| 00:00.20 | linagee | ManxPower: so an SMS service center is "dialed out" to send out messages, but what if you want to receive them for your DIDs too? just not possible? heh. |
| 00:00.57 | linagee | oh yikes. please enter your info. you will be tracked. heh |
| 00:01.15 | *** part/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 00:01.37 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 00:02.00 | linagee | hehehe |
| 00:02.55 | linagee | first number just kept ringing |
| 00:03.46 | linagee | bizarre. are these numbers not meant to be rung? heh. they all don't actually pick up. heh |
| 00:04.13 | linagee | probably some sort of "ISDN" thing or out of band signaling thing i guess |
| 00:06.18 | *** join/#asterisk Dr-Linux|home (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk) |
| 00:07.54 | Dr-Linux|home | asterisk text to speech works fine, but it's machine voice, is there anyway to make this voice good or something good ? |
| 00:08.19 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:08.46 | ManxPower | Dr-Linux: Which specific one? |
| 00:09.29 | linagee | ManxPower: the british guy. :) |
| 00:10.03 | ManxPower | linagee: I assume that would be Festivel |
| 00:10.45 | linagee | ManxPower: unless there's another freeloader, er i mean open source text to speech "renderer" |
| 00:10.50 | linagee | :) |
| 00:11.03 | ManxPower | I was not aware Dr-Linux said anything about open source. |
| 00:11.14 | linagee | Dr-Linux|home: which one then? |
| 00:11.23 | linagee | ManxPower: not enough info to continue. <kernel panic> |
| 00:11.33 | wunderkin | press any key to continue |
| 00:11.38 | ManxPower | linagee: exactly. |
| 00:11.45 | ManxPower | I call that "lazy" |
| 00:12.03 | linagee | ManxPower: understatement. :) |
| 00:12.27 | Dr-Linux|home | ManxPower: yeah, i just tried fastivel weather |
| 00:12.32 | ManxPower | linagee: http://smsforum.net/smf/index.php?PHPSESSID=862713296e3d851c574703eab4d99313&topic=2117.0 |
| 00:13.14 | linagee | ManxPower: interesting. "you must pay" |
| 00:13.16 | linagee | ManxPower: so... |
| 00:13.18 | Dr-Linux|home | linagee: i wanna use it within my IVR, but when i need text2speech, that has kinda machine voice, so too much different |
| 00:13.33 | Dr-Linux|home | so was looking for something good |
| 00:13.40 | linagee | ManxPower: it's like DNS. lol. its just data or a database or whatever. but you have to pay. :) you can't just stick your server out there listening |
| 00:14.37 | linagee | Dr-Linux|home: weather probably does this by just reading out numbers or something. certain "symbols" they have recorded. |
| 00:14.49 | linagee | Dr-Linux|home: a trick to make text to speech "look better" |
| 00:15.52 | linagee | ManxPower: why do we even bother with things that say "there's no free lunch" when there are open protocols like TCP/IP where you *can* just stick your server out listening on a port? heh |
| 00:15.54 | Dr-Linux|home | linagee: it's tatally machine voice :S |
| 00:16.04 | ManxPower | Dr-Linux: there are better voices available, but they are not much better. Your best bet is to spend the money on a commercial TTS package like Cepstral |
| 00:16.08 | linagee | Dr-Linux|home: can't understand. cannot compute |
| 00:16.26 | Dr-Linux|home | hhm.. ok |
| 00:16.33 | linagee | ManxPower: or do the hackish thing of recording tons of "symbols" (ie, phrases of things that your IVR will ever say) |
| 00:16.41 | Dr-Linux|home | how about this one: |
| 00:16.42 | Dr-Linux|home | http://nerdvittles.com/index.php?p=134 |
| 00:17.04 | ManxPower | linagee: Oh, I'm sure you could set up your own SMS network to message between all your clients. but if you want to connect to someone else's clients, like Verizon or T_Mobile, then they will make you pay |
| 00:17.26 | linagee | Dr-Linux|home: so that uses "Flite". go fer it. tell us how it goes. |
| 00:17.35 | wunderkin | luckilly eventually they will all be one carrier so you only have to pay for one.. |
| 00:17.42 | linagee | ManxPower: yep. like DNS then. :) |
| 00:17.45 | ManxPower | Dr-Linux: perhaps a search of the mailing list archives will help you. |
| 00:17.55 | linagee | ManxPower: you could set up your own DNS "club", etc etc. |
| 00:18.07 | Dr-Linux|home | linagee: but it says, Flite is for Asterisk@home |
| 00:18.15 | Dr-Linux|home | i don't use A@home |
| 00:18.19 | linagee | Dr-Linux|home: so try it. break something. let us know how it goes. |
| 00:18.34 | linagee | Dr-Linux|home: remember when you try new things though, always make backups. |
| 00:19.18 | Dr-Linux|home | linagee: Thanks, but i've a couple of test servers |
| 00:19.23 | J4k3 | bah, practice makes perfect... just set it back up! :) |
| 00:19.28 | linagee | ManxPower: "but what do you mean i have to round out this square to make it fit into the circle hole". hah. or you could hire a consultant to do it for you. |
| 00:19.31 | Dr-Linux|home | and KVM |
| 00:19.35 | J4k3 | oh, that only applies during business hours during the week |
| 00:19.48 | J4k3 | ;) |
| 00:21.25 | linagee | ManxPower: lol. i love that paragraph, "direct to carrier" |
| 00:21.44 | linagee | ManxPower: you should just call up the CEO of cingular and tell him you have a message to deliver to one of his client's mobile phones. :-> |
| 00:22.00 | wunderkin | linagee... at&t... they are all... at&t... |
| 00:22.09 | linagee | wunderkin: er, AT&T. yes |
| 00:22.17 | linagee | wunderkin: AT&T still "owns" the internet. ;-) |
| 00:22.26 | linagee | or do i even need quotes. hehe |
| 00:22.38 | polerin | http://pastebin.ca/429025 line 10 is saying menu-start isn't a lable. :/ |
| 00:22.46 | wunderkin | all of the tubes that al gore invented? |
| 00:23.06 | Innatech | I saw some kind of asterisk tool for controlling cell phones over USB cables. That + free phone + unlimited text package (~15.99/mo) seems like the best solution for quick and dirty SMS. |
| 00:23.15 | polerin | i know i'm doing something wrong, but ... |
| 00:23.15 | ManxPower | polerin: DON'T PUT IN EXTRA SPACES |
| 00:23.24 | linagee | ma bell ownz joo |
| 00:23.48 | polerin | ManxPower: lol |
| 00:24.01 | polerin | ManxPower: I think I deserved the caps ;P |
| 00:24.32 | polerin | before/around the ? or where? |
| 00:24.44 | polerin | oh, wait I tried it without the space in front of the lable, and with () around it |
| 00:24.47 | polerin | same result |
| 00:25.06 | polerin | sorry, thought I had reset before binning |
| 00:25.34 | linagee | ManxPower: what if i have a product that just might "take over" at&t? will i get a black van at my house soon? hah |
| 00:25.46 | ManxPower | exten => s,n,Gotoiftime(08:30-19:59?incoming-menu,s,menu-start) -- assumiung you want context = incoming-menu, extension=s and priority=menu-start |
| 00:26.32 | polerin | linagee: you've got a product that is going to replace millions of miles of copper and fiber? neat, when did you get the ansible working? |
| 00:26.45 | linagee | polerin: :-D |
| 00:26.46 | ManxPower | well label=menu-start of course. |
| 00:26.51 | polerin | yeah, |
| 00:26.55 | linagee | polerin: think about what you just said for a few minutes |
| 00:26.57 | polerin | I think I tried that too, but resetting |
| 00:27.07 | polerin | ... |
| 00:27.08 | polerin | hehehe |
| 00:27.21 | polerin | yeah the black van boys might be interested in an ansible |
| 00:28.04 | linagee | polerin: where did you get that from? ender's game? heh |
| 00:28.17 | ManxPower | GotoIfTime(<times>|<weekdays>|<mdays>|<months>?[[context|]exten|]priority) |
| 00:28.35 | ManxPower | notice how times, weekdays, mdays, and months are NOT optional |
| 00:29.44 | ManxPower | This is also wrong, of course: exten => s,n menu-start,Wait(1) |
| 00:29.59 | ManxPower | you would need exten => s,n(menu-start),Wait(1) |
| 00:30.06 | ManxPower | there you go with the extra spaces again |
| 00:30.08 | polerin | ManxPower: actually I was getting warnings on *,*,* |
| 00:30.27 | polerin | yeah actually I mentioned that above, it was a test I thought I had reset |
| 00:30.57 | polerin | linagee: yes, though I've read the rest of the series as well |
| 00:31.02 | ManxPower | And I'm not so sure I know what "exten => s,n,Gotoif($[${IN_MENU} >3]?:t,1)" is doing. |
| 00:31.32 | ManxPower | looks like go to exten t, priority 1 if IN_MENU is NOT greater than 3 |
| 00:31.34 | polerin | I thought you could leave off the context on the destination |
| 00:31.52 | polerin | if you are inside of the same context |
| 00:32.03 | polerin | sec |
| 00:32.08 | ManxPower | polerin: that is correct, but I have not said that you can't. |
| 00:32.40 | ManxPower | You can even leave of the extension if you want to stay within the same extension, but go to a different priority/label |
| 00:32.43 | polerin | yeah, wanted too loop through the menu 3 times and then go to the t extention (ie goodbye) |
| 00:32.57 | linagee | polerin: me too |
| 00:33.09 | polerin | linagee: I assumed so as you knew what it was ;) |
| 00:33.12 | linagee | polerin: ender gets the world in a huge conflict/fight by using newsgroups. LOL |
| 00:33.22 | ManxPower | Well you have an empty place yo go if the evaluation is true. i.e. ?:t,1 |
| 00:33.31 | ManxPower | since you have the : |
| 00:33.34 | linagee | polerin: maybe he just stopped the flow of porn and it pissed people off. LOL |
| 00:33.56 | polerin | ManxPower: doesn't it just go to the next step if dest1 is blank? |
| 00:34.02 | linagee | polerin: where's my alt.binaries.pics!!! *hits nuclear meltdown button* |
| 00:34.23 | polerin | ... oh gods I'm not going to even THINK about that linagee. |
| 00:34.47 | ManxPower | GotoIf($[condition]?truedest:falsedest) |
| 00:34.48 | linagee | polerin: what if someone crashed the newsgroups and it spread to every other server. hehehe |
| 00:35.01 | polerin | sec. |
| 00:35.05 | ManxPower | since your truedest is empty it will go to the next priority, your falsedest goes to t,1 |
| 00:35.32 | polerin | yeah, that was the intent |
| 00:36.01 | ManxPower | Just making sure. Might be more readable if you did exten => s,n,Gotoif($[${IN_MENU} < 3]?t,1) |
| 00:36.20 | linagee | ManxPower: oic.... "So, our advice is simple.. unless your idea is the size of something like Big Brother, Pop Idol etc.. i.e something that has the potential of millions of premium-rate SMS votes etc., then forget the carrier." |
| 00:36.55 | linagee | ManxPower: so that's how they do those stupid "vote now for $1.99, just text 1234" (and it's an unusually small number) |
| 00:36.59 | ManxPower | your current line basically says "If IN_MENU is greater than 3, then go to the next priority. If IN_MENU is less than or equal to 3 then go to t,1 |
| 00:37.19 | polerin | ManxPower: er.. thought it said the oppsite? |
| 00:37.27 | polerin | i thought it was truedest:falsedest |
| 00:37.48 | polerin | unless I messed up my </> because I was on th ephone :P |
| 00:38.43 | ManxPower | Correct. If IN_MENU is greater then 3, and IN_MENU is 4 then it will go to the next priority by default because your line has NO truedest. It has an empty truedest. |
| 00:38.44 | polerin | .. Oh bugger I DID mess it up. should be as it is but without the : |
| 00:38.59 | ManxPower | polerin: that is what I have been trying to say for the past 15 mins |
| 00:39.12 | polerin | sorry I have someone who is demanding my attention with aremote in my ribs :D |
| 00:39.49 | ManxPower | polerin: prolly best to not try to do complex technical stuff while distracted. |
| 00:40.54 | ManxPower | you could tear a hole in the space/time continuum. |
| 00:41.00 | ManxPower | or something |
| 00:41.08 | polerin | ManxPower: it's the only chance I get. I actually built a modular OO php framework during and in-between back to back phone calls :P |
| 00:41.25 | polerin | it just meens you have to triple check your work to make sure it behaves as intended |
| 00:41.43 | ManxPower | not with your current attention to detail you are not. |
| 00:42.10 | polerin | didn't say I did it quickly ;P |
| 00:42.13 | ManxPower | please tell me you are not coding airplane or power plant systems. |
| 00:42.19 | polerin | lol |
| 00:42.20 | polerin | no |
| 00:42.49 | polerin | not to mention that I am in the process of phasing out that framework. |
| 00:46.22 | polerin | ok other than extra spaces and the brainfart in the operand, anything horribly wrong with the setup? |
| 00:54.04 | *** join/#asterisk Kumbang (n=macan@router-wmi.paume.itb.ac.id) |
| 01:05.29 | *** join/#asterisk MrTelephone (n=test@bas13-toronto63-1242371209.dsl.bell.ca) |
| 01:05.50 | MrTelephone | does anyone have any knowledge about collect calls and how to handle them? |
| 01:07.54 | *** join/#asterisk djs307 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 01:14.36 | MrTelephone | exit |
| 01:20.53 | *** join/#asterisk Mportnoy (n=test@200.122.158.88) |
| 01:21.23 | Mportnoy | hello, I have a problem the voicemail is telling me that is full but I dont have any Voicemail... what could be the problem ? |
| 01:26.47 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 01:31.11 | Dimitripietro | Mportnoy, tcheck Linux permission on the voicemail folder |
| 01:32.26 | bkruse_home | bowling night in hsv! |
| 01:32.50 | Qwell | wii bowling? :p |
| 01:33.21 | bkruse_home | Qwell: real bowling! :D |
| 01:35.57 | *** join/#asterisk MrTelephone (n=test@bas13-toronto63-1242371209.dsl.bell.ca) |
| 01:36.21 | MrTelephone | anyone alive here? |
| 01:36.29 | bkruse_home | MrTelephone: never! |
| 01:36.32 | bkruse_home | wuts up bud |
| 01:36.33 | MrTelephone | hah |
| 01:36.35 | MrTelephone | not much |
| 01:36.41 | Qwell | bah! |
| 01:36.43 | MrTelephone | im just trying to setup some sip clients.. |
| 01:36.51 | MrTelephone | and the problem I'm facing is call-limit! |
| 01:37.03 | MrTelephone | call-limit has to be set to 2 for callwaiting to work |
| 01:37.21 | bkruse_home | well ya, that is 2 calls. |
| 01:37.27 | MrTelephone | that works for analog adaptors but it still lets people with softphones to use 2 lines? |
| 01:37.39 | MrTelephone | I'm trying to decide how to do this |
| 01:37.54 | MrTelephone | so I should set the clients phone to a single line softphone (its a polycom 501) |
| 01:38.14 | MrTelephone | so set call-limit = 2 but set the phone to only one line? |
| 01:38.46 | MrTelephone | there's not much on collect calls either :( |
| 01:39.12 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 01:39.27 | Qwell | MrTelephone: should be able to tell if it's collect by looking at the ANI II |
| 01:39.32 | Qwell | Strom_M: right? |
| 01:40.03 | Strom_M | no |
| 01:40.03 | MrTelephone | ANI II, is that supported in asterisk? |
| 01:40.07 | Dr-Linux|home | bkruse_home: hey there :) |
| 01:40.08 | Qwell | oh |
| 01:40.14 | Qwell | I stand corrected then ;) |
| 01:40.18 | bkruse_home | Dr-Linux|home: wut up! |
| 01:40.31 | Strom_M | the II digits determine the class of service of the originating station; not whether it's a collect call |
| 01:40.43 | Strom_M | all you'll be able to tell is maybe whether it's been handled by an operator |
| 01:40.43 | Qwell | figured collect would be a CoS |
| 01:40.48 | Strom_M | nope |
| 01:40.56 | Qwell | lame |
| 01:41.19 | MrTelephone | if people are using asteirsk to sell service then how do you manage collect calls? |
| 01:41.26 | Dr-Linux|home | bkruse_home: you were about to giving me an example for socket connect perl/agi :) |
| 01:42.37 | Cybertoy | how do I jump to priority n+101 when using extensions.ael ??? |
| 01:42.49 | Qwell | Cybertoy: You don't. |
| 01:42.55 | Qwell | n+101 is discouraged |
| 01:43.02 | Cybertoy | ok |
| 01:43.10 | Cybertoy | so I Have a Dial(); command in extensions.ael ... |
| 01:43.17 | MrTelephone | hmmm |
| 01:43.21 | Qwell | check ${DIALSTATUS} |
| 01:43.26 | Cybertoy | the next one can be goto(s-${dialstatus}) ? |
| 01:43.28 | Cybertoy | ok |
| 01:43.29 | Cybertoy | tnx |
| 01:43.34 | MrTelephone | Strom, any idea how collect calls are handled? |
| 01:43.37 | Qwell | no, DIALSTATUS, not dialstatus |
| 01:43.41 | Cybertoy | yeah ... |
| 01:43.47 | Cybertoy | was too lazy with the caps.. :) |
| 01:45.25 | MrTelephone | poopy :( |
| 01:46.03 | MrTelephone | im using perl to scan through NPANA codes, takes forever for billing |
| 01:46.04 | Strom_M | MrTelephone: what are you trying to do, exactly |
| 01:46.18 | MrTelephone | im just worried about how to handle collect calls |
| 01:46.50 | MrTelephone | im going to get a phone bill and i'll have to manually find out where the collect calls belong too by comparing asterisk cdr and the telcos bill |
| 01:47.03 | MrTelephone | there won't be too many but it will still suck |
| 01:47.10 | Strom_M | ok, so don't answer my question then |
| 01:48.16 | MrTelephone | maybe I'm lagged :( |
| 01:48.24 | Strom_M | MrTelephone: what are you trying to do, exactly |
| 01:49.43 | MrTelephone | to have collect calls marked in the cdr. |
| 01:50.12 | MrTelephone | I havn't had any collect calls yet but I'm guessing that they'll look like regular incoming calls |
| 01:50.42 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 01:50.50 | Strom_M | what kind of entrance facilities do you have? |
| 01:51.56 | MrTelephone | a t1/pri into a sangoma pri pci card |
| 01:52.00 | MrTelephone | ? |
| 01:52.10 | Strom_M | who is the PRI provider? |
| 01:52.15 | MrTelephone | bell canada |
| 01:52.19 | MrTelephone | running a dms100 switch |
| 01:53.31 | Strom_M | do you get ANI II digits on other inbound calls? |
| 01:53.33 | MrTelephone | shit another cop got shot in the states.. gotta ban those guns |
| 01:53.48 | MrTelephone | strom how can I tell. pri debug? |
| 01:54.18 | Strom_M | perhaps. It depends on how Bell Canada is sending it |
| 01:54.30 | Strom_M | or whether ANI II digits are even relevant in canada |
| 01:55.18 | MrTelephone | is ANI II supported? i see something here on the web about ${ANII} |
| 01:55.29 | MrTelephone | ${ANIII} i mean |
| 01:55.45 | Strom_M | well, look at your PRI debug first |
| 01:57.03 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 01:57.25 | linagee | once you advertise your logo to an extent where it's globally recognized, what's really the point in pasting it everywhere? /me makes reference to AT&T |
| 01:57.54 | MrTelephone | I'll see something ANI II in the debug? |
| 01:58.10 | Dr-Linux|home | what's ANIII ? :S |
| 01:59.04 | kn0x | Information Indicator.. i think it stands for |
| 01:59.20 | kn0x | it decribes what type of phone the call originated from |
| 01:59.47 | kn0x | 00 POTS line |
| 01:59.55 | kn0x | 23 payphone |
| 02:00.11 | ber_ | hi, im getting error 407 "Proxy Authentication Required" when i try to send a call to my sip termination provider |
| 02:00.26 | ber_ | does anyone know how i can authenticate the invite? |
| 02:00.34 | MrTelephone | ok what are the different IE codes |
| 02:01.10 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 02:01.12 | Dr-Linux|home | kn0x: thanks, i thought something new feature for asteris, since i'm using pri's and fxo's :) |
| 02:01.27 | kn0x | MrTelephone- http://www.nanpa.com/number_resource_info/ani_ii_assignments.html ? |
| 02:01.32 | Dr-Linux|home | as he mentioned : <MrTelephone> ${ANIII} i mean |
| 02:01.46 | kn0x | yeah, asterisk supports them on pri i believe |
| 02:02.01 | kn0x | ive never had the luxury of access to a PRI so I have no idea |
| 02:02.18 | *** join/#asterisk LasaK (n=mypain@203.117.213.88) |
| 02:02.34 | LasaK | hi all |
| 02:04.44 | MrTelephone | i hope so |
| 02:04.58 | MrTelephone | what is the code for collect.. i don't think its that simple |
| 02:06.22 | LasaK | how many user on you asterisk you have guys ? |
| 02:06.24 | Dr-Linux|home | Qwell: when that 7935 patch is going to be work for asterisk statble version? :) |
| 02:06.37 | Qwell | when at least one person has tested it and says it works... |
| 02:07.26 | MrTelephone | i have like 20 users |
| 02:07.35 | MrTelephone | and at most so far only 5 channels have been used at the same time |
| 02:07.42 | MrTelephone | 5 pri channels |
| 02:07.47 | Dr-Linux|home | Qwell: how that person will test? will setup a different asterisk box for SVN version? |
| 02:09.10 | LasaK | i found strange behavior in my instalation |
| 02:09.13 | *** join/#asterisk pck1 (i=Parker@ool-18bd7ea2.dyn.optonline.net) |
| 02:09.43 | pck1 | with asterisk can u spoof caller id's? |
| 02:09.59 | LasaK | once a time whole my user got disconnect, status changes to unreach |
| 02:10.00 | Dr-Linux|home | Qwell: if yes, then someday give me a few time, i'll setup a setup an asterisk box for svn version |
| 02:10.02 | LasaK | whole of them |
| 02:10.04 | Dr-Linux|home | to test out my phone |
| 02:10.10 | LasaK | but there is nothing with the network |
| 02:10.34 | Dr-Linux|home | LasaK: users are on LAN? |
| 02:11.16 | LasaK | nope |
| 02:11.18 | pck1 | what voip service s hould i use |
| 02:11.19 | LasaK | over internet |
| 02:11.20 | pck1 | with astreix |
| 02:11.21 | pck1 | so spoof |
| 02:11.40 | Dr-Linux|home | pck1: internet is fine? |
| 02:12.05 | MrTelephone | spoof callerids? |
| 02:12.06 | MrTelephone | heh |
| 02:12.35 | Dr-Linux|home | LasaK: use qualify=yes in sip.conf and the "sip reload" |
| 02:12.45 | MrTelephone | Qwell phoned me and the ANI II code showed up as 29??? |
| 02:12.47 | LasaK | yes |
| 02:13.01 | LasaK | i think its not the matter |
| 02:13.14 | MrTelephone | 29 Prison/Inmate Service - the ANI II digit pair 29 is used to designate lines |
| 02:13.17 | LasaK | becoz when i registered one by my self |
| 02:13.20 | MrTelephone | haha |
| 02:13.22 | Qwell | MrTelephone: yes, I spoof ANI II as 29 |
| 02:13.26 | LasaK | it got registered in cli |
| 02:13.43 | MrTelephone | I'll bring you a carton of ciggs during visiting hour tomorrow |
| 02:13.45 | pck1 | @ Dr-linux |
| 02:13.46 | LasaK | but when i issue sip show peers, its still got status unreach |
| 02:13.48 | pck1 | yes internet is what i want |
| 02:13.50 | LasaK | whole of them |
| 02:14.09 | pck1 | do u know of a good internet voip service i could use with astreix to spoof? |
| 02:14.09 | MrTelephone | pck1, goto #hacking |
| 02:14.24 | pck1 | no ones there |
| 02:14.34 | MrTelephone | why do you want to spoof? |
| 02:14.36 | Qwell | what the hell is astreix? |
| 02:14.42 | MrTelephone | can't you block your caller name and id through ur phone company? |
| 02:14.50 | pck1 | bc spoofing can be fun |
| 02:14.59 | MrTelephone | ok thats rediculous |
| 02:15.04 | MrTelephone | what do you have to hide |
| 02:15.26 | MrTelephone | are you trying to pick up a girl by using brad pitts phone number? |
| 02:15.28 | pck1 | to prank friends |
| 02:15.38 | Dr-Linux|home | LasaK: do sip debug for the peer |
| 02:15.40 | MrTelephone | if you have a pri you can set anything you want |
| 02:15.44 | Dr-Linux|home | and see what's happening |
| 02:15.48 | MrTelephone | if you don't have one then don't bother |
| 02:16.07 | MrTelephone | 1000$ a month to prank your friends.. if your dad is bill gates why not? |
| 02:16.19 | LasaK | the whole peers ? |
| 02:16.37 | pck1 | why would it be 1000 a month... |
| 02:16.46 | Dr-Linux|home | LasaK: unreachable doesn't mean unkown |
| 02:17.49 | LasaK | yupe |
| 02:17.57 | MrTelephone | i payed some guys in new jersey for voip service |
| 02:17.59 | MrTelephone | it was pretty good |
| 02:18.04 | MrTelephone | connect.voicepulse.com |
| 02:18.07 | MrTelephone | or something |
| 02:18.28 | MrTelephone | they let you change your callerid on the web |
| 02:18.40 | pck1 | thank you |
| 02:18.44 | red9012 | In using streaming music on hold. Does each music on hold class result in a new stream, or one stream can be shared among all classes? |
| 02:19.01 | J4k3 | pck1: vitelity doesn't seem to adjust my outgoing CID |
| 02:19.30 | J4k3 | I forgot to set it and ended up sending my extension numbers out for a few days |
| 02:19.33 | J4k3 | oops. |
| 02:19.51 | J4k3 | my cellphone suddenly shouts out "call from 401 *riiiiiing*" |
| 02:19.56 | J4k3 | and I'm like... whowhatwhy |
| 02:20.09 | pck1 | lol |
| 02:20.24 | pck1 | who do u use for service? |
| 02:20.32 | LasaK | any one here can help me with an url howto for asterisk 1.4.2 ? |
| 02:20.46 | J4k3 | vitelity.net |
| 02:21.04 | pck1 | thank you |
| 02:22.00 | J4k3 | the service seems decent so far and the pricing is alright... |
| 02:22.19 | J4k3 | when I get 5k+ minutes/month I'll go shopping for better. |
| 02:22.52 | *** join/#asterisk h3x0r (n=hex@64.192.116.17) |
| 02:22.54 | pck1 | im not gonna use many minutes |
| 02:22.59 | pck1 | this is perfect thanks alot |
| 02:23.07 | h3x0r | !seen zoa |
| 02:23.07 | hermuli | <PROTECTED> |
| 02:23.23 | h3x0r | well, thats a bizzznatch |
| 02:23.24 | h3x0r | haha |
| 02:24.02 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 02:24.11 | Dovid | anyone here using .1.4.X ? |
| 02:25.35 | *** join/#asterisk rnovotny22 (n=rnovotny@70-56-172-170.mpls.qwest.net) |
| 02:26.14 | *** join/#asterisk brussel_ (n=brussel@cpe-72-130-172-213.san.res.rr.com) |
| 02:29.47 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-12-203.hsd1.fl.comcast.net) |
| 02:32.15 | *** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-126-167-188.washdc.fios.verizon.net) |
| 02:32.19 | SomeOne1 | <PROTECTED> |
| 02:32.25 | SomeOne1 | why does asterisk satart doing that? |
| 02:32.30 | SomeOne1 | <PROTECTED> |
| 02:32.35 | SomeOne1 | it randomly starts going crazy |
| 02:32.43 | SomeOne1 | just sitting there, while NOTHING is happening |
| 02:32.48 | SomeOne1 | it starts using so many resources |
| 02:34.54 | kn0x | Dovid- its 1.4.X |
| 02:35.05 | kn0x | yes, i am using 1.4.2 |
| 02:36.29 | MrTelephone | i made a table in mysql with a - and now it won't let me do NOTHING to it |
| 02:36.30 | MrTelephone | :( |
| 02:36.36 | MrTelephone | can't rename it or anything |
| 02:37.53 | SomeOne1 | someone help! :( |
| 02:40.50 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 02:42.53 | Dovid | kn0x: how has it been working for u ? |
| 02:43.06 | Dovid | kn0x: i am scared to touch it |
| 02:44.54 | *** join/#asterisk rnovotny22 (n=rnovotny@70-56-172-170.mpls.qwest.net) |
| 02:45.52 | SomeOne1 | seems like after a call ended, asterisk automatically re-loaded everything for some reason |
| 02:46.00 | SomeOne1 | and after that it started gobbing up CPU and memory |
| 02:46.03 | SomeOne1 | for no reason |
| 02:46.08 | SomeOne1 | even though it was still functional |
| 02:46.15 | SomeOne1 | no error/debug messages, nothing |
| 02:46.20 | SomeOne1 | does anyone else experience this? |
| 02:47.11 | Dovid | nope. i never had it |
| 02:47.21 | Dovid | hmm. r u runnin mpg123? |
| 02:48.03 | kn0x | Dovid- no problems |
| 02:48.07 | SomeOne1 | no im not |
| 02:48.23 | kn0x | i just upgraded from 1.2.X(11 i think) |
| 02:48.23 | SomeOne1 | no extra modules or anything at all |
| 02:48.40 | kn0x | all my configs worked just fine no issue |
| 02:50.12 | Dovid | cause the bug tracker has been full |
| 02:50.35 | SomeOne1 | Dovid you talkin to me? |
| 02:50.39 | *** join/#asterisk littleball (n=littleba@bb220-255-71-61.singnet.com.sg) |
| 02:50.42 | Dovid | someone1: What do u see on the debug ? and in the logs ? |
| 02:50.58 | SomeOne1 | a call that was 25 hours long |
| 02:51.00 | SomeOne1 | thats all |
| 02:51.04 | SomeOne1 | nothing unsual otherwise |
| 02:51.18 | littleball | hello, i cannot get CALLERID(dnid) in IAX channel. who can help? |
| 02:52.01 | wunderkin | littleball, i don't think i ever have either |
| 02:52.38 | Dovid | someone1: what version of asterisk ? |
| 02:52.47 | Dovid | littleball: how r u trying to get it ? |
| 02:53.08 | SomeOne1 | 1.2.17 |
| 02:53.18 | littleball | mobile -->asterisk A with PSTN-->IAX2-->Asterisk Box B. i try to get CALLERID(dnid) in B |
| 02:53.47 | littleball | i can get CALLERID(dnid) within box A |
| 02:54.09 | littleball | but when use IAX2 forward the call to Box B, CALLERID(dnid) is empty then |
| 02:59.41 | polerin | choppy Playback() but not on every Playback, and the same file won't neccisarly do it twice in a row. no warnings the timing device/etc. Softphone shows GSM, and the files are .gsm's so... |
| 02:59.48 | polerin | any suggestions? |
| 03:00.05 | polerin | googled for it but no luck.. most of what I see is MOH stuff |
| 03:04.42 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:09.33 | InHisName | any Sunrocket.com users connected up right now ? |
| 03:12.07 | polerin | oh, no breakup on echo test either, so I don't think it's network |
| 03:12.35 | *** join/#asterisk Caplain (n=Shayne@c-68-60-62-95.hsd1.mi.comcast.net) |
| 03:15.49 | Dovid | that is this warning ? |
| 03:15.50 | Dovid | /usr/src/zaptel-1.2.16/ztd-eth.c:189: warning: initialization from incompatible pointer type |
| 03:15.50 | Dovid | <PROTECTED> |
| 03:15.53 | *** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net) |
| 03:19.25 | *** part/#asterisk S2AnGeL (n=S2AnGeL@CPE0014bf103d31-CM000039529869.cpe.net.cable.rogers.com) |
| 03:24.40 | *** join/#asterisk riddlebox (n=victoria@75-132-215-110.dhcp.stls.mo.charter.com) |
| 03:39.55 | nybble | greetings |
| 03:45.14 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:48.00 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 03:50.59 | *** join/#asterisk daveburr (i=Miranda@49.sub-70-193-239.myvzw.com) |
| 03:53.15 | *** join/#asterisk bmg505 (n=leon@196.209.180.81) |
| 04:02.23 | *** join/#asterisk asterisknerds (n=logger@66.7.124.15) |
| 04:02.23 | asterisknerds | <PROTECTED> |
| 04:02.54 | [TK]D-Fender | asterisknerds, You don't say! |
| 04:04.22 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 04:04.36 | Dovid | hehe |
| 04:04.40 | Dovid | TK: r u using 1.4.X? |
| 04:05.04 | [TK]D-Fender | Dovid, Not yet, but I think I should convert pretty soon... |
| 04:05.11 | [TK]D-Fender | Dovid, new things I need to test. |
| 04:05.37 | Dovid | TK: I want to use it on a test server for a client but I am scared shitless. |
| 04:05.46 | Dovid | bug tracker is still getting lots of complaints. |
| 04:05.49 | [TK]D-Fender | Dovid, promised some people to report back on SLA on Polyom and the new Devstate stuff |
| 04:06.06 | [TK]D-Fender | Dovid, What key points for the complaints? |
| 04:06.41 | Dovid | i dont know any specific. but when ever i go on to bug tracker there are new bug reports. mainly for 1.4.x |
| 04:07.01 | Dovid | and i dont wana use it and loose a client |
| 04:08.33 | [TK]D-Fender | Dovid, imperfect world. I'm sure there are bugs with 1.2.x, its jsut that with more and more people converting, you've just watching a population shift. I'm not sure we can attribute it to 1.4 being so different so much as being much more in focus |
| 04:09.56 | Dovid | i know when i was on 1.0.X when 1.2 came out there were lots of bugs in the begining and it went down. |
| 04:13.30 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 04:14.16 | Dovid | i am off to bed, (its 7 AM here) |
| 04:14.16 | Dovid | night |
| 04:21.37 | *** part/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 04:21.55 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 04:29.17 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 04:33.21 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 04:35.16 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:39.00 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 04:42.46 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 04:45.38 | *** join/#asterisk marcansoft (n=marcanso@160.10.7.121) |
| 04:46.30 | *** join/#asterisk marcan (n=marcanso@160.10.7.121) |
| 04:51.37 | *** join/#asterisk asterisknerds (n=asterisk@66.7.124.15) |
| 04:51.39 | asterisknerds | <PROTECTED> |
| 04:52.48 | bkruse_home | owned. |
| 04:53.47 | *** join/#asterisk asterisknerds (n=asterisk@66.7.124.15) |
| 04:53.49 | asterisknerds | <PROTECTED> |
| 04:54.07 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 04:57.15 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 05:00.49 | bkruse_home | asterisknerds: what are you trying to do? |
| 05:00.49 | bkruse_home | ............. |
| 05:00.50 | *** join/#asterisk Innatech (n=nospam@cpe-76-167-129-44.socal.res.rr.com) |
| 05:02.21 | *** join/#asterisk fab5freddy (n=vmware@bas1-montreal19-1177819133.dsl.bell.ca) |
| 05:03.44 | *** kick/#asterisk [asterisknerds!n=north@pdpc/sponsor/digium/Qwell] by Qwell (Fix your logbot^Wirc client.) |
| 05:06.25 | fab5freddy | how does one test out the ivr? |
| 05:06.32 | Qwell | fab5freddy: call it |
| 05:06.48 | fab5freddy | QWell: But which extension? |
| 05:06.53 | bkruse_home | lol |
| 05:06.55 | Qwell | whatever extension you gave it |
| 05:06.56 | Qwell | ~wikis |
| 05:06.59 | jbot | i heard wikis is http://www.voip-info.org |
| 05:06.59 | Qwell | ~book |
| 05:07.01 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:07.01 | bkruse_home | just quit while your ahead. |
| 05:07.02 | bkruse_home | lol |
| 05:07.02 | Qwell | You've got some reason to do... |
| 05:07.57 | Qwell | reading* |
| 05:07.59 | Qwell | wtf |
| 05:08.13 | Qwell | I'm usually pretty on top of typos |
| 05:09.54 | Qwell | *CLI> marko show birthday |
| 05:09.54 | Qwell | Happy 30th birthday Marko! |
| 05:09.54 | Qwell | *CLI> |
| 05:10.08 | Qwell | w00t, it worked |
| 05:11.08 | bkruse_home | nice! |
| 05:13.17 | Qwell | let's see how flamed I get on the -dev list for posting that there :P |
| 05:14.20 | bkruse_home | bleh |
| 05:14.26 | bkruse_home | was it merged into the 1.4 branch? |
| 05:15.06 | Qwell | no, just trunk |
| 05:15.11 | Qwell | and only if you enable it in menuselect |
| 05:15.29 | bkruse_home | i think its fine |
| 05:15.32 | bkruse_home | and cool :D |
| 05:15.39 | Qwell | it's been there for weeks, heh |
| 05:15.47 | bkruse_home | ya, i remember when it was committed |
| 05:16.45 | *** join/#asterisk jnc (n=jnc@205.234.240.46) |
| 05:17.35 | fab5freddy | what config files point to the extension of the ivr? sip.conf? |
| 05:17.46 | Qwell | extensions.conf |
| 05:18.10 | bkruse_home | ~book |
| 05:18.12 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:18.12 | bkruse_home | ~wiki |
| 05:18.18 | bkruse_home | ~thebook |
| 05:18.19 | jbot | rumour has it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:18.37 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 05:18.44 | jnc | asterisk-now gui making my head spin, trying to get it operational on an debian etch install. I've seen it work on an AsteriskNOW install, want to try on my debian box |
| 05:18.58 | bkruse_home | jnc: |
| 05:18.59 | bkruse_home | its not hard |
| 05:19.03 | bkruse_home | join /#asterisk-gui |
| 05:19.07 | jnc | oh sweet |
| 05:19.52 | [TK]D-Fender | fab5freddy, pastebin your dialplan and tell us what context your phone uses in there. |
| 05:19.53 | [TK]D-Fender | ~pb |
| 05:19.54 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 05:21.32 | fab5freddy | [TK]D-Fender: i am reading through extensions.conf now.. |
| 05:21.49 | fab5freddy | [TK]D-Fender: i also finally got my did and am ready to roll |
| 05:21.57 | bkruse_home | nice |
| 05:22.11 | [TK]D-Fender | fab5freddy, Hope you get your money's worth... |
| 05:23.01 | fab5freddy | [TK]D-Fender: if you go to a bar and spend $6 on a drink.. but spend $3.5/month on a phone line, where are you getting more value? |
| 05:23.23 | [TK]D-Fender | fab5freddy, easy answer... she was TOTALLY worth it ;) |
| 05:24.10 | *** join/#asterisk dos000 (n=ymo@CPE000f66912f92-CM0018c0c6147e.cpe.net.cable.rogers.com) |
| 05:24.15 | dos000 | howdy |
| 05:24.29 | [TK]D-Fender | doody |
| 05:24.49 | dos000 | anyone can tell me why i keep getting res_odbc: Error SQLConnect=-1 errno=2003 [unixODBC][MySQL][ODBC 3.51 Driver]Can't connect to MySQL server on '82.103.139.35' (4) |
| 05:24.59 | *** join/#asterisk PBXtech (i=Miranda@106.sub-70-193-67.myvzw.com) |
| 05:25.15 | dos000 | i made sure i can connect to that server from the command line |
| 05:25.32 | *** join/#asterisk asterisknerds (n=asterisk@66.7.124.15) |
| 05:25.32 | asterisknerds | <PROTECTED> |
| 05:25.42 | bkruse_home | dude, whats your problem. lol |
| 05:26.14 | PBXtech | sorry dont ban it. ill fix it. |
| 05:26.34 | bkruse_home | PBXtech: thats you? asterisknerds? |
| 05:26.45 | bkruse_home | what is it? |
| 05:26.49 | dos000 | i knwo for sure that 1) the server is running 2) the password is connect |
| 05:26.52 | PBXtech | yea just logging to a web page so i can read this chan offline |
| 05:27.17 | bkruse_home | PBXtech: oh, cool |
| 05:27.19 | dos000 | at least if it told me why it just cant connect .. is there a way to get verbose output ? |
| 05:27.19 | bkruse_home | nvm then :] |
| 05:27.36 | Qwell | dos000: reason 2003 - should be googlable |
| 05:27.39 | PBXtech | just need to fix the onjoin park :/ |
| 05:29.40 | fab5freddy | Does Asterisk enable a ivr by default? |
| 05:30.20 | Qwell | fab5freddy: only a demo |
| 05:31.09 | fab5freddy | Qwell: Now you have me that wiki to all the information.. but there was too much to deal with, do you have a link to a tutorial on how to create an ivr? |
| 05:32.03 | ezway | crime de grosse jrne encore ; |
| 05:32.11 | ezway | sorry |
| 05:33.27 | ezway | k |
| 05:35.46 | *** join/#asterisk ksteward (n=ksteward@71.174.94.243) |
| 05:36.46 | dos000 | qwell: unfortunately google is no help so far :) |
| 05:37.26 | dos000 | the actual error (4) does not seem to be out there on gooogle land |
| 05:37.59 | ksteward | anyone concerned for |
| 05:38.20 | ksteward | ... Asterisk based on the Verizon / Vonage suit? |
| 05:38.48 | dos000 | i cant figure what the hell the patent was for |
| 05:40.04 | ksteward | aside from creating some fear, it seems like Verizon can't really sue an open source project, right? |
| 05:40.29 | PBXtech | anyone can sue anyone |
| 05:41.31 | dos000 | if they sue software makers that will be crazy .. they usually go for servie providers |
| 05:41.59 | ksteward | but there is no single company behind Asterisk, or could / would they go after Digium; and could that halt Asterisk???? |
| 05:52.31 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 05:52.54 | luke-jr | ksteward: who are you kidding? |
| 05:53.03 | luke-jr | there certainly is a single company behind Asterisk |
| 05:53.06 | luke-jr | it's called Digium |
| 05:53.13 | ksteward | yes, i mentioned them. |
| 05:53.25 | luke-jr | ah, skipped that part XD |
| 05:53.27 | bkruse_home | ksteward: i dont think you understand the relationship between digium and asterisk |
| 05:53.43 | luke-jr | anyhow, what does either Verizon or Vonage have to do w/ Asterisk? |
| 05:54.14 | bkruse_home | voip? |
| 05:54.16 | bkruse_home | thats it. |
| 05:54.20 | luke-jr | heh |
| 05:54.23 | ksteward | if Verizon wins against Vonage due to it patent claims, then there is a precedant for further suits. |
| 05:54.33 | luke-jr | what patent claims? |
| 05:54.36 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 05:54.40 | bkruse_home | verizon has 0 grounds to even consider suing digium |
| 05:54.43 | bkruse_home | did you even read the suites? |
| 05:54.45 | luke-jr | when do we get to overthrow the corrupt US gov't? :p |
| 05:55.12 | ksteward | Could it then sue Digium for violation of Verizon patents? |
| 05:55.28 | luke-jr | ksteward: what patents?? |
| 05:55.37 | ksteward | i read someone else's review and evaluation of the patents |
| 05:55.54 | ksteward | hang on i'll find the web page again... |
| 05:57.03 | ksteward | http://ipurbia.com/2007/03/verizon-patent-analysis.html |
| 05:57.29 | bkruse_home | ksteward: your way over your head, stop why your ahead |
| 05:57.32 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 05:57.41 | bkruse_home | verizon would sue the voip industry at large, not just vonage |
| 05:57.59 | bkruse_home | vonage did step on toes, but asterisk was built from scratch according to rfc's, its COMPLETELY different than vonage |
| 05:58.02 | bkruse_home | just please, stop talking |
| 05:58.08 | ksteward | right, and so would that include Digium and/or Asterisk? |
| 05:58.22 | ManxPower | What I wonder is if vonage actually developed the technology that "violates" these patents. |
| 05:58.26 | bkruse_home | vonage == itsp asterisk == open source pbx vonage != digium asterisk != itsp |
| 05:58.35 | ManxPower | If not, who did and why are they not being sued? |
| 05:58.56 | J4k3 | you could possibly use asterisk to get yourself sued by verizon |
| 05:59.03 | J4k3 | but asterisk/digium is under no risk |
| 05:59.13 | J4k3 | is my basic review of the situation |
| 05:59.13 | Qwell | J4k3: You could probably use asterisk to get yourself sued by a lot of people |
| 05:59.14 | ksteward | J4k3, good! |
| 05:59.25 | J4k3 | Qwell: ;) |
| 06:00.35 | ManxPower | Somehow I suspect that Vonage did not actually develop anything. |
| 06:01.17 | ksteward | but then consider the legal issues/confusion that surfaced when SCO started going after LINUX (IBM esp.) |
| 06:01.40 | ksteward | could something similar happen with Asterisk (Digium)??? |
| 06:02.04 | ManxPower | ksteward: anyone can sue anyone. Fortunatly, Digium now has the cash to defend itself if required. |
| 06:02.16 | ksteward | obviously LINUX prevailed and SCO was defeated, but it wasn't until there were a lot of nervous people. |
| 06:02.58 | Qwell | ksteward: no they weren't |
| 06:02.59 | ksteward | SCO was puny compared to Verizon. It eventually ran out of $$$ |
| 06:03.10 | Qwell | all of SCO's suits are still going |
| 06:03.41 | J4k3 | ksteward: you can always sit around and wait for a reason to pee your pants. |
| 06:03.45 | ksteward | I thought IBM pretty much silenced SCO, and SCO is almost out of business. |
| 06:03.56 | Qwell | sure, but the suits are still ongoing |
| 06:04.04 | J4k3 | SCO was almost out of business before the suit, too. |
| 06:04.04 | ManxPower | Just wait until Verizon sues someone like Comcast or Cox Communications (cable TV companies with VoIP offerings) |
| 06:04.15 | J4k3 | the big cheque that microsoft sent them helped a bit (at least thats what I read...) |
| 06:04.16 | Qwell | ManxPower: that'll be fun |
| 06:04.23 | Qwell | try suing somebody who actually makes money, heh |
| 06:04.33 | bkruse_home | lol!!!!!! |
| 06:04.44 | J4k3 | ibm vs sco is tard vs tard. |
| 06:04.51 | bkruse_home | yep |
| 06:04.56 | J4k3 | two lame dying futureless companies |
| 06:05.00 | J4k3 | that won't just admit they suck and go away. |
| 06:05.11 | J4k3 | they gotta waste down all the cash first. |
| 06:05.12 | Qwell | IBM isn't going away any time soon |
| 06:05.16 | ksteward | is Digium really that flush with $$$ to survive the confusion that a Verizon law suite would cause? |
| 06:05.26 | bkruse_home | lol |
| 06:05.33 | J4k3 | Qwell: sadly... it'd be nice if they would. |
| 06:05.41 | h3x0r | manx: They aren't selling VoIP. They are selling VoATM |
| 06:05.50 | Qwell | I'm going to bed. This conversation is pointless. |
| 06:05.50 | ManxPower | ksteward: They are building a big new HQ. I hope they have some cash laying around. |
| 06:05.55 | h3x0r | Cable uses the Cisco MGX platform for VoATM |
| 06:05.57 | J4k3 | ATM over cable? eww. |
| 06:06.04 | ksteward | BUT... |
| 06:06.07 | bkruse_home | Qwell: totally agree, flame war |
| 06:06.11 | bkruse_home | cya bud |
| 06:06.13 | h3x0r | yes, most all packet cable networks are ATM |
| 06:06.22 | Qwell | heh |
| 06:06.31 | ksteward | how many companies would buy Digium products if they feared Verizon would put Digium out of business? |
| 06:06.33 | Qwell | I once said "I'm on an ATM ATM, ATM", and meant it |
| 06:06.35 | bkruse_home | lies! |
| 06:06.53 | ManxPower | J4k3: Huh? Cable TV is great for ATM. Scads of bandwidth so a 30% cell tax doesn't kill you. |
| 06:06.54 | bkruse_home | ksteward: we arent even in the same freaking business! |
| 06:06.59 | J4k3 | ksteward: only an idiot would think so, and idiots can't afford digium's products. |
| 06:07.24 | bkruse_home | thats like saying what happends between dell computers and people that fix computers |
| 06:07.53 | ManxPower | I can easily see how the Verizon patents might try to be applied to Digiums TDM products. |
| 06:08.20 | ksteward | so the consensus here is that Digium and Asterisk have absolutely no worries about Verizon. Good to hear! |
| 06:08.38 | coppice | at face value one of those verizon patents would seem to affect any voice system interconnecting the PSTN and a packet switched network. you need to read the details many times to see just how all encompassing something like this really is |
| 06:08.57 | ManxPower | Verizon has a patent on "ring all" feature. Asterisk has that feature. Digium paid the programmers to create the tools to create that feature. |
| 06:09.01 | J4k3 | Verizon mostly wanted to play at competition control. |
| 06:09.04 | bkruse_home | coppice: exactly, but not aimed directly at digium is the point |
| 06:09.38 | coppice | but law suits are always aimed at someone in particular :-) |
| 06:09.38 | ManxPower | ATM is, of course, a packet switched network |
| 06:09.58 | coppice | ATM isn't quite packet switched |
| 06:10.04 | h3x0r | no its a cell switched network |
| 06:10.06 | J4k3 | ManxPower: so I guess Verizon can sue anyone whos plugged more than one phone into a pots line. |
| 06:10.17 | h3x0r | packets have variable length |
| 06:10.21 | h3x0r | atm cells are always 53 bytes |
| 06:10.48 | h3x0r | you can also encapsulate TDM inside of ATM |
| 06:10.57 | h3x0r | and you can do LANE to map something like ethernet to ATM |
| 06:11.12 | h3x0r | ATM competes with SONET more directly than any "packet switched" network |
| 06:11.28 | coppice | ATM is determinstic. packet switched systems are inherent not so, though some have QoS features to try to fake determinism |
| 06:11.30 | h3x0r | VoATM is AAL5 voice |
| 06:11.31 | ManxPower | So no Voice over X.25? |
| 06:11.39 | h3x0r | yeah |
| 06:11.54 | h3x0r | Why dosen't asterisk have a VoATM module yet... hgeh |
| 06:12.04 | h3x0r | VoFR is useless |
| 06:12.29 | ManxPower | h3x0r: We expect to be doing VoFR soon. |
| 06:12.36 | h3x0r | wtf for? |
| 06:12.39 | ManxPower | Well, at least TRY to VoFR. 8-) |
| 06:12.45 | ksteward | well, given the confidence expressed here that Digium/Asterisk are untouchable by Verizon lawsuites, i'm going to go to bed and sleep easy now. |
| 06:12.55 | ManxPower | h3x0r: because most of the WAN is FR |
| 06:12.56 | ksteward | thanks |
| 06:13.01 | J4k3 | and the thing is |
| 06:13.02 | h3x0r | kste: digium dosent have enough cash |
| 06:13.05 | J4k3 | if digium disappears |
| 06:13.07 | h3x0r | for verizon to steal |
| 06:13.09 | J4k3 | who cares, the support is worldwide |
| 06:13.21 | J4k3 | if some jackass judge does something completely retarded, it shouldn't change anything much |
| 06:13.27 | bkruse_home | J4k3: agreed....... |
| 06:13.31 | J4k3 | and rememeber to quit voting for dipshits. |
| 06:13.34 | J4k3 | :| |
| 06:13.57 | h3x0r | or just base your company out of a different country |
| 06:13.58 | h3x0r | that dosent suck |
| 06:14.00 | h3x0r | like the US |
| 06:14.02 | bkruse_home | lol |
| 06:14.04 | bkruse_home | nigeria? |
| 06:14.11 | ManxPower | Canada |
| 06:14.11 | J4k3 | latvia |
| 06:14.18 | J4k3 | canada just sucks you dry. |
| 06:14.30 | h3x0r | canada lets you export crypto everywhere! |
| 06:14.31 | h3x0r | heh |
| 06:14.51 | ManxPower | Any country that has decriminalized marijuana is a good country in my book. |
| 06:14.57 | J4k3 | launch a sat to export crypto from. |
| 06:15.10 | J4k3 | "decriminialization" doesn't mean 'legalization' so its fairly worthless. |
| 06:15.17 | J4k3 | plus canadian pot is still overpriced. |
| 06:15.23 | h3x0r | the satellite is probably still under the jurisdiction of the country that launched it |
| 06:15.27 | J4k3 | so it didn't change anything much |
| 06:15.31 | h3x0r | just like a plane or boat |
| 06:15.51 | J4k3 | h3x0r: middle of the ocean, and don't leave fingerprints. |
| 06:15.56 | h3x0r | hahaha |
| 06:16.26 | J4k3 | these days that'd be pretty much asking to start WW3 |
| 06:19.06 | nybble | mmm.... crypto |
| 06:21.19 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 06:22.33 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 06:23.39 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 06:41.17 | h3x0r | yes! |
| 06:41.26 | h3x0r | its 100% france telecom compliant! |
| 06:41.26 | h3x0r | haha |
| 06:46.13 | jnc | maybe you guys are familiar, how to call asterisk from a sip softphone within the lan asterisk runs on |
| 06:46.21 | *** join/#asterisk Ankleteeth (n=chatzill@main.toble.com) |
| 06:46.27 | jnc | I connect, asterisk says the number is not in service, hangs up on me |
| 06:46.54 | jnc | (which is an improvement from earlier today when it spat garbled audio at me and hung up anyways) |
| 06:47.18 | jnc | should I be adding a service provider for random SIP calls? |
| 06:50.36 | pfn | damnit, why does my 7960 keep saying I have a parse error in SIPDefault.cnf |
| 06:52.14 | pfn | http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx |
| 06:52.18 | pfn | ah, that seems to have info |
| 06:58.04 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:13.24 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
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| 08:31.18 | DrCron | hmm, what would be a good codec for low power devices |
| 08:32.21 | coppice | you mean in the sub micro-watt range? :-) |
| 08:33.09 | DrCron | more in the ~200mhz range |
| 08:33.15 | DrCron | pocket pc/cell phone |
| 08:34.05 | coppice | sadly the cellphones contain great codecs, and you can't get to them |
| 08:34.26 | DrCron | at least yet |
| 08:34.45 | DrCron | or do you mean the hardware gsm codecs, |
| 08:35.02 | *** join/#asterisk cuco (n=diegoloc@62.90.10.53) |
| 08:35.12 | cuco | tzafrir_laptop: ping |
| 08:35.39 | cuco | tzafrir: ping? |
| 08:35.43 | coppice | the GSM or equivalent CDMA codecs are locked out from applications on most phones. some are starting to change, but slowly |
| 08:35.45 | tzafrir_laptop | here |
| 08:38.13 | *** join/#asterisk friedrich| (n=friedric@e177250237.adsl.alicedsl.de) |
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| 10:43.04 | *** part/#asterisk Geert (i=geert@irssi/staff/geert) |
| 10:49.59 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
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| 10:52.11 | EmleyMoor | Is there a variable which equals the context of the phone being used? |
| 10:54.47 | EmleyMoor | I'm also getting mains hum occasionally on my Zap phones |
| 10:57.59 | *** join/#asterisk nemski (n=nemesis@65.111.176.146) |
| 11:00.07 | tzafrir_laptop | EmleyMoor, ${CONTEXT} ? |
| 11:00.13 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 11:06.56 | EmleyMoor | tzafrir_laptop: Isn't that likely to be the context of the actual line in the dialplan? |
| 11:07.43 | tzafrir_laptop | yes. If you want a previous context, save it in a temporary var |
| 11:08.01 | EmleyMoor | I need the default context of the phone being used |
| 11:08.20 | EmleyMoor | As far as I can tell, there is no opportunity to save it in the first place |
| 11:10.12 | EmleyMoor | Ah, ${CONTEXT} does show the right context, even if the line is in a lower included one |
| 11:10.22 | EmleyMoor | Thanks |
| 11:15.46 | EmleyMoor | Now I can prevent people returning calls to numbers they cannot normally dial :-) |
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| 11:36.22 | *** join/#asterisk voltagex (n=voltagex@124-254-121-155-dsl.ispone.net.au) |
| 11:36.29 | voltagex | I found a bug in asterisk |
| 11:44.19 | EmleyMoor | voltagex: Describe it |
| 11:44.41 | voltagex | asterisk -r then type console dial then type console dial again |
| 11:45.58 | voltagex | crash |
| 11:47.38 | garreel | it's true |
| 11:53.40 | voltagex | EmleyMoor: 1.4.12 |
| 11:54.43 | voltagex | oops |
| 11:54.45 | voltagex | hang on |
| 11:54.59 | voltagex | version 1.4.2 |
| 12:00.46 | *** join/#asterisk friedrich| (n=friedric@e177250237.adsl.alicedsl.de) |
| 12:04.52 | voltagex | ... |
| 12:13.37 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
| 12:14.05 | voltagex | anyone interested in the bug I discovered in 1.4.2? |
| 12:15.00 | polerin | submitted a ticket yet? |
| 12:15.20 | voltagex | polerin: never done that before, how do I do that? |
| 12:15.47 | EmleyMoor | ... and is not likely to for the moment either |
| 12:16.13 | voltagex | bah, can't checkout the latest SVN |
| 12:16.55 | polerin | yeah that's a good start :P |
| 12:17.07 | *** join/#asterisk friedrich| (n=friedric@e177246183.adsl.alicedsl.de) |
| 12:17.16 | voltagex | doesn't seem to be reported yet |
| 12:17.21 | voltagex | well, the latest svn is broken |
| 12:17.26 | voltagex | collect2: ld returned 1 exit status |
| 12:19.25 | polerin | http://www.asterisk.org/developers/bug-guidelines |
| 12:19.50 | voltagex | hang on, something funky is going on here |
| 12:19.59 | voltagex | checking svn out again |
| 12:20.41 | voltagex | does that mean I can't report a bug in 1.4.2? That I have to do my testing against svn? |
| 12:21.02 | polerin | you just said that the hatest svn is broken corret? |
| 12:21.37 | voltagex | polerin: I'm not so sure now :S |
| 12:21.50 | voltagex | polerin: just realised my box may be a bit unstable |
| 12:21.56 | voltagex | it's hotter than I thought it was |
| 12:21.56 | polerin | heh |
| 12:22.16 | polerin | well, I'd test it for you but I'm playing with 1.2 |
| 12:22.20 | voltagex | only a PIII 1ghz so configuring takes a while |
| 12:22.40 | voltagex | in 1.2, try typing dial then dial again |
| 12:22.49 | voltagex | I wonder if it's in 1.2 as well |
| 12:23.04 | voltagex | Checked out revision 60708. |
| 12:23.06 | voltagex | ok |
| 12:23.18 | polerin | http://bugs.digium.com/main_page.php |
| 12:23.56 | voltagex | yes, I was reading that, but the way I interpreted that was they only wanted bugs reported for SVN? |
| 12:24.56 | polerin | tbh i'm not ever sure if that'sa current bug tracker link ;P |
| 12:25.18 | polerin | i havent' had coffee yet, but I'd work on finding out were to set up a ticket. |
| 12:25.23 | voltagex | ok |
| 12:25.34 | voltagex | well, I'm going to see if it's been fixed in SVN |
| 12:28.37 | voltagex | hope you're not drinking instant, bleargghh |
| 12:28.42 | polerin | where not were. Unless your ticket is going to turn into a person and start biting people |
| 12:29.13 | voltagex | well it bit me |
| 12:29.37 | voltagex | "I'm just trying to dial something from the console"..."ok" *click* |
| 12:32.46 | voltagex | shit... can't edit |
| 12:32.47 | voltagex | http://bugs.digium.com/view.php?id=9500 |
| 12:32.49 | voltagex | bad bug report |
| 12:32.57 | voltagex | but it has enough info in there to reproduce |
| 12:34.35 | voltagex | nope, even with verbose 100 and debug 10 there's no extra debug output : |
| 12:48.56 | polerin | is it crashing just the console or the entire thing? |
| 12:49.54 | voltagex | entire server |
| 12:50.03 | voltagex | the process dies |
| 12:55.22 | omm | Is there a way to automatically pause a member of a queue when they dont answer the phone? |
| 12:59.57 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
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| 13:57.56 | nsphang88 | any assistance around? |
| 13:58.31 | EmleyMoor | nsphang88: What kind of assistance? |
| 13:58.48 | nsphang88 | would like to know if there is anyone here who does freelance asterisk server setup |
| 14:04.59 | bulle | nsphang88: better state the region you want the service in aswell, this is an international channel |
| 14:05.22 | *** join/#asterisk friedrich| (n=friedric@e177249164.adsl.alicedsl.de) |
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| 14:16.03 | brian | Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) |
| 14:16.25 | brian | I keep getting flooded with that after: -- agent_call, call to agent 'justin' call on 'SIP/proxy01.sipphone.com-08164120' |
| 14:17.26 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 14:23.48 | tzafrir_laptop | I'm trying to build h323 of asterisk 1.2.17 , and I can't figure out what provides the "opt" target for the directory h323 |
| 14:28.55 | *** join/#asterisk rnovotny22 (n=rnovotny@70-56-172-170.mpls.qwest.net) |
| 14:33.57 | tzafrir_laptop | how nice. h323 been broken for about a month |
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| 14:49.24 | *** part/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 14:49.31 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 14:49.33 | blitzrage | wow... I had no idea this was valid: Goto(context,extension,+5) |
| 14:54.45 | *** join/#asterisk SmaxTheFrog_ (n=M_A_X@i577A447F.versanet.de) |
| 14:54.48 | SmaxTheFrog_ | hi |
| 14:55.40 | SmaxTheFrog_ | is it possible to use timetable for exsample to switch on/off answeringmaschine using realtime ext with mysql ? |
| 14:59.38 | nsphang88 | would like to know if there is anyone here who does freelance asterisk server setup - remotely installation |
| 15:00.15 | brian | where is bkw ;( |
| 15:00.15 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 15:00.15 | SmaxTheFrog_ | nsphang88 no problem with config on .conf basis |
| 15:01.01 | nsphang88 | SmaxTheFrog_: i mean the entire installation including the software + configurations |
| 15:02.03 | SmaxTheFrog_ | nsphang88 if drivers for isdn etc. are available ... jep |
| 15:03.54 | SmaxTheFrog_ | nsphang88 if you are interested send a mail to info@dasin.de |
| 15:04.56 | SmaxTheFrog_ | anyone has an idea concerning my realtime ext problem with timetables ? |
| 15:08.26 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:12.15 | brian | Asterisk won't play any of my sound files... |
| 15:14.20 | *** join/#asterisk voltagex (n=voltagex@124-254-121-155-dsl.ispone.net.au) |
| 15:15.33 | ManxPower | Brian: do you have any Digium (or other brand) cards installed? |
| 15:16.06 | brian | ManxPower: no |
| 15:16.30 | brian | ManxPower: do i need a timer for that to work? |
| 15:17.31 | ManxPower | Brian: no, but there is a bug where if you have a T-1/E-1 card installed and configured but no line connected asterisk will not play audio on VoIP calls. |
| 15:17.54 | ManxPower | What user are you running asterisk as? |
| 15:18.44 | Qwell | I'm betting NAT issue |
| 15:18.46 | brian | asterisk |
| 15:19.06 | brian | Qwell: it transmits my voice fine |
| 15:19.40 | ManxPower | Brian: and I assume all the /usr/lib/asterisk/sounds (maybe it is /var/lib/asterisk/sounds, I can never remebmer) is readable by the asterisk user? |
| 15:19.47 | Qwell | var |
| 15:19.49 | brian | It's not in there |
| 15:20.08 | brian | IT's in /home/asterisk/asterisk/callManager/sounds |
| 15:20.12 | Qwell | brian: what does it do/say when it tries to play them? |
| 15:20.26 | brian | Apr 8 15:11:22 WARNING[14693]: chan_sip.c:2575 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) |
| 15:20.30 | brian | Lots of that |
| 15:20.36 | ManxPower | Brian: are you giving Playback the full patch? |
| 15:20.56 | ManxPower | Brian: there is 10 mins of my lfe I will never get back |
| 15:20.56 | ManxPower | ~codecs |
| 15:21.06 | jbot | i heard codecs is http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=ulaw' and see if that works. Anyone that tells you to use 'allow=all' is an idiot as it usually causes audio problems, or Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc |
| 15:21.06 | brian | I'm using AGI to play it |
| 15:21.08 | brian | And yes I'm giving it the full path |
| 15:21.14 | ManxPower | Brian: not when you are troubleshooting you are not. |
| 15:21.23 | brian | I already tried messing with codecs. |
| 15:21.28 | brian | I have disallow=all |
| 15:21.40 | brian | and then allow=ulaw, allow=alaw, allow=slin, allow=gsm |
| 15:21.54 | brian | when I had just allow=ulaw I had that problem |
| 15:22.04 | brian | Then when I added allow=slin it went away |
| 15:22.09 | brian | And then I added allow=gsm and it came back |
| 15:22.11 | ManxPower | Brian: do not do that. allow=ulaw (or allaw=alaw if you are not int he USA) |
| 15:22.15 | brian | But regardless, it never played the audio file |
| 15:22.20 | ManxPower | remove the allow=sln you never want to allow=sln |
| 15:22.41 | ManxPower | Brian: I think you have multiple problems. We have to fix them all |
| 15:23.14 | brian | It worked on my FreeBSD server. |
| 15:23.24 | ManxPower | Brian: make it disallow=all and allow=ulaw remove all the other allow= lines, and as I said you never want to allow=sln |
| 15:23.28 | brian | And then I migrated to Amazon EC2 (Linux) and problems started to happen |
| 15:23.32 | brian | ManxPower: done |
| 15:23.59 | ManxPower | Brian: now do you get an error messages at all when trying to run a playback from the Dialplan? |
| 15:24.22 | brian | haven't tried to do it from the dial plan next |
| 15:24.26 | brian | that was going to be my next try |
| 15:24.38 | ManxPower | is this a production machine? |
| 15:24.42 | brian | Not yet |
| 15:24.46 | ManxPower | good. |
| 15:24.50 | brian | I still have lots to do |
| 15:24.54 | brian | To make it "production" |
| 15:25.00 | ManxPower | try using playback to play the file. |
| 15:25.42 | brian | btw... |
| 15:25.58 | brian | On a SIP to SIP call the sound files play fine |
| 15:26.22 | brian | It's only when I call using the toll free number there is problems |
| 15:26.22 | ManxPower | what specific calls does it not play? |
| 15:26.42 | brian | The "Agent logged in" sound file plays fine |
| 15:26.48 | ManxPower | lets see what happens with playback |
| 15:27.34 | ManxPower | Qwell: I'm betting it is a supervision issue, but we'll know when he tries Playback |
| 15:28.47 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 15:28.52 | brian | Playback worked fine |
| 15:29.03 | brian | Its when I try to do STREAM FILE from AGI it doesn't work |
| 15:29.37 | *** join/#asterisk darken_darken (n=marco@201.173.77.83.cust.bluewin.ch) |
| 15:29.39 | ManxPower | Brian: now try running Answer in the dialplan as the priority before your AGI |
| 15:29.55 | brian | Already is like that |
| 15:30.19 | ManxPower | ok. stop asterisk and start asterisk as "asterisk -cvvv" BTW, what version of asterisk are you running? |
| 15:30.26 | brian | And my host supports early media anyways. |
| 15:30.27 | ManxPower | then try the AGI and watch the console |
| 15:31.57 | brian | Asterisk 1.2.14 |
| 15:32.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:32.06 | brian | It's patched |
| 15:32.45 | ManxPower | stderr from AGIs always go to the tty asterisk is attached to. when starting asterisk as a daemon and then connecting to it via asterisk -rvvv the AGI erres will not be on the console you are watching,. "asterisk -cvvv" fixes that issue when debugging. |
| 15:33.31 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 15:37.03 | brian | http://rafb.net/p/KlnjRw87.html |
| 15:37.15 | brian | There is no error |
| 15:39.00 | brian | I found this in the docs |
| 15:39.01 | brian | Note: streamFile is apparently unstable in AGI, may want to use |
| 15:39.01 | brian | execute( 'PLAYBACK', ... ) instead (according to the Wiki) |
| 15:39.42 | ManxPower | Brian: The wiki is a cesspool of misinformation, but sometimes it is correct. Give that a try. |
| 15:40.05 | ManxPower | btw, you are running FastAGI, not AGI. |
| 15:41.32 | brian | That worked |
| 15:41.34 | brian | Yeah I know |
| 15:41.43 | brian | It's kind of the same thing though |
| 15:41.51 | brian | Using execute('PLAYBACK ...') fixed it |
| 15:42.04 | brian | apparently streamFile is screwy on Linux but works fine on FreeBSD |
| 15:42.09 | brian | weird |
| 15:42.17 | brian | Apr 8 15:41:11 WARNING[15144]: chan_sip.c:2575 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) |
| 15:42.21 | brian | I'm still being flooded with that |
| 15:42.24 | brian | There is at least 20 of those |
| 15:42.34 | brian | It happens after: |
| 15:42.35 | brian | <PROTECTED> |
| 15:42.35 | brian | <PROTECTED> |
| 15:42.59 | ManxPower | Brian: is there ANY errors from your AGI? |
| 15:43.02 | brian | no |
| 15:43.08 | brian | the only error is that WARNING |
| 15:43.14 | mmartinn | IIRC the stream file thing was an issue, but it got resolved |
| 15:43.41 | brian | It still doesn't work on Gentoo Linux Asterisk...I guess they are behind. |
| 15:43.47 | ManxPower | Brian: no, I mean in stderr of the AGI, which would NOT show in the asterisk console, since you are using a socket.; |
| 15:43.59 | mmartinn | brian: I run Gentoo and have no problems with it |
| 15:44.02 | brian | I'm using asterisk -cvvv like you said |
| 15:44.06 | ManxPower | Brian: hundreds of people run gentoo |
| 15:44.09 | brian | mmartinn: it might be because I'm on Xen? |
| 15:44.28 | ManxPower | Brian: "asterisk -cvvv" will no you no good if you are running AGI via a socket |
| 15:44.28 | mmartinn | There's too many possibilities for me to even try to speculate :) |
| 15:44.48 | brian | ManxPower: I have debugging on my FastAGI |
| 15:44.50 | brian | ManxPower: No errors |
| 15:45.07 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-76-107.dsl.peoril.sbcglobal.net) |
| 15:45.40 | ManxPower | Brian: I'm sure it is something very simple, but I am out of ideas. Your issue is not a typical one, but your setup is not typical either. |
| 15:45.54 | brian | It's running on an Amazon EC2 instance |
| 15:46.07 | mmartinn | brian: Are you using an AGI framework that is trying to mangle the file? |
| 15:46.11 | ManxPower | I've never hear of Amazon EC2 |
| 15:46.20 | brian | mmartinn: It's not trying to mangle the file |
| 15:46.34 | brian | mmartinn: It just sends "STREAM FILE..." to Asterisk |
| 15:46.48 | mmartinn | brian: okay, not sure myself then |
| 15:47.04 | mmartinn | brian: perhaps try a framework? =P |
| 15:47.18 | brian | I'm using a framework |
| 15:48.14 | brian | EXEC PLAYBACK /home/asterisk/asterisk/callManager/sounds/jtv-greeting works |
| 15:48.18 | mmartinn | In my experience, some of those frameworks don't "just send STREAM FILE..." |
| 15:48.20 | brian | while STREAM FILE doesn't |
| 15:48.29 | brian | mmartinn: I just told you it did. |
| 15:48.33 | brian | mmartinn: I know for a fact. |
| 15:48.41 | mmartinn | brian: okay, okay... |
| 15:48.50 | brian | It's not trying to do magic. |
| 15:49.00 | brian | It's just a simple interface to Fast AGI (or AGI) |
| 15:49.00 | mmartinn | brian: I've seen some that do try to do magic. |
| 15:49.18 | brian | ugh |
| 15:49.23 | brian | so what about |
| 15:49.30 | brian | Apr 8 15:41:11 WARNING[15144]: chan_sip.c:2575 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) |
| 15:49.33 | brian | How do I fix that? |
| 15:49.38 | brian | I only have disallow=all and allow=ulaw |
| 15:50.07 | brian | It happens when the beep sound plays on the agent's side to let them know there is a new caller |
| 15:51.02 | ManxPower | Brian: I have never seen 1 specific thing that causes that error. |
| 15:51.28 | brian | Well, when I added allow=slin it stopped displaying that error. |
| 15:51.40 | brian | But you told me that was bad. |
| 15:51.56 | mmartinn | brian: Have you tested stream file and exec playback with a file from /var/lib/asterisk/sounds? |
| 15:51.59 | ManxPower | Brian: but sip clients do not understand sln so it would be pointless to try allowing that codec. |
| 15:52.13 | ManxPower | mmartinn: good suggestion |
| 15:52.22 | brian | mmartinn: not yet |
| 15:52.45 | Qwell | Why are you calling an agent directly? |
| 15:54.26 | ManxPower | Wow! I have finally come up with a reason to use an answering machine! |
| 15:54.34 | mmartinn | brian: only ask because I've never attempted to play files outside that location, nor have I used absolute paths. |
| 15:54.40 | brian | Qwell: Because there is only one agent |
| 15:54.58 | brian | Qwell: And this isn't really being used as a "technical support" line or whatever |
| 15:55.14 | ManxPower | mmartinn: full path lets you play files from anywhere. |
| 15:55.26 | Qwell | calling agents does funky things |
| 15:55.32 | mmartinn | ManxPower: right... I just never tried it :) |
| 15:55.42 | ManxPower | but it is still a good suggestion. Perhaps his sound files are stereo, for example. |
| 15:55.50 | brian | It's worked fine for me ManxPower |
| 15:56.10 | brian | I only have one agent. |
| 15:56.21 | Qwell | try not using an agent, and it'll probably work fine |
| 15:56.33 | brian | If I don't use an agent how do I queue up? |
| 15:56.58 | ManxPower | Brian: you set member= lines in queues.conf |
| 15:57.16 | brian | ManxPower: You still need to Agentlogin |
| 15:57.33 | brian | Just being a member of the queue isn't good enough. |
| 15:57.40 | ManxPower | Brian: not in my experience |
| 15:58.10 | ManxPower | I'll put my actual produiction queues.conf on pastebin for you |
| 15:58.11 | brian | If I'm just a member, it hangs up on me. |
| 15:58.23 | ManxPower | we NEVER use aqentlogin |
| 15:58.39 | brian | But if I login using Agentlogin, it doesn't hang up on me. |
| 15:58.54 | *** join/#asterisk ming_zym (n=ming_zym@124.254.53.141) |
| 15:58.57 | brian | So I figured I'd use Agentlogin since the idea is for it not to hang up on me. |
| 15:59.21 | brian | I'm all for a better solution though because agent channels really suck they don't even listen to DTMF |
| 15:59.52 | brian | So I can't even add cool stuff for the "agent" to do, like block the caller, etc. |
| 15:59.53 | ManxPower | Brian: http://pastebin.ca/429913 |
| 16:00.19 | Qwell | ManxPower: what do the f,b,c stand for? |
| 16:00.24 | Qwell | lines on a phone? |
| 16:00.37 | brian | Where do you get SIP/0004f201d497-f from anyways? |
| 16:00.50 | Qwell | brian: sip.conf |
| 16:01.02 | brian | See the problem with this solution is that they need to be able to change the phone number it calls. |
| 16:01.10 | ManxPower | Brian: That is the 6th line appearance on the SIP device with the MAC of 0004f201d497 |
| 16:01.40 | ManxPower | Brian: you can use chan_local if you want to. |
| 16:01.56 | brian | Please tell me how to do that and I'll do it! |
| 16:01.59 | brian | At least point me to a tutorial. |
| 16:02.01 | ManxPower | since SIP accounts ARE NOT EXTENSIONS, I see no reason to make them look like extensions |
| 16:02.04 | Qwell | brian: I told you last night |
| 16:02.07 | brian | I'm still learning about Asterisk... |
| 16:02.12 | Qwell | there is a text file in the 1.4 doc/ dir |
| 16:02.13 | ManxPower | Qwell: I'll paste an example |
| 16:02.15 | brian | Qwell: I tried to do that |
| 16:02.21 | ManxPower | Qwell: he is using 1. |
| 16:02.23 | ManxPower | 1.2 |
| 16:02.26 | brian | Qwell: It did not work |
| 16:02.34 | brian | Of course I'm using 1.2 it's stable. |
| 16:02.51 | brian | 1.4 is good if you want Asterisk to crash all the time. :( |
| 16:03.00 | Qwell | Have you reported any bugs? |
| 16:03.10 | brian | I never tried 1.4 |
| 16:03.17 | brian | Because of all the crash bugs |
| 16:03.23 | Qwell | name one |
| 16:03.28 | brian | I don't know. |
| 16:03.32 | brian | Someone told me not to use it. |
| 16:04.03 | ManxPower | brian: http://pastebin.ca/429918 |
| 16:04.31 | brian | Besides 1.4 isn't in Gentoo portage yet |
| 16:04.40 | brian | And if it's not in portage that usually means it's not ready yet. |
| 16:04.49 | Qwell | don't use asterisk packages |
| 16:04.50 | Qwell | they all suck |
| 16:05.24 | brian | It's just Asterisk with patches. |
| 16:05.25 | ManxPower | Brian: asterisk is one of like 3 pieces of software I always compile by hand and do not use packages |
| 16:05.33 | Qwell | ManxPower: what are the other 2? |
| 16:05.41 | Qwell | and if you say libpri/zaptel, I'm gonna smack you ;P |
| 16:05.53 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 16:05.53 | brian | Gentoo doesn't really add extras though. |
| 16:06.00 | brian | It just adds patches, like for the SIP DOS vulnerability |
| 16:06.01 | Qwell | No, they just butcher it |
| 16:06.11 | Qwell | brian: you mean...the one that's...already in 1.2.17? |
| 16:06.15 | ManxPower | Qwell: there are 40 pending issues in bugs.digium.com for 1.4. I strongly doubt all of them are trivial issues |
| 16:06.28 | Qwell | ManxPower: and the other 400 are 1.2? ;) |
| 16:06.55 | ManxPower | Qwell: ClamAV and Spamassassin and (I forgot this one) Anomy Sanatizer. |
| 16:07.03 | Qwell | ahh, yeah |
| 16:07.30 | ManxPower | i.e. all programs that are under heavy active development and change for the better frequently |
| 16:08.49 | brian | Okay well |
| 16:09.10 | brian | This isn't really heavy duty or anything so I can cheat a little and use Asterisk from portage. |
| 16:09.31 | ManxPower | Qwell: I upgraded from CVS to 1.2.4 then to 1.2.13. I'm not usually in the mood to be beaten to death by users so I try to minimize upgrading |
| 16:09.53 | brian | Hay guise! |
| 16:10.00 | brian | can u halp me |
| 16:10.28 | ManxPower | Qwell: I also have to convert all my macros and dialplans to 1.2 syntax at least |
| 16:10.48 | brian | I still have no idea based on your example how to do it. |
| 16:11.01 | brian | So what if I dial into a local channel then what? |
| 16:11.16 | brian | How is the queue supposed to find this local channel? |
| 16:11.31 | brian | Do I add the local channel as a queue member, or the person I'm calling with SIP? |
| 16:11.55 | ManxPower | member=Local/extension@context |
| 16:12.09 | brian | And what exactly does that do? |
| 16:12.14 | ManxPower | then you can manage the dialed number in that. |
| 16:12.53 | brian | Does that basically just go to my extensions.conf |
| 16:13.00 | ManxPower | Brian: Instead of sending the call to a specific agent, it sends it to a dialplan extension so you can do things like look up the required destination for Dial from database or something |
| 16:13.16 | brian | So I could have that context be an AGI? |
| 16:13.45 | brian | ManxPower: The problem with that is... |
| 16:13.48 | ManxPower | Brian: anything you can put in the dialplan |
| 16:14.01 | brian | ManxPower: Won't that hang up and redial everytime? |
| 16:14.09 | ManxPower | huh? |
| 16:14.25 | brian | ManxPower: After they talk to one person waiting in the queue, won't it hang up and then redial them for the next person waiting? |
| 16:14.35 | *** join/#asterisk friedrich| (n=friedric@e177249164.adsl.alicedsl.de) |
| 16:14.47 | ManxPower | Brian: yes. Instead of that utter silly and stupid way of the agent not hanging up |
| 16:14.54 | brian | That's not what I want! |
| 16:15.09 | brian | ManxPower: There is only *one member* |
| 16:15.11 | ManxPower | Brian: then you can't use chan_local |
| 16:15.13 | brian | Do you realize how ridiculous that would be? |
| 16:15.21 | ManxPower | Huh? Not at all |
| 16:15.34 | brian | There is going to be at least 20 people in the queue |
| 16:15.38 | brian | guaranteed |
| 16:15.51 | brian | that means his phone will hang up and then ring 20 times |
| 16:15.57 | ManxPower | you just press the answer button on your phone and use a heasset. |
| 16:15.58 | brian | what if someone else calls him while asterisk is trying to call him? |
| 16:15.58 | ManxPower | headset too |
| 16:16.10 | brian | everything will screw up |
| 16:16.37 | ManxPower | we don't have that problem because we never send normal calls to the same call appearance as queue calls. |
| 16:16.37 | brian | There has to be an easy way to do this |
| 16:16.50 | ManxPower | Asterisk's queue system sucks. |
| 16:17.11 | brian | I'm starting to think I'd be better off queueing in AGI. |
| 16:18.02 | brian | Is there anyway to get chan_agent to respond to DTMF? |
| 16:18.08 | ManxPower | What specifically are you using AGI for right now? |
| 16:18.17 | brian | Not anything really yet. |
| 16:18.28 | brian | Pretty much everything I did in FastAGI I could of just put in the dial plan. |
| 16:18.38 | ManxPower | and do you have the unable to write frame messages when you don't use AGI? |
| 16:18.51 | brian | But it's going to get more complicated |
| 16:18.58 | brian | ManxPower: Why would AGI cause a codec problem? |
| 16:19.07 | brian | ManxPower: That doesn't even make sense |
| 16:19.51 | *** join/#asterisk darken_darken (n=marco@201.173.77.83.cust.bluewin.ch) |
| 16:19.57 | ManxPower | Brian: You are correct, but that is about the only thing we have not tried yet. BTW, did playing a sound in /var/lib/asterisk/sounds instead of your own sound files still cause the error |
| 16:20.26 | ManxPower | and the error message might not be cause by a codec issue |
| 16:20.47 | brian | ManxPower: You don't get it. |
| 16:21.03 | brian | ManxPower: The problem occurs when asterisk plays the agent beep noise |
| 16:21.33 | brian | ManxPower: The beep that lets the agent know that there is a caller on the line. |
| 16:22.33 | ManxPower | Brian: so YOU are not playing the beep? |
| 16:23.22 | ManxPower | I assume you have looked in /etc/asterisk/asterisk.conf to confirm the asterisk sounds are actually installed where asterisk.conf says they are? |
| 16:23.50 | brian | ManxPower: The beep plays |
| 16:24.04 | brian | ManxPower: But 20 warning messages scroll when the beep plays |
| 16:24.55 | ManxPower | so you actually hear the beep? |
| 16:25.24 | brian | yes |
| 16:26.09 | ManxPower | messages to the console are async, so those error messages could be from something right after the beep. |
| 16:26.30 | ManxPower | the only thing we have not done is eliminate the AGI |
| 16:26.58 | brian | No |
| 16:27.03 | brian | It's from my PLAYBACK of my sound file |
| 16:27.05 | brian | Not the Beep |
| 16:27.12 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 16:27.23 | brian | I commented out the playback and it stopped doing it |
| 16:27.32 | ManxPower | Brian: and if you PLAYBACK an ASTERISK sound file do you still get the same problems |
| 16:27.41 | brian | I'm going to check right now |
| 16:30.57 | brian | No it doesn't happen when I play a asterisk sound file |
| 16:31.03 | brian | But I can't stream file a asterisk sound file either |
| 16:31.08 | brian | So streamFile is apparently broken |
| 16:32.06 | ManxPower | not in 1.2.14 it isn't |
| 16:33.05 | brian | well that is the version I'm using |
| 16:33.09 | brian | and it doesn't work |
| 16:33.17 | brian | so that would make it broken would it not |
| 16:34.25 | *** join/#asterisk friedrich| (n=friedric@e177249164.adsl.alicedsl.de) |
| 16:35.00 | ManxPower | what happens when you do a: su -lc "ls -l /var/lib/asterisk/sounds/vm-login.*" asterisk |
| 16:35.46 | ManxPower | Brian: if it was broken we would have hundreds of reports of it. |
| 16:35.58 | brian | It asks me for a password. |
| 16:36.07 | ManxPower | do it as root |
| 16:36.21 | brian | -rw-r----- 1 asterisk asterisk 3993 Apr 6 20:52 /var/lib/asterisk/sounds/vm-login.gsm |
| 16:36.30 | brian | my permissions are correct |
| 16:36.47 | brian | The only thing is that asterisk is in the wheel group and not asterisk group |
| 16:36.54 | ManxPower | Brian: paste the streamfile line |
| 16:37.36 | brian | <PROTECTED> |
| 16:37.38 | brian | That works |
| 16:37.48 | brian | agi.streamFile('tt-monkeys') |
| 16:37.50 | brian | That didn't work |
| 16:37.58 | ManxPower | what language? |
| 16:38.10 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 16:38.50 | ManxPower | well, what programming language |
| 16:39.44 | *** part/#asterisk Xteven (n=deepstar@void.singularity.be) |
| 16:41.23 | brian | ManxPower: Python |
| 16:41.29 | brian | ManxPower: That's pretty irrelevant though |
| 16:41.56 | ManxPower | Brian: not really. It could easily be a bug in the Python AGI library |
| 16:42.18 | brian | NFO:FastAGI:Send Command: "STREAM FILE tt-monkeys '' 0" |
| 16:42.36 | brian | That's all it does when you use the streamFile method |
| 16:44.50 | brian | Why do codec errors occur when I play *my* sound file though? |
| 16:44.50 | ManxPower | what is that 0 and extra quote? |
| 16:44.50 | brian | That's part of the parameters |
| 16:45.07 | brian | Within the quotes, would be the escape keys |
| 16:45.10 | ManxPower | (11:30:44) brian: No it doesn't happen when I play a asterisk sound file |
| 16:45.11 | ManxPower | (11:30:49) brian: But I can't stream file a asterisk sound file either |
| 16:45.21 | ManxPower | apparently you can't streamfile an asterisk sound file either |
| 16:45.43 | brian | Maybe the API changed? |
| 16:46.00 | brian | I was using 1.2.13 on the FreeBSD box |
| 16:46.31 | brian | ~agi |
| 16:46.41 | jbot | somebody said agi was the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 16:47.02 | wunderkin | if it is a new install why arent you using 1.2.17? oh because it is too hard to install manually right? :P |
| 16:47.14 | brian | no |
| 16:47.16 | brian | i'm lazy |
| 16:47.27 | brian | okay?!? |
| 16:48.18 | ManxPower | what Python AGI lib are you using? |
| 16:49.50 | *** join/#asterisk darken_darken (n=marco@201.173.77.83.cust.bluewin.ch) |
| 16:52.11 | JaviG | can I ask a basic question here? |
| 16:53.04 | Nugget | You just did! :) |
| 16:53.38 | brian | ManxPower: StarPy |
| 16:54.35 | JaviG | jaja... ok. what linux distro or OS do u recommend for asterisk? |
| 16:54.51 | russellb | windows vista |
| 16:55.21 | Qwell | bah, 3.11 |
| 16:55.41 | JaviG | I know any Linux or UNIX will work, and I have used CentOS and Fedora in the past |
| 16:55.43 | brian | Usage: STREAM FILE <filename> <escape digits> [sample offset] |
| 16:55.57 | brian | Yup, the syntax is right ManxPower |
| 16:56.01 | JaviG | I wanted to know what you people recommend... |
| 16:56.12 | brian | you people?!?! |
| 16:56.21 | ManxPower | JaviG: The one you are most familiar with |
| 16:56.46 | GreyFoxx | JaviG: If there is a particular distro you are most familiar with, stick to it |
| 16:57.51 | JaviG | ok. thanks for the answer. |
| 16:57.55 | ManxPower | Brian: the extra quote still looks bad. In the asterisk console do an "agi debug" and paste the output of the call and AGI on something like pastebin |
| 16:57.58 | *** join/#asterisk [shodan] (n=shodan@ip115.99-113-216.pppoe4.joliette.intermonde.net) |
| 17:00.37 | brian | AGI Rx << EXEC STREAM FILE /home/asterisk/asterisk/callManager/sounds/jtv-greeting "" 0 |
| 17:00.37 | brian | <PROTECTED> |
| 17:00.38 | brian | Apr 8 17:00:12 WARNING[16118]: res_agi.c:1115 handle_exec: Could not find application (STREAM) |
| 17:01.09 | brian | apparently the application stream isn't there |
| 17:01.26 | ManxPower | look at that. |
| 17:01.44 | ManxPower | It is trying to EXEC the APPLICATION streamfile. there is no such application |
| 17:01.54 | russellb | s/EXEC// |
| 17:01.58 | brian | No |
| 17:02.03 | brian | I did that. |
| 17:02.07 | ManxPower | it should read something like AGI Rx << STREAM FILE /home/asterisk/asterisk/callManager/sounds/jtv-greeting "" 0 |
| 17:02.38 | ManxPower | do it again, only correctly this time |
| 17:05.58 | brian | http://rafb.net/p/FyHHbm84.html |
| 17:07.51 | ManxPower | <PROTECTED> |
| 17:08.33 | brian | yes |
| 17:09.31 | MACscr | hmm, when someone calls my ivr, its not recognizing their inputs. Any idea what it could be? I have had this issue before when calling from a sip phone to someone elses ivr, but never with a regular phone |
| 17:09.42 | ManxPower | in your agi do whatever you do in python to just sleep for 30 seconds. I want to make sure we are not seeing those errors in the wrong place. |
| 17:10.11 | ManxPower | MACscr: usually it is caused by your zaptel gains being too high or too low |
| 17:10.45 | MACscr | is zaptel still used for sip? |
| 17:10.55 | ManxPower | Brian: heck, sleep for 10 seconds before and after |
| 17:11.06 | ManxPower | MACscr: you did not say you are using sip |
| 17:11.20 | MACscr | sry, its all sip |
| 17:11.36 | MACscr | i just meant calling in from a regular phone to my sip did |
| 17:11.38 | MACscr | DID |
| 17:11.44 | ManxPower | MACscr: what other important information did you leave out? |
| 17:12.18 | MACscr | my apologies. I unfortunately automatically think sip when i think of asterisk |
| 17:12.23 | MACscr | its my fault |
| 17:12.28 | ManxPower | MACscr: many providers have problems with SIP and DTMF. I have no suggestions on how to fix that |
| 17:12.40 | MACscr | thanks |
| 17:12.47 | brian | ManxPower: I slept for 10 seconds same shit |
| 17:13.22 | ManxPower | Brian: no change in the order of the messages? |
| 17:13.28 | MACscr | correcto mundo |
| 17:13.54 | brian | I think it *could* be the single quotes instead of double quotes |
| 17:14.45 | brian | But I seriously doubt it. |
| 17:15.35 | ManxPower | uh, in the pastebin they are single quotes, in the paste to the channel they are double quotes. |
| 17:15.37 | ManxPower | which are you suing |
| 17:15.40 | ManxPower | using that is |
| 17:16.00 | brian | Did you not look at the paste? |
| 17:16.22 | ManxPower | AGI Rx << STREAM FILE /home/asterisk/asterisk/callManager/sounds/jtv-greeting '' 0 |
| 17:16.50 | ManxPower | silly me, I assumed you did not change the quoting when you fixed the EXEC stream file problem |
| 17:17.35 | brian | I doubt that's the problem. |
| 17:18.02 | ManxPower | Brian: even though the actual AGI docs for stream file specifically say to use double quotes? |
| 17:18.13 | ManxPower | pbx-1*CLI> show agi stream file |
| 17:18.15 | brian | But it worked fine on 1.2.13 |
| 17:18.35 | ManxPower | well then go back to 1.2.13!!! |
| 17:18.37 | brian | why would they change it in 1.2.14 to break if you use single quotes |
| 17:19.26 | brian | 1.2.13 = SIP DOS |
| 17:19.26 | ManxPower | Brian: it is documented to require double quotes. perhaps accepting single quotes was a bug since it was not what was documented. |
| 17:19.45 | brian | I'll try to change it |
| 17:19.48 | brian | but I doubt it will help |
| 17:19.54 | ManxPower | Brian: that is better than AGI not Work. |
| 17:20.25 | ManxPower | personally I doubt that going back to 1.2.13 sill make it work and if it does then you can file a bug report |
| 17:20.47 | wunderkin | you're telling someone to submit a bug when not using the latest version? shame! shame on you :D |
| 17:20.54 | ManxPower | but the report won't be accepted unless you test it with the latest version of 1.2 asterisk |
| 17:21.01 | wunderkin | hehe |
| 17:21.21 | ManxPower | wunderkin: My experience with reporting bugs has been less than good. |
| 17:23.17 | russellb | the code only handles double quotes |
| 17:23.20 | russellb | and that has not changed |
| 17:24.17 | brian | Still doesn't work. |
| 17:24.24 | brian | It made no difference ata ll. |
| 17:24.29 | ManxPower | russellb: you mean we have to follow the docs! |
| 17:24.43 | wunderkin | docs?? z0mg |
| 17:25.00 | russellb | it's crazy |
| 17:26.42 | Corydon76-home | brian: what's wrong with it? |
| 17:26.57 | russellb | are you waiting for a response from STREAM FILE before running EXEC Queue? |
| 17:26.57 | Corydon76-home | The output you posted appeared to be successful |
| 17:27.07 | brian | russellb: oops |
| 17:27.14 | russellb | i win |
| 17:27.23 | brian | The Queue is playing over the streamed file isn't it |
| 17:27.54 | russellb | well i'm thinking you may not be giving it a chance |
| 17:28.05 | russellb | i'm just guessing, but ... |
| 17:28.09 | *** join/#asterisk danp (i=danp@elmer.glueless.net) |
| 17:28.34 | danp | anyone having trouble with outbound calls through vitelity? |
| 17:30.53 | *** join/#asterisk beeb (n=b@220-253-28-46.VIC.netspace.net.au) |
| 17:30.54 | brian | russellb: you were right |
| 17:31.05 | Corydon76-home | <ding> |
| 17:31.06 | brian | russellb: the library is async I/O so it wasn't playing the file i feel like an idiot |
| 17:31.18 | russellb | yay! |
| 17:31.31 | russellb | what do i win? |
| 17:31.37 | Qwell | a brand new car! |
| 17:31.42 | russellb | ta-da! |
| 17:31.46 | Qwell | matchbox |
| 17:31.51 | russellb | that's fine |
| 17:31.53 | russellb | those pwn |
| 17:31.57 | Qwell | crap |
| 17:32.02 | Qwell | I wasn't prepared for that |
| 17:32.04 | russellb | :-p |
| 17:32.39 | brian | well i'm still being flooded for Apr 8 17:30:46 WARNING[16545]: chan_sip.c:2575 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) |
| 17:32.48 | brian | how do I make that annoying crap stop? |
| 17:35.37 | brian | When I don't play my sound file (which is signed linear format), I don't get any warning messages. |
| 17:36.32 | brian | Help? :( |
| 17:37.17 | brian | I'd convert it to ulaw but I don't know how and haven't found anything about converting wav to ulaw. |
| 17:37.30 | ManxPower | Dare I even ask why they are in SLN format? |
| 17:37.38 | brian | I read on the wiki that it was the preferred format. |
| 17:37.49 | brian | so I figured it would work flawlessly. |
| 17:37.54 | brian | apparently not |
| 17:38.01 | ManxPower | No, the preferred format is whatever codec your calls are. |
| 17:38.06 | brian | ulaw |
| 17:38.37 | brian | If I use ulaw codec and someone from Europe calls will it screw up? |
| 17:38.40 | ManxPower | and .wav is not .sln |
| 17:38.46 | brian | ]i know |
| 17:38.48 | russellb | asterisk 1.4 can convert files for you, heh |
| 17:38.56 | brian | The original file is wav |
| 17:38.58 | Qwell | russellb: he's using like 1.2.13 |
| 17:39.01 | Qwell | ... |
| 17:39.03 | russellb | sudo asterisk -rx "convert myfile.slin myfile.ulaw" |
| 17:39.10 | ManxPower | Brian: only if they call over IP and even then not if you allow alaw |
| 17:39.26 | russellb | well, it's a bug in any case |
| 17:39.31 | ManxPower | russellb: so what would be the equiv sox command? |
| 17:39.36 | russellb | i'm just talking about hacked up ways to fix it |
| 17:39.38 | russellb | ManxPower: i have no idea |
| 17:39.45 | brian | Okay well |
| 17:39.50 | brian | Yeah a sox command would be helpful |
| 17:39.51 | brian | :( |
| 17:39.57 | Qwell | russellb: what's a bug? |
| 17:39.57 | ManxPower | Brian: now that you have a file being played, does playing a standard asterisk gsm file also cause that message? |
| 17:39.59 | brian | But I have the original WAV. |
| 17:40.07 | brian | ManxPower: I'm not sure |
| 17:40.10 | russellb | Qwell: the spewing of messages about incorrect formats |
| 17:40.15 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:40.15 | brian | ManxPower: probably not |
| 17:40.18 | Qwell | yeah, fixed since 1.2.13 :P |
| 17:40.23 | ManxPower | Brian: try it and see just in case |
| 17:40.27 | russellb | Qwell: lol, nice |
| 17:40.38 | Qwell | if he had upgraded like I told him over an hour ago... |
| 17:40.40 | russellb | brian: upgrade, silly goose |
| 17:40.50 | brian | nooooo |
| 17:40.56 | brian | neeeeever |
| 17:40.57 | russellb | dude. |
| 17:41.01 | russellb | upgrade or i'll kick you |
| 17:41.04 | brian | why |
| 17:41.05 | brian | :( |
| 17:41.08 | ManxPower | russellb: he is one of those that think using prepackaged asterisk is a good idea. |
| 17:41.22 | russellb | because you are asking about a bug that has been fixed already |
| 17:41.28 | brian | what bug |
| 17:41.30 | Qwell | /msg gentoo-devs Please stop packaging asterisk if you aren't going to keep up. |
| 17:41.31 | Qwell | :P |
| 17:41.35 | ManxPower | he is using some gentoo build of asterisk or something like that |
| 17:41.51 | russellb | the bug that caused spewing those messages |
| 17:42.38 | brian | Well when I play a asterisk gsm file it doesn't happen |
| 17:42.58 | ManxPower | good to know |
| 17:43.27 | brian | so how do I convert my wav files |
| 17:43.36 | russellb | by upgrading asterisk |
| 17:43.44 | brian | what if I want to use 1.2 |
| 17:43.48 | brian | and not smelly 1.4 |
| 17:43.51 | Qwell | UPGRADE ANYWAYS |
| 17:43.54 | russellb | then use 1.2.18 or whatever |
| 17:43.56 | russellb | good lord |
| 17:43.57 | Qwell | You're using something that's like 3 months old |
| 17:43.59 | ManxPower | Brian: so upgrade to the latest 1.2.x |
| 17:44.08 | Qwell | with a *KNOWN* bug, that has been *FIXED* |
| 17:44.13 | brian | Qwell: It's actually a lot newer, it's patched! |
| 17:44.18 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:44.20 | *** mode/#asterisk [+b %brian!*@*] by russellb |
| 17:44.23 | danp | heh |
| 17:44.27 | russellb | brian: you can't talk until you upgrade!! |
| 17:44.29 | Qwell | ha, beat me to it by like half a second |
| 17:44.50 | *** mode/#asterisk [-b %brian!*@*] by russellb |
| 17:44.59 | brian | haha tricked you i'll never upgrade eveeer |
| 17:45.00 | brian | jk |
| 17:45.12 | brian | okay so is there any configuration changes |
| 17:45.17 | brian | in 1.2.18 |
| 17:45.18 | Qwell | no, none |
| 17:45.27 | russellb | except that we removed AGI support |
| 17:45.52 | Juggie | hah |
| 17:46.01 | brian | what |
| 17:46.17 | russellb | yeah ... got tired of it |
| 17:46.21 | brian | you better be joking |
| 17:46.43 | russellb | no way, dead serious |
| 17:46.43 | danp | something i've noticed about this 1.2 svn checkout (along with asterisk-addons) is that realtime seems to be trying to load extra columns in my SIP users table as options |
| 17:46.58 | Qwell | danp: well, yeah |
| 17:47.24 | danp | it makes sense i guess but i don't recall seeing messages about it before |
| 17:47.28 | danp | http://pastie.caboo.se/52398 |
| 17:48.39 | brian | ok it's compiling |
| 17:48.50 | Qwell | uninstall the package first |
| 17:48.53 | brian | i did |
| 17:48.56 | brian | i'm not an idiot |
| 17:49.00 | rudholm | looks like the latest in Gentoo's package db is 1.2.14 |
| 17:49.16 | rudholm | the Gentoo "developers" are behind on a lot of stuff, not just Asterisk. |
| 17:49.22 | brian | Gentoo is always behind |
| 17:49.27 | rudholm | although, asterisk does rev pretty briskly. |
| 17:49.42 | rudholm | yes, and I use "developers" mockingly. |
| 17:49.43 | brian | But Gentoo was the best EC2 image available |
| 17:49.49 | rudholm | they're packagers, not developers. |
| 17:49.57 | brian | All the other ones were like CentOS and icky ones |
| 17:50.06 | russellb | rudholm: well played |
| 17:50.07 | brian | And I was waaaaay too lazy to make my own |
| 17:50.57 | file | hello class |
| 17:51.03 | rudholm | russellb: seriously, they run around calling themselves "developers" but unless they're working on Portage itself, or the installer, or some actual coding, they're just building packages for Portage, and that's not "Development", that's "packaging" |
| 17:51.31 | russellb | yeah, same with any distro ... |
| 17:51.36 | rudholm | Debian makes that distinction |
| 17:51.42 | brian | this won't overwrite my configuration files will it |
| 17:51.47 | russellb | oh? cool |
| 17:51.47 | rudholm | they don't call anyone who comes near the distro a "Developer" |
| 17:51.58 | rudholm | but I think Gentoo is run by teenagers without any professional experience |
| 17:52.33 | *** join/#asterisk ChkDigit (n=mrw@static24-72-71-175.regina.accesscomm.ca) |
| 17:52.45 | mmartinn | I know at least one Gentoo developer IRL, and he has submitted code via Gentoo to tons of projects, so I wouldn't generalize that all Gentoo "developers" are packagers |
| 17:52.45 | russellb | yay gentoo trolling |
| 17:53.00 | rudholm | oh, I actually use Gentoo, this isn't a troll :) |
| 17:53.02 | tzanger | russellb: like shooting fish in a barrel |
| 17:53.04 | rudholm | how do you think I know about it? :) |
| 17:53.15 | brian | hay guise i funroll-loops |
| 17:53.20 | rudholm | hahaha |
| 17:53.24 | rudholm | that website is gone :( |
| 17:53.24 | mmartinn | lol =/ |
| 17:53.25 | Juggie | use whatever distro you want |
| 17:53.30 | Juggie | just use asterisk source. |
| 17:53.33 | brian | rudholm: what ;( |
| 17:53.36 | rudholm | yeah |
| 17:53.40 | rudholm | bummed me right out |
| 17:53.41 | russellb | i miss that website ... |
| 17:53.43 | Qwell | I actually suggested -funroll-loops as a solution to a problem :D |
| 17:53.44 | brian | say it isn't sooooooo |
| 17:53.52 | rudholm | go see |
| 17:53.53 | ChkDigit | Now... speaking of asterisk... |
| 17:53.54 | brian | Qwell: were you high |
| 17:53.57 | wunderkin | fruit rollups... mmm |
| 17:54.00 | *** join/#asterisk friedrich| (n=friedric@e177248114.adsl.alicedsl.de) |
| 17:54.04 | Qwell | brian: No, it was a legitimate solution |
| 17:54.23 | ChkDigit | Anybody know of an application or function I can use to check a channel for the called number? |
| 17:54.25 | Qwell | looping through some loops repeatedly, and each of the loops had a static count |
| 17:54.52 | russellb | ChkDigit: like ... ${EXTEN} |
| 17:54.53 | ChkDigit | ... and this would be another channel, one that I'd like to possibly hangup... |
| 17:55.07 | russellb | i don't understand the question :) |
| 17:55.23 | mmartinn | Is there a good way to figure out what SIP user dialed a channel, other than by looking at the channel name? |
| 17:55.23 | brian | ok so |
| 17:55.34 | ChkDigit | I want to have a macro check the a congested SIP gateway when dialling 911. |
| 17:55.48 | mmartinn | ChkDigit's question reminded me of my own :) |
| 17:55.52 | ChkDigit | And hangup the channel, unless it is already calling 911. |
| 17:55.55 | brian | now I need a init script for gentoo |
| 17:55.56 | brian | :( |
| 17:56.30 | rudholm | brian: what do you need the init script to do? |
| 17:56.31 | mmartinn | brian: You can grab one out of portage's latest asterisk package |
| 17:57.06 | russellb | ChkDigit: I don't know a way to do that from the dialplan |
| 17:57.07 | brian | rudholm: it needs to make vroom vroom noises when I'm playing with my toy cars |
| 17:57.17 | rudholm | oh, that's easy |
| 17:58.02 | *** join/#asterisk phr0z3n (n=phr0z3n@216.186.159.215) |
| 17:58.14 | *** part/#asterisk phr0z3n (n=phr0z3n@216.186.159.215) |
| 17:58.32 | brian | i can't find the init script for asterisk for gentoo |
| 17:58.36 | brian | and i'm lazy |
| 17:59.08 | ChkDigit | Well, maybe I'll just use a DB(emergencycall/SIPxyz)=1 |
| 17:59.31 | mmartinn | brian: Look in /usr/portage/net-misc/asterisk/files/1.2.0/asterisk.(confd/rc6) |
| 17:59.34 | rudholm | brian: here, I'll paste one: |
| 17:59.38 | rudholm | /usr/sbin/asterisk |
| 17:59.39 | rudholm | there |
| 18:00.35 | mmartinn | Is there a way to tell from the dialplan or in an AGI what two channel identifiers are bridged, other than by the name of the bridged channel? |
| 18:03.38 | brian | I need a sample asterisk.conf |
| 18:04.06 | ManxPower | Brian: /path/to/src/asterisk/configs |
| 18:04.15 | brian | I already deleted the souce |
| 18:04.20 | brian | I don't have enough space on this ec2 instance |
| 18:04.23 | ManxPower | Brian: it sucks to be you. |
| 18:04.25 | blitzrage | ManxPower: actually, I think that file is generated from the 'make config' |
| 18:04.40 | brian | It said it would overwrite my existing config files |
| 18:04.46 | brian | so i didn't do it |
| 18:04.51 | Qwell | make config != make samples |
| 18:04.57 | tzafrir_laptop | can chan_misdn be built vs. mISDNuser v. 1.1.2? |
| 18:04.57 | blitzrage | errrr |
| 18:05.00 | blitzrage | yah... make samples :) |
| 18:05.12 | Qwell | You shouldn't need asterisk.conf |
| 18:05.17 | blitzrage | also true |
| 18:05.25 | ManxPower | make config just installs the init script for your system doesn't it. |
| 18:05.32 | blitzrage | ManxPower: yes -- I mean make samples |
| 18:05.32 | Qwell | ManxPower: yes |
| 18:06.18 | tzafrir_laptop | brian, actually there's no sample asterisk.conf there. The asterisk.conf is usually generated on 'make samples'. However take a look at the doxygen docs, there's a document there about it |
| 18:07.00 | ManxPower | tzafrir_laptop: don't bother. He's trying to set himself up for failure. |
| 18:08.20 | brian | doxygen has so many dependencies. |
| 18:08.21 | tzafrir_laptop | brian, if you look for a decent init script for $DISTRO , start with the init script provided in the asterisk package of $DISTRO . At least in the case of Debian and SUSE those are better than the defualt Asterisk ones. Not sure about gentoo and freebsd |
| 18:08.58 | tzafrir_laptop | brian, luckily the docs themselves are availble on-line |
| 18:09.27 | tzafrir_laptop | http://asterisk.org/doxygen/ |
| 18:11.02 | ManxPower | Yay! AL will start getting civilized weather on wed. |
| 18:13.15 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 18:13.15 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:13.30 | mmartinn | ManxPower: What do you consider civilized? |
| 18:13.42 | ManxPower | mmartinn: anything over 65F |
| 18:14.11 | mmartinn | ManxPower: Ah; I'm in North FL, and we got the same cold spell; it's only supposed to be warmer from here out |
| 18:14.23 | Qwell | it actually snowed here on Friday |
| 18:14.32 | mmartinn | Qwell: You're in AL as well? |
| 18:14.39 | Qwell | yeah, Huntsville |
| 18:14.44 | mmartinn | Brrr |
| 18:14.50 | ManxPower | Qwell: It snowed on the mountian as well. Ick |
| 18:15.10 | mmartinn | I had all these new plants now that it was "warm" and then it gets cold again |
| 18:15.11 | mmartinn | =/ |
| 18:15.30 | rudholm | it was overcast here yesterday. brrr. |
| 18:21.16 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 18:21.50 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 18:22.01 | linagee | did anyone notice voicepulse go out yesterday? |
| 18:22.07 | linagee | yesterday afternoon i guess |
| 18:22.18 | linagee | i had an angry client. hah |
| 18:22.29 | linagee | "all i got was a busy signal" |
| 18:22.36 | linagee | oh wait.... |
| 18:22.37 | linagee | yikes |
| 18:23.14 | linagee | ManxPower: might have been when i was playing with upgrading the server. :) |
| 18:23.17 | linagee | whoops. :) |
| 18:23.24 | blitzrage | "playing" |
| 18:23.31 | linagee | then i rebooted. things probbaly worked after that |
| 18:23.36 | linagee | blitzrage: exactly that |
| 18:23.40 | linagee | lol |
| 18:24.03 | linagee | blitzrage: doing yum install to some modules, i was kind of O_o that it would change around a zaptel module |
| 18:24.09 | linagee | and it upgraded asterisk. ack |
| 18:24.44 | linagee | now i am running 1.2.17 and have no idea how well that plays with freepbx. :( |
| 18:25.25 | linagee | and again, and again |
| 18:26.40 | danp | linagee: i'm investigating freepbx for a project and i'm using a 1.2 svn checkout from yesterday...seems to be fine |
| 18:26.57 | ManxPower | danp: we don't support FreePBX here |
| 18:26.57 | linagee | lol |
| 18:26.58 | linagee | great |
| 18:27.04 | danp | did i ask for support? |
| 18:27.13 | ManxPower | danp: no, but you will 8-) |
| 18:27.20 | linagee | danp: did i ask yum to start upgrading modules on me? :( |
| 18:27.28 | danp | i won't, trust me |
| 18:35.32 | *** join/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net) |
| 18:36.03 | trevarthan | Ugh. What happens if I Gosub() out of a macro? I was hoping it would return to the macro context, but I think it freaks out and dies. |
| 18:37.12 | trevarthan | Ultimately, I'd prefer to use Gosub() instead of Macro() for everything. But the MACRO_* variables aren't available in a Gosub() context, are they? |
| 18:38.05 | trevarthan | In addition, I don't think I can pass arguments to subroutines. That's annoying. |
| 18:41.36 | blitzrage | trevarthan: try latest SVN, that was fixed yesterday I believe |
| 18:42.12 | trevarthan | ugh.... ok. Well, I can't use SVN, but it's good to know that it's actually a bug. |
| 18:42.36 | linagee | blitzrage: yay. i got it back. :) |
| 18:43.13 | tzanger | whee, billing |
| 18:44.05 | trevarthan | blitzrage: can I nest macros? Or does that screw things up too? |
| 18:45.04 | russellb | you can, but only so deep |
| 18:45.10 | russellb | i think it is limited to 7 levels deep |
| 18:46.01 | mmartinn | Are Macros supposed to be like your dialplan applications? I get that sense because it seems like they are treated that way by the calling context |
| 18:46.39 | mmartinn | Like when you get DTMF or whatever and you don't jump inside your macro, but jump to the priority in the calling scope |
| 18:48.16 | trevarthan | mmartinn: not really. because pressing a key doesn't occur in the macro context, it occurs in the parent context. |
| 18:49.01 | trevarthan | mmartinn: macros are supposed to be shortcuts for a bunch of operations. Instead of having to type them 20 different times in 20 different places you call a macro. |
| 18:49.02 | mmartinn | trevarthan: And actual applications grab their keys inside themselves and not in the parent context? |
| 18:49.12 | trevarthan | mmartinn: yeah. |
| 18:49.25 | mmartinn | trevarthan: that helps me understand it better :) |
| 18:50.58 | trevarthan | mmartinn: macros are crap, IMO. Gosub() is much better. But, like I said, you don't get MACRO_* variables in a subroutine and you can't do arguments. So Gosub()s are crap too. Really, the ideal replacement would be Gosub() behavior combined with argument functionality and the MACRO_* variables, IMO. |
| 18:51.24 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 18:51.26 | trevarthan | Currently I use a combination of Macro() and Gosub() in my applications, but it's hella confusing. |
| 18:52.17 | trevarthan | Basically I use Gosub() everywhere I can because it behaves more like a typical "function" in a real programming language. But when I need arguments or MACRO_* vars then I make do with Macro(). |
| 18:53.32 | blitzrage | or you could use the version from svncommunity that has args |
| 18:54.21 | blitzrage | I don't bother with Macro() anymore |
| 18:57.44 | tzanger | yeah you're a subroutine weenie now |
| 18:58.29 | mmartinn | What did the dialplan syntax evolve from anyway? Is AEL the "new way" or has it not been adopted as much as hoped? |
| 18:59.02 | trevarthan | Honestly, I should probably just be using AGI for this application. Probably anything over 100 lines is too complicated for extensions.conf. |
| 18:59.33 | blitzrage | mmartinn: AEL is simply an alternative |
| 18:59.34 | trevarthan | I'm up to 430 right now, and I feel like I'm dancing on swiss cheese. |
| 19:00.15 | blitzrage | -= 259 extensions (1738 priorities) in 60 contexts. =- |
| 19:00.43 | blitzrage | AEL still converts everything to a dialplan syntax |
| 19:00.45 | trevarthan | I particularly dislike how I keep forgetting to include 'i' and 'h' extensions in my subroutines and bad things happen. |
| 19:00.56 | mmartinn | Ah |
| 19:01.04 | mmartinn | So where did that syntax come from originally? |
| 19:01.06 | blitzrage | trevarthan: #include common/post-call-cleanup.inc |
| 19:01.10 | mmartinn | Or did it start with Asterisk? |
| 19:01.12 | blitzrage | mmartinn: which syntax? |
| 19:01.21 | blitzrage | mmartinn: dialplan syntax is unique to Asterisk |
| 19:01.23 | mmartinn | The dialplan syntax... the plain one |
| 19:01.34 | wunderkin | it came from.. brains... brrraaaainnns |
| 19:01.36 | tzanger | you fucking pansies... |
| 19:01.37 | tzanger | -= 3577 extensions (3850 priorities) in 73 contexts. =- |
| 19:01.38 | tzanger | :-) |
| 19:01.39 | mmartinn | Was it designed or did it evolve? |
| 19:02.02 | blitzrage | tzanger: nice :) |
| 19:02.11 | blitzrage | mmartinn: it was designed and has continued to evolve |
| 19:02.12 | tzanger | blitzrage: and nary a gosub |
| 19:02.22 | blitzrage | tzanger: that's why you have so much crap :) |
| 19:02.30 | tzanger | <PROTECTED> |
| 19:02.31 | trevarthan | blitzrage: yeah. I feel like it should be a required part of the subroutine definition though. Do you really want asterisk to hang up when the user presses a key you aren't expecting? Really? |
| 19:02.49 | mmartinn | interesting |
| 19:02.49 | tzanger | trevarthan: that's what 'i' is ofr |
| 19:03.00 | blitzrage | trevarthan: all the subroutine does is a Goto() and sets a channel variable to know where to Return() to |
| 19:03.17 | trevarthan | tzanger: I know, but you have to define it in every context. If you forget one, boom. |
| 19:03.28 | tzanger | trevarthan: define your contexts smarter :-) |
| 19:03.32 | blitzrage | agreed |
| 19:03.37 | blitzrage | I only have mine in a couple contexts |
| 19:03.47 | blitzrage | you don't need it everywhere |
| 19:03.56 | trevarthan | you don't? |
| 19:04.11 | trevarthan | I just left it out of a subroutine and asterisk hung up when I pressed a key. |
| 19:04.18 | trevarthan | how am I supposed to avoid that? |
| 19:04.20 | blitzrage | when would 'i' get hit in a context you're using simply to process some logic that a user isn't interacting with? |
| 19:04.41 | trevarthan | I use subroutines for prompts. |
| 19:04.56 | trevarthan | the subroutine says text |
| 19:05.07 | blitzrage | _X.,1,GoSub(incorrect-button,s,1()) |
| 19:05.17 | blitzrage | 1,1,GoSub(button-that-works,s,1()) |
| 19:05.29 | blitzrage | 1 is a more specific match, and thus would hit before the pattern match |
| 19:05.37 | blitzrage | do better error control in your dialplan logic |
| 19:05.55 | blitzrage | actually _X!,1,.... is better than _X.,1,... |
| 19:06.12 | tzanger | blitzrage: !? |
| 19:06.26 | trevarthan | 1,1,Gosub(sub-say-some-text|s|1) |
| 19:06.33 | blitzrage | tzanger: ! matches zero or more characters, . matches 1 or more |
| 19:06.48 | trevarthan | then in [sub-say-some-text] you need to remember the 'i' extension. That's my point. |
| 19:06.51 | tzanger | ... that isn't true |
| 19:06.52 | blitzrage | trevarthan: oh yah -- I'm using GoSub() version that supports arguments |
| 19:06.59 | tzanger | or rather, it wasn't |
| 19:06.59 | blitzrage | hence the GoSub(foo,s,1()) format |
| 19:07.07 | blitzrage | GoSub(foo,s,1(arg1,arg2)) |
| 19:07.20 | tzanger | _X. matched any 1-or-more digit |
| 19:07.23 | trevarthan | blitzrage: yeah, I get it. read my reply. |
| 19:07.23 | blitzrage | trevarthan: this is what the #include is for |
| 19:07.29 | *** join/#asterisk Slingky (n=Slingky@modemcable199.182-200-24.mc.videotron.ca) |
| 19:07.30 | blitzrage | #include error_control.inc |
| 19:07.45 | trevarthan | blitzrage: but you have to remember to put that include in EVERY subroutine. |
| 19:07.46 | blitzrage | tzanger: no, _X. matches 2 or more |
| 19:07.56 | Slingky | does somebody knows if "company directory" is enabled by default on freepbx/trixbox |
| 19:07.59 | blitzrage | trevarthan: uhhh... yah -- it's pretty obvious to do |
| 19:08.07 | Slingky | cause when i press "#", i get invalid option |
| 19:08.20 | blitzrage | Slingky: see topic --> #freepbx |
| 19:08.35 | trevarthan | blitzrage: my point is this: do you *ever* want asterisk to hang up if you forget and the user presses a key? Do you? Really? Then why *does* it? |
| 19:09.01 | blitzrage | trevarthan: because GoSub() is not something fancy special -- its equivelent to a Goto(), and you need to handle it just like a regular context |
| 19:09.24 | blitzrage | thus you need to put the 'i' in the context, just like every other context you want to do 'i' handling in |
| 19:09.29 | Slingky | blitzrage: sorry, nobody answer me there |
| 19:09.40 | trevarthan | I understand that is how it's implemented. But logically it needs to be more intelligent because it's making the programmer work harder than he should have to. |
| 19:09.40 | blitzrage | Slingky: sorry to hear that |
| 19:09.51 | mmartinn | If I was an evil call center manager, I would totally want to hang up on as many customers as possible :) |
| 19:10.27 | Slingky | blitzrage: are you aware of way to do that ? |
| 19:10.44 | Slingky | blitzrage: maybe i can change a config file manually |
| 19:10.44 | blitzrage | Slingky: I don't use freepbx |
| 19:11.16 | blitzrage | trevarthan: *shrug* |
| 19:11.32 | blitzrage | doesn't seem to affect my dialplan logic |
| 19:11.46 | blitzrage | I figure I gotta do error control in every context to some extent |
| 19:12.12 | trevarthan | I see you understand where I'm coming from though. You might forget to add the 'i' some day and then you'll be a little annoyed. I'm just saying it could be implemented better. |
| 19:12.22 | blitzrage | I guess |
| 19:12.32 | blitzrage | I don't really find it that big a deal |
| 19:12.42 | blitzrage | I don't implement untested contexts :) |
| 19:13.24 | trevarthan | Well, that's why I say I should probably be using AGI for this 430 line application anyway. Then I could implement default behavior the way I want. It's my own fault that I'm not using AGI. |
| 19:17.50 | *** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net) |
| 19:18.42 | Hmmhesays | anyone from the uk in here? |
| 19:18.52 | trevarthan | I've got another example that's a little bit more compelling, if less likely to actually happen in practice: Say you have a subroutine that doesn't actually output any audio. It just runs a bunch of system commands. However, those system commands take a finite amount of time. Most of the time the user waits patiently, but one day the user presses a key in the middle of the subroutine by accident. If you forgot the 'i' extension, |
| 19:20.00 | trevarthan | Again, that's probably another reason to use AGI for largish applications. |
| 19:20.24 | trevarthan | It's just a shame that extensions.conf doesn't provide a better way to handle that. |
| 19:21.27 | ManxPower | uh, that should not be the case. |
| 19:21.51 | trevarthan | should not, or is not? |
| 19:21.58 | ManxPower | that should only be the case if you are using Background, Read, Authenticate, etc. |
| 19:22.09 | *** join/#asterisk friedrich| (n=friedric@e177240050.adsl.alicedsl.de) |
| 19:22.10 | trevarthan | it seems to be the case from my testing under 1.2.x and Gosub(). |
| 19:22.16 | ManxPower | and since your subroutine does not use that dtmf processing should not happen. |
| 19:22.44 | ManxPower | It is easy to test using a simple gosub |
| 19:22.58 | ManxPower | BTW, just what are you doing that takes so long? |
| 19:23.46 | trevarthan | ManxPower: well, when it trips up, I'm using Flite() for TTS. Are you saying that if I don't output any audio that it won't listen for keypresses? |
| 19:24.56 | trevarthan | Perhaps it's just that I'm using Flite(). That would make sense. |
| 19:25.17 | ManxPower | trevarthan: no, I'm saiying that if you don't use any apps that expect user input that should not happen, but I shall test it with playback and wait right now. |
| 19:25.56 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 19:26.07 | ManxPower | nope, it does not process dtmf |
| 19:26.26 | trevarthan | I was taking the behavior I saw using Flite() and assuming that the same behavior would continue without Flite(). Thanks. Good to know. |
| 19:26.33 | ManxPower | perhaps you are running something in your subroutine that is expecting DTMF |
| 19:26.45 | trevarthan | Yes, Flite() listens for DTMF. |
| 19:27.09 | ManxPower | well then you had better be prepared to process that dtmf |
| 19:27.35 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:28.38 | trevarthan | yeah. thanks. |
| 19:37.41 | *** join/#asterisk Fieldy (i=RqZyMxzk@gentoo/contributor/Fieldy) |
| 19:39.04 | omm | does autologoff work with statick agents? |
| 19:39.23 | omm | static |
| 19:42.56 | *** part/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net) |
| 19:46.22 | *** join/#asterisk ManxPower (n=manxpowe@33.sub-75-203-140.myvzw.com) |
| 19:53.35 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 19:53.48 | *** join/#asterisk [shodan] (n=shodan@ip039.99-113-216.pppoe4.joliette.intermonde.net) |
| 19:54.55 | *** join/#asterisk darken_darken (n=marco@240.129.77.83.cust.bluewin.ch) |
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| 19:55.48 | *** join/#asterisk sudhir492 (n=sudhir@c-71-63-59-45.hsd1.va.comcast.net) |
| 19:56.02 | sudhir492 | which version of spandsp works for asterisk 1.4? |
| 19:57.19 | ManxPower | sudhir492: that info would be on the spandsp website |
| 19:58.48 | sudhir492 | I tried to find that, however, it is not clear which one should I use |
| 19:59.17 | ManxPower | actually spandsp does not link to asterisk at all so it does not matter what version you use |
| 19:59.56 | sudhir492 | but apps in asterisk uses spandsp, doesn't it? |
| 20:00.35 | ManxPower | rxfax and txfax those are the ones that care about what version of asterisk you are using, but they are not part of spandsp |
| 20:00.46 | sudhir492 | ok |
| 20:02.40 | *** join/#asterisk daveburr (i=Miranda@15.sub-70-192-143.myvzw.com) |
| 20:04.32 | *** join/#asterisk asterisknerds (n=asterisk@66.7.124.15) |
| 20:04.32 | asterisknerds | <PROTECTED> |
| 20:08.43 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 20:11.04 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 20:12.48 | danp | hmm, i'm trying to make some talkswitch analog phones work with asterisk. for the most part they're just normal analog phones but they also support on-hook paging and intercom |
| 20:12.57 | danp | i can't seem to find any hints online for getting that to work. any hints? |
| 20:14.32 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 20:14.33 | Strom_M | how many pairs do they require for the extra features? |
| 20:15.02 | danp | pairs to the phone? they only use one as far as i know |
| 20:15.17 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net) |
| 20:15.19 | danp | but maybe that's part of my problem |
| 20:16.04 | Strom_M | yeah, i imagine that if they're analog, then the extra features mean they behave more like 1A2 sets where there's a separate pair for control signals |
| 20:16.22 | Strom_M | or, probably a better example, Merlin :) |
| 20:16.35 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 20:18.26 | danp | with systems like that, is there normally a separate FXS port for each phone for that or are they all wired to a control "party line"? |
| 20:19.00 | Strom_M | no, each would have to be a separate port |
| 20:19.45 | Strom_M | im not familiar with talkswitch, but for example, in 1A2, everything is associated primiarily with the line card associated with the incoming telephone line |
| 20:20.32 | Strom_M | so all the A leads for line appearances associated with 555-2368 are wired to the same line card, for example |
| 20:23.08 | danp | hmm |
| 20:26.11 | Strom_M | in Merlin, though, it's a bit different |
| 20:26.33 | Strom_M | each set plugs into its own port on the KSU, and the line appearances are multipled across all phones |
| 20:27.02 | danp | sounds like i'm out of luck for making this work |
| 20:27.19 | danp | all the other basics seem to work fine |
| 20:29.37 | danp | though i am having trouble getting the message waiting count to disappear after clearing the mailbox |
| 20:33.23 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:35.48 | omm | I'm trying to setup my static agents to puase automatically when they miss a call... Is this possible and does anyone have documentation that shows me how to do it? |
| 20:36.40 | *** join/#asterisk voltagex (n=voltagex@124-254-121-155-dsl.ispone.net.au) |
| 20:40.01 | blitzrage | omm: you could control that via the AstDB or using func_odbc to a relational database |
| 20:40.24 | blitzrage | I've never done it, but that's probably how I'd do it |
| 20:40.38 | blitzrage | not a bad idea... will write that down to implement :) |
| 20:47.00 | *** join/#asterisk codejunky (n=jan@c206165.adsl.hansenet.de) |
| 20:48.16 | codejunky | Hello, I am trying to user asterisk as a sipgate client. To accomblish this I added the following line to the sip.conf file in [general]: register => SIPID:PASSWD@sipgate.de/SIPID |
| 20:49.09 | codejunky | sip show registry in the asterisk cli shows that it is Registered. My question is, how can I handle incoming calls? What should I add to my dialplan that a sip client in my network gets the call? |
| 20:49.49 | *** join/#asterisk teun (i=teun@lanfear.moonblade.net) |
| 20:50.04 | Cybertoy | do you have a sip phone connected to asterisk already? |
| 20:50.09 | codejunky | Yes. |
| 20:50.33 | Cybertoy | ok ... so you will need to create an extension SIPID that rings your sip phone |
| 20:50.58 | Cybertoy | since you register SIPID to sipgate |
| 20:51.05 | codejunky | Okay. |
| 20:51.29 | Cybertoy | exten => SIPID,1,Dial(SIP/yourphone) |
| 20:51.32 | Cybertoy | for example |
| 20:51.35 | codejunky | Ahh |
| 20:52.32 | codejunky | Cool it is working :) |
| 20:52.33 | codejunky | Thanks! |
| 20:52.41 | Cybertoy | np |
| 20:58.45 | *** join/#asterisk Glanzmann (i=sithglan@faui08.informatik.uni-erlangen.de) |
| 20:59.11 | Glanzmann | Hello. How can I configure asterisk that it makes a backup copy (recording) of every telephone conversation? |
| 21:00.05 | [TK]D-Fender | GiantPickle, "show application monitor" |
| 21:00.56 | [TK]D-Fender | Glanzmann, "show application monitor" |
| 21:02.27 | Glanzmann | Okay. Thanks. I got this. |
| 21:07.58 | *** join/#asterisk doolph (n=ubuntu@200.75.244.19) |
| 21:11.05 | doolph | hi |
| 21:13.08 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 21:15.31 | *** join/#asterisk paavum (n=Dorphals@pcsp163-73.supercabletv.net.co) |
| 21:15.48 | paavum | hello |
| 21:15.55 | codejunky | hi |
| 21:16.04 | paavum | I'm trying to get app_rxfax and app_txfax to compile with * 1.4 |
| 21:16.18 | paavum | I downloaded them from http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/ |
| 21:16.28 | doolph | and? |
| 21:16.30 | paavum | however I am getting an error when compiling |
| 21:16.58 | paavum | !pastebin |
| 21:16.58 | doolph | are you trying to send/receive fax? |
| 21:17.12 | paavum | well I need to... |
| 21:17.22 | paavum | recieve faxes |
| 21:18.58 | doolph | with a fax machine? |
| 21:20.37 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 21:21.27 | paavum | recieve with fax 2 email |
| 21:21.30 | paavum | http://pastebin.ca/430267 |
| 21:21.55 | paavum | here is the output |
| 21:25.13 | paavum | helllo? |
| 21:27.28 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 21:28.48 | paavum | I'm trying to get app_rxfax and app_txfax to compile with * 1.4 |
| 21:28.59 | paavum | however I am getting an error when compiling them |
| 21:29.06 | paavum | something about conflitcing types |
| 21:42.46 | *** part/#asterisk daveburr (i=Miranda@15.sub-70-192-143.myvzw.com) |
| 21:50.04 | *** join/#asterisk jebba (n=jebba@220-179-89-200.fibertel.com.ar) |
| 21:53.19 | *** join/#asterisk ShakespeareFan00 (n=chatzill@dyn-62-56-121-158.dslaccess.co.uk) |
| 21:53.27 | ShakespeareFan00 | Hello |
| 21:53.53 | ShakespeareFan00 | Is there an experienced Asterisk user here that would be willing to assist on another project? |
| 21:55.41 | [TK]D-Fender | Sure, I do great collages... you should see what I can do with a tube a Super Glue...... |
| 21:56.04 | paavum | I'm trying to get app_rxfax and app_txfax to compile with * 1.4, however I am getting an error when compiling them something about conflitcing types |
| 21:56.57 | [TK]D-Fender | paavum, try anothre version of SpanDSP |
| 21:57.15 | paavum | any suggestions? |
| 21:57.22 | paavum | I'm using spandsp3 |
| 22:01.33 | *** join/#asterisk THX2000 (n=bob@netblock-208-127-94-59.dslextreme.com) |
| 22:01.54 | *** join/#asterisk zapata (n=user@chello213047080026.4.14.vie.surfer.at) |
| 22:02.08 | THX2000 | Anyone have experience running asterisk on embedded platforms? Particularly the WRAP? |
| 22:03.04 | THX2000 | Trying to see if ztdummy will run on the WRAP or if im just spinning my wheels |
| 22:03.12 | tzanger | no, but I will be running it on a custom MCF5282 board just to see what i can do |
| 22:03.26 | tzanger | WRAP is just x86 though so there shouldn't be any funniness |
| 22:03.34 | tzanger | ztdummy requires HPET or RTC, neither of which WRAP has, IIRC |
| 22:04.02 | paavum | does spandsp2 work wih * 1.4? |
| 22:04.03 | THX2000 | iirc? |
| 22:04.49 | THX2000 | Well, i guess my next question is...would there be a way to get meetme to run on the wrap then using some sort of other timer? |
| 22:05.11 | *** part/#asterisk ShakespeareFan00 (n=chatzill@dyn-62-56-121-158.dslaccess.co.uk) |
| 22:06.18 | *** join/#asterisk zimdog (n=zimdog@63-227-112-12.hlrn.qwest.net) |
| 22:11.30 | zimdog | Hello All. I have been messing with freepbx but feel it may be more hassle than I need when trying to do custom stuff. I would like to know if I can accomplish the following with just a plain asterisk system. I want a main server to handle inbound routes to several other servers. So far I was able to have the other servers dial out through the main servers trunks. I can also receive calls coming in to the different servers through the main |
| 22:11.30 | zimdog | server. The problem I ma having is when I setup different inbound DIDs on a secondary server the main server does not seem to be passing this information. I would like a secondary server to match on the DID and then deleiver the call to an extension. I am using sip mostly but have IAX2 trunks setup between the different servers |
| 22:13.38 | [TK]D-Fender | zimdog, You can do just about anything if you do it yourself. |
| 22:16.54 | tzafrir_laptop | THX2000, maybe a USCH usb controller? |
| 22:17.02 | zimdog | [TK]D-Fender: I was thinking that was the case. I just wanted to make sure that this scenario was possible before scrapping my current stuff that was so close to working and except for the DID routing to find out that this cannot be done |
| 22:17.39 | [TK]D-Fender | zimdog, Just sending calls from server A to B.... no big deal... |
| 22:18.47 | tzafrir_laptop | 10$ ATA? http://www.rowetel.com/blog/?p=26 |
| 22:20.29 | tzafrir_laptop | (actually: not exactly an ATA. basically something of the sort of S100U) |
| 22:20.39 | zimdog | [TK]D-Fender, So this type of implentation is possible? IAX2 will pass the DID it matched on to the second server ? |
| 22:21.00 | [TK]D-Fender | Sure. |
| 22:22.25 | zimdog | Ok I must be missing domething then. I was calling out with the following exten => _.,1,Macro(dialout-trunk,6,${EXTEN},,) and then the second server would not match on the DID |
| 22:23.05 | zimdog | Maybe I need to pass a different variable the ${EXTEN} |
| 22:23.25 | [TK]D-Fender | zimdog, who said anything about how FreePBX's stupid macro's work, or if IT was designed to handle what you want? |
| 22:23.59 | Dimitripietro | <zimdog> you should refer to #freepbx |
| 22:24.05 | [TK]D-Fender | zimdog, FreePBX is a canned little world, and if you want to think "outside of the box", don't expect too much. |
| 22:24.16 | [TK]D-Fender | Dimitripietro, we're not there yet.... |
| 22:25.54 | zimdog | [TK]D-Fender, I understand. I see the limitaitons that is why I wanted to see if I could do it with just plain asterisk |
| 22:26.43 | [TK]D-Fender | zimdog, And you can... |
| 22:27.48 | zimdog | I was also looking at having an interface that would allow me to do the basics without adjusting the configs manually |
| 22:29.18 | [TK]D-Fender | zimdog, Nope. FreePBX is a canned POS and if you don't like it, you'll have to do it yourself. You said as much coming in yte you seem to have to keep coming to this conclusion over and over. Stick with it already.... |
| 22:30.01 | zimdog | Any gui that can work with a base asterisk install? |
| 22:30.24 | [TK]D-Fender | zimdog, nothing free that I've ever heard of. |
| 22:30.50 | zimdog | what are your thoughts on asterisknow? |
| 22:31.30 | [TK]D-Fender | zimdog, Same crap, different smell. only works on 1.4, closed source IIRC, uses craptastic wierd configs and the users.conf with 1.4 |
| 22:32.59 | zimdog | That is what I thought. Guess I need to give up on a GUI and see if I can do what I want the manual way |
| 22:33.30 | [TK]D-Fender | zimdog, well I've already confirmed that you can, so its just a matter of you doing it. |
| 22:34.09 | zimdog | Thanks for the help in setting me straight |
| 22:37.10 | tzanger | [TK]D-Fender: did you recommend bon cop bad cop to me? |
| 22:37.15 | tzanger | saw it a couple weeks ago, it is pretty good :-) |
| 22:37.21 | red9012 | In using streaming music on hold. Does each music on hold class result in a new stream, or one stream can be shared among all classes? |
| 22:38.39 | Dimitripietro | red9012 if you want to share, why don't you use the same classes ? |
| 22:42.01 | [TK]D-Fender | tzanger, Heardit was good... |
| 22:53.10 | brian | does anyone in here run asterisk 1.4 in production |
| 22:54.52 | paavum | I'm trying to get app_rxfax and app_txfax to compile with * 1.4, however I am getting an error when compiling them something about conflitcing types... I've already tried with spandsp 0.02 0.03 and 0.04 pre1 |
| 22:54.59 | paavum | and I get the same error |
| 22:55.09 | paavum | can anybody please give me a hand |
| 22:55.22 | paavum | I've even looked at the C code ... and strangely enough its the same |
| 22:55.55 | paavum | so there should be no conflict |
| 22:57.04 | pfn | what's overlap dialing? |
| 23:02.04 | SplasPood | Hey, is there any way to massage my CDRs in such a way that the dst field shows the actual dialed number rather than the final exten in my context... Stuff coming into my IVR tends to show as dst == 's' |
| 23:02.32 | SplasPood | Or do I need to shove the dialed # into the userfield and post-process later |
| 23:03.12 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 23:06.30 | *** join/#asterisk mrdigital (n=jkjkj@pool-72-81-78-68.phlapa.east.verizon.net) |
| 23:06.44 | mrdigital | anyone here recommend a x100p clone? |
| 23:10.08 | Qwell | mrdigital: no |
| 23:10.20 | blitzrage | mrdigital: x100p (and their clones) are very bad hardware |
| 23:10.31 | mrdigital | bad as in? |
| 23:10.33 | Qwell | blitzrage++ |
| 23:10.37 | Qwell | ~x100p |
| 23:10.47 | jbot | i guess x100p is an obsolete card. You don't want to bother trying to make it (or any of the "digium compatible" clones) work. Get a TDM01B, and you will save your sanity, your hair, and countless other things. |
| 23:11.02 | mrdigital | how much are those? im on a budget? |
| 23:11.15 | mrdigital | that last part wasnt a question |
| 23:12.48 | GreyFoxx | hehe hmmm $15 on ebay for a x100p, or $150 for a TDM01B.....I think I'd spend the effort to try and get it working :) |
| 23:12.56 | mrdigital | are there any cards for 20-50? |
| 23:12.59 | GreyFoxx | Assuming homeuser use |
| 23:13.45 | *** join/#asterisk Growly (n=himself@125-238-233-134.broadband-telecom.global-gateway.net.nz) |
| 23:14.20 | blitzrage | if you're on a budget, get an account from an ITSP |
| 23:14.25 | mrdigital | im building a call center for a online clothing company their budget is not big |
| 23:14.29 | blitzrage | money better spent |
| 23:14.34 | mrdigital | and they dont want to spend monthly |
| 23:14.49 | blitzrage | they? x100p SHOULD NOT be used for production |
| 23:14.52 | blitzrage | home hobbyist at best |
| 23:15.04 | blitzrage | x100p introduces echo into the line |
| 23:15.12 | GreyFoxx | Yeah, if it's for business use go for something better |
| 23:15.22 | Qwell | and every line within 3 miles of it |
| 23:15.22 | Strom_M | it's the classic "but why should we spend money on anything related to infrastructure?" uber-cheap client from which you should RUN, not walk. |
| 23:15.24 | Qwell | :P |
| 23:15.36 | blitzrage | Strom_M: AMEN!! |
| 23:15.40 | Strom_M | also, hi |
| 23:15.49 | GreyFoxx | Especially since you will need several lines for a callcenter, and the x100p generate a lot of interrupts |
| 23:16.04 | mrdigital | its a small call center |
| 23:16.08 | mrdigital | 3 people |
| 23:16.52 | GreyFoxx | If it's a business they should be able to fork out for a better card. |
| 23:17.09 | mrdigital | the owner is a dumbass and cheap |
| 23:17.19 | GreyFoxx | potential business downtime due to cheap hardware is likely more than the cost of the card |
| 23:17.21 | [hC] | mrdigital: you'll regret doing x100p, trust us |
| 23:17.29 | [hC] | its not even usable, really |
| 23:17.37 | [hC] | its that bad |
| 23:17.46 | mrdigital | how bout i describe what we're using asterisk for? |
| 23:18.04 | [hC] | it really doesnt matter, if you are using an x100p for calls, its GOING to be painful |
| 23:18.08 | mrdigital | would that help finding a cheaper but good card? |
| 23:18.24 | [hC] | i would suggest a sangoma a200, personally |
| 23:18.33 | [hC] | they work great and they arent too expensive. |
| 23:18.36 | mrdigital | how much? |
| 23:18.49 | [hC] | less than 500 bucks for 2 lines i think. |
| 23:19.25 | Qwell | 500? yikes |
| 23:19.26 | mrdigital | we need 1 line |
| 23:20.06 | mrdigital | well 2 actually |
| 23:20.19 | Strom_M | a three person call center with only two lines? |
| 23:20.23 | [hC] | pardon me its $210 on telephonyware |
| 23:20.38 | Strom_M | this sounds like a recipe for disaster |
| 23:20.51 | Qwell | Strom_M: You mean they can't put one on call waiting?! |
| 23:20.56 | Strom_M | omg |
| 23:21.02 | Strom_M | also cocks |
| 23:21.19 | Qwell | eh? |
| 23:21.22 | mrdigital | Strom_M: its more of a automated system |
| 23:21.57 | Strom_M | how so? |
| 23:22.15 | mrdigital | for people to call in and check their order statuses. etc |
| 23:22.24 | mrdigital | with the option of talking to a customer rep |
| 23:22.28 | mrdigital | we arent a big company |
| 23:22.35 | mrdigital | therefore we dont get calls after calls |
| 23:23.12 | Strom_M | are you in north america? |
| 23:23.17 | mrdigital | the boss is looking for a low budget system (i got a free system that'll do it just need a fxo card) |
| 23:23.20 | mrdigital | yes |
| 23:23.50 | Strom_M | ADSL + ITSP |
| 23:24.05 | Strom_M | srsly. |
| 23:24.18 | mrdigital | ITSP? |
| 23:24.27 | Strom_M | internet telephony service provider |
| 23:24.30 | Strom_M | here, read this |
| 23:24.32 | Strom_M | ~101 |
| 23:24.41 | jbot | extra, extra, read all about it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 23:24.41 | Strom_M | ~wikis |
| 23:24.42 | jbot | methinks wikis is http://www.voip-info.org |
| 23:24.43 | Strom_M | ~book |
| 23:24.44 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:25.09 | mrdigital | i know asteriosk |
| 23:25.16 | mrdigital | im just trying to find a low cost card |
| 23:25.26 | *** join/#asterisk shodan- (n=shodan@ip078.96-113-216.pppoe1.joliette.intermonde.net) |
| 23:25.50 | Strom_M | if your company is that cheap, phone lines sound like too much of an expense |
| 23:26.00 | mrdigital | we already have the lines and system |
| 23:26.03 | Strom_M | just get ADSL and an ITSP like teliax that charges you only per minute |
| 23:26.08 | mrdigital | we just need a low cost card |
| 23:26.10 | Strom_M | well then get a TDM02B |
| 23:26.18 | Strom_M | trust me, you'll hate the low cost card |
| 23:26.21 | mrdigital | how much? |
| 23:26.30 | Strom_M | actually, trust me and the twelve other people who are telling you the same thing |
| 23:27.05 | mrdigital | too much the budget is $20-50 |
| 23:27.13 | Strom_M | um |
| 23:27.19 | Strom_M | don't even bother, then, |
| 23:27.26 | mrdigital | alright |
| 23:27.33 | *** part/#asterisk mrdigital (n=jkjkj@pool-72-81-78-68.phlapa.east.verizon.net) |
| 23:27.40 | Qwell | wtf |
| 23:27.54 | Strom_M | i feel sorry for their employees. |
| 23:28.05 | Qwell | he's gonna spend more on a single analog line *PER MONTH* than that |
| 23:28.27 | Strom_M | one thing ive learned is that you can't reason with a chronically stingy tightwad |
| 23:29.25 | Strom_M | speaking of toys, I need to find me an ISDN telephone |
| 23:29.40 | Qwell | I'm surprised you don't have one |
| 23:29.52 | Strom_M | I have an NT1 and a TA, but not an actual voice terminal |
| 23:34.38 | Strom_M | http://cgi.ebay.com/Lucent-ISDN-8510T-Voice-Terminal-NEW_W0QQitemZ110084550950QQihZ001QQcategoryZ58344QQrdZ1QQssPageNameZWD1VQQcmdZViewItem |
| 23:34.57 | Strom_M | it's bricktacular |
| 23:35.11 | Qwell | god that's ugly |
| 23:35.36 | Qwell | looks exactly like the phones we had at the bank |
| 23:36.06 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 23:36.31 | *** join/#asterisk plantseeker (n=chatzill@host86-145-78-219.range86-145.btcentralplus.com) |
| 23:39.04 | blitzrage | well... good luck to him |
| 23:39.17 | blitzrage | should be fun to watch him keep coming back for help on the x100p |
| 23:41.14 | Strom_M | heh |
| 23:43.25 | *** join/#asterisk Fieldy (i=OrHb1ovW@gentoo/contributor/Fieldy) |
| 23:45.10 | *** join/#asterisk mroli (n=mroli@wsip-64-58-154-130.oc.oc.cox.net) |
| 23:45.13 | mroli | hello all |
| 23:45.23 | pfn | x100p sucks, heh |
| 23:45.50 | mroli | having a brainfart.. can someone remind me where is that directory that asterisk scans, where we can drop a script file to be executed automatically? |
| 23:48.02 | mroli | here it is |
| 23:48.19 | mroli | . /var/spool/asterisk/outgoing |