00:00.07 | drray | now that cisco is wising up |
00:00.07 | Strom_C | can you run SIP on the 70s now? |
00:00.13 | CerealVore | i don't think so yet |
00:00.13 | drray | 8.2 firmware |
00:00.14 | drray | yes |
00:00.21 | CerealVore | wow, they finally released it |
00:00.24 | Strom_C | oooooooooh |
00:00.31 | Iamtheman | tks |
00:00.51 | CerealVore | is there anything particularly special about the 7970s? apart from having a funky background on your phone? |
00:00.56 | drray | color |
00:01.03 | drray | and 5 menu buttons |
00:01.04 | CerealVore | yeah |
00:01.08 | CerealVore | but they don't do video |
00:01.09 | drray | so no |
00:01.12 | CerealVore | so why would you bother? |
00:01.19 | drray | but I still want one |
00:01.50 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
00:01.50 | *** mode/#asterisk [+o russellb] by ChanServ |
00:01.52 | CerealVore | heh |
00:01.57 | CerealVore | i can't deny, they are funky |
00:02.52 | Iamtheman | and what's the best quality/price/ease of use model ? |
00:03.05 | Strom_C | 7940 / 7960 |
00:03.08 | Strom_C | :) |
00:03.18 | drray | the ciscos are a pain getting sip firmwares |
00:03.21 | drjones1 | yea |
00:03.25 | drjones1 | but asfterr you do |
00:03.26 | drjones1 | they are great |
00:03.28 | drray | yeah |
00:03.31 | drjones1 | i was recommened some polycomms outta here |
00:03.36 | drjones1 | i bout 15 of those bitches |
00:03.38 | drjones1 | just the other day |
00:03.40 | drjones1 | i picked one up |
00:03.45 | drjones1 | fried transitors |
00:03.45 | Strom_C | the polycoms are a huge pain in the ass to set up |
00:03.48 | drjones1 | i didn't even touch anything |
00:03.50 | drjones1 | fuck a pollycom |
00:03.56 | drray | cisco is a good solid phone |
00:03.59 | drjones1 | indeed |
00:04.02 | drjones1 | thanks guys |
00:04.02 | drjones1 | later |
00:04.12 | Iamtheman | and 3Com ? |
00:04.12 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
00:04.12 | drray | but any voip phone will be persnickity |
00:05.32 | *** join/#asterisk coppice (n=chatzill@18.162.17.210.dyn.pacific.net.hk) |
00:05.36 | dlynes_home | so i guess no such thing as a voip phone with firmware that works reliably? |
00:05.39 | Iamtheman | what's the difference btwn G and G-GE, only Gigabit ? |
00:06.08 | Strom_C | dlynes_home, if you want reliability in a phone, get a Western Electric :) |
00:06.19 | dlynes_home | Iamtheman: nbx works with sip? |
00:06.21 | drray | and an adit |
00:06.23 | drray | :) |
00:06.42 | dlynes_home | Strom_C: what's western electric? an analog phone? |
00:06.56 | Strom_C | dlynes_home, yow, and you call yourself a phone person :) |
00:07.05 | Strom_C | western electric was the manufacturing arm of the Bell System |
00:07.08 | dlynes_home | Strom_C: i call myself a computer guy |
00:07.18 | dlynes_home | I know dick about American manufacturers |
00:07.26 | Strom_C | made every piece of AT&T's hardware from 188x through 1984 |
00:07.37 | dlynes_home | ah. |
00:07.42 | drray | I miss my old southwestern bell phone |
00:07.50 | drray | that was a solid device |
00:07.52 | dlynes_home | northern telecom is better, anyways |
00:07.58 | drray | exhibit a your honor |
00:08.04 | dlynes_home | That's who made almost every single phone I've ever used |
00:08.07 | Strom_C | became "AT&T Technologies" after divestiture, and today they're called "Lucent" |
00:08.14 | Strom_C | Nortel is awesome |
00:08.17 | h3x0r | and then Avaya |
00:08.17 | dlynes_home | well now |
00:08.18 | Qwell | ugh |
00:08.20 | dlynes_home | I've heard of Lucent |
00:08.28 | dlynes_home | and AT&T Bell Labs |
00:08.32 | dlynes_home | but never western electric |
00:08.35 | Strom_C | h3x0r, Avaya is the business systems division of western electric |
00:08.37 | h3x0r | and nortel sold the phone stuff to Aastra |
00:08.40 | Strom_C | or was, anyway |
00:08.47 | Strom_C | dlynes_home, how old are you? |
00:08.53 | dlynes_home | Strom_C: 37 |
00:09.01 | Qwell | old men... |
00:09.06 | dlynes_home | old? |
00:09.07 | dlynes_home | wtf? |
00:09.10 | Strom_C | heh, and never heard of western electric |
00:09.11 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
00:09.14 | Strom_C | thats funny |
00:09.16 | dlynes_home | Strom_C: like i said |
00:09.20 | dlynes_home | Strom_C: i'm not american |
00:09.44 | Strom_C | ok, but you heard of something called "Northern Electric" at one time, right? |
00:09.54 | dlynes_home | nope |
00:10.00 | dlynes_home | Northern Lights, yeah |
00:10.06 | Strom_C | Northern Electric -> Northern Telecom -> Nortel |
00:10.36 | dlynes_home | how many decades ago was it called Northern Electric? |
00:10.46 | Strom_C | name change happened in 1975 |
00:11.00 | dlynes_home | yeah...I was 6 at that time |
00:11.11 | Strom_C | but there was tons of Northern Electric stuff in service for years after that |
00:11.24 | coppice | Bell Northern Research + Northern Telecom -> Nortel |
00:11.32 | Qwell | Strom_C: how much of that is now currently in your house? |
00:11.32 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
00:11.33 | Qwell | ;) |
00:11.34 | dlynes_home | Strom_C: in the 70's, we didn't have a phone line, electricity, plumbing, or any of that stuff |
00:11.47 | Qwell | silly canadians |
00:11.49 | Strom_C | Northern Electric was once a division of Western Electric |
00:11.50 | Strom_C | Qwell, all of it |
00:12.07 | dlynes_home | Strom_C: coppice sounds more correct |
00:12.10 | Iamtheman | good place to buy cisco phones in Canada ? |
00:12.13 | dlynes_home | Strom_C: nortel was never an american company |
00:12.19 | Qwell | Strom_C: how large IS your collection? |
00:12.40 | coppice | companies are so jumbled up if they are old. daimler chrysler can't call a car a daimler, as ford have the tracemark for that |
00:12.42 | h3x0r | its hard to find cisco routers in canada let alone phones |
00:12.52 | Strom_C | dlynes_home, Nortel was once Northern Electric, a subsidiary of Western Electric, and Bell Canada was once a subsidiary of AT&T. |
00:12.56 | dlynes_home | h3x0r: you must not look very hard |
00:13.15 | dlynes_home | Strom_C: bell canada is a subsidiary of BCE |
00:13.27 | h3x0r | i spent much time trolling bell canada, telegloba, etc |
00:13.30 | h3x0r | teleglobe |
00:13.32 | h3x0r | back in the daty |
00:13.35 | h3x0r | and uunet canada |
00:13.40 | Strom_C | dlynes_home, you're not thinking far enough back in time |
00:13.45 | h3x0r | everything was nortel or 3com |
00:13.53 | dlynes_home | Strom_C: BCE also owns teleglobe, HK Telecom, UK Telecom, Transcanada Pipelines, ... |
00:14.09 | dlynes_home | h3x0r: uunet canada |
00:14.14 | dlynes_home | h3x0r: those guys were a major ripoff |
00:14.19 | h3x0r | yep |
00:14.26 | coppice | HK telecom is owned by the Li family |
00:14.33 | coppice | and its called PCCW now |
00:14.38 | Qwell | The Li Family is owned by BCE |
00:14.40 | Qwell | :D |
00:14.54 | dlynes_home | coppice: perhaps, but last time i checked my annual report, BCE owned 20 or 30% of HK T&T |
00:14.54 | dongs | what |
00:15.06 | Strom_C | BCE is owned by my left ring finger |
00:15.23 | Iamtheman | what's a good price for a used 7940 and a 7960 & |
00:15.28 | h3x0r | bce can suck my left nut |
00:15.36 | Qwell | Iamtheman: $50 |
00:15.39 | Strom_C | Iamtheman, I got one for free once |
00:15.39 | h3x0r | telus can suck my right nut |
00:15.54 | dlynes_home | h3x0r: they did...that's why they moved to First Canadian Place from Montreal :p |
00:15.54 | Strom_C | h3x0r, so where does that leave MTS? |
00:16.03 | h3x0r | licking my asshole? |
00:16.10 | russellb | h3x0r: enough |
00:16.11 | coppice | Its sad when your main telco ends up with a name like Pacific Century Cyber Works. quite embatassing, really :-) |
00:16.36 | dlynes_home | Cyber works? wtf? |
00:17.34 | dlynes_home | h3x0r: you don't like telus? wtf is wrong with you man? |
00:17.45 | h3x0r | i didnt say that |
00:17.48 | Strom_C | dlynes_home, maybe he just isnt fond of GTD-5 EAX :) |
00:17.52 | h3x0r | i just said they could balance out the nut sucking |
00:17.55 | coppice | The run the worlds only successful IPTV business. successful means they get lots of subs by more or less giving the ervice free with broadband :-) |
00:17.59 | dlynes_home | Strom_C: ok, whatever that is |
00:18.18 | Strom_C | dlynes_home, class 5 digital switching office that GTE used a lot |
00:18.24 | Strom_C | dlynes_home, and since BC TEL was a subsidiary of GTE... |
00:18.24 | dlynes_home | Strom_C: ah |
00:18.28 | h3x0r | im thinking about moving to vancouver |
00:18.31 | dlynes_home | Strom_C: yeah...that's old news |
00:18.36 | dlynes_home | h3x0r: dood |
00:18.40 | dlynes_home | h3x0r: stay in Toronto |
00:18.44 | dlynes_home | h3x0r: we don't want you here :p |
00:18.49 | h3x0r | then i can pee on telus |
00:18.51 | h3x0r | i live in vegas |
00:18.55 | dlynes_home | h3x0r: oh |
00:18.56 | h3x0r | i used to work in montreal for a while |
00:18.56 | dlynes_home | heh |
00:19.05 | Strom_C | jeez, how can you make telephone jokes in #asterisk and have people not get them?!:) |
00:19.22 | Strom_C | h3x0r, so how's Embarq these days? |
00:19.22 | dlynes_home | Strom_C: i don't know if you noticed or not |
00:19.23 | coppice | telus is a pretty dumb name too |
00:19.32 | dlynes_home | Strom_C: but a lot of us are computer geeks, not telco geeks :) |
00:19.32 | coppice | Telus - we enjoy a good laugh! |
00:19.37 | h3x0r | it reminds me of telstra |
00:19.49 | Strom_C | dlynes_home, bah! |
00:19.52 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
00:20.30 | dlynes_home | Strom_C: the only thing i like about at&t |
00:20.43 | dlynes_home | Strom_C: is when they freely released korn shell :p |
00:22.05 | JoseBravo | Im getting this error triying to call from ext to ext. Jun 19 04:22:59 WARNING[23803]: app_dial.c:803 dial_exec_full: Dial requires an argument (technology/number) |
00:23.46 | SkramX | anyone going to HOPE6? Or even heard of it :P |
00:24.12 | *** join/#asterisk Strom_C (n=strom@12.150.239.132) |
00:24.20 | orlock | Eurgh telstra |
00:24.20 | Strom_C | did I mention this hotel's wifi sucks balls? |
00:24.27 | SkramX | Strom_C: :( |
00:24.35 | Iamtheman | how to change the default ring sound with eyebeam or xten ? |
00:24.36 | Strom_C | so we were on AT&T and ksh when I dropped off |
00:24.43 | SkramX | I have been on the road the last week (just got home) and I totally know what you mean |
00:24.45 | Strom_C | what'd I miss |
00:24.49 | Strom_C | hey SkramX |
00:24.50 | SkramX | I said: |
00:24.51 | SkramX | anyone going to HOPE6? Or even heard of it :P |
00:25.18 | Strom_C | SkramX, dunno. HOPE is a long way away from los angeles |
00:25.21 | SkramX | I need to make my final decision if I want to present about Asterisk in a VPS environment or not |
00:25.32 | hads | JoseBravo: So what does your dial command look like? |
00:25.33 | SkramX | Strom_C: I hear ya.. Im coming from Central Texas |
00:28.13 | Iamtheman | how to change the default ring sound with eyebeam or xten ? |
00:34.38 | Strom_C | SkramX, go to defcon instead, so you can see my talk :) |
00:34.48 | Qwell | Strom_C: get me in free :p |
00:34.58 | SplasPood | Heh, I think I've completely rewritten asterisk's agent system /w realtime, app_addon_mysql, some macros and some dialplan |
00:35.14 | Qwell | SplasPood: good.. publish it, so we can remove chan_agent :P |
00:35.21 | SplasPood | Qwell: heh |
00:35.47 | SplasPood | it'd have to be hacked up a bit to be ripped away from the rest |
00:36.00 | SkramX | Strom_C: about what? |
00:36.01 | coppice | is this a secret agent system? |
00:36.09 | SplasPood | coppice: I suppose |
00:36.21 | SkramX | I think greyarea is presenting.. |
00:36.22 | Strom_C | SkramX, fedex kinko's hack, plus another talk which has yet to be approved |
00:36.39 | SkramX | hmm sounds good |
00:36.46 | SkramX | im only going to hope since i have family up there |
00:36.48 | Qwell | What kind of hack? |
00:36.51 | SplasPood | if only asterisk's realtime extensions wasn't so lame |
00:36.54 | Strom_C | Qwell, free xeroxes! |
00:36.58 | Qwell | ahh |
00:36.58 | SkramX | hehe |
00:37.10 | Qwell | reverse engineer the card system? |
00:37.13 | Strom_C | yes |
00:37.16 | SkramX | im deciding if i should do a talk about asterisk and vpses, etc |
00:37.18 | Qwell | nice |
00:37.31 | coppice | xeroxes are almost free. the toner on the other hand..... |
00:37.40 | Strom_C | asterisk + vps == ew |
00:37.47 | SkramX | Strom_C: why do you say that? |
00:37.50 | Qwell | anything + vps == eww |
00:37.54 | Strom_C | yes, exactly |
00:37.55 | SkramX | A number of my clients like it |
00:38.08 | SkramX | vps == Virtual Private Server == wee |
00:38.11 | SkramX | wee == good |
00:38.13 | SplasPood | VPS for dev == great |
00:38.19 | Qwell | SplasPood: That's about it |
00:38.25 | Qwell | but, even then... |
00:38.36 | SkramX | I actually agree, but it can also be used in the production environment |
00:38.40 | SkramX | but, its debatable |
00:38.46 | SplasPood | I actually love running VMs on my macbook in parallels |
00:38.48 | SplasPood | got asterisk in one |
00:38.52 | Strom_C | if you're a cheapskate, sure |
00:38.55 | SplasPood | openser in the other... |
00:39.04 | SkramX | Strom_C: ah, I guess so |
00:39.20 | SkramX | Meh, dont tell our customers!! |
00:39.25 | SplasPood | I wish I could figure out how to put them all on their own subnet, and then route that through the macbook as a gw |
00:40.23 | SplasPood | ~paste |
00:40.25 | jbot | hmm... paste is see http://paste.husk.org |
00:41.18 | drray | do you use vmware for vps? |
00:41.26 | SkramX | drray: nopers. |
00:41.31 | drray | what do you use? |
00:41.44 | SkramX | linux-vserver.org |
00:41.57 | SkramX | still in some development but we have it stabilized |
00:42.00 | drray | thank you |
00:42.03 | SkramX | sure |
00:45.28 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
00:48.05 | SplasPood | when I do a NoCDR() I get these console warnings: |
00:48.06 | SplasPood | Jun 18 20:46:10 WARNING[29399]: cdr.c:443 ast_cdr_free: CDR on channel 'SIP/C1000-000f34fa1d16-1-6539' not posted |
00:48.06 | SplasPood | Jun 18 20:46:10 WARNING[29399]: cdr.c:445 ast_cdr_free: CDR on channel 'SIP/C1000-000f34fa1d16-1-6539' lacks end |
00:48.11 | SplasPood | is that something to be concerned about? |
00:54.10 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
00:54.18 | *** part/#asterisk pigpen (n=mark@fw.seamans.cc) |
00:58.05 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
00:59.14 | *** join/#asterisk Strom_C (n=strom@12.150.239.132) |
01:01.05 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
01:01.35 | paolob | Hi guys! Where do I specify the format (gsm, ulaw, etc.) of the sound files to play? thank you! |
01:03.40 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
01:05.00 | russellb | paolob: you don't specify it |
01:05.15 | russellb | paolob: asterisk will try to find the best format based on what format the calling channel is using |
01:05.16 | *** join/#asterisk Strom_C (n=strom@m215e36d0.tmodns.net) |
01:05.21 | Strom_C | ok |
01:05.24 | Strom_C | to hell with wifi |
01:05.27 | Strom_C | edge for the win |
01:05.52 | paolob | russellb, I get a "Unable to open MenuEspanol (format ulaw): No such file or directory" error: what does it mean? |
01:06.08 | paolob | do I lack the ulaw format file MenuEspanol? |
01:06.55 | russellb | paolob: it means it couldn't find that file at all |
01:07.30 | russellb | file: go party |
01:08.35 | paolob | russellb, I don't know what I changed in a config file, that file is in /usr/share/asterisk/sounds/es. Before the change asterisk found it, now it doesn't. However in sip.conf I have specified language=es. Shall I specify somewhere else? |
01:09.03 | russellb | try putting it in /var/lib/asterisk/sounds/es |
01:09.49 | SplasPood | hrm... I wonder how I can somehow pass the name of the queue making the call to the Local/ channel I have setup as a member |
01:10.34 | russellb | SplasPood: you can probably do it using variable inheritance |
01:10.55 | russellb | before a caller enters the Queue, do ... Set(__QUEUENAME=whatever) |
01:11.00 | russellb | that is two underscords |
01:11.08 | russellb | s/underscords/underscores/ |
01:11.14 | SplasPood | russellb: hrm, lemme try |
01:11.36 | russellb | then, in the extension for the local channel, NoOp(Coming from queue: ${QUEUENAME}) |
01:11.54 | *** join/#asterisk CoffeeIV (i=rgr@cpe-70-112-100-20.austin.res.rr.com) |
01:12.53 | SplasPood | russellb: bingo, thanks |
01:13.03 | russellb | s/chers/cheers/ |
01:13.05 | russellb | i can't type |
01:13.12 | russellb | you're welcome :) |
01:13.21 | Strom_C | i believe you have my vocoder |
01:13.22 | Qwell | "Do you belieeeevve .." |
01:13.38 | russellb | eep! |
01:14.12 | *** join/#asterisk neilbags-work (n=neilbags@149.171.94.134) |
01:14.28 | CoffeeIV | I am about to set up an * server to try some stuff out, if it works it will go into production -- any reason to stick to Fedora Core 4 instead of using FC 5 ? |
01:14.55 | Qwell | CoffeeIV: none |
01:15.14 | russellb | unless you have FC4 cds and not FC5 |
01:15.28 | CoffeeIV | cool -- that's what I thought -- I have to download the CDs either way |
01:15.40 | Qwell | in that case...get debian :P |
01:15.46 | russellb | word :-p |
01:15.51 | Qwell | (or gentoo..) |
01:16.01 | paolob | russellb, no the issue is that it looks for it in /usr/share/asterisk/sounds: if I put the sound file there it find it. |
01:16.27 | CoffeeIV | I prefer Debian or Slackware, but the other guy on this project is used to redhat/fedora systems |
01:16.42 | russellb | paolob: try adding Set(LANGUAGE()=es) before playing the file |
01:17.19 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
01:20.23 | SplasPood | Nice, I now have Cepstral based queue agent announcements "Call in from <queue name>, press 1 to accept" |
01:20.32 | russellb | SplasPood: nice |
01:20.41 | SplasPood | russellb: ALL THANKS TO YOU! :P |
01:20.48 | russellb | woohoo |
01:21.01 | russellb | at least i accomplished something |
01:21.01 | *** join/#asterisk Iamtheman222 (n=fddfffsa@modemcable237.246-83-70.mc.videotron.ca) |
01:21.09 | Iamtheman222 | I have a problem with the callback |
01:21.11 | russellb | my Makefile changes aren't quite as successful here ... |
01:21.20 | Qwell | russellb: sucks |
01:21.26 | russellb | Qwell: i'm pretty close, though .. |
01:21.28 | Qwell | cool |
01:21.40 | Qwell | commit it! "close" is close enough ;) |
01:21.53 | russellb | but the default paths lost their prefixes ... so running asterisk -r complains that there is no file /run/asterisk.pid |
01:21.58 | Qwell | ahh |
01:21.59 | SplasPood | Iamtheman222: I think people are gonna need a lil more info.. |
01:22.05 | Qwell | that could be an issue |
01:22.05 | Iamtheman222 | when I try to use the callback, it seems to look for callback.config file |
01:22.05 | russellb | Qwell: that's kinda bad :) |
01:22.24 | russellb | Iamtheman222: there is no such thing as callback.config in asterisk |
01:22.27 | SplasPood | Iamtheman222: what is 'the callback' ? |
01:22.56 | Iamtheman222 | callback function that I have added in the extensions.conf to call me back |
01:23.05 | russellb | and perhaps you should say hi first :) IRC is a social place |
01:23.22 | Iamtheman222 | Hi |
01:23.27 | russellb | yay |
01:23.40 | russellb | Iamtheman222: pastebin your extensions.conf entry ... www.pastebin.ca |
01:24.34 | Iamtheman222 | http://pastebin.ca/67014 |
01:25.06 | Iamtheman222 | the callback function has been found in freepbx... |
01:25.19 | Qwell | ~freepbx |
01:25.23 | jbot | methinks freepbx is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
01:25.24 | paolob | russellb, There is something strange with my problem with playing sound file: the sound file is looked for in the sounds directory on incoming calls, while it is looked for in sounds/es if I call from an extension. Why? |
01:25.54 | Iamtheman222 | I know but what's the problem, it's not about freepbx |
01:26.10 | russellb | paolob: because the language isn't getting set on the channel for incoming calls |
01:26.12 | Qwell | <russellb> paolob: try adding Set(LANGUAGE()=es) before playing the file |
01:26.46 | paolob | russellb, isn't there a way to get it set for incoming calls? |
01:26.53 | *** part/#asterisk Iamtheman222 (n=fddfffsa@modemcable237.246-83-70.mc.videotron.ca) |
01:27.02 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
01:27.02 | russellb | geez, impatient |
01:27.15 | russellb | paolob: yeah, you just have to get it into the right entry of your config |
01:27.16 | Qwell | russellb: he's been doing that all day |
01:27.18 | SplasPood | paolob: your sip peer in sip.conf, that the call comes in on |
01:27.22 | SplasPood | paolob: or iax.. |
01:27.37 | paolob | SplasPood, iax...? |
01:27.46 | SplasPood | how do your inbound calls come in |
01:27.51 | SplasPood | or zap for that matter.. |
01:27.58 | russellb | or ... mgcp |
01:28.04 | SplasPood | yea, but who uses that :P |
01:28.05 | Qwell | or ski..wait, nm |
01:28.06 | *** join/#asterisk tlowe_ (n=tlowe@omfg.wtf.no) |
01:28.11 | russellb | Qwell: ;) |
01:28.19 | Qwell | actually though.. |
01:28.35 | Qwell | I have briefly thought about making chan_skinny a client implementation too :) |
01:28.40 | paolob | SplasPood, russellb, but in sip.conf I have set language=es , the comment says "Default language setting for all users/peers". Why doesn't it work for incoming calls? |
01:28.51 | Qwell | russellb: How incredibly pointless would that be? |
01:28.51 | russellb | Qwell: people would use it, i'm sure |
01:28.53 | SplasPood | paolob: hrm, good point |
01:29.07 | russellb | Qwell: what about for extra features for an existing call manager install? |
01:29.20 | Qwell | russellb: yeah, perhaps |
01:29.31 | russellb | paolob: perhaps it's a bug |
01:29.35 | russellb | paolob: is this 1.2 ? |
01:29.51 | russellb | and are you using allowguest=yes ? |
01:29.52 | paolob | russellb, 1.2.7.1 |
01:30.16 | paolob | russellb, it's commented out |
01:30.19 | russellb | Qwell: i think it would be cool. lots of people have asked for the client implementation of mgcp |
01:30.35 | Qwell | it's fairly easy to do, I think |
01:30.42 | Qwell | we already know all the messages |
01:31.10 | SplasPood | paolob: uncomment it? |
01:31.31 | paolob | SplasPood, what does it makes precisely? |
01:31.46 | russellb | no, that probably won't help |
01:31.52 | russellb | i was just trying to think places there could be a bug |
01:32.55 | JoseBravo | I have lost all my day triying to call from ext to ext with astbill. But I get this f....k error, Jun 19 05:32:41 WARNING[30085]: app_dial.c:803 dial_exec_full: Dial requires an argument (technology/number). Plese if anoye know why is that, I'll be very thanks. |
01:33.07 | russellb | paolob: is this an installation from source? i have a patch for you, i think ... |
01:33.20 | paolob | russellb, no, it's the debian package |
01:33.35 | paolob | russellb, will the patch enter the debian archive? |
01:34.03 | russellb | paolob: eventually, yes, if it fixes your problem |
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01:34.20 | russellb | paolob: if you want to download source, then i have a patch you can try ... |
01:34.36 | paolob | JoseBravo, You must put dial(SIP/resource,60,Tt), for example |
01:34.47 | russellb | paolob: if you have svn installed .... svn co http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
01:34.48 | Qwell | DO NOT USE tT! |
01:34.50 | Qwell | ever |
01:34.58 | Qwell | well...mostly |
01:35.00 | russellb | Qwell: oh shush |
01:35.10 | paolob | russellb, no, thanks, I'm going to put the set language in the incoming extension |
01:35.12 | Qwell | russellb: make me :p |
01:35.19 | russellb | paolob: alright |
01:35.26 | Qwell | wait, no, forget I said that |
01:35.34 | *** mode/#asterisk [+b %Qwell!*@*] by russellb |
01:35.39 | russellb | pwned |
01:35.46 | Strom_C | ha |
01:35.54 | *** mode/#asterisk [-b %Qwell!*@*] by russellb |
01:35.55 | Qwell | :D! |
01:36.50 | JoseBravo | paolob exctly? |
01:36.56 | *** part/#asterisk tlowe_ (n=tlowe@omfg.wtf.no) |
01:37.59 | paolob | JoseBravo, I put an example, see http://www.asteriskguru.com/tutorials/dial.html |
01:38.01 | russellb | paolob: well i'm pretty sure i found the bug, and i'm committing the fix ... just fyi |
01:38.25 | paolob | russellb, ok, thank you, what version will it be fixed? |
01:38.34 | russellb | paolob: 1.2.10 |
01:38.53 | paolob | russellb, ok, you're fantastic!!!! |
01:38.58 | russellb | thanks :) |
01:39.16 | Qwell | russellb rocks :D |
01:39.46 | paolob | russellb, (excuse me) when will 1.2.10 be released? |
01:39.48 | [TK]D-Fender | Oh God.. another release within a week? |
01:39.52 | Strom_C | oh, speaking of which, 1.4 is in beta now, right? |
01:40.05 | russellb | paolob: sooner if you test my fix :D |
01:40.13 | [TK]D-Fender | Strom_C :Yup, and due to be released with Windows Vista I hear ;) |
01:40.14 | paolob | russellb, :-( |
01:40.17 | russellb | Strom_C: not yet |
01:40.34 | paolob | guys, and what about stun support in asterisk? is there anyone working on it? |
01:40.37 | *** join/#asterisk DMark (n=kk7cu@pool-70-105-217-25.scr.east.verizon.net) |
01:40.42 | russellb | paolob: it's in trunk already |
01:41.02 | russellb | (to be 1.4) |
01:41.15 | paolob | russellb, "trunk" (I'm not english speaking) what is it? |
01:41.22 | JoseBravo | paolob it was a account addedwith AstBill |
01:41.22 | [TK]D-Fender | russellb : A lot of remarkable stuff going on for 1.4... |
01:41.25 | Strom_C | will chan_skinny be in 1.4? (I assume the answer is no) |
01:41.32 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
01:41.33 | russellb | paolob: the development branch of code that will be released as asterisk 1.4 in the near future |
01:41.45 | [TK]D-Fender | russellb : Whats the status of SIP-B Shareld Line Appearances? |
01:41.46 | russellb | [TK]D-Fender: we hope so :) |
01:41.56 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
01:42.11 | russellb | [TK]D-Fender: that was something that was talked about a lot at our developers conference ... no code in trunk yet, i don't think |
01:42.25 | russellb | it's pretty complicated to do |
01:42.26 | gambolputty | Hi. Can * detect if an incoming call is already muted? |
01:42.27 | [TK]D-Fender | russellb : DAMN... thats a huge selling point... |
01:42.44 | Qwell | Strom_C: the answer is yes, actually |
01:42.50 | Strom_C | Qwell, oh awesome! |
01:42.54 | russellb | Strom_C: it went in today, because Qwell r0x0rz |
01:42.56 | Qwell | Strom_C: which is why I committed the branch today :) |
01:43.20 | russellb | ok, i better log off of IRC before people find more bugs |
01:43.21 | Strom_C | double awesome on a stick with sugar and caramel and pralines and holy shit now I want ice cream |
01:43.33 | Strom_C | is there a good ice cream place in huntsville? :/ |
01:43.43 | Qwell | Strom_C: are you in huntsville? |
01:43.46 | Strom_C | yep |
01:43.52 | Qwell | fun |
01:44.03 | Strom_C | I go back to los angeles on tuesday |
01:44.29 | Strom_C | Qwell, you can do shared line appearances with skinny, cant you |
01:45.34 | Qwell | nope..not yet |
01:45.45 | Strom_C | ah ok |
01:46.56 | Qwell | Strom_C: easy |
01:47.04 | Qwell | asterisk should automatically create your dialplan for you |
01:48.02 | Strom_C | will that be res_bippity_boppity_boo.so? |
01:56.06 | littleball | hello, i want to enable "invite" from any sip client without authorization. How to? |
02:03.47 | *** join/#asterisk pbd (n=pbd@c-67-163-20-134.hsd1.il.comcast.net) |
02:04.14 | pbd | Evening, all. |
02:04.58 | *** join/#asterisk dlynes_home (i=1000@S0106001217014b92.vc.shawcable.net) |
02:05.14 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
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02:12.41 | *** join/#asterisk rbd (n=rbd@cpe-066-057-011-095.nc.res.rr.com) |
02:13.13 | rbd | hey guys, anyone using asterisk on debian with debian's asterisk packages? any idea if ztdummy comes precompiled anywhere, or if I have to compile that from source? |
02:13.47 | Qwell | rbd: It's best not to use the packages.. |
02:14.57 | rbd | Qwell: seems like it, I'll grab the source |
02:15.28 | Strom_C | hmm, is there a #asterisk FAQ page? |
02:15.38 | neilbags-work | Qwell: why is it best not to use packages? are the packages broken? |
02:15.43 | pbd | ~wiki |
02:15.46 | Qwell | neilbags: usually, yes |
02:16.13 | neilbags-work | Qwell: so who is the maintainer? is he/she doing a poor job? |
02:16.15 | pbd | They're usually old, if nothing else. |
02:16.20 | neilbags-work | ok |
02:16.26 | Qwell | neilbags-work: there is at least one maintainer for each distro |
02:16.46 | Qwell | and they all have wildly different ways of packaging it |
02:16.57 | neilbags-work | Qwell: well i use gentoo and the ebuilds work great and they are totally up to date, i guess other distros lag |
02:17.22 | Qwell | !!! All ebuilds that could satisfy "asterisk" have been masked. |
02:17.24 | Qwell | sorry, no |
02:17.55 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
02:19.20 | neilbags-work | Qwell: well i mean the ones in ~x86 work great, and are up to date. 1.0.11_p1 is the latest unmasked version on x86, are you using a differerent arch? |
02:19.45 | Qwell | 1.0.11.1 is old |
02:19.51 | Qwell | I mean...it's new...but it's old code |
02:20.33 | pbd | A 1.0 release? Ick. Current stable is 1.2. A loooong road in between. |
02:20.45 | neilbags-work | Qwell: i know, but the 1.2.9.1 ebuild works fine ... there would be no reason not to use the ebuild ... |
02:20.56 | Qwell | except that asterisk packages suck :) |
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02:22.27 | neilbags-work | Qwell: its bad form to compile it from source yourself, i write my own ebuild for things that don't exist, if they suck then they should be fixed, not avoided |
02:22.46 | neilbags-work | Qwell: i havn't come across any 'sucking' yet so i havn't had to make any changes to the ebuilds |
02:22.54 | littleball | hello, i want to enable "invite" from any sip client without authorization. How to? |
02:22.56 | Qwell | it's not really possible to make good asterisk packages. |
02:23.15 | Qwell | it makes it incredibly difficult, by design |
02:23.22 | littleRalphie | looking for a quick tip |
02:23.32 | neilbags-work | Qwell: how come? what about just a basic-ebild that does ./configure && make && make install? |
02:23.57 | Strom_C | littleRalphie, just ask your question |
02:24.02 | Qwell | because it's far more complex than that |
02:24.08 | neilbags-work | Qwell: surely its better to use the ebuild even if only to track what files have been installed! |
02:24.10 | Qwell | especially now, with menuselect |
02:24.22 | littleRalphie | I added gnome to my trixbox 1 system, but I dont get how to start the gui once booted |
02:24.29 | Qwell | ~trixbox |
02:24.53 | Strom_C | #freepbx |
02:24.58 | Strom_C | ~freepbx |
02:25.00 | jbot | i guess freepbx is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
02:25.08 | neilbags-work | Qwell: why not have an ebuild call menuselect? |
02:25.18 | Qwell | ~trixbox |
02:25.19 | jbot | hmm... trixbox is NOT supported here! People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP) |
02:25.33 | Qwell | neilbags-work: which would do what? |
02:26.02 | neilbags-work | Qwell: huh? i don't quite understand what you're asking |
02:29.41 | neilbags-work | i'm interested to hear any good reasons why using the gentoo ebuilds is bad compared to compiling manually ... i'd be really surprised if any reasons warranted not modding the ebuild. |
02:30.05 | drray | why not just use the svn? |
02:30.14 | drray | what do you gain? |
02:30.17 | Qwell | neilbags-work: because it would be impossible to get every single option in make menuselect, as a USE flag |
02:31.21 | JoseBravo | Why im getting this error, ast_channel_make_compatible: No path to translate from SIP/dominet-9b29(256) to SIP/70103-ed68(4)? |
02:31.28 | JoseBravo | Triying to do a out call |
02:31.31 | neilbags-work | Qwell: ok. but if menuselect is so important, why not let the ebuild run menuselect? at least then you can keep track of what files asterisk is installing for ease of removal/upgrades |
02:31.48 | Qwell | neilbags-work: because no other ebuilds are interactive |
02:31.50 | file | JoseBravo: one side is G729, one side is ULAW, and your Asterisk can't transcode between them |
02:32.24 | Qwell | file: You aren't allowed to remember codec numbers |
02:32.35 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
02:32.36 | file | :( |
02:32.37 | Qwell | only 4 and 8..no more |
02:33.04 | neilbags-work | Qwell: yes valid, but but even if asterisk is an exception, why not make that exception? if the gentoo people don't like it then keep it as an 'unofficial' ebuild in an asterisk overlay. |
02:33.21 | Qwell | because then it becomes an unofficial ebuild |
02:33.36 | SplasPood | muahahhaa... I can now have queue agents on cell phones |
02:33.36 | neilbags-work | Qwell: i still don't think that arguement warrants dodging portage to install asterisk |
02:33.37 | Qwell | which, NOBODY will use |
02:35.16 | *** join/#asterisk RF_MIA (n=Administ@69-172-194-16.miamfl.adelphia.net) |
02:35.48 | neilbags-work | Qwell: i would ... wouldn't you? ... if menuselect is preventing you from using portage isn't it better to have an ebuild that is a little non-standard (not like its going to break anything) than doing it manually from source? |
02:36.10 | Qwell | neilbags-work: no |
02:36.11 | neilbags-work | Qwell: gentoo is designed to be all-encompassing ... if asterisk needs an interactive build then it should have it |
02:36.16 | SplasPood | mailbags: asterisk isn't all that messy with it's files |
02:36.18 | drray | again, what is wrong with svn? |
02:36.26 | SplasPood | and I think just no one cares enough about gentoo to .. care |
02:36.32 | Qwell | neilbags-work: Then why doesn't gentoo let you compile your kernel, from just emerge? :) |
02:37.10 | pbd | (while being supportive of the 'compile from source' bunch- I have to say that, adding svn to a machine is a PITA, with dependancies and, if left alone, mangled httpd.conf when complete) |
02:37.21 | neilbags-work | drray: when you compile a kernel it doesn't scatter its files through /usr and become a mess in the future |
02:37.38 | neilbags-work | sorry, i meant Qwell not drray |
02:37.57 | Qwell | neither does asterisk |
02:38.36 | neilbags-work | yes it does, it puts plenty in /usr/lib and /usr/share |
02:38.57 | drray | so you've been using asterisk how long? |
02:39.31 | neilbags-work | Qwell: i'm not saying the asterisk build is bad, its really clean ... but i still don't see how you justify dodging portage |
02:40.42 | SplasPood | pbd: wtf are you smoking? |
02:40.51 | JoseBravo | file Thanks |
02:41.22 | pbd | Not to be purely argumentative here, but why *should* the asterisk-dev community support any particular distro over another? |
02:41.24 | SplasPood | neilbags-work: I justified it... no one cares enough to deal with it... You could... |
02:42.01 | pbd | Splas- I've installed svn on several different distros, versions, kernels, and machines- and it was never 'smooth'. |
02:42.23 | SplasPood | pbd: Odd, I've installed it everywhere and its been perfectly smooth.. which distros |
02:42.34 | neilbags-work | SplasPood: i guess i'm more careful than most with my gentoo boxes ... i won't put *anything* in /usr unless portage installs it |
02:42.37 | SplasPood | pbd: as a client, I mean |
02:42.50 | pbd | Mosty slack and suse- also some FC. |
02:42.54 | SplasPood | neilbags-work: I don't run gentoo |
02:43.11 | SplasPood | pbd: heh 3 distributions I haven't run in quite some time |
02:43.12 | pbd | Only as a client- haven't delved into using it as a repository for my own work yet. |
02:43.22 | SplasPood | pbd: I use it for both, on debian |
02:43.28 | SplasPood | as a client on macosx (darwin) |
02:43.30 | pbd | Splas: Hence the difficulty. :) |
02:43.33 | SplasPood | debian |
02:43.34 | SplasPood | centos |
02:43.39 | neilbags-work | SplasPood: 'i justified it' ? this implied you were using gentoo |
02:43.48 | littleRalphie | ok, well trixbox seems cool, but is there a better approach to speed up my learning curve? |
02:43.56 | SplasPood | neilbags-work: No I justified why there's no working ebuild or whatever.. by saying "no one cares" |
02:44.37 | SplasPood | I think the problem is that an asterisk compile doesn't A) take long enough or B) have enough ansi eye candy to warrant it's own ebuild |
02:44.54 | drray | heh |
02:44.56 | neilbags-work | SplasPood: there is a working ebuild, but it doesn't run menuselect for you to fiddle with ... i'm just wondering whether its worth having an ebuild that does run it |
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02:45.15 | SplasPood | portage is only for 24+hr seizure inducing ansi nightmares |
02:45.29 | SplasPood | menuselect? |
02:45.53 | drray | would portage take longer than SVN asterisk? |
02:46.08 | neilbags-work | drray: define 'take longer' |
02:46.16 | SplasPood | drray: depends on how much "optimization" you want and how "k-rad" you wish to feel |
02:46.27 | SplasPood | www.funroll-loops.org |
02:46.38 | neilbags-work | SplasPood: Qwell said he doesn't use ebuilds because of menuselect |
02:46.57 | Qwell | no, that is one reason |
02:47.03 | Qwell | I have many other reasons...such as being a developer |
02:47.22 | neilbags-work | Qwell: well that makes sense for sure, i'm talking about production use |
02:47.34 | SplasPood | Chief among them is not caring, I bet |
02:47.43 | SplasPood | PRODUCTION MUST HAVE ANSI |
02:47.59 | Qwell | I use svn trunk in prod |
02:48.07 | Qwell | That's just how I roll |
02:48.24 | SplasPood | word. |
02:48.35 | neilbags-work | do you remove prior versions before an update or install over the top? |
02:48.42 | drray | over |
02:48.49 | pbd | Qwell likes to live on the edge. :) |
02:48.55 | SplasPood | if you keep copying binaries over one another there's buildup |
02:48.57 | SplasPood | can't have that |
02:49.36 | Qwell | neilbags-work: I just make install, and it works |
02:49.44 | Qwell | perhaps I just know what I'm doing |
02:49.48 | SplasPood | if asterisk wasn't such a moving target i'd be all for proper binary packages |
02:50.15 | neilbags-work | SplasPood: gentoo doesn't use binary packages |
02:50.25 | pbd | If asterisk wasn't such a moving target- it would be like thousands of other OSS projects out there. |
02:50.26 | neilbags-work | Qwell: well you know the source and build better than me |
02:50.27 | rbd | guys, I try to appexec festival, and I get the voice but it's very choppy |
02:50.30 | SplasPood | neilbags-work: I know, hence my use of the word 'proper' |
02:50.50 | SplasPood | rbd: you wan cepstral |
02:50.54 | SplasPood | s/wan/want |
02:54.10 | neilbags-work | Qwell: askerisk is just one piece of software i maintain on gentoo boxes i can't know every file that every package installs (thats what portage is for!) so not being an asterisk developer, there is no reason for me to treat asterisk any different. if i need to write an ebuild that checks out svn and runs menuselect then so be it |
02:55.33 | SplasPood | neilbags-work: you're probably better off with asterisk in a box anyway |
02:56.02 | *** part/#asterisk RF_MIA (n=Administ@69-172-194-16.miamfl.adelphia.net) |
02:56.06 | rbd | SplasPood: cepstral? googled it..looks interesting, looking for a trial/free link |
02:56.47 | Corydon76-home | You can download any Cepstral voice for free |
02:57.02 | SplasPood | rbd: you can use it as a demo, but it adds "blah blah, this isn't paid for" to your stuff |
02:57.05 | Corydon76-home | The only thing is, all phrases are prefixed with a demo phrase |
02:57.27 | *** join/#asterisk Cerlyn (i=ALEIN@pdpc/supporter/sustaining/Cerlyn) |
02:57.28 | SplasPood | its reasonably priced tho |
02:57.41 | Corydon76-home | Can't argue too much with $30/voice |
02:57.42 | neilbags-work | SplasPood: well i'm happy with the way things are going now, i just didn't know about the ebuilds 'sucking'. i don't have dedicated boxes for asterisk anyway. |
02:57.42 | SplasPood | I wonder how the price drops off when you get into volume, I didn't look |
02:57.56 | Corydon76-home | No, the price is actually cheaper one at a time |
02:57.57 | SplasPood | neilbags-work: who said they suck? |
02:58.03 | neilbags-work | Qwell |
02:58.23 | Corydon76-home | I truly love domain names for $9.99/year, but 19.99/2years |
02:58.23 | SplasPood | Corydon76-home: I saw something about buying "ports" ... seemed to be simultaneous use stuff |
02:58.39 | Corydon76-home | Don't they realize that the price for multiple years is more expensive? |
02:58.58 | *** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
02:59.02 | Corydon76-home | SplasPood: the same is true for Cepstral, but it's only pennies |
02:59.08 | *** join/#asterisk Strom_C (n=strom@m615e36d0.tmodns.net) |
02:59.14 | Strom_C | no no, the answer is "cake" |
03:00.19 | h0 | if i have a TDM400P then i can use any linux network interface card to conect to the asterisk server with a softclient corect? |
03:00.21 | drray | the answer is always cake |
03:00.30 | Strom_C | h0, yes |
03:00.34 | h0 | k thanx |
03:00.36 | drray | you need to rethink the question if that does not work for you |
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03:06.16 | *** part/#asterisk pbd (n=pbd@c-67-163-20-134.hsd1.il.comcast.net) |
03:07.22 | *** join/#asterisk erwinism (i=erwin@61.9.118.37) |
03:07.29 | erwinism | hello |
03:08.19 | erwinism | i have asterisk server, one x100p single FXO card, can anyone help me to make my asterisk work. :) |
03:09.03 | Strom_C | erwinism, sure. I can do everything for you at my standard hourly rate :) |
03:10.03 | erwinism | Strom_C, does that mean its hard to configure this thing? |
03:10.08 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
03:10.22 | Strom_C | no, it means that you are encouraged to read the documentation :) |
03:10.29 | SplasPood | erwinism's question is topic worthy |
03:11.01 | Corydon76-home | The difficulty of configuring Asterisk increases only with the scale of complexity of what you want it to do |
03:11.11 | file | and the IQ requirement increases too |
03:11.22 | Strom_C | so if you just want it to route calls, easy. If you want it to bake cake, tough. |
03:11.33 | file | res_oven! |
03:11.34 | erwinism | haha, ok, im not a tech guy. so maybe i need to read more |
03:12.17 | Strom_C | erwinism, if you have specific questions, we're more than happy to answer them |
03:12.19 | erwinism | i just want to make a simple pbx at my office using PC, and be able to accept calls from PSTN |
03:12.26 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
03:12.43 | SplasPood | erwinism: I encourage you to checkout trixbox or one of the other asteriskInABox type things |
03:12.43 | Strom_C | but I doubt you will find anyone here willing to regurgitate the entire asterisk book from memory :) |
03:12.47 | erwinism | Strom_C, ok, sorry :) i have to read the manual :) |
03:12.55 | Strom_C | and I encourage you not to do that |
03:13.04 | Strom_C | GUIs are the devil |
03:13.05 | SplasPood | Strom_C: Why not? |
03:13.07 | SplasPood | true |
03:13.10 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
03:13.11 | file | I will do it, but I will say exactly what happens in the core |
03:13.14 | Strom_C | especially for something like asterisk |
03:13.17 | file | right down to the concepts and API calls |
03:13.33 | SplasPood | Strom_C: Seems to be exactly what this person wants tho.. |
03:13.50 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
03:13.53 | Strom_C | SplasPood, yes, but the problem is that he'll learn to do things the wrong way |
03:13.59 | erwinism | SplasPood, ok i will search about trixbox |
03:14.02 | vivek | hello all |
03:14.13 | Strom_C | damnit, there goes another one |
03:14.16 | Strom_C | hello vivek |
03:14.27 | erwinism | hahaha |
03:14.28 | Strom_C | erwinism, i would advise you not to waste your time with trixbox |
03:14.44 | SplasPood | Strom_C: Sorry |
03:14.45 | vivek | a little off topic. Can i transfer a call from pstn to sip numbers using a sipura or some other sip hardware ? |
03:14.51 | erwinism | ok, im reading the version2 of the asterisk handbook |
03:14.56 | SplasPood | erwinism: trixbox won't teach you anything.. but at first it'll be "easy" |
03:15.01 | file | ~thebook |
03:15.08 | Strom_C | ~book |
03:15.11 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
03:15.11 | file | darn it I do that every time |
03:15.32 | pjchilds | ~just-buy-a-key-system |
03:15.42 | pjchilds | hmm.... :) |
03:15.48 | Strom_C | pjchilds, ooh, Western electric 1A2? :) |
03:16.17 | Strom_C | file, 5ESS or 5ESS-2000? |
03:16.29 | file | just 5ESS |
03:16.33 | Strom_C | ah ok |
03:16.42 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
03:17.20 | file | Cerlyn: 'tis a telco switch |
03:17.35 | pjchilds | or the 'VCDX' the (very compact digital exchange...) |
03:17.50 | Cerlyn | You mean they don't just fill it up with a bunch of SPA-2000's? :) |
03:17.53 | pjchilds | minimum footprint... three cabinets 6 feet heigh... |
03:18.16 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
03:18.31 | Strom_C | feh. DMS-10 for the win |
03:18.47 | file | Strom_C: can you only do centrex off a DMS100? |
03:19.00 | Strom_C | file, I think DMT can do centrex as well |
03:19.17 | file | I've been trying to figure out what my telco uses |
03:19.33 | file | well, not really trying... just randomly wondering |
03:19.35 | Strom_C | file, PM me your NPA/NXX |
03:19.53 | file | I'm Canadian don't forget :D |
03:20.28 | Strom_C | claims DMS-100 |
03:20.38 | Strom_C | Alliant Telecom |
03:20.41 | file | thought so |
03:22.55 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
03:23.25 | Strom_C | droooooops! |
03:28.50 | littleRalphie | questin about sipp_additional.conf? |
03:29.04 | *** part/#asterisk justdave (n=dave@unaffiliated/justdave) |
03:30.03 | littleRalphie | I can connect to telregister to telasip.com with a sip client but not in asterisk |
03:30.21 | rene- | ~amp |
03:30.22 | jbot | it has been said that amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
03:30.53 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
03:30.53 | *** mode/#asterisk [+o russellb] by ChanServ |
03:32.01 | rene- | littleRalphie: you could try to see what errors are you getting by issuing a sip reload from your asterisk CLI (asterisk -r and then sip reload) |
03:32.17 | erwinism | SplasPood, you there? |
03:34.04 | littleRalphie | just reattemmpts, is this loggd somewhere? |
03:35.45 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
03:36.09 | littleRalphie | REGISTER attempt 81 to rshumway@gw4.telasip.com |
03:36.09 | littleRalphie | REGISTER attempt 81 to ralphshumway@yahoo.com@sip.stanaphone.com |
03:36.10 | littleRalphie | asterisk1*CLI> |
03:36.32 | littleRalphie | this is alli see |
03:36.57 | TheCops | Someone have 1.6.6 firmware of polycom SIP phone?! |
03:37.07 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
03:37.53 | rene- | before that you can enable a sip trace with sip debug on, and that will give you the error you are getting |
03:39.02 | rene- | is stanaphone working for you? |
03:39.08 | littleRalphie | nope |
03:39.14 | rene- | telasip? |
03:39.18 | littleRalphie | debug on now |
03:39.22 | littleRalphie | neither |
03:40.15 | litage | is it possible to limit the size of an extension's voicemail box? |
03:40.26 | rene- | but you are using AMP/FreePBX/ or something like that right? |
03:40.42 | sevard | as far as i know, no.. but you can limit the max amount of messages in said box |
03:42.00 | rene- | littleRalphie: go here for an example: http://forum.stanaphone.com/viewtopic.php?t=2342&highlight=asterisk |
03:42.52 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
03:43.07 | rene- | you should be able to find sample configurations for your other voip provider unless it is very new or very little known |
03:43.27 | rene- | ~google |
03:43.28 | jbot | i heard google is a search engine found at http://www.google.com/ |
03:44.07 | littleRalphie | I followed mr nerdvittles docs, but perhaps I missed somthing |
03:44.43 | rene- | some of the users posted a working configuration |
03:45.01 | rene- | try to match that to your specific acct details |
03:45.31 | SplasPood | erwinism: yes |
03:47.11 | *** join/#asterisk dlynes_home (i=1000@S0106001217014b92.vc.shawcable.net) |
03:49.06 | erwinism | SplasPood, i installed trixbox :) |
03:49.31 | sevard | sad. |
03:51.06 | erwinism | i have to make everythings working before wondering how these things works hehe |
03:52.12 | sevard | that's true, i started with AMP and after I had a vague understanding deleted it |
03:58.32 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
03:58.41 | CrashHD | how can I use a gotoif to tell if a file exists? |
03:59.15 | *** part/#asterisk rene- (n=rene@201.152.34.100) |
03:59.18 | CrashHD | STAT doesn't seem to be registered |
04:01.14 | CrashHD | how is EXISTS used? |
04:03.20 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
04:12.34 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
04:15.42 | Corydon76-home | STAT is only in trunk |
04:15.52 | CrashHD | ahh |
04:16.08 | Qwell | heh, I used a little hack in 1.2 |
04:16.30 | Qwell | System(ls filename) with SYSTEMSTATUS |
04:16.36 | russellb | you dialplan h4xor |
04:16.43 | Qwell | :D |
04:16.53 | russellb | hey, that's a pretty cool way to do it |
04:17.11 | Qwell | I like func_backtick... |
04:17.16 | Qwell | haven't used it, but I like the idea |
04:17.34 | russellb | what's the idea |
04:17.55 | Qwell | backtick.. like running a subcommand from a shell |
04:18.06 | Qwell | ls -l /lib/modules/`uname -r`/ |
04:18.10 | Qwell | so, like |
04:18.11 | Corydon76-home | Isn't that what System does? |
04:18.18 | Qwell | Corydon76-home: it doesn't return a value though. |
04:18.30 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
04:18.32 | russellb | ok, yeah, i gotcha |
04:18.33 | Corydon76-home | Ah... |
04:18.41 | Qwell | exten => s,1,System(ls -l ${BACKTICK(uname -r)}) |
04:18.45 | Qwell | or something |
04:19.01 | russellb | heh, would be fairly straight forward to write |
04:19.13 | littleRalphie | I currently run a sunrocket gizmo direc to my cable modem, culd that be my issue since the device uses port 5060? |
04:19.26 | Qwell | probably |
04:19.29 | Corydon76-home | You could have done ${CURL(http://localhost/exists.cgi?${filename})} |
04:19.51 | Qwell | Corydon76-home: heh, hack |
04:19.55 | Qwell | I like it :D |
04:21.15 | russellb | i think the MacroExclusive app is a cool idea |
04:21.21 | Qwell | huh? |
04:21.27 | Qwell | I think I saw that... on -dev? |
04:21.28 | russellb | it was something discussed on the dev list |
04:21.34 | russellb | but it's on mantis now, i think |
04:21.37 | russellb | haven't looked at the code, yet |
04:22.18 | Corydon76-home | You could do the same thing with Groups |
04:22.43 | Corydon76-home | nonblocking, though |
04:23.12 | Corydon76-home | That's always nice |
04:30.19 | *** join/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au) |
04:31.04 | P-NuT | Hi all, has anybody in australia got an x100p card to sell me? |
04:31.49 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
04:36.49 | *** part/#asterisk Cerlyn (i=ALEIN@pdpc/supporter/sustaining/Cerlyn) |
04:49.57 | *** join/#asterisk Strom_C (n=strom@m615e36d0.tmodns.net) |
04:50.23 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
04:52.43 | Qwell | brimstone: y0 |
04:53.05 | Strom_C | brimstone! |
04:57.13 | *** join/#asterisk janeNarak (i=chonlada@jane.lru.ac.th) |
04:57.48 | janeNarak | hi all, Asterisk are support Skype? |
05:00.35 | Strom_C | not natively |
05:04.58 | CrashHD | how can I stop the dial function from bridging the call (I'm trying to use the M option but there is about a second where the channels are bridged)???? |
05:05.12 | CrashHD | before the macro starts |
05:06.09 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
05:10.06 | erwinism | in order to accept calls from pstn and forward to extention, what configuration should i edit? i have x100p already installed |
05:11.01 | vivek | janeNarak: do you have some skype hardware ? if you don't you can do without skype ... |
05:13.49 | *** join/#asterisk eset (n=eset@203-114-177-203.dsl.sta.inspire.net.nz) |
05:14.00 | eset | does speex work in a conf room w asterisk? |
05:14.38 | Qwell | yes |
05:15.03 | sevard | it ought to |
05:15.54 | Strom_C | wheeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee |
05:17.07 | sevard | happy drugs? |
05:17.25 | Strom_C | even better |
05:17.30 | Strom_C | ive got a date on friday |
05:17.38 | sevard | mailman? |
05:17.39 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
05:18.09 | Strom_C | what? |
05:18.20 | sevard | did you get a date with the mailman? |
05:18.25 | Strom_C | no |
05:18.41 | sevard | cop? |
05:18.44 | Strom_C | no |
05:18.52 | sevard | anyone from the village people |
05:19.01 | Strom_C | let me check |
05:19.02 | Strom_C | no |
05:19.05 | sevard | wtf |
05:19.21 | sevard | I thought you liked uniforms. |
05:19.59 | Strom_C | only if the uniform has a telephone company logo on it |
05:21.30 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
05:22.03 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
05:25.12 | *** part/#asterisk neilbags-work (n=neilbags@149.171.94.134) |
05:28.33 | *** join/#asterisk Eggplant (n=none@dsl-72-19-42-179.cascadeaccess.com) |
05:33.14 | erwinism | Strom_C, i only have x100p card, what should i name it for outgoing channel? |
05:33.34 | Strom_C | ZAP/1 |
05:33.53 | erwinism | ok i will try that one ;) |
05:34.01 | sevard | "Sandy" |
05:34.04 | sevard | or HAL |
05:35.28 | Strom_C | dave |
05:35.32 | Strom_C | i'm sorrry dave |
05:35.37 | Strom_C | your call did not go through |
05:36.08 | sevard | hahaha |
05:36.21 | sevard | daisy daisy, your sip client won't register |
05:36.30 | sevard | k mine wasn't as good. |
05:36.47 | Strom_C | seriously, how cool would it be to get Douglas Rain to do voice prompts for asterisk |
05:36.58 | sevard | who the hell is that |
05:37.14 | Strom_C | the guy who did the voiceover work for HAL9000 |
05:37.15 | Strom_C | duh |
05:37.21 | sevard | jesus like i'm fucking imdb |
05:37.38 | sevard | but no, that wouldn't be cool. that would be scary. |
05:37.46 | sevard | there's a reason why we have cheerful women doing our prompts. |
05:38.15 | erwinism | Strom_C, i already named it ZAP/1 when i tried calling 9 plus 2971274 it tells me "busy" ...my pstn is not busy, maybe i missed something |
05:38.34 | sevard | it's your pstn? |
05:38.35 | Strom_C | erwinism, are you using asterisk or are you using trixbox? |
05:39.09 | erwinism | Strom_C, i installed trixbox, but i configured it manually |
05:39.50 | Strom_C | erwinism, pastebin your extensions.conf, zaptel.conf, and zapata.conf files |
05:39.53 | Strom_C | ~pb |
05:39.55 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
05:40.04 | erwinism | Strom_C, ok :) |
05:41.47 | erwinism | Strom_C, brb, boss is calling me hehe |
05:42.03 | *** join/#asterisk neilbags-work (n=neilbags@149.171.94.134) |
05:42.42 | CrashHD | I can't seem to get mp3's to play for music on hold, anything special I need to do? |
05:42.53 | Strom_C | did you install mpg123 |
05:42.55 | CrashHD | (I just uploaded one of my mp3's I usually listen to for testing) |
05:42.58 | CrashHD | no I did not |
05:42.58 | *** join/#asterisk znoG (n=gs@205-17-235-201.fibertel.com.ar) |
05:42.59 | Qwell | or asterisk-addons |
05:42.59 | CrashHD | running 1.2 |
05:43.05 | CrashHD | no asterisk-addons either |
05:43.11 | Qwell | well... |
05:44.32 | Strom_C | this is kind of like asking "why won't my car start?" when you have never purchased gasoline for it |
05:44.39 | CrashHD | lol |
05:45.00 | CrashHD | if I knew a car ran on gasoline that would make sense, but if I didn't...I would still be where I'm at now... |
05:45.19 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
05:46.47 | CrashHD | mod=quitemp3 |
05:46.58 | CrashHD | this will not work by default |
05:47.02 | CrashHD | I'm guessing |
05:47.03 | CrashHD | ? |
05:47.26 | Strom_C | well, especially if you misspell mode like that |
05:47.35 | CrashHD | *nods* |
05:50.21 | Qwell | You mean it doesn't just know what you want? |
05:50.46 | Strom_C | maybe he forgot to compile res_esp.so |
05:50.54 | Qwell | must be it |
05:55.20 | Strom_C | OOOH |
05:55.28 | Strom_C | There's Waffle House in Arizona! |
05:55.45 | Strom_C | at this rate, there will eventually be one in Los Angeles |
05:56.28 | sevard | ... |
05:56.48 | dlynes_home | sevard: ow! wtf was that for? |
05:57.25 | sevard | HOLY CRAP MAN |
05:57.35 | sevard | that was delayed... how many hours?! |
05:57.52 | *** join/#asterisk aclark78 (n=aclark@cpe-70-116-103-202.houston.res.rr.com) |
05:58.35 | mitcheloc | Strom_C: we already have ihop and arthurs ;) |
05:58.53 | Strom_C | "arthurs"? |
05:59.14 | mitcheloc | what city are you in? |
05:59.45 | Strom_C | also, can you walk into an ihop, order hash browns scattered, covered, diced, and be greeted by someone who even has a clue what the hell you're yammering on about? no! :) |
06:00.04 | Strom_C | mitcheloc, I live in Los Angeles, though right now I'm in Huntsville AL |
06:00.26 | mitcheloc | arthurs is in whittier, are you visiting digium? |
06:00.58 | Strom_C | oh ok, i'll have to schlep out to Whittier when I get back |
06:00.58 | Strom_C | yes, I am |
06:01.53 | sevard | Strom_C: the only ihop i was at was in las vegas and our waitress was clearly a huge meth head |
06:02.08 | Strom_C | doesn't surprise me in the least |
06:02.16 | Strom_C | that's vegas for you |
06:02.21 | sevard | she had about 5 teeth left |
06:03.03 | *** join/#asterisk mrtwister (n=manopulu@107.250.broadband5.iol.cz) |
06:03.21 | mitcheloc | Strom_C: i'm supposed to visit there next month ;) |
06:03.28 | Strom_C | mitcheloc, ah cool |
06:03.39 | Qwell | mitcheloc: huntsville? |
06:03.53 | mitcheloc | Qwell: yep |
06:04.01 | Qwell | what timeframe? |
06:04.30 | mitcheloc | the 6th to the 10th, why do you ask? |
06:04.42 | Qwell | because I'll be there on the 9th.. |
06:04.54 | Qwell | seems like a happenin' place, heh |
06:05.14 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
06:05.22 | mitcheloc | ah, yea, are you going on business? (i'm just going for fun) |
06:05.27 | Strom_C | there's waffle house and bumpers and...um...what else is there to do here besides phones and grease and booze and 8-ball? |
06:06.28 | sevard | you're a coke fiend? |
06:06.42 | Strom_C | no |
06:06.47 | Strom_C | neither coca-cola nor cocaine |
06:06.47 | sevard | liar. |
06:11.41 | dlynes_home | sevard: i was sleeping..sheesh |
06:11.51 | Strom_C | waffle house! |
06:11.56 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
06:12.51 | mitcheloc | heh Strom_C is it really that good? i'll have to check it out then when i go ;) |
06:13.14 | Strom_C | mitcheloc, yes |
06:13.17 | Strom_C | it's addictive |
06:13.49 | dlynes_home | mitcheloc: the coke, or the meth? |
06:14.04 | Strom_C | the meth waffles |
06:14.13 | Strom_C | and the coke hash browns |
06:14.21 | mitcheloc | sounds good then, it's worse weather though isn't it? =( |
06:14.30 | Strom_C | hot and muggy |
06:14.32 | Strom_C | bring shorts |
06:14.33 | Strom_C | tshirts |
06:14.38 | Strom_C | and no underwear |
06:14.47 | mitcheloc | aww/eww |
06:14.56 | Qwell | w00t - 2.6.17 == niagara support |
06:15.08 | dlynes_home | niagara? |
06:15.16 | Qwell | new Sun chip |
06:15.18 | dlynes_home | you can go over the falls in a barrel with linux now? |
06:15.19 | dlynes_home | oh |
06:15.25 | *** join/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au) |
06:15.34 | P-NuT | Hi all. |
06:15.58 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
06:16.18 | P-NuT | Does anyone know what the cheapest FXO card is I can get in Australia? |
06:16.27 | dlynes_home | P-NuT: x100p |
06:16.29 | P-NuT | single line. |
06:16.46 | dlynes_home | you didn't say how many ports |
06:16.48 | Strom_C | of course, the X100P will make you stab your eyes out with white-hot knitting needles |
06:16.49 | dlynes_home | and you said cheapest |
06:17.06 | mitcheloc | what i never understood is why asterisk doesn't support regular old modems... |
06:17.20 | dlynes_home | mitcheloc: it does |
06:17.26 | dlynes_home | mitcheloc: but only two specific models |
06:17.33 | Strom_C | because most regular old modems don't really work all that well for voice communications |
06:17.50 | dlynes_home | mitcheloc: which is what the x100p, and x101p cards are |
06:17.54 | neilbags-work | dlynes_what models are supported, and do they work well? |
06:17.57 | sevard | what about voice modems? |
06:18.03 | mitcheloc | dlynes_office: yep but i can't pick those up at Fry's |
06:18.13 | dlynes_home | neilbags-work: intel and motorola |
06:18.26 | Strom_C | mitcheloc, just buy a tdm400 and quit kvetching :) |
06:19.23 | dlynes_home | P-NuT: if you don't want x100p, i'd suggest a sangoma a200 or a200d with 1 to 12 fxo modules (2-24 fxo ports) |
06:19.26 | mitcheloc | nah, the real truth is a conspiracy to get us to buy tdm400's ;) |
06:19.40 | dlynes_home | P-NuT: a200 is the regular card; a200d is the same card, but with hardware echo cancellation |
06:23.44 | sevard | freaka you |
06:23.46 | sevard | freaka me |
06:24.23 | P-NuT | well, I have a x100p and it works totally fine for me. |
06:24.30 | P-NuT | I've had no issues at all. |
06:24.50 | neilbags-work | hi, is anyone using chan_bluetooth? is it un the current stable asterisk releases? |
06:25.08 | P-NuT | But I want to buy a card for a mate's asterisk box, but x100p's are hard to come by here is Oz. Anybody wanna sell me one? lol |
06:25.25 | Strom_C | P-NuT, the x100p has been discontinued |
06:26.21 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
06:26.35 | neilbags-work | P-NuT: i'm having a weird problem with my x100p on telstra, if you call in just after the line has been hung up (say 10 seconds later) asterisk answers the call and emits a high-pitched tone ... considering trying different hardware |
06:27.05 | Strom_C | TDM400P with single FXO module will be reasonably inexpensive and work well for you |
06:29.17 | P-NuT | neilbags-work: I'm going to write an article about x100p and telstra soon, so hang out for a few weeks. |
06:29.27 | orlock | just blame telstra |
06:29.35 | P-NuT | a few settings you need in zapata.conf apparently |
06:29.36 | orlock | there wont be anybody there knowledgable enough to deny its their fault |
06:29.38 | orlock | :) |
06:29.40 | neilbags-work | P-NuT: do you know about this problem? |
06:30.11 | neilbags-work | P-NuT: i get an error saying that the caller ID module exited with an error, something like that |
06:30.21 | P-NuT | no, but I had a few weird issues with telstra and x100p until I stuffed about with zapata.conf |
06:30.28 | P-NuT | try this... |
06:31.01 | neilbags-work | is the x100p obsoleted by x100m, or am i confused? |
06:31.17 | Strom_C | x100p is replaced by TDM400P |
06:31.38 | mitcheloc | or you could use a t100p to create a channel bank |
06:31.45 | neilbags-work | ok, P-NuT i have a TDM400P with 1 FXO module, not a x100p |
06:31.54 | mitcheloc | and yes i'm serious, i've got one for my house ;) |
06:32.01 | neilbags-work | i think |
06:34.10 | neilbags-work | P-NuT: i used http://www.voip-info.org/wiki/view/Australia+Asterisk+Details, is this info correct? |
06:36.59 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
06:38.58 | *** join/#asterisk littlejohn (n=little@82.53.4.87) |
06:39.21 | P-NuT | neilbags-work: I use this in zapata.conf |
06:39.23 | P-NuT | http://pastebin.ca/67058 |
06:40.26 | P-NuT | try that. |
06:40.37 | P-NuT | but change the context to your context. |
06:41.22 | neilbags-work | P-NuT: yeah but there is nothing in there that looks like it'll solve my obscure problem |
06:41.29 | neilbags-work | here's mine: http://pastebin.ca/67059 |
06:41.47 | nextime | aleeee |
06:41.51 | nextime | ops |
06:42.10 | P-NuT | well, mine looks kinda different.. |
06:42.11 | neilbags-work | what does callwaitingcallerid=yes do? |
06:42.32 | neilbags-work | i have call-waiting turned off so it shouldn't matter |
06:42.40 | P-NuT | why dont you try my config? but.... where is your context? |
06:43.06 | neilbags-work | yeah they are different but mine does work in every other situation .... default context |
06:43.23 | P-NuT | hmm.. |
06:43.34 | P-NuT | give mine a try. You have nothing to lose.. |
06:44.10 | neilbags-work | i'll try it next time i'm on site, cant break things in business hours :( |
06:44.30 | neilbags-work | i just can't see anything in there that would solve my problem |
06:44.36 | neilbags-work | what does callreturn=yes do? |
06:44.46 | P-NuT | Ahh yes... |
06:44.49 | P-NuT | umm... |
06:44.51 | P-NuT | yes.. |
06:45.11 | P-NuT | no idea. |
06:45.15 | neilbags-work | and i assume all the callwaiting stuff is irrelavent since i have it disabled on the line |
06:45.37 | neilbags-work | my gains are different, but again, they work fine for me |
06:45.42 | P-NuT | but all I know is all of my weird call hanging issues, low volume and not hanging up properly were magically solved by that/ |
06:45.59 | neilbags-work | by what exactly? |
06:46.57 | P-NuT | changing the zapata conf to that |
06:47.09 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
06:47.11 | *** join/#asterisk akke (n=akke@85.158.211.235) |
06:47.11 | neilbags-work | P-NuT: ok, where did you get that config from? |
06:47.15 | P-NuT | but hey, I hope it works for you. |
06:47.17 | P-NuT | Ahh |
06:47.24 | P-NuT | bits from here... |
06:47.33 | P-NuT | http://www.loligo.com/asterisk/current/ |
06:47.43 | P-NuT | great config example that one.. |
06:47.44 | neilbags-work | P-NuT: can you confirm that you don't have the same problem as me? try calling the line as soon as it has been hung up ... |
06:47.49 | akke | can anyone recommend me a provider that offers a 'dial-in number' from belgium? |
06:48.23 | P-NuT | I do that and it's still offhook. asterisk doesn't hangup the call imediatley |
06:48.40 | P-NuT | so therefore I dont get the beeping that you have. |
06:48.48 | P-NuT | But hey, it could always be dodgy hardware. |
06:49.08 | P-NuT | so what has replaced the x100p now? What's the new el-cheapo card? |
06:49.12 | neilbags-work | P-NuT: yes thats what i'm currently putting it down to |
06:49.28 | P-NuT | I'd just do that |
06:49.36 | neilbags-work | P-NuT: its more like a screaming than a beeping :( |
06:50.31 | neilbags-work | P-NuT: so if asterisk hasn't hung up the line yet, what happens, do you get a busy signal? how long does asterisk take to hang up your line? why the delay? |
06:53.21 | P-NuT | busy ssignal for about 30 secs to 1 min |
06:53.44 | P-NuT | then asterisk resets the line and we're good to go again. |
06:54.02 | P-NuT | so where do you get your VOIP hardware from? |
06:54.12 | neilbags-work | P-NuT: hmmm .... i don't think i have a delay like that but i've never really tested it well. |
06:54.28 | neilbags-work | P-NuT: i'm pretty sure my asterisk hangs up straight away |
06:54.49 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
06:54.57 | neilbags-work | i got my TDM400P from techtopia (i think) and I buy linksys PAP2s from a computer stop down the street |
06:56.08 | P-NuT | got a link for techtopia? |
06:56.51 | neilbags-work | google it |
06:56.53 | P-NuT | how hard are the linksys's to setup? and how many lines do they take? |
06:57.11 | neilbags-work | there is 2 FXS ports on each device |
06:57.20 | P-NuT | so.. 2 incoming lines? |
06:57.40 | neilbags-work | 2 sip lines, 2 handsets |
06:58.19 | neilbags-work | i havn't changed anything from the defaults on the linksys pap2s except for IP address, sip proxy and dial plan |
06:59.24 | *** join/#asterisk lorinc (n=ang@caracas-0691.adsl.interware.hu) |
07:01.01 | P-NuT | cool |
07:01.57 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
07:06.06 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
07:06.35 | FuriousGeorge | appears to work |
07:06.45 | FuriousGeorge | someone say my name |
07:06.48 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
07:07.14 | FuriousGeorge | please? |
07:07.29 | Strom_C | FuriousBill |
07:07.36 | Strom_C | FuriousJames |
07:07.44 | FuriousGeorge | :| |
07:07.47 | hads|home | FuriousTom |
07:08.15 | FuriousGeorge | :| |
07:08.41 | FuriousGeorge | seriously, i wanna see what the client does |
07:09.41 | hads|home | FuriousGeorge! |
07:09.48 | FuriousGeorge | thanks hads|home |
07:09.49 | *** join/#asterisk prh (n=paul@X80.mjr.org) |
07:09.55 | hads|home | :) |
07:09.59 | Strom_C | FuriousJorge! |
07:10.04 | FuriousGeorge | now say my name again, but this time while you talk dirty to me |
07:10.15 | Strom_C | oh FuriousGeorge, dial me harder |
07:10.18 | *** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
07:10.20 | FuriousGeorge | lol |
07:10.28 | hads|home | ha |
07:10.36 | Strom_C | hello, FuriousGeorge, i'm here to punch my big tool into your connecting block |
07:11.07 | FuriousGeorge | i buiilt gaime with my new gcc 4.1 based toolchain and now it gets all core-droppy |
07:11.26 | Strom_C | it doesnt like you misspelling its name |
07:11.54 | FuriousGeorge | that must be it |
07:12.07 | hads|home | gaym |
07:13.25 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
07:13.34 | lilo | FuriousGeorge: ember |
07:13.35 | lilo | ? |
07:13.42 | *** join/#asterisk Sonderblade (n=mah@host-213.131.147.169.addr.tdcsong.se) |
07:14.26 | akke | can anyone recommend me a provider that offers a 'dial-in number' from belgium? |
07:14.37 | *** join/#asterisk jonnysupersonic (n=jonny@dsl-145-56-131.telkomadsl.co.za) |
07:14.46 | FuriousGeorge | lilo: arent you famous or something |
07:14.50 | x86 | akke: that'd be DID |
07:14.54 | x86 | akke: what's the country code? |
07:14.59 | lilo | FuriousGeorge: no, I'm not a boot loader :) |
07:15.06 | akke | x86: +32 |
07:15.21 | FuriousGeorge | i was thinking the disney cartoon chacter |
07:15.28 | FuriousGeorge | lilo: what you mean ember |
07:15.28 | lilo | I'm named after a favorite science fiction character, in John Varley's THE OPHIUCHI HOTLINE. I am *not* named after the LInux LOader, and I didn't write it! :) -- lilo, winter 1993 || Nor am I named after the little girl in the movie, nor the air mattress, nor Last In Last Out. ;) -- lilo, summer 2003 |
07:15.36 | lilo | FuriousGeorge: cartoon humor, ignore it 8) |
07:16.08 | FuriousGeorge | fair enough |
07:16.10 | FuriousGeorge | :) |
07:16.27 | x86 | akke: i'm looking now |
07:16.33 | akke | x86: thanks |
07:16.46 | x86 | akke: you just need termination in the US, and origination in belgium? |
07:17.23 | x86 | for example, people can call you locally from belgium, but you only dial the US |
07:17.50 | akke | x86: i live in belgium but I want some extra numbers that come into my asterisk box. I don't need to dial out using that number... |
07:18.06 | x86 | ah ok |
07:18.09 | x86 | i can hook you up :) |
07:18.15 | x86 | akke: talk in private? |
07:18.22 | akke | okay |
07:22.08 | SkramX | eeeks, I need to be up forwork in ~4 hours |
07:22.09 | SkramX | peace out |
07:26.32 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
07:29.27 | *** join/#asterisk Shoragan (n=shoragan@134.169.175.72) |
07:35.00 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
07:41.12 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
07:43.11 | *** join/#asterisk cypromis (n=michal@voiceworks.pl) |
07:43.39 | *** join/#asterisk astar` (n=astar@ANantes-154-1-7-185.w81-53.abo.wanadoo.fr) |
07:51.07 | *** join/#asterisk QuAtRo[NL] (n=QuAtRo_@dsl-083-247-051-039.solcon.nl) |
07:51.34 | QuAtRo[NL] | Hi all |
07:51.51 | QuAtRo[NL] | I'm searching for a Astrisk 'command' which rings all phones... |
07:52.01 | mitcheloc | dial |
07:52.22 | *** join/#asterisk martijn_ (n=martijn@dbcorp.xs4all.nl) |
07:52.24 | QuAtRo[NL] | In that case i have to specify all phones, haven't i? |
07:52.37 | QuAtRo[NL] | don't i? |
07:52.47 | dpryo | Set up a queue with ringall strategy? |
07:54.06 | QuAtRo[NL] | I'm developping my own Control Panel voor Asterisk, i want a default rule for all numbers which no dialplan assigned to them.. |
07:54.20 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
07:54.27 | FuriousGeorge | call group? |
07:54.37 | *** join/#asterisk MooseAble (n=ismoose@203-59-195-5.dyn.iinet.net.au) |
07:54.51 | QuAtRo[NL] | In that case i still have to assign the users to a group... |
07:55.13 | QuAtRo[NL] | It should be enough to create an extension for an user... |
07:55.16 | QuAtRo[NL] | Is that possible |
07:55.17 | QuAtRo[NL] | ? |
07:55.28 | FuriousGeorge | dynamically? |
07:55.46 | MooseAble | I'm trying to solve an extention registration problem at the moment... i would like to know what "Auto destroying call" means in the asterisk logs first off... can't find documentation anywhere covering this. |
07:55.46 | QuAtRo[NL] | Yes |
07:56.05 | FuriousGeorge | a meetme and an api script that makes it happen |
07:56.31 | QuAtRo[NL] | Ok, thanks! |
07:56.48 | *** join/#asterisk saftsack (n=saftsack@p54A7FCEB.dip.t-dialin.net) |
08:00.41 | motu | how do i make my fxo device detect pickups? right now it just times out even if someone answers |
08:11.08 | af_ | there is something could I put in the dialplan like autohangup <tme> ? |
08:19.08 | *** join/#asterisk Modcuts (n=bob@82.133.98.155) |
08:22.05 | erwinism | when i add this line "channel => 1" to my zapata.conf, it gives me error chan_zap.c: Unable to open channel 1: No such device or address |
08:22.12 | erwinism | what did i miss? |
08:24.47 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
08:24.52 | motu | what does the various wink settings in zapata.conf mean? |
08:25.13 | dec | ;) |
08:25.46 | motu | where can i find documentation on them |
08:26.08 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
08:26.38 | kmilitzer | Grrr, never book an EasyJet flight ... |
08:27.14 | saftsack | lo |
08:27.15 | saftsack | l |
08:27.37 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
08:43.18 | MooseAble | anyone managed to get a adsl router with built in voip to register on their LAN hosted asterisk server? Just having a little trouble here. |
08:43.46 | MooseAble | whilst the asterisk server registers to an external sip server for inbout/outbound calls that is |
08:45.35 | MooseAble | router keeps reporting "TRYING:No Response" and asterisk keeps giving it "Auto destroying call 'DC4C-D416-8172465-DD2A58ED28D6-0008@203.59.xxx.xxx'" (despite the fact that the routers lan address is "10.1.1.1", 203.59.xxx.xxx is the wan ip address) |
09:01.40 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
09:08.17 | *** join/#asterisk pappu (n=giridhar@60.254.116.238) |
09:09.39 | pappu | hi every on .. i need some help on the AGI .. i have written a agi script which does some database check and return some value |
09:10.06 | pappu | how do i capture that in dialplan ? |
09:10.58 | Eric-xx | -- Got SIP response 302 "Moved temporarely" back from 194.221.62.207 |
09:11.03 | Eric-xx | does anyone know what this means? |
09:12.04 | pappu | how do i capture the return value of a agi script in a dailplan ? |
09:12.34 | QuAtRo[NL] | Are there any some known problems with timeouts, which seem not work? |
09:16.55 | erwinism | hello, i have Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface. but when i run ztcfg i got the error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
09:17.01 | erwinism | any ideas? |
09:17.04 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
09:22.19 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
09:23.38 | *** join/#asterisk andrebarbosa (n=andrebar@62.48.215.93) |
09:27.25 | andrebarbosa | asterisk does g729b encoding\decoding? cant find any info about this.. |
09:27.36 | andrebarbosa | can make pass-thru, tough |
09:28.20 | RoyK | rotfl http://karlsbakk.net/fun/kebab.jpg |
09:28.41 | RoyK | andrebarbosa: don't think there's such a thing as a g.729b codec for * |
09:29.10 | Luke-Jr | how about h264? =p |
09:30.29 | RoyK | Luke-Jr: methinks libavcodec integration would be neat |
09:30.50 | andrebarbosa | ok, thanks RoyK |
09:30.51 | andrebarbosa | :) |
09:35.23 | tzafrir | erwinism, a "Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface" can be one of quite a few cards. Is it a X100P FXO card? |
09:35.50 | tzafrir | pastebin the output of the following: |
09:35.55 | *** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr) |
09:36.02 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
09:36.08 | Dr-Linux | hi guys |
09:36.19 | tzafrir | lsmod | grep zaptel; cat /etc/zaptel.conf |
09:36.22 | erwinism | yes, it works when i "modprobe wcfxo" do i have to do this everytime i start asterisk? |
09:36.23 | Dr-Linux | what these Alarms mean? http://pastebin.com/718016 |
09:36.44 | tzafrir | erwinism, what distro do you use? |
09:36.46 | Dr-Linux | tzafrir: please have a look, i just installed new digium te210p calrds |
09:36.55 | *** join/#asterisk Pj_ (n=pj@fernande.happycoders.org) |
09:36.56 | erwinism | tzafrir, centos |
09:37.03 | Dr-Linux | s/calrds/cards |
09:37.04 | Pj_ | G'morning people |
09:37.41 | tzafrir | erwinism, if you use the standard zaptel.init init script, just set MODULES=wcfxo in /etc/sysconfig/zaptel |
09:38.09 | Eric-xx | -- Got SIP response 302 "Moved temporarely" back from 194.221.62.207 |
09:38.10 | Eric-xx | does anyone know what this means? |
09:38.13 | [pyro] | evening :) |
09:38.24 | Dr-Linux | tzafrir: any clue for my question? |
09:38.30 | Pj_ | evening too yeah ;) |
09:38.30 | tzafrir | Dr-Linux, not a PRI expert, but "red" basically means something like "no cable", right? |
09:39.01 | Dr-Linux | tzafrir: i'm not sure my friend |
09:39.11 | erwinism | tzafrir, thank you very much |
09:40.40 | Dr-Linux | tzafrir: but my FXO asterisk server, it shows 1 line for each card |
09:40.41 | Dr-Linux | Wildcard TDM400P REV I Board 1 OK 0 0 0 |
09:40.41 | Dr-Linux | Wildcard TDM400P REV I Board 2 OK 0 0 0 |
09:41.08 | Dr-Linux | but not sure how PRI works in this matter |
09:41.45 | Dr-Linux | tzafrir: would you like to see my zaptel.conf? |
09:43.27 | nextime | red for pri is no cable or d-channel down |
09:45.47 | Dr-Linux | nextime: can i show you my zaptel.conf? |
09:46.26 | nextime | Dr-Linux sure |
09:47.51 | Dr-Linux | nextime: have a look >> http://pastebin.com/718024 |
09:48.19 | Dr-Linux | nextime: i have two TE210P card installed |
09:48.23 | Dr-Linux | 2 port each |
09:49.05 | erwinism | tzafrir, i have another problem, when i call from my voip to pstn, i can hear my own voice echoing at one time only, how can i fix this? |
09:49.17 | RoyK | Dr-Linux: http://karlsbakk.net/fun/kebab.jpg |
09:49.21 | nextime | Dr-Linux use only one time source, so, set the first span to 1, the second to 2 and others to 0 |
09:49.55 | nextime | your have now all 4 spans set to 1 that mean "use this span as a primary timiming source" |
09:49.58 | Dr-Linux | nextime: i didn't understand :S |
09:50.27 | nextime | you have span=1,1,0,esf,b8zs |
09:50.37 | nextime | and span=2,1,0 |
09:50.42 | nextime | span=3,1,0 |
09:51.07 | *** join/#asterisk UlbabraB (n=UlbabraB@host-84-222-44-158.cust-adsl.tiscali.it) |
09:51.08 | Dr-Linux | nextime: yes |
09:51.13 | nextime | use the first one as span=1,1,0, the second one span=2,2,0 the third and the fourth span=X,0,9, |
09:51.14 | Dr-Linux | so how should i use? |
09:51.36 | *** join/#asterisk RoyKa (n=roy@80.239.107.70) |
09:51.57 | nextime | the second parameter is the timing source, you need to have only one time source, and for failover purpose a secondary one |
09:52.07 | Dr-Linux | wht do you mean by span=X,0,9, ? |
09:52.14 | nextime | so, set the span 1 to be the primary time source, the span 2 to secondary, others to 0 |
09:52.44 | nextime | Dr-Linux : span=3,0,0,esf,b8zf and span=4,0,0... |
09:53.37 | nextime | i don't see any other error in your config, are you sure that 24 is the right d-channel? |
09:53.57 | nextime | ( i don't have t1 pri, i'm in europe ) |
09:55.41 | Dr-Linux | nextime: check now please >> http://pastebin.com/718034 |
09:55.44 | Dr-Linux | is tht what you said? |
09:55.46 | *** join/#asterisk pjo (n=pjo@212.88.98.114) |
09:56.13 | nextime | Dr-Linux : yes, exactly |
09:56.16 | Dr-Linux | nextime: yes 24 is the right d-channel |
09:57.43 | pjo | hi all, when the docs say 's' isn't a "catchall extension" does it mean doing something like exten => 12345,1,Dial(IAX2/HOSTNAME/s@default) will not just go through the default context on iax peer HOSTNAME? |
09:57.57 | Pj_ | I've been testing the BT-100 and I was wondering if you had any recommandation for other phones in that price range... Those looks cheap, act "user-unfriendly" (speaker button) and offer only 10-baseT connectivity |
09:57.58 | nextime | Dr-Linux : sorry i was thinking that you have one quad-pri card, as i can see now you have 2 double span cards, so, you can set a primary and a secondary time source for any card, it's meaning that you can set span 1 to 1, span 2 to 2, span 3 to 1, span 4 to 2 |
09:58.27 | nextime | ( not a must, but you can if you prefer ) |
09:58.42 | nextime | anyway, your config is correct |
09:59.11 | Dr-Linux | ok |
09:59.20 | nextime | if you get a red error, it mean that you cable isn't right connected or your dchannel is down |
09:59.51 | Dr-Linux | nextime: what you prefer, should i change them back to the previous setting? |
10:00.39 | nextime | Dr-Linux : no, previus setting was all 4 spans to primary time source, isn't correct. Only one primary source for card, one primary and one secondary for card |
10:01.45 | Dr-Linux | okey great i understand |
10:01.59 | Dr-Linux | nextime: now would you like to see my zapata.conf? |
10:02.20 | MooseAble | anyone know what "Jun 19 05:59:06 DEBUG[1388]: Auto destroying call 'DC4C-D416-25900427-67DA7E70EEDE-0019@203.59.xxx.xxx'" means? |
10:02.25 | nextime | Dr-Linux it's not related with red alarm, but yes, paste it |
10:02.55 | Dr-Linux | nextime: it's my new setup, so just wanna verify.. |
10:03.04 | nextime | Dr-Linux : ok |
10:03.50 | Dr-Linux | http://pastebin.com/718048 |
10:04.06 | Dr-Linux | nextime: looks like there are few errors, please verify |
10:06.22 | Dr-Linux | what should be here: |
10:06.23 | Dr-Linux | # |
10:06.24 | Dr-Linux | pridialplan=unknown |
10:06.24 | Dr-Linux | # |
10:06.24 | Dr-Linux | prilocaldialplan=unknown |
10:08.06 | nextime | rxgain and txgain are obsolete if i remember right, and you need to specify context ( one for all groups, or one for group, as you prefer ), for the latest group you can use channel => 49-71,73-95, i suggest to add overlapdial=yes, pridialplan and prilocaldialplan are ok to unknown, or you can set national / local |
10:09.08 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
10:09.11 | Bert- | hello there |
10:09.34 | Dr-Linux | nextime: so should i remeve rxgain/txgain? |
10:09.47 | Dr-Linux | and what it does? overlapdial=yes |
10:10.32 | Bert- | Well When I call someone through my *, when the called party hangup, Asterisk don't, and maintains the connection between my sipphone and himself. Is there a way to make * asterisk detecting disconnection plz ? |
10:10.57 | Bert- | timeout issue ? |
10:10.58 | nextime | Dr-Linux : overlapdial is for calls come from old analogic phone source, to catch the right extension on your primary, if you don't specify overlapdial, calls come from old analog lines come all to your shirt number |
10:11.06 | nextime | s/shirt/short |
10:11.41 | Dr-Linux | ok great |
10:12.00 | Dr-Linux | what about rxgain/txgain, i'm using them in other servers with FXO's |
10:12.13 | Dr-Linux | don't i need them with PRI setup? |
10:12.23 | QuAtRo[NL] | nextime saw too many soccer |
10:12.25 | QuAtRo[NL] | :P |
10:13.03 | nextime | Dr-Linux : no, you don't need them |
10:14.04 | Dr-Linux | nextime: what if i'd need increase or decrease rxgain/txgain ... then what i will do? :S |
10:14.17 | Dr-Linux | nextime: or i won't need that with PRI? |
10:14.52 | nextime | Dr-Linux : in my opinion you don't need them on a isdn channel, pri or bri is indifferent |
10:14.55 | Bert- | hmm noone can help me ? |
10:15.18 | nextime | anyway, you can use it if you think, no matter about that |
10:15.19 | Bert- | Is a way to detect disconnection on a SIP trunk ? |
10:15.32 | Dr-Linux | i see |
10:16.15 | Dr-Linux | nextime: actually i'm not sure if "cables" are plugged to my cards or not, that's why i have doubt about RED alarams |
10:16.28 | Dr-Linux | nextime: i'm in pakistan and my servers are in the US datacenter |
10:17.04 | Dr-Linux | nextime: i just wanna know, what possible things can make it "RED Alarm" |
10:17.19 | Dr-Linux | ? |
10:18.16 | nextime | Dr-Linux : a bad cable, a too long cable ( if it's too long, you can set the third parameter of span= to something different to 0 ), or a dchannel down from carrier |
10:18.31 | nextime | or, of course, a disconnected cable |
10:19.30 | nextime | or a wrong dchannel number, but as you say it isn't your case |
10:20.20 | Dr-Linux | nextime: i see |
10:20.28 | nextime | or a wrong framing/coding combination, but i don't think that is your case, cause your first span is ok |
10:20.46 | Bert- | <PROTECTED> |
10:20.58 | Dr-Linux | but didn't understand the logic if the cable is too long i can set third parameter of span= to somehting different than 0 :S |
10:21.04 | Bert- | it is my softphone which doesn't detect disconnect ... :) |
10:21.43 | Dr-Linux | nextime: well, my first Span is "OK" but not others |
10:21.54 | Dr-Linux | nextime: so what you think, what could be the reason |
10:22.13 | Dr-Linux | as you saw, my zap config is fine. that what's new :S |
10:23.14 | Dr-Linux | nextime: if the cable is not plugged it should show the 1st span also "RED" but in my case 1 span is OK but 2nd is RED :S |
10:23.23 | nextime | Dr-Linux : if your cable is very long, you can receive a "low signal" on that. So, using a different set on the third parameter you say to zaptel to be more "sensitive", you must use 1 if your cable is from 133 to 266 feet, 2 from 267 to 399 and so on |
10:24.26 | nextime | Dr-Linux : you will have 4 pri with 4 different cable, if your first cable is plugged and the others 3 cable aren't connected, you get ok on the first span and red on the others 3 spans |
10:24.54 | nextime | any span have a different cable |
10:25.06 | Dr-Linux | nextime: i understand |
10:25.07 | Mw3 | hm, i hear double ringing on my sip device when dialing out on a PRI. what can the cause of this? |
10:25.28 | Dr-Linux | but really the long/short cables matter is very good to know :S |
10:25.50 | nextime | Mw3 : maybe are you answer before the dial command to call over pri? |
10:26.23 | MooseAble | anyone know what "Jun 19 05:59:06 DEBUG[1388]: Auto destroying call 'DC4C-D416-25900427-67DA7E70EEDE-0019@203.59.xxx.xxx'" means? |
10:26.39 | Pj_ | that resistence is futile |
10:26.56 | nextime | we will be assimilated? |
10:27.22 | MooseAble | your head a-splode |
10:28.13 | pjo | can anyone recomend a good IAX2 client for macos (no not * .. that's a server :-D ) |
10:28.21 | pjo | err OS X |
10:28.27 | nextime | pjo : wengophone |
10:28.33 | pjo | nextime: thx. |
10:28.40 | nextime | oh |
10:28.44 | nextime | you say IAX |
10:28.49 | nextime | sorry wengo is sip |
10:29.02 | pjo | ahh.. yes. i meant iax |
10:29.07 | nextime | so, no idea for iax, sorry |
10:29.33 | pjo | k. |
10:30.18 | Mw3 | nextime: no, i dont. my outgoing dialplan is just a SetAMAFlags(billing) and a Dial(ZAP/g1/${EXTEN},60) |
10:30.29 | Eric-xx | how do i change my User-Agent: Asterisk PBX ? |
10:32.53 | nextime | ok, lunch time, bye all |
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10:36.48 | andrebarbosa | Eric-xx, on sip.conf |
10:36.58 | andrebarbosa | under [general] |
10:37.18 | andrebarbosa | useragent=XXXXXX |
10:37.19 | andrebarbosa | ;) |
10:37.44 | pappu | hi how do we enable echo canlellation on zaptel card ? |
10:42.45 | pjo | pappu: echocancel=yes in zapata.conf |
10:43.05 | erwinism | tzafrir, i have another problem, when i call from my voip to pstn, i can hear my own voice echoing at one time only, how can i fix this? |
10:43.19 | erwinism | anyone can help me? |
10:44.10 | Eric-xx | thanks andrebarbosa |
10:46.23 | tzafrir | erwinism, for starters, set opermode (the parameter to the driver) appropriate to your country |
10:46.41 | tzafrir | once that is done, read about echo cancelling |
10:47.10 | tzafrir | e.g: echocancell=yes in zapata.conf |
10:47.50 | erwinism | i already have that echocancell=yes, tzafrir , i am in philippines right now |
10:50.21 | pjo | I have a voicetronix openswitch 12 card.. setting callerid=yes in vpb.conf results in a lot of static noises on the FXS ports which doesn't go away till I reload the driver module. Any ideas on how to fix that? |
10:52.10 | erwinism | tzafrir, how can i set opermode? |
10:55.48 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
10:56.40 | erwinism | by the way, i use the clode x100p card |
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11:01.25 | tzafrir | it is a module parameter. You can set it manually when loading the module or through modprobe.conf |
11:03.20 | erwinism | ok thank you :) |
11:07.05 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
11:16.27 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
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11:29.22 | zwelch | puzzled: yes, sort of (in answer to your question in #minisip) |
11:30.22 | puzzled | zwelch: great, let me get back there |
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11:54.51 | MooseAble | well, looks like nobody knows teh answer. seems I'll never get this router to register with asterisk o_O |
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11:59.12 | fbucher | hi |
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12:03.08 | Mavvie | anybody here familair with the Outlook Dialer from Third lane? |
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12:11.51 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
12:12.12 | Dr-Linux | anybody using spa-2100 or 2000? |
12:13.04 | Dr-Linux | [TK]D-Fender: can i access spa-2100 device from the local network? |
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12:19.01 | [TK]D-Fender | Dr-Linux : Should be able to (for admin), bup SIP is only functional on the WAN interface which I why I would never buy it. |
12:19.09 | *** part/#asterisk Ludo_ (n=Ludo@obelix.zoxx.net) |
12:20.16 | Dr-Linux | [TK]D-Fender: i can access spa-2100 device when i plug a cable in PC port of device, wheather device is on network, but i can't access it from the local network. |
12:22.42 | *** join/#asterisk tomtom_ (n=root@83.217.70.166) |
12:22.50 | tomtom_ | Hi |
12:22.54 | *** join/#asterisk Webboarder (n=rubentul@dsl-083-247-051-039.solcon.nl) |
12:22.55 | littleRalphie | anyone have an x100p o sell? |
12:23.14 | tomtom_ | Anyone can help me with DID's in the Phillipines? pvt msg pls |
12:23.34 | [TK]D-Fender | Dr-Linux : maybe you didn't enable outside access to it... |
12:26.43 | Webboarder | Hello, I have a problem with queue's. I am logged in as an agent at queue 'support'. But when i log off as an agent, my phone still rings when someone is in the queue 'support'. How can i solf this problem? |
12:27.13 | Dr-Linux | [TK]D-Fender: i see, i'd need to see from where can i enalble this access :S |
12:28.00 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
12:30.13 | clive- | littleralph ebay is your friend |
12:31.55 | [TK]D-Fender | Webboarder : Pastebin your queues.conf, agents.conf, and extensions.conf |
12:32.42 | [TK]D-Fender | ~pb |
12:33.00 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
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12:39.25 | Webboarder | [TK]D-Fender : http://pastebin.com/718264 |
12:40.14 | Dr-Linux | [TK]D-Fender: please have a look and let me know, why 3 span's alarms are "RED" http://pastebin.com/718048 |
12:40.36 | Dr-Linux | http://pastebin.com/718268 << here |
12:42.15 | *** join/#asterisk speedwagon (n=FreePBX5@70.46.87.158) |
12:43.09 | iCEBrkr | W3rd. |
12:46.46 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
12:46.51 | [TK]D-Fender | Webboarder : How are you "logging out"? |
12:48.05 | *** join/#asterisk Darthclue (n=Darthclu@adsl-69-152-233-213.dsl.snantx.swbell.net) |
12:48.27 | Webboarder | [TK]D-Fender : I am calling 1000, then pressing 1, enterring my extension number followed by '#'. And finally when asked to enter a new extension number I only enter a '#'. |
12:48.50 | Webboarder | sorry i'm pressing 2 instead of 1 |
12:50.50 | [TK]D-Fender | Webboarder : Hmmm... |
12:51.08 | [TK]D-Fender | Dr-Linux : Have you tried a loopback test? |
12:51.45 | Dr-Linux | [TK]D-Fender: sir, how can i do that? |
12:53.17 | iCEBrkr | [TK]D-Fender: You're like Asterisk Tech support 24/7 :P |
12:53.27 | [pyro] | lol |
12:53.35 | *** join/#asterisk hongtien (n=hongtien@203.162.100.12) |
12:53.45 | Dr-Linux | he glads to help :) |
12:54.14 | hongtien | Hi |
12:54.57 | [TK]D-Fender | Dr-Linux : Go read up on how to make a loopback connector dongle. |
12:55.33 | hongtien | Hi all |
12:56.03 | hongtien | I want to use Asterisk with Voice Gateway |
12:56.11 | [TK]D-Fender | Dr-Linux : May want to verify your wiring as well. When in doubt plug your other lines onto port 1 and see if they come up. |
12:56.11 | Dr-Linux | [TK]D-Fender: ok, and your thoughs about my configs? |
12:56.30 | Dr-Linux | ok |
12:57.06 | [TK]D-Fender | Dr-Linux : Not sure on your 2nd cards timing, but go verify the PRI's first |
12:57.20 | hongtien | ie: Asterisk didn't act Media Gateway |
12:58.09 | hongtien | How many concurent call can make on that |
12:58.11 | hongtien | ? |
12:59.46 | *** part/#asterisk hongtien (n=hongtien@203.162.100.12) |
13:00.37 | *** join/#asterisk hongtien (n=hongtien@203.162.100.12) |
13:00.50 | *** part/#asterisk hongtien (n=hongtien@203.162.100.12) |
13:02.45 | Dr-Linux | [TK]D-Fender: what's possible things that cause of "RED" alarm? |
13:03.31 | [TK]D-Fender | Dr-Linux : Nothing connected to the port. Use your imagination and test what I jsut told you to. |
13:04.07 | Dr-Linux | [TK]D-Fender: ok thanks |
13:04.10 | QuAtRo[NL] | Problem of Webboarder is solved |
13:05.57 | _4d4m_ | i've just been reading around about options for LCR on *. I'm unsure what I need to use.. Should I run MySQL or execute it through AGI? 4 points of termination max, and dont envisage altering this structure for a while. |
13:06.15 | _4d4m_ | any assistance appreciated |
13:08.00 | speedwagon | _4d4m_, I would use your dialing rules, maybe use something like dialparties.agi But your dialing rules can do this for you. |
13:08.03 | *** join/#asterisk acrg (n=aragon@decoder.geek.sh) |
13:08.06 | acrg | hi |
13:08.16 | speedwagon | morning everyone |
13:08.22 | inv_Arp | ~seen ariel |
13:08.42 | jbot | ariel <n=kvirc@host253.200-82-113.telecom.net.ar> was last seen on IRC in channel #debian, 47d 11h 39m 37s ago, saying: 'Does anyone know how to capture my current X session to any video format ?'. |
13:08.52 | acrg | can anyone confirm that sending/receiving faxes works with asterisk and a sangoma pri card ? |
13:09.15 | speedwagon | inv_Arp, how are you doing? |
13:09.16 | acrg | from spandsp |
13:09.43 | inv_Arp | speedwagon: ello... |
13:09.45 | [TK]D-Fender | acrg : SpanDSP seems to. Fax machines behind a channel-bank are "OK" |
13:10.09 | speedwagon | inv_Arp, also the nick is ariel_, or me speedwagon... or abatista one and the same.. |
13:10.18 | inv_Arp | ahh ok... lol |
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13:10.57 | acrg | Fender thanks |
13:10.59 | inv_Arp | been lookin through my career builder and I thought I seen something you might like... |
13:11.05 | speedwagon | acrg, it should I have a customer using a sagoma pri card. |
13:11.18 | speedwagon | inv_Arp, OK |
13:11.38 | acrg | speedwagon and they send faxes over it with something like asterfax ? |
13:12.23 | *** join/#asterisk Dovid (n=none@barak.cellcom.co.il) |
13:12.35 | Dovid | Can anyone help me with a meetme issue ? |
13:13.11 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
13:13.29 | *** join/#asterisk speedwagon (n=Ariel@70.46.87.158) |
13:13.41 | *** part/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
13:13.55 | Dovid | ? |
13:14.32 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
13:15.10 | speedwagon | inv_Arp, sorry network issues. |
13:15.12 | [TK]D-Fender | ~suggestions |
13:15.16 | jbot | [suggestions] 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite and patient. 6) ... |
13:15.40 | Dovid | Why does meetme dump the call if the extension enterd is in valid (I.e. Exten _5xxx,1,Meetme) |
13:16.18 | [TK]D-Fender | Dovid : Pastebin the code that isn't working, and evidence of its failure in CLI. |
13:16.51 | [TK]D-Fender | Dovid : And if its dialplan related, the entire context and linked contexts. |
13:17.56 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
13:18.12 | Dovid | K, on dial up. Give me a sec :) |
13:20.27 | Dovid | http://pastebin.com/718381 |
13:22.14 | *** part/#asterisk acrg (n=aragon@decoder.geek.sh) |
13:22.34 | [TK]D-Fender | Dovid : Sorry, can't help with realtime... |
13:22.43 | Dovid | k |
13:23.01 | Dovid | Can u take a guess as to y it dumps the call ? |
13:24.07 | *** join/#asterisk Faithful (n=Faithful@202.6.145.116) |
13:25.28 | Dovid | ${EXTEN}:1 will subtract the first number from the exten ? |
13:25.35 | tzanger | no |
13:25.38 | tzanger | ${EXTEN:1} will |
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13:26.59 | Dovid | thx |
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13:56.16 | *** join/#asterisk kshumard_ (n=kshumard@gateway.digium.com) |
13:56.38 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
13:57.00 | Katty | morning |
13:59.30 | *** join/#asterisk Pointy (n=chowell@brain.xilogix.net) |
13:59.35 | *** join/#asterisk bkervaski (n=bkervask@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
13:59.55 | bkervaski | Hi all. Is it possible to have a voicemail delivered to multiple mailboxes without copying the file manually with a script? i.e., in extensions.conf? |
14:01.11 | Pointy | anyone have success using the follow-me option in * version 1.2.9.1? Having an issue where the timeout seems to be ignored. |
14:01.13 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
14:01.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:03.28 | Hmmhesays | Morning |
14:03.44 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
14:03.47 | PakiPenguin | hello everyone |
14:04.01 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
14:05.34 | [TK]D-Fender | Katty : Mew. |
14:05.58 | [TK]D-Fender | bkervaski : Yes, its all part of the Voicemail application |
14:06.16 | [TK]D-Fender | bkervaski : go read the instructions again |
14:06.42 | *** join/#asterisk JoseBravo (i=JoseBrav@200.119.32.47) |
14:07.21 | Hmmhesays | We saw a kickass live band this weekend |
14:07.28 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.141.Dial1.SanJose1.Level3.net) |
14:08.18 | Dovid | Using meetme in real time with _5XXX if invalid room is enterd meetme dups the call. Anyone know y ? |
14:08.44 | JoseBravo | I have a sip peer for outgoing calls that use g729 codec, and my sip clients use x-lite that don't have g729 codec. Then when I tried to do a call, I get this message: channel.c:2693 ast_channel_make_compatible: No path to translate from SIP/dominet-f844(256) to SIP/70103-6092(4). Where dominet is my provider and 70103 is the client. |
14:08.54 | Ahrimanes | http://bash.org/?654797 |
14:08.59 | [TK]D-Fender | Dovid : Maybe because the room is invalid.. you should try to avoide passing applications garbage values you know.... |
14:09.20 | Dovid | Well the room is invalid |
14:09.22 | [TK]D-Fender | JoseBravo : Did you pay for licenses at Digium.com? |
14:09.54 | Katty | [TK]D-Fender: mew. |
14:10.06 | vader-- | is it normal to see asterisk say B-channel 0/23 successfully restarted on span 1 |
14:10.06 | Dovid | Client enters *5000 to start a room. Problem is if a client of his calls and enters the wrong room he gets dumped. I am tryin to send em back to the main menu instead |
14:10.10 | vader-- | for all the channels |
14:10.13 | vader-- | every so often |
14:10.14 | Katty | [TK]D-Fender: my company is moving. |
14:10.18 | Katty | [TK]D-Fender: and i get my own office! |
14:10.21 | Katty | [TK]D-Fender: with a door! |
14:10.24 | file | vader--: that's normal |
14:10.24 | JoseBravo | [TK]D-Fender I need to pay licences? |
14:10.29 | drray | good morning fender |
14:10.29 | Dovid | yay |
14:10.35 | Dovid | yes |
14:10.38 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
14:10.41 | Dovid | To digium $10.00 per channel |
14:10.43 | Katty | file: i get my own office :>>>> |
14:10.55 | file | Katty: yay!!! |
14:11.00 | Katty | i know. |
14:11.06 | Katty | i feel all speshul. |
14:11.15 | file | and now... I run away! |
14:11.23 | [TK]D-Fender | Katty : Cool... |
14:11.53 | JoseBravo | I need to pay $10 per use sip peer to aoutgoing calls? |
14:11.54 | Dovid | [TK]D-Fender: If room is invalid it automaticly dumps the call ? Can I set it to go to the next pri. In the dial plan ? |
14:12.14 | Dovid | yes |
14:12.17 | [TK]D-Fender | JoseBravo : G.729 is a patented codec and * will not translate it without paying a licensing fee. |
14:12.20 | Katty | file: byebye |
14:12.25 | Katty | file: CALL ME |
14:12.30 | Dovid | It works 2 ways. Pretty much one channel per concurent call 2 eays |
14:12.44 | *** join/#asterisk jeremib (n=netnameu@c-71-203-209-162.hsd1.tn.comcast.net) |
14:13.10 | Dovid | [TK]D-Fender: did u get my last ? |
14:14.05 | znoG | did anyone else get spammed by LisaZhang? |
14:14.12 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
14:14.13 | znoG | from Telecomoutsourcing in China or something |
14:14.16 | Dr-Linux | Jun 19 07:13:00 NOTICE[10627]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
14:14.16 | Dr-Linux | <PROTECTED> |
14:14.18 | Dovid | For which list ? |
14:14.25 | bkervaski | Thanks, Fender. |
14:14.28 | bkervaski | Helpful as always. |
14:14.29 | Dr-Linux | why can't i dialout via PRI line? :S |
14:15.13 | jeremib | how can I troubleshoot my iax2 trunks not being listed in "iax2 show registry"? I have three different register= lines in my iax config file connected to 3 different servers. When i do iax2 reload it shows it parsed the file, but if i do iax2 show regstry nothing is listed |
14:15.18 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
14:15.28 | jeremib | and obviously my iax2 connections aren't being made |
14:15.40 | Dovid | Iax2 debug |
14:15.48 | *** join/#asterisk feld (n=feld@66-188-108-178.static.mdsn.wi.charter.com) |
14:15.55 | [TK]D-Fender | Dovid : nO IDEA. |
14:15.56 | file | jeremib: or make sure they're in the right place in iax.conf |
14:16.14 | Dovid | Thanks tk |
14:16.23 | [TK]D-Fender | Dr-Linux : wHATS THE LINE STATUS? |
14:16.40 | JoseBravo | Dovid no other solution for not pay per channels? |
14:16.50 | [TK]D-Fender | JoseBravo : Stop using G.729 |
14:16.56 | jeremib | file... you're a genious :) |
14:17.01 | Dovid | nope |
14:17.13 | Dovid | Unless u just pass thru |
14:17.18 | jeremib | i'm using freepbx, and it was loading my _custom.conf file before the _additional.con file, which i guess was messing it up |
14:17.23 | Dovid | Tk: You are using caps LOCK |
14:17.24 | [TK]D-Fender | Dovid : Which he can't... |
14:17.29 | jeremib | switched it around and it worked |
14:17.31 | jeremib | thanks!! |
14:17.33 | JoseBravo | [TK]D-Fender but its depends for my provider. |
14:17.43 | [TK]D-Fender | Dovid :WhAt ArE YoU TaLkInG AbOuT?!?! |
14:17.51 | Dovid | hehe |
14:17.55 | PakiPenguin | Dr-Linux, got a pri ? |
14:18.06 | file | [TK]D-Fender: Hi TkDeFeNdEr HoW aRe YoU? |
14:18.19 | [TK]D-Fender | JoseBravo : They don't support G.711? I'd be surprised. |
14:18.30 | znoG | can anyone get to www.freshmeat.net? it gets content from falkag.net which doesn't seem to respond, quickly |
14:18.33 | bkervaski | Hey tk: give me a hint on the voicemail->multiple mailbox delivery.. coming up short.. probably not sure what to search for.. I've looked through voicemail.conf, no dice.. (* 1.2.x) |
14:19.08 | [TK]D-Fender | file : Feeling colourful |
14:19.14 | file | [TK]D-Fender: uh oh |
14:19.37 | [TK]D-Fender | bkervaski : " |
14:19.45 | [TK]D-Fender | bkervaski : "show application voicemail" |
14:19.46 | Dr-Linux | PakiPenguin: yes |
14:19.56 | bkervaski | Thanks. |
14:19.57 | PakiPenguin | Dr-Linux, ptcl ? cool ? what for ? |
14:20.50 | bkervaski | Sometimes, it's just too easy to fathom... Thanks, TK |
14:21.06 | [TK]D-Fender | bkervaski : SCARY isn't it :) |
14:21.13 | Dr-Linux | PakiPenguin: not here, i got PRI in USA |
14:21.16 | Dr-Linux | PTCL sux |
14:21.28 | PakiPenguin | yup ;) |
14:21.32 | JoseBravo | [TK]D-Fender g723 will help to me? |
14:21.33 | PakiPenguin | it does! |
14:22.06 | *** part/#asterisk jeremib (n=netnameu@c-71-203-209-162.hsd1.tn.comcast.net) |
14:22.09 | Dr-Linux | PakiPenguin: PakiPenguin look here what PTCL did > www.syednetworks.com |
14:23.26 | *** part/#asterisk clive- (n=pirch@dsl-165-165-00.telkomadsl.co.za) |
14:25.35 | JoseBravo | [TK]D-Fender g723 will help to me? |
14:26.25 | *** join/#asterisk copland (n=stonecol@209.216.65.10) |
14:26.42 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
14:26.43 | copland | is there a way to have a h323 ext in asterisk |
14:27.37 | JoseBravo | I need pay licences for use g723? |
14:27.52 | file | JoseBravo: Asterisk currently can't transcode G723.1 |
14:28.32 | RoyK | JoseBravo: I beleive a g.723.1 license starts at about $100k or so |
14:28.41 | *** join/#asterisk ruza (n=ruza@81.0.238.58) |
14:29.48 | JoseBravo | My provider says that g711 use much bandwidth |
14:31.10 | JoseBravo | Thats right? |
14:31.13 | file | yes |
14:31.30 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
14:32.57 | Hmmhesays | Someone needs to read some more |
14:33.18 | [TK]D-Fender | JoseBravo : No, even worse |
14:33.49 | [TK]D-Fender | JoseBravo : So go pay for a lincense or two from Digium. |
14:34.12 | [TK]D-Fender | Hmmhesays : EVERYBODY needs to read more. Just most more than some ;) |
14:34.28 | Hmmhesays | true |
14:34.28 | *** join/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com) |
14:34.33 | file | [TK]D-Fender: Professor [TK]D-Fender, how much do I need to read?!? |
14:36.00 | harpermood | Can anyone help me with a connection problem.... I have a digium TE110P and am attempting to connect with a Channelized T1 with 6 channels using E&M signalling with wink |
14:36.12 | file | oooooooh |
14:36.59 | *** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net) |
14:37.36 | JoseBravo | [TK]D-Fender ok, im purchasing |
14:39.36 | *** join/#asterisk ToyMan (n=stuq@74-32-6-50.dsl1.mdl.ny.frontiernet.net) |
14:40.10 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.141.Dial1.SanJose1.Level3.net) |
14:40.16 | *** part/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com) |
14:41.07 | Dr-Linux | anybody have a look >> http://pastebin.com/718487 |
14:41.10 | Dr-Linux | not sure why i'm getting this |
14:41.39 | Hmmhesays | uclinux is not fun |
14:41.55 | [TK]D-Fender | Dr-Linux : And is the port still red? |
14:42.01 | *** join/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com) |
14:42.02 | SplasPood | hrm, if i wanted to test what will become 1.4, whats the current best route? |
14:42.23 | Dr-Linux | [TK]D-Fender: i'm getting this from "OK" port |
14:42.24 | [TK]D-Fender | SplasPood : Install chan_fluxcapacitor |
14:42.38 | SplasPood | [TK]D-Fender: :P You know what I meant ;) |
14:43.08 | [TK]D-Fender | Dr-Linux : I believe 34 is a notice that the remote end is busy |
14:43.15 | Dr-Linux | [TK]D-Fender: for not i forgot other 3 spans, but if it should dialout via 1 span, but it gives me this error |
14:43.24 | *** part/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com) |
14:43.42 | [TK]D-Fender | SplasPood : And you know what I meant, thus is symmetry attained! |
14:44.56 | file | SplasPood: wait until the beta is out |
14:45.24 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
14:45.34 | [TK]D-Fender | file : Whats the ETA looking like for it these days? (R, not RC) |
14:46.19 | Dr-Linux | [TK]D-Fender: by dialing any number, i get the same "Notice" |
14:46.24 | Dr-Linux | Jun 19 07:37:52 NOTICE[10694]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
14:46.24 | Dr-Linux | <PROTECTED> |
14:46.55 | [TK]D-Fender | Dr-Linux : Maybe you should pastebin the relevent parts of you config, and not jsut the 1 line error.... |
14:47.14 | Katty | [TK]D-Fender: corporate made a terrible, terrible decision today. |
14:47.24 | [TK]D-Fender | Katty : rut roh.... |
14:47.27 | Katty | [TK]D-Fender: they want me to take down the asterisk box and put up a vonexus box. |
14:47.33 | Katty | [TK]D-Fender: because microsoft is /clearly/ better. |
14:47.42 | Katty | [TK]D-Fender: save me :< |
14:48.23 | file | [TK]D-Fender: this year! |
14:48.25 | tzanger | Katty: put it on a 10mbit hub with a p2p client |
14:48.35 | pjo | lol |
14:49.17 | *** join/#asterisk harpermood (n=harpermo@66-215-122-11.dhcp.atsc.ca.charter.com) |
14:49.32 | [TK]D-Fender | Katty : And why would they want such a thing? |
14:50.03 | Dr-Linux | [TK]D-Fender: i have nothing more then this >> http://pastebin.com/718500 |
14:50.13 | harpermood | I need to setup my phone system for my company... got my TE110P card on Saturday, but can't seem to dial out or in |
14:50.17 | Dr-Linux | this time i pasted while trying different number |
14:50.46 | [TK]D-Fender | Dr-Linux : perhaps you aren't awake yet... I said PASTE YOUR CONFIG FOR IT |
14:50.53 | [TK]D-Fender | </subtle> |
14:51.03 | Dandan | [TK]D-Fender u r being sooo nice today :) |
14:51.04 | Dr-Linux | ok |
14:51.08 | Dandan | is it your birthday? :D |
14:51.14 | Katty | [TK]D-Fender: they seem to think that since i'm an mcp, working on my mcse, and because sell microsoft products.....that we should be using them. |
14:51.33 | Dandan | there is nothing wrong with mcse and asterisk |
14:51.46 | *** join/#asterisk anonymouz666 (i=anonymou@200.218.196.5) |
14:51.52 | file | Katty: how works your Asterisk install right now anyway? |
14:52.03 | Katty | file: it works. |
14:52.08 | [TK]D-Fender | Katty : and MS hosts on Linux servers... go figure... your admins are looking to waste money for nothing... what you REALLY need to do it ditch that hybrid channel bank BS you've got and just go PRI. |
14:52.08 | Katty | file: it's a neat as a button too |
14:52.33 | Katty | [TK]D-Fender: corporate doesn't share my views. |
14:52.35 | Dandan | [TK]D-Fender: yeah, and their experts.microsoft.fr has been hacked into and defaced :) |
14:52.36 | Dr-Linux | [TK]D-Fender: here >> http://pastebin.com/718506 |
14:52.36 | *** join/#asterisk wunderkin (n=wunderki@69.26.192.234) |
14:52.46 | Katty | [TK]D-Fender: no surprise there tho (= |
14:52.47 | Dandan | that's what you get for hosting stuff on windows... :) |
14:53.01 | Dandan | over the weekend :) |
14:53.22 | harpermood | Is there anyone that can help me with my TE110P problem.. I am really stuck. |
14:53.39 | Dandan | harpermood: as I am with sangoma and zapata.conf :) |
14:53.52 | Dandan | and even asterisk-users can't help :/ |
14:55.25 | [TK]D-Fender | Dr-Linux : that isn't your config... thats just the dialpland and I can SEE what its dialing... I want to see what its USING. |
14:55.26 | Katty | [TK]D-Fender: in other mews, my banking is getting better :> |
14:55.53 | Katty | file: wrong banking, deary. |
14:56.01 | Dr-Linux | [TK]D-Fender: you wanna see zap configs? |
14:56.10 | *** join/#asterisk salviadud (n=ralfalfa@201.145.29.99) |
14:56.13 | [TK]D-Fender | Katty : Laws of Star Trek : "Bank left, lurch right" |
14:56.27 | [TK]D-Fender | Dr-Linux : ..... perhaps some coffee is in order.... |
14:56.36 | Katty | [TK]D-Fender: lurch? |
14:56.51 | Dr-Linux | [TK]D-Fender: http://pastebin.com/718048 |
14:57.04 | harpermood | I am on the phone now with Digium support... |
14:57.13 | harpermood | They are having phone system troubles HAH ;) |
14:57.30 | [TK]D-Fender | Katty : a jerking lean. |
14:58.26 | file | harpermood: yeah... we are... |
14:58.28 | [TK]D-Fender | Dr-Linux : Dunno... verify that your lines aren't all in use, try calling those #'s from other lines to confirm they're OK... |
14:58.52 | Dr-Linux | [TK]D-Fender: okey thanks |
14:59.33 | harpermood | Phone system troubles happen.. it is just ironic, and a bit embarrassing, I would guess !! |
14:59.56 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
15:00.34 | Dandan | I am looking for a GOOD knowledge of PRI for hire for ONE (SMALL?) glitch? |
15:00.38 | harpermood | so, file, if I am in the queue, listenening to some fantastic music, how long might I expect to stay there? |
15:00.39 | Dandan | blah |
15:00.44 | Dandan | *expert of course |
15:00.54 | file | harpermood: less then a year |
15:02.12 | harpermood | got someone!! |
15:02.12 | harpermood | it has been a short year :) |
15:02.32 | JoseBravo | I have a problem registering my g729 codec, because my interface is not eth0. Then I can't register my codec? |
15:02.58 | file | JoseBravo: are you having an actual problem or are you just asking |
15:03.26 | Katty | [TK]D-Fender: what's that? |
15:03.31 | Katty | [TK]D-Fender: that does not parse. |
15:03.32 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
15:04.38 | [TK]D-Fender | Katty : Picture getting cross-checked by an NFL quarterback, then you'll understand "lurch".... |
15:04.55 | [TK]D-Fender | Katty : Or try http://dictionary.reference.com/browse/lurch |
15:07.02 | JoseBravo | file I have an actual problem. |
15:07.13 | JoseBravo | file Unable to determine hostid. You must have at least one ethernet card |
15:07.39 | [TK]D-Fender | JoseBravo : You don't have a NIC in that box? |
15:07.50 | file | what OS and what's the interface name? |
15:08.34 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
15:08.59 | JoseBravo | The OS is Fedora Core 4. But I patched the kernel with OpenVZ. Then Its a VPS. The interface name is venet0:0 |
15:09.10 | file | interesting |
15:10.00 | Ahrimanes | hm maybe the : that confuses it? |
15:11.11 | JoseBravo | The alias of the first |
15:11.12 | JoseBravo | <PROTECTED> |
15:11.12 | JoseBravo | <PROTECTED> |
15:11.12 | JoseBravo | <PROTECTED> |
15:11.29 | JoseBravo | Sorry for the flood, it wasn't my intension. |
15:11.40 | file | yeah I'm just trying to think of how you can make it work |
15:11.56 | JoseBravo | Thanks file |
15:12.10 | *** part/#asterisk harpermood (n=harpermo@66-215-122-11.dhcp.atsc.ca.charter.com) |
15:12.18 | [TK]D-Fender | file : Alias it to ETH0 ? |
15:12.47 | Ahrimanes | or look at the freebsd reg util? it doesnt look for eth0.. |
15:12.59 | JoseBravo | I can't create new virtual interfaces into VPS |
15:13.10 | file | registration utility is the same, just compiled for FreeBSD |
15:14.07 | Ahrimanes | hm but eth0 is not present on freebsd... |
15:14.14 | file | it uses a different method to grab the info |
15:14.35 | file | really think the API calls would be the same between Linux and BSD? :D |
15:16.13 | lunk | man asterisk is so much faster without all that A@H overhead |
15:17.00 | *** join/#asterisk feld (n=feld@12.148.212.157) |
15:17.09 | mitcheloc | lunk: go post that on the tb forums ;) |
15:17.15 | mitcheloc | evangelize! |
15:17.22 | *** join/#asterisk mzeltner (n=eaon@62.96.102.155) |
15:17.24 | *** join/#asterisk akke (n=akke@85.158.211.235) |
15:17.25 | lunk | tb? |
15:17.43 | mitcheloc | aah=tb |
15:18.04 | file | mzeltner: hope you're having a good time |
15:18.08 | lunk | oh |
15:18.08 | mitcheloc | mzeltner: that's just mean! |
15:18.14 | lunk | that new project with the stupid name? |
15:18.17 | akke | anyone here can offer flat fee SIP/IAX dial-out to belgium landlines? |
15:18.18 | mitcheloc | yes |
15:18.23 | lunk | the name is the reason i stopped using it |
15:18.31 | mitcheloc | good for you |
15:18.34 | lunk | can't go into a client's office with a 'trixbox' |
15:18.45 | *** part/#asterisk cods (n=cods@tuxee.net) |
15:18.53 | mitcheloc | well.... there are worse names forprojects |
15:19.07 | lunk | Ms Bob? |
15:19.17 | mzeltner | file: Yeah, it's nice, would've expected more people though - maybe tomorrow |
15:19.19 | mitcheloc | like yahoo or google |
15:19.29 | [TK]D-Fender | mitcheloc : Another sign of * GUI underacheivement ;) |
15:19.40 | file | mzeltner: well it's spread out across three places, so everyone doesn't have to go to one location... |
15:19.47 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
15:19.49 | mitcheloc | [TK]D-Fender: what are you referring to? |
15:21.06 | [TK]D-Fender | mitcheloc : concerning their failure to have come up with the WORST possible name for TrixBox :) |
15:21.56 | mitcheloc | ahh, well, name's rarely matter for products/projects as long as they are solid |
15:22.06 | rene- | it is pretty lame tho |
15:22.13 | rene- | (the name) |
15:22.13 | mitcheloc | the thing that kicks me is that you can't use "asterisk" in your project name |
15:22.18 | mitcheloc | *** or aren't supposed to |
15:22.26 | [TK]D-Fender | rene- : its flaming |
15:23.09 | mitcheloc | [TK]D-Fender: do you have any name suggestions? i'm working on a gui, and i need a name |
15:23.34 | *** join/#asterisk dgilmore (n=dennis@fedora/dgilmore) |
15:23.46 | rene- | i was going to work on a GUI and i was going to name it Masterisk but i dont know when i am going to ship this if ever so if you like it |
15:24.14 | mitcheloc | rene-: what kind of gui? |
15:24.22 | *** mode/#asterisk [-o file] by ChanServ |
15:24.28 | mitcheloc | M* doesn't really give it away ;) |
15:24.29 | MikeJ__ | heh |
15:24.35 | MikeJ__ | file you got -o'd |
15:24.40 | file | ChanServ restarted ;( |
15:24.59 | rene- | just a front for Asterisk Realtime Mysql tables |
15:25.26 | rene- | but i found a really good one what it is already in my language of choice so, i much better use that one |
15:25.33 | mitcheloc | that's a ton of work =/ |
15:25.46 | rene- | you can find it under rubyforge telephony apps |
15:25.52 | rene- | it is very well done |
15:26.05 | Dr-Linux | file: question, what i suppose to get, if PRI cable is not connect and i try do dialout? |
15:26.07 | mitcheloc | [TK]D-Fender: i'd still love to hear your name suggestion ;) |
15:26.09 | akke | anyone here can offer flat fee SIP/IAX dial-out to belgium landlines? |
15:26.32 | file | Dr-Linux: out of a zaptel channel? no clue |
15:27.01 | Dr-Linux | file: in FXO case, it makes me conneted to the channel, even phone line is not connected |
15:27.31 | file | FXO case is different, for a PRI it would know it's in red alarm and probably refuse to allocate a channel |
15:27.31 | Dr-Linux | currently i gives me "notice" |
15:27.32 | Dr-Linux | <PROTECTED> |
15:27.35 | file | with whatever reason code |
15:27.42 | tomtom_ | so nobody can help me with philipine did's? |
15:28.17 | Dr-Linux | file: but my first span is not "RED" it's "OK" |
15:28.31 | file | Dr-Linux: well you never told me what you tried to dial out on |
15:29.00 | [TK]D-Fender | mitcheloc : Pick something vaguely normal is business-like like "QuickPBX", "InstaPhone", or something catchy that doesn't sounds some gay punk created it in his basement... |
15:29.33 | Dr-Linux | file: have a look > http://pastebin.com/718048 |
15:30.01 | Dr-Linux | and |
15:30.01 | mitcheloc | [TK]D-Fender: so i take it, you don't like "druid" (the name of an * web gui)? |
15:30.09 | Dr-Linux | file: i'm trying this >> http://pastebin.com/718506 |
15:31.03 | file | I would go there except it's not loading |
15:31.07 | file | ah there we go |
15:31.09 | [TK]D-Fender | mitcheloc : Cheap twist on "Wizard" (TM'd? that'd be stupid), and the entire concept of quick setups as "wizardry" is what perpetuates stupid users. Then again, thats what GUI's do so.... hrm |
15:31.24 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
15:31.41 | file | you're dialing out on group 3, which I assume is down |
15:31.50 | *** join/#asterisk momelod (n=momelod@HSE-Montreal-ppp133997.qc.sympatico.ca) |
15:31.53 | asterboy | dawson |
15:31.56 | momelod | hello peoples |
15:32.04 | file | well, down in that all the spans part of the group are down |
15:32.15 | Dr-Linux | file: when i dialout throug group 1 , it gives me same busy/congestion notic though |
15:32.16 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
15:33.02 | mitcheloc | [TK]D-Fender: i'm not keen on gui's either for configuring asterisk, however some people would argue that modern software should have gui's, so who knows on that one, that's why i like to work on guis for end users and not administrators |
15:33.30 | file | Dr-Linux: so try to dial out an exact channel on the span that's up and see |
15:33.42 | file | and try out the group that supposedly has a span that's up, and pastebin it too |
15:33.50 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
15:33.56 | momelod | i've got a question about my zaptel card. I have been experiencing echo on my telephone when i call out on a zap interface. The person im calling doesnt hear the echo, only i hear my own voice echoed back to me on my reciever.. I used ztmonitor as described here: http://www.voip-info.org/wiki/view/Asterisk+PSTN+interface+debugging and saw that my transmit signals are way too high. How can i manipulate the transmit signal stren |
15:34.20 | Dr-Linux | file: ok, wait |
15:34.30 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
15:34.35 | feld | momelod: zapata.conf i believe in your asterisk folder |
15:34.41 | feld | you can do rxgain and txgain settings |
15:34.51 | momelod | thanx |
15:34.52 | feld | s/folder/directory/ |
15:35.00 | rene- | mitcheloc: the way i see it, asterisk is a pbx toolkit, and it you are using a subset of it for specific functionality then a GUI is a must, since you cant realistically expect you users to be fluent in technologies like asterisk and linux, so a GUI is the way to go for specific purpose apps, and the name should let people know what the app is about |
15:35.14 | feld | nobody says "folder" except when they're using windows... which i do too often....... |
15:35.49 | coppice | i thought a folder was a dot com |
15:36.12 | asterboy | gui for asterisk? fuck that |
15:36.12 | coppice | a manila folder is surely a bankruptcy in the philipinnes |
15:36.41 | Dr-Linux | file: here i dialout through group 1, but same >> http://pastebin.com/718578 |
15:36.57 | mitcheloc | rene-: well, for specific purpose apps yes, but general purpose, not so much.. and that's just my opinion |
15:37.06 | JoseBravo | How can I contact ro digium for fix my problem? |
15:37.13 | mitcheloc | asterboy: you would say no to a gui for users? |
15:37.16 | l-fy | JoseBravo > call at 500 |
15:37.17 | asterboy | pickup a phone and call them. |
15:37.24 | asterboy | yep |
15:37.39 | asterboy | no gui for users...for asterisk anyway. |
15:38.00 | mitcheloc | asterboy: by user's i mean people sitting at workstations...not admins |
15:38.35 | asterboy | maybe the flashop stuff would be nice. |
15:38.45 | mitcheloc | asterboy: would you rather be using telnet for irc =P |
15:40.21 | Dr-Linux | file: no clue? ... how can i dialout through s specific channel? exten => _91NXXXXXXXXX,1,Dial(Zap/1-1/${EXTEN:1}) |
15:40.24 | asterboy | a web interface is sufficient...but then I'd argue that it's not gui in that you don't have to run XWindows or something intensive on resources |
15:40.27 | Dr-Linux | is that correct? |
15:40.29 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
15:40.34 | file | Dr-Linux: Zap/1/blah |
15:40.46 | akke | anyone here can offer flat fee SIP/IAX dial-out to belgium landlines? |
15:40.58 | l-fy | akke > not me |
15:41.09 | pjo | akke: check voip-info |
15:41.11 | mitcheloc | asterboy: a "web interface" for asterisK? if so you can run it on a different machine (and should) |
15:41.23 | akke | ok |
15:41.24 | Dr-Linux | exten => _91NXXXXXXXXX,1,Dial(Zap/1/${EXTEN:1}) < is this correct? :S |
15:41.25 | mitcheloc | of coures it has to be programmed correctly |
15:41.30 | Dr-Linux | or i need to mentioned group as well? |
15:41.31 | asterboy | how do you mean run on different machine? |
15:41.41 | asterboy | different binaries? |
15:41.44 | akke | anyone tried voipcheap.com? supports SIP protocol but i wonder how usable it is ? |
15:41.45 | file | yes that's fine as far as I know, but as I said before... I don't do zaptel :P |
15:42.08 | mitcheloc | asterboy: i mean, lets say for a call log analyzer, like ast-stat, it just needs a mysql connection |
15:42.08 | Dr-Linux | lol |
15:42.17 | mitcheloc | asterboy: both pieces can be on a seperate server |
15:42.20 | asterboy | ah, yes...thats nice |
15:42.34 | Dandan | need an PRI expert for hire... please /msg me |
15:43.01 | pjo | akke: nope, but to save you some searching http://www.voip-info.org/wiki-VOIP+Service+Providers has quite a list. (and i think there are some user comments at the end) |
15:43.04 | mitcheloc | asterboy: the problem is most people won't bother setting up the second machine =/ |
15:43.19 | Hmmhesays | bah |
15:43.22 | Dr-Linux | aww |
15:43.23 | Dr-Linux | this time: |
15:43.24 | Dr-Linux | NOTICE[10843]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
15:43.24 | Dr-Linux | <PROTECTED> |
15:43.34 | mitcheloc | Hmmhesays: ? |
15:45.32 | *** join/#asterisk LH-euhost (n=LH-euhos@82.131.200.80.pool.invitel.hu) |
15:45.54 | Katty | Hmmhesays: the company is /finally/ moving! we're getting a new building and i'm getting my own office (= |
15:46.09 | Hmmhesays | cool |
15:46.14 | Hmmhesays | i'm still getting the shaft here |
15:46.19 | Katty | damn :< |
15:46.36 | Hmmhesays | ohwell |
15:47.15 | Dandan | need an PRI expert for hire... please /msg me |
15:47.33 | salviadud | the shaft? |
15:48.01 | Hmmhesays | Dandan: what a cryptic message |
15:49.42 | *** join/#asterisk catlee (n=catlee@Z-pc1-959-S1.gw2.tor1.rogerstelecom.net) |
15:49.45 | catlee | Good morning |
15:50.21 | SplasPood | Hey all, question.. Say I had a call listening to MusicOnHold() and then I wanted to command asterisk to drop MusicOnHold and instead bridge it to another channel.. possible? |
15:51.50 | catlee | If I wanted to use * to connect to my POTS at home, would things like call-waiting, and voicemail if the line is busy still work? |
15:56.10 | Dr-Linux | file: have a look, maybe it helps > http://pastebin.com/718611 |
16:00.24 | [TK]D-Fender | catlee : yes |
16:00.55 | catlee | How does voicemail work if the line is busy? |
16:01.00 | [TK]D-Fender | catlee : if by that you mean a phone connected to *. if the call never makes it to * because of something telco related well that out of its hands. |
16:01.25 | [TK]D-Fender | catlee : Depends on your definition of "line is busy". You referring to a PHONE on your system, or an incoming "line"? |
16:01.39 | catlee | well, let's say I have one incoming POTS line |
16:01.44 | catlee | and I'm using it to talk to somebody |
16:02.03 | [TK]D-Fender | catlee : then no. Forget call waiting on that line, and if its busy, TFB |
16:02.33 | catlee | right now, if somebody else calls me, I will be notified that there's another caller on the line, and if I don't answer they will be sent to my voicemail |
16:03.03 | catlee | ok, that's what I thought |
16:03.24 | catlee | but different with incoming calls via SIP/IAX? |
16:03.59 | [TK]D-Fender | catlee : depends. |
16:04.13 | JoseBravo | The mail support of digium is fast? |
16:04.56 | [TK]D-Fender | catlee : think of it in terms that your telco (whatever tech) HAS to be able to send the enxt call to *. That means and extra PSTN line or being permitted multiple VoIP calls simultaneously. Every call takes a line. |
16:05.51 | catlee | ah, so that's why VoIP providers list # of simultaneous calls? |
16:06.08 | [TK]D-Fender | catlee : In typical ITSP's you get to have 2 calls at a time. If you're using an analog phone you can switch/conference between them by use of callwaiting style functionality. On a SIP phone you may be able to treat them more naturally like in a normal PBX. |
16:06.14 | [TK]D-Fender | catlee : Yup... |
16:06.29 | [TK]D-Fender | Because if you want * to do the VM it obviously has to be the one answering that call. |
16:06.34 | catlee | yup |
16:07.09 | *** part/#asterisk faberk64 (n=faberk@213.199.15.249) |
16:07.22 | *** join/#asterisk faberk64 (n=faberk@213.199.15.249) |
16:11.22 | catlee | So that's why it may make sense to have voicemail provided by the telco |
16:11.55 | [TK]D-Fender | catlee : Nope... thats why it makes sense to have more lines than calls :) |
16:12.06 | [TK]D-Fender | catlee : Or feel safe in knowing people will call you back. |
16:12.24 | [koss] | are there any turn-key asterisk solutions i can buy to replace a PBX for about 50 phones and 10 telco lines? |
16:12.42 | dlynes_office | [TK]D-Fender: btw...just thought I'd let you know |
16:12.54 | dlynes_office | [TK]D-Fender: everything seems to be working fine (for the most part) with the a200d |
16:13.03 | dlynes_office | in a 2.6.15.5 kernel, too |
16:13.41 | dlynes_office | the only thing that doesn't seem to be working is the HWEC |
16:13.56 | catlee | [TK]D-Fender: that's probably not an option for many residential users, is it? |
16:14.03 | dlynes_office | Do I not use echocancel=yes in zapata.conf if i'm using the hwec? |
16:14.27 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
16:14.32 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
16:15.41 | akke | anyone here can offer flat fee SIP/IAX dial-out to belgium landlines? |
16:15.44 | Egonis | I am using a TDM400P w/ 2 FXS Cards, and I notice that I get lots of cracks/beeps during conversations -- when I load ALSA, it gets worse.. although i have noload => chan_alsa.so -- has anyone else experienced this? |
16:16.32 | dlynes_office | Egonis: what's your cpu load like? what kinda cpu are you using? |
16:16.32 | Egonis | when I try load => chan_oss.so and noload => chan_alsa.so and the reverse, it makes no difference |
16:16.45 | catlee | although it makes sense for businesses, and most businesses should already have enough lines |
16:17.28 | dlynes_office | catlee: probably not an option for residential users, no |
16:17.29 | Egonis | dlynes_office: 99.3% idle, PIII 1.0 |
16:17.40 | Dandan | Egonis: and your irq? |
16:17.46 | Egonis | Dandan: HOw do I find out? |
16:17.51 | dlynes_office | catlee: but then again, pbx is probably unnecessary for a residential user, too :) |
16:17.57 | Dandan | static is usually caused by irqs and buffer underruns |
16:18.04 | Dandan | egonis with stuff like mpstat |
16:18.07 | Dandan | afair |
16:18.17 | Egonis | Dandan: mpstat? |
16:18.28 | dlynes_office | Egonis: lspci -vv |
16:18.34 | dlynes_office | Egonis: cat /proc/interrupts |
16:18.34 | Dr-Linux | dlynes_home: hey :) |
16:18.37 | dlynes_office | hey |
16:18.40 | Dandan | yeaf, from sysstat package |
16:18.42 | Dandan | re dlynes_ |
16:18.43 | Dandan | re dlynes_office |
16:18.44 | Dandan | :) |
16:19.02 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
16:19.03 | Dandan | i offered some money for anyone willing to resolve my issue |
16:19.05 | Dandan | with PRI |
16:19.18 | dlynes_office | Egonis: oh..you're using bsd, not linux? |
16:19.31 | dlynes_office | Dandan: which problem was that again? |
16:19.33 | Egonis | dlynes_office: And what am I looking for in cat /proc/interrupts? I get: |
16:19.33 | Egonis | 193: 781748155 IO-APIC-level wctdm |
16:19.34 | catlee | dlynes_home: true :) I was thinking it would be cool to set up so I could phone my family using VoIP...And maybe even use each server as a termination point into its local area code |
16:19.40 | Dandan | PRI calls going out, not coming in |
16:19.42 | Egonis | dlynes_office: Gentoo Linux |
16:19.43 | Dandan | BLAH |
16:19.53 | Dandan | PRI calls *NOT* going out, but coming in... |
16:19.54 | dlynes_office | Dandan: ah...and did you get the problem fixed? |
16:19.55 | Dr-Linux | dlynes_home: my both TE21OP card has been succesfully installed. but can't dialout |
16:20.10 | Dr-Linux | dlynes_home: http://pastebin.com/718611 |
16:20.11 | dlynes_office | Dr-Linux: can peeps call in? |
16:20.22 | Dandan | dlynes_home: no, i just posted a follow up asking to step forward if you know PRI and I will pay you to have it resolved |
16:20.31 | dlynes_office | ah |
16:20.32 | Dr-Linux | dlynes_home: didn't check that yet, don't have numbers, still waiting for |
16:20.39 | Dr-Linux | dlynes_home: also have a look > http://pastebin.com/718048 |
16:20.55 | dlynes_office | Dr-Linux: if you don't have numbers yet, what makes you think the pri is even functional? |
16:21.14 | Dandan | dlynes_home: http://lists.digium.com/pipermail/asterisk-users/2006-June/156266.html |
16:21.24 | Dr-Linux | dlynes_home: i mean, i don't know number yet, but it should dialout |
16:21.24 | Egonis | dlynes_office: So what should I check next? |
16:21.28 | Dr-Linux | but i gives .. : |
16:21.49 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
16:21.49 | dlynes_office | Egonis: zttest |
16:21.51 | Dr-Linux | <PROTECTED> |
16:21.52 | Dr-Linux | <PROTECTED> |
16:22.09 | Egonis | dlynes_office: Should I shutdown asterisk prior? |
16:22.16 | dlynes_office | Dr-Linux: yeah...sounds to me like your pri is down |
16:22.21 | dlynes_office | Dr-Linux: type zap show status |
16:22.51 | Dr-Linux | dlynes_home: that should 1 span "OK" other 3 "Alarm" |
16:23.01 | dlynes_office | Dr-Linux: and what span are you trying to dial out on? span 1 or span 2? |
16:23.06 | Dr-Linux | sorry "RED" |
16:23.07 | dlynes_office | span 2 is in alarm |
16:23.15 | Dr-Linux | dlynes_home: span 1 |
16:23.20 | Dr-Linux | the "OK" one |
16:23.22 | dlynes_office | and both spans are down |
16:23.27 | dlynes_office | not up |
16:23.33 | Dr-Linux | Description Alarms IRQ bpviol CRC4 |
16:23.35 | Dr-Linux | T2XXP (PCI) Card 0 Span 1 OK 0 0 0 |
16:23.35 | Dr-Linux | T2XXP (PCI) Card 0 Span 2 RED 0 0 0 |
16:23.44 | dlynes_office | Dr-Linux: read your pri show span 1 |
16:23.51 | Dr-Linux | dlynes_home: the one i pasted in ? |
16:23.53 | dlynes_office | Status: Provisioned, Down, Active |
16:23.57 | Dr-Linux | yeah, you are right |
16:24.02 | dlynes_office | The 'down' means it's down |
16:24.06 | Dr-Linux | but why it showing "OK" the span 1 ? |
16:24.06 | dlynes_office | as in it's not operational |
16:24.13 | dlynes_office | Because it's not in alarm |
16:24.17 | Egonis | dlynes_office: Does zttest end? I am getting mostly 100.00% readings, but worst appear to be 85% |
16:24.30 | dlynes_office | iow, your eq and your telco's eq are talking to each other |
16:24.37 | dlynes_office | but you're not getting service |
16:24.45 | Dr-Linux | dlynes_office: can you explain a bit, "bcoz it's not in alarm" ? |
16:24.47 | dlynes_office | Egonis: it should run for about 2-3 minutes |
16:25.11 | dlynes_office | Egonis: it should run at 99.875% or higher |
16:25.13 | Egonis | dlynes_office: Okay, thanks.. should I specify for it to not run on the zap pseudo interface? should I force it to measure on a Wildcard Channel? |
16:25.14 | Dr-Linux | dlynes_office: what do you mean it's not in alarm? |
16:25.29 | Egonis | dlynes_office: How would I adjust it to run above or at that level? |
16:25.32 | dlynes_office | Egonis: iow, you're probably having irq issues |
16:25.38 | dlynes_office | Egonis: you don't |
16:25.42 | dlynes_office | Dr-Linux: hold your horses |
16:25.48 | dlynes_office | Dr-Linux: i've only got one keyboard |
16:25.53 | Egonis | dlynes_office: nice... any suggestions for fixing irq issues? |
16:26.00 | Dr-Linux | heh :) ok |
16:26.04 | dlynes_office | Egonis: how many pci slots do you have? |
16:26.20 | Egonis | dlynes_office: two, both full -- one is an IBM ServeRAID, the other is the TDM400P |
16:26.23 | *** join/#asterisk JoseBravo (i=JoseBrav@200.119.32.47) |
16:26.51 | dlynes_office | Egonis: ok, does your bios allow you to force slots onto certain irqs? |
16:26.58 | Egonis | dlynes_office: nope! :( |
16:27.06 | [TK]D-Fender | dlynes_office : Yes, you are supposed to use echocancel=yes for the HWEC. pastebin your wanpipe1.conf |
16:27.53 | dlynes_office | Egonis: yeah...the digiums are really picky about interrupts; the apic is only masking the problem |
16:27.55 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
16:28.05 | dlynes_office | Egonis: can you try the card in a different machine? |
16:28.17 | asterboy | ~echo |
16:28.19 | jbot | well, echo is an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting, or of ... |
16:28.21 | Egonis | dlynes_office: Theoretically, yes.. but with lots of work, this is my only asterisk box |
16:28.28 | Dr-Linux | dlynes_office: look for your quick and last answer, that's why i'm looking for then i need to run. |
16:28.43 | Egonis | dlynes_office: Is there a kernel option I should use? |
16:28.46 | dlynes_office | Dr-Linux: there's not much you can do...once you get your dids, try it again |
16:29.00 | dlynes_office | Dr-Linux: if it's still not working, then complain to your telco that your pri is not up yet |
16:29.12 | asterboy | I just fixed my digim blues. |
16:29.26 | Dr-Linux | dlynes_office: yes i understand that, but why first span is "OK" |
16:29.27 | dlynes_office | asterboy: you bought a sangoma? |
16:29.30 | asterboy | You really need a tight kernel, no acpi |
16:29.31 | Dr-Linux | that's my questin |
16:29.33 | dlynes_office | Dr-Linux: it's not ok |
16:29.35 | dlynes_office | Dr-Linux: it's down |
16:29.38 | asterboy | have sangoma as well |
16:29.38 | Dr-Linux | <dlynes_office> Because it's not in alarm << please explain it |
16:29.52 | Egonis | asterboy: So if I have crack/beep noises during calls, I should kill acpi? i.e. noacpi in kernel options? |
16:29.55 | Dr-Linux | dlynes_office: i know it's down, but it shows "OK" but other span not |
16:30.06 | asterboy | hell ya |
16:30.08 | dlynes_office | Dr-Linux: both spans are down, one is in alarm |
16:30.16 | dlynes_office | Dr-Linux: alarm means the other end is not connected to anything |
16:30.18 | Egonis | asterboy: And that can possibly fix irq issues? |
16:30.26 | asterboy | if zttest has 99.975 or lower ... you'll get that |
16:30.33 | asterboy | yep |
16:30.35 | Egonis | asterboy: Mine bounces from 100 to 85 |
16:30.41 | asterboy | 85? |
16:30.41 | dlynes_office | Egonis: yeah...sangoma has almost no issues with sharing interrupts |
16:30.43 | Dr-Linux | awwwwwwwwwwwwww nice answer :) |
16:30.43 | asterboy | yikes! |
16:30.51 | Egonis | dlynes_office: Yeeeah, I've heard |
16:30.52 | asterboy | you have a serious problem with 85 |
16:30.57 | Dr-Linux | dlynes_office: thanks, |
16:31.00 | Egonis | asterboy: I would tend to agree.. :) |
16:31.04 | asterboy | Egonis, follow this: |
16:31.06 | asterboy | ~echo |
16:31.07 | jbot | well, echo is an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting, or of ... |
16:31.16 | asterboy | especially the last entry |
16:31.16 | Dr-Linux | dlynes_office: what i suppose to get while dialing out, if my PRI cables are not connected? |
16:31.29 | dlynes_office | Dr-Linux: you can't dial out...your pri is down |
16:31.34 | JoseBravo | I need to go to take a coffe or go to vacations until Digium Support response my mails? |
16:31.37 | dlynes_office | Dr-Linux: what part of 'down' do you not understand? |
16:31.53 | asterboy | In BIOS I turned off every unnecessary option |
16:31.56 | Dr-Linux | dlynes_office: yes, i understand :) |
16:32.02 | dlynes_office | Egonis: ideally you shouldn't have acpi or apic enabled |
16:32.03 | asterboy | Sound, Para Port, Serial Ports, blah blah |
16:32.07 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
16:32.08 | Dr-Linux | dlynes_office: i'm asking things just for learning purpose |
16:32.12 | dlynes_office | Egonis: also, you should enable dma on your hard drives |
16:32.19 | dlynes_office | Dr-Linux: I've already told you everything |
16:32.28 | dlynes_office | [TK]D-Fender: http://pastebin.com/718701 |
16:32.30 | asterboy | Then compiled the kernel with no mouse, no everything I did not need. |
16:32.31 | Dr-Linux | dlynes_office: bcoz i used FXO not PRI, and they are kinda different |
16:32.44 | Dr-Linux | dlynes_office: ok thanks |
16:32.46 | dlynes_office | Dr-Linux: umm...you were showing me pri, not fxo |
16:32.46 | Dr-Linux | brb |
16:32.49 | asterboy | That brought my zttest scores within 5 9's |
16:33.01 | asterboy | now the system is working perfectly. |
16:33.18 | Dr-Linux | dlynes_office: yes, but in the FXO case if the phone line is not connected, it will still connect you to the channel, |
16:33.20 | asterboy | otherwise you get crack, drops, noice, bad dtmf |
16:33.24 | dlynes_office | JoseBravo: no, not usually |
16:33.37 | dlynes_office | JoseBravo: emailing them is generally not very efficient |
16:33.38 | Dr-Linux | dlynes_office: but PRI doesn't go this way :S |
16:33.42 | dlynes_office | JoseBravo: try calling them |
16:33.55 | dlynes_office | Dr-Linux: no it won't |
16:34.02 | dlynes_office | Dr-Linux: it'll tell you the line is congested also |
16:34.28 | JoseBravo | What is the digium support line? |
16:34.41 | Dr-Linux | dlynes_office: yes, correct, that's what i was trying to asking since 6 hours :) |
16:34.45 | dlynes_office | JoseBravo: it's on their website, but you can also call them via iax |
16:34.49 | [TK]D-Fender | asterboy : If you aren't 5-9's, you aren't using enough decimal places :D bound to be a few more 9's in the next hundred digits or so of decimal precision.... |
16:34.54 | Dr-Linux | i knew you will have an answer :) |
16:35.06 | dlynes_office | Dr-Linux: and i gave you an answer so get over it :p |
16:35.22 | Dr-Linux | dlynes_office: yeah, thanks |
16:36.05 | [TK]D-Fender | dlynes_office : Pastebin.com is being completely slow for me... can you use .ca.... |
16:36.20 | dlynes_office | [TK]D-Fender: yeah..one sec...going to have to find the ip address for it first |
16:36.30 | dlynes_office | [TK]D-Fender: stupid dns servers that use ipv6 anyways :p |
16:36.48 | CunningPike | Morning all |
16:36.54 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
16:37.44 | dlynes_office | morning, cp |
16:38.12 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
16:38.29 | CunningPike | Hey, dlynes_office |
16:39.07 | akke | anyone here can offer flat fee SIP/IAX dial-out to belgium landlines? |
16:39.16 | dlynes_office | [TK]D-Fender: http://159.18.52.69/index.php |
16:39.22 | dlynes_office | erm |
16:39.32 | dlynes_office | [TK]D-Fender: http://159.18.52.69/67245 |
16:39.49 | terrapen | I wonder, is there any way to set up Asterisk server roaming for softphone users? Like, where they get the local asterisk server for the office that they happen to be in at the time |
16:39.58 | terrapen | I suppose it could be done with DNS but that's kind of ugly |
16:40.23 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
16:40.43 | dlynes_office | terrapen: Yeah, you could set something like that up with an ettercap or ethereal parser, or something like that |
16:40.44 | [TK]D-Fender | dlynes_home : Has Sangoma logged in to confirm the DSP is up and running? |
16:40.51 | dlynes_office | [TK]D-Fender: nope |
16:40.57 | terrapen | what does ethereal have to do with it? |
16:41.06 | dlynes_office | [TK]D-Fender: the logs all indicate the hwec is up |
16:41.23 | dlynes_office | [TK]D-Fender: however, I get an error during bootup |
16:41.32 | [TK]D-Fender | terrapen : set them to use DNS at whatever site they are at. |
16:41.49 | *** join/#asterisk zamba (i=marius@2001:730:5:21e0:0:0:0:1) |
16:42.06 | dlynes_office | terrapen: yeah...another way you can do it too, is if the phones support SRV entries in the DNS |
16:42.08 | terrapen | d-fender, yeah, that's the only solution i can come up with...i was wondering if there was something more elegan |
16:42.09 | terrapen | t |
16:42.37 | zamba | i'm planning to set up a voicemail service in an already running telephony environment.. i want one of the extensions to point to an asterisk server.. what kind of hardware do i need for this? |
16:42.37 | dlynes_office | terrapen: srv entries are more elegant, but not all sip phones support them |
16:42.38 | [TK]D-Fender | terrapen : Slightly less elegent, but you could just set up 2-3 peers and let them hand-pick. |
16:43.04 | [TK]D-Fender | zamba : Depends how you want to interface the 2 systems. |
16:43.13 | zamba | [TK]D-Fender: what are my alternatives? |
16:43.24 | [TK]D-Fender | zamba : Typically you'd use analog ports, or preferably T1/E1 |
16:43.37 | terrapen | I'm probably going to be using X-LITE and Ekiga |
16:43.39 | zamba | E1 is the same as ISDN, right? |
16:43.41 | [TK]D-Fender | zamba : How many ports do you want for it? |
16:43.42 | dlynes_office | zamba: what phone system, specifically? |
16:43.49 | zamba | dlynes_office: some alcatel-stuff.. |
16:44.00 | dlynes_office | zamba: e1 is pri, not bri |
16:44.02 | [TK]D-Fender | zamba : E1 is ISDN PRI for EU. |
16:44.04 | zamba | dlynes_office: "no one" in the office knows about it, it has just been running there for years :) |
16:44.28 | zamba | [TK]D-Fender: "many ports" means how many will be able to call the service at the same time, right? |
16:44.36 | [TK]D-Fender | zamba : Correct. |
16:45.00 | zamba | [TK]D-Fender: just one to begin with, but i want the system to scale |
16:45.14 | [TK]D-Fender | zamba : I had an old Nortel phone system with a 4 port VM which obvious how many people can be picking up / leaving VM at a time. That usit also served as our IVR |
16:45.33 | zamba | [TK]D-Fender: too many abbrievations :) VM and IVR? |
16:45.35 | [TK]D-Fender | zamba : What kind of ports do you have available on that PBX? |
16:45.50 | *** join/#asterisk websae (n=websae@209-252-79-66.ip.mcleodusa.net) |
16:45.56 | [TK]D-Fender | zamba : VoiceMail. Interactive Voice Response (auto-attendant) |
16:45.57 | zamba | [TK]D-Fender: i'm not quite sure, i'll have a look at it later today.. take some pictures and stuff.. |
16:46.01 | zamba | ah |
16:46.08 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
16:46.10 | [TK]D-Fender | zamba : Pictures = no good... need line specs. |
16:46.18 | zamba | yeah, i'll get that as well |
16:46.49 | [TK]D-Fender | zamba : T1/E1 users RJ48 which is indistinguisable from Ethernet & Norteles proprietary stuff really... |
16:47.17 | zamba | i think it's rj11 |
16:47.23 | [TK]D-Fender | dlynes_home : I take it you compiled the HWEC tools seperately?> |
16:47.24 | zamba | so analogue? |
16:47.31 | [TK]D-Fender | zamba : Quite likely. |
16:47.33 | dlynes_office | [TK]D-Fender: ummm...no? |
16:47.51 | zamba | [TK]D-Fender: but what are the alternatives for hardware on the asterisk box? |
16:47.56 | dlynes_office | [TK]D-Fender: i unzipped the main sangoma tarball, and then i unzipped the hwec tarball into the same directory |
16:48.06 | [TK]D-Fender | zamba : For that you'd use an analog TDM card. Probably 4-port cardwould serve your needs to start |
16:48.15 | zamba | [TK]D-Fender: got some names for me? |
16:48.17 | dlynes_office | [TK]D-Fender: and then ran the main setup |
16:48.26 | [TK]D-Fender | dlynes_home : Go inspect before I start swinging the trout :) |
16:48.36 | dlynes_office | [TK]D-Fender: go inspect what? |
16:48.42 | [TK]D-Fender | zamba : Digium TDM400P, Sangoma A200 |
16:48.50 | [TK]D-Fender | dlynes_home : the HWEC tools packacge |
16:49.00 | dlynes_office | what should i be looking at in it? |
16:49.10 | [TK]D-Fender | dlynes_home : just go look.... |
16:49.39 | dlynes_office | [TK]D-Fender: you mean the wan_ec subfolder off of /etc/wanpipe? |
16:49.47 | dlynes_office | [TK]D-Fender: or the original directory I installed from? |
16:51.08 | zamba | oh shit, those cards are expensive :) |
16:51.29 | dlynes_office | zamba: well, they're not video cards :) |
16:51.32 | zamba | hehe |
16:51.34 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
16:51.41 | generalhan | whats up all |
16:51.53 | dlynes_office | ni zao, generalhan |
16:52.22 | generalhan | I still cant get this stupid MWI issue fixed with 2 of my 7960s .... |
16:52.23 | [TK]D-Fender | dlynes_home : Look in the wanpipe/util/wan_ec from your source folder for some of the diagnosis binaries. |
16:52.26 | generalhan | its driving me insane ! |
16:52.41 | [TK]D-Fender | generalhan : Get a pair of pliers and yank them out.. |
16:53.12 | generalhan | lol ... for that matter i can just turn them off by removing the 'mailbox=' line in sip.conf. the goal is to get it to function properly ! lol |
16:53.25 | dlynes_office | generalhan: thought you said you were gonna try chan_skinny as soon as you got home yesterday? |
16:53.49 | generalhan | dlynes_home: wasnt me ... i havent been on since Friday ( i only get on while im at work ) |
16:54.02 | [TK]D-Fender | generalhan : Go make sure the VM box has no files in it... |
16:54.03 | generalhan | i try as hard as i can to forget about everything related to work when i leave |
16:54.16 | dlynes_office | generalhan: ah...anyways...if you check trunk, Qwell's committed a bunch of changes to chan_skinny |
16:54.29 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
16:54.30 | dlynes_office | generalhan: it might be much more stable than trying to run sip on a cisco now |
16:54.35 | dlynes_office | speak of the devil |
16:54.42 | dlynes_office | your ears must've been ringing, qwell |
16:54.45 | generalhan | [TK]D-Fender: nothing in it ... and ive even removed that line from voicemail.conf reloaded ... then put it back in so that it would create a new VM folder |
16:55.19 | generalhan | that stupid red light just wants to stay on |
16:55.37 | dlynes_office | [TK]D-Fender: what should I supply as a parameter for the devname for wan_ec_client? |
16:55.43 | dlynes_office | [TK]D-Fender: the if_name is obvious |
16:56.15 | generalhan | i thought that it was odd because they all have the same firmware .. so i actually downgraded that phone, and then reloaded the new version and still the stupid red light stays on |
16:58.09 | [TK]D-Fender | dlynes_office : Don't know the details.. I just remember the tech using that do diagnose problems with the DSP not firing up properly. |
16:58.24 | dlynes_office | [TK]D-Fender: ah..ok |
16:58.27 | dlynes_office | [TK]D-Fender: thx |
17:01.15 | *** join/#asterisk LokeshIndian (n=lokesh_k@estrela.nortenet.pt) |
17:02.21 | *** join/#asterisk JoseBravo (i=JoseBrav@200.119.32.47) |
17:02.24 | *** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
17:02.36 | LokeshIndian | Hello, can anyone please help me??? |
17:02.39 | paryl | does anyone know what "Don't know what to do if second ROSE component is of type 0x6" means? |
17:03.15 | [TK]D-Fender | LokeshIndian : www.drphil.com |
17:03.35 | inv_Arp | [TK]D-Fender: lol |
17:04.14 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
17:04.32 | dlynes_office | paryl: i've searched all over the damned place on asterisk-dev, asterisk-users, google, ... I haven't been able to find a damned thing on it, either |
17:04.54 | LokeshIndian | [TK]D-Fender: If my softphone is off and i try to call it then i gets some wierd entries in asterisk cdr |
17:05.10 | LokeshIndian | [TK]D-Fender: how i can get rid off with that ? |
17:05.23 | Hmmhesays | probabably because by defaults your cdrs reflect parts of your dialplan |
17:05.27 | dlynes_office | LokeshIndian: weird entries....good description |
17:06.01 | LokeshIndian | dlynes_office:hang on plz i m pasting them here |
17:06.16 | dlynes_office | ~pb |
17:06.18 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
17:06.18 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
17:07.09 | LokeshIndian | dlynes_office:-- Executing Hangup("Local/4365@from-sip-ec13,2", "") in new stack |
17:07.09 | LokeshIndian | <PROTECTED> |
17:07.38 | CunningPike | LokeshIndian: Looks like normal call progression to me........ |
17:07.39 | zamba | [TK]D-Fender: Intel Dialogic D/4-PCI ok? |
17:07.42 | dlynes_office | no kidding |
17:07.51 | dlynes_office | zamba: dialogic is absolute crap |
17:07.57 | dlynes_office | zamba: probably worse than digium |
17:08.04 | zamba | digium is crap as well? |
17:08.13 | zamba | what's no crap(tm)? |
17:08.18 | dlynes_office | zamba: not really, but it does have problems |
17:08.25 | dlynes_office | zamba: dialogic is horrible though |
17:08.29 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
17:08.31 | [TK]D-Fender | zamba : Avoid Dialogic.... barely usable at best... |
17:08.38 | LokeshIndian | CunningPike: if the softphone is off then i had 60-10 lines like i pasted here in the asterisk logs |
17:08.41 | [TK]D-Fender | zamba : Whats your idea of "expensive"? |
17:08.53 | CunningPike | zamba: Digium can be fussy to set up (some people have no problems, some do). The only real issue we had with Digium cards is the on-board EC |
17:08.54 | LokeshIndian | and all those entries are also gets logged in asterisk cdr database |
17:09.05 | zamba | [TK]D-Fender: well, since this is merely for testing, i'd prefer to be under 100$ |
17:09.34 | paryl | i'm getting some weird things with IAX... i've got two asterisk boxes, and we make calls back and forth without any issues |
17:09.34 | dlynes_office | zamba: go with an ATA or an analog gateway then |
17:09.54 | [TK]D-Fender | zamba : Would you like fires with that sir? :) Were talking about PBX's here.... |
17:10.01 | paryl | but in the last couple days , during calls, we've been getting just dead air... like 5-10 seconds, totally dead, and the call comes back and proceeds normally |
17:10.07 | paryl | nothing is in the logs |
17:10.09 | zamba | [TK]D-Fender: i know, but i'm new to this, and i really have no budget :) |
17:10.15 | [TK]D-Fender | zamba : around $200 for 2 line Sangoma A200 |
17:10.26 | zamba | [TK]D-Fender: and since i can't guarantee success it's hard for me to sell this to my company :) |
17:10.27 | [TK]D-Fender | fries* |
17:10.46 | dlynes_office | zamba: you can get Yoda G620's, Sipura 2000's (ATA), Sipura 3000 (analog gateway) |
17:10.53 | [TK]D-Fender | zamba : How about you just replace the entire PBX with *.... that'd be effective... |
17:10.58 | dlynes_office | zamba: but they're not hardware cards...they're external units |
17:11.03 | LokeshIndian | CunningPike: its not normal call progression..can u plz help me |
17:11.06 | zamba | [TK]D-Fender: then i definitely can't guarantee success :) |
17:11.15 | [TK]D-Fender | Talks does Yoda funny hhmmmMMMMMM?!?! |
17:11.19 | dlynes_office | LokeshIndian: it is normal call progression |
17:11.46 | dlynes_office | CunningPike: you mean the lack of an onboard ec? |
17:11.54 | zamba | dlynes_office: how do i connect those to asterisk? LAN? |
17:12.01 | dlynes_office | zamba: exactly |
17:12.08 | zamba | interesting |
17:12.18 | dlynes_office | zamba: they're sip devices |
17:12.25 | dlynes_office | zamba: the yoda devices can also do h323 |
17:12.30 | *** join/#asterisk Gabriel25 (n=gabe@user-12ld5f7.cable.mindspring.com) |
17:12.34 | CunningPike | dlynes_office: No - we found the VPM less effective than software EC, and it introduced call quality issues (clicks and dropouts) |
17:12.34 | Gabriel25 | Hi hys |
17:12.39 | Gabriel25 | [root@server zaptel]# make |
17:12.39 | Gabriel25 | You do not appear to have the sources for the 2.6.16-1.2133_FC5smp kernel installed. |
17:12.39 | Gabriel25 | make: *** [linux26] Error 1 |
17:12.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-80-10.red.bezeqint.net) |
17:12.51 | Gabriel25 | I have this error but I installed kernel-devel |
17:12.54 | dlynes_office | Gabriel25: pretty self explanatory, don't you think? |
17:12.56 | Gabriel25 | any idea ? |
17:13.20 | Gabriel25 | dlynes_home I installed kernel-devel |
17:13.26 | CunningPike | Gabriel25: symlink in /usr/src? |
17:13.32 | dlynes_office | Gabriel25: obviously not hte right kernel-dev |
17:13.41 | dlynes_office | Gabriel25: you probably installed the wrong version of kernel-dev |
17:13.52 | JoseBravo | Gabriel try to install kernel-smp-devel |
17:14.16 | tzafrir_laptop | the kernel-devel package has the symlink /lib/modules/`uname -r`/build ? |
17:14.31 | dlynes_office | tzafrir_laptop: one would think |
17:14.48 | Gabriel25 | [root@server gabriel]# uname -r |
17:14.48 | Gabriel25 | 2.6.16-1.2133_FC5smp |
17:14.48 | Gabriel25 | [root@server gabriel]# rpm -qa kernel-devel |
17:14.48 | Gabriel25 | kernel-devel-2.6.16-1.2133_FC5 |
17:15.03 | [TK]D-Fender | CunningPike : I hear the VPM rev 2 is a noticable improvement. |
17:15.13 | dlynes_office | Gabriel25: see the lack of the 'smp' appended to that kernel version? |
17:15.14 | [TK]D-Fender | CunningPike : And that a REAL DSP is pending. |
17:15.20 | CunningPike | [TK]D-Fender: So is a Ditech box ;) |
17:15.34 | dlynes_office | Gabriel25: like i said...you don't have the correct version of kernel-dev installed....listen to JoseBravo |
17:15.41 | Gabriel25 | ok |
17:15.43 | [TK]D-Fender | CunningPike : So is my Otasic ;) |
17:15.48 | JoseBravo | Gabriel25 yum isntall kernel-smp-devel |
17:15.50 | CunningPike | [TK]D-Fender: Heh heh |
17:16.54 | *** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69) |
17:16.57 | Gabriel25 | ok |
17:17.07 | Gabriel25 | thank you so much |
17:17.13 | CunningPike | Actually, here is a question for the group - all of a sudden, it seems that the far end of our PRIs are receiving any CID number any more - anyone ever seen that happen? |
17:17.37 | CunningPike | All the 'pri intense debug' looks normal, but the far end isn't seeing a number |
17:17.46 | CunningPike | It's weird |
17:18.12 | dlynes_office | CunningPike: is it every far end? or just one particular location? |
17:18.29 | CunningPike | dlynes_office: Every - our Nortel, and the telco |
17:18.36 | CunningPike | dlynes_office: Name is fine - just number |
17:18.46 | dlynes_office | CunningPike: try giving me a call through it...see if I get your caller id |
17:18.51 | CunningPike | OK |
17:19.04 | dlynes_office | CunningPike: you've got my cell number, right? |
17:19.38 | lunk | how do you setup internal routing between SIP extensions? (or a link to some info) |
17:19.46 | lunk | i'm missing something :/ |
17:20.03 | Gabriel25 | thank you guys is working |
17:20.06 | Gabriel25 | stupid me ! |
17:26.28 | JoseBravo | ANoye have installed AstBill without all expample info? |
17:27.16 | CunningPike | lunk: Explain? |
17:27.30 | *** join/#asterisk wingman_sg (n=Wingman@bb219-74-103-65.singnet.com.sg) |
17:28.30 | lunk | CunningPike: i plead ignorance |
17:28.52 | CunningPike | lunk: Explain what it is you are trying to do |
17:29.16 | wingman_sg | anybody know how to instal/configure l Dialogic card into Asterisk platform ? |
17:29.21 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
17:29.26 | lunk | i just want ext 500 to call ext 501 |
17:29.33 | lunk | i don't have any internal routing |
17:29.46 | rene- | is it possible to have agents defined like agent/johnny as opposed to numeric only agent channel names? |
17:29.53 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
17:29.54 | CunningPike | lunk: So, have you set up a basic dialplan? |
17:30.05 | lunk | that must be what i'm missing |
17:30.14 | lunk | i can call out through voipjet, but not internally |
17:30.35 | CunningPike | rene-: It sure is |
17:31.05 | CunningPike | lunk: So, you need to have each UA in sip.conf, and then a dialplan that handles calls between them |
17:31.26 | [TK]D-Fender | lunk : ... |
17:31.27 | [TK]D-Fender | ~book |
17:31.34 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
17:31.43 | rene- | thanks, question 2, thereis agentlogin and agentcallbacklogin, can i use addqueuemember and still have the agentlogin functionality? (always in line, new calls preceeded by beep, as opposed to telephone onhook new call is new ring) |
17:32.26 | JoseBravo | lunk |
17:32.28 | lunk | CunningPike: do you have simple dial plan example? I'm looking on voip-info, but they're all pretty complex |
17:32.39 | *** join/#asterisk ManxPower (i=ewieling@53.sub-70-219-19.myvzw.com) |
17:32.42 | ManxPower | Jun 19 12:28:51 WARNING[902]: chan_zap.c:8394 pri_dchannel: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. |
17:32.43 | ManxPower | yippee |
17:32.57 | CunningPike | lunk: The samples that come with asterisk are good, and I highly recommend The Book |
17:33.01 | CunningPike | ~book |
17:33.03 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
17:33.03 | ManxPower | according to "show channels" channel 0/2 was NOT in use. |
17:33.07 | JoseBravo | You need to add exten => 500,1,Dial(SIP/sip_user) |
17:33.12 | rene- | ~rene- |
17:33.14 | jbot | i guess rene- is THE MAN |
17:33.22 | rene- | i agree |
17:33.38 | CunningPike | jbot, rene- is vain |
17:33.39 | jbot | ...but rene- is already something else... |
17:34.02 | CunningPike | jbot, rene- is also vain |
17:34.03 | jbot | CunningPike: okay |
17:34.07 | CunningPike | :D |
17:34.08 | [TK]D-Fender | ~[TK]D-Fender |
17:34.10 | jbot | methinks [tk]d-fender is rockin' the casbah !!! |
17:34.14 | [TK]D-Fender | :D |
17:34.20 | mitcheloc | i forgot mine... |
17:34.21 | mitcheloc | ~mitcheloc |
17:34.22 | jbot | somebody said mitcheloc was your master |
17:34.31 | mitcheloc | heh, i like that |
17:34.33 | CunningPike | Settle down, children |
17:34.59 | *** join/#asterisk sg_wingman (n=Wingman@bb219-74-103-65.singnet.com.sg) |
17:35.11 | mitcheloc | ~CunningPike |
17:35.15 | generalhan | !^caret Outcast (Goodie Mobb) - They Don't Dance No More.mp3 |
17:35.20 | generalhan | HAHAHA ! |
17:35.20 | mitcheloc | ~<insert nick here> |
17:35.30 | generalhan | i Love random spam mess. on mIRC |
17:35.48 | sg_wingman | CLEAR |
17:35.53 | mitcheloc | jbot, CunningPike is invisible |
17:35.54 | jbot | mitcheloc: okay |
17:35.57 | CunningPike | :D |
17:36.04 | lunk | JoseBravo: that's all it was, thanks dude, knew it was something small |
17:36.04 | mitcheloc | ~CunningPike |
17:36.06 | jbot | i heard cunningpike is invisible |
17:36.09 | rene- | jbot, CunningPike is also ledgar se cae |
17:36.11 | jbot | rene-: okay |
17:36.15 | mitcheloc | where'd he go! |
17:36.18 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
17:36.32 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
17:37.14 | paolob | Hi guys! Is there a string in asterisk a function returning the lenght of a variable? |
17:37.20 | rene- | this is funny :) |
17:37.25 | syzygybsd | I have a couple polycoms nated behind routers. When I connect to asterisk (not nated) and try to make a call it is trying to send it to my local 192.168 addresses instead of my external address |
17:37.40 | syzygybsd | I am connecting via sip and have nat=1 in the sip.conf |
17:37.46 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
17:37.47 | rene- | paolob: most likely, try show applications like len in the asterisk cli |
17:38.07 | dlynes_office | rene-: and show functions |
17:38.23 | dlynes_office | rene-: such as for 'if'; it's a function, not a dialplan application |
17:38.36 | *** join/#asterisk thock (n=thock@63.133.144.2) |
17:38.39 | dlynes_office | no idea what the difference is but... |
17:38.43 | rene- | jbot, CunningPike went to watch edgar se cae http://www.youtube.com/watch?v=0Ab9ERPhLh8 |
17:38.51 | ManxPower | syzygybsd, Dial by peer name, not IP address |
17:39.01 | paolob | rene-, I can't see it |
17:39.13 | thock | Anyone here pretty knowledgeable with E&m/PRI's that i can bug for a few minutes? |
17:39.20 | CunningPike | rene-: What does ledgar se cae mean? |
17:39.25 | syzygybsd | ManxPower: it happens if I dial out zap from my sip phone too |
17:39.38 | rene- | i meant edgar se cae: http://www.youtube.com/watch?v=0Ab9ERPhLh8 |
17:39.48 | syzygybsd | or if I do dial(sip/Gary) |
17:39.50 | dlynes_office | paolob: show functions, you'll see 'LEN' |
17:40.04 | paolob | dlynes_home, ok, thank you! |
17:40.47 | dlynes_office | paolob: if you don't, it's because you don't have one of the func_*.so modules loaded |
17:44.41 | paolob | well, the command Dial(SIP/${EXTEN:LEN(${VAR})}@skypho,60,t) doesn't work. What am I wrong? |
17:45.53 | CunningPike | rene-: So which one am I? The mean orange guy, or the poor red one? :) |
17:46.17 | thock | Here's something really strange. my LDT1 is constantly calling my extension every minute and a half |
17:46.19 | thock | leaves a message |
17:46.21 | thock | and then hangs up |
17:46.32 | thock | CID shows as "unknown" |
17:47.53 | paolob | dlynes_home, could you tell me if I can put Dial(SIP/${EXTEN:LEN(${VAR})}) ? |
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17:48.38 | [TK]D-Fender | paolob : Do you have any clue what that does? Basically its returns a BLANK. |
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17:48.57 | rene- | i dont know you get to pick victim or saddistic cousin |
17:49.02 | [TK]D-Fender | paolob : Actualy... what's {var}.. I missed something there.. |
17:49.21 | CunningPike | rene-: Probably victim. [TK]D-Fender is the sadistic cousin ;) |
17:49.26 | rene- | heh |
17:49.32 | [TK]D-Fender | paolob : And you need to call a function inside of ${} |
17:50.25 | [TK]D-Fender | paolob : And what are you trying to strip a variable # of chars off an exten? |
17:50.33 | paolob | [TK]D-Fender, I want to use it to strip away a code (like #123545) of variable lenght. The code is used to permit international calls only to the persons that receive the code |
17:50.47 | Dr-Linux | dlynes_home: around? |
17:51.25 | [TK]D-Fender | paolob : pastebin your code segment (everything related) |
17:51.34 | paolob | [TK]D-Fender, for example I set the code at "#567", and I give it to the person I permit to make itnl calls, marking #567 to be able to make the code |
17:52.06 | [TK]D-Fender | paolob : Show me exactly how you're going to dial it. |
17:53.21 | Egonis | FYI: I asked for help earlier about IRQ issues with my TDM400P, I added 'noacpi, acpi=off, and routeirq' to my kernel options, and disabled assign IRQ to VGA Controller in the bios -- and bingo.. works like a charm |
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17:55.03 | paolob | [TK]D-Fender, http://pastebin.com/718909 |
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18:02.13 | paolob | [TK]D-Fender, I added the result in the CLI |
18:02.36 | *** join/#asterisk dec_ (n=tom@ppp147-40.lns3.adl2.internode.on.net) |
18:02.55 | Gabriel25 | Jun 19 14:04:41 ERROR[6260] chan_zap.c: Unable to load config zapata.conf |
18:02.55 | Gabriel25 | Jun 19 14:04:41 WARNING[6260] loader.c: chan_zap.so: load_module failed, returning -1 |
18:02.55 | Gabriel25 | Jun 19 14:04:41 WARNING[6260] loader.c: Loading module chan_zap.so failed! |
18:03.00 | thock | what would be the first thing to look for when you get a channel unavailable message? |
18:04.55 | rene- | ~rene |
18:04.56 | jbot | [rene] always happy to learn new words! what's antsy? |
18:05.08 | rene- | ~rene- |
18:05.10 | jbot | it has been said that rene- is THE MAN, or vain |
18:05.29 | rene- | jbot, rene- is not vain |
18:05.30 | jbot | ...but rene- is already something else... |
18:05.52 | rene- | damn |
18:09.45 | tzafrir_laptop | jbot, no, rene- is just a nick floating around on #asterisk |
18:09.47 | jbot | okay, tzafrir_laptop |
18:10.51 | tzafrir_laptop | Gabriel25, touch /etc/asterisk/zapata.conf |
18:11.10 | tzafrir_laptop | alternatively, maybe this is a permissions problem? |
18:11.31 | tzafrir_laptop | (touch: in case it didn't exist) |
18:12.38 | *** join/#asterisk h3x (i=hex@ip70-189-236-254.lv.lv.cox.net) |
18:12.41 | PakiPenguin | jbot, tzafrir_laptop's company makes astribanks |
18:13.18 | PakiPenguin | ah not listening to me |
18:17.04 | Corydon-w | jbot: astribank is <reply>Ask tzafrir about the Astribank. I dunno anything about it. |
18:17.05 | jbot | okay, Corydon-w |
18:17.48 | Corydon-w | ~botsnack |
18:17.48 | jbot | Corydon-w: aw, gee |
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18:28.43 | gandhijee | has anyone check the dlfnc.c file? |
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18:29.24 | gandhijee | the static int isValidStatus(struct dlstatus *status) looks kinda jacked |
18:34.21 | [TK]D-Fender | paolob : You are using a constant, not a variable, so why not just use a constand for your trim as well. And you aren't calling the LEN function properly. |
18:38.48 | gandhijee | i should say it seems jacked for x-compiles |
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18:39.22 | [TK]D-Fender | paolob : You also seem to not have a "1" priority for that exten... another "no-no" |
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18:40.21 | *** join/#asterisk clive- (n=pirch@dsl-145-34-91.telkomadsl.co.za) |
18:41.38 | clive- | is " PCadach" here ?....under another Nick |
18:42.49 | clive- | found him |
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18:46.21 | TheMonoTone | so I've been trying out various echo cancellation methods and they all seem to get rid of the echo |
18:46.24 | TheMonoTone | but are adding noise |
18:46.35 | TheMonoTone | is there some setting I should be adjusting in zapata.conf ? |
18:46.51 | TheMonoTone | I'm using a pair of zaptel tdm400 cards |
18:49.42 | SplasPood | it'd be nice to be able to do ifAppisLoaded(app_milliwatt.so,something,something) |
18:49.44 | feld | TheMonoTone: look into the fxotune tool to "tune" your ports to the analog lines and also try the echolearning or whatever it is |
18:50.01 | feld | echolearning is a setting in zapata.conf i believe |
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18:53.38 | TheMonoTone | feld: I'll check in to fxotune first |
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18:55.09 | vader-- | do you guys know if there is a sound file that comes with asterisk that says please record your message? |
18:55.41 | dlynes_office | vader--: it'd be /var/lib/asterisk/sounds/vm-something-or-other.gsm |
18:55.48 | [TK]D-Fender | SplasPood : There are ways.... not elegent though... |
18:56.23 | SplasPood | [TK]D-Fender: yea... |
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19:06.29 | Navire | Someone how I test SIP Post Dial delay? |
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19:15.06 | dlynes_office | Has anyone encountered a problem whereby all incoming iax2 calls and all incoming zap calls are not passing audio? |
19:15.20 | dlynes_office | I can't seem to figure out the problem, for the life of me |
19:15.31 | dlynes_office | sip to sip is working just fine |
19:15.49 | dlynes_office | The iax2 is set up exactly the same way I have it on another box, that's working just fine |
19:15.55 | ghenry | doesn't sip pass audo via rtp? |
19:15.59 | dlynes_office | yep |
19:16.11 | ghenry | so they do audio indepant of * |
19:16.26 | dlynes_office | yeah? |
19:16.46 | ghenry | maybe your sound card then |
19:16.50 | TheMonoTone | ugh, why is there so much noice in this phone |
19:16.57 | TheMonoTone | a regular phone line doesn't have this much noice |
19:16.58 | TheMonoTone | *noise |
19:17.00 | dlynes_office | ghenry: sound card? huh? I'm not even using a soundcard |
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19:17.23 | dlynes_office | ghenry: it's two sip hardphones |
19:17.36 | ghenry | ah right |
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19:18.00 | shaun2222 | any of you guys used the linksys WRT54GP2 |
19:18.16 | shaun2222 | does it require vontage or can i use it to connect via sip to my asterisk server |
19:18.17 | ghenry | do you only need a sound device for ivr? |
19:19.01 | dlynes_office | ghenry: you only need a sound device for paging |
19:19.20 | dlynes_office | ghenry: i.e. for chan_alsa, chan_oss, and I think chan_phone as well |
19:19.30 | ghenry | paging? |
19:19.30 | dlynes_office | ghenry: chan_phone is a softphone built into asterisk I think |
19:19.36 | ghenry | ah |
19:19.39 | dlynes_office | ghenry: yeah...paging out over a loudspeaker |
19:19.41 | Dr-Linux | dlynes_home: why this now? >> Jun 19 12:18:22 WARNING[13423]: loader.c:554 load_modules: Loading module chan_zap.so failed! |
19:19.45 | ghenry | gotya dlynes_home |
19:23.19 | dlynes_office | Dr-Linux: i have no idea...you're not showing me the whole error |
19:23.37 | thock | Dr-Linux: pastebin your zaptel.conf? and possibly the last bits of /var/log/asterisk/full ? |
19:23.51 | dlynes_office | Dr-Linux: and zapata.conf |
19:23.55 | thock | that too |
19:24.25 | Dr-Linux | dlynes_office: thanks, i figured out |
19:24.37 | dlynes_office | Dr-Linux: jumped the gun? |
19:24.42 | Dr-Linux | dlynes_office: now it showing my 2 ports "OK" |
19:24.46 | thock | [TK]D-Fender: can i pick your brain a bit re: sangoma stuff |
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19:27.08 | Dr-Linux | dlynes_office: |
19:27.09 | Dr-Linux | Description Alarms IRQ bpviol CRC4 |
19:27.09 | Dr-Linux | T2XXP (PCI) Card 0 Span 1 OK 0 0 0 |
19:27.09 | Dr-Linux | T2XXP (PCI) Card 0 Span 2 OK 0 0 0 |
19:27.36 | Dr-Linux | dlynes_office: i skiped the 2nd span |
19:28.04 | thock | Dr-Linux: is ztdummy running? |
19:28.40 | dlynes_office | thock: he's not having an issue now....he figured it out |
19:28.52 | thock | oh. |
19:28.55 | Dr-Linux | thock: lsmod doesn't show "ztdummy" is it not running right? |
19:29.07 | Dr-Linux | dlynes_home: noooooooo |
19:29.16 | thock | what was the original problem? |
19:29.34 | Dr-Linux | i just figured out the error after changing the zaptel.conf structure |
19:30.07 | Dr-Linux | thock: when i dialout, i get busy congested |
19:30.16 | thock | Ah. |
19:30.24 | thock | I'm having that same problem with my LDT1 |
19:30.47 | thock | but it's probably the fault of my insessent newbitizim editing my dialplan :< |
19:31.09 | Dr-Linux | thock: mine is not dialplan issue, |
19:31.36 | Dr-Linux | looks like my pri lines are connected, but my telco didn't put the DID's or D channels stuff yet :S |
19:31.48 | thock | Dr-Linux: open up the CLI and type pri show span # |
19:31.51 | thock | and then the span of your PRI |
19:32.37 | Dr-Linux | thock: that shows down, active |
19:32.39 | Dr-Linux | lemme show ya |
19:32.42 | thock | pm it |
19:33.39 | Dr-Linux | ok |
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19:42.18 | pagec | what if i want to show if person x is on the phone to phone y? |
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20:04.18 | JoseBravo | Im triying to do a call from 70103 tp 71462 extension. But its my CLI output: http://pastebin.com/719296 any idea? |
20:06.58 | Bullseye_Network | I cant get anything to come up on pastebin.com hmmm... |
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20:08.57 | TripleFFFF | can we have playback use mp3 ? |
20:11.32 | *** join/#asterisk Skumlos (n=youl@nat.kollegienet.dk) |
20:11.59 | Skumlos | mjallo |
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20:16.00 | TripleFFFF | >?? |
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20:20.46 | *** join/#asterisk MedozasSVR (n=MedozasS@p549B8087.dip0.t-ipconnect.de) |
20:20.55 | MedozasSVR | Hi guys! |
20:21.17 | MedozasSVR | Is anybody here with knowledge of asterisk realtime? |
20:21.46 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
20:22.11 | TripleFFFF | yyeah |
20:22.34 | vivek | hello all, can i use tor & privoxy on a linksys router to get across the voip blocks put up by some countries like uae ? |
20:23.04 | LoRez | not if you want to actually use VoIP |
20:23.43 | vivek | hmmz ok i don't get why though ... |
20:23.50 | MedozasSVR | tripleffff: you mean you have knowledge with asterisk realtime? |
20:24.41 | JoseBravo | How I see the agi debug? |
20:24.56 | MedozasSVR | @jose bravo: agi debug |
20:25.14 | MedozasSVR | thats the command |
20:25.38 | vivek | LoRez: lets say i have a spa-2k connected the the linksys router and a gizmo or some other voip provider. and linksys passes all the info via the http port ... it won't work ? |
20:26.35 | MedozasSVR | i have varoius questions regarding it: |
20:26.35 | MedozasSVR | 1. how about support for "s" and "i" extensions within realtime - are they now builtin to 1.2.9.1 or still only availiable via SVN? |
20:26.35 | MedozasSVR | 2. are there any implementations regarding conference rooms (meetme.conf) in realtime except from asterisk realtime static (like sip table or so)? |
20:27.02 | dlynes_office | [TK]D-Fender: found out the problem...there was a stale socket file for wanpipe in /var/run |
20:27.05 | TripleFFFF | no idea |
20:28.26 | dlynes_office | [TK]D-Fender: btw...for future reference, it's like wan_ec_config wanpipe1 w1g1 stats |
20:28.42 | [TK]D-Fender | dlynes_home : Thought so.. I did mention that to you a while ago.. from a lockup quite likely |
20:28.57 | dlynes_office | [TK]D-Fender: ah...didn't see you mention the stale file |
20:29.05 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
20:29.06 | dlynes_office | [TK]D-Fender: i got that from David |
20:29.09 | [TK]D-Fender | dlynes_office : That was like 2 weeks ago :) |
20:29.12 | dlynes_office | oh |
20:29.13 | dlynes_office | hahaha |
20:29.22 | dlynes_office | i was having problems with sangoma in general then |
20:29.24 | [TK]D-Fender | dlynes_home : Before I got stage 1 working for you :) |
20:29.37 | dlynes_office | so i wasn't even looking at ec at that point |
20:30.14 | [TK]D-Fender | yeah... echo would be a sign of life back then :) |
20:30.21 | dlynes_office | exactly |
20:30.32 | dlynes_office | anyways...it's up and running now, with hwec enabled |
20:30.45 | dlynes_office | I guess when it says it's using mg1 echo canceller, you can ignore that, right? |
20:31.07 | dlynes_office | as long as the hwec reports back correctly? |
20:31.40 | justinu|laptop | i'd guess you'd want to disable any software EC in zaptel |
20:31.49 | [TK]D-Fender | dlynes_home : exactly... the Otasic kicks in when the wanpipe drive sees the call for EC. |
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20:32.29 | [TK]D-Fender | dlynes_office : which is why you need to specify "echocancel=yes", not nothing regarding echotraining, and you'll never have to screw around with gains again. |
20:33.33 | [TK]D-Fender | Zaptel EC never really gets called. Wanpipe sees to that (a change in the more modern firmwares) |
20:34.34 | justinu|laptop | cool |
20:34.46 | [TK]D-Fender | ok, heading home, back in a few. |
20:35.56 | MedozasSVR | another question: are there any tags for noise reduction for sip and/or zap? |
20:36.16 | MedozasSVR | to reduce background noise |
20:36.45 | dlynes_office | MedozasSVR: for sip, you can replace your phone |
20:36.57 | dlynes_office | MedozasSVR: for zap, you can try adjusting the gains |
20:37.04 | dlynes_office | MedozasSVR: also for sip, you can try a different codec |
20:37.13 | justinu|laptop | or replace those irritating CPU fans with quiet models |
20:38.10 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
20:38.45 | MedozasSVR | phone: a good cisco, no way :) --- codec is 729 with a 1 m/bit line with traffic shaping (module hsfc) |
20:39.38 | *** part/#asterisk terrapen (n=cjs@166.70.183.108) |
20:39.43 | MedozasSVR | well its actually not because of cpu fans - much more because other people talking in the background, and some people are irritated by that |
20:40.09 | justinu|laptop | hmm, mic is too sensitive? |
20:40.31 | justinu|laptop | or you txgain on your zap channels is too high |
20:40.37 | justinu|laptop | if you're doing sip->zap calls |
20:40.49 | MedozasSVR | its same with sip>sip |
20:41.23 | justinu|laptop | in that case you can only blame the phones, since asterisk is just forwarding rtp, not modifying it |
20:41.26 | MedozasSVR | i know the 7960 can be changed a bit in its sensitivity, but are there any tags for asterisk? would save time.... |
20:41.37 | *** join/#asterisk GarethTheGreat (n=gareth@unaffiliated/gareththegreat) |
20:41.39 | justinu|laptop | no, asterisk doesn't modify the rtp packets unless it's transcoding |
20:41.52 | MedozasSVR | darn, thanks! |
20:41.54 | justinu|laptop | np |
20:42.02 | GarethTheGreat | anyone here had problems building on Fedora Core 4? |
20:42.36 | MedozasSVR | @justinu: as you seem to know a lot - do you know things of asterisk realtime? can you help me there? |
20:42.57 | GarethTheGreat | /usr/bin/ld: cannot find -lssl |
20:43.03 | GarethTheGreat | getting that when trying to make install |
20:43.09 | GarethTheGreat | though openssl is installed alright |
20:43.11 | MedozasSVR | you should install openssl |
20:43.17 | MedozasSVR | openssl-devel |
20:43.34 | justinu|laptop | yep, you'll need a lot of *-devel packages perhaps |
20:43.48 | MedozasSVR | i can give you an exact listing - one moment |
20:43.51 | GarethTheGreat | compiling now |
20:45.14 | MedozasSVR | gcc, ncurses-devel, openssl-devel, patch (not neccessarily), bison, zlib-devel, kernel-source, kernel-syms |
20:45.33 | MedozasSVR | @gareth: did it work now? |
20:45.48 | sevard | damnit |
20:46.05 | sevard | If somebody enters 4000# how would I strip off the # |
20:46.27 | MedozasSVR | ${EXTEN:-1} |
20:46.42 | MedozasSVR | sorry, wrong |
20:46.42 | sevard | i've already tried that |
20:46.43 | sevard | hmm |
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20:46.58 | justinu|laptop | sevard: get that ATA? |
20:46.58 | MedozasSVR | ${EXTEN:0:4} |
20:47.09 | MedozasSVR | if 4 digits |
20:47.22 | sevard | justinu|laptop: Yes I did, did you catch my email from yesterdayand the one from today? |
20:47.30 | sevard | MedozasSVR: what if it's an undefined amount od digits |
20:47.37 | sevard | s/od/of/g |
20:48.01 | MedozasSVR | hmm |
20:48.13 | MedozasSVR | then it should be able to work with LEN |
20:48.38 | justinu|laptop | servard: not until just now :) |
20:49.00 | justinu|laptop | so you got it working well? |
20:49.02 | justinu|laptop | that's awesome |
20:49.10 | sevard | like ${EXTEN:$LEN:{MATH:LEN-1:}} |
20:49.13 | sevard | or something crazy? |
20:49.19 | sevard | justinu|laptop: yeah :D:D |
20:49.23 | justinu|laptop | werd |
20:49.28 | sevard | justinu|laptop:I played with it for a couple of hours |
20:49.32 | MedozasSVR | im testing at one system at the moment |
20:49.42 | MedozasSVR | give me some minutes, ok? |
20:49.52 | sevard | right on |
20:50.07 | sevard | justinu|laptop: if you come across another one I can show you how to do it |
20:50.31 | justinu|laptop | i will take you up on that |
20:50.48 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
20:51.18 | sevard | justinu|laptop: apparently it will take a day or two for funds to be transfered, my first time using paypal ;/ |
20:51.33 | sevard | justinu|laptop: i'm really appreciative though man, it's pretty awesome of you. |
20:52.14 | MarcPtz | Hi all , one question , when working with a2billing all generated calls are by default redirected to context [from-sip-external] ? |
20:52.43 | justinu|laptop | it's ok, i'm not waiting on your money to eat or anything :) |
20:53.05 | sevard | hehe |
20:53.06 | justinu|laptop | just glad it has a good home :) |
20:53.22 | sevard | :D it's awesome |
20:55.43 | Qwell[] | justinu|laptop: what, no guilt trip? |
20:56.06 | sevard | justinu is a good guy |
20:56.17 | Qwell[] | bah! :p |
20:56.33 | justinu|laptop | nah, i wouldn't guilt him even if he didn't pay... life's to short to be pissed off over 12 bucks |
20:56.43 | Qwell[] | oh, heh |
20:56.50 | justinu|laptop | but i wasn't gonna tell him that before he paid :) |
20:56.57 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
20:57.50 | rene- | what is the deal with realtime agents? is it something that will be available for * 1.4? is it available for head? |
20:58.11 | Qwell[] | rene-: I believe it's in trunk..which means it'll be in 1.4 |
20:58.50 | justinu|laptop | be a man, backport it to 1.2 |
20:58.59 | mog | righttttttttttttt |
20:59.03 | Qwell[] | real men backport to a pre 2005 cvs |
20:59.19 | Qwell[] | mog: Did you see, did you see?! |
20:59.22 | justinu|laptop | heh |
20:59.28 | mog | wha Qwell ? |
20:59.30 | justinu|laptop | i backported res_snmp to 1.2 |
20:59.32 | Qwell[] | pfft |
20:59.34 | justinu|laptop | took about 30 minutes |
20:59.36 | Qwell[] | didn't even see... |
20:59.42 | Qwell[] | mog: skinny in trunk :p |
20:59.48 | mog | OH that |
20:59.50 | mog | yes i did |
20:59.55 | Qwell[] | :D |
21:00.45 | rene- | thx |
21:01.09 | justinu|laptop | so who's running gentoo? |
21:01.15 | sevard | nobody important |
21:01.17 | justinu|laptop | lol |
21:01.19 | Qwell[] | justinu|laptop: You know I do :p |
21:01.20 | sevard | :) |
21:01.25 | sevard | i rest my case. |
21:01.26 | Qwell[] | why? because that's what real men run! |
21:01.33 | justinu|laptop | gentoo seems pretty slick |
21:01.46 | Qwell[] | mog: I don't have access to the "other" file ;) |
21:01.52 | CunningPike | justinu|laptop: yes - oil slick ;) |
21:01.57 | justinu|laptop | lots of good docs for running on laptops |
21:01.58 | *** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
21:02.01 | justinu|laptop | lots of control |
21:02.18 | Qwell[] | (CREDITS, that is) |
21:02.21 | mog | ohhh |
21:02.22 | mog | yeah |
21:02.49 | file | eh? |
21:02.52 | Qwell[] | I wish he did too |
21:02.58 | MedozasSVR | @sevard: hmmm - tricky |
21:03.11 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
21:03.22 | MedozasSVR | you can make one thing: match with if or gotoif |
21:03.32 | MedozasSVR | is better anyways |
21:03.57 | MedozasSVR | so you can match with ${EXTEN:-1:1} if this is '#' |
21:04.11 | MedozasSVR | and use if or gotoif as case what to do |
21:10.48 | *** join/#asterisk alystair (i=Alystair@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com) |
21:11.15 | Poincare | anyone knows why 'call-limit' won't work with some sip-providers? |
21:11.53 | generalhan | Poincare: what ver. of * you using ? |
21:12.16 | Poincare | generalhan: Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q |
21:12.37 | Poincare | it works fine with a test account on a friends * |
21:13.00 | Poincare | but fails when I try limit calls one of those sip-providers |
21:13.40 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-254-10.telkomadsl.co.za) |
21:14.27 | generalhan | Poincare: in your sip.conf are your phone entried listed as "friend" or "peer" ? |
21:15.05 | Poincare | peer, it aren't phones but accounts on another server |
21:16.11 | generalhan | Poincare: im not familiar with this issue ... but read here and see if this seems to be your issue http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+incominglimit |
21:16.40 | Poincare | almost know that page by hard :-) |
21:16.42 | generalhan | Poincare: all the way to the bottom they use a group function to make it work |
21:17.08 | generalhan | are you trying to only allow one or so calls at a time through one provider ? |
21:17.10 | sevard | I have a question about phone.conf |
21:17.14 | sevard | Do we really even need it? |
21:17.38 | *** join/#asterisk Lino` (n=Lino@i577BD2A6.versanet.de) |
21:17.43 | Poincare | that's the idea yes, to make sure i don't make a second outgoing call through a provider |
21:18.22 | alystair | anyone here heard of Comwave? |
21:21.23 | *** join/#asterisk zeppelin_ (n=zeppelin@201-35-78-44.paebv700.dsl.brasiltelecom.net.br) |
21:21.54 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3n74.cable.mindspring.com) |
21:22.24 | Mattwj2005 | hey guys I just wanted to let you guys know about something that I discovered |
21:23.07 | Mattwj2005 | one of the programs that comes with mythtv, mythfrontend conflicts with Asterisk....that is why I was getting all those errors before |
21:25.07 | *** join/#asterisk los415 (n=los415@sfca-office.corp.race.com) |
21:33.06 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
21:35.04 | *** join/#asterisk rollot (n=rollotom@c-68-37-168-10.hsd1.pa.comcast.net) |
21:35.44 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
21:37.04 | justinu|laptop | hey paki, ltns |
21:37.15 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
21:37.44 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
21:38.02 | smackus | am i going to have any issues using asterisk and zaptel on a 64 bit machine? I will be running Red Hat Enterprise |
21:38.04 | dlynes_office | sevard: i think that's for using the cli 'dial' application |
21:40.16 | *** part/#asterisk rollot (n=rollotom@c-68-37-168-10.hsd1.pa.comcast.net) |
21:43.26 | *** part/#asterisk smackus (n=smackus@63.149.122.94) |
21:45.06 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
21:46.08 | flujan | guys, I'm trying to use chan_spy and monitor a specific channel... is it possible? I want do record the conversation and chan_spy the channel at the same time. |
21:47.35 | *** join/#asterisk existx (i=existx@sniff.ttyp.net) |
21:48.39 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
21:49.17 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
21:51.18 | redder86 | Hi. We were using Asterisk 1.0 up until a few weeks ago at which point I upgraded to 1.2.7.1 (and now 1.2.9.1). After doing so we started having some problems where the call audio would be grossly distorted. I'm ready to revert the installation to use 1.0 again, but I would be delighted to work with someone to root-out the problem. |
21:51.54 | dlynes_office | are you using zaptel? |
21:54.17 | redder86 | yes |
21:54.28 | X-Rob | ?centosbug |
21:54.33 | Qwell[] | ~ |
21:54.40 | X-Rob | ~centosbug |
21:54.41 | jbot | [centosbug] a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
21:54.41 | redder86 | The OS is Linux 2.4.20 on RedHat 9. |
21:54.53 | X-Rob | even |
21:54.58 | Qwell[] | heh |
21:55.08 | Qwell[] | I love how those get incrementally more complex |
21:55.32 | redder86 | what is that centos bug? |
21:55.36 | Qwell[] | ..that |
21:55.48 | X-Rob | I re-wrote to auto-fix the correct file |
21:55.53 | Qwell[] | X-Rob: nice |
21:56.01 | X-Rob | <-- sick |
21:57.03 | dlynes_office | ~redhatbug |
21:57.04 | jbot | redhatbug is probably is a problem with the latest RedHat Enterprise Linux and CentOS kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
21:57.54 | redder86 | why are we messin' with kernel headers? |
21:58.01 | redder86 | are the kernel headers faulty? |
21:58.05 | Qwell[] | becayse they're broke |
21:58.10 | X-Rob | yes |
21:58.17 | X-Rob | spinlocks don't work on RHEL |
21:58.26 | X-Rob | (without that fix) |
21:58.28 | redder86 | and they make zaptel audio quality crappy? |
21:58.35 | Qwell[] | no |
21:58.37 | X-Rob | no, zaptel doesn't complile |
21:58.38 | X-Rob | 8) |
21:58.38 | Qwell[] | they make zaptel not compile |
21:58.45 | redder86 | zap compiles fine |
21:59.53 | redder86 | ls: /usr/src/kernels/2.4.20-8smp-i686/include/linux/spinlock.h: No such file or directory |
22:00.18 | dlynes_office | redder86: what do you get when you run zttest? |
22:00.18 | X-Rob | you're not running centos 4.2 or 4.3 |
22:00.29 | redder86 | I didn't say when I run zttest. |
22:00.38 | dlynes_office | redder86: are you getting any %'ages lower than 99.975? |
22:01.46 | redder86 | --- Results after 27 passes --- |
22:01.46 | redder86 | Best: 100.000000 -- Worst: 99.975586 -- Average: 99.989601 |
22:02.10 | MikeJ[Laptop] | greetings redder86... |
22:02.12 | redder86 | We're using Asterisk as a PRI bridge between the telco and a Patton 2977 for faxing |
22:02.16 | redder86 | MikeJ: hi |
22:02.39 | dlynes_office | redder86: are you sharing any interrupts? |
22:02.54 | redder86 | after upgrading to 1.2 Asterisk will make the audio in the fax call turn very ugly about 1 in 10 calls ... it varies. |
22:03.26 | dlynes_office | redder86: and did you let zttest run for 2 or 3 minutes? |
22:03.29 | redder86 | <PROTECTED> |
22:03.29 | redder86 | <PROTECTED> |
22:03.30 | redder86 | <PROTECTED> |
22:03.30 | redder86 | <PROTECTED> |
22:03.30 | redder86 | <PROTECTED> |
22:03.30 | redder86 | <PROTECTED> |
22:03.32 | redder86 | <PROTECTED> |
22:03.34 | redder86 | <PROTECTED> |
22:03.34 | dlynes_office | redder86: also, have you run patlooptest? |
22:03.36 | redder86 | <PROTECTED> |
22:03.38 | redder86 | <PROTECTED> |
22:03.40 | redder86 | NMI: 0 0 |
22:03.42 | redder86 | LOC: 92637658 92637657 |
22:03.44 | redder86 | ERR: 0 |
22:03.47 | redder86 | MIS: 0 |
22:03.48 | redder86 | 27 passes is much less than 2 or 3 minutes |
22:03.49 | dlynes_office | ummmm |
22:03.52 | redder86 | but remember ... things were fine with 1.0 |
22:03.54 | dlynes_office | ~pb |
22:03.55 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
22:04.36 | redder86 | what is patlooptest? |
22:04.42 | dlynes_office | yeah...things were fine for me with 1.0, too |
22:04.44 | brad_mssw | what version of the zaptel driver are you using ? |
22:04.46 | dlynes_office | but 1.0 was next to useless |
22:04.59 | redder86 | zaptel-1.2.6 |
22:05.03 | robin_sz | redder86, probably a pathlooptest spelt wrong |
22:05.06 | dlynes_office | patlooptest is another utility that doesn't get compiled by default with zaptel |
22:05.34 | dlynes_office | it runs a pattern loop test on each span of your pri crd |
22:05.36 | dlynes_office | it runs a pattern loop test on each span of your pri card |
22:06.03 | dlynes_office | you need a loop module connect to each span you want to test |
22:06.25 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
22:06.47 | dlynes_office | and you need to use clear x-n, where x is the min chan number and n is the max chan number for the particular span (make sure you've got the b chans and d chan commented out) |
22:06.52 | redder86 | dlynes_office: so you had the same kind of problems as I'm having after upgrading to 1.2? screwed up audio (that begins at the beginning from the time the call is first bridged) on only some small percentage of the calls? |
22:06.58 | dlynes_office | then reissue a ztcfg before running patlooptest |
22:07.13 | dlynes_office | redder86: i had issues with the pri in general after upgrading |
22:07.30 | dlynes_office | redder86: i've traced it down to a bad card that never reared its head in 1.0, but did in 1.2 |
22:07.45 | redder86 | you replaced the zap card and all is well? |
22:07.59 | dlynes_office | redder86: i'm going to be replacing hte card within the next day or so |
22:08.07 | redder86 | how do you know it's a bad card? |
22:08.37 | dlynes_office | redder86: because i switched spans, and found out span 1 was totally fubar (patlooptest threw up an endless stream of errors on it) |
22:09.05 | dlynes_office | redder86: and span 2, 3 and 4 i get less than 99.875% the odd time on zttest |
22:09.43 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
22:09.55 | dlynes_office | for all i know, there might not be anything wrong with 2, 3 and 4, but because 1 is bad, it's affecting the other three |
22:09.58 | dlynes_office | i don't really know |
22:10.04 | dlynes_office | i just know it needs to be fixed |
22:10.11 | Dr-Linux | hi again |
22:10.16 | dlynes_office | bye again |
22:10.21 | Dr-Linux | CunningPike: hey |
22:10.25 | Dr-Linux | dlynes_office: bye |
22:10.31 | dlynes_office | Dr-Linux: don't disturb him |
22:10.34 | dlynes_office | Dr-Linux: he's sleeping |
22:10.49 | Dr-Linux | dlynes_office: with whome? :S |
22:10.52 | redder86 | why wouldn't the problem have come up in 1.0? |
22:10.56 | dlynes_office | Dr-Linux: you know how those government employees are |
22:11.03 | dlynes_office | Dr-Linux: they never do any real work :) |
22:11.29 | dlynes_office | redder86: i have no idea, but I can't stay on 1.0, myself |
22:11.31 | Dr-Linux | dlynes_office: i see, i thought Canada guys work hard :P |
22:11.34 | dlynes_office | redder86: it's lacking too many features |
22:12.01 | Dr-Linux | dlynes_home: i got DID's from my telco, but my problem is still there |
22:12.02 | dlynes_office | redder86: so instead, i broke down and bought a sangoma pri card instead |
22:12.10 | dlynes_office | Dr-Linux: so phone up your telco and bitch |
22:12.26 | dlynes_office | Dr-Linux: are you able to dial in to your asterisk box, using one of the dids? |
22:13.28 | redder86 | hehe ... well, that's not going to tell you for-certain that the card was defective or whether the fault was a built-in Digium issue |
22:13.31 | Dr-Linux | dlynes_office: when i call on the DID, it do not come to my server |
22:13.33 | Dr-Linux | ivr1*CLI> pri show span 1 |
22:13.35 | Dr-Linux | Primary D-channel: 24 |
22:13.35 | Dr-Linux | Status: Provisioned, Down, Active |
22:13.42 | dlynes_office | BECAUSE YOU'RE STILL DOWN |
22:14.16 | Dr-Linux | dlynes_office: yes, is it my configuration fault or telco fault? :S |
22:14.18 | dlynes_office | phone up your telco, and tell the idiots that your pri is still down |
22:14.46 | dlynes_office | it looks like it's a problem on their end, to me |
22:15.05 | dlynes_office | but you better hurry up |
22:15.13 | dlynes_office | i think their office might have already closed |
22:15.30 | Dr-Linux | dlynes_home: no they are 24/7 |
22:15.42 | Dr-Linux | i have opened a ticket to the Datacenter |
22:15.45 | dlynes_office | ah...even their business center? |
22:15.53 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
22:16.02 | Dr-Linux | dlynes_home: i asked them 4 question in the ticket |
22:16.06 | dlynes_office | ah |
22:16.15 | Dr-Linux | questions |
22:16.20 | dlynes_office | yeah...our crappy telco's business center is only open 9-6 |
22:16.23 | dlynes_office | m-f |
22:16.48 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
22:16.52 | Dr-Linux | dlynes_office: that's a Datacenter |
22:17.11 | Dr-Linux | dlynes_office: i asked them: |
22:17.11 | dlynes_office | oh...a colo? |
22:17.21 | Dr-Linux | 1. PRI cables connectvity |
22:17.26 | Dr-Linux | 2. PRI numbers/DID's |
22:17.43 | dlynes_office | Forget all that crap |
22:17.48 | dlynes_office | Just tell them your pri is down |
22:18.01 | Dr-Linux | 3. Framing and coding (currently i'm using esf,8bzo something) |
22:18.16 | dlynes_office | yeah...you can tell them your framing and coding |
22:18.18 | Dr-Linux | 4. it's only inbound or outbound as well |
22:18.24 | dlynes_office | Dr-Linux: ummm |
22:18.32 | dlynes_office | Dr-Linux: what else is there besides inbound and outbound? |
22:18.43 | Dr-Linux | i asked them those 4 question |
22:19.06 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
22:19.09 | Dr-Linux | dlynes_home: maybe the allow us only inbound calls |
22:19.18 | dlynes_office | Dr-Linux: ummm |
22:19.23 | dlynes_office | Dr-Linux: YOUR INBOUND IS NOT WORKING, EITHER |
22:19.33 | Dr-Linux | dlynes_home: yes |
22:19.39 | Dr-Linux | dlynes_office: |
22:20.42 | *** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
22:20.47 | Dr-Linux | dlynes_office: they gave me that number beside 22 more numbers |
22:21.46 | *** join/#asterisk philv (n=bleep@cowpig.ca) |
22:21.56 | Dr-Linux | dlynes_home: when i call this number, it says "all circuit are busy please try later" |
22:22.03 | philv | Anyone here ever used an i2004 with chan_unistim? |
22:22.58 | AlexCTI | Hi Dr-Linux, Are you familiar with the varible ${PRIORITY} ? |
22:23.31 | Dr-Linux | AlexCTI: nope, sorry |
22:24.17 | MedozasSVR | can anyone tell me how to turn on debug mode to also see mysql queries been done with asterisk realtime? |
22:24.36 | Dr-Linux | dlynes_office: PRI down/up is also related with framing and coding? |
22:24.59 | dlynes_office | Dr-Linux: don't think so, but then again, i don't know enough about pris, either |
22:25.30 | dlynes_office | philv: nope...how well does it work though? |
22:25.40 | dlynes_office | philv: i'd love to have those working with asterisk |
22:25.50 | Dr-Linux | dlynes_home: CunningPike told me that i can't do anything untill my PRI status is UP and green. |
22:25.53 | dlynes_office | philv: but i don't want to spent $600 to find out it's a paperweight, either |
22:26.08 | dlynes_office | Dr-Linux: isn't that what I've been telling you ALL DAY, too? |
22:26.18 | philv | dlynes_office: Well, I have the i2050 working with no troubles, but my i2004 tries to locate the server, does so, and sends a few bad UDP packets |
22:26.31 | philv | dlynes_office: I'm fortunate, I picked it up on fleaBay for 20 bucks ;) |
22:26.39 | dlynes_office | philv: ah...how well does the i2050 work? |
22:26.43 | Dr-Linux | dlynes_office: yes, but i was still thinking that my configuration is wrong. |
22:26.48 | dlynes_office | that's probably why it was $20 |
22:26.53 | philv | dlynes_office: the i2050 softphone works very well |
22:26.54 | dlynes_office | it probably won't work on a bcm, either :p |
22:27.02 | dlynes_office | oh...it's a softphone, not a real phone |
22:27.03 | philv | Oh it does :P |
22:27.26 | dlynes_office | It does sip though too, doesn't it? |
22:27.26 | philv | The i2050 works well on Asterisk, and the i2004 works on a BCM50 no trobules. |
22:27.28 | philv | *troubles. |
22:27.33 | dlynes_office | or do you need a super expensive module for sip? |
22:27.43 | philv | I think the i2050 doesn't do SIIP |
22:27.45 | philv | *SIP |
22:27.49 | philv | MCS does. |
22:28.11 | dlynes_office | how much does the i2050 cost? |
22:28.13 | philv | Actually, MCS is entirely SIP, iirc, with just a few of our special proprietary options tacked on. |
22:28.18 | philv | That I'm not sure of. |
22:28.30 | dlynes_office | ah....you've got a pirated copy? |
22:28.38 | philv | Haha no no. |
22:28.41 | philv | I work for Nortel :) |
22:37.17 | X-Rob | ~centosbug |
22:37.18 | jbot | centosbug is probably a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
22:38.56 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
22:41.18 | Qwell[] | philv: heathen ;) |
22:42.08 | MedozasSVR | i have one weird thing i have a question for: |
22:42.08 | MedozasSVR | 200/200 192.168.254.37 D 5060 OK (6 ms) |
22:42.08 | MedozasSVR | 100/100 192.168.254.75 D 1720 OK (150 ms) |
22:42.08 | MedozasSVR | SIPGATE_VOIP/1999XXX 217.10.79.9 N 5060 UNKNOWN |
22:42.08 | MedozasSVR | 301/301 192.168.250.25 D 18680 OK (132 ms) |
22:42.46 | MedozasSVR | following: im sitting about 100 km away from this asterisk box - i have a vpn connection to there |
22:43.02 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
22:43.18 | MedozasSVR | my location is 192.168.250.0/24, the other location is 192.168.254.0/24 |
22:43.41 | MedozasSVR | both are completely switched 100 M/bit networks |
22:44.41 | MedozasSVR | when i ping 192.168.254.75, i get a respone time of (from the other location!) 112 ms |
22:45.31 | MedozasSVR | why is the response time from within the network itself (192.168.254.75 <> 192.168.254.0/24) slower than it is even through a vpn from completeley somewhere else? |
22:45.44 | MedozasSVR | the switches are fine |
22:46.08 | MedozasSVR | PING 192.168.254.75 (192.168.254.75) 56(84) bytes of data. |
22:46.08 | MedozasSVR | 64 bytes from 192.168.254.75: icmp_seq=1 ttl=128 time=2.41 ms |
22:46.08 | MedozasSVR | 64 bytes from 192.168.254.75: icmp_seq=2 ttl=128 time=2.42 ms |
22:46.51 | MedozasSVR | and even ping from the >same< machine with the listing of peers |
22:47.07 | MedozasSVR | is looking fine |
22:47.11 | MedozasSVR | anybody can help? |
22:47.32 | X-Rob | sounds like the phone is crappy |
22:48.53 | MedozasSVR | hmmm - but what does asterisk use to measure this? |
22:49.01 | MedozasSVR | not a simple ICMP? |
22:49.08 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
22:49.08 | MedozasSVR | or SIP response time? |
22:49.28 | X-Rob | sip response |
22:49.39 | MedozasSVR | thanks |
22:52.18 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
22:52.31 | *** join/#asterisk sumdingwong (n=sumdingw@adsl-66-137-178-195.dsl.spfdmo.swbell.net) |
22:53.46 | sumdingwong | Does anyone know how asterisk knows to send mwi light off? |
22:54.33 | *** join/#asterisk dant (n=dan@84.9.188.2) |
22:55.12 | sumdingwong | is there anyone else on this channel? |
22:55.33 | *** part/#asterisk sumdingwong (n=sumdingw@adsl-66-137-178-195.dsl.spfdmo.swbell.net) |
22:58.54 | MedozasSVR | here |
23:01.45 | *** join/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com) |
23:02.12 | harpermood | I have a question, regarding my TE110P card. |
23:02.16 | harpermood | I can dial out... |
23:02.25 | harpermood | but when I try to dial in, I get all busy trunks. |
23:02.39 | harpermood | the Telco guy, whom I trust, says everything works up to my card.. |
23:02.49 | harpermood | zttool shows no change to any of those bits. |
23:03.05 | harpermood | how can I get low level debug information from the driver or card to see what signales I have. |
23:03.32 | Dr-Linux | question: i'm doing "zttest" it's doing and doing, so should i stop it manually or it will be stopped by it self? |
23:04.05 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
23:04.47 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
23:13.57 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
23:14.30 | RoyK | <PROTECTED> |
23:15.13 | *** part/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
23:25.33 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
23:28.25 | *** join/#asterisk Eggplant (i=No@dsl-72-19-44-253.cascadeaccess.com) |
23:28.27 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
23:29.52 | harpermood | What does it mean when my Card doesn't return a wink to the telco? |
23:31.00 | Qwell[] | harpermood: that the telco needs to buy it another beer |
23:31.33 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
23:33.16 | *** part/#asterisk Beighto (n=chatzill@64.160.113.130) |
23:36.37 | *** join/#asterisk MoutaPT (n=MoutaPT@a83-132-239-109.cpe.netcabo.pt) |
23:37.01 | MoutaPT | Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event) , combined - 0x1 (telephone-event) |
23:37.18 | MoutaPT | i'm getting this msg in sipdebug with voip provider |
23:37.31 | MoutaPT | any one has any idea what might be wrong? |
23:37.33 | harpermood | any idea why my card wouldn't return the wink to the telco (and I have had plenty of beer ;) |
23:37.54 | file | MoutaPT: why do you think there is something wrong? |
23:38.03 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
23:38.29 | MoutaPT | because after that i get congestion message |
23:38.30 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
23:38.40 | MoutaPT | CHANUNAVAILable |
23:38.45 | P-NuT | Hi all. |
23:38.49 | file | well pastebin the complete sip debug and console output |
23:38.57 | MoutaPT | ok brb |
23:38.57 | CunningPike | Dr-Linux: ctrl-C |
23:39.18 | Dr-Linux | CunningPike: already done |
23:39.30 | Dr-Linux | --- Results after 918 passes --- |
23:39.31 | Dr-Linux | Best: 99.987793 -- Worst: 99.975586 -- Average: 99.976639 |
23:39.41 | P-NuT | Can anyone recommend a cheap 2 port FXO gateway for asterisk? |
23:39.55 | Dr-Linux | CunningPike: someone told me it's bad result |
23:40.14 | CunningPike | Dr-Linux: It's not that bad......... |
23:40.34 | CunningPike | Dr-Linux: It's only a bad result if you are having problems |
23:40.37 | Dr-Linux | CunningPike: i see |
23:41.07 | Dr-Linux | CunningPike: i'm having problems, but not sure what's down :S |
23:41.37 | CunningPike | Dr-Linux: Your PRI is. The problems I was referring to are call quality problems, not a Down ORI |
23:41.48 | CunningPike | s/ORI/PRI/ |
23:42.29 | Qwell[] | P-NuT: tdm400p |
23:44.08 | *** join/#asterisk iq|mobile (n=iq@71-215-58-212.omah.qwest.net) |
23:45.52 | *** join/#asterisk rikstah (n=quirc@213.205.195.254) |
23:47.05 | MoutaPT | file: http://pastebin.com/719777 |
23:47.16 | P-NuT | think so? |
23:47.23 | P-NuT | does it detect without drivers? |
23:47.30 | P-NuT | is it widely supported? |
23:47.32 | Qwell[] | no, you need zaptel instaled |
23:47.34 | Qwell[] | installed |
23:47.53 | MoutaPT | a=rtpmap:101 telephone-event/8000 is this event a problem? |
23:47.58 | file | no |
23:48.08 | file | that's DTMF |
23:48.13 | Qwell[] | only if you think getting dtmf is problematic |
23:48.32 | MoutaPT | no problem with that |
23:48.37 | file | MoutaPT: and you're hearing a congestion? |
23:48.43 | MoutaPT | yes |
23:48.49 | file | it's coming from your provider |
23:48.50 | MoutaPT | i mean |
23:48.52 | file | or wherever this is going |
23:48.53 | P-NuT | hmm.. |
23:49.02 | philv | Any of you kind folks ever use chan_unistim with a Phase I i2004? |
23:49.02 | MoutaPT | not hearing congestion tones |
23:49.10 | P-NuT | anybody have 2 x100p's they want to sell me? |
23:49.11 | MoutaPT | jus no chan available |
23:49.12 | MoutaPT | msg |
23:49.13 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
23:49.18 | *** part/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com) |
23:49.27 | Qwell[] | P-NuT: $75 each |
23:49.27 | file | provide me with the complete console output too |
23:49.31 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
23:49.46 | P-NuT | Qwell[]: is that $AUD? |
23:49.48 | MoutaPT | brb |
23:49.55 | Qwell[] | P-NuT: USD |
23:50.02 | P-NuT | hmm.. |
23:50.12 | P-NuT | Do you have the dodgy asian knock-offs? |
23:50.15 | P-NuT | the clones? |
23:50.25 | Qwell[] | That is a clone |
23:51.03 | *** join/#asterisk oceanlan|dustin (n=info@cpe-69-133-109-130.woh.res.rr.com) |
23:51.24 | oceanlan|dustin | Hey all...I have an outdated version of AAH (Asterisk @ Home) |
23:51.33 | Qwell[] | oceanlan|dustin: see topic |
23:51.40 | oceanlan|dustin | How can I update it in the CLI? where should I look? |
23:51.46 | *** join/#asterisk adker (n=adker@70-97-137-155.dsl1.glv.ny.frontiernet.net) |
23:51.47 | Qwell[] | in the channel topic |
23:51.55 | oceanlan|dustin | ? channel topic? |
23:51.59 | Qwell[] | type /topic |
23:52.13 | oceanlan|dustin | i did.. |
23:52.24 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
23:52.25 | file | read it |
23:52.35 | oceanlan|dustin | nothing is happening.. |
23:52.54 | file | FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
23:53.07 | orlock | Hmm.. |
23:53.11 | oceanlan|dustin | neat. thanks!! |
23:53.15 | orlock | do Cisco's have a habit of locking up with the sip firmware? |
23:53.19 | oceanlan|dustin | what is the /topic thing though? |
23:53.19 | P-NuT | AUD$100 + postage for a clone! |
23:53.21 | P-NuT | damn. |
23:53.23 | Qwell[] | P-NuT: yep |
23:53.35 | Qwell[] | or, just slightly more for a "real" fxo |
23:53.36 | rikstah | orlock not here |
23:53.48 | P-NuT | wouldnt the spa3000 be a cheaper one? |
23:53.58 | Qwell[] | P-NuT: two of them |
23:54.11 | P-NuT | yeah |
23:54.18 | Qwell[] | at $75 each? |
23:54.25 | Qwell[] | $75-1000 |
23:54.28 | P-NuT | touch? |
23:54.33 | P-NuT | hmm... |
23:54.34 | Qwell[] | touch what? |
23:54.38 | P-NuT | LOL |
23:54.41 | P-NuT | never miond. |
23:54.50 | P-NuT | but thanks. |
23:54.54 | P-NuT | haha |
23:55.27 | X-Rob | ~centosbug |
23:55.30 | jbot | centosbug is, like, a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
23:55.36 | X-Rob | Blah. |
23:55.36 | oceanlan|dustin | #freepbx is a dead chat...anyone know where I can find a walkthough on upgrading *@home via the CLI? |
23:55.44 | rikstah | no apostrophy ! |
23:55.55 | X-Rob | oceanlan|dustin, you know, it does say 'wait' |
23:56.39 | oceanlan|dustin | what? says wait? |
23:56.45 | X-Rob | when you joined the channel |
23:57.18 | oceanlan|dustin | ooohhh...I gotcha |
23:58.10 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |