00:00.15 | pipwerk | I guess actually correctly interpreting what you read is harder for some then for others |
00:01.14 | *** join/#asterisk rnovotny22 (n=ro085181@h460da7b0.area2.spcsdns.net) |
00:03.16 | Strom_M | pipwerk: what we need is drool-proof asterisk |
00:03.39 | Strom_M | so simple, even the totally wasted who can't even order at Denny's can set it up! |
00:04.19 | Strom_M | holy god no |
00:04.46 | Strom_M | that guy is just way too queer even for me |
00:04.46 | pipwerk | I wouldn't know what denny's is, but I get the point |
00:04.59 | Strom_M | pipwerk: 24 hour greasy spoon diner type place |
00:05.03 | pipwerk | and yes, * was harder that I expected when I got started |
00:05.15 | Strom_M | the menu has many pictures so you can order just by pointing if you're so inclined |
00:05.21 | pipwerk | :) |
00:05.41 | file | Strom_M: if you are ever in HSV and Robinson invites you out for a midnight Denny's group gathering... go insane |
00:05.49 | pipwerk | as I said, I get the point |
00:06.12 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
00:06.12 | *** mode/#asterisk [+o anthm] by ChanServ |
00:06.19 | pipwerk | time for bed, later |
00:06.34 | Strom_M | later |
00:06.47 | Strom_M | file: I will indeed do that |
00:06.58 | Strom_M | seeing as how i'm gonna be in lolabama in a few weeks... |
00:08.57 | [TK]D-Fender | Strom_M, I take it he spammed your in PM then gave only you the pastbin? |
00:09.34 | Strom_M | yes |
00:09.47 | Strom_M | how did you know ?!?!?!??!!?!?!?!?!?!?!!!!!!!!!!!!!?!?!!?!??!?!?!?!!?!!!?!?1//11/1/1/1 |
00:09.56 | *** join/#asterisk rnovotny22 (n=ro085181@h460da7b0.area2.spcsdns.net) |
00:10.39 | rnovotny22 | Anyone have any ideas where to start looking for an intermittant no receive audio problem using voip? |
00:10.56 | Strom_M | the router |
00:11.01 | Strom_M | the network |
00:11.04 | Strom_M | the pbx |
00:11.07 | Strom_M | the phone |
00:11.19 | rnovotny22 | the provider? |
00:11.28 | Strom_M | sure, that too |
00:12.06 | rnovotny22 | That narrows it down a little. |
00:12.43 | denon | the user :) |
00:13.16 | rnovotny22 | Well, I think I can rule that one out. |
00:13.28 | denon | well, if you have to ask ... |
00:13.31 | denon | I wouldn't rule that one out :) |
00:13.48 | Strom_M | never rule out the user |
00:13.56 | sevard | OMG LOL BOOM |
00:14.05 | Strom_M | hahahahahaha |
00:14.19 | Strom_M | sevard: did you just see the biggest explosion evar>? |
00:14.58 | sevard | i've regressed into my 12 year old state, i'm currently hiding in my treehouse with crates of ak47 ammo and viagra |
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00:29.19 | [TK]D-Fender | sevard, s'ok... you have only 1 year to take back ;) |
00:30.53 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
00:33.49 | sevard | OOOOOOO there's TK with the really well thought out and very prompt insults. |
00:34.05 | sevard | man, they have so much DEPTH, they just cut DEEP |
00:42.27 | [TK]D-Fender | The Wiltshire Wit strikes again! Never dulls! |
00:43.12 | centrex | wow, I thought the asterisk documentation was pretty bad, but freespeech just doesn't have any |
00:43.19 | centrex | freeswitch |
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00:49.24 | anthm | why don't you go to that channel and complain! |
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01:02.08 | adeeln | hi, i'm running asterisk 1.2.17 and i can't seem to get my sip clients to use any zap devices, and vice versa |
01:02.34 | harlequin516 | adeeln: Why not? |
01:02.47 | harlequin516 | What happens? |
01:02.51 | adeeln | harlequin516: i don't know...it just won't let me dial out on it |
01:02.53 | harlequin516 | Wheres the log? |
01:02.57 | adeeln | let me pastebin it |
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01:03.44 | guille1983 | hi |
01:03.53 | Strom_M | better yet, pastebin sip.conf and extensions.conf |
01:03.59 | Strom_M | I bet you forgot an include :) |
01:04.13 | guille1983 | i would like to know important companies that are using asterisk... would you please tell me some of them ? |
01:04.33 | denon | cisco, avaya, the white house |
01:04.46 | denon | pretty sure the FBI uses it |
01:04.52 | denon | I think ATT runs it |
01:05.07 | guille1983 | denon: really? white house and FBI ? |
01:05.12 | MoutaPT | oh really white house?:P |
01:05.13 | denon | well, no |
01:05.15 | denon | but that'd be nice |
01:05.16 | harlequin516 | Anyone know how to play a .ulaw file from the linux commandline? |
01:06.03 | guille1983 | i need to prove who are using * |
01:06.12 | Strom_M | guille1983: why? |
01:06.23 | denon | he's in sales |
01:06.26 | denon | trying to convice a manager |
01:06.34 | denon | or owner |
01:06.38 | guille1983 | Strom_M: an assigment in my class |
01:06.45 | denon | haha |
01:06.51 | denon | homework help in #asterisk |
01:06.58 | denon | never thought I'd see the day |
01:07.08 | guille1983 | :D now you see it |
01:08.53 | adeeln | here's the sip.conf file http://www.pastebin.ca/586790 |
01:09.16 | Strom_M | oh fucking hell, not trixbox again |
01:09.23 | Strom_M | ~trixbox |
01:09.23 | jbot | somebody said trixbox was a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
01:09.55 | harlequin516 | adeeln: We want to see your CLI output or logs from your asterisk server. |
01:10.09 | adeeln | harlequin516: i'll paste that as well |
01:10.15 | Strom_M | harlequin516: it's trixbox; don't bother trying. :) |
01:11.17 | Strom_M | you'll only get a headache |
01:11.19 | harlequin516 | adeeln: HAHAH. OSI |
01:13.26 | adeeln | osi? |
01:14.59 | adeeln | harlequin516: http://www.pastebin.ca/586795 is the cli output...when i call from zap to sip, it rings the sip phone, but drops when i pick it up...calling from sip to zap, nothing happens |
01:18.33 | MoutaPT | zap show channels |
01:18.40 | MoutaPT | what it gives u? |
01:20.01 | adeeln | zap show channels -- http://www.pastebin.ca/586800 |
01:21.48 | adeeln | sip to zap call -- http://www.pastebin.ca/586801 |
01:23.42 | harlequin516 | I don't really understand how/why you have 12 Zap channels? |
01:23.50 | harlequin516 | Whats your hardware? |
01:24.14 | adeeln | opvxa1200 .... it's a 12 port card |
01:24.19 | adeeln | 3 fxo, 9 fxo |
01:24.25 | adeeln | err 9 fxs |
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01:25.34 | adeeln | everything used to work great, but i recently migrated machines, and then upgraded asterisk from 1.2.15 to 1.2.17 and then this started happening |
01:25.54 | Strom_M | adeeln: that's your punishment for using that horrible gui |
01:26.08 | adeeln | Strom_M: any other recommendations for a gui? |
01:26.19 | Strom_M | vim |
01:26.33 | adeeln | haha vim really isn't a gui |
01:26.50 | adeeln | although gvim is, but that's irrelavent |
01:26.55 | Strom_M | what the hell do you need a gui for, anyway? :) |
01:27.05 | adeeln | well, it used to make my life easier |
01:27.29 | adeeln | and it's a headless machine |
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01:28.03 | adeeln | an http interface was supposed to be the easiest for us |
01:28.26 | Strom_M | ssh was too complicated or something? |
01:28.46 | adeeln | no, but i don't always have an ssh client on the machines i'm using |
01:29.00 | adeeln | ssh is setup, and that's how i do debugging if i need to, but i normally don't need to |
01:29.21 | Hymie | put a copy of putty on an internal website |
01:29.29 | Hymie | then you can download it to any windows box in 1 second |
01:29.33 | Strom_M | put a copy of putty on a thumbdrive |
01:29.39 | Hymie | heh, or that too |
01:29.44 | Strom_M | or hell, just download it from their website every time |
01:30.22 | adeeln | because i'm rarely at my desk |
01:30.34 | adeeln | or in the office for that matter |
01:30.48 | adeeln | http also allows me to remote admin from my phone |
01:30.54 | Strom_M | sounds like you need...a laptop |
01:31.03 | adeeln | its in the shop |
01:31.15 | Hymie | that's it's, but anyhow.. look |
01:31.16 | Hymie | just |
01:31.17 | Hymie | download |
01:31.17 | Hymie | putty |
01:31.18 | Hymie | and |
01:31.19 | Hymie | stop |
01:31.20 | Hymie | complaining ;) |
01:31.26 | Strom_M | welcome to the excuses hour, with our special guest adeeln |
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01:31.29 | adeeln | but you guys think it's a config issue? |
01:31.32 | Hymie | it takes 10 seconds to download |
01:31.37 | Strom_M | adeeln: it's a freepbx issue |
01:31.44 | Strom_M | toss that crap and build a real asterisk system |
01:32.15 | adeeln | i also don't hear any prompts being played...like *69 will tell me the digits, but there are some other voice prompts that don't work anymore |
01:32.21 | adeeln | that a freepbx or asterisk issue? |
01:32.21 | Hymie | er |
01:32.27 | Hymie | that's broken, they should all work or none |
01:32.38 | adeeln | the logs say they're playing, but i don't hear anything |
01:32.49 | Hymie | do they all not work, or some don't work |
01:32.55 | Hymie | on the same call |
01:33.07 | adeeln | all the digit related ones work, but anything like 'please enter your pin' don't |
01:33.23 | Hymie | well, that does sound like a messed up dialplan |
01:33.35 | Hymie | internal asterisk apps seem to find them |
01:33.39 | Hymie | but the dialplan doesn't seem to |
01:33.57 | adeeln | is there a need to compile the alsa/oss module for asterisk? |
01:34.13 | Hymie | some of the sounds work |
01:34.18 | Hymie | therefore, it's not that |
01:34.20 | Hymie | and no |
01:34.24 | Hymie | no oss needed |
01:34.41 | Hymie | what phones are you using, just curious |
01:34.50 | Hymie | not really related to your prob |
01:34.53 | adeeln | what's the use for the alsa module? transcoding? playback on the local machine through speakers? |
01:35.06 | Hymie | yes, your local box can become an extension |
01:35.24 | Hymie | alsa isn't needed for transcoding |
01:35.34 | adeeln | well we have a bunch of analog cordless phones plugged into the zap channels, 2 grandstream bt-120's, & a bunch of different soft phones (x-lite, phonerlite, etc) |
01:36.06 | Hymie | interesting collection |
01:36.06 | Hymie | well |
01:36.12 | Strom_M | sounds like you think you can get away with a business-grade PBX for $20 |
01:36.16 | Hymie | I've never used anything buy pure asterisk |
01:36.21 | Hymie | but |
01:37.16 | Hymie | anyhow, it sounds like your install is borked, but.. I don't know if you know enough to debug it, if you get what I mean |
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01:37.24 | Hymie | it might be better for you long term, to learn asterisk |
01:37.32 | Hymie | save you time over 10 years |
01:37.41 | adeeln | Strom_M: i'd hardly say that's the case...this is more of a proof of concept machine; fax doesn't work reliably through asterisk (at least i've had a hell of a time getting it to work) and we're trying to figure out what combination is right for us |
01:38.04 | Strom_M | adeeln: i'm surprised that openvox crap works at all, tbh |
01:38.06 | Hymie | adeeln: I can't say enough good about the polycoms |
01:38.25 | adeeln | there's no point in spend 10,000 on a demo system |
01:38.28 | Strom_M | adeeln: fax is a sensitive one; you need to set things up in specific ways |
01:38.40 | Strom_M | grandstream is crap |
01:38.52 | adeeln | the grandstreams are crap, and we're aware of that, but if the crap works well, then the expensive stuff will be worth the investment |
01:38.53 | Strom_M | poor build quality, poor feature set, poor reliability |
01:39.42 | adeeln | anyways, thanks for the help |
01:39.45 | __DAW | good or bad, you should test with what you will be using. Things vary greatly between phones. |
01:40.46 | Strom_M | "how to play ostrich" by adeeln |
01:41.04 | __DAW | :) |
01:41.57 | Hymie | http://toronto.ctv.ca/servlet/an/local/CTVNews/20070623/contaminated_chocolate_070623/20070623?hub=TorontoHome |
01:42.09 | Strom_M | OH NO NOT THE CHOCOLATE |
01:42.13 | Hymie | my god.. to steal chocolate bars being disposed of an reselling them? |
01:42.17 | Strom_M | ANYTHING BUT THE CHOCOLATE |
01:42.27 | Hymie | so little kids die, I mean, come on, that's freaking beyond low |
01:42.37 | Strom_M | Hymie: people are greedy, cheap, and stupid. you should know this by now. |
01:42.39 | Hymie | I bet those guys go beat up midgets for weekened entertainment |
01:43.11 | Hymie | Strom_M: the day I become unastonished by such horrid greed, is the day that my morality changes |
01:43.16 | Hymie | and, that is not acceptable |
01:43.29 | Hymie | the astonishment, the anger, it remains in the face of such horrid actions |
01:43.48 | Hymie | that does not mean I do not know about it... merely that I am still angered by it |
01:43.52 | Hymie | as you should be |
01:44.01 | Strom_M | so you'd rather live in fantasy land and believe that people aren't greedy and selfish and moronic? |
01:44.30 | Hymie | that statement makes no sense, based upon what I've just said to you |
01:44.43 | Hymie | I already stated I realise that such people exist, this is not news to me |
01:44.58 | Hymie | I simply refuse to accept that behaviour as something that does not anger me |
01:45.02 | Strom_M | and what has this got to do with phones, anyway |
01:45.13 | denon | Hymie: I agree. |
01:45.19 | Strom_M | please check the number and dial again |
01:45.21 | Hymie | Strom_M: maybe someone spread that chocolate onto a polycom before you received it! |
01:45.34 | Strom_M | or ask your operator for assistance |
01:45.39 | Strom_M | delicious, poisonous assistance |
01:45.45 | Strom_M | this is a recording |
01:45.49 | Strom_M | 213-4 |
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01:50.16 | __DAW | Any major problems with 1.4.5? |
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02:29.38 | paragonc | do SMS chargers only effect the end user, or are there origination fees from VoIP providers ? |
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03:31.25 | ez` | i need a solar satelite voip phone |
03:34.26 | JT | why on earth would you want it to be voip? |
03:35.50 | coppice | he doesn't want it to be on earth |
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03:41.43 | plasmid | I setup extension 5 on my IVR to be emergency which should ring all phones. I created a ringroup called "emergency" which should dial all my extensions: 1,2,3,4 .. Is the correct syntax in the extensions box: 1# 2# 3# 4# OR 1#,2#,3#,4# ? |
03:41.53 | plasmid | extensions 3 and 4 are misc destinations = my cell and my partner's cell. |
03:42.18 | JT | plasmid: this channel is for asterisk support |
03:42.26 | plasmid | JT: I know. |
03:42.43 | JT | your problem does not sound like an asterisk problem |
03:42.51 | plasmid | I was hoping the underlying foundation is similar. |
03:43.07 | JT | freepbx? |
03:43.28 | JT | well asterisk doesn't have an "extensions box" for one |
03:43.35 | plasmid | On asterisk, how would u setup a ring group to ring all all extensions? |
03:44.16 | JT | the dial command can dial multiple extensions at once |
03:44.23 | plasmid | I don't think it's a problem per se, but more a syntax issue. |
03:44.44 | plasmid | this dial command you are reffering to... what is the file to modify? |
03:44.48 | JT | ~freepbx |
03:45.59 | jbot | freepbx is probably unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
03:45.59 | JT | the dialplan, extensions.conf |
03:45.59 | plasmid | so in extensions.conf the dialplan is called say: emergency and the syntax should be 1# followed by a comma or a space or? |
03:46.04 | JT | ~thebook |
03:46.05 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
03:46.16 | plasmid | yes...yes. I am aware of that. |
03:46.16 | JT | you need to read the book |
03:46.22 | JT | especially of the dial command |
03:46.30 | JT | you are? then you should have no problems |
03:46.43 | plasmid | no i meant the referral. |
03:47.11 | plasmid | I dont use the IVR that much so I was hoping to get a quick solution to the matter. |
03:47.24 | plasmid | then again I should peruse through the documentation once more |
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04:03.09 | ech0dish | hi |
04:05.37 | ech0dish | im a lil curious about asterisk. i've been trying to read a lil about it but its kinda confusing. i was wondering if this could be used to turn my home phone line into a voip that i can use on other computers on the net? |
04:06.01 | ech0dish | to make pc to land line calls? |
04:06.47 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
04:09.01 | ech0dish | cause i got a laptop and a desktop... it would be cool if i could use my desktop as a voip server connected through my modem to my home phone and then make calls from my laptop at whatever wifi spot and it would show up as if i was calling from my home phone on the caller id? |
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04:14.16 | fx0 | not cool |
04:14.16 | ech0dish | ? |
04:15.05 | ech0dish | not cool what? |
04:15.32 | JT | zeros in nicks ;) |
04:16.01 | ech0dish | k |
04:16.19 | ech0dish | hi, u know about asterisk any? |
04:17.04 | JT | do I know any asterisk? Yes. |
04:18.03 | coppice | I know some asterisk. I know "rm -rf /*" |
04:18.26 | ech0dish | sigh |
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04:18.56 | JT | asking dumb questions doesn't get people far, who knew? |
04:19.21 | coppice | Please, please, I know this one. I know this one! |
04:21.29 | plasmid | If my ring group is "5" (where it will ring all my extensions: 1,2,3,4) do I have to modify the dialplan on those extensions and put callgroup = 5? |
04:26.00 | plasmid | ack.. nv. Callgroup is for picking up an extesion if the other doesn't pick up by punching for example *8 |
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04:40.26 | kusznir | Hi all: I'm trying to debug an asterisk problem that claims to be call routing, but I don't understand how it can be such. I'm running asterisk 1.2.16 on openwrt linux, configured by hand. I just rebuilt my system and copied the config files that used to work back. |
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04:41.04 | kusznir | Now, whenever I attempt an IAX call, I get "unable to create channel of type 'IAX2' (cause 3 - no route to destination). internal sip-sip calls work fine. |
04:42.19 | kusznir | But it shows it has sucessfully registered to the remote host, and the line before it shows it is trying to pass the call to the correct host. The only thing I see is an error setting CALLERID a few lines before. Could that cause the "no route" error? |
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04:53.49 | J4k3 | B |
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05:14.57 | JT | kusznir: no, it couldn't |
05:15.56 | kusznir | Yea, I've been troubleshooting that one, and that's fixed. I think there's an issue with IAX peers. I'm at a loss on how to debug this further, though. |
05:16.26 | kusznir | iax2 debug shows registrations (about every 15 sec, I think) on all 3 of my iax peers. Yet iax2 show peers shows them all as "UNMONITORED". |
05:16.48 | kusznir | reload shows "registered to peer..." messages, which all look right. |
05:17.13 | kusznir | I have verified I can receive an outside call via SIP, but I cannot receive calls via IAX. |
05:19.03 | b1shop | my whole config is hosed. |
05:19.07 | Strom_M | what does 'iax2 show registry' show? |
05:19.17 | JT | qualify is not set to yes if they're unmonitored. |
05:19.24 | b1shop | i can call ext 12 from ext 10. but ext 10 cant call 12 |
05:19.26 | b1shop | etc |
05:19.34 | JT | it has no relation to registration |
05:20.25 | kusznir | so is that an option in the peer stanza in iax.conf? |
05:20.35 | Strom_M | kusznir: what does 'iax2 show registry' show? |
05:20.36 | kusznir | show registry shows 3 peers registered. |
05:20.37 | JT | yes |
05:20.53 | kusznir | eg, 64.2.142.22:4569 kusznir 66.233.138.60:4569 60 Registered |
05:22.24 | b1shop | i bought a sangoma 200 card. i was pissed when i saw the ports son't accept standard phone line ends. how do you get around that? |
05:22.43 | coppice | they supply the cables you need |
05:22.50 | b1shop | yeah. but only 4ft |
05:23.10 | b1shop | wont reach from the punchdown to my rack |
05:23.28 | coppice | the strange plug is a headset plug, so they are easily available as parts, but not as leads |
05:23.49 | coppice | I understand this was a cockup by the developers, and not part of any master plan |
05:24.02 | b1shop | i have a crimper. just @ a loss as to where to buy the ends |
05:24.34 | coppice | sorry, handset plug, not headset plug |
05:24.53 | b1shop | yup. understand the end exactly |
05:25.10 | b1shop | aside from cutting the ends and splicing.. i have no idea what to do |
05:25.16 | kusznir | I have found small quantities available at Lowes and Home Depot. |
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05:25.43 | b1shop | kusznir: they cromp ok with a rj-45 crimper? |
05:25.47 | b1shop | crimp* |
05:25.58 | kusznir | I've seen them referred to as "4p4c" connectors in bulk from places like digikey. |
05:26.39 | b1shop | i'm kinda screwed. moved offices this weekend. was hoping to have new system built out by then |
05:26.43 | b1shop | not looking good |
05:26.46 | coppice | something that rolls of the toungue nicely, like that |
05:26.53 | kusznir | If the crimper has the insert for the smaller connector. There are only 4 pins, and the body doesn't have any "spare" locations, so the crimper die has to have exactly 4 pins. |
05:27.35 | kusznir | A lot of RJ-45 crimpers that also do RJ-11 can do the headset. I think I got a cheap one form Wal-mart years ago.... |
05:27.42 | b1shop | if anything can possibly go wrong. it has with this install |
05:28.10 | coppice | I have a headset crimper in the cupboard next to me, and a few plug. now, if you can arrange crimp over IP....... |
05:28.21 | b1shop | dcc? ;-) |
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05:29.31 | b1shop | i might be forced to try and make my old norstar system work |
05:30.03 | coppice | the reason I have a crimper which can do the headset plugs is because I also have some sangoma A200 cards :-) |
05:30.27 | b1shop | engineers screwed up and ran two phone and one data instead of 2 data and one phone. and they used old cat3.. so it's useless |
05:30.57 | b1shop | at&t ran the lines into the basement and the bldg engineer punched them to block in new office. but i can;t get dialtone off them |
05:31.28 | JT | what's wrong with cat3? |
05:31.36 | b1shop | and of course i bought POE phones... so no one will have PC connections due to lack of ports |
05:32.16 | syle | fuckit install wireless router and wireless cards or bitch at at&t |
05:32.30 | JT | wireless? evil |
05:32.46 | JT | PoE phones usually have a switch, with a PC port |
05:32.50 | syle | yeah i hate them to, i use it for laptop only |
05:33.11 | b1shop | yup. had to hack a wireless together |
05:33.29 | JT | b1shop: what phones did you buy? |
05:33.34 | b1shop | unfortunately. the polycom 320's i bought don;t have the extra ethernet port |
05:33.47 | JT | well you should've checked that first |
05:33.55 | JT | it's like an extra $10 for an ethernet port |
05:34.18 | b1shop | didn;t know iw as goign to be an issue. the install order was for 2-data 1phone. not the other way around |
05:34.38 | JT | so can't it be changed? |
05:35.16 | b1shop | sure. but they used old cat3 for phone. and it does't help me on mon morning |
05:35.39 | JT | cat3 can carry data |
05:36.01 | b1shop | plus. engineers seemed to have cut power for a 6 cubes in the office |
05:36.19 | b1shop | they have data. no power |
05:36.35 | kusznir | lock the switch port to 10Mbps and you'll be fine...two of my buildings were wired exclusively with Cat-3 due to stupid people working in the contracting office...(we can save $$$ by downgrading CAT-5 to CAT-3...nobody will notice....) |
05:36.38 | b1shop | JTcat3 do 100mb or only 10mb? i was wondering about that |
05:36.58 | JT | depends on length |
05:37.19 | JT | i've had success with 100Mbit/s on <50m runs |
05:37.21 | coppice | a lot of e-mail tells me that |
05:37.25 | kusznir | My experiences have been at about 10 feet, cat-3 start to get a lot of errors at 100Mbps. |
05:37.32 | b1shop | also have 2ton ac unit plugged into new server room. that blew a fuse tonight and my 5TB NAS box overheated |
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05:38.18 | b1shop | like i said. NOTHING with this move is working right |
05:39.37 | b1shop | my * install is working like crap also. |
05:40.13 | b1shop | provisioned phones with tftp. some can dial extensions... others cannot |
05:40.17 | b1shop | but they all ring |
05:40.27 | JT | eww tftp |
05:40.40 | b1shop | ftp better? |
05:41.13 | b1shop | this is my 1st install. so if you can do it wrong. i prob am |
05:41.17 | JT | ftp or http, or the secure versions |
05:41.29 | JT | tftp is not recommended by polycom even |
05:42.30 | b1shop | i tried ftp. created user/pass with the polycom defaults. it seemed to always fail to log in |
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05:43.40 | russellb | provisioning via ftp works great for me. |
05:45.01 | coppice | sucker |
05:45.20 | b1shop | lol |
05:45.36 | b1shop | anything is better then not having phones on mon morning |
05:46.00 | coppice | ah, the sense of peace sounds blissful |
05:46.10 | kusznir | So, any more thoughts on what could be causing my issue? (although its far less important than b1shop's....my * is for personal learning and experimenting) |
05:46.35 | b1shop | i'm thinkign about pissed off clients and loss of $ |
05:46.55 | b1shop | not like i don;t have 20 other things wrong with the move i could spend time on |
05:47.09 | coppice | if they can't call you, then were is the problem? :-) |
05:47.46 | coppice | cue Monty Python..... |
05:47.48 | coppice | "Always look on the bright side of life..." |
05:47.54 | b1shop | oh. i dunno. missed deadlines. no new business. pissed off clients that send new projects elsewhere, etc.. |
05:48.19 | coppice | there you go. focusing on the downside |
05:49.00 | b1shop | heh. i even went out to radio shack today and bought a $10 phone just to punch in so i had SOMETHING that rang. i still could not get dialtone |
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05:50.13 | coppice | see, that's the business you should be in. Those phones leave the factory at around $1, and you pay $10. |
05:50.28 | b1shop | odd thing is that my internet punched down fine |
05:51.10 | kclaussen | well you know shipping :) |
05:51.41 | coppice | if I really knew shipping, I'd be buying $1 phones and reselling them :-) |
05:52.05 | sevard | yeah, $1 phones, big business, lots and lots of money to be made there. |
05:53.20 | coppice | there's a lot more in that than buying $10 IP phones and reselling them at $50 |
05:54.42 | kusznir | So, I think I've traced a part of it down now...I'm pretty sure its coming back to CallerID (I think my peers dont' accept calls without CID info set or something). Anyway, I'm using: Set(CALLERID(number)=<my DID>), but I get: ERROR[7420]: pbx.c:1437 ast_func_write: Function CALLERID not registered. How do I fix this? |
05:56.20 | kusznir | b1shop: Well, a good chunk of the problems are not your fault: Insufficient wiring for the AC, data/phone wiring not to spec, insufficient power for cubes, etc. So even if asterisk did work, you'd still not be able to provide reliable service due to others failing to do their job properly. |
05:56.56 | b1shop | yeah. i know. |
05:57.05 | coppice | kusznir: wheta he is experiencing seems pretty normal for a move, so a lack of planning for it might well be his fault |
05:58.06 | b1shop | oh. some if it is def. my fault. murphy's law. week before move is always busiest with projects |
05:58.33 | b1shop | not to mention having a crappy admin |
05:58.45 | b1shop | her days are numbered.... |
05:58.55 | coppice | crappy admin == janitor? :-\ |
05:59.32 | b1shop | yeah. i can seem my interview requirements dropping now. |
05:59.53 | b1shop | Qiestion 1: can you actually make a decent pot of coffee? |
06:00.12 | coppice | Answer: No. I work at Starbucks |
06:00.30 | *** join/#asterisk jamon (n=jamon@ppp-70-225-71-139.dsl.covlil.ameritech.net) |
06:01.21 | b1shop | she got the requirements. it was her job to make sure everything go done |
06:01.46 | b1shop | she spent hrs over there last week. always came back and said everything looked great. |
06:02.09 | b1shop | she WAS right. it looks awesome.. nothing works... but ti looks sweet as hell |
06:02.15 | coppice | she probably referred to the decor :-\ |
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06:03.48 | b1shop | lol. they delivered new cubes thurs evening. all the surfaces are a nice slate grey. except for the end pieces. tyey are a funny blue color |
06:04.43 | jamon | I'm trying to write a script to automatically pickup inbound routes and send a DTMF 2 seconds after picking up, then wait 2 seconds, then transfer it to an extension... do I need to do an Answer() or something before my first Wait() and SendDTMF()? |
06:04.56 | *** join/#asterisk GothAlice (n=amcgrego@200.75.246.175) |
06:06.07 | GothAlice | I have a Nokia N95, fully updated, which is successfully registering with my Asterisk server. If I attempt to make a local-extension Echo test call (ext. 613) the connection is made, Asterisk begins playback of a "this is a test" message then switches to the echo test, but I hear nothing on my Nokia. I've tested every option for the NAT= option, to no avail. |
06:07.17 | GothAlice | Neermind! |
06:07.28 | GothAlice | Had the wrong IP on the "externip" option! |
06:07.33 | jamon | :) |
06:07.43 | jamon | i had that happen with all my inbound calls the other day :p |
06:07.57 | jamon | i'm going to write a script to auto update it for me :p |
06:08.29 | jamon | wow, that's nice :P |
06:08.32 | jamon | now i'm jealous |
06:09.33 | GothAlice | Trust me, the N95 -isn't- worth it. I just happen to have one because a local (cough: chinese) retailer gave me the runaround. Some things in Panama are good, some things are very, very bad. Electronics retailers are bad. |
06:09.58 | jamon | ah :p |
06:10.01 | jamon | thanks for the heads up :) |
06:10.15 | jamon | what are you doing to connect it to freepbx? |
06:10.21 | jamon | does it have a SIP client built in? |
06:10.25 | jamon | or are you running a java one or something? |
06:10.26 | GothAlice | The interface is non-intuitive to the max, the OS is slow and doesn't have enough RAM (get out of RAM errors all the time), the GPS doesn't work here (at all - it thinks I'm in Egypt), and the mic picks up all surrounding noise. |
06:10.33 | GothAlice | SIP built-in. |
06:10.45 | jamon | that's cool, if it worked, huh? :) |
06:10.54 | jamon | impressive featureset |
06:11.09 | jamon | that's a new one for nokia though... I've always found that their stuff usually doesn't have as many features, but is more stable |
06:11.11 | GothAlice | It's grossly expensive, the feel is of a $20 Linksys VoIP phone, not a $1K gold brick. |
06:11.14 | GothAlice | Etc. :^) |
06:11.30 | GothAlice | The settings to get SIP working are stupid. |
06:11.36 | jamon | of course I bought a Motorola Razr v3i the day they were sold in america for $300 + 2 yr contract |
06:11.38 | jamon | so I'm an idiot :P |
06:11.58 | GothAlice | Proxy: your IP, asterisk realm, username, password. Register server: asterisk is the host, no user, no pass, no realm. WTF? |
06:12.01 | jamon | that was before I realized it was a piece of garbage with a song limit via itunes and a USB 1.0 interface :) |
06:12.01 | b1shop | heh. i'll sell ya my old one!? |
06:12.08 | GothAlice | XD |
06:12.39 | jamon | who the hell creates a device with 512 meg of storage for music and puts it on usb 1? |
06:12.46 | jamon | someone should be kicked in the head for that |
06:13.05 | GothAlice | XD True. |
06:13.09 | b1shop | same ppl who put 50 song limit |
06:13.17 | jamon | it's 100 for v3i, but still, way way too low |
06:13.26 | jamon | so I just use the java based mp3 player instead |
06:13.28 | jamon | and let it drain my battery |
06:13.33 | coppice | jamon: you simply misunderstand the phone market. the goal is to piss off the customer |
06:13.46 | b1shop | i have v3i. mine caps @ 50 songs |
06:13.46 | jamon | apparently so |
06:13.51 | jamon | bishop: does it? |
06:13.55 | jamon | maybe I'm misremembering |
06:13.56 | b1shop | yup |
06:13.58 | coppice | if you make a music oriented phone, you give it a 2.5mm jack socket, or a proprietary row of pins |
06:14.00 | jamon | it's been a year since I used itunes |
06:14.13 | jamon | razr 2 looks much better |
06:14.17 | jamon | but i've already been burned once |
06:14.24 | coppice | if you have USB, you use a propritary row of gold pads that get too dirty to ever work |
06:14.40 | b1shop | i ditched my v3 for a blackberry |
06:14.49 | coppice | if you have something hot, like the tx PA, you put it right under their ear |
06:14.58 | jamon | hehe |
06:15.11 | jamon | though recently my v3i has been rebooting itself randomly |
06:15.15 | jamon | white screen for a second, then reboots |
06:15.28 | b1shop | sounds like i should sell ya mine! ;-) |
06:15.33 | coppice | the V3 sounds amazingly good, though |
06:15.42 | jamon | yeah, it's sound quality is great |
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06:15.55 | jamon | both music and talking |
06:15.57 | jamon | i'll give it that |
06:16.04 | jamon | peopel can't tell i'm on a cellphone usually |
06:16.15 | coppice | people complain a lot about codecs, but half the time its the phone at fault. The V3 shows what the GSM codecs can really do |
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06:17.27 | jamon | any of you ever use grandcentral? :) |
06:18.08 | jamon | it's a relaying thingy for phones... they give you a local number someplace, and it forwards to your other phones... or gizmo account |
06:18.17 | jamon | I've got mine setup to forward to gizmo,a nd gizmo to forward to asterisk :) |
06:18.32 | jamon | it actually doesn't sound bad... and it's a free DID anywhere in america pretty much =) |
06:18.57 | jamon | only problem is that when they call you... they play a stupid message and ask if you want to pickup the call |
06:19.07 | jamon | so i'm trying to get asterisk to answer it right away, and immediately accept the call |
06:19.15 | jamon | it works most of the time, but sometimes it doesn't |
06:19.34 | jamon | and it ALWAYS plays the first part of the message, even if it did work |
06:19.52 | jamon | which is very odd to me :p |
06:25.31 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
06:26.50 | jamon | anyone know what the timeout does on senddtmf()? |
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07:03.37 | kusznir | what causes asterisk to suddenly show all peers as "Host (Unspecified)" after a reload? |
07:03.46 | kusznir | (iax2 show peers command) |
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07:36.37 | waKKu | evening... |
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07:43.12 | rvhi0 | how to do multiple registration for one extension? |
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07:51.41 | rvhi0 | anyone here? |
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08:04.56 | Dovid | any one know where I can rent space on a SBC router or any recomendations for a good SBC router/swtich ? |
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08:45.41 | DeadlyData | is any one around that could help me? |
08:48.09 | tzafrir | ~anyone |
08:48.09 | jbot | *** anyone: No such nick/channel - and yes, there probably is someone, somewhere, who knows or runs it; that doesn't mean /I/ do. |
08:48.57 | tzafrir | DeadlyData, just ask your question |
08:50.13 | tzafrir | Dovid, rent space on a SBC? SBC as in Single Board Computer? |
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08:52.25 | JT | tzafrir: i'd say session border controller is much more likely |
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09:01.21 | Dovid | tzafrir: a router in a DC |
09:01.38 | JT | an sbc is not quite a router |
09:01.55 | Dovid | we are having lots of NAT issues as well as I understand from an SBC we can have it re-route the RTP stream even if behind NAT |
09:02.01 | Dovid | JT so SBC = ? |
09:02.25 | coppice | SBC == session border controller == VoIP box with VoIP box crossed out, SBC written in crayon, and the price increased at least 5X |
09:05.26 | JT | exactly |
09:05.30 | JT | sbcs are for suckers |
09:07.20 | coppice | SBCs are for innovators. people at the cutting edge. people who need enterprise scalability.... oh, sorry, you just said that |
09:08.17 | Dovid | hehe |
09:08.26 | Dovid | so an SBC wont help me at all ? |
09:09.00 | Dovid | my bigest issue is 1) NAT and 2) RTP Stream - I want it to by pass the server even if the calling client is behind NAT |
09:12.50 | JT | Dovid: no |
09:12.55 | JT | Dovid: you are dreaming |
09:13.11 | JT | if you don't want to deal with rtp traffic, get out of the voip business :) |
09:19.58 | syle | nat is not a big issue, but if you don;t want to handle rtp traffic, you gonna loose CDR records, so you can;t bill crap to anyone |
09:20.41 | JT | and end users may get nat problems too |
09:27.37 | Dovid | JT: why can astrisk just handle the set up and tear down of the call ? |
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09:29.26 | JT | because nat exists |
09:29.37 | JT | and i assume you want accurate billing data |
09:34.50 | DeadlyData | what would this error mean if I was attempting to connect to my asterisk system usng SIP, Rejected registration for 'CENSORED' with cause 'bearercapability notauth' |
09:43.46 | *** join/#asterisk DragoraN (n=dragoran@217.67.19.74) |
09:43.47 | DragoraN | hi |
09:44.08 | DragoraN | does this work? if not, how can i do this? exten => 20,1,SetGlobalVar(PARTY_EXT=${EXTEN}) |
09:44.26 | DragoraN | i want to save originally called extension to variable for using in other context |
09:44.37 | DragoraN | not context, sorry, on other extension |
09:44.46 | DragoraN | which i jump to with goto |
09:46.40 | DragoraN | please |
09:48.55 | Dovid | huh ? |
09:49.04 | Dovid | u want to set the exten of the person that is calling ? |
09:49.13 | Dovid | y would u do global ? |
09:49.26 | Dovid | u can use just Setvar and use it any where in the system on the same call |
09:49.54 | *** join/#asterisk YonahW-Work (n=YonahW-W@genie03-173-74.inter.net.il) |
09:50.58 | Dovid | Shalom Yonah |
09:56.51 | YonahW-Work | Shalom Dovid, How are you |
10:03.01 | Dovid | :) |
10:03.13 | Dovid | ani roeh she atah me-yisrael ? |
10:04.23 | YonahW-Work | I am (I believe this is an English only channel) |
10:04.28 | YonahW-Work | I see you are a local as well |
10:04.54 | Dovid | yup |
10:05.02 | Dovid | there is also #asterisk-il |
10:05.35 | YonahW-Work | did not know that! is it active at all? |
10:05.57 | Dovid | here and there |
10:06.06 | Dovid | not many of us on at once |
10:06.17 | YonahW-Work | so I see |
10:06.49 | YonahW-Work | I am on the mailing list as well but dont really see any action there |
10:07.19 | YonahW-Work | were you at the IsrAsterisk conference a few months back? |
10:07.38 | Dovid | nope. didnt know about it :( |
10:07.53 | Dovid | where was it ? and i asume it was hebrew only which I am not the best at |
10:08.02 | *** join/#asterisk YonahW-Work (n=YonahW-W@genie03-173-74.inter.net.il) |
10:08.14 | Dovid | where was it ? and i asume it was hebrew only which I am not the best at |
10:08.51 | YonahW-Work | It was in Petach Tikva at the IBM building and was mainly Hebrew with parts in English |
10:09.15 | Dovid | argh !!! |
10:09.22 | Dovid | when was it ? and how many people showed up ? |
10:09.29 | Dovid | I know VON is coming up - i hope to be at it |
10:11.06 | YonahW-Work | funnily enough I was there and don't remember when exactly it was |
10:11.23 | YonahW-Work | I want to say around April and there were probably 50-100 people |
10:11.29 | YonahW-Work | when is VON? |
10:11.33 | Dovid | http://www.von.com/israel07/ |
10:11.46 | Dovid | in oct. |
10:12.02 | Dovid | 14-15th of Oct. |
10:12.22 | Dovid | wanted to go to astricon in the US but it is on Sukkot |
10:12.25 | YonahW-Work | cool October is a good time for me |
10:12.29 | Dovid | same here |
10:12.30 | YonahW-Work | yeah I noticed that |
10:12.46 | Dovid | maybe in 2 years |
10:12.58 | Dovid | or i will go to the one in Europe |
10:15.23 | DragoraN | how can i capture entered digits? |
10:16.13 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
10:17.11 | YonahW-Work | DragoraN: check out the IVR page on the wiki http://www.voip-info.org/wiki-Asterisk+tips+IVR+menu |
10:17.31 | DragoraN | thx |
10:39.43 | *** join/#asterisk eatmypiano (n=eatmypia@host86-132-181-229.range86-132.btcentralplus.com) |
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11:03.26 | DragoraN | how to play music on hold just in background and let ringing the calling party? |
11:03.41 | DragoraN | when part will answer, the moh will stop.. |
11:03.53 | *** join/#asterisk X-Rob (n=rob-x@dsl-124-150-69-250.vic.westnet.com.au) |
11:07.27 | Dovid | anyone know how I can change the packetization time for g729 ? |
11:07.59 | YonahW-Work | Dragora: try looking into the Background cmd |
11:18.05 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
11:31.15 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
11:36.36 | DragoraN | YonahW-Work: thx |
11:50.06 | YonahW-Work | anyone know how I can figure out why 1.4.5 is not building chan_zap.so? |
11:50.34 | DrukenLPY | is your zaptel current? |
11:51.38 | YonahW-Work | zaptel 1.4.3 |
11:56.23 | tzafrir | is it installed? |
11:57.11 | YonahW-Work | yes, I am currently installing again |
11:57.46 | tzafrir | ./configure |
11:57.51 | tzafrir | once again... |
12:01.55 | YonahW-Work | lets see how it goes |
12:04.20 | YonahW-Work | ok compiled no errors and I see chan_zap.so |
12:04.34 | tzafrir_laptop | YonahW-Work, look at build_tools/menuselect-deps in the asterisk source tree |
12:05.08 | tzafrir_laptop | this is how menuselect figures out if a certain external dependency (E) is available |
12:05.15 | YonahW-Work | cool thanks for the tip, I will look into this further |
12:05.34 | tzafrir_laptop | it is updated by ./configure |
12:05.58 | DragoraN | how do you call in english when someone just rings you once or twice then he hangs up? |
12:07.25 | YonahW-Work | DragoraN: I think you mean a prank call |
12:08.18 | DragoraN | yeah, thx |
12:09.17 | *** join/#asterisk gardo (n=gardo@121.97.178.163) |
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12:36.36 | sof76 | Hi all, does anyone know how to install asterisk wihout installing zaptel in a clean way? Is there an option to give during asterisk installation? |
12:37.36 | Strom_M | zaptel and asterisk are completely separate packages |
12:37.56 | Strom_M | you don't even have to download zaptel to build asterisk |
12:38.40 | sof76 | thank you |
12:39.08 | Strom_M | don't thank me - thank obviousness |
12:41.38 | DrukenLPY | now Strom_M, be nice... not everyone has the brain power of a common mouse |
12:43.03 | Strom_M | the hi/cocks protocol |
12:44.18 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
12:45.08 | DrukenLPY | Strom_M: you know if anyone has had any luck with a voice activated ivr? |
12:47.11 | Strom_M | yeah |
12:47.22 | Strom_M | lumenvox apparently makes it easy |
12:47.51 | DrukenLPY | yeah? will have to check that out |
12:48.40 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
12:56.35 | DragoraN | how to detect something like this 555444#200 |
12:56.43 | DragoraN | to get #200 from the number |
12:57.03 | Strom_M | DragoraN: during an IVR menu? |
12:57.09 | DragoraN | no |
12:57.18 | DragoraN | when calling for example from my cell phone to my asterisk |
12:57.32 | DragoraN | i want to route to other destination when #200 is entered at the end of number |
12:57.35 | Strom_M | are you dialing # before you set the call up? |
12:57.50 | DragoraN | no |
12:57.55 | DragoraN | or 555444p200 |
12:58.10 | Strom_M | grr, answer my question |
12:58.15 | Strom_M | at which point do you dial #200 |
12:58.22 | Strom_M | after asterisk answers, or before? |
13:00.19 | Strom_M | i guess "at what point in time are you sending DTMF digits?" is too difficult a question |
13:01.18 | *** join/#asterisk CVirus (n=GoD@196.205.193.169) |
13:11.23 | *** join/#asterisk cappiz (n=cappiz@c85-196-102-30.static.sdsl.no) |
13:11.42 | cappiz | is it possible to make the "on hold music" not restart after a user has been announced his/her queue number? |
13:24.00 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
13:35.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
13:40.01 | DrukenLPY | uhmm... why? |
13:40.10 | DrukenLPY | you want them to just sit there in silence? |
13:41.09 | cappiz | well |
13:41.23 | cappiz | cant it pick up from where is stoped? |
13:41.42 | cappiz | kind of bad to only hear 30 seconds of the song |
13:41.47 | cappiz | over and over again |
13:42.02 | DrukenLPY | oh... |
13:42.14 | DrukenLPY | yeah, it's in the settings i guess... i've never had that problem |
13:42.35 | DrukenLPY | mine stays at it's current position until someone else goes into the moh |
13:42.55 | cappiz | hum.. settings under what? the music on hold or the queue |
13:43.06 | cappiz | wel, i tried with 2 in queue |
13:43.18 | DrukenLPY | music on hold.... |
13:43.24 | cappiz | and each one of the callers heard the song for only 30 seconds, |
13:43.25 | cappiz | ok |
13:43.31 | DrukenLPY | moh has nothing todo with the queue |
13:43.40 | cappiz | ok |
13:44.03 | cappiz | could it be that i only have 1 song in the moh category? |
13:46.34 | DrukenLPY | well, that's kinda dumb... hehehe yeah you should have a minimum of 2 |
13:46.41 | cappiz | ok |
13:46.41 | cappiz | :> |
13:46.55 | DrukenLPY | even if it's the same song |
13:48.31 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
13:52.49 | DrukenLPY | personally, i have several moh contexts... |
13:53.09 | cappiz | hehe |
13:53.11 | DrukenLPY | of course,... i also do multipul ivr's on a single system... and seperate pbx's |
13:53.23 | cappiz | yeah... works after i uploaded more than one song :) |
13:53.30 | cappiz | now it picks up |
13:53.31 | cappiz | :) |
13:53.33 | DrukenLPY | wholly crap :) |
13:56.16 | cappiz | is it possible to make freepbx/asterisk record calls done from DIAS? |
13:58.39 | Uatec | WHY AM I IN WORK ON SUNDAY?! |
13:58.56 | cappiz | hehe... you should know :P |
14:01.37 | DrukenLPY | Uatec: because your a bitch? |
14:03.12 | Dovid | hehe. i work on sunday too. goto bring home the money |
14:05.01 | *** join/#asterisk coppice (n=chatzill@163.201.17.210.dyn.pacific.net.hk) |
14:06.53 | *** join/#asterisk _WildPikachu_ (n=WildPika@about/linux/staff/wildpikachu) |
14:07.13 | _WildPikachu_ | I'm trying to strip down the number of modules loaded by asterisk, is there a simple list I can explicitly define to load? |
14:07.19 | _WildPikachu_ | I think i'm missing a few from my list |
14:07.45 | Strom_M | look in /var/lib/asterisk/modules |
14:07.56 | _WildPikachu_ | yea ... I'm loading about 10 so far |
14:08.20 | Corydon76-home | s/var/usr/ |
14:08.21 | _WildPikachu_ | but when I try dial ... I get engaged without any logs generated, think I'm forgetting something |
14:08.24 | Strom_M | er yes |
14:08.34 | _WildPikachu_ | one sec, I paste my list on pastebin |
14:08.34 | Strom_M | /usr/lib/asterisk/modules |
14:08.43 | Strom_M | it be early here |
14:08.58 | Corydon76-home | Yes, what are you doing up at 7am on a Sunday? |
14:09.04 | Strom_M | jet lag |
14:09.09 | _WildPikachu_ | http://pastebin.ca/587330 |
14:09.18 | Corydon76-home | Ah... |
14:09.20 | _WildPikachu_ | anything obvious I'm missing? |
14:09.48 | Strom_M | the CDR module, i'd imagine |
14:10.14 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
14:16.11 | _WildPikachu_ | ah |
14:16.41 | _WildPikachu_ | hectic, theres a whole bunch of cdr modules |
14:17.34 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
14:19.39 | Corydon76-home | _WildPikachu_: func_cdr |
14:19.51 | _WildPikachu_ | yea ... was trying out one by one |
14:22.22 | _WildPikachu_ | hrmmmmm |
14:22.22 | _WildPikachu_ | i'm missing something else |
14:22.35 | _WildPikachu_ | i dont think my extensions.conf is getting loaded |
14:23.13 | Corydon76-home | Why aren't you using autoload=yes ? |
14:24.23 | Corydon76-home | I would suggest an alternate approach... Use autoload=yes, then strip out various modules with noload => whatever.so until something breaks |
14:24.35 | Corydon76-home | extensions.conf is loaded with pbx_config.so |
14:25.08 | _WildPikachu_ | well ... I only want what I need, and would like to understand what each module does :) |
14:25.28 | Corydon76-home | formats are needed to load sound files from disk |
14:25.43 | _WildPikachu_ | doubt i'm going to need that |
14:25.56 | _WildPikachu_ | i basically just want to route voice calls comming into this server to various providers |
14:26.02 | Corydon76-home | You aren't playing messages (such as pbx-invalid.gsm)? |
14:26.08 | _WildPikachu_ | nope |
14:26.14 | _WildPikachu_ | (not yet) |
14:26.31 | Corydon76-home | Formats are also needed for recording |
14:26.38 | Corydon76-home | or monitoring |
14:26.40 | _WildPikachu_ | don't need that yet :o) |
14:26.54 | Corydon76-home | codecs are needed to translate |
14:27.25 | _WildPikachu_ | yea, figured that out and the formats .... just trying to get something to show up in the logs now |
14:27.31 | _WildPikachu_ | let me try the pbx_config quick |
14:27.37 | Corydon76-home | Well, if you're not playing anything from disk, then you don't need app_playback |
14:28.17 | _WildPikachu_ | excellent, I have some logs :) |
14:28.20 | _WildPikachu_ | let me go play around |
14:34.53 | _WildPikachu_ | hrmmm, Corydon76-home quick question ... which format_*.so provides ulaw btw? |
14:35.04 | Corydon76-home | pcm |
14:35.27 | _WildPikachu_ | :) |
14:41.47 | _WildPikachu_ | btw, where are asterisk sounds supposed to be installed? |
14:43.11 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
14:43.11 | *** mode/#asterisk [+o anthm] by ChanServ |
14:53.08 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
15:01.37 | Crimethinker | mine are in /var/lib/asterisk/sounds |
15:03.25 | *** join/#asterisk kusznir (n=kusznir@66-233-138-60.lew.clearwire-dns.net) |
15:03.47 | rob0 | No they're not. Weasels have eaten those sound files. |
15:04.57 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
15:05.21 | syle | ferrets are bad ass to |
15:05.56 | Strom_M | weasels have eaten your ferrets |
15:07.56 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
15:10.16 | *** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
15:12.46 | *** join/#asterisk af_ (n=getsmart@81-174-9-236.dynamic.ngi.it) |
15:13.10 | *** join/#asterisk lukketto (n=lukketto@host45-89-dynamic.1-87-r.retail.telecomitalia.it) |
15:15.55 | kusznir | Hi: I'm having trouble with setting outbound caller ID info. I'm getting the error: |
15:15.58 | kusznir | Jun 24 08:15:00 ERROR[2343]: pbx.c:1437 ast_func_write: Function CALLERID not registered |
15:16.37 | Strom_M | what line of dialplan code are you executing? |
15:16.43 | kusznir | The line generating that error is: exten => _X.,1,Set(CALLERID(num)=<my num>) |
15:16.55 | Strom_M | which version of asterisk? |
15:17.04 | kusznir | asterisk 1.2.16 |
15:17.08 | Strom_M | that's odd, |
15:17.39 | Strom_M | you compiled from source, right> |
15:17.40 | Strom_M | ? |
15:18.02 | kusznir | Yes (through a build-kit -- OpenWRT embedded linux) |
15:18.19 | Strom_M | hm |
15:18.43 | Strom_M | i don't know if they've slimmed it down so severely so as to not include the callerid function |
15:18.55 | kusznir | Most of the app_* stuff was not installed to save space, but I didn't see any that looked related. |
15:19.17 | kusznir | I grabbed the app_setcallerid.so, but it appears that's deprecated. |
15:19.33 | kusznir | Where would the callerid function be included? |
15:20.14 | Strom_M | func_callerid.so perhaps? |
15:20.43 | *** part/#asterisk lukketto (n=lukketto@host45-89-dynamic.1-87-r.retail.telecomitalia.it) |
15:24.22 | kusznir | Yep, that was it. I didn't have any func_* modules on my client, so I didn't even notice that catagory of modules existed. Thanks! |
15:25.23 | `Sean | wow |
15:25.27 | `Sean | must suck using it on wrt tho |
15:25.42 | `Sean | seeing how you wont have half the functions and not to mention no way to install a FXS port |
15:30.55 | kusznir | I was looking to use it for IP phones only. I don't have enough CPU power to run the codec's or do inline DTMF, but otherwise things seem to work well. |
15:31.07 | kusznir | I'm only running 3-5 phones from it anyway. |
15:31.49 | kusznir | I think its really cool how much I am able to get from such a small box! (I haven't tried musiconhold yet, but that is part of the minimal install, so it might be doable.) |
15:31.56 | kusznir | I do have voicemail working. |
15:34.15 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
15:34.30 | kusznir | I do have another question for everyone: I'm having issues with the timestamp on voicemail. The embedded system doesn't have /usr/share/zoneinfo, but it does have its zone info in /etc/TZ (uclibc-based). Asterisk console and the rest of the system properly show localtime, but voicemail is always in UTC. |
15:36.18 | Strom_M | you have to set the time zone in voicemail.conf |
15:36.34 | Strom_M | since your voicemail users can be in multiple time zones |
15:36.45 | kusznir | I've attempted to...I'm trying some more tests now with other formats. tz=pacific did NOT work. |
15:37.06 | Strom_M | odd, because it works everywhere else |
15:37.13 | Strom_M | you're sure it's in the right spot in the file? |
15:37.26 | kusznir | Do I have to reload anything on asterisk to make voicemail.conf changes take effect? |
15:37.35 | Strom_M | reload app_voicemail.so |
15:37.40 | tzafrir_laptop | <PROTECTED> |
15:37.41 | [TK]D-Fender | kusznir, You did this on the box definition line? |
15:37.49 | kusznir | yes |
15:38.44 | kusznir | 70 => <PIN>,Jim Kusznir,kusznir@kusznir.net,tz=PST8PDT |
15:38.54 | kusznir | I previously had tz=pacific |
15:39.04 | Strom_M | wrong spot |
15:39.21 | Strom_M | pin,name,email,pager_email,options IIRC |
15:40.27 | Uatec | what extension should i use if i want to catch all other numbers that i haven't already specified extensions for? |
15:40.32 | Strom_M | _X. |
15:40.41 | Strom_M | or _X! |
15:44.44 | Uatec | apparently it can't match that |
15:44.45 | Uatec | ? |
15:44.58 | Strom_M | why not? |
15:45.08 | Uatec | i don't know |
15:45.13 | Strom_M | _X! will match anything 1 or more digits |
15:45.15 | Uatec | oh wait |
15:45.17 | Uatec | DAMMIT |
15:45.19 | Uatec | of course |
15:45.23 | Uatec | 1 or more digits |
15:45.30 | Uatec | only works if your phone company aren't retards |
15:45.43 | Strom_M | let me guess: GTE |
15:45.47 | Uatec | nah |
15:45.49 | Uatec | NTL |
15:45.52 | Uatec | i'm in the UK |
15:45.58 | Uatec | they're presenting 0 digits on this line |
15:46.04 | Strom_M | hahahahahaha |
15:46.17 | Strom_M | match on s then |
15:46.37 | kusznir | On the voicemail timezone issue: If I use pacific as my timezone, I get disconnected with a "file vm-received not found". If I remove the tz or set it to PST8PDT, it finds vm-received just fine. |
15:46.44 | *** join/#asterisk soylentgreen (n=fgast@193.238.89.34) |
15:46.48 | kusznir | pacific is defined in zonemessages as: pacific=America/Los_Angeles|'vm-recieved' Q 'digits/at' IMp |
15:46.51 | Uatec | still doesn't want to match s |
15:47.45 | *** join/#asterisk soylentgreen (n=fgast@193.238.89.34) |
15:48.30 | kusznir | Ok, corrected the typo, and it now plays back :) Unfortunately, its still using UTC. |
15:49.08 | kusznir | My guess is the difference between glibc and uclibc's handling of timezones. |
15:53.15 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
16:04.22 | Uatec | when dialing out with misdn, what does: mISDN/4-u7 |
16:04.24 | Uatec | <PROTECTED> |
16:04.33 | Uatec | what's the 4 and the u7? |
16:05.19 | ai-a | who / what told you to use 4-u7 ? |
16:05.44 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
16:06.04 | Uatec | i didn't |
16:06.06 | Uatec | that's from my log |
16:06.37 | Uatec | i used misdn/g:isdn/${EXTEN:1} |
16:06.58 | Uatec | the u number appears to be incrementing every time i dial |
16:07.17 | ai-a | line number used ? |
16:08.01 | Uatec | that's what i thought initially |
16:08.10 | Uatec | but i've got 4 lines plugged in |
16:08.22 | Uatec | 3 of them are flashing red, which i took to mean that they were not working |
16:08.35 | Uatec | and i get 4- and 2- |
16:08.43 | Uatec | *sigh* |
16:10.15 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
16:11.41 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
16:15.27 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
16:17.23 | Uatec | FUCK |
16:17.29 | Uatec | i just rebooted |
16:17.32 | Uatec | and it's all gone |
16:17.35 | Uatec | all my hardware |
16:17.39 | Uatec | raid array failed |
16:17.40 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
16:22.50 | syle | that'll teach ya to check your system logs |
16:23.13 | syle | try loosing 3 years worth of programming, not fun |
16:26.19 | syle | so replace the 2 downed drives, reinstall OS, restore backups, reconfigure system, 5 hours maybe, go for a beer |
16:27.21 | Uatec | syle, how can i check my system logs |
16:27.27 | Uatec | the array is fucked |
16:27.35 | Uatec | 5 hours work |
16:27.36 | Uatec | youre right |
16:27.40 | Uatec | on a fracking sunday night |
16:27.41 | Uatec | for no reason |
16:30.01 | syle | shit happens dude, its how you deal with it that makes you a person |
16:30.52 | syle | course if you just needed to vent that works to :) |
16:32.54 | syle | tell ya one thing though, i;d deal with a downed array anyday of the week instead of going through a divorce again lol |
16:33.58 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
16:34.17 | [TK]D-Fender | "Divorce : a word derived from latin meaning 'to rip a man's heart out through his wallet'" |
16:35.55 | Strom_M | "Polycom: a word derived from Algonquin for 'telephone which takes five minutes to reboot'" |
16:39.07 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
16:39.33 | syle | Child Support: way to do it for next 18 years |
16:40.15 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
16:42.50 | Uatec | lol |
16:42.53 | Uatec | i'm fortunate |
16:42.57 | Uatec | noone can divorce me |
16:43.42 | *** join/#asterisk Greek-Boy (n=g@196.45.144.42) |
16:43.46 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
16:44.42 | Greek-Boy | what is an open-wire infrastructure? |
16:44.49 | Greek-Boy | first time i hear of that in telecoms... |
16:45.27 | Strom_M | Greek-Boy: open-wire carrier as opposed to twisted-pair insulated wiring |
16:46.00 | [TK]D-Fender | Strom_M, Exaggerated... takes a hair over 2 to boot :) |
16:46.00 | Greek-Boy | hmmm |
16:46.05 | Greek-Boy | i've tried to google it |
16:46.12 | Uatec | Strom_M, you mean cables that are metal open to the air? |
16:46.51 | Strom_M | sigh |
16:46.53 | Greek-Boy | no results, I'd like to see some technical specs of open wire |
16:47.10 | Strom_M | welcome to #asterisk, where you apparently don't need to know a thing about telephony to screw up a phone system |
16:48.09 | *** join/#asterisk Greenbox (n=Brett@c-68-59-20-153.hsd1.sc.comcast.net) |
16:48.33 | Uatec | .... |
16:49.25 | syle | to only be a cunt hair of an error to make the whole thing work again |
16:51.28 | Uatec | like your expensive new raid mirror which appears to be less stable than the cheapo 10gig hdd you pulled out of your PC you had 7 years ago |
16:53.02 | syle | but we can all agree on at least raid mirrors being more stable than some of the women out there |
16:54.44 | Uatec | oh yes |
16:56.05 | *** join/#asterisk javar (n=javar@69.79.134.24) |
16:58.34 | syle | then again you might be lucky and just power supply or raid controller died |
16:59.35 | *** join/#asterisk Metfan2007 (n=metfan@189.146.139.155) |
17:00.31 | Metfan2007 | Hi, I have installed a Asterisk 1.4.5 and Zaptel 1.4.2 but I cannot use any ZAP command in the Asterisk CLI |
17:00.35 | Metfan2007 | any clue? |
17:01.57 | russellb | you have to install zaptel before asterisk |
17:02.46 | *** join/#asterisk marv (n=Tim@c-71-228-189-127.hsd1.al.comcast.net) |
17:03.27 | marv | what was the url to the latest iax2 rfc draft? |
17:03.57 | russellb | marv: http://tools.ietf.org/id/draft-guy-iax-03.txt |
17:03.58 | syle | anyone actually running 1.4 branch in production? |
17:04.23 | russellb | plenty of people are |
17:05.56 | Uatec | hey, in SIP.conf is is possible to allocate multiple values to the mailbox? |
17:06.24 | russellb | Uatec: no |
17:06.26 | syle | why |
17:07.07 | russellb | just kidding, yes |
17:07.19 | marv | russellb: thanks, for some reason google wasn't finding it, it found the iax enum draft though |
17:07.22 | russellb | you can put multiple mailboxes, separated by commas |
17:07.50 | russellb | marv: yeah, i always search buy "draft-guy-iax" |
17:07.53 | russellb | by* |
17:08.22 | russellb | Uatec: you can put multiple mailboxes separated by commas |
17:09.14 | russellb | which you would have seen if you had looked in sip.conf.sample ... |
17:09.16 | Metfan2007 | anyone has compiled succesfully sangoma cards with zaptel 1.4.2????? |
17:10.24 | russellb | maybe a few more question marks will help :-p |
17:11.14 | marv | why is the word wikipedia in brackets after the first sentence of section 1.1? format conversation error? |
17:11.26 | Metfan2007 | hehe, sorry, I am frustrated |
17:11.35 | *** join/#asterisk gardo (n=gardo@121.97.208.93) |
17:11.41 | russellb | marv: i have no idea ... |
17:12.08 | Metfan2007 | someone using sangoma digital cards here? |
17:15.53 | *** join/#asterisk Greek-Boy (n=g@196.45.144.42) |
17:17.23 | syle | haven;t tried 1.4 yet, worth it? |
17:17.24 | Uatec | :( |
17:17.26 | Uatec | P[ 2] GOT IGNORE SETUP |
17:17.26 | Uatec | P[ 2] CC_RELEASE_COMPLETE|CONFIRM [TE] |
17:17.26 | Uatec | P[ 2] release_chan: Ch not found! |
17:17.39 | Uatec | i'm getting that whenever i dial in to my B410P BRI isdn card |
17:17.45 | Uatec | i have no idea what it means |
17:17.49 | Uatec | what does it mean? |
17:17.54 | syle | suppose i;d have to recode all my c modules now, hopefully not much has changed |
17:29.20 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
17:31.52 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
17:39.05 | *** join/#asterisk saftsack (n=saftsack@pD9E0453C.dip.t-dialin.net) |
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18:10.29 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
18:10.30 | __DAW | errr |
18:12.02 | DeL3e7 | yea |
18:13.53 | Strom_M | cocks |
18:14.01 | russellb | ... |
18:14.12 | Strom_M | i mean pones |
18:14.14 | Strom_M | phones |
18:14.46 | DeL3e7 | this is too mutch for me to handle |
18:15.00 | Strom_M | ? |
18:15.12 | Strom_M | i mean, I know I'm fun to hug and all, but wow |
18:15.33 | DeL3e7 | i mean asterrisk |
18:15.35 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
18:15.35 | *** mode/#asterisk [+o anthm] by ChanServ |
18:15.48 | Strom_M | ah |
18:15.51 | Strom_M | well that's different then |
18:15.57 | Strom_M | and that's why we're here! :) |
18:16.01 | DeL3e7 | its installed on my linksys |
18:16.09 | Strom_M | wrt54g? |
18:16.12 | DeL3e7 | yea |
18:16.14 | DeL3e7 | GS |
18:16.50 | Strom_M | ive got one of those that runs asterisk |
18:16.58 | Strom_M | or that ran asterisk anyway |
18:17.05 | Strom_M | i dont think i've installed it on the latest reflasg |
18:17.07 | Strom_M | reflash |
18:17.13 | DeL3e7 | what could i use it for a gateway? |
18:17.29 | Strom_M | for a gateway for what? |
18:17.50 | DeL3e7 | a main asterisks box with alittle more beef |
18:17.58 | Strom_M | sure |
18:18.10 | DeL3e7 | i dunno ive never used this software before |
18:18.12 | Strom_M | you could use it as an IAX/SIP gateway if you configured it right |
18:18.19 | DeL3e7 | im reading the huge pdf's |
18:18.30 | Strom_M | but I'd say start with a basic setup and then go from there |
18:18.33 | Strom_M | do you have thebook? |
18:19.07 | DeL3e7 | asterisksTFOT.pdf |
18:19.21 | DeL3e7 | http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
18:19.25 | Strom_M | well, s/asterisks/asterisk/ |
18:19.29 | Strom_M | but yes, that's a good book |
18:22.34 | DeL3e7 | kewl asterisk -c |
18:23.33 | DeL3e7 | http://voip-info.org/wiki/ |
18:24.54 | Uatec | hey, does anybody know how to use the distinctive ring feature on the linksys SPA922? |
18:32.36 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
18:32.36 | *** mode/#asterisk [+o mog] by ChanServ |
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18:54.51 | mikpel | how can a legacy phone from ericsson be connected to asterisk, and stil use its functionsbuttons? |
18:57.23 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
18:59.03 | Cyber-Dogg | so... if I wanted to have my system make calls out to someone... what am I looking for? |
19:04.05 | Greek-Boy | any gsm network techies around? |
19:04.10 | Uatec | having just finished rolling out a 16 phone asterisk system, but not having a clue how he did it |
19:07.24 | *** join/#asterisk _WildPikachu_ (n=WildPika@about/linux/staff/wildpikachu) |
19:12.56 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
19:15.40 | *** join/#asterisk skoenman (n=dbgksdfg@dsl-241-186-126.telkomadsl.co.za) |
19:15.48 | skoenman | hey all |
19:15.57 | Cyber-Dogg | howdy |
19:16.02 | *** part/#asterisk MikeJ (n=MikeJ@d149-67-175-107.try.wideopenwest.com) |
19:16.29 | skoenman | can anybody tell me if its possable to use asterisk to make a pbx with software phones??? |
19:18.12 | waKKu | exactly what it is |
19:18.39 | skoenman | he??? |
19:18.57 | skoenman | k so you can have the pbx and use software phones with it??? |
19:19.13 | russellb | yes |
19:19.26 | skoenman | cool whats the softwares name ??? |
19:19.35 | skoenman | i have been looking but theres nothing |
19:23.14 | skoenman | ??? |
19:24.21 | Iamnacho | x-lite, idefisk are 2 that i like |
19:24.35 | skoenman | k cool |
19:24.41 | skoenman | do they run on windows??? |
19:24.47 | Iamnacho | yes both of those do |
19:24.54 | skoenman | k cool |
19:24.55 | Iamnacho | i think idefisk runs cross platform |
19:24.58 | skoenman | thanks |
19:25.28 | Iamnacho | you should look at this page http://www.voip-info.org/wiki-Asterisk+phones |
19:25.48 | Iamnacho | down a ways is a soft phone thing |
19:40.25 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
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20:05.16 | *** join/#asterisk dagrim (n=_dagrim_@unaffiliated/dagrim) |
20:05.33 | *** join/#asterisk NOT_guru (n=chatzill@209.145.181.55) |
20:06.34 | NOT_guru | hello anyone here using a linksys pap2? how well does it work? are you using both ports as 2 different extensions? |
20:07.14 | NOT_guru | I am thinking of getting a couple of these and breaking my house into 4 "zones" |
20:07.16 | waKKu | NOT_guru: yes |
20:08.07 | NOT_guru | wonderfull waKKu I have played with a grandstream a buddy had and didn't like it much... are the linksys ones pretty straight forward? |
20:08.09 | waKKu | each linksys pap2 haves 2 FXS ports.. u can put 2 differentes numbers and, of course, 2 atas |
20:08.57 | waKKu | NOT_guru: yeah.. i believe linksys a best choice for this kind of equip ;) |
20:09.14 | waKKu | however.. linksys is Cisco SOHO |
20:09.17 | NOT_guru | wonderfull thanks man I appreciate the information |
20:09.33 | *** part/#asterisk mikpel (i=mikpel@h-195-210.A183.cust.bahnhof.se) |
20:09.40 | NOT_guru | I found a place I trust that has some for $25 |
20:09.54 | Qwell | locked, no doubt |
20:10.22 | NOT_guru | I don't think so but I guess I will find out |
20:10.31 | waKKu | uow.. NOT_guru $25 is cheaper SO MUCH... |
20:10.48 | *** join/#asterisk swift99 (n=dave@ip70-178-9-136.ks.ks.cox.net) |
20:10.51 | NOT_guru | either of you wanna look at the link? |
20:11.00 | waKKu | i have seem some pap2 for $100 $120 ... |
20:11.02 | NOT_guru | see what you think |
20:11.06 | waKKu | ok |
20:12.00 | NOT_guru | you get that? and if so please let me get mine purchased before you spread that =) |
20:12.26 | waKKu | yeah.. its really a linksys pap2 ... very very cheap.. |
20:12.28 | swift99 | Would someone be willing to walk a newbie (me) through some of the pros and cons of an asterisk based system for a small office (12 phones)? |
20:12.48 | NOT_guru | looks unlocked to me |
20:13.28 | waKKu | NOT_guru: if I not live in brazil.. i'll purchase a dozen :P |
20:13.35 | NOT_guru | LOL |
20:15.29 | Qwell | NOT_guru: link? |
20:16.14 | NOT_guru | again please refrain from spreading that till tomorrow |
20:16.21 | DragoraN | what softphone on windows do you use? |
20:16.24 | NOT_guru | I just have to see if my buddies want some before I buy |
20:16.35 | NOT_guru | I personally use x-lie |
20:16.41 | NOT_guru | x-lite |
20:16.46 | _VoiceMeUp_COM | xlite is ok |
20:16.48 | _VoiceMeUp_COM | anyone tried bria ? |
20:16.52 | _VoiceMeUp_COM | from same company |
20:16.59 | waKKu | DragoraN: idefisk is a good choice, and free |
20:17.06 | DragoraN | yes |
20:17.12 | waKKu | and.. one of few that support IAX |
20:17.22 | NOT_guru | I am willing to try anything, as I am looking to buy some softphone software |
20:18.09 | NOT_guru | unless I find something wonderfull and free |
20:18.19 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
20:18.27 | swift99 | I've just been asked to help choose and configure a small office PBX, from the ground up. At this point, I don't know what questions to ask. Asterisk has been suggested as a possible solution. Would someone with experience be willing to bring me up to speed on what questions to ask? |
20:18.36 | swift99 | Thanks! |
20:18.50 | _VoiceMeUp_COM | failover/emergency |
20:19.03 | _VoiceMeUp_COM | then backup solution.. example net down |
20:19.19 | *** join/#asterisk dagrim (n=_dagrim_@unaffiliated/dagrim) |
20:19.42 | _VoiceMeUp_COM | so for tooll frees you could setup an alternative btn for them in case pri down |
20:19.42 | _VoiceMeUp_COM | etc |
20:19.51 | waKKu | swift99: use asterisk and be happy :) |
20:20.09 | swift99 | What kind of equipment would we need to purchase? |
20:20.15 | waKKu | asterisk was made to do exactly what u want.. |
20:20.18 | _VoiceMeUp_COM | then theres always the budget and , does anyone live byt he phones ? .. example.. would you set up an hospital emgency reposnce unit with asterisk ? |
20:20.40 | waKKu | swift99: this depends.. what type of equip. do u want use in the end's ? |
20:20.43 | _VoiceMeUp_COM | so how vital is the system in first place, and do they want 80% up time or 99.99999 |
20:20.56 | swift99 | _VoiceMeUp_COM: I'll ask my neighbor - he manages the hospital network :o) |
20:20.58 | waKKu | ATA's ? Softphones ? did u have an PABX ? |
20:20.59 | _VoiceMeUp_COM | actualy i think 99.99999 is too much |
20:21.09 | _VoiceMeUp_COM | 5 9's = 1 sec downtime in 1 year |
20:21.19 | swift99 | Right now, this is a brand new building. There is nothing. |
20:22.05 | swift99 | We have limited resources to support the tool, but the resources we do have are world class. |
20:22.30 | swift99 | Ideally, we want the tool to just run like any turnkey PBX that we would purchase. |
20:23.09 | swift99 | I was hoping to run it on two midsized ubuntu servers, with a load balancer to provide automatic failover. |
20:23.33 | swift99 | What kinds of phones and telco interface hardware would we need? |
20:23.58 | swift99 | ... for an ideal configuration, that is. |
20:24.41 | *** join/#asterisk sof76 (n=chatzill@BAAd99e.baa.pppool.de) |
20:26.29 | swift99 | By nothing, I mean two walls and a roof have been built. We need to decide what sort of internal phone system to use between the roof going up andthe ceiling going in. |
20:27.33 | swift99 | waKKu: One of the things we need to decide are what we want on the ends. I was thinking cisco VOIP phone. |
20:28.17 | sof76 | Hello, does someone know why when I launch safe_asterisk, sometimes the stop command is not recognized in the asterisk console? |
20:28.32 | swift99 | But, I'm an apps programmer, not a telecom tech. I don't know what the alternatives are for end points. |
20:29.30 | waKKu | swift99: well.. if u really choose for IP-Phones and SIP protocol, u wont need any other hardware on asterisk.. |
20:30.06 | swift99 | What about connection to the telco side? |
20:30.13 | waKKu | but if u need comunicate asterisk with other PBX, u'll need some PCI card (E1 or ISDN or FXO ports) .. |
20:31.33 | swift99 | At this point the simpler the system, the better it is. We have *nix admins and network techs, but little telecom experience in our team. |
20:33.57 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
20:34.36 | swift99 | waKKu: what do you recommend for connecting to the telco side of the pbx? |
20:35.59 | waKKu | swift99: i cant say.. i dont know ur structure and i dont live in your country (probably.. i'm from brazil)... Here we have some PBX with R2 or E1 connectors.. |
20:37.14 | swift99 | wakku: I'm USA. So, I'd need to talk to my telco, see what connector they expect, then get the appropriate card for that connection, I guess. |
20:38.39 | swift99 | waKKu: At that point, if we use SIP phones, we should be able to handle everything. |
20:38.57 | swift99 | waKKu: what about using load balancers with SIP Phones? |
20:39.46 | swift99 | waKKu: Would they work, providing an automatic failover, or would they mess up the assumptions of the system? |
20:40.27 | waKKu | swift99: hm.. really dunno.. i never used that.. |
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20:41.06 | waKKu | but i thought that its possible .. :) |
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20:51.40 | swift99 | How long would you expect it to take to configure a basic system with SIP phones, provided linux and asterisk were installed? |
20:55.38 | waKKu | thinking out of structure and users - one day, two on max |
20:55.58 | bbryant | swift99, there's not too much you have to do except add sip peers in sip.conf and config your phones |
20:57.23 | *** join/#asterisk Dovid (n=Dovid@bzq-82-81-163-218.red.bezeqint.net) |
20:58.12 | swift99 | Sounds good. We'll have to set up a demo system. |
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21:17.00 | *** join/#asterisk codey (i=codec@iglu.paranoid-penguin.de) |
21:17.05 | codey | hi all |
21:17.16 | codey | anyone using app-fax? |
21:17.20 | codey | (faxreceive) |
21:22.56 | codey | http://slexy.org/paste/3235 |
21:23.00 | codey | maybe someone is able to help |
21:24.09 | Dovid | i donr know the agi. but it is failing after |
21:24.09 | Dovid | AGI Tx >> 200 result=1 (default) |
21:24.20 | Dovid | did u look at the docs for it. maybe u missed a step some where |
21:24.42 | Dovid | a hunch is this |
21:24.56 | Dovid | you may need to put in exten => s,1,ANswer |
21:24.58 | Dovid | and then |
21:25.14 | Dovid | exten => s,2,,AGI(receive_fax,receive) |
21:25.17 | codey | well, the docs say you need the part "[recipient foobar]" |
21:25.23 | codey | and then faxrecipient has to be "foobar" |
21:25.33 | Dovid | as well as |
21:25.33 | Dovid | exten => h,2,Set(MY_CALLERIDNUM=${CALLERIDNUM}) |
21:25.33 | Dovid | exten => h,3,DeadAGI(receive_fax,deliver) |
21:25.46 | Dovid | should be h,1 and h,2 but that should not be affecting it |
21:25.59 | Dovid | goobar needs to set |
21:26.02 | Dovid | its just an example |
21:26.07 | codey | yes, i know |
21:26.10 | codey | I've set it to "default" |
21:26.19 | codey | (i've tried a different name before - same result) |
21:26.34 | Dovid | do u know y its set to default ? |
21:26.53 | Dovid | also are there any other packages that need to be installed ? |
21:27.24 | codey | i've installed all needed perl-modules |
21:27.28 | codey | and all the postscript stuff the agi uses |
21:27.31 | Dovid | my hunch is that one of the variables sint set |
21:28.46 | Dovid | isnt set* |
21:29.27 | codey | the FAXRECIPIENT= stuff doesn't matter as long as you have a section with the same name in the faxreceive.conf |
21:29.39 | Dovid | that may matter |
21:29.43 | sof76 | Hi everybody, I encountered a bug with asterisk, sometimes asterisk does not recognize the stop commands, it answers : No such command 'stop now' (type 'help' for help), can you help me please? |
21:29.47 | codey | it *is* there |
21:30.01 | codey | (read my paste - i've included all the needed config stuff) |
21:30.08 | Dovid | u may need to set it differently. like i said i dont know the agi so i cant say for sure |
21:30.13 | Dovid | u can try emailing the users list |
21:30.35 | Dovid | sof76: what version ? i doubt its a bug. others would of reported it here |
21:30.41 | Dovid | can u replicate it ? |
21:31.20 | sof76 | I have the latest one, 1.4.5 |
21:31.33 | codey | sof76: what kind of terminal do you use? maybe there are some strange sequences because you did a backspace or whatever (i experience this on my mac sometimes) |
21:31.35 | Dovid | codey: what codec r u using ? i doubt thats ths issue |
21:31.43 | Dovid | sof76: can u replicate it ? |
21:31.51 | sof76 | I have this problem only on asterisk installed on a virtual server without zaptel and libpri |
21:31.54 | codey | Dovid: RxFAX works fine for me |
21:32.38 | Dovid | codey: also try agi debug. see if anything more comes up |
21:33.03 | codey | the output of my paste IS agi debug |
21:33.19 | sof76 | it does not happen automatically, but when it happens command stop does not exist on the help menu |
21:33.41 | Dovid | ah ok. sorry i cant help u |
21:33.46 | Dovid | like i said try the users lsit |
21:33.53 | Dovid | hmm |
21:34.00 | Dovid | if its not in help then it may be a bug |
21:34.07 | Dovid | try asking one of the room ops when the come in |
21:34.26 | sof76 | what is the room ops? |
21:34.41 | Dovid | any of the mods in the room |
21:34.48 | Dovid | TK: also knows a lot |
21:42.22 | sof76 | If I launch asterisk with asterisk -c, then sometimes it stops at this level : pbx_dundi.c:437 reset_global_eid: No ethernet interface found for seeding global EID You will have to set it manually. |
21:42.24 | sof76 | ............................ |
21:42.41 | _VoiceMeUp_COM | hey guys need a regex expert im a noob regex : sip:78|2@1.2.3.4:5060;blah=udp i need to extract the part between sip: and the | so 78 in this example could be from null to 8 chars |
21:42.44 | _VoiceMeUp_COM | if one can help |
21:42.58 | sof76 | and then the stop commands are not in the help menu and not recognized by asterisk |
21:43.09 | Dovid | i am off to bed. |
21:45.52 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
21:46.06 | *** part/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
21:46.08 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
21:49.11 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
21:49.42 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
21:49.48 | *** join/#asterisk dudes (n=nixtux@66-216-227-31.dhcp.stcd.mn.charter.com) |
21:52.58 | dudes | Any suggestions on a good residential DID provider inside the US that is reliable and would run less than $60/mth per did? |
21:53.41 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
21:58.29 | Crimethinker | I'd love to find one that actually bills with dead trees and a stamp at any price at all |
21:58.46 | dudes | yea I here you there |
21:59.22 | Strom_M | i'd love to find an IRC user that can tell the difference between homophones |
21:59.32 | dudes | sadly, I'm thinking about getting raped by Qwest and buying a TDM card /w a couple FXO ports |
22:00.38 | Strom_M | dudes: what's your expected call traffic? |
22:01.27 | dudes | not much |
22:01.40 | dudes | probably like 10 calls a day at most |
22:01.41 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
22:01.51 | dudes | per DID that is |
22:02.34 | *** join/#asterisk obnauticus (n=admin@c-71-59-162-60.hsd1.wa.comcast.net) |
22:02.47 | Strom_M | "not much" doesn't tell me how many minutes |
22:03.12 | dudes | no more than 500 a month per DID |
22:04.19 | *** join/#asterisk droops (n=root@adsl-074-245-001-031.sip.jan.bellsouth.net) |
22:04.26 | [TK]D-Fender | Strom_M, homophones!??!?! I run a *STRAIGHT* PBX thank-you! |
22:04.55 | waKKu | LOL |
22:04.56 | waKKu | hhaioeha |
22:04.59 | Strom_M | homophones are words that sound alike but which suck each other off in the men's restroom |
22:05.12 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
22:05.18 | [TK]D-Fender | Strom_M, An apple a day keeps the mixed-metaphor away ;) |
22:05.23 | Strom_M | heheh |
22:05.35 | Crimethinker | snap crackle pop toyota? |
22:05.38 | [TK]D-Fender | <- literate bastard |
22:06.02 | Crimethinker | does any voip provider bill with dead trees & stamp, and accept payment by same? |
22:06.30 | Strom_M | dudes: teliax is $5 a month for the DID and 2c a minute for the traffic, which would put you at $15 a month per DID |
22:06.53 | Crimethinker | I'm trying to deduct this crap as a business expense, but my accountant is unhappy with printouts of webpages as documentation |
22:07.02 | kombi | when CLI says: "No application 'MeetMe' for extension ()" when in fact there is one, what might that be? |
22:07.07 | Strom_M | well tell your accountant that it's the 21st century now |
22:07.18 | __DAW | Crimethinker: PM me |
22:07.56 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
22:10.18 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
22:11.33 | kombi | I understand there has to be [rooms] conf=>123 and exten=>123,2,MeetMe(123), correct? |
22:13.11 | *** join/#asterisk saftsack (n=saftsack@pD9E0453C.dip.t-dialin.net) |
22:13.50 | [TK]D-Fender | kombi, You need to have zaptel installed BEFORE compiling *. |
22:14.03 | [TK]D-Fender | kombi, Forget the conf.. your APP didn't build for lack of a timing source |
22:14.47 | kombi | f* me.., just managed to compile ztdummy against sources, didn't know * needs remaking too.. thank Fender! |
22:14.58 | kombi | thanks, I meant.. |
22:15.55 | [TK]D-Fender | kombi, NP, * needs to see Zaptel being ready becfoe compiling the app. |
22:16.03 | [TK]D-Fender | before* |
22:16.04 | [TK]D-Fender | kl;asjdkl;asjd |
22:16.33 | [TK]D-Fender | <- just in from a 5 hour bikeride around the West Island. |
22:18.13 | kombi | that being a racing bike? |
22:19.12 | kombi | or street bike, thing your race on streets with with narrow tyres |
22:19.44 | kombi | what is it really called in your lingo? |
22:20.49 | kombi | should do make -j3.. |
22:21.28 | kombi | maybe make clean wasn't necessary there.. |
22:29.20 | [TK]D-Fender | Hybrid |
22:29.20 | kombi | flat bar? |
22:29.20 | [TK]D-Fender | not originally, but replaced custom, yes |
22:29.20 | [TK]D-Fender | was a "W" like hybrid bar, heavy street (light off-road capable) tires, T bar (loo long for me now, I need a shorter & taller riser post)) |
22:29.20 | [TK]D-Fender | More of a light off-road touring bike class now. |
22:29.20 | [TK]D-Fender | I'm totally f'n wiped right now... |
22:29.21 | kombi | so what is the proper name for racing bike for the street? |
22:29.21 | [TK]D-Fender | kombi, damned if I know! I just ride! |
22:29.21 | kombi | lol.. still trying to pick up english better |
22:29.21 | [TK]D-Fender | kombi, used to be the old "10-speed" which looks rather similar still. But names are so grey these days |
22:29.21 | kombi | since there 30 speeds now..;) |
22:29.21 | kombi | btw, does it make sense to install progdocs? |
22:29.21 | [TK]D-Fender | kombi, Your's is considerably better that some native english speakers I know :) |
22:29.21 | *** join/#asterisk nephfl (i=nephilim@wsip-70-184-144-158.ga.at.cox.net) |
22:29.21 | kombi | thankyou! (flattered) |
22:29.21 | [TK]D-Fender | I never installed any extra docs personally |
22:29.21 | [TK]D-Fender | kombi, Whats your native tongue? |
22:29.21 | kombi | german |
22:29.21 | [TK]D-Fender | kombi, neato, any other languages you speak? |
22:29.21 | JT | vas is das? |
22:29.21 | JT | i mean hi |
22:29.21 | kombi | JT: was ist das |
22:29.21 | [TK]D-Fender | Haagen Dasz? ;) |
22:29.21 | kombi | ..means What's that? |
22:29.21 | kombi | lol.. |
22:29.21 | JT | kombi: it's not vas? |
22:29.21 | kombi | nope |
22:29.21 | kombi | common misconception..;) |
22:29.23 | JT | is it at least pronounced as vas? |
22:29.23 | kombi | also hardly ever said as a phrase |
22:29.23 | kombi | JT: that is completely right |
22:29.35 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
22:29.47 | nephfl | anyone here setup a wrtp54g with asterisk? |
22:29.59 | kombi | vas is das? |
22:30.03 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
22:30.37 | kombi | Fender: some french, but just enough to get by without being beaten up |
22:32.18 | kombi | after succesfully modprobing ztdummy and re-compiling *, should there be some meetme options in CLI? |
22:33.39 | [TK]D-Fender | kombi, Excellent, trilingual = very cool. |
22:34.00 | [TK]D-Fender | kombi, You should see it via "show application meetme" |
22:34.49 | kombi | "your application is not registered" ;( |
22:34.51 | [TK]D-Fender | kombi, Now before touching * I would have suggested doing "modprobe ztdummy" (you DID specifically check that off in make menuselect, right?) and "modprobe zaptel" |
22:35.56 | kombi | Fender: actually compiled it debian style with apt-get zaptel-sources and m-a a-i zaptel sources |
22:36.43 | kombi | modprobe ztdummy went without complains, maybe I try again with modprobe zaptel.. |
22:37.07 | kombi | oh, no, never mind, zaptel's loaded (lsmod) |
22:37.33 | [TK]D-Fender | kombi, I really can't comment on your style of install, I've only done pure-source |
22:38.11 | kombi | make menuselect in /usr/src/asterisk you mean? (I've only ever done that to do with kernels so far) |
22:38.24 | [TK]D-Fender | kombi, in zaptel. |
22:38.34 | kombi | oh, ok.. |
22:38.36 | [TK]D-Fender | kombi, in ther you can choose whic modules to complie |
22:39.26 | kombi | modules seem to have come out ok, at least they loaded are nicely listed in lsmod |
22:39.30 | kombi | and.. |
22:40.20 | saftsack | hi, is a via 1ghz cpu capable of handling about 20 calls? (no codec tranlsation) |
22:41.49 | kombi | doing make menuselect in asterix shows meetme XXed out, so there is still in issue with zaptel.. |
22:42.53 | [TK]D-Fender | kombi, do a ./configure first |
22:43.01 | [TK]D-Fender | You'll need to pick up your changes. |
22:43.10 | kombi | just did, still.. |
22:43.11 | [TK]D-Fender | saftsack, sure |
22:43.15 | [TK]D-Fender | :/ |
22:43.26 | kombi | weired.. |
22:43.50 | saftsack | has the via board a long lifetime? |
22:44.00 | saftsack | or will they propably die after about 3 years |
22:44.33 | [TK]D-Fender | saftsack, Oh, I presumes that 1 GHZ was on a POS CPU |
22:44.45 | [TK]D-Fender | wasn't |
22:44.48 | [TK]D-Fender | C3 = crap |
22:44.54 | [TK]D-Fender | more like a P550-600 |
22:45.10 | [TK]D-Fender | saftsack, It'll probably do the job but.... ICK |
22:45.31 | saftsack | oh ok. so it is more recommened to buy a soekris board for long life duration? |
22:45.41 | [TK]D-Fender | I can buy a 1 GHZ PC complete for 100$ easy these days with all sorts of extras... |
22:45.57 | [TK]D-Fender | saftsack, Buy a real computer |
22:46.16 | saftsack | its for home usage ... |
22:46.30 | saftsack | and no ... i have no 20 persons at home :-P |
22:46.40 | [TK]D-Fender | saftsack, So's mine, and a real PC will cost you less and survive more. |
22:47.16 | saftsack | why will it cost less? a normal pc takes about 100W |
22:48.49 | [TK]D-Fender | saftsack, How much does a Soekris cost? |
22:49.55 | saftsack | about 200$ |
22:50.04 | saftsack | + psu i think |
22:50.32 | [TK]D-Fender | saftsack, I can get a real PC for under half of that which includes expansion options, and no extras required like flash, etc. |
22:50.54 | saftsack | but with active cooling i think |
22:52.07 | [TK]D-Fender | saftsack, sure, feel free to guesstimate your energy costs, effort, etc. I gave you "my" math. Go see how it applies to you. |
22:52.22 | *** join/#asterisk Corydon76-home (i=two@pdpc/supporter/sustaining/Corydon76-home) |
22:52.22 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
22:52.35 | saftsack | yes your calculation is cool but i dont think that your pc will life longer than my soekris |
22:55.09 | kombi | saftsack: Is that thing completely noiseless and easy to run * on? |
22:55.22 | saftsack | i have to find out .... |
22:56.42 | JT | people have run asterisk on soekris before |
22:57.09 | JT | in fact, some setups have a full pri with a suitable pci card (no transcoding) |
23:00.15 | saftsack | how to get pci cards in this minipci port? are there adapters available? |
23:00.40 | JT | err |
23:01.29 | *** join/#asterisk obnauticus (n=admin@c-71-59-162-60.hsd1.wa.comcast.net) |
23:01.29 | JT | you buy the soekris net4801 |
23:01.35 | JT | it has a full pci slot |
23:01.35 | saftsack | i know that the soekris has a pci port on the side but what if i want tot use the minipci port? i saw an adapter but it was only available in the usa |
23:01.51 | JT | yes you can buy adapters online |
23:01.54 | saftsack | i have an ar7 here with a minipci port and i want to test this first |
23:02.00 | saftsack | where? :) |
23:02.07 | JT | what's the point? |
23:02.11 | JT | google for them |
23:02.21 | JT | if you have a pci card... use the pci slot? |
23:06.00 | saftsack | no my router just has a minipci slot |
23:06.12 | saftsack | i got 1 entry in google but they dont sell :( they just produce |
23:06.30 | JT | well you bought the wrong model |
23:06.38 | JT | what card do you want to attach? |
23:07.37 | saftsack | hfc-s card |
23:07.55 | saftsack | this cards have a one byte fifo so the irqs in the router will be fullfilled but i want to give it a try |
23:08.10 | JT | you can get minipci bri cards, but they cost a lot more than those hfc-s cards ;) |
23:08.51 | saftsack | thats the point ;) |
23:08.57 | saftsack | those adapters are at 60 bucks |
23:09.13 | JT | so next time what willl you do... buy the correct soekris model? :) |
23:09.42 | kombi | Fender: got it! debian's zaptel is no good.. |
23:09.43 | saftsack | i havent no soekris ... i have an asus wl-500gp |
23:10.51 | JT | i see |
23:11.28 | saftsack | http://www.interfacemasters.com/products/pci_tools/im380/index.html |
23:13.20 | saftsack | http://www.interfacemasters.com/products/pci_tools/mini_pci_to_pci/index.html |
23:13.29 | saftsack | this is what i want .... but where to buy? :( |
23:21.36 | *** join/#asterisk remmo (n=junk@203.62.147.3) |
23:23.10 | *** join/#asterisk Stridernzl (n=neville@125-239-186-131.jetstream.xtra.co.nz) |
23:27.31 | dlynes_laptop | saftsack: how about here? http://www.google.ca/search?hl=en&q=where+can+I+get+interface+masters+im300%3F&btnG=Search&meta= |
23:29.37 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
23:30.49 | lesouvage | . |
23:31.32 | lesouvage | sorry,just trying irc client on e61 |
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23:47.15 | DeL3e7 | IAX |
23:47.27 | DeL3e7 | how do i do that ;) |
23:49.55 | lesouvage | del37e: what do you mean or looking for |
23:50.35 | JT | dlynes_laptop: i was more interested in performance on the Netra T1 than anything else |
23:50.40 | obnauticus | hey umm |
23:50.48 | obnauticus | I got an inbound trunk on SIP |
23:50.51 | obnauticus | but im answering through IAX2 |
23:50.52 | JT | i'm sure installation isn't that hard unless you're trying to use a pci card |
23:50.55 | obnauticus | -- IAX2/obnautsoft-6 answered SIP/66.54.140.46-081d6690 |
23:51.04 | obnauticus | I can talk TO the person who's calling me |
23:51.08 | obnauticus | but they cannot talk to me |
23:51.11 | obnauticus | is there any reason why that is? |
23:52.33 | JT | nat or firewall |
23:52.41 | obnauticus | How would i fix that ? |
23:52.43 | obnauticus | like |
23:52.47 | obnauticus | im running a BSD based firewall... |
23:55.03 | JT | depends on where the problem lies i guess |
23:55.40 | lesouvage | Abnauticus: udp 5060 and rtp 10000-20000 should be open . |
23:56.10 | JT | udp 10000-20000 you mean... rtp runs on top of that |
23:56.14 | obnauticus | I can only forward TCP and UDP |
23:56.15 | obnauticus | lol |
23:56.24 | obnauticus | udp 10000-20000 |
23:56.25 | obnauticus | wow |
23:56.27 | obnauticus | lol. |
23:56.51 | JT | you shouldn't need to forward anything unless people from outside are connecting to your asterisk as a server |
23:56.56 | JT | why all the lols? |
23:57.01 | obnauticus | i lol all the time |
23:57.01 | obnauticus | irl too |
23:57.13 | obnauticus | I'm 15. |
23:57.17 | lesouvage | If you don t expect lots of connections you can lower the range |
23:57.23 | JT | making excuses now ;) |
23:57.31 | obnauticus | Where does it start binding sockets? |
23:57.31 | obnauticus | like |
23:57.35 | obnauticus | 10000 and up |
23:57.38 | obnauticus | starting at 10001 |
23:57.44 | JT | 5060 is bound for sip |
23:57.51 | JT | the rest are chosen per call |
23:58.06 | obnauticus | im not running 10000 clients so umm |
23:58.07 | obnauticus | is |
23:58.09 | obnauticus | 300 ok? |
23:58.12 | JT | err |
23:58.13 | obnauticus | 10000 10300 |
23:58.25 | JT | who really cares, unless you have a reason to, leave it at defaults |
23:58.34 | obnauticus | ya |
23:59.10 | obnauticus | it still no workie |
23:59.20 | obnauticus | -- Hungup 'IAX2/obnautsoft-7' |
23:59.20 | obnauticus | == Spawn extension (inbound, 508, 2) exited non-zero on 'SIP/66.54.140.46-081d6690' |
23:59.26 | JT | umm |
23:59.39 | JT | i think the problem is probably in the asterisk configuration |
23:59.47 | obnauticus | codec's? |
23:59.51 | JT | but first of all, what is the setup functionally? |
23:59.52 | JT | no. |
23:59.59 | obnauticus | Just for stupid stuff |