irclog2html for #asterisk on 20041211

00:00.07tessierNugget: That's what I was hoping. Because this Dell box doesn't seem to have power connectors for this card anyhow. But then that means 2 of the modules on this card are probably bad. :(
00:01.37*** join/#asterisk infinity005 (brendon@adsl-68-126-236-227.dsl.pltn13.pacbell.net)
00:02.00infinity005wow. busy channel.
00:02.48tzangeryeah it's usually a lot busier
00:02.50infinity005i'm new to voice on linux, but i had this idea. I would like to setup a bluetooth earpiece and setup a bluetooth gateway for it via a PC
00:03.11infinity005is that possible with asterik? if not, anyone know where to go for something like that?
00:03.42gigagodwhere?
00:04.03infinity005where?
00:04.25infinity005bad idea? good idea?
00:04.58infinity005i thought it was a good idea :)
00:05.06tzangeryeah I use that
00:05.18tzangerMotorola headset and Linksys BT100 and Firefly
00:05.32tzangerworks pretty good except that Windows Bluetooth support stinks, and Linux Bluetooth support isn't much better
00:05.32infinity005how about a bluetooth earpiece, connecting to linux/asterisk and VoIP to vonage?
00:05.36infinity005ahhh
00:05.46*** join/#asterisk bilo (nabeelj@69.158.63.195)
00:05.52infinity005is the bt100 the voip gateway?
00:05.57tzangerno
00:06.00tzangerit's a bluetooth usb adaptor
00:06.08_RaYmAn_I seem to remember people talking about a chan_bluetooth at some point... but dunno status
00:06.17tzangeryeah it was on the list recently
00:06.24Nuggetyeah, there's a chan_bluetooth in development but it's still very raw
00:06.29biloi wanted a simple windows-based GUI interface to monitor my * server (not for configuration)... any suggestions?
00:06.40Nuggetbilo: http://asternic.org/
00:06.51infinity005tzanger: so one you have the earpiece and bluetooth adapter, how do you get on the phone?
00:06.51Nuggetflash-based, works in windows and osx.  probably elsewhere
00:07.12jeronite
00:08.00bilonugget: looks good, i'll try it
00:08.01tessieryeay
00:08.09tessierTurns out the daughter boards were not seated well
00:08.13tessierReseated them and now all 4 line work
00:08.26tessierBut at least their tech support did recommend reseating them
00:09.10NuggetI like simple solutions.
00:09.18Atacommlol.....digium's tech support?
00:09.20kramall boards leave digium tested, it's possible it came lose in shipment
00:09.28Atacommor the resellers?
00:09.34kramwe test all the boards in the actual configuration they are sold in
00:09.40tessierNow to figure out why when calling in on the zap lines my s extension answers but when I dial 102 I get a timeout in the dialplan...
00:09.42kramso if you order a TDM22B it gets tested as a 22B
00:09.52tessierkram: Indeed, I am sure you are right. All of my other digium stuff has worked great so far.
00:10.23tzangerkram: is there an official status on the TDM4xxp power problems?  It ain't the system power supply in a lot of cases, I can tell you that much.
00:10.33bilonugget: does asternic show some kind of log to see failed sip calls, missing contexts, etc?
00:10.38Nuggetbilo: no
00:10.39tzangeri.e. why do these damn boards need to be reloaded (modules) every 2-6 weeks
00:10.55bilonugger: i needed something like that, more technical data ;)
00:11.05bilonot necessarily web-based
00:11.08bilo*nugget
00:11.29tzangerkram: I want to push the TDM4XXP cards, I really really do, but they simply are not reliable
00:11.42jeroi have a problem with digium quad-fxo/fxs boards. quite often an analog phone plugged on an fxs port starts working and the only noise heard from the phone is "crrhrhshchcshsssshh". The only way to recover it is to stop asterisk, unload, reload zaptel, restart asterisk. What can I do ?
00:11.52tzangercase in point.
00:12.02tessierdoh...I used playback instead background
00:12.47kramtzanger: what do you mean they have to be reloaded
00:12.50kramwhat is the symptom?
00:12.57gigagodYeah I have spent 3 weeks trying to get that card to work correctly
00:13.22tzangerkram: jero's exact symptom on FXS, FXO problem is more it stops answering or won't hang up
00:13.24kramdid you call tech support?
00:13.38kramtzanger: yah, we found a couple of people with FXO issues and have a fix for those
00:13.46kramit seems to be system specific and only started showing up recently
00:13.48jerotzanger, same fxs problem so.
00:13.50bilowell, i have ssh to my asterisk server, but after 5 or so mins of me not typing anything, the ssh connection seems to freeze... after another few minutes, it disconnects... how can i set it up so the ssh connection stays alive
00:13.53biloso i can watch *
00:14.18tzangerkram: yes -- this is more an FXS issue with me (I haven't done much with FXO yet) -- one of the machines is a server-class (triple-redundant power) Xeon system with a T100P in it too (T100P works great)
00:14.25jerokram, I have been experiencing this problem since I got the first card in july (asterisk was not 1.0 yet)
00:14.45ariel_bilo, it's your router timing out. linksys does that allot here.
00:14.59kramjero: is this fxs or fxo?
00:15.03jerokram, fxs
00:15.06tzangerI've had my 100MHz DSO on the +12V line when this is ocurring (power on module #1, resetting! type error) -- that line is +12V within a couple dozen mV -- it's not dipping
00:15.08kramhrm...
00:15.18biloariel: i have another ssh window with vi open, and it doesn't time out?
00:15.21Atacommhas the FXO module even cleared Part68 yet?  I looked up Digium's Part68 filings a week ago and didnt see it
00:15.29biloariel: ssh to the same server :|
00:15.32jeroit occurs at least 1 time a day actually, because we use this analog phone more often
00:15.34kramit's in part 68
00:15.41tzangerthe other machine I have is a P3/733 (nothing special) -- works great for 4-6 weeks then starts crackling and basically not working... module reload and it's good to go for another while
00:15.45kramit passed we're just waiting on paperwork
00:16.04kram(and already passed 15, again waiting on paperwork)
00:16.36tessierhmm...now there is just a tiny bit of echo on my zap lines...
00:16.40ManipuraHello everyone
00:16.46Atacommah, ok, was curious.....i keep hearing problems from customers, and was beginning to get worried because we pass it off as certified and i didnt see it on the Part68 website
00:16.48tzangertessier: I find that is almost impossible to get rid of
00:16.51jeroreloading the module fixes then I guess its possible to fix it with software
00:17.11tessiertzanger: You may be right. echocancel=yes is turned on in zapata.conf etc.
00:17.19tzangerjero: that's my impression as well -- either the chip's getting into a weird state and the driver is (incorrectly) restoring it, or something about the reinitialization of the chip fixes it
00:17.42twisted[work]WAY OFF TOPIC - any php gurus around that want lend a learner a hand?
00:17.43tzangerwe get echo on some of our calls out our PRI -- 99% are fine but some specific numbers in specific exchanges are just horrible
00:17.45kramwe've had a handful of customer call about the fxo getting lost but we finally were able to duplicate it and fix it
00:17.54tzangerkram: excellent news!
00:17.55biloariel: any other ideas?
00:18.05twisted[work]kram, same issue we were having?
00:18.10tzangerkram: is there anything us mere mortals can do to help you with the FXS issues?
00:18.11Manipuratwisted[work] I wouldn't call myself a guru
00:18.16kramyes, the one you had
00:18.19tessiertzanger: Yeah, that's a problem on the other persons end though, right? If it were a pure digital call whether voip or tdm there should be no echo. It's just the analog portion that causes issues right?
00:18.23twisted[work]kram, sweet!
00:18.25Manipurabut I might be able to help
00:18.29kramdid christian ever get the boards with the extra caps?
00:18.39twisted[work]yea, we shipped them
00:18.54kramyah that should fix it, it did for mog
00:18.57tzangertessier: not always -- I am guessing that the echo canceller is getting disabled on those calls (like it does if it detects fax) -- I have to debug a little more
00:19.06twisted[work]kram, good :
00:19.07twisted[work]er ;)
00:19.11tzangerkram: what caps need to be augmented?  I am an EE :-)
00:19.16twisted[work]kram, gonna redesign the boards to include the extra cap?
00:19.43jerotzanger, I'm experiencing the problem once a day
00:19.46jeroat least
00:19.49gigagodPASS / FAIL   ---- DON'T CARE....  I WANT MY BOX RUNNING.
00:19.51tzangerjero: WOW... mine's NEVER that bad
00:20.07tzangerjero: get a ticket with digium and see if they'll warrant the boards or if there is something else they can do
00:20.13tzangerjero: you're not sharing interrupts or anything are you?
00:20.58jeroI dont think so, i'll check
00:21.16jeroI had this problem with my first board, then got it again with the second one
00:21.43tzangerhmm I've *never* had that bad a problem -- maybe you really do have a bad power supply :-)  What kind of system and what kind of PS?
00:21.47gigagodWhat other cards are there out there that asterix will support
00:21.50jero/proc/interrupts
00:21.56jero<PROTECTED>
00:22.06jpabloanybody in sipphone.com can help me dialing 1-747-6675276 .. ?
00:22.10tzangerjero: yikes
00:22.20jerotzanger, !
00:22.24tzangerboth TDM cards and ethernet on one interrupt...  that's not good :-)
00:22.29Atacommgigagod: voicetronix i hear, very nice analog cards
00:22.46jeroCan I force the zaptel driver to use another int ?
00:23.08tzangerjero: that's a fight between your BIOS and the PCI subsystem
00:23.22tzangeri.e. you need to shuffle cards around, see if you can force them to separate IRQs inside the BIOS, etc.
00:23.28jeroThis tends to be boring on PCs
00:23.50asuffieldsharing pairs are usually tied to a given slot. you can change the interrupt, but the ethernet device will change too
00:23.54asuffieldmove 'em
00:24.01gigagodWhat are these TDM120's I have been hearing about.  Anyone know anything about them?
00:24.33tzangergigagod: yup we're discussing them right now :-)
00:24.43gigagodcool
00:24.43jerotzanger, It seems the problem is proportional to the # of times the fxs port is used
00:24.50tzangerasuffield: unless they're all assigne the same IRQ in the BIOS
00:24.55file[laptop]fooooooooood is here
00:25.02asuffieldthat would be highly abnormal
00:25.06tzangerjero: well yeah -- first things first --g et those cards on separate IRQs from each other and ethernet
00:25.12*** join/#asterisk r0d3nt (~RatMan@64-60-114-35.cust.telepacific.net)
00:25.14jeroyup
00:25.24r0d3ntHello everybody =)
00:25.26Atacommgigagod: we're expecting prototypes in a few weeks, we've turned the TDM120 T1/E1/J1 boards over to our contract mfg for prototyping
00:25.34tzangerasuffield: yes but you tell the BIOS to do everything auto and you get weird results sometimes
00:25.39r0d3ntWhat is considered "normal" delay time to initiate a SIP Asterisk phone call to a T1/PRI ?????
00:25.55r0d3nt5+ seconds ?
00:26.09asuffieldsecond or so, sometimes less
00:26.12gigagodnice... When can I get my hands on several!
00:26.46r0d3ntasuffield, why would it take more then 4-5 seconds to initiate a SIP call ?
00:26.54asuffieldbecause it's broke
00:26.54r0d3ntsometimes up to 8 seconds
00:26.56file[laptop]cool, they gave me 5 strips instead of 4
00:27.05tessierhmm...I don't understand how this dialplan stuff on the snom220 is supposed to work. Looks like it will only let me make one dialplan string per line.
00:27.07r0d3ntasuffield, well then my asterisk vendor is a POS.
00:27.13jerosoon a new softphone to be released
00:27.15tessierAnd even that one isn't working but that's my fault I'm sure.
00:27.17Atacommgigagod: soon i hope :)  the thing is tiny, its about 1 cm longer than the TE410P.... even with all the processing power onboard
00:27.32KalD|Workanyone have issues with IAX calls between different versions of asterisk?  specifically I have two * boxes, one w/ a T1 the other with IAXy clients;  I call from an IAXy client to the * box which talks IAX to the other * box which dials out the PRI - the problem is the * w/ the IAXy connection reports that the call has ended when it is still going.
00:28.31r0d3ntasuffield, you really thing something is broken ??
00:28.39gigagodthats ok... I have 17" of space in that RLX server
00:28.46jerois anybody using softphones ?
00:29.22tzangerjero: I've used Xlite and am using Firefly
00:29.24gigagodthat Wildcard just doesn't like CenturyTel's T1s
00:29.44Atacommgigagod: our main card is 13 cm, our daughter card is 10 cm
00:29.46jerowe'll release a new one in some days
00:30.29gigagodI use a softphone on my laptop... X10, can't get it to work though
00:30.57gigagodsweet size... That will go beautifully in the RLX boxes
00:31.33Atacommbut remember, the expansion card uses a PCI slot, even though it doesnt connect into the slot....  due to component height and ports/bracket
00:31.40r0d3ntWhat is considered "normal" delay time to initiate a SIP Asterisk phone call to a T1/PRI ??
00:31.52tzangerr0d3nt: seconds
00:31.52ScaredyCatnone
00:31.57tzangerlike 1 or 2
00:32.02ScaredyCatMAX
00:32.08gh0stoi r0d3nt
00:32.15r0d3nttzanger, under 5 seconds ?
00:32.15gh0stfancy seeing you here
00:32.21r0d3ntya.. imagine that =)
00:32.23tzangerdo you have osm ekind of "overlap" dialing mode on the SIP device (not hte right term)
00:32.24gh0stheh
00:32.27gigagodDoes it draw any power from the PCI slot?  Is there a website that I can go to to find out info about the TDM???
00:32.39r0d3nttzanger, i do not believe so...
00:32.47tzangerr0d3nt: definitely, it's under 2 seconds with my damn KSU -> * (PRI)
00:33.02r0d3nttzanger, ok.. so it sounds like something is incorrectly setup...
00:33.06Atacommgigagod: no public site on it yet... the daughtercard does not draw power from the slot, it draws power from the ribbon connector to the main card... (which does draw power from the slot)
00:33.09r0d3ntit takes almost 5-8 seconds @ times...
00:33.19r0d3ntthis is a dual xeon 2.4ghz with a gig of ram....
00:33.27kramtzanger: it's a 200pf across reset to ground
00:33.47gigagodThen I definately want that card!!!
00:33.51kramas bizarre as this seems, it keeps ESD from causing a partial reset
00:34.03tzangerkram: which pin is reset (on the module or on the TJ320)
00:34.17kramon the module
00:34.26kramit should be as close to the reset pin as possible
00:34.36gigagodWhat do you think Kram?
00:34.42tzangerkram: that makes a lot of sense actually
00:35.00tzangerkram: I am going to give that a shot and let you know what I find here
00:36.14r0d3nttzanger, I'm using Cisco 7940G's, * and a CT1 from TelePacific....  almost 10 seconds until the phone starts ringing....
00:36.26r0d3nti need to remember my password for r0d3nt|m
00:36.37tzangerr0d3nt: are you sure you don't have som ekind of huge digittimeout and are dialing "live" ?
00:37.45r0d3ntnomad_, and yes... I press new call, then dial the number, then press Dial, and then it takes up to 10+ seconds to complete the call and i hear it ringing....
00:38.01r0d3nttzanger, No .. i'm not sure... and Yes, I am dialing "live"
00:38.43*** join/#asterisk coolschool (~dan@22.247.adsl.brightview.com)
00:38.44tzangerr0d3nt: that'll be your digittimeout killing you
00:38.57tzangerr0d3nt: show application digittimeout and put that at the start of your dialplan
00:38.57coolschoolhi all
00:38.58gigagodwill the documentation for the TDM be added in to the Asterisk docs
00:39.07r0d3ntwhich conf file ? extensions ? or sip .conf ?
00:39.19r0d3ntok i'll check it out...
00:39.27r0d3nttzanger, I appreciate the info.. lemme check it...
00:39.35ScaredyCatr0d3nt: watch the * console, how long b4 asterisk tries to dial out
00:39.36*** join/#asterisk mr_monkey (~root@201.129.239.6)
00:39.37Atacommgigagod: dont know, we will document and support everything ourselves, we havent approached anyone on the asterisk project officially, there has been some backchannel conversations
00:39.47r0d3ntI didn't build this system.. i paid someone else to.. and I'm finding out all these problems....
00:39.54r0d3ntScaredyCat, OK
00:40.00mr_monkeydoes exists a manual to configure asterisk with postgres ?
00:40.49*** part/#asterisk coolschool (~dan@22.247.adsl.brightview.com)
00:40.51*** join/#asterisk r1 (~erwan@www.thiscow.com)
00:41.14tzangerr0d3nt: damn I hope you had some kind of satisfaction clause
00:42.35*** join/#asterisk znoG (gs@134-134-126-200.fibertel.com.ar)
00:43.00*** join/#asterisk _Vile (~vile@90.b160.bendtel.net)
00:43.04_VileMUHAHAHA
00:43.34redder86Anyone got a link for Linux QoS that is different than this one: http://www.voip-info.org/tiki-index.php?page=QoS%20Linux
00:43.50tzangerredder86: I use this script (which is on that wiki page) www.mixdown.ca/~andrew/dump/rc.tc
00:44.21gigagodQOS: http://qos.ittc.ukans.edu/
00:44.57*** part/#asterisk eKo1 (~abc@63.245.57.70)
00:45.00redder86tzanger: that script looks awfully involved.
00:45.06tzangerredder86: nah it's very straightforward
00:45.17redder86what is tc ?
00:45.24agronquitcl
00:45.30mr_monkeydoes exists a manual to configure asterisk with postgres ?
00:45.41tzangerbasically on the ADSL uplink: IAX2 traffic goes directly to the interface (this is kind of bad but it works -- no delay) -- everything else gets put into a HTB tree
00:46.08tzangersimiarly on the ethernet output I shift all the default ;priority maps "down" one spot so that I can tell it that IAX2 traffic gets priority 0
00:46.10*** join/#asterisk brettnem (~Brett@216-60-162-174.ded.swbell.net)
00:46.24brettnemhey all
00:46.34gigagodno idea... just googled "linux qos"
00:46.35brettnemhey, does anyone use setgroup and check group?
00:46.52redder86tzanger: okay, is this for incoming traffic as well as outgoing?
00:47.20brettnemdoesn't look like setgroup ever decrements the counters when the channel hangs up.. hmmm
00:47.34tzangerredder86: not really -- you can't shape your incoming traffic, you already have it.  The best you can do is drop excessive ***TCP*** traffic in an effort to get the sender to slow down
00:47.51tzangerredder86: consequenty I also drop excessive UDP traffic in case the application does some kind of application-layer throttling
00:48.27gigagodBest solution:  Increase the bandwith
00:48.58ScaredyCatminimise hops
00:49.08brettnemany ideas for setgroup??
00:49.28ScaredyCatbrettnem: what about it?
00:50.06r0d3ntdoesn't look like there is a digittimeout specified....
00:50.12redder86gigagod: increasing the bandwidth in lieu of QoS requires you to keep up with the Joneses.  As long as you have a bigger net connection than the guy on the other end then you will still have bandwidth left for VoIP.  When you stop upgrading and they overcome you, then you will either need to upgrade again or use QoS.
00:50.13brettnemwell I'm using it.. it increments a counter, but never decrements it when the channel is released
00:50.30redder86gigagod: methinks one must do both
00:51.00redder86gigagod: I don't think that QoS is really optional unless the interface is but rarely used for anything other than VoIP.
00:51.01file[laptop]Purolator is such a nice company, they even call to reschedule deliveries and stuff
00:51.07tzangerr0d3nt: specify like 3 seconds
00:51.19r0d3nttzanger, OK.. i'll do that on a test extension
00:51.20tzangerfile[laptop]: :-)
00:51.24r0d3ntthanks
00:51.26brettnemhmm...looks like hangups might not being detected
00:51.42file[laptop]and on my last delivery the guy came back a second time to try
00:51.57redder86Purolator?
00:52.12file[laptop]shipping company
00:52.22UmaroOkay guys, i'm trying to put * between a E1/multiplexer and a siemens phone system, and am having nothing but problems. I'm using a TE410P. Should I be using straight through cables, or reversed?
00:52.42ScaredyCatcrossed if it's e1 >E1
00:53.19UmaroScaredyCat, how do you mean?
00:53.43ScaredyCatif you are connecting an E1 to an E1 on the siemens then cross cable
00:53.46tzangerPurolator is a private company that is owned by Canada Post (the federal mail carrier) -- talk about conflict of interest
00:53.51ScaredyCatis required
00:54.16redder86ah, that's why I don't know Purolator.  I'm not in Canada.
00:54.59*** join/#asterisk visik7 (~ciao@host161-36.pool80182.interbusiness.it)
00:55.03UmaroScaredyCat, erm.. normally, the E1 is plugged into the siemens. I'm trying to put a * between them so I can send their long distance over voip.
00:55.26visik7where can I find a single gsm gateway (h323 or serial better h323)
00:55.52UmaroScaredyCat, So the connection between the E1 and * should be straightthrough, right?
00:56.05tzangerUmaro: I believe so yes
00:56.09tzangeryou need cross to go to channel banks
00:56.16tessierCool, so I have my snom220 properly dialing the local 3 digit extensions without having to press enter.
00:56.28tessierNow to figure out how to make it recognize a 9 and then a 10 digit number.
00:56.33tzangerif it lights up green it's seeing the other end (and hte other end is seeing you) -- your data layer protocol might be fucked but the bits are being seen :-)
00:56.37redder86tzanger: this guy: http://www.voip-info.org/tiki-index.php?page=Linux%20Router doesn't even use tc.  Is he wrong?
00:56.47brettnemso my setgroup checkgroup problem I think is here: http://pastebin.ca/2866
00:56.55Umarotzanger: well, the "E1" is from the multiplexer. the multiplexer is connecting to all the isdn lines
00:57.03tessierUnfortunately it sort of looks like I can only have one dialplan entry per line which is rather lame.
00:57.13brettnemI have checkgroup set to 2.. and it shows 2 channels up, but sip show inuse shows no channels in use.. any ideas??
00:57.26tzangerredder86: he's using basic marking -- no queuing at all
00:57.35tzangerit'll work too
00:57.42tzangerjust not as robust / configurable
00:58.04brettnemlooks like my answer supervision isn't working somewhere..
00:58.28redder86tzanger: as for the configurations he poses versus the one you pose, what is the effective difference?
00:58.48brettnemany ideas?
00:58.49tzangerredder86: effective difference is that he can't shape or limit bandwidth
00:59.12tzangerredder86: if it's just you it should work fine so long as you reduce the queue length on the interfaces (I use 10, default is 1000 I think)
00:59.23tzangerthen so long as you don't totally hose your link you should be fine
00:59.36tzangerI have 35 customers behind my connection, I needed something a little more :-)
00:59.57cypromisÂ/window 34
01:00.01cypromishumpf
01:00.23ScaredyCatuseless
01:00.32cypromisyeah
01:00.35redder86tzanger: it's not just me
01:01.00redder86tzanger: well, here it is, but not at the server where I'm worried about it
01:02.22Umarotzanger: could you please look at my configs? http://pastebin.ca/2867
01:02.32redder86tzanger: why does it make a difference how many users there are behind the connection?
01:02.42tzangerredder86: ask yourself what you need to do
01:02.44tzangerthen tell me
01:02.55*** join/#asterisk Darwin35 (~darwin35@pool-68-162-181-243.pitt.east.verizon.net)
01:03.37redder86tzanger: I just want to dedicate an adequate portion of bandwidth to IAX2
01:03.45tzangerUmaro: looks fine to me, the only thing I'd do (personal preference, dont' think it makes a whit of difference to *) is to say bchan=1-15,17-32
01:03.55tzangerthat is traffic shaping, you need tc
01:04.02redder86tzanger: okay, thanks.
01:04.13tzangerif yuou just want to give highest priority to iax2 you *should* be able to get away with straight fwmark
01:04.23tzangerbut all bets are off if you're capable of flooding the link
01:04.37Umarotzanger: ok
01:04.39tzangerUmaro: ztcfg -vvv (everything looks fine?)
01:04.47Umarotzanger: yep, no errors
01:04.55redder86tzanger: ah, okay, I think I understand the difference between prioritization and traffic shaping now
01:04.56Umarotzanger: zttool tells me span 1 is OK
01:05.07*** join/#asterisk ariel_ (~Ariel@ssf-office.corp.race.com)
01:05.07tzangerredder86: my script does both
01:05.11tzangerUmaro: the *span* is fine
01:05.16tzangeryou're having a problem on the data link
01:05.22tzangeri.e. the D channel isn't where you think it is
01:05.33tzangeris there any way to query the existing Siemens config to see how it's set up?
01:05.38Umarobut doing a 'pri intense debug span 1' doesn't give me anything except saying it's sending a SABME request
01:05.41UmaroNo..
01:05.55Umarowell, I think it's channel 16, and when I try it on any other channel, it can't configure channel 16 as a bchan
01:06.06ScaredyCatyou do have one end as cpe and one as net?
01:06.09Umaroit says the signalling is wrong
01:06.12Umarospan 1 is cpe
01:06.20UmaroThat's all i'm trying to get working for the moment
01:06.24tzangerUmaro: well like I said I would start with bchan=1-31, dchan=32 (and simialr config in zapata.conf), then ztcfg -vvv, start * and see... then go bchan=1-30,32/dchan=31 and try again
01:06.49*** join/#asterisk ApEtc (apetc@ip68-99-136-197.ph.ph.cox.net)
01:06.50tzangeryou reran ztcfg -vvv after every change to /etc/zaptel.conf?
01:06.57Umarotzanger: yep
01:07.05tzangerUmaro: hmm
01:07.13tzangershort of calling your telco, I don't know.
01:07.27Umarothere isn't a telco.. at least not for the E1
01:07.28tzangerfrom what I understand of E1 (I'm in Canada, we use T1), dchan on E1 is typically channel 16
01:07.30Umaroit's a multiplexer
01:07.38tzangerUmaro: uh
01:07.42tzangerUmaro: the MUX is PRI??
01:07.44UmaroI'm told the d channel is on 16 in this case too
01:08.00Umarotzanger: :)
01:08.01tzangerUmaro: every MUX I've dealt with is straight E1 or T1, not PRI
01:08.09Umarotzanger: so then what signalling should I use?
01:08.25tzangerand then your signalling would be fxs_ls or fxs_ks generally
01:08.52tzangerUmaro: so you have a DS3 being split off in a MUX or what
01:08.57tzangerwhere's this MUX
01:09.35Umarono, there's like however many ISDN lines coming in
01:09.38Umarolike BRIs
01:09.42tzangeroh
01:09.47tzangerwow man I really don't know in this case
01:10.00tzangerBRI just blows my mind :-)
01:10.33Umaroit says "signalling requested is FXS kewlstart but line is in PRI signalling signalling"
01:10.40Umarowhen I try to start asterisk
01:11.07tzangerUmaro: well if it's a buch of BRIs coming in I doubt it's CAS E1
01:11.14tzangersorry for the confusion
01:11.25tzangerand if tha't shte error you did not configure zaptel.conf
01:11.27tzangerand rerun ztcfg
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01:11.57ariel_ok I am back.  Argh network problems.
01:11.57Umarotzanger, no, I did run ztcfg, and didn't get errors
01:12.23Umaroztcfg says each of the channels are "Invidual clar channel"
01:12.26Umaroariel_, wb
01:12.34Umaroer Individual, lol
01:12.51ariel_Umaro, your still having E1 problems?
01:13.01tzangerariel_: yeah he is
01:13.22tzangerit's beyond me as I know nothing about E1 and even less about E1 BRI->PRI muxes
01:13.29Umaroariel_: I figure if i'm not dead from old age when I get this going finally, i'll kill myself after I get it working to end the horror.
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01:13.58ariel_Umaro, have you called digium?
01:14.11Umaroyes, until they closed
01:14.14tzangerariel_: it's not a digium problem
01:14.19tzangerit's a ocnfiguration issue from what I can gather
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01:14.59ariel_ah I see... (I will then go back to my working area).
01:16.43FaithfulHey guys what CPU would I need to handle 2 incomming ISDN lines?
01:16.55tzangerFaithful: anything
01:17.00Weezeyanyone have any SPA-3000s?
01:17.05ariel_anythign will do for just 2
01:17.32mutilatori suggest 200mhz
01:17.37FaithfulPentium 133 with 32Mb ram?
01:17.56tzangerFaithful: that's getting on the low side
01:17.59redder86tzanger: does your rc.tc get run automatically by something else, or do you just run it from rc.local ?
01:18.00Umarotzanger, here's the info I got from the multiplexer guy: framing:  CAS Signaling:  edss1 line data on channel 16 coding:  hdb3
01:18.08tzangerI was able to hadnle a POTS line and FXS port wiht a P90 (no MXX) but I could not do iLBC
01:18.15tzangerredder86: run it from rc.local
01:18.28FaithfulOk got a few PII 300 Celerons around
01:18.33tzangerFaithful: that's plenty
01:18.44tzangerUmaro: CAS signalling but D channel data on line 16?  wuh?
01:19.09tzangerUmaro: try signalling fxs_ks on channels 1-15,17-32
01:19.10ariel_what is a PII celeron?
01:19.17tzangerariel_: they exist
01:19.20tzangerPII with 1/2 the cache
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01:19.43tzangerUmaro: I'm guessing that channel 16 is something weird so I'm skipping it :-)
01:19.44ariel_Those were the first generation Celeron's then.
01:19.47tzangerariel_: yes
01:20.19Umarotzanger: fair enough
01:21.05FaithfulSo I have a little VIA Eden 533Mhz in a little alloy ITX case... that should be heaps... It won't play DVDs (which I bought it for)
01:21.18WeezeyFXS = line or extenstion?
01:21.24ariel_Faithful, yes that will work.
01:21.33tzangerFaithful: it can but be careful of the VIAs -- you must disable MMX in zaptel (and asterisk?)
01:21.39FaithfulI'm just looking for a low power solution since it will be 24x7
01:21.47tzangerWeezey: FXS port = plug a phone into it.  FXO port = plug a phone LINE into it
01:21.48ariel_FXS is a line you plug in a analog phone to.
01:22.36FaithfulWhich linux software phone is the best with *
01:22.44ariel_I hope we don't get into the fxs fxo thing again... Station or office please.
01:22.51tzangerFaithful: I haven't found one that works well
01:23.44FaithfulBoooo!
01:24.11Umarotzanger: gives me an error when I start * that says "Signalling requested is FXS Kewlstart but line is in PRI Signalling signalling"
01:24.43gh0stUmaro: signalling = pri_cpe
01:25.00Faithfultzanger:  are you saying that all MMX must be disabled or only with VIA
01:25.27Umarogh0st: tried that, don't get anything when I do pri intense debug span 1 except something talking about SABME
01:25.37tzangerUmaro: you did not run ztcfg, or you did not configure zaptel.conf
01:25.39Umarowhich i'm told means the d-channel isn't syncing up
01:25.46tzangerFaithful: only with VIA, it's broken on VIA
01:26.08Umarotzanger: dude... I totally did. I wish that was the problem, but i've tried running it every single time I change anything
01:26.12Umaroback in a few
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01:26.33[Outcast]yo twisted[work]
01:26.42WeezeyIf I get a SPA-3000 they have two FXS and one FXO.  How many calls can an asterisk box handle?
01:26.49Weezeysimultaneously
01:26.59ariel_Weezey, only one fxs and one fxo
01:27.04gh0stUmaro: what channel is the d-channel?
01:27.20tzangershow me your zaptel.conf now
01:27.21ariel_Weezey, one inbound one outbound only via the sipura.
01:27.52Weezeyariel: how many sipura's could I have making calls at once?
01:28.14DrukenHMEhas anyone used a generic version of the t100p ?
01:28.37tzangerDrukenHME: there's a generic version of the single-span T1 card?
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01:29.24DrukenHMEtzanger: i belive so...
01:29.32tzangerDrukenHME: URL?
01:30.01ariel_Weezey, that really depends on your setup. But if it's a normal asterisk server doing no transcoding 40 to 50 if they do transcoding well no more then about 20 to 25 in my use.
01:31.16DrukenHMEtzanger: maybe not... i dunno, i thought i seen one....
01:31.29Weezeyawesome.  Is a 1.5GHz P4 enough?
01:32.04ariel_Weezey, in most case yes more then enough.
01:32.26DrukenHMEmy 400 handles voip.. hehehe
01:33.36WeezeyI need an inexpensive SIP phone, does anyone have any good recommendations?
01:33.50tzangerWeezey: BT101
01:33.51ariel_what do you call inexpensive?
01:34.01Weezey$100 ish
01:34.14ariel_I am waiting for the Sipura one. It's on backorder till the 15th they told me.
01:34.31WeezeyI just got the email from Voxilla, it's now January
01:35.22ariel_Weezey, I have used the snom and the polycom's there good fones. But there in the 140 to 200 mark.
01:35.49Weezeywhere did you get them from?
01:38.49DrukenHMEwww.voipsupply.com has voip phones under 100 USD
01:39.54WeezeyI don't have a huge budget to work with, but bottom line is, I need a coupla phones.