00:00.07 | tessier | Nugget: 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) |
00:02.00 | infinity005 | wow. busy channel. |
00:02.48 | tzanger | yeah it's usually a lot busier |
00:02.50 | infinity005 | i'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 |
00:03.11 | infinity005 | is that possible with asterik? if not, anyone know where to go for something like that? |
00:03.42 | gigagod | where? |
00:04.03 | infinity005 | where? |
00:04.25 | infinity005 | bad idea? good idea? |
00:04.58 | infinity005 | i thought it was a good idea :) |
00:05.06 | tzanger | yeah I use that |
00:05.18 | tzanger | Motorola headset and Linksys BT100 and Firefly |
00:05.32 | tzanger | works pretty good except that Windows Bluetooth support stinks, and Linux Bluetooth support isn't much better |
00:05.32 | infinity005 | how about a bluetooth earpiece, connecting to linux/asterisk and VoIP to vonage? |
00:05.36 | infinity005 | ahhh |
00:05.46 | *** join/#asterisk bilo (nabeelj@69.158.63.195) |
00:05.52 | infinity005 | is the bt100 the voip gateway? |
00:05.57 | tzanger | no |
00:06.00 | tzanger | it'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 |
00:06.17 | tzanger | yeah it was on the list recently |
00:06.24 | Nugget | yeah, there's a chan_bluetooth in development but it's still very raw |
00:06.29 | bilo | i wanted a simple windows-based GUI interface to monitor my * server (not for configuration)... any suggestions? |
00:06.40 | Nugget | bilo: http://asternic.org/ |
00:06.51 | infinity005 | tzanger: so one you have the earpiece and bluetooth adapter, how do you get on the phone? |
00:06.51 | Nugget | flash-based, works in windows and osx. probably elsewhere |
00:07.12 | jero | nite |
00:08.00 | bilo | nugget: looks good, i'll try it |
00:08.01 | tessier | yeay |
00:08.09 | tessier | Turns out the daughter boards were not seated well |
00:08.13 | tessier | Reseated them and now all 4 line work |
00:08.26 | tessier | But at least their tech support did recommend reseating them |
00:09.10 | Nugget | I like simple solutions. |
00:09.18 | Atacomm | lol.....digium's tech support? |
00:09.20 | kram | all boards leave digium tested, it's possible it came lose in shipment |
00:09.28 | Atacomm | or the resellers? |
00:09.34 | kram | we test all the boards in the actual configuration they are sold in |
00:09.40 | tessier | Now 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... |
00:09.42 | kram | so if you order a TDM22B it gets tested as a 22B |
00:09.52 | tessier | kram: Indeed, I am sure you are right. All of my other digium stuff has worked great so far. |
00:10.23 | tzanger | kram: 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. |
00:10.33 | bilo | nugget: does asternic show some kind of log to see failed sip calls, missing contexts, etc? |
00:10.38 | Nugget | bilo: no |
00:10.39 | tzanger | i.e. why do these damn boards need to be reloaded (modules) every 2-6 weeks |
00:10.55 | bilo | nugger: i needed something like that, more technical data ;) |
00:11.05 | bilo | not necessarily web-based |
00:11.08 | bilo | *nugget |
00:11.29 | tzanger | kram: I want to push the TDM4XXP cards, I really really do, but they simply are not reliable |
00:11.42 | jero | i 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 ? |
00:11.52 | tzanger | case in point. |
00:12.02 | tessier | doh...I used playback instead background |
00:12.47 | kram | tzanger: what do you mean they have to be reloaded |
00:12.50 | kram | what is the symptom? |
00:12.57 | gigagod | Yeah I have spent 3 weeks trying to get that card to work correctly |
00:13.22 | tzanger | kram: jero's exact symptom on FXS, FXO problem is more it stops answering or won't hang up |
00:13.24 | kram | did you call tech support? |
00:13.38 | kram | tzanger: yah, we found a couple of people with FXO issues and have a fix for those |
00:13.46 | kram | it seems to be system specific and only started showing up recently |
00:13.48 | jero | tzanger, same fxs problem so. |
00:13.50 | bilo | well, 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 |
00:13.53 | bilo | so i can watch * |
00:14.18 | tzanger | kram: 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) |
00:14.25 | jero | kram, I have been experiencing this problem since I got the first card in july (asterisk was not 1.0 yet) |
00:14.45 | ariel_ | bilo, it's your router timing out. linksys does that allot here. |
00:14.59 | kram | jero: is this fxs or fxo? |
00:15.03 | jero | kram, fxs |
00:15.06 | tzanger | I'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 |
00:15.08 | kram | hrm... |
00:15.18 | bilo | ariel: i have another ssh window with vi open, and it doesn't time out? |
00:15.21 | Atacomm | has the FXO module even cleared Part68 yet? I looked up Digium's Part68 filings a week ago and didnt see it |
00:15.29 | bilo | ariel: ssh to the same server :| |
00:15.32 | jero | it occurs at least 1 time a day actually, because we use this analog phone more often |
00:15.34 | kram | it's in part 68 |
00:15.41 | tzanger | the 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 |
00:15.45 | kram | it passed we're just waiting on paperwork |
00:16.04 | kram | (and already passed 15, again waiting on paperwork) |
00:16.36 | tessier | hmm...now there is just a tiny bit of echo on my zap lines... |
00:16.40 | Manipura | Hello everyone |
00:16.46 | Atacomm | ah, 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 |
00:16.48 | tzanger | tessier: I find that is almost impossible to get rid of |
00:16.51 | jero | reloading the module fixes then I guess its possible to fix it with software |
00:17.11 | tessier | tzanger: You may be right. echocancel=yes is turned on in zapata.conf etc. |
00:17.19 | tzanger | jero: 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 |
00:17.42 | twisted[work] | WAY OFF TOPIC - any php gurus around that want lend a learner a hand? |
00:17.43 | tzanger | we get echo on some of our calls out our PRI -- 99% are fine but some specific numbers in specific exchanges are just horrible |
00:17.45 | kram | we've had a handful of customer call about the fxo getting lost but we finally were able to duplicate it and fix it |
00:17.54 | tzanger | kram: excellent news! |
00:17.55 | bilo | ariel: any other ideas? |
00:18.05 | twisted[work] | kram, same issue we were having? |
00:18.10 | tzanger | kram: is there anything us mere mortals can do to help you with the FXS issues? |
00:18.11 | Manipura | twisted[work] I wouldn't call myself a guru |
00:18.16 | kram | yes, the one you had |
00:18.19 | tessier | tzanger: 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? |
00:18.23 | twisted[work] | kram, sweet! |
00:18.25 | Manipura | but I might be able to help |
00:18.29 | kram | did christian ever get the boards with the extra caps? |
00:18.39 | twisted[work] | yea, we shipped them |
00:18.54 | kram | yah that should fix it, it did for mog |
00:18.57 | tzanger | tessier: 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 |
00:19.06 | twisted[work] | kram, good : |
00:19.07 | twisted[work] | er ;) |
00:19.11 | tzanger | kram: what caps need to be augmented? I am an EE :-) |
00:19.16 | twisted[work] | kram, gonna redesign the boards to include the extra cap? |
00:19.43 | jero | tzanger, I'm experiencing the problem once a day |
00:19.46 | jero | at least |
00:19.49 | gigagod | PASS / FAIL ---- DON'T CARE.... I WANT MY BOX RUNNING. |
00:19.51 | tzanger | jero: WOW... mine's NEVER that bad |
00:20.07 | tzanger | jero: get a ticket with digium and see if they'll warrant the boards or if there is something else they can do |
00:20.13 | tzanger | jero: you're not sharing interrupts or anything are you? |
00:20.58 | jero | I dont think so, i'll check |
00:21.16 | jero | I had this problem with my first board, then got it again with the second one |
00:21.43 | tzanger | hmm 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? |
00:21.47 | gigagod | What other cards are there out there that asterix will support |
00:21.50 | jero | /proc/interrupts |
00:21.56 | jero | <PROTECTED> |
00:22.06 | jpablo | anybody in sipphone.com can help me dialing 1-747-6675276 .. ? |
00:22.10 | tzanger | jero: yikes |
00:22.20 | jero | tzanger, ! |
00:22.24 | tzanger | both TDM cards and ethernet on one interrupt... that's not good :-) |
00:22.29 | Atacomm | gigagod: voicetronix i hear, very nice analog cards |
00:22.46 | jero | Can I force the zaptel driver to use another int ? |
00:23.08 | tzanger | jero: that's a fight between your BIOS and the PCI subsystem |
00:23.22 | tzanger | i.e. you need to shuffle cards around, see if you can force them to separate IRQs inside the BIOS, etc. |
00:23.28 | jero | This tends to be boring on PCs |
00:23.50 | asuffield | sharing pairs are usually tied to a given slot. you can change the interrupt, but the ethernet device will change too |
00:23.54 | asuffield | move 'em |
00:24.01 | gigagod | What are these TDM120's I have been hearing about. Anyone know anything about them? |
00:24.33 | tzanger | gigagod: yup we're discussing them right now :-) |
00:24.43 | gigagod | cool |
00:24.43 | jero | tzanger, It seems the problem is proportional to the # of times the fxs port is used |
00:24.50 | tzanger | asuffield: unless they're all assigne the same IRQ in the BIOS |
00:24.55 | file[laptop] | fooooooooood is here |
00:25.02 | asuffield | that would be highly abnormal |
00:25.06 | tzanger | jero: 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) |
00:25.14 | jero | yup |
00:25.24 | r0d3nt | Hello everybody =) |
00:25.26 | Atacomm | gigagod: we're expecting prototypes in a few weeks, we've turned the TDM120 T1/E1/J1 boards over to our contract mfg for prototyping |
00:25.34 | tzanger | asuffield: yes but you tell the BIOS to do everything auto and you get weird results sometimes |
00:25.39 | r0d3nt | What is considered "normal" delay time to initiate a SIP Asterisk phone call to a T1/PRI ????? |
00:25.55 | r0d3nt | 5+ seconds ? |
00:26.09 | asuffield | second or so, sometimes less |
00:26.12 | gigagod | nice... When can I get my hands on several! |
00:26.46 | r0d3nt | asuffield, why would it take more then 4-5 seconds to initiate a SIP call ? |
00:26.54 | asuffield | because it's broke |
00:26.54 | r0d3nt | sometimes up to 8 seconds |
00:26.56 | file[laptop] | cool, they gave me 5 strips instead of 4 |
00:27.05 | tessier | hmm...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. |
00:27.07 | r0d3nt | asuffield, well then my asterisk vendor is a POS. |
00:27.13 | jero | soon a new softphone to be released |
00:27.15 | tessier | And even that one isn't working but that's my fault I'm sure. |
00:27.17 | Atacomm | gigagod: soon i hope :) the thing is tiny, its about 1 cm longer than the TE410P.... even with all the processing power onboard |
00:27.32 | KalD|Work | anyone 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. |
00:28.31 | r0d3nt | asuffield, you really thing something is broken ?? |
00:28.39 | gigagod | thats ok... I have 17" of space in that RLX server |
00:28.46 | jero | is anybody using softphones ? |
00:29.22 | tzanger | jero: I've used Xlite and am using Firefly |
00:29.24 | gigagod | that Wildcard just doesn't like CenturyTel's T1s |
00:29.44 | Atacomm | gigagod: our main card is 13 cm, our daughter card is 10 cm |
00:29.46 | jero | we'll release a new one in some days |
00:30.29 | gigagod | I use a softphone on my laptop... X10, can't get it to work though |
00:30.57 | gigagod | sweet size... That will go beautifully in the RLX boxes |
00:31.33 | Atacomm | but remember, the expansion card uses a PCI slot, even though it doesnt connect into the slot.... due to component height and ports/bracket |
00:31.40 | r0d3nt | What is considered "normal" delay time to initiate a SIP Asterisk phone call to a T1/PRI ?? |
00:31.52 | tzanger | r0d3nt: seconds |
00:31.52 | ScaredyCat | none |
00:31.57 | tzanger | like 1 or 2 |
00:32.02 | ScaredyCat | MAX |
00:32.08 | gh0st | oi r0d3nt |
00:32.15 | r0d3nt | tzanger, under 5 seconds ? |
00:32.15 | gh0st | fancy seeing you here |
00:32.21 | r0d3nt | ya.. imagine that =) |
00:32.23 | tzanger | do you have osm ekind of "overlap" dialing mode on the SIP device (not hte right term) |
00:32.24 | gh0st | heh |
00:32.27 | gigagod | Does it draw any power from the PCI slot? Is there a website that I can go to to find out info about the TDM??? |
00:32.39 | r0d3nt | tzanger, i do not believe so... |
00:32.47 | tzanger | r0d3nt: definitely, it's under 2 seconds with my damn KSU -> * (PRI) |
00:33.02 | r0d3nt | tzanger, ok.. so it sounds like something is incorrectly setup... |
00:33.06 | Atacomm | gigagod: 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) |
00:33.09 | r0d3nt | it takes almost 5-8 seconds @ times... |
00:33.19 | r0d3nt | this is a dual xeon 2.4ghz with a gig of ram.... |
00:33.27 | kram | tzanger: it's a 200pf across reset to ground |
00:33.47 | gigagod | Then I definately want that card!!! |
00:33.51 | kram | as bizarre as this seems, it keeps ESD from causing a partial reset |
00:34.03 | tzanger | kram: which pin is reset (on the module or on the TJ320) |
00:34.17 | kram | on the module |
00:34.26 | kram | it should be as close to the reset pin as possible |
00:34.36 | gigagod | What do you think Kram? |
00:34.42 | tzanger | kram: that makes a lot of sense actually |
00:35.00 | tzanger | kram: I am going to give that a shot and let you know what I find here |
00:36.14 | r0d3nt | tzanger, I'm using Cisco 7940G's, * and a CT1 from TelePacific.... almost 10 seconds until the phone starts ringing.... |
00:36.26 | r0d3nt | i need to remember my password for r0d3nt|m |
00:36.37 | tzanger | r0d3nt: are you sure you don't have som ekind of huge digittimeout and are dialing "live" ? |
00:37.45 | r0d3nt | nomad_, 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.... |
00:38.01 | r0d3nt | tzanger, No .. i'm not sure... and Yes, I am dialing "live" |
00:38.43 | *** join/#asterisk coolschool (~dan@22.247.adsl.brightview.com) |
00:38.44 | tzanger | r0d3nt: that'll be your digittimeout killing you |
00:38.57 | tzanger | r0d3nt: show application digittimeout and put that at the start of your dialplan |
00:38.57 | coolschool | hi all |
00:38.58 | gigagod | will the documentation for the TDM be added in to the Asterisk docs |
00:39.07 | r0d3nt | which conf file ? extensions ? or sip .conf ? |
00:39.19 | r0d3nt | ok i'll check it out... |
00:39.27 | r0d3nt | tzanger, I appreciate the info.. lemme check it... |
00:39.35 | ScaredyCat | r0d3nt: watch the * console, how long b4 asterisk tries to dial out |
00:39.36 | *** join/#asterisk mr_monkey (~root@201.129.239.6) |
00:39.37 | Atacomm | gigagod: dont know, we will document and support everything ourselves, we havent approached anyone on the asterisk project officially, there has been some backchannel conversations |
00:39.47 | r0d3nt | I didn't build this system.. i paid someone else to.. and I'm finding out all these problems.... |
00:39.54 | r0d3nt | ScaredyCat, OK |
00:40.00 | mr_monkey | does 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) |
00:41.14 | tzanger | r0d3nt: 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 | _Vile | MUHAHAHA |
00:43.34 | redder86 | Anyone got a link for Linux QoS that is different than this one: http://www.voip-info.org/tiki-index.php?page=QoS%20Linux |
00:43.50 | tzanger | redder86: I use this script (which is on that wiki page) www.mixdown.ca/~andrew/dump/rc.tc |
00:44.21 | gigagod | QOS: http://qos.ittc.ukans.edu/ |
00:44.57 | *** part/#asterisk eKo1 (~abc@63.245.57.70) |
00:45.00 | redder86 | tzanger: that script looks awfully involved. |
00:45.06 | tzanger | redder86: nah it's very straightforward |
00:45.17 | redder86 | what is tc ? |
00:45.24 | agronqui | tcl |
00:45.30 | mr_monkey | does exists a manual to configure asterisk with postgres ? |
00:45.41 | tzanger | basically 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 |
00:46.08 | tzanger | simiarly 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) |
00:46.24 | brettnem | hey all |
00:46.34 | gigagod | no idea... just googled "linux qos" |
00:46.35 | brettnem | hey, does anyone use setgroup and check group? |
00:46.52 | redder86 | tzanger: okay, is this for incoming traffic as well as outgoing? |
00:47.20 | brettnem | doesn't look like setgroup ever decrements the counters when the channel hangs up.. hmmm |
00:47.34 | tzanger | redder86: 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 |
00:47.51 | tzanger | redder86: consequenty I also drop excessive UDP traffic in case the application does some kind of application-layer throttling |
00:48.27 | gigagod | Best solution: Increase the bandwith |
00:48.58 | ScaredyCat | minimise hops |
00:49.08 | brettnem | any ideas for setgroup?? |
00:49.28 | ScaredyCat | brettnem: what about it? |
00:50.06 | r0d3nt | doesn't look like there is a digittimeout specified.... |
00:50.12 | redder86 | gigagod: 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. |
00:50.13 | brettnem | well I'm using it.. it increments a counter, but never decrements it when the channel is released |
00:50.30 | redder86 | gigagod: methinks one must do both |
00:51.00 | redder86 | gigagod: I don't think that QoS is really optional unless the interface is but rarely used for anything other than VoIP. |
00:51.01 | file[laptop] | Purolator is such a nice company, they even call to reschedule deliveries and stuff |
00:51.07 | tzanger | r0d3nt: specify like 3 seconds |
00:51.19 | r0d3nt | tzanger, OK.. i'll do that on a test extension |
00:51.20 | tzanger | file[laptop]: :-) |
00:51.24 | r0d3nt | thanks |
00:51.26 | brettnem | hmm...looks like hangups might not being detected |
00:51.42 | file[laptop] | and on my last delivery the guy came back a second time to try |
00:51.57 | redder86 | Purolator? |
00:52.12 | file[laptop] | shipping company |
00:52.22 | Umaro | Okay 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? |
00:52.42 | ScaredyCat | crossed if it's e1 >E1 |
00:53.19 | Umaro | ScaredyCat, how do you mean? |
00:53.43 | ScaredyCat | if you are connecting an E1 to an E1 on the siemens then cross cable |
00:53.46 | tzanger | Purolator is a private company that is owned by Canada Post (the federal mail carrier) -- talk about conflict of interest |
00:53.51 | ScaredyCat | is required |
00:54.16 | redder86 | ah, 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) |
00:55.03 | Umaro | ScaredyCat, 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. |
00:55.26 | visik7 | where can I find a single gsm gateway (h323 or serial better h323) |
00:55.52 | Umaro | ScaredyCat, So the connection between the E1 and * should be straightthrough, right? |
00:56.05 | tzanger | Umaro: I believe so yes |
00:56.09 | tzanger | you need cross to go to channel banks |
00:56.16 | tessier | Cool, so I have my snom220 properly dialing the local 3 digit extensions without having to press enter. |
00:56.28 | tessier | Now to figure out how to make it recognize a 9 and then a 10 digit number. |
00:56.33 | tzanger | if 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 :-) |
00:56.37 | redder86 | tzanger: this guy: http://www.voip-info.org/tiki-index.php?page=Linux%20Router doesn't even use tc. Is he wrong? |
00:56.47 | brettnem | so my setgroup checkgroup problem I think is here: http://pastebin.ca/2866 |
00:56.55 | Umaro | tzanger: well, the "E1" is from the multiplexer. the multiplexer is connecting to all the isdn lines |
00:57.03 | tessier | Unfortunately it sort of looks like I can only have one dialplan entry per line which is rather lame. |
00:57.13 | brettnem | I have checkgroup set to 2.. and it shows 2 channels up, but sip show inuse shows no channels in use.. any ideas?? |
00:57.26 | tzanger | redder86: he's using basic marking -- no queuing at all |
00:57.35 | tzanger | it'll work too |
00:57.42 | tzanger | just not as robust / configurable |
00:58.04 | brettnem | looks like my answer supervision isn't working somewhere.. |
00:58.28 | redder86 | tzanger: as for the configurations he poses versus the one you pose, what is the effective difference? |
00:58.48 | brettnem | any ideas? |
00:58.49 | tzanger | redder86: effective difference is that he can't shape or limit bandwidth |
00:59.12 | tzanger | redder86: 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) |
00:59.23 | tzanger | then so long as you don't totally hose your link you should be fine |
00:59.36 | tzanger | I have 35 customers behind my connection, I needed something a little more :-) |
00:59.57 | cypromis | Â/window 34 |
01:00.01 | cypromis | humpf |
01:00.23 | ScaredyCat | useless |
01:00.32 | cypromis | yeah |
01:00.35 | redder86 | tzanger: it's not just me |
01:01.00 | redder86 | tzanger: well, here it is, but not at the server where I'm worried about it |
01:02.22 | Umaro | tzanger: could you please look at my configs? http://pastebin.ca/2867 |
01:02.32 | redder86 | tzanger: why does it make a difference how many users there are behind the connection? |
01:02.42 | tzanger | redder86: ask yourself what you need to do |
01:02.44 | tzanger | then tell me |
01:02.55 | *** join/#asterisk Darwin35 (~darwin35@pool-68-162-181-243.pitt.east.verizon.net) |
01:03.37 | redder86 | tzanger: I just want to dedicate an adequate portion of bandwidth to IAX2 |
01:03.45 | tzanger | Umaro: 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 |
01:03.55 | tzanger | that is traffic shaping, you need tc |
01:04.02 | redder86 | tzanger: okay, thanks. |
01:04.13 | tzanger | if yuou just want to give highest priority to iax2 you *should* be able to get away with straight fwmark |
01:04.23 | tzanger | but all bets are off if you're capable of flooding the link |
01:04.37 | Umaro | tzanger: ok |
01:04.39 | tzanger | Umaro: ztcfg -vvv (everything looks fine?) |
01:04.47 | Umaro | tzanger: yep, no errors |
01:04.55 | redder86 | tzanger: ah, okay, I think I understand the difference between prioritization and traffic shaping now |
01:04.56 | Umaro | tzanger: zttool tells me span 1 is OK |
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01:05.07 | tzanger | redder86: my script does both |
01:05.11 | tzanger | Umaro: the *span* is fine |
01:05.16 | tzanger | you're having a problem on the data link |
01:05.22 | tzanger | i.e. the D channel isn't where you think it is |
01:05.33 | tzanger | is there any way to query the existing Siemens config to see how it's set up? |
01:05.38 | Umaro | but doing a 'pri intense debug span 1' doesn't give me anything except saying it's sending a SABME request |
01:05.41 | Umaro | No.. |
01:05.55 | Umaro | well, I think it's channel 16, and when I try it on any other channel, it can't configure channel 16 as a bchan |
01:06.06 | ScaredyCat | you do have one end as cpe and one as net? |
01:06.09 | Umaro | it says the signalling is wrong |
01:06.12 | Umaro | span 1 is cpe |
01:06.20 | Umaro | That's all i'm trying to get working for the moment |
01:06.24 | tzanger | Umaro: 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 |
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01:06.50 | tzanger | you reran ztcfg -vvv after every change to /etc/zaptel.conf? |
01:06.57 | Umaro | tzanger: yep |
01:07.05 | tzanger | Umaro: hmm |
01:07.13 | tzanger | short of calling your telco, I don't know. |
01:07.27 | Umaro | there isn't a telco.. at least not for the E1 |
01:07.28 | tzanger | from what I understand of E1 (I'm in Canada, we use T1), dchan on E1 is typically channel 16 |
01:07.30 | Umaro | it's a multiplexer |
01:07.38 | tzanger | Umaro: uh |
01:07.42 | tzanger | Umaro: the MUX is PRI?? |
01:07.44 | Umaro | I'm told the d channel is on 16 in this case too |
01:08.00 | Umaro | tzanger: :) |
01:08.01 | tzanger | Umaro: every MUX I've dealt with is straight E1 or T1, not PRI |
01:08.09 | Umaro | tzanger: so then what signalling should I use? |
01:08.25 | tzanger | and then your signalling would be fxs_ls or fxs_ks generally |
01:08.52 | tzanger | Umaro: so you have a DS3 being split off in a MUX or what |
01:08.57 | tzanger | where's this MUX |
01:09.35 | Umaro | no, there's like however many ISDN lines coming in |
01:09.38 | Umaro | like BRIs |
01:09.42 | tzanger | oh |
01:09.47 | tzanger | wow man I really don't know in this case |
01:10.00 | tzanger | BRI just blows my mind :-) |
01:10.33 | Umaro | it says "signalling requested is FXS kewlstart but line is in PRI signalling signalling" |
01:10.40 | Umaro | when I try to start asterisk |
01:11.07 | tzanger | Umaro: well if it's a buch of BRIs coming in I doubt it's CAS E1 |
01:11.14 | tzanger | sorry for the confusion |
01:11.25 | tzanger | and if tha't shte error you did not configure zaptel.conf |
01:11.27 | tzanger | and rerun ztcfg |
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01:11.43 | *** join/#asterisk ariel_ (~Ariel@ssf-office.corp.race.com) |
01:11.57 | ariel_ | ok I am back. Argh network problems. |
01:11.57 | Umaro | tzanger, no, I did run ztcfg, and didn't get errors |
01:12.23 | Umaro | ztcfg says each of the channels are "Invidual clar channel" |
01:12.26 | Umaro | ariel_, wb |
01:12.34 | Umaro | er Individual, lol |
01:12.51 | ariel_ | Umaro, your still having E1 problems? |
01:13.01 | tzanger | ariel_: yeah he is |
01:13.22 | tzanger | it's beyond me as I know nothing about E1 and even less about E1 BRI->PRI muxes |
01:13.29 | Umaro | ariel_: 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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01:13.58 | ariel_ | Umaro, have you called digium? |
01:14.11 | Umaro | yes, until they closed |
01:14.14 | tzanger | ariel_: it's not a digium problem |
01:14.19 | tzanger | it's a ocnfiguration issue from what I can gather |
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01:14.59 | ariel_ | ah I see... (I will then go back to my working area). |
01:16.43 | Faithful | Hey guys what CPU would I need to handle 2 incomming ISDN lines? |
01:16.55 | tzanger | Faithful: anything |
01:17.00 | Weezey | anyone have any SPA-3000s? |
01:17.05 | ariel_ | anythign will do for just 2 |
01:17.32 | mutilator | i suggest 200mhz |
01:17.37 | Faithful | Pentium 133 with 32Mb ram? |
01:17.56 | tzanger | Faithful: that's getting on the low side |
01:17.59 | redder86 | tzanger: does your rc.tc get run automatically by something else, or do you just run it from rc.local ? |
01:18.00 | Umaro | tzanger, here's the info I got from the multiplexer guy: framing: CAS Signaling: edss1 line data on channel 16 coding: hdb3 |
01:18.08 | tzanger | I was able to hadnle a POTS line and FXS port wiht a P90 (no MXX) but I could not do iLBC |
01:18.15 | tzanger | redder86: run it from rc.local |
01:18.28 | Faithful | Ok got a few PII 300 Celerons around |
01:18.33 | tzanger | Faithful: that's plenty |
01:18.44 | tzanger | Umaro: CAS signalling but D channel data on line 16? wuh? |
01:19.09 | tzanger | Umaro: try signalling fxs_ks on channels 1-15,17-32 |
01:19.10 | ariel_ | what is a PII celeron? |
01:19.17 | tzanger | ariel_: they exist |
01:19.20 | tzanger | PII with 1/2 the cache |
01:19.34 | *** part/#asterisk shifter (~shifter@ip18.unival.com) |
01:19.43 | tzanger | Umaro: I'm guessing that channel 16 is something weird so I'm skipping it :-) |
01:19.44 | ariel_ | Those were the first generation Celeron's then. |
01:19.47 | tzanger | ariel_: yes |
01:20.19 | Umaro | tzanger: fair enough |
01:21.05 | Faithful | So 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) |
01:21.18 | Weezey | FXS = line or extenstion? |
01:21.24 | ariel_ | Faithful, yes that will work. |
01:21.33 | tzanger | Faithful: it can but be careful of the VIAs -- you must disable MMX in zaptel (and asterisk?) |
01:21.39 | Faithful | I'm just looking for a low power solution since it will be 24x7 |
01:21.47 | tzanger | Weezey: FXS port = plug a phone into it. FXO port = plug a phone LINE into it |
01:21.48 | ariel_ | FXS is a line you plug in a analog phone to. |
01:22.36 | Faithful | Which linux software phone is the best with * |
01:22.44 | ariel_ | I hope we don't get into the fxs fxo thing again... Station or office please. |
01:22.51 | tzanger | Faithful: I haven't found one that works well |
01:23.44 | Faithful | Boooo! |
01:24.11 | Umaro | tzanger: gives me an error when I start * that says "Signalling requested is FXS Kewlstart but line is in PRI Signalling signalling" |
01:24.43 | gh0st | Umaro: signalling = pri_cpe |
01:25.00 | Faithful | tzanger: are you saying that all MMX must be disabled or only with VIA |
01:25.27 | Umaro | gh0st: tried that, don't get anything when I do pri intense debug span 1 except something talking about SABME |
01:25.37 | tzanger | Umaro: you did not run ztcfg, or you did not configure zaptel.conf |
01:25.39 | Umaro | which i'm told means the d-channel isn't syncing up |
01:25.46 | tzanger | Faithful: only with VIA, it's broken on VIA |
01:26.08 | Umaro | tzanger: dude... I totally did. I wish that was the problem, but i've tried running it every single time I change anything |
01:26.12 | Umaro | back in a few |
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01:26.33 | [Outcast] | yo twisted[work] |
01:26.42 | Weezey | If I get a SPA-3000 they have two FXS and one FXO. How many calls can an asterisk box handle? |
01:26.49 | Weezey | simultaneously |
01:26.59 | ariel_ | Weezey, only one fxs and one fxo |
01:27.04 | gh0st | Umaro: what channel is the d-channel? |
01:27.20 | tzanger | show me your zaptel.conf now |
01:27.21 | ariel_ | Weezey, one inbound one outbound only via the sipura. |
01:27.52 | Weezey | ariel: how many sipura's could I have making calls at once? |
01:28.14 | DrukenHME | has anyone used a generic version of the t100p ? |
01:28.37 | tzanger | DrukenHME: there's a generic version of the single-span T1 card? |
01:28.43 | *** join/#asterisk Mezenv (Elshar@ip205-69.oregonfast.net) |
01:29.24 | DrukenHME | tzanger: i belive so... |
01:29.32 | tzanger | DrukenHME: URL? |
01:30.01 | ariel_ | 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. |
01:31.16 | DrukenHME | tzanger: maybe not... i dunno, i thought i seen one.... |
01:31.29 | Weezey | awesome. Is a 1.5GHz P4 enough? |
01:32.04 | ariel_ | Weezey, in most case yes more then enough. |
01:32.26 | DrukenHME | my 400 handles voip.. hehehe |
01:33.36 | Weezey | I need an inexpensive SIP phone, does anyone have any good recommendations? |
01:33.50 | tzanger | Weezey: BT101 |
01:33.51 | ariel_ | what do you call inexpensive? |
01:34.01 | Weezey | $100 ish |
01:34.14 | ariel_ | I am waiting for the Sipura one. It's on backorder till the 15th they told me. |
01:34.31 | Weezey | I just got the email from Voxilla, it's now January |
01:35.22 | ariel_ | Weezey, I have used the snom and the polycom's there good fones. But there in the 140 to 200 mark. |
01:35.49 | Weezey | where did you get them from? |
01:38.49 | DrukenHME | www.voipsupply.com has voip phones under 100 USD |
01:39.54 | Weezey | I don't have a huge budget to work with, but bottom line is, I need a coupla phones. |