irclog2html for #asterisk on 20041116

00:00.11darkskiezbowman: i hate the way they change the names in every bloody bit of software
00:00.52bowmandarkskiez: I'm not talking about linphone, I'm talking about Gnophone ;) anyways, I used exactly the options given in the gnophone example at the wiki page
00:01.13darkskiezBoRiS: what does asterisk say in the logs
00:01.22darkskiezs/BoRis/bowman
00:01.25*** join/#asterisk matobago (~matobago@65-77-23-11.ptp.ezeronetworks.net)
00:01.31matobagohi everyone
00:01.40matobagohttp://news.zdnet.co.uk/communications/networks/0,39020345,39169076,00.htm
00:02.19*** part/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net)
00:02.27_Vilehe likes to show me this things
00:02.30_Vileskdfk
00:03.09darkskiezis there any good examples of a working enum configuration that doesnt bork up ?
00:03.23*** join/#asterisk m-00kie (3704558@pcp09297172pcs.arlngt01.va.comcast.net)
00:03.44bkw_ok why does drama always find me?
00:03.45darkskiezlike checks the enum for what you dialled and falls back properly if it doesnt work
00:03.46bkw_WHY OH WHY?
00:03.46bowmandarkskiez: it says pure void , nothing :-)
00:04.04darkskiezbowman: set debug 15
00:04.07darkskiezbowman: set verbose 15
00:04.19ZX81_AFKmaybe cos u r a drama queen?
00:04.22ZX81_AFK:-)
00:04.45*** join/#asterisk bonbon-home (~happy@81-86-185-223.dsl.pipex.com)
00:05.05*** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net)
00:09.09bowmanscratch it. asterisk doesn't even react to traffic from gnophone, whereas gnophone sends exactly 2 packets to the server which contain pure garbage.
00:09.37rene-hey all, does anybody know what happened to zapteldoc.blogspot.com? it hasnt been updated for ages....
00:10.36ZX81_AFKdunno
00:10.38ZX81_AFKsux a
00:11.45darkskiezWTF does inheritance mean with regards the SetVar applicaiton
00:12.37*** join/#asterisk Corydon76-home (gray@pcp08665860pcs.500ash01.tn.comcast.net)
00:12.44redder86if SetVar's parent dies then all of the parent's environment will belong to SetVar
00:12.55redder86;-)
00:15.03blitzrageis notransfer=yes known to work in Asterisk?  I can't seem to stop the calls from being natively bridged...
00:15.40file[laptop]blitzrage: always works for me
00:16.06blitzragefile[laptop]: I can't seem to get it to work at all.  Very confusing and annoying
00:16.11blitzrageunless I'm missing something stupid simple
00:16.29file[laptop]just put it in the two entries... hrm yes
00:17.19blitzrageI've placed notransfer=yes in the [general] section of iax.conf on all 3 computers.  I'm calling from *1 <--> *2 <--> *3 and *2 keeps natively bridging the calls.
00:17.26ChulJinI thought to prevent [attempts at] native bridging it was 'canreinvite=no'
00:17.31ionixI get an error when i receive a FWD call on my asterisk gateway :/
00:17.32ionixlook
00:17.32ionix<PROTECTED>
00:17.32ionix<PROTECTED>
00:17.32ionix<PROTECTED>
00:17.32ionixNov 15 19:16:09 NOTICE[-1253721168]: app_dial.c:714 dial_exec: Unable to create channel of type 'SIP'
00:17.33ionix<PROTECTED>
00:17.35ionix<PROTECTED>
00:17.39_Vilepastebin.ca
00:17.43ionixoks orry
00:17.44ionixsorry
00:17.45blitzrageionix: NO FLOODING please
00:17.57file[laptop]ChulJin: different things, different things
00:18.00JonR800SIP/l100????
00:18.01Optic2sip sip sip sip mooo
00:18.03ionixyeh
00:18.04ChulJinah, I apologise.
00:18.07_VileSIP/l100?
00:18.07ionixl100 is the extension
00:18.09rowterbbits possible to add ilbc support to cisco 7960, anyone has heard about it?
00:18.21file[laptop]notransfer=yes will prevent your IAX2 legs from directly connecting, put it in their entries... try not global
00:18.22_Vilesip debug
00:18.23ionixI have [l100] in sip.conf configured
00:18.29file[laptop]canreinvite=no does the same thing, prevents it from directly connecting
00:18.31JonR800oh.
00:18.37file[laptop]they will still native bridge the audio if both sides are using the same codec
00:18.44blitzragedamnit!
00:18.47blitzragenot what I want
00:18.53*** join/#asterisk Elshar (~Elshar@just.another.lame.unix-admin.com)
00:18.58blitzrageI need *2 to stay in the channel path.
00:19.15JonR800ionix: do a sip show peers.. and check that it's registered.
00:19.22filenative bridge the audio... as in, not transcode
00:19.26_Vileis your l100 phone registered, ion?
00:19.55blitzragefile[laptop]: what do you mean?
00:20.02rowterionix, it seems its not registered, try to add l100 to name and shortname as in authname to test it.
00:20.11ionixstatus is Unmonitored
00:20.22ElsharHey, has anyone had a problem with the zaptel drivers? In particular, I keep getting the error "zaptel: unknown symbol crc_ccitt_table". When trying to modprobe a cleanly compiled version from cvs
00:20.29blitzrageionix: qualify=yes to monitor
00:20.34ionixok
00:20.35fileblitzrage: native bridging = asterisk just passing the audio from one leg to the other because of the same codec
00:20.50blitzragefile: right, but I don't want that :)
00:21.00ionixstatus is UNKNOWN
00:21.10fileblitzrage: can you go into a conf?
00:21.19blitzragefile: sure, where?
00:21.30ionixaccording to the SPA-2000 Registration State: Registered
00:21.32implicitman even my regular cdrs are acting weird
00:21.35blitzragefile: just /msg it to me
00:21.39_Vile<PROTECTED>
00:21.49implicitdoes someone have a bit of time to take a look at them with me?
00:22.05blitzrage_Vile: I get those every once in a while too - not sure the exact cause
00:22.14blitzrage_Vile: doesn't seem to affect much...
00:22.16_Vileme either, but it's simply annoying...
00:22.19_Vileyeah, no complaints here
00:22.25fileblitzrage: get it?
00:22.56JonR800ionix: does it list an ip in sip show peers?
00:23.12ionixUnspecified
00:23.13ionixhmm
00:23.15implicitdoes anyone know how to get cdr's to be put out for all the legs of a call?
00:23.21JonR800it doesn't sound like it's registered.
00:23.23Blackthornon a two pri setup can you have both pri's set to primary sync? and would that be a reason that shows the secondary in red alarm
00:23.24implicitI am dialing a DID that goes into my asterisk box
00:23.26mishehuwhat does the zaptel watchdog actually do if a zap device stops taking interrupts?
00:23.28ionixwhy is that so ?
00:23.31JonR800err not registered
00:23.34JonR800sorry typo
00:23.45ionixsipura says: Registration State: Registered
00:23.52ionixshould I restart the spa ?
00:24.17ionixahh
00:24.19ionixlol my error
00:24.29implicitthat rings my 7960 which forwards back to the local context which uses a voip provider to my cell phone, but the only records that go in are the DID record and the Local dial, there is no dialout to the voip provider
00:24.42implicitanyone know how to change that ?
00:25.16ionixRegistered SIP 'line1' at 65.94.160.215 port 62263 expires 3600
00:25.46ionixhowever I still cannot receive a call
00:25.46Blackthornwell got my primary up thats good enophe for the night thanks for everyoens help
00:25.49ionix<PROTECTED>
00:25.49ionix<PROTECTED>
00:26.05ionixso at this point problem is the SPA ?
00:26.42ionixok works
00:26.52ionixI love you guys
00:27.12JonR800I'm glad someone does.
00:27.18Elsharhehe :)
00:27.26implicit??????
00:27.28implicitlol
00:27.30*** join/#asterisk sleepy_one__ (~chatzilla@dhcp16632045.neo.rr.com)
00:29.31*** join/#asterisk sleepy_one____ (~chatzilla@dhcp16632045.neo.rr.com)
00:29.35blitzragefile: thanks again for the tip - you're the man!
00:30.40*** join/#asterisk nem (~nemisis@66.220.23.4)
00:30.42ElsharAnyone have any ideas on that unknown symbol thing? ;)
00:30.55bonbon-homeif you have a quality=yes setting in your iax entity, then wouldn't you expect * to move from one line of extensions.conf to the next in the event that iax2 connection is down?
00:31.36bonbon-homebecause it doesn't seem to work like that
00:31.38bonbon-home:-(
00:32.08*** join/#asterisk nem (~nemisis@66.220.23.4)
00:32.45bonbon-homeit jumps to "busy" context
00:32.59*** join/#asterisk krw (~ken@209.242.52.25)
00:33.46*** join/#asterisk trelane (trelane@lan.trelane.net)
00:33.53*** join/#asterisk sleepy_one_____ (~chatzilla@dhcp16632045.neo.rr.com)
00:34.23*** part/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net)
00:34.46implicitbonbon-home: sucks :(
00:35.17trelaneanyone have a good FSX interface recommendation besides digium's?  Ambient/Intel chipset modems are noted on voip-info.org but I've not seen anyone being particularly open about what chipsets are on their modems, does anyone have any experience or feedback with specific vendor models?
00:35.50mengeris ftp.digium.com broken? seems to be very slow for me
00:36.07file[laptop]it's always slow
00:36.15file[laptop]it's the ftp.digium.com way
00:36.26bonbon-homeimplicit: have you got it to work?
00:36.36sleepy_one_____try to wget asterisk v1.0.0 and zaptel and libpri instead using http
00:36.57mengersleepy_one____ url?
00:37.45sleepy_one_____wget http://www.asterisk.org/asterisk-1.0.0.tar.gz; wget http://www.asterisk.org/libpri-1.0.0.tar.gz; wget http://www.asterisk.org/zaptel-1.0.0.tar.gz
00:37.46*** join/#asterisk Moc (~mochouina@modemcable212.49-80-70.mc.videotron.ca)
00:37.49Mochi all
00:39.27sleepy_one_____hey Moc
00:39.39ChulJinhey Moc
00:39.49ChulJinça va?
00:40.42Mocpas pire toi ?
00:40.43sleepy_one_____tres bien merci
00:40.56sleepy_one_____pardon my french ;)
00:41.09Moc;)
00:41.17sleepy_one_____pun intended
00:41.30Mocyou said it right ;)
00:42.07sleepy_one_____I know ;)
00:42.24sleepy_one_____he was addressing you tho, I was kidding around
00:43.33*** join/#asterisk enzo123 (~jason@cpe-024-211-181-116.nc.rr.com)
00:46.18enzo123sup all
00:46.28enzo123anyone using a spa3000
00:47.53*** join/#asterisk sleepy_one______ (~chatzilla@dhcp16632045.neo.rr.com)
00:48.07ChulJinenzo: yes, but not the pstn side (yet)...so not prepared to answer questions about that part of its functionality
00:48.27sleepy_one______anyone know what the ETA is on the S100I v2.0 ?
00:49.47trelaneanyone have a good FSX interface recommendation besides digium's?  Ambient/Intel chipset modems are noted on voip-info.org but I've not seen anyone being particularly open about what chipsets are on their modems, does anyone have any experience or feedback with specific vendor models?
00:50.00trelanes/FXS FXS
00:50.29*** join/#asterisk myconid (myconid@155.42.19.149)
00:50.31myconidHello.
00:50.32enzo123ive got a clone
00:50.42myconidI am trying to connect to my remote * system..
00:50.48myconidvia SIP (xlite)
00:50.50enzo123dont remember the chipset tho
00:50.58myconidbut it just times out.. and im not sure how to see if sip port is open
00:51.20*** join/#asterisk Legend (~legend@24.244.142.134)
00:51.51trelaneenzo123, know what manufacturer made the modem?
00:52.55*** join/#asterisk sleepy_one (~chatzilla@dhcp16632045.neo.rr.com)
00:53.12sleepy_onehello again everyone
00:55.39implicittrelane fxs with which modems??
00:56.27myconidhow do I bind multiple ports in sip.conf?
00:58.35enzo123dont remember
00:58.44enzo123ebay for x100p
00:58.52enzo123there are tons of clones
00:58.57implicitfxo?
00:59.21implicitx100[p is fxo
00:59.27WilliamKprobably 50-100 on ebay right now
00:59.29enzo123FXO
00:59.32WilliamKjust looked about 20 mins ago
00:59.35enzo123PSTN
00:59.39implicitnever heard of modems doing fxs
00:59.46enzo123they dont
00:59.52implicitya
01:00.05implicittrelane: did u ,mean fxs or jiust typo?
01:00.19enzo123got a old quicknet card
01:00.27enzo123souunds like crap but it works
01:00.43enzo123for fxs
01:00.54folssonmyconid : You don't
01:01.18impliciti used the tdm400p and iaxy
01:01.24implicitnot too pleased with either's sound quality
01:01.29implicitso i just bought a 7960
01:02.00implicitthe sound quality is not too bad either
01:02.04implicitjust not as good as i'd like
01:02.16*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
01:02.22implicithi
01:02.28puzzledhello
01:02.33implicitwhats going on?
01:02.36enzo123my 7960 doesn't sound as good as my sipura
01:02.53implicitoh yeah?
01:02.59Mocenzo123, I wouldn't say that ;)
01:02.59implicitwhich sipura?
01:03.12Mocbut my polycom sound better than my cisco
01:03.17enzo123haha
01:03.20enzo123prob so
01:03.29puzzledMoc: polycom kept the good stuff for themselves :)
01:03.38krwI'm trying to use ser to forward to a voice app on asterisk.  I'm rewriting the URI and forwarding the call correctly to asterisk, but I need to get the origonal dialed number in the "To:" sip header.  Found RDNIS but that appears to be adopting an old cisco "Diversion" tag, anyone have any thoughts on this?
01:03.41*** join/#asterisk Darwin35 (~darwin35@c-24-3-204-71.client.comcast.net)
01:03.44implicityou think i should sell cisco and get the polycom?
01:04.08puzzledthey are cheaper, that's for sure. and cisco's are basically licensed/OEM'ed polycom technology
01:04.09Mocimplicit, Well you can get 2 polycom for the price of 1 cisco ;)
01:04.18enzo123im using my cisco on 729 for pass , but even the vm sounds kinda crapy
01:04.20implicitwhat is the best polycom?
01:04.24Mocimplicit, you ever saw polycom ?
01:04.27implicitseen
01:04.29implicitbut not used
01:04.29puzzledimplicit: IP600
01:04.33implicitnot seen in person
01:04.33Mocimplicit, IP 500 are good, but IP 600 is the best
01:04.39MocI just ordered the ip 600 yesterday
01:04.44enzo123my 7960 was only 200
01:05.15Moceven if 7960 and polycom was the same price, I would still go to polycom
01:05.21implicitmine was 250 with headset and power cube
01:05.42Mocimplicit, you ever saw polycom config file ?
01:05.49puzzledhehe
01:05.53implicitnope
01:05.56puzzledxml stuff right?
01:05.58Mocimplicit: check http://www.freedomphones.net/polycom/files/Admin_Guide-SoundPoint_IP_SIP_2004-06-16.pdf
01:05.59Mocyes
01:06.17Mocthis PDF will show you just how good polycom phone are
01:06.22*** join/#asterisk johaan (~johaan@ip24-56-24-181.ph.ph.cox.net)
01:06.27enzo123hey got a question for everyone that has nufone
01:06.28ChulJin~seen jerjer
01:06.31jbotjerjer <~mine@d2-236.rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 3d 17h 56m 37s ago, saying: 'no config necessary'.
01:06.39Mocenzo123, go
01:06.45ChulJin~seen jerjer[mobile]
01:06.46jbotjerjer[mobile] <~jj@mail.nufone.net> was last seen on IRC in channel #asterisk, 8h 40m 55s ago, saying: 'paging'.
01:06.54enzo123whats up with nufone ?
01:07.19ChulJinhaha that's rather broad :P
01:07.20Mocwhat wrong with them ?
01:07.21redder86enzo123: whaddya mean?
01:07.23ChulJin(though I know what you mean)
01:07.50enzo123all support emails to them go unanswered
01:07.59johaanHello everyone,  Running Fedora Core2,  I did make config to have asterisk start up on boot.  And the script started like 14 Asterisk proccess
01:08.01johaanroot      2115  0.0  0.4 115396 5020 tty1    S    03:31   0:00 asterisk -vvvg -c
01:08.29enzo123service asterisk start
01:08.32johaanpreviously I was just have one running..  is this an error or is it suppose to look like that.
01:08.44Mocimplicit, what do you think now ? ;)
01:08.56Moccan you do all that with your cisco ? ;)
01:09.05*** join/#asterisk syslod (~sysglod@65.114.0.198)
01:09.21enzo123anyone else able to get support fomr them ?
01:09.31syslodhi
01:09.36redder86enzo123: yup
01:09.59enzo123humm then must just ignore my emails
01:10.10redder86enzo123: how long has it been?
01:10.27trelaneimplicit, I meant fxs, that was indeed a typo
01:10.33enzo1233 weeks this thr
01:10.36ChulJinenzo: once I overcame my indignance, I found and talked to JerJer. my DID's not up, but he did have some encouraging words about the progress.
01:10.43trelaneanyone have a good FXS interface recommendation besides digium's?  Ambient/Intel chipset modems are noted on voip-info.org but I've not seen anyone being particularly open about what chipsets are on their modems, does anyone have any experience or feedback with specific vendor models? (amended for implicit)
01:10.57ChulJinenzo: one thing to notice (that I overlooked) was 7-10 business days.
01:11.01Mocdamn you can even use Windows .fon file on the polycom phone ..
01:11.02puzzledenzo123: me too. wonder if they have "a prepay as you go" thing instead of those plans
01:11.29redder86treland: the Ambient/Intels are for FXO.
01:11.35enzo123umm oct28th is over 10 buz days
01:11.51trelaneredder86, I was looking for specific model recommendations (I've had signal to noise issues on mine)
01:11.59enzo123i dont really care about the time.. just reply saying " hey we are working on it "
01:12.04redder86treland: FXO or FXS?
01:12.13redder86enzo123: what's the problem?
01:12.15trelaneFXS
01:12.28trelaneto terminate a phoneline
01:12.35redder86trelane: FXO, then
01:12.36johaanI should only see one asterisk process running right?
01:12.41trelanedo I have it backassword?
01:12.56enzo123ordered and paid for a 800 DID waiting for it to be turned up  3 weeks thr
01:12.58redder86trelane: you plug phone lines into FXO ports and phones into FXS ports
01:13.16redder86enzo123: he had mine up in a matter of hours.  I didn't order vanity, though.
01:13.23BoRiSfrom enzo?
01:13.30enzo123vanity
01:13.58redder86enzo123: did you pay yet?
01:14.10trelaneredder86, then I need FX0
01:14.14trelanesigh
01:14.18trelaneI need a new keyboard
01:14.21redder86trelane: how many ports?
01:14.22trelaneI honestly can type
01:14.22enzo123redder86 yes
01:14.23trelaneone
01:14.26trelanesingle line
01:14.27trelanevoicemail system
01:15.12ChulJinenzo: the idea is to forego indignance, find JerJer on here, and just PM him.
01:15.15ChulJinHe's actually quite nice.
01:15.37enzo123my point is that i should have to chase down someone on IRC to get support
01:15.43enzo123shouldn't
01:15.46redder86trelane: if you don't want to pay for the dev kit from Digium (X100P) then find an Intel/Ambient that uses the MD3200 chipset.  AMI-IA92/IE92 but make darn sure that you check with the vendor on the chipset first.  The i537EP chipset is *not* what you want.
01:15.55*** join/#asterisk JunK-Y (~junky@modemcable144.95-37-24.mc.videotron.ca)
01:16.18redder86enzo123: in my experience he's been good about responding to e-mail
01:16.35trelaneredder86, the intel 837 is also supposed to work, I've got one, it does, signal quality is poor, I think because of inferior modem quality, I got it used for $1
01:16.38redder86enzo123: as long as it got into his "system"
01:16.54*** join/#asterisk AlexZander (~AlexZande@spc2-bolt4-4-0-cust107.bagu.broadband.ntl.com)
01:16.57trelaneredder86, my question wasn't so much one of compatibility but of personal experience with a given part
01:16.57redder86trelane: I'll bet you money, though that it's not the i537EP chipset.
01:17.24myconidanyone use a IAX softphone/
01:17.35enzo123you can make the call via e164.org lookup , but not via pstn
01:17.46redder86trelane: I've used two different AMI-IA92/IE92 modems that I purchased from two different vendors, both with no problem.
01:17.57trelanenoted
01:18.01trelaneredder86, that's what I needed to haer
01:18.01trelanehear
01:18.20redder86trelane: one I paid $15 for and the other $65.
01:18.42trelaneany audible difference in quality, or issues with call pickup etc?
01:19.08*** join/#asterisk riksta (~rick@81-178-241-195.dsl.pipex.com)
01:19.11redder86trelane: nope.  In fact, the cards are identical, except that the vendor of the $65 one rubbed out the chipset ID silkscreen with sandpaper.
01:19.30trelanejerks
01:19.31redder86trelane: I purchased that one directly from Tiger Jet.
01:19.51redder86trelane: the other I purchased from eebuy.com
01:20.17redder86trelane: I hear that eebuy.com is now listing i537EP cards under the AMI-IA92/IE92 label, though.
01:20.34trelanenoted
01:20.35redder86trelane: and those don't work, believe me.  If you want some I can sell you a few.
01:21.07enzo123mine works like a champ
01:21.12enzo12315 bucks on ebay
01:21.26redder86enzo123: it's not an i537EP then
01:21.29trelaneredder86, any intel with speaker+mic is a nogo best I can tell
01:21.56redder86trelane: the i537EPs that I have do not have speaker or mic
01:22.39redder86trelane: I must correct myself.  There was a slight difference between the two cards that I have.
01:23.03redder86trelane: the PCI ID codes vary due to some differences in the resistors on the PCI ID portion of the card.
01:23.21redder86trelane: the PCI IDs only affect "auto detection"
01:23.43trelanethat's not an issue, I can make just about anything work, I just can't deal iwth a loud whirr throughout the call
01:23.59enzo123here
01:24.01enzo123http://i12.ebayimg.com/02/i/02/a9/31/db_1.JPG
01:24.10enzo123thats the one i have
01:24.34redder86enzo123: an "Intel 537", but note that it has the MD3200 chipset.
01:24.43puzzledenzo123: got mine new in the shop for something like €9.95
01:24.44redder86enzo123: not the i537EP
01:25.11*** join/#asterisk florz (nobody@I9593.i.pppool.de)
01:25.14Syncros<PROTECTED>
01:26.01redder86enzo123: notice also the "AMI-IA92/IE92" silkscreening.  I don't think that you will see that on the i597EP cards, although the *vendor* may incorrectly sell you an i537EP under that name.
01:26.49redder86Syncros: I went through the effort of looking up "Tiger Jet Network Inc." and it led me to cuphone.com, which is where I bought the $65 version that was adulterated.
01:26.56enzo12300:0e.0 Communication controller: Individual Computers - Jens Schoenfeld Intel 537
01:27.17enzo123yeah
01:27.25enzo123i got my from a buddy that got 30 of them
01:27.34enzo123all 6 bucks each
01:29.15redder86enzo123: there were numerous vendors of these things.  The PCI ID was completely configurable on the vendor-level, and so there's a bunch of different entries in the Linux PCI ID device list... just depending on what particular PCI ID your card uses.
01:29.35enzo123guess i got lucky
01:29.55enzo123picked up a spa3000 today ill she how that does me
01:30.00*** join/#asterisk vexorg (~vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
01:30.17redder86enzo123: add/remove resistors to the joints on the top-left (in the picture you gave) to twiddle with the PCI ID value.
01:30.55myconidlol
01:40.00*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
01:50.12*** join/#asterisk t3t (~t3t@207.67.0.20)
01:52.38redder86ssssh!
01:53.41*** join/#asterisk Darwin_35 (~darwin35@c-24-3-204-71.client.comcast.net)
01:54.03ariel_wow seems like there is just plain too much going on here tonight.
01:54.14syslodI'm telling ya.
01:54.19syslodLots of activity
01:54.42syslodWell guess I'll liven it up a bit with a ongoing problem.
01:54.59ariel_problem?
01:55.03syslodAnyone know why MWI would not clear on a poly 300?
01:55.19Optic2anybody know when budgetone will support MWI? :)
01:55.20Optic2hehe
01:55.24syslodReboots don't help nothing.  Somehow it thinks I have a message.
01:55.34redder86Optic2: mine already does.
01:55.38syslodgrandstream MWI works.
01:55.40Optic2redder86: whoa!
01:55.49ariel_syslod, sometimes there is some mistry message left in the box.
01:55.52Optic2is there a faq somewhere?
01:55.57syslodpoly works too well.  It won't turn off.
01:56.00Nuggethttp://voip-info.org/
01:56.13redder86Optic2: I have had a Grandstrem BT100 since June and it has always supported MWI
01:56.27syslodI checked all the boxes.  Any debug commands or way to have MWI info in the debug?
01:56.53syslodGrandstreams seem to work well except for forgetting they are registered.
01:56.58ariel_are you sure it's not pointing to the wrong box area? other then default?
01:57.01syslodPolys work even better.
01:57.16redder86Polys are a pain to configure, but work well, yes.
01:57.18syslodI've checked it a million times.
01:57.50syslodI even setup a budgettone then configured the poly for the same sip account.  Budget tone works poly doens't.
01:58.08redder86sounds like a Poly issue
01:58.28syslodAlso is there a way to have the message button or message menu go directly to voicemail password prompt?
01:58.48syslodAnyone have a working config for the poly they mind sending me?
01:59.12rikstawhat clever things can i do with the menu screen of a cisco 7940 ?? :)
01:59.21redder86syslod: you can configure all of the buttons, including the message one, to dial whatever you need it to
02:00.01syslodBut what do I program it to?  IE voicemail is ext 999 do I just do 999 then mailbox?
02:00.09syslodSeems to screw up 80% of the time.
02:00.14BoRiSOptic2: Are you using a Grandstream with realtime config (res_config?)
02:00.17syslodIs there a pause
02:00.20Optic2nope
02:00.29BoRiSok
02:00.52*** join/#asterisk Verliba (~Miranda@65-100-56-22.ptld.qwest.net)
02:02.14redder86syslod: is your poly configured via ftp?
02:02.24syslodYea.
02:02.55syslodI've played with the menus too.
02:03.03syslodftp seems to have more options.
02:03.46redder86I can't very well send you my cfg without DCC, now can I?
02:04.25syslodnope.  I'll try and figure out what broke my dcc or return with working client. BRB.
02:05.00redder86http://pastebin.ca/2115
02:05.37*** join/#asterisk syslod (0@65.114.15.106)
02:05.39AlexZanderSorry to but in but are you talking about Polycom handsets - Last time I looked they weren;t too useable with Asterisk - have things changed now?
02:05.59syslodK now in working client.
02:06.40syslodPolys seem to work very well.  300's and 500's here
02:07.57*** join/#asterisk nullogic (~nullogic@c-24-98-72-110.atl.client2.attbi.com)
02:08.59syslodbeen awhile since I've been on IRC at the console.
02:09.50AlexZandersyslod: Do you use sip or h323 with the polys?
02:10.02syslodsip
02:11.50AlexZandernice - I'll have to take a look at them again - I really liked the look of the Poly handsets but was told that I a) couldn;t get reliable sip to work with asterisk and b) you couldn;t configure them fully so they worked with asterisk without compromise
02:12.26syslodI think most ppl are using poly
02:13.14AlexZanderIt's been quite a long time since I last frequented this channel and looked at asterisk
02:13.27*** join/#asterisk smoothjim (~jim@24-159-238-38.jvl.wi.charter.com)
02:13.51AlexZanderBack then, people were going with snom over poly not because they felt they were better but just because they worked - I'm obviously way behind.
02:13.52syslodIt looks like its a replacment for most keys.
02:14.14syslodWe are looking at virtual hosted PBX's
02:14.41redder86If Snom works then Polycoms rock.  Snoms suck IMO.
02:15.05syslodredder86: Hopefully things are working now if you don't minding sending again.
02:15.21ariel_snom work fine. Polycom are better but are alittle harder to configure. But both are good phones.
02:16.12redder86I've had no end of firmware glitches with my Snom 190
02:16.19redder86I have to reboot it frequently.
02:16.56redder86Polycom are quite difficult to configure in comparison with, say, a Grandstream.
02:18.04syslodexit
02:18.23*** join/#asterisk tclark (~TC@S0106000c413a1c61.gv.shawcable.net)
02:18.35*** join/#asterisk hades_ (~hades_@200-203-038-195.paemt7006.dsl.brasiltelecom.net.br)
02:19.12*** join/#asterisk syslod (~sysglod@65.114.0.198)
02:21.41*** join/#asterisk mlh407 (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net)
02:23.10*** join/#asterisk QRPartner (~tenikiwon@ns1.accu-com.com)
02:23.22AlexZanderredder86: I guess though that once you;ve configured one successfully, it's quite easy to do it again - Plus you get so much more out of the phone than you do with a Grandstream - A phone that interacts with Asterisk like a traditional PBX is worth the premium.
02:23.31*** join/#asterisk dima202 (~chatzilla@ool-18b9039c.dyn.optonline.net)
02:23.37dima202»ey guys
02:23.56dima202I am running ppc, would it be possible to run atericks on this processor
02:23.58AlexZanderI'm looking at all the poly stuff on voip-info.org - very happy - none of this was there last time I looked.
02:24.39dima202It would be great if someone could point me to the direction
02:25.29funkknobPPC Mac or PPC IBM?
02:25.40dima202mac, sorry
02:26.00funkknobYou want to run under MacOS or Linux?
02:26.10dima202I am running Mac right now
02:26.26funkknobOS X I think you should be able to, or load linux.
02:26.42dima202I tried make
02:26.47JamesDotComyeah, there's a binary package of asterisk for os x floating around somewhere
02:26.48funkknob'Cuz OS X is basically unix now
02:26.52dima202but I get an error, I will paste
02:27.33dima202for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x depend || exit 1 ; done
02:27.42dima202make[1]: Nothing to be done for `depend'.
02:27.54JamesDotComthat's not an error
02:27.58dima202cd editline && unset CFLAGS LIBS && test -f config.h || ./configure
02:28.01JamesDotComoh
02:28.09dima202make -C editline libedit.a
02:28.14dima202and td.
02:28.24dima202I mean so on
02:28.27JamesDotComno errors in there so far
02:28.36dima202gmm
02:28.38JamesDotCommake install it
02:28.46dima202ast_expr.y:110: unrecognized: %locations
02:28.48JamesDotComthen try run "asterisk -vvvc"
02:29.14dima202if [ -d CVS ] && ! [ -f .version ]; then echo 1.0.0 > .version; fi
02:29.26JamesDotComwell anyway, as i said, there's a compiled version of asterisk for os x floating about a few places
02:29.45dima202unknown command
02:29.54JamesDotComdid you make install?
02:29.58dima202yes
02:30.04JamesDotComdid that error?
02:30.28dima202Yes, it occured
02:30.34dima202make: *** [ast_expr.c] Error 1
02:30.38dima202here is 1 error
02:31.00*** part/#asterisk smoothjim (~jim@24-159-238-38.jvl.wi.charter.com)
02:31.01JamesDotComgo to pastebin.ca or something and paste all the output from make
02:31.06*** join/#asterisk Marlow (~marlow@217.67.139.197)
02:31.48dima202Argh, it's the damn client, won't let me paste
02:32.27JamesDotComwhich client?
02:32.51dima202from mozilla
02:32.59file[laptop]kram: how are you muffin man?
02:33.05kramchillin
02:33.06kramu?
02:33.23dima202WOuld you be able to accept txt version?
02:33.42file[laptop]these reality TV shows are scary
02:33.47JamesDotComdima, just get something substantial to look at somewhere :D
02:34.16JamesDotComscariest one i've seen here is a reality tv show based on ballroom dancing
02:36.41file[laptop]creepy
02:37.15paulcI saw Wife Swap for the first time the other night.. HOLY, that program is CRAZY!
02:37.16JamesDotComvery
02:37.23file[laptop]that was on earlier
02:37.24JamesDotComit's got sms voting and all
02:37.37*** join/#asterisk Peanut486 (~dustin@net24-164-112-183.neo.rr.com)
02:38.13JamesDotComhaha, they even showed that show over here in australia
02:38.30JamesDotComthere was another one just the same, trading spouses? i think
02:38.35file[laptop]paulc: conf?
02:38.43paulcSure, pick a number?
02:38.47JamesDotComdima202: any luck?
02:38.48file[laptop]666
02:38.57paulccos you're evil personified ;-)
02:39.03file[laptop]yup
02:39.05dima202yeah, I think I found something
02:39.21dima202http://www.apple.com/downloads/macosx/unix_open_source/asteriskinstallpackageformacosx.html
02:39.26JamesDotComyeah
02:39.30JamesDotComvery old version though
02:39.35JamesDotComi'm sure you'll find newer ones about
02:39.43JamesDotComVersion:  CVS 10-28-03
02:41.50dima202how do u run it?
02:41.53dima202lol
02:42.16dima202no man pages
02:42.39JamesDotComasterisk
02:42.44*** join/#asterisk wolfson (~hehe@208.25.254.124)
02:42.47JamesDotComasterisk -vvvc for some verbosity
02:42.58dima202hmm
02:43.34dima202it tells me it's running
02:44.04JamesDotComasterisk -r
02:44.59dima202unable to connect to remote asterisk
02:45.35dima202hmm
02:48.06QRPartnerI downloaded the *1.0.0.tar.gz file and I seem to unzip them
02:49.06slePPanyone have a DVG-1120S?
02:49.25*** join/#asterisk Verliba (~Miranda@65-100-56-22.ptld.qwest.net)
02:50.00file[laptop]slePP: loan me one!
02:51.11slePPheh
02:51.14slePPthese things suck ass
02:51.52dima202what command is to list current processes
02:52.00funkknobps -ef
02:52.03slePPfile: keep your PAP2-NA's. don't ever let them go
02:52.06dima202thanks
02:52.13slePPi'd like to toss it out the window, but, alas.... it's the best i can do atm
02:52.20slePPmind you, it costs more than the SPA-2000
02:52.25slePPand the SPA-2000 is much cooler
02:52.32dima202there is another command
02:52.34slePPwe're still waiting on our RMA of one of the 5 PAP2's we had
02:52.48paulcI love the SPA-3000
02:52.51slePPand we have 21 more on order.. but, yeh. right.... 'they'll be in in 3 days!' 'bullshit you whoremongeringbitchfacedliar!'
02:53.05paulcand I'm itching to get my hands on a new Sipura 841 phone
02:53.15slePPer.
02:53.15file[laptop]paulc: 666 you ding bat
02:53.16slePPmouse
02:53.51file[laptop]mmm.... Swiss Chalet....
02:54.06funkknobI haven't used IRC for more than 10 yrs - what's a good windoze client?
02:54.17BoRiSdrive thru?
02:54.25BoRiSmirc
02:54.34puzzledgaim
02:54.36mlh407chatzilla
02:54.38funkknobSorry, I mean free :D
02:54.40file[laptop]I'm in 666 for all those who know what it is
02:54.44JamesDotComirssi4life
02:54.53paulcI use X-Chat and like it a lot.. www.xchat.org I think
02:55.06JonR800I'm with irssi.
02:55.28Peanut486I am trying to have the FXO channel on voice1 server ring the FXS on voice2 server. I keep getting "Rejected connect attempt from <HOST>"
02:55.37*** join/#asterisk wolfson` (~hehe@208.25.254.124)
02:55.52funkknobk I'll check those out thx.... right now using HydraIRC but I have to copy/paste URLs :(
02:56.32dima202guys, there is something besides ps
02:57.01funkknobWhat do you need to see?
02:57.08dima202all processes
02:57.29funkknobps -ef did not work?
02:57.29dima202on unix I remember there was a command, just can't remember it
02:57.32JamesDotComps aux
02:57.35dima202it is same for mac
02:57.36JamesDotComon os x
02:57.52dima202yes, yyou are correct, there is also a live one
02:57.59funkknobtop
02:58.03JamesDotComoh
02:58.05JamesDotComyeah top
02:58.07dima202hell yeah
02:58.10funkknobheehee :D
02:58.10dima202Thanks guys
02:58.18JamesDotComi love os x
02:58.47funkknobMe too, my powerbook is back in the states :(
02:58.49dima202mine is sexier
02:58.59JamesDotComnothing is sexier than a 12" powerbook
02:59.15dima202yeah, a 17"
02:59.18dima202hehe
02:59.22JamesDotComnah
02:59.25JonR800the 12" screen needs updating bad..
02:59.29JamesDotComi'd take the smaller one any day
02:59.32JamesDotComyeah
02:59.37JamesDotComi wish it could do more than 1024x768
02:59.40JamesDotCombut i'm happy
02:59.42JonR800I'm hoping January revisions are WS
02:59.53JonR800WS G5 12".. mm ;)
02:59.56JamesDotComos x looks a lot better than other os's with a smaller resolution
03:00.13*** join/#asterisk skeeziks (~skeeziks@r80h128.res.gatech.edu)
03:00.19JamesDotComheh, i dont need anything more powerful than this, all i run is terminals and a browser
03:00.23JamesDotComshould last me a good 5 years at least
03:00.40dima202I like to game some time
03:00.51dima20217  is great for gaming sometimes
03:00.54JamesDotComthat's what the xbox is for :)
03:01.02dima202Don't have it
03:01.08dima202I had a choice one day, a car or p b
03:01.26JamesDotCompb! :D
03:01.29dima202no xbox
03:01.34dima202yeah...
03:01.38JonR800the 17" is beautiful.
03:01.44dima202Thats how crazy I am
03:01.47JamesDotComtoo big
03:01.49JamesDotComhaha
03:01.51JamesDotComi'd do the same
03:01.53JonR800i wish the 12" and 15" were as thin as the 17"
03:01.56JamesDotCombut i live in the middle of the city
03:01.59JamesDotComdont need a car
03:02.00*** join/#asterisk IOscanner (~IOscanner@24.0.186.72)
03:02.20JamesDotComthe 12" is thin enough
03:02.24JamesDotComit's all of an inch?
03:02.30JonR800more
03:02.57JamesDotCom1.18
03:02.57*** join/#asterisk jake_ (~jake@gw-kit.locore.ca)
03:02.58JamesDotComclose enough
03:03.06dima202BTW, aluminum is much better then TI
03:03.12JonR800that .18 is a lot in person :)
03:03.12JamesDotComamen
03:03.16dima202It doesn't wear out
03:03.17skeeziksDoes this mean anything to anyone here?
03:03.21skeeziksNov 15 21:58:42 WARNING[1092078512]: channel.c:1445 ast_indicate: Unable to handle indication 3 for 'SIP/6782380063-a6da'
03:03.24IOscannerwell it is how you use it...
03:03.38IOscannersorry I think I missed a few things....
03:03.54JamesDotComonly complaint with mine is that i've scratched the screen and the paint is wearing under where my palms sit
03:04.17JamesDotCombut i also love it for the fact that when i got the pb and started using the keyboard, my rsi went away
03:04.28dima202can someone explain how to use asterisk?
03:04.37dima202Asterisk Ready.
03:04.39JonR800lol
03:04.41JamesDotComdima202: what stage are you at?
03:04.42*** join/#asterisk edguy (~edguy2@host-24-225-213-218.patmedia.net)
03:04.51JonR800dima202: carefully.
03:04.54dima202I don't even know what this does yet
03:04.54JamesDotComwell, start reading voip-info.org and start editing the configs in /etc/asterisk/
03:04.57funkknobHe just got the process running ;)
03:05.11dima202But I have an idea how to make free* calling
03:05.26dima202and if that works, I will post
03:05.32funkknobDo you have a phone?
03:05.40funkknobIP phone?
03:06.05dima202I don't I thought this is soft. replacement>?
03:06.19JamesDotComhrmm
03:06.25funkknobWell you can run a softphone
03:06.27dima202I was wrong?
03:06.32JonR800You have a lot of reading to do.
03:06.40funkknobAsterisk is a softswitch
03:06.43dima202Argh! I knew it!
03:06.49JamesDotComthere's an x-lite client for os x
03:06.50funkknobNot a phone
03:06.59JamesDotComget that, and you'll be able to get it to call to an asterisk server
03:07.21JonR800dima202: http://www.voip-info.org/wiki-Asterisk
03:08.10*** join/#asterisk michael12345 (~mick@staff2.tsn.cc)
03:08.24michael12345I am told asterisk can auth off radius
03:08.47michael12345is this true
03:09.48mishehuI heard rumors
03:09.52JonR800http://lists.digium.com/pipermail/asterisk-users/2004-March/040837.html
03:10.17mishehudammit, can't seem to find a heatsink that fits this mobo+proc
03:10.31dima202guys, lets say my isp offers voip through cable modem
03:10.34dima202and I have server
03:11.03freestyle_networanyone here experienced with SIP DTMF?
03:11.56mikegrb:D
03:12.38*** join/#asterisk daveGS (~none@CPE000625dbadc2-CM014280007905.cpe.net.cable.rogers.com)
03:13.02funkknobdima202: no need to subscribe to your provider's voip service
03:13.21funkknobThere are plenty of providers around
03:13.27funkknobAnd free solutions too
03:13.27dima202I know that
03:13.29scrubbany broadvoicers around?
03:13.38dima202free solutions?
03:13.40scrubbtryign to patch against todays tree.
03:13.42dima202already available?
03:13.54funkknobhttp://fwd.pulver.com
03:13.57funkknobI think
03:14.20*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
03:14.36funkknobYou can get free inbound numbers in US, UK and several others
03:14.53funkknobCalling to PSTN you can get flat-rate packages
03:15.30dima202WHat is astericks for then guys?
03:15.52funkknobhttp://www.ipkall.com/
03:16.01funkknobIt's a phone switch
03:16.16funkknoband pbx
03:16.20dima202WHat is the main funtion? briefly explained?
03:16.37funkknobIt routes calls between phones
03:16.57funkknobJust like the switch at your PSTN or GSM provider
03:17.17funkknobSo a network would look like this:
03:18.00funkknobIP Phone ----- Asterisk box --------provider(s) ------- PSTN or other IP subscriber
03:18.26funkknobAlso it can provide voicemail, conference calling, etc.
03:18.27dima202So do I really need it?
03:18.33funkknobProbably not
03:18.36JonR800dima202: how can we tell you that?
03:18.37dima202or is it like an addon
03:18.37funkknobBut it can be fun
03:19.14funkknobYou can use an IP phone at home, send calls to the PSTN using a Digium card or via IP to many different networks
03:19.50funkknobYou can have multiple IP phones at home and ring diff. extensions within your house/business
03:19.54JonR800It makes the thing with the handset go ring ring!
03:20.00skeeziksHow stable is chan_sip2 vs. chan_sip?
03:20.02mlh407if I have 3 phone janks all on one line, and someone is on jack #1, can I disconnect the call from jack #2.  I have tried crossing the pairs but that does not do it, it only "flashes" the line.  Is there anyway to disconnect the call?
03:20.03funkknobAnd have voicemail, IVR, etc.
03:20.03*** join/#asterisk jcrock7 (~jcrock7@209.130.128.34)
03:20.08freestyle_networanyone use the nagios plugin for asterisk?
03:20.36*** join/#asterisk gregk (gregk@CPE0080c832cd7b-CM014340017512.cpe.net.cable.rogers.com)
03:20.46funkknobmlh407: yes, but you need a special little inline adaptor
03:21.06funkknobmlh407: on each phone
03:21.19mlh407funkknob: how do I make/buy one?
03:21.30funkknobHmmm it's been awhile, lemme look
03:21.58*** join/#asterisk int19h (Miranda@219.95.154.87)
03:23.40daveGShi there, anyone know how to use a softphone with vonage without paying for there extra softphone line
03:24.05daveGSThey gave me a Linksys RT31P2 Device
03:24.08mikegrbdaveGS: haw haw
03:24.34daveGSis that a yes i know how haw haw or a good luck youll need it one
03:25.49mikegrbnumber two
03:26.03*** join/#asterisk PilotPTK-Home (~trillian@pcp02587377pcs.shlb1201.mi.comcast.net)
03:27.25funkknobmlh407: Still looking, it's been years since I saw those things....
03:27.32mlh407funkknob: thanks
03:27.44tzangerfunkknob: what the things that disconnect other phones when a call is being made?
03:28.03ChulJinit seems perhaps the opposite...
03:28.15ChulJindisconnected a call in progress on another jack on the same line
03:28.36tzangersounds like some kind of device priority module
03:29.28mlh407I need it for my kids
03:29.45tzangermlh407:  why... TDM440P and you have 4 lines :-)
03:30.13mlh407tzanger: I use asterisk at work, I really don't want to use it at home.  But you are right, that would also work ... "soft hangup ..."
03:30.24tzangerno no
03:30.36tzangerI mean you could carry 4 simultaneous conversations
03:30.48tzangerprovided you only needed one POTS line
03:31.11AlexZanderAll these companies that provide local telephone numbers - are they just getting a bank of ISDN lines and hooking them up to an Asterisk type box to act as the bridge between voip and POTS or is it more complicated than that?
03:31.13dima202I am having a problem registering everything right to the prog
03:31.16mlh407I need to disconnect then when they are on late
03:31.26*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net)
03:31.34dima202All they give me is SIP Phone Number, SIP Proxy, e-maail and pw
03:31.41dima202Is that enough info to recieve calls?
03:31.51tzangermlh407: meh, I just instill the fear of god in my kids
03:32.00tzanger"I catch you on the phone late and you lose it."
03:32.22mlh407yes but I sleep too well for that....
03:32.42tzangerheh
03:32.57tzangerwell you could always put a relay in line with the tip or ring
03:32.59tzangerhell X10 it
03:33.05funkknobmlh407: http://www.sandman.com/lineshar.html - Maybe something there you can use, scroll down to exclusion privacy devices
03:33.30tzangerthe rat shack modules would work too
03:33.36dima202guys, is that sufficient info>?
03:33.36tzangerjust pick up your phone to disconnect the others
03:33.39PilotPTK-Homeuse a script.  absolutetimeout(23:00 - now)
03:33.41PilotPTK-Homebasically...
03:33.49*** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de)
03:33.55PilotPTK-Homeand then have the context they dial out from only include during the hours you want them to be able to call.
03:33.56funkknobmlh407: You may just need to rewire your house, and put a switch in your bedroom
03:33.57dima202I put username and Authorization username the phone number I was assigned
03:34.01PilotPTK-Homethen you can still allow them 911 and other things after hours.
03:34.42PilotPTK-Homethat way, at 11 (assuming thats the time you want to use), the phone would disconnect, and they could only dial a few selected numbers that you allow.
03:34.48tzangeryeah speaking of 911, I didn't realize that 911 was not working since we moved in June... oopsie
03:35.16doughecka_tzanger: oh well, that person who had the heart attack wasnt liked anyway
03:35.22tzangerdoughecka_: heh
03:35.31tzangerit's not like there aren't a dozen cell phones in the place
03:35.35doughecka_yea
03:35.42doughecka_till the cell towers are taken out
03:35.44tzangerand it's working now :-)
03:35.45doughecka_:P
03:35.54tzangerwell when the towers are taken out there will be bigger issues
03:36.06doughecka_yea :P
03:36.19tzangerif your cell towers disappear it's likely your internet connectivity will die too
03:36.56tzangerI should play with DUNDi more...  how stable is it in terms of call quality?  I mean if you're just using anyone's lines I imagine there'd be a lot of change in quality from call to call
03:36.58doughecka_this week I am in vmware training, and theres this guy that works in nashville who is in my same class... he was talking about hot sites and stuff like that...
03:37.06doughecka_actually I have never used dundi :P
03:37.23doughecka_well, basicly, if the hospital gets taken out... what use is the client data gonna be?
03:37.26tzangervmware training?
03:37.30tzangerwhat exactly do they train you on?
03:37.34doughecka_eh
03:37.37doughecka_vmware ESX server
03:37.43tzangerok :-)
03:38.05tzangeris there that much of a speedup from eliminating the host OS with a slim vmware scheduler and I/O abstractor?
03:38.13doughecka_speedup?
03:38.24tzangerdifference between GSX and ESX
03:38.29doughecka_oh
03:38.38doughecka_well, ESX is redhat 7.2
03:38.49funkknobdima202: You can receive calls from other users on the system, if you want to receive PSTN calls, try http://www.voip-info.org/wiki-Telesthetic
03:38.50doughecka_with ANOTHER kernel sittin beside it doing direct hardware stuff
03:38.53tzangerIIRC GSX runs on a host OS still, whereas ESX uses a thin kernel that pretty much abstracts I/O only
03:38.58tzangerahh maybe I have GSX and ESX mixed up
03:39.05doughecka_no, you are correct
03:39.17doughecka_but ESX is a contained package
03:39.21doughecka_gsx is vmware only
03:39.23dima202You mean it is not a real number?
03:39.26tzangerahh
03:39.28dima202just like a nick
03:39.38doughecka_esx is 80%-98% cpu speed
03:39.53doughecka_gsx is more like 60%-80%
03:39.56dima202wonder why they speak of washington numbers currently free
03:40.13tzangerdoughecka_: GSX is slower than ESX?  WTF's the advantage then?
03:40.28doughecka_esx is more expensive
03:40.41tzangerbut esx runs on a host OS?
03:40.43doughecka_gsx is far cheaper
03:40.47tzanger<-- confused
03:41.01tzangerperhaps I need some vmware training :-)
03:41.02doughecka_ok, you have 3 products
03:41.05doughecka_workstation
03:41.07doughecka_gsx
03:41.09doughecka_and esx
03:41.36tzangeryes I use workstation at least twice a week
03:41.38funkknobdima202:
03:41.41doughecka_gsx == a better workstation
03:41.42funkknobdima202: http://phone.ipkall.com/ipphone/
03:41.54tzangerplayed briefly with either GSX or ESX on slackware
03:41.56doughecka_with a web gui
03:42.01dima202I am getting time outs when  i TRY TO log in
03:42.06doughecka_then you played with gsx
03:42.08dima202though the phone prog
03:42.24tzangerand ESX's scheduler/abstraction is just RH7.2??
03:42.50doughecka_well, yes and no
03:43.03doughecka_it runs redhat 7.2 as the base OS, for basic booting...
03:43.06funkknobXlite?
03:43.11dima202yes
03:43.14dima202can I pm you?
03:43.21doughecka_but theres a vmkernel that runs beside the redhat kernel
03:43.22funkknobsure
03:43.37doughecka_and it accesses most hardware directly
03:44.01doughecka_NICs, drives, SANs, long range photon cannons, etc
03:45.41tzangerheh
03:46.03doughecka_so the vmkernel adds a whole level of speed increase
03:46.07daveGSwhat is a good voip provider that doesnt have vonages restrictions
03:46.13tzangerdaveGS: nufone
03:46.26tzangerI use them pretty much exclusively with vpc as a backup
03:46.30daveGStzanger: thanks
03:46.30doughecka_while GSX (running on a windows/linux host OS) offers a cheaper, but less performance product
03:46.34tzangerand then my PRI as a backup backup :-)
03:46.41tzangerdoughecka_: right i Understand now
03:47.17doughecka_GSX DOES offer more hardware support, anything the host OS can support, but ESX supports fiberchannels and junk like that.
03:48.21tzangerright right
03:48.37*** join/#asterisk brettnem (~brettnem@user-0ccsr4b.cable.mindspring.com)
03:48.38h3x0resx is for those who want to spend a shitload of money to save money doing virtual server colos
03:48.38h3x0rheh
03:48.58doughecka_well...
03:49.10doughecka_and other people
03:51.42*** join/#asterisk Kb1_Kanob (johnsmith@sec2d7.dial.uniserve.ca)
03:52.09int19hword up all
03:52.09int19h, is anyone anywhere in the world having trouble getting calls through IAXtel, in either direction?
03:53.24Kb1_Kanobevening all.
03:53.39Kb1_KanobHas mantis started eating it's own children? I seem to have lost a bugreport...
03:54.48gregki've given up on IAXtel a week ago.  then again I never had it running before, so maybe I was doing it wrong
03:55.47mishehuI'm going to kill intel.
03:56.19gregkAny bug marshalls out here?  How do I go about having a bug reopened?
03:56.36mlh407What bug #
03:56.38redder86gregk: just repost to the bug report, it will reopen
03:56.39JunK-Ygrefk: which #?
03:56.45int19hhmmm, I had it working great... till my ISP (pretty much the only broadband ISP in malaysia) decided to grciously 'upgrade' all it's customers... 384->512... 512->1M.... and since IAXtel been broken
03:56.47int19hbah
03:57.07gregkbug_id=0001822
03:57.07kramwhat's wrong with iaxtel btw/
03:57.12kramwhat is the symptom?
03:57.23kramit's on my list to work on today
03:58.16gregkmlh407, Junk-Y: bug_id=0001822 " Quick Net Internet Phonejack Goes Bezerk"
03:58.26twistedkram: it's up and down and up and down and up and down
03:58.33twistedand generally nobody can make calls through it
03:58.53JunK-Ygregk: maybe twisted could re-open it.
03:58.57JamesDotComis there a way to get more debug output from the h.323 driver?
03:59.10twistedgregk, one sec
03:59.13BoRiShey kram, Can you check the callerid column from a database in realtime config? It doesn't seem to be taking the name to show as callerid (SIP dialing). Also, MWI doesn't work with realtime config. :(
03:59.32redder86Use iaxtel for real calls?  Hah!
04:00.00twistedgregk, i closed that out over 2 months ago, because it was inactive for almost a month before that.
04:00.08int19hkram: this is as far as I can get... Called iaxtel-outbound/18009797900@iaxtel... and then WARNING[1096813488]: chan_iax2.c:1473 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/iaxtel-outbound/16384 (type = 6, subclass = 9, ts=12610, seqno=2)
04:00.43twistedgregk, i'll re-open it, but it will get closed again if nothing happens for awhile.
04:01.04twistedgregk, it's open.
04:01.13*** join/#asterisk adavee (adavee@S01060090f8009e11.va.shawcable.net)
04:01.18redder86yup... iaxtel is down
04:01.43gregktwisted:I can reproduce it very easily.  We'll see
04:01.51gregktwisted: thanks
04:01.51twistedgregk, okay. post away :)
04:02.09adaveeis it ok for a newbie to ask a couple questions?
04:02.15*** join/#asterisk outtolunc (~chatzilla@adsl-69-110-40-71.dsl.pltn13.pacbell.net)
04:03.44Kb1_Kanoba good evening question - Is incoming audio data from a zaptel interface passed through a high-pass filter anywhere to remove DC offset effects? I have noticed the "base" signal level on my calls seems to vary wildly - yet my end is PRI. So, the serving switch must be passing me a line artifact from the other parties analog line, no?
04:03.50h3x0ri should set up a iaxtel clone, plus DIDs
04:03.57h3x0rits probably not a bad idea
04:04.13h3x0ri've got a couple local PRIs im not doing anything with
04:04.21Kb1_Kanobadavee: usually best to just ask, albeit politely...
04:05.22adaveeI just signed up for Voip and am severing my two telco lines (one for us and one for the kids).  I'd like to get some kind of simple setup where calls to the one voip # could be redirected to the kids phone.  I thought a simple PBX setup might work.  Would Asterisk work well for this kind of setup?
04:05.30Kb1_Kanobfollowing on from the "dc offset", I get complaints of bad audio on some calls that are pumped over a gsm codec. Not all though - could a reduced dynamic range be a source of problems?
04:05.32*** join/#asterisk abombss (~abombss@c-67-163-3-0.client.comcast.net)
04:06.05redder86adavee: yes
04:06.12h3x0radavee: many voip providers lock you to their hardware and won't let you use your own software/hardware
04:06.12Kb1_Kanobadavee: how would you differentiate between the calls?
04:06.19h3x0rsuch as vonage
04:06.21adaveeI just wasted some $$ on a VoicePro 206 that doesn't seem to work....so searching around led me to Asterisk
04:07.12adaveesomeone would call my voip # and then the voicemail attendant would ask .."for the kids, hit 5" or something like that??
04:07.35redder86adavee: yes, if you wanted, you could do that
04:08.48Kb1_Kanobadavee: troll through the voip-info.org wiki a bit. Look at implementing an "IVR"
04:09.06adaveethanks, I will
04:09.14redder86adavee: your VP206 doesn't work?
04:09.23adaveeno, it doesn't seem to
04:09.36adaveeit just sits there with the power light on
04:10.12redder86adavee: what's it supposed to do?
04:10.55adaveeits supposed to give me a dial tone when I pick up a phone...and then it can direct calls to other extensions in my house - it has 5 phoune outlets
04:11.07adaveebut it does nothing
04:11.19redder86got it on eBay?
04:11.44adaveeno, not directly
04:12.25redder86well, I see them on eBay for $170.  You'll spend at least double that for your setup with Asterisk.
04:12.59adaveeI paid $100 for it but a lesson learned - I'd rather do it the "right" way  :)
04:13.31redder86adavee: you'll need one or two FXO ports (~$20 each), and then an ATA device or two ($100 each).
04:13.51redder86adavee: you have broadband?
04:14.00adaveehigh speed cable
04:14.18adaveeI have an ATA device from Broadvox - you mean that?
04:14.26redder86possibly
04:14.40redder86is it a SIP device?
04:15.05adaveehmm, not sure what that means..
04:15.22tzangerany recommended places to purchase Polycom IP300 phones?  Looking for about a dozen of 'em
04:15.24redder86What is the model number information?
04:15.47adaveeMediatrix 2102
04:16.18Mavvieseems that my question with regarding to modem emulation for SIP/IAX was too difficult :-/
04:16.32*** join/#asterisk rje (~rje@c-67-160-57-179.client.comcast.net)
04:16.52Kb1_Kanobanyone familiar with the concept of "padding" a PSTN line? either digital or analog forms?
04:17.13redder86adavee: "Deployable in SIP VoIP networks"  that's a yes
04:17.32adaveeyou just searched for that?  :)
04:17.51redder86adavee: www.voip-info.org, then I went to the manufacturer's website, yet
04:17.52redder86yes
04:18.19redder86Mavvie: what are you doing?
04:18.37redder86adavee: do you want to keep your existing PSTN number?
04:18.50adaveeno, that doesn't matter
04:18.52redder86adavee: do your kids have their own number that they solicit?
04:19.25redder86adavee: well you wouldn't need Asterisk, then, unless you wanted it.  You should be able to set up your Mediatrix with a VoIP provider.
04:19.37adaveeI was thinking that the kids would just get an ext. # off the main voip #
04:19.40redder86adavee: they would provide you a number, called DID
04:20.04Mavvieredder86: http://lists.digium.com/pipermail/asterisk-users/2004-November/072668.html
04:20.38Mavvieredder86: the idea is to have some kind of modem emulator device which does do the calls via a SIP link instead of a PSTN link
04:20.39redder86Mavvie: okay, gotcha, but what are you trying to do with the emulated modem?
04:21.00redder86slePP: around?
04:21.13Mavvieredder86: I'm trying to get rid of a BRI card ;-)
04:21.19Mavviewell, that's the effect of it.
04:21.20rjeADSI question... After posting to the users list and thy helped me part of the way I still cannot get a greeting/idle screen to show up on my 390.  I've got the slots set right and the keys show up when i go off hook.  Just no idle gretting.  Anyone have some ideas?
04:21.21adaveeok, I'll read up a bunch at the listed sites!
04:21.23redder86Mavvie: for faxing?
04:21.35Mavviedialin / dialout and faxing.
04:22.31Mavvieredder86: it's just that it's easier for fall-over that I don't have to worry about too many ISDN connections. One PRI is cheaper than four+ BRIs.
04:22.39redder86Mavvie: most of the code for faxing exists already in spandsp.  slePP was working on getting an IAX client to connect with the spandsp libraries, which already has a T.31 modem available, and then interface that with a device node.
04:23.17Mavvieoh! that's interesting.
04:23.17int19hhey, has anyone seen an * app that can apply audio filters to a call? like reverb or distortion or whatever?
04:23.17redder86Mavvie: as for data calls, I don't know anything about that.
04:23.42Mavviebut only fax. data calls is then still an issue.
04:24.08redder86Mavvie: coppice, the spandsp author, was going to make a pseudo-modem channel driver instead.  So it would provide pseudo-modems on the same system as Asterisk.
04:24.27redder86Mavvie: spandsp may work for data, but I don't know.  I only concern myself with fax.
04:24.41Mavvieredder86: aha, that sounds like a way to go for legacy apps which want to talk to /dev/something.
04:25.09*** join/#asterisk rustyb (~rustyb@68.235.250.116)
04:25.21redder86Mavvie: the only difference, in the end-result, I think between slePP's and coppice's approach is that slePP's would provide device nodes on remote systems.
04:25.36Mavvieredder86: I'm going to figure out if I can get it up and running, will at least get rid of three of the four cards if I get faxing running.
04:25.50redder86Mavvie: coppice worries about frame slips if the machine is remote
04:26.10redder86Mavvie: I don't think that there is anything in a finished state, yet.
04:26.38dima202Discovered Full Cone NAT Firewall
04:26.41dima202is that bad?
04:27.11redder86Mavvie: right now to integrate Asterisk and faxing you've got to either connect a separate fax device to an FXS port on Asterisk, or you have to use spandsp's txfax/rxfax.
04:28.07daveGSnufone sux they have horrible service, cant even sign up for a phone number
04:28.21redder86daveGS: NuFone's been good to me.
04:28.31twisteddaveGS, if you say so
04:28.38twistedi've not had a problem
04:28.42adaveeone last question, can I use my existing cordless phones with Asterisk, or do I need specific phones?
04:28.59h3x0rhttp://www.knopsterisk.com/
04:29.01tzangerdaveGS: you just asked me not even an hour ago who to use ...  and now you're an expert on nufone?
04:29.04h3x0rthats violating so many gpl's
04:29.07h3x0ri dont know where to begin
04:29.08redder86adavee: the purpose of an ATA device is to allow you to use regular old phones with VoIP
04:29.23adaveehehe, ok
04:29.47dima202so guys what does this mean? Discovered Full Cone NAT Firewall
04:29.56dima202and Discovered Blocked Firewall
04:29.58redder86h3x0r: how does it violate the GPL?
04:30.07heragmy vsp gave me a patch to chan_sip.c the other day and is shoving it down my throat, if I just patch and recompile that one file, can I just replace it's respective .so in the /var/lib/asterisk/modules dir and just restart *? or do I have to go through the entire recompile process?
04:30.35tzangerahh herag's another happy broadvoice customer
04:30.43heragya, no frick
04:30.47h3x0ryou can't download it
04:31.03heragI'm getting a little pissed
04:31.18redder86h3x0r: does the GPL require it to be downloadable?
04:31.45heragI finally had a little time to work on configuration of my extensions, and now they go about screwing with my registration functions....my server won't even register with them properly now
04:32.01tzangerherag: in theory you can just do the one file, unload chan_sip.so or whatever it is and reload it without taking all of * down
04:32.04redder86herag: try the former and if that doesn't work then the latter
04:32.21tzangeranyway it is my beddy bye time, later all
04:32.49redder86herag: is Broadvoice going to submit a bug report on that thing or what?
04:33.07Kb1_Kanobtzanger: gnight.
04:33.08heragyes, actually, they claim their patch will be incorporated into the cvs
04:33.15*** join/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net)
04:33.32redder86herag: that's what their e-mail claims, but they have to jump through Digium's hoops to get it there.  I wonder if they will.
04:33.40tzangerKb1_Kanob: :-)  Didn't see you there, 'night
04:33.59Kb1_Kanobquietly slipped in the back door...
04:34.09heragredder86: well, supposedly, the patch was made by astricon devs, so I dunno
04:34.18tzangerherag: well it was a * dev who made the patch for them and the patch is good, I don't doubt it is already in CVS HEAD
04:34.36tzangerin fact I am certain I saw email about it in the asterisk-cvs list
04:34.50skeeziksAnybody have this problem with SIP? WARNING[1092078512]: channel.c:1445 ast_indicate: Unable to handle indication 3 for 'SIP/6785550063-dee1'
04:34.54redder86herag: the noise on the -users was confusing
04:35.07tzangersomething about "fixing olle's sneakiness" or sommat :-)
04:35.09heragolle johansson and steve sokol wrote the patch
04:35.11tzangeranyway 'night for real now :-)
04:35.38heragthis is the error I'm getting now when trying to dial out: app_dial.c:743 dial_exec: Unable to create channel of type 'SIP'
04:35.44heragwhat the frick does that mean?
04:35.47redder86I would have thought that if it were in CVS HEAD that Broadvoice would have told their customer base to upgrade to CVS HEAD rather than distributing a patch to all of them.
04:36.07redder86herag: it means that SIP isn't working
04:36.28heragredder86: is that a source problem, or registration problem?
04:37.01redder86herag: not sure
04:37.04heraggrr
04:37.16redder86herag: I would guess a SIP configuration problem
04:37.28redder86kram: was that Broadvoice patch integrated into CVS HEAD?
04:37.29heraghmm
04:38.00*** join/#asterisk adavee (adavee@S01060090f8009e11.va.shawcable.net)
04:39.32Mavvieh3x0r: I'm not really sure how it violates the GPL...
04:40.04Mavviethey say it's plain asterisk, and their FAQ shows how to make one yourself.
04:40.20heragsigh...
04:40.31heragit's the patch's fault
04:40.35heragI gotta recompile everything
04:40.40heragfricken bv
04:40.49redder86hehe... there's like 100 votes' difference in the gubenatorial race in WA
04:42.17adaveeok, I'm looking at pic of a FXO module @ Digium - I've read that I'll need one - it doesn't look like it goes in a PCI or ISA slot, - how are the connected?
04:42.54redder86adavee: those go on the TDM cards.  You want the X100P
04:43.10adaveeok
04:43.44adaveeahh, there it is
04:45.06redder86adavee: the X100P is an Ambient/Intel winmodem (AMI-IA92/IE92) for which Asterisk has its own driver and comes with installation/setup support from Digium
04:46.12*** join/#asterisk tetraz (~asko@S010600067b03be4c.vs.shawcable.net)
04:46.16adaveedoes anyone have any user websites that show some detailed steps in setting up a small home steup - even just one line?  I'm just so new to all this.  But I'm eager to learn.  I should only need one X100P with one line correct?
04:46.29*** join/#asterisk charlesIII (~charlesII@c-67-161-110-59.client.comcast.net)
04:46.36redder86adavee: yes, one x100P for one line
04:46.46tetrazIs there an asterisk newbie channel?
04:46.53adaveehaha
04:46.54charlesIIIanyone have spandsp working with freebsd?
04:46.56PilotPTK-Hometetraz: www.voip-info.org
04:47.08adaveeI'm reading lots there right now
04:47.28adaveestill quite confusin |)
04:47.31JunK-Ytetraz: ask ur questions.
04:47.48tetraz:) How do I know if asterisk recognized my soundcard?
04:47.58adaveein a nutshell this would be my simple setup:
04:48.03redder86adavee: unfortunately the entry level learning curve requirement for Asterisk is quite high
04:48.20tetrazredder86: No kidding :)
04:48.35redder86adavee: voip-info.org has some good links.  Don't trust everything you read there (or anywhere else, either), though.
04:49.01*** join/#asterisk florz (nobody@I811c.i.pppool.de)
04:49.04JonR800does anyone else have trouble with voipjet taking about a full minute to dial out???
04:49.24redder86adavee: even if you're in-the-know with Linux and telephony you'll probably spend many, many hours getting a basic Asterisk system going
04:49.42adaveeinternet connection --> cable modem --> hub/switch --> one connection to Asterisk box in --> Asterisk box out to my cordless phones
04:50.17adaveeother connection --> router --> my main computer
04:50.20charlesIIIfreebsd --- spandsp... anyone?
04:50.58redder86adavee: this looks promising: http://www.surfcity.com/Asterisk/default.htm
04:51.43redder86charlesIII: s/freebsd/linux/ ?
04:52.07charlesIIIone can hope, you know?
04:52.18charlesIIInah, specifically looking for freebsd
04:52.36BoRiSAnyone know why when I try to dial an H323 phone (exten => 123,1,Dial(H323/1800xxxxxxx@192.168.x.x). When I dial 123, My console keeps showing "Nov 15 22:53:34 NOTICE[16306]: pbx.c:1341 pbx_extension_helper: Cannot find extension context 'default'" even though I just want it to dial?
04:53.12redder86charlesIII: the way that I see it, the box is there for the purpose of running the applications, and if the application that you need to run is Asterisk+spandsp, then well, Linux is the OS you should use.  Likewise, if you want to run MS Access... well, then you're going to need to run Windows.
04:53.39charlesIIII don't think inquiring about ports of software is too weird, considering asterisk runs well on freebsd
04:53.45redder86BoRiS: you need to have a context where it can dial from
04:54.09redder86charlesIII: but spandsp doesn't
04:54.23charlesIIII am asking if someone has ported it or is in the process...
04:54.28redder86oh
04:54.56redder86your message "freebsd --- spandsp... anyone?" was unclear about that
04:55.01charlesIIIsorry
04:57.51*** join/#asterisk hmodes (hmodes@dsl092-231-153.phl1.dsl.speakeasy.net)
04:58.41hmodesoy
05:02.20abombssanyone in here use kdevelop with *
05:02.49abombssor anyone else use any other ide?
05:04.30kramwell you can restart yah
05:05.57dima202hey guys, is this site for real? http://phone.ipkall.com/ipphone/
05:06.19dima202DO they really give you a phone number and can be reached from land line phone for free?
05:06.26*** part/#asterisk PilotPTK-Home (~trillian@pcp02587377pcs.shlb1201.mi.comcast.net)
05:06.43kFuQdima202:  yes
05:06.45brettnemdima202: I used them.. it works
05:06.50kFuQi have 2 #'s
05:06.53dima202I dont understand it
05:06.54kFuQthru them
05:07.07dima202I configured everything properly to make calls
05:07.16dima202but not to recieve using their service
05:07.31brettnemI think they are termination only..
05:07.36brettnemer.. origination
05:07.43dima202360-519-5157
05:07.50brettnemmeaning they send calls to your did
05:07.50dima202here is my number they issued
05:08.18dima202I dont even know where to include the info config of SIP Proxy an d SIP Phone, PW
05:11.35dima202How do I tell them to use my farward number so I can reach the call?
05:15.28dima202can anyone help me set this up
05:19.30*** join/#asterisk Moc (~mochouina@modemcable212.49-80-70.mc.videotron.ca)
05:19.45Mochail
05:19.54Dr_RayMoc yesterday you talked about a dial around 411 provider, any name of a company?
05:20.10MocDr_Ray, I just play a prompto use 411.ca
05:20.25Mocsomeone talked about a provider that sold in bulk or something
05:20.40Dr_Rayoops, thought that was you
05:20.41Dr_Raymy bad
05:21.30Moclet me check my logs
05:22.37Mocok found it
05:22.48daveGStzanger: ya i actually went to sign up with them, their web app doesn't register any phone numbers either michigan or toll free, theres no number for customer support
05:22.57daveGSthats why i say they suck
05:23.09Moc<AMAG> You can redirect them to a toll-free 411 service for like $0.30/call with no real volume commitment
05:23.40Mocthat all I see
05:23.58daveGSand they didnt even register me under the correct plan
05:24.30Dr_Raythanks Moc
05:25.29bkw_YAY asterlink is live
05:25.31bkw_:P
05:27.18Mocso what new today ?
05:27.37daveGSi just signed up for it with ipkall, its working fine
05:28.20*** join/#asterisk yaout (eric@CPE-65-30-220-56.wi.rr.com)
05:28.43Dr_RayDave in Seattle?
05:28.56MocI hope to get my polycom at the end of the month.. REally hope
05:29.10JunK-Ymoc: ip500?
05:29.11Mocbut I guess I'll have it by new year ..
05:29.18MocJunK-Y: IP 600
05:29.20JunK-Ynice
05:29.27JunK-Ygoing to bed, tty tomorrow moc.
05:29.29Mocyea I decided to try it out
05:29.34Mocallright cya
05:29.36JunK-Yhehe nice guess
05:29.39JunK-Y:)
05:31.08*** join/#asterisk nowork (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com)
05:32.41_Vilemmmmm
05:32.46_VileHI
05:33.48_Vilehi
05:33.50_Vilesdjld
05:33.53outtolunchowdy
05:35.27freestyle_networi have a 600 ..love it
05:36.57*** join/#asterisk ender (~me@ender.fedora)
05:37.40fOSSiLanyone know how to upgrade IAXy firmware on an early-release box?
05:38.05ManxPowerfOSSiL: You have to send it to Digium.
05:38.32ManxPowerI think digium can tell you based on the srial number of they have to do the first firmware upgrade
05:39.10ManxPowerI think Digium can tell you, based on the serial number, if they have to do the first firmware upgrade.
05:39.49_Vileman I keep taking these billing tickets and they don't seem to be stopping.. damn the top 10 newest unowned tickets..
05:39.54fOSSiLyeah, i got that, thanks :)
05:40.48fOSSiLi think this one definetely needs a digium upgrade -- the MAC addy is d00d00f00dXX
05:41.02_Viled00d f00d
05:41.13fOSSiLya :P
05:41.25ManxPowerfOSSiL: Yes.  That MAC address made me spend 2 hours tracking it down.
05:41.25*** join/#asterisk Guest^DJ (~mmmm@219.94.64.226)
05:41.32fOSSiLlol
05:41.58ManxPowerIt was on an unmanaged switch connected to an unmanaged switch connected to our backbone managed switches.
05:42.41fOSSiLheh, how the heck did you find it then?
05:42.43_Vilefun
05:43.07fOSSiLthe cable it was hooked up with did not have any dust on it? :)
05:43.09ManxPowerfOSSiL: Pinged the IP, then started unplugging stuff
05:43.14_Vilehahahaha
05:43.21_Vilei was just thinking that
05:43.40ManxPowerWe localzed it to MIS/accounting/marketing
05:44.27outtoluncsounds localized to me <G>
05:45.10Guest^DJanyone knows where to get FXS clones for * ?
05:45.15fOSSiLall network hubs and switches need a "send 10,000 volts to <this> port" function
05:46.13robl^at 10,000 watts
05:46.13NuggetI am in full support of that and also SITFoIP technology.  (Stab In The Face over IP)
05:46.13ManxPowerGuest^DJ: They don't exist.
05:46.13_VileGuest, you'll be happier with a t100p and a channel bank full of fxs's
05:46.13*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:46.13Dr_Raydigium derves the money
05:46.13Dr_Rayer, deserves
05:46.24ManxPowerouttolunc: 8-)
05:46.38ManxPowerCan I call Guest^DJ "cheap"?  Can I?  Can I?
05:46.48Guest^DJsure u can, ManxPower lol
05:47.01Guest^DJwell, place i am in T1 dont exist
05:47.02_Vileguest, they don't exist, and if they do, you'll have a bitch of a time with echo..
05:47.09_VileGuest, doesn't matter
05:47.26*** join/#asterisk |Blaze| (dirc@d142-59-247-192.abhsia.telus.net)
05:47.31_VileAnalog Lines <-> FXS Ports on Channel Bank <-> T1 <-> Asterisk
05:47.49_Viledepending on # of lines you're looking for
05:47.49ManxPowerGuest^DJ: A Channel bank splits a T-1 to Asterisk into 24 analog ports.
05:48.02_Vileyeah that
05:48.03ManxPowerTheefore T-1 no connection to the telco needed
05:48.03_Viletoo
05:48.05*** part/#asterisk int19h (Miranda@219.95.154.87)
05:48.21Guest^DJhmmmm, i am in asia, so only E1 is available here
05:48.43Dr_Raye1 gives you 32 ports
05:48.51Mavvie30
05:48.53Dr_Ray30
05:48.57Dr_Ray:)
05:49.07Dr_Raywhich I was tempted to use here in the states
05:49.15Guest^DJyup, i have a T410P
05:50.16Guest^DJDr_Ray, i am going nuts over *
05:50.24Dr_Raygood nuts?
05:50.50Guest^DJafter whole night of reading, dont know what the FXXXX is going on
05:51.00Dr_Raymore reading
05:51.07Dr_Rayhitchikers guide to asterisk
05:51.10ManxPowerUseful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/
05:51.13ManxPowerTo search the Asterisk mailing list archive go to www.google.com and put site:lists.digium.com in addition to your other query terms.
05:51.27Guest^DJthen i am gonna change from good nut to bad nut
05:52.55_Vilewhat are you trying to do, guest?
05:54.17Guest^DJjust trying to built 2 *, 1 with 3 FXO and 4 FXS on each end, and connect both * using IP
05:55.31_Vileand you have 1 T410P?
05:56.16Guest^DJoh, that 1 T410P is already running in my firm, it was built by a developer from here
05:56.42Himekonot all of asia is t1
05:56.45Himekoer e1
05:56.52Guest^DJnow trying to built analog based
05:56.57Himekobut most is
05:57.04Guest^DJHimeko, we have E1/t1 here
05:57.48Himekoi wouldn't think they woudl have both
05:58.02Himekot1 is NA and japan
05:58.05Himekoe1 everywhere else
05:58.22Corydon76-homeI thought Japan was J1
05:58.35ManxPowerJapan is J1
05:59.00Guest^DJE1 is common here, but T1 is also available here, but telco guys are trained with E1 knowledge
05:59.03Himekothey might call it that too
05:59.30Himekoi seen it called t1-Jsomthing
05:59.40Corydon76-homeIsn't J1 basically the same as T1, but with enough incompatibilities introduced to ensure that T1 equipment will not work there?
05:59.53Himekoprobably
06:00.02outtoluncthe question is not what you can do for your country, the question is what you can do with your channel
06:00.38Guest^DJwhy cant they just sit down and agree on the same standard
06:01.04Himekohehe
06:01.08outtolunci'll take ego's for $100
06:03.18HimekoThere are just a few minor
06:03.19Himekodifferences, designed purely to make it incompatible. They just changed
06:03.19Himekosome alarm flags and CRC patterns, if I recall correctly.
06:03.48outtoluncjust look at the ITU specs
06:04.05Himekoi guess taiwan used to use it too
06:04.13Himekocallignit a T1M
06:04.37outtoluncnot saying that US providers follow any better
06:04.38Himekobut when the telecom was deregulated there they use standard t1 and e1
06:05.14outtolunci faught with SBC abount SIT tones ..'hard' about 7 months ago
06:05.59outtoluncseems they have forgetten that 'some people' still require inband compliance
06:07.16outtoluncwouldn't they just love my publishing their in-compliance
06:07.59Guest^DJhi
06:08.06Himekoi guess 2 bits have to be twidlled in the framer for a t1
06:08.15Himekobut the isdn spec is slightly off too
06:10.22*** join/#asterisk brc_ (~bob@brc.base.supporter.pdpc)
06:10.23ZX81can you have host=dynamic with a peer? i.e. and have him register for peer and user account with one line? or do I have to create two register lines (i.e. 1 for peer and one for user?)
06:12.23ZX81ah well type=friend then
06:13.47ZX81compliance with
06:13.57outtoluncread up
06:14.01ZX81i.e. certifications / permits?
06:14.03ZX81ok
06:14.20ZX81ah yeah
06:15.01_Vilei don't like sbc
06:15.03_Vilenor qwest
06:15.26outtolunclike or dislike have nothing to do with this
06:16.53_Vileincompetancy and care-free ...? getting warmer?
06:16.59_Vilebad companies
06:17.34outtoluncis your phone provider SBC under the hood, is it working?
06:17.57*** join/#asterisk clive- (~pirch@myw-stp-66-18-85-82.sentechsa.net)
06:19.36outtoluncmaybe you aren't aware of the issues with 'detecting SIT'
06:19.52outtoluncinband
06:21.43outtoluncwell
06:26.59outtoluncITU q.35, e.182, e.181 should help
06:27.26WilliamKso who is everyone's favorite 1800 number provider?
06:27.57outtolunc_vile: any comments?
06:29.13ZX81does anyone know what the snom 190 is expecting for the music on hold server?
06:32.05outtoluncanyone else want to go head to head on issues regarding SIT detecting and thier dislike of thier local provider?
06:32.07*** join/#asterisk alphaque (~Alphaque@218.208.238.245)
06:32.25*** join/#asterisk davegsx (fuzzy@CPE000d6d60d-CM014280007905.cpe.net.cable.rogers.com)
06:33.53*** join/#asterisk serdiehard (~serdiehar@202.65.128.18)
06:35.24pfnSIT detection?
06:36.44outtoluncgoogle is your friend http://66.102.7.104/search?q=cache:FoltOTpWaR0J:www.ahk.com/Special%2520Information%2520Tones.pdf+itu+SIT+&hl=en&start=3
06:37.12outtoluncever heard of 'TRI-Tone'
06:37.32outtoluncda da dee
06:38.29outtoluncwe're sorry, the person you attempted to call has fled the country or is no longer available <G>
06:39.57outtoluncyou can't tell me all you were born after inband detection
06:44.08outtoluncthen again, i guess you can be <G>
06:45.18outtolunci seem to forget i'm a bit of a old fart here sometimes
06:48.21outtoluncand given that i've been the only one stating anything for what 20 lines, ... nevermind i'm just talking to my damn self
06:48.33outtoluncthought so
06:49.09outtoluncgoing back to other channels
06:57.20*** join/#asterisk scratchrf (~ryan@67-40-182-169.tukw.qwest.net)
06:57.33*** join/#asterisk ard (ard@goatse.kwaak.net)
06:58.21*** join/#asterisk wolfson (hehe@cpe-68-187-190-045.man.nc.charter.com)
07:00.04scratchrfif i start * from a "regular" login, then connect from my win box with putty and run asterisk -r, how do I disconnect from the -r session w/o killing * completely?
07:00.32ard^D maybe?
07:00.40ardquit?
07:00.57scratchrfthx, i'll try and let you know
07:01.09ardto get asterisk down you'll have to type down or something like that
07:01.12Slimeyquit or exit
07:01.27heragcan I kill an active call/channel?
07:02.04scratchrfquit works....
07:05.09ZX81herag: soft hangup <tab>
07:06.27heragit's cool the zombie channel just died on its own eventually
07:06.34scratchrfif i can do everything in webvmail but save preferences, probably a cookie issue?  i can see new vm's, move, etc. but w/o saving preferences I can't actually listen to them...
07:07.13scratchrf..boots me back to "login incorrect" login screen
07:07.37ZX81ctrl c
07:12.10brc_~seen jerjer
07:12.12jbotjerjer <~mine@d2-236.rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 4d 2m 18s ago, saying: 'no config necessary'.
07:12.34brc_~seen jerjer[mobile[
07:12.35jboti haven't seen 'jerjer[mobile[', brc_
07:12.37brc_~seen jerjer[mobile]
07:12.38jbotjerjer[mobile] <~jj@mail.nufone.net> was last seen on IRC in channel #asterisk, 14h 46m 47s ago, saying: 'paging'.
07:13.46heragwhat does the ACL A flag mean when I do sip show peers?
07:15.03*** join/#asterisk SuperMMan (~SuperMMan@clgrtnt5-port-123.dial.telus.net)
07:15.38SuperMManEvening all, I was wondering if anyone knows of any companies that are offering in/out dids in Edmontion Alberta?
07:19.14*** part/#asterisk abombss (~abombss@c-67-163-3-0.client.comcast.net)
07:20.27Himekocontact netmonks.ca
07:22.26ZX81anyone know why music on hold would not be able to start? with default mp3's?
07:22.32SuperMManHimeko,  ok thanx willd o
07:25.36*** join/#asterisk znoG (~speedy@134-134-126-200.fibertel.com.ar)
07:31.10skeeziksCan anybody give me a hand with gdb and Asterisk?  I'm having trouble getting symbol lookups to work for the various modules.
07:32.03*** join/#asterisk Stephie3 (Stephie@203.81.192.172)
07:32.14Stephie3hello
07:32.54Stephie3anybody know about quintum A800 configuration? plz msg me
07:38.06clive-stephie do they do sip yet those quintum boxeS?
07:43.04*** join/#asterisk RoyK (~roy@110.80-203-29.nextgentel.com)
07:47.41*** join/#asterisk juice (~juice@mo-205-240-40-86.dyn.sprint-hsd.net)
07:48.46_Vileclive, no sip
07:50.24*** join/#asterisk EisNerd (eisnerd@outpost.cyberadaptor.de)
07:51.26EisNerdis there a german channel?
07:53.14florzDunno about any, but how about opening one if there is none?
08:02.28*** join/#asterisk jas_williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com)
08:04.35Guest^DJ~seen Essobi
08:04.37jbotessobi <kstone@75.137.26.216.host.teledvance.com> was last seen on IRC in channel #asterisk, 129d 1h 48m 23s ago, saying: 'Freak. ;)'.
08:07.34*** join/#asterisk los415 (~los415@adsl-69-104-178-68.dsl.pltn13.pacbell.net)
08:10.00SuperMMan~seen TestMasTer
08:10.01jbottestmaster <~testmaste@S0106000ea67573e2.cg.shawcable.net> was last seen on IRC in channel #asterisk, 102d 11h 15m 19s ago, saying: 'sivana:  there is a company from here in alberta that says they can do it, but won`t give me a price lol'.
08:10.24*** join/#asterisk kivi (~martin@djz.net)
08:16.10*** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it)
08:22.43*** join/#asterisk RoyK (~roy@80.239.107.80)
08:27.54*** join/#asterisk scannachiappolo (~scannachi@81-174-16-211.f5.ngi.it)
08:32.14*** join/#asterisk libpcp (libpcp@210.16.20.5)
08:32.20*** join/#asterisk soulz2 (~Soulz-@host-137-132-45-241.imcb.nus.edu.sg)
08:32.57libpcphi everyone
08:33.12libpcpanyone has an asterisk write-ups? where can i get it?
08:33.59*** join/#asterisk ed0wn (ed0wn@dontvisit.us)
08:34.18*** join/#asterisk Michiel1983 (~Michiel19@81-17-62-133.dsl.uwadslprovider.nl)
08:34.48ed0wnI was wondering with asterisk...how does camophone.com use no special hardware?
08:35.24Dr_Rayclick to call
08:35.52Dr_Rayit calls you when it makes the bridge
08:35.53tafazziHi all. I would like my asterisk to find a free extension before answering the incoming call, to avoid having the caller paying the call to hear ringing the default extensions. Anybody has a dialplan suggestion for me?
08:36.44libpcpanyone has an asterisk write-ups? where can i get it?
08:40.09RoyKanyone that have used the queue_log here?
08:40.30*** join/#asterisk skeeziks (~skeeziks@r80h128.res.gatech.edu)
08:41.06skeeziksDoes anybody have oej's version of the chan_sip2.c patch that integrates the BroadVoice patch?
08:42.16skeeziksThe latest bugnotes says that he integrated the BV patch, but there's not a new attachment yet
08:42.47skeeziksI'm having trouble with call progress on a SIP channel once it's been answered, so I thought I'd give his patch a atry
08:42.51Michiel1983Hello all. I get a segmentation fault when I make a dail from my SIP softphone. I've a Winbond ISDN card with i4l driver the error messages are after a few seconds when I make a call  "Executing Dial("SIP/msn2-fed4", "Modem/ttyI1:...") in new stack" "Segmentation fault (core dumped)" Did anyone know what the problem is or where I can find information?
08:46.52*** join/#asterisk monst3r (~monst3r@adsl-59-29.swiftdsl.com.au)
08:47.40*** join/#asterisk CleanerX (AC142907@nat-ph3-wh.rz.uni-karlsruhe.de)
08:47.55skeeziksRoyK: Do you know of any sources of more info about chan_sip2?
08:48.57*** join/#asterisk Pryk (~tmalkut@host-ip226-209.crowley.pl)
08:50.01pfncvs
08:51.20RoyKpfn, I don't think sip2 is in cvs
08:51.32RoyKpfn, stuff from sip2 is being backported to cvs, that's all
08:51.40skeeziksYeah, it's not there
08:52.02*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net)
08:52.26skeeziksHave you guys heard of any problems with call progress once a SIP channel has been answered?
08:52.33pfnchan_sip2 is in cvs
08:52.36skeeziksI'm looking for a bug report but can't find anything
08:52.36pfnsearch for it
08:52.41skeeziksOK, looking
08:52.48pfnthere is no call progress once a channel has been answered
08:52.58*** join/#asterisk zoa (~zoa@213.16.46.130)
08:53.12pfn'cept hanging it up, maybe
08:53.21pfnor inband
08:54.05skeeziksWell, I answer a SIP call and prompt the user for a number, then dial out again on SIP, and I get no ringing or busy or anything, and this error:
08:54.32skeeziksNov 16 03:28:44 WARNING[14215]: channel.c:1451 ast_indicate: Unable to handle indication 3 for 'SIP/6785550063-7fe9'
08:55.28skeeziksAnd I don't see sip2 anywhere in CVS - where did you find it?
08:56.07pfnchan_sip2
08:56.16skeeziksOh, it's a module...
08:56.16skeeziksOK
08:56.30pfnoops, doid I say cvs
08:56.38zoaits on mantis
08:56.40pfnI meant mantis
08:56.44zoabugs.digium.com
08:57.41skeeziksYeah, I found it there, but oej mentions that he's made some changes that I want since the last version was posted - I guess I'll have to wait
08:57.50ed0wnsoftphone isnt working to test my damn asterisk setup
08:57.53ed0wni wonder what the hell i did wrong
08:57.55monst3rhow are we all this evening
08:58.19skeezikspfn: Any idea on that warning?  If I dial out a zap channel I hear ringing and stuff properly...
09:01.09pfnthe ringing should happen on the other channel, then
09:01.13pfnand you should get ringing just fine
09:01.20pfnfigure out what indication 3 is and why you can't play it
09:01.41skeeziksI've been trying to trace things through GDB but I'm having trouble getting all the symbols available
09:02.42skeeziksindication 3 is AST_STATE_DIALING
09:03.06pfnast_control_ringing
09:03.53pfnasterisk can't find the ringing tone for the zone that the channel belongs to
09:04.36skeeziksOK
09:04.42skeeziksSo what should I do?
09:05.39pfnI dunno, figure out your indication situation
09:07.35skeeziksBut you think the zone thing is the underlying issue?
09:10.15*** join/#asterisk darkskiez (~mbryars@usergc137.dsl.pipex.com)
09:11.06darkskiez<PROTECTED>
09:12.45posteldarkskiez: heh, funky idea
09:13.07*** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f)
09:14.01darkskiezYay, there is even Hal9000: (From 2010)There is a message for you
09:14.07monst3ris there away to make asterisk if you dial any number thats not part of the dial plan goto a default context, i thought using the bogon-calls context did this.. have i missed something?
09:14.21funkknobCalling voicemail, is this a codec issue? Nov 16 16:06:29 WARNING[1168520128]: file.c:550 ast_readaudio_callback: Failed to write frame
09:14.40darkskiezmonst3r, with the 'i' extension?
09:15.19monst3rhmmm ok im not using that so using the i, and tell it where to go will do that?
09:18.14*** join/#asterisk monkey- (user3@port-219-88-128-149.orcon.net.nz)
09:31.52darkskiezDISA times out far too quickly, the dialtone lasts for a couple of seconds and then dies.
09:32.39*** part/#asterisk monkey- (user3@port-219-88-128-149.orcon.net.nz)
09:33.19*** join/#asterisk r1 (~erwan@www.thiscow.com)
09:34.26*** join/#asterisk pif (~pif@zenon.apartia.fr)
09:44.21*** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it)
09:45.41Michiel1983Hello all. I get a segmentation fault when I make a dail from my SIP softphone. I've a Winbond ISDN card with i4l driver the error messages are after a few seconds when I make a call  "Executing Dial("SIP/msn2-fed4", "Modem/ttyI1:...") in new stack" "Segmentation fault (core dumped)" Did anyone know what the problem is or where I can find information?
09:48.28funkknobHi Michiel1983 - sorry I don't know
09:48.37funkknobSeems everyone's gone to sleep
09:51.54funkknobDoes anyone know the syntax for all the codecs in sip.conf and h323.conf? "show codecs" in the CLI does not - i.e. g.711a syntax is "allow=alaw"
09:53.22*** join/#asterisk jaXon` (~jaxon@ip116.66.1311D-CUD12K-03.ish.de)
09:54.41funkknobHi jaXon
09:54.50jaXon`hi
09:55.12funkknobDo you know the allow= for g.729a in h323.conf?
09:55.52funkknobI keep getting:
09:55.57funkknobNov 16 16:46:37 WARNING[1076298368]: Cannot allow unknown format 'g.729a'
09:56.47jaXon`what about allow=g729 ?
09:56.57funkknobthx I'll try
09:57.55funkknobHey that works! thx
09:58.32jaXon`np
09:59.21funkknobThe * process started with no errors, but the h323 segment is still using g.711u :(
10:00.03RoyKprolly cause it's h323 :P
10:01.24funkknobI'm using * as a protocol contverter, SIP phone ------ Proxy --------- * -------- AS5350 (H.323)
10:05.30*** join/#asterisk shadebob (~shadebob@ll81-134-144-192-81.ll81.iam.net.ma)
10:21.50Michiel1983Hello all, I want to call with a SIP softphone to an extern number with an ISDN card my dial command in extensions.conf is: exten => 21,1,Dial(Modem/ttyI0/69:00123456789). When I make a call I get:  "Executing Dial("SIP/msn2-e371", "Modem/ttyI0/69:00123456789") in new stack" - "WARNING[1144486832]: chan_modem.c:828 modem_request: Requested device 'ttyI0/69' does not exist" - " NOTICE[1144486832]: app_dial.c:743 dial_exec: Unable
10:23.00skeezikspfn: ARGH!  I was missing indications.conf and res_indications.so.
10:23.06skeezikspfn: Thanks for your help.
10:33.30*** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com)
10:41.36RoyKhi
10:47.29*** join/#asterisk pif (~pif@zenon.apartia.fr)
10:48.30ZX81how were you missing those files?!
10:48.49ZX81:-)
10:49.21RoyKlol
10:51.27RoyKdoes anyone know if asterisk supports T.38?
10:55.02*** join/#asterisk Mike_tk (~Mike_@213.180.245.62)
11:00.43[Sim]royk: it does not support t38 yet
11:01.10*** part/#asterisk PCadach (~paul@www.east.telecom.kz)
11:01.43RoyK[Sim], any idea how much it'll take to make it?
11:02.15[Sim]sorry, no idea
11:06.14*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net)
11:06.39*** join/#asterisk Jayjay (~Jasonrive@host213-120-116-134.in-addr.btopenworld.com)
11:06.45Jayjayhi peeps
11:07.08Jayjayanyone awake?
11:07.36RoyKnah
11:07.37RoyKsleeping
11:07.40Jayjayl0l
11:08.05ZX81playing counterstrike
11:08.10Jayjayi'm on Red Hat 9 (don't ask why yet) and i'm having some problems installing Asterisk....
11:08.20RoyKSIP:0xc0ffee
11:08.30Jayjay...while i make Zaptel, i get "zaptel.c:6213: variable `zt_fops' has initializer but incomplete type"
11:08.39RoyKJayjay, redhat kernel or stock kernel?
11:08.58RoyKThou Shalt Not Use Distribution Specific Kernels With Zaptel
11:09.19darkskiezAnd Jesus cried.
11:09.39RoyKisn't he dead?
11:09.47darkskiezs/Jesus/Jayjay/
11:09.47*** join/#asterisk oller (oller@nemec.strul.net)
11:09.52RoyK:)
11:09.58JayjayRoyK, i'm not entirely sure, its been on this machine for err... a couple of years, stuffed on a shelf
11:09.59ZX81jayjay is dead?
11:10.04ZX81nope
11:10.06ZX81:-)
11:10.21Jayjayi can only assume RedHat Kernel
11:10.28RoyKJayjay, uname -r
11:10.42ZX810.0.0.1a
11:10.50RoyKrotflmao
11:10.55Jayjayjust says 2.4.20-8
11:11.03ZX81cant be that old
11:11.06RoyKthat's the redhat standard
11:11.10RoyKredhat 9 standard
11:11.14Jayjayyup
11:11.16RoyK...and it REALLY REALLY SUCKS
11:11.23Jayjayi know, its not my machine
11:11.34Jayjayif i had my way, it would be debian or gentoo
11:11.38RoyKupgrade to 2.4.27 or 2.6.9 from kernel.org
11:11.51Jayjaybut i'm under orders of *they take too long to install so we're sticking with this for now*
11:11.59*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net)
11:12.19Jayjayok
11:12.25RoyKJayjay, then tell Them to Stick The RedHat CD Up Their Behind
11:12.31Jayjayl0l
11:12.35Jayjayits my old mans ;)
11:12.42Jayjaymight not go down too well
11:13.09Jayjaywhat am i going for 2.4 or 2.6
11:13.10Jayjay?
11:13.10RoyKwell. then gently tell them that Asterisk Can Not Be Installed In A Hurry On The RedHat Standard Kernel
11:13.15RoyKwhatever
11:13.23RoyK2.4 will be smoothest
11:13.29RoyK2.6 scales better, is better etc
11:13.38RoyKbut will require some other updates
11:16.22Jayjaydo we know what updates?
11:18.02RoyKread Documentation/Changes in the kernel source tree
11:18.14RoyKthat gives a brief list
11:18.28RoyKand a complete list...
11:19.40Dr_Rayis there a problem with fedora core 3 and asterisk? kernel 2.6?  I just installed it yesterday
11:19.54*** join/#asterisk speedwagon (~Ariel@fl-nked-ubr2-c6a-125.miamfl.adelphia.net)
11:25.27Michiel1983Hello all, I want to call with a SIP softphone to an extern number with an ISDN card my dial command in extensions.conf is: exten => 21,1,Dial(Modem/ttyI0/69:00123456789). When I make a call I get:  "Executing Dial("SIP/msn2-e371", "Modem/ttyI0/69:00123456789") in new stack" - "WARNING[1144486832]: chan_modem.c:828 modem_request: Requested device 'ttyI0/69' does not exist" - " NOTICE[1144486832]: app_dial.c:743 dial_exec: Unable
11:33.42PoWeRKiLLkram are you here ?
11:33.57*** join/#asterisk zyke (~xirak@host-212-158-223-3.bulldogdsl.com)
11:35.06ionixanyone knows how to add a "*" before an extension ? Trying to parse the 1-800 numbers
11:35.30brc_.
11:35.40ionixi.e if asterisk receives _1800xxxxxxx, I want it to send *18001234567 to SIP/fwd
11:35.43brc_.
11:35.53ionixI don't understand brc
11:35.57brc_no you don't
11:36.11ionixI have
11:36.12ionixexten => _1800.,1,Dial(SIP/#*${EXTEN}@fwdmax)
11:36.12ionixexten => _1800.,2,Macro(fastbusy)
11:36.12brc_why would you want to send a asterisk?
11:36.21ionixbecause fwd needs an asterisk
11:36.33brc_k
11:40.05Jas_Williamsionix: change your dial string to exten => _1800.,1,Dial(SIP/*${EXTEN}@fwdmax
11:40.45brc_buzz
11:41.10ionixok thx
11:41.31ionixI tought # was required. Thx
11:42.00ionixbtw, any SIP provider that offers long distance without subscription ? Like I can fuel a card 10$ and make outgoing call to them without having a phone number ?
11:42.26ionixI would like my gateway to use my phone number for local call and that sip provider for long distance. + use 1800 with fwd
11:42.43ionixi.e plenty of rules to save money (the real reason is to have fun but shhht)
11:42.45brc_~nufone
11:42.45jbothmm... nufone is Visit http://www.nufone.net for an excellent, native IAX termination service.
11:42.50brc_2.0
11:42.59ionixI'll look into that
11:47.37Makenshi~sipgate
11:47.49Makenshioh well :>
11:48.53ZX81~fish
11:48.54jbothmm... fish is FISHFISHFISH! DO THE FISH DANCE! "Give a man a fish and you'll feed him a day. Teach him how to fish and he'll feed himself for the rest of his life." This is so appropriate, instead of asking us to tell you exactly what to do, why not read some docs, then come back and ask specific questions which aren't covered?, or ...
11:48.56ionixI signed up for nufone.net seems nice the pay as you go
11:49.06ZX81lol
11:49.23Jas_Williams~docs
11:49.23jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
11:49.38Jas_Williams~sipgate
11:50.09ZX81~cow
11:50.11jbotI am a cow, hear me moo. I eat grass and weigh twice as much as you.
11:50.15ZX81lol
11:50.34Jas_Williams~sipgate is http://www.sipgate.de
11:50.35jbotJas_Williams: okay
11:50.40Jas_Williams~sipgate
11:50.41jbot[sipgate] http://www.sipgate.de
11:50.43ZX81~pig
11:50.44jbot*oink* *oink* *oink*
11:50.48Jas_Williamsbetter
11:50.55Jas_Williams~lart ZX81
11:51.00ZX81lol
11:51.10ZX81~thwap Jas_Willia
11:51.12jbotACTION bends over Jas_Willia and grins happily
11:51.20ZX81y:-)
11:51.31Jas_Williamstooche'
11:52.56ZX81~daily asterisk news
11:53.25ZX81~daily asterisk news is http://www.sineapps.com/news.php for html and http://www.sineapps.com/rssfeed.php for rss
11:53.26jbotokay, ZX81
11:53.28ZX81~daily asterisk news
11:53.29jbotdaily asterisk news is probably http://www.sineapps.com/news.php for html and http://www.sineapps.com/rssfeed.php for rss
11:53.44ZX81~news
11:53.54darkskiezdoes anyone know where the asterisk news site is?
11:54.02ZX81~daily asterisk news
11:54.03jbotsomebody said daily asterisk news was http://www.sineapps.com/news.php for html and http://www.sineapps.com/rssfeed.php for rss
11:54.04ZX81:-)
11:54.06brc_are you dense? or do you just pretend to be
11:54.23zoahello there
11:54.26brc_damn
11:54.32ZX81where's there?
11:54.41darkskiezbrc_, sorry, easily amused.
11:54.45ZX81:-)
11:55.06brc_~asterisk daily news is <reply> see daily asterisk news
11:55.07jbotbrc_: okay
11:55.14brc_~adn is <reply> see daily asterisk news
11:55.15jbot...but adn is already something else...
11:55.19brc_~adn
11:55.20jbotmethinks adn is AcidoDexosirriboNucleico
11:55.25brc_~no adn is <reply> see daily asterisk news
11:55.26jbotokay, brc_
11:55.35brc_~news is <reply> see asterisk daily news
11:55.45brc_~seen Makenshi
11:55.47jbotmakenshi is currently on #asterisk (2h 42m 40s).  Has said a total of 2 messages.  Is idling for 7m 58s
11:56.39*** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it)
11:57.00ZX81:-)
11:57.08ZX81i died
11:57.15ZX81headshot while jumping
11:57.17ZX81sucky
11:58.00brc_halo2?
11:58.11ZX81:-)
11:58.13ZX81nah
11:58.17ZX81still stuck in CS
11:58.20ZX81:-)
11:58.24brc_CS
11:58.26brc_HA
11:58.32ZX81lol sad a
11:58.57ionixSilly question but can I use my nufone account to place 2 simultanous call ?
11:59.06ionixi.e I have 2 lines on my spa-2000...
11:59.14ionixOr do I need to register 2 accounts
12:00.32PatrickDKtwoaccounts
12:01.17ZX81dunno, voicepulse i was doing 4
12:01.18brc_PatrickDK, su
12:01.35brc_ionix, of course you don't need two accounts with nufone
12:01.43PatrickDKoh heh
12:01.46RoyKsudo rm -rf /
12:01.49PatrickDKno, one nufone account
12:02.01brc_touch rm\ \-rf
12:02.02PatrickDKI didn't see the first part of the question
12:02.35brc_riiiiiiiiiight
12:02.58ionixok thx :)
12:06.21darkskiez<PROTECTED>
12:09.15ionixAlso, how will I know if there are insuffisent funds in the nufone account ? I mean if balance is insuffisent, how can I notify the caller on my pbx that long distance calls cannot be dialed at the moment ?
12:09.47RoyKmm
12:10.12RoyKis it possible to turn off echo cancellation for certain calls?
12:10.16RoyKon zap?
12:10.54*** join/#asterisk SuB_X (Sahm_X@adsl-135-24-192-81.adsl.iam.net.ma)
12:10.55ZX81you can have it on or off per line i think, but not dynamicly
12:11.09PatrickDKna, it turns it off for fax's I know
12:11.36ZX81k
12:11.42ZX81so then must be possible
12:11.53PatrickDKya, just don't know if there is a command to do it
12:12.00SuB_Xhi, plz i'm searching a frenchspeaking webdevelopper,
12:12.03PatrickDKor if it's hardcoded into the fax detection stuff
12:12.03ZX81grep source
12:12.11ZX81my wife is french
12:12.19SuB_Xhi, plz i'm searching a frenchspeaking webdevelopper,, could you help me
12:12.19ZX81from lyon
12:12.20SuB_X?
12:12.23FuzzyCatSuB_X: rentacoder
12:12.39ZX81i am a developer and my wife is french, i don't speak a word of it tho
12:12.41ZX81:-)
12:12.41SuB_Xrentacoder ?
12:12.48SuB_Xlol
12:13.02FuzzyCatwww.rentacoder.com
12:13.03SuB_Xcould you help
12:13.04SuB_X.?
12:13.39RoyKPatrickDK, how can I signal that THIS IS FAX?
12:13.46RoyK...if it doesn't detect it, I mean
12:13.57PatrickDKyou don't
12:14.01RoyKwell...
12:14.13RoyKbut is fax detection supposed to really work?
12:14.15ZX81put faxdetect=both in zapata.conf
12:14.22ZX81or incoming
12:14.24ZX81or whatever
12:14.35RoyKproblem is outgoing faxes...
12:14.48RoyKare there any echo cancellation in chan_sip?
12:14.56PatrickDKno
12:14.58ZX81nope
12:15.09PatrickDKwhat are you using to connect the fax to asterisk?
12:15.21ZX81ygsm codec
12:15.23ZX81:-)
12:15.45RoyKfax -> SIP ATA -> SIP/IAX2 gw (*) -> IAX2/PSTN gw (*) -> ISDN PRI
12:15.57RoyKusing ALAW
12:16.37PatrickDKhmm, the problem is here is no fax detect logic on the sip side
12:17.06PatrickDKinside of asterisk
12:17.55ZX81what defines the time a box re-registers at?  like some of my peer servers are around every 10 mins, some every hour or so
12:18.12ZX81all the same in iax.conf
12:18.20PatrickDKit depends on the peer
12:18.30ZX81what do they set?
12:18.38PatrickDKanything they want
12:18.46ZX81i.e. what variable
12:18.50ZX81xxx=xxx
12:19.04PatrickDKdunno, it's different
12:19.08ZX81k
12:19.23PatrickDKI mean, it would be different on a supira200, a snom phone, and a cisco 7960
12:19.32ZX81no, all iax
12:19.32PatrickDKyou can't control it in iax.conf
12:19.44ZX81intracompany boxes
12:19.48PatrickDKyou can only control how often you register to someone else
12:19.50Makenshibrc_, had you confused, my bad
12:19.52ZX81hmmm...strange
12:20.30PatrickDKyou can control registeration timeouts
12:20.39ZX81ah maybe it is not registering the other box becuase of qualify
12:20.46ZX81so it already knows
12:20.48tzangerany recommended places to purchase Polycom IP300 phones?  Looking for about a dozen of 'em
12:25.51ionixSuB_X: I am french heh
12:26.40ionixtzanger: check ebay or www.froogle.com