| 00:00.35 | [r]evolution | your as in who works for you |
| 00:01.02 | [r]evolution | or your as in a guy whom you're working with and belongs to another company providing your pri line? |
| 00:01.22 | watchy2 | i just started with a VOIP company in LR arkansas |
| 00:01.26 | watchy2 | hes a guy I work with |
| 00:01.38 | watchy2 | they setup and had working a * box. i dunno wtf they did |
| 00:01.50 | watchy2 | but now everytime they start * the pri quits working |
| 00:02.00 | [r]evolution | you check the zap config make sure it's setup properly? |
| 00:02.04 | watchy2 | no incoming/outgoing calls when earlier it worked fine |
| 00:02.24 | watchy2 | i don't think i'm gonna beable to check the tech decided to reinstall elastics |
| 00:02.40 | [r]evolution | p.s. have zero experience with zap but im looking for excuse to get my mind off asterisk rejecting calls b/c they're sending as G729a instead of G729 |
| 00:02.46 | [r]evolution | O_o |
| 00:02.47 | [TK]D-Fender | sky_drive: Perhaps you could try providing what I asked for half an hour ago... |
| 00:02.54 | [r]evolution | when in doubt... blow it away and start over? |
| 00:03.05 | watchy2 | rev: in my eyes? hell no |
| 00:03.12 | watchy2 | in these guys i work with, yes |
| 00:03.26 | [TK]D-Fender | [r]evolution: Blow 'em away.... great idea! |
| 00:03.27 | watchy2 | if you don't fix it this time next time your not going to have t he answer |
| 00:03.30 | [r]evolution | yeah i know... it was kinda of rhetorical sarcasm. :) |
| 00:03.46 | [r]evolution | didnt really want you to answer... was just being a cunt. |
| 00:03.49 | [r]evolution | i agree. |
| 00:03.56 | [r]evolution | so why dont you kick him off and you do it? |
| 00:03.59 | [r]evolution | be like FUCK OFF CUNT! |
| 00:04.02 | watchy2 | tk: i hate to ask you this but if i can drop you to a shell on the box you mind looking at it before they blow it away? |
| 00:04.49 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:04.51 | watchy2 | well i have alot of my plate today. #1 its my bday #2 i aint officially started #3 i'km about to go eat alot of sushi for my bday |
| 00:05.04 | [TK]D-Fender | watchy : PM me the info and I'll take a look |
| 00:05.08 | [r]evolution | i heart sushi. |
| 00:05.10 | watchy2 | i drove 2 hours up here to eat sushi today |
| 00:05.18 | watchy2 | i'm not cancelling those plans |
| 00:05.23 | watchy2 | thanks tk. lemme get t he info from the tech |
| 00:05.57 | watchy2 | awesome his phone goes straight to vmail |
| 00:06.26 | [r]evolution | you dont have ssh access? |
| 00:06.33 | [r]evolution | gay... if you're gonna fix it... im sayin |
| 00:07.01 | fujin_ | do a password recovery on it |
| 00:07.01 | watchy2 | the tech does. i'm not working this system so i have no idea of what the info is |
| 00:07.09 | watchy2 | hes onsite |
| 00:07.14 | watchy2 | i'm off site |
| 00:07.44 | watchy2 | we moved this customer from Rhino channel banks to Xorcom USB channel banks |
| 00:07.58 | watchy2 | anyone that wants to kick me in the nuts go ahead. i'll try take it and cry |
| 00:09.21 | watchy2 | i dont wanna talk bad about xorcom tzairaif or whatever works for em |
| 00:09.47 | [TK]D-Fender | watchy2: http://go-cry-emo-kid.ytmnd.com/ |
| 00:09.54 | *** join/#asterisk mardum_ (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 00:09.56 | watchy2 | haha |
| 00:10.48 | watchy2 | ok the techs emailing me the info tk. soon as i get it i'll give it to ya |
| 00:11.25 | sky_drive | [TK]D-Fender did you read my question about setting canreinvite=yes ? |
| 00:11.52 | [TK]D-Fender | sky_drive: Did you see my reiteration that you haven't provided what I asked for over half an hour ago? |
| 00:13.15 | watchy2 | anyways I really wanted to use some media gateways for the channel banks which was a suggestion from TK |
| 00:13.33 | watchy2 | but the guy i work with wanted to try Xorcom because of their white papers |
| 00:14.42 | *** join/#asterisk mbranca (i=daemon@mi-gw1.voismart.net) |
| 00:14.43 | *** part/#asterisk wacker (n=wacker@wb2flw.octothorp.org) |
| 00:14.56 | [TK]D-Fender | watchy2: Whats on your T1? |
| 00:15.32 | [TK]D-Fender | watchy2: WTF, you're acting as a timing source to your telco? |
| 00:15.35 | craigk | quick question about AMI events .... I am getting most of the events (like newchannel and link) but am not getting hold and unhold events - any ideas/suggestions ? |
| 00:15.39 | [TK]D-Fender | watchy2: Thats jsut wrong... |
| 00:16.06 | watchy2 | tk: hahaha |
| 00:16.17 | [TK]D-Fender | jsut got kicked |
| 00:16.17 | watchy2 | tk: hey tk. CHANGE WHATEVER YOU SEE THAT NEEDS CHANGING |
| 00:16.19 | [TK]D-Fender | from * CLI |
| 00:16.30 | watchy2 | he mighta restarted * |
| 00:16.35 | watchy2 | i dunno why |
| 00:16.41 | watchy2 | i'm not logged in |
| 00:17.00 | watchy2 | I didnt set this box up. I havent even seen its configuration |
| 00:17.47 | watchy2 | i wish it was a straight up * box and not some elastix freepbx crap |
| 00:18.27 | *** join/#asterisk saftsack (n=saftsack@pD9E075E7.dip.t-dialin.net) |
| 00:19.03 | *** join/#asterisk jtexter3 (n=jamest@ip67-90-136-204.z136-90-67.customer.algx.net) |
| 00:19.19 | [TK]D-Fender | watchy2: WTF, chan_zap didn't autoload |
| 00:19.28 | watchy2 | why not? |
| 00:19.36 | [TK]D-Fender | watchy2: no idea. |
| 00:19.54 | [TK]D-Fender | watchy2: and libpri stuff doesn't seem in... |
| 00:19.55 | watchy2 | could it be my tech is a idiot and broke something? |
| 00:21.14 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 00:21.15 | [TK]D-Fender | modules.conf looks normal(-ish) and shouldn't be stopping it, but its not loading on start |
| 00:22.16 | [TK]D-Fender | and "pri show span 1" fails |
| 00:22.29 | watchy2 | haha |
| 00:22.31 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:22.31 | *** part/#asterisk jtexter3 (n=jamest@ip67-90-136-204.z136-90-67.customer.algx.net) |
| 00:22.38 | watchy2 | lemme ssh in and check some of this |
| 00:22.48 | craigk | nvm my quick question ... found it :) |
| 00:25.48 | watchy2 | my god is that admin login shit annoying |
| 00:26.02 | [TK]D-Fender | watchy2: Whats loggin in all the time? |
| 00:26.15 | watchy2 | fucking freepbx cock sucking shit does that |
| 00:26.16 | [TK]D-Fender | watchy2: you should run astmanproxy |
| 00:26.51 | watchy2 | <PROTECTED> |
| 00:26.51 | watchy2 | <PROTECTED> |
| 00:26.51 | watchy2 | <PROTECTED> |
| 00:26.59 | watchy2 | you change anything? |
| 00:29.06 | [TK]D-Fender | nothing major, why? |
| 00:29.17 | watchy2 | it seems to be accepting dialin now i think |
| 00:29.31 | [TK]D-Fender | watchy2: jsut fixed your timing, but if you're running freepbx you'll have to update it in your interface. |
| 00:29.37 | watchy2 | i think its doing more then it was |
| 00:29.55 | [TK]D-Fender | watchy2: Also tell your telco to sent you *10* digit DID's |
| 00:30.21 | *** join/#asterisk sjobeck (n=sjobeck@72-34-70-131.skyriver.net) |
| 00:30.21 | watchy2 | instead of 9? |
| 00:30.56 | [TK]D-Fender | watchy2: Set("Zap/1-1", "__FROM_DID=9725") <-- 4 |
| 00:30.58 | *** join/#asterisk sjobeck (n=sjobeck@72-34-70-131.skyriver.net) |
| 00:31.09 | watchy2 | oh |
| 00:31.18 | watchy2 | thats how he sets up extensions |
| 00:31.28 | watchy2 | using his awesomeness in freepbx |
| 00:31.36 | watchy2 | he just puts the lst 4 digits |
| 00:31.39 | watchy2 | instead of the whole # |
| 00:31.48 | watchy2 | <PROTECTED> |
| 00:31.52 | watchy2 | its sending 10 |
| 00:32.08 | watchy2 | he just likes it to answer the last 4. i dunno why |
| 00:32.11 | [TK]D-Fender | watchy : No, its not... |
| 00:32.29 | watchy2 | oh |
| 00:32.31 | *** join/#asterisk Yourname` (i=Myztic@unaffiliated/yourname/x-837320) |
| 00:32.33 | [TK]D-Fender | watchy2: 5019415084 is the CALLERID. 9725 is the DID |
| 00:32.39 | watchy2 | ohg |
| 00:32.41 | watchy2 | i see |
| 00:32.42 | Yourname` | I'm going laptop shopping! |
| 00:32.46 | [TK]D-Fender | watchy2: kick your telco's ass |
| 00:33.03 | watchy2 | our telco i think is TimeWarner |
| 00:33.18 | watchy2 | atleast at this location |
| 00:33.54 | coppice | you let the animaniacs run your phone service? :-\ |
| 00:34.55 | fujin_ | Yourname`: don't buy a laptop! |
| 00:35.06 | fujin_ | they depreceate even worse than normal PC hardware! |
| 00:35.58 | watchy2 | haha |
| 00:36.02 | watchy2 | get a cheap 1330 |
| 00:36.13 | [hC] | unless you buy an apple! |
| 00:36.15 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-15940c19b43d364a) |
| 00:36.22 | fujin_ | lol apple |
| 00:36.24 | fujin_ | don't joke |
| 00:36.38 | fujin_ | they depreceate at the same rate, it's the exact same hardware |
| 00:36.44 | coppice | he's right. people use their apple notebooks for years |
| 00:36.49 | Yourname` | I know, but I can't help it.. :( |
| 00:37.09 | Yourname` | Apple just looks good. I don't know about performance. |
| 00:37.29 | Yourname` | blitzrage: Didn't you get some laptop from futureshop a few months ago? |
| 00:37.31 | _charly_ | i have my compaq laptop since about 6 years now |
| 00:37.38 | blitzrage | Yourname`: nope |
| 00:37.43 | *** join/#asterisk salzh (n=salzh@124.77.15.177) |
| 00:37.49 | blitzrage | I got MacBook Pro while I was in Arizona |
| 00:37.54 | Yourname` | Oh, sorry.. that was a TV? Right? |
| 00:38.20 | Yourname` | Nvmd. Anyway, yeah.. an associate says lenovo.ca is where I have to look, lol |
| 00:38.24 | watchy2 | the tech said hes been in telco 3 years |
| 00:38.32 | watchy2 | and sees no reason for 10 digit dids |
| 00:38.54 | [TK]D-Fender | watchy2: twit |
| 00:38.59 | `Sauron | watchy: Tell the tech he's fired. |
| 00:39.02 | watchy2 | he also said he likes freepbx and said "freepbx musta messed the config up when i added soime stuff" |
| 00:39.13 | watchy2 | but if it adds shit and breaks your shit |
| 00:39.17 | watchy2 | where your can fucking fix it |
| 00:39.30 | watchy2 | UNLESS YOU DELETE EVERYTHING YOU ADDED IN GOD DAMN FREEPBX |
| 00:39.34 | watchy2 | WHAT FUCKING GOOD IS IT |
| 00:39.44 | watchy2 | he went through and deleted ALL EXTENSIONS |
| 00:39.48 | Yourname` | All this because an HP-DV2000 cannot 'downgrade' to XP from it's vista :( |
| 00:39.49 | watchy2 | ALL ZAPTEL SHIT |
| 00:39.52 | watchy2 | and now it works |
| 00:40.14 | watchy2 | so now he gets to readd 120 extensions back |
| 00:40.19 | blitzrage | watchy2: turn down the language pls |
| 00:40.22 | watchy2 | that he added about 4 hours ago |
| 00:40.51 | watchy2 | no wonder this guy wants to goto cisco callmanaqger |
| 00:41.32 | watchy2 | he doesnt wanna take feedback from anyone |
| 00:41.59 | watchy2 | hes replaced this phone system at this company 5 times in the last 3 weeks |
| 00:42.13 | *** join/#asterisk ZX81 (n=matt@222-155-41-34.jetstream.xtra.co.nz) |
| 00:42.34 | `Sauron | I told you. Fire him. |
| 00:42.58 | ZX81 | Man Digium support rocks!!!!! Machine has stopped kernel panicking by using i686 hpec instead of i386 hpec (i thought 686 would just contain optimisations) |
| 00:43.00 | watchy2 | yes if i could hes been the only phone guy at this place for the past year or two |
| 00:43.25 | watchy2 | anyone wanna move to LR and take his place? |
| 00:43.44 | `Sauron | You couldn't pay me enough. |
| 00:44.00 | watchy2 | yea he only makies like $60/yr |
| 00:44.05 | watchy2 | makes |
| 00:44.42 | watchy2 | im heavily thinkin about not starting with this company |
| 00:44.45 | Yourname` | That's a decent amount to be paid for googling. |
| 00:44.56 | watchy2 | itd be easier to keep my current job without the hassle |
| 00:45.14 | watchy2 | yourname: if he didnt use freepbx he probably wouldnt be having these issues |
| 00:45.25 | watchy2 | because WHEN he has the damn issues he can't read the code to fix them |
| 00:45.35 | watchy2 | he just deletes everything in freepbx and starts over |
| 00:46.03 | watchy2 | whats wrong with gentoo + asterisk + zaptel :/ |
| 00:46.17 | watchy2 | oh yea no pretty gui |
| 00:46.49 | watchy2 | zomg were it workers in something susposily advanced but were using guis 5 year olds could use to setup phone systems |
| 00:46.52 | watchy2 | makes me wonder |
| 00:47.34 | watchy2 | so if anyone wants a job in little rock setting up FreePBX systems paying $60/yr lemme know |
| 00:47.53 | Yourname` | lol |
| 00:48.03 | watchy2 | clicking a gui for $60k is pretty good |
| 00:48.23 | Yourname` | Just make it a remote job ;) |
| 00:48.46 | Yourname` | bbl |
| 00:48.48 | watchy2 | i wanted to. but this guy is the main tech so i gotta put up with his freepbx installds |
| 00:49.38 | watchy2 | he aint my boss or anything but he does the main imps. I have * + lotsa unix/windows net admin exp |
| 00:49.43 | watchy2 | but not alot of telephony |
| 00:51.58 | watchy2 | man i want some sushio |
| 00:57.10 | *** join/#asterisk thinko (i=jdoe6alp@smaug.rackdragon.com) |
| 01:00.48 | [TK]D-Fender | watchy2: $60/yr? That wouldn't pay for the free support I've given you so far if I was billing :) |
| 01:04.25 | *** join/#asterisk saftsack (n=saftsack@pD9E075E7.dip.t-dialin.net) |
| 01:06.44 | *** join/#asterisk lepine (n=lepine@dsl-147-89.aei.ca) |
| 01:07.29 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 01:08.32 | lepine | sorry to bother you guys with an almost OT question which might result in my using a "competing" product ... but ... if i wanted to run a simple voip server for under 20 users ... tomostly have chat room, à la ventrilo ... i have no doubt asterisk could do it ... but as a newbie to voip as asterisk ... is asterisk overkill in my case? |
| 01:09.08 | lepine | that didn't come out too well ... i can clarify if need be |
| 01:09.15 | [TK]D-Fender | lepine: Guess ti depends on what you'd consider "overkill" |
| 01:09.48 | [TK]D-Fender | lepine: I' mean sure, you acn do just about anything with *, but do you already have an alternative thats dead easy to use and free? |
| 01:10.37 | lepine | overkill would simply be prolonged setup time ... eg, do i really need to spend more than a half hour on this project? |
| 01:10.59 | lepine | and not, i don't have an alternative yet ... asterisk is the first and only thing that came to mind |
| 01:11.00 | [TK]D-Fender | lepine: lol, HELL YEAH! |
| 01:11.26 | lepine | [TK]D-Fender: ^ @ setup time? |
| 01:11.28 | [TK]D-Fender | lepine: Or you could pay someone to set it up for you. Then again for your needs, Trixbox is right up your alley. |
| 01:11.45 | [TK]D-Fender | lepine: Setup no... learning curve = YES |
| 01:12.20 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 01:12.42 | lepine | indeed ... learning curve is what i feared |
| 01:12.57 | lepine | i'll be looking into trixbox then :) |
| 01:13.06 | [TK]D-Fender | ~trixbox |
| 01:13.06 | jbot | [~trixbox] Trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 01:13.18 | [TK]D-Fender | lepine: you'll need a dedicated PC for it |
| 01:13.31 | lepine | geh, i'm running debian on a cheapo vps |
| 01:13.52 | lepine | trixbox gets ruled out i guess :-/ |
| 01:14.54 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 01:15.10 | [TK]D-Fender | lepine: well at that point pretty much every solution I see is more work that you're looking forward to.. |
| 01:15.12 | xai | does anyone know how to re-set to factory a voxpath phone? |
| 01:15.44 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 01:19.59 | lepine | [TK]D-Fender : you're probably right ... otoh, this doesn't keep me from hacking asterisk ... and perhaps making a .deb or config set for stupid setups like mine |
| 01:20.24 | [TK]D-Fender | lepine: installing * takes a few minutes, CONFIGURING it is another matter |
| 01:21.18 | *** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 01:23.38 | lepine | hence hacking the conf and distributing it + scripts maybe :) |
| 01:25.07 | objective | knowing nothing, the fastest way to a dialtone is probably trixbox or switchvox-free-edition |
| 01:25.17 | *** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 01:25.27 | bmd | ~switchvox |
| 01:25.29 | objective | if you're contemplating writing scripts then you've already decided to spend a few days on it |
| 01:25.56 | lepine | the investment is worth it ... if not only for bragging rights |
| 01:26.32 | objective | then that's different than what you said when you first came in.... |
| 01:26.33 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 01:26.37 | [TK]D-Fender | lepine: * has already been packaged for Debian several times over. The cofigs for it all fit in 1 folder hence nothing to "install". |
| 01:26.59 | [TK]D-Fender | lepine: Your projects fits the description of "nearly non-existant" |
| 01:28.16 | lepine | bleh |
| 01:30.13 | *** join/#asterisk dlynes (n=chatzill@d154-20-45-103.bchsia.telus.net) |
| 01:37.53 | [TK]D-Fender | lepine: your project would be more challenging actually since on a VPS you probably couldn't (or only with great difficulty and uncertain results) run Zaptel which is required for your conference room. That'd mean you'd likely be compiling in 3rd party apps too... |
| 01:44.34 | *** join/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 01:51.20 | *** join/#asterisk khronos (n=khronos@c-66-229-159-175.hsd1.fl.comcast.net) |
| 01:51.47 | khronos | Hi, trying to setup an Aastr 933i to connect to my Asterisk server. |
| 01:51.59 | khronos | When I plug the phone in it tries to get an address from dhcp, but fails. |
| 01:52.18 | khronos | Whenb it sends the request it asks for vlan 100 and p 0. |
| 01:52.30 | khronos | None of my other machines do this. |
| 01:52.58 | khronos | How do I get the standard linux dhcp server to respond with the vlan the phone wants and also the p value? |
| 01:53.13 | khronos | I tried hard coding a host entry in for the mac address, but this didn't seem to wrok. |
| 01:53.14 | d-k-t | so, it's talking with packets tagged as vlan100? but your dhcp server isn't on this vlan? |
| 01:53.38 | d-k-t | the best way would probably to reset the phone to defaults |
| 01:53.49 | khronos | I don't have any vlans setup in my switch at all. |
| 01:54.01 | d-k-t | then it should default to vlan0 and be able to talk to your kit ok |
| 01:54.37 | d-k-t | ok, and the phone was used somewhere else before? |
| 01:54.47 | khronos | Is there a way I can setup the dhcp to be on vlan 100 so I can make this change in the phone? |
| 01:55.01 | d-k-t | what sort of switch do you have? |
| 01:55.07 | xai | Will a voip phone refuse an IP if it doesn't get a server to register to? |
| 01:55.14 | *** part/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 01:55.17 | khronos | I can't use the phone's interface unless I have exact key presses to reset the phone. |
| 01:55.32 | xai | Fo some reason this voxpath doesn't accept the IP that the dhcp server gives it. no idea why. |
| 01:55.34 | khronos | Dumb linksys rtp300 router. |
| 01:55.59 | d-k-t | khronos, most phones I've used have a reset sequence |
| 01:56.20 | d-k-t | xai, does it need additional options in the DHCP response? |
| 01:56.44 | *** join/#asterisk mardum (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 01:58.22 | xai | d-k-t: im not sure.. I can't find a manual for it.. It asks for an IP address, but it doesn't seem to like the ones we're trying to give it. |
| 01:58.47 | xai | d-k-t: maybe it does.. not sure.. what type of other response could it want? |
| 01:58.54 | xai | or options... |
| 01:59.26 | d-k-t | xai, some devices I've used will only accept the offer from the DHCP server if they have a special option included in the response, the special option varies by device and manufacturer |
| 02:00.13 | xai | maybe that is it.. Do generic voip phones like sipura do that too? |
| 02:01.06 | d-k-t | not used sipura phones |
| 02:01.14 | d-k-t | what phone are you using? |
| 02:01.26 | xai | voxpath vip-2400, seems very rare |
| 02:01.38 | xai | maybe its tied to a proprietary pbx |
| 02:03.01 | d-k-t | having a look |
| 02:04.06 | d-k-t | can you open this url? http://goharmonica.net/docs/ViP-2400%20DS.pdf |
| 02:04.12 | xai | d-k-t: nope.. |
| 02:04.16 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:04.19 | xai | gets stuck.. can you? |
| 02:04.57 | d-k-t | no, xai, how about http://64.233.167.104/search?q=cache:TY3b6Ox9hHEJ:goharmonica.net/docs/ViP-2400%2520DS.pdf+voxpath+vip-2400&hl=en&ct=clnk&cd=1&client=firefox-a ? |
| 02:05.27 | d-k-t | google cache is blocked here so I can't get into that |
| 02:05.31 | xai | yea, but I didn't see anything usefull. |
| 02:06.20 | xai | Maybe its defective, who knows. |
| 02:06.50 | d-k-t | ok, so, has voxpath folded? |
| 02:07.10 | d-k-t | at least none of their websites seem to work |
| 02:08.42 | xai | must have.. it was a $3 phone.. |
| 02:09.01 | xai | They'll let me exchange it.. for something. |
| 02:10.13 | xai | http://tinyurl.com/yuuqbj |
| 02:12.43 | d-k-t | hmm |
| 02:13.23 | xai | I'll bring my laptop over to their shop and see if they all act like that. |
| 02:13.30 | xai | They have a few boxes full of em. |
| 02:13.40 | d-k-t | voxpath was previously known as UINTAH MOUNTAIN COPPER COMPANY |
| 02:13.41 | d-k-t | <PROTECTED> |
| 02:13.55 | d-k-t | as a company, they seem confused |
| 02:14.16 | xai | indeed.. |
| 02:15.00 | xai | d-k-t: it was worth a shot.. $3 is a good price for a voip phone, it only it worked. |
| 02:16.35 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:16.35 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 02:22.09 | khronos | Anybody have the reset sequence for the Aastr 933i? |
| 02:22.59 | khronos | The phone manual says it ships by default with dhcp enabled. |
| 02:23.23 | khronos | I'm seeing this, but it doesn't say anything about it shipping asking for a different vlan than default. |
| 02:24.47 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:24.47 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 02:28.01 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 02:31.09 | d-k-t | khronos, I'm slowly looking for info, internet connectivity from china is painful |
| 02:33.32 | fujin_ | khronos: it probably makes use of CDP to guess the voice vlan |
| 02:33.59 | fujin_ | they shouldn't come preprogrammed to join a specific vlan, and even if they did it would only affect you if you had that vlan running |
| 02:34.13 | fujin_ | should be able to see what's going on with a careful Hub and wireshark/tshark |
| 02:34.47 | *** part/#asterisk mog (i=mog@nat/digium/x-f367bab683f1ccec) |
| 02:38.20 | *** join/#asterisk mardum_ (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 02:43.24 | *** join/#asterisk cjs (n=cjs@d131.GtokyoFL2.vectant.ne.jp) |
| 02:43.57 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:43.57 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:47.05 | d-k-t | khronos, options, phone status, restore defaults |
| 02:47.26 | d-k-t | then all defaults and # to confirm |
| 02:49.34 | dlynes | khronos: options -> phone status -> reset -> default all, options -> phone status -> reset -> reset config, options -> phone status -> restart phone |
| 02:50.14 | d-k-t | vlan is disabled by default |
| 02:50.28 | d-k-t | so the phone must have been configured with a vlan by someone else |
| 02:51.49 | dlynes | d-k-t: or perhaps there's some kind of dhcp option that specifies the vlan, and the 9133's getting it from that |
| 02:52.16 | d-k-t | dlynes, but his problem is that his phone is talking on vlan100 and isn't able to reach the DHCP server |
| 02:52.24 | fujin_ | I don't believe there's a registered dhcp option to set the VLAN, and, VLAN setup is pre-DHCP anyway. |
| 02:52.28 | fujin_ | that's where CDP comes in |
| 02:52.30 | fujin_ | (pre DHCP) |
| 02:52.40 | dlynes | fujin_: ah...don't know much about vlans, so ... |
| 02:53.03 | fujin_ | although you are correct, you *can* send a dhcp option for some phones (Mitel springs to mind) to re-set the vlan |
| 02:53.08 | fujin_ | i.e; run TWO dhcp servers |
| 02:53.21 | d-k-t | avaya phones do that |
| 02:53.56 | d-k-t | boot in default vlan, get lease, see option that says they should be on a different vlan, release, change vlan tagging, go for dhcp again |
| 02:54.19 | fujin_ | Yeah. |
| 02:54.22 | fujin_ | That's what the Mitels do, aswell |
| 02:54.26 | fujin_ | it's *terrible*. |
| 02:54.51 | d-k-t | but then again, relying on Cisco Discovery Protocol on non-cisco devices is pretty terrible too ;) |
| 02:55.16 | d-k-t | the Avaya phones at least remember the vlan they are supposed to be in across reboots with more recent firmware though |
| 02:55.24 | fujin_ | ah, that's handy |
| 02:55.39 | fujin_ | we were looking at a full avaya setup here |
| 02:55.40 | fujin_ | including pbx |
| 02:56.06 | d-k-t | one not so good feature is that by default, if they are unable to contact the gateway within 60 seconds, they will mark the VLAN as bad and not try it again |
| 02:56.25 | fujin_ | christ |
| 02:56.27 | fujin_ | that's ridiculous |
| 02:56.53 | d-k-t | so when say, the gateway is rebooted, all the phones drop onto vlan0, find they can get to the gateway as it's now back up and stay there |
| 02:56.56 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 02:57.07 | fujin_ | I had major issues with a batch of mitel 5224s, they apparently supported cdp but woudln't work |
| 02:57.12 | fujin_ | so we had to do the dual dhcp thing |
| 02:57.18 | fujin_ | and then they'd error out when downloading firmware and cook themselves |
| 02:57.21 | fujin_ | epic fail |
| 02:57.36 | d-k-t | when you are working with a /24 for PCs and a /24 for phones and have 200 seats, it doesn't work too well when that happens |
| 02:58.06 | d-k-t | I use a single DHCP server that's in both vlans |
| 02:58.34 | fujin_ | have had no issues with the Linksys ones, they support CDP, so, two vlans = easy |
| 02:58.37 | d-k-t | well, failover pair, but... |
| 02:58.43 | fujin_ | single dhcp server listening on the correct vlan |
| 02:58.49 | fujin_ | well (n+1 everything) |
| 02:59.17 | d-k-t | Linksys == cisco, so proper support of CDP would be a good move ;) |
| 03:00.32 | fujin_ | indeed |
| 03:00.37 | fujin_ | ~= cisco, anyway :] |
| 03:01.04 | d-k-t | avaya switches are, erm, primitive |
| 03:03.07 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 03:03.26 | d-k-t | one point had to get a firmware upgrade done on some avaya cajun P333TPWR switches, 1/3rd died with a failure to write to flash followed by a reboot and nothing |
| 03:04.16 | d-k-t | back to the avaya they went |
| 03:04.30 | d-k-t | since replaced them all with c3750s |
| 03:04.38 | fujin_ | nasty |
| 03:04.44 | fujin_ | yeah, we're on 3650's here |
| 03:04.45 | fujin_ | poe ones |
| 03:04.51 | fujin_ | they're very, very nice. very suitable |
| 03:04.54 | d-k-t | yep |
| 03:05.56 | d-k-t | and there's even high power models of the 48 port ones now that allow you to fully populate them with 15W devices |
| 03:07.27 | d-k-t | also a lot cheaper than the avaya kit |
| 03:07.30 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:08.14 | fujin_ | yeah?? |
| 03:08.18 | fujin_ | i would have assumed it to be the other way |
| 03:08.38 | d-k-t | 24 port poe switch from avaya was $3000, 2 x 1000baseT uplink module, $3000 |
| 03:09.53 | d-k-t | 24 port poe 3750 with built in 2 x 1000 and built in stacking was less than the price of the switch alone |
| 03:12.22 | fujin_ | damn |
| 03:12.23 | fujin_ | nice. |
| 03:13.21 | d-k-t | shame there was the whole multiple PoE standards issue, otherwise we'd have been able to avoid the avaya switches from the start, but, even the 3550 PoE switches were the cisco guess at what the final standard would be |
| 03:13.44 | fujin_ | thank god it's been ratified now |
| 03:13.50 | fujin_ | I have a/c power backups on all of my phones |
| 03:13.55 | fujin_ | for power-loss redundancy anyways |
| 03:14.04 | fujin_ | although dual-upsed PoE switches is probably enough |
| 03:14.37 | d-k-t | we don't go that far :) |
| 03:14.53 | d-k-t | avaya kit doesn't support it anyway |
| 03:15.20 | d-k-t | their power bricks are inline power injectors that cost $300 each |
| 03:15.34 | fujin_ | o_0 |
| 03:15.36 | fujin_ | That's ridiculous. |
| 03:16.22 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 03:16.22 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 03:18.15 | d-k-t | everything avaya is rediculously expensive, standard avaya set up, for each desk, you need your PoE switch port or power brick, $500 for the phone, $200 for the ip phone license on the PBX, $100 for the port license on the PBX, $75 for the voicemail license, $100 if you want call center functionality for that desk etc... |
| 03:18.30 | fujin_ | o_0 |
| 03:18.38 | fujin_ | I thought Cisco was rich with CCM |
| 03:18.45 | fujin_ | that's terrible~ |
| 03:19.13 | d-k-t | and you can't buy they license direct from avaya and apply it yourself, no, you need to go through a reseller/maintainer who insists on charging $1200 consultancy to apply the license pack |
| 03:20.15 | d-k-t | you may have noticed, I don't really like working with avaya kit |
| 03:20.21 | fujin_ | heh |
| 03:20.30 | fujin_ | I much prefer my handbuilt systems :D |
| 03:23.08 | d-k-t | oo, this one will make you laugh, guess how many parties can join a meetme conference on avaya kit? |
| 03:23.41 | fujin_ | there's a limit? |
| 03:23.51 | d-k-t | yep |
| 03:24.45 | d-k-t | 6 |
| 03:24.57 | fujin_ | that sucks |
| 03:25.02 | fujin_ | I've actually not had any use for meetme, at all. |
| 03:25.22 | fujin_ | my users have been happy with three-way calling |
| 03:25.33 | fujin_ | which the SPA942 does, without any asterisking. |
| 03:25.44 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-207-168.rgv.res.rr.com) |
| 03:25.59 | d-k-t | threeway calling on avaya kit allows you to conference up to 5 other people into a call, for a total of 6 |
| 03:26.07 | TrentCreek | Who does Prepaid? |
| 03:26.07 | d-k-t | their meetme implementation has the same limit |
| 03:26.14 | TrentCreek | cards that is |
| 03:26.33 | [TK]D-Fender | d-k-t: Thats like anal rape without the courtesy K-Y |
| 03:27.13 | TrentCreek | yeah..its Petrolum Jelly |
| 03:27.21 | [TK]D-Fender | d-k-t: Oh.... and my had office drank the Avaya IP Office Kool-Aid ;) |
| 03:27.30 | [TK]D-Fender | head* |
| 03:28.10 | d-k-t | [TK]D-Fender, indeed, haha, we managed to fool you into paying $300000 for the initial setup, now you'll have to keep paying if you want to actually use it |
| 03:29.57 | d-k-t | at least VoIP is opening doors to potentially being able to break out of the loop of then always having to buy avaya |
| 03:30.39 | TrentCreek | hey..don't put down Avaya! |
| 03:31.06 | TrentCreek | They got ot pay that gay black guy a lot of money ;-) |
| 03:32.00 | [TK]D-Fender | d-k-t: You mean SIP ;) UNISTIM FTW! |
| 03:37.23 | blitzrage | I'm doing my part to save the environment... I switched to a dark theme :) |
| 03:38.44 | [TK]D-Fender | blitzrage: yeah and "I'm not racist... I have a coloured TV" :p |
| 03:38.57 | blitzrage | o.O |
| 03:44.01 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 03:45.40 | *** join/#asterisk mardum (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 03:47.06 | *** join/#asterisk Dont_Panic_42 (n=user1@c-24-61-239-60.hsd1.ma.comcast.net) |
| 03:47.29 | Dont_Panic_42 | hello all |
| 03:48.43 | Dont_Panic_42 | has anyone had any trouble with wget when you were downloading form digum? |
| 03:51.43 | russellb | what kind of problem are you having? |
| 03:52.12 | SwK | probably the counter script :P |
| 03:56.15 | *** part/#asterisk jake1932 (n=Jake@pool-71-185-133-72.phlapa.east.verizon.net) |
| 04:12.54 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 04:17.25 | *** join/#asterisk d-k-t (n=dt@125.120.139.187) |
| 04:23.34 | *** join/#asterisk bkw_ (n=brian@ppp-70-128-124-90.dsl.tulsok.swbell.net) |
| 04:25.12 | *** join/#asterisk cjs (n=cjs@x239088.ppp.asahi-net.or.jp) |
| 04:26.46 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 04:26.48 | *** join/#asterisk idefine (n=panaii@c-24-99-1-240.hsd1.ga.comcast.net) |
| 04:26.52 | teknoprep | hey i thought you guys might want to know this |
| 04:27.08 | idefine | anyone know of any irc channels that talk about SIP and IMS? |
| 04:27.23 | teknoprep | XO is offering T1 ineternet access with unlimited SIP channels to a PBX for 479$ per month |
| 04:27.36 | teknoprep | thats alot of calls if you use 729 |
| 04:27.39 | teknoprep | g729 |
| 04:27.51 | russellb | nice .. |
| 04:28.33 | russellb | g729 and iax2 trunking :) |
| 04:28.59 | teknoprep | ? |
| 04:29.05 | teknoprep | XO doesn't offer iax2 |
| 04:29.14 | teknoprep | why would you use IAX2 over SIP tho? |
| 04:29.20 | teknoprep | i prefer sip |
| 04:29.33 | teknoprep | IAX2 is really nice for NAT tho |
| 04:29.40 | russellb | well, you'd get more calls that way, that's why i said it |
| 04:30.02 | teknoprep | really? |
| 04:30.08 | teknoprep | iax2 uses less bandwidth then sip ? |
| 04:30.25 | russellb | yeah, and _much_ less if you enable trunking |
| 04:30.39 | teknoprep | what do you mean enabled trunking? |
| 04:30.43 | russellb | which is putting the audio frames of a bunch of calls all into the same packet |
| 04:30.47 | russellb | it's a feature of the protocol |
| 04:30.53 | russellb | stick a bunch of calls in the same packet |
| 04:30.56 | teknoprep | ahh |
| 04:30.59 | russellb | and save on a bunch of IP overhead |
| 04:31.08 | teknoprep | can i do that to voicepulse ? |
| 04:31.15 | russellb | not sure, you'd have to ask them |
| 04:31.23 | russellb | it's something you specifically enable |
| 04:31.23 | teknoprep | well i could always just try |
| 04:31.28 | teknoprep | where? |
| 04:31.35 | teknoprep | whats the option to do this |
| 04:31.41 | russellb | trunking=yes in iax.conf i think |
| 04:31.44 | teknoprep | oh |
| 04:31.45 | teknoprep | thats it? |
| 04:32.00 | JT | that and make sure you have zap timing |
| 04:32.04 | JT | and not too many calls |
| 04:32.05 | russellb | er, trunk=yes |
| 04:32.09 | blitzrage | you need zaptel timing too (ztdummy at leastA) |
| 04:32.19 | blitzrage | if you have hardware already, then you're fine |
| 04:32.32 | russellb | ah yes .. |
| 04:34.04 | teknoprep | i use elastix with freepbx |
| 04:34.17 | teknoprep | so it already comes with ztdummy |
| 04:35.14 | JT | just remember ztdummy will never be as good as real hardware |
| 04:36.21 | teknoprep | yeah i know |
| 04:36.21 | teknoprep | but i have a 100% voip setup |
| 04:36.55 | JT | you need zaptel timing for iax trunking |
| 04:37.04 | JT | regardless of the fact it is voip |
| 04:38.07 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 04:38.13 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:45.26 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 04:46.38 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 04:53.08 | *** join/#asterisk Op3r (n=Op3r@222.127.42.216) |
| 04:54.32 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:54.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:55.29 | neax | http://crap.teurasporsaat.org/archive/5851.jpg |
| 05:04.31 | *** join/#asterisk Strom_M (n=strom@pool-71-243-217-40.lsanca.fios.verizon.net) |
| 05:07.53 | *** join/#asterisk peanut- (n=tokarev@50ae.net) |
| 05:08.34 | BBHoss | neax, lol |
| 05:10.06 | *** part/#asterisk peanut- (n=tokarev@50ae.net) |
| 05:10.44 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 05:11.07 | MrTelephone | have any of you dealt with sip clients that support their own forwarding, callerid block, threewaycalling? |
| 05:11.27 | MrTelephone | I have these clients that are meant to run on a proxy and they handle their own forwarding and crap |
| 05:11.42 | MrTelephone | but thats going to conflict with asterisks features |
| 05:14.48 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 05:15.53 | grimsy | how do queue members become paused? |
| 05:16.12 | grimsy | i don't have anything that i can see in my dialplan that calls PauseQueueMember |
| 05:16.37 | grimsy | but 5 out of 8 members are currently paused and not receiving calls |
| 05:21.02 | Op3r | maybe their softphone is in dnd mode? |
| 05:21.20 | [TK]D-Fender | MrTelephone, possibly |
| 05:22.10 | grimsy | Op3r: all physical phones and not in dnd |
| 05:22.27 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 05:22.30 | grimsy | thanks for the suggestion though :) |
| 05:23.11 | grimsy | restarting asterisk has un-paused them all now, but just interested as to how they got that way in the first place |
| 05:24.26 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:24.26 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 05:25.42 | [TK]D-Fender | grimsy, Go check your basic queue params for auto-pause.... |
| 05:25.59 | craigk | am i right in thinking that asterisk does not provide things like 'call forward' and 'do not disturb' but rather provides a means for me to implement them ? |
| 05:26.17 | grimsy | [TK]D-Fender: all set to autopause=no |
| 05:27.37 | MrTelephone | when my t1 card is RED alarm asterisk won't start :( |
| 05:29.41 | [TK]D-Fender | craigk, in a way. You can implement all those through the dialplan, or digital phones may be capable of making up their own mind and telling the server what to do and deciding when and how to ring |
| 05:30.06 | MrTelephone | ahh i had to run ztcfg -vv |
| 05:30.30 | craigk | [TK]D-Fender: thanks .... so I need to know what features my phones will implement and which ones I have to implement :) |
| 05:30.37 | MrTelephone | craigk, cable modem atas provide those features |
| 05:30.47 | MrTelephone | polycom 501s have a lot of forwarding features |
| 05:31.22 | grimsy | [TK]D-Fender: also, there's nothing about autopause in the book in the queues.conf section. am i looking in the wrong place? |
| 05:31.23 | MrTelephone | if your sip clients do all the work then you can use a sip proxy like openser and use asterisk for a pstn gateway or voicemail |
| 05:31.26 | [TK]D-Fender | craigk, and which side you want you users to work with to put into effect |
| 05:31.42 | [TK]D-Fender | grimsy, Give a good read to the sample config |
| 05:31.58 | grimsy | cheers, will do |
| 05:32.18 | [TK]D-Fender | MrTelephone, and conversely, if you leave it up to the phone and the phone drops out for whatever reason, POOF, no more forwarding to voicemail :) |
| 05:32.55 | [TK]D-Fender | I leave basic forwarding to the phone, nothing else. I do it in the dialplan if I want to be able to override it remotely though. |
| 05:34.43 | MrTelephone | yeah fender has a good point |
| 05:34.57 | MrTelephone | i don't like to depend on the client side connectivity |
| 05:35.06 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:35.06 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:35.43 | MrTelephone | i have a slight issue you might have seen before fender |
| 05:35.43 | teknoprep | i am tried of having echo and choppiness problems with VoIP providers |
| 05:36.06 | nestAr | buy a pri. ;) |
| 05:36.34 | teknoprep | why? |
| 05:36.39 | teknoprep | they are expensive |
| 05:36.41 | MrTelephone | when you call a sip client and it just recently dropped connetivity.. you don't hear a ring and it still takes 20 seconds to goto voicemail |
| 05:37.04 | nestAr | expensive, but the sound quality is great. |
| 05:37.10 | MrTelephone | it won't goto voicemail automatically unless asterisk marks the client as unreachable |
| 05:37.11 | teknoprep | lol |
| 05:37.58 | [TK]D-Fender | teknoprep, Yes, we can relate to your plight and your inability to escape it! You are welcome here to share your incorrigable woes! |
| 05:38.15 | teknoprep | ty |
| 05:38.20 | teknoprep | i appreciate that |
| 05:38.29 | [TK]D-Fender | MrTelephone, Because its the first attempt and there is no response to the call. |
| 05:39.47 | *** part/#asterisk hacim (n=micah@debian/developer/micah) |
| 05:40.03 | [TK]D-Fender | teknoprep, now go grab a bucket of Häagen-Dazs and have yourself a good cry! |
| 05:40.09 | teknoprep | dude |
| 05:40.22 | teknoprep | i already had a snickers ice cream cone |
| 05:40.27 | [TK]D-Fender | teknoprep, c'mon it'll make you feel better! |
| 05:40.30 | teknoprep | and am working on a bottle of vodka |
| 05:40.36 | MrTelephone | fender, do you know of a way aroudn that? |
| 05:40.50 | [TK]D-Fender | MrTelephone, if its the first call thats going to fail, then no. |
| 05:41.18 | MrTelephone | qualify doesn't work with the dlink routers I find |
| 05:41.29 | MrTelephone | I still have to set the clients to register every 30 seconds |
| 05:41.37 | [TK]D-Fender | MrTelephone, typically thats what qualify helps with. if you leave your hosts dynamic and they fail a qualify or a call they stay listed as unreachable until they reregister. |
| 05:41.52 | [TK]D-Fender | MrTelephone, D-Link routers don't work I find ;) |
| 05:42.02 | MrTelephone | they are buggy |
| 05:42.03 | [TK]D-Fender | MrTelephone, And yes, that IS the problem. |
| 05:42.14 | MrTelephone | but it still works |
| 05:42.16 | [TK]D-Fender | MrTelephone, Thanksfully it is the cheapest thing to replace. |
| 05:42.26 | MrTelephone | it still works though |
| 05:42.33 | MrTelephone | just have to register every 30 secs |
| 05:42.36 | [TK]D-Fender | MrTelephone, And No, it quite clearly isn't working out for you |
| 05:42.40 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 05:42.53 | Maliuta | anyone know where I can find a list of people doing VoIP provision to business in .au? |
| 05:42.53 | MrTelephone | i tried switching register to 120 secs and turn qualify on |
| 05:42.55 | [TK]D-Fender | "work around" != "work" |
| 05:44.12 | russellb | in case anyone is bored ... http://bugs.digium.com/svnstats/asterisk/trunk/ |
| 05:44.18 | russellb | just playing around |
| 05:46.06 | MrTelephone | thats a lot of lines |
| 05:46.19 | russellb | indeed |
| 05:47.19 | MrTelephone | count me in for 4 lines |
| 05:48.33 | russellb | chan_sip.c is almost to 20k lines |
| 05:48.35 | russellb | that's insane |
| 05:49.12 | MrTelephone | i wrote this invoices.pl and its almost 2k lines :( |
| 05:49.40 | MrTelephone | and i find it hard to navigate |
| 05:49.50 | MrTelephone | russellb, goto line 3000 |
| 05:50.09 | MrTelephone | and write me a patch to check nonce and username before pulling out the Authoriztion: header :) |
| 05:50.13 | MrTelephone | if your bored.. |
| 05:50.14 | MrTelephone | heh |
| 05:51.00 | russellb | ha, bored, but too tired for real coding |
| 05:51.29 | MrTelephone | i was debating to add a seperate get_header routine or add extra parameters to that one |
| 05:54.20 | MrTelephone | ah it was for some shitty little bug in the client anyways |
| 05:58.44 | *** join/#asterisk nirz (n=nir@bzq-79-178-22-251.red.bezeqint.net) |
| 05:59.10 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1096656572.dsl.bell.ca) |
| 06:02.11 | *** join/#asterisk KeltusEx (i=Keltus@c-76-21-119-166.hsd1.ca.comcast.net) |
| 06:04.43 | *** join/#asterisk mardum_ (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 06:05.06 | *** join/#asterisk idefine (n=panaii@c-24-99-1-240.hsd1.ga.comcast.net) |
| 06:10.09 | TrentCreek | Who does pre-paid??? |
| 06:10.33 | TrentCreek | I bet Mr Obvious does |
| 06:14.35 | MrTelephone | DNS SRV doesn't seem to work well |
| 06:15.24 | TrentCreek | I dont need DNS info ;-) |
| 06:15.43 | TrentCreek | trying to fiqure out some billing calcs |
| 06:15.53 | TrentCreek | Seems to be getting into a nightmare |
| 06:16.46 | *** join/#asterisk shinao1 (n=shinao1@196.207.1.30) |
| 06:20.11 | MrTelephone | billing calcs? |
| 06:20.41 | *** join/#asterisk ManxPower (n=manxpowe@110.sub-70-220-13.myvzw.com) |
| 06:27.25 | TrentCreek | yes |
| 06:27.36 | TrentCreek | have you done that? |
| 06:28.26 | MrTelephone | i export the cdr to mysql and bill a flat rate |
| 06:28.32 | MrTelephone | for long distance |
| 06:30.02 | SwK | what kinda billing calcs |
| 06:30.08 | SwK | they are pretty easy |
| 06:30.15 | SwK | depends on what bill increment you are doing |
| 06:30.26 | SwK | and how you are rating |
| 06:30.30 | TrentCreek | its per mintue |
| 06:30.35 | SwK | 60/60? |
| 06:30.53 | TrentCreek | I am trying to make a spread sheet that shows me how to make money |
| 06:30.55 | TrentCreek | yes |
| 06:30.59 | TrentCreek | 60/60 |
| 06:31.11 | *** join/#asterisk neax (n=newdle@125-237-23-251.jetstream.xtra.co.nz) |
| 06:31.13 | SwK | well then ceil(billsec/60) |
| 06:31.22 | SwK | then * rate |
| 06:31.32 | TrentCreek | oh no..that is not the problem..that is all taken care of |
| 06:31.57 | TrentCreek | it's the calcilations I need to do to figure out how to mak emoney |
| 06:32.04 | SwK | heh |
| 06:32.13 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-f57f1e9fa612a342) |
| 06:32.40 | *** join/#asterisk gardo (n=gardo@124.106.65.1) |
| 06:34.38 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 06:42.39 | MrTelephone | just charge 5 bucks a minute |
| 06:44.52 | SwK | MrTelephone, will you give me $5/minute flat rate to anywhere in the world? |
| 06:45.29 | TrentCreek | hey..that is a good way to go out of biz real fast |
| 06:46.08 | SwK | I wish someone would give me $5/minute flat rate anywhere in the world |
| 06:46.19 | MrTelephone | hahah |
| 06:46.23 | SwK | I need a really good route like that for irridium traffic :P |
| 06:46.23 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 06:46.24 | MrTelephone | why, whereare you phoning? |
| 06:46.29 | TrentCreek | I could do that |
| 06:46.51 | SwK | iridium is like $15 to $20/minute |
| 06:46.51 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:46.51 | TrentCreek | but you may not like the qaulity |
| 06:49.21 | Mavvie | http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Voicemail+Integration <- anybody ever done this? |
| 07:00.42 | *** join/#asterisk mistermocha (n=espresso@adsl-75-22-54-73.dsl.irvnca.sbcglobal.net) |
| 07:03.06 | TrentCreek | Maybe tey asking in a few hours when more people online |
| 07:18.57 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:29.56 | *** join/#asterisk harpal (n=harpal@124.125.255.24) |
| 07:36.53 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:39.00 | ManxPower | http://www.theregister.co.uk/2007/12/05/swat_conspiracy_guilty_pleas/ |
| 07:44.34 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:57.47 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 07:58.51 | *** join/#asterisk tsearle (n=torrey@97.110-246-81.adsl-static.isp.belgacom.be) |
| 07:59.29 | *** join/#asterisk shtoom (n=shtoom@59.93.116.155) |
| 08:01.04 | shtoom | hi is it possible to make a failover shift without lossing any of the SIP calls that are in progress? |
| 08:10.46 | *** join/#asterisk nirz (n=nir@mail2.tikalnetworks.com) |
| 08:11.44 | TrentCreek | ? |
| 08:12.08 | elzapp | 09:02 < _uplinkd_> +gotta 1337 speak something like porn... |
| 08:12.08 | elzapp | 09:02 < SixNein> +pr0n |
| 08:12.08 | elzapp | 09:02 < SixNein> +lol |
| 08:12.11 | elzapp | eek |
| 08:12.14 | elzapp | sorry |
| 08:13.10 | blitzrage | shtoom: sorry, not possible |
| 08:13.18 | blitzrage | (well, I'm sure it's possible, but you'd have to do some coding |
| 08:15.41 | TrentCreek | is it possible to have asterisk to phone every phone on the planet at the same time? |
| 08:15.44 | TrentCreek | ;-) |
| 08:16.09 | shtoom | blitzrate: what is the usual setup recomended for failover of SIP calls, as far as E1/T1s are concerned I've seen documents of red-fone phone bridge with Linux HA , but I am looking for a best setup in case of SIP |
| 08:17.29 | shtoom | TrentCreek:Its just a matter of a determined spammer getting the asterisk :D |
| 08:18.46 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:19.47 | *** join/#asterisk mardum (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 08:25.45 | watchy | anyone here |
| 08:25.50 | TrentCreek | hehe |
| 08:26.03 | watchy | anyone here do channel banks for FXS's? |
| 08:26.13 | watchy | say like for a place with alot of analogs |
| 08:26.57 | TrentCreek | who would want to use outdated and expensive analog? |
| 08:27.25 | Op3r | me |
| 08:27.26 | watchy | a nursing home? |
| 08:27.34 | Op3r | and some other cheapo |
| 08:27.36 | watchy | where old folks plug in there phone |
| 08:27.43 | watchy | from their home? |
| 08:27.51 | watchy | old folks can use digital phones |
| 08:27.53 | TrentCreek | why when you can get 12 incoming lines for only $35 a month |
| 08:27.55 | watchy | they are old and stupid |
| 08:28.05 | watchy | u igmo |
| 08:28.10 | watchy | pri coming in |
| 08:28.19 | watchy | analog in rooms |
| 08:28.35 | TrentCreek | oh |
| 08:28.40 | Op3r | who would want to use ip phones at home? |
| 08:28.45 | TrentCreek | you can convert it to digial |
| 08:29.03 | TrentCreek | me..its a LOT cheaper than analog |
| 08:29.14 | watchy | so what would you use for the channel banks? |
| 08:29.50 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:30.06 | TrentCreek | good question |
| 08:30.24 | Op3r | call centers |
| 08:30.39 | TrentCreek | big ATA? |
| 08:31.20 | watchy | i'm thinking a media gateway |
| 08:31.24 | watchy | like a Mediatrix |
| 08:31.27 | TrentCreek | for digital phones I think it would be easier |
| 08:31.53 | TrentCreek | then just get a MUX device to plug into the box to handle all those channels |
| 08:31.55 | watchy | well we already tried Rhino channel banks and they didnt work |
| 08:32.08 | watchy | then we tried Xorcoms with no luck, timing issues |
| 08:32.13 | watchy | they are freakin USB |
| 08:32.21 | JT | why don't you try quality |
| 08:32.27 | JT | like Adtran or Adit |
| 08:32.31 | JT | or CAC |
| 08:32.44 | watchy | #1 wasnt my project |
| 08:32.51 | TrentCreek | I think it would be just cheaper to buy a bunch of ATAs and get volume discount.. |
| 08:32.55 | watchy | #2 the guy on project picked rhino and xorcom |
| 08:33.02 | watchy | #3 guy is kicked off project |
| 08:33.09 | watchy | #4 unfortunately its my turn |
| 08:33.23 | TrentCreek | how many phones? |
| 08:33.27 | JT | TrentCreek: that's shit for manageability |
| 08:33.44 | tzafrir | watchy, timing issues? no can be :-) |
| 08:33.44 | watchy | 120 or so Analogs |
| 08:33.48 | JT | watchy: get T1 cards and a channel bank of the brand i specified, or get a SIP to FXS channel bank |
| 08:33.58 | tzafrir | Should have asked me :-) |
| 08:33.59 | watchy | tzafrir: fix devins issue then |
| 08:34.05 | JT | like Audiocodes, Mediatrix, or Vegastream |
| 08:34.08 | watchy | hes about to shoot himself |
| 08:34.16 | TrentCreek | you can easily set up 120 ATAs and forget about them |
| 08:34.23 | JT | TrentCreek: still a really shit idea |
| 08:34.29 | JT | incredibly bad |
| 08:34.44 | TrentCreek | would be cheaper than the other ideas that got the others fired |
| 08:34.49 | watchy | JT: i'm thinking of looking at Mediatrix SIP to FXS, what do you think? |
| 08:34.54 | JT | not really |
| 08:35.03 | JT | unreliable and a pain more like it |
| 08:35.14 | TrentCreek | probably get them for $30 a piece |
| 08:35.15 | watchy | tk recommended mediatrix |
| 08:35.18 | TrentCreek | or less |
| 08:35.24 | JT | watchy: oh, and one other brand is Patton |
| 08:35.29 | JT | TrentCreek: who cares |
| 08:35.33 | JT | it's a dumb idea |
| 08:35.38 | JT | a wall of 120 ATAs |
| 08:35.39 | JT | ... |
| 08:35.44 | TrentCreek | huh??? |
| 08:35.52 | JT | it's stupid |
| 08:35.52 | TrentCreek | now who is being stupid? |
| 08:35.57 | JT | the man hours to set them up |
| 08:36.01 | watchy | trend: i need 120 analogs |
| 08:36.03 | JT | will outstrip silly cost savings |
| 08:36.11 | TrentCreek | yiu out them in each room and run cabls to differen swtiches |
| 08:36.23 | JT | and using consumer grade ATAs instead of business grade media gateways is daft |
| 08:36.35 | JT | err they already have cabling suitable for analogue phones to every room |
| 08:36.56 | watchy | jt: so you think media gateways are nice? |
| 08:37.15 | TrentCreek | well okay...have a cabinet of 120 ATA..run off the old wires |
| 08:37.15 | JT | yeah they're very flexible a solution |
| 08:37.20 | watchy | i've been driving for 5 hours today. its 2:37 |
| 08:37.23 | watchy | i'm kinda outta JT |
| 08:37.28 | JT | channel banks with T1 cards maybe a little cheaper |
| 08:37.32 | watchy | so if i ask something more thenonce bare with me bro |
| 08:37.44 | JT | but maybe not if the channel banks are brand new |
| 08:38.00 | JT | because Adtran, Adit and CAC are carrier grade companies generally |
| 08:38.26 | watchy | well i'm new to using channel banks |
| 08:38.51 | JT | channel banks are pretty much T1 RBS only |
| 08:38.57 | JT | 24 port a T1 |
| 08:39.11 | TrentCreek | does Digium have a solution? |
| 08:39.13 | watchy | im going to bed guys, i have mega early morning appointments |
| 08:39.39 | JT | TrentCreek: if they had it their way, they'd make you buy five billion TDM2400Ps (and servers...) |
| 08:40.08 | JT | as they recommend neer using more than one digium card per server |
| 08:40.12 | JT | never |
| 08:40.15 | JT | err |
| 08:40.17 | JT | s/one/two/ |
| 08:40.29 | JT | but in reality the limit is often one |
| 08:40.30 | TrentCreek | yeah I noticed that Digium has been making Asterisk to direct you to buy their stuff |
| 08:40.46 | JT | but do they have a channel bank solution? nope |
| 08:40.53 | Op3r | cos they made asterisk? |
| 08:41.07 | TrentCreek | yeah..they would not sell many units if they did |
| 08:41.13 | TrentCreek | yeah.. |
| 08:41.46 | TrentCreek | i wonder how much longer before they kick up free loaders off ;-) |
| 08:42.01 | Op3r | when they got acquired by MS? |
| 08:42.04 | JT | come on, it's open source software |
| 08:42.21 | TrentCreek | yes, but for how long? Then no mor upgrades |
| 08:42.26 | JT | kicking people off for not buying digium wares is biting the hand that feeds |
| 08:42.35 | JT | so people will fork it if that happens? big deal |
| 08:42.43 | Op3r | its already been forked |
| 08:42.45 | JT | yes |
| 08:43.04 | JT | the forks will become more popular or more numerous |
| 08:43.10 | TrentCreek | the big companies will, but us small frys wont |
| 08:43.25 | JT | i don't understand rhetoric about freeloaders |
| 08:43.28 | Op3r | I heard the reason why it was forked because of asterisk is dependent on mark spencer |
| 08:43.34 | JT | it is open source software |
| 08:43.36 | JT | TrentCreek: ? |
| 08:43.41 | JT | Op3r: probably somewhat true |
| 08:44.28 | TrentCreek | Us small frys dont have the fiancial abilty to fork out thousands on phone systems hence VIOP is heavily used |
| 08:45.01 | JT | i don't know what that has to do with the conversation :) |
| 08:45.33 | TrentCreek | about Digium closing off Astrisk |
| 08:45.50 | JT | be more specific? |
| 08:46.46 | TrentCreek | If they get to become a powerhouse with a lot of installs and decided to to to propritary |
| 08:47.14 | JT | i don't see how it will change anything other than them reducing their customer level |
| 08:47.40 | Op3r | yeah and everyone flocking to callweaver |
| 08:48.07 | JT | or <insert name of open telephony project here> |
| 08:48.29 | TrentCreek | hehe |
| 08:48.46 | TrentCreek | well look what happened to M$ |
| 08:49.06 | Op3r | they were proprietary in the first place |
| 08:49.11 | TrentCreek | it was "open source" for a while ;-) |
| 08:49.41 | Op3r | I was a baby when bill gates send out that letter |
| 08:51.01 | TrentCreek | and look what happened since then...BSA and Windows registering |
| 08:53.30 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 08:55.12 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:01.40 | nexilus | ugh.. i thought i smelled "expensive" somewhere... took a while to find what channel it was |
| 09:02.54 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 09:09.27 | *** join/#asterisk cappiz (n=cappiz@217.68.105.61) |
| 09:09.57 | cappiz | someone knows of an adapter/hardware that allows you to use a mobile/sim-card as a trunk? |
| 09:12.14 | *** join/#asterisk grEvenX (n=even@1mldj72.ip.ssc.net) |
| 09:12.48 | R1ck | I'm trying to install a Junghanns QuadBRI card, but I get the following in /var/log/asterisk/full: chan_zap.c: Failed to read gains: Invalid argument - any idea how to fix that? |
| 09:13.24 | Op3r | ./genzaptelconf is a cool toy |
| 09:17.52 | agx | With Digium 4 BRI what are the suggested poll= and dsp_poll= value to receive faxes? |
| 09:19.52 | *** join/#asterisk mardum_ (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 09:21.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 09:22.20 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 09:24.08 | *** join/#asterisk oej (n=olle@193.126.30.214) |
| 09:25.37 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 09:27.25 | tzafrir | R1ck, sounds like the version of zaptel.h Asterisk was built with and the version of the zaptel kernel module don't match |
| 09:27.30 | tzafrir | e.g: 1.2 vs. 1.4 |
| 09:28.01 | yxa | hi i'm using a full E1 PRI. Each line/number is matched to a SIP fone. When I dial out using the sip phone using the whole zap group, opposite's party callerid displays the main number. How can make it show the DID of the SIP phone? |
| 09:31.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 09:32.12 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:32.15 | mosty | yxa, did you set CALLERID(num) before dialing the zap group? |
| 09:32.28 | yxa | mosty no i did not |
| 09:33.43 | yxa | how do i do it? Set(CALLERID(num))=? |
| 09:36.09 | TrentCreek | I think that method has expired |
| 09:36.15 | yxa | i already have callerid="Name" <666> set in sip.conf for all sip phones. does that matter? |
| 09:36.17 | TrentCreek | at least on the newest version |
| 09:36.40 | yxa | TrentCreek pls advise |
| 09:37.07 | TrentCreek | I have 1.4.11 and I set it that way and I see a message when I make a call out. |
| 09:37.53 | TrentCreek | it says "bla blas has been depreciated please use ......." |
| 09:38.31 | TrentCreek | or is it .10 I have..i cant recall..it is shut dhown now |
| 09:38.33 | R1ck | tzafrir: hmm, I compiled and installed from the BRIstuff packages |
| 09:38.37 | R1ck | -s |
| 09:38.52 | TrentCreek | I dont know what version theyhave |
| 09:38.53 | tzafrir | R1ck, what version of bristuff? |
| 09:39.13 | R1ck | 0.3.0-PRE-1y-m |
| 09:39.34 | yxa | TrentCreek i just need the opposite party to see the caller's DID and not the main number |
| 09:39.42 | R1ck | it includes asterisk 1.2.25 and zaptel 1.2.22 |
| 09:39.58 | tzafrir | ah, ok |
| 09:40.25 | TrentCreek | yes..so it should still be fine with that command |
| 09:40.26 | tzafrir | asterisk -rx 'show version' |
| 09:41.10 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 09:41.29 | TrentCreek | I also found out that caller ID name does not always function |
| 09:41.39 | R1ck | Unable to connect tzafrir, its not running because of the error, but asterisk -V says 'Asterisk 1.2.25-BRIstuffed-0.3.0-PRE-1y-m' |
| 09:41.42 | mosty | TrentCreek, set it before your dial the zap channel |
| 09:42.15 | mosty | yxa, rather |
| 09:42.32 | TrentCreek | No.. |
| 09:42.57 | TrentCreek | Some providers do some function that .... |
| 09:43.07 | TrentCreek | does a database lookup |
| 09:43.15 | tzafrir | ls -l /usr/sbin/asterisk /usr/lib/asterisk/modules/chan_zap.so |
| 09:43.28 | tzafrir | Both of generally the same time? |
| 09:43.37 | tzafrir | hmm... sorry, silly me: |
| 09:43.41 | tzafrir | surely they are |
| 09:43.43 | TrentCreek | And may over ride the name you give it |
| 09:44.09 | R1ck | tzafrir: ya exactly the same time |
| 09:46.23 | R1ck | cat /proc/zaptel/1 says : Span 1: ztqoz/1/1 "quadBRI PCI ISDN Card 1 Span 1 [TE] (cardID 0) Layer 1 DEACTIVATED (F4)" AMI/CCS |
| 09:46.35 | R1ck | why DEACTIVATED ? |
| 09:46.41 | R1ck | or is that normal |
| 09:51.00 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:52.15 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:58.38 | *** join/#asterisk nexilus (n=nexilus@gate.compodium.se) |
| 09:59.04 | nexilus | hm.. |
| 09:59.18 | R1ck | cat /proc/zaptel/1 says : Span 1: ztqoz/1/1 "quadBRI PCI ISDN Card 1 Span 1 [TE] (cardID 0) Layer 1 DEACTIVATED (F4)" AMI/CCS |
| 09:59.28 | R1ck | how do I get ACTIVATED ? |
| 09:59.37 | R1ck | does the line need to be plugged in? right now its not |
| 10:01.44 | *** join/#asterisk nexilus (n=nexilus@nexilus.com) |
| 10:06.32 | tzafrir | R1ck, layer 1 is down. Either you have wrong line parameters (ccs,ami), or the ports is disconnected |
| 10:07.03 | tzafrir | ami/ccs should work. |
| 10:07.07 | R1ck | they're disconnected allright, no lines have been plugged in |
| 10:07.21 | R1ck | they're set to ami/css |
| 10:07.36 | R1ck | but shouldnt asterisk just start up anyway? |
| 10:07.44 | tzafrir | One thing you can try is set one port to be NT and try a loopback |
| 10:08.40 | R1ck | well, can I disable those ports somehow that I dont use? |
| 10:08.50 | R1ck | or should I just not specify them in the zapata.conf |
| 10:09.22 | *** join/#asterisk mardum (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 10:09.28 | *** join/#asterisk shinao1 (n=shinao1@196.207.1.30) |
| 10:10.02 | mort_gib | A question... Transferring calls.. How do you do that?? Blind transfer or park calls?? |
| 10:12.37 | *** join/#asterisk Beave (n=beave@DCC.SEND.startkeylogger.000.telephreak.org) |
| 10:17.23 | *** part/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:17.31 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:17.58 | *** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu) |
| 10:19.34 | *** join/#asterisk shinao1 (n=shinao1@196.207.1.30) |
| 10:27.03 | *** join/#asterisk saftsack (n=saftsack@pD9E0480F.dip.t-dialin.net) |
| 10:34.00 | McDouglas | i'm having some trouble with the setting call-limit=1 in sip.conf. If i set it the user can only receive one call and he wont hear call waiting indication. Thats what i want. But if the user initiates a call, and someone else calls him he will receive the call waiting indication. (he wants me to disable it) |
| 10:36.06 | mosty | McDouglas, you can turn that off on the phone, usually |
| 10:36.24 | McDouglas | this is a panasonic dect phone with a sip ata |
| 10:36.37 | McDouglas | i cant find an option to turn it off in the ata |
| 10:36.47 | McDouglas | neither in the phone (its a really simple one) |
| 10:37.19 | *** join/#asterisk davyach (n=davide@pat-20m.hypergrid.it) |
| 10:37.20 | *** join/#asterisk harpal (n=harpal@124.125.255.24) |
| 10:37.29 | davyach | exit |
| 10:38.02 | mosty | McDouglas, you can use the GROUP_COUNT function to see if the phone has a call before dial'ing to it |
| 10:42.37 | *** join/#asterisk davyach (n=davyach@pat-20m.hypergrid.it) |
| 10:42.43 | agx | Is there a way to limit the standard Pickup (sometimes i use Pickup from bristuff) only to incoming call? actually it also pickup outgoing calls |
| 10:43.11 | *** part/#asterisk davyach (n=davyach@pat-20m.hypergrid.it) |
| 10:43.16 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 10:44.35 | *** join/#asterisk saftsack (n=saftsack@pD9E0480F.dip.t-dialin.net) |
| 10:45.11 | *** join/#asterisk davyach (n=davyach@pat-20m.hypergrid.it) |
| 10:45.45 | *** join/#asterisk mkl1525 (n=qwertz@p5098c328.dip0.t-ipconnect.de) |
| 10:46.43 | davyach | Is there anybody that is using the new AA50 asterisk appliance from digium ?? |
| 10:47.56 | mkl1525 | Hi, I'd like to store the caller id in a astdb value using "exten => 997,n,Set(DB(Agent_SIP_${CALLERID(num)})=1)" but after this extension was executed database show doesn't show the entry - so am I missing something? |
| 10:48.17 | mosty | mkl1525, why do you want that in astdb? |
| 10:50.46 | mkl1525 | mosty I need to check if an agent is already logged in on a telefon, so that no telefon can have two agents at the same time - any better idea on how to check for this? |
| 10:52.06 | mosty | i personally would use a postgres database, with realtime agents, then you can setup a database constraint so that an agent is only ever in one queue |
| 10:55.53 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 10:57.50 | mkl1525 | mosty thanks but problem isn't that an agent is only in one queue (normally we want to have some agents in two or more queues) - example: 1. agent logs in on sip phone 800, 1. agent goes to lunch, 2. agent sees free phone 800 and tries to logon - now * remembers only the last agent, 2. agent can log off from the phone, 1. agent (returning from lunch) can't log off anymore |
| 10:59.12 | mosty | you want a constraint that a phone is only ever in one queue maximum? |
| 11:00.37 | *** join/#asterisk rob_w (n=rob_w@gw.suellberg-hamburg.de) |
| 11:00.40 | mkl1525 | mosty no I'd like a 1 phone = 1 agent at a time constraint |
| 11:00.41 | rob_w | hi all |
| 11:01.10 | mosty | mkl1525, why don't you use sip accounts instead of agents? |
| 11:01.27 | *** join/#asterisk grEvenX (n=even@1mldj72.ip.ssc.net) |
| 11:02.18 | rob_w | can i get the old AT command able devices ttyI[0-9] at mISDN based interfaces |
| 11:03.16 | mkl1525 | mosty we're using snom phones and afaik you have to configure the sip accounts on the phone that would be a hazzle |
| 11:03.46 | mosty | you already have a sip account configured on the snom phones if you're using them now |
| 11:06.53 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 11:06.58 | R1ck | is it possible, for testing purposes, to connect a quadbri card in NT mode to a Siemens HiPath ISDN PBX (as a phone kind of) |
| 11:10.59 | *** join/#asterisk myiagy (n=myiagy@200.215.59.133) |
| 11:11.31 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:12.16 | agx | To avoid a phone picking up another phone making an outgoing calls i've to put in sip.conf for everyphone: pickupgroup=1 but callgroup=2 ? |
| 11:22.00 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-c14877e4e46b86d8) |
| 11:23.06 | *** part/#asterisk davyach (n=davyach@pat-20m.hypergrid.it) |
| 11:24.27 | *** join/#asterisk mardum_ (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 11:26.10 | McDouglas | mosty: i suppose i have to put the sip user into a group to be able to use the GROUP_COUNT, right? how is that done? |
| 11:26.29 | mosty | there's a GROUP function |
| 11:30.36 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-c14877e4e46b86d8) |
| 11:32.50 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:33.05 | billybongo | what's the best way to go about diagnosing media problems - I've got crappy audio from one upstream provider |
| 11:34.03 | mosty | what kind of crappy? |
| 11:34.12 | billybongo | distorted and dropping out |
| 11:34.48 | billybongo | I would say packet loss, but since they are only 1ms away from us it shouldn't be |
| 11:34.58 | billybongo | mtr running for a while returns no packet loss issues |
| 11:34.59 | mosty | dropping out as in cutting in and out? that usually means the network bandwidth issues |
| 11:35.04 | mosty | what codec are you using? |
| 11:35.10 | billybongo | g729a |
| 11:35.38 | billybongo | which of course sounds a bit crappy to start with |
| 11:36.06 | mosty | perhaps the latency/qos on the link is no good |
| 11:36.44 | billybongo | it should be fine - I'm on the same ISP as the provider |
| 11:37.01 | billybongo | 100mbits/sec ethernet between the two |
| 11:37.43 | mosty | can you try using g711? |
| 11:38.10 | billybongo | I could but it will take a while for them to change it, and they are currently investigating the issue |
| 11:38.21 | billybongo | I'm wondering if there's somewhere I can get reports on the media coming in my end |
| 11:41.26 | billybongo | for instance, provider asks me if I'm seeing lots of media errors; where would I look for those? |
| 11:41.44 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:41.56 | mosty | ifconfig |
| 11:42.44 | billybongo | 0 errors on tx and rx apparently |
| 11:43.25 | *** join/#asterisk dbaio (n=danilo@unaffiliated/dbaio) |
| 11:44.16 | cappiz | i use trixbox, but i'm wondering if asterisk has a webportal aswel? or is that a module i need to install? |
| 11:44.37 | cappiz | i want to setup asterisk in the same way as trixbox, but not using trixbox itself |
| 11:44.50 | billybongo | cappiz: take your pick |
| 11:44.58 | cappiz | ? :) |
| 11:45.13 | billybongo | there are quite a few out there |
| 11:45.22 | billybongo | have a look at asterisk now |
| 11:46.08 | cappiz | isnt that something i can install without installing a new OS? meaning, i can use asterisk now under ubuntu? |
| 11:46.30 | dbaio | hi... how can i get a password with a agi-bin ? i need some like this... when a sip will make a call, the asterisk say: hi, "dial you password". so, i check the password and ask what number the sip want to call...... i dont know how to read the digits in the midle of agi-bin..... any help me ? |
| 11:46.32 | cappiz | isnt/is* |
| 11:46.58 | billybongo | cappiz: well asterisk now uses asteriskgui, which I think is something you can just add on |
| 11:47.07 | cappiz | k |
| 11:47.09 | billybongo | I've never used a gui |
| 11:47.16 | billybongo | probably a bad person to advise you |
| 11:47.22 | billybongo | ignore me |
| 11:47.37 | mosty | cappiz, just install trixbox, then disable the web interface? |
| 11:47.48 | cappiz | i dont want to install trixbox |
| 11:47.56 | cappiz | i want the same kind of setup as trixbox, using ubuntu |
| 11:47.57 | cappiz | :) |
| 11:48.14 | tzafrir | cappiz, what do you mean by "exactly"? |
| 11:48.16 | mosty | you mean you want the trixbox interface, on top of ubuntu? |
| 11:48.23 | tzafrir | sorry: "same"? |
| 11:48.35 | cappiz | yeah |
| 11:48.42 | cappiz | but not installing the centos OS |
| 11:49.23 | mosty | cappiz, that coule be a lot of effort, there are probably easier interfaces you can put on top of ubuntu instead |
| 11:49.36 | cappiz | mosty, as log as there is a gui :) |
| 11:51.13 | mosty | actually, come to think of it maybe they are all complex in that way |
| 11:51.41 | cappiz | hehe :) |
| 11:53.03 | mosty | maybe try asking #trixbox |
| 11:53.14 | cappiz | yeah |
| 11:56.00 | *** join/#asterisk dbaio (n=danilo@unaffiliated/dbaio) |
| 11:58.57 | RoyK | is it possible to allow for unauthenticated INVITE if client is already authenticated with REGISTER? |
| 12:00.56 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 12:02.34 | *** join/#asterisk DJ_InstincT (n=marcus@87-194-124-8.bethere.co.uk) |
| 12:02.44 | *** join/#asterisk bantu (n=Miranda@p54A32CD1.dip0.t-ipconnect.de) |
| 12:10.28 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:13.24 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:21.32 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 12:23.03 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:23.40 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 12:26.50 | *** join/#asterisk monstertruck (n=monstert@68-240-111-240.area5.spcsdns.net) |
| 12:26.59 | monstertruck | hi |
| 12:26.59 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 12:27.25 | monstertruck | is there a way to conserve the value of a variable set in the t extension |
| 12:27.40 | monstertruck | for the next time the t extension gets called? |
| 12:28.24 | *** join/#asterisk saftsack (n=saftsack@pD9E0480F.dip.t-dialin.net) |
| 12:28.39 | R1ck | is it possible, for testing purposes, to connect a quadbri card in NT mode to a Siemens HiPath ISDN PBX (as a phone kind of) |
| 12:28.55 | dijungal | how do i get mysql support in Asterisk 1.4.15? I want to save my cdr and queuelog to mysql |
| 12:29.24 | mosty | you need asterisk-addons, i believe |
| 12:29.57 | monstertruck | or you can do it with agi |
| 12:30.24 | monstertruck | and then access the db from any supported language |
| 12:31.01 | dijungal | mosty: do i install this before or after i install asterisk, because i do not get the option in "make menuselect" |
| 12:31.21 | mosty | after |
| 12:31.39 | mosty | it's a seperate package |
| 12:32.20 | dijungal | k |
| 12:32.22 | dijungal | thanks much |
| 12:32.46 | dijungal | last time i installed asterisk 1.4, 1.2 was running on the machine, so i think i may have caused some issues |
| 12:33.13 | dijungal | and i mean actually running in memory, while i was doing the make distclean, make menuconfig... etc.. :S |
| 12:33.21 | dijungal | so i'm redoing the install |
| 12:33.28 | dijungal | with asterisk not running this time.. lol |
| 12:35.56 | dijungal | so how will i know if the mysql support is there? will there be a module?? mysql.so or something? |
| 12:36.02 | dijungal | or i can do a "show modules" |
| 12:36.04 | dijungal | ? |
| 12:36.54 | monstertruck | people, does anybody have any idea? im about 5 minutes away from just throwing it into a table and reading from it every time |
| 12:37.50 | mosty | monstertruck, you can use astdb, or you could write to a file, or you could write to an sql db |
| 12:38.06 | mosty | i would not recommend writing to a file |
| 12:38.37 | monstertruck | mosty, yeah, im just trying to avoid using the db for every sucker that calls in, gets the menu and lets it timeout |
| 12:39.14 | monstertruck | i was hoping i could somehow conserve a variable between subsequent calls to the timeout extension... |
| 12:39.30 | mosty | within the same call? |
| 12:39.35 | monstertruck | yes |
| 12:39.41 | mosty | just use a channel variable |
| 12:39.54 | monstertruck | doesnt work |
| 12:40.21 | mosty | odd |
| 12:40.28 | monstertruck | every time it times out, the variable is reset to zero |
| 12:40.42 | mosty | what version of asterisk? |
| 12:41.14 | *** join/#asterisk Rob782 (n=Rob@cpc4-sout2-0-0-cust715.sotn.cable.ntl.com) |
| 12:41.49 | monstertruck | 1.4 |
| 12:41.56 | mosty | monstertruck, perhaps you can avoid using the timeout extension, code the timeouts into your dialplan |
| 12:42.06 | mosty | then a channel variable would definitely work |
| 12:42.33 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 12:43.13 | Rob782 | Hey, any recommendations on a rack server - needs to support upto 250 light users.. Can't decide between dell, hp or supermicro... Not sure on what spec. Any suggestions? |
| 12:44.33 | RoyK | Rob782: ibm.com is a good place to start looking :) |
| 12:44.49 | dijungal | can i store extra cdr information for a call? for example caller name, talktime, call outcome? |
| 12:44.54 | mosty | dell will probably be the cheapest |
| 12:45.27 | Rob782 | well i really wanted dual psu for redundancy.. does dell support that? |
| 12:45.40 | mosty | yes |
| 12:45.53 | RoyK | Rob782: we have a few of these |
| 12:45.59 | RoyK | Rob782: http://www-03.ibm.com/systems/x/rack/x3550/index.html |
| 12:46.17 | RoyK | good stuff (tm) |
| 12:46.33 | Rob782 | just looking :) |
| 12:47.00 | dijungal | can i store extra cdr information for a call? for example caller name, talktime, call outcome? |
| 12:47.09 | Rob782 | The real problem i have is knowing that a PRI card will fit |
| 12:47.20 | RoyK | dijungal: use the userfield |
| 12:47.45 | Rob782 | RoyK, "The page you requested cannot be displayed" |
| 12:47.48 | RoyK | dijungal: also, callername is stored, talktime is stored as 'billsec', call outcome is stored as disposition |
| 12:47.59 | RoyK | http://www-03.ibm.com/systems/x/rack/x3550/index.html |
| 12:48.01 | RoyK | that one?? |
| 12:48.05 | RoyK | works for me (tm) |
| 12:48.14 | Rob782 | worked then. odd |
| 12:48.36 | dijungal | RoyK: userfield, to store many more stuff? |
| 12:48.52 | mosty | Rob782, PRI cards aren't large, i've never had problems fitting them in rack servers |
| 12:48.52 | dijungal | RoyK: lets say i wanna store 4 more new variables |
| 12:49.16 | Rob782 | mosty, cool |
| 12:52.51 | RoyK | dijungal: what do you want to store? |
| 12:53.20 | RoyK | Rob782: I have 5-6 1U boxes with 4 PRI each |
| 12:54.06 | RoyK | Rob782: just that sort of server, too, except and older model (xSeries 336) |
| 12:57.25 | *** join/#asterisk duckz (n=duckz@85.204.47.228) |
| 12:58.56 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 12:59.26 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 13:01.53 | Rob782 | RoyK, And how many concurrent calls is each box routing? And how much RAM :) |
| 13:03.06 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 13:03.39 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 13:03.47 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 13:09.51 | *** join/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 13:23.12 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:23.34 | RoyK | Rob782: RAM shouldn't be an issue unless memory leaks are around |
| 13:23.55 | RoyK | Rob782: concurrent calls varies, but peaks at 120 or so on this box |
| 13:24.49 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 13:24.49 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 13:25.04 | dijungal | i get this error when i try to install asterisk-addons-1.4.5 |
| 13:25.20 | dijungal | cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory |
| 13:25.20 | dijungal | make[1]: *** [install] Error 1 |
| 13:25.26 | *** join/#asterisk clive- (n=pirch@dsl-241-207-110.telkomadsl.co.za) |
| 13:25.56 | clive- | does anyone have any expereince with fastagi ?... I am having trouble passing the callerid to the agi |
| 13:27.47 | clive- | nexilus :) the calerid is not being picked up in the script...its baffling me |
| 13:31.54 | nexilus | as previously stated, i use deadagi, and there i get it just fine :P |
| 13:32.31 | R1ck | can I connect a regular ISDN phone to a quadbri isdn card? |
| 13:32.49 | tzafrir | no |
| 13:32.53 | tzafrir | You need an ISDN phone |
| 13:33.18 | tzafrir | Also note that for an ISDN phone you'll need to provide it power from the card. See the manual |
| 13:33.42 | nexilus | $agivar[agi_callerid] <-- that works fine for callerid for me :) |
| 13:34.20 | nexilus | (ofc, $agivar is a var i myself write from stdin in my case) |
| 13:34.53 | tzafrir | R1ck, oops, you wrote "ISDN phone": basically yes. You'll need to provide some external power source to the card. |
| 13:34.59 | tzafrir | IIRC |
| 13:35.30 | tzafrir | And set the port as NT, of course |
| 13:36.11 | R1ck | ah |
| 13:36.16 | R1ck | ok, i've already set it to NT |
| 13:36.25 | R1ck | should I get a dialtone? |
| 13:36.52 | R1ck | Span 2: ztqoz/1/2 "quadBRI PCI ISDN Card 1 Span 2 [NT] (cardID 0) Layer 1 DEACTIVATED (G2)" AMI/CCS |
| 13:36.56 | JT | did you set the jumpers/dip switches? |
| 13:37.06 | R1ck | thats where I plugged in the phone |
| 13:37.12 | R1ck | the 3 ports all say "In use" |
| 13:37.26 | R1ck | i should check the diagram about the powering thing |
| 13:39.00 | *** join/#asterisk mltlnx (n=mltlnx@64.3.170.41.ptr.us.xo.net) |
| 13:40.29 | tzafrir | R1ck, is the phone powered? |
| 13:41.01 | tzafrir | "(In use)" means Asterisk uses the channels. |
| 13:46.46 | R1ck | tzafrir: i cant tell :) |
| 13:46.56 | R1ck | i just dont get a dialtone |
| 13:47.13 | tzafrir | I suspect it isn't |
| 13:47.30 | R1ck | and the led is red, meaning "Layer 1 down" |
| 13:47.51 | R1ck | i toggled a switch on the board, but it doesnt seem to power it |
| 13:49.55 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.125.196) |
| 13:50.26 | mosty | is there a way to somehow see all the channel variables etc for a call going out a PRI line? i can call from my sip phone via the pri line, but i can't set a phone to redirect to the same number (caller just gets the congested tone) |
| 13:55.03 | dijungal | i get this error when trying to compile asterisk-addons-1.4.5, any reasons why? |
| 13:55.03 | dijungal | cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so |
| 13:55.03 | dijungal | cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory |
| 13:55.03 | dijungal | make[1]: *** [install] Error 1 |
| 13:55.21 | R1ck | ah, it seems I need a "PFM module" to power NT ports |
| 13:56.44 | mosty | dijungal, i'm no psychics, but i guess that file does not exist |
| 13:57.14 | dijungal | mosty: but how do i get around that... and if the file does not exists why does this CRASH the install??? |
| 13:58.05 | mosty | either install the package that provides it, or disable chan_ooh323 from the build |
| 13:59.00 | dijungal | how do i disable it from the build?? |
| 13:59.31 | dijungal | i prefer to disable it, i'm not using ooh323 |
| 13:59.33 | mosty | my guess would be with a configure flag, or in make menuselect |
| 13:59.43 | dijungal | ahhh menuselect |
| 14:00.09 | dijungal | ahh that worked :) |
| 14:00.10 | dijungal | thanks |
| 14:00.37 | iratik | who's the best termination/origination provider ... in terms cost vs. quality vs. reliability... reliable, high quality and cheap... any recommendations? |
| 14:01.06 | JT | to terminate where? |
| 14:01.06 | dijungal | ahhhh but that did not help my purpose... when i enter make menuselect, i cannot select cdr_Addon_mysql... |
| 14:01.19 | iratik | ah... US/Canada Termination |
| 14:01.20 | mosty | dijungal, you need mysql dev libs installed |
| 14:01.25 | dijungal | ok thanks |
| 14:02.21 | *** join/#asterisk quelo (n=quelo@host8-184-dynamic.2-79-r.retail.telecomitalia.it) |
| 14:03.36 | RoyK | Is it possible to allow for unauthenticated INVITE if client is already authenticated with REGISTER? |
| 14:04.07 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:04.07 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:04.19 | iratik | any ideas? |
| 14:04.42 | iratik | recommendations.... i mean if i said termination.com was the best... or teliax.com was the best--- i would start an argument over who is the best |
| 14:05.49 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 14:06.16 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:06.44 | iratik | I'm leaning towards voicepulse? |
| 14:06.47 | iratik | any objections? |
| 14:08.50 | killfill | hey |
| 14:09.17 | killfill | ${EXTEN} is the extension a user calls. how do i get the user who is calling? |
| 14:09.31 | killfill | where is he calling from |
| 14:10.01 | [TK]D-Fender | iratik: Take a look through these guys and compare : |
| 14:10.07 | [TK]D-Fender | ~itsplist-us |
| 14:10.07 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com |
| 14:10.09 | [TK]D-Fender | ~itsplist-ca |
| 14:10.10 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 14:10.12 | iratik | thanks |
| 14:10.37 | [TK]D-Fender | killfill: that would be CALLERID |
| 14:11.20 | [TK]D-Fender | iratik: everything depends on exactly where you are calling, how many channels needed, etc. |
| 14:14.01 | *** join/#asterisk mardum (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 14:14.18 | Maliuta | ~itsplist-au |
| 14:15.46 | [TK]D-Fender | Maliuta: do you have recommendations for AU? |
| 14:16.07 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:18.58 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 14:20.25 | *** join/#asterisk cjk (n=loic@80.92.64.103) |
| 14:21.19 | cjk | hi, in which context do i need to put my "hint" extensions? into subscribecontext or context? |
| 14:22.03 | mocker | cjk: In the context your phones subscribe to. |
| 14:22.28 | mocker | cjk: What I did, was just make a hints.txt that has all of them and just include that. |
| 14:22.45 | mocker | (Also makes it easy to test if you have it in the right place, because you just move the line around) |
| 14:22.57 | cjk | mocker, so subscribecontext is correct and this context does not need any dial or what every dommands, just a list of the hint priorities ie enough? |
| 14:23.39 | *** part/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 14:23.53 | mocker | Well, you probably want your phones to be able to dial out. |
| 14:23.56 | mocker | And receive calls. |
| 14:25.09 | cjk | mocker, well subscribecontext and context are different |
| 14:25.11 | cjk | in my case |
| 14:25.20 | quelo | Hi to all |
| 14:25.28 | cjk | so i have hte hint in subscribecontext and the dialout logic in context |
| 14:25.31 | cjk | is this a problem |
| 14:25.36 | cjk | should they be the same? |
| 14:26.07 | quelo | I'm going to setup a trunk to join a trixbox asterisk server to an avaya ip 406 |
| 14:26.21 | JT | ~trixbox |
| 14:26.22 | jbot | [~trixbox] Trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 14:27.09 | [TK]D-Fender | cjk: Doesn't matter. |
| 14:27.09 | mocker | cjk: The wya I have it is sip phone [100] has context=foo in the sip.conf |
| 14:27.28 | mocker | And then the hints include is in foo in extensions.conf |
| 14:28.03 | [TK]D-Fender | cjk: subscribecontext is only something you fill in if the hints aren't accessible to the normal context it uses |
| 14:28.22 | mocker | morning [TK]D-Fender |
| 14:28.47 | [TK]D-Fender | mocker: mornin' |
| 14:28.57 | cjk | ok thanks for the confirmation so i can split.... now i have another question. if the light blinks on my phone and i push that button to pickup the call. then the phone needs to dial an extension which is in context. i know that grandstream dial **EXTEN. is there any common standart for this? |
| 14:30.15 | *** join/#asterisk skirmisha (i=skirmish@90.154.200.195) |
| 14:30.18 | skirmisha | guys |
| 14:30.36 | skirmisha | anyone here with experience in openser+ asterisk realtime? |
| 14:31.32 | [TK]D-Fender | cjk: No, if the GS is capable of sending 1 exten if its blinking, and another if it isn't, then that is unique in my experience. |
| 14:31.43 | [TK]D-Fender | cjk: And maybe the one smart thing they've ever done |
| 14:32.19 | cjk | [TK]D-Fender, honestly grandstream rock in features, only problem is that you need to order the triple of the quantity you need because they stop working fast |
| 14:32.34 | mocker | ~grandstream |
| 14:32.34 | jbot | i heard grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 14:32.37 | skirmisha | anyone here that can help me???? |
| 14:32.42 | mocker | :P |
| 14:33.00 | mocker | skirmisha: Sorry never messed w/ openser. |
| 14:33.06 | mocker | Or realtime for that matter. |
| 14:33.13 | mocker | Only DB I do is for VM. |
| 14:33.36 | [TK]D-Fender | skirmisha: Ask a SPECIFIC question and maybe someone will answer. |
| 14:33.45 | mocker | ~question |
| 14:33.46 | jbot | it has been said that question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 14:33.55 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:34.16 | skirmisha | i need to know when i have view of table in asterisk DB which is coming from openser db, do i need to update that view with users and passwords that are stored in real asterisk table |
| 14:34.58 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 14:36.10 | quelo | I'm going to setup a trunk between an asterisk box and an avaya ip office 406 that use h323 protocol |
| 14:36.18 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:36.18 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:36.25 | *** part/#asterisk clive- (n=pirch@dsl-241-207-110.telkomadsl.co.za) |
| 14:37.02 | quelo | now from an asterisk extension to an avaya extension I can call properly, but the contrary doesn't work |
| 14:37.37 | [TK]D-Fender | skirmisha: if * auths calls that pass in from SER, then I'd say yes. |
| 14:38.12 | quelo | If I try to call from an avaya extension for example 67248 (where 248 is an asterisk extension and 67 is the function code that route vs asterisk) |
| 14:38.42 | skirmisha | ok thanks guys |
| 14:39.00 | quelo | I have this log in trixbox... |
| 14:39.10 | [TK]D-Fender | quelo: Sorry Trixbox is NOT supported here, please refer to their channels and other resources for your issues |
| 14:39.52 | RoyK | oej: Is it possible to allow for unauthenticated INVITE if client is already authenticated with REGISTER? |
| 14:39.56 | quelo | http://paste.debian.net/44297 |
| 14:40.19 | oej | RoyK: No, those are two unrelated transactions |
| 14:40.32 | oej | You might re-try with the previous nonce, but still need to authenticate |
| 14:40.38 | quelo | I wrong isn't a trixbox |
| 14:40.49 | quelo | is an asterisk server |
| 14:41.10 | RoyK | oej: I just see that quite a few other sip products have this functionality |
| 14:42.01 | [TK]D-Fender | quelo: You told us twice already, don't expect us to start believing different. And forget about jsut trying to send us debug dumps like that, we'd need to see the complete CLI output at verbose 10 and H.323 debug enabled. |
| 14:42.19 | JT | quelo: ooh323 sucks anyway with asterisk |
| 14:42.25 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:42.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:43.37 | quelo | ok i try to enable verbos mode at 10 and h323 log (how I can activate h323 log?) |
| 14:53.15 | *** join/#asterisk codejunky (n=jan@codejunky.org) |
| 14:54.52 | codejunky | Hi, I am connecting with asterisk to my sip provider. Is the connection per default encrypted or unencrypted? Does asterisk support encrypted connections? |
| 14:55.05 | *** join/#asterisk etfonhomey (n=chatzill@12.169.248.226) |
| 14:55.09 | codejunky | I mean for the sip portocol |
| 14:55.40 | JT | unencrypted |
| 14:56.18 | coppice | ah, Tales from the Encrypt |
| 14:59.17 | *** join/#asterisk phsdshft (n=noone@204.56.88.111) |
| 15:02.29 | [TK]D-Fender | quelo: http://www.voip-info.org/wiki/view/Asterisk+CLI |
| 15:02.48 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:04.05 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 15:05.19 | phsdshft | Good morning.. I'm using Broadvoice.. I'm behind NAT (and have NAT configured in sip.conf..) I register successfully but when I initiate a call (send an invite) broadvoice replies back with 401 unauthorized... What are the likely reasons for this? |
| 15:07.13 | *** join/#asterisk af_ (n=getsmart@88-149-241-31.dynamic.ngi.it) |
| 15:07.47 | *** join/#asterisk saftsack (n=saftsack@pD9E0480F.dip.t-dialin.net) |
| 15:08.32 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 15:09.14 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 15:09.26 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 15:10.25 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:10.29 | brodiem | Anyone know if it's possible to stop app_queue from logging "RINGNOANSWER" events when the result of Dial is congestion/busy? Or just disable it altogether since logging as no answer when busy majorly fluffs reporting anyway? |
| 15:10.33 | [TK]D-Fender | phsdshft: Bad user / pass. Thats it, thats all |
| 15:10.50 | [TK]D-Fender | phsdshft: it means what it says. Go follow their guides and pay close attention. |
| 15:11.49 | McDouglas | how do i globaly disable call waiting feature in asterisk? |
| 15:12.41 | tzafrir | McDouglas, it is channel-dependent |
| 15:12.53 | McDouglas | well, i want to disable it for my sip channels |
| 15:13.20 | [TK]D-Fender | McDouglas: then thats typically phone based and you had to prevent the phone from doing it. |
| 15:13.42 | [TK]D-Fender | McDouglas: * can't stop them. |
| 15:14.13 | McDouglas | [TK]D-Fender: the phone is connected to a sip ata, and the ata does support call waiting, but unfortunately doesnt seem to allow the user to disable it |
| 15:14.18 | [TK]D-Fender | McDouglas: You can try using the call-limit options in sip.conf or checking to see if they're on a channel already. |
| 15:14.29 | McDouglas | i already use the call-limit |
| 15:14.32 | [TK]D-Fender | McDouglas: crappy ATA then. |
| 15:14.40 | McDouglas | and it works strange |
| 15:15.38 | McDouglas | my user can only receive one call, thats fine, but if the user initiates a call, he can receive one more (at least the phone will signal the call waiting) |
| 15:16.10 | McDouglas | [TK]D-Fender: can you recommend a good ata then? |
| 15:16.48 | [TK]D-Fender | McDouglas: that doesn't make sense. |
| 15:16.54 | *** join/#asterisk shinao1 (n=shinao1@196.207.1.30) |
| 15:16.56 | phsdshft | Fender: it registers.. if it was the wrong password, it wouldn't register.. |
| 15:17.07 | McDouglas | [TK]D-Fender: well, it still hapens |
| 15:17.14 | [TK]D-Fender | phsdshft: Registering has NOTHING to do with placing a call. |
| 15:17.36 | *** join/#asterisk techie (n=techie@adsl-76-214-29-238.dsl.lsan03.sbcglobal.net) |
| 15:17.38 | cappiz | im very new at pbx/asterisk and stuff.... but i have a analog line - if i want to have that associated with my PBX, what kind of hardware do i need? |
| 15:17.52 | *** join/#asterisk slowshutt (n=d@196.211.34.2) |
| 15:18.11 | [TK]D-Fender | cappiz: only 1 line planned? |
| 15:18.15 | slowshutt | hi there can one dial more than one sip phone at a time? |
| 15:18.32 | cappiz | [TK]D-Fender, yeah - i also have SIP |
| 15:18.34 | [TK]D-Fender | slowshutt: Yes, read Dials instructions : show application dial |
| 15:18.42 | cappiz | but only one line |
| 15:18.47 | [TK]D-Fender | cappiz: For home use basically? |
| 15:18.57 | cappiz | you could say so |
| 15:19.02 | mocker | cappiz: What's your goal? If you're just trying to learn Asterisk I would play w/ straight voip. |
| 15:19.04 | slowshutt | thx TK great help |
| 15:19.09 | mocker | Don't have to buy any hardware or anything. |
| 15:19.09 | phsdshft | Fender: Broadvoice confirmed that the password is correct... It also works correctly when NAT is not used (using the same config w/o the NAT).. |
| 15:19.14 | mocker | (well, besides the computer) |
| 15:19.50 | [TK]D-Fender | cappiz: Good bet would probably be the Linksys SPA-3102 ATA. It'll let you take in your line AND let yuo use 1 analog phone as a SIP device. $75 +/- |
| 15:19.57 | cappiz | mocker, i need a line - need to setup DISA. and that analog line is a part of a business plan |
| 15:20.14 | [TK]D-Fender | cappiz: this is an external ethernet device |
| 15:20.16 | mocker | cappiz: Ahh, so not just for play then. :) |
| 15:20.23 | cappiz | mocker, nope :) |
| 15:21.00 | cappiz | [TK]D-Fender, so it communicates with the PBX via ethernet? |
| 15:21.26 | [TK]D-Fender | cappiz: yes, SIP over UDP |
| 15:21.44 | teknoprep | hey all |
| 15:22.00 | [TK]D-Fender | cappiz: no need to muck around with cards & drivers and of course that means you can place it anywhere you want relative to your server |
| 15:22.10 | cappiz | analog-line <-> ATA <-> asterisk ? |
| 15:22.38 | mocker | cappiz: Yeah, you should probably read up on FXO vs FXS if you haven't yet. |
| 15:22.42 | cappiz | and it does handle traffic in both ways? |
| 15:22.44 | cappiz | k :) |
| 15:22.44 | mocker | That way you don't get tripped up. |
| 15:22.45 | [TK]D-Fender | cappiz: Yes. The 3102 has a FXO port for your line, and an FXS port for an analog phone. Both ports operate completely independant of eachother |
| 15:23.03 | McDouglas | just wondering, can i use a fax machine with a sip ata to connect it to asterisk? |
| 15:23.08 | mocker | ~thebook |
| 15:23.09 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 15:23.21 | mocker | cappiz: ^^^ good reference |
| 15:23.36 | cappiz | thanks :) |
| 15:23.44 | cappiz | i'll read up on it :) |
| 15:24.16 | slowshutt | Dial(SIP/100[&SIP/200][|20][|tT]) does this look right Tk? |
| 15:24.32 | mocker | Why the [] stuff? |
| 15:24.33 | [TK]D-Fender | slowshutt: remove all the []'s |
| 15:24.40 | slowshutt | thx |
| 15:24.56 | [TK]D-Fender | mocker: because he doesn't understand how [] is used to seperate nested option parameters |
| 15:25.15 | slowshutt | Dial(SIP/100&SIP/200|20|tT) does this look right Tk? |
| 15:25.18 | [TK]D-Fender | slowshutt: Which hopefully you do now |
| 15:25.28 | [TK]D-Fender | slowshutt: Yes, 2nd try, not bad |
| 15:25.32 | slowshutt | learning slow but surely |
| 15:25.45 | [TK]D-Fender | slowshutt: No, not bad so far. |
| 15:25.45 | slowshutt | if it wasn't for this channel i would be lost |
| 15:25.55 | phsdshft | fender: can you review my broadvoice sip.conf file just to be sure that it is correct? I'm using one of the many available templates.. |
| 15:26.29 | slowshutt | can you use the , instead of the | sign? |
| 15:26.47 | slowshutt | Dial(SIP/100&SIP/200,20,tT) will aslo work? |
| 15:26.49 | [TK]D-Fender | phsdshft: pastebin THEIR sample in 1 PB, and your's in another masking only passwords. In yours please include the full CLI output of your failed call at verbose 10 |
| 15:26.51 | [TK]D-Fender | ~pb |
| 15:26.51 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:26.52 | dijungal | is the agents module still flawed or have issues in 1.4 asterisk? |
| 15:26.54 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 15:27.34 | Qwell | dijungal: the callbacklogin mode was and still is very much flawed |
| 15:27.44 | [TK]D-Fender | slowshutt: Typically yes. for a long time the "|" and "," were both valid parameter delimiters. This is changing, so I highly advise you use "," everywhere |
| 15:27.48 | coppice | mocker: that's unkind. all the other coloed servers will laugh at it |
| 15:27.59 | mocker | coppice: Psh. |
| 15:28.03 | mocker | They won't even notice it! |
| 15:28.10 | mocker | I can hid it behind a power strip. |
| 15:28.13 | mocker | er, hide |
| 15:30.45 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 15:31.07 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 15:31.24 | cjk | hi, all my labels are 'n'. how can i goto n+1 |
| 15:31.39 | [TK]D-Fender | cjk: use labels. Go read the book. |
| 15:31.58 | grandpapadot | cjk: labels |
| 15:32.32 | cjk | ok thanks |
| 15:33.55 | mocker | Anyone know if the feature for MeetMe to get the participants name before joining is built-in, or is that all custom dialplan stuff? |
| 15:34.05 | mocker | Googled around and can't find much on hit. |
| 15:34.10 | Qwell | mocker: show application meetme |
| 15:35.31 | mocker | Qwell: Maybe I'm just blind. |
| 15:35.40 | mocker | I don't see the 'Ask for participant name' option. |
| 15:35.52 | mocker | announce user join/leave ? |
| 15:35.59 | mocker | Or is that just a beep when they come into the room? |
| 15:36.14 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:37.56 | slowshutt | thx Tk works like a charm |
| 15:38.35 | [TK]D-Fender | slowshutt: np |
| 15:38.41 | *** join/#asterisk aiurea (n=aiurea@83.166.220.142) |
| 15:38.42 | mocker | Qwell: Looks like that's the option I needed. |
| 15:38.47 | [TK]D-Fender | mocker: look at the PARAMETERS |
| 15:38.50 | slowshutt | here i ga again, can one use the flash to transfer calls like on legasy pbx's? |
| 15:38.57 | mocker | For some reason I always thought that was just beeping. :) |
| 15:39.05 | [TK]D-Fender | slowshutt: "flash"? On what exactly? |
| 15:39.42 | slowshutt | you know on normal pbx you use flash to transfer calls, can one do this with asterisk? |
| 15:40.10 | [TK]D-Fender | slowshutt: use WHAT device exaclty, connected to * HOW? |
| 15:40.24 | slowshutt | i see infeatures.conf you have an option to set it default # |
| 15:40.29 | slowshutt | all sip phones |
| 15:40.34 | [TK]D-Fender | slowshutt: this is the point where you permanently realize that everything depends. |
| 15:40.45 | slowshutt | have multitech fxo/fxs 8 port gateway |
| 15:40.47 | [TK]D-Fender | slowshutt: depends... on EXACTLY what hardware you are using. |
| 15:41.12 | mocker | [TK]D-Fender: Hmm, I must be missing where you're seeing this. |
| 15:41.18 | [TK]D-Fender | slowshutt: Go read your Multitech's manual. I would suspect this to be a yes, but go verify |
| 15:41.20 | mocker | Still from the console? |
| 15:41.26 | aiurea | what's the best way to make asterisk(1.2) route all h323 connections via SIP to another box? |
| 15:41.31 | [TK]D-Fender | mocker: "show application meetme" |
| 15:41.43 | mocker | [TK]D-Fender: Right. :) |
| 15:42.01 | slowshutt | k Tk |
| 15:42.04 | mocker | I just misunderstood announce user join/leave |
| 15:42.14 | mocker | I thought that was just a beep on join/part for some reason. |
| 15:42.27 | slowshutt | is dialing multiple sip using the dial the same as group dial? |
| 15:42.53 | [TK]D-Fender | slowshutt: "group dial" is not a valid and unique term. |
| 15:43.40 | slowshutt | in your sip.conf you can add it to the the sip clients info group=1 ect |
| 15:43.55 | *** join/#asterisk [r]evolution (n=spmcatch@208.6.94.10) |
| 15:43.55 | phsdshft | fender: just sent it to www.pastebin.com |
| 15:44.05 | phsdshft | fender: thank you again for assisting |
| 15:44.10 | [TK]D-Fender | phsdshft: LINKS to them please and identify one from the other |
| 15:44.24 | JT | phsdshft: and we will use telepathy to work out where in pastebin.com you sent it? |
| 15:44.24 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:44.24 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:44.31 | [r]evolution | Hey Andrew -- you ever get a chance to check that issue we spoke upon last night? |
| 15:44.51 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 15:45.02 | [r]evolution | i.e. if the polycom phones sent the name of G729 as G729a (like linksys) and thus = rejected |
| 15:45.46 | [TK]D-Fender | [r]evolution: Nope. |
| 15:45.59 | [TK]D-Fender | [r]evolution: no chance this week. |
| 15:46.10 | [r]evolution | you suck lots :( |
| 15:46.11 | phsdshft | ... It would help if I pasted the URL, wouldn't it? :) http://pastebin.com/d398e1972 |
| 15:46.27 | [r]evolution | doesn't matter anyways... I made a patch to rtp.c and it works fine now :) |
| 15:46.47 | [r]evolution | though I confess -- I know shit of C... and I've never created a patch before... but everything is up and groovy |
| 15:47.03 | phsdshft | fender: I put it into one big pastebin, they are identified in the pastebin.. would you like me to post separate files? |
| 15:47.07 | phsdshft | err pastebins |
| 15:47.14 | JT | phsdshft: WHAT IS THE URL? |
| 15:47.25 | mocker | WHAT IS YOUR FAVORITE COLOR? |
| 15:47.40 | JT | ah nm |
| 15:47.43 | JT | you pasted it ;) |
| 15:47.58 | mocker | awww, no monty python fans. :) |
| 15:49.02 | cappiz | [TK]D-Fender, so that ATA acts as a SIP-extension towards the PBX? |
| 15:50.54 | phsdshft | JT: Sorry about not including it before.. I hit paste but it wasn't in my clipboard lol.. You have it now though right? http://pastebin.com/d398e1972 |
| 15:51.21 | JT | yep |
| 15:51.23 | [TK]D-Fender | cappiz: for the FXO and the FXS seperately, yes. And please don't use the term "extensions" for the term "device" |
| 15:51.35 | cappiz | hehe, np =) |
| 15:51.38 | [TK]D-Fender | psh reading |
| 15:51.53 | [TK]D-Fender | mocker: I don't know! |
| 15:52.01 | [TK]D-Fender | aaaaaarrrrrrrghhhhhhhhhhh |
| 15:52.06 | mocker | woo! |
| 15:52.37 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-239-242.dhcp.embarqhsd.net) |
| 15:52.48 | mocker | asterisk-users is killing my outlook. |
| 15:52.48 | blitzrage | BLUE! |
| 15:52.48 | [TK]D-Fender | phsdshft: did you set up your HOSTS file as they told you to? |
| 15:53.31 | phsdshft | yes |
| 15:53.48 | phsdshft | 147.135.32.221 sip.broadvoice.com |
| 15:54.16 | JT | that's retarded that you have to do that |
| 15:54.19 | JT | :) |
| 15:54.26 | phsdshft | yes.. pretty much |
| 15:54.54 | DJ_InstincT | any1 here doing VoiP on uk ADSL? |
| 15:55.16 | phsdshft | I'm also running Asterisk 1.2.24 |
| 15:55.25 | phsdshft | (if that matters) |
| 15:55.53 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:56.08 | [TK]D-Fender | phsdshft: http://pastebin.com/m67c42c4d <- replacement [general] section |
| 15:56.35 | [TK]D-Fender | phsdshft: the rest matches up pretty much. double check that you don't have typos |
| 16:02.47 | tzafrir | mocker, don't use outlock. And you'll help asterisk-users by not killing threads |
| 16:04.08 | slowshutt | how does the callgroup work regarding sip? |
| 16:04.14 | skirmisha | guys can i set in www_authorize to check in mysql view instead of table? |
| 16:04.48 | skirmisha | sorry wrong window |
| 16:06.30 | De_Mon | killing threads? |
| 16:07.24 | tzafrir | causing problems with theading |
| 16:08.06 | slowshutt | Tk if one used # to transfer how does one make the call come back to the sip phone that transfered the call? |
| 16:08.30 | *** join/#asterisk mog (i=mog@nat/digium/x-e54d558a9496a969) |
| 16:08.30 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:08.41 | slowshutt | if the phone where you transfer is busy? |
| 16:08.53 | phsdshft | fender: still not working.. is there anything else you see that could affect it? |
| 16:11.34 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:11.34 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:12.31 | [TK]D-Fender | phsdshft: Not that I can see offhand |
| 16:13.09 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 16:13.17 | [TK]D-Fender | slowshutt: time to read the book on *'s transfers |
| 16:13.28 | [TK]D-Fender | slowshutt: I never use them. |
| 16:13.51 | slowshutt | where is the book? |
| 16:13.59 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-ae0b9d128161d638) |
| 16:14.25 | [TK]D-Fender | ~book |
| 16:14.26 | jbot | book is, like, Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:15.01 | slowshutt | thx |
| 16:15.23 | [TK]D-Fender | blitzrage: your link = fail |
| 16:16.08 | blitzrage | [TK]D-Fender: oh? the SRV record stuff? |
| 16:16.12 | blitzrage | oh! |
| 16:16.15 | blitzrage | tfot.leifmadsen.com |
| 16:16.34 | blitzrage | hrmmmm... I'll look into it. I just changed IPs |
| 16:17.03 | [TK]D-Fender | blitzrage: And I'll have to retrain jbot for formatting :) |
| 16:17.20 | mocker | [TK]D-Fender: Any way to export jbot to html? |
| 16:17.29 | mocker | jbot2faq? |
| 16:18.11 | [TK]D-Fender | mocker: not that I know of and I e-mailed Tim about that.... no answer |
| 16:18.48 | [TK]D-Fender | mocker: I have done or redone the majority of the common stuff used and keep off-site copies handy in case of corruption/loss |
| 16:23.49 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 16:27.13 | *** join/#asterisk mltlnx (n=mltlnx@mbf5f36d0.tmodns.net) |
| 16:27.35 | *** join/#asterisk catpants (n=pREIXK@12-202-220-194.client.mchsi.com) |
| 16:27.47 | *** join/#asterisk ManxPower (n=manxpowe@64.sub-75-202-176.myvzw.com) |
| 16:28.21 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:32.25 | mocker | jbot: [TK]D-Fender++ |
| 16:33.34 | *** part/#asterisk harpal (n=harpal@124.125.255.24) |
| 16:35.18 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 16:36.41 | mocker | [TK]D-Fender: Or maybe just having Tim(?) do a nightly copy of the DB to someplace web accessable. |
| 16:36.47 | mocker | Then we can all mirror. |
| 16:36.58 | mocker | sorta like voip-info, voluntary mirrors. |
| 16:37.59 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 16:37.59 | *** mode/#asterisk [+o twisted] by ChanServ |
| 16:40.02 | *** join/#asterisk Op3r (n=Op3r@222.127.44.217) |
| 16:41.35 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177584269.dsl.bell.ca) |
| 16:42.06 | [TK]D-Fender | mocker: except of course, you can't pmirror voip-info |
| 16:42.53 | mocker | [TK]D-Fender: No, I think the admin got kinda lazy. |
| 16:42.59 | mocker | He just gives rsync instructions.. |
| 16:43.05 | mocker | :) |
| 16:43.09 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:43.25 | slowshutt | does anyone know if you can send a notify to eyebeam if someone has left you a message? |
| 16:43.57 | slowshutt | cant seem to get any info from google |
| 16:44.15 | [TK]D-Fender | slowshutt: what kind of nitofy? |
| 16:44.35 | [TK]D-Fender | notify* |
| 16:44.44 | ManxPower | slowshutt: Oh, sure Asterisk can send it. No idea if that softphone will accept the notify, however. |
| 16:44.45 | mocker | [TK]D-Fender: I guess I didn't know we were all being logged too: http://ibot.rikers.org/%23asterisk/20071205.html.gz |
| 16:45.35 | slowshutt | how would one go about sending a notify if i know how then i can see if eyebeam can accept it |
| 16:46.09 | ManxPower | slowshutt: mailbox=vmbox@vmcontext in the sip.conf entry for the softphone |
| 16:46.18 | ManxPower | then leave a message in vmbox |
| 16:46.29 | slowshutt | k thx |
| 16:48.34 | slowshutt | you lost me ManxPower |
| 16:48.41 | slowshutt | the voicemail works |
| 16:48.42 | mocker | Wow, tons of things I didn't know #asterisk had. |
| 16:48.46 | mocker | http://ibot.rikers.org/stats/asterisk.html.gz |
| 16:49.52 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 16:50.56 | lirakis | funn |
| 16:51.19 | lirakis | *funny .. ive never seen _Sam-- looks like he's here when im not |
| 16:51.32 | mocker | I just saw him the first time yesterday. |
| 16:51.35 | lirakis | oh wait |
| 16:51.38 | lirakis | yeah me too |
| 16:51.44 | lirakis | i forgot.. he was talking about grandstreams |
| 16:53.09 | *** join/#asterisk techie (n=techie@adsl-76-214-29-238.dsl.lsan03.sbcglobal.net) |
| 16:53.14 | [TK]D-Fender | slowshutt: so you get the VMI , what EXACTLY are you looking to "notify" aside fromt hat, and how? |
| 16:53.17 | ManxPower | slowshutt: setting the mailbox= option will make asterisk send notifies to the device to tell it there is new messages in that voicemailbox. |
| 17:00.21 | *** join/#asterisk novinder (n=Novinder@CPE000f664f0f37-CM0014045a95ea.cpe.net.cable.rogers.com) |
| 17:01.58 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 17:03.51 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 17:06.19 | dlynes | !seen flauto |
| 17:07.33 | dlynes | ~seen flauto |
| 17:07.35 | jbot | flauto <n=zhao@71.194.141.225> was last seen on IRC in channel #asterisk, 32d 23h 30m 37s ago, saying: 'and email'. |
| 17:07.58 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 17:09.49 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:10.50 | lirakis | ~seen dlynes |
| 17:10.52 | jbot | dlynes is currently on #asterisk-dev (15h 40m) #debian (15h 40m) #asterisk (15h 40m) #asterisk-bugs (15h 40m). Has said a total of 32 messages. Is idling for 3m 19s, last said: '~seen flauto'. |
| 17:11.09 | lirakis | i thought i might get a cool loop.. where it was using your last seen message |
| 17:11.14 | lirakis | heh heh |
| 17:11.17 | lirakis | no go |
| 17:12.55 | [TK]D-Fender | ~seen jbot |
| 17:12.57 | jbot | jbot is currently on #asterisk-doc (9d 19h 36m 39s) ##t42 (9d 19h 36m 39s) #how (9d 19h 36m 39s) #ol (9d 19h 36m 39s) #flyspray (9d 19h 36m 39s) #asterisk-dev (9d 19h 36m 39s) #asterisk (9d 19h 36m 39s) #byumug (9d 19h 36m 39s) #orkut (9d 19h 36m 39s) #nslu2-linux (9d 19h 36m 39s) ... |
| 17:13.24 | teknoprep | in asterisk 1.4 |
| 17:13.25 | *** join/#asterisk Renacor (n=kvirc@daimler.farheap.net) |
| 17:13.30 | teknoprep | how do i do... sip show channels |
| 17:13.33 | teknoprep | or sip show peers |
| 17:13.47 | [TK]D-Fender | teknoprep: same |
| 17:13.51 | teknoprep | oh nvm |
| 17:13.55 | teknoprep | i just typed it in wrong |
| 17:13.56 | teknoprep | lol |
| 17:15.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:17.44 | *** join/#asterisk tdi (n=tdi@gvf90.internetdsl.tpnet.pl) |
| 17:17.50 | *** part/#asterisk tdi (n=tdi@gvf90.internetdsl.tpnet.pl) |
| 17:17.57 | Renacor | how do you decrease verbosity in the asterisk CLI ? |
| 17:18.59 | [r]evolution | core set verbose X |
| 17:19.24 | Renacor | cool thanks! |
| 17:19.33 | [r]evolution | rtfm :) |
| 17:19.42 | Renacor | heh yeah did right before you said that =P |
| 17:19.52 | [r]evolution | word. :) |
| 17:20.19 | [r]evolution | someone using SPA-941 and G729 w/ 1.4.15 needs to appear |
| 17:20.36 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 17:20.37 | Renacor | poof! |
| 17:20.45 | [r]evolution | not laughing :) |
| 17:20.54 | [r]evolution | reaaadddyyy not funny |
| 17:21.00 | [r]evolution | what is xai |
| 17:21.01 | Renacor | hehe sorry couldn't help it |
| 17:22.30 | xai | Do voxpath phones need some special key or option when then get their dhcp ? The vip-2400 doesn't seem to accept my dhcp offers. I see the request. |
| 17:23.37 | *** join/#asterisk coolfreecode (n=jimmy@190.41.82.1) |
| 17:24.01 | *** join/#asterisk SwK_ (n=SwK@user-24-214-55-149.knology.net) |
| 17:25.27 | coolfreecode | hey guys how to use variables as a datetime,exten with Monitor() |
| 17:25.39 | slowshutt | thx ManXPower you are a genuis |
| 17:25.47 | slowshutt | thx ManXPower you are a genius |
| 17:25.49 | ManxPower | slowshutt: I know. |
| 17:26.05 | slowshutt | message notification works like a charm |
| 17:26.07 | ManxPower | But you are welcome to send money via paypal to eric@fnords.org |
| 17:26.09 | ManxPower | 8-) |
| 17:26.16 | slowshutt | lol |
| 17:29.37 | blitzrage | [TK]D-Fender: link back up -- seemed that on my server the default route got changed |
| 17:30.51 | [TK]D-Fender | <PROTECTED> |
| 17:30.54 | [TK]D-Fender | ~book |
| 17:30.55 | jbot | book is, like, Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 17:30.58 | [TK]D-Fender | blitzrage: thx |
| 17:31.01 | *** join/#asterisk mvanbaak (i=michiel@vanbaak.xs4all.nl) |
| 17:32.34 | ManxPower | 14,000 new messages in this user's e-mailbox. Time for drastic measures. |
| 17:32.39 | [r]evolution | readdddyyyyy stop stroking Manx's already massive ego ;) |
| 17:33.04 | [r]evolution | O_o |
| 17:36.26 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 17:36.28 | Uatec | Hiya |
| 17:36.55 | *** join/#asterisk Laureano (n=Laureano@200.59.172.38) |
| 17:36.58 | [r]evolution | dammit |
| 17:37.04 | [r]evolution | FUCK |
| 17:37.07 | Laureano | ~seen eliel |
| 17:37.09 | jbot | eliel <n=eliel@151-202-114-200.fibertel.com.ar> was last seen on IRC in channel #asterisk-dev, 13h 12m 41s ago, saying: 'in ast_waitfor_nandfds_complex the last for(i=0;i<25;i++) could be for(i=0;i<res;i++)'. |
| 17:37.12 | *** join/#asterisk masus (n=ethemc@88.248.14.186) |
| 17:37.21 | Uatec | thankyou |
| 17:37.51 | Uatec | i know that when you want to transfer and pickup calls you dial *2 and *8... |
| 17:37.59 | Uatec | is there anyway i can change what asterisk does when i dial *2 |
| 17:38.05 | [r]evolution | features.conf |
| 17:38.22 | Uatec | that allows you to change what to dial to transfer a call, or pick it up |
| 17:38.27 | Uatec | i don't want to change what you dial |
| 17:38.31 | Uatec | i want to change what it does.. |
| 17:38.41 | ManxPower | Uatec: that is ALL controled in features.conf |
| 17:38.46 | Uatec | i want to change the way a call is transfered, by performing the transfers myself |
| 17:38.48 | Uatec | really? |
| 17:38.48 | [r]evolution | if your trans and pickup are based on *2 and *8 |
| 17:38.54 | ManxPower | in fact Asterisk has no default for these things for SIP. |
| 17:38.57 | [r]evolution | id' say you've already edited your features.conf |
| 17:39.06 | [r]evolution | default transfer = # |
| 17:39.07 | Uatec | [r]evolution, no i haven't |
| 17:39.08 | [r]evolution | not *2 |
| 17:39.14 | Uatec | *2 is attended transfer |
| 17:39.27 | ManxPower | Uatec: So change it. |
| 17:39.39 | Uatec | I didn't see anything in features.conf that would allow me to set what *2 does |
| 17:39.53 | [r]evolution | look at the bottom |
| 17:39.57 | Uatec | i mean, i could override *2 to do bxfer or pickup... but nothinng user defined |
| 17:39.57 | [r]evolution | somewhere under applicationmap |
| 17:40.00 | codejunky | Hello, if I call my asterisk from outside, I hear no dial tone. How can I fix this? |
| 17:40.05 | Uatec | ooooooh |
| 17:40.18 | [r]evolution | you'll see syntax etc etc |
| 17:40.27 | Yourname`` | inrainbows.com by radiohead, free download, if anyone cares. |
| 17:40.28 | [r]evolution | <FeatureName> => <DTMF_sequence>, etc |
| 17:40.31 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:40.45 | ManxPower | codejunky: you are not supposed to hear a dialtone when you call asterisk |
| 17:40.53 | Uatec | i'm assuming that ";testfeature => #9,callee,Playback,tt-monkeys" is outdated syntax |
| 17:41.10 | ManxPower | Uatec: No need to guess. look in UPGRADE.txt |
| 17:41.18 | Uatec | i would do something like "testfeature => #9,callee,Playback(tt-monkeys)"? |
| 17:41.32 | ManxPower | Uatec: what does features.conf.sample show? |
| 17:41.35 | [r]evolution | i think he's looking for an internal dial-tone when calling from an external line manxy. |
| 17:41.39 | Uatec | umm |
| 17:41.41 | [r]evolution | what does RTFM do? :) |
| 17:41.42 | codejunky | ManxPower: Yeah, sorry, I was not really precise. I have the following in my dialplan: exten => 3075866,1,Answer() exten => 3075866,2,Dial(SIP/jan) exten => 3075866,3,Hangup(), my siphone rings, no problem. But the caller hears no ringtone. |
| 17:41.46 | Uatec | i have no features.conf.sample |
| 17:41.59 | ManxPower | [r]evolution: maybe so, but he only gave some vague question |
| 17:42.01 | [r]evolution | you do Uatex... |
| 17:42.07 | [r]evolution | it comes in the tarball |
| 17:42.17 | Uatec | i'm using asterisk business edition |
| 17:42.20 | ManxPower | codejunky: don't do the Answer |
| 17:42.21 | Uatec | so i don't have the tarball |
| 17:42.22 | [r]evolution | hey code -- why are you answering? |
| 17:42.33 | codejunky | ManxPower: Ah, ok! |
| 17:42.36 | [r]evolution | so go download the tarball :) |
| 17:42.37 | ManxPower | codejunky: and make sure you have /etc/asterisk/indications.conf |
| 17:42.52 | ManxPower | Uatec: we can't help you if you don't have the source. |
| 17:42.59 | [r]evolution | http://www.asterisk.org |
| 17:43.17 | [r]evolution | Hey Manx -- got the G729/G729a issue sorted :) |
| 17:43.26 | [r]evolution | quick edit to rtp.c -- recompile -- works great. |
| 17:43.34 | ManxPower | [r]evolution: did you report the big? |
| 17:43.39 | [r]evolution | patch submitted to bug-tracked... but didnt have disclaimer on file :( |
| 17:43.50 | [r]evolution | http://bugs.digium.com/view.php?id=11483 |
| 17:44.17 | [r]evolution | i need test dummies other than my boxes here now :( |
| 17:44.18 | codejunky | ManxPower: Thanks, not it works |
| 17:44.31 | codejunky | Argh, now it works |
| 17:45.52 | ManxPower | codejunky: You should not Answer() unless you know you need to. Once a call is answered, Asterisk looks in /etc/asterisk/indications.conf for information on generating tones. Before the call is answered Asterisk just tells the device "generate ringing sounds" and the device figures out what has to be done. |
| 17:46.46 | codejunky | ManxPower: Ah, ok. Then I understood it wrong before. |
| 17:47.18 | [r]evolution | Answer would be more along the lines of streaming audio as soon as the call is connected |
| 17:47.23 | [r]evolution | like if you were calling into an IVR |
| 17:47.29 | ManxPower | codejunky: most things that need the line answered will answer it automatically, so you seldom need an actual Answer() in the dialplan |
| 17:47.33 | [r]evolution | and you didnt want the first couplea words to get chopped off :) |
| 17:47.37 | Uatec | ManxPower, what has source got to do with it? |
| 17:47.49 | ManxPower | Uatec: it has all the documentation you seem to be missing. |
| 17:47.53 | [r]evolution | b/c if you had the source you'd have the samples :) |
| 17:48.16 | ManxPower | Uatec: you are running a GUI aren't you?!?! |
| 17:48.31 | [r]evolution | psst eff that. |
| 17:48.31 | outtolunc | stop swearing G<> |
| 17:48.50 | Uatec | ManxPower, no i am not running a gui |
| 17:48.54 | [r]evolution | chop != swear unless your name is John Bobbit |
| 17:48.59 | ManxPower | outtolunc: if he's running a GUI, you'll be hearing a lot more swearing. |
| 17:49.05 | [r]evolution | ouch. |
| 17:49.06 | outtolunc | hehe |
| 17:49.07 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:49.13 | ManxPower | Uatec: /etc/asterisk/features.conf does not just magically appear |
| 17:49.32 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 17:49.35 | Uatec | no it doesn't. It's put there when you install asterisk |
| 17:49.41 | ManxPower | Uatec: NO IT DOES NOT. |
| 17:50.00 | ManxPower | It puts it there if you do a "make samples", that is true, but it is not put there when you do a "make install" |
| 17:50.31 | ManxPower | [r]evolution: ABE users should not really be here anyway. They should work directly with Digium. |
| 17:50.37 | Uatec | ManxPower, when you install asterisk business edition it does... |
| 17:50.49 | ManxPower | Uatec: Asterisk Business Edition is not Asterisk. |
| 17:51.02 | Uatec | oh, i'm sorry |
| 17:51.05 | ManxPower | If you are using Asterisk Business Edition you should contact Digium for support. |
| 17:51.12 | Uatec | i thought that having Asterisk in the name implied that it was asterisk |
| 17:51.18 | [r]evolution | He does have a point there Uatec... you basically paid digium for support of the open-source project |
| 17:51.19 | Uatec | you fucking donkey |
| 17:51.22 | Uatec | irc is fucking retarded |
| 17:51.25 | Uatec | i'm out of here |
| 17:51.26 | [r]evolution | you should probably take advantage of that |
| 17:51.27 | Uatec | dick weed |
| 17:51.29 | *** part/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 17:51.30 | [TK]D-Fender | Uatec: Or jsut copy it over from a compatible source tarball |
| 17:51.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:51.34 | [r]evolution | wow... |
| 17:51.41 | [r]evolution | talk about anger management |
| 17:51.49 | ManxPower | Uatec: Oh, sure it is Asterisk under the hood, but all the installer and docs, and all that stuff is different. Really very little we tell you will apply to ABE. |
| 17:51.54 | [r]evolution | Manx he left. |
| 17:52.09 | [r]evolution | * Uatec (n=uatecuk@adsl.ntsols.com) has left #asterisk |
| 17:52.15 | ManxPower | [r]evolution: No big loss. We could not really help him anyway. |
| 17:52.39 | [r]evolution | Eh. I'm sure we could've... given that one comes from the other... but none-the-less |
| 17:52.41 | ManxPower | We know Asterisk's defaults. We don't know ABE's defaults, and it's obvious they are different. |
| 17:52.47 | [r]evolution | if you're going to pay for ABE... |
| 17:52.55 | [r]evolution | which basically means paying for Digium's support |
| 17:53.01 | [r]evolution | why not take advantage of what you paid for? |
| 17:53.04 | ManxPower | [r]evolution: we could help people with AsteriskGUI too. It would just take ten times as long. |
| 17:53.12 | [r]evolution | lol. |
| 17:53.15 | [r]evolution | gui :( |
| 17:53.38 | [r]evolution | <ManxPower> If you are using Asterisk Business Edition you should contact Digium for support. |
| 17:53.38 | [r]evolution | <Uatec> i thought that having Asterisk in the name implied that it was asterisk |
| 17:53.38 | [r]evolution | <[r]evolution> He does have a point there Uatec... you basically paid digium for support of the open-source project |
| 17:53.38 | [r]evolution | <Uatec> you fucking donkey |
| 17:53.38 | [r]evolution | <Uatec> irc is fucking retarded |
| 17:53.39 | [r]evolution | <Uatec> i'm out of here |
| 17:53.41 | [r]evolution | <[r]evolution> you should probably take advantage of that |
| 17:53.43 | [r]evolution | <Uatec> dick weed |
| 17:53.45 | [r]evolution | i love that. |
| 17:53.55 | [r]evolution | im so keeping that because it cracks me up... sudden explosion of anger |
| 17:53.57 | [TK]D-Fender | [r]evolution: Stop spamming, we all saw it |
| 17:54.01 | ManxPower | We KNOW Asterisk does not have default files in /etc/asterisk unless you do a "make install". We tell him to refer to UPGRADE.txt but he either does not have UPGRADE.txt or he does not know where ABE installed that file. |
| 17:54.05 | [r]evolution | shut your face Andrew :p |
| 17:54.17 | [TK]D-Fender | [r]evolution: watch it...... |
| 17:54.18 | ManxPower | So really most of the information we gave him did not apply to ABE. |
| 17:54.42 | [TK]D-Fender | ManxPower: Sure it does. |
| 17:54.49 | [TK]D-Fender | ManxPower: features.conf is the same everywhere |
| 17:54.52 | [r]evolution | not quite sure why he objected so fiercely to just going and DLing the tarball |
| 17:54.55 | [r]evolution | and reading the features.conf.sample |
| 17:55.05 | [r]evolution | i mean really... its 5 seconds out of your day. |
| 17:55.05 | [TK]D-Fender | ManxPower: if he stayed half a second to read my advise he'd have been fine |
| 17:55.07 | ManxPower | [TK]D-Fender: I can guarntee you my features.conf is different from yours. |
| 17:55.16 | [TK]D-Fender | ManxPower: I'm talking stock here. |
| 17:55.18 | [r]evolution | i think he means the base |
| 17:55.31 | [TK]D-Fender | ManxPower: You shouldn't give him samples based on your custom stuff anyways :) |
| 17:55.49 | ManxPower | [TK]D-Fender: Do you know for a fact that the /etc/asterisk/features.conf that is insalled by ABE the same as /path/to/src/asterisk/configs/features.conf ? |
| 17:56.10 | ManxPower | The problem is that he does not even have the default file for us to refer to. |
| 17:56.24 | [TK]D-Fender | ManxPower: ABE still has normal configs you know... it just has some (C) modules |
| 17:56.45 | file | it is the same. |
| 17:56.50 | ManxPower | file would be the one that could find out. |
| 17:57.10 | [r]evolution | all hail. |
| 17:57.22 | ManxPower | file: where does ABE install UPGRADE.txt ? |
| 17:57.34 | file | ManxPower: that I do not know |
| 17:57.59 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:58.15 | [TK]D-Fender | ManxPower: All he needs to do is grab a source tarball dated around the same |
| 17:58.37 | [r]evolution | file: what about the sample confs? still come with ABE? i.e. is it more likely that Uatec the angstful just needed without bothering to look around for himself |
| 17:58.57 | ManxPower | file: if ABE installs the same file as OS Asterisk, and Uatac did not modify features.conf then all his features.conf stuff would be disabled as the .sample file has them disabled. |
| 17:59.05 | file | I don't know, I don't do packaging of BE... I just have access to the tree |
| 17:59.22 | file | I can find out though |
| 17:59.25 | ManxPower | The other issue is that if Digium wants to charge for Asterisk then we should not be giving free support to the ABE users. |
| 17:59.31 | [r]evolution | also -- didnt think the default for AT trans was *2 |
| 17:59.46 | [r]evolution | though I havent used 'make samples' in many moons... so it could've changed |
| 17:59.49 | ManxPower | [r]evolution: the default is for all items in features.conf to be commented out.l |
| 18:00.00 | phsdshft | I cannot initiate outbound calls over a SIP connection to broadvoice through NAT (1 to 1 static NAT, all ports being allowed in/out), although I register correctly. I have verified my password, etc. with Broadvoice and the same config works correctly in an environment that is not NAT'd. Pastebin of templates from Broadvoice, what I'm using and debug output are at http://pastebin.com/d398e1972 |
| 18:00.05 | [r]evolution | well yeah I know that Manx... but I meant post commented out state |
| 18:00.27 | ManxPower | I don't have a real issue with Digium selling ABE, but we should not help them support it. |
| 18:00.41 | [r]evolution | i.e. default for B Trans = # default for Disconn = * etc. |
| 18:00.46 | [TK]D-Fender | ManxPower: thats a little mean.... sometimes the only reason people pick ABE int he first place is to shut up their bosses and placate their concrns of viability |
| 18:01.09 | ManxPower | [TK]D-Fender: Then maybe Digium should pay us to support it. |
| 18:01.23 | [r]evolution | eh... im not against helping the ABE users... but they ARE paying Digium for support... |
| 18:01.31 | [r]evolution | so it only makes sense for them to... ask Digium for help. |
| 18:01.46 | [r]evolution | why pay for something you don't intend to use? that = retarded idea. |
| 18:01.49 | [TK]D-Fender | ManxPower: maybe they should charge you for using it so long :) We'll all here by choice... or not ;) |
| 18:01.52 | ManxPower | [r]evolution: ABE does not come with source code, AFIK |
| 18:02.30 | ManxPower | phsdshft: you set localnet= and externip= in sip.conf [general] |
| 18:02.48 | [r]evolution | doesn't matter... they're still paying digium for support so they may as well use what they pay for. |
| 18:03.02 | [r]evolution | ABE can't be *that* different from * -- yes? |
| 18:03.29 | ManxPower | [r]evolution: oh I'm sure ABE is almost exactly the same as the GPL Asterisk. But "almost" is not the same as "exactly" |
| 18:03.43 | [r]evolution | his refusing to even consider the tarball of * is just a foolish mistake on his part. |
| 18:04.08 | [TK]D-Fender | phsdshft: I gave you a rebuilt [general] section. USE IT |
| 18:04.25 | [r]evolution | true... but I'm fairly certain you could assist an ABE user who wasn't being belligerent as easily as a GPL user |
| 18:04.57 | [TK]D-Fender | [r]evolution: ManxPower is an equal opportunity agitator ;) |
| 18:05.37 | ManxPower | [r]evolution: Uatec is a perfect example of why that is not always the case. Someone/something created a different features.conf from the one that we would refer to. |
| 18:05.41 | [TK]D-Fender | and according to Strom_M, I'm "irascible" :) |
| 18:05.53 | lirakis | if you have NAT=yes on a peer basis, do you need to set externip? |
| 18:06.04 | ManxPower | ergo he had never worked with features.conf before. |
| 18:06.20 | ManxPower | lirakis: nat==yes is for REMOTE clients behind NAT. |
| 18:06.31 | ManxPower | externip and localnet is for ASTERISK being behind NAT |
| 18:06.36 | [TK]D-Fender | lirakis: it shouldn't be in that peer. |
| 18:07.04 | lirakis | ManxPower: ahh .. right .. |
| 18:07.08 | [r]evolution | eh... I think Manx is just a habitual button-pusher |
| 18:07.13 | [r]evolution | thats ok -- I am too... ;) |
| 18:07.15 | [TK]D-Fender | ManxPower: I will say that he frankly DOESN'T read and only learns by asking for complete hand-holding.... |
| 18:07.38 | ManxPower | [TK]D-Fender: so even if he was using the tarball of asterisk we would not want him here. |
| 18:07.46 | [r]evolution | and i've not noticed you to be prone to sudden explosions of calling people 'fucking donkeys' |
| 18:08.20 | [TK]D-Fender | ManxPower: well you stone-walled him hard & fast and he's easy enough to set off. This was somewhat predictable |
| 18:08.26 | ManxPower | [TK]D-Fender: You know I expect dinner and drinks before any handholding. |
| 18:08.47 | *** join/#asterisk umay (n=chris@71-208-162-10.hlrn.qwest.net) |
| 18:09.03 | [r]evolution | you're an expensive date Manx. |
| 18:09.26 | [r]evolution | i'll settle with drinks... :) though that can quickly get expensive. |
| 18:09.48 | [r]evolution | eh... the answers were right there in front of him [TK] -- it was just too much like work to look for them |
| 18:09.56 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:10.03 | lirakis | lol .. just scrolled up and read uatec's retarded rant |
| 18:10.06 | [r]evolution | no sense in wasting time cuddling him to the answers. |
| 18:10.29 | [r]evolution | which part lira? the fucking donkeys or before that? |
| 18:10.47 | [r]evolution | i really seriously like that way too much... its the most random insult i've heard today |
| 18:11.09 | [r]evolution | kinda taking it as an abstract way to call someone a jackass |
| 18:11.13 | *** join/#asterisk hohum_ (n=dcorbe@h-74-1-66-114.lsanca54.covad.net) |
| 18:11.24 | lirakis | .. "you f*ing doney.. irc is retarded .. im out of here" .. lol .. thats seriously funny stuff |
| 18:11.32 | [TK]D-Fender | [r]evolution: that or he's ESL and running IRC through a translator :) |
| 18:11.45 | lirakis | <PROTECTED> |
| 18:12.07 | [r]evolution | nah... i think he was just a silly little cunt TK ;) |
| 18:12.25 | [r]evolution | i agree... at first it seemed like a semi-legit question... then you realize he just hadn't bothered to look around |
| 18:12.59 | [TK]D-Fender | [r]evolution: please refrain from name-calling... |
| 18:14.06 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177584269.dsl.bell.ca) |
| 18:14.17 | [r]evolution | you're just really opposed to 'vulgar' language arent you |
| 18:15.16 | *** join/#asterisk hohum (n=dcorbe@h-74-1-66-114.lsanca54.covad.net) |
| 18:15.47 | [r]evolution | annnnnnnd the channel falls silent. |
| 18:15.51 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-c3743b2264897e74) |
| 18:15.51 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 18:16.58 | [TK]D-Fender | [r]evolution: its keeping the bile flowing. just stop. He left. Let it end. |
| 18:17.18 | [r]evolution | O_o |
| 18:17.47 | [r]evolution | i forget everyone is as twisted as I am... I get a kick out of people throwing temper tantrums... to me its more about humor than negativity |
| 18:18.00 | bkruse | file: 2 more days :D |
| 18:18.10 | [r]evolution | though that tends to not go well with my girlfriend when she gets mad and I tell her she's silly. ;x |
| 18:18.40 | file | bkruse: yup |
| 18:19.34 | *** part/#asterisk masus (n=ethemc@88.248.14.186) |
| 18:20.47 | [r]evolution | maybe that just makes me a jerk :( |
| 18:22.24 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 18:22.51 | *** join/#asterisk shinao1 (n=shinao1@196.207.1.30) |
| 18:27.48 | De_Mon | oiy there are 1000ms in a second? |
| 18:28.11 | *** part/#asterisk shtoom (n=shtoom@59.93.116.155) |
| 18:28.16 | De_Mon | what kinda of second only has 100 of something? |
| 18:28.26 | De_Mon | microseconds? |
| 18:29.40 | [r]evolution | micro? nano? pico? |
| 18:31.08 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 18:36.24 | dlynes | I'm just curious...what is it that you need the D channel for? |
| 18:37.51 | [r]evolution | encouraging me to D-stroy existance because of random expenses for things :( |
| 18:38.39 | [TK]D-Fender | Centi-second |
| 18:39.05 | [TK]D-Fender | dlynes: to pass call progress indications on |
| 18:39.40 | dlynes | [TK]D-Fender: so, why don't you need a D channel when it's just a regular 24 channel T1, then? |
| 18:40.31 | [TK]D-Fender | dlynes: Ask yourself why you don't have DID's and use inband progress on them, and you'll know why |
| 18:41.51 | dlynes | [TK]D-Fender: inband progress is quite unreliable, I'm guessing? |
| 18:42.19 | [TK]D-Fender | dlynes: Feel free to sit around and hope to synch on a "busy" tone :p |
| 18:42.30 | *** join/#asterisk saftsack (n=saftsack@pD9E0480F.dip.t-dialin.net) |
| 18:42.49 | [TK]D-Fender | dlynes: its a poor or dumb schmuck how uses "analog" T1 |
| 18:42.50 | dlynes | [TK]D-Fender: oh...asterisk won't know if the channel is busy on a regular T1? |
| 18:42.51 | *** join/#asterisk RoyK (n=roy@ti211310a080-6805.bb.online.no) |
| 18:43.21 | [TK]D-Fender | dlynes: And no DID's typically (some wink-start via DTMF, but yeah, thats why we're using a T1 .. so we can use ANALOG signalling...) |
| 18:44.14 | *** join/#asterisk mardum_ (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 18:45.10 | dlynes | I was under the impression that the only disadvantage of a regular T1 over a PRI, is that a regular T1 couldn't do DIDs |
| 18:45.51 | ManxPower | For one thing most non-PRI stuff does not provide answer indication, so all calls are considered answered as soon as dialing is finished. |
| 18:46.12 | ManxPower | I should say most FXO signalled (T-1 or analog) |
| 18:46.25 | dlynes | ManxPower: ah...so inaccurate billing, then? |
| 18:46.38 | ManxPower | dlynes: correct |
| 18:46.48 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 18:46.55 | dlynes | ManxPower: but if you have answer and disconnect supervision on the t1 or analog line, that solves that issue, right? |
| 18:46.56 | ManxPower | also can't do parallel dial without crude hacks |
| 18:47.12 | dlynes | ManxPower: meaning two outgoing calls at the same time? |
| 18:47.17 | ManxPower | dlynes: it solves the answer when dialing is done at least. |
| 18:47.32 | ManxPower | dlynes: Dial(Zap/g1/5551515&SIP/otherperson) |
| 18:47.43 | ManxPower | the zap channel would be considered answered as soon as dialing is done. |
| 18:47.55 | dlynes | ManxPower: unless you've got answer supervision, right? |
| 18:48.23 | [TK]D-Fender | dlynes: Oh you mean "disconnectmycallsatrandom=yes"? :) |
| 18:48.26 | ManxPower | dlynes: I just said answer supervision is not available for most types of PSTN signalling |
| 18:48.35 | dlynes | ManxPower: ah |
| 18:48.37 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:48.54 | dlynes | [TK]D-Fender: no, i mean ordering the 'answer supervision' feature on the line...not that buggy option in zaptel |
| 18:49.13 | [TK]D-Fender | dlynes: Not familir with * supporting anything else |
| 18:49.26 | [TK]D-Fender | dlynes: and sounds very telco specific |
| 18:49.31 | dlynes | [TK]D-Fender: it's a telco option, not a zaptel option |
| 18:49.36 | [TK]D-Fender | dlynes: jsut like a variety of "wink" analog options. |
| 18:49.52 | [TK]D-Fender | dlynes: and if zaptel doesn't support it, you're doa |
| 18:49.54 | lesouvage | oej: are you there? (erik) |
| 18:51.56 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 18:52.33 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 18:52.36 | coolfreecode | hello i put exten = s,1,Monitor(wav,${DATETIME}-${EXTEN},mb) exten = s,2,Dial(zap/g1/3872594) but in the directory /var/spool/asterisk/monitor/ only recored "-.wav" help plz |
| 18:54.01 | ZX81 | coolfreecode: look in /usr/src/asterisk/doc/channelvariables.txt |
| 18:54.10 | ZX81 | datetime is deprecated |
| 18:54.12 | [TK]D-Fender | coolfreecode:${EXTEN] ="s", and I believe ${DATETIME} has been replaced. Go read channelvariables.txtx like I told you to |
| 18:54.53 | lesouvage | I'm searching for info to inergrate asterisk and Edirectory of novell. Any pointer is more then welcome. |
| 18:55.46 | ZX81 | edirectory == ldap |
| 18:56.06 | ZX81 | theres a few things you can do but better to search for asterisk ldap on google |
| 18:56.25 | lesouvage | zx81: so I should google on "asterisk ldap"? |
| 18:56.31 | ZX81 | there are some patches that have been worked on forever - don't know if they've made it in yet |
| 18:56.35 | ZX81 | yeah |
| 18:56.44 | coolfreecode | thx |
| 18:57.39 | [TK]D-Fender | lesouvage: Integrate how? For what purpose? |
| 18:59.08 | ZX81 | lesouvage: there is http://insects.digium.com/view.php?id=5768 (not yet accepted into mainline code) |
| 18:59.20 | Qwell | insects still exists? |
| 18:59.24 | Qwell | russellb: ^^? O.o |
| 18:59.33 | russellb | heh |
| 18:59.38 | russellb | goes to the same place as bugs ... |
| 18:59.40 | Qwell | wasn't that like a test install? |
| 18:59.41 | Qwell | oh |
| 18:59.49 | russellb | now it does, anyway |
| 18:59.53 | Qwell | I see |
| 19:00.42 | [TK]D-Fender | "And henceforth all fatal bugs shall be referred to as 'Special Features'" |
| 19:01.13 | ZX81 | :) I liked insects more than bugs - sounds less "crashy" |
| 19:03.01 | ZX81 | hey btw how come the i386 binary crashes my i686 machine (i thought it would just not have optimisations)? |
| 19:03.07 | ZX81 | *hpec |
| 19:04.43 | dijungal | hello |
| 19:05.05 | dijungal | should i do this before or after the dial command line: "Set(CDR(userfield)=Inbound)" |
| 19:05.27 | ZX81 | either |
| 19:05.36 | ZX81 | either before or in the h extension |
| 19:06.47 | [TK]D-Fender | dijungal: Before |
| 19:07.00 | dijungal | k |
| 19:07.22 | dijungal | reason i asked, if beause i have it before and i'm not seeing the "Inbound" in my CDR |
| 19:07.26 | dijungal | *because |
| 19:07.54 | ZX81 | dijungal: using mysql or csv or what? |
| 19:07.56 | ManxPower | dijungal: are you using Asterisk's Realtime stuff? |
| 19:08.09 | dijungal | mysql |
| 19:08.12 | dijungal | real time stuff??? |
| 19:08.17 | ManxPower | I didn't think the database driver supported User Fields |
| 19:08.29 | ZX81 | make sure you have userfield=1 in cdr_mysql.conf |
| 19:09.06 | dijungal | :| |
| 19:09.51 | dijungal | ZX81: done... lets see if it works |
| 19:09.57 | ZX81 | sweet |
| 19:11.21 | dijungal | ZX81: did not work |
| 19:11.28 | dijungal | table field still empty |
| 19:11.49 | ZX81 | may need to do: |
| 19:11.50 | dijungal | Set(CDR(userfield)="Inbound") |
| 19:12.02 | ZX81 | reload |
| 19:12.04 | ZX81 | or |
| 19:12.14 | dijungal | did a reload |
| 19:12.31 | dijungal | i hope not restart.. :s |
| 19:13.37 | ZX81 | you could do: |
| 19:13.41 | ZX81 | module unload cdr_addon_mysql.so |
| 19:13.44 | ZX81 | module load cdr_addon_mysql.so |
| 19:16.32 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.144.19) |
| 19:18.11 | *** join/#asterisk grEvenX (n=even@1mldj74.ip.ssc.net) |
| 19:18.27 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) |
| 19:18.27 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:19.11 | hackeron | hey, is there anyway to have asterisk answer a call and instead of playing say mp3 music or on hold music - open an input channel like a microphone? |
| 19:19.21 | ZX81 | yeah |
| 19:19.23 | ZX81 | chan_oss |
| 19:19.41 | ZX81 | think rizzo's been doing a bit of work on it |
| 19:19.50 | [TK]D-Fender | hackeron: For how long? Under waht circumstances? |
| 19:20.12 | ZX81 | or stream a connection from vlc |
| 19:20.29 | hackeron | [TK]D-Fender: I have motion installed that alerts me about intruders in my home, it would be cool if I could phone my asterisk box and listen in |
| 19:20.44 | ZX81 | yeah chan_oss |
| 19:20.58 | ZX81 | or xlite on a windows pc with auto answer turned on |
| 19:21.01 | hackeron | ZX81: sweet, thanks :) -- is there a chan_alsa or chan_pulse? |
| 19:21.11 | ZX81 | alsa i fink |
| 19:21.14 | [TK]D-Fender | hackeron: Yes, SF + AA |
| 19:21.32 | [TK]D-Fender | hackeron: That sould like the easiest way |
| 19:21.57 | hackeron | [TK]D-Fender: Soldier Fortune + America's Army? |
| 19:22.02 | ZX81 | :D |
| 19:23.45 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:23.49 | lesouvage | [TK]D-Fender: a kick out crieria in a tender. I suppose they want to use the edirectory info as a kind of phonebook. |
| 19:24.06 | hackeron | [TK]D-Fender: could you kindly tell me what SF + AA is? :) |
| 19:24.17 | [TK]D-Fender | hackeron: ZX81's idea |
| 19:24.27 | hackeron | [TK]D-Fender: ah, I see, brilliant |
| 19:24.58 | Qwell | SF? |
| 19:25.04 | Qwell | SPH maybe? |
| 19:25.06 | [TK]D-Fender | lesouvage: SoftPhone <-------- |
| 19:25.14 | [TK]D-Fender | Qwell: rather.. |
| 19:25.21 | Qwell | give up now :p |
| 19:25.47 | tzafrir | hackeron, I think someone is/was working on chan_console that also include pulse_audio support. But I might be confusing things |
| 19:26.10 | [r]evolution | i dream of a day where i dont lose everything |
| 19:27.01 | lesouvage | [r]evolution: loosing everything everyday garantees a fresh start of every new day |
| 19:27.36 | [r]evolution | well fresh starts are all well and good... |
| 19:27.44 | [r]evolution | but when I need something from the day before... |
| 19:27.50 | [r]evolution | maybe not so good :( |
| 19:27.50 | lesouvage | [TK]D-Fender: what o you man by "Softphone" |
| 19:27.52 | *** join/#asterisk grEvenX (n=even@1mldj74.ip.ssc.net) |
| 19:28.19 | lirakis | hackeron: AA was sick .. before they discontinued the linux client |
| 19:28.21 | [TK]D-Fender | ~softphone |
| 19:28.22 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam |
| 19:28.32 | hackeron | lirakis: lol, tell me about it |
| 19:28.39 | *** join/#asterisk bmcghee (n=brentmcg@d66-183-250-149.bchsia.telus.net) |
| 19:29.17 | *** join/#asterisk grEvenX (n=even@1mldj74.ip.ssc.net) |
| 19:29.18 | lesouvage | [r]evolution: like the Dutch former soccer player Cruijf is saying "every advantage has its disadvantage" (sound much better in Dutch) |
| 19:30.25 | lesouvage | [TK]D-Fender: I know wat oftphone is but what is the link between softphone and ldap integration of asterisk? |
| 19:30.37 | lirakis | hackeron: can you believe that one guy (iccalus) ported that whole damn game to linux & osx? |
| 19:31.02 | [TK]D-Fender | lesouvage: and right below you could see I corrected who my comment was intended for. Not you. |
| 19:31.08 | hackeron | iulius: really? - wow |
| 19:31.15 | lirakis | *icculus |
| 19:32.27 | lirakis | hackeron: http://icculus.org/ |
| 19:32.46 | iulius | what now? |
| 19:34.00 | lirakis | iulius: not you .. icculus ;P |
| 19:34.09 | lesouvage | [TK]D-Fender: Sorry, english is not my native language, thought it was a comment on another question. Don't get to touchy (although you have a reputation to keep up) |
| 19:34.52 | [TK]D-Fender | lesouvage: Wasn't being bitchy, sorry if I came across taht way |
| 19:35.38 | De_Mon | [TK]D-Fender he just called you out, you have a reputation to uphold now! |
| 19:36.21 | lesouvage | [TK]D-Fender: Don't worry, its great the way you answer all this questions day after day. I guess you gain the right to be picky once in a while. |
| 19:36.49 | ZX81 | :) |
| 19:37.20 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-3753cf7fd876a977) |
| 19:37.20 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 19:37.22 | ZX81 | mmmmmm cofffeeeeeeee |
| 19:38.20 | ZX81 | lol if he was pissed I'm sure your local phones would go down :D |
| 19:39.03 | De_Mon | lesouvage if he still talks to you, you haven't pissed him off yet, don't worry I'm sure you will some day |
| 19:39.21 | ZX81 | ~seen jbot |
| 19:39.24 | jbot | jbot is currently on #asterisk-doc (9d 22h 3m 6s) ##t42 (9d 22h 3m 6s) #how (9d 22h 3m 6s) #ol (9d 22h 3m 6s) #flyspray (9d 22h 3m 6s) #asterisk-dev (9d 22h 3m 6s) #asterisk (9d 22h 3m 6s) #byumug (9d 22h 3m 6s) #orkut (9d 22h 3m 6s) #nslu2-linux (9d 22h 3m 6s) ##ducleague (9d 22h ... |
| 19:40.15 | De_Mon | grawr I hate vista~ |
| 19:40.39 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:40.42 | khronos | Anybody have any trouble getting Aastra 9133i phones to register to Asterisk? |
| 19:40.42 | ZX81 | De_Mon: lol install 3.11 again |
| 19:40.53 | ZX81 | khronos: check sip debug |
| 19:41.06 | De_Mon | ZX81 I was looking at TinyXP yesterday I might give that a try |
| 19:41.20 | Deeewayne | khronos: mine works. |
| 19:41.35 | lesouvage | De_Mon: don't worry, I have my share. |
| 19:41.49 | ZX81 | LOL!!! TinyXP installation only uses 390Mb of hard-disk space |
| 19:41.52 | ZX81 | heh |
| 19:42.03 | ZX81 | my Asterisk installation uses like 32mb |
| 19:42.04 | ZX81 | :) |
| 19:42.13 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 19:42.43 | ZX81 | I'd need 39 x 10Mb hard drives for tinyxp! |
| 19:44.01 | De_Mon | considering it plays games (why else whould you use windows!) its a far better deal than the full XP that takes up 2gb |
| 19:44.08 | ZX81 | agreed |
| 19:44.20 | De_Mon | or the 10 gigs that is the pice of crap known as vista ultimate |
| 19:44.21 | ZX81 | cept directx is prolly 100gb |
| 19:44.47 | *** join/#asterisk rajkosto (i=lolwut@cable-87-116-180-142.dynamic.sbb.co.yu) |
| 19:44.50 | rajkosto | hello |
| 19:46.09 | ZX81 | hi |
| 19:46.36 | Qwell | De_Mon: ultimate, hell. home basic takes up 10g |
| 19:47.44 | rajkosto | any way i can set up a sip server just so i can access skype from my sip phone ? |
| 19:47.57 | ZX81 | anyone use linux on their laptop for office work? does it slow down after 6 months like the registry stuff on windows? |
| 19:48.13 | ZX81 | rajkosto: chan_skype |
| 19:48.23 | rajkosto | but that is pay |
| 19:48.26 | ZX81 | yep |
| 19:48.41 | [TK]D-Fender | ~skype |
| 19:48.42 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 19:48.48 | ZX81 | skype == yuck |
| 19:48.49 | khronos | In my syslog I get from the phone after it is booted. |
| 19:48.50 | khronos | Dec 6 14:47:58 10.200.34.12 00/1/1 00:00:24.38 3 endpoint.c:1566 processCmdQ: EPTCMD_SETDEVmac:00-08-5D-18-9D-D3^M |
| 19:48.53 | khronos | <PROTECTED> |
| 19:48.55 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:49.11 | khronos | This comes up in the log about every 5 or 10 secs. |
| 19:49.24 | ZX81 | google it |
| 19:49.36 | dijungal | lol! |
| 19:51.10 | ZX81 | man why is there an on sound and no off sound |
| 19:52.17 | De_Mon | ~chanskype |
| 19:53.17 | blitzrage | chanskype... gross |
| 19:54.10 | De_Mon | yeah I just looked at their webpage... |
| 19:54.55 | *** join/#asterisk Op3r (n=edwin@203.177.230.56) |
| 19:55.33 | *** join/#asterisk klyrelion (n=Kevin@boyne.demon.co.uk) |
| 19:59.29 | fujin_ | chanskype? |
| 20:00.42 | *** part/#asterisk klyrelion (n=Kevin@boyne.demon.co.uk) |
| 20:02.58 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 20:03.39 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 20:05.29 | [r]evolution | don't trust canadians who have names resembling a 3yr olds spelling of Life ;x |
| 20:08.16 | blitzrage | harsh |
| 20:09.03 | [r]evolution | nah |
| 20:09.22 | [r]evolution | you know everyone loves the guys who bring all the esoteric * documentation into one easy to locate volume |
| 20:10.27 | bmcghee | sup all |
| 20:13.07 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 20:13.27 | De_Mon | a 3yr olds spelling of Life? |
| 20:13.40 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 20:14.38 | *** join/#asterisk aod3 (n=aod@demon.gbp.com) |
| 20:14.43 | aod3 | Hello |
| 20:15.06 | aod3 | Can anyone help me with an IAX jitterbuffer issue? |
| 20:15.28 | [r]evolution | what's blitzrage's first name de_mon? |
| 20:15.35 | De_Mon | blitz |
| 20:15.40 | putnopvut | heh |
| 20:15.57 | blitzrage | Dr. Blitz Rage |
| 20:16.34 | De_Mon | yup it's blitz |
| 20:17.04 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 20:18.02 | aod3 | Can someone tell me why I am seeing 0 lost packets on only one side of an IAX2 connection (other side shows lost packets correctly) when I do an iax2 show netstats? I can even do something like iax2 test losspct 20 and it still shows no packets lost on that end? I'm confused. |
| 20:18.40 | aod3 | Oh, and when I do the test there are definitely lost packets because I can hear it. |
| 20:18.50 | [r]evolution | ... |
| 20:18.53 | [r]evolution | :( |
| 20:19.21 | aod3 | I've searched around and can't seem to find anything about this issue. |
| 20:19.22 | [r]evolution | anyone want to come take my place tonight? im supposed to go to a christmas parade and i do not particularly wish to do so |
| 20:20.44 | aod3 | No ideas? :( |
| 20:22.10 | [r]evolution | apparently no one has any ideas about being more for the night |
| 20:22.13 | [r]evolution | come on its not that bad |
| 20:22.53 | *** join/#asterisk curtn (n=curtis@cl-451.trn-01.it.sixxs.net) |
| 20:22.56 | [r]evolution | usually its pretty fun... but dammit why does geek = 'guy who works on the computers of any family member at any time because he has nothing else to do and doesnt get tired of looking at them all day/night long' |
| 20:23.04 | aod3 | The bad thing about this problem is that the jitter buffer isn't working correctly on that side of the connection because of it. |
| 20:24.10 | curtn | I still have some echo problem on my SPA3102... anyone have a good url ? |
| 20:24.26 | [TK]D-Fender | curtn: www.voxilla.com <- check out their forums on this |
| 20:25.18 | curtn | [TK]D-Fender: gain and impedance on the FXO seems to be very difficult to tune... |
| 20:25.33 | [TK]D-Fender | curtn: They've got a lot of good guides. |
| 20:28.20 | curtn | is it possible to have something like a spectrum analyser connected to asterisk ? (or to the SPA3102 ?) |
| 20:28.31 | ZX81 | aod3: maybe jitter buffer is off? |
| 20:28.43 | ZX81 | on one side? |
| 20:28.49 | ZX81 | i.e. voip -> voip |
| 20:29.18 | ZX81 | curtn: record a file and then use wavelab or audacity or something |
| 20:29.19 | aod3 | ZX81: in iax.conf I have jitterbuffer=yes and trunktimestamps=yes on both ends |
| 20:29.33 | aod3 | It works on one end, but not the other. |
| 20:29.33 | ZX81 | just while testing force it on |
| 20:29.51 | ZX81 | forcejitterbuffer=no |
| 20:29.54 | ZX81 | except yes |
| 20:29.55 | ZX81 | :D |
| 20:30.03 | aod3 | Ahh, let me try that. Thanks. |
| 20:30.08 | ZX81 | np |
| 20:30.08 | [r]evolution | its never nice to force things around |
| 20:30.13 | [r]evolution | just because you can doesnt mean you should :( |
| 20:30.14 | ZX81 | heh indeed |
| 20:30.18 | ZX81 | :) |
| 20:31.12 | aod3 | Well, that did it |
| 20:31.16 | ZX81 | :) |
| 20:31.18 | aod3 | I don't quite understand why though |
| 20:31.31 | ZX81 | because you only do jitter buffer when the endpoint shouldn't |
| 20:31.42 | ZX81 | i.e. the jitter should be passed to the end voip connection |
| 20:31.46 | ZX81 | and it should dejitter |
| 20:32.01 | ZX81 | zap <--> iax should dejitter |
| 20:32.10 | ZX81 | because the telephone network can't dejitter |
| 20:32.19 | ZX81 | but iax <--> iax shouldn't |
| 20:32.20 | aod3 | Oh, sorry, forgot to mention |
| 20:32.26 | aod3 | this is an iax <-> iax trunk |
| 20:32.29 | ZX81 | yeah |
| 20:32.33 | aod3 | why shouldn't it? |
| 20:32.44 | ZX81 | because the thing at the other end should |
| 20:32.46 | aod3 | if I'm dropping packets on one end, i would want it to dejitter on that end, no? |
| 20:33.14 | ZX81 | so analogue phone (dejitter) --> IAX --> IAX --> phone line (dejitter) |
| 20:33.25 | ZX81 | in the middle dejittering doesn't happen |
| 20:33.36 | *** join/#asterisk oej_ (n=olle@193.126.30.214) |
| 20:33.53 | ZX81 | yay my morning meetings got cancelled! |
| 20:34.14 | aod3 | I guess I'm confused then. Right now I have a call from a SIP phone to hold music on another PBX from via an IAX trunk |
| 20:34.31 | aod3 | if I introduce packet loss on the end where the hold music is playing, the jitter buffer does its magic |
| 20:34.32 | ZX81 | SIP phone should dejitter |
| 20:34.37 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 20:34.42 | ZX81 | depending on type |
| 20:34.44 | [r]evolution | i rock rough and stuff with my afro-puffs. |
| 20:34.49 | aod3 | if i introduce it on the pbx that has the sip phone connected, it goes crazy |
| 20:34.55 | aod3 | it is a Cisco 7970 |
| 20:35.13 | ZX81 | yeah cos the sip phone should be getting rid of any jitter by the time it gets there |
| 20:35.29 | aod3 | unfortunately, it doesn't. :( |
| 20:35.36 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 20:35.42 | ZX81 | really? no setting for jb in the 7970? |
| 20:35.51 | aod3 | not that i know of |
| 20:35.52 | ZX81 | the linksys phones do I think |
| 20:35.59 | ZX81 | and grandstream |
| 20:36.12 | [TK]D-Fender | Polycom > ALL |
| 20:36.13 | [r]evolution | O_O |
| 20:36.14 | [r]evolution | ZX |
| 20:36.19 | [r]evolution | you have linksys phones? |
| 20:36.24 | ZX81 | spa942 |
| 20:36.27 | [r]evolution | :-D |
| 20:36.30 | ZX81 | :) |
| 20:36.31 | [r]evolution | you could be my new best friend |
| 20:36.33 | *** join/#asterisk fukz (n=basti@p5B0620C3.dip.t-dialin.net) |
| 20:36.37 | ZX81 | lol |
| 20:36.39 | [r]evolution | you have a testing server? |
| 20:36.45 | ZX81 | a server |
| 20:36.49 | ZX81 | :) |
| 20:36.51 | [r]evolution | is it production? |
| 20:37.15 | ZX81 | akl.venturevoip.com, chch.venturevoip.com, dndn.venturevoip.com, www.venturevoip.com for free accounts |
| 20:37.24 | ZX81 | all in New Zealand though :) |
| 20:37.33 | ZX81 | can register to any of them or all |
| 20:37.34 | [r]evolution | lol dont need accounts |
| 20:37.49 | [r]evolution | i need someone with another Linksys phone to try to make a G729 call on Asterisk 1.4.15 |
| 20:37.57 | aod3 | oh, and I had one other strange thing happen as well. I had someone on that PBX call my extension and the jitter buffer was working correctly on their end still, but still not working on my end. |
| 20:38.00 | aod3 | Is that normal as well? |
| 20:38.15 | [r]evolution | b/c it rejects the SPA941... the SPA2100 and the SPA2002 |
| 20:38.29 | [r]evolution | b/c they send the codec name as G729a (which is the proper name) |
| 20:38.33 | fujin_ | have you filed a bug [r]evolution? |
| 20:38.35 | ZX81 | [r]evolution you mean passthrough |
| 20:38.40 | [r]evolution | where Asterisk wants it to be G729 |
| 20:38.54 | [r]evolution | nono... you need a license free or otherwise |
| 20:38.56 | ZX81 | aod3: yeah because both ends should be dejittering in a voip->voip |
| 20:39.11 | [r]evolution | sooo... i need someone with other brands to test and see if its kicking them too |
| 20:39.14 | [r]evolution | or if its just linksys |
| 20:39.15 | ZX81 | [r]evolution just hack the source |
| 20:39.18 | [r]evolution | or if its just these brands |
| 20:39.20 | [r]evolution | i did. |
| 20:39.20 | fujin_ | heh, yeah |
| 20:39.27 | ZX81 | [r]evolution and? |
| 20:39.30 | fujin_ | should be able to fix it with codec_g729.c |
| 20:39.32 | [r]evolution | which leads to the next statement -- if it IS rejecting you too |
| 20:39.41 | [r]evolution | i want you to apply this patch and see if it fixes you |
| 20:39.44 | [r]evolution | no fujin. |
| 20:39.55 | ZX81 | I'm not in the office at the mo, but will be in like 4 hours :) |
| 20:40.01 | aod3 | ZX81: One thing I did notice, with jitter buffer forced the call quality suffered even more, so maybe the 7970 is doing jitter buffering afterall |
| 20:40.02 | [r]evolution | your breathing rights were revoked... go off somewhere and die slow. |
| 20:40.04 | [r]evolution | bah. |
| 20:40.09 | [r]evolution | :( |
| 20:40.14 | ZX81 | aod3: I would have thought so |
| 20:40.17 | [r]evolution | i wont be here in four hours but ill send you the patch regardless |
| 20:40.17 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:40.19 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:40.20 | fujin_ | submit your patch to the bugtracker |
| 20:40.22 | ZX81 | [r]evolution lol |
| 20:40.25 | fujin_ | stop faggoting around in here |
| 20:40.28 | fujin_ | you're doing it wrong |
| 20:40.35 | ZX81 | [r]evolution: matt@venturevoip.com |
| 20:40.36 | [r]evolution | Once Again... Captain Obvious Fujin is only a day behind. |
| 20:40.52 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 20:41.12 | fujin_ | well why are you pissing around with it in here trying to get people to fix? use the process already defined |
| 20:41.14 | aod3 | I guess I was just retarded as to how the jitter buffer is supposed to work. :) |
| 20:41.25 | [r]evolution | the process defined is to have more than one person test it |
| 20:41.33 | lesouvage | I'm rading a tender paper. They demand support for 1 gb full duplex ethernet on the sip phone. Does this make any sence? (I don't think so). |
| 20:41.35 | fujin_ | i.e.; 1) identify problem 2) submit to bugtracker 3) patch, or wait for someone else to test 4) wait for a dev to pick it up |
| 20:41.40 | fujin_ | so point him to the bugtracker page? |
| 20:41.44 | fujin_ | muppet |
| 20:41.45 | [r]evolution | i understand you must not realize that... but you don't just make something and say |
| 20:41.47 | [r]evolution | YAY IT WORKS |
| 20:42.06 | fujin_ | sure, I do, I test locally, leave it on a bugtracker and wait for the appropriate people to find it |
| 20:42.11 | fujin_ | instead of trying to ++ego in IRC |
| 20:42.31 | ZX81 | can you do that? |
| 20:42.33 | [r]evolution | you mean instead of trying to test on your own -- you'd rather let someone else do the leg-work? |
| 20:42.35 | ZX81 | ++dick size |
| 20:42.37 | ZX81 | meh |
| 20:42.39 | ZX81 | didn't work |
| 20:42.40 | ZX81 | :) |
| 20:42.43 | fujin_ | no, I said, I test it locally |
| 20:42.47 | ZX81 | shit now I've got two! |
| 20:42.53 | ZX81 | damn forgot the speech marks |
| 20:42.54 | fujin_ | ZX81: lol |
| 20:42.55 | ZX81 | :) |
| 20:42.56 | fujin_ | donotwant |
| 20:42.58 | [r]evolution | chya... " " |
| 20:43.01 | [r]evolution | lolcat |
| 20:43.11 | aod3 | you must have typoed *= instead of ++ :( |
| 20:43.20 | ZX81 | :) |
| 20:43.39 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 20:43.57 | aod3 | thanks for the help zx81. I think the 7970's jitter buffer isn't quite as good as the one in asterisk, but I believe there is definitely one hidden in there |
| 20:44.07 | ZX81 | yep |
| 20:44.15 | ZaVoid | anyone ever have a problem with RINGING sounding completely staticky? |
| 20:44.26 | ZX81 | not I |
| 20:44.27 | *** part/#asterisk mardum_ (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 20:45.13 | ZX81 | brb getting power |
| 20:48.32 | *** join/#asterisk saftsack (n=saftsack@p4FC74499.dip.t-dialin.net) |
| 20:52.27 | lesouvage | any coment on the 1 gb full duplex ethernet ort on a sip phone? |
| 20:52.46 | [TK]D-Fender | lesouvage: which? |
| 20:52.50 | [r]evolution | fly with a sledgehammer? |
| 20:53.22 | lesouvage | [TK]D-Fender: I'm rading a tender paper. They demand support for 1 gb full duplex ethernet on the sip phone. Does this make any sence? (I don't think so). |
| 20:53.47 | [r]evolution | fly with a sledgehammer |
| 20:53.49 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:53.53 | [TK]D-Fender | lesouvage: Yes. Only high-end Cisco's have that that I've seen |
| 20:53.54 | [r]evolution | thats like killing a fly with a sledgehammer. |
| 20:54.18 | [r]evolution | oh look... this needs 100K(ish) for maximum quality codec. |
| 20:54.24 | ZaVoid | fender you ever see that? |
| 20:54.26 | [r]evolution | lets put a gig-port in it. |
| 20:54.30 | lesouvage | [TK]D-Fender: But what is the point of 1 gb on a phone? |
| 20:54.34 | [r]evolution | exactly. |
| 20:54.36 | [TK]D-Fender | lesouvage: Frankly anybody needing good bandwidth to the pass-through like that should wire their phones seperately anyways and be using PoE. |
| 20:54.38 | [r]evolution | fly with a sledgehammer. |
| 20:54.42 | ZaVoid | call is coming g.723 to the asterisk(pass through) to the client device |
| 20:55.04 | [TK]D-Fender | ZaVoid: its either jitter, or the codec |
| 20:55.18 | ZaVoid | yeah its not jitter i ruled that out |
| 20:55.18 | *** join/#asterisk RoyK (n=roy@ip-78-1-149-91.dialup.ice.no) |
| 20:55.25 | [TK]D-Fender | lesouvage: the point is when used in PASS-THROUGHT with a PC |
| 20:55.52 | ZaVoid | and codec i don't see how.... maybe the far end is negotiating the g723 right.... i guess that could be it |
| 20:56.22 | ZX81 | lesouvage: with two ports and the pc on the other side if they need 1gb on the pc it will need 2 1gb ports on the phone |
| 20:56.27 | ZX81 | really in shower now |
| 20:56.34 | ZX81 | and its 9:56am here :) |
| 20:56.36 | [r]evolution | eh... for the avg. company I personally don't see the need for gig-network at this point in life :) |
| 20:56.44 | [r]evolution | damn. thought you guys were like 12 hours behind |
| 20:56.59 | fujin_ | ^5 ZX81 |
| 20:57.03 | fujin_ | nz represent |
| 20:57.45 | [r]evolution | really and truly... how many average companys you know rocking an OC24 connection? |
| 20:58.01 | fujin_ | uh |
| 20:58.01 | fujin_ | none |
| 20:58.05 | lesouvage | [TK]D-Fender: But what normal desk pc needs 1 gb bandwidth. For word, excel and powerpoint (or openoffice) you certainly don't. I know offices where they still use 10 mb etwork without problems. |
| 20:58.15 | *** join/#asterisk Seldon75 (n=chatzill@69.77.161.3) |
| 20:58.38 | [TK]D-Fender | lesouvage: are YOU defining your clients needs? |
| 20:58.40 | [r]evolution | but you know this is kinda a massive digression... |
| 20:58.53 | [r]evolution | les -- did you just come in to debate the logic behind gig-networks? |
| 20:58.58 | [r]evolution | or was there a point to all this? |
| 20:59.06 | [TK]D-Fender | lesouvage: and maybe they're trying to validate their Gig-E investment and don't want the phones slowing that down. |
| 20:59.36 | fujin_ | lesouvage: word/excel/powerpoint over DAV will certainly have performance increases in all gige network |
| 20:59.59 | [r]evolution | eh -- maybe TK... but then some people in life would get primarily gig-net just for the simple principle of saying they have it. |
| 21:00.04 | [r]evolution | would/will/do. |
| 21:00.22 | [r]evolution | even if their backbone is only a T1 |
| 21:00.46 | [TK]D-Fender | [r]evolution: Why do you assume internal LAN bandwidth has anything to do with INTERNET? |
| 21:01.01 | [r]evolution | i don't :) |
| 21:01.09 | lesouvage | [TK]D-Fender: no, I'm just checkking if I'm 1 years behind. If they want 1 gb sip phones I'm sure they can get it. Imho it is a wast of money to end up with high end cisco phones while snom or linksys of EUR 200,- will do. |
| 21:01.11 | [TK]D-Fender | [r]evolution: Tel that to my marketing guys working on 400 meg Adobe CS files! |
| 21:01.33 | [r]evolution | 400 meg adobe files... |
| 21:01.40 | [TK]D-Fender | lesouvage: Linksys is a LOT cheaper than that. |
| 21:01.49 | [r]evolution | how long does that take them to transfer on a 10/100 network to a local FTP? |
| 21:01.53 | [r]evolution | couple minutes? |
| 21:02.01 | [TK]D-Fender | lesouvage: and you could pay for completely new wiring cheaper than the cost of the phones alone |
| 21:02.12 | [TK]D-Fender | [r]evolution: FUGLY..... |
| 21:02.17 | [TK]D-Fender | [r]evolution: leave it at that... |
| 21:02.22 | lesouvage | [TK]D-Fender: there has to be a profit somehere ;-) |
| 21:02.56 | lesouvage | thnks for the input. |
| 21:03.43 | [r]evolution | eh. point being is that we transfer several gig files all the time over the 10/100 here... |
| 21:03.50 | [r]evolution | no complaints. |
| 21:04.03 | [r]evolution | but to each their own I suppose |
| 21:08.12 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-58-187.pskn.east.verizon.net) |
| 21:09.20 | ZaVoid | m=audio 18730 RTP/AVP 4 101 |
| 21:09.20 | ZaVoid | a=rtpmap:4 G723/8000 |
| 21:09.20 | ZaVoid | a=fmtp:4 annexa=no |
| 21:09.21 | ZaVoid | a=rtpmap:101 telephone-event/8000 |
| 21:09.21 | ZaVoid | a=fmtp:101 0-16 |
| 21:09.21 | ZaVoid | a=silenceSupp:off - - - - |
| 21:09.22 | *** join/#asterisk saftsack (n=saftsack@p4FC76884.dip.t-dialin.net) |
| 21:09.23 | ZaVoid | a=ptime:30 |
| 21:09.25 | ZaVoid | a=sendrecv |
| 21:09.29 | ZaVoid | that does't look right and sorry for flooding... |
| 21:15.30 | [r]evolution | what doesnt look right about it? |
| 21:19.47 | *** join/#asterisk hohum (n=dcorbe@h-74-1-66-114.lsanca54.covad.net) |
| 21:21.40 | fujin_ | ~pb |
| 21:21.41 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:21.45 | fujin_ | please, for next time |
| 21:25.23 | ZaVoid | the g723/8000 |
| 21:25.34 | ZaVoid | no bitrate in the a= lines either |
| 21:25.46 | Qwell | bitrate isn't required |
| 21:26.07 | [r]evolution | no pretty sure the rtpmap is normal |
| 21:26.20 | [r]evolution | did you pull that from the asterisk debug or from a wireshark cap? |
| 21:26.38 | [r]evolution | a=rtpmap:18 G729a/8000 |
| 21:26.38 | [r]evolution | see |
| 21:31.07 | ZaVoid | <PROTECTED> |
| 21:31.13 | ZaVoid | asterisk debug |
| 21:31.23 | *** join/#asterisk objective (n=objectiv@ool-43536c3b.dyn.optonline.net) |
| 21:31.24 | ZaVoid | http://bugs.digium.com/view.php?id=11062 |
| 21:31.29 | ZaVoid | sounds similar to this is what i'm seeing |
| 21:31.56 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:32.05 | ZaVoid | minus the h323 part |
| 21:32.07 | *** join/#asterisk Hadi- (n=AlanS@208.113.18.155) |
| 21:32.10 | Hadi- | Hello |
| 21:32.15 | ZaVoid | hi Hadi- |
| 21:32.19 | objective | who are you guys using for general usage int'l calls these days? |
| 21:32.35 | Hadi- | quick question.. is it possible to force asterisk to send the call g729 bytes 30? |
| 21:33.25 | Hadi- | I think by default it is bytes 20 |
| 21:33.25 | ZaVoid | not sure Hadi- i've never had to on asterisk |
| 21:33.25 | ZaVoid | on a cisco sure :) |
| 21:33.25 | [r]evolution | yeah... pretty sure you do that in the sip.conf |
| 21:33.25 | [r]evolution | could be wrong... but pretty sure |
| 21:33.34 | Hadi- | ZaVoid: my cisco is rejecting the call |
| 21:33.45 | Hadi- | because our carrier is only supporting |
| 21:33.45 | ZaVoid | which AS cisco? |
| 21:33.52 | ZaVoid | 5350 5400? |
| 21:33.56 | Hadi- | its not AS |
| 21:34.00 | Hadi- | its 2851 |
| 21:34.08 | Hadi- | we are using it IP-to-IP |
| 21:34.42 | ZaVoid | ah |
| 21:34.48 | Hadi- | our PRI is a SIP PRI |
| 21:34.57 | Hadi- | and its only supporting g729 bytes 30 |
| 21:35.05 | Hadi- | when in trying to send the call from astersk |
| 21:35.07 | Qwell | SIP PRI? |
| 21:35.09 | Hadi- | its rejecting |
| 21:35.22 | Hadi- | my provider is telling me that you are sending the call g729 byte 20 |
| 21:35.26 | Hadi- | and thats why its rejecting |
| 21:35.38 | bkruse | Qwell: its the newest kind of PRI, it rocks. I also have an h323 PRI and an iax pri! |
| 21:35.45 | Qwell | Hadi-: google for rtp packetization |
| 21:35.55 | Qwell | actually, there should be an example in the asterisk configs |
| 21:36.31 | [r]evolution | i swear it was somewhere in sip.conf qwell... but i can not remember. |
| 21:36.38 | Qwell | doc/rtp-packetization.txt |
| 21:36.47 | [r]evolution | HAH |
| 21:36.48 | [r]evolution | yes |
| 21:37.50 | ZaVoid | Hadi-: whats your origination device? |
| 21:38.07 | [r]evolution | hey qwell -- how long does it typically take to get the license to post patches etc approved? |
| 21:38.20 | Qwell | a day |
| 21:38.26 | Qwell | they're only checked once a day, so... |
| 21:39.05 | [r]evolution | oh ok |
| 21:39.28 | ZaVoid | so this is my vcall flow... client -- spx -- asterisk -- transcoder device -- carrier.... when ringing 183 comes back form the transcoder.. it passes it to asterisk which passes it t the client.. and i get static ringing.... makes no sense |
| 21:39.40 | [r]evolution | word. |
| 21:39.59 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:40.34 | JT | ZOMG SIP PRI |
| 21:40.40 | JT | stfu about the BS PRI already |
| 21:40.58 | JT | every time you say sip pri, another person wants to shoot you |
| 21:41.03 | [TK]D-Fender | JT : unload chan_bile.so |
| 21:41.08 | [r]evolution | lolol |
| 21:41.13 | bkruse | jbot: [TK]D-Fender++ |
| 21:41.15 | bkruse | you earned that one. |
| 21:41.33 | JT | i don't know how many times i've told him there is NO SUCH THING as a SIP PRI |
| 21:41.44 | [r]evolution | i laughed... twice. |
| 21:42.03 | [TK]D-Fender | JT: unload chan_brokenrecord.so |
| 21:42.07 | JT | Hadi-: you cannot use that provider with asterisk. |
| 21:42.18 | JT | Hadi-: asterisk supports 20ms rtp packetisation only |
| 21:42.24 | Qwell | JT: wrong |
| 21:42.29 | JT | since when? |
| 21:42.36 | Qwell | at least 9 months |
| 21:42.38 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 21:42.38 | ZaVoid | lol |
| 21:42.45 | JT | in what version? |
| 21:42.47 | [r]evolution | i think someone may need to take JT some valium... he may be close to giving himself an aneurysm |
| 21:42.50 | [TK]D-Fender | Hadi-: * supports only a single packetization rate per codec as per codecs.conf |
| 21:43.04 | Yourname`` | Hi. When I simply get disconnected from a CLI, it usually means Asterisk crashes, right? |
| 21:43.10 | [TK]D-Fender | Hadi-: (for the fixed rate ones) |
| 21:43.10 | ZaVoid | codecs.conf? don't have codecs.conf |
| 21:43.15 | [r]evolution | depends... on if you typed exit or not |
| 21:43.23 | Qwell | JT: 1.4, so over a year |
| 21:44.02 | JT | hooray |
| 21:44.03 | Hadi- | hum |
| 21:44.07 | Hadi- | lame MCI Canada |
| 21:44.09 | Hadi- | thats the issue |
| 21:44.19 | JT | and sip pris for lols and rofls |
| 21:44.26 | Hadi- | nothing but problems with them |
| 21:44.39 | [TK]D-Fender | hrmm.... can't see it in mine... thts where I remember it... |
| 21:44.51 | [r]evolution | ready......... find a different carrier... go! |
| 21:45.01 | [r]evolution | rofflecopter? lollerblades? |
| 21:45.18 | [r]evolution | unload res_retarded.so :( |
| 21:46.39 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:46.58 | bkruse | [r]evolution: Segfault |
| 21:47.24 | Hadi- | i guess allow=g729:30 |
| 21:47.30 | Hadi- | is only supported in asterisk 1.4? |
| 21:48.46 | [r]evolution | pretty sure qwell wouldn't have said that came about as of asterisk 1.4 if it werent... |
| 21:48.49 | [r]evolution | but maybe thats just me. |
| 21:51.14 | [hC] | im pretty sure there was no way to specify the g729 bit rate before 1. |
| 21:51.15 | [hC] | 1.4 |
| 21:51.53 | [r]evolution | but maybe thats just you |
| 21:54.32 | _x86_ | -- Hungup 'Zap/27-1<MASQ>' |
| 21:54.36 | _x86_ | what's the <MASQ> mean? |
| 21:55.36 | bkruse | channel masquerade? |
| 21:55.51 | [hC] | Meat and Sauce Queen! |
| 21:57.32 | [r]evolution | that just sounds really vile hc |
| 21:58.20 | [r]evolution | im sayin like... the chick at the center of a bukkake gang |
| 21:58.23 | [r]evolution | ewwwww |
| 22:00.53 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-112-82.lns10.syd6.internode.on.net) |
| 22:01.31 | JT | quite ot, but who says she dislikes it? |
| 22:03.31 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 22:03.38 | Assid | yoza |
| 22:03.55 | Assid | office just got some polycom 601's :P |
| 22:04.03 | Assid | i wonder if ishould update the firmware on it |
| 22:04.41 | [TK]D-Fender | Assid: Yes, to 1.6.7 absolute minimum. I recommend 2.2.0 |
| 22:04.52 | [TK]D-Fender | Assid: IP601's come stock with 1.6.3 |
| 22:05.01 | Assid | yeah |
| 22:06.15 | [r]evolution | yuck... |
| 22:08.15 | JT | what's yuck? |
| 22:08.31 | [r]evolution | Meat/Sauce Queen :) |
| 22:08.46 | JT | sure |
| 22:08.51 | JT | made up term i think |
| 22:11.02 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 22:11.58 | muiro | question: I'm attempting to concatenate a string together, but I need to put ":" in the string. I've tried putting in it double quotes, I've tried escaping it, but asterisk only seems to want to operate on it. How can I do this? |
| 22:12.19 | Yourname`` | Hi. When I simply get disconnected from a CLI, it usually means Asterisk crashes, right? |
| 22:12.20 | [TK]D-Fender | muiro: paste what you tried |
| 22:12.34 | [TK]D-Fender | Yourname`: or restarted for whatever reason |
| 22:12.42 | [r]evolution | or maybe you turned the computer off by accident |
| 22:13.18 | muiro | [TK]D-Fender: sepparated spaces: ":" : \: "\:" |
| 22:13.33 | [TK]D-Fender | muiro: please paste the exact code you're using |
| 22:13.39 | muiro | bah, hold on |
| 22:13.41 | *** join/#asterisk Connor (n=billy@198-144-165-66.knx.tn.nxs.net) |
| 22:14.15 | Connor | I'm having a problem getting a Asterisk to router a VoIP call to a Sipura 3000 PSTN port.. |
| 22:14.21 | Connor | Anyone done this before? |
| 22:15.01 | [TK]D-Fender | Connor: www.voxilla.com <- go check their forums for guides on configuring it |
| 22:15.24 | Connor | I have. Not working quiet right.. other wise.. I wouldn't be here. |
| 22:15.48 | Connor | I've got asterisk routing the call to the SPA3000, but.. it's not sending it out the PSTN correctly.. |
| 22:15.55 | Connor | I think I have a issue with the SPA dialplan |
| 22:15.55 | [TK]D-Fender | Connor: well apstebin what you've done so far and SIP debug of failed attemps |
| 22:16.12 | [TK]D-Fender | Connor: if its on the SPA, then you'll have to refer to Voxilla |
| 22:16.29 | lesouvage | Des anybody know about seamles transfer of sip calls to gsm, when walking out of range of the access point the voip connection is taken over by a gsm conection without the caller even noticing the transfer (a demand in a tender paper) |
| 22:17.17 | blitzrage | lesouvage: I think that'd only be possible if you were in control of the GSM connection along with the WiFi connection so that you can do the call handoff |
| 22:17.35 | blitzrage | i.e. not impossible... just not... practical unless you own a cell network.... |
| 22:17.48 | blitzrage | at least that's how I see it. I'm not expert in the area. |
| 22:18.11 | [r]evolution | agh... crap there was a company out in Cali that was supposedly working on that |
| 22:18.15 | [r]evolution | the whole dual-mode thing |
| 22:18.34 | [r]evolution | b/c in all honesty blitz -- even being in control of GSM and WiFi/VoIP |
| 22:18.44 | [r]evolution | it still doesnt happen too cleanly |
| 22:18.51 | blitzrage | oh no... definitely not |
| 22:18.56 | [r]evolution | read : not at all ;x |
| 22:19.05 | [r]evolution | but there this was company in california |
| 22:19.07 | [r]evolution | motehr.. |
| 22:19.08 | [r]evolution | fuxxxx |
| 22:20.04 | [r]evolution | i forgot the name... its driving me nuts now |
| 22:20.06 | [r]evolution | something with a k |
| 22:20.11 | blitzrage | I mean, even the straight handoff between WiFi APs isn't very common. I think Cisco has something that does it. Basically all the APs are connected together via a physical connection (LAN/WAN/MAN), and they talk to each other to hand off the calls |
| 22:20.15 | [r]evolution | you might want to look at http://www.calypsowireless.com as well lesou. |
| 22:21.00 | lesouvage | blitzrage: I had the same idea, I can imagin that a call is taken over by a kind of call back or transfer with a pause but seamlessly sounds very far in the future. |
| 22:21.16 | blitzrage | lesouvage: well, it certainly wouldn't be trivial |
| 22:21.17 | muiro | [TK]D-Fender: ah, figured it out. I was putting everything inside $[]. I took it out so now it won't try to operate the : |
| 22:21.37 | blitzrage | the : in $[ ] means to do a regex I think |
| 22:21.47 | muiro | yeah, it does |
| 22:21.55 | muiro | I was just trying to concatenate ":" into a string |
| 22:22.00 | blitzrage | you can probably escape it |
| 22:22.08 | blitzrage | try \\\: |
| 22:22.12 | muiro | I wasn't sure from the documentation whether string concat needed to go into $[] |
| 22:22.20 | [r]evolution | dammit it is seriously pissing me off... |
| 22:22.21 | muiro | but, anywayn, I eliminated the $[] and now it works |
| 22:22.21 | [r]evolution | :( |
| 22:22.22 | blitzrage | muiro: oh -- to concat -- no, you don't |
| 22:22.31 | blitzrage | ya, just comparisons need $[ ] |
| 22:22.43 | muiro | blitzrage: yeah, there we no code samples so I wasn't sure. No I know. Thanks. |
| 22:23.52 | codejunky | How can I get a call history? |
| 22:29.08 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 22:30.55 | [r]evolution | exit |
| 22:30.56 | [r]evolution | er |
| 22:30.57 | [r]evolution | fuck |
| 22:30.59 | [r]evolution | wrong screen |
| 22:31.02 | [r]evolution | peace out hookers. |
| 22:31.04 | *** part/#asterisk [r]evolution (n=spmcatch@208.6.94.10) |
| 22:36.47 | jer | codejunky, asterisk-stat is one way |
| 22:37.12 | jer | codejunky, http://www.areski.net/asterisk-stat-v2/about.php |
| 22:42.33 | *** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
| 22:42.52 | markit | hi, how can I know what codec a call is currently using from CLI? |
| 22:43.11 | [TK]D-Fender | markit: for sip : sip shoe channels |
| 22:43.15 | [TK]D-Fender | markit: for sip : sip show channels |
| 22:43.22 | markit | thanks [TK]D-Fender, I try |
| 22:44.01 | markit | [TK]D-Fender: perfect, thanks a lot! |
| 22:45.43 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 22:49.08 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:50.20 | lesouvage | is an uptime for a telephone solution of 99,999% realistic. That's 5 minutes off time a year? (still reading the tender paper) |
| 22:50.39 | *** join/#asterisk cjs (n=cjs@d131.GtokyoFL2.vectant.ne.jp) |
| 22:51.20 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 22:51.59 | ManxPower | lesouvage: Maybe, but you would have to spend a lot of money. |
| 22:52.43 | *** join/#asterisk nitrus (n=ntisog@72-34-76-86.skyriver.net) |
| 22:53.28 | nitrus | anyone here have an adtran 750 with a te110p or equivalent? i need to know the signalling for fxo and fxs you're using and echo cancelling settings/compilation changes for zaptel |
| 22:53.41 | nitrus | im having delays in channel bridging and occassional echo problems |
| 22:53.46 | [TK]D-Fender | lesouvage: No, it isn't |
| 22:53.59 | JT | nitrus: is it just FXS ports on the adtran? |
| 22:54.21 | Greek-Boy | whose got SS7 running in ast 1.4? |
| 22:54.23 | nitrus | i have half and half except i only use 1 FXO port for output emergency |
| 22:54.25 | [TK]D-Fender | lesouvage: think that on any PBX in the course of a year a board may fry which will take you a long time searching through drawers to find a replacement for :) |
| 22:54.30 | nitrus | everything else is FXS |
| 22:54.54 | Greek-Boy | when i say SS7 i mean libss7 or anything else... |
| 22:54.58 | lesouvage | ManxPower: that wa my idea too, what is a reasonable demand for uptime in not live treatening or national security treatening situations |
| 22:55.08 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 22:55.11 | [TK]D-Fender | Greek-Boy: You're jsut about the only person here who even speaks of it |
| 22:55.31 | Greek-Boy | seems like it |
| 22:55.43 | Greek-Boy | I got a link up and running but not stable at all |
| 22:55.45 | JT | nitrus: the calls with echo, are they to other phones on the pstn? |
| 22:55.54 | Greek-Boy | I'm still determined to find something solid |
| 22:56.04 | Greek-Boy | 1.2 was more stable with it |
| 22:56.17 | nitrus | they're usually a call inbound from sip bridged to an internal analog phone on fxs |
| 22:57.12 | lesouvage | Greek-Boy: Are you sure your provider is doing what he is supposed to do, offering a standard compliant connection. I have wasted lots of time lately ending with the conclusion that the provider didn't had everything in place (they agree with that) |
| 22:57.46 | Greek-Boy | its possible |
| 22:58.31 | Greek-Boy | why do u have an SS7 provider lesouvage? SS7 is usually meant for interconnecting telco providers. |
| 22:58.45 | Greek-Boy | are u a provider? :P |
| 22:58.57 | JT | nitrus: is the other end hearing the echo, or are you guys? |
| 22:58.57 | nitrus | JT: there is no echo phone to phone in the office |
| 22:59.15 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:59.15 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:59.44 | nitrus | the person on the FXS hears themselves echo, and the person on the other end hears themselves echo |
| 23:00.01 | nitrus | neither people hear either partys echos |
| 23:00.17 | lesouvage | no I'm not a provider, but what I understand from isdn 30 conection is that if you use isdn30 for phone and data connection ss7 is there to have it work smoothly. I have had lots of trouble with a isdn30 conection. |
| 23:00.17 | JT | weird |
| 23:00.32 | Greek-Boy | i c |
| 23:01.01 | nitrus | and there is always a delay when someone answers the FXS the other party wont hear them say hello if the FXS answering party doesnt wait to speak |
| 23:01.48 | [TK]D-Fender | BBIAB |
| 23:02.25 | lesouvage | Greek-Boy: But I start to read and study when the problem was their so there is still a change that I'm talking nonsense because I misunderstood it comletely. see: http://www.commsdesign.com/showArticle.jhtml;jsessionid=MYKUAQ5BSSUMEQSNDLPSKH0CJUNN2JVN?articleID=16501900 |
| 23:02.53 | JT | nitrus: that's probably a provider problem |
| 23:05.10 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 23:06.28 | JT | lesouvage: ISDN30/E1 PRI uses a D channel that uses Q.931 |
| 23:06.30 | JT | not SS7 |
| 23:06.58 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 23:07.12 | JT | lesouvage: 5 nines availability is realistic only if you have class 5 spec gear |
| 23:09.01 | nitrus | JT: is kewl start the right signaling to use if im on an adtran 750 with a te110p? |
| 23:09.26 | lesouvage | JT: see figure 12 on the link. I'm sure you are right but from that I understand that in case of data and vocie ss7 comes in place. I have still lots to read |
| 23:09.27 | JT | yeah i guess if it's configured for it |
| 23:09.37 | JT | kewlstart is an extension to loopstart |
| 23:09.48 | *** join/#asterisk saftsack (n=saftsack@p4FC7780C.dip.t-dialin.net) |
| 23:09.58 | nitrus | one of the channels i have is configured as ground start because i'm using it with an intercom system |
| 23:09.59 | JT | lesouvage: data and voice, HUH? |
| 23:10.23 | nitrus | the others i think are just set to traditional, im not sure if kewlstart is actually listed in the adtran's options |
| 23:15.55 | *** join/#asterisk PodMan99a (n=keith@82-34-164-205.cable.ubr02.maid.blueyonder.co.uk) |
| 23:16.38 | PodMan99a | hey guys... issue with * supprisingly.... i am unable to hear anything both ways on an asterisk call away from office ?? any ideas? |
| 23:16.39 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:17.15 | lesouvage | JT: what do you mean by class 5 spec gear, see: http://en.wikipedia.org/wiki/Class_5_telephone_switches class 5 is a functional clasifaction and says nothing about availibility. |
| 23:19.06 | PodMan99a | connecting through SIP sorry forgot that bit |
| 23:19.54 | JT | lesouvage: well i was refering to more high availability hardware and software |
| 23:20.02 | JT | asterisk is not high availability |
| 23:22.23 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 23:28.13 | *** join/#asterisk idefine (n=panaii@c-24-99-1-240.hsd1.ga.comcast.net) |
| 23:33.41 | nitrus | what is e&m signalling |
| 23:33.57 | outtolunc | ear and mouth |
| 23:34.38 | outtolunc | http://www.atis.org/tg2k/_e_m_signaling.html |
| 23:35.20 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 23:35.27 | generalhan | hey all !~ |
| 23:36.50 | generalhan | how difficult would it be to setup 2 SIP phones in a remote location, both behind the same router ? i have only ever set up larger remote offices behind a 2nd * server. |
| 23:50.58 | *** join/#asterisk Maliuta (n=nikolai@203.201.152.211) |
