| 00:00.16 | *** join/#asterisk Smi|k (~Ling@adsl-66-159-200-157.dslextreme.com) |
| 00:00.25 | JerJer | where? |
| 00:00.42 | Smi|k | if I get a PRI T1 with many DID's can I have ring-down service across all 23 lines from each of those numbers? |
| 00:01.01 | JerJer | if your carrier puts the number in a trunk group on the PRI |
| 00:01.13 | JerJer | which is normal - but there are some pretty lame carriers out there |
| 00:01.16 | *** part/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 00:01.16 | Smi|k | and is it allocated, or not? |
| 00:01.36 | Smi|k | i.e. if I have one number that gets called a LOT during the day, and one a LOT during the night, can the 23 lines be leveraged to both of them |
| 00:01.41 | Smi|k | only a confilict when the 23 runs out |
| 00:01.45 | JerJer | sure |
| 00:02.01 | JerJer | but that's totally up to the carrier that delivers you that PRI |
| 00:02.01 | meppl | gute nacht |
| 00:02.08 | Smi|k | so its 23 active incoming calls and 10 diff numbers to bring them in, each of which can have 23 lines etc..etc.. |
| 00:02.13 | Katty | nini meppl |
| 00:02.19 | meppl | ;) |
| 00:02.30 | Smi|k | and then the asterisk box can detect which DID and play the appropriate message |
| 00:02.38 | Smi|k | so many small businesses can share a single PRI t1 |
| 00:02.45 | JerJer | sure |
| 00:02.47 | Smi|k | especially if it is one that operates at night and another during the day |
| 00:02.57 | Smi|k | I see, very very good |
| 00:03.20 | Katty | JerJer: http://www.brick.net/~izaah/pictures/me/webcamarchive/ |
| 00:03.48 | Katty | it's updated every...uh...when i get around to it |
| 00:04.09 | *** join/#asterisk kusznir (~kusznir@nsl.evergreen.edu) |
| 00:04.53 | kusznir | Hello all: I have some questions about phone numbers and E911 as related to asterisk. |
| 00:05.44 | Katty | http://www.brick.net/~izaah/webcam.jpg <- shiny new katrs |
| 00:05.49 | Katty | i mean kats |
| 00:06.29 | kusznir | Specifically, if I run an asterisk system providing POTS termination (~1000 lines, all DID) and SIP trunking, can I somehow get my own block of phone numbers (US -- Washington)? If so, how? |
| 00:06.32 | P-NuT | careful she's nude.. |
| 00:07.13 | Katty | haha |
| 00:07.15 | Katty | you wish |
| 00:07.18 | P-NuT | LOL |
| 00:07.28 | P-NuT | I don't know you so no I don't. |
| 00:07.39 | makhtar | i do |
| 00:07.44 | Katty | good |
| 00:07.50 | P-NuT | LOL |
| 00:07.55 | P-NuT | maybe I do. ;-) |
| 00:08.04 | Katty | ... |
| 00:08.36 | makhtar | the women of asterisk calendar is great |
| 00:08.55 | Katty | obviously |
| 00:09.12 | makhtar | you are april, i believe? |
| 00:09.25 | P-NuT | there's a calendar? |
| 00:09.41 | *** join/#asterisk sedwards50 (~chatzilla@adsl-64-171-118-72.dsl.sndg02.pacbell.net) |
| 00:10.15 | P-NuT | do we have a url? |
| 00:10.43 | kusznir | My other question: how does one provide E911 service through SIP "trunking"? |
| 00:10.50 | kusznir | (or does one?) |
| 00:12.07 | sedwards50 | What is the "busiest" time of day on this IRC? I need some volunteers to call my "800" number so I can confirm trunk group rollover |
| 00:12.21 | *** join/#asterisk beto75 (~beto75@201.133.243.30) |
| 00:12.34 | beto75 | hello guys |
| 00:12.34 | Katty | makhtar: you think i'm april? heh |
| 00:13.05 | beto75 | guys I need a AMP guru ,, anyone here |
| 00:13.26 | *** join/#asterisk pjz (~pj@place.org) |
| 00:13.32 | pjz | howdy all |
| 00:13.55 | Smi|k | do any ISP's specialize in giving you a PRI T1, standard bandwidth, and space in rack for your asterisk server? |
| 00:14.29 | robin_sz | OK, so i *still* dont get dialtone |
| 00:14.30 | pjz | how do I make a context where if the user dials a '*' they get forwarded to another context? |
| 00:14.50 | pjz | is that possible? |
| 00:14.50 | makhtar | katty: yes, you are the one looking all pouty while configuring the boot server for some polycom ip500s. |
| 00:15.25 | Katty | i'm afraid you are mistaken |
| 00:15.35 | Katty | try again |
| 00:15.41 | makhtar | are they ciscos? |
| 00:15.41 | Katty | also, post gifs |
| 00:15.54 | Katty | jpgs optional |
| 00:15.56 | Katty | png too |
| 00:16.25 | Smi|k | how many DID's does a PRI T1 come with? 10? |
| 00:16.49 | robin_sz | OK, I found my dialtone .. panic over |
| 00:16.59 | *** join/#asterisk jmav (~jmav@201.243.100.68) |
| 00:17.04 | JunK-Y | Smi|k: as many as u want. |
| 00:17.24 | JunK-Y | u can have 300 DIDs with a T1 if ya want. |
| 00:17.30 | PatrickDK | hmm, you can't have a pri t1 |
| 00:17.34 | PatrickDK | it's pri, or t1 |
| 00:17.41 | PatrickDK | t1=data, pri= voice |
| 00:17.53 | JerJer | JunK-Y: if the carrier allows it |
| 00:17.59 | JunK-Y | PatrickDK: he probably means pri. |
| 00:18.07 | PatrickDK | and they came with as many did as you buy |
| 00:18.16 | JerJer | Patrick: PRI rides on T-1 signalling |
| 00:18.32 | JunK-Y | JerJer: of course, but many carrier will support almost anything, cause its more money for 'em. |
| 00:18.48 | PatrickDK | jerjer, ya, just I hate people using the wrong terms all the time |
| 00:19.16 | JunK-Y | and a t1 can be for voice if you're not using dchan. |
| 00:19.37 | JerJer | you can have a PRI on 12 channels and clear channel data on the other 12 |
| 00:19.39 | JerJer | we do it all day long |
| 00:19.53 | PatrickDK | ya, that is what I was going get, intergrated-t1 |
| 00:20.01 | JunK-Y | theres a lot of possibilities yeah. |
| 00:20.04 | PatrickDK | but then they told me, they wouldn't do it for any price |
| 00:20.13 | Smi|k | but I cant get a standard T1 and make it PRI |
| 00:20.16 | PatrickDK | no t1/pri to the locateion I needed |
| 00:20.19 | JerJer | find a real carrier then |
| 00:20.23 | JerJer | or move |
| 00:20.28 | JunK-Y | mouhaha |
| 00:20.33 | Smi|k | need to sign up and pay for a PRI t1 right? |
| 00:20.33 | PatrickDK | jerjer, kind of hard to move the business |
| 00:20.43 | JerJer | Smi|k: standard? |
| 00:20.49 | PatrickDK | smilk, you need pri, not t1 |
| 00:20.49 | Smi|k | data' |
| 00:20.50 | JerJer | what's 'standard'? |
| 00:20.55 | Smi|k | yep, I need pri |
| 00:20.57 | PatrickDK | you ask for t1, you will get data link, and no voice |
| 00:21.09 | Smi|k | any experience moving cell phone numbers to PRI DID's? |
| 00:21.20 | Smi|k | xo told me it was grey and they didnt know of anyone who does it |
| 00:21.22 | PatrickDK | good luck |
| 00:21.29 | pjz | will exten => _*X.,1,Goto(internal,${EXTEN:1},1) |
| 00:21.36 | pjz | what will that do? |
| 00:21.43 | PatrickDK | smilk, I doubt you will find anyone to do it |
| 00:21.47 | dant | PatrickDK, but you might work out it wasn't going to plan when they asked for the b end location for the t1 |
| 00:22.07 | robin_sz | OK, so .. my phoen can dial into * and all that .. and call voicemail, but it seems voicemail cant "hear" my username and password .. is it possible that im sending out-of-band DTMF thats fone for * but not for the voicemail? |
| 00:22.22 | *** part/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 00:22.25 | PatrickDK | dant, what? |
| 00:22.43 | dant | <PatrickDK> you ask for t1, you will get data link, and no voice |
| 00:23.01 | dant | pri goes to the telco, t1 goes from point a to b |
| 00:23.11 | dant | hence my comment :) |
| 00:23.27 | robin_sz | Incorrect password '' for user '5000' (context = <any>) |
| 00:23.35 | PatrickDK | dant, the b end was them, for inet access |
| 00:23.41 | PatrickDK | so that wasn't my issue |
| 00:23.42 | robin_sz | sigh ... but I am typing my pw |
| 00:23.43 | JerJer | sepcify a context |
| 00:24.02 | Smi|k | does telco have rack space available? |
| 00:24.08 | PatrickDK | smilk, yep |
| 00:24.11 | Smi|k | I need to get PRI T1 -> Asterisk -> DATA T1 |
| 00:24.30 | Smi|k | is the cheapest way to do that through telco rack space, or 3rd party with low-distance from telco, or someone like xo |
| 00:24.40 | PatrickDK | smilk, depends on your volume |
| 00:24.49 | Smi|k | volume of what? |
| 00:25.00 | JerJer | your iPod |
| 00:25.16 | Smi|k | how did you know I had an ipod? |
| 00:25.35 | Smi|k | kidding. volume of what though? calls? isint all 23 lines mine? outgoing I want to use voip providers, incoming pri t1 |
| 00:25.35 | JunK-Y | everyone has an ipod bro. |
| 00:25.35 | JunK-Y | :) |
| 00:25.43 | dant | Smi|k, totally depends on your provider |
| 00:26.10 | PatrickDK | telco space is the most expensive place to rent |
| 00:26.32 | *** join/#asterisk orlon_ (~apathy@office-fw.iexec.net.au) |
| 00:26.38 | PatrickDK | so it depends on exactly how many calls you make, or money you get to pay for it |
| 00:26.52 | orlon_ | mornin' all |
| 00:26.53 | PatrickDK | cause if you at the telco, the pri is practically free |
| 00:26.54 | Smi|k | expensive, but T1's have loop fee of $$$$$/month, I assume there is no loop fee if you get rack space at telco? |
| 00:27.00 | dant | Smi|k, that said, it would potentially be more cost effective to get a server colocated with a decent internet connection and a pri than it would be to get the same circuits to a remote location |
| 00:27.05 | Smi|k | no loop on pri t1, or data t1 |
| 00:27.24 | Smi|k | I see, thanks dant |
| 00:27.29 | PatrickDK | dant, you have any idea how hard it has been to get a phone line to my colo? let alone a pri? :) |
| 00:27.40 | orlon_ | anyone using res_config_odbc and #include's? or familiar with the res_config_odbc code? |
| 00:27.50 | Smi|k | what colos are set up with telco in mind |
| 00:28.04 | dant | PatrickDK, it all falls into the 'depends on the provider' bit :) |
| 00:28.05 | Smi|k | i.e. they put colo next to telco for low loop fees and they run multiple pri t1's into colo etc |
| 00:28.30 | PatrickDK | hmm, my colo doesn't deal with anything that small is the problem |
| 00:28.32 | JerJer | there shouldn't be a loop fee if you colo |
| 00:28.34 | PatrickDK | it's all oc48 or larger |
| 00:28.39 | JerJer | or they are raping you if they do charge one |
| 00:28.45 | Smi|k | no loop fee on a pri t1? |
| 00:28.59 | Smi|k | I thought colo uses data t1 and telco uses pri t1 |
| 00:29.08 | JerJer | if they drop it into a cabinet down the hall from their switch |
| 00:29.11 | PatrickDK | I hope a colo doesn't use a t1 |
| 00:29.18 | Smi|k | you know what I mean |
| 00:29.27 | PatrickDK | they normally run optic |
| 00:29.27 | Smi|k | are colo's connected to telco's or other data networks |
| 00:29.30 | JerJer | yeah lets hope they deliever you a DS-3 and a mux to break them out into T-1s |
| 00:29.30 | JerJer | then u can scale |
| 00:29.31 | blitzrage | Ariel_: how about turning off that auto-announce |
| 00:29.44 | Smi|k | i.e. colo cannot install a PRI T1 in my office |
| 00:29.46 | Smi|k | telco can |
| 00:29.56 | Smi|k | but can a colo install a pri t1 in their own office, or does the telco have to |
| 00:30.06 | JerJer | you are highly confused |
| 00:30.09 | Smi|k | I know |
| 00:30.11 | dant | Smi|k, colo's would tend to have plenty of telco presence in the building |
| 00:30.19 | Smi|k | got it dant |
| 00:30.46 | robin_sz | OKm .. and whats wrong with: |
| 00:30.48 | robin_sz | exten => 5100,(SIP/robin&SIP/elaine,10,t) |
| 00:30.50 | Smi|k | now, the DID I want is on a cell phone, how do I find a colo that has experience with # portability to DID's? |
| 00:31.02 | PatrickDK | rabin, all of it |
| 00:31.06 | JerJer | robin_sz: the word Dial |
| 00:31.15 | JerJer | and a priority |
| 00:31.16 | Smi|k | and would the colo have to be in the same exchange as the mobile phone number? |
| 00:31.28 | robin_sz | hmmm ... |
| 00:31.28 | PyroSteve | robin_sz: and the priority |
| 00:31.33 | dant | Smi|k, speak to a telco about colo, or, speak to a telco about services to a 3rd party colo facility |
| 00:31.47 | bewest | those priorities will get ya every time |
| 00:31.50 | robin_sz | OK, ... |
| 00:31.52 | PatrickDK | I thought they made priority optional? but you still need the cama right? |
| 00:32.17 | robin_sz | so what I want to do is ring all the phones in a group ... |
| 00:32.45 | PatrickDK | your can define groups, unless your using zaptel |
| 00:32.59 | PatrickDK | can't |
| 00:33.25 | PatrickDK | The only trick to get around that, is to use LOCAL/ |
| 00:33.26 | robin_sz | well, I want to ring 3 extensions if no one lese picks it up |
| 00:33.41 | pjz | PatrickDK: does'ting ringing SIP/ext1&SIP/ext2&SIP/ext3 simulate a group ring? |
| 00:33.53 | pjz | PatrickDK: or does it ring them one at a time? |
| 00:33.54 | PatrickDK | pjz, simulate yes, defines a group, no |
| 00:33.59 | *** join/#asterisk adiao (~adiaowudi@60.176.213.186) |
| 00:34.02 | PatrickDK | it rings all, first to pick up gets it |
| 00:34.04 | JerJer | PatrickDK: you can do groups without zaptel |
| 00:34.07 | JerJer | live in the now man |
| 00:34.07 | pjz | PatrickDK: okay |
| 00:34.13 | robin_sz | right .. thats what I want to do |
| 00:34.23 | robin_sz | but I get: |
| 00:34.27 | robin_sz | Invalid priority '(SIP/robin&SIP/elaine' at line 271 |
| 00:34.31 | PatrickDK | jerjer, you can? heh, I haven't upgrade for awhile, been to busy working |
| 00:34.44 | PatrickDK | robin, 2 people told you how to fix that now |
| 00:34.45 | pjz | I programatically generate my extensions.conf, so simulation is the same as reality ofr me :) |
| 00:34.45 | Nivex | robin_sz: sounds like you have other problems on your Dial line |
| 00:35.04 | robin_sz | Patrick, no .. two people might think thye have ... |
| 00:35.21 | PatrickDK | robin, we will not spell out the whole line for you |
| 00:35.22 | Smi|k | who does the actual portability of cell # to DID for pri t1 though? the colo or the telco or? |
| 00:35.26 | PatrickDK | scroll back and read again |
| 00:35.29 | pjz | robin_sz: you need 'Dial' |
| 00:35.37 | *** join/#asterisk bobessutio (~william@c-67-180-96-152.hsd1.ca.comcast.net) |
| 00:35.41 | Nivex | robin_sz: you need 5100,1,... instead of 5100,... |
| 00:35.45 | pjz | robin_sz: JerJer told you straight up |
| 00:35.46 | Smi|k | telco's pretend to have no clue how to move a cell number to a DID |
| 00:35.53 | PatrickDK | pjz, heh, I use local for my groups, so I can dynamically add/remove people from the group |
| 00:36.14 | pjz | PatrickDK: I'm too much of a newbie to know that trick |
| 00:36.19 | robin_sz | doh! ... dial |
| 00:36.25 | PyroSteve | robin_sz: why dont you give asterisk@home a try |
| 00:36.26 | pjz | PatrickDK: if I do exten => _*X.,1,Goto(internal,${EXTEN:1},1) |
| 00:36.36 | pjz | PatrickDK: will that... work? |
| 00:36.53 | dant | Smi|k, telco deals with telephony, colo deals with server colocation, but, both could be the same company |
| 00:36.59 | PatrickDK | pjz, that works for flow |
| 00:37.08 | pjz | PatrickDK: right |
| 00:37.09 | PatrickDK | but doesn't let you do other things with the call |
| 00:37.18 | pjz | PatrickDK: well, the 'internal' context is all my extensions |
| 00:37.21 | PatrickDK | see, I have stuff I do for each phone a ring |
| 00:37.25 | pjz | PatrickDK: this is for outgoing calls |
| 00:37.28 | PatrickDK | you can't do different stuff for each phone |
| 00:37.36 | PatrickDK | without using LOCAL to loop back to call each phone |
| 00:37.58 | PatrickDK | like one phone might have callforwarding, another DND |
| 00:38.07 | JerJer | asterisk@home is a joke |
| 00:38.47 | pjz | PatrickDK: so folk can pick up and dial either * and get to a context where all my extensions are defined or they can just dial the outbound number they want |
| 00:38.50 | PyroSteve | well it might be to most of us, but for someone who doesn't understand how to write a line in the dialplan |
| 00:38.51 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
| 00:39.06 | PyroSteve | , its probably everything in the world |
| 00:39.33 | PatrickDK | people need to look at the examples more |
| 00:39.36 | PatrickDK | there are enough of them |
| 00:40.14 | PatrickDK | and you would also thing they would ring one phone, before they attempt ringing two at a time |
| 00:40.18 | pjz | any of them with a good explanation attached? |
| 00:40.34 | robin_sz | sigh |
| 00:40.39 | PatrickDK | heh, explanations optional :) |
| 00:40.43 | pjz | :P |
| 00:41.03 | PatrickDK | if you can see something that works, start with it, and modify it till it's broken |
| 00:41.06 | pjz | this goes live Monday, I have to have it at least minimally functional |
| 00:41.08 | robin_sz | yes, thankyou I did ring one phone before attempting to ring two |
| 00:41.11 | PatrickDK | then change it back, and try again :) |
| 00:41.12 | *** join/#asterisk tiko_ (~root@218.108.175.19) |
| 00:41.20 | robin_sz | yes thankyou I can write a line a in a dialplan |
| 00:41.34 | pjz | is there an example of how to make a recording extension? |
| 00:41.48 | orlon_ | pjz: search voip-info.org |
| 00:41.48 | PatrickDK | robin_sz, well, you had two very big mistakes in it, and wouldn't take out help when you asked |
| 00:41.51 | pjz | I need a way to record the voiceprompts |
| 00:42.00 | PatrickDK | pjz, there used to be |
| 00:42.17 | PatrickDK | I would assume there still is |
| 00:43.05 | robin_sz | PatrickDK: Im sure you mean well, but .. consider for a moment that someone saying "the word dial" might not exactly trigger an understanding of "you missed out the function Dial" |
| 00:43.49 | pjz | PatrickDK: if I've got an exten => _X.,... and also an exten => *,... will the * ever match? |
| 00:43.50 | robin_sz | it not a question of not accepting help .. its a question of the help not being understood .. if yo uknow the answer, its obvious. if you dont .. ... |
| 00:44.20 | PatrickDK | [20:31] <JerJer> robin_sz: the word Dial |
| 00:44.20 | PatrickDK | [20:31] <JerJer> and a priority |
| 00:44.41 | robin_sz | means someting to me now, meant *NOTHING* ot me at the time |
| 00:44.53 | PatrickDK | pjz, hmm, last I knew * wasn't valid |
| 00:44.59 | PatrickDK | oh wait, heh |
| 00:45.11 | pjz | PatrickDK: ... |
| 00:45.17 | PatrickDK | pzj, yes it will, cause _X. matchs digits not * |
| 00:45.49 | pjz | PatrickDK: ahh... okay |
| 00:46.44 | blitzrage | X matches numbers 0-9 |
| 00:46.56 | PatrickDK | I guess X doesn't match ABCD |
| 00:47.06 | PatrickDK | I wonder though about that |
| 00:47.08 | robin_sz | pjz: still need the record thing? |
| 00:48.28 | blitzrage | no, it doesn't match ABCD |
| 00:49.08 | robin_sz | ooh, and I am hating this grandstream even more ... |
| 00:49.36 | robin_sz | the not sending digits till you press send is a PITA |
| 00:49.54 | bewest | it sends it |
| 00:49.59 | bewest | after 6 seconds or so? |
| 00:50.04 | bewest | I think that's common |
| 00:50.05 | robin_sz | hmm .. wait .. |
| 00:50.07 | bewest | dunno though |
| 00:50.13 | bewest | you can also press # |
| 00:50.18 | Nivex | cell phones don't dial until you hit send |
| 00:50.22 | bewest | I'm so used to it, I do it on normal phones too |
| 00:50.50 | *** join/#asterisk shidan (~shidan@CPE000625dbadc2-CM014280007905.cpe.net.cable.rogers.com) |
| 00:50.50 | robin_sz | heh |
| 00:50.58 | robin_sz | yeah .. 6 seconds or so |
| 00:51.32 | bewest | I like having to hit #/send |
| 00:51.42 | bewest | gives you a chance to add a little flourish |
| 00:52.17 | *** join/#asterisk Nukemizer (~Nuke@160.7.249.15) |
| 00:52.22 | shidan | anyone know if its possible to do billing/cdrs when using sip reinvites, not only asterisk but in general |
| 00:53.01 | robin_sz | i thought reinvites basically bypassed * ?? |
| 00:53.19 | cluecon[file] | no |
| 00:53.28 | cluecon[file] | a reinvite just renegotiates where your audio goes |
| 00:53.33 | robin_sz | ahh |
| 00:53.35 | cluecon[file] | the signalling (hangups/etc) still go through asterisk |
| 00:53.41 | robin_sz | but the control still goes through * |
| 00:53.45 | robin_sz | right |
| 00:53.54 | cluecon[file] | on IAX2 however, native transfers take the middle asterisk box totally out of the loop |
| 00:54.24 | shidan | so if the control goes thru asterisk u should be able to do billing with sip on reinvites |
| 00:54.43 | shidan | but then when using ser why do they say u have to use a B2BUA?? |
| 00:54.43 | cluecon[file] | yes |
| 00:55.41 | cluecon[file] | shidan: just accept it... if you wanna learn more, read up on what everything is |
| 00:56.07 | robin_sz | not really had chance to play with it as an exchange ... just used it to deliver live audio. the dialplan is *fun* ;) |
| 00:56.54 | *** join/#asterisk at561 (~angry@68.71.213-254.atlsfl.adelphia.net) |
| 00:57.02 | at561 | asterisk should join the summer of code program |
| 00:57.21 | shidan | so in other words u dont know cluecon why in ser u need a B2BUA |
| 00:57.27 | JerJer | at561: what are we going to code? |
| 00:57.27 | shidan | cools |
| 00:57.33 | cluecon[file] | shidan: no I know |
| 00:57.41 | cluecon[file] | but why should I spend my time explaining SIP internals and what everything is? |
| 00:57.43 | cluecon[file] | when you can look it up |
| 00:57.46 | *** join/#asterisk Jazzholio (~jazzholio@61.197.221.203.velocitynet.com.au) |
| 00:57.57 | cluecon[file] | free help doesn't pay my bills :) |
| 00:57.57 | robin_sz | at561: a web-based client please. |
| 00:58.11 | shidan | I never asked u but u can just shutup and let someone else who feels like answering tell me instead of sayin just accept it |
| 00:58.40 | cluecon[file] | lol |
| 00:58.44 | newl | <JunK-Y> and a t1 can be for voice if you're not using dchan. <-- if it's used for voice, where does the call data from the exchange to the remote end come from if the d channel isn't used? <g> |
| 00:58.53 | at561 | yeah, one of the tedious but simple projects is good for students |
| 00:58.58 | shidan | its all good file ;) |
| 00:59.21 | at561 | make them document everything or write a configurator |
| 00:59.29 | JerJer | make menuconfig |
| 00:59.34 | cluecon[file] | JerJer: I'm trying to get egold transferred into my Paypal, but god help me... |
| 01:00.15 | JerJer | hmm egold |
| 01:00.30 | *** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
| 01:00.32 | JerJer | interesting |
| 01:00.39 | cluecon[file] | apparently I have like 5 ounces of gold? |
| 01:00.59 | cluecon[file] | I dunno, cause the site is down |
| 01:01.08 | shidan | The e-gold system is temporarily offline |
| 01:01.11 | shidan | hahahaha |
| 01:04.27 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 01:05.13 | Jazzholio | ok.. so who are the SIP transfer experts here? ;) |
| 01:05.23 | at561 | maybe the students could implement E911 |
| 01:05.30 | at561 | how are the LECs going to provide access? |
| 01:05.38 | cluecon[file] | Jazzholio: I didn't raise my hand, honest |
| 01:06.15 | Jazzholio | I'm just trying to find out if attended xfers will work in 1.0.7-stable.. or do i need to suck down a CVS build? |
| 01:06.18 | JerJer | the LECs are going to let cLECs specialize in PSAP interconnections |
| 01:06.21 | robin_sz | wahey ... voicemail sorted :) phone was set to in-band DTMF, * was set to SIP Info |
| 01:07.11 | Jazzholio | no takers? :( |
| 01:09.26 | shidan | what do u mean specialize |
| 01:09.30 | orlon_ | JerJer: who'd be the best person to talk to wrt a res_config_odbc issue? |
| 01:09.48 | robin_sz | nighguys ... |
| 01:09.57 | orlon_ | i don't want to submit a bug in Mantis until i'm sure it's not my own stupid fault |
| 01:11.11 | shidan | theyre still going to do all the ani/ali controlling? |
| 01:11.13 | *** join/#asterisk file[mac] (~jcolp@mctn1-4363.nb.aliant.net) |
| 01:11.23 | at561 | that doesn't sound very organized JerJer |
| 01:12.53 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 01:13.32 | *** join/#asterisk Mavvie (edwin@dialin-83.barnet.com.au) |
| 01:17.23 | *** join/#asterisk Druken (~druken@CPE00119539b9cc-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 01:17.50 | *** join/#asterisk DEEZED (deezed@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 01:19.06 | *** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3771041.sympatico.ca) |
| 01:22.11 | *** part/#asterisk Jazzholio (~jazzholio@61.197.221.203.velocitynet.com.au) |
| 01:23.42 | JerJer | res_config is not the solution either |
| 01:24.11 | file[mac] | there's only one way to make your asterisk box scale far |
| 01:24.17 | file[mac] | currently. |
| 01:24.25 | pjz | buy a bigger box? |
| 01:24.33 | file[mac] | pfft |
| 01:24.51 | orlon_ | JerJer: was that in response to my question? |
| 01:26.19 | JerJer | that's my response, duno if it was the response you were looking for |
| 01:26.57 | *** join/#asterisk dersteer (~travis@24-231-151-119.dhcp.aldl.mi.charter.com) |
| 01:27.47 | orlon_ | umm..ok |
| 01:31.59 | pjz | file[mac]: and that way is? |
| 01:32.11 | pjz | file[mac]: or are you just going to keep us plebes in suspense? |
| 01:33.18 | file[mac] | suspense. |
| 01:40.41 | *** join/#asterisk adiao (~adiao@218.108.175.19) |
| 01:40.57 | adiao | exit |
| 01:42.08 | timecop | <PROTECTED> |
| 01:42.09 | timecop | noice? |
| 01:42.31 | orlon_ | noise |
| 01:42.40 | timecop | perhaps someone should fix the typo |
| 01:45.34 | JerJer | timecop what file rtp.c ? |
| 01:49.56 | JerJer | fixed in -head |
| 01:51.41 | *** join/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com) |
| 01:55.21 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 01:58.10 | *** join/#asterisk santiago (~santiago@63.245.86.198) |
| 01:59.17 | Godsey | I've found SER |
| 01:59.36 | Godsey | it should integrate well w/ my asterisk setup and provide nice radius accounting |
| 01:59.53 | JerJer | if you say so |
| 02:00.05 | JerJer | SER rocks, but not radius |
| 02:00.14 | JerJer | for VoIP |
| 02:02.05 | Godsey | radius rocks |
| 02:02.30 | JerJer | for what it was designed for, most def |
| 02:02.38 | JerJer | but not for VoIP accounting |
| 02:02.46 | shidan | what about cdrs for ser |
| 02:02.47 | Godsey | voip accounting is no different |
| 02:02.54 | JerJer | once you've been bitten by it you will wish you listened |
| 02:02.59 | JerJer | bullshit |
| 02:03.09 | shidan | u ever used it with reinvites only? |
| 02:03.20 | Godsey | same piece of hardware is used for voip and modems |
| 02:03.22 | Godsey | same PRIs |
| 02:03.25 | JerJer | voip accounting is hardcore different than say wireless device or modem accounting |
| 02:03.28 | Godsey | I don't see how accounting should be any different |
| 02:03.51 | JerJer | continue to deploy radius with voip and you will see |
| 02:04.14 | JerJer | but then you will have production traffic and will not be able to change things |
| 02:04.17 | JerJer | so you will be stuck with it |
| 02:04.25 | JerJer | and then will begin to hate yourself |
| 02:04.28 | JerJer | trust me |
| 02:04.29 | Godsey | it works fine |
| 02:04.35 | Godsey | we have 5 as5400s |
| 02:04.49 | Godsey | all on lan, it's not like the traffic is a problem |
| 02:04.58 | JerJer | if you say so |
| 02:05.23 | shidan | have u tried to use ser with reinvites only and do billing with it |
| 02:05.34 | shidan | I cant get the cdr to fill |
| 02:05.36 | timecop | holy fuck |
| 02:05.43 | timecop | so i downloaded asterisk_prepaid thing |
| 02:05.55 | doolph | hello |
| 02:06.24 | timecop | http://rafb.net/paste/results/QJaJzN18.html |
| 02:06.32 | shidan | writing a prepaid app isnt hard for asterisk I dont like any of the ones out there |
| 02:06.34 | timecop | can someome please tell me i'm not insane? |
| 02:06.40 | timecop | why hte fuck |
| 02:06.42 | timecop | would this even be released |
| 02:06.46 | timecop | with bugs like these? |
| 02:07.14 | Godsey | have you not noticed asterisk is riddled w/ bugs? :) |
| 02:07.18 | shidan | timecop: do u care if its an AGI or a module the prepaid app |
| 02:07.19 | timecop | heh |
| 02:07.22 | Katty | just to annoy the hell out of you |
| 02:07.36 | timecop | shidan: at this point, no, i'm just trying to get something together to demonstrate. |
| 02:07.46 | timecop | and i'm guessing i'll probably end up writing my own sooner or later. |
| 02:07.50 | Katty | if you think you can do a better job, why don't you rewrite the whole thing, timecop |
| 02:07.53 | JerJer | timecop: because whoever wrote that app has no clue? |
| 02:07.57 | timecop | JerJer: well yeah |
| 02:08.01 | timecop | but sitll |
| 02:08.04 | shidan | ok I have a prepaid app that works fine |
| 02:08.12 | timecop | i mean i found one bug like that, and i was like, ok, must have ben a mistake |
| 02:08.12 | Godsey | heck ser can do radius auth too |
| 02:08.20 | timecop | then I keep looking down and its same shit evertywhere |
| 02:08.22 | timecop | wtf |
| 02:08.26 | shidan | I modified astcc to handle multiple channels properly |
| 02:08.35 | shidan | send me a note if u want to take a look sometime |
| 02:08.40 | timecop | shidan: astcc, its a app_something? |
| 02:08.50 | timecop | i think that what im fuckign with actually |
| 02:08.54 | shidan | no its the sample agi prepaid app |
| 02:08.55 | shidan | no |
| 02:08.59 | timecop | ah |
| 02:09.05 | *** join/#asterisk nifter (~nifter@190-141.SPEEDe.golden.net) |
| 02:09.06 | shidan | your using the worst piece of garbage out there |
| 02:09.10 | timecop | hehehe |
| 02:09.14 | Godsey | I love AGI |
| 02:09.30 | shidan | Im going to recode it as a module soon tho |
| 02:09.39 | timecop | what's it in now? perl I guess? |
| 02:09.43 | shidan | AGI is clean and simple |
| 02:09.47 | JerJer | too bad agi does not scale |
| 02:09.48 | shidan | no i rewrote it in python |
| 02:10.01 | Godsey | jerjer why not? |
| 02:10.04 | timecop | lol @ python |
| 02:10.06 | timecop | no thanks |
| 02:10.14 | timecop | i'll keep unfucking this app_ then until it works |
| 02:10.15 | doolph | ... |
| 02:10.28 | shidan | haha why lol @ python |
| 02:10.34 | JerJer | lets see... parse compile, execute times 150 channels |
| 02:10.40 | Godsey | yes? |
| 02:10.46 | shidan | its true agi doesnt scale |
| 02:10.55 | shidan | weve already outgrown it |
| 02:10.56 | Godsey | we have 3100 channels |
| 02:11.30 | Godsey | i don't think we've hit maximum capasity yet but at least 1000 |
| 02:12.15 | Godsey | I use perl for my AGI scripts |
| 02:12.34 | timecop | i prefer to use sometihng I can at least understand. |
| 02:12.35 | timecop | like C. |
| 02:12.45 | Nuxi | or asm |
| 02:12.50 | Nuxi | much better than perl |
| 02:12.51 | shidan | ok u can understand C but not python or perl?? |
| 02:13.01 | shidan | thats messed up |
| 02:13.07 | timecop | how is that messed up? |
| 02:13.07 | Nuxi | python is ok. |
| 02:13.27 | shidan | because if u understand c then python should be like reading english |
| 02:13.36 | shidan | I like C too |
| 02:13.37 | timecop | except python sucks. |
| 02:13.51 | shidan | how so |
| 02:14.00 | Nuxi | because if u understand c then <insert language here> should be like reading english |
| 02:14.06 | Nuxi | because if u understand c then cobol should be like reading english |
| 02:14.12 | timecop | idont fucking get it. with so many bugs in this code, how the fuck was this ever released |
| 02:14.13 | shidan | thats not true |
| 02:14.16 | timecop | am I looking at some old version or somethign |
| 02:14.26 | shidan | try reading haskell with your c skills |
| 02:14.31 | shidan | doesnt work out that well |
| 02:14.35 | timecop | asterisk-prepaid-0.3.1 |
| 02:14.40 | timecop | ^^ latest? |
| 02:14.43 | Godsey | I still don't see how perl or whatever else doesn't scale |
| 02:14.51 | Godsey | network will give up before memory or cpu |
| 02:14.56 | shidan | perl does scale agi doesnt Godsey |
| 02:15.06 | shidan | I dont know the status of res_perl |
| 02:15.08 | *** part/#asterisk jeffik (~Jeff@69.158.21.177) |
| 02:15.08 | Nuxi | does fastagi scale? |
| 02:15.09 | shidan | if it works or not |
| 02:15.12 | shidan | but that should work |
| 02:15.18 | shidan | for scaling |
| 02:15.21 | Godsey | all I know is I fork perl all the time w/ no problem |
| 02:15.23 | shidan | if it works that is |
| 02:15.38 | Godsey | I have 8gigs in my machine |
| 02:15.54 | Godsey | if I hit ram problems I'll toss in another 8gig |
| 02:15.57 | JerJer | plus the AGI interface is very limiting |
| 02:16.09 | Godsey | I only set variables w/ agi for the most part |
| 02:16.15 | JerJer | i prefer to use the asterisk C language API and get the job done right the first time |
| 02:16.39 | doolph | Anyone have any sample code out there to register with an h323 gateway |
| 02:16.39 | doolph | (like avaya ip office 403) as an h323 trunk? |
| 02:16.41 | shidan | Theres an API ?? LOL |
| 02:16.44 | Nuxi | Only setting variables doesn't scale. If you have to call your agi more than once, you might as well put some dial logic in there too. |
| 02:16.50 | shidan | well I guess technically theres one |
| 02:16.54 | *** part/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3771041.sympatico.ca) |
| 02:17.14 | Godsey | Nuxi: I could just don't know how |
| 02:17.24 | Godsey | I use dial plan to goto and jump around |
| 02:17.32 | Godsey | set time remaining on account etc... |
| 02:18.42 | Godsey | I'm not a C programmer or I would probably take that route |
| 02:18.52 | Godsey | I'm just getting into C# |
| 02:19.00 | Godsey | I like asp.net 2 ;) |
| 02:19.02 | orlon_ | doolph: you using chan_h323 or chan_oh323? |
| 02:19.10 | Nuxi | As the kids said, if you program it in c the first time, it's done. |
| 02:19.26 | doolph | oh323 |
| 02:19.29 | Godsey | I started w/ asm68k |
| 02:19.30 | Godsey | :) |
| 02:19.36 | Godsey | then pascal |
| 02:19.41 | shidan | C# is the best language around by far |
| 02:19.51 | Godsey | and modula2 |
| 02:19.56 | Nugget | nah, everyone should be using modula-2 |
| 02:20.06 | Godsey | I didn't have a c compiler for my amiga :P |
| 02:20.19 | Godsey | till later and I was into some compiled basic language |
| 02:20.33 | Godsey | gwbasic I think |
| 02:20.37 | Nugget | that's becaue amigas sucked. |
| 02:20.43 | Nugget | I had a c compiler for my atari st! |
| 02:20.48 | Godsey | i had an st too |
| 02:20.51 | Nuxi | hmmmmm. I think asteriskwin32 with qbasic for agi should scale...right? |
| 02:20.57 | Godsey | but didn't like the b/w :) |
| 02:21.11 | Nugget | b/w? |
| 02:21.42 | Godsey | black / white |
| 02:21.50 | Nugget | oh, you didn't have a color monitor? |
| 02:22.01 | Godsey | nog gwbasic, gfa basic |
| 02:22.31 | Godsey | heh I guess not |
| 02:22.32 | Nuxi | my first agi on windows was a batch file. |
| 02:22.35 | *** join/#asterisk swk (~ken@12-219-156-206.client.mchsi.com) |
| 02:22.38 | Druken | there's still monocrome monitors in exsistance? |
| 02:22.45 | kimo_sabe | Nuxi: eww |
| 02:22.57 | Godsey | I'd like to use webservices for agi! |
| 02:23.03 | shidan | my first cgi was an awk file |
| 02:23.23 | Godsey | my first cgi was perl 4 |
| 02:23.26 | Nuxi | I wanted to give phpagi a windows compatibility test on asteriskwin32. Woo hoo, it worked. |
| 02:23.38 | shidan | Godsey: theres already an xml-rpc interface to asterisk |
| 02:24.05 | Godsey | I don't even know if I actually need asterisk if I get SER working |
| 02:24.17 | shidan | what do u want to use ser for |
| 02:24.22 | shidan | Im trying to set it up too |
| 02:24.24 | Godsey | accounting |
| 02:24.27 | shidan | well I have three times |
| 02:24.30 | Nuxi | You can't play blackjack with just ser. |
| 02:24.38 | Godsey | I am writing a service |
| 02:24.45 | Godsey | Nuxi: I was thinking of making a zork agi |
| 02:24.48 | shidan | but one potential client wants ser set up with reinvites only |
| 02:24.48 | Druken | ser is hard to setup ? |
| 02:25.00 | shidan | for their cdr record as well |
| 02:25.05 | Godsey | stripped down zork w/ only 9 options |
| 02:25.16 | shidan | any way to do that |
| 02:25.17 | shidan | ? |
| 02:25.25 | Nuxi | If you get it working you should post it to the wiki or put it as example for the toolkit you are using. |
| 02:25.29 | Godsey | I only found ser this evening |
| 02:25.39 | doolph | omg exists asterisk windows |
| 02:25.51 | Godsey | Nuxi: you can parse those zork and many other files w/ a perl module |
| 02:25.51 | Nuxi | asteriskwin32.com |
| 02:25.58 | Godsey | infocom I think it's called |
| 02:26.07 | shidan | Its just a cygwin port no? |
| 02:26.28 | shidan | U might as well use Yate if u want something on windows |
| 02:26.33 | Nuxi | yup. You'll have to build php in cygwin for phpagi to work. perl is already installed in it. |
| 02:26.43 | orlon_ | i think the win32 * stuff is running under CoLinux |
| 02:26.54 | Nuxi | yup, that too. |
| 02:27.09 | Godsey | I love asterisk's dialplan flexibility |
| 02:27.16 | Nuxi | The cygwin port doesn't support eagi. |
| 02:27.21 | Druken | shidan: is there some underlying reason why the toronto asterisk group always meets in a bar?? :) |
| 02:27.30 | shidan | ya we like drinking |
| 02:27.48 | Druken | hehe at least yer honest |
| 02:28.07 | shidan | what do u mean at least its a great reason to meet in a bar |
| 02:28.54 | Druken | i guess hehe |
| 02:29.04 | Druken | personally i haven't attended yet |
| 02:29.05 | timecop | opensource fucking sucks. |
| 02:29.07 | shidan | have u been to any meetings |
| 02:29.15 | shidan | ahh ok |
| 02:29.21 | shidan | its a good crowd |
| 02:29.33 | Nuxi | That's why I only used closed source * |
| 02:30.20 | Sedorox | Hmmmm |
| 02:30.29 | Druken | shidan: one of these days i'll make it to a meeting... but i can't see it being in the near future |
| 02:30.33 | shidan | so can anybody tell me how sip reinvites work if control stays with me then why doesnt my ser box get any of the bye's |
| 02:31.10 | orlon_ | shidan: have you tried running a tcpdump infront of one of the endpoints. see where it's sending it's BYE to? |
| 02:31.11 | Druken | a reinvite removes the middle from the path |
| 02:31.12 | shidan | we are having a dundi workshop soon by blitzrage |
| 02:31.12 | orbi | orlon_: how can you tell? I mean, most trolls are given away by the tell-tale sign of not bathing. Many geeks on coding binges are known not to bathe. |
| 02:31.27 | shidan | totally removes it |
| 02:31.31 | Druken | yeah |
| 02:31.43 | Druken | once the call is reinvited you cannot track the CDR |
| 02:31.51 | shidan | thats what I thaught but then file said its not so |
| 02:31.56 | orlon_ | orbi: good point =] |
| 02:31.57 | shidan | I thaught so |
| 02:32.01 | JerJer | closed sourced * ?????? |
| 02:32.10 | Druken | it's been my experince that you cannot |
| 02:32.16 | shidan | so why is sip any good |
| 02:32.20 | cluecon[file] | record routing |
| 02:32.31 | JerJer | timecop: then dont' fucking use it |
| 02:32.43 | cluecon[file] | your box has to record route itself in the SIP messages to ensure that the messages then travel through it again |
| 02:32.58 | timecop | heh |
| 02:33.00 | cluecon[file] | otherwise it'll go direct |
| 02:33.01 | timecop | hey jerjer |
| 02:33.09 | timecop | anything technically preventing H323 from listening on more than one network interface? |
| 02:33.10 | shidan | record routing is statefull but the rtp stream is separate |
| 02:33.14 | shidan | ahhh i gotcha |
| 02:33.16 | doolph | Anyone have any sample code out there to register with an h323 gateway |
| 02:33.18 | doolph | (like avaya ip office 403) as an h323 trunk? |
| 02:33.32 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 02:33.35 | Godsey | shidan: I think some sip devices continue sending information after reinvite, others don't |
| 02:33.35 | cluecon[file] | the term 'reinvite' means a new INVITE with the same callid, but with different SDP data |
| 02:33.38 | *** join/#asterisk AlexCeli (~Alex@200.37.85.91) |
| 02:33.48 | cluecon[file] | which contains different RTP address/port info |
| 02:34.16 | Druken | so it can work kinda like FXP ? |
| 02:34.24 | cluecon[file] | FXP? |
| 02:34.39 | shidan | ok I see I was thinking messed up the B2BUA is only staying in between the sip not the rtp |
| 02:34.40 | shidan | cools |
| 02:34.46 | Druken | uhmm... control is on the client, but two servers send data to each other |
| 02:34.50 | cluecon[file] | ah yes |
| 02:34.56 | cluecon[file] | as for a B2BUA, depends |
| 02:35.17 | cluecon[file] | in asterisk B2BUA world, they are two separate calls two separate rtp streams |
| 02:35.21 | timecop | <timecop> hey jerjer |
| 02:35.21 | timecop | <timecop> anything technically preventing H323 from listening on more than one network interface? |
| 02:35.31 | cluecon[file] | but with a reinvite, instead of having two separate rtp streams... you can have them go direct to eachother |
| 02:35.40 | cluecon[file] | still two separate calls mind you |
| 02:35.51 | cluecon[file] | acts as a user agent on both sides.. |
| 02:35.55 | Druken | sounds like a CDR nightmare |
| 02:35.56 | cluecon[file] | thus the name, Back to Back User Agent |
| 02:36.03 | cluecon[file] | B2BUAs make CDRs easy |
| 02:36.12 | JerJer | timecop: open source fucking sucks, says you ... then u want support on my open source driver? |
| 02:36.13 | cluecon[file] | the signalling will ALWAYS go through you |
| 02:36.21 | cluecon[file] | because it's two separate calls :) |
| 02:36.28 | timecop | JerJer: :( |
| 02:36.32 | shidan | ok that makes sense |
| 02:36.35 | cluecon[file] | if you're acting as a proxy, then it's not for certain... |
| 02:36.42 | cluecon[file] | unless you record route, and that stuff is honored properly |
| 02:37.32 | timecop | JerJer: no really |
| 02:37.49 | shidan | file: what do people do for prepaid apps in ser?? |
| 02:38.41 | cluecon[file] | I have my own platform |
| 02:38.42 | JerJer | ser doesn't do that |
| 02:38.46 | shidan | Now that I know your not talking shit Ill serioulsy paypal u 5 bucks |
| 02:38.48 | shidan | hahahaha |
| 02:39.02 | shidan | oh ok |
| 02:39.03 | cluecon[file] | I got paid lots of money to write it :P |
| 02:39.08 | shidan | nice |
| 02:39.09 | JerJer | ser is a sip proxy, registrar or redirect server |
| 02:39.12 | cluecon[file] | because, as JerJer said, ser is not designed to do it |
| 02:39.29 | shidan | but your app works with ser right? |
| 02:39.32 | shidan | do u have a link to it |
| 02:39.45 | cluecon[file] | uh no |
| 02:39.51 | cluecon[file] | it exists only on three boxes in existence :P |
| 02:40.05 | cluecon[file] | and it's not for sale |
| 02:40.13 | shidan | ahh ok |
| 02:40.50 | Godsey | ser can use radius |
| 02:41.02 | Godsey | so any isp management software can handle it |
| 02:41.15 | JerJer | blah |
| 02:41.21 | Druken | asterisk can use radius |
| 02:41.30 | cluecon[file] | you guys don't understand what it takes |
| 02:41.35 | timecop | so jerjer, anythign technically preventing h323 from listening on more than one network interface? |
| 02:42.18 | PyroSteve | <PROTECTED> |
| 02:42.23 | PyroSteve | <PROTECTED> |
| 02:42.28 | PyroSteve | <PROTECTED> |
| 02:42.29 | timecop | nice. |
| 02:42.33 | PyroSteve | opps |
| 02:42.35 | PyroSteve | sorry |
| 02:42.35 | Druken | uhmm... |
| 02:42.58 | PyroSteve | didn't know the crap on my desk could type |
| 02:43.18 | shidan | file : your prepaid app, is it a ser module u built, or do u use asterisk for that? |
| 02:43.44 | PyroSteve | I guess the crap on my desk grew arms and fingers instead of legs |
| 02:43.46 | cluecon[file] | it's not a prepaid app |
| 02:44.04 | shidan | what is it then |
| 02:44.04 | cluecon[file] | it's a platform is modularized so you can easily add a prepaid thing in if needed... |
| 02:44.10 | cluecon[file] | and it does not use asterisk |
| 02:44.12 | PyroSteve | hopefully this gets me flamed and not banned but would somone be interested in backdoring asterisk ? |
| 02:44.14 | shidan | oh ok |
| 02:44.14 | cluecon[file] | er that is |
| 02:44.23 | shidan | so its like a whole system |
| 02:44.26 | shidan | like asterisk |
| 02:44.35 | Godsey | there is always sipx :P |
| 02:44.59 | *** join/#asterisk iq (~iq@204-26-74-86.omah.qwest.net) |
| 02:45.02 | cluecon[file] | no, asterisk is a PBX... |
| 02:45.11 | Druken | can't the sip implimentation in asterisk be improved? |
| 02:45.32 | *** join/#asterisk mrplum (~mrplum@24-52-166-190.lndnnh.adelphia.net) |
| 02:45.38 | JerJer | define improved |
| 02:45.43 | shidan | its also a gateway and a media server ;) |
| 02:45.59 | Druken | uhmm.... unbroken ? |
| 02:46.42 | JerJer | what's broken about it? we do hundreds of thousands of SIP calls thru asterisk boxes without issue |
| 02:47.07 | Druken | is the invites not broken? |
| 02:47.10 | cluecon[file] | the SIP implementation is good enough that for most things, it works fine |
| 02:47.17 | Godsey | Druken: I've not had a problem |
| 02:47.37 | Juggie | works fine for me |
| 02:47.38 | Godsey | wouldn't invites be a problem w/ the end hardware? |
| 02:48.45 | shidan | ya it doesnt work with some gateways |
| 02:48.59 | Nuxi | such as? |
| 02:49.26 | shidan | give me 5 mins Ill find out |
| 02:50.24 | shidan | well of the top of my head I know it doesnt work with versatel |
| 02:50.30 | Juggie | theres no way i can tell a phone to place a call through sip is there... |
| 02:50.31 | JerJer | figure out why |
| 02:50.38 | shidan | but thats not the big one |
| 02:50.39 | JerJer | instead of just bitching about it |
| 02:50.42 | Juggie | eg, the phone just goes on speaker phone and places the call. |
| 02:50.51 | *** join/#asterisk PBXtech (nik@63-226-102-79.slkc.qwest.net) |
| 02:51.38 | shidan | its flaky with mci sometimes I dont know what they use thats just my experiences but I know others have had issues Nuxi |
| 02:52.08 | Druken | Juggie: you mean make the phone dial without physically touching it? |
| 02:52.50 | Juggie | yeah.... |
| 02:53.07 | Juggie | the mitel voip implementation we have at the office does that... i assume it logs into the phone and tells it to do so |
| 02:53.33 | Juggie | i know i could do it by telnetting into the phone and telling the phone to do it, but i was wondering if theres someway to ask the phone to do someting via sip |
| 02:53.35 | Godsey | are you sure it's sip? |
| 02:53.37 | Juggie | thats standardized |
| 02:53.42 | Druken | what exactly would the benefit of that be ? |
| 02:53.43 | Godsey | I know it's easy to do that w/ mgcp |
| 02:53.44 | *** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-94.dsl.spfdmo.swbell.net) |
| 02:53.46 | Juggie | well no, the mitel voip implementation isnt sip |
| 02:53.53 | Juggie | i'm just trying to recreate the effect :) |
| 02:54.02 | Godsey | Druken: integration w/ desktop apps |
| 02:54.03 | Nuxi | There isn't even a standardize way to reboot a phone via sip. |
| 02:54.14 | Ayano | anyone in here SER users. I need a good link to start learning it and testing it. |
| 02:54.21 | Juggie | thats what i was thinking, mitel has a unified messaging thing which sucks |
| 02:54.26 | JerJer | Ayano: good luck |
| 02:54.38 | Godsey | I know shoretel uses mgcp too |
| 02:54.41 | shidan | soon this will be #ser haha |
| 02:54.41 | Juggie | i am going to reimplement it for * as a web interface |
| 02:54.51 | Ayano | JerJer; that bad huh? |
| 02:55.01 | Nuxi | With our phones, we mix sip, snmp, and telnet to get them to do what we want. |
| 02:55.14 | Ayano | Does ser replace most of the features for asterisk or what? |
| 02:55.18 | kimo_sabe | Juggie: some phones can be setup to autoanswer |
| 02:55.24 | Juggie | Nuxi, i know i can build modules to telnet into the phones one for each phone type we support |
| 02:55.29 | Juggie | mitel, cisco etc |
| 02:55.38 | kimo_sabe | Juggie: so you'd call the phone's autoanswer thing and connect it to your other call |
| 02:55.38 | Juggie | but i was hopeing to avoid that by some standardized sip command :) |
| 02:55.56 | Juggie | kimo_sabe, i dont want each phone having two extensions like that. |
| 02:56.04 | Nuxi | The sip standard is open. Nothing like that is covered. |
| 02:56.07 | Juggie | and autoanswer cant be managed remotely |
| 02:56.30 | Juggie | at least not on cisco phones |
| 02:56.33 | Nuxi | http://www.faqs.org/rfcs/rfc3261.html |
| 02:56.39 | Godsey | autoanswer can be configured on my polycom phones |
| 02:56.44 | Godsey | Juggie: why not use mgcp? |
| 02:56.51 | Juggie | Godsey, why would i not use sip |
| 02:56.55 | Juggie | i want a standard |
| 02:57.00 | Juggie | not mgcp |
| 02:57.01 | Godsey | mgcp is a standard |
| 02:57.02 | cluecon[file] | MGCP is a standard |
| 02:57.08 | Godsey | and it does what you want |
| 02:57.10 | Juggie | ok, a commonly used standard :) |
| 02:57.10 | mrplum | Where can I pick up a FXO card-100XP for Asterisk? Besides eBay? |
| 02:57.25 | Godsey | I think most commercial voip pbx systems are mgcp |
| 02:57.25 | JerJer | TDM01B from DIgium |
| 02:57.31 | Juggie | none of the sip phones i have do mgcp with the exception of the cisco 7960 |
| 02:57.39 | Juggie | mitel is a h323 variant |
| 02:58.45 | Juggie | anyways, you see i want to create a web interface which shows the incomming/outgoing calls, missed calls, voicemail for a given extension |
| 02:58.50 | Godsey | I think Cisco Call Manager controled the phones w/ mgcp |
| 02:59.12 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 02:59.12 | *** mode/#asterisk [+o twisted] by ChanServ |
| 02:59.13 | Godsey | there are such beasts now |
| 02:59.23 | Juggie | click on a phone number to call it... as well allow you to manage your phone, set it to ring a group of numbers, or one at a time etc. |
| 02:59.31 | Godsey | flash operator panel may be a good place to start |
| 02:59.33 | cluecon[file] | call file! |
| 02:59.41 | Juggie | that flash operator pannel is gay |
| 02:59.57 | Juggie | no way :) this will be all web. |
| 02:59.59 | Godsey | why are you asking questions if you have seen how it works? |
| 03:00.12 | Juggie | i was just asking if there was some way to ask a phone to place a call |
| 03:00.13 | Juggie | via sip |
| 03:00.19 | Juggie | i thought no, but i was double checking |
| 03:00.27 | Godsey | use call files |
| 03:00.32 | JerJer | Cisco Call Manager uses SCCP |
| 03:00.34 | Godsey | it's quite easy |
| 03:00.35 | Juggie | that is having * make a call |
| 03:00.37 | JerJer | not MGCP |
| 03:00.44 | Godsey | JerJer: thanks. |
| 03:01.03 | Godsey | I haven't really looked back at call manager since switching to asterisk |
| 03:01.16 | Juggie | the 7960 has sip/mgcp/sccp FW |
| 03:01.26 | Godsey | just miss the brain dead easy accounting :) |
| 03:01.28 | Juggie | its unfortunate that the screen doesnt really do anything |
| 03:01.32 | JerJer | and iax |
| 03:01.34 | JerJer | oops |
| 03:01.35 | Juggie | nothing exciting |
| 03:01.47 | Juggie | iax? |
| 03:01.52 | *** join/#asterisk bkw__ (~brian@adsl-69-154-144-52.dsl.tulsok.swbell.net) |
| 03:02.03 | Godsey | Juggie: what screen? |
| 03:02.09 | Juggie | the screen on the 7960 |
| 03:02.11 | Godsey | fop flash? |
| 03:02.13 | Godsey | oh |
| 03:02.23 | Juggie | its nice, but not really used to its potential |
| 03:02.28 | Godsey | you can make the screen do something sending it messages right? |
| 03:02.34 | Juggie | yah |
| 03:02.35 | Juggie | xml |
| 03:02.40 | Juggie | i dont like the phone |
| 03:02.45 | Juggie | the mitel 5220 is much nicer |
| 03:02.50 | JerJer | most of the power is only available on the SCCP firmware load |
| 03:02.56 | Godsey | we have polycom ip300 for most of our techs |
| 03:03.06 | cluecon[file] | JerJer: you're awfully active today |
| 03:03.10 | Godsey | which are super simple no frill |
| 03:03.12 | JerJer | crisco crippled the phones power with the other firmware loads |
| 03:03.24 | JerJer | cluecon[file]: don't feel like coding or dealing with users |
| 03:03.33 | Godsey | at home I've removed sip phones :) |
| 03:03.37 | Godsey | and just use PAP2-NA now |
| 03:03.57 | *** join/#asterisk axscode (~1sdfgsdfg@203.177.235.188) |
| 03:04.21 | Godsey | I found I didn't need any special features, and with asterisk I can do all the parking call managment stuff w/ *xxx if I need to |
| 03:04.21 | Juggie | jerjer, with the sip load, my phone is broken |
| 03:04.28 | Juggie | the phone got set to a invalid vlan, and theres no way to change it unless thep hone boots |
| 03:04.28 | kimo_sabe | the new Grandstreams don't seem to suck that hard |
| 03:04.35 | axscode | can i use the modem to connect my phone? for external call? |
| 03:04.41 | Juggie | and the phone wont boot because it doesnt get a dhcp response |
| 03:04.56 | Juggie | i have lots of mitels now so i dont even care :) |
| 03:05.07 | dr123 | can anyone help me find Cisco 7960 6.3 Firmware |
| 03:05.09 | dr123 | or have it? |
| 03:05.24 | dr123 | not for production just testing |
| 03:05.27 | dr123 | I have 7.4 |
| 03:05.27 | JerJer | don't use vlans then |
| 03:05.33 | Godsey | Microsoft is working on a sip platform right? |
| 03:05.38 | dr123 | but I want to compare |
| 03:05.51 | dr123 | 6.3 does SIp correct |
| 03:06.07 | Godsey | wonder if it will let me work c# magic |
| 03:06.10 | shidan | Godsey: Exchange is the closest as far as I know |
| 03:06.30 | timecop | holy shit, can someone recommend a nonfailing prepaid app |
| 03:06.31 | Nuxi | Microsoft is working on destroying the VoIP market, yes. |
| 03:06.37 | timecop | this asterisk-prepaid shit is garbage. |
| 03:06.59 | Juggie | anyways, my next app is a * phone control platform in php |
| 03:07.05 | JerJer | timecop: i could have told you that many months ago |
| 03:07.07 | Juggie | conf server is almost complete. |
| 03:07.10 | cluecon[file] | stop complaining people |
| 03:07.18 | cluecon[file] | it's free, so you get what you pay for |
| 03:07.20 | cluecon[file] | oh wait - you didn't pay |
| 03:07.21 | Godsey | I will like MS more for helping me w/ voip :) |
| 03:07.31 | cluecon[file] | Clippy on your phone... |
| 03:07.32 | Godsey | there are some really nice text2speech platforms on windows |
| 03:07.36 | timecop | JerJer: well, got anything better? |
| 03:07.42 | axscode | hey guyz i made my asterisk working on my LAN... is there anyway i can connect my existing telephone from a telco so that i can use external calls. |
| 03:07.42 | cluecon[file] | "It appears you are dialing a phone number. Would you like to bring up your address book?" |
| 03:07.58 | Nuxi | Godsey, * works on windows http://www.asteriskwin32.com |
| 03:08.11 | Godsey | Nuxi: I will never use that software |
| 03:08.18 | timecop | haha cygwin |
| 03:08.21 | Godsey | I hope they're sued severly |
| 03:08.21 | timecop | what a fucking waste |
| 03:08.46 | axscode | where is much better.. a win32 or a unix base asterisk!?!? |
| 03:08.51 | timecop | win32, assuming it was properly coded. |
| 03:08.58 | timecop | which means not a cygwin hack. |
| 03:09.10 | Godsey | win32 ver on that site is criminal |
| 03:09.12 | JerJer | timecop: nothing that we distribute, no |
| 03:09.25 | timecop | Godsey: why? |
| 03:09.30 | Nuxi | win32 and coded properly don't go togther. |
| 03:09.33 | Godsey | they don't hand out the source |
| 03:09.38 | timecop | sure they do |
| 03:09.42 | timecop | did you even look at the fucking download link? |
| 03:09.44 | Godsey | ye |
| 03:09.50 | timecop | http://www.asteriskwin32.com/sourcecode.php |
| 03:09.51 | timecop | whats this? |
| 03:09.57 | *** part/#asterisk bkw__ (~brian@adsl-69-154-144-52.dsl.tulsok.swbell.net) |
| 03:10.05 | Godsey | download it |
| 03:10.54 | timecop | looks fine to me? |
| 03:10.55 | Godsey | win32 console ver won't build |
| 03:11.15 | timecop | well, thats no surprise. |
| 03:11.22 | timecop | who says it has to build? |
| 03:11.40 | timecop | cygwin is shit anyway, if anyone was goign to seriously port asterisk to windows they would make it a native app. |
| 03:11.47 | Nuxi | Godsey, it will build. |
| 03:11.51 | Nuxi | I built it. |
| 03:11.55 | *** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net) |
| 03:11.58 | Juggie | http://www.linuxgazette.com/issue63/misc/backpage/sv8538548.jpg |
| 03:12.04 | axscode | win32 vs unix base asterisk? what do u think!? |
| 03:12.08 | Juggie | check that |
| 03:12.11 | Sedorox | unix |
| 03:12.12 | Juggie | best thing making fun of clippy |
| 03:12.21 | timecop | ya very funny |
| 03:12.33 | timecop | i'd like to see opensores office apps do better. |
| 03:13.41 | Juggie | i cant wait to start on the * phone control pannel or whatever i call it |
| 03:13.42 | Nuxi | Remeber even M$ uses open source. |
| 03:13.46 | shidan | ms isnt in the voip market they care about broadcasting thats what theyre spending their money on |
| 03:13.47 | OnlyMe | Sedorox got it up and running |
| 03:14.01 | OnlyMe | Sedorox my tdm400P 11B is good |
| 03:14.08 | shidan | Ya like their tcp stack |
| 03:14.09 | Juggie | its going to rock, see all your call history, manage voicemail, manage what happens when your phone is busy, or no answer, ow many rings until its considered no answer... etc... |
| 03:14.32 | axscode | win32 vs unix base asterisk? what do u think!? ? guyz come on.. |
| 03:14.39 | *** join/#asterisk bkw__ (~brian@adsl-69-154-144-52.dsl.tulsok.swbell.net) |
| 03:14.43 | Juggie | unix |
| 03:14.45 | Nuxi | Juggie, if you open source it, you might want to hide it under a rock so that people don't complain about it. lol |
| 03:14.48 | OnlyMe | axscode come on |
| 03:14.56 | Nuxi | unix all the way. |
| 03:14.58 | axscode | hehhe |
| 03:14.59 | kimo_sabe | axscode: win32 is teh suX0r |
| 03:15.03 | OnlyMe | axscode keep windows for gaming like solitaire |
| 03:15.05 | kimo_sabe | axscode: better? |
| 03:15.11 | Juggie | Nuxi, i'll use phpagi too :) |
| 03:15.14 | axscode | ok unix u say.. im running it on freeBSD |
| 03:15.17 | axscode | hehe |
| 03:15.19 | Godsey | http://blog.tmcnet.com/blog/tom-keating/voip/microsoft-live-communications-server-2005.asp |
| 03:15.20 | axscode | just a question... |
| 03:16.08 | OnlyMe | axscode my 486 with kernel 1.2 is better them p4 running XP |
| 03:16.08 | axscode | guyz.. i have an existing telephone line.. how can i route my VoIP to that line? so that my Local Area can CALL using the telecom |
| 03:16.35 | Sedorox | OnlyMe: Cool |
| 03:16.37 | OnlyMe | axscode pots ?? |
| 03:16.40 | Corydon76-home | Get an FXO card |
| 03:16.42 | OnlyMe | Sedorox like you say |
| 03:16.43 | OnlyMe | :) |
| 03:17.08 | Corydon76-home | axscode: or you could get a T1 card and get the telco to drop a PRI to you |
| 03:17.20 | Corydon76-home | But that might be slightly expensive |
| 03:17.28 | Corydon76-home | "slightly" |
| 03:17.33 | OnlyMe | hihihihihi |
| 03:17.47 | *** join/#asterisk Esteli (~psolomon@69-165-217-96.atlsfl.adelphia.net) |
| 03:18.07 | Sedorox | axscode: my suggestion.. look at a Sipura SP-2000 or whatever it is.. it can bring in a normal phone.. and send one out... |
| 03:18.08 | Sedorox | whops... |
| 03:18.14 | Godsey | or you could get a t1 card and mux hardware |
| 03:18.18 | OnlyMe | Jun 1 23:10:50 NOTICE[14971]: chan_zap.c:5728 ss_thread: Got event 2 (Ring/Answered)... |
| 03:18.19 | Corydon76-home | Nice of him to stick around for the answer |
| 03:18.43 | OnlyMe | I get this only on my distective ring number |
| 03:19.17 | OnlyMe | but it works |
| 03:19.48 | *** join/#asterisk axscode (~1sdfgsdfg@203.177.235.188) |
| 03:19.59 | Godsey | microsoft is a perfect voip canditate |
| 03:20.05 | axscode | hmm.. i hate this.. always disconnected |
| 03:20.11 | Godsey | I bet they have sweet contracts w/ l3 ;) |
| 03:20.17 | Sedorox | [23:18] <Sedorox> axscode: my suggestion.. look at a Sipura SP-2000 or whatever it is.. it can bring in a normal phone.. and send one out... |
| 03:20.19 | Godsey | w/ msn.net dialup pools |
| 03:20.33 | axscode | SP-2000? gadget? |
| 03:21.24 | axscode | im thinking of emulating a modem.. |
| 03:21.34 | axscode | then.. throu that modem.. maybe it can pass in or passout the VoIP? |
| 03:21.39 | axscode | u think it can!? |
| 03:21.40 | Godsey | the spa-2000 can be FXO? |
| 03:21.47 | Juggie | axs, emulate a modem how? |
| 03:21.51 | Godsey | I thought it was 2 fxs ports |
| 03:21.53 | axscode | dont know |
| 03:21.55 | Juggie | a modem driver for * would be sweet :) go write one. |
| 03:22.01 | kimo_sabe | Godsey: one FXO, one FXS, might be the 3000 |
| 03:22.02 | Sedorox | there is one... |
| 03:22.05 | Sedorox | that has both... |
| 03:22.11 | Godsey | my asterisk uses a modem for pots |
| 03:22.12 | Juggie | no there isnt |
| 03:22.13 | Sedorox | so you can have a normal phone on it |
| 03:22.19 | axscode | whats pots? |
| 03:22.29 | Sedorox | and then one for normal lin |
| 03:22.33 | Sedorox | ~pots |
| 03:22.34 | jbot | [pots] Plain Old Telephone Service as in "Old Analogue Crap" |
| 03:22.47 | Juggie | i mean a modem as in allow someone to dial into * with a modem and get a connection |
| 03:22.52 | Juggie | over a pri |
| 03:23.02 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 03:23.05 | axscode | ic.. POTS.. |
| 03:23.13 | axscode | how did u do that Godsey? |
| 03:23.29 | OnlyMe | axscode no you can't smoke that stuff |
| 03:23.32 | Godsey | uh buy it from digium |
| 03:24.05 | OnlyMe | weed==POTS in french |
| 03:24.13 | Godsey | ok spa-3000 does fxs+fxo |
| 03:25.43 | *** join/#asterisk axscode (~1sdfgsdfg@203.177.235.188) |
| 03:25.50 | axscode | god... what D.. |
| 03:25.57 | axscode | packin ISP |
| 03:26.15 | axscode | what do i need to call outside? like a PBX? |
| 03:26.29 | Sedorox | depends how you wanna do that.. |
| 03:27.00 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 03:27.06 | axscode | i dont have any device.. so morelikely software base? |
| 03:28.25 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 03:28.29 | Sedorox | you need a device.. at least one.. the computer * would run on... and nothing else... IF you use a VoIP provider and VoIP phones... |
| 03:29.02 | *** part/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 03:29.31 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 03:29.31 | *** mode/#asterisk [+o twisted] by ChanServ |
| 03:29.48 | axscode | i have VoIP HardPhones and SoftPhone, i have existing telephone lines. i have modem. router. access point. of course i have computers.. |
| 03:30.12 | axscode | thats all i got for now.. so just learning asterisk with just that.. |
| 03:30.32 | axscode | and im using different modem/router for my DSL connection. |
| 03:30.42 | Sedorox | well.. only certain modems will allow you to use a normal telephone with Asterisk... |
| 03:30.57 | blitzrage | no modems work as an FXS afaik |
| 03:30.59 | axscode | ic.. whats that? i have PCI internal modem though.. |
| 03:31.17 | Sedorox | yea.. but you need a certain Intel chipset softmodem for it to work |
| 03:31.19 | axscode | whats the meaning of "afaiks" |
| 03:31.27 | blitzrage | only certain models (using Intel 537 chipset) are able to work as an FXO |
| 03:31.31 | blitzrage | as far as I know |
| 03:31.38 | JerJer | and most of them suck |
| 03:31.58 | Sedorox | As Far As I Know |
| 03:32.00 | Juggie | most of them have problems with hangup detection |
| 03:32.00 | axscode | hmm how would i know that.. |
| 03:32.03 | Juggie | why i dont know |
| 03:32.06 | Juggie | but they do |
| 03:32.09 | blitzrage | the modem doesn't have the circuitry to work as an FXS |
| 03:32.13 | JerJer | only the original motorolla chipset works properly |
| 03:32.17 | blitzrage | thus, you can't use it to control a phone, only a phone line |
| 03:32.28 | blitzrage | MD5200 or something like that |
| 03:32.34 | JerJer | hangup detection, horrid echo, cannot deal with on-hook audio |
| 03:32.36 | JerJer | etc |
| 03:32.44 | blitzrage | agreed |
| 03:32.47 | blitzrage | its shitty |
| 03:32.49 | Juggie | just go pri :) |
| 03:32.53 | axscode | whats the pri? |
| 03:32.55 | blitzrage | I wouldn't even both |
| 03:32.56 | Juggie | pots sucks |
| 03:32.57 | blitzrage | bother* |
| 03:33.03 | blitzrage | axscode: you have a lot of reading to do :) |
| 03:33.04 | blitzrage | ~pri |
| 03:33.05 | jbot | rumour has it, pri is 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, etc. |
| 03:33.30 | axscode | ic... |
| 03:33.33 | *** part/#asterisk orbi (~dantate@pcp08696782pcs.500ash01.tn.comcast.net) |
| 03:33.39 | axscode | ive been reading a lot lately.. |
| 03:33.45 | *** join/#asterisk irv999 (~irv999@ool-44c241c7.dyn.optonline.net) |
| 03:33.46 | irv999 | lo all |
| 03:33.49 | blitzrage | axscode: it doesn't end |
| 03:33.52 | axscode | at least.. i made my asterisk works.. |
| 03:34.00 | axscode | i know blitzrage.. |
| 03:34.08 | axscode | and i cant sleep without reading.. |
| 03:34.22 | blitzrage | anyone know how to explain sendani so it makes sense? |
| 03:34.36 | irv999 | Does anyone know if there is a way (when dialing an extension or transferring to an extension) for the sending phone to receive a message that someone is on the line? or the phone is off hook? |
| 03:35.05 | axscode | so for the feature that i made is voicemail.. |
| 03:35.19 | Sedorox | I guess call it first.. then when it rings.. transfer it |
| 03:35.29 | axscode | the conferencing still sucks.. dont have a zaptel.. ztdummy i cant find a downloadable source |
| 03:36.00 | axscode | i really need a zaptel for it.. |
| 03:36.25 | *** join/#asterisk dudeox (~boanthrap@dva215.resnet.neu.edu) |
| 03:36.30 | dudeox | Hey |
| 03:36.36 | axscode | hey what? |
| 03:36.37 | blitzrage | axscode: huh? its in zaptel |
| 03:36.57 | dudeox | sorry... newbie |
| 03:37.07 | dudeox | ;) |
| 03:37.07 | axscode | blitzrage: dont know. icant coference.. it says no channel exist.. though i made it on the cnfig.. |
| 03:37.21 | axscode | i did visit here last time. it says. it needs a ztdummy.. |
| 03:37.43 | blitzrage | axscode: ztdummy is part of the zaptel drivers and will give you a timing source |
| 03:37.52 | axscode | though i cant find one that is .tgz coz. its always on a cvs type.. |
| 03:37.54 | blitzrage | axscode: you really need to buy my book :) |
| 03:38.02 | axscode | wow. u made a book? |
| 03:38.10 | axscode | e-book.. i download it.. thanks.. |
| 03:38.19 | axscode | ill* |
| 03:38.23 | axscode | hehehe... |
| 03:38.29 | blitzrage | export CVSROOT=:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot |
| 03:38.32 | blitzrage | cvs co zaptel |
| 03:38.38 | blitzrage | cd zaptel |
| 03:38.42 | blitzrage | vim Makefile |
| 03:38.49 | blitzrage | uncomment ztdummy |
| 03:38.58 | blitzrage | make ; make install |
| 03:39.15 | blitzrage | modprobe zaptel |
| 03:39.19 | blitzrage | modprobe ztdummy |
| 03:39.56 | axscode | thats why.. im on FreeBSD.. |
| 03:40.06 | blitzrage | then you're screwed |
| 03:40.09 | Sedorox | no |
| 03:40.11 | blitzrage | :) |
| 03:40.15 | Sedorox | ztdummy works in fbsd :p |
| 03:40.20 | Sedorox | my first * box was fbsd |
| 03:40.21 | blitzrage | oh it does? cool :) |
| 03:40.21 | Sedorox | ANYWAY... |
| 03:40.30 | Sedorox | yea.. its a different version of zaptel tho |
| 03:40.39 | Sedorox | cd /usr/port/misc/zaptel ; make install |
| 03:40.54 | Sedorox | kldload (or whatever it is) zaptel |
| 03:40.55 | axscode | sad think is. i dont have an internet at home. |
| 03:40.57 | Sedorox | then the same for ztdummy |
| 03:41.05 | Sedorox | hold... |
| 03:41.15 | axscode | so its like.. i have to download the source then... save it on USB |
| 03:41.21 | axscode | so its like.. i have to download the source then... save it on USB Flashdisk |
| 03:41.30 | Sedorox | http://www.portaone.com/~sobomax/zaptel-freebsd-0.8.tar.gz |
| 03:41.34 | Sedorox | save.. take home |
| 03:41.35 | Sedorox | and compile |
| 03:41.45 | axscode | yeah.. but cant find the download |
| 03:41.50 | axscode | can u point me.. |
| 03:41.50 | Sedorox | ... |
| 03:41.55 | Sedorox | [23:41] <axscode> so its like.. i have to download the source then... save it on USB Flashdisk |
| 03:41.55 | Sedorox | [23:41] <Sedorox> http://www.portaone.com/~sobomax/zaptel-freebsd-0.8.tar.gz |
| 03:41.55 | Sedorox | [23:41] <Sedorox> save.. take home |
| 03:42.00 | axscode | there |
| 03:42.32 | axscode | is there a dependencies for this? |
| 03:42.54 | Sedorox | Umm |
| 03:42.56 | Sedorox | shouldn't be... |
| 03:43.09 | Sedorox | actually... |
| 03:43.10 | Sedorox | hold on |
| 03:43.18 | axscode | ok.. thanks Sedorox.. quite a help |
| 03:43.51 | Sedorox | http://www.portaone.com/~gonzo/zaptel-bsd-trunk.tar.gz |
| 03:43.55 | Sedorox | get that instead |
| 03:44.12 | dudeox | I am just beginning to look at Asterisk and want to find out if it can perform one particular job |
| 03:44.17 | Sedorox | anything below 8.3 ( I think thats it) has a nasty bug will it'll lock the kernel |
| 03:44.28 | Sedorox | dudeox: and that would be..? |
| 03:44.45 | axscode | ok.. so how to install this Sedorox? |
| 03:44.49 | *** topic/#asterisk by drumkilla -> Asterisk: The Open Source PBX || 1.0.8 RC bug #4424 || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm || The FCC ruled for forced E911 by pstn-terminated voip carriers - http://www.fcc.gov |
| 03:44.54 | axscode | tar -zxvf file |
| 03:44.56 | axscode | cd file |
| 03:45.04 | axscode | make install clean |
| 03:45.11 | axscode | or there is something i have to do? |
| 03:45.11 | Sedorox | well... cd <whatever the directory is> |
| 03:45.17 | dudeox | I want to know if transfering from line A to line B, if line A would still be let go |
| 03:45.18 | JerJer | rm -rf /boot ; reboot -n |
| 03:45.49 | Sedorox | you should just be able to do make install.. and it'll do everything |
| 03:45.56 | Sedorox | dudeox: do you want it to drop the line or not? |
| 03:46.07 | dudeox | I would like to drop the line |
| 03:46.21 | axscode | ok nice Sedorox.... |
| 03:46.23 | Sedorox | yea.. most phones will either do the transfer.,... or * can do it |
| 03:46.41 | axscode | then after that.. there is a thing on linux.. like modrobe.. |
| 03:46.55 | Sedorox | Ummm |
| 03:46.56 | Sedorox | fbsd... |
| 03:47.15 | Sedorox | kldload zaptel |
| 03:47.19 | Sedorox | kldload ztdummy |
| 03:47.20 | axscode | will i have to do that.. kld |
| 03:47.37 | axscode | hmm ok.. so ill just do that.. |
| 03:47.43 | axscode | ok ok. hehhee.. cant wait to conference.. |
| 03:47.54 | Sedorox | eh |
| 03:47.57 | Sedorox | its nothing special |
| 03:47.57 | axscode | just a thought. of phone s*x orgy |
| 03:47.59 | dudeox | so, for example, I can recieve a call on my single home phone line, transfer it to my cell phone, and then have my home phone free to take more calls again |
| 03:48.02 | Sedorox | 0_o |
| 03:48.13 | Sedorox | dudeox: no... |
| 03:48.40 | Sedorox | but you would only be able to do that if you had two phone lines.. or a VOIP account that allows several in/out calls at the same time |
| 03:49.23 | axscode | can i use cellfone for VoIP? |
| 03:49.27 | axscode | how would u do that? |
| 03:49.36 | Sedorox | not really... |
| 03:49.48 | dudeox | by two phone lines, you mean two lines going to my house, right? |
| 03:49.54 | Sedorox | yes |
| 03:50.04 | Sedorox | or like.. one line in your house.. and another being a VOIP provider |
| 03:50.19 | axscode | hey guyz i have an AccessPoint.. a Wireless reciever.. can u use CellFone for Asterisk!? |
| 03:50.23 | dudeox | Thanks a lot. This finishes off a 12 hour search for that answer. |
| 03:50.35 | Sedorox | axscode: not a cellphone... |
| 03:50.40 | Sedorox | hehe |
| 03:50.45 | axscode | PDA maybe right. |
| 03:51.00 | Sedorox | if the PDA has a speaker and mic |
| 03:51.16 | Sedorox | and can run the software |
| 03:51.25 | axscode | 6600 use JAVA platform. maybe there is a way to convert a SoftPhone to midlets.. |
| 03:51.32 | axscode | just a thought |
| 03:51.41 | axscode | Nokia6600* |
| 03:51.57 | dr123 | does anyone have cisco firmware 6.3 with sip support |
| 03:51.58 | opus__ | Hey guys when you build asterisk, what version of GCC do you use? |
| 03:51.58 | dr123 | does anyone have cisco firmware 6.3 with sip support |
| 03:52.36 | axscode | what device can let the VoIP to Telephone LineS? |
| 03:52.51 | Sedorox | FXS |
| 03:52.59 | axscode | ~FXS |
| 03:53.00 | jbot | fxs is probably foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 03:53.04 | blitzrage | so what does allowguest= actually do in SIP? |
| 03:53.06 | blitzrage | for instance, if I have a [guest] definition in sip.conf, and I do allowguest=no, does that disable it? |
| 03:53.20 | kimo_sabe | chan_bluetooth! :) |
| 03:56.53 | *** part/#asterisk Ayano (~erik_leee@adsl-70-245-190-94.dsl.spfdmo.swbell.net) |
| 03:57.19 | *** join/#asterisk Mavvie (edwin@dialin-83.barnet.com.au) |
| 03:58.16 | axscode | if im running an 1.7Ghz processor.. how many user can asterisk handle |
| 03:58.53 | opus__ | -msse2 breaks asterisk |
| 03:59.12 | Katty | i'm thinking it's bedtime |
| 04:04.22 | *** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net) |
| 04:05.28 | *** join/#asterisk sudhir492 (~sudhir@4.7.58.146) |
| 04:05.45 | mmlj4 | axscode: how much ram? and what codecs, are you using SIP/AIX/zap/whatever? |
| 04:05.53 | mmlj4 | er, IAX |
| 04:06.04 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 04:06.26 | axscode | im using SIP then i have about 256 of RAM and im using ulaw and alaw codecs |
| 04:07.01 | mmlj4 | ok, how many users do you have? maybe that's something I can actually answer |
| 04:07.53 | axscode | hmm.. just 5 of us.. but... actually im starting to let my other friends giving an acount on it.. then theyll use a softfone. |
| 04:08.01 | axscode | so im thinking about 100 - 200 users. |
| 04:08.05 | axscode | is that fine? |
| 04:08.34 | axscode | i have a 712KBPS speed |
| 04:08.43 | axscode | of internet |
| 04:09.01 | mmlj4 | hmm... you might want to ask someone else, since you're going to have that many users |
| 04:09.43 | mmlj4 | but i would add more RAM to that server, regardless of what you're going to do with it |
| 04:10.46 | axscode | ok.. anyways i have a dual processor CPU.. runing a 3GHZ Pentium 4, SuperMicro 5013C-M8 Model RackMount. with a 1GBPS Ethernet Adapter. running a FreeBSD 5.4 |
| 04:11.30 | opus__ | pbx_wilcalu.c .. was this removed in the latest CVS? |
| 04:11.37 | axscode | so im planning to use that for VoIP Gateway... so how many it can allow do u think? just a guess |
| 04:12.16 | pjz | if I want internal extensions to have to dial * to reach other extensions, should I exten => *,1,Goto(context_with_extensions,s,1), or exten => _*.,Goto(context_with_extensions,${EXTEN:1},1) ? |
| 04:12.18 | opus__ | axscode - for a small company or a entire telecom? a call center or a home office |
| 04:12.23 | kimo_sabe | axscode: more than you can offortd |
| 04:13.07 | axscode | <PROTECTED> |
| 04:13.34 | axscode | kimo can u gimme a digits for it?! |
| 04:14.50 | kimo_sabe | axscode: nope, that's more than I can afford to test |
| 04:15.14 | OnlyMe | axscode voip calculator |
| 04:15.32 | axscode | there is a VoIP Calculator? |
| 04:15.38 | axscode | cool.. whre can i find that? |
| 04:15.40 | OnlyMe | voip-calculator dot com |
| 04:16.35 | OnlyMe | another happy customer |
| 04:16.37 | OnlyMe | ;) |
| 04:16.38 | axscode | hehe |
| 04:16.40 | axscode | yeah |
| 04:16.43 | axscode | very nice |
| 04:17.16 | opus__ | that calculator is great |
| 04:17.21 | OnlyMe | yeap |
| 04:17.29 | opus__ | for seeing what it was like in 2001 |
| 04:17.49 | axscode | hehe... but... do its not an opern source |
| 04:18.05 | pjz | I guess what I'm asking is: how fast is context execution? |
| 04:18.18 | pjz | but I'm not sure that makes sense |
| 04:18.29 | opus__ | almost immedately |
| 04:18.45 | santiago | hi, anyone here knows about the zapata library status? |
| 04:18.54 | axscode | is there a device for VoIP Gateway? for VAST user |
| 04:18.54 | pjz | can you transfer someone to a context without transferring them to an extension? |
| 04:19.20 | pjz | ie. can you sent them to some context's 's' extension even if it's not defined? |
| 04:21.38 | pjz | and is there any diff between the two exten lines I entered above? |
| 04:25.06 | opus__ | pjz - I'm not sure, you would have to look at the source |
| 04:25.24 | opus__ | if you send somebody to an extenion , I don't believe 's' will be used anyway because it's already answered. |
| 04:26.38 | pjz | is there a multi-level menu example someplace? |
| 04:27.31 | opus__ | pjz - not really |
| 04:27.36 | opus__ | look for asterisk config examples. |
| 04:27.46 | opus__ | pjz - really, Goto is powerful |
| 04:28.39 | opus__ | uh oh. does spandsp patch work with CVS HEAD? |
| 04:28.42 | axscode | ~GSM |
| 04:28.43 | jbot | gsm is, like, Groupe Spécial Mobile |
| 04:28.44 | axscode | ~GPRS |
| 04:28.45 | jbot | [gprs] an abbreviation that stands for "General Packet Radio Service" which is now the ever-so-popular method for accessing the Information Super Highway through a telecommunication device, preferrably a/an EGSM/GSM transceiver which is capable of this service. There is some information on how to get this service working on your iPAQ whilst establishing a ... |
| 04:29.48 | axscode | can i use asterisk for GSM? |
| 04:29.58 | opus__ | no |
| 04:30.05 | axscode | ok... |
| 04:30.10 | opus__ | you can use it for the gsm codec though |
| 04:30.17 | opus__ | not to make cell phone calls |
| 04:30.49 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 04:30.57 | kimo_sabe | opus__: maybe with chan_bluetooth |
| 04:31.16 | kimo_sabe | opus__: well, if you have a non-crippled BT capable phone |
| 04:31.27 | pjz | hrm, anyone know of an ARM-linux softphone? |
| 04:31.45 | pjz | I just got to the linux prompt on my cellphone :) |
| 04:31.58 | kimo_sabe | opus__: I think I've dialed out of * through my T610, but it's been a while so I'm not positive |
| 04:32.27 | timecop | running lunix on a phone is probably the most retarded shit ever. |
| 04:33.13 | kimo_sabe | timecop: nah, there's phone with Windows on them ;) |
| 04:33.23 | iheartcanada | timecop: you have to be pretty retarded to say a thing like that |
| 04:33.34 | iheartcanada | and i don't even like linux :) |
| 04:34.03 | iheartcanada | linux can do real time in many soft-real time versions, it's an excellent way to do phone |
| 04:34.31 | kimo_sabe | iheartcanada: some phones already have hard-realtime and non-realtime parts anyway |
| 04:35.07 | kimo_sabe | iheartcanada: my T610 does. The slow, twitchy GUI runs on a separate processor than the GPS processing |
| 04:35.09 | iheartcanada | kimo_sabe: no doubt |
| 04:35.11 | kimo_sabe | err, GSM |
| 04:35.45 | pjz | yeah, I'm pretty sure the linux on mine is on a separate processor than the GSM stuff |
| 04:39.58 | *** join/#asterisk Juggie (~agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 04:42.44 | axscode | anyone happen to know any device that cellfone can communicate going to a PC |
| 04:43.47 | timecop | iheartcanada: oh fucking please. all the lunix phones i've seen are fucking horrible. |
| 04:44.01 | santiago | axscode, what do you mean? |
| 04:44.30 | axscode | i want to communicate to a cellfone |
| 04:44.33 | axscode | nope wrong |
| 04:44.38 | OnlyMe | timecop i sen nice one but they dont work |
| 04:44.41 | axscode | i want a cellfone communicate to a PC |
| 04:44.47 | timecop | you dont. |
| 04:44.57 | kimo_sabe | axscode: most phones arn't very talkative |
| 04:45.10 | axscode | hehe |
| 04:45.13 | santiago | :) |
| 04:45.28 | axscode | so maybe any device aside using bluetooth and infrared? |
| 04:45.33 | kimo_sabe | axscode: seriously, doubly so if they are anywhere near Sprint or Verizon |
| 04:46.19 | axscode | hehhe just thinking of something i can integrate to VoIP |
| 04:46.29 | *** join/#asterisk wshs (screwy@69-168-246-252.bflony.adelphia.net) |
| 04:46.47 | santiago | axscode, I connected an cell phone "trunk" to and fxo |
| 04:47.17 | axscode | i have my own asterisk? how will i connect my asterisk to another asterisk? or another VoIP gateway? so that my local can dial that network? |
| 04:47.20 | kimo_sabe | axscode: I know I had my headset working, but i'm not positive about the phone itself, using a custom hacked chan_bluetooth |
| 04:47.28 | kimo_sabe | ...I should update that patch and send it upstream.... |
| 04:47.46 | axscode | bluetooth is a very short range |
| 04:47.56 | timecop | asxcode is a known troll |
| 04:47.58 | OnlyMe | axscode IAX Inter Asterisk Exchange |
| 04:48.01 | timecop | plz ignore, people. |
| 04:48.08 | axscode | ~IAX |
| 04:48.10 | jbot | iax is probably 4569 and 5036, or pronounces "Eeks" |
| 04:48.20 | kimo_sabe | axscode: long enough to go from the server to the phone kept by the window to it actually gets reception |
| 04:48.59 | axscode | yeah... but i want to deploy it |
| 04:49.02 | axscode | all over the city |
| 04:49.04 | *** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-94.dsl.spfdmo.swbell.net) |
| 04:49.07 | axscode | which i prolly cant |
| 04:49.37 | Ayano | how many sip phones can an asterisk box support using the default signaling? |
| 04:50.04 | pjz | what's signaling got to do with the number of supportable phones? |
| 04:50.40 | Ayano | hold on, I might have used the wrong term for that, good point... |
| 04:53.00 | Ayano | sorry, codecs is the word I meant... how many sip phones can an asterisk box support using the default codecs? |
| 04:55.30 | Qwell | Ayano: There are far too many factors |
| 04:55.37 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 04:56.15 | pjz | Ayano: also, if you think about it, you are probabbly really asking not how many phones, but how many simultaneous calls |
| 04:56.28 | pjz | Ayano: since the load is more if people are talking onthe line than if they're not |
| 04:57.11 | pjz | Ayano: alos,, it's clearly different per machine - I wouldn't expect to run too many people on my ancient 486/33... |
| 04:57.35 | pjz | Ayano: so I suspect that once you get to where you've defined the question enough to be answerable, you'll be able to answer it yourself |
| 04:57.37 | Qwell | I was able to get 98 calls with iaxcomm a few days ago, on a fairly cheap (by todays standards) desktop. I have a feeling iaxcomm was the weak point |
| 04:59.50 | Ayano | pjz, I see what your getting at, I know what you mean. the question is too vague. I guess the question is what happens if i hit the max? How do I get more? Another asterisk server? |
| 05:00.07 | Qwell | Ayano: new/better hardware, or sure, another box |
| 05:00.52 | Ayano | What is the best thing you can do for redundancy as well? |
| 05:01.49 | pjz | buy good quality hardware |
| 05:04.22 | Ayano | Say I have 2 * boxes. What is the best way to balance the two? |
| 05:04.34 | *** join/#asterisk Ahewes (~rsb@adsl-69-107-50-127.dsl.pltn13.pacbell.net) |
| 05:05.12 | Ayano | Put half the users on one, and the other on number 2? |
| 05:08.29 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:13.19 | Qwell | Please DO NOT turn your MoH ALL the way up! |
| 05:14.57 | *** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 05:15.43 | Qwell | especially when you have music that is high pitched...jesus |
| 05:15.56 | Qwell | this is painful |
| 05:17.20 | denon | yeah .. just set your txgain to 200 |
| 05:18.47 | Smi|k | what ms ping should a colo be able to offer for voip to not have a problem |
| 05:18.49 | Smi|k | under what # |
| 05:20.04 | Qwell | denon: cellphone |
| 05:20.21 | Qwell | Smi|k: the lower the better |
| 05:20.27 | Smi|k | whats low and whats high |
| 05:20.28 | Qwell | 60 should be pretty good |
| 05:20.34 | Smi|k | 80 is too high? |
| 05:20.36 | Qwell | I get 80ms to nufone, and its still good |
| 05:21.00 | Smi|k | I'm having asterisk box on colo, and then for outgoing colo will connect to voip provider for long distance |
| 05:21.06 | Smi|k | so I think there is 2 legs of ping involved |
| 05:21.09 | Qwell | why? |
| 05:21.20 | blitzrage | jbot_: iax is port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". |
| 05:21.29 | Smi|k | one for ip phone to colo asterisk, one from colo asterisk to ld voip provider |
| 05:21.42 | Smi|k | each location will only have ip phones, asterisk server at colo |
| 05:21.45 | Qwell | You'll definitely want do use SIP the whole way and allow reinvites, to lower the delay |
| 05:22.03 | Smi|k | I cant go direct from phone to voip provider right? |
| 05:22.08 | blitzrage | hrmmm, guess I need to /msg jbot to update it :) |
| 05:22.09 | Qwell | blitzrage: no, x is |
| 05:22.12 | blitzrage | oh well, fixed |
| 05:22.28 | Smi|k | because asterisk handles the selection of which voip to use, failover, etc..etc.. |
| 05:22.49 | Smi|k | or is there some mini-local-asterisk just to select outgoing calls from each office and the main asterisk box at colo does incoming only? |
| 05:22.55 | blitzrage | Qwell: yah, but it normally replies, "but x is something else" |
| 05:23.01 | Qwell | oh |
| 05:23.42 | blitzrage | didn't that time though :) |
| 05:23.48 | blitzrage | did when I /msg'd him directly though |
| 05:23.54 | Qwell | strange |
| 05:24.58 | Qwell | ho hum, ho hum |
| 05:25.10 | blitzrage | blah! :) |
| 05:25.16 | Qwell | You guys should test bug 4403... Even if you don't use the odbc voicemail stuff. |
| 05:25.27 | Smi|k | is there a website that pings an ip from say 10 diff places and gives you the results of each? |
| 05:25.42 | Qwell | Smi|k: perhaps something at broadbandreports.com |
| 05:25.50 | Smi|k | if not, can a few people ping adnc.com and visuallink.com and tell me what you get for each as average? |
| 05:27.42 | dr123 | Hey I just got my Cisco 7960 in the Mail I NEEEEEED the 6-3 Firmware PLZ |
| 05:27.57 | Qwell | dr123: cisco.com |
| 05:28.15 | Qwell | Don't msg me |
| 05:28.34 | gordonjcp | hey Qwell |
| 05:28.40 | Qwell | gordonjcp: y0 |
| 05:29.05 | orlon_ | dr123: it's not free, go buy a Smartnet contract then you can download 7.4 |
| 05:29.11 | Qwell | gordonjcp: I'm gonna hack up a patch to indications, so you can use midi notes or frequencies... |
| 05:29.23 | Qwell | and it'll convert them on the fly to freqs |
| 05:29.46 | gordonjcp | Qwell: hahaha sick |
| 05:29.49 | denon | hey, anyone running 7.4 .. what happened to the clock on top? |
| 05:29.55 | denon | the date/time, on 7960s |
| 05:30.17 | blitzrage | does anyone know what 'mask' does in sip.conf? |
| 05:30.19 | *** join/#asterisk oej (~oej@213.204.186.40) |
| 05:30.34 | *** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 05:30.55 | dr123 | I heard 7.4 had a problem and that 6.3 is the one you want to use |
| 05:31.08 | denon | dr123: you heard wrong |
| 05:31.22 | gordonjcp | dr123: but - 7.4 is 1.1 louder, innit? |
| 05:31.31 | denon | blitzrage: as in subnet mask? |
| 05:31.42 | *** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 05:31.43 | blitzrage | denon: no idea - I can only assume, but no idea why or how I would use it :) |
| 05:31.46 | blitzrage | mask= |
| 05:31.59 | blitzrage | there is no example in sip.conf.samples either |
| 05:32.02 | blitzrage | its just listed |
| 05:32.05 | gordonjcp | blitzrage: subnet mask |
| 05:32.15 | dr123 | what i heard was that is was buggy w/ sip |
| 05:32.17 | blitzrage | I'm not sure what I would use it for though? |
| 05:32.17 | gordonjcp | you don't use it any more |
| 05:32.35 | denon | dr123: you heard wrong |
| 05:32.35 | gordonjcp | uhm, to tell Asterisk what subnet it is using? |
| 05:32.53 | denon | 7.4 has been just fine for us |
| 05:33.01 | dr123 | cool then |
| 05:33.01 | denon | but the clock and date missing on the top sucks |
| 05:33.13 | dr123 | really that took that out... that does suck |
| 05:33.15 | blitzrage | gordonjcp: I guess.... is that related to the localnet, etc..? because localnet has the form ip/mask |
| 05:33.24 | blitzrage | gordonjcp: so you say its just not used anymore? |
| 05:33.24 | gordonjcp | blitzrage: yes |
| 05:33.33 | denon | dr123: its weird.. its there when it boots up, but then it goes away, or somethin |
| 05:33.34 | gordonjcp | well, going by an example I saw, no |
| 05:33.37 | denon | cant figure it out |
| 05:33.38 | blitzrage | gordonjcp: much obliged, exactly what I was looking for :) |
| 05:33.47 | blitzrage | I only asked because I'm documenting SIP |
| 05:33.48 | gordonjcp | blitzrage: possibly a debug message or something |
| 05:33.50 | blitzrage | thanks guys |
| 05:34.09 | gordonjcp | ISTR *something* telling me to use x.x.x.x/n format |
| 05:34.40 | *** join/#asterisk LOT (~Methos@S0106000f6694b86f.ed.shawcable.net) |
| 05:34.41 | *** join/#asterisk cmk (~cmk_@p54A3F981.dip.t-dialin.net) |
| 05:34.53 | Qwell | blitzrage: document my patch in bug 4403! :p |
| 05:36.16 | blitzrage | Qwell: pay me :) |
| 05:36.18 | blitzrage | lol |
| 05:36.27 | blitzrage | Qwell: what is it? |
| 05:36.49 | Qwell | its a good addition, heh |
| 05:36.58 | twisted | oh my gawd |
| 05:37.00 | Qwell | nobody uses the odbc voicemail stuff though |
| 05:37.03 | blitzrage | odbc! |
| 05:37.07 | blitzrage | bah! :D |
| 05:37.08 | twisted | you're wanting someone to document your code for you? |
| 05:37.15 | Qwell | twisted: all the time :p |
| 05:37.16 | twisted | what do you think this is? |
| 05:37.34 | twisted | "i'll pay ya ten bucks to do my homework" |
| 05:37.36 | Qwell | by document, I mean test ;] |
| 05:37.40 | blitzrage | twisted: I need to get someone to add a code review section that checks if adequate documentation has been supplied |
| 05:37.48 | blitzrage | if no - no patch to CVS for j00 |
| 05:37.58 | twisted | blitzrage, haha... talk to matt |
| 05:38.15 | blitzrage | twisted: I will :D |
| 05:38.17 | twisted | and blitzrage - wrong! if the code is good and solid, it's oing in ;) |
| 05:38.17 | blitzrage | Matt F? |
| 05:38.20 | twisted | going, too. |
| 05:38.23 | blitzrage | twisted: fuck that |
| 05:38.41 | twisted | fuck what? |
| 05:38.55 | blitzrage | putting in undocumented code :) |
| 05:39.07 | blitzrage | thats how we end up with things that make no sense and no one knows how to use it |
| 05:39.16 | Qwell | blitzrage: You remind me of somebody... |
| 05:39.17 | blitzrage | or rather, undocumented features |
| 05:39.23 | Qwell | ahh, right, our tech writer at work. :D |
| 05:39.29 | blitzrage | I remind me of me |
| 05:39.50 | MikeJ[Laptop] | you do? |
| 05:39.56 | Qwell | it really is a shame that there aren't more people willing to do documentation though |
| 05:40.08 | gordonjcp | I'd do it if I knew more about it though |
| 05:40.20 | MikeJ[Laptop] | or more people to test, or more people to backport bigfixes to stable or...... |
| 05:40.22 | Qwell | gordonjcp: I think thats part of the problem |
| 05:40.28 | Qwell | chicken-egg scenario |
| 05:40.34 | Qwell | MikeJ[Laptop]: of course... |
| 05:40.35 | blitzrage | Qwell: uhhhh, yah :) |
| 05:40.42 | MikeJ[Laptop] | or more people to create doxygen docs... |
| 05:40.46 | twisted | blitzrage, we know how to use what we have |
| 05:40.48 | blitzrage | theres like... 5 of us, including twisted and drumkilla :) |
| 05:40.51 | twisted | (generally speaking) |
| 05:40.56 | MikeJ[Laptop] | or more coders in general to asterisk |
| 05:41.09 | Qwell | MikeJ[Laptop]: I'm trying to become a bit active with coding |
| 05:41.11 | twisted | hehe |
| 05:41.16 | twisted | 5 of us what? |
| 05:41.17 | blitzrage | whats Remote-Party-ID ? |
| 05:41.18 | MikeJ[Laptop] | losts of I wants... not so many I do's |
| 05:41.22 | Qwell | small features here and there, and I'll eventually try to get bug fixes |
| 05:41.23 | blitzrage | twisted: who have written docs |
| 05:41.26 | twisted | oh |
| 05:41.26 | twisted | yea |
| 05:41.32 | twisted | teehee |
| 05:41.50 | twisted | it hurt every minute, too :P |
| 05:41.52 | MikeJ[Laptop] | qwell.. you want to get into some code? |
| 05:41.59 | blitzrage | and I should say 6 actually, I'll make file and honorary documenteur |
| 05:42.07 | Qwell | MikeJ[Laptop]: when time permits. I'm just trying to learn how everything works right now |
| 05:42.07 | blitzrage | s/and/an |
| 05:42.11 | blitzrage | he knows his shit |
| 05:42.24 | twisted | blitzrage, i still like the hostmask i threw at ya :) |
| 05:42.28 | MikeJ[Laptop] | Qwell, there are 7 bugs in pending stable that need patches for stable |
| 05:42.44 | Qwell | MikeJ[Laptop]: backport? |
| 05:42.53 | *** part/#asterisk beto75 (~beto75@201.133.243.30) |
| 05:43.00 | Qwell | or do the patches not exist anywhere at all? |
| 05:43.04 | MikeJ[Laptop] | 3 that are really just waiting for patches |
| 05:43.12 | MikeJ[Laptop] | just backporting |
| 05:43.26 | Qwell | lemme take a look |
| 05:43.30 | MikeJ[Laptop] | 3833, 3971, 3867 |
| 05:43.37 | Qwell | those top 7? |
| 05:43.43 | blitzrage | twisted: me too :D |
| 05:43.50 | MikeJ[Laptop] | those are the 3 ready to have patches |
| 05:43.51 | twisted | hehe |
| 05:43.54 | blitzrage | twisted: I've had comments |
| 05:44.00 | blitzrage | people love it :) |
| 05:44.17 | MikeJ[Laptop] | and 4406 |
| 05:44.21 | MikeJ[Laptop] | forgot that one |
| 05:44.31 | twisted | blitzrage, hehe ya |
| 05:44.43 | MikeJ[Laptop] | they all have the ...HeadCommit.txt attached, that is what needs to get backported |
| 05:44.48 | Qwell | hmm |
| 05:44.57 | *** join/#asterisk outofjungle (~outofjung@61.17.134.200) |
| 05:46.07 | MikeJ[Laptop] | 3833 should be fairly easy |
| 05:46.27 | Qwell | looking at it now |
| 05:46.31 | MikeJ[Laptop] | as shoule 3971 |
| 05:49.38 | MikeJ[Laptop] | should that is |
| 05:50.02 | *** join/#asterisk rue_mohr (~dan@ip-216-123.ppp.ucc-net.ca) |
| 05:51.04 | *** join/#asterisk goldenolden (~goldenold@c-67-160-85-227.hsd1.wa.comcast.net) |
| 05:51.05 | goldenolden | hi |
| 05:51.13 | goldenolden | anyone have any clue who makes the bestelco wireless phones? |
| 05:51.21 | oej | Morning |
| 05:51.35 | MikeJ[Laptop] | standard cordless phones? panasonic probably |
| 05:51.41 | MikeJ[Laptop] | oej: Morning |
| 05:51.48 | MikeJ[Laptop] | welcome back from von land |
| 05:52.05 | oej | Thank you! |
| 05:52.10 | MikeJ[Laptop] | :) |
| 05:52.20 | oej | Going into Astricon Land! |
| 05:52.25 | MikeJ[Laptop] | yep |
| 05:52.56 | MikeJ[Laptop] | so I hear |
| 05:53.15 | MikeJ[Laptop] | no one to sponsor my travel :( |
| 05:53.46 | goldenolden | take a look at this wireless IP phone... $179, I have one sitting in front of me http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1225720&CatId=1626 |
| 05:53.50 | MikeJ[Laptop] | you coming to the US any time soon? |
| 05:54.48 | MikeJ[Laptop] | hmmm |
| 05:54.51 | MikeJ[Laptop] | no idea |
| 05:55.26 | oej | MikeJ: Coming to Kansas and Denver in July for two weeks |
| 05:56.16 | *** join/#asterisk wfu (~wfu@203.131.175.66) |
| 05:57.06 | *** join/#asterisk pbx123 (~honeyrt@203.177.171.242) |
| 05:57.17 | wfu | can i ask who in here has developed a parser for asterisk configuration files? |
| 05:58.19 | Qwell | MikeJ[Laptop]: 3833 uploaded |
| 05:58.25 | rue_mohr | *the* or *a* ? |
| 05:58.32 | Qwell | I should probably be using cvs stable, shouldn't I? |
| 06:01.09 | MikeJ[Laptop] | ummm |
| 06:01.11 | MikeJ[Laptop] | I don't |
| 06:01.23 | Qwell | oh, just use current stable? like 1.0.7? |
| 06:01.54 | MikeJ[Laptop] | qwell.. it needs to be against current 1-0 tree, not 1.0.7... |
| 06:02.08 | MikeJ[Laptop] | checkout with -r v1-0 |
| 06:02.18 | Qwell | I kinda figured that would be the case, after I was done. heh |
| 06:02.20 | MikeJ[Laptop] | your 1.0.7 patch may still apply |
| 06:02.28 | MikeJ[Laptop] | check it and find out |
| 06:02.31 | Qwell | will do |
| 06:03.07 | pjz | I don't have a parser, but I'm working on a generator :) |
| 06:03.09 | MikeJ[Laptop] | those files don't change much.. so I bet it will |
| 06:03.30 | pjz | I've got a pile of python that I'm writing that generates my config files |
| 06:03.51 | rue_mohr | wfu: ? |
| 06:03.52 | pjz | including stuff in /var/lib/tftpboot that my phones tftp-download when they boot up |
| 06:04.13 | Qwell | MikeJ[Laptop]: I think even the patch that was already up might apply. |
| 06:04.21 | MikeJ[Laptop] | it might have |
| 06:04.28 | MikeJ[Laptop] | I didn't try it yet |
| 06:05.15 | iheartcanada | why are sip phones $179? too bloody expensive |
| 06:05.23 | MikeJ[Laptop] | Qwell, try that... |
| 06:05.38 | MikeJ[Laptop] | if it does, just comment the bug that it does and good to go |
| 06:08.16 | Qwell | it failed on the .h file...no clue why though |
| 06:08.21 | pbx123 | I'm using a speedtouch 511e modem for my VOIP asterisk box. I enable the port forwarding of the modem to assign all requests to my asterisk box. I get to route the calls through the system, but the problem is that there's only one way audio. You cannot hear the callers voice on the other line, but the caller hears you. |
| 06:08.21 | pbx123 | I use it as NAT. I already created the static translation to forward the SIP requests to 10.0.0.2 (w/c is the IP assiged to my box). But still it doesn't work. |
| 06:08.21 | pbx123 | What do I need to do to be able to hear the other person on the other line? Is there is something else I still need to enable? |
| 06:08.40 | pbx123 | please help me |
| 06:09.06 | pbx123 | im using G729 codec |
| 06:09.11 | Qwell | heh, because of extra spaces |
| 06:09.17 | MikeJ[Laptop] | nice |
| 06:09.33 | Qwell | thats funny |
| 06:09.55 | Qwell | yeah, two spaces at the end of a comment line made it break |
| 06:10.00 | shido | whats up pbx123? |
| 06:10.08 | pbx123 | shido I need help |
| 06:10.16 | pbx123 | please read my problem |
| 06:10.28 | shido | got a license for that g729, sir? |
| 06:10.35 | pbx123 | yes sir |
| 06:10.38 | pbx123 | i got it |
| 06:10.40 | shido | and how many, sir? |
| 06:10.44 | pbx123 | 8 |
| 06:10.55 | shido | ok, is your asterisk system behind a nat? |
| 06:11.00 | pbx123 | yes it is |
| 06:11.03 | Qwell | MikeJ[Laptop]: I'll just upload a new patch. Can you delete the first one I did? |
| 06:11.14 | shido | and do you have access to the router or linux box that is doing the natting? :) |
| 06:11.48 | pbx123 | yes I do, i already enabled the port forwarding |
| 06:11.55 | shido | for what ports, sir? |
| 06:12.06 | MikeJ[Laptop] | yes |
| 06:12.06 | pbx123 | 5060 specifically |
| 06:12.14 | shido | and what about the audio, sir? |
| 06:12.16 | shido | rtp |
| 06:12.32 | shido | and did you forward 5060 UDP or TCP or both? |
| 06:12.33 | pbx123 | hmmmm |
| 06:12.37 | pbx123 | UDP sir |
| 06:12.46 | shido | im just messing with you on the sir stuff |
| 06:13.04 | pbx123 | LOL |
| 06:13.12 | pbx123 | :) |
| 06:13.50 | pbx123 | do i still need the rtp? even if im not using IAX? |
| 06:13.52 | shido | u may need to open up some more ports |
| 06:14.10 | *** join/#asterisk B2382F29 (~tripled@dsl-084-058-141-066.arcor-ip.net) |
| 06:14.15 | pbx123 | can you please teach me |
| 06:14.21 | pbx123 | what do i need to do |
| 06:14.32 | shido | HS1WKWTFTAD |
| 06:15.22 | pbx123 | what is that? |
| 06:15.44 | Qwell | MikeJ[Laptop]: all done :p |
| 06:15.49 | MikeJ[Laptop] | thanks |
| 06:17.34 | shido | hire some 1 who knows what the frog they are doing |
| 06:17.47 | shido | :) |
| 06:18.27 | B2382F29 | Hi, i have a problem with ISDN (using chan_capi). Incoming calls are accepted, so far no problem. Now i created an extension (7) which should dial a number on the ISDN card. exten => 7,1,Dial(CAPI/g1/0179XXXXXX) (not actually XXXX....) but when i dial that extension from an SIP connection.... i get "Called g1/0179XXXX" "CAPI Hangingup" "== No one is available to answer at this time" |
| 06:20.04 | timecop | is there a capi debug or something? |
| 06:20.07 | timecop | that might give you more info. |
| 06:20.35 | B2382F29 | i used asterisk -vvvvc .... how do i get the capi debug? |
| 06:21.47 | MikeJ[Laptop] | Qwell, thanks.. keep it up :) |
| 06:22.02 | cypromis | I thought your sleeping ? |
| 06:22.12 | cypromis | *gg |
| 06:22.15 | Smi|k | is there any standard number if DID's you get with a PRI T1? |
| 06:22.21 | Smi|k | and how much do they usually charge for more DID's? |
| 06:22.49 | Smi|k | i.e. would it be hard to get a PRI and request a block of 1000-1999 |
| 06:26.14 | B2382F29 | Sorry.... i found the error .... error 3301 ... i forgot to plug in the cable D'OH!!!! |
| 06:26.33 | Qwell | MikeJ[Laptop]: 3971 |
| 06:27.42 | Qwell | the formatting of that file is UGLY in v1-0 |
| 06:28.07 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 06:30.47 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:30.48 | tzafrir | what iax client do you recommend that supports rsa auhtentication? |
| 06:37.31 | Esteli | Has anyone been divorced after finaly figuring out asterisk? |
| 06:37.59 | oej | Esteli: I think a lot of us have been close. It's a piece of software that engulfs you totally. |
| 06:38.05 | Esteli | Ok |
| 06:38.10 | oej | Esteli: :-) |
| 06:38.13 | Esteli | Just checking ... Cheesh |
| 06:38.43 | tzafrir | Essobi, I figure it is covered by the disclaimer |
| 06:39.35 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 06:42.13 | tzafrir | iaxcomm and wxiax don't seem to support rsa keys. Any iax soft phone that does? |
| 06:44.22 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:44.33 | *** join/#asterisk Astonished (~Minestron@host190-14.pool8257.interbusiness.it) |
| 06:45.01 | *** join/#asterisk speakman (~speak@t2o30p10.telia.com) |
| 06:45.05 | speakman | hi people |
| 06:45.38 | Esteli | Hey Speakman |
| 06:45.45 | speakman | how are ya? :) |
| 06:46.20 | Esteli | Im trying to do dial plans with my eyes closed |
| 06:46.55 | Esteli | at 2:47AM EST |
| 06:47.06 | speakman | lol, not a sleepwalker, but sleepconfigurer.. :) |
| 06:47.31 | speakman | oh! here's 08:47AM CET |
| 06:47.37 | tzafrir | Esteli, just use a different TZ. |
| 06:47.43 | Astonished | Hello guys... I have some problem in going out through my isdn card. The server is online and can be connected to using "maint - password" it is also possible to try to make phone calls since the nat has been properly configured. The address is cosmapack.dyndns.org whould you help? |
| 06:47.44 | speakman | just arrived to work |
| 06:47.46 | tzafrir | export TZ=GMT |
| 06:48.10 | tzafrir | and suddenly you'll have saner a saner hour |
| 06:48.22 | speakman | btw, has anyone here used the app_pppd ? |
| 06:48.28 | speakman | or ZapRAS or anything :) |
| 06:49.29 | speakman | or 'pppd' at all along with Asterisk ;P |
| 06:49.41 | speakman | none? |
| 06:49.50 | speakman | not even digital calls at all? :) |
| 06:50.02 | speakman | on zap interface with PRI though.. |
| 06:50.29 | Esteli | Sorry |
| 06:50.30 | tzafrir | any software besides asterisk uses IAX RSA keys? |
| 06:51.15 | speakman | np.. I just have some really strange problem with my pppd connection |
| 06:51.19 | speakman | wanna hear anyway? :D |
| 06:51.34 | speakman | U may have ideas that could fix this ;) |
| 06:52.07 | Esteli | Sure |
| 06:53.10 | Esteli | tzafrir - hybrid with SSH? |
| 06:54.35 | tzafrir | Esteli, IAX2 can authenticate either using a password or using an RSA key |
| 06:55.36 | tzafrir | e.g: a default asterisk installation comes with keys for FWD and iaxtel |
| 06:55.50 | speakman | When I dial my ISDN ISP for Internet Access, it works between 0-2 minutes.. then it stop responding to both PPP EchoRequests and ICMP ECHO_REQUESTS (ping) |
| 06:55.51 | Qwell | MikeJ[Laptop]: 4406 |
| 06:55.55 | tzafrir | (their public keys, that is) |
| 06:56.40 | tzafrir | I'd like to use that, rather than passwords, for authentication over the internet |
| 06:56.44 | speakman | But when pppd tells the serial link is down, it does a CORRECTLY disconnect (the other peer gets the message!) |
| 06:56.51 | *** join/#asterisk TheEmperor (~TheEmpero@202.179.113.161) |
| 06:57.15 | TheEmperor | hello |
| 06:57.34 | TheEmperor | does anyone know if the digium boards can tell the difference between a direct line and a hunting line? |
| 06:57.40 | *** join/#asterisk outsidefactor (chrismarti@203-206-241-250.dyn.iinet.net.au) |
| 06:58.30 | Qwell | MikeJ[Laptop]: I'm not even gonna try to touch the endianess one... |
| 07:01.53 | *** join/#asterisk jeffik (~Jeff@69.158.21.177) |
| 07:02.19 | speakman | Cya people ! |
| 07:03.51 | Esteli | tzafrir makes sense, authentication is weak. It brings visions of PBX fraud toll calls |
| 07:04.00 | Esteli | to mind, like a blast from the past. |
| 07:04.28 | Esteli | The 10,000 in 900 numbers and Intl calls |
| 07:04.38 | Esteli | Yes RSA would be nice in a client |
| 07:09.38 | *** join/#asterisk ManxPower (~eric@32.199-78-194.adsl-fix.skynet.be) |
| 07:09.56 | Qwell | ManxPower: good <insert timezone greeting> |
| 07:10.11 | Qwell | morning? |
| 07:10.24 | Qwell | 9ish perhaps? |
| 07:11.31 | ManxPower | Ja |
| 07:12.21 | *** join/#asterisk Broesmeli (~broesme@195.65.2.68) |
| 07:13.20 | *** join/#asterisk Ogun (~johangrip@h236n2fls34o865.telia.com) |
| 07:15.37 | *** part/#asterisk B2382F29 (~tripled@dsl-084-058-141-066.arcor-ip.net) |
| 07:16.47 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 07:16.55 | *** join/#asterisk indego (~chris@floyd.gms.lu) |
| 07:19.29 | wfu | sorry i was gone for a while.. we had some discussion about asterisk |
| 07:21.00 | wfu | im tasked to create a config file generator for asterisk.. |
| 07:21.23 | wfu | using php |
| 07:23.47 | Broesmeli | got anyone a clue why it doesnt use the outgoing callerid i set? im in switzerland here, and the phone line provider said it should be possible, but it clearly isnt... im using a beronet bn8s0 with chan_misdn... intern (lan interface) everything works fine :x |
| 07:27.15 | ManxPower | SetCallerID(My Name <12345>) |
| 07:27.26 | ManxPower | notice the lack of quotes and the lack of () or - in the number |
| 07:27.39 | ManxPower | you also cannot set the name, only the number for most providers |
| 07:27.51 | Broesmeli | well, i tried with SetCallerID and SetCIDNum |
| 07:27.57 | Broesmeli | yeah |
| 07:28.25 | Broesmeli | but SetCallerID(0448789050) should work too, shouldnt it? because when i give out the callerid after setting it, it says this number |
| 07:29.10 | ManxPower | I don't think so. kill the leading 0 and MAYBE the country code. |
| 07:29.13 | timecop | hey ManxPower you got any suggestions for a NONFAILING calling card app? |
| 07:29.24 | ManxPower | 0 is not part of the phone number, it's a digit to indicate to the telco it's a non-local call. |
| 07:29.27 | ManxPower | timecop, no |
| 07:30.25 | timecop | damn |
| 07:30.32 | timecop | ive been fucking with asterisk-prepaid all day. |
| 07:30.43 | Broesmeli | well, if you'd call the number from another country, it'd be 0041448789050 (or +41448789050) |
| 07:30.55 | *** join/#asterisk Mavvie (edwin@dialin-196.barnet.com.au) |
| 07:31.22 | indego | Is anyone using Realtime with mysql out there? |
| 07:31.27 | Qwell | country code in cidnum? ugly |
| 07:31.42 | ManxPower | yeah, so don't put the leading 0s or country code int he callerid |
| 07:31.44 | Broesmeli | well i dunno |
| 07:32.12 | Broesmeli | i once tried kinda everything, cut 1 by 1 digits, down to only the last 2 |
| 07:32.22 | Broesmeli | coz the standard numba is 40 at the end |
| 07:34.08 | ManxPower | when you set the callerid you normally do not include the dialing prefix (0 or 00) or the country code. |
| 07:34.34 | indego | I am trying to move my config files to mysql but am having a little trouble with the mysql interface. The 'debug' file that should log the errors is not being created and therefore I see no useful errors |
| 07:35.23 | Qwell | off to bed |
| 07:37.12 | *** join/#asterisk g729 (troubled@d235-143-242.home1.cgocable.net) |
| 07:37.28 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 07:38.15 | g729 | hey, quick question. If I grab the g.729 licenses, should I grab some for any IAXy's I order? Or do they even support that codec? |
| 07:38.33 | timecop | i dont think so? |
| 07:38.38 | timecop | i think iaxy is only 711 |
| 07:39.08 | g729 | have you used it before? Does it make any sound difference? good or bad? |
| 07:39.21 | timecop | i have several licensed 729 channels. |
| 07:39.25 | timecop | its good for low bandwidth links. |
| 07:39.32 | timecop | i can do 2 channels of 729 over 64k isdn with iax. |
| 07:40.01 | g729 | how does iax work for voice on the net? run smooth? choppy? |
| 07:40.13 | g729 | consdiering ordering a dev card and a few iaxy's |
| 07:40.14 | timecop | its excellent. |
| 07:40.45 | g729 | elaborate please, in hopes of me buying more goodies to help the community:) |
| 07:40.46 | timecop | well, you understand the purpose of iaxy right? |
| 07:41.07 | g729 | you dont need a machine with cards in it |
| 07:41.07 | timecop | its ATA that speaks IAX protocol, incase you want to connect some analog phones |
| 07:41.10 | timecop | right |
| 07:41.21 | timecop | if thats your applicaiton, then it would work great for it. |
| 07:41.34 | g729 | want to be able to give them out to a few coworkers to chat with |
| 07:41.49 | timecop | on lan or over internet? |
| 07:41.50 | g729 | without buying two dev cards and a dosen machines |
| 07:41.54 | g729 | internet |
| 07:42.02 | g729 | network in house though |
| 07:42.18 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 07:42.50 | timecop | sure. it would work great for that purpose. |
| 07:42.59 | g729 | how is the voice quality over the internet with those things do you think? |
| 07:43.17 | g729 | not too lagged? (or course depending on traffic) |
| 07:43.35 | timecop | well, 711u is 64kbit codec |
| 07:43.41 | timecop | so, if you dont have that much upstream it'll be bad. |
| 07:43.50 | g729 | got about 1mbit up |
| 07:43.59 | timecop | should be good then |
| 07:44.21 | g729 | only plan on 2 for now, maybe 3 or 4 if it works well |
| 07:44.26 | timecop | its uncompressed PCM, 8khz/16bit/mono, so thats hwo good it'll sound. |
| 07:44.33 | g729 | ah |
| 07:45.08 | g729 | ever tried routing iax over encrypted links before? |
| 07:45.19 | timecop | like VPN? |
| 07:45.22 | g729 | ya |
| 07:45.29 | timecop | i have iax running over ipsec VPN here just fine |
| 07:45.38 | timecop | but ipsec is handled by hardware routers |
| 07:45.41 | g729 | k |
| 07:45.44 | timecop | so i dont need to do anything about that |
| 07:46.02 | g729 | what about full duplex? can both people interrupt each other easily while talking? |
| 07:46.18 | timecop | it is full duplex. |
| 07:46.19 | g729 | ie: latency |
| 07:46.38 | timecop | that would largely depend on your upstream/network latency. |
| 07:47.06 | g729 | I guess what im asking is, how does typical <= 100ms latency sound? |
| 07:47.24 | g729 | with the full duplex |
| 07:47.46 | g729 | like a normal phone call or different? |
| 07:48.06 | timecop | sure. i havent noticed much in the sense of delay. unless I know i'm talking with a lagged remote. |
| 07:48.24 | g729 | (sorry, would like to visualize the overall quality of the setup) |
| 07:48.55 | g729 | but does it sound like a typical lan line phone call? or some cell phone call? |
| 07:49.25 | timecop | eh, about in the middle. |
| 07:49.30 | g729 | k |
| 07:49.45 | g729 | just wondering if I should expect some tinny or echo'y effect |
| 07:50.13 | timecop | you can tune lateancy/jitter with QOS on the server(s). |
| 07:50.18 | g729 | cool |
| 07:50.30 | timecop | within limits, of course. |
| 07:50.33 | timecop | but yeah. |
| 07:51.18 | g729 | now, I was also considering using this for a voice mail box as well. I assume that the normal internal phone to lan line calls sound like a noormal call would? (ie: FXS -> FXO) |
| 07:51.29 | g729 | ie: realtime? |
| 07:51.53 | timecop | yeah. |
| 07:52.06 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 07:52.13 | blop | u got native zap bridgind |
| 07:52.18 | blop | -d+g :p |
| 07:52.37 | g729 | whats that? |
| 07:53.12 | blop | its the automatic way * is linking calls from a fxs to an fxo :p |
| 07:53.18 | g729 | i assume all the call overhead of fxs -> fxo is taken care of on the card and not subject to latency of the system? |
| 07:54.27 | timecop | dont have any mixed fxs/fxo setups, so I couldnt tell you |
| 07:54.30 | timecop | i'd imagine so. |
| 07:54.44 | timecop | digium stuff doesnt suck. |
| 07:54.54 | g729 | i should hope not |
| 07:54.58 | g729 | about to buy some |
| 07:55.32 | g729 | can asterisk use a full duplex sound card as a call source? or more specifically, a usb headset? |
| 07:55.50 | timecop | i think there's something wiht chan_oss/chan_alsa |
| 07:56.17 | timecop | to use soundcard |
| 07:56.33 | Esteli | Souldnt I be able to play any soundfile in the default sound location? |
| 07:56.44 | Esteli | At least thats what the docs lead me to believe |
| 07:56.57 | timecop | Esteli: sure. if the file is there. with .wav/.gsm .g729/wahtever extension |
| 07:57.10 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 07:57.11 | Esteli | I leave out the extention |
| 07:57.12 | g729 | i assume that the asterisk server on the net should be subject to standard firewalling rules? |
| 07:57.46 | timecop | iax is all UDP, SIP is 5060 tcp/udp + RTP ports, the rest doesnt really matter. |
| 07:57.55 | timecop | asterisk can also run as nonroot after some fucking around. |
| 07:58.50 | g729 | ah, so I assume that every packet's auth info is transmitted during each udp packet to assure only small dropouts? |
| 07:59.16 | g729 | btw, does skype or anything suppourt sip connections to asterisk? |
| 07:59.28 | timecop | no, skyke is some proprietary shit. |
| 07:59.50 | blop | we're all waiting for a chan_skype :D |
| 07:59.51 | g729 | is there any windows voip clients that do support sip to asterisk? |
| 07:59.56 | timecop | unrelated to asterisk. i think there's a few grand bounty on getting it working, but with hundreds of better voip providers, i dont really undrstand why anyone would want skyke anyway. |
| 08:00.09 | blop | g729 of course u got sip, iax win32 clients |
| 08:00.13 | g729 | looking for suggestions |
| 08:00.16 | timecop | yea there' a bonch |
| 08:00.20 | kajtzu | timecop: skype is proprietary shit intermingled with sip :) |
| 08:00.20 | timecop | we use eyebeam here |
| 08:00.23 | timecop | from xten |
| 08:00.29 | timecop | its good. |
| 08:00.30 | kajtzu | timecop: the skypeout service uses sip to talk with the pstn gateways |
| 08:00.49 | blop | timecop would be usefull to call from asterisk someone who's only capable of installing skype :D |
| 08:00.54 | timecop | kajtzu: shrug, its all kiked up anyway. there's no point for skyke, when there are tons of other providers. |
| 08:00.58 | kajtzu | timecop: yup |
| 08:01.03 | timecop | blop: then you dont call them |
| 08:01.06 | timecop | huhu |
| 08:01.16 | blop | :) |
| 08:01.31 | g729 | just looking to know if theres an easy to download/free win32 client that I could give url's to so they can get in on phone calls |
| 08:01.40 | timecop | xlite is free |
| 08:01.46 | blop | firefly works well too |
| 08:01.49 | timecop | if you're on lan, setup should be paintless |
| 08:02.06 | timecop | does firefly do SIP yet? it worked with IAX like a year ago when I tried it last |
| 08:02.34 | blop | yeah it does iax && sip && (firefly network) |
| 08:02.46 | timecop | neat. |
| 08:02.51 | blop | could be used in g729 too :p |
| 08:02.57 | timecop | does it still look like shit? |
| 08:03.00 | blop | :) |
| 08:03.01 | timecop | hm time to check it out again. |
| 08:03.48 | g729 | which can use g729? firefly? |
| 08:04.24 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 08:04.33 | timecop | pretty much any softphone you pay for will have g729 |
| 08:04.39 | timecop | eyebeam, xpro have it. |
| 08:04.41 | blop | u need to download a .dll iforgotwhere to use g729 in firefly |
| 08:05.32 | timecop | the only point to use 729 is if you got low bandwidth connections. |
| 08:05.38 | g729 | with sip, if there any provisions made with features like whiteboards etc? |
| 08:05.55 | Esteli | In my sounds directory I have national-weather-service, I put exten => s,5,Playback(National-weather-service) |
| 08:06.01 | Esteli | in my extensions.conf |
| 08:06.03 | timecop | Esteli: national-weather-service.wav? |
| 08:06.11 | Esteli | and it doesnt work |
| 08:06.42 | Esteli | gsm |
| 08:06.44 | Esteli | .gsm |
| 08:06.52 | timecop | how did oyu save it in that format? |
| 08:07.01 | timecop | did you reload hte extensions? |
| 08:07.08 | timecop | "extensions reload" or restart asterisk |
| 08:07.10 | Esteli | I didnt save it its just one of the default sound files in here |
| 08:07.13 | timecop | oh |
| 08:07.14 | Esteli | Yes |
| 08:07.16 | Esteli | <PROTECTED> |
| 08:07.22 | timecop | <PROTECTED> |
| 08:07.26 | timecop | perhaps its case sensitive |
| 08:07.29 | timecop | National->national |
| 08:07.39 | Esteli | Let me try that |
| 08:07.41 | Esteli | thanks |
| 08:07.47 | timecop | and hceck console |
| 08:07.52 | timecop | wiht -vvvc for asterisk |
| 08:07.56 | timecop | it should show if it couldnt find that file |
| 08:08.09 | Esteli | I did |
| 08:08.12 | Esteli | <PROTECTED> |
| 08:08.15 | Sato1 | chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?) <-- that means the timer is not loaded? |
| 08:08.17 | Esteli | Let me try the case |
| 08:08.22 | timecop | yea sounds like case sensitive issue. |
| 08:10.02 | Esteli | WARNING[2556]: file.c:475 ast_openstream: File national-weather-service does not exist in any format |
| 08:10.06 | Esteli | case is correct |
| 08:10.14 | Esteli | interesting |
| 08:10.18 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 08:10.41 | Esteli | I can get it to play the prompts like tt-weasels |
| 08:11.05 | Esteli | but it wont play most of the others |
| 08:13.25 | timecop | heh wtf |
| 08:13.30 | timecop | permissions? |
| 08:14.47 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:15.43 | indego | Is there anything that you need to do to get astersikt to log to /var/log/asterisk/debug |
| 08:17.14 | indego | Jun 2 10:02:04 WARNING[8671]: res_config_mysql.c:379 config_mysql: MySQL RealTime: Failed to query database. Check debug for more info. |
| 08:17.15 | indego | Jun 2 10:02:04 NOTICE[8671]: chan_sip.c:10329 reload_config: Unable to load config sip.conf |
| 08:17.15 | indego | Jun 2 10:02:04 WARNING[8671]: loader.c:393 __load_resource: chan_sip.so: load_module failed, returning -1 |
| 08:17.15 | indego | <PROTECTED> |
| 08:17.15 | indego | Segmentation fault (core dumped) |
| 08:17.32 | indego | I get no debug log.... |
| 08:17.36 | Zeeek | see logging.conf |
| 08:17.45 | indego | ok |
| 08:17.57 | Zeeek | and restart afetr changes |
| 08:18.43 | *** join/#asterisk Assid (~assid@203.115.64.56) |
| 08:18.53 | tzafrir | well, skype is propritary shit. How do I connect remote users to my office |
| 08:18.57 | tzafrir | ? |
| 08:18.58 | indego | Zeeek: cool thanks |
| 08:19.11 | tzafrir | Any recommended soft phone for that? |
| 08:19.17 | tzafrir | What do folks here use? |
| 08:19.20 | Assid | tzafrir xlite? |
| 08:19.28 | blop | any iax,sip client :p |
| 08:19.36 | blop | there are tons of it :) |
| 08:19.40 | tzafrir | SIP is kind of firewalling pain, isn't it? |
| 08:19.46 | gordonjcp | I've had a lot of success with firefly on windows, and iaxcomm |
| 08:19.54 | Assid | firefly gives me problems |
| 08:20.00 | gordonjcp | tzafrir: if it's a remote office, you should be using a VPN *anyway* |
| 08:20.39 | tzafrir | It's not a remote office. I currently just need to allow peole to "dial" remotely |
| 08:20.48 | tzafrir | Sadly, currently skype is used for that |
| 08:21.04 | tzafrir | Which I take as a personal failure |
| 08:21.06 | blop | lol :) |
| 08:21.21 | blop | remplace it with xlite || firefly 3rd party |
| 08:21.32 | blop | which work in iax |
| 08:21.45 | Assid | tzafrir: frankly.. skype aint that bad for me |
| 08:21.48 | Assid | hrmm |
| 08:21.53 | Assid | anywyas.. |
| 08:22.05 | Assid | what do i need to do for mp3 format for voicemail? |
| 08:22.17 | Assid | i tried mp3 as a format.. doesnt work |
| 08:22.19 | tzafrir | I wanted to use rsa authentication. I kind of don't trust passwords. |
| 08:22.47 | Assid | tzafrir: officially.. you vpn.. then softphone |
| 08:23.27 | tzafrir | Assid, well, skype manages to avoid that. IAX should be able to avoid that. So I would like to avoid that |
| 08:23.56 | tzafrir | Assid, why mp3? What's wrong with wav? |
| 08:24.10 | tzafrir | mp3 writing is expensive |
| 08:24.14 | *** part/#asterisk outsidefactor (chrismarti@203-206-241-250.dyn.iinet.net.au) |
| 08:24.20 | Assid | too big |
| 08:24.27 | tzafrir | not to mention patent infriging |
| 08:24.38 | tzafrir | Assid, gsm? |
| 08:24.42 | Assid | arent there any free codecs out there? |
| 08:25.10 | tzafrir | Assid, vorbis |
| 08:25.30 | tzafrir | But I figure it would be just as cpu-intensive |
| 08:25.32 | JerJer | don't forget LPC-10 |
| 08:25.38 | Assid | hrmm |
| 08:25.47 | Assid | okay.. how do i make it email the gsm file instead of wav? |
| 08:26.53 | blop | which codec should be used to send fax over ip ? :) |
| 08:27.05 | JerJer | none |
| 08:27.13 | JerJer | fax is not reliable over ip |
| 08:27.22 | blop | thats what i tought :) |
| 08:27.47 | tzafrir | Assid, it seems that my default configuration writes both wav, WAV and gsm voicemail messages |
| 08:28.03 | blop | why not having a dedicated codec for fax ? |
| 08:28.19 | tzafrir | have you lookeds at /var/spool/asterisk/voicemail/INBOX ? |
| 08:29.00 | gordonjcp | blop: uhm, because the switch at the far end would need to support it too |
| 08:34.07 | *** join/#asterisk isam (~isam@81.10.126.2) |
| 08:34.25 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 08:34.30 | tzafrir | does iax password authentication use any decent chalange-response? or will sniffing the traffic be useful for faking authentication? |
| 08:34.45 | JerJer | depends on auth method used |
| 08:34.58 | JerJer | plain text, sure |
| 08:35.04 | JerJer | md5, sure |
| 08:35.13 | JerJer | both are subject to the replay attack |
| 08:35.16 | JerJer | but RSA is not |
| 08:35.17 | isam | it seems that the WAIT doesn't work on a line before it is ANSWERed .. I wanted to delay the answer for a couple of rings, and once I put WAIT before ASNWER even if it is WAIT (5) the line is never answered |
| 08:35.21 | isam | any idea why ? |
| 08:35.55 | tzafrir | So which clients support RSA authentication? |
| 08:36.19 | oej | tazfrir: Asterisk :-) |
| 08:36.54 | tzafrir | I can't be the first genius to think about this. This should be an important feature for iax phones |
| 08:37.06 | gordonjcp | yeah |
| 08:37.31 | gordonjcp | tzafrir: why not just use a vpn though? |
| 08:37.42 | RoyK | gordonjcp: vpn adds latency |
| 08:38.31 | tzafrir | vpn adds setup headaches. I also aim for a simple setup. Not only for us, but for others |
| 08:38.45 | gordonjcp | RoyK: no biggie |
| 08:38.59 | gordonjcp | tzafrir: it's a fair point though |
| 08:39.03 | gordonjcp | iax-ssl |
| 08:39.12 | tzafrir | ~iax-ssl |
| 08:39.24 | tzafrir | ~google iax-ssl |
| 08:40.50 | tzafrir | google suggests looking for aix-ssl |
| 08:41.22 | gordonjcp | yes |
| 08:41.35 | gordonjcp | I don't think iax-ssl exists |
| 08:41.39 | gordonjcp | but there's no good reason for that |
| 08:41.49 | tzafrir | SSL is a layer on top of TCP |
| 08:42.14 | gordonjcp | yup |
| 08:42.18 | tzafrir | How can it relate to a UDP protocol like IAX? |
| 08:42.43 | gordonjcp | it doesn't have to be *exactly* the same as IAX |
| 08:42.48 | tzafrir | If you want to encrypt IAX's data, you can't use SSL |
| 08:43.12 | gordonjcp | well then we're back to tunnelling |
| 08:43.18 | tzafrir | But I was talking just about the initial authentication |
| 08:43.36 | gordonjcp | you could in theory encrypt the stuff way before it goes over the wire |
| 08:43.36 | tzafrir | (this won't prevent anybody from slashing into an existing session |
| 08:43.44 | gordonjcp | yes |
| 08:43.49 | RoyK | there are someone working on IAX crypto |
| 08:48.02 | *** join/#asterisk da_monumental_1 (~da_monume@cpe-065-191-085-021.nc.res.rr.com) |
| 08:50.25 | outofjungle | cdr_mysql or cdr_odbc? what is recomended? |
| 08:51.51 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 08:52.08 | Esteli | tzafrir http://www.kauss.org/Stephan/ziaxphone/ |
| 08:52.13 | Esteli | did you look at that one? |
| 08:52.44 | Esteli | <PROTECTED> |
| 08:52.54 | Esteli | Its alpha but ... |
| 08:53.20 | RoyK | outofjungle: if you're using mysql, I'd recommend cdr_mysql |
| 08:53.35 | outofjungle | RoyK: thanks. |
| 08:53.39 | RoyK | i really don't see the point of adding another abstraction layer between asterisk and the db |
| 08:53.41 | *** join/#asterisk pat_lehem (~lehem@212.150.210.2) |
| 08:53.49 | pat_lehem | hi all. |
| 08:54.08 | pat_lehem | I have a question about NVLineDetect, is anyone familiar with it here? |
| 08:55.11 | *** join/#asterisk psycodad (~obiwan@2001:4060:4419:b1:0:0:0:2) |
| 08:56.31 | *** join/#asterisk wasim (~wasim@203.81.201.188) |
| 08:56.36 | *** part/#asterisk wasim (~wasim@203.81.201.188) |
| 09:01.40 | Maksim | Hi. Where can I read about "No authority found" with iax? |
| 09:05.14 | *** join/#asterisk _kno_ (~kno@80-28-236-136.adsl.nuria.telefonica-data.net) |
| 09:05.29 | _kno_ | hi |
| 09:08.15 | *** join/#asterisk lehel (~max@86.125.98.100) |
| 09:08.58 | *** join/#asterisk jmorn (~jmorn@hst8.kraftnat.aland.fi) |
| 09:10.09 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
| 09:11.18 | *** join/#asterisk wasim (~wasim@203.81.201.188) |
| 09:15.24 | *** join/#asterisk Assid (~assid@203.115.64.56) |
| 09:15.34 | Assid | stupid isp |
| 09:18.23 | *** part/#asterisk lehel (~max@86.125.98.100) |
| 09:18.47 | doolph | sup |
| 09:18.59 | Assid | nm |
| 09:19.08 | Assid | gotta go help someone install some softwares |
| 09:19.10 | Assid | :| |
| 09:19.16 | Assid | stupid windows users |
| 09:19.21 | Assid | bbiab |
| 09:19.36 | doolph | lol |
| 09:20.35 | *** join/#asterisk JerJer (~JerJer@jerjer.bronze.supporter.pdpc) |
| 09:20.43 | _kno_ | which hardware should I use if I whant to manage 20 analog lines? |
| 09:22.29 | wasim | _kno_: tdm04 x5 , or a te100 + cb |
| 09:22.46 | RoyK | _kno_: don't listen to wasim |
| 09:23.20 | _kno_ | ??? |
| 09:23.44 | doolph | lol |
| 09:23.55 | Zeeek | don't listen to RoyK either! |
| 09:23.57 | doolph | 20 analog incoming right? |
| 09:23.58 | wasim | _kno_: /me kicks RoyK on his diverting leg |
| 09:24.06 | _kno_ | doolph: yes |
| 09:24.24 | doolph | tdm400 x 5 or te100 + Channel bank |
| 09:24.27 | doolph | with fxo modules |
| 09:24.37 | wasim | vindicated! |
| 09:26.03 | *** join/#asterisk Blackvel (~blackvel@dsl-084-057-124-053.arcor-ip.net) |
| 09:26.15 | *** part/#asterisk Blackvel (~blackvel@dsl-084-057-124-053.arcor-ip.net) |
| 09:28.45 | _kno_ | and somebody knows a vendor in spain? |
| 09:29.39 | RoyK | milk vendor? |
| 09:29.43 | RoyK | or shoes? |
| 09:30.32 | wasim | sales@avanzada7.com |
| 09:30.51 | wasim | but don't listen to me! |
| 09:31.40 | *** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr) |
| 09:31.57 | Ahrimanes | anyone know the pricing for the various cisco hardphones? |
| 09:32.02 | Martohtar | yvebe71j |
| 09:32.14 | Martohtar | bah, wrong window, sorry... |
| 09:32.25 | Ahrimanes | oh really |
| 09:32.59 | bublbobl | For information, fax over IP (G711-alwa or ulaw) work well with asterisk, I'm quite happy of my first experiments. :-) |
| 09:32.59 | _kno_ | wasim: are you from avanzada7? |
| 09:34.12 | Zeeek | Martohtar what's the username that goes with that password |
| 09:34.57 | wasim | _kno_: not that I'm aware of, but better ask RoyK |
| 09:35.07 | Martohtar | Zeeek: pass? nah, that's good morning in my native tounge ;) |
| 09:35.18 | Zeeek | with 71? |
| 09:35.33 | Zeeek | oh, wrong character set |
| 09:35.49 | wasim | Zeeek: 71 is tl |
| 09:36.14 | Zeeek | what is tl ? |
| 09:36.25 | Zeeek | 71? |
| 09:36.40 | wasim | 31337 speek |
| 09:38.41 | Zeeek | I'm too old for this! |
| 09:39.08 | JerJer | age is a state of mind |
| 09:39.26 | wasim | this is your brain on * |
| 09:40.18 | Zeeek | you have to have a mind in the first place in order for it to have a state |
| 09:40.35 | Zeeek | I think I just receive a free IAX hardphone |
| 09:40.49 | wasim | lucky bum |
| 09:41.06 | Zeeek | the bad news is, it isn't the best kind |
| 09:41.18 | wasim | then its not a farfon |
| 09:41.19 | Zeeek | but I should test it and see if it works |
| 09:41.31 | Zeeek | it's white as a matter of fact |
| 09:41.43 | wasim | someone take me away and lock me up, please |
| 09:41.51 | Zeeek | wasim - that was amusing but I ned more coffee to really guffaw |
| 09:42.08 | Zeeek | come to Paris for lunch |
| 09:42.14 | JerJer | fresh out of slePP's oven |
| 09:42.22 | wasim | someday, mon ami ... someday |
| 09:45.34 | Zeeek | porridge... that's something out of a fairy tale |
| 09:45.43 | Zeeek | no one eats porridge |
| 09:45.58 | gordonjcp | porridge is good |
| 09:46.17 | wasim | porridge rocks, especially freshly roasted in the morning |
| 09:46.17 | Zeeek | so after 6 weeks of our DSL connection stopping every 8 minutes, the problem mysteriously disappeared |
| 09:46.27 | Zeeek | Weirdest thing I've ever seen |
| 09:47.25 | gordonjcp | rusty wire |
| 09:47.41 | gordonjcp | was it *bang on* 8 minutes? |
| 09:47.50 | Zeeek | wasn't great for asterisk, had to move that box to a different connection |
| 09:48.30 | Zeeek | It was exactly every 8min +2seconds. Every piece of hardware was changed, the two DSL lines come in on the same cable, one stopped every 8m one was fine |
| 09:48.49 | Zeeek | I was pretty sure it was at the DSLAM |
| 09:49.09 | gordonjcp | exactly 482 seconds? |
| 09:49.22 | gordonjcp | yeah, some sort of misconfiguration |
| 09:49.28 | Zeeek | the phone co here is a mafia and they never want to fix their stuff because they're forced to allow other companies to provde DSL on the "their" lines |
| 09:50.05 | Zeeek | as close as I could measure, it was 8m02s but it was prolly 8.02.45635243 or something |
| 09:51.04 | Zeeek | because it didn't stay in *perfect* sync. But it wasn't synced with our equip. Turning off the power here didn't change the expected time of interruption. Syncho wasn't lost by the modem, but all data stopped for around 5-10 second |
| 10:02.29 | tzafrir | Does iax support ChanIsAvail? |
| 10:03.35 | RoyK | think so, in HEAD |
| 10:04.18 | *** join/#asterisk switch (~switch@61.206.115.5.user.ad.il24.net) |
| 10:05.19 | *** join/#asterisk kks (~kks@202.73.9.10) |
| 10:05.52 | *** join/#asterisk switch (~switch@61.206.115.5.user.ad.il24.net) |
| 10:11.16 | JerJer | tzafrir: but why? |
| 10:11.20 | JerJer | just attempt to dial the call |
| 10:11.39 | JerJer | then examine ${DIALSTATUS} to determine if it failed |
| 10:11.45 | JerJer | and act accordingly |
| 10:16.07 | JerJer | does anyone else get these messages very very regularly: |
| 10:16.09 | JerJer | Jun 2 05:14:51 NOTICE[2820]: pbx.c:1595 pbx_extension_helper: Cannot find extension context 'aasterisk' |
| 10:16.23 | *** join/#asterisk DT-V (~sjaaknabu@fia254-108-100.dsl.mxposure.nl) |
| 10:16.27 | gordonjcp | 'aasterisk' might be a typo |
| 10:16.30 | JerJer | or do I have another lamer with no clue trying to send a call to the extension 'aasterisk' |
| 10:16.56 | JerJer | 'aasterisk' is not in any of my dialplan |
| 10:17.18 | wasim | there should be catchall context option |
| 10:17.39 | Zeeek | an incompetent hacker? |
| 10:18.19 | Zeeek | do like they do with domain names: make a context assterisk and aasterisk with porno sites as destinations |
| 10:18.51 | JerJer | the problem is it is on a very highly utilized box so adding additional debug to that statement is going to be non-trivial :( |
| 10:19.07 | Zeeek | I remember once typing altavist.com by accident and seeing a full beaver spread with "IS THIS WHAT YOU'RE LOOKINGFOR?" |
| 10:19.45 | Zeeek | happened the other days with macromedai.com |
| 10:20.50 | JerJer | lol |
| 10:22.19 | *** join/#asterisk B2382F29 (~tripled@dsl-084-058-141-066.arcor-ip.net) |
| 10:23.09 | Zeeek | what pisses me off the most is stuff like "Find HP15243 at eBay!" |
| 10:23.37 | gordonjcp | yes |
| 10:23.48 | gordonjcp | pages and pages and pages of it |
| 10:23.52 | Zeeek | so you look up "Warning: unexpected character PHP4" in google and it lists Find "Warning: unexpected character PHP4" at eBay! |
| 10:24.04 | Zeeek | fuck those ad people |
| 10:24.07 | gordonjcp | ah, no, I mean the sales link farms |
| 10:24.18 | Zeeek | yeah exactly |
| 10:24.36 | gordonjcp | "FIND XXX BEST PRICE XXX WE HAVE KDJ-11A BEST PRICE ALL COMPUTER PARTS" |
| 10:24.38 | Zeeek | "find cure for herpes at eBay" |
| 10:24.48 | gordonjcp | when you try to search for PDP11 jumper settings |
| 10:25.14 | Zeeek | also, if you do look for an HP50 or something,n half the links say "e are currently out of HP50" and then list other stuff |
| 10:25.24 | Zeeek | yeah |
| 10:27.54 | *** part/#asterisk B2382F29 (~tripled@dsl-084-058-141-066.arcor-ip.net) |
| 10:30.45 | psycodad | can I dynamically set the value of a sip friends fromuser= setting ? If yes, which variable would I have to set ? |
| 10:31.49 | *** join/#asterisk flotox (jovan@host156-75.pool80183.interbusiness.it) |
| 10:36.58 | *** join/#asterisk tzafrir_laptop (~tzafrir@62.90.10.53) |
| 10:43.38 | *** join/#asterisk B2382F29 (~tripled@dsl-084-058-138-126.arcor-ip.net) |
| 10:44.51 | B2382F29 | Hi, i have a problem with chan_capi, when calling asterisk, the standard demo is running, but when i press a key, i get this: "chan_capi.c:875 capi_read: Fax detected, but no fax extension" |
| 10:51.07 | JerJer | create a fax extension |
| 10:51.29 | JerJer | or it smells like chan_capi is borken |
| 10:51.44 | B2382F29 | but that is not a fax, it misrecognizes dialtones as fax |
| 10:51.53 | JerJer | borken |
| 10:54.12 | B2382F29 | known issue? nobody else had that problem? |
| 10:55.15 | JerJer | i'm on the wrong side of the big pond, so i don't use capi |
| 10:55.16 | JerJer | sorry |
| 10:55.32 | JerJer | turn off fax detect |
| 10:55.36 | JerJer | as a quick fix |
| 10:55.55 | blop | where do we turn fax detect on/off ? |
| 10:56.27 | B2382F29 | JerJer, yes, how do i turn it off? |
| 10:57.12 | JerJer | comment out the dsp function in chan_capi |
| 10:57.27 | JerJer | perhaps around line 875 ? |
| 10:58.24 | B2382F29 | softdtmf=1 is in capi.conf, that should disable hardware dtmf detection.... |
| 10:59.14 | blop | yes but dtmf detection is used for other things too |
| 11:00.25 | blop | zapata.conf:faxdetect=no for myself :) |
| 11:03.43 | *** join/#asterisk cmk (~cmk_@p54A3F981.dip.t-dialin.net) |
| 11:04.15 | B2382F29 | found the error .... gentoo had chan_capi in version 0.4.xxx... i installed 0.3.5 and it worked.... |
| 11:07.01 | B2382F29 | btw, someone knows an SIP-Client for linux (not Linphone)...? |
| 11:07.15 | Ahrimanes | B2382F29: kphone |
| 11:07.44 | JerJer | asterisk |
| 11:08.40 | *** join/#asterisk meppl (mephisto@p54AAC0E7.dip.t-dialin.net) |
| 11:08.53 | B2382F29 | JerJer ... hmmm... good idea ... you mean calling from console/alsa .... but it lacks a GUI |
| 11:10.10 | Ahrimanes | B2382F29: kphone works quite good, can even have video with webcam |
| 11:11.45 | B2382F29 | Ahrimanes, sounds good, i will try, but i use gnome as primary desktop... so a native gnome client would be even better ... ok, thanks for all, bye... |
| 11:11.53 | *** part/#asterisk B2382F29 (~tripled@dsl-084-058-138-126.arcor-ip.net) |
| 11:17.01 | JerJer | gooies are for MIS morons |
| 11:17.07 | *** join/#asterisk Lo0 (~root@support.variant6.bg) |
| 11:17.26 | JerJer | a real geek doesn't require a gooie to simply make a telephone call |
| 11:17.53 | Lo0 | Hello |
| 11:18.26 | Lo0 | can I ask you is there any way to proxy between H.323 and NCS? |
| 11:20.47 | *** join/#asterisk stefanocarlini (~stefano@213.233.11.14) |
| 11:23.22 | Lo0 | or anybody using asterisk for PacketCable operations? |
| 11:23.44 | stefanocarlini | hello |
| 11:24.04 | stefanocarlini | i'm writing a paper about * for my floss master |
| 11:24.29 | stefanocarlini | I nedd info about the project organization: how many developers |
| 11:24.40 | stefanocarlini | leadership |
| 11:24.53 | tzafrir | JerJer, is there any decent non-gui sip soft-phone? |
| 11:24.53 | stefanocarlini | and so on... any one can help me? |
| 11:26.23 | tzafrir | stefanocarlini, some interviews: http://www.sineapps.com/news.php |
| 11:27.26 | tzafrir | gnome-meeting's latest versions support sip, btw |
| 11:27.38 | tzafrir | but that B@-whatever has left |
| 11:28.04 | stefanocarlini | thanks tzafrir |
| 11:30.02 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 11:34.04 | stefanocarlini | how many developers work at Asterisk? I'm trying to find out this info form the sources but isn't so simple |
| 11:35.40 | skeffling | stefanocarlini, www.digium.com are the primary developer |
| 11:36.45 | stefanocarlini | yes I know. But I think there is not only Mark Spencer working on the code |
| 11:37.30 | skeffling | no, I'm not sure either but there are lots of contributers all over the world |
| 11:39.46 | stefanocarlini | yes. I'm finding interesting the sineapps interviews |
| 11:43.59 | MikeJ[Laptop] | who do you consider an asterisk developer? |
| 11:44.22 | MikeJ[Laptop] | anyone who submits code? anyone with more than 1000 lines of code? |
| 11:44.58 | MikeJ[Laptop] | depending on how you look at it there are anywhere between 6 and 25 active devs or so |
| 11:45.47 | tzafrir | OT: where can I find information regarding "push mode" html? basically I want to create a dynamic status page and naturally would rather avoid flash/java/local-client |
| 11:47.04 | JerJer | good luck |
| 11:47.07 | cypromis | Lo0: look at asterisk.urtho.net |
| 11:47.16 | tzafrir | is there any such existing interface? FOP is flash :-( |
| 11:47.28 | JerJer | tzafrir: you can fake it out by constantly refreshing the page |
| 11:47.32 | JerJer | but how sane is that? |
| 11:47.49 | tzafrir | JerJer, may work for one or two users. Won't scale |
| 11:47.57 | JerJer | hell no it won't scale |
| 11:48.06 | JerJer | so you are pretty muched fucked - java |
| 11:48.14 | JerJer | much |
| 11:48.43 | tzafrir | JerJer, the basic idea is to simply never tell the browser you finished loding the page |
| 11:48.59 | JerJer | that won't cut it either |
| 11:49.06 | tzafrir | and keep pushing update javascript instructions |
| 11:49.16 | JerJer | you will burn up peoples memory like mad |
| 11:49.37 | psycodad | is there a way to increase the dtmf timeout for entering extensions when transferring calls ? |
| 11:49.39 | tzafrir | Some chat clients work that way |
| 11:50.36 | doolph | where can I find demo scripts of oh323 trunks |
| 11:53.25 | doolph | noone? |
| 11:53.26 | gordonjcp | there |
| 11:53.34 | gordonjcp | now, let's have no more silly talk of h323 |
| 11:53.38 | gordonjcp | <shudder> |
| 11:56.06 | JerJer | asterisk/channels/h323/h323.conf.sample and README |
| 12:00.03 | *** join/#asterisk BerndR (~konversat@mich2-145-8.utaonline.at) |
| 12:00.19 | BerndR | hey all |
| 12:00.34 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l03v-23-230.d4.club-internet.fr) |
| 12:00.55 | PoWeRKiLL | hi |
| 12:01.09 | PoWeRKiLL | someone send fax with asterisk ? |
| 12:01.24 | doolph | not me |
| 12:01.26 | doolph | but why |
| 12:04.49 | stefanocarlini | I'm interested in realize a fax server with asterisk |
| 12:04.49 | BerndR | in one section of my extensions i try to make an 'i' extension which looks for the extension in an other section? |
| 12:04.49 | BerndR | ??? |
| 12:05.14 | BerndR | ok, the caller is in section "foo" |
| 12:05.17 | postel | Since i only roll cvs installations for maself i have no clue of cd asterisk-in-a-box setups, i need to give someone a cd to test asterisk for himself, he knows the basics and have used my setup, any recomendations? |
| 12:05.26 | BerndR | presses i.e 7 |
| 12:05.41 | BerndR | in section foo there is no extension for 7 |
| 12:05.54 | BerndR | so the extension i gets active |
| 12:05.59 | JerJer | app_txfax works beautifully for us |
| 12:06.23 | JerJer | rxfax seems to have trouble receiving faxes from time to time, but overall pretty damn good |
| 12:06.31 | BerndR | now i asterisk should look for the extension 7 in section "bar" |
| 12:06.41 | BerndR | any idea? |
| 12:06.57 | JerJer | there is also another fax app based on the spandsp libs - but that link is on another computer at the moment |
| 12:07.12 | stefanocarlini | what is app_txfax? |
| 12:09.12 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:09.12 | postel | Well? 246 people and nobody has a demo cd that works alright to recommend? |
| 12:09.44 | JerJer | a cd? |
| 12:09.54 | RoyK | stefanocarlini: application to send a fax.... in spandsp |
| 12:10.08 | stefanocarlini | ok. thanks |
| 12:10.08 | JerJer | postel: you expect us to do all of that work for you? |
| 12:10.11 | JerJer | for nothing? |
| 12:10.27 | Ahrimanes | postel: http://www.automated.it/asterisk/ |
| 12:10.28 | JerJer | and its 248 people |
| 12:10.32 | Ahrimanes | hehe |
| 12:10.34 | newl | Yes, chop chop! :) |
| 12:10.35 | postel | JerJer: no. not at all, i just want you to tell me "GOOGLE FOR $THIS" |
| 12:10.44 | stefanocarlini | poster: look for asterisk @ home |
| 12:10.47 | JerJer | cvs co asterisk |
| 12:10.49 | JerJer | make |
| 12:10.51 | JerJer | make install |
| 12:10.53 | stefanocarlini | or xorcom |
| 12:11.06 | JerJer | then use your fav editor to bang out config files |
| 12:11.06 | postel | jeremywhiting: i said i use cvs myself, i want a cd for somebody else |
| 12:11.12 | postel | stefanocarlini: thank you |
| 12:11.24 | JerJer | doesn't work like that |
| 12:11.53 | postel | JerJer: what doesnt work like that? |
| 12:12.01 | Ahrimanes | postel: http://www.automated.it/asterisk/ <- has a livecd download |
| 12:12.32 | postel | thanks |
| 12:12.38 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:13.27 | JerJer | lol |
| 12:13.39 | JerJer | they require a 64 meg flash minimum |
| 12:13.48 | JerJer | talk about bloat |
| 12:14.15 | stefanocarlini | another livecd is here: http://www.knopsterisk.com/ |
| 12:14.45 | *** join/#asterisk The_Duke (~the_duke@80.92.64.103) |
| 12:14.49 | postel | interesting |
| 12:15.41 | PoWeRKiLL | JerJer how do you create the tiff file for sending ? |
| 12:15.54 | *** join/#asterisk squirrelv5 (~squirrelv@202.57.81.146) |
| 12:15.56 | JerJer | lol that one wants 10 bucks |
| 12:16.02 | Maksim | which is the best linux distro you can recommend to use asterisk with fast upgrade and etc? (-: |
| 12:16.09 | JerJer | PoWeRKiLL: scanner? app_rxfax ? |
| 12:16.12 | squirrelv5 | hello everybody |
| 12:16.17 | JerJer | Maksim: asterisk doesn't care |
| 12:16.21 | squirrelv5 | elmer from philippines here |
| 12:16.32 | JerJer | what's up doc? |
| 12:16.47 | *** join/#asterisk cpm (~Chip@cpm.sustaining.supporter.pdpc) |
| 12:16.51 | The_Duke | hello, how can i tell my asterisk to sent the from field as sip:12345678@my.host.com instead of sip:12345678@111.222.333.123 |
| 12:17.01 | PoWeRKiLL | when i receive a fax with rxfax then i send it back it's ok but i want to make a printer like to create the tiff file and I don't find specification |
| 12:17.20 | JerJer | pdf2tiff |
| 12:17.55 | Maksim | JerJer, what is your distro? |
| 12:18.03 | JerJer | my own |
| 12:19.19 | squirrelv5 | i need help |
| 12:19.36 | squirrelv5 | are there some issues on sip on nat behind firewall |
| 12:19.53 | RoyK | squirrelv5: is the sip client or the asterisk server behind firewall? |
| 12:19.54 | JerJer | busybox+glibc+linux2.4+iptables+openssl+zaptel+asterisk = my distro |
| 12:20.40 | squirrelv5 | the server and some clients too |
| 12:20.52 | squirrelv5 | is there a number i can test |
| 12:21.10 | squirrelv5 | i successfully setup IAX2 |
| 12:21.13 | squirrelv5 | via FWD |
| 12:22.52 | *** join/#asterisk TonyM (~softins@adsl-solo-80-168-224-238.claranet.co.uk) |
| 12:23.56 | TonyM | anyone here familiar with Firefly? |
| 12:24.10 | squirrelv5 | i cant make it to work with iptel |
| 12:24.26 | JerJer | isn't firefly an insect? |
| 12:24.55 | TonyM | hehe - I want to know the general IAX url format for initiating guest calls |
| 12:24.59 | JerJer | newl: i installed asterisk on a iPAQ |
| 12:25.11 | JerJer | or was it a treo - hmmm |
| 12:25.15 | RoyK | firefly, family Lampyridae |
| 12:25.19 | RoyK | http://en.wikipedia.org/wiki/Firefly |
| 12:25.30 | JerJer | some handheld i was given for a weekend to play with |
| 12:25.56 | JerJer | that already had linux with a telnetd client :) |
| 12:26.22 | gordonjcp | JerJer: you sick sick weirdo |
| 12:26.25 | JerJer | asterisk started up and was able to deal with like one inbound SIP call playing a prompt, but that's all we could really make it do |
| 12:26.59 | JerJer | just wait - asterisk running on an IP phone :D |
| 12:27.02 | gordonjcp | JerJer: I'm going to have to try running asterisk on my Workpad Z50 now... |
| 12:28.47 | *** part/#asterisk TonyM (~softins@adsl-solo-80-168-224-238.claranet.co.uk) |
| 12:29.50 | gordonjcp | JerJer: one of those little gumstix dinky wee boards would rock |
| 12:30.10 | bjohnson | I've posted a message to the mgetty mailing list that describes how to fax out from openoffice on a linux standalone or lan environment |
| 12:30.26 | bjohnson | tzanger: you were interested in that ^^ |
| 12:30.38 | *** join/#asterisk P-NuT (~P-NuT@CPE-60-225-214-14.nsw.bigpond.net.au) |
| 12:30.44 | P-NuT | Hey all. |
| 12:30.54 | P-NuT | Need some ubuntu asterisk help |
| 12:31.43 | P-NuT | Anyone installed it on ubuntu or Debian? |
| 12:32.24 | P-NuT | Is anyone here tonight? |
| 12:32.32 | pjz | good morning |
| 12:32.33 | gordonjcp | it's the middle of the day, I'm at work |
| 12:32.36 | pjz | :) |
| 12:32.43 | P-NuT | Oh cool. |
| 12:32.46 | pjz | it's morning, I'm not at work yet :) |
| 12:32.49 | gordonjcp | heh |
| 12:32.49 | P-NuT | it's 10:30 here |
| 12:32.59 | gordonjcp | am or pm? |
| 12:32.59 | P-NuT | Cool. |
| 12:33.16 | P-NuT | So I'm installing asterisk on a fresh ubuntu system.... |
| 12:33.19 | P-NuT | pm |
| 12:33.19 | squirrelv5 | help on iptel SIP pls |
| 12:33.34 | squirrelv5 | cant work it behind a firewall |
| 12:34.00 | squirrelv5 | and on my iptel account page displays a my 1234@private ip |
| 12:34.03 | squirrelv5 | why is that |
| 12:36.47 | *** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 12:38.12 | squirrelv5 | anyone online? |
| 12:38.51 | Ariel_ | Hello everyone |
| 12:39.43 | *** join/#asterisk meppl (mephisto@p54AAD9E9.dip.t-dialin.net) |
| 12:40.11 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 12:40.42 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 12:49.01 | *** join/#asterisk B2382F29 (~tripled@dsl-084-058-138-126.arcor-ip.net) |
| 12:50.18 | B2382F29 | hi, again, i have a problem with Wait and chan_capi... with Wait,1 the line gets answered after 1 second, but with Wait,9 it doesn't answer even after 20 sec. |
| 12:53.32 | B2382F29 | Wait,4 works also most of the time, but values > 6 never work |
| 12:56.28 | inspired | best way to load balance asterisk? |
| 12:56.31 | Katty | miow. |
| 12:56.45 | Ariel_ | hello Katty hope this morning is going well. |
| 12:58.39 | Katty | i'm still drippy from teh shower, but good..yes. |
| 13:00.11 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 13:00.19 | lehel | hello |
| 13:00.43 | Katty | hihi |
| 13:01.39 | gordonjcp | hello lehel, Katty |
| 13:01.47 | lehel | what do you think people.. if i have 13 different locations not so far away (50 miles max) |
| 13:02.06 | lehel | what could be the solution to connect them with asterisk? |
| 13:03.02 | gordonjcp | have they got broadband? |
| 13:03.47 | gordonjcp | lehel: if they had line-of-site, you could use wifi |
| 13:07.02 | *** join/#asterisk zztopper (~me@ip70-177-50-126.br.br.cox.net) |
| 13:07.14 | lehel | gordonjcp: let's assume they have broadband, i don't know for sure yet |
| 13:07.27 | gordonjcp | mmmm |
| 13:07.35 | *** join/#asterisk irv999 (~irv999@ool-44c241c7.dyn.optonline.net) |
| 13:07.36 | gordonjcp | softphones, or real ones? |
| 13:07.37 | irv999 | Lo all |
| 13:08.17 | irv999 | Can someone answer a logiscitcs issue on a polycom phone via call management? |
| 13:08.36 | *** join/#asterisk vandien (~stephan@p5091A574.dip0.t-ipconnect.de) |
| 13:08.38 | irv999 | I am really stuck at a roadblock.. and I need to solve this. I dont know if my programmer is capapble |
| 13:09.26 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:09.47 | lehel | softphone, you mean with headset? < yes, not necessary real phones |
| 13:09.57 | zztopper | i have a general question about asterisk... we're evaluating it .... i'm somewhat concerned about all the 'difficulties' i see on forums... does this stuff work ok in a standard installtion if u use recommended OS and machine? |
| 13:10.14 | irv999 | zztopper: yes |
| 13:10.29 | *** join/#asterisk Mavvie (edwin@edwin.adsl.barnet.com.au) |
| 13:10.33 | irv999 | zz: as long as you use recommended hardware.. you are ok.. you don't need a powerful machine |
| 13:10.45 | irv999 | zzz: a cheap dell works.. lots of ram.. pentium, |
| 13:10.52 | gordonjcp | lehel: well, IAX2 is less hassle to get through NAT |
| 13:11.56 | lehel | what could be the ideal connection and solution? |
| 13:12.38 | gordonjcp | lehel: well, my gf connects to my asterisk server over 1Mbps cable, with no problems at all |
| 13:13.05 | gordonjcp | one of my friends connects from Turkey, and only has problems 'cos the mike socket on his laptop is crap |
| 13:13.15 | gordonjcp | he's on IIRC 256kbps ADSL |
| 13:14.07 | RoyK | anyone here using SER? |
| 13:14.56 | lehel | gordonjcp: should be enough one asterisk server? with what kind card? (wildcard?) ..zap/sip/iax2 ..? |
| 13:15.15 | gordonjcp | lehel: depends what you want to do |
| 13:15.32 | gordonjcp | how many users? |
| 13:17.33 | lehel | for each location ..'round 10 > means 130 users ? |
| 13:18.45 | *** part/#asterisk stefanocarlini (~stefano@213.233.11.14) |
| 13:23.30 | lehel | gordonjcp: i'm kinda' beginner in asterisk (however i have local a running asterisk), but i don't know what is the solution to connect 13 location.. as they can talk each other.. voip |
| 13:24.42 | blitzrage | lehel: DUNDi perhaps |
| 13:24.49 | blitzrage | lehel: in conjunction with IAX |
| 13:24.56 | gordonjcp | lehel: give IAX a try |
| 13:25.00 | gordonjcp | blitzrage: heh |
| 13:25.06 | blitzrage | gordonjcp: :D |
| 13:25.13 | blitzrage | gordonjcp: great minds think a like |
| 13:25.20 | gordonjcp | lehel: if your asterisk box is behind NAT you will only need to open up one port |
| 13:25.23 | DeeJayTwo | Has anybody got problems with zaptel drivers freezing the system? |
| 13:25.30 | MikeJ[Laptop] | which great minds? |
| 13:25.35 | blitzrage | MikeJ[Laptop]: ! |
| 13:25.40 | gordonjcp | MikeJ[Laptop]: shurrit |
| 13:25.40 | DeeJayTwo | We got this problem with TE410P |
| 13:26.49 | blitzrage | went to bed at 3:30, got up at 8:30 without an alarm, what the hell am I doing! |
| 13:28.08 | OnlyMe | blitzrage getting old |
| 13:28.09 | OnlyMe | :) |
| 13:28.34 | blitzrage | OnlyMe: ... I'm 24 :) |
| 13:29.05 | blitzrage | I'm nearly half way to 50! :) |
| 13:29.24 | OnlyMe | blitzrage yeap but ...counting |
| 13:29.24 | MikeJ[Laptop] | whatever |
| 13:29.26 | OnlyMe | ;) |
| 13:29.27 | MikeJ[Laptop] | :p |
| 13:29.45 | RoyK | anyone here using SER? |
| 13:29.47 | irv999 | I am really stuck at a roadblock.. and I need to solve this. I dont know if my programmer is capapble |
| 13:30.03 | irv999 | anyone an expert on polycom |
| 13:30.04 | irv999 | ? |
| 13:30.13 | lehel | okay gordonjcp, thanks. |
| 13:30.21 | blitzrage | well, suppose I should go for a run before everyone starts calling me |
| 13:30.30 | Ariel_ | irv999, experts hummm I don't thin experts but I have a few up and running |
| 13:30.41 | blitzrage | back latah |
| 13:31.10 | irv999 | ariel: ok.. we have ip600's.. with 3 buttons programmed |
| 13:31.47 | irv999 | ariel: the secretaries are having issues with the beeps (indicating new calls are coming in) interrupts the calls they are on |
| 13:32.46 | irv999 | ariel: we need a way to have the phone ring to hear when no one is on the phone talking.. and if someone is on the phone talking it does not beep in the handset |
| 13:32.50 | DeeJayTwo | Are zaptel 1.0 drivers supposed to be very stable? |
| 13:32.55 | Ariel_ | irv999, ahh yes this is normal that is why we don't use call waiting on ours just one line setup. |
| 13:33.05 | DeeJayTwo | or should I use a specific recommended CVS version? |
| 13:33.38 | irv999 | ariel: yeah.. we have 3 types of calls here (dr's office) patients, dr's and emergencies.. need 3 buttons programmed |
| 13:33.51 | irv999 | ariel: if it was one button we could queue the calls up.. |
| 13:34.01 | irv999 | ariel: but we dont have that luxury |
| 13:34.31 | BerndR | do someone know an url of some navigation strategies for phone applications? |
| 13:35.05 | MikeJ[Laptop] | RoyK, echocancell=yes :) |
| 13:35.41 | RoyK | ??? |
| 13:36.00 | MikeJ[Laptop] | [09:14] <RoyK> anyone here using SER?, [09:29] <RoyK> anyone here using SER? |
| 13:36.26 | MikeJ[Laptop] | :p |
| 13:36.51 | Ariel_ | irv999, we use FOP and set up call groups via AMP when a call comes in via the line for one co. it appens INTER: /CO2 or other before our caller ID |
| 13:37.16 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 13:37.16 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:37.25 | irv999 | ariel: FOP? |
| 13:37.26 | Ariel_ | irv999, so our recp is able to pickup the line and say this is onestep or this is kasi international. |
| 13:37.31 | RoyK | ~fop |
| 13:37.32 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/ |
| 13:37.32 | bkw_ | holy shit son... RMS emailed me |
| 13:37.41 | Ariel_ | irv999, FLASH OPERATOR PANEL |
| 13:37.58 | RoyK | ~foip |
| 13:38.04 | irv999 | ariel: do you have a good recommendation for a FOP? does yours run on a computer or is it separate? |
| 13:38.06 | irv999 | sidecar |
| 13:38.23 | irv999 | amp? (sorry the abbrev don't work for me) |
| 13:38.31 | Ariel_ | fop is a windows flash web. |
| 13:38.45 | irv999 | ariel: can't do that here.. computers are dos based.. can't have something running for this.. |
| 13:38.47 | RoyK | jbot_: FoIP is Fax over IP. See http://www.voip-info.org/wiki-FoIP |
| 13:38.51 | Ariel_ | and we setup the asterisk box with AMP Asterisk Management Portal |
| 13:39.00 | irv999 | ariel: secretaries do so mcuh can't add that to their list |
| 13:39.22 | irv999 | ariel: good thought though |
| 13:39.33 | Hmmhesays | i tried to get amp running on a p233 |
| 13:39.42 | Hmmhesays | I was drinking at the time |
| 13:40.13 | Katty | Hmmhesays: i want a 2005 mustang gt coupe, in black |
| 13:40.16 | Katty | Hmmhesays: find one for me |
| 13:40.47 | Hmmhesays | Katty, got plenty around these parts, but I don't like the new body style |
| 13:40.57 | Hmmhesays | wheels are to small, makes it look top heavey |
| 13:41.02 | Katty | :< |
| 13:41.09 | Hmmhesays | *heavy |
| 13:41.09 | gordonjcp | Ariel_: I did a funky hack to get the name of the line and the incoming number to come up |
| 13:41.11 | MikeJ[Laptop] | bkw_, RMS? |
| 13:41.13 | Katty | i think it's /cute/ |
| 13:41.21 | irv999 | ariel: any other thoughts |
| 13:41.26 | Hmmhesays | i like the '04 svt |
| 13:41.39 | gordonjcp | Katty: can do you a *really* good deal on a 1968 Mustang, just recently finished being restored |
| 13:41.45 | *** join/#asterisk stefanocarlini (~stefano@213.233.11.14) |
| 13:41.52 | gordonjcp | and in fact it's *not* *quite* finished 'cos it's still in primer |
| 13:41.59 | gordonjcp | so you get to choose the paint |
| 13:41.59 | Katty | ... |
| 13:42.03 | Hmmhesays | Horsepower : 390 hp @ 6000 rpm |
| 13:42.03 | Hmmhesays | Torque : 390 lb-ft @ 3500 rpm |
| 13:42.17 | Hmmhesays | almost 400 ft/lbs of torque at 3500 rpm is nuts |
| 13:42.18 | Katty | gordonjcp: i don't want a 68 mustang. |
| 13:42.23 | Hmmhesays | that's on regular gasoline |
| 13:42.24 | *** join/#asterisk stevek (~stevek@slim-eth0.horizonlive.net) |
| 13:42.47 | Katty | gordonjcp: would you paint it pink? |
| 13:43.37 | Hmmhesays | old body styles with a modern drive train is the way to go |
| 13:43.45 | RoyK | ~foip |
| 13:44.00 | stefanocarlini | Hello, anyone can tell me where I can find information about the worldwide diffusion of *? how many users? how many organizations use *? |
| 13:44.02 | Hmmhesays | computer control is a very very good thing |
| 13:44.14 | Hmmhesays | stefanocarlini google |
| 13:44.22 | gordonjcp | Hmmhesays: my old Citroen CX 25DTR used to put out 330bhp at 2100rpm |
| 13:44.32 | stefanocarlini | I'm googling ;) |
| 13:44.34 | Katty | gordonjcp: would you paint it pink? |
| 13:44.41 | gordonjcp | Katty: I could get someone else to spray it pink, my shed's too small for a spray booth |
| 13:44.56 | Katty | YOU"D RUIN IT?!?! |
| 13:44.57 | Hmmhesays | gordonjcp: what kind of torque did you get out of that |
| 13:45.00 | Katty | !!!!11oneone11!!! |
| 13:45.08 | Katty | :<<<<<<<<<<<<<<<<<<<<<<<<<<< |
| 13:45.17 | RoyK | ~lart Katty |
| 13:45.26 | gordonjcp | Hmmhesays: bah, not 330bhp, 330lb/f, about 130bhp |
| 13:45.29 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 13:45.34 | zoa | yooo |
| 13:45.49 | Hmmhesays | you must have had that geared real funny |
| 13:45.56 | gordonjcp | Hmmhesays: just stock gearing |
| 13:46.03 | zoa | any requests for an asterisk performance / stability / security speech ? |
| 13:46.08 | gordonjcp | about 33mph/1000rpm in 5th IIRC |
| 13:46.16 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
| 13:46.17 | Katty | zoa: no, but vegan food would be nice |
| 13:46.19 | Hmmhesays | maybe i'm thinking of something else, i shall google |
| 13:46.22 | zoa | :) |
| 13:46.30 | RoyK | zoa: yo! |
| 13:46.31 | gordonjcp | Katty: a vegan cat? crazy behaviour |
| 13:46.50 | newl | unnatural..cats are meat eaters by nature ;) |
| 13:46.54 | Katty | gordonjcp: i'm a female in an asterisk channel. |
| 13:47.00 | Katty | gordonjcp: and you're just realizing i'm crazy? |
| 13:47.02 | gordonjcp | newl: indeed |
| 13:47.09 | Hmmhesays | http://www.citroenz.com/CX/hist_cxcentre.html sfw <-- gordonjcp |
| 13:47.14 | Hmmhesays | and she's got all the guys under her thumb |
| 13:47.17 | Hmmhesays | except me of course |
| 13:47.20 | Hmmhesays | ;) |
| 13:47.21 | gordonjcp | Katty: well, a couple of our comms dudes are girls |
| 13:47.41 | Katty | yeah, except Hmmhesays ...he prefers to be under teh boots |
| 13:47.59 | Hmmhesays | that cause you're standing on my shoulders? |
| 13:48.01 | gordonjcp | Hmmhesays: that's the older Series 1, mine was the S2 with plastic bumpers and wider front track |
| 13:48.09 | Katty | Hmmhesays: well how else am i going to reach anything? |
| 13:48.13 | gordonjcp | Hmmhesays: oh, and a bigger turbocharger |
| 13:48.17 | gordonjcp | and a few other things done |
| 13:48.20 | gordonjcp | ;-) |
| 13:48.22 | Hmmhesays | gordonjcp what year? |
| 13:48.25 | gordonjcp | '88 |
| 13:48.31 | Hmmhesays | Katty: climbing gear? |
| 13:48.41 | gordonjcp | same as my current 22TRS which is much saner |
| 13:48.50 | Katty | Hmmhesays: woah, they make climbing gear to reach the top shelf of the cabinents? |
| 13:48.51 | Hmmhesays | speaking of gear, I was out diving yesterday and it really sucked |
| 13:48.56 | gordonjcp | Hmmhesays: it was running an unholy amount of boost |
| 13:49.10 | Hmmhesays | stock? |
| 13:49.19 | gordonjcp | not very, no |
| 13:49.24 | Hmmhesays | Katty, darkwing duck style |
| 13:49.39 | Katty | Hmmhesays: well that's /it/! i have to find me a cape |
| 13:49.41 | Hmmhesays | you don't run into problems with pre-ignition? |
| 13:49.49 | Hmmhesays | I amm the terror the flaps in the night |
| 13:49.50 | gordonjcp | Hmmhesays: it's a diesel |
| 13:50.00 | *** part/#asterisk pat_lehem (~lehem@212.150.210.2) |
| 13:50.03 | Hmmhesays | yes I know |
| 13:50.05 | Katty | pre-ignition problems...that sounds like a personal problem |
| 13:50.06 | gordonjcp | hence the torque low down |
| 13:50.19 | gordonjcp | you don't really get pre-ignition with a diesel |
| 13:50.39 | gordonjcp | on the basis that there is no fuel at all in the cylinder until a couple of degrees before TDC |
| 13:50.42 | Hmmhesays | you can if you run to much boost.... it's not called pre-ignition though |
| 13:50.54 | gordonjcp | Hmmhesays: not pre-ignition as suck |
| 13:50.56 | gordonjcp | such |
| 13:51.04 | Hmmhesays | yes, you know what i am referring to though |
| 13:51.05 | gordonjcp | you *do* get the intercoolers blowing apart |
| 13:51.37 | gordonjcp | well yeah, you can get the charge so hot that there isn't enough lap after injection |
| 13:51.42 | *** join/#asterisk perlmonky (~perlmonky@pix.benchmark-systems.com) |
| 13:51.47 | Hmmhesays | to much air in the cylinder cause the cylinder to fire before it was designed to |
| 13:51.52 | B2382F29 | hi, someone knows something about ISDN and the Wait command? |
| 13:51.59 | gordonjcp | Hmmhesays: but it can't fire until there's fuel... |
| 13:52.14 | Hmmhesays | ahh true |
| 13:52.24 | gordonjcp | and the fuel isn't injected until a couple of degrees BTDC |
| 13:52.27 | Hmmhesays | turbo diesels are definately not my thing anyway |
| 13:52.32 | gordonjcp | Hmmhesays: what you *do* get is this |
| 13:52.33 | Hmmhesays | yeah I missed that |
| 13:52.59 | gordonjcp | there's a fuel control diaphram on top of the injector pump, which increases the fuelling when it comes on boost |
| 13:53.16 | gordonjcp | above around 2.8 bar, the diaphram sticks down |
| 13:53.26 | gordonjcp | effectively giving you a stuck wide open throttle |
| 13:53.42 | gordonjcp | and because the engine is going flat out, the turbo stays up and the boost stays up |
| 13:53.47 | Hmmhesays | ahh so you don't run to lean with the high boost |
| 13:53.48 | gordonjcp | ... holding the fuel on |
| 13:53.59 | gordonjcp | Hmmhesays: diesels don't really have "lean" and "rich" |
| 13:54.06 | Hmmhesays | geebus |
| 13:54.08 | Hmmhesays | they have to |
| 13:54.14 | gordonjcp | there's no throttle as such, you're always throwing as much air down it as you can |
| 13:54.14 | Hmmhesays | to much air, not enough fuel |
| 13:54.35 | *** join/#asterisk Ogun (~johangrip@h236n2fls34o865.telia.com) |
| 13:54.38 | gordonjcp | no, it doesn't need to be a precise ratio like with petrol |
| 13:55.08 | Hmmhesays | damn diesels, messing things all up, heh |
| 13:55.26 | gordonjcp | for low power you only deliver a little bit of fuel, for full power you deliver *just* a bit less than the smoke point |
| 13:55.46 | gordonjcp | if you see black smoke from a diesel, it's 'cos there's just a little too much fuel for the amount of air available |
| 13:55.51 | gordonjcp | probably a blocked air filter |
| 13:56.12 | Katty | you've insaned. |
| 13:56.14 | Hmmhesays | same with most fuels |
| 13:56.34 | Hmmhesays | fossil fuels anyway |
| 13:56.36 | gordonjcp | Hmmhesays: yes, but unlike petrol, it *will* actually run even when severely overfuelled |
| 13:56.43 | *** join/#asterisk claude2005 (~claude@kcassidy.plus.com) |
| 13:56.52 | gordonjcp | that was not a very standard one, though |
| 13:56.54 | gordonjcp | :-) |
| 13:57.06 | Hmmhesays | well... severly over fueled is relative to the type of engine I suppose |
| 13:57.15 | gordonjcp | I once towed a Saab 9000 on a very heavy trailer at around 95mph |
| 13:57.18 | gordonjcp | forgot it was there |
| 13:57.26 | gordonjcp | thought the car seemed a little slower than usual... |
| 13:57.39 | gordonjcp | put it down to crappy fuel |
| 13:58.36 | *** join/#asterisk __a ([X7JVD8epc@212.154.32.104) |
| 13:58.47 | Hmmhesays | I once burned a half gallon of off road diesel in a fire |
| 13:58.59 | Hmmhesays | never do that again |
| 13:59.01 | gordonjcp | hehy |
| 13:59.09 | gordonjcp | I used to just run mine on waste veg oil |
| 13:59.11 | __a | guys, how can i specify the time asterisk waits for a user to input number digits on zap channels? |
| 13:59.25 | gordonjcp | woah, back on topic with a bang! |
| 13:59.36 | __a | when dialing a number i mea |
| 13:59.37 | __a | n |
| 13:59.53 | bkw_ | go read the documentation |
| 13:59.56 | bkw_ | :P |
| 14:00.07 | newl | And look for DigitTimeout |
| 14:00.09 | B2382F29 | exten => s,3,DigitTimeout,5 |
| 14:00.11 | bkw_ | no |
| 14:00.11 | zoa | hey ho bkw |
| 14:00.19 | B2382F29 | ResponseTimeout ? |
| 14:00.24 | bkw_ | that doesn't apply when you're handed a simple switch from going offhook |
| 14:00.33 | __a | cool, just point me what to read |
| 14:01.02 | bkw_ | it applies to menus and IVRs but not like say FXS ZAP |
| 14:01.21 | __a | bkw_, and what should I read for FXS? |
| 14:01.28 | bkw_ | use app_disa? |
| 14:01.45 | B2382F29 | WaitExten |
| 14:01.57 | __a | also, how can I get rid of # in dialed numbers? |
| 14:01.58 | __a | <PROTECTED> |
| 14:02.04 | __a | see the # at the end? |
| 14:02.06 | bkw_ | get a sip phone that isn't on crack |
| 14:02.22 | __a | bkw_ I'm talking about a zap-sip channels |
| 14:02.30 | MikeJ[Laptop] | heh |
| 14:02.31 | bkw_ | why are you pressing #? |
| 14:02.34 | bkw_ | don't press # |
| 14:02.36 | bkw_ | its just that simple |
| 14:02.49 | bkw_ | :P |
| 14:02.58 | __a | it's users and i can't explain it to everyone |
| 14:03.09 | __a | i know how not to press # |
| 14:03.12 | Katty | send out a mass email |
| 14:03.17 | bkw_ | strip it |
| 14:03.21 | Hmmhesays | look at your dialplan variables |
| 14:03.23 | bkw_ | use regexp function to detect it |
| 14:03.24 | gordonjcp | irv999: were you trying to get caller ID to display the incoming line and incoming caller ID? |
| 14:03.32 | bkw_ | and ${EXTEN:1} the bastard |
| 14:03.41 | *** join/#asterisk simonides (simon@byte.unitycode.org) |
| 14:03.42 | __a | it's a callshop, users could be any friggin tourist from peru or united kingdom |
| 14:03.42 | Hmmhesays | i didn't want to come right out and say it |
| 14:04.07 | bkw_ | oh wait |
| 14:04.07 | Katty | :< |
| 14:04.07 | __a | bkw_, cool, thanks for reminding me :1 |
| 14:04.11 | bkw_ | __a, you violated a rule |
| 14:04.21 | __a | what rule? |
| 14:04.22 | bkw_ | you failed to say Hi before busting in and asking questions |
| 14:04.29 | __a | igh |
| 14:04.31 | MikeJ[Laptop] | kick kick kick |
| 14:04.31 | bkw_ | hehe |
| 14:04.34 | *** join/#asterisk Cadu20 (~Cadu83@200-215-114-219.fnsce701.e.brasiltelecom.net.br) |
| 14:04.39 | bkw_ | na |
| 14:04.44 | bkw_ | just letting em know |
| 14:04.45 | MikeJ[Laptop] | hehe |
| 14:04.48 | MikeJ[Laptop] | paypal? |
| 14:04.55 | bkw_ | ya mine? |
| 14:04.56 | __a | i feel like adam prefect now |
| 14:05.02 | __a | Hi! |
| 14:05.05 | bkw_ | hhe |
| 14:05.15 | Hmmhesays | *ahem* hello, thank you, that is all |
| 14:05.19 | bkw_ | I started getting on people for that.. they would join and demand answers |
| 14:05.26 | MikeJ[Laptop] | bkw_ always says: "I'll be right back" ;) |
| 14:05.34 | bkw_ | No I don't |
| 14:05.36 | bkw_ | thats kram |
| 14:05.38 | Ariel_ | bkw_, you have been doing that for over a year now. |
| 14:05.47 | Cadu20 | Hi, the Asterisk Realtime Architeture is working properly? Can I do MySQL SIP peers? |
| 14:05.48 | newl | bkw_: Then leave 10 seconds later because someone doesn't answer their question immediately? |
| 14:05.55 | __a | Cadu20: you can |
| 14:05.56 | bkw_ | newl, yep |
| 14:06.03 | newl | Those annoy me the most. :) |
| 14:06.03 | *** part/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 14:06.07 | bkw_ | hahahha |
| 14:06.25 | Hmmhesays | annoying is asking questions that are easily answered by the wiki |
| 14:06.29 | __a | so there's no simple solution for making * wait for digits on Zap channels?! |
| 14:06.38 | Cadu20 | __a, Thanks! |
| 14:06.57 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 14:07.05 | newl | Impatient bastards. If they've got the time to use IRC, make the effort of coming on a channel, they can afford to idle ~5-10 minutes while doing something else. Surely most would offer an answer or a suggestion.</mini rant> |
| 14:07.14 | Cadu20 | Another question, can Asterisk support a large number of peers? Like 200.000... |
| 14:07.23 | bkw_ | Cadu20, NO |
| 14:07.25 | __a | Cadu20: err, no. |
| 14:07.30 | bkw_ | not just no.. BUT WTF? |
| 14:07.30 | perlmonky | has anyone encountered an issue with polycom IP500 phones with no sound when videosupport is enabled? |
| 14:07.34 | Cadu20 | Whats the limit? |
| 14:07.39 | newbieidiot | somone tell me how to set up asterisk for my business.. I'm in a rush |
| 14:07.46 | bkw_ | you don't use asterisk if you need that many |
| 14:07.53 | bkw_ | or you split it up in to a cluster |
| 14:07.55 | newl | newbieidiot: Throw lots of cash our way! B) |
| 14:07.57 | newbieidiot | cmon... |
| 14:08.01 | Hmmhesays | heh just be glad there's no java client on digium's website |
| 14:08.02 | newbieidiot | damn |
| 14:08.04 | *** part/#asterisk newbieidiot (~ircatjerr@mi.origenfinancial.com) |
| 14:08.08 | perlmonky | newbieidiot: asterisk@home |
| 14:08.10 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 14:08.11 | Ariel_ | newbieidiot, pay me money and I will set it up for you. |
| 14:08.11 | MikeJ[Laptop] | hehe |
| 14:08.12 | bkw_ | repeat after me.. ASTERISK DOES NOT SCALE |
| 14:08.13 | perlmonky | start there... |
| 14:08.14 | newl | :) |
| 14:08.21 | __a | ASTERISK DOES NOT SCALE |
| 14:08.28 | Cadu20 | ASTERISK DOES NOT SCALE |
| 14:08.30 | MikeJ[Laptop] | the spoon does not exist |
| 14:08.34 | Cadu20 | Hmm... got it. |
| 14:08.37 | gordonjcp | LINUS DOES NOT SCALE |
| 14:08.41 | bkw_ | it doe NOT cluster either. N+1 IS NOT clustering. |
| 14:08.41 | Ariel_ | Asterisk does in lots of ways scale. but there is a limit. |
| 14:08.45 | Zeeek | astrisk does not skale |
| 14:08.51 | newl | These are not the dtmf tones you are looking for. |
| 14:08.53 | Hmmhesays | where's ol' whatsisname |
| 14:09.00 | bkw_ | hahaha |
| 14:09.03 | Hmmhesays | started with an s |
| 14:09.10 | bkw_ | no that said.. we are working on those issues. |
| 14:09.37 | Cadu20 | Even if there is just 100 calling... ? |
| 14:09.51 | __a | Ariel_: you mean they're rewriting it from scratch? |
| 14:09.56 | MikeJ[Laptop] | heh |
| 14:10.02 | Ariel_ | __a, not me |
| 14:10.09 | bkw_ | haha |
| 14:10.16 | bkw_ | it has a few design flaws |
| 14:10.22 | bkw_ | asterisk for the most part is good stuff |
| 14:10.26 | bkw_ | but it has a few fatal flaws |
| 14:10.29 | Cadu20 | Does SER support these 200.000 peers? |
| 14:10.32 | Hmmhesays | lies |
| 14:10.34 | bkw_ | Cadu20, yes |
| 14:10.38 | Hmmhesays | my world is crushed |
| 14:10.43 | bkw_ | haha |
| 14:10.49 | bkw_ | the fun part is we are working on them :P |
| 14:11.03 | Cadu20 | Hmm... And the answers are coming from the mountain to clarify my mind! |
| 14:11.07 | Ariel_ | but you can set up some asterisk boxes with hartbeat and qfs mounts and have some redundancy now. but limited |
| 14:11.09 | MikeJ[Laptop] | Cadu20, if that is a . after the 200 and not a , |
| 14:11.11 | newl | On the flaws or the good stuff? :D |
| 14:11.17 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 14:11.30 | bkw_ | the flaws :P |
| 14:11.33 | Cadu20 | 200 hunred thousand... dammed pointing differences between countries. |
| 14:11.38 | bkw_ | converting the flaws to good stuff |
| 14:11.42 | bkw_ | :P |
| 14:11.44 | MikeJ[Laptop] | Ariel_, or use dundi and stuff |
| 14:11.47 | Cadu20 | or 2 thousand hundred... i dont know.. |
| 14:11.53 | bkw_ | Cadu20, use SER |
| 14:11.53 | Cadu20 | :P bad english |
| 14:11.55 | MikeJ[Laptop] | but PRI failover is still a pain |
| 14:12.02 | Ariel_ | MikeJ[Laptop], yes dundi is a part of it. |
| 14:12.10 | bkw_ | mix some asterisk in there for voicemail |
| 14:12.15 | bkw_ | and IVR goodness |
| 14:12.21 | bkw_ | but it by no means could ever do 200k peers |
| 14:12.25 | bkw_ | without first blowing up |
| 14:12.30 | Katty | bkw_: way to not even say hi |
| 14:12.40 | newl | Oh no!</lemmings> |
| 14:12.41 | Katty | bkw_: like some sort of antisocial GEEK |
| 14:12.45 | bkw_ | Katty, hahahahahaha |
| 14:12.46 | MikeJ[Laptop] | bkw_, yeah.. you jerk |
| 14:12.47 | Cadu20 | Thanks bkw_... i´ll try that. |
| 14:12.59 | Katty | thx, all better |
| 14:13.05 | MikeJ[Laptop] | :) |
| 14:13.08 | Hmmhesays | I wouldn't want to enter even 200 peers in sip.conf |
| 14:13.12 | newl | s/HI/NEXT!/ |
| 14:13.30 | bkw_ | do you need some help with your large system? |
| 14:13.32 | bkw_ | ;) |
| 14:14.30 | Katty | bkw_: you can setup my large system of 13 phones. |
| 14:14.36 | Katty | bkw_: help yourself. |
| 14:14.39 | bkw_ | 13 phones.. mmmmmmmm baby |
| 14:14.51 | Katty | i know |
| 14:15.07 | bkw_ | I have two more boxen to update today |
| 14:15.07 | bkw_ | damn it |
| 14:15.10 | bkw_ | haha |
| 14:15.32 | perlmonky | what is reasonable to expect from asterisk (single system) 100+ 200+ |
| 14:15.44 | perlmonky | say dual xeons... gig of ram |
| 14:15.47 | Ariel_ | perlmonky, that is a loaded question. |
| 14:15.58 | perlmonky | no transcodeing |
| 14:15.59 | Hmmhesays | anyone using cron to schedule callbacks in asterisk? |
| 14:16.00 | Ariel_ | it really depends on codec used and many other thngs. |
| 14:16.10 | perlmonky | all sip |
| 14:16.26 | bkw_ | you can do 600-700 calls on a dual xeon with ZERO transcoding |
| 14:16.40 | bkw_ | you can do 5500+ on that same box if you DO NOT carry the media |
| 14:16.41 | jeremywhiting | but sip isn't a codec, it's a protocol |
| 14:16.46 | zoa | brian, we stresstested rtpproxy |
| 14:16.57 | bkw_ | zoa and? |
| 14:16.57 | perlmonky | right.. i understand that |
| 14:16.58 | Ariel_ | perlmonky, no transcoding we have some boxes doing about 750 to 1000 calls. |
| 14:16.59 | zoa | you can get the same, not more on rtpproxy |
| 14:17.04 | Essobi | bkw_ Dual xeon what? and with how much ram? |
| 14:17.09 | perlmonky | that is why i mensioned no transcoding... |
| 14:17.15 | perlmonky | gig |
| 14:17.18 | bkw_ | I did my test on a 3ghz HT with 512 |
| 14:17.28 | bkw_ | so i'm sure it would give you more breathing room |
| 14:17.29 | Essobi | Nice. |
| 14:17.30 | perlmonky | ok that works for me... |
| 14:17.37 | bkw_ | zoa kewl |
| 14:17.48 | perlmonky | only planning on 75-100 phones and two t1's |
| 14:17.49 | Hmmhesays | what was your testing procedure? |
| 14:18.40 | Essobi | Shew, that's a lot of legs. |
| 14:18.46 | Essobi | bkw_ IAX2 or SIP? |
| 14:19.14 | bkw_ | run far and fast from IAX2 |
| 14:19.18 | bkw_ | its a great protocol |
| 14:19.20 | Essobi | lol |
| 14:19.21 | Essobi | ;) |
| 14:19.23 | bkw_ | but when you have alot of calls |
| 14:19.25 | bkw_ | it sounds like ass |
| 14:19.26 | Essobi | I don't use it. |
| 14:19.26 | bkw_ | total ass |
| 14:19.36 | bkw_ | its one of the things mark is working on |
| 14:19.44 | Essobi | I'm slowly getting away from H323 now. |
| 14:19.56 | Hmmhesays | i still use h.323 sometimes |
| 14:19.59 | Essobi | I want to give that new driver a spin. |
| 14:20.04 | Hmmhesays | on occasion |
| 14:20.09 | Essobi | I still have quite a few endpoints on it. |
| 14:20.30 | Hmmhesays | I won't bother, it's no pain having a few h.323 endpoints out there |
| 14:20.42 | Essobi | I have more then a "few" |
| 14:20.52 | cpm | sounds like ass? |
| 14:21.06 | Essobi | *COUGH*JITTER*COUGH* |
| 14:21.12 | gordonjcp | Hmmhesays: I was kind of thinking of playing with H.323 but then I attempted it |
| 14:21.41 | Hmmhesays | if you dont' have a decent reason to use it then don't bother |
| 14:22.07 | Hmmhesays | but if you've got it working already it's really of no consequence to leave it |
| 14:22.11 | Ariel_ | Hmmhesays, I 2nd that. |
| 14:22.35 | *** part/#asterisk The_Duke (~the_duke@80.92.64.103) |
| 14:22.45 | Hmmhesays | cheers |
| 14:23.17 | Hmmhesays | that said... debian has a nice package of chan_oh323 if you are just curious |
| 14:23.39 | gordonjcp | Hmmhesays: well, my Avaya 4602 came with H.323 firmware, but I reflashed it to use SIP |
| 14:23.52 | gordonjcp | although - when I use SIP I can't use the "message waiting" light |
| 14:24.11 | *** join/#asterisk santiago (~santiago@63.245.86.198) |
| 14:24.16 | Hmmhesays | is that a known issue? or just a config problem |
| 14:24.36 | gordonjcp | no idea |
| 14:24.43 | gordonjcp | googling suggests that it's a known problem |
| 14:25.00 | *** join/#asterisk dsfr (~dsfr@207.111.174.1) |
| 14:25.07 | gordonjcp | what happens is that if I specify a "voicemail=<this>" number in sip.conf, it registers for NOTIFY |
| 14:25.10 | gordonjcp | and indeed the light works |
| 14:25.32 | gordonjcp | but after a few minute it returns an error, possibly when the phone re-REGISTERs |
| 14:25.44 | gordonjcp | and then it doesn't work again until you reset |
| 14:25.57 | OnlyMe | anyone know if netweb X401 is FCC aproved |
| 14:26.09 | gordonjcp | taking "voicemail=6001" out of sip.conf stops the problem, but also means the light doesn't work |
| 14:27.10 | Hmmhesays | mailbox= |
| 14:27.25 | bkw_ | FYI all signate has no clue what the hell they are talking about |
| 14:27.41 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 14:29.01 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 14:29.21 | gordonjcp | Hmmhesays: yes, mailbox= |
| 14:29.50 | Hmmhesays | number@context |
| 14:29.59 | gordonjcp | Hmmhesays: I will reproduce the problem and let you see the exact error, if you like |
| 14:30.03 | Katty | hmm. |
| 14:30.04 | gordonjcp | but not until I get home |
| 14:30.19 | Hmmhesays | if you like, no idea if I'll be around or not |
| 14:30.39 | gordonjcp | it may be that the 4602 is a bit funny about things like that |
| 14:30.40 | Hmmhesays | or if I'll even care at that point |
| 14:30.44 | gordonjcp | heh |
| 14:30.47 | Hmmhesays | j/k ;) |
| 14:30.48 | Katty | in my case it's user error. |
| 14:30.49 | gordonjcp | well, maybe tomorrow or something |
| 14:30.56 | gordonjcp | no rush |
| 14:31.29 | Hmmhesays | ugh rain, go away |
| 14:31.40 | Katty | Hmmhesays: i rather like it |
| 14:31.53 | Hmmhesays | Katty: it's been raining all month |
| 14:31.58 | Hmmhesays | gordonjcp, that's fine |
| 14:31.59 | Katty | so? |
| 14:32.11 | docelm0 | bkw, what do you mean signate has no clue? |
| 14:32.19 | Hmmhesays | so I'm ready for some sunshine |
| 14:32.23 | blitzrage | bkw_: I got that impression from their book |
| 14:32.28 | Hmmhesays | I gotta get this farmers tan going on |
| 14:32.42 | gordonjcp | Hmmhesays: hehe, I used to have one of those |
| 14:32.50 | gordonjcp | now I'm just a pale geek working in a helpdesk |
| 14:32.52 | bkw_ | blitzrage, http://lists.digium.com/pipermail/asterisk-users/2005-January/086734.html |
| 14:32.54 | Katty | Hmmhesays: no you don't. pale is pretty |
| 14:32.55 | blitzrage | Hmmhesays: me too - cleaned the office, now heading out for a run to try and get some sun - should be a bit stronger now that its 10:30 |
| 14:32.59 | bkw_ | blitzrage, http://lists.digium.com/pipermail/asterisk-users/2005-May/109077.html |
| 14:33.05 | bkw_ | blitzrage, http://lists.digium.com/pipermail/asterisk-users/2005-April/099317.html |
| 14:33.41 | Katty | someone thought i was some chick off a calendar last night |
| 14:33.41 | Hmmhesays | blitzrage: if i tried to run I would cough up a lung and possibly die |
| 14:33.49 | gordonjcp | Katty: heh |
| 14:33.53 | gordonjcp | Katty: and are you? |
| 14:33.54 | Katty | april, i think they said..on a "Women of Asterisk" calendar |
| 14:34.05 | Katty | gordonjcp: you tell me |
| 14:34.08 | zoa | wait i have another quote here |
| 14:34.09 | Hmmhesays | Katty: pale from the middle bicep up? |
| 14:34.25 | bkw_ | zoa do share |
| 14:34.29 | Katty | Hmmhesays: sure. |
| 14:34.31 | bkw_ | These guys are in business.. man |
| 14:34.32 | bkw_ | how |
| 14:34.33 | Katty | Hmmhesays: wait, me or you? |
| 14:34.33 | bkw_ | they LIE |
| 14:34.36 | bkw_ | LIE LIE LIE |
| 14:34.45 | Hmmhesays | me... hence the farmers tan |
| 14:34.57 | docelm0 | um, ok? |
| 14:34.58 | blitzrage | bkw_: LOL!!! yah, because CPU is useless when you're transcoding, and the 64KB/s doesn't include IP overheard |
| 14:35.01 | blitzrage | overhead* |
| 14:35.05 | Katty | i guess you could just sprawl out naked in the back yard |
| 14:35.12 | blitzrage | Katty: sure, lets go :) |
| 14:35.14 | bkw_ | KB is not a measurement of throughput |
| 14:35.20 | Katty | blitzrage: uhh, no |
| 14:35.35 | gordonjcp | blitzrage: well, he does say "The simple answer is..." |
| 14:35.40 | blitzrage | Katty: lol - just trying to replace tzanger while he's away :) |
| 14:35.41 | Hmmhesays | Katty: I'll save the neighbors from bleeding eyes and refrain from that |
| 14:35.45 | Katty | bkw_: is there a Women Of Astierks calendar? |
| 14:35.46 | *** join/#asterisk file[class] (~root@66.199.241.90) |
| 14:35.53 | file[class] | ah whoops |
| 14:35.55 | blitzrage | haha.. there should be :) |
| 14:35.57 | blitzrage | LOL |
| 14:36.12 | *** join/#asterisk _omer (dfsdf@202.147.167.213) |
| 14:36.17 | *** join/#asterisk file[class] (~jcolp@66.199.241.90) |
| 14:36.24 | file[class] | better. |
| 14:36.25 | blitzrage | file, that was classic |
| 14:36.29 | Hmmhesays | I dunno, like guys in IT, a lot of women not so much attractive |
| 14:36.31 | Katty | blitzrage: k |
| 14:36.33 | Katty | Hmmhesays: :< |
| 14:36.53 | blitzrage | anyways, iPod should be charged enough now, heading for that run :D |
| 14:36.54 | file[class] | blitzrage: indeed |
| 14:36.54 | Katty | Hmmhesays: pfft, i'm not going to smack you. that's too easy |
| 14:36.58 | file[class] | nooooooo |
| 14:36.59 | file[class] | dont' go |
| 14:37.01 | bkw_ | ok |
| 14:37.04 | bkw_ | leave Katty alone |
| 14:37.08 | bkw_ | don't run her off boys |
| 14:37.10 | blitzrage | file[class]: I must, I must, I must increase my bust - Lord of Acid |
| 14:37.21 | Hmmhesays | heh, I'll keep my comments to myself about that |
| 14:37.22 | bkw_ | hhahaha |
| 14:37.31 | zoa | in their book they write |
| 14:37.33 | file[class] | blitzrage: gah |
| 14:37.38 | Katty | Hmmhesays: i'll con you into setting up another asterisk box for me if you're not nice :P |
| 14:37.39 | zoa | Asterisk supports iax2, sip and BGCP |
| 14:37.45 | file[class] | zoa: Joachim! |
| 14:37.45 | blitzrage | LOL |
| 14:37.49 | zoa | yes ? |
| 14:37.53 | file[class] | hi. |
| 14:37.56 | Hmmhesays | You got more help than you need in this chan now ;) |
| 14:38.00 | bkw_ | what is odd is I like that song "Pussy - Lords of Acid" |
| 14:38.00 | blitzrage | zoa: yah, I bet you find the same typo on the wiki ;) |
| 14:38.02 | bkw_ | figure that one out |
| 14:38.15 | *** join/#asterisk coppice (~chatzilla@14.198.17.210.dyn.pacific.net.hk) |
| 14:38.21 | blitzrage | bkw_: you're a closet heterosexual! |
| 14:38.22 | zoa | lords of acid -> belgian group |
| 14:38.25 | bkw_ | hahha |
| 14:38.27 | Katty | Hmmhesays: so? that doesn't mean i don't have my favorites |
| 14:38.27 | bkw_ | nope |
| 14:38.30 | __a | wtf is bgcp? |
| 14:38.33 | file[class] | bkw, heterosexual, WHAT |
| 14:38.37 | bkw_ | NEVER |
| 14:38.37 | docelm0 | MGCP |
| 14:38.39 | blitzrage | should be MGCP |
| 14:38.42 | file[class] | that's crazy talk |
| 14:38.44 | zoa | yeah |
| 14:38.44 | docelm0 | another version of it. |
| 14:38.45 | docelm0 | :) |
| 14:38.48 | blitzrage | haha |
| 14:38.48 | file[class] | MGCP is a nice protocol btw |
| 14:38.50 | newl | It's BGP on crack. |
| 14:38.51 | RoyK | ~mgcp |
| 14:38.52 | jbot | i heard mgcp is Media Gateway Control Protocol |
| 14:38.52 | Hmmhesays | ;) |
| 14:38.52 | blitzrage | aye |
| 14:38.53 | coppice | he likes women in wardrobes? :-\ |
| 14:38.57 | zoa | just a secm i will find you the exact quote |
| 14:38.59 | blitzrage | newl: LOL, I was thinking that too :) |
| 14:39.07 | newl | 8) |
| 14:39.09 | bkw_ | Tomb Raider Chick |
| 14:39.11 | bkw_ | <PROTECTED> |
| 14:39.14 | bkw_ | she's HAWT |
| 14:39.14 | coppice | MGCP seems pointless |
| 14:39.15 | zoa | Asterisk supports every possible kind of telephone technology. The technologies include VoIP, SIP, H.323, IAX, and BGCP (for gateways and phone) |
| 14:39.17 | zoa | - |
| 14:39.18 | blitzrage | ok, I'm gone, back in a couple hours |
| 14:39.19 | zoa | Exact quote |
| 14:39.29 | bkw_ | Angelina Jolie |
| 14:39.34 | Hmmhesays | video game tomb raider chick or movie |
| 14:39.35 | newl | New Tomb Raider..will have to buy that. |
| 14:39.36 | blitzrage | zoa: thats fuckin' hilarious |
| 14:39.41 | zoa | its a good thing that besides SIP and h323 it also supports VOIP |
| 14:39.46 | docelm0 | bk, and then some |
| 14:39.48 | blitzrage | yah, good thing |
| 14:39.53 | coppice | pity * doesn't do MGCP gateway. I need to implement it :-\ |
| 14:39.53 | RoyK | anyone here that could help me with setting up SER in front of a few asterisk servers? |
| 14:39.54 | __a | zoa: where do they write it? |
| 14:40.00 | zoa | in chapter 2 |
| 14:40.05 | bkw_ | haha |
| 14:40.14 | file[class] | RoyK: for free? no |
| 14:40.15 | newl | That'll give me something to do when I eventually finish GT4. hehe |
| 14:40.18 | bkw_ | signate should be dragged out and beat with a cluebat |
| 14:40.18 | Hmmhesays | um... what is bgcp? |
| 14:40.25 | file[class] | bkw_: calllllll |
| 14:40.32 | docelm0 | newl.. You working on it also? |
| 14:40.34 | bkw_ | file dear I can't its too early |
| 14:40.40 | docelm0 | GT4 that is |
| 14:40.41 | file[class] | lies, all lies |
| 14:40.51 | bkw_ | sure calling at 7:40 AM works out great |
| 14:40.56 | bkw_ | ya ya lets do that |
| 14:40.57 | file[class] | yes it does! |
| 14:40.58 | bkw_ | NOT!!! |
| 14:41.14 | file[class] | I really need to get my plane ticket soooooon |
| 14:41.23 | bkw_ | file yes you do |
| 14:41.24 | newl | docelm0: yeah, I'm only a couple hundred days in. |
| 14:41.27 | file[class] | seats are dwindling, prices are going up |
| 14:41.37 | docelm0 | I play when I have time.. |
| 14:41.40 | bkw_ | talk to anthm when he gets here |
| 14:41.42 | file[class] | I may be able to buy it soon... and get whoever to reimburse me |
| 14:41.53 | file[class] | well I did |
| 14:42.02 | bkw_ | twat did he say? |
| 14:42.03 | file[class] | he said talk to David, so guess what I did - I talked to David |
| 14:42.04 | newl | docelm0: same here. I've got mates that've already completed everything. Though, they're single.. :) |
| 14:42.12 | file[class] | and then David said sure I can probably do that |
| 14:42.18 | file[class] | but it's David, so it hasn't gotten done |
| 14:42.22 | docelm0 | Very true.. |
| 14:42.40 | bkw_ | file[class], you must saddle up and ride his ass |
| 14:42.57 | bkw_ | bug him every day |
| 14:42.58 | bkw_ | haha |
| 14:43.00 | file[class] | but I don't wanna |
| 14:43.00 | Katty | file[class]: yes, what bkw_, and i want a copy of the dvd |
| 14:43.14 | bkw_ | haha |
| 14:43.15 | Katty | s/what bkw_/what bkw_ said/ |
| 14:43.25 | bkw_ | Katty, I read it like that |
| 14:43.51 | Katty | bkw_: like what? |
| 14:43.55 | bkw_ | god I till hurt from falling off the porch |
| 14:44.05 | Katty | then don't fall off the porch you dingbat |
| 14:44.12 | bkw_ | Katty, I read it with the word said in it.. without it actually being there |
| 14:44.21 | Katty | k'then |
| 14:44.27 | bkw_ | brainwaves grrl |
| 14:44.39 | bkw_ | or in this case gaymarays |
| 14:44.40 | coppice | how can you fall off a porch? aren't they usually at ground level? |
| 14:44.45 | bkw_ | yes |
| 14:44.46 | Katty | mmm, gaymarays |
| 14:44.50 | bkw_ | my foot went sideways |
| 14:44.54 | bkw_ | I spun around |
| 14:44.57 | bkw_ | and smacked into the pavement |
| 14:45.01 | bkw_ | wasn't a pretty site |
| 14:45.04 | Katty | ouch :< |
| 14:45.04 | bkw_ | er sight |
| 14:45.07 | Ariel_ | LOL |
| 14:45.11 | jsharp | Gravity wins again! |
| 14:45.14 | coppice | and didn't get video? darn! |
| 14:45.16 | Ariel_ | bkw_, thanks for the picture |
| 14:45.19 | bkw_ | whiped my ass fo sure |
| 14:45.25 | Katty | kinky |
| 14:45.29 | bkw_ | I was going up the steps |
| 14:45.41 | bkw_ | when I hit the ground.. I did a complete 180 before hitting it.. so my face hit first |
| 14:45.45 | bkw_ | had a mouth full of dirt |
| 14:45.46 | Katty | you fell up the steps...and off the porch? |
| 14:45.50 | bkw_ | BOY that was fucking fun as hell |
| 14:46.04 | bkw_ | no it spun my ass around and smacked me into the pavement |
| 14:46.11 | Katty | i see |
| 14:46.21 | bkw_ | its funnynow that I think abou tit |
| 14:46.26 | bkw_ | but I was in mucho pain when it happened |
| 14:46.34 | _omer | http://pastebin.ca/13181 <---- anybody please have a look at my problem... |
| 14:46.35 | bkw_ | I pulled stuff i didn't realize existed |
| 14:46.37 | Katty | did you hurt teh noggin? |
| 14:46.47 | bkw_ | na |
| 14:46.53 | Katty | :> |
| 14:46.54 | bkw_ | my hat protected me |
| 14:47.00 | gordonjcp | heh |
| 14:47.08 | gordonjcp | there's a great video of me tripping over a fence |
| 14:47.23 | bkw_ | they have this great video of me drunk at astricon |
| 14:47.25 | bkw_ | its out there |
| 14:47.27 | bkw_ | quite funny stuff |
| 14:47.47 | Katty | oh, i should go pester about cluecon |
| 14:48.12 | bkw_ | hehe |
| 14:48.32 | Katty | or not, seeing that the boss's door is shut |
| 14:48.55 | *** join/#asterisk mkrufky (~mike@68.160.103.76) |
| 14:49.44 | *** part/#asterisk B2382F29 (~tripled@dsl-084-058-138-126.arcor-ip.net) |
| 14:49.46 | _omer | http://pastebin.ca/13181 <---- |
| 14:49.46 | file[class] | bkw_: do I HAVE to bug David? |
| 14:51.02 | Katty | there should be an asterisk calendar |
| 14:51.28 | Katty | do we have any photographers? |
| 14:51.57 | gordonjcp | my gf |
| 14:52.24 | Katty | there has to be at least one photographer of 254 people in here |
| 14:52.58 | Goshen | my wife is a professional photographer |
| 14:53.18 | Katty | we she take pictures for an asterisk calendar? |
| 14:53.20 | Ariel_ | _omer, did you upgrade or downgrade your asterisk from a different version? |
| 14:53.23 | Katty | s/we/would/ |
| 14:53.26 | Goshen | you want to make a "Girls of #Asterisk" calendar? |
| 14:53.39 | Katty | not #asterisk |
| 14:53.41 | Katty | just asterisk in general |
| 14:53.49 | _omer | yesterday I recompiled and downloaded it.. |
| 14:53.56 | Goshen | Sure, but we are in Salt Lake City, Utah |
| 14:53.57 | Ariel_ | head or stable |
| 14:54.02 | Katty | it might help sponser a few things |
| 14:54.18 | Ariel_ | Katty, why would people get a calendar of geeks |
| 14:54.40 | Katty | it was an idea, i didn't say it was a good one |
| 14:54.43 | _omer | Ariel_: not sure about head....but I did export CVS: ...bla bla |
| 14:54.47 | *** join/#asterisk dos000 (~dos000@66.11.173.123) |
| 14:54.54 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:54.54 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:55.02 | Goshen | Geek girls perhaps :) I have seen some nice BSD, Linux girls :) |
| 14:55.16 | Katty | bkw_: are there enough girls to make a female asterisk calendar? |
| 14:55.22 | gordonjcp | my gf is a nice Linux geek girl ;-) |
| 14:55.35 | file[class] | anthm: can I go and throw stuff at David? |
| 14:55.42 | [TK]D-Fender | <Goshen> you want to make a "Girls of #Asterisk" calendar? <- There's more than one?! ;0 |
| 14:55.50 | Ariel_ | _omer, I would remove all the old modules and download the asterisk again. then do the make clean ,make and make install again. |
| 14:55.51 | zoa | we have bkw, he can also be on the girl calendar :) |
| 14:55.56 | Nugget | my girlfriend is a linux girl, but I'm trying to fix that. |
| 14:56.09 | gordonjcp | Nugget: Solaris? |
| 14:56.17 | Katty | 3 of tweleve...hrmm. |
| 14:56.19 | file[class] | anthm: yay!!! |
| 14:56.33 | Nugget | I got her to move her server to freebsd, so I'm making progress, but she still runs some bletcherous gentoo box for her desktop |
| 14:56.45 | Nugget | I'm pushing for osx :) |
| 14:56.49 | Goshen | This guy could be on the cover - http://www.angelfire.com/ill/thursdayclub/asterisk_flyer.JPG |
| 14:57.15 | Katty | hmm. |
| 14:57.17 | Nugget | getting her hooked on world of warcraft is phase 1 |
| 14:57.24 | _omer | Ariel_: yesterday I did all this....would you like to give it a test? |
| 14:59.10 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc0gh.dialup.mindspring.com) |
| 15:00.41 | *** join/#asterisk sretooh (sretooh@63.252.229.9) |
| 15:02.04 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 15:03.43 | bkw_ | Katty, what? |
| 15:03.47 | bkw_ | Grrls of asterisk? |
| 15:03.49 | bkw_ | haha |
| 15:04.00 | mutilator | :O |
| 15:04.29 | mutilator | girlsofasterisk.com? |
| 15:04.29 | bkw_ | BAH the canadians do it again |
| 15:04.34 | bkw_ | I say "Bacon, Egg, McMuffin" |
| 15:04.42 | bkw_ | since when does "Bacon" mean canadian bacon? |
| 15:04.50 | mutilator | heh |
| 15:05.17 | mutilator | i perfer the bacon egg bagel |
| 15:05.19 | bkw_ | this is why I always get "Sausage Biscuits" |
| 15:05.23 | mutilator | the mcmuffin crap dun do it for me |
| 15:05.34 | Katty | bkw_: exploited moocow muffin |
| 15:05.40 | gordonjcp | I prefer food |
| 15:05.42 | bkw_ | is this real meat? |
| 15:05.46 | bkw_ | and real biscuits? |
| 15:05.57 | Katty | bkw_: there should be a grrrrls of asterisk calender |
| 15:05.59 | bkw_ | thought it was simulated cardboard |
| 15:06.08 | bkw_ | Katty, ok |
| 15:06.19 | Katty | yes, hot sexy particle board on cardboard action |
| 15:06.31 | file[class] | bkw_: OMG BECKY |
| 15:06.36 | Katty | baked to stale perfection |
| 15:06.41 | mutilator | .. |
| 15:06.45 | mutilator | the bacon is real bacon.. |
| 15:06.51 | mutilator | you can't fake bacon |
| 15:06.56 | Katty | i can |
| 15:07.00 | Katty | using the power of SOY |
| 15:07.04 | cpm | what about sizzlean? |
| 15:07.08 | Katty | and a little cayenne pepper |
| 15:07.12 | mutilator | pff |
| 15:07.15 | Katty | and tamari sauce |
| 15:07.24 | mutilator | thats not bacon |
| 15:07.25 | gordonjcp | Katty: I suspect that would taste like slightly spicy soy |
| 15:07.28 | cpm | heh |
| 15:07.41 | Katty | i prefer the bacon to oink |
| 15:07.46 | Katty | in a non harmed fashion |
| 15:07.55 | bkw_ | haha |
| 15:07.57 | Nugget | http://cockeyed.com/pranks/menu/menu01.html <-- bacon |
| 15:08.19 | *** join/#asterisk squirrelv5 (~squirrelv@202.57.81.146) |
| 15:08.25 | squirrelv5 | hello everyone |
| 15:08.27 | gordonjcp | hello |
| 15:08.28 | Katty | gordonjcp: well there are other things, like molasses and bbq sauce that's put in there too |
| 15:08.42 | gordonjcp | Katty: hmm |
| 15:08.51 | gordonjcp | bet it tastes good, but not much like bacon |
| 15:09.00 | Katty | i wouldn't remember |
| 15:09.06 | Katty | haven't had bacon in over a year |
| 15:09.13 | squirrelv5 | ive got a cisco 2621Xm router connected via leased line, is it possible to integrate it with asterisk? |
| 15:09.25 | mutilator | o man |
| 15:09.27 | Katty | i don't like eating piglets. |
| 15:09.32 | bkw_ | I don't think a single person including me is qualified to speak about res_perl.. anthm is the only person I would let talk about it. |
| 15:09.38 | file[class] | how many times a day do we get, "can I do this... can I do that... yada yada yada" |
| 15:09.40 | mutilator | i love getting that peppered bacon thats black cause theres so much pepper in it |
| 15:09.44 | mutilator | thats some good stuff |
| 15:09.46 | file[class] | yes, anthm is res_perl's daddy |
| 15:09.48 | gordonjcp | Katty: you're probably not eating piglets, they'd need to be at least 2 years old |
| 15:09.57 | gordonjcp | possibly 3 |
| 15:10.05 | Katty | gordonjcp: that's beside the point. i don't want a 2/3 year old piglet either |
| 15:10.10 | gordonjcp | otherwise they'd be all fatty and bony |
| 15:10.12 | squirrelv5 | oh sorry is this the correct channel to ask questions? |
| 15:10.27 | file[class] | squirrelv5: yes but I've just heard the same things over and over that it's getting to me |
| 15:10.48 | file[class] | and I'm gone |
| 15:10.48 | squirrelv5 | any references/resource to get into that? |
| 15:10.51 | mutilator | file's havin a breakdown |
| 15:10.57 | file[class] | squirrelv5: Google, http://www.voip-info.org/ |
| 15:11.00 | Katty | file[class]: go have a massage or something |
| 15:11.00 | file[class] | ~useful asterisk docs |
| 15:11.01 | jbot | hmm... useful asterisk docs is it has been said that useful asterisk docs is (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unnoficial Links") and http://www.voip-info.org/wiki-Asterisk (the Wiki), and http://www.fnords.org/~eric/asterisk (ManxPower's site), and http://asteriskdocs.org, also, read all files in /usr/src/asterisk/doc |
| 15:11.37 | mutilator | i need to find a job, anyone hiring? |
| 15:11.45 | mutilator | this job is teh sux |
| 15:12.37 | Katty | you mean teh sukc |
| 15:12.47 | Katty | i could use an assistant |
| 15:12.57 | mutilator | yep that too |
| 15:12.59 | Katty | mutilator: want to handle all the windows boxes? |
| 15:13.03 | otmar | mutilator: you're locate where? |
| 15:13.12 | mutilator | Katty, sure |
| 15:13.21 | Katty | you're crazy. |
| 15:13.24 | mutilator | otmar: michigan, i'll move anywhere tho |
| 15:13.24 | Katty | we don't hire crazy people |
| 15:13.28 | mutilator | ;P |
| 15:13.45 | Katty | gordonjcp: yeah, with a big sledge hammer |
| 15:13.46 | gordonjcp | stick a *real* OS on them, job done |
| 15:14.01 | otmar | I know a company who is looking for asterisk-capable techies. |
| 15:14.06 | gordonjcp | otmar: cool |
| 15:14.09 | _omer | which is the folder to update HEAD? |
| 15:14.13 | Katty | gordonjcp: yeah, but then our clients would be all omgwtfisthislolzkthxbi |
| 15:14.14 | mutilator | where? |
| 15:14.35 | otmar | The catch: Innsbruck, Austria. German language skills required. |
| 15:14.53 | mutilator | o, yea, my german is about as good as my alienese |
| 15:14.53 | gordonjcp | Katty: tough, they'd have the same problem if you "upgraded" from 2000 to XP |
| 15:15.06 | Katty | gordonjcp: :P |
| 15:15.22 | gordonjcp | Katty: seriously, we've just gone from 2000 to XP at work, it's a bloody nightmare |
| 15:15.29 | Katty | <client> oh noe!!11oneone11! i can't find my PROGRAMS |
| 15:15.38 | gordonjcp | first, right onto an unfamiliar OS |
| 15:15.41 | mutilator | why did you upgrade? |
| 15:15.50 | gordonjcp | mutilator: no idea, not my company |
| 15:15.58 | gordonjcp | ask Sam Palmisiano |
| 15:16.06 | Katty | i hate working for other companys |
| 15:16.06 | otmar | switch XP to old-style GUI and it's not that different. |
| 15:16.08 | _omer | which is the folder to update HEAD?? |
| 15:16.14 | mutilator | i still prefer 2k to xp |
| 15:16.15 | Katty | and i /really/ hate that attitude that it's My Fault if it doesn't work |
| 15:16.16 | gordonjcp | otmar: but just different enough to make it difficult |
| 15:16.18 | Katty | ooh, that just pisses me off to no end |
| 15:16.18 | irv999 | does anyone know any good sidecars for asterisk? (FOP) non computer based |
| 15:16.24 | mutilator | tho xp does have it's uses |
| 15:16.30 | gordonjcp | otmar: especially if you're not used to either OS |
| 15:16.37 | gordonjcp | irv999: what exactly do you want to do? |
| 15:16.38 | otmar | indeed |
| 15:16.59 | gordonjcp | this is the first time I've really used Windows for any length of time |
| 15:17.19 | Katty | :< |
| 15:17.21 | gordonjcp | it's OK, I wouldn't want to have to put up with it for my day-to-day home use |
| 15:17.35 | Katty | that's cause you're spoiled. |
| 15:17.47 | Katty | actually, i was a windows user. |
| 15:17.52 | Katty | i was raised on it, so to speak. |
| 15:17.54 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 15:18.02 | Katty | but, being an irc addict, someone showed me irssi |
| 15:18.04 | Katty | and ssh |
| 15:18.07 | Katty | and SCREEN |
| 15:18.12 | bkw_ | where did file go |
| 15:18.13 | bkw_ | damn it |
| 15:18.18 | gordonjcp | Katty: it's the only way to fly |
| 15:18.22 | Katty | i died and when to kitty heaven |
| 15:18.23 | bkw_ | I got a task to do today and I need him .. haha |
| 15:18.23 | *** join/#asterisk RoyK (~roy@host-81-191-165-149.bluecom.no) |
| 15:18.33 | gordonjcp | Katty: I kind of "grew up" using Unix |
| 15:18.43 | Katty | lucky you |
| 15:18.54 | gordonjcp | pretty much went straight from DOS to SCO Unix and very, very old SunOS on machines at Uni |
| 15:19.00 | mutilator | i still like my win2k for a desktop |
| 15:19.02 | gordonjcp | back when SCO really were a software company |
| 15:19.15 | RoyK | file[mac]: ping |
| 15:19.16 | irv999 | gordon: view all calls parked.. |
| 15:19.22 | irv999 | gordon: with a device not a computer |
| 15:19.25 | gordonjcp | irv999: ah... |
| 15:19.33 | irv999 | similar to a key system.. |
| 15:19.35 | gordonjcp | irv999: any reason *not* to use a computer? |
| 15:19.44 | irv999 | gordon: takes up to much desk space |
| 15:20.02 | gordonjcp | you've got someone who has a phone but no computer? |
| 15:20.05 | jsharp | Get bigger desks? |
| 15:20.12 | gordonjcp | smaller computers? |
| 15:20.16 | irv999 | gordion: they have a computer.. but their software is dos based.. |
| 15:20.24 | Katty | skip the desk, just get a laptop and use the floor |
| 15:20.34 | Katty | or cowch |
| 15:20.35 | gordonjcp | irv999: can't it run in dosemu or something? |
| 15:20.51 | irv999 | so we can't put it on their machines because it will continually switch between their dos program and the viewer |
| 15:20.58 | irv999 | gordon: nope |
| 15:21.15 | gordonjcp | just doesn't work? |
| 15:21.21 | gordonjcp | hrmmm |
| 15:21.27 | gordonjcp | small laptop? |
| 15:21.33 | irv999 | goron: nope.. the dos program requires hardware integration which the emu does not work with |
| 15:21.41 | gordonjcp | ok, what's the dos software? |
| 15:21.44 | irv999 | gordon: A little expensive for laptop |
| 15:21.56 | irv999 | gordon: it is a terminal based software.. similar to wyse terminals |
| 15:22.05 | lehel | could you tell me how to set the permissions ok in /var/spool/asterisk/voicemail ?? after my calls are saved, //localhost/cgi-bin/vmail (Comedian Mail) .. cannot read them! i have to set allways manually |
| 15:22.12 | gordonjcp | irv999: uhm, is it really just a terminal emulator? |
| 15:22.32 | gordonjcp | ok, this is getting into the realms of scope creep |
| 15:22.42 | gordonjcp | but - what *exactly* is the DOS software? |
| 15:23.04 | irv999 | gordon: yes |
| 15:23.21 | irv999 | gordon: ADS medical billing system it usess kermit to communicate |
| 15:23.48 | irv999 | gordon: if I can figure out the call flow issues I can avoid using this computer based solution.. |
| 15:24.26 | gordonjcp | irv999: you get kermit for *everything* |
| 15:24.26 | gordonjcp | have you tried using kermit on a more modern OS? |
| 15:24.26 | _omer | where I need to do "Make update" ??? folder path plz? |
| 15:24.42 | irv999 | gordon: yes.. it does not work 100$ |
| 15:24.43 | irv999 | 100% |
| 15:25.06 | gordonjcp | hmmm |
| 15:25.07 | irv999 | gordon: If I can figure out how to turn off rining in the headset for a polycom phone I can solve this problem ASAP |
| 15:25.11 | gordonjcp | ok |
| 15:25.25 | gordonjcp | can't help you with the polycom phone, may be able to help with kermit |
| 15:25.32 | _omer | where I need to do "Make update" ??? folder path plz? |
| 15:25.53 | irv999 | gordon: I am always willing to listen on that one.. |
| 15:25.53 | zoa | make update is wherever you installed asterisk before |
| 15:26.03 | zoa | ./usr/src/asterisk for example |
| 15:26.14 | _omer | alright.... |
| 15:26.51 | gordonjcp | irv999: ok, well let me know what you've tried and what's wrong, and I'll see what I can suggest |
| 15:28.41 | irv999 | gordon: the software kermit32 does not run under xp.. I have not tried in a while but from what I remember it does not recognize the serial port.. |
| 15:29.52 | gordonjcp | irv999: ah, Windows |
| 15:30.11 | gordonjcp | I know *nothing* about Windows, I've never installed it, and I've only been using it for about a year |
| 15:30.39 | irv999 | gordon: so I have to keep buying old computers because new machines have difficulty running dos with hyper threading.. |
| 15:31.00 | gordonjcp | irv999: ever considered one of the free Unix-alikes and Kermit? |
| 15:31.26 | irv999 | gordon: not at this office.. unfortunatly.. VERY computer illiterate people |
| 15:31.35 | gordonjcp | irv999: so? |
| 15:31.36 | Qwell | but they can use DOS? come now |
| 15:31.54 | irv999 | no.. there is a shortcut on the windows 98 desktop.. it runs their program.. |
| 15:31.54 | Qwell | try out minicom on Linux or something |
| 15:32.05 | Qwell | put a shortcut to it in X |
| 15:32.09 | irv999 | then when they want to use office they type alt tab and go for it |
| 15:32.13 | gordonjcp | they don't *need* to be computer literate |
| 15:32.35 | gordonjcp | no more so than they do to use XP |
| 15:33.01 | irv999 | gordon: well they need xp to be able to purchase modern computers |
| 15:33.29 | gordonjcp | ? |
| 15:33.42 | gordonjcp | I don't understand what you just said |
| 15:33.46 | *** join/#asterisk sm7syx (~kvirc@212-162-184-20.skbbip.com) |
| 15:33.51 | *** join/#asterisk kFuQ (~somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 15:34.39 | sm7syx | Hi all, please help with a small error when starting asterisk. |
| 15:35.23 | irv999 | gordon: this is true |
| 15:35.40 | irv999 | they dont really use dos.. they click |
| 15:35.41 | irv999 | that is all |
| 15:35.51 | gordonjcp | irv999: well there you go |
| 15:35.51 | irv999 | follow instructions like a monkey |
| 15:36.06 | Qwell | so write up instructions for Linux with minicom on a newer machine |
| 15:36.07 | *** join/#asterisk Nukemizer (~Nuke@67.137.28.165) |
| 15:36.13 | gordonjcp | common misconception that your end users need to somehow be computer geniuses to use Linux |
| 15:36.18 | irv999 | qwell: ok for the future |
| 15:36.38 | gordonjcp | it's all down to how you set it up |
| 15:36.50 | irv999 | goron: it does not have anything to with being a genius |
| 15:36.59 | irv999 | it has to be with what is similar to what they are used to |
| 15:37.41 | gordonjcp | irv999: well, it will be |
| 15:37.42 | irv999 | gordon: for the future |
| 15:37.58 | gordonjcp | it will be the exact same program, running in a slightly nicer-looking terminal |
| 15:40.39 | irv999 | gordon: I agree |
| 15:41.46 | *** join/#asterisk maik (~maik@bfs.cs.uni-sb.de) |
| 15:41.55 | irv999 | so anyone else have experience with polycom? ariel: you were a help.. Still need to figure this out |
| 15:43.05 | [TK]D-Fender | I'm working with them now |
| 15:43.36 | *** join/#asterisk yxa (empty@cm121.gamma228.maxonline.com.sg) |
| 15:49.05 | _omer | how do I know that which peer is on the call..if SIP SHOW INUSE is not working in my asterisk box. |
| 15:49.24 | mutilator | sip show channels.. |
| 15:51.39 | *** join/#asterisk JerJer (~JerJer@jerjer.bronze.supporter.pdpc) |
| 15:52.07 | _omer | actually in my scenario. peers log into the queue... |
| 15:52.20 | *** join/#asterisk jburdine (~jburdine@208.2.145.2) |
| 15:52.48 | _omer | simply, I dont see anything when I do , sip show channels... |
| 15:52.54 | _omer | and I have peers taking calls.. |
| 15:53.07 | mutilator | then they aren't using sip channels.. |
| 15:53.20 | _omer | they are using....SIP |
| 15:53.26 | mutilator | do a 'show channels' |
| 15:53.35 | mutilator | if that still shows nothing, they're all faking their calls |
| 15:53.36 | mutilator | heh |
| 15:53.50 | mutilator | or you're in the wrong box |
| 15:54.34 | _omer | alright....multilator |
| 15:54.36 | _omer | I got it .. |
| 15:55.00 | _omer | sorry my mistake...no call were active. |
| 15:55.18 | _omer | but I dont see anything when I do SIP SHOW INUSE.... |
| 15:56.02 | mutilator | don't see anything as in, nothing at all or no active calls? |
| 15:57.36 | _omer | there are active calls...when I do SIP SHOW INUSE ALL .........I see ZEROsssssssssssssssss.... |
| 15:57.43 | _omer | means...none is inuse |
| 15:58.17 | mutilator | yea, i think it's just broken |
| 15:58.22 | mutilator | never worked for me either |
| 15:58.28 | mutilator | i always look at sip show channels |
| 15:58.42 | _omer | ok what you get when you do SIP SHOW INUSE ? |
| 15:58.51 | _omer | * User name In use Limit |
| 15:58.52 | _omer | * Peer name In use Limit |
| 15:58.55 | _omer | here is what I get... |
| 15:58.58 | mutilator | yea |
| 15:58.59 | _omer | that's it.....:( |
| 15:59.03 | mutilator | o |
| 15:59.07 | mutilator | i get a listing of all my users |
| 15:59.11 | mutilator | it's all 0's tho |
| 15:59.28 | mutilator | Username incoming Limit outgoing Limit |
| 15:59.28 | mutilator | 9898266870 0 N/A 0 N/A |
| 15:59.59 | _omer | why my listing is different that yours? |
| 16:00.07 | mutilator | what version ya using? |
| 16:00.11 | _omer | it was same as yours last 2 days before... |
| 16:00.34 | _omer | Asterisk CVS-HEAD-06/02/05-08:36:40 built by root@MYLINUX on a i686 running Linux |
| 16:00.42 | mutilator | i have head 1/24 |
| 16:00.58 | _omer | can I get the older one? |
| 16:01.10 | mutilator | no.. |
| 16:01.11 | _omer | http://pastebin.ca/13181 <--- please have a look at this... |
| 16:01.26 | *** join/#asterisk jamest (~jamest@adsl-208-191-42-201.dsl.tpkaks.swbell.net) |
| 16:01.27 | mutilator | someone probably changed it recently |
| 16:01.48 | mutilator | wait a cpl more days, update your head and see if it's fixed |
| 16:02.57 | docelm0 | Does anyone know if the DialStatus variable in Asterisk is broke? |
| 16:03.01 | _omer | alright... |
| 16:03.01 | docelm0 | Cause it used to work |
| 16:03.51 | _omer | any how...you have given me another way to see who is taking call...thanks :) |
| 16:04.13 | mutilator | i like the channels method more myself |
| 16:04.19 | mutilator | better detailed info |
| 16:04.29 | _omer | yes....it's better than INUSE |
| 16:05.09 | docelm0 | I just need to find the status of the channel when they hang up. I was getting ANSWER, BUSY, etc.. But now I get nothing |
| 16:05.29 | *** join/#asterisk Paul[NOC] (~paul@66.195.243.254) |
| 16:06.26 | jamest | hello, at some point in the future i was going to look into the feasability replacing our Strata 424i setup with something based upon asterisk |
| 16:06.33 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 16:06.37 | *** join/#asterisk hypa7ia (~leigh@64.223.135.10) |
| 16:07.01 | *** join/#asterisk Falle (falstaf@voip-forum.se) |
| 16:08.05 | cyburdine | hey asterisk channel, I'm curious to know if anyone knows how to test if asterisk realtime is connecting to the database? |
| 16:08.28 | docelm0 | I am using it now.. Just make a call |
| 16:08.35 | cyburdine | is there a cli command that might allow me to test an odbc connection? |
| 16:08.43 | jamest | but yesterday figured out that to add more phones we need to buy more hardware for the strata and the owner isn't sure he wants to do that if we can instead use something like asterisk to add more features to our callcenter. I've been swiming in the docs since then but have a dew questions i can't seem to find |
| 16:08.48 | docelm0 | no |
| 16:10.01 | cyburdine | hmm because it doesn't seem to work I have my sip phone in the DB and it fails to register. |
| 16:10.02 | jamest | what I want to do is integrate our call center app with an IP phone on the desk. |
| 16:10.47 | jamest | so that I can track the phone numbers associated with a customer and pop up their info when they call in, or when they are transfered from one internal rep to another |
| 16:11.00 | cyburdine | I've been following the steps on voip-info.org, but that hasn't yielded a working sip phone |
| 16:11.34 | jamest | i'd also really like to record all calls and associate them with the customer records (to keep on file for a month) |
| 16:11.52 | cyburdine | is there a line that I can look for when asterisk starts up that will confirm that the DB is indeed being contacted? |
| 16:12.58 | cyburdine | it says that it's binding, but I don't see the extensions that I have added in the DB |
| 16:13.07 | jamest | but I can't figure out if an external application on our phone reps desk can associate the current call # with a claim, or cause asterisk to start/stop recording information |
| 16:13.24 | jamest | is such a thing possible? |
| 16:13.59 | *** join/#asterisk mrkyr (~bviitanen@h24-207-80-55.cst.dccnet.com) |
| 16:15.15 | jsharp | You can use the Monitor application to record calls. |
| 16:15.40 | jsharp | And you can connect to the Asterisk manager port to watch call info as calls come in. |
| 16:18.56 | cyburdine | does anyone have a cookbook to get realtime working with postgrep (or any db) via odbc |
| 16:19.08 | cyburdine | err postgreSQL |
| 16:19.23 | [TK]D-Fender | jamest : Very possible, I've done it. |
| 16:19.47 | [TK]D-Fender | jamest : the CallerID thing that is |
| 16:20.13 | zoa | we do it with pgsql |
| 16:20.14 | [TK]D-Fender | jamest : Gets a little tricky handling multiple phone #'s for the same comany when they have a lot of lines though |
| 16:20.22 | zoa | working on it for a week now |
| 16:20.23 | zoa | :) |
| 16:20.46 | jamest | well, i've got 3 types of people calling in |
| 16:20.58 | cyburdine | I know our odbc drivers are working |
| 16:21.01 | Katty | _omer sure pissed me off |
| 16:21.19 | cyburdine | what are the files that you needed to "touch" to get it working. |
| 16:21.27 | zoa | i dont know didnt do it myself |
| 16:21.32 | zoa | but we will post a tutorial on how to do it |
| 16:21.35 | jamest | insurance companies (our clients) from a small set of numbers, their insureds that we're handling work for, shops around the US that do the work |
| 16:21.35 | *** join/#asterisk [Outcast] (~knoppix@c-24-218-94-11.hsd1.ma.comcast.net) |
| 16:21.44 | cyburdine | that would be awesome! |
| 16:21.57 | blop | how could i bridge a zap fxo 2 a zap fxs channel automatically when the PC is off? :) is there any hardware solution ? |
| 16:21.58 | [Outcast] | twisted: you going to spain? |
| 16:22.25 | jamest | the 1st and 3rd are limited numbers, but the middle group changes constaintly |
| 16:22.25 | blop | how would a TDM behave? |
| 16:22.53 | [Outcast] | blop: won't a spa-3000 do that? |
| 16:23.10 | blop | yeah i saw that, but i already got digium hardware :) |
| 16:23.16 | *** join/#asterisk s-ndh-c (~s-ndh-c@der-bastard.net) |
| 16:23.20 | [Outcast] | blop: hehe |
| 16:23.21 | s-ndh-c | hi |
| 16:23.30 | [Outcast] | s-ndh-c: greetings |
| 16:23.45 | ipso | Is there a way to control how often Asterisk registers with its IAX peers? In hopes to help keep the route from going stale? |
| 16:23.52 | [Outcast] | so anyone here going to spain? |
| 16:23.59 | *** join/#asterisk ldav15 (~ldavis@208.2.145.2) |
| 16:24.00 | jamest | but it looks like Monitor and the Manager interface would work as long as the apps "manager" login can be limited as to what it can do |
| 16:24.24 | blop | whats in spain? |
| 16:24.34 | [Outcast] | astricon europe |
| 16:24.36 | zoa | im going |
| 16:24.58 | blop | :) |
| 16:25.10 | [Outcast] | zoa: i am just trying to find my drinking buddies early |
| 16:25.24 | jamest | the other thing that i've not figured out is if I can do a voice over the hold music, or maybe play a song, then a msg from us, then a song |
| 16:25.59 | jamest | it seems that only random playing is allowed or a timed amount of hold, then some other hold, then start of new song |
| 16:26.04 | [Outcast] | jamest: what not setup a shoutcast server and setup playlist for your music on hold |
| 16:26.15 | blop | Admission: $650.00 :p |
| 16:26.19 | [Outcast] | jamest: s/what/why |
| 16:26.30 | [Outcast] | blop: work is paying |
| 16:26.40 | blop | ure lucky |
| 16:26.46 | [Outcast] | blop: yep |
| 16:27.02 | jamest | [Outcast]: cause i didn't know I could? :) |
| 16:27.15 | [Outcast] | jamest: hehe |
| 16:27.17 | coppice | $650 for admission, and I bet there are no dancing girls :-) |
| 16:27.23 | [Outcast] | jamest: i will get you the link |
| 16:27.31 | jamest | [Outcast]: thanks |
| 16:28.31 | s-ndh-c | i would like to connect my old p2 to my ascom phone thingy, cause those ascom phones are too expensive, i would like to connect new phones to the isdn bus without buying new ascom phones and pay the support guys to activate the ports and stuff, can asterisk do something like that for me? |
| 16:29.01 | [Outcast] | jamest: http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf and look at the section on shoutcast music on hold |
| 16:30.10 | jamest | thanks |
| 16:30.17 | [Outcast] | jamest: np |
| 16:31.06 | Cresl1n | coppice: are you planning on going? |
| 16:31.16 | zoa | hey creslin |
| 16:31.24 | s-ndh-c | can i connect phones to my compouter and interconnect it with my existing phone asset? |
| 16:31.33 | coppice | if its moved to asia, i'll probably go :-) |
| 16:34.16 | coppice | its only $40 for the ballet tomorrow night, and I bet they dancing girls there :-) |
| 16:37.10 | *** join/#asterisk SteveL (~stephen@216.62.85.65) |
| 16:38.27 | bkw_ | what cluecon? |
| 16:38.34 | bkw_ | if I have to get dancing girls to get people there.. I WILL!!! |
| 16:38.38 | bkw_ | or dancing guys... |
| 16:38.42 | bkw_ | haha |
| 16:39.36 | *** part/#asterisk s-ndh-c (~s-ndh-c@der-bastard.net) |
| 16:39.41 | SteveL | Hi, I'm having trouble getting MeetMe to work in asterisk. In meetme.conf I have: conf => 500 Then in extensions.conf I have: exten => 500,1,MeetMe(500||) |
| 16:39.47 | *** join/#asterisk brycec (~brycec@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:39.54 | SteveL | Everytime I call extension 500 it says invalid conference. |
| 16:40.11 | SteveL | actually 'That is not a valid conference number.' |
| 16:40.25 | bkw_ | bet you don't have zap hardware |
| 16:40.56 | SteveL | i tried installing the ztdummy but can't get it to work |
| 16:41.02 | bkw_ | use a 2.6 kernle |
| 16:41.07 | SteveL | i am |
| 16:41.08 | coppice | bkw_: dancing girls, and a location in asia? :-) |
| 16:41.21 | bkw_ | SteveL, ztdummy should "just work" |
| 16:41.31 | bkw_ | coppice, hrm no asian part |
| 16:41.33 | bkw_ | :( |
| 16:41.37 | bkw_ | I would love to have you there |
| 16:41.41 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 16:41.53 | SteveL | bkw: so is it installed by default? I don't have to insmod? |
| 16:42.07 | bkw_ | modprobe ztdummy |
| 16:42.09 | coppice | it sounds better when straight guys say that :-) |
| 16:42.24 | bkw_ | insmod doesn't pickup dependancies |
| 16:42.29 | SteveL | tried that... |
| 16:42.34 | bkw_ | what was the error? |
| 16:42.36 | SteveL | FATAL: Module ztdummy not found. |
| 16:42.40 | bkw_ | depmod -a |
| 16:42.45 | bkw_ | uname -a |
| 16:42.47 | cyburdine | [asterisk] |
| 16:42.47 | cyburdine | dsn => PGSQL-asterisk |
| 16:42.47 | cyburdine | username => fmsuser |
| 16:42.47 | cyburdine | password => fmsuser |
| 16:42.47 | cyburdine | pre-connect => yes |
| 16:42.47 | bkw_ | <PROTECTED> |
| 16:42.56 | *** kick/#asterisk [cyburdine!~bkw_@bkw.developer.and.friend.of.asterisk] by bkw_ (USE PASTEBIN) |
| 16:43.22 | SteveL | Linux pbx 2.6.11-1.27_FC3 #1 Tue May 17 20:27:37 EDT 2005 i686 i686 i386 GNU/Linux |
| 16:43.56 | SteveL | ran depmod -a and tried modprobe again and got same error |
| 16:44.23 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.8 RC bug #4424 || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm || 1PM CDT Developers Conference Call IAX2/guest@switch-3.asterlink.com/996 |
| 16:44.27 | bkw_ | 1PM CDT Developers Conference Call IAX2/guest@switch-3.asterlink.com/996 |
| 16:44.30 | bkw_ | everyone take note |
| 16:44.33 | bkw_ | pass it around |
| 16:44.36 | *** join/#asterisk cyburdine (~jburdine@208.2.145.2) |
| 16:44.44 | cyburdine | sorry about that |
| 16:44.47 | MikeJ[Laptop] | yeah yeah yeah |
| 16:45.49 | cyburdine | my question was going to be what does the [asterisk] context refer to in res_odbc.conf |
| 16:46.02 | cyburdine | can I name that whatever I want? |
| 16:46.06 | bkw_ | the data source name you'll ref in extconfig.conf |
| 16:46.16 | bkw_ | so yes you can name it jimbobsdatabase |
| 16:46.21 | bkw_ | or what ever you wish |
| 16:47.17 | puppet | bkw_: how long time til conferense start? |
| 16:47.26 | puppet | bkw_: whats 1pm cdt in gmt? |
| 16:48.11 | cpatry | +7 i think. |
| 16:48.48 | puppet | ok |
| 16:48.57 | SteveL | bkw_: how can I get modprobe to find that file? Is there a way I can tell it to search in /usr/src/zaptel? |
| 16:49.11 | cyburdine | so in extconfig.conf it should look like: extensions => odbc,asterisk,extensions |
| 16:49.28 | cyburdine | or should I have [asterisk] above my extensions => odbc,asterisk,extensions line |
| 16:49.46 | anthm | -5 |
| 16:52.06 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net) |
| 16:52.39 | Dovid | morning all |
| 16:53.38 | Dovid | any one know if digium has some kind of refferal list |
| 16:54.05 | Dovid | i have a potential client that wants to ask around b4 he gets it. is there any places that he can ask about asterisk ? |
| 16:55.20 | Dovid | anyone ? |
| 16:57.20 | bkw_ | no |
| 16:57.23 | bkw_ | -5 |
| 16:57.23 | bkw_ | haha |
| 16:57.49 | bkw_ | is in one hour, 3 min. |
| 16:58.04 | puppet | ty |
| 16:59.46 | *** join/#asterisk l-fy (~pchitescu@l-fy.developer.yate) |
| 17:00.17 | *** join/#asterisk bofh42 (~bofh42@p54821A9D.dip0.t-ipconnect.de) |
| 17:00.30 | doolph | anyone here know how to use oh323 |
| 17:00.31 | twisted[work] | bkw_, WHEEE |
| 17:01.22 | cyburdine | so I'm still a bit confused when I start asterisk I get: Binding sip.conf to odbc/asterisk/sipusers |
| 17:01.57 | cyburdine | but further down I get :Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) |
| 17:02.35 | cyburdine | my extconfg.conf contains: sip.conf => odbc,asterisk,sipusers |
| 17:03.02 | cyburdine | which is a valid driver,db, and table |
| 17:03.14 | sm7syx | Hello all, anyone here that can help with an "illegal instruction" ? |
| 17:04.33 | gordonjcp | sm7syx: sure |
| 17:04.42 | gordonjcp | sm7syx: go and drill holes in your neighbour's door |
| 17:05.51 | sm7syx | gordonjcp, proble ! No neighbour !! |
| 17:07.57 | sm7syx | So I guess I'm stuck with my little error ! |
| 17:08.04 | OnlyMe | <PROTECTED> |
| 17:08.27 | johnnyb | Don't give illegal instructions. I'll have to call the police. |
| 17:08.46 | johnnyb | But my Vonage 911 service isn't working. |
| 17:09.09 | sm7syx | OnlyMe, I get that error after the line ' Regist... gsmtolin from format ... cost 9 |
| 17:09.10 | gordonjcp | johnnyb: git up git git down, 911's a joke in yo' town |
| 17:09.57 | Sato1 | anyone having problems compiling cdr_mysql? i m getting this error: http://pastebin.ca/13192 |
| 17:10.02 | johnnyb | sm7syx: I'm guessing that you compiled with options that aren't valid for your processor. |
| 17:10.08 | sm7syx | Sorry, I didn't get it ;-) |
| 17:10.12 | OnlyMe | sm7syx i'm so new in this .... maybe try http://pastebin.ca some ppl in channel read them all i'm sure |
| 17:10.36 | johnnyb | sm7syx: either that or the binary file got corrupted. |
| 17:10.47 | johnnyb | sm7syx: did you compile from source, or download a binary? |
| 17:10.55 | SteveL | anyone have any hints on how to get the modprobe ztdummy to work? |
| 17:11.07 | bkw_ | WEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEE |
| 17:11.10 | sm7syx | jonnyb, i use gentto dist and I have never before encountered that problem, but there should always be a first time. |
| 17:11.12 | gordonjcp | SteveL: uhm, "modprobe ztdummy" |
| 17:11.12 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 17:11.18 | puppet | bkw_: :o |
| 17:11.23 | bkw_ | twisted whats up? |
| 17:11.29 | SteveL | FATAL: Module ztdummy not found. |
| 17:11.39 | bkw_ | someone didn't make install |
| 17:11.40 | gordonjcp | have you compiled it? |
| 17:11.40 | bkw_ | in zaptel |
| 17:11.48 | bkw_ | does modprobe zaptel work? |
| 17:11.51 | *** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com) |
| 17:12.01 | johnnyb | sm7syx: what processor are you using? |
| 17:12.05 | johnnyb | (cat /proc/cpuinfo) |
| 17:12.10 | bkw_ | an uberproc2000 |
| 17:12.17 | bkw_ | it has 1 gig of L2 cache |
| 17:12.18 | bkw_ | it rocks |
| 17:12.21 | bkw_ | its only 50mhz |
| 17:12.30 | twisted[work] | what, why am I happy? |
| 17:12.36 | bkw_ | yes why? |
| 17:12.43 | twisted[work] | because a certain company fixed this certain POS that was causing certain headaches |
| 17:13.04 | bkw_ | oh good to hear the certain POS was fixed by a certain company... |
| 17:13.05 | sm7syx | jonnyb, processor is Pentium Pro and -mcpu and -march is pentiumpro ! |
| 17:13.08 | denon | 1 gig of L2 cache? wonder who let bkw hit the pipe again |
| 17:13.12 | twisted[work] | bkw_, you should know what i'm talking about |
| 17:13.15 | twisted[work] | if not, aim me |
| 17:13.17 | bkw_ | I know |
| 17:13.20 | bkw_ | i'm being silly ya know |
| 17:13.22 | bkw_ | damn boi |
| 17:13.34 | twisted[work] | okay |
| 17:13.34 | puppet | bkw_: where do i get thoose cpus? i want 1gb l2 to! |
| 17:13.46 | puppet | bkw_: then ill mod it and put it on my 486, the power! |
| 17:13.47 | bkw_ | 47 more min till the dev conf |
| 17:13.50 | sm7syx | jonnyb, machine has dual cpu ! (pentium pro 166 ...) |
| 17:13.52 | Sato1 | http://pastebin.ca/13192, anyone? :D |
| 17:14.23 | denon | puppet: strap a solid state drive to your proc .. close 'nuff |
| 17:14.33 | puppet | denon: woot ;P |
| 17:14.44 | puppet | that would rock ;P |
| 17:14.46 | puppet | or not ;P |
| 17:14.53 | denon | New AMD Opteron with FC RAID CONTROLLER! |
| 17:15.19 | puppet | all people would buy it ;P |
| 17:15.20 | bkw_ | Asterisk Ready. |
| 17:15.20 | bkw_ | *CLI> show version |
| 17:15.20 | bkw_ | Asterisk CVS-HEAD-04/28/05-15:09:00 built by brian@imac.local on a Power Macintosh running Darwin |
| 17:15.27 | bkw_ | NO NO this is where its at |
| 17:15.30 | bkw_ | Power mac |
| 17:15.31 | bkw_ | baby |
| 17:15.34 | bkw_ | mac mac mac |
| 17:15.45 | denon | bkw just likes the fruity colors |
| 17:15.49 | puppet | pfft ;P |
| 17:15.52 | puppet | Asterisk 1.0.7 built by puppet@insitu on a i686 running Linux |
| 17:16.00 | [TK]D-Fender | Like.... GREY..... bestest colour EVER! |
| 17:16.24 | denon | Asterisk 2.0.2 built by denon@pbx12 on a i686 running Linux |
| 17:16.28 | *** part/#asterisk l-fy (~pchitescu@l-fy.developer.yate) |
| 17:16.29 | [TK]D-Fender | :O |
| 17:16.32 | puppet | 2.0.2 :o |
| 17:16.39 | *** join/#asterisk kFuQ (~somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 17:16.46 | puppet | cheat |
| 17:16.56 | denon | comes with full sccp support! |
| 17:16.56 | puppet | denon: cheater! :( |
| 17:16.59 | gordonjcp | Asterisk 1.0.7 built by root@ishikawa on a i386 running NetBSD |
| 17:17.14 | denon | 12 9's redundancy |
| 17:17.14 | gordonjcp | I really need to make that properly |
| 17:17.21 | denon | and native clustering |
| 17:17.54 | puppet | i think someone can make a list of free Dids ;P |
| 17:17.59 | bkw_ | fruity colors? |
| 17:18.03 | bkw_ | my imac is WHITE |
| 17:18.06 | bkw_ | not fruity |
| 17:18.09 | bkw_ | WHITE |
| 17:18.10 | puppet | wtb numbers in more contries ;P just havegermany/uk/us ;P |
| 17:18.20 | denon | it only achieved 12 9's of redundancy because they couldnt get the cpu to cycle any faster, and its technically down between cycles |
| 17:18.41 | cyburdine | minty white? |
| 17:18.46 | denon | bkw_: wow, apple makes a white imac? I thought the only reason people bought em was 'cause of their fruity colors |
| 17:18.49 | bkw_ | shiny |
| 17:18.54 | bkw_ | no |
| 17:19.15 | puppet | bkw_: shiny milk white? |
| 17:19.27 | [TK]D-Fender | Just auto-provisioned a new IP 600 :D yay |
| 17:19.40 | cyburdine | ah prefer beige... it was a level playing field when things were beige |
| 17:19.44 | Katty | that's...interesting |
| 17:19.45 | denon | Ive gotta play with some IP600s |
| 17:19.50 | Katty | 12:22 < puppet> bkw_: shiny milk white? |
| 17:20.06 | Katty | hmm, just coincidence |
| 17:20.25 | Katty | bkw_: i don't wanna go back to work :< |
| 17:20.55 | denon | ooh, the ip600s are kinda pricey |
| 17:20.56 | puppet | http://cheston.com/pbf/PBF049ADChewBoy.jpg ;P |
| 17:21.08 | denon | I thought the whole idea was a cool cheaper phone, but at this price, may as well go cisco |
| 17:22.29 | [TK]D-Fender | Denon, the only reason the Cisco's look cheaper is because they are unlicensed and yu get NO support and can't DL firmware or anything from Cisco. |
| 17:23.01 | [TK]D-Fender | I've seen IP600's aroun 280$US which is still cheaper that the lowest 7960G I've seen at 299. |
| 17:23.43 | denon | yeah .. spose, but for the price difference, the 7960's been really solid for us |
| 17:23.47 | [TK]D-Fender | And Polycom supports REAL PoE (not just Cisco's proprietary PoE). If you need a power brick for Cisco.. $$$ MORE |
| 17:23.57 | denon | 7960 does real PoE now too |
| 17:24.04 | [TK]D-Fender | Finally? |
| 17:24.13 | Nugget | for quite some time. |
| 17:24.18 | denon | afaik the 7960g did already |
| 17:24.23 | denon | like a couple years ago heh |
| 17:24.30 | Nugget | the only reason the old ciscos don't do "real" poe is because "real" poe didn't exist then. |
| 17:24.31 | cyburdine | so when asterisk realtime starts should I see it load extensions like it would if it was loading from the conf file? |
| 17:24.33 | bkw_ | does it do standard POE? |
| 17:24.36 | [TK]D-Fender | but then again thats unlicensed. add up almost 50% of the unit price for proper support or you'll be running them illegally (which for home use I'm sure most won't care about) |
| 17:24.43 | Nugget | bkw_: yes |
| 17:25.08 | cyburdine | dang... |
| 17:25.37 | denon | of course, polycomm isnt exactly forthcoming with their firmware either |
| 17:25.45 | [TK]D-Fender | News to me, but hey, why not.... either way the net cost of a legit Cisco >> Polycom and Polycom is more "standards" driven. from what I can tell. Cisco is solid stuff though I'm sure. |
| 17:26.28 | [TK]D-Fender | denon : You should get it through your auth'd reseller. You can even go through a Q&A multiple choice type deal to get yourself "authorized" |
| 17:26.40 | denon | yep |
| 17:26.40 | bkw_ | you sure? |
| 17:26.49 | bkw_ | I have two 7960G's that won't do it |
| 17:26.52 | denon | but I'm saying, its not like they just have a link on their site |
| 17:27.00 | [TK]D-Fender | Friend of mine is a frequent supplier of my company and is going throgh it now for me. |
| 17:27.19 | Nugget | 7960g is ieee 802.3af poe |
| 17:27.58 | bkw_ | mine isn't |
| 17:28.20 | Nugget | http://lnk.nu/cisco.com/2xy.shtml |
| 17:28.27 | Nugget | send it back to cisco as defective, then. |
| 17:28.33 | *** join/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
| 17:28.37 | cyburdine | in theory I can delete extensions.conf if I have it defined in extconfig.conf? |
| 17:28.46 | [TK]D-Fender | bkw_ : I remember the photo's you sent of your home office a year ago, nice setup. I'm looking for a similar layout myself. |
| 17:28.50 | bkw_ | cyburdine, NO |
| 17:29.01 | bkw_ | [TK]D-Fender, home office? |
| 17:29.05 | cyburdine | hmm same for sip.conf? |
| 17:30.52 | *** part/#asterisk sretooh (sretooh@63.252.229.9) |
| 17:31.02 | [TK]D-Fender | When you were renovating. Glass table, pivot lamp, 7960 on the desk, dual LCD |
| 17:31.51 | Marlow | [TK]D-Fender: the glass table is usually a bad idea .. |
| 17:31.56 | *** part/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 17:32.00 | [TK]D-Fender | Yeah for optical mice.... |
| 17:32.01 | Marlow | [TK]D-Fender: it doesn't look too good for long .. |
| 17:32.02 | Nugget | I'm too much of a slob to use a glass tabletop. |
| 17:32.18 | Marlow | [TK]D-Fender: No .. also scratches, etc .. |
| 17:32.19 | [TK]D-Fender | And the Windex bill's *shudder* |
| 17:32.24 | [TK]D-Fender | :D |
| 17:32.27 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 17:32.50 | Marlow | [TK]D-Fender: but the other stuff .. that's actually in place .. nearly :) |
| 17:33.36 | *** join/#asterisk ansel (~ansel@out.emointernet.com) |
| 17:35.18 | Marlow | [TK]D-Fender: http://www.marlow.dk/site.php/pics/image/gallery//200503-Move_to_Galway/img_0917.jpg |
| 17:35.35 | Marlow | [TK]D-Fender: that ISDN phone is replaced by a Cisco 7960 a while ago :) |
| 17:37.06 | sm7syx | jonnyb, You where right ! I did have 'mmx' option set and processor did not support it. Recompile and it runs ! Thanks ! |
| 17:37.08 | Marlow | TFT can be turned, for relaxing on the couch, seeing a movie .. and there is a remote for the pc :) |
| 17:37.33 | [TK]D-Fender | I'm way too cheap to buy much for myself and don't really have the need. At home its enough that I've got a TDM22B running 2 phones for me, and soo my SPA2000 in service for a cordless & desk phone extra |
| 17:37.38 | puppet | 25min to conferense ;P |
| 17:37.45 | Nugget | http://lnk.nu/slacker.com/2y0 |
| 17:38.34 | *** join/#asterisk Lee__ (~Lee__@cpe-69-203-211-144.nyc.res.rr.com) |
| 17:38.35 | Marlow | Nugget: i prefer 2 screens, don't like widescreen .. |
| 17:38.46 | Nugget | I hate two screens. :) |
| 17:38.54 | *** join/#asterisk jsolares (~jsolares@200.30.141.86) |
| 17:38.54 | Marlow | Nugget: i'm looking at replacing that 19" CRT and 19" TFT with 2 20" ones |
| 17:39.09 | Nugget | I never use the secondary display. it just acts as a neglected window ghetto |
| 17:39.30 | Marlow | Nugget: actually .. there are 3 screens on the desk :) |
| 17:40.07 | *** join/#asterisk zztopper2 (~me@ip70-177-50-126.br.br.cox.net) |
| 17:40.10 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 17:40.12 | Marlow | Nugget: one 17" connected to the KVM, for monitoring machines that are being build, debugging and stuff :) |
| 17:40.34 | [TK]D-Fender | Nugget : Nice screen..... |
| 17:40.42 | [TK]D-Fender | That the 23" model? |
| 17:40.49 | Nugget | no, the 30" |
| 17:40.53 | [TK]D-Fender | :O |
| 17:41.09 | [TK]D-Fender | "Size doesn't matter" *bullshit* :D |
| 17:41.14 | Nugget | heh |
| 17:41.21 | *** join/#asterisk irv999 (~irv999@ool-44c241c7.dyn.optonline.net) |
| 17:41.24 | irv999 | lo all |
| 17:41.57 | *** join/#asterisk bewest (~bewest@h48.204.255.206.cable.srcy.cablelynx.com) |
| 17:42.17 | irv999 | Still looking for help on polycom phones.. |
| 17:42.20 | irv999 | :-( |
| 17:42.52 | [TK]D-Fender | What do you need to know? |
| 17:43.47 | zztopper2 | who is available for consulting on asterisk? |
| 17:43.48 | irv999 | tk: I need to know how to turn off the calll waiting (beeping) feature on a ip600.. When somoene is on the phone NO ring in the headset |
| 17:43.55 | blitzrage | zztopper2: what do you need? |
| 17:44.20 | blitzrage | irv999: thats not just done in the GUI ? |
| 17:44.25 | zztopper2 | call center in phils |
| 17:45.12 | zztopper2 | distributed call centers... phils.. + US |
| 17:45.53 | blitzrage | zztopper2: sent you a contact in a PM |
| 17:46.06 | irv999 | blitz: the menu on the polycom phones? not that I knowof |
| 17:46.52 | [TK]D-Fender | You need to set your lines to 1 call each which will effectively disable call-waiting. |
| 17:47.19 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 17:47.37 | [TK]D-Fender | My IP 600's Allocate 6 line keys (all) to a single registration with 1 call max each. That way it auto-cascades to the next available line key. |
| 17:47.46 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 17:49.15 | *** join/#asterisk Silik0n (~krice@newrso.suspicious.org) |
| 17:49.27 | Silik0n | anyone using polycoms and Alert Info headers? |
| 17:52.28 | *** part/#asterisk ldav15 (~ldavis@208.2.145.2) |
| 17:53.57 | irv999 | tk: what happens when you are on the phone and the next button rings? do you hear a beep in the handset? |
| 17:54.42 | *** join/#asterisk Veto (mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 17:55.00 | [TK]D-Fender | I think it does beep once. Not sure If thats disableable |
| 17:55.13 | [TK]D-Fender | It should only ring. Will test now |
| 17:55.35 | *** join/#asterisk tangel (tangel@64.135.81.8) |
| 17:55.50 | irv999 | tk: that would be awesome.. ours keep beeping.. Although Ours cascade 2 per button.. |
| 17:56.47 | [TK]D-Fender | Ok, it beeps once, then flashes until my Dial command gives up. |
| 17:57.10 | irv999 | tk: hrmmmm ok |
| 17:57.14 | [TK]D-Fender | set the # of calls to 1 and it'll just fall to the next button for each. Call waiting with multiple mile keys is silly. |
| 17:57.19 | *** part/#asterisk ansel (~ansel@out.emointernet.com) |
| 17:57.29 | [TK]D-Fender | s/mile/line |
| 17:57.43 | puppet | the conference is muted right? |
| 17:58.08 | blitzrage | which conf ? |
| 17:58.23 | puppet | ./topic |
| 17:58.33 | blitzrage | puppet: nope, not muted |
| 17:59.22 | puppet | ok |
| 18:00.18 | irv999 | tk: ok.. I will try |
| 18:03.03 | *** part/#asterisk jeffik (~Jeff@69.158.21.177) |
| 18:05.29 | shmaltz | anybody from nufone here? |
| 18:05.55 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:05.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:10.25 | SteveL | ok i have ztdummy installed but meetme still won't work |
| 18:10.32 | *** join/#asterisk zztopper (~me@ip70-177-50-126.br.br.cox.net) |
| 18:10.35 | irv999 | tkL can I pick your brain for a sec? |
| 18:10.38 | irv999 | tk: |
| 18:10.39 | SteveL | looking at the log file it is looking for /dev/zap but it doesn't exist |
| 18:10.48 | SteveL | ztdummy shows up in a lsmod |
| 18:11.57 | *** join/#asterisk meppl (mephisto@p54AAD9E9.dip.t-dialin.net) |
| 18:15.45 | *** join/#asterisk AndrewKT (~mirc@nc-69-68-178-225.sta.sprint-hsd.net) |
| 18:18.50 | *** join/#asterisk cluecon (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 18:19.08 | cluecon | ~cluecon |
| 18:19.09 | jbot | it has been said that cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
| 18:19.32 | cyburdine | still banging my head against realtime... where does this context get set? |
| 18:19.35 | cyburdine | Setting global variable 'switch' to 'Realtime/@extensions' |
| 18:19.41 | *** join/#asterisk Godsey (lanny@204.17.223.9) |
| 18:19.42 | puppet | hahahahahahahahahahahahahahahaha, sneaklistens to conferense ;P |
| 18:19.56 | Godsey | so who said microsoft isn't getting into voip? :) |
| 18:20.08 | cyburdine | or where is "extensions" referencing? |
| 18:20.16 | Godsey | I was just invited to a ms voip conference in bellevue |
| 18:21.15 | docelm0 | Godsey ya.. Same here.. I said hell NO! |
| 18:21.31 | Godsey | I said yes, it's free copy of live communication server and 25 cals :) |
| 18:21.48 | Godsey | 100% of my ms server software is free from seminars |
| 18:22.08 | Godsey | I only wish they would put on a MSDN seminar :P |
| 18:22.13 | Godsey | I want free msdn dammit |
| 18:22.21 | docelm0 | I have MSDN.. :) |
| 18:22.24 | cyburdine | from what I am reading it should be referring to extconfig.conf which contains extconfig.conf:extensions => odbc,asterisk,ast_extensions |
| 18:23.00 | cyburdine | but for some reason I get Jun 2 12:15:06 WARNING[21553]: pbx.c:3670 ast_merge_contexts_and_delete: Requested contexts didn't get merged |
| 18:23.34 | *** part/#asterisk Moc_ (~mochouina@64.235.210.66) |
| 18:23.59 | blitzrage | anyone know how to stop stund from displaying the *****'s on the linux console? I'm not even using the -v verbose flag :) |
| 18:24.06 | *** join/#asterisk implicit (~implicit@dhcp-144188.mobile.uci.edu) |
| 18:26.05 | puppet | how many here are in the closet? ;P |
| 18:26.35 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 18:28.05 | blitzrage | just russell :) |
| 18:28.11 | puppet | haha ;P |
| 18:28.33 | blitzrage | damnit! Can't seem to keep stund from displaying *'s on the console |
| 18:28.41 | *** join/#asterisk ZX81 (matt@222-153-114-70.jetstream.xtra.co.nz) |
| 18:28.42 | puppet | -d ? |
| 18:28.47 | puppet | wait |
| 18:28.48 | puppet | ill check |
| 18:29.06 | blitzrage | -b to set it in the background, but does nothing |
| 18:29.15 | blitzrage | I even added & to the end |
| 18:29.15 | blitzrage | :) |
| 18:29.20 | blitzrage | still shows the *'s |
| 18:29.28 | puppet | hmm |
| 18:29.38 | blitzrage | ZX81: zup |
| 18:30.19 | *** join/#asterisk alerios (~alerios@63.245.86.184) |
| 18:30.27 | *** join/#asterisk Moc_ (~mochouina@64.235.210.66) |
| 18:30.53 | meppl | guten abend |
| 18:32.14 | Marlow | w*** the f*** ... wasn't aware, that virbiage has a IAX2 and SIP capable ATA now, that does G729 |
| 18:33.46 | ZX81 | blitzrage: hey |
| 18:34.03 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 18:34.37 | *** join/#asterisk bprice20 (~brandon@Unassigned-216.120.255.29.hrwebservices.net) |
| 18:36.28 | blitzrage | so no suggestions for my stund situation? :) |
| 18:36.37 | tzafrir_laptop | what's stund? |
| 18:36.55 | blitzrage | a STUN server by vovida.org |
| 18:37.10 | blitzrage | can't seem to get it to stop displaying *'s on the console, even when run in the background |
| 18:37.23 | tzafrir_laptop | blitzrage, 2/dev/null ? |
| 18:37.32 | tzafrir_laptop | blitzrage, >/dev/null ? |
| 18:37.44 | blitzrage | ahhhh... hrmmm, I should try that :) |
| 18:37.50 | *** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il) |
| 18:38.19 | tzafrir_laptop | Anyway, a daemon should detach itself from its terminal |
| 18:38.59 | *** part/#asterisk lehel (~lehel@82.79.20.17) |
| 18:39.26 | blitzrage | you'd think so :) |
| 18:39.57 | Hmmhesays | anyone use one of those cheap linksys wireless ethernet bridges? |
| 18:40.15 | tzafrir_laptop | Hmmhesays, used one of those |
| 18:40.17 | blitzrage | tzafrir_laptop: wow, no change |
| 18:40.59 | Hmmhesays | tzafrir: how'd it work? |
| 18:45.05 | ZX81 | ~ping |
| 18:45.07 | jbot | pong |
| 18:45.14 | Hmmhesays | haha tzafrir died |
| 18:45.54 | Marlow | muhahahaa |
| 18:46.11 | Marlow | blitzrage: eh ... just ordered new toy :) |
| 18:46.28 | *** join/#asterisk P-Chan (~jpfingstm@68.142.66.200) |
| 18:46.33 | P-Chan | Hello |
| 18:46.47 | Marlow | toys are good |
| 18:47.01 | ZX81 | ~seen bkw_ |
| 18:47.03 | jbot | bkw_ is currently on #asterisk (5h 9m 47s). Has said a total of 208 messages. Is idling for 1h 18m 2s |
| 18:47.03 | blitzrage | Marlow: ooooo, what'd you order?> |
| 18:47.16 | P-Chan | Unable to open IAX timing interface: Permission denied - Despite having 666 on /dev/zap/* and /proc/zaptel showing Span 1: ZTDUMMY/1 "ZTDUMMY/1 1". Ideas? |
| 18:47.18 | Marlow | blitzrage: virbiage ata .... does IAX2 and SIP |
| 18:47.25 | blitzrage | neato |
| 18:47.26 | blitzrage | ! |
| 18:47.37 | *** join/#asterisk cmk (~cmk_@p54A3F924.dip.t-dialin.net) |
| 18:47.39 | jsolares | it says it'll do SIP with a new firmware in the future |
| 18:48.06 | Marlow | blitzrage: jup .. i didn't know, that they finally had got themselves together to launch anything, but it seems for once to be for real |
| 18:48.21 | blitzrage | hehe |
| 18:48.36 | Marlow | blitzrage: and it does, what the IAXy doesn't: G729, G726, GSM-FR and iLBC |
| 18:49.41 | Marlow | blitzrage: ilbc later though .. |
| 18:49.59 | blitzrage | I don't worry about bandwidth... G.711u baby! :) |
| 18:50.22 | Hmmhesays | hrm can I get by on a p4 2.8ghz machine transcoding 25 sip calls? |
| 18:50.38 | blitzrage | I'd think so |
| 18:50.42 | mutilator | ya |
| 18:50.45 | Marlow | Hmmhesays: sure .. |
| 18:50.53 | P-Chan | Hmmhesays: Yeah, we run about 30-40 calls on a 2.4 ;) |
| 18:51.01 | Hmmhesays | transcoding 30/40? |
| 18:51.27 | P-Chan | Hmmhesays: IAX2 |
| 18:51.36 | tzafrir_laptop | P-Chan, what about the directory /dev/zap itself? |
| 18:51.43 | Marlow | Hmmhesays: anyhow .. the thumb-rule says .. dual P4 2.4 GHz for transcoding 120 channels .. should be ok |
| 18:52.02 | Marlow | Hmmhesays: so 25 og a single P4 2.8 should be ok .. |
| 18:52.05 | P-Chan | tzafrir_laptop: 666 there too |
| 18:52.14 | Hmmhesays | yes one would think, thanks for the info |
| 18:52.26 | Marlow | Hmmhesays: there is only one way to prove it :) |
| 18:52.26 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 18:52.40 | Hmmhesays | true |
| 18:52.54 | Marlow | Hmmhesays: but one of my boxes is a dial PRI with P4 2.667 ghz .. and haven't had problems .. most calls transcode G.729 |
| 18:53.00 | tzafrir_laptop | P-Chan, it is a directory. It should be executable |
| 18:53.10 | Marlow | Hmmhesays: s/dial/dual/ |
| 18:53.13 | P-Chan | tzafrir_laptop: ok, I'll try that. ;) |
| 18:53.18 | Hmmhesays | yeah i caught that ;) |
| 18:53.40 | Marlow | Hmmhesays: just give your box also enough memory |
| 18:54.22 | *** join/#asterisk AlexCeli (~Alex@200.37.85.96) |
| 18:54.35 | Hmmhesays | yeah, this one is price sensitive, so i'm trying to cut corners |
| 18:55.09 | Marlow | Hmmhesays: memory is inexpensive .. don't be cheap on that :) |
| 18:55.16 | Marlow | Hmmhesays: or it will hurt you .. |
| 18:55.21 | claude2005 | hello there |
| 18:55.25 | Hmmhesays | ahhh I was just saying in general |
| 18:55.27 | Hmmhesays | :) |
| 18:55.28 | claude2005 | Internal call keeps ringing even though the external call has hungup |
| 18:55.37 | Hmmhesays | nope we don't want rtp in the swap file |
| 18:55.45 | Marlow | Hmmhesays: ehh .. right .. |
| 18:55.48 | claude2005 | i am using TDM22P and BT uk Land line |
| 18:55.51 | claude2005 | can any one help |
| 18:55.54 | blop | virbiage ata ? |
| 18:55.57 | P-Chan | tzafrir_laptop: That did the trick! Thanks alot! |
| 18:56.02 | Marlow | claude2005: you can't be helped :) |
| 18:56.06 | Marlow | blop: yap |
| 18:56.14 | claude2005 | why not |
| 18:56.20 | claude2005 | is there no soloution to this |
| 18:56.21 | Marlow | claude2005: it's BT :) |
| 18:56.30 | Marlow | claude2005: run and hide |
| 18:56.37 | claude2005 | i know but i live in the UK and i can only use BT |
| 18:56.47 | Marlow | LOL |
| 18:56.51 | blop | Marlow where did u ordered ? :) |
| 18:57.02 | Marlow | blop: on their website .. pre-order . |
| 18:57.17 | Marlow | blop: they are selling them at 99 something AUD |
| 18:57.22 | blop | i see :) |
| 18:57.25 | claude2005 | Marlow: any idea what i can do |
| 18:57.31 | blop | oh, thats $AUD :D |
| 18:57.35 | Marlow | claude2005: what's the problem .. |
| 18:57.39 | *** part/#asterisk AndrewKT (~mirc@nc-69-68-178-225.sta.sprint-hsd.net) |
| 18:57.44 | Marlow | blop: shouldn't make it worse .. |
| 18:57.58 | blop | looks great :) |
| 18:58.00 | Marlow | claude2005: you need to be more specifik .. usually it works |
| 18:58.03 | Marlow | blop: indeed |
| 18:58.14 | claude2005 | the internal phone keeps on ringing even if the external caller has hungup |
| 18:58.17 | Marlow | blop: and it does both IAX2 and SIP .. so .. can't go wroong |
| 18:58.28 | claude2005 | internal to internal there is no problem |
| 18:58.33 | Marlow | claude2005: that's a common problem .. no hangup detection .. |
| 18:58.59 | claude2005 | what is the way round a hangup detection |
| 18:59.17 | tzafrir_laptop | claude2005, timeout :-( ? |
| 18:59.39 | *** join/#asterisk Blake0PS (~blake@blakeops.com) |
| 18:59.40 | claude2005 | i have a time out of 30 seconds then go to voice mail |
| 18:59.40 | *** join/#asterisk AndrewKT (~mirc@nc-69-68-178-225.sta.sprint-hsd.net) |
| 18:59.50 | claude2005 | but i keep geting blank voice mails |
| 19:00.18 | blop | Marlow i would buy one if i had no iaxy yet :p |
| 19:00.42 | Marlow | blop: i do have a pre-production iaxy |
| 19:00.48 | blop | preproduction ? |
| 19:00.52 | Marlow | claude2005: http://lists.digium.com/pipermail/asterisk-users/2004-November/070862.html deals with it .. |
| 19:01.08 | Marlow | claude2005: try to search the list .. |
| 19:01.13 | Marlow | claude2005: you might find more .. |
| 19:01.35 | Marlow | blop: yep .. the first series that went out to a few people |
| 19:01.57 | blop | humm, u mean the blue&orange ones ? |
| 19:01.59 | *** join/#asterisk stevek (~stevek@slim-eth0.horizonlive.net) |
| 19:02.29 | blop | coz i got a blue&orange version (which is older than the new black version) |
| 19:02.40 | Marlow | blop: there was a limited series they send out before they started selling them in amounts, but yes, blue and orange .. |
| 19:02.53 | blop | :) |
| 19:03.11 | Marlow | blop: was hard to get by in the start .. you had to punch the resellers real hard to get one .. |
| 19:03.16 | blop | :) |
| 19:03.25 | *** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3771041.sympatico.ca) |
| 19:03.26 | brad_mssw | they'd be better if they had a built in 2port switch, so you could daisy chain it to a local workstation ... but ... |
| 19:03.30 | jsolares | what advantages does the new black version have over the blue/orange? or it's just to shave costs? |
| 19:03.31 | DaLion | <PROTECTED> |
| 19:03.31 | DaLion | but after a couple of hours I can not recieve a incoming call until I make a |
| 19:03.31 | DaLion | outgoing call first. |
| 19:03.39 | DaLion | do i need a qualify in sip.conf ? |
| 19:03.52 | DaLion | seems it unregisters it self |
| 19:03.57 | Marlow | jsolares: the blinking light for voicemail :) |
| 19:04.06 | blop | i bought one just before the end of stocks, i think the b&o version is more beautiful than the black one :) |
| 19:04.15 | blop | (excuse my poor english :p) |
| 19:04.31 | jsolares | :o, that's neat, i'm waiting for the day it'll support g729 since bw is quite expensive here :\ |
| 19:04.47 | jsolares | i do have 5 for testing tho :) |
| 19:04.47 | *** join/#asterisk jeffik (~jeffik@69.158.21.177) |
| 19:05.03 | jeffik | Hello all, anyone using sixtel? |
| 19:05.26 | DaLion | anyone ? |
| 19:05.27 | Marlow | jsolares: that's the problem .. i don't think the IAXy will get that .. |
| 19:05.52 | jsolares | :( |
| 19:05.56 | Marlow | jsolares: if it had the power to do the transcoding, they would have done it quite a while ago |
| 19:06.04 | Marlow | jsolares: just my personal opinion .. |
| 19:06.35 | jsolares | i was hoping they might've upgraded the "cpu" in the black version |
| 19:06.38 | Marlow | jsolares: that's one of the reasons why i'm not buying more iaxy's .. |
| 19:06.40 | blop | :p |
| 19:06.43 | Marlow | jsolares: maybe .. maybe not .. |
| 19:06.47 | jsolares | i guess the atcom might do |
| 19:07.27 | blop | iaxy only support IAX1 isnt it ? |
| 19:07.32 | *** part/#asterisk P-Chan (~jpfingstm@68.142.66.200) |
| 19:07.47 | Druken | do the iaxy's work good?, i've personally only used the linksys equipment |
| 19:07.50 | Marlow | blop: nope .. iax2 |
| 19:08.01 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 19:08.04 | blop | hum cool :) |
| 19:08.10 | *** join/#asterisk mager_ (~mager@ua-83-227-134-28.cust.bredbandsbolaget.se) |
| 19:08.36 | Marlow | Druken: they are only use for in office use .. |
| 19:08.48 | Marlow | Druken: because they only support ulaw and adpcm |
| 19:08.57 | Marlow | Druken: 64k codecs .. |
| 19:09.03 | Druken | that's fine... no biggie |
| 19:09.14 | jsolares | that is my only problem with it |
| 19:09.24 | blop | it works fine, but does heat a lot |
| 19:09.26 | Druken | i was just wondering what the quality of service on them was |
| 19:09.46 | *** join/#asterisk bprice20 (~brandon@Unassigned-216.120.255.29.hrwebservices.net) |
| 19:09.51 | Druken | i'm used the the linksys pap2 and rt31p2 |
| 19:09.54 | Marlow | Druken: then they usually do, what you want .. you provision them from inside asterisk |
| 19:10.00 | *** join/#asterisk implicit (~bayan@209.80.0.43) |
| 19:10.15 | *** part/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 19:10.57 | puppet | bkw_: is it maybe me that needs update? ;P |
| 19:11.45 | blop | RT31P2 == 2fxs ? |
| 19:11.58 | *** join/#asterisk nn (~anonymous@ip-wv-68-119-133-020.charterwv.net) |
| 19:12.36 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 19:12.42 | Druken | yeah, rt31p2 == router for 3 + 2 FXS |
| 19:14.01 | blop | and rt31p2 = pap2 + router capabilities and 3ports switch? |
| 19:14.35 | *** join/#asterisk sergiovel (~sergio@200.68.89.177) |
| 19:14.37 | shido | the pap2 is kinda nice looking |
| 19:14.47 | Druken | shido: i love them :) |
| 19:14.49 | shido | mine is nat'd and works great |
| 19:15.03 | shido | so now Im going to sell them |
| 19:15.05 | sergiovel | hello everyone |
| 19:15.08 | blop | pap2 looks great , but its SIP only :-( |
| 19:15.15 | shido | sip is fine |
| 19:15.19 | tzafrir_laptop | anybody managed to confiugre pap2 from lynx? |
| 19:15.25 | Druken | shido: what you going to sell them for?? :) |
| 19:15.30 | tzafrir_laptop | it seems to require javascript |
| 19:15.45 | *** join/#asterisk fugitivo (~ajf@168.226.247.166) |
| 19:15.47 | fugitivo | hello |
| 19:15.56 | tzafrir_laptop | hello -m |
| 19:17.41 | sergiovel | hello guys, I have a quick question: I have a x100p one port fxo card connected to asterisk. the box sees it fine and so does asterisk. The thing is when I plug an extention from my pbx the card opens the port... |
| 19:17.51 | sergiovel | so when I call the extension the asterisk is busy |
| 19:17.56 | sergiovel | anyone seen that? |
| 19:19.38 | *** join/#asterisk aeinet (~jcorgan@64-142-68-61.dsl.static.sonic.net) |
| 19:19.38 | fugitivo | did you try connecting the line directly to the x100p? |
| 19:19.53 | sergiovel | yes |
| 19:19.58 | sergiovel | same thing |
| 19:19.58 | *** part/#asterisk AndrewKT (~mirc@nc-69-68-178-225.sta.sprint-hsd.net) |
| 19:20.14 | fugitivo | try |
| 19:20.16 | Marlow | sergiovel: what if the cable is connected and the box switched off ? |
| 19:20.21 | Marlow | sergiovel: still busy ? |
| 19:20.29 | sergiovel | yes |
| 19:20.36 | sergiovel | hi marlow |
| 19:20.37 | Marlow | sergiovel: yep .. seen it .. |
| 19:20.45 | fugitivo | do you see any error with asterisk -rvvvvvvvvv? |
| 19:20.50 | Marlow | sergiovel: and trust me .. it's not your X100P .. |
| 19:21.00 | Marlow | sergiovel: find another landline it works :) |
| 19:21.24 | sergiovel | it is weird |
| 19:21.24 | Marlow | sergiovel: had the same problem .. unfortunatly never found the reason really .. |
| 19:21.25 | sergiovel | caused i remember having the problem |
| 19:21.30 | Marlow | sergiovel: i came from a place in Dublin .. box worked ... moved to Galway .. box didn't work .. |
| 19:21.32 | sergiovel | and solving it some months ago |
| 19:21.41 | sergiovel | cant remember what i did :-) |
| 19:21.46 | Marlow | sergiovel: got upgraded to ISDN .. x100p in pstn a/b .. works .. |
| 19:21.56 | Marlow | sergiovel: without changes |
| 19:22.30 | *** join/#asterisk Lee__ (~Lee__@cpe-69-203-211-144.nyc.res.rr.com) |
| 19:22.46 | fugitivo | what's the difference between a switch with qos and one without wos? |
| 19:22.48 | fugitivo | qos |
| 19:22.49 | sergiovel | so you think it is the line |
| 19:23.02 | sergiovel | maybe the card doesn like the tone from my panasonic |
| 19:23.03 | Marlow | no no .. i know it's the line . |
| 19:23.12 | sergiovel | hmm |
| 19:23.17 | Marlow | because the line is also busy, when the machine is OFF |
| 19:23.18 | sergiovel | I have another pbx |
| 19:23.21 | sergiovel | i will try that one |
| 19:23.26 | Marlow | yep .. |
| 19:23.43 | Marlow | i tried to cable the x100p into a fxs port on a tdm .. worked |
| 19:23.50 | sergiovel | ok, i never tried with the box off |
| 19:23.51 | Marlow | just not with that particular landline .. |
| 19:23.58 | Marlow | sergiovel: try it .. |
| 19:24.03 | sergiovel | thanks |
| 19:24.05 | Marlow | sergiovel: if so .. it's the line .. :) |
| 19:24.13 | sergiovel | right |
| 19:24.16 | Marlow | sergiovel: and other kit might work .. |
| 19:24.27 | Marlow | sergiovel: but x100p .. simply not .. |
| 19:24.33 | sergiovel | amazing |
| 19:24.41 | Marlow | indeed .. |
| 19:24.44 | fugitivo | sergiovel: where are you located? |
| 19:24.45 | Marlow | unfortunatly .. |
| 19:24.47 | sergiovel | maybe that is what I did, i used the line directly from pstn |
| 19:24.51 | sergiovel | earlier |
| 19:24.55 | sergiovel | in buenos aires |
| 19:24.57 | sergiovel | argentina |
| 19:25.07 | fugitivo | sergiovel: i'm in buenos aires too, working with x100p perfectly |
| 19:25.15 | sergiovel | really? |
| 19:25.19 | fugitivo | yes |
| 19:25.24 | sergiovel | do you have it connected to a pbx? |
| 19:25.33 | fugitivo | no, directly to the phone line |
| 19:25.46 | sergiovel | ok, i will try that...but i dont like that setup |
| 19:25.56 | fugitivo | but a friend is doing that, and it works |
| 19:25.58 | sergiovel | i want to be able to use the pbx with * |
| 19:25.59 | Marlow | sergiovel: try it anyway |
| 19:26.12 | sergiovel | I will. |
| 19:26.19 | sergiovel | what i really want: |
| 19:26.25 | fugitivo | sergiovel: i only had hungup problems, but it's solved, some echo the first 5 seconds of the call |
| 19:26.33 | Marlow | sergiovel: sure .. but it seems to be a compatibility problem sometimes . |
| 19:26.42 | sergiovel | is to have the lines going into the panasonic and forwarded to a couple of extension of the pbx |
| 19:26.51 | sergiovel | them into the 100 card |
| 19:26.55 | sergiovel | cards |
| 19:27.13 | *** join/#asterisk lunchbox08 (~geoff@66-193-73-162.gen.twtelecom.net) |
| 19:27.14 | PBXtech | anyone use that DIAX dialer? |
| 19:27.18 | lunchbox08 | Hello |
| 19:27.20 | fugitivo | sergiovel: my friend is using a panasonic pbx too, aren't you my friend? ;) |
| 19:27.31 | sergiovel | yes, i have it too |
| 19:27.47 | sergiovel | I have a lot of experience with panasonics, but this one is not working |
| 19:27.55 | lunchbox08 | I am having some issues with SIP Register / Inbound Calls - anyone want to lend a hand? |
| 19:28.09 | sergiovel | i installed asterisk with panasonics that have inband dtmf |
| 19:28.10 | blop | [21:26:00] <sergiovel> i want to be able to use the pbx with * => put the pbx behind asterisk :) |
| 19:28.16 | sergiovel | and it works nice |
| 19:28.20 | blop | thats what i'm doing here |
| 19:28.20 | sergiovel | with digium cards |
| 19:28.44 | sergiovel | you can do that blop |
| 19:29.00 | PBXtech | trash the panasonic |
| 19:29.17 | sergiovel | i know, pbxtech, thanks for the tip, I will eventually |
| 19:29.31 | Druken | asterisk is my hero |
| 19:29.44 | sergiovel | there has to be something with the tone they spit out |
| 19:29.56 | sergiovel | fugitivo where do you work? |
| 19:30.06 | sergiovel | can we talk over the phone? |
| 19:30.51 | PBXtech | fugitivo is shy |
| 19:31.29 | sergiovel | i know and he lives here in Buenos Aires too ;-) |
| 19:32.23 | sergiovel | pbxtech, do you have problems with diax? |
| 19:32.26 | lunchbox08 | Anyone have any ideas SIP authenicating based on hostname instead of username? |
| 19:32.49 | PBXtech | no im interested in it |
| 19:32.49 | *** join/#asterisk voipguy (~voipguy@196.200.25.253) |
| 19:32.50 | jsolares | it's a tough sell to replace an existing pbx with asterisk, even if the asterisk does everything the previous pbx did... unless they are cheap bastards hehe |
| 19:33.16 | voipguy | hi guys |
| 19:33.19 | PBXtech | cheep bastards wont buy asterisk either, cause it costs money |
| 19:33.28 | voipguy | anyone terminated on teliax? |
| 19:33.40 | sergiovel | hang on |
| 19:33.51 | jsolares | they will if the need to do something that'll cost too much with the existing pbx, but yeah it's a tough sell nonetheless |
| 19:34.12 | sergiovel | last time i used diax it worked fine. |
| 19:34.21 | sergiovel | it is not pretty looking. but it worked |
| 19:34.25 | [TK]D-Fender | Hello all (back from afk) got q question maybe someone could help me with : |
| 19:34.37 | sergiovel | i will be back |
| 19:34.51 | voipguy | i'm shoping around for voip providers terminating on IAX |
| 19:35.10 | Druken | to ? |
| 19:35.13 | lunchbox08 | voipguy I use netlogic.net |
| 19:35.17 | voipguy | and wanted some advice from the community |
| 19:35.23 | [TK]D-Fender | I'm looking to implement a "follow-me" feature from the dialplan where I'd like it to be able to transfer the call to a geiven exten & context. How would I do this? |
| 19:36.10 | Druken | [TK]D-Fender: transfer(ext@context) ?? |
| 19:36.53 | cluecon | TKD: so you want to be able to dial into an extension, enter a username / password that corresponds to a extension a and have all calls forwarded to extension c? |
| 19:37.47 | [TK]D-Fender | Druken : Yeah that looks like what I want, but I didn't see the context implication |
| 19:37.50 | ZX81 | ~ping |
| 19:37.52 | jbot | pong |
| 19:37.58 | [TK]D-Fender | cluecon : more or less. |
| 19:39.41 | cluecon | i think you are stalking me or something. i'm planning on doing something very similar. it is basically a roaming agent with a default extension if the agent is registered. |
| 19:40.13 | tzafrir_laptop | drumkilla, any reason not to take zonedata.c from HEAD and use it for 1.0.8? |
| 19:40.25 | *** join/#asterisk Paul[NOC] (~paul@66.195.243.254) |
| 19:40.55 | drumkilla | tzafrir_laptop: just a general rule not to include new features |
| 19:40.56 | [TK]D-Fender | We must be psychotically linked! |
| 19:41.06 | [TK]D-Fender | ....err psychicly ;) ... err... BOTH! |
| 19:41.49 | tzafrir_laptop | drumkilla, in the worst case, you add new mal-functioning tone-zones. Take a look at the patch |
| 19:42.01 | [TK]D-Fender | Druken : Transfer didn't entirely work :/ |
| 19:42.04 | tzafrir_laptop | diff, I meant |
| 19:42.35 | Druken | entirely? |
| 19:43.01 | [TK]D-Fender | I see the call, but it didn't continue the actions the extension should have triggered |
| 19:43.20 | Paul[NOC] | Questions for you all, Using AgentCallbackLogin and it only requests User (that matches the CallerID so agent is same as extension) |
| 19:43.28 | Paul[NOC] | Any reason it wouldnt step onto Password? |
| 19:43.50 | drumkilla | tzafrir_laptop: did you make a diff for this? |
| 19:44.07 | blop | shido what are u selling ? pap2? |
| 19:44.18 | tzafrir_laptop | I have a diff. post it anywhere? |
| 19:44.46 | *** join/#asterisk bprice20 (~brandon@Unassigned-216.120.255.29.hrwebservices.net) |
| 19:45.01 | tzafrir_laptop | drumkilla, though you should be able to make the same diff with one simple cvs command |
| 19:45.09 | *** join/#asterisk edynaddy (anonymous@202.134.129.237) |
| 19:45.13 | drumkilla | yeah, I guess I don't really need a diff |
| 19:45.15 | edynaddy | Hi |
| 19:45.18 | drumkilla | I could just copy the file over |
| 19:45.24 | edynaddy | Hello experts :) |
| 19:45.49 | edynaddy | Could someone explain what do people mean by DID? Direct Inward Dial? What does it mean to a layman |
| 19:46.00 | *** join/#asterisk danett_ (unknown@rijn068.athome227.wau.nl) |
| 19:46.02 | danett_ | heya. |
| 19:46.05 | Druken | DID's are phone numbers |
| 19:46.11 | danett_ | How can i forward a call? |
| 19:46.34 | drumkilla | tzafrir_laptop: I'll make you a deal ... |
| 19:46.35 | edynaddy | Druken:phone numbers and ... |
| 19:46.42 | drumkilla | tzafrir_laptop: post it to the bug tracker, that way I won't forget |
| 19:46.48 | [TK]D-Fender | Druken : I get - Executing Transfer("SIP5234-259a", "912345@full) in new stack |
| 19:47.05 | [TK]D-Fender | Druken : And it doesn't actually do what it would had I dianled it manually |
| 19:47.08 | drumkilla | tzafrir_laptop: and if you don't mind, we might as well update indications.conf as well |
| 19:47.20 | edynaddy | hey i run an asterisk gateway |
| 19:47.28 | drumkilla | tzafrir_laptop: and add a note in the bug that you've already talked to me about it so nobody closes it out as a new feature |
| 19:47.29 | JerJer | wow me too |
| 19:47.37 | JerJer | edynaddy: ^ |
| 19:47.39 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 19:47.42 | edynaddy | someone just asked me that he is looking for iax termination in my state |
| 19:47.48 | edynaddy | can i make some money out of this? |
| 19:47.49 | Druken | [TK]D-Fender: if that's what you want, then use either a dial(LOCAL/ext@context) or a goto |
| 19:47.55 | danett_ | i would like to transfer a call (incoming on sip) to my mobile phone (Trought my carrier). how would one do this? |
| 19:48.09 | edynaddy | he wants me to send the IPs of my asterisk server |
| 19:48.50 | [TK]D-Fender | Druken : I think Dial is more appropriate. I want them to be able to dial any number the user had access to. Basically so the user can forward to an internal extension or to an external # equally. |
| 19:48.56 | [TK]D-Fender | Druken : Will try now |
| 19:49.16 | *** join/#asterisk tsume (~tsume@tsume.user) |
| 19:50.32 | *** join/#asterisk Patrick^ (~patrickm@pc-0-34.mountaincable.net) |
| 19:50.58 | Blake0PS | Can I have someoen dial IAX2/somedynamictext@my.host and have * catch 'somedynamictext' and do something with it, or do iax contexts have to be static? |
| 19:51.20 | [TK]D-Fender | Druken : Works decent, thanks :) |
| 19:51.35 | [TK]D-Fender | Just need to work the variables a bit. |
| 19:52.32 | Druken | uhmm.... yeah... ok :) |
| 19:52.46 | [TK]D-Fender | To make it user-friendly. I did a dirty test is all...... |
| 19:53.10 | blop | To those having a linksys pap2 running with Asterisk: i've read some information talking about some locked pap2 models, which can only be used with vonage service and cannot be linked with *. Should i try buying one or there's a way to know before? or maybe there's a firmware upgrade or so .. |
| 19:54.29 | [TK]D-Fender | blop : I haven't found any reason to ever touch a PAP2 with whats on the market currently.... Whats your price on it? |
| 19:54.44 | Druken | i don't know anyone that had managed to unlock a vonage pap2 yet |
| 19:56.26 | Druken | blop: the unlocked versions are pap2-na |
| 19:56.50 | blop | [TK]D-Fender the fact is that linksys products are more easily available here in europe, but maybe you've a specific model to buy in place ? i've already looked at digium, virbiage and sipura hardware |
| 19:58.07 | blop | soyo has some nice products too, but its quite unavailable at the moment |
| 19:58.47 | tsume | anyone here in east TN and near Knoxville which needs a job as a phone technician? |
| 20:02.06 | *** join/#asterisk Pazzo (~Pazzo@host130-250.pool8172.interbusiness.it) |
| 20:03.15 | Hmmhesays | hmmm callback with call files, or using cron to run a script that originates from the manager |
| 20:03.20 | Hmmhesays | decisions decisions |
| 20:03.25 | [TK]D-Fender | Sipura should be pretty cheap, but I couldn't confirm for EU. An SPA-2000 runs around 65$US on low |
| 20:04.44 | Hmmhesays | is there any good arguement as to one over the other? |
| 20:05.25 | Hmmhesays | i'll take that as a no |
| 20:05.39 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 20:06.26 | Hmmhesays | well maybe I'll try out both |
| 20:07.12 | *** part/#asterisk nn (~anonymous@ip-wv-68-119-133-020.charterwv.net) |
| 20:08.03 | *** join/#asterisk SimonR (dhcp@cobae1.consultronics.on.ca) |
| 20:08.36 | blop | i'll look at that :) |
| 20:09.46 | blop | also, i found my zap channel was setting CALLERIDNUM and CALLERIDNAME but it should only set NUM coz i'm not receiving the name from my landline provider, |
| 20:10.08 | blop | the fact is, its setting the NUM well, and NAME=NUM in place of leaving NAME unset or empty |
| 20:10.29 | *** part/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
| 20:11.21 | blop | dunno if it should be considered as a bug or whatever :) |
| 20:12.37 | fugitivo | is any list of sounds for asterisk? i need to record all the sounds, and i want to skip listening all the sounds to know what i have to do |
| 20:13.04 | jontow | fugitivo; check the wiki.. there is a good list there. |
| 20:13.06 | shido | "/var/lib/asterisk/sounds" |
| 20:14.12 | fugitivo | shido: i know, but for example, what is vm-Cust1.gsm? i need to listen to it to know that, i want to skip that if it's possible |
| 20:14.17 | fugitivo | jontow: thanks |
| 20:14.41 | *** join/#asterisk clive- (~pirch@rrba-146-109-202.telkomadsl.co.za) |
| 20:15.19 | cluecon | ~cluecon |
| 20:15.24 | jbot | rumour has it, cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
| 20:15.24 | *** join/#asterisk Paul[NOC] (~paul@66.195.243.254) |
| 20:15.32 | Paul[NOC] | Anyway to display things on a SIP/Grandstream phone using Asterisk |
| 20:16.19 | shido | http://www.voip-info.org/wiki-Asterisk+sound+files+additional |
| 20:16.37 | shido | http://www.voip-info.org/wiki-Asterisk+sound+files |
| 20:17.13 | fugitivo | shido: great, thanks |
| 20:18.01 | blop | Paul[NOC] u should be able to display caller id&&name on it |
| 20:21.41 | Paul[NOC] | blop, yea thats no problem |
| 20:21.49 | Paul[NOC] | Its changing to other things such as queue length |
| 20:21.54 | Paul[NOC] | Displaying any variable on it |
| 20:24.07 | blop | dunno :) |
| 20:24.39 | *** join/#asterisk meppl (mephisto@p54AAD9E9.dip.t-dialin.net) |
| 20:30.45 | *** join/#asterisk kkkkk (~jcorgan@64-142-68-61.dsl.static.sonic.net) |
| 20:31.47 | bjohnson | anyone tried using multiple lines to win a radio contest? |
| 20:31.47 | bannerman | I'm having strange fax problems. My setup something like PSTN <-> ZAP <-> (asterisk) <-> ULAW/SIP SPA-1001 <-> FAX. It works great, all my testing in and out was flawless and fast. Occasionally faxes just don't send (transmission error) or receive. |
| 20:32.00 | jeffik | hello all, anyone using sixtel? |
| 20:32.05 | bjohnson | I am |
| 20:32.14 | bjohnson | my dids aren't working anymore |
| 20:32.42 | jeffik | bjohnson: my outgoing is extremely jittery |
| 20:32.43 | *** join/#asterisk meppl (mephisto@p54AAD9E9.dip.t-dialin.net) |
| 20:33.31 | bannerman | Some days it's difficult to get a fax to go out at all. I don't understand how it can sometimes work flawlessly, and other times not work at all. |
| 20:34.21 | blop | fax arent made to be encoded over ip :p |
| 20:34.26 | jeffik | bjohnson: when did you start having problems? |
| 20:34.31 | jsharp | bjohnson: I've thought about it, but never had the time to try it out. |
| 20:34.48 | bannerman | would I be better off using an FXS to connect to my fax machine? |
| 20:34.55 | jsharp | Yes. |
| 20:34.56 | Paul[NOC] | Can anyone provide a example context for the AgentCallBackLogin function |
| 20:35.19 | bannerman | how likely am I to continue to have the same problems with an FXS? :-P |
| 20:35.33 | bannerman | boy, that question is hardly english, sorry |
| 20:35.35 | jeffik | anyone else using sixtel? |
| 20:35.45 | blop | bannerman it should works with a fxs2fxo bridging than with fax over SIP |
| 20:35.51 | blop | +better :p |
| 20:36.06 | bannerman | wow, the answer was even worse |
| 20:36.09 | blop | :) |
| 20:36.11 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 20:36.16 | bannerman | I've found my home. |
| 20:36.34 | Blake0PS | Is it possible to dial(IAX2/somedynamictext@my.pbx.host) and have asterisk catch somedynamictext and do something with it? |
| 20:36.43 | *** join/#asterisk devel (~devel@wiggum.digitalcoven.com) |
| 20:36.58 | bjohnson | jeffik: We just use them for incoming. A few days ago a client called on another line to say they just keep ringing |
| 20:37.11 | bjohnson | jeffik: I filed a report and await a reply |
| 20:37.12 | bannerman | better.. you're insinuating that to make it work like a normal fax machine, I'd want to connect it directly to a phone line? |
| 20:37.20 | bjohnson | while I wait, I look for alternatives |
| 20:38.13 | blop | bannerman not directly, but .. i think the probleme is due to the fact its encoded over sip, using a specifiq codec |
| 20:38.38 | jeffik | bjohnson: I'm going to try tollfreedeal |
| 20:39.02 | jeffik | bjohnson: let me know what you find |
| 20:39.09 | bjohnson | I have CDN DIDs so there isn't as much selection |
| 20:39.22 | jeffik | what city? |
| 20:39.31 | bannerman | hm, maybe lan activity is causing my problem .. after hours when I do my testing it works flawlessly |
| 20:39.31 | *** join/#asterisk mikewho2 (~Schnappi@ip68-105-227-82.lu.dl.cox.net) |
| 20:39.33 | blop | bannerman i think its not the case when u're dialing from an fxs througt a fxo directly on the asterisk box |
| 20:39.40 | jeffik | bjohnson: where are you? |
| 20:39.48 | mikewho2 | to dial an extension, do you just i.e. 'dial 200 for ext 200?' |
| 20:39.52 | mikewho2 | i got 2 phones up and working |
| 20:39.54 | bjohnson | jeffik: right here |
| 20:39.55 | mikewho2 | and im trying to call one another |
| 20:39.58 | jeffik | bjohnson: I'm in Toronto |
| 20:40.00 | bjohnson | jeffik: sitting down |
| 20:40.13 | jeffik | bjohnson: very funny |
| 20:40.15 | bjohnson | I'm in SW Ontario too |
| 20:40.19 | mikewho2 | i wana try and talkt o someone, anyone wana help me with that? |
| 20:40.22 | mikewho2 | i got asterisks running |
| 20:40.26 | bjohnson | my DIDs are for Kitchener and London |
| 20:40.30 | mikewho2 | and 2 7960's as extensions |
| 20:40.38 | jeffik | bjohnson, did you look at Unlimitel? |
| 20:40.56 | bjohnson | I think I did at one time but they didn't have DIDs where I needed them |
| 20:41.03 | *** join/#asterisk meppl (mephisto@p54AAD9E9.dip.t-dialin.net) |
| 20:41.36 | jeffik | bjohnson: I use them, i think they have Kitchner now. they have been good for me |
| 20:41.46 | *** join/#asterisk robin_sz (~robin@adsl.redpoint.org.uk) |
| 20:41.52 | robin_sz | meep? |
| 20:41.55 | blop | bannerman i mean, using zap native bridging of 2 ports, u shouldnt have any loss of information which are causing fax errors depending maybe of the content of the fax itself |
| 20:42.09 | *** join/#asterisk hopper` (~hopper@81.56.188.180) |
| 20:42.21 | bannerman | blop: ok, thanks |
| 20:42.26 | bannerman | is there a cheap single port fxs? |
| 20:42.33 | bannerman | or do I have to get another tdm400 thingy |
| 20:42.36 | bjohnson | jeffik: yeah. They're Kitchener DIDs are from tzanger iirc |
| 20:42.44 | blop | your tdm is already full ? :) |
| 20:42.58 | bjohnson | bannerman: another module for your tdm is cheapest |
| 20:43.02 | hopper` | anyone know how correct sip channels stuck ? |
| 20:43.15 | hopper` | format: unknow(d) |
| 20:43.15 | bjohnson | bannerman: next cheapest is an ATA like a SPA or a PAP2-NA |
| 20:43.36 | mikewho2 | anyone know why i cant call my other extension on *? |
| 20:43.37 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 20:43.45 | blop | yeah but ata means codec .. shouldnt help with fax |
| 20:44.02 | *** join/#asterisk meppl (mephisto@p54AAD9E9.dip.t-dialin.net) |
| 20:44.11 | Paul[NOC] | exten => 700,1,AgentCallbackLogin(|${CALLERIDNUM}@default), Problems? |
| 20:44.20 | hopper` | on a sip show history the last event is: 7. TxResp SIP/2.0 200 OK / 53159 BYE |
| 20:44.22 | SteveL | if i have ztdummy installed why do i not have a /dev/zap directory? |
| 20:44.24 | shmaltz | anybody here has terminal software to connect to an Avaya difinity G3? |
| 20:44.47 | bannerman | My TDM400 is full, I'll need another... that's unfortunate. $125, just to see if it works. |
| 20:44.48 | mikewho2 | when i dial the extension, it just goes too a buzy signal |
| 20:44.51 | Hmmhesays | mikewho2: you farked it up |
| 20:45.21 | blitzrage | where might I go look in the source to determine what the dependencies for each module is? For example, I'm trying to load app_adsiprog.so as the only module, but it just exits with no error, and I'm not sure where to look to determine why its failing. |
| 20:45.35 | mikewho2 | Hmmhesays well, whats the deal with it? |
| 20:45.35 | blop | bannerman u could swap the fax machine with another fxs device u've connected to the tdm, to test :) |
| 20:45.41 | mikewho2 | i wana at least try to talk between lines |
| 20:45.46 | mikewho2 | or if one of you guys will let me call you |
| 20:45.47 | mikewho2 | i wana try it |
| 20:46.10 | puppet | can people get into the conferense |
| 20:46.22 | mikewho2 | ookay. let me see if i can get meetme working |
| 20:46.22 | puppet | IAX2/guest@switch-3.asterlink.com/996 |
| 20:46.30 | cluecon | we need test subjects to join the conference. |
| 20:46.30 | puppet | need to test some |
| 20:46.31 | *** join/#asterisk KristinG (~KristinG@fozzie.geekgirls.us) |
| 20:46.33 | blop | bannerman: but i can tell you the only time my fax failed was when i tried to send it over iax, and it works well over zap .. :) |
| 20:46.33 | Nethab | hey guys can you all call the conference |
| 20:46.33 | mikewho2 | puppet want me to add that trunk? |
| 20:46.35 | KristinG | good day |
| 20:46.40 | bkw_ | everyone update to CVS-HEAD and call IAX2/guest@switch-3.asterlink.com/996 |
| 20:46.42 | Nethab | we need to test the jitterbuffer under load |
| 20:46.45 | hopper` | SteveL kernel 2.6 ? |
| 20:46.48 | bkw_ | so we can load test the new jitter buffer |
| 20:46.51 | SteveL | yes it is |
| 20:46.51 | Hmmhesays | paste the extension mikewho2 |
| 20:46.54 | bkw_ | just stay muted |
| 20:46.56 | cluecon | ~cluecon |
| 20:46.57 | jbot | hmm... cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
| 20:47.07 | puppet | bkw_: do i need to update to CVS-HEAD? |
| 20:47.07 | KristinG | has anyone managed to get 1.0.7 to build under FreeBSD with freetds? |
| 20:47.10 | mikewho2 | Hmmhesays its just extension 200/202 with my IP |
| 20:47.14 | mikewho2 | i dont even know what ports to forward |
| 20:47.19 | bkw_ | yes so you can have the latest jitterbuffer patches |
| 20:47.24 | Nethab | you need at least a version in the last week or so |
| 20:47.25 | Hmmhesays | paste the extension line out of your extensions.conf |
| 20:47.47 | mikewho2 | Hmmhesays what path is that under |
| 20:47.52 | puppet | ok ill try and fix that fast then running an dooold version :/ been a kind busy :/ |
| 20:47.58 | Hmmhesays | lemme guess you're using asterisk@home |
| 20:48.03 | puppet | dont laugh ;P |
| 20:48.05 | mikewho2 | howd u know! |
| 20:48.06 | puppet | Asterisk 1.0.7 built by puppet@insitu on a i686 running Linux |
| 20:48.16 | Hmmhesays | mikewho2: do yourself a favor and format that drive |
| 20:48.23 | mikewho2 | im just trying to get it to work man |
| 20:48.24 | mikewho2 | just to test it |
| 20:48.26 | jeffik | puppet: i use aah with success |
| 20:48.27 | Hmmhesays | do a clean install |
| 20:48.31 | mikewho2 | what path is extensions.conf in' |
| 20:48.37 | Hmmhesays | and learn how to configure the *.conf files |
| 20:48.42 | puppet | jeffik: aint aah |
| 20:48.43 | mikewho2 | okay, i will |
| 20:48.46 | mikewho2 | where are they loated |
| 20:48.47 | mikewho2 | located |
| 20:48.49 | KristinG | would that be a no then? |
| 20:48.57 | Hmmhesays | that way when asterisk@home farks up the *.conf files you at least have a shot at fixing it |
| 20:49.15 | mikewho2 | Hmmhesays not if i dont know where they are located |
| 20:49.17 | puppet | they are saying the alphabet ;P |
| 20:49.49 | puppet | im uising 1.0.7 and it works great here but i havent spoken so |
| 20:49.50 | Hmmhesays | http://www.voip-info.org/wiki-Asterisk mikewho2 |
| 20:49.51 | jeffik | puppet: sorry |
| 20:49.58 | Hmmhesays | start there, not with *@home |
| 20:50.04 | Hmmhesays | you will thank me |
| 20:50.08 | mikewho2 | thanks manb, that sure did help alot more that just telling me where the conf files are |
| 20:50.23 | Hmmhesays | teach a man to fish.... blah blah blah, you know the rest |
| 20:50.24 | puppet | ok i need to upgrade ;P |
| 20:50.41 | mikewho2 | teach me where the conf files are, and blahblahblah |
| 20:50.53 | KristinG | hello?? |
| 20:51.05 | Hmmhesays | did you have a question? |
| 20:51.09 | KristinG | has anyone managed to get 1.0.7 to build under FreeBSD with freetds? |
| 20:51.10 | hopper` | SteveL you have add zaptel def on 50-udev.rules ? |
| 20:51.48 | Hmmhesays | mikewho2 don't be a lazy bastard http://www.voip-info.org/wiki-Asterisk+config+files |
| 20:52.08 | Hmmhesays | sorry KritinG no joy there for me |
| 20:52.11 | *** join/#asterisk CoffeeIV (~rristroph@mail.airlinksystems.com) |
| 20:52.12 | Hmmhesays | never tried |
| 20:52.38 | CoffeeIV | Is the cheapest IP Phone around still the Budgetone ? |
| 20:52.43 | puppet | where do i find a guid for CVS usage? |
| 20:52.45 | KristinG | ok thanks. guess it is time to file a bug report |
| 20:52.51 | SteveL | hopper: no i haven't seen anything about that in the documentation |
| 20:52.51 | puppet | like never used cvs real ;P |
| 20:52.59 | KristinG | seeing that res_odbc needs freetds |
| 20:53.01 | Hmmhesays | KristinG: have you checked the mailing list? |
| 20:53.14 | KristinG | yes, it is usually useless for info |
| 20:53.28 | Hmmhesays | there's a lot of good stuff on there |
| 20:53.29 | mikewho2 | Hmmhesays now what do we need from my extensions.conf? |
| 20:53.33 | *** part/#asterisk Blake0PS (~blake@blakeops.com) |
| 20:53.33 | KristinG | mainly n00bs whining about configs |
| 20:53.51 | hopper` | SteveL http://asterisk.espia-net.net/horde/chora/co.php/zaptel/README.udev?r=1.1&asterisksess=8d11c59df30a14d8122324fe62458250 |
| 20:53.53 | *** join/#asterisk Blake0PS (~blake@blakeops.com) |
| 20:54.04 | Hmmhesays | I was serious about formatting that drive mikewho2 |
| 20:54.07 | KristinG | the wiki is usually rather helpful |
| 20:54.16 | KristinG | if you use insecure linux that is :D |
| 20:54.24 | CoffeeIV | puppet: search on "Pers Cederqvist" and "CVS Manual". If you read the first 25 pages or so you will know enough. |
| 20:54.30 | mikewho2 | seems like i twould be easier to do a test call now than format my drive and reinstall/compile everything |
| 20:54.33 | puppet | coffeeiv: ^^ |
| 20:54.53 | puppet | cederqvist sounds swedish |
| 20:55.00 | Hmmhesays | mikewho2 you want a simple test call? |
| 20:55.04 | mikewho2 | yes |
| 20:55.08 | Hmmhesays | paypal me 20 bucks and i'll set it up for you |
| 20:55.09 | mikewho2 | exactamundo |
| 20:55.18 | mikewho2 | pissoff |
| 20:55.30 | mikewho2 | i dont have that kind of money! |
| 20:55.33 | Hmmhesays | hey I don't see anyone else in here paying any attention |
| 20:55.33 | mikewho2 | im using opensource! |
| 20:55.47 | puppet | opensource != free |
| 20:55.50 | puppet | ;P |
| 20:55.53 | Hmmhesays | i suggest you log onto the asterisk cli and see wtf is going on |
| 20:55.54 | puppet | www.google.com |
| 20:56.01 | CoffeeIV | puppet: for a shorter/faster guide, you can look at my notes here: ale.freeshell.org/articles/cvs-getting-started/ (Note the server seems down at the moment, it will hopefully be back soon) |
| 20:56.03 | puppet | www.voip-info.org |
| 20:56.12 | puppet | coffeeiv: ok nice, thanks :) |
| 20:56.38 | puppet | mikewho2: www.voip-info.org / www.asterisk.org / www.google.com / www... etc :) |
| 20:56.45 | puppet | mikewho2: there is free knowledge :) |
| 20:57.09 | Hmmhesays | we don't give out information that's easily attainable on the wiki here for free |
| 20:57.33 | puppet | ~wiki |
| 20:57.37 | mikewho2 | man, im starving |
| 20:57.47 | mikewho2 | imma go grab something to eat. |
| 20:57.54 | mikewho2 | asterisks is a nuts program, much to learn |
| 20:57.55 | Hmmhesays | food is good |
| 20:58.02 | mikewho2 | way too many conf files |
| 20:58.04 | Hmmhesays | mikewho2: you need to move to SER then |
| 20:58.08 | mikewho2 | ser? |
| 20:58.11 | Hmmhesays | xD |
| 20:58.13 | Nethab | i don't use but 7 of them |
| 20:58.18 | Nethab | i removed the rest |
| 20:58.22 | mikewho2 | really? |
| 20:58.24 | Hmmhesays | asterisk is childs play for config man |
| 20:58.24 | Nethab | yeah |
| 20:58.30 | Hmmhesays | it's the cats meow |
| 20:58.31 | mikewho2 | i mean, even the sipmac.cnf is a beotch |
| 20:58.31 | Hmmhesays | lol |
| 20:58.33 | Nethab | but i don't use zap |
| 20:58.42 | mikewho2 | no zaptel? |
| 20:58.50 | Nethab | no i am pure ip |
| 20:58.53 | Hmmhesays | mikewho2: if you do what I told you to do. I'd consider giving you some help |
| 20:59.04 | mikewho2 | i cant just format this drive bro |
| 20:59.06 | jeremywhiting | hi all |
| 20:59.15 | Hmmhesays | remove asterisk then |
| 20:59.18 | Hmmhesays | and download cvs-head |
| 20:59.22 | jeremywhiting | anyone here know the default username/password for http setup on polycom phones? |
| 20:59.23 | Hmmhesays | compile it. |
| 20:59.35 | Nethab | admin/4567 |
| 20:59.36 | mikewho2 | ill graba bite first. |
| 20:59.39 | Nethab | er 456 |
| 20:59.39 | mikewho2 | thx for offering tho |
| 20:59.47 | Hmmhesays | make sure you clean out /etc/asterisk |
| 20:59.54 | Nethab | jeremywhiting: polycom password is 456 |
| 21:00.09 | Hmmhesays | asterisk@home is not for n00bies in reality |
| 21:00.30 | jeremywhiting | Nethab: for web access when it asks for username/password? |
| 21:00.39 | blop | *@home just rox ! :) |
| 21:00.46 | Nethab | admin/456 |
| 21:00.48 | Hmmhesays | if you know what you're doing already it's pretty good |
| 21:01.08 | jeremywhiting | tried that, maybe just cant access it from links or something |
| 21:01.20 | Hmmhesays | if you have no idea what's going on there's no hope of you ever fixing a problem... other than mucking it up more with endless point and click |
| 21:01.33 | Nethab | let me login to mine |
| 21:02.49 | bannerman | blop: thanks fro the help |
| 21:03.01 | shido | anything wrong with cvs? |
| 21:03.17 | blop | bannerman u're welcome |
| 21:03.28 | shido | @home has brought me a ton of students |
| 21:03.59 | jeremywhiting | Nethab: I've tried the ftp username/password, and the phone's sip username/password, but to no avail |
| 21:04.02 | hopper` | anyone know where find the new firmware for grandstrean bt100 http://www.grandstream.com/BETATEST/ not working now |
| 21:04.06 | jeremywhiting | admin/456 didn't work either |
| 21:04.31 | puppet | hmm i cant login to the cvs |
| 21:04.33 | Nethab | polycom/456 |
| 21:04.49 | Nethab | checked the manual |
| 21:05.18 | jeremywhiting | oh, cool, thanks |
| 21:06.01 | *** join/#asterisk trizzo (~troy@money.doogles.com) |
| 21:06.34 | jeremywhiting | nope, still no dice |
| 21:06.35 | SteveL | hopper: thanks for that link...that's exactly what i needed. It's working great now. |
| 21:06.37 | Nethab | capital P it says |
| 21:06.46 | jeremywhiting | oh, that's probably my mistake |
| 21:07.03 | puppet | cvs [login aborted]: connect to cvs.digium.com(66.250.69.240):2401 failed: Connection refused |
| 21:07.17 | trizzo | anyone know if jerjer is around? |
| 21:07.21 | bkw_ | why? |
| 21:07.36 | trizzo | need some assistance |
| 21:07.39 | MikeJ[Laptop] | yes, jerjer is around |
| 21:07.56 | Nethab | round, i thought he was square |
| 21:08.09 | mikewho2 | can someone give me a url to make a testcall too? |
| 21:08.18 | CoffeeIV | can asterisk send an outgoing fax ? I mean from itself, not from a fax machine hooked up to an extension |
| 21:08.35 | Hmmhesays | spandsp |
| 21:08.35 | Nethab | IAX2/guest@switch-3.asterlink.com/996 |
| 21:08.47 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 21:08.50 | mikewho2 | Nethab do i just type that in my phone to dial? |
| 21:08.53 | mikewho2 | as a url? |
| 21:09.02 | Nethab | it's a dial string in asterisk |
| 21:09.12 | Nethab | does it support iax? |
| 21:09.15 | Hmmhesays | mikewho2 you are going to irritate a lot of people using that |
| 21:09.25 | Hmmhesays | ;) |
| 21:09.29 | mikewho2 | soooo... how do i use a dialstring? |
| 21:09.32 | Hmmhesays | that said.... |
| 21:09.35 | mikewho2 | add it as a trunk? |
| 21:09.38 | mikewho2 | or put it in the configs? |
| 21:09.41 | Hmmhesays | erase that *@home |
| 21:09.42 | Nethab | no, as a Dial |
| 21:09.50 | puppet | mikewho2: www.voip-info.org |
| 21:09.53 | puppet | mikewho2: start reading there |
| 21:09.56 | puppet | mikewho2: PLEASE |
| 21:10.08 | Nethab | if you want a url try sip:613@fwd.pulver.com |
| 21:10.16 | Hmmhesays | mikewho2: rm /etc/asterisk/* |
| 21:10.20 | blop | :D |
| 21:10.28 | Hmmhesays | seriously |
| 21:10.32 | Hmmhesays | get ride of those config files |
| 21:10.41 | *** join/#asterisk stkn (nobody@stkn.developer.gentoo) |
| 21:11.02 | Hmmhesays | get cvs head, compile it, and generate the bare configs |
| 21:11.06 | Hmmhesays | it takes 10 minutes |
| 21:11.45 | blop | its longer reading the docs than configuring asterisk itself :) |
| 21:12.11 | Hmmhesays | the base config files are heavily commented |
| 21:12.11 | Nethab | i thought that was cause your illiterate |
| 21:12.24 | Nuxi | wiki == wandom information kontained in-here. |
| 21:12.52 | Nethab | wanton idiots keep iquiring |
| 21:12.53 | Hmmhesays | even better if you are using *@home rm -r /etc/asterisk |
| 21:13.12 | bannerman | So much hate. |
| 21:13.15 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 21:13.25 | Nuxi | Nethab, needs some bad spelling and grammer. |
| 21:13.37 | Nethab | you mean speling and gammer |
| 21:13.39 | *** join/#asterisk bprice20 (~brandon@Unassigned-216.120.255.29.hrwebservices.net) |
| 21:14.23 | Hmmhesays | i thought schpelen had a 'ch' in it |
| 21:14.24 | bannerman | Echo cancellation is supposed to be disabled while transmitting a fax, right? |
| 21:14.33 | Nethab | yes |
| 21:15.07 | Nuxi | otherwise you get an inverted echo. |
| 21:15.28 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 21:15.48 | Hmmhesays | mix that with a little acid and you got one helluva trip |
| 21:16.02 | *** join/#asterisk DiAbLe666 (diable@lynux.xdsl.openweb.be) |
| 21:16.41 | *** join/#asterisk Connor- (~billy@198-144-174-5.knx.tn.nxs.net) |
| 21:16.58 | Connor- | Hey guys, is there a way to monitor how long someone has been on hold in the cdr's or anything? |
| 21:17.16 | *** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 21:17.20 | Nuxi | pick up the line and ask them. ;) |
| 21:17.27 | Danett | how can i check if asterisk registere correctly with a sip server? |
| 21:17.58 | Hmmhesays | pick up a phone and make a call? |
| 21:18.08 | Danett | works correctly |
| 21:18.10 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 21:18.16 | bannerman | I checked "zap show channel 2" from the CLI while sending a fax (successfully) and echo cancellation was ON the entire time. |
| 21:18.18 | *** join/#asterisk Champi (Champi@ogmios.iplab.info) |
| 21:18.30 | Danett | i have an id at a service provider. don't seem to be able to call it |
| 21:18.32 | jeremywhiting | Hi all, getting strange errorhi all, anyone know what 302 Moved Temporarily means in a sip debug message? |
| 21:18.34 | Danett | (softphone) |
| 21:18.37 | Nethab | did the fax go through? |
| 21:18.40 | bannerman | yes. |
| 21:18.49 | Nethab | then you don't have a problem |
| 21:19.05 | bannerman | it's just the other 5 out of 10 faxes that don't go through that cause the problem. |
| 21:19.06 | Hmmhesays | moved temporarily means you ain't getting your money |
| 21:19.37 | jeremywhiting | Hmmhesays: the phone is online, I can connect to it and everything, don't know why I get this message |
| 21:20.02 | Hmmhesays | that was a joke, I have no idea what the cause of that error would be |
| 21:20.05 | jeremywhiting | I can ping it, open it's web config page, just can't call it |
| 21:20.07 | jeremywhiting | oh, ok |
| 21:20.10 | jeremywhiting | thanks anyway |
| 21:20.18 | bannerman | I wonder if ugprading to CVS-HEAD would improve things. |
| 21:20.24 | blop | is echocancel enabled or disabled by default in zapata.conf ? |
| 21:20.24 | Hmmhesays | if you don't laugh in IT your head will explode |
| 21:20.26 | Nethab | it always does |
| 21:20.32 | bannerman | blop: enabled |
| 21:20.37 | blop | k |
| 21:21.14 | jeremywhiting | that's true |
| 21:21.22 | Hmmhesays | jeremywhiting: the mailing list has some info on 302 errors |
| 21:21.48 | jeremywhiting | and the console error is saying 'No such extension/context levi@ourphones creating local channel' |
| 21:21.53 | jeremywhiting | on voip-info.org? |
| 21:22.00 | Hmmhesays | no the mailing list |
| 21:22.24 | Hmmhesays | http://www.digium.com/index.php?menu=mailing_list search at the bottom, or subscribe |
| 21:22.54 | *** join/#asterisk file[mac] (~jcolp@mctn1-142166196157.nb.aliant.net) |
| 21:26.13 | bannerman | the 'Fax Handled' thing in zap show channel # only applies to inbound calls, correct? |
| 21:27.40 | blop | bannerman u can configure on incoming, outgoing, both, or none |
| 21:27.45 | blop | look at zapata.conf |
| 21:27.46 | bannerman | gotcha, I just found that too |
| 21:27.46 | bannerman | thanks |
| 21:27.52 | blop | :) |
| 21:30.12 | robin_sz | ick. well I found out what crashed my Grandstream ... dodgy ethernet cable. |
| 21:30.44 | drumkilla | that's impressive that a cable crashed the phone |
| 21:30.55 | robin_sz | the interface must rely heavily on the embedded processor I guess |
| 21:31.16 | drumkilla | ~thwack file[mac] |
| 21:31.18 | jbot | ACTION beats file[mac] on the foot with a UNIX Manual |
| 21:31.40 | cluecon | ~frag drumkilla |
| 21:31.41 | jbot | ACTION readies the nuke launcher and fires some rounds at drumkilla |
| 21:31.44 | dsmouse | ~thwack himself |
| 21:31.45 | jbot | ACTION smacks himself on the eye with a Cisco Manual |
| 21:31.53 | cluecon | ~cluecon |
| 21:31.55 | jbot | well, cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
| 21:32.47 | *** join/#asterisk rephorm (~brian@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 21:33.05 | robin_sz | cluecon: can I take a wild WILD guess that you learnt your oh-so-subtle advertising techniques from a group of viagra salesmen? |
| 21:33.50 | cluecon | robin: i have a sign posted that warns all viagra salesmen that they will be shot on site. |
| 21:33.59 | cluecon | er...sight even. |
| 21:34.01 | robin_sz | ugh |
| 21:34.05 | robin_sz | better |
| 21:34.33 | cluecon | it's been a long day. far too long. |
| 21:34.36 | robin_sz | can I do my lose/loose rant? |
| 21:34.38 | *** join/#asterisk jdb1968 (~jdb1968@S0106000f3d016dd2.cg.shawcable.net) |
| 21:34.46 | openfly | ? |
| 21:34.58 | openfly | okay i figure i will ask one more time.. |
| 21:35.07 | openfly | does anyone have any spare cisco power cubes? |
| 21:35.09 | cluecon | robin: go for it. the conf call appears to have ended and well, i could use the entertainment. |
| 21:35.09 | openfly | =/ |
| 21:35.13 | jdb1968 | anyone know when Digium's cvs server will be back up |
| 21:35.28 | *** join/#asterisk claude2005 (~claude@kcassidy.plus.com) |
| 21:35.46 | puppet | wtb cvs server |
| 21:36.04 | robin_sz | cluecon: its not particularly entertaining .. people mix up site and sight are almost as annoying as thois who mix up lose and loose ... or worse, brake and break. |
| 21:36.23 | Danett | hmm. I am able to make calls trought my provider, however, sip show registry shows 0 |
| 21:36.30 | Danett | isn't that a little weird |
| 21:36.46 | Nethab | you can make calls while unregisterd |
| 21:36.50 | Nethab | but not recieve them |
| 21:37.11 | Nethab | registration is just to tell them where to send the calls to you |
| 21:37.15 | cluecon | robin: as i stated, long day. normally i'm not loose with my lingo and very rarely do i apply the break in order to avoid braking the garage door. |
| 21:37.16 | robin_sz | Nethab: presumably only in default context? |
| 21:37.19 | Danett | hmm |
| 21:37.19 | jeremywhiting | hi all, having trouble with one phone out of 5, can't dial it, keep getting their voicemail |
| 21:37.33 | jeremywhiting | asterisk is saying 'No such extension/context' |
| 21:37.34 | *** part/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 21:37.40 | Danett | so everyone who knows my sip gateway ip can make calls? even when they are not allowed to? |
| 21:37.42 | Hmmhesays | typo |
| 21:37.56 | Hmmhesays | check your syntax |
| 21:38.02 | jeremywhiting | and 'Unable to create Local channel for call forward to ...' |
| 21:38.03 | Nethab | the incoming context is defined either in the type = peer definition, or in the [general] section |
| 21:38.13 | robin_sz | danett; only of you allow outgoing calls from yor defualt cotnext I suspect |
| 21:38.55 | *** join/#asterisk jas_williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com) |
| 21:39.08 | JerJer | jeremywhiting: create a extension/context then |
| 21:39.20 | jeremywhiting | it says trying to create channel levi@ouroffice, but there is no extension levi only levi1 and levi2 |
| 21:39.33 | jeremywhiting | don't know where asterisk is getting the name levi with no line numbers |
| 21:39.46 | jeremywhiting | all the other sip accounts and phones work fine the same way |
| 21:39.52 | Nethab | danett: with IAX if you don't have a guest account in iax.conf you won't get anonymous calls |
| 21:40.08 | Nethab | everyone will have to login |
| 21:40.43 | Danett | ah |
| 21:40.44 | Danett | i understand |
| 21:40.50 | Danett | still the register problem exists |
| 21:40.56 | Danett | i have |
| 21:41.07 | Danett | register => user:pass@domain/number |
| 21:41.09 | *** join/#asterisk TrevorSHarrison (~trevorsha@24.49.36.218) |
| 21:41.31 | Danett | i am able to make calls on MY account, but no register status |
| 21:41.49 | Nethab | you will be able to make calls even if unregistered |
| 21:41.54 | openfly | man... netgear fs108p switches are too smart for their own good. |
| 21:42.02 | jeremywhiting | sip debug shows 302 Moved Temporarily and |
| 21:42.11 | Nethab | that means your username/password in the type= peer section is correct |
| 21:42.11 | jeremywhiting | Diversion: reason="deflection" |
| 21:42.32 | *** join/#asterisk Marlow (~marlow@159-134-144-30.as1.mvw.galway.eircom.net) |
| 21:42.41 | Danett | Nethab: hmm |
| 21:42.53 | Romik_ | anybody know FXO USB device working with asterisk? |
| 21:42.57 | Danett | i have type=friend |
| 21:43.01 | Danett | to enable inbound |
| 21:43.21 | Nethab | sip or iax |
| 21:43.41 | Danett | sip |
| 21:44.17 | Nethab | with sip i get away with having only a peer entry, friends are only usefull for people that register to You like phones |
| 21:44.27 | jdb1968 | digium has confirmed it's down, waiting for an ETA... |
| 21:44.42 | JerJer | Romik_: i saw new wcusb code get committed - for a new SLIC so there might be something on the horizon |
| 21:44.48 | Danett | Nethab: ok. |
| 21:45.03 | Romik_ | jerjer: what is new SLIC? |
| 21:45.13 | Danett | i register line should be sufficient to receive calls right? |
| 21:45.15 | JerJer | subscriber line interface circuit - i presume |
| 21:45.25 | Nethab | the register line plus a peer entry |
| 21:45.34 | *** join/#asterisk gatty (~agatward@beauchief.plus.com) |
| 21:45.36 | Danett | define peer entry |
| 21:45.39 | JerJer | jdb1968: i'll bite...what's down ? |
| 21:45.45 | jdb1968 | digium's cvs server |
| 21:45.57 | gatty | was just gonna ask about that |
| 21:46.01 | jdb1968 | they say 1/2 hour ETA... cheers |
| 21:46.14 | Nethab | [peer] |
| 21:46.17 | Nethab | type = peer |
| 21:46.37 | Danett | i have it |
| 21:46.49 | Nethab | does the hostname match the register line |
| 21:46.57 | Danett | Nethab: when it's only a peer, do you need the context line then? |
| 21:47.02 | Danett | yes. it does |
| 21:47.07 | Nethab | yes |
| 21:47.20 | Nethab | the context line directs incoming calls to that context |
| 21:47.33 | Danett | ok. that's all good then |
| 21:48.27 | robin_sz | hmmm ... so .. to make an extension such that a user can log into their own mailbox, from their phone, with a single key-press rather than all that "mail-box, ... password" nonsense ... there must be a neat way of doing this?? |
| 21:48.30 | Nethab | does sip show registry show "unregistered" or nothing at all |
| 21:48.45 | Danett | nothing at all |
| 21:48.50 | puppet | robin_sz: stand in the wiki somewhere |
| 21:48.52 | Nethab | then you need to either sip reload |
| 21:48.57 | Nethab | or restart asterisk |
| 21:48.57 | puppet | robin_sz: ive seen it |
| 21:49.06 | gatty | is it ok to mix CVS zaptel with stable (1.0.7) asterisk? |
| 21:49.06 | Danett | did that for like 10 times |
| 21:49.11 | jas_williams | robin_sz: there is but what about security, you should still keep the passsword prompt |
| 21:49.34 | johnnyb | robin_sz: It's not hard. The extension you call from is one of the variables you can use. |
| 21:49.51 | Nethab | is the hostname or username or anything in sip show registry |
| 21:49.52 | robin_sz | jas_williams: nah, this is only for if they already have access to the physical phone |
| 21:49.55 | johnnyb | robin_sz: Just pass in the extension to the VoicemailMain function |
| 21:49.59 | Danett | there is *nothing* |
| 21:50.06 | Danett | i enabled debugging now |
| 21:50.13 | Nethab | then it hasn't read the sip.conf file |
| 21:50.13 | robin_sz | johnnyb: ta |
| 21:50.24 | Nethab | what about sip show users |
| 21:50.30 | cyburdine | is there any command I need to run in order for asterisk to see a unixODBC driver? |
| 21:50.31 | Nethab | or sip show peers |
| 21:50.40 | cyburdine | besides add it to res_odbc.conf? |
| 21:50.49 | Danett | sip show users doesn't give me the voip provider |
| 21:50.51 | jas_williams | robin_sz: what about somebody (security) walking upto the phone and playing ? |
| 21:51.05 | Danett | show peers gives me the provider |
| 21:51.34 | robin_sz | jas_williams: if they are wandering around in our pffices, we have WAY more problems than them getting our voicemail |
| 21:51.52 | Nethab | does sip show registry show the header line "host username refresh" |
| 21:51.56 | Danett | yes |
| 21:52.38 | Nethab | is the register line in the [general] section or the peer section |
| 21:52.40 | cyburdine | I have a feeling that asterisk doesn't know how to talk to my db :( |
| 21:53.01 | robin_sz | jas_williams: and, our current answering machine doesnt have a password ... |
| 21:53.57 | jas_williams | anyway this is a sample line so it connects with no mailbox question exten => 8500,1,VoicemailMain(${CALLERIDNUM}) |
| 21:54.08 | gatty | my university's PBX is crap - doesn't do kewlstart or groundstart and drops back to dialtone at end of call. Is there any way to get the zap channel to detect the call has ended in this situation? |
| 21:54.13 | Danett | erm |
| 21:54.18 | Danett | in none of all? :) |
| 21:54.22 | robin_sz | jas_williams: thats simple, thanks :) |
| 21:54.29 | Danett | i just put it somewhere |
| 21:54.47 | Nethab | put it at the top underneath the [general] header |
| 21:54.55 | jas_williams | 8500 being the voicemail extension |
| 21:54.55 | jas_williams | or |
| 21:54.59 | bkw_ | doh cvs is fixed |
| 21:55.34 | jas_williams | or exten => 8500,1,VoicemailMain(${CALLERIDNUM},s) skips the password prompt as well |
| 21:56.02 | robin_sz | right, I'll consider that carefully. maybe only on *certain* extensions |
| 21:56.05 | robin_sz | thnaks |
| 21:56.20 | puppet | bkw_: why did u break it? :( |
| 21:56.40 | robin_sz | infact, stuff it, they can enter their password, I'll skip it when the whining outweighs the security ;) |
| 21:56.40 | Nethab | because mark took off his cluecon link |
| 21:56.42 | bkw_ | I killed the wrong box |
| 21:56.50 | bkw_ | Nethab, you think i'm that shallow |
| 21:56.51 | bkw_ | haha |
| 21:57.03 | puppet | lol bkw_ ;P |
| 21:57.09 | Nethab | no but that was a very interesting conversation |
| 21:57.36 | robin_sz | bkw_: congrats, now .. learnt o type uname -a *before* reboot |
| 21:57.36 | puppet | is it ok to use same config files for cvs head as for 1.0.7? |
| 21:57.50 | Nethab | i set my prompt with the hostname |
| 21:57.54 | Nethab | much easier |
| 21:58.16 | Nethab | puppet: config file yes, extensions.conf maybe not |
| 21:58.28 | puppet | shit my sister talks in the phone ;P |
| 21:58.35 | puppet | wonder if she will wine if i kill the pbx ;P |
| 21:58.38 | blitzrage | hey, does app_intercom.so load? |
| 21:58.39 | Nethab | soft hangup |
| 21:58.48 | jas_williams | puppet: there are a few changes new options but check for new options in sip.conf and a few other places |
| 21:58.49 | cypromis | blitzrage: app_intercom is dead |
| 21:58.51 | cypromis | and buried |
| 21:58.54 | puppet | ok jas_williams |
| 21:58.59 | blitzrage | haha, then it should be removed from CVS :) |
| 21:59.03 | *** part/#asterisk jamest (~jamest@adsl-208-191-42-201.dsl.tpkaks.swbell.net) |
| 21:59.21 | puppet | and new IAX functions now ;P |
| 21:59.22 | bkw_ | blitzrage, it doesn't compile |
| 21:59.40 | blitzrage | bkw_: its in my /usr/lib/asterisk/modules/ directory... |
| 21:59.48 | file[mac] | bkw_: you make me smile! yes you do! |
| 21:59.49 | puppet | old module maybe? |
| 21:59.51 | puppet | since it was dead? |
| 21:59.53 | bkw_ | remove them |
| 21:59.55 | puppet | since pre-dead |
| 21:59.55 | blitzrage | nope, brand new machine, first time |
| 21:59.57 | puppet | even |
| 21:59.59 | bkw_ | thats why CVS-HEAD tells you to remove them |
| 22:00.08 | blitzrage | bkw_: brand new machine, first time compiled |
| 22:00.45 | blitzrage | no old modules, just downloaded from CVS a couple hours ago |
| 22:00.49 | blitzrage | I'm pretty sure anyways |
| 22:01.22 | blitzrage | actually, yah, the guy installed the machine for me, thus it wasn't used before |
| 22:01.22 | bannerman | do you have to restart asterisk to reload zapata.conf? |
| 22:01.40 | gatty | oh for BRI / PRI termination of my PBX line :( |
| 22:01.51 | MikeJ[Laptop] | blitzrage, what os? |
| 22:02.24 | blitzrage | RHEL3 |
| 22:02.30 | jas_williams | bannerman: you could try reload chan_zap.so works with cvs |
| 22:02.46 | bannerman | jas_williams: thanks |
| 22:02.52 | cyburdine | I'll be somebody's best friend if they'll get me past this DB problem with asterisk realtime! ;) |
| 22:02.57 | gatty | if I have a loopstart (only) capable PBX line that drops back to dialtone when the other end hangs up, is there any way to do hangup detection? |
| 22:02.59 | jeremywhiting | aaaah, this is making me mad, now the one phone won't pick up or even ring still, but I'm getting 302, then 486 sip responses |
| 22:03.11 | jeremywhiting | anyone ever experienced this kind of trouble with polycom phones ever? |
| 22:03.30 | blitzrage | gatty: don't think so - only on koolstart |
| 22:03.40 | robin_sz | eep! ... that weird .. the console has "red alert" messages scrolling past at a million miles an hour ... |
| 22:03.45 | file[mac] | frozen water is good |
| 22:04.08 | *** join/#asterisk Marlow (~marlow@159-134-144-30.as1.mvw.galway.eircom.net) |
| 22:04.14 | jas_williams | robin_sz: what's the message |
| 22:04.22 | gatty | blitzrage: arse... the PBX we have is so old it won't do that - means I get lots of dialtone on the end of voicemail messages. |
| 22:04.54 | blitzrage | gatty: yep, it'll do that :) |
| 22:05.00 | robin_sz | jas_williams: FXO PCI Master Abort |
| 22:05.01 | *** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
| 22:05.09 | blitzrage | gatty: thats the whole thing with koolstart - far end disconnect supervision |
| 22:05.34 | gatty | blitzrage: I thought as much - been about 8 years since I've done "traditional" voice stuff so a bit rusty now |
| 22:05.36 | jas_williams | robin_sz: yuck.... Nasty |
| 22:05.53 | gatty | robin: I've been getting those too with zaptel-1.0.7 and CVS |
| 22:06.09 | blitzrage | gatty: yah, I haven't done much, but I've documented ls, gs and ks enough to have been told that ls doesn't do disconnect supervision |
| 22:06.29 | jas_williams | gatty: get a digital interface for the pbx, |
| 22:06.36 | robin_sz | gatty: this is 1.0.7 on debian |
| 22:06.39 | gatty | blitzrage: kinda sucks cos they've stopped making line cards for the PBX (told you it was old) |
| 22:06.50 | blitzrage | gatty: yah, that does suck :( |
| 22:06.52 | JohnnyC | anyone using Festival ? |
| 22:07.12 | gatty | robin_sz: 1.0.7 on gentoo... but I did put the card in a PCI-X slot so wondered if it might be that |
| 22:07.29 | jas_williams | gatty: Just replace the PBX :) with asterisk |
| 22:07.30 | blitzrage | JohnnyC: nope - <comic_book_guy>worst TTS evar</comic_book_guy> - Use cepstral via a script and the text2speech application |
| 22:07.35 | JohnnyC | I get a strange error not in access list |
| 22:07.37 | robin_sz | gatty: yeah ... this is an "almost the same as digium" card ;) |
| 22:07.43 | danett_ | Nethab : sorry i was disconnected |
| 22:07.46 | danett_ | you still there? |
| 22:08.16 | cyburdine | dectalk is a cool festival replacement as well. |
| 22:08.24 | gatty | jas_williams: would love to... unfortunately its one of those systems that they refuse to update. They can't even source E1 cards for it now :( |
| 22:08.27 | robin_sz | gatty: going to get a real Digium card anyway, so it may just be me being cheap on hardware ;) |
| 22:08.48 | gatty | robin_sz: mine's a real digium card but works fine in other boxes |
| 22:08.54 | *** join/#asterisk danett- (none@dslam7-21-59-81.dyndsl.versatel.nl) |
| 22:08.59 | danett- | fucking versatel |
| 22:09.01 | *** join/#asterisk cluecon[file] (~file@mctn1-142166196157.nb.aliant.net) |
| 22:09.37 | JohnnyC | anyone using Festival ? |
| 22:09.51 | jas_williams | gatty: could you try using the dodgy progress detection and set dial tone as a hangup tone |
| 22:09.56 | fugitivo | try cepstral, it costs $30 and it's much better |
| 22:09.58 | drumkilla | ~[6~[5~[5~[6~[6~[6~[6~[6~[6~[6~ |
| 22:10.04 | jsharp | Bless you. |
| 22:10.28 | *** join/#asterisk likwid-- (~likwid@nc-67-76-110-70.dyn.sprint-hsd.net) |
| 22:10.38 | gatty | jas_williams: could try... will have to wait until tomorrow morning (my time) though, am at home at the mo installing asterisk on what is to be the new server |
| 22:10.47 | danett- | Nethab: ok. put it in [general] it's registred now |
| 22:11.03 | jas_williams | gatty: where are you (on the globe) |
| 22:11.14 | gatty | jas_williams: 30 miles west of London, UK. |
| 22:11.22 | cyburdine | what is the diff between res_odbc.conf and cdr_odbc.conf? |
| 22:11.41 | Nuxi | 3 letters? |
| 22:11.46 | cyburdine | nice |
| 22:11.49 | jas_williams | gatty: reading ? |
| 22:11.57 | puppet | drumkilla: you dont mean that;P |
| 22:11.58 | gatty | jas_williams: yep |
| 22:12.08 | cyburdine | what does res and cdr stand for? |
| 22:12.14 | robin_sz | cyburdine: cdr is prurely related to call logging, res is related to system setup |
| 22:12.21 | cyburdine | gotcha |
| 22:12.24 | danett- | are cdr records recorder by standard? |
| 22:12.37 | puppet | johnnyc: some for testmenus |
| 22:12.44 | jas_williams | gatty: Drop by here tomorrow and I should be here It's time for bed now ZZZ... |
| 22:13.16 | jas_williams | gatty: I'm 20 miles south of Bristol |
| 22:13.35 | gatty | jas_williams: ok - will hopefull have got rid of my FXO PCI Master abort and Failed to initailize DAA, giving up... by tomorrow too with this new hardware ;) |
| 22:13.43 | robin_sz | jas_williams: Bath? |
| 22:14.02 | jas_williams | robin_sz: No Shepton Mallet Near Wells |
| 22:14.06 | gatty | I'd have guessed Taunton |
| 22:14.06 | robin_sz | 'k |
| 22:14.22 | robin_sz | $wife is at the "Bath and West" today ... |
| 22:14.35 | robin_sz | and I'm working. arse. |
| 22:14.53 | jas_williams | she was causing all the traffic problems then |
| 22:15.02 | robin_sz | probably. |
| 22:16.26 | jas_williams | robin_sz: & gatty Check you interrupts cat /proc/interrupts and also make sure you have the latest zaptel drivers |
| 22:16.40 | gatty | I have latest zaptel from CVS and interrupts look fine |
| 22:17.17 | gatty | already checked that one - but as I say, the card's getting moved tomorrow to a box with Intel chipset instead of Serverworks so should stand a better chance of behaving. |
| 22:17.18 | robin_sz | I think I may be using the xorcom ones with the deian release |
| 22:17.19 | robin_sz | hmmm |
| 22:18.26 | jas_williams | gatty: I'm usually here during the working day if you need any ideas. |
| 22:18.34 | robin_sz | I forget *why* I sed the .deb of someone elses zaptel-modules ... oh, wait, it was a ztdummy thing |
| 22:18.44 | danett- | Nethab? |
| 22:18.54 | Nethab | huh |
| 22:18.56 | danett- | ah |
| 22:18.59 | gatty | jas_williams: great, thanks :) Will speak to my colleague who runs the PBX to see if he's got a spec sheet for the line cards. |
| 22:19.01 | danett- | nl.voipgate.nl:5060 4518 105 Registered |
| 22:19.05 | danett- | it shows up now |
| 22:19.08 | Nethab | yay |
| 22:19.15 | danett- | should it also show up in sip show users? |
| 22:19.20 | Nethab | no |
| 22:19.23 | danett- | ok |
| 22:19.27 | puppet | lol alot of stuff that been replaced with set instead of functions |
| 22:19.30 | danett- | thanks for the help |
| 22:19.32 | Nethab | users was just to see if it was reading the sip;.conf at all |
| 22:20.15 | danett- | is it better to put in [general] or in [peer] |
| 22:20.26 | Nethab | you need to put it in general |
| 22:20.35 | Nethab | in the peer it never works |
| 22:20.37 | danett- | ok. very good |
| 22:20.42 | danett- | so i noticed :) |
| 22:21.25 | danett- | when i have an inbound call in the context, will a simple Dial command will be enough to bridge the connections? (forward) |
| 22:22.01 | jas_williams | puppet: I don't loike the set command its far too cryiptic |
| 22:22.54 | Nethab | yes, once the incoming call makes it to a context you Dial to your phone with Dial(SIP/<peer or friend> |
| 22:23.24 | danett- | very good. thank you again :) |
| 22:23.57 | jeremywhiting | hi all, anyone here know what "Diversion: <sip:...>; reason="deflection" is, and how to fix it? |
| 22:24.27 | puppet | <PROTECTED> |
| 22:24.30 | puppet | Jun 3 00:23:35 ERROR[5331]: pbx.c:1365 ast_func_write: Function TIMEOUT not registered |
| 22:24.49 | Nethab | did you load res_functions.so |
| 22:25.15 | Nethab | type show functions |
| 22:25.29 | Druken | ~pastebin |
| 22:25.30 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 22:25.42 | danett- | Nethab: would you like to test the inbound for me from a sip url? |
| 22:25.54 | Nethab | i can try |
| 22:26.02 | puppet | hmm it aint there wait a sec |
| 22:26.04 | danett- | 4518@nl.voipgate.nl |
| 22:27.51 | puppet | res_features haventbeen copiedto libdir |
| 22:28.13 | Nethab | it's in res_functions |
| 22:28.17 | Nethab | not featyres |
| 22:28.26 | Nethab | i get service unavailable |
| 22:28.48 | danett- | to bad |
| 22:28.56 | danett- | i don't see you on my asterisk box also |
| 22:29.11 | puppet | there aint no res_functions.c |
| 22:29.40 | Nethab | pbx_functions.so |
| 22:29.49 | puppet | there it is ;P |
| 22:29.50 | Nethab | stupid naming people |
| 22:30.32 | Nethab | i get a busy signal |
| 22:30.38 | robin_sz | hmmm .. so .. this girl that does the asterisk voice-overs? |
| 22:30.44 | Nethab | allison |
| 22:30.57 | robin_sz | and she does commercial stuff huh? |
| 22:31.34 | robin_sz | ie you send money, she sends back a recording? |
| 22:31.46 | Nethab | yep |
| 22:31.57 | Nethab | you wouldn't believe the stuff they've paid her to say |
| 22:32.01 | danett- | lol |
| 22:32.05 | robin_sz | oh .. I would ;) |
| 22:33.01 | devel | hmmm.... i normally pay people to _not_ say things.... |
| 22:33.50 | Druken | i'm sure she could make good money doing phonesex |
| 22:34.09 | Nethab | "love ya bitch" |
| 22:34.10 | robin_sz | Druken: nah, she faisl the basic test |
| 22:34.12 | danett- | Nethab: with this information: nl.voipgate.nl:5060 4518: you should be able to call me trought a sip url right? |
| 22:34.15 | Druken | make a really dig voice recongnition system with her recordings :) |
| 22:34.30 | Lee__ | what utility is good for munging these gsm encoded .wav files asterisk records? |
| 22:34.40 | robin_sz | Lee__: asterisk |
| 22:34.42 | Nethab | that may be just your authentication username not your full number on their system |
| 22:34.47 | cyburdine | sox |
| 22:34.54 | danett- | hmm |
| 22:34.57 | robin_sz | Lee__: oh, munging, sorry, not * |
| 22:35.00 | danett- | can you try this one? |
| 22:35.20 | danett- | 0707508108@nl.voipgate.nl |
| 22:35.36 | robin_sz | Druken: by law, all phone sex operatives must sound like thay are 18 to 25, but actually be 60 with bad teeth. |
| 22:35.37 | Lee__ | I can put a .wav (gsm encoded) into sox and get a .wav (pcm encoded) out? |
| 22:35.46 | puppet | ok got it up running withcvs head gone check out thenew stuff now |
| 22:35.50 | Lee__ | or even better a raw .gsm out? |
| 22:35.58 | cyburdine | yup it's like the Imagemagick of audio |
| 22:36.05 | Lee__ | excellent. |
| 22:36.31 | Druken | robin_sz: are you a 60 year old with bad teeth ? |
| 22:37.01 | robin_sz | no, im actually an 18 yr old high scolly preppy, I just sound like a 60 yr old with bad teeth |
| 22:37.09 | Nethab | still same |
| 22:37.14 | robin_sz | scolly? ... school. |
| 22:37.38 | puppet | force jutterbuffer right? |
| 22:38.11 | Nethab | jutterbiffer |
| 22:38.19 | puppet | jitterbuffer ;P omg im tired ;D |
| 22:38.24 | Lee__ | word. sox is rad! |
| 22:38.41 | cyburdine | awe yeah... |
| 22:38.48 | puppet | worse would be force jutterbutter! ;P |
| 22:38.49 | bkw_ | the jitterbuffer fixes ROCK |
| 22:39.09 | cyburdine | apparently so is asterisk realtime... but I'll be damned if I can get it working... ;) |
| 22:39.10 | puppet | bkw_: i just enabled it |
| 22:39.34 | Lee__ | cyburdine: that's the advantage of the realtime-ness? |
| 22:39.47 | *** join/#asterisk apardo (~apardo@175.Red-83-44-181.pooles.rima-tde.net) |
| 22:39.56 | Qwell | Lee__: the configs are realtime |
| 22:39.58 | robin_sz | not *that* realtime |
| 22:40.12 | Nethab | no silly storing users and dialplans in a database |
| 22:40.22 | Nethab | and accessing them in real-time |
| 22:40.36 | apardo | hi. anybody knows how register to a sip server with outbound proxy in asterisk ? |
| 22:40.39 | cyburdine | so is there a cookbook with E |
| 22:40.41 | Lee__ | oh, neato. that'll eliminate a lot of SQL programming for us users. |
| 22:40.48 | cyburdine | Z steps to walk ya through setting it up |
| 22:41.52 | cyburdine | voip-info.org just ain't doin it for me... (please don't tell me that's my only hope.. I'll just cry) |
| 22:41.58 | Lee__ | OT: this call recording feature of AMP is pretty rad. |
| 22:42.01 | danett- | Nethab: i found the problem |
| 22:42.26 | blitzrage | cyburdine: I started to document it, and found it to be such a pain in the ass, that I stopped trying |
| 22:42.29 | danett- | since the register is in the [general] part it want to use the context of general to make the call |
| 22:42.38 | Nethab | yes |
| 22:42.48 | danett- | that's kinda stupid, isn't it |
| 22:42.54 | cyburdine | oh man... that sucks to hear |
| 22:42.59 | Qwell | bkw_: You around? |
| 22:43.06 | Qwell | or drumkilla |
| 22:43.08 | blitzrage | danett-: what do you mean? |
| 22:43.17 | cyburdine | this seems SO useful how can it possibly be this hard? |
| 22:43.48 | Juggie | anyone intreasted in voicemail in a database test bug 4403 and comment plz |
| 22:43.51 | cyburdine | how are people building GUIs for this thing? manipulating the confs? |
| 22:44.03 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 22:44.10 | Qwell | Juggie: also people who don't use it, should test it, to make sure it doesn't break anything else |
| 22:44.12 | Lee__ | cyburdine: are you talking about asterisk in general? |
| 22:44.21 | cyburdine | yeah... |
| 22:44.39 | Lee__ | I'm using amportal.sf.net although it's huge bloatware at the moment. |
| 22:44.42 | cyburdine | I am building an ASR app for this, and want to have the DB functionality |
| 22:44.58 | Katty | hmm |
| 22:45.01 | Lee__ | it manages extensions nicely and has some bells and whistles. |
| 22:45.05 | Qwell | guess not..okay, bbl |
| 22:45.08 | puppet | do i have to lod an module for it to call the things in outgoing spool dir? |
| 22:45.09 | cyburdine | so I'm turning to realtime... but man I've spent days on this |
| 22:45.23 | Lee__ | but it dominates apache. |
| 22:45.56 | danett- | in the debug, is see that there is incoming |
| 22:45.59 | cyburdine | so how does it manipulate the configs? |
| 22:46.01 | danett- | Looking for s in unreg |
| 22:46.29 | Lee__ | AMP stores state info in MySQL, uses Perl to parse the configs and php to call the parsing scripts :) |
| 22:46.44 | Juggie | amp sucks |
| 22:46.51 | Juggie | use realtime |
| 22:47.03 | Lee__ | that's what lots say. it may suck but it's functional. |
| 22:47.10 | Katty | hmmmmmmmmmmmmmmmmmmmmmmmmm. |
| 22:47.16 | puppet | no one uses .call files here? |
| 22:47.19 | Lee__ | obviously katty doesn't agree |
| 22:47.37 | Katty | i don't agree with a lot of things |
| 22:47.40 | cyburdine | "use realtime" I'd love to... spent days trying to figure out how... |
| 22:47.43 | Lee__ | hmmmmmmmmmmmmmm |
| 22:48.07 | Juggie | read the wiki |
| 22:48.21 | Lee__ | I'd recommend AMP for someone with a lot of experience setting up web applications with php + mysql + apache |
| 22:48.26 | cyburdine | scoured it... it leads you in circles |
| 22:48.30 | Lee__ | but it's by no means user friendly |
| 22:48.31 | danett- | http://pastebin.com/294234 |
| 22:48.37 | danett- | this is scrolling my screen when debug enabled |
| 22:49.00 | gatty | AMP looks like a sysadmin headache... probably beer-inspired coding. |
| 22:49.28 | Lee__ | gatty: yup. I turned to setting up virtual machines with xen and giving each pbx it's own AMP installation. |
| 22:50.04 | danett- | i have exten => s, blabla in my [unreg] |
| 22:50.20 | danett- | hmm |
| 22:50.23 | puppet | app_queuecall.so hmm |
| 22:50.24 | danett- | hello chad? |
| 22:50.39 | danett- | my mic broke |
| 22:50.41 | Nethab | hello |
| 22:50.45 | Nethab | it worked |
| 22:50.51 | danett- | yes. sounded good :) |
| 22:51.12 | danett- | what did you call? |
| 22:51.15 | gatty | hmm... "make: warning: Clock skew detected. Your build may be incomplete." |
| 22:51.22 | Nethab | your IP address |
| 22:51.54 | danett- | the sip url i assume? |
| 22:51.55 | Nethab | so we know it's not your system per se |
| 22:52.02 | danett- | nope. |
| 22:52.12 | Nethab | s@137. |
| 22:52.30 | danett- | that one is working fine indeed |
| 22:52.32 | puppet | have pbx_outgoing been replaced by anthing? |
| 22:52.36 | Nethab | s@137.224.227.68 |
| 22:52.46 | danett- | yep. that's my asterisk box |
| 22:53.03 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.215) |
| 22:53.19 | danett- | thank you for the testing |
| 22:54.02 | Nethab | oh i see something, in your type= peer |
| 22:54.13 | Nethab | i think you have nat = yes |
| 22:54.18 | danett- | yes i have |
| 22:54.19 | Nethab | you should remove that |
| 22:54.39 | Nethab | nat = yes is only when the other side is behind nat |
| 22:54.52 | Nethab | they are not behind nat you are |
| 22:55.28 | danett- | i am also not behind nat |
| 22:55.41 | Nethab | then you don't need it at all |
| 22:55.49 | danett- | ok |
| 22:56.12 | danett- | All the calls wich use NAT are still in the buffer, is there a way to delete them? |
| 22:56.22 | doolph | anyone can help me to use oh323 as trunk |
| 22:56.24 | Nethab | soft hangup <tab> |
| 22:57.07 | danett- | no active channels : |
| 22:57.09 | danett- | :) |
| 22:57.31 | gatty | does the exten => bit in extensions.conf have to be left-aligned or can I indent it to make readability better? |
| 22:58.13 | Nethab | whitespace makes no differeence |
| 22:58.25 | gatty | marvellous :) |
| 22:58.53 | danett- | Nethab: they are still scrolling of the screen. weird stuff |
| 22:59.04 | danett- | destroying call blabla |
| 22:59.28 | Nethab | those are acknowledgements to voipgate.nl |
| 22:59.32 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 22:59.35 | Nethab | they must be sending packets to you |
| 22:59.38 | Ariel_ | hello everyone |
| 22:59.41 | Nethab | hello |
| 22:59.50 | swk | Anyone using TDM400s seen one keep causing a Dell 2850 to alarm on PCI Parity Errors? |
| 23:00.21 | Ariel_ | swk, I use them in a few dells mainly the SC420's. |
| 23:01.09 | swk | yeah I'm not overly concerned with that alarm as that box is mainly for VM and meetme, and never really has issues |
| 23:01.19 | gatty | swk: out of interest, does that model dell use the serverworks chipset? |
| 23:01.22 | swk | and it seems the parity errors follow the tdm400 |
| 23:01.48 | swk | gatty I think so... its a 2850 Dual HT Xeon box |
| 23:01.54 | Ariel_ | swk, can you set the card in it's own irq |
| 23:02.04 | swk | yeah it has its own irq |
| 23:02.10 | swk | just really usuing it for timing |
| 23:02.19 | Ariel_ | I had to turn of ht on my systems. |
| 23:02.25 | swk | not actually passing a call thru it |
| 23:02.44 | swk | i have 2 more boxes with a similar config right next to that one that never have errors |
| 23:02.44 | danett- | damn. i keep on smoking sigarettes |
| 23:02.53 | danett- | just can't stop :) asterisk is stressing me :P |
| 23:03.00 | swk | danett- : switch too weed then |
| 23:03.06 | swk | it'll calm your nerves |
| 23:03.11 | swk | y0 kram |
| 23:03.17 | danett- | I kinda stopt doing that |
| 23:03.26 | danett- | stoned every night is not o.k. |
| 23:03.43 | swk | TDM400s seen one keep causing a Dell 2850 to alarm on PCI Parity Errors? |
| 23:04.52 | `Sauron | Mmm. |
| 23:04.56 | `Sauron | zap goodness |
| 23:05.46 | *** join/#asterisk twilson (~terry@63.77.68.11) |
| 23:06.19 | swk | if you say so heh |
| 23:07.27 | *** join/#asterisk bkw__ (~brian@adsl-69-154-144-52.dsl.tulsok.swbell.net) |
| 23:07.38 | swk | bkw you whore |
| 23:09.22 | danett- | to bad my local number in holland doesn't work :( |
| 23:09.26 | swk | hah |
| 23:09.27 | danett- | dying to try it |
| 23:09.35 | swk | my local number in beverly hills works |
| 23:10.24 | danett- | when you put a Dial command in a extension, is a Hangup afterwards needed? |
| 23:10.29 | *** join/#asterisk anthm (~anthm@000-447-501.area4.spcsdns.net) |
| 23:10.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:11.45 | *** join/#asterisk iq (~iq@70-59-162-171.omah.qwest.net) |
| 23:14.41 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 23:14.42 | Nethab | no, because asterisk won't be the one hanging up |
| 23:14.45 | Nethab | the user will |
| 23:14.59 | robl^ | seen atacomm |
| 23:15.20 | Ariel_ | ~seen atacomm |
| 23:15.38 | jbot | atacomm <~dan@69.54.45.98> was last seen on IRC in channel #asterisk, 4d 20h 10m 41s ago, saying: 'lol, well that would be the more specific place, but last time i chceked there's alot of talk about bugs in here, lol...'. |
| 23:15.39 | Nethab | you use hangup when you haven't dialed anything yet |
| 23:15.59 | robl^ | Ariel_, ahh! thanks |
| 23:16.03 | shido | hangup is not needed if everything is working right |
| 23:16.13 | danett- | like in an ivr |
| 23:16.13 | shido | I noticed it doesnt hang up in the uk however on some lines |
| 23:16.19 | Nethab | or when you want to hangup on specific callerids |
| 23:16.27 | Nethab | *cough* ex wife |
| 23:17.46 | danett- | hehe |
| 23:18.04 | danett- | well. since the inbound from voipgate uses the general section other users wich are not registred on the system will always follow 's' extention |
| 23:19.10 | Ariel_ | robl^, no problem. |
| 23:20.08 | danett- | in the context of [general] |
| 23:21.50 | Nethab | if you put the /<exten> on your register line it will ring that extension, so you can use s for guests |
| 23:22.29 | danett- | that's smart :) |
| 23:23.10 | danett- | does 's' precede the extention? of the other way arouond |
| 23:23.20 | danett- | (this is the last question :) then i will go to sleep hehe) |
| 23:23.44 | Nethab | no s is the default extension triggered if no other extension is specified |
| 23:23.53 | Nethab | it's the 'start' extension |
| 23:24.23 | danett- | you should write the documentation :) |
| 23:24.39 | Nethab | i should rewrite asterisk |
| 23:25.06 | *** join/#asterisk Dus10 (~Dus10@adsl-69-210-15-227.dsl.ipltin.ameritech.net) |
| 23:25.13 | Dus10 | hello all |
| 23:25.20 | Nethab | hello dustin |
| 23:25.32 | drumkilla | Nethab: maybe you should rebuild the internet, too |
| 23:25.39 | Nethab | yes i should |
| 23:25.42 | drumkilla | since both things are so trivial. |
| 23:25.47 | Nethab | no more of this IP address bs |
| 23:25.51 | danett- | it's good to have goals in life :) |
| 23:26.11 | danett- | would exten => _.,1,Congestion be the same as 's'? |
| 23:26.17 | Dus10 | I am new to asterisk, and phone systems as a whole. I am downloading asterisk@home first, and I will be trying it out. Is it possible to use standard fax/modems to connect to POTS and analog phones? |
| 23:26.33 | danett- | Dus10: if you have 2 modems, yes |
| 23:26.39 | Dus10 | cool |
| 23:26.55 | danett- | 1 modem of the telephone line, and one for the analog phone |
| 23:27.00 | Dus10 | so one to connect to my POTS line, and one to hook an analog phone into, and that should do it? |
| 23:27.01 | Nethab | no, modems don't work well for fxs pors |
| 23:27.02 | Dus10 | sweet |
| 23:27.08 | danett- | they don't? |
| 23:27.12 | Dus10 | not at all? |
| 23:27.13 | danett- | hmm. misinformed then |
| 23:27.14 | Nethab | not for fxs |
| 23:27.25 | Dus10 | I just wanna play with it, not use it in production |
| 23:27.26 | Nethab | there are FXO ports and FXS ports |
| 23:27.38 | danett- | fxs is for ringing the phone right |
| 23:27.49 | Nethab | FXO is what connects to the phone company, and FXS is for plugging into your phone |
| 23:28.00 | Dus10 | fxo is fine with any old modem? |
| 23:28.10 | Sedorox | ~FXO |
| 23:28.11 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 23:28.11 | Sedorox | ~FXS |
| 23:28.12 | jbot | i guess fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 23:28.20 | Dus10 | I could use a soft phone to play with it, I guess |
| 23:28.21 | Nethab | a modem can't act like the phone company, so plugging a phone into it won't work well |
| 23:29.38 | *** join/#asterisk AlexCeli (~Alex@200.37.85.91) |
| 23:29.48 | puppet | hmm |
| 23:29.55 | jeremywhiting | apardo: with a register => line in sip.conf |
| 23:30.10 | robl^ | does anyone know if Atacomm still has his VoIP store? the URL I have no longer works |
| 23:30.35 | puppet | why doesnt the recordfilename get set to CALLFILENAME while using *1 |
| 23:30.45 | Dus10 | has anyone here setup asterisk to work with vonage? |
| 23:31.00 | Dus10 | heh |
| 23:31.00 | danett- | well. |
| 23:31.02 | danett- | <PROTECTED> |
| 23:31.07 | danett- | thanks for the help Nethab |
| 23:31.08 | Dus10 | I cannot believe this |
| 23:31.14 | Nethab | vonage specifically block asterisk from registering |
| 23:31.29 | Dus10 | I am going to download 555MB work of data in 13 minutes... |
| 23:31.32 | Dus10 | my DSL rocks |
| 23:31.40 | Dus10 | hmm |
| 23:31.42 | Dus10 | that stinks |
| 23:31.44 | claude2005 | Is there a way of picking up a call of extension that is ringing on another extension that is not ringing |
| 23:31.52 | Dus10 | that is what one of my friends wants |
| 23:31.56 | danett- | pickup group :) |
| 23:32.02 | danett- | dunno if asterisk can do it |
| 23:32.08 | drumkilla | how are you going to pick up a call for an extension that isn't ringing? :) |
| 23:32.18 | drumkilla | danett-: it does |
| 23:32.28 | danett- | pick group owns |
| 23:32.29 | Juggie | drum, can u code review 4403 |
| 23:32.36 | Dus10 | what about with using fxs to connect like an analog device? |
| 23:32.39 | drumkilla | Juggie: calm your nerves, son! |
| 23:32.42 | Juggie | :) |
| 23:32.45 | drumkilla | you've asked about that like 50 times today |
| 23:32.48 | danett- | Dus10: you need an fxs card |
| 23:32.52 | danett- | they are pretty expensive |
| 23:32.52 | drumkilla | we'll get to it :) |
| 23:32.57 | Juggie | patience! :) |
| 23:33.00 | drumkilla | there is a lot more stuff that has been there longer than that one |
| 23:33.02 | *** join/#asterisk dudes (~dudes@12-215-34-84.client.mchsi.com) |
| 23:33.15 | Dus10 | really? |
| 23:33.21 | Dus10 | I saw those generic ones for $7 |
| 23:33.40 | Nethab | he wants to steal a call that's ringing one extension and bring it to his extensin |
| 23:33.47 | Dus10 | he is operating a business and is willing to pay a reasonable amount of money for equipment |
| 23:34.15 | Dus10 | he just does not want to spend $7K for a phone system for five phones |
| 23:34.38 | puppet | Jun 3 01:29:07 WARNING[5747]: res_monitor.c:154 ast_monitor_start: Could not create file /var/spool/asterisk/monitor/auto-1117754947-1-Unknown-in |
| 23:34.58 | Nethab | so he doesn't have a PRI or BRI |
| 23:35.08 | Dus10 | nope, vonage |
| 23:35.08 | *** join/#asterisk kf4zmt (~kf4zmt@68-64-224-138.ironoh.adelphia.net) |
| 23:35.16 | puppet | -- Executing Set("SIP/0850007222-e14f", "CALLFILENAME=Unknown_20050603-012859") in new stack |
| 23:35.24 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 23:35.27 | puppet | can someone explain why res_monitor tries to set the other filename? |
| 23:35.44 | kf4zmt | hello |
| 23:38.20 | *** join/#asterisk irv999 (~irv999@ool-44c241c7.dyn.optonline.net) |
| 23:38.22 | irv999 | lo all |
| 23:38.40 | irv999 | darn.. NG is not around.. |
| 23:39.12 | *** join/#asterisk sedwards50 (~chatzilla@adsl-64-171-118-72.dsl.sndg02.pacbell.net) |
| 23:39.20 | irv999 | Does anyone know how I can make my polycom 600 accept or be routed ONLY 1 call at a time per button instead of 2 (which is it s max) |
| 23:43.32 | *** part/#asterisk kf4zmt (~kf4zmt@68-64-224-138.ironoh.adelphia.net) |
| 23:44.48 | irv999 | lalallaall |
| 23:46.52 | swk | ivr999: upgrade to the latest firmware |
| 23:49.45 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@adsl-69-209-152-90.dsl.sfldmi.ameritech.net) |
| 23:49.58 | Qwell | MikeJ[Laptop]: I failed. :p |
| 23:49.59 | puppet | bah not getting one touch record to work |
| 23:50.03 | puppet | im going to sleep :( |
| 23:50.41 | MikeJ[Laptop] | yes, but all failure leads to future sucess |
| 23:50.48 | Ariel_ | it's very strange. But I have followed the setup on the wiki for the Sipura 3000 and I still can't get the damm pstn side working. |
| 23:50.48 | Qwell | MikeJ[Laptop]: perhaps |
| 23:50.55 | *** join/#asterisk tessier (~treed@wsip-68-224-172-77.sd.sd.cox.net) |
| 23:50.57 | tessier | Hello all! |
| 23:51.00 | MikeJ[Laptop] | do you get why? |
| 23:51.07 | tessier | What is the default pass on the Cisco 7912? |
| 23:51.09 | Qwell | MikeJ[Laptop]: get why what? |
| 23:51.22 | MikeJ[Laptop] | get what was wrong with the patch |
| 23:51.22 | puppet | night all |
| 23:51.28 | Qwell | yeah, hascallerid = callerid |
| 23:51.30 | Ariel_ | anyone here has gotten the sipura 3000 working with the pstn for inbound? |
| 23:51.39 | MikeJ[Laptop] | drumkilla said it did not even apply |
| 23:51.39 | Qwell | The formatting was also wrong, but it was correct...heh |
| 23:51.45 | Qwell | erm? |
| 23:51.47 | MikeJ[Laptop] | :) |
| 23:52.01 | Qwell | his note just said it wouldn't compile |
| 23:52.21 | MikeJ[Laptop] | did you try to compile it? |
| 23:52.29 | Qwell | MikeJ[Laptop]: no, I know what was wrong with the compilation |
| 23:52.34 | Qwell | I didn't realize it wouldn't even apply though... |
| 23:53.03 | MikeJ[Laptop] | maybe I misunderstood what he said... |
| 23:53.25 | Qwell | it should apply, it was a straight cvs diff -u |
| 23:53.41 | Qwell | I totally hosed the e->hascallerid lines though |
| 23:53.43 | MikeJ[Laptop] | it may have |
| 23:53.48 | Nethab | yes i use SPA-3000 |
| 23:53.55 | MikeJ[Laptop] | no biggy... that's how you learn |
| 23:53.56 | Nethab | and have inbound calling |
| 23:54.01 | Qwell | erm, callwaiting...whatever it was |
| 23:54.20 | Qwell | e->hascallwaiting = hascallwaiting isn't the same as e->hascallwaiting = callwaiting, heh |
| 23:54.28 | *** join/#asterisk jdg (~jdg@CA03F85F.adsl.mana.pf) |
| 23:54.46 | Nethab | Ariel_: I use the SPA-3000 for inbound PSTN calls |
| 23:55.14 | Ariel_ | Nethab, ok so what is the secret to get it working for inbound calls? |
| 23:55.26 | Nethab | does it register? |
| 23:55.32 | Ariel_ | no |
| 23:55.39 | Ariel_ | only the line 1 does |
| 23:55.40 | Nethab | is it a static IP |
| 23:55.54 | Ariel_ | no but it's on the same network no nat |
| 23:56.09 | dr123 | I am going to need some help in 2 minute when i get this new Cisco 7960 Plugged in on flashing the firmware i have the firmware files!!!! |
| 23:56.16 | Nethab | ok go to admin / advanced |
| 23:56.26 | *** join/#asterisk Rez (lorez@lorez.staff.freenode) |
| 23:57.23 | Ariel_ | Nethab, ok |
| 23:57.52 | Nethab | line enable yes? |
| 23:58.12 | Ariel_ | yes |
| 23:58.15 | Nethab | go to PSTN line |
| 23:59.21 | Ariel_ | Nethab, you can put more then one thing on the line....Yes.... |
| 23:59.23 | sedwards50 | Anybody care to take a guess on when this channel is "busiest?" I need some volunteers to help generate about 130 calls to my 800 number so I can confirm the t1's are rolling over correctly. |
| 23:59.53 | Nethab | the trick is to have it register as a user phone, and set a default dialplan for incoming calls, to automatically go to an extension/context in asterisk |
