| 00:00.14 | ariel_ | CosmicRay, do you have a windows pc to try out on the network to see if it does the same? |
| 00:00.31 | ruiner | ariel it's a router |
| 00:00.38 | ruiner | i've updated firmware so it has sip options |
| 00:00.44 | ruiner | i think it's just my config is off |
| 00:00.53 | ruiner | and the cisco site is so convoluted |
| 00:00.57 | CosmicRay | ariel_: yeah, I could try that |
| 00:01.11 | CosmicRay | ariel_: what do you think it would suggest if it still has trouble? |
| 00:01.43 | sivana | ariel_: did you get any compile errors? |
| 00:01.57 | ariel_ | ruiner, then it should be easy to configure. But without one here I can't even start to see why it's a problem. Now you can post the setup info you have for it on pastebin.ca like your sip.conf and extensions.conf |
| 00:02.18 | ruiner | well, i think i'm done messing with it for this evening :) |
| 00:02.20 | ariel_ | CosmicRay, either a problem with the asterisk box or network problems. |
| 00:02.24 | ruiner | think i'll go get drunk :) |
| 00:02.28 | Moc | hi all |
| 00:02.37 | ariel_ | sivana, it never works the first time. |
| 00:02.39 | CosmicRay | ariel_: I am quite certain I am not having network problems |
| 00:03.01 | CosmicRay | I have observed, btw, that the gsm codec is far worse with this than ulaw |
| 00:03.16 | CosmicRay | also, that iaxcomm starts the popping sooner and is more consistent, while kphone gets worse over time |
| 00:03.16 | Moc | sivana, I probably asked you already, but you could unblock the # for US also ? |
| 00:03.39 | ariel_ | ulaw is un compressed the gsm is compressed. Then look at the asterisk box to see if your resources are going down. |
| 00:04.07 | Moc | try g726 maybe, it take more ressources than GSM I think |
| 00:04.15 | Moc | but more bandwidth |
| 00:04.15 | sivana | Moc: Yes :) |
| 00:04.22 | sivana | Moc: US/CA same rate |
| 00:04.25 | Moc | ok |
| 00:04.29 | sivana | ariel_: tiffiop.h |
| 00:04.36 | sivana | is that a lib I need? |
| 00:04.54 | Moc | sivana, if you use the lastest package, it doesnt ask you for tiffiop.h |
| 00:05.02 | sivana | which one? |
| 00:05.04 | sivana | 0.02? |
| 00:05.08 | ariel_ | sivana, go to the spandsp site and they have a very good faq section with the correct things to do. |
| 00:05.08 | Moc | yes |
| 00:05.19 | sivana | ariel_: ok, thanks :) |
| 00:05.29 | ariel_ | use .02pre10 |
| 00:05.34 | sivana | Moc: it said beta so I was kinda worried |
| 00:05.38 | sivana | ok |
| 00:05.52 | ariel_ | its all beta look at the release number .02 |
| 00:05.56 | Moc | sivana, .02pre10 is the one Ive used too, it working great |
| 00:06.06 | Moc | sivana, btw dont try spandsp over IAX, it doesnt work |
| 00:06.13 | Moc | but spandsp work fine over SIP |
| 00:06.30 | ariel_ | Moc, I have it working....over iax ..... |
| 00:06.33 | sivana | ya.. it's coming in over TDM anyways |
| 00:06.51 | ruiner | later all |
| 00:06.52 | drsperm | ok...I have asterisk running...and 2 phones associated... |
| 00:06.59 | ariel_ | I have even gotten 80% working via sip and a sat connection. |
| 00:07.12 | drsperm | when I pickup the handset and dial from ext. 103 to 102..I get a busy. |
| 00:07.29 | drsperm | ideas? |
| 00:07.34 | ariel_ | drsperm, are they in the correct context? what does the cli say |
| 00:07.37 | Moc | ariel_, I always get some codec error when I try it via iax provider |
| 00:07.45 | sivana | I should offer a mirror for soft-switch.org... it's painfully slow |
| 00:07.49 | Moc | converting between unknown and ulaw |
| 00:07.56 | drsperm | well, I was trying to be smart and put them in different context...guess not. |
| 00:09.03 | ariel_ | Moc, I am able to use iax for faxing with VoicePulse, VoipJet and nuFone no problems and also race.com but using ulaw only. |
| 00:09.29 | Moc | ariel_, faxing using spandsp ? |
| 00:09.40 | ariel_ | well I take back the voipjet it's sometimes gets cut off. But 99 % of the time it works with them. |
| 00:09.49 | ariel_ | Moc, yes |
| 00:09.51 | drsperm | cool....thanks. |
| 00:09.52 | Moc | I had to loopback myself in SIP for it to work with my iax provider |
| 00:10.32 | sivana | oh.. Moc.. and that's CAD too |
| 00:10.38 | Moc | I'll try it again |
| 00:10.46 | Moc | sivana, ?? what |
| 00:10.50 | sivana | tollfree did |
| 00:11.50 | Moc | it wasnt last time ? ;) |
| 00:12.04 | sivana | not sure.. just clarifying :) |
| 00:12.19 | Moc | well I just want for the register page ;) |
| 00:12.22 | Moc | want = wait |
| 00:12.51 | sivana | :) |
| 00:12.59 | *** join/#asterisk p0lar (~p0lar@sjcc176x190.sjccnet.com) |
| 00:13.12 | buddah | anyone here familiar with SER? |
| 00:13.51 | roamer323 | : ariel - did the test with voipjet's NYC term point go well? thx |
| 00:13.53 | p0lar | Ok, so I approached SysMaster again today |
| 00:13.56 | p0lar | here at VON |
| 00:14.10 | ariel_ | roamer323, it's back up working at least it was 2 hours ago. |
| 00:14.30 | p0lar | Once again -- vehement denial of * use. But, now they claim that the use Digium tdm cards and drivers. |
| 00:14.53 | Moc | urm.... ok |
| 00:14.57 | p0lar | And they sent me a link that they think proves it. |
| 00:15.00 | p0lar | I am amazed. |
| 00:15.16 | roamer323 | ariel_ - thx :-) |
| 00:15.24 | ariel_ | roamer323, np |
| 00:15.37 | p0lar | If I had the time, Iw ould wander back up there and really turn up the heat. |
| 00:15.49 | Darwin35 | ok who has spandsp compiling on fbsd |
| 00:15.53 | Moc | it aint over yet ;) |
| 00:15.56 | Darwin35 | I am having a issue |
| 00:16.08 | p0lar | Moc: Let's hope so, the tone they took with me today -- AS A CUSTOMER -- was appalling. |
| 00:16.12 | p0lar | I would never buy their products. |
| 00:16.14 | Moc | you damn BSD people, always whining.. |
| 00:16.14 | p0lar | Never |
| 00:16.17 | Moc | ;) |
| 00:16.43 | ariel_ | Darwin35, sorry |
| 00:17.06 | Moc | visiting system master website send me to a chat window !! |
| 00:17.21 | p0lar | Here is what I got Googling around it.. |
| 00:17.30 | p0lar | krish: Here is what I got Googling around it.. |
| 00:17.36 | p0lar | krish: http://www.sineapps.com/news.php?rssid=314 |
| 00:17.39 | jakepdev | ariel - does Asterisk H323 with support hookflash? |
| 00:17.45 | p0lar | krish: read it for yourself |
| 00:17.46 | Moc | what ? first it was asterisk, then zaptel, then g729, now spandsp... |
| 00:18.02 | Moc | and I probably missed other some ;) |
| 00:18.03 | *** join/#asterisk wankel (nobody@ohno.mrbill.net) |
| 00:18.07 | jakepdev | should read - does Asterisk with H323 with support hookflash? |
| 00:18.09 | sivana | I needed libtiff-devel |
| 00:18.16 | jakepdev | just can't type tonight |
| 00:18.57 | jakepdev | does Asterisk with H323 support hookflash? |
| 00:19.21 | ariel_ | jakepdev, I don't use h323 in fact I run far away when I hear it. |
| 00:19.43 | jakepdev | does it support SIP with hookflash? |
| 00:19.54 | sivana | rxfax = spandsp? |
| 00:20.22 | Moc | I wonder why we can't make a simple h323 * driver.. |
| 00:20.38 | ariel_ | h323 is not simple that is why |
| 00:21.09 | roamer323 | <- thinks that h323 is the legacy Telco's way of slowing down voip adoption - it worked for a decade - but * and SER changed all of that |
| 00:21.23 | ariel_ | jakepdev, flash hook is mainly for a zap channels in my view. |
| 00:21.59 | jakepdev | I should explain more - I have a Hardware PBX that I want to xfer calls into Asterisk |
| 00:22.08 | ariel_ | ~seen JunK-C |
| 00:22.11 | jbot | junk-c is currently on #asterisk |
| 00:22.11 | jakepdev | then xfer the calls back when their done |
| 00:22.39 | jakepdev | may not be called flash hook in SIP |
| 00:23.48 | Darwin35 | seen god |
| 00:23.56 | Darwin35 | ~seen god |
| 00:23.57 | jbot | god <~xenixgod@19-40.69-92-cpe.cableone.net> was last seen on IRC in channel #debian, 57d 1h 56m 9s ago, saying: 'err I am better than vader'. |
| 00:24.33 | ariel_ | Darwin35, did you not hear there is no GOD.... |
| 00:25.01 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 00:25.06 | CosmicRay | is anyone else having trouble connecting to iaxtel, or is it just me? |
| 00:25.20 | jakepdev | http://www.1800dialgod.com/ |
| 00:25.46 | roamer323 | CosmicRay - everybody in the universe have problem with iaxtel |
| 00:25.52 | CosmicRay | heh |
| 00:25.56 | yaboo | hi trying to call from xlite to a sipura 2000, but xlite keeps stating when dialing the sipura, error 404 |
| 00:26.08 | yaboo | even thou both devices are registered? |
| 00:26.17 | ariel_ | CosmicRay, iaxtel is not 100% up and running it's maybe 80% of the time working. |
| 00:26.19 | roamer323 | CosmicRay - at last count - it was one server and 6000 users |
| 00:26.20 | CosmicRay | roamer323: is there any other service that can connect to toll-free numbers for free? |
| 00:26.26 | Darwin35 | god id the dog that pooped out this planet and called it earth |
| 00:26.27 | CosmicRay | roamer323: yeow |
| 00:26.38 | roamer323 | CosmicRay - use FWD or Sipphone - both great |
| 00:26.39 | CosmicRay | roamer323: let me gues, a pii/500, eh? |
| 00:26.58 | CosmicRay | roamer323: really, FWD can connect to PSTN 1-800 numbers? |
| 00:27.00 | ariel_ | yaboo, use the same codec or put canreinvite=no on the setups for the devices. |
| 00:27.28 | roamer323 | CosmicRay - yes, click around their website - you'll find the info |
| 00:27.30 | yaboo | ariel_, thanks will do |
| 00:27.37 | CosmicRay | roamer323: nice. |
| 00:27.39 | ariel_ | CosmicRay, fwd works for toll free as well. |
| 00:27.50 | Darwin35 | grr it should be compilin |
| 00:27.52 | Darwin35 | gg |
| 00:28.27 | ariel_ | jakepdev, how about 976evil.com |
| 00:28.46 | puppet | anyoe here using oppanel? |
| 00:28.57 | Darwin35 | it 1666deamon1 |
| 00:29.12 | *** part/#asterisk Beave (~beave@vistech.org) |
| 00:29.21 | yaboo | ariel_, sipura uses alaw, and the xlite uses rfc2822 codecs, which codec should I choose, I intend soon to add a cisco 7940 to the system also |
| 00:29.58 | ariel_ | yaboo, use both with alaw or ulaw. cisco uses ulaw as well. |
| 00:30.06 | *** join/#asterisk NoCAT (NoCAT@c-24-9-32-2.client.comcast.net) |
| 00:30.07 | jakepdev | ariel - i tried dialing that but the it said I had to wait an hour for a customer service agent |
| 00:30.10 | NoCAT | hello, |
| 00:30.26 | ariel_ | ROFL |
| 00:30.27 | yaboo | ariel_, can the xlite use ulaw and alaw also? |
| 00:30.33 | ariel_ | yaboo, yes |
| 00:30.39 | yaboo | ariel_, thanks |
| 00:30.42 | ariel_ | it's called g711u g711a |
| 00:35.11 | Primer | Any opinions on the D-Link 1402? |
| 00:35.29 | Primer | $50 rebate on that at newegg |
| 00:35.35 | *** join/#asterisk justinnnn (~dsf@solid.mpa.net.au) |
| 00:35.40 | Primer | comes out to $44 |
| 00:39.29 | puppet | im not getting this to work :/ |
| 00:39.39 | *** join/#asterisk Trepalium (~chadk@wnpgmb02dc1-59-91.dynamic.mts.net) |
| 00:39.41 | puppet | ANyone have had trouble with redirecting calls with op panel? |
| 00:40.22 | puppet | calling form analog phone to ipphone, connecting to internal ipphone, trying to pull the connection to anotehr phone but it disconnects |
| 00:40.46 | ariel_ | puppet, if you go to there web site I think there is a note on this. |
| 00:41.22 | puppet | checking again now |
| 00:42.46 | *** join/#asterisk geekster (~Klenert@pcp08940256pcs.trentn01.nj.comcast.net) |
| 00:43.07 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 00:43.19 | *** join/#asterisk djMax (~djMax@dsl093-190-107.nyc2.dsl.speakeasy.net) |
| 00:43.30 | sivana | Moc: you there? |
| 00:43.39 | *** join/#asterisk mesi (~player@dsl-082-083-145-010.arcor-ip.net) |
| 00:44.03 | djMax | I've mulled on this a bit before, but has anybody out there implemented "snatch call from voicemail"? |
| 00:44.26 | ariel_ | djMax, what is snatch call? |
| 00:44.34 | *** join/#asterisk Luke-Jr (~luke-jr@207.192.219.246) |
| 00:44.35 | djMax | call pickup essentially |
| 00:44.36 | denon | pull it back to your handset |
| 00:44.39 | djMax | right |
| 00:44.43 | Luke-Jr | How can I put multiple channels in a call file? :/ |
| 00:44.47 | denon | in the middle of them leaving a voicemail |
| 00:44.47 | djMax | I think it's important for home apps |
| 00:45.09 | djMax | (so is the Sipura3k supporting call waiting, but that's a whole different mess) |
| 00:45.10 | denon | djMax: semi-common in business scenarios too .. sales people just getting off one call, wanting to take the next |
| 00:45.29 | djMax | true. I think there's some way with the manager api, but not sure if others may have done it. |
| 00:45.41 | denon | yeah .. could definitely do it with the manager api |
| 00:46.01 | denon | not sure how you'd do it otherwise .. aside from maybe some crazy thing like each call having a parked ID while it's in vmail |
| 00:46.24 | djMax | so the process would be deskset dials *1# or whatever, that runs an AGI (?) that calls the monitor api, finds the matching call, and transfers it? |
| 00:46.26 | sivana | well shit... I don't see rxfax.c |
| 00:46.29 | mesi | Luke: Sorry, I have no clue. I'm afraid you can only put one channel in a call file at a time. |
| 00:46.31 | puppet | ariel_: not finding :/ |
| 00:46.41 | djMax | or do I not need AGI |
| 00:46.41 | denon | djMax: sure, that would work .. |
| 00:46.51 | denon | wouldnt really need AGI, but could be an easy way |
| 00:46.57 | ariel_ | puppet, it has to do the incorrect context setup. |
| 00:47.00 | denon | easier than coding the C |
| 00:47.21 | mesi | Luke: Idea: try putting two call files at once in outgoing which both connect to a conference. |
| 00:47.23 | denon | djMax: you could also do something with zapbarge |
| 00:47.34 | denon | and dump the voicemail |
| 00:47.53 | djMax | dump as in ignore? |
| 00:47.55 | puppet | ariel_: aha |
| 00:47.56 | ariel_ | sivana, have you read the readme that comes with the spandsp tar file? |
| 00:48.02 | Luke-Jr | mesi: What channel for a conference? |
| 00:48.03 | denon | well .. stop recording or stop playing the OGM |
| 00:48.18 | sivana | ariel_: yes, doesn't match the site :) |
| 00:48.22 | Luke-Jr | mesi: And everything I've seen suggests I need a Zaptel card for a conf |
| 00:48.43 | mesi | Luke: You have to define one. There are two apps for conferences which I know of, Conference and MeetMe() |
| 00:48.52 | denon | djMax: ultimately, at a glance, I think this is probably something that should be built into voicemail .. callers in your vmail each have a parking ID, if you dial that ID, the vmail connects you to them and backs off |
| 00:48.57 | djMax | is there a corresponding sipbarge? |
| 00:49.09 | mesi | Luke: Ah, no. You can use ztdummy.o instead of a zaptel card. |
| 00:49.45 | *** join/#asterisk NormAst (HydraIRC@70.49.168.83) |
| 00:49.46 | mesi | djMax: What the hell is a sipbarge? |
| 00:49.50 | denon | djMax: to do it right, you'd need to whip out a fair bit of code, I think |
| 00:50.02 | Luke-Jr | mesi: but then, it will dial the destination before the source picks up |
| 00:50.16 | djMax | ok, maybe I'll start by just writing a script to go to the manager api and push a call somewhere |
| 00:50.33 | denon | nod |
| 00:50.41 | djMax | not sure how I would "join" to the person who pressed *1# since they are on the line? |
| 00:51.05 | mesi | Luke: why that? What do you mean? |
| 00:51.08 | MicH323 | Hi all, strange problem with Broadvoice... If I have the secret=PASSWWD in the [Broadvoice] section in sip.conf I dont recieve calls. If I tae it out I rtecieve calls but cant mae them!!! :( |
| 00:51.09 | denon | lots of ways .. could dump them into a meetme, for one |
| 00:51.11 | yaboo | anyone know how to unlock a line in a sipura 2000 |
| 00:51.25 | sivana | hey norm |
| 00:51.37 | puppet | ariel_: a bit closer now ;p |
| 00:51.39 | denon | *1# or whatever drops them into a meetme, then goes out and gets the other call into the meetme too |
| 00:51.40 | djMax | yeah, true. |
| 00:51.43 | denon | dynamic |
| 00:51.52 | puppet | ariel_: but now i just get to main menu ;p not to internal phones but im getting closer ;D |
| 00:52.00 | denon | kinda ugly. . but the best solution isnt easily or elegantly obtainable, I dont think |
| 00:52.26 | denon | heh .. all voicemail could actually be users bridged to a dynamic meetme. . |
| 00:52.30 | djMax | how does call pickup do it? |
| 00:52.31 | denon | and you could hop in at any time |
| 00:52.55 | denon | then all you need to do is drop the vmail leg |
| 00:53.06 | djMax | ooh. interesting thought. so I'd setup some silent dynamic meetme |
| 00:53.14 | mesi | MicH323: Can I sign up with broadvoice for free or will I have to pay? |
| 00:53.18 | denon | nod |
| 00:53.49 | denon | would be handy, in that someone could spy on someone leaving voicemail .. |
| 00:53.53 | denon | without actually taking over the call |
| 00:54.04 | denon | then if they choose to take over the call .. so be it |
| 00:54.51 | sivana | well.. I'm missing something here |
| 00:54.58 | ariel_ | mesi, pay |
| 00:55.00 | djMax | and when they hang up the meetme and the vm will both go away somehow? |
| 00:55.15 | mesi | Ariel: Ah ok. |
| 00:55.23 | NormAst | Hay.. |
| 00:55.53 | denon | djMax: I'm guessing you could get ,h, to handle that |
| 00:56.02 | mesi | Who would like to go to the sipphone.com conference room for a conference test? |
| 00:56.04 | denon | djMax: I'm not suggesting this is the best method .. merely a brainstorm |
| 00:56.13 | djMax | yeah, understood. |
| 00:56.18 | denon | hart attach? that some kind of debugging method? :) |
| 00:56.23 | denon | [heart attack] :) |
| 00:56.56 | ariel_ | I gess 6 servers in my home office is too much. |
| 00:56.59 | djMax | random second question, if you want a dialplan entry to hit a web page and return ok/not ok, what would be the easiest way? agi and wget? |
| 00:57.05 | ariel_ | ./gess guess |
| 00:57.08 | NormAst | Is there a way to have a allowed list of CallerID's in *. I want my iax users only to send CallerID that is approved by us. |
| 00:57.22 | denon | NormAst: yes, google for the blacklist stuff |
| 00:57.29 | NormAst | Thanks. |
| 00:57.38 | harryvv | denon have you ever seen or tried asterisk with a 2 way radio repeater before simular to autopatch on a hamradio repeater? |
| 00:58.07 | mesi | djMax: Easy to use wget with System() call and check for the exit status. |
| 00:58.08 | denon | nope, I havent .. havent had much need to play with ham stuff since the advent of wireless and real bandwidth .. |
| 00:58.11 | puppet | ariel_: Do you have time to explain some thing? |
| 00:58.18 | djMax | ok, thx |
| 00:58.18 | harryvv | sure |
| 00:58.39 | denon | djMax: not sure I'd trust wget to timeout properly etc if something weird happens though |
| 00:58.43 | denon | keep that in mind .. :) |
| 00:58.48 | ariel_ | puppet, maybe I am not feeling too good right now but go for it. |
| 00:58.52 | shmaltz | ~seen tzanger |
| 00:58.54 | jbot | tzanger is currently on #asterisk. Has said a total of 142 messages. Is idling for 3h 16m 58s |
| 00:58.54 | denon | say, a looping 302 redir or such |
| 00:58.58 | djMax | for home that's ok |
| 00:58.58 | puppet | ariel_: http://pastebin.ca/7128 |
| 00:59.10 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
| 00:59.13 | puppet | ariel_: thats how my config file looks like, i can connect to "main" fine but not the internal |
| 00:59.16 | harryvv | I came up with the idea for a company that has a 1k employee base that use alot of radios but also the added cost of cell phones. This would be a way to keep the cell cost down for non sensitive calls across a radio network. |
| 00:59.43 | hardwire | I got the NEC NEAX IPS programming manuals emailed to me |
| 00:59.44 | denon | harryvv: hmm? every employee has a ham license? |
| 00:59.48 | hardwire | over 1000 pages of complete crud |
| 00:59.54 | harryvv | denon no, commercial radios. |
| 00:59.59 | hardwire | why is it that an asterisk box can take care of this in around 50 pages of docs? |
| 01:00.19 | harryvv | Canada largest security agency and is also a global agency in three countries. |
| 01:00.36 | denon | hardwire: because nobody has gotten around to writing the other 950 pages of * documentation that SHOULD exist <G> |
| 01:00.42 | hardwire | hah |
| 01:00.50 | hardwire | well |
| 01:00.52 | ariel_ | puppet, I have to get back to you. I have a small problem here. |
| 01:00.59 | hardwire | I just found out that I need some more hardware just to program this stupid PBX |
| 01:01.07 | harryvv | It would not suprise me if the total employee base was 10 thousand officers. |
| 01:01.07 | puppet | ariel_: ok, hope it solves :) |
| 01:01.11 | hardwire | I think we are going to kill the lease.. sell it.. rip the wire out of the walls and go all VoIP |
| 01:01.18 | *** join/#asterisk mesi (~player@dsl-082-083-145-010.arcor-ip.net) |
| 01:01.22 | mesi | re |
| 01:01.29 | hardwire | bar |
| 01:06.40 | harryvv | mmm |
| 01:06.57 | KalD|Work | anyone know what protocol the MITEL SX-200 speaks for IP stuff? h.323? |
| 01:09.16 | MikeJ[Jayden] | !google:Mitel sx-200 |
| 01:09.20 | tzanger | shmaltz: I'm here |
| 01:09.35 | MikeJ[Jayden] | ~google: MITEL SX-200 |
| 01:09.48 | MikeJ[Jayden] | tzanger! |
| 01:09.52 | tzanger | what |
| 01:09.59 | MikeJ[Jayden] | hello :) |
| 01:10.11 | tzanger | hello :-) |
| 01:10.16 | Darwin35 | Yellow |
| 01:10.33 | Darwin35 | firewire rocks on bsd |
| 01:10.37 | tzanger | I have 8 firewire drives |
| 01:10.45 | Darwin35 | coool |
| 01:10.54 | tzanger | the kernel detects 9 firewire devices (8 + the host card) |
| 01:10.58 | tzanger | but only one fucking drive shows up |
| 01:11.15 | tzanger | and if I force a rescan nothing changes unless I force all scsi channels, at which point I get 8 of the same drive |
| 01:11.24 | tzanger | even if only 4 are plugged in |
| 01:11.35 | Darwin35 | hmm |
| 01:11.48 | tzanger | I'm building 2.6.11 now to see if that helps at all |
| 01:12.15 | *** part/#asterisk eKo1 (~bernd@207.42.191.67) |
| 01:13.19 | tzanger | haha |
| 01:13.20 | tzanger | Ontario Provincial Police are looking for a tractor-trailer that was stolen from a gas station off Highway 401 in Napanee, west of Kingston. The cargo was just peanuts ? 18,000 pounds of shelled nuts in 20 tote bags. The truck's driver was headed for the Kraft Foods processing plant in Montreal when his rig went missing. |
| 01:13.32 | tzanger | first the beer bandit and now this |
| 01:15.04 | harryvv | crooks are generally idiots |
| 01:15.05 | harryvv | hehe |
| 01:15.37 | harryvv | I should sell my aprs as a commercial version to those truck drivers. |
| 01:15.39 | *** join/#asterisk Bacon (~Bacon@thorin.nplus1.net) |
| 01:16.26 | *** part/#asterisk mesi (~player@dsl-082-083-145-010.arcor-ip.net) |
| 01:19.27 | *** join/#asterisk miguellinux (~miguellin@200.47.223.190) |
| 01:20.28 | ariel_ | puppet, this is what the flash operator panel doc's say. Context: if the extension is not reachable from the default context in your dialplan, you should also specify its context. If you have extension number 100 inside the 'from-sip' context, then you should write 100 for the extension and from-sip for the context |
| 01:21.21 | shepherd | when are they going to certify extensions.conf as it's own programming language? |
| 01:21.24 | shepherd | heh |
| 01:21.57 | ariel_ | shepherd, it's own programming lang. It's just a script area. |
| 01:22.09 | shepherd | IT SHOULD BE! |
| 01:22.16 | shepherd | i learn something new everyday! |
| 01:22.21 | Trepalium | Makes it a pain to parse from any non-* tools. |
| 01:22.34 | AgiNamu | Does asterisk go into an infinite loop if you dial your own extension from your own extension? |
| 01:22.36 | ariel_ | shepherd, but the hard part is the agi and other app's. |
| 01:22.54 | mishehu | einal is what it shall be called. it'll stand for einal is not a language |
| 01:23.00 | shepherd | yeah.. agi is easier than extensions.conf (in my opinion) |
| 01:23.00 | ariel_ | AgiNamu, yes sometimes it does. |
| 01:23.07 | mishehu | the e coming from "extensions" |
| 01:23.27 | ariel_ | shepherd, I don't know agi nor perl too much but I can sure do lots in the extension.conf |
| 01:23.43 | shepherd | heh |
| 01:23.49 | puppet | ariel_: have gotten that right now with connection between phones but im loosing the call when im connecting it back to main menu ;p |
| 01:24.03 | shepherd | we should add more crap to "einal" so we can do agi with it |
| 01:24.43 | ariel_ | puppet, put it in debug mode and see what it says. |
| 01:24.50 | shepherd | ariel: btw.. you can use php for agi |
| 01:24.51 | jakepdev | when you say AGI is hard - hard like C programmming - or hard like little harder than configuring * |
| 01:24.52 | shepherd | it's fun |
| 01:25.16 | ariel_ | shepherd, yes I know. I do some agi and perl as well but I am not a programmer. |
| 01:25.34 | shepherd | the best programmers know how the cut and paste! |
| 01:25.40 | shepherd | the = to |
| 01:25.56 | shepherd | congrats! |
| 01:26.04 | jakepdev | shepherd - sounds like you have been using AGI - have you found it to be stable? |
| 01:26.04 | shepherd | i learned agi before everything else |
| 01:26.10 | ariel_ | 3 years ago we had to go to the coded to see what was going on. |
| 01:26.16 | shepherd | and i do mean everything |
| 01:26.32 | shepherd | for the most part |
| 01:26.35 | shepherd | use fastagi! |
| 01:26.41 | jakepdev | I want to throw about 60 simultanious calls at it |
| 01:27.00 | shepherd | hmm.. |
| 01:27.05 | puppet | ariel_: really the first problem is that when i call in, the line says my own phonnumber and not the one that is calling not until i press one and connects to internal phones then it says real phonnumber |
| 01:27.23 | shepherd | agi has always had problems with mass calls |
| 01:27.36 | shepherd | i don't know if it has been fixed yet with fastagi |
| 01:27.39 | jakepdev | what's the limit you would use? |
| 01:27.45 | shepherd | it should be though |
| 01:27.49 | harryvv | anyone seen a case where merdian ivr voice is tripping over its self? Ie before it finishes saying one word the next starts in its place? |
| 01:27.58 | shepherd | it use to be like really low.. like 20 |
| 01:27.59 | shepherd | heh |
| 01:28.07 | shepherd | but i'm sure it's high than that now |
| 01:28.11 | shepherd | er |
| 01:28.35 | shepherd | yeah |
| 01:28.40 | shepherd | are you running head? |
| 01:28.41 | ariel_ | puppet, I just started to use fop a month ago. So I am also new to it. |
| 01:28.42 | jakepdev | is there any other alternative you'd suggest for a production environment? |
| 01:28.57 | ariel_ | but I think there is a channel here for it. called #asterisk-fop |
| 01:29.03 | jakepdev | w/ about 60 calls simul) |
| 01:29.15 | shepherd | c can handle it for sure |
| 01:29.29 | jakepdev | ugh :) |
| 01:29.33 | shepherd | c always has been able to |
| 01:29.35 | shepherd | but! |
| 01:29.40 | harryvv | jak, a low cpu asterisk box has been tested with 700 calls |
| 01:29.40 | puppet | ariel_: but the first look of it it looks good |
| 01:29.49 | shepherd | php should do a good job |
| 01:29.53 | shepherd | same with perl |
| 01:29.57 | shepherd | i think fast agi fixed a lot |
| 01:30.02 | NormAst | 55 Calls very low cpu.. Next to nothing. |
| 01:30.04 | jakepdev | harry - but with AGI? |
| 01:30.11 | NormAst | c... |
| 01:30.36 | *** join/#asterisk zignig (~simon@203.217.15.10) |
| 01:30.41 | NormAst | haryvv: No echo cancellation on the 700 calls. |
| 01:30.53 | shepherd | heh |
| 01:30.57 | jakepdev | Norm - right - but with AGI? |
| 01:31.04 | jakepdev | ok |
| 01:31.08 | *** join/#asterisk madclicker (~icechat5@static-90-68.dsl.tht.net) |
| 01:31.10 | jakepdev | soory - saw the response too late |
| 01:31.33 | NormAst | jakepdev: how many incomming lines will you have? |
| 01:31.37 | madclicker | SOS 7960 SIP firmware required |
| 01:31.47 | harryvv | sorry has to leave for a little no i read this one some site but dont recall all the details. It was load tested with 700 calls. |
| 01:31.55 | jakepdev | Norm - figuring on 60 |
| 01:32.24 | jakepdev | k Harry - tnx |
| 01:32.36 | NormAst | jakepdev: agi should be okay.. BUT... If you want to go bigger then you really need to write a c loadable module. |
| 01:32.42 | ariel_ | I have to go and re-do my test machine. I just killed my hdd. argh I hate it when cheap old drives die. |
| 01:32.50 | sivana | ark |
| 01:32.54 | Bacon | Anyone have any rhel3/centos/wbel rpms of Asterisk? |
| 01:33.17 | madclicker | anyone has a cisco 7960 phone? |
| 01:33.19 | harryvv | I wonder what a 1.8 amd op with 1 gig would do concerning a load. 1500 or greater calls? |
| 01:33.36 | shepherd | bacon: even if rpms were out.. we would suggest you use head |
| 01:33.55 | NormAst | I have looped a quad card at 120 channels on a single machine. It's the echo canneller that really kills the CPU. |
| 01:34.06 | Bacon | shepherd: Why is that? |
| 01:34.12 | tzanger | ariel_: nonsense |
| 01:34.14 | NormAst | More taps.. More cpu. |
| 01:34.18 | tzanger | ariel_: HEAD is almost always stable |
| 01:34.20 | RoyK | ~lart digium for giving horrible support |
| 01:34.21 | harryvv | btw, what ide or sata drives have a really high mtbf ratings? |
| 01:34.33 | jakepdev | it's talking to another PBX a few feet away, will echo cann still be an issue? |
| 01:34.40 | harryvv | no |
| 01:34.51 | NormAst | Shoudl be fine. |
| 01:34.52 | tzanger | harryvv: the ones that say "SCSI" on them |
| 01:35.10 | geekster | does anyone notice a bad echo when calling from one asterisk pbx to another over the PSTN ? |
| 01:35.11 | Trepalium | lol |
| 01:35.13 | NormAst | jakepdev: PRI -> * -> PRI -> PBX? |
| 01:35.22 | Trepalium | SCSI is nice. The price is not. |
| 01:35.26 | jakepdev | PRnorm - exactly |
| 01:35.30 | tzanger | Trepalium: then deal with IDE |
| 01:35.30 | harryvv | echo is more a function of line impedence then anything else. Tzangr a scci drive is a ide drive with scci elctronics on it. |
| 01:35.44 | shepherd | bacon: /whois tzanger |
| 01:35.45 | NormAst | jakepdev: make sure you turn off echocanwhenbride=no |
| 01:35.46 | shepherd | dasdfasdf |
| 01:35.51 | tzanger | Trepalium: seriously... IDE drives in RAID1 should gie you plenty of warning |
| 01:35.56 | shepherd | hehe. ignore that |
| 01:35.56 | tzanger | harryvv: you are *dead* wrong |
| 01:36.13 | jakepdev | tnx - Norm - I'll try that |
| 01:36.17 | Trepalium | I know. Nothing beats being able to hot swap a dead drive in a production system, though. |
| 01:36.22 | tzanger | harryvv: show me a 15krpm IDE drive. Show me an IDE drive with a 5 year warranty |
| 01:36.24 | harryvv | Thats what I have read from some tech documentation. |
| 01:36.45 | tzanger | harryvv: your tech documentation is either wrong or it's talking about the bottom of the barrel SCSI drives |
| 01:36.47 | harryvv | tzanger, ballance is obviosly a issue at those higher speeds. |
| 01:36.53 | tzanger | harryvv: obviously |
| 01:37.05 | tzanger | as are bearings and heat |
| 01:37.18 | jakepdev | oops - Norm - I misread - it's actually PRI -> PBX -> * -> PBX |
| 01:37.51 | NormAst | jakepdev: yea.. set echocanwhenbridged=no |
| 01:37.53 | Trepalium | The media might be exactly or nearly the same, but I imagine the media with fewer errors are reserved for the SCSI drives, whereas the ones with more defects are branded IDE. |
| 01:37.59 | jakepdev | tnx norm |
| 01:38.02 | harryvv | yea i know. I am a prior jet engine aircraft tech were our compressor spools spin at 60k rpms and studied those drives. |
| 01:38.13 | harryvv | those berrings |
| 01:38.14 | harryvv | :) |
| 01:38.25 | tzanger | Trepalium: I think you're mistaken except maybe on the bargain scsi drives |
| 01:38.34 | NormAst | Anyone know if I can set the rxgrain and txgrain when a call is bridged? ie: rxgrainwhenbridge= x and txgrainwhenbridged=x |
| 01:38.34 | tzanger | harryvv: right on |
| 01:38.36 | harryvv | but if anything ballance is very critical at those speeds. |
| 01:39.54 | harryvv | Even a change in the rotor blade weights of a few ounces in our 150 pound roror blades exibit vibration in the airframe. |
| 01:39.55 | NormAst | jakepdev: on your pri you need to set one as pri_cpe and the other as pri_net |
| 01:40.06 | Trepalium | Then again, I had one of our sales people try to tell me that the industry was going to drop SCSI in favor of Serial ATA. He said he was told this by a supplier. I was scared. |
| 01:40.30 | tzanger | harryvv: yup I used to know the physics... it's interesting stuff |
| 01:40.51 | jakepdev | the PRI goes into the PBX - does it still make a difference? |
| 01:40.57 | harryvv | tzanger, helo dynamics is very facinating. Far far more complex then fixed wing. |
| 01:41.03 | tzanger | SATA and even PATA have been getting more and more of the featurs of SCSI (hotswap, TCQ, etc.) but there always seems to be the difference in performance |
| 01:41.06 | RoyK | <PROTECTED> |
| 01:41.14 | RoyK | ~lart digium for giving horrible support |
| 01:41.32 | NormAst | Royk: bad support = Great FREE software. |
| 01:41.37 | RoyK | ~sangoma? |
| 01:41.38 | jbot | extra, extra, read all about it, sangoma is a company that makes PRI cards the way Digium should have done it in the first place.... |
| 01:41.55 | RoyK | NormAst: free as in beer - not as in speech |
| 01:41.58 | tzanger | ~lart royk for giving my chlamydia |
| 01:42.15 | jakepdev | is there open source speech rec? |
| 01:42.15 | RoyK | NormAst: see aefirion for the free and good stuff |
| 01:42.15 | Trepalium | To who? |
| 01:42.21 | RoyK | ~lart tza |
| 01:42.24 | RoyK | ~lart tzanger |
| 01:42.27 | tzanger | hahaha |
| 01:42.41 | RoyK | ~lart tzanger |
| 01:43.18 | tzanger | hahaha |
| 01:43.23 | tzanger | jeez what did I hit a nerve? |
| 01:44.06 | NormAst | okay... enough jbot. |
| 01:44.16 | NormAst | ~jbot dies |
| 01:44.17 | jbot | i do? |
| 01:45.03 | *** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) |
| 01:45.07 | NormAst | Royk: I do like the sangoma Pri cards.. Have 4 of them now. |
| 01:45.11 | Trepalium | What makes Sangoma cards better than Digium's? |
| 01:45.37 | NormAst | normast: 3.3 volt 5 volt. just for starters. |
| 01:46.34 | NormAst | trepalium: They are able to handle PRI timing from different T1 providers on the same system. |
| 01:46.47 | NormAst | without giving you HDLC errrors. |
| 01:47.14 | NormAst | Trepalium: and the one I like the most is..... |
| 01:47.40 | yaboo | constantly getting call failed between a sipura 2000 and xlite, any reasons why? |
| 01:47.47 | NormAst | Trepalium: They don't tell you it's your motherboard when you have an issue and tell you to replace it. |
| 01:47.50 | NormAst | :) |
| 01:48.23 | Trepalium | Okay, given the fact that I'm in the market for 2x T1 line capable card(s), this is good to know. |
| 01:48.29 | *** part/#asterisk NoCAT (NoCAT@c-24-9-32-2.client.comcast.net) |
| 01:49.32 | NormAst | Trepalium: I use the 2x t1 card as a call record for a client.. PRI to * and PRI out to PBX. |
| 01:49.39 | NormAst | recorder |
| 01:51.39 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
| 01:51.45 | tclark | NormAst: :), another convert, .. |
| 01:52.10 | NormAst | yup. |
| 01:52.12 | ManxPower | ~docs |
| 01:52.13 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 01:52.22 | Trepalium | For those cards, channelized, or unchannelized, what's the difference? |
| 01:53.23 | NormAst | Trepalium: You can setup say 10 channels for voice and 13 channels for data, using there wanpipe drivers. |
| 01:55.11 | *** join/#asterisk syslod (~yurplsl@65.114.0.198) |
| 01:55.30 | *** join/#asterisk Newbie___ (some@60.48.49.82) |
| 01:55.42 | Newbie___ | bjohnson: hi |
| 01:55.58 | Trepalium | I see. Thanks. |
| 01:56.14 | syslod | Hello. |
| 01:56.28 | Newbie___ | hi all |
| 01:57.04 | syslod | Anyone know how make the callerid on "non" callerid to be something other than asterisk on inbound call? |
| 01:57.32 | modulus_ | source code? |
| 01:58.03 | yaboo | hmm fixed my problem a bit more but seems I can dial from the sipura to the xlite but not the other way around |
| 01:58.33 | syslod | Isn't there an equiv of setcallerid for inbound? |
| 01:59.41 | *** join/#asterisk mes (~mes@70.66.246.248) |
| 01:59.57 | NormAst | syslod: why not just do a exten => s,1,SetCallerID(1231231323) |
| 02:00.02 | NormAst | exten => s,2,Answer |
| 02:00.05 | NormAst | etc. |
| 02:00.22 | syslod | I tried that it still says "asterisk" when calling |
| 02:00.45 | NormAst | SetCallerIDName |
| 02:00.57 | NormAst | SetCallerName or something like that. |
| 02:01.05 | syslod | This is on an inbound call. Does that work on a inbound call? |
| 02:01.15 | NormAst | should do. |
| 02:03.19 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 02:04.04 | ManxPower | syslod, What is your specific problem? |
| 02:04.15 | NormAst | Anyone know if the grandstream 101 are any better then the BudgeTone-100? |
| 02:05.21 | ManxPower | NormAst, I didn't know there was such a thing as a BT100. I thought there was only a BT101. Anyway, their product sheet will tell you. |
| 02:05.45 | MikeJ[Jayden] | Manx... I can probably get my IRC client to do ~docs every 5 minutes... it would save you some typing |
| 02:05.46 | NormAst | Yea.. I have one of the very FRIST grandstreams.. |
| 02:06.08 | ManxPower | MikeJ[Jayden], Every 30 mins is more than often enough. |
| 02:06.22 | MikeJ[Jayden] | :) |
| 02:06.32 | ManxPower | MikeJ[Jayden], I type it when I join the channel. And any time I see lots of newbies in the channel. |
| 02:07.13 | MikeJ[Jayden] | if only we could get a bot smart enough to di that automatically the first time somone new spoke in the channel |
| 02:10.03 | BrianR___ | Hmm.. The Varion quad pri cards are much cheaper than the digium ones.. |
| 02:12.24 | ManxPower | syslod, s/,1,SetCIDName(Secret Agent) |
| 02:12.46 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 02:13.03 | puppet | Does anyone know a websystem for reconnecting calls etc more then op panel? |
| 02:15.23 | madclicker | anyone has a 7960 cisco phone ? |
| 02:15.38 | ManxPower | madclicker, 23 million people, according to Cisco. |
| 02:16.01 | madclicker | this channel i nean |
| 02:16.04 | madclicker | mean |
| 02:16.55 | madclicker | if only one in a million....:( |
| 02:18.21 | puppet | madclicker: looking for firmware? |
| 02:19.09 | madclicker | puppet: uh-huh, got three of them from the -bay no firmware....buhhhhh |
| 02:19.26 | dsmouse | hey, anyone know of a way for asterisk to get data from ldap |
| 02:19.30 | dsmouse | like dialplan stuff? |
| 02:19.46 | *** join/#asterisk PHILLTH (~email@ool-45734e5f.dyn.optonline.net) |
| 02:19.53 | potter | whats the priority code for no answer? |
| 02:20.14 | potter | n+101? |
| 02:21.18 | shepherd | anyone know if it is possible to transcode ulaw to gsm ? |
| 02:21.26 | NormAst | yes. |
| 02:21.30 | riksta | of course |
| 02:22.02 | NormAst | cost cpu |
| 02:22.55 | PatrickDK | not much, ask asterisk |
| 02:23.05 | PatrickDK | show transcoding,or something like that |
| 02:23.21 | *** join/#asterisk pUmkInhEd (~nospam@s142-179-184-59.ab.hsia.telus.net) |
| 02:23.28 | *** join/#asterisk kks (~kks@203.115.210.253) |
| 02:23.28 | riksta | translations |
| 02:23.55 | BrianR___ | tzanger: Been playing with your norstar / asterisk much lately? |
| 02:25.09 | tzanger | no not at all BrianR___ ... been sick and just no time |
| 02:26.29 | BrianR___ | Aah. |
| 02:27.13 | BrianR___ | This "E News" cable channel has re-enactments of the michael jackson mollestation trial.. |
| 02:27.21 | tzanger | you're kidding |
| 02:27.23 | BrianR___ | From the transcripts. |
| 02:27.24 | tzanger | that is fucking disgusting |
| 02:27.28 | BrianR___ | It's pretty gross. |
| 02:27.57 | Trepalium | Uhg. Terrible... And all for ratings, I imagine. |
| 02:28.01 | PHILLTH | i saw that the other day i thought i was watching mad tv or something |
| 02:28.09 | BrianR___ | <Lawyer Guy> Were you in bed with michael jackson while he was touching your brother's penis while he was masterbating? |
| 02:29.05 | BrianR___ | The actor they have playing michael jackson is hillarious too |
| 02:30.00 | BrianR___ | She has on tons of make up and a very fake looking black wig... |
| 02:30.51 | harryvv | brian thats why most of the time i keep the tv turned off. |
| 02:31.04 | BrianR___ | Almost lost my dinner :( |
| 02:31.37 | harryvv | ohh thats nothing.. I can tell you storis my brother encounters at a hospital when it comes to the mentaly disturbed. |
| 02:32.10 | BrianR___ | TV would be better if there were more naked women and less shows about michael jackson and little boys... |
| 02:32.35 | harryvv | not for this familly man :) |
| 02:32.45 | CoaxD | BrianR: The sad thing is, ratings do not show that to be true |
| 02:32.55 | CoaxD | BrianR: Everybody wants scandals. Dirty, rotten scandals |
| 02:33.03 | CoaxD | involving famous people, especially. they love those |
| 02:33.19 | CoaxD | this is for the same reason that howard stern's radio show took off so good way back when |
| 02:33.28 | BrianR___ | harryvv: Boobs are good for kids... |
| 02:33.40 | BrianR___ | rectal prolapse due to excessive buggering by the king of pop? Not for kids. |
| 02:33.53 | *** join/#asterisk p0lar (~p0lar@dhcp64-134-126-92.sjca.sjc.wayport.net) |
| 02:33.59 | CoaxD | BrianR: Yeah, i hate it when that happens |
| 02:34.00 | drsperm | question, how might I turn down the music on hold...even with the vol on the phone all the way down..it is still loud. |
| 02:34.06 | harryvv | I know of some one that would disagree with that. |
| 02:34.20 | CoaxD | BrianR: Brings back the constant reminder of when michael jackson molested me :( |
| 02:34.39 | jakepdev | he sure does get around |
| 02:34.40 | greg_work | CoaxD: and when there are no scandals.. stick some people in a house, throw around the world "reality" and make them fight |
| 02:34.43 | CoaxD | BrianR: I wasn't sure i should come forward until all these other people did. Now, I feel better about it |
| 02:35.01 | CoaxD | greg_work: Yeah, thats about right |
| 02:35.18 | Trepalium | I stopped subscribing to cable tv because there was just so little worth watching. |
| 02:35.19 | p0lar | Don't forget about the $1M prize |
| 02:35.33 | CoaxD | Trepalium: I watch a few shows. thats it |
| 02:35.45 | *** join/#asterisk Newbie___ (some@218.111.158.18) |
| 02:35.49 | p0lar | Has anyone used any of these USB SIP phones? |
| 02:35.53 | CoaxD | Trepalium: (SG1 and SGA, Judging Amy, The Outer Limits, and a couple others.) |
| 02:35.56 | BrianR___ | netflix was worth the $$ though... |
| 02:35.58 | CoaxD | p0lar: They're *REALLY* not worth it, sir |
| 02:36.16 | CoaxD | p0lar: You're better off spending a few extra bucks and getting an ATA |
| 02:36.27 | p0lar | yeah, I thoguht about that, but I'm in an airport right now |
| 02:36.35 | Trepalium | If I had a Tivo or similar, I would probably find more utility in having cable tv, but I don't. |
| 02:36.44 | p0lar | whipping out my ata + phone is a little.. I dunno.. strange |
| 02:36.46 | BrianR___ | Unfortunately there's a few older flicks I'd like to see that aren't on DVD.. Maybe I'll get the laserdiscs and a player on eBay.. |
| 02:36.53 | CoaxD | p0lar: The only thing they do is generate a sound interface with inputs/outputs that which a phone can understand |
| 02:37.01 | p0lar | ah |
| 02:37.19 | p0lar | I need something compact that can share my wireless connection here |
| 02:37.33 | CoaxD | p0lar: Get an IAXy or somesuch |
| 02:37.39 | CoaxD | p0lar: or a grandstream phone or something |
| 02:37.52 | p0lar | I guess I could piggyback it to the eth port on the laptop |
| 02:38.00 | p0lar | and do nat onto the other interface |
| 02:38.08 | p0lar | or not, since I run through a VPN tunnel |
| 02:38.09 | CoaxD | oh. you could also use a sipura 2100 for that |
| 02:38.21 | p0lar | true, I could just run my laptop into the 2100... |
| 02:38.29 | p0lar | then auth its mac to the server here |
| 02:38.38 | p0lar | or I could pinch someone else's mac when they leave to get on a flight |
| 02:38.41 | p0lar | :D |
| 02:38.48 | CoaxD | heh |
| 02:38.55 | p0lar | or buy a TMobile account.. *puke* |
| 02:39.01 | CoaxD | you could also get a celphone and eliminate the need for the whole mess :) |
| 02:39.10 | Trepalium | Too easy! |
| 02:39.14 | CoaxD | (and forward your DID there via asterisk) |
| 02:39.22 | p0lar | Got one... but my plan isn't friendly for US travel |
| 02:39.32 | CoaxD | p0lar: Get a new plan |
| 02:39.39 | p0lar | c'mon, let me do something geeky, damn it |
| 02:39.42 | CoaxD | p0lar: You need a celphone with nationwide |
| 02:39.49 | CoaxD | p0lar: t-mobile does not suck, btw. |
| 02:40.00 | CoaxD | p0lar: They have good, reliable service. (And no, they're not paying me.) |
| 02:40.03 | CoaxD | p0lar: AHHHHHH. |
| 02:40.07 | tzanger | okay |
| 02:40.07 | tzanger | wtf |
| 02:40.20 | CoaxD | p0lar: That brings things into perspective |
| 02:40.22 | p0lar | T-mobile data services have not been good to me |
| 02:40.24 | p0lar | Yeah... |
| 02:40.34 | tzanger | what would cause asterisk to originate IAX2 frames with out of order framestamps? The call source is zap, going to an iax2 peer |
| 02:40.37 | p0lar | 's ok, I can cope with just having IRC, I didn't feel like being social anyway |
| 02:40.41 | CoaxD | p0lar: GPRS should work okay.. it really depends on what hardware you're trying to do it on |
| 02:40.48 | tzanger | tcpdump on the * box that took the zap call shows OOO framestamps |
| 02:40.54 | CoaxD | tzanger: piss poor network |
| 02:40.58 | tzanger | CoaxD: no |
| 02:40.59 | tzanger | FRAMEstampd |
| 02:41.03 | tzanger | not packet timestamps |
| 02:41.12 | p0lar | security measure |
| 02:41.18 | CoaxD | tzanger: piss poor network would cause anything |
| 02:41.22 | tzanger | as in the iax2 mini frames are being sent out the network out of order |
| 02:41.30 | p0lar | randomized ip ids & frames, hehehe |
| 02:41.30 | AgiNamu | What does "CNG" mean? |
| 02:41.44 | tzanger | CoaxD: I'm recording this on the interface sourcing those packets though |
| 02:41.50 | tzanger | i.e. tcpdump on the * box sending them |
| 02:42.09 | tzanger | and the network card is solid... intel gige running at 10mbit/hd |
| 02:42.14 | BrianR___ | Heh.. Get Verizon's EVDO service :) 2mbps wireless in most major US markets... |
| 02:42.18 | BrianR___ | 144kbps nationwide. |
| 02:42.19 | CoaxD | tzanger: perhaps since * probably does some checking, it perhaps might be retransmitting a packet that wasn't received correctly |
| 02:42.31 | tzanger | CoaxD: it only does that for full frames |
| 02:42.35 | tzanger | these are mini frames |
| 02:42.44 | tzanger | only full frames are ACK'd |
| 02:42.45 | CoaxD | tzanger: I dont know squat about IAX2, man |
| 02:43.09 | CoaxD | tzanger: i'm just saying, its udp. If ASTERISK isn't doing something correctly, i'd classify that as a BUG. |
| 02:43.29 | tzanger | CoaxD: I'd agree but I can't say for sure that that's it yet |
| 02:43.47 | *** part/#asterisk zignig (~simon@203.217.15.10) |
| 02:44.15 | *** join/#asterisk Gronker (~Gronker2@adsl-220-75-161.ags.bellsouth.net) |
| 02:44.50 | tzanger | anyway I'm going to bed |
| 02:44.51 | tzanger | later |
| 02:45.05 | p0lar | damn.. > 750kbit/s through my vpn connection over this wifi connection.. heh |
| 02:45.06 | BrianR___ | tzanger: tty; |
| 02:45.06 | p0lar | sweet. |
| 02:45.44 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 02:45.46 | p0lar | ~300kbit/s up, not shabby |
| 02:45.51 | Mw3 | p0lar: openvpn ? |
| 02:45.55 | p0lar | you know it. ;) |
| 02:46.20 | p0lar | I have two conn-types |
| 02:46.35 | p0lar | first is on 53/UDP and 2nd is http 'connect' proxy-type |
| 02:48.50 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 02:49.53 | FaithX | anyone got acucobol in here? |
| 02:50.54 | potter | whats the priority code for no answer? |
| 02:50.55 | potter | n+101? |
| 02:51.12 | CoaxD | potter: There is no priority code for no answer. It just goes on in context |
| 02:51.51 | CoaxD | tho there might be a flag to tell Dial() to exit with a priority code upon a no answer |
| 02:52.41 | ManxPower | You can check the value of ${DIALSTATUS} to know what happened. |
| 02:53.33 | *** join/#asterisk Damin_Mobile (~pocketirc@112.sub-70-214-23.myvzw.com) |
| 02:54.01 | p0lar | hmnn.. ~105ms.. quake in an airport? is it doable? |
| 02:54.05 | p0lar | bbiab |
| 02:54.56 | CoaxD | that'd be modem latencies |
| 02:54.58 | CoaxD | but i spose |
| 02:55.11 | hardwire | hi |
| 02:57.04 | ManxPower | Always do right. This will gratify some people and astonish the rest. -- Mark Twain |
| 02:57.05 | modulus_ | isn't she lovelyyyy |
| 02:57.10 | modulus_ | this hooooollywood giiiiiirl |
| 02:58.00 | *** join/#asterisk [hC] (~turnerd@69.180.96.113) |
| 02:58.16 | [hC] | Is the res_mysql support found in asterisk-addons only available via CVS? |
| 02:59.30 | ManxPower | [hC], I think that is correct. |
| 02:59.50 | *** join/#asterisk jero (~jero@modemcable040.12-81-70.mc.videotron.ca) |
| 02:59.54 | jero | hello |
| 03:00.16 | modulus_ | jello |
| 03:00.27 | jero | lol |
| 03:00.54 | DyOS | anyone use bellster? |
| 03:01.54 | p0lar | jero: you live in QC? |
| 03:03.20 | jero | p0lar, yes |
| 03:03.29 | jero | p0lar, you too ? |
| 03:03.34 | modulus_ | vnc uses 2 byte encrypted passwords |
| 03:03.45 | modulus_ | wtf |
| 03:03.47 | p0lar | Yep.. well, not right at this moment, but sometime around 10h00 tomorrow I will be, heh |
| 03:04.13 | jero | heh, are you from here or are you coming for an occasional feature ? |
| 03:04.38 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 03:04.45 | yaboo | hi trying to dial from xlite to a sipura 2000, but the debug message tells me that 404 not found from the sipura |
| 03:04.57 | yaboo | the sipura can dial the xlite without fail |
| 03:05.55 | p0lar | jero: haha 1) I know what occasional feature in QC means, but 2) no, I live there.. heh.. MTL that is |
| 03:06.05 | jero | lol |
| 03:06.15 | jero | mtl too |
| 03:06.22 | p0lar | jero: still nasty weather? |
| 03:06.54 | jero | quite cold, around -20/-30c today, some snow around. but the next days will be sunny |
| 03:07.14 | Qwell | jero: care to trade? |
| 03:07.24 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 03:07.26 | jero | trade what ? |
| 03:07.30 | Qwell | weather |
| 03:07.42 | p0lar | didn't call the wife today.. d'oh |
| 03:08.08 | p0lar | cat /dev/random | excuse > /dev/wife | grep "COUCH" && locate couch |
| 03:08.14 | jero | where are you in, p0lar and Qwell |
| 03:08.15 | Qwell | jero: its about 30c here |
| 03:08.15 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 03:08.19 | jero | p0lar, lol |
| 03:08.22 | p0lar | jero: San Jose, CA |
| 03:08.32 | Qwell | a bit further south then him |
| 03:08.42 | jero | in mexico ? |
| 03:08.49 | Qwell | just a tad higher |
| 03:09.04 | jero | okay :) |
| 03:09.30 | p0lar | jero: I get in ~10h20ish tomorrow morning and have to work all day. :( |
| 03:09.40 | jero | rude |
| 03:09.42 | p0lar | flying all night.. it'll suck |
| 03:09.51 | [hC] | Im reading some of the RealTime pages on voip-info. they seem to contradict each other alot. One says something (I think?) about not being able to support nat keepalives or MWI when you use RealTime modules. Another says you can store some things in sip.conf and some in sipfriends, another says you cant mix the two.. Which one is right? |
| 03:10.04 | p0lar | hate west coast -> east coast flights |
| 03:10.46 | jero | they make me knocked-out |
| 03:10.57 | *** join/#asterisk krilloz (majestic@220-253-7-238.VIC.netspace.net.au) |
| 03:11.04 | p0lar | first leg is 5.5 hours.. |
| 03:11.29 | p0lar | I'm hoping it isn't cramped |
| 03:11.33 | jero | :) |
| 03:12.11 | Eight | Well... I swear I'm miss-using the Dial command. |
| 03:12.12 | Grooby | w000t!!!!! |
| 03:12.15 | Grooby | got HD3000 to work! |
| 03:12.55 | ManxPower | [hC], As I understand it, sipfriends is a 1.0.x thing, whereas Realtime is a CVS-HEAD thing (at least right now) |
| 03:13.22 | Eight | The MeetMe conference is working fine. But when I have "exten => 314,1,Dial(SIP/me)" it rings, and answers, but there's no sound. |
| 03:13.50 | Juggie | [hC] neither |
| 03:13.52 | *** join/#asterisk soundguy (~soundguy@soundguy.id.au) |
| 03:14.21 | Juggie | you can use MWI with realtime now, but you need to get the lastest version of the sip config file |
| 03:14.29 | ManxPower | Eight, You don't have a bandwidth= line or an allow=all in sip.conf, do you? |
| 03:14.49 | Eight | I have allow=gsm, ulaw, alaw. |
| 03:14.53 | Juggie | and in there, there is a flag to put the database of sip peers into asterisk memory structures |
| 03:14.55 | Eight | ManxPower: the thing is, we can talk FINE in meetme. |
| 03:14.58 | Juggie | thus making MWI work |
| 03:14.58 | *** join/#asterisk JustinSan (~just@user-11216cl.dsl.mindspring.com) |
| 03:17.23 | puppet | lol i have bashquotes now |
| 03:17.30 | puppet | festival is semifun |
| 03:17.30 | puppet | ;p |
| 03:17.52 | [hC] | Ah I see where im getting confused. I dont understand the difference between a database peer/user and a static peer/user. Time to dig more. :) |
| 03:18.17 | [hC] | I thought a database peer/user meant a peer/user that is stored in database config, but that doesnt seem to be the case |
| 03:18.22 | *** join/#asterisk cero64 (ruiner@fantab.ulo.us) |
| 03:18.37 | Eight | Is it possible there's some issue with the NAT that only crops up when Asterisk dials out to a SIP client, but doesn't present a problem when all the dialing is done by the SIP client? |
| 03:18.41 | Juggie | [hC] originally with realtime 'sip show peers' at the CLI woudnt show you the peers in the database |
| 03:18.42 | Juggie | however |
| 03:18.51 | Juggie | there is a new setting to change that |
| 03:18.57 | Juggie | which now makes * aware of its peers |
| 03:19.04 | Juggie | (instead of just when they are needed) |
| 03:19.09 | Juggie | and hence makes it all work |
| 03:19.17 | ManxPower | Eight, Is Asterisk behind NAT? |
| 03:19.41 | Eight | ManxPower: Yes. But 5060 and 10k-20k are forwarded, and ExternIP is set in sip.conf |
| 03:19.52 | ManxPower | Eight, localnet= too? |
| 03:19.55 | Eight | ManxPower: yup. |
| 03:20.06 | ManxPower | Eight, Are you SURE the SIP client is set for 10k-20k too? |
| 03:20.56 | yaboo | dailing from xlite to sipura 2000 now get 403 forbidden, anyone know why? |
| 03:20.56 | Eight | the SIP client is listening on 8k (X-lite), but that shouldn't require forwarding on the Ast' end... |
| 03:21.04 | Eight | yaboo: user/pass? |
| 03:21.16 | yaboo | Eight on which client |
| 03:21.35 | Eight | yaboo: Sipura 2000. |
| 03:21.39 | yaboo | Eight, both register, and the sipura 2k can dial the xlite without any problems |
| 03:22.27 | Eight | yaboo: SIP seems to use HTTP style errors. 404 not found, 403 forbidden. 403 forbidden usually implies that the resource exists (you're specifying it properly) but isn't available to you (for whatever reason. User/pass, or just not allowed at all). |
| 03:22.50 | Eight | yaboo: er, wait. You're going through asterisk right? |
| 03:23.18 | yaboo | Eight or errors on the debug I get when the xlite dials the sipura 2k are |
| 03:23.31 | Eight | yaboo: And they can both do the echo test fine? |
| 03:23.46 | yaboo | Got sip response 404 "Not found" back from 137.172.63.147 |
| 03:24.02 | yaboo | SIP/3004-a83f is circuit-busy |
| 03:24.16 | yaboo | Everyone is busy/congested at this time |
| 03:24.26 | yaboo | Eight echo test? |
| 03:24.28 | Eight | Sounds like your destination isn't currently registered. |
| 03:24.40 | yaboo | the sipura 2k |
| 03:25.21 | Eight | put what I just pasted into the context the SIP connections end up in. |
| 03:25.42 | Eight | Assuming it's not the demo context already, which has that. |
| 03:25.59 | Eight | Then dial 600 with each device. Make sure they devices can talk to Asterisk, by themselves. |
| 03:26.18 | yaboo | the sipura 2k needs nothing special for it to work |
| 03:26.29 | yaboo | only line 1 is domain locked thou |
| 03:26.34 | Eight | If one of them doesn't hear the announce, or get the echo back, then you can narrow down your issues. |
| 03:26.44 | yaboo | ok |
| 03:26.49 | puppet | damn so useless function i did |
| 03:26.59 | puppet | why do u EVER want loosy bashquotes in your phone? |
| 03:27.00 | puppet | lol |
| 03:27.21 | Eight | yaboo: keep in mind, you're taking advice from someone who can't get two things to dial eachother either =) |
| 03:28.02 | Eight | well, the dialing works. it's the sound after the pickup that doesn't =/ |
| 03:29.00 | *** join/#asterisk jmhunter (~jmhunter@64.77.199.223) |
| 03:29.00 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 03:29.01 | yaboo | Eight seems the xlite echotest works |
| 03:29.31 | yaboo | Eight, echo test on sipura works |
| 03:29.38 | Eight | oh god. |
| 03:29.47 | yaboo | what Eight |
| 03:29.47 | Eight | Apparently Dial(SIP/name) doesn't work. |
| 03:29.54 | Eight | I stuck a 30s timeout and Tt on it and it works. |
| 03:30.12 | Eight | I spent hours last night trying to fix what I thought were NAT issues =/ |
| 03:30.25 | modulus_ | did you try nat=yes? |
| 03:30.28 | modulus_ | in your sip.conf |
| 03:30.32 | Eight | modulus_: I wasn't having nat issues =) |
| 03:30.39 | modulus_ | did you try forcing ulaw? |
| 03:31.01 | modulus_ | did you try canreinvite=no |
| 03:31.13 | Eight | modulus_: is that a bot script? =p |
| 03:31.15 | modulus_ | 'cause i don't even know what kind of device you have |
| 03:31.23 | modulus_ | no i just type really fast |
| 03:31.27 | modulus_ | and you're just really slow |
| 03:31.40 | Eight | modulus_: well, you're ignoring what I'm saying. |
| 03:31.47 | Eight | modulus_: I am not, and never was, having NAT issues. |
| 03:31.54 | modulus_ | *shrug* |
| 03:32.06 | modulus_ | Dial(SIP/name) works for me |
| 03:32.13 | modulus_ | sip.conf: |
| 03:32.16 | modulus_ | [1000] |
| 03:32.21 | modulus_ | username=whateveruser |
| 03:32.27 | modulus_ | Dial(SIP/1000) |
| 03:32.30 | modulus_ | there |
| 03:32.50 | Eight | well, Dial(SIP/name) rang the user, but had no sound. |
| 03:32.58 | Eight | Dial(SIP/name,30,Tt) rang the user, and had sound. |
| 03:33.19 | modulus_ | did you try canreinvite=no in your sip.conf? |
| 03:33.35 | Eight | nope. |
| 03:33.44 | jmhunter | hey did we ever decide if the sipura 3000's fxo port is *able |
| 03:33.46 | modulus_ | that's another option you can mess with |
| 03:33.49 | modulus_ | in the sip.conf |
| 03:34.00 | modulus_ | just fuck around with all the options until it works |
| 03:34.10 | Eight | modulus_: I did, and now it does =) |
| 03:34.10 | modulus_ | b/c you're obviously not going to read what all the options do |
| 03:34.19 | modulus_ | that fixed it? |
| 03:34.29 | Eight | modulus_: I *have* been reading what the options do. |
| 03:34.37 | modulus_ | canreinvite works? |
| 03:35.14 | Eight | I've spent hours reading the Wiki, and actually 'solved' my problem without suggestions from you or anyone else. Don't give me crap for not RTFM. |
| 03:35.23 | modulus_ | what fixed it? |
| 03:35.33 | Eight | You're the one who isn't listening. |
| 03:35.35 | dsmouse | what was the problem? |
| 03:36.36 | modulus_ | it should work w/o using the ,30,Tt |
| 03:36.38 | Eight | dsmouse: When two parties go into a MeetMe conference they can hear eachother fine. When one dials the other with "Dial(SIP/name)" it rings, connects, but no sound. |
| 03:36.53 | Eight | I switched it to "Dial(SIP/name,30,Tt)" and now it works. |
| 03:37.06 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 03:37.06 | Juggie | never had that problem here. |
| 03:37.07 | modulus_ | what happens when you remove 30? |
| 03:37.12 | modulus_ | juggie, me neither |
| 03:37.20 | dsmouse | weird |
| 03:37.21 | Eight | modulus_: I haven't gotten that far, yet. |
| 03:37.23 | SexyKen | Anyone here know of any good syncronization programs for Windows? |
| 03:37.32 | Eight | modulus_: been busing being defensive in here =p |
| 03:37.35 | modulus_ | sexyken, FreeBSD works really well |
| 03:37.38 | [hC] | What exactly is a 'database channel' |
| 03:37.54 | SexyKen | modulus -> Eat my ass. |
| 03:38.12 | modulus_ | sexyken -> freebsd will sync your programs perfectly |
| 03:38.34 | SexyKen | modulus -> FreeBSD isn't a program either. |
| 03:38.38 | Eight | modulus_: Ah, Thanks for pointing me at canreinvite. |
| 03:38.47 | SexyKen | modulus -> And it isn't for windows. |
| 03:38.54 | Eight | I see why Tt fixed it. |
| 03:38.56 | modulus_ | sexyken -> your arrow notation sucks |
| 03:39.01 | Eight | Or, I suspect. |
| 03:39.02 | modulus_ | eight, you're welcome |
| 03:39.21 | SexyKen | modulus -> Would you like a spoon? |
| 03:39.31 | modulus_ | sexyken -> there is no spoon |
| 03:39.50 | SexyKen | rEtard. |
| 03:40.10 | modulus_ | jbot nickometer SexyKen |
| 03:40.26 | SexyKen | jbot nickometer modulus_ |
| 03:40.34 | modulus_ | jbot lart SexyKen for being an 31337 rEtard. |
| 03:41.21 | scrubb | lart? |
| 03:41.27 | modulus_ | jbot lart? |
| 03:41.28 | jbot | methinks lart is Luser Attitude Re-adjustment Tool |
| 03:41.36 | scrubb | lol |
| 03:44.13 | Juggie | jbot nick Juggie |
| 03:44.27 | Juggie | jbot nickometer Juggie |
| 03:46.14 | cero64 | jbot nickometer ruienr |
| 03:46.21 | cero64 | jbot nickometer ruiner |
| 03:46.34 | cero64 | yay, even with a spelling error it's not latme |
| 03:46.44 | cero64 | shit, i really can't type today |
| 03:47.19 | jmhunter | wheres the wiki these days |
| 03:48.04 | shepherd | not helpful |
| 03:48.05 | shepherd | hehe |
| 03:49.11 | jmhunter | oops, there it is... anyone know baout race.com |
| 03:49.29 | shepherd | anyone want to conf? |
| 03:49.30 | shepherd | heh |
| 03:52.24 | ManxPower | If you are going to play with the bot it's polite to do so in private and wash your hands after. |
| 03:54.05 | jmhunter | hey manx |
| 03:56.40 | *** join/#asterisk Mneumonic (Mnemonic@ool-18ba58b4.dyn.optonline.net) |
| 03:57.15 | Mneumonic | hey, im trying to set up overhead paging thru oss...i got the extension to auto-answer but anything i say into the mic isnt comin out the speakers... any help on this? |
| 03:57.48 | ManxPower | Mneumonic, Um, overhead paging is EASY. |
| 03:58.15 | Mneumonic | well i got extension 880 dialing CONSOLE/dsp |
| 03:58.25 | Mneumonic | and it answers, but no sound |
| 03:58.31 | Mneumonic | so i i guess im retarted |
| 03:58.42 | ManxPower | Can you play sounds via that device? |
| 03:59.02 | Mneumonic | never tried, assumed linux picked up the sound card... how can i test? |
| 03:59.18 | Mneumonic | im not very linux savvy, but i am getting the hang of it |
| 03:59.19 | cero64 | try using mpg123 to play an mpg |
| 03:59.22 | ManxPower | Mneumonic, Let me find my configs to post. |
| 03:59.45 | ManxPower | Mneumonic, You really need to make sure sound is working without Asterisk. |
| 04:00.09 | Mneumonic | trying mpg123 now |
| 04:01.12 | spackle | you should be able to dial an extension from the soundcard and hear it just like a phone |
| 04:01.51 | Eight | How long does it usually take for Broadvoice to kick in? |
| 04:01.57 | Eight | registering a new account. |
| 04:02.23 | Mneumonic | yes |
| 04:02.26 | ManxPower | Mneumonic, http://pastebin.ca/7147 |
| 04:02.30 | Mneumonic | playing a mp3 now and it works |
| 04:02.38 | Mneumonic | thanx |
| 04:02.43 | *** part/#asterisk Popdog (daniel@edtn014064.hs.telusplanet.net) |
| 04:03.18 | modulus_ | eight, i don't recommend broadvoice |
| 04:03.27 | Eight | modulus_: too late =p |
| 04:04.35 | Mneumonic | ManxPower - copies your settings and it doent work... :( |
| 04:04.43 | *** join/#asterisk cf_man (~cfman@c-67-176-47-241.client.comcast.net) |
| 04:04.48 | Mneumonic | OSS/dsp andwered... but no sound |
| 04:06.26 | cf_man | greatings everyone |
| 04:08.20 | p0lar | Can anyone get to www.openh323.org? |
| 04:08.58 | cero64 | sup cf_man |
| 04:09.46 | cf_man | just installed slackware on a fresh box and wanted to get asterisk on it |
| 04:10.35 | cero64 | www.asterisk.org |
| 04:10.47 | cf_man | wgetting the files now |
| 04:10.48 | cero64 | more specifically, http://www.asterisk.org/index.php?menu=download |
| 04:11.00 | cf_man | got it |
| 04:11.04 | cf_man | thanks |
| 04:11.10 | cero64 | pretty simple as far as installing, gunzip, tar -xf, cd into directory, make, make install |
| 04:11.21 | cf_man | sweet |
| 04:11.28 | cero64 | configuration is another issue :) |
| 04:11.43 | *** join/#asterisk Speer (~Speer@pool-70-20-123-87.pitt.east.verizon.net) |
| 04:11.55 | Eight | now i've gone and forgotten, is it the zaptel stuff or asterisk itself that wants %make linux26; for that kernel? or both? |
| 04:12.26 | Qwell | Eight: its not needed anymore, I don't think |
| 04:12.37 | modulus_ | php pastebin sucks |
| 04:12.51 | Speer | Hi, Can anyone tell me if asterisk will work on sun microsystems' architecture? |
| 04:13.07 | cf_man | any really cool options I should looking enabling |
| 04:13.09 | Eight | Qwell: oh? I was having troubles with something yesterday, that got resolved by make linux26... or maybe that's because /sbin wasn't in my path ::curses fresh install:: |
| 04:13.11 | shepherd | speer: yes it will |
| 04:13.48 | Eight | Speer: The software itself runs on just about any unix (I ran it earlier on Mac OS X). Support for the zaptel cards is a different matter. |
| 04:13.52 | Qwell | Eight: dunno, I was looking at a Makefile the other day, and it did the check on its own if you do just make |
| 04:14.05 | Eight | Qwell: cool. I'll stop mentioning it then =) |
| 04:14.22 | Mneumonic | ManxPower - It seems i am loading the module chan_oss.so to handle the paging... i see u use alsa, so what module do i need to load to use alsa? |
| 04:14.40 | Speer | I have a Cobalt Qube laying around I was gonna try it with just out of curiosity, Thanks for your help! |
| 04:14.40 | Qwell | chan_alsa |
| 04:15.06 | Mneumonic | is it just chan_alsa? or chan_alsa.so? |
| 04:15.08 | shepherd | i think someone made it work with sunos |
| 04:15.15 | shepherd | zaptel |
| 04:16.23 | Mneumonic | loanding chan_alsa or chan_alsa.so failed.. grrrr |
| 04:19.33 | cf_man | cer064 do you recomend A.M.P for a management platform? |
| 04:20.27 | *** join/#asterisk jawong (~jawong@adsl-67-114-131-119.dsl.sntc01.pacbell.net) |
| 04:21.34 | niZon | I wonder if this will get me access to the cisco SIP firmware: http://www.insight.ca/apps/productpresentation/index.php?product_id=CIS411873 |
| 04:21.49 | jawong | sorry to be a bother - but where's the url for how to connect asterisk to a standard phone line? (ie, give it a real telephone number). I'm looking for the entire guide |
| 04:22.30 | Qwell | ~docs |
| 04:22.31 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 04:22.35 | Qwell | jawong: somewhere in there |
| 04:22.56 | jawong | excellent - thx! |
| 04:25.19 | wildcard0 | hmm. anyone try iaxcomm? |
| 04:25.55 | *** part/#asterisk jawong (~jawong@adsl-67-114-131-119.dsl.sntc01.pacbell.net) |
| 04:26.13 | Qwell | wildcard0: yeah, works good for me |
| 04:27.04 | wildcard0 | i can't see to dial up a softphone. it registers ok and i can dial into the asterisk box, but doing iax2 show peers shows it as unreachable. should i not be using a qualify statement? |
| 04:27.08 | *** join/#asterisk Thelrax (~Thelrax@cpe-67-11-252-96.satx.res.rr.com) |
| 04:27.45 | *** part/#asterisk Thelrax (~Thelrax@cpe-67-11-252-96.satx.res.rr.com) |
| 04:29.08 | *** join/#asterisk mog_home (~mog_home@146.229.176.173) |
| 04:29.19 | p0lar | Wow.. SJPhone -> h323 -> openvpn tunnel -> wireless -> http proxy -> internet -> openvpnserver -> softswitch at the office and it works. :S |
| 04:29.43 | wildcard0 | damn |
| 04:29.48 | wildcard0 | that's a lot of arrows |
| 04:31.51 | *** join/#asterisk Funbags (~Funbags@ool-18be223d.dyn.optonline.net) |
| 04:32.54 | Funbags | anyone know why or how to fix asterisk from not passing the caller id of a incoming call from broadvoice to a sipura device ( I see the caller id in the debug, so its getting sent from broadvoice) |
| 04:35.37 | shepherd | http://www.digium.com/handbook-draft.pdf is great! |
| 04:37.58 | puppet | cf_man: ampf? |
| 04:38.05 | puppet | cf_man: AMP* ? |
| 04:39.44 | slePP | who knows anything about thermodynamics? |
| 04:39.53 | shepherd | heh |
| 04:39.56 | shepherd | i hated that class |
| 04:40.13 | slePP | well, my question is: is heat transfer rate proportional to the temperature differential of materials? |
| 04:40.24 | drsperm | Could anyone tell me how to reduce the volume on the music on hold... |
| 04:40.26 | slePP | or is it a constant rate as it nears equilibrium? |
| 04:42.20 | Beirdo | oh good God |
| 04:42.31 | Beirdo | the one class I blocked from my memory |
| 04:42.56 | Funbags | proportional from what i recall |
| 04:43.03 | Beirdo | and, incidentally, the one class with my lowest mark ever :) |
| 04:43.35 | Mneumonic | anyone know why the chan_alsa.so module wouldnt be in the modules directory? i successfully installed alsa... |
| 04:43.41 | *** join/#asterisk bandrew (~Snak@c-67-184-114-237.client.comcast.net) |
| 04:45.10 | slePP | Funbags: so -100C and 0C will equalize at the same rate 100C and 200C will (say in a fluid), but -50 and 0 will equalize at a rate that is not exactly 50% of -100 to 0? |
| 04:45.58 | shepherd | slepp: i'm going to give you an answer |
| 04:46.02 | shepherd | but it probably won't be right |
| 04:46.02 | Funbags | slePP, outa my league dude :) |
| 04:46.47 | slePP | it's really a fairly simple question, i just don't know the answer at all. given a two fluids of different temperatures, will half the initial temperature difference equate to exactly half the time needed for double the initial temperature. or something ;> |
| 04:46.53 | bandrew | Hi folks, I'm looking to buy a switch for my VOIP asterisk network.. something reliable that won't break the bank. Anyone ever heard of TRENDnet? They have a QoS switch for $250. Or can anyone else recommend a good one in that price range? |
| 04:46.57 | *** join/#asterisk alexns (~alex@acs-24-154-114-15.zoominternet.net) |
| 04:47.07 | slePP | 0 + 100deg = 15 minutes, 0 + 50 deg == 7.5 minutes |
| 04:47.10 | slePP | true or false :> |
| 04:47.38 | shepherd | poe switch? |
| 04:47.43 | bandrew | not poe |
| 04:47.54 | shepherd | so just a swtich |
| 04:47.59 | bandrew | POE would be nice but the ones I saw started at around $1000. |
| 04:48.02 | shepherd | like.. basic networking switch |
| 04:48.04 | bandrew | A switch with QoS |
| 04:48.05 | alexns | anyone interested getting xcapi to work with oh323 interface in asterisk ?? |
| 04:48.10 | shepherd | oh |
| 04:48.10 | shepherd | hah |
| 04:48.12 | alexns | will pay |
| 04:48.20 | bandrew | you need QoS, right? |
| 04:48.45 | slePP | okie, well, i'm gonna run off and try to figure this out.. i just thought about it on the way home from dinner and now i can't stop thinking about it :> |
| 04:49.35 | *** join/#asterisk soundguy (~soundguy@soundguy.id.au) |
| 04:49.44 | alexns | <PROTECTED> |
| 04:50.07 | shepherd | yuck, is all i can say |
| 04:51.00 | bandrew | wow, you can get a Gigafast QoS switch for $70. Why do Cisco switches start at $700? What's so great about them? |
| 04:51.22 | alexns | cisco name |
| 04:51.36 | denon | BS. they run a real OS |
| 04:51.41 | alexns | true |
| 04:51.44 | denon | there's tons of functionality in a catalyst .. |
| 04:51.52 | denon | if all you need is a big Y adapter, get the gigafast crap |
| 04:52.15 | denon | if you need powerful vlan stuff, lots of spanning, trunking, nice snmp, etc |
| 04:52.20 | denon | go cisco |
| 04:52.44 | alexns | ios upgrades suck, |
| 04:53.00 | alexns | too bad they always want that service contract... |
| 04:53.21 | bandrew | denon: what's spanning trunking or snmp? Are they important for a phone system? |
| 04:53.29 | drsperm | I love Cisco..but I must say...the high end Dell's are nice (I think by Data Foundry) |
| 04:53.35 | denon | google |
| 04:54.09 | cero64 | i wish i could figure out how to get my cisco to work right |
| 04:54.30 | drsperm | what is the issue? |
| 04:54.35 | alexns | denon, do you know of any decent cisco telephony dealers? |
| 04:55.05 | cero64 | drsperm: i can't figure out how to get it to work with asterisk so that if i dial an extension i will dial out from my router |
| 04:55.07 | channan | It's all about what you want.. Cisco has great stuff with high price... I used to not like it but I started getting like it now |
| 04:55.14 | alexns | i am in pa, its hard to get them to return a call |
| 04:55.31 | cero64 | i have a 3640 with two fxo ports in it, one hooked into an analog line with all calls from that line going to asterisk box, other port plugged into an fxs |
| 04:55.58 | cero64 | so i'm basically, for testing, trying to dial into the analong line, then hit an extension that will then let me dial out to whatever number i want |
| 04:56.06 | drsperm | yeah...sorry...I am a bit new to the voice side... |
| 04:56.20 | cero64 | calls get into asterisk fine, but i think i need some weird conf in the router to allow incoming sip connections to dial back out |
| 04:56.52 | *** join/#asterisk alexns (~alex@acs-24-154-114-15.zoominternet.net) |
| 04:56.53 | bandrew | channan: Who do you recommend as a good bang for the buck? |
| 04:56.58 | krilloz | whats the smallest unit someone has got asterisk to run on, can someone point me in the right direction... |
| 04:57.08 | krilloz | like some kind of embedded platform perhaps |
| 04:58.40 | drsperm | What issues will I have if my voip phones are beind firewall "A" and * is behind firewall "B" |
| 04:59.07 | drsperm | I know that 5060/udp needs to be open to the server.... |
| 04:59.11 | drsperm | but any other issues? |
| 04:59.45 | shepherd | sip needs a proxy |
| 04:59.55 | alexns | i used xlite at one office with nat & firewall to connect to my * server @ home |
| 05:00.10 | drsperm | sip for the client side? |
| 05:00.13 | alexns | it seems to work well |
| 05:00.23 | [hC] | Do many of you guys use mysql for RES data? |
| 05:00.36 | [hC] | Or.. Anyone at all? :) |
| 05:00.36 | alexns | but i don't leave it on that often, on the client side there may be an issue with registration |
| 05:00.40 | drsperm | alexns: isn't xlite a softphone... |
| 05:00.53 | alexns | what is it ? |
| 05:00.57 | puppet | give me something to code :/ |
| 05:01.04 | puppet | bash.org thing was useless ;p |
| 05:01.06 | alexns | yes |
| 05:01.13 | alexns | sorry read wrong |
| 05:01.20 | drsperm | so a sip proxy... |
| 05:01.21 | alexns | i have also used polycom |
| 05:01.24 | alexns | 500 |
| 05:01.28 | drsperm | that is what I have. |
| 05:01.42 | drsperm | so I need a local sip proxy server.... |
| 05:01.52 | drsperm | local being near the client side. |
| 05:02.09 | channan | bandrew-Sorry I didn't follow the thread from the begining. what is your requirements? |
| 05:02.13 | alexns | i pointed mine directly to asterisk box |
| 05:02.17 | shepherd | yeah.. or a sip friendly nat :) |
| 05:02.24 | shepherd | i think they are selling appliances now |
| 05:02.34 | drsperm | hmm...I run linux firewalls... |
| 05:02.40 | drsperm | using iptables... |
| 05:02.42 | alexns | iptables |
| 05:02.45 | alexns | same as me |
| 05:02.55 | alexns | i was just forwarding ports to asterisk machine |
| 05:03.03 | drsperm | 5060 right? |
| 05:03.15 | alexns | my config isn't the best |
| 05:03.25 | alexns | 1 sec |
| 05:03.28 | alexns | let me check ports |
| 05:03.46 | shepherd | http://www.digium.com/handbook-draft.pdf |
| 05:03.47 | shepherd | sdfasdf |
| 05:03.53 | shepherd | 5060 and 5061 OPEN |
| 05:04.12 | alexns | yes 5060 |
| 05:04.17 | *** join/#asterisk dontmsgme (~none@207.215.252.80) |
| 05:04.35 | alexns | also 10000-20000 rtp |
| 05:04.39 | alexns | big hole |
| 05:04.42 | drsperm | k...so I might be able to get by without a sip proxy right... |
| 05:04.44 | alexns | not very optimal |
| 05:04.45 | alexns | yes |
| 05:04.45 | dontmsgme | I've got a Windows machine on wireless/all ports forwarded, I'm trying to ssh into my linux/asterisk box to start Asterisk but it won't connect any ideas anyone? |
| 05:04.56 | drsperm | wow...10k ports... |
| 05:04.57 | drsperm | nice. |
| 05:04.59 | alexns | hehe |
| 05:05.02 | shepherd | http://www.voip-info.org/wiki-NAT+and+VOIP |
| 05:05.14 | [hC] | nobody does res data in mysql? |
| 05:05.20 | alexns | i guess i could cut that range & specify in asterisk config |
| 05:05.36 | channan | dontmsgme-what's the error? what linix flavor? |
| 05:05.44 | SexyKen | what is a modulus? |
| 05:05.44 | cero64 | dontm: if you telnet to port 22 doesthe connection open at all? |
| 05:05.46 | SexyKen | what is a modulus_? |
| 05:05.52 | *** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net) |
| 05:05.53 | cf_man | puppet what do u mean? |
| 05:06.00 | SexyKen | what is modulus_? |
| 05:06.07 | SexyKen | jbot, what is a modulus_? |
| 05:06.09 | jbot | I think you lost me on that one, SexyKen |
| 05:06.15 | odie_flocon | allo all. |
| 05:06.27 | SexyKen | jbot, what is modulus_? |
| 05:06.28 | jbot | modulus_ is your god sexyken, or a rapist |
| 05:06.42 | dontmsgme | Cero how can I test that |
| 05:06.46 | dontmsgme | Im using putty |
| 05:07.05 | channan | putty should work fine |
| 05:07.09 | cero64 | yeah |
| 05:07.12 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 05:07.16 | cero64 | possible ssh version problem |
| 05:07.34 | *** join/#asterisk NoCAT (NoCAT@c-24-9-32-2.client.comcast.net) |
| 05:07.37 | dontmsgme | Everything is forwarded but how do I disable NAT |
| 05:07.39 | puppet | cf_man: what manager? :) im looking on a manager now to, sitting with flasp operator now, but its kinda messy |
| 05:07.41 | elric | If my SIP phone has Auto Answer function how can I implement Paging and Intercom announcement functionalities? |
| 05:07.48 | cero64 | you disable nat in your wireless router |
| 05:07.51 | alexns | what about firewall preventing ssh on linux machine |
| 05:07.55 | NoCAT | how many t1s are there in a t3? |
| 05:07.57 | cero64 | but you probably don't want to do that |
| 05:08.22 | alexns | 29 |
| 05:08.25 | cero64 | NoCat: essentially 3 |
| 05:08.26 | cero64 | 30 |
| 05:08.29 | elric | I want to make one way announcements. |
| 05:08.39 | dontmsgme | Alexns, I disable firewall on Linksys's settings |
| 05:08.56 | alexns | what about iptables config on linux machine |
| 05:09.03 | NoCAT | 30 t1s? |
| 05:09.15 | cero64 | it's not like they bond 30 T1s, but it's the same bandwidth |
| 05:09.27 | cero64 | a T3 (ds3) is 45mbps, a T1 is 1.5 |
| 05:09.29 | NoCAT | what about voice lines? |
| 05:09.34 | shepherd | cero: i've seen it |
| 05:09.36 | shepherd | hehe |
| 05:09.42 | NoCAT | is there a pri t3? |
| 05:09.47 | cero64 | well, they _could_ but generally not |
| 05:09.48 | cero64 | heh |
| 05:09.51 | shepherd | sometimes it's cheaper to get a t3 broken up into t1s |
| 05:09.56 | dontmsgme | What shoudl I look for on the ipconfig tables? |
| 05:09.58 | cero64 | you can get channelized ds3s i do believe |
| 05:10.07 | drsperm | cero64: yes. |
| 05:10.16 | drsperm | Mine will be installed in 2 weeks. |
| 05:10.18 | cero64 | 24 channels per t1, so 720 per ds3 |
| 05:10.25 | alexns | make sure port 22 is not blocked |
| 05:10.31 | cero64 | if it's a pri though they use one channel for signaling iirc |
| 05:10.45 | alexns | iptables -A INPUT -p tcp -i eth0 --dport 22 -j ACCEPT |
| 05:10.47 | cero64 | if pri, the 24 channels are only 56k, i t hink |
| 05:10.52 | cero64 | er, if not pri |
| 05:11.04 | cero64 | i always get it confused |
| 05:11.08 | NoCAT | so 3x24? |
| 05:11.12 | NoCAT | or 30x24? |
| 05:11.13 | cero64 | 30x24 |
| 05:11.14 | cero64 | yes |
| 05:11.16 | cero64 | 720 |
| 05:11.16 | NoCAT | YEAH? |
| 05:11.20 | NoCAT | !! |
| 05:12.30 | *** join/#asterisk JerJer[mobile] (~jj@65.173.197.109) |
| 05:12.41 | cero64 | NoCAT: yeah, that's a lot of voice channels :) |
| 05:13.28 | drsperm | Looks like if your firewall will support NAT Transversal, and your phone can support knowledge of the public ip address...it should work fine... |
| 05:13.33 | drsperm | ...at least with 1 phone. |
| 05:13.49 | cf_man | anybody have any thoughts on the asterisk / CD (asterisk@home) |
| 05:13.50 | tuxinator_linux | Good Evening guys |
| 05:13.58 | dontmsgme | How do you disable NAT |
| 05:13.59 | tuxinator_linux | cf_man: I hear it is good |
| 05:13.59 | cero64 | sup tux |
| 05:14.03 | dontmsgme | You dont use 192.168.1.1? |
| 05:14.07 | alexns | nat on what |
| 05:14.10 | tuxinator_linux | Packing for VON and Meet * |
| 05:14.26 | drsperm | NAT-T on what ever... |
| 05:14.34 | drsperm | not everything supports natt |
| 05:14.38 | cf_man | tux: sup? |
| 05:14.38 | drsperm | sorry..NAT-T |
| 05:14.51 | bandrew | hey do most of these VOIP phones like the soundpoint 600 have intercoms? |
| 05:15.51 | cero64 | intercom as in like speakerphone? |
| 05:16.28 | drsperm | Am I correct that the newer polycom's support STUN ? |
| 05:16.35 | bandrew | yeah, and intercom to page other people in the office |
| 05:18.37 | alexns | yes my 500 does |
| 05:18.54 | drsperm | yeah...that is what I got...I remember seeing it in the config. |
| 05:19.00 | *** join/#asterisk jpayne (~jpayne@baconhouse.sackheads.org) |
| 05:19.16 | alexns | there is an article on asterisk-wiki that talks about intercom with polycom 500 & 600 |
| 05:19.19 | drsperm | so with that we just define the outside ip address, open 5060/udp on the server side...and it is ready to go... |
| 05:19.57 | shepherd | drsper: did you read that article? |
| 05:20.09 | shepherd | http://www.voip-info.org/wiki-NAT+and+VOIP |
| 05:20.21 | drsperm | yep...and a few others. |
| 05:20.49 | drsperm | STUN and NATT is the answer...other than a vpn |
| 05:21.02 | alexns | anybody wit h323 experiance ?? |
| 05:21.08 | jpayne | anyone seen bkw recently? |
| 05:22.46 | shepherd | he's at von |
| 05:23.33 | shepherd | it works |
| 05:23.34 | shepherd | i've seen it |
| 05:23.56 | NoCAT | how do i order a t3 for voice what do i ask for? |
| 05:24.06 | Qwell | a t3 for voice |
| 05:24.32 | NoCAT | yeah is there a such thing as a pri for voice? |
| 05:24.33 | techie | haha. |
| 05:24.43 | NoCAT | t3 |
| 05:24.46 | Qwell | This is where research might come in handy. |
| 05:24.57 | NoCAT | yeah, i'm trying |
| 05:25.12 | Mavvie | that reminds me that I still have a spare E3 module. |
| 05:25.30 | shepherd | nocat: first you make a call |
| 05:25.36 | NoCAT | could you use a t3 like a t1 pri? |
| 05:25.36 | shepherd | to bell |
| 05:25.42 | shepherd | and they help you out with the rest, hehe |
| 05:25.44 | techie | first you pick up the phone |
| 05:26.09 | ManxPower | Generally T-1s and T-3s, etc are considered groups of 64K channels |
| 05:26.25 | *** join/#asterisk vinmohnj (~vinmohfx@pcp0010311885pcs.avenel01.nj.comcast.net) |
| 05:26.27 | shepherd | nocat: somewhere you're going to have to split up that t3 into t1s |
| 05:26.36 | NoCAT | so a t3 is 720 64k channels? |
| 05:26.39 | ManxPower | NoCAT, PRI us just a signaling protocol. In theory you could use PRI over IPX/SPX. |
| 05:27.03 | ManxPower | NoCAT, Why would it not be? |
| 05:27.13 | ManxPower | Of course the physical INTERFACE is different. |
| 05:27.38 | vinmohnj | hi all |
| 05:28.15 | vinmohnj | I'm new to Asterisk and I need help to setup sip connection.Could any one help me in this ? |
| 05:28.16 | NoCAT | manxpower how much on average does the physical interface to a pc cost for a t3? |
| 05:28.29 | cero64 | NoCat: you might want to ask for a channelized ds3/t3 |
| 05:28.37 | cero64 | and it's going to be quite expensive, probably |
| 05:28.55 | NoCAT | cero64, how expensive? |
| 05:28.58 | cero64 | NoCat: i just saw a digium t1/e1 card for like $589 |
| 05:29.10 | NoCAT | cero64 can i use that for a t3? |
| 05:29.14 | cero64 | NoCat: really depends on telco, but hundreds if not thousands per month |
| 05:29.18 | cero64 | NoCat: probably not |
| 05:29.27 | NoCAT | more then 5 thousand a month? |
| 05:29.32 | cero64 | ask your telco |
| 05:29.49 | shepherd | you can probably get a t3 cheaper from sprint |
| 05:29.53 | shepherd | than bell |
| 05:29.58 | cero64 | yeah, most likely |
| 05:30.05 | NoCAT | qwest is currently my provider |
| 05:30.09 | ManxPower | The last DS3/T-3 I ordered came in on TWO coax cables and cost US$30,000/month |
| 05:30.11 | shepherd | MORE THAN LIKELY |
| 05:30.17 | shepherd | location is the biggest issue |
| 05:30.21 | t3t | NoCAT: Your loop will probably be > $5k/mo |
| 05:30.26 | vinmohnj | Any one in here to help me setting up sip connection ? I have to set up this week or else My job will be in trouble |
| 05:30.33 | shepherd | if you are sitting next door to sprint |
| 05:30.37 | ManxPower | But that was the year that the Olympics were in Atlanta, so it's been a while. |
| 05:30.39 | *** join/#asterisk p0lar (~p0lar@dhcp64-134-126-92.sjca.sjc.wayport.net) |
| 05:30.40 | shepherd | your t3 will be damn cheap |
| 05:30.49 | cero64 | Manx: 96 :) |
| 05:30.49 | NoCAT | manx $30,000/month... |
| 05:31.16 | ManxPower | NoCAT, It was a connection to the internet for the Siggraph show that year in New Orleans. |
| 05:31.29 | NoCAT | t3t why do you think my loop would be 5k a month? |
| 05:31.53 | ManxPower | NoCAT, Anyway, there are no T-3/DS3 interfaces for PCs that I know of that are supported by Asterisk. |
| 05:32.01 | *** join/#asterisk Sedorox (~Sed@pcp01339110pcs.wilog101.pa.comcast.net) |
| 05:32.09 | shepherd | manx: he would have to mux t1s |
| 05:32.10 | NoCAT | no? |
| 05:32.13 | cero64 | you would probably need a cisco router to handle that |
| 05:32.16 | ManxPower | NoCAT, It was a VERY good deal for the time. |
| 05:32.18 | p0lar | I'd *love* to see a T3 interface for *..haha |
| 05:32.23 | NoCAT | can you get 30 t1s? |
| 05:32.34 | cero64 | sure |
| 05:32.39 | shepherd | yeah |
| 05:32.41 | techie | sure. |
| 05:32.42 | cero64 | you could ima them |
| 05:32.44 | vinmohnj | potter: r you trying to set up a VOIP through Asterisk ? |
| 05:32.46 | techie | i have 24 |
| 05:32.54 | potter | vinmohnj: yes |
| 05:32.55 | shepherd | so.. you might as well buy the t1s seperately |
| 05:32.57 | vinmohnj | I mean a remote one ? |
| 05:32.57 | p0lar | What codec? |
| 05:33.03 | NoCAT | yeah, |
| 05:33.07 | shepherd | and you can get t1s from different locations too :) |
| 05:33.14 | potter | vinmohnj: am trying to terminate h323 calls to SIP |
| 05:33.20 | potter | vinmohnj: via asterisk |
| 05:33.33 | vinmohnj | potter:Me too ..I'm new to this |
| 05:33.46 | t3t | NoCAT: because that's what the telcos charge for most anything beyond a few hundred feed of a well-connected CO |
| 05:33.49 | potter | vinmohnj: i got it setup ... it connects, rings, answered ... but i get dead air |
| 05:34.06 | vinmohnj | potter: oh great |
| 05:34.25 | potter | vinmohnj: dead air boths sides |
| 05:34.36 | potter | vinmohnj: dunno why for now |
| 05:34.38 | vinmohnj | :potter how did you set up ? I couldint get any rings or dial ton |
| 05:34.39 | NoCAT | how much does something cost which splits everything from a t3 to t1s? what is something like that called? |
| 05:34.52 | t3t | channel bank |
| 05:34.57 | vinmohnj | potter: dead air ? |
| 05:35.01 | shepherd | haha |
| 05:35.02 | potter | vinmohnj: i got this setup ... AS5300 --------> asterisk ---------> audiocodes |
| 05:35.03 | shepherd | channel bank |
| 05:35.05 | shepherd | hasdhfasdf |
| 05:35.06 | shepherd | MUX |
| 05:35.15 | NoCAT | what kind of channel bank? |
| 05:35.17 | NoCAT | mux? |
| 05:35.23 | NoCAT | mux channel bank.. ok thanks |
| 05:35.26 | shepherd | not a channel bank |
| 05:35.29 | potter | vinmohnj: dead air .... no voice ... blank |
| 05:35.31 | cero64 | mux is short for multiplexer |
| 05:35.45 | *** join/#asterisk krilloz (majestic@220-253-7-238.VIC.netspace.net.au) |
| 05:36.01 | vinmohnj | Potter: oops I'm new to this too |
| 05:36.13 | potter | vinmohnj: thats why am trying my luck here |
| 05:36.15 | *** join/#asterisk lattice (~lattice@S010600045ad57bb6.vc.shawcable.net) |
| 05:36.19 | potter | vinmohnj: still trying though |
| 05:36.28 | vinmohnj | potter: could you help me in configuring the sip thing ? |
| 05:36.47 | vinmohnj | :potter I'm in big trouble cause I have to set up this wek itself ..uhh |
| 05:36.53 | NoCAT | you said you had 24 t1s? how much are you paying per t1? |
| 05:36.54 | potter | vinmohnj: whats your setup |
| 05:36.55 | t3t | shepherd: you could use a channel bank for an OC-12 |
| 05:36.59 | potter | vinmohnj: whats the call flow |
| 05:37.05 | potter | vinmohnj: whats machines involved |
| 05:37.27 | shepherd | we want to keep it digital, i'm sure :) |
| 05:37.59 | NoCAT | i only want voip devices wifi |
| 05:38.51 | vinmohnj | Potter: I have a small test bed with 4 regular PSTN which connects to FXO @ channel bank and 4 extentions terminating at FXZS channel bank ..it works fine but I dont know how to set up a VOIP externally by which we can give an extention to our guest house through Internet |
| 05:39.34 | vinmohnj | potter:Channel bank connects to asterisk server throught T1 PCI |
| 05:40.22 | vinmohnj | Any one in here who could helm Petter + vinmohnj to set up+ troubleshoot SIP / Asterisk > |
| 05:40.26 | alexns | h323 knowledge ?? |
| 05:40.41 | alexns | developers interested in project ? |
| 05:40.45 | t3t | I know h323 sucks.. si that enough? |
| 05:40.51 | alexns | yes |
| 05:40.54 | alexns | i hate it too |
| 05:41.19 | NoCAT | vinmohnj your running * at home.. your a stunner |
| 05:41.32 | *** join/#asterisk Tarox (someone@pD9E7ABEA.dip.t-dialin.net) |
| 05:41.46 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
| 05:42.10 | NoCAT | brb |
| 05:42.25 | vinmohnj | : noCAT I'm running Asterisk at one of start up company |
| 05:42.34 | alexns | i have a distributor of a product that happens to use h323 who wants it to work with asterisk.. i guess whoever gets it working could get a licence fee for each installation |
| 05:42.34 | techie | get me a DS3 |
| 05:42.39 | jakepdev | i just installed *@home - can dial to 1234 (test app), looks good, but can't hear anything |
| 05:42.50 | alexns | the system it integrates with is pretty serious |
| 05:43.44 | shepherd | jake: codec problem maybe? |
| 05:44.05 | jakepdev | would it say something in console? |
| 05:44.18 | shepherd | sometimes it does |
| 05:44.18 | shepherd | heh |
| 05:44.25 | jakepdev | lol |
| 05:44.27 | zapa | thanks for all silla |
| 05:44.31 | shepherd | i had a problem with ulaw and gsm today |
| 05:44.39 | *** part/#asterisk zapa (zapa@200.77.116.158) |
| 05:44.56 | jakepdev | using dta310 - says I can use g711 on that |
| 05:45.16 | jakepdev | should i put gsm in conf file? |
| 05:46.12 | jakepdev | no luck with that |
| 05:46.17 | shepherd | try ulaw |
| 05:46.21 | jakepdev | ok |
| 05:46.22 | shepherd | or alaw |
| 05:46.31 | jakepdev | i'll add both |
| 05:47.22 | *** part/#asterisk alexns (~alex@acs-24-154-114-15.zoominternet.net) |
| 05:47.33 | jakepdev | nope - still no audio |
| 05:47.50 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:48.04 | jakepdev | if it picks up the DTMF, does that mean the code is ok? |
| 05:48.21 | jakepdev | codec |
| 05:48.21 | Eight | er, silly question... where's the authoritative Asterisk changelog? |
| 05:48.43 | *** join/#asterisk Bruns (bruns@pool-141-153-151-58.nwrk.east.verizon.net) |
| 05:48.53 | Eight | there doens't seem to be one in the tar, or on the asterisk.org site... or voip-info wiki... |
| 05:48.53 | t3t | Eight: the source |
| 05:49.06 | Eight | t3t: You mean read the patches? =/ |
| 05:49.22 | t3t | Eight: seriously, you can find changes in a module by browsing cvs |
| 05:49.47 | t3t | As far as the whole app is concerned, I don't know if there is a master change log that contains everything |
| 05:50.00 | Eight | well, that's disappointing =/ |
| 05:50.11 | Silik0n | when doing a gotoif is there a good way to say do something like |
| 05:50.13 | t3t | What are you looking for specifically? |
| 05:50.33 | Eight | Do you know if the 'broadvoice patch' made it into 0.6? |
| 05:51.07 | t3t | For SIP. |
| 05:51.07 | Silik0n | exten => _X.,n,GotoIf($[$blah : 1|2|3|*]?something:somethingelse) |
| 05:51.08 | Silik0n | ? |
| 05:51.31 | Silik0n | the * causes it to fail every time... any hints? |
| 05:51.32 | JerJer[mobile] | that is just nasty |
| 05:51.35 | Bruns | hey, anyone have a sec to help me figure out something? I've got two separate nufone accounts that I need to terminate on the same asterisk box. However, its rejecting the second incoming 866 number as failed to authenticate |
| 05:51.36 | Eight | t3t: That's a yes? |
| 05:51.53 | *** join/#asterisk alexns (~alex@acs-24-154-114-15.zoominternet.net) |
| 05:51.59 | t3t | It was supposed to be a question, but my fingers got the better of me |
| 05:52.03 | Eight | t3t: ah. |
| 05:52.10 | Eight | t3t: ya, the chan_sip.c patch. |
| 05:52.12 | JerJer[mobile] | Silik0n: why not write a C API based app to do that logic for you? |
| 05:52.15 | t3t | was that a sip patch? |
| 05:52.29 | *** part/#asterisk alexns (~alex@acs-24-154-114-15.zoominternet.net) |
| 05:52.41 | Silik0n | JerJer[mobile]: that would benice, but theres a monster AGI driving everything |
| 05:52.46 | NoCAT | check this out says its a t3 pci card for linux http://www.linuxdevices.com/news/NS8878561328.html |
| 05:53.02 | Silik0n | a agi that someone else wrote |
| 05:53.25 | JerJer[mobile] | Bruns: first off this is not a nufone support channel. second, you either have to use the same secret on both accounts or setup an RSA key |
| 05:53.33 | Eight | t3t: I grepped a few + lines from http://edvina.net/broadvoice/broadvoicesip.txt in my downloaded sources. It looked good but I don't know if they're the relevant ones. |
| 05:53.33 | JerJer[mobile] | Silik0n: that's even worse |
| 05:53.36 | Silik0n | I need to see if the called presses * on the prompt to back up a menu |
| 05:53.41 | *** join/#asterisk djin (~djin@213.84.95.241) |
| 05:53.44 | JerJer[mobile] | oh god |
| 05:53.48 | JerJer[mobile] | use the new ivr logic |
| 05:53.51 | JerJer[mobile] | screw agi |
| 05:53.53 | Silik0n | hah |
| 05:54.03 | jakepdev | jejer - new ivr logic? |
| 05:54.05 | Eight | t3t: I'm running 1.0.6 (I don't like the idea of running stuff from CVS =) |
| 05:54.11 | Bruns | jerjer: I know, a friend suggested I ask here. You actually answered my question perfectly :) |
| 05:54.16 | *** join/#asterisk dan2 (~beta3@dan2.active.supporter.pdpc) |
| 05:54.41 | t3t | Eight: It looks like the 'patch' was introduced on 12/30/04 into the 1.0 branch... so I would guess yes. |
| 05:54.42 | NoCAT | SBE's wanPMC-C1T3 board |
| 05:54.54 | Silik0n | ok like I have time on this project to completely reimplement thisentire thing |
| 05:55.04 | jakepdev | what is this new IVR logic? |
| 05:55.10 | JerJer[mobile] | see app_ivrdemo.c |
| 05:55.11 | Eight | t3t: thanks man. I was thinking it was there but I wanted someone else to confirm. |
| 05:55.20 | jakepdev | tnx jerjer |
| 05:55.33 | t3t | np, Eight |
| 05:55.56 | JerJer[mobile] | Bruns: I own nufone :) |
| 05:56.01 | Bruns | oh |
| 05:56.02 | Bruns | jeremy |
| 05:56.18 | t3t | JerJer[mobile]: How do dem nufones work again? |
| 05:56.26 | CpuID | haha |
| 05:56.31 | jakepdev | jus like de old ones |
| 05:56.45 | JerJer[mobile] | powered by Asterisk |
| 05:56.54 | t3t | JerJer[mobile]: I wish that I could have stayed for your talk at WISPNOG |
| 05:57.02 | JerJer[mobile] | wasn't there |
| 05:57.06 | Bruns | I didn't recognize you. You were at LWE 2004 right? |
| 05:57.22 | t3t | I thought you were going? |
| 05:57.47 | JerJer[mobile] | t3t: we had 2 inches of ice on the ground |
| 05:57.57 | t3t | bummer |
| 05:58.10 | Qwell | JerJer[mobile]: Do your talk here. :p |
| 05:58.23 | JerJer[mobile] | no way in hell I was even leaving my house much less to commute to Chicago |
| 05:58.33 | jakepdev | jerjer - are you saying ivrdemo.c is more stable than agi? |
| 05:58.40 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 05:58.41 | JerJer[mobile] | how about more sane |
| 05:58.51 | jakepdev | ok - i'll take that |
| 05:59.01 | JerJer[mobile] | not sure about stability, yet |
| 05:59.12 | vinmohnj | Any one here who could help us on fixing Sip connection ? |
| 05:59.33 | JerJer[mobile] | the one app i hammered out so far that uses it seems to work quite nicely, so far |
| 05:59.50 | dan2 | Eight: whats the issue |
| 05:59.59 | jakepdev | is there any tested solution - pushed to the limits that you know about? |
| 06:00.08 | dan2 | Eight: I'm a software engineer for broadvoice, not that I can help you right now, I'm fucking daed tired |
| 06:00.13 | JerJer[mobile] | broadvoice doesn't seem to be very friendly twoards asterisk |
| 06:00.27 | t3t | dan2: so what was your impression? |
| 06:00.33 | Bruns | they aren't |
| 06:00.39 | Bruns | they made some sort of a change the other day |
| 06:00.41 | t3t | JerJer[mobile]: How far out of the city are you? |
| 06:00.46 | Bruns | which crippled our outgoing call abilities |
| 06:00.48 | JerJer[mobile] | lol - 5 hours |
| 06:01.00 | Bruns | we couldn't authenticate to their proxies |
| 06:01.03 | JerJer[mobile] | plus traffic |
| 06:01.07 | Eight | dan2: Ah, heh. I just signed up and got no confirmation e-mail. And my account doesn't seem to be active either. |
| 06:01.18 | dan2 | JerJer[mobile]: just not friendly to non sip compliant devices and applications |
| 06:01.21 | t3t | I thought you were in the city for some reason |
| 06:01.25 | dan2 | Eight: when did you sign up |
| 06:01.30 | Eight | dan2: couple hours ago. |
| 06:01.32 | [hC] | God damn this wiki page on voip-info makes no sense |
| 06:01.42 | dan2 | Eight: name? |
| 06:01.45 | CpuID | hey JerJer[mobile], you got plans to expand your servers outside north america in future? |
| 06:01.48 | [hC] | the asterisk realtime page has a paragraph about database peers and static peers, i dont get it. |
| 06:02.03 | Newbie___ | dan2: is it ture that broadvoice do not accept non USA credit card? |
| 06:02.23 | dan2 | Newbie___: no clue |
| 06:02.25 | Qwell | CpuID: What part of oz? |
| 06:02.30 | CpuID | gold coast here |
| 06:02.37 | CpuID | like i know i can call AU already |
| 06:02.43 | CpuID | but...got latency alright :) |
| 06:02.47 | CpuID | tried it as a test once |
| 06:02.55 | Qwell | yeah, latency back and forth over the ocean would suck |
| 06:03.00 | `Sauron | Hum |
| 06:03.00 | CpuID | lol exactly |
| 06:03.08 | CpuID | has to be expected going across the pacific twice :) |
| 06:03.11 | `Sauron | now we recompile *, and see if the BV patch made it in yet |
| 06:03.29 | `Sauron | They changed chan_sip.c, so chances are that they did |
| 06:03.37 | JerJer[mobile] | [hC] i don't think many get realtime at all |
| 06:04.12 | JerJer[mobile] | i cannot believe mark even implented something like that, to be honest |
| 06:04.19 | *** part/#asterisk JerJer[mobile] (~jj@65.173.197.109) |
| 06:04.33 | t3t | `Sauron: According to CVS it was added to the 1.0 tree in Dec. so it should be there |
| 06:04.38 | *** join/#asterisk JerJer[mobile] (~jj@65.173.197.109) |
| 06:04.52 | JerJer[mobile] | stupid x |
| 06:04.56 | `Sauron | t3t: I'm talking about the patch that surfaced yesterday |
| 06:05.00 | dan2 | JerJer[mobile]: mark didn't, oej and I did |
| 06:05.01 | `Sauron | it wasn't applied back in december |
| 06:05.18 | JerJer[mobile] | dan2: then i cannot believe he approved it |
| 06:05.20 | `Sauron | realtime is sort of mostly backwards |
| 06:05.40 | t3t | `Sauron: you talking about something other than the one 'fixing' 2917 |
| 06:05.48 | JerJer[mobile] | sort of ? |
| 06:06.10 | `Sauron | t3t: I'm talking about the one that 'fixes' the INVITE authentication changes that BV did in the last 2-3 days |
| 06:06.28 | t3t | Aren't they getting friendly |
| 06:06.35 | dan2 | JerJer[mobile]: its a step towards RFC compliance, its been in SIP since 1.0, it was a necessary feature |
| 06:06.43 | JerJer[mobile] | !? |
| 06:06.57 | Qwell | You two are talking about two different things I'd bet. |
| 06:06.58 | JerJer[mobile] | what does realtime have to do with sip ? |
| 06:07.03 | `Sauron | jerjer: are you talking about the invite auth, or realtime? |
| 06:07.06 | `Sauron | That's what I thought. |
| 06:07.13 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
| 06:07.14 | dan2 | JerJer[mobile]: I'm referring to the broadvoice patch |
| 06:07.26 | JerJer[mobile] | yeah ok - i could care less about sip |
| 06:07.30 | JerJer[mobile] | break it all you want |
| 06:07.32 | `Sauron | dan2: do you know if it got committed to HEAD today? |
| 06:07.40 | shepherd | broadvoice should just offer iax |
| 06:07.41 | dan2 | `Sauron: its in stable 1.0.4 |
| 06:07.44 | shepherd | it's not that hard |
| 06:07.45 | shepherd | :) |
| 06:07.48 | dan2 | shepherd: never |
| 06:08.08 | JerJer[mobile] | broadvoice is too legacy |
| 06:08.12 | *** join/#asterisk marshall (~test@S0106000f66563988.wp.shawcable.net) |
| 06:08.16 | shepherd | dan2: you will one day |
| 06:08.19 | `Sauron | dan2: Doubt it. CVS-HEAD didn't work as of last night. |
| 06:08.22 | Qwell | flamewar! :p |
| 06:08.28 | dan2 | `Sauron: did you follow the instructions? |
| 06:08.35 | Juggie | whats broken in cvs-head now? |
| 06:08.41 | `Sauron | add the authuser thing, etc etc? |
| 06:08.55 | dan2 | `Sauron: hold a moment |
| 06:09.01 | `Sauron | alright |
| 06:09.22 | JerJer[mobile] | Juggie: i had an odd problem early this morning, but it seemed to go away on its own - reallly strange |
| 06:09.38 | dan2 | `Sauron: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup |
| 06:09.41 | tuxinator_linux | Any of you at VON? |
| 06:09.59 | Juggie | JerJer[mobile], the iax problem we talked about in #*-dev? |
| 06:10.09 | p0lar | I was .. leaving as soon as my plane gets here though |
| 06:10.11 | p0lar | <forever> |
| 06:10.19 | tuxinator_linux | forever? |
| 06:10.21 | `Sauron | Sigh. |
| 06:10.27 | p0lar | || 23h45.. :( |
| 06:10.29 | JerJer[mobile] | Juggie: yeah |
| 06:10.34 | `Sauron | Did y'all ever decide, is it account ID or phone number, or either? |
| 06:10.37 | Juggie | that just disapeared? thats odd |
| 06:10.46 | p0lar | BTW, I approached the SysMaster guys again today |
| 06:10.47 | tuxinator_linux | p0lar: whre are you going back to |
| 06:10.50 | p0lar | MTL |
| 06:10.56 | p0lar | sorry, Montreal, QC |
| 06:11.02 | dan2 | `Sauron: accountid = phone number |
| 06:11.03 | JerJer[mobile] | p0lar: hell yeah - dish :) |
| 06:11.10 | dan2 | `Sauron: its how we index our systems |
| 06:11.12 | tuxinator_linux | I am going over to San Jose tonight |
| 06:11.13 | p0lar | Those guys were full of it |
| 06:11.21 | Juggie | p0lar, who are the SysMaster guys |
| 06:11.26 | `Sauron | dan2: accountid on the web thing is some large number != phone number |
| 06:11.28 | p0lar | they were ducking and dodging.. claimed they used digium equipment and their drivers, etc... but not the software |
| 06:11.32 | p0lar | I laid it on pretty thick |
| 06:11.41 | *** join/#asterisk cero64- (ruiner@fantab.ulo.us) |
| 06:11.43 | *** join/#asterisk dontmsgme (~none@69-175-234-120.vnnyca.adelphia.net) |
| 06:11.43 | p0lar | they said, "if you can show us one asterisk developer who says that, we'll concede" |
| 06:11.47 | p0lar | I figure they ahve a binding agreement |
| 06:11.50 | p0lar | that prevents such |
| 06:11.50 | dontmsgme | Why can I ping my asterisk box from windows but I cant telnet in? |
| 06:11.52 | dan2 | `Sauron: what you see there is really group id != account id |
| 06:11.53 | dontmsgme | Or Ssh |
| 06:12.03 | `Sauron | Heh. Alright. |
| 06:12.29 | dan2 | `Sauron: use phone number |
| 06:12.29 | tuxinator_linux | dontmsgme: try ssh |
| 06:12.29 | `Sauron | okay |
| 06:12.29 | JerJer[mobile] | p0lar: tell them Brian West and Jeremy McNamara have both independantly verified Asterisk is used |
| 06:12.29 | p0lar | Juggie: sysmaster = Asterisk on industrial-looking garbage-hardware |
| 06:12.37 | `Sauron | are y'all going to fix the /etc/hosts hack at some point? It makes me feel dirty. |
| 06:12.40 | p0lar | Yeah, I sent a guy named Krish links |
| 06:12.58 | p0lar | he sent back the link where Mark didn't deny it, but didn't claim it either. |
| 06:13.06 | p0lar | I haven't responded yet, but I can bcc you on it if you're curious |
| 06:13.09 | JerJer[mobile] | sure |
| 06:13.12 | p0lar | k |
| 06:13.17 | dan2 | `Sauron: I just update and replaced with phonenumber |
| 06:13.26 | p0lar | Those guys piss me off.. I'm the one holding the money, damn it. |
| 06:13.35 | Juggie | p0lar, what are you trying to prove? |
| 06:13.41 | JerJer[mobile] | it was on slashdot! :P |
| 06:13.45 | marshall | Hi everyone, I'm working with a Polycom 300 which can receive calls perfectly, and call other phones but with no outbound audio - I'm racking my brains |
| 06:14.07 | dontmsgme | Tuxinator It wont ssh in via putty for example |
| 06:14.10 | JerJer[mobile] | Juggie: non-compliance |
| 06:14.15 | Qwell | dontmsgme: is ssh running? |
| 06:14.16 | JerJer[mobile] | gpl |
| 06:14.18 | p0lar | My favourite line of the eMail: "We are here to start a new business relationship and I would like it to be mutually trustworthy rather than being suspicious." |
| 06:14.27 | p0lar | mutually trustworth = redefined to fit their needs I guess |
| 06:14.30 | dontmsgme | Qwell how do I do that |
| 06:14.36 | p0lar | Juggie: I have nothing to prove, THEY DO. |
| 06:14.40 | dontmsgme | Does ssh need to be running to simply connect from windows with Xlite to the linux box? |
| 06:15.02 | dontmsgme | Forgetting about putty |
| 06:15.02 | p0lar | I'm the customer. I hold the money, they hold the product -- or do they? |
| 06:15.02 | t3t | marshall: is there a NAT between the * and the polycom? |
| 06:15.02 | Juggie | p0lar, i'm just asking |
| 06:15.05 | marshall | yes |
| 06:15.13 | shepherd | if they ever modify the code, they can't call it asterisk |
| 06:15.15 | Qwell | dontmsgme: no |
| 06:15.16 | t3t | where is ti? |
| 06:15.26 | p0lar | google://sysmaster asterisk gpl |
| 06:15.37 | marshall | Nat is on the phone end, * is straight into internet |
| 06:15.41 | JerJer[mobile] | you mean they've taken linux, busybox, gnugk, asterisk, and some php to create a multi-thousand dollar product |
| 06:15.54 | t3t | marshall: do you have nat=yes in sip.conf? |
| 06:15.55 | JerJer[mobile] | and claim it is completly their own work |
| 06:15.58 | marshall | yes |
| 06:16.17 | `Sauron | dan2: I followed the instructions, and I still get the failed to authenticate on INVITE ... error |
| 06:16.21 | `Sauron | Grumble. |
| 06:16.26 | t3t | marshall: and on the phone config (sip config I think)? |
| 06:16.29 | JerJer[mobile] | shepherd: they still have to disclose |
| 06:16.35 | p0lar | Agreed. |
| 06:16.37 | JerJer[mobile] | and provde code |
| 06:16.43 | p0lar | Why is it they won't confess? |
| 06:16.49 | p0lar | What's wrong with using *? |
| 06:17.04 | marshall | yes, however there is a place on the phone to specify a NAT device - can't find any documentation on the polycom web interface |
| 06:17.14 | JerJer[mobile] | even if it is as smple as: cvs -d:pserver:anon@cvs.digium.com co asterisk :) |
| 06:17.22 | JerJer[mobile] | -D <some date> |
| 06:17.28 | t3t | marshall: just a sec |
| 06:17.44 | Juggie | is distributing pre build machines with pre built binaries, considered distributing a modified binary? i would assume so eh. |
| 06:17.46 | Eight | `Sauron: Is the accountID 10 digit or 11 digit? |
| 06:17.59 | JerJer[mobile] | they want to give people that warm fuzzy feeling by claiming it is all custom development |
| 06:18.05 | JerJer[mobile] | and absolutely not open-source |
| 06:18.11 | p0lar | Anyway, I'll shoot an eMail tomorrow to respond to their responses.. amusing at best, what it is. |
| 06:18.19 | `Sauron | Eight: It's your phone number |
| 06:18.27 | `Sauron | 10 digits |
| 06:18.28 | p0lar | Anyway, as an open-source advocate and supporter -- I want to make sure my $$ is well-placed. |
| 06:18.30 | Eight | `Sauron: Thanks. |
| 06:18.31 | JerJer[mobile] | smells likes its time for another round of noce |
| 06:18.36 | JerJer[mobile] | noise |
| 06:18.48 | Eight | `Sauron: I'm not quite that far along right now, but that's something I don't want to mess with =p |
| 06:18.49 | JerJer[mobile] | this time something less geeky than slashdot |
| 06:19.06 | Juggie | cnet news? |
| 06:19.07 | p0lar | The GPL needs a public defender. :D |
| 06:19.32 | p0lar | layeth the smacketh downeth on those who violateth.. |
| 06:19.32 | dan2 | p0lar: its called the FSF |
| 06:19.34 | JerJer[mobile] | i'm thinkin Wall Street Journal |
| 06:19.44 | Eight | p0lar: ya, what dan2 said =) |
| 06:19.51 | JerJer[mobile] | that would smack em up side the head once |
| 06:20.00 | Juggie | submit a story to cnet, i used to have a contact there, but i dont think he works there anylonger |
| 06:20.06 | p0lar | dan2,Eight: heh.. yeah.. that. :D |
| 06:20.39 | Eight | This is what I'd get if my account doesn't exist, right? "Registration for '10DIGITS@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again" |
| 06:20.48 | Eight | I mean, so long as we're talking about BV =) |
| 06:20.55 | `Sauron | Yes. |
| 06:21.03 | dan2 | Eight: you received your password in the email? |
| 06:21.19 | Eight | dan2: nope. Remember, I haven't gotten *any* e-mails from BV. |
| 06:21.23 | p0lar | ok.. I'm out -- time to pass out on an airplane for 5 hours. :( |
| 06:21.26 | dan2 | Eight: then it won't work |
| 06:21.41 | Eight | dan2: It doesn't just use the account password I entered? |
| 06:21.45 | dan2 | Eight: nope |
| 06:21.50 | Eight | ah. |
| 06:21.53 | dontmsgme | Is xlite made for FC2? |
| 06:22.45 | t3t | marshall: have you set "<outboundProxy" values in sip.conf for the phone? |
| 06:22.56 | marshall | negative |
| 06:23.08 | t3t | Give that a shot.. here's the syntax... |
| 06:23.37 | t3t | <outboundProxy voIpProt.SIP.outboundProxy.address="your.ip.address" voIpProt.SIP.outboundProxy.port="5060"/> |
| 06:23.54 | marshall | I'll try it, one minute |
| 06:24.16 | Eight | well, so much for 24/7 phone support =/ |
| 06:24.20 | Essobi | MAhaha |
| 06:24.28 | Juggie | p0lar, http://news.com.com/2040-1096_3-0.html?tag=ne.ft.si.con |
| 06:24.34 | Essobi | that's 24 hours a week, 7 months a year. |
| 06:24.39 | t3t | Eight: you got the auto attendant, right? |
| 06:24.47 | t3t | Doesn't that count? |
| 06:24.50 | Eight | t3t: heh. |
| 06:25.05 | Eight | I think the idea is that you can stay on hold until 9AM when people show up to work =) |
| 06:25.13 | Eight | Or poke random employees on IRC, of course =p |
| 06:25.21 | dan2 | Eight: nah, the head of support staff is still at von |
| 06:25.37 | Essobi | I'm mad. |
| 06:25.47 | Essobi | My boss went to the conference. :| |
| 06:25.54 | dan2 | I went to the conference |
| 06:25.56 | dan2 | it wasn't bad |
| 06:25.57 | Essobi | :P |
| 06:26.03 | Essobi | Next year he better take me. |
| 06:26.17 | dan2 | Essobi: I have an exhibitor pass |
| 06:26.20 | *** join/#asterisk krilloz (majestic@220-253-7-238.VIC.netspace.net.au) |
| 06:27.27 | Essobi | Bah. |
| 06:27.50 | dan2 | we have a big ass booth at the convention center |
| 06:27.55 | dan2 | Essobi: someone drove in a broadvoice van |
| 06:28.04 | dan2 | 6 days all the way from boston massachusetts |
| 06:28.15 | `Sauron | dan: do you know how far behind y'all are on port requests? :) |
| 06:28.26 | Eight | it took them 6 days to cross the country? pffft. |
| 06:28.29 | dan2 | no idea |
| 06:28.42 | Essobi | Y'know.. |
| 06:28.56 | dan2 | `Sauron: it takes roughly a month so I hear |
| 06:28.57 | Essobi | I've had one way voice problems before.. |
| 06:29.02 | Essobi | I mean.. who doesn't.. |
| 06:29.06 | `Sauron | dan2: yeah, I keep hearing that. |
| 06:29.09 | Essobi | but one way DTMF? Pssh.. |
| 06:29.13 | JerJer[mobile] | wholly crazy road trip batman |
| 06:29.27 | `Sauron | the FCC was shooting for 4 days, and most wireless carriers can do it in < 8 hours |
| 06:29.45 | `Sauron | I guess the baby bells are more reluctant to give up customers |
| 06:29.56 | JerJer[mobile] | if i would have known Digium was in a pinch for gear I would have hopped into the twin cessna I have access to and flown out some gear |
| 06:29.57 | Essobi | JerJer[mobile] You want to see this h323 debug and tell me my cisco 3300 is broxored? :) |
| 06:30.09 | Essobi | hehe |
| 06:30.39 | JerJer[mobile] | coulda been there in 8 or 10 flight hours - maybe less |
| 06:31.33 | vinmohnj | hi |
| 06:32.07 | Essobi | JerJer[mobile] Ignore the php tags. :) http://www.pastebin.com/251810 |
| 06:32.19 | dontmsgme | Why would I ping my linux box from windows but not ssh in / |
| 06:32.44 | marshall | t3t - same result with the outbound proxy |
| 06:32.44 | JerJer[mobile] | dontmsgme: cuz you are smokin crack? |
| 06:32.48 | Essobi | MAhaha |
| 06:32.59 | t3t | which side are you missing audio from? |
| 06:33.03 | Qwell | JerJer[mobile]: Think you could fly me out a pizza? |
| 06:33.08 | marshall | calling out |
| 06:33.12 | cero64 | dontmsgme: possibly ssh isn't running? |
| 06:33.20 | marshall | when I phone to it from both PRI and other voip phones it works perfectly |
| 06:33.22 | JerJer[mobile] | Qwell: i don't think the pie would survive the flight |
| 06:33.34 | *** join/#asterisk file (~file@251.134.218.209.transedge.com) |
| 06:33.39 | JerJer[mobile] | esp since i missed dinner tonight |
| 06:34.03 | t3t | marshall: so when you place a call, you can't hear either side or just not the outgoing audio? |
| 06:34.03 | Qwell | JerJer[mobile]: I'll pay for 2...you can eat one on the way. |
| 06:34.15 | marshall | nothing - completely dead |
| 06:34.18 | marshall | but the other phone rings |
| 06:34.29 | t3t | marshall: what other phone? |
| 06:34.39 | Essobi | JerJer[mobile] When I push a button on a phone (only on one side this occurs) I get two RTP NTE's for digits pushed being sent and the first, being the proper one, the second being a "null" or something non-sensical to rtp.c. Did I mention it's your h323 lastest? |
| 06:34.43 | t3t | marshall: oh, the one you called... |
| 06:34.47 | marshall | yep |
| 06:35.08 | Essobi | The otherside of the NTE's which are being generated in * from a sip client, works perfectly. |
| 06:35.22 | Essobi | the Cisco side is acting retarded.. Heh. |
| 06:36.17 | marshall | t3t have you got a working sip.cfg/phone1.cfg file? |
| 06:36.40 | marshall | at least then I could narrow it down to the server settings |
| 06:36.46 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
| 06:36.56 | file | hi everyone, anyone come by the Digium booth today |
| 06:36.57 | file | ? |
| 06:37.16 | t3t | marshall: sure do.. working through nat too |
| 06:37.32 | Qwell | file: Nope, but I'll be there tomorrow if you get me a plane ticket |
| 06:37.33 | Essobi | file My boss did. |
| 06:37.34 | Essobi | :) |
| 06:37.37 | Qwell | assuming there is a tomorrow :p |
| 06:37.46 | Essobi | KABOOOM |
| 06:37.48 | marshall | t3t - are you port forwarding 5060 into the phone? |
| 06:37.54 | JerJer[mobile] | ekk |
| 06:37.58 | JerJer[mobile] | port fowardng - why? |
| 06:38.00 | t3t | marshall: try this: go to Menu/Settings/SIP config/ |
| 06:38.02 | t3t | no |
| 06:38.18 | JerJer[mobile] | its called register to proxy and have a stateful edge device |
| 06:38.21 | file | Essobi: who was your boss? |
| 06:38.25 | marshall | yep Im there t3t |
| 06:38.30 | t3t | the phone should establish the UDP connection |
| 06:38.38 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 06:39.13 | t3t | marshall: Do you see anything after "Outbound Add..."? |
| 06:39.13 | JerJer[mobile] | Essobi: what kind of crazy device is this |
| 06:39.35 | marshall | the internal ip of my local router |
| 06:39.39 | JerJer[mobile] | most of the problems that people are having with chan_h323 are now very very typical H.323 interop problems |
| 06:39.51 | t3t | marshall: that should be the IP of your * box |
| 06:40.00 | marshall | ahhh |
| 06:40.04 | marshall | ok one second |
| 06:40.05 | JerJer[mobile] | the ITU left too much open to interpretation |
| 06:40.05 | Essobi | It's a cisco skinny phone on an CCM3.2 box connected VIA a 33XX H323 <-> H323 leg. |
| 06:40.12 | Essobi | cisco 33XX |
| 06:40.26 | file | JerJer[mobile]: Snom got yelled at today in the 'IAX2 Better Than SIP' talk |
| 06:40.29 | t3t | so should "server address" obviously |
| 06:40.34 | Essobi | They won't let me drop an h323 straight into the CCM anymore. |
| 06:40.39 | marshall | yep |
| 06:40.43 | file | JerJer[mobile]: said there should be one protocol, SIP |
| 06:40.54 | JerJer[mobile] | Essobi: mmkay - why |
| 06:40.57 | `Sauron | Essobi: did you ever do SIP ccm <-> * ? |
| 06:41.00 | file | and one guy kept saying 'eye axe' |
| 06:41.00 | Essobi | *H323 rebooting the CCM cluster last time. |
| 06:41.11 | Essobi | Sauron No sip CCM |
| 06:41.13 | Essobi | just H323.. |
| 06:41.17 | shepherd | heh |
| 06:41.17 | `Sauron | hum |
| 06:41.23 | shepherd | so basically they are worried about iax |
| 06:41.32 | shepherd | and too lazy to come up with a product for it |
| 06:41.41 | JerJer[mobile] | there is SIP for CCM but its still listed as early adoptment and is only for trunkng |
| 06:41.48 | Essobi | Yea.. mid day crash due to something wierd in *h323 made about 5000 phones go down t5o the failover server. |
| 06:41.57 | t3t | is h.323 udp only? |
| 06:42.03 | JerJer[mobile] | lol no |
| 06:42.03 | Essobi | Nope |
| 06:42.09 | Essobi | S |
| 06:42.18 | JerJer[mobile] | h.323 is a huge umbrella protocol of shit |
| 06:42.21 | t3t | but it mainly uses udp, right? |
| 06:42.27 | Essobi | So they figured it'd be better to reboot a router, then the CCM box. :) |
| 06:42.37 | `Sauron | h.323 took ISDN and tried to turn it into a protocol |
| 06:42.41 | dontmsgme | I tried to get a Connect Voicepulse account for a DID like 2 weeksa go and they never emailed me back with my login information is their policy a lot more strict now |
| 06:42.43 | Essobi | Q931 |
| 06:42.44 | `Sauron | higher than layer 1 or 2 |
| 06:42.49 | Essobi | lovely |
| 06:43.01 | Essobi | JerJer[mobile] Anyways.. WTF is that thing doing? |
| 06:43.28 | t3t | their main problem was they they wanted to incorporate L1-3 protocols in an L4 protocol |
| 06:43.42 | Essobi | I was looking at rtp.c and I saw some support for a "cisco NTE" on payload 121 but.. I wasn't sure it that is what I was seeing there.. I was about to crack the code open again. |
| 06:43.54 | Essobi | t3t "wanted"? |
| 06:43.56 | Essobi | heh |
| 06:43.58 | t3t | What I was getting at was if they use UDP and the systems are accessible to the net, you could probably take them out with spoofed packets |
| 06:44.25 | dontmsgme | Does FWD supply DIDs? |
| 06:44.29 | t3t | untraceable... from anywhere |
| 06:44.31 | `Sauron | nope |
| 06:44.36 | `Sauron | just FWD numbers |
| 06:44.48 | t3t | no wonder nist wants separate networks for voip and data... stupid |
| 06:45.14 | Qwell | dontmsgme: try nufone, I hear they're good |
| 06:45.15 | JerJer[mobile] | no idea |
| 06:45.39 | Essobi | JerJer[mobile] is it just sending a retarded verion of rfc end-of-message NTE header, I'm wondering.. MHMMM. |
| 06:45.45 | dontmsgme | Are they like Voicepulse, never going to email me my DID login |
| 06:45.54 | t3t | I heard the same about nf... |
| 06:46.02 | Essobi | I'm going to get out tethereal tomarrow me thinks |
| 06:46.02 | Qwell | erm, MI DIDs only for now... |
| 06:46.24 | Qwell | always forget about that part |
| 06:46.36 | Qwell | and US48 toll free |
| 06:47.45 | t3t | JerJer[mobile]: are you guys going to jump on the LNP band wagon any time soon? |
| 06:48.37 | JerJer[mobile] | we can do LNP all day long |
| 06:48.46 | JerJer[mobile] | for Michigan DIDs |
| 06:49.05 | t3t | how about for a state with a population :) |
| 06:49.15 | JonR800 | like nebraska? |
| 06:49.23 | JerJer[mobile] | or montana? |
| 06:49.24 | Qwell | or idaho? |
| 06:49.32 | JonR800 | or north dakota? |
| 06:49.32 | Eight | ... or Minnesota =) |
| 06:49.35 | Qwell | Utah, perhaps |
| 06:49.36 | dontmsgme | Anyone know some way to make one of those disposable cellphone's really powerful antenna wise |
| 06:49.47 | dontmsgme | Can you unscrew that antenna off and put one on from radio shack like 8 times as long? |
| 06:49.59 | JonR800 | im not sure which is less populated.. north or south dakota.. |
| 06:50.08 | Qwell | I've heard of people from ND |
| 06:50.24 | Qwell | the only thing I know about SD, is that, thats where Mt Rushmore is |
| 06:50.34 | JonR800 | well that solves it in my mind, SD it is |
| 06:50.35 | Qwell | oh, and a city Sioux Falls |
| 06:50.42 | Eight | North is 642k |
| 06:50.48 | *** join/#asterisk jjg (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net) |
| 06:50.49 | Qwell | ND has Fargo, Dickenson...etc |
| 06:50.49 | Eight | South is 754k |
| 06:50.51 | jjg | hi |
| 06:50.52 | JonR800 | hahah |
| 06:50.53 | Qwell | ooo |
| 06:50.57 | Qwell | I lose |
| 06:50.58 | JonR800 | someone knows their google |
| 06:51.07 | Eight | JonR800: Wikipedia =) |
| 06:51.24 | JonR800 | ahh |
| 06:51.37 | Eight | MN is 4.9M =) |
| 06:51.46 | Eight | so get MN DIDs =) |
| 06:51.48 | *** join/#asterisk jmhunter (~jmhunter@64.77.199.223) |
| 06:51.48 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 06:52.00 | t3t | Wikipedia is too up to date. those are probably tomorrow morning's pop numbers. I want a 10-year census figure... |
| 06:52.09 | Eight | t3t: hehe. |
| 06:52.10 | *** mode/#asterisk [+o brc_] by jmhunter |
| 06:52.14 | jjg | i was wondering what the simplest "good" way to use a T-1 card as an interface to some sort of FXS hookup...for something like a 24 phone outbound setup using analog telephones...any recommendations? |
| 06:52.23 | Eight | I remember looking at wikipedia for tsunami info... that day. |
| 06:52.25 | JerJer[mobile] | jjg: TA-750 |
| 06:52.30 | JonR800 | lol |
| 06:52.36 | jjg | ok |
| 06:52.42 | jjg | i've seen those |
| 06:52.43 | JerJer[mobile] | Adtan |
| 06:52.59 | jmhunter | mmmm beer |
| 06:53.01 | jjg | do the fxs cards have rf11 ports on them or do i need somthing else? |
| 06:53.11 | Eight | rj11, afaik. |
| 06:53.15 | jjg | s/rf11/rj11 |
| 06:53.18 | JerJer[mobile] | my new house has a TE410P + TA-750 powering it - each room in the house has its own phone line |
| 06:53.20 | Essobi | Baag. |
| 06:53.22 | JerJer[mobile] | why? because I can |
| 06:53.30 | Essobi | dont' you hate it when you're googling for days |
| 06:53.32 | jjg | JerJer : ok, thanks for the tip |
| 06:53.36 | Essobi | and cursing in here |
| 06:53.46 | Eight | TA-750? |
| 06:53.55 | jjg | JerJer : so the person who told me i needed some sort of break out panel was wrong, right? |
| 06:53.56 | Essobi | and you find a match to a google, and it's the IRC logs on a webserver somwhere. :| |
| 06:54.09 | jjg | JerJer : that is a TA-750 with a break panel |
| 06:54.49 | Essobi | ta-750 is a PRI to Analog channel bank, if I remember right. |
| 06:54.57 | JerJer[mobile] | jjg: the adtran is going to gve you a amphenol connector and u run t into a 110 block |
| 06:55.21 | JerJer[mobile] | then just punch down the cable ends just like one would see in a busness |
| 06:55.29 | jjg | ok, right |
| 06:55.33 | Essobi | JerJer[mobile] So, you got any suggestions besides hacking rtp.c to make it happy? |
| 06:55.44 | JerJer[mobile] | nope - sorry |
| 06:55.50 | Essobi | Roger that. |
| 06:55.54 | JerJer[mobile] | haven't bothered to learn dtmf crap |
| 06:56.00 | Essobi | I've never heard of one way DTMF before. :) |
| 06:56.27 | JerJer[mobile] | i dig how IAX does it |
| 06:56.35 | Essobi | Anyways.. I'll get it mangled sooner or later.. probaby start bright and early |
| 06:56.37 | Essobi | night all |
| 06:56.58 | JerJer[mobile] | l8r |
| 06:57.01 | JerJer[mobile] | i need to crash myself |
| 06:57.14 | jjg | JerJer[mobile] the amphenol, where is a good web source for such connectors? |
| 06:57.14 | JerJer[mobile] | this workin first shift crap is not fun |
| 06:57.15 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 06:57.34 | Essobi | heh |
| 06:58.21 | jakepdev | i'm working through *@home. I can talk from phone to phone through *, but no audio from * when it says it's playing voice files... any ideas? |
| 06:58.29 | jjg | also, JerJer, when you were talking about punching down the cable ends, were you talking about the amphenol cable, or the phone cable? |
| 06:58.41 | jjg | srry for my basic questions :^\ |
| 06:59.00 | JerJer[mobile] | the house wiring |
| 06:59.19 | jjg | is there a name for that type of phone cabling? |
| 06:59.19 | JerJer[mobile] | i ran all of my lines in a star pattern (home-run) |
| 06:59.23 | JerJer[mobile] | cat 3 |
| 06:59.47 | JerJer[mobile] | usually, but i just ran cat 5 everywhere |
| 06:59.52 | jjg | wouldn't it have to be a star with the break out panel in the middle? |
| 07:00.19 | JerJer[mobile] | well i went nutz, you don't have to |
| 07:00.29 | Eight | So what does everyone prefer for 1 or 2U servers? I'm poking around for something single CPU, RAID 1 capable. |
| 07:00.43 | JerJer[mobile] | Dell 1750/1850 |
| 07:01.00 | Silik0n | Eight: for colo? |
| 07:01.01 | SexyKen | How can I configure cdr_addon_mysql to group by account code? |
| 07:01.03 | jjg | JerJer : you running Fedora on those or homebrew? |
| 07:01.09 | jjg | or RHEL? |
| 07:01.16 | JerJer[mobile] | jjg: all of the phone jack wires go back to the server room in my basement |
| 07:01.20 | Silik0n | GENTOO IS FOR RICERS |
| 07:01.27 | Eight | Silik0n: client site. |
| 07:01.34 | t3t | jjg: try http://www.monoprice.com/products/subdepartment.asp?c_id=105&cp_id=10515&style= for parts |
| 07:01.44 | jjg | t3t : thanks |
| 07:01.52 | JerJer[mobile] | where they get punched down into a 66 block... each on their own interface to the T-1 |
| 07:01.55 | JerJer[mobile] | well channel |
| 07:02.02 | Silik0n | Eight: Proliants are nice if they pay for them... Remote Insite Lights Out means console access from a web browser etc |
| 07:02.17 | jjg | is the punch down block a dumb panel or does it have electronics in it? |
| 07:02.19 | JerJer[mobile] | jjg: i have rolled my own distro of linux |
| 07:02.22 | JerJer[mobile] | dumb |
| 07:02.23 | jjg | lemme find one |
| 07:02.55 | Eight | Silik0n: ya... I have a political-ish aversion to HP atm. |
| 07:03.07 | jjg | JerJer : do you request 80 pin connectors so that you can run UDMA up to 6? |
| 07:03.13 | Silik0n | and I'm not talking just IP-KVM stuff with the remote insite... its just like standing there at the kb attached to the ox cept you cant swap hardware |
| 07:03.27 | JerJer[mobile] | jjg: we haven't, no |
| 07:03.30 | t3t | jjg: dumb |
| 07:03.57 | Silik0n | i dont care for HP personally but the features they inherited from compaq are pretty freakin nice... specially if you have to remotely manage a buncha boxes |
| 07:04.15 | jjg | JerJer : on the dell, do you just go basic config ... ata drive, no dma? |
| 07:04.21 | JerJer[mobile] | we run with a simple software mirror and the redunant power supps |
| 07:04.36 | JerJer[mobile] | we run dma |
| 07:04.43 | jjg | just llike 2 or so? |
| 07:04.52 | jjg | with standard 40 pin connects on SATA? |
| 07:04.57 | JerJer[mobile] | oh hell no |
| 07:04.58 | JerJer[mobile] | scsi |
| 07:05.01 | jjg | ok |
| 07:05.03 | Silik0n | scsi++ |
| 07:05.31 | JerJer[mobile] | and just a software raid mirror - no need to be all hardcore with raid controllers and JBODs |
| 07:05.32 | JerJer[mobile] | blah |
| 07:05.59 | jmhunter | actually its a restaurant |
| 07:07.53 | *** part/#asterisk JerJer[mobile] (~jj@65.173.197.109) |
| 07:08.13 | jjg | tft : i'm looking at that page for the fxs blocks ( what's the terminology? ) ... which one of these models should i be looking at for use with the amphenol to the T100 |
| 07:08.38 | jjg | glad they are cheap |
| 07:08.57 | Silik0n | jjg: what are you doing? |
| 07:09.49 | jjg | wanting to run up to a 24 phone outbound rig with analog phone -> fxs breakout panel -> cat2? -> T100 -> VoIP |
| 07:10.28 | jjg | Silikon : oops missed the TA-750 |
| 07:10.29 | jjg | heh |
| 07:10.35 | jjg | so that would be : |
| 07:10.42 | Newbie___ | hi, i try to connect 2 * using IAX, when i make the call i get Rejected connect attempt , any idea ? |
| 07:10.57 | Silik0n | iax.confisnt correct? |
| 07:11.13 | jjg | phone -> FXS breakout panel -> amphenol connect -> TA-750 -> cat2? -> T100 -> * |
| 07:11.30 | Newbie___ | seems to be, i use voip-info.org example |
| 07:12.05 | Silik0n | jjg: just go to your local phone supply house and get a 66 block w/ a amphenol connector prewired then get a male-male amphenol cable to connect the 66 block to the adtran |
| 07:12.41 | jjg | does the cat2 between the 750 and the T100 need to be crossover? |
| 07:12.44 | Silik0n | you should have like a greybar or anixter around there |
| 07:12.49 | Silik0n | yes |
| 07:12.50 | Newbie___ | client log into remote successful, just cant make calls |
| 07:13.18 | Silik0n | jjg: both the 750 and the T100 T1 ports are wired CPE style |
| 07:13.31 | jjg | CPE as in customer premise equip? |
| 07:13.36 | Silik0n | yeah |
| 07:13.56 | Silik0n | just make you a "T1 cross over cable" its a little different from a ethernet cross over |
| 07:14.03 | jjg | not familiar with what that means exactly, but do understand that i need a xover |
| 07:14.23 | jjg | Silikon : ok, thanks for that |
| 07:14.31 | jjg | then this seems VERy accessible |
| 07:14.52 | jjg | so the setup sounds like this now ( bit more detail ) |
| 07:14.55 | Silik0n | http://www.voip-info.org/wiki-crossover+T1+cable |
| 07:16.01 | Silik0n | and you can make that cross over cable with CAT3 or CAT5 cable it doesnt have to be cat2 |
| 07:16.11 | jjg | analog phone -> ( rj-11 / cat3 ) -> FXS breakout panel prewired with amphenol -> male2male amphenol connect -> TA-750 -> T1 crossover with ( cat2 or cat5 ) -> T100 -> * |
| 07:16.23 | Silik0n | yeah |
| 07:16.29 | jjg | HEELLZZ yah |
| 07:16.45 | jjg | and the lord shall keep oonn givin |
| 07:16.51 | Silik0n | hah |
| 07:17.01 | jjg | i mean good lord |
| 07:17.05 | jjg | shall |
| 07:17.05 | jjg | heh |
| 07:17.27 | jjg | any quality drop with the cat3/5 vs the cat2? |
| 07:17.37 | jjg | a penny in time ... |
| 07:19.00 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 07:19.52 | Silik0n | none to worry about |
| 07:20.37 | Silik0n | unless you plan on putting the damned channelbank a mile from your asterisk box (then you're gonna have drive issues from the line drivers in the chanbank and the t100 anyway |
| 07:22.37 | SexyKen | Anyone know why I'd get this: res_config_mysql.c:418: storage size of `mysql_engine' isn't known |
| 07:23.03 | jjg | anyone have an recommendation on a model for a good cheap analog phone? |
| 07:23.08 | *** join/#asterisk Nivex (kjotte@user-0ce2jqe.cable.mindspring.com) |
| 07:23.22 | jjg | a phone that may be destroyed by college dudes |
| 07:23.29 | jjg | but with decent performance |
| 07:23.35 | jjg | so cheap |
| 07:23.36 | jjg | heh |
| 07:26.00 | shepherd | jjg: anything $9 |
| 07:26.19 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 07:27.52 | jakepdev | shepherd - do you know if there's a different place to conifgure audio for what * says vs a conversation between two phones? I get audio between the 2 phones through *, but no audio from *. |
| 07:28.26 | shepherd | explain |
| 07:28.35 | tuxinator_linux | ~weather ksjc |
| 07:28.49 | file | it's nice outside |
| 07:28.49 | jjg | shepard : ok thanks |
| 07:28.59 | shepherd | i hate you all |
| 07:29.00 | shepherd | hehe |
| 07:29.04 | jakepdev | 8000 DTA calls 8001 softphone through * - I hear audio and dtmf |
| 07:29.18 | shepherd | i wish it was like 40 |
| 07:29.32 | tuxinator_linux | file: You're in San Jose? |
| 07:29.42 | jakepdev | I see on console "Playing 'vm-login' |
| 07:29.53 | file | yes |
| 07:29.53 | jjg | anybody played with linphone on ipaq and asterisk? if so, what codec? u/alaw was terrible |
| 07:29.55 | jakepdev | (when I dial *98) |
| 07:30.07 | file | I'm in a hotel room... with bkw, drumkilla, and krammy boy at the moment |
| 07:30.13 | jjg | i heard there was a voip thing in san jose soon |
| 07:30.13 | jakepdev | but I don't hear anything |
| 07:30.14 | tuxinator_linux | I will be there in 8 hours |
| 07:30.18 | file | we're watching The Matrix and looking at bugs |
| 07:30.23 | djorange | hello got a couple question.. what port do i open up in my router for some1 to connect to my * box |
| 07:30.26 | file | while I try to stay concious |
| 07:30.47 | jjg | the embedded conference in frisco today was boring as hell |
| 07:30.52 | tuxinator_linux | file: Mind if I come find you today? |
| 07:30.53 | jjg | the exhibitions were assy |
| 07:30.57 | shepherd | djorange: which protocol |
| 07:31.15 | *** join/#asterisk jas_williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 07:31.20 | file | tuxinator_linux: tomorrow you mean? |
| 07:31.21 | djorange | i don't know what do i need to do. i got some1 who wants to connect to my * |
| 07:31.37 | tuxinator_linux | well, ya, it's 12:31 here, so it is Thursday |
| 07:31.59 | file | ah you're coming down to SJC? sneaking into VON or just wanna show up? |
| 07:32.20 | tuxinator_linux | sneaking into VON, and also going to Meet * |
| 07:32.35 | shepherd | HAHAHAH |
| 07:32.44 | djorange | isn't like port 5080 or something? |
| 07:32.47 | shepherd | i just saw this on voip-info.org |
| 07:32.47 | tuxinator_linux | file: I'm staying at the Marriott |
| 07:32.52 | djorange | i remmebre theres like 3 |
| 07:32.58 | file | we're in the Hyatt... |
| 07:33.05 | shepherd | News |
| 07:33.06 | shepherd | This section is for news, ie news reports, press releases, product release announcements etc |
| 07:33.06 | file | but if you wanna visit the Digium booth I'll be there |
| 07:33.11 | shepherd | # 2005-03-09 - voip-info readers annoyed by vonage spam |
| 07:33.21 | file | so will everyone else... swing by |
| 07:33.29 | tuxinator_linux | will do |
| 07:33.37 | file | can't miss it, big turning Digium/Asterisk... orange shag carpet... |
| 07:33.49 | tuxinator_linux | shag, ya baby |
| 07:33.54 | file | it's pretty |
| 07:34.07 | tuxinator_linux | APC said they will give me a VON ticket, we'll see |
| 07:34.14 | file | goody |
| 07:34.27 | tuxinator_linux | otherwise, is there a back door? |
| 07:34.35 | tuxinator_linux | underground passage? |
| 07:34.49 | tuxinator_linux | I could repel from the ceiling |
| 07:34.59 | jjg | how much are the VON tickets? |
| 07:35.03 | tuxinator_linux | 200 |
| 07:35.06 | file | I just walk in with my pass, silly you |
| 07:35.23 | tuxinator_linux | nah, to easy |
| 07:35.24 | jjg | 200! you gotta be fucking kidding me |
| 07:35.37 | jjg | no comment |
| 07:35.38 | tuxinator_linux | 200 for exhibit only |
| 07:36.17 | tuxinator_linux | jjg: http://www.von.com/register.html |
| 07:36.25 | jakepdev | anyone using *@home? |
| 07:36.42 | tuxinator_linux | jakepdev: I hear it works |
| 07:37.00 | jakepdev | any special config to get IVR prompts working? |
| 07:37.24 | file | I wonder where p0lar is, he swung by the booth today |
| 07:37.39 | tuxinator_linux | file: I talked to him earlier, he went home |
| 07:37.52 | file | k |
| 07:38.05 | file | I can't feel my foot |
| 07:38.18 | tuxinator_linux | bite it |
| 07:39.27 | djorange | do you know what port i need to open on my router for people to connect to my * with xlite? |
| 07:39.36 | jjg | i wanna see who is running von, cause that is the funniest thing i've ever heard |
| 07:39.52 | jjg | oh but wait wait |
| 07:40.03 | tuxinator_linux | file: (23:21:24) p0lar: ok.. I'm out -- time to pass out on an airplane for 5 hours. :( |
| 07:40.20 | jjg | the 2695.00 for the package DOES include access to the Town Hall Meeting!!!!! |
| 07:40.27 | jjg | haaaaaaaaaaaaaaaaaa |
| 07:40.33 | jjg | ok, i'm done |
| 07:41.20 | Juggie | http://www.ottawabusinessjournal.com/311916555482103.php |
| 07:41.20 | jjg | so is anyone paying 200 dollars to go see the exhibitions? |
| 07:41.45 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 07:41.49 | Juggie | canada fed gov "notifies" telecom industry its going to be buying a ton of phones/servers for voip in the next 1-3 years |
| 07:41.52 | PTG123 | hey anyone have an older copy of asterisk-addons |
| 07:41.53 | tuxinator_linux | jjg: seems kinda silly |
| 07:42.02 | Juggie | it basically means they will buy 1/4million phones |
| 07:42.02 | jjg | hahahaha |
| 07:42.03 | jjg | ha |
| 07:42.03 | jjg | haha |
| 07:42.06 | jjg | i mean |
| 07:42.10 | jjg | hahaha |
| 07:42.15 | jjg | im sry |
| 07:42.17 | jjg | really |
| 07:42.24 | jjg | i'm sure it will be worth every penny |
| 07:42.28 | tuxinator_linux | jjg: need some sleep? |
| 07:42.32 | jjg | uh yah |
| 07:42.43 | jjg | had 3 hours and spent all day at the embedded conference today |
| 07:42.46 | jjg | which was free |
| 07:42.48 | jjg | heh |
| 07:43.08 | Juggie | jjg we have nothing to do with von |
| 07:43.11 | Juggie | thats pulver |
| 07:43.15 | jjg | oh i know |
| 07:43.15 | tuxinator_linux | I need to go get embedded with my wife in a few minutes |
| 07:43.16 | Juggie | www.pulver.com |
| 07:43.17 | Eight | Nobody's had any issues with the TDM400P cards on PCI-Express, right? |
| 07:43.17 | *** join/#asterisk three55ml (~who@cpe-66-68-110-140.austin.res.rr.com) |
| 07:43.23 | jjg | i'm laughing my ass off at whoever does |
| 07:43.30 | jjg | pulver |
| 07:43.31 | three55ml | Anyone have any experience with the DVG-1120? |
| 07:43.37 | PTG123 | anyone have an old version? :) anyone anyone? |
| 07:43.40 | jjg | that guy has ballz |
| 07:43.49 | djorange | can some1 help me? |
| 07:43.50 | Juggie | eight, pci express slots of chipset you mean? |
| 07:43.51 | djorange | with port? |
| 07:43.51 | jjg | next year they'll pay you |
| 07:44.04 | *** join/#asterisk ynn77 (~Ming2k4@65.75.172.100) |
| 07:44.04 | Juggie | *of=or |
| 07:44.07 | djorange | i need to open open what ports on the router for * to work? |
| 07:44.12 | jjg | including access to the Town Hall Meeing |
| 07:44.16 | jjg | hahahah |
| 07:44.17 | jjg | :D |
| 07:44.20 | Juggie | jjg, get bent.... |
| 07:44.25 | Juggie | stop your rant and go away |
| 07:44.25 | Eight | Juggie: TDM400P card in a PCI-Express slot |
| 07:44.34 | jjg | i'm a lil upset that it is 200 |
| 07:44.36 | jjg | can you tell? |
| 07:44.37 | jjg | i mean |
| 07:44.41 | jjg | i'd REALLY like to go |
| 07:44.42 | Juggie | eight, if it fits, it will work :) |
| 07:44.43 | shepherd | djorange: www.voip-info.org |
| 07:44.55 | jjg | but, uh...i can't |
| 07:45.00 | shepherd | search for nat |
| 07:45.01 | Juggie | i dont know if pci express is backword compat with pc, never read up on it. |
| 07:45.04 | Eight | Juggie: Ya, I figured as much. But stranger things have happened =) |
| 07:45.05 | Juggie | *pci |
| 07:45.20 | Juggie | is it? |
| 07:45.30 | Juggie | do any pci cards go in a pci-express slot? |
| 07:45.34 | Eight | http://en.wikipedia.org/wiki/PCI-Express |
| 07:45.36 | *** join/#asterisk IronHelix (~irc@ool-182c8f9f.dyn.optonline.net) |
| 07:46.17 | shepherd | PCI-E is a completely different beast, with a different form-factor. |
| 07:46.17 | shepherd | Traditional PCI cards won't fit into the slot. |
| 07:46.34 | jjg | wait, Juggie...did you pay two hundies? |
| 07:46.40 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 07:46.52 | djorange | shepherd |
| 07:47.07 | Juggie | thats what i was thinking but ive been wrong before |
| 07:47.23 | djorange | shepherd: do u know which ports i need to open to connect to my * box? |
| 07:47.43 | shepherd | djorange: depends on the protocol |
| 07:47.47 | jas_williams | djorange: What protocol ? |
| 07:48.01 | Zeeek | jas_williams don't you ever sleep? |
| 07:48.03 | djorange | shepherd: im using sip |
| 07:48.09 | *** join/#asterisk ikey (~kirankuma@202.54.37.186) |
| 07:48.11 | Eight | Aha, was confusing PCI-X with PCI-E. |
| 07:48.17 | jas_williams | Zeeek: Just got up ;-) |
| 07:48.18 | Eight | surprise surprise. |
| 07:48.31 | Zeeek | jas_ you use dhcp and tftp ? |
| 07:48.48 | jas_williams | Yes |
| 07:49.02 | djorange | shepherd: what rtsp ports do i need to open |
| 07:49.15 | Zeeek | I have tftp-hpa running (and dhcp) : both work independently |
| 07:49.18 | *** join/#asterisk bonbon-home (~happy@81-86-0-190.dsl.pipex.com) |
| 07:49.26 | jas_williams | DHCP on my wireless router and tftp on my asterisk |
| 07:49.33 | Zeeek | but I can't get formware to load unsing next-server and filename |
| 07:49.34 | bonbon-home | anyone know the best way of remotely detecting a deadlock ? |
| 07:49.47 | ikey | hi can any one help me in asterisk...need some clarifications |
| 07:49.59 | Zeeek | clarificationsRus |
| 07:50.19 | jas_williams | Zeeek: Don't use next server use option tftp server |
| 07:50.25 | ikey | hi Zeeek can u help me |
| 07:50.34 | tuxinator_linux | ask your question ikey |
| 07:50.36 | Zeeek | aha! that would be slick if it's the anwser |
| 07:50.50 | Zeeek | I'll let you know from the office - thanks for the direction:) |
| 07:50.56 | Zeeek | ikey Ask! |
| 07:51.01 | Zeeek | someone will help |
| 07:51.17 | ikey | yeah does asterisk act as SIP server and H323 gateway also? |
| 07:51.23 | jjg | so nobody at VON is presenting papers? |
| 07:51.23 | Zeeek | yes |
| 07:51.37 | shepherd | can't be a sip proxy though |
| 07:51.51 | Zeeek | aha anticipation |
| 07:51.55 | Zeeek | good. |
| 07:52.02 | jjg | i see a bunch of people's names as speakers, but no papers? |
| 07:52.11 | ikey | what are the features that asterisk sip server can do |
| 07:52.26 | shepherd | asterisk can do * |
| 07:52.27 | shepherd | :) |
| 07:52.28 | Zeeek | Starter tutorial: |
| 07:52.28 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 07:52.28 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 07:52.28 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 07:52.28 | Zeeek | THE reference of the moment: |
| 07:52.28 | Zeeek | http://www.asteriskdocs.org |
| 07:52.38 | shepherd | just not sip proxy |
| 07:52.40 | Zeeek | ikey read the first two |
| 07:52.45 | jjg | The legendary All Conference Party on Wednesday night |
| 07:52.53 | SexyKen | Anyone know why I'd get this: res_config_mysql.c:418: storage size of `mysql_engine' isn't known |
| 07:52.53 | ikey | ok great |
| 07:53.04 | SexyKen | I can't not figure out for the life of me how to fix that...but I can't compile asterisk-addons |
| 07:53.13 | jjg | there are a lot of exhibitors atleast |
| 07:53.23 | shepherd | sexyken: is that with head? |
| 07:53.35 | ikey | and in case if we need to distribute asterisk commercially how to go about it |
| 07:53.51 | shepherd | ikey: sales@digium.com |
| 07:53.56 | shepherd | or |
| 07:53.59 | shepherd | be@digium.com |
| 07:54.08 | SexyKen | shepherd - I think |
| 07:54.16 | shepherd | sexy: try stable |
| 07:54.21 | shepherd | head might have broken it |
| 07:55.05 | ikey | ok great |
| 07:55.07 | PTG123 | ok |
| 07:55.12 | PTG123 | add-ons will not compile |
| 07:55.18 | PTG123 | with the latest cvs |
| 07:55.23 | ikey | does asterisk support Voice Recognition and text to speech? |
| 07:55.27 | PTG123 | AST_LIST_REMOVE in asterisk has 3 params, in addons it needs 4 |
| 07:55.52 | shepherd | ikey: yes (as an addon i believe) and yes |
| 07:56.18 | bonbon-home | shpherd - voice recognition using what? |
| 07:56.44 | shepherd | maybe not, heh |
| 07:56.58 | shepherd | i'm pretty sure someone was doing something in that area |
| 07:57.49 | ikey | how to getit working ..we were sucessful in using it as IVRS and when some one calls up ivrs and selects language options they need to key in the number of the option on their phone but we need to activate it on Voice Recog(dsp) in such a way that instead if pressing number they say "english" for selecting option english |
| 07:58.22 | ta[i]nted | ikey i think u want speech to text |
| 07:58.48 | ta[i]nted | and voice recognition sucks |
| 07:58.53 | ta[i]nted | people hate it |
| 07:58.54 | SexyKen | So what do we do? |
| 07:59.05 | shepherd | i don't think voice recognition is free |
| 07:59.16 | shepherd | but someone out there is doing it |
| 07:59.16 | ta[i]nted | why can't your users just press a button |
| 07:59.17 | ikey | yeah u are rt i need speech to text and text to speech |
| 07:59.40 | ta[i]nted | if they are that lazy, maybe they should put down the phone to conserve energy |
| 07:59.53 | bonbon-home | shpeherd - yes there are companies doing it |
| 08:00.13 | *** join/#asterisk sale1357 (~ss@pcp09129941pcs.arlngt01.va.comcast.net) |
| 08:00.50 | ikey | yeah we have deployed 400 e1 infrastructure across diffeent countries on premium services and we have a competitor who is working on NMS solutions who offers speech to text |
| 08:01.11 | ikey | so we are now forced to convert all the infra into wuch kind |
| 08:01.31 | ikey | did any one installed speech to text on asterisk.. |
| 08:03.04 | Qwell | ikey: I think you'll probably want to look into sphinx |
| 08:03.08 | sale1357 | does anyone know anything about the channel hanging when you try and record with *77? |
| 08:08.37 | *** join/#asterisk ph_matrix (potchy_fem@203.115.169.48) |
| 08:10.00 | ph_matrix | hello.. can i ask simply question ? newbie here |
| 08:10.14 | sale1357 | oh you can ask |
| 08:10.29 | sale1357 | whether anyone will answer you or not is another question. |
| 08:10.37 | ph_matrix | :) |
| 08:12.00 | shepherd | ?? |
| 08:12.06 | ph_matrix | my q is about the asterisk FXO and FXS.. those module is not toggble to act as an FXO or FXS ? |
| 08:12.16 | shepherd | no |
| 08:12.22 | Eight | ph_matrix: you have to swap modules on the card. |
| 08:12.22 | shepherd | they are not |
| 08:13.01 | ph_matrix | ic... what about in the case for Wildcard E1/T1 ? |
| 08:13.07 | *** join/#asterisk soulz- (~soulz@host-137-132-45-89.imcb.nus.edu.sg) |
| 08:13.27 | shepherd | e1/t1's usually don't use either |
| 08:13.27 | Eight | that doesn't work as an FXS *or* and FXO =) |
| 08:13.50 | soulz- | hi all, is there any bug on Asterisk CVS-HEAD-03/07/05-17:14:42 for bridging calls? |
| 08:14.07 | Eight | ph_matrix: But you can run a channel bank or talk to the telco with any port, which is probably more what you're interested in. |
| 08:14.19 | Eight | with the Wildcard T/E1 cards. |
| 08:14.56 | ph_matrix | ic... i have to read a lot about this.. im interested in putting up a voip gateway in my country.. |
| 08:15.09 | Eight | ph_matrix: Ya, there's a phenomenal amount of reading to be done. |
| 08:15.20 | Eight | The voip-info wiki is a good place to start. |
| 08:15.46 | *** join/#asterisk iamx (~DmD@pppoe50-99-luxdsl-220.pt.lu) |
| 08:15.51 | iamx | Hi |
| 08:16.02 | ph_matrix | il already setup an asterisk server in my linux box already and tested it wid a SIP phone.. it works.. my problem now is how to connect my box to the telco.. |
| 08:16.25 | Eight | ph_matrix: well, you're looking at the right cards. |
| 08:16.35 | ph_matrix | :) |
| 08:16.55 | ph_matrix | the cards cost a lot i think.. |
| 08:17.02 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 08:17.18 | tuxinator_linux | 500 something USD |
| 08:17.20 | tuxinator_linux | not bad |
| 08:17.36 | tuxinator_linux | well, the 4 port are 1500 USD I think |
| 08:17.49 | Eight | ya, but that's 4x24 lines. |
| 08:17.58 | Eight | That's alot of lines =) |
| 08:18.00 | tuxinator_linux | Cost per line, is LOW |
| 08:18.40 | Eight | That's 15.50 per line. |
| 08:18.40 | tuxinator_linux | 4 lines would cost about 2000 USD |
| 08:18.41 | ph_matrix | yup.. it is very cheap in ratio |
| 08:19.05 | tuxinator_linux | 2000 / (23 * 4) |
| 08:19.11 | tuxinator_linux | = per line monthly |
| 08:19.19 | tuxinator_linux | well, not ture |
| 08:19.28 | Eight | tuxinator_linux: wait, what are you tlaking about now? |
| 08:19.35 | *** join/#asterisk Delmar (~Delmar@222-152-57-78.adsl.inspire.net.nz) |
| 08:19.37 | tuxinator_linux | 2000 / ((24*4)-1) |
| 08:19.45 | ph_matrix | doest E1/T1 cards need a lot of memory for the linux box ? |
| 08:19.48 | tuxinator_linux | PRI lines to connect to the cards |
| 08:20.05 | tuxinator_linux | ph_matrix: You need more CPU power than anything |
| 08:20.07 | iamx | humm, i have a problem with asterisk, everything works fine, trunks to other peers, internal calling, but just the internal things like festival, or any other internal playback don't work, did anyone have the same problem ? |
| 08:20.37 | Eight | iamx: Festival flaked out on me, but sound playback works still. |
| 08:20.40 | ph_matrix | is 2.5Ghz athlon 512 ddr mem can do ? |
| 08:20.54 | *** join/#asterisk Alexis (~alexis@www.trim.it) |
| 08:20.54 | Eight | ph_matrix: it depends very specifically on what you're doing. |
| 08:20.56 | jjg | anyone here used linphone on an ipaq successfully with asterisk? |
| 08:20.59 | Delmar | ph_matrix yep. |
| 08:21.12 | jjg | i think i asked that eralier, but can't remember |
| 08:21.32 | Delmar | ph_matrix even if you had say.. E1 (30 channels) all running g729 at once, it would be fine dude. |
| 08:21.33 | tuxinator_linux | ph_matrix: A little week |
| 08:21.46 | ph_matrix | ic.. what can u recomend ? |
| 08:21.48 | Eight | If you're running all 95 external lines, doing meetme sessions, transcoding a bunch for SIP->SIP conversations... not so much =) |
| 08:22.02 | *** join/#asterisk godsmoke (giovani@66-108-159-216.nyc.rr.com) |
| 08:22.23 | iamx | humm, it doesn't even give out an error message and when i 1234 it should read back my number but i just hear little 500 ms chunks of sound every 5 seconds or so |
| 08:22.29 | *** join/#asterisk fjoe (~fjoe@samodelkin.net) |
| 08:22.29 | ph_matrix | a gig of memory ? |
| 08:22.31 | fjoe | re |
| 08:22.33 | Delmar | hey does anyone here know what the hell I can do to increase the TX_Gain in zapata.conf without causing huge echo? its driving me nuts. |
| 08:22.42 | Eight | ph_matrix: the memory isn't so much the issue. |
| 08:22.46 | fjoe | anyone using quadBRI with *? |
| 08:22.46 | tuxinator_linux | ph_matrix: At least a dual pent 4 or xeon with 2-4Gigs of RAM. But that may be overkill or not enough |
| 08:22.52 | Eight | ph_matrix: it's the CPU power of the transcoding. |
| 08:22.59 | ph_matrix | ic.... |
| 08:23.05 | tuxinator_linux | fjoe: BRI or PRI? |
| 08:23.27 | fjoe | tuxinator_linux HFC-S-based quad_BRI_ cards by Junghanns.net |
| 08:23.31 | Eight | ph_matrix: but unless you start getting real serious most moderately well equipped boxes are plenty. |
| 08:23.43 | tuxinator_linux | fjoe: Sorry, can't help you there |
| 08:24.05 | tuxinator_linux | What I am still doing away, need to get some sleep before my flight in a 7 hours |
| 08:24.25 | *** join/#asterisk Alexis (~alexis@www.trim.it) |
| 08:24.26 | ph_matrix | ic .. well ok il upgrade as demands rise.. |
| 08:24.29 | Alexis | hi |
| 08:24.56 | tuxinator_linux | ph_matrix: You need a High Availible setup |
| 08:25.01 | fjoe | 4 |
| 08:25.03 | *** join/#asterisk schurig (~schurig@p54B296F1.dip0.t-ipconnect.de) |
| 08:25.08 | soulz- | anyone have a billing solution for asterisk? |
| 08:25.13 | tuxinator_linux | Night guys |
| 08:25.15 | Delmar | When a call comes in via the FXO, and goes to voicemail, the audio/volume is way to low. When I access the voicemail internally ie. from a SIP phone, the voicemail prompts and such sound fine. can the voicemail itself have the volume increased? |
| 08:25.43 | jakepdev | still can't get this working - I can't hear any voice prompts from *- there must be a config option somewhere. Any ideas anyone? |
| 08:25.52 | ph_matrix | yup.. have any site also that teach like ideal calls per CPU power ? |
| 08:26.04 | Delmar | really what I wanna do is increase the TX_Gain, but when I do that, the self-echo on the SIP client device is shockingly bad. |
| 08:26.12 | Juggie | Delmar, is the volume low for everything |
| 08:26.53 | Delmar | Juggie, yeah the tx volume on the FXO is generally low.... |
| 08:26.55 | iamx | jakepdev i think i have the same problem |
| 08:27.00 | Delmar | but if i increase it... im in the poo with echo. |
| 08:27.02 | Juggie | change your gain then delmar. |
| 08:27.09 | Juggie | did you enable echo detection? |
| 08:27.14 | jakepdev | iamx - are you using *@home? |
| 08:27.15 | Delmar | tried... set to 10 now and its crappy. |
| 08:27.21 | fjoe | tuxinator_linux do you use PRI with asterisk? |
| 08:27.26 | Delmar | and even at 10, the voicemail prompts are still crappy. |
| 08:27.33 | Juggie | Delmar, 10 is TOO much, voicemail is low? |
| 08:27.35 | iamx | jakepdev yes |
| 08:27.57 | Delmar | yeah, i mean.... |
| 08:28.37 | jakepdev | iamx - I've been able to talk between two phones at least. Did you try that? |
| 08:28.39 | iamx | everything works fine just the voiceprompts and festival things don't work |
| 08:28.45 | jakepdev | ok |
| 08:28.52 | Delmar | Juggie i had the tx_gain set at a level where the echo was still there, but the canceller would kick in after a few secs and kill it off... and at that level.. sometimes ppl say.. they can hardly hear me.. so i just have to have the phone piece right up at my face.... |
| 08:29.07 | Delmar | so at that level.. its bareable.. BUT.. voicemail is a joke as far as volume goes.... |
| 08:29.19 | Delmar | so really.. i need to find middle ground by increasing voicemail volume..... |
| 08:29.22 | *** part/#asterisk jjg (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net) |
| 08:29.24 | iamx | yes, i can communicate internally and even have peerings with other providers, everything works, just not the voice thingy |
| 08:29.51 | *** join/#asterisk kram (~mark@kram.digium.sponsor.pdpc) |
| 08:29.51 | *** mode/#asterisk [+o kram] by ChanServ |
| 08:29.53 | Delmar | frankly... i have wicked echo issues which are slightly improved by lowering the TX_Gain, but other things suffer. |
| 08:29.54 | jakepdev | iamx - strange thing is I had it working with astwind at one point (w/voice prompts) - must be a config issue.. wish someone on here knew |
| 08:30.00 | shepherd | mark |
| 08:30.30 | Delmar | jakepdev, when u say voicemail isnt working... how are u connecting to it to test it? |
| 08:30.30 | iamx | hummm |
| 08:30.50 | Delmar | jakepdev i had issues like yours.. i might be able to help. |
| 08:30.50 | jakepdev | i tried through SJPhone, and my analog phone to DTA310 |
| 08:30.51 | Juggie | Delmar, when u make a sip->pstn call, how is it? |
| 08:31.05 | jakepdev | dialed *98 |
| 08:31.23 | jakepdev | says playing vm-login |
| 08:31.27 | jakepdev | but no sound |
| 08:31.37 | iamx | jakepdev does 1234 read back your number ? |
| 08:31.40 | Delmar | Juggie heh, and that is the wierd thing.. its usually very good... sip->pstn i really dont have an issue with... incomming pstn is echo big time. |
| 08:31.42 | jakepdev | let me see |
| 08:31.55 | Delmar | jakepdev ok what application u using now .. or shat sip client? |
| 08:32.03 | jakepdev | on the screen it does :) |
| 08:32.07 | Delmar | jakepdev, the issues I had were purely codec issues. |
| 08:32.14 | iamx | but no sound ? |
| 08:32.17 | jakepdev | right |
| 08:32.36 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:32.40 | Delmar | jakepdev oh and sometimes NAT can play a part...how are you connected.. .local lan stuff or what? |
| 08:32.49 | iamx | hmm k, so it's exactly the same problem as is have |
| 08:32.54 | Juggie | Delmar, i run with a gain of 3.0, and echocancel and echotraining set to yes |
| 08:32.57 | Juggie | and i am all good. |
| 08:32.58 | jakepdev | * installed inside VMWare |
| 08:33.07 | jakepdev | bridged mode |
| 08:33.09 | iamx | me too |
| 08:33.18 | jakepdev | on same subnet |
| 08:33.35 | *** join/#asterisk amer (~aaa@adsl-64-174-95-188.dsl.sntc01.pacbell.net) |
| 08:33.40 | iamx | i don't think it's a nat issue because normal communications work |
| 08:33.40 | Delmar | sounds good. ignore nat issues. look at codec issues. |
| 08:33.43 | jakepdev | (no firewalls or anything like that) |
| 08:33.53 | Delmar | do "show translation" |
| 08:33.57 | Delmar | on the * console |
| 08:34.09 | jakepdev | ok |
| 08:34.12 | amer | Has anyone used ast-ax-snmpd? |
| 08:34.13 | Delmar | what codec is the sip client trying to use? |
| 08:34.32 | amer | I need to download it but all links are dead |
| 08:34.34 | jakepdev | i told it to use ulaw |
| 08:34.50 | Delmar | Juggie yeah.. and echotraining=800 here |
| 08:34.57 | Juggie | wow... thats high |
| 08:34.57 | jakepdev | disallow=all allow=ulaw |
| 08:35.05 | Juggie | set it just to yes |
| 08:35.08 | Delmar | juggie, and im using the MARK3 echo canceler.. i have tried every other damn thing. |
| 08:35.08 | iamx | i used ulaw and gsm, both don't change anything |
| 08:35.09 | Juggie | which is 128 |
| 08:35.20 | Delmar | nah allow=ulaw is ok |
| 08:35.21 | Juggie | Delmar, what hardware are you using |
| 08:35.36 | Delmar | juggie, yuk X100P clone. |
| 08:35.40 | Juggie | :) |
| 08:35.54 | jakepdev | ok - i gues that's fine then |
| 08:36.01 | Delmar | juggie, yep. but it should be ok. |
| 08:36.06 | Juggie | well, i cant say much about analog stuff never used it.... |
| 08:36.19 | Juggie | but default cource no mods for echo stuff... i am ok |
| 08:36.22 | Juggie | *source |
| 08:36.29 | Juggie | sip to sip is ok right? |
| 08:36.34 | Delmar | Juggie yeah im really wanting a channel bank and a T1 card. |
| 08:36.36 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:36.38 | Delmar | yup |
| 08:36.40 | jakepdev | yep |
| 08:36.48 | Delmar | sip to sip, sip to iax and all that crap |
| 08:36.53 | jakepdev | sip -> * -> sip is fine |
| 08:37.02 | nirs | ~seen kram |
| 08:37.03 | jbot | kram is currently on #asterisk (7m 12s) |
| 08:37.03 | Delmar | and even incomming FXO seems to work great.... |
| 08:37.27 | Juggie | hrmm.... |
| 08:37.31 | Juggie | well a gain of 10 is alot |
| 08:37.35 | Juggie | thats db remember |
| 08:37.41 | Juggie | unless u said 10% |
| 08:37.50 | nirs | kram, if you're in here, please respond |
| 08:38.03 | Delmar | Juggie yeah that was a test just now. echo like hell on an incomming call, and didnt really improve the voicemail volume much at all... |
| 08:38.13 | Delmar | no .. 10. not 10% |
| 08:38.25 | Juggie | well 10db gain is alot |
| 08:38.28 | Juggie | thats what i am saying |
| 08:38.30 | Juggie | i run with 3 |
| 08:38.59 | Delmar | jakepdev, place a call to your voicemail extension.. then on the * console do .. sip show channels |
| 08:38.59 | Juggie | regardless, did you try echowhenbridged? |
| 08:39.12 | jakepdev | ok |
| 08:39.25 | Delmar | echotraining=800 |
| 08:39.25 | Delmar | echocancel=yes |
| 08:39.25 | Delmar | echocancelwhenbridged=yes |
| 08:39.33 | jakepdev | 2 active channels |
| 08:39.34 | *** part/#asterisk schurig (~schurig@p54B296F1.dip0.t-ipconnect.de) |
| 08:39.36 | amer | I want to monitor Asterisk via snmp. Is there anyway I can do this? |
| 08:39.37 | Delmar | show me |
| 08:39.43 | jakepdev | ulaw |
| 08:40.08 | Delmar | mine has something like.... |
| 08:40.11 | Juggie | Delmar, and with all that, you still have bad echo? |
| 08:40.18 | jakepdev | Peer User/ANR Call ID Seq (Tx/Rx) Format |
| 08:40.18 | jakepdev | 192.168.1.41 8000 50f94-ca2d3 00101/00100 ulaw |
| 08:40.18 | jakepdev | 192.168.1.40 8001 7BB2600E-0C 00101/00002 ulaw |
| 08:40.18 | jakepdev | 2 active SIP channel(s) |
| 08:40.19 | Delmar | 192.168.1.128 grandstrea ea37236b32b 00101/43028 ulaw |
| 08:40.34 | Juggie | ------------ please use pastebin.ca --------- |
| 08:40.49 | Delmar | juggie, yep. and i have replaced the card, the box, the power... not the location yet.. i want to try another phone line.. see if its some impedance issue. |
| 08:41.13 | Juggie | it could be just a crummy line |
| 08:41.26 | Juggie | but other then that i cant think of anything |
| 08:41.32 | Delmar | now.. what was it that i remember .. about the whole.. voicemail thingie actually being gsm recorded.... |
| 08:41.51 | Juggie | all asterisk prompts are gsm i think |
| 08:41.56 | jakepdev | .41 is the one I was using when I did sip show channels |
| 08:42.09 | Delmar | ok |
| 08:42.44 | Delmar | jakepdev, when u do .. show translation .. you do have a number .. for gsm to ulaw right? |
| 08:42.49 | GMsoft | hello all |
| 08:43.07 | jakepdev | says 2 gsm-ulaw |
| 08:43.19 | Juggie | Delmar, you can try echowhenbridged off, but i dont know other then that.... |
| 08:43.23 | jakepdev | (and 4) |
| 08:43.24 | iamx | does the machine running asterisk need a soundcard to playback the internal prompts like voicemail and festival etc. ? |
| 08:43.30 | Juggie | no |
| 08:43.39 | Juggie | thats something i should do, convert all those gsm files to ulaw |
| 08:43.44 | Juggie | so * doesnt have to transcode |
| 08:44.01 | Qwell | What do you do to convert? |
| 08:44.10 | Juggie | use sox |
| 08:44.13 | jakepdev | should I paste in show translation? |
| 08:44.13 | Juggie | read the wiki |
| 08:44.17 | Juggie | no |
| 08:44.19 | Juggie | not to here |
| 08:44.20 | Qwell | Juggie: will do |
| 08:44.22 | Juggie | use www.pastebin.ca |
| 08:44.25 | Delmar | Juggie hrm. ok. |
| 08:44.39 | Juggie | wiki has instructions for how to do all the files in one directory |
| 08:44.45 | Delmar | Juggie, its gotta work somehow.. i mean.. its ok when a call goes out.. wtf is wrong with a call coming in.... |
| 08:45.05 | Juggie | Delmar, it could be an issue not related to * |
| 08:45.07 | Delmar | Juggie, ok the only thing that happens when a call comes in ... is the line spikes for a ring... it must screw up the card |
| 08:45.14 | Juggie | did u try putting a call on that line |
| 08:45.14 | jakepdev | http://pastebin.ca/7155 |
| 08:45.20 | Juggie | with a regular pone |
| 08:45.23 | Juggie | *phone |
| 08:45.38 | nirs | Is there a way to capture a dead-lock on asterisk? and run gdb at that second ? |
| 08:45.41 | Delmar | Juggie, for sure. normal non-* stuff works as expected. |
| 08:45.52 | Juggie | nirs, try #asterisk-dev |
| 08:46.16 | jakepdev | tnx Juggie - didn't know about that |
| 08:46.27 | Delmar | jakepdev, what is the SIP client you are using to connect to * and test your voicemail? |
| 08:46.32 | Juggie | jakepdev, everyone prefers u use that rather then spam the channel |
| 08:46.53 | jakepdev | two different SIP clients - SJPhone, and DTA310 |
| 08:47.06 | Delmar | software ... |
| 08:47.11 | SexyKen | Anyone know what the hell the account_code is? |
| 08:47.12 | Delmar | just thinking out of the box... |
| 08:47.21 | Delmar | have u got any software firewalls running at all? |
| 08:47.32 | jakepdev | nope - I disabled all firewall stuff |
| 08:47.35 | Juggie | Delmar, have u tried to receive a call on that line with a regular phone? |
| 08:47.39 | Delmar | like.. on the box running the sip software? |
| 08:48.02 | Delmar | juggie, yep. normal phone on the line and it all works as expected. |
| 08:48.37 | Delmar | juggie, the echo and issues are to do with incomming calls on these X100P cards. its either an * problem .. or its the card screwing up when a call comes in. |
| 08:48.54 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 08:48.56 | Juggie | Delmar, not sure then, if you try here during north amercain daytime hours there will be more knowledgable people around |
| 08:49.11 | Delmar | what sux is.... the way this whole VoIP world works.. sure.. an outgoing call via the FXO works but.. how often am I gonna be doing that? id rather have it screw up the other way around.. haha |
| 08:49.29 | Delmar | yeh i might jump on tomorrow morning. |
| 08:49.45 | jakepdev | (3 hours more for me) :) |
| 08:49.47 | Delmar | dunno what yur time is there but its almost 10pm here |
| 08:49.48 | Juggie | everyone is at a conference though so u never know ;) |
| 08:49.57 | Juggie | i'm in the east and its almost 4am |
| 08:50.08 | jakepdev | yep - I'm also there |
| 08:50.17 | jakepdev | (near Phila) |
| 08:50.22 | Delmar | jakepdev, ok so all firewall stuff is gone... |
| 08:50.31 | jakepdev | yep - no firewall |
| 08:50.38 | Juggie | if u showed up like 1-2est there should be some more people around.... so that is 10hours from now |
| 08:50.49 | Juggie | get up early ;) 7-8am |
| 08:50.57 | jakepdev | I can run a network trace to see traffic |
| 08:51.05 | Delmar | jakepdev, your * is on a linux dist under vmware... |
| 08:51.13 | jakepdev | that is correct |
| 08:51.13 | Delmar | effectivly its a real system on the same subnet... |
| 08:51.18 | jakepdev | yep |
| 08:51.19 | Delmar | so ignore all that crap too.... |
| 08:51.28 | Delmar | so its back to * I guess... |
| 08:51.31 | Juggie | jakepdev, i am going to bed, but try a sip debug |
| 08:51.39 | Juggie | to see where your packets are being sent |
| 08:51.41 | jakepdev | ok - tnx Juggie |
| 08:51.49 | Delmar | yeh thats a good idea. |
| 08:51.51 | Delmar | turn that on. |
| 08:52.00 | Delmar | then place the call to VM and see. |
| 08:52.01 | SexyKen | Hey guys, how do you set the account code in asterisk for cdr? |
| 08:52.10 | Delmar | ooo thats fun stuff... |
| 08:52.16 | amer | in sip.conf |
| 08:52.21 | Juggie | SexyKen, its a dialplan function or in sip.conf |
| 08:52.25 | jakepdev | i can do one better - - I can do a netowrk trace between the VM and main PC |
| 08:52.35 | Juggie | jakepdev, try * first |
| 08:52.38 | Delmar | under sip.conf .. under the specific section ... use accountcode=blah |
| 08:52.39 | jakepdev | ok |
| 08:52.40 | Juggie | it puts it in a more readable format |
| 08:52.42 | Juggie | and decodes it |
| 08:53.17 | Delmar | and what happens is.. all the cdr stuff for that sip user... or users if u add it to more than one.. is bundled into files ...in /var/log/asterisk/cdr-csv |
| 08:53.57 | *** join/#asterisk shantanoo (~shantanoo@shantanoo.user) |
| 08:54.28 | jakepdev | http://pastebin.ca/7156 |
| 08:54.54 | *** part/#asterisk fjoe (~fjoe@samodelkin.net) |
| 08:56.08 | Luke-Jr | wow... I've actually thought of a good use for dialup over VoIP o.o |
| 08:57.26 | SexyKen | cd /usr/local/apache |
| 08:57.28 | SexyKen | oops |
| 08:57.52 | jakepdev | wrong window error |
| 08:57.52 | shepherd | if you can get dialup to work with voip |
| 08:58.03 | shepherd | ni ni |
| 08:58.42 | Luke-Jr | shepherd: Plug an ATA into a modem? |
| 08:59.24 | *** join/#asterisk claint (~claint@195.174.26.218) |
| 08:59.38 | Delmar | jakepdev, you have canreinvite=yes in the sip.conf section for the sip clients 8000/8001? |
| 08:59.56 | jakepdev | let me see |
| 09:00.20 | jakepdev | nope |
| 09:00.24 | Delmar | :P |
| 09:00.30 | jakepdev | is that it? |
| 09:00.36 | Delmar | give that a try please. |
| 09:00.45 | Delmar | was just lookin at your debug output. |
| 09:01.03 | Delmar | was just seeing.. invitations.... |
| 09:01.10 | Delmar | so thats one thing to do... |
| 09:01.19 | Delmar | whenever u are non-Nat.. you can do canreinvite=yes |
| 09:01.32 | Delmar | but if you had say.. nat=yes, you should have canreinvite=no |
| 09:01.39 | jakepdev | ok |
| 09:01.54 | Delmar | did u do a pastebin of one of your sip.conf user sections ? |
| 09:02.04 | jakepdev | nope -i'll put that up |
| 09:02.05 | Delmar | if not, throw me one (remove passwords). |
| 09:02.17 | jakepdev | ok |
| 09:02.34 | *** join/#asterisk mamcinty (~mamcinty@adsl-068-209-174-113.sip.int.bellsouth.net) |
| 09:03.00 | *** join/#asterisk RoyKa (~roy@83.80-203-29.nextgentel.com) |
| 09:03.08 | jakepdev | http://pastebin.ca/7157 |
| 09:03.19 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 09:03.44 | jakepdev | (I just added canreinvite) |
| 09:06.25 | Delmar | ok what does that do |
| 09:06.31 | jakepdev | no luck |
| 09:06.34 | Delmar | dont forget to do a reload |
| 09:06.51 | jakepdev | ok |
| 09:06.54 | Delmar | and i mean.. sometimes i like to drop asterisk out the arse, then fire it up. not just "reload" at the console. |
| 09:07.06 | Delmar | especially when im dicking around with zapata and stuff. |
| 09:07.24 | jakepdev | i'll see a REINVITE in the debug now? |
| 09:07.38 | Delmar | maybe. |
| 09:07.58 | Delmar | i cant see anything else much going on/wrong with your sip.conf so... im gonna move on from there..... |
| 09:08.00 | jakepdev | you're saying to stop asterisk |
| 09:08.01 | jakepdev | ? |
| 09:08.05 | Juggie | Delmar, see priv message. |
| 09:08.09 | jakepdev | then start |
| 09:08.24 | Delmar | yup. restart it. |
| 09:08.29 | Delmar | sorry Juggie, lookin |
| 09:08.54 | claint | how fast a system do i need for asterisk? for testing at home that is. |
| 09:09.02 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 09:09.02 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 09:09.06 | SexyKen | 1. 2005-03-10 03:11:53 IAX2/te... 6507846652 6507842252 Playback customers/1.closed closed ANSWERED 00:20 |
| 09:09.06 | SexyKen | 2. 2005-03-10 03:11:37 IAX2/vo... Dial SIP/1.201|20|tr s ANSWERED 00:08 |
| 09:09.12 | Juggie | claint, anything p3 should be decent |
| 09:09.40 | SexyKen | THose are two calls ... according to the database ... but they're actually the same actual call...is there anyway to store this in the db so it knows that the two calls are the same ? |
| 09:09.44 | claint | what about a p2/266? will just try to connect and get it working... |
| 09:10.02 | Juggie | sure |
| 09:10.04 | Juggie | go nuts |
| 09:10.08 | Juggie | that will work |
| 09:10.11 | jakepdev | I did a stop and start |
| 09:10.24 | mamcinty | Does anyone know where I might aquire a data source that would let me determine the City, State from NPA-NXX? I have a project in mind. |
| 09:10.25 | jakepdev | i'll run the debug on it |
| 09:10.40 | Delmar | ok |
| 09:11.09 | Delmar | jakepdev, in voicemail.conf u have format=wav49|gsm|wav etc ? |
| 09:11.21 | Delmar | nah thats just for writing... thats not it. |
| 09:11.30 | Eight | grrr... voip-info is down? |
| 09:11.42 | Delmar | no, your squid cache died :P |
| 09:11.49 | jakepdev | http://pastebin.ca/7158 |
| 09:12.00 | amer | I want to monitor Asterisk via snmp. Is there anyway I can do this? |
| 09:12.22 | Delmar | Eight, no go here either. must be dude. |
| 09:12.29 | Eight | Delmar: thanks for the confirmation. |
| 09:12.52 | SexyKen | 2. 2005-03-10 03:11:37 IAX2/vo... Dial SIP/1.201|20|tr s ANSWERED 00:08 |
| 09:12.54 | SexyKen | THose are two calls ... according to the database ... but they're actually the same actual call...is there anyway to store this in the db so it knows that the two calls are the same ? |
| 09:12.56 | jakepdev | amer - http://puck.nether.net/npa-nxx/ |
| 09:13.08 | jakepdev | (just googled on what you wrote) |
| 09:13.36 | Newbie___ | hmmm voip-info.org web site is down |
| 09:14.07 | Juggie | yes, yes it is |
| 09:14.20 | Delmar | jakepdev, u have Xlite? i dont know anything about those other apps... install Xlite. take u 5mins. |
| 09:14.33 | jakepdev | i think i have it here already... |
| 09:14.38 | SexyKen | Damn -- I just want some help |
| 09:15.19 | Eight | so, since the wiki is down... |
| 09:15.19 | mamcinty | thank you! |
| 09:16.01 | Eight | In sip.conf: Username is what's in [], and authorization user is what's after username=, right? |
| 09:16.23 | Eight | going with x-lite terminology here. |
| 09:17.50 | djorange | question trival: what phone code do you press for directory |
| 09:18.59 | *** join/#asterisk djin (~djin@62.58.40.196) |
| 09:20.11 | SexyKen | What's the userfield in the cdr for |
| 09:20.31 | amer | jakepdev: thanks, I have searched on google. This link is not related to * |
| 09:20.42 | RoyK | ~lart Zeeek |
| 09:21.05 | jakepdev | amer - it was meant for someone else - sorry bout that |
| 09:21.08 | Delmar | jakepdev, so install that, and make sure all the codecs on the "lcd" screen are all lit up, and give it a go. |
| 09:21.22 | jakepdev | how do i get back into config on this thing? |
| 09:21.30 | Zeeek | I never met a client I didn't like.... to sue |
| 09:21.37 | Delmar | heh. the icon to the right of the "clear" on the panel... |
| 09:21.49 | Delmar | then system settings |
| 09:22.01 | Delmar | and sip proxy |
| 09:22.03 | Delmar | etc. |
| 09:22.07 | Zeeek | why is alaw better than ulaw? |
| 09:22.08 | Delmar | got it? |
| 09:22.10 | Zeeek | anyone? |
| 09:22.19 | jakepdev | i'm so embarrased |
| 09:22.23 | jakepdev | i don't see it |
| 09:22.28 | jakepdev | i see CLR |
| 09:22.39 | Zeeek | these things happen to all men - don't be |
| 09:22.45 | Delmar | ok.. ther is an icon/button thingie to the right of "CLEAR" on the main phone panel |
| 09:22.49 | iamx | Voicemail and '1234' prompts don't play. |
| 09:22.59 | jakepdev | i have a slide to the right of CLR |
| 09:23.00 | Delmar | looks like a notepad. |
| 09:23.06 | iamx | there's a bug thingy on the dev page of asterisk@home |
| 09:23.14 | Delmar | umm |
| 09:23.16 | Delmar | what version u have? |
| 09:23.17 | jakepdev | i'll find it |
| 09:23.18 | shantanoo | can asterisk be used on intranet without any telephony card? |
| 09:23.26 | jakepdev | just pulled it from the site |
| 09:23.33 | Delmar | ah ok |
| 09:23.34 | Eight | shantanoo: with SIP phones, sure. |
| 09:23.35 | iamx | delmar, jakepdev http://sourceforge.net/forum/forum.php?thread_id=1244196&forum_id=420324 |
| 09:23.40 | jakepdev | oh - it threw a wied skin on here |
| 09:24.12 | shantanoo | Eight: no phones involved. only computers with sound card and mikes |
| 09:24.26 | *** join/#asterisk naif (~User@host250-27.pool62110.interbusiness.it) |
| 09:24.29 | Eight | shantanoo: That's what I'm doing right now, works great. |
| 09:24.29 | jakepdev | did you try this? |
| 09:24.35 | naif | . |
| 09:24.36 | shantanoo | i was wondering how can i ring them? |
| 09:24.48 | Eight | shantanoo: dial their extension from another computer. |
| 09:25.00 | iamx | it seems to be a general problem on *@home |
| 09:25.14 | jakepdev | does yours work now? |
| 09:25.23 | shantanoo | and the extensions are configurable? |
| 09:25.26 | Delmar | thats a new one on me. |
| 09:25.33 | iamx | no, didn't find a solution yet and i'm searching since a week |
| 09:25.42 | Delmar | u guys arent using * from stable cvs? |
| 09:25.52 | jakepdev | i pulled *@home iso |
| 09:25.59 | Delmar | oh ok |
| 09:26.00 | iamx | <- too |
| 09:26.01 | jakepdev | should I pull stable |
| 09:26.03 | Delmar | i like cvs. |
| 09:26.10 | jakepdev | ? |
| 09:26.10 | Eight | ~wiki is back up |
| 09:26.17 | Delmar | oo thanks Eight |
| 09:26.19 | Eight | err.. |
| 09:26.20 | Eight | ~wiki |
| 09:26.29 | Eight | ok, I don't know how to use jbot =p |
| 09:26.33 | Eight | jbot wiki? |
| 09:26.34 | jbot | [wiki] http://www.voip-info.org |
| 09:26.37 | Eight | tada! |
| 09:26.53 | jakepdev | i'm going to reinstall x-lite (no config button on this skin) |
| 09:27.05 | Zeeek | ~wiki |
| 09:27.21 | Zeeek | ~got milk |
| 09:27.22 | jbot | ACTION chugs down a carton |
| 09:27.29 | Zeeek | ~jbot wiki |
| 09:27.44 | Zeeek | one out of three isn't bad |
| 09:28.39 | jakepdev | now i got the right skin |
| 09:28.52 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:29.29 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 09:29.46 | SexyKen | Hey guys, what is the 'userfield' for in mysql cdr db? |
| 09:29.52 | *** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg) |
| 09:30.19 | claint | how do i find out if my soundcard is full-duplex? |
| 09:30.25 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:30.57 | Eight | claint: was it purchased in the last decade? |
| 09:31.08 | claint | Eight: aye ;-) |
| 09:31.16 | Eight | It'd be a good trick to find one that ISN'T anymore. |
| 09:31.59 | claint | does OSS do full-duplex? or do i need Alsa strictly. [would still use alsa with oss simulation i guess] |
| 09:33.12 | jakepdev | i got x-lite installed - working on config |
| 09:34.35 | SexyKen | Does uniqueid stay the same for the same call? For instance, in an ivr system...a user calls an incoming line. They're connected to the ivr. They press 1 for sales. Asterisk connects them to an agent. |
| 09:34.47 | SexyKen | In the cdr, how am I supposed to know that those two calls are connected to the same user? |
| 09:36.22 | *** join/#asterisk wasim (~wasim@203.81.217.160) |
| 09:36.47 | Delvar | SexyKen: there is no easy way, i use userfeild to tag it with my own id, that also realtes to a seperat table with more data |
| 09:38.19 | jakepdev | i'm going to update from CVS |
| 09:38.25 | SexyKen | Delvar -> How'd you do it? |
| 09:38.34 | jakepdev | maybe this was a bug |
| 09:39.01 | Delvar | it was a while ago, i had to modify the headers before compile |
| 09:39.33 | shantanoo | Eight: which packages are required from asterisk? |
| 09:39.46 | iamx | humm jakepdev how will you do that ? |
| 09:40.07 | SexyKen | Delvar -> Anyway you can help me or no? |
| 09:40.18 | shantanoo | there are around 9 in debian sarge |
| 09:40.21 | jakepdev | i remember reading on one of the sites how to do that.. I'm going to try and find it, I'll paste the link back |
| 09:40.50 | iamx | ok thanks jakepdev |
| 09:41.21 | jakepdev | http://www.voip-info.org/wiki-Asterisk+Download |
| 09:41.27 | jakepdev | near the bottom |
| 09:42.01 | iamx | k tnx |
| 09:42.30 | jakepdev | delmar - thank you for the time you put in to this thing. If you're still around, i'll let you know if the upgrade fixed it |
| 09:43.40 | *** part/#asterisk Luke-Jr (~luke-jr@207.192.219.246) |
| 09:43.51 | jakepdev | iamx - i would copy your 2 conf files if you haven't already (just in case) |
| 09:44.17 | iamx | yup i've done it thanks |
| 09:44.22 | jakepdev | np |
| 09:45.44 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 09:46.08 | SexyKen | Delvar -> Anyway you can help me or no? |
| 09:50.56 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 09:56.31 | *** join/#asterisk afe ([puTPUFQYz@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 09:57.46 | iamx | jakepdev i finished upgrading but i still have the same problem. hummm |
| 10:00.01 | ikey | does any know about sphinx? |
| 10:00.21 | ikey | or any one implimented sphinx with asterisk |
| 10:04.15 | iamx | it isn't a voicemail specific problem, it's every playback function coming from *, festival too, the script is called up correctly and runs, but no sound, when calling 1234 where it should read back my phone number is just her small chunks of sound, 500 ms or so |
| 10:05.56 | jakepdev | iamx - no luck yet either - still trying |
| 10:07.51 | Eight | Other than editing the source... is there any way to alter the voice prompts in the default voice mail system? |
| 10:08.12 | iamx | i've mailed one of the users on the asterisk@home forum asking him if he has found a solution |
| 10:08.14 | Eight | it just says vm-password to start with, I want it to do pls-enter-vm-password =/ |
| 10:11.05 | Eight | I suppose I could do the auth myself, but that seems silly =/ |
| 10:11.57 | *** join/#asterisk bmilanov (~bm@CPE-61-9-217-156.qld.bigpond.net.au) |
| 10:12.26 | bmilanov | hello I have a question about asterisk hardware |
| 10:12.42 | bmilanov | pulse dialing |
| 10:13.08 | bmilanov | is there anyone who have used pulse dialing with asterisk? |
| 10:13.33 | jakepdev | pulse dialing - that's pretty funny |
| 10:13.38 | bmilanov | lol |
| 10:14.00 | bmilanov | well it is still in use in some forgoten places qround the world |
| 10:14.28 | bmilanov | I am searchin for a hardware which can do the pulse dialing |
| 10:14.49 | bmilanov | a regular FXO doesn't support pulse dialing |
| 10:14.57 | bmilanov | as long as I know - never tried |
| 10:15.28 | bmilanov | can you provide me with a link? |
| 10:15.34 | bmilanov | where could I find it? |
| 10:16.07 | bmilanov | oh I though this is a brand name :) |
| 10:16.09 | shantanoo | bmilanov: i am newbie with asterisk. i can courier you the device ;) |
| 10:16.46 | bmilanov | have you used pulse dialing? |
| 10:17.38 | shantanoo | bmilanov: yes. over here in India, pulse as well as tone dialling is available. |
| 10:18.04 | shantanoo | but as I said, I am complete newbie regarding asterisk |
| 10:18.04 | bmilanov | can you tell me what kind of hardware are you using? |
| 10:18.23 | bmilanov | or you have never used pulse dialing with asterisk :) |
| 10:18.42 | iamx | here in luxembourg pulse dialing works too |
| 10:19.21 | bmilanov | anyone who has connected asterisk to a pulse PSTN? |
| 10:19.27 | elric | shantanoo :) |
| 10:19.37 | shantanoo | mooooooooooooooooooooooooooooooooooooooooooooooooooooooooooooo elric :) |
| 10:19.44 | bmilanov | so only shantano is the hero |
| 10:19.46 | shantanoo | shit. |
| 10:19.47 | bmilanov | :) |
| 10:19.47 | elric | how are you mate? |
| 10:20.30 | shantanoo | heheh |
| 10:20.30 | shantanoo | bmilanov: from which country you are? |
| 10:21.13 | bmilanov | many :) Originaly from Bulgaria but now I am in Australia |
| 10:21.16 | amer | I want to monitor Asterisk via snmp. Is there anyway I can do this? |
| 10:21.39 | elric | bmilanov, my house mate is of bulgarial origin |
| 10:21.44 | elric | and in sydney |
| 10:21.46 | elric | are you him? |
| 10:21.47 | elric | heh |
| 10:21.55 | elric | bulgarian |
| 10:22.01 | bmilanov | I doubt I am him :) |
| 10:22.06 | bmilanov | I am in Brisbane |
| 10:23.01 | elric | ah ok |
| 10:23.18 | elric | have you had any problems with Onramp |
| 10:23.29 | elric | we are about to deploy a lot of lines for a client. |
| 10:23.31 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 10:23.36 | bmilanov | what is Onramp |
| 10:23.46 | Damin | Hey...... |
| 10:23.50 | Damin | Whhat up? |
| 10:23.52 | *** join/#asterisk Jistah (~Jistah@80.72.89.162) |
| 10:24.00 | elric | Telstra ISDN |
| 10:24.57 | *** join/#asterisk vagwin (~vagwin@mk-ns500-1.uk.tiscali.com) |
| 10:32.05 | ikey | any one worked on sphinx with asterisk |
| 10:32.06 | ikey | ? |
| 10:34.10 | Damin | I am eating Jack In the Box from the corner of Santa Clara anf S4th ST in San Jose. If I die, you willl act least know who too look at.... |
| 10:36.11 | Zeeek | where's the beef ? |
| 10:43.10 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 10:43.32 | shadebob | hi, someone have a complete zapata.conf for a TE110P card? |
| 10:46.58 | antifuchs | hm. are sipfriends that are in a realtime database not treated via qualify? |
| 10:47.56 | antifuchs | if I add the user to sip.conf, I get "Status : OK (35 ms)", but if I add the same user to the realtime table, I get "Status: UNKNOWN" after the sip phone registers. |
| 10:52.53 | Darwin35 | ok got spandsp to compile |
| 10:52.57 | Darwin35 | yes |
| 10:53.44 | naif | anyone integrated asterisk + hylafax in a decent way? |
| 10:55.26 | RoyK | I beleive so |
| 10:55.28 | RoyK | see the wiki |
| 10:55.33 | Zeeek | no |
| 10:58.31 | RoyK | no? |
| 10:58.39 | RoyK | ~lart Zeeek |
| 11:01.13 | Zeeek | not |
| 11:07.20 | Zeeek | <PROTECTED> |
| 11:07.23 | Zeeek | heh heh |
| 11:08.31 | tzafrir | chrooted or anything? |
| 11:08.53 | Zeeek | " /dev/dull" |
| 11:09.16 | tzafrir | BTW: I believe bash has sepcial handling for redirection to /dev/null even if /dev/null does not exist, but I'm not sure |
| 11:09.28 | Zeeek | it's a type |
| 11:09.33 | Zeeek | typO |
| 11:09.36 | Zeeek | dull |
| 11:09.45 | Darwin35 | ok now on to festival |
| 11:09.52 | tzafrir | Ah, you mean the extra space before the "/"? ;-) |
| 11:09.53 | Darwin35 | this will be fun |
| 11:09.57 | Zeeek | sounds so festive and fun |
| 11:10.02 | Zeeek | but I hear it isn't |
| 11:10.23 | Darwin35 | it is when your getting working on FBSD and * |
| 11:10.32 | Darwin35 | I have got alot working |
| 11:10.33 | Zeeek | aha great |
| 11:10.45 | Darwin35 | spandsp works |
| 11:10.45 | Zeeek | that's good - a lot of people are interested including me |
| 11:11.22 | Darwin35 | well thats todat project |
| 11:11.39 | Darwin35 | so it will be fun |
| 11:11.42 | tzafrir | Darwin35, what version of spandsp? |
| 11:11.59 | Darwin35 | pre10 |
| 11:12.57 | Darwin35 | bbiab time to take the roomie to work |
| 11:15.32 | *** join/#asterisk mbranca (~matteo@80.152.73.227) |
| 11:15.33 | Zeeek | why doesn't this work? |
| 11:15.36 | Zeeek | tftp dgram udp wait nobody /usr/sbin/tcpd in.tftpd |
| 11:15.47 | Zeeek | in inetd.conf ? |
| 11:17.44 | *** join/#asterisk szlwzl (simon@81.144.188.147) |
| 11:18.55 | szlwzl | have a question re: meet me - is it possible for external callers to be greeted with a prompt to enter a passcode which then automatically joins them to a conference? |
| 11:24.28 | PatrickDK | yes |
| 11:24.33 | PatrickDK | many ways to do that |
| 11:25.37 | szlwzl | excellent stuff - I presumed there would be. |
| 11:28.01 | szlwzl | any howto links from the top of your head? |
| 11:28.46 | *** join/#asterisk sunil (~sunil@202.54.37.183) |
| 11:29.03 | shantanoo | see it already proved it |
| 11:29.22 | sunil | hello can any one help me in installing astGUICLIENT |
| 11:30.19 | Zeeek | szlwzl start here :http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20meetme.conf |
| 11:31.31 | Damin | Cy upieast |
| 11:31.55 | Zeeek | cnergeda? |
| 11:32.13 | Zeeek | Kartengs hareng |
| 11:34.00 | szlwzl | Zeeek: thank you |
| 11:34.07 | Zeeek | np |
| 11:35.55 | szlwzl | the meeting creation is no probs really - but how would a external caller be prompted to join it? |
| 11:36.14 | Zeeek | you have to send them in |
| 11:36.22 | *** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM] |
| 11:36.22 | szlwzl | so someone internal has to do that |
| 11:36.24 | szlwzl | ? |
| 11:36.45 | Zeeek | no you could present a menu that said "enter blah for conf" |
| 11:37.05 | Zeeek | and to do that yopu might look here |
| 11:37.08 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
| 11:37.08 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 11:38.14 | Zeeek | there is a classic example of a IVR menu there |
| 11:38.14 | szlwzl | brilliant - cheers :) |
| 11:38.34 | Zeeek | now you know two places to look |
| 11:38.39 | modulus_ | zeeek, what is ivr? |
| 11:38.43 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 11:38.56 | szlwzl | Zeeek: will go and research further :) |
| 11:39.41 | Zeeek | Interactive Voice Relatives ? |
| 11:40.17 | Zeeek | <PROTECTED> |
| 11:40.42 | Zeeek | another example of a dialplan for IVR |
| 11:42.21 | *** join/#asterisk pratik (~root@202.149.48.204) |
| 11:44.34 | pratik | pratik says hi to everyone |
| 11:44.54 | Eight | Relatives? Response? |
| 11:45.12 | Zeeek | everyone is busy saying hi to other people wanting to say hi |
| 11:45.14 | pratik | i am facing problems in making calls to FWD |
| 11:45.21 | Zeeek | still? |
| 11:45.50 | pratik | ya now i am trying through the iax |
| 11:45.50 | Zeeek | you are right to face them, rather than just being in denial |
| 11:46.01 | Zeeek | what's wrong? |
| 11:46.13 | pratik | i configured only the fwd part with iax |
| 11:46.22 | Zeeek | yes? |
| 11:46.37 | pratik | no when i make calls to fwd an engage tone comes |
| 11:46.52 | Zeeek | and you copied the info from here? http://www.freeworlddialup.com/advanced/iax |
| 11:47.00 | pratik | i removed all the fwd details from the sipo.conf |
| 11:47.10 | pratik | sorry sip.conf |
| 11:47.13 | Zeeek | and checked iax in the web page config |
| 11:47.52 | pratik | i copiued it from http;??www.freeworlddialup.com/content/view/full/1501/ |
| 11:48.05 | pratik | i copiued it from http://www.freeworlddialup.com/content/view/full/1501/ |
| 11:48.53 | Zeeek | you try to call 612 ? |
| 11:49.01 | pratik | ya i tried |
| 11:49.15 | pratik | but again the same engage tone came |
| 11:49.27 | Zeeek | and the console message says what? |
| 11:49.40 | pratik | well should i paste it |
| 11:49.46 | Zeeek | ~pastebin |
| 11:49.47 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 11:50.02 | pratik | ya just a minute |
| 11:50.03 | Zeeek | paste your configs |
| 11:50.28 | Zeeek | from iax.conf - just the fwd part |
| 11:51.34 | *** join/#asterisk fitzel (~flint@p3EE3978E.dip0.t-ipconnect.de) |
| 11:52.14 | Zeeek | exit |
| 11:52.18 | Zeeek | not |
| 11:52.26 | pratik | ya i have pasted the iax on pastebin.ca/7171 |
| 11:52.26 | *** join/#asterisk Umaro (~umaro@209.140.74.64) |
| 11:52.34 | pratik | and the console on 7172 |
| 11:52.55 | Umaro | Hey guys, getting this odd "Unknown RTP codec 72 recieved" message.. any hints? I've searched google and the mailing lists, no luck there |
| 11:52.57 | pratik | in iax i have only the fwd |
| 11:53.40 | Zeeek | I would say that the Dial command is missin a closing parenthesis |
| 11:53.55 | Zeeek | <PROTECTED> |
| 11:55.23 | Zeeek | Umaro how are your codecs looking? |
| 11:56.06 | Zeeek | paratik there is a line that wraps in default - fix that and it should be ok |
| 11:56.13 | Zeeek | pratik |
| 11:56.16 | Newbie___ | Zeeek: i am learning how to connect 2 * together, any idea where i can get examples except voip-info.org i tried that |
| 11:56.16 | Zeeek | pranav |
| 11:56.23 | pratik | yes tell me |
| 11:56.42 | Zeeek | under default - FIX that line that wraps to a new line |
| 11:56.45 | pratik | the closing parenthesis is continued on the next line |
| 11:56.54 | Zeeek | no good - must be one line |
| 11:57.10 | Zeeek | go through all your files and make sure there are only SINGLE LINES |
| 11:57.11 | pratik | ok fine i'll do it |
| 11:57.19 | pratik | ok |
| 11:57.36 | Zeeek | Newbie_ there should be plenty about IAX trunking |
| 11:59.25 | Newbie___ | ok been trying to learn that all day |
| 11:59.37 | Zeeek | the name? ya not obvious |
| 11:59.50 | Zeeek | also you need hardware timer to do it |
| 12:00.26 | Newbie___ | Zeeek: u mean i need hardware timer? |
| 12:00.34 | *** join/#asterisk jmav (~jmav@201.243.76.158) |
| 12:00.46 | Zeeek | for iax trunking yes |
| 12:01.01 | shadebob | Hi, I have a "Ouch ... error while writing audio data: : Broken pipe" message when I launch asterisk server :s |
| 12:01.03 | Zeeek | here's some interesting stuff |
| 12:01.05 | Zeeek | <PROTECTED> |
| 12:01.18 | Newbie___ | damn, and i never even heard of hardware timer |
| 12:01.35 | Zeeek | it comes with the Digium hardware |
| 12:02.29 | jmav | Hi I am new in this thing of asterisk... I would like to know if its some way that a line from pstn could be use for dialout for tell the asterisk to not be used for incoming calls |
| 12:02.41 | Newbie___ | confused as usual, anyway will try to look into IAX trunk first |
| 12:02.56 | pratik | ya i have changed it |
| 12:03.08 | *** join/#asterisk Inv_arp (junya@adsl-8-232-168.mia.bellsouth.net) |
| 12:03.14 | jmav | hello everybody |
| 12:03.22 | Poincare | jmav: send it to a context where it won't pickup the line? |
| 12:03.33 | pratik | the whole of the 2nd which was splitting into 2 lines i made it into single line |
| 12:03.48 | jmav | thx |
| 12:04.42 | Zeeek | jmav : |
| 12:04.44 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
| 12:04.44 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 12:05.08 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 12:05.20 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 12:05.28 | pratik | i have used type=user, should i change it to type=friend |
| 12:05.51 | jmav | thx zeeek... and thx poincare |
| 12:06.01 | Zeeek | peer pratik, peer |
| 12:06.32 | Zeeek | or user |
| 12:06.44 | Zeeek | I have both in fact |
| 12:07.23 | pratik | ok i changed it but no effect |
| 12:07.26 | Zeeek | to dial out I used peer |
| 12:07.48 | pratik | i can only dial out, incomiong calls are not allowed in INDIA |
| 12:09.22 | Zeeek | http://pastebin.ca/7173 |
| 12:09.28 | Zeeek | this is what I have |
| 12:09.50 | pratik | ok i'll check it out |
| 12:10.42 | pratik | that is what all u have in iax.conf for fwd |
| 12:10.56 | Zeeek | your Dial line is still wrong I'd say |
| 12:11.16 | pratik | i copied it from the site, exactly the same thing |
| 12:11.30 | pratik | copy, paste |
| 12:12.22 | pratik | if the dial line is wrong then instead of that what should i write |
| 12:12.33 | Zeeek | try this, filling in the obvious data |
| 12:12.35 | Zeeek | exten => _393.,2,Dial(IAX2/YOURNUMBER:YOURPASS@the_name_in_iax.conf/NUMBER_TO_CALL,45,r) |
| 12:13.13 | Zeeek | the name in iax.conf is [fwdout] for example |
| 12:13.34 | Zeeek | if you want to debug, put 612 instead of NUMBER_TO_CALL |
| 12:13.45 | pratik | ok let me try |
| 12:13.55 | Zeeek | put your number and password in directly - you can add variable when it works |
| 12:17.14 | pratik | exten=>_393.,2,Dial(IAX2/${607191}:${password}@iaxfwd/${612},45,r) |
| 12:17.21 | pratik | is this ok |
| 12:17.27 | Zeeek | not even close |
| 12:17.41 | Zeeek | ${} this means variable name |
| 12:17.47 | pratik | in dint get u |
| 12:17.59 | pratik | ok i'll remove them |
| 12:18.03 | Zeeek | remove all the ${} |
| 12:18.08 | riksta | *sigh* :) |
| 12:18.29 | Zeeek | don't post it just fill in your real number and password, extensions reload and try it |
| 12:18.36 | Eight | Hrmm... that's weird. I'm using Playtones(dial), but I only hear the tone if I'm breaking the squelch on x-lite. |
| 12:19.00 | Zeeek | try setting transmit silence to YES - it needs to be there anyway |
| 12:19.09 | Eight | Zeeek: ya, I just changed that... |
| 12:19.10 | Inv_arp | ahh extension reload never used that |
| 12:19.26 | Eight | Zeeek: I don't see why it should be a problem, though. |
| 12:19.29 | Zeeek | it should be called e r |
| 12:19.46 | Inv_arp | heh |
| 12:19.53 | Eight | Zeeek: ya, that fixed it. |
| 12:19.54 | Zeeek | it is a problem using X-Lite |
| 12:20.02 | Eight | I'd rather fix it on the Asterisk end, though. |
| 12:20.11 | Eight | one less configuration option to tell users about. |
| 12:20.12 | Zeeek | it's an X-Lite problem |
| 12:20.17 | pratik | exten=>_393.,2,Dial(IAX2/607191:password@iaxfwd/612,45,r) |
| 12:20.31 | Zeeek | don't show us, just bloody try the sucker! |
| 12:20.32 | Eight | alright, I'd rather *workaround it* on the Asterisk side =p |
| 12:20.49 | Zeeek | Eight i dont think you can |
| 12:20.50 | pratik | i tried it still i am getting the error |
| 12:21.01 | Inv_arp | Eight: sure... submit a patch |
| 12:21.22 | Eight | Inv_arp: heh, you don't want *my* C code =) |
| 12:21.29 | Eight | Python code, sure... not C code. |
| 12:21.38 | Zeeek | pratik, I'm afraid you need help, but maybe someone in person |
| 12:21.39 | pratik | see this is the error i get in the console screen |
| 12:21.45 | Inv_arp | Eight: heh |
| 12:21.46 | pratik | : app_dial.c:884 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) |
| 12:21.47 | Zeeek | is it new? |
| 12:21.52 | Zeeek | aha |
| 12:22.16 | Alexis | on witch OS * has better performances ? |
| 12:22.38 | Alexis | a Linux or a BSD ? |
| 12:22.42 | pratik | someone in person:what does this mean |
| 12:23.57 | Eight | Anyone happen to know of a place that'll reliably ship something overnight, same day it's ordered? |
| 12:24.06 | Eight | I mean, that sells TDM400P cards =) |
| 12:24.20 | pratik | anyone else i can help , i have pasted all my configurations in pastebin.ca/7171 |
| 12:24.40 | Zeeek | hmmm that a tough one Eight |
| 12:24.53 | Alexis | Is it easier to use Asterisk on a Linux or on FreeBSD ? |
| 12:24.56 | Zeeek | you could ask digium they may stock them |
| 12:24.59 | Inv_arp | Alexis: linux |
| 12:25.01 | Zeeek | Alexis linux |
| 12:25.07 | Alexis | withc one ? |
| 12:25.16 | Alexis | what distrib ? |
| 12:25.20 | Eight | Alexis: Fedora didn't give me any problems. |
| 12:25.21 | Zeeek | Alexis |
| 12:25.21 | Inv_arp | Alexis: any |
| 12:25.22 | Newbie___ | Zeeek: any idea what happen ? request '23701169@trunk' does not exist |
| 12:25.28 | Alexis | ok tx |
| 12:25.34 | Eight | Alexis: But then, neither did Mac OS X =p |
| 12:25.43 | Zeeek | Newbie_ absolutelt the number doesn't exist |
| 12:25.54 | sunil | anyone can help me installing asterisk astGUICLIENT |
| 12:25.54 | Alexis | but i love so much FreeBSD.... |
| 12:26.05 | Newbie___ | hmmm |
| 12:26.11 | Inv_arp | Alexis: it does work fine under BSD |
| 12:26.14 | pratik | sunil:till where have u reached |
| 12:26.25 | Alexis | :( |
| 12:26.29 | Inv_arp | Alexis: but it had to be ported to work |
| 12:26.34 | tzafrir | Alexis, Debian. |
| 12:26.34 | Newbie___ | i should look under extensions.conf right ? |
| 12:26.42 | Alexis | ok |
| 12:27.15 | *** join/#asterisk casterman (~casterman@63.240.97-84.rev.gaoland.net) |
| 12:27.15 | jmav | still having the same problem I have 2 pstn lines but i want to use 1 for dialout and dialin and the other one just for dialout how can i do that ? |
| 12:27.15 | tzafrir | Specificly, on Rapid you'd have * up and running in 10 to 20 minutes |
| 12:27.15 | elric | It runs very easily on FreeBSD |
| 12:27.15 | Eight | Alexis: If you're used to working on BSD, then you're already accustomed to Linux software that isn't *quite* as at home under FreeBSD. Asterisk is fairly normal on that scale. |
| 12:27.15 | elric | i use it |
| 12:27.15 | Zeeek | Newbie_ yes, the message is kind of obvious - it finds no such number in that context |
| 12:27.43 | Eight | Alexis: If you're accustomed to FreeBSD, you're probably going to have more trouble with the Linux-ism than any Asterisk/BSD issues. |
| 12:27.48 | Newbie___ | but that is how i dial 23701169 and is ok |
| 12:28.12 | Inv_arp | jmav: in your extensions.conf |
| 12:28.14 | elric | I am waiting for TE405P drivers to be out for FreeBSD |
| 12:28.17 | Zeeek | apparently not. It's just a machine executing your dialplan you know :) |
| 12:28.23 | elric | then I can run it on a production system. |
| 12:28.37 | Newbie___ | grrr |
| 12:29.18 | Zeeek | jmav - if I may be so bold - you need to read about and understand dialplans |
| 12:29.23 | jmav | thx |
| 12:29.29 | Zeeek | context, extension, priority - |
| 12:29.35 | Zeeek | it will all become crystal clear |
| 12:29.40 | Newbie___ | roger that |
| 12:30.31 | Zeeek | after a year or so :) |
| 12:30.31 | Inv_arp | lol |
| 12:30.31 | jmav | ohhh great |
| 12:30.31 | Zeeek | what, don't have a year? |
| 12:30.52 | jmav | no |
| 12:30.54 | Zeeek | It should take about a week or less for it to sink in if you have no experience at all with phones |
| 12:31.01 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 12:31.03 | Zeeek | or, you can pay someone to help |
| 12:31.15 | Zeeek | but in your case, I'd recommend snuggling up to the docs |
| 12:31.19 | Zeeek | such as they are |
| 12:31.41 | jmav | No i prefer to learn and do it my self |
| 12:31.50 | jmav | thx i will read it in details |
| 12:32.11 | Zeeek | The number one way to start up the learning curve is to download http://asteriskdocs.org PDF book |
| 12:32.26 | Inv_arp | ~docs |
| 12:32.27 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:32.31 | *** join/#asterisk becks (~becks@stepbuild.com) |
| 12:32.33 | Zeeek | I'm might be stupid, but I read it cover to cover about three times in 24 hours |
| 12:32.50 | Newbie___ | hahah, fuck it worked |
| 12:32.59 | Zeeek | and that was before they put the naked pictures in |
| 12:33.02 | Newbie___ | define wrong context |
| 12:33.05 | pratik | zeek:do you recomend me to read any docs |
| 12:33.17 | Zeeek | no protik, don't read anything |
| 12:33.30 | Newbie___ | trial and error |
| 12:33.32 | Zeeek | find someone near you that can help |
| 12:33.33 | jmav | thx zeeek |
| 12:33.52 | pratik | here there is no one near that i can find |
| 12:33.53 | Zeeek | jmav you'll be answering questions next tuesday |
| 12:34.00 | *** join/#asterisk MatsK (~NNSCRIPT@246.80-202-58.nextgentel.com) |
| 12:34.26 | Inv_arp | pratik: where? |
| 12:34.26 | pratik | seee my netrworkn is behind some firewall |
| 12:34.27 | Zeeek | Idja |
| 12:34.46 | *** join/#asterisk tih (tih@athene.hamartun.priv.no) |
| 12:35.17 | pratik | Inv_arp:my asterisk is working properly buit the fwd calls are not going |
| 12:35.17 | Zeeek | pratik have you tried to use FWD with X-Lite client to see if that works? |
| 12:36.09 | pratik | no |
| 12:36.15 | Zeeek | why not? |
| 12:36.28 | Zeeek | it will help eliminate other possiblme problems like firewall |
| 12:36.32 | pratik | what is X-Lite, |
| 12:36.57 | jobi | hi all |
| 12:36.57 | pratik | is it a sip provider |
| 12:37.10 | jobi | I'm trying to setup a SIP / ISDN BRI gateway |
| 12:37.18 | jobi | using zaphfc |
| 12:37.55 | jobi | each time I try to make a call from Asterisk I get Unable to create channel of type 'ZAP' |
| 12:38.19 | jobi | is there a way to get some more debug info? |
| 12:38.37 | Zeeek | any messages when you start * ? |
| 12:39.25 | Zeeek | drivers all modprobed and working ok? |
| 12:39.25 | pratik | X-Lite is a softphone |
| 12:39.25 | jobi | no messages about zap |
| 12:39.25 | jobi | and the modules are loaded ok |
| 12:39.48 | Zeeek | jobi sometimes that message just means it can't reach the ZAP device |
| 12:40.00 | Zeeek | at least with SIP and IAX2 that can mean that |
| 12:40.27 | Zeeek | pratik, ya download and play with X-Lite and see if you can get it to work with FWD before using asterisk |
| 12:40.51 | Zeeek | It actually works very well with FWD |
| 12:41.14 | pratik | ok i'll try it,will it get downloaded on linux |
| 12:41.31 | Zeeek | you have any windows machines? |
| 12:41.54 | pratik | ya in my network i have both linux and windows |
| 12:41.57 | Eight | X-Lite seems to be common enough, but it doesn't seem like it's very good, and I certainly detest the interface. Is there another SIP soft client people like? |
| 12:42.15 | Zeeek | Eight it works better (for me) than all the rest |
| 12:42.15 | Eight | Oh, and I've tried SJPhone as well. |
| 12:42.36 | jobi | I got NOTICE[6060]: chan_zap.c:7786 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 12:42.38 | Eight | Zeeek: ya, that's about all I can say for it myself =/ |
| 12:42.39 | Zeeek | the interface is clunky |
| 12:43.15 | Inv_arp | Eight: firefly is a fav around here also |
| 12:43.15 | Zeeek | but I'm talking windows clients - I never got anything to work on linux |
| 12:43.36 | Zeeek | ff was ok for IAX, good for newbies (little config) |
| 12:43.51 | Inv_arp | iny kphone works in *nix linphone kinda buggy |
| 12:44.03 | Inv_arp | err only |
| 12:46.23 | *** join/#asterisk ckruetze (~ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 12:50.06 | *** join/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
| 12:51.10 | Eight | Firefly looks interesting... thanks for the heads up. |
| 12:51.20 | *** join/#asterisk TheEmperor (TheEmperor@218.111.49.132) |
| 12:55.18 | *** join/#asterisk Jas_Williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 12:55.26 | *** join/#asterisk Nix (~Nix@dsl81-214-9283.adsl.ttnet.net.tr) |
| 12:57.10 | *** join/#asterisk hawk-irc (~Domain@host194-67.pool8248.interbusiness.it) |
| 12:57.49 | *** join/#asterisk Brumle (~brumle@brumle.com) [NETSPLIT VICTIM] |
| 12:58.25 | hawk-irc | hi all |
| 12:59.05 | Inv_arp | hey |
| 12:59.38 | Zeeek | bouhaaaa |
| 13:00.32 | jobi | I'm using zapbri and have configured signalling=bri_cpe |
| 13:00.55 | jobi | but when I launch asterisk I still get -- Registered channel 1, PRI Signalling signalling |
| 13:01.03 | *** join/#asterisk BuzzBud (~adoroar@90.62-97-254.bkkb.no) [NETSPLIT VICTIM] |
| 13:01.08 | Zeeek | you restarted ? |
| 13:01.14 | jobi | yes |
| 13:01.34 | Zeeek | well, I shot my wad then :) |
| 13:03.50 | Zeeek | I know people in the USA pay for calls they receive on their cellphones; does calling a cellphone cost more than calling a landline? |
| 13:04.38 | Inv_arp | nope |
| 13:05.13 | Zeeek | Ok, so the only downside of me calling their cell is that they're dumb enough not to have a landline therefore paying to talk to me? |
| 13:05.39 | Inv_arp | correct :) |
| 13:05.44 | Zeeek | that shit is well-marketed though |
| 13:05.58 | Zeeek | Whenever I ask if it costs when I call they say "no, just minutes" |
| 13:06.08 | Zeeek | duh! minutes are free? |
| 13:06.25 | Zeeek | cellphones are a license to print money |
| 13:06.41 | Inv_arp | heh after minutes run out... pay per min |
| 13:07.09 | Zeeek | ok, since it's so quit, who's the linux resident support person available in #asterisk queue ? |
| 13:07.25 | tzanger | ha |
| 13:07.27 | tzanger | what's up |
| 13:07.28 | Zeeek | I have one final issue with my DHCP/TFTPD stuff |
| 13:07.30 | tzanger | I know all |
| 13:07.35 | Zeeek | I believ you |
| 13:07.37 | Inv_arp | heh |
| 13:07.51 | Zeeek | I want to start these two daemons at boot, naturally |
| 13:07.56 | tzanger | ... I see a corrupted reiser partition in your future, make your backups now... |
| 13:08.09 | Zeeek | so... first attempt, uincomment tftp in /etc/inetd.conf |
| 13:08.17 | tzanger | yes |
| 13:08.25 | Zeeek | doesn't run the server |
| 13:08.27 | Zeeek | then... |
| 13:08.29 | Zeeek | wait for it |
| 13:08.39 | tzanger | have you HUPped your inetd lately? |
| 13:08.39 | Zeeek | to get the full ignorance at once... |
| 13:08.51 | Zeeek | there is no inetd in ps aux |
| 13:08.58 | tzanger | you are not running inetd then |
| 13:08.59 | Zeeek | HEY, MAYBE THAT'S IT! |
| 13:09.03 | tzanger | (I never do either) |
| 13:09.03 | Zeeek | heh |
| 13:09.08 | Zeeek | ok so next |
| 13:09.10 | tzanger | I just usually run |
| 13:09.16 | Zeeek | I triesd to put it in rc.d/rc.local |
| 13:09.23 | tzanger | /usr/sbin/in.tftpd -l /tftpboot in rc.local |
| 13:09.29 | Zeeek | the effect of which is no console |
| 13:09.41 | tzanger | you'll notice the -l flag |
| 13:09.50 | tzanger | -l = listen, daemonize, background |
| 13:09.50 | Zeeek | yeah I think I used it |
| 13:09.57 | tzanger | what distro |
| 13:09.59 | Zeeek | lemee seeeheah |
| 13:10.04 | Zeeek | Slack 9.1 |
| 13:10.16 | Eight | Slack is up to 9.x? wow. |
| 13:10.19 | Inv_arp | *woah he does know all* :) |
| 13:10.20 | tzanger | works just fine for me |
| 13:10.30 | tzanger | I am running slack 10 |
| 13:10.34 | Zeeek | what about dhcpd ? |
| 13:10.59 | tzanger | dhcpd is runnign on the wrt54g |
| 13:11.02 | tzanger | but it's the same |
| 13:11.05 | tzanger | dhcpd eth0 |
| 13:11.07 | Zeeek | I had this: /usr/sbin/dhcpd -q -d |
| 13:11.08 | tzanger | is all you need to run |
| 13:11.20 | tzanger | <PROTECTED> |
| 13:11.25 | tzanger | it's running on my systme :-) |
| 13:11.32 | Zeeek | how can I chain from the router to the tftp server? |
| 13:11.36 | tzanger | /usr/sbin/in.tftpd -l /tftpboot |
| 13:11.44 | tzanger | Zeeek: ahhh young grasshopper |
| 13:11.50 | tzanger | you must learn the bootp options |
| 13:11.52 | Zeeek | I didn't see any way |
| 13:11.59 | *** join/#asterisk kamran (~kamran@mbl-82-51-9.dsl.net.pk) |
| 13:12.07 | tzanger | siaddr ip.of.tftp.server |
| 13:12.18 | Zeeek | whazzzat? |
| 13:12.19 | tzanger | boot_file /tftpboot/file.to.boot |
| 13:12.26 | tzanger | that must be in your dhcpd.conf |
| 13:12.35 | tzanger | (well I use udhcpd, dhcpd's options may be slightly different) |
| 13:12.39 | Zeeek | the router has no file? |
| 13:12.42 | tzanger | and it depends on if you're doing dhcp or bootp |
| 13:12.47 | tzanger | the router file looks like this |
| 13:13.20 | tzanger | start 192.168.1.100 |
| 13:13.20 | tzanger | end 192.168.1.149 |
| 13:13.37 | tzanger | option dns 192.168.1.1 |
| 13:13.41 | tzanger | option domain mixdown.ca |
| 13:13.41 | tzanger | siaddr 192.168.1.9 |
| 13:13.41 | tzanger | boot_file /tftpboot/pxelinux.0 |
| 13:13.46 | tzanger | now |
| 13:13.47 | tzanger | as I said |
| 13:13.54 | tzanger | it depends on if you're doing bootp or dhcp to tftp |
| 13:13.56 | tzanger | I'm using the former |
| 13:13.57 | Zeeek | ??? on my linksys NAT router? |
| 13:14.15 | Zeeek | where's the slot for the diskette? |
| 13:14.24 | tzanger | my myth box uses bootp to get an IP and the udhcpd server gives it 192.168.1.9:/tftpboot/pxelinux.0 as the next step |
| 13:14.32 | tzanger | Zeeek: well you need a better router |
| 13:14.36 | tzanger | get a wrt54g :-) |
| 13:14.49 | Zeeek | I have one at home - I don't remember any options like that |
| 13:14.55 | Zeeek | but here it's a WAG54g |
| 13:14.56 | tzanger | you have to add it manually |
| 13:15.04 | Zeeek | which is similar but with a DSL modem |
| 13:15.10 | tzanger | Zeeek: wha? |
| 13:15.13 | kamran | hi all any one using asterisk-oh323. i have problem in routing my calls to gungk gatekeeper |
| 13:15.14 | tzanger | wag54g eh? |
| 13:15.17 | Zeeek | MANUALLY ? Nevah! |
| 13:15.23 | Zeeek | as in wag the dog |
| 13:15.35 | tzanger | I'm running sveasoft's firmware but openwrt would work too |
| 13:15.39 | tzanger | and then you wouldn't have to add it manually |
| 13:15.42 | Zeeek | the thing disconnects every time you change something |
| 13:15.52 | tzanger | I did not know wag54g existed |
| 13:15.59 | tzanger | that could be the solution to my problem |
| 13:16.10 | Zeeek | well, back to the real world, all I need to do is get dhcpd to run at boot |
| 13:16.10 | tzanger | use it to do the rc.tc script I have |
| 13:16.12 | tzanger | verrrrry nice |
| 13:16.25 | tzanger | getting dhcpd to run at boot isn't a big problem |
| 13:16.28 | Zeeek | I think it's only sold here in Eu |
| 13:16.28 | tzanger | dhcpd eth0 |
| 13:16.35 | tzanger | and make sure /etc/dhcpd.conf is set up right |
| 13:16.36 | Inv_arp | heh every ques that has h323 also has problem in it |
| 13:16.45 | Zeeek | possibly because some US providers are very anal about what modem/router you use |
| 13:16.53 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 13:17.38 | Zeeek | when I run dhcpd --q -d in rc.local there is no console |
| 13:17.45 | Zeeek | -q -d |
| 13:19.52 | tzanger | did I say to use -q -d ? |
| 13:19.59 | Zeeek | no sir |
| 13:20.05 | tzanger | thou needst to consult thy man pages |
| 13:20.08 | Zeeek | and there's only one interface |
| 13:20.11 | tzanger | anyway gotta get these kids to school |
| 13:20.18 | Zeeek | I did but that's what it appears to suggest |
| 13:20.28 | Zeeek | bye |
| 13:21.17 | *** join/#asterisk bunny_700 (~ced_fou@213-193-168-25.adsl.easynet.be) |
| 13:28.30 | darkskiez | is the TE110P dual voltage? |
| 13:29.43 | Zeeek | who has a long test number for iAX? didn't someone have a local weathr station or something? anyone? |
| 13:29.53 | Zeeek | a number that plays a long thing to test |
| 13:29.59 | darkskiez | digium |
| 13:30.12 | darkskiez | its part of example dial plan |
| 13:30.45 | Zeeek | k |
| 13:31.13 | darkskiez | quiet in here! |
| 13:31.23 | *** join/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
| 13:31.45 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:32.10 | Zeeek | it was real noisy a few moments ago! |
| 13:32.28 | Zeeek | you shut everyone up with your hi-falutin advanced question |
| 13:32.43 | Zeeek | we're all scared to talk now :) |
| 13:33.54 | *** part/#asterisk casterman (~casterman@63.240.97-84.rev.gaoland.net) |
| 13:38.09 | *** join/#asterisk marak (~twist@ndn-165-130-148.telkomadsl.co.za) |
| 13:39.15 | marak | hi all, anybody can help me on call transfer problems ? |
| 13:39.50 | Zeeek | go for it |
| 13:40.21 | marak | when calls come in most of the time we can transfer without hassles. but if we dial out there is no way we can transfer ? |
| 13:40.37 | Zeeek | what phone and how are you transfering? |
| 13:40.47 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34) |
| 13:40.49 | marak | using firefly softphones using # |
| 13:40.59 | marak | # plus extension |
| 13:41.08 | marak | normaly after pushing # it says extension please |
| 13:41.13 | Zeeek | what dial options? you have the 'T' ? |
| 13:41.28 | marak | no only ,tr |
| 13:41.35 | marak | should i use the capital |
| 13:42.10 | marak | this is my dial command : exten => _9.,1,Dial(Zap/g1/${EXTEN:1},20,tr) |
| 13:42.11 | Zeeek | find the part that explains the dial application |
| 13:42.27 | Zeeek | one way to do this is to type show application dial |
| 13:42.50 | Zeeek | <PROTECTED> |
| 13:42.53 | marak | i have done that it tells me the same thing for both t and T |
| 13:43.44 | Zeeek | then something has distorted the universe horribly |
| 13:44.01 | marak | ahhh sorry just reread it |
| 13:44.07 | Zeeek | maybe dial is broken in the version you have |
| 13:44.15 | Zeeek | yes, good |
| 13:45.19 | *** join/#asterisk ncjp (~switch@61.206.115.5.user.ad.il24.net) |
| 13:46.52 | marak | would 'm' be to supplement 'r' ? |
| 13:47.16 | Zeeek | why not read carefully what is said there |
| 13:47.28 | Zeeek | and make a few tests |
| 13:51.10 | *** join/#asterisk tih (tih@athene.hamartun.priv.no) |
| 13:51.19 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 13:56.12 | tzanger | whirrrrrrrrred |
| 13:56.17 | marak | thanks all |
| 13:56.19 | *** part/#asterisk marak (~twist@ndn-165-130-148.telkomadsl.co.za) |
| 14:01.10 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 14:01.58 | shadebob | Hi, how I can prompt a user to send his agentID/password and after make a agentlogin in my extension.conf? |
| 14:02.14 | *** join/#asterisk blankman (~chatzilla@h000d88a1570c.ne.client2.attbi.com) |
| 14:02.39 | blankman | Hey guys. Is there anyone from NuFone on? |
| 14:03.21 | tzanger | not from nufone but I'm an avid user and supporter of... :-) |
| 14:04.20 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
| 14:04.32 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 14:05.09 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 14:05.35 | blankman | k. So I use them as well, and I like them, but recently I have been having an issue and I am trying to track down if anything changed on their side in the last week or so. |
| 14:05.44 | Katty | morning |
| 14:05.48 | tzanger | whirred |
| 14:05.50 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 14:06.28 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 14:07.11 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 14:07.19 | tzanger | fix yer client W1thdraw |
| 14:07.21 | *** join/#asterisk ph_matrix (~potchy_fe@203.115.169.48) |
| 14:11.10 | blankman | tzanger, what version of * are you running? |
| 14:11.25 | blankman | Head? |
| 14:11.36 | nirs | is there a reason why would zaptel think an FXO module is actually an FXS module ? |
| 14:11.49 | tzanger | blankman: yes |
| 14:11.59 | tzanger | nirs: on what hardware |
| 14:12.08 | blankman | nirs, you modprobe'd in the wrong order maybe :-) |
| 14:12.13 | nirs | ah ? |
| 14:12.22 | nirs | it's a simple pentium 4 box |
| 14:12.53 | tzanger | nirs: what hardware is the TDM hardware... T1, TDM400P, what |
| 14:13.12 | Katty | gosh, no one answered. |
| 14:13.23 | blankman | Hey Katty. |
| 14:13.26 | blankman | That better ;-) |
| 14:13.28 | tzanger | Katty: I did so |
| 14:13.31 | nirs | TDM400P |
| 14:13.36 | Katty | mad, i didn't see it |
| 14:13.38 | nirs | there are 3xFXO modules on it |
| 14:13.42 | nirs | and 1xFXS module |
| 14:14.08 | tzanger | nirs: ok. what does dmesg say |
| 14:14.08 | tzanger | paste teh 4 lines here |
| 14:14.16 | Katty | i've obviously insaned. |
| 14:14.29 | nirs | the funny part is that dmesg says 3xfxo + 1xfxs |
| 14:16.09 | ta[i]nted | how much are virtual DIDs? |
| 14:16.11 | nirs | any ideas ? |
| 14:16.28 | bjohnson | Katty: if good manners and human interaction were our strong points, we wouldn't be geeks |
| 14:16.42 | blankman | ta[i]nted: that depends on who you buy you line from. |
| 14:17.09 | nirs | the stupid part that if I indicate that the modules are fxx for the fxo modules on zaptel it loads up nicely |
| 14:17.15 | ta[i]nted | blankman u mean my carrier line? |
| 14:17.20 | blankman | If you are using an ITSP it is anywhere from free-50 bucks |
| 14:17.53 | ta[i]nted | blankman i thought places like XO sells DIDs |
| 14:17.58 | nirs | tzanger |
| 14:17.59 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4463:7269:1:20e:a6ff:fe66:c5a3) |
| 14:18.00 | nirs | odule 0: Installed -- AUTO FXS/DPO |
| 14:18.00 | nirs | Module 1: Installed -- AUTO FXO (FCC mode) |
| 14:18.00 | nirs | Module 2: Installed -- AUTO FXO (FCC mode) |
| 14:18.00 | nirs | Module 3: Installed -- AUTO FXO (FCC mode) |
| 14:18.15 | tzanger | nirs: and what's wrong with that |
| 14:18.18 | blankman | ta[i]nted: your DID's have to be tied to a trunk (pri, digital, etc). |
| 14:18.19 | tzanger | 1 FXS, 3 FXO |
| 14:18.20 | nirs | that is fine |
| 14:18.21 | tzanger | just like you said |
| 14:18.30 | nirs | well, modprobe is just fine |
| 14:18.40 | ta[i]nted | blankman right - the trunk i've got handled.. |
| 14:18.40 | nirs | but when I run ztcfg, that fucks up big time |
| 14:18.51 | tzanger | nirs: is your zaptel.conf file right for that card? |
| 14:19.03 | nirs | well, I think it is |
| 14:19.11 | tzanger | i.e. channel 1 is fxo signalled and 2-4 fxs signalled? |
| 14:19.18 | blankman | You have to neg. with the provider for the pricing, but usually in my experience it is about 10 bucks a month for about 100. |
| 14:19.20 | nirs | fxsks=1 |
| 14:19.20 | nirs | fxoks=2-4 |
| 14:19.25 | tzanger | wrong |
| 14:19.29 | ta[i]nted | blankman but if my trunk is IP-based, do I have to go with ITSP DIDs or can I get DID through other means? |
| 14:19.36 | nirs | oh ? |
| 14:19.43 | tzanger | nirs: it's mentioned in zaptel.conf and in zapata.conf that fxs devices use fxo signaling and vice-versa |
| 14:19.51 | nirs | oops |
| 14:19.55 | ta[i]nted | blankman who do you go through? |
| 14:19.55 | nirs | << feels silly |
| 14:20.14 | nirs | actually, coming to think about it, it makes sense |
| 14:21.14 | blankman | Nine different providers ;-) But yes you can from both ITSP and "hardline" providers. |
| 14:22.10 | blankman | nirs, also, you sure the order is: fxo,fxs,fxs,fxs? |
| 14:22.16 | bjohnson | Katty: by the lack of reply I assume you were called away or you too, are a true geek |
| 14:25.49 | nirs | yes |
| 14:25.51 | nirs | it works fine |
| 14:25.53 | nirs | just checked |
| 14:25.55 | nirs | thanks |
| 14:26.49 | *** join/#asterisk Conductor (~thomas@62.8.240.132) |
| 14:27.04 | Conductor | how would you solve this: when dialing 123 i want the caller to be added to a conference room. after that, ${EXTEN:1} should be added to the conference also. |
| 14:28.43 | bjohnson | the first part is covered by numerous examples |
| 14:28.50 | bjohnson | the second part I don't understand |
| 14:30.08 | *** join/#asterisk gmcinnes (~gmcinnes@67.71.63.9) |
| 14:30.14 | *** join/#asterisk dercol (~ercolani@sei.yacme.com) |
| 14:30.22 | gmcinnes | hi everyone. I have a hardware question. |
| 14:30.40 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 14:30.50 | mgth | Is the bugtracker down? |
| 14:31.05 | gmcinnes | I have a tdm400p which needs a ide power supply. All the power on my server is backplane though. |
| 14:31.13 | bjohnson | the bugs crawled off with it |
| 14:31.32 | gmcinnes | Has anyone used a dell 2600 poweredge with a tdm400p? |
| 14:35.02 | nirs | WHAT THE F*** HAPPENED TO voicemail.conf ???? |
| 14:35.33 | nirs | voicemail.conf now has a similar number of options to extensions.conf |
| 14:35.35 | nirs | crazy |
| 14:35.47 | *** join/#asterisk kant (~bernd@63.245.57.70) |
| 14:36.52 | shadebob | when I use agentcallbacklogin, i don't known what is new extension? Someone can help me? |
| 14:37.09 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:37.59 | Zeeek | money? |
| 14:38.03 | `Sauron | Eight: Now what? |
| 14:38.07 | Eight | Nothing new. |
| 14:38.19 | Eight | but it's 9:30 AM and there's still nobody answering the '24/7' support line =p |
| 14:38.34 | `Sauron | I thought it was staffed 24/7 |
| 14:38.41 | Eight | `Sauron: That's the impression I got, too. |
| 14:38.47 | Zeeek | yeah but at 7 they're all there, all 24 |
| 14:38.49 | Eight | Well, before I called it. |
| 14:41.35 | Conductor | bjohnson, the first part is easy, right. |
| 14:41.46 | kant | 24/7 means 24 minutes every hour for 7 hours. |
| 14:41.55 | Conductor | bjohnson, ill try to explain the second part: |
| 14:41.59 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 14:42.19 | *** part/#asterisk blankman (~chatzilla@h000d88a1570c.ne.client2.attbi.com) |
| 14:42.19 | *** part/#asterisk bunny_700 (~ced_fou@213-193-168-25.adsl.easynet.be) |
| 14:42.33 | Conductor | bjohnson, after the caller has been added to the conference, another call shall be triggered and joined to the conference also. |
| 14:42.38 | `Sauron | I managed to educated guess my way to what happened with the sip INVITE authentication thing |
| 14:42.47 | `Sauron | and why there wasn't any prior notice |
| 14:43.11 | Conductor | bjohnson, a function like the "Originate"-Action with *-manager... |
| 14:44.07 | Eight | `Sauron: care to enlighten the rest of us? |
| 14:44.48 | bjohnson | kant: I thought it was 24 minutes every day, 7 days a week |
| 14:45.00 | `Sauron | My guess was that there was some big security vuln. - so they had to upgrade their gear, and as a side effect of the upgrade, it tightened down on RFC compliance on some SIP stuff |
| 14:45.17 | `Sauron | * didn't have the compliance, and thus the loud screaming |
| 14:45.23 | bjohnson | Conductor: so you want to call somone else and when they answer, dump them into an active conference? |
| 14:46.08 | `Sauron | sigh |
| 14:46.20 | `Sauron | it's sooo hard to sit with proper posture in these chairs |
| 14:46.26 | `Sauron | they lend themselves all too well to slouching |
| 14:46.34 | antifuchs | good chairs are hard to find /-: |
| 14:46.43 | antifuchs | good posture even more so (: |
| 14:46.44 | hawk-irc | hi to all... anybody knows a guy with "akholsmith" as email username? |
| 14:46.52 | hawk-irc | i met him here but forgot his nick :( |
| 14:47.28 | Conductor | bjohnson, yes |
| 14:48.15 | puzzled | hawk-irc: search the -users mailinglist for his email address. he posts tons of messages |
| 14:48.43 | jakepdev | anyone know why I can talk between SIP phones using *, and get audio, but I never hear * voiceprompts? |
| 14:49.01 | hawk-irc | where should i carry on that search? |
| 14:49.13 | `Sauron | You forget to call answer() before you start sending audio in *, maybe |
| 14:49.42 | jakepdev | Sauron - it happens even with the 1234 demo |
| 14:50.45 | `Sauron | YEah, I noticed some of the demos weren't accurate |
| 14:51.03 | jakepdev | good point... -i'll check that |
| 14:51.05 | bjohnson | Conductor: I haven't done it but I think I've read about it (on the wiki) being done with the agi |
| 14:51.39 | jakepdev | nope It says executing Answer |
| 14:52.35 | jakepdev | Sauron - any other ideas? |
| 14:52.44 | `Sauron | Nope |
| 14:52.58 | Zeeek | jakeppdev tried on different phones? |
| 14:52.59 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
| 14:53.19 | puzzled | hawk-irc: search on google for his name |
| 14:53.22 | _THEEND_ | hi! |
| 14:53.26 | Conductor | bjohnson, do you remember in which context? what do i have to search for? |
| 14:53.52 | jakepdev | yep - SJPhone and DTA310 - smae results - can talk to each other through *, but no voice prompts when connecting to * |
| 14:54.14 | _THEEND_ | someone could help me pls!? |
| 14:54.24 | Zeeek | BEGIN |
| 14:54.43 | hawk-irc | puzzled: found!!! it's tzanger |
| 14:55.04 | Zeeek | he had to drive the kids to school |
| 14:55.14 | Zeeek | ~seen tzanger |
| 14:55.16 | jbot | tzanger is currently on #asterisk. Has said a total of 267 messages. Is idling for 35m 33s |
| 14:55.19 | tzanger | I'm here |
| 14:55.25 | Zeeek | tsk, tsk, left the box on |
| 14:55.28 | Zeeek | aha |
| 14:55.35 | Zeeek | thx your answers fixed everything |
| 14:55.40 | Zeeek | but you left too soon |
| 14:56.22 | _THEEND_ | someone uses web interface to configure asterisk? |
| 14:56.53 | jakepdev | THEEND - I used it.. what's the question? |
| 14:57.14 | tzanger | Zeeek: good, I am glad :-) |
| 14:57.24 | *** part/#asterisk mtmachen (~matthewma@cable-68-113-71-35.grd.al.charter.com) |
| 14:58.07 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
| 14:59.28 | *** join/#asterisk bacondoublechz (~bacon@69-162-37-142.stcgpa.adelphia.net) |
| 15:00.07 | _THEEND_ | no question i'm looking for a web interface |
| 15:00.20 | _THEEND_ | what you suggest? |
| 15:00.40 | tzanger | _THEEND_: have you looked at AMP? I won't touch it as it's PHP but I imagine it works well enough |
| 15:00.55 | jakepdev | THE END - AMP |
| 15:01.23 | bjohnson | Conductor: no. can't find it now |
| 15:01.31 | jakepdev | THEEND - I used it and it seems to work just fine |
| 15:01.38 | santiago | _THEEND_, try destar |
| 15:01.52 | santiago | _THEEND_, http://developer.berlios.de/projects/destar/ |
| 15:01.55 | Conductor | bjohnson, ok thanks anywa |
| 15:03.20 | gmcinnes | Has anyone used a dell 2600 server for asterisk? |
| 15:04.45 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
| 15:04.56 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 15:07.09 | TheBear | using a snom200 what do I put in extensions.conf to get intercom working. I have applied the chan_sip.c hack given, but still can't get it to work ? |
| 15:07.18 | TheBear | any help appriecated |
| 15:08.50 | ariel_ | good morning all |
| 15:09.18 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 15:10.12 | TheBear | hi |
| 15:14.13 | Eight | Hmm... anyone know how to get the 'transfer' feature in x-lite working happily with Asterisk? |
| 15:14.36 | Zeeek | It's disbled in x6lite, isn't it? |
| 15:14.49 | Eight | ah, maybe that's it =) |
| 15:14.50 | Zeeek | only for X-Pro |
| 15:15.28 | *** join/#asterisk ruiner (ruiner@ruiner.netslacking.net) |
| 15:18.57 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 15:19.17 | TheBear | anyone know a good link for dialplan that has intercom=true for snom 200s ? |
| 15:20.27 | puppet | hmm |
| 15:20.39 | puppet | Can I do in sip.comf a new [] with another register? |
| 15:20.56 | bjohnson | TheBear: have you checked the wiki? |
| 15:21.05 | puppet | f.ex. under [default] my main and then a new [germany] and take my sipgatelink? |
| 15:21.38 | bjohnson | puppet: you can have numerous contexts and numerous registers (registers do not go in the general section .. not in the context) |
| 15:21.58 | Eight | rather, registers DO go on the general section. |
| 15:22.14 | bjohnson | the register just tells the other server what IP address you have |
| 15:22.26 | TheBear | bjohnson: I have looked everywhere I can think of. I wiki I find the hack to chan_sip.c which I have applied, but nothing about what extensions.conf must look like |
| 15:22.27 | jakepdev | I give digium support an A+ |
| 15:22.30 | SexyKen | Does uniqueid stay the same for the same call? For instance, in an ivr system...a user calls an incoming line. They're connected to the ivr. They press 1 for sales. Asterisk connects them to an agent. |
| 15:22.31 | bjohnson | Eight: yes .. sorry for the confusion |
| 15:22.39 | puppet | Cuase im playing with Op Panel |
| 15:22.51 | puppet | and I want it when I call one link I get a inc button with light |
| 15:22.56 | SexyKen | In the cdr, how am I supposed to know that those two calls are connected to the same user? |
| 15:22.58 | puppet | but right now it just goes on sip/* |
| 15:26.12 | puppet | Hmm |
| 15:26.19 | puppet | should I add /sipgate |
| 15:26.24 | puppet | after the register if i get it right? |
| 15:28.31 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:28.31 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:32.11 | *** join/#asterisk JunK-Y (~grepmoo@65.39.228.5) |
| 15:32.31 | xkev | is there some magic way to get rid of channels that won't soft hangup? |
| 15:32.52 | xkev | <PROTECTED> |
| 15:33.00 | xkev | kinda broken there |
| 15:33.54 | Conductor | is there a way to originate a call with a regular asterisk cmd? |
| 15:33.58 | anthm | you could patch the softhangup cli to take -f and do a real hanfup buy 8/10 times it would crash the box |
| 15:34.15 | Conductor | it is possible using the manager... but this is much harder. |
| 15:34.16 | xkev | anthm yeah, like when one forces a zap channel to die |
| 15:34.33 | xkev | conductor, do you mean from the cli? |
| 15:34.44 | Conductor | xkev, no in extensions.conf |
| 15:34.48 | xkev | Dial() |
| 15:34.52 | xkev | ? |
| 15:35.01 | xkev | if you are in extensions.conf, you already have a channel |
| 15:35.04 | Conductor | xkev, but this would not let me execute any more commands |
| 15:35.16 | xkev | you can do the /var/spool/asterisk/outgoing/ dance |
| 15:35.35 | xkev | conductor, Dial() can take a 'M(macro^arg1^arg2^...)' upon answer |
| 15:35.38 | Conductor | xkev, I want to call somone else and when they answer, go into an active conference |
| 15:35.46 | anthm | come to cluecon and start to learn to code these and many more apps =D |
| 15:35.53 | *** join/#asterisk [ro]nic3try (~iancu@81.181.199.39) |
| 15:35.56 | [ro]nic3try | re all |
| 15:36.11 | xkev | or 'g' to continue on if called party hangs up, but there is no way you can continue on if the caller hangs up, as you no longer have a channel structure |
| 15:36.20 | xkev | conductor, you want M() |
| 15:36.32 | xkev | I use that for some findme prompting, etc |
| 15:36.55 | xkev | pbx*CLI> show application dial, see M(x[^arg]) |
| 15:37.15 | Conductor | xkev, so i write a macro which calls the conference and when this is done it jumps back and Dials another number? |
| 15:37.16 | TheBear | perhaps progress. I have the setvar(_VXML.... and then Dial(SIP/2205) the phone rings, but doesn't auto answer, like a paging or intercom. |
| 15:37.21 | xkev | ..conductor oh, I see where you want the caller and the called to join the same meetme |
| 15:37.30 | Conductor | xkev, yes |
| 15:38.35 | TheBear | would I need to have load => app_intercom.so in modules or is this done automatically ? |
| 15:38.37 | puppet | <PROTECTED> |
| 15:38.49 | jakepdev | is the command to unload zaptel "modprobe -r zaptel"? |
| 15:38.59 | xkev | thebear app_intercom is deprecated, chan_oss is used now |
| 15:39.00 | zipp | jakepdev, rmmod |
| 15:39.16 | xkev | thebear, but you want a sip ua to do an auto-answer right? |
| 15:39.21 | xkev | ..there are solutions for that on the wiki |
| 15:39.29 | jakepdev | zipp - "rmmod zaptel"? |
| 15:39.41 | xkev | rmmod <the card driver> first though :) |
| 15:39.58 | jakepdev | tnx |
| 15:40.12 | Conductor | xkev, do you have an idea? |
| 15:40.13 | TheBear | xkev: I see that in modules.conf, just trying to find why this won't work ? |
| 15:40.19 | xkev | conductor what phone is it |
| 15:40.41 | Conductor | softphone |
| 15:40.42 | xkev | chan_oss is for using a sound card as an overhead pager |
| 15:40.49 | xkev | conductor, that is dependent on the phone |
| 15:40.58 | xkev | there is no auto-answer standard for sip |
| 15:41.03 | anthm | you could add another option to Dial G(<exten>) whereby when the call is successful instead of bridge you send both parties to the specific exten |
| 15:41.21 | xkev | anthm hmmm, if only I had more time :) |
| 15:41.43 | xkev | busy feature-bloating my app_queue :) |
| 15:41.44 | Conductor | xkev, youre * developer? |
| 15:41.51 | anthm | paypal me 50 bux and i'll have it in mantis in 30 min |
| 15:41.56 | xkev | I fix/add things that I need, conductor |
| 15:42.21 | xkev | anthm hehe, I'll do it for 29 mins for $100 :) |
| 15:42.23 | Conductor | xkev, you know you really NEED this option! |
| 15:42.27 | ManxPower | Don't use the Intercom Application. It will be going away at some point. Set up an OSS or ALSA console phone (see /etc/asterisk/alsa.conf or /etc/asterisk/oss.conf), set it to auto answer and then use the Dial application to call the port for overhead paging. |
| 15:42.39 | *** part/#asterisk santiago (~santiago@63.245.86.95) |
| 15:42.47 | *** join/#asterisk viLeR (1000@ip-33-7.telesat.com.co) |
| 15:42.49 | anthm | ok |
| 15:43.10 | FaithX | anyone got hfczap working? |
| 15:43.14 | Conductor | anthm, you also need this option. how can you even live without it? ;) |
| 15:43.29 | xkev | anthm, but my version will probably segfault |
| 15:43.48 | *** join/#asterisk DevilFish (~me@staff211.qtm.net) |
| 15:43.50 | epoch | hrmmm |
| 15:43.53 | anthm | easy cos my conference app has attended add ppl to the conf feature |
| 15:44.02 | epoch | any Polycom SoundPoint IP300/500/600 users around? |
| 15:44.02 | riksta | gentoo portage people are on absolute crack, they mask and unmask zaptel 1.0.4 and revert to 1.0.1 like every two days, for the past 2 months.....wtf |
| 15:44.13 | xkev | anthm, is that what I'm thinking of? the new app_conference? |
| 15:44.22 | xkev | ..wrt dial-out to bring people in |
| 15:44.24 | anthm | that is stevek's |
| 15:44.29 | TheBear | ManxPower: I kind of follow what you're saying is there a wiki page or something on that ? |
| 15:44.36 | anthm | his goes for efficiency mine is a retooling of meetme |
| 15:44.44 | ManxPower | TheBear: That is an excersize for the reader. |
| 15:44.49 | anthm | but i may someday adopt some of his efficiency techniques |
| 15:44.52 | xkev | anthm, where are the sounds for join/leave anyway |
| 15:44.57 | DevilFish | I have a situation where after about 4 hours calls from the PSTN (SIP channel) I'm getting really bad chop and calls start dropping.. looking for some ideas on this |
| 15:44.59 | ManxPower | The I was talking about overhead paging, not phone to phone intercom. |
| 15:45.00 | shadebob | hi, i have problem with parked call... When I press # on my budgetone no action... how I can configure parked call? |
| 15:45.12 | anthm | you mean in meetme ? |
| 15:45.14 | xkev | y |
| 15:45.19 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
| 15:45.22 | anthm | in header files |
| 15:45.24 | DevilFish | I can see where the provider sends then INVITE and asterisk is not sending back an ACK |
| 15:45.33 | ManxPower | shadebob, shadebob # only works for calles that happen because of Dial. |
| 15:45.35 | DevilFish | then the calls drop |
| 15:45.46 | TheBear | ManxPower: I have the default alsa and oss .confs which already have autoanswer=yes |
| 15:46.08 | ManxPower | TheBear, You need to mak e sure sound works on the box without Asterisk first. |
| 15:46.12 | anthm | apps/enter.h |
| 15:46.17 | shadebob | ManxPower : and how can I parked a call? |
| 15:46.44 | ariel_ | shadebob, do you have a flash key on the phone? |
| 15:46.47 | xkev | oh god, it's hardcoded |
| 15:46.57 | shadebob | yes |
| 15:47.03 | shadebob | ariel_ yes |
| 15:47.03 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 15:47.09 | ManxPower | shadebob, Well is the call getting to the phone via a Dial line? |
| 15:47.23 | ariel_ | the set features.conf up and just flash type the exten number like 700 and it's parked. |
| 15:47.35 | shadebob | ManxPower : SIP to SIP |
| 15:47.36 | DevilFish | epoch: I use polycoms 500s and 600s |
| 15:47.40 | xkev | anthm, during the developer conference last week, I noted the enter sound was less annoying. I've been pining. :) |
| 15:47.43 | ManxPower | ariel_, He's using a barbie tone |
| 15:47.50 | ariel_ | argh |
| 15:48.01 | ManxPower | shadebob, You are dialing between phones without Asterisk? |
| 15:48.02 | anthm | that is cos we run my retooled meetme |
| 15:48.08 | anthm | app_confcall |
| 15:48.35 | xkev | anthm, I have a conf in an hour, where can I slerp and give it a whirl |
| 15:49.07 | shadebob | ManxPower : no, sip asterisk sip |
| 15:49.19 | ManxPower | shadebob, Then you have a Dial line in extensions.conf the dials the call, right? |
| 15:49.30 | neopher | I know it is possible to go from text to speech with festival, is it possible to go from speech to text? |
| 15:49.40 | ManxPower | neopher, no. |
| 15:49.58 | DevilFish | I thought that was what sphinx was for |
| 15:50.03 | puppet | It is evil :/ |
| 15:50.07 | TheBear | ManxPower: how would I go about checking that ? |
| 15:50.08 | shadebob | ManxPower : yes |
| 15:50.16 | eKo1 | Hmm...looks like my multi-homing setup is fucked up. |
| 15:50.16 | ManxPower | devel, Correct, but you can't do it with festivle. |
| 15:50.25 | neopher | is there anythink that will go from speech to text for asterisk? |
| 15:50.29 | ManxPower | TheBear, Stop Asterisk, play an mp3 using mpg123 |
| 15:50.29 | DevilFish | ahh yes youre right there |
| 15:50.58 | ManxPower | neopher, See the mailing list. You'll have to build what you want from scratch. |
| 15:51.06 | ManxPower | neopher, And it's not going to work very well |
| 15:51.29 | DevilFish | I'd think it'd be a pretty rough project building that |
| 15:51.48 | xkev | there's been random bits about using sphynx via eagi, but I've never heard the results |
| 15:52.04 | DevilFish | anyone got anything on this at all ..... I have a situation where after about 4 hours calls from the PSTN (SIP channel) I'm getting really bad chop and calls start dropping.. looking for some ideas on this |
| 15:52.32 | DevilFish | I can see where the provider sends then INVITE and asterisk is not sending back an ACK |
| 15:52.35 | DevilFish | then the calls drop |
| 15:53.53 | ariel_ | DevilFish, could they have a limit on the call time? |
| 15:54.00 | [ro]nic3try | has anyone succesfully used MeetMe ? |
| 15:54.17 | ManxPower | Allthe time |
| 15:54.21 | DevilFish | well "they" are me and we are using a Metaswitch 3500 |
| 15:54.23 | DevilFish | so no limit |
| 15:54.41 | ManxPower | DevilFish, a 4 hour long call? |
| 15:54.45 | DevilFish | just get bad choppy sound after 4 hrs, sound is fine on the PSTN side |
| 15:54.55 | DevilFish | no just 4 hours of asterisk running |
| 15:55.03 | DevilFish | roughly 4hrs |
| 15:55.22 | DevilFish | we are 0.3 ms to the meta with no packetloss |
| 15:55.47 | DevilFish | its just that for whatever reason this chop starts and then asterisk will not ACK the INVITE |
| 15:55.57 | ManxPower | devel, Any firewall or nat? |
| 15:56.02 | ManxPower | ..er...devil |
| 15:56.10 | DevilFish | no, public IPs all the way on this one |
| 15:56.53 | ManxPower | DevilFish, CVS-HEAD or 1.0.x stable? |
| 15:56.59 | DevilFish | will a packet sniff like ethreal show me more than a standard SIP debug in asterisk? |
| 15:57.03 | *** join/#asterisk VOIP_enthused (~Tony@ip70-187-201-105.dc.dc.cox.net) |
| 15:57.06 | DevilFish | ver 1.0.6 |
| 15:57.17 | ManxPower | devel, Have you tried 1.0.7rc1? |
| 15:57.20 | DevilFish | this has been happening on all versions so far though |
| 15:57.43 | ManxPower | DevilFish, I've not seen the problem in any of my asterisk servers. |
| 15:57.48 | ManxPower | DevilFish, no IRQ shareing? |
| 15:57.59 | VOIP_enthused | Can someone tell me how scalable the Asterisk system is, in terms of maximum number of concurrent teleophony sessions per PBX |
| 15:58.17 | DevilFish | I'm wondering if it is a problem with the Metaswitch but...not sure how to tell |
| 15:58.17 | ManxPower | VOIP_enthused, That depdns on about 40,000 different things. |
| 15:58.19 | epoch | VOIP_enthused: it all depends on hardware |
| 15:58.23 | *** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 15:58.24 | xkev | VOIP_enthused, depends on transcoding, hardware, etc. |
| 15:58.31 | DevilFish | especially when I can clearly see astrisk not ACKing |
| 15:58.48 | *** join/#asterisk dan2 (~beta3@dan2.active.supporter.pdpc) |
| 15:58.48 | xkev | not acking or not 200 OKing? |
| 15:58.49 | ManxPower | DevilFish, Well it's not a general problem. |
| 15:59.07 | Darwin35 | ok porting fetival is harder on 1.95 then 1.4.3 |
| 15:59.20 | DevilFish | yeah I pretty much can see that now, and I'm really scraping the barrel looking for some clue and just turning up nothing |
| 15:59.33 | ManxPower | DevilFish, No IRQ shareing? |
| 15:59.46 | DevilFish | does anyone or do you know of anyone using a Metaswitch 3500? |
| 16:00.04 | TheBear | ManxPower: I can run 'mpg123 mymusic.mp3' I get no errors I have no speakers connected so I have no way of knowing if anything played |
| 16:00.12 | DevilFish | IRQ sharing? what do you mean? |
| 16:00.33 | ManxPower | DevilFish, pput the output of cat /proc/interrupts to pastebin.ca |
| 16:00.34 | VOIP_enthused | How about a Dual processor Xeon, 1G Memory, SIP PHones running G729a, how many concurrent sessions can the PBX manage for sessions within the PBX? |
| 16:00.46 | DevilFish | ok just a sec |
| 16:00.51 | ManxPower | VOIP_enthused, About 40 I would guess. |
| 16:01.13 | ManxPower | VOIP_enthused, G729 is a CPU intensive codec. Amount of memory doesn't really matter as long as the system is not swapping. |
| 16:01.20 | xkev | thebear are you trying to do overhead paging or just intercom between phones |
| 16:01.30 | epoch | VOIP_enthused: keep in mind, though, that most SIP phones support "reinvite" |
| 16:01.35 | DevilFish | ManxPower: here we are http://pastebin.ca/7185 |
| 16:01.46 | xkev | VOIP_enthused, 729 will also cost you $10/concurrent channel |
| 16:01.46 | epoch | VOIP_enthused: which can take the audio path away from the PBX.... |
| 16:01.52 | *** part/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
| 16:01.58 | Darwin35 | madplayer seems not to lock up like mpg123 |
| 16:02.03 | epoch | though that's only really useful if these are phones on the same LAN, or if they're directly reachable |
| 16:02.08 | xkev | I prefer sip<-ulaw->zap for nearly nothing load |
| 16:02.42 | ManxPower | I always ue ulaw if the phones are on the local lan with the Asterisk server. |
| 16:02.53 | VOIP_enthused | That's not very scalable? Why would the PBX care about the G729a compression if it's done on the SIP telephone device level. Won't it just manage the call set up? |
| 16:02.55 | DevilFish | not quite sure what I'm looking at here though ... http://pastebin.ca/7185 |
| 16:03.01 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 16:03.30 | xkev | VOIP_enthused, if it's just passthrough, then it doesn't care, but if it has to convert to/from gsm/wav/ulaw then it needs to recrunch the data |
| 16:03.33 | ManxPower | DevilFish, nevermind. You are doing VoIP only. |
| 16:03.49 | DevilFish | yeah sip to asterisk and sip to metaswitch |
| 16:04.03 | DevilFish | oh, I see where you were headed now |
| 16:04.19 | VOIP_enthused | OK then if it's pass through, what's the high limit of SIP phones the PBX can handle on Pass through? |
| 16:04.33 | xkev | e.g. playing a menu, taking voice mail, will require transcoding 729 to slin, etc. (and then slin to gsm or slin to ulaw, or whatever * decides is most efficient) |
| 16:05.05 | xkev | voip_enthused you could probably throw hundreds concurrent at that box |
| 16:05.35 | xkev | but if you only need passthrough, you might as well just use SER and have it proxy the calls and keep the media stream between endpoints |
| 16:05.36 | DevilFish | ManxPower: do you no if an ethreal packet sniff yields more info than a standard asterisk sip debug? |
| 16:05.54 | VOIP_enthused | thousands? Has anyone done any benchmarking on the high limit scalabilty? |
| 16:05.58 | epoch | VOIP_enthused: it's really hard to come up with a useful number -- this is something that you can only really know by testing for your particular uses... |
| 16:06.37 | ManxPower | DevilFish, Yes, of course. but I don't know if the additional info is USEFUL. |
| 16:06.42 | epoch | VOIP_enthused: there is some stuff on benchmarks on voip-info.org -- I suggest looking it up |
| 16:06.43 | xkev | yeah, it really depends how much voodoo you cook up. asterisk isn't a static beast that can really be benchmarked, it all depends on what you do |
| 16:06.53 | xkev | ..with it. |
| 16:07.23 | DevilFish | ManxPower: yeah I suppose... any idea on the types of things that might stop an asterisk box from ACKing an invite? |
| 16:07.28 | epoch | man, I'm getting FTP connections once a minute from these polycom ip500s... I wonder why... |
| 16:07.33 | Skysky | hi, is there anybody tried phpconfig b4 arround? |
| 16:07.36 | xkev | epoch, new 1.4.1? |
| 16:07.37 | ManxPower | DevilFish, No. If I did I would have said something. |
| 16:07.38 | VOIP_enthused | I've seen other systems that claim 20,000 users, and 2000 concurrent sessions. Is it possible to build a system with * to scale to this number? |
| 16:07.40 | epoch | xkev: yeah |
| 16:07.48 | xkev | they poll periodically for new wares |
| 16:07.50 | epoch | xkev: though I think it was happening with 1.3.1 too |
| 16:07.54 | xkev | and refuse dto reboot on check-sync :) |
| 16:07.54 | ManxPower | epoch, They want to download their configs. |
| 16:08.07 | epoch | xkev: oh, really? |
| 16:08.08 | xkev | epoch, then it could be uploading their logs, etc |
| 16:08.15 | epoch | I"m not so sure |
| 16:08.21 | epoch | cuz they're not actually uploading anything |
| 16:08.26 | *** join/#asterisk amsterdam (~ak@xdsl-213-196-213-157.netcologne.de) |
| 16:08.28 | epoch | they're just connecting and disconnecting |
| 16:08.33 | *** join/#asterisk ckruetze (~ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 16:08.37 | amsterdam | hi |
| 16:08.44 | xkev | epoch, I have a 1.3.4 that responds to check-sync fine, but 1.4.1 only checks its sip.ld, etc. if I set it to always reboot on check-sync (whatever that option is) it checks a few more files, but never reboots |
| 16:09.02 | xkev | ..1.3.4 does a LIST, where 1.4.1 actually slerps the file and aborts it if it doesn't want it |
| 16:09.28 | xkev | epoch, run tethereal and see what they're actually doing |
| 16:09.29 | ManxPower | xkev, Just powercycle the phones. That's what I do. |
| 16:09.30 | *** join/#asterisk carbon60 (~adam@gw.techsupport.ca) |
| 16:09.35 | carbon60 | Morning all. |
| 16:09.55 | xkev | manx, but it's much nicer to sip notify polycom-check-cfg <list of everything> |
| 16:10.11 | ManxPower | xkev, of course. |
| 16:10.20 | amsterdam | does someone know where i can find information about snom 190 and asterisk ? |
| 16:10.26 | *** part/#asterisk Moc__ (~mochouina@64.235.210.66) |
| 16:10.26 | VOIP_enthused | Anyone pushing FAX through * with no problems? |
| 16:10.29 | epoch | xkev: yeah, doing that... |
| 16:10.31 | ManxPower | ~docs |
| 16:10.32 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 16:10.47 | ManxPower | VOIP_enthused, Don't expect fax to work to work via VoIP. It might work. Just don't expect it. |
| 16:10.49 | amsterdam | especially about sip url for call pickup ? |
| 16:11.05 | xkev | amsterdam, snom also has an asterisk integration pdf |
| 16:11.18 | xkev | amsterdam, um pickup? :) |
| 16:11.27 | amsterdam | yes pickup ... |
| 16:11.28 | *** join/#asterisk ckruetze_ (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 16:11.35 | carbon60 | I have a client whose main mailbox gets almost 100 messages a night. When checking those messages from a PSTN phone (via a SIP provider), the system becomes unresponsive to DTMF after going through approx. 20 messages. Any ideas where to look first? |
| 16:11.36 | xkev | * doesn't have parking orbits like snom expects |
| 16:12.22 | xkev | you are kinda stuck with the 700/7xx park/pickup thing |
| 16:12.25 | amsterdam | page 32 of the snome.com snom190 doc |
| 16:12.31 | epoch | ahhhhhh |
| 16:12.37 | amsterdam | snom.com... |
| 16:12.49 | epoch | xkev: you were right -- they're trying to upload logs... |
| 16:12.59 | xkev | but you have no perms on the dir? :) |
| 16:13.05 | epoch | xkev: the reason that I wasn't seeing anything in the xferlog was because they're getting a 451 ;) |
| 16:13.09 | epoch | ;yep |
| 16:13.09 | epoch | haha |
| 16:13.14 | xkev | been there :) |
| 16:13.16 | epoch | -e |
| 16:13.29 | epoch | whoa, hold on |
| 16:13.33 | epoch | the perms are right |
| 16:13.34 | xkev | make sure they can write their mac-directory.xml and mac-phone.cfg too |
| 16:14.00 | epoch | er, nm... not fs perms, ftp append perms |
| 16:14.07 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 16:14.15 | xkev | ain't secure-by-default a bitch :) |
| 16:14.31 | epoch | good old proftpd |
| 16:14.45 | VOIP_enthused | thanks all! You've been really helpful. Hope to contribute once we get our * platform up and running |
| 16:15.27 | xkev | voip_enthused, good luck; have fun. I've been working ours for 6 months, about to roll live monday |
| 16:15.37 | CosmicRay | I am confused about caller ID handling in Asterisk. Supposedly *67 from a phone disables caller ID... but I can't find references to that in any config files execpt zapata.conf. If my call is terminated to PSTN via an IAX link, how exactly to I tell asterisk that *67 disables caller ID? |
| 16:16.35 | xkev | dropping to pri? |
| 16:17.03 | xkev | pbx*CLI> show application setcallerpres |
| 16:17.04 | Nugget | if you want asterisk to emulate the pstn "*67" behavior, you'll have to put that in your dialplan. |
| 16:17.07 | Nugget | it's not inherent. |
| 16:17.10 | xkev | ..and that too |
| 16:17.17 | Nugget | but perhaps I misunderstand what you're asking |
| 16:17.33 | CosmicRay | Nugget: I think you do understand... but I haven't found any examples of exactly how to accomplish that |
| 16:17.45 | CosmicRay | Nugget: what about the other *xx features, like call waiting on/off, etc? |
| 16:17.51 | Nugget | you'll have to add dialplan entries for *67NXXNXXXXXX or whatever. |
| 16:17.51 | xkev | iax to a provider, or your own termination box |
| 16:18.04 | Nugget | and then set callerid to something else in those situations |
| 16:18.19 | Nugget | presuming your iax provider lets you determine your own callerid |
| 16:18.32 | CosmicRay | xkev: I'm thinking of a provider like voipjet |
| 16:19.25 | CosmicRay | Nugget: hmm. thta's a lot of extra dialplan entries, but ok. what about things like *78/*79 (enable/disable do not disturb)? |
| 16:19.32 | xkev | if they've implemented some way to use setcallerpres (which can block callerid, but still allow ani, essentially), then they'll have to provide you some target to dial that will turn that flag on. |
| 16:19.35 | CosmicRay | I can't find them in any config files either |
| 16:19.44 | xkev | otherwise, as nugget said, just setcidnum("") or something |
| 16:19.45 | Nugget | it's only a lot of entries if your dialplan is poorly designed. |
| 16:20.00 | gmcinnes | hi all. Any dell server users here? |
| 16:20.24 | *** join/#asterisk ikey (~kirankuma@202.54.37.186) |
| 16:20.27 | CosmicRay | Nugget: well, I would be routing long distance one way, toll-free another way, and local a third way. toll-free alone requires 4 entries.... |
| 16:20.34 | eKo1 | gmcinnes: many |
| 16:20.37 | pigpen | gmcinnes: depnds... |
| 16:20.43 | Nugget | 4 is not a lot. :) |
| 16:20.44 | ManxPower | Dialplans are complicated. It's as simple as that. |
| 16:20.49 | xkev | 800 is a lot |
| 16:20.52 | CosmicRay | heh |
| 16:20.57 | ikey | did any one worked with voicexml and asterisk |
| 16:21.00 | ikey | ? |
| 16:21.15 | Juggie | CosmicRay, i'll set you up for 50$ an hour :) |
| 16:21.16 | ManxPower | ikey, I don't think Asterisk supports VXML |
| 16:21.33 | CosmicRay | Juggie: pfft, I can run emacs myself, thanks :-) |
| 16:21.45 | Juggie | asterisk does not, but there is a openvxml that supports sip i beleive |
| 16:21.51 | Juggie | check out www.sipfroundry.org |
| 16:21.54 | CosmicRay | I guess I'm jsut surprised that this isn't in the sample config files |
| 16:22.04 | gmcinnes | eKo1: I have a poweredge 2600. I need a special cable to get ide power off the board for a tdm400p |
| 16:22.08 | CosmicRay | it papears that the zaptel config has some special support for it automatically, somehow? |
| 16:22.11 | Juggie | you are only doing pattern matching, its not that hard. |
| 16:22.17 | CosmicRay | I'm confused about how its *67 interacts with the rest of the system |
| 16:22.31 | Juggie | on the sip side *67 does nothing |
| 16:22.34 | gmcinnes | eKo1: Have you ever seen such a thing? Dell denies its existance :) |
| 16:22.37 | Juggie | its only if you have phones on a zap card |
| 16:22.39 | anthm | ok all done who is gonna fund the development |
| 16:22.41 | anthm | 'G(context^exten^pri)' -- If the call is answered transfer both party to the specified exten. |
| 16:23.03 | eKo1 | gmcinnes: Not sure. My Dell came preinstalled with a quad E1 already. |
| 16:23.17 | CosmicRay | Juggie: so I'm adding *67 to my extensions.conf to support the SIP side, but if I only cared about client phones on zap, I wouldn't need to? |
| 16:23.37 | Juggie | CosmicRay, i think *67 is part of chan_zap unless you override it via the dialplan |
| 16:23.39 | ManxPower | CosmicRay, chan_zap has support for a lot of features built into it. |
| 16:23.53 | Juggie | i've never used analog phones so i cant say for sure |
| 16:24.01 | Juggie | check features.conf and the wiki |
| 16:24.13 | Juggie | however, your dialplan is easy |
| 16:24.35 | Juggie | 3 main paths, the only one you need to do work for is the list of toll free numbers |
| 16:24.51 | CosmicRay | yeah, that makes sense. |
| 16:25.24 | Juggie | if you are going to have internal extensions, then dont forget you'll need to have people dial 8 or 9 or something to get out for a local call. |
| 16:25.31 | Juggie | unless u want dialplan crossover |
| 16:25.32 | CosmicRay | right. |
| 16:26.09 | ikey | is there any application addon which support speech to text and text to speech in asterisk |
| 16:26.21 | CosmicRay | Juggie: what's dialplan crossover? |
| 16:26.24 | Juggie | text to speech, yes. |
| 16:26.30 | ikey | ok |
| 16:27.02 | ikey | yeah but does it have accent change feature |
| 16:27.25 | Juggie | crossover may not be the right term, but say you have an internal extension 4534 |
| 16:27.30 | ikey | say UK english and US english have two different accents |
| 16:27.52 | ManxPower | Juggie, "pattern overlap" is the term I use. |
| 16:28.02 | Juggie | ManxPower, probally a more correct term. |
| 16:28.10 | Juggie | anyways, then someone dials a local number |
| 16:28.19 | Juggie | your pattern matching for local is NXXXXXX |
| 16:28.32 | Juggie | so, when you dial 4534, its going to wait |
| 16:28.37 | CosmicRay | right, gotcha. |
| 16:28.38 | Juggie | becaue it thinks it may see 3 more digits |
| 16:28.51 | anthm | no you can use regex too if you have HEAD |
| 16:28.56 | anthm | i mean now you |
| 16:29.00 | CosmicRay | so in that case, one might decide to make internal extensions start with 1 or sometime |
| 16:29.03 | gmcinnes | eKo1: hmm. ok. Thanks anyway. I may have to send the server back and get something else. |
| 16:29.06 | CosmicRay | s/sometime/something/ |
| 16:29.27 | ManxPower | CosmicRay, In the USA toll calls are dialed as 1NXXNXXXXXX |
| 16:29.37 | CosmicRay | right, so same problem. |
| 16:30.01 | ManxPower | There is a REASON most real PBXs require 9 (or 0) for an outside line. |
| 16:30.01 | CosmicRay | actually there are so few numbers that are a local call from my area that I know which prefixes are safe for extensions :-) |
| 16:30.07 | *** part/#asterisk ikey (~kirankuma@202.54.37.186) |
| 16:30.36 | tzanger | ManxPower: yeah they're fucking gayass pieces of shit |
| 16:30.46 | tzanger | I miss not having to dial 9 |
| 16:30.50 | Darwin35 | what ver of sphinx has been tested with * is it 2 ot 3 |
| 16:31.42 | *** part/#asterisk Alexis (~alexis@www.trim.it) |
| 16:32.10 | Conductor | monitor executing ( nice -n 19 soxmix "/var/spool/asterisk/monitor/test-in.wav" "/var/spool/asterisk/monitor/test-out.wav" "/var/spool/asterisk/monitor/test.wav" && rm -f "/var/spool/asterisk/monitor"/test-* ) & |
| 16:32.18 | Conductor | look at the last command |
| 16:32.25 | Conductor | rm -f "/var/spool/asterisk/monitor"/test-* |
| 16:32.41 | Conductor | the " are set at the wrong place... |
| 16:32.56 | anthm | xkev, http://66.250.68.190/eg/sample.txt |
| 16:32.56 | Conductor | is this a known bug? any workarounds? |
| 16:34.44 | xkev | anthm, when I said $100 I meant I'd take $100 and do it in 29 mins, not pay you :) |
| 16:34.53 | xkev | I dont' need it :) (yet anyway) |
| 16:35.20 | xkev | I think I should find something to send you regardless though |
| 16:35.44 | *** join/#asterisk randu (~randy@pool-70-16-112-36.scr.east.verizon.net) |
| 16:35.46 | Juggie | hah, what did you get him to do :P |
| 16:35.55 | xkev | how about a Cisco PA-8E? |
| 16:36.07 | TheBear | Ok I'm able to transfer calls from pstn -> snom1 -> snom2 with the CNF/TRAN button on the snom 200. anyone using the snom 200 with intercom=true ? |
| 16:37.21 | xkev | thebear, intercom=true? |
| 16:37.52 | randu | Good Morning Or Afternoon y'all :-) I have an asterisk box using broadvoice as incoming line. Every couple of days when I try to call the number I get a message saying, this user is busy leave a message at the tone. I then have to reboot asterisk box or sometimes just restart asterisk to get it working again. I have to do this every couple of days, any idea why? |
| 16:37.54 | *** join/#asterisk HuangDi (TheEmperor@218.111.49.132) |
| 16:37.56 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 16:38.10 | Conductor | anthm, how did you do this? |
| 16:38.40 | anthm | <PROTECTED> |
| 16:39.02 | Conductor | anthm, what is this G? |
| 16:39.04 | xkev | anthm, was that just a couple of async_gotos? |
| 16:39.23 | anthm | its an option to app_dial |
| 16:39.54 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 16:40.08 | Conductor | anthm, its not documented, is it? |
| 16:40.11 | TheBear | when I dial ext 20 which is set with exten => 20,1,SetVar(_VXML_URL=intercom=true) and exten => 20,2,Dial(SIP/2205) I get Forbidden: 20 on the snom LCD ? |
| 16:40.26 | anthm | it's only on my copy, I just coded it 5 min ago |
| 16:40.40 | Conductor | i understand... |
| 16:40.40 | xkev | conductor, you were the guy an hour ago that wanted to send both legs to a meetme, right? |
| 16:40.50 | Conductor | xkev, yes |
| 16:41.16 | TheBear | is there something else I'm missing ? |
| 16:41.22 | Conductor | xkev, i can do a workaround with agi and the manager... |
| 16:41.25 | Eight | xkev: Now that you mention it, I'd kinda like that too. |
| 16:41.31 | Conductor | xkev, but this is not very nice... |
| 16:41.44 | anthm | I need someone to fund to development effort required to make the patch and add it to mantis =D |
| 16:42.02 | *** join/#asterisk Remowylliams (~Mare@168.215.138.106) |
| 16:42.02 | xkev | anthm, I'm searching my office for something you might want |
| 16:42.07 | Conductor | what is mantis anyway? |
| 16:42.18 | xkev | http://bugs.digium.com/ |
| 16:42.21 | xkev | aka "the source" |
| 16:42.21 | puppet | ANyone here that can OP Panel? |
| 16:42.27 | Remowylliams | Good morning / afternoon all. |
| 16:42.33 | Nugget | when did "OP Panel" become a verb? |
| 16:42.37 | anthm | the place where you can dl the path while it waits like a bill on capitol hill to be added to asterisk |
| 16:42.38 | xkev | (in poor humor matrix terms) |
| 16:42.39 | Conductor | oh i see. the cvs server |
| 16:42.49 | xkev | not exactly |
| 16:43.09 | Conductor | a patch server? |
| 16:43.15 | xkev | check it out |
| 16:43.24 | xkev | bugtracker, but features go there too |
| 16:43.50 | Conductor | anthm, when will you upload it? |
| 16:44.05 | xkev | when one of us gives him $50 or equivalent crap :) |
| 16:44.16 | puppet | nugget: caused its Operator panel, and since the files are named op_xxxxx, OP Panel |
| 16:44.19 | xkev | I'm eyeing a 7960, but I'm not sure if I need it yet or not |
| 16:44.35 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 16:44.55 | puppet | xkev: of course u need it ;p |
| 16:45.25 | epoch | heh, nobody really *needs* a 7960 ;) |
| 16:45.35 | Nugget | puppet: so what the hell are you asking? "OP Panel" is still not a verb. |
| 16:46.04 | puppet | nugget: ahh saw now it was swenglish ;p can == knows ;D |
| 16:46.13 | Nugget | ahhh |
| 16:46.14 | Nugget | :) |
| 16:46.26 | Nugget | I've used it, but not much. |
| 16:46.39 | puppet | nugget having problems getting an 100% config on buttons, i cant redirect 100% |
| 16:47.13 | Remowylliams | I have tried to use X-light to connect to my asterisk server. I can connect with the G711w just fine but I'm being refused 488 when I try to connect with GSM. Can some one help point me in the direction for an answer. Also if someone can recommend a good windows client for Asterisk other than X-lite I'd like to know. |
| 16:47.37 | pigpen | Can anyone think of a reason why I would not want to use a Cisco ATA 186 for a FX0 with Asterisk? |
| 16:48.16 | Remowylliams | Pigpen Sorry I don't have any experience with either of those devices. |
| 16:48.30 | pigpen | cool. |
| 16:49.05 | anthm | remo, firefly is the easiest one you can jump back and forth between sip and iax with 1 raido button |
| 16:49.25 | TheBear | I'm getting chan_sip.c:8042 handle_request: Failed to authenticate user <sip:2202@192.168.2.15>;tag=017do3q in the * console, and yet sip show peers shows my sip phones ? |
| 16:49.26 | *** join/#asterisk Tili (~Tili@202-133-67-112-dialup.sat.net.pk) |
| 16:49.35 | Remowylliams | Anthm Very cool thank you I've not heard of it before. |
| 16:49.56 | Remowylliams | I'm using Asterisk@home 0.6 by the way. |
| 16:50.27 | anthm | http://www.virbiage.com/firefly/download/firefly-thirdparty.exe |
| 16:50.58 | TheBear | how can a SIP phone (snom 200) suddenly go from status OK to status forbidden ? |
| 16:51.19 | ruiner | changes in sip.conf? |
| 16:51.47 | Zeeek | ruiner - got it working? |
| 16:51.54 | ruiner | no :( |
| 16:51.59 | Zeeek | sorry I mentioned it ;) |
| 16:52.05 | Zeeek | "it" |
| 16:52.06 | ruiner | I need to find an example config for my Cisco, I think |
| 16:52.11 | ruiner | I'm pretty sure that's the problem |
| 16:52.17 | ruiner | But, I am learning other things at least |
| 16:52.22 | Zeeek | wouldn't there be a Crisco user community somewhere? |
| 16:52.40 | ruiner | I should just convince my boss that we should just use * boxes everywhere instead |
| 16:52.40 | Zeeek | unenet, maybe ? |
| 16:52.47 | ruiner | Zeeek: I could check, yeah |
| 16:52.47 | Zeeek | Usenet |
| 16:53.01 | ruiner | I loathe usenet, though |
| 16:53.02 | ruiner | heh |
| 16:53.30 | Zeeek | see, on the Usenet dealies, you got the crusty old guys that have used cisco for 25 years and know every magic signet ring and handshake (but watch out for the net nazis) |
| 16:53.46 | Zeeek | PLEASE DO NOT TOP POST!!!! |
| 16:54.21 | Nugget | http://slacker.com/~nugget/stuff/circular.txt <-- usenet |
| 16:54.22 | Zeeek | Usenet is like recorded IRC |
| 16:54.38 | tzanger | top posting sucks |
| 16:54.40 | Zeeek | heh right away wioth the PDP |
| 16:54.42 | TheBear | ok came right by itself no changes nothing just waited a few minutes ????? weird ? |
| 16:55.02 | Zeeek | all posting sucks |
| 16:55.29 | *** join/#asterisk file (~file@251.134.218.209.transedge.com) |
| 16:55.36 | tzanger | haha |
| 16:55.43 | pigpen | TheBear: network issues? |
| 16:55.49 | Zeeek | why don't we start quoting on IRC |
| 16:55.49 | ruiner | so is this festival thing pretty cool? |
| 16:56.01 | Nugget | festival sucks. |
| 16:56.06 | TheBear | pigpen: don't know why ?? |
| 16:56.07 | Nugget | I have no idea how people tolerate it. |
| 16:56.18 | Nugget | it sounds like a speak and spell |
| 16:56.19 | CosmicRay | because there is no readily-available better alternative? |
| 16:56.20 | Zeeek | ruiner there is a problem with all that stuff |
| 16:56.30 | TheBear | once I make a call to console/dep how do I pass this call/message to active phones ? |
| 16:56.34 | xkev | when meetme prompts for a name to record, is that the 'i' option, or the 'T' doing that? |
| 16:56.37 | Zeeek | the average caller really don't want to hear that |
| 16:58.05 | puppet | ruiner: its cool CAN be usefull, but i used it to the most useless thing reading bash.org quotes in phone ;D |
| 16:58.20 | ruiner | haha |
| 16:58.27 | ruiner | so it's just a text to speech type thing? |
| 16:58.32 | puppet | yeah |
| 16:58.43 | ruiner | so it sounds like ass probably |
| 16:58.44 | ruiner | ? |
| 16:58.51 | puppet | ruiner: u can call and check at me? |
| 16:59.10 | CosmicRay | hell, it sounds ilke ass even when not being played over a telephone. |
| 16:59.28 | ruiner | yeah, give me digits |
| 16:59.38 | ruiner | or IP or whatever |
| 16:59.50 | puzzled | CosmicRay: cool. i just compiled it on ppc |
| 17:00.00 | puppet | oh got to add it in extensions |
| 17:00.50 | Zeeek | ruiner what was that cisco model number? |
| 17:00.57 | ruiner | Zeeek: 3640 |
| 17:01.03 | Zeeek | recent? |
| 17:01.05 | Zeeek | I guess |
| 17:01.12 | CosmicRay | puzzled: there were a lot of scary x86-only warnings on the wiki, but as of yet, I haven't run into any arch problems |
| 17:01.14 | ruiner | how do you mean? |
| 17:01.17 | Remowylliams | Sorry work got me :) |
| 17:01.20 | CosmicRay | puzzled: we will see when my x100d card arrives, tho |
| 17:02.28 | puzzled | CosmicRay: the only thing I bumped into was that the $(PROC) stuff in codecs/lpc10/Makefile doesn't work and chose -marchi386 on a ppc |
| 17:04.37 | Zeeek | ruiner - there's a lot of stuff about that router but nothing at all about FXO or voip |
| 17:04.44 | ruiner | yeah, i know |
| 17:04.56 | puzzled | a 3640 is ancient afaik |
| 17:05.00 | Zeeek | like hierarchical token buckets |
| 17:05.21 | Zeeek | is this a plug in card that gives those interfaces? |
| 17:05.29 | ruiner | yeah |
| 17:05.33 | Zeeek | aha |
| 17:05.39 | ruiner | i'm trying to remember what it's called |
| 17:05.43 | ruiner | VIC-FXO2 or something |
| 17:05.46 | Zeeek | I have a problem with router 3640 - when I set transport input to ssh on |
| 17:05.46 | Zeeek | line vty 0 4 I can't connect to router through tacacs server |
| 17:05.49 | Zeeek | " I have a problem with router 3640 - when I set transport input to ssh on" |
| 17:06.16 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 17:06.21 | Zeeek | ruiner this: VIC-2FXO-M1 card |
| 17:06.45 | Zeeek | check out comp.dcom.sys.cisco |
| 17:06.55 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 17:06.57 | Zeeek | I haven't seen a single top posting complaint yet |
| 17:07.17 | ruiner | what's top posting? |
| 17:07.41 | Zeeek | posting above the quote |
| 17:07.44 | Nugget | people who reply to emails or usenet articles with their text on top and the quote below. |
| 17:07.45 | ruiner | oh |
| 17:07.57 | Zeeek | which is more common in email |
| 17:08.17 | Nugget | http://mailformat.dan.info/quoting/ <-- an excellent, excellent page on the subject |
| 17:08.18 | Zeeek | unedited quoting is irritating as hell, top or bottom |
| 17:08.28 | Darwin35 | ok sphinx installs on fbsd and shuld work fine |
| 17:08.43 | Darwin35 | festival is going to take soome major work |
| 17:08.53 | zipp | I much prefer top posting |
| 17:09.02 | Nugget | zipp: read that page. |
| 17:09.04 | Zeeek | apparently you can plug the FXOx2 card into a lot of their routers |
| 17:09.19 | Juggie | Darwin35, festival isnt hard... just use the latest festival and use the text2wave tool |
| 17:09.31 | Darwin35 | this is on fbsd |
| 17:09.36 | Darwin35 | not linux |
| 17:09.47 | Juggie | does festival compile? |
| 17:10.04 | Darwin35 | not yet I am working on patching it |
| 17:10.29 | Darwin35 | sphinx-2 compiles and installs |
| 17:10.34 | Darwin35 | thats a good thing |
| 17:10.54 | Juggie | Yah, i never got around to trying that, did you get it working in a dialplan? |
| 17:11.11 | Darwin35 | well I just installed it |
| 17:11.21 | Darwin35 | have to read about it next in the wiki |
| 17:11.24 | Pinhole | has anybody done anything useful with sphinx2? |
| 17:11.31 | TheBear | in the snom web interface you can set auto answer yes/no. can this be done for a single line ? |
| 17:11.39 | file | ohhhhhhhhhh say can you seeeeeeeeeee |
| 17:11.41 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 17:11.42 | file | down in the row double zeeeeeeee |
| 17:12.20 | Darwin35 | its the only ver in my ports tree |
| 17:12.35 | puppet | yeah |
| 17:12.37 | Darwin35 | so I have to use it sphinx3 is otu I know but I hear it is worse |
| 17:12.39 | puppet | skip patching and shit |
| 17:12.43 | puppet | go with AGI instead |
| 17:12.51 | Darwin35 | ? |
| 17:13.05 | puppet | use AGI instead of patching festival :) |
| 17:13.20 | puppet | then u can use 1.95 of festival to |
| 17:13.20 | Juggie | yah, thats what i did |
| 17:13.20 | Juggie | wrote a perl script for it |
| 17:13.24 | Pinhole | festival or swift work very well without patching from agi. |
| 17:13.35 | Juggie | whats swift? |
| 17:13.35 | puppet | pinhole: yeah it does |
| 17:13.45 | puppet | i started to wonder that now to ;p |
| 17:13.48 | Darwin35 | well at the min festival needs patching for fbsd |
| 17:13.56 | Darwin35 | 195 does not compile |
| 17:13.59 | Pinhole | it's another tts, quality is much better, but its not free. |
| 17:14.43 | Pinhole | also known as cepstral |
| 17:14.46 | puppet | aha cepstral |
| 17:14.54 | Zeeek | I have cepstral |
| 17:15.01 | puppet | Cepstral sounds like a drug tho |
| 17:15.07 | Zeeek | it's good for $30 but not great |
| 17:15.14 | Darwin35 | yeah but it is not open src so I cant port it to fbsd |
| 17:15.25 | Zeeek | there is one tts that is great - they have a demo somewhere |
| 17:15.26 | Darwin35 | that means linux emu |
| 17:15.34 | puppet | zeeek: better then festival? |
| 17:15.34 | Pinhole | From what I understand, you have to have a license for each simultaneous voice. |
| 17:15.36 | Zeeek | but it is way expensive |
| 17:15.49 | Juggie | AT&T is the best one |
| 17:15.56 | Zeeek | Pinhole not simultaneos, just one for each voice |
| 17:15.58 | puppet | juggie: cost to? |
| 17:16.01 | Juggie | no idea |
| 17:16.04 | Pinhole | If you chop off the first 51000 from the demo, you get a non-demo ^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H |
| 17:16.26 | Juggie | just look up AT&T tts on google you'll find it |
| 17:16.28 | Darwin35 | what is the best text2wav tool |
| 17:16.43 | Darwin35 | is it text2wav if so niot in the ports |
| 17:17.02 | puppet | pinhole: chop of the first 51000? :) |
| 17:17.08 | Juggie | test2wav is a part of festival |
| 17:17.19 | Darwin35 | ahh ok |
| 17:17.36 | Darwin35 | well I will spend time working on 195 |
| 17:17.38 | Pinhole | When you stream file the output of swift, use 51000 as the offset and the demo message is gone. |
| 17:17.42 | Hmmhesays | should a sip bye message be sent when you use hangup in the dialplan? |
| 17:17.44 | Darwin35 | trying to get it working |
| 17:17.47 | puppet | pinhole: oh |
| 17:17.50 | Darwin35 | shower time |
| 17:18.04 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net) |
| 17:18.22 | puppet | pinhole: is it better then festival? |
| 17:18.24 | Pinhole | Please don't use it if you don't pay. I used it to show my boss the difference in quality without the annoying message. he didn't see fit to spend $$$ |
| 17:18.34 | Goshen | ok, so I am on a call, and another call comes in, how does Asterisk handle call waiting? just flash like the telco? |
| 17:18.39 | Pinhole | I think it is better than festival. There are choices of voices too. |
| 17:18.49 | Goshen | I am on hold right now, and had a call come in, and heard it beep |
| 17:19.15 | Juggie | festival has many voices too if you can figure out how to use them |
| 17:19.26 | Juggie | Goshen, your sip phone has lines |
| 17:19.41 | Juggie | * will send calls until your sip phone has no more lines |
| 17:19.47 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-212-218.dsl.scarlet.be) |
| 17:20.18 | SexyKen | Hey guys, How am I supposed to know which calls belong to who? For instance, in an IVR setup. Someone calls the Toll-Free #asterisk and gets the IVR Context. They then select 1. They get directed to sales. CDR makes this 2 calls, if the caller is on hold, they make it even more calls..... |
| 17:20.27 | SexyKen | ....how do I know it's from the same originiating call? |
| 17:23.21 | pigpen | Question: How would faxing be handled by * ? Or would I just load up hylafax and have it email the fax to the end user? |
| 17:23.44 | zipp | pigpen, t.38 isn't supported |
| 17:23.57 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 17:24.02 | Hmmhesays | it seems asterisk doesn't send any sip messages out when you call hangup |
| 17:24.02 | pigpen | so fax would be a seperate project? |
| 17:24.23 | Zeeek | pigpen check this out: http://scottstuff.net/scott/archives/cat_asterisk.html |
| 17:24.32 | pigpen | thanks. |
| 17:24.46 | Zeeek | it great piece of work |
| 17:25.19 | Zeeek | asterisk can emailt he fax without hylafax btw |
| 17:25.20 | carbon60 | SexyKen: I think it's suppose to be a single CDR entry. What version? |
| 17:25.31 | puppet | pigpen: http://freshmeat.net/projects/astfax/ |
| 17:25.37 | carbon60 | I have a client whose main mailbox gets almost 100 messages a night. When checking those messages from a PSTN phone (via a SIP provider), the system becomes unresponsive to DTMF after going through approx. 20 messages. Any ideas where to look first? |
| 17:26.20 | Zeeek | tell her to get more lines! |
| 17:26.25 | SexyKen | carbon60 CVS-HEAD-03/09/05-01:20:42 |
| 17:26.27 | Zeeek | (or less clients) |
| 17:26.39 | carbon60 | Don't know then. |
| 17:26.51 | carbon60 | SexyKen: But I get a single CDR entry for most calls, I think. |
| 17:27.09 | pigpen | Zeeek: so would I need any special hardware? |
| 17:27.35 | Zeeek | pigpen, take a loog at the wiki there's plenty of ink about all that |
| 17:27.54 | Zeeek | http://www.voip-info.org/wiki-Asterisk+spandsp |
| 17:28.07 | pigpen | k |
| 17:28.08 | pigpen | thanks. |
| 17:28.13 | Zeeek | np |
| 17:28.24 | *** join/#asterisk Sedorox (~Sed@pcp01339110pcs.wilog101.pa.comcast.net) |
| 17:28.25 | pigpen | oh...sweet. |
| 17:29.03 | ruiner | ooh yeah, just got another router thrown into the mix with another vic2fxo card |
| 17:29.05 | ruiner | yay! |
| 17:29.21 | Zeeek | now to get either one to work... |
| 17:29.26 | ruiner | no kidding |
| 17:29.50 | Zeeek | how about looking up that FXO card in google groups |
| 17:29.55 | Katty | Zeeek: Zeeek Zeeek Zeeek |
| 17:29.58 | ruiner | what we eventually want to do is have a router in every city we provide access to (we're an ISP) and have the FXO cards in it |
| 17:30.04 | Conductor | anthm, when will you upload it? |
| 17:30.18 | anthm | upload what? |
| 17:30.24 | ruiner | we want to roll out a program where we can sell service to our broadband customers and let them dial any city we provide access in for a fixed fee per month |
| 17:30.25 | Katty | lunctime! |
| 17:30.27 | Zeeek | {{{{Katty}}}} |
| 17:30.28 | Katty | i mean lunch |
| 17:30.31 | Katty | gosh |
| 17:30.33 | Zeeek | me? |
| 17:30.38 | Katty | all, ripply wavey effect |
| 17:30.38 | Sedorox | hmmmmm |
| 17:30.39 | *** part/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 17:30.41 | ariel_ | hello Katty yes it is lunch time. |
| 17:30.45 | Katty | hi ariel! |
| 17:30.47 | Katty | bye ariel! |
| 17:30.58 | ruiner | so instead of putting an asterisk box in each city, we'll just get an few $80 fxo cards for our routers |
| 17:31.28 | Zeeek | ruiner the idea is good - now get it to work |
| 17:31.35 | ruiner | yeah no doubt |
| 17:32.00 | shido6 | then all the other backend systems for provisioning and communicating with your own team |
| 17:32.05 | shido6 | billing |
| 17:32.26 | ruiner | yeah, it's going to be a big headache |
| 17:32.26 | shido6 | paper billing / electronic billing |
| 17:32.36 | shido6 | good luck with that |
| 17:32.40 | ruiner | well the billing isn't going to be such a big thing |
| 17:32.52 | ruiner | we already have a billing system in place for our customers anyway, we'll just add a charge to their account |
| 17:33.06 | Zeeek | ruiner check this: |
| 17:33.10 | Zeeek | http://groups.google.fr/groups?hl=fr&lr=&client=firefox-a&rls=org.mozilla:en-US:official&threadm=760aba4d.0407022355.7552ca30%40posting.google.com&rnum=2&prev=/groups%3Fq%3Dfxo%26hl%3Dfr%26lr%3D%26group%3Dcomp.dcom.sys.cisco%26client%3Dfirefox-a%26rls%3Dorg.mozilla:en-US:official%26selm%3D760aba4d.0407022355.7552ca30%2540posting.google.com%26rnum%3D2 |
| 17:33.14 | Zeeek | http://groups.google.fr/groups?hl=fr&lr=&client=firefox-a&rls=org.mozilla:en-US:official&threadm=760aba4d.0407022355.7552ca30%40posting.google.com&rnum=2&prev=/groups%3Fq%3Dfxo%26hl%3Dfr%26lr%3D%26group%3Dcomp.dcom.sys.cisco%26client%3Dfirefox-a%26rls%3Dorg.mozilla:en-US:official%26selm%3D760aba4d.0407022355.7552ca30%2540posting.google.com%26rnum%3D2 |
| 17:33.16 | Zeeek | <PROTECTED> |
| 17:33.23 | puppet | spam ;p |
| 17:33.55 | Zeeek | <PROTECTED> |
| 17:33.58 | Zeeek | <PROTECTED> |
| 17:34.03 | Nivex | Zeeek: tinyurl!!!! |
| 17:34.06 | Zeeek | can't paste too long |
| 17:34.12 | Nugget | ow |
| 17:34.13 | tzanger | dammit pastebin that shit :-) |
| 17:34.14 | Zeeek | oooops |
| 17:34.17 | tzanger | pastebin for a URL |
| 17:34.20 | Nugget | use lnk.nu for that |
| 17:34.33 | Zeeek | The window was screwed up I didn't see it wokred |
| 17:34.39 | Zeeek | a thousand pardons! |
| 17:35.10 | puppet | haha ;o) |
| 17:35.18 | Zeeek | Let's see: http://lnk.nu/groups-beta.google.com/1qc |
| 17:35.20 | Nugget | http://lnk.nu/groups.google.fr/1qd |
| 17:35.42 | Zeeek | OMG I shit all over #asterisk.... |
| 17:35.57 | Zeeek | your fault, ruiner |
| 17:36.08 | ruiner | haha |
| 17:36.08 | Nugget | ruiner ruined everything! |
| 17:36.25 | Zeeek | anyway this is the path to enlightenment - there's a few threads about FXO |
| 17:36.32 | ruiner | they don't call me ruiner for nothing |
| 17:36.36 | puppet | hahaha |
| 17:36.39 | ruiner | Zeeek: I appreciating |
| 17:36.40 | Zeeek | the holy grail has to be in there somewhere |
| 17:36.42 | ruiner | er, appreciate it |
| 17:36.50 | Zeeek | Appreciate THIS |
| 17:37.03 | puppet | *reads THIS* |
| 17:37.05 | Remowylliams | Well firefly isn't being my friend anthm. fialed to network 200 (408) |
| 17:37.44 | Zeeek | that thread was worth finding - it's answered by the grisly old cisco engineer |
| 17:38.35 | puppet | does there exist any asterisk place on usenet? |
| 17:39.23 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 17:39.47 | *** join/#asterisk mesi (~player@dsl-082-083-150-235.arcor-ip.net) |
| 17:41.00 | mesi | About the topic... what happend when I call that IAX number? |
| 17:41.21 | Zeeek | you have the privilege of listening to a brilliant team of specialists |
| 17:41.31 | Zeeek | talking about Mark |
| 17:41.32 | mesi | And can I talk? |
| 17:41.36 | Zeeek | Nooooooo |
| 17:41.44 | mesi | Ah, ok. |
| 17:41.46 | mesi | :-) |
| 17:41.49 | Zeeek | the good side, you can belch and fart |
| 17:41.52 | mesi | I'm calling... |
| 17:41.56 | Zeeek | they can't hear you |
| 17:42.12 | mesi | Yes, that's right. But on the other hand... how can I be SURE they don't hear me? |
| 17:42.19 | Zeeek | you can't really |
| 17:42.30 | Zeeek | I think Mark listens to them talking about him, too |
| 17:43.24 | JunK-Y | mesi: ya'll be notify if you're muted. |
| 17:43.37 | mesi | Hm... There's a conference there! |
| 17:43.46 | Zeeek | but how does he know it isn't a fake mute message |
| 17:43.48 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 17:43.59 | Zeeek | Dev Conf |
| 17:44.00 | bannerman | Geez, never use TelIAX. |
| 17:44.12 | JunK-Y | call 2 times and scream like hell when you're muted, ya'll seee. |
| 17:44.15 | mesi | Zek, junk: It's a conference there. Nobody is online. |
| 17:44.24 | bannerman | Fortunately I'm just doing testing and configuration right now, not using this for production, but my 888 number now goes to some recording in Hebrew. |
| 17:44.40 | Zeeek | convenient if you are in Egypt |
| 17:44.57 | JunK-Y | mesi: i know its at 1pm. |
| 17:45.35 | mesi | junk: Ah, I see. I am in a conference where I am muted. :-) |
| 17:45.43 | mesi | junk: But there was no mute message. Anyway. |
| 17:45.54 | Zeeek | you are conferring |
| 17:46.06 | Zeeek | but not conferencing |
| 17:46.13 | mesi | zeeek: Yes, I get it now. It is an open conference room for some developers or so. |
| 17:46.26 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 17:46.53 | JunK-Y | mesi: huH? |
| 17:47.01 | Zeeek | yes but it is open to the public audience and can be interesting |
| 17:47.06 | Sedorox | Does mpg123 have a problem with SMP machines? |
| 17:47.17 | Zeeek | plus t's a good test of your voip setup if you listen for like 2 hours |
| 17:47.27 | mesi | junk: Yes, there is no mute message. Only that I am the only one is said :-) |
| 17:47.36 | JunK-Y | mesi: come back |
| 17:47.43 | JunK-Y | just tried it. |
| 17:47.43 | Zeeek | well if you talk to yourslef you won't know you're muted |
| 17:47.45 | mesi | Zeek: Yes, I will test this :-) |
| 17:48.01 | Essobi | WAAAAAASAAAABIIIII |
| 17:48.03 | *** join/#asterisk afe ([kZT0x7ttI@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 17:48.04 | *** join/#asterisk roamer323 (~sing@Toronto-HSE-ppp3680763.sympatico.ca) |
| 17:48.07 | Zeeek | I listened tot ha last one for almost 90 minutes |
| 17:48.18 | Zeeek | had some wasabi yesterday |
| 17:48.20 | JunK-Y | Zeeek: are ya in? |
| 17:48.23 | Zeeek | no |
| 17:48.28 | mesi | zeeek: I can take a second handset and call this conference. Then talk a bit and when I can hear myself, I am NOT muted ;-) |
| 17:48.48 | Zeeek | maybe it's only muted when the REAL guys get there |
| 17:48.54 | mesi | junk: Yes, I am in. |
| 17:49.14 | *** join/#asterisk emitrax (~emitrax@host209-51.pool80181.interbusiness.it) |
| 17:49.17 | emitrax | hi |
| 17:49.30 | JunK-Y | moooo |
| 17:49.46 | mesi | Ok, there's somebody there now. |
| 17:49.51 | JunK-Y | yes |
| 17:49.52 | mesi | He's typing on his keyboard :-) |
| 17:49.55 | JunK-Y | hehehe |
| 17:49.58 | Zeeek | it's you |
| 17:50.10 | Essobi | Someone testing a conference? |
| 17:50.11 | Essobi | :) |
| 17:50.17 | JunK-Y | ya, ites me |
| 17:50.20 | JunK-Y | its me |
| 17:50.24 | *** part/#asterisk emitrax (~emitrax@host209-51.pool80181.interbusiness.it) |
| 17:50.41 | mesi | junk: say something! |
| 17:50.46 | JunK-Y | booo |
| 17:50.47 | Essobi | I got like 5 phones sitting next to me.. Want a lil purple haze feedback? :) |
| 17:50.48 | mesi | junk: say: hello or so ;-) |
| 17:51.04 | mesi | Ok, go for it. |
| 17:51.57 | Remowylliams | Well Firefly is up for iax2 but I'm going to have to tinker with sip it seems. |
| 17:52.13 | Zeeek | I never got SIP working on FF |
| 17:52.21 | Essobi | FF? |
| 17:52.28 | Zeeek | FireFlop |
| 17:52.29 | Remowylliams | Zeeek thanks for the feedback. |
| 17:52.34 | Essobi | oh fire |
| 17:52.37 | Zeeek | but that was way back |
| 17:52.37 | Ayano | has anyone ever tried to use the xpl connector in asterisk@home? |
| 17:52.43 | Essobi | it crashed my machine the last time I installed it |
| 17:53.02 | Ayano | essobi: xpl? |
| 17:53.10 | Essobi | fireflop |
| 17:53.18 | Zeeek | FF is decent in IAX, esp good to give to people who don't care about configuring clients |
| 17:53.18 | Ayano | oh, |
| 17:53.42 | Essobi | hows that? |
| 17:53.53 | Zeeek | wait for the IAX hardphone their building... shipping in a few weeks - for the last year literally |
| 17:53.54 | Essobi | I never messed with IAX and/or firefly |
| 17:54.01 | Remowylliams | I know alot depends on network congestion and cpu power.. But I couldn't find any solid requirements or recommendations for how much CPU and all was needed for Asterisk. Currenly I've got it running on a 300 Mhz PII with 160 Megs of ram |
| 17:54.26 | Zeeek | voip only, one channel, no prob |
| 17:54.40 | Essobi | Remowylliams Oh.. go look up dimensioning on voip-info |
| 17:54.56 | Essobi | I don't think anyone has any "how small" diminsioning, but rather "how big" |
| 17:54.57 | Remowylliams | Dimentioning? |
| 17:54.59 | Zeeek | I think people have used PentiumI-90 witough MM |
| 17:55.04 | Zeeek | dimentai ! |
| 17:55.08 | Zeeek | dimentia |
| 17:55.24 | lucca | dementia, you mean :p |
| 17:55.26 | Essobi | I want a mini-itx I can put 2-4 FXOs on. |
| 17:55.41 | Zeeek | ya, thanks - must be the drugs I take to keep it under control |
| 17:55.43 | Essobi | FXS would be nice too |
| 17:55.50 | Essobi | SHHH! |
| 17:55.53 | Remowylliams | can I get a url for voip-info ? |
| 17:55.54 | Darwin35 | cool * now loads no warnings |
| 17:56.04 | Zeeek | .org |
| 17:56.05 | Remowylliams | Never mind |
| 17:56.06 | Essobi | Tell your sweater to stop talking to me. The red speaks to loud. |
| 17:56.24 | Darwin35 | everyone time out |
| 17:56.32 | Darwin35 | take your chill pils |
| 17:56.35 | Darwin35 | pills |
| 17:56.40 | Zeeek | go register with a register that isn't a register |
| 17:56.43 | Remowylliams | I've been getting some snapping and crackling with my setup here and there |
| 17:56.47 | Darwin35 | and a sip of the martinie |
| 17:56.58 | Zeeek | look up RiceKrispies on the wiki |
| 17:57.11 | Zeeek | you'll need hardware for the Pop |
| 17:57.22 | Essobi | Do what what? |
| 17:57.24 | Zeeek | [or ztdully] |
| 17:57.32 | Essobi | heh |
| 17:57.33 | Remowylliams | Zeeek thanks on the RiceKrispies |
| 17:57.56 | Essobi | there seriously isnt an entry for RiceKrispies |
| 17:57.57 | Zeeek | Remo it may be the small footprint of the system or a lot of other things |
| 17:58.05 | Zeeek | there should be |
| 17:58.11 | Essobi | heh |
| 17:58.12 | Zeeek | and also "bad cellphone quality" |
| 17:58.41 | *** join/#asterisk KrimHum (~barry@mercury.santabarbararealty.net) |
| 17:58.43 | Remowylliams | I'm not swapping and watching the load it's mostly idle. I'll look around some more. |
| 17:59.19 | Essobi | what's your setup Remo? |
| 17:59.30 | Essobi | "snapping and crackling" tends to come from analog |
| 17:59.39 | mesi | Zeeek: Great conference room :-) |
| 17:59.48 | Zeeek | ok, wait a second.... |
| 17:59.52 | Zeeek | ... use anal-safe toys; they're marked in the GV catalog and web site with an |
| 17:59.52 | Zeeek | asterisk. . |
| 18:00.00 | Essobi | where as, echo and ev-er-y-wo-r-d-ju-mp-s tends to be voip |
| 18:00.02 | Zeeek | c'mon guys |
| 18:00.51 | Remowylliams | Asterisk@home 0.6, on a 300 Mhz PII with 160 megs of ram sitting on a 100 Mbit lan, I've DSL for my internet 1.3 Mbit / 320 Kb |
| 18:00.51 | puzzled | anyone know a g.729 codec for 32 bit PPC? |
| 18:01.11 | Essobi | So what're you using for phones? |
| 18:01.23 | Essobi | puzzled Mac? |
| 18:01.38 | Remowylliams | The finest headset and boom mike I can find just now. :) |
| 18:01.51 | puzzled | Essobi: RS6000 |
| 18:02.05 | Essobi | Essobi Umm.. I don't know if any of the pirated once of a PPC but look at the wiki and check around |
| 18:02.12 | Essobi | s/once/ones |
| 18:02.26 | puzzled | the pirated ones are only for intel cpu's |
| 18:02.27 | Essobi | Remowylliams Analog sound card? |
| 18:02.39 | Essobi | puzzled Welp.. get to compiling one then. |
| 18:02.48 | Essobi | Good luck on that too. ;) |
| 18:02.55 | puzzled | heheh |
| 18:03.02 | Essobi | WTF are you running * on? |
| 18:03.06 | Essobi | SGI? |
| 18:03.11 | JunK-Y | i wonder is there any standard concerning the PDD ? |
| 18:03.13 | puzzled | IBM RS/6000 43p-150 |
| 18:03.18 | Essobi | Ah, nice. |
| 18:03.20 | Remowylliams | Yes for my extention |
| 18:03.28 | Essobi | Remowylliams Umm. Ther'es your answer. |
| 18:03.38 | Essobi | Your sound card is shite, or the headset is. |
| 18:04.00 | Remowylliams | This is audio being played in my conference room |
| 18:04.02 | KrimHum | FYI, the new asterisk ebuild for gentoo doesn't seem to work with the hardened USE flag set. the gsm codec doesn't install correctly. |
| 18:04.10 | Remowylliams | Or audio comeing to me from the ivr |
| 18:04.15 | Essobi | KrimHum Gentoo is for ricers. |
| 18:04.28 | KrimHum | I drive a 78 Impala |
| 18:04.41 | Essobi | Yet, you want to rice up your linux install. |
| 18:05.00 | Remowylliams | It's not horrible mind you. thanks for the feedback though |
| 18:05.01 | Essobi | I remember the 78 imp.. looks like a box. |
| 18:05.09 | Essobi | Remowylliams NP. |
| 18:05.10 | angler_ | compile everything for that little extra horsepower |
| 18:05.17 | KrimHum | No, I want source installs like BSD with better package management, but I've seen the web site you're obviously quoting. |
| 18:05.31 | *** part/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
| 18:05.37 | Essobi | WOOOO! I GOT 2% more CPU BY OVER OPTIMIZING EVERYTHING! |
| 18:05.47 | Essobi | KrimHum I've got chuck tattooed on my left arm. |
| 18:05.50 | Essobi | :) |
| 18:05.54 | angler_ | Essobi, lol |
| 18:05.56 | Essobi | Go run freebsd. |
| 18:05.59 | Essobi | Be happy. |
| 18:06.29 | KrimHum | err. straw man. wrong logical fallacy. |
| 18:06.51 | PTG123 | If you want something like freebsd, why not just run freebsd? :) |
| 18:07.01 | Essobi | Exactly my point. |
| 18:08.04 | Essobi | I just don't see the point in recompiling my entire ports tree every two weeks, trying to keep up with package updates, that are irrelevent, and present creeping problems in the OS, LIBs and Packages. |
| 18:08.06 | Godsey | because freebsd blows :) |
| 18:08.12 | Essobi | MAhaha. |
| 18:08.28 | KrimHum | I never ran FreeBSD, but I ran OpenBSD for a while. |
| 18:08.35 | Godsey | bsd people critisize linux for being bastardized mix of sysv and bsd |
| 18:08.39 | Godsey | where freebsd is no different |
| 18:08.40 | KrimHum | In the end, I liked Linux better, but appreciated ports just the same. |
| 18:08.42 | Godsey | only not as good |
| 18:08.59 | Essobi | I think Yahoo, IMDB, cdrom.com, and slashdot would protest that statement. |
| 18:09.51 | KrimHum | If they made a movie for the OS Wars, everyone would be on the Dork Side. |
| 18:09.59 | Essobi | In the end, FreeBSD has a few things more down pat, and more robust for serious load, and packets per second. |
| 18:10.07 | Godsey | lies |
| 18:10.13 | Essobi | Is it? |
| 18:10.17 | Godsey | solaris is the clear leader for network and disk io |
| 18:10.22 | Juggie | fbsd is about 10x better packet wise then linux |
| 18:10.27 | Essobi | :) |
| 18:10.43 | Essobi | Gods who wais anything about solaris? |
| 18:10.46 | Essobi | Said.. |
| 18:10.50 | Juggie | which would stand to reason that a fbsd sip server would wipe the ass of linux |
| 18:10.57 | Juggie | should be able to do many more c alls |
| 18:11.10 | Essobi | Go take your pills, and come back when the voices stop talking to you. :) |
| 18:11.17 | puppet | PII 233 / 128mb ram |
| 18:11.21 | Essobi | Heh. |
| 18:11.21 | puppet | to little to run asterisk? |
| 18:11.25 | Godsey | I don't think you can find any benchmarks which rate fbsd 5.x over linux 2.6 for network io |
| 18:11.28 | Zeeek | 49 minutes and the real guys will start |
| 18:11.38 | PTG123 | FreeBSD is 10x better for everything |
| 18:11.42 | KrimHum | Little voices talk to me every time I enter IRC. |
| 18:11.45 | Juggie | puppet, should be ok for a few calls. |
| 18:11.46 | Essobi | Haha |
| 18:11.47 | PTG123 | Godsey: yes they do, under load, freebsd wins every time |
| 18:11.51 | Godsey | locking up on smp hardware freebsd tops it |
| 18:11.55 | puppet | juggie: few calls 6+? |
| 18:11.57 | Essobi | Hahaha. |
| 18:12.04 | Essobi | Godsey, have you ever ran FreeBSD? |
| 18:12.06 | puppet | juggie: or 10- |
| 18:12.07 | Godsey | yes |
| 18:12.16 | PTG123 | If you want to run a desktop use linux :) if you want a server run freebsd |
| 18:12.16 | Essobi | Sounds like a personal problem to me then. |
| 18:12.20 | *** part/#asterisk Remowylliams (~Mare@168.215.138.106) |
| 18:12.27 | Essobi | PTG123 DING DING DING! We have a winner. |
| 18:12.33 | Godsey | I began using freebsd in 95 |
| 18:12.41 | Zeeek | me too |
| 18:12.41 | Godsey | when it WAS better than linux |
| 18:12.46 | PTG123 | Hah |
| 18:12.49 | Juggie | puppet, depends on what your doing, without any transcoding it may be possible. |
| 18:12.51 | Godsey | :) |
| 18:13.02 | PTG123 | Godsey: i can crash a linux box in 2 seconds flat, it can't even create 1000 threads without dying |
| 18:13.11 | Zeeek | linux is so 1968 |
| 18:13.16 | Essobi | Hah. |
| 18:13.20 | PTG123 | godsey: you have conflicts out the ass managing software |
| 18:13.27 | Godsey | how many cluster filesystems are there for linux? |
| 18:13.31 | PTG123 | and thats not even scratching the surfice on problems |
| 18:13.31 | Godsey | or freebsd? |
| 18:13.46 | PTG123 | Godsey: the only reason you would want to support the MOST hardware is if you run a desktop |
| 18:13.56 | PTG123 | if you run a server you can afford to buy the exact rihgt hardware the os supports the best |
| 18:14.01 | Essobi | Or your servers are all junk piles. :) |
| 18:14.21 | PTG123 | So more hardware support is an argument for a desktop os :) |
| 18:14.27 | Juggie | take a look at, http://geri.cc.fer.hr/~ivoras/web2/papers/osbench.html |
| 18:14.35 | Godsey | I asked about cluster filesystems |
| 18:15.32 | Essobi | what kind of dingus uses a clusstering filesystem over NFS? |
| 18:15.32 | Godsey | nfs is sloppy |
| 18:15.32 | Godsey | doesn't handle locks properly |
| 18:15.32 | Godsey | isn't replicated to multiple hosts |
| 18:15.32 | Essobi | FFS, get a SCSI-over-ip implementation with a jbod raid then. |
| 18:15.48 | Essobi | Gods If your programs are wrote correctly, there is no problem with locking. |
| 18:15.49 | Godsey | that's fine, you still don't have a cluster filesystem to mount it on multiple hosts |
| 18:16.02 | Essobi | Sure, it's called NFS. |
| 18:16.10 | bannerman | Anyone have any comments on Nufone's reliability and service? |
| 18:16.15 | Godsey | single point of failure is the nfs server |
| 18:16.18 | Essobi | Read my lips, NETAPPS |
| 18:16.20 | Zeeek | great and excellent |
| 18:16.37 | bannerman | Thanks, Zeeek. |
| 18:17.05 | Godsey | why would I need netapps? |
| 18:17.18 | *** join/#asterisk n3tar (~geno@201.254.31.232) |
| 18:17.41 | Essobi | Raid 5+1 with redundant cache coherent heads, and lookie there.. a cross platform, fail-over, redundant file storage.. Windows, Sun, BSD, Linux, MacOSx, and the all play happy together. |
| 18:17.54 | Delvar | rar! gust wrote a dialplan macro to give back an available channel from a list of channels and call limits |
| 18:17.56 | *** join/#asterisk coldfeet (~cf@dsl-80-46-109-145.access.as9105.com) |
| 18:18.04 | coldfeet | hi all |
| 18:18.14 | Godsey | I use raid6 |
| 18:18.14 | Delvar | ie only allow 20 calls down one sip account, 1 down another and 45 down an IAX, also checks if the chanel is available suing chanisavail |
| 18:18.14 | MikeJ[Jayden] | hey clodfeet.. you working yet? |
| 18:18.19 | PTG123 | Multii access disk arrays are a poort way to go, they don't scale properly, then applications written to be cells instead |
| 18:18.44 | Godsey | on top of raid6 is cluster file system |
| 18:18.47 | puppet | how do I make so friends jsut can get one call or two calls, or so? and not as many as the sip phone says it can take? |
| 18:18.47 | Essobi | Y-cables suck. :) |
| 18:18.53 | Godsey | which gives read/write to all systems in the cluster |
| 18:19.05 | Godsey | front end clients fail over to multiple backend servers |
| 18:19.12 | Godsey | hourly snapshots |
| 18:19.17 | coldfeet | yeah it seems to be, I removed the registry entry for xlite reinstalled and voila it works :-) |
| 18:19.25 | MikeJ[Jayden] | cool |
| 18:19.28 | coldfeet | have a question bout dialplans |
| 18:19.32 | MikeJ[Jayden] | I knew it was somthing weird.. |
| 18:19.44 | Qwell | puppet: Thats a good question. I don't think I've seen a limit option in any of the configs |
| 18:19.44 | coldfeet | what I have is 3 inbound 0845 numbers eg 08451 08452 08453 |
| 18:20.02 | Delvar | puppet: look up http://www.voip-info.org/wiki-Asterisk+cmd+CheckGroup |
| 18:20.17 | coldfeet | for each one of these I need to change the CallerID, and then dial the SIP extension, however I cant work out a nice way of writing this into a dialplan |
| 18:20.20 | Qwell | ahh, groups...they can do anything |
| 18:20.26 | Godsey | I've found several bugs w/ freebsd and twice was blamed on user error |
| 18:20.27 | Delvar | puppet: they took out incomming/outgoing limit from sip.conf in favor of this |
| 18:20.34 | Godsey | when infact it was kernel problem |
| 18:21.07 | coldfeet | I wanna do something like exten => 0845_ ,1, SETCIDNAME (name based on 0845_...pulled from somewhere) |
| 18:21.24 | Qwell | coldfeet: agi that hits a DB? |
| 18:21.25 | coldfeet | what I have write now is a section for each extension, is that the correct way |
| 18:21.36 | Essobi | Godsey You sound a bit jaded. |
| 18:22.14 | Essobi | Godsey So what distro do YOU use? |
| 18:22.21 | Godsey | years of #freebsd and freebsdhelp on efnet did it :) |
| 18:22.27 | Essobi | haha |
| 18:22.32 | Essobi | They are all assholes. |
| 18:22.38 | Essobi | I've met half of those peckers. |
| 18:22.40 | Godsey | I use rhel, gentoo, and solaris10 :) |
| 18:22.51 | Essobi | No wonder you're talking about GFS |
| 18:22.54 | Godsey | but not redhat |
| 18:22.55 | Essobi | RHEL.. ehh. |
| 18:22.58 | Godsey | I use white box |
| 18:23.05 | MikeJ[Jayden] | you can do lookup into astdb if you want, like the blacklist stuff does, and then you can do it without seperate sets for each |
| 18:23.16 | MikeJ[Jayden] | plus change it real easy with somthing web |
| 18:23.32 | Godsey | whitebox is a source build of rhel3 |
| 18:23.33 | Essobi | You know the sistina GFS was wrote for irix originally, and was being ported to linux and free BSD.. |
| 18:23.50 | coldfeet | web I will get to..eventually, how do u call astdb from the extensions file...is it on wiki |
| 18:23.51 | Godsey | I use veritas on solaris for it |
| 18:24.02 | Godsey | and for linux I'm using peerfs, not gfs |
| 18:24.09 | Essobi | Ah. |
| 18:24.14 | Essobi | GFS is cool. |
| 18:24.20 | Godsey | if not for peerfs I'd try IBM's gpfs |
| 18:24.27 | Essobi | FCAL fabric networks for drives are neeet |
| 18:24.32 | Godsey | oracle's ocfs2 will be interesting |
| 18:24.37 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 18:24.49 | Godsey | peerfs is a simplified gfs from setup/admin standpoint |
| 18:24.53 | Essobi | I still want a SCSI-over-IP implementation running on 10G |
| 18:24.58 | MikeJ[Jayden] | http://www.voip-info.org/wiki-Asterisk+cmd+DBget |
| 18:25.09 | Godsey | have you seen etherdrives? |
| 18:25.23 | Essobi | cant say I've looked |
| 18:25.36 | Godsey | http://www.coraid.com/ |
| 18:25.44 | Godsey | they've been advertising in linux journal |
| 18:25.47 | MikeJ[Jayden] | you can use that for all kinds of interesting things |
| 18:26.14 | Godsey | oh one last fbsd/linux thing |
| 18:26.20 | Godsey | someone said they could forkbomb linux |
| 18:26.36 | Essobi | maybe a 2.4 kernel |
| 18:26.37 | Essobi | Heh. |
| 18:26.38 | Godsey | you can easily do the same w/ freebsd if you alter login.conf |
| 18:26.47 | Essobi | forkbombing is SO old skool |
| 18:26.55 | Qwell | forkbomb? |
| 18:27.02 | Essobi | forkbomb |
| 18:27.09 | Qwell | wassat? |
| 18:27.09 | Nugget | linux is poo. |
| 18:27.20 | Essobi | heh |
| 18:28.13 | Godsey | I developed a product around peerfs and linux :) |
| 18:28.37 | Godsey | using raid6, lvm2, and peerfs on top |
| 18:28.43 | KalD|Work | anyone know if it is possible to send other information like text etc down an IAX connection - and how asterisk will deal with it? |
| 18:28.49 | coldfeet | does the database get cleaned on restart ...I am guessing now since its in a file |
| 18:28.55 | Godsey | online backups of customer data |
| 18:28.56 | Godsey | and full archive of in/out email |
| 18:29.09 | Godsey | it takes hourly snapshots for 2 days |
| 18:29.42 | gr8nash | anyone running BV sucessfully here? |
| 18:29.45 | Godsey | hum not 2 days worth, only 18 hours |
| 18:29.49 | gr8nash | sorry broad voice |
| 18:30.04 | puppet | who says Freebsd is best choice for a PII233 / 128mb ram Over Linux for a Asterisk machine |
| 18:30.05 | Godsey | gr8nash: I have it setup at work for outbound calls |
| 18:30.15 | modulus_ | broadvoice really sucks bad |
| 18:30.19 | gr8nash | yeah.. i can call out.. just cant receive calls! |
| 18:30.32 | Godsey | gr8nash: I was able to get it to work in 1 direction |
| 18:30.33 | gr8nash | heh.. techsupport last night.. broke calling out.. hehe |
| 18:30.36 | Godsey | either in or out, not both |
| 18:30.45 | Godsey | I figured it was user error :) |
| 18:30.56 | Godsey | for my home setup I use ipkall for inbound and nufone for out |
| 18:31.12 | Godsey | I'm thinking of getting livevoice for inbound tollfree |
| 18:31.22 | gr8nash | Godsey, what error were you getting? |
| 18:31.41 | Godsey | I didn't get an error if I recall right |
| 18:31.45 | gr8nash | is it kinda a pain to have one for outbound.. 1 for inbound? |
| 18:31.48 | coldfeet | okay got the callerID inserted and looking up |
| 18:31.50 | Godsey | but when I'm able to place calls |
| 18:32.02 | Godsey | I try to call the # and get a message from broadvoice |
| 18:32.15 | gr8nash | 401 unahtorized? in the incoming proxy |
| 18:32.21 | coldfeet | what variable contains the lookedup callerid so that I can use it in setcallerid |
| 18:32.26 | Godsey | I saw that broadvoice had some patches for asterisk |
| 18:32.30 | PTG123 | anyone in here use livevoip? |
| 18:32.36 | gr8nash | those are old patches i was told |
| 18:32.36 | Godsey | I gave up quickly |
| 18:32.37 | loud | i do |
| 18:32.45 | zipp | so what makes ipkall want to offer such a service? |
| 18:32.49 | Godsey | I think livevoip is what I want to use for 888, not livevoice |
| 18:33.01 | Godsey | zipp it's a clec and trying to ballance traffic |
| 18:33.26 | Godsey | only thing I can think of :) |
| 18:33.34 | PTG123 | yah i just wonder if anyone has any complaints for livevoip |
| 18:33.37 | Godsey | they have lots of dial ports through nocharge.com |
| 18:33.55 | Godsey | I ordered ata device from totalaccess.net 2 days ago |
| 18:34.02 | PTG123 | what i like about them is they don't have to terminate via asterisk, they can drop you sip right from their switch |
| 18:34.05 | Godsey | when they arrive I'll setup livevoip account |
| 18:34.09 | *** join/#asterisk citats (~james@duff.gnuinter.net) |
| 18:34.10 | PTG123 | what i don't like is everythin comes out of new york |
| 18:34.12 | gr8nash | Godsey, would you use IAX or SIP |
| 18:34.27 | Godsey | I'll use IAX on the ata |
| 18:34.34 | Godsey | I use IAX for nufone |
| 18:34.40 | *** join/#asterisk fjoe (~fjoe@samodelkin.net) |
| 18:34.46 | fjoe | hi |
| 18:34.51 | Godsey | and work uses SIP for broadvoice |
| 18:34.55 | *** join/#asterisk brimstone (me@146.229.186.157) |
| 18:35.05 | Godsey | I think if I register w/ broadvoice I can't make calls |
| 18:35.20 | Godsey | it could totally be user error |
| 18:36.10 | gr8nash | maybe.. except there tech support was clueless |
| 18:36.23 | Godsey | heh when I called it sounded like he was doing dishes :) |
| 18:36.25 | gr8nash | i spent an hour on the phone with them.. they only broke what i had working before |
| 18:36.25 | Qwell | gr8nash: most tech support is |
| 18:37.19 | Godsey | can you use voiceplus w/ asterisk? |
| 18:37.44 | gr8nash | i guess ill try another company.. just try em all till i find one.. |
| 18:37.50 | gr8nash | i wonder whats better about IAX |
| 18:37.58 | zipp | everything |
| 18:38.17 | Godsey | firewalls/nat doesn't interfere |
| 18:38.23 | Godsey | http://store.totalaccess.net/oscommerce-2.2ms2/catalog/product_info.php?products_id=190&osCsid=6e0aefee352dc17f8cbf31a38a3f65e5 |
| 18:38.28 | Godsey | here are the devices I purchased |
| 18:38.37 | Godsey | tho I'm a bit unhappy w/ the order progress :) |
| 18:38.46 | Godsey | I got confirmation but nothing else so far |
| 18:39.21 | zipp | I guess you purchased for price? |
| 18:39.23 | gr8nash | zipp.. well let me ask it this way.. what difference will the user see |
| 18:39.33 | Godsey | nope |
| 18:39.36 | Godsey | iaxtalk is cheaper |
| 18:39.37 | fjoe | I have a lame question: I have zaptel drivers loaded, zap show channels shows 8 channels, but why show channels shows 0 active channels for me? |
| 18:39.44 | zipp | Godsey, why not an iaxy? |
| 18:39.54 | zipp | gr8nash, is the user behind nat? |
| 18:40.04 | Godsey | I think this device is better than an iaxy |
| 18:40.06 | fjoe | I mean "zap show channels" and "show channels" respectively |
| 18:40.16 | Godsey | Support G.711 a/u ,G.723.1 5.3/6.3, G.729A/B/AB/ gsm610 |
| 18:40.44 | zipp | yes, iaxy only supports 1 codec |
| 18:40.59 | Godsey | this has comfort noise |
| 18:41.26 | Godsey | that's really it |
| 18:41.43 | zipp | I suppose if you have no nat issues |
| 18:41.57 | Godsey | what do you mean? |
| 18:42.08 | gr8nash | yes im behind NAT |
| 18:42.15 | zipp | sip behind nat sucks |
| 18:42.20 | Godsey | oh yes :) |
| 18:42.45 | Godsey | <- less than optimal at splitting out 2 conversations sorry |
| 18:42.55 | *** join/#asterisk ManxPower (eric@106.sub-166-145-133.myvzw.com) |
| 18:43.09 | MikeJ[Jayden] | lookedup caller id? |
| 18:43.15 | MikeJ[Jayden] | the return from dbget you mean? |
| 18:43.21 | *** join/#asterisk YoYo (YoYo@dilbert.psknet.com) |
| 18:43.40 | YoYo | ok, which kernel these days? 2.4 or 2.6? |
| 18:43.47 | Qwell | 2.6 works fine |
| 18:43.53 | zipp | 2.6 here |
| 18:44.07 | Godsey | a co-worker ordered a color laser printer from dell |
| 18:44.09 | Mw3 | both sucks :( |
| 18:44.11 | Godsey | and they keep delaying the ship date |
| 18:44.12 | puppet | hmm |
| 18:44.20 | zipp | Linux debian 2.6.8-2-686 #1 Mon Jan 24 03:58:38 EST 2005 i686 GNU/Linux |
| 18:44.21 | Godsey | she just called and after talking for 10 minutes |
| 18:44.22 | Nugget | windows vs freebsd: http://lnk.nu/gallery.622mbit.org/1qf.jpg |
| 18:44.22 | Nugget | linux vs freebsd: http://lnk.nu/linuxisforbitches.com/1qg.php |
| 18:44.30 | Godsey | "I NEED THE FUCKING PRINTER TO PRINT MY SUICIDE NOTE" |
| 18:45.11 | puppet | I got a question regarding Record(), IF the person hangs up instead of pressing # is there any good way to recode it to mp3 and mail it? like if someone hangs up during record it goes to number+100 ? |
| 18:45.21 | gr8nash | zipp what ports do you open for IAX on a NAT/firewall |
| 18:45.27 | Qwell | Nugget: nsfw warnings, please :P |
| 18:45.30 | zipp | gr8nash, non |
| 18:45.37 | Nugget | heh |
| 18:45.44 | zipp | which means you can have multiple iaxy devices behind nat |
| 18:45.47 | gr8nash | zipp, to receive you dont have to open ports? |
| 18:45.53 | zipp | gr8nash, no |
| 18:45.59 | gr8nash | WOW |
| 18:46.00 | gr8nash | thats cool |
| 18:46.17 | zipp | iax2 is using only 1 port |
| 18:46.42 | zipp | when you register it is kept alive, so asterisk has a route back behind nat |
| 18:47.09 | gr8nash | so who provides IAX2.. i have seen only IAX |
| 18:47.20 | Qwell | gr8nash: everybody calls it just IAX |
| 18:47.23 | Qwell | afaik |
| 18:47.25 | zipp | IAX is common name |
| 18:47.30 | gr8nash | ohh cool |
| 18:47.45 | zipp | really it is IAX2, but most won't/don't remember IAX1 |
| 18:48.13 | zipp | read the wiki about IAX2 |
| 18:48.38 | Godsey | I think my project for this week should be getting asterisk working w/ sql for extensions |
| 18:49.16 | gr8nash | so IAX is stable enough to use for business .. we dont want to drop customer calls.. =) |
| 18:49.18 | zipp | Godsey, wait for 1.2 |
| 18:49.33 | zipp | gr8nash, IAX is better then sip imho |
| 18:49.44 | Qwell | by leaps and bounds |
| 18:49.46 | zipp | I use it every day for business, and I would trust it more then sip |
| 18:50.07 | *** join/#asterisk _tommyg_ (~tom@vsat-148-64-73-166.c119.t7.mrt.starband.net) |
| 18:50.10 | coldfeet | Guys I am trying earlier suggestion on LookUPCIDName, the lookup works fine...I think, but how can I now use the returned variable, |
| 18:50.12 | zipp | what were there on the last IAX dev conf call, 50 people? |
| 18:50.34 | Qwell | cst is what, gmt-6? |
| 18:50.35 | gr8nash | heh.. cool |
| 18:50.35 | MikeJ[Jayden] | 30+ on the last call... |
| 18:50.53 | gr8nash | ill look at.. http://www.livevoip.com/ and see if they setup easy |
| 18:50.57 | MikeJ[Jayden] | it is 10 till 1 cst right now |
| 18:51.16 | Qwell | think I might join in this time |
| 18:51.17 | PTG123 | use iax if you don't have a choice, but with someone like livevoip they can give you sip direct to their switch.. so why use iax? |
| 18:51.20 | zipp | MikeJ[Jayden], actually 9 till 1 :) |
| 18:51.36 | MikeJ[Jayden] | ~troutslap zipp |
| 18:51.38 | jbot | ACTION slaps zipp around with a large trout |
| 18:52.06 | zipp | PTG123, if you are behind nat, or if you are connecting with a device that supports IAX |
| 18:52.08 | Godsey | zipp: asterisk 1.2? |
| 18:52.27 | zipp | Godsey, a few [6] months away |
| 18:52.53 | Godsey | is there something that will bite me in the butt w/ it now? |
| 18:52.56 | zipp | will have a common interface to db's, conf files, ... |
| 18:53.09 | PTG123 | zipp: behind nat maybe, but a dvice that supports iax no, b/c then you are adding an extra piece3 of equipment on their end |
| 18:53.11 | Godsey | oh |
| 18:53.23 | zipp | Godsey, read this: http://www.voip-forum.com/news.php?p=166&more=1 |
| 18:53.34 | Godsey | thanks |
| 18:53.47 | zipp | PTG123, I imagine it is at least fast ethernet to their switches |
| 18:54.07 | PTG123 | zipp: but if their server has problems you go down |
| 18:54.11 | _tommyg_ | Can ya point me in the direction of MCGP configuration |
| 18:54.17 | PTG123 | they are doing SIP->THEIRIAX |
| 18:54.24 | PTG123 | why wouldn't you just do SIP->YOU |
| 18:54.57 | zipp | because they are doing SIP -> IAX over a local network, and you are doing IAX to you over the internet |
| 18:55.08 | _tommyg_ | errr, I meant MGCP |
| 18:55.45 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 18:55.47 | zipp | PTG123, IAX trunking, smaller protocol, no rtp... |
| 18:56.49 | *** join/#asterisk Tili (~Tili@202-133-65-5-dialup.sat.net.pk) |
| 18:57.25 | Qwell | stop typing on the call :P |
| 18:57.32 | shmaltz | http://story.news.yahoo.com/news?tmpl=story&ncid=1211&e=10&u=/nm/20050310/wr_nm/tech_internet_powerline_dc&sid=95573372 |
| 18:57.35 | PTG123 | zipp: on a good network who cares |
| 18:58.08 | *** join/#asterisk adorah (~jack@80.179.34.21.forward.012.net.il) |
| 18:58.25 | zipp | PTG123, I don't think a dsl/cable line constitutes a good network |
| 18:58.55 | PTG123 | ah |
| 18:59.04 | Godsey | zipp, I'm anxious to play w/ it :) I'll grab head |
| 18:59.09 | adorah | grrrr... |
| 18:59.11 | Godsey | humm that didn't sound good |
| 18:59.15 | Qwell | Godsey: That sounded really bad |
| 18:59.15 | zipp | ha |
| 18:59.25 | shmaltz | zipp, why not? |
| 18:59.30 | zipp | s/head/cvshead/ |
| 18:59.33 | Qwell | mind Qwell > gutter |
| 18:59.45 | zipp | shmaltz, relative to a oc12 or something :) |
| 18:59.51 | shmaltz | cable is very stable and gives you in some places up to 10mbps download speed and 1 mbps upload |
| 18:59.53 | *** part/#asterisk fjoe (~fjoe@samodelkin.net) |
| 18:59.55 | shmaltz | :) |
| 19:00.18 | CosmicRay | shmaltz: where? |
| 19:00.21 | shmaltz | of course oc12 is better, but it's not a soho/small business solution |
| 19:00.31 | shmaltz | central New Jersey |
| 19:00.31 | CosmicRay | shmaltz: I have never seen a "stable" cable modem |
| 19:00.43 | zipp | t1, up = down and you can trust it is stable |
| 19:00.44 | shmaltz | CosmicRay, maybe in your area, |
| 19:01.08 | shmaltz | overhere it is considered the most stable solution outside T-1 services |
| 19:01.10 | Godsey | I'm moving to a place w/ 3mbps cable |
| 19:01.11 | Godsey | sync |
| 19:01.13 | shmaltz | of for the price |
| 19:01.14 | CosmicRay | shmaltz: I've never heard of anyone having a stable one either |
| 19:01.16 | zipp | cable is a shared medium, if anything dsl > cable, read docis 2.0 standards |
| 19:01.24 | CosmicRay | right |
| 19:01.27 | Godsey | the only complaint I have is they won't let me do bgp :) |
| 19:01.28 | CosmicRay | that's one problem with it |
| 19:01.36 | *** join/#asterisk Torgo (~Torgo@c-66-41-135-254.mn.client2.attbi.com) |
| 19:01.56 | shmaltz | zipp, I agree about dsl, but in my area cable is much better |
| 19:02.02 | CosmicRay | I had @home in indianapolis for awhile... outage about once a month or so. once comcast took over, they capped download speed at 1/3 of what I used to get with it |
| 19:02.02 | Godsey | I can get fiber to my house for $550/mo |
| 19:02.28 | mogorman | *1 to talk right? |
| 19:02.29 | Godsey | cosmicray: I used to live in Syracuse, NY w/ rr.com and got over 700K/sec down :) |
| 19:02.49 | *** join/#asterisk _Vile (~vile@90.b160.bendtel.net) |
| 19:02.52 | CosmicRay | we have business-class cable modem at work in a different state now... and we still have outages about once every 2-3 months |
| 19:03.04 | shmaltz | CosmicRay, the capping is something that in some areas it very popular, however I know a case here wher I live, someone called optimum that the speed wasnt too fast (only around 200 kbps) and they send someone down to fix it |
| 19:03.10 | CosmicRay | not to mention that cox periodically has routing issues |
| 19:03.26 | Torgo | Anyone got a moment for a compete noob having NAT issues? |
| 19:03.48 | zipp | Torgo, sip I imagine? |
| 19:03.57 | Godsey | I have DSL now, but the co is closed in the area where I want to move (just sold house) |
| 19:03.57 | adorah | tprgo: I'll join you in y'r request.. |
| 19:04.03 | shmaltz | in my area Verizon DSL goes down (it's actualy their ATM loop that goes down, and every dsl provider is effected) around every six weeks, cable never |
| 19:04.18 | zipp | shmaltz, no WISP's? |
| 19:04.20 | CosmicRay | I've got dsl now. not any great speed, but still, quite reasonable, and once they discovered that they had also given my IP to someone else, rock solid |
| 19:04.20 | shmaltz | Torgo, use IAX. thats my moment |
| 19:04.21 | Inv_arp | adorah: whats he prob ? what provider? etc........ |
| 19:04.28 | Torgo | Yep. I've installed asterisk@home... seems towork just fine with a POTS line and a TDM400 card. But I'm trying to get an outgoing call to use a BroadVoice account... |
| 19:04.29 | adorah | Nat Traversal.. |
| 19:04.29 | Inv_arp | s/he/th |
| 19:04.45 | shmaltz | zipp, whats WISP? Wireless? |
| 19:04.49 | zipp | shmaltz, yes |
| 19:04.51 | Inv_arp | adorah: ok...err symtoms? |
| 19:05.03 | zipp | shmaltz, www.nxwi.com/temp (still working on it) |
| 19:05.10 | CosmicRay | shmaltz: the same thing happened to me when I had dsl in dallas |
| 19:05.13 | Torgo | "stopping retransmission blahblah@192.168.4.200" (the machine's private IP address) |
| 19:05.23 | CosmicRay | shepherd: SBC popped the atm fabric, poof, dsl is down in the metro area |
| 19:05.31 | CosmicRay | s/shepherd/shmaltz/ |
| 19:05.40 | shmaltz | you can drive around and within 30 seconds you should be able to pick one up, from residential, but no commercial, unless you drive down to Barnes and Nobels |
| 19:05.43 | zipp | Torgo, does your router have a dmz function? |
| 19:05.58 | scrubb | anyone here usinga Sipura SPA? I need to know if they support a NAT setting. |
| 19:06.02 | Torgo | zipp, sure does. |
| 19:06.03 | adorah | the good news is that I call a remote sip ext. seamlessly via my router..How ever once the remote user too has a router - no voice streaming..hardly a ring.. |
| 19:06.12 | zipp | Torgo, you tried that? |
| 19:06.12 | *** part/#asterisk _tommyg_ (~tom@vsat-148-64-73-166.c119.t7.mrt.starband.net) |
| 19:06.17 | CosmicRay | zipp: can't he just set nat = yes, the ip address, and canreinvite=no in his sip.conf? |
| 19:06.23 | Torgo | zipp: nope. Didn't know I should! |
| 19:06.54 | zipp | CosmicRay, sometimes that works with stun |
| 19:07.00 | Godsey | zipp: I was thinking of trying to do wisp in my fairly small town |
| 19:07.13 | BrianR___ | finally found my QoS problem.. There was another device sneaking packets in after my traffic shaper. Grr. |
| 19:07.14 | *** join/#asterisk jlewis (~jlewis@solo.atlantic.net) |
| 19:07.18 | Godsey | but a cable company is rolling out free wireless in a town about 40 miles away |
| 19:07.25 | jlewis | has anyone else hacked enumlookup.agi to do random call distribution (instead of simultaneous dialing) of equal order/priority naptr records? |
| 19:07.26 | mogorman | sorry i think that was me |
| 19:07.31 | Godsey | seems like access is going to be dirt cheap before long |
| 19:07.37 | shmaltz | zipp, nxwi.com, they use WiFi or Satelite? |
| 19:07.39 | jlewis | I just did...and wanted to see if maybe there was a better way |
| 19:07.43 | *** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net) |
| 19:07.48 | yaaar | word |
| 19:07.53 | adorah | <CosmicRa: I've already done that+ externip..no avail |
| 19:08.01 | Torgo | I've been using AMP... but it's not perfect. How can I restart asterisk or have it re-read conf files? kill -HUP? |
| 19:08.01 | yaaar | hey fella's. what's your pick for a good gui software phone for linux? |
| 19:08.04 | zipp | shmaltz, custom mac (access layer) wifi (nxwi is me ) |
| 19:08.14 | puppet | Ffs, are they stupid, swedish Antipiracy org. something and 3 other music/filmcompanies leaved in a paper to the court and got right to SHUT DOWN an ISP in sweden to look after THREE IPs?! This due to 4 moves and 8 albums :s |
| 19:08.24 | shmaltz | oh, anyplans on coming to NY Metro area? |
| 19:08.34 | shmaltz | zipp, thats you? |
| 19:08.47 | zipp | shmaltz, yes |
| 19:08.47 | shmaltz | theres a browser error when browsing the page |
| 19:08.51 | coldfeet | guys on lookupcaller ID if my inbound number already has one assigned will nething be returned |
| 19:08.52 | mogorman | bkw |
| 19:09.02 | BrianR___ | Has anyone hacked up a dialplan fragment for alliance-style conferencing using asterisk's meetme? |
| 19:09.06 | zipp | shmaltz, as I said, still working on the page, notice the /temp in url |
| 19:09.07 | BrianR___ | before I go reinvent the wheel.. |
| 19:09.07 | shmaltz | http://www.nxwi.com/temp/index.php?page=business&sub=access |
| 19:09.15 | shmaltz | oh, sorry |
| 19:09.55 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 19:10.00 | shmaltz | zipp I like the logo |
| 19:10.21 | zipp | shmaltz, thank my designer :) I chose it though... |
| 19:10.24 | mogorman | hey now |
| 19:10.37 | mogorman | talkin smack |
| 19:10.41 | shmaltz | send him a copy of theis message |
| 19:10.42 | shmaltz | :) |
| 19:11.32 | afe | Howdy! Could wctdm sharing IRQ with uhci_hcd cause problems? |
| 19:11.49 | shmaltz | afe, I would guess that yes |
| 19:12.02 | shmaltz | since zaptel is very egoistic when it comes to irq |
| 19:12.04 | Torgo | CasmicRay, adding your suggestions still shows @...@192.168.4.200" on the outgoint request |
| 19:12.13 | Torgo | CosmicRay even |
| 19:12.19 | afe | uhci_hcd is USB, right? |
| 19:12.24 | shmaltz | yep |
| 19:12.30 | mogorman | is anyone in the dev con on irc |
| 19:12.32 | mogorman | just say hi |
| 19:12.37 | shmaltz | do you have an external HD plugged in? |
| 19:12.45 | Inv_arp | oh i forgot lemme log in |
| 19:12.55 | afe | nope, I have nothing connected to usb |
| 19:13.00 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 19:13.05 | shmaltz | then just unload the Module |
| 19:13.11 | shmaltz | and disable it if possible |
| 19:13.36 | afe | ok, thanks. I'll take a look in bios and see if I can disable it |
| 19:13.47 | *** join/#asterisk stefh (~stef@65.39.228.5) |
| 19:13.47 | shmaltz | nope, do a rmmod |
| 19:13.51 | zipp | mogorman, I am in, but muted :) |
| 19:14.05 | mogorman | muted people need a voice |
| 19:14.05 | shmaltz | as well as bios |
| 19:14.56 | zipp | anyone listening to the dev conf, it is 80f here, sunny and clear :) |
| 19:14.59 | *** join/#asterisk clive- (~pirch@rrba-146-115-158.telkomadsl.co.za) |
| 19:15.03 | Mneumonic | anyone know why i dont have chan_alsa.so in my modules dir? or how to get it? |
| 19:15.04 | Inv_arp | im in |
| 19:15.05 | afe | would that be like "rmmod uhci_hcd, or should I use any options? |
| 19:15.05 | Qwell | mogorman: You can still talk... |
| 19:15.09 | Qwell | to yourself... |
| 19:15.18 | mogorman | lol |
| 19:15.19 | mogorman | ouch |
| 19:15.30 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 19:15.31 | Qwell | zipp: You must live in southern california. ;] |
| 19:15.49 | adorah | the good news is I connect a remote sip ext. seamlessly via my router when he is connected directly with a modem. ..However once the remote user too has a router - no voice streaming..hardly a ring.. |
| 19:15.58 | zipp | Qwell, http://maps.google.com/maps?q=78373&hl=en |
| 19:16.05 | Qwell | way off |
| 19:16.12 | stefh | Hello, I want to start safe_asterisk as non root user. If I use the command "su the_user -c asterisk", asterisk is started correctly, but the same command with safe_asterisk won't work |
| 19:16.19 | afe | well, it unloaded without crashing anything :) |
| 19:16.34 | Qwell | oh, not too far off |
| 19:16.44 | Qwell | AZ and TX would have been my next guesses. :p |
| 19:17.17 | *** join/#asterisk critch (critch@steven.basesys.com) |
| 19:18.20 | critch | anyone here been contacted by a guy in NY named Allen needing asterisk help? |
| 19:18.38 | Qwell | Nope. Are you Allen? |
| 19:18.50 | adorah | Anyone can help me with my NAT Traversal problem? |
| 19:18.50 | Torgo | zipp, I put the asterisk server in the DMZ. I still can't ping it from the outside. |
| 19:18.51 | critch | no. My boss took the call earlier |
| 19:18.54 | Qwell | hmm...Steven. Brother of Allen? |
| 19:19.08 | zipp | Torgo, you are pinging your router public ip right |
| 19:19.16 | Torgo | zipp, yep |
| 19:19.24 | zipp | blocking icmp? |
| 19:19.48 | Torgo | zipp, dunno. It's a linksys befsr41. If I set the DMZ, do I still have to forward ports? |
| 19:19.50 | critch | Torgo: ping is not a good tool when nat is involved. look at trying to use telnet to something like the manager port |
| 19:19.58 | Torgo | excellent |
| 19:20.25 | stefh | anyone has an experiance with asterisk as non-root. http://www.voip-info.org/wiki-Asterisk+non-root don't mention my problem |
| 19:20.32 | Torgo | Hey that worked. thanks, critch. |
| 19:20.39 | critch | Torgo: np |
| 19:21.11 | Torgo | now, how do I restart asterisk? (: |
| 19:21.19 | critch | Torgo: be aware that by being in the DMZ, anything the router doesn't answer your linux machine will. You essentially are not firewalled anymore |
| 19:21.28 | *** join/#asterisk eKo1 (~bernd@207.42.191.67) |
| 19:21.39 | Torgo | critch; well aware. I have my private network behind ANOTHER router. |
| 19:21.39 | zipp | Torgo, asterisk -rc reload |
| 19:21.44 | zipp | -rx sorry |
| 19:21.44 | adorah | If I can connect a remote sip thru my router when he uses a modem only but no voice when he is behind a router: Any suggestions 4 help? |
| 19:22.02 | zipp | adorah, rtp packets not getting through |
| 19:22.21 | adorah | Soo is it my router or his? |
| 19:22.33 | zipp | both sip behind? |
| 19:22.38 | adorah | I opend all possible ports both ends.. |
| 19:22.44 | adorah | both behind |
| 19:22.45 | critch | adorah: are you hitting the double nat problem? |
| 19:22.51 | *** join/#asterisk ikey (ikey@220.226.30.41) |
| 19:22.55 | adorah | works fine with 1 router only |
| 19:23.23 | adorah | I guess critch..:( |
| 19:23.28 | Torgo | critch, zipp; still no luck: Got SIP response 404 "Not Found" back from 147.135.0.128 |
| 19:23.54 | coldfeet | Mike I worked it out, when a caller calls from say 12345 (his number) to 08451 I want the callerID replaced based upon the 08451 NOT the 12345 number which is what it i doing |
| 19:24.42 | Torgo | SIP/broadvoice-35d0 is circuit-busy |
| 19:24.44 | adorah | <critch: Any advice? |
| 19:24.54 | *** join/#asterisk gpowers (~glenn@static-68-162-84-101.phil.east.verizon.net) |
| 19:25.37 | Torgo | <PROTECTED> |
| 19:26.38 | stefh | everyone starts asterisk as root here ? |
| 19:26.51 | Nugget | are you asking us or telling us? |
| 19:27.16 | eKo1 | I certaintly as hell don't. |
| 19:27.19 | stefh | it's a question |
| 19:27.21 | bannerman | What's the story with voipjet? Good service, good reliability? |
| 19:27.23 | TheBear | with the default answer exten for incoming pstn calls. Is there anyway to delay that it only activates after the 5 or 6 ring ? |
| 19:27.47 | gpowers | I'm quite happy with voipjet. cheap. fast. works. |
| 19:27.55 | tzanger | TheBear: Wait() |
| 19:28.17 | Inv_arp | bannerman: me 2 use them for outgoing, support gsm,ilbc,alaw |
| 19:28.22 | PTG123 | so does anyone here use livevoip? |
| 19:28.35 | gpowers | tried it. didn't work. |
| 19:28.42 | Inv_arp | PTG123: used them also never had a pob |
| 19:28.48 | TheBear | tzanger: I have wait(4) but it stops ringing and waits then moves on |
| 19:28.54 | PTG123 | Inv_arp: you don't use them any more? |
| 19:28.55 | JerJer[mobile] | the cock suckers at NuFone won't return my calls |
| 19:28.59 | *** join/#asterisk Moc____ (~mochouina@64.235.210.66) |
| 19:29.04 | Qwell | JerJer[mobile]: ? |
| 19:29.08 | Moc____ | hail everyone |
| 19:29.11 | tzanger | TheBear: what is it you want to do? |
| 19:29.13 | Torgo | What's with all the registration timeouts in the log? |
| 19:29.28 | Qwell | JerJer[mobile]: They're all off playing foosball |
| 19:29.33 | TheBear | tzanger: to continue ringing then only answer after the 5th or 6th ring |
| 19:29.41 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 19:29.43 | Qwell | I told you that you shouldn't have bought that table for them. :P |
| 19:29.49 | tzanger | ... |
| 19:29.50 | JerJer[mobile] | Qwell: no they're off gambling and gettng drunk |
| 19:29.53 | Inv_arp | PTG123: they dont have any miami DID's right now... i bought the wrong DID at first b/c it started with 305 which can be keywest or miami |
| 19:30.04 | Qwell | JerJer[mobile]: maybe your office is being overrun my iguanas? |
| 19:30.06 | tzanger | so someone is calling you and you do not want to pick up until the 5th or 6th ring? Wait(). |
| 19:30.10 | tzanger | i.e. |
| 19:30.15 | Inv_arp | PTG123: i had for a week worked perfect |
| 19:30.16 | Qwell | by* |
| 19:30.17 | tzanger | exten => s,1,Wait(20) |
| 19:30.21 | tzanger | exten => s,2,Answer |
| 19:30.42 | Qwell | I think he's saying he wants to answer the PSTN with a normal phone, and if he does so, tell * not to answer |
| 19:30.50 | Qwell | TheBear: correct me if I'm wrong, please |
| 19:30.59 | Qwell | ie; without an FXS |
| 19:31.07 | PTG123 | Inv_arp: ah who do you use now? |
| 19:31.19 | tzanger | TheBear: is Qwell right? |
| 19:31.32 | Qwell | I should be a translator. |
| 19:31.40 | Qwell | I think I just found my new career. |
| 19:31.45 | Nugget | heh |
| 19:31.48 | PTG123 | Qwell: who is that? |
| 19:31.55 | Qwell | PTG123: who is what? |
| 19:31.56 | Nugget | one of those fax line sharer device things would work. |
| 19:32.05 | TheBear | I spli my incoming line to a std. phone and to * server. I want it to ring, on both. if yes I answer on the std. phone then * can ignore the call. if after 5 rings I have no answered on the std. phone then I want * to pick up the call and dial my SIP phones |
| 19:32.09 | Inv_arp | PTG123: Broadvoice/incoming voipjet/outgoing |
| 19:32.10 | PTG123 | Qwell: er what is tht yuour new carreer :) |
| 19:32.10 | Qwell | Nugget: a simple relay would do it... |
| 19:32.15 | TheBear | I'll try a bigger wait(20) |
| 19:32.17 | Qwell | PTG123: translator? |
| 19:32.18 | PTG123 | Inv_arp: ah :) |
| 19:32.21 | PTG123 | Qwell: ah :) |
| 19:32.26 | Qwell | TheBear: It don't work like that... |
| 19:32.36 | tzanger | TheBear: yuck... Asterisk doesn't work so well in those conditions, |
| 19:32.41 | Torgo | Anyone got broadvoice outgoing working? |
| 19:33.01 | *** join/#asterisk afe ([jFHSCP2mm@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 19:33.11 | Qwell | PTG123: I often find people talking about 2 different things, and I'm able to bridge the gap |
| 19:33.19 | TheBear | why not ? |
| 19:33.28 | *** part/#asterisk stefh (~stef@65.39.228.5) |
| 19:33.30 | bannerman | Qwell: so you're more of a bridge |
| 19:33.34 | Qwell | TheBear: You'd need some sort of hardware. |
| 19:33.39 | bannerman | Qwell: Good marketing technique, you can sell somoene a bridge! |
| 19:33.46 | Qwell | bannerman: umm, I prefer english-english translation |
| 19:33.52 | Qwell | marketing? hell no |
| 19:34.10 | tzanger | TheBear: because Asterisk is a PBX, not an answering machine |
| 19:34.22 | TheBear | ok |
| 19:34.33 | tzanger | TheBear: there is (currently) no way to have Asterisk see a ring and then tell it "don't worry about it" if it goes away |
| 19:34.37 | adorah | tzange: LOL |
| 19:34.53 | Qwell | TheBear: The EASIEST way to do it, would be to get an FXS, let * answer, and immediately transfer to the FXS |
| 19:35.00 | Qwell | then if it isn't answer, * will take control again |
| 19:35.10 | tzanger | TheBear: the way to do it normally is to have an FXO and FXS interface and have * route the call to the FXS interface by Dial()ing it |
| 19:35.18 | Qwell | what he said |
| 19:35.20 | tzanger | and if there's no answer on the FXS interface after 20s or whatever, dump to IVR |
| 19:35.44 | TheBear | ok I can do that I have the X100P and TDM10B cards |
| 19:35.49 | CosmicRay | TheBear: why don't you just get a ATA for your analog phone/ |
| 19:35.50 | Qwell | is a ring always x seconds apart? |
| 19:35.51 | tzanger | TheBear: perfect |
| 19:35.57 | PTG123 | hey I have a channel Local/303 that shows 347 entries in the cdr.. yet i only have received 50 calls today.. anyone know what Local/303 is refering to, and why the number is so high? |
| 19:35.58 | tzanger | in your dialplan |
| 19:36.03 | CosmicRay | ah |
| 19:36.09 | CosmicRay | someone already suggested that, never mind :-) |
| 19:36.12 | tzanger | exten => s,1,Ring(Zap/2,20) |
| 19:36.18 | tzanger | exten => s,2,Answer |
| 19:36.25 | Qwell | ring? hmm |
| 19:36.29 | tzanger | exten => s,3,VoiceMail(whatever) |
| 19:36.32 | tzanger | exten => s,4,Hangup |
| 19:36.32 | TheBear | ok I'll try that thanks tzanger: |
| 19:36.45 | Qwell | tzanger: Whats Ring() do? I'm not showing it in the CLI |
| 19:36.48 | tzanger | Asterisk wants to be in charge of the call, not subservient to it |
| 19:36.50 | eKo1 | Quickie: Is [default] the the default context for entries that don't have context=... in them? |
| 19:36.51 | tzanger | er |
| 19:36.52 | TheBear | I'm sure he means dial(zap/2 |
| 19:36.55 | tzanger | Qwell: not Ring, sorry, Dial() |
| 19:36.59 | Qwell | oh, ok, heh |
| 19:37.14 | adorah | voice thru router-router problem: Any advice? |
| 19:37.16 | Inv_arp | Ring sounds better :) |
| 19:38.08 | puppet | Can I make so calls that comes in on one registry goes to one phone directly? |
| 19:38.12 | *** join/#asterisk GordonF (GordonF@rrba-146-64-37.telkomadsl.co.za) |
| 19:38.15 | Nugget | of course. |
| 19:38.20 | GordonF | Hi all |
| 19:38.21 | puppet | got three SIP registrys, want to have one to go to my fax directly |
| 19:38.29 | Nugget | so do that. |
| 19:38.51 | TheBear | Qwell: tzanger: do either of you use snom 200s and intercom feature ? |
| 19:38.53 | puppet | nugget: But how I didnt get how to do it exactly read some about register => user:pass@host/name |
| 19:38.56 | Qwell | nope |
| 19:38.59 | puppet | nugget: but what does that /name does? |
| 19:39.05 | Nugget | that's unrelated. |
| 19:39.13 | puppet | nugget: oh ok |
| 19:39.18 | Qwell | TheBear: But, if you were to send one over, I'd be glad to test it for you. :) |
| 19:39.21 | puppet | nugget: im doing it in extensions then? |
| 19:39.26 | GordonF | Lame, Noob question here..... Will asterisk run on Knoppix? First time user so I wanna play with it a bit |
| 19:39.31 | Nugget | yes, it's done via the dialplan. |
| 19:39.32 | Qwell | hmm, better make it two |
| 19:39.38 | Nugget | in the context for that iax or sip entry. |
| 19:39.44 | Qwell | GordonF: If Knoppix is Linux, sure, it should |
| 19:39.55 | GordonF | Sweet thanx :) |
| 19:40.01 | Qwell | it would suck to have to redo your configs every time you boot though, heh |
| 19:40.06 | GordonF | Debian I think |
| 19:40.18 | GordonF | was going to do an HDD install |
| 19:40.25 | Qwell | yeah, I was kidding :p |
| 19:40.43 | GordonF | :) |
| 19:41.01 | puppet | nugget: hmm where do i define context for a sipentry? :o im i just stopid now? :) |
| 19:41.04 | shepherd | sms mark |
| 19:41.05 | shepherd | heh |
| 19:41.38 | Nugget | perhaps you'd be well-served by spending more time with the documentation. |
| 19:41.46 | GordonF | cya's later time to break some more software :) |
| 19:41.49 | Nugget | your question is fast approaching "do it for me" territory |
| 19:42.00 | *** part/#asterisk GordonF (GordonF@rrba-146-64-37.telkomadsl.co.za) |
| 19:42.16 | puppet | nugget: ill check documention again |
| 19:44.01 | adorah | Any idea how to stream voice via 2 routers? |
| 19:44.21 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 19:45.31 | *** join/#asterisk bile_one (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 19:46.03 | Inv_arp | adorah: double nat? |
| 19:46.17 | puppet | hmm |
| 19:46.24 | puppet | nugget: i was on the right track wasnt i? |
| 19:46.31 | adorah | <Inv_arp>: Yup:( |
| 19:46.57 | puppet | nugget: if i add /2000 on one f.ex the fax and then under default do an d 2000,1,Dial(SIP/faxnum) |
| 19:47.02 | puppet | nugget: aint that correct? |
| 19:47.48 | Nugget | what is a "f.ex"? |
| 19:47.52 | puppet | for example |
| 19:48.06 | Nugget | add a /2000 where? |
| 19:48.14 | afe | anyone here tried a softphone on a PocketPC with WiFi? |
| 19:48.16 | adorah | Inv_arp: any advice? |
| 19:48.17 | bile_one | does anyone have access to ManxPower's Cepstral setup and configuration pages? He says he has donated them to the Astrisk Documentation Project, but I have not found them there. Any clues? |
| 19:48.27 | puppet | in sip.conf |
| 19:48.34 | Nugget | no, you don't get to choose that. |
| 19:48.39 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
| 19:48.40 | Qwell | afe: I think it was sjphone that worked on a pocketpc? I'd have to ask my friend... |
| 19:48.44 | Nugget | unless your provider is doing weird things |
| 19:48.54 | puppet | nugget: two providers |
| 19:49.00 | Nugget | you do it in extensions.conf, like I said before. |
| 19:49.14 | gr8nash | Qwell X-lite people make a pocketpc phone.. |
| 19:49.23 | *** join/#asterisk tbye (~chatzilla@69.27.4.131) |
| 19:49.32 | afe | Qwell, Xten has a X-Pro for PocketPC, but I was mainly wondering about quality |
| 19:49.44 | Qwell | afe: oh, he said it was working great |
| 19:49.54 | tbye | Is there a tome of all things relating to "All circuits are busy"? |
| 19:49.57 | zipp | diax? |
| 19:50.14 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 19:50.32 | afe | There are some nice PDA/phones available now, and it would be kinda cool to be able to use them with asterisk as well :) |
| 19:51.30 | Inv_arp | adorah: are you port forwarding correctly? |
| 19:53.07 | *** join/#asterisk alex_asterisk (~alex@200.94.71.170) |
| 19:54.08 | adorah | Inv_arp: I believe so..opened all possible ports both ends |
| 19:54.24 | eKo1 | General question: Which country has the least amount of digits in their dialing plan? |
| 19:54.35 | JerJer[mobile] | that one |
| 19:55.50 | *** join/#asterisk dfuller (~dfuller@natty.paycom.net) |
| 19:56.11 | eKo1 | Hmm...I remember seeing some German numbers which are 5 digits long. |
| 19:56.22 | *** join/#asterisk peted20 (~chatzilla@24-113-67-25.wavecable.com) |
| 19:56.46 | Inv_arp | adorah: so any pc on the net can access internal machine? |
| 19:56.54 | dfuller | has anyone gotten atxfer (*2) to work? |
| 19:57.58 | alex_asterisk | dfuller I have it working, but since i'm using spftphone i changed it to ** in order to avoid the timeout between digits |
| 19:58.35 | puppet | nugget: i got right on it I was on right track ;p |
| 19:59.21 | Nugget | I'm so happy for you. |
| 19:59.55 | dfuller | I just performed a brand new install with the latest from CVS but doesn't work for me on a snom 220 |
| 19:59.58 | *** part/#asterisk critch (critch@steven.basesys.com) |
| 20:00.02 | adorah | Inv_ar: right' 4 now.. |
| 20:00.06 | Nugget | CVS HEAD is not guaranteed to work. |
| 20:00.21 | adorah | Inv_arp: yup. right 4 now |
| 20:00.55 | adorah | Inv_arp: if you have a user/pass can use my server to call anywhere |
| 20:01.37 | alex_asterisk | dfuller: straight install from Head didn't work for me, but installing stable then Head works |
| 20:02.07 | dfuller | alex_asterisk: thanks a bunch will give that a try |
| 20:02.08 | mogorman | yeah you guys should read |
| 20:02.13 | mogorman | so muted users can speak |
| 20:02.34 | mogorman | ouch |
| 20:02.42 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 20:03.52 | alex_asterisk | my current setup is all softphone clients. When i dial a number that does not exist i get an error on the softphone, but what i need is to listen and record the telco message that says the number does not exist...any ideas? |
| 20:03.55 | hardwire | anybody used a low bandwidth * box under a uml? |
| 20:04.03 | hardwire | going to attempt doing some IAX trunking via a linode :) |
| 20:04.11 | Inv_arp | adorah: what provider? incoming and outgoing dont work? |
| 20:05.02 | zipp | hardwire I use * on a linode |
| 20:05.07 | hardwire | groovy |
| 20:05.12 | hardwire | did you compile zaptel? |
| 20:05.18 | hardwire | how did you get their source? |
| 20:05.23 | *** join/#asterisk Aviator (~ask@ip-65-111-77-10.customer.accelacom.net) |
| 20:05.27 | *** join/#asterisk jesse_132 (~chatzilla@207.246.72.150) |
| 20:06.07 | zipp | hardwire, you cannot load modules on a linode |
| 20:06.07 | adorah | Inv_arp: I'm outside y'r continent..Israel..the sip sf is logged in when I dial from server get a ring and VV..no voice is heard both ways |
| 20:06.13 | zipp | insecure on uml |
| 20:06.15 | jesse_132 | my phones are behind NAT, and I have NAT=yes in their sip file... but when I do sip show peers it says Nat: N for them... any clues? |
| 20:06.19 | zipp | so caker says no |
| 20:06.21 | hardwire | zipp.. ah.. so no zaptel timer |
| 20:06.26 | zipp | nope |
| 20:06.28 | hardwire | is there another pseudo timer * can use? |
| 20:06.31 | zipp | nope |
| 20:06.38 | hardwire | then I need to use app_conference |
| 20:06.41 | hardwire | vs app_meetme |
| 20:06.47 | zipp | yes |
| 20:06.53 | hardwire | actually.. doesn't IAX trunking require a timer? |
| 20:06.59 | zipp | or get a dedicated server, I got one from serverbeach.com |
| 20:07.01 | zipp | hardwire, yes |
| 20:07.06 | hardwire | well damnit |
| 20:07.08 | hardwire | "_ |
| 20:07.10 | hardwire | err |
| 20:07.11 | hardwire | :) |
| 20:07.20 | zipp | ask caker about it, I know I have |
| 20:07.21 | hardwire | I wonder if linude can include a ztdummy for me |
| 20:07.26 | zipp | he told me no |
| 20:07.30 | hardwire | hmm |
| 20:07.32 | hardwire | those fuckers |
| 20:07.43 | zipp | irc.oftc.net #linode, look for caker |
| 20:07.46 | zipp | he runs it all |
| 20:07.58 | *** join/#asterisk DaLion (~DaLion@Quebec-HSE-ppp224577.qc.sympatico.ca) |
| 20:08.02 | *** join/#asterisk Jackfiber (Jackfiber@82.99.197.209) |
| 20:08.21 | Jackfiber | hello any body is using Handytone 486 behind NAT adsl ? |
| 20:08.28 | hardwire | zipp.. hmm.. ok |
| 20:08.36 | *** join/#asterisk sigmounte_ (~sigmounte@lns-vlq-42-lil-82-252-93-170.adsl.proxad.net) |
| 20:08.41 | sigmounte_ | hi all !! |
| 20:08.43 | zipp | we need a #asterisk-sipnat channel :) |
| 20:09.01 | Jackfiber | yes zipp |
| 20:09.06 | dfuller | alex_asterisk: I get the message on our hard phones as well |
| 20:09.14 | shepherd | #asterisk-omguseiax |
| 20:09.33 | Jackfiber | anybody is using grandstream phone behind nat ? |
| 20:09.35 | eKo1 | Dealing with NAT is easy if everything is on your own network. |
| 20:09.47 | sigmounte_ | just a question , can a use cisco ipphone with asterisk ?? |
| 20:09.48 | Jackfiber | yes it's |
| 20:09.50 | zipp | or you have asterisk convert it to IAX on your network |
| 20:09.57 | zipp | sigmounte_, yes |
| 20:10.07 | zipp | Jackfiber, then you aren't behind nat |
| 20:10.08 | sigmounte_ | YEEES ! thanks ! |
| 20:10.38 | shepherd | or.. sip->nat->asterisk->sip phones |
| 20:10.48 | Jackfiber | the problem here for NAT is the outgoing is fine but no incoming because NAT opens 60695 port for outgoing and asterisk use it for or like that port for incoming while 5060 is default and is forwarded |
| 20:11.04 | *** join/#asterisk jero (~boo@199.243.85.90) |
| 20:11.06 | jero | hello |
| 20:11.11 | hardwire | zipp: http://www.linode.com/irc/logs/linode.log-2004-09-25 |
| 20:11.29 | tzanger | hardwire: no |
| 20:11.30 | tzanger | http://www.craigslist.org/about/best/sfo/60286784.html |
| 20:11.33 | Jackfiber | zipp the phone is |
| 20:11.35 | tzanger | THAT, my friend, is the URL |
| 20:11.42 | sigmounte_ | what else do i need with asterisk and my ipphhone for all to works ? |
| 20:12.01 | eKo1 | sigmounte_: You need someone to call. |
| 20:12.08 | hardwire | tzanger: uh |
| 20:12.11 | coldfeet | does anyone know howto use database deltree it doesnt seem to clear my entries |
| 20:12.18 | Jackfiber | hey any expert is here? |
| 20:12.23 | alex_asterisk | dfuller: any ideas on how to pass on the audio from the telco recording so we can record the message instead of getting the error on the client side? |
| 20:12.25 | sigmounte_ | eKo1, lol |
| 20:13.36 | hardwire | ok |
| 20:13.36 | hardwire | well |
| 20:13.38 | zipp | hardwire, look at the top of that |
| 20:13.42 | zipp | eurozip, that is me :) |
| 20:13.46 | hardwire | the questino is.. if you can derive the timings from the RTC |
| 20:13.56 | hardwire | whats stopping asterisk from being able to do that itself? |
| 20:14.28 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 20:14.56 | Jackfiber | anyone with grandstream handy tone or sip phone? |
| 20:15.32 | eKo1 | Who here has * working with in a multi-homed setup? |
| 20:15.37 | Inv_arp | Jackfiber: oh me me |
| 20:15.57 | hardwire | haha |
| 20:16.01 | zipp | Jackfiber, I simply set up an ipip tunnel from behind nat to the asterisk server, putting the phone and * on the same net |
| 20:16.02 | hardwire | ztdummy requires usb? |
| 20:16.03 | hardwire | damnit |
| 20:16.11 | eKo1 | GS sucks. |
| 20:16.12 | zipp | and it works for me with sip, otherwise, I use IAX2 |
| 20:16.36 | eKo1 | Although someone yesterday was raving about their new phone showcased at VON. |
| 20:16.50 | Jackfiber | zipp, r u using VPN ? |
| 20:16.58 | Jackfiber | I don't wanna use VPN |
| 20:17.01 | zipp | hardwire, ztdummy needs uhci in 2.4, 2.6 it needs no hardware |
| 20:17.06 | zipp | Jackfiber, ipip is not vpn |
| 20:17.11 | hardwire | zipp: thats only for trunking IAX right.. not typical IAX usage. |
| 20:17.24 | zipp | hardwire, and meetme |
| 20:17.30 | hardwire | well fuck meetme then |
| 20:17.36 | zipp | ha |
| 20:17.46 | sigmounte_ | is there somewhere a quick setup guide ? |
| 20:17.50 | Jackfiber | I got X-lite to work it's working properly but handy tone sends out using a port other than 5060 behind NAT !!! |
| 20:18.10 | Jackfiber | do u know how to enforce it not to use anything other than 5060 |
| 20:18.13 | zipp | Jackfiber, look in the handy-tone config, random ports |
| 20:18.14 | AgiNamu | Man, Virbiage doesn't get back to you about anything eh? |
| 20:18.16 | shmaltz | hardwire, you only need usb if you are running 2.4 run 2.6, or get digium hardware |
| 20:18.28 | AgiNamu | I wrote about their USB phones, wrote about G729, wrote about OEM licensing... nary a peep. |
| 20:18.32 | shmaltz | hardwire, any decent app in any os needs a timing source |
| 20:18.37 | hardwire | shmaltz: the issue is that I am attempting to do this within a UML hosted at Linode |
| 20:18.51 | *** join/#asterisk aw (~aw@dialin-145-254-141-169.arcor-ip.net) |
| 20:18.52 | hardwire | shmaltz: for some reason I thought that was what the RTC was for. |
| 20:18.54 | Jackfiber | Zipp, it's set to NO |
| 20:18.54 | adorah | Can anyone reccomend a free IAX softfone client on windows? |
| 20:18.58 | hardwire | I guess I was wrong :) |
| 20:19.04 | Jackfiber | adorah, X-lite |
| 20:19.06 | AgiNamu | adorah: www.virbiage.com -- FireFly |
| 20:19.09 | zipp | adorah, iaxclient |
| 20:19.13 | hardwire | X-lite |
| 20:19.19 | hardwire | :) |
| 20:19.19 | AgiNamu | X-line doesnt do IAX..... |
| 20:19.21 | shmaltz | I think that the RTC is only use by the CPU |
| 20:19.23 | AgiNamu | X-lite sucks anywyas |
| 20:19.26 | hardwire | I know.. its fun to say |
| 20:19.26 | adorah | Jackfiber: does not support IAX only sip |
| 20:19.26 | Jackfiber | X-lite is Sip |
| 20:19.27 | AgiNamu | stupid interface :@ |
| 20:19.35 | AgiNamu | Other than that, it's a cool think. |
| 20:19.35 | hardwire | shepherd: ERTC in 2.6 |
| 20:19.38 | AgiNamu | thing. |
| 20:19.45 | zipp | iaxcomm, iaxclient, diax |
| 20:19.48 | hardwire | zaprtc (a module) also exists to interperate the RTC into ZAP Timings |
| 20:19.53 | Jackfiber | for IAX u need to use iaxclient |
| 20:19.54 | zipp | iaxclient.sf.net |
| 20:19.54 | ariel_ | hardwire, if your running kernel 2.6.X you should be able to use ztdummy without a usb. |
| 20:19.55 | AgiNamu | Zipp: aren't all of those ugly? |
| 20:19.59 | adorah | Thx |
| 20:20.01 | AgiNamu | What's wrong with FireFly? |
| 20:20.05 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 20:20.08 | AgiNamu | Firefly works great, and is attractive, nicely designed too. |
| 20:20.10 | hardwire | ariel_: the problem being that linode does not have module loading enabled on their umls |
| 20:20.15 | Jackfiber | Zipp, do u have handytone ? |
| 20:20.20 | zipp | AgiNamu, I use iaxcli from tkphone in the console, so I don't know |
| 20:20.27 | zipp | Jackfiber, I have a budgetone |
| 20:20.37 | Jackfiber | zipp, so they r similar |
| 20:20.37 | hardwire | or an RTC for that matter.. damn |
| 20:20.53 | Jackfiber | zipp, have u set NAT traversal to YES ? |
| 20:20.56 | ariel_ | hardwire, then don't use trunking nor meetme's |
| 20:21.00 | jluk | hardwire: if you've not got usb or zap hardware, use 2.6. I think ztdummy is needed for MOH as well as meetme |
| 20:21.05 | zipp | Jackfiber, as I said, I use an IPIP tunnel |
| 20:21.12 | *** join/#asterisk xD (~BombTrack@200.126.66.66) |
| 20:21.15 | jakepdev | in the Wiki, it specifies 2 implementations of H.323 but it only give the link of one. Where is this "source tree" for *? |
| 20:21.19 | shmaltz | hardwire, what you trying to do? testing? |
| 20:21.19 | hardwire | jluk: thanks.. heh.. wow.. never though of it that way :) |
| 20:21.22 | Jackfiber | IPIP tunnel is kinda VPN |
| 20:21.23 | ariel_ | moh does not need ztdummy nor any timing |
| 20:21.28 | zipp | jluk, he can't use ztdummy at all, linode won't let you load modules |
| 20:21.28 | aw | Does anyone know how to send a busy tone over capi without answer the call? The Hangup command don't work as expected. |
| 20:21.33 | shmaltz | or setting up a confferincing server? |
| 20:21.35 | Jackfiber | or just u forwarded ports ? |
| 20:21.40 | hardwire | shmaltz: just testing |
| 20:21.47 | *** part/#asterisk xD (~BombTrack@200.126.66.66) |
| 20:21.48 | hardwire | nothing more |
| 20:21.57 | shmaltz | get a stupid old computer off ebay, and test it |
| 20:21.59 | AgiNamu | So anyone here know the details on the SGI+Asterisk bundle? |
| 20:22.03 | hardwire | trying to get a server in the Lower 48 with a good ping time to the sip providers |
| 20:22.05 | zipp | hardwire, you cannot get a timer on a linode |
| 20:22.09 | hardwire | linode isn't going to be a good solution |
| 20:22.12 | AgiNamu | "the sip providers"? |
| 20:22.13 | AgiNamu | heheh |
| 20:22.15 | shmaltz | you trying to save electricity as well? |
| 20:22.16 | hardwire | zipp: I am noticing this |
| 20:22.18 | shmaltz | hehe |
| 20:22.26 | hardwire | shmaltz: no. |
| 20:22.33 | ariel_ | hardwire, how about vmware? |
| 20:22.37 | AgiNamu | Yea, those dang sip providers |
| 20:22.49 | hardwire | shold on |
| 20:22.51 | hardwire | phone |
| 20:23.23 | DaLion | guys |
| 20:23.25 | tbye | in zapata.conf is it supposed to be channels => 4 or channels=4? (I've got a tdm400p with the module in tel4) |
| 20:23.34 | coldfeet | does asterisk have alimit on the length of username, I seem to be able to do 8 chars but not 9 |
| 20:23.57 | DaLion | how can we return a var from a perl or php initiated manager telnet session back to php or perl /? |
| 20:24.01 | shmaltz | you can still use your test machine from home, and then connect your machine to your uml, and have your uml connect to sip providers |
| 20:24.04 | hardwire | ok |
| 20:24.07 | jakepdev | ariel - just a note on vmware - need to disable zaptel - or no audio from * |
| 20:24.08 | boch | http://pastebin.ca/7196 <- do you know why this happens? |
| 20:24.09 | hardwire | why do I need less ping times to the sip providers |
| 20:24.13 | hardwire | because |
| 20:24.14 | *** join/#asterisk Xander77 (~Alex@exten-halls-243.soton.ac.uk) |
| 20:24.25 | hardwire | I plan on using IAX trunking from alaska.. and starband that terminates in Georgia |
| 20:24.31 | hardwire | as well as another VSAT provider in utah |
| 20:24.41 | AgiNamu | No, im saying "the sip providers" |
| 20:24.41 | DaLion | whoever talking shold lower volume |
| 20:24.41 | DaLion | hardwire |
| 20:24.46 | hardwire | and going from utah to anchorage where I am.. all the way to new york isn't my idea of fun |
| 20:24.50 | hardwire | or georgia |
| 20:24.54 | shmaltz | hardwire, what connection to you have to the intarweb? |
| 20:25.20 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 20:25.37 | hardwire | shmaltz: from all three points .. very laggy |
| 20:25.49 | shmaltz | I c |
| 20:25.52 | hardwire | more lag if I have to go up to anchorage again just to go back to the lower 48 |
| 20:25.53 | hardwire | so |
| 20:25.53 | shmaltz | get a colo then |
| 20:26.00 | *** part/#asterisk Jackfiber (Jackfiber@82.99.197.209) |
| 20:26.05 | hardwire | shmaltz: like I said.. linode was just a test |
| 20:26.07 | hardwire | this was only a test |
| 20:26.08 | NK123 | hi kiran r u there |
| 20:26.14 | hardwire | had this been a real need.. I would have colocated :) |
| 20:26.18 | shmaltz | so for testing, no meetme |
| 20:27.15 | DaLion | Anyone knows AGI well enough ? to answer how can we return a var from a perl or php initiated manager telnet session back to php or perl /? |
| 20:27.27 | jakepdev | i just built an AGi test |
| 20:28.00 | DaLion | so php -> manager -> * --> AGI --and abck |
| 20:28.24 | jakepdev | * -> AGI |
| 20:28.27 | DaLion | basically i need a WAITEXTEN.. from agi wich i need returned ..it always exits on zero |
| 20:28.42 | jakepdev | AGI processess in perl |
| 20:28.45 | DaLion | any ideas ? |
| 20:29.11 | jakepdev | hmm |
| 20:29.14 | DaLion | whatever .. i mean.. how to .. RETURN vars from AGI when it ends |
| 20:29.24 | JunK-Y | DaLion: waitexten for what exactly? |
| 20:29.31 | DaLion | like WAITEXTEN(2) |
| 20:29.38 | DaLion | enter 23 in phone |
| 20:29.44 | DaLion | and AGI would exit with 23 |
| 20:29.51 | hardwire | shmaltz: and no iax trunking |
| 20:29.53 | JunK-Y | why not GET DATA? |
| 20:29.53 | DaLion | so i can grab the exit code of agi from manager |
| 20:29.56 | hardwire | manditory for my testing |
| 20:30.06 | AgiNamu | you can't get the exit code. |
| 20:30.08 | AgiNamu | Use a channel var. |
| 20:30.09 | AgiNamu | that's it. |
| 20:30.13 | DaLion | well.. hmm.. |
| 20:30.25 | DaLion | can u see var form manager port ? |
| 20:30.25 | AgiNamu | At least, from the dialplan. Maybe the manager works better. |
| 20:30.31 | JunK-Y | AgiNamu: with GET DATA, ya can get the results yeah. |
| 20:30.38 | DaLion | CONF guys.. any idea ? |
| 20:30.40 | shmaltz | hardwire, I didn't know IAX need a timing source, but it makes sense. |
| 20:30.45 | gr8nash | anyone tell me where you create "a new context" is it iax.conf? |
| 20:30.55 | DaLion | gr8 |
| 20:31.00 | DaLion | try exensions.conf |
| 20:31.02 | shmaltz | gr8nash, RTFM |
| 20:31.15 | DaLion | GET DATA, |
| 20:31.20 | DaLion | in manager ? cool |
| 20:31.20 | AgiNamu | gr8nash, actually, you do it on google. |
| 20:31.35 | DaLion | is get data docu'ed ? |
| 20:31.59 | jakepdev | http://home.cogeco.ca/~camstuff/agi.html |
| 20:32.01 | JunK-Y | DaLion: |
| 20:32.04 | JunK-Y | ~agi apui |
| 20:32.05 | jakepdev | dalion - here's agi docs |
| 20:32.05 | JunK-Y | ~agi api |
| 20:32.06 | jbot | methinks agi api is at http://home.cogeco.ca/~camstuff/agi.html |
| 20:32.22 | JunK-Y | a new version should be make too. |
| 20:33.01 | DaLion | omg |
| 20:33.03 | DaLion | thanks |
| 20:33.08 | DaLion | iou |
| 20:33.39 | jakepdev | where is the source tree for *? |
| 20:33.54 | DaLion | depends on install |
| 20:33.55 | zipp | jakepdev, in cvs, /join #asterisk-dev |
| 20:34.00 | jakepdev | ok |
| 20:34.06 | gr8nash | DaLion, thanks btw |
| 20:34.06 | DaLion | ./usr/src/asterisk |
| 20:34.07 | jakepdev | Asterisk@home |
| 20:34.13 | DaLion | n/p |
| 20:34.34 | gr8nash | shmaltz, why does this channel exist? |
| 20:34.42 | gr8nash | political debates? |
| 20:34.45 | yaaar | anybody in here know what Linux softphones are available that work well with asterisk? |
| 20:34.48 | gr8nash | coffe talk? |
| 20:34.49 | shmaltz | for help |
| 20:34.55 | zipp | yaaar, iaxclient |
| 20:35.00 | Qwell | yaaar: iaxcomm |
| 20:35.01 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 20:35.06 | gr8nash | was my question out of bounds?? |
| 20:35.07 | shmaltz | gr8nash, you were using it as a manual |
| 20:35.11 | shmaltz | yep |
| 20:35.17 | gr8nash | heh.. |
| 20:35.31 | shmaltz | it wasnt a qustions for help, it was a question that said I'm lazy |
| 20:35.32 | yaaar | cool |
| 20:35.47 | gr8nash | im part of many channels.. dont know why this one is the one with the most attitude |
| 20:35.50 | shmaltz | it's clearly in the asterisk handbook |
| 20:35.57 | mogorman | hey |
| 20:36.00 | mogorman | thats really cold |
| 20:36.01 | shmaltz | ~docs |
| 20:36.02 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:36.21 | Inv_arp | gr8nash: create a context to do? |
| 20:36.26 | KalD|Work | anyone got time to help me solve a stupid dialplan bug? |
| 20:36.37 | shmaltz | gr8nash, if the channel would be for such questions like yours, than we wouldn't be able to make asterisk run |
| 20:36.58 | shmaltz | KaID|Work, shoot |
| 20:36.59 | KalD|Work | I'm trying to setup a simple dialplan that prompts for creating or joining a conferenece and it just hangs up on me =( here is the pastebin: http://pastebin.ca/7197 |
| 20:37.00 | gr8nash | nope.. it was a simple question. .not everything makes sense to everybody |
| 20:37.18 | yaaar | zipp; Qwell; you guys wouldn't happen to know whether those are part of a gentoo package? or one that is? |
| 20:37.25 | shmaltz | gr8nash, have you read the handbook? |
| 20:37.27 | gr8nash | i have read a small book so far on astrisk.. you dont have a clue how much i work |
| 20:37.28 | yaaar | i'm willing to use SIP instead of IAX if necessary |
| 20:37.45 | shmaltz | gr8nash, the handbook? |
| 20:37.45 | gr8nash | shmaltz, i have read the wike and the "docs" |
| 20:37.50 | gr8nash | not sure.. |
| 20:37.59 | shmaltz | gr8nash, the handbook? |
| 20:38.04 | shmaltz | not sure, RTFM |
| 20:38.09 | yaaar | shmaltz: gr8nash don't you guys think you're spending an odd amount of time on the question of whether the question is ok? |
| 20:38.10 | shmaltz | that's my point |
| 20:38.36 | gr8nash | yaar i agree |
| 20:38.40 | shmaltz | well, yaar, if this will make gr8nash read the handbook, then i saved lots of time |
| 20:38.47 | machinehd | when using ztdummy what needs to be changed in zapata.conf? |
| 20:38.52 | gr8nash | im done.. my point is rudeness sucks.. mean people blow |
| 20:39.09 | shmaltz | gr8nash, I am not mean |
| 20:39.10 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 20:39.19 | shmaltz | I just gave you a hint where you can find the info |
| 20:39.20 | KalD|Work | shmaltz, did you see the pastebin? |
| 20:39.26 | gr8nash | maybe not but the capslock was unecesary |
| 20:39.26 | *** join/#asterisk w0w0 (~w0w0@80-28-166-80.adsl.nuria.telefonica-data.net) |
| 20:39.28 | shmaltz | it's in the handbook, which in short is RTFM |
| 20:39.41 | shmaltz | abrivs always go in CAPS |
| 20:39.52 | shmaltz | ~RTFM |
| 20:39.53 | jbot | i guess rtfm is read the f*cking manual... try asking me about "FAQ" |
| 20:40.11 | shmaltz | ~caps |
| 20:40.13 | jbot | caps is, like, Don't write words all in capital letters unless they're abbreviations. To emphasize words put _*_ on both sides of them. |
| 20:40.37 | shmaltz | sorry, RTFM again this time about caps |
| 20:40.48 | MikeJ[Jayden] | ~rtfw |
| 20:40.49 | jbot | rtfw is probably Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 20:41.11 | *** join/#asterisk TokyoJimu (~jimmy@198.51.175.64) |
| 20:41.26 | KalD|Work | ~jbot |
| 20:41.27 | jbot | from memory, jbot is the shipboard computer, but you may call me eddie if it helps you relax |
| 20:41.29 | shmaltz | KaID|Work, nope give it to me again |
| 20:41.31 | gr8nash | no i handt seen that.. thanks for the help |
| 20:41.41 | KalD|Work | shmaltz, http://pastebin.ca/7197 |
| 20:41.46 | shmaltz | gr8nash, thanks |
| 20:41.57 | shmaltz | sorry if you felt I was mean |
| 20:42.06 | gr8nash | yeah me too.. |
| 20:42.51 | CosmicRay | anyone here use voipuser.org? |
| 20:43.14 | TokyoJimu | What happened to app_qcall? Was it replaced by something else? |
| 20:43.30 | shmaltz | KaID|Work, the dialplan has no rule after s,5 |
| 20:43.38 | KalD|Work | yeah |
| 20:43.43 | shmaltz | it has nothing to do so it falls through |
| 20:43.45 | KalD|Work | so after s,5 I wanna enter in 1 |
| 20:43.58 | shmaltz | you need: |
| 20:44.00 | shmaltz | exten => s,6,Goto(1,1) |
| 20:44.11 | KalD|Work | then it would always goto 1 |
| 20:44.15 | KalD|Work | I might want to enter 2 |
| 20:44.17 | KalD|Work | or something else |
| 20:44.21 | shmaltz | or you can change 1,1 to s,6, 1,2 to s,7 ans so on |
| 20:44.33 | shmaltz | so do cmd waitexten |
| 20:44.46 | KalD|Work | ... It is supposed to answer the phone and say press 1 for so and so - press 2 for whatever... then I enter 1 or 2 |
| 20:45.03 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 20:45.07 | KalD|Work | ... brb |
| 20:45.25 | shmaltz | waitexten will wait for 1 or 2 to be entered. |
| 20:45.43 | shmaltz | KaID|Work, waitexten will wait for 1 or 2 to be entered. |
| 20:45.44 | DaLion | Seems its always same questins here ;) |
| 20:45.48 | *** join/#asterisk tbye (user@69.27.4.138) |
| 20:45.50 | *** part/#asterisk dfuller (~dfuller@natty.paycom.net) |
| 20:46.19 | *** part/#asterisk alex_asterisk (~alex@200.94.71.170) |
| 20:46.26 | tbye | Is it possible to probe a tdm400p to see if it can detect if the PSTN is connected or not? |
| 20:46.52 | shmaltz | KaID|Work, change the playback to Background, this way you can enter extensions when playing the message |
| 20:50.10 | jakepdev | has anyone got H323 working in *? |
| 20:52.43 | johnnyb | Does asterisk support G728? I couldn't find any information about such support, but I also couldn't find any information (apart from obviously developer time) which would prevent such support (patents, etc.). Anyone know anything about it? |
| 20:53.12 | eKo1 | There's a G.728? |
| 20:53.18 | dan2 | no |
| 20:53.25 | dan2 | johnnyb: do you mean g729? |
| 20:53.37 | eKo1 | johnnyb: Stop smoking dem chiba. |
| 20:53.38 | johnnyb | Nope, G728. It's low-latency AND low-bandwidth. |
| 20:53.49 | johnnyb | It's ships w/ the grandstream phones |
| 20:53.59 | johnnyb | http://www-mobile.ecs.soton.ac.uk/speech_codecs/standards/g728.html |
| 20:54.22 | johnnyb | This has a sample implementation, which is why I was surprised that Asterisk didn't support it. |
| 20:54.32 | machinehd | with ztdummy do I delete zapata.conf? |
| 20:55.07 | eKo1 | g728 is 16kbps |
| 20:55.09 | DaLion | if you dont get answer.. try the teliax forum.. http://www.teliax.com/forum/ not much yet... but if you dont get your answers here.. might be worth try there.. |
| 20:55.10 | eKo1 | that sucks |
| 20:55.27 | dan2 | johnnyb: g728 uses an extreme amount of processor |
| 20:55.38 | eKo1 | might is well use adpcm 16 |
| 20:55.46 | *** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au) |
| 20:55.51 | boch | http://pastebin.ca/7196 <- do you know something about this? |
| 20:56.28 | dan2 | adpcm 16 is owned by M$ |
| 20:57.11 | dan2 | g728 is relatively new as well, it was only released as of January last year iirc |
| 20:57.39 | dan2 | hmm never mind |
| 20:57.54 | dan2 | johnnyb: grandstreams are the only one that support this codec |
| 20:59.27 | *** join/#asterisk gst (~gst@wireless.sysfrog.org) |
| 20:59.29 | dan2 | now, ilbc is a GOOD codec |
| 20:59.41 | dan2 | and so is speex for that matter |
| 20:59.53 | eKo1 | yeah right. |
| 21:01.01 | clive- | cooo, just got my phone talking iax2 |
| 21:01.29 | zipp | clive-, using an iaxy? |
| 21:01.30 | clive- | bye bye sip and nat |
| 21:01.33 | DaLion | is anyone from conf READING THIS ? |
| 21:01.37 | DaLion | gaim is GREAT IDEA ! |
| 21:01.45 | clive- | zipp its a pa168 ip phone |
| 21:01.51 | DaLion | new eyebeam has chat enabled also |
| 21:01.56 | *** join/#asterisk phantasis (~phantasis@c68.190.174.244.eau.wi.charter.com) |
| 21:01.56 | zipp | clive-, ah, you can get them from iaxtalk.com |
| 21:02.06 | zipp | that firmware isn't extremely stable yet |
| 21:02.17 | clive- | zippp, have you tried them? |
| 21:02.21 | santiago | Hi, only to be sure, with QoS, I have only to set more priority to the 4569 upd port if the protocol used between two * is iax2, isn't it? |
| 21:02.32 | zipp | clive-, what I have heard, never had a pa168 phone |
| 21:02.34 | clive- | I just loaded the firmware like 15 minutes ago |
| 21:02.35 | dan2 | twisted[work]: yo |
| 21:02.37 | DaLion | y |
| 21:02.44 | DaLion | santiago si senior |
| 21:02.45 | phantasis | what would I need to do to use asterisk with nearly all analog phones (2000) and a handful of voip phones |
| 21:02.53 | clive- | so I cant say how stable it is, but it works..:))) |
| 21:02.56 | zipp | phantasis, channel banks |
| 21:02.59 | LoRez | phantasis: lots of channel banks |
| 21:02.59 | santiago | DaLion, gracias |
| 21:03.01 | DaLion | siantiago mihgt add 5060 too while there |
| 21:03.23 | zipp | santiago, you only need 1 port for iax2 |
| 21:03.33 | LoRez | phantasis: E1 chanbanks at that. |
| 21:03.33 | santiago | iax2 is great |
| 21:03.34 | clive- | zipp what else did you hear about the pa168's ? |
| 21:03.34 | DaLion | santiago de nada.. se fui muy rapido |
| 21:03.38 | Darwin35 | ok asterisk got updated to 1.0.6 today |
| 21:03.41 | tbye | anyone have a slick way to check an fxo on a tdm400p to see if it hears dialtone? |
| 21:03.42 | Darwin35 | in ports ye |
| 21:03.44 | Darwin35 | s |
| 21:03.52 | clive- | santiago iax2 is great, excpet very few phones support it |
| 21:04.12 | phantasis | correct if wrong, a channel bank would take 24 lines and but them into a T1 to interface with a te410p? |
| 21:04.46 | gst | hmmm... i try to get a sip phone with a dynamic ip to work with asterisk, but when the phone tries to REGISTER asterisk just responds with 403 forbidden. the section of the phone in the sip.conf contains the host=dynamic, etc. stuff - but it still fails :/ any hints? |
| 21:05.10 | clive- | gst try auth=md5 |
| 21:05.20 | Darwin35 | what type of phone |
| 21:05.34 | Darwin35 | did you set the secret and the username |
| 21:05.50 | Darwin35 | and the exten # |
| 21:06.25 | gst | Darwin35: snom100 |
| 21:06.42 | Darwin35 | hmm |
| 21:06.53 | Godsey | I don't need to run cvshead, I can just use AGI to pull data from mysql in my dialplan |
| 21:07.00 | Godsey | I thought about it over lunch :) |
| 21:07.05 | gst | [client-gst] |
| 21:07.05 | gst | type=friend |
| 21:07.05 | gst | context=from-sip-gst |
| 21:07.05 | gst | username=gst |
| 21:07.05 | gst | secret=test |
| 21:07.07 | gst | host=dynamic |
| 21:07.09 | gst | nat=yes |
| 21:07.12 | *** join/#asterisk lordcian (~john@209.194.32.60) |
| 21:07.14 | gst | auth=md5 |
| 21:07.15 | gst | this is the section in my sip.conf |
| 21:07.37 | gst | asterisk logs: Mar 10 22:06:12 NOTICE[28613]: chan_sip.c:7654 handle_request: Registration from '"gst" <sip:gst@eris.sysfrog.org>' failed for '62.116.93.254' |
| 21:08.01 | lordcian | hi, im getting same issue |
| 21:08.07 | gst | i also tried some of the examples in the sip.conf file which didn't work either :/ |
| 21:08.39 | Darwin35 | you only put gst in the phone not client-gst I bet |
| 21:09.17 | Darwin35 | you should have fallowed the snom base setup in the sip.conf sample |
| 21:09.24 | lordcian | do you mean the peer definition area in the sip.conf by 'client-gst'? |
| 21:09.29 | phantasis | it'd be better to get a channel bank that supports ethernet instead of t1 in a lan environment where alot of analogs are going to be used, correct? |
| 21:09.35 | yaaar | anybody in here use x-lite on windows to talk to asterisk? how the hell do you reconfigure it once it's running? I can't find any buttons that seem like they would work' |
| 21:09.50 | tzanger | it's in the middle |
| 21:09.53 | tzanger | between pickup and hangup |
| 21:09.55 | lordcian | download the new x-lite client from the home site... |
| 21:10.02 | tzanger | it looks like a box with smaller boxes in it IIRC |
| 21:10.06 | tzanger | I haven't used xlite in quite a while |
| 21:10.11 | lordcian | then use the menu button, click around till you find the 'default' section |
| 21:10.28 | tzanger | I fucking HATE these skinned apps, so fucking hard to use |
| 21:10.33 | tzanger | but they're "snazzy" so they stay... ugh |
| 21:11.06 | gst | lordcian: tnx - that was the problem 'client-gst' vs. 'gst' |
| 21:11.07 | gst | :) |
| 21:11.24 | Darwin35 | II thought it might be |
| 21:11.28 | Darwin35 | heheh |
| 21:11.35 | clive- | detective darwin:) |
| 21:11.36 | lordcian | wait, mine still isnt working.. :( |
| 21:11.44 | yaaar | tzanger: i don't get it....but there is nothing in between the buttons labelled 'dial' and 'hangup' |
| 21:11.54 | tzanger | there are three icons in there I think |
| 21:12.06 | tbye | If asterisk sees a call coming in on an fxo is there a log entry created? where? |
| 21:12.07 | lordcian | you have the wrong version of xlite |
| 21:12.12 | yaaar | oh |
| 21:12.16 | yaaar | hrm |
| 21:12.23 | lordcian | yours is square box? |
| 21:12.26 | Nugget | <PROTECTED> |
| 21:12.29 | lordcian | (yaar?) |
| 21:12.34 | lordcian | heh |
| 21:12.41 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-129-44-136-89.buff.east.verizon.net) |
| 21:12.43 | eKo1 | Nugget: hehe |
| 21:12.45 | ManxPower | YES!!!!!!!!!!!!! Sipura has released an updated firmware for the SPA-841 (and it fixes the last major problem with the phone that I have) |
| 21:12.46 | SuPrSluG | hello |
| 21:12.47 | antifuchs | Nugget: audio cocks are one of jwz's greater inventions |
| 21:13.00 | eKo1 | ManxPower: What problem? |
| 21:13.10 | Nugget | actually jwz didn't invent audiocock technology. |
| 21:13.16 | Nugget | Makali did. |
| 21:13.21 | ManxPower | eKo1, too low microphone volume for handset and speakerphone and it was not adjustable. |
| 21:13.25 | shmaltz | ManxPower someone else on the list was complaining about the 841 |
| 21:13.34 | shmaltz | gtg |
| 21:13.36 | shmaltz | bey |
| 21:13.37 | antifuchs | Nugget: it was on his journal though, or am I misremembering? |
| 21:13.42 | shmaltz | I meant bye |
| 21:13.49 | Nugget | http://www.jwz.org/doc/linuxvideo.html |
| 21:14.00 | SuPrSluG | why do I get this error? Mar 10 16:13:42 NOTICE[23028]: chan_sip.c:8343 handle_request: Failed to authenticate user "2001" <sip:2001@192.168.1.1:5060 for SUBSCRIBE |
| 21:14.01 | Darwin35 | man this rocks the new pbx board and the mini drive and I have a sip pbx in a box |
| 21:14.11 | Darwin35 | but I have to fight snom now |
| 21:14.22 | Darwin35 | the boxes look the same |
| 21:14.53 | SuPrSluG | phone=polycom 500 |
| 21:15.26 | Darwin35 | well first check username exten name and passwd |
| 21:15.55 | Darwin35 | make sure they match phone for sip.coonf |
| 21:16.19 | SuPrSluG | k |
| 21:16.44 | phantasis | if i am connecting analof phones to a channel bank I want FXS or FXO |
| 21:16.54 | Darwin35 | FXS |
| 21:17.03 | cypromis | xfs |
| 21:17.05 | Darwin35 | fxo is for phoneline to pbx |
| 21:17.31 | Darwin35 | FXS is for putting a phone into * |
| 21:17.35 | Nugget | FXO talks to a dial tone. FXS creates a dial tone. |
| 21:17.39 | Darwin35 | aka the FXS creates dial tone |
| 21:17.55 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
| 21:18.14 | Darwin35 | Nugget your on the same wave length as me |
| 21:18.22 | Nugget | <-- sine wave |
| 21:18.25 | Darwin35 | stop hogging the bandwidth |
| 21:18.25 | phantasis | so asterisk would be the FXO in that situation? |
| 21:18.41 | Nugget | my traffic is shaped - - I can't hog bandwidth! |
| 21:18.57 | TheBear | trying for the 1st time with AGI. I have 'use Asterisk::AGI; (from festival weather config) yet I get the error "Can't locate Asterisk/AGI.pm in @INC (@INC contains: /etc/perl.... " what did I do worng ? |
| 21:18.58 | Darwin35 | phan what card do you have |
| 21:19.08 | phantasis | just planning/thinking |
| 21:19.13 | phantasis | what kind would I need |
| 21:19.14 | ManxPower | TheBear, You need to INSTALL asterisk-perl. |
| 21:19.14 | phantasis | 410? |
| 21:19.29 | clive- | nugget they shape my bandwidth too,,,,where you from? |
| 21:19.37 | ManxPower | SuPrSluG, For the most part Asterisk does not support SUBSCRIBE. |
| 21:19.41 | Nugget | where am I from or where do I live? |
| 21:19.41 | TheBear | ManxPower: from where ? |
| 21:19.41 | eKo1 | TheBear: You didn't do anything and that is wrong. |
| 21:20.01 | ManxPower | TheBear, I don't recall. Check the digium docs page. |
| 21:20.02 | TheBear | I have asterisk-addons |
| 21:20.04 | ManxPower | ~docs |
| 21:20.06 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:20.09 | clive- | :)...where is your shaped bandwidth from |
| 21:20.16 | SuPrSluG | ManxPower:what's trying to subscribe? |
| 21:20.23 | Nugget | I shape it. |
| 21:20.35 | ManxPower | SuPrSluG, The device at 192.168.1.1 |
| 21:20.37 | Darwin35 | TheBear this is s eprate part yo have to fetch |
| 21:20.41 | RaYmAn-Bx | Nugget: by hand? :P |
| 21:20.47 | Darwin35 | heheh |
| 21:20.52 | Nugget | I pass it through a Play-doh fun factory. |
| 21:20.59 | clive- | wel in south africa the isp shapes it,,,the scum |
| 21:21.01 | Nugget | it comes out looking like a star |
| 21:21.19 | Darwin35 | hehe mine looks like a long hose |
| 21:21.52 | eKo1 | Hmm...I completely forgot about todays conference. |
| 21:22.09 | Darwin35 | what time |
| 21:22.21 | eKo1 | Oh well, next time. |
| 21:22.39 | Darwin35 | grr missed it |
| 21:22.54 | TheBear | ManxPower: http://asterisk.gnuinter.net/files/asterisk-perl-0.08.tar.gz |
| 21:23.04 | Darwin35 | thats it |
| 21:23.55 | yaaar | will asterisk fail to run if there aren't any outbound channels? I figured I'd try to get calls working between extensions first, but none of my extensions could connect.....turns out asterisk isn't listening, and the logs don't show anything except warnings about lack of outgoing channels |
| 21:24.25 | yaaar | when i run '/etc/init.d/asterisk start' it says it starts, but the only process it adds to ps waux is an mpg123 process |
| 21:24.42 | phantasis | each TDM400P can only support 4 analog lines per card? |
| 21:25.51 | *** join/#asterisk xantus (~david@66.165.228.13) |
| 21:25.57 | mog_home | yes |
| 21:26.13 | yaaar | my full asterisk log is at http://www.pastebin.com/252749 |
| 21:26.23 | mog_home | yaaar run asterisk -vvvvc, see why it is crashing |
| 21:26.29 | Darwin35 | yar did you make libpri and zaptel |
| 21:26.30 | yaaar | thx one sec |
| 21:26.37 | Darwin35 | did you fallow the directions |
| 21:27.00 | *** join/#asterisk bah (048830696@ACAA21EA.ipt.aol.com) |
| 21:27.04 | Darwin35 | mail call |
| 21:27.25 | puppet | Is there any way to sneaklisten to channels? Sip not zap, i saw that chanspy command didnt exist anymore? |
| 21:27.30 | Darwin35 | beck luan stacy tracy steve dan ron terry |
| 21:27.44 | yaaar | Darwin35: well, after a fashion. I was working from the AMP installation howto, mostly, and installed almost everything from .debs on ubuntu hoary |
| 21:28.00 | lordcian | yaaar, and DONT forget ztcfg |
| 21:28.17 | yaaar | lordcian: yeah, ran that, although i was defining 0 channels... |
| 21:28.17 | Bacon | Anyone here using Asterisk with BroadVoice? |
| 21:30.14 | mesi | Anybody on for a chat on a conference room? |
| 21:31.50 | yaaar | ok, the output of asterisk -vvvvc is at http://www.pastebin.com/252751 ....it all looks fine to me (albeit I don't exactly know what the hell I'm doing) until the last couple of lines that say "Ouch ... error while writing audio data: : Broken pipe" |
| 21:32.10 | ManxPower | puppet, chan_spy was NEVER part of the official Asterisk source. |
| 21:32.27 | CosmicRay | yaaar: I wouldn't work from the AMP howto for a debian install |
| 21:32.38 | CosmicRay | yaaar: I'd apt-get install asterisk and then use the quickstart guide on the wiki |
| 21:32.42 | CosmicRay | or on asteriskdocs.org |
| 21:32.49 | yaaar | CosmicRay: well, I changed it where appropriate.... |
| 21:33.07 | puppet | manxpower: oh ok |
| 21:33.41 | Darwin35 | wow asterisk on ubuntu |
| 21:33.47 | Darwin35 | thats a feet in itself |
| 21:34.02 | yaaar | Darwin35: yeah, i know....kind of funny. but it was a box that was already installed and doing a whole lot of nothing |
| 21:34.02 | dan2 | asterisk has been in debian for sometime now |
| 21:34.11 | CosmicRay | hell, asterisk is in *stable* |
| 21:34.21 | dan2 | CosmicRay: :) |
| 21:34.21 | yaaar | Darwin35: of course, it *would* be a feat....if it worked |
| 21:34.27 | Darwin35 | heheh |
| 21:34.36 | dan2 | CosmicRay: if asterisk isn't stable at a 1.0 release how stable can it be in woody |
| 21:34.48 | CosmicRay | yaaar: I just did an install on my alpha 2 days ago, first time ever asterisk install for me |
| 21:34.50 | CosmicRay | worked great |
| 21:34.54 | Darwin35 | well I had to be the porter to fbsd and not given up yet and now 98% of it works |
| 21:34.55 | yaaar | but where can I look to see what's causing this audio data error? |
| 21:35.18 | CosmicRay | yaaar: edit /etc/default/asterisk, uncomment the PARAMS line with all the -vvv in it |
| 21:35.23 | CosmicRay | then /etc/init.d/asterisk restart |
| 21:35.26 | Darwin35 | I have * running on a Dec alpha 21264/dual 600 |
| 21:35.27 | CosmicRay | debugging will be dumped to your console |
| 21:35.27 | yaaar | k |
| 21:35.40 | Darwin35 | in the loft in toronto |
| 21:35.46 | CosmicRay | Darwin35: sweet. mine is a 21164a 600MHz (LX164) |
| 21:35.53 | Darwin35 | nice |
| 21:36.07 | CosmicRay | I bought it new probably 8 years ago |
| 21:36.09 | Darwin35 | mine is on fbsd and I will keep it thre |
| 21:36.15 | CosmicRay | it was a top-of-the-line workstation back then |
| 21:36.31 | Darwin35 | mine I bought on ebay a while back |
| 21:36.54 | yaaar | CosmicRay: I got no debugging from that....just said 'starting asterisk PBX' and then gave my prompt back. also started that same mpg123 process, but no others |
| 21:37.06 | Darwin35 | was going to make a master server out of it now it servs as the loft phone and x server |
| 21:37.13 | phantasis | how does asterisk run on solaris 10/sparc? |
| 21:37.44 | Darwin35 | not played with sparc yet |
| 21:37.49 | puppet | Can I restart the musiconhold mp3 stream without stop now? |
| 21:37.55 | Darwin35 | and I will not touch solaris 10 |
| 21:37.59 | CosmicRay | yaaar: that's wacky. what version asterisk have you installed? (dpkg -l asterisk) |
| 21:38.04 | TheBear | anyone using the festival weather config ? I'm getting illegal port command ? |
| 21:38.05 | CosmicRay | Darwin35: I will not touch solaris. |
| 21:38.11 | phantasis | why not 10? I installed 10 and it works great |
| 21:38.15 | TheBear | thanks ManxPower: I now have Asterisk::AGI |
| 21:38.16 | phantasis | on sparc tho |
| 21:38.28 | Darwin35 | netbsd or linux |
| 21:39.16 | yaaar | CosmicRay: version 1.0.2-3 |
| 21:39.32 | phantasis | well I was thinking of converting an E3500 with 8 Ultra 400Mhz and about 10GB RAM into a central IAX switch |
| 21:39.46 | phantasis | how many IAX calls could that hold? |
| 21:39.51 | dan2 | twisted[work]: I know you are at von! |
| 21:39.56 | phantasis | or SIP even |
| 21:39.57 | dan2 | :) |
| 21:40.04 | Darwin35 | ok just started the remote update of the ports on the Toronto server |
| 21:40.08 | *** join/#asterisk DaLion (DaLion@70.49.214.54) |
| 21:40.26 | Darwin35 | and 1.0.6 is in ports on the alpha |
| 21:40.27 | Darwin35 | yes |
| 21:40.36 | CosmicRay | yaaar: hmm, is that the version that ubuntu packaged? |
| 21:40.38 | DaLion | i already do my $number = $AGI->get_data("number"); in my AGI |
| 21:40.43 | DaLion | but its not dumping the result |
| 21:40.47 | CosmicRay | yaaar: 1.0.5 is in sid, and 1.0.6 was uploaded in the last couple of days |
| 21:40.52 | CosmicRay | yaaar: you might try grabbing those |
| 21:41.04 | DaLion | print STDERR "$number\n"; |
| 21:41.04 | DaLion | return $number;print STDERR "$number\n"; |
| 21:41.04 | DaLion | return $number; |
| 21:41.06 | CosmicRay | but I don't really know... I can just say it worked for me |
| 21:41.43 | *** join/#asterisk search_learn2005 (~Miranda@209.68.139.150) |
| 21:41.53 | Darwin35 | my decalpha serves 25 phones and a fax mailbox for each person in my loft |
| 21:42.04 | CosmicRay | nice |
| 21:42.18 | antifuchs | hi CosmicRay. I used cscvs on asterisk's CVS today, and got great results (: |
| 21:42.26 | search_learn2005 | Suggestions for a $100-$140 IAX (preffered) or SIP phone? |
| 21:42.39 | yaaar | CosmicRay: yeah, it's the ubuntu package. |
| 21:42.43 | CosmicRay | antifuchs: nice |
| 21:42.47 | Darwin35 | polycom |
| 21:42.55 | dan2 | CosmicRay: what are you using asterisk for? |
| 21:43.00 | Darwin35 | $100 2 lines |
| 21:43.11 | antifuchs | there's this one CVS revision that seems to work pretty nicely with what we have installed here (: |
| 21:43.25 | yaaar | also, i just grepped my log for error, and found several lines complaining about no channel 1.....like i said, i have not defined any outgoing channels because I don't have anything to hook them up to yet.....does that confuse asterisk? |
| 21:43.29 | CosmicRay | dan2: just my house, for now |
| 21:43.29 | Darwin35 | next I want to get a door buzzer and get it to work with * |
| 21:43.44 | dan2 | CosmicRay: sipuras or digium hardware |
| 21:43.51 | CosmicRay | dan2: if that goes well, I'll start talking about it at work |
| 21:43.51 | Darwin35 | and a good intercom phone |
| 21:43.58 | CosmicRay | dan2: I ordered the SPA-841 |
| 21:44.03 | CosmicRay | dan2: if it works well, I'll get 3 more |
| 21:44.07 | Darwin35 | spa are good also |
| 21:44.14 | dan2 | CosmicRay: nice phone, I have a dozen of them |
| 21:44.18 | ruiner | anyone ever used ciscos with voice ports in them? |
| 21:45.00 | Darwin35 | dan I have a file for you |
| 21:45.07 | Darwin35 | extesions.conf |
| 21:45.09 | antifuchs | hm, these polycom phones look nice... |
| 21:45.09 | Darwin35 | loaded |
| 21:45.12 | clive- | learn-search, I just got my pa168 working with iax2...I am pretty impressed |
| 21:45.22 | lordcian | is there someplace do download a sample dialplan that makes more sense then the make samples version? I must just be to stupid to figure out extensions.conf |
| 21:45.32 | search_learn2005 | clive: where can I get that pa168? |
| 21:45.53 | Darwin35 | grr I have to wait to pull it over |
| 21:45.54 | clive- | search-learn, from china, but they are available in the usa |
| 21:45.58 | mesi | lordcian: it is tricky, the dialpaln. |
| 21:46.10 | search_learn2005 | clive: any webaddress in the usa? |
| 21:46.13 | SuPrSluG | ManxPower:could this happen because I use friend instead of peer+user? |
| 21:46.20 | Hmmhesays | is it tricky to rock a rhyme? to rock a rhyme that's right on time? |
| 21:46.31 | SuPrSluG | ManxPower:the SUBSCRIBE issue |
| 21:46.55 | Darwin35 | asterisk is the answer to years of not having vm on unix |
| 21:47.05 | clive- | http://ipphone.eezeephone.com/ |
| 21:47.29 | Corydon-w | Unix doesn't have Virtual Memory? |
| 21:47.37 | BrianR___ | voicemail.. |
| 21:47.39 | Darwin35 | voicemail |
| 21:47.41 | bjohnson | I haven't had a good quality outgoing call yet through livevoip |
| 21:47.43 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 21:47.44 | bjohnson | arrr |
| 21:48.00 | KalD|Work | shmaltz, ok - I am looking for the 1.0 way of things ... I see it auto-falls thru now |
| 21:48.30 | search_learn2005 | clive: Have you started using one of these? |
| 21:49.33 | Darwin35 | we have a iax2 phone |
| 21:49.36 | xantus | is the max concurrent calls i can get via t1/pri 24/23? I'm looking into having 1000 concurrent calls |
| 21:49.36 | Darwin35 | I am in love |
| 21:49.40 | clive- | well I have been using them with SIP, until today, ..cant handle any more NAT-SIP troubles |
| 21:49.53 | jontow | xantus; yes.. there are only that many channels on a T1 :) |
| 21:49.57 | gpowers | Darwin35: I'm happy for you! |
| 21:50.04 | LoRez | xantus: yes. get lots of T1's |
| 21:50.12 | jontow | just get a DS3 or a few :) |
| 21:50.15 | xantus | so, is there some kind of multiplex equipment that can bring down the price? |
| 21:50.21 | Darwin35 | man I have been waiting for a iax2 based phone for a long time |
| 21:50.21 | xantus | T1's are expensive |
| 21:50.29 | *** join/#asterisk TwoSchubert (~0x746F6F7@twoschubert.user) |
| 21:50.38 | BrianR___ | xantus: You need to get a bigger pipe.. Or use a PSTN<->VOIP gateway service and data compression to get more channels out of an internet data t1 |
| 21:50.39 | yaaar | xantus: how many T's do you need? |
| 21:50.46 | *** part/#asterisk TwoSchubert (~0x746F6F7@twoschubert.user) |
| 21:50.58 | yaaar | xantus: you can get a DS-3 MUX for a hell of a lot cheaper than the 28 T-1's it provides.... |
| 21:51.09 | TheBear | anyone using the festival weather config ? |
| 21:51.09 | opus___ | what is the best choice for the Asterisk GUI? |
| 21:51.12 | clive- | xantus, I will probably get shouted down, but 1000 simultaneous calls sounds like too much for asterisk to handle imho |
| 21:51.20 | LoRez | xantus: you could get a pair of T3s (28 T1's bundled) and demultiplex it on your end, but you can't pipe that many T1's into a single chassis |
| 21:51.37 | xantus | well, it looks like 43 T1's :p |
| 21:51.42 | xantus | lol |
| 21:51.55 | jontow | damn :P |
| 21:51.58 | tzanger | you wouldn't want to |
| 21:52.18 | search_learn2005 | Darwin35: Which phone do you mean when you say we have a IAX2 phone? Brand, Model, Resource? |
| 21:52.22 | opus___ | hey darwin, do you use a GUI? |
| 21:52.27 | BrianR___ | clive-: Asterisk on a single PC can run a pretty large number of concurrent calls... Testing shows it's somewhere on the order of 300 for a single cheapie PC. |
| 21:52.29 | xantus | does anyone here have * systems that handle 1000 or more concurrent calls? |
| 21:52.38 | LoRez | xantus: what will you need 1k concurrent calls for? |
| 21:52.43 | Darwin35 | not at the min I am going to look into amp |
| 21:52.47 | xantus | conf system |
| 21:52.49 | clive- | serach the other iax phone is farfon, but its been very quiet from the farfon stable |
| 21:52.54 | BrianR___ | clive-: Certainly there's voip gateway services that are running it on a massive scale. More than one box to handle all the load, of course. |
| 21:52.55 | yaaar | xantus: do they need multiplexed? or you just need that data capacity? you could easily just get DS-3's and plug them into a cisco or something and do it that way |
| 21:52.56 | bjohnson | someone was testing 1500 on a dual cpu machine |
| 21:53.02 | Darwin35 | I dont have much to change on my system |
| 21:53.11 | Darwin35 | its a full blown pbx as it is |
| 21:53.11 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 21:53.33 | xantus | i need a stable conference call system |
| 21:53.39 | Darwin35 | I am missing some functions I think but for the most part it does what we need it to |
| 21:53.50 | mesi | BrainR: The nice thing is, that - if you only need a small home PBX - you can use a very old PC from the former days, like my P75 :-) |
| 21:54.05 | BrianR___ | I've been testing asterisk's meetme stuff with voice channels brought in over IAX from sixtel and nufone... |
| 21:54.09 | Darwin35 | asterisk has meetme |
| 21:54.15 | xantus | about 1000 calls at once, but i don't know my provider options...pstn->voip, too many T1's or multiplexed |
| 21:54.16 | clive- | brian , I guess so, but it all depends if your doing transcoding etc etc |
| 21:54.17 | Darwin35 | its a confrance system |
| 21:54.20 | BrianR___ | Once I fixed the QoS problem at my gateway, it's been pretty decent. |
| 21:54.29 | *** part/#asterisk santiago (~santiago@63.245.86.95) |
| 21:54.31 | xantus | yeah, i know about *, I use it at home |
| 21:54.37 | modulus_ | [root@asterisk asterisk]# wc -l extensions.conf `grep '#include' extensions.conf|awk '{print $2}'` |
| 21:54.37 | modulus_ | <PROTECTED> |
| 21:54.37 | modulus_ | <PROTECTED> |
| 21:54.37 | modulus_ | <PROTECTED> |
| 21:54.37 | modulus_ | <PROTECTED> |
| 21:54.38 | modulus_ | <PROTECTED> |
| 21:54.40 | modulus_ | w00t |
| 21:54.45 | xantus | i've setup a 23 line system with a PRI |
| 21:54.58 | LoRez | xantus: 1000 connections into the same conference or multiple? |
| 21:55.06 | BrianR___ | clive-: Run 'show translation' and look for the biggest number. That'll give you a pretty good idea of how many calls you can run in the worst transcoding case. |
| 21:55.07 | xantus | and i wrote POE::Component::Client::Asterisk::Manager |
| 21:55.25 | xantus | LoRez: both |
| 21:55.34 | Darwin35 | modulus cool |
| 21:55.36 | *** join/#asterisk yaout (~eric@CPE-65-30-220-56.wi.rr.com) |
| 21:56.03 | Darwin35 | modulus want to help add to the extenion.conf project |
| 21:56.15 | Darwin35 | add fuctions that are missing |
| 21:56.30 | Darwin35 | we are making 1 husr extenisons.conf to be had by all |
| 21:56.34 | clive- | brian, if I have 1000 simultaneous customers, I wont worry about trying to run everything on a single old pc:) |
| 21:56.42 | Darwin35 | then you turn off what you dont need |
| 21:57.10 | xantus | no, we'll use a few pcs |
| 21:57.12 | yaaar | how can i tell if an mp3 file has a flexible bitrate? |
| 21:57.34 | xantus | you mean VBR |
| 21:57.36 | mesi | yaaar: I think mp3info <filename> tells you. |
| 21:57.36 | modulus_ | try bending it |
| 21:57.36 | LoRez | yaaar: run file on it |
| 21:57.43 | modulus_ | variable bit rate? |
| 21:57.44 | modulus_ | hahahahahaaaa |
| 21:57.48 | modulus_ | flexible |
| 21:57.50 | modulus_ | nice touch |
| 21:57.55 | *** join/#asterisk paulc (paulc@176.134.218.209.transedge.com) |
| 21:58.07 | BrianR___ | There's another project called app_conference which has special optimizations for the cases where a large number of IP clients are connected to the conference.. Avoids unnecessary transcoding. |
| 21:58.11 | BrianR___ | http://voip-info.org/wiki-Asterisk+app_conference |
| 21:58.53 | xantus | can anyone recommend any PSTN<->IAX/SIP providers? |
| 21:59.11 | hardwire | I wish I could intercom from the meetme driver to a list of phones |
| 21:59.24 | clive- | nufone works well with iax2 |
| 21:59.25 | BrianR___ | xantus: I've done testing with both sixtel and nufone. Both seem to work OK. |
| 21:59.27 | xantus | .oO( maybe vonage can provide bulk ) |
| 21:59.48 | xantus | large capacity? |
| 21:59.53 | CosmicRay | xantus: http://www.voip-info.org/wiki-VOIP+Service+Providers+Residential |
| 22:00.01 | CosmicRay | err, strike "+Residential" |
| 22:00.03 | lordcian | mesi? |
| 22:00.09 | Darwin35 | has anyone setup overhead paging with a soundcard yet ? |
| 22:00.13 | BrianR___ | xantus: In my brief testing, I wasn't able to bring up more channels than they could provide. |
| 22:00.14 | xantus | ah yeah, good place |
| 22:00.15 | CosmicRay | I saw several that offered discounts for using over 1 million minutes |
| 22:00.25 | clive- | for 1000 calls, just call qwest or someone |
| 22:00.34 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 22:00.52 | xantus | usworst |
| 22:00.53 | Darwin35 | there is nothing in the wiki for overhead paging |
| 22:00.53 | xantus | heh |
| 22:00.56 | CosmicRay | xantus: maybe http://www.livevoip.com/ |
| 22:01.01 | xkev | is there some way to keep a conference open? |
| 22:01.08 | xkev | meetme |
| 22:01.13 | xkev | ..with nobody in it |
| 22:01.21 | Darwin35 | its always there |
| 22:01.29 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 22:01.34 | *** join/#asterisk tsetane (~tsetane@pppoecl72252.minlos.no) |
| 22:01.40 | xkev | oh duh I need to do meetme.conf |
| 22:01.45 | Darwin35 | anyone with overhead paging pls |
| 22:01.54 | Darwin35 | yes |
| 22:02.03 | Darwin35 | xkev has seen the light |
| 22:02.10 | Darwin35 | and it was yellow |
| 22:02.12 | CosmicRay | xantus: is this scaling high enough? :-) http://www.livevoip.com/index.php?subject=1&content=usaCanadaRates |
| 22:02.12 | xkev | I've been doing dynamics :) |
| 22:02.40 | CosmicRay | xantus: down to .85 cents per minute if you buy more than 1 million minutes :-) |
| 22:02.50 | TheBear | what would cause 'Illegal Port Command' trying to d/l the file "ftp://weather.noaa.gov/data/forecasts/fl/tampa.txt" it exists but I get 'Illegal Port Command' or 'Can't build data connection:' |
| 22:03.02 | CosmicRay | TheBear: failing to use passive mode |
| 22:03.16 | opus___ | thebear - firewall? |
| 22:03.19 | opus___ | iptables, |
| 22:03.20 | opus___ | ? |
| 22:03.40 | hardwire | CosmicRay: those rates kinda freak me out |
| 22:03.45 | CosmicRay | heh |
| 22:04.29 | BrianR___ | Like $8k for more phone service than one person could ever use.. |
| 22:04.43 | CosmicRay | BrianR___: large companies could is it, I suspect |
| 22:04.54 | TheBear | ok what do I need to change in the firewall ? |
| 22:05.05 | BrianR___ | CosmicRay: Yes... But it's frightening how cheap phone service has become... |
| 22:05.17 | TheBear | opus: or CosmicRay: what needs to change ? |
| 22:05.32 | hardwire | CosmicRay: do you use them? |
| 22:06.00 | Corydon-w | Pretty much anybody with a PRI running into Asterisk |
| 22:06.04 | CosmicRay | hardwire: not yet, but as soon as I get my SIP phone, I think I'll sign up |
| 22:06.09 | hardwire | ok |
| 22:06.11 | CosmicRay | TheBear: what ftp client are you using? |
| 22:06.13 | hardwire | I am shopping lately |
| 22:06.18 | hardwire | i hate their site.. is that evil of me? |
| 22:06.25 | hardwire | its just plain ugly. |
| 22:06.26 | BrianR___ | A million minutes is almost 700 days of continuous talk time. |
| 22:06.27 | CosmicRay | no, websites suck all the time |
| 22:06.29 | TheBear | trying from AGI script and also windows ftp |
| 22:06.32 | CosmicRay | hardwire: at least it doesn't require flash |
| 22:06.37 | xantus | :o |
| 22:06.41 | hardwire | nope |
| 22:06.51 | hardwire | just a bigger monitor |
| 22:06.51 | CosmicRay | TheBear: if you're using windows ftp, type "pass" before you run your "geT" |
| 22:07.02 | CosmicRay | I hate sites that require flash |
| 22:07.03 | hardwire | its header is that of a klingon. |
| 22:07.07 | CosmicRay | heh |
| 22:07.30 | yaaar | is it normal for asterisk not to show up on an nmap? I've gotten it to claim it's listening (according to logs) and to not die on startup, but nmaping it from another machine still doesn't show any ports open beyond 22 and 80 |
| 22:07.30 | CosmicRay | yaaar: you're probably not probing udp ports. |
| 22:07.33 | opus___ | yaar -- netstat -a --programs | grep 5060 |
| 22:07.47 | opus___ | or | grep asterisk |
| 22:07.59 | hardwire | I think I am going to use this company I just found |
| 22:08.05 | hardwire | or somebody poked me an upsale too |
| 22:08.05 | yaaar | excellent. thanks for the tips |
| 22:08.11 | CosmicRay | hardwire: which one is that? |
| 22:08.12 | TheBear | CosmicRay: Now I get 500 Illegal PORT Command |
| 22:08.15 | hardwire | because they are in my hometown :) |
| 22:08.19 | hardwire | tel iax :) |
| 22:08.28 | CosmicRay | TheBear: from which, pass or get? |
| 22:08.33 | hardwire | they atleast have the capacity and are on the same networks I need to use down in colorado |
| 22:08.45 | TheBear | CosmicRay: get after I had done pass sucessful |
| 22:08.46 | hardwire | as well as they are going to be able to colocate a machine for me.. to interconnect to them |
| 22:08.57 | CosmicRay | hardwire: pfft! 2 cents per minute! what a ripoff! :-) |
| 22:09.07 | slePP | http://pastebin.ca/draw.php -- i'm disappointed in all of you :P |
| 22:09.11 | hardwire | CosmicRay: I am sure its backed up with quality. |
| 22:09.13 | yaaar | cool. yeah, it's listening now |
| 22:09.25 | yaaar | unfortunately, it still won't let my clients register |
| 22:09.32 | hardwire | if it isn't then holy shit.. what a ripoff.. and tada.. you switch providers.. the wonders of voip eh? |
| 22:09.38 | CosmicRay | hardwire: well, that is important. |
| 22:09.52 | CosmicRay | hardwire: so I figure I will start with the cheapo ones and move up if they suck :-) |
| 22:09.58 | hardwire | I wouldn't mind talking to livevoip.. but they seem to ambitious from their site |
| 22:10.07 | hardwire | which is sliglhty scary from a stability standpoint |
| 22:10.16 | CosmicRay | I've also heard good things about voipjet.com |
| 22:10.21 | CosmicRay | though their ToS is *scary* |
| 22:10.24 | TheBear | CosmicRay: "wget ftp://weather.noaa.gov/data/forecasts/city/fl/tampa.txt" also gets Invalid PORT. ? |
| 22:10.29 | hardwire | cause I'm.. talking.. on a voipjet plane. |
| 22:10.35 | hardwire | don't know when i'll hang up again.. |
| 22:10.40 | hardwire | heh |
| 22:10.43 | hardwire | I need coffee |
| 22:10.55 | moonwick | what's scary about their TOS? |
| 22:11.11 | CosmicRay | moonwick: take a look here: https://www.voipjet.com/tos.php |
| 22:11.14 | hardwire | it must not exist |
| 22:11.14 | hardwire | heh |
| 22:11.20 | CosmicRay | moonwick: basically, you agree to never tell anyone that you use voipjet |
| 22:11.29 | hardwire | oh |
| 22:11.32 | hardwire | Terms of Service |
| 22:11.34 | hardwire | not Type of Service |
| 22:11.35 | CosmicRay | moonwick: then you must agree to never use it to discuss financial or medical affairs |
| 22:11.47 | CosmicRay | moonwick: and you must agree to never use it for anything "important", whatever that means |
| 22:12.03 | moonwick | huh, I haven't even seen their ToS |
| 22:12.10 | Nugget | hat's nutty. |
| 22:12.19 | CosmicRay | yeah. |
| 22:12.21 | hardwire | hehe |
| 22:12.31 | hardwire | I need to find out about origination failover via PSTN |
| 22:12.36 | hardwire | I should email this guy |
| 22:13.07 | opus___ | asterisk.gnuinter.net seems to be down... |
| 22:13.20 | opus___ | does anybody have a copy of asterisk-perl-0.08.tar.gz that I can get a copy of? |
| 22:13.49 | BrianR___ | nufone has the nice pstn failover for their toll free inbound numbers. If it can't reach your asterisk box it'll call a PSTN number (extra $0.02/min charge applies though) |
| 22:14.09 | slePP | opus___: i do. but you have to enter my draw first :> |
| 22:14.19 | opus___ | ha |
| 22:14.23 | slePP | http://pastebin.ca/draw.php |
| 22:14.29 | slePP | http://netmonks.ca/asterisk-perl-0.08.tar.gz |
| 22:14.57 | outtolunc | http://asterisk.gnuinter.net/files/ is up |
| 22:16.00 | xantus | CosmicRay: .0085 cents per min |
| 22:16.14 | opus___ | Of course it came up after i asked :) |
| 22:16.15 | opus___ | thanks |
| 22:16.49 | zipp | xantus, what is that price for? |
| 22:17.14 | TheBear | CosmicRay: what would I need to change to get past the Illegal PORT Command |
| 22:17.19 | xantus | 1000000 minutes |
| 22:17.36 | xantus | 8.5k |
| 22:17.39 | yaaar | aaaarg |
| 22:18.11 | zipp | xantus, where? |
| 22:18.11 | xantus | livevoip |
| 22:18.11 | zipp | xantus, and when do they expire? |
| 22:18.11 | hardwire | I want the incredibles! |
| 22:18.11 | jontow | thebear; passive ftp mode? |
| 22:18.12 | xantus | http://www.livevoip.com/index.php?subject=1&content=usaCanadaRates |
| 22:18.22 | yaaar | so, now all i'm getting from iaxcomm is 'registration rejected' and all i'm getting in the asterisk log is 'No registration for peer '200' (from <ip>) |
| 22:18.32 | xantus | zipp: no idea |
| 22:18.32 | TheBear | jontow: how I add $ftp->pasv to the agi script and still doesn't work |
| 22:18.32 | yaaar | anyone point me to where else to look for why? |
| 22:18.36 | slePP | hardwire: then fill in the form :P |
| 22:18.43 | JerJer[mobile] | yaar: host=dynamic |
| 22:18.46 | slePP | right now, chances of winning are 1 in 17 |
| 22:18.49 | slePP | since no one seems to want to enter :P |
| 22:19.05 | yaaar | JerJer[mobile]: sorry, i'm a bit slow....where's that option go? |
| 22:19.12 | JerJer[mobile] | RTFM |
| 22:19.17 | yaaar | right... |
| 22:19.20 | slePP | heh |
| 22:19.43 | *** join/#asterisk twilson (~terry@63.77.68.11) |
| 22:19.46 | DaLion | ??? |
| 22:19.46 | DaLion | where ? |
| 22:19.46 | DaLion | Thebear what u trying to ftp ? |
| 22:19.59 | DaLion | Slepp where the contest ? |
| 22:20.00 | xantus | but what i am looking for is DID in and they charge 1.1 cents per min |
| 22:20.06 | slePP | DaLion: http://pastebin.ca/draw.php |
| 22:20.08 | TheBear | DaLion: weather.noaa.gov/data/forecasts/city/fl/tampa.txt |
| 22:20.23 | TheBear | to get festival weather config to work |
| 22:20.26 | DaLion | ah lol |
| 22:21.02 | JerJer[mobile] | xantus: wholy corncobbing batman |
| 22:21.10 | BrianR___ | xantus: Nufone seems to offer unlimited DID's for a fixed amount per month. I bet if you had 10000 calls coming in they'd cap the number of calls or total minutes though. :) |
| 22:21.19 | JerJer[mobile] | BrianR___: no |
| 22:21.29 | JerJer[mobile] | all we care about is the number of simultaneous calls |
| 22:21.37 | xantus | 10k or 1k? |
| 22:22.24 | zipp | BrianR___, I am quite sure on local DID's (michigan I think) you are limited to 4 concurrent incoming lines |
| 22:22.30 | xantus | yeah, i'm checking nufone and sixtel |
| 22:22.31 | JerJer[mobile] | zipp no |
| 22:22.38 | ManxPower | xantus, I think with Nufone you can have as many incoming calls as you want, as long as it's not more than one. 8-) |
| 22:22.48 | JerJer[mobile] | NO |
| 22:23.01 | JerJer[mobile] | if you do not know the answer, just keep quiet |
| 22:23.16 | zipp | JerJer[mobile], so I can have a michigan did and unlimited incoming calls for free |
| 22:23.17 | xantus | nufone website looks like it was done by a 12yo |
| 22:23.27 | JerJer[mobile] | thank you |
| 22:23.35 | zipp | xantus, come on now, no need to be like that |
| 22:23.36 | xantus | :p |
| 22:23.36 | slePP | heh |
| 22:23.39 | xantus | hehe |
| 22:23.40 | slePP | JerJer[mobile]: users rule, huh? |
| 22:23.59 | zipp | JerJer[mobile], can you explain the incoming michigan did's w/ nufone? |
| 22:24.06 | DaLion | added |
| 22:24.34 | xantus | ManxPower: haha |
| 22:24.35 | JerJer[mobile] | xantus: we could go back to the flash driven website that crashed lots of browsers |
| 22:24.41 | xantus | nooo |
| 22:24.43 | DaLion | MAnxpower teliax too.. as much channles as you need.. in PAYG plan |
| 22:25.01 | JerJer[mobile] | or we can keep the current one that has enough content for those with a clue to figure out what we provide |
| 22:25.08 | xantus | true |
| 22:25.20 | Godsey | the # of simultanious calls is a function of how many lines are in the hunt group :) |
| 22:25.21 | TheBear | ok with wget --passive-ftp I can get the file, so how to I change this in the weather.agi script ? |
| 22:25.26 | xantus | the yellow throws my eyes when on white |
| 22:25.35 | xantus | thats just me tho |
| 22:25.37 | JerJer[mobile] | it is designed to anony |
| 22:25.46 | JerJer[mobile] | to keep the riff raff away |
| 22:26.11 | xantus | :p |
| 22:26.43 | Godsey | I've had 20 simultanious calls using ipkall so far :) |
| 22:26.58 | JerJer[mobile] | if you really want to look at something pretty then look here: http://ww2.nufone.net |
| 22:27.02 | zipp | JerJer[mobile], so, for 7.95 a month I can have unlimited inbound calls? |
| 22:27.02 | JerJer[mobile] | but don't expect much |
| 22:27.18 | JerJer[mobile] | zipp: how about 7.50 |
| 22:27.32 | JerJer[mobile] | and we absolutely do not use the word unlimited |
| 22:27.41 | Godsey | you can get unlimited inbound free from ipkall :) |
| 22:27.44 | Godsey | tho there is no support |
| 22:28.00 | Godsey | and the quality probably sucks in comparison |
| 22:28.09 | yaaar | ok, i must be missing something here; i've tried adding either host=dynamic or defaultip=000.000.000.000 (based on a mailing list posting) and either way the log output (and output from iaxcomm) is the same....registration rejected on the client and 'no registration from peer 200 (ip)' on the server |
| 22:28.30 | zipp | JerJer[mobile], I like www better then ww2 |
| 22:28.35 | *** join/#asterisk metrogtiguy (~a@dsl093-086-034.det1.dsl.speakeasy.net) |
| 22:28.54 | metrogtiguy | Can anyone help me with the remote call pickup function? |
| 22:29.54 | metrogtiguy | I'm not sure if I have the extensions set up wrong, or using the wrong syntax |
| 22:30.01 | yaaar | now, if bindaddr=0.0.0.0 in iax.conf, it should listen on all ips, right? |
| 22:30.36 | DaLion | ok.. question... -- Playing 'number' (language 'en') |
| 22:30.36 | DaLion | 1234 |
| 22:30.46 | *** join/#asterisk optix (optix@dsl254-066-144.nyc1.dsl.speakeasy.net) |
| 22:30.49 | DaLion | AGI Script blah.agi completed, returning 0 |
| 22:30.58 | DaLion | how do i get that from my manager ? |
| 22:32.27 | yaaar | hey, wait that's odd.....i'm getting a log message saying bindaddr is an unknown directive. i didn't put that in the file, it was already there...... |
| 22:32.55 | *** join/#asterisk pixer (~dotto@socks4.fastwebnet.it) |
| 22:33.00 | pixer | hi to all |
| 22:33.16 | *** join/#asterisk NirS_HOME (Nir@192.117.110.178) |
| 22:36.12 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 22:36.21 | gr8nash | GRRRR |
| 22:38.53 | Beirdo | gr8nash: what's yer damage? |
| 22:39.43 | pixer | hi to all! I have a problem with one digium wildcard... they are not ignited the leds with kernel module loaded.. someone knows to help me? thanks |
| 22:39.52 | gr8nash | hey Beirdo nuttin.. just not able to receive calls still.. i just switched to livevoip |
| 22:39.59 | Beirdo | ah |
| 22:40.22 | optix | Does anyone know if the Cisco Wireless IP Phone 7920 is compatible with Asterisk? |
| 22:42.16 | johnnyb | I've had problems w/ Cisco IP Phones. Be sure that they say they work with a SIP server, and not just that they are SIP compatible. |
| 22:42.39 | johnnyb | pixer: what's the output of dmesg |
| 22:42.43 | Beirdo | anybody use the PrivacyManager application? |
| 22:43.06 | search_learn2005 | Can I use the T1 line that brings internet to my school to serve VOIP to around 40 teachers? Or do I have to get PRI? |
| 22:43.50 | xantus | JerJer[mobile]: hot damn that looks better! |
| 22:43.53 | *** join/#asterisk dontmsgme (~none@69-175-234-120.vnnyca.adelphia.net) |
| 22:44.09 | optix | What would you guys recommend for a VoIP hardphone to work with Asterisk that has intercom capabilities? |
| 22:44.10 | DaLion | xantus where |
| 22:44.18 | dontmsgme | I'm habing problems with my router I think because of the firmware installed which does not allow for NAT destriction has anyone ever had this happen? |
| 22:44.22 | DaLion | optix polycom ip600 or 500 |
| 22:44.30 | xantus | http://ww2.nufone.net |
| 22:45.15 | optix | Soundpoint IP? |
| 22:45.50 | *** join/#asterisk cripito (~ncripito@68.216.32.57) |
| 22:46.14 | cripito | hola |
| 22:46.28 | optix | DaLion: anything smaller? |
| 22:46.58 | cripito | anyone known where 2 buy fxs gateway in 40 - 50 price ranges? |
| 22:47.04 | DaLion | sure |
| 22:47.11 | DaLion | loke in voupsupply |
| 22:47.18 | DaLion | s/u/i |
| 22:47.19 | *** join/#asterisk pixer2 (~dotto@socks4.fastwebnet.it) |
| 22:47.25 | dontmsgme | Has anyone ever had a problem with NAT firewalls because their router's firmware didn't allow for disabling NAT |
| 22:47.32 | optix | DaLion: this is going to be for a nightclub |
| 22:47.42 | *** join/#asterisk masuda (~masuda@pcp04490438pcs.brmngh01.mi.comcast.net) |
| 22:47.45 | DaLion | donttmsgme.. yes.. over a sat ocnnection in africa .. a friend working there had probs... |
| 22:47.58 | DaLion | optix u mean ??? |
| 22:48.29 | optix | DaLion: the application for these voip phones are for staff in a nightclub |
| 22:48.35 | optix | to communicate internally |
| 22:48.36 | masuda | asterisk@home0.6, right after I installed it I was able to connect with X-Lite, now it appears Asterisk has stopped listening on port 5060 & X-Lite can't connect. |
| 22:48.37 | DaLion | dontmsgme try playing with port forwarding of 4569 and 5060 .. check/uncheck uPNP.. also.. |
| 22:48.47 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 22:48.48 | DaLion | also... try to force to 5060 on client |
| 22:49.08 | masuda | sip.conf has port=5060, but netstat -an shows no listening on 5060, and tcpdump shows asterisk bouncing icmp with port unreachable |
| 22:49.15 | DaLion | masuda .. grep '5060' /etc/asterisk/*.conf |
| 22:49.20 | DaLion | will tell you file to check |
| 22:49.21 | *** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 22:49.40 | DaLion | maybe.. port already bound |
| 22:49.48 | DaLion | or |
| 22:49.50 | DaLion | hmm |
| 22:49.54 | DaLion | reboot ;) |
| 22:50.18 | Skysky | hi, i wonder if anyone received the same warning as me when using NoCDR(), it is "WARNING[11533]: cdr.c:114 ast_cdr_free: CDR on channel 'SIP/111-2b74' not posted, WARNING[11533]: cdr.c:116 ast_cdr_free: CDR on channel 'SIP/111-2b74' lacks end" |
| 22:50.19 | masuda | . /etc/asterisk/sip_additional.conf:port=5060 |
| 22:50.26 | Skysky | because my NoCDR isn't working right now |
| 22:50.31 | masuda | just rebooted |
| 22:51.00 | *** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
| 22:51.05 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 22:51.16 | DaLion | masuda.. ah i see AMP again .. |
| 22:51.18 | DaLion | lol |
| 22:51.26 | masuda | AMP ? |
| 22:51.40 | RoyK | ~amp? |
| 22:51.41 | jbot | it has been said that amp is an Audio MPEG Player. [non-free] |
| 22:51.42 | DaLion | asterisk management portal |
| 22:51.45 | RoyK | ~rtfm? |
| 22:51.46 | jbot | extra, extra, read all about it, rtfm is read the f*cking manual... try asking me about "FAQ" |
| 22:51.54 | masuda | ahh |
| 22:51.56 | RoyK | ~lart masuda |
| 22:52.01 | *** join/#asterisk zapa (~kasokda@201.135.137.236) |
| 22:52.06 | DaLion | ~google AMP |
| 22:52.09 | shmaltz | anybody here using Thirdlane? |
| 22:52.11 | Nivex | ~fgi |
| 22:52.15 | DaLion | bah |
| 22:52.18 | RoyK | ~lart DaLion |
| 22:52.18 | Nivex | Guess it doesn't know that one. |
| 22:52.23 | masuda | AMP came default with asterisk@home |
| 22:52.25 | *** join/#asterisk Kinyobi (~Ladius@lumiere.lasierra.edu) |
| 22:52.36 | shmaltz | ~masuda |
| 22:52.49 | DaLion | man this shitty line has been around since 1982 |
| 22:53.03 | DaLion | large torut my a... |
| 22:53.11 | DaLion | s/orut/rout |
| 22:53.33 | ariel_ | masuda, what is your question I missed it. I got discconected from the network. |
| 22:53.51 | optix | DaLion: what do you think about the BudgeTones? |
| 22:53.55 | optix | (Grandstream) |
| 22:54.11 | xantus | can anyone provide a DID in chile? |
| 22:54.18 | masuda | port 5060's not open, not listening |
| 22:54.20 | ariel_ | optix, nick name is barbie tone does that kinda give you an Idea. |
| 22:54.32 | *** join/#asterisk dontmsgme (~none@69-175-234-120.vnnyca.adelphia.net) |
| 22:54.42 | ariel_ | masuda, asterisk@home does not close the ports in fact it has no firewall setup on it. |
| 22:54.47 | masuda | X-Lite can't connect to it |
| 22:54.51 | optix | ariel_: heh |
| 22:54.58 | masuda | yea it's not a firewall issue. netstat -an|grep 5060 shows nothing |
| 22:55.08 | optix | ariel_: I'm looking for something about that size, but maybe a bit more respectable? :P |
| 22:55.31 | ariel_ | optix, I have been using now the Sipura 841 there good for the price. |
| 22:55.36 | xantus | i have a snom 200 |
| 22:55.40 | optix | how much? |
| 22:55.44 | ariel_ | 85 |
| 22:55.45 | xantus | works...but i'd go with a cisco |
| 22:55.55 | ariel_ | Cisco are too costly |
| 22:56.01 | optix | xantus: I've been looking at Cisco |
| 22:56.05 | xantus | you get what you pay for |
| 22:56.06 | optix | and for the ammount that I need |
| 22:56.09 | optix | it'd cost $28k |
| 22:56.16 | xantus | for how many phones?. |
| 22:56.18 | *** join/#asterisk SimonR (~SimonR@static-1M-b1-14.highspeed.eol.ca) |
| 22:56.19 | optix | 40 |
| 22:56.23 | *** join/#asterisk atmel (~vlad@wireless-am4.ucsd.edu) |
| 22:56.24 | xantus | !! |
| 22:56.30 | ariel_ | masuda it should work if you set the ext correctly in the GUI |
| 22:56.37 | xantus | toooo much |
| 22:56.39 | pixer2 | Hi to all! I have a problem with one digium wildcard... they are not ignited the leds with kernel module loaded and interrupts assigned.. someone can help me, please? thanks! |
| 22:56.48 | xantus | are you getting a cisco pbx with it? |
| 22:56.51 | optix | xantus: nope |
| 22:56.54 | xantus | wtf |
| 22:56.57 | optix | planning on running * on it :) |
| 22:56.59 | xantus | power supplies? |
| 22:57.19 | xantus | the 7960's are $300 aren't they |
| 22:57.23 | optix | xantus: I don't think that was included, this is the 7920's |
| 22:57.27 | ariel_ | optix, if you want something better then the low end and in my view as good or better then Cisco get the Polycom IP-500 |
| 22:57.41 | optix | ariel_: but thats gonna be too much deskspace |
| 22:57.58 | xantus | a lower model.. |
| 22:58.10 | optix | ariel_: this is gonna be for a waitress station |
| 22:58.12 | SimonR | Does anyone know a good bet for large-volume VoIP termination? |
| 22:58.14 | Darwin35 | ok evryone on your knees and praise the lord * |
| 22:58.19 | Darwin35 | heheh |
| 22:58.20 | ariel_ | masuda, how about a FW on the windows system your using? is it installed? |
| 22:58.23 | DaLion | Simon whats large volume ? |
| 22:58.28 | masuda | here's my ext setting: http://www.pastebin.com/252817 |
| 22:58.29 | xantus | optix: cordless? |
| 22:58.38 | DaLion | large =2 200 20000 2000000000000000000 minutes ? |
| 22:58.49 | xantus | oh, i assumed they'd be the 7960 series, my bad |
| 22:58.52 | SimonR | by large volume, I mean hundreds of simultaneous calls, although our traffic is unpredictable. |
| 22:58.52 | pixer2 | please help me.. i have read the manual but I have not found null :\ |
| 22:58.56 | masuda | ariel_, when I run tcpdump on the asterisk server it shows the windows system sending UDP request, and the asterisk server responding ICMP: port 5060 unreachable |
| 22:59.26 | optix | xantus: ya |
| 22:59.30 | optix | <PROTECTED> |
| 22:59.40 | xantus | $503.25 a piece |
| 22:59.45 | xantus | at lacc.com |
| 22:59.51 | optix | Xander77: I found them for $300 something |
| 23:00.12 | cripito | i need a device btw 30 - 50 price range |
| 23:00.15 | cripito | anything btw that? |
| 23:00.19 | xantus | you mean me right? |
| 23:00.25 | optix | xantus: ya |
| 23:00.27 | optix | sorry. |
| 23:00.32 | optix | damn nickcomp :\ |
| 23:00.35 | jontow | cripito; no.. $75-80 though, and you can get a grandstream budge-tone or handytone ATA |
| 23:00.48 | xantus | $300 @ 40 is 12k |
| 23:00.53 | *** part/#asterisk DaLion (DaLion@70.49.214.54) |
| 23:00.54 | xantus | why twice the price |
| 23:00.58 | optix | xantus: hold on |
| 23:01.02 | optix | lemme double check the quote |
| 23:01.03 | optix | :P |
| 23:01.13 | cripito | :) we want 2 buy 40 - 50 device :D |
| 23:01.31 | doughecka | buy cisco |
| 23:01.33 | yaaar | alright....time to pack it up and go home. thanks for all the help everybody |
| 23:01.34 | doughecka | its the best |
| 23:01.45 | doughecka | 250+21 for powersupply |
| 23:01.48 | optix | actually, it's about $550, I had some other gear added in that I didn't see. |
| 23:02.04 | cripito | i am pretty happy with sipura at this time |
| 23:02.06 | cripito | ;) |
| 23:02.08 | optix | (Cisco Aironet) |
| 23:02.11 | cripito | 65 |
| 23:02.20 | cripito | sipura 1001 |
| 23:02.27 | hardwire | optix: I use SPA-3000's |
| 23:02.29 | hardwire | I love them |
| 23:02.34 | hardwire | they are amazing happy little devices. |
| 23:02.40 | cripito | the 1001? yeap |
| 23:02.51 | cripito | i like it more over the 2000 or 3000 |
| 23:03.05 | cripito | but i need 2 buy a lot :D |
| 23:03.09 | optix | what about the SPA-841 |
| 23:03.31 | cripito | optix.. i am so happy with the 1001 that i even try it :D |
| 23:03.46 | cripito | i need at least 20 more :D |
| 23:03.59 | optix | I need an actual device. |
| 23:04.12 | optix | (i.e. an actual hard VoIP phone) |
| 23:04.31 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
| 23:05.03 | TheBear | ok got past getting the file. are you supposed to have /var/lib/asterisk/sounds/tts ? I don't yet the weather script calls for it ? |
| 23:05.08 | cripito | well it depends in what are u plans.. i try 2000, 3000, 1000, 1001 |
| 23:05.11 | cripito | i love 1001 |
| 23:05.32 | optix | cripito: this is going to be for waitstaff to call internally around a nightclub |
| 23:05.46 | johnnyb | Does anyone else get crackely sound when using sox to convert wav to gsm files? |
| 23:06.17 | cripito | the staff will be moving or in just 1 place? |
| 23:06.38 | cripito | 3000 is out b/c is 1 fxo 1 fxs u are making internal calls |
| 23:06.56 | optix | cripito: yeah, they're gonna be constantly moving |
| 23:06.57 | cripito | 2000 have 2 fxs so u can have 2 phones in the places |
| 23:07.05 | optix | cripito: it's basically going to be an intercom system |
| 23:07.21 | optix | with some direct calling |
| 23:07.27 | cripito | then why don't get intercom instead just phones? i think there is device for that |
| 23:07.44 | optix | cripito: because there's gonna be some outbound calling as well |
| 23:07.46 | TheBear | I have subscribe to asterisk-users@digium but not received any email to activate the subscription ?why? |
| 23:08.16 | xantus | optix: sounds like a good way to go IF they are willing to spend the $$ |
| 23:08.28 | optix | xantus: they make over $7m a night. |
| 23:08.36 | xantus | $$ |
| 23:08.41 | xantus | damn |
| 23:08.54 | xantus | million not thousand right? |
| 23:08.58 | xantus | :P |
| 23:08.59 | optix | million. |
| 23:09.13 | xantus | jez, casino too? |
| 23:09.19 | optix | xantus: not that I know of |
| 23:09.25 | cripito | :D |
| 23:09.25 | ruiner | how do you specify a sip trunk a dial? for instance, my cisco router i've setup a voice port as trunk 1, when i make my dial, do i just dial SIP/1@cisco? |
| 23:09.40 | ruiner | i'm not even sure i'm using the right terminology here |
| 23:09.41 | ruiner | heh |
| 23:09.47 | ruiner | but i'm really at wit's end |
| 23:09.51 | cripito | optix: if u need 2 lines in the place 2000 |
| 23:09.55 | cripito | if u need 1 1001 |
| 23:10.07 | optix | cripito: it's mostly gonna be internal traffic |
| 23:10.13 | cripito | 2000 is 8x |
| 23:10.19 | cripito | 1001 is 6x |
| 23:10.35 | optix | cripito: there are going to be 40 different "extensions" |
| 23:10.37 | optix | so to speak |
| 23:10.41 | optix | internally |
| 23:11.06 | optix | but I think 1-2 outside lines |
| 23:11.09 | optix | should be enough. |
| 23:11.25 | cripito | 1001 see atacomm usually maybe u can get a discount. for the device fxs device.. |
| 23:12.30 | optix | cripito: how am I going to deal with the internal phones tho? |
| 23:13.01 | cripito | see ur pvt |
| 23:16.57 | zapa | hi all , i have a E1 Pri, i am having a lot of echo from my asterisk voip side with Polycom and Cisco 7940 phones, the pstn donīt hear the echo only the asterisk side any clue ? i alredy active echo canceler and training at zapata.conf this just happen when i made call to pstn |
| 23:18.20 | jontow | zapa; i had a lot of problems with that.. never quite got it solved before the project was given up on.. we tuned for hours though.. that kinda sucked :) |
| 23:18.41 | *** part/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
| 23:18.41 | jontow | a lot can be done with gain.. |
| 23:18.44 | doughecka | I get a TINY bit of echo on my cisco phoen |
| 23:18.48 | doughecka | with a pstn card |
| 23:18.50 | doughecka | analog |
| 23:19.02 | doughecka | and thats with training turned on and thats it |
| 23:19.43 | zapa | jontow: but is problem from the e1 anda zapata.conf ? |
| 23:20.01 | jontow | it was explained to me that it was more in the cisco phones.. |
| 23:20.02 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 23:20.08 | jontow | picking up noise from either end |
| 23:20.13 | xantus | how is support for the S100U nowdays? |
| 23:20.14 | jontow | they've got really sensitive audio components.. |
| 23:21.03 | *** join/#asterisk Damin_Mobile (~pocketirc@10.sub-70-214-224.myvzw.com) |
| 23:21.05 | zapa | jontow: itīs funny just i hear the echo but pstn people dont |
| 23:21.36 | Damin_Mobile | Von is winding down. |
| 23:21.51 | xantus | vonage? |
| 23:22.02 | Damin_Mobile | Sittinh here w kram |
| 23:22.18 | jontow | yep |
| 23:22.49 | Damin_Mobile | zoa and BRIAN capouch liteninng into the conversation |
| 23:23.02 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 23:23.07 | Damin_Mobile | about the draft rfc for IAX. |
| 23:23.17 | jontow | awesome :) |
| 23:24.00 | Damin_Mobile | One of Brian's students is working on it... |
| 23:24.40 | Damin_Mobile | T |
| 23:24.40 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 23:24.59 | Damin_Mobile | That was a major issue that w |
| 23:26.00 | Damin_Mobile | s brought forth by the vendors froom the IAX breakout group... |
| 23:27.05 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 23:27.34 | shadebob | hi, I search how I can implement transfert in my diaplan.. Someone can help me? |
| 23:28.47 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 23:31.41 | *** join/#asterisk xyharley (~daecon@xyharley.dsl.xmission.com) |
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| 23:39.30 | opus___ | asterisk-addon fails to build :( |
| 23:39.41 | opus___ | app_addon_sql_mysql.c:164:77: macro "AST_LIST_REMOVE" passed 4 arguments, but takes just 3 |
| 23:39.56 | opus___ | from the asterisk-addons-1.0.6.tar.gz from asterisk.org |
| 23:40.00 | opus___ | has anyone had this problem? |
| 23:40.42 | opus___ | if I comment out the line it works |
| 23:40.46 | ManxPower | opus___, Sounds like you are using asterisk-addons-1.0.6 with CVS-HEAD |
| 23:40.55 | opus___ | ManxPower thanks |
| 23:41.09 | opus___ | is CVS-HEAD stable is the question now |
| 23:41.14 | ManxPower | If that's the case, well don't do that. 1.0.6 is for 1.0.6 |
| 23:41.21 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-39-201.no.no.cox.net) |
| 23:41.47 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 23:43.01 | stustu | What's the deal with the pound sign and dialling extensions over SIP? I can't make calls to extenstions like #55# using SIP from X-Lite, but it works on my Zap devices. |
| 23:43.35 | ManxPower | stustu, Most SIP devices assume # means "I'm done dialing" |
| 23:43.43 | ManxPower | And strips it off, of course |
| 23:44.17 | stustu | Would you know if this is a part of SIP, or if it's just a client's idea? |
| 23:44.50 | Kinyobi | anyone got any links hand re avaya R9SI/S8500 sip and *? having a problem getting the avaya sales people to fess up about their sip compliance... |
| 23:45.19 | stustu | (I vaguely remember some very old services that were accessed over a split speed 75/1200 modem using # as an equivalent to ENTER. Maybe there's a connection? |
| 23:46.18 | *** part/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 23:49.38 | *** join/#asterisk ctooley (~ctooley@rrcs-24-153-228-2.sw.biz.rr.com) |
| 23:49.54 | ctooley | I'm having some issues with a queue. It seems to not want to allow people to join it. |
| 23:50.07 | opus___ | awesome... AMP just overwrote my configs |
| 23:50.31 | ctooley | 2005-03-10 17:44:55 VERBOSE[8916]: -- Executing Queue("IAX2/pplay1@pplay1/2", "datenumber") ---- 2005-03-10 17:44:55 WARNING[8916]: Unable to join queue 'datenumber' |
| 23:51.01 | ctooley | opus___, that's cool. I want AMP to do that for me. |
| 23:54.07 | ctooley | Anyone wanna take a look at the queue config for me? |
| 23:55.07 | opus___ | sip show queue? |
| 23:55.21 | opus___ | show queue |
| 23:55.21 | opus___ | even |
| 23:56.38 | ctooley | proxy1*CLI> show queue datenumber |
| 23:56.38 | ctooley | datenumber has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s |
| 23:56.38 | ctooley | <PROTECTED> |
| 23:56.52 | ctooley | I just logged agent 101 out, it was logged in earlier |
| 23:58.22 | ctooley | it does say "Agent/101 (unavailable) has taken no calls yet" even after -- Agent '101' logged in (format ulaw/slin) |
| 23:58.26 | hardwire | I think I need to install VNC on every single persons computer in this office and just have it log it to jpegs on an archive server |
| 23:58.33 | hardwire | so I can figure out what the hell people are talking about. |
| 23:58.45 | *** join/#asterisk mandreko (mandreko@12-222-3-81.client.insightBB.com) |
