| 00:00.05 | DogBoy | oh wait he's dead already |
| 00:05.29 | DogBoy | http://www.voip-info.org/wiki/view/Asterisk+G.729+Licensing |
| 00:05.44 | DogBoy | "Under patent law, it is a legitimate use to study or experiment with a patented technology without paying for a patent license." |
| 00:06.19 | WIMPy | Ich which countries? |
| 00:06.41 | DogBoy | lol |
| 00:06.45 | DogBoy | what |
| 00:06.51 | [TK]D-Fender | DreamLand :) |
| 00:07.05 | WIMPy | :-) |
| 00:07.10 | *** join/#asterisk [Outcast] (~anonymous@24-181-235-14.dhcp.oxfr.ma.charter.com) |
| 00:11.35 | DogBoy | http://forums.whirlpool.net.au/forum-replies-archive.cfm/481223.html |
| 00:11.45 | DogBoy | "Free vs Paid G729 codec for asterisk." |
| 00:12.27 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 00:16.59 | xheliox | Not that I'm indifferent to breaking software licenses, but when's the last time (if ever) that someone has gone to jail for that? |
| 00:19.28 | [TK]D-Fender | xheliox: My bet would be : nasty fine |
| 00:20.17 | xheliox | probably more likely a civil judgement |
| 00:20.35 | xheliox | and the patent holder would have to determine damages |
| 00:24.13 | *** join/#asterisk _Speedy2k (~speedy_2k@modemcable187.150-57-74.mc.videotron.ca) |
| 00:27.06 | _Speedy2k | Can someone help me to manualy desactivate the hardware echo cancelation of my TDM400p, i know Rhino have a line to add into the rhino.modprobe.conf, but don't know how to do it for a digium card ? |
| 00:27.57 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 00:29.42 | ChannelZ | it should be in your /etc/dahdi/system.conf |
| 00:30.02 | *** join/#asterisk JerJer (~PhatJ@asterisk/original-h323-guy/JerJer) |
| 00:30.20 | _Speedy2k | for the hardware echo cancelation to be activated what it should look lie ? |
| 00:31.02 | _Speedy2k | here is my current system.conf file |
| 00:31.03 | _Speedy2k | # Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER) |
| 00:31.03 | _Speedy2k | fxsks=1-4 |
| 00:31.03 | _Speedy2k | # Global data |
| 00:31.03 | _Speedy2k | loadzone= us |
| 00:31.03 | _Speedy2k | defaultzone= us |
| 00:33.16 | ChannelZ | echocanceller=something |
| 00:33.23 | ChannelZ | I'm not sure what the hardware ec is called |
| 00:34.11 | ChannelZ | dahdi_scan might tell you |
| 00:34.39 | WIMPy | If I understood it right, hw-ec is activated automatically, if detected. |
| 00:34.51 | _Speedy2k | [1] |
| 00:34.51 | _Speedy2k | active=yes |
| 00:34.51 | _Speedy2k | alarms=OK |
| 00:34.51 | _Speedy2k | description=Wildcard TDM410P Board 1 |
| 00:34.51 | _Speedy2k | name=WCTDM/0 |
| 00:34.52 | _Speedy2k | manufacturer=Digium |
| 00:34.52 | _Speedy2k | devicetype=Wildcard TDM410P (VPMADT032) |
| 00:34.53 | _Speedy2k | location=PCI Bus 04 Slot 01 |
| 00:34.53 | _Speedy2k | basechan=1 |
| 00:34.54 | _Speedy2k | totchans=4 |
| 00:34.54 | _Speedy2k | irq=66 |
| 00:34.55 | _Speedy2k | type=analog |
| 00:34.55 | _Speedy2k | port=1,FXO |
| 00:34.56 | _Speedy2k | port=2,FXO |
| 00:34.59 | ChannelZ | STOP PASTING THAT SHIT |
| 00:35.01 | ChannelZ | ~pb |
| 00:35.02 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 00:36.17 | _Speedy2k | http://asterisk.pastey.net/138801 |
| 00:36.25 | _Speedy2k | I want to disable it |
| 00:37.47 | ChannelZ | do echocancel=off |
| 00:39.08 | ChannelZ | actually that would go in your /etc/asterisk/chan_dahdi.conf |
| 00:39.59 | ChannelZ | the 'echocanceller' in system.conf just selects which ones to enable, you can cofigure what channels use it at all in chan_dahdi.conf |
| 00:43.42 | _Speedy2k | Ok but digium doesn't have a line to add in the module loading file to disable the hardware board like rhino ? |
| 00:44.04 | ChannelZ | who cares? |
| 00:47.07 | _Speedy2k | I want to do a fxotune, but they tell to disable echo cancelation when doing that, but if asterisk if shut off, how-do i know th eboard down't do any echo cancelation ? |
| 00:50.40 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
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| 00:55.14 | ChannelZ | who are 'they'? |
| 00:55.53 | WIMPy | Don't you know THEM? |
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| 00:57.46 | Katty | Qwell: i was thinking of redditrouletting |
| 00:58.10 | Katty | Qwell: there's only 2 people on. |
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| 01:02.15 | leifmadsen | HI! |
| 01:02.42 | Katty | hi leif |
| 01:02.50 | leifmadsen | Katty: heyo! |
| 01:02.52 | Katty | no one wants to roulette with me |
| 01:02.53 | Katty | :< |
| 01:03.20 | leifmadsen | Katty: I'd totally "roulette" with you |
| 01:03.24 | leifmadsen | (whatever that means) |
| 01:03.38 | leifmadsen | that, and I have no idea what "redditroulette" is |
| 01:03.52 | Katty | it's like chatroulette |
| 01:03.59 | Katty | but for reddit users. so much smaller population |
| 01:04.56 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
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| 01:06.58 | leifmadsen | Katty: and less penis? |
| 01:07.49 | Katty | lol yes, that's the point |
| 01:07.54 | Katty | btn |
| 01:07.57 | Katty | brb |
| 01:09.38 | JerJer | yawns |
| 01:09.52 | leifmadsen | JerJer: heyo! |
| 01:09.59 | JerJer | sup sup |
| 01:14.44 | leifmadsen | nada much -- just printing off some forms so I can get my SCSI HD order through |
| 01:18.38 | drmessano | I saw your tweet about those drives earlier. Sounded like the usual (1) Does someone have these? and (2) Which kidney do you want? |
| 01:19.39 | leifmadsen | drmessano: ya, I found a site that has them, and I'm just filling out the forms for which kidney I have to donate |
| 01:20.19 | JerJer | leifmadsen: i might know another source too |
| 01:20.29 | leifmadsen | well I've already placed the order |
| 01:20.35 | JerJer | d'oh |
| 01:20.36 | leifmadsen | I'm just filling out the CC info now |
| 01:20.41 | leifmadsen | $29 a drive |
| 01:20.47 | leifmadsen | didn't seem TOO bad |
| 01:20.53 | JerJer | yeah |
| 01:21.00 | drmessano | I had to buy a couple drives for HP servers. $500 for a 146GB "Drive kit" from HP or $235 for the original OEM drive and $1.50 for the spline driver to replace the drive in the shell |
| 01:21.09 | leifmadsen | and now my brscan app isn't being detected by gscan2pdf |
| 01:22.36 | TJNII | $29 a drive? How big? |
| 01:23.49 | leifmadsen | 36GB SCSI U160 80pin |
| 01:24.05 | TJNII | That's not bad. |
| 01:25.08 | TJNII | I actually salvaged a pile of 36G SCSI drives from some decomissioned servers this week. Haven't done anything with them, yet. |
| 01:25.32 | drmessano | Use the .90 mil garbage bags. Less tearing |
| 01:27.26 | TJNII | Haha. I got my hands on a FC JBOD I used instead of them, so they are probably going back out for the recycle guys. |
| 01:36.06 | *** join/#asterisk QubeZ (~nkasu@68.204.67.110) |
| 01:36.12 | QubeZ | hello all |
| 01:36.40 | QubeZ | I just installed the UT50 Sangoma usb adapter and compile wanpip-voicetime source but my dahdi_test is still running very low timings: 99.971% etc... Any ideas? |
| 01:41.51 | *** join/#asterisk jql (~jql@12.9a.344a.static.theplanet.com) |
| 01:42.01 | pabelanger-lap | QubeZ: low? |
| 01:42.25 | QubeZ | pabelanger-lap: isn't 99.998% the expected value when using a dedicated timing hardware? |
| 01:44.20 | pabelanger-lap | Hmmm, you should be good until 99.975 from what I've read. So, you do need to bump it up a little |
| 01:44.52 | QubeZ | pabelanger-lap: not sure why its not working properly |
| 01:45.07 | pabelanger-lap | perhaps an IRQ issue? |
| 01:45.20 | pabelanger-lap | Are you sharing the IRQ? |
| 01:45.29 | QubeZ | checking |
| 01:45.38 | pabelanger-lap | try a different slot |
| 01:47.00 | QubeZ | its a usb port |
| 01:48.27 | pabelanger-lap | QubeZ: do you need it? If not, disable it in your bios |
| 01:48.44 | pabelanger-lap | otherwise, like i said, try another slot |
| 01:48.52 | QubeZ | pabelanger-lap: no i mean, the timing device is a usb device |
| 01:49.03 | QubeZ | i enabled apic in the kernel and rebooting now |
| 01:49.19 | pabelanger-lap | QubeZ: ok |
| 01:49.48 | QubeZ | thanks for getting me on the right path (irq issue) |
| 01:53.56 | pabelanger-lap | np |
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| 02:15.46 | mmlj4 | are there docs for splitting voice and data channels on a digium T1 card? I can't find any |
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| 02:18.45 | pabelanger-lap | mmlj4: yes, in DAHDI sample files |
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| 03:30.41 | mmlj4 | nethdlc=13-24 # trivial, hah |
| 03:31.26 | phix | hi gang |
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| 03:35.46 | WIMPy | To the Polycom fans in here: Is teh SoundPint ip 430 SIP a decent model for evaluation or is it more mlike low end? |
| 03:38.35 | carrar | it's a excellent phone |
| 03:39.08 | WIMPy | Ok, then I will try to get one of them. |
| 03:39.16 | mmlj4 | if you can stand polycom, anyway |
| 03:39.34 | WIMPy | That's what I wanted to find out. |
| 03:40.10 | WIMPy | May people seem to like them so I want to find out if I do as well. |
| 03:40.20 | carrar | though the 430 has been replace by the 450 |
| 03:40.46 | carrar | but 430 still makes a great desktop (none g722) phone |
| 03:41.04 | WIMPy | But that wouldn't make sense if it's a model likely to give me a bad impression. |
| 03:41.19 | carrar | Do you want g722? |
| 03:42.16 | WIMPy | Don't care. It really just to find out about Polycoms. So if that's the biggest point, it does not matter. |
| 03:42.32 | WIMPy | Oterwise G722 is a good thing, off course. |
| 03:42.43 | carrar | Why don't you read their marking information so that you can make the correct choice |
| 03:42.48 | carrar | marketing |
| 03:43.18 | WIMPy | Marketing can tell you a lot... But not what it's really like. |
| 03:44.48 | carrar | we use a lot of Polycom, they're great phones. Also if you need support on their phones for bug fixes we've gotten new code for fixes. |
| 03:45.03 | carrar | Not gonna get that from Cisco |
| 03:45.59 | WIMPy | Well, some people also like Cisco, but I'm not so sure about their expertise in that area. |
| 03:46.07 | carrar | yeah I like the look and feel of cisco |
| 03:46.20 | carrar | the 7941, 7970 |
| 03:46.30 | carrar | they are decent phones |
| 03:47.25 | WIMPy | Problem is their licencing for SIP firmware. |
| 03:47.38 | carrar | and their support |
| 03:47.54 | WIMPy | That's understood :-) |
| 03:51.24 | carrar | http://www.polycom.com/products/voice/comparison/desktop_phone_matrix.html |
| 03:53.25 | WIMPy | I don't need a phone. It's just that a bunch of those 430s are on ebay and I thought that it might be a good chance to find out about Polycom and if they're really good. |
| 03:54.12 | boodu | bye |
| 03:54.15 | WIMPy | wonders if that information will rise the prices... |
| 03:54.25 | mmlj4 | how much are they going for? |
| 03:54.26 | carrar | I like polycom, they have a ton a features and work great in a mass deployment |
| 03:55.03 | WIMPy | Some are still waiting for the first Euro to be bid. |
| 03:55.41 | carrar | $89? |
| 03:56.15 | WIMPy | The highest is 6,50. |
| 03:56.22 | WIMPy | With 5h to go. |
| 03:57.36 | carrar | i don't see it |
| 03:57.53 | carrar | But then again I suck at eBay |
| 03:58.07 | ChannelZ | ebay sucks |
| 03:58.15 | carrar | thats probably why |
| 03:58.25 | WIMPy | Did you search worldwide? |
| 03:58.41 | carrar | not if thats the default |
| 03:59.01 | WIMPy | It isn't. |
| 03:59.28 | carrar | ah |
| 03:59.30 | carrar | woah |
| 03:59.40 | carrar | $1.29 |
| 03:59.41 | carrar | hahah |
| 04:00.04 | carrar | Is that phone from Nigeria? |
| 04:00.39 | WIMPy | Nobody knows... |
| 04:01.20 | carrar | Using that bidding tool that bids up a few cents right at the last min? |
| 04:01.39 | WIMPy | Doing that by hand. |
| 04:01.51 | carrar | I tried using one of those once and it didn't work |
| 04:02.22 | carrar | was pissed cause it was a awesome deal on a coffee grinder |
| 04:02.22 | phix | I like my snom |
| 04:02.49 | WIMPy | Ja, Snom ist the best I've seen so far. |
| 04:02.58 | carrar | not a fan of Snom |
| 04:03.37 | WIMPy | Actually it's the only voip phone Ive seen so far, I like at all. |
| 04:03.56 | carrar | woah you need to try more phones |
| 04:04.13 | WIMPy | Well, the linksys SPA 962 hardware with Snom Software would be ace :-) |
| 04:04.54 | WIMPy | Maybe I find another kind I find usable? |
| 04:05.03 | WIMPy | It's always good to have choice. |
| 04:05.09 | carrar | it is |
| 04:05.35 | phix | WIMPy: ja! |
| 04:09.16 | *** join/#asterisk iscsi (~light@78.108.73.46) |
| 04:17.01 | *** join/#asterisk pabs3 (~pabs@ppp121-45-240-95.lns20.per2.internode.on.net) |
| 04:19.55 | pabs3 | how do I make the voicemail unavailable message not include a reading of the extension that was dialed? |
| 04:20.08 | pabs3 | s/voicemail/default voicemail/ |
| 04:26.32 | ChannelZ | I think it only does that when you have no greeting actually recorded? (IE it's all generic) |
| 04:28.07 | ChannelZ | hmm or not. Mine doesn't. Guess I'm not sure what exactly you're referring to then. |
| 04:28.45 | ChannelZ | or are you running FreePBX? probably it does a bunch of extra nonsense |
| 04:29.37 | ChannelZ | nope I stand corrected! Calling VoiceMail with the 'u' flag but no custom greeting recorded will say that. |
| 04:29.45 | WIMPy | No, standard asterisk also reads the extension (or mailbox#?) unless anything is recorded. |
| 04:29.48 | ChannelZ | So record a greeting. |
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| 04:30.48 | pabs3 | ChannelZ: I want to the change the default, not the one for a particular mailbox |
| 04:32.22 | pabs3 | tries without the 'u' flag |
| 04:33.27 | ChannelZ | That will just play the instructions. |
| 04:34.44 | pabs3 | yeah, not exactly what I want |
| 04:34.54 | ChannelZ | well what is it you want it to say? |
| 04:35.27 | ChannelZ | You could re-record vm-intro (or kludge it together from other existing prompts you want it to say) |
| 04:35.39 | ChannelZ | otherwise you're off to the races hacking the source |
| 04:35.55 | ChannelZ | or building the entire thing with MiniVM |
| 04:38.00 | pabs3 | meh, will just leave it with |u for now |
| 04:38.47 | pabs3 | I wanted something like "The person you have called is unavailable" |
| 04:39.05 | ChannelZ | Make it say whatever you want.. Playback() that, then call VoiceMail with the 's' flag |
| 04:40.29 | pabs3 | but then if a user has recorded their own greeting, they'll get both greetings :) |
| 04:42.09 | ChannelZ | life is full of ups and downs |
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| 05:12.42 | josexato | Hello, where can i find information about how dsp.c is called during a call? |
| 05:13.30 | josexato | does any body knows if it's working constantly |
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| 05:38.14 | pabs3 | ChannelZ: a few playbacks and gotoifs later, I have something much closer to what I wanted, thanks for the advice |
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| 06:27.24 | KingDavidNYC | hello everybody |
| 06:28.00 | KingDavidNYC | anybody here? |
| 06:29.19 | josexato | hi |
| 06:29.58 | KingDavidNYC | there your are, here is one guy! |
| 06:30.15 | KingDavidNYC | is everybody else sleep? |
| 06:31.23 | cmendes0101 | maybe |
| 06:32.40 | KingDavidNYC | my friend, do you know trixbox? |
| 06:34.01 | cmendes0101 | me? not specifically, but if you have a question just ask. Someone will probably answer |
| 06:35.20 | KingDavidNYC | this is a trixbox that has a programmed queue, I need to change the order of the extensions in the queue programtically |
| 06:35.56 | KingDavidNYC | can anybody tell me where trixbox stores the queue information? |
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| 06:54.47 | ChannelZ | only the pimp knows |
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| 07:25.25 | davido1 | hello room... Can someone help me to understand how hints and notifycid work? |
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| 08:07.24 | davido1 | So anyway, I have some Snom phones here. My SIP peers all use the context "from_intern" to dial. In the context "from_intern", I have hints for the peers configured. But when, say, the peer 3 calls the peer 5, all I see in my display is: "From: 5 To: 5" |
| 08:07.30 | davido1 | Any help? |
| 08:10.26 | Aqituado | sorry, nothing comes to mind (but i am new at this) |
| 08:10.34 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:10.57 | Aqituado | What does: "Really destroying SIP dialog" mean.... (from sip debug).... |
| 08:14.36 | *** join/#asterisk mpe_ (~mpe@gate.ipvision.dk) |
| 08:15.07 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 08:22.00 | ChannelZ | well SIP is a dialog in that most messages need a reply. So they are sort of cached until whatever action is complete and the SIP conversation as it were is over, and then destroyed (forgotten) |
| 08:22.42 | *** join/#asterisk garymc (~chatzilla@host81-148-29-236.in-addr.btopenworld.com) |
| 08:22.49 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 08:23.06 | Aqituado | the reason im asking, is that my calls are being "cut off" |
| 08:23.27 | Aqituado | Really destroying SIP dialog '6d45ef5f21eb0bcd528aabbb73dedb49@93.167.108.92' Method: OPTIONS |
| 08:23.27 | Aqituado | <PROTECTED> |
| 08:23.31 | garymc | good morning all |
| 08:23.34 | Aqituado | and then my calls end :( |
| 08:23.55 | Aqituado | it seems more frequent with offsite phones (nat) than internal phones. |
| 08:24.02 | Aqituado | but that could just be random |
| 08:24.06 | Aqituado | morning garymc |
| 08:24.23 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 08:25.47 | Aqituado | could it be that im loosing packages, and somehow asterisk thinks the dialog is over, and calls a hangup ? |
| 08:25.49 | ChannelZ | Well they might be related but it's not necessarily that particular dialog being destroyed that is doing it. OPTIONS is usually just Asterisk 'pinging' the peer |
| 08:26.02 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 08:26.57 | Aqituado | where should i start looking when asterisk calls "hangup" on my calls randomly.... (within a few minutes per call) |
| 08:27.22 | ChannelZ | do you have a call timeout set? |
| 08:27.37 | Aqituado | dunno, where can i find it ? |
| 08:28.20 | Aqituado | but it does sound promissing |
| 08:28.27 | ChannelZ | Whatever Dial() command used should appear on the console |
| 08:28.52 | ChannelZ | Do all your calls get undesireably ended? Is it the same duration each time? |
| 08:29.16 | ChannelZ | Does the media stream choke and then after a time the call terminates or is the whole thing just bang, disconnected? |
| 08:29.32 | Aqituado | its not exactly the same duration |
| 08:29.44 | Aqituado | could be a choke..... |
| 08:30.21 | Aqituado | I dont see any timeout .... (did a search in the dump ive made with v30 d30 and sip debug |
| 08:30.26 | *** join/#asterisk lost_soul (~shawn@cpe-67-241-66-112.twcny.res.rr.com) |
| 08:32.48 | Aqituado | can i send the dump (500 lines) to you ? so you could just "scan" it through to se if you se any abnormalityes ??? |
| 08:33.07 | ChannelZ | pastebin it |
| 08:33.34 | Aqituado | check =) |
| 08:34.14 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 08:34.56 | Aqituado | http://pastebin.com/rYi6KEtp |
| 08:36.12 | Aqituado | ive got one more conversation with similar debug info that i could paste .... |
| 08:38.50 | mort_gib | Aqituado is this SIp - to - SIP? |
| 08:39.02 | Aqituado | yes |
| 08:39.06 | mort_gib | ok |
| 08:39.17 | mort_gib | network and NAT problems would be a guess |
| 08:39.38 | Aqituado | but the funny thing is, that the problems just came out of random.... |
| 08:40.01 | Aqituado | i havent touched anything. have been working for 1½ years.... now ... 1 month ago or so... it began doing "funny" things.... |
| 08:40.29 | Aqituado | but ive also exsperienced it Onsite -> PSTN ... (witch is also SIP..... :S) |
| 08:41.00 | mort_gib | Aqituado Setup smokeping to monitor SIP devices that have issues |
| 08:41.18 | Aqituado | what am i looking for ? |
| 08:41.22 | *** join/#asterisk geemee (~ocs@mailhost.exterity.com) |
| 08:41.33 | mort_gib | high latency, lost packages |
| 08:41.54 | Aqituado | shouldnt i also be noticing this in the conversation ? |
| 08:42.22 | Aqituado | the conversation is "smooth" no delays and so on... |
| 08:42.43 | mort_gib | But somehow packages are dropped |
| 08:42.56 | geemee | Morning / Evening all. I have joined 2 asterisk boxes together and have set the dial plan so extenstions at both sites can call each other. How can I setup so that if someone externally phones into 1 trixbox I can transfer the call to the other box? |
| 08:42.57 | mort_gib | I had a similar issues with a Sangoma card |
| 08:43.14 | mort_gib | A firmware update cleared the issue |
| 08:43.16 | ChannelZ | So in the example you pasted, SIP/409 is calling a bunch of other peers but you're answering SIP/407 ? |
| 08:43.30 | Aqituado | yes... 409 is calling a ringgroup. |
| 08:43.40 | Aqituado | 207, 307, 407 |
| 08:43.46 | Aqituado | and 407 pickes up |
| 08:43.57 | ChannelZ | geemee: make an extension that Dial()s an extension on 'the other side'.. transfer to that extension |
| 08:44.59 | geemee | Channelz: Ahhh.. so I make say an extension 400 on trixbox 1 and this dials()s extension 500 on asterisk server 2 |
| 08:45.02 | Aqituado | is there some way i can make asterisk more "resiliant" agaings lost packages ? |
| 08:45.38 | geemee | therefore to transfer to 500 I would transfer the call to 400 |
| 08:46.08 | ChannelZ | geemee: yup |
| 08:46.34 | geemee | excellent, cheers for you help.. I may need to work on logic for the extensions now but that can be sorted out. |
| 08:47.34 | ChannelZ | typically you designate a range of extensions (IE 1xx is office 1, 2xx is office 2) or some other pattern that you can easily pick up anything dialed with a certain prefix and just barf it to the other server |
| 08:48.29 | Aqituado | well.. thats just extention logic, and not realy the issue here =) but in my head, i picked 1xx ring group, 2xx softphone, 3xx tablephone, 4xx laptop softphone |
| 08:49.02 | geemee | ChannelZ: thats fine for creating a new setup but its a different story when you inherit lots of independant offices that were never linked before and had their own structure |
| 08:49.06 | geemee | :) |
| 08:49.08 | Aqituado | might not be the normal way to do it, but its been doing the job for 4 years now =) ..... granted ive only got 6 "persons" |
| 08:49.32 | ChannelZ | Aqituado: Well I don't see anything in the dialog to indicate that either end is terminating the call on purpose, so I'm not sure what is going on -- probably what mort suggests, some other network issue is causing a breakdown. |
| 08:50.49 | Aqituado | ok |
| 08:50.53 | ChannelZ | geemee: well you can do it however you feel like, that's just a common way so you don't have to write crazy big dialplans to handle being able to call each extension at each remote office. Basically make an "area code" that applies to other offices. |
| 08:51.01 | Aqituado | ill setup smokeping and start monitoring the connection |
| 08:51.09 | *** join/#asterisk ndemir (~ndemir@155.223.46.171) |
| 08:51.30 | Aqituado | and i gues there is no way of making asterisk more resiliant agaings packet los ? |
| 08:51.32 | Aqituado | loss* |
| 08:51.37 | ndemir | how can i run asterisk 1.4 on tcp port? It runs on udp. |
| 08:51.57 | geemee | Channelz: yeah thats what Im thinking.. essentially changing from 3digitextensions to 4 digit extensions for transferring to other offices. |
| 08:52.07 | ChannelZ | So if Person@Office-A wants to call Person@Office-B, he dials "02xxx" or something where 'xxx' is Person@Office-B's local extension. |
| 08:52.19 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 08:52.36 | geemee | Yep Makes perfect sense and easy structure for people to remember |
| 08:52.46 | ChannelZ | you use the 02 as a way to match a pattern but strip it off and just send the call to the correct office with the plain extension |
| 08:52.51 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 08:55.23 | Aqituado | is there some debug/verboose lvl i can set that tells me when there is packet loss ? and why it calls the Hangup ??? |
| 08:55.32 | ChannelZ | Aqituado: Do you have notices turned on for console logging, and some verbosity turned on? It seems to me there is no real indication that anything other than a normal call termination has happened. |
| 08:56.08 | Aqituado | i have debug 30, verbose 30 and sip debug on the two extentions |
| 08:57.34 | ChannelZ | but is notice turned on in logger.conf? I don't see any. |
| 08:57.56 | Aqituado | ill look |
| 08:58.14 | Aqituado | whats the path for logger.conf again ? |
| 08:58.40 | ndemir | how can i run asterisk 1.4 on tcp port? It runs on udp. how can i run asterisk 1.4 on tcp port? It runs on udp.how can i run asterisk 1.4 on tcp port? It runs on udp. |
| 08:58.47 | ChannelZ | /etc/asterisk in normal world, you're running trixbox (?) which I have no clue what they might have done |
| 08:58.52 | Aqituado | ChannelZ found =) |
| 08:59.22 | ChannelZ | ndemir: maybe google maybe google maybe google |
| 08:59.24 | Aqituado | ChannelZ the paste i gave you was from console, so there is no notice there..... because i wanted to have SIP debug |
| 08:59.47 | Aqituado | ill make a dump from the full log and pastebin it... give me a moment |
| 08:59.55 | tzafrir_laptop | ndemir, not 1.4. That is: not without heavy-duty patching, IIRC |
| 09:00.02 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 09:00.39 | ChannelZ | Aqituado: well all I can tell is the call seems to end normally, Asterisk sends a BYE to the phone.. |
| 09:00.58 | Aqituado | so its not because of packet loss ? |
| 09:01.19 | tzafrir_laptop | It's of the type of fixes that if you didn't know how to search for in google, you probably have no chance of successfully backporting ;-) |
| 09:01.24 | ChannelZ | Aqituado: Perhaps trixbox is doing some weird shit I'm not seeing, setting a call timeout somehow with one of it's AGIs... or I see some references to call recording, maybe there is a limit set somewhere there, or your disk is full... I have no idea |
| 09:01.44 | Aqituado | ok |
| 09:02.06 | Aqituado | lots of disk free |
| 09:02.14 | ChannelZ | with notices/errors turned on you can try another call and see if it says anything interesting right before it ends the call |
| 09:02.26 | Aqituado | thx =) i might be back :D |
| 09:02.28 | ChannelZ | but based on this I don't really see what is going on. |
| 09:03.01 | Aqituado | ill try and pastebin the full log and come back.... just gota snip' the right part . |
| 09:04.52 | *** part/#asterisk ndemir (~ndemir@155.223.46.171) |
| 09:06.26 | ChannelZ | I did just notice line 520 which looks odd to me ("xpires: 180I>") but maybe that's a bad paste? |
| 09:07.30 | Aqituado | ill check my dump.... |
| 09:08.16 | ChannelZ | I think it must be just a console vomit, the I> is from the console prompt srv-trix*CLI> |
| 09:08.35 | Aqituado | yes... that seems likely |
| 09:08.39 | ChannelZ | like it's overlayed. dunno where the E went but I'm sure it's just fluke |
| 09:08.51 | Aqituado | i have a BUTLOAD of: |
| 09:09.05 | Aqituado | [Jul 22 10:15:46] DEBUG[17525] channel.c: :::=== Now have 1 locks (recursive) |
| 09:09.05 | Aqituado | [Jul 22 10:15:46] DEBUG[17525] channel.c: ::::==== Channel SIP/407-0000002d was locked |
| 09:09.05 | Aqituado | [Jul 22 10:15:46] DEBUG[17525] channel.c: ::::==== Unlocking AST channel SIP/407-0000002d |
| 09:09.05 | Aqituado | [Jul 22 10:15:46] DEBUG[17525] channel.c: ::::==== Channel SIP/407-0000002d was unlocked |
| 09:09.05 | Aqituado | [Jul 22 10:15:46] DEBUG[17525] channel.c: ====:::: Locking AST channel SIP/407-0000002 |
| 09:09.19 | Aqituado | we are talking .... 200mb of that... in 1½ hour.... |
| 09:09.44 | Aqituado | that is not "normal" :) and might have something to do with the problem.... not sure.... im still looking for the conversation ive pasted.. |
| 09:12.46 | ChannelZ | well yeah that's what you get for turning on debug |
| 09:12.56 | Aqituado | ohhh :) so thats normal ? |
| 09:13.10 | ChannelZ | [Jul 22 10:15:46] DEBUG[17525] |
| 09:13.13 | ChannelZ | <PROTECTED> |
| 09:13.24 | Aqituado | but it seems to be the same two extentions doing it. |
| 09:14.08 | ChannelZ | every random thought going through Asterisk's head is getting logged. It's debug, not an error. It's meant to trace just about every step * is doing |
| 09:14.12 | Aqituado | i mean,... its like.... 200 lines for each second... only those two extentions.... :D |
| 09:14.24 | Aqituado | k |
| 09:14.40 | ChannelZ | you almost never need debug on and certainly not on maximum |
| 09:14.55 | Aqituado | ok |
| 09:14.58 | Aqituado | ill keep that in mind |
| 09:15.08 | Aqituado | ive just msg'ed you |
| 09:15.20 | Aqituado | thought it be better than not spamming the channel with log |
| 09:15.47 | *** part/#asterisk davido1 (~davido1@p54B0A6CC.dip0.t-ipconnect.de) |
| 09:19.18 | tzafrir_laptop | ~pb |
| 09:19.19 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 09:19.34 | tzafrir_laptop | Some more spam to the channel |
| 09:20.35 | Aqituado | tzafrir_laptop =) |
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| 10:20.57 | ndemir | what parameters should i pass to AJAM for click2call? |
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| 10:52.40 | addeswe | I have a short question about FreePBX... I've made some extensions (sip) and clicked save (submit) and also clicked Apply Configuration Changes. They show up in the webinterface but when i run a sip show users in CLI, I got no users. /etc/asterisk/* has writingpermissions and im running out of ideas. I guess it stores all values in the mysql database. But how do i make it write the configfiles? Sorry for my english, my na |
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| 10:56.19 | Aurs | addeswe: I think you should try that question in #freepbx |
| 10:58.35 | addeswe | Aurs : Never crossed my mind. Thanks for the tip! :-) |
| 10:59.13 | Aurs | np! |
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| 11:06.13 | oryxtec | hi all |
| 11:06.28 | oryxtec | on asterisk CLI i am getting these error msgs |
| 11:06.29 | oryxtec | [Jul 22 07:00:59] ERROR[4555]: chan_sip.c:15359 sipsock_read: We could NOT get the channel lock for SIP/sipdialers-08224dd0! |
| 11:06.30 | oryxtec | [Jul 22 07:00:59] ERROR[4555]: chan_sip.c:15360 sipsock_read: SIP transaction failed: 73f7b70b7258f6ee607f35ae72601638@109.169.28.5 |
| 11:06.34 | oryxtec | please any help on this |
| 11:06.35 | oryxtec | ? |
| 11:09.06 | oryxtec | help!!!!!!!! plz on this |
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| 12:00.11 | garymc | Hi, I know this may be a silly question but i will ask it anyway. |
| 12:00.23 | garymc | I have isdn30 with 8 channels |
| 12:00.54 | garymc | does this mean I can have one call coming in and 7 waiting in the que at max. Or can i still get 30 in the que? |
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| 12:03.21 | fibres | Hi all. |
| 12:03.34 | telnettech | garymc...... if you have 8 Voice channels then that is all the calls you can have..... |
| 12:03.43 | fibres | Does anyone know a way to block a certain ip range from accessing my asterisk server? I have someone trying to hack my asterisk. |
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| 12:04.39 | garymc | How many channels of voip can one have? |
| 12:05.27 | garymc | cos im looking at expanding my call center to 10 staff and im going to need possibly 60 channels to handle incoming calls |
| 12:05.38 | garymc | thats so people can que |
| 12:05.39 | drmessano | Depends on bandwidth, and if the plan you have with your ITSP is limited to a specific number of channels |
| 12:05.50 | garymc | and wait to get through to a staff clerk |
| 12:06.09 | Chainsaw | If any transcoding is required, the amount of CPU power on the Asterisk server comes into play. |
| 12:06.14 | drmessano | Unlimited plans tends to limit to a specific number of channels |
| 12:06.23 | Chainsaw | But if not, it's indeed just down to what your ITSP contract specifies. |
| 12:06.42 | drmessano | But metered is generally "if you have money and calls, we can help you spend it" |
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| 12:07.07 | garymc | cheers |
| 12:07.18 | garymc | theres nothing more I like more than spending money |
| 12:07.33 | drmessano | Gotta spend money to make money |
| 12:10.41 | coppice | but I mostly need to make money to replace what my family spends |
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| 12:40.14 | Jinxed- | how easy would it be to implement voice mail with asterisk in an existing voip system? |
| 12:41.28 | [TK]D-Fender | Jinxed-: meaning? |
| 12:41.40 | BarthezZ | you mean asterisk as a voicemail for an other voip system? |
| 12:41.48 | [TK]D-Fender | Jinxed-: * VM is * VM. |
| 12:42.04 | [TK]D-Fender | Jinxed-: * gets a call. Yuo call voicemail. The end. |
| 12:42.42 | garymc | [TK]D-Fender : Is there a way i can stop a missed call showing on other extensions that didnt answer a call that was answered on another extension. As it shows a missed call that wasnt actually missed? |
| 12:43.02 | BarthezZ | garymc depends on the phone you are using |
| 12:43.06 | [TK]D-Fender | ^^^ |
| 12:43.07 | garymc | im using polycom phones |
| 12:43.14 | [TK]D-Fender | garymc: Go read the admin guide. |
| 12:43.16 | BarthezZ | read the docs to check for the feature :) |
| 12:43.45 | garymc | ok you know if its able to do so, may save me an hour or so reading through to find its not possible |
| 12:44.01 | garymc | what would the feater be listed as? |
| 12:44.06 | garymc | *feature |
| 12:44.58 | [TK]D-Fender | garymc: Perhaps you should consider using the powerful text search options in most PDF readers.... |
| 12:45.09 | [TK]D-Fender | garymc: and look for the obvious ones |
| 12:45.20 | garymc | hmmm didnt know that i could do that |
| 12:45.21 | *** join/#asterisk FILLVAIO3 (~v_agarkov@79.165.89.20) |
| 12:45.29 | FILLVAIO3 | Hello people! |
| 12:46.13 | FILLVAIO3 | Does anybody know good visual cdr log analizer/collector? |
| 12:46.39 | [TK]D-Fender | FILLVAIO3: MS Excel |
| 12:48.06 | Jinxed- | BarthezZ: correct asterisk as a voicemail system for another voip system |
| 12:49.02 | BarthezZ | Jinxed- shouldn't be hard, create the extensions in asterisk to just go to voicemail directly, and on the other voip system after $timeout transfer the call to the extension on asterisk, done :P |
| 12:49.12 | [TK]D-Fender | Jinxed-: I still don't see what this other system has to do with * in any special way. Voicemail is voicemail. What makes your need SPECIAL? |
| 12:49.39 | [TK]D-Fender | Jinxed-: Send the call to *. Dump into Voicemail. THE END. |
| 12:49.47 | FILLVAIO3 | [TK]D-Fender: but what about web interface for cdr? I have find many of this, but what is the best, i don't know. And i don't have a chance to install all of them. |
| 12:50.00 | [TK]D-Fender | FILLVAIO3: Areski seems decent |
| 12:50.05 | garymc | anyone give me a tip as what to input in the search field for my problem? |
| 12:50.09 | Jinxed- | [TK]D-Fender: Sorry... im just asking.... I have never used * before |
| 12:51.09 | [TK]D-Fender | garymc: how about words like MISSED <--- geez.... |
| 12:51.28 | garymc | yeah its not giving me much <--------- geeezus |
| 12:52.28 | [TK]D-Fender | garymc: Guess after looking up about 3-4 key works you should be able to either find it, or conclude that it does not exist |
| 12:52.34 | [TK]D-Fender | wrods* |
| 12:52.35 | FILLVAIO3 | [TK]D-Fender: thanx a lot! Do you think this is the best of all? |
| 12:52.52 | garymc | ok im looking at something here, not quite sure how Id go about tit though |
| 12:52.58 | [TK]D-Fender | FILLVAIO3: Not yet. That means I'd have to have TRIED them all |
| 12:53.48 | garymc | is this what i should be looking at? : Specify per-registration whether all missed-call events or only |
| 12:53.49 | garymc | remote/server-generated missed-call events will be displayed |
| 12:54.14 | garymc | am i on the right track here? |
| 12:54.24 | [TK]D-Fender | garymc: Servers don't tell phones that they missed calls. |
| 12:54.32 | [TK]D-Fender | garymc: completely backwards |
| 12:54.37 | garymc | ok |
| 12:54.40 | garymc | :( |
| 12:55.04 | [TK]D-Fender | garymc: Phone sees a call. You don't answer and the phone say YOU MISSED ANOTHER ONE |
| 12:55.46 | garymc | yes but i have some phone lines set to ring three phones. So if one person answers the call another phone shows it as missed, but it wasnt missed |
| 12:56.08 | Jinxed- | BarthezZ: is * just for linux |
| 12:56.08 | garymc | this is really starting to piss people off in the office |
| 12:57.14 | [TK]D-Fender | garymc: https://issues.asterisk.org/view.php?id=16928 |
| 12:58.07 | tzafrir_laptop | Jinxed-, Asterisk is most commonly used on Linux. It is also known to run on some other systems, such as FreeBSD |
| 13:00.37 | *** join/#asterisk mmlj4 (~jkelly@ip70-171-94-246.no.no.cox.net) |
| 13:00.47 | garymc | [TK]D-Fender : Im looking at your link and it seems a bit jumbled to me. What exactly does this tell me to do? |
| 13:00.57 | Jinxed- | tzafrir_laptop: how compatible is it with ubuntu? |
| 13:01.20 | garymc | I can see there is a patch file. But I dont know how to install it or configure it |
| 13:01.42 | [TK]D-Fender | Jinxed-: fine |
| 13:01.42 | tzafrir_laptop | Jinxed-, Ubuntu is a Linux distribution |
| 13:03.30 | garymc | Oh well im no nearer to fixing this issue. I need some plain english guidance |
| 13:03.56 | garymc | step by step would be nice but......... |
| 13:05.32 | *** join/#asterisk Fefeu (~Felipe@unaffiliated/fefeu) |
| 13:07.03 | garymc | OK TK where would I place this patch file to make it work? |
| 13:07.24 | [TK]D-Fender | 09:00]<Jinxed->tzafrir_laptop: how compatible is it with ubuntu? <- Ubuntu = Linux. Asterisk runs on linux. |
| 13:07.56 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:09.53 | [TK]D-Fender | garymc: It refers to something you should be LOOKING AT |
| 13:10.25 | garymc | im looking for a folder called apps on my asterisk server but cant find it. Am i on the right track |
| 13:11.21 | tzafrir_laptop | garymc, do you see the link to 'wget patch'? |
| 13:12.15 | tzafrir_laptop | Add ' --dry run' to the end of the command-line it gives you, just to be on the safe side |
| 13:12.32 | [TK]D-Fender | tzafrir_laptop: He doesn't actually NEED it... this is something for him to read to see SOMETHING ELSE. |
| 13:13.24 | oryxtec | hi all... when i try to make manual call i get this error msg |
| 13:13.25 | oryxtec | [Jul 22 09:12:14] ERROR[4555]: chan_sip.c:15359 sipsock_read: We could NOT get the channel lock for SIP/sipdialers-0823dd00! |
| 13:13.25 | oryxtec | [Jul 22 09:12:14] ERROR[4555]: chan_sip.c:15360 sipsock_read: SIP transaction failed: 1a52429320121c0f685931f50bcb3b3a@x.x.x.x |
| 13:13.25 | oryxtec | [Jul 22 09:12:14] ERROR[4555]: chan_sip.c:15359 sipsock_read: We could NOT get the channel lock for SIP/sipdialers-0823dd00! |
| 13:13.25 | oryxtec | [Jul 22 09:12:14] ERROR[4555]: chan_sip.c:15360 sipsock_read: SIP transaction failed: 1a52429320121c0f685931f50bcb3b3a@x.x.x.x |
| 13:13.29 | tzafrir_laptop | ~pb |
| 13:13.30 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 13:13.35 | oryxtec | please could any one guide me |
| 13:13.35 | [TK]D-Fender | oryxtec: No NOT flood in here |
| 13:13.40 | oryxtec | sorry |
| 13:13.54 | [TK]D-Fender | oryxtec: What version of * are you using exactly? |
| 13:14.11 | oryxtec | 1.4.16.1 |
| 13:14.25 | [TK]D-Fender | oryxtec: ANCIENT. Loaded with bugs which this is likely one of |
| 13:14.26 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 13:14.32 | [TK]D-Fender | oryxtec: UPGRADE |
| 13:15.09 | oryxtec | i was having same issue on asterisk 1.2 soo i upgrade it 1.4 but still same soo u think this is a bug |
| 13:15.14 | oryxtec | which ver should i use |
| 13:15.34 | garymc | yzafir i do see that link |
| 13:15.38 | garymc | tzafir |
| 13:15.41 | leifmadsen | oryxtec: uhhh.... 1.4.34 is coming out today |
| 13:15.46 | [TK]D-Fender | ^^ |
| 13:15.48 | leifmadsen | 1.4.16 is like... 2 years old or something? |
| 13:15.55 | oryxtec | humm |
| 13:16.04 | [TK]D-Fender | oryxtec: 18 releases old <- |
| 13:16.06 | leifmadsen | how did you even decide on that version? |
| 13:16.09 | garymc | after runnning that command do i need to do anything else? |
| 13:16.13 | oryxtec | from where i can download 1.4.34? |
| 13:16.14 | [TK]D-Fender | oryxtec: and that is just in that BRANCH |
| 13:16.17 | oryxtec | can u pass me link |
| 13:16.24 | leifmadsen | oryxtec: have you tried looking at http://www.asterisk.org ? |
| 13:16.24 | [TK]D-Fender | oryxtec: my guess.... www.asterisk.org |
| 13:16.37 | oryxtec | let me check |
| 13:16.39 | leifmadsen | like i said, 1.4.34 is coming out today, which implies it is not yet out |
| 13:16.43 | [TK]D-Fender | Quick... who makes Microsoft Excel? |
| 13:16.45 | leifmadsen | 1.4.33.1 is out though |
| 13:16.50 | leifmadsen | [TK]D-Fender: Linus! |
| 13:16.52 | [TK]D-Fender | What colour was Napoleon's white horse!??! |
| 13:16.58 | leifmadsen | [TK]D-Fender: Black! |
| 13:16.59 | [TK]D-Fender | WHO SHOT J.R>?! |
| 13:17.05 | leifmadsen | [TK]D-Fender: you totally did it |
| 13:17.09 | [TK]D-Fender | LIES |
| 13:20.45 | garymc | tzafrir_laptop I ran that command in putty on my asterisk server and It says unknown bash command etc |
| 13:23.23 | Naikrovek | WIMPy: polycom conversation last night: yes polycom phones are awesome |
| 13:23.40 | mmlj4 | or not |
| 13:23.46 | Naikrovek | i'm replacing every non-polycom phone in the house with the polycom variety |
| 13:23.58 | mmlj4 | the things feel flimsy in my hand |
| 13:24.01 | Naikrovek | there's no "or" here. Polycom > all |
| 13:24.40 | Katty | herroes |
| 13:24.49 | [TK]D-Fender | Just for one day... |
| 13:25.17 | garymc | [TK]D-Fender : i put the right command in now. It now says "File to Patch:" what do i do here? |
| 13:25.30 | [TK]D-Fender | garymc: NOTHING. |
| 13:25.39 | Katty | mmlj4: buy something you're happy with. |
| 13:25.40 | garymc | do i just hit enter? |
| 13:25.48 | [TK]D-Fender | [09:12]<[TK]D-Fender>tzafrir_laptop: He doesn't actually NEED it... this is something for him to read to see SOMETHING ELSE. |
| 13:26.05 | garymc | god damn |
| 13:26.08 | Katty | mmlj4: for most of us it's been polycom. but you're entitled to your own opinion too. |
| 13:26.34 | Naikrovek | he's entitled to his own opinions, but not his own facts. |
| 13:26.38 | *** join/#asterisk Jinxed- (93b138ea@gateway/web/freenode/ip.147.177.56.234) |
| 13:26.39 | Naikrovek | fact is, polycom > all |
| 13:26.41 | Naikrovek | :P |
| 13:26.51 | [TK]D-Fender | \Everyone is entitled to my own opinion ;) |
| 13:27.00 | Naikrovek | heh |
| 13:27.06 | Katty | Naikrovek: oh shush. |
| 13:27.22 | drmessano | Per the ticket, if he actually READ it, it points to the 'c' option for DIAL.. the patch was shown to be useless because the 'c' option was doing what it was supposed to, which happens to be a demo of EXACTLY what garymc needs |
| 13:27.29 | Jinxed- | how easy is it to remove *b from ubuntu if you don't like it |
| 13:27.44 | [TK]D-Fender | drmessano: Oh sure... jsut #&*$^&#$^ HAND it to him... |
| 13:27.45 | drmessano | Jinxed-: rm -Rf /* |
| 13:27.57 | Jinxed- | drmessano: lol... |
| 13:28.07 | drmessano | [TK]D-Fender: He couldnt find it if you slapped him with it |
| 13:28.19 | mmlj4 | Katty: I do |
| 13:28.20 | drmessano | [TK]D-Fender: May as well, or this could go on for hours |
| 13:28.21 | Jinxed- | nah i'll skip any type of rm -rf |
| 13:28.22 | [TK]D-Fender | drmessano: True. But what does that have to do with this? ;) |
| 13:28.34 | [TK]D-Fender | drmessano: MY SYSTEM AM SCREW! |
| 13:28.38 | Katty | infobot: crittercam |
| 13:28.39 | infobot | crittercam is probably Katty's live broadcast of The Nut House at http://ustre.am/8H5d |
| 13:28.42 | mmlj4 | linksys (now it's cisco) |
| 13:28.43 | drmessano | Exactly! |
| 13:29.15 | garymc | drmessano so how do i do it? |
| 13:29.17 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 13:29.24 | [TK]D-Fender | drmessano: Dear God.. |
| 13:29.34 | [TK]D-Fender | drmessano: Can't even lead this horse to water... |
| 13:29.51 | [TK]D-Fender | is still pretty sure he can hold its head under though... |
| 13:30.05 | Katty | so much negativity this morning |
| 13:30.18 | Naikrovek | "crittercam" that's what I call C-SPAN. Congresscritters. |
| 13:30.22 | [TK]D-Fender | Katty: You POSITIVE of that? ;) |
| 13:30.43 | drmessano | garymc: 'c' option for Dial |
| 13:30.50 | Katty | well i'm still hopeful people are going to be nice. |
| 13:30.54 | Katty | even if i'm the only one (= |
| 13:31.01 | drmessano | garymc: Thats what ALL OF THIS is pointing you to |
| 13:31.05 | garymc | how do i do that drmessano |
| 13:31.12 | [TK]D-Fender | spins up some Eric Carmen jsut for Katty |
| 13:31.23 | [TK]D-Fender | drmessano: FAIL |
| 13:32.57 | garymc | come on guys how do i implement the 'c' option |
| 13:32.58 | drmessano | garymc: One of the examples in the ticket shows you. You're making this impossibly difficult and the example is RIGHT THERE |
| 13:33.10 | drmessano | It's IN THE TICKET he LINKED YOU |
| 13:33.33 | garymc | yeah im looking but im not as clever as you lot |
| 13:33.52 | drmessano | There's no 'CLEVER' here... you want to know how to do it.. it's THERE |
| 13:34.40 | oryxtec | [TK]D-Fender: i ve upgrade asterisk to 1.4.33.1... which latest i found on asterisk.org |
| 13:34.49 | oryxtec | now i m getting this error msg ERROR[4674]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe |
| 13:34.58 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 13:35.26 | oryxtec | and ERROR[5141]: utils.c:968 ast_carefulwrite: write() returned error: Connection reset by peer |
| 13:35.43 | leifmadsen | oryxtec: means audio was playing when the call was hung up |
| 13:35.49 | leifmadsen | I see that sometimes |
| 13:35.53 | oryxtec | humm |
| 13:36.00 | leifmadsen | what is the actual problem? |
| 13:36.02 | oryxtec | :S |
| 13:36.44 | garymc | Im looking at the ticket and nothing there tells me where the C option is located.. |
| 13:36.48 | oryxtec | the problem is b4 i was getting some error msgs coz that my asterisk was giving me ltos on pro. |
| 13:37.01 | oryxtec | soo i upgrade it to 1.4.33.1 |
| 13:37.18 | oryxtec | now those msgs are gone but this one are now showing up on asterisk cli |
| 13:37.33 | garymc | and how i turn it on or off |
| 13:37.43 | Katty | hugs leifmadsen |
| 13:38.04 | leifmadsen | hugs Katty back |
| 13:38.23 | tzafrir_laptop | Jinxed-, aptitude purge asterisk #? |
| 13:38.27 | oryxtec | how can i fix this error msg |
| 13:38.39 | oryxtec | write() returned error: Broken pipe |
| 13:40.08 | Naikrovek | garymc: i want to avoid yelling at you but the 'c' option is shown right there on that asterisk issues page |
| 13:40.13 | Naikrovek | https://issues.asterisk.org/view.php?id=16928 |
| 13:40.28 | Naikrovek | observe this part: Dial(SIP/2024,30,c) |
| 13:40.34 | Naikrovek | note the 'c' |
| 13:40.43 | Naikrovek | now ask again where the 'c' option is |
| 13:40.45 | Naikrovek | please |
| 13:40.47 | Naikrovek | :| |
| 13:41.00 | garymc | yes ok... maybe it cos im running freepbx ? |
| 13:41.05 | garymc | im confused |
| 13:41.08 | Naikrovek | ............... |
| 13:41.14 | Naikrovek | shoots himself in the brain |
| 13:41.17 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:41.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:41.27 | Jinxed- | is the best way to install * for ubuntu just from the built in software center? |
| 13:41.40 | Naikrovek | Jinxed-: define "best way" |
| 13:41.41 | Katty | Jinxed-: i prefer from the tarballs. |
| 13:41.43 | garymc | I woulda done that for you naikrovec |
| 13:41.59 | Jinxed- | Ok best way for someone who has never touched * |
| 13:42.07 | mmlj4 | apt-get install asterisk |
| 13:42.10 | mmlj4 | if it's there |
| 13:42.11 | Naikrovek | Jinxed-: does it HAVE to be ubuntu |
| 13:42.23 | Katty | Jinxed-: would you like me to pastebin some stuff for you? |
| 13:42.26 | Naikrovek | recommends AsteriskNOW for people wanting to try it out |
| 13:42.35 | mmlj4 | ok, so what's wrong with ubuntu? don't like group hugs? |
| 13:42.37 | Jinxed- | Well I already have it set up with other things |
| 13:42.43 | Naikrovek | ubuntu is fine |
| 13:42.48 | Naikrovek | i like ubuntu i use it everywhere |
| 13:43.05 | Jinxed- | also do i need to install gastman with asterisk? |
| 13:43.08 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 13:43.08 | Naikrovek | but AsteriskNOW as a distribution is better for just testing asterisk, I think. unless you REALLY don't want a gui for it |
| 13:43.32 | Jinxed- | Naikrovek: isn't gastman a gui? |
| 13:43.40 | Katty | Jinxed-: when i was first learning, i did it from tarballs. but as long as it's well documented i'd think just about any way would be suitable for someone new. just make sure there are lots of instructions. |
| 13:43.40 | Naikrovek | if you want asterisk and you do NOT want a gui, then yeah "apt-get install asterisk" should probably do the trick |
| 13:43.57 | Naikrovek | Jinxed-: the only asterisk GUI worth mentioning is FreePBX |
| 13:44.40 | Jinxed- | the voip system im trying to connect to was all command line |
| 13:44.50 | Jinxed- | so im not overally opposed to it, obviously a gui would be nice |
| 13:44.51 | Naikrovek | asterisk is all cmdline and config files |
| 13:45.02 | Naikrovek | vanilla asterisk, i should say |
| 13:45.23 | Naikrovek | if you plan to do a lot of pbx stuff or have complicated requirements freepbx probably won't get you where you want to go |
| 13:45.26 | Jinxed- | I was just going to try to see if I could convince others to take a serious look at * |
| 13:45.33 | Jinxed- | and I figured if I could get voice mail working |
| 13:45.36 | Jinxed- | for at least 1 phone |
| 13:45.37 | Katty | Jinxed-: a GUI is nice for end users to update a few things, like names associated with extensions... but i wouldn't use it as something to configure the bulk of your asterisk boxes. |
| 13:45.39 | Jinxed- | by a demo this afternoon |
| 13:45.41 | Naikrovek | Jinxed-: easy |
| 13:45.46 | Jinxed- | then it would be worth it |
| 13:45.56 | Naikrovek | Jinxed-: AsteriskNOW can have you up and working with voicemail in 30 minutes |
| 13:46.05 | oryxtec | voice is getting really bad... now these msgs are on asterisk cli channel.c:952 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/8600051@default-81e2,1 |
| 13:46.08 | oryxtec | plz help :( |
| 13:46.14 | Naikrovek | Jinxed-: you can have 1000 extensions set up in an hour |
| 13:46.18 | Katty | Jinxed-: if you'd like, i can pastebin my exact setup process. |
| 13:46.20 | Naikrovek | or less |
| 13:46.37 | Katty | Jinxed-: but it doesn't include configuration specifics. |
| 13:46.41 | *** part/#asterisk oej (~olle@ns.webway.se) |
| 13:46.54 | Naikrovek | oryxtec: bandwidth all used up? |
| 13:46.59 | Jinxed- | Katty: that would be nice |
| 13:47.14 | oryxtec | no |
| 13:47.19 | Jinxed- | Naikrovek: I really wish I could try it, but I don't have a comp that I can spare to put the distro on |
| 13:47.31 | Naikrovek | Jinxed-: virtual machine, then |
| 13:47.41 | oryxtec | naikrovek: i m using only one ext for dialing |
| 13:47.50 | oryxtec | and voice is really braking out |
| 13:47.52 | mmlj4 | Jinxed-: vmware |
| 13:48.10 | Naikrovek | oryxtec: what is the bandwidth between you and the place you're calling |
| 13:48.15 | *** join/#asterisk Mhaddog_Mac (~anonymous@adsl-32-170-204.mia.bellsouth.net) |
| 13:48.19 | *** join/#asterisk basty (~basty@212.218.65.249) |
| 13:48.20 | basty | Hi |
| 13:48.26 | Katty | hi basty |
| 13:49.04 | Naikrovek | Jinxed-: VMware player is free and will let you install AsteriskNOW without any extra hardware |
| 13:49.06 | oryxtec | 512 |
| 13:49.41 | oryxtec | 512 KB |
| 13:49.43 | Naikrovek | oryxtec: is anything else (other than IRC) happening over that link as you try to make a call |
| 13:49.43 | basty | I am trying to code a little agi script (in bash) for my asterisk. How can I tell asterisk to use the variable...the output of my script ? For right now I am trying something in bash with : "echo "SET VARIABLE test \"${CIDNAME}\"" <- but if I noop ${test} its empty |
| 13:49.55 | oryxtec | nop |
| 13:50.03 | Jinxed- | Naikrovek: I have never used vmware... what is it? |
| 13:50.47 | Naikrovek | Jinxed-: http://en.wikipedia.org/wiki/Operating_system-level_virtualization |
| 13:50.59 | Naikrovek | ah wait that's not right |
| 13:51.12 | Naikrovek | Jinxed-: http://en.wikipedia.org/wiki/Full_virtualization |
| 13:51.38 | Naikrovek | tl;dr - it allows you to run an operating system entirely within another operating system |
| 13:51.40 | oryxtec | by the i have asterisk installed on sun virtual machine |
| 13:51.47 | oryxtec | naikrovek:.. |
| 13:51.59 | Naikrovek | oryxtec: what kind of virtual machine; what hardware are you on |
| 13:52.29 | oryxtec | i ve server on 2.2 dual core 1gb ram |
| 13:52.37 | oryxtec | on that server i ve installed |
| 13:52.40 | Naikrovek | if you have some 32-bit cpu then that could cause issues. 64-bit hardware should be fine |
| 13:52.45 | oryxtec | 32 bit |
| 13:52.56 | Naikrovek | there's your issue |
| 13:52.58 | Naikrovek | probably |
| 13:53.02 | hrhrhr | what's the best way to route a wan trunk... is iax2 a measurably better performer than say... sip? |
| 13:53.14 | oryxtec | humm |
| 13:53.21 | oryxtec | wht should i do now |
| 13:53.21 | oryxtec | ? |
| 13:53.22 | Naikrovek | hrhrhr: for multiple calls, iax2 uses considerably less bandwidth than sip |
| 13:53.39 | Naikrovek | hrhrhr: and iax2 can be encrypted on a protocol level |
| 13:53.41 | Naikrovek | unlike sip |
| 13:53.43 | hrhrhr | thanks Naikrovek. is there anything better than iax2? |
| 13:53.58 | Naikrovek | hrhrhr: what does 'better' mean |
| 13:54.04 | hrhrhr | i need to route from .uk to .sg |
| 13:54.05 | *** join/#asterisk jhirley (~jhirley@c-75-74-13-194.hsd1.fl.comcast.net) |
| 13:54.11 | hrhrhr | just wondering the best way to go about it |
| 13:54.19 | Naikrovek | use iax2 |
| 13:54.28 | hrhrhr | alrighty |
| 13:54.38 | hrhrhr | now i just need to find a provider out there that supports it :D |
| 13:54.46 | Naikrovek | there are a few but not many |
| 13:54.52 | Naikrovek | SIP is widespread and very common |
| 13:55.18 | hrhrhr | can you quantify how much better it is for say, 5 calls? |
| 13:55.25 | hrhrhr | or even 2 |
| 13:55.31 | Naikrovek | minimal |
| 13:55.39 | leifmadsen | you need to do like 20+ |
| 13:55.42 | leifmadsen | or 10+ |
| 13:56.06 | Naikrovek | but will use less bandwidth than sip for any more than 1 simultaneous call. obviously 100 calls would save more bandwidth than just 2 |
| 13:58.30 | Naikrovek | http://i.imgur.com/gTNuM.jpg |
| 13:59.23 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 13:59.37 | *** join/#asterisk iCEBrkr (~icebrkr@cyberdyne.org) |
| 14:00.24 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 14:00.56 | *** join/#asterisk digitalml (digitalml@wsip-24-234-120-155.lv.lv.cox.net) |
| 14:07.41 | digitalml | im using asterisk and freepbx and would like to modify extensions_custom.conf for a custom outbound dial paln. i've created an extension (1000) and I've pointed my .call file that I'm placing the the outgoing directory to use this extension. the problem is i dont know how to specify the right call context in the extensions_custom.conf file to get used |
| 14:07.49 | digitalml | can anyone help please |
| 14:08.43 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 14:09.54 | *** join/#asterisk Arsenick (~y@modemcable230.231-70-69.static.videotron.ca) |
| 14:11.27 | Jinxed- | Naikrovek: any tut on how to go about intalling *now within vmware? |
| 14:12.22 | [TK]D-Fender | digitalml: So what do you want on each side of this call? |
| 14:12.37 | [TK]D-Fender | Jinxed-: Its an ISO. You install it like any other |
| 14:12.49 | [TK]D-Fender | Jinxed-: Or... you could jsut install on the Ubuntu you already have |
| 14:13.06 | digitalml | [TK]D-Fender: im just trying to place an outbound call with some pre recorded messages and wait() inbetween |
| 14:13.22 | Jinxed- | [TK]D-Fender: that is waht Im planning on doing is just installing vmware and then trying to put *now on that |
| 14:13.30 | [TK]D-Fender | digitalml: You want to call out using FreePBX's normal call processing rules? |
| 14:13.41 | Jinxed- | how much space do i need for *now |
| 14:13.45 | [TK]D-Fender | Jinxed-: What do you want to do with *? |
| 14:13.50 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 14:14.05 | [TK]D-Fender | Jinxed-: And is your Ubuntu system running all the time you need your * system running? |
| 14:14.13 | Jinxed- | Right now I just want voicemail for an existing voip demo system (not* call manager) |
| 14:14.30 | Jinxed- | [TK]D-Fender: it is just for a demo, so no |
| 14:14.34 | digitalml | [TK]D-Fender i think so yes |
| 14:14.38 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:14.56 | Jinxed- | my goal is to show that * is worth looking into a bit |
| 14:15.18 | [TK]D-Fender | digitalml: Channel: Local/thenumberasyouwoulddialit@from-internal/n |
| 14:15.18 | Jinxed- | but the demo is in ~3 hrs |
| 14:15.49 | digitalml | and then modify [from-internal] |
| 14:15.54 | digitalml | in extensions_custon.conf |
| 14:15.55 | [TK]D-Fender | Jinxed-: You've never installed * before and you expect to learn enough to even be processing calls and have VM's set up.... your odds are frighteningly low |
| 14:15.58 | digitalml | ? |
| 14:16.16 | [TK]D-Fender | digitalml: I jsut gave you the CHANNEL line for your call file |
| 14:16.35 | [TK]D-Fender | digitalml: The rest is normal as to where you DUMP them onece they answer |
| 14:16.56 | [TK]D-Fender | digitalml: context, extension, priority <- |
| 14:17.14 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 14:18.09 | Jinxed- | [TK]D-Fender: [09:46] <Naikrovek> Jinxed-: AsteriskNOW can have you up and working with voicemail in 30 minutes |
| 14:18.19 | Naikrovek | Jinxed-: install vmware. reboot. download asterisknow ISO image. create new virtual machine in vmware, tell it to boot to the ISO you downloaded. observe as you install an operating system inside another operating system |
| 14:18.42 | [TK]D-Fender | Jinxed-: There is a difference between POSSIBLE, and "can do this knowing NOTHING" |
| 14:18.47 | Naikrovek | Jinxed-: set up the virtual machine to use BRIDGED networking (default is NAT) |
| 14:19.56 | Naikrovek | Jinxed-: get the IP address of the virtual machine, then http://ip.add.re.ss/ and proceed in setting up an extension and turning voicemail on |
| 14:19.57 | Naikrovek | done |
| 14:19.57 | Jinxed- | haha sounds good Naikrovek |
| 14:19.57 | Jinxed- | I will update as I go |
| 14:19.57 | [TK]D-Fender | Jinxed-: Yes, I could get a full system running in under an hour, but thats experience. And to get it running, boxes configured, etc... FreePBX wasn't meant to have generic boxed normally either. handling GENERIC SIP calls? oh boy. We don't even know WHAT they will look like coming from your system. That is a lot of testing right there. |
| 14:20.04 | Naikrovek | this probably isn't #asterisk fare, keep me update in #asterisknow |
| 14:20.54 | [TK]D-Fender | Naikrovek: You know how to setup a VM-only "extension" in FreePBX? |
| 14:21.29 | Naikrovek | uhh, set up an extension, turn voicemail on, don't connect a phone to it :) |
| 14:21.32 | Naikrovek | other than that, no |
| 14:21.43 | [TK]D-Fender | Naikrovek: Or are you suggesting making an "extension" that no-one will actually log into and that's all? |
| 14:21.50 | [TK]D-Fender | Naikrovek: Guess so. FUGLY |
| 14:21.55 | Naikrovek | well |
| 14:22.02 | Naikrovek | i never claimed to be an expert |
| 14:22.11 | Naikrovek | if i do it the wrong way i do it the wrong way |
| 14:22.34 | [TK]D-Fender | Naikrovek: FreePBX is the completely wrong way :) |
| 14:22.42 | Naikrovek | well for now i'm stuck with it |
| 14:22.52 | [TK]D-Fender | Naikrovek: for HIM. |
| 14:23.10 | [TK]D-Fender | Naikrovek: You... well it may be fine for you. Are you actually using it like it's MEANT to be used? |
| 14:23.12 | Naikrovek | he has 3 hours |
| 14:23.24 | Naikrovek | just tryin to help |
| 14:23.43 | Naikrovek | christ when will my receptionist learn to use the effing phone |
| 14:23.45 | [TK]D-Fender | [10:23]<Naikrovek>he has 3 hours <- he shouldn't leave things to teh last minute and set unrealistic goals |
| 14:23.50 | Naikrovek | [TK]D-Fender: true |
| 14:24.27 | [TK]D-Fender | Naikrovek: So, Are you using FreePBX for the things it is meant to do? Generic phone setup, route calls from normal resources in normal ways, etc? |
| 14:24.39 | Naikrovek | yeah |
| 14:24.51 | Naikrovek | i don't have any thing spectacular going on |
| 14:25.47 | [TK]D-Fender | Naikrovek: Then its right for you. It isn't a tool to try and implement just back-end stuff. that creats SIP users it shouldn't, etc.. and there is likely a bit of extra trickery jsut to get away with this context-wise depending on how his existing system even hands off the call to *. |
| 14:25.50 | *** join/#asterisk edguy3 (~edguy@ool-43521c56.dyn.optonline.net) |
| 14:25.55 | [TK]D-Fender | Naikrovek: I'm betting this gets ugly, fast |
| 14:26.20 | *** join/#asterisk jpeeler (~jpeeler@asterisk/developer/jpeeler) |
| 14:27.18 | Naikrovek | probably |
| 14:27.35 | [TK]D-Fender | grabs a soda and a medium popcorn and sits down for the show. |
| 14:28.31 | Naikrovek | why not large? long movie |
| 14:28.56 | Naikrovek | i'm not good at regular asterisk stuff. i've never done it, mostly |
| 14:29.16 | Naikrovek | so my direction was to help in a way i knew how |
| 14:29.31 | Naikrovek | that would meet his stated goals within the stated timeframe |
| 14:29.33 | Naikrovek | that's it |
| 14:29.43 | Naikrovek | any requirements or goals beyond those are not considered |
| 14:29.47 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 14:30.14 | [TK]D-Fender | Naikrovek: Except... given his goals and circumstances aren't normal he'll likely spend more time fighting with FreePBX just to get a minimal broken idea working.... |
| 14:30.22 | [TK]D-Fender | grabs another fistful of popcorn |
| 14:31.07 | Naikrovek | maybe, but maybe not |
| 14:31.14 | Naikrovek | maybe he'll get it working |
| 14:31.23 | Naikrovek | maybe his demo will be successful |
| 14:31.28 | Naikrovek | maybe it won't |
| 14:31.30 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 14:31.44 | Naikrovek | but i've given him all he needs - some google mixed in and he'll be there |
| 14:31.52 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:31.53 | Naikrovek | i don't see what the problem is, really |
| 14:31.59 | *** join/#asterisk iCEBrkr (~icebrkr@cyberdyne.org) |
| 14:32.04 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 14:32.05 | Naikrovek | maybe it's not the IDEAL solution but whatever |
| 14:32.19 | Naikrovek | what percentage of solutions even approach ideal |
| 14:32.27 | [TK]D-Fender | Naikrovek: LOTS :) |
| 14:32.39 | Naikrovek | it's a freaking demo |
| 14:32.48 | Naikrovek | it's not even going to make or receive calls |
| 14:32.57 | [TK]D-Fender | Naikrovek: Plain * would take SO much less time to set up that FreePBX for this technically... |
| 14:33.03 | Naikrovek | then tell him how to do it |
| 14:33.05 | Naikrovek | i don't know how |
| 14:33.11 | Naikrovek | i have the desire to help, you have the know how |
| 14:33.18 | [TK]D-Fender | Naikrovek: Assuming being at least slightly familiar with it |
| 14:33.44 | Naikrovek | well i can write a custom java network server in an hour, and when people need help with it i help |
| 14:33.45 | *** join/#asterisk iCEBrkr (~icebrkr@cyberdyne.org) |
| 14:33.52 | Naikrovek | not saying you don't help |
| 14:33.55 | Naikrovek | but i WANT to help |
| 14:33.57 | Naikrovek | so i helped |
| 14:34.48 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 14:34.50 | Naikrovek | nevermind |
| 14:35.04 | Naikrovek | you finish typing your witty retort, hit enter and i'll ignore it and we can move on |
| 14:35.23 | SuPrSluG | anyone heard of an issue with mwi in asterisk 1.6.2? |
| 14:35.40 | Naikrovek | Katty: can you send me that config doc you offered whats-his-nose earlier |
| 14:35.41 | SuPrSluG | no notify messages are sent |
| 14:39.10 | *** join/#asterisk iCEBrkr (~icebrkr@cyberdyne.org) |
| 14:39.11 | *** join/#asterisk sulex (~sulex@office.blindata.ch) |
| 14:44.05 | *** join/#asterisk joobie (~joobie@CPE-124-181-130-3.vic.bigpond.net.au) |
| 14:44.44 | sulex | hello. Card: Digium TE410P; Kernel: Linux ubuntu 2.6.32-21-server; libpri: 1.4.11.3; dahdi: dahdi-linux-complete-2.3.0.1+2.3.0. After the knightrider blinking the leds turn off, all the module are loaded but no PRI cable are attached to the card. The strange thing that is happening is that dahdi_tool reports all the four spans as "OK". I already called Digium but they say they will not give me support on that card because it's out of warranty. Question |
| 14:44.44 | sulex | <PROTECTED> |
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| 14:46.18 | *** join/#asterisk Infin1ty (~Infin1ty@pdpc/supporter/active/infin1ty) |
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| 14:47.01 | digitalml | [TK]D-Fender: you around? |
| 14:47.52 | [TK]D-Fender | digitalml: Yes |
| 14:48.41 | [TK]D-Fender | sulex: You have no cable attached. What is that actual PROBLEM here? |
| 14:49.22 | joobie | [TK]D-Fender, get with it bro - cable is old school ;P |
| 14:49.48 | digitalml | [TK]D-Fender: so in my .call file i have: Channel: SIP/trunkname/numberToCall Extension: 100 Context: testcontext |
| 14:50.10 | digitalml | and in my extensions_custon.conf i have a [testcontext] section |
| 14:50.21 | digitalml | with a bunch of exten => crap |
| 14:50.32 | digitalml | the call gets placed |
| 14:50.46 | [TK]D-Fender | digitalml: ok.... |
| 14:50.59 | digitalml | but all i hear is a "goodbye" instead of whats in my extensions_custom.conf |
| 14:51.08 | digitalml | like something went wrong? |
| 14:51.26 | digitalml | any ideas? |
| 14:51.31 | [TK]D-Fender | digitalml: I guess so. Maybe you should LOOK at the call file and the CALL, and your configs. |
| 14:52.09 | digitalml | what do you mean, i am looking at them |
| 14:52.13 | digitalml | but i cant see what is wrong |
| 14:52.24 | *** join/#asterisk genin (~phrame@ANice-252-1-74-55.w83-201.abo.wanadoo.fr) |
| 14:52.50 | *** part/#asterisk genin (~phrame@ANice-252-1-74-55.w83-201.abo.wanadoo.fr) |
| 14:52.59 | [TK]D-Fender | digitalml: What do you expect us to say? WE can't see it. |
| 14:53.07 | [TK]D-Fender | digitalml: PASTEBIN is your friend <- |
| 14:53.09 | [TK]D-Fender | ~pb |
| 14:53.10 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 14:53.12 | [TK]D-Fender | ^^^^^^^^ |
| 14:53.18 | joobie | i just realised |
| 14:53.24 | joobie | u have a lot of patience tk |
| 14:53.42 | [TK]D-Fender | joobie: Yeah, I haven't killed anyone in DAYS.... |
| 14:53.50 | joobie | heh |
| 14:53.59 | sulex | [TK]D-Fender: with no cable attached, I'm supposed to have the span allarmed or at least not shown as OK in dahdi_tool. with asterisk turned on and cable attahced I'm getting a D-channel not available. of course if i attach those E1 to another box(as5350 from cisco) everything is ok. the point is that the card before the dist-upgrade was working great. so this is why I ask |
| 14:54.03 | leifmadsen | he likes his shift key though :_) |
| 14:54.44 | [TK]D-Fender | leifmadsen: INDEED! I don't do any of this lazy-ass caps-lock BS either. No, I hold it down MYSELF! |
| 14:55.07 | [TK]D-Fender | leifmadsen: Cause these peeps.. they're like... TOTALLY worth it you know ;) |
| 14:56.33 | leifmadsen | [TK]D-Fender: snap :) |
| 14:58.48 | WIMPy | Can someone tell me what CCSS means? |
| 14:59.43 | pabelanger | WIMPy: Context? |
| 15:00.12 | WIMPy | Found it on a slide from a talk about new features in 1.8. |
| 15:00.24 | WIMPy | http://www.amoocon.de/archives/pictures/1417/original/slide-15.png?1276023498 |
| 15:00.54 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 15:02.01 | garymc | what command do i use to show what version of asterisk im using? |
| 15:06.13 | ChannelZ | core show version |
| 15:07.35 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 15:11.34 | digitalml | [TK]D-Fender: this is what isn't working... http://pastebin.com/4yJrf1MB |
| 15:12.49 | [TK]D-Fender | digitalml: I don't see the failed call... |
| 15:13.18 | [TK]D-Fender | DigiAnd I don't see you specifying the PRIORITY in your call-file either |
| 15:13.25 | pabelanger | WIMPy: Call Completion Supplementary Services |
| 15:13.46 | pabelanger | WIMPy: https://reviewboard.asterisk.org/r/523/ |
| 15:15.09 | *** join/#asterisk wcselby (~wcselby@208.180.112.123) |
| 15:16.28 | WIMPy | Ah. Thanks. |
| 15:16.56 | wcselby | o/ |
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| 15:28.48 | *** join/#asterisk BANSAL (~bansal@117.199.123.162) |
| 15:32.05 | kn0x | cdr mysql (addon) is inserting blank rows |
| 15:32.20 | kn0x | everything is default. Csv file has correct values |
| 15:32.33 | wcselby | are you getting any rows? |
| 15:32.37 | wcselby | i mean, rows with any data |
| 15:36.31 | kn0x | no data at all |
| 15:36.50 | [TK]D-Fender | kn0x: Cool. |
| 15:36.56 | *** join/#asterisk Da-Geek (~Da-Geek@87-194-2-213.bethere.co.uk) |
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| 15:38.31 | *** part/#asterisk Fefeu (~Felipe@unaffiliated/fefeu) |
| 15:38.43 | kn0x | [TK]D-Fender: not a bit :\ |
| 15:39.15 | [TK]D-Fender | kn0x: Feel free to do something to fix it then... |
| 15:40.23 | kn0x | thats why i brought it up in the channel... |
| 15:41.10 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 15:41.52 | geemee | Hey folks. Using a test server with AsteriskNOW 1.7 I cant get CDR to log any calls. Any ideas? |
| 15:43.11 | geemee | Its a fresh install. |
| 15:43.12 | *** join/#asterisk radic (~radic@dslb-094-216-253-214.pools.arcor-ip.net) |
| 15:44.26 | [TK]D-Fender | knkn0And didn't tell us what * version, what Addon's version, no call debug, no configs, no details on what backend actually.... |
| 15:44.45 | [TK]D-Fender | kn0x: When you feel like doing sonething to help people help you... let us know ;) |
| 15:45.43 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 15:46.25 | kn0x | asterisk 1.6.2.9 cdr_addon_mysql |
| 15:46.27 | kn0x | asterisk-addons-1.6.2.1 |
| 15:46.56 | radic | is there a way to synchronize the in and out -records? |
| 15:47.56 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 15:48.03 | geemee | kn0x: sure I installed the asterisk addons but not the cdr addon. Will give that a go. |
| 15:48.04 | kn0x | radic: not using ForkCDR() |
| 15:48.05 | asteriskATmarmuD | morning |
| 15:48.26 | radic | kn0x: I dosn't used it anywhere |
| 15:48.44 | kn0x | geemee: sorry, that wasnt to you.. |
| 15:48.48 | asteriskATmarmuD | how to check if a sip-phone is off/on hook |
| 15:48.53 | geemee | ah ok :) |
| 15:49.00 | kn0x | geemee: but what are you trying to do? |
| 15:49.06 | asteriskATmarmuD | with analog ones I got a message on the CLI in the past |
| 15:49.38 | geemee | essentially have a clean asterisknow install on test server but no call logs showing. Curious where to start. |
| 15:49.44 | Naikrovek | bmoraca_work: ping |
| 15:50.11 | pabelanger | asteriskATmarmuD: device hints |
| 15:50.27 | kn0x | csv ? check /etc/asterisk/cdr.conf |
| 15:50.31 | kn0x | and look at [csv] |
| 15:50.36 | *** join/#asterisk Joe_CoT (~joecot@pdpc/supporter/active/joe-cot) |
| 15:50.37 | geemee | Hi again Naikrovek :) |
| 15:50.41 | Naikrovek | hola |
| 15:50.42 | asteriskATmarmuD | <PROTECTED> |
| 15:50.46 | kn0x | should be in /var/log/asterisk/cdr-csv/ |
| 15:50.59 | bmoraca_work | what's up, Naikrovek ? |
| 15:51.08 | Naikrovek | bmoraca_work: can i /msg you about a cisco question |
| 15:51.15 | kn0x | [TK]D-Fender: and theres nothing on the debugs about the mysql cdr |
| 15:51.16 | Joe_CoT | Is there anyway to tell when a meetme recording starts, or when it ends, besides keeping track of everyone that joins and leaves and the meetmecount? |
| 15:51.24 | geemee | kn0x: ah.. file exists but is blank.. could explain it. |
| 15:51.33 | bmoraca_work | Naikrovek: I don't see why not...but i can't promise to answer :) |
| 15:51.40 | kn0x | what about cdr.conf |
| 15:51.42 | Naikrovek | fair enough |
| 15:52.16 | kn0x | [TK]D-Fender: cdr mysql status tells me its writing them.. the counter goes up |
| 15:52.27 | geemee | the file has nothing in it |
| 15:52.42 | pabelanger | asteriskATmarmuD: best you got is to setup a device hint (extension.conf) for the SIP connection. On/off hook events don't really exist for SIP. |
| 15:53.01 | asteriskATmarmuD | <PROTECTED> |
| 15:53.10 | geemee | kn0x: nothing in the cdr-csv either |
| 15:53.35 | pabelanger | ~hint |
| 15:53.52 | pabelanger | slaps infobot with a trout |
| 15:53.58 | pabelanger | asteriskATmarmuD: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 15:54.14 | kn0x | geemee: add |
| 15:54.15 | kn0x | [cdr] |
| 15:54.15 | kn0x | loguniqueid=yes ;log uniqueid |
| 15:54.16 | kn0x | loguserfield=yes ;log user field |
| 15:54.20 | asteriskATmarmuD | <PROTECTED> |
| 15:55.51 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 15:57.19 | geemee | kn0x: need to restart anything? |
| 15:59.31 | asteriskATmarmuD | <PROTECTED> |
| 15:59.40 | kn0x | geemee: module reload |
| 16:00.02 | asteriskATmarmuD | <PROTECTED> |
| 16:00.09 | wcselby | asteriskATmarmuD - hints are indeed in asterisk 1.4 |
| 16:00.35 | asteriskATmarmuD | wcselby: ok, first panic calmed |
| 16:00.36 | wcselby | asteriskATmarmuD - but like pabelanger stated, there's no real on-hook / off-hook with sip phones |
| 16:01.01 | asteriskATmarmuD | wcselby: can I get this info in some way? for example using hints |
| 16:01.28 | asteriskATmarmuD | wcselby: I need to know if a phone is on/off hook |
| 16:01.28 | wcselby | asteriskATmarmuD - not that I know of |
| 16:01.35 | asteriskATmarmuD | wcselby: great ;) |
| 16:01.43 | asteriskATmarmuD | wcselby: thx for the info |
| 16:02.35 | geemee | kn0x: hmmm still nothing.. sql appears ok.. |
| 16:02.41 | geemee | as in service running |
| 16:02.52 | geemee | looking through full logs |
| 16:03.43 | geemee | think I have found my problem : Failed to connect to mysql database |
| 16:05.13 | kn0x | geemee: that will only enable csv... |
| 16:05.37 | geemee | There is no CSV file also.. |
| 16:07.23 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
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| 16:19.15 | digitalml | can anyone help as to why i'm getting the error listed here please: http://pastebin.com/AjgYHMHn |
| 16:19.32 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-32-170-204.mia.bellsouth.net) |
| 16:20.14 | Joe_CoT | Is there anyway to tell when a meetme recording starts, or when it ends, besides keeping track of everyone that joins and leaves and the meetmecount? Or is there a way to tell whether asterisk is currently writing to a recording file? |
| 16:20.35 | *** join/#asterisk outtolunc (~me@c-98-248-105-248.hsd1.ca.comcast.net) |
| 16:21.04 | leifmadsen | digitalml: did you #include the correct file? |
| 16:21.27 | leifmadsen | did you reload the dialplan after modifying it? |
| 16:21.31 | leifmadsen | not sure what else it might be |
| 16:21.52 | digitalml | im using free pbx and it was my understanding that the extensions_custom.conf is already #included |
| 16:21.54 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-32-170-204.mia.bellsouth.net) |
| 16:22.10 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 16:22.44 | digitalml | i guess i wouldnt know hwo to reload the dialplan since i simply edited the conf file outside of freepbx |
| 16:23.22 | Joe_CoT | digitalml, from the asterisk shell, dialplan reload |
| 16:25.02 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 16:26.47 | *** join/#asterisk yonahw (~user@75.99.93.178) |
| 16:27.08 | digitalml | reloading the dial plan worked |
| 16:27.09 | digitalml | woo hoo |
| 16:27.10 | digitalml | thanks guys |
| 16:29.42 | *** part/#asterisk paparent (~paparent@unaffiliated/paparent) |
| 16:33.11 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:46.18 | *** join/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
| 16:50.37 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 16:51.24 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 16:53.25 | *** join/#asterisk Pritam (~chatzilla@unaffiliated/pritam) |
| 17:03.15 | *** join/#asterisk b14ck (~b14ck@dsl-lfkn-207-70-143-25.consolidated.net) |
| 17:03.18 | *** join/#asterisk gadams999 (~gadams999@173-165-184-27-atlanta.hfc.comcastbusiness.net) |
| 17:13.37 | LemensTS | sip show peer voipinnovations ( Codec Order : (g729:20,ulaw:20) ) |
| 17:13.46 | LemensTS | that does mean g729 is rated higher right? |
| 17:14.01 | LemensTS | it seems to go to ulaw no matter whatn unless i take ulaw out |
| 17:14.12 | LemensTS | then g729 works fine |
| 17:14.39 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 17:17.31 | timholum | good morning |
| 17:18.18 | andrebarbosa | LemensTS, what is the codec offered by the other end? |
| 17:18.22 | andrebarbosa | maybe is only g711 |
| 17:19.13 | LemensTS | andrebarbosa: ulaw by default, g729 if i request it in sdp |
| 17:20.57 | andrebarbosa | you need to enable sip debug, and see the logs |
| 17:21.11 | andrebarbosa | if one end offers only g711, asterisk will avoid transcodding |
| 17:21.23 | [TK]D-Fender | LemensTS: just because * preferes it doesn't mean its preference is what counts the most. |
| 17:32.14 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 17:36.51 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 17:45.36 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 17:49.13 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 18:20.10 | LemensTS | TKD-Fender: Voip-innovations says if I give g729 priority they will do that instead of g711. |
| 18:23.34 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 18:29.02 | philipp64|laptop | coppice: regarding the GSM v. GCC-4.2 stuff... I opened a new bug for that... 17688... but was wondering; did anyone ever investigate how the object code differed between GCC 4.2 and GCC 4.3? it can't be that different... you'd think that diff'ing disassembling listings would give a smoking gun. |
| 18:30.10 | coppice | people don't want proper answers. they remove the opportunity to whine and moan from a position ot ignorance |
| 18:31.27 | philipp64|laptop | ouch! |
| 18:32.16 | philipp64|laptop | well, I don't know enough about GSM encoding for the assembly to make much sense to me... and it's been a while since I stared at x86 instructions (I think DOS 3.0 was out at the time). |
| 18:32.31 | *** part/#asterisk RobH (~robh@wikimedia/RobH) |
| 18:33.34 | philipp64|laptop | our compiler tools team at Cisco has some slick tools that would "diff" two builds side by side and highlight how object changed. very handy for identifying regressions introduced by compiler version bumps. |
| 18:33.46 | philipp64|laptop | but I don't have access to any tools like that now. |
| 18:34.15 | philipp64|laptop | what about making the codec_gsm be built externally to asterisk against Spandsp? You seem to not be a fan of that idea. |
| 18:35.12 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 18:35.53 | coppice | looking at the different code only helps a little. it doesn't tell you which changes are just because the compilers work differently, and which are because the optimiser is being told the wrong thing. I expect there is a sea of code generation changes, which would make the error like a needle in a haystack |
| 18:36.14 | coppice | if you obey the licencing, you can do what you like |
| 18:42.04 | Katty | peers. |
| 18:44.37 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 18:44.39 | [sr] | hi |
| 18:46.16 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-96-224-34-135.nycmny.east.verizon.net) |
| 18:46.23 | KingDavidNYC | hello everybody |
| 18:46.45 | Katty | leifmadsen: ping. |
| 18:46.53 | Katty | KingDavidNYC: howdy. |
| 18:47.08 | pabelanger | Q: astctlowner in asterisk.conf, anybody give me the 30 word or less description? |
| 18:47.10 | Katty | leifmadsen: there is a certain pork tenderloin recipe i intend to weasel out of you. |
| 18:47.14 | raden_work | hugs katty !!!!!!!!!!!!!! |
| 18:47.21 | Katty | :> |
| 18:47.24 | Katty | HAI RADEN! |
| 18:47.26 | Katty | hugs raden_work |
| 18:47.31 | raden_work | where u been :P |
| 18:47.31 | wcselby | o/ |
| 18:47.36 | KingDavidNYC | hi Kathy |
| 18:48.00 | Katty | hugs wcselby |
| 18:48.08 | Katty | raden_work: i thought i told you |
| 18:48.13 | Katty | raden_work: or perhaps i'm loosing my mind |
| 18:48.18 | Katty | raden_work: single again? |
| 18:48.20 | Katty | raden_work: moved. |
| 18:48.23 | Katty | raden_work: etc. |
| 18:48.36 | raden_work | Katty, yes you did just never thought we'd see so lil of you : P |
| 18:48.47 | Katty | raden_work: well i had a lot going on :< |
| 18:48.51 | KingDavidNYC | can anybody help with a quick question in trixbox? I am making changes to the database, and then I reload in the CLI, but it doesn't seem to take the chages |
| 18:50.13 | Katty | KingDavidNYC: not too familiar with trixbox i'm afeered :< |
| 18:50.17 | mmlj4 | good luck getting an answer |
| 18:50.34 | KingDavidNYC | I need it |
| 18:50.40 | Naikrovek | ask in #trixbox |
| 18:50.48 | KingDavidNYC | cool |
| 18:50.55 | KingDavidNYC | thanks guys |
| 18:52.39 | leifmadsen | Katty: oh ya? :) |
| 18:55.39 | leifmadsen | Katty: Diana's BBQ sauce (any flavour -- I used Smoked Hickory I think). Brush on and marinate in fridge for 2-3 hours. BBQ on full heat -- sear each side for about 1-2 minutes each on the grill directly above the flame. Brush on more sauce then place on top rack of BBQ at 400F for 20 minutes rotating once (I did it with with 7:30 left). Use additional spices if you want (I added roasted garlic and pepper). After 20 minut |
| 18:55.39 | leifmadsen | es inside should be 170F. Take off BBQ and wrap in tinfoil for 10 minutes to absorb the juices. Slice into dimes and eat! :) |
| 18:57.24 | *** join/#asterisk b14ck (~b14ck@dsl-lfkn-207-70-143-25.consolidated.net) |
| 19:00.40 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
| 19:02.06 | *** join/#asterisk bn-7bc (bjarne@mac.wlan.noare-1.holmedal.net) |
| 19:08.00 | raden_work | Sounds Nummy |
| 19:12.37 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 19:14.38 | *** join/#asterisk hardwire (~spencersr@69-161-26-211.static.acsalaska.net) |
| 19:23.32 | leifmadsen | it was quite juicy :) |
| 19:32.01 | philipp64|laptop | coppice: we could narrow things down by compiling one module at a time with optimization, until it breaks. |
| 19:32.30 | philipp64|laptop | then we'd know which module to look at. as I remember, though, the generated code actually doesn't change that much from one compiler version to another. |
| 19:32.38 | carrar | PICS!! |
| 19:32.40 | carrar | of BBQ |
| 19:33.05 | philipp64|laptop | in 12 million lines of code, when we changed from gcc 3.3 to 4.1, less than 200 files had different object. |
| 19:33.20 | philipp64|laptop | out of about 40,000. |
| 19:40.28 | *** join/#asterisk DelphiWorld (~Delphi@41.200.0.115) |
| 19:40.30 | DelphiWorld | hi |
| 19:40.36 | DelphiWorld | anyone own grandstream gxp1200? |
| 19:44.57 | *** join/#asterisk lost_soul (~shawn@cpe-67-241-66-112.twcny.res.rr.com) |
| 19:45.18 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 19:46.12 | *** join/#asterisk NEEDINGHELP123 (Mordi@v58.sgsvr.com) |
| 19:47.04 | NEEDINGHELP123 | hello guys, i really need some help with H323, i am doing a project and i'd like to understand the some things about the project, i will be willing to pay for the help.. please PM me if your available |
| 19:47.18 | NEEDINGHELP123 | not the H323 in asterisk, but H323 in general.. i need a pro |
| 19:55.19 | DelphiWorld | hi NEEDINGHELP123 |
| 19:55.26 | *** join/#asterisk timholum (~chatzilla@64-91-67-5.stat.centurytel.net) |
| 19:55.35 | *** join/#asterisk godfather_ (~godfather@30.Red-88-6-253.staticIP.rima-tde.net) |
| 19:56.26 | *** part/#asterisk DelphiWorld (~Delphi@41.200.0.115) |
| 19:56.38 | Katty | <- just got flowers delivered to work. |
| 19:56.50 | *** join/#asterisk simond (~simon@syria.uc.org) |
| 19:57.18 | simond | In asterisk 1.4, is there some way to see whether a dahdi channel is in alarm within the dialplan? ChanIsAvail does not seem to know. |
| 19:57.26 | pabelanger | NEEDINGHELP123: Or just ask your question here |
| 19:58.03 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 19:59.32 | *** join/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
| 19:59.40 | Qwell | Katty: got a stalker? |
| 19:59.53 | Katty | Qwell: so much for being single a couple months eh? |
| 20:00.01 | Qwell | guess so |
| 20:00.11 | Katty | Qwell: can't even be single for 3 damn weeks without someone knockin at the door |
| 20:00.14 | Qwell | no Qwelldate(TM) either. :P |
| 20:00.23 | Katty | Qwell: well i tried to get you to roulette last night! |
| 20:00.26 | Katty | Qwell: you snob. |
| 20:00.27 | Qwell | did you? |
| 20:00.35 | Qwell | I wondered why this channel was blue this morning.. |
| 20:00.38 | Katty | Qwell: too busy sitting in ER i guess! without me! |
| 20:00.50 | Qwell | I was killing computer monsters |
| 20:00.52 | Katty | Qwell: they are very pretty flowers tho. |
| 20:01.03 | Katty | Qwell: it's all white. |
| 20:01.12 | Katty | Qwell: i recognize white roses in there. |
| 20:01.19 | Qwell | note to self: randomly send flowers when interested |
| 20:02.14 | Katty | Qwell: flowers are nice, but i'm getting a slightly creepy stalker vibe off it |
| 20:02.14 | Nugget | What am I gonna do with 40 subscriptions to Vibe?! |
| 20:02.31 | Katty | Qwell: perhaps might be cause i didn't ever get any flowers from ryan. ever. in 3 years. |
| 20:02.36 | Qwell | note to self: DON'T randomly send flowers when interested |
| 20:02.59 | Katty | Qwell: i think it's okay to randomly send flowers. |
| 20:03.03 | Katty | Qwell: just not...anything big. |
| 20:05.25 | Qwell | note to self: DO be creepy, but only slightly. check! |
| 20:07.20 | Katty | *hee* |
| 20:09.00 | Katty | Qwell: i'm sure it's all dependant on the girl. |
| 20:09.10 | Katty | Qwell: personally, i would have prefered something slightly more geeky. |
| 20:09.21 | Katty | Qwell: like a bonsai with Tetris Pieces hanging on it |
| 20:18.59 | [sr] | WIMPy: LCR could take controls of the led light's on each port on the cards, like DAHDI does.. green on OK, blinking red on failure/not configured |
| 20:24.05 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
| 20:28.43 | NEEDINGHELP123 | Hi Guys |
| 20:28.49 | raden_work | LMAO |
| 20:28.59 | NEEDINGHELP123 | I need help I need to understand some thing for a uni project that I AM totally stuck with |
| 20:29.28 | raden_work | Qwell, if we were to write a book of do's and dont's about pursuing a woman, it would totally contradict itself :) |
| 20:29.30 | Qwell | If your university requires you to learn H323, you should drop out. |
| 20:29.34 | *** join/#asterisk dailylinux (~fedora@s21-00210.dsl.no.powertech.net) |
| 20:29.38 | Qwell | raden_work: that's the point |
| 20:29.47 | pabelanger | ~ask |
| 20:29.48 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:29.49 | NEEDINGHELP123 | I need to udnerstand how to make a call |
| 20:29.53 | Qwell | ~book |
| 20:29.54 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:29.59 | Qwell | NEEDINGHELP123: read that. |
| 20:30.02 | raden_work | ~brain |
| 20:30.03 | infobot | brain is probably a very useful device. It is advised that you ensure your brain is switched to the "on" position before continuing. If you need assistance, please consult your directory. This is a recording. |
| 20:30.32 | raden_work | NEEDINGHELP123, Picking up the phone and dialing would be a good start |
| 20:30.47 | NEEDINGHELP123 | :) |
| 20:30.52 | NEEDINGHELP123 | I understand a sip call |
| 20:31.07 | NEEDINGHELP123 | I understand IAX call |
| 20:31.14 | NEEDINGHELP123 | and how it initates and the codecs etc etc |
| 20:31.25 | raden_work | drops head on desk |
| 20:31.26 | NEEDINGHELP123 | what I dont understand and cant get help with is the way an h323 call is made |
| 20:31.53 | NEEDINGHELP123 | can you help me please? |
| 20:32.22 | NEEDINGHELP123 | i.e. give me an example packet of an h323 call |
| 20:32.26 | NEEDINGHELP123 | how to initate it |
| 20:32.57 | pabelanger | NEEDINGHELP123: read the book |
| 20:34.53 | NEEDINGHELP123 | which book man? |
| 20:34.59 | raden_work | ~book |
| 20:34.59 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:35.00 | NEEDINGHELP123 | I'm looking for help |
| 20:35.19 | NEEDINGHELP123 | I dont have that much time |
| 20:35.31 | NEEDINGHELP123 | I need some one to be serious and helpful that is not too much to ask for I hope |
| 20:35.35 | NEEDINGHELP123 | ? |
| 20:36.30 | leifmadsen | NEEDINGHELP123: hire a consultant |
| 20:36.41 | NEEDINGHELP123 | Thats what I am looking for |
| 20:36.41 | leifmadsen | if time is really that critical |
| 20:36.46 | NEEDINGHELP123 | who wants to help? |
| 20:36.49 | leifmadsen | asterisk-biz |
| 20:36.52 | NEEDINGHELP123 | I am more than willing to pay |
| 20:36.54 | NEEDINGHELP123 | for this service |
| 20:36.57 | leifmadsen | see asterisk-biz mailing list |
| 20:37.07 | NEEDINGHELP123 | and to finish with answers that wont help me |
| 20:38.08 | leifmadsen | I have a 1 million dollar pre-paid retainer for all services |
| 20:38.16 | pabelanger | NEEDINGHELP123: Ask your questions, be specific. You may get an answer. |
| 20:38.17 | raden_work | LMAO |
| 20:38.32 | raden_work | leifmadsen, im way to damn cheap at 55/hr then |
| 20:39.03 | leifmadsen | raden: write a book! charge whatever you want! |
| 20:39.28 | NEEDINGHELP123 | pabelanger |
| 20:39.33 | NEEDINGHELP123 | listen good |
| 20:39.39 | NEEDINGHELP123 | your not helping your just bullying here |
| 20:39.49 | NEEDINGHELP123 | sort it out please specifically |
| 20:39.58 | pabelanger | <PROTECTED> |
| 20:40.13 | NEEDINGHELP123 | I am looking for the equivilent of a invite packet for H3232 |
| 20:40.17 | NEEDINGHELP123 | H323 |
| 20:40.20 | NEEDINGHELP123 | do you understand that |
| 20:40.26 | NEEDINGHELP123 | and can you help please |
| 20:40.28 | NEEDINGHELP123 | ? |
| 20:40.30 | NEEDINGHELP123 | thank you |
| 20:41.50 | pabelanger | NEEDINGHELP123: https://secure.wikimedia.org/wikipedia/en/wiki/H323 |
| 20:42.33 | leifmadsen | NEEDINGHELP123: there are several links to papers and such here: http://en.wikipedia.org/wiki/H.323#H.323_Network_Signaling |
| 20:42.35 | pabelanger | specifically H.225.0 |
| 20:42.54 | leifmadsen | such as http://hive.packetizer.com/users/packetizer/papers/h323/h323_protocol_overview.pdf |
| 20:43.03 | russellb | leifmadsen charges 1 million? I'll cut a deal at 950k |
| 20:43.32 | leifmadsen | russellb: what do you know.. you haven't even co-authored a book! :) |
| 20:43.44 | russellb | yet! |
| 20:43.45 | leifmadsen | russellb: oh? you wrote large chunks of asterisk?! pffft |
| 20:44.05 | leifmadsen | russellb: please, that might impress the ladies, but not I! |
| 20:44.19 | russellb | throws https://issues.asterisk.org/svnstats/asterisk/trunk/user_russell.html at leifmadsen |
| 20:45.01 | leifmadsen | russellb: all that proves is that you're lazy at 4am and on Sundays |
| 20:45.08 | russellb | true statement |
| 20:45.12 | leifmadsen | DEDICATION MAN! |
| 20:45.58 | leifmadsen | russellb: heh :) https://issues.asterisk.org/svnstats/asterisk/trunk/user_lmadsen.html |
| 20:46.19 | russellb | you're calling me lazy on Sunday? |
| 20:46.21 | russellb | you suck all weekend |
| 20:46.43 | raden_work | LMAO |
| 20:46.45 | raden_work | ROFL |
| 20:47.26 | pabelanger | Didn't know that existed |
| 20:47.34 | russellb | it's sekret |
| 20:47.34 | leifmadsen | russellb: apparently I have a Leif :) |
| 20:47.40 | russellb | terrible joke |
| 20:47.43 | leifmadsen | <-- troll! :) |
| 20:48.02 | leifmadsen | russellb: sign my releases! |
| 20:48.07 | russellb | on a conf call |
| 20:48.09 | *** join/#asterisk Gershwin (~fake@unaffiliated/gershwin) |
| 20:48.55 | pabelanger | markster 2007-05-23 21:23 <- most recent commit |
| 20:50.44 | kn0x | :\ |
| 20:51.37 | Gershwin | anyone here have an EOC (ethernet over copper) circut provisioned? |
| 20:51.45 | *** join/#asterisk edguy3 (~edguy@c-98-221-27-224.hsd1.nj.comcast.net) |
| 20:52.00 | russellb | pabelanger: he has people for that now :-p |
| 20:52.23 | Gershwin | either point to point or w/a CLEC/ILEC/ISP, etc |
| 20:52.32 | pabelanger | <PROTECTED> |
| 20:55.46 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 20:58.33 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 21:01.52 | NEEDINGHELP123 | Hi guys |
| 21:02.05 | NEEDINGHELP123 | I need an example of an h323 invite packet |
| 21:02.53 | raden_work | I believe hes wondering if someone has a example of a debug they can show him from a initiated call .. |
| 21:03.16 | telnettech | needinghelp: everyone is helping you....go to the wiki and it will give you everything you need |
| 21:03.30 | telnettech | it isnt rocket science you know |
| 21:03.48 | NEEDINGHELP123 | its not giving me what I need |
| 21:03.58 | NEEDINGHELP123 | and I cannot find an example of this packet |
| 21:04.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:04.04 | NEEDINGHELP123 | for h323 |
| 21:04.04 | telnettech | keep reading....IT WILL!!! |
| 21:04.11 | russellb | "H323 INVITE" does not compute |
| 21:04.41 | NEEDINGHELP123 | but surely if you guys know someting you would be willing to put me out of misery and assist me |
| 21:04.56 | NEEDINGHELP123 | rather than have banter and fight m way for an answer |
| 21:05.20 | russellb | H.225.0 is what you're looking for within H.323 ... and it's based on ISDN Q.931 |
| 21:05.28 | russellb | the call setup message is "SETUP" |
| 21:05.37 | drmessano | ~shoot NEEDINGHELP123 |
| 21:06.02 | russellb | "H.323 Network Signaling" on this page gives you an overview: http://en.wikipedia.org/wiki/H.323 |
| 21:06.56 | *** join/#asterisk garymc (~chatzilla@host86-162-166-186.range86-162.btcentralplus.com) |
| 21:07.48 | garymc | Hi people, I want to make a backup of my Asterisk before I upgrade to 1.6 anyone help me do this? |
| 21:07.48 | NEEDINGHELP123 | I simply need the tcp packet that initates a call in h323 |
| 21:07.54 | NEEDINGHELP123 | I have no idea how to do it |
| 21:08.00 | NEEDINGHELP123 | I have no idea how to see it |
| 21:08.08 | NEEDINGHELP123 | and I simply am at my wits end |
| 21:08.11 | NEEDINGHELP123 | I am reading |
| 21:08.23 | NEEDINGHELP123 | as much information as everyone is giving me |
| 21:08.27 | russellb | blinks |
| 21:08.44 | russellb | what do you mean you need the TCP packet? |
| 21:09.02 | russellb | are you familiar with the OSI model for how network protocols operate? |
| 21:09.26 | NEEDINGHELP123 | yes |
| 21:09.38 | NEEDINGHELP123 | I need the application layer protocl spefication |
| 21:09.39 | NEEDINGHELP123 | s |
| 21:10.40 | russellb | http://www.h323forum.org/standards/ |
| 21:10.42 | Qwell | so download it? |
| 21:12.31 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:12.48 | NEEDINGHELP123 | thanks |
| 21:13.09 | russellb | you're welcome. |
| 21:13.41 | russellb | sends a bill |
| 21:14.18 | Katty | hungry :< |
| 21:14.54 | NEEDINGHELP123 | but that's a hell of a lot of data ;-). Look, I have to write an application that tests my h323 phones connection. I don't need the entire protocol but I'm having a very hard time finding the information I need for this exactly. And I'm on a tight deadline as well... |
| 21:15.16 | russellb | i'm not writing code for your school project, heh |
| 21:16.54 | NEEDINGHELP123 | I don't want anyone to write the code, but I'm just looking for easy documents that describe the structures involved of the data that is being sent and what the responses are. |
| 21:17.05 | russellb | that's what the standards are |
| 21:17.15 | russellb | it's not a simple standard |
| 21:17.24 | NEEDINGHELP123 | I can tell ;) |
| 21:17.46 | Gershwin | sorry, had to step away |
| 21:17.49 | Gershwin | you still there pabelanger? |
| 21:18.27 | Gershwin | did you mean to say you've got an EOC circuit at one of your locations? |
| 21:19.42 | NEEDINGHELP123 | the situation is this |
| 21:19.58 | NEEDINGHELP123 | My class teacher has a project |
| 21:20.05 | NEEDINGHELP123 | and he wants us to develop the software |
| 21:20.16 | russellb | your teacher is getting you to do his work? :-) |
| 21:20.26 | NEEDINGHELP123 | to dial his h323 phone |
| 21:20.29 | NEEDINGHELP123 | that is the competition |
| 21:20.35 | raden_work | I was just going to say ..... |
| 21:20.53 | raden_work | get a h.323 phone :) |
| 21:21.07 | NEEDINGHELP123 | I know but the problem is is I cant do that |
| 21:21.11 | NEEDINGHELP123 | :-) |
| 21:21.19 | russellb | install asterisk? |
| 21:21.23 | NEEDINGHELP123 | I would use a call generator |
| 21:21.23 | russellb | and tell asterisk to make a call? |
| 21:21.43 | NEEDINGHELP123 | thats not the idea the idea is that we can create the packet in order to dial that phoen we have the details and all authenticaiton |
| 21:21.53 | NEEDINGHELP123 | so all is easy |
| 21:22.01 | NEEDINGHELP123 | but I must use a packet that we created |
| 21:22.16 | NEEDINGHELP123 | if I could pull apert an open source h323 phone and see what the packet it |
| 21:22.22 | raden_work | I cannot believe are phone are working |
| 21:22.24 | raden_work | 13 64.74.178.102 (64.74.178.102) 1019.066 ms 1020.974 ms 1020.831 ms |
| 21:22.25 | russellb | write an app against this library then http://www.h323plus.org/ |
| 21:22.27 | raden_work | to vitelity |
| 21:22.51 | NEEDINGHELP123 | I would but I dont have the time and dont have the knowledge either |
| 21:23.02 | russellb | well you're screwed |
| 21:23.21 | drmessano | Let me guess, there's a monetary prize? |
| 21:23.30 | NEEDINGHELP123 | but to send a packet across the unis network and dial a phone when I know the IP and have authentication |
| 21:23.33 | NEEDINGHELP123 | should be easy |
| 21:23.37 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:23.41 | russellb | no, it shouldn't be easy |
| 21:23.44 | NEEDINGHELP123 | its more an issue of respect |
| 21:23.50 | NEEDINGHELP123 | and my teachig appreciating me |
| 21:23.51 | russellb | it gets no easier that writing a trivial application against an existing protocol stack |
| 21:23.56 | raden_work | Whats her name ? |
| 21:24.00 | raden_work | :) |
| 21:24.10 | NEEDINGHELP123 | Mrs Wright |
| 21:24.18 | NEEDINGHELP123 | ;) |
| 21:24.22 | raden_work | lol |
| 21:24.23 | kn0x | NEEDINGHELP123: http://wiki.wireshark.org/SampleCaptures?action=AttachFile&do=view&target=rtp_example.raw.gz |
| 21:24.26 | NEEDINGHELP123 | professor balanksy |
| 21:24.33 | drmessano | NEEDINGHELP123: Well, considering that this channel is logged and searchable with google, I hope your teacher doesn't find where you came here to solicit a SHORTCUT |
| 21:24.37 | drmessano | So much for Respect |
| 21:24.53 | drmessano | F minus minus |
| 21:24.53 | NEEDINGHELP123 | it is innovative |
| 21:25.06 | raden_work | drmessano, this channel logged to a website or something ? |
| 21:25.08 | NEEDINGHELP123 | and if she does hcek then I will be more than happy to explain |
| 21:25.21 | drmessano | raden_work: Yeah |
| 21:25.41 | raden_work | interesting .. |
| 21:25.55 | kn0x | women dont know anything about voip anyway... |
| 21:25.57 | drmessano | HAHAHAH |
| 21:26.00 | drmessano | http://www.medhelp.org/posts/STDs/Bump-on-penis/show/245922 |
| 21:26.01 | NEEDINGHELP123 | even if I use wireshark |
| 21:26.04 | drmessano | Check the first post |
| 21:26.08 | drmessano | NEEDINGHELP123 |
| 21:26.11 | drmessano | :( |
| 21:26.13 | kn0x | drmessano: NSFW |
| 21:26.32 | drmessano | It's educational, not porn |
| 21:26.39 | NEEDINGHELP123 | if I use wireshark how will I analyze the protocl |
| 21:26.43 | NEEDINGHELP123 | ? |
| 21:27.07 | russellb | facepalms |
| 21:27.12 | drmessano | If can help H323 help and some cream for that bump, he's good to go |
| 21:27.25 | drmessano | GAH |
| 21:27.44 | drmessano | s/If can help/If he can get/ |
| 21:27.51 | drmessano | brain/keyboard fail |
| 21:28.01 | kn0x | :D |
| 21:28.15 | drmessano | Either way, google is your friend |
| 21:28.26 | raden_work | lmao |
| 21:29.20 | drmessano | Last post was that he was trying to get closer pics with his digital camera and would check back :( |
| 21:30.00 | drmessano | NEEDINGHELP123: If you use Wireshark, you ARE analyzing the protocol |
| 21:30.14 | russellb | drmessano: heh, i don't know about that ... |
| 21:30.27 | russellb | wireshark is decoding it, it's up to the human to analyze :-) |
| 21:31.07 | russellb | (so no) |
| 21:31.15 | drmessano | NEEDINGHELP123: If you use Wireshark, and your eyes aren't CLOSED, you ARE analyzing the protocol |
| 21:31.32 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 21:31.42 | NEEDINGHELP123 | analysing |
| 21:31.57 | NEEDINGHELP123 | is anyalysing |
| 21:32.10 | NEEDINGHELP123 | but understanding it is better |
| 21:32.10 | kn0x | NEEDINGHELP123: you look at the packets captured... thats your call flow. |
| 21:32.16 | drmessano | Analysis is the process of breaking a complex topic or substance into smaller parts to gain a better understanding of it. |
| 21:32.24 | NEEDINGHELP123 | because then I can get the packet =and understand where to put my details and win the frigging competition |
| 21:32.28 | russellb | ~roulette |
| 21:32.28 | infobot | ACTION watches russellb pull the trigger: Click! |
| 21:32.42 | drmessano | and win the frigging competition <--- |
| 21:32.42 | kn0x | ~roulette |
| 21:32.43 | infobot | ACTION watches kn0x pull the trigger: Click! |
| 21:32.45 | thansen | I have an asterisk server registered as sip endpoint to a provide. I just restared (without making any changes to sip.conf) and now I keep getting SIP/2.0 407 Proxy Authentication Required |
| 21:32.50 | drmessano | NOW we get to the bottom of it |
| 21:33.01 | russellb | thansen: that's normal for the first response you get |
| 21:33.09 | drmessano | What is in it for ME, NEEDINGHELP123 ? |
| 21:33.22 | russellb | drmessano: some pics |
| 21:33.23 | thansen | russellb: I've been getting it for 1.5 hours :( killing business |
| 21:33.49 | drmessano | russellb: ROFL |
| 21:34.01 | russellb | thansen: try reloading? *CLI> module reload chan_sip.so |
| 21:34.05 | drmessano | Ba-da-BUMP |
| 21:34.30 | drmessano | russellb, since you're the drum master.. rimshot, plz |
| 21:34.42 | thansen | russellb: same thing |
| 21:34.44 | russellb | cracks a mean rimshot |
| 21:34.52 | russellb | thansen: restart again? :-/ |
| 21:35.00 | *** join/#asterisk PTorres (~PTorres@200.68.87.148) |
| 21:35.03 | drmessano | Did you reboot 3 times? |
| 21:35.11 | Corydon76-dig | thansen: are you sure you made no changes? |
| 21:35.12 | *** join/#asterisk bjhaid (~IceChat7@41.220.68.9) |
| 21:35.23 | thansen | yeah |
| 21:35.34 | Corydon76-dig | Like, checked the timestamp on the file? |
| 21:35.43 | drmessano | NEEDINGHELP123: Do NOT PM me |
| 21:36.14 | thansen | keep getting [Jul 22 15:35:42] WARNING[24405]: chan_sip.c:12675 check_auth: username mismatch, have <alarmguard-801>, digest has <alarmguard-802> |
| 21:36.26 | drmessano | No, I will not help you.. and no you cannot afford my asking price |
| 21:37.02 | bjhaid | i installed dahdi via sudo apt-get dahdi on my ubuntu machine, just want to know if it would work well |
| 21:37.13 | NEEDINGHELP123 | so why did you ask what is in it for me? |
| 21:37.17 | NEEDINGHELP123 | is that not just an ass |
| 21:37.20 | NEEDINGHELP123 | of a thing to say |
| 21:37.21 | NEEDINGHELP123 | ? |
| 21:37.28 | NEEDINGHELP123 | and then to put me down |
| 21:37.28 | russellb | ~enter |
| 21:37.29 | infobot | the enter key is not a substitute for punctuation. Use a period '.', exclamation mark '!', question mark '?', comma ',', colon ':', semi-colon ';' emdash '--', or ellipsis '...' instead. |
| 21:37.33 | kn0x | NEEDINGHELP123: $150/hour |
| 21:37.39 | kn0x | prepaid paypal |
| 21:37.47 | [TK]D-Fender | NEEDINGHELP123: ... [17:33]<drmessano>What is in it for ME, NEEDINGHELP123 ? <-- he asked what what was in it for HIM |
| 21:37.57 | [TK]D-Fender | NEEDINGHELP123: not YOU |
| 21:38.06 | russellb | ~roulette |
| 21:38.07 | infobot | ACTION watches russellb pull the trigger: Click! |
| 21:38.12 | russellb | come on infobot, kill me nowww |
| 21:38.19 | kn0x | haha russellb does infobot every kill anyone? |
| 21:38.23 | [TK]D-Fender | ~die |
| 21:38.24 | infobot | ACTION takes two shots to the head and crumples to the ground, lifeless. |
| 21:38.26 | Corydon76-dig | russellb: so you can go home |
| 21:38.28 | [TK]D-Fender | :D |
| 21:38.29 | Corydon76-dig | ? |
| 21:38.30 | drmessano | ~shoot NEEDINGHELP123 |
| 21:38.31 | Qwell | ~kill kn0x |
| 21:38.32 | NEEDINGHELP123 | let me ask it straight as a dice |
| 21:38.37 | russellb | ~roulette |
| 21:38.38 | infobot | ACTION watches russellb pull the trigger: Click! |
| 21:38.38 | NEEDINGHELP123 | becasue I am not getting anything here |
| 21:38.52 | drmessano | NEEDINGHELP123: Let me tell you, as lumpy as a bump.. |
| 21:39.09 | [TK]D-Fender | NEEDINGHELP123: No-one is going to waste their time on a useless project for charity for a person who doesn't execpt to have to do any work. |
| 21:39.09 | russellb | NEEDINGHELP123: I have given you links to protocol overview documents, the specifications for the protocol, and an implementation of the protocol. someone else gave you a packet capture. |
| 21:39.09 | NEEDINGHELP123 | drmessano |
| 21:39.10 | russellb | what else could you get? |
| 21:39.16 | NEEDINGHELP123 | do you know the answer to my question |
| 21:39.18 | NEEDINGHELP123 | ? |
| 21:39.24 | drmessano | Yes I do |
| 21:39.28 | Corydon76-dig | NEEDINGHELP123: I'd say you're well out of your depth. You probably need to take a few training courses, first |
| 21:39.48 | russellb | Corydon76-dig: heh, clearly, but he's trying to impress his female teacher and win a competition or something |
| 21:39.59 | Qwell | what does the winner get? |
| 21:40.05 | russellb | i don't know, but i want my cut |
| 21:40.11 | Qwell | and do I have to be a student to enter? |
| 21:40.31 | Qwell | ~cut russellb |
| 21:40.33 | Corydon76-dig | NEEDINGHELP123: how old are you? |
| 21:40.35 | Qwell | stupid bot |
| 21:40.36 | NEEDINGHELP123 | no you can enter via me as a proxy |
| 21:40.37 | NEEDINGHELP123 | :) |
| 21:40.38 | russellb | ~thwack Qwell |
| 21:40.47 | [TK]D-Fender | NEEDINGHELP123: You clearly lack the skills that would make worthy of being considered "impressive". Doing this for false pride as an act of charity is a waste of our time. |
| 21:40.50 | NEEDINGHELP123 | 19 |
| 21:41.09 | NEEDINGHELP123 | there is nothing false about it |
| 21:41.15 | NEEDINGHELP123 | I need to dial her phone |
| 21:41.18 | NEEDINGHELP123 | no matter how I do it |
| 21:41.21 | [TK]D-Fender | NEEDINGHELP123: it is when everyone else hands you the answer |
| 21:41.45 | drmessano | He just wants to ask her out |
| 21:41.47 | Corydon76-dig | NEEDINGHELP123: if you want to impress her, do your homework |
| 21:41.57 | [TK]D-Fender | NEEDINGHELP123: Am I impressive because I hire a professional boxer to fight my personal fights? |
| 21:41.57 | NEEDINGHELP123 | no your wrong |
| 21:42.02 | Corydon76-dig | consistently and on time |
| 21:42.04 | drmessano | Yeah, and get some cream for that 5 yr old bump |
| 21:42.08 | russellb | it's "you're", not "your" |
| 21:42.22 | russellb | drmessano: it's funny that he never told you that wasn't him, heh |
| 21:42.25 | NEEDINGHELP123 | you come from a world of thought that is taught to you |
| 21:42.29 | NEEDINGHELP123 | line breeding |
| 21:42.29 | drmessano | russellb: Indeed |
| 21:42.42 | NEEDINGHELP123 | I need the answers that I need so I can move on |
| 21:42.46 | NEEDINGHELP123 | I am willing to pay for that |
| 21:42.56 | drmessano | NEEDINGHELP123: Maybe of us are self taught nerds.. They don't teach "VoIP" in school |
| 21:42.57 | kn0x | NEEDINGHELP123: not enough. |
| 21:42.58 | russellb | asterisk-biz mailing list |
| 21:42.59 | russellb | kthxbye |
| 21:43.04 | NEEDINGHELP123 | will somebody please take my F******* money and help with the ansqwer and give me the exact specs of what I need? |
| 21:43.09 | russellb | lol |
| 21:43.25 | kn0x | slaps % around with a large trout. NEEDINGHELP123 |
| 21:43.29 | kn0x | oop |
| 21:43.30 | kn0x | s |
| 21:43.32 | russellb | fail. |
| 21:43.33 | drmessano | LOL |
| 21:43.38 | Corydon76-dig | What you need is a mathematics degree |
| 21:43.38 | drmessano | ~trout NEEDINGHELP123 |
| 21:43.44 | drmessano | ~slap NEEDINGHELP123 |
| 21:43.57 | [TK]D-Fender | NEEDINGHELP123: http://www.rfc-archive.org/getrfc.php?rfc=4123 |
| 21:43.59 | drmessano | NEEDINGHELP123: What did the 5 fingers say to the face? |
| 21:44.00 | drmessano | SLAP |
| 21:44.09 | russellb | ~frag drmessano |
| 21:44.12 | drmessano | I'M RICK JAMES |
| 21:44.16 | russellb | infobot forgot a lot of commands :-( |
| 21:44.27 | drmessano | Yeah, WTF happened to it? |
| 21:44.28 | raden_work | infobot loose his brain ? |
| 21:44.33 | russellb | guess so |
| 21:44.38 | Corydon76-dig | ~timriker |
| 21:44.38 | infobot | timriker is probably my owner http://rikers.org/ mailto:Tim@Rikers.org mailto:TimR@Debian.org maintainer of BZFlag, member of a ton of open source projects http://www.advogato.com/person/timriker/ http://sourceforge.net/users/timriker/ the guy who GPL'd SCO's ABI files, giving every Linux user the right to use them ;-), or a very cool guy. |
| 21:44.42 | drmessano | ~happyclownpbx |
| 21:44.43 | infobot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for its core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 21:44.50 | drmessano | YAY |
| 21:45.02 | russellb | ~drumkilla |
| 21:45.02 | infobot | somebody said drumkilla was Russell Bryant, the Asterisk release branch maintainer <russelb@clemson.edu>, or someone who should be ph33r3d, or russellb |
| 21:45.09 | *** join/#asterisk moy (~moy@74.12.99.17) |
| 21:45.11 | russellb | that's ancient |
| 21:45.12 | drmessano | LOL |
| 21:45.17 | drmessano | ~drmessano |
| 21:45.17 | infobot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily |
| 21:45.18 | russellb | so someone must have pruned silly stuff |
| 21:45.30 | drmessano | heh |
| 21:45.38 | Corydon76-dig | ~corydon |
| 21:45.39 | infobot | [corydon] one of the most l33t Asterisk developers around. He has been around longer than you have. |
| 21:45.46 | drmessano | LOL |
| 21:45.56 | russellb | ~russellb |
| 21:45.57 | infobot | you are, like, Russell Bryant <russell@digium.com> or <russell@russellbryant.net>, the Asterisk project lead. Blog @ http://www.russellbryant.net/ |
| 21:46.06 | russellb | infobot is a valley girl? |
| 21:46.19 | drmessano | ~trashbox |
| 21:46.23 | drmessano | ~trixbox |
| 21:46.24 | infobot | well, trixbox is SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 21:46.28 | russellb | lol |
| 21:46.29 | *** part/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 21:46.33 | drmessano | ~pbxpilaf |
| 21:46.36 | Corydon76-dig | What's odd is that I haven't edited mine |
| 21:46.37 | raden_work | lol, im outta here |
| 21:46.45 | russellb | waves to raden_work |
| 21:46.52 | drmessano | chow |
| 21:49.30 | *** join/#asterisk b14ck (~b14ck@173.219.15.94) |
| 21:49.41 | NEEDINGHELP123 | I helped myself |
| 21:49.52 | NEEDINGHELP123 | with some nicotine and coffeee |
| 21:50.06 | NEEDINGHELP123 | now I still need some help |
| 21:50.09 | NEEDINGHELP123 | drmessano |
| 21:50.26 | NEEDINGHELP123 | russellb |
| 21:50.34 | russellb | good luck with that |
| 21:50.34 | NEEDINGHELP123 | I offer a bounty of 500 USD for this |
| 21:51.15 | kn0x | lmao |
| 21:51.27 | NEEDINGHELP123 | what more can I try guys |
| 21:51.33 | NEEDINGHELP123 | you can see that I need the assistance |
| 21:51.34 | russellb | you can try more money if you want |
| 21:51.43 | NEEDINGHELP123 | and you can see that I'm not getting anywhere |
| 21:51.46 | kn0x | NEEDINGHELP123: okay, let me give you my US checking account number.. |
| 21:51.47 | NEEDINGHELP123 | I will pay a price |
| 21:51.54 | NEEDINGHELP123 | no issues |
| 21:51.56 | [TK]D-Fender | NEEDINGHELP123: You just offered to one person who doesnt want your money, and another who I believe might have a confict of interest in accepting it. |
| 21:52.07 | kn0x | NEEDINGHELP123: ...please let me know what nigerian bank you will be wiring from |
| 21:52.16 | NEEDINGHELP123 | lol |
| 21:52.20 | NEEDINGHELP123 | thats funny |
| 21:52.26 | NEEDINGHELP123 | :-) |
| 21:52.36 | NEEDINGHELP123 | but seriously |
| 21:52.45 | NEEDINGHELP123 | I cant get too much sense from anyone here |
| 21:52.54 | NEEDINGHELP123 | I am offering a service and looing for a service |
| 21:52.59 | Katty | yelllllowwwwwwwwwww. |
| 21:53.03 | NEEDINGHELP123 | what can be better than this |
| 21:53.12 | Katty | cookies. |
| 21:53.13 | Katty | obviously. |
| 21:53.16 | NEEDINGHELP123 | you guys have the knowledge and I have resources |
| 21:53.18 | Katty | possibly kittens. |
| 21:53.46 | NEEDINGHELP123 | it seems totally crazy that this is the case |
| 21:53.56 | Katty | no i think kittens are simply awesome. |
| 21:54.04 | NEEDINGHELP123 | so please step forward any one that is will to help me |
| 21:54.15 | Katty | gonna say it now--i'm out. |
| 21:54.24 | kn0x | asterisk 1.6.2.4 addons 1.6.2.1.. occassionally I am getting cdr_addon_mysql.c: Unable to query table description!! |
| 21:54.25 | NEEDINGHELP123 | thaks katty |
| 21:54.29 | Katty | i'm too busy, gotta meet Qwell in ER. |
| 21:54.35 | Katty | or that's my excuse. |
| 21:54.36 | NEEDINGHELP123 | ;) |
| 21:54.46 | NEEDINGHELP123 | blowdrying your hair |
| 21:54.51 | Katty | god no. |
| 21:54.59 | Katty | blowing drying your hair does an insane ammount of damage to it |
| 21:55.01 | NEEDINGHELP123 | shalom |
| 21:55.11 | Katty | then you have to oil it for 2 weeks straight to get the proper moisture back in there. |
| 21:58.31 | NEEDINGHELP123 | I did not think that it would be this hard to get a tcp packet in a protocol that is used ww on a dialy basis |
| 21:58.41 | NEEDINGHELP123 | my tutor was having a big laugh with us |
| 21:58.46 | NEEDINGHELP123 | :) |
| 22:01.40 | kn0x | NEEDINGHELP123: what is your assignment anyway? |
| 22:02.44 | kn0x | NEEDINGHELP123: why dont you ask your classmates how they are tackling the issue |
| 22:02.54 | Qwell | Katty: going without me?! I see how it is. |
| 22:03.59 | *** part/#asterisk PTorres (~PTorres@200.68.87.148) |
| 22:06.32 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:06.54 | *** join/#asterisk timholum (~chatzilla@64-91-67-5.stat.centurytel.net) |
| 22:08.00 | kn0x | is mysql through odbc better-supported than cdr_addon_mysql? |
| 22:08.01 | drmessano | kn0x: His ASSignment is to annoy an IRC channel into revealing that H323 is extinct |
| 22:08.04 | drmessano | Fine, you GOT US |
| 22:08.08 | drmessano | H323 is EXTINCT |
| 22:08.12 | drmessano | There, WE SAID IT |
| 22:10.08 | NEEDINGHELP123 | how can it be extinct Dr? |
| 22:10.20 | Qwell | He's a Dr. Don't question him. |
| 22:10.28 | NEEDINGHELP123 | it is the most widely used protocol? |
| 22:10.37 | Qwell | H.323? Widely used? No. |
| 22:10.40 | kn0x | drmessano: lol |
| 22:11.03 | NEEDINGHELP123 | I'm in a nightmare |
| 22:11.07 | NEEDINGHELP123 | a real one |
| 22:11.13 | NEEDINGHELP123 | please give me some insight? |
| 22:11.31 | NEEDINGHELP123 | (regardless, I need to know about the packet) |
| 22:12.51 | Qwell | NEEDINGHELP123: Please leave. You've been given more than enough information. We aren't here to do your homework for you. |
| 22:13.03 | drmessano | H.323 isn't widely used. It's widely extinct |
| 22:13.20 | NEEDINGHELP123 | please don't mistake questions for completeing my homework |
| 22:13.29 | NEEDINGHELP123 | I have no intention to insult anybody |
| 22:13.40 | NEEDINGHELP123 | so far it is harmless banter |
| 22:13.42 | drmessano | Besides which, those bumps |
| 22:13.46 | NEEDINGHELP123 | and I have only been asking questions |
| 22:13.54 | drmessano | Yes, and we have ALL answered you |
| 22:13.58 | NEEDINGHELP123 | so I dont understand why you would ask me to leave |
| 22:14.06 | drmessano | A DEVELOPER gave you very specific info |
| 22:14.09 | NEEDINGHELP123 | thank you for being decent human beings |
| 22:14.17 | NEEDINGHELP123 | and assisting another with his problems |
| 22:14.17 | drmessano | Shalom, my friend |
| 22:14.30 | NEEDINGHELP123 | lol |
| 22:15.10 | NEEDINGHELP123 | anything I say is not going to help |
| 22:15.18 | NEEDINGHELP123 | or get my the right answer here |
| 22:15.19 | NEEDINGHELP123 | correct? |
| 22:15.27 | NEEDINGHELP123 | me* |
| 22:15.29 | drmessano | You got lots of answers |
| 22:15.32 | NEEDINGHELP123 | for sure |
| 22:15.38 | NEEDINGHELP123 | I was told to go and read |
| 22:15.43 | NEEDINGHELP123 | and that h232 is dead |
| 22:15.44 | NEEDINGHELP123 | :) |
| 22:15.45 | Qwell | Correct. Go and read. |
| 22:15.48 | Qwell | and H.323 is dead. |
| 22:16.26 | drmessano | [17:05] <russellb> H.225.0 is what you're looking for within H.323 ... and it's based on ISDN Q.931 |
| 22:16.26 | drmessano | [17:05] <russellb> the call setup message is "SETUP" |
| 22:16.26 | drmessano | [17:06] <russellb> "H.323 Network Signaling" on this page gives you an overview: http://en.wikipedia.org/wiki/H.323 |
| 22:16.28 | *** join/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
| 22:16.32 | drmessano | Thats what russellb told you |
| 22:16.36 | drmessano | Now roll with that |
| 22:16.46 | drmessano | That was well over an hour ago |
| 22:16.50 | drmessano | DUH much? |
| 22:17.09 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 22:17.12 | tompaw | Hello. |
| 22:17.28 | NEEDINGHELP123 | okay guys |
| 22:17.38 | *** join/#asterisk _zoom_ (~zoom@41.218.33.171) |
| 22:17.39 | NEEDINGHELP123 | IF i dont get what I need I'll come back |
| 22:17.48 | drmessano | Please don't |
| 22:17.49 | NEEDINGHELP123 | thank you for your help |
| 22:18.16 | drmessano | This isn't #h323, this is #asterisk.. You got your info |
| 22:18.23 | drmessano | Which was off topic, mostly |
| 22:18.24 | bmoraca_work | i can plz has halp plz! |
| 22:18.28 | _zoom_ | hi guys, have u ever faced a problem of passing .729 over openvpn? |
| 22:18.34 | drmessano | Now roll with that, and good luck with that bump |
| 22:18.37 | tompaw | Guys, do you know of any software that would produce a graph of an audio file? (.wav) |
| 22:18.57 | drmessano | tompaw: An audio editor? |
| 22:19.01 | bmoraca_work | tompaw: nero wave editor does it...i'm sure any of the pinnacle software would as well |
| 22:19.07 | tompaw | I want to protect my switch against FAS and I was considering passing the calls through asterisk, monitoring them and producing a graph outputs |
| 22:19.25 | tompaw | erm... but I need something open source that I can run from *nix console |
| 22:19.30 | tompaw | like wav2png ;P |
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| 22:36.23 | boodu | hello |
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| 22:37.58 | Trab | hello, I'm hoping someone can help me. I'm considering using asterisk as our new phone system, but I need to know if it's capable of doing a few things |
| 22:38.12 | bmoraca_work | Trab: ask your questions |
| 22:38.28 | Qwell | Trab: the answer to any question you may have is "yes". |
| 22:38.35 | Trab | ha. |
| 22:38.43 | Qwell | I challenge anybody to come up with something that Asterisk cannot do. |
| 22:39.09 | Trab | well, we want to have multiple phone lines come in, and then when they make a call, be able to pick which number to use out. seems pretty basic to me... |
| 22:39.17 | bmoraca_work | Qwell: queues shared between multiple boxes over a large geographic area :P |
| 22:39.43 | Qwell | bmoraca_work: simple. couple lines of dialplan. |
| 22:40.05 | rbd_ | hi guys, running asterisk 1.4.30. on a linode (Xen hypervisor) instance and ubuntu 9.10. dahdi_dummy was the stock ubuntu one, and after switching to the linux-xen kernel, compiled and loaded fine. I can even get a caller to join a meetme conference. However, after some time (around 30 seconds?) the user is kicked from the conference and Asterisk prints out: app_meetme.c:803 build_conf: Unable to open pseudo device. seems like dahdi/asterisk |
| 22:40.15 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 22:40.19 | bmoraca_work | Qwell: and queue position is maintained over all boxes to any agent on any box? |
| 22:40.22 | Trab | does asterisk work with ringcentral? |
| 22:40.28 | Qwell | bmoraca_work: sure |
| 22:40.35 | Qwell | Trab: is it SIP? |
| 22:40.49 | bmoraca_work | Trab: it's certainly possible...though you'll go crazy if you try to implement that with analog lines |
| 22:41.21 | Trab | well, here's our problem. (and please be kind, I certainly didn't design or endorse the current system) |
| 22:41.45 | Trab | we have vontage phone lines right now. and their service to us, sucks. we have spent hours working on it, and cant determine why |
| 22:42.13 | Trab | so, they're switching to cox analog lines instead. (we just moved, have 50mbit connection, barely using it, and vontage still sucks) |
| 22:42.41 | Trab | I believe they just had ring-central numbers forwarding to vontage. |
| 22:42.55 | Trab | what they want to do, is when they make a phone call out, be able to use the ring-central number |
| 22:42.59 | Trab | (for caller id) |
| 22:43.05 | Trab | but have that integrated into the phone. |
| 22:43.10 | bmoraca_work | Trab: cancel the move to cox analog lines and signup with a SIP provider or something that will give you the ability to specify outbound callerid over a multi-channel connection |
| 22:43.27 | Trab | does ringcentral offer that service? |
| 22:43.36 | bmoraca_work | i don't know what ringcentral is |
| 22:44.04 | bmoraca_work | do you just use them for toll-free? |
| 22:44.24 | [TK]D-Fender | Trab: I've never heard of them either in the many years I've been here |
| 22:44.29 | Trab | no. it's kinda like a google voice from what I gather. |
| 22:44.37 | Trab | http://www.ringcentral.com/ |
| 22:44.43 | bmoraca_work | ah...so basically pointless in the context of asterisk, then |
| 22:44.48 | [TK]D-Fender | Trab: I'd suggest porting to a decent provider that lets yous et your CallerID as other have suggested |
| 22:45.27 | Trab | bmoraca_work maybe. I'm slightly confused. this seems needlessly complicated.. |
| 22:45.30 | [TK]D-Fender | Trab: yes, Vonage sucks. Feel free to ditch them |
| 22:45.37 | Trab | (not asterisk, but the current system |
| 22:45.43 | Trab | that they have in place |
| 22:46.01 | bmoraca_work | Trab: yes, it does seem needlessly complicated. there's no reason to have ringcentral when you have asterisk. i can see how they might want that in other situations, but asterisk can do all that by itself |
| 22:46.22 | Trab | bmoraca_work as far as I'm aware asterisk isn't a phone service though, right? |
| 22:46.24 | Jumpie | man inception was badass |
| 22:46.29 | Trab | so who would be providing our phone lines? |
| 22:46.34 | bmoraca_work | Trab: the bottom line is that with an analog line (whether they're provided via an ATA, over cable, or direct copper), you cannot specify callerid |
| 22:46.41 | [TK]D-Fender | Trab: Correct, * is not a SERVICE |
| 22:46.53 | bmoraca_work | Trab: anyone you want can provide your phone lines. but if you use analog, you are severely restricted. |
| 22:46.53 | [TK]D-Fender | Trab: * is a PBX a telephony toolkit <- |
| 22:47.00 | [TK]D-Fender | and* |
| 22:47.16 | Trab | bmoraca_work so who would provide a digital line? |
| 22:47.27 | [TK]D-Fender | Trab: Another ITSP <- |
| 22:47.34 | [TK]D-Fender | ~itsplist-us |
| 22:47.35 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 22:47.45 | bmoraca_work | Trab: your local ILEC or CLECs can provide PRIs if you need that kind of capacity...or you can use an ITSP |
| 22:48.46 | Trab | is there one vs the other that will likely work better? |
| 22:48.58 | Trab | I guess my fear of this whole thing is: my boss was giving me a crapstorm over vontage sucking. |
| 22:49.04 | bmoraca_work | i can't stand callcentric...so, in my mind, anyone but them is good. |
| 22:49.06 | Trab | they're trying to blame our router, our network, etc etc. |
| 22:49.09 | Qwell | Trab: That would be because Vonage sucks. |
| 22:49.14 | Trab | and it's really, vonage |
| 22:49.25 | bmoraca_work | however, i usually use myself as an ITSP and don't work with any others outside of wholesaling with globalpops |
| 22:49.27 | Trab | so with an ITSP we're not gonna have those issues? |
| 22:49.51 | bmoraca_work | Trab: you may have other issues, but likely not the same ones as with vonage. |
| 22:49.58 | [TK]D-Fender | Trab: You can. Vonage is an ITSP |
| 22:50.05 | [TK]D-Fender | Trab: You seem to miss the big print. |
| 22:50.12 | [TK]D-Fender | Trab: the point is they SUCK |
| 22:50.29 | Trab | okay. so even if vonage is an ITSP, I shouldn't judge all ITSP's on vonage sucking |
| 22:50.43 | Trab | bmoraca_work what is a PRI? |
| 22:50.49 | [TK]D-Fender | ~pri |
| 22:50.50 | infobot | methinks pri is [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 22:51.28 | [TK]D-Fender | Trab: PRI is a signallng used on T1 (and similar) circuits to your telco |
| 22:51.44 | [TK]D-Fender | Trab: note this is NOT a "data" circuit |
| 22:51.53 | [TK]D-Fender | Trab: TDM to the telco |
| 22:52.10 | Trab | got it. |
| 22:52.37 | Jumpie | messing with euro circuits got confusing when i worked at sprint |
| 22:52.43 | Jumpie | the heirarchy is different |
| 22:53.13 | Trab | okay, so if I go with a better ITSP and use asterisk, i should have much more control. my next question is this: which ITSP can you recommend? http://www.teliax.com is the first one I'm looking into |
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| 22:56.59 | [TK]D-Fender | Trab: I'd start with voip.ms , and then les.net |
| 22:57.02 | bmoraca_work | Trab: yeah, a PRI is a digital voice T1...it'll give you very good control over your system, but is generally intended for higher-volume installations. |
| 22:57.39 | [TK]D-Fender | Trab: PRI is best, but requires a more expensive link to your telco, and a hardware interface investment |
| 22:58.22 | bmoraca_work | if you have a good quality internet connection and router, you can have nearly as good service from an ITSP. |
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| 23:17.15 | Jumpie | yep |
| 23:21.54 | bmoraca_work | mmmmm hmmmmmmmmm |
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