| 00:00.28 | *** join/#asterisk lyroy (~lyroy@modemcable117.123-202-24.mc.videotron.ca) |
| 00:01.16 | lyroy | Does someone know how to configura a Cisco 7940 Phone with asterisk (a working .cnf sample) please |
| 00:01.20 | lyroy | configure |
| 00:01.24 | StilexIP | eKo1: here's the sip debug http://pastebin.ca/5604 |
| 00:04.50 | StilexIP | scratch that |
| 00:04.58 | StilexIP | i'm getting 404's |
| 00:05.40 | Nethab | your client isn't sending credentials after * sends 401 auth required |
| 00:07.22 | liquidno2 | anybody have any tips for configuring a dial plan to allow a client to easily adjust an auto attendant? sort of like the voicemailmain? |
| 00:07.23 | *** join/#asterisk PTG123 (PTG123@ip68-106-19-249.ph.ph.cox.net) |
| 00:07.37 | Nethab | what do you mean adjust |
| 00:07.49 | liquidno2 | rerecord I guess is the easiests |
| 00:08.22 | Nethab | you mean change the voice prompts, like "listen carefully as our prompts have changed" |
| 00:08.49 | liquidno2 | Or just make it easy for them to change the autoattendant message quickly and easily |
| 00:09.10 | Nethab | if it's just one message file use Record |
| 00:09.24 | liquidno2 | the customer wants to have callers here the current network outages in the autoattendant |
| 00:09.33 | liquidno2 | I suggested having them press an option |
| 00:09.41 | liquidno2 | to take them to a voicemail box |
| 00:09.50 | liquidno2 | but they want it to be the second thing they hear |
| 00:10.04 | Nethab | well you can set an extension to playback a certain sound file, and set another to record to that file |
| 00:10.05 | liquidno2 | I guess the record application is the only way then |
| 00:10.51 | greg_work | liquidno2: theres AMP, if you want a web-based admin to configure autoattendant |
| 00:11.13 | liquidno2 | greg_work: I played with that a little... but I couldn't get spandsp to compile in debian... |
| 00:11.35 | liquidno2 | and it seems that some of the AMP stuff expected that to be there |
| 00:11.59 | greg_work | if you just want to play network status though, i'd recommend just putting in a Background(networkstatus) in the autoattendant, then using exten => *991,1,Record(networkstatus) exten => *992,1,Playback(networkstatus) |
| 00:12.10 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 00:12.14 | greg_work | they can just use *991 every day to change it |
| 00:12.29 | greg_work | liquidno2: the AMP install is particularly crappy |
| 00:12.34 | liquidno2 | greg_work: Yeah... I was looking for something a little more "hand holding" |
| 00:12.50 | greg_work | but offhand, i don't think it REQUIRES spandsp |
| 00:13.02 | liquidno2 | maybe some voice prompts before the record... |
| 00:13.06 | liquidno2 | hmmm |
| 00:13.11 | liquidno2 | maybe an AGI for that. |
| 00:13.36 | greg_work | liquidno2: well then .. exten => *991,1,Playback("about-to-record-network-status-message") ,2,Record(networkstatus) |
| 00:13.45 | greg_work | you dont need an agi, you could do it all in dialplan |
| 00:14.04 | greg_work | unless you wanted to have an automatic network status thing .. but that would probably be more of a pain in the ass than anything ;P |
| 00:14.11 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 00:14.19 | liquidno2 | greg_work: I was thinking like an autoattendant inside that... |
| 00:14.29 | *** join/#asterisk afrosheen (afrosheen@txprotoa8.august.net) |
| 00:14.30 | greg_work | i don't think there's an allison recording that says "a backhoe operator dug up our fiber line" |
| 00:14.32 | afrosheen | hello |
| 00:14.39 | liquidno2 | like you dial *999 and then press 1 to hear the current message and then 2 to record a new message |
| 00:14.49 | afrosheen | greg_work: are you sure? there's one about a weasel and monkeys |
| 00:14.53 | greg_work | you could do that in extensions.conf |
| 00:14.58 | greg_work | afrosheen: ;) |
| 00:14.59 | liquidno2 | greg_work: I actually have the walt disney world monorail guy :) |
| 00:15.11 | liquidno2 | he will say anything I want him to :) |
| 00:15.27 | afrosheen | anyone know how to handle call throughs? i.e. dial into the * server from a landline phone and use an extension to dial out |
| 00:15.36 | mikegrb | greg_work: actually there is one about a back hoe |
| 00:15.41 | afrosheen | lol |
| 00:15.46 | mikegrb | greg_work: check the extra sounds package |
| 00:16.21 | afrosheen | mikegrb: what's the name |
| 00:17.04 | mikegrb | afrosheen: check the txt, search for back hoe or back-hoe |
| 00:17.50 | afrosheen | don't see it in sounds-extra.txt |
| 00:17.57 | greg_work | yeah i don't see it either |
| 00:18.09 | mikegrb | I don |
| 00:18.16 | liquidno2 | nope not here |
| 00:18.17 | afrosheen | suckers, all of us |
| 00:18.18 | Nethab | i keep looking for ho's but haven't found one that i like |
| 00:18.22 | mikegrb | 't either, could have sworn that was one of the ones in the recent update |
| 00:18.22 | liquidno2 | totally |
| 00:18.27 | liquidno2 | I feel cheap and used... |
| 00:18.31 | liquidno2 | what else is new |
| 00:18.41 | afrosheen | so what about the call through, any clues |
| 00:19.13 | greg_work | afrosheen: fairly simple |
| 00:19.15 | Nethab | if you have an fxo gateway registered to your * box give it a context with exten => to dial out |
| 00:19.22 | afrosheen | greg_work: keep going |
| 00:19.22 | greg_work | afrosheen: you have an autoattendant ? |
| 00:19.31 | liquidno2 | afrosheen: so you want to pick and extension to use as basis for your callerID outbound? |
| 00:19.31 | afrosheen | greg_work: yeah my ivr is setup |
| 00:19.43 | afrosheen | no no wrong idea |
| 00:19.51 | greg_work | afrosheen: you'd want to do something like exten => 23853,1,goto(dialout,s,1) |
| 00:20.08 | greg_work | where 23853 is the 'secret code' to dialout |
| 00:20.09 | afrosheen | I wanna dial through * like pbx systems you dial into, enter a pin, then dial out |
| 00:20.10 | liquidno2 | damn it man... make up your mind! :) |
| 00:20.42 | tzanger | stevekstevek: you there yet :-) |
| 00:20.44 | Nethab | sipura spa-3000 has that built in |
| 00:21.32 | afrosheen | I used the example here http://www.voip-info.org/wiki-Asterisk+tips+call+through BUT ext. 500 is invisible to the ivr, works from the inside though |
| 00:22.03 | mikegrb | %to-report-system-network-down.gsm%To report a system or network down |
| 00:22.09 | mikegrb | that would should be useful |
| 00:22.43 | liquidno2 | why-no-answer-mystery is a personal favorite |
| 00:22.53 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 00:22.56 | mikegrb | %step-in-stream.gsm%You step in the stream, but the water has moved on. |
| 00:23.39 | liquidno2 | don't believe it... it is a trick! |
| 00:24.18 | yebo | anyone know of any SMS gateways one can peer with? |
| 00:24.40 | yebo | kram: heya |
| 00:25.19 | Mavvie | hmmm... enum code doesn't work on 1.0.5, but works on HEAD. Annoying. |
| 00:25.19 | greg_work | afrosheen: you need to put that exten=>500 line in the context of the IVR |
| 00:25.31 | yebo | kram: ive seen little to nothing in the listservs about the verisign sip7 service.. people have successfully used it with asterisk though right? |
| 00:25.40 | *** join/#asterisk bobx (~bobx@206.124.165.14) |
| 00:25.42 | afrosheen | greg_work: yeah I knew I needed to make it aware, thanks :) |
| 00:26.36 | *** join/#asterisk Dr-Linux (~sshah@202.125.141.6) |
| 00:26.41 | mikegrb | %moron.gsm%Moron! (said by an angry Allison) |
| 00:26.47 | Dr-Linux | hi all |
| 00:28.55 | sezuan | Very OT, does anybody succesfully uploaded a ringtone to a Giptel G100? |
| 00:30.42 | afrosheen | ringtones on hardware phones, lol |
| 00:30.53 | afrosheen | I've been trying to do that with my polycoms for months |
| 00:31.00 | greg_work | hehe, i like this telemarketer torture thing |
| 00:31.09 | Dr-Linux | anybody can help me, i wanna put a song, in unavailable msg and onhold ? |
| 00:31.17 | greg_work | i think i'll have to add that when I put * at home.. |
| 00:31.44 | greg_work | keep a blocked list and redirect everyone on the list to that dialplan. |
| 00:32.00 | sezuan | afrosheen: Yeah, I want the CTU ring :-) |
| 00:32.08 | greg_work | 800xxxxxxx for example ... |
| 00:32.09 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 00:32.10 | redder86 | greg_work: I just send my blocked telemarketers to my voice mail. |
| 00:32.17 | afrosheen | sezuan: get the cisco 7960 then :) |
| 00:32.22 | greg_work | redder86: oh but that script looks so much more fun :p |
| 00:32.34 | redder86 | greg_work: they never leave messages |
| 00:32.45 | liquidno2 | greg_work: yeah I read that last night |
| 00:32.46 | sezuan | afrosheen: I have it in the office, but for home use it's a bit to expensive. |
| 00:33.15 | liquidno2 | pretty funny stuff... and a good example how not to make an auto attendant :) |
| 00:33.15 | redder86 | greg_work: as soon as I signed up for the do-not-call list I stopped getting telemarketers almost entirely. But beginning about a month ago I started getting a lot more again. |
| 00:34.01 | liquidno2 | see |
| 00:34.09 | liquidno2 | I knew it was a trick |
| 00:34.10 | redder86 | greg_work: they're all using "TOLL FREE" as the Caller*ID Name and some toll-free number as the Caller*ID Number. |
| 00:35.14 | redder86 | greg_work: I think that they learned that people block them when they don't send any Caller*ID at all, so they started sending this bogus stuff. |
| 00:36.07 | PTG123 | ~seen czero |
| 00:36.08 | jbot | czero <~me@CPE0090f800c5b0-CM001225704b6e.cpe.net.cable.rogers.com> was last seen on IRC in channel #asterisk, 16d 22h 1m 46s ago, saying: 'blah bateris in my wireles mouse are dead :)'. |
| 00:36.15 | PTG123 | i am beginning to think he is dead :) |
| 00:36.28 | mikegrb | his batteries died too |
| 00:36.40 | PTG123 | must have gotten electrocuted putting them in |
| 00:36.44 | Nethab | i think the batteries in his mouse died and he can't click on IRX |
| 00:36.44 | Nethab | c |
| 00:37.10 | redder86 | that's what you get for typing IAX too much, you can't type IRC |
| 00:37.39 | liquidno2 | haha |
| 00:37.48 | redder86 | the hands develop a habit of following patterns, almost subconciously |
| 00:38.07 | liquidno2 | Yeah I have the same problem |
| 00:38.16 | *** join/#asterisk Tough_Nuts (~Tough_Nut@204.110.228.254) |
| 00:38.27 | liquidno2 | I go to type microsoft and end up typing shithead ever time. It is weird |
| 00:38.27 | Nethab | i keep typeing iac though |
| 00:38.28 | redder86 | if you type something over and over and over and over and over again and then try to type it slightly differently afterwards, it's almost impossible to do it with any accuracy |
| 00:39.17 | bjohnson | how do you comment out a section in html again? <! > ? |
| 00:39.33 | redder86 | bjohnson: <!-- blah --> |
| 00:40.01 | bjohnson | thns |
| 00:40.04 | bjohnson | thnx |
| 00:40.36 | liquidno2 | <script> /* blah */ </script> :) |
| 00:40.59 | liquidno2 | in the pike position |
| 00:40.59 | Nethab | <span style="display:none">blah</div> |
| 00:41.11 | Sedorox | lol |
| 00:41.14 | liquidno2 | Nethab: hehe |
| 00:41.21 | redder86 | Nethab: </span> |
| 00:41.23 | liquidno2 | although |
| 00:41.26 | liquidno2 | right :) |
| 00:41.29 | Nethab | no i like to close my spans with a div |
| 00:41.39 | redder86 | is that legit? |
| 00:41.39 | liquidno2 | that isn't correct |
| 00:41.57 | liquidno2 | no it isn't since a <div> is a bounding box and span isn't |
| 00:42.07 | Nethab | silly head, i was joking |
| 00:42.14 | liquidno2 | ah... Okay :) |
| 00:42.27 | liquidno2 | this is a * channel not #html or #css |
| 00:42.31 | liquidno2 | so I have to make sure :) |
| 00:42.44 | redder86 | anyone here got a color fax machine nearby? |
| 00:43.00 | afrosheen | greg_work: sweet, it's working, now how do I actually password protect that extension |
| 00:43.03 | FuRR_ | color fax machine? |
| 00:43.04 | Nethab | well i'm sure i have one within around 10 miles of me, it's not mine though |
| 00:43.06 | FuRR_ | is there such a beast |
| 00:43.14 | redder86 | FuRR_: of course |
| 00:43.16 | liquidno2 | redder86: I do... it is really just an inkjet officejet |
| 00:43.26 | FuRR_ | redder86: interesting |
| 00:43.32 | greg_work | afrosheen: change 500 to something unknown |
| 00:43.33 | redder86 | liquidno2: can you send me a fax? 8084475658 |
| 00:43.34 | FuRR_ | redder86: never dealt that much with fax machines |
| 00:43.47 | afrosheen | greg_work: security through obscurity? no thanks |
| 00:43.49 | FuRR_ | redder86: why would you want to fax in color anyway |
| 00:43.57 | liquidno2 | redder86: hmm... not sure if I can send in color or not... I never tried |
| 00:44.02 | redder86 | FuRR_: to get color pictures on faxes |
| 00:44.11 | FuRR_ | redder86: why wouldnt you just email it? |
| 00:44.17 | PTG123 | nor redder i will not fax you a color copy of my ass :) |
| 00:44.19 | greg_work | afrosheen: or, copy the [callthrough] context to [callthrough-password] |
| 00:44.23 | redder86 | FuRR_: because e-mail and fax are different |
| 00:44.34 | redder86 | PTG123: ;-) |
| 00:44.44 | FuRR_ | redder86: but if you can afford a color fax, couldnt you afford some sort of internet connection |
| 00:44.49 | Nethab | almost no one has a color fax machine so sending color faxes are less than econimical |
| 00:45.00 | FuRR_ | seems almost dumb |
| 00:45.10 | redder86 | FuRR_: why are you trying to argue this with me? people use fax machines still. Get over it. |
| 00:45.24 | greg_work | afrosheen: and have 500 go to that .. then remove the existing _X and # lines, and put in .. exten=>2563,1,Goto(callthrough,s,1) exten=>4836,1,Goto(callthrough,s,1) |
| 00:45.25 | FuRR_ | all that shit that gets sent to my fax machine everyday -- from day trades to home loans, id hate to waste color ink or toner on that.... |
| 00:45.25 | greg_work | etc |
| 00:45.35 | greg_work | you could put as many of those as you want and give people different pins |
| 00:45.48 | greg_work | oh, and instead of playing privacy-prompt, play just vm-password or something |
| 00:45.49 | redder86 | FuRR_: your fax machine is an open bin, then. If you get spam faxes you must filter just like you do with e-mail. |
| 00:46.10 | mikegrb | afrosheen: check out app_disa it supports authentication |
| 00:46.12 | FuRR_ | redder86: not arguing, just question why someone would own such a device, seems kinda pointless |
| 00:46.32 | liquidno2 | redder86: do you still need that color fax? |
| 00:46.56 | FuRR_ | redder86: im curious though, does it take any longer to fax a document |
| 00:47.21 | redder86 | FuRR_: in many industries fax is still the easiest and fastest way to communicate hard-copy signed or handwritten documents. Using a fax machine tends to be quicker and more reliable than scanning and e-mailing. Plus, the former tends to be more legally accepted than the latter. |
| 00:47.23 | PTG123 | you know redder develops hylafax, so he probably wants to test???!?!?! :) |
| 00:47.30 | mikegrb | afrosheen: it can even have a filename given with list of pin codes and will set the account code in the cdr to the pin number entered |
| 00:47.31 | redder86 | liquidno2: sure |
| 00:47.35 | afrosheen | mikegrb: cool thanks |
| 00:47.36 | redder86 | liquidno2: fax me still |
| 00:47.42 | PTG123 | ok i disagree with faxing quicker and easier then scanning :) |
| 00:47.43 | liquidno2 | redder86: number again? |
| 00:47.56 | PTG123 | since good xeroxers you could just ender in an email, click scan and be done ;) |
| 00:48.10 | redder86 | FuRR_: color faxes are JPEG compressed (well, or JBIG). Just depends upon the size of the image. But yes, color faxes tend to take longer than black 'n white. |
| 00:48.17 | redder86 | liquidno2: 8084475658 |
| 00:48.37 | liquidno2 | damn it |
| 00:48.39 | liquidno2 | hold on |
| 00:48.40 | FuRR_ | redder86: just seems kinda funny, considering unified messaging emails your fax to you anyway |
| 00:48.51 | redder86 | FurRR_: yup |
| 00:49.00 | FuRR_ | and like PTG123 said, most of the xeroxs out there allow you point and scan |
| 00:49.26 | PTG123 | well except of course they wouldn't do it in color, thats just silly |
| 00:49.26 | PTG123 | heh |
| 00:49.26 | redder86 | FuRR_: it's just a different way to communicate. |
| 00:49.50 | *** join/#asterisk WiFiGuy (~wilco@d009.max6-3.mke2.ticon.net) |
| 00:49.52 | redder86 | FuRR_: I think that you'll find that more offices have fax machines than internet connections. |
| 00:49.53 | *** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
| 00:50.09 | *** part/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
| 00:50.10 | liquidno2 | redder86: on its way |
| 00:50.14 | redder86 | :-) |
| 00:50.29 | FuRR_ | redder86: its just kinda funny when you think about both sides having unified messaging and faxing back and forth |
| 00:50.41 | redder86 | FuRR_: typically both sides don't, though |
| 00:50.53 | *** join/#asterisk brycec (~brycec@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 00:51.11 | FuRR_ | redder86: true, but in this application, say you were sending to * -- would it be admissable in court as a fax and not an email |
| 00:51.25 | redder86 | FuRR_: just think of a fax machine as a scanner+MUA built-together. It makes it really easy to scan+email. ... well, that's the end result, anyway. |
| 00:51.48 | brycec | Can someone help me with a T1 issue? I'm getting an OOF error and I don't know how to reframe, or whatever is required to fix it |
| 00:51.53 | FuRR_ | redder86: i guess i just get annoyed when end users will print out an email i sent to them, respond to it and fax it back to me |
| 00:51.59 | liquidno2 | redder86: how did it come out? |
| 00:52.06 | liquidno2 | I am curious myself |
| 00:52.13 | redder86 | FuRR_: the legality of fax depends upon state law. But I suspect that some states will respect a fax no matter how it is ultimately delivered to the recipient. |
| 00:52.24 | redder86 | liquidno2: chekcing |
| 00:52.43 | liquidno2 | was that an actual fax machine or asterisk with spandsp? |
| 00:52.52 | Dr-Linux | [classes] |
| 00:52.53 | Dr-Linux | default => quietmp3:/var/lib/asterisk/mohmp3 |
| 00:52.53 | Dr-Linux | ;loud => mp3:/var/lib/asterisk/mohmp3 |
| 00:52.53 | Dr-Linux | ;random => quietmp3:/var/lib/asterisk/mohmp3,-z |
| 00:52.53 | Dr-Linux | ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 |
| 00:52.53 | Dr-Linux | ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 |
| 00:53.15 | Dr-Linux | which line i should use for holdmusic ? |
| 00:53.34 | redder86 | liquidno2: it was HylaFAX using a Patton 2977 in Class 1. |
| 00:53.47 | liquidno2 | did it come out in color? |
| 00:53.54 | redder86 | liquidno2: it came through well. The X is red and words below it. Would you like me to send it to you by e-mail? |
| 00:54.03 | brycec | ok, nevermind my question. seems to have solved itself... |
| 00:54.26 | liquidno2 | redder86: sure |
| 00:54.32 | redder86 | liquidno2: email? |
| 00:54.36 | liquidno2 | liquidno2: tom@expresshosting.net |
| 00:54.48 | mikegrb | redder86: you should put it on www |
| 00:55.09 | liquidno2 | email is for the truely lazy :) |
| 00:55.23 | redder86 | liquidno2: do you want me to post it on http? |
| 00:55.39 | liquidno2 | redder86: I couldn't care less :) |
| 00:55.45 | liquidno2 | what ever is easiest |
| 00:55.58 | redder86 | well, I already e-mailed it, but mikegrb wants to see |
| 00:56.02 | mikegrb | :D |
| 00:56.20 | liquidno2 | Damn mikegrb... he didn't pay for the call :) |
| 00:56.25 | liquidno2 | I PAID! :) |
| 00:56.37 | mikegrb | heh |
| 00:57.11 | liquidno2 | I have wasted more money on experiemnts than that $0.05 |
| 00:58.05 | Mavvie | so. 1.0.5 now works too. |
| 00:58.23 | redder86 | liquidno2: mikegrb: http://faxservice.realestate808.com/fax000054278.pdf |
| 00:58.33 | *** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
| 00:58.45 | liquidno2 | damn... that came out rather good. |
| 00:58.53 | redder86 | liquidno2: thank you |
| 00:59.02 | kimo_sabe | so is there some reason I'm missing that asterisk doesn't allow sending a text part of "To's" in SIP INVITEs? |
| 00:59.15 | liquidno2 | redder86: most certainly welcome |
| 00:59.16 | redder86 | you can't do *that* with spandsp, I tell ya! |
| 00:59.23 | liquidno2 | good lesson in how to fax in color too |
| 00:59.44 | Dr-Linux | my softphone is onhold , and i'm listening music, but this music file is not in mohmp3 ? |
| 01:00.10 | liquidno2 | Dr-Linux: you must restart asterisk to have asterisk pull a new moh file |
| 01:00.13 | Sedorox | lol. I had that before |
| 01:00.27 | liquidno2 | unless there is something else that does that |
| 01:00.33 | Dr-Linux | Liquide: restart or reload ? |
| 01:00.35 | liquidno2 | because reload doesn't |
| 01:00.46 | redder86 | liquidno2: I've been able to test with HPs and Brothers, but nothing else yet. I don't know if other manufacturers commonly make color faxes. |
| 01:01.10 | liquidno2 | redder86: the only other company I can think of would be Lexmark |
| 01:01.13 | *** join/#asterisk Gronker (~Gronker2@adsl-220-89-99.ags.bellsouth.net) |
| 01:01.36 | Dr-Linux | Liquide: but there are only 3 mp3 files in mohmp3, but this holdmusic which is playing not in them ? |
| 01:01.46 | FuRR_ | so im wondering if you sent a color fax to a non color fax, would it be grey scale or black and white |
| 01:01.49 | Dr-Linux | Liquide: from where this file being detect ? |
| 01:02.13 | redder86 | FuRR_: the sending fax machine would have to convert the color to black & white. Dithered. Not greyscale. |
| 01:02.24 | FuRR_ | yuck |
| 01:02.33 | greg_work | can we just move in to the 1990's please? |
| 01:02.39 | *** join/#asterisk lyroy (~lyroy@modemcable117.123-202-24.mc.videotron.ca) |
| 01:02.57 | redder86 | FuRR_: there is also a "greyscale" support. But most fax machines prefer dithered MMR over grescale JPEG. |
| 01:02.59 | FuRR_ | so basically, you would have to know that the called party has a color fax |
| 01:03.00 | liquidno2 | I took two 1980's and went straight to Y2K |
| 01:03.04 | Dr-Linux | Liquide: from where this file being detect ? |
| 01:03.09 | redder86 | greg_work: color fax was 1990s. |
| 01:03.16 | liquidno2 | Dr-Linux: did you read the wiki? |
| 01:03.45 | greg_work | redder86: so was a million other ways of transporting images and text that is better than FAX |
| 01:03.46 | redder86 | FuRR_: usually the calling fax machine won't "scan in" the image until after the call is placed ... if you tell it to do color. THen it will scan it in while it performs the prologue handshaking and stuff. |
| 01:03.54 | FuRR_ | redder86: methinks that the whole color fax thing is really only for those stupid real estate flyers i get daily |
| 01:04.18 | greg_work | s/is/are/ |
| 01:04.35 | redder86 | greg_work: I haven't said that fax is the best way to do this kind of stuff. But people have fax machines, and people want to use fax machines, and people have their reasons to use fax machines. So I make a living by helping people use their fax machines. |
| 01:04.55 | redder86 | FuRR_: get Caller*ID on your fax line and start blocking calls. |
| 01:05.10 | liquidno2 | actually |
| 01:05.21 | FuRR_ | thats if my enterprise would spring to get caller id on their incoming trunks |
| 01:05.28 | redder86 | FuRR_: or report them to the FCC |
| 01:05.28 | liquidno2 | unsolicitated phone faxes is a federal crime |
| 01:05.34 | greg_work | redder86: well, fair enough. i just think .. color fax.. why bother |
| 01:05.37 | liquidno2 | $500 per instance |
| 01:05.48 | liquidno2 | I sued and won against InDemand |
| 01:06.17 | liquidno2 | BellSouth gave me a line that use to be the local Comcast fax number |
| 01:06.25 | liquidno2 | InDemand constantly sent me stuff... |
| 01:06.29 | FuRR_ | muwahahaha |
| 01:06.32 | liquidno2 | I called repeatedly and it didn't stop |
| 01:06.44 | liquidno2 | So I hooked up a fax machine... stocked it with paper |
| 01:06.49 | liquidno2 | and collected the faxes |
| 01:07.02 | liquidno2 | took them to a lawyer who was more than happy to represent me :) |
| 01:07.06 | greg_work | i pity the person that gets our old fax number |
| 01:07.42 | greg_work | it plays a 'number has been changed' recording for 6 months .. but after that... |
| 01:07.51 | redder86 | greg_work: there is a beauty in faxing in that both the receiver and the sender negotiate the formatting of the image. It almost guarantees that the receiver can view the image properly. Sometimes in e-mailing scans that's not possible. |
| 01:08.16 | liquidno2 | yeah... mime encoding is always funky |
| 01:08.21 | liquidno2 | digium is a case in point |
| 01:08.25 | liquidno2 | their billing system |
| 01:08.30 | redder86 | exactly. |
| 01:08.32 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 01:08.32 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 01:08.34 | liquidno2 | uses PEAR::Mime-Mail |
| 01:08.35 | greg_work | redder86: yeah, thats true. scanning technology is pretty crappy really |
| 01:08.41 | *** join/#asterisk Evanrude (~david@ip68-96-125-184.lu.dl.cox.net) |
| 01:08.50 | liquidno2 | and they use the method in Mime-Mail |
| 01:09.04 | liquidno2 | it butchers the mime header so badly that nobody can read it |
| 01:09.28 | greg_work | redder86: but theres also a lot of stuff that's in electronic form to begin with .. just send it as .txt or .pdf ... |
| 01:09.41 | liquidno2 | I wrote my own php class to handle mime mail... it is really easy once you understand it |
| 01:09.57 | redder86 | greg_work: well, fax isn't used for that kind of stuff as much any more. It's more for handwritten or signed documents. |
| 01:10.24 | greg_work | redder86: we get a lot of stuff that really doesn't need to be faxed |
| 01:10.52 | redder86 | greg_work: I'm sure that's true. But it gets faxed because that's what people know and are accustomed to. |
| 01:11.02 | greg_work | but yeah,it is fine for signed documents. but my original point . do we really need color? why bother? |
| 01:11.13 | PTG123 | all i know is we use digital docs for everything :) and email them whenever possible |
| 01:11.38 | *** join/#asterisk PMantis (~PrayingMa@65-37-0-159.nrp2.roc.ny.frontiernet.net) |
| 01:11.51 | redder86 | greg_work: I can imagine a few uses for signed or handwritten color documents. |
| 01:12.04 | redder86 | greg_work: granted, it's not going to be common. |
| 01:12.15 | FuRR_ | redder86: other than real estate listings...what else? |
| 01:12.22 | redder86 | greg_work: most documents aren't color to begin with |
| 01:12.58 | redder86 | FuRR_: a programmer may hand-draw a flowchart with color coding |
| 01:13.11 | FuRR_ | redder86: and then fax it??? |
| 01:13.36 | redder86 | FuRR_: if the receiver is at a color fax machine, then possibly, yes |
| 01:13.50 | FuRR_ | redder86: and why is he hand drawing...what 2bit company is he working for that cant afford OpenOffice |
| 01:13.54 | Sedorox | Feb 10 18:10:10 WARNING[32281]: asterisk.c:343 listener: Accept returned -1: Bad file descriptor ?????????? |
| 01:14.00 | Sedorox | I got 20 of them and it locked up a box |
| 01:14.28 | redder86 | FuRR_: don't misunderstand me - I'm not saying that it's the best way to communicate this kind of material. I'm not in the business of forcing people to do things the best way. I'm in the business of providing options to my clients. |
| 01:14.36 | PMantis | Anyone use BroadVoice with Asterisk? |
| 01:14.45 | PMantis | (that's on tonight?) |
| 01:14.49 | FuRR_ | redder86: i understand, but its one of those things that just boggles my mind |
| 01:15.00 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 01:15.03 | FuRR_ | redder86: like the warning on coffee saying "Caution, contents may be hot" |
| 01:15.13 | redder86 | FuRR_: okay, but I was just trying to explain their point of view |
| 01:16.26 | dan2 | PMantis: I work for broadvoice, does that count? |
| 01:16.34 | liquidno2 | heh |
| 01:16.45 | redder86 | FuRR_: I strongly believe that fax has it's place in some things, it gets overused, yes, but e-mail plain sucks with all of the filtering and spam and stuff to compete with. There is far, far less fax spam than e-mail spam. |
| 01:16.53 | JerJer | Caution, contents will burn your crotch |
| 01:17.09 | liquidno2 | Do not taunt happy fun ball. |
| 01:17.17 | dan2 | PMantis: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup |
| 01:17.31 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
| 01:18.19 | PMantis | dan2, I'm having continuouos reliability issues... I just got off the phone with Ron D... he's researching. |
| 01:18.24 | syslod | Hello |
| 01:18.31 | redder86 | FuRR_: if I send you an e-mail I have absolutely no way of knowing with any degree of surety that you received it correctly. If you don't reply, then it may have got caught in a spam filter or somehow or other got lost otherwise. |
| 01:18.42 | liquidno2 | too funny... I looked at my * logs when I did that fax to redder86... zapchan detected the fax but didn't know what to do with it... I am pretty sure I have faxdetect set to incoming only... |
| 01:18.45 | PMantis | dan2, Like... I can't ping proxy.bos.broadvoice.com.... no dNS response |
| 01:18.51 | FuRR_ | redder86: read_reciept |
| 01:18.57 | redder86 | FuRR_: with fax there is a degree of assurance that the image got through. |
| 01:19.13 | Sedorox | <PROTECTED> |
| 01:19.22 | redder86 | FuRR_: read_receipt doesn't nearly work half the time - depending on the MUA configuration |
| 01:19.24 | liquidno2 | FuRR_: read receipts aren't universally supported either |
| 01:19.41 | liquidno2 | neither are things like delivery confirmations |
| 01:19.45 | dan2 | PMantis: you shouldn't be using bos anyway |
| 01:19.48 | liquidno2 | at least not all smtps support it |
| 01:19.54 | dan2 | PMantis: thats my lab server, its not meant to be used by the public |
| 01:19.58 | FuRR_ | redder86: what assurance does a fax give the the user got it? |
| 01:19.59 | liquidno2 | sendmail does... and that is about the entire list :) |
| 01:20.08 | PMantis | dan2, LOL Ron *told* me to use it. |
| 01:20.16 | dan2 | PMantis: Ron D? is he one of our support guys? |
| 01:20.24 | FuRR_ | redder86: alot of times i faxed stuff and its gotten "lost" |
| 01:20.30 | redder86 | FuRR_: there is a confirmation signal that the receiver sends to the sender. |
| 01:20.32 | *** join/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 01:20.37 | MrEntropy | yo |
| 01:20.48 | FuRR_ | redder86: thats great, so the machine got it, but the document isnt done yet |
| 01:21.00 | PMantis | dan2, Yes.. Just got off the phone with him. |
| 01:21.12 | FuRR_ | redder86: the document needs to reach its intended victim..err recipient |
| 01:21.37 | FuRR_ | redder86: most faxes you still have to follow up with the age old phone call "did you get my fax?" |
| 01:21.47 | PMantis | dan2, I'm getting registration timeouts from chi, right now. |
| 01:21.55 | dan2 | PMantis: all asterisk guys are supposed to deal with me, not support guys |
| 01:22.05 | PMantis | dan2, Hah |
| 01:22.06 | dan2 | PMantis: support guys don't have a clue about asterisk atm |
| 01:22.07 | syslod | Any shell script genius out there? |
| 01:22.16 | dan2 | PMantis: ok, follow that link |
| 01:22.36 | MrEntropy | syslod: try #bash for the geniuses |
| 01:22.42 | syslod | :) |
| 01:22.43 | PMantis | dan2, I read that exact text elsewhere... Tried that setup. |
| 01:22.44 | dan2 | PMantis: can you msg me? |
| 01:22.52 | liquidno2 | syslod: what do you need? |
| 01:22.55 | dan2 | PMantis: then you didn't follow the directions |
| 01:23.27 | syslod | how do I take output of wget which will be 0 or 1 and do a condional on it? |
| 01:23.49 | liquidno2 | syslog the previously executed command will always return the exit status in $? |
| 01:23.55 | liquidno2 | so you can do EXIT=$? |
| 01:24.07 | liquidno2 | and then if $EXIT eq etc... |
| 01:24.10 | syslod | Yea but this isn't exit this is output of 0 or 1. |
| 01:24.45 | liquidno2 | is this to stdout? |
| 01:24.49 | syslod | wget is outputing string 0 or 1. Could be anything |
| 01:24.52 | syslod | Yes. |
| 01:24.56 | liquidno2 | then $1 |
| 01:25.23 | syslod | I was thinking $result = wget http://test.php would work but it doesn't |
| 01:25.48 | liquidno2 | syslod: then you aren't seeing stdout as that would catch it |
| 01:25.55 | liquidno2 | or you aren't calling it correctly |
| 01:26.13 | Darwin35 | <PROTECTED> |
| 01:26.16 | syslod | I'd like to call wget within the script and store what it returns in a var. |
| 01:26.18 | liquidno2 | are you calling it VAR=${wget......} or VAR=`wget.....` |
| 01:26.18 | Darwin35 | thats a nice find |
| 01:26.29 | syslod | var = wget |
| 01:26.39 | liquidno2 | well that will never work |
| 01:26.43 | syslod | :) |
| 01:26.45 | liquidno2 | you need to execute wget |
| 01:26.50 | FuRR_ | syslod: http://www.tldp.org/LDP/abs/html/ |
| 01:26.53 | liquidno2 | so do that with either the backticks |
| 01:27.00 | liquidno2 | or the new method |
| 01:27.18 | syslod | new method? is that $command? |
| 01:27.26 | liquidno2 | ${command} |
| 01:27.28 | liquidno2 | but yeah |
| 01:27.29 | FuRR_ | syslod: and i believe $? stores the return of the last command run |
| 01:27.39 | liquidno2 | that is the exit code |
| 01:27.50 | liquidno2 | he wants to capture the stdout output I think |
| 01:28.04 | liquidno2 | although I am not sure as 0 or 1 doesn't sound like stdout output to me |
| 01:28.08 | liquidno2 | at least from wget |
| 01:28.23 | liquidno2 | but then I don't know all the silent switches for wget by heart either |
| 01:28.25 | FuRR_ | unless the php script returns 0 or 1 |
| 01:28.25 | syslod | I have a php page that just sends a 0 or 1. |
| 01:28.36 | FuRR_ | aaaah |
| 01:28.43 | FuRR_ | thats a lil bit easer |
| 01:28.49 | liquidno2 | exit(1) |
| 01:28.50 | liquidno2 | ; |
| 01:29.08 | liquidno2 | now I am really confused though |
| 01:29.14 | liquidno2 | are you using PHP or bash? |
| 01:29.28 | FuRR_ | $VARIABLE = wget -v -o - |
| 01:29.30 | liquidno2 | because they both do the same thing (php is the more flexible of the two) |
| 01:29.52 | syslod | Ok. OK. I have a php script on one machine. I would like to have shell script on another that ran and depending on what wget to the php page return either do something or not. |
| 01:29.59 | FuRR_ | $VARIABLE = `wget -v -o - http://test.php` |
| 01:30.11 | syslod | FuRR_: Tried both of those. |
| 01:30.30 | FuRR_ | bah |
| 01:30.33 | liquidno2 | syslod: why not use ssh for something like this? |
| 01:30.34 | FuRR_ | do it in perl |
| 01:30.53 | liquidno2 | setup a shared key between the two servers... |
| 01:31.14 | liquidno2 | and allow the local machine to ssh into the remote machine and run the command and then you can capture the exit code locally |
| 01:31.28 | syslod | Shell can't do it? |
| 01:31.33 | liquidno2 | sure it can |
| 01:31.40 | liquidno2 | just do an ssh command |
| 01:31.56 | liquidno2 | google up remote ssh command and you will find a couple of how tos |
| 01:31.59 | FuRR_ | hehehehe |
| 01:32.05 | liquidno2 | SSH it the POWAH! |
| 01:32.09 | syslod | well i have 99% already done and ssh sucks cause it gives too much access the machine with a key. |
| 01:32.10 | FuRR_ | $VARIABLE = `wget -q -O - http://test.php` |
| 01:32.25 | liquidno2 | syslod: so createa specific user and lock it down with sudo |
| 01:32.42 | liquidno2 | no spaces |
| 01:32.53 | mikegrb | http://thegrebs.com/documentation/?&itemID=13 <-- read this |
| 01:33.03 | liquidno2 | VARIABLE=`wget -q -O http://test.php` |
| 01:33.04 | mikegrb | syslod: ^ that even says how to limit the key to only running one command |
| 01:33.35 | liquidno2 | mikegrb: thank you... I was feeling particularly lazy :) |
| 01:33.39 | FuRR_ | thanks Liquide |
| 01:33.46 | mikegrb | liquidno2: I wrote it ;) |
| 01:33.47 | FuRR_ | stupid tab completion |
| 01:34.06 | liquidno2 | FuRR_: like I said... i are the bash freak |
| 01:34.17 | liquidno2 | I wrote a load balancer for spamassassin in bash :) |
| 01:34.22 | FuRR_ | ewww |
| 01:34.30 | liquidno2 | worked really well actually |
| 01:34.31 | mikegrb | I have a bash book |
| 01:34.37 | syslod | ./pubast: line 3: =wget:: command not found |
| 01:34.39 | liquidno2 | used a combination of netcat and other stuff. |
| 01:34.44 | FuRR_ | i try not to use bash for anything more than changing ownership of directories when other peoples bash scripts go crazy |
| 01:34.52 | liquidno2 | syslod: whereis wget |
| 01:34.55 | harryvv | No idea mark spencer was such a young guy :) |
| 01:35.00 | liquidno2 | use the fully qualified path |
| 01:35.35 | liquidno2 | FuRR_: I have my bash script of the day competition with myself |
| 01:35.36 | syslod | Guess something simple turns complex in a hurry. :( |
| 01:35.55 | modulus_ | bleh |
| 01:36.17 | FuRR_ | liquidno2: yeah, i tend not to play with myself at work |
| 01:36.28 | liquidno2 | haha |
| 01:36.33 | liquidno2 | work from home |
| 01:36.43 | liquidno2 | nooners rule! :) |
| 01:36.46 | FuRR_ | and be home with the wife and kids? you crazy |
| 01:36.59 | liquidno2 | lock the door |
| 01:37.01 | FuRR_ | id rather face the slings and arrows of outrageous end users |
| 01:37.13 | liquidno2 | I love working from home |
| 01:37.34 | modulus_ | shakespear |
| 01:37.47 | modulus_ | hamlet |
| 01:37.55 | FuRR_ | modulus_: 2 points for you |
| 01:38.06 | modulus_ | it's "suffer the slings..." not "face" |
| 01:38.27 | liquidno2 | geek alert! :) |
| 01:38.43 | FuRR_ | yeah, i took some liberties with it |
| 01:38.56 | *** join/#asterisk forrestc (~fwc@206.127.78.199) |
| 01:39.07 | forrestc | Anyone from NuFone on? |
| 01:40.16 | syslod | For those that care: if you do VAR=(command) it works. |
| 01:40.27 | *** join/#asterisk robf (~robf@SP3-24.207.240.3.charter-stl.com) |
| 01:41.19 | hermie | liquideno2: you know that NO2 is never a liquid right? |
| 01:41.32 | liquidno2 | hermie: hence the irony of it all |
| 01:41.33 | hermie | well, almost never... its there in small amounts |
| 01:41.50 | liquidno2 | my other nick on efnet is stupidnic |
| 01:41.58 | hermie | good ol equilibria! |
| 01:42.26 | liquidno2 | and according to tests... I am far from that :) |
| 01:42.58 | hermie | an equilibrium system??? |
| 01:43.19 | *** join/#asterisk cobryce (~cobryce@ip68-99-209-169.ph.ph.cox.net) |
| 01:44.04 | Dr-Linux | <PROTECTED> |
| 01:44.40 | liquidno2 | hermie: no... stupid :) |
| 01:44.54 | liquidno2 | ie my nick |
| 01:44.57 | liquidno2 | stupidnic |
| 01:45.29 | hermie | liquideno2: I was kidding... I knew what you were saying :) |
| 01:45.35 | joebaker | Do I need asterisk running to get kphone to work with Vonage? |
| 01:45.44 | liquidno2 | and I am pretty sure I have stupidnic on freenode... I just can't remember my ident :) |
| 01:46.17 | *** join/#asterisk Feral_Kid (~feral_kid@red-corp-200.56.96.178.telnor.net) |
| 01:46.19 | liquidno2 | So I tried to come up with a nickname that was equally ironic. |
| 01:47.47 | Feral_Kid | Does anyone know if there are wifi phone available that allow for more than one proxy (ITSP) provider at a time? Or should I just set it up to go through my *, and set up a dialplan that will allow for multiple providers... |
| 01:48.05 | PMantis | seen dan2 |
| 01:51.38 | *** part/#asterisk PTG123 (PTG123@ip68-106-19-249.ph.ph.cox.net) |
| 01:53.03 | Dr-Linux | <PROTECTED> |
| 01:53.10 | *** join/#asterisk thepdakid (~vtrandal@c-24-8-106-135.client.comcast.net) |
| 01:53.55 | *** part/#asterisk JunK-Y (~junky@modemcable144.95-37-24.mc.videotron.ca) |
| 01:57.49 | *** join/#asterisk adrianhensler (~chatzilla@blk-222-123-93.eastlink.ca) |
| 01:58.09 | harryvv | Why would placing options wcfxo debug=1 |
| 01:58.37 | harryvv | eliminate the imagines IRQ conflic issue with the x100p? |
| 01:58.55 | harryvv | That option was placed in modules.conf and it resolved this issue. |
| 01:59.32 | harryvv | very strange. |
| 02:00.18 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 02:03.35 | *** join/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
| 02:17.57 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 02:25.12 | Darwin35 | damnit the broadvoice patch needs to be fixed for 1.0.5 |
| 02:25.15 | Darwin35 | grrr |
| 02:25.36 | iMediax | ? |
| 02:25.40 | file[laptop] | what if I told you it's already in there? |
| 02:25.53 | denon | wtf is broadvoice deviating from the standards? |
| 02:25.58 | iMediax | ya i thought it was |
| 02:26.03 | file[laptop] | denon: they're not |
| 02:26.07 | denon | so why the patch? |
| 02:26.18 | file[laptop] | it cleaned up chan_sip behavior slightly |
| 02:26.24 | denon | ah |
| 02:26.27 | file[laptop] | allows it to reuse previous authorization data in registrations |
| 02:26.34 | Darwin35 | the patch is not iin asterisk-1.0.5 release |
| 02:26.35 | file[laptop] | so instead of going REGISTER -> Unauthorized -> REGISTER -> OK |
| 02:26.39 | denon | ah ic |
| 02:26.41 | Darwin35 | it might be in the cvs |
| 02:26.45 | denon | is that spec? |
| 02:26.55 | file[laptop] | you can reuse authorization details |
| 02:27.00 | denon | huh |
| 02:27.00 | file[laptop] | ATAs do it sometimes |
| 02:27.05 | denon | makes sense |
| 02:27.24 | file[laptop] | it just lessened the traffic on their servers |
| 02:27.43 | Darwin35 | I just wish they would pass the rining off to the sip devices better |
| 02:28.02 | Darwin35 | I hate when you dial and get dead air till some one answers or you get a vm |
| 02:28.17 | file[laptop] | Darwin35: but yes, the broadvoice patch is in there... |
| 02:28.28 | file[laptop] | it's been in stable for a good, 2 months |
| 02:28.47 | file[laptop] | actually 1 month 10 days |
| 02:29.01 | Darwin35 | ? |
| 02:29.03 | harryvv | is this your only * box? |
| 02:29.26 | Darwin35 | no I have 2 now but its my main 1 |
| 02:29.35 | file[laptop] | Darwin35: you said the patch is not in the asterisk-1.0.5 release, I'm telling you it is :) |
| 02:29.48 | Darwin35 | hmm ok |
| 02:31.56 | harryvv | file, Two days ago had some serios digium 100p problems. JUST got it resolved now. Now im getting absolutly no errors or warnings and three notice messages. |
| 02:32.19 | harryvv | file talking to you :) |
| 02:33.18 | adrianhensler | are the clone digium FXO cards (ebay under $10) okay or are they junk? |
| 02:33.27 | harryvv | ad...stay away from them |
| 02:33.32 | Darwin35 | ok I am on 1.0.3 for some reason . the upgrade did not happen |
| 02:33.36 | harryvv | try and get the real thing used. |
| 02:34.15 | *** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au) |
| 02:35.37 | adrianhensler | I'll keep my eyes open then, thanks |
| 02:35.46 | harryvv | have the seller get a pic of it. also, before asking that ask him what city he lives in then if he says newark nj then ask him to put it on top of the towns paper and get a pic of it. That way he is not passing off somone elses photo of another card. I dont think this is a digium card I have yet it has a digium sticker on the back. |
| 02:37.26 | harryvv | adianhelsler, also you need a fks port or a ipphone to check voicemail. |
| 02:37.32 | harryvv | Or a softphone like xlite. |
| 02:37.41 | harryvv | better to get the phone. |
| 02:37.47 | *** join/#asterisk Chrisf0rd (~chris@dhcp024-209-236-189.cinci.rr.com) |
| 02:38.03 | Chrisf0rd | Hello, I got one for ya's |
| 02:38.16 | adrianhensler | yes, I am just at the playing stage now. got asterisk installed and running and working with a softphone. |
| 02:38.40 | Chrisf0rd | why is when I change my greeting I have no access to my 7777 Calling card system from an outside line. |
| 02:39.12 | Chrisf0rd | 7777 => Ext 7777 cannot access it now that I have chanegs teh opening greeting |
| 02:41.30 | adrianhensler | I guess you stumped everyone Chrisf0rd |
| 02:41.56 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 02:42.27 | adrianhensler | I wonder if that changes a file permission incorrectly. long shot; even I can't see that happening |
| 02:43.24 | Darwin35 | man I wish max would get back on the ball and update the pport |
| 02:43.34 | Darwin35 | the patches keep failing |
| 02:44.14 | liquidno2 | damn you miserable excuses for a patch, why must you be so mediocre |
| 02:44.46 | Darwin35 | no this is the fbsd port |
| 02:44.52 | Darwin35 | its still at 1.0.3 |
| 02:44.57 | Darwin35 | in the ports tree |
| 02:45.08 | *** join/#asterisk hardocp (~hardocp@ool-44c1f7b4.dyn.optonline.net) |
| 02:45.31 | hardocp | good evening |
| 02:45.41 | Darwin35 | I keep a fbsd ver and a linux ver running at all times |
| 02:45.44 | liquidno2 | [H] in the house |
| 02:46.34 | hardocp | does anyone know how to limit the amount of concurrent iax connections between my * server and my voip provider? |
| 02:46.35 | Mavvie | Darwin35: send-pr the patches for the update and max will be happier. |
| 02:46.38 | forrestc | Hello everyone. |
| 02:47.00 | forrestc | Anyone know of a level3 reseller (or someone else with Helena, MT DID's) which does online setup? |
| 02:47.29 | forrestc | I.E acct is setup immediately? |
| 02:48.47 | *** join/#asterisk ManxPower (~eric@72-254-142-96.client.stsn.net) |
| 02:48.52 | Darwin35 | I have to go threw the current patches and find out wich are needed and wich are not |
| 02:49.10 | Darwin35 | to get 1.0.5 to compile |
| 02:49.22 | liquidno2 | ouch and so max is the port mainter |
| 02:49.23 | Darwin35 | going to be some work |
| 02:49.41 | Darwin35 | maxsobila is |
| 02:49.44 | Darwin35 | yes |
| 02:50.51 | thepdakid | Does anyone here recommend buying a prebuilt IP PBX (like VoIP PBX in a Box) versus installing Asterisk "from the ground up"? |
| 02:51.09 | liquidno2 | I do it because it is fun |
| 02:51.10 | Mavvie | thepdakid: finger pointing is easier when you buy. |
| 02:51.15 | liquidno2 | and I have a need for it |
| 02:51.33 | liquidno2 | So I get to tinker... and I get to play... |
| 02:51.34 | hardocp | have you seen switchvox yet? |
| 02:51.38 | liquidno2 | and that works for me |
| 02:51.43 | Darwin35 | asterisk is better makes you learn as you admin |
| 02:52.08 | hardocp | its an all in one asterisk solution |
| 02:52.17 | hardocp | but |
| 02:52.24 | hardocp | its tied to hardware purhcase |
| 02:52.31 | liquidno2 | eh |
| 02:52.32 | hardocp | at least it seems to be for now |
| 02:52.47 | hardocp | they sell you a dell server and their asterisk gui |
| 02:52.56 | forrestc | Geez I hate the phone company ... |
| 02:53.04 | forrestc | *78 service sucks. |
| 02:53.24 | Darwin35 | what asterisk gui ? |
| 02:53.27 | hardocp | you could also try asteriskathome |
| 02:53.32 | hardocp | its their own |
| 02:53.38 | Darwin35 | grrr |
| 02:53.44 | hardocp | its pretty nice |
| 02:53.46 | hardocp | actually |
| 02:54.32 | hardocp | does anyone know about limiting iax trunks here? |
| 02:54.33 | Darwin35 | I need a asterisk gui for my pbx hardware/box I made |
| 02:54.39 | Darwin35 | i needs a web side |
| 02:54.45 | hardocp | amp |
| 02:54.52 | hardocp | is pretty solid |
| 02:54.56 | Darwin35 | it uses a serial port/term right now |
| 02:55.03 | hardocp | nice gui |
| 02:55.06 | Darwin35 | url for amp |
| 02:55.27 | hardocp | amp.voxbox.ca |
| 02:55.28 | *** join/#asterisk Q-At-Home (~Queue@S0106000c41bb87af.ed.shawcable.net) |
| 02:55.32 | liquidno2 | I gave up on amp |
| 02:55.36 | hardocp | huh? |
| 02:55.38 | hardocp | ahy? |
| 02:55.40 | liquidno2 | the install is horid |
| 02:55.43 | hardocp | ya |
| 02:55.51 | liquidno2 | very rh specific |
| 02:55.52 | hardocp | asteriskathome makes it alot easier |
| 02:55.59 | hardocp | the install that is |
| 02:56.02 | forrestc | Who in there right mind would design a service which you couldn't specify a Toll free number as the destination? |
| 02:56.28 | forrestc | Qwest sucks. |
| 02:56.40 | hardocp | liquidno what do you use? |
| 02:56.45 | hardocp | if anything at all |
| 02:57.57 | Darwin35 | ok reading about it now thnks |
| 02:59.42 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.133) |
| 03:00.28 | *** join/#asterisk Pressy (~Pressy@pD9E17C19.dip.t-dialin.net) |
| 03:00.49 | ManxPower | Well the Stealth Asterisk Install didn't go terribly well. |
| 03:02.03 | liquidno2 | man you can't leave us hanging with an intro like that ManxPower |
| 03:02.10 | adrianhensler | hardocp: http://www.voip-info.org/wiki-Asterisk+bounty+IAX+incoming-outgoing+limit ? |
| 03:02.17 | ManxPower | I have a question. Does anyone know how to do a clear cannel cross connect between channels on a Digium T-1 card? |
| 03:02.27 | ManxPower | liquidno2, Of course I can. |
| 03:03.09 | liquidno2 | ManxPower: what are you doing... it sounds like fun... like a T-Bird analyzer in a box or something fun like that |
| 03:03.20 | hardocp | adrian does that mean that this is not going to be possible to do? |
| 03:03.35 | ManxPower | liquidno2, putting Asterisk between the telco and the PBX. |
| 03:03.49 | liquidno2 | For what purpose? |
| 03:03.58 | liquidno2 | like a CSU? |
| 03:03.58 | ManxPower | BUT the telco is providing internet service on some of the channels on the PBX that we have to pass directly to the PBX. |
| 03:04.06 | liquidno2 | AHHHH |
| 03:04.11 | liquidno2 | so a channel demuxer |
| 03:04.11 | adrianhensler | sorry man - I just did a quick google - I don't know enough to even guess. The way I read that it looks like there is no easy way. seems like that would be something people would have implemented right away to me? |
| 03:04.11 | ManxPower | Liquide, So we can do clever toll bypass and call routing. |
| 03:04.51 | hardocp | ya its a pita |
| 03:05.30 | hardocp | it seems like some of the voip providers have started limiting the iax channels on their end to like 5 max |
| 03:05.42 | adrianhensler | hardocp - look here for another poosilbity: http://lists.digium.com/pipermail/asterisk-users/2005-January/083227.html |
| 03:05.42 | hardocp | but other seem to imply an ulimited amount |
| 03:05.47 | hardocp | which would not be good |
| 03:06.01 | ManxPower | hardocp, Generally if you pay per-min you get unlimited calls. |
| 03:06.53 | adrianhensler | I'm in Canada and having a hard time deciding on a provider for some cheap play-time. looking at iax.cc . |
| 03:07.38 | Q-At-Home | greets all |
| 03:07.40 | hardocp | voipjet gives some free time |
| 03:07.43 | hardocp | adrian |
| 03:07.50 | hardocp | if you want to play |
| 03:07.54 | adrianhensler | saw that ty though. I had forgotten |
| 03:08.07 | bjohnson | adrianhensler: hi |
| 03:08.27 | bjohnson | what's that bounty link? |
| 03:08.50 | Q-At-Home | anyone know how to filter caller id based on the name instead of the number? ala "s/4422281,5,Goto(blah)" but use a name instead? |
| 03:09.03 | hardocp | hmm interesting adrian |
| 03:09.11 | *** join/#asterisk Dickson (~loti@203.118.44.103) |
| 03:09.14 | mikegrb | Q-At-Home: don't believe it is possible to do it like that, you could use GotoIf |
| 03:09.27 | Q-At-Home | hrmm |
| 03:09.29 | adrianhensler | it does look promising. odd that it wasn't mentioned in the first link? damn, I wish I knew more |
| 03:09.41 | bjohnson | hardocp: if you're trying to limit how many voip calls happen .. the superdial macro is all set up for that |
| 03:09.50 | Q-At-Home | we have a fax spammer here going hardcore at my phone number.. and the local telco wont do anything about it |
| 03:10.00 | Q-At-Home | for now I just redirect fax to hangup |
| 03:10.05 | mikegrb | Q-At-Home: so filter by number |
| 03:10.12 | Q-At-Home | the number is from a large pool |
| 03:10.19 | Q-At-Home | its never the same twice |
| 03:10.25 | mikegrb | oh |
| 03:10.25 | ManxPower | Q-At-Home, Why not redirect the fax to the local FBI office, or even better yet the FCC Fax number? |
| 03:10.25 | mikegrb | that sucks |
| 03:10.26 | bjohnson | and the name would be? |
| 03:10.32 | tzanger | Q-At-Home: I bet you have a good idea what the pool is |
| 03:10.33 | tzanger | use some scripting |
| 03:10.41 | bjohnson | dump them into an ivr |
| 03:10.44 | Q-At-Home | but the name is always "DATAMED SOFTWR" |
| 03:10.54 | ManxPower | Well the canadian equivs of those, at least. |
| 03:10.58 | bjohnson | what country? |
| 03:11.02 | Q-At-Home | Canada |
| 03:11.03 | ManxPower | Q-At-Home, Doesn't .ca have a junk fax law? |
| 03:11.15 | Q-At-Home | sure, between 9-6pm only |
| 03:11.21 | Q-At-Home | after 9-6 not allowed |
| 03:11.53 | ManxPower | Q-At-Home, Change your phone number? |
| 03:11.57 | Q-At-Home | hell no :) |
| 03:12.22 | hardocp | how does super dial work? |
| 03:12.25 | tzanger | find datamed's' fax number and use app_txfax to send them 5000 pages of black |
| 03:12.30 | Q-At-Home | mmmmm |
| 03:12.36 | bjohnson | hardocp: you got the wiki page? |
| 03:12.38 | Q-At-Home | that would be fun |
| 03:12.45 | Q-At-Home | super dial? |
| 03:12.49 | tzanger | outside 9-5, of course |
| 03:12.51 | bjohnson | hardocp: give me the url so we're looking at the same thing |
| 03:13.00 | hardocp | url of what? |
| 03:13.08 | bjohnson | the superdial macro on the wiki |
| 03:13.20 | bjohnson | because it has instructions right it in |
| 03:13.24 | adrianhensler | http://www.voip-info.org/tiki-index.php?page=Superdial%20macro |
| 03:13.29 | bjohnson | in it |
| 03:13.31 | hardocp | http://www.voip-info.org/tiki-index.php?page=Superdial%20macro |
| 03:13.36 | hardocp | lol |
| 03:14.20 | bjohnson | see the usage instructions? and the description of the args? |
| 03:14.26 | Dickson | I am trying to make a SIP call through my cisco 2600 pstn, but I get a "488 Media Not Acceptable" error. Anyone knows what's wrong? |
| 03:14.46 | file[laptop] | you're using a codec that it won't speak |
| 03:14.51 | bjohnson | now .. are there any questions? |
| 03:14.57 | hardocp | shweet |
| 03:14.58 | hardocp | wow |
| 03:15.04 | file[laptop] | thus, Media Not Acceptable |
| 03:15.05 | hardocp | so that solves 2 problems for me |
| 03:15.20 | hardocp | i can also use this to group iax trunks |
| 03:15.23 | hardocp | right? |
| 03:15.35 | Dickson | file[laptop] : hmm .. so i need to enable SIP on the cisco pstn then |
| 03:15.50 | file[laptop] | Dickson: you need to use a codec that the 2600 supports |
| 03:16.08 | *** join/#asterisk sudoer (~sudoer@65.75.148.190) |
| 03:16.13 | Dickson | codec as in voice codec ? |
| 03:16.17 | sudoer | is asterisk supposed to act like a piece of shit? |
| 03:16.17 | file[laptop] | yes |
| 03:16.18 | adrianhensler | protocol does not equal codec |
| 03:16.25 | Dickson | thanks |
| 03:16.48 | bjohnson | hardocp: not exactly group them .. you would have to dial each trunk in sequence .. but it will limit the number of calls (grouped in whatever way you want .. but I use pstn and voip) |
| 03:16.50 | Q-At-Home | depends if you run the "act-like-a-piece-of-shit" application in your dial plan |
| 03:17.06 | *** join/#asterisk charles___ (~charles@64.35.168.55) |
| 03:17.26 | Q-At-Home | so suddenly my vonage caller ID starts working on port 1 today |
| 03:17.29 | bjohnson | or live the "act-like-a-piece-of-shit" attitude |
| 03:17.33 | Q-At-Home | I figure they did a mass firmware update |
| 03:17.41 | netsurfer | anyone seen this before: NOTICE[15972]: app_dial.c:911 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) |
| 03:18.14 | sudoer | I switched out one 4 port fxo card for a brand new one becuase the other one was flaky and digium sent me a new one, now its acting all stupid , i call form my cell phone to here and i can answer, I call form internal to our external number and the phone rings, but I answer from another phone but the phone I dialed from continues to ring |
| 03:18.21 | bjohnson | not sure if the "cause" is the same but I've seen that for a variety of reasons |
| 03:19.06 | Q-At-Home | sudoer: are you doing an "answer" in your dial plan? |
| 03:19.08 | Mavvie | you should answer the phone which rings. |
| 03:19.15 | sudoer | all the phones here ring |
| 03:19.34 | Q-At-Home | sudoer: are you doing an "answer" in your dial plan? |
| 03:19.35 | sudoer | yes ,there is an answer, everythign was working fine, I only changed a card out for the exact same one |
| 03:19.37 | bjohnson | damn .. openoffice devs are taking years to implement my suggested features. Don't they recognize brilliance? |
| 03:19.49 | sudoer | exten=>s,1,Dial(SIP/main&SIP/con&SIP/mike&SIP/speaker,10) |
| 03:20.08 | bjohnson | "All your phone are belong to us" |
| 03:20.14 | Silik0n | why dont you implement them and submit a patch? |
| 03:20.26 | netsurfer | if exten=>s,1 is Dial then what is answer ? |
| 03:20.42 | adrianhensler | the end of Lawnmower Man. all the phones in the world ring. |
| 03:20.46 | Q-At-Home | I'm guessing the answer isnt in the default context |
| 03:20.51 | Q-At-Home | thus, it never picks up |
| 03:20.54 | harryvv | Whats a common reason why after making three seperate calls that the other party looses the voice link with me as if the system goes simplex ? |
| 03:21.02 | harryvv | I hear them thay dont hear me. |
| 03:21.02 | bjohnson | netsurfer: is that a trick question? |
| 03:21.26 | Q-At-Home | harryvv: are the phones reinviting? |
| 03:21.26 | netsurfer | bjohnson - shouldnt answer come before dial ? |
| 03:21.40 | sudoer | it is in the default context, this setup has been working for months, I didnt change any config files, just changed out a card |
| 03:21.43 | harryvv | what do you mean by reinviting? |
| 03:21.44 | bjohnson | netsurfer: 1. not if you're dialing out |
| 03:21.48 | iMediax | why would u put the 2 on the same exten? |
| 03:21.49 | liquidno2 | oh I am sorry all answers must be in the form of a question. "THEN WHY THE HELL DO YOU CALL IT AN ANSWER DAMN IT!" |
| 03:21.55 | bjohnson | netsurfer: 2. not if you don't want to answer first |
| 03:21.59 | Q-At-Home | sudoer: can you humor us and check |
| 03:22.17 | Q-At-Home | I think your goal is to get it working, not to debate the merits of an "answer" statement |
| 03:22.20 | liquidno2 | good thing it is light weight |
| 03:22.41 | sudoer | I justed checked again, it is |
| 03:22.41 | sudoer | it is in default |
| 03:22.50 | Q-At-Home | I do tech support all day for users that I have to beat answers out of... I dont want to do the same here for free |
| 03:23.00 | bjohnson | you DO NOT have to answer before you dial extensions to have them ring |
| 03:23.05 | Q-At-Home | paste your extens to pastebin.ca |
| 03:23.08 | netsurfer | forget it |
| 03:23.12 | Q-At-Home | bjohnson: no, you dont |
| 03:23.15 | bjohnson | if someone answers the ringing phone .. THAT is the answer |
| 03:23.25 | hardocp | bjohnson do you do * consulting at all? |
| 03:23.29 | Q-At-Home | but you do need to answer BEFORE you pass the call to another sip device no? |
| 03:23.32 | harryvv | Q-at-home tech support for what technoligy? |
| 03:23.38 | ManxPower | You seldom ACTUALLY need an run Answer |
| 03:23.44 | Q-At-Home | sadly, computers |
| 03:23.45 | bjohnson | Q-At-Home: no |
| 03:23.46 | ManxPower | Q-At-Home, Why? |
| 03:23.56 | Q-At-Home | I've never had mine work without it |
| 03:24.01 | harryvv | Q thats okay :) |
| 03:24.03 | Q-At-Home | my pots are on a tdm 4 port fxo |
| 03:24.05 | liquidno2 | Answer is good for faxdetect |
| 03:24.11 | ManxPower | Q-At-Home, Then your SIP devices suck. 8-) |
| 03:24.17 | sudoer | mine is dial, then answer, then background |
| 03:24.18 | Q-At-Home | hahah, yes they probably do |
| 03:24.24 | bjohnson | Q-At-Home: here the answers are easily available .. here you have to beat the questions out of people |
| 03:24.27 | iMediax | why would you put Answer and Dial on the same exten? |
| 03:24.47 | iMediax | uhhh nm... thought of a reason |
| 03:24.49 | Q-At-Home | in default, the first s should be an answer |
| 03:24.51 | ManxPower | Running Answer + Wait(1) is useful before Backgroun() and Playback() if the audio is cut off for the first second or so. |
| 03:24.52 | liquidno2 | iMediax: self torture :) |
| 03:25.04 | Q-At-Home | or am I wrong |
| 03:25.14 | harryvv | btw, I think its possible to link 2,3,4 or more pstn to voip calls together though say iax.cc? that would really make my familly think ive broken into the phone company or something ;) |
| 03:25.16 | liquidno2 | Anybody used that NVBackground yet? |
| 03:25.22 | iMediax | no it dont have to be first |
| 03:25.23 | ManxPower | Since Background and Playback both ANSWER THE LINE AUTOMAGICALLY |
| 03:25.30 | Q-At-Home | hahaha |
| 03:25.35 | ManxPower | In fact so does VoiceMail and VoicemailMain! |
| 03:25.36 | liquidno2 | the guy that published it wants you to email him |
| 03:25.39 | bjohnson | Q-At-Home: just wait(), dial(),vm(b) .. if someone answers then good .. use a timeout in the dial to go to unavailable vm. |
| 03:25.40 | liquidno2 | to get the code |
| 03:26.03 | harryvv | Q-at-home tell me what is this caretoinvite? |
| 03:26.13 | Q-At-Home | no idea |
| 03:26.19 | harryvv | reinvite |
| 03:26.21 | harryvv | I mean |
| 03:26.28 | liquidno2 | wait(for-it),wait(for-it)... |
| 03:26.34 | Q-At-Home | two sip devices will try to talk directly |
| 03:26.39 | Q-At-Home | NAT breaks this |
| 03:26.41 | bjohnson | yes |
| 03:26.42 | Q-At-Home | (or used to) |
| 03:26.57 | Q-At-Home | oh wait |
| 03:27.08 | bjohnson | NAT and sip don't cooperate |
| 03:27.09 | Q-At-Home | I answer() so the telco voicemail wont take over :) |
| 03:27.13 | Q-At-Home | thats why I have it |
| 03:27.16 | harryvv | Again, my connection has dropped 1/2 of the conversation at times where I can hear the called party but thay cannot hear me. |
| 03:27.17 | bjohnson | NAT can be overcome in some cases |
| 03:27.26 | Q-At-Home | harryvv: what sort of devices? |
| 03:28.02 | harryvv | Q-at-home xlite-> * -> iax.cc > Qwest |
| 03:28.15 | Q-At-Home | xlite isnt iax |
| 03:28.19 | Q-At-Home | ohh |
| 03:28.20 | Q-At-Home | sorry |
| 03:28.21 | bjohnson | harryvv: meetme is the call conferencing app in *. It will connect many calls from whatever channels you have into one big group |
| 03:28.27 | harryvv | okay its sip |
| 03:28.36 | Q-At-Home | I'm biased... I hate xlite |
| 03:28.38 | Q-At-Home | :) |
| 03:28.39 | bjohnson | the dev meeting today was such with up to 23 concurrent participants |
| 03:28.43 | liquidno2 | you don't say? :) |
| 03:28.53 | harryvv | bj, so iax.cc could call many different pstn calls and link them together then. |
| 03:28.53 | liquidno2 | bjohnson |
| 03:29.06 | liquidno2 | bjohnson: did anybody actually record it? :) |
| 03:29.13 | liquidno2 | So the rest of us could hear it? |
| 03:29.23 | bjohnson | Liquide: I have no idea |
| 03:29.29 | Q-At-Home | who runs iax.cc |
| 03:29.35 | bjohnson | harryvv: yes |
| 03:29.35 | harryvv | Q-at-home why is that. |
| 03:29.43 | bjohnson | Q-At-Home: sixtel |
| 03:29.50 | Q-At-Home | ok, who runs sixtel |
| 03:29.54 | Q-At-Home | :) |
| 03:30.00 | Q-At-Home | I like to know my vendors |
| 03:30.09 | bjohnson | take them out to dinner |
| 03:30.21 | liquidno2 | and now vendor number one tell us a little about yourself |
| 03:30.49 | liquidno2 | "well I am a voip vendor with plans for global domination." |
| 03:30.50 | Q-At-Home | I like long walks... |
| 03:30.53 | Q-At-Home | hahaha |
| 03:30.56 | Q-At-Home | no |
| 03:31.08 | Q-At-Home | I just want cheap IAX termination in Edmonton, Alberta |
| 03:31.24 | hardocp | i want cheap iax with lnp |
| 03:31.26 | Q-At-Home | from a place with some "big backing" instead of joe six-pack |
| 03:31.30 | bjohnson | Q-At-Home: termination (outgoing) or DID (incoming)? |
| 03:31.39 | Q-At-Home | both |
| 03:31.42 | liquidno2 | Joe six is the best telco evah! |
| 03:31.48 | Q-At-Home | I use nufone for outgoing |
| 03:31.53 | Q-At-Home | works well |
| 03:31.54 | bjohnson | termination is easy .. DIDs are harder to find |
| 03:32.00 | hardocp | ya |
| 03:32.05 | bjohnson | so use nufone for termination |
| 03:32.08 | hardocp | and what about LNP |
| 03:32.12 | hardocp | and iax |
| 03:32.15 | harryvv | What are the procedures to link several pstn calls to iax.cc and then to a * soft/hardphone setup ? |
| 03:32.16 | bjohnson | and see what you can find for DIDs |
| 03:32.17 | Q-At-Home | I could care less about LNP |
| 03:32.27 | Q-At-Home | I pay for a telco forward on my main number |
| 03:32.29 | adrianhensler | q-at-home - did you check direct.freeworldtel.com? |
| 03:32.31 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4117547.sympatico.ca) |
| 03:32.36 | NormAst | Hi all |
| 03:32.40 | liquidno2 | lnp |
| 03:32.41 | liquidno2 | haha |
| 03:32.49 | hardocp | ?? |
| 03:32.55 | Q-At-Home | adrianhensler: problem is none of these people provide "big backing" info |
| 03:33.00 | liquidno2 | I just laugh at lnp |
| 03:33.03 | hardocp | bc |
| 03:33.06 | Q-At-Home | dont get me wrong, I like the local guy trying to make it |
| 03:33.06 | liquidno2 | it is suck a non issue |
| 03:33.09 | liquidno2 | such |
| 03:33.14 | liquidno2 | fredian slip there :) |
| 03:33.22 | hardocp | i dont understand |
| 03:33.23 | Q-At-Home | I just need reliable phone lines |
| 03:33.30 | hardocp | what do i do with my current number |
| 03:33.35 | hardocp | that eveyone calls me on |
| 03:33.40 | hardocp | just ditch it? |
| 03:33.41 | Q-At-Home | shaw cable is supposed to sell voip |
| 03:33.45 | harryvv | bjohnson I might know but will look into a extentions and see how it can be done. |
| 03:33.48 | liquidno2 | Just give them the new one |
| 03:33.53 | liquidno2 | I mean come on |
| 03:33.57 | Q-At-Home | people will call your old one |
| 03:34.04 | liquidno2 | are you that self absorbed? |
| 03:34.07 | Q-At-Home | get the telco 404 not found |
| 03:34.09 | liquidno2 | big deal |
| 03:34.11 | hardocp | what if i am a business |
| 03:34.15 | Q-At-Home | and figure out the new number |
| 03:34.15 | hardocp | then what? |
| 03:34.16 | liquidno2 | if they need me... they will find me |
| 03:34.24 | Silik0n | <PROTECTED> |
| 03:34.25 | harryvv | Q-at-home what is it about xlite yu dont like? |
| 03:34.26 | Q-At-Home | pay for a telco forward like I do |
| 03:34.30 | Silik0n | ack! |
| 03:34.31 | Silik0n | misfire |
| 03:34.35 | Silik0n | NSFW! |
| 03:34.37 | bjohnson | Q-At-Home: the problem is .. the big boys don't want voip .. so they don't back voip providers |
| 03:34.37 | harryvv | silk, we dont need to see that here. |
| 03:34.39 | Q-At-Home | I think their network code is crappy |
| 03:34.44 | Q-At-Home | yeah |
| 03:34.47 | liquidno2 | Man that is bash.org worthy |
| 03:34.48 | Q-At-Home | I miss my PRI |
| 03:34.49 | Silik0n | it was a misfire |
| 03:34.54 | Silik0n | as in OOOOPS |
| 03:34.57 | Q-At-Home | I had big plans for it |
| 03:34.59 | Silik0n | jesus |
| 03:35.11 | *** join/#asterisk jdims (~sleepbsd@24-119-121-6.cpe.cableone.net) |
| 03:35.15 | sudoer | is it normal for a * box with zaptel hardware in it to not start up after a softreboot, but instead the machine has to stay down for a few minutes before turnign back on |
| 03:35.28 | twisted | harryvv, chill |
| 03:35.33 | Silik0n | no shit |
| 03:35.34 | sudoer | these are the shit problems I have to deal with * |
| 03:35.36 | Q-At-Home | sudoer: nope, mine always comes back |
| 03:35.43 | twisted | it's not like it used to be in here, so you should be happy for that ;) |
| 03:35.45 | Q-At-Home | sudoer: I'd blame your hardware first |
| 03:35.53 | sudoer | hey just send me a new card today |
| 03:35.54 | harryvv | no, i have familly here last thing is thay want to see that. |
| 03:35.55 | twisted | bkw and I used to make this channel look like #wackypornotalk |
| 03:35.59 | sudoer | now the whole system is f*ed up |
| 03:36.05 | twisted | then don't click on the link that says LESBIANS!.jpg |
| 03:36.08 | bjohnson | hardocp: it's useful to have at least one line you can count on in emergancy situations. Get a fxo to connect to it and it can become part of your voip system |
| 03:36.10 | liquidno2 | I blame pi... that nonending nonrepeating floating point |
| 03:36.14 | Silik0n | yeah what twisted said |
| 03:36.15 | harryvv | I dont |
| 03:36.30 | Q-At-Home | pi has to end some day |
| 03:36.36 | liquidno2 | pi is just a fad |
| 03:36.39 | Silik0n | thats the great thing about links you dont have to click them and I was NICE enuff to say OOPS NSFW! |
| 03:37.06 | hardocp | can you exlain |
| 03:37.15 | hardocp | more fully how that would work please |
| 03:37.26 | Q-At-Home | hey, I know those LESBIANS! |
| 03:37.30 | bjohnson | mmmmmmmm pi |
| 03:37.35 | liquidno2 | Silik0n: well thanks for the link anyways... |
| 03:37.38 | liquidno2 | it rocks |
| 03:37.43 | liquidno2 | 3.1415926535 |
| 03:37.51 | *** join/#asterisk Chuji (Chuji@pcp09929633pcs.tulipgrove.tn.nash.comcast.net) |
| 03:37.51 | hardocp | bjohnson how does it become part of my system |
| 03:37.54 | liquidno2 | how sad is it that I have that memorized? |
| 03:38.07 | bjohnson | it just becomes another way in or out |
| 03:38.10 | hardocp | aha |
| 03:38.41 | *** join/#asterisk seanner (~ss@d36-161-141.home1.cgocable.net) |
| 03:38.42 | hardocp | but then i lose the advantage of iax |
| 03:38.58 | bjohnson | or .. keep it and use call forwarding .. little tidbit I picked up in here is that call forwarding allows multiple concurrent calls (if the number it forwards to can handle them) |
| 03:39.03 | *** join/#asterisk harryvetch (~noyb@S010600055d210201.vs.shawcable.net) |
| 03:39.11 | hardocp | interesting |
| 03:39.16 | harryvetch | bsod and memory. |
| 03:39.31 | bjohnson | hardocp: why do you lose iax? iax is just another way in and out of your system |
| 03:39.43 | hardocp | i meant the concurrent call |
| 03:39.47 | hardocp | part |
| 03:40.00 | hardocp | but you just explained it |
| 03:40.06 | bjohnson | one problem with voip is that it is not as dependable as pstn |
| 03:40.26 | harryvetch | bj, what have you seen. |
| 03:40.37 | bjohnson | most users here have more than one voip provider and a dialplan that auto-fails to the secondary one if the first isn't working |
| 03:40.50 | bjohnson | harryvetch: I have seen the light |
| 03:41.05 | harryvetch | no, the issues that make voip less reliable. |
| 03:41.09 | bjohnson | hardocp: eg .. pstn lines work if the power goes out |
| 03:41.15 | twisted | http://www.teencyberfetish.com/b/f4d/15.jpg |
| 03:41.15 | twisted | oops |
| 03:41.17 | twisted | misfire! |
| 03:41.21 | twisted | NSFW |
| 03:41.26 | hardocp | ya but my * wont work with the power out |
| 03:41.31 | hardocp | so whats the point |
| 03:41.39 | bjohnson | harryvetch: just more things need to be working for it to work .. your computer, your internet, your isps internet |
| 03:41.44 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 03:41.52 | harryvetch | bj, no reason why a asterisk network can work with a battery bank and inverters. |
| 03:42.08 | twisted | it's called "get a UPS" |
| 03:42.13 | Q-At-Home | my whole home lan is UPS'ed |
| 03:42.17 | twisted | mine too |
| 03:42.22 | twisted | my rack is on an 8 hour backup |
| 03:42.29 | hardocp | really? |
| 03:42.32 | hardocp | whoa |
| 03:42.35 | Q-At-Home | some of us are hardcore |
| 03:42.38 | twisted | and my lappie, of course, has a battery |
| 03:42.41 | harryvetch | I have a 120 amp hour battery here that would make my system laste at least 12 hours or more. |
| 03:42.43 | twisted | and then there's the desk in the other room |
| 03:42.52 | charles___ | hey |
| 03:43.03 | hardocp | 120 amp battery? |
| 03:43.06 | hardocp | damm |
| 03:43.07 | twisted | which has a 1200va backup |
| 03:43.12 | charles___ | How do you get the status of a T over a 401T card |
| 03:43.17 | twisted | dont' stick your tounge to THAT battery :) |
| 03:43.17 | sudoer | <PROTECTED> |
| 03:43.23 | Nethab | 120 amps worth for one hour |
| 03:43.25 | charles___ | I remember something like Blue or Red signal |
| 03:43.40 | harryvetch | net, or longer if the amperage is very low. |
| 03:44.04 | Q-At-Home | swap line 50 with 51 |
| 03:44.07 | twisted | that means 120 amps worth of pull will last 1 hour |
| 03:44.12 | harryvetch | yes |
| 03:44.14 | twisted | so if' it's 60 amps, that's 2 hours. |
| 03:44.23 | twisted | 30 amps, 4 hours. |
| 03:44.25 | twisted | you do the math :) |
| 03:44.26 | harryvetch | and so on. |
| 03:44.32 | Nethab | even a fully loaded 15amp circuit would last 8 hours or so |
| 03:44.38 | Q-At-Home | put your answer before your dial |
| 03:44.44 | twisted | yup |
| 03:44.47 | *** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk) |
| 03:45.14 | harryvetch | nethab plenty of time for the power company to bring the power backup unless its a wide spread problem. |
| 03:45.21 | twisted | hehe |
| 03:45.25 | sudoer | Q-At-Home: i had the answer before dial and same results |
| 03:45.26 | Q-At-Home | and if you want your background to ever play... put it before the dial, but after the answer |
| 03:45.29 | twisted | we've overloaded the circut at the office |
| 03:45.32 | bjohnson | hardocp: a sipura SPA 3000 has one fxo and one fxs. they auto connect if the power goes out .. so the phone connected to the fxs will still work (if it is a style that can run on the line power) |
| 03:45.33 | Q-At-Home | did you reload? |
| 03:45.35 | Q-At-Home | :) |
| 03:45.40 | forrestc | my stuff is in a colo with a backup generator and telco-grade inverted power. |
| 03:45.42 | twisted | some dumbass wired the entire fucking office to one 20A breaker |
| 03:45.47 | sudoer | Q-At-Home: I want it after the dial |
| 03:45.54 | twisted | so every day, we throw the breaker about 5 times |
| 03:45.56 | sudoer | yes, I stop and start, not reload |
| 03:46.01 | Q-At-Home | twisted: I hear ya |
| 03:46.04 | charles___ | hey guys |
| 03:46.10 | Q-At-Home | twisted: my old office was like that |
| 03:46.15 | charles___ | How do I get a status of a T channel on zaptel ? |
| 03:46.18 | twisted | yea... it sucks. |
| 03:46.20 | Q-At-Home | so I paid $1000 for a 30 amp to be brought in |
| 03:46.20 | loud | what does this mean ? No translator path exists for channel type Zap (native 68) to 256 |
| 03:46.25 | Silik0n | god damnit this softlicnet is fucking pissing me the fuck of |
| 03:46.27 | Q-At-Home | moved out 3 months later |
| 03:46.30 | forrestc | If the power goes out, they have something like 12 hours to make sure the generator started. |
| 03:46.32 | twisted | but fortunately, when the breaker is thrown, nothing turns off ;) |
| 03:46.35 | Silik0n | soft client even |
| 03:46.41 | twisted | soft-core porn? |
| 03:46.44 | twisted | what? |
| 03:46.48 | Silik0n | yeah that too |
| 03:46.50 | Q-At-Home | UPS's just bitch loudly |
| 03:46.52 | Chuji | twisted : When you going to move back here and come work for me? |
| 03:46.55 | Q-At-Home | brb |
| 03:47.03 | Chuji | twisted : I need an * developer |
| 03:47.08 | harryvetch | My * psu is 5 amps so 120/5 is 24 hours of power to the server. |
| 03:47.08 | Chuji | bad |
| 03:47.15 | twisted | Chuji, heh. |
| 03:47.22 | harryvetch | I probebly would not bet on it though. |
| 03:47.23 | twisted | if i have trouble here |
| 03:47.27 | twisted | i'll let you know :) |
| 03:47.36 | twisted | in the mean time, you can contract us out ;) |
| 03:47.45 | Chuji | twisted : Know anyone here besides critch and Corydon? |
| 03:47.57 | twisted | well, there's Silik0n in memphis |
| 03:47.59 | sudoer | digium also told me they were going to send a ups envelope to ship back the the bad card to the mfor free |
| 03:48.12 | Chuji | Hell Memphis is as far as you |
| 03:48.14 | harryvetch | im outa here for a while. |
| 03:48.23 | twisted | yeah, but he might relocate if the coniditions are right ;) |
| 03:48.25 | Silik0n | what memphis is only 2 hours to nash |
| 03:48.33 | Silik0n | and yeh I might reloc |
| 03:48.33 | bjohnson | it's not just battery backups |
| 03:48.36 | twisted | hsv is only 1.5/2 depending |
| 03:48.41 | sudoer | it also took them a few weeks to even send the card, becuase they wouldnt call back right away |
| 03:48.49 | tzanger | Feb 10 22:47:13 NOTICE[23658]: app_dial.c:927 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) |
| 03:48.49 | bjohnson | my isp lost their internet provider connection for a day |
| 03:48.51 | tzanger | wtf does that mean? |
| 03:48.53 | sudoer | it seems like digium sucks as a commercial companty |
| 03:48.57 | Chuji | How's your C skills Silik0n ? |
| 03:49.04 | Silik0n | they are ok |
| 03:49.09 | bjohnson | so voip didn't work |
| 03:49.15 | Chuji | Got any karma on mantis ? |
| 03:49.16 | Chuji | :) |
| 03:49.19 | Silik0n | a little |
| 03:49.21 | bjohnson | hardocp: you got that? |
| 03:49.29 | Silik0n | i been working on some non* stuff |
| 03:49.38 | Silik0n | well non* in that its not going into CVS yet |
| 03:50.02 | Chuji | Hmm, well I'll keep that in mind |
| 03:50.10 | *** join/#asterisk elee (~erik_leee@adsl-66-51-208-150.dslextreme.com) |
| 03:50.53 | *** part/#asterisk twisted (~twisted@twisted-active-pdpc.developer.and.friend.of.asterisk) |
| 03:50.53 | *** join/#asterisk twisted (~twisted@twisted-active-pdpc.developer.and.friend.of.asterisk) |
| 03:50.53 | *** mode/#asterisk [+o twisted] by ChanServ |
| 03:50.56 | twisted | d'oh |
| 03:50.57 | twisted | wrong button |
| 03:51.03 | elee | Hey everyone is this chat always this quiet? |
| 03:51.15 | twisted | quiet? |
| 03:51.17 | brc_ | yes |
| 03:51.17 | brc_ | it is |
| 03:51.18 | twisted | you just joined |
| 03:51.19 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 03:51.25 | elee | Yes I did |
| 03:52.02 | Nethab | it's better than being flooded |
| 03:52.09 | elee | True. |
| 03:52.22 | elee | I was hoping to listen in on some conversations about asterisk |
| 03:52.31 | Nethab | it's better than getting kicked in the nuts |
| 03:52.32 | twisted | you can't listen on irc |
| 03:52.36 | twisted | unless it's reading it off to you |
| 03:53.01 | twisted | Nethab, but wait! |
| 03:53.05 | elee | So true |
| 03:53.10 | twisted | Nethab, it's better than dying of a sherry enema, too! |
| 03:53.41 | Nethab | twisted, actually that doesn't sound too bad, is sherry blonde? |
| 03:53.52 | twisted | Nethab, it's a liqour |
| 03:54.09 | Nethab | she's a licker eh, hmm all the better |
| 03:54.12 | bjohnson | sherry is Harvey |
| 03:54.28 | hardocp | do iax providers charge for incomming minutes on dids |
| 03:54.36 | Nethab | not usually |
| 03:54.46 | bjohnson | hardocp: first .. it is voip providers |
| 03:54.50 | Nethab | incoming calls are usually paid for by the person making the call |
| 03:54.52 | hardocp | right |
| 03:54.56 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 03:55.02 | bjohnson | SOME you connect to with iax |
| 03:55.11 | hardocp | iax.cc seems to charge |
| 03:55.19 | twisted | hardocp, depends. |
| 03:55.31 | bjohnson | and voip packages usually include free incoming but limit how many concurrent calls |
| 03:55.33 | twisted | connect.voicepulse.com you pay $7.95/mo for the did, and no charge on incoming minutes |
| 03:55.41 | hardocp | aha |
| 03:55.45 | charles___ | Please |
| 03:55.45 | bjohnson | ones that are pay as you go typically charge for incoming |
| 03:55.59 | twisted | bjohnson, not always true. |
| 03:56.05 | Mavvie | I tell them: your username is "foo". They fill in: "foo bar". |
| 03:56.09 | charles___ | Can I do a cat /dev/zapchannel to see the status of a T1 ? |
| 03:56.19 | twisted | charles___, try zttool. |
| 03:56.21 | bjohnson | twisted: one way of the other .. you pay for it |
| 03:56.28 | Mavvie | I tell them: "your sip domain is foo.bar". They fill in "sip.foo.bar" |
| 03:56.34 | Mavvie | how can I live with these people? |
| 03:56.36 | hardocp | so what serivce offers cheap did with free incoming that you would recommend |
| 03:56.43 | twisted | bjohnson, if you say so |
| 03:56.51 | bjohnson | twisted: ie .. vp at $8 with free incoming vs iax at $2 and you pay |
| 03:57.36 | twisted | i can guarantee i would spend more than $8 in incoming calls. |
| 03:57.50 | twisted | besides, no limit on concurrent calls |
| 03:57.52 | twisted | AND |
| 03:57.58 | bjohnson | so that might be why you would use vp .. it doesn't apply to everyone |
| 03:58.00 | twisted | it's cheaper than a bare copper pair |
| 03:58.12 | twisted | gimme a fucking break |
| 03:58.17 | twisted | if you can't afford $8 a month, get a new hobby |
| 03:58.18 | bjohnson | AND they don't offer DIDs outside of US |
| 03:58.27 | bjohnson | so they ain't gonna work for hardocp |
| 03:58.36 | hardocp | i am in the us |
| 03:58.39 | twisted | HAH |
| 03:58.41 | twisted | ROFL |
| 03:58.41 | hardocp | actually? |
| 03:58.47 | hardocp | sorry |
| 03:58.47 | bjohnson | hardocp: I thought you said Edmondton |
| 03:58.52 | bjohnson | maybe that was Q-At-Home |
| 03:58.55 | hardocp | that was the other guy |
| 03:58.56 | adrianhensler | 1-800 is an option |
| 03:58.59 | twisted | yea |
| 03:59.14 | hardocp | hey bjohnson has been very helpful |
| 03:59.16 | twisted | toll-free incoming you will pay for minutes on incoming 99% of the time |
| 03:59.23 | hardocp | no reason to get nasty |
| 03:59.23 | bjohnson | 800 that works in canada .. check the pricing .. usually starts at $0.08 per minute |
| 03:59.33 | adrianhensler | yes - so I have found out! *curses* |
| 04:00.00 | bjohnson | crtc |
| 04:00.06 | adrianhensler | iax.cc 1-800 incoming to Canada is 6.5c US a minute |
| 04:00.13 | twisted | OUCH |
| 04:00.22 | bjohnson | adrianhensler: except they don't actually have any |
| 04:00.22 | twisted | that's like child rape |
| 04:00.33 | JerJer | we are forced to charge 10 cpm |
| 04:00.41 | bjohnson | listen to the MAN |
| 04:00.42 | adrianhensler | yes; and I am in 902 (Nova Scota) so it is next to impossible to find someone with a DID offering here |
| 04:00.46 | sudoer | piece of shit asterisk |
| 04:00.52 | *** join/#asterisk PBXtech (~nik@71-32-207-98.slkc.qwest.net) |
| 04:01.03 | twisted | sudoer, ? |
| 04:01.08 | adrianhensler | sorry bjohnson? don't have any? |
| 04:01.23 | PBXtech | how do i set the verbose level in the manager app so it only send minimal event information? |
| 04:01.24 | bjohnson | sudoer: really? the next version was going to be named FANTASTERICK |
| 04:01.25 | Mavvie | poor sudoer. so young and so frustrated. |
| 04:02.27 | bjohnson | adrianhensler: sorry .. haven't seen any. You could still use voip for outgoing though |
| 04:02.47 | twisted | word to the wise: subscribe to asterisk-biz |
| 04:02.55 | twisted | there are constantly providers offering numbers in odd areas there |
| 04:02.56 | adrianhensler | iax.cc says they have 1-800 to Canada. unless I am missing something |
| 04:03.26 | bjohnson | adrianhensler: they don't have any |
| 04:03.39 | bjohnson | adrianhensler: or at least they didn't 3 weeks ago |
| 04:04.06 | bjohnson | all though there web site said the same thing then as it does now |
| 04:04.12 | bjohnson | sorry .. their |
| 04:04.20 | samueltc | hi |
| 04:04.39 | adrianhensler | hm - I think you are right; but they sort of fib on their page: Toll free Number (US50/CA) $1.09/month US48 2¢, CA/HW 6.5¢ |
| 04:04.39 | samueltc | hey, anyone know were I can virtual host an asterisk system? |
| 04:04.56 | bjohnson | well .. gotta go before I get things "twisted" too |
| 04:05.03 | twisted | ehh? |
| 04:05.53 | samueltc | i need some reliable hosting for a low volume number |
| 04:06.13 | twisted | samueltc, all voip traffic? |
| 04:06.32 | samueltc | yes |
| 04:06.34 | adrianhensler | all voip maybe at linode? http://www.linode.com/products/linodes.cfm |
| 04:07.11 | twisted | yeah, you can get away with doing that at most places like linode |
| 04:07.22 | *** part/#asterisk joebaker (~joebaker@rrcs-67-52-9-110.west.biz.rr.com) |
| 04:07.37 | adrianhensler | not cheap though |
| 04:07.39 | Nethab | ok this background thing isn't working |
| 04:07.56 | elee | I could probly do it in like a month? |
| 04:09.31 | elee | How soon were you looking to get it taken care of? |
| 04:10.24 | samueltc | two weeks ago |
| 04:10.51 | elee | oh. All I have right now is business dsl w/ static |
| 04:11.47 | *** part/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 04:11.52 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 04:12.20 | charles___ | twisted, ztcfg -vvvvvvv returns no problems |
| 04:12.31 | elee | 1500/768 |
| 04:12.35 | charles___ | twisted, but I see the led of the 405t is red |
| 04:12.41 | elee | close to a t1 |
| 04:12.54 | charles___ | twisted, is there a way to see the red alarm in the system ? |
| 04:13.49 | pcm | cat /proc/zaptel/span_number |
| 04:13.50 | twisted | charles___, didn't I just say to use zttool? |
| 04:14.00 | twisted | pcm, you can do that, but zttool is pretty :) |
| 04:14.22 | pcm | twisted: but you can do 'remote loopback' hehe |
| 04:14.28 | sudoer | what is the command for zaptel to run to do basic configuration of line noise? |
| 04:14.29 | pcm | if you have colors screwd |
| 04:14.46 | charles___ | twisted, I don't have that zttool |
| 04:14.53 | charles___ | twisted, where does it came from ? |
| 04:14.55 | twisted | pcm, sure, it's fun :) |
| 04:14.57 | pcm | you need ncurses |
| 04:15.01 | twisted | charles___, it's in zaptel. you need newt |
| 04:15.11 | pcm | well maybe newt :) |
| 04:15.12 | pcm | heh |
| 04:15.13 | charles___ | twisted, I did in the past a cat on something |
| 04:15.16 | charles___ | twisted, but I can't remember |
| 04:15.26 | twisted | yeah, you must have done the cat /proc/zaptel/<span> |
| 04:15.27 | JerJer | pcm: newt-dev to be specific |
| 04:15.27 | twisted | thing |
| 04:15.28 | pcm | look in the history :) |
| 04:16.00 | pcm | JerJer: right, right, includes ... |
| 04:17.16 | Mavvie | http://bugs.digium.com/bug_view_page.php?bug_id=0003558 |
| 04:17.20 | Mavvie | yay! |
| 04:18.19 | v-office | anyone tell me why I'm getting disagreeing symbol versions after rebuilding zaptel? Nothing on the system changed except a setting in the zaptel Makefile |
| 04:19.00 | sudoer | ok, I put the old4 port fxo card back in, now when calls go through, there is a horrible clicking noise |
| 04:19.36 | sudoer | the way asterisk works changes for everytime you take out the fxo card and put it back in???!?!?!??! |
| 04:19.54 | Mavvie | sudoer: the clicking noise is the FBI |
| 04:20.12 | twisted | lol |
| 04:21.12 | v-office | ah, found it. subsequent make installs are not overwriting old versions of wct1xxp.o |
| 04:21.37 | sudoer | I have 2 fxo cards here, why does the behior dramatically change when I put a different card in, seriously, this is sucha retarded problem |
| 04:21.44 | sudoer | behavior |
| 04:21.51 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 04:21.51 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 04:24.17 | *** join/#asterisk freddy (~jason@martin.pnc.com.au) |
| 04:24.29 | freddy | anyone have time to help me with a calleridname problem? |
| 04:24.43 | pcm | what's the prob ? |
| 04:25.05 | sudoer | anyone? |
| 04:25.11 | sudoer | I cant leave until this is fixed :( |
| 04:25.33 | freddy | the variable CALLERIDNAME is set to the local ext rather than the caller. It was working but is not now. |
| 04:25.41 | pcm | what technology is the caller ? |
| 04:26.03 | pcm | what asterisk version do you use ? |
| 04:27.11 | freddy | SetCIDName |
| 04:27.11 | PMantis | sudoer, Describe the change - then perhaps someone will be more willing to respond |
| 04:27.11 | freddy | Asterisk CVS-v1-0-02/07/05-22:32:26 |
| 04:27.19 | sudoer | PMantis: the change is now all calls have the sound of a tv blasting at fukll volume on a channel with no reception |
| 04:27.34 | sudoer | and I cant call out form here either |
| 04:27.50 | PMantis | sudoer, Static on a Zap device? |
| 04:27.52 | freddy | pcm: someone suggested that I need to use the 'o' option with Dial. |
| 04:28.14 | sudoer | when I use the other brand new card, the calls are all messed up in another way |
| 04:28.18 | freddy | pcm: but it did not seem to work. but perhaps I put it in the wrong place. |
| 04:28.26 | sudoer | yes, static on zap, that wasnt here about an hour ago |
| 04:29.10 | PMantis | I don't get it. Anyony else? Why would sudoer be hearing static on a zaptel device? |
| 04:29.42 | Chuji | The one thing that my job has brought me is a great salary and benifites, and less of me wants to take a risk like when I was younger |
| 04:30.23 | PMantis | sudoer, I have to go soon, but hopefully someone will have a clue about this. Perhaps reboot the server as a last resort? |
| 04:30.41 | sudoer | ok, I'm rebooting now |
| 04:31.57 | redder86 | static on a Zap device? How do you "hear" static on a zap device? Is there some way to connect speakers to a Zap device? |
| 04:32.43 | freddy | pcm: I don't think the version I am using requires the 'o' option. |
| 04:33.00 | Chrisf0rd | anybody know where I can buy a block of DID Numbers |
| 04:33.11 | Chrisf0rd | or did virtual numbers |
| 04:33.42 | sung | yeah |
| 04:33.43 | sung | level3 |
| 04:33.49 | sung | just commit to $80k/mo |
| 04:34.14 | Chrisf0rd | HOLY DOLLAR SIGNS |
| 04:34.19 | sudoer | asterisk has just completey wen tcrazy |
| 04:34.30 | sudoer | nothing works at all here |
| 04:34.37 | sung | Chrisf0rd: are you looking in a single area |
| 04:34.39 | sung | or across the US? |
| 04:34.47 | Chrisf0rd | both |
| 04:34.51 | Chrisf0rd | select areas |
| 04:34.55 | charles___ | hey |
| 04:34.56 | sung | ahh. |
| 04:35.04 | sung | you'll need to find a nation-wide clec |
| 04:35.08 | sung | or a nation-wide clec reseller |
| 04:35.17 | sung | recommend XO/LVLT if you've got the biz |
| 04:35.19 | Chrisf0rd | Cincinnati Dayton Texas FLor. |
| 04:35.26 | Chrisf0rd | I have the biz |
| 04:35.38 | sung | how many minutes? |
| 04:35.44 | sung | in each of the areas |
| 04:35.48 | sung | per mo |
| 04:35.52 | Chrisf0rd | right now we are using 17000 |
| 04:36.01 | Chrisf0rd | across teh board |
| 04:36.16 | sung | assuming equal distribution |
| 04:36.25 | sung | you're talking 4250 minutes per area |
| 04:36.27 | Chrisf0rd | what is the XO/LVLT web site |
| 04:36.27 | Chrisf0rd | yeah |
| 04:36.30 | sung | contact a smaller CLEC in each of htose areas |
| 04:36.40 | sung | www.xo.com |
| 04:36.42 | sung | www.level3.com |
| 04:36.50 | sung | wouldn't advise speaking to lvlt |
| 04:36.56 | sung | they'll tell you to pound sand |
| 04:37.02 | sung | perhaps XO will be willing to work with you |
| 04:37.11 | sung | or you can dealler with an xo/level3 reseller |
| 04:37.11 | Chrisf0rd | ok |
| 04:37.12 | file[laptop] | level3 won't talk to you unless you are doing A LOT of traffic |
| 04:37.26 | sung | i <3 level3 |
| 04:37.32 | Chrisf0rd | I only have 368 customers |
| 04:37.34 | sung | i need to buy some of their stock tomorrow |
| 04:37.43 | sung | it's a good start |
| 04:37.49 | sung | talk to level3 when you've got the minutes |
| 04:37.56 | Chrisf0rd | I am puting in a DS3 with aT&T in 3 weeks |
| 04:38.04 | sung | ds3 with att? |
| 04:38.07 | Chrisf0rd | we had the technical interview today |
| 04:38.09 | sung | ...for termination? |
| 04:38.13 | sung | domestic or intl? |
| 04:38.20 | Chrisf0rd | Domestic Ohio |
| 04:38.30 | sung | that's a fairly decent amount of traffic |
| 04:38.32 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 04:38.54 | Chrisf0rd | well I also route for broadvoice |
| 04:39.06 | Chrisf0rd | starting next month |
| 04:39.06 | sung | you can push, like, seven million minutes. |
| 04:39.09 | sung | good luck with that mang. |
| 04:39.19 | sung | ( non sarcastic ) |
| 04:39.22 | Chrisf0rd | Well the line only cost $5900 a month |
| 04:39.29 | sung | yeah |
| 04:39.40 | sung | they may charge you $$$ to turn up each of the t1's on the ds3 |
| 04:39.44 | Chrisf0rd | I remember My 1st T-1 it cost $3200 in 19956 |
| 04:39.45 | sung | all i can say is read it all very carefully |
| 04:39.46 | Chrisf0rd | oops |
| 04:39.47 | Chrisf0rd | 1995 |
| 04:39.58 | sung | are you going to be talking ss7 with att |
| 04:40.05 | sung | or doing PR? |
| 04:40.07 | sung | PRI? |
| 04:40.11 | Chrisf0rd | PRI |
| 04:40.21 | sung | mmm |
| 04:40.30 | sung | what kind of switch are you plugging the ds3 into? |
| 04:40.42 | Chrisf0rd | Cisco 3745 |
| 04:40.43 | Silik0n | what up sung |
| 04:40.51 | sung | Silik0n: not much |
| 04:41.07 | sung | Silik0n: do i know you? :P |
| 04:41.17 | Chrisf0rd | Sung what do you know about the Credit Card program |
| 04:41.25 | sung | what credit card program? |
| 04:41.31 | Chrisf0rd | ASTERICC |
| 04:41.35 | Chrisf0rd | i belive |
| 04:41.37 | sung | oh, nothing. |
| 04:42.02 | sung | i don't do very much with asterisk...just play with it a lot. |
| 04:42.07 | Chrisf0rd | we changed the main greeting on one of the * boxes and damned if you can't type 7777 and get teh callign card system |
| 04:42.27 | Chrisf0rd | to start up anymore |
| 04:42.34 | Chrisf0rd | but from internal it starts right up |
| 04:44.41 | Essobi | Anyone know an URL for a third party CNAME database? |
| 04:45.22 | Mavvie | Essobi: eu.org |
| 04:45.37 | Mavvie | Essobi: dyndns.org |
| 04:45.40 | Mavvie | or .com |
| 04:45.45 | Essobi | umm |
| 04:45.53 | Essobi | not a DNS CNAME |
| 04:46.00 | Mavvie | oh. |
| 04:46.04 | Essobi | DNIS to NAME |
| 04:46.09 | sung | heh. |
| 04:46.13 | sung | hahahaha. |
| 04:46.14 | JerJer | google |
| 04:46.17 | sung | LIDB/CNAM PLZ |
| 04:46.30 | Essobi | CNAM CNAME whatever |
| 04:46.36 | brc_ | bwhahahahahahaha |
| 04:46.38 | brc_ | "As we progress, running a casino will become more and more like playing SimAnt. " |
| 04:46.42 | brc_ | simant...I loved that game |
| 04:46.44 | Essobi | Who do you use jer jer? |
| 04:46.51 | JerJer | we don't use |
| 04:46.54 | brc_ | googol |
| 04:49.16 | Essobi | Mmm. |
| 04:49.39 | Essobi | I need a static CNAM database to load up. I got a customer who wants us to hand them a PRI and provide CNAM on it. |
| 04:50.00 | sudoer | can anyone help me with this zaptel problem |
| 04:50.14 | *** join/#asterisk brimstone (me@146.229.186.157) |
| 04:50.18 | sudoer | the whole freakin system is acting 100 crazt, it does not work at all now |
| 04:52.01 | sudoer | i have 2 zaptel cards here, one is falky, the other is a brnad new one digium sent to me,they are both 4 port fxo cards |
| 04:53.55 | sudoer | I switched the flaky one out and put in the new one, when i call into office from outside, and let it go to the answering machine, it plays only the last 2 seconds of the message, when i call form sip phone to the external number of this office, then answer from another sip phone, the original phone keeps ringing |
| 04:55.02 | sudoer | then i put switched cards again , to put flaky one bac kin, and now there is a very very loud static noise on all calls, so i switched back the other card, and the same problems again, switching back and forth between the cardsm asterisk changes its behavior completely |
| 04:55.53 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 04:56.01 | Essobi | bad psu? |
| 04:56.14 | elee | whoa that's some crazyness |
| 04:56.16 | netsurfer | sudoer - just a suggestion.. check the chips on the cards are the same |
| 04:57.44 | *** join/#asterisk Chrisf0rd (~chris@dhcp024-209-236-189.cinci.rr.com) |
| 04:58.09 | sudoer | netsurfer: ok, ill check, but even if they werent the same, asterisk wouldnt start becuase of the zapata.conf file, you have to tell it what kind of card you have and if it detects the wrong chip, * wont start at all |
| 05:02.01 | sudoer | psu is power supply? |
| 05:02.52 | netsurfer | yup |
| 05:04.11 | *** join/#asterisk Flood_of_SYNs (~cory@wbar13.chi1-4.27.95.222.chi1.dsl-verizon.net) |
| 05:07.06 | soulz- | hello all |
| 05:07.22 | elee | what's up soulz |
| 05:07.24 | sudoer | follow up to problem, when i call like this sip1->zap->sip2 i answer on sip2, and sip1 keeps ringing, then after about 5 seconds the call is connected, so that is similar to not asnwering and hearing the last 2 seconds of message, becuase asterisk answers the phone,s but the sip phones dont recieve an answer until 5 seconds later, why?!?!?! |
| 05:09.23 | sudoer | then after who knows how long,(i just noticed, so i odnt know) the call was disconnected without me hanging it up |
| 05:10.04 | Flood_of_SYNs | what kind of console msgs ru getting? |
| 05:11.12 | sudoer | when i really do answer the phone, i get an answered message: -- SIP/main-5b92 answered Zap/2-1 |
| 05:11.25 | *** join/#asterisk kvanals (~kvanals@kvanals.org) |
| 05:11.35 | freddy | okay, the CALLERID is correct in the first Macro, but when the second runs it is wrong. |
| 05:11.36 | sudoer | but after the answer, the callign phone still continues to ring for about 5 mroe seconds before the call actually connects |
| 05:12.28 | Flood_of_SYNs | on the asterisk box type this asterisk -cr |
| 05:13.14 | sudoer | I using start with -vvvvvvvvvvvvvvvgggc |
| 05:13.24 | Flood_of_SYNs | hehe |
| 05:13.24 | Mavvie | you're missing the -d |
| 05:13.25 | soulz- | anyone have any problems with broadvoice now? |
| 05:13.26 | sudoer | while i am debugging |
| 05:13.51 | Flood_of_SYNs | verbose enough? :-) |
| 05:14.08 | freddy | pcm: you there? |
| 05:14.41 | pcm | sure |
| 05:15.03 | sudoer | i notied something else, when i hangup the sip2 call ,the sip1 call still thinks its running because the counter is still going |
| 05:15.19 | freddy | does asterisk join to channels to make a call? |
| 05:15.38 | freddy | pcm: I get -- Executing NoOp("SIP/2042-6907", ""Andrew Leahy"<341>") in new stack |
| 05:15.40 | elee | and it only happens like that with the new one, do you get any delay in the crappy one? |
| 05:15.48 | freddy | pcm: then -- Executing NoOp("SIP/2028-dff5", "2028") in new stack |
| 05:15.57 | freddy | pcm: so the callerid has changed! |
| 05:16.12 | sudoer | no, in the crappy one, it is a an extremely loud screech, its unusable |
| 05:16.24 | freddy | sorry I meatn two channels, not 'to' |
| 05:16.57 | freddy | pcm: is that two different channels? |
| 05:17.48 | pcm | freddy: the incoming channel is separate from outgoing |
| 05:18.11 | pcm | freddy: two diffrent channels, right ... asterisk bridges them in app_dial.c |
| 05:19.11 | *** join/#asterisk Chrisf0rd (~chris@dhcp024-209-236-189.cinci.rr.com) |
| 05:21.05 | freddy | pcm: so what would have changed. |
| 05:21.21 | freddy | pcm: that second line that says 2028 use to have the guys name. |
| 05:21.43 | sudoer | i just want * to go back to normalk |
| 05:21.58 | freddy | pcm: is there a way to get the callerid into the second channel perhaps a variable? |
| 05:23.17 | Flood_of_SYNs | sudoer: could it be that the sound levels on the clients are set to high? |
| 05:23.51 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 05:23.51 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 05:25.54 | soulz- | hello all |
| 05:25.55 | soulz- | anyone have any problems with broadvoice now? |
| 05:26.22 | v-office | you mean right this second, or in general? |
| 05:26.30 | pcm | freddy: the app_dial should copy the callerid data ... |
| 05:26.31 | soulz- | this second |
| 05:26.37 | sudoer | Flood_of_SYNs: no, they are on medium by default and people here complain to set default to high because the volume is too low |
| 05:26.40 | v-office | dunno, I'll check in a second |
| 05:26.44 | sudoer | im going to try switching cards again |
| 05:26.56 | soulz- | v-office : when i dial to toll free numbers + bv numbers it works |
| 05:27.02 | soulz- | when i dial others, it just doesn't work |
| 05:27.08 | v-office | wierd... |
| 05:27.24 | soulz- | tell me about it |
| 05:27.37 | Flood_of_SYNs | sudoer: is the whole system pure voip or are you doing analog phones? |
| 05:28.26 | freddy | pcm: so if its not what could that be |
| 05:29.27 | sudoer | sip phones on internal office, to connect to outside, go through sip |
| 05:29.36 | sudoer | the card is for sip |
| 05:29.41 | sudoer | zaptel i mean |
| 05:30.33 | pcm | freddy: is your asterisk HEAD ? or some 1.0.x ? |
| 05:30.49 | elee | sudoer: are you going to be on tommarrow during the day? I want to find out what is happening. |
| 05:31.02 | Flood_of_SYNs | sudoer: it could be a misconfiguration of the zaptel driver. |
| 05:31.19 | sudoer | I wish it is a misconfiguration of zaptel driver |
| 05:31.55 | sudoer | elee, I was not planning on it, but now if the phones dont work ,I have to be here, this company depends on phones |
| 05:32.01 | Flood_of_SYNs | sudoer: this site has helped me out alot http://www.voip-info.org |
| 05:32.21 | sudoer | Flood_of_SYNs: thanks, I have read the site a million times, |
| 05:32.41 | sudoer | im not a super newb, (but not an expert either) |
| 05:33.21 | *** part/#asterisk elee (~erik_leee@adsl-66-51-208-150.dslextreme.com) |
| 05:33.26 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
| 05:33.30 | RoyK | morning |
| 05:33.44 | pcm | what's the time ? |
| 05:33.56 | RoyK | 0633 |
| 05:33.57 | sudoer | omfuckingg, I put the flaky card back in, and it seems like its working somewhat normal |
| 05:34.00 | RoyK | CET |
| 05:34.07 | freddy | pcm: CVS-v1-0-02/07/05-22:32:26 |
| 05:34.19 | pcm | Oh ok .... |
| 05:34.23 | RoyK | pcm: getting up early while people aren't talking so much on the phone, working :P |
| 05:34.46 | pcm | yeah; how late are you going to sleep usually ? |
| 05:35.35 | RoyK | I've been up an hour or so now |
| 05:35.45 | RoyK | usually I get to work between nine and ten |
| 05:36.14 | RoyK | that's just 10km from here, 25minutes on the bike (unless it's too cold) |
| 05:37.14 | *** join/#asterisk Chrisf0rd (~chris@dhcp024-209-236-189.cinci.rr.com) |
| 05:39.40 | Flood_of_SYNs | join |
| 05:40.06 | RoyK | part |
| 05:40.55 | Flood_of_SYNs | for got the "/" :-) |
| 05:41.26 | AgiNamu | oh guys |
| 05:41.28 | AgiNamu | sigh... |
| 05:41.50 | AgiNamu | "I'm using a Sipura SPA-3000 and connecting using IAX to FWD and DTMF doesn't work." |
| 05:42.59 | RoyK | AgiNamu: lol |
| 05:43.16 | RoyK | AgiNamu: soldered his own IAX client? |
| 05:44.55 | *** part/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 05:53.41 | thepdakid | Has anyone used Mahler's book "Voip Telephony with Asterisk" successfully with a soft phone? |
| 05:57.31 | *** join/#asterisk Chrisf0rd (~chris@dhcp024-209-236-189.cinci.rr.com) |
| 05:58.22 | thepdakid | I ask because it refers to Cisco and Snom hard IP phones (that I don't have) |
| 06:07.50 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 06:10.18 | AgiNamu | What's the standard way to handle out of memory in Asterisk C code? |
| 06:10.37 | AgiNamu | like i realloc something, and get null... do i just die? |
| 06:10.37 | WildPikachu | can u record all incoming calls in asterisk? |
| 06:10.39 | *** join/#asterisk crazyhick (ldjvwl@cbn-24.182.133.243.cbnstl.net) |
| 06:10.43 | AgiNamu | wild yes |
| 06:10.45 | WildPikachu | for say security purposes? |
| 06:10.51 | crazyhick | lo all |
| 06:10.54 | WildPikachu | which app would handle that? :) |
| 06:10.56 | *** join/#asterisk Chrisf0rd (~chris@dhcp024-209-236-189.cinci.rr.com) |
| 06:11.51 | AgiNamu | monitor |
| 06:12.00 | AgiNamu | google is a marvellous thing :) |
| 06:12.29 | AgiNamu | should I gracefully exit the routine |
| 06:12.54 | AgiNamu | or ast_log(LOG_ERROR, "Fuck, we're out of memory. Gonna go segfault now.") and then segfault. |
| 06:13.20 | WildPikachu | heh |
| 06:13.27 | Qwell | segfaults are always fun |
| 06:13.30 | WildPikachu | #define segfault free(1) |
| 06:15.02 | charles___ | hey guys |
| 06:15.12 | AgiNamu | hey you |
| 06:15.46 | charles___ | does anybody have been working with adtran + 405t |
| 06:15.51 | charles___ | 10 channels on the adtran |
| 06:16.01 | charles___ | I don't know how to select the b channel |
| 06:17.57 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:19.24 | charles___ | how can I get the zttool ? |
| 06:22.28 | datareactor | charels it in bristuff |
| 06:23.06 | charles___ | cool |
| 06:27.32 | glock | anyway i can increase the volume on a zap interface |
| 06:27.41 | glock | i can hardly hear anything |
| 06:29.22 | *** join/#asterisk adrianhensler (~chatzilla@blk-222-123-93.eastlink.ca) |
| 06:30.39 | charles___ | glock, gain |
| 06:31.05 | *** join/#asterisk kb1_kanobe (~krisbouti@border.scrd.bc.ca) |
| 06:31.11 | kb1_kanobe | evening all. |
| 06:31.33 | Nethab | good morning |
| 06:33.01 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 06:34.29 | kb1_kanobe | Any opinions on running * stable on kernel 2.6.x in production? |
| 06:34.41 | glock | chaoscon: thanks, will try it |
| 06:34.50 | glock | charles___: err, :) |
| 06:34.57 | Nethab | i run * on 2.6 |
| 06:35.17 | kb1_kanobe | Nethab: are you using any zap hardware? |
| 06:35.36 | Nethab | no, i use sip based fxo/fxs gateways |
| 06:35.52 | kb1_kanobe | ok. Worked well for you? |
| 06:36.09 | Nethab | yes, i haven't been down since october |
| 06:36.23 | Nethab | that's when i started |
| 06:36.57 | kb1_kanobe | Alright - that qualifies as well. :-) |
| 06:37.42 | Nethab | i don't have a pri or bri, so i can't give any comparison's for you |
| 06:37.58 | glock | charles___: How can i reload with the gain settings. I've made it rxgain=1.0 in zapata.conf, and rerun ztcfg and asterisk. still soft thou. anything im missing? |
| 06:39.38 | kb1_kanobe | glock: did you "restart now" or "reload" asterisk? |
| 06:40.13 | glock | kb1_kanobe: stop now, then restarted it |
| 06:40.16 | soulz- | anyone have any problems with broadvoice now? |
| 06:41.14 | Nethab | right this second? |
| 06:41.15 | Nethab | no |
| 06:41.27 | kb1_kanobe | glock: that should have done it then. What was it before? |
| 06:41.50 | soulz- | nethab: something weird is happening to me |
| 06:41.59 | soulz- | nethab : when i dial to toll free numbers + bv numbers it works |
| 06:42.05 | soulz- | when i dial others, it just doesn't work |
| 06:42.48 | glock | kb1_kanobe: there was none set. still veeery soft now |
| 06:43.03 | Nethab | how long ago did you sign up |
| 06:43.18 | kb1_kanobe | glock: ah. 1.0 is the default gain. try increasing to, say, 3.0 and reloading. |
| 06:43.27 | soulz- | nethab: 5 months ago, but my hdd crashed, so i had to redo my config |
| 06:43.43 | Nethab | what other numbers are you trying |
| 06:44.04 | Nethab | international? |
| 06:44.11 | soulz- | us |
| 06:44.13 | soulz- | <PROTECTED> |
| 06:44.32 | soulz- | dialing 1-978-418-7300 works |
| 06:44.40 | Nethab | have you looked at your broadvoice CDR records |
| 06:45.26 | soulz- | let me check dude:) |
| 06:46.14 | kb1_kanobe | Anyone else running * stable w/zaptel hardware on 2.6.x kernel? |
| 06:47.33 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
| 06:48.01 | Mavvie | kb1_kanobe: yes |
| 06:48.15 | kb1_kanobe | Mavvie: all good? |
| 06:48.30 | Mavvie | kb1_kanobe: can't complain, except that my second ethernet card doesn't get recognized. |
| 06:49.15 | kb1_kanobe | Mavvie: I'm trying to decide between patching 2.4.x w/low latency patches or moving up to 2.6.x... thoughts? |
| 06:49.43 | Mavvie | ask somebody with linux-skills, not somebody with bsd-skills :-) |
| 06:49.53 | kb1_kanobe | Ah. :-) |
| 06:52.37 | glock | kb1_kanobe: Thanks for that. I think it sounds a bit better. :) |
| 06:53.25 | kb1_kanobe | glock: +3db will double the volume, -3db will halve. Adjust accordingly, but beware of excessive volumes interfering w/echo cancellation if you use it. |
| 06:55.43 | glock | kb1_kanobe: kk, thanks :) |
| 06:57.54 | *** join/#asterisk tih (tih@athene.hamartun.priv.no) |
| 06:58.04 | AgiNamu | im trying to add a filter to logger.c |
| 06:58.23 | AgiNamu | ast_log takes varags and uses va_start and end, if I want a temp buffer, so i can check the output |
| 06:58.38 | AgiNamu | can I do va_start/vsnprintf/va_end? |
| 06:58.45 | glock | Has anyone got fax detection working? I think my logic is a bit messed up. For an incoming, i need an exten => s value. For fax i need a fax option. How do i take the call, detect if it is a fax and send it to zap/4 and if it is not a fax, ring on zap/1? |
| 07:02.52 | redder86 | glock: have faxdetect=incoming in your zapata.conf, and then in your dialplan Answer() and Wait(3) or something. |
| 07:03.46 | kb1_kanobe | glock: see http://www.voip-info.org/wiki-Asterisk+fax |
| 07:13.13 | glock | thanks, will try it some more. |
| 07:18.53 | AgiNamu | Well, my ast_log filter is working! |
| 07:21.42 | Beirdo | anyone here have any experience using nufone? I'm considering them for my VOIP provider, and was just looking for user experiences. |
| 07:28.18 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 07:29.46 | redder86 | Beirdo: they're good |
| 07:30.09 | Beirdo | not many service interruptions, etc? |
| 07:31.29 | moonwick | not really |
| 07:31.34 | moonwick | nufone's good people |
| 07:31.50 | *** join/#asterisk Inv_arp (junya@adsl-3-251-94.mia.bellsouth.net) |
| 07:31.59 | Beirdo | good good. |
| 07:32.21 | Beirdo | and of course, I'm likely talking to people who run nufone :) |
| 07:33.04 | *** join/#asterisk X-Gen (~x-gen@rrba-146-67-74.telkomadsl.co.za) |
| 07:33.28 | Beirdo | but IAX connection... paypal... DID... |
| 07:33.30 | Inv_arp | does one have to use the wan side of the HT 486? just plug lan portion to switch |
| 07:33.35 | Beirdo | seems to be a good combo |
| 07:35.12 | X-Gen | g'day freaks© |
| 07:35.18 | X-Gen | :) |
| 07:35.40 | AgiNamu | hey will someone try my ast_log filter out and tell me if it works for them? |
| 07:36.02 | AgiNamu | http://bugs.digium.com/bug_view_page.php?bug_id=0003561 |
| 07:37.10 | Mavvie | make it a regexp. |
| 07:37.20 | AgiNamu | ooh thats a good idea. |
| 07:38.24 | AgiNamu | added a bugnot |
| 07:38.26 | AgiNamu | thanks |
| 07:39.10 | AgiNamu | well im shit tired |
| 07:39.12 | AgiNamu | so |
| 07:39.15 | AgiNamu | im gonna go now |
| 07:39.19 | AgiNamu | good night everyone |
| 07:39.29 | AgiNamu | and good luck in your asterisk journeys |
| 07:39.34 | Mavvie | sleep well |
| 07:40.48 | jarrod | hmm |
| 07:40.50 | jarrod | i like SER |
| 07:47.04 | *** join/#asterisk murangd (~nukaidc@pool-162-84-152-73.ny5030.east.verizon.net) |
| 07:47.11 | murangd | hello is anyone awake |
| 07:47.24 | murangd | I have a few questions about the asterisk and dundi network |
| 07:47.28 | murangd | can anyone spear 5 minutes |
| 07:47.51 | murangd | anyone alive? |
| 07:48.01 | Qwell | fire away |
| 07:48.20 | murangd | quel: ok please bare with me if I sound a little silly |
| 07:48.37 | murangd | I was reading about the DUNDI network and the sterisk homepage |
| 07:48.43 | murangd | I am confused as what DUNDI actually does |
| 07:48.46 | Qwell | no need to direct the questions at me, I know nothing about dundi |
| 07:48.56 | murangd | ok about VOIP then |
| 07:49.14 | Qwell | not really, heh |
| 07:49.24 | Qwell | people are around though, lurking |
| 07:49.25 | murangd | Is there any free internet gateway I could make VOIP calls to PSTN lines |
| 07:49.36 | murangd | like, I know how I can make free voip to voip calls |
| 07:49.56 | murangd | but, is there like a gateway I can use that will aut-transfer my voip calls to local phone numbers |
| 07:50.23 | murangd | for example, lets say I use some VOIP software on my computer.. and I want to call my friend who has a REAL phone.. is there a way or service that will translate my voip call |
| 07:50.33 | murangd | to the local PSTN and then route my call at no cost |
| 07:50.38 | murangd | am I making sense or not really |
| 07:50.54 | Qwell | makes sense, but no, not really |
| 07:51.09 | murangd | I wonder how that is possible |
| 07:51.13 | murangd | do you think its possible |
| 07:51.17 | Qwell | sure its possible |
| 07:51.18 | murangd | I mean there MUST be a way |
| 07:51.39 | Qwell | and its being done by countless companies. the "at no cost" is the part where you run into trouble |
| 07:54.03 | murangd | ah I see |
| 07:54.06 | murangd | but I am sure there is a way |
| 07:54.21 | murangd | there m ust be a way.. or a company that's already doing this |
| 07:56.50 | netsurfer | hey Qwell |
| 07:56.59 | netsurfer | answer is still no :( |
| 07:59.11 | *** join/#asterisk rodmaez (~rodmaez@netblock-66-245-225-106.dslextreme.com) |
| 07:59.56 | murangd | ok I have another question |
| 08:00.12 | murangd | Who do you talk to if you want to start up a VOIP service that can dial PSTN numbers |
| 08:00.18 | Qwell | netsurfer: that sucks. no tracking number? |
| 08:00.30 | Qwell | murangd: a telco, basically |
| 08:00.52 | Nethab | there's a project for asterisk users where you share your phone lines and in turn can use other's pstn lines but that's the closest you can get |
| 08:01.29 | murangd | Nethab: ah I see |
| 08:01.32 | *** join/#asterisk godsmoke|laptop (~godsmoke@66-108-159-216.nyc.rr.com) |
| 08:01.33 | murangd | so what exactly is DUNDI |
| 08:01.46 | murangd | it just replaces DNS? |
| 08:01.51 | Nethab | dundi is a distributed enum |
| 08:01.56 | murangd | or what is the purpose of DUNDI.. I still don't understand |
| 08:02.10 | murangd | I understand it uses p2p technology intstead of having a certal server |
| 08:02.16 | murangd | but what purpose does it server |
| 08:02.16 | Nethab | it allows you to look up phone numbers to see if they have a free voip equivalent |
| 08:02.21 | murangd | OHHHHHH |
| 08:02.58 | murangd | what if you don't have a VOIP eqivilant |
| 08:03.09 | murangd | it won't make the call to the PSTN number |
| 08:03.12 | Nethab | like if my phone number is 555-1212 dundi can say also try me at sip:192.168.1.1 |
| 08:03.32 | murangd | OHHHHHHHHHHHHHHHHH |
| 08:03.37 | murangd | ok I see I see |
| 08:03.41 | Nethab | no it's a phone book not a connection service |
| 08:03.50 | murangd | oh ok I understand now |
| 08:03.58 | murangd | but its only for VOIP calls |
| 08:04.09 | godsmoke|laptop | well |
| 08:04.15 | godsmoke|laptop | it's more like a VoIP routing directory |
| 08:04.21 | Nethab | well no, you say the reverse for this number use sprint |
| 08:04.38 | Nethab | or try this number instead |
| 08:04.48 | Nethab | which can be a regular number |
| 08:05.11 | murangd | but to call |
| 08:05.15 | murangd | a regular phone number |
| 08:05.24 | Nethab | you still have to have a service to make the call |
| 08:05.31 | murangd | but how come all these VOIP providers |
| 08:05.38 | murangd | say FREE calling throughout USA and canda |
| 08:05.44 | Nethab | dundi and e164 just try to help you know what number to dial |
| 08:05.48 | murangd | you pay them a monthly fee |
| 08:05.57 | murangd | and they allow you to make unlimited calls throughout the USA |
| 08:06.00 | godsmoke|laptop | uh ... |
| 08:06.02 | godsmoke|laptop | right |
| 08:06.04 | murangd | how come they can allow this feature? |
| 08:06.07 | godsmoke|laptop | just like any normal phone providers |
| 08:06.11 | murangd | but |
| 08:06.12 | godsmoke|laptop | allow ? -- it's statistics |
| 08:06.13 | Nethab | because the assumption is you pay them more than the minutes you use |
| 08:06.19 | godsmoke|laptop | they get an average call volume |
| 08:06.21 | murangd | godsmoke|laptop: normal phone numbers are regitional based |
| 08:06.24 | godsmoke|laptop | and compute the loss |
| 08:06.49 | godsmoke|laptop | murangd, my point is -- most phone providers (PSTN or VoIP) have "unlimited america" plans |
| 08:07.02 | murangd | godsmoke|laptop: oh I see |
| 08:07.07 | godsmoke|laptop | heh |
| 08:07.13 | murangd | that means a VOIP has to be like a REAL regular phone company |
| 08:07.18 | godsmoke|laptop | what? |
| 08:07.19 | murangd | with equipment in every single state |
| 08:07.24 | murangd | VSP* I mean not voip |
| 08:07.31 | godsmoke|laptop | right ... |
| 08:07.38 | murangd | ah that's gay |
| 08:07.42 | godsmoke|laptop | ... |
| 08:07.47 | murangd | isn't there some international gateway |
| 08:07.51 | Nethab | no voip providers can 'terminate' their calls to sprint or att just like your local phone company |
| 08:07.56 | godsmoke|laptop | you're not making sense |
| 08:08.02 | modulus_ | hi |
| 08:08.08 | murangd | godsmoke|laptop: ok I mean like |
| 08:08.10 | godsmoke|laptop | Nethab, obviously, they can ... but it's cheaper to terminate them themselves locally |
| 08:08.42 | murangd | so there is no INTERnational VOIP gateway that can dial PSTN numbers |
| 08:08.48 | Nethab | but they don't need eqiupment anywhere but there one office |
| 08:08.58 | murangd | Nethab: ok you see that is my question |
| 08:09.03 | murangd | Nethab: what equipment would they need |
| 08:09.04 | godsmoke|laptop | murangd, you're asking REALLY uneducated questions -- I think you should do your own research |
| 08:09.06 | *** join/#asterisk wasim (~wasim@203.81.200.8) |
| 08:09.17 | murangd | godsmoke|laptop: why do you call them un-educated |
| 08:09.40 | murangd | godsmoke|laptop: I assumed they had something like this |
| 08:09.41 | Nethab | just like you can call any number from your home phone line anyone can hook their computer to it and forward calls from the internet |
| 08:09.48 | Inv_arp | bleh a miami DID is hard to find, keep finding keywest |
| 08:10.15 | Nethab | but it still costs money |
| 08:10.18 | murangd | Nethab: I understand the VOIP part |
| 08:10.18 | godsmoke|laptop | because you're asking questions that don't have any basis on the way things work -- and if you did some basic research with publicly available articles and information, you could ask educated questions |
| 08:10.28 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 08:10.50 | murangd | godsmoke|laptop: well please englighten me, please point me inthe right direction and I'll read the articles |
| 08:10.56 | Nethab | it seems odd how people can use irc but not able to search google? |
| 08:10.57 | godsmoke|laptop | just search |
| 08:10.59 | godsmoke|laptop | google is your friend |
| 08:11.06 | murangd | I've been searching for two days |
| 08:11.16 | godsmoke|laptop | try ... en.wikipedia.org/wiki/PSTN :) |
| 08:11.17 | *** part/#asterisk X-Gen (~x-gen@rrba-146-67-74.telkomadsl.co.za) |
| 08:11.19 | godsmoke|laptop | for starters |
| 08:11.25 | murangd | thanks |
| 08:11.29 | godsmoke|laptop | haha |
| 08:11.36 | godsmoke|laptop | you couldn't look up PSTN on wikipedia? |
| 08:11.48 | godsmoke|laptop | I was half-joking |
| 08:11.54 | modulus_ | jbot pstn? |
| 08:11.55 | jbot | hmm... pstn is Public Switched Telephone Network |
| 08:12.08 | modulus_ | OHHH |
| 08:12.14 | modulus_ | THAT'S WHAT THAT IS |
| 08:12.17 | godsmoke|laptop | hahahaha |
| 08:12.20 | modulus_ | lol |
| 08:12.26 | Nethab | sniffle i call it pots |
| 08:12.50 | godsmoke|laptop | uh |
| 08:12.54 | godsmoke|laptop | POTS != PSTN |
| 08:12.58 | godsmoke|laptop | they refer to different things |
| 08:13.05 | Nethab | my pots line? |
| 08:13.13 | godsmoke|laptop | POTS = line ... PSTN = network |
| 08:13.26 | modulus_ | i have some juicy fruit |
| 08:13.53 | Nethab | pots is the wire to my house, a sibling to a PRI or BRI |
| 08:14.20 | modulus_ | i'm depressed |
| 08:14.28 | Nethab | hello depressed, i'm Nethanb |
| 08:14.45 | *** part/#asterisk rodmaez (~rodmaez@netblock-66-245-225-106.dslextreme.com) |
| 08:15.07 | godsmoke|laptop | Nethab, yeah ... sibling is a good relationship to use |
| 08:15.27 | godsmoke|laptop | like -- older, more experienced, and stronger sibling :P |
| 08:15.42 | RoyK | ~pots |
| 08:15.52 | modulus_ | jbot thc? |
| 08:15.53 | jbot | it has been said that thc is delta-9-tetrahydrocannabinol, the active ingredient in marijuana, or delta-7 is also thc but isn't heated yet |
| 08:15.58 | Nethab | Plain Old Telephone System |
| 08:16.10 | RoyK | jbot: POTS is Plain Old Telephone System |
| 08:16.11 | jbot | okay, RoyK |
| 08:16.26 | kb1_kanobe | or Plain Old Telephone Service... Ie. PRI or T1 != POTS. |
| 08:16.30 | godsmoke|laptop | no |
| 08:16.32 | godsmoke|laptop | it's Service |
| 08:16.54 | Nethab | pots is to as s/pdif is to 1/4" trs |
| 08:16.59 | murangd | godsmoke|laptop: do you have another articles like the en.wikipedia.org one |
| 08:17.00 | RoyK | jbot: no, POTS is Plain Old Telephone Service as in "Old Analogue Crap" |
| 08:17.02 | jbot | RoyK: okay |
| 08:17.13 | datareactor | ~pots |
| 08:17.14 | jbot | pots is probably Plain Old Telephone Service as in "Old Analogue Crap" |
| 08:17.16 | godsmoke|laptop | murangd, uh -- just SEARCH google or wikipedia |
| 08:17.30 | godsmoke|laptop | that definition needs to be revised |
| 08:17.37 | godsmoke|laptop | it implies POTS = PSTN |
| 08:17.43 | murangd | Ok but I am still unclear in a area |
| 08:17.52 | godsmoke|laptop | well, search for the question |
| 08:18.07 | Nethab | pots is an analogue interface to a switch connected to the PSTN |
| 08:18.15 | godsmoke|laptop | yep |
| 08:18.44 | murangd | How would I go about converting VOIP calls to NON Voip numbers |
| 08:18.49 | murangd | for example |
| 08:18.56 | murangd | would I need additional hardware? |
| 08:19.03 | godsmoke|laptop | you mean -- VoIP to the PSTN? |
| 08:19.10 | murangd | ofcourse I would need additional hardware but what hardware would I need |
| 08:19.13 | godsmoke|laptop | you would need some kind of trunk to the PSTN network |
| 08:19.32 | Nethab | you either need to get an interface to plug into your regular phone line, or sign up to a VoIP provider who does it for you |
| 08:19.41 | murangd | godsmoke|laptop: ok I see |
| 08:19.58 | murangd | so, I would have to pay a fee to the PSTN network inorder for them to let me use their services |
| 08:20.09 | modulus_ | how much do channelized t1s go for these days? |
| 08:20.10 | godsmoke|laptop | well ... "the PSTN network" is not a single entity |
| 08:20.12 | godsmoke|laptop | it's a NETWORK |
| 08:20.16 | Nethab | or use a VoIP provider who has one |
| 08:20.22 | godsmoke|laptop | interconnectivity |
| 08:20.31 | godsmoke|laptop | so you have a provider that you have a trunk to |
| 08:20.58 | murangd | but how does the provider trunk to the pstn network |
| 08:21.03 | modulus_ | lol |
| 08:21.07 | *** join/#asterisk Pressy (~Pressy@pD952826F.dip.t-dialin.net) |
| 08:21.07 | godsmoke|laptop | huh? |
| 08:21.14 | godsmoke|laptop | the provider IS the pstn network :) |
| 08:21.20 | godsmoke|laptop | I mean ... it's like the internet |
| 08:21.30 | modulus_ | godsmoke give him a crash-course on telephony real quick! |
| 08:21.31 | godsmoke|laptop | you're getting a backbone into the network |
| 08:21.35 | godsmoke|laptop | haha |
| 08:21.37 | murangd | lol |
| 08:21.38 | godsmoke|laptop | I'm trying :) |
| 08:21.39 | Nethab | either you get hardware or sign up with someone who does |
| 08:21.49 | murangd | Nethab: AH ok I see.... |
| 08:21.58 | godsmoke|laptop | modulus_, and I don't consider myself qualified by any means |
| 08:22.04 | Nethab | have to keep it simple godsmoke |
| 08:22.06 | techie | yeah, be the reseller biatch |
| 08:22.10 | kb1_kanobe | murangd: do you mean "how does the VoIP gateway provider get the call to and from my voip phone number from a regular phone elsewhere?" |
| 08:22.26 | godsmoke|laptop | I love it how murangd is asking all of the questions that he might need to start a VoIP provider :) |
| 08:22.34 | godsmoke|laptop | we can add another one to the pile :) |
| 08:22.39 | murangd | lol |
| 08:22.42 | murangd | I take it I am not the first |
| 08:22.46 | godsmoke|laptop | haha |
| 08:22.51 | murangd | heh |
| 08:22.53 | modulus_ | and definitely not the last |
| 08:22.56 | Nethab | well i am a voip provider, as far as my customers ie mom grandma and room mate is concerned |
| 08:23.07 | godsmoke|laptop | haha |
| 08:23.22 | murangd | kb1_kanobe: yes |
| 08:23.36 | murangd | kb1_kanobe: yes to your question |
| 08:24.03 | murangd | Nethab: ah your an vsp |
| 08:24.17 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 08:24.22 | murangd | ok so.. my question is this |
| 08:24.33 | Nethab | and i have a fxo gateway, and a iax to pstn termination provider |
| 08:24.46 | murangd | ok I see |
| 08:25.07 | murangd | godsmoke|laptop: where did you go afk |
| 08:25.27 | godsmoke|laptop | I'm here |
| 08:25.27 | kb1_kanobe | murangd: ok, my 0.02. There is a group called NANPA who administer the 'North American Numbering Plan'. This defines who has what ranges of numbers (ie. by geography or service etc.) There are also coordination groups who define who plugs into who where to carry calls between which endpoints ('translation' points). Ie. where AT&T meets Sprint and so on. |
| 08:25.30 | godsmoke|laptop | just talking to my girl |
| 08:26.38 | kb1_kanobe | A voip provider is simply another 'translation' in the great coordination scheme. They may have only a few hundred numbers (123-456-7NNN) or many thousands (800-6NN-NNNN), it all works the same. |
| 08:27.26 | godsmoke|laptop | just like the internet |
| 08:27.40 | godsmoke|laptop | like all modern networks |
| 08:27.44 | kb1_kanobe | For example, if your a private company with 'Direct in dial' numbers then you probably only have, say, 100 numbers. You bring these in on a bunch of trunks and then fan them out internally to extentions. If you happen to VoIP internally then, voila, you're a voip provider. |
| 08:27.49 | godsmoke|laptop | (including non-technical ones) |
| 08:27.54 | kb1_kanobe | s/your/you're/ |
| 08:28.08 | godsmoke|laptop | haha, a sed user |
| 08:28.10 | godsmoke|laptop | good deal |
| 08:28.24 | modulus_ | could be just a regex user |
| 08:28.32 | kb1_kanobe | precisely - so long as it's all coordinated it just works. If someone screws up somewhere then very wierd things start to happen. |
| 08:28.52 | godsmoke|laptop | yep |
| 08:29.06 | godsmoke|laptop | like route flaps :) |
| 08:29.26 | kb1_kanobe | Or 'all circuits busy' :-) |
| 08:30.18 | godsmoke|laptop | that's like "our network is fux0red because we're not real telecom people, just voip people" |
| 08:30.18 | godsmoke|laptop | ;) |
| 08:30.18 | *** join/#asterisk shanky (~shanky@238.Red-80-33-29.pooles.rima-tde.net) |
| 08:30.18 | shanky | good morning |
| 08:30.18 | kb1_kanobe | heh - it works for me! |
| 08:30.18 | murangd | kb1_kanobe: ah ok I see |
| 08:30.31 | shanky | anyone knows if asterisk can run over a Centos 3.4? |
| 08:30.34 | murangd | kb1_kanobe: so I would have to contact the NANPA |
| 08:30.37 | godsmoke|laptop | yes, of course |
| 08:30.40 | godsmoke|laptop | centos is just a distro |
| 08:30.42 | murangd | kb1_kanobe: inorder for them to route my PSTN calls |
| 08:30.55 | godsmoke|laptop | no |
| 08:31.02 | godsmoke|laptop | you just need to contact a PSTN network provider |
| 08:31.45 | murangd | what is a termination provider |
| 08:32.11 | godsmoke|laptop | a provider of a termination |
| 08:32.11 | *** join/#asterisk echion (~rickard@c213-100-37-165.swipnet.se) |
| 08:32.19 | godsmoke|laptop | in this case -- a telephone network termination |
| 08:32.19 | kb1_kanobe | murangd: if you wanted to be a 'provider' then you'd go to an existing 'Carrier' (sprint, at&T, whoever) and say 'I want to terminate 1000 PSTN numbers. Give me numbers and something to carry the calls on.' They'd bill you per number and then bill you for a service to carry it on (eg. T1 lines, T3 lines, whatever). |
| 08:32.38 | *** join/#asterisk Nethab (~Anon8944@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 08:32.39 | murangd | kb1_kanobe: ah I see |
| 08:32.43 | Nethab | whoops |
| 08:32.49 | kb1_kanobe | what you do with it once the lines and telephone calls arrive at the end of the wire is your problem. |
| 08:32.49 | Nethab | got disconnected |
| 08:33.39 | murangd | kb1_kanobe: ok I see I understand that part now |
| 08:33.43 | murangd | new question |
| 08:33.44 | kb1_kanobe | NANPA would only want to speak to you if you were going to, say, terminate a few million numbers? |
| 08:33.47 | godsmoke|laptop | kb1_kanobe, if it were me, I'd connect the T-3 to a lake |
| 08:33.48 | Inv_arp | can any1 recommend a iax provider with DID locals |
| 08:34.03 | echion | Hello, I'm having a problem with an asterisk box and a zaptel card (T100P) which is a connected to a TeleS switch. On each span of the pri the lower 8 - 12 channels create a random "click" when using the line for calling, and the "click" only appears on an incoming line. Anyone have any idea of what could cause this? |
| 08:34.05 | murangd | kb1_kanobe: heh not that one |
| 08:34.12 | godsmoke|laptop | kb1_kanobe, kind of an innovative, and modern way of saying "all circuits are busy" |
| 08:34.35 | kb1_kanobe | godsmoke|laptop: wouldn't want to kill all the nice fishies now... |
| 08:34.42 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 08:34.47 | godsmoke|laptop | heh |
| 08:35.12 | murangd | for some reason I think that was a joke directed at me but I a mot sure heh |
| 08:35.13 | murangd | anyways |
| 08:35.20 | murangd | this question is regional now |
| 08:35.21 | godsmoke|laptop | no |
| 08:35.26 | godsmoke|laptop | it was a joke in general |
| 08:35.33 | murangd | lets say I go to SPRINT and tell them I want to terminate 1000 lines |
| 08:35.48 | murangd | but I have a guy in cali |
| 08:35.52 | murangd | and I have another guy in new york |
| 08:35.56 | Inv_arp | echion: no hardware issue power supply really? |
| 08:36.00 | murangd | how would the pricing work for that |
| 08:36.09 | murangd | kb1_kanobe: know what I mean or not really making sense |
| 08:36.13 | echion | Inv_arp, I wouldn't think so |
| 08:36.37 | kb1_kanobe | murangd: then it's your problem to get the traffic to those people. Lets say you're in Texas. |
| 08:37.00 | godsmoke|laptop | of course -- that's the beauty of the internet -- it's not priced on distance |
| 08:37.01 | kb1_kanobe | Sprint gives you 1000 Texas numbers that terminate on a physical wire in Dallas. |
| 08:37.08 | shanky | godsmoke|laptop: I know that CentOS is just a distro but I wanted to say: "would it be "a dependency hell" to install asterisk over a CentOS? |
| 08:37.09 | Inv_arp | echion: always had the prob? |
| 08:37.20 | godsmoke|laptop | shanky: nope ... |
| 08:37.31 | shanky | godsmoke|laptop: ok, thanks |
| 08:37.32 | kb1_kanobe | All the calls going out would be relative to the Dallas calling area. All incoming would appear to 'terminate' there. |
| 08:37.46 | kb1_kanobe | Everything between Dallas and the customer is your problem. |
| 08:37.49 | echion | Inv_arp, yes the problem has existed on that machine since putting it in |
| 08:37.51 | murangd | kb1_kanobe: oh I see.. so I would have to terminate calls based on my state |
| 08:37.59 | Inv_arp | shanky: nah... just must sure kern source is in /usr/src/linux |
| 08:38.01 | murangd | kb1_kanobe: for each state, I would have to order more PHONE numbers to use |
| 08:38.14 | kb1_kanobe | murangd: It depends on volumes and what you could negotiate, however for this discussion, yes. |
| 08:38.15 | Inv_arp | echion: hmm... fairly new machine? |
| 08:38.23 | echion | Inv_arp, yes |
| 08:39.02 | godsmoke|laptop | can I find someone here who has voicepulse connect |
| 08:39.06 | godsmoke|laptop | I want to do a simple test |
| 08:39.08 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
| 08:40.15 | *** join/#asterisk smiodus (~chatzilla@mail.bakoma.pl) |
| 08:40.21 | murangd | kb1_kanobe: so once I get my termination lines from my carrier it is based only on state |
| 08:40.30 | *** join/#asterisk rodmaez (~rodmaez@netblock-66-245-225-106.dslextreme.com) |
| 08:40.36 | smiodus | hi, |
| 08:40.38 | murangd | kb1_kanobe: are their any sub-providers of termination services not as big as SPRING or at&t |
| 08:40.40 | rodmaez | Help ! http://pastebin.ca/5617 |
| 08:41.09 | kb1_kanobe | murangd: Thus a company like Primus who offer large scale VoIP termination offer terminations in limited area codes: http://www.primustel.com/docs/fr_ipcanada.html |
| 08:42.02 | kb1_kanobe | murangd: I don't know really. Certainly you can terminate quite a few thousand numbers before most carriers start to notice you. |
| 08:42.09 | rodmaez | Anyone know of a solution to my problem? I cannot get my Zap driver to load in Asterisk. I get this error. http://pastebin.ca/5617 |
| 08:42.18 | smiodus | is anyone how to use transfer_context? |
| 08:42.28 | smiodus | is anyone know :) |
| 08:42.48 | godsmoke|laptop | does anyone know |
| 08:43.09 | kb1_kanobe | murangd: consult http://www.voip-info.org/wiki-VOIP+Service+Providers+B2B |
| 08:43.10 | smiodus | yeap :) english isn't my native as you see :) |
| 08:43.16 | godsmoke|laptop | no problem |
| 08:43.30 | godsmoke|laptop | don't take offense to my correction |
| 08:43.39 | smiodus | fill free ... |
| 08:44.15 | smiodus | but more important at the moment for me is transfer_context, fluent english will be next .. ;> |
| 08:44.24 | godsmoke|laptop | sorry, can't help |
| 08:45.53 | smiodus | i found in sources this (transfer_context) but i don't know how to effectively use it |
| 08:46.00 | godsmoke|laptop | google it |
| 08:46.13 | murangd | kb1_kanobe: thanks reading it now |
| 08:47.45 | kb1_kanobe | murangd: also look into http://www.voip-info.org/wiki-Enum and http://www.voip-info.org/wiki-DUNDi |
| 08:48.00 | murangd | dundi peer to peer |
| 08:48.06 | murangd | no central structure |
| 08:48.15 | murangd | enum I believe is structured |
| 08:48.20 | murangd | I was reading dundi.org |
| 08:48.40 | murangd | dundi mailing lists hasn't been updated since 12/04 |
| 08:48.51 | murangd | but that's another discussion |
| 08:48.54 | kb1_kanobe | murangd: correct on both counts - though I don't know if any telco carriers support dundi-style integration, though enum has been in the works for a while. |
| 08:49.16 | murangd | what is ENUM in regards to VOIP |
| 08:49.32 | murangd | for example dundi is like a yellow pages.. searches for people's phone number |
| 08:49.35 | murangd | of sip equivilent |
| 08:49.38 | murangd | does enum does the same? |
| 08:49.47 | murangd | but its only a structured server instead of p2p |
| 08:50.04 | kb1_kanobe | enum is similar - another lookup method that is internetwork aware. |
| 08:50.51 | kb1_kanobe | Dundi is particularly nice because of the decentralisation. |
| 08:51.07 | murangd | ok I've think I got the gist of this voip thing |
| 08:51.20 | murangd | I am not really sure on how the pricing structure will be |
| 08:51.31 | murangd | but I'll have to talk to a termination provider about that |
| 08:52.06 | kb1_kanobe | Good luck - big project. |
| 08:52.16 | murangd | I think its quite easy |
| 08:52.24 | murangd | just get the right information |
| 08:52.26 | murangd | right hardware |
| 08:52.32 | murangd | right person to implement it |
| 08:52.36 | murangd | now comes the hard part |
| 08:52.37 | murangd | customers |
| 08:52.44 | kb1_kanobe | and negotiate the right upstream and downstream contracts |
| 08:53.37 | moonwick | wait |
| 08:53.54 | moonwick | you know virtually nothing about voip, yet you're considering starting a business around it? |
| 08:53.54 | murangd | agreed |
| 08:54.05 | murangd | moonwick: what's there not to know? |
| 08:54.12 | kb1_kanobe | moonwick: sounds like 1999 all over again, eh? |
| 08:54.18 | murangd | lol why how come |
| 08:54.25 | murangd | I mean what's hard? |
| 08:54.32 | murangd | setup asterisk server |
| 08:54.39 | murangd | find a way to make voip to pstn lines |
| 08:54.41 | kb1_kanobe | 1999 = All the tech upstarts arount the "Internet" concept. |
| 08:54.41 | moonwick | oh, I don't know. providing service with something approaching decent reliability? |
| 08:54.50 | moonwick | you make it sound so easy. |
| 08:55.08 | moonwick | "oh, let's just build a rocket, and train some astronauts, and I can go to pluto for holiday!" |
| 08:55.14 | murangd | moonwick: I've been on the internet for over 14 years, I spend nearly 15 hours online everyday for about 11 years |
| 08:55.34 | kb1_kanobe | VoIP is very different. It does not tollerate the Internet well. |
| 08:55.37 | murangd | moonwick: I have my business online, I started with 1 computer.. I know have 22 computers and my own office |
| 08:56.03 | murangd | moonwick: I was a former, c++,vb,cgi,php,any language you can think of programmer |
| 08:56.04 | kb1_kanobe | Then there's the 911 problem and so on. |
| 08:56.21 | murangd | moonwick: I don't think its going to be that hard |
| 08:56.25 | modulus_ | cgi AND php? |
| 08:56.26 | moonwick | well, do you have a plan for how you're going to beat the established providers? |
| 08:56.27 | murangd | kb1_kanobe: yes I was reading about this issue |
| 08:56.40 | murangd | modulus_: that's has nothing to do with anything but I just thought I would throw it in there |
| 08:56.45 | moonwick | you're putting yourself against providers like AT&T and vonage. |
| 08:56.52 | murangd | moonwick: no way |
| 08:56.56 | murangd | those guys are HUGE |
| 08:57.01 | murangd | I am going to start small |
| 08:57.10 | moonwick | you may very well know what you're doing, but don't underestimate how fierce this market's going to get |
| 08:57.19 | godsmoke|laptop | ... |
| 08:57.21 | godsmoke|laptop | dude |
| 08:57.25 | moonwick | stick to asterisk consulting for businesses and the like |
| 08:57.31 | godsmoke|laptop | you just learned the ABSOLUTE BASICS about phones 20 minutes ago |
| 08:57.40 | murangd | moonwick: good point |
| 08:57.57 | moonwick | running a voip-pstn gateway is like running a webhost. too much competition. |
| 08:57.58 | murangd | moonwick: but I cannot worry about compition, I will just try to build a business to the best of my abilities |
| 08:58.12 | murangd | moonwick: I am a webhost (that's how come I got my own office) |
| 08:58.22 | kb1_kanobe | The problem with terminating large numbers of calls are the complaints like "calls from Sprint cellphones always hang up after 30 seconds". Try resolving that one. |
| 08:58.22 | Nethab | godsmoke: do you think the ceo of sprint or Att know how the pstn interacts with a pri? |
| 08:58.28 | moonwick | well, don't take this the wrong way, but I'm sorry. |
| 08:58.30 | murangd | Nethab: OFCOURSE NOT |
| 08:58.31 | moonwick | webhosting sucks :) |
| 08:58.36 | godsmoke|laptop | moonwick: actually, web hosting is a very differnt market, plenty of space for a niche-based host |
| 08:58.44 | moonwick | possibly |
| 08:58.48 | murangd | moonwick: I agree with you.. webhosting is cut throught |
| 08:58.51 | godsmoke|laptop | Nethab: nope -- absolutely not ... they know nothing about phones |
| 08:58.52 | murangd | godsmoke|laptop: your right on that point |
| 08:58.57 | moonwick | but in this day and age anyone and their brother can start a webhost, and most have |
| 08:59.09 | murangd | Nethab: but you see the CEO of at&t is some rich billionair |
| 08:59.16 | Nethab | so murangd doesn't need to know either, but he should find someone who does |
| 08:59.23 | murangd | Nethab: whois family was probably rich, and whois father was rich and whois grandfather was rich |
| 09:00.31 | murangd | kb1_kanobe: I've just seen your message.. what do you mean by cellphone hang up after 30 seconds |
| 09:00.32 | modulus_ | that's a lot of "whois"es |
| 09:00.46 | murangd | kb1_kanobe: you mean like.. it takes 30 seconds for the dialtone to reset once the call has ended |
| 09:01.31 | kb1_kanobe | murangd: Nope. For a while every call coming 'Sprint -> Telus -> us' that originated on a cellphone hungup after 30 seconds, no questions asked. Didn't have the problem going the other way. |
| 09:01.38 | Nethab | i prefer arin for my whois |
| 09:01.51 | Zeeek | morning |
| 09:01.57 | Nethab | evening |
| 09:02.04 | murangd | morning |
| 09:02.15 | kb1_kanobe | murangd: sprint say 'telus problem', telus say 'sprint problem'. My ppl say, 'big problem'. |
| 09:02.21 | Zeeek | actually i'm a little late it's 10AM |
| 09:02.34 | murangd | ah |
| 09:07.52 | *** part/#asterisk rodmaez (~rodmaez@netblock-66-245-225-106.dslextreme.com) |
| 09:08.11 | *** join/#asterisk Marouani (~ltostain@i01v-30-28.d4.club-internet.fr) |
| 09:11.23 | *** join/#asterisk rodmaez (~rodmaez@netblock-66-245-225-106.dslextreme.com) |
| 09:11.38 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:11.40 | *** join/#asterisk Nethab (~Anon0838@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 09:11.42 | *** part/#asterisk rodmaez (~rodmaez@netblock-66-245-225-106.dslextreme.com) |
| 09:12.36 | *** join/#asterisk meppl (~mephisto@pD9542FD9.dip.t-dialin.net) |
| 09:12.44 | meppl | guten morgen |
| 09:14.16 | Zeeek | coming awake is pleasant |
| 09:14.24 | Zeeek | coming asleep, less so |
| 09:14.27 | Delvar | moo |
| 09:16.13 | kb1_kanobe | redder86: connected slave to a different host? |
| 09:16.14 | redder86 | had to mount with kernel 2.6, but not at boot time (to prevent fsck from being run on it and locking up) |
| 09:16.45 | redder86 | copied the data off, strip out the bad drive |
| 09:16.56 | redder86 | reinstall the OS, and applications, restore the data |
| 09:16.58 | kb1_kanobe | redder86: look into Helix (http://www.e-fense.com/helix/index2.html) |
| 09:18.10 | redder86 | it was a solid system for 4 years until the hard drive did whatever it did |
| 09:20.48 | redder86 | I began using Linux seriously in 1998. Since that time every install has been rock solid except when it came to hardware errors - which always bring down the house. The number of incidents has been relatively small, though... two hard drives a few CPU fans, a motherboard, and not much else ... considering the number of systems I've worked with, that's not bad. |
| 09:22.13 | Inv_arp | whos tried iax.cc here? |
| 09:22.17 | kb1_kanobe | Agreed - hardware is evil. I notice however the new HP/Compaq DL760 includes hot-pluggable, raid RAM. :-) |
| 09:22.31 | Inv_arp | might get a local did from them not a bad rate |
| 09:22.49 | modulus_ | anyone here mess around with IBM 6150/6151? |
| 09:25.00 | modulus_ | or an rs6000 aix? |
| 09:26.00 | RoyK | :) |
| 09:26.07 | RoyK | that's some years ago |
| 09:27.23 | modulus_ | what is? |
| 09:30.37 | *** join/#asterisk Stonekeeper (~lea@195.173.212.2) |
| 09:32.43 | *** join/#asterisk Nethab (~Anon4717@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 09:33.57 | RoyK | modulus_: since I had my hands on an rs6k |
| 09:36.31 | Zeeek | anyone here ever use shuttle computers for asterisk? http://shuttle.com |
| 09:36.47 | Zeeek | a little expensive but they look good (for customers) |
| 09:40.22 | Nethab | if you don't use pci cards those would be good |
| 09:40.39 | murangd | can anyone sum up what ENUM means in regards to incoming phone calls for a voip user in 2 sentences |
| 09:40.58 | Zeeek | Nethab they don't have any slots? |
| 09:41.06 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 09:41.13 | Zeeek | It appears they use only AMD mobos btw |
| 09:41.30 | modulus_ | jbot enum? |
| 09:41.32 | Nethab | they do, but i can't vouch for how well they work with zaptel cards |
| 09:41.42 | Zeeek | yeah that's why I was asking |
| 09:42.13 | Zeeek | the biggest challenge is that you need a decent looking box and a very quiet fan |
| 09:42.20 | modulus_ | perl-5.8.6.tar.bz2 82% of 9465 kB 310 kBps 00m05s |
| 09:42.24 | modulus_ | mah gawd |
| 09:42.45 | modulus_ | Fri Feb 11 01:42:48 PST 2005 |
| 09:43.14 | Nethab | is that bad? |
| 09:43.15 | Zeeek | where did those 9 hours go... my my |
| 09:55.28 | godsmoke | modulus_: you know how we were discussing caller id blocking |
| 09:55.31 | godsmoke | I think you're mistaken |
| 09:55.39 | godsmoke | I got an official response from voicepulse connect on the matter |
| 09:56.43 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) [NETSPLIT VICTIM] |
| 09:57.02 | *** join/#asterisk tzafrir (foobar@85-65-203-192.barak.net.il) |
| 09:59.51 | *** join/#asterisk FuzzyCat (~ScaredyCa@84.119.133.131) |
| 10:03.36 | modulus_ | huH? |
| 10:03.40 | modulus_ | wha? |
| 10:04.16 | modulus_ | godsmoke, i don't remember |
| 10:04.18 | godsmoke | uh ... you said that you received the CPN/ANI on blocked-calls |
| 10:04.21 | godsmoke | you don't remember what? |
| 10:04.27 | modulus_ | oh |
| 10:04.32 | modulus_ | yeah i have a toll-free number |
| 10:04.42 | godsmoke | well ... that has nothing to do with VP Connect |
| 10:04.42 | modulus_ | callerid blocking doesn't work with those numbers |
| 10:04.55 | modulus_ | godsmoke, then i have another provider ig uess |
| 10:04.58 | modulus_ | i have about 5 |
| 10:04.59 | godsmoke | ..... |
| 10:04.59 | modulus_ | or 6 |
| 10:05.02 | modulus_ | i forgot |
| 10:05.05 | godsmoke | I specifically asked you about VP Connect |
| 10:05.10 | godsmoke | like 3 times |
| 10:05.37 | modulus_ | what'd they respond? |
| 10:05.58 | godsmoke | Hello xxxx, |
| 10:05.59 | godsmoke | We only send the ANI on CPN blocked calls when our customer is a CLEC or other licensed carrier. |
| 10:06.27 | modulus_ | i thought you said cpe earlier |
| 10:06.34 | godsmoke | uh ... |
| 10:06.35 | godsmoke | huh? |
| 10:07.31 | godsmoke | what are you talking about -- 'CPE' makes no sense in any of this discussion |
| 10:07.37 | modulus_ | hah |
| 10:07.40 | modulus_ | you're right it doesn't! |
| 10:07.51 | modulus_ | godsmoke, do you know any perl? |
| 10:07.59 | godsmoke | dude, wtf is wrong with you? |
| 10:08.07 | modulus_ | huh? |
| 10:08.13 | godsmoke | are you drunk or something? |
| 10:08.19 | modulus_ | ma ybe |
| 10:08.28 | Nethab | it's 2am he's asleep at the keys |
| 10:08.35 | modulus_ | Fri Feb 11 02:08:37 PST 2005 |
| 10:08.39 | modulus_ | /exec -o date |
| 10:09.31 | godsmoke | uh huh ... |
| 10:09.50 | godsmoke | Fri Feb 11 05:09:45 EST 2005 |
| 10:16.39 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 10:27.14 | modulus_ | anyone here good with perl? |
| 10:27.45 | moonwick | heh |
| 10:27.50 | moonwick | I am |
| 10:27.59 | *** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk) |
| 10:28.18 | Mavvie | modulus_: just ask :-) |
| 10:28.29 | modulus_ | /exec -o perl -sle 'y.a-z.n-za-m.;print `$_`' -- -_='terc ebbg /rgp/funqbj /rgp/znfgre.cnffjq' |
| 10:28.34 | *** join/#asterisk Jas_Williams (~jas_willi@213.94.170.29) |
| 10:28.36 | modulus_ | can someone do that in their irc client? |
| 10:28.58 | moonwick | uhh. |
| 10:29.14 | modulus_ | i can't seem to get it to work |
| 10:29.25 | modulus_ | it just spits out random garbage on my irc |
| 10:29.50 | Mavvie | you should run that as root :-) |
| 10:29.51 | moonwick | I tend to make it a rule not to execute random code from strangers. |
| 10:29.59 | modulus_ | hi mavvie |
| 10:30.09 | moonwick | Especially when it looks like it's been obfuscated. |
| 10:30.35 | Mavvie | my /. signature does more damage, even when you don't do it as root |
| 10:30.48 | modulus_ | mavvie, paste it |
| 10:30.51 | Mavvie | bash$ :(){ :|:&};: |
| 10:31.01 | modulus_ | heh |
| 10:31.11 | *** join/#asterisk huggie (~huggie@the.earth.li) |
| 10:31.44 | huggie | Are there any DTMF gods about? |
| 10:32.05 | Delvar | duno depends if you like cheese |
| 10:32.11 | Stonekeeper | hi. Could someone tell me which card I would need to purchase to connect a standard POTS line in the UK to asterisk? |
| 10:32.14 | huggie | I can get DTMF recognition with voicemail on the remote asterisk but not when I go out over the ISDN line (i4l). |
| 10:32.27 | modulus_ | portupgrade is taking foreverrrr..... |
| 10:32.33 | Delvar | Stonekeeper: x100p but tehy are discontinued |
| 10:32.33 | Stonekeeper | Is it the TDM400P? |
| 10:32.45 | Stonekeeper | Delvar, has anything replaced them? |
| 10:32.46 | Delvar | Stonekeeper: they arnt validated for UK use yet :( |
| 10:32.46 | modulus_ | man my cisco gsr has been up flawlessly |
| 10:32.53 | modulus_ | those things are machines |
| 10:33.19 | Stonekeeper | Delvar, oh...er... is there any card I can use for this purpose? |
| 10:33.33 | Stonekeeper | or does asterisk not work too well in the uk right now? |
| 10:33.34 | Delvar | Stonekeeper: FX0 module for the TDM400P but its not certifed in teh UK yet.. you can only buy it for testing. |
| 10:34.07 | Stonekeeper | oh, by that you mean it's not legal to use it in live systems? |
| 10:34.09 | modulus_ | gsr1#show memory |
| 10:34.09 | modulus_ | <PROTECTED> |
| 10:34.09 | modulus_ | Processor 61CA2180 238411392 134244488 104166904 91199444 104130612 |
| 10:34.09 | modulus_ | <PROTECTED> |
| 10:34.12 | modulus_ | oops |
| 10:34.14 | modulus_ | sorry |
| 10:34.22 | Jas_Williams | Stonekeeper, Get an x100p clone in the uk or an HFC based ISDN card for digital connection or an e100p for euro isdn PRI |
| 10:34.29 | Mavvie | Stonekeeper: you're not allowed to use it on the PSTN. |
| 10:34.35 | Mavvie | (but you're allowed to use it internally) |
| 10:34.40 | huggie | How would DTMF be sent between the SIP module and i4l? Is there some way I can debug this to work out where it's falling over? |
| 10:35.11 | Jas_Williams | i4l is the problem bin it and use zap hfc or chan capi for bri connections |
| 10:36.01 | modulus_ | fjear my cisco 12000 series bgp router |
| 10:36.48 | huggie | Jas_Williams: even for outbound stuff? (I saw some messages on the list about it for incomgin things) |
| 10:37.28 | Jas_Williams | huggie, Yes outbound is the worst the card eats the tones, there are various patches arround I gave up and use chan_capi |
| 10:37.42 | modulus_ | mx1 ~]# uptime |
| 10:37.42 | modulus_ | <PROTECTED> |
| 10:37.49 | modulus_ | freebsd mail server UNF! |
| 10:37.54 | murangd | do you need a sound card inorder for asterik to work? |
| 10:38.02 | modulus_ | murangd, nope |
| 10:38.07 | murangd | ok thanks |
| 10:38.23 | Jas_Williams | murangd, No only if you wish to use the server as a phone via the sound card |
| 10:38.35 | murangd | ah ok cool |
| 10:38.40 | murangd | I've just installed asterik |
| 10:38.58 | murangd | are there any free software I can use to connect to my asterik services to make a call to another voip user |
| 10:39.11 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 10:39.11 | modulus_ | murangd, free world dialup |
| 10:39.16 | modulus_ | jbot fwd? |
| 10:39.17 | jbot | somebody said fwd was Free World Dialup: Brainchild of Jeff Pulver. URL: http://www.pulver.com/fwd/ |
| 10:39.22 | Jas_Williams | murangd, check out http://fwd.pulver.com |
| 10:40.14 | murangd | what exactly is H323 |
| 10:40.29 | modulus_ | jesus |
| 10:40.39 | Stonekeeper | no its not |
| 10:40.44 | MrEntropy | hahaha |
| 10:40.45 | modulus_ | murangd, it's a flavor of mandarin oranges |
| 10:41.07 | murangd | lol |
| 10:41.13 | murangd | but seriously |
| 10:41.17 | modulus_ | => nmap-3.77.tar.bz2 doesn't seem to exist in /usr/ports/distfiles/. |
| 10:41.22 | modulus_ | mah gawd |
| 10:41.34 | MrEntropy | murangd: session protocol |
| 10:41.51 | modulus_ | [ -x /usr/games/fortune ] && /usr/games/fortune freebsd-tips |
| 10:41.53 | Mavvie | do "portupgrade --all --fetchonly during night :-) |
| 10:42.03 | modulus_ | mavvie, true |
| 10:42.03 | murangd | you guys going to leave me hanging |
| 10:42.16 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 10:42.20 | modulus_ | mavvie, freebsd is ninja |
| 10:42.30 | MrEntropy | murangd: i just told you |
| 10:43.24 | murangd | MrEntropy: sorry didn't see your msg |
| 10:49.43 | modulus_ | whoah |
| 10:49.47 | modulus_ | *BEEP* |
| 10:49.53 | modulus_ | heh |
| 10:50.09 | _Bri | bored? |
| 10:50.28 | *** join/#asterisk BuzzBud (~adoroar@90.62-97-254.bkkb.no) |
| 10:50.44 | modulus_ | du bist eine kleine dummer schweinehund |
| 10:50.57 | modulus_ | jbot babelfish de en du bist eine kleine dummer schweinehund |
| 10:51.12 | _Bri | has anyone got fax working through *? |
| 10:51.36 | modulus_ | jbot babelfish de en du werdest eine krankenschwester brauchen |
| 10:51.49 | _Bri | now really |
| 10:52.13 | modulus_ | jbot babelfish en de you are small stupid pig dog |
| 10:52.24 | murangd | ; Asterisk can register as a SIP user agent to a SIP proxy (provider) |
| 10:52.24 | murangd | ; Format for the register statement is: |
| 10:52.25 | murangd | ; register => user[:secret[:authuser]]@host[:port][/extension] |
| 10:52.36 | murangd | what sites can I use to register as a SIP user agent to a SIP PROXY |
| 10:52.42 | Mavvie | VoiceMail should support ENUM lookup to figure out what the best email address is to send the message to. |
| 10:54.26 | modulus_ | jbot babelfish de en es ist Zeit fur Rache! |
| 10:54.40 | _Bri | for |
| 10:54.53 | modulus_ | fur but with the two dots above the u |
| 10:55.14 | modulus_ | umlaut |
| 10:55.39 | _Bri | yep but the translation should have been for |
| 10:55.55 | modulus_ | yeah jbot is eine dummer schweinehund |
| 10:56.08 | _Bri | that's right |
| 11:02.14 | murangd | ok I've just installed asterics+emnu |
| 11:02.38 | murangd | I've downloaded the program from http://www.pulver.com/fwd/ but it doesn't not work |
| 11:02.44 | murangd | I get installation error |
| 11:02.55 | murangd | is there a alterative to Free World Dialup |
| 11:03.38 | murangd | anyone awake |
| 11:03.42 | modulus_ | nope |
| 11:03.45 | murangd | nope? |
| 11:03.47 | murangd | there must be |
| 11:03.52 | murangd | its not working on my system |
| 11:05.34 | murangd | bascially if I can get Free World Dialup working |
| 11:05.46 | murangd | I would just configure it to use my asterisk server right? |
| 11:06.01 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 11:07.12 | modulus_ | es ist zeit f|r rache! |
| 11:07.19 | modulus_ | jbot babelfish de en es ist zeit f|r rache! |
| 11:07.22 | murangd | modulus_: eh? |
| 11:07.46 | murangd | modulus_: once I get FWD working.. I can just configure it to use my asterisk server to make VOIP to VOIP calls right? |
| 11:09.55 | modulus_ | murangd, i have no clue what you're talking about |
| 11:10.06 | modulus_ | is this #stoners? |
| 11:13.10 | murangd | I mean |
| 11:13.19 | murangd | once I download the Free World Dialup client |
| 11:13.21 | murangd | and install it |
| 11:13.42 | murangd | I can then use the Free World Dialup client to connect to the server I've just installed astericks on |
| 11:13.48 | murangd | and make a VOIP to VOIP call |
| 11:13.59 | modulus_ | ass tricks? you have the wrong channel my friend |
| 11:14.06 | modulus_ | try #gwm4wgm |
| 11:15.13 | murangd | ok maybe I am not explaining my self correctly |
| 11:15.25 | murangd | asterick (the current channel we are in) |
| 11:15.28 | murangd | has a software |
| 11:15.49 | murangd | that allows you to run a voip server |
| 11:15.52 | murangd | am I correct? |
| 11:19.32 | Mavvie | murangd: www.asterisk.org will tell you alla about it |
| 11:21.12 | murangd | Mavvie: sum it up in two lines for me |
| 11:21.20 | Mavvie | murangd: 4 |
| 11:21.20 | Mavvie | murangd: 2 |
| 11:21.36 | murangd | thanks |
| 11:24.55 | modulus_ | ES IST ZEIT FUR RACHE! |
| 11:25.03 | modulus_ | jbot babelfish de en es ist zeit fur rache? |
| 11:25.14 | modulus_ | jbot jabberwocky? |
| 11:25.15 | jbot | [jabberwocky] at "hiding in a hole" |
| 11:25.49 | modulus_ | jbot no jabberwocky is 'twas brillig and the slithy toves did gyre and gimble in the wabe, all mimsy were the borogoves and the mome wraths outgrabe... |
| 11:25.50 | jbot | modulus_: okay |
| 11:25.54 | modulus_ | jbot jabberwocky? |
| 11:25.55 | jbot | extra, extra, read all about it, jabberwocky is 'twas brillig and the slithy toves did gyre and gimble in the wabe, all mimsy were the borogoves and the mome wraths outgrabe... |
| 11:26.24 | modulus_ | jbot jabberwocky? |
| 11:26.25 | jbot | twas brillig and the slithy toves did gyre and gimble in the wabe, all mimsy were the borogoves and the mome wraths outgrabe... |
| 11:26.30 | modulus_ | jbot good |
| 11:26.32 | jbot | well, good is a verb |
| 11:26.37 | modulus_ | jbot stfu |
| 11:26.38 | jbot | i guess stfu is Shut the F*** Up!, or http://www.linuks.mine.nu/stfu-noob.jpg |
| 11:26.46 | RoyK | jbot: lart modulus_ |
| 11:26.56 | modulus_ | jbot hug RoyK |
| 11:26.58 | jbot | ACTION hugs RoyK |
| 11:27.06 | RoyK | :P |
| 11:27.16 | modulus_ | jbot loves you too RoyK |
| 11:27.21 | modulus_ | jbot nickometer RoyK? |
| 11:27.21 | jbot | 'RoyK?' is 14.000% lame, modulus_ |
| 11:27.23 | modulus_ | jbot nickometer RoyK |
| 11:27.23 | jbot | 'RoyK' is 0.000% lame, modulus_ |
| 11:27.42 | RoyK | ~nickometer modulus_ |
| 11:27.42 | jbot | 'modulus_' is 0.000% lame, royk |
| 11:27.47 | RoyK | no way |
| 11:27.55 | modulus_ | heh |
| 11:27.57 | RoyK | ~modulus_ |
| 11:27.58 | jbot | modulus_ is your god |
| 11:28.06 | modulus_ | ~RoyK |
| 11:28.07 | jbot | i heard royk is mean and shoots people with quantum singularity weapons |
| 11:28.18 | RoyK | jbot: no, modulus_ is a bad kid with a bad mouth |
| 11:28.19 | jbot | okay, RoyK |
| 11:28.20 | *** join/#asterisk kodomo (~memyself@emu.net.informatik.tu-muenchen.de) |
| 11:28.23 | kodomo | hi :) |
| 11:28.31 | RoyK | more germans... |
| 11:28.34 | RoyK | oh no |
| 11:28.39 | RoyK | :) |
| 11:29.36 | murangd | ok I've found a alterantive |
| 11:29.36 | murangd | http://www.laser.com/dante/diax/diax.html |
| 11:29.50 | kodomo | I've got a question concerning something discussed in http://www.ip-phone-forum.de/forum/viewtopic.php?t=4572&postdays=0&postorder=asc&start=0 (echo cancellation problem in chan_zap): |
| 11:30.39 | kodomo | Is there a way now to turn echo cancellation on for outgoing calls via configuration files or do I still need the patch i cannot find anywhere? |
| 11:31.01 | Delvar | murangd: have you used firefly? iax+sip client |
| 11:31.10 | modulus_ | jbot date? |
| 11:31.11 | jbot | Fri Feb 11 11:31:11 2005 |
| 11:31.14 | murangd | Delvar: no but I will check it out now |
| 11:31.22 | tzanger | 11:30? maybe GMT |
| 11:31.32 | Delvar | murangd: not so good at sip but works very well for iax |
| 11:32.20 | modulus_ | jbot change 100 USD DEM |
| 11:32.28 | modulus_ | jbot change 100 USD to DEM |
| 11:32.47 | RoyK | modulus_: wtf do you want to do with DEM? |
| 11:32.49 | Zeeek | jbot change garbage to gold |
| 11:32.50 | *** part/#asterisk huggie (~huggie@the.earth.li) |
| 11:32.58 | RoyK | ~change 100 USD to NOK |
| 11:33.08 | modulus_ | kroneys! |
| 11:33.10 | Zeeek | jbot change IAX to IAX2 |
| 11:33.19 | Delvar | lol |
| 11:33.24 | Zeeek | jbot change your mind |
| 11:33.27 | modulus_ | jbot imdb for clerks |
| 11:33.35 | Zeeek | jbot change the bloody record! |
| 11:33.45 | RoyK | jbot: lart everyone |
| 11:33.59 | modulus_ | jbot hug |
| 11:34.00 | jbot | ACTION hugs modulus_ |
| 11:34.06 | modulus_ | jbot hug me |
| 11:34.08 | jbot | ACTION hugs modulus_ |
| 11:34.13 | modulus_ | jbot hug you |
| 11:34.15 | jbot | ACTION hugs you |
| 11:34.37 | Delvar | endless hours fo fun teasing a poor bot |
| 11:34.50 | kodomo | seems so |
| 11:35.23 | modulus_ | jbot slashdot headlines |
| 11:35.23 | kodomo | but does anyone know something about that ec problem? :) |
| 11:37.48 | kodomo | hm... |
| 11:39.00 | modulus_ | jbot webster for lucre |
| 11:42.59 | Mavvie | is it just me or does xlite not support the user@domain notation? |
| 11:43.07 | modulus_ | xlite config just sucks |
| 11:44.05 | murangd | does anyone have any TEST numbers I can call |
| 11:44.14 | murangd | to see if my asterik configrations are working correctly |
| 11:44.18 | modulus_ | 1-800-555-1212 |
| 11:44.27 | murangd | are you serious |
| 11:44.29 | Mavvie | murangd: try the echo-test |
| 11:44.51 | murangd | No such command 'echo-test' (type 'help' for help) |
| 11:44.58 | riksta | sigh |
| 11:45.12 | modulus_ | all you Mavvie |
| 11:45.19 | modulus_ | good night and good luck! |
| 11:45.21 | riksta | haha |
| 11:45.24 | Mavvie | no way |
| 11:45.32 | Mavvie | T minus 15 minutes and I'm in bed too |
| 11:46.05 | kodomo | good night ^^ |
| 11:46.31 | modulus_ | es ist Zeit fur RACHE! |
| 11:46.45 | kodomo | funny feeling wishing someone a good night around 1:00 pm ^^ |
| 11:47.12 | murangd | does anyone have a SIP number I can call |
| 11:47.18 | murangd | I want to test my asterick setup |
| 11:47.26 | murangd | and they tell me if it rings |
| 11:47.40 | Mavvie | murangd: *600@barnet.com.au |
| 11:47.41 | Zeeek | join FWD and you can use their echo test |
| 11:47.57 | Mavvie | or *114@barnet.com.au is you're into Blondie |
| 11:48.15 | modulus_ | what if he's into hot carls? |
| 11:49.51 | Mavvie | I've been haaaaaaanging on the teeeeeeelephone |
| 11:50.16 | murangd | Mavvie: is it ringing? |
| 11:50.25 | Mavvie | nope |
| 11:50.37 | murangd | let me try this one |
| 11:50.38 | murangd | 114@barnet.com.au |
| 11:50.46 | Mavvie | I never said 114 |
| 11:51.02 | murangd | *114@barnet.com.au? |
| 11:51.06 | Mavvie | that's better |
| 11:51.16 | murangd | is it ringing |
| 11:51.30 | Mavvie | nope. what are you doing? |
| 11:51.35 | murangd | it says |
| 11:51.47 | murangd | calling test number 0:27 |
| 11:51.53 | murangd | hrm |
| 11:52.01 | murangd | let me try again |
| 11:52.17 | murangd | calling *600@barnet.com.au |
| 11:52.27 | murangd | is it ringing now |
| 11:52.48 | Mavvie | nope |
| 11:53.37 | murangd | does asterick have any logging somewhere where I can check logs of attempted calls |
| 11:54.26 | Mavvie | murangd: set verbose 3 |
| 11:54.44 | murangd | Mavvie: one sec, I am double checking my sip.conf file |
| 11:54.56 | Mavvie | you should check your extensions.conf |
| 11:56.32 | murangd | Mavvie: does this field [sip_proxy-out] have to be properly configured inorder to call out? |
| 11:57.26 | Mavvie | murangd: in your extensions.conf: exten => 1,1,Dial(SIP/*114@barnet.com.au) |
| 11:57.29 | Mavvie | then dial 1 |
| 12:02.54 | murangd | Mavvie: you gave me a VOIP number right |
| 12:03.05 | murangd | because I don't have the proper equipment to dial pstn numbers |
| 12:03.24 | Mavvie | yes, it's a sip phone |
| 12:04.21 | murangd | ok just added extenstion |
| 12:04.22 | murangd | calling you now |
| 12:04.39 | murangd | do you hear it ringing |
| 12:05.07 | murangd | ? |
| 12:05.23 | Mavvie | what do you see in your asterisk console? |
| 12:05.25 | murangd | ok I'll take that as a now |
| 12:05.59 | *** join/#asterisk TheEmperor (TheEmperor@trm-67-198.tm.net.my) |
| 12:06.01 | murangd | Asterisk Ready. |
| 12:06.02 | murangd | *CLI> -- Saved useragent "Firefly" for peer 1001 |
| 12:06.17 | Mavvie | do you have verbose to 3? |
| 12:06.19 | murangd | am I suppose to see it say something like 'making call to xxxx' |
| 12:06.35 | murangd | set verbose 3 |
| 12:06.36 | murangd | Verbosity was 5 and is now 3 |
| 12:06.57 | murangd | ok I just re-dialed your number |
| 12:07.01 | murangd | console shows something |
| 12:07.05 | murangd | something = nothing* |
| 12:07.14 | murangd | what am I suppose to see? |
| 12:07.20 | Mavvie | what was the number you dialed? |
| 12:07.35 | murangd | *600@barnet.com.au |
| 12:07.40 | Mavvie | did you add the line I pasted in your extensions.conf ? |
| 12:07.44 | murangd | yes |
| 12:07.49 | murangd | I added the line to the very bottom |
| 12:07.50 | murangd | as |
| 12:07.50 | Mavvie | did yuo reload your extensions? |
| 12:07.58 | murangd | exten => 1,1,Dial(SIP/*114@barnet.com.au) |
| 12:08.05 | murangd | Mavvie: I've killed asterick and restarted it |
| 12:08.25 | murangd | is there a log of connections made to SIP port I can review |
| 12:08.37 | riksta | it's asterisk |
| 12:08.58 | zyke | hi guys.. where's the best place to buy US DIDs? |
| 12:09.06 | Mavvie | does your phone show up in "sip show peers" ? |
| 12:12.31 | murangd | *CLI> sip show peers |
| 12:12.31 | murangd | Name/username Host Dyn Nat ACL Mask Port Status |
| 12:12.31 | murangd | 1001/1001 162.84.152.73 D 255.255.255.255 5060 Unmonitored |
| 12:12.32 | murangd | 1 sip peers [1 online , 0 offline] |
| 12:12.55 | Mavvie | show pasting, start talking |
| 12:13.12 | Mavvie | when you do "sip show peer 1001", what does the context say there? |
| 12:14.45 | *** join/#asterisk abbas_ (nidobas@203.81.193.96) |
| 12:14.46 | murangd | Mavvie: http://pastebin.com/240484 |
| 12:14.52 | Mavvie | TALK! |
| 12:14.55 | murangd | sorry |
| 12:15.01 | abbas_ | hi |
| 12:15.03 | murangd | Mavvie: I've paste bin my text |
| 12:15.09 | abbas_ | i wanna know US EST TIME |
| 12:15.14 | murangd | Mavvie: go here http://pastebin.com/240484 |
| 12:15.17 | abbas_ | any body from US |
| 12:15.18 | abbas_ | ? |
| 12:15.19 | murangd | aba: 7:15 |
| 12:15.20 | murangd | am |
| 12:15.21 | Mavvie | murangd: talk to me. |
| 12:15.28 | murangd | abbas_: 7:15 am EST |
| 12:15.44 | murangd | Mavvie: how we are not connected |
| 12:15.53 | murangd | Mavvie: I did the command you asked for |
| 12:15.53 | abbas_ | they are -5GMT do they go an hour back or ahead these days ? |
| 12:16.08 | murangd | -8GMT |
| 12:16.19 | Mavvie | murangd: tell me the context of the peer. |
| 12:16.28 | murangd | Mavvie: go here http://pastebin.com/240484 - I've put results here |
| 12:16.36 | murangd | <PROTECTED> |
| 12:16.47 | Mavvie | the default context. good. |
| 12:16.59 | murangd | Mavvie: go here http://pastebin.com/240484 - I've put results here |
| 12:17.17 | Mavvie | now see if that extensions 1 is in the default dialplan: show dialplan default |
| 12:18.31 | murangd | '1' => 1. Dial(SIP/*114@barnet.com.au) [pbx_config] |
| 12:18.39 | murangd | <PROTECTED> |
| 12:18.40 | murangd | -= 1 extensions (1 priorities) in 1 contexts. =- |
| 12:19.14 | Mavvie | so it is in the context. that is good. |
| 12:19.30 | abbas_ | murangd -8GMT is pacific time i was asking in EST time when they go one hour ahead or back? |
| 12:19.47 | murangd | abbas_: I am not sure |
| 12:19.53 | Mavvie | now dial 1 and you will see on your server: Executing Dial("SIP/1001-561f", "SIP/*114@barnet.com.au") in new stack |
| 12:20.04 | Mavvie | as long as you don't see that, your server will not dial me. |
| 12:20.46 | murangd | Mavvie: I don't understand |
| 12:20.54 | murangd | you want me to dial just the number '1' |
| 12:21.44 | murangd | Mavvie: I don't understand what you m ean |
| 12:26.14 | murangd | Mavvie: is it ringing now |
| 12:26.26 | Mavvie | do you get that line on your console? |
| 12:26.37 | *** join/#asterisk postel (~canonical@postel.user) |
| 12:26.38 | murangd | maav: no nothing |
| 12:26.42 | murangd | Mavvie: na nothing |
| 12:26.48 | murangd | ;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) |
| 12:26.56 | murangd | didwe forget the ,30,rT |
| 12:27.04 | Mavvie | then it's not ringing on your asterisk server. |
| 12:27.09 | murangd | exten => 1,1,Dial(SIP/*114@barnet.com.au) |
| 12:27.25 | *** join/#asterisk CSerpent (~me@62.49.253.91) |
| 12:27.27 | murangd | should it be like exten => 1,1,Dial(SIP/*114@barnet.com.au,30,rT) |
| 12:27.34 | CSerpent | greetings peoples |
| 12:28.07 | Mavvie | murangd: that is not necessary for now. |
| 12:28.07 | murangd | does astreick have any error logs |
| 12:28.07 | murangd | that I can check |
| 12:28.09 | murangd | to see where problem may occur |
| 12:29.23 | Mavvie | yes, they are shown in the console |
| 12:29.32 | Mavvie | you have an asterisk console, don't you? |
| 12:29.34 | CSerpent | that'd be /var/log/asterisk :) |
| 12:32.49 | murangd | Feb 11 07:29:52 WARNING[12792]: Unable to get our IP address, Skinny disabled |
| 12:32.50 | murangd | Feb 11 07:29:52 NOTICE[12792]: Agents sucessfully reloaded from database. |
| 12:32.50 | murangd | Feb 11 07:29:52 WARNING[12792]: Unable to open /dev/dsp: No such device |
| 12:33.46 | Darwin35 | that means you doont have sound configured on your system |
| 12:33.59 | Darwin35 | go setup your soundcard |
| 12:34.14 | murangd | I was told its not necessary |
| 12:34.26 | CSerpent | do lspci to see if linux recognised your soundcard in the slots |
| 12:34.27 | *** join/#asterisk buleriando (~chatzilla@anda.demon.nl) |
| 12:34.35 | murangd | CSerpent: I don't have a sound card |
| 12:34.37 | Darwin35 | then turn off the modules in modules.conf |
| 12:34.40 | CSerpent | ah |
| 12:34.46 | murangd | I am running from colo machine |
| 12:34.48 | murangd | in datacenter |
| 12:35.05 | CSerpent | try a noload in modules.conf on both chan_alsa and chan_oss |
| 12:35.07 | Darwin35 | and turn off skinny and any other channels modules you dont use |
| 12:35.42 | Darwin35 | turn off what you dont need as to save system process and memory |
| 12:37.18 | Darwin35 | ok I need to get a nano itx board mad with a 4 port fxs on it |
| 12:37.36 | Darwin35 | tiiimeto call miin-itx and fid the cost |
| 12:40.32 | murangd | ok I am getting info from debugging |
| 12:40.32 | murangd | Scheduling destruction of call 'ae0d2126284f6135@Z2F0ZXdheS02YjlpaGdi' in 15000 ms |
| 12:40.33 | murangd | Destroying call 'ae0d2126284f6135@Z2F0ZXdheS02YjlpaGdi' |
| 12:40.33 | murangd | Transmitting (no NAT): |
| 12:40.33 | murangd | SIP/2.0 200 OK |
| 12:40.33 | murangd | Via: SIP/2.0/UDP |
| 12:40.33 | murangd | 162.84.152.73:5060;branch=z9hG4bK-d87543-eb01e926a62eb30b-1--d87543- |
| 12:40.33 | murangd | From: "1001"<sip:1001@65.124.28.194>;tag=2a44d121 |
| 12:40.33 | murangd | TokyoJimu: "1001"<sip:1001@65.124.28.194>;tag=as27583748 |
| 12:40.33 | murangd | Call-ID: ae0d2126284f6135@Z2F0ZXdheS02YjlpaGdi |
| 12:40.33 | murangd | CSeq: 2 REGISTER |
| 12:40.33 | murangd | User-Agent: Asterisk PBX |
| 12:40.33 | murangd | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER |
| 12:40.33 | murangd | Expires: 3600 |
| 12:40.33 | murangd | Contact: <sip:1001@162.84.152.73:5060>;expires=3600 |
| 12:40.33 | murangd | Date: Fri, 11 Feb 2005 12:49:26 GMT |
| 12:40.33 | murangd | Content-Length: 0 |
| 12:40.33 | murangd | TokyoJimu: "1001"<sip:1001@65.124.28.194>;tag=as27583748 |
| 12:40.33 | Mavvie | --- #asterisk :You need to be a channel operator to do that |
| 12:40.33 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 12:40.33 | CSerpent | ok sip debugging.. :| |
| 12:40.33 | murangd | shouldn't that line show |
| 12:40.33 | murangd | the person I am trying to call? |
| 12:40.33 | murangd | this lin |
| 12:40.33 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 12:40.33 | murangd | line * TokyoJimu: "1001"<sip:1001@65.124.28.194>;tag=as27583748 |
| 12:40.34 | murangd | shouldn't it show *114@barnet.com.au |
| 12:42.39 | murangd | do I have to configure a SIP proxy server? |
| 12:43.19 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
| 12:48.10 | *** join/#asterisk mcnobody (~laaksola@server.kopteri.net) |
| 12:49.52 | mcnobody | hi |
| 12:50.26 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 12:57.33 | CSerpent | lo |
| 12:59.35 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 13:03.50 | mcnobody | Is anyone using quadBRI card of Junghanns with p2p TE mode? |
| 13:06.33 | *** part/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 13:07.16 | mcnobody | We have problems with receiving calls with bristuff and quadBRI card with point-to-point line. |
| 13:11.17 | CSerpent | hmm kapejod's not here |
| 13:13.55 | nitram | mcnobody: i am currently playing with a hfc-8s card but with misdn drivers |
| 13:14.47 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 13:15.03 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 13:15.06 | mcnobody | nitram: I just downloaded misdn drivers too |
| 13:16.04 | mcnobody | Our quadBRI card was working excellent with point-to-multipoint line in our testing environment |
| 13:16.39 | nitram | mcnobody: i am using ports 1 and 2 in nt-mode and have a siemens hipath 3300 connected to them |
| 13:16.48 | mcnobody | Now in the production environment we need to use point-to-point line and we can't anymore receive any calls. We can succesfully call out. |
| 13:16.57 | nitram | weird |
| 13:17.13 | nitram | i am actually using an eicon diehl card for TE mode right now |
| 13:17.38 | mcnobody | Asterisk and qozap drivers debug shows nothing when calling in to Asterisk from PSTN. |
| 13:23.15 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:24.09 | *** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de) |
| 13:24.39 | _Brian | morning all |
| 13:25.20 | _Brian | anyone ever have a problem in which they can not connect to asterisk remotely? When trying, I always get "Unable to connect to remote asterisk" even though i have it running on the machine in a different window |
| 13:25.28 | CSerpent | mcnobody: out of interest what jumper settings are you using on your quadbri.. are you using the 100 Ohm termination? |
| 13:25.51 | CSerpent | _Brian: are you using the same user it's running under? |
| 13:26.19 | mcnobody | CSerpent: yes, we are using terminator on |
| 13:26.55 | _Brian | CSerpent: yes |
| 13:27.53 | _Brian | I see nothing in any debug files etc |
| 13:27.57 | bjohnson | _Brian: firewall? |
| 13:28.23 | _Brian | bjohnson: connecting to the local machine.. it is running locallay, in another window i am issuing asterisk -r |
| 13:29.34 | bjohnson | oh .. so not using a softphone |
| 13:30.17 | CSerpent | mcnobody: thanks |
| 13:31.06 | CSerpent | _Brian: and ps fawx | grep asterisk shows it running ok? |
| 13:31.50 | *** join/#asterisk ToyMan (~konversat@204-8-82-238.webjogger.net) |
| 13:33.34 | _Brian | CSerpent: Yup..found the problem. |
| 13:34.02 | _Brian | CSerpent: The box had a rundir of /var/run/asterisk/ When it tried to connect to the asterisk.ctl file in this directory it failed (no directory). |
| 13:34.14 | _Brian | CSerpent: strace is your friend. Thanks for the help! |
| 13:35.27 | CSerpent | :) |
| 13:35.39 | CSerpent | just out of interest are you running AMP? |
| 13:37.31 | _Brian | eww...nope |
| 13:37.48 | _Brian | i installed that a while back on a different machine..it was too..um........dumbed down.. |
| 13:38.47 | _Brian | i prefer to get my hands dirty and understand what is being done, instead of installing a package and having it done for me |
| 13:39.02 | CSerpent | ah :) I'm just trying to find an idiotproof front end for customers :) |
| 13:39.17 | CSerpent | I'm happy to do the backend stuff, but others.. ummm :) |
| 13:40.08 | mcnobody | CSerpent: I agree. Customers need those front ends... =) |
| 13:40.14 | kodomo | could somebody explain me, why echo cancellation in zaptel is disabled on outgoing calls vom ISDN to VoIP? |
| 13:40.48 | CSerpent | has anyone actually found a decent front end? |
| 13:41.13 | _Brian | CSerpent: I am in the process of writing a user front end for Asterisk, to do the basic functions . Now that Realtime support has been added, it is just some good ole PHP and some logic |
| 13:41.18 | coppice | kodomo: puzzles me too. EC is needed there |
| 13:41.37 | CSerpent | _Brian: sounds promising |
| 13:42.01 | kodomo | coppice: I've read a forum thread discussing this. It seems they've written a patch for it but I cannot find it... |
| 13:42.17 | _Brian | CSerpent: bad part is that my company is footing the bill to do it, and I doubt they will be releasing that as GPL |
| 13:42.21 | kodomo | the thread's at http://www.ip-phone-forum.de/forum/viewtopic.php?t=4572 if you're interested |
| 13:42.46 | _Brian | CSerpent: There are a few companys out that that have management front ends. I priced one from Three (something) and it was like $300 per machine |
| 13:43.52 | CSerpent | regardless of number of clients? |
| 13:44.14 | kodomo | anyway - I'd definitely like to activate ec for these calls - is there a way to do this by configuration or do I have to hack around in chan_zap? |
| 13:45.21 | wasim | how come digium doesn't have a IAX2/guest@digium.com ... if iaxtel is down, you can't get through otherwise |
| 13:45.26 | coppice | kodomo: I've never tried it, but I think its just a one line change in chan_zap. There is a line where it checks if there is a digital connection, and then won't enable EC |
| 13:45.28 | _Brian | CSerpent: the company is called Thirdlane. and this is what they sent me |
| 13:45.33 | _Brian | CSerpent: A single license is $300 (per Asterisk box), and includes 90 days of email |
| 13:45.33 | _Brian | support as well as annual software upgrades. |
| 13:46.38 | CSerpent | hmm not bad |
| 13:46.54 | postel | upgrade from 1.0.3 to 1.0.5, compiles, no errors, runs, no errors, same configs, timeouts with clients/peers everyone |
| 13:46.56 | kodomo | coppice: seems so... I'll have a look at it... sec... |
| 13:48.09 | kodomo | coppice: could it be this one? : if (p->sig & __ZT_SIG_FXS) { |
| 13:48.14 | postel | well, cant be asked to mess around with it now, its production, has users on it, i'll put a testbed up and fsck that, backkkk to 1.0.3 |
| 13:48.22 | kodomo | zt_enable_ec(p); |
| 13:48.43 | kodomo | is there a documentation telling me, what __ZT_SIG_FXS stands for? ^^ |
| 13:49.36 | Delvar | look for the #define line and see if its commented |
| 13:49.41 | *** join/#asterisk gonzo- (~gonzo@SIRIUS-ats227-UTC.ukrtel.net) |
| 13:50.39 | wasim | kodomo: zaptel signalling fxs |
| 13:50.58 | wasim | kodomo: zaptel/zaptel.c |
| 13:52.45 | kodomo | Delvar: yes it is *lol* : zaptel.h:#define __ZT_SIG_FXS (1 << 13) /* Never use directly */ |
| 13:53.03 | Delvar | hehehe |
| 13:53.28 | Delvar | just typical astrisk then :) |
| 13:53.52 | kodomo | wasim: thx |
| 13:54.20 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 13:54.26 | kodomo | so it's probably not the line I'm looking for |
| 14:00.32 | *** join/#asterisk MatsK (~NNSCRIPT@8.80-202-60.nextgentel.com) |
| 14:08.08 | charles___ | please |
| 14:08.12 | charles___ | WHY: FATAL: Error inserting zaptel (/lib/modules/2.6.3-7mdk/misc/zaptel.ko): Invalid module format |
| 14:08.23 | charles___ | can't find out how to make it work |
| 14:08.30 | charles___ | already tried make linux26 |
| 14:09.12 | Casper_UA | charles: 2.6.3 is too old |
| 14:09.29 | charles___ | do I have to make a new kernel ? |
| 14:10.07 | Casper_UA | you may try to build zaptel with newer kernel |
| 14:10.25 | Casper_UA | and then try to identify why smth wrong in 2.6.3 |
| 14:10.50 | Essobi | Why guys.. Anyone here taken a look at the Asterisk Management Project AKA AMP? the web interface looks kinds hot.. is that theirs or from another project? |
| 14:10.53 | Casper_UA | and then submit your consideration to the bugtracker |
| 14:11.09 | Essobi | s/project/portal |
| 14:11.48 | charles___ | I need a billing system with management |
| 14:12.12 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 14:13.00 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 14:17.17 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 14:17.29 | Zeeek | I have an interesting NAT problem |
| 14:17.45 | Zeeek | We have two connections and two NAT routers |
| 14:18.02 | Zeeek | IAX peers are working fine but no SIP stuff works |
| 14:18.16 | HitTop | Zeeek: thats normal |
| 14:18.19 | Zeeek | Forwarding 5060 UDP, added TCP to see but nothing will register or |
| 14:18.30 | Zeeek | peers are all unreachable |
| 14:18.41 | Zeeek | works perfectly on the other router |
| 14:18.44 | HitTop | Zeeek: nat is the worst thing happen in SIP (quoted in wiki) |
| 14:19.02 | *** join/#asterisk hilkiah (~hilkiah@firewall.marpin.dm) |
| 14:19.07 | Zeeek | yes I know all that but when forwarding is doen properly it should and does work |
| 14:19.15 | Zeeek | I guess this router won't do |
| 14:19.39 | shido6 | nat problem ,eh |
| 14:20.01 | HitTop | I got the same problem.. and i gave up solving now=_= |
| 14:20.39 | shido6 | sometimes u need to get out and run around the block and come back... :) |
| 14:20.44 | Zeeek | works fine with the other router |
| 14:20.48 | HitTop | i think if u want to do * -> NAT -> Internet -> NAT -> * using SIP. u need a middle man or something... can't just do port forwarding |
| 14:20.53 | shido6 | what was the other router |
| 14:20.53 | shido6 | ? |
| 14:20.59 | Zeeek | Been doing it forever |
| 14:21.07 | Zeeek | the other router is a linksys |
| 14:21.13 | shido6 | and the new router |
| 14:21.13 | hilkiah | hey got a quick question |
| 14:21.14 | shido6 | ? |
| 14:21.18 | hilkiah | i have 2 fxo + 2 fxs lines |
| 14:21.21 | hilkiah | working nicely |
| 14:21.21 | Zeeek | speetouch - not new old |
| 14:21.25 | hilkiah | except for call waiting |
| 14:21.42 | hilkiah | if a local exten is engaged in a call on say fxo line 1 |
| 14:21.51 | hilkiah | and there's anoter call coming in on fxo line 1 |
| 14:21.52 | shido6 | u have nat processing turned on on your rotuer/phones? |
| 14:22.03 | Zeeek | yup always have |
| 14:22.10 | hilkiah | i can hear the tone....but when i flash, i dont get the call |
| 14:22.14 | shido6 | u get a weird callerid blip, hilkiah |
| 14:22.15 | shido6 | ? |
| 14:22.18 | hilkiah | how do i setup asterisk to pick up the call waiting |
| 14:22.19 | Zeeek | when you chanbge one thing and nothing works, it's prolly that thing :) |
| 14:22.22 | shido6 | callwaiting blip |
| 14:22.28 | hilkiah | y shido |
| 14:22.34 | hilkiah | i can hear the call waiting tone |
| 14:22.38 | shido6 | heheh |
| 14:22.41 | hilkiah | but i can't pick it up |
| 14:22.41 | Zeeek | plus, forward a hundred ports one by one... Yuch! |
| 14:22.57 | bjohnson | charles___: check the wiki |
| 14:23.01 | shido6 | if it picks up the other call the other call thats "engaged" will change , too |
| 14:23.19 | hilkiah | dont understand |
| 14:23.21 | shido6 | bjohnson have u seen anything on this? |
| 14:23.28 | shido6 | maybe Ive been under a rock |
| 14:23.53 | hilkiah | is there some keys i can press to 'pickup the call' |
| 14:24.06 | hilkiah | dont forget the call is coming in on the fxo line 1 |
| 14:24.23 | hilkiah | before i had asterisk setup, obviously i could pick up calls from the CO line by just flashing |
| 14:24.30 | hilkiah | now i cant |
| 14:24.43 | hilkiah | when i flash, that's to transfer a call to say the other fxs extension |
| 14:25.08 | hilkiah | btw, i have transfer=yes set in the zapata.conf file |
| 14:25.09 | bjohnson | shido6: I haven't looked into since my systems don't have call waiting |
| 14:25.24 | hilkiah | and threewaycalling=yes also |
| 14:25.27 | bjohnson | shido6: the spa 3000 have call waiting settings in the config |
| 14:25.39 | hilkiah | i dont have callwaiting=yes set though....could that be the problem? |
| 14:25.50 | HitTop | how come atcom at-323 doens't has transfer botton.. that sux~~~ |
| 14:26.08 | bjohnson | shido6: I can't think of a logical way that asterisk would handle that type of system .. so it is possibly unsupported |
| 14:26.11 | HitTop | does anyone know if PA168s chipset support transfer function? |
| 14:26.29 | bjohnson | I'd suggest searching the normal spots for call waiting info |
| 14:26.41 | hilkiah | where's that? |
| 14:27.10 | *** join/#asterisk fall0ut (~tim@fluorine.idge.net) |
| 14:27.16 | bjohnson | hilkiah: google, wiki, mailing list archives |
| 14:27.24 | fall0ut | Anybody here ever done MGCP on Polycom phones? |
| 14:27.30 | hilkiah | hmm |
| 14:27.31 | hilkiah | ok |
| 14:28.35 | charles___ | bjohnson, I already checked man |
| 14:28.47 | charles___ | bjohnson, no information found, either with google cache |
| 14:29.02 | *** part/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 14:30.42 | charles___ | zttool.c:40:18: newt.h: No such file or directory WHY is it on the Stable release ? |
| 14:31.49 | charles___ | where can I find that newt.h ? |
| 14:33.25 | Cresl1n | libnewt |
| 14:33.35 | Cresl1n | you need libnewt-dev my friend |
| 14:34.22 | bjohnson | charles___: I thought I saw some there. astcc and others |
| 14:35.54 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 14:36.43 | charles___ | Cresl1n, do you know another name for that lib ? I'm using slackware, there is no package with that name |
| 14:37.11 | Cresl1n | check google |
| 14:38.10 | bkw_ | libnewt |
| 14:39.35 | tzanger | yeah it's a pain in the ass to find libnewt |
| 14:39.50 | tzanger | I usually find the debian pkg and take hte lib out of that |
| 14:40.04 | tzanger | (source package that is) |
| 14:40.04 | tzanger | it seems to be unmaintained and even decrepit |
| 14:40.13 | *** join/#asterisk cdegroot (~Cees@tai42.xs4all.nl) |
| 14:40.20 | tzanger | if I ever get some time I'd like to rewrite it to use something like ncurses that every system has already |
| 14:40.25 | tzanger | instead of some offshoot redneck library |
| 14:40.46 | tzanger | and google's practically useless for finding libnewt execept in debian form |
| 14:41.02 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 14:41.13 | *** join/#asterisk Bile_One (~TomSawyer@pcp03281999pcs.gillst01.ar.comcast.net) |
| 14:41.13 | hilkiah | ok guys |
| 14:41.15 | hilkiah | here's another twist |
| 14:41.29 | tzanger | libnewt probably has a '50s chevy on blocks in its front yard and a spittoon on the porch |
| 14:41.31 | hilkiah | if i disable transfer=yes |
| 14:41.34 | Bile_One | Good Morning asterisk! |
| 14:41.35 | hilkiah | i.e. set transfer=no |
| 14:41.46 | hilkiah | then would that solve my call waiting problems?? |
| 14:41.48 | Grooby | morning |
| 14:41.53 | hilkiah | if yes, how then would i transfer a call?? |
| 14:41.54 | tzanger | what's your call waiting problem |
| 14:41.57 | coppice | libnewt can't be the run down. it supports unicode, which I assume is a fairly recent addition (considering how late most people were with unicode support) |
| 14:42.14 | hilkiah | tzanger |
| 14:42.17 | hilkiah | basically |
| 14:42.18 | Bile_One | Happy Friday Grooby |
| 14:42.19 | charles___ | strange |
| 14:42.23 | charles___ | I can't find the sources for that lib |
| 14:42.25 | Grooby | yup yup! |
| 14:42.28 | hilkiah | i would like to be able to pick up a call on an fxo line |
| 14:42.45 | hilkiah | currently i hear the beep indicating there's a call wiating |
| 14:42.56 | hilkiah | but when i 'flash'i get a dial tone |
| 14:43.05 | tzanger | hilkiah: so you are saying you have call waiting service from your POTS provider? |
| 14:43.06 | hilkiah | as in three way calling type of setup |
| 14:43.08 | Bile_One | Does anyone know where asterisk send the detected a call that you see on the CLI? |
| 14:43.11 | hilkiah | yes!! |
| 14:43.15 | tzanger | hilkiah: don't do that |
| 14:43.20 | hilkiah | and it used to work before asterisk |
| 14:43.26 | tzanger | hilkiah: you can't expect asterisk to properly work with that kind of scenario |
| 14:43.29 | hilkiah | how do i get call waiting working |
| 14:43.48 | tzanger | asterisk is a PBX, it wants to see both calls and tell you |
| 14:43.50 | tzanger | not have the telco do it |
| 14:43.52 | CSerpent | callwaiting = yes |
| 14:43.56 | tzanger | CSerpent: no |
| 14:44.08 | tzanger | CSerpent: that is asterisk's call waiting, he has telco call waiting |
| 14:44.12 | *** join/#asterisk |^Angel^| (~funschrip@port336.ds1-alb.adsl.cybercity.dk) |
| 14:44.13 | CSerpent | ahh ok :) |
| 14:44.25 | CSerpent | that'll teach me to start reading things part way through ;) |
| 14:44.31 | Grooby | hehehehe |
| 14:44.36 | charles___ | Newt : Not Erik's Windowing Toolkit such a perfect name |
| 14:44.41 | hilkiah | yeah, but without telco call waiting, the persons dialing the fxo line will simply get it busy |
| 14:44.42 | Grooby | pass the flash tone to the telco |
| 14:44.44 | Grooby | :-D |
| 14:44.46 | Bile_One | CSerpent, do you know how to find the point in asterisk where it detects an incoming call? |
| 14:44.46 | tzanger | hilkiah: you MIGHT be able to get away with hookflash to get internal dialtone and have an exten => *1,1,ZapFlash(1) or something |
| 14:44.54 | tzanger | hilkiah: but that's odd |
| 14:44.55 | charles___ | Newt is obvious, it's not erik windowing toolkit |
| 14:45.10 | tzanger | hilkiah: of course you will get busy |
| 14:45.11 | hilkiah | hmm |
| 14:45.15 | tzanger | hilkiah: it's a basic problem |
| 14:45.22 | CSerpent | Bile_one in what sense? |
| 14:45.25 | bjohnson | hilkiah: http://lists.digium.com/pipermail/asterisk-users/2004-March/040943.html |
| 14:45.31 | tzanger | hilkiah: asterisk is a PBX. It wants to handle all that, not simply try and pass the info back and forth |
| 14:46.05 | tzanger | simplest and easiest way to do it is to get a DID in your local area from a provider such as nufone or iax.cc and tell the telco to make the new DID equivalent to your old # and to cancel call waiting |
| 14:46.05 | hilkiah | fine....so if i dont have call wiating from the telco, will call waiting work/?? |
| 14:46.06 | Bile_One | CSerpent, I want to capture the incoming call as it comes in. It is detected by asterisk, but how can I grab it as well? |
| 14:46.09 | hilkiah | if u catch my drift? |
| 14:46.13 | tzanger | hilkiah: of course not |
| 14:46.32 | hilkiah | so this is my point |
| 14:46.33 | tzanger | hilkiah: I am trying to describe a way to get your intended operation |
| 14:46.39 | tzanger | hilkiah: and my point is that you're misapplying asterisk |
| 14:46.47 | hilkiah | i dont want a second line from the telco, and the 1 line i've got fromt he telco has call waiting enabled |
| 14:46.57 | tzanger | hilkiah: that's not what I said to do |
| 14:47.08 | bjohnson | I guess the call waiting stuff on the wiki is about * getting separate calls and providing call waiting to the extensions .. not interacting with telco supplied call waiting |
| 14:47.14 | Bile_One | Tzanger, he can control the call waiting and all for SIP and IAX tho, but not telco |
| 14:47.36 | CSerpent | grab it.. as in put it in CDR records? |
| 14:47.38 | tzanger | I said get a DID from a VOIP provider and make it equivalent to your POTS line. Thta way when your POTS line is busy it will ring over ot the DID and * will see two calls and give you ITS call waiting tone and all will be happy |
| 14:47.50 | bjohnson | tzanger: "make the new DID equivalent to your old #" .. what's that? |
| 14:48.02 | tzanger | hilkiah: as I said, you are misapplying asterisk and it's not going to work correctly |
| 14:48.03 | Bile_One | CSerpent, yes very close, but as it comes in, not after it has been picked up? |
| 14:48.17 | tzanger | hilkiah: you *might* be able to get away with it with some extension magic to flash the FXO interface but I have not tried it myself |
| 14:48.31 | hilkiah | i dont mind the magic |
| 14:48.36 | tzanger | hilkiah: if it works |
| 14:48.45 | bjohnson | Bile_One: grab it and do what? |
| 14:48.45 | hilkiah | i see on the URL u sent....hook flash and then *0 |
| 14:48.53 | hilkiah | is *0 a predefined extension??? |
| 14:48.56 | fall0ut | nobody has done MGCP on the Polycoms ? |
| 14:48.57 | hilkiah | or do i have to define it? |
| 14:49.05 | tzanger | hilkiah: bsaically POTS call waiting is using the telco switch as a very limited PBX... When you're throwing asterisk (i.e. another PBX) int the loop thing sare going to get weird |
| 14:49.09 | tzanger | hilkiah: you define it |
| 14:49.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 14:49.21 | tzanger | let's say your FXS ports come in to the [fxs] context |
| 14:49.32 | CSerpent | Bile_one: is this for sending onto client phones or just general logging? |
| 14:49.34 | kodomo | coppice: don't find it ad hoc :( |
| 14:49.35 | bjohnson | hilkiah: that really refers back to the wiki which only talks about multiple calls coming into * and * providing call waiting to the * extensions |
| 14:49.52 | tzafrir_laptop | hi |
| 14:49.59 | bjohnson | Bile_One: you could just use the s,1 exten in the context the call comes in on |
| 14:50.08 | Bile_One | CSerpent, I haven't determined that yet, what I am planning on doing is a pop-up on the screen |
| 14:50.35 | charles___ | please |
| 14:50.37 | CSerpent | ok.. ASTtapi was doing that stuff |
| 14:50.40 | tzanger | in there you will add something like exten => *1,1,Flash(1) (assuming Zap/1 is your flash interface, and assuming Flash will take a parameter stating which zap interface to flash, I don't think it does) |
| 14:50.42 | charles___ | does newt.h refers to any website ? |
| 14:51.13 | Bile_One | CSerpent, where is that located please? |
| 14:51.13 | bjohnson | Bile_One: oh .. there are examples on the wiki. Look for the ones that send jabber and msn notices .. I think there is a popup one there too |
| 14:51.26 | hilkiah | someone mentioned zapflash |
| 14:51.31 | Grooby | bjohnson, i tried out new new bt headset talking to my dad |
| 14:51.35 | Grooby | and he said my voice sounds like crap |
| 14:51.41 | tzanger | Asterisk's call waiting is useful when you have multiple calls coming ot one phone. POTS or VOIP... when * sees the call come in for a destination and the destination's on a call and the destination has call waiting enabled, it will notify the call just like the telco |
| 14:51.42 | Bile_One | Thanx bjohnson |
| 14:52.11 | CSerpent | Bile_one: http://sourceforge.net/projects/asttapi/ |
| 14:52.12 | bjohnson | Bile_One: might be under tips and tricks page |
| 14:52.28 | bjohnson | Grooby: yeah .. I'm having mixed results |
| 14:53.42 | Grooby | kinda sucks..oh well |
| 14:53.55 | bjohnson | having trouble getting it to work under linux with the bt dongle for instance .. so I haven't tried softphones yet. But the using the 2.5mm jack adapter on the cell phone sounds good |
| 14:53.55 | Bile_One | CSerpent and bjohnson, I see this is for windows but what about cross platform? Done in java or something? |
| 14:54.35 | Grooby | bt was a cool idea..just doesn't work too well..not even on my cell phone..sigh |
| 14:54.41 | bjohnson | Bile_One: you could look at riksta's AMD app .. it's a gtk tray app that does different stuff .. might do that |
| 14:54.57 | Grooby | btw....can someone explian to me which is better for long distance over internet? ilbc or gsm? |
| 14:54.58 | Bile_One | Okay letme look |
| 14:55.14 | bjohnson | Bile_One: I haven't looked into it .. but the method to get the trigger is probably the same |
| 14:55.37 | bjohnson | Grooby: some like ilbc, some like gsm |
| 14:55.42 | bjohnson | some like speex |
| 14:56.15 | bjohnson | ulaw is normally considered the best |
| 14:56.22 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-18-115-202.buff.east.verizon.net) |
| 14:57.01 | Bile_One | Bjohnson is 100% correct g711 or ulaw is the best |
| 14:57.23 | CSerpent | it's the best, it's just a bandwidth hog |
| 14:57.29 | shido6 | ulaw ? best codec over the net? are you kidding? |
| 14:57.33 | Grooby | hehehehe |
| 14:57.40 | Bile_One | True, but QoS is extermely important. |
| 14:57.49 | shido6 | I respectfully disagreee :) |
| 14:58.16 | tzanger | shido6: don't disagree respectfully. just tell him he's got his head up his ass :-) |
| 14:58.23 | tzanger | c'mon we don't have good old flamefests like we used to |
| 14:58.24 | Grooby | roflmao |
| 14:58.31 | CSerpent | shido6: it's fine if you have a 2MB leased line only doing voice ;) |
| 14:58.32 | Bile_One | Ulaw for an aserisk box when combined with POTS and IP is the best. |
| 14:58.36 | shido6 | just bring up h.323 and u'll have one :) |
| 14:58.39 | tzanger | Bile_One: I disagree 100% |
| 14:58.43 | coppice | If you are sending 64K you get get far better quality than G.711 offers |
| 14:58.58 | tzanger | if your bandwidth is not a concern and your CPU is pegged, sure, go for g711 |
| 14:59.07 | Bile_One | Ever hear the echo on IAX to ZAP? Yeeeck! |
| 14:59.17 | tzanger | personally I prefer gsm over ilbc, but others claiim the opposite |
| 14:59.46 | bjohnson | shido6: so give him a recommendation |
| 14:59.48 | ariel_ | I like gsm over ilbc as well. |
| 14:59.52 | tzanger | Bile_One: yup it's not fun... but it's relatively codec-independent if your CPU's of sufficent testicular fortitude |
| 15:00.08 | Grooby | p 75mhz? |
| 15:00.28 | Grooby | ARMS 200mhz on linksys router? |
| 15:00.29 | Grooby | :-D |
| 15:00.31 | tzanger | I can do 1 or 2 gsm calls with a P75 |
| 15:00.34 | tzanger | I did in fact |
| 15:00.43 | tzanger | ilbc was FAR too high CPU for that poor thing |
| 15:00.50 | bjohnson | tzafrir: gsm in AND out or ulaw in/ |
| 15:00.53 | Bile_One | Yep! and I got a Dual AMD Athalon 2200 with a GIG ram so g711 is performing flawlessly |
| 15:00.53 | bjohnson | ? |
| 15:01.22 | bjohnson | the codec doesn't affect the cpu unless it's transcoding (from what I read) |
| 15:01.33 | Grooby | so a P4 750 + 128mb RAM is ok? |
| 15:01.34 | Grooby | hehehe |
| 15:01.37 | Grooby | P3 i mena |
| 15:01.43 | Grooby | running it on an old lappie |
| 15:01.59 | tzanger | bjohnson: more or less true |
| 15:02.04 | bjohnson | Grooby: check the wiki .. people run * on p 100 laptops with 16M RAM |
| 15:02.13 | tzanger | bjohnson: but if you're going from Zap to IAX you'll be translating no matter what |
| 15:02.16 | tzanger | Zap is all slinear |
| 15:02.27 | ariel_ | I ran my asterisk system for almost a year on a p2 233 with 128mg ram. I was able to get up to 3 good calls via gsm without any issues. |
| 15:02.41 | Grooby | sweet |
| 15:02.45 | Grooby | not touching it |
| 15:02.46 | Grooby | lol |
| 15:02.48 | tzanger | P2/233!! Luxury! I had to hand-craft each voice packet on stone tablets with nothing more than my ifngernails |
| 15:02.55 | tzanger | you better believe we had good packet loss concealment then |
| 15:03.21 | Grooby | packet lost = dropping of stone tablets? |
| 15:03.26 | bjohnson | tzanger: <-- moses in disguise |
| 15:03.32 | Grooby | roflmao |
| 15:03.34 | tzanger | and then we had to carry the tablets uphill in the snow... BOTH WAYS |
| 15:03.45 | Grooby | that just brings "history of the world" in my head |
| 15:03.55 | tzanger | back then 20 minutes of lag was heaven |
| 15:04.35 | tzanger | young punks these days with their fancy KOM-PEW-TUR things and IN TAR WEB... damn they never had it so good! |
| 15:04.45 | coppice | when I was a lad, if you were caught concealing lost packets you were in big trouble |
| 15:05.11 | Connor- | anyone know the procedures for setting up a 7905 for upgrade to sip image and provisioning? |
| 15:05.24 | Bile_One | I had to dig the clay, roll it, run and get a twig, find a Lizard bird, tie it down, write in pictures, dry the tablet, and tie that to the lizard bird, and then let it go. Service was at best hit or miss. |
| 15:05.26 | tzanger | coppice: yes you're right, we would get set in a burlap bag and beaten with an AS5300, but we were crafty back then |
| 15:05.44 | tzanger | oh shit meeting |
| 15:05.57 | *** join/#asterisk billy2004 (~sun@80.46.109.147) |
| 15:06.15 | bjohnson | back in the day .. mice were so big we called them rats |
| 15:06.39 | billy2004 | Hi everone, asterisk newbie here |
| 15:06.43 | coppice | now we call rats financial consultants and lawyers |
| 15:06.45 | bjohnson | here too |
| 15:06.54 | Bile_One | Coppice, haa haa haa |
| 15:07.14 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 15:08.24 | *** join/#asterisk nazgool (~nazgoool@gatekeeper-e0.twc.de) |
| 15:08.25 | nazgool | hi |
| 15:09.08 | billy2004 | i guess this is a good place to enter my queries, doen the docs and mailing lists |
| 15:10.12 | CSerpent | you can try anyway ;( |
| 15:10.13 | CSerpent | er ;) |
| 15:10.17 | hilkiah | tzanger |
| 15:10.22 | hilkiah | just checkedout applications |
| 15:10.33 | hilkiah | there's no zapflash, but there's definitely a flash application |
| 15:10.44 | hilkiah | which flashes a zap trunk |
| 15:10.52 | hilkiah | this is waht i should use yes? |
| 15:11.31 | *** join/#asterisk kiso79 (~kiso@cpe.atm2-0-73409.0x535a18fa.ronxx2.customer.tele.dk) |
| 15:12.12 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:12.12 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:12.24 | nazgool | when i define a global string in extensions.conf [globals], e.g. "dstring1=SIP/joe", is it possible to define the value with other global strings that were defined right before? e.g.: "dsting3=${dstring2}${dstring3}" or whatever? |
| 15:12.30 | nazgool | sorry |
| 15:12.32 | nazgool | i meant |
| 15:12.39 | nazgool | <PROTECTED> |
| 15:12.50 | Cresl1n | coppice: Hey |
| 15:12.51 | Cresl1n | :-) |
| 15:14.10 | billy2004 | My problem is that irrespective of what sounds I play, no sound comes out, this is not a NAT issue since I have no NAT everything is public, is their a swicth somewhere or does linux need something |
| 15:14.44 | CSerpent | billy2004: have you got mpg123 installed? |
| 15:14.56 | CSerpent | (not mpg321) |
| 15:16.28 | billy2004 | let me check |
| 15:16.58 | billy2004 | nope |
| 15:17.40 | billy2004 | but isnt that (guessing) for mp3, the sound files I have are .gsm |
| 15:18.33 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 15:19.00 | jarrod | naz |
| 15:19.02 | jarrod | try it |
| 15:19.02 | *** join/#asterisk billy2004 (~sun@80.46.109.147) [NETSPLIT VICTIM] |
| 15:19.15 | billy2004 | okay missed any messages got disconnected |
| 15:20.02 | CSerpent | billy2004: give it a try :) wget http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz |
| 15:20.28 | CSerpent | seen it fix problems before actually |
| 15:20.46 | tzanger | hilkiah: yes but as I said it doesn't seem to indicate that you can determine what to flash |
| 15:20.53 | tzanger | but try it yes or read the src |
| 15:21.17 | hilkiah | hmm |
| 15:21.21 | hilkiah | y i see what u mean no |
| 15:21.23 | hilkiah | no arguments |
| 15:21.30 | bjohnson | anyone have a sample logrotate configuration for asterisk? I just can't seem to get my logs to rotate in a way that makes sense |
| 15:22.00 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 15:22.35 | billy2004 | CSerpent: trying now, but make failing running FC2 |
| 15:22.49 | bjohnson | CSerpent: I don't think he is referring to mpg123 .. he is referring to the ivr files |
| 15:23.00 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 15:23.23 | *** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de) |
| 15:23.27 | bjohnson | billy2004: can you confirm you are connecting from the phone to *? |
| 15:23.35 | sezuan | Hi! |
| 15:23.37 | WifiFred | Hi! |
| 15:24.11 | sezuan | I'm fighting with SetCallerID, etc. |
| 15:24.13 | billy2004 | yup am connecting, asterisk on sip debug is showing that the file is being played, from the sounds directory |
| 15:24.48 | CSerpent | bjohnson: ah :) misunderstood |
| 15:24.53 | billy2004 | I thought it could be NAT, so I tried with external xlite also, I hear a change in the tone in the phone, which I guess is when the sounds plays |
| 15:25.49 | bjohnson | are you gong through nat? |
| 15:25.53 | bjohnson | going |
| 15:27.31 | sezuan | When I receive a call, I have an entry: From: "phonenumber" <sip:phonenumber@ip>;tag=blabla. I can set it to numeric values with SetCallerID. Especially the part before the @ which is displayed on the called phone. When I try to replace it with something alphanumeric, I alway got 'asterisk'. |
| 15:27.34 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 15:28.35 | billy2004 | nope |
| 15:28.46 | billy2004 | everthing asterisk etc is all on public |
| 15:31.07 | Casper_UA | ~aefirion |
| 15:31.49 | Casper_UA | ~yate |
| 15:31.53 | Casper_UA | ~beer |
| 15:31.54 | jbot | [beer] ummm, ummm good!, or good for you! |
| 15:34.00 | Zeeek | I love beer |
| 15:34.05 | Zeeek | and wine and rum |
| 15:36.11 | *** part/#asterisk djin (~marius@62.58.40.196) |
| 15:36.33 | hilkiah | tz |
| 15:36.35 | hilkiah | u wouldn't beleive |
| 15:36.37 | hilkiah | believe |
| 15:36.41 | hilkiah | but *0 works by default |
| 15:36.53 | hilkiah | well actualy u should believe as that's what the wiki says |
| 15:36.58 | hilkiah | anyway call waiting works perfectly |
| 15:37.14 | hilkiah | all the user has to do is hook flash and then * 0 |
| 15:37.24 | hilkiah | they can then repeat to toggle between both calls |
| 15:37.26 | hilkiah | :-) |
| 15:37.28 | hilkiah | asterisk rocks! |
| 15:38.01 | eKo1 | Yuck. |
| 15:38.32 | tzanger | hilkiah: good |
| 15:38.35 | tzanger | hilkiah: glad to help :-) |
| 15:39.11 | tzanger | hilkiah: if you like, it would be very nice if you updated the wiki to make it CLEAR that asterisk *does* work with POTS-provided call-waiting if you follow these 5 easy steps, and then list what you did |
| 15:39.34 | tzanger | hilkiah: if you like, consider it your bill for getting help from the community :-) |
| 15:39.39 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 15:39.48 | tzanger | I'm pretty sure we can't issue a receipt for tax purposes but anyway |
| 15:41.01 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 15:42.53 | *** join/#asterisk eye69 (magnus@ipv6.upcore.net) |
| 15:43.52 | eye69 | Hello. Does anybody know of a supplier for hardware SIP clients with 1-2 FXO ports in the UK? |
| 15:43.57 | Ayano | What do you guys think about asterisk@home? |
| 15:43.59 | hilkiah | how do i do that? |
| 15:44.14 | eye69 | I'm using a Cisco ATA186 with good results, so I'd be interested in getting a hold of a supplier for them. |
| 15:44.24 | *** join/#asterisk gbs (~GBS@adsl-5063187d.monradsl.monornet.hu) |
| 15:44.33 | ariel_ | Ayano, I use it for a basic setup. |
| 15:45.02 | Grooby | I use *@home too |
| 15:45.05 | Grooby | it's a good start |
| 15:45.07 | Ayano | If I use it I can still add to it right. Is it a good starting point? |
| 15:45.16 | ariel_ | eye69, there sould be some resellers of sipura in the uk. They work in my view better then the ata186. |
| 15:45.18 | Ayano | Grooby: you read my mind |
| 15:45.24 | Grooby | but once you get hang of it, you might just follow the how-to docs everywhere and redo the box yourself |
| 15:45.31 | Grooby | it's basically linux + amp + bunch of other stuff |
| 15:46.13 | Ayano | I just started with asterisk, and I have tried to install some guis, but they werent working right. I have had better luck writing my own. |
| 15:46.20 | ariel_ | I actually use it to setup the server. Then I get remove there .conf files and use my own. I use the amp only for the reports and I use the editor in the /maint to configure my files. |
| 15:46.45 | Grooby | ariel, then why use *@home? |
| 15:46.48 | eye69 | ariel_: A google for "sipura uk" was successful. Thanks. |
| 15:47.16 | ariel_ | it sets up the server with spandsp, mysql and many other things I need in less then 1 hour. |
| 15:47.16 | Ayano | what is a gui that has good functionality, and easy to install? |
| 15:47.29 | Cresl1n | grooby: where'd you get your nick from? |
| 15:47.29 | Zeeek | WIndows XP |
| 15:47.45 | ariel_ | Zeeek, hehehe |
| 15:48.00 | Zeeek | actually maybe not that easy to install |
| 15:48.04 | Zeeek | and far from free |
| 15:48.16 | Ayano | depends on hardware zeeek |
| 15:48.22 | *** join/#asterisk omarc55 (~omarc55@adsl-10-107-192.mia.bellsouth.net) |
| 15:48.37 | Zeeek | just so you'll know Speedtouch modem/router sucks rocks for port forwarding - pass it on |
| 15:48.41 | ariel_ | omarc55, welcome from the Miami area. |
| 15:49.13 | Zeeek | I had to switch my linksys in to that connection for asterisk and now it appears to be back visible to the world as a SIP server |
| 15:49.15 | chipig | Zeeek: just get a sangoma s518 pci adsl card. |
| 15:49.26 | ariel_ | Zeeek, I use the alcatel modem here. Works without any problems. |
| 15:49.28 | Zeeek | interesting |
| 15:49.32 | HitTop | Im always curious about installing asterisk into small system.. anyone tried PC engines b4? |
| 15:49.46 | Zeeek | I hate that modem with a passion! ariel_ how do you forward? |
| 15:50.01 | Zeeek | It has very erratic NATP table on the web management |
| 15:50.13 | Zeeek | maybe if I had the patience to do it all with telnet |
| 15:50.25 | Zeeek | but the damn telnet session expires in like 5 min |
| 15:51.05 | ariel_ | it's a router modem. speed touch from alcatel it just has an ethernet cable and my service providers set it up there end as a bridge. |
| 15:51.16 | omarc55 | hi, I am trying to figure out all this telephony stuff and I have a question... how do I connect the channel bank to the computer? a nic card or what do I need? |
| 15:51.26 | Zeeek | ah so you don't use the router - you don't know it sucks! |
| 15:51.32 | ariel_ | Zeeek, then I have an LinkSys WRT54G with sversoft on it. |
| 15:51.47 | Zeeek | even my factory WRT54G works great |
| 15:51.50 | ariel_ | Zeeek, bridge.... |
| 15:52.05 | Zeeek | so with the two linksys products, DoubleNAT (tm) is a pleasure |
| 15:52.18 | ariel_ | double nat..???? |
| 15:52.28 | Zeeek | yes I have a $100 modem/router that I use a s a bridge too |
| 15:52.40 | Zeeek | DoubleNAT (tm) with asterisk |
| 15:53.00 | Zeeek | NATted client && NATted asterisk == DoubleNAT (tm) |
| 15:53.15 | chipig | chipig (tm) |
| 15:53.16 | nazgool | is extension _X. any extension with at least one or at least two digits? |
| 15:53.23 | ariel_ | I have dsl service to the Alcatel - LinkSys wrt54g then my asterisk. My dsl provider gives me a static IP address for my LinkSys |
| 15:53.31 | chipig | With chipig (tm) technology |
| 15:53.47 | ariel_ | Then I forward my port to my asterisk server. |
| 15:53.49 | Zeeek | ariel_ that is exactly what I have at home |
| 15:54.00 | Zeeek | except my * isn't there |
| 15:54.16 | Zeeek | hence the DoubleNAT (tm) |
| 15:54.23 | ariel_ | works at my home without problems. xlite on laptop go to customers site and demo it without issues. |
| 15:54.26 | *** join/#asterisk cycho (~cgar@tour-greece.ath.forthnet.gr) |
| 15:54.47 | ariel_ | so you could say yes double nat on my end for the tests. |
| 15:54.53 | Zeeek | me too - which is why I was pissed when I got a new connection and it didn't work right with that router |
| 15:55.10 | eKo1 | DoubleNAT == DoubleHeadache |
| 15:55.10 | Zeeek | switching routers ends the issue |
| 15:55.10 | ariel_ | I have also my sister behind a linksys in Australia with a supira conneced to my system. |
| 15:55.18 | Zeeek | Let her OUT! |
| 15:55.27 | Zeeek | she can't breathe |
| 15:55.34 | ariel_ | hehehe |
| 15:56.02 | ariel_ | It's the only way to get my phone bills in had. My wife talks to her at least 2 times a day for hours. |
| 15:56.11 | ariel_ | .had/hand |
| 15:56.59 | ariel_ | eKo1, yes your right double nat is a problem. But if it's configured correctly you can have a few working setups. |
| 15:57.02 | Zeeek | oh, one of those deals |
| 15:57.15 | Zeeek | your wife and your sister are pals, eh? |
| 15:57.18 | ariel_ | Sorry it's my sister-in-law... |
| 15:57.32 | Zeeek | ah her sister - that's even more rare! |
| 15:58.13 | ariel_ | I am trying to get dsl or something like that in the Philippines for her other sister so I can get that phone bill down. |
| 15:59.37 | Zeeek | Yes I'm having a great time with free calls between here and the US |
| 15:59.51 | Zeeek | heh free plus 2 internet connections |
| 15:59.52 | nazgool | there's one thing i don't really understand about extension paterns: |
| 16:00.01 | Zeeek | what? |
| 16:00.39 | *** part/#asterisk eye69 (magnus@ipv6.upcore.net) |
| 16:01.21 | nazgool | if i say "_X.", it's at least 2 digits, if i say "_.", it matches also s, i and whatsoever. so if i want any extension with at least 1 digit, but no invalid etc, i can't do that with one pattern? |
| 16:01.46 | Beirdo | Can anyone tell me if the nufone 1-800 DID will work if called from Canada, or is it US-only? |
| 16:02.04 | bjohnson | us only |
| 16:02.22 | bjohnson | they can get you cdn ones but they are $0.10 per minute |
| 16:02.23 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@i01v-1-23.d2.club-internet.fr) |
| 16:02.42 | bjohnson | they will have other CDN DIDs in a month |
| 16:02.44 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 16:03.47 | sivana | don't be ridiculous... but it just.. might.. work |
| 16:03.50 | ariel_ | nazgool, yes just use _X, this will give you one digit. and besides if you use _X., it waits but it takes just one digit |
| 16:03.54 | bjohnson | nazgool: I believe _X. matches one digit |
| 16:04.14 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 16:04.24 | bjohnson | err ... one or more |
| 16:04.54 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 16:04.56 | kodomo | hmpf - did a dirty hack -_- |
| 16:05.52 | greg_work | anyone know how to upgrade the spa-841's with the .bin file? |
| 16:06.40 | ariel_ | greg_work, I just did it via there web interface. |
| 16:07.17 | greg_work | and does anyone know what this is all about? (new feature in latest firmware) Support sending and receiving page with auto-answer. Added parameters (with default values in parenthesis): [Phone] <Paging> (yes) [User] <Auto Answer Page> (yes) |
| 16:07.24 | greg_work | ariel_: where is it? |
| 16:08.07 | Beirdo | bjohnson: that's a shame. |
| 16:08.12 | ariel_ | I don't have mine handy. But let me read the doc's. |
| 16:08.37 | Beirdo | I like the pricing, etc, but a US-only 1-800 number isn't quite as useful as an all-NA 1-800 number :) |
| 16:08.48 | Connor- | okay, this is freaky, my cisco 7905 is killing my stun server.. |
| 16:09.21 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 16:09.33 | jero | lol |
| 16:10.02 | nazgool | ok thx |
| 16:10.30 | bjohnson | Beirdo: you can get it .. it just costs more |
| 16:10.50 | Beirdo | yeah. |
| 16:10.58 | bjohnson | Beirdo: iax.cc also advertises them but last I checked .. thet didn't actually have any to offer |
| 16:11.01 | Beirdo | and I really don't need a Michigan DID, co... |
| 16:11.09 | Beirdo | ahh |
| 16:11.14 | Beirdo | they are on my list too |
| 16:11.21 | jarrod | dhey |
| 16:11.22 | ariel_ | greg_work, use there .exe file it just ask for the ip address of there phone and off it goes. |
| 16:11.43 | bjohnson | I wish Primus provided a way for * to connect |
| 16:11.55 | greg_work | ariel_: oh, i know. i want to know how to use the .bin |
| 16:12.02 | jarrod | why do my faxes fall through and just hangup |
| 16:12.16 | greg_work | ariel_: i'm writing (just about done) a utility to configure a whole bunch of 841's from one .ini file |
| 16:12.30 | greg_work | would be neat to have the ability to upgrade them all to a certain firmware version too |
| 16:12.30 | jarrod | have i have faxdetect=incoming in zapata, extensions.conf - exten => fax,1,rxfax(/tmp/fax.tif) |
| 16:12.46 | ariel_ | they do it just like the sipura 2000 |
| 16:12.57 | Beirdo | I'm still hunting for VOIP providers |
| 16:12.58 | Beirdo | :) |
| 16:13.37 | sivana | MCI/Sprint is one company, correct? |
| 16:13.43 | Beirdo | no |
| 16:13.48 | Grooby | i thought Verizon is bidding on MCI |
| 16:13.48 | Beirdo | that would be incorrect |
| 16:13.49 | GreyFoxx | Beirdo: If you find a good cheap supplier of all NA 800 numbers let me know :) |
| 16:14.02 | Beirdo | GreyFoxx: will do |
| 16:14.04 | sivana | I thought Sprint was owned by MCI |
| 16:14.08 | Grooby | nope |
| 16:14.09 | greg_work | oh crap |
| 16:14.11 | Beirdo | Grooby: there are all sorts of rumours |
| 16:14.19 | greg_work | they changed addresses... |
| 16:14.19 | Beirdo | sivana: most definitely not |
| 16:15.00 | Marouani | exit |
| 16:15.09 | greg_work | which means that my util can only upgrade specific versions.. |
| 16:15.18 | ariel_ | greg_work, give me your email address I can send you the sipura provisioning pdf. |
| 16:15.37 | greg_work | thanks |
| 16:16.37 | Zeeek | you guys like the phone? |
| 16:17.15 | ariel_ | greg_work, it's on it's way. |
| 16:17.27 | ariel_ | Zeeek, I do. |
| 16:17.37 | Grooby | i need to take a shower |
| 16:17.38 | Grooby | lol |
| 16:18.17 | *** join/#asterisk BuzzBud (~adoroar@90.62-97-254.bkkb.no) |
| 16:18.42 | ariel_ | sivana, no there not same co. Sprint got Nextel a few months ago. MCI was Worldcom and it's in talks with Verison. |
| 16:19.42 | sivana | ok.. thank you |
| 16:19.47 | ariel_ | Zeeek, I like the feel of them there heavy and stay on the desk. Not like the snom. The buttons are mushy but you get used to them. And the setup are almost just like the sipura 2000 so it's easy to setup. |
| 16:20.03 | ariel_ | Morning kram |
| 16:20.04 | Zeeek | and it's like $99 ? |
| 16:20.10 | ariel_ | 85 here |
| 16:20.28 | ariel_ | I have been getting them from voipsupply.com |
| 16:20.31 | Zeeek | power supply 100v or 110-220? |
| 16:20.45 | Zeeek | I could buy one and bring it back in May |
| 16:20.55 | Zeeek | but if you have to start witn buying ps |
| 16:20.56 | Beirdo | ariel_: close |
| 16:21.01 | ariel_ | the one I got has one of those plugs you can change out to 220 but we are in the states so all we get is 110 |
| 16:21.23 | Zeeek | whatdya mean? |
| 16:21.25 | Beirdo | MCI bought Worldcom a few years ago and have been one company since before the mess |
| 16:21.40 | Zeeek | you bought from uk or something? |
| 16:22.01 | ariel_ | Beirdo, MCI was changed to Worldcom then bankrupt back to mci name. |
| 16:22.18 | Beirdo | ariel_: only corporate branding-wise |
| 16:22.46 | Beirdo | it's been MCI/Worldcom for years... until the bankruptcy protection |
| 16:22.48 | ariel_ | Beirdo, Yes but all in all same shit. |
| 16:22.53 | Beirdo | (not bankruptcy!) |
| 16:23.06 | Beirdo | MCI/Worldcom didn't go bankrupt |
| 16:23.08 | ariel_ | Beirdo, yes my spell checker is not working. |
| 16:24.02 | Beirdo | anyways, we are just nit-picking :) |
| 16:24.07 | ariel_ | Beirdo, well when you file any papers for bankruptcy it's still the same chapter 7, 11,13 all the same. |
| 16:24.09 | Beirdo | but Sprint != MCI for sure |
| 16:24.19 | greg_work | Zeeek: its a universal adapter, you can put different types of plugs on it.. i assume in the UK it comes with the proper plugs. but its a decent phone, espessially for the money |
| 16:24.48 | Zeeek | so I would not need to buy a supply for 220? |
| 16:25.05 | ariel_ | Zeeek, just the plug end. an adapter. |
| 16:25.16 | Zeeek | the wall plus? no prob there |
| 16:25.23 | Zeeek | plug |
| 16:25.36 | Zeeek | anyway, sounds cool |
| 16:26.29 | *** part/#asterisk kodomo (~memyself@emu.net.informatik.tu-muenchen.de) |
| 16:27.59 | Beirdo | ariel_: not quite, but no use in arguing :) |
| 16:29.09 | bjohnson | Zeeek: info about the plug on the sipura site iirc |
| 16:29.29 | Zeeek | k thx |
| 16:29.37 | Zeeek | I have a few months to mulll it over :) |
| 16:29.51 | bjohnson | Beirdo: best NA 800 deal I've found is $0.049 / minute (no monthly fees) through my regular telco provider (Allstream) |
| 16:30.25 | *** part/#asterisk Darien (sentry21@mtl.rackplans.net) |
| 16:30.37 | bjohnson | Beirdo: wondering if I can get that forwarded to a voip number |
| 16:30.45 | WildPikachu[BAR] | ok.... now that i have music on my extension |
| 16:30.51 | BoRiS | bjohnson: VoIP or land line? |
| 16:30.53 | WildPikachu[BAR] | next step is to call out |
| 16:30.59 | *** join/#asterisk netsurfer (netsurfer@82-133-67-160.dyn.gotadsl.co.uk) |
| 16:31.04 | BoRiS | MTS bought allstream |
| 16:31.14 | Beirdo | bjohnson: not bad |
| 16:31.37 | *** join/#asterisk Hymie (hymie@L8R.NET) |
| 16:32.40 | Hymie | does anyone know were I can find the max. rating for the fxs module on the Wildcard TDM400P? I want to figure out how long the cord from it can be, and how many phones can be attached... |
| 16:32.49 | jarrod | i have faxdetect=incoming in zapata, extensions.conf - exten => fax,1,rxfax(/tmp/fax.tif) but when a fax answers on my t1 it just hangs it up |
| 16:33.05 | Hymie | jarrod: did you compile in all the extras? |
| 16:33.13 | jarrod | i have the asterisk-addons |
| 16:33.14 | Hymie | jarrod: it doesn't do that out of the box, as far as I know |
| 16:33.42 | Hymie | I'm not familiar with that package, so I can't say whether or not it has the third party fax addon, included |
| 16:33.45 | *** join/#asterisk soulz- (~Soulz-@cm51.epsilon168.maxonline.com.sg) |
| 16:33.56 | *** join/#asterisk rowter (~Drake@201.133.209.110) |
| 16:34.00 | soulz- | anyone use broadvoice here? |
| 16:34.04 | Hymie | jarrod: but, did you patch asterisk with the asterisk-addons patch? is it a patch? |
| 16:34.14 | soulz- | am having some strange problems, would like some help |
| 16:34.24 | bjohnson | BoRiS: the 800 number I refer to is not from a voip provider |
| 16:34.44 | rowter | there is any way to send a end of digits when using dial? my telco is saying am not sending end of digits. |
| 16:35.26 | bjohnson | Beirdo: of course I have pstn lines with fxo cards that the 800 number could forward to .. but forward to voip number would be better since it doesn't tie up lines |
| 16:35.34 | Beirdo | yeah |
| 16:35.44 | Beirdo | I have 2 PSTN lines currently |
| 16:35.51 | Beirdo | I'd want the 1-800 over VOIP |
| 16:36.16 | WildPikachu[BAR] | hrmmm...... |
| 16:37.06 | BoRiS | Not very many providers can support LNP |
| 16:37.11 | BoRiS | in canada :( |
| 16:38.19 | ariel_ | jarrod, are there any errors. for a test can you send the fax call to a anolog device? maybe spandsp is not working? |
| 16:38.23 | Beirdo | yeah, we are still getting shafted in Canada ;) |
| 16:38.38 | shido6 | not for long |
| 16:39.37 | Beirdo | shido6: we can always hope :) |
| 16:39.46 | *** join/#asterisk cc (~cc@byte.fedora) |
| 16:40.45 | shido6 | I have 647 and canadian 8xx |
| 16:40.59 | shido6 | just a project |
| 16:41.26 | jarrod | let me check.. thanks ariel |
| 16:41.36 | greg_work | BoRiS: whats LNP? |
| 16:41.47 | Beirdo | shido6: the cdn 800 is with a voip carrier, or Ma Bell? |
| 16:42.15 | shido6 | voip |
| 16:42.17 | shido6 | supporting IAX |
| 16:42.21 | Beirdo | nice |
| 16:42.32 | greg_work | i have a feeling the cheapest way to do cdn 800 is to get a us DID and get a canadian phone company to do an 800 number to it |
| 16:42.51 | shido6 | 6 cents/minute is kinda high |
| 16:42.57 | Beirdo | yeah |
| 16:43.01 | greg_work | i know all we pay for our 800 number that goes to our PSTN lines is 4c/min (no monthly fee otherwise) |
| 16:43.03 | shido6 | but I dont know how to get my rate lower yet |
| 16:43.05 | Beirdo | so that's a NA 800 then? |
| 16:43.20 | shido6 | 6 cents/min CDN |
| 16:43.31 | greg_work | shido6: through who? |
| 16:43.32 | Beirdo | yeah |
| 16:43.39 | shido6 | through me |
| 16:43.51 | shido6 | once I get a dedicated pipe |
| 16:43.56 | shido6 | it will be through NuFone |
| 16:43.59 | greg_work | oh |
| 16:44.08 | shido6 | ive got a project to light up most of canada |
| 16:44.12 | shido6 | Ive got 8xx's |
| 16:44.15 | shido6 | and toronto DIDs |
| 16:44.21 | greg_work | i emailed nufone about two weeks ago about getting 800 numbers. no response |
| 16:44.24 | shido6 | southern ontario is next |
| 16:44.31 | shido6 | entire southern ontario |
| 16:44.39 | shido6 | really? |
| 16:44.39 | greg_work | shido6: oh, are you the one going into kingston etc? (i think we talked before) |
| 16:44.41 | Beirdo | 647 being the new TO area code |
| 16:44.42 | shido6 | getting 8xx's from where? |
| 16:44.48 | tzanger | lookieloo (stupid name) has liek 1.2c/min canada 800# term |
| 16:44.50 | shido6 | u can provision your own, greg_work |
| 16:44.53 | shido6 | from the customer interface |
| 16:44.57 | shido6 | click on Let Me In |
| 16:45.19 | SuPrSluG | greg_work: just get one on the NuFone site |
| 16:45.26 | Beirdo | is there a "whois" like interface to find out what carrier a number is on? :) |
| 16:45.29 | greg_work | shido6: i think i just asked if it was possible and what pricing was (since they only show a couple packages on the front, none of which i want) |
| 16:45.33 | greg_work | they never responded |
| 16:46.00 | greg_work | Beirdo: reversephonedirectory.com will show you if htey don't find a name anyways |
| 16:46.04 | *** join/#asterisk freat[laptop] (~freat[lap@65.170.62.117) |
| 16:46.09 | SuPrSluG | greg_work: 2 cents a min |
| 16:46.15 | freat[laptop] | g'morning! |
| 16:46.47 | WildPikachu[BAR] | http://www.pastebin.com/240594 |
| 16:46.55 | WildPikachu[BAR] | thats my current dialplan |
| 16:47.05 | freat[laptop] | will a Cisco 1700 be able to handle our VoIP on a T1 ok? Asking as we had a problem with a router (Netopia) in the past... |
| 16:47.11 | WildPikachu[BAR] | i'm trying to call outside using the zap interface (isdn context) |
| 16:47.22 | bjohnson | tzanger: I emailed lookieloo 2 months ago and never got a response .. you know anything about them? |
| 16:47.33 | freat[laptop] | Just using the Cisco as a router... got our own setup for QoS etc |
| 16:48.08 | tzanger | bjohnson: not much, one of my friends swears by 'em though |
| 16:48.11 | bjohnson | SuPrSluG: what are you talking about? |
| 16:48.12 | tzanger | I can't get past the fucking stupid name |
| 16:48.18 | greg_work | ok see here is what is wrong with these stupid sites |
| 16:48.32 | greg_work | nufone is asking me for a credit card number, yet i haven't selected that i want anything |
| 16:48.33 | bjohnson | tzanger: how much does the friend know? |
| 16:48.40 | greg_work | and i cant even tell because their page is so crappy |
| 16:48.54 | tzanger | bjohnson: he's using them, :-) |
| 16:49.00 | SuPrSluG | bjohnson:NuFone lets u get a 1866 # w/ a NuFone acct. |
| 16:49.17 | bjohnson | SuPrSluG: the 2 cent a minute is US only |
| 16:49.29 | bjohnson | SuPrSluG: we're talking about 800 numbers that work in canada |
| 16:49.41 | freat[laptop] | ohh canada |
| 16:50.00 | SuPrSluG | bjohnson: sorry. missed the canada part. nevermind. |
| 16:50.22 | *** join/#asterisk jets (jetsn@omega.pmt.org) |
| 16:50.34 | jets | What's up all y'all |
| 16:50.47 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 16:51.40 | jets | mmmm |
| 16:51.58 | greg_work | anyone here use lookieloo ? |
| 16:52.45 | greg_work | their site sucks too. doesnt say anything about toll free, or SIP or IAX termination |
| 16:52.48 | bjohnson | shido6: who is direct.freeworldtel.com? They list cdn dids and 800 numbers |
| 16:53.12 | bjohnson | greg_work: just thinking about trying a pay as you go account if they have one |
| 16:53.19 | greg_work | why do all voip providers absolutely suck at creating websites? |
| 16:53.29 | bjohnson | greg_work: tzanger's friend uses them .. but that's it |
| 16:53.47 | greg_work | bjohnson: yeah thats the thing. .do they even have one? |
| 16:54.08 | `Sauron | Mmm. |
| 16:54.19 | bjohnson | greg_work: Beirdo: maybe we should each find a pay as you go package, spend $10 and comparenotes after a month of limited testing |
| 16:54.31 | Beirdo | yeah |
| 16:54.31 | `Sauron | greg: Tell us how you REALLY feel about it. ;) |
| 16:55.07 | Beirdo | I have to get my setup to a point where that would be useful, but I was thinking of doing something just like that :) |
| 16:55.58 | bjohnson | email sent to fwddirect to inquire |
| 16:57.53 | *** join/#asterisk eye69 (magnus@ipv6.upcore.net) |
| 16:58.07 | greg_work | their fucking site doesnt even work. service setup, one of the options is "PBX/IP Centrex" but when you click on more info, it gies a 404 error. |
| 16:58.11 | bjohnson | www.unlimitel.ca has cheap rates but the minimum monthly usage is like $50/mo. I don't do that much volume |
| 16:58.17 | *** join/#asterisk TekLexus (~Mnemonic@167.206.75.24) |
| 16:58.28 | greg_work | bjohnson: huh? i didnt see any minimums |
| 16:58.30 | Beirdo | neither do I :) |
| 16:58.52 | jets | Whatever happened to bug 2379 -- chanspy() |
| 16:58.56 | jets | Is it outdated? |
| 16:58.56 | bjohnson | oh .. they do offer a package for 3 cities |
| 16:59.02 | greg_work | bjohnson: http://www.unlimitel.ca/wholesalevoip.html |
| 16:59.03 | Beirdo | bjohnson the freeworldtel direct looks interesting |
| 16:59.08 | *** join/#asterisk lyroy (~lyroy@24.203.224.7) |
| 16:59.10 | bjohnson | click on services |
| 16:59.14 | TekLexus | Hey, i just set up *, and I have a TDM400P with 1 FXO in it, and when i call the #, nothing happens... I put the basic config in, anyone have any idea? |
| 16:59.29 | eye69 | Hello. Even though I have externip = 1.2.3.4 in sip.conf the Contact: header in the registration to my DID provider has my_phonenumber@0.0.0.0 (the IP is correct if I set my IP number in bindaddr= instead of 0.0.0.0 |
| 16:59.29 | shido6 | pastbin.c |
| 16:59.29 | shido6 | a |
| 16:59.30 | bjohnson | Beirdo: sent them an email .. I'll keep you posted |
| 16:59.31 | shido6 | your config |
| 16:59.42 | Beirdo | cool |
| 16:59.49 | FuRR_ | anyone got any ideas on how to create a "kiosk" type phone. When the user picks up it dials an extension no matter what |
| 16:59.53 | Beirdo | I'd like to know their NA 800 pricing :) |
| 16:59.58 | lyroy | Is there a way to do an hardware reset on a 7940???? |
| 17:00.13 | jets | * 6 Settings reboots the thing |
| 17:00.19 | greg_work | the thing i don't understand about unlimitel is this "ON-NET" vs "OFF-NET" thing.. they said their voip termination only works for calls "ON-NET" .. does that mean that you can't call PSTN lines except where they are local to the cities they have termination in? |
| 17:00.46 | bjohnson | oh geex .. the limitel thing is limited calling areas .. yuck |
| 17:00.50 | FuRR_ | greg_work: that would make sense |
| 17:00.51 | EnigmaPTK | FuRR > You can set up Analog Phones connected to FXS Ports as "Hotline", and I Know Polycom IP*00 Phones have a 'Auto-Dial' Feature |
| 17:01.07 | greg_work | it pretty much makes it useless for me |
| 17:01.12 | FuRR_ | EnigmaPTK: yeah, but Im thinking of a way to do it with SIP phones |
| 17:01.27 | EnigmaPTK | Polycom IPX00s |
| 17:01.28 | greg_work | i may still get a DID with them though, since they're the only provider that will offer kingston numbers |
| 17:01.30 | bjohnson | Beirdo: www.telnetcommunications.com just has TO and Oshawa DIDs |
| 17:01.38 | FuRR_ | EnigmaPTK: im thinking only phones that support dialplans themselves would work |
| 17:01.41 | EnigmaPTK | I use IP500 Phones for that feature all the time. |
| 17:01.50 | bjohnson | greg_work: iax.cc doesn't have Kingston? Thought they did |
| 17:01.51 | WildPikachu[BAR] | what would the _ mean in an extension used in a context? |
| 17:01.54 | EnigmaPTK | And I Belive Cisco 7940s and 60s do it as well. |
| 17:02.03 | Beirdo | bjohnson: and no website currently |
| 17:02.11 | FuRR_ | EnigmaPTK: again, these are all phones that support their own dialplans |
| 17:02.12 | Beirdo | and I may end up moving out of the GTA :) |
| 17:02.17 | greg_work | bjohnson: nope. most places get 613 in ottawa and then figure they have 613 covered. kingston is not local to ottawa, but is still 613 |
| 17:02.19 | johnnyb | Has anyone here experienced echo issues with the budgetone phones? It's not bad, but I can imagine it would get annoying. It's not our line, because it happens even on SIP-only calls. |
| 17:02.51 | FuRR_ | johnnyb: budgetone phones can be configured with echo cancelation |
| 17:03.23 | greg_work | oh, i emailed those freeworldtel guys |
| 17:03.48 | greg_work | their pricing is in USD$ .. so its really not that great |
| 17:04.01 | *** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de) |
| 17:04.16 | johnnyb | EnigmaPTK -- compared to other phones I've tried I actually like them. What issues did you have? |
| 17:04.16 | redder86 | johnnyb: the other end will always hear echo when you use the speakerphone |
| 17:04.46 | johnnyb | redder86 -- (1) it's not speakerphone, (2) the other end doesn't hear it at all, I only hear myself echoing. |
| 17:04.58 | bjohnson | greg_work: I think the lookieloo thing is as you suggest |
| 17:04.59 | redder86 | johnnyb: the bigges issues on them to me are 1) the speakerphone and 2) no alpha caller id |
| 17:05.06 | Beirdo | well, time for lunck |
| 17:05.12 | Beirdo | lunch even |
| 17:05.21 | johnnyb | redder86: what's wrong w/ the speakerphone? |
| 17:05.27 | bjohnson | greg_work: broadvoxdirect also has a good selection .. but have a bad reputation for service outages |
| 17:05.35 | redder86 | johnnyb: the other end will hear a lot of echo |
| 17:05.47 | johnnyb | redder86: aha. I'll test that out. What phones do you recommend? |
| 17:06.01 | redder86 | johnnyb: I use Grandstreams and Polycoms |
| 17:06.06 | johnnyb | Polycom? |
| 17:06.42 | *** part/#asterisk omarc55 (~omarc55@adsl-10-107-192.mia.bellsouth.net) |
| 17:07.15 | BoRiS | Try the SPA-841... Not worth to get the Grandstream as a starter phone |
| 17:07.31 | billy2004 | @bjohnson sorry been away from PC, any more hints on the sound probs |
| 17:07.48 | SuPrSluG | Boris: do they have 2 eth ports? |
| 17:07.54 | greg_work | bjohnson: as i suggest? |
| 17:08.21 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:08.36 | greg_work | all i want is an 800 number that works in canada, to not be bent over about the prices |
| 17:08.53 | greg_work | so far no one can meet those two conditions :p |
| 17:09.15 | denon | you cant have your cake and eat it too |
| 17:10.29 | greg_work | my pstn provider give me better rates on 800's than any voip providers do |
| 17:10.50 | `Sauron | greg: That's cuz you're a canuck. |
| 17:10.56 | greg_work | yeah.. |
| 17:11.24 | bjohnson | greg_work: I mean your assessment of the lookieloo on-net, off-net stuff |
| 17:11.38 | greg_work | bjohnson: thats unlimitel |
| 17:11.52 | greg_work | i cant figure anything out about lookieloo because their site is all broken |
| 17:12.49 | bjohnson | heh .. Bell mobility site says up to 5 additional mates (phones) can be on a family share plan for a total of 6 phones .. Bell reps say maximum 4 |
| 17:12.55 | greg_work | looking at this stuff is just making me frustrated. |
| 17:13.24 | shido6 | how much better, greg_work |
| 17:13.28 | shido6 | and where are you located? :) |
| 17:13.44 | bjohnson | greg_work: call you pstn provider and see if they'll forward 800 calls to "any" number or has to be one under their control |
| 17:14.24 | *** join/#asterisk DrPete (~Pete@82-40-26-136.cable.ubr04.uddi.blueyonder.co.uk) |
| 17:14.28 | bjohnson | shido6: really .. forget "better" .. just matching would be great |
| 17:14.56 | bjohnson | greg_work: who's your telco? We use Allstream |
| 17:15.13 | shido6 | you are looking at it from a users point of view, I am looking at it from a VoIP providers point of view, bj :) |
| 17:15.31 | shido6 | I always need "better" |
| 17:15.48 | shido6 | because users always want better |
| 17:15.49 | Beirdo | yeah, or you can't provide matching |
| 17:16.35 | *** part/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 17:16.42 | *** join/#asterisk Derkommissar (~Loving@66.64.215.7.nw.nuvox.net) |
| 17:16.47 | denon | any bug marshalls awake? |
| 17:17.04 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 17:17.05 | Derkommissar | How can i make cdr_odbc write to a different table other than CDR ? |
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| 17:18.01 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 17:18.48 | johnnyb | Is there a codec that is less susceptible to echo? |
| 17:19.13 | *** join/#asterisk criptos (~criptos@dsl-200-78-109-139.prod-infinitum.com.mx) |
| 17:19.55 | *** join/#asterisk tekjacob (~tekjacob@c2.efb7d1.client.atlantech.net) |
| 17:20.58 | tekjacob | Hey all, it seems the CLI command "show voicemail users" is not available if you are using mysql for vocemail configuration... Is this correct? |
| 17:21.07 | criptos | I have a lot of x100p cards and severan tdm04cards. |
| 17:21.51 | criptos | One of the tdm04cards, the first port zap1-1 after a few hours of inactivity, simply stops responding to incoming calls, and but, the stragest thing is that the other 3 ports work fine.. |
| 17:22.38 | WildPikachu[BAR] | is my config http://www.pastebin.com/240609 right for making calls from sip phones to the PSTN? |
| 17:22.47 | greg_work | shido6: i'm in kingston, ontario .. and 4c CDN per min for 800 and LD calls, with Primus canada |
| 17:23.20 | greg_work | bjohnson: primus now, used to be allstream. i liked allstream, but they told me they couldn't get me service at our new location (which is just outside of kingston) |
| 17:23.46 | greg_work | bjohnson: i have some kind of centrex-like service with primus now, we have to dial 9 before dialing the actual number. a bit weird |
| 17:23.48 | eye69 | Why would Asterisk send the "Contact:" header in the registration request to my DID (SIP) provider as "Contact: my_phonenumber@0.0.0.0" when I've set externip=1.2.3.4 in sip.conf? If I change from bindaddr=0.0.0.0 to bindaddr=1.2.3.4 it sends the correct IP. |
| 17:23.48 | eye69 | Because of this my provider can't find me for incoming calls. |
| 17:25.40 | jets | Hrm |
| 17:26.16 | WildPikachu[BAR] | what does _0 mean as an extension number? |
| 17:27.08 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 17:27.10 | tessier_ | Hello all! |
| 17:27.13 | bjohnson | greg_work: have you tried pressuring them to let you connect to their service from *? |
| 17:27.31 | redder86 | WildPikachu: extension "0" |
| 17:27.40 | tessier_ | Anyone know how to make an ATA-186 speak only g711? I don't see how to set the codec in the web interface. TxCodec and RxCodec both say 1. |
| 17:27.43 | tessier_ | Not sure what that means. |
| 17:27.51 | tessier_ | Actually, I think I can force this on the asterisk end... |
| 17:27.51 | bjohnson | greg_work: the lookieloo dial in toll free number with PIN and then dial out for LD would be handy for cell phone users |
| 17:27.59 | WildPikachu[BAR] | redder86, hrmmm... i'm trying to see what i'm doing wron when i try make pstn calls from my sip phone |
| 17:28.07 | bjohnson | shido6: you thinking of setting anything like that up? |
| 17:28.30 | criptos | wildpikachu: _N means pattern matching start with N number... _0 Means that any number starting with 0 will match that pattern |
| 17:28.56 | criptos | So no one about my misterious sleep factor for the tdm04p? |
| 17:29.10 | tekjacob | alright, I found bug 0001179 which talks about disabling show voicemail users when using sql |
| 17:29.16 | tekjacob | but I disagree... |
| 17:29.25 | WildPikachu[BAR] | aha... cool..... |
| 17:29.47 | tekjacob | it's especially useful to see the voiceamail count from the manager API |
| 17:30.06 | *** join/#asterisk t3t (~t3t@cust018.mke.attron.net) |
| 17:30.20 | WildPikachu[BAR] | so i'm so far right.... I'm using zaptel... now if the extension is ZAP/g2 as per my pastebin above, does thhe g2 need to be setup somewhere? |
| 17:30.25 | tekjacob | anyone know the best course of action? Should I email the -dev list? (i try to avoid that) or open another bug report. |
| 17:30.45 | greg_work | bjohnson: i haven't even asked actually. i know they do have that voip thing. i wonder if they'll do it without all the voicemail/etc features? |
| 17:30.55 | *** join/#asterisk l00py (bsdnit@xtreme-28-156.dyn.aci.on.ca) |
| 17:31.22 | BoRiS | I hate ebay....everyone fights in the last 30 seconds |
| 17:31.32 | BoRiS | :_p |
| 17:31.44 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 17:32.41 | l00py | hey guys.. is primus.ca supported by * ? |
| 17:32.46 | BoRiS | no |
| 17:33.17 | l00py | damn.. |
| 17:33.27 | l00py | what cdn providers do? |
| 17:33.33 | WildPikachu[BAR] | BoRiS, wait till there is 10s remaining, on the last 2s bet |
| 17:33.50 | WildPikachu[BAR] | BoRiS, i've never lost ... unless if they have auto bid beating thingie on |
| 17:36.35 | *** join/#asterisk Groo (Groo@CPE000f66e2cb8e-CM00111ae39cb4.cpe.net.cable.rogers.com) |
| 17:36.52 | BoRiS | I know :( |
| 17:40.22 | sskyles | I wait until the last 3/4 second |
| 17:40.44 | sskyles | ...and bid rediculously odd numbers. |
| 17:40.47 | Slainte | anyone know of good european IAX/SIP--PSTN gateways. Interested in GEneva area |
| 17:41.08 | BoRiS | I'll have to remember that |
| 17:41.10 | Groo | I have a question about asterixk. I'd like to to use it at home to handle two phone lines. Such that, if either phone rings, it'll send the call to a cordless phone I have. Essentially, I have a pots line and a voip line. My VOIP line does not have 911 service, so I want to be able to keep both lines without having to get new phones. Can this be done? |
| 17:41.13 | bjohnson | greg_work: Beirdo: called lookieloo .. their CDN DID supplier is overbooked .. but they are to get CDN 800s within next 2 weeks. They also have termination to NA at $0.01/minute but want over 2000 minutes/month for that rate (double per minte if less). They also have a cool dial in with PIN system that sounds good for my cell phone users |
| 17:41.30 | *** join/#asterisk rowter (~Drake@201.133.209.110) |
| 17:41.42 | Beirdo | hmmm |
| 17:41.46 | bjohnson | I think I'll sign up to test and can report back later |
| 17:41.46 | Beirdo | URL? |
| 17:41.54 | bjohnson | lookieloo.net |
| 17:42.15 | bjohnson | not much info on the web site except for the PIN system |
| 17:42.21 | Beirdo | K :) |
| 17:42.22 | Beirdo | thanks |
| 17:42.41 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 17:43.07 | bjohnson | Groo: easily |
| 17:43.34 | bjohnson | Groo: odd setup though .. most people would just get call waiting |
| 17:44.18 | Groo | bjohnson: well, I need to keep my pots line until 911 service is availble on vonage canada |
| 17:44.25 | Slainte | Groo: Plus you can set it up so if your VOIP provider goes down, you can make all your LD calls on your PSTN line |
| 17:45.00 | bjohnson | Groo: the second line is vonage? |
| 17:45.02 | Slainte | I personally dont think phone services should be allowed to be sold unless there is a 911 gateway |
| 17:45.07 | Groo | ok, great, so I guess I need to buy one of the those digium cards |
| 17:45.15 | bjohnson | Groo: you're gonna need 2 |
| 17:45.47 | Groo | ya second line is vonage |
| 17:45.51 | twisted[work] | grr |
| 17:45.54 | Groo | what cards in particlar? |
| 17:46.04 | bjohnson | Groo: the vonage hardware is locked so it talks directly to vonage .. won't talk to *. You need a fxo for the pstn and a fxo for the vonage adapter |
| 17:46.24 | Slainte | and an fxs for the phone |
| 17:46.31 | bjohnson | you also need a fxs for the cordless |
| 17:46.33 | bjohnson | :) |
| 17:47.07 | *** part/#asterisk tekjacob (~tekjacob@c2.efb7d1.client.atlantech.net) |
| 17:47.08 | Groo | ok, sounds like a good start |
| 17:47.11 | bjohnson | you could try using a fxo for the pstn and forwarding over internet to vonage who then sends back to your vonage hardware .. not sure if that's possible though |
| 17:47.12 | Slainte | So get a developmint kit and an extra FXO port |
| 17:47.54 | bjohnson | Groo: search the wiki and the mailing list for possible hits on sending from * directly to vonage |
| 17:48.19 | bjohnson | you maybe just need 1 fxo .. but then double the internet traffic |
| 17:49.19 | Groo | great, I just wanted to make sure that what I wanted to do was possible before I started searching the mailing list and wiki |
| 17:49.34 | Groo | I wish that vonage had an input port for a pots line |
| 17:49.38 | Slainte | Groo, it is a good start for you to learn about * |
| 17:50.04 | Groo | the old voip carrier I was using had a port so if I dialed # it'd switch over |
| 17:50.24 | *** join/#asterisk Hmmhesays (~Hmmmhesay@66.173.103.108) |
| 17:51.15 | Hmmhesays | is there anyway I can continue on to another priority after using hangup in the dialplan? |
| 17:51.44 | Slainte | I need to start IAX peering my LD traffic. Can someone recommend some good providers. I am based in bermuda, and the best rate I can get is 22cents a minute. |
| 17:52.14 | johnnyb | Can an asterisk conversion between two different codecs introduce echo? Further tests reveal that my budgetone only echos when calling a FireFly phone, but not when calling an outside line. |
| 17:52.57 | johnnyb | The firefly is only set up for GSM, and I think the Budgetone using G.711U |
| 17:53.47 | Slainte | johnnyb do you have any echo cancel in your zapata.conf? |
| 17:54.12 | *** join/#asterisk expousr (~expousr@66.101.10.149) |
| 17:54.16 | expousr | afternoon al. |
| 17:54.18 | expousr | all. |
| 17:54.19 | Slainte | Hmmh, thats a good question. Try using softhangup |
| 17:54.31 | Slainte | I dont knwo if it will work. Just create a NoOp |
| 17:54.37 | Slainte | and watch the CLI |
| 17:54.50 | Hmmhesays | yeah, i was looking at that, i think I will give it a shot |
| 17:54.58 | WildPikachu[BAR] | ok... if i add an extension _0 saying Dial(ZAP/g2/${EXTEN}) i get error SIP/2.0 484 Address Incomplete |
| 17:55.11 | WildPikachu[BAR] | if i dial say... 08244445555 |
| 17:55.22 | bjohnson | Slainte: how much traffic? |
| 17:55.57 | Slainte | bj, it depends on my customers. Some will be about 500 minutes others will be 4000 minutes a month |
| 17:55.58 | bjohnson | find a friend in N.A. and route your calls through their * to use NA to NA pricing |
| 17:56.28 | Slainte | cant do that. I need a proper company that does this, for legal and accountability reasons |
| 17:57.01 | l00py | what providers are good in canada? |
| 17:57.10 | l00py | can anyone suggest something? |
| 17:57.37 | Slainte | I live in bermuda and am replacing systems for Suisse banks etc |
| 17:57.42 | bjohnson | Slainte: www.lookieloo.net .. don't know anything about them .. just was looking at them. NA to Bermuda is 0.0899 |
| 17:58.00 | bjohnson | based out of British Columbia, Canada |
| 17:58.18 | bjohnson | don't know if reverse princing would be the same |
| 17:58.25 | Slainte | one sec |
| 17:58.38 | *** part/#asterisk sudoer (~sudoer@65.75.148.190) |
| 17:58.58 | bjohnson | l00py: for outgoing or for DIDs? |
| 17:59.05 | WildPikachu[BAR] | if you use _0 will EXTEN include 0 ? |
| 17:59.10 | expousr | Anyone know what the state of SS7 on * is? |
| 17:59.20 | WildPikachu[BAR] | in other words is the 0 chopped off or left? |
| 17:59.32 | bjohnson | left |
| 17:59.35 | Hmmhesays | good call Slainte |
| 17:59.38 | bjohnson | unless you chop it off |
| 18:01.59 | WildPikachu[BAR] | ok... must be my ZAP/g2 |
| 18:02.43 | *** join/#asterisk citats (~james@duff.gnuinter.net) |
| 18:03.04 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 18:03.33 | ctooley | Ok, so now I'm confused... again... it was a short trip |
| 18:03.37 | bjohnson | ${EXTEN:1} chops the first digit off an extension |
| 18:03.44 | ctooley | Asterisk seems to be eating the # |
| 18:04.23 | Slainte | ctooley what type of phone is it |
| 18:05.11 | ctooley | Slainte, uhhh it's cell <----> IAX Termination Provider <----> Asterisk |
| 18:05.19 | *** join/#asterisk thieumS (~darkmind@nanterre-7-82-229-210-142.fbx.proxad.net) |
| 18:06.11 | Slainte | when you say * is eating the # do you even see the # comming in? |
| 18:06.18 | ctooley | Slainte, nope |
| 18:06.30 | Hmmhesays | hmmm, that doesn't continue the priority if the end user hangs up |
| 18:06.48 | Hmmhesays | I need to continue on to the next priority after the user hangs up |
| 18:07.28 | Slainte | ctooley, is it at a menu it is not seeing it or in mid call? |
| 18:08.17 | Slainte | for example, my polycom phones # work fine in voicemail, but does not work in mid call for call parking |
| 18:08.46 | *** join/#asterisk Sc00t3r (~smc@smc.dsl.istop.com) |
| 18:09.38 | Beirdo | Heya, Sc00t3r |
| 18:09.47 | Sc00t3r | Hey Beirdo :) |
| 18:09.56 | ctooley | Slainte, it's in an AGI |
| 18:10.11 | Beirdo | bjohnson: another canuck... |
| 18:10.24 | Sc00t3r | That's right, eh |
| 18:11.56 | Sc00t3r | This is my first time dropping by the channel. Been playing with * at home and loving it so far. |
| 18:13.47 | Slainte | ctooley, are all other characters comming through? |
| 18:13.56 | ctooley | Slainte, yes |
| 18:14.03 | *** join/#asterisk pointer-gaim (~pointer@router.cathey.us) |
| 18:17.40 | Hmmhesays | hmmm is there any way for the dial plan to continue on the priority list after a user calling in hangs up the call? |
| 18:17.49 | Hmmhesays | softhangup didn't do it |
| 18:17.51 | *** join/#asterisk Gronker (~Gronker2@adsl-220-89-99.ags.bellsouth.net) |
| 18:18.07 | jets | What's the easiest way to stream the actively written files in /var/spool/asterisk/monitor/* If you start playing them from a browser and it's an active call -- it downloads it -- but doesn't keep streaming it i suppose |
| 18:18.24 | bjohnson | Hmmhesays: t I think |
| 18:18.39 | bjohnson | extension t |
| 18:18.40 | *** join/#asterisk Mneumonic (Mnemonic@ool-18ba58b4.dyn.optonline.net) |
| 18:18.56 | anthm | besides exten 'h' ? |
| 18:19.08 | Hmmhesays | extension t is for time outs |
| 18:19.09 | bjohnson | err .. yeah .. 'h' |
| 18:20.16 | *** join/#asterisk MrClean (~seabrook@store-fw.porchlight.ca) |
| 18:20.26 | Hmmhesays | exten => h,1,goto(s,20) exten => s,20,wait(5) it won't go onto exten => s21 |
| 18:20.34 | eye69 | jets: You'll probably have to have a system that streams it to the browser. |
| 18:20.46 | Hmmhesays | exten => s,21 |
| 18:20.51 | anthm | you cahn t do that |
| 18:20.52 | eye69 | jets: You could check if icecast could help. |
| 18:21.07 | anthm | you have the wrong idea about goto |
| 18:21.19 | Hmmhesays | ? |
| 18:21.37 | Hmmhesays | goto(extension, priority) |
| 18:21.48 | `Sauron | goto(context,extension,priority) |
| 18:21.52 | anthm | all goto does is set the channel's next indended destination pointers |
| 18:21.59 | anthm | if you are hungup its moot |
| 18:22.16 | Hmmhesays | yeah, so I can't set a goto with h |
| 18:22.40 | MrClean | Anyone have any idea why GotoIfTime() would be consistently off by 2 minutes? For example, given GotoIfTime(8:00-23:00|*|*|*?open,s,1), incoming calls won't be sent to open,s,1 until the system time is >= 23:02. |
| 18:22.41 | Hmmhesays | Sauron, you can omit the context if you are going to the same context |
| 18:22.47 | `Sauron | Ah. |
| 18:22.56 | `Sauron | I've never gone to the same context. :) |
| 18:23.00 | anthm | you can do all your business there in h tho |
| 18:23.05 | bjohnson | Beirdo: greg_work: from FWTdirect .. We do have toll free numbers that work in Canada. But they are on a huge backorder. It may take as long as 21 days to hook them up. 3c/min USD |
| 18:23.12 | *** join/#asterisk sergiovel (sergiovel@194-167-126-200.fibertel.com.ar) |
| 18:23.21 | Beirdo | that's not too shabby |
| 18:23.34 | Beirdo | although the waiting list sucks |
| 18:23.53 | bjohnson | shit .. you've been waffling over this for at least that long |
| 18:23.59 | eye69 | MrClean: Is it the same in the other end? |
| 18:24.06 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 18:24.06 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 18:24.10 | eye69 | MrClean: As in 08.02? |
| 18:24.12 | bjohnson | probably a small monthly fee too |
| 18:24.21 | Beirdo | hehe |
| 18:24.37 | Beirdo | I don't even have asterisk setup completely yet. :) |
| 18:24.45 | Beirdo | that's weekend work |
| 18:25.11 | MrClean | You mean for the caller? Hmm, the time is a little fast for the calling computer, actually. Wouldn't it still go by the system time on the asterisk box? |
| 18:26.00 | *** join/#asterisk opticalcarrier (~bo@optical.neteng.earthlink.net) |
| 18:26.02 | expousr | anyone doing any ss7->* gatewaying? looking for hardware recommendations. |
| 18:26.25 | eye69 | MrClean: I actually meant if it doesn't stop sending to open,s,1 until 08.02 |
| 18:26.39 | opticalcarrier | is there anyway to * to periodically send out NOTIFY if a mailbox has messages? |
| 18:26.51 | MrClean | Oh. Good question. I'll simulate that now... |
| 18:26.51 | opticalcarrier | instead of just once |
| 18:26.53 | echion | expousr, I haven't tried it but I might... http://www.sangoma.com/products/p_csu_cards.htm |
| 18:27.26 | eye69 | MrClean: I guess you NTP sync the time on the server? |
| 18:27.57 | MrClean | No, I don't think we have any synching going on on that server. |
| 18:28.27 | eye69 | MrClean: You really should. In fact, servers should always be NTP synced. |
| 18:28.34 | Beirdo | amen |
| 18:28.48 | MrClean | Oh. It's possible that I'm ignorant. I'm not actually the admin for that box. |
| 18:28.49 | eye69 | It's so easy to do there's no reason to not do it. |
| 18:29.13 | eye69 | It's quite easy to setup a local ntpd too that all the other servers can use. |
| 18:30.30 | expousr | agreed! *all* servers should be ntp synced. Even our phones are ntp synced. |
| 18:30.46 | Nugget | yup. |
| 18:31.13 | Qwell | even my pots phone is ntp synced |
| 18:31.24 | MrClean | Hmm, it works correctly for the other end of the range. Given, GotoIfTime(13:30-23:00|*|*|*?open,s,1), it sends me to open,s,1 at >= 13:30. |
| 18:31.28 | eye69 | My mobile phone is NTP synced. :) |
| 18:32.20 | MrClean | It sounds like that box is actually ntp synched. |
| 18:32.21 | eye69 | Ok, that is a bit weird. |
| 18:32.27 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 18:33.03 | *** part/#asterisk gbs (~GBS@adsl-5063187d.monradsl.monornet.hu) |
| 18:34.15 | expousr | echion thanks - I've seen those, curious as to if anyone's actually using 'em. :) |
| 18:35.17 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
| 18:35.26 | MrClean | That time it was only off by one minute though. Does the function round up by one minute (ie. 08:00 to 22:59 is fair game until >= 23:00)? |
| 18:36.38 | Essobi | What kernel should I use for a 1.0 build? |
| 18:36.48 | Essobi | 2.6.X is fairly safe still right? |
| 18:37.07 | bjohnson | Beirdo: FTWdirect Random Toll-Free Rates US50/Canada: $2.49/month, and 3¢/min. |
| 18:37.46 | Beirdo | in USD? |
| 18:39.27 | bjohnson | y |
| 18:39.57 | bjohnson | yes |
| 18:41.20 | Beirdo | OK, just wanted to make sure |
| 18:41.24 | Beirdo | that's not all that bad |
| 18:41.49 | Beirdo | especially with the USD tanked |
| 18:44.48 | terrapen | i want to find some offshore hosting |
| 18:44.54 | terrapen | in a place that does not have copyright laws |
| 18:44.59 | terrapen | and start a massive bittorrent site |
| 18:45.02 | terrapen | and make a fucking mint. |
| 18:45.44 | Beirdo | wonder what the registry rules are in the Cook Islands? |
| 18:46.01 | terrapen | well, it has to be somewhere w/ good bandwidth |
| 18:46.13 | Beirdo | but then you wouldn't get .co.ck |
| 18:46.17 | terrapen | heh |
| 18:46.44 | Beirdo | I couldn't believe it when someone showed me www.google.co.ck |
| 18:46.46 | terrapen | i need a lawless place that is in close proximity to a lawless place |
| 18:46.49 | mikegrb | Beirdo! |
| 18:46.59 | Beirdo | heya, mikegrb |
| 18:47.12 | mikegrb | :D |
| 18:47.17 | terrapen | i wonder if i could set up a Somali data center |
| 18:47.25 | terrapen | and get bandwidth from kenya |
| 18:47.28 | Beirdo | bad idea, it would get bomed |
| 18:47.49 | terrapen | i wouldn't be the least bit surprised if the MPAA hired someone to bomb it |
| 18:47.59 | terrapen | Somalia is, for all purposes, lawless |
| 18:48.13 | Beirdo | yeah |
| 18:48.18 | terrapen | cayman islands maybe? |
| 18:48.32 | *** join/#asterisk gr0mit_home (~wendolene@router1.txrx.org.uk) |
| 18:48.36 | Beirdo | silly boy |
| 18:51.08 | terrapen | i wonder if such a place exists |
| 18:51.11 | terrapen | Denmark maybe? |
| 18:51.32 | Beirdo | heh |
| 18:51.51 | Beirdo | if you think Denmark has lax copyright laws, I think you aren't paying attention :) |
| 18:52.00 | terrapen | i thot it did at one time |
| 18:52.10 | shido6 | can someone tell me what is wrong with http://pastebin.ca/5632 |
| 18:52.18 | Beirdo | Europe is pretty strong on copyright laws last I heard |
| 18:52.21 | terrapen | surely some place in Europe has lax laws |
| 18:52.23 | sivana | http://pastebin.ca/5632 |
| 18:52.54 | terrapen | ugh pastebin.ca is always slow |
| 18:52.58 | terrapen | why not use pastebin.com? |
| 18:53.08 | shido6 | no I mean the info in it |
| 18:53.14 | shido6 | my zaptel fxo is not working |
| 18:53.16 | terrapen | i can't read that info |
| 18:53.17 | shido6 | asterisk wont use it |
| 18:53.20 | shido6 | cant see it |
| 18:53.35 | terrapen | try moving PCI slots |
| 18:53.45 | shido6 | did that |
| 18:53.48 | shido6 | once already |
| 18:53.56 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 18:54.00 | zoa | helloooo |
| 18:54.09 | Beirdo | McFly? |
| 18:54.25 | sivana | shido6: how many devices do you have in the box? |
| 18:55.04 | shido6 | http://pastebin.ca/5633 |
| 18:55.06 | shido6 | there |
| 18:55.06 | *** join/#asterisk adnans (~adnans@linuxgoeroe.demon.nl) |
| 18:56.19 | Hmmhesays | hmm it seems I cannot do this without agi |
| 18:56.21 | Hmmhesays | damnit. |
| 18:56.52 | redder86 | Hmmhesays: you know, I find that I hate to have to do things in AGI, but after I do, I'm always glad that I did because I have better control of it. |
| 18:57.17 | Hmmhesays | I just want simple as possible. user calls in, inputs callback number, hangs up |
| 18:57.24 | Hmmhesays | asterisk waits 5 seconds and calls it back |
| 18:57.49 | Hmmhesays | but when the user hangs up, that destroys the channel, and I can't figure out how to spawn a new one |
| 18:58.23 | *** part/#asterisk opticalcarrier (~bo@optical.neteng.earthlink.net) |
| 19:01.00 | *** join/#asterisk Mother__ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 19:01.47 | Mother__ | hi all |
| 19:01.59 | Mother__ | anyone here upgraded a Cisco 7912 to SIP? |
| 19:03.07 | sezuan | How can I convice asterisk to transmit the real caller name and not to replace it with 'asterisk'? |
| 19:03.48 | denon | sezuan: show application SetCIDName |
| 19:04.19 | *** join/#asterisk delta407 (~wglynn@65.124.195.162) |
| 19:04.56 | delta407 | I'd like some help with a dialplan-related issue I'm having |
| 19:05.28 | delta407 | specifically: I want to place outbound calls over DUNDi when possible, but use local trunks when not |
| 19:06.12 | *** join/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 19:06.27 | delta407 | this works fine most of the time -- however, when someone advertises a route, I can't figure out how to fail back to the local trunks when the route is circuit-busy |
| 19:06.45 | *** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
| 19:06.54 | sezuan | denon: hmm.. I now that option. But I'm not sure if helps me. When I receive a call throu a SIP/PSTN gateway I the the caller number. When my receives a call directly, I see the caller name in the display. But if I receive a call from another sip client throu asterisk, I see 'asterisk' |
| 19:07.23 | denon | you can set it either way - in sip.conf or with that app |
| 19:07.29 | Firestrm | bjohnson, are you around? i need some spa-3000 help, you seem to know alot about them |
| 19:07.44 | paulc | Firestrm: I can help if it's a quickie |
| 19:07.55 | Firestrm | paulc, new toy, wont register |
| 19:08.09 | denon | sezuan: put a callerid="Foo" <55> in your sip.conf for the calling user |
| 19:08.11 | Firestrm | i have the proxy, username and secret set correctly. |
| 19:08.36 | denon | gotta run, cya |
| 19:09.09 | *** join/#asterisk jgrant (~jgrant@office1.visionpointsystems.com) |
| 19:09.10 | paulc | do you see failed registration attempts on the asterisk console? |
| 19:09.17 | sezuan | denon: the user is not registered at my asterisk. |
| 19:09.56 | denon | sezuan: you want to change the incoming callerid of just that person? |
| 19:10.04 | denon | or you just dont want the default to be "asterisk"? |
| 19:10.18 | denon | if default, you can do something like this in sip.conf |
| 19:10.21 | denon | [general] |
| 19:10.28 | Firestrm | paulc, ya i get 403 forbiddon.. the relivent portions of my sip.conf are here http://pastebin.ca/5634 |
| 19:10.28 | denon | callerid=Unknown |
| 19:10.47 | denon | and that'll replace "asterisk" |
| 19:10.58 | *** join/#asterisk Uajal (~icechat5@ool-182e86f3.dyn.optonline.net) |
| 19:11.01 | sezuan | denon: no, I want to see the callerid of the person that calls me, but I only see 'asterisk' |
| 19:11.40 | WildPikachu[BAR] | hrmmmmm |
| 19:11.59 | denon | sezuan: if they arent registered in your box, you cant force them to give it to you .. |
| 19:12.04 | Uajal | Hi, did anybodyworked with Broadvoice. They say some patch should be used and then asterisk should be recompiled. Is it true? |
| 19:12.05 | denon | you can set it to whatever you want, using the app I told you about |
| 19:12.08 | sezuan | denon: Wenn the calls passes my asterisk box, the callerid disappears. |
| 19:12.09 | denon | or you can set the default in the config |
| 19:12.12 | terrapen | does anybody here have Speakeasy DSL? |
| 19:12.12 | WildPikachu[BAR] | i setup a context for dialing out from my sip phone, i setup _0 to Dial(ZAP/g1/${EXTEN}), but i get this message... SIP/2.0 484 Address Incomplete |
| 19:12.15 | WildPikachu[BAR] | any idea why? |
| 19:12.21 | denon | I gotta go |
| 19:12.21 | denon | cya |
| 19:12.46 | paulc | Firestrm: What's the user set to in SPA3000? It should be 2002 (not whatever you've got username set to) |
| 19:12.53 | bjohnson | Firestrm: sip.conf looks ok. is for the pstn or the line1? |
| 19:13.00 | sezuan | denon: The calls arrive with 'From: "Caller Name" caller.name@domain and leaves as From: "Caller Name" asterisk@domain |
| 19:13.24 | sezuan | denon: When I receive the call directly, I see caller.name in the display. |
| 19:13.38 | Firestrm | bjohnson, i have pstn disabled, i dont even want to play with that untill i get line1 working |
| 19:13.52 | paulc | Firestrm: I gotta head out for lunch, got an appointment, I'll leave you in the capable hands of Dr B Johnson :-) |
| 19:14.19 | Firestrm | paulc, you might be on to something.. i have username set to sipura1 |
| 19:14.51 | bjohnson | Firestrm: for most people .. there's like 3 things you need to change in the Line1 tab |
| 19:15.01 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 19:15.03 | shido6 | because it is |
| 19:15.06 | shido6 | incomplete WildPikachu |
| 19:15.13 | shido6 | look at your exten line |
| 19:15.16 | shido6 | no wildcards |
| 19:15.22 | shido6 | nuffin |
| 19:15.22 | shido6 | there |
| 19:15.29 | shido6 | show me the full exten line |
| 19:15.33 | WildPikachu[BAR] | hrmm..... |
| 19:15.41 | bjohnson | Firestrm: Display Name, User ID, and Password: |
| 19:15.48 | WildPikachu[BAR] | exten => _0,1,Dial(${PSTN}/${EXTEN}) |
| 19:15.56 | shido6 | err |
| 19:15.59 | shido6 | so when u press 0 |
| 19:16.06 | shido6 | u want it go out the zap interfaces |
| 19:16.14 | WildPikachu[BAR] | i dial in sip... sip:0823333723@asteriskip.whateva |
| 19:16.18 | bjohnson | Firestrm: in sip.conf I have the device section and the username set to the same |
| 19:16.26 | WildPikachu[BAR] | ywp |
| 19:16.32 | WildPikachu[BAR] | yep, zap interface is group=1 |
| 19:16.39 | WildPikachu[BAR] | PSTN=ZAP/g1 |
| 19:16.39 | bjohnson | Firestrm: make sure no 2 devices use the same secret of * gets confused |
| 19:16.50 | Firestrm | bjohnson, displayname=sipura1, username =sipura1, password=password |
| 19:17.19 | Strom_C | good morning channel |
| 19:17.19 | Firestrm | bjohnson, ahhh maybe thats it.. i will change the secret |
| 19:17.30 | bjohnson | Firestrm: sip show peers .. does it show the ip |
| 19:18.38 | WildPikachu[BAR] | shido6, am i doing summin wrong? :) |
| 19:18.42 | shido6 | yes |
| 19:18.48 | shido6 | how do you want do dial |
| 19:19.33 | Uajal | Excuse me for repeating: Should asterisk be recompiled with some patch for broadvoice (broadvoice is traffic terminator)? |
| 19:19.46 | WildPikachu[BAR] | well.... dial 0 then 0823333723 |
| 19:20.01 | WildPikachu[BAR] | i'm taking this setup step by step, and dialing out is my next step :) |
| 19:20.16 | shido6 | then u need to do |
| 19:20.25 | Firestrm | bjohnson, 2002/sipura1 (Unspecified) D 255.255.255.255 0 Unmonitored |
| 19:20.29 | shido6 | exten => _0N.,1,Dial(${PSTN}/${EXTEN}) |
| 19:20.46 | WildPikachu[BAR] | AHA |
| 19:20.53 | WildPikachu[BAR] | what does the N do? number? |
| 19:21.07 | *** join/#asterisk Bonbon (~bonbon@83.146.53.34) |
| 19:21.30 | Strom_C | N == [2-9] |
| 19:21.46 | bjohnson | Firestrm: oops .. I forgot to mention Proxy (and make sure use outbound proxy is no) |
| 19:21.57 | WildPikachu[BAR] | hrmm... but i need to dial 0 :( |
| 19:22.14 | Uajal | RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v |
| 19:22.14 | Uajal | retrieving revision 1.564 |
| 19:22.14 | Uajal | diff -u -r1.564 chan_sip.c |
| 19:22.42 | Uajal | Does anybody know about the following cvs update (before recompiling asterisk for broadvoice)? |
| 19:23.13 | bjohnson | supposed to be in current stable |
| 19:23.24 | WildPikachu[BAR] | yeehaa! |
| 19:23.38 | WildPikachu[BAR] | thanks shido6 |
| 19:23.55 | WildPikachu[BAR] | if i have _0.,1 ..... it works |
| 19:24.01 | WildPikachu[BAR] | aha |
| 19:24.10 | sezuan | denon: Ok, when I use: exten => marieber,1,SetCallerID,454545 Asterisk will send: From: "454545" <sip:454545@domain..> |
| 19:24.33 | WildPikachu[BAR] | wait... it works with N aswell..... i now understand that it doesn't operate like a conventional PBX, you don't have to dial 0 first you just dial the number |
| 19:24.41 | Firestrm | bjohnson, that was the magical switch.. Registered SIP '2002' at 192.168.2.10 port 5060 expires 180 |
| 19:24.43 | sezuan | denon: but when i use: exten => marieber,1,SetCallerID,"Foobar" Asterisk will send: From: "Foobar" <sip:asterisk@domain..> |
| 19:24.50 | Firestrm | bjohnson, thanks :) |
| 19:24.53 | WildPikachu[BAR] | what does the . mean in _0N. ? |
| 19:24.59 | Strom_C | . == |
| 19:25.00 | Strom_C | rt |
| 19:25.01 | Strom_C | er |
| 19:25.03 | Strom_C | . == |
| 19:25.03 | _Brian | I was curious if there is a way to block a 302 Moved temp message on a per phone basis? If I setup a DID Number to receive the call, and then have remote call forwarding (RCF) enabled, it sends back a 302 with the new extension (for outbound termination). The original extension that sent back the 302 appears lost..... |
| 19:25.05 | Uajal | bjohnson: Does it mean that I shouldn't recompile asterisk if I will install it tomorrow? |
| 19:25.08 | WildPikachu[BAR] | rt? |
| 19:25.12 | Strom_C | . == "match anything" |
| 19:25.12 | Delvar | nn all |
| 19:25.16 | WildPikachu[BAR] | aha |
| 19:25.29 | Strom_C | WildPikachu, i just woke up. accuracy at hitting keys is not at full speed yet |
| 19:25.30 | WildPikachu[BAR] | cool man.... i'll go fiddle some more and come back if i get stuck for a few hours :-) |
| 19:25.30 | Strom_C | :) |
| 19:25.44 | WildPikachu[BAR] | Strom_C, no prob bud.... *hands u some nice filter coffee* |
| 19:26.24 | Strom_C | blech, coffee...i prefer a nice tall glass of milk to get the day started :) |
| 19:26.37 | _Brian | I would like to block this on the outbound call like by calling a AGI for each call passed, but I am unable to find a comman link between the inbound call leg and the outbound call leg when a 302 is involved |
| 19:26.40 | Beirdo | milk? interesting choice |
| 19:27.04 | shido6 | bbl |
| 19:28.17 | bjohnson | Uajal: I don't really know. i don't use Broadvoice yet |
| 19:28.23 | _Brian | I guess that would be a no :) |
| 19:29.05 | bjohnson | WildPikachu: read the extensions.conf pages on the wiki. _ means pattern match . means any digit or digits |
| 19:29.26 | bjohnson | _0. means 0 or anything that starts with 0 |
| 19:30.19 | Qwell | Where can one get a list of vars that are set? |
| 19:30.31 | bjohnson | try to avoid large patterns matches like that or _. or _X. becuase they tend to catch things even when you don't them to |
| 19:30.42 | junky[work] | <PROTECTED> |
| 19:30.42 | junky[work] | <PROTECTED> |
| 19:30.42 | junky[work] | <PROTECTED> |
| 19:30.48 | bjohnson | Qwell: there's a readme in the source |
| 19:30.53 | junky[work] | why isnt reading the damn file in fr/? |
| 19:31.11 | Qwell | bjohnson: Any idea of the path to it? |
| 19:31.22 | bjohnson | do you have french gsm files? |
| 19:31.36 | bjohnson | Qwell: no .. /usr/src/asterisk/ ? |
| 19:31.55 | junky[work] | bjohnson: ya, /var/lib/asterisk/sounds/fr/user-pass.gsm |
| 19:32.08 | bjohnson | check file permissions? |
| 19:32.35 | _Brian | i guess i will try again later..thanks guys |
| 19:32.41 | bjohnson | ok |
| 19:32.48 | junky[work] | same shit. |
| 19:33.06 | junky[work] | DAMN |
| 19:33.07 | junky[work] | my bad. |
| 19:33.25 | *** join/#asterisk DARP (~diegoramo@65.167.56.182) |
| 19:33.48 | DARP | hi |
| 19:33.58 | DARP | need help with two issues |
| 19:34.39 | *** join/#asterisk ionix (~ioni@66.38.219.151) |
| 19:34.39 | ionix | Hey |
| 19:34.41 | DARP | one somebody has configurated a weeltech FXO as sip client to manage PSTN lines |
| 19:34.46 | Qwell | bjohnson: looks like its README.variables(who would have thought) |
| 19:34.49 | ionix | anyone knows how I can get reverse lookup database or service ? |
| 19:34.57 | Uajal | which cheap traffic terminator would anybody recommend to use with asterisk? |
| 19:34.57 | ionix | i.e I would like to resolve ANI into name |
| 19:34.59 | Mother__ | I have the SIP image for the 7905, will this work straight with the 7912 too, or is there a specific image for the 7912? |
| 19:36.52 | ionix | Uajal: nufone.net or sixtel |
| 19:38.31 | DARP | somebody has configurate an external FXO gateway with * |
| 19:38.33 | DARP | ???? |
| 19:38.47 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 19:42.41 | DARP | hey nobody can help me??????????? |
| 19:42.53 | ionix | what you think ?????????????? |
| 19:43.01 | ionix | explain your problem rather than poking people |
| 19:43.13 | DARP | ok |
| 19:43.28 | DARP | i have a welltech 6 FXO |
| 19:43.44 | Qwell | Is it supported by *? |
| 19:43.46 | DARP | i configured each channel as sip |
| 19:43.56 | DARP | i dont know really |
| 19:44.08 | Qwell | ignore me, continue |
| 19:44.36 | DARP | when i call to a channel the gateway give me tone |
| 19:44.59 | DARP | but it doesn't receive the keys to call output |
| 19:47.48 | sezuan | denon: OK, I guess I got it. ast_callerid_parse removes non phonenumber locations. |
| 19:49.27 | bjohnson | DARP: keys to call output/ |
| 19:49.29 | bjohnson | ? |
| 19:49.45 | _Brian | I was curious if there is a way to block a 302 Moved temp message on a per phone basis? If I setup a DID Number to receive the call, and then have remote call forwarding (RCF) enabled, it sends back a 302 with the new extension (for outbound termination). The original extension that sent back the 302 appears lost..... |
| 19:49.50 | bjohnson | can;t you just send it to the FXO sip device as part of the dial string? |
| 19:50.08 | Firestrm | what is the dial format for dialing two extensions at once, is it DIAL(SIP/1001 & SIP/1002) ? |
| 19:50.16 | bjohnson | Firestrm: yes |
| 19:50.32 | Firestrm | hmm i doesnt work when i try that.. goes to voicemail |
| 19:51.09 | bjohnson | set verbose 3 and watch the messages .. look for errors from 1001 and 1002 |
| 19:51.25 | bjohnson | Firestrm: 1001 and 1002 show up wit sip show peers? |
| 19:51.38 | *** join/#asterisk clord (~clint@166.70.78.242) |
| 19:52.49 | ionix | anyone knows how I can get reverse lookup database or service ? |
| 19:52.57 | ionix | automatically for asterisk |
| 19:53.03 | ionix | I get the ANI but I would like to provide name |
| 19:53.33 | DARP | ok when i heard the tone i try to press the numbers to try out a call |
| 19:54.27 | ionix | does asterisk send your number to an other proxy or a FXO interface ? |
| 19:54.44 | bjohnson | tone on a voip phone or a fxs phone? |
| 19:55.22 | anti | ugh, the default music on hold mp3s sound aweful.. |
| 19:55.45 | bjohnson | did anyone get mplayer of a stream working for moh |
| 19:55.47 | bjohnson | ? |
| 19:57.07 | Firestrm | bjohnson, i can call 2002 from 2001 but i cant call 2001 from 2002, says somthing about cant find path from slin to g729, even though i have bought a license. when it picks up an incoming call, extension 1001 goes though, but not extension 1002.. i have the relivant parts of my extensions.conf and the debug info at http://pastebin.ca/5636 |
| 19:57.58 | bjohnson | do both 2001 and 2002 support g729? |
| 19:58.03 | Firestrm | yes |
| 19:58.09 | bjohnson | wouldn't that require 2 licences on *? |
| 19:58.21 | bjohnson | one for each connection |
| 19:58.42 | Firestrm | hmm.. ok well that explains the extension thing.. |
| 19:58.59 | bjohnson | if for a lan .. most people use ulaw (g711u) |
| 19:59.10 | Firestrm | i dont know why its not ringing the phone on incoming call though |
| 19:59.42 | bjohnson | you have an extensions set up for both 1001 and 1002? |
| 20:00.07 | Firestrm | yes |
| 20:00.32 | DARP | i call from a sip softphone to an xxx extension there asterisk call a sip channel that is the line in the FXO gateway |
| 20:00.33 | Firestrm | they are identical, except in 1002 im ringing both phones |
| 20:00.48 | *** part/#asterisk FuzzyCat (~ScaredyCa@84.119.133.131) |
| 20:01.25 | jarrod | anyone familiar with 3com nbx switches |
| 20:01.41 | anti | Anyone have any idea when resampling to 8000hz using mpg123, the audio is horrible? hehe |
| 20:01.44 | jarrod | im wondering if those things support standard sip |
| 20:02.03 | bjohnson | Firestrm: try without spaces before and aft & |
| 20:04.19 | *** join/#asterisk jayden (~ircatjerr@65.170.43.34) |
| 20:04.21 | bjohnson | Firestrm: 2001 and 2002 show up wit sip show peers? |
| 20:04.40 | *** join/#asterisk pointer-gaim (~pointer@router.cathey.us) |
| 20:04.42 | jayden | quick pool.. what OS are most using for production systems? |
| 20:04.58 | denon | pool? |
| 20:04.58 | bjohnson | fc2 .. but I'd suggest centos |
| 20:05.02 | denon | like swimming? |
| 20:05.14 | bjohnson | many people run debian or slackware |
| 20:05.21 | denon | we run a lot of debian |
| 20:05.24 | jayden | no, pool, like mavis beacons teaches typing |
| 20:05.35 | bjohnson | mix of others, gentoo, bsd, etc |
| 20:05.36 | Firestrm | bjohnson, thats what it was.. removed spaces works like a charm.. now only one last problem... for now.. |
| 20:05.50 | jayden | y |
| 20:06.00 | denon | bjohnson: we run a lot of fbsd too, but not for * |
| 20:06.06 | bjohnson | denon: maybe "betting pool" |
| 20:06.45 | Firestrm | bjohnson, * doesnt seem to detect immediatly when incoming caller has hung up.. it hangs on for 15 sec or so before giving up |
| 20:07.31 | jayden | and everyone recomends installing X on an asterisk box right |
| 20:07.46 | shido6 | well |
| 20:07.51 | jayden | :) |
| 20:07.51 | shido6 | if its a zap fxo |
| 20:08.00 | shido6 | the t1 and e1 cards hangup quickly |
| 20:08.20 | jayden | sorry, sometimes my hands just type things... like in that movie... |
| 20:08.27 | shido6 | I hae X on my box :) but I dont recommend it for the newb's |
| 20:08.28 | bjohnson | Firestrm: I haven't noticed that. Look for silence detection settings in sip.conf and spa config screens |
| 20:08.30 | *** join/#asterisk mountainm2k (~freenodei@cbit-98.bullseye9.com) |
| 20:09.07 | Firestrm | bjohnson, i am not using spa for fxo yet.. my head is still spinning after getting spa-3000 fxs working |
| 20:09.38 | Firestrm | bjohnson, but i will check the wiki on silence detection |
| 20:10.19 | bjohnson | fxo is harder |
| 20:10.44 | bjohnson | info on the wiki under spa 3000 |
| 20:11.17 | *** join/#asterisk brycec (~brycec@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 20:12.19 | Firestrm | bjohnson, ya i came across it.. im going to wait until i fully intergrate the data into my neurons |
| 20:12.35 | *** join/#asterisk omarc55 (~omarc55@adsl-10-107-192.mia.bellsouth.net) |
| 20:12.37 | *** join/#asterisk Samoied (~samoied@200.247.141.111) |
| 20:13.13 | Samoied | How I view current logged sip users? |
| 20:13.22 | Hmmhesays | sip show peers |
| 20:13.26 | Hmmhesays | and RTFM |
| 20:13.39 | Samoied | Hmmhesays: This show all users |
| 20:13.49 | Samoied | not logged |
| 20:14.11 | Hmmhesays | are your users dynamic? |
| 20:14.27 | Hmmhesays | if so it should show when they are logged in with an IP |
| 20:14.30 | Samoied | Hmmhesays: yes |
| 20:14.53 | Hmmhesays | otherwise it says "unspecified" |
| 20:16.05 | wwalker | How do I turn off "# transfer" mode? I need users to be able to hit # without it being grabbed by the * server. |
| 20:16.17 | Hmmhesays | it doesn't do that by default |
| 20:16.36 | wwalker | I've found references to patches, but nothing current |
| 20:16.37 | hardwire | heya |
| 20:16.39 | hardwire | folkey dokes |
| 20:16.43 | Hmmhesays | you must have Tor t in your exten |
| 20:18.07 | Uajal | any live examples of calling Asterisk commands from perl? |
| 20:19.05 | anti | Anyone have any ideas why my music on hold is very crappy, I've tried using various mp3s, and mpg123 and madplay, its very loud crackly and breaking.. |
| 20:19.11 | wwalker | Hmmhesays don't have T or t in the extension, but when I call GET DATA and press #, it returns an empty string, when I press * it returns *. |
| 20:19.36 | Derkommissar | People im thinking of espending 3K on an asterisk Training |
| 20:19.40 | Derkommissar | is it worth it ? |
| 20:19.43 | Firestrm | bjohnson, i dont think my hangup problem is a sip problem.. its a zap problem. zap is not detecting hangup events |
| 20:19.59 | Derkommissar | 3K plus 19 hours on a plane to go to sweeden |
| 20:20.48 | ionix | hey |
| 20:20.52 | DrPete | anyone know a voip provider that had really good UK rates? |
| 20:20.58 | ionix | when someone calls me with Asterisk, they don't hear the ring |
| 20:21.09 | ionix | from exterior I mean |
| 20:21.14 | wwalker | anti, I'm a newb, but I'd try converting the mp3's to gsm files, then there is no on the fly encoding |
| 20:21.53 | Firestrm | anti, are you using mpg123 or mpg321. if your using redhat, they like to use mpg321 wiht a link to mpg123 just to make it extra confusing |
| 20:22.14 | Samoied | wwalker: Use this: |
| 20:22.41 | anti | Firestrm: nah its real mpg123, plus I've also tried using madplay, its very loud, breaks a lot. I've even tested it with my speakers it seems when I resample to 8000hz, it turns to poop. |
| 20:22.49 | Samoied | sod => custom:/usr/share/asterisk/moh/sod,/usr/bin/sox_wrapper |
| 20:22.50 | Uajal | I checked the commands available through AGI. There is channel ststus. How can I do such things like silence detection, voice-barge-in detection, answering machine detection? |
| 20:22.58 | Firestrm | anyone here have any idea why my x100p zap channel is not detecting hangup of remote party? |
| 20:23.37 | DARP | why when i use sip outside my lan i can connect with asterisk and sip clients but i cannot heard them??? |
| 20:23.57 | *** join/#asterisk Poincare (~jefffnode@dD5779B07.access.telenet.be) |
| 20:26.18 | bjohnson | Derkommissar: hotel and food included in that? |
| 20:26.20 | Connor- | anyone have a idea how to unlock a 7905? |
| 20:26.36 | Derkommissar | bjohnson, nope :-/ thats what held me back |
| 20:27.09 | bjohnson | send me .. I'll let you know if it's worth it when I get back |
| 20:27.33 | Derkommissar | is http://www.astricon.net/ part of digium ? |
| 20:27.34 | bjohnson | DARP: NAT |
| 20:27.40 | Derkommissar | You wisth |
| 20:27.45 | Derkommissar | :-P |
| 20:28.14 | Qwell | Derkommissar: Send me too, I'll keep him in line. |
| 20:28.14 | bjohnson | like hell you will |
| 20:28.14 | anthm | Get a Clue.. http://www.pbxclue.com |
| 20:28.14 | Qwell | bjohnson: Theres this great bar I know of in Sweeden :P |
| 20:28.16 | Qwell | erm, -e |
| 20:28.46 | Meznev | Derkommissar: I don't think so, doesn't even have the same contact info as the digium site. |
| 20:29.07 | bjohnson | Derkommissar: I think it is |
| 20:29.28 | Derkommissar | wierd cause mark is going to be in miami that week |
| 20:29.33 | Derkommissar | in the TMC conference |
| 20:29.42 | Derkommissar | so who is giving the training ? |
| 20:29.49 | Derkommissar | better be someone from Digium |
| 20:33.29 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 20:33.30 | ariel_ | Derkommissar, hello how are you doing? |
| 20:33.46 | Derkommissar | ariel_, |
| 20:33.49 | Derkommissar | how are you? |
| 20:33.52 | ariel_ | fine |
| 20:34.02 | Derkommissar | havent seen you around in a while |
| 20:34.16 | ariel_ | are you going to the internet telephone conference in miami? |
| 20:34.31 | ariel_ | Derkommissar, I have been here. It's you that I have not seen. |
| 20:34.55 | ariel_ | are you here in Miami or out of town? |
| 20:34.58 | Derkommissar | yes im |
| 20:35.01 | anti | hmm I think my music on hold problems have something to do with I have no zaptel devices I'm using with asterisk... |
| 20:35.27 | Derkommissar | im in miami |
| 20:35.28 | Firestrm | anyone here have X100P hardware working in canada? Im having hangup detect problems, it may be my telco.. |
| 20:35.33 | ariel_ | anti, moh does not use zap for timing only iax trunk and meetme |
| 20:36.31 | Derkommissar | ariel_, are you working ? |
| 20:36.42 | bjohnson | Astricon training .. Olle E. Johansson & Steven M. Sokol |
| 20:36.43 | ariel_ | on my own here and there |
| 20:37.06 | bjohnson | Firestrm: i do |
| 20:37.27 | bjohnson | well .. I did for while |
| 20:37.31 | bjohnson | a while |
| 20:38.28 | Firestrm | bjohnson, did you have any problems with hangup detect? |
| 20:38.38 | bjohnson | no |
| 20:38.46 | Firestrm | bjohnson, mine completely ignores handup |
| 20:39.09 | Firestrm | bjohnson, which telco? Smellus? |
| 20:39.15 | bjohnson | mines a genuine Digium (tm) X100P though |
| 20:39.32 | bjohnson | I've used it on Bell and Allstream |
| 20:40.06 | Firestrm | bjohnson, im on smellus (tellus).. maybe they are not sending hangup signal |
| 20:40.19 | Uajal | Does current stable of asterisk include call progress detection or it is only in CVS? |
| 20:40.28 | Firestrm | bjohnson, or maybe i have to get on with the FXO portion of my spa-3000 install |
| 20:40.52 | bjohnson | Uajal: sorry. don't know |
| 20:41.36 | Uajal | Did anybody work with call progress detection? |
| 20:42.14 | Uajal | I mean (answering machine detection, voice barge-in detection, silence detection) |
| 20:44.50 | *** join/#asterisk Flood_of_SYNs (~cory@wbar13.chi1-4.27.95.222.chi1.dsl-verizon.net) |
| 20:46.26 | *** join/#asterisk robf (~robf@208.188.247.3) |
| 20:50.48 | denon | any bug marshalls awake yet? |
| 20:51.49 | junky[work] | pretty busy at this time. |
| 20:55.26 | jayden | anthm, pbxclue, nice. |
| 20:56.32 | *** join/#asterisk LarsAC (~LarsAC@pD9501463.dip0.t-ipconnect.de) |
| 21:00.09 | *** join/#asterisk Inv_arp (junya@adsl-3-245-112.mia.bellsouth.net) |
| 21:02.00 | bjohnson | anyone have a script to sum usage in a cdr file? |
| 21:02.27 | LarsAC | re florz |
| 21:03.05 | *** join/#asterisk clord (~clint@166.70.78.242) |
| 21:03.38 | *** part/#asterisk SickNix (~Nix@81.213.125.220) |
| 21:04.18 | clord | Has anyone use NMS AG-T1 cards with Asterisk? |
| 21:04.41 | outtolunc | anyone notice if bkw ever posted the link to the dev-conf recording? |
| 21:05.23 | clord | or any NMS card for that matter? |
| 21:06.06 | mrempire | exten => 550,1,Dial(Modem/ttyI0/00235407616:0650506652) if i dial 550 then asterisk should connect me to the 06number? |
| 21:06.35 | file | woot my plane tickets for VON arrived today |
| 21:06.52 | outtolunc | file when you gonna be out here? |
| 21:07.00 | bjohnson | ok .. anynow how to sum a file containing one number per row? |
| 21:07.03 | bjohnson | in bash |
| 21:07.59 | wwalker | awk 'BEGIN{x=0}{x=x+$1}END{print $x}' |
| 21:08.25 | wwalker | bjohnson awk 'BEGIN{x=0}{x=x+$1}END{print $x}' filename |
| 21:08.42 | wwalker | bjohnson awk 'BEGIN{x=0}{x=x+$1}END{print x}' filename |
| 21:09.18 | file | outtolunc: 6th to the 12th |
| 21:10.27 | anthm | cat file | perl -e '$x += $_ while <STDIN>; print $x' |
| 21:11.32 | outtolunc | you gonna swing by? |
| 21:12.18 | outtolunc | (i should be done with the capitola office move, by the 3-4th) |
| 21:12.30 | file | where are you? |
| 21:12.36 | outtolunc | campbell |
| 21:12.42 | file | hrm |
| 21:12.44 | file | lemme see here |
| 21:13.32 | file | you're not that far away... hmm |
| 21:13.41 | outtolunc | nope <G> |
| 21:14.02 | bjohnson | cat Master.csv | awk '{ FS = ","; s +=$14; } END { print "Sum is:", s}' |
| 21:14.05 | file[laptop] | mayyyyyyybe |
| 21:14.16 | outtolunc | k |
| 21:14.29 | bjohnson | using that you can grep for a date and calc number of minutes |
| 21:14.45 | bjohnson | (or date range .. like a month) |
| 21:17.40 | file[laptop] | outtolunc: the 12th I'm not doing anything really |
| 21:18.01 | file[laptop] | so if you just want to meet up somewhere, so be it |
| 21:18.15 | outtolunc | k |
| 21:18.51 | file[laptop] | I just need to run to the Government New Brunswick building and get some ID |
| 21:18.55 | file[laptop] | then I shall be set |
| 21:19.04 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-032-081.arcor-ip.net) |
| 21:19.20 | Blackvel | is there any problem with g729 and sipura for asterisk 1.0.2? |
| 21:20.18 | Darwin35 | update to 1.0.5 |
| 21:21.48 | *** join/#asterisk blake0ps (BlakeOPS@c-24-245-24-84.mn.client2.attbi.com) |
| 21:22.41 | *** join/#asterisk jbresler (jbresler@jbresler.cs.uiuc.edu) |
| 21:22.56 | *** join/#asterisk Mneumonic (~Mnemonic@206.231.230.230) |
| 21:22.59 | jbresler | <PROTECTED> |
| 21:23.16 | mrempire | what does this mean?? ;request '551@default' does not exist |
| 21:23.21 | jbresler | I'm not getting any noticable latency when calling with a SIP client |
| 21:23.37 | file[laptop] | mrempire: it means extension 511 in context default does not exist |
| 21:23.40 | blake0ps | Why am I getting this error message? :: Timeout, but no rule 't' in context 'default' |
| 21:24.03 | file[laptop] | blake0ps: something timed out. |
| 21:24.09 | jbresler | I'm writing a custom IAX2 client for use in a research project and need to improve the latency |
| 21:24.16 | blake0ps | file[laptop]: How do I fix it? |
| 21:24.29 | file[laptop] | blake0ps: well what are you doing to cause it? |
| 21:24.35 | mrempire | thanks file[laptop] |
| 21:24.53 | *** join/#asterisk search_learn (~Miranda@209.68.139.150) |
| 21:24.58 | anthm | do something before it times out that will fix it! |
| 21:25.32 | file[laptop] | :) |
| 21:25.43 | blake0ps | file[laptop]: I place a call over a DUNDi network |
| 21:27.50 | eKo1 | Hmm...for some reason, * is storing a blank for the accountcode of two of my sip peers which happen to be quintum voip gateways. |
| 21:28.14 | eKo1 | This makes no sense because I have the accountcode set in sip.conf. |
| 21:30.55 | ionix | quintum eek |
| 21:32.11 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 21:32.42 | WildPikachu[BAR] | cool.... is it easy for anyone to call a UK 0844 number that you can get on www.voipuser.org.. if you from uk? |
| 21:32.55 | search_learn | hardware advice needed!!!!! |
| 21:33.27 | eye69 | search_learn: I can advice you to hold back on the exclamation marks. |
| 21:34.06 | search_learn | ok no problem |
| 21:34.42 | bjohnson | hehe |
| 21:34.47 | bjohnson | ask question |
| 21:34.59 | eye69 | get answer (hopefully) |
| 21:35.33 | Bile_One | Anyone want a drink of this wine? |
| 21:36.35 | eye69 | Bile_One: I've got beer, but thanks anyway. |
| 21:36.52 | Blackvel | darwin: why 1.0.5? i have bristuff |
| 21:37.02 | Bile_One | It is a Port Wine, very stout |
| 21:37.06 | Qwell | Bile_One: Depends. Are we talking like Chardoney, or maddog 20/20? heh |
| 21:37.38 | Bile_One | No mad dog... I drank that once and tried to hitch hike to dalls, bare foot |
| 21:38.06 | Beirdo | Bile_One: free port? I'm there, dude |
| 21:38.20 | Qwell | Bile_One: heh, maddog will do that to you |
| 21:40.01 | *** join/#asterisk murangd (~nukaidc@pool-162-83-230-193.ny5030.east.verizon.net) |
| 21:40.30 | Blackvel | is this problem with g729 and sipura for asterisk 1.0.2 well known? 1.0.5 fixes it? |
| 21:40.43 | bjohnson | search_learn: did we scare you off? |
| 21:43.11 | bjohnson | but enough about me |
| 21:43.59 | ariel_ | bjohnson, I know there are allot of good cigars down here. |
| 21:43.59 | WildPikachu[BAR] | lol |
| 21:45.02 | Beirdo | bjohnson: you are into cigars and port to, eh? :) |
| 21:45.03 | WildPikachu[BAR] | i must find a nice chick to talk on my pabx |
| 21:45.04 | WildPikachu[BAR] | heh |
| 21:45.31 | mountainm2k | Quick newbie question -- I was trying to "demo" asterisk without connecting any trucks, etc -- was just going to use a soft-phone... |
| 21:45.32 | Beirdo | being in Canada has its advantages for Cuban cigars |
| 21:45.46 | bjohnson | where the hell did search_learn disappear to? |
| 21:45.47 | mountainm2k | but I wasn't able to find any good soft-phones that I could get to work... |
| 21:45.50 | mountainm2k | What am I missing? |
| 21:46.04 | bjohnson | Beirdo: I have mostly cuban but actually prefer Honoduran |
| 21:46.10 | bjohnson | Honduran |
| 21:46.12 | outtolunc | so, is there gonna be a meetme setup for VON like it was at astricon? |
| 21:46.14 | ariel_ | Beirdo, you think better then the ones we get here. |
| 21:46.25 | bjohnson | ariel_: where is that? |
| 21:46.26 | Beirdo | my fave is Dominican actually :) |
| 21:46.34 | ariel_ | miami florida. |
| 21:46.40 | tzanger | wow |
| 21:46.43 | bjohnson | ahh .. ex-cuban |
| 21:46.48 | Beirdo | Cuban is good for once in a while, but just too strong a lot of the time |
| 21:46.52 | bjohnson | Little Havana |
| 21:46.52 | tzanger | that's the first time I've been able to hear echo in a zapscan() |
| 21:47.03 | Beirdo | ariel_: they are legal here, not there. :) |
| 21:47.13 | ariel_ | so you think |
| 21:47.27 | bjohnson | Beirdo: Miami is home to many displaced cuban expatriotes |
| 21:47.30 | Beirdo | so the chances of getting a *real* one legally at a shop is higher here |
| 21:47.38 | Beirdo | bjohnson: I realize that |
| 21:47.50 | Beirdo | but Cuban goods are still illegal in the US |
| 21:47.54 | bjohnson | Beirdo: also, castro nationalized tobacoo industry |
| 21:47.58 | Beirdo | and therefore people do get scammed |
| 21:48.05 | bjohnson | Beirdo: not cuban goods .. cuban made |
| 21:48.06 | Beirdo | bjohnson: so? |
| 21:48.11 | ariel_ | Beirdo, yes but we have ways to get all of them here. |
| 21:48.12 | Beirdo | no. |
| 21:48.18 | ariel_ | Including the ones not for export. |
| 21:48.27 | *** join/#asterisk GodThor (~ninja@212.110.67.6) |
| 21:48.30 | Beirdo | Cuban goods. if it comes from Cuba, it's illegal in the US, full stop |
| 21:48.44 | GodThor | i need help? |
| 21:48.52 | Beirdo | stupid laws, but it's not my government (this time) |
| 21:48.54 | GodThor | which nick give a help |
| 21:48.57 | bjohnson | Beirdo: where do you think the cigar makers ended up when castro took over |
| 21:49.03 | ariel_ | godsmoke, ask the question |
| 21:49.05 | bjohnson | there's a bunch of them in Miami |
| 21:49.10 | Beirdo | bjohnson: all over the place. |
| 21:49.12 | GodThor | :))) |
| 21:49.20 | GodThor | no smoke, thor |
| 21:49.33 | sskyles | smokesmoke |
| 21:49.40 | *** join/#asterisk search_learn4004 (~Miranda@209.68.139.150) |
| 21:49.51 | Beirdo | but anyways, I still prefer Dominican for a smoother smoke :) |
| 21:49.58 | ariel_ | Beirdo, I know a customer of mine that makes the cigar here puts the label from cuba then sends it to canada. |
| 21:50.01 | GodThor | when i compile asterisk its says to me " some kind lib missing -lidn |
| 21:50.21 | ariel_ | GodThor, we need the exact error |
| 21:50.21 | Beirdo | ariel_: yeah, there are scammers everywhere |
| 21:50.27 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 21:50.44 | GodThor | " /usr/bin/ld cannot find -ling |
| 21:50.44 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 21:50.48 | GodThor | -lind |
| 21:51.15 | GodThor | "/usr/bin/ld: cannot find -lidn |
| 21:51.15 | GodThor | collect2: ld returned 1 exit status |
| 21:51.48 | GodThor | something with libind library? |
| 21:51.57 | Beirdo | argh, now I want a cigar. |
| 21:52.31 | *** join/#asterisk elge (~elge@l01m-35-111.d4.club-internet.fr) |
| 21:53.05 | GodThor | anyone? |
| 21:53.07 | Beirdo | ariel_: cool. the skills I am sure came from Cuba, but the flavor is different from DR. even from the same seed |
| 21:53.25 | ariel_ | Beirdo, yes your correct. |
| 21:53.37 | GodThor | hmm gods those guys saying about cigar and me ... |
| 21:53.40 | ariel_ | But still most of the cigar makers in the Dr are cubans |
| 21:53.46 | GodThor | :)))) |
| 21:53.54 | Beirdo | and I like the subtleties of the DR flavor better |
| 21:54.04 | Beirdo | and I personally have nothing against Cuba :) |
| 21:54.10 | Beirdo | anyways... |
| 21:54.19 | *** join/#asterisk Asta2 (~asta2@66.180.175.16) |
| 21:54.25 | Beirdo | GodThor: what library? |
| 21:54.28 | techie | a good cuban cigar is like a whore, cost you a lot and you only get to enjoy it for a few minutes |
| 21:54.33 | GodThor | aaa at least |
| 21:54.51 | Beirdo | techie: a few minutes? what kinda piddly cigars are you smoking? |
| 21:55.09 | mountainm2k | Recommend a Windows soft-phone to test Asterisk with??? |
| 21:55.09 | GodThor | what error is this: /usr/bin/ld: cannot find -lidn???/// |
| 21:55.18 | Asta2 | I tried smoking cigars lots of times, but never got any pleasure |
| 21:55.39 | Beirdo | GodThor: looks like you are missing libidn |
| 21:56.04 | Beirdo | which is DNS relates I think |
| 21:56.22 | GodThor | but when i type rpm -qa it says that i have that libind |
| 21:56.32 | Beirdo | huh? |
| 21:56.39 | Beirdo | which is it? |
| 21:56.44 | Beirdo | libidn or libind? |
| 21:57.03 | GodThor | libind ,sorry syntax error |
| 21:57.06 | GodThor | :) |
| 21:57.08 | Beirdo | and you may need the -devel package |
| 21:57.27 | GodThor | so?? what i must do? |
| 21:57.33 | mountainm2k | bind-devel |
| 21:58.05 | Beirdo | and is it libind or libbind? ;) |
| 21:58.10 | GodThor | dont under. |
| 21:58.24 | *** join/#asterisk [1]Flood_of_SYNs (~cory@wbar13.chi1-4.27.95.222.chi1.dsl-verizon.net) |
| 21:59.06 | Beirdo | try what mountainm2k said |
| 21:59.12 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 21:59.22 | Beirdo | mountainm2k: I'd suggest X-lite as a starter |
| 21:59.56 | GodThor | what is bind-devel?explain |
| 21:59.57 | Beirdo | GodThor: and if it's actually libidn, you will need to install libidn |
| 22:00.21 | Beirdo | it's a package |
| 22:00.40 | GodThor | aaaa |
| 22:00.51 | mountainm2k | I don't have anything named libidn on my box and I was able to compile Asterisk... |
| 22:01.40 | mountainm2k | bind-devel : You might need to install that package... |
| 22:01.46 | GodThor | no libidn, libind |
| 22:01.51 | mountainm2k | Although I don't have it and I was able to compile it... |
| 22:02.11 | HitTop | Can anyone give me some guide on how to install asterisk into wrt54g? |
| 22:03.09 | Bile_One | Bjohnson, you got to try Mayo Family Port really good. |
| 22:03.18 | GodThor | but this is something about DNS? why i need DNS? |
| 22:03.19 | mountainm2k | HitTop: Isn't that a Linksys wireless access point? |
| 22:03.39 | *** join/#asterisk HeadachesAbound (user76@wsip-68-99-73-32.tu.ok.cox.net) |
| 22:03.46 | Beirdo | GodThor: I dunno |
| 22:03.52 | tzanger | HitTop: please *try* a little research yourself |
| 22:03.56 | GodThor | tsnami |
| 22:03.59 | GodThor | tsunami |
| 22:04.02 | tzanger | I got relevant results with an obvious google search |
| 22:04.09 | tzanger | first result even |
| 22:04.12 | *** join/#asterisk jarrod (jarrod@dipole.informationwave.net) |
| 22:04.16 | jarrod | exten => 7135554141,1,Answer , exten => 7135554141,2,Wait,1 , exten => fax,1,rxfax(/tmp/fax.tif) |
| 22:04.30 | Bile_One | Bjohnson, try this: http://www.mayofamilywinery.com/zinfandel-port.html |
| 22:04.31 | jarrod | anything wrong with that for receiving a fax to 7135554141 if spandsp is installed and such |
| 22:05.09 | tzanger | jarrod: nothing obvious |
| 22:05.22 | Beirdo | Bile_One: is it made in Portugal? :) |
| 22:05.29 | Beirdo | if not, it's not port, AFAIK |
| 22:05.32 | *** join/#asterisk zigman (~zigman@irc.zigman.de) |
| 22:05.33 | GodThor | thanks Beirdo,m----2k |
| 22:05.37 | tzanger | do you have faxdetect=incoming or =both in your zapata.conf above the channel= that this is coming in on? |
| 22:05.43 | zigman | anyone got a phising website handy ? |
| 22:05.52 | Bile_One | Beirdo, no california |
| 22:05.53 | zigman | i want to try my virusfilter;) |
| 22:05.59 | jarrod | yea |
| 22:06.36 | Beirdo | Bile_One: then it ain't real port :) kinda like there is no real champagne made in California, only in the correct region of France :) |
| 22:06.52 | Beirdo | betcha its still good though :) |
| 22:07.02 | ionix | anyone ever used Nortel Symposium ? |
| 22:07.30 | *** join/#asterisk darkskiez (~mhb@host-84-9-102-60.bulldogdsl.com) |
| 22:07.36 | Beirdo | ahhh!!!! he said the "n" word! |
| 22:08.21 | Silik0n | i'm sor tired of Political Correct bullshit |
| 22:08.36 | Beirdo | Silik0n: hmmm? |
| 22:08.50 | darkskiez | Silik0n: you racist then? |
| 22:08.53 | Silik0n | the "n" word or the "F" bomb |
| 22:09.03 | Beirdo | the "n" word here was Nortel |
| 22:09.04 | tzanger | Beirdo: yeah I read that news with a little bit of a chuckle |
| 22:09.04 | Silik0n | negro please i grew up in the hood |
| 22:09.15 | tzanger | negro please? hahahahahaha |
| 22:09.19 | tzanger | now THAT is politically correct |
| 22:09.29 | tzanger | the term is "nigger please, [objection here]" |
| 22:09.30 | GodThor | web interface for asterisk? |
| 22:09.38 | Silik0n | little hard to be racist when most your friends are all of different races and creeds |
| 22:09.40 | tzanger | or are you going for the painfully white |
| 22:09.46 | greg_work | bjohnson: i haven't even asked actually. i know they do have that voip thing. i wonder if they'll do it without all the voicemail/etc features? |
| 22:09.53 | greg_work | (uh, ignore taht .. sorry) |
| 22:09.58 | darkskiez | Silik0n: that must be hard for you |
| 22:10.01 | Silik0n | tzanger actually is nigga please |
| 22:10.08 | tzanger | Silik0n: you're of course correct |
| 22:10.13 | Beirdo | jeeez |
| 22:10.14 | tzanger | as you can tell, I *am* painfully white |
| 22:10.17 | Silik0n | hah |
| 22:10.23 | tzanger | I can't dance, either |
| 22:10.27 | greg_work | thuogh, btw, i called primus .. only thing they offer is the two voip packages, but they will do toll-free to arbitrary numbers |
| 22:10.28 | Beirdo | or jump |
| 22:10.34 | tzanger | hahaha |
| 22:10.42 | tzanger | dammit now I'm gonna have to go watch robin hood men in tights tonight |
| 22:10.53 | Beirdo | heh |
| 22:10.53 | jarrod | tzanger weird |
| 22:10.56 | jarrod | it just hangs up |
| 22:10.59 | Silik0n | yes i'm a horible racist and i hate gay people (oh no I just told 75% of the people i hang out with that I hate them)_ |
| 22:11.07 | MatsK | GodThor: search www.voip-info.org and you will fins several |
| 22:11.07 | darkskiez | So is someone recording the asterisk samples in ebonics? |
| 22:11.09 | jarrod | -- Executing Wait("Zap/1-1", "1") in new stack == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN' |
| 22:11.23 | Silik0n | screw ebonics i was redneck samples |
| 22:11.26 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 22:11.29 | *** part/#asterisk GodThor (~ninja@212.110.67.6) |
| 22:11.37 | tzanger | jarrod: try s,1,Answer, s,2,rxfax(/tmp/blah.tiff) |
| 22:11.39 | tzanger | does it still hang up |
| 22:11.44 | tzanger | reload of course |
| 22:11.45 | jarrod | lemme try |
| 22:11.53 | Silik0n | complete with "boy you shore do gotta perrrrty mouth" |
| 22:12.12 | Cresl1n | heh :-) |
| 22:12.23 | Silik0n | and "squeeeellll like a pig boy" |
| 22:12.25 | MatsK | GodThor: http://www.voip-info.org/wiki-Asterisk+GUI |
| 22:12.42 | darkskiez | I'm going to make really sarcastic samples when someone presses a wrong button. |
| 22:12.42 | Silik0n | we can call it the huntsville set |
| 22:12.47 | bjohnson | redneck sample: Yous all reached this here MAASHEENE. Leave your goddamn message at the gun shot. |
| 22:12.52 | HeadachesAbound | Thank you for the calling the redneck roadhouse. Today's special is Moose on a Semi with a side of City Slicker Coon. |
| 22:12.58 | Silik0n | darkskiez there are some really good ones out there already |
| 22:13.01 | Beirdo | darkskiez: make sure it calls them a dumbass |
| 22:13.13 | bjohnson | Silik0n: need urls |
| 22:13.20 | darkskiez | need to look into this |
| 22:13.30 | jarrod | wow that time it worked when i did that |
| 22:13.35 | Silik0n | darkskiez: there's one of Allyson saying "Due to high call volume, our PBX is fucked and we will hang up on you as soon as our box segfaults" |
| 22:13.42 | bjohnson | darkskiez: there are some done by Alison if you look around |
| 22:13.43 | ionix | so |
| 22:13.47 | ionix | for the [n]ortel |
| 22:13.49 | ionix | anyone ever used it |
| 22:14.12 | ionix | I want to make an app so that it prompts for card id when people call and pass that to the agent computer |
| 22:14.15 | Silik0n | nortel is over rated unless you need a M1 |
| 22:14.26 | ionix | I know asterisk and avaya do that with the manager interface |
| 22:14.38 | ionix | and I am trying to move from symposium to asterisk in fact ;) |
| 22:14.45 | ionix | but for now, anyone has a slight idea ? |
| 22:15.01 | ionix | I hate nortel BTW :P |
| 22:15.15 | Silik0n | hah |
| 22:15.23 | jarrod | tzanger: it doesnt like my exten => fax, |
| 22:15.29 | Silik0n | URL for what? |
| 22:15.36 | jarrod | if i put exten => XX,2,rxfax(file) |
| 22:15.37 | jarrod | it works |
| 22:15.41 | Hmmhesays | ahh fax |
| 22:15.44 | blake0ps | Is there a way to record call logs without a database (just plaintext suits me fine, I can write an app to parse it) |
| 22:15.47 | jets | Hrm I should be able to redirect in the asterisk manager api to an application, instead of a channel :) |
| 22:16.09 | Hmmhesays | i'm should be about to get a hot chick to go on a date with me |
| 22:16.40 | Silik0n | just redirect to an extension that calls an app |
| 22:17.12 | Silik0n | Hmmhesays thats wasy, just get out the yellow pages and look under "Escort Services" |
| 22:17.29 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 22:19.45 | *** join/#asterisk muntz (~msh@acheron.ne.client2.attbi.com) |
| 22:20.10 | muntz | I can't remember how to access my voicemail on asterisk. |
| 22:20.39 | muntz | My voicemail.conf shows me my mailbox number and passwd |
| 22:20.53 | muntz | but that digits do I dial to get the menu? |
| 22:21.08 | Beirdo | that would be in the extensions.conf, would it not? |
| 22:21.15 | muntz | ah |
| 22:21.42 | ionix | Silik0n: You know symposium TAPI interface ? |
| 22:21.58 | Silik0n | nope |
| 22:22.02 | ionix | hmm k |
| 22:22.31 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 22:22.45 | buddah | how do i reset a zap channel, or t1 card, without resetting the whole box? |
| 22:23.29 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 22:26.27 | Silik0n | hah |
| 22:29.06 | Silik0n | anyone know what class in C# is for MAPI access? |
| 22:29.09 | Silik0n | what up kram |
| 22:29.24 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 22:30.23 | kram | chillin |
| 22:30.40 | sivana | how do you check out based on date? |
| 22:30.45 | sivana | latest CVS has issues |
| 22:30.50 | sivana | I need to go back to the 1st |
| 22:31.01 | eKo1 | head or stable? |
| 22:31.04 | sivana | head |
| 22:31.13 | sivana | i meant, the 1st of Feb |
| 22:31.38 | eKo1 | 'man cvs' and check. |
| 22:31.43 | sivana | lol |
| 22:31.45 | sivana | thanks |
| 22:33.02 | sivana | :) |
| 22:33.05 | eKo1 | Dang it, this blurry ass monitor is giving me a headache. |
| 22:33.15 | Silik0n | time to toss that thing isnt it? |
| 22:33.17 | buddah | does anyone know how to reset a pri without restarting asterisk? |
| 22:33.34 | eKo1 | I tried to steal my co-workers monitor but the bastard caught me in the act. |
| 22:33.42 | eKo1 | So much for that. |
| 22:34.03 | Silik0n | hah |
| 22:34.04 | eKo1 | I hate geting stuck with these puny 14-inchers. |
| 22:34.18 | Silik0n | and who said sized doesnt matter? |
| 22:34.43 | eKo1 | I wish I had a nice LCD flat-screen. |
| 22:34.55 | eKo1 | Or an OLED flat-screen. |
| 22:35.04 | *** join/#asterisk wildcard0 (~generic@S0106006097e16040.vc.shawcable.net) |
| 22:35.35 | wildcard0 | hey. does anyone happen to have any experience with resync'ing the sipura spa-2000 via sip notify? |
| 22:35.45 | *** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
| 22:36.17 | darkskiez | does anyone in here use the cisco 7940/60 ? I just want to check something with the startup sequence.. |
| 22:44.56 | *** join/#asterisk florz (nobody@odnb-d9baa409.pool.mediaWays.net) |
| 22:45.10 | Hymie | http://groups-beta.google.com/group/rec.arts.wobegon/msg/5a460541df793f1a?dmode=source |
| 22:45.13 | *** join/#asterisk joelipgn (~joel@209.163.136.163) |
| 22:46.41 | *** part/#asterisk joelipgn (~joel@209.163.136.163) |
| 22:46.49 | *** join/#asterisk JTPU (~JT@evrtwa1-ar9-4-65-255-236.evrtwa1.dsl-verizon.net) |
| 22:49.10 | *** join/#asterisk Feral_Kid (~feral_kid@red-corp-200.56.96.178.telnor.net) |
| 22:49.18 | JTPU | Anyone available to help a newbie with a question? |
| 22:50.13 | Blackvel | sure just task |
| 22:50.14 | Blackvel | ;) |
| 22:50.17 | Blackvel | ask |
| 22:50.43 | buddah | yeah, they are good with that |
| 22:50.50 | buddah | got a lot answered before i got an RTFM |
| 22:51.14 | JTPU | I have a new installation of Asterisk with a single x100p card at this time. We are going to expand as soon as I can prove it will do what we want. |
| 22:51.21 | JTPU | The biggest thing is in the area of call logging. |
| 22:51.37 | JTPU | Is there a GUI application to help with the setup of that? |
| 22:51.48 | Blackvel | call logging...you mean for billing? |
| 22:52.00 | Blackvel | not that I know...there is asterisk@home |
| 22:52.06 | ariel_ | JTPU, if you want to get started look up asterisk@home. It has it all build in. |
| 22:52.06 | Blackvel | not sure what it provides to you |
| 22:52.21 | ariel_ | It has a gui, call records and reports. |
| 22:52.26 | JTPU | not to actually bill, but for legal issues... Proving that a person called and attemped to go through our IVR |
| 22:52.46 | Feral_Kid | So does anyone have a report on the feasibility of FWDOut... I was thinking about getting that running... |
| 22:53.32 | JTPU | ok, we basically get ID numbers from the callers and pass them to a perl script to do what we want... We could also then just do database inserts with that data, but I wasn't sure if there was something built right in for that |
| 22:53.44 | *** join/#asterisk GodThor (~ninja@212.110.67.6) |
| 22:54.20 | ariel_ | JTPU, asterisk has a full cdr and can also put that info into a mysql database. |
| 22:54.46 | JTPU | that's what i read a brief into to, but I just haven't looked into how difficult it is to setup... |
| 22:54.50 | mountainm2k | Here's a question, re GUI... Is there a GUI to set up extensions, etc? I came from the Lucent Definity world, but I can't seem to get this quite right... |
| 22:55.27 | mountainm2k | Trying to make X-Lite work on a vanilla (Read: just built) asterisk... |
| 22:55.27 | JTPU | i guess that would have been a better question to ask.. What's the easiest way to setup CDR with mysql |
| 22:55.32 | Blackvel | check out DEstar, I have not played around with it, but I guess it does everything yet |
| 22:55.34 | ariel_ | mountainm2k, not built into asterisk out of the box. unless you use setups like AMP or Asterisk@home. |
| 22:55.57 | ariel_ | JTPU, have you seen the wiki? |
| 22:56.18 | *** join/#asterisk wwalker (~wwalker@wwalker.sustaining.supporter.pdpc) |
| 22:56.37 | JTPU | parts of it... I was a major help with one issue we had getting it setup... I will go back to that |
| 22:56.55 | wwalker | OK, I give, what is magic about 26 seconds? I make a bridged call (via AGI->Exec('Dial'...)) and 26 seconds later the AGI says call ended (though I'm still successfully using the bridged call.... |
| 22:57.28 | ariel_ | wwalker, do you have canreinvite=no? |
| 22:58.43 | JTPU | Thank you everyone! |
| 22:59.25 | wwalker | ariel_ I think it's set to yes. I'll check, I'm writing AGI's, the Asterisk admin had the gall to work on other things.... |
| 23:00.29 | ariel_ | wwalker, if the asterisk release the call to your device then the agi thinks it's gone |
| 23:00.30 | jets | Are there still asterisk developer meetme's? |
| 23:01.37 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 23:04.42 | *** join/#asterisk implicit (~bayan@209.80.0.43) |
| 23:04.45 | implicit | whats up |
| 23:04.58 | *** part/#asterisk wildcard0 (~generic@S0106006097e16040.vc.shawcable.net) |
| 23:04.59 | Beirdo | my blood pressure? |
| 23:05.15 | wwalker | ariel_ yes. Harder to track minutes usage (our charges don't change, just whether or not we're int the loop... |
| 23:14.26 | mountainm2k | OK, I downloaded AMP and Asterisk@home but both look like they're designed for a dedicated asterisk server (which would be logical, but not what I'm doing at this moment) |
| 23:15.00 | mountainm2k | Is there a quick-and-dirty way to make SIP-softphone X-Lite work so I can play with it? |
| 23:15.17 | mountainm2k | Heh, trying to jump-start my learning curve of Asterisk... |
| 23:15.19 | greg_work | AMP will install on a non-dedicated server .. but beware, its install is very messy and hardcoded |
| 23:15.28 | Feral_Kid | What is the URL for Asterisk@home |
| 23:15.30 | mountainm2k | Yes I noticed... |
| 23:15.32 | greg_work | take a good look at the directory structure to see where its going to put everything.. |
| 23:15.33 | Beirdo | mountainm2k: it wasn't hard when I tried it |
| 23:15.45 | mountainm2k | Feral_Kid: http://asteriskathome.sourceforge.net/ |
| 23:15.55 | Beirdo | #2 on the google list |
| 23:16.09 | ariel_ | asterisk@home uses amp. |
| 23:16.13 | Beirdo | jbot: google asterisk@home |
| 23:16.37 | ariel_ | but you can configure your .conf files your self and still have the other done fast for you. |
| 23:17.35 | ariel_ | mountainm2k, it takes about 3 or 4 hours to install amp via the instructions. |
| 23:17.40 | mountainm2k | I'm just trying to get it going with one or two extensions so I can play with it... |
| 23:17.46 | ariel_ | it take less then a hour to setup asterisk@home |
| 23:18.00 | ariel_ | mountainm2k, then get asterisk@home |
| 23:18.18 | ariel_ | it's easyer then amp for making changes |
| 23:18.19 | mountainm2k | Thought I considered myself a competent telecom-guy / unix guy, but appearently not smart enough to follow all the asterisk docs... |
| 23:20.11 | mountainm2k | Looked like asterisk@home wanted me to build a new box (IE full wipe/install) |
| 23:21.25 | *** join/#asterisk Evanrude (~david@ip68-96-125-184.lu.dl.cox.net) |
| 23:21.58 | Mavvie | yay! patch accepted. |
| 23:22.13 | FuRR_ | mountainm2k: it does |
| 23:22.19 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 23:22.31 | FuRR_ | mountainm2k: and it sets your TZ to EST |
| 23:22.55 | tessier_ | Is there any way to make a Cisco phone just start dialing when you press a number with the phone on the hook insteread of having to press new call every time? |
| 23:23.07 | tessier_ | That is completely unlike any phone I have ever used and it confuses clients. |
| 23:23.15 | FuRR_ | tessier_: buy the yellow book, or setup the dialplan |
| 23:23.36 | ChulJin | ????? ????? |
| 23:23.37 | ChulJin | oops |
| 23:23.40 | ChulJin | yellow book? |
| 23:23.43 | ChulJin | (forgive my ignorance) |
| 23:23.48 | loud | the signate one ? |
| 23:24.03 | FuRR_ | ChulJin: the Signate book, its got a whole chapter on the cisco phones, great read |
| 23:24.08 | FuRR_ | more than enough info |
| 23:24.16 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 23:24.26 | nitram | FuRR_: i thought that "feature" was only available on the sccp load? |
| 23:24.33 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-166.sbi.ct.charter.com) |
| 23:24.39 | loud | good, i was just looking for an excuse to buy it |
| 23:24.41 | FuRR_ | nitram: its actually just handled in the phones dialplan |
| 23:24.55 | nitram | interesting |
| 23:24.55 | FuRR_ | nitram: its an xml file the phone tftp gets at boot time |
| 23:25.28 | nitram | FuRR_: you don't happen to have a sample of a dialplan with that feature incorporated? |
| 23:26.01 | ChulJin | oh that book |
| 23:26.07 | ChulJin | I rather thought that was widely reviled |
| 23:26.23 | ChulJin | or perhaps I'm confusing it with some other book. |
| 23:26.38 | nitram | FuRR_: when i read the dialplan.xml documentation in the sip admin guide of the cisco i did not see anything about this behaviour iirc |
| 23:27.08 | FuRR_ | nitram: I dont have my book on me, it might be on the wikki |
| 23:27.25 | srt | i dont think the dialplan helps with that one |
| 23:27.35 | nitram | FuRR_: are we really talking about hte same thing here? |
| 23:27.38 | srt | you still have to press new call at first |
| 23:27.44 | FuRR_ | srt: it does, i rememeber beause the book actually mentioned it |
| 23:27.49 | srt | of take the phone off hook |
| 23:27.51 | nitram | wow |
| 23:28.14 | FuRR_ | srt: basically it said "if you dont use a dial plan this happens" |
| 23:28.20 | srt | hmm |
| 23:28.56 | srt | i have a dialplan setup and it doesnt work that way |
| 23:29.18 | srt | but for my four digit internal numbers it dials immediately after the 4th digit |
| 23:29.34 | srt | without having to press the dial button afterwards or waiting for the timeout |
| 23:30.03 | FuRR_ | something like "9..." type="phone" |
| 23:30.38 | srt | <PROTECTED> |
| 23:32.17 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 23:32.23 | FuRR_ | you have to make a seperate template match for each scenario |
| 23:32.46 | FuRR_ | so say a 10 digit dial, would be "9,1.........." |
| 23:32.53 | file[laptop] | BoRiS!!!!!!!!!!!!!! |
| 23:33.04 | FuRR_ | and a regualar 7 digit "9......." |
| 23:33.15 | BoRiS | file!!!!!!!!!!!! |
| 23:33.22 | FuRR_ | the matches are exact |
| 23:33.22 | srt | sure but you still can't dial when on hook ;) |
| 23:33.35 | file[laptop] | BoRiS: waddup |
| 23:33.56 | srt | it only saves pressing dial or waiting for the timeout |
| 23:34.10 | FuRR_ | srt: i thought his issue was dialing without having to touch the dial button, didnt know he wanted to dial on hook, |
| 23:34.18 | *** join/#asterisk salmandr (~salmandr@h216-170-207-50.216-170.unk.tds.net) |
| 23:34.46 | nitram | hehe |
| 23:34.59 | nitram | srt: thanks for clearing that up again |
| 23:35.14 | nitram | FuRR_: afaik the sccp load can do exactly that |
| 23:35.53 | *** join/#asterisk Gronker (~Gronker2@adsl-220-89-99.ags.bellsouth.net) |
| 23:43.53 | Asta2 | How do I make asterisk forward all incoming connections to another sip-phone at another server? |
| 23:44.39 | ScythelX | anyone currently using ser with asterisk? I was wondering if someone can look at something for me quickly |
| 23:50.36 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 23:52.05 | Blackvel | cu all |
