| 00:00.04 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 00:00.13 | *** mode/#asterisk [+o denon] by ChanServ |
| 00:00.41 | *** join/#asterisk rkeels (n=chatzill@99.eedinc.com) |
| 00:01.35 | rkeels | Hey does anyone know of a function with asterisk that would off hook auto dial capabilities using a sip phone |
| 00:02.48 | rkeels | And the beast slumbers |
| 00:03.36 | rkeels | Oh well I'll come back later... Thanks All... Sorry I didn't make it to AstriCon... I was in Turkey... Couldn't pass up the family reunion |
| 00:06.00 | TJNII | Hmmm... Perhaps I'm going about this the wrong way. |
| 00:06.50 | TJNII | I have a script, and when it executes I want it to make Asterisk call multiple phones. When one is answered, I want the user to hear a recording and the other phones to stop ringing. |
| 00:07.04 | TJNII | I have this working using a call file, but that only rings one phone. |
| 00:07.07 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 00:07.28 | TJNII | I also have a queue that calls all the said phones, but I don't know if that can be used. (I'm thinking no.) |
| 00:11.41 | citats | TJNII: you could try to have your channel dialed by your call file be a Local channel that essentially does a Dial(Zap/g1/12345&Zap/g1/123456&Zap/g1/98765) |
| 00:12.06 | citats | TJNII: or have the local channel go into your queue |
| 00:13.07 | zil2 | Hi, I was wondering if there are any instructions on getting a cisco 7910 to connect to asterisk? Ive tried a few things I have seen online but it just aint working! I keep getting a message on the phone "registration rejected" |
| 00:14.41 | [hC] | Qwell: ping! |
| 00:14.52 | Qwell | pung! |
| 00:15.01 | [hC] | :) |
| 00:15.14 | [hC] | I just registered a 7960+7914 and a 7970 to chan_skinny in 1.4.11 :) |
| 00:15.16 | [hC] | finally... |
| 00:16.04 | [hC] | i did notice a couple things though, not sure if its chan_skinny or me doing something wrong... Pressing hold i dont believe puts the caller on hold (and doesnt give any indication that hold has been activated) and pushing transfer does nothing as well. I do have the moh and transfer stuff enabled in skinny.conf |
| 00:18.13 | TJNII | citats: That worked! Thanks! |
| 00:18.26 | citats | TJNII: no problem |
| 00:18.32 | TJNII | Though the queue doesn't, but that's OK as I understand why. |
| 00:19.11 | TJNII | Now my computer will call me when I need to change backup DVDs. Awesome. |
| 00:19.55 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 00:20.33 | *** join/#asterisk andresmujica (n=andresmu@correo.seaq.com.co) |
| 00:20.37 | Qwell | [hC]: neat, a 7914 was completely untested |
| 00:20.45 | Qwell | it should work just fine though |
| 00:20.54 | Qwell | do you get all the lines? |
| 00:21.43 | andresmujica | hi, anyone knows which card should i use to connect an * box to a PBX with an E1 Channel? i mean without PRI (no D-CHANNEL).?? |
| 00:21.55 | [hC] | still have to test that. so far it just turns on :) baby steps... but i will find out tomorrow for sure |
| 00:22.06 | Qwell | cool |
| 00:22.20 | [hC] | know what might be up with the hold/transfer? |
| 00:23.02 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.136.192) |
| 00:23.21 | Qwell | transfer needs trunk I think |
| 00:23.24 | Qwell | hold should work |
| 00:23.39 | [hC] | pushing hold/transfer both just do nothing |
| 00:23.45 | [hC] | im running 1.4.11 on there |
| 00:23.57 | Qwell | hold softkey? |
| 00:24.04 | Qwell | should work fine in 1.4 |
| 00:24.30 | [hC] | oh pardon me |
| 00:24.38 | [hC] | it put the caller on hold, but the phone showed no indication that you had done so |
| 00:24.56 | Qwell | ahh |
| 00:25.03 | Qwell | hmm |
| 00:25.06 | [hC] | is that expected? |
| 00:25.11 | Qwell | might be :P |
| 00:25.15 | [hC] | :) |
| 00:25.22 | Qwell | I don't have * source here.. give me a second |
| 00:25.26 | [hC] | I dont know how active you are in the development, but if you want, ill be your test monkey. |
| 00:26.15 | [hC] | hey also, for chan_mobile, to start by testing this out, attaching a BT dongle to an asterisk box, can i pair any phone that support bt headset? or should i specifically go find a certain model? I was gonna try with my blackberry, or my iphone first |
| 00:26.27 | Qwell | it should blink an icon |
| 00:26.27 | [hC] | yes thats right, i come out of the woodwork and pelt you with questions :) |
| 00:26.41 | [hC] | Qwell: yeah it wasnt doing anything.. on either the 7970 or the 7960 |
| 00:26.50 | Qwell | I'll have to try it out tomorrow |
| 00:26.55 | [hC] | ok |
| 00:27.06 | [hC] | anything youd like me to test, just say the word |
| 00:27.13 | Qwell | and any phone *should* work... |
| 00:27.22 | Qwell | of course, whether they actually do is an entirely different story :) |
| 00:27.25 | [hC] | kay :) |
| 00:27.31 | Qwell | I think a blackberry has been tested though |
| 00:27.33 | rpm | [hC], just got home.. that was fun. :) |
| 00:27.43 | rpm | tow truck drivers are retardedly slow. |
| 00:27.48 | [hC] | Qwell: headset and sms? |
| 00:28.03 | [hC] | rpm: no kidding! your car gonna be alright? any idea what happened? |
| 00:28.13 | Qwell | yeah, blackberry is listed in the chan_mobile.txt doc |
| 00:28.52 | [hC] | neet. okay... so the machine that does the bluetooth connectivity, does it actually have to have asterisk on it, or can it be a client that just sends information back to a * server? |
| 00:28.54 | Qwell | [hC]: a "mobile show devices" will say whether SMS is supported |
| 00:28.59 | rpm | [hc], not sure. its a diesel so it could be anything.. the starter turns but doesn't grab the flywheel.. so the engine could be seized. |
| 00:29.01 | Qwell | has to have asterisk |
| 00:29.21 | Qwell | I don't know if there is any remote dongle support in bluez |
| 00:29.27 | [hC] | Qwell: ahh okay.. so this isnt really ready for an office deployment yet. I guess maybe the new event arch will help that? |
| 00:29.37 | [hC] | er yeah i guess it would have to be bluez |
| 00:29.42 | [hC] | or a helper app that can speak to * |
| 00:29.52 | [hC] | rpm: yikes. warranty? |
| 00:30.07 | [hC] | rpm: i guess insurance covers you at the very least |
| 00:30.32 | rpm | yeah. |
| 00:30.48 | rpm | well, it was good talking.. i gotta take off to p.g. |
| 00:34.17 | *** join/#asterisk Road-rnnr (n=RoadPutz@66.119.167.162) |
| 00:37.32 | CCFL_Man2 | yeah yeah yeah!, two 684A subset ringers on ebay |
| 00:37.56 | *** part/#asterisk ecdpalma (n=ecdpalma@201-27-192-60.dsl.telesp.net.br) |
| 00:38.48 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 00:41.51 | hug1 | How do you guys feel about epygi? |
| 00:44.27 | *** join/#asterisk pLr (n=acer@unaffiliated/plr) |
| 00:44.58 | Nugget | I think it looks hard to spell. beyond that, I have no idea what epygi is. |
| 00:46.18 | Qwell | looks like a proprietary pbx to me |
| 00:46.21 | [hC] | im not sure how to pronounce it either. |
| 00:46.22 | Qwell | what's to feel? |
| 00:46.32 | [hC] | i want to say e-piggle |
| 00:46.44 | [hC] | oh wait thats an i |
| 00:46.45 | Qwell | e-piggy |
| 00:46.48 | [hC] | so e-piggy |
| 00:46.54 | [hC] | font makes it hard to tell :) |
| 00:47.00 | pLr | eppigea |
| 00:47.25 | Qwell | where you getting the ea from? |
| 00:47.27 | [hC] | eppijee |
| 00:49.04 | CCFL_Man2 | piggy piggy |
| 00:51.04 | *** part/#asterisk andresmujica (n=andresmu@correo.seaq.com.co) |
| 01:04.25 | *** join/#asterisk wglenncamp (n=wglennca@c-69-139-126-170.hsd1.ky.comcast.net) |
| 01:05.03 | wglenncamp | can someone direct me to a place where I can find a way to load my modprobe commands automatically on boot? |
| 01:06.11 | flenders | wglenncamp: #linux |
| 01:06.53 | flenders | I run my modprobes on my startup scripts |
| 01:07.24 | flenders | load the modules and then start asterisk, but that's just me |
| 01:07.45 | wglenncamp | okay, did you create a new script for your modprobe, or did you add them to an existing script? |
| 01:08.20 | wglenncamp | you are talking about in your init.d directory right? |
| 01:10.27 | hug1 | lol |
| 01:10.37 | hug1 | right i just stepped out of a meeting |
| 01:10.40 | hug1 | hehe |
| 01:10.45 | hug1 | thank you for ripping them off |
| 01:11.15 | hug1 | what i was looking for was... let me see |
| 01:11.30 | hug1 | what would the response be if I had to say Grandstream |
| 01:12.04 | wglenncamp | ((chuckle)) |
| 01:13.20 | hug1 | common ppl, u had a lot to say about "e-piggy" |
| 01:14.09 | Sci_05 | ~gs |
| 01:14.09 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 01:15.01 | wglenncamp | They are horrible quality wise... And they can't ever get their firmware right.. I only use them in our lab, but that's as far as it gets |
| 01:15.43 | hug1 | right, now, the same way the jbot dissed Grandstream because of past experience etc, is there anybody here who has worked with Epygi and have a opinion on whether they are good or bad? |
| 01:16.18 | hug1 | all right |
| 01:16.26 | hug1 | thank you wglenncamp |
| 01:17.30 | wglenncamp | And support from Grandstream is hit and miss. Example: new firmware posted to their site one day? Next day... Gone poof! FTP Server empty... |
| 01:18.19 | wglenncamp | I would recommend Polycom, and possibly Aastra Phones.. But that's my opinion |
| 01:18.31 | hug1 | right let me clarify, wglenncamp, u are speaking of grandstream right, not epygi |
| 01:18.53 | hug1 | yeah im looking for a FXO gateway |
| 01:19.02 | wglenncamp | Grandstream.. I have never dealt with epygi |
| 01:19.05 | hug1 | with more than one FXO port |
| 01:19.24 | hug1 | I proved the concept using a Grandstream |
| 01:19.41 | hug1 | now Im looking for something that can handle more than just oneoutgoing line |
| 01:19.52 | hug1 | one outgoing line* <sorry> |
| 01:20.16 | hug1 | I have asterisk for sip, bit its sip only |
| 01:20.27 | hug1 | so one sip phone can phone another |
| 01:20.40 | hug1 | I now want to give them access to the outside |
| 01:20.44 | hug1 | iow, PSTN |
| 01:21.10 | wglenncamp | Why not a card? |
| 01:21.16 | hug1 | dont tell me to use cards coz Im running OBSD |
| 01:21.21 | hug1 | and I know |
| 01:21.30 | hug1 | I inherited it just before the comments come |
| 01:21.44 | hug1 | and I cant change it |
| 01:21.53 | wglenncamp | Ah, I came in late on the conversation... Didn't know |
| 01:22.10 | hug1 | nah all good glenn |
| 01:22.13 | wglenncamp | How much are you looking to spend? |
| 01:22.16 | hug1 | I didnt mention that before |
| 01:22.26 | hug1 | money is not an object |
| 01:22.31 | hug1 | functionality is |
| 01:22.38 | hug1 | so i have lots |
| 01:22.56 | hug1 | but of course i dont want to rip the chicken through its you know what |
| 01:23.28 | hug1 | basically im looking for a fxo gateway that can deliver 6+ lines |
| 01:23.34 | hug1 | PSTN lines that is |
| 01:24.23 | [TK]D-Fender | hug1 : AudioCodes MP-114 4 FXO |
| 01:24.47 | wglenncamp | I was getting ready to mention that one.. |
| 01:24.51 | wglenncamp | Or a Mediatrix |
| 01:24.52 | hug1 | it must be reliable and fairly easy to configure, if nor well documented and supported |
| 01:25.00 | [TK]D-Fender | hug1, Or if you're sure you're always going to treat your lines homogeneously : Linksys SPA-400 |
| 01:25.31 | wglenncamp | Audiocodes or Mediatrix from what I heard are good. |
| 01:25.41 | wglenncamp | Never used one before.. And they are a little pricey |
| 01:25.42 | [TK]D-Fender | Mediatrix high density FXS gateways are a breeze, haven't tried their FXO's yet though. |
| 01:26.00 | wglenncamp | compared to grandstream that is. |
| 01:26.02 | wglenncamp | :) |
| 01:26.07 | hug1 | so is there an |
| 01:26.13 | [TK]D-Fender | ~gs |
| 01:26.13 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 01:26.13 | hug1 | sorry |
| 01:26.14 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 01:26.41 | wglenncamp | Will cost you about $500 - $600 for the gateway (Audiocodes or Mediatrix) |
| 01:26.58 | [TK]D-Fender | yup, thats what quality will do to you.... |
| 01:27.35 | hug1 | D-Fender: what do you mean by homogenously |
| 01:29.36 | [TK]D-Fender | hug1, meaning you don't want to say "line 1 goes HERE, line 2-4 go THERE), etc |
| 01:33.21 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 01:39.09 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:46.37 | *** join/#asterisk mcquaid (n=mcquaid@toronto-hs-216-138-233-79.s-ip.magma.ca) |
| 01:48.38 | mcquaid | what is the term used to describe when asterisk just basically negotiates the handshake of the call and backs out vs asterisk staying in between the call (for onhold music or whatever) |
| 01:49.06 | *** join/#asterisk ELINGE25 (n=pregunt@189.154.15.154) |
| 01:49.18 | ELINGE25 | alguien que hable espa;ol |
| 01:49.21 | *** join/#asterisk hug1 (n=hugo@electr319.lnk.telstra.net) |
| 01:49.30 | hug1 | hello |
| 01:49.57 | ELINGE25 | i need help any body here speak spanish? |
| 01:50.32 | hug1 | something strange is going on |
| 01:50.35 | *** join/#asterisk red9012 (n=marc3234@76-10-149-62.dsl.teksavvy.com) |
| 01:50.55 | red9012 | anyone here? |
| 01:50.58 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 01:51.20 | CCFL_Man2 | whats a good open sorce softphone? |
| 01:51.39 | ELINGE25 | soy nuevo en asterisk y en mi trabajo me solicitaron crear un proyecto de un callcenter .net alguien que me ayude |
| 01:51.57 | CCFL_Man2 | ELINGE25: engrish only plz |
| 01:52.28 | ELINGE25 | sorry ccfl_man2 but my english is very little |
| 01:52.56 | ELINGE25 | i need to create a callcenter whit C#.NET and i new in asterisk |
| 01:53.04 | CCFL_Man2 | ELINGE25: i unfortunately cannot help you, i only know wnglish |
| 01:53.15 | mcquaid | CCFL_Man2, twinkle, ekiga. |
| 01:53.18 | hug1 | D-Fender you there |
| 01:53.31 | mcquaid | linphone is a little rough aroudn the edges but also has a console client |
| 01:53.36 | CCFL_Man2 | mcquaid: twinkly build on osx? |
| 01:53.42 | red9012 | I would like to do the following: I want to be able to get a voice prompt before I accept a call. |
| 01:53.51 | mcquaid | oh. oh don't know. it's a kde3 app |
| 01:53.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:53.59 | CCFL_Man2 | ahh |
| 01:54.07 | mcquaid | well qt, i don't think it has kde deps |
| 01:54.16 | CCFL_Man2 | it doesn't |
| 01:54.35 | red9012 | for example, an incoming call is automatically forwarded to my cell using the dial cmd. I would to have a voice prompt telling me that this call is coming from my pbx, |
| 01:55.02 | red9012 | anyone in here knows how I can accomplish this |
| 01:55.06 | CCFL_Man2 | ELINGE25: as far as i know C#.net won't be of any help |
| 01:55.20 | mcquaid | what is the function to make asterisk bow out gracefully from calls and not sit inbetween the call for music on hold or whatever |
| 01:55.41 | CCFL_Man2 | maybe i'll try limphone |
| 01:56.11 | mcquaid | and i found linphone the best at getting around nat issues |
| 01:56.45 | mcquaid | i have a box behind a nat where ekiga asterisk, and twinkle only receiving incoming calls with no audio but ilnphone works fine |
| 01:58.17 | CCFL_Man2 | ahh, k |
| 01:58.36 | ectospasm | what protocol is linphone using? |
| 02:01.40 | CCFL_Man2 | sip |
| 02:03.20 | *** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au) |
| 02:03.21 | ectospasm | funny... SIP doesn't handle NAT very well... you usually gotta jump through hoops to get it working |
| 02:03.52 | hmmhesays | sometimes |
| 02:04.00 | hmmhesays | depends on the nat really |
| 02:09.35 | *** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl) |
| 02:09.52 | CrazyTux | ELINGE25, babblefish english might help |
| 02:11.53 | puzzled | evening all |
| 02:12.30 | *** join/#asterisk saftsack (n=saftsack@pD9E07EA7.dip.t-dialin.net) |
| 02:13.01 | puzzled | I get a "No Authority" error on anonymous incoming iax2 connections. Anyone care to enlighten me why? http://pastebin.ca/726315 |
| 02:17.01 | ELINGE25 | any body here speak spanish |
| 02:22.55 | CrazyTux | ELINGE25, www.systransoftusa.com |
| 02:23.11 | CrazyTux | ELINGE25, http://www.systransoft.com/ rather. |
| 02:23.31 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 02:24.54 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 02:30.28 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 02:30.28 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 02:33.17 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 02:33.37 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 02:34.14 | *** join/#asterisk PepOSX (n=pepOSX@190.72.149.171) |
| 02:43.24 | *** join/#asterisk mcquaid (n=mcquaid@toronto-hs-216-138-233-79.s-ip.magma.ca) |
| 02:46.37 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-112-81.knology.net) |
| 02:47.31 | mcquaid | what is the name of the cmd/option so asterisk just does the 'handshaking' and is not in between through the duration of the call? |
| 02:51.48 | Nugget | that's the default behavior. |
| 02:52.10 | mcquaid | ok... but what is it called to enable it then? |
| 02:52.42 | Nugget | start here: http://www.voip-info.org/wiki/view/Asterisk+SIP+media+path |
| 02:52.55 | mcquaid | or is it autoenabled when you use something like music on hold (which obviously needs it) |
| 02:53.17 | mcquaid | i remember reading there was a cfg option to force it (can't recall if it was to force it on or off) |
| 02:53.23 | mcquaid | but it's been awhile |
| 02:53.35 | Nugget | it's more complicated than that. |
| 02:53.59 | mcquaid | ok |
| 02:54.10 | mcquaid | but ah thx for the link, the term I was trying to think of was reinvite |
| 02:54.21 | Nugget | actually it's canreinvite. |
| 02:54.34 | Nugget | although you'll encounter many sample configs online that have it wrong |
| 02:55.33 | mcquaid | ah, ya, just reading that at the site you provided. thx again |
| 02:56.00 | mcquaid | i've been investigating other pbxes like the online one (pbxes.org) and their free account limits calls to 60 minutes |
| 02:56.23 | mcquaid | and that made me think it's always sitting inbetween the call (potentially adding latency) |
| 02:56.47 | *** join/#asterisk smgua (n=smelgar@190.56.109.27) |
| 02:56.56 | mcquaid | and I couldn't find any info on their site, tried to remember the asterisk cfg option for that |
| 02:57.27 | smgua | Anyone, is it safe to upgrade from 1.4.5 to 1.4.12 ¿? |
| 02:58.46 | ELINGE25 | any body here speak spanish |
| 02:59.45 | smgua | si |
| 03:00.27 | *** join/#asterisk hug1 (n=hugo@electr319.lnk.telstra.net) |
| 03:06.02 | hmmhesays | drupal sure seems like a pain in the ass |
| 03:06.29 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:09.21 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:14.27 | *** part/#asterisk mitcheloc (n=mitchel@adsl-67-121-104-74.dsl.irvnca.pacbell.net) |
| 03:14.36 | *** join/#asterisk asdx (n=foo@adsl-145-245.click.com.py) |
| 03:14.56 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:16.38 | smgua | again: anybody with 1.4.5 to 1.4.11,1.4.12 upgrade experience? |
| 03:19.28 | *** part/#asterisk smgua (n=smelgar@190.56.109.27) |
| 03:20.10 | *** join/#asterisk gremzoid (n=gremzoid@d58-111-173-16.rdl5.qld.optusnet.com.au) |
| 03:20.23 | *** join/#asterisk Dalbaech (n=Dalbaech@c-98-200-244-16.hsd1.tx.comcast.net) |
| 03:23.09 | *** join/#asterisk J4k3- (i=J4k3@28.sub-70-218-232.myvzw.com) |
| 03:27.44 | ELINGE25 | any body here speak spanish |
| 03:34.00 | hug1 | hey does anybody know of any other FXO gateway other than SPA400 that will work with asterisk |
| 03:34.29 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 03:35.53 | gremzoid | apart from sip and iax, can h323 peers/users be configured from mysql? |
| 03:37.04 | dan__t | Hrm, looks like using a Polycom phone with a SIP provider is shoddy at best. |
| 03:37.07 | dan__t | Anyone ever done that before? |
| 03:37.37 | hug1 | sorry dan nope |
| 03:37.46 | hug1 | hey does anybody know of any other FXO gateway other than SPA400 that will work with asterisk |
| 03:37.51 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:41.09 | bintut | anyone here able to make gtalk communication work on asterisk 1.4.11? |
| 03:41.50 | *** part/#asterisk bfrance (n=brian@adsl-68-72-34-207.dsl.ipltin.ameritech.net) |
| 03:41.59 | gremzoid | hug1, is that a response to me? |
| 03:42.24 | dan__t | I'm trying to get it to work with Teliax. |
| 03:42.41 | dan__t | I'm going to circumvent Asterisk for the time being, just for the sake of getting some connectivity here. |
| 03:42.55 | Dalbaech | I haven't used it bintut.... but there's a howto, but it suggests using trunk |
| 03:42.56 | Dalbaech | http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk |
| 03:43.03 | *** join/#asterisk saftsack (n=saftsack@pD9E05C43.dip.t-dialin.net) |
| 03:43.12 | Dalbaech | http://www.voip-info.org/wiki/view/Asterisk+Google+Talk |
| 03:43.30 | Dalbaech | so dunno |
| 03:45.07 | Dalbaech | dan: I've heard Polycom phones are generally evil. |
| 03:45.27 | Dalbaech | I used a few, and NAT is a nightmare |
| 03:45.34 | bintut | Dalbaech: i tried the latter site that you gave but doesn't work.. i mean, when i check if it's connected using the command "jabber show connected", it says it is |
| 03:46.29 | hug1 | again: hey does anybody know of any other FXO gateway other than SPA400 that will work with asterisk |
| 03:46.29 | Dalbaech | not sure bintut, I've never done it |
| 03:46.35 | Dalbaech | I might on the next rebuild of my pbx |
| 03:46.44 | Dalbaech | but for now, it works, so I'm not doing anything to it. |
| 03:46.51 | Dalbaech | soon, I'm shutting it down and starting from scratch |
| 03:49.04 | dan__t | Dalbaech, seems like the phones are solid, but very picky. |
| 03:49.11 | Dalbaech | indeed |
| 03:49.48 | dan__t | Like there's two SIP configuration sections. |
| 03:49.56 | dan__t | I can't differentiate between the two, honestly. |
| 03:50.13 | dan__t | I found a few half-assed hacked together bits of information on forums and such, but that's it. |
| 03:50.40 | Dalbaech | hehe |
| 03:52.40 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 03:57.06 | *** join/#asterisk xezz (n=phob@trust-it.gr) |
| 04:00.22 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 04:04.27 | dan__t | Too bad. |
| 04:06.11 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 04:07.32 | AlcateLXpert | hey, I am at the very begining of configuring asterisk. I can register my sip user, and in the documentation, it says that I can call extension 100 or 611 to have an echo of myself.. my x-lite says that the number can't be reach.. is there anything to do for that to work ? |
| 04:09.22 | flenders | do you have those extensions on the dialplan? |
| 04:09.40 | flenders | like 'exten => 611,1,echo()' |
| 04:11.29 | AlcateLXpert | what file is this in ? |
| 04:11.41 | flenders | extensions.conf? |
| 04:11.43 | AlcateLXpert | at the very begining in the doc, they talk about that, but not which file it s supposed to be in |
| 04:11.48 | hug1 | again: hey does anybody know of any other FXO gateway other than SPA400 that will work with asterisk |
| 04:12.01 | flenders | AlcateLXpert: what doc? |
| 04:12.16 | *** part/#asterisk hug1 (n=hugo@electr319.lnk.telstra.net) |
| 04:12.21 | AlcateLXpert | flenders, http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip |
| 04:13.20 | flenders | ~book |
| 04:13.21 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 04:14.14 | *** join/#asterisk hug1 (n=hugo@electr319.lnk.telstra.net) |
| 04:14.17 | flenders | AlcateLXpert: clearly you're missing something |
| 04:14.28 | AlcateLXpert | in thew doc ? |
| 04:14.30 | flenders | which chapters did you skip? |
| 04:14.41 | AlcateLXpert | bottom of page 71 |
| 04:15.12 | hug1 | lol, sorry guys got dc'd there, if you a message to my question please just post the question again.... the question was |
| 04:15.22 | AlcateLXpert | flenders, never mind |
| 04:15.36 | hug1 | has anbody installed or worked with FXo gateway s other PSA400 |
| 04:15.39 | flenders | AlcateLXpert: the thing is, it doesn't come pre-configured |
| 04:15.48 | flenders | maybe if you did a make samples |
| 04:16.33 | AlcateLXpert | i did |
| 04:16.41 | AlcateLXpert | i ll try again tomorrow |
| 04:16.44 | AlcateLXpert | it s getting late |
| 04:16.45 | flenders | what the book explains very well is how the whole thing works, I'm sure it doesn't tell you to create a sip account on sip.conf and dial 611 |
| 04:17.01 | flenders | maybe that sip account is on a different context |
| 04:17.39 | *** join/#asterisk arcanine (n=saxon_m2@203.82.44.179) |
| 04:17.46 | flenders | on extensions.conf you need to have the 'exten => 611,1,dosomething' on the same [context] as context= on sip.conf |
| 04:18.11 | AlcateLXpert | ok, let me try to add an internal context |
| 04:18.33 | flenders | if you pastebin your dialplan and your sip.conf I can see where you made a mistake |
| 04:18.47 | flenders | dialplan, I mean extensions.conf |
| 04:19.04 | flenders | get used to calling that file as dialplan |
| 04:20.42 | hug1 | so then i take it that everybody here has worked with the linksys SPA400 FXO gateway |
| 04:20.47 | hug1 | and with nothing else |
| 04:26.21 | AlcateLXpert | flenders, the [internal] exten ... should i put that in the sip.conf, or extension.conf ? |
| 04:26.28 | AlcateLXpert | from the doc, it looks like sip.conf ? |
| 04:26.38 | flenders | extensions.conf |
| 04:26.57 | flenders | on sip.conf, on your sip account details, you need to add 'context=internal' |
| 04:27.12 | AlcateLXpert | i have that under my user |
| 04:27.47 | AlcateLXpert | ok |
| 04:27.49 | AlcateLXpert | working now |
| 04:27.54 | AlcateLXpert | but no echo on what i m saying |
| 04:28.26 | flenders | is the x-lite on the same network as asterisk? |
| 04:28.40 | AlcateLXpert | yup |
| 04:29.59 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 04:30.05 | dan__t | All I want is to make SIP calls cha cha cha |
| 04:30.15 | FireMac | hi where can i get mpg123? |
| 04:30.55 | *** part/#asterisk gremzoid (n=gremzoid@d58-111-173-16.rdl5.qld.optusnet.com.au) |
| 04:31.04 | AlcateLXpert | flenders, must be my stupid windows, it s working on my mac. |
| 04:31.44 | flenders | macs are great |
| 04:31.46 | flenders | :D |
| 04:32.36 | AlcateLXpert | yeah. ok.. looks like it was a conflict with skype |
| 04:32.38 | AlcateLXpert | it s working no |
| 04:32.45 | arcanine | hi |
| 04:32.54 | AlcateLXpert | i ll continue reading the doc, and see how cool is it to have a SIP trunk with my Alcatel LAB |
| 04:32.55 | arcanine | wat is the cpu requirement for asterisk |
| 04:32.57 | AlcateLXpert | Thanks for the help |
| 04:33.21 | Qwell | arcanine: it depends |
| 04:33.24 | Qwell | at least a 486 |
| 04:33.51 | arcanine | right now im using pentium 3 |
| 04:34.01 | Qwell | that's plenty, depending on what you want to do |
| 04:34.11 | Qwell | obviously you can't expect to send hundreds of calls at it |
| 04:34.20 | Qwell | but a few dozen would probably be okay |
| 04:34.35 | arcanine | with 256 memory, but when simultaneuous call of 20.. calls are dropped |
| 04:35.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:35.55 | flenders | so you mean, 20 simultaneous calls are fine, but it can't handle 21? |
| 04:36.38 | arcanine | 19 simultanous calls are ok but 20 calls drop |
| 04:37.22 | arcanine | cant handle 20 calls and up |
| 04:37.36 | flenders | well, there you go. |
| 04:38.02 | arcanine | do i need to change cpu? |
| 04:38.16 | flenders | how's the system load while you have 19 calls? |
| 04:38.55 | flenders | how many users you have? |
| 04:39.27 | flenders | and what sort of CPU is it? |
| 04:41.14 | dan__t | Anyone ever been successful in setting up a PolyCom phone for use with an external SIP provider? I'd like to use the phone with Teliax for a bit here until I get * working as I'd like it to. |
| 04:42.00 | arcanine | i hav 25 users, cpu is advantech pentium 3 1ghz, 256 mb sdram |
| 04:42.51 | flenders | when you say 19 simultaneous calls, are these calls to the outside world? |
| 04:43.00 | flenders | or even calls between users |
| 04:44.10 | xezz | hello, can i connect a Siemens HIPath 3700 with asterisk ? is this possible ? |
| 04:45.44 | flenders | dan__t: I'm sure it works |
| 04:45.57 | flenders | I've seen people doing it, though, haven't done it myself |
| 04:47.32 | dan__t | I never doubted that it does not work. |
| 04:47.44 | dan__t | It just seems that the setup involved is tricky, one I've not managed to hammer out. Yet. |
| 04:49.24 | arcanine | 19 simultaneous calls to the outside world |
| 04:49.46 | flenders | arcanine: you have a PRI? |
| 04:53.38 | J4k3 | arcanine: my P3-700/100 (1:1 internal cache version) asterisk box chokes really quick, suprisingly so |
| 04:54.48 | flenders | J4k3: even with no transcoding? |
| 04:56.37 | arcanine | yes |
| 04:56.44 | [TK]D-Fender | dan__t, I've done it once. |
| 04:57.05 | dan__t | Remember any of it? heh |
| 04:57.16 | arcanine | do i need to change box, from p3 1ghz to core2duo |
| 04:57.58 | arcanine | is it advisable to change box |
| 04:58.20 | [TK]D-Fender | dan__t, Nothing to remember. IP/host, user, pass. End of story. |
| 04:58.32 | dan__t | ok, there's a SIP settings dialog, and a per-line SIP dialog. |
| 04:58.33 | AlcateLXpert | anyone tried the Linksys SPA941 SIP Phone with sterisk ? |
| 04:58.41 | flenders | AlcateLXpert: I have a few |
| 04:58.52 | AlcateLXpert | flenders, r u happy with them ? or do you have anything better ? |
| 04:59.12 | flenders | AlcateLXpert: Polycoms are better |
| 04:59.17 | AlcateLXpert | which one ? |
| 04:59.30 | flenders | AlcateLXpert: but the SPA941s are the cheapest phones I would use in production |
| 04:59.38 | flenders | anything cheaper than that is shit |
| 04:59.48 | flenders | polycoms 330s are good |
| 04:59.50 | AlcateLXpert | lol |
| 04:59.50 | AlcateLXpert | ok |
| 04:59.59 | flenders | I have a few 430s too |
| 05:00.06 | lonekazoo | I'm very happy with polycom 601's. a couple of quirks, but overall very nice. |
| 05:00.24 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 05:00.26 | dan__t | Do you remember which settings apply to which dialog, [TK]D-Fender? |
| 05:00.29 | flenders | 601s are too expensive |
| 05:00.40 | flenders | most people don't need that |
| 05:00.57 | AlcateLXpert | flenders, hey, do you have any idea if it s easy to create a sip trunk toward a real-world pbx ? |
| 05:01.08 | flenders | you only need anything more than 430 if you have more than 2 simultaneous calls |
| 05:01.13 | flenders | which most users dont |
| 05:01.16 | AlcateLXpert | my issue is that for every sip user on the asterisk, I need their extension to show up on the PBX that they are calling |
| 05:01.27 | lonekazoo | i picked up a boatload of 601's on ebay for 159, new in the box. |
| 05:01.36 | [TK]D-Fender | AlcateLXpert, Being in North America there is virtuall no reason for you to consider anything other than Polycom... |
| 05:01.41 | flenders | lonekazoo: nice |
| 05:01.41 | J4k3 | sexy phones are a must. |
| 05:01.47 | flenders | I probably missed that one |
| 05:02.12 | J4k3 | 1, I'm switching to Cingular... 2, it should kick ass as a wifi sip handset |
| 05:02.19 | [TK]D-Fender | dan__t, its just those 3, and by dialog its sounding like you are configurin your via the web interface. That is a true poor choice... |
| 05:02.36 | dan__t | I'm sure it is a poor choice. Good thing I'm not doing it. |
| 05:02.44 | flenders | AlcateLXpert: can you explain it? |
| 05:02.55 | dan__t | Ok, SIP dialog and per-line dialogs. That makes two. Am I missing a third? |
| 05:03.03 | AlcateLXpert | where on the polycom website are those 330/440 ? |
| 05:03.04 | [TK]D-Fender | flenders, And no, you don't need a bigger phone to handle more than 2 calls. I had my IP 430 set to handle *10* |
| 05:03.14 | dan__t | Which takes precedence, and which should contain my SIP provider's login info? |
| 05:03.20 | [TK]D-Fender | AlcateLXpert, www.telephonydepot.com |
| 05:03.25 | flenders | [TK]D-Fender: well done! and thanks for the info |
| 05:04.04 | lonekazoo | on the 601's, you cant configure more than 1 sip server registration, you have to use the web interface or xml |
| 05:04.15 | [TK]D-Fender | flenders, 2 registrations, each getting 1 line-key, each linekey set to accept up to 5 simultaneous calls. |
| 05:04.34 | dan__t | Was that for me, lonekazoo? |
| 05:04.49 | dan__t | Good to know regardless. |
| 05:04.55 | flenders | [TK]D-Fender: that's a lot of calls mate |
| 05:06.24 | Qwell | [TK]D-Fender: I once had a cisco doing 100+ :p |
| 05:06.24 | Qwell | that was funny |
| 05:06.24 | flenders | [TK]D-Fender: I wonder how you would switch to different calls |
| 05:06.27 | [TK]D-Fender | flenders, never HAD that many, but was configured for it. Always look at how you span your regs', line-keys, and CALLS-PER line-key |
| 05:06.45 | [TK]D-Fender | flenders, using the cursor keys... looks even NICER actually. |
| 05:06.46 | AlcateLXpert | flenders, I need to setup a solution for over 4,000 users |
| 05:06.46 | Qwell | ciscos have a never-ending scrollable list |
| 05:07.00 | *** join/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
| 05:07.01 | AlcateLXpert | AlcateLXpert, can't sspend too much money. so i was thinking about using Asterisk |
| 05:07.13 | [TK]D-Fender | flenders, you kno when a CW beep call comes in you only see the name on the bottom> |
| 05:07.14 | [TK]D-Fender | ? |
| 05:07.15 | AlcateLXpert | flenders, then connect the asterisk to an alcatel |
| 05:07.24 | flenders | yeah |
| 05:07.26 | Qwell | 4,000 seats is going to be expensive.. |
| 05:07.30 | AlcateLXpert | flenders, and have the alcatel send and receinve the calls to the PRI |
| 05:07.35 | Qwell | figure $150+ per seat |
| 05:07.44 | AlcateLXpert | Qwell, with alcatel, yeah.. so i need a cheaper solution |
| 05:07.59 | Qwell | with anything. asterisk will be less expensive, sure |
| 05:08.12 | [TK]D-Fender | flenders, When you allow more than 1 call per key, you get BOTH in a "top half / bottom half" way and get to see it better. You scroll to the call like Cisco's. |
| 05:08.23 | Qwell | but don't expect to get away with only spending a few thousand dollars (if you include phones) :) |
| 05:08.24 | AlcateLXpert | so i was thinking about doing IP trunking |
| 05:08.26 | AlcateLXpert | or sip trunking |
| 05:08.33 | AlcateLXpert | or use the alcatel as a sip gateway |
| 05:08.38 | flenders | [TK]D-Fender: nice, I'll do it on mine! |
| 05:08.49 | karleeto | [TK]D-Fender: ever done an intercom type thing with the auto-answer function on polycoms? |
| 05:08.54 | [TK]D-Fender | flenders, using the up/down keys. |
| 05:08.58 | dan__t | Bah. Forget it. |
| 05:09.10 | [TK]D-Fender | karleeto, Yup, I've done just about everything on them except VLAN's |
| 05:10.13 | [TK]D-Fender | dan__t, in provisioning, just look at reg.1.(blah). about 6 or so fields you'll want to fill in to set your account up and configure your linekey allocation. |
| 05:11.06 | AlcateLXpert | i ll work on that tomorrow morning. gonig to bed tired. thanks flanders |
| 05:11.23 | dan__t | Yea, I'm trying to do it all from the phone. Like I said, the phone itself has two SIP-related dialogs, the "Main" configuration and one configuration per each line. I will note that the "Main" configuration doesn't ask for authentication paramaters to be set etc etc, where as the per-line configurations do. |
| 05:11.52 | flenders | dan__t: I don't think you can set it up on the phone |
| 05:12.52 | flenders | AlcateLXpert: no worries |
| 05:12.58 | [TK]D-Fender | dan__t, under "SIP" you set the server type / IP, etc. under "LINES" you pick one to configure and add the user, pass, and line-key setup. |
| 05:13.26 | dan__t | Don't I need to specify the local external IP address per NAT workarounds etc etc? |
| 05:13.32 | [TK]D-Fender | dan__t, and yes you CAN set it up from the phone. But just wait for 10 reboots as you putz your way through those screens :) |
| 05:13.41 | [TK]D-Fender | dan__t, generally no. |
| 05:13.47 | karleeto | [TK]D-Fender: do you think that the polycom auto-answer page on voip-info is the way to go? |
| 05:13.49 | dan__t | Ok.... |
| 05:13.49 | dan__t | hrm |
| 05:14.18 | [TK]D-Fender | karleeto, pretty close. Watch out for the new way to set the headers based on * version |
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| 05:30.36 | luke-jr | â‘ |
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| 05:42.37 | karleeto | [TK]D-Fender: i know _3XX would mean 3 and any two digits, but what would _3ZX mean? |
| 05:43.20 | [TK]D-Fender | karleeto, http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
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| 06:19.59 | Chris-NB | hi |
| 06:20.14 | Chris-NB | anyone using a sangoma card with hwec running? |
| 06:20.33 | Chris-NB | i'm getting problems with digital calls (data calls) when the hwec is running |
| 06:21.27 | karleeto | This is my site config: http://rafb.net/p/jLdi0D65.html |
| 06:22.23 | karleeto | will this work for polycom auto-answer intercom? i wasnt fond of editing my sip.cfg, so i thought i'd just include it in my site.cfg, which should override anything from sip.cfg, right? |
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| 06:33.20 | arekm | any chan_zap.c guru available? |
| 06:33.24 | Grandfrere | Anybody know anything about altering inbound call volume on a Sip channel? |
| 06:33.57 | arekm | chan_zap.c ignores pulsedial=no in zapata.conf and changes that to yes runtime, why is that? |
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| 06:56.04 | kaldemar | arekm: pastebin your zapata.conf and tell what channels should have pulsedial=no |
| 06:58.00 | arekm | kaldemar: every channel should have pulsedial=no |
| 06:58.15 | arekm | kaldemar: anyway I've bugreported it http://bugs.digium.com/view.php?id=10894 |
| 06:58.30 | arekm | now I'm waiting to see: closed won't fix or something ;) |
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| 07:43.43 | tzafrir_home | arekm, pulsedial=yes/no affects how you dial out (through FXO), not if you identify pulse tones |
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| 07:54.03 | Raky100 | wooooooah |
| 07:54.06 | Raky100 | that's a lot of users |
| 07:54.18 | Raky100 | hey guys, i've been having a bit of an issue and was hoping someone could help |
| 07:54.25 | Raky100 | I've trunked two machines, one in AU the other in the US |
| 07:54.44 | Raky100 | calls between AU->AU are fine, and US->US are fine. |
| 07:55.06 | Raky100 | However, when i make a direct call to a US extension, from the AU line the quality is bad for people on the AU side. However, people in the US can here me fine. |
| 07:55.25 | Raky100 | I have to ask them to goto the conference line on the US side, and that works fine too, i can hear them perfectly then. It's only when i call them directly that is plays up. |
| 07:55.31 | Raky100 | Any idea as to what could be going on? |
| 07:55.44 | reaxion | Hi. |
| 07:56.15 | sevard | Hi |
| 07:56.45 | Raky100 | Hi. |
| 07:56.49 | reaxion | If I put more .call files in /var/spool/asterisk/outgoing than I have channels, what will Asterisk do? Will it queue the additional calls or ignore them? I'm testing out a travel alerts service so would prefer if it queues |
| 07:59.20 | tzafrir_home | Raky100, what type of call? PSTN? VoIP? |
| 07:59.34 | Raky100 | They are VoIP calls. |
| 07:59.41 | *** join/#asterisk tamp4x (n=roger@nimbuzz.xs4all.nl) |
| 07:59.56 | tzafrir_home | Asterisk will retry them |
| 08:00.01 | tzafrir_home | (The call files) |
| 08:00.14 | Raky100 | I just don't get how the quality sounds weird over a direct call. But then I goto the meetme line with them, and it works fine? |
| 08:00.46 | reaxion | tzafrir_home: Retry? So they'll fail the first time because of no available channels? |
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| 08:02.46 | tzafrir_home | reaxion, but some of them succeed, right? |
| 08:03.33 | reaxion | All that can fit will succeed, yes |
| 08:04.12 | reaxion | Interesting one. Asterisk doesn't move/delete the call file until the call is complete? |
| 08:04.34 | arekm | tzafrir_home: why then pulsedial is changed to yes if pulse comes in? |
| 08:05.38 | arekm | is there a way to put additional (my own) field in zapata.conf/sip.conf for some channel like mymessage=abc and access it from dial plan later? |
| 08:07.38 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.51) |
| 08:08.44 | arekm | Or is there a way to know which Zap is calling to me? |
| 08:09.01 | arekm | Well, the problem is actually that I need separate callerid for outgoing connections and separate for local calls |
| 08:10.12 | arekm | I was thinking about setting callerid=id_for_outgoing and adding mylocalcallerid=22 and then in dialplan do something like: if localcall then fetch(mylocalcallerid, for zap/22) |
| 08:10.21 | arekm | and set that as new callerid |
| 08:13.11 | tzafrir_home | arekm, you can set channel variables from sip.conf (and from zapata.conf in trunk) |
| 08:13.38 | tzafrir_home | You can also use accountcode |
| 08:14.29 | *** join/#asterisk dimgr (n=btsadmin@193.92.98.181) |
| 08:14.34 | dimgr | hi |
| 08:14.40 | dimgr | does asterisk detects my card? |
| 08:14.47 | dimgr | core show channels |
| 08:14.48 | dimgr | Channel Location State Application(Data) |
| 08:14.48 | dimgr | 0 active channels |
| 08:14.49 | dimgr | 0 active calls |
| 08:15.00 | dimgr | capi info |
| 08:15.01 | dimgr | Contr1: 8 B channels total, 8 B channels free. |
| 08:15.54 | luke-jr | ☺ |
| 08:16.08 | dimgr | no? |
| 08:20.16 | arekm | tzafrir_home: setvar=var=value? in extensions it's just ${var} then? |
| 08:20.23 | *** join/#asterisk stony (n=stony@p57B38B35.dip0.t-ipconnect.de) |
| 08:20.27 | stony | hi |
| 08:20.44 | tzafrir_home | arekm, yes |
| 08:21.12 | stony | i can't get my spa |
| 08:21.12 | stony | 901 from linksys to work |
| 08:21.15 | stony | are there any howtos or something ? |
| 08:21.36 | *** join/#asterisk appelza (n=d@dsl-240-129-197.telkomadsl.co.za) |
| 08:24.12 | arekm | tzafrir_home: ok, trying exten => _XX,9,IF(LEN(${callerid_local} > 0)?SET(CALLERID(num)=${callerid_local})) |
| 08:25.46 | arekm | uh, bad |
| 08:27.36 | sparq | Holy crap! There is a VoIP application for the Nintendo DS! |
| 08:27.54 | sparq | http://libw11.free.fr/svsip/ |
| 08:28.07 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:28.38 | arekm | [Oct 5 10:27:32] WARNING[8550]: pbx.c:1797 pbx_extension_helper: No application 'If' for extension (from-local, 41, 10) |
| 08:28.42 | arekm | huh? |
| 08:30.35 | arekm | ok, seeing now, cannot be used directly :-/ |
| 08:32.09 | tzafrir_home | GotoIf ? |
| 08:34.34 | appelza | Hi guys, I have a wireless phone that rings when the number is dialed, but then asterisk female voice tells the caller that the number you have dialed is not in service.. |
| 08:34.35 | appelza | any ideas? |
| 08:35.28 | Nugget | what's the asterisk console say? |
| 08:36.16 | appelza | I'll paste |
| 08:36.21 | appelza | (I cant figure it out) |
| 08:36.22 | Nugget | use a pastebin |
| 08:36.35 | Nugget | ~pastebin |
| 08:36.36 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 08:36.38 | appelza | ok |
| 08:36.58 | dimgr | how do you know if asterisk really detects your card? core show channels says 0 active channels |
| 08:37.36 | arekm | tzafrir_home: is > or < supported at all? http://pastebin.com/m6c3a1021 |
| 08:38.04 | appelza | http://pastie.caboo.se/104006 |
| 08:39.27 | *** join/#asterisk xezz (n=pagalos@athedsl-50891.home.otenet.gr) |
| 08:40.19 | tzafrir_home | GotoIf(Len(${callerid_local}) > 0?from-local,${EXTEN},19:from_local,${EXTEN},20) |
| 08:40.35 | tzafrir_home | from-local and from_local are two different names |
| 08:40.47 | tzafrir_home | arekm, ===^ |
| 08:40.56 | Nugget | good eye, tzafrir |
| 08:41.41 | Nugget | appelza: paste your dialplan too, I guess. hard to tell from just the console log |
| 08:42.14 | *** join/#asterisk blq (n=Bl@dslb-088-067-021-081.pools.arcor-ip.net) |
| 08:42.32 | arekm | tzafrir_home: ah, right, fixed that but still dials with prio 19 http://pastebin.com/m527ce3f8 |
| 08:42.42 | Nugget | http://forums.digium.com/viewtopic.php?p=36064 might be meaningful. |
| 08:43.18 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 08:46.32 | xezz | hello,i would like to connect asterisk(digium te110p) with siemens Hipath 3700(2 pri card), any idea ? |
| 08:50.17 | *** join/#asterisk qdk_ (n=qdk@85.235.253.139) |
| 08:55.00 | arekm | tzafrir_home: setvar=callerid_local=999 |
| 08:55.28 | arekm | tzafrir_home: exten => _XX,10,GotoIf(LEN(${callerid_local}) .... and -- Executing [41@from-local:10] GotoIf("SIP/101-08239098", "LEN() > 0?from-local |
| 08:55.37 | arekm | tzafrir_home: so it even doesn't see this variable due to some reason |
| 09:04.04 | *** join/#asterisk LeFallen (n=fallen@mail.stabat.com) |
| 09:04.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:05.19 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:07.00 | *** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 09:07.26 | Zeeek | ok |
| 09:19.24 | ai-a | is it PBX or PABX ? |
| 09:20.02 | *** join/#asterisk Kandinsky (n=cristi@perla2.tm.ew.ro) |
| 09:24.24 | LeFallen | I have a question if I may: When I get "ast_channel_bridge: Can't make SIP/XXX and SIP/YYY compatible", is that a codec issue? |
| 09:28.59 | *** join/#asterisk Conductor (n=Conducto@i59F78139.versanet.de) |
| 09:29.14 | Conductor | i have a problem receiving calls from umts networks |
| 09:29.20 | Conductor | sometimes you can see the call on the CLI but sometimes you dont (the handset says: number unknown) |
| 09:29.38 | Conductor | is there a way to watch the incoming data on a zaptel device? |
| 09:31.05 | ai-a | LeFallen: are they on the same network ? |
| 09:31.28 | *** join/#asterisk PSU_Boss_1 (n=Eric@unaffiliated/psuboss/x-309451) |
| 09:31.48 | ai-a | Conductor: should all go via the pbx... is this ext -> internal calls or internal -> internal ? sure your not calling the sip device directly ? |
| 09:32.48 | Conductor | ai-a, hmmm... what sip phone? |
| 09:33.05 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:33.25 | ai-a | a sip phone is a sip device. |
| 09:33.25 | Conductor | ai-a, but it's external-><whatever> |
| 09:33.44 | ai-a | Conductor: ok, what debug level you got enabled ? |
| 09:34.13 | Conductor | you mean core set verbose? |
| 09:34.30 | ai-a | asterisk -rvvvvvvvvvvvvvvv |
| 09:35.10 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:35.29 | Conductor | asterisk -rvvvvvvvvvvvvvvv |
| 09:35.32 | Conductor | yes |
| 09:35.41 | Zeeek | LeFallen one way to see would be to set both ends to ulaw |
| 09:37.04 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:37.13 | Conductor | ai-a, good. i call but i cant see anything on the cli |
| 09:37.33 | Conductor | i wonder if the call comes in on the zaptel card |
| 09:37.45 | Conductor | maybe asterisk just doesn't accept it... |
| 09:37.59 | Conductor | how can i find out? |
| 09:38.52 | Zeeek | Conductor you sau zaptel? |
| 09:38.59 | Zeeek | s/sau/say/ |
| 09:39.26 | Conductor | zeedo, well... its a digium TE420. so it is zaptel right? |
| 09:40.45 | Conductor | how can i make this zaptel channel more verbose? |
| 09:41.01 | Zeeek | what does CLI say when a call comes in now? |
| 09:41.06 | Conductor | nothing |
| 09:41.10 | Conductor | that's the problem |
| 09:41.20 | Conductor | from gsm it works |
| 09:41.28 | Conductor | from umts it only works sometimes |
| 09:41.36 | Zeeek | and from gsm what is on the CLI? |
| 09:41.44 | LeFallen | ai-a: Sorry, thanks... Yes they are. |
| 09:42.04 | Conductor | <PROTECTED> |
| 09:42.04 | Conductor | <PROTECTED> |
| 09:42.04 | Conductor | <PROTECTED> |
| 09:42.04 | Conductor | <PROTECTED> |
| 09:42.44 | Zeeek | my asterisk died last night. I ran to the office to see if it was on fire. In fact, it was the router that died. asterisk just basically froze with no internet connection. |
| 09:43.10 | LeFallen | Zeeek: I have configured the server to only allow alaw/ulaw but it doesn't help any. |
| 09:43.32 | Zeeek | LeFallen then use sip debug to see what's happening |
| 09:44.17 | Zeeek | Conductor if there's no "Starting simple sw" from CLI from utms, then it's something in the way the card is configured or the card itself. Call or email digium support |
| 09:44.41 | Zeeek | or wait until the US wakes up and more people are around |
| 09:45.15 | Conductor | Zeeek, OK thanks... |
| 09:45.50 | Conductor | Zeeek, i heard of some way to watch the whole zaptel traffic (with ethereal)... do you know anything about that? |
| 09:46.26 | Zeeek | No, I don't because ethereal sees the LAN traffic, not the zap |
| 09:46.40 | dj_instinct | hi all - do I _need_ a soundcard to monitor / increase the call volume on a digitum fxo card? |
| 09:46.48 | LeFallen | Zeeek: I did that but I'm not so good at understanding what I'm seeing. :( Kinda got lumped with this. |
| 09:47.16 | Zeeek | dj_instinct you need the zaptel utility to set the gain |
| 09:47.44 | LeFallen | Zeeek: All I get is some "codec translation path from g729 to alaw" errors, followed by a "no path to translate from SIP/XXX(256) to SIP/YYY(8)" |
| 09:47.51 | Zeeek | LeFallen post the appropriate debug stuff (cut trest out) to pastebin |
| 09:48.20 | Conductor | Zeeek, so there is no way to find out what actually comes in on the digium card? |
| 09:48.39 | Zeeek | Conductor I don't know, sorry |
| 09:48.47 | Conductor | ok thanks |
| 09:48.55 | Zeeek | LeFallen so it IS a codec issue because that's what asterisk tells you |
| 09:50.05 | Zeeek | btw do you have g729 installed? |
| 09:50.57 | *** join/#asterisk michael-i (n=michael-@141.41.40.186) |
| 09:53.15 | LeFallen | Zeeek: That's what I figured. I don't have g729 codec listed in "show translation" but I don't know how to stop it from trying to use it. Could it be the hardphone @ the answer point requesting it? |
| 09:53.20 | LeFallen | Zeeek: I set disable=all allow=alaw;ulaw but it still seems to request g729 |
| 09:53.35 | LeFallen | Zeeek: Thanks a lot for the help btw |
| 09:53.37 | Zeeek | LeFallen could be the phone indeed |
| 09:54.00 | LeFallen | Zeeek: http://pastebin.org/4164 < |
| 09:54.45 | BiG^DoG | is there a good cordless SIP phone? |
| 09:54.56 | Zeeek | LeFallen that's a MESS! |
| 09:56.02 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:57.52 | LeFallen | Zeeek: Sorry :S |
| 09:58.48 | Zeeek | you have to make the other end NOT use g729 |
| 10:02.21 | Conductor | when setting DEBUG=YES in /etc/sysconfig/zaptel where do the debug messages go? |
| 10:08.08 | LeFallen | Zeeek: So that is the problem then ... OK thanks heaps for your assistance :) |
| 10:08.30 | Zeeek | I hope it's worth the price :) |
| 10:09.52 | *** join/#asterisk IvanV3835 (n=Miranda@styx.mcn.ru) |
| 10:19.49 | tzafrir_home | to answer his question: it sets the "debug" module parameter (and as a side effect - fails to loading of xpp modules...) |
| 10:24.27 | Zeeek | hi tzafrir_home |
| 10:24.39 | tzafrir_home | hi |
| 10:26.52 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:27.11 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 10:49.44 | *** join/#asterisk saftsack (n=saftsack@pD9E05C43.dip.t-dialin.net) |
| 10:52.54 | *** join/#asterisk ming_zym (n=ming_zym@124.254.56.252) |
| 10:55.13 | Zeeek | . |
| 10:55.17 | Zeeek | so quiet |
| 10:59.21 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582170.dsl.bell.ca) |
| 11:01.33 | Maliuta | well, I would go to bed (at 21:00 on a friday) and get my weekend started early, but I don't think I could cope with the disappointment |
| 11:02.00 | Zeeek | which disappointment is that? |
| 11:03.44 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 11:03.46 | luke-jr | Whee |
| 11:03.59 | Maliuta | my weekend |
| 11:04.02 | luke-jr | Caller ID names from my live address book stored in IMAP ☺ |
| 11:07.49 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:10.41 | *** join/#asterisk gremzoid (n=gremzoid@d58-111-173-16.rdl5.qld.optusnet.com.au) |
| 11:11.16 | gremzoid | hmmm can anyone point me in the right direction, i'm trying to get asterisk to load it's configuration from SQL |
| 11:11.33 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 11:11.38 | gremzoid | i've spent hours building a database and a pretty little php UI for it... |
| 11:12.19 | gremzoid | i start asterisk and get good signs: Binding iaxusers to mysql/chps/asterisk_iax_conf |
| 11:12.53 | gremzoid | however it dosn't seem to work (read: what do i do now?) |
| 11:13.01 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.136.195) |
| 11:13.19 | JerJer | is there some way to send something other than 489 Bad Event back when someone sends the standard notify nat ping? |
| 11:15.30 | Zeeek | hey JerJer |
| 11:15.39 | Zeeek | up late? |
| 11:15.51 | Zeeek | or awake early |
| 11:25.40 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 11:25.42 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
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| 11:26.11 | JerJer | dpm |
| 11:26.13 | JerJer | grr |
| 11:26.26 | JerJer | sleep is for the weak |
| 11:26.30 | Zeeek | heh |
| 11:27.01 | *** join/#asterisk zsilak (n=spam@195.230.180.186) |
| 11:27.06 | zsilak | hi all |
| 11:27.50 | zsilak | I have an issue with asterisk / PRI on a digium te110p card |
| 11:28.12 | JerJer | ok and? |
| 11:28.20 | JerJer | lots of ppl have lots of issues |
| 11:28.33 | JerJer | those that ask specific questions have a chance of solving their issues |
| 11:28.43 | Zeeek | JerJer welcome back! :) |
| 11:28.48 | zsilak | I'll pastebin the log-file so it's easier to explain |
| 11:29.39 | zsilak | http://pastebin.ca/726629 |
| 11:30.20 | zsilak | i want to know if it is possible (and how) to change the pri setup message |
| 11:30.24 | zsilak | interface implicitly identified |
| 11:30.29 | zsilak | and B-channel selection: exclusive |
| 11:32.44 | zsilak | i know it's somehow very deep into the material :D can anybody help? |
| 11:32.48 | JerJer | that's over my head |
| 11:33.00 | zsilak | same here :D |
| 11:33.10 | Zeeek | I don't have any head |
| 11:33.24 | Zeeek | at least not recently :( |
| 11:35.17 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 11:48.08 | zsilak | found it ! |
| 11:48.18 | Zeeek | what was it |
| 11:48.28 | zsilak | if somebody asks in the future: |
| 11:48.40 | zsilak | if you comment out in zapata.conf |
| 11:48.41 | zsilak | [trunkgroups] |
| 11:48.41 | zsilak | ;trunkgroup => 1,16 |
| 11:48.41 | zsilak | ;spanmap => 1,1 |
| 11:49.04 | zsilak | it will do it not explicitly, but implicitly |
| 11:49.49 | zsilak | ( i think i mean the channel association ) |
| 11:50.15 | zsilak | cu (-: |
| 11:50.33 | gremzoid | what crap... i spent all afternoon setting up databases and cfg files... and all i get is: Realtime mapping for 'iaxpeers' found to engine 'mysql', but the engine is not available |
| 11:51.02 | Zeeek | a little like clicking on "contact us" and getting 404 page not found |
| 12:03.40 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 12:05.36 | thewiizle | anyone know of an AGI script that can check a trunk is working |
| 12:05.50 | *** join/#asterisk coppice (n=chatzill@142.204.17.210.dyn.pacific.net.hk) |
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| 12:07.39 | *** part/#asterisk Zeeek_ (n=Heh@213.223.114.24) |
| 12:11.46 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:11.59 | lirakis | morning everyone |
| 12:12.38 | *** join/#asterisk rogerz (n=highvolt@nucleabio.com) |
| 12:16.25 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:21.55 | Aurs | can i find hi-res asterisk logos somewhere? |
| 12:23.32 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 12:25.28 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:28.14 | key2 | someone knows how much HPEC uses the CPU ? |
| 12:30.26 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 12:35.55 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:41.37 | coppice | Hits Processor's Every Cycle |
| 12:41.57 | tzanger | good morning coppice |
| 12:42.15 | coppice | hi |
| 12:42.15 | tzanger | HPEC's supposed to be pretty "heavy" code in that regard |
| 12:43.04 | tzanger | coppice: have I told you lately how much I love that sliptest app? I've done some mods to it to send arbitrary data for detection of corruption but all the same, I love that util, and I want to thank you for it |
| 12:43.43 | coppice | tzafrir keeps complaining it won't work for him |
| 12:44.10 | _x86_ | tzanger: HPEC or HWEC? |
| 12:44.30 | tzanger | coppice: won't work as in what |
| 12:44.47 | tzanger | _x86_: HPEC runs on the host CPU. HWEC (to me) is hardware echo can |
| 12:44.48 | coppice | he says he never gets a stable lag value |
| 12:44.51 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 12:45.43 | _x86_ | tzanger: i guess i'm unclear to what HPEC is |
| 12:45.59 | *** join/#asterisk duckz (n=duckz@81.180.83.75) |
| 12:46.02 | *** join/#asterisk DataCompBoy (n=datacomp@213.187.250.34) |
| 12:46.07 | tzanger | HPEC is (IIRC) a g.168-compliant software echo can module for Asterisk |
| 12:46.08 | DataCompBoy | Hi all! :) |
| 12:47.12 | coppice | I find most cans pretty echoey |
| 12:47.43 | DataCompBoy | Does anybody know issue, when calls from some (one!) operator via ZAP dies after 16 seconds? ISDN provider see no problem on his side, the only see that problem with only one block of phones... |
| 12:48.26 | DataCompBoy | Dies not all calls... So, often he calls onece, was interrupted, redial and talk okay |
| 12:48.45 | tzanger | coppice: you need to open both ends |
| 12:53.49 | coppice | chilli does that for most people |
| 12:56.00 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
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| 13:00.28 | *** join/#asterisk smgua (n=smelgar@168.234.226.66) |
| 13:01.26 | smgua | any issues to connsider before upgrading from 1.4.5 -> 1.4.12? |
| 13:01.48 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:01.48 | *** join/#asterisk _ShrikE (n=ShrikE@74.185.215.60) |
| 13:02.41 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 13:03.33 | tru_`z24 | I heard 1.4.xxx wasn't ready for production? |
| 13:03.59 | tru_`z24 | Does this myth need to be busted? |
| 13:05.07 | coppice | Who ya gonna call? |
| 13:05.08 | coppice | Myth busters..... |
| 13:05.30 | deeperror | would like to see that episode |
| 13:05.37 | tru_`z24 | lol |
| 13:06.05 | deeperror | nothing could beat blowing up a cement mixer though |
| 13:06.21 | coppice | will it blend? |
| 13:06.23 | DataCompBoy | tru_`z24: i'm use in production, fine |
| 13:06.56 | smgua | wich version? |
| 13:07.37 | *** join/#asterisk mltlnx (n=mltlnx@pool-96-224-1-190.nycmny.east.verizon.net) |
| 13:07.37 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-51-160.dsl.tul2ok.sbcglobal.net) |
| 13:07.37 | DataCompBoy | tru_`z24: just you need some recompile and change in additional modules, in compare with 1.2 |
| 13:07.51 | tru_`z24 | Are channel banks outdated? I was going to order some, but I had a guy tell me that those are "old school" |
| 13:08.02 | bkw_ | no |
| 13:08.05 | bkw_ | they are not outdated |
| 13:08.12 | tru_`z24 | He says this because of how dialogic has a MSI board that comes with their cards... |
| 13:08.20 | tru_`z24 | therefor saving "space" |
| 13:08.27 | aiksa[LV] | you mean rhino channel banks? |
| 13:08.39 | coppice | dialogic's MSI boards suck |
| 13:08.49 | bkw_ | yo coppice |
| 13:09.03 | tru_`z24 | Well, is there a comparative technology with digium ? |
| 13:09.07 | DataCompBoy | tru_`z24: dunno what you mean, i'm use one zaptel TE card with 2*T1 :) |
| 13:09.28 | tru_`z24 | What are you doing to break the digital lines into 24 analog ones? |
| 13:09.30 | tru_`z24 | A channel bank? |
| 13:09.44 | tru_`z24 | or are you voip only ? :-) |
| 13:10.23 | *** part/#asterisk smgua (n=smelgar@168.234.226.66) |
| 13:10.44 | deeperror | we use rhino banks here |
| 13:10.57 | DataCompBoy | tru_`z24: i have 2*T1 and VoIP, when ground lines ends :) |
| 13:11.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:11.14 | _x86_ | ugh, what's the username on polycom phones? isn't it just "Polycom" ? |
| 13:11.15 | DataCompBoy | tru_`z24: 2*T1 via zaptel card. |
| 13:11.30 | _x86_ | nvm, had the password wrong ;) |
| 13:11.39 | tru_`z24 | DataCompBoy: yeah, but i'm more curious if anyone is using analog phones with anything other than a channel bank |
| 13:11.58 | aiksa[LV] | yes - atas |
| 13:12.30 | _x86_ | is it possible to set the TFTP server from a polycom's web interface? |
| 13:12.31 | aiksa[LV] | if a company wants to preserve their phones in new location where there is only etehrnet available |
| 13:12.46 | _x86_ | aiksa[LV]: sounds expensive |
| 13:12.53 | deeperror | depends on the setup |
| 13:13.04 | _x86_ | well, cheaper than SIP phones, I guess |
| 13:13.27 | coppice | preserve? I've never heard of anyone pickling phones before |
| 13:13.34 | bkw_ | haha |
| 13:13.44 | _x86_ | rofflecopter |
| 13:13.45 | aiksa[LV] | coppice: I have :) |
| 13:13.51 | _x86_ | where is TK when i need him? :P |
| 13:13.59 | coppice | or phone jerky, maybe |
| 13:14.04 | _x86_ | Qwell: you around? |
| 13:14.27 | bkw_ | coppice, why do people confuse G.722 and G.722.1/G722.2? |
| 13:14.45 | aiksa[LV] | they told that they are used to the actual feeling of the phones ... (I didnt comment on that one - it aint a good practice to laugh about a customer during sales) |
| 13:14.58 | _x86_ | bkw_: same reason people confuse g.723 and g.723.1 ;) |
| 13:14.59 | coppice | bkw_: why do people confuse faxing over VoIP with T.38? |
| 13:15.47 | jcanfield | Anyone know the polycom <dialplan> settings(s) to prevent the phone from asking for more digits? |
| 13:15.50 | aiksa[LV] | coppice: as a guru of T.38, please enlight me: is T.38 sip specificf? |
| 13:15.54 | _x86_ | why do people confuse "having AOL installed" with "having the Internet version 9.0"? |
| 13:16.16 | bkw_ | haha |
| 13:16.20 | coppice | T.38 was originally H.323 specific, but these days its specified for use with SIP and MGCP |
| 13:16.21 | bkw_ | good point guys |
| 13:16.29 | _x86_ | jcanfield: if you find out, let me know ;) |
| 13:16.29 | Katty | moo. |
| 13:16.31 | bkw_ | MGCP ewwww |
| 13:16.34 | aiksa[LV] | coppice: thanks. |
| 13:17.00 | jcanfield | _x86_: Will do...it's one of my goals today. |
| 13:17.19 | coppice | MGCP is a really brain dead protocol, but the architecture it was designed for is very sane |
| 13:17.27 | _x86_ | ugh, i want to tell this Polycom phone it's new TFTP server, but my choices are do it from the web interface, or walk an employee through it (giving them the password) |
| 13:17.48 | Qwell | coppice: it looks like it was designed by brain dead people too :p |
| 13:17.50 | aiksa[LV] | coppice: is there a t.38 guide for dummies out there |
| 13:18.01 | _x86_ | heya Qwell |
| 13:18.06 | aiksa[LV] | i mean less technical junk, but more overall concepts? |
| 13:18.12 | michael-i | i have a asterisk-users etiquette question. I've seen others announce Asterisk related software releases on -users (and have done it once myself). Is this frowned upon? Is another list more appropriate? |
| 13:18.12 | jcanfield | bkw_: Aren't you a fellow okie? |
| 13:18.26 | coppice | aiksa[LV]: like some kinda specially trained labrador, you mean? |
| 13:18.50 | coppice | he's more of a dokie than an okie |
| 13:19.13 | aiksa[LV] | :)) in broad terms describing what is going on in T.38 session |
| 13:19.31 | aiksa[LV] | rather than referencing individual packets, checksuums etc. |
| 13:20.12 | lirakis | _x86_: you can do it via DHCP "option 66" |
| 13:20.25 | coppice | dunno really. I can't remember seeing one. "T.38 sucking dummies" |
| 13:20.28 | bkw_ | jcanfield, yes |
| 13:20.30 | _x86_ | lirakis: even if the phone is setup static and there is no DHCP server in that office? :P |
| 13:20.35 | aiksa[LV] | :) |
| 13:21.03 | aiksa[LV] | thats sad. Wanted to start reading by more general overall process description |
| 13:21.12 | lirakis | _x86_: setting up a dhcp server will probably be easier than telling X employees how to do it .. then going and fixing it for the 50%+ that did it wrong |
| 13:21.32 | _x86_ | lirakis: we're only talking about a single remote phone |
| 13:21.37 | jcanfield | bkw_: I thought so...haven't heard your name in quite a while. I remember your dealing with the feds back when i worked for a tax software company. Guess you are quite the asterisk guru now. |
| 13:21.53 | bkw_ | let me chat with you in private :) |
| 13:21.59 | bkw_ | check your pm |
| 13:22.01 | lirakis | _x86_: oh.. it didnt sound like that... well in that case.. (shrug) .. it doesnt sound too painful at all |
| 13:22.22 | _x86_ | lirakis: but the user (having the password) may fudge some stuff up later on... |
| 13:22.27 | aiksa[LV] | but the idea of T.38 is? If sending fax over t.38 from ata to PBX (which would terminate it on PSTN) what happens exactly? |
| 13:22.36 | _x86_ | AFAIK, there is no way to provision the admin password to the phone, eh? |
| 13:22.44 | gremzoid | *sigh* such crap documentation |
| 13:22.53 | Katty | left side of nose works, right side no workith :< |
| 13:23.00 | aiksa[LV] | does PBX receive the whole fax (emulating) the receiving party and then transmits it over the PSTN |
| 13:23.30 | aiksa[LV] | or t.38 allows both actual devices to "see" each other during comunication? |
| 13:23.37 | coppice | the modem signals are demodulated, the digital data sent across in packets, and the result remodulated at the outgoing end (unless, of course, one end is actually terminating the T.38 itself) |
| 13:23.59 | lirakis | _x86_: yeah |
| 13:24.33 | aiksa[LV] | coppice: so as far as faxing equipment at both endpoints is regarded they see each other over "transaprent" channel |
| 13:25.09 | aiksa[LV] | and the magic lies in demodulating/modulating signal during the transmission over IP network. |
| 13:25.09 | Qwell | coppice: can a T.38 session be saved and "played back"? |
| 13:25.16 | coppice | it tried to be as transparent as possible. it has to be. the timing of the fax protocol (T.30) doesn't allow much timing latitude |
| 13:25.29 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 13:25.44 | coppice | Qwell: no more or less than any other FAX session can |
| 13:26.35 | aiksa[LV] | "it tried to be as transparent" - this means T.38 is as reliable as the ATA capability of demodulation and PSTN terminators :)) ability to modulate? |
| 13:27.11 | coppice | well, any communication is only as reliable as the elements in the chain |
| 13:27.21 | aiksa[LV] | well, yes. |
| 13:27.30 | aiksa[LV] | i just wanted to make sure that i understand. ;) |
| 13:28.59 | aiksa[LV] | so first Fax Machine from visual image modulates the sound signal and within 3 feet from that another device tries to demodulate that? |
| 13:30.24 | coppice | if the fax is plugged straight into an ATA, then yes. there are very few native T.38 machines that can do things directly |
| 13:30.56 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:30.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:32.14 | aiksa[LV] | coppice: sounds like native t.38 fax machine should be a more obvious choice. (there is saying in russian - less figures/parties makes the game easier) |
| 13:33.35 | coppice | I haven't seen a native T.37 or T.38 machine. there are internet capable fax machines, but they tend to send attachments to e-mails in a way that is not consistent with T.37 |
| 13:34.17 | aiksa[LV] | Oki Fax 5950 T.38 ? |
| 13:36.09 | iCEBrkr | Phreakz! |
| 13:36.23 | coppice | there is a T.37 option card for some of the OKIs, but I think someone told me they lie, and it doesn't really follow T.37 |
| 13:36.25 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 13:37.02 | coppice | oh, doing a google it looks like that have a separate T.38 option as well, now |
| 13:37.04 | devel | greetings all. anybody here who uses realtime in the dialplan? |
| 13:37.40 | aiksa[LV] | coppice: looking for a price of that baby |
| 13:38.53 | coppice | $3290 + the options |
| 13:39.14 | aiksa[LV] | $2173. but it doesnt make it any better :( |
| 13:39.31 | coppice | having separate option kits for the two protocols is pretty sucky |
| 13:40.03 | coppice | $3290 was for the 5980. the 5950 is obsolete |
| 13:40.23 | coppice | still two separate option kits, though |
| 13:40.42 | aiksa[LV] | the brice is ouch ouch ouch though |
| 13:41.19 | aiksa[LV] | heres another: SAGEM IP PhoneFax 49A (SIP) |
| 13:41.24 | coppice | a $50 thermal paper fax, and a granstream ATA will do just fine :-) |
| 13:41.29 | aiksa[LV] | EUR 300 |
| 13:41.51 | aiksa[LV] | gs can t.38? |
| 13:42.59 | coppice | yes, with the right firmware version |
| 13:43.16 | aiksa[LV] | http://www.sagem-communications.com/index.php?id=27&L=0 |
| 13:43.52 | coppice | oh, the sagem is a cordless VoIP phone with a fax machine built in. interesting |
| 13:44.37 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 13:44.40 | aiksa[LV] | and with extremely usefull "SUDOKU FUNCTION" for those days when fax just dont work! |
| 13:44.52 | *** join/#asterisk elriah (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 13:44.56 | coppice | its one of those nasty ink film machines, though |
| 13:46.05 | coppice | 229,99 EUR - some VoIP phones cost that much |
| 13:47.07 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 13:49.08 | deeperror | Would changing the dtmfmode setting in sip.conf change behavior of VLDTMF being played? |
| 13:49.11 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 13:49.25 | aiksa[LV] | coppice: here is a nice niche for new fax machines the i suppose |
| 13:50.21 | rantsh | hello |
| 13:50.23 | rantsh | anyone can give me a hint with some transcoding crap yet again? |
| 13:51.11 | rantsh | I don't have control over my gateway, all I know is it only allows g729, alaw and ulaw |
| 13:51.25 | rantsh | my * box is using the b2bua perl script |
| 13:52.16 | rantsh | and my sip.conf is set (as I learned from this wonderful channel) to allow ilbc and disallow g729 |
| 13:52.53 | bkw_ | why? |
| 13:52.57 | rantsh | but for some reason the call won't gp through and it won't say much on the console, |
| 13:52.58 | bkw_ | doe syour gateway not do ilbc? |
| 13:53.03 | bkw_ | sip debug |
| 13:53.05 | bkw_ | check the SDP |
| 13:53.13 | *** join/#asterisk Buglouse (i=FreeNode@my.body.is.so-relax.com) |
| 13:53.20 | bkw_ | what gateway you have? |
| 13:53.33 | bkw_ | chances are it doesn't do anything but g729 |
| 13:53.52 | rantsh | my gateway is a 3rd party equipment, I have no control over it, nor I know what it is, it's a black box for me |
| 13:54.13 | rantsh | quintum |
| 13:54.30 | bkw_ | ok check the SDP on the invite to the gateway |
| 13:54.36 | bkw_ | and see what it says it supports that way |
| 13:55.03 | rantsh | I might just be killing myself in some n00bie error, but I can't see where it is |
| 13:55.17 | bkw_ | asterisk -r |
| 13:55.18 | bkw_ | sip debug |
| 13:55.20 | bkw_ | make a call |
| 13:55.21 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 13:55.27 | bkw_ | put the sip messages on pastebin |
| 13:55.32 | rantsh | 'k, give me a sec |
| 13:55.33 | rantsh | I will |
| 13:55.46 | agx | Hi, with mISDN can i use master_clock on every TE port if i've a digium 4 bri ? |
| 13:56.33 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.157.180) |
| 13:56.49 | rantsh | http://pastebin.com/m7784b19a |
| 13:57.32 | aiksa[LV] | agx: i somehow remember that master clock could be used on only 1 port |
| 13:57.33 | bkw_ | well one don't just allow ilbc |
| 13:57.46 | bkw_ | rantsh, try adding ulaw and alaw to that .. and g729 |
| 13:58.00 | bkw_ | if its quintum I know it doesn't support ilbc |
| 13:58.36 | rantsh | yup, but I need my asterisk box to transcode the ilbc to g729 |
| 13:58.43 | Katty | anonymouz666: :> |
| 13:58.51 | anonymouz666 | Katty !!!!!! |
| 13:58.56 | Katty | anonymouz666: herro. :> |
| 13:58.58 | bkw_ | ok you need allow=ilbc on the one facing the phoen and allow=g729 on the one facing the gateway |
| 13:59.05 | bkw_ | so you'll need to setup a peer entry in sip.conf |
| 13:59.07 | bkw_ | for the gateway |
| 13:59.14 | bkw_ | then dial sip/number@peername |
| 13:59.24 | Katty | ^_^ |
| 13:59.38 | bkw_ | Katty, hey girl |
| 13:59.49 | Katty | bkw_: hello!! |
| 14:00.07 | anonymouz666 | Katty: how goes? |
| 14:00.18 | Katty | anonymouz666: umm, that's a very complicated question :< |
| 14:00.26 | Katty | anonymouz666: this morning i'm doing well :> |
| 14:01.16 | Zeeek | Hi {{Katty}} |
| 14:01.20 | anonymouz666 | :) |
| 14:01.32 | Katty | hi Zeeek :> |
| 14:01.36 | rantsh | thanks, so I'll need 2 contexts in sip.conf |
| 14:02.22 | rantsh | bkw_: I'll try that |
| 14:02.29 | rantsh | thanks for the help |
| 14:02.45 | bkw_ | np |
| 14:03.14 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 14:03.35 | Zeeek | NEXT! |
| 14:03.46 | Katty | ^- (tm) |
| 14:03.50 | Zeeek | yes |
| 14:03.58 | Katty | :> (tm) |
| 14:04.02 | Zeeek | it was said tounge in... |
| 14:04.21 | Zeeek | tongue in chic |
| 14:04.28 | Katty | ahem. |
| 14:04.42 | Katty | also! jinx++ |
| 14:05.07 | Katty | i got a trixbox, rhino server in the other day. |
| 14:05.13 | Katty | it's purrty red. |
| 14:05.43 | Katty | ceros thingy. |
| 14:06.04 | coppice | wouldn't rhinos usually be served by other rhinos |
| 14:07.06 | Zeeek | prejudice in the workplace? |
| 14:10.26 | Katty | no, postjudice. |
| 14:11.00 | rantsh | bwk_: btw, how did you know it didn't accept the call because of codec issues? |
| 14:11.02 | coppice | ooh, look. a see through throat |
| 14:11.04 | Maliuta | post-judice-purance? that's what the mortician does if you have the death penalty :) |
| 14:11.15 | Maliuta | prudance even |
| 14:11.15 | rantsh | bwk_ : I mean on the debug output? |
| 14:11.18 | anthm | about time |
| 14:11.35 | coppice | in what? |
| 14:11.37 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 14:11.42 | Qwell | batter, hopefully |
| 14:11.45 | Katty | pure sunshine, i hope ^_- |
| 14:11.47 | bkw_ | rantsh, look at the full sip debug of the call going to the gatway |
| 14:11.51 | Maliuta | could be chocolate |
| 14:11.52 | Katty | no additives please! |
| 14:12.10 | Maliuta | Katty: not even sugar? sugar. |
| 14:12.12 | bkw_ | rantsh, this is one of those hard ones to solve without being on the machine to see it when it happens |
| 14:12.22 | Katty | Maliuta: pfft, sugar. |
| 14:12.27 | Katty | Maliuta: splenda :P |
| 14:12.35 | Qwell | Maliuta: high fructose corn syrup? |
| 14:12.40 | Katty | Maliuta: nobody cooks with sugar anymore. |
| 14:12.42 | Maliuta | Katty: nah, I need that calories |
| 14:12.48 | Katty | oh, i see. |
| 14:12.55 | rantsh | bkw, I did... I see the decline but I can't see the reason why it didn't |
| 14:13.09 | bkw_ | rantsh, that debug you posted didn't have the outbound invite in it |
| 14:13.14 | rantsh | bkw_, I understand... althought I very much appreciate your help |
| 14:13.14 | bkw_ | can you capture it again and pastebin it? |
| 14:13.54 | Zeeek | ~seen russellb |
| 14:13.56 | jbot | russellb is currently on #asterisk-dev (19h 30m 47s) #asterisk (19h 30m 47s) #asterisk-bugs (19h 30m 47s). Has said a total of 57 messages. Is idling for 7m 21s, last said: 'M10406'. |
| 14:13.59 | rantsh | bkw_ sure give me a sec |
| 14:14.33 | russellb | Zeeek: pong |
| 14:14.50 | Katty | beer pong? |
| 14:14.56 | Katty | i mean, malt beverage pong. |
| 14:15.11 | rantsh | I know why it wasn't there, most the "magic" goes through an agi b2bua script my boss has |
| 14:15.38 | Zeeek | pink? |
| 14:15.43 | Maliuta | beer only pongs when it's been left out overnight |
| 14:15.52 | Katty | i can't stand beer. |
| 14:16.04 | Katty | especially amberbock. |
| 14:16.07 | anthm | did you try hacker pschorr ? |
| 14:16.07 | Qwell | russellb: I found a very humorous orange sticker in the bathroom yesterday |
| 14:16.07 | Maliuta | and I think he meant ICMP: Echo-rely |
| 14:16.19 | Qwell | they wrote on it "Clicking noise" |
| 14:16.21 | anthm | grolsch ? |
| 14:16.35 | russellb | Qwell: ha ... as if that noise is easy to ignore |
| 14:16.39 | Qwell | yeah |
| 14:16.53 | Qwell | I think somebody is trapped in the wall, and sending morse code SOS |
| 14:17.35 | Katty | i don't think i want to know... |
| 14:17.54 | Maliuta | anthm: it's not Kriek or Duval |
| 14:17.57 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:18.41 | russellb | Qwell: lol |
| 14:19.14 | anthm | beer is like linux they all work but not as potent as the real stuff |
| 14:19.22 | Qwell | I'm gonna get an orange sticker, and stick it on of of the other ones... |
| 14:19.27 | Katty | wow. |
| 14:19.28 | Qwell | and write "orange stickers everywhere" |
| 14:19.35 | russellb | nice |
| 14:19.36 | Katty | anthm: way to be the first person i've ever seen to compare linux and beer. |
| 14:19.39 | Maliuta | anthm: and the real stuff is? |
| 14:19.42 | Katty | anthm: /clapclap |
| 14:19.43 | Zeeek | Orange is now my favorite color |
| 14:19.51 | Zeeek | since I went to IP Convergence |
| 14:20.02 | anthm | dunno whiskey rum vodka |
| 14:20.14 | Zeeek | Orange was giving away free coffee and they had the drop dead beautiful hostesses servibng it |
| 14:20.15 | Qwell | Katty: I've seen purple |
| 14:20.26 | Katty | anthm: well then what's the rum of the computer world then if linux is beer? |
| 14:20.33 | Zeeek | and they had on little tight t-shirts that said "Hello" |
| 14:20.37 | Zeeek | in orange |
| 14:20.41 | rantsh | bwk, my new sip file is here http://pastebin.com/d7787b17 |
| 14:20.42 | Maliuta | Katty: sure, but it's not as cool as black, with black buttons and black lights |
| 14:20.52 | harryr | Zeeek: I just bought a bright orange rackmount server |
| 14:20.52 | Katty | who said. |
| 14:21.08 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:21.10 | Katty | i think purple in the dark with lots of pink fans and things looks wonderful :P |
| 14:21.16 | Zeeek | nice, orange server |
| 14:21.18 | bkw_ | rantsh, I need the full sip debug of a call going out the gateway |
| 14:21.22 | Katty | maybe a cute little dragon imoogi etched in the side :> |
| 14:21.22 | bkw_ | can you paste bin that for me |
| 14:21.30 | rantsh | bkw, it still doesn't work, I'm pretty sure is because I have a newbie mistake somewhere |
| 14:21.34 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:21.36 | Maliuta | Katty: isn't purple the colour of sexual frustration? ;P |
| 14:21.44 | Katty | Maliuta: ^_- |
| 14:21.49 | bkw_ | rantsh, get me the sip invite that hits the gateway and its response |
| 14:21.51 | Katty | Maliuta: i have no idea. |
| 14:21.52 | rantsh | bwk_, ooops sorry I forgot, give me a sec |
| 14:22.13 | grandpapadot | %s/rabble/asterisk/g |
| 14:22.14 | Maliuta | Katty: if you say so :) |
| 14:22.22 | bkw_ | rantsh, let me give you a hint |
| 14:22.23 | *** join/#asterisk Aeudian (n=somewher@ip67-94-157-98.z157-94-67.customer.algx.net) |
| 14:22.28 | bkw_ | [GATEWAY_IP] <-- do not use IP address for peer names |
| 14:22.30 | anthm | Katty, you should have tried that asssie rum nix had at my house that one time you were there |
| 14:22.34 | bkw_ | use names NOT IP's |
| 14:22.35 | Katty | Maliuta: what are you getting at? :P |
| 14:22.41 | anthm | i still have the rest |
| 14:22.50 | bkw_ | doesn't need to be a valid dns name either.. just something to specify it |
| 14:22.51 | Katty | anthm: you still have the rest?! |
| 14:22.56 | Katty | anthm: that stuff is OLD! |
| 14:22.59 | *** join/#asterisk karlh626 (n=kharris@IP-66.249.227.5.indigital.net) |
| 14:23.05 | Katty | anthm: damn, i wasn't even 21 back then >.< |
| 14:23.13 | Katty | anthm: i turn 23 on sunday |
| 14:23.17 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:23.17 | Aeudian | Does anyone have the link or mirror for hudlite 1.4.5? It says its on trixbox's site but i searched all over it and google and can only find older version. I am trying to intergrate hudlite with a standalone asterisk system |
| 14:23.18 | rantsh | bkw, ok |
| 14:23.26 | Katty | anthm: 4 YEAR OLD RUM MIX! |
| 14:23.45 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 14:23.50 | devel | greetings all. anybody here who uses realtime in the dialplan? |
| 14:24.01 | Maliuta | Katty: at the moment? not much. |
| 14:24.19 | Maliuta | anthm: did you mean aussie rum? like Bundaberg? |
| 14:24.27 | anthm | ya |
| 14:24.33 | anthm | exactly |
| 14:24.38 | Maliuta | Katty: pfft 23 is young |
| 14:24.42 | Maliuta | anthm: ewww |
| 14:24.52 | anthm | Katty, happy bday |
| 14:24.52 | Maliuta | anthm: have you _been_ to Bundy? |
| 14:25.05 | anthm | next time you come to chicago we'll get you properly smashed |
| 14:25.13 | Maliuta | it tastes like crap and it comes from a hole |
| 14:25.14 | anthm | first week of aug |
| 14:25.16 | bkw_ | hehe |
| 14:25.19 | Katty | anthm: thanks :> |
| 14:25.25 | Katty | Maliuta: yes yes it is... |
| 14:25.35 | rantsh | bkw_, so it should be a context and then modify the agi script to dial SIP/###@sip_context ? ? ? |
| 14:25.41 | Katty | Maliuta: but i was /so/ young when i met anthm, they had to take me to a strip joint that didn't serve alchahol. >.< |
| 14:25.53 | bkw_ | rantsh, yes |
| 14:26.00 | bkw_ | but they aren't called context's in sip .conf |
| 14:26.05 | bkw_ | they are users or peer entries |
| 14:26.09 | Maliuta | Katty: in the US that's only like 21 |
| 14:26.23 | Maliuta | Katty: here in .au you can drink legally at 18 |
| 14:26.33 | Katty | anthm: wait. you didn't go to the stripper place with us, did you? |
| 14:26.58 | anthm | no |
| 14:27.03 | *** join/#asterisk mltlnx (n=mltlnx@pool-96-224-1-190.nycmny.east.verizon.net) |
| 14:27.04 | Katty | what a snob. :> |
| 14:27.04 | anthm | my wife was not too keen on the idea |
| 14:27.14 | Katty | oh, that's right. she went with you. |
| 14:27.17 | Katty | i remember now. |
| 14:27.26 | anthm | nobody got her drunk enuf to pass out so i could |
| 14:27.33 | Katty | heh. |
| 14:27.38 | bkw_ | darn /me adds that to his task list for next year |
| 14:27.47 | coppice | anthm: you were too lazy to? |
| 14:27.49 | Katty | leave her at home next year :P |
| 14:28.16 | Maliuta | coppice: well he could always be under the thumb ;) |
| 14:28.55 | anthm | i try |
| 14:29.28 | anthm | coppice, it was also 5am |
| 14:29.41 | anthm | and i am supposed to be running the place |
| 14:29.52 | anthm | hic |
| 14:30.12 | anthm | so we compromised and put half barrels in the classroom |
| 14:30.19 | bkw_ | that was great |
| 14:30.33 | bkw_ | I don't drink beer but damn rolling those out at noon the next day was fun |
| 14:30.38 | anthm | paid for by digium, how sweet of them to make such a nice gesture |
| 14:31.00 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.227) |
| 14:31.03 | Maliuta | they can make that kind of gesture in my direction if they like :) |
| 14:31.16 | tmccrary | Can anyone recommend a good IAX call termination provider? |
| 14:31.40 | Zeeek | voicepulse connect |
| 14:32.03 | bkw_ | tmccrary, honestly IAX on a ITSP isn't very good... if you try to use it in that env it'll fall over |
| 14:32.11 | Maliuta | I swear my cisco IP phone just blinked a light at me |
| 14:32.22 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 14:32.53 | tmccrary | bkw_: So are you recommending SIP then? I've used a few |
| 14:32.58 | bkw_ | tmccrary, yes |
| 14:33.04 | tmccrary | Regardless, does anyone have a favorite SIP provider? |
| 14:33.13 | bkw_ | I hightly recommend SIP.. we do sip at Asterlink |
| 14:33.31 | bkw_ | I had to really discontinue IAX due to quality of audio... |
| 14:33.48 | coppice | if you are seriously dehydrated I suggesting doing more than SIP |
| 14:34.01 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 14:34.04 | tmccrary | bkw_: Ah, I have never used IAX before, only SIP. I was just interested in seeing what the difference was |
| 14:34.08 | bkw_ | mixing media and signalling is ok for small scale is ok.. but when you start to do it on larger scales it starts to fall over |
| 14:34.12 | *** part/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 14:34.16 | pepse | iax works great for me |
| 14:34.27 | Maliuta | tmccrary: depends on where you are and what you want. I use pennytel for my SIP DID |
| 14:34.30 | bkw_ | pepse, small scale its fine.. but if you do it on high scale it doesn't work well |
| 14:35.03 | bkw_ | you have 150 calls from 150 differnt IP's all hitting the same port |
| 14:35.04 | tmccrary | I'm in Michigan, US |
| 14:35.20 | deeperror | MI-2 |
| 14:35.20 | bkw_ | the iax stack has to sort the signalling from the media and do it quickly |
| 14:35.25 | pepse | ah, possibly. but anywhere from 1 to 20 it's been fine |
| 14:35.37 | bkw_ | pepse, yah i'm talking higher scale than that |
| 14:35.39 | Maliuta | tmccrary: that's along way from me, but I think pennytel has a presence in the states |
| 14:35.51 | tmccrary | Maliuta: I think pennytel would be a little expensive for me, but thanks for the link and reminding me to specify where I'm at :) |
| 14:36.21 | deeperror | tmccrary: voicepulse? |
| 14:36.48 | tmccrary | deeperror: Thanks, I will check them out |
| 14:37.00 | pepse | any of you guys have a favorite Windows Mobile 6 or Palm voip client? |
| 14:37.07 | pepse | err softphone |
| 14:37.15 | Maliuta | tmccrary: well at $0.08c untimed to any fixed line in .au, .ca, .uk and the US I'm not complaining |
| 14:37.51 | Maliuta | tmccrary: and that's $0.08AU (not that the US dollar is worth much these days) |
| 14:37.59 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 14:38.28 | tmccrary | voicepulse doesn't look bad, the online reviews seem okay |
| 14:38.55 | Zeeek | I've used VP COnnect for three years, works well, good price (outbound) |
| 14:39.20 | deeperror | i've got a callcenter running some voip testing over them right now |
| 14:39.21 | coppice | what does 0.08 untimed mean? :-\ |
| 14:39.28 | pepse | tmccrary: Where are you located? |
| 14:39.38 | pepse | coppice: the call is 8 cents no matter how long you talk |
| 14:39.41 | tmccrary | pepse: Michigan, US |
| 14:39.51 | deeperror | troy mi here |
| 14:39.53 | coppice | weird |
| 14:40.00 | pepse | tmccrary: Have you ever checked out Voipjet, Broadvoice, les.net, or jnctn.net? |
| 14:40.13 | tmccrary | deeperror: Same here ;) |
| 14:40.29 | tmccrary | pepse: I've used Broadvoice and they have strange problems with their service |
| 14:40.36 | pepse | coppice: It's not so weird. Most countries that charge per min have an evening rate where after 8pm or 9pm or midnight or whatever, all calls (no matter how long) are one flat rate |
| 14:40.40 | Maliuta | it's late-ish |
| 14:40.48 | deeperror | tmccrary: we should get the users group going haha |
| 14:41.02 | pepse | tmcrary: yeah, but they are cheap and have unlimited. voipjet and jnctn are pretty good tho. |
| 14:41.09 | coppice | pepse: well, I used to make 30 day calls, back when I used modems |
| 14:41.47 | pepse | yeah, we're spoiled in the US |
| 14:42.00 | coppice | I'm not in the US |
| 14:42.14 | bkw_ | voipjet is just a reseller of other people |
| 14:42.24 | pepse | ah, sorry, got confused :) |
| 14:42.35 | pepse | bkw: Really? well they resell some good quality voip :) |
| 14:42.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:43.07 | pepse | i like how they give you 25 cents for free when you sign up |
| 14:43.09 | bkw_ | pepse, yah never know anything about his quality |
| 14:43.11 | pepse | i still haven't used it |
| 14:43.22 | Zeeek | Jnctn is good, too |
| 14:43.24 | pepse | err still haven't used it up |
| 14:43.38 | pepse | I use jnctn for my 800 number. Also very good. |
| 14:43.50 | pepse | just expensive for outgoing calls |
| 14:43.55 | Zeeek | If you need mission critical stuff, go to Asterlink |
| 14:43.57 | tmccrary | deeperror: is there an asterisk user group around? |
| 14:44.13 | deeperror | not according to voip-info |
| 14:44.47 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:45.14 | deeperror | there is a local linux users group but it's more down in the city |
| 14:46.06 | pepse | when the lights.. go down... in the citeh... |
| 14:46.19 | deeperror | blah |
| 14:50.48 | Dan0maN_Work | heh |
| 14:54.10 | *** join/#asterisk Aeudian (n=somewher@ip67-94-157-98.z157-94-67.customer.algx.net) |
| 14:54.23 | *** part/#asterisk Aeudian (n=somewher@ip67-94-157-98.z157-94-67.customer.algx.net) |
| 14:55.36 | JerJer | is there any way to send a 'better' response than 'Bad Event' on the standard notify nat pings ? |
| 14:57.45 | *** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com) |
| 15:09.20 | iCEBrkr | Asterlink? |
| 15:10.10 | iCEBrkr | They don't even have a 'Forgot password' link and they're trying to bill a CC I no longer have and I keep getting invoices instead of just cancelling the account. |
| 15:12.50 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 15:13.40 | bkw_ | iCEBrkr, lets take care of you |
| 15:14.17 | bkw_ | iCEBrkr, check our PM |
| 15:14.31 | bkw_ | er your |
| 15:14.41 | *** join/#asterisk shido6 (n=shido6@74-130-59-184.dhcp.insightbb.com) |
| 15:15.53 | bkw_ | iCEBrkr, you there? |
| 15:17.23 | iCEBrkr | I'm here |
| 15:17.36 | shido6 | where can I buy a 4gb iphone? |
| 15:17.39 | iCEBrkr | bkw_: damn brian, you're still slumming? |
| 15:18.40 | bkw_ | slumming? haha |
| 15:18.50 | bkw_ | shido6, try the apple clearance page on apple.com |
| 15:18.51 | Katty | moo. |
| 15:19.34 | iCEBrkr | bkw_: Yeah, didn't you take your ball and go home? LOL It's ok, I know you still love #asterisk :) |
| 15:20.36 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 15:20.46 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 15:21.03 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-c34616a71c17211a) |
| 15:21.11 | rantsh | bkw_, can I bother you on |
| 15:21.18 | pigpen | Hi all: |
| 15:21.24 | pigpen | I am getting: chan_iax2.c:6521 socket_read: Out of idle IAX2 threads for I/O, pausing! |
| 15:21.25 | rantsh | bkw_ once more, sorry... |
| 15:21.33 | russellb | pigpen: what version? |
| 15:21.38 | pigpen | with my iaxthreadcount=500 and iaxmaxthreadcount=2000 |
| 15:21.38 | russellb | pigpen: if not 1.4.12, try that. |
| 15:21.40 | iCEBrkr | rantsh: He'll bother you back. |
| 15:21.52 | pigpen | 1.4.11 |
| 15:21.59 | russellb | pigpen: also, build with DEBUG_THREADS enabled, and when it happens, get the output of the CLI command "core show locks" |
| 15:22.06 | rantsh | ¿?¿??¿¿?¿?¿? |
| 15:22.27 | russellb | rantsh: ... ? |
| 15:22.32 | pigpen | k. were there some fixes for this in 1.4.12? |
| 15:22.37 | Zeeek | anyone here in charge of http://www.voipuser.org |
| 15:22.39 | russellb | pigpen: a lot of them .. |
| 15:22.42 | pigpen | k. |
| 15:22.51 | pigpen | tks again...and again...and again. |
| 15:23.11 | russellb | np |
| 15:23.20 | twisted | bkw_...? |
| 15:23.24 | rantsh | well he had helped me a lot this morning with some transcoding issues on my b2bua asterisk |
| 15:24.45 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
| 15:24.57 | bkw_ | twisted, yes? |
| 15:25.42 | twisted | hi |
| 15:25.45 | russellb | ha |
| 15:26.16 | Zeeek | wow, the wayback machine is on |
| 15:26.29 | russellb | twisted: don't have _too_ much fun ... |
| 15:26.39 | twisted | oh, i've made her say the things she wouldn't say |
| 15:26.48 | iCEBrkr | twisted: pervert :) |
| 15:27.02 | russellb | someone told allison at astricon something like that |
| 15:27.04 | rantsh | I'm trying to do transcoding a b2bua asterisk box, I got to the part where I can make the asterisk transcode |
| 15:27.15 | twisted | well, the only thing *I* know of she wouldn't say is the c-bomb |
| 15:27.15 | russellb | they told her that they made her say "oooh" and would listen to it on a loop |
| 15:27.17 | rantsh | but there's no audio on the call |
| 15:27.19 | rantsh | http://pastebin.com/d6bd7304a |
| 15:27.29 | russellb | creeped her out |
| 15:27.34 | *** join/#asterisk Comradin (n=marcus@e177148127.adsl.alicedsl.de) |
| 15:27.36 | twisted | lol |
| 15:27.39 | iCEBrkr | I'm sure it did |
| 15:27.53 | twisted | that's funny, considering the conversations we've had with her |
| 15:28.06 | russellb | heh |
| 15:28.10 | *** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com) |
| 15:28.23 | iCEBrkr | bkw has a nice collection of odd things he's made her say. :) |
| 15:28.27 | russellb | twisted: welcome back to #asterisk, btw :) |
| 15:28.32 | twisted | yea |
| 15:28.44 | twisted | russellb : i'm slumming :P |
| 15:28.50 | russellb | heh, sweet |
| 15:29.21 | twisted | btw |
| 15:29.25 | twisted | how's the new building? |
| 15:29.36 | russellb | it's awesome |
| 15:29.41 | twisted | you in it now? |
| 15:29.42 | russellb | except for not being able to see my screens from glare |
| 15:29.43 | russellb | yeah |
| 15:29.49 | russellb | i'm wearing sunglasses |
| 15:29.52 | twisted | lol |
| 15:29.54 | twisted | i should be |
| 15:30.01 | twisted | 4 1/2 pitchers of beer last night |
| 15:30.02 | russellb | cool, feel free to come on by and check it out |
| 15:30.04 | bkw_ | twisted, I have too.. the allison voice.. she says that one word really well |
| 15:30.17 | twisted | bkw_: yeah, almost like she recorded that word specifically... |
| 15:30.22 | bkw_ | yep |
| 15:30.38 | Qwell | bkw_: ha, I didn't even think to try that |
| 15:30.47 | Qwell | she would be *pissed* if you sent that to her :p |
| 15:30.49 | bkw_ | its the first thing I had her say |
| 15:30.59 | bkw_ | told her that at astricon too |
| 15:31.02 | bkw_ | she cracked up |
| 15:31.04 | twisted | bkw_: lol... me 2 |
| 15:31.05 | Qwell | nice |
| 15:31.16 | iCEBrkr | haha |
| 15:31.19 | Qwell | russellb: that word Allison doesn't say :p |
| 15:31.23 | twisted | lol |
| 15:31.23 | russellb | Qwell: oh, ha |
| 15:31.24 | Qwell | the cepstral voice says it |
| 15:31.32 | twisted | it says it PERFECTLY |
| 15:31.40 | russellb | nice |
| 15:32.10 | twisted | her cepstral voice is one of the best cepstral voices i've heard |
| 15:32.24 | twisted | not a lot of breakage in pronounciation like most ot the others |
| 15:32.27 | iCEBrkr | Allison has a cepstral voice? |
| 15:32.36 | twisted | Hi, welcome to 2007 |
| 15:32.40 | iCEBrkr | twisted: haha |
| 15:32.58 | iCEBrkr | I haven't touched cepstral in, nearly a year. |
| 15:33.01 | twisted | heh |
| 15:33.16 | iCEBrkr | I got the sdk/dev kit deal.. Tinkered with it.. and forgot about it |
| 15:33.21 | twisted | heh |
| 15:33.47 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 15:33.59 | twisted | cepstral is fun |
| 15:34.16 | rantsh | any help on my transcoding b2bua would be very much appreciated |
| 15:35.19 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:36.36 | twisted | wtf |
| 15:36.42 | twisted | who tries to recruit through myspace |
| 15:36.46 | twisted | seriously |
| 15:39.01 | Zeeek | try recruting through "second wife" |
| 15:39.16 | jcanfield | http://cepstral.com/demos/ |
| 15:39.27 | jcanfield | fun |
| 15:42.33 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:42.45 | iCEBrkr | I got 2nd life, cuz I don't have a 1st life. |
| 15:43.37 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:43.42 | *** join/#asterisk schue (n=ean@pitch.brainfood.com) |
| 15:43.51 | schue | howdy. |
| 15:44.33 | *** join/#asterisk Trionnis (n=blah@65-117-172-195.dia.static.qwest.net) |
| 15:44.34 | schue | Any thoughts on where I should RTFM for routing multiple inbound DIDs to different extensions on an IAX connection to Asterisk. |
| 15:44.36 | *** join/#asterisk jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 15:44.46 | Bladerunner05 | hi all I'm looking for a good agi script to check if a password is valid or not |
| 15:44.47 | schue | IAX2, actually, I think., |
| 15:44.52 | Bladerunner05 | any suggestion ? |
| 15:44.59 | Trionnis | schue: what tech is the incoming channel? |
| 15:45.10 | schue | Trionnis: IAX2? |
| 15:45.12 | iCEBrkr | Bladerunner05: I'd use func_odbc in place of AGI |
| 15:45.28 | Trionnis | ok, wasn't clear if incoming was iax, or the extensions were iax :) |
| 15:45.43 | schue | Trionnis: its a Junction Networks connection. |
| 15:45.50 | tzafrir_home | Bladerunner05, what do you eman by "valid"? test it vs. what? How do you get the password? |
| 15:46.16 | Trionnis | hm, not sure right offhand how to do it with iax... sip is pretty easy |
| 15:46.18 | schue | Trionnis: who, amusingly, gave me someone else's phone number. |
| 15:46.22 | iCEBrkr | I want a AGI script to come into work for me and do my work. |
| 15:46.47 | schue | Trionnis: hmmm. |
| 15:47.18 | schue | so... when you have your inbounds.... |
| 15:47.29 | schue | exten => _1NXXNXXXXXX,1,Goto(mainextension|s|1) |
| 15:47.34 | Trionnis | with sip, the header "To" is the DNIS |
| 15:47.44 | Trionnis | I'm looking at iax to see if there's anything similar |
| 15:47.50 | schue | that _1NXXNXXXXXX pattern matches the inbound number... right? |
| 15:48.20 | Trionnis | hm |
| 15:48.23 | Trionnis | kinda |
| 15:48.26 | schue | Trionnis: yeah. i saw some stuff about that. |
| 15:48.56 | Bladerunner05 | <iCEBrkr>: so using a php script query it into db, I need only example for exchangin variables |
| 15:50.16 | *** join/#asterisk mltlnx (n=mltlnx@64.3.170.41.ptr.us.xo.net) |
| 15:50.35 | Trionnis | I suppose you could just use an extension for it |
| 15:51.02 | Trionnis | but it wouldn't really let you do different things based on the number, 'cause I don't *think* that's set in a variable anywhere |
| 15:52.17 | *** join/#asterisk STeven_elvisda (n=Steven_E@202.47.107.60) |
| 15:52.19 | schue | Trionnis: using a different pattern didn't seem to work. |
| 15:52.48 | schue | If that pattern does match the inbound number then I don't really understand why that is useful. |
| 15:53.04 | schue | Or, more useful than the callee would be... |
| 15:53.16 | schue | but I guess most people run an asterisk server for a single business. |
| 15:53.43 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 15:54.09 | Trionnis | well, I do something very similar to what you're doing for separate menus for different companies |
| 15:54.17 | Trionnis | thing is, my incoming trunks are all sip, so it's easy :) |
| 15:55.01 | Trionnis | I'm not really seeing anything in IAX that can easily do the same thing though... likely someone else here that's a bit more familiar with it would have better luck |
| 15:55.39 | *** part/#asterisk karlh626 (n=kharris@IP-66.249.227.5.indigital.net) |
| 15:56.38 | schue | Trionnis: it surprises me that this isn't easily done with IAX. |
| 15:56.44 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 15:56.46 | Trionnis | it very well might be |
| 15:56.52 | Trionnis | I'm just an IAX noob :) |
| 15:57.23 | iCEBrkr | Bladerunner05: I'd still use func_odbc and throw the query into the .conf file. |
| 15:57.35 | iCEBrkr | Bladerunner05: It's faster, it's less overhead and easier to maintain |
| 15:58.32 | mocker | Having a problem w/ sip regcontext w/ softphones. When the softphone is running, it works fine, but if the softphone exits, it's no longer in regcontext so calls don't go to voicemail they just die. |
| 15:58.44 | mocker | Anyone worked around that before? |
| 15:59.03 | iCEBrkr | mocker: You have the dialplan setup correctly? |
| 15:59.28 | *** join/#asterisk xlyz (n=scoma@81-174-26-100.static.ngi.it) |
| 15:59.42 | mocker | iCEBrkr: Well, it isn't working like I want it to, so now. :) |
| 15:59.47 | mocker | er, so no. |
| 15:59.47 | iCEBrkr | mocker: hehe.. |
| 15:59.51 | thewiizle | hi, how do i call the agi_extension into a php script |
| 16:00.09 | thewiizle | im trying to define it but its warning |
| 16:00.19 | schue | Trionnis: I think it may be ${EXTEN} |
| 16:00.22 | iCEBrkr | mocker: well dial() returns a status, just cover your bases and have it all dump to vmail |
| 16:00.46 | mocker | iCEBrkr: But it doesn't see that as a valid extension to even go to dial. |
| 16:00.57 | mocker | Because it sees w/ regcontext where the extension is. |
| 16:01.03 | iCEBrkr | mocker: You saying it doesn't get to dial()? |
| 16:01.16 | iCEBrkr | I dunno what regcontext is.. |
| 16:01.33 | mocker | iCEBrkr: right, because regcontext NoOps the priority(1) so I can do a dundi lookup for where the phone lives. |
| 16:01.35 | mpruett | Howdy everyone! I am having trouble getting xlite to work with asterisk on video calls. |
| 16:01.36 | mocker | Ahh. |
| 16:01.45 | hmmhesays | why is that? |
| 16:01.48 | Trionnis | anyone able to tell me if it's possible to statically set the bulk of the sip/iax options for connections in a config file, and have just the username and password pulled in with realtime? |
| 16:02.01 | mpruett | I can only get video in one direction to work correctly. |
| 16:02.39 | Trionnis | to explain a bit, I'd like to pull the user and password data from another unrelated table instead of having to keep tables in sync all the time |
| 16:02.47 | iCEBrkr | mocker: welp, that's out of my realm of knowledge.. I haven't touched dundi nor anything outside of a simple extension.conf :P |
| 16:04.13 | mocker | iCEBrkr: Thanks for trying. :) |
| 16:06.55 | mpruett | Anyone have any ideas why I can only get video to work in one direction using xlite? |
| 16:07.05 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-193-236.dsl.irvnca.pacbell.net) |
| 16:07.55 | hmmhesays | nat? |
| 16:09.42 | mpruett | I have nat=yes and regular calls work fine. We are able to connect and get audio to work fine but only one person can send video - It is a coin flip on who can actually send video correctly, usually it is the calling party |
| 16:10.31 | deeperror | mpruett: 10000-20000 port forwarding? |
| 16:10.44 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 16:10.46 | Zeeek | Voip Users Conference is in 15 minutes in the Havana Room, folks |
| 16:10.49 | mpruett | No - I can try that |
| 16:11.07 | devel | greetings all. anybody here who uses realtime in the dialplan? |
| 16:11.32 | deeperror | Zeeek: where? |
| 16:11.35 | mpruett | Thanks guys |
| 16:11.49 | Zeeek | http://www.VoipUsersConference.org for details |
| 16:11.51 | Zeeek | bring your cigars and whiskey glasses |
| 16:13.05 | *** part/#asterisk Ebola (i=ebola@goatse.co.uk) |
| 16:13.34 | UnixDog | zeek you around |
| 16:14.17 | UnixDog | hey what the irc channel again and why on the site does it not show us schedualed for today |
| 16:14.30 | UnixDog | it says th 12th |
| 16:14.32 | Zeeek | it does |
| 16:14.33 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:14.44 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 16:14.52 | Zeeek | http://voipusersconference.org/topics.php |
| 16:15.18 | Mimmus | good evening, does someone use this channel-bank: http://spidermux.com ? |
| 16:15.44 | Mimmus | It converts FXS/FXO to TDMoE instead of T1/E1 |
| 16:15.51 | Mimmus | and this seems a good idea |
| 16:16.00 | *** join/#asterisk ibob63_ (n=james@bb-87-82-14-140.ukonline.co.uk) |
| 16:16.05 | UnixDog | ok and the irc channel ? |
| 16:16.45 | *** join/#asterisk jengelh (n=jengelh@sovereign.computergmbh.de) |
| 16:17.24 | Zeeek | <PROTECTED> |
| 16:17.24 | UnixDog | zeek whats the irc chanel |
| 16:17.28 | UnixDog | ok |
| 16:17.29 | Zeeek | <PROTECTED> |
| 16:17.43 | jengelh | Qwell: |
| 16:17.48 | [TK]D-Fender | Mimmus, YUCK |
| 16:18.32 | Mimmus | [TK]D-Fender: I already have a Rhino channel-bank but I'd like to save a T1 port on my Digium card! |
| 16:19.12 | *** part/#asterisk jengelh (n=jengelh@sovereign.computergmbh.de) |
| 16:20.01 | ibob63_ | can anyone recommend a European PSTN gateway provider who use IAX? |
| 16:20.34 | Mimmus | http://www.voip4biz.it/ |
| 16:21.21 | ibob63_ | Mimmus: have you used them before? |
| 16:21.42 | Mimmus | ibob63_: only 50 Euro of pre-paid credit! |
| 16:21.46 | [TK]D-Fender | Mimmus, then buy a SIP gateway |
| 16:21.56 | Mimmus | [TK]D-Fender: tell me more... |
| 16:22.21 | [TK]D-Fender | Mimmus, AudioCodes MP-124 , Mediatrix 1124, etc |
| 16:23.03 | [TK]D-Fender | Mimmus, That Spidermux unit doesn't scale and is directly restricted to * an its uptime. |
| 16:23.20 | [TK]D-Fender | Mimmus, it is a dead-end non-recyclable mistake. |
| 16:23.52 | Mimmus | [TK]D-Fender: ahhhh. I don't understand well but I will meditate on your words |
| 16:25.15 | syzygyBSD | careful a samurai doesn't sneak up behind you while you meditate |
| 16:25.23 | [TK]D-Fender | Mimmus, TDMoE requires a direct ehternet link to *. If * dies, IT dies. Also if you stop liking * your unit becomes worthless |
| 16:25.31 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:25.44 | [TK]D-Fender | Mimmus, It also means it has to be physically close to *. |
| 16:26.27 | Mimmus | [TK]D-Fender: same consideration for a traditional channel-bank? |
| 16:26.42 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-a339e02d74e16ce4) |
| 16:26.42 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:27.27 | *** join/#asterisk mltlnx (n=mltlnx@64.3.170.41.ptr.us.xo.net) |
| 16:27.37 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:28.11 | [TK]D-Fender | Mimmus, No, at least a channel bank can be used in any T1 scenario. That means dozens of other solutions. it has the same proximity problem, but is reusable. SIP gateways can be used by most solutions and don't require T1 at all thus saving wiring and cost. |
| 16:28.22 | [TK]D-Fender | Mimmus, that is the best way to handle voice FXS |
| 16:28.44 | Mimmus | [TK]D-Fender: OK, I'm thinking to a media gateway also for PRI |
| 16:28.54 | [TK]D-Fender | Mimmus, SIP gateways can have redundant links to failover servers and those 2 models have analog failover options BUILT-IN |
| 16:29.25 | Mimmus | [TK]D-Fender: now I'm using internal digium/sangoma cards but I'd like to have no hw on * servers |
| 16:29.31 | rantsh | hey I got the transcoding up with softphones |
| 16:29.35 | rantsh | thanks guys |
| 16:30.38 | rantsh | anyways, I get this error message when I try to use my granstream box WARNING[14479]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (38), any pointers????? |
| 16:32.43 | Mimmus | [TK]D-Fender: both of AudioCodes and Mediatrix products work with Asterisk? |
| 16:32.59 | [TK]D-Fender | Mimmus, yes. |
| 16:33.54 | hmmhesays | i've used mediatrix extensively with asterisk |
| 16:35.07 | Mimmus | does Asterisk see analog channels as Zap devices, like with channelbanks? |
| 16:35.49 | [TK]D-Fender | Mimmus, No... perhaps you didn't hear me say this the first 10 times .. **SIP** |
| 16:36.21 | Mimmus | [TK]D-Fender: OK, sorry |
| 16:36.48 | *** join/#asterisk Kandinsky (n=Kandinsk@perla2.tm.ew.ro) |
| 16:37.10 | *** join/#asterisk PSU_Boss_1 (n=Eric@unaffiliated/psuboss/x-309451) |
| 16:37.28 | [TK]D-Fender | Mimmus, And naturally this means you have no need of special hardware |
| 16:38.38 | Mimmus | [TK]D-Fender: this is useful during Asterisk upgrades! I don't need to upgrade drivers, etc |
| 16:39.37 | Mimmus | I'm looking for resellers |
| 16:39.41 | Mimmus | in Italy |
| 16:39.50 | *** join/#asterisk ToTo (n=ToTo@host75-142-dynamic.8-87-r.retail.telecomitalia.it) |
| 16:40.06 | *** part/#asterisk ming_zym (n=ming_zym@124.254.56.252) |
| 16:40.11 | xheliox | anyone had a problem with persistent queue members disappears since upgrading to 1.4.12? |
| 16:40.22 | xheliox | disappearing too. |
| 16:43.14 | *** part/#asterisk ibob63_ (n=james@bb-87-82-14-140.ukonline.co.uk) |
| 16:47.16 | Mimmus | [TK]D-Fender: now a suggestion for SIP-PRI gateways... |
| 16:47.26 | Mimmus | Patton? |
| 16:48.22 | [TK]D-Fender | Mimmus, No personaly experience with them. I suggest you visit some forums and do a bit of research first. AudioCodes is a little complex but very powerful, and Mediatrix is dead-easy. |
| 16:48.39 | [TK]D-Fender | (Sorry, was talking analog there) |
| 16:48.54 | [TK]D-Fender | Mimmus, I've only worked with AudioCodes PRI gateways once. |
| 16:49.01 | thewiizle | Tk what are you like on AGI scripting? |
| 16:49.08 | [TK]D-Fender | Mimmus, but I suggest you take this as a start to your research |
| 16:49.20 | [TK]D-Fender | thewiizle, /dev/null ;) |
| 16:49.21 | Mimmus | [TK]D-Fender: ok, but why 'complex'? what do you can expect from a product like these? |
| 16:49.37 | thewiizle | :P |
| 16:49.43 | thewiizle | damnit |
| 16:49.51 | thewiizle | im smashing my head into the wall here |
| 16:50.06 | *** join/#asterisk jetlagmk2 (i=jetlag@70.17.41.110) |
| 16:50.16 | [TK]D-Fender | Mimmus, Some are easier to configure than others, better documented, etc. That can be a starting factor in your choice, but in the end go with quality, flexibilty... |
| 16:50.27 | [TK]D-Fender | thewiizle, pastebin it and someone will have an idea. |
| 16:50.36 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 16:50.51 | thewiizle | heh i dont have an idea myself :P |
| 16:51.59 | [TK]D-Fender | thewiizle, hence why you should PASTEBIN it so others can see and help... |
| 16:52.09 | Mimmus | [TK]D-Fender: now I have 3 Asterisk servers in 3 sites of the company, everyone of these has PRI/BRI cards for local PSTN access, 1 Rhino channelbank to accomodate residual analog devices, ... |
| 16:52.32 | Mimmus | and now another site with analog devices to slowly migrate... |
| 16:52.54 | Mimmus | uff... I'm looking for manageability |
| 16:53.00 | [TK]D-Fender | Mimmus, For lower density analog FXS I suggest the Linksys SPA-8000. 8 ports @ < $300 USD |
| 16:53.24 | Mimmus | no, more ports... |
| 16:53.57 | [TK]D-Fender | Mimmus, I've already told you the 2 models I see most used and have worked with |
| 16:54.25 | Mimmus | yes, I already asked for an offer to my local resellers! I believe you! |
| 16:54.49 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 16:56.53 | Mimmus | in the future, I hope to have one site with one PRI AND 1 FXS gateway, now I seems a VoIP shop |
| 16:56.59 | Mimmus | (sorry for chatting...) |
| 16:58.41 | [TK]D-Fender | Mimmus, PRI gateways are very pricey but certain redundant setups depend on them (like SER multi-homed). |
| 16:59.07 | GreggB | I've got a PRI which goes down (according to the "Status" line from "pri show span 1") at roughly the same time just about every day (20:06-20:09 US/Pacific), though I have no cron processes which could be doing something at that time. Is there any possibility * could be doing something itself? |
| 16:59.56 | [TK]D-Fender | Mimmus, You can get an 4-port HWEC card & server for the price of a 1 port Patton PRI gateway |
| 17:01.05 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:04.42 | Mimmus | [TK]D-Fender: OK, price is important but sometime, especially at enterprise level, manageability is more important |
| 17:05.03 | Mimmus | now I'm not upgrading Asterisk because I down't want to upgrade Sangoma drivers |
| 17:05.22 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 17:05.37 | [TK]D-Fender | Mimmus, in larger scenarios most end up running SER as their core routing service, and * as a back-end application server |
| 17:07.07 | Mimmus | [TK]D-Fender: I understand... I have 350-400 users, I don't think to need SER |
| 17:08.08 | [TK]D-Fender | Mimmus, Actually you're not too far off. it depends how independant each site is capable of being from the collective |
| 17:08.47 | [TK]D-Fender | Mimmus, if you are heavily linked with roaming employees, things start to look very different. These are all points to be thought over in "the big picture" |
| 17:08.52 | Mimmus | [TK]D-Fender: where is the limit? now I have 5-15 contemporary calls, load is 0.05! |
| 17:10.57 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
| 17:11.23 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
| 17:11.27 | [TK]D-Fender | Mimmus, then in your case I wouldn't even consider a PRI gateway or SER. |
| 17:11.45 | [TK]D-Fender | Mimmus, local PTS/PRI card in a * per site. |
| 17:12.18 | [TK]D-Fender | Mimmus, but if you're using FXS for phones, go for high-density SIP gateways. |
| 17:12.29 | Mimmus | [TK]D-Fender: i.e my setup... but in the near future we'll have ONE site with all employees |
| 17:13.48 | [TK]D-Fender | Mimmus, then 1 4-port HWEC PRI card & server (duplicate server for backup). |
| 17:14.12 | *** join/#asterisk killfill (n=killfill@pc-164-134-45-190.cm.vtr.net) |
| 17:14.15 | killfill | hi. |
| 17:14.48 | Mimmus | [TK]D-Fender: ok, this is my setup in every site. thank you for your suggestions |
| 17:14.53 | killfill | i need that when an incomming call comes in, people popups the clients info, taken from the callerId. |
| 17:14.59 | Mimmus | now I'm going home, it's 19:15 in Italy |
| 17:15.06 | killfill | any recomendations for this setup? |
| 17:15.36 | Mimmus | bye |
| 17:15.39 | *** join/#asterisk mjgraves (n=mgraves@65.14.229.26) |
| 17:16.00 | killfill | its for a callcenter |
| 17:16.56 | [TK]D-Fender | killfill, make a server app polling for AMI queue call messages and have it push to a local client on their PC. |
| 17:17.26 | killfill | [TK]D-Fender: AMI? |
| 17:18.45 | killfill | ah.. :P manager |
| 17:18.58 | killfill | [TK]D-Fender: how would i put that into to the clients?.. |
| 17:19.30 | killfill | actually we plan to save all in sugarcrm. i could just see whats the phone. map it to an url, and send that url to the clients. |
| 17:19.32 | jcanfield | Anyone aware of an operator panel that would work well with a touch screen? |
| 17:19.42 | killfill | but how could one do that?.. send something to the client. |
| 17:19.48 | *** join/#asterisk StevenElvisda_ (n=Steven_E@202.47.107.60) |
| 17:20.16 | [TK]D-Fender | killfill, write an app to run on the client and push from the server to the client and have that pop the URL |
| 17:21.11 | killfill | hm.. |
| 17:21.14 | *** join/#asterisk _Krieger_ (n=warsword@91.102.176.6) |
| 17:21.27 | killfill | i would need to map the phone number to a user/pass or something like that. |
| 17:21.45 | killfill | so the client application, can have user/pass... |
| 17:21.49 | killfill | to diferenciate.. |
| 17:22.08 | _Krieger_ | if asterisk box is behind NAT relatively to users, what workaround is enough? |
| 17:22.34 | [TK]D-Fender | _Krieger_, .... |
| 17:22.36 | [TK]D-Fender | ~sipnat |
| 17:22.37 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:22.38 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 17:22.56 | _Krieger_ | thx :) |
| 17:23.16 | iCEBrkr | DEEEEEEEEEEEEEEEEEEFNDR |
| 17:24.01 | [TK]D-Fender | iCEBrkr, y0 |
| 17:24.19 | iCEBrkr | [TK]D-Fender: Did ya go to Astricon? |
| 17:24.48 | [TK]D-Fender | iCEBrkr, Nope... no passport yet, too far, and too expensive, with little net return on investment for me :) |
| 17:24.49 | iCEBrkr | I really need to make it to one of those. |
| 17:25.40 | *** join/#asterisk telamon (n=telamon@bridge.isn.net) |
| 17:26.28 | telamon | Anyone know where I can find the GXP-2000 1.1.1.14 firmware? It doesn't seem to be on the Grandstream site, and I need it in order to upgrade to newer firmwares. |
| 17:27.46 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-3b5b9e7d3a062c2b) |
| 17:29.03 | *** join/#asterisk javar (n=javar@200.118.168.197) |
| 17:29.25 | iCEBrkr | zzzzzzzzzzzzzz |
| 17:29.31 | iCEBrkr | I ate too much at lunch |
| 17:30.32 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 17:34.41 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 17:39.05 | hmmhesays | thats no good |
| 17:39.09 | hmmhesays | i'm playing with drupal |
| 17:41.55 | thewiizle | man |
| 17:41.58 | thewiizle | this shit aint even fun |
| 17:42.04 | telamon | Anyone know where I can find the GXP-2000 1.1.1.14 firmware? It doesn't seem to be on the Grandstream site, and I need it in order to upgrade to newer firmwares. |
| 17:42.05 | thewiizle | not like the rest of asterisk |
| 17:42.09 | thewiizle | asterisk was fun |
| 17:42.11 | thewiizle | agi is not |
| 17:43.00 | *** join/#asterisk IP_FIX (n=ip_fix@c953074b.virtua.com.br) |
| 17:43.32 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-93-91-14.dsl.hstntx.swbell.net) |
| 17:43.43 | *** join/#asterisk Lithium_Ion (n=asd@d38-41-195.commercial1.cgocable.net) |
| 17:44.10 | *** join/#asterisk mdumi (n=Mdumi@dsl-242-239-240.telkomadsl.co.za) |
| 17:45.16 | Lithium_Ion | Can someone help me troubleshoot fax. Most of my faxes go through but certain machines on outgoing faxes respond with a short beep that sounds like a handshake then nothing. |
| 17:45.20 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:46.10 | *** part/#asterisk mdumi (n=Mdumi@dsl-242-239-240.telkomadsl.co.za) |
| 17:49.34 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:00.47 | *** part/#asterisk xlyz (n=scoma@81-174-26-100.static.ngi.it) |
| 18:03.12 | *** join/#asterisk mltlnx (n=mltlnx@216.213.96.210) |
| 18:05.08 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 18:07.55 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:10.32 | hmmhesays | oh voip problems today |
| 18:17.28 | Katty | voip problems everyday |
| 18:18.21 | jingles | yup. |
| 18:18.28 | jingles | broadvoice is having some itchews. |
| 18:18.38 | jingles | saw an error today I'd never seen before. 409 *conflicts* |
| 18:18.58 | hmmhesays | I can't push this gateway past 150 calls before I start getting dead air on some calls |
| 18:23.16 | *** join/#asterisk kay2 (n=two@gob75-7-82-247-113-230.fbx.proxad.net) |
| 18:23.44 | kay2 | is there any way to decrease the volume of a SIP channel ? |
| 18:26.43 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:27.02 | [TK]D-Fender | kay2, not currently. 1.6 will have ways |
| 18:27.26 | *** join/#asterisk captiancrash (n=thejonmo@70.159.118.70) |
| 18:28.05 | kay2 | [TK]D-Fender: so I guess I need to create a pseudochannel and do it myself ? |
| 18:29.06 | [TK]D-Fender | kay2, right now there is no live way except to pass it through ZAP. |
| 18:30.03 | kay2 | [TK]D-Fender: you mean to put a cross cable, and make it go through it ? |
| 18:30.04 | [TK]D-Fender | TE120P + loopback = 12 channels of volume rigging goodness! |
| 18:30.35 | [TK]D-Fender | actually... I'm not sure that'd work... Would for sure on a 2 port card... |
| 18:32.25 | *** join/#asterisk BiG^DoG (n=BiG^DoG@c-71-204-211-58.hsd1.de.comcast.net) |
| 18:33.25 | Katty | mew. |
| 18:33.32 | [TK]D-Fender | Katty, Mew. |
| 18:35.41 | syzygyBSD | isn't Voip kind of a misnomer? I mean, it isn't over ip.... it is over a network |
| 18:36.09 | bkw_ | yes its over the ip protocol |
| 18:36.30 | citats | syzygyBSD: so call it voipon then :) |
| 18:37.17 | syzygyBSD | bkw_: it uses IP, but I wouldn't say it is over it |
| 18:37.36 | bkw_ | what ever |
| 18:37.44 | citats | syzygyBSD: i think your a few years to late for this argument |
| 18:37.46 | bkw_ | you're just being pedantic |
| 18:39.01 | syzygyBSD | bkw_: an easy argument for anything you don't want to think about |
| 18:39.26 | bkw_ | its a layer on top of the IP stack |
| 18:39.34 | bkw_ | so in the iso model its OVER ip |
| 18:40.43 | syzygyBSD | thanks, a very good explination |
| 18:41.09 | syzygyBSD | s/explination/explanation/ |
| 18:41.09 | coppice | its a part of the 7-layer burrito |
| 18:41.15 | iCEBrkr | Did someone say burrito? |
| 18:41.38 | iCEBrkr | BurritoOIP |
| 18:41.52 | iCEBrkr | [TK]D-Fender: hey hey |
| 18:42.12 | iCEBrkr | mmm good. func_odbc built. |
| 18:42.22 | iCEBrkr | Now I can let the phone-spam begin! |
| 18:42.27 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-68-89.dsl.irvnca.pacbell.net) |
| 18:42.29 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 18:42.30 | *** join/#asterisk ru{b}y (n=ruy@201.22.56.237.static.gvt.net.br) |
| 18:42.31 | syzygyBSD | hmm, I have never heard the burrito way of remembering the stack? what do the letters stand for? there are seven... |
| 18:42.33 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 18:44.55 | tzafrir_home | http://www.itworld.com/Net/3303/071004jajah/ - claiming eBay is blocking competitors of Skype |
| 18:44.56 | coppice | doppy |
| 18:44.59 | coppice | sleepy |
| 18:45.03 | coppice | happy |
| 18:45.14 | [TK]D-Fender | sleezy |
| 18:45.15 | Qwell | don't forget sneezy |
| 18:45.19 | Qwell | or sleezy |
| 18:45.24 | [TK]D-Fender | behold... the 7 deadly dwarves! |
| 18:45.30 | iCEBrkr | [TK]D-Fender: you'd know sleezy.. eh? |
| 18:45.31 | iCEBrkr | :P |
| 18:45.38 | Qwell | [TK]D-Fender: nice |
| 18:47.36 | deeperror | what would CDR on channel lacks start indicate? |
| 18:48.25 | codefreeze | deeperror: it indicates strangeness. your asterisk version? |
| 18:48.39 | deeperror | 1.4.11 |
| 18:48.53 | codefreeze | deeperror: ok, and in what situation is this happening? |
| 18:49.13 | deeperror | i'm not 100% sure i'm unable to reproduce myself and no one is complaining so hard for me to really see what is causing it |
| 18:49.24 | deeperror | i just see it in the logs quite often |
| 18:49.27 | *** join/#asterisk ctooley (n=ctooley@209.33.109.249) |
| 18:50.30 | codefreeze | deeperror: at first, I thought, some reset might do it, but a reset usually sets the start time... if you can find a pattern, let me know. |
| 18:50.57 | deeperror | i've got heavy logging enable, still looking for patterns |
| 18:51.08 | deeperror | i actually get 2 warnings and a notice at the same time |
| 18:51.39 | codefreeze | deeperror: all from the cdr subsystem, no doubt |
| 18:52.37 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 18:53.19 | deeperror | http://pastebin.ca/727068 |
| 18:55.28 | deeperror | here is what cli is showing http://pastebin.ca/727070 |
| 18:57.01 | codefreeze | deeperror: you need to start the snippet when zap/5 was picked up, till when it hung up and issued those messages. |
| 18:57.26 | deeperror | codefreeze: could this be caused by having very verbose logging enabled and several calls being performed at the same time? |
| 18:57.45 | deeperror | maybe messages file is locked or can't be written |
| 18:58.37 | codefreeze | deeperror: no, I've tried to break the logging on purpose. couldn't do it. it locks, etc. and works pretty good. |
| 18:59.05 | deeperror | let me get the full call log |
| 18:59.08 | *** part/#asterisk captiancrash (n=thejonmo@70.159.118.70) |
| 18:59.21 | deeperror | i think its occuring on 3way calls mainly but not 100% |
| 19:00.42 | *** join/#asterisk ekiczek (n=ekiczek@h-72-245-66-3.cmbrmaor.covad.net) |
| 19:12.14 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 19:14.26 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:14.26 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 19:14.32 | [hC] | JavaMan! |
| 19:16.03 | blitzrage | High C! |
| 19:16.39 | [hC] | :) |
| 19:17.07 | Corydon76-dig | Spoo- oh, wait, you didn't... |
| 19:17.09 | [hC] | I decided that since it was friday and i am comfy, im just going to work from home today in a super comfy chair in the living room |
| 19:17.43 | Corydon76-dig | GET... the COMFY CHAIR!!! |
| 19:17.51 | [hC] | hahha |
| 19:18.09 | [hC] | I guess THIS is one of those benefits they talk about with regard to "being your own boss" |
| 19:18.24 | [hC] | :) |
| 19:18.31 | iCEBrkr | uh huh. |
| 19:18.39 | Corydon76-dig | [hC]: I dunno, I'm in a padded chair in my house... |
| 19:18.43 | iCEBrkr | Slackass |
| 19:18.53 | iCEBrkr | Corydon76-dig: Sure that's not a padded room? |
| 19:19.26 | blitzrage | I'm in my home office |
| 19:19.28 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-17.dq1sn.easystreet.com) |
| 19:19.30 | Corydon76-dig | iCEBrkr: even if it was, it wouldn't help. The server cabinet next to me has sharp edges |
| 19:19.59 | iCEBrkr | yeah, no good. |
| 19:20.14 | Corydon76-dig | Well, sharp corners anyway |
| 19:20.28 | [hC] | blitzrage: how are you liking your mac? |
| 19:20.39 | blitzrage | pretty cool so far |
| 19:20.43 | blitzrage | i actually do like OSX |
| 19:20.45 | Corydon76-dig | He likes it better than gay sex |
| 19:20.57 | blitzrage | I like pretty much everything more than gay sex |
| 19:21.03 | sjobeck | hi all, how are things? It has been so long since I updated Zaptel on this one server of mine that today, just now, when I did, I'm getting this error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) and I cant remember how we resolved it last time. not sharing interupt, power cable, tried calibrate=off, to no effect. ideas? |
| 19:21.18 | [hC] | yeah, <3 osx. I got a powerbook in like... late 2003 early 2004, i forget.. i opened it up and abandoned my linux desktop that day, and havent used anything since |
| 19:21.19 | Corydon76-dig | blitzrage: see? I was right |
| 19:21.19 | karleeto | is there a way to do a mass reboot on polycom phones? |
| 19:21.37 | blitzrage | Corydon76-dig: yes... you were right, heh |
| 19:21.46 | [hC] | karleeto: issue the command 'sip notify polycom-check-cfg <exten number>' in the asterisk console for each phone |
| 19:21.51 | sjobeck | karleeto, you can write a script to send sip reboot command to each of them & then run that script to automate sending it to every device |
| 19:22.10 | karleeto | ok, thanks |
| 19:22.10 | [hC] | karleeto: depending on your sip.cfg you may need to enable the option which makes it so they always reboot when you send a check config, otherwise they will only reboot if their config files have changed |
| 19:22.11 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 19:22.24 | _pepo_ | hi friends |
| 19:22.33 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 19:22.47 | sjobeck | polycoms dont reboot automatically on config file change, they dont know about that change |
| 19:22.48 | [hC] | I found the coolest thing the other day... a Free .Mac replacement, that uses your servers instead of apples.. exactly what i wanted, integrates as though it were .mac precisely |
| 19:23.00 | _pepo_ | How do I can forwarding a call to any extension in system A (running Asterisk) if it did not answear to the voice mail in the system B (other Asterisk)? so later users of system A can check the remote messages |
| 19:23.12 | sjobeck | any one have any ideas for me on that ZT_CHANCONFIG failed on channel 1: No such device or address (6) error |
| 19:23.18 | [hC] | sjobeck: i never said they did. I said if you issue a sip notify polycom-check-cfg from inside asterisk, it will check the files, and by default they will reboot if they had changed. |
| 19:23.32 | sjobeck | hC: ahh, I see, thx |
| 19:24.04 | sjobeck | hC: and that is basically what his bash script needs to do, call asterisk -rx, and then send that to each device. |
| 19:24.19 | [hC] | yeah, i use this line |
| 19:24.26 | sjobeck | russellb ? |
| 19:24.39 | sjobeck | file ? |
| 19:25.02 | sjobeck | should I call Digium Support about this TDM card being a bugger for me today? |
| 19:25.40 | sjobeck | i know I got it to compile & ztcfg -vvvvvvvvvvvvd last time but cant dont know what we did to get it to do so, something about this card that didnt like the default compile options. |
| 19:27.16 | [TK]D-Fender | sjobeck, modprobe your card(s)' module(s) and verify with "cat /proc/interrupts" then "ztcfg -vvvv". Then start * manually. |
| 19:27.43 | sjobeck | thx, did that |
| 19:28.18 | *** join/#asterisk wiljacket (n=wilson@cpe-76-173-243-4.socal.res.rr.com) |
| 19:28.26 | russellb | sjobeck: yes, call them |
| 19:29.37 | sjobeck | all: I am so embarassed, totally my fault, I was running my own start script which was supposed to modprobe |
| 19:29.48 | sjobeck | it didnt, I did, and it worked |
| 19:30.02 | sjobeck | thx so much! it was just the process of talking out loud that kicked my brain over |
| 19:30.04 | sjobeck | thx all! |
| 19:30.05 | sjobeck | great |
| 19:30.18 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:30.33 | [TK]D-Fender | np |
| 19:33.16 | Corydon76-dig | Strom_C: do you normally explore reasons? |
| 19:34.08 | jameswf | I am fat I dont explore anything :) |
| 19:34.28 | Strom_C | Corydon76-dig: perhapssibly |
| 19:39.54 | *** join/#asterisk Servergod (n=Maverick@70.97.159.120) |
| 19:44.23 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 19:52.14 | *** join/#asterisk iPod-nano (n=iPod-nan@c-68-43-60-193.hsd1.mi.comcast.net) |
| 19:52.34 | iPod-nano | When I get an incoming call, if I hit a number it hangs up. |
| 19:52.49 | iPod-nano | Can I change that? |
| 19:53.03 | lirakis | hmm .. using FOP .. i get "Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 7396." when i try to conference |
| 19:53.55 | iPod-nano | I just got GrandCentral and I love having it call me, but I have to press 1 to accept the call, which causes my connection to hang up. |
| 19:59.27 | putnopvut | Is "ringing" a valid option for the Playtones() application? |
| 19:59.45 | putnopvut | assuming a default indications.conf built from make samples? |
| 20:00.39 | putnopvut | Hmm, I think it may actually be "ring" |
| 20:02.28 | putnopvut | Yup, it's "ring" |
| 20:04.07 | iPod-nano | Why does it hang up if I press a number? |
| 20:04.35 | deeperror | what hangs up? |
| 20:05.59 | [TK]D-Fender | iPod-nano, show us the CLI output of a call where this is happening with SIP debug enabled |
| 20:06.56 | iPod-nano | Could you remind me how to enable it? |
| 20:08.10 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.227) |
| 20:08.46 | iPod-nano | I got it, OK hold on. |
| 20:08.56 | *** part/#asterisk javar (n=javar@200.118.168.197) |
| 20:13.37 | deeperror | codefreeze: still around? |
| 20:13.44 | iPod-nano | There's so much output I don't even know where to start copying. |
| 20:15.47 | [TK]D-Fender | iPod-nano, try... the BEGINNING |
| 20:16.47 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
| 20:17.30 | iPod-nano | http://rafb.net/p/9KGUZW13.html |
| 20:18.32 | Servergod | tail +f /var/log/asterisk/full > call.txt |
| 20:18.36 | Servergod | and make the call |
| 20:18.50 | [TK]D-Fender | Oct 5 16:15:42 WARNING[2498]: channel.c:2380 set_format: Unable to find a codec translation path from ilbc to ulaw |
| 20:18.56 | [TK]D-Fender | iPod-nano, Codec mismatch...... |
| 20:19.28 | iPod-nano | The call connected and I heard the automated voice. |
| 20:19.32 | iPod-nano | I pressed 1. |
| 20:19.38 | iPod-nano | Call disconnected. |
| 20:19.54 | [TK]D-Fender | iPod-nano, And why do I not see your phone being dialed or answered? |
| 20:20.48 | deeperror | features.conf? |
| 20:20.54 | iPod-nano | Couldn't tell you. That's everything that happened from the point I called to the point it hung up. |
| 20:21.14 | Katty | yay! |
| 20:21.19 | Katty | tonight is my birthday dinner :> |
| 20:21.27 | Katty | eta 3 hours! |
| 20:21.30 | twisted | nice :) |
| 20:21.43 | Netgeeks | happy birthday, Katty |
| 20:21.50 | mishehu | Katty: happy bday |
| 20:21.55 | Katty | thank you thank you :> |
| 20:22.00 | Katty | i'm all grown up :> |
| 20:22.02 | mishehu | with a kandle on it. |
| 20:22.03 | Katty | again :> |
| 20:22.29 | Katty | sounds spektakular. |
| 20:22.37 | mishehu | Katty: being grown up is a lot of work! |
| 20:22.43 | Katty | mishehu: i know :/ |
| 20:23.09 | [TK]D-Fender | iPod-nano, no, we are missing the first part of the invite we see onscreen, and clearly cannot see a single dialplan app being called. |
| 20:24.09 | [TK]D-Fender | iPod-nano, Crank up your verbose, and show EVERYTHING. |
| 20:25.45 | Katty | mishehu: how is your birdie? :> |
| 20:25.51 | Katty | mishehu: taking off the visitors arms i hope :> |
| 20:28.07 | *** join/#asterisk NirS (n=chatzill@87.68.157.27) |
| 20:30.30 | mishehu | Katty: nah, she's been good. |
| 20:30.43 | mishehu | Katty: I've not been taking her out a lot the past two months though, been WAY too busy. |
| 20:30.55 | mishehu | but she's in the living room so at least she sees people. |
| 20:32.11 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-a339e02d74e16ce4) |
| 20:32.30 | Katty | mishehu: :> |
| 20:37.42 | *** join/#asterisk dexpdx (n=jason@66-162-134-242.static.twtelecom.net) |
| 20:42.46 | *** join/#asterisk pots_line (n=bryan@66-43-34-50.misn.com) |
| 20:47.33 | *** join/#asterisk mltlnx (n=mltlnx@m125f36d0.tmodns.net) |
| 20:50.37 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:55.54 | Katty | i wanna go home!!! >.< |
| 20:57.53 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 20:57.58 | ussrback | Hi all |
| 20:58.05 | ussrback | I have Asterisk 1.4 |
| 20:58.17 | ussrback | how can i unload modules |
| 20:59.11 | ussrback | i add noload => some_module.so |
| 20:59.15 | ussrback | in modules.conf |
| 20:59.23 | ussrback | bit it still loads this module |
| 21:01.15 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 21:01.45 | ussrback | hello motooooo |
| 21:01.55 | ussrback | anybodyyy alive? |
| 21:02.25 | sheppard | they're dead Jim |
| 21:03.11 | Katty | hi. |
| 21:03.15 | ussrback | hi |
| 21:03.22 | Katty | i was turning in my grave. |
| 21:03.40 | ussrback | ghghghghg |
| 21:03.45 | Katty | really. |
| 21:03.45 | ussrback | Like a Gogol |
| 21:03.55 | Katty | no, like a murloc. |
| 21:04.11 | Katty | ...from louisiana. |
| 21:04.20 | ussrback | so hows under ground :) |
| 21:04.36 | Katty | lonely. |
| 21:04.41 | Katty | no one is ever in Undercity. |
| 21:04.48 | Katty | They're always in Orgrimmar. |
| 21:05.15 | ussrback | oh |
| 21:06.15 | ussrback | ill put my * there now |
| 21:06.37 | Katty | They don't need your asterisk box. |
| 21:06.49 | Katty | No one makes phone calls there. They either get a port or a summon to their party. |
| 21:07.41 | ussrback | yes but i put it there cause its dead |
| 21:11.04 | Katty | ^_- |
| 21:11.13 | Katty | The dead go to the spirit healer. |
| 21:11.17 | Katty | The undead go to undercity. |
| 21:11.22 | Katty | silly rabbit. |
| 21:11.27 | Katty | clearly you've never ran an instance or raid. |
| 21:11.51 | Katty | apparently my World of Warcraft humor is going unappreciated :< |
| 21:12.10 | ussrback | LoL |
| 21:14.46 | Katty | anthm: ping. |
| 21:16.48 | ussrback | nick XQZME |
| 21:18.22 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 21:20.16 | funxion | anyone know if using immediate=yes in zapata.conf on a E&M T1 would cause outbound billsec to include the time from the inbound portion of the call |
| 21:21.19 | XQZME | probably they kno, but they r dead |
| 21:21.33 | funxion | lol |
| 21:21.41 | funxion | nobody's here huh |
| 21:21.58 | tzafrir_home | Why would you use immediate=yes? |
| 21:22.29 | funxion | I have 2 different trunk groups on 1 incoming t1 |
| 21:22.55 | funxion | its also the termination point using the same 2 trunk groups for different COS |
| 21:23.44 | funxion | since I dont need callerid I use immediate-yes and s,1 to save a second or 2 on call setup |
| 21:23.54 | funxion | call setup takes a while its going over a satellite |
| 21:24.04 | funxion | so every little bit helps |
| 21:24.53 | funxion | tzafrir_home do you know if it would cause outbound billsec to include the time from the inbound portion of the call |
| 21:24.56 | funxion | ? |
| 21:26.09 | tzafrir_home | why not just callerid=no |
| 21:26.16 | tzafrir_home | (or sothing similar) |
| 21:26.39 | *** join/#asterisk ekiczek (n=ekiczek@h-72-245-66-3.cmbrmaor.covad.net) |
| 21:27.01 | funxion | but I need to just go of hook and start playing prompts when a call comes in I dont want to wait for digits I identify cos by the channel #asterisk |
| 21:27.15 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 21:33.03 | *** join/#asterisk tristanbob (n=tristan@oalug/member/tristanbob) |
| 21:36.16 | *** join/#asterisk rkeels (n=chatzill@99.eedinc.com) |
| 21:37.01 | Strom_C | dan__t: ? |
| 21:37.08 | rkeels | can anyone here help me to create an automatic ring down hotline. ie go off hook and the phone automatically calls a preprogramed number |
| 21:37.30 | Strom_C | rkeels: easy depending on your hardware |
| 21:37.47 | rkeels | Polycom phones and Asterisk 1.4 |
| 21:37.49 | dan__t | I'm trying to configure this phone to be used with a SIP provider, namely Teliax. |
| 21:37.58 | rkeels | Poly 430 to be exact |
| 21:38.11 | dan__t | I'd like to configure it through the phone itself, and not have to boot a config file. |
| 21:41.20 | dan__t | Is this something you might have experience in, Strom_C? |
| 21:41.24 | _x86_ | "Good health" is merely the slowest rate at which one can die. |
| 21:41.28 | _x86_ | fortune cracks me up |
| 21:41.42 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 21:41.56 | Strom_C | dan__t: I tend to configure the phones using config files; they're really designed for deployments moreso than individual installs |
| 21:42.04 | dan__t | I understand. |
| 21:42.15 | Strom_C | using the config files is a breeze |
| 21:42.21 | [TK]D-Fender | _x86_, This is clearly for you then : http://www.despair.com/pessimistsmug.html |
| 21:43.43 | [TK]D-Fender | rkeels, Yes you can set your PHONE to auto-dial a number on going off-hook. Go grab the latest firmware & admin guide. Its all in there. |
| 21:43.54 | gremzoid | do i need odbc support to use res_config_mysql and extconfig ? |
| 21:43.58 | rkeels | Not Really |
| 21:44.29 | [TK]D-Fender | twit |
| 21:44.58 | bkw_ | is name calling really needed? :P |
| 21:45.13 | _x86_ | There are two kinds of pedestrians... the quick and the dead. |
| 21:45.15 | _x86_ | rofl |
| 21:45.21 | dan__t | haha. |
| 21:45.29 | bkw_ | tranlsation smart and the stupid |
| 21:46.28 | [TK]D-Fender | bkw_, No, but it comes bundled with the package ;) |
| 21:46.34 | _x86_ | haha |
| 21:46.36 | gremzoid | so can anyone tell me why i can't get asterisk/mysql to work? |
| 21:46.57 | gremzoid | ie followed this to every dotted I and crossed T: http://www.asteriskguru.com/tutorials/realtime_pgsql.html |
| 21:47.22 | gremzoid | and all i get is: Realtime mapping for 'iaxpeers' found to engine 'mysql', but the engine is not available |
| 21:48.24 | _x86_ | I wish there was a knob on the TV to turn up the intelligence. There's a |
| 21:48.26 | _x86_ | knob called "brightness", but it doesn't seem to work. -- Gallagher |
| 21:48.35 | _x86_ | so true ;) |
| 21:49.55 | funxion | anyone have a clue why my billsec would = my duration in my cdr? |
| 21:50.50 | Katty | mew. |
| 21:51.12 | Corydon76-dig | funxion: if the first thing that happened was an Answer, it's possible |
| 21:51.32 | _x86_ | The problem with the gene pool is that there is no lifeguard. |
| 21:51.35 | _x86_ | lol |
| 21:51.40 | funxion | no |
| 21:52.24 | Katty | is that the metallicy tasting icky stuff people like to put into mexican dishes? |
| 21:52.25 | [TK]D-Fender | funxion, I think your dilaplan issuing "Answer" + "immediate=yes" sums it up <- |
| 21:52.37 | funxion | I do have immediate=yes |
| 21:52.42 | funxion | there we go |
| 21:52.46 | funxion | but only on one side |
| 21:52.50 | funxion | that sux |
| 21:52.52 | Katty | hi JunK-Y (= |
| 21:53.07 | [TK]D-Fender | funxion, remove the "answer" and see if letting it "ring through while still technically processing helps |
| 21:53.27 | [TK]D-Fender | JunK-Y, don't hurt me bebe! ;) |
| 21:53.28 | JunK-Y | whats up? |
| 21:53.31 | funxion | I need the channel to forward to s1 when it goes off hoook |
| 21:53.39 | Katty | JunK-Y: trying to figure out what 'cilantro' is. |
| 21:53.49 | [TK]D-Fender | JunK-Y, mom's coming in town gearing up for thanksgiving... the usuall.. |
| 21:53.53 | JunK-Y | [TK]D-Fender: stop hurting me baby! |
| 21:53.57 | [TK]D-Fender | Katty, a herb |
| 21:54.00 | funxion | thats why I was immediate=yes |
| 21:54.07 | Katty | [TK]D-Fender: what /kind/ of herb. |
| 21:54.14 | funxion | lol |
| 21:54.15 | Katty | [TK]D-Fender: is it a parsley herb. |
| 21:54.19 | JunK-Y | Katty: not something you can smoke! |
| 21:54.20 | Katty | [TK]D-Fender: or a sage/thyme herb. |
| 21:54.24 | _x86_ | [TK]D-Fender: thanksgiving already? |
| 21:54.25 | Katty | ha |
| 21:54.40 | Corydon76-dig | Would you consider cloves an herb? |
| 21:54.42 | _x86_ | she's like... more than a month early ;) |
| 21:54.53 | _x86_ | Corydon76-dig: smokeable herb ;) |
| 21:54.59 | *** join/#asterisk angom (n=angom@201.143.89.82) |
| 21:55.04 | [TK]D-Fender | Katty, its another name for coriander |
| 21:55.05 | Katty | i can smoke anything.... in an oven. |
| 21:55.06 | Corydon76-dig | How about banana peels? |
| 21:55.07 | _x86_ | Corydon76-dig: just like some other nice herbs... |
| 21:55.14 | Katty | [TK]D-Fender: uhmmm, coriander is brown. |
| 21:55.18 | funxion | other than immediate=yes how can I get * to forward the channel to s,1 without disturbing my billsec |
| 21:55.24 | Corydon76-dig | Katty: the smoke alarm is NOT a cooking timer |
| 21:55.27 | putnopvut | Katty: Cilantro is the mature coriander seed. |
| 21:55.28 | Katty | [TK]D-Fender: a very fine powder. NOT an herb :P |
| 21:55.29 | [TK]D-Fender | Katty, http://en.wikipedia.org/wiki/Cilantro <------- |
| 21:55.37 | Katty | argh. |
| 21:55.39 | putnopvut | The plant it grows into. |
| 21:55.40 | Katty | but how does it taste! |
| 21:55.47 | Katty | or is it just decoration :P |
| 21:55.49 | putnopvut | Katty: god awful! |
| 21:55.50 | _x86_ | like chicken, of course |
| 21:55.56 | putnopvut | I HATE cilantro |
| 21:55.59 | Katty | _x86_: /bonk |
| 21:56.06 | [TK]D-Fender | Katty, its all in there... |
| 21:56.17 | _x86_ | omgwtfbbq i just got bonked by a chick... i gotta tell my wife! |
| 21:56.27 | Katty | is it that metallicy herb they put into mexican stuff? |
| 21:56.29 | [TK]D-Fender | Katty, coriander is a key component of curry. Rather musky |
| 21:56.33 | Katty | [TK]D-Fender: yes, but wiki is not interactive enough. |
| 21:56.34 | putnopvut | Katty: bingo! |
| 21:56.39 | Katty | eww :< |
| 21:56.43 | Katty | i think i'm allergic to that stuff. |
| 21:56.59 | [TK]D-Fender | Katty, your digenstion-fu is weak! |
| 21:57.07 | gremzoid | is there a descent asterisk manual anywhere? or do i have to configure things at best guesstimate based on the crap outdated documentation i've read from 6 different websites? |
| 21:57.17 | [TK]D-Fender | ~book |
| 21:57.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 21:57.19 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 21:57.23 | [TK]D-Fender | gremzoid, there |
| 21:57.24 | gremzoid | ... and i thought siemens manuals where crap |
| 21:58.47 | _x86_ | Hollerith, v.: What thou doest when thy phone is on the fritzeth. |
| 21:58.55 | blitzrage | yay... someone updated the ~book link |
| 21:59.01 | _x86_ | ~book |
| 21:59.02 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 21:59.18 | _x86_ | omg i just bought 1st edition like 3 months ago! |
| 21:59.47 | gremzoid | yey! downloading a 20mb pdf on 33.3k! |
| 21:59.47 | [TK]D-Fender | _x86_, Procrastination : The art of keeping up with yesterday |
| 22:00.07 | _x86_ | *nod* |
| 22:00.31 | [TK]D-Fender | gremzoid, cry me a river.... so I can hold your head under :D |
| 22:00.34 | _x86_ | interesting, ORA has a T1 Survival Guide |
| 22:01.12 | blitzrage | indeed |
| 22:01.47 | carrar | I am sure a T1 cable could come in handy in a servival type situation |
| 22:01.52 | carrar | survival |
| 22:02.42 | _x86_ | that sentance was all kinds of lame |
| 22:03.07 | _x86_ | there is no such thing as a "T1 cable" heh |
| 22:03.17 | gremzoid | [TK]D-Fender, lets just say my exp with * documentation over the last 2 days is starting to piss me off |
| 22:03.30 | carrar | We there certainly is not a DS1 cable |
| 22:03.31 | _x86_ | gremzoid: read Teh Book |
| 22:03.42 | carrar | T1 there is |
| 22:03.45 | _x86_ | carrar: DS1 == T1 |
| 22:03.52 | carrar | better read that book |
| 22:04.11 | _x86_ | carrar: Digital Signalling level 1 == T1 with digital signalling |
| 22:04.25 | blitzrage | gremzoid: if that's the case, then you better toughen up if you want to use Asterisk :) |
| 22:04.28 | [TK]D-Fender | gremzoid, http://www.jeremy-mcnamara.com/index.php/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 22:04.32 | funxion | without using immediate yes on a cas t1 how can I get asterisk to just pick up and forward call to s,1 in the groups context without giving dialtone |
| 22:04.33 | Servergod | anyone gotten far with drbd and heartbeat with 2.3.0.2? |
| 22:04.45 | [TK]D-Fender | gremzoid, Decent minimalistic quicky-setup |
| 22:04.46 | _x86_ | carrar: a T1 can ride over a lot of different types of cable, from cat3 all the way to cat7, i've even see T1's run on fiber [sic] |
| 22:04.52 | gremzoid | [TK]D-Fender, i know how to configure... it's just the lack of documentation on more advanced things |
| 22:04.56 | _x86_ | Servergod: wrong channel |
| 22:04.59 | gremzoid | like configuring from sql... |
| 22:05.09 | Servergod | whoops srry |
| 22:05.10 | [TK]D-Fender | funxion, I didn't say to remove that... I said knock off the ANSWER as step 1..... |
| 22:05.19 | carrar | like I said |
| 22:05.21 | carrar | cable |
| 22:05.24 | *** part/#asterisk Servergod (n=Maverick@70.97.159.120) |
| 22:05.31 | [TK]D-Fender | gremzoid, that is advanced stuff.... walk before you run |
| 22:05.44 | _x86_ | carrar: like i said, there is no definitive "T1 cable" |
| 22:05.55 | carrar | Did I say cat5? |
| 22:05.57 | carrar | cat3 |
| 22:05.58 | carrar | fiber? |
| 22:06.06 | funxion | TK I dont have answer |
| 22:06.13 | _x86_ | carrar: you lose ;) |
| 22:06.16 | funxion | just immediate yes in zapata |
| 22:06.22 | carrar | You miss interpided |
| 22:06.35 | _x86_ | you misspeelded |
| 22:06.39 | _x86_ | ;) |
| 22:06.40 | funxion | its padding my billsec by like a minute per call |
| 22:06.57 | [TK]D-Fender | funxion, pastebin it |
| 22:07.08 | funxion | what default context of zap |
| 22:07.12 | funxion | of=or |
| 22:07.18 | gremzoid | [TK]D-Fender, errr it would still be nice to have documentation for it |
| 22:07.19 | [TK]D-Fender | funxion, the one thats used |
| 22:07.27 | [TK]D-Fender | gremzoid, BOOK <------ |
| 22:08.47 | gremzoid | [TK]D-Fender, slowly wgeting.... :P |
| 22:09.35 | funxion | [TK]D-Fender http://pastebin.com/d3fb57148 |
| 22:10.26 | *** part/#asterisk Braxus (n=bhsieh@66.147.214.164) |
| 22:12.05 | [TK]D-Fender | funxion, Ah, funny lookin' script you got there! ;) |
| 22:12.21 | [TK]D-Fender | funxion, Lemme guess.... the system dialing in passes that in-line, doesn't it? |
| 22:12.43 | [TK]D-Fender | funxion, the pin+*, right? |
| 22:13.47 | [TK]D-Fender | funxion, 1,2,4 are wasted priorities.... |
| 22:14.08 | funxion | I realize that |
| 22:14.18 | Qwell | use n |
| 22:14.30 | Qwell | and don't use the MYSQL function. Use func_odbc |
| 22:14.36 | Qwell | it's far better |
| 22:14.47 | [TK]D-Fender | funxion, but keep in mind that yoru input script answers the call pretty much immediately... so passing the PIN = answered call... |
| 22:14.49 | funxion | this is really old code |
| 22:14.55 | [TK]D-Fender | Qwell, he's on **1.0** |
| 22:15.07 | funxion | I'm jsut trying to adapt it to an analog environment |
| 22:15.10 | Qwell | upgrade |
| 22:15.14 | funxion | I plan to |
| 22:15.20 | funxion | but havent had the chance |
| 22:15.27 | funxion | going to start next week |
| 22:15.36 | [TK]D-Fender | funxion, but the issue is that the call is answered by that script... we'd need to manipulate the billing CDR |
| 22:16.00 | [TK]D-Fender | funxion, starinput = answer :/ |
| 22:16.16 | [TK]D-Fender | funxion, not sure how to reset CDR billsec in 1.0 |
| 22:16.32 | funxion | whats weird is I use this same context with a pri and I get the correct billsec |
| 22:17.00 | [TK]D-Fender | funxion, billsec should = duration - 1s (the wait) pretty much... |
| 22:17.53 | funxion | I need billsec to = the time from when * passes the call out and is answered to hang up |
| 22:18.15 | funxion | not the begining portion where authentication is processed |
| 22:19.34 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 22:21.26 | funxion | the call flows from channel bank => * => carrier |
| 22:21.40 | funxion | need billsec to be the answered duration between * and carrier |
| 22:22.00 | funxion | this works with PRI |
| 22:22.06 | funxion | but not with cas t1 |
| 22:22.10 | funxion | I dont get it |
| 22:22.51 | *** part/#asterisk gremzoid (n=gremzoid@d58-111-173-16.rdl5.qld.optusnet.com.au) |
| 22:23.10 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581883.dsl.bell.ca) |
| 22:23.46 | [TK]D-Fender | funxion, I think the issue is that with E&M + that script, the answer counts as a channel answer as opposed to an offset by using Dial before having answered in * |
| 22:27.15 | funxion | know of any work aournd? |
| 22:27.18 | funxion | around |
| 22:28.05 | *** join/#asterisk tripps (n=ss@116.sub-70-216-117.myvzw.com) |
| 22:28.42 | [TK]D-Fender | funxion, with it having to dial in like that you can either calculate the time it takes to process, and back that out, or perhaps upon completion dial a Local channel which will set an account code and use THAT call for billing. |
| 22:29.05 | [TK]D-Fender | funxion, I suspect the latter would be more accurate and usable. |
| 22:29.22 | [TK]D-Fender | funxion, exten => s,34,Goto(${EXTEN},4) <-- replacing this |
| 22:33.20 | *** part/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 22:37.19 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-a339e02d74e16ce4) |
| 22:37.19 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 22:37.47 | dan__t | I would whore myself out for a week straight if I could find a dumbed up Polycom GUI config tool. |
| 22:37.53 | dan__t | That sounds like a project... |
| 22:39.37 | devel | greetings all. anybody here who uses realtime in the dialplan? |
| 22:44.25 | CrazyTux | Anyone here done polycom provisioning before? |
| 22:44.29 | CrazyTux | HTTP preferably |
| 22:45.35 | Strom_C | crazyTux: I do it all the time |
| 22:45.57 | CrazyTux | Strom_C, any TIPS? |
| 22:46.08 | Strom_C | crazytux: any specific questions? |
| 22:47.02 | [TK]D-Fender | dan__t, There is a plenty-good guide on the WIKI actually... |
| 22:47.33 | dan__t | Just started going through that one again. |
| 22:47.45 | CrazyTux | Strom_C, I want to keep it simple like linksys/sipura, simply point to a URL i.e. http://somehost/$MAC, or some what, and then that way I can auto (parse/create) a file for the polycom to dl and provision, is this possible? |
| 22:48.06 | Strom_C | i'm sure it is |
| 22:49.11 | [TK]D-Fender | dan__t, my rates are very acccessable ;) |
| 22:49.21 | dan__t | I'm sure they are. |
| 22:49.33 | dan__t | I won't learn anything if I don't do it myself. |
| 22:49.35 | dan__t | I'm sure you understand. |
| 22:52.48 | [TK]D-Fender | dan__t, I do.... you don't want help, aren't "getting it", and are merely here to vent your spleen on us ;) |
| 22:54.38 | CrazyTux | Strom_C, what method do you use all the time? |
| 22:54.47 | Strom_C | crazytux: tftp |
| 22:58.58 | [TK]D-Fender | crazyTux : there are Polycom auto config creators & GUI's out there already. You can use them to provision a number of ways |
| 22:59.14 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:59.17 | [TK]D-Fender | crazyTux: ftp, ftps, tftp, http, https |
| 22:59.22 | CrazyTux | [TK]D-Fender, mind recommending one? |
| 22:59.27 | CrazyTux | [TK]D-Fender, I just want something quick and dirty |
| 22:59.35 | CrazyTux | Thus I don't deal with polycoms much. |
| 23:00.38 | [TK]D-Fender | crazyTux : I think trixbox comes with one, check that out. SIPX does as well. I've never used ANY of them personally. |
| 23:00.52 | [TK]D-Fender | crazyTux: I do all mind from scratch |
| 23:01.19 | CrazyTux | [TK]D-Fender, I think its easier from scratch, but want simple, HTTP provisioning. |
| 23:01.35 | CrazyTux | Trixbox has one, but not really my cup of tea. |
| 23:01.46 | CrazyTux | I think I'm just going to end up doing the research and making something quick |
| 23:02.09 | [TK]D-Fender | crazyTux: sorry, can't offer more there... how many do you have to prepare? |
| 23:02.55 | CrazyTux | [TK]D-Fender, not even alot, just 25 or so |
| 23:04.31 | [TK]D-Fender | crazyTux, not a huge task.... |
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| 23:28.59 | d4rkf1br | ? If all i need is a purely voip config of asterisk then I wouldn't need to install the libpri or zaptel packages right ? |
| 23:29.16 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-a339e02d74e16ce4) |
| 23:31.46 | devel | correct, d4rkf1br |
| 23:33.39 | *** join/#asterisk d4rkf1br (n=d4rkf1br@ip68-226-154-34.lf.br.cox.net) |
| 23:33.58 | d4rkf1br | thx devel |
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| 23:47.32 | tzafrir_home | d4rkf1br, libpri: sure. Zaptel might be useful as a timing source |
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