFFFj/k
17:16.03TripleFFFFanyone have pap2 t na ?
17:16.16TripleFFFFim getting probs with the 2nd line ref
17:16.17TripleFFFFreg
17:17.01Stephniehi
17:17.09nortexTripleFFFF, don't have one, but are both lines try to reg using the 5060 port.
17:17.40TripleFFFFno
17:17.42TripleFFFF5061
17:17.44TripleFFFFfor 2
17:17.46*** join/#asterisk mdiehl (n=mdiehl@c-69-252-219-76.hsd1.nm.comcast.net)
17:18.04TripleFFFFand is Bridged Call(SIP/blah) normal ?
17:18.07mdiehlHi all.
17:18.08TripleFFFFbridged i mean
17:18.11TripleFFFFhi
17:18.25nortexhi
17:18.39mdiehlGot a question.
17:19.09nortexTripleFFFF, You should see bridged call when asterisk connects the two parts of the call
17:19.16Qwell[]mdiehl: we've got dumbfounded looks
17:19.17nortexmdiehl, Fire away
17:19.20mdiehlI just removed a tdm card from my * server, now call transfer doesn't work, voicemail doesn't work.
17:19.31mdiehlAnd conferenceing STILL doesn't work.
17:19.44mdiehlI've got ztdummy loaded.
17:20.03mdiehl...says that voicemailbox can't accept new messages.
17:20.04*** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net)
17:20.09FlyboySR22g us for it.
17:20.24TripleFFFFk
17:20.44TripleFFFFso ..line 1 connect.. line 2 cant connect to server...
17:20.44TripleFFFFif i make line 2 login onto line 1 ..it sworks
17:20.46Dr-Linuxquestion about spa 2100, what should be "NAT Keep Alive Intvl:?" by default it's 15
17:20.54TripleFFFFso the pap just doens want to use line 2 reg
17:20.57mdiehlThings wer working better with the card installed.... eventhough it was sharing an irq with 3 nic's.
17:21.07*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.196)
17:21.13Dr-Linuxsometime line goes dead, anybody can tell me if this is a reason?
17:21.26rob0Qwell[]: you work for Radio Shark? :)
17:22.15mdiehlAny ideas as to where to begin to fix this?
17:22.49dlynes_officevariable_office: nah...it's not that...i was just busy in the office :p
17:22.59rob0Problem with IRC: the dumbfounded looks are hard to communicate. Is there an emoticon for that?
17:23.07Qwell[]rob0: ":)"
17:23.11nortexTripleFFFF, can you pastebin your sip.conf file?
17:23.30Stephniehttp://pastebin.ca/73259  . . . .anyone?
17:24.09rob0No, that's *happy* -- when you catch a supposed tech person who's pretending, they're seldom happy about that.
17:24.28dlynes_officevariable_office: can you give me a bit more information as to what your moh environment is like?  i.e. what's your configuration, ...?
17:25.32TripleFFFFnortex not really
17:25.40TripleFFFFi dfoudn hte prob
17:25.40TripleFFFFthanks
17:25.44Stephnieanyone using ASTCC?
17:25.48_problem_Stephnie: there are many prepaid proprietary solutions available to use with asterisk ..u can check them on voip-info
17:26.19vader--anyone have any issues where their sip phones won't do anything in the features.conf?
17:26.21vader--i have cisco 7940G phones
17:26.21nortexTripleFFFF, Good deal!
17:26.22Stephnie_problem_ : what about astcc?
17:26.31rnovotny22Can incoming PSTN and SIP calls be routed to the same context in extensions.conf?
17:26.37TripleFFFFyes
17:26.46nortexrnovotny22, yep
17:26.48*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
17:27.02_problem_Stephnie: no i dont know abt that..never used that
17:27.02TripleFFFFjust make the ztconf etc use same exten
17:27.02TripleFFFFlol
17:27.10dlynes_office_problem_: i don't think she's really wanting to know what's available...just what's reliable for a heavy load, specifically astcc, and if not astcc, then an equivalent
17:27.13TripleFFFFso...
17:27.21TripleFFFFzap i mean
17:27.37_problem_dlynes_office: ok
17:27.38Stephniedlynes_office :  yeah ..
17:28.08dlynes_officeStephnie: astcc's just a perl script, right?
17:28.42Stephniedlynes_office : right....and in my production server I'll use AGI with Perl...is it gud?
17:29.08StephnieAGI + PERL + MYSQL .....
17:29.08dlynes_officeStephnie: yeah, should be fine
17:29.12dlynes_officeStephnie: but lose the mysql
17:29.23dlynes_officeStephnie: use postgresql instead...it's much more robust
17:29.36StephnieI thought to go for MS-SQL
17:29.37dlynes_officeStephnie: as soon as you start doing any kind of joins, you'll see what i mean
17:29.47dlynes_officeStephnie: mssql's fine, too
17:29.49Stephnieaaa aou....okey
17:29.56*** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net)
17:30.07dlynes_officeStephnie: but if you want to handle a lot of traffic, and you want to be able to handle business logic
17:30.18TripleFFFFanyone familiar with cdrool froma g crap
17:30.20TripleFFFF?
17:30.21dlynes_officeStephnie: imho, mysql is a horrible database; it's not terribly scalable
17:30.30TripleFFFFdlynes_office what ?
17:30.33*** join/#asterisk skraelings001 (n=skraelin@201.230.140.95)
17:30.37dlynes_officeTripleFFFF: cdrool?
17:30.38TripleFFFFdlynes_office use clustering engine
17:30.38salaudAnyone here have experience with H.323 and asterisk?  I could use some help interfacing IAX channels and H.323
17:30.45TripleFFFFdlynes_home yes
17:30.46dlynes_officeTripleFFFF: you didn't listen to me
17:30.51StephnieI see!! then ? u suggest me to use POSTGRESQL
17:30.53TripleFFFFit's not terribly scalable
17:30.55dlynes_officeTripleFFFF: mysql's join support is absolutely horrible
17:30.55h3xsala: try Yate for that
17:31.01TripleFFFFoh
17:31.02TripleFFFFyeah
17:31.06dlynes_officeTripleFFFF: you only listened to half of what i said
17:31.10TripleFFFFany db sucks.. imho
17:31.13rnovotny22nortex: Thanks, having problems with incoming sip calls getting busy all the time.
17:31.15h3xSupposedly Yate does a better job with H.323
17:31.28salaudh3x: Yate?
17:31.30TripleFFFFrnovotny22 .. what provider
17:31.31dlynes_officeStephnie: I only suggested postgresql because i assumed you wanted to go opensource
17:31.33rpmi can't find the kb article that talks about having multiple digium cards in a single machine, can someone point me to the url?
17:31.35TripleFFFFwhat hardware
17:31.36salaudh3x: A different PBX?
17:31.36h3xbut personally I would use H.323 to SIP than H.323 to IAX
17:31.37dlynes_officeStephnie: mssql's fine, too
17:31.41h3xSort of yes
17:31.47TripleFFFFrpm dont do it
17:31.57dlynes_officeStephnie: postgresql is about the closest you can get to oracle in the opensource world
17:32.07rnovotny22TripleFFFF:  BroadVoice
17:32.10h3xwhoracle
17:32.20Stephniedlynes_office : thanks ....
17:32.23TripleFFFFand a sip ? or asterisk
17:32.24salaudh3x:  My thing is that I have an existing IAX channel structure to VOIP providers and I just need to do H.323 with another phone system
17:32.25nortexrnovotny22, are you registering to your SIP provider?
17:32.36TripleFFFFsip debug peer (PERNAMEOFBROADVOICE)
17:32.45TripleFFFFthen make call
17:32.49dlynes_officeStephnie: basically what you want is something scalable, and that can handle a heavy load
17:32.49TripleFFFFcheck to see hwat happens
17:33.02h3xwell keep in mind that SIP and H.323 both use RTP
17:33.04h3xand IAX2 dosent
17:33.06dlynes_officeStephnie: oracle is probably total overkill for you, though
17:33.08TripleFFFFme thinks stephanie looking for scratch card calling card solution
17:33.13rnovotny22nortex: yes, current config is incoming on pstn, outgoing on sip.  Would like to use sip for incoming also.
17:33.25Stephniedlynes_office : so do u think that using AGI + PERL + POSTGRESQL or MYSQL is a professional way to use in production server for heavy traffic as they wont mess up the billing......
17:33.28salaudh3x: I'm trying to put asterisk in the middle to do the translation layer...
17:33.42salaudh3x: The solution only has to work for a few weeks until some PRI's come in
17:33.46dlynes_officeStephnie: like i said...take mysql out of the solution...it's horrible for business logic
17:33.51TripleFFFFhttp://www.xtenn.com/
17:33.58TripleFFFFWOW.. nice copryight issue
17:33.58h3xso stick yate in the middle of it
17:34.09h3xor find some other software that does h.323 -> sip like ummm
17:34.10rnovotny22dlynes_office:  Whats wrong with mysql.  It rivals oracle and is open source?
17:34.12StephnieTripleFFFF: yeah..... Pinless service...(authentication on Caller ID)
17:34.19h3xIn any case, asterisk and H.323 is awful
17:34.34TripleFFFFk
17:34.40TripleFFFFgood idea
17:34.46salaudh3x: So H.323 PBX -> Yate (SIP) -> Asterisk  -> VOIP Providers (IAX) ?
17:34.49TripleFFFFStepni
17:34.52Stephniedlynes_office : okey....forget about DATABASE .... what about AGI + PERL
17:35.08dlynes_officemovotny22:  mysql rivals oracle?  are you joking?
17:35.14h3xor scrap the h.323 pbx and use asterisk :)
17:35.28salaudh3x: Client is currently using asterisk...
17:35.33dlynes_officeStephnie: yeah, agi+perl is fine
17:35.39rnovotny22dlynes_office: nope, use it quite heavily here with no problems.
17:35.41salaudh3x: Unfortunately they are idiots and bought some Altigen solution
17:35.49h3xew
17:35.55dlynes_officemovotny22:  you don't use joins, do you?
17:35.59salaudh3x: But... they still want to pay us to bridge the two systems
17:36.14Stephniedlynes_office : great!!! so I think I should change ASTCC Scripts to use them in my production server....
17:36.18salaudh3x: Stupid people with money... never ceases to amaze me
17:36.26h3xThats like somebody wanting to use uucp for email in 2006
17:36.27h3xheh
17:36.27Stephniedlynes_office : what about oracle?
17:36.40salaudh3x: I'm thinking smoke signals instead of e-mail
17:36.44h3xhaha
17:36.46rnovotny22dlynes_office:  You just have to set them up correctly, then they are quite fast.  Faster in my case than oracle.
17:36.47dlynes_officeStephnie: oracle will definitely do the job, but as I said, it's total overkill for what you need
17:37.00salaudh3x: Is there a package (debian) for Yate?
17:37.13dlynes_officemovotny22:  Yeah, I know mysql can be faster than Oracle
17:37.17h3xi dont know, probably
17:37.32dlynes_officemovotny22:  but my problem with it, is if I do any kinda complex queries in mysql, it slows to a crawl
17:37.33Stephniedlynes_office : then I should go for mysql ...
17:37.36salaudh3x: Ok... I'll start digging around for Yate...
17:37.55salaudh3x: But you are pretty sure the H.323 channels for asterisk won't work at all?
17:38.04*** join/#asterisk lars-ut (n=lars-ut@70.103.228.158)
17:38.05h3xthey work but not very well
17:38.12h3xcapacity is a problem for sure
17:38.22salaudh3x:   We only have 8 channels
17:38.31h3xthe max capacity ive heard is like 10
17:38.32salaudh3x: I mean 12 channels
17:38.34h3xso thats even pushing it
17:38.44salaudh3x: But.. Yate doesn't have that limitation?
17:38.45rnovotny22dlynes_office:  I use joins all the time with multiple tables and have not seen any slowdown.  Several tables have close to a million records.
17:40.29*** join/#asterisk InfraRed (n=subhi@bb-87-81-46-122.ukonline.co.uk)
17:40.31InfraRedhi all
17:40.33h3xthey claim it dosent
17:40.41h3xbut it still sucks im sure just because its openh323
17:40.41*** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
17:40.43InfraRedanyone here using sip registration from behind nat ?
17:40.51Juggieeveryone?
17:40.53[TK]D-FenderInfraRed : Yup, plenty of us
17:40.53h3xopenh323 is huge
17:41.03dlynes_officeJuggie: ?
17:41.03h3xit takes about 4 hours to compile on a relatively fast machine
17:41.07h3xto give you an idea of how bloated it is
17:41.14InfraRedok what voodoo must i do to make it work ?
17:41.28Juggiedlynes_home, my answer was everyone, almost everyone uses sip behind nat.
17:41.34dlynes_officeJuggie: oh
17:41.34[TK]D-FenderInfraRed : Forward 2 port ranges, add 4 settings to sip.conf and thats it
17:41.36JuggieInfraRed, nat=yes for your client.
17:41.40dlynes_officeJuggie: Yeah...I did before
17:41.44dlynes_officeJuggie: don't now
17:41.45InfraRedi added the * server as DMZ
17:42.08JuggieInfraRed, so you have * behind nat, but DMZ on your router
17:42.10InfraRedJuggie: excellent idea
17:42.12Stephniedlynes_office : thank you for your suggestions....
17:42.12Juggieand then also clients behind nat
17:42.16Juggiecorrect?
17:42.19InfraRedJuggie: yes
17:42.24InfraReddouble nat
17:42.26Juggieset externip= in sip.conf
17:42.26[TK]D-FenderInfraRed : Then all you need is "localnet", "externip", and "nat=yes" added to your [general] section of sip.conf
17:42.29Stephniethanks everybody...:) tc...Bye
17:42.31Juggieto your real external ip.
17:42.35Juggieand nat=yes for your clients
17:42.38Juggiealso qualify=yes
17:42.38InfraRedfantastic
17:42.40Juggieand you should be good to go
17:43.05InfraRedwhat is the "localnet" directive?
17:43.13salaudh3x: That doesn't sound good
17:43.18dlynes_officeor qualify=300 so that you're compatible with most home office grade routers out there (qualify=150, for some really crappy cisco routers)
17:43.33salaudh3x: Is there anything that doesn't suck?
17:43.37Juggiedont worry about that unless you have clients on your local network
17:43.38*** join/#asterisk nortex (n=nortex@64.136.65.142)
17:43.38h3xI think vodiva has a h323 to sip stack
17:43.42Juggiebut they are on a different subnet
17:43.55salaudh3x: is that commercial?
17:43.55[TK]D-FenderInfraRed : For SIP to work, * has to forge your external IP address into the headers.  Any client calling you OUTSIDE your "local" networks will receive the forged one specified in "externip"
17:44.24*** join/#asterisk Damin (n=damin@nucleus.nacs.net)
17:44.30Juggieyou dont need to touch local unless your internal network is more then one subnet.
17:44.30*** part/#asterisk Damin (n=damin@nucleus.nacs.net)
17:44.39*** join/#asterisk Damin (n=damin@nucleus.nacs.net)
17:44.53Juggieeg, if your * box is 192.168.1.100 but you also have 192.168.2.x as a local network
17:44.55InfraRedok
17:44.57h3xhttp://www1.cs.columbia.edu/~kns10/research/gw/
17:45.48h3xthat looks like a clean implementation
17:46.29salaudh3x: So it is essentially only necessary to translate signaling and then everything is RTP from that point.
17:46.38h3xright
17:46.48Juggiertp is cake so long as your nat implementation doesnt suck
17:46.54h3xasterisk does alright at sip rtp to iax2 usually
17:47.09*** join/#asterisk lunk (n=lunk@negative-influence.com)
17:47.32lunkdo iax trunks support the dtmfmode option?
17:47.36salaudWell.... I'll try a few things...  Yate looks like an entire infrastructure to itself built on openh323
17:48.02salaudh3x: I thought there was an asterisk channel driver based on openh323 also
17:48.05SpaceBassanyone know of a way to mimic the Cisco VT video stuff on os x? IE call someone with a hard phone and have a video softphone start automaticallly?
17:48.18h3xthere is
17:48.22h3xthats the one im talking about :)
17:48.27h3xthe builtin one is a joke
17:49.06sonic69i am trying to dial with my polycom ip 4000 phone and it says ressource full!!! and when i am doing sip show peer in the CLI i see that the phone is connected
17:49.07salaudh3x: Ok... So the one in Yate and the openH323 based channel driver for asterisk are equivalent, in your mind?
17:49.58h3xwell the last i knew the openh323 stuff for asterisk was a dangling project
17:50.07salaudh3x: ah..
17:50.17Juggiebug jerjer.
17:50.26h3xjerjer is going to say
17:50.29h3x"dont use h323"
17:50.38Juggieprobally
17:50.44Juggiebut didnt he write it? or at least one implementation of it?
17:50.46h3xand then some explicitives about how you should go F your mother
17:50.51h3xyes
17:50.53*** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net)
17:51.16salaudh3x: This is where software "available" link goes on the sip323 implementation you put the link to.. http://www.sipquest.com/
17:51.18h3xhe wrote it so people could do h.323 with nufone
17:51.26Spy000007I thought jerjer was busy driving to Mexico to avoid paying his Nufone bills...
17:51.32salaudh3x: looks like it is commercial now... not 100% sure
17:51.51h3xavailable?
17:51.52h3xi clicked on software
17:51.53DaminJuggie: Not that I can see..
17:52.01h3xohyeah
17:52.04h3xTHAT software link
17:52.15JuggieDamin, answer me in the correct channel! :)
17:53.02salaudh3x: right... software link sends you to another page with an "available" link which goes to a commercial site... where it is unavailable
17:54.01anthmwe use woomera for h323 into asterisk sometimes
17:55.12h3xwell thats a good idea
17:55.18h3xi didnt know woomera did h323
17:55.43salaudh3x: is woomera another PBX?
17:55.54salaudanthm: is it easier to setup than Yate?
17:56.02h3xThe Woomera protocol, designed by Craig Southeren of OpenH323 fame, makes it possible to put your
17:56.06anthmwoomera is openh323 running as a daemon
17:56.07h3xhaha well that makes sense
17:56.16*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
17:56.39salaudanthm: cool...  does it translate to SIP?
17:56.46anthmin it's own process then it abstracts it down to a mega simple voice proto that asterisk can handle
17:57.14salaudanthm: how does it interface to asterisk?  via a channel module?
17:57.21anthmya chan_woomera
17:57.50salaudthat might be the most stripped down solution... can it do 12 channels?
17:57.50h3xyou could just use that asterisk fork project
17:58.13h3xfreepbx.org
17:58.16salaudh3x: I can't replace the existing asterisk infrastructure... well... I guess I COULD...but...
17:58.21Qwell[]h3x: openpbx..
17:58.26Qwell[]~freepbx
17:58.27jbotrumour has it, freepbx is NOT supported here!  People using it should join #freepbx (FreePBX is the new name of AMP)
17:58.32h3xoh
17:58.37Qwell[]~openpbx
17:58.38jboti guess openpbx is an asterisk fork without asterisk's limitations of using other GPLed code. see http://openpbx.org/ for more info, or join #openpbx
17:58.39dlynes_officeJuggie: yeah...he only wrote one implementation of it...the paid implementation (chan_ooh323c, the one in asterisk-addons)
17:58.47*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-125-116.red.bezeqint.net)
17:59.06anthmyah probably it can do 12 chans easily it's going to switch to opal lib sometime soon for now it does h323
17:59.13salaudwhere do you get chan_woomera?
17:59.28MACscrlol, how can something be a fork of a GPLed project and not use its license
17:59.32MACscrthat doesnt make sense
17:59.46*** join/#asterisk MatsK (i=MatsK@83.233.97.229)
17:59.48salaudha! found it
17:59.51Qwell[]salaud: pbxfreeware?
18:00.00salaudQwell[]: no chan_woomera
18:00.01*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
18:00.13Qwell[]...
18:00.24Qwell[]pbxfreeware
18:00.41coppiceMACscr: what doesn't make sense?
18:02.32anthmi think what they mean is they are not opposed to using gpl libraries where asterisk actually will not use any so they can still own it all but you would have to ask them to be sure
18:02.37salaudso... basically... H.323 sucks for asterisk
18:02.50*** join/#asterisk hohum (n=dcorbe@12.195.58.235)
18:03.09MACscrcoppice: i was thinking asterisk was gpl, thus anything fork of it, etc, would have to be gpl'ed as well
18:03.33Qwell[]MACscr: it is gpl
18:03.40coppiceMACser you completely miss the point. asterisk is not GPL, and is unable to use anything GPL
18:03.54anthmasterisk is gpl+cake+eat
18:04.06MACscrwhat license is asterisk then
18:04.11Qwell[]MACscr: gpl
18:04.16Qwell[]and commercial
18:04.16coppicegpl+something to make it totally incompatible with anything gpl
18:04.21rob0H.323 is an Equal Opportunity Suck. It sucks for all. :)
18:04.28anthmthe cake-and-eat-it-too license
18:04.30MACscrlol, coppice is confusing me
18:05.35MACscrlol, i guess so
18:05.35coppiceasterisk is GPL+something that means asterisk cannot use any GPL code
18:05.35salaudMaybe * has the "go F yourself telcos" license
18:05.35*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)
18:05.35salaudie... we are gonna get this job done
18:05.40coppiceso asterisk has a lousy SIP, a lousy RTP, a lousy almost everything as it all has to be built from scratch
18:06.10*** join/#asterisk mroth_imm (n=chatzill@63.65.26.220)
18:06.17*** join/#asterisk hads|home (n=hads@mail.nice.net.nz)
18:06.19salaudcoppice: I imagine those are fighting words... but I'm a lover not a fighter  ;)
18:06.33mroth_immcould anyone tell me what output i should expect from lspci for a Sangoma A200?
18:06.33anthmh.323 is actually the closest to perfect voip proto of them all it's just that ppl dont realize they need some of what it has to offer till later when they learn more
18:06.48*** part/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net)
18:07.09coppiceeven the developers will admit that things like chan_sip are horrible junk. its not exactly a point of contention :-)
18:07.36salaudanthm:  It seems more sophisticated... but like other sophisticated stuff it is more in the commercial domain... which makes it useless
18:07.43MACscrhttp://www.voip-info.org/wiki/view/Asterisk+GPL+Compliance
18:07.53MACscrcheck out the editline library part
18:08.16anthmit's a vicious circle really
18:08.20coppicesalaud: not really. SIP started simple but has now surpassed everything else for complexity, and without a clean design
18:08.30salaudcoppice: As a non-voip-developer, but fairly successful implementer.. I would say... it works
18:08.56coppicethey have finally realised that SIP is a disaster, and demand TCP but don't require UDP
18:09.17mroth_immsalaud, i've got a big single server installation (355 concurrent SIP calls yesterday) and i can tell you, asterisk is not exactly stable yet
18:09.24anthmyou start with telco circuit and think of a way to make it go over ip and not lose anything then you think, nah lets just get it to work, then you say oops i need that
18:10.01anthmit's like playing 7 degrees of separation to why h323 was not so crazy when they did it that way
18:10.11salaudanthm: Betamax
18:10.20salaudanthm: nuff said
18:10.22*** join/#asterisk hads|home (n=hads@mail.nice.net.nz)
18:10.23salaud;)
18:10.27coppiceSIP has still got things wrong. they changed from UDP to TCP at a time when  SCTP had become mature enough to use. SCTP is the way to communicate signalling
18:10.53anthmya and now they are saying "we should compress the signalling to a binary format" =D
18:10.59salaudmroth_imm: scalability is an issue for sure.. need multiple servers and that sucks
18:11.15rpmare there any polycom configuration tools out there?
18:11.24rpmfor ip-601s/501s?
18:11.29Qwell[]rpm: vi
18:11.35coppiceanthm: really? i hadn't heard that. i've heard suggestions of streaming XML
18:11.36rpmdirty.
18:11.41mroth_immit scaled fine...we were 35% to 45% idle...the problem is that that many calls shakes out bugs
18:11.51mroth_immit seems that sometimes chan_sip just goes away
18:11.53anthmyah they are looking for ways to compress it now
18:12.09anthmthey wanted text to begin with cos it's "easy to debug" =p
18:12.45salaudmroth_imm: maybe 35% idle is still too much....  asterisk definitely breaks down with high volume to one server... no doubt
18:12.50coppicethe text is easy to debug argument is really brain dead. text causes most of the damn bugs in the frist place :-)
18:13.36anthmyah especially with tethereal
18:13.48mroth_immnoone out there with a Sangoma A200...i just need to know how it identifies itself so i can compare it to my lspci output
18:13.52anthmumm it turs it into text for you
18:14.11dlynes_officemroth_imm: one sec
18:14.26coppicehow come ethereal has better protocol analysis than any of the packages people try to debug with it? :-)
18:14.43anthmyah no kidding
18:15.02Juggieethereal isnt called ethereal anymore
18:15.14dlynes_officeit's called wraith now?
18:15.23coppiceits called loan shark or something
18:15.49*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:16.01dlynes_officemroth_imm: if it's got a hwec on it, it looks like thus:
18:16.03dlynes_office00:08.0 Network controller: Unknown device 1923:0040
18:16.03dlynes_office<PROTECTED>
18:16.20mroth_immdlynes_office: many thanks dlynes
18:16.25anthmthey should make it into a lib you can link against to read protocols into your app
18:16.39mroth_immit does not show up on my 6850...and it has the added bonus of making my NIC invisible too :(
18:17.00coppiceso gradually all network communication will be promiscuous :-)
18:17.06dlynes_officemroth_imm: i would check to make sure your pci card is fully seated (after shutting your machine off, of course)
18:17.45Juggieethereal => wireshark
18:17.51mroth_immdlynes_office: i'll do it again, but i'm starting to think the Dell PEs just don't play nice with certain PCI devices
18:18.06mroth_immI couldn't get a Digium TDM400P to show up in it either...
18:18.15coppicei think loan shark would have been a better name :-)
18:18.18anthmanyway the moral is that h.323 is actually not as bad as asterisk makes it appear.
18:18.33dlynes_officemroth_imm: ah...I would tend to think it's possibly a bios configuration issue, then
18:18.42dlynes_officemroth_imm: are you using plug and pray support in your bios?
18:18.50coppiceH.323 sucks, but SIP sucks even more
18:18.51mroth_immi will start sticking paper clips into it and see if that helps :)
18:18.54De_Monanyone familiar with xtunnels.org?
18:19.02anthmthey all have to suck that is not negotiable
18:19.22Juggiefrom a developement standpoint, sip sucks alot less then h323
18:19.28mroth_immdlynes_office: i'll have to check that, any other ideas while i'm looking at the BIOS (should PaP be on or off)?
18:19.31anthmwe would run out of things to bitch about then and that would be no fun
18:19.43coppicewhy are they interested in compressing SIP signalling, when the media uses RTP :-)
18:19.50dlynes_officemroth_imm: i would turn it off, to see if that helps the situation
18:20.07mroth_immi'm running Fedora, and with most things kudzu just picks it up and adds it to the hwdb, but the telephony cards are causing premature balding for me
18:20.08dlynes_officemroth_imm: if you bios allows you to see what devices are configured in each slot, see if your bios sees the card
18:20.08*** join/#asterisk saftsack (n=saftsack@p54A7FFFD.dip.t-dialin.net)
18:20.11anthmimagine it: "good thing evey protocol is perfect all I have to do is sit here looking out the window ho hum..."
18:20.25*** join/#asterisk Vni (n=chatzill@adsl-69-235-247-151.dsl.irvnca.pacbell.net)
18:20.30mroth_immit does indeed and it shows that slot as empty...
18:20.56anthmcoppice, cos ppl like to get bothered by moot points
18:21.16coppicei really hate having to learn stuff to support 2 kids in the room next door
18:21.30xachenYou know... unloading MOH shouldn't have to crash asterisk
18:21.34anthmlike how much db performance you lose using odbc to connect asterisk to a db dispite the fact that if you top out odbc asterisk would have been dead back at the 10% mark
18:21.44*** join/#asterisk dsully (i=daniel@electricrain.com)
18:21.58coppiceanthm: you mean because its easier to play around with SIP, than to attack something containing DSP, right? :-)
18:22.01dsullyg'morning.
18:22.17VniI messed up my extensions.conf and I didnt back it up does somebody have one so I can fix mine.  Im running Fedora core 4 with astguiclient 1.1.11 and vicidialer???  Please my bosses is gonna fire me
18:22.25salaudH.323 stuff in yate (routing especially) looks a bit complicated :(
18:22.29dsullyany one have thoughts on why, after asterisk has been up for a while (1-2 days), I start seeing this on the console:
18:22.32dsullypri_dchannel: Ring requested on channel 0/1 already in use on span 1.
18:22.35Juggieanthm, are both openpbx & freeswitch being actively developed? or is all work being done on freeswitch now?
18:22.51dsullyand all inbound & outbound calls on the PRI stop?
18:23.11coppiceJuggie: well openpbx can now do T.38 :-)
18:23.19anthmcoppice, or then play with iax cos sip is too hard like the old lady who swallowed a fly
18:23.29jbalcombHow about this one? Our Polycom SoundStation conference phone attached to a GrandStream HT-386 decides to put on the hold music and then disconnect the call.
18:23.35Juggiecoppice, so can asterisk, at least passthrough.
18:23.41mroth_immdlynes_office: thank you for your help... :)
18:23.47Vni>I messed up my extensions.conf and I didnt back it up does somebody have one so I can fix mine. Im running Fedora core 4 with astguiclient 1.1.11 and vicidialer??? Please my bosses is gonna fire me .  No zaptel cards
18:23.59coppiceJuggie: yeah, but that's just the mickey mouse stuff :-)
18:24.23dlynes_officemroth_imm: it sounds like maybe that slot's defective
18:24.31anthmahh dont say that, i just spent 12 days with mickey mouse paying him money at every turn
18:24.32jbalcombVni: you can get one off the wiki or in the asterisk source directory.
18:24.33dlynes_officemroth_imm: have you tried throwing hte card into a different slot?
18:25.01*** join/#asterisk Katty (n=aisaacs@64.82.232.54)
18:25.04mroth_immdlynes_office: i will give it a shot
18:25.07anthmjuggie so ya, it appears openpbx is, I can attest freeswitch is based on my terminal full of code i am looking at
18:25.12dlynes_officeKatty: Meowwwrrr!
18:25.21salaudI appreciate everyone's help on the H.323 ... I'm going to take my best pass at Yate or Woomera and then it's off to a bunch of Sipura 2000's doing analog/digital and back to analog again into this f*'n altigen system as a final option
18:25.29Kattydlynes_office: heya
18:25.31coppiceanthm: I can see disneyland from here. that's as close as i want to get
18:25.36smackusok, so here is an ftp/polycom question for you. I have the ftp server set up to provision the phones, but they will not boot. here is my boot log for the phone http://pastebin.ca/73313
18:25.47smackuscan anyone make a suggestion?
18:25.59smackusi have played with permissions and such, no changes
18:26.16smackuswhich file exactly is the bootrom.ld
18:26.24smackusSIP.cfg?
18:26.33smackusno, that woulnt be it
18:26.48smackussip.ld
18:26.50smackusright?
18:27.02CunningPikesmackus: Do you have a bootrom.ld file?
18:27.10*** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6)
18:27.11*** join/#asterisk albertito (n=net@host199.201-252-24.telecom.net.ar)
18:27.11smackusnot named as such
18:27.13anthmturning h323 into sip is probably yate's best trick in thier list of bullet points
18:27.32CunningPikesmackus: You need two files: bootrom.ld is the bootrom and sip.ld is the SIP application
18:27.40anonymouz6663x0 :)
18:28.06smackusok, and it has to be named bootrom.ld?
18:28.09*** part/#asterisk skraelings001 (n=skraelin@201.230.140.95)
18:28.20Juggieanthm, this signup process to get into the tracker is 'special'
18:28.25Juggieto say the leasdt.
18:28.27Juggie*least.
18:28.43anthmfor what?
18:28.55Juggietrac.freeswitch.org
18:29.04jbalcombOur Polycom SoundStation conference phone attached to a GrandStream HT-386 decides to put on the hold music and then disconnect the call. Any ideas why or what direction to head?
18:29.24anthmit's unified all you do is sign up and you get an account for everything
18:29.35Juggieyeah iknow its jsut a weird signup process.
18:29.47smackushere are all of the files in the spip zip file, should there be more? http://pastebin.ca/73315
18:29.50anthmit's cos it's derived from our isp platform
18:29.54*** part/#asterisk dsully (i=daniel@electricrain.com)
18:29.54coppiceanthm: I thought it was to make people suspicious and go away :-)
18:29.55Juggieah.
18:30.03anthmyah that too
18:30.26Juggiegrr.
18:30.28CunningPikesmackus: Nope - you're missing your bootrom
18:30.29Juggiei'm impatient
18:30.33Juggieits slow sending me my email
18:31.02anthmyou dont even have to wait really
18:31.35Juggiewhat will my default pass be then?
18:32.21Vnianyone know how I would put IVR on outbound
18:33.11[TK]D-Fendersmackus : odds are you won't want to mess with your bootrom
18:33.41CunningPike[TK]D-Fender: Still needs one though :)
18:33.43[TK]D-Fenderjbalcomb : And I doubt your SS decided to go "on-hold" then hangup...
18:33.54[TK]D-FenderCunningPike : And the phone CAME with one...
18:34.06CunningPike[TK]D-Fender: You'd think........
18:34.09smackusok... well, at one point I had this working. I have all of my phones running 1.6.5, from the link i have from the wiki, I cannot get that version bootrom. Is there a better link?
18:34.15[TK]D-FenderCunningPike : I haven't seen a stock phone that can't run through 1.6.6 stock...
18:34.40[TK]D-Fender1.6.6 is the SIP version, not the bootrom version.  What are you running?
18:34.42CunningPikesmackus: You can use the bootrom from the Polycom web site
18:35.04smackusso 3.1.0?
18:35.04[TK]D-Fendersmackus : If you are running 2.6.1 or better, don't touch it.
18:35.08Vnianyone know how I would put IVR on outbound???
18:35.12CunningPikesmackus: Latest is 3.1.3
18:35.14[TK]D-Fendersmackus : Did you check on the phone direct?
18:35.22jbalcomb[TK]D-Fender: How do you mean? Are you suggested my users have done this?
18:35.33Juggieanthm, somethings broken, i still havnt got my email, unless it has to be approved or something.
18:35.46[TK]D-Fenderjbalcomb : Ask yourself how an analog phone puts a call on hold...
18:35.55[TK]D-Fenderjbalcomb : Esp on taht ATA
18:35.57smackusApplication, main: Label=BOOT, Version=2.6.1.0003 04-Dec-04 14:38
18:36.11*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
18:36.12[TK]D-Fendersmackus : thats your BR version and its fine.
18:36.14Juggie[TK]D-Fender, whats stopping analog ports from going on hold?
18:36.20Juggieif the call is bridged through *
18:36.32[TK]D-Fendersmackus : So whats the actual issue at this point?
18:36.34jbalcomb[TK]D-Fender: I am. Line noise? Misinterpretted signal? ATA thinks there was a *<hold> or <flash>?
18:36.51[TK]D-Fenderjbalcomb : Dunno... gotta wonder...
18:37.03smackusthe phone wont boot completely, it gets the error Failed to load bootrom.ld.  Check filename & FTP parameters and reboots
18:37.19[TK]D-FenderEW
18:37.22[TK]D-FenderNOT good.
18:37.24jbalcomb[TK]D-Fender: it's a conference room phone so it's not likely they are messing with it past dialing into the conference.
18:37.31CunningPikesmackus: Your issue is that if you are using FTP provisioning, the phone needs a bootrom.ld to check against
18:37.34[TK]D-FenderMight be "bricked" from a bad upgrade
18:37.39jbalcomb[TK]D-Fender I've heard the complaint before on other phones.
18:37.53smackusok, so back to the question... can I use bootrom 3.1.0?
18:38.00[TK]D-Fendersmackus : Go DL any bootrom equal or newer to the one you have and try to force upgrade it.
18:38.12[TK]D-Fendersmackus : What model are you running?
18:38.16jbalcomb[TK]D-Fender However, I am thinking it's a good opeurtunity to PO a Polycom SoundPoint IP conference phone
18:38.19smackus301 and 501
18:38.34[TK]D-Fendersmackus : Ok, sure... you should get 3.1.3 though as its more current.
18:38.40*** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154)
18:38.42gbodemantvhey all
18:38.52dlynes_officemroth_imm: anyways...i'm out for most of the day now...catch ya later
18:38.52gbodemantvI am in a bind
18:38.52[TK]D-FenderI should check to see if SIP 2.0 is full release now.
18:38.56rpmim running 3.1.0 heh
18:39.02gbodemantvI have been asked to impkement queue reports
18:39.03rpmi need to get a new bootrom and sip image
18:39.10jbalcomb[TK]D-Fender: Isn't there some way I can see if they hit the hold button or what the phone system thinks was going on with that extension?
18:39.12smackuswhere can i get the newer version? it will only let me dl old bersions
18:39.31gbodemantvI am wanting to use Queue Statitics from Asterisk Gurus but I am having a hell of a time with the implementation
18:39.37[TK]D-Fenderjbalcomb : "hold" on an analog phone introduces no signalling to the ATA.  That can't be it.
18:39.39CunningPikesmackus: What's the most recent they have?
18:39.40gbodemantvHave never install PHP or PostGres
18:39.41rpmyou gotta be a polycom sales partner or something.. i've been trying to get the images.. you gotta pay for them
18:39.54gbodemantvany guides or walkthroughs anyone knows of
18:39.57jbalcomb[TK]D-Fender That's what I was thinking because the hold music came on.
18:40.00[TK]D-Fenderrpm : No you don't....
18:40.11CunningPikerpm: You can get them for free from your reseller.......
18:40.15smackus3.1.0
18:40.18[TK]D-Fenderjbalcomb : hold on an analog phone only mutes everything
18:40.22CunningPikesmackus: Good enough
18:40.24jbalcomb[TK]D-Fender right
18:40.25smackusok
18:40.26*** join/#asterisk nortex (n=nortex@64.136.65.142)
18:40.45jbalcomb[TK]D-Fender so for the hold music to come on asterisk has to think something happened and i should be able to see it in the logs right?
18:40.47smackusjust stick it right in the directory with the sip.ld and such, right?
18:41.02[TK]D-Fenderjbalcomb : Perhaps... I don't know the logging system too well.
18:41.02jbalcombCunningPike rpm: you can also find them on the internet just like i did
18:41.23[TK]D-Fendersmackus : Yes, but you need to set up your <mac>.cfg file to point to them so it picks up.
18:41.53jbalcombsmackus: try 'freedomphones'
18:43.05jbalcomb[TK]D-Fender: ok, i've created /var/log/asterisk/rediculous.asterisk for tailing. It has all log levels going.
18:43.41jbalcomb[TK]D-Fender: it even shows DTMF events..
18:43.56[TK]D-FenderWWF Events? pr0n PPV events?
18:44.09*** join/#asterisk MatsK (i=MatsK@83.233.97.229)
18:44.57jbalcomb[TK]D-Fender: Yes, you can monitor the World Wildlife Fund and PayPerView pr0ns. Asterisk is rather multifaceted<SP>.
18:45.56[TK]D-FenderWWE....
18:45.59[TK]D-Fenderclose enough
18:46.11gbodemantvanybody using Queue Statistics and Asterisk?
18:46.52[TK]D-Fendergbodemantv : I do on my GUI package...
18:50.13gbodemantvhave been told I have a week to implement queues and reporting
18:50.19gbodemantvgot realtime ques working
18:50.28gbodemantvbut need to get reporting up and running
18:50.38[TK]D-Fendergbodemantv : Reporting has several packages out there already.
18:50.40gbodemantvAsterisk Stats from asterisk gurus looks perfect
18:50.58gbodemantvbut their install is not at all specific
18:51.05gbodemantvI am having a hell of a time
18:51.20gbodemantvcan't even get postgres off the ground
18:51.44gbodemantvtrying to get ahold of them to implement it
18:51.50gbodemantvbut no one has responded
18:51.54jbalcombgbodemantv: QueueMetrics is decent
18:52.15gbodemantvdon't need that much info
18:52.26jbalcombgbodemantv: Asterisk CDR is good for the reporting
18:52.47gbodemantvhow difficult is it to implement
18:52.52jbalcombgbodemantv: Maybe you should contract [TK]D-Fender to get it going for you
18:53.18*** join/#asterisk backblue (n=moo@87-196-47-160.net.novis.pt)
18:53.19gbodemantvif he is interested I might
18:53.31jbalcombgbodemantv: I would say not difficult because I got it working.
18:53.55jbalcombgbodemantv: you should /msg him. =)
18:54.22jbalcombgbodemantv: I've contracted him and he is effective and friendly.
18:55.35gbodemantvjbalcomb: does it report realtime queues?
18:55.58jbalcombgbodemantv: it reports every call thats been hung up and accounted for if that what you mean.
18:56.08*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
18:56.29gbodemantvhttp://www.asteriskguru.com/tools/queue_stats.php
18:56.34gbodemantvthis is what i am trying to use
18:56.38jbalcombgbodemantv: there maybe a matter of adjusting some line in the source and recompiling so that additional information is logged.
18:56.52gbodemantvand I know that it will work
18:57.02gbodemantvjust cant get it all to come together
18:57.02jbalcombgbodemantv: We added some system calls to echo lines to the log as well.
18:57.32*** join/#asterisk Johnnie (i=odysseus@pdpc/supporter/active/Johnnie)
18:58.31jbalcombgbodemantv: Ah, ok. Asterisk CDR is not the comprable product I think. I beleive what we are doing with QueueMetrics is what your looking to do with Asteriskguru Queue Statistics
18:59.13drraycan I axe why you don't just sed/awk/perl a solution?
18:59.24jbalcombQueueMetrics has a 30 or 90 dial trial and is fairly easy to set up. The only troubling matter is setting up tomcat for the java machine.
18:59.36[TK]D-Fender"Too much trouble"
18:59.43jbalcombdrray: "too much trouble"
19:00.07jbalcombman, I gots to take typing course.. too slow.
19:00.07*** join/#asterisk oej (n=oej@212.17.152.81)
19:00.15drraythat sounds a lot like people who use asterisk at home, because asterisk is "too much trouble" but hey, roll on
19:00.44jbalcombdrray: one can only be so /into/ maintain custom code and systems.
19:01.19jbalcombdrray: I'm putting together a IP phone management system right now and I very much dread the maintanence of it.
19:01.21*** join/#asterisk oej_ (n=olle@212.17.152.81)
19:01.21drraythat's a fair point, and I'd be on your side if I did not think that the turn-key solutions lacked something
19:01.28drray:)
19:01.40drrayI hear you
19:03.03jbalcombdrray: I would think the thing to do is /fix/ the turn-key solution and submit ones code changes for addition
19:03.46drrayI've always found it easier to roll my own than fix what I don't like about someone elses
19:04.00drraybut I wrote my own crappy pvr because I did not like mythtv crashing
19:04.56*** join/#asterisk InfraRed (n=subhi@bb-87-81-46-122.ukonline.co.uk)
19:05.07*** join/#asterisk Monkey13 (n=jcheuvro@69.7.217.140)
19:05.23jbalcombAnyone have opinions on the Polycom SoundStation IP 3000?
19:06.14*** join/#asterisk NullC (n=greg@wikimedia/Mindspillage)
19:06.14*** part/#asterisk mog (i=ejabberd@68.62.237.103)
19:06.17InfraRedhi all
19:06.25InfraRedthanks for the help earlier with the nat
19:06.43*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
19:06.55NullCI have a pair of asterisks servers, IAX 2 trunk between them. It works fine, except it doesn't truck (put multiple calls in one packet) in ONE DIRECTION.. one way trunks, the other doesn't. Help!
19:07.10ManxPowerI didn't realize that wasps can nest inside a LNB housing...until today.
19:07.31drraywasps can nest anywhere
19:07.36drrayeven inside of beehives
19:07.41ptinsleyManxPower, i had the same thing on my old one dirt dobbers I think it was I had tough
19:07.44ptinsleythough
19:07.52InfraRednow i am having stream issue, during a call i can 'transmit' but not receive. the asterisk box is behind nat in dmz, the phone is behind another nat behind the asterisk box. any ideas?
19:07.57*** join/#asterisk mog (n=mogorman@gateway.digium.com)
19:08.00ManxPowerNullC, sounds like 1) you don't have zaptel timing on one of the two servers or 2) the incoming connection does not correctly match a user/friend
19:08.04drrayI ran a mile one time when they infested my tool box
19:08.17InfraRedwhat i mean by transmit is voice, they can hear me when i can, but i can't hear them
19:08.26InfraRedwhen i call
19:08.29ptinsleydrray, lol
19:08.42ManxPowerMy aunt and my brother are both allergic.  Apparently I'm not yet
19:09.01*** join/#asterisk jgoo (n=jgoo@ppp129-197.adsl.forthnet.gr)
19:09.15ptinsleymy dad almost died when he was a kid by mowing over a hole full of the little bastards
19:09.30NullCHmm ... Ztdummy is on both...
19:09.45ManxPowerNullC, but is it LOADED?
19:10.29jgoohello all, I am about to partake in the joy that is bristuff install for a card, and i have come up against loveable terms such as NT and TE, which i kind of undertstand...
19:11.01jgoothe Install docs talks about the script downloading and patching the asterisk build from cvs ... is the cvs-ftp server still running on digium_
19:11.03*** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)
19:11.19jgoothe card is a HFC single port BRI
19:11.28ManxPowercvs has been turned off, svn has been used for a while.
19:11.53jgoothat was my specific question - how can I test without breaking things if this install is ok?? voip wiki was a little out of date I fear
19:12.01jgoohas anyone recently setu pone o fthese babies-
19:12.13jgoo(new keyboard >.< )
19:12.14rob0Maybe that CVS - SVN thing should be in /topic
19:12.31drrayit's been a year?
19:12.35rob0Yes, I have set up new keyboards before. :)
19:12.42rob0wow, a year?
19:12.55rob0Seems like several questions every day here.
19:13.16jgoo=] it is a swedish keyboard, and I am not swedish, so it seems most illogical, but then, so do most swedish people ÖP
19:14.07jgooso, setting up a BRI card ... the pstn digium was easy enough, but this is an unbranded (rebranded as crypto) hfc card
19:14.59jgooI have 3 concerns: 1) it is plugged into a S0 port on the ISDN box. is that normal?
19:15.23[TK]D-Fenderjbalcomb : Strangely I would suggest the IP version.. go for the SS 2 W (wireless) on an ATA... Works great and not locked to any system.
19:15.32jgoo2) the junghanns install for this bristuff, is it the only way forward for these hfc cards? and if so is there updated info =]
19:15.51NullCManxPower So changing my extensions.conf so that user:pass@hostname rather than user:pass@ipaddr  fixed it.
19:16.38florzjgoo: What is an "ISDN box"?
19:17.12florzjgoo: And no, you could also use misdn, capi or i4l
19:17.14jgoogood question, I am an isdn virgin here (except that one time in germany, but that doesn't count I was drunk)
19:17.31jgooflorz - aha, you are famous, i read something about your patch
19:17.36jgoothat was my 3rd concern, florz
19:17.39jbalcomb[TK]D-Fender: well, if i don't go IP there'd be no point. i think i'll atleast switch it to my sipura ATA rather than the ht-386
19:17.42jgoo=]
19:18.27jbalcomb[TK]D-Fender I was just thinking the IP phone would behave better than the ATA
19:19.48florzjgoo: As far as that bristuff-stuff is concerned: instead of using the download/build script you just as well can apply the patches manually - the script doesn't so all that much anyway ...
19:20.01florzs/so/do/
19:21.47[TK]D-Fenderjbalcomb : I would sooner think its a GS device at fault than a Polycom one :)
19:23.21kristalinois it possible to text to speech in french ?
19:23.30*** part/#asterisk NullC (n=greg@wikimedia/Mindspillage)
19:24.08*** join/#asterisk crich1999 (n=crich@port-212-202-198-145.dynamic.qsc.de)
19:26.09ptinsley[TK]D-Fender, i second that one, GS == bugggggy
19:27.11*** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl)
19:29.07smackusjust a general question.. I do not want to try to make this work now, but can i dial an agent login like an extension?
19:29.10smackusis that even possible
19:30.25SplasPoodanyone know teliax's inbound rate for 800 DIDs?
19:30.53*** join/#asterisk Nodren (n=nodren@adsl-75-8-201-246.dsl.frs2ca.sbcglobal.net)
19:30.57Nodren~centosbug
19:30.58jboti guess centosbug is a problem with the latest Centos kernels (4.2 and 4.3).  To fix it, paste everything inside the quotes into a root shell:  "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h"
19:31.17SplasPoodkristalino: Cepstral supports french
19:31.23SplasPoodkristalino: I'm pretty sure
19:32.12jgooflorz - do you know of any good steps to follow for this patches so I don't miss one?
19:32.36kristalinoSplasPood,what is cepstral ?
19:32.47SplasPoodkristalino: TTS system..
19:33.32*** join/#asterisk dangerarea (n=kevin@kingfisherwalk.demon.co.uk)
19:34.07florzjgoo: The script that's in the bristuff package :-)
19:34.08kristalinook, thanks
19:34.33dangerareaevening
19:34.41*** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal)
19:35.40dangerareaanyone know the rough spec of the standard HP proliant server these days?
19:35.55dangerareamore to the point how many simultanious calls it could handle with asterisk
19:36.11vader--anyone have any issues where their sip phones won't do anything in the features.conf?
19:36.13vader--like *1 for recording and stuff
19:36.14vader--i have cisco 7940G phones
19:36.21InfraReddangerarea: there is a wiki page about 'dimentioning an asterisk server'
19:36.30dangerareaooh, ta InfraRed
19:36.46InfraRednp
19:38.25*** part/#asterisk NotJohnDavid (i=dave@c-68-47-199-178.hsd1.tn.comcast.net)
19:38.55InfraRedi have a problem with asterisk and nat, i have adsl router with nat, asterisk is set as DMZ in the nat, the phones are behind asterisk on another nat (2 nats), the asterisk server also acts as nat server for another subnet seperate from the phones, when i make a phonecall, the call receiver can hear me but i cannot hear anything, any ideas? i have externip in each phone config and nat=1 too
19:39.06[TK]D-Fendervader-- : Your Dial statements all enable those features? "wWtT", etc?
19:39.38InfraRedi'm using sip btw
19:39.42[TK]D-FenderInfraRed : Externip belongs in [general], not in the phone config
19:39.52[TK]D-FenderInfraRed : And did you set up your localnet clauses?
19:39.58InfraRedi have it in general
19:40.08[TK]D-FenderInfraRed : pastebin your sip.conf
19:40.11[TK]D-Fender~pb
19:40.22jbotmethinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
19:40.22InfraRedok sec
19:40.55*** join/#asterisk iq (n=iq@unaffiliated/iq)
19:41.39InfraRedis the localnet for the phones subnet ?
19:41.40jbalcomb[TK]D-Fender I would agree. I've gotten us down to just two HT-386s. If I get around to switch the conf phone to the SPA-2000 that'll drop us to one.
19:42.03jbalcomb[TK]D-Fender I've been getting a broken PB all day
19:42.27*** join/#asterisk mog (n=mogorman@gateway.digium.com)
19:42.55*** part/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl)
19:43.35sparkleytoneanyone ever seen an issue where every time you reboot an asterisk machine the /var/run/asterisk has been deleted?
19:43.43iqHi
19:44.33InfraRed[TK]D-Fender: localnet was what missing
19:44.50InfraRedthanks for your help :)
19:45.03sparkleytonei keep having to recreate the directory...
19:45.37InfraRedsparkleytone: i think /var/run and /tmp get cleared on every reboot
19:45.45sparkleytonehrrrm
19:46.02InfraRedtry setting the pid file to file inside /var/run rather than /var/run/asterisk/
19:47.21*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
19:47.36[TK]D-Fenderjbalcomb : PB?
19:47.43[TK]D-FenderInfraRed : ywc
19:47.56sparkleytonedoes this if statement work for directories?  `if [ ! -e "/var/run/asterisk" ]`
19:48.12sparkleytoneas much as i can admin a box, i am allergic to scripting
19:48.14sparkleytoneand coding
19:48.21sparkleytonewill that return properly?
19:48.39jbalcomb[TK]D-Fender PasteBin
19:49.21lunkhas anyone else experienced difficulty with DTMF with VoipJet?
19:49.50lunkand/or can anyone recommend a good, cheap, bulk termination service
19:49.59lunksip preferrably..
19:51.07Qwell[]sparkleytone: -d
19:51.35sparkleytonethx Qwell
19:51.36wese103sparkleytone: -e will tell you that the file or directory exists, but will not discern between the two.
19:51.52drrayI don't think good and cheap go together on that
19:52.22*** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
19:54.08dangerarealunk: i'm looking for one too
19:55.10dangerareai'm looking for someone to terminate 1000 simultanious calls if anyone knows of anyone
19:55.41drrayyour local telco wont do that for you?
19:56.11*** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com)
19:56.24InfraReddangerarea: depends on your location
19:56.32dangerarealondon
19:56.53InfraReddangerarea: magrathea(they rock), gradwell
19:56.54*** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
19:57.21dangerareawell they might but I'm looking for a price for someone to terminate them to one non-geo number deliver them over SIP/AIX/IAX2 and then us to do IVR
19:57.47InfraRedmagrathea or gradwell
19:57.52dangerareahence trying to find out about maximum call handling
19:58.12dangerareanot looked at magrathea, i shall do now
19:58.16dangerareacheers again InfraRed
19:58.20InfraRedtheir website is shite
19:58.31dangerareaas all good companies should be
19:58.49dangerareait means they have techies that know what they're doing
19:58.49InfraRedbut they deliver one of the best voip service sin the uk
19:59.00dangerarearather than building websites
19:59.06dangerareaall techies hate websites
19:59.08dangerarea:)
19:59.15InfraRedyep
19:59.25InfraRedthey can deliver geo and non-geo number
19:59.26InfraReds
19:59.57dangerareacool
20:00.14InfraRedof any area code in the uk also port number from pstn to voip
20:00.27dangerareai've had a look at that wiki page on simultanious calls
20:00.28InfraRedwhich is very handy
20:00.41dangerareabut still not really any the wiser
20:01.12*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
20:01.47InfraRedwell
20:01.54InfraRedare you doing pure voip ?
20:02.10dangerareahere's the full brief...
20:02.12dangerarea1000 calls in
20:02.16dangerareaivr handles them
20:02.29dangerareasay 30 out to another phone system
20:02.44dangerarea(avaya ip office)
20:02.52InfraReddo you have the bandwidth ?
20:03.24dangerareanot at the mo but we have tonnes of fibre in to our building
20:03.46InfraRed1000 x 80Kbps per call
20:03.46dangerareacan be done at 30 mins notice
20:03.47InfraRed~ 80Mbps
20:03.50drraysee if they will coloc your asterisk box
20:03.59dangerareareally 80k per call?
20:04.06InfraReddepends on the codec
20:04.14InfraRedi use 711u
20:04.20InfraRedsince i find 729 is shite
20:04.50dangerareawell since most of the network uses 711u/711a we'd be using that
20:04.59InfraRedsorry, 711a
20:06.22dangerareayep
20:08.27*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)
20:09.52terrapenso, is there a way to configure password-less agents?
20:10.10terrapeni'm trying to ring all phones in this store when a call comes in and I don't want anyone to have to log onto the queue, ever
20:10.21Vorondilokay, just out of curiosity: since 711u is often refered to as u-law, which is pronounced "mu law" (the greek letter), is 711a pronounced "alpha law" or "a law"?
20:10.27CunningPiketerrapen: We use members instead: member => SIP/1234
20:10.43jbalcombHrmm.. the Polycom SoundStation IP 4000 is $200.00 USD cheaper that the IP 3000.
20:10.55terrapencunningpike, thanks
20:11.05jbalcombVorondil no one says mu-law
20:11.30Eonzguys is possible record the calls with asterisk
20:11.45jbalcombterrapen have you tried the meetme hack from the wiki?
20:11.54jbalcombEonz yep, check the wiki for details
20:12.09InfraRedterrapen: you dont need agents
20:12.12[TK]D-Fenderjbalcomb : Stay analog, go WIRELESS ;)
20:12.13InfraRedjust use extensions
20:12.24Eonzjbalcomb: k thanks
20:12.25InfraRedthen google the wiki for extensions configuration
20:12.31jbalcomb[TK]D-Fender strange that its not the ip phone thats wirelss
20:12.34InfraRedyou can have a ring all facility
20:12.57jbalcomb[TK]D-Fender i do want the linksts wifi sip phone for myself though
20:13.10jbalcomb[TK]D-Fender i gotta finish revamping our wifi network first though
20:15.10jbalcomb[TK]D-Fender I finished my perl script that reads the arp.dat from ArpWatch and put the MAC and IP in a MySQL DB!!
20:15.24Vorondiljbalcomb: nobody? i guess i'm just weird then.  i mean, it *is* the greek letter mu, right?
20:15.56wese103The ITU-T G.711 spec actually has the greek letter mu in it.
20:15.57jbalcombVorondil: You are likely a bit /unique/ and I have no idea if its the greek letter mu.
20:16.06wese103And for alaw, it uses simply an 'a'.
20:16.12wese103Well, an 'A'.
20:16.18*** join/#asterisk mtaht4 (n=m@adsl-75-10-213-145.dsl.pltn13.sbcglobal.net)
20:16.37jbalcombits "yew-law" and "aye-law" that i use and hear used.
20:17.00wese103We've always said "mu" and "aye" here.
20:17.11*** join/#asterisk japerry (n=japerry@216.231.51.208)
20:17.31japerryanyone here well inveloped within Asterisk, and lives in the seattle area?
20:17.37drrayI do
20:17.39jbalcombI'm not the autority on anything though cause people throw things at me something because i say "ga-nu" for GNU.
20:18.09jbalcombTom Robbins lives in Seattle.
20:18.27japerrynice nice.. we're looking for someone to consult our asterisk system
20:18.55wese103Yea, I actually try to avoid verbalizing GNU.  :)
20:19.09jbalcombjaperry: I'll take the cheap flight if'n need you can't find anyone else.
20:19.13drraywese103 - How do you say voip?
20:19.26japerryjbalcomb: thanks
20:19.51japerryjbalcomb: checking local refs first, I'll keep you in mind though if we can't :-)
20:19.56jbalcombdrray Are you looking at "voyp" rather than "v-o-i-p"?
20:19.56MACscrv-oy-p
20:19.59wese103I would incline to say it with an "oy" sound.  But I work with a lot of Germans who like to say it with an "oh" sound.
20:20.12drrayvoyp annoys me
20:20.24MACscrto bad its the correct way
20:20.25InfraRedi saw vooopah
20:20.26wese103I find myself mostly saying the entire "voice-over-ip" phrase instead.
20:20.33drrayI say voice over ip
20:20.45jbalcombis the e starting ethernet long or short? I hear short and I think they sound like a hick.
20:20.55rob0I say toe-may-toe.
20:21.00jbalcombagreed
20:21.13jbalcombthough its almost more ta-may-toe
20:21.15*** part/#asterisk acrg (n=aragon@decoder.geek.sh)
20:21.20*** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk)
20:22.18wese103I get poked a little bit for my pronounciations.  It makes life a little more colorful.
20:22.27rob0I'm really uncomfortable with how I say [un]comfortable. I leave out a syllable. I have a large syllabic deficit by now.
20:22.42jbalcombOf course, I should note that I'm from Cleveland, OH which is considered to nearly be the heart of what is known as proper pronunciation of US/Microsoft English.
20:22.59rob0nonono Walter Cronkite is from Nebraska!
20:23.13rob0And that's the way it is.
20:23.42jbalcombWalter Cronkite.. feh. That guy talks like somebody who needs a punch in the face.
20:23.59*** join/#asterisk pa (n=paolo@unaffiliated/pa)
20:24.00wese103I work with a guy from India, and his English is impeccibly the Queen's.  We have long discussions comparing my American with his proper English.
20:24.36jbalcombwese103: My Japanese Sensei is from India as well and we have similar amusements.
20:24.58jbalcombwese103: The best so far has been the Japanese word 'mafura
20:25.13wese103meaning?
20:25.31jbalcombwhich is the English word Muffler but really it's the British word for scarf as opposed to the US English sound reduction device under most vehicles.
20:25.54wese103Heh.  :)
20:26.12wese103I had a similar miscommunication with a German coworker about the workd "beamer".
20:26.17wese103He said the boss agreed to get us a "beamer".
20:26.23wese103I was floored.
20:26.26wese103"Wow!"
20:26.28jbalcomband you were like HELL YEAH!
20:26.39wese103Turns out he meant an overhead projector to connect to a VGA port.
20:26.57jbalcombhaha... oh man, now that is the def. of disappointment.
20:26.58wese103It was rather anticlimactic.
20:28.00wese103Ok, back on topic... anybody here work with Digium's TE110P card?
20:28.23NuggetIt's weird to see someone say "def." to obscure the fact that they can't spell "definition."  Usually people say "def." because they're embarassed by not being able to spell "definitely"
20:28.52*** join/#asterisk giesen (i=giesen@dirtypackets.net)
20:28.55CunningPikewes103: We have one - what's up?
20:29.24wes103Is there any way I can verify clocking source?
20:29.28CunningPikejaperry: I'm in Vancouver, BC - so is dlynes_office
20:29.35giesenIs there a way in asterisk queues to have asterisk prompt the agent receiving the call telling them they have to hit # to accept the call
20:29.35wes103Any way to poll it and see how many slip errors it might be getting?
20:30.18wes103I have an E1 link, and I have set both ends so that neither is clock source.  I expect to see some errors, but don't know where to look.
20:30.22*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
20:30.31CunningPikewes103: Your CLI (-vvv) will usually show HDLC errors if you are getting timing errors - what makes you think you are?
20:30.40wes103Oh, nothing.
20:30.44wes103I am just testing out the card.
20:30.54wes103Haven't gotten asterisk fully configured yet.  Just zaptel drivers.
20:31.06wes103Trying to connect it to an IMACS channel bank.
20:31.26wes103No alarms, so I have framing and carrier.
20:31.35CunningPikewes103: Ah - you may see HDLC errors then when you get asterisk running, unless there is a timing source on the PRI
20:31.43wes103This is CAS.
20:31.58*** join/#asterisk mindwarp (i=mindwarp@silenceisdefeat.org)
20:32.45wes103Well, I will wait until I get the entire system up, and then see if I "need to worry about it".
20:32.47wes103:)
20:33.02CunningPikewes103: Good plan :D
20:33.47wes103But, in zttool, no matter how I set the timing source, it always shows sync source as internal.
20:34.04ptinsleyok guys, my problem with my pri has come back, even after changing the trunk to descending as I assumed it would
20:34.16ptinsleyis there ANYTHING i should run on it before i restart asterisk to get debug info
20:34.26ptinsleyoutbound calls work
20:34.29ptinsleyall inbound calls fail
20:35.17justinu|laptopwes103: i think that's a very irritating known bug
20:35.23wes103Ah.
20:35.26ptinsleyJun 27 15:35:17 WARNING[17647]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/13 already in use on span 1.  Hanging up owner.
20:35.27ptinsleyJun 27 15:35:17 WARNING[17647]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/14 already in use on span 1.  Hanging up owner.
20:35.32wes103I tried looking in google for it, but didn't find anything.
20:35.45wes103I will let it rest for now.
20:35.51wes103Thanks.
20:36.48CunningPikewes103: It may be because there isn't a timing source on the line, so it's failing back to internal
20:37.19wes103Shouldn't there be a message somewhere to that affect though?
20:37.25ptinsleyoh well, restarted, everything works again, till next time :/
20:37.35wes103Switching the source would cause alarms in a Telco.
20:38.01*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
20:38.14trelane_are there any module options for wctdm24xxp?  I need to steer it to a different IRQ
20:38.48justinu|laptopno, you'll have to move the card
20:39.05CunningPikewes103: You know - I don't know that there is......
20:39.15wes103ok
20:39.23CunningPikewes103: Your telco should be the timing source......
20:39.54wes103I have a good source coming into the IMACS from the primary E1.  I have the secondary E1 going to this TE110P card in a testing system.
20:40.09justinu|laptopthe timing source is either the inbound framed DS1 signal from your telco, or the internal oscillator
20:40.10wes103I guess I need to find out if the IMACS is passing it through.
20:40.26wes103Hence, my wondering about slip error counts.  :)
20:40.46CunningPikewes103: Yup - doesn't sound like it is right now.....
20:42.10wes103Well, troubleshooting that is out of scope here.  It sounds like zttool and /proc/zaptel won't report slip errors.  So I have to get asterisk configured to see how it sounds.
20:42.38wes103Time to bolt.
20:42.50vader--so once a user records a call where does it go? and how can they retrieve it?
20:46.28*** join/#asterisk ncef (n=cef@38.119.128.203)
20:46.38*** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
20:46.44*** part/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
20:46.53jbalcombvader--: Have your set up the Asterisk ARI?
20:47.04InfraRedARI?
20:47.22Bullseye_Networkwhen doing a show channel Zap/whatever there are some level1 CDR Variables listed can I call them in the dial plan like ${dstchannel} or ${lastdata} I didnt see it in the wiki
20:47.25jbalcombAsterisk Recording Interface; lets people retrieve voicemail and call recordings.
20:48.30luke-jr_Anyone know a good VoIP provider similar to SellVoIP (except with actual support)?
20:48.33jbalcombBullseye_Network: not sure if you can do it natively but if not you can probably make a system call to a script that returns a value to use
20:49.24vader--na i haven't set that up
20:49.28vader--i have voicemail setup
20:49.29vader--but not that
20:49.32Spy000007luke-jr_: try connect.voicepulse.com
20:49.51jbalcombI have the MAC addresses and IP addresses of all my IP phones. Now I need to get the extension and username.
20:50.02jbalcombWhat is a good way to look that up?
20:50.26InfraRedsnmp
20:50.27InfraRed:)
20:50.55jbalcombI'm thinking I could pull the asterisk db into an array, search for the IP, and regex the extension out. Once I have the extension I can grep the sip.conf to get the username.
20:51.43InfraRed3-1 to france
20:51.48jbalcombvs. ?
20:52.08InfraRedspain
20:52.22luke-jr_Spy000007: per-minute outgoing looks nice, but that's about it :p
20:52.23*** join/#asterisk zotz (n=zotz@24.244.133.115)
20:52.26InfraRedthey'll play brazil next
20:52.32Bullseye_Networkjbalcomb: think I found it ${CDR(<name>)} does that look familiar?
20:52.50Spy000007luke-jr_: You asked for an ITSP with actual support
20:52.58Spy000007luke-jr_: Or were you looking to get it for free?
20:54.21luke-jr_Spy000007: similar to SellVoIP
20:54.26jbalcombBullseye_Network: I haven't used it but its listed on the wiki.
20:54.26jbalcombTurns out you can't use Set(CDR(<name>)=value) for anything but userfield and accountcode.
20:54.26jbalcombThese fields are read-only.
20:54.36Spy000007luke-jr_: ok, good luck
20:54.43luke-jr_Spy000007: I was looking for per-minute incoming w/ DIDs themselves being cheap
20:54.45jbalcombhttp://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr
20:55.37vader--ARI seems like it's something for AMP or ASterisk@home
20:56.08Bullseye_Networkjbalcomb: I only need to read them
20:56.23jbalcombvader--: noe
20:56.35jbalcombBullseye_Network: I imagine that means you're all set then
20:58.36vader--is ARI built into asterisk default package?
20:59.28nortexvader--, ARI is from littlejohnconsulting.com and is part of the FreePBX package, but can be run on any Asterisk install.
20:59.44vader--gotcha
20:59.52vader--there is no native way to playback these files through asterisk?
21:00.04vader--or have them emailed to the user when the call monitoring is finished
21:00.47*** join/#asterisk speedwagon (n=Ariel@70.46.87.158)
21:02.09ariel_I am trying to setup a callerID change but it's not working can you guys look at this and help me see what I am doing wrong? http://pastebin.ca/73438
21:02.27*** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com)
21:04.51*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
21:05.11directoryariel_: ${CALLERID(number)}
21:05.18*** join/#asterisk marv[work] (n=timr@64.89.118.139)
21:05.25directorythat's if you want to get the callerid number...
21:06.44*** join/#asterisk brijn (n=brijnier@204.244.176.116.net-conex.com)
21:06.59*** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
21:08.04jbalcombvader-- not that i know of. you may try google and/or the wiki.
21:08.34jbalcombIs there really no one doing any sort of IP phone system management?
21:09.32jbalcombI've only got 150 phones and I'm sick of maintaining the phones and users. Surely someone must have done something...
21:12.11ariel_directory, thanks
21:19.06*** join/#asterisk electus (i=electus@113.129.8.217.in-addr.arpa)
21:19.34electusAnyone here that has some experience with the Originate command in asterisk API?
21:19.39electusgot some problems
21:20.06electusWARNING[21212]: pbx.c:2353 __ast_pbx_run: Channel 'SIP/62-d5d9' sent into invalid extension 's' in context 'default', but no invalid handler
21:20.39electuswhen I try to call a number with my intern number 62
21:21.13*** join/#asterisk colinm_ (n=colol@VDSL-130-13-11-67.PHNX.QWEST.NET)
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21:23.56electusI got one context I use wich is 'internt'
21:24.29electusanyone?
21:24.56*** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net)
21:26.02electusgot a very basic config, so maybe Im missing something
21:26.03Drukenanyone want my headache?
21:26.15FuriousGeorgeanyone using snom 360's notice that when they either assign a park orbit button (may only work with a patch) or transfer to the park extension, the user doesnt hear what "spot" the call was parked on?
21:28.01FuriousGeorge~seen shmaltz
21:28.18jbotshmaltz <n=mybox@mail.dmaven.com> was last seen on IRC in channel #asterisk, 22h 51m 36s ago, saying: 'anybody seen any problems with queus when the members are Sip/sipaccount?'.
21:28.37*** part/#asterisk MACscr (n=MACscr@66.73.154.70)
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21:30.55mountainm2kanybody heard of Advantatech ?  Polycom referred me to them...
21:33.35mountainm2kquiet today...
21:38.01FuriousGeorgeanyone using snoms notice users dont hear what extensiona  call is parked on?
21:38.22Strom_CFuriousGeorge: that will happen with any phone if you do a blind transfer
21:39.05CunningPikejbalcomb: Give me ssh access and I'll do it for you :)
21:39.57*** join/#asterisk benno|b1 (n=ben@88-96-30-86.dsl.zen.co.uk)
21:40.19CunningPikeDruken: What's your headache, apart from being named after an NHL player who never realized his full potential?
21:40.29*** join/#asterisk viler (i=1000@200.114.70.228)
21:41.02FuriousGeorgeStrom_C: i set uyp a park orbit button, but lemme try attended.  if i try to call the extension first the phone says "address incomplete" and the CLI says nothing.  im pretty sure this happened before i applied the metermaid patch to ge the LEDs going
21:41.41FuriousGeorgein fact i applied the patch because parking was not telling me the number as it would not let me call the park extension directly
21:41.49FuriousGeorgefor atxfer
21:42.35Strom_C*shrug* I've used call parking with attented transfers just fine without any patches
21:47.41nortexStrom_C, We do a blind transfer to parking, but it is a PBX transfer with the # key.
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22:01.03nortexI have seen a little bit on this, but I have a dumb question about Sangoma Echo cancelation cards, I have a A104D and yet there are still times that there is echo on the calls, but only on my side. Should there be any echo with a hardware echo cancelling card?
22:01.19Qwell[]nortex: there can be, sure
22:02.02nortexQwell, Is there anything I can do about it?
22:03.39*** part/#asterisk tlow (n=tlowe@bgp.terrorist.net)
22:06.38mindwarpHi, I have a problem with my SPA3000 apparently not passing inbound calls to Asterisk. I have a phone connected to the Sipura's "phone" port, and the "line" port is connected to PSTN. When PSTN rings, the phone connected to the Sipura rings, but Asterisk shows no signs of "seeing" the call (no activity in the CLI with extreme verbosity turned on). I did configure the PSTN-to-VOIP bridge on the Sipura. Does anybody know what may be causing this is
22:07.28[hC]zaptel suddenly doesnt want to compile... any ideas why i'd be getting this? http://pastebin.ca/73487
22:07.39[hC]seems like a libc error?
22:09.33benjamin7062Can anyone give me a real life benchmark on recorded calls... does a decent powered machine handle... say.. 40+ monitors at the same time?
22:09.58justinuime, yes
22:11.13*** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)
22:12.08[hC]well for the future
22:12.16[hC]current libc in debian unstable breaks zaptel.
22:12.27[hC]i downgraded and it works now.
22:13.12X-Rob_nortex, you should be speaking _to sangoma_ -- their tech support is amazingly good.
22:14.37nortexX-Rob_, I did a few weeks back on a fax over PRI issue and got no where. I'm really just curious if there is a problem or the norm.
22:15.02brijnbenjamin7062, Google for asterisk benchmarks.. Somebody did a bunch of test runs and put it all on a page
22:15.04X-Rob_well, that's kinda reasonable. fax over voip is a silly thing to do.
22:15.17X-Rob_getting good echo cancellation is not unreasonable.
22:15.30X-Rob_yes, you can still get echo, but you shouldn't notice it enough to bring it up in channel 8)
22:16.23benjamin7062brijn, Good Call.. I could just turn it on and break everyone if it fails.  =)
22:16.52benjamin7062Our callcenter gets a crap ton of traffic... don't want the IO to hurt the machine.  I'll look for the site.  TY
22:16.59nortexThanks, I'll check into it with them. The fax thing was not VOIP, it was PRI to channel bank on the same card and failed terribly,
22:17.35CunningPikemindwarp: Is it registered?
22:18.15terrapenanybody ever seen a Polycom minibrowser app that can show a list of parked calls?
22:18.25dahunter3Guys, I have a question about merging an inbound phone call with an outbound phone call.  The volume is really low and it's extremely hard for both sides to hear each other.  I assume I need to mess with the gain settings.  Is there a way to only do this for inbound + outbound calls?
22:19.00terrapenmerging?  you mean forwarding?
22:19.02nortexterrapen, Yes, look on voip-info there is a pack of XML files that can be twiked
22:19.04*** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net)
22:19.09terrapeni've never heard of "merging"
22:19.13SexyKenHey guys -- is there anyway to see how many channels have been in use at one time?
22:19.14terrapenthanks nortex
22:19.29*** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it)
22:19.40dahunter3I'm not sure what the right terminology is for it.  A call comes in and the dialplan connects it with an outside number.
22:19.44CunningPikeSexyKen: You mean zap channels
22:19.55nortexSexyKen, show channels will give the current stat.
22:19.57SexyKenWell, No -- becuase I'm using PURE VOIP
22:20.02SexyKenI dont want current stat
22:20.16SexyKenI want to know the highest number of hcannels that were in use at one time
22:20.40mindwarpCunningPike: i tried both with and without registration (i.e. both registering it and setting it to host=dynamic and doing "call/answer without registration" setting it to a static ip)
22:20.44terrapennortex, sweet, found what i needed.  shoulda looked there before askin'
22:20.51heath__run a crazy assed query against your cdrs?
22:20.56nortexterrapen, No problem
22:21.21terrapeni'll have to re-write it though.  i hate PHP :)
22:21.31dahunter3terrapen: So, I guess that's forwarding.
22:21.34mindwarpCunningPike: i should also note that if i dial with the console ('dial extension@context') then it will work as it should and the console will tell me a bunch of stuff about the incoming call, not sure if that really means anything
22:21.35terrapenI'll write a simple CGI in C or Perl or somethin
22:21.46terrapendahunter: "bridging"
22:22.08nortexterrapen, I've modified it, but the operators here struggle to use the phone, let alone anything cool like a microbrowser.
22:22.22*** join/#asterisk mog (n=mogorman@gateway.digium.com)
22:23.21CunningPikemindwarp: Not really :D At least we know that your dialplan works. It's likely to be something in your ATA's setup
22:23.29terrapennortex, i'm afraid that's what I might be up against
22:23.55terrapennortex, I wonder if there is a way to make the microbrowser interface allow an agent to pick up a parked call directly from the UI
22:24.16mindwarpCunningPike: the PSTN-to-VoIP gateway in the Sipura is using a dialplan that looks like (S0<:192.168.0.x) (address of the asterisk server) ... but the asterisk server doesn't seem to be given anything
22:24.23terrapeni suppose i could do it with the manager API but it would be nicer to do it straight from within the microbroser
22:24.59CunningPikemindwarp: Ah - that doesn't look like a valid dialplan to me....... hang on a sec
22:25.25mindwarpCunningPike: thanks
22:25.38terrapendahunter, you might be able to find an app that will allow you to adjust the gain
22:25.51terrapenso before you call Dial(), you'd call the gain adjustment app
22:25.54*** join/#asterisk nagl (n=nagl@86.59.54.237)
22:25.56terrapendunno if that is available
22:26.21dahunter3terrapen: Interesting.  Am I pretty much on the right track that it is a gain issue?
22:26.52terrapenprobably
22:27.02benjamin7062terrapen, the Microbrowser is easy to code against... and connecting the parked calls works well with the Manager API... fyi
22:27.36terrapeni'd explain your problem on the list (make sure you mention that you are trying to adjust the gain on two bridged channels) and see if there is a gain adjustment app
22:27.52terrapenbenjamin, any chance you could let me see your API code that connects them?
22:28.00terrapeni'm just lazy and it would save me some time :)
22:28.08smackusI have started getting this error: Jun 27 16:19:34 WARNING[10304]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x77f5b0', 10 retries! I have been reading around and have only learned how to log more verbose output for the error. Is there a fix for this? I just started happening. I must have changed something. but I dont know what.
22:29.35*** join/#asterisk jvictorfc (n=jvictorf@201009011114.user.veloxzone.com.br)
22:30.02jvictorfchi all
22:30.17jvictorfcplease i give a help
22:31.38jvictorfci need a md3200, I am having problems when I go up the module wcfxo for the md3200
22:31.50CunningPikemindwarp: Try this: (xx.)
22:31.57mindwarpCunningPike: trying
22:32.12*** join/#asterisk japerry (n=japerry@216.231.51.208)
22:33.09mindwarpCunningPike: same behavior -- the line rings, the phone connected to the Sipura rings, but Asterisk remains utterly silent
22:33.24jvictorfcits a problem: ZT_CHANCONFIG failed on channel 1: No such device or address (6)
22:33.25jvictorfcFATAL: Error running install command for wcfxo
22:33.45jvictorfccan anybody help myself?
22:33.51terrapenheh
22:34.04CunningPikemindwarp: Do you have the admin manual? There are some good examples in it.....
22:34.08dahunter3terrapen: Thnk you for your help.
22:34.13terrapennp
22:34.55mindwarpCunningPike: I guess I should really read that... I haven't been able to find it on the Sipura site yet, but I've heard it's supposed to be there
22:35.12mindwarpthe dialplan above was taken from a tutorial which supposedly is for what I'm trying to do
22:35.16mindwarpbut the settings seem to do nothing
22:35.34CunningPikemindwarp: It is - the admin guide was helpful to me
22:36.43mindwarpCunningPike: do you happen to have a link for it handy or could you tell me where you found it? I can't see it on the support page for the SPA3000?
22:37.29mindwarpCunningPike: ah, nevermind, I found it :)  Thanks for your help, I may ask for more once I RTFM
22:37.55CunningPikemindwarp: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf - yup, sure, no problem
22:38.29*** join/#asterisk mjh001 (n=mjh001@c-68-37-78-102.hsd1.nj.comcast.net)
22:43.59nortexI'm looking over the Polycom reboot script on voip-info and get the impression that it will only reboot one phone, or am I missing something? I want to nightly reboot all my Polycom phones.
22:44.02*** join/#asterisk rnovotny22 (n=rnovotny@198.57.19.126)
22:44.32terrapennortex, yeah, you'd have to hax0r something that got the info from 'sip show peers'
22:44.46terrapenthat might be able to be accessed through the manager api, I'm not sure...
22:45.43mindwarpuhm, should 'sip show registry' give me a list of registered peers? because it returns nothing (an empty list) and I thought I had 2 peers registered
22:46.30directorymindwarp: sip show peers
22:46.30nortexIf I had a list of the peer names I could just pass them to the asterisk cli with sip notify though right.
22:46.41terrapensip show peers
22:46.47*** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)
22:46.48mindwarpdirectory: thanks
22:49.42*** part/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)
22:50.15*** join/#asterisk phigwork (n=phigan@71-209-152-225.phnx.qwest.net)
22:50.46phigworkhi, my pbx is being weird with local outgoing calls (out the fxo)
22:51.21phigworkif I have this: exten => _9.,1,Macro(dialout,${TRUNK},${EXTEN:1})
22:51.36phigworkit doesn't actually dial everything but the 9.. or it's leaving something else out
22:53.28CunningPikenortex: I wrote a simple shell script that uses 'sip show peers' - want it?
22:55.08phigworkif I leave out the :1, it seems to work
22:55.09phigworkbut why
22:55.32*** join/#asterisk Nurstonix (i=fwuser@200.27.54.240)
22:56.41*** join/#asterisk RoyK[no] (n=roy@svg-acs.ipzone.no)
22:56.42salaudquick question... how do you / can you setup hunt groups for sip in asterisk?
22:56.54salauddoes the trunk methodology only work for PRI or Zap?
22:57.09salaudcan do something like SIP/trunkid and have it hunt?
22:59.34nortexCunningPike, Sure that would be great.
23:02.53benjamin7062phigwork, Good question.. =)
23:03.22RoyK[se]<PROTECTED>
23:04.58*** join/#asterisk quadrata (n=quadrata@ool-44c61ecb.dyn.optonline.net)
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23:10.07phigworkhehe, no ideas?
23:10.10phigworkmaybe just a bug?
23:11.52slideroolhello. does anyone here have experience using app_swift for realtime text to speech or would know what is causing these errors: "app_swift.c: Poll timed out/errored out" "app_swift.c: No more data."
23:13.04giesenIs there a way in asterisk queues to have asterisk prompt the agent receiving the call telling them they have to hit # to accept the call
23:14.43giesenphigwork: that looks right, so if it doesnt work, it's probably a bug
23:15.04giesenmaybe your macro has a bug in it
23:15.11giesendoes it work with a plain jane Dial cmd?
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23:36.35japerryin /var/log/asterisk/messages?
23:38.14CunningPikejaperry: Yes - or event_log
23:43.12japerryCunningPike: yup
23:45.50CunningPikeSorry - brain fart - there is another setting that deals with call supervision......
23:46.43*** join/#asterisk goldsmurf (n=rgoldber@64-13-22-231.dul.clearwire-dns.net)
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23:48.57CunningPikejaperry: Nope - sorry, red herring
23:49.29CunningPikeI think you're going to have to get lucky enough to get some PRI debug output from one of these dropped calls
23:51.17CunningPikejaperry: You can try messages => notice,warning,error,debug in your logger.conf and restart asterisk.
23:54.55japerryCunningPike: okay, I'll try that
23:55.34CunningPikeOK - let me know how you make out. On the CID thing, was it inbound or outbound?
23:55.47*** join/#asterisk iq|mobile (n=iq@unaffiliated/iq)
23:57.10japerryCunningPike: Both.. CID incoming doesn't wrok, and outside phones don't see our number either
23:57.22mrdigitalCunningPike?
23:57.55japerryCunningPike: I keep getting those debug messages every second to different phones
23:58.29CunningPikejaperry: From the zap channel?
23:58.34CunningPikemrdigital: ?
23:58.58mrdigitalpm?
23:59.35*** part/#asterisk benno|b1 (n=ben@88-96-30-86.dsl.zen.co.uk)
23:59.38CunningPikejaperry: That's coming from chan_sip, I notice - not from chan_zap......
23:59.44japerryCunningpike: DEBUG[23371] chan_sip.c: Auto destroying call '302ea438a1df97f5@10.0.6.105' and I get DEBUG[23371] chan_sip.c: Stopping retransmission on '4cc86356543fc6ff2d410ac168fc8b31@10.0.6.14' of Request 102: Match Found

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