| 00:03.54 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:07.46 | *** join/#asterisk PepOSX (n=pepOSX@190.72.147.76) |
| 00:13.05 | *** join/#asterisk Strom_M (n=strom@m660e36d0.tmodns.net) |
| 00:17.34 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 00:23.33 | Yourname`` | [Nov 29 16:44:35] WARNING[32380]: translate.c:163 framein: no samples for ulawtolin |
| 00:23.34 | Yourname`` | [Nov 29 16:40:39] WARNING[26711]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '4a5042636c589e7c02c4f1906f1ba6e8@6x.xx.xx.xx'. Giving up. |
| 00:23.43 | Yourname`` | No dice on these fo |
| 00:23.47 | Yourname`` | <PROTECTED> |
| 00:26.34 | *** part/#asterisk ak_hepcat (n=ak_hepca@patroler.gci.net) |
| 00:27.25 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 00:30.39 | *** join/#asterisk Strom_C (n=strom@m390e36d0.tmodns.net) |
| 00:30.46 | Qwell | Strom_C: ! |
| 00:31.01 | Qwell | Strom_C: I have a doc for you to review :D |
| 00:31.22 | Qwell | if you're interested, that is |
| 00:31.29 | mocker | So I learned in my Aspect course today how predictive dialer voice machine detection works. |
| 00:31.44 | mocker | Much simpler than I though, it just listens for the lenght of time for a greeting. |
| 00:31.45 | Strom_C | Qwell: sure |
| 00:31.52 | *** join/#asterisk pdd_ (n=pdd@71-83-188-204.dhcp.lnbh.ca.charter.com) |
| 00:32.00 | Qwell | Strom_C: callerid stuff |
| 00:32.02 | mocker | i.e. "Hello?" vs. "Thanks for calling so-and-so" |
| 00:32.17 | Strom_C | Qwell: sure, send it on over |
| 00:38.57 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 00:43.53 | Yourname`` | When lines are all static sounding, what could it be? |
| 00:46.58 | remmo | interference |
| 00:47.31 | Yourname`` | Like what? |
| 00:50.40 | *** join/#asterisk Strom_M (n=strom@m750e36d0.tmodns.net) |
| 00:52.52 | pdd_ | anyone familiar with asterisk-java manager api? I'm having trouble playing a message ONLY after the call has been answered |
| 00:54.06 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 00:54.21 | MrTelephone | hey Qwell, who would program me a little patch for my Authorization Header problem? |
| 00:59.47 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 01:05.25 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 01:07.58 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-eeec13a0f0850a28) |
| 01:09.36 | MrTelephone | the people in #cisco are a bunch of jerks |
| 01:14.11 | Yourname`` | Every smartass is a jerk. |
| 01:24.21 | Qwell | MrTelephone: You'd be a jerk too if you had to deal with Cisco's BS every day |
| 01:26.32 | Yourname`` | Is there a reference to what I see when I type show queues? For example, how it says "Not in use" or "In use" or "Unavailable" or "Invalid" etc? |
| 01:26.35 | Qwell | Yourname``: it depends on the channel - they can mean multiple things |
| 01:28.48 | Yourname`` | Qwell: So -> SIP/14 (dynamic) (Not in use) has taken 59 calls (last was 6 secs ago) -> That means SIP/14 is waiting for the next call in the queue? |
| 01:29.03 | Qwell | no, it means that if a call were sent to it, it would ring |
| 01:29.17 | Qwell | at that very moment |
| 01:29.45 | Yourname`` | Qwell: And it means the agent is not on call? |
| 01:29.52 | Qwell | what agent? |
| 01:30.05 | Yourname`` | Qwell: SIP/14 |
| 01:30.10 | Qwell | that isn't an agent |
| 01:30.21 | Yourname`` | Qwell: A device? |
| 01:30.25 | Yourname`` | Qwell: A channel? |
| 01:30.28 | Qwell | a queue member |
| 01:30.40 | Yourname`` | Qwell: lol, ok, so it means the queue member is not on call? |
| 01:30.45 | Qwell | no, I never said that |
| 01:31.07 | Qwell | it literally means, that if a call were sent to SIP/14 at that very moment, it would ring. |
| 01:31.32 | Yourname`` | Qwell: Ok, so it would ring. But, just curious, what else could it be doing other than having to rin? |
| 01:31.34 | Yourname`` | ring* |
| 01:31.37 | Qwell | the user could have a busylevel higher than 1 set, and he could very well already be talking to a caller |
| 01:32.12 | Yourname`` | Ah. |
| 01:32.16 | Yourname`` | Gotcha. |
| 01:32.30 | Qwell | whatever SIP reports as the state, the queues will happily use |
| 01:32.38 | Yourname`` | While I have your attention, Qwell, can I ask you a couple other things too please? |
| 01:32.44 | Qwell | no, I have a box to open |
| 01:32.49 | Yourname`` | lol |
| 01:32.58 | Qwell | (no, really) |
| 01:33.01 | Yourname`` | Ok, [Nov 29 16:44:35] WARNING[32380]: translate.c:163 framein: no samples for ulawtolin -> this comes up quite often.. what does it usually do? |
| 01:33.14 | *** join/#asterisk powell2 (n=cullen@c-75-66-240-62.hsd1.tn.comcast.net) |
| 01:33.16 | powell2 | hey |
| 01:33.16 | Yourname`` | I have to wait till xmas to open boxes. :( |
| 01:33.24 | Qwell | HP sent me a brand new laptop :D |
| 01:33.34 | Qwell | because I'm such a nice guy |
| 01:33.43 | Qwell | s/I.*/they screwed up majorly/ |
| 01:33.48 | Yourname`` | Because you broke it! |
| 01:33.51 | Qwell | basically |
| 01:33.56 | Yourname`` | There you go, truth be told! |
| 01:34.15 | Qwell | nah, I sent mine in for repair, and a month later, still hadn't gotten it back |
| 01:34.29 | Qwell | then when I *did* get it back, the problem wasn't fixed, and there was stuff missing O.o |
| 01:34.35 | Qwell | solution: new laptop |
| 01:35.22 | Yourname`` | My brother in law called me asking if his "shipped-with-vista" HP DV2000 can install XP on it as Vista is no good for him.. |
| 01:35.36 | Qwell | vista is no good for anybody |
| 01:36.26 | Yourname`` | I never agreed to upgrading to Vista although I had to go buy 3 copies for fellas in my office cuz they were acting like they are kids in a candy store. |
| 01:36.37 | Nugget | OS X > * :) |
| 01:36.47 | Yourname`` | 6 months later, one by one, I was request to get XP pro for them, lol |
| 01:38.30 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:40.16 | MrTelephone | xp is hard to install on the new sata controllers |
| 01:40.20 | MrTelephone | u need a boot floppy |
| 01:40.28 | MrTelephone | then u need a floppy drive |
| 01:40.42 | Yourname`` | Qwell : Done opening your box? |
| 01:40.53 | Yourname`` | MrTelephone: Are you being serious? |
| 01:41.36 | MrTelephone | yeah |
| 01:41.40 | MrTelephone | why |
| 01:41.53 | Corydon76-dig | Because he loves Vista |
| 01:42.01 | MrTelephone | all the new toshibas im ordering won't install xp without drivers for the sata controller |
| 01:42.18 | MrTelephone | i had a dell dimension that wouldn't even take the drivers |
| 01:42.37 | MrTelephone | and I phoned dell and he had me do a system restore.. what if the restore frigs up? |
| 01:42.47 | MrTelephone | this whole shit is getting f'd up |
| 01:43.17 | Nugget | migrate to macs. :) |
| 01:43.28 | Nugget | I have an office full of mac minis and it's great. |
| 01:44.53 | Yourname`` | lol I dont love it Corydon76-dig |
| 01:44.54 | MrTelephone | might have to |
| 01:45.05 | Yourname`` | I was going to do it for the bro in law thinking i was easy. |
| 01:45.06 | Yourname`` | it* |
| 01:45.27 | Nugget | os x is fantastic. it's unix, you can ssh into your machines, but it still runs office and doesn't totally suck for your users. |
| 01:52.04 | Qwell | heh |
| 01:52.13 | Qwell | I was able to convince my sister-in-law to get a mac |
| 01:52.14 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
| 01:52.19 | Qwell | she just got back from the apple store :D |
| 01:52.51 | r0d3nt | i'm sorry. |
| 01:53.06 | r0d3nt | over priced boutique x86 " pretty " hardware. |
| 01:53.09 | r0d3nt | IMO. =) |
| 01:53.18 | Qwell | can you run osx on your laptop? |
| 01:53.20 | r0d3nt | btw, i own a mini and a powerbook. =) |
| 01:53.22 | r0d3nt | sure can. |
| 01:53.28 | r0d3nt | on my dell and my sony |
| 01:53.42 | r0d3nt | 600m and sony vgn-SZ650N/C |
| 01:53.52 | Qwell | now go get a real processor |
| 01:53.54 | Qwell | :p |
| 01:54.00 | r0d3nt | howz that ? |
| 01:54.06 | Qwell | those != amd |
| 01:54.08 | r0d3nt | dual core 2.2ghz ?? not good enough for you ? |
| 01:54.09 | r0d3nt | haha |
| 01:54.25 | r0d3nt | fuck amd. especially while abudahbi owns 10% of them. |
| 01:54.45 | r0d3nt | but hey, opinions are like assholes =) we all have'm =) and they all stinky =) |
| 01:54.51 | r0d3nt | s/stinky/stink/ |
| 01:54.59 | r0d3nt | jbot: thanks. |
| 01:54.59 | jbot | de rien, r0d3nt |
| 01:55.03 | km- | Qwell: I loaded osx on my dell |
| 01:55.06 | km- | Qwell: latitude D620 |
| 01:55.14 | km- | my boss just shook his head and called me a nerd |
| 01:55.18 | r0d3nt | ahaha |
| 01:55.38 | *** join/#asterisk sergey (n=sergey@sergey.iks.ru) |
| 01:55.42 | r0d3nt | yeah i prefer win32/64 over OSX anyways.. but i use'm all and support'm all.. |
| 01:55.57 | r0d3nt | anyways, Qwell GL with your sister-N-law's computer |
| 01:56.10 | r0d3nt | be sure to sell her on " no viruses ! " |
| 01:56.17 | r0d3nt | hahah |
| 01:56.18 | km- | after almost 14 years of using linux, I pretty much use windows on the desktop, but if it involves something that remotely resembles a server process, it occurs in a variant of *nix. |
| 01:56.24 | [hC] | you use and support both, and still prefer win32? |
| 01:56.33 | [hC] | and you made a point of even saying you prefer win64? |
| 01:56.36 | [hC] | you are clearly out of your mind |
| 01:56.42 | km- | haha |
| 01:56.42 | r0d3nt | km-: absolutely, but i still have to use win2k3 servers and so on... but yeah... |
| 01:56.59 | r0d3nt | [hC]: well no, i hate 64 driver bullshit. |
| 01:57.01 | km- | I just find the jobs I have to do are easier to perform on windows, when it's desktop related |
| 01:57.02 | r0d3nt | but i support them all |
| 01:57.07 | r0d3nt | and i prefer windows over a linux desktop |
| 01:57.12 | r0d3nt | and over a OSX desktop |
| 01:57.18 | r0d3nt | my servers are gentoo or freebsd |
| 01:57.34 | r0d3nt | my desktops are OSX for apple land, and win32/Vista/XP for gaming and general usage... |
| 01:57.36 | [hC] | so clearly you are just a glutton for punishment all around |
| 01:57.40 | [hC] | im surprised you didnt say openbsd |
| 01:57.42 | [hC] | :) |
| 01:57.45 | r0d3nt | fuck openbsd. |
| 01:57.49 | km- | yeah, I ran gnome and beryl for a while but went back to windows, I found myself getting angry that certain features windows had, gnome didn't. |
| 01:58.04 | km- | haha, lets encrypt the swap! |
| 01:58.05 | km- | woo hoo |
| 01:58.26 | *** part/#asterisk sergey (n=sergey@sergey.iks.ru) |
| 01:58.28 | r0d3nt | linux is cool and all, but yeah... |
| 01:58.46 | Qwell | you don't like getting things done? |
| 01:59.01 | r0d3nt | wtf are you talking about ? |
| 01:59.05 | km- | I find windows is more efficient for desktop-related GSD. |
| 01:59.23 | km- | X-based desktop environments, in their current form, are less efficient for similar tasks. |
| 01:59.40 | r0d3nt | i use vista business on my work laptop, and XP on my personal laptop.. |
| 01:59.49 | km- | now I wouldn't run vista, that's for sure |
| 01:59.50 | km- | heh |
| 01:59.58 | r0d3nt | XP get's EOL'd in 2008 |
| 02:00.06 | r0d3nt | 2nd or 3rd quarter if it doesn't get pushed out. |
| 02:00.14 | r0d3nt | and you have a short memory... |
| 02:00.19 | r0d3nt | XP was crap too when it first came out |
| 02:00.27 | r0d3nt | not until SP2 did it become usable and widely accepted. |
| 02:00.28 | Qwell | it was nowhere near as bad as vistas |
| 02:00.30 | km- | yeah, I ran win2k until it stabilized :) |
| 02:00.31 | r0d3nt | vista will be no different. |
| 02:00.32 | Qwell | vista is* |
| 02:00.45 | km- | I still run win2k on vmware a lot |
| 02:00.46 | Qwell | XP wasn't 50% slower than 2k |
| 02:01.15 | km- | since when does an OS suggest you have a whizbang graphics adapter? |
| 02:01.35 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 02:01.41 | r0d3nt | XP required more video and ram. |
| 02:01.48 | km- | some of us don't want 3 pcix slots taken by graphics cards so we can click a fucking icon |
| 02:01.50 | km- | negative |
| 02:01.54 | r0d3nt | haha |
| 02:01.56 | km- | I ran XP on the same equipment I ran win2k on |
| 02:02.02 | r0d3nt | sure |
| 02:02.08 | Nugget | for desktop usage it boils down to this canard: Linux is for people who like to do things TO their computers. OS X is for people who like to do things WITH their computers. Windows is for gaming. :) |
| 02:02.15 | km- | I can't run vista on the hardware I've run XP on. |
| 02:02.32 | r0d3nt | Nugget: ahahahah pretty much =) windows is for business and gamign yes =) i agree |
| 02:02.54 | Qwell | Nugget: wine |
| 02:03.07 | Qwell | windows is irrelevant |
| 02:03.25 | Qwell | games that I play in wine run FASTER in wine than they do in windows, on the same hardware |
| 02:03.36 | Nugget | on the server as long as you're using some flavor of Unix you're all right in my book, but I personally find that Linux tends to create more headaches than it solves. I'm one of those bearded, suspender-wearing BSD relics. |
| 02:03.41 | Qwell | now, where's my ubuntu CD, so I can wipe this windows install |
| 02:03.55 | km- | now that eve released a linux client |
| 02:03.59 | file | Nugget: suspenders? really? |
| 02:04.11 | km- | maybe I'll go back to linux on the desktop for a while :) |
| 02:04.12 | Nugget | give me a unix that's built for admins and bofhs, not one that's trying to moonlight as a PVR |
| 02:04.28 | Qwell | hmm, are there any linux distros that can boot from a usb key? |
| 02:04.38 | km- | yep |
| 02:04.39 | km- | a bunch |
| 02:04.42 | km- | plop, dsl |
| 02:04.59 | km- | ubuntu as well |
| 02:05.02 | Qwell | any that are 64-bit? I'm gonna be bootstrapping an install |
| 02:05.11 | Qwell | km-: really? |
| 02:05.15 | km- | I dont think the 64bitness matters |
| 02:05.22 | km- | you just need to fire up syslinux |
| 02:05.34 | km- | to toolchain the kernel in and then its aces |
| 02:05.40 | Qwell | what's the image for ubuntu called? |
| 02:05.57 | km- | just search for ubuntu usb flash drive |
| 02:06.05 | km- | its not an image so much as a procedure |
| 02:06.09 | *** join/#asterisk perf3kt (n=perf3kt@adsl-68-73-146-81.dsl.ipltin.ameritech.net) |
| 02:06.21 | Qwell | bootable? |
| 02:06.26 | Qwell | ahh |
| 02:06.27 | km- | but I did see it on ubuntu's site |
| 02:06.32 | km- | so it's legit |
| 02:06.36 | Yourname`` | <PROTECTED> |
| 02:06.40 | Yourname`` | What the hell is this! :( |
| 02:06.44 | Yourname`` | No google help either. |
| 02:06.49 | Qwell | 256mb good? |
| 02:07.04 | km- | Qwell: one of the methods of doing it says you should copy the iso to the usb stick |
| 02:07.11 | km- | Qwell: but there are a couple ways to do it on the page |
| 02:07.24 | km- | Yourname``: are calls being dropped? |
| 02:07.33 | km- | Yourname``: a lot of bellyaching asterisk does in verbose mode can be safely ignored |
| 02:07.55 | Qwell | crap, need 1gb+ |
| 02:08.13 | km- | you can get 2gb usb keys for pretty cheap, $30 some dollars |
| 02:08.19 | km- | at least around here (suburban philadelphia) |
| 02:08.28 | Qwell | I can buy a blank CD for cheaper :) |
| 02:08.49 | grimsy | Qwell: tryied DSL? http://damnsmalllinux.org/ |
| 02:08.54 | Yourname`` | km-No calls going in :( |
| 02:09.31 | km- | Qwell: this is true |
| 02:09.49 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 02:09.52 | km- | Yourname``: where are the calls coming from and what are they going to, hardware wise? |
| 02:10.17 | km- | are we talking TDM to a sip client on a PC? |
| 02:11.01 | Yourname`` | km-: Just SIP trunks.. and softphone behind a NAT. |
| 02:11.12 | WilliamK | anyone know what would cause this to occur? there isn't a 102 in the sip.conf --- SIP/102 (dynamic) (Invalid) has taken no calls yet |
| 02:11.33 | km- | WilliamK: you have dynamic sip turned on? |
| 02:12.13 | WilliamK | what's the diff between having dynamic sip turned on vs having peers that are dynamic? |
| 02:12.28 | WilliamK | the others are fully registered and have entries |
| 02:13.07 | WilliamK | and 102 doesn't show up in the sip show peers either |
| 02:14.38 | km- | WilliamK: do you have allowguest=yes in your config file? |
| 02:15.05 | km- | Yourname``: why is there a transcoding going on for your call, I wonder? Why's it trying to convert ulaw to lin |
| 02:16.47 | WilliamK | nope |
| 02:16.53 | Yourname`` | km-: What's lin? |
| 02:17.34 | km- | linear pcm |
| 02:18.04 | km- | where are the sip trunks coming from? |
| 02:18.20 | km- | try asserting a different codec via disallow=all, allow=something |
| 02:18.28 | km- | if your source supports gsm, for instance, try forcing gsm. |
| 02:18.48 | km- | you really don't want ulaw over a broadband connection anyway, it's incredibly bandwidth intensive. |
| 02:22.53 | km- | does anyone here use a sip client with video capability with asterisk? |
| 02:24.06 | r0d3nt | http://www.candelatech.com/lanforge_v3/datasheet.html |
| 02:24.18 | r0d3nt | this has been pretty helpful |
| 02:24.28 | r0d3nt | testing and breaking and what not. |
| 02:27.08 | r0d3nt | and with that, good night and good luck. |
| 02:27.09 | km- | one thing that's important to note about measurements of bandwidth. g711u is 64kbps per direction. |
| 02:27.16 | r0d3nt | yup. |
| 02:27.18 | km- | add tcp overhead and it's a whopping 144kbps |
| 02:27.23 | km- | that's a lot of banwdidth. |
| 02:27.26 | r0d3nt | yup. |
| 02:27.28 | km- | s'why I use g729! |
| 02:27.29 | km- | hehe |
| 02:27.48 | km- | nite rodent |
| 02:27.49 | [hC] | So for hosted pbx, when you have a bunch of people logging in to one pbx, and everyone wants to have an extension 100, thats easy to solve with contexts, but what about registrations? how do you guys deal with that? |
| 02:28.06 | [hC] | register phones as something other than the exten number, or is that another workaround im not thinking about? |
| 02:28.09 | km- | [hC]: there's no law that says sip usernames have to be the same as the extension. |
| 02:28.33 | [hC] | Only other thing I can think of is multiple copies of asterisk running on different ips, either on the same box or in virtual machines |
| 02:28.38 | km- | if you're multihomed, it's probably a good idea to use descriptive alphanumeric usernames. |
| 02:28.39 | [hC] | and vm's seem like a disaster waiting to happen |
| 02:28.53 | km- | nah, vm and directory are both safely multihomeable |
| 02:28.58 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 02:29.10 | km- | just pass the right vm or directory context to the individual apps and voila |
| 02:29.14 | km- | you'll never have an issue with it. |
| 02:29.19 | [hC] | vm = virtual machine |
| 02:29.20 | [hC] | not voicemail. |
| 02:29.28 | km- | oh |
| 02:29.51 | km- | I've run asterisk in vmware and xen |
| 02:30.02 | [hC] | alphanumeric usernames seems the best way to go |
| 02:30.11 | km- | but what you're talking about is easily done in a single asterisk instance |
| 02:30.15 | km- | yeah, that's what I would do. |
| 02:30.24 | [hC] | yep.. thinking about it some more it would even help to identify the client |
| 02:30.29 | [hC] | clientname-exten as the username |
| 02:30.35 | [hC] | superclient-100, superclient-101 |
| 02:30.36 | km- | yeppers. |
| 02:30.52 | km- | or even superclient-jeff, superclient-sandy, etc. |
| 02:31.00 | [hC] | yeah |
| 02:31.09 | [hC] | id get away from that cause names change a lot, but i see your point. |
| 02:31.11 | km- | there's no law that says you need an "extension number" in the username |
| 02:31.31 | km- | that's true |
| 02:31.44 | km- | in a volatile environment it'd be one less thing to change |
| 02:32.30 | km- | I need to make a default landing pad for my isn link |
| 02:35.47 | *** join/#asterisk khronos (n=khronos@c-66-229-159-175.hsd1.fl.comcast.net) |
| 02:36.32 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 02:37.50 | WilliamK | km, just looked at something... appears that allowguest=yes by default unless explicty set to allowguest=no in the conf |
| 02:38.12 | WilliamK | no reason for autocreatepeer=yes is there? |
| 02:42.12 | km- | autocreatepeer is the culprit |
| 02:42.14 | km- | turn that off |
| 02:42.36 | km- | you have no need of that if you're going to explicitly define sip users |
| 02:45.16 | WilliamK | by default I assumed settings were pretty tight, appears that things need to be explictly defined |
| 02:46.40 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:47.15 | MrTelephone | qwell |
| 02:52.35 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177808409.dsl.bell.ca) |
| 02:55.05 | WilliamK | hmmm, nice error when upgrading to the latest svn.... |
| 02:55.08 | WilliamK | channel.c:3013 set_format: Unable to find a codec translation path from unknown to gsm |
| 03:08.17 | pdd_ | anyone familiar with asterisk-java manager api? I'm having trouble playing a message ONLY after the call has been answered |
| 03:08.40 | MrTelephone | can someone code me a fix in asterisk :( |
| 03:09.15 | [TK]D-Fender | pdd_, "show application dial" |
| 03:09.54 | pdd_ | tried Dial with option G and X but they don't work |
| 03:11.43 | pdd_ | sorry Dial with option M(macro^ext^pri) |
| 03:16.45 | MrTelephone | does the nonce have to change all the time? |
| 03:17.47 | [TK]D-Fender | pdd_, describe the call flow. |
| 03:19.00 | Qwell | ~freepbx |
| 03:19.01 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 03:19.05 | pdd_ | my java app --> originateAction calling a local channel wich places the Dial call with option M(macro^ext^pri) |
| 03:19.40 | [TK]D-Fender | pdd_, and once it answers, where do you send it? |
| 03:20.33 | pdd_ | when the Dial is answer is supposed to execute the macro |
| 03:20.36 | pdd_ | but it doesn't |
| 03:22.05 | [TK]D-Fender | pdd_, hold off on that part. You Originate a call that will dial person X via a Local channel. Upon answering what exactly do you want it to do? |
| 03:22.40 | pdd_ | all I need to accomplish is to place a call through my app, and deliver a message when the call is answered |
| 03:23.47 | pdd_ | I'm not sure I should use local channel |
| 03:24.25 | [TK]D-Fender | pdd_, ok, call out, answer, play message, then what? |
| 03:24.34 | pdd_ | hangup |
| 03:24.39 | MrTelephone | qwell when and who decides to work on a bug or feature that was reported on bugs.digium.com? |
| 03:25.01 | Qwell | we do |
| 03:25.21 | [TK]D-Fender | pdd_, Ok, simple. You dial your local channel like normal. then you should have set the context,exten,priority to SEND that "callee" into UPON answering. in there you do Answer, Playback, hangup. |
| 03:26.30 | [TK]D-Fender | pdd_, read the rest of the Originate parameters for the exact fields |
| 03:26.40 | pdd_ | send the callee? I'm not following |
| 03:26.51 | pdd_ | you mean the originateAction? |
| 03:28.56 | [TK]D-Fender | pdd_, http://pastebin.com/m7983608a |
| 03:29.37 | [TK]D-Fender | pdd_, when that local channel is answered, they get sent INTO the dialplan in the place specified |
| 03:29.44 | *** join/#asterisk commandLine (n=root@cpe-071-070-229-009.nc.res.rr.com) |
| 03:31.22 | pdd_ | right ... the problem is that the playback occurs before the call is answered (if you let it ring a few secs) |
| 03:32.17 | [TK]D-Fender | pdd_, not in my sample. |
| 03:32.26 | [TK]D-Fender | pdd_, DEPENDING. |
| 03:32.31 | pdd_ | uhm... let me double check |
| 03:32.50 | [TK]D-Fender | pdd_, See the key thing to keep in mind is when is that channel ACTUALLY considered "answered" by Asterisk? |
| 03:32.54 | pdd_ | is there a global setting I might be missing that forces that behaviour? |
| 03:33.10 | [TK]D-Fender | pdd_, So I have to ask : What exactly are you dialing out from on that? |
| 03:33.49 | pdd_ | I'm dialing to PSTN |
| 03:33.56 | [TK]D-Fender | pdd_, what hardware exactly? |
| 03:33.58 | pdd_ | the thing is I'm using the local channel to dial out |
| 03:34.07 | [TK]D-Fender | pdd_, tech matters <- |
| 03:34.07 | pdd_ | TDM |
| 03:34.13 | [TK]D-Fender | pdd_, ANALOG? |
| 03:34.16 | pdd_ | yeap |
| 03:34.18 | pdd_ | Zapata |
| 03:34.21 | [TK]D-Fender | pdd_, THAT is problematic |
| 03:34.27 | pdd_ | uhm |
| 03:34.34 | file | an analog line is considered answered as soon as it is dialed |
| 03:34.39 | [TK]D-Fender | pdd_, See typically a zaptel channel is considered answerd the moment it is placed. |
| 03:34.48 | [TK]D-Fender | </echo> |
| 03:34.48 | [TK]D-Fender | :p |
| 03:34.59 | pdd_ | there you go...finally i get some answers :-) |
| 03:35.15 | pdd_ | good point... I appreciate it |
| 03:35.25 | [TK]D-Fender | pdd_, Now you COULD set your channel to "callprogress=yes", but that can cause other issues |
| 03:35.47 | [TK]D-Fender | pdd_, that parameter is also known as "dropmycallsrandmly" |
| 03:37.35 | pdd_ | I also was trying to create a loop that monitors the DIALSTATUS for ANSWER then break out of the loop and play the message |
| 03:38.04 | pdd_ | but it's useless because (as you mentioned) the ANSWER already took place |
| 03:39.23 | [TK]D-Fender | pdd_, yup.... if you're not running around screaming like the rest of us... you simply haven't grasped the full depth of the problem ;) |
| 03:39.54 | file | analog is good for calling... if you don't need call progress detection... fast dialing... DID information... CID transmission... |
| 03:40.06 | [TK]D-Fender | pdd_, Now if you were using PRI or a voip carrier that passed on call progress similarly you'd be OK. |
| 03:40.24 | pdd_ | I'm started to get the concept :-) ... thanks for your help tough... I should probably look into diff hardware |
| 03:40.33 | JT | s/hardware/lines/ |
| 03:41.36 | [TK]D-Fender | JT : Cause you can't have one without the... OOOOTTTHHHERRRR! |
| 03:41.48 | MrTelephone | if (!strncasecmp("Authorization",name,13) |
| 03:41.51 | MrTelephone | is that legal? |
| 03:42.07 | MrTelephone | or do you have to set 13 to a int var first |
| 03:42.20 | *** join/#asterisk cli4me (n=shizm@cpe-071-070-229-009.nc.res.rr.com) |
| 03:43.23 | pdd_ | do you guys know of a relatively cheap service to terminate my Sip calls into PSTN ? |
| 03:43.36 | pdd_ | I looked into chanskype but doesn't seem reliable |
| 03:43.48 | [TK]D-Fender | pdd_, now go try that "callprogress=yes" + "busydetect=yes", etc I suggested and if it doesn't wreak even MORE havok on your system... then you're welcome :) |
| 03:43.52 | *** join/#asterisk Strom_M (n=strom@m440e36d0.tmodns.net) |
| 03:44.01 | [TK]D-Fender | pdd_, What kind of volume? |
| 03:44.17 | pdd_ | really, really low (for now at least) |
| 03:44.28 | pdd_ | I'll look into those parameters... |
| 03:44.50 | MrTelephone | ~pastebin |
| 03:44.50 | jbot | [~pastebin] 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 |
| 03:45.13 | [TK]D-Fender | pdd_, give that a try first. For the rest, cal the /min charge estimate vs a monthly type deal. Ask yourself if you multiple simultaneous callouts, etc. |
| 03:45.30 | MrTelephone | http://paste.debian.net/43802 |
| 03:45.34 | MrTelephone | do you think that will work? |
| 03:46.20 | pdd_ | great... I have enough to digest I think - I'll try and research my options...thank you guys for the good tips |
| 03:46.45 | [TK]D-Fender | pdd_, np, test your analog first, calc out the rest. Where are you terminating to? |
| 03:46.55 | MrTelephone | someone with some c code experience :P |
| 03:47.15 | pdd_ | what do you mean? |
| 03:47.19 | MrTelephone | *pointer = intvar; |
| 03:47.26 | [TK]D-Fender | pdd_, What location are you calling? |
| 03:47.30 | pdd_ | PSTN... our customer are supposed to receive the call |
| 03:47.44 | [TK]D-Fender | pdd_, PSTN is a medium, not a location :) |
| 03:48.09 | [TK]D-Fender | pdd_, Heck, not even a MEDIUM per-se! |
| 03:48.14 | pdd_ | this is for a web site that sends a confirmation number to your phone to finalize your site registration |
| 03:48.36 | pdd_ | so you register... then get a call with a number that you need to enter into the web site |
| 03:48.37 | [TK]D-Fender | pdd_, Lemme simplify this for you a bit more... calling ANTARCTICA? |
| 03:48.50 | pdd_ | us customer :-) |
| 03:48.58 | pdd_ | customers |
| 03:48.59 | [TK]D-Fender | pdd_, U-S-A? |
| 03:49.06 | pdd_ | the good old USA |
| 03:49.19 | pdd_ | we are located in CA |
| 03:49.22 | [TK]D-Fender | pdd_, Good... you're right, that wasn't like squeezing blood from a stone... |
| 03:49.24 | *** join/#asterisk techie (n=techie@adsl-76-214-21-156.dsl.lsan03.sbcglobal.net) |
| 03:49.28 | [TK]D-Fender | ~itsp-us |
| 03:49.46 | [TK]D-Fender | ~itsp |
| 03:49.47 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 03:49.55 | [TK]D-Fender | ~itsplist-us |
| 03:49.56 | 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 |
| 03:50.01 | [TK]D-Fender | there |
| 03:50.18 | pdd_ | great |
| 03:50.19 | [TK]D-Fender | forgot the new bot codes.. |
| 03:50.49 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 03:51.32 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:54.12 | koszik | MrTelephone: yes, it will work |
| 03:57.02 | MrTelephone | i added a debug line in there |
| 03:57.21 | MrTelephone | I set the *start pointer correctly? |
| 03:58.14 | MrTelephone | this is the first time i got this error though.. wtf |
| 03:58.15 | MrTelephone | configure: error: installation or configuration problem: C compiler cannot create executables. |
| 03:58.58 | *** join/#asterisk Odie_flocon (n=odie@S0106000f3d5eae81.cg.shawcable.net) |
| 04:00.37 | Odie_flocon | Hey all. |
| 04:01.32 | *** join/#asterisk rasix (n=rasix@ip70-110.cbn.net.id) |
| 04:02.47 | MrTelephone | if its not one thing its another |
| 04:02.50 | MrTelephone | damn |
| 04:10.53 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 04:15.36 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-183-43.dsl.irvnca.pacbell.net) |
| 04:15.49 | rasix | hi all... how do we know that realtime is enabled or not within asterisk? |
| 04:19.11 | MrTelephone | whats asterisk port 4520 |
| 04:21.39 | Nugget | that's the port we use to call sex hotlines using your phone line. |
| 04:21.57 | km- | IN YER ASTERIX, SEXXORIN YOUR GF. |
| 04:21.59 | km- | lawlz. |
| 04:23.04 | *** join/#asterisk Strom_C (n=strom@m0a0e36d0.tmodns.net) |
| 04:23.16 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 04:29.19 | MrTelephone | crap |
| 04:29.29 | MrTelephone | my longdistance bill was 500 bucks |
| 04:36.57 | [TK]D-Fender | MrTelephone, .... was it good for you? |
| 04:37.19 | JayTee52 | lol |
| 04:43.20 | *** join/#asterisk techie (n=techie@adsl-76-214-21-156.dsl.lsan03.sbcglobal.net) |
| 04:45.37 | MrTelephone | yeah not too bad i guess |
| 04:45.38 | *** join/#asterisk saftsack (n=saftsack@pD9E0402C.dip.t-dialin.net) |
| 04:46.35 | MrTelephone | my girlfriend isn't into asterisk km- |
| 04:47.44 | [TK]D-Fender | MrTelephone, Yeah.. you wan't someone who'll got the extra mile..... not just 0.625 miles ;) |
| 04:48.10 | MrTelephone | :P |
| 04:48.42 | MrTelephone | hey fender have you seen this error much? 481 "Call Leg/Transaction Does Not Exist" |
| 04:49.37 | [TK]D-Fender | MrTelephone, now and again yeah... no clue why |
| 04:51.10 | MrTelephone | why you were writin that i found a good explanation |
| 04:51.26 | MrTelephone | UA got a CANCEL or BYE for a call that didn't exist |
| 04:58.44 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 05:00.39 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 05:00.52 | MrTelephone | fender you lost your route |
| 05:06.16 | *** join/#asterisk AnYoNeSNM (i=ircap8@190.173.217.135) |
| 05:06.21 | AnYoNeSNM | hello |
| 05:06.34 | AnYoNeSNM | heelloo |
| 05:08.18 | AnYoNeSNM | animore |
| 05:08.19 | AnYoNeSNM | :P |
| 05:12.33 | MrTelephone | debian testing is worst then vista |
| 05:12.56 | AnYoNeSNM | [MrTelephone] hello |
| 05:13.18 | MrTelephone | hi |
| 05:13.32 | AnYoNeSNM | the questions.. |
| 05:13.36 | AnYoNeSNM | you used asterisk ? |
| 05:14.28 | MrTelephone | i used it once |
| 05:15.05 | AnYoNeSNM | ah oki.. |
| 05:15.18 | AnYoNeSNM | used cards pbx or pbo ? |
| 05:16.21 | AnYoNeSNM | you where from ? |
| 05:16.34 | AnYoNeSNM | where you from.. |
| 05:16.42 | AnYoNeSNM | my inglish es bad............. |
| 05:21.54 | *** part/#asterisk AnYoNeSNM (i=ircap8@190.173.217.135) |
| 05:23.50 | MrTelephone | yeah |
| 05:23.52 | MrTelephone | oops |
| 05:23.52 | *** join/#asterisk oej (n=olle@85.19.87.179) |
| 05:23.57 | MrTelephone | i wasn't fast enough |
| 05:24.04 | MrTelephone | sorry im re inventing chan_sip.c at the moment |
| 05:26.26 | linagee | is it possible that XO and AT&T are having routing problems on the same day? (today) |
| 05:27.15 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 05:27.16 | linagee | is there a forum for that? internet downtime dot com or something? |
| 05:34.36 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 05:43.49 | *** part/#asterisk dr1 (n=dr1@adsl-70-135-219-177.dsl.mrdnct.sbcglobal.net) |
| 05:45.57 | De_Mon | linagee internethealthreport.com or somesuch |
| 05:46.20 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 05:46.20 | *** mode/#asterisk [+o mog] by ChanServ |
| 05:46.36 | De_Mon | looks like qwest is having issues, but thats about it |
| 05:47.41 | De_Mon | MrTelephone yea don't do testing, unstable or stable, not testing -- when it breaks it stays broke |
| 05:50.55 | MrTelephone | i just made some changes to chan_sip and i screwed everything up |
| 05:51.06 | De_Mon | chan_sip is a mess |
| 05:52.28 | MrTelephone | can you c program? |
| 05:56.53 | MrTelephone | what do you make of this |
| 05:56.54 | MrTelephone | <PROTECTED> |
| 05:56.54 | MrTelephone | <PROTECTED> |
| 05:56.54 | MrTelephone | <PROTECTED> |
| 05:56.54 | MrTelephone | <PROTECTED> |
| 05:56.54 | MrTelephone | <PROTECTED> |
| 05:56.56 | MrTelephone | <PROTECTED> |
| 05:58.08 | MrTelephone | if Authorization = name then loop through headers from start value.. if the header begins with name then change start to current x value |
| 05:58.50 | JT | ~pb |
| 05:58.51 | 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 |
| 06:02.10 | MrTelephone | sorry noone was speaking and I had to keep the channel quota going |
| 06:06.40 | *** join/#asterisk watchy (n=watchy@h200.176.255.206.cable.cmdn.cablelynx.com) |
| 06:06.45 | watchy | hey tk you there? |
| 06:07.10 | watchy | anyone recommend isymphony? |
| 06:10.17 | MrTelephone | hard to test this without screwin up crap hardcore |
| 06:11.12 | *** join/#asterisk murdmath (n=vircuser@c-24-10-190-87.hsd1.co.comcast.net) |
| 06:37.51 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 06:48.34 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 06:53.12 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:53.17 | rasix | anyone here can help me how to enable realtime asterisk through mysql.. i'm newbie on asterisk |
| 06:53.24 | rasix | i did everthing on the wiki |
| 06:53.34 | rasix | but it doesnt work |
| 06:53.39 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:59.16 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 06:59.20 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4f41703ab8c1ec19) |
| 07:01.04 | *** join/#asterisk Somebee (n=Somebee@80.232.5.97) |
| 07:04.02 | *** part/#asterisk rasix (n=rasix@ip70-110.cbn.net.id) |
| 07:04.11 | MrTelephone | finally |
| 07:04.13 | MrTelephone | yay |
| 07:04.24 | pdd_ | great :-) |
| 07:06.42 | MrTelephone | i got my little program working but it debug outputs everytime there is a REGISTER message |
| 07:06.46 | MrTelephone | that wasn't pretty |
| 07:13.31 | MrTelephone | strcasecmp() returns 0 if there is a match.. it throws you off when you see if(!strcasecmp()) because you think.. if there isn't a match do if statement |
| 07:17.30 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
| 07:18.09 | *** join/#asterisk nirz (n=nnscript@bzq-79-181-140-77.red.bezeqint.net) |
| 07:21.47 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 07:27.54 | *** join/#asterisk fukz (n=johns@p4FC2E889.dip.t-dialin.net) |
| 07:29.04 | *** join/#asterisk lesterc (n=lesterc@158.27.96.58.exetel.com.au) |
| 07:30.50 | *** join/#asterisk suvir (n=suvir@ppp-124.120.245.212.revip2.asianet.co.th) |
| 07:38.56 | *** join/#asterisk suvir (n=suvir@ppp-124.120.245.212.revip2.asianet.co.th) |
| 07:45.00 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 07:45.00 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:46.16 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:47.43 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:52.32 | *** join/#asterisk cypherdelic (n=cypher@p5B27DE4F.dip.t-dialin.net) |
| 07:53.36 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581834.dsl.bell.ca) |
| 07:55.50 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 08:04.30 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:07.12 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 08:09.30 | *** join/#asterisk samdell33 (n=antispam@nap-mt2a.airnet.net.nz) |
| 08:11.06 | samdell33 | #11423 - just me, or am I going mad ? |
| 08:13.21 | yangvnc | Is there a manual on how to connect my ekiga client to the asterisk, which is linked to a VOIP provider with the SIP account ? |
| 08:13.50 | yangvnc | I get authorisation failed if I want to use the same user/pass |
| 08:16.18 | yangvnc | <PROTECTED> |
| 08:30.47 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:32.17 | dandre | Hello, |
| 08:34.26 | dandre | Is there any alternative syntax for zapata.conf? For instance, intead of: |
| 08:34.27 | dandre | [channels] |
| 08:34.27 | dandre | ... |
| 08:34.27 | dandre | channel => 1 |
| 08:34.27 | dandre | write something like: |
| 08:34.27 | dandre | [channels] |
| 08:34.29 | dandre | Defaults go here |
| 08:34.31 | dandre | [1] |
| 08:34.33 | dandre | ... |
| 08:37.55 | dandre | anyone ? |
| 08:40.09 | mort_gib | Dandre: Why?? |
| 08:42.24 | dandre | I find it more convenient and as all the other config files are used this way, it is easier for my config tool to handle zapata.conf that way |
| 08:43.07 | mort_gib | Dandre good point, but then sip.conf and extensions.conf are different too -right?? |
| 08:45.10 | *** join/#asterisk Somebee (n=Somebee@80.232.5.97) |
| 08:45.44 | dandre | not really, in those files, there are global (default) serctions and items description section. Not in zapata.conf where the "item section" is defined by channel =>... |
| 08:45.44 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 08:48.45 | dandre | and what really mess up is that definition for on channel depends on the previous channel definition. This is not error prone or consistent |
| 08:53.49 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 08:55.37 | mort_gib | Dandre: I just started with * |
| 08:55.49 | dandre | ok |
| 08:56.12 | mort_gib | So I also find the conf files somewhat inconsistent |
| 08:56.49 | mort_gib | Top down execution is not what I normally see :-) |
| 09:00.55 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:05.24 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 09:13.32 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 09:15.22 | *** join/#asterisk Somebee (n=Somebee@80.232.5.97) |
| 09:18.14 | *** join/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 09:18.57 | *** join/#asterisk slavon_net (n=slavon@slavon.bigtelecom.ru) |
| 09:19.00 | slavon_net | hello |
| 09:19.16 | slavon_net | how to simple check in AEL that exten in context is exists? |
| 09:19.31 | slavon_net | that function may do that? |
| 09:19.40 | slavon_net | i not like use IsChanAvay |
| 09:22.15 | slavon_net | anyone? |
| 09:23.46 | *** join/#asterisk guomi (n=francois@c2cpc3.camptocamp.com) |
| 09:26.27 | *** join/#asterisk qdk (n=qdk@193.164.155.7) |
| 09:26.40 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 09:30.30 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:32.18 | *** join/#asterisk currach (n=currach@194.125.21.98) |
| 09:38.16 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4f41703ab8c1ec19) |
| 09:38.47 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:49.37 | R1ck | does AsteriskNOW support BRI cards? |
| 09:50.16 | *** join/#asterisk leikh (n=chatzill@port162.ds1-amb.adsl.cybercity.dk) |
| 09:51.37 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 09:52.11 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 09:54.16 | leikh | My asterisk is set up as a sip-client (just to receive voicemail) behind nat. After approx 15 minutes, my asterisk can't receive incoming calls (but sip show registry says it's connected). If I do a reload, everything is fine... Is it a problem with keep-alive? |
| 09:55.36 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 09:56.00 | fukz | leikh, Maybe it is a problem with your firewall/nat router. Try to use portforwarding. |
| 09:58.47 | leikh | fukz: If i do a reload every 15 minutes, it's ok, so i guess it has to be an asterisk issue... |
| 10:00.01 | leikh | refresh is 105 and state is registered |
| 10:01.23 | leikh | what's this? pbx_dundi.c:451 reset_global_eid: No ethernet interface found for seeding global EID You will have to set it manually. |
| 10:12.04 | *** join/#asterisk tzafrir_laptop (n=tzafrir@89.130.6.136) |
| 10:16.06 | kaldemar | leikh: dundi takes its entity id from the first ethernet interface it finds when it's not set in dundi.conf. for some reason it can't get it automatically and it tells you to edit your dundi.conf. |
| 10:17.15 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 10:17.31 | kaldemar | leikh: if you're not using dundi, disable it. |
| 10:19.47 | leikh | kaldemar: how do I disable it? |
| 10:21.37 | leikh | kaldemar: just comment it out from extensions.conf? |
| 10:21.39 | kaldemar | leikh: put "noload => pbx_dundi.so" in /etc/asterisk/modules.conf |
| 10:22.24 | kaldemar | extensions.conf is for configuring dialplan, it doesn't affect any modules getting loaded. |
| 10:22.28 | leikh | kaldemar: ok, thx |
| 10:31.38 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 10:36.26 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
| 10:38.52 | R1ck | does AsteriskNOW support BRI cards? |
| 10:40.05 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 10:41.21 | tzafrir_laptop | R1ck, Only misdn ones, AFAIK |
| 10:43.17 | *** join/#asterisk ming_zym (n=ming_zym@124.14.236.162) |
| 10:44.43 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 10:49.36 | *** join/#asterisk bantu (n=Miranda@rz-du-ubx-140-93.rz.uni-karlsruhe.de) |
| 10:50.27 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:55.21 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:57.22 | *** join/#asterisk qdk (n=qdk@213.150.62.99) |
| 11:00.24 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 11:00.45 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 11:03.57 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:04.21 | *** join/#asterisk dijungal (n=kdaniel@209.59.110.36) |
| 11:04.25 | dijungal | hello |
| 11:04.35 | dijungal | how do you remove asterisk 1.2 to upgrade to 1.4 ? |
| 11:11.09 | harpal | dijungal, no need to remove old asterisk Just install new asterisk and read that upgrade text file before upgrading |
| 11:11.50 | dijungal | hmmm |
| 11:12.06 | dijungal | right... the new files from the new asterisk should overwrite the old ones |
| 11:14.22 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 11:17.06 | *** join/#asterisk bantu (n=Miranda@rz-du-ubx-140-93.rz.uni-karlsruhe.de) |
| 11:20.39 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 11:20.42 | msetim | hi guys |
| 11:22.09 | msetim | how can I make recording a queue agente on-the-fly. I would like to set a recording to a specif agent while he is speaking |
| 11:23.05 | slavon_net | how to simple check in AEL that exten in context is exists? |
| 11:30.56 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 11:31.30 | *** join/#asterisk fukz (n=johns@p4FC2E65A.dip.t-dialin.net) |
| 11:33.27 | *** join/#asterisk currach (n=currach@194.125.21.98) |
| 11:38.56 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 11:55.21 | *** join/#asterisk billybongo (n=rich@82.153.23.79) |
| 11:57.43 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 12:00.33 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 12:02.39 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 12:08.51 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 12:12.33 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 12:17.32 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 12:21.04 | *** join/#asterisk qdk (n=qdk@193.164.155.7) |
| 12:24.30 | *** join/#asterisk TUNTAP (n=error@BSN-77-58-164.dsl.siol.net) |
| 12:32.15 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 12:32.27 | flujan | hi guys... |
| 12:32.41 | flujan | some sip peers of my pbx have the call-limit flag set to 1 |
| 12:32.57 | flujan | sometimes I it locks the users preventing them to place calls... |
| 12:33.17 | flujan | Even if I restart the computer the user still could not place calls . |
| 12:33.23 | flujan | any ideas? |
| 12:43.15 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 12:45.51 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) [NETSPLIT VICTIM] |
| 12:45.51 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) [NETSPLIT VICTIM] |
| 12:45.51 | *** join/#asterisk qdk (n=qdk@193.164.155.7) [NETSPLIT VICTIM] |
| 12:45.51 | *** join/#asterisk currach (n=currach@194.125.21.98) [NETSPLIT VICTIM] |
| 12:45.51 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) [NETSPLIT VICTIM] |
| 12:45.51 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk atisss (n=atisss@193.238.212.171) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk mbranca (i=daemon@mi-gw1.voismart.net) |
| 12:45.52 | *** join/#asterisk gardo (n=gardo@121.97.108.105) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 12:45.52 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk linuxmaniac (n=linuxman@118.Red-88-27-159.staticIP.rima-tde.net) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 12:45.52 | *** join/#asterisk cy3o3 (n=cy@is.trapped.in.themetaverse.org) |
| 12:45.52 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 12:45.52 | *** join/#asterisk ViKing78 (n=ViKing78@cerberus.franklinamerican.com) [NETSPLIT VICTIM] |
| 12:45.52 | *** join/#asterisk stephbul (n=stephbul@ks33394.kimsufi.com) [NETSPLIT VICTIM] |
| 12:45.53 | *** join/#asterisk js_ (i=js@194.17.31.204) [NETSPLIT VICTIM] |
| 12:45.53 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 12:45.54 | *** join/#asterisk Haq (n=liquid@6a.ab.5446.static.theplanet.com) [NETSPLIT VICTIM] |
| 12:45.54 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 12:45.54 | *** join/#asterisk orcimrepus (n=orcimrep@74-130-123-207.dhcp.insightbb.com) [NETSPLIT VICTIM] |
| 12:45.54 | *** join/#asterisk _mm_ (n=mmclain@cpe-75-80-238-180.dc.res.rr.com) [NETSPLIT VICTIM] |
| 12:45.54 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) [NETSPLIT VICTIM] |
| 12:45.54 | *** join/#asterisk Corydon76-home (i=six@pdpc/supporter/bronze/Corydon76-home) |
| 12:45.54 | *** mode/#asterisk [+oooo twisted codefreeze angler Corydon76-home] by irc.freenode.net |
| 12:45.59 | *** join/#asterisk billybongo (n=rich@82.153.23.79) [NETSPLIT VICTIM] |
| 12:45.59 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) [NETSPLIT VICTIM] |
| 12:45.59 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk PepOSX (n=pepOSX@190.72.147.76) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk hwt (i=hwt@195.139.204.163) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk smithj (n=smithj@foresight/developer/smithj) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk MihiNomenEst (i=b7Fi@66.255.220.17) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk De_Mon (i=de_mon@fl-71-52-101-157.dhcp.embarqhsd.net) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk variable_office (n=variable@208.73.60.2) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk JackEStorm (n=no@ip68-225-77-136.no.no.cox.net) [NETSPLIT VICTIM] |
| 12:46.00 | *** join/#asterisk zirman (i=zirman@ip194.207.107.216.seg.net) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk _ys (i=yuri@91.151.196.254) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk guomi (n=francois@c2cpc3.camptocamp.com) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk wothinn (n=Allfathe@vs1.svartalfheim.net) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk hypa7ia (i=hypatia@judecca.aculei.net) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk thugwh0re (i=plutten@pimping.at.manwh0re.se) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk Siya (n=djerk@217-195-248-251.dsl.easynet.nl) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk mace (n=mace@debian/developer/mace) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk tc3driver (n=huh@rrcs-24-199-16-118.west.biz.rr.com) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk Kobaz (i=kobaz@its.kobaz.net) [NETSPLIT VICTIM] |
| 12:46.02 | *** join/#asterisk eonz (n=Icarus@irc.americatelnet.com.pe) [NETSPLIT VICTIM] |
| 12:47.14 | *** join/#asterisk linxroute (n=linxrout@125.235.0.122) |
| 12:53.13 | *** join/#asterisk saftsack (n=saftsack@pD9E0402C.dip.t-dialin.net) |
| 12:57.12 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 12:58.13 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 13:04.07 | *** join/#asterisk ming_zym (n=ming_zym@124.14.236.162) |
| 13:07.08 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 13:10.31 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 13:13.02 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) [NETSPLIT VICTIM] |
| 13:13.02 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) [NETSPLIT VICTIM] |
| 13:13.02 | *** join/#asterisk ming_zym (n=ming_zym@124.14.236.162) [NETSPLIT VICTIM] |
| 13:13.02 | *** join/#asterisk saftsack (n=saftsack@pD9E0402C.dip.t-dialin.net) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk currach (n=currach@194.125.21.98) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk atisss (n=atisss@193.238.212.171) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk mbranca (i=daemon@mi-gw1.voismart.net) |
| 13:13.03 | *** join/#asterisk gardo (n=gardo@121.97.108.105) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 13:13.03 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk linuxmaniac (n=linuxman@118.Red-88-27-159.staticIP.rima-tde.net) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM] |
| 13:13.03 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 13:13.03 | *** join/#asterisk cy3o3 (n=cy@is.trapped.in.themetaverse.org) |
| 13:13.03 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 13:13.04 | *** join/#asterisk ViKing78 (n=ViKing78@cerberus.franklinamerican.com) [NETSPLIT VICTIM] |
| 13:13.04 | *** join/#asterisk stephbul (n=stephbul@ks33394.kimsufi.com) [NETSPLIT VICTIM] |
| 13:13.05 | *** join/#asterisk js_ (i=js@194.17.31.204) [NETSPLIT VICTIM] |
| 13:13.05 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 13:13.05 | *** join/#asterisk Haq (n=liquid@6a.ab.5446.static.theplanet.com) [NETSPLIT VICTIM] |
| 13:13.05 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 13:13.05 | *** join/#asterisk orcimrepus (n=orcimrep@74-130-123-207.dhcp.insightbb.com) [NETSPLIT VICTIM] |
| 13:13.05 | *** join/#asterisk _mm_ (n=mmclain@cpe-75-80-238-180.dc.res.rr.com) [NETSPLIT VICTIM] |
| 13:13.05 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) [NETSPLIT VICTIM] |
| 13:13.05 | *** join/#asterisk Corydon76-home (i=six@pdpc/supporter/bronze/Corydon76-home) |
| 13:13.05 | *** mode/#asterisk [+oooo twisted codefreeze angler Corydon76-home] by irc.freenode.net |
| 13:13.09 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:13.57 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) [NETSPLIT VICTIM] |
| 13:15.23 | *** join/#asterisk saftsack (n=saftsack@pD9E0402C.dip.t-dialin.net) |
| 13:16.57 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:17.19 | flujan | hey guys.. someone alive here? |
| 13:18.50 | *** join/#asterisk saftsack (n=saftsack@pD9E0402C.dip.t-dialin.net) |
| 13:18.53 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
| 13:20.49 | cpm | all are dead. Welcome to the chamber of the zombie, you see a light off to your left |
| 13:21.24 | lirakis | yeah |
| 13:22.48 | [TK]D-Fender | cpm: left ; left ; talk orc |
| 13:23.07 | cpm | :P |
| 13:24.05 | cpm | hey [TK]D-Fender, I have a bunch of Polycom Viewstations here at work. Obsolete stock, know anyone who may be interested? |
| 13:24.38 | [TK]D-Fender | What models? |
| 13:24.54 | cpm | I'm gonna go get a list right now. I'll pastebin |
| 13:25.09 | [TK]D-Fender | cpm: sure |
| 13:25.20 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 13:28.54 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 13:33.20 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 13:35.00 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 13:37.03 | flujan | cpm: don't die guys.. :D |
| 13:37.12 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 13:37.12 | flujan | [TK]D-Fender: hi TK.... how are you doing? |
| 13:37.15 | lirakis | cpm: |
| 13:37.21 | lirakis | cpm: i might be interested |
| 13:37.23 | nicox | Hallo, ist jemand aus Österreich hier? |
| 13:37.47 | [TK]D-Fender | flujan: meh... |
| 13:37.58 | flujan | guys are you experiencing random crashs with asterisk 1.4.14. |
| 13:38.24 | flujan | I am having a lot of crashes with this version.... |
| 13:38.41 | flujan | sip peers been locked because of call-limit |
| 13:38.49 | flujan | an so forth.. |
| 13:39.00 | flujan | 1.4.15 does not solve it too.. |
| 13:39.13 | lirakis | cpm: i am particularly looking for a few pvs-1419's |
| 13:39.14 | flujan | I switch back to a svn version before 1.4.14 release. :D |
| 13:40.23 | cpm | [TK]D-Fender, http://pastebin.ca/802609 |
| 13:40.50 | cpm | lirakis, same same |
| 13:41.18 | *** join/#asterisk masus (n=ethemc@88.248.14.186) |
| 13:41.23 | cpm | There is also one large-ish box labeled Polycom that I didn't open. |
| 13:41.49 | *** part/#asterisk masus (n=ethemc@88.248.14.186) |
| 13:41.51 | lirakis | cpm: .. so are these the 14XX style? |
| 13:42.24 | *** join/#asterisk anujsingh (n=Administ@59.90.65.14) |
| 13:42.26 | cpm | lirakis, yes |
| 13:42.30 | lirakis | ah cool |
| 13:42.54 | cpm | 1419s are just the isdn dongles, yes? |
| 13:42.55 | anujsingh | hi |
| 13:43.44 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 13:44.10 | anujsingh | i configured two asterisk servers , want to share dialplan so that users of nodeA can call nodeB , |
| 13:44.29 | anujsingh | vice versa |
| 13:45.15 | anujsingh | i creates users in iax.conf registering each other aster servers. |
| 13:46.00 | anujsingh | but unable to call, from users of nodeA to users of nodeB |
| 13:46.44 | anujsingh | please give me the link for a tutorial so that i can share two dial plans |
| 13:49.01 | [TK]D-Fender | anujsingh: lookup "asterisk dual servers" on the WIKI |
| 13:49.03 | [TK]D-Fender | ~wikis |
| 13:49.04 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 13:56.21 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 14:02.22 | *** join/#asterisk currach (n=currach@194.125.21.98) |
| 14:06.54 | anujsingh | Thanks [TK]D-Fender |
| 14:07.12 | anujsingh | Does astguiclient works in autodial mode ? |
| 14:07.22 | cpm | lirakis, you gonna just use'em in house? |
| 14:07.39 | lirakis | cpm: yes .. just in house |
| 14:07.42 | cpm | k |
| 14:07.44 | anujsingh | i am using astguiclient , and manual dialing working fine, but autodial is not working |
| 14:08.06 | lirakis | cpm: weve got one here already and like it.. and we want to send 2 out to our offices elsewhere. |
| 14:08.58 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:08.58 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:12.18 | lirakis | has a lot changed for queues in 1.4? |
| 14:12.52 | *** part/#asterisk roe_ (n=roe___@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 14:14.01 | afritz | morning |
| 14:15.17 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:15.44 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 14:16.12 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:21.03 | anujsingh | my astguiclient is not working with autodial mode, with manual dialing it is working smoothly |
| 14:21.35 | anujsingh | what can be the cause |
| 14:26.45 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.6) |
| 14:26.47 | *** join/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 14:29.01 | rantsh | Hi guys, |
| 14:29.07 | rantsh | how's everyone doing? |
| 14:30.11 | rantsh | I have a fast and easy question, if using asterisk as a SIP b2bua can you increase (or decrease for tha matter) the volume of the sound? |
| 14:30.38 | rantsh | I know it's possible for zapata, no clue for a sip channel (though I'm guessing you can) |
| 14:31.45 | [TK]D-Fender | rantsh: Currently no. |
| 14:32.23 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:32.23 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:32.34 | dandre | hello |
| 14:33.14 | dandre | in * 1.4 how many channel groups can I have? 32 or 64 ? |
| 14:33.55 | rantsh | [TK]D-Fender, thanks... I tought you could as a transcoding kind of thing.... guess it'll come on a future release (maybe 1.6?)... anyway, thank you dude |
| 14:34.03 | [TK]D-Fender | dandre: How many do you think you'll need? |
| 14:35.00 | [TK]D-Fender | rantsh: 1.6 is supposed to have an entirely new bridging scheme where you can do dynamic conferencing, spying, DSP-stuff, etc. So I'm realtively certain that a volume tweaker would be one of the first applications of this. |
| 14:35.13 | dandre | I don't know but it could be over 32 |
| 14:35.42 | [TK]D-Fender | dandre: How many PRI's you looking to split? |
| 14:37.22 | file | it's already there, func_volume |
| 14:37.48 | cpm | lirakis, I'm pretty sure I can check out two of those units, the FX and one of the others, and make sure they work and can talk to each other. There are some 'howevers' |
| 14:37.58 | dandre | has group anything to do with phone interfaces or are they used only for trunking? |
| 14:38.11 | [TK]D-Fender | file: in trunk? |
| 14:38.40 | [TK]D-Fender | dandre: the only "group" is for zaptel channels |
| 14:38.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 14:39.01 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 14:39.18 | lirakis | cpm: okay.. well we arent looking for anything rediculous.. |
| 14:39.20 | blitzrage | isn't func_volume in 1.4? or am I on crack? |
| 14:39.27 | lirakis | err ridiculous |
| 14:39.27 | dandre | yes I know |
| 14:39.29 | [TK]D-Fender | dandre: in that sense. What exactly are you looking to "group", and what actions are you planning on doing to it? |
| 14:39.30 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-5004e68593727285) |
| 14:39.31 | cpm | lirakis, look me up next week. |
| 14:39.37 | lirakis | cpm: as long as they 'work' |
| 14:39.41 | lirakis | cpm: will do |
| 14:39.42 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:39.45 | cpm | okay, the however's. |
| 14:40.00 | cpm | if you are a 'polycom' person, this will be like christmas. |
| 14:40.07 | cpm | if you aren't. you'll be fsck'd |
| 14:40.22 | cpm | reason, these are just the settop box, and a power supply. No remote. |
| 14:40.25 | cpm | no other stuff |
| 14:40.39 | cpm | and, if you worked with viewstations, you can't really do much without the remote |
| 14:40.40 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 14:40.59 | cpm | however, if you can get yer hands on a remote easily enough, then you'll get the deal of the day. :) |
| 14:43.19 | dandre | I must set a limit in group definition for my config tool so that the user cant define more than x groups. I just wandered to know what to use. 32 as stated in voipinfo or 64 as I have read somewhere |
| 14:45.43 | [TK]D-Fender | dandre: Again you aren't specifying what KIND of groups. Call group, pickupgroup, zaptel channel group, etc. |
| 14:45.51 | [TK]D-Fender | dandre: they are not all the same max. |
| 14:46.36 | jer | if i've got a usb system on irq 18, and an x100p on irq 18, nothing is used on that particular usb bus, should there be a problem with getting that card loaded? (just a home system here, so i don't need a lecture on the x100p =]) |
| 14:47.20 | dandre | so could you tell me the size of each. |
| 14:47.20 | dandre | for the moment I need zaptel channel group but in the future I will need the others |
| 14:47.43 | file | [TK]D-Fender: yes |
| 14:48.30 | [TK]D-Fender | dandre: this is all ont he WIKI, and for whatever you want to double-check jsut go see what all the other GUI's are doing. |
| 14:49.51 | dandre | in the wiki all the max are set to 32 |
| 14:50.31 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 14:50.34 | [TK]D-Fender | jer: disable the USB in your BIOS. |
| 14:50.49 | jer | [TK]D-Fender, hard to do from here, but i'll try that when i get home |
| 14:51.09 | [TK]D-Fender | jer: Never ask for help when you can't see it through :) |
| 14:51.19 | jer | well, i have access to the box just not the bios =] |
| 14:51.24 | jer | and my apologies |
| 14:51.43 | [TK]D-Fender | jer: Well.... are you experiencing problems with your card right now? |
| 14:52.13 | jer | [TK]D-Fender, yeah, the driver won't attach. ztcfg fails with a "Device not configured" error (meaning the driver isn't attached) |
| 14:52.40 | jer | "device_attach: wcfxo0 attach returned 5" in the kernel logs |
| 14:52.42 | [TK]D-Fender | jer: do you see the module loaded? |
| 14:53.01 | jer | yeah |
| 14:53.18 | [TK]D-Fender | jer: Hrm.... what kind of system? |
| 14:53.37 | jer | [TK]D-Fender, freebsd 6.2 OS, little core 2 duo system, nothing fancy |
| 14:53.53 | jer | linux wouldn't recognize either the wireless card in it or the network card so, i went with something else |
| 14:55.03 | _x86_ | asterisk was designed from the ground up to be run on linux |
| 14:55.38 | _x86_ | it's been reported to gimp along on macosx, freebsd, etc, but it will always work best on linux |
| 14:55.39 | jer | _x86_, i know; but i don't have a problem with asterisk, it's working fine calling one sip phone to another sip phone, and calling out over a sip trunk works; i just want to add the zap channel |
| 14:55.50 | _x86_ | right |
| 14:56.02 | _x86_ | zap kernel drivers are actively supported on linux |
| 14:56.04 | [TK]D-Fender | jer: Zaptel on FBSD too? You ARE a masochist... |
| 14:56.09 | jer | [TK]D-Fender, =] |
| 14:56.12 | _x86_ | yeah good luck |
| 14:56.21 | _x86_ | better install linux and get your network card working ;) |
| 14:56.40 | jer | well, if i have to, i'll cross compile a linux kernel on freebsd with the driver for the realtek 8101e built into it so i can go from there |
| 14:56.57 | _x86_ | system calls wont be the same |
| 14:57.03 | _x86_ | doubt it will work properly |
| 14:57.05 | jer | but first i need to find one (haven't looked horribly hard, i'm sure one exists that doesn't depend on ndis) |
| 14:57.23 | _x86_ | what card do you have? |
| 14:57.23 | jer | _x86_, well, it's an experiment. if it fails, i'll just throw more hardware at it (something i know is supported in linux) |
| 14:57.36 | jer | _x86_, the network card is a realtek 8101e |
| 15:01.31 | _x86_ | 8139too wont cut it? |
| 15:01.55 | _x86_ | every realtek nic i've ever used worked fine with that driver |
| 15:05.21 | mort_gib | _X86_:That just gave away your age! |
| 15:06.23 | jer | _x86_, 8139too doesn't support the 8101e (pci-express) |
| 15:06.41 | jer | i tried all of them before i gave up on linux |
| 15:06.42 | jer | =] |
| 15:09.16 | flujan | [TK]D-Fender: do you how can I check the locked sip channels? I have call-limit 1and some phones keep with the counter to 1 even if I restart the machine or reload sip on the asterisk. |
| 15:09.46 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 15:10.24 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:10.59 | [TK]D-Fender | flujan: AstDB <- |
| 15:13.07 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:15.00 | *** part/#asterisk simond (n=simon@208.68.95.5) |
| 15:17.14 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 15:17.28 | teknoprep | hey all |
| 15:17.34 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-57-3.dsl.tul2ok.sbcglobal.net) |
| 15:22.49 | flujan | [TK]D-Fender: I search documentation and found nothing about how can I check it on astdb... |
| 15:22.55 | flujan | could you give any hint? |
| 15:23.07 | [TK]D-Fender | flujan: database show |
| 15:23.25 | *** join/#asterisk SirWhit (n=sirjames@blk-222-38-6.eastlink.ca) |
| 15:23.34 | teknoprep | [TK]D-Fender, i find 1.4.* asterisk is really much better for voip channels |
| 15:23.37 | teknoprep | over the inet |
| 15:23.40 | teknoprep | why is this? |
| 15:24.00 | [TK]D-Fender | teknoprep: Where do I begin with such an unqualified statement.... hmmm... |
| 15:24.22 | flujan | [TK]D-Fender: thanks... it shows all my sip peers |
| 15:24.29 | teknoprep | lol |
| 15:24.31 | *** join/#asterisk Somebee (n=Somebee@80.232.5.97) |
| 15:24.48 | teknoprep | well our voip sounds better since i have upgraded to 1.4.13 |
| 15:24.54 | teknoprep | from 1.2. |
| 15:25.15 | [TK]D-Fender | teknoprep: ......... |
| 15:25.22 | teknoprep | ..... |
| 15:25.34 | [TK]D-Fender | teknoprep: Could you provide a little LESS detail please? |
| 15:25.36 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:25.53 | teknoprep | ok... we don't get delays or choppyness anymore |
| 15:25.57 | teknoprep | ever |
| 15:25.58 | teknoprep | not one |
| 15:26.05 | teknoprep | its like a perfect phone call everytime |
| 15:26.27 | teknoprep | was something done different with SIP ? jitter buffering ? |
| 15:26.52 | flujan | [TK]D-Fender: http://pastie.caboo.se/123743 |
| 15:26.53 | [TK]D-Fender | teknoprep: Oh wow, I even know what protocol for call setup! Yippy! |
| 15:27.13 | teknoprep | www.bandwidth.com |
| 15:27.18 | teknoprep | sip termination |
| 15:27.45 | SirWhit | is there a way to know if my digium card is broken? |
| 15:28.11 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:28.22 | hmmhesays | is it not completing calls? |
| 15:28.44 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 15:28.54 | flujan | [TK]D-Fender: peers that already log out are still on the database... |
| 15:28.57 | SirWhit | I had it hooked up to a D4 AMI T1 for the past couple weeks and now it doesn't want to connect to it |
| 15:29.02 | flujan | here is how looks like a peer configuration |
| 15:29.13 | SirWhit | it goes into alarm then out then in then out then... |
| 15:29.41 | SirWhit | I hooked the same T1 into a differnt pbx and it works fine.. I just am not sure how to troubleshoot |
| 15:30.18 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 15:30.29 | [TK]D-Fender | flujan: Not sure why but I somehow suspect your pastebin isn't "complete" |
| 15:30.58 | *** part/#asterisk bkw_ (n=brian@adsl-70-142-57-3.dsl.tul2ok.sbcglobal.net) |
| 15:31.42 | *** join/#asterisk epaulin (n=epaulin@61.144.111.197) |
| 15:33.05 | flujan | [TK]D-Fender: full pastie |
| 15:33.06 | flujan | http://pastebin.com/m6431643e |
| 15:33.40 | [TK]D-Fender | flujan: perhaps not in astDB.... hrm. |
| 15:34.21 | flujan | blitzrage: do you why the sip channel keep stuck? |
| 15:34.33 | blitzrage | no |
| 15:34.38 | Zefk | Hi. My asterisk console and calls hangs on 1.4.14 and 1.4.15. With 1.4.13 everything works fine. Any hints? |
| 15:35.41 | blitzrage | Zefk: file a bug with as much information as possible |
| 15:36.17 | *** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 15:41.02 | flujan | [TK]D-Fender: do you already saw this problem? |
| 15:42.00 | [TK]D-Fender | flujan: nope, I don't limit my calls |
| 15:42.32 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:42.32 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:47.04 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 15:47.15 | flujan | [TK]D-Fender: There is not related to the limit |
| 15:47.16 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:47.19 | flujan | I update the limit to 3 |
| 15:47.21 | *** join/#asterisk ming_zym (n=ming_zym@124.14.236.162) |
| 15:47.22 | flujan | and sip reloaded |
| 15:47.24 | flujan | no help |
| 15:47.28 | flujan | the problem still happens |
| 15:51.29 | *** join/#asterisk cesar_CR (n=cesar@201.192.34.34) |
| 15:54.11 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:54.11 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:57.34 | epaulin | How to resolve low volume issue? Probably cause by ECHO cancellation due to this issue occurs after two days running. |
| 15:58.41 | [TK]D-Fender | epaulin: What cards, and does this correct on reboot? |
| 15:59.28 | epaulin | [TK]D-Fender: TMP400P, it does not. |
| 16:00.07 | darkskiez | whats the script/tool that will log in to irc and report whom is talking in a meetme conf? |
| 16:00.33 | [TK]D-Fender | epaulin: TMP400P? |
| 16:00.35 | epaulin | [TK]D-Fender: sorry, TDM400P |
| 16:00.41 | epaulin | yess |
| 16:01.03 | [TK]D-Fender | epaulin: Ok, well that card does not have EC, nor have I ever heard of Zaptel EC dropping gains after a while. |
| 16:01.09 | reber | hi all |
| 16:01.33 | [TK]D-Fender | epaulin: And if rebooting doesn't help, then something else is happening. A permanent change seems to have gone into effect. |
| 16:01.59 | reber | i have the feeling that sipdiscount, voipbuster, internetcalls, etc. are the *very same* company or at least very similar sites/prices. True ? Feeling ? |
| 16:02.04 | [TK]D-Fender | darkskiez: Never heard of one, but doesn't sound terribly different to write |
| 16:02.40 | *** join/#asterisk huey23 (n=_huey23_@64.192.209.132) |
| 16:02.42 | darkskiez | [TK]D-Fender: nope, i have the time to play with an existing script but not write one, but agreed, not hard in principal :) |
| 16:02.53 | darkskiez | [TK]D-Fender: I was sure i've seen one in use here for an asterisk conference |
| 16:03.19 | [TK]D-Fender | darkskiez: Go check the WIKI & get Google-ing |
| 16:03.31 | darkskiez | [TK]D-Fender: i did that first, of course =] |
| 16:03.39 | epaulin | [TK]D-Fender: is that means ECHO cancellation training done, and talking effect? |
| 16:03.42 | [TK]D-Fender | darkskiez: Then continue in earnest :p |
| 16:04.16 | [TK]D-Fender | epaulin: All echo training does is stop audio for the specified time to detect if EC is needed. This does not change the actual routine used. |
| 16:04.30 | [TK]D-Fender | epaulin: What EC routine are you using, and have you tried others? |
| 16:04.58 | reber | i know this question is not *stricly* related to asterisk, but rather related to sip, but ... |
| 16:05.03 | huey23 | any suggestions to why 1 users voicemail cuts out mid sentence? i increased the maxsilence but didn't help |
| 16:05.15 | epaulin | [TK]D-Fender: I didn't change the default conf |
| 16:05.55 | epaulin | [TK]D-Fender: rxgain=0.0 txgain=0.0 |
| 16:06.12 | [TK]D-Fender | epaulin: wasn't what I asked... |
| 16:06.55 | epaulin | [TK]D-Fender: sorry, I'm new to asterisk |
| 16:07.15 | epaulin | [TK]D-Fender: what is EC routine? |
| 16:08.13 | [TK]D-Fender | epaulin: enter "ztcfg -vvvv" and it will tell you what echo canceller you are using |
| 16:08.39 | *** join/#asterisk Somebee (n=Somebee@cm-84.208.125.244.getinternet.no) |
| 16:08.54 | epaulin | [TK]D-Fender: Echo Canceller: MG2 |
| 16:10.09 | coppice | anyone here tried zed-3 phones? |
| 16:10.19 | [TK]D-Fender | epaulin: go read the BOOK or WIKI to see about trying out another routine and see if it works better for you. Now this should not normally have to do with gains. first disable EC altogether (echocancel=no) and see if that helps the gain issue. If not, re-enable it and start playing with your gains a bit |
| 16:11.24 | epaulin | [TK]D-Fender: tnx a lot, I'll do that. |
| 16:11.48 | huey23 | [TK]D-Fender: you are in rare form today, did you get laid? |
| 16:12.31 | Zeeek | off? |
| 16:12.35 | *** join/#asterisk wglenncamp (n=wglennca@c-69-139-126-170.hsd1.ky.comcast.net) |
| 16:13.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:13.49 | wglenncamp | Can someone tell me if you can run FOP on an asterisk appliance? If not, is there another solution out there where I can view the statuses of calls in the parking lot on a large displayed LCD monitor hanging on the wall? |
| 16:13.57 | [TK]D-Fender | coppice: their higher end definitely looks better than the base (resembles the old IPDIALOG phones). Kinda "brick-like". Good price point? |
| 16:14.09 | wglenncamp | Possibly another server watching the asterisk app. |
| 16:14.52 | wglenncamp | (and don't say trixbox) ;) |
| 16:14.57 | Zeeek | in just about 45 minutes, the VOIP Users COnference, codenamed "Brassy Monkeys" will begin: http://x2z.eu |
| 16:15.20 | coppice | [TK]D-Fender: dunno about the prices, but I saw some in a shop. the look solid but fully up to industry standards for ugly. they run linux, but I can't see any source code being provided |
| 16:15.32 | *** join/#asterisk mihinomenest (i=ZJ8r@66.255.220.17) |
| 16:15.59 | [TK]D-Fender | coppice: So, par for the course :p |
| 16:16.19 | [TK]D-Fender | coppice: LCD usability looks crappy too? Was hoping for pics |
| 16:16.21 | Zeeek | can the appliance run apache? |
| 16:17.47 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:18.21 | wglenncamp | y'know, I'm not sure what the appliance is running. But, I am concerned since it's an embedded system, it doesn't have the power to run FOP, which is why I am looking for a solution. |
| 16:18.46 | wglenncamp | Ideally I would like to run another box to process the FOB and dedicate the appliance only to call processing. |
| 16:18.55 | wglenncamp | I may be asking for the impossible though |
| 16:19.08 | wglenncamp | FOP I mean |
| 16:21.23 | [TK]D-Fender | coppice: user's manul describes the web config in detail but not a SINGLE thing the USER would do :) |
| 16:21.32 | Zeeek | wglenncamp you can use the manager to watch asterisk |
| 16:22.02 | [TK]D-Fender | wglenncamp: Go write a small app to connect via AMI. |
| 16:22.16 | coppice | [TK]D-Fender zed-3 was apparently started by the guy who started Zultys, after that went under. it looks like similar code in new cases |
| 16:22.44 | [TK]D-Fender | coppice: Zultys.... yup... the other king of fugly :) |
| 16:22.54 | [TK]D-Fender | coppice: and massively overpriced |
| 16:23.14 | coppice | these are less ugly than zultys |
| 16:23.35 | wglenncamp | Yes, I understand about the API, is there something out there that's already written. I don't have the time to come up with something. This is for a customer of ours. |
| 16:24.00 | wglenncamp | Nevermind, I think that I found some info on voip-info Thanks |
| 16:24.42 | huey23 | any suggestions to why 1 users voicemail cuts out mid sentence? i increased the maxsilence but didn't help...i have also searched for this issue and found nothing :( |
| 16:26.15 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:26.24 | *** join/#asterisk saftsack (n=saftsack@pD9E0402C.dip.t-dialin.net) |
| 16:28.50 | Zeeek | maybe i'll just jump to 1.6 in production! But then, nah! |
| 16:29.11 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 16:29.40 | codefreeze | brave man, that Zeeek |
| 16:29.59 | Zeeek | no, I already chickened out |
| 16:30.08 | codefreeze | ;) |
| 16:30.11 | Zeeek | 1.4 tomorrow. I promise. Yeah, tomorrow. |
| 16:30.52 | codefreeze | I'm astounded at times, how long it takes to reach 'tomorrow'. |
| 16:31.34 | Zeeek | it's been a looong time comin' that's for sure |
| 16:32.38 | Zeeek | can't toast. Nothing to drink. |
| 16:32.54 | coppice | you need bread for toast |
| 16:33.05 | codefreeze | oj/milk/water will do in a pinch |
| 16:33.11 | huey23 | coffee? |
| 16:33.13 | Zeeek | no, no and no |
| 16:33.25 | codefreeze | and coffee too, I'd imagine. |
| 16:33.28 | Zeeek | toast = champagne or whiskey |
| 16:33.37 | Qwell | whiskey toast? |
| 16:33.46 | Zeeek | only the finest of either will do |
| 16:34.07 | huey23 | cow urine? |
| 16:34.35 | Zeeek | The following is a channel floo^H^H^H important announcement. |
| 16:34.49 | codefreeze | hmmm. You have to draw a line somewhere. I personally draw mine way, way before that, huey23 !! |
| 16:35.05 | Zeeek | The VOIP Users COnference begins at 12 noon EST, i.e., in 25 minutes. |
| 16:35.14 | file | codefreeze: you are a fan of hot chocolate aren't you? |
| 16:35.17 | *** join/#asterisk Enigma (n=enigma@globalways/developer/enigma) |
| 16:35.39 | codefreeze | file: just finished off a cup as an experiment, as a matter of fact |
| 16:35.43 | huey23 | :P |
| 16:35.52 | codefreeze | hot choc, that is! |
| 16:36.16 | Qwell | codefreeze: Did you bring enough for the entire class? |
| 16:36.32 | Zeeek | Champagne for the class |
| 16:36.51 | codefreeze | Well, if I open the new sack, there probably would be... |
| 16:37.17 | *** join/#asterisk jhb (n=joerg@81-5-139-2.dsl.eclipse.net.uk) |
| 16:37.42 | Zeeek | We have a special guest this week, an IP Magician |
| 16:37.55 | Qwell | Zeeek: eh? |
| 16:38.06 | Zeeek | although his spelling leaves room for improvement |
| 16:38.10 | jhb | hi *. Just trying to dial from agi. using 'Exec Dial "SIP/%s|8|r|o|G(default,777%s,1)" |
| 16:38.17 | Qwell | can he make entire class As disappear? |
| 16:38.41 | jhb | now, the call connects, but does not transfer both parties to 777xxx |
| 16:38.45 | jhb | any ideas? |
| 16:38.48 | Zeeek | he can change audio that is breaking up into zombiz channels |
| 16:39.44 | *** join/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 16:40.03 | Qwell | Zeeek: who? |
| 16:40.23 | [TK]D-Fender | jhb: Nows the point that options are not 1 parameter EACH <--- |
| 16:40.23 | supjigator | Anyone have a Maxtnt manual? |
| 16:42.44 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 16:42.46 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:44.04 | jhb | [TK]D-Fender, what do you mean by not one param each? |
| 16:44.36 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:45.00 | *** join/#asterisk bytesnagger (n=bytesnag@c-71-205-52-205.hsd1.mi.comcast.net) |
| 16:45.12 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:45.30 | Zeeek | r o G |
| 16:45.50 | *** part/#asterisk bytesnagger (n=bytesnag@c-71-205-52-205.hsd1.mi.comcast.net) |
| 16:46.44 | huey23 | any voicemail masters in the house? |
| 16:50.51 | blitzrage | huey23: just go ahead and ask your question :) |
| 16:50.55 | blitzrage | there may or may not be |
| 16:51.16 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:51.30 | huey23 | i have ... twice :) |
| 16:51.36 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 16:51.50 | Zeeek | blitzrage hey now |
| 16:52.04 | blitzrage | Zeeek: hey hey now now! |
| 16:52.23 | Zeeek | time to join us at http://x2z.eu |
| 16:52.33 | blitzrage | unfortuantely I gotta go eat food and do some running around :( |
| 16:52.40 | blitzrage | so with that... I'm off to get fooooood |
| 16:52.48 | Zeeek | awwwww |
| 16:53.06 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:54.40 | _x86_ | fewd |
| 16:56.03 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:56.22 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 16:57.01 | Zeeek | Last call for participants before the door is locked: http://x2z.eu |
| 16:57.04 | *** join/#asterisk _pepo_ (n=c9eea608@190.10.187.20) |
| 16:57.13 | _pepo_ | hi friends |
| 16:57.14 | Zeeek | You can listen to file type, he's really quick! |
| 16:57.28 | Zeeek | and he sings! |
| 16:57.40 | iratik | Anyone know of any softphones that can be triggered to call using a URL-like protocol? not to mention skype.... but you can create skype urls that launch skype and begin a call ... e.g: skype:+14178675309 ..... are there any softphones compatible with SIP out there that have similar capabilities? |
| 16:59.03 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 17:00.16 | *** join/#asterisk gabbernaster (n=jshanks@69.10.147.2) |
| 17:00.29 | iratik | let me rephrase that |
| 17:00.47 | iratik | Does anyone know of a softphone that takes command line arguments to launch it that enables it to start dialing a number? |
| 17:02.39 | lirakis | does the "autologoff" setting take effect on a reload or does it only work after a restart? |
| 17:03.03 | ManxPower | lirakis: And which autologoff option might you be referring to? |
| 17:03.08 | [TK]D-Fender | iratik: what is your full goal? |
| 17:03.29 | lirakis | ManxPower: in agents.conf |
| 17:03.56 | iratik | TK-D a javascript client initiated auto-dialing system |
| 17:04.04 | ManxPower | lirakis: I am not aware of any non-hardware config option requiring a restart rather than a reload. |
| 17:04.26 | lirakis | ManxPower: moh usually requires a restart |
| 17:04.29 | lirakis | i thought |
| 17:04.35 | lirakis | ManxPower: changes to the files.. |
| 17:04.47 | *** join/#asterisk Chris-NB (n=chris@213162066135.public.t-mobile.at) |
| 17:05.01 | ManxPower | lirakis: no, modern files based MoH (i.e. non-mpg123, etc) should not require a restart. |
| 17:05.10 | lirakis | ok |
| 17:05.12 | lirakis | cool thanks |
| 17:06.53 | _pepo_ | please, I am using asterisk with PostgreSQL like voicemail system... but, How do I disable voicemail forwarding? |
| 17:07.59 | [TK]D-Fender | iratik: You mean user dials from a web page? |
| 17:08.10 | ManxPower | _pepo_: A security issue has been found with Postgress SQL that was announced today. |
| 17:08.56 | ManxPower | _pepo_: I think the term you are looking for is ARI (Asterisk Realtime Interface). |
| 17:10.06 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-57-3.dsl.tul2ok.sbcglobal.net) |
| 17:11.55 | iratik | [TK]D-Fender: I've already got it working with skype.. but skype sucks and i'd rather use the ipbx system i've set up |
| 17:12.14 | iratik | its like some hack job version of predictive/power dialing |
| 17:12.51 | [TK]D-Fender | iratik: Should I take that as a "yes"? |
| 17:13.23 | iratik | yeah |
| 17:13.38 | SwK | so wtf is up w/ asterisk-biz being +k'd? |
| 17:13.56 | iratik | i'm just looking for an sip softphone with the ability to take a phone number as an argument |
| 17:14.03 | *** join/#asterisk hijacked (i=8LEs@66.255.220.17) |
| 17:15.17 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:15.58 | [TK]D-Fender | iratik: I'd suggest using a soft-phone that supports auto-answer, and then using an AMI originate. |
| 17:16.01 | flujan | finally... :D |
| 17:16.14 | flujan | I stayed all the day look for a problem on my e1. |
| 17:16.27 | iratik | AMI originate? |
| 17:16.28 | iratik | googling |
| 17:16.29 | flujan | According to the telco guy the E1 closes something like a loop |
| 17:16.41 | flujan | and it not releases unless i restart the machine |
| 17:16.58 | flujan | ztcfg or reload chan_zap solves the issue. |
| 17:17.17 | flujan | E enabled the debug on that span and now i can have the information necessary to detect the problem. |
| 17:17.35 | iratik | thanks |
| 17:18.41 | SwK | iratik, ikega (someone correct me if I am wrong) but thats an opensouce softphone based on OPAL supports several protocols... you should be easily able to mod that to do what you what.. |
| 17:18.48 | SwK | actually I want a Flash/Flex soft phone |
| 17:19.57 | iratik | thanks |
| 17:24.33 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:29.55 | _pepo_ | Is ARI=ARA ? |
| 17:31.49 | flujan | [TK]D-Fender: could you see this for me? |
| 17:31.50 | flujan | http://pastebin.com/m1a616aae |
| 17:32.02 | flujan | I am not totally fluent on the pri debug... :) |
| 17:32.19 | [TK]D-Fender | _pepo_: Typically ARI = Asterisk Recording Interface. ARA = Asterisk Realtime Archetecture. 2 clearly different things |
| 17:33.10 | russellb | Asterisk Realtime Interface ? |
| 17:33.10 | russellb | :-p |
| 17:33.44 | _pepo_ | <ManxPower> _pepo_: I think the term you are looking for is ARI (Asterisk Realtime Interface). |
| 17:33.47 | _pepo_ | <ManxPower> _pepo_: I think the term you are looking for is ARI (Asterisk Realtime Interface). |
| 17:34.05 | _pepo_ | sorry I read: <ManxPower> _pepo_: I think the term you are looking for is ARI (Asterisk Realtime Interface). |
| 17:34.25 | russellb | o.O |
| 17:36.07 | [TK]D-Fender | russellb: some is clearly Not Usually That Sane.... |
| 17:36.10 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) |
| 17:37.10 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 17:37.41 | ManxPower | I wasn't aware Asterisk had a Recording Interface. |
| 17:37.53 | [TK]D-Fender | ManxPower: 3rd party junk |
| 17:38.19 | ManxPower | Oh. |
| 17:38.58 | ManxPower | Maybe we need to start replacing "Asterisk" with "Digium", for example the Digium Realtime Interface to distinguish it from 3rd party crap. |
| 17:39.08 | _x86_ | anyone use bugzilla? i need to merge two bugs (not just mark one as a dup of the other) |
| 17:39.49 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:40.36 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 17:40.45 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:44.54 | flujan | russellb: I am having a strange behavior with my digium board... |
| 17:45.14 | flujan | is is closing a loop to the telco guy and I think I discover the issue that triggers it... |
| 17:45.51 | flujan | when I receive a call on number 44882120 on span 1 I answer it and dial to span 2 4567909 |
| 17:46.04 | flujan | I call this follow-me to track users on vacantion... |
| 17:46.15 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 17:46.19 | _pepo_ | How do I disable voicemail forwarding (option 8 in the user-voicemail menu)? |
| 17:46.20 | flujan | span 1 = telco 1 |
| 17:46.25 | flujan | span 2 = telco 2 |
| 17:46.43 | flujan | could it be a zaptel bug? |
| 17:46.58 | flujan | I dunno with who I can speak about it. |
| 17:48.19 | russellb | flujan: support@digium.com |
| 17:48.31 | flujan | russellb: ok... Thanks for the tip. |
| 17:48.33 | russellb | np. |
| 17:49.16 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 17:49.22 | ManxPower | _pepo_: very few people here use databases with voicemail. Your search of the mailing list archives and the Wiki was not helpful? |
| 17:50.02 | ManxPower | flujan: "closing the loop" on a T-1/E-1 makes NO sense. |
| 17:50.12 | flujan | hi ManxPower |
| 17:50.25 | ManxPower | Each T-1/E-1 loop supports 24 or 30 channels. |
| 17:50.43 | flujan | ManxPower: the telco guys said the the link just loops for he. |
| 17:50.59 | ManxPower | When you refer to "the loop" you are talking about shorting the line and all 24 or 30 channels. |
| 17:50.59 | _pepo_ | Ive read that usedirectory is the option that was looking for ... ill try it |
| 17:51.32 | flujan | Now I solved this issue sending the link to a legacy pbx with dialog boards and this old pbx sends me the calls. |
| 17:51.42 | flujan | ManxPower: the entire linke 30 channels stop working to receive or place calls. |
| 17:51.53 | jhb | ... I use agi to do a 'Exec Dial "SIP/%s@de-out|8|rG(default^777%s^1)"' % (sip,confno)', but the calling party just gets disconnected, while the called party still stays in. Any idea why? |
| 17:52.02 | jhb | (python agi) |
| 17:52.22 | ManxPower | jhb: For one thing % really is not valid |
| 17:52.26 | flujan | ManxPower: I am not a pri master but according to the pastie it is working fine... |
| 17:52.30 | tripps | what is the default setting of jitterbuffer for iax2 trunks when it isn't set in iax.conf? |
| 17:52.38 | flujan | Dunno why it just stops working. |
| 17:52.41 | ManxPower | tripps: none, I think. |
| 17:53.00 | ManxPower | jhb: also an 8 second timeout is pretty short. |
| 17:53.31 | ManxPower | jhb: as far as I can tell that is NOT a valid line. |
| 17:53.41 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 17:53.54 | tripps | ManxPower: i've got a customer backhauling all voice over iax to a PRI SIP gateway and they're getting call quality issues - like voice cutting in and out. did some research and saw that perhaps jitterbuffer steps on echo cancelation but looks like i don't have a jitter buffer |
| 17:54.00 | ManxPower | unless you have an exten => %s,1,Whatever in your destnation servers dialplan |
| 17:54.07 | tripps | should i enable the jitter buffer? |
| 17:54.24 | ManxPower | tripps: asterisk does not do echo cancel on Voip links. |
| 17:54.45 | ManxPower | tripps: I have no idea. We don't use IAX anymore |
| 17:55.44 | tripps | ManxPower: thought maybe the endpoints were doing it in hardware |
| 17:55.59 | jhb | ManxPower, ok, sorry, after python string substituion it will be something like: AGI Script Executing Application: (Dial) Options: (SIP/004952144694640@de-out|8|rG(default^777566^1)) |
| 17:56.09 | jhb | (cli output) |
| 17:56.18 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 17:56.42 | jhb | ManxPower, I have an extension _777XXX |
| 17:56.57 | ManxPower | tripps: I would not know anything about that, but I do know that echo cancel on a voip stream is silly and doesn't work. |
| 17:56.59 | *** join/#asterisk rnovotny22 (n=rnovotny@70.13.57.232) |
| 17:56.59 | jhb | and the phone gets picked up |
| 17:57.25 | ManxPower | jhb: put the CLI output of a failed call on pastebin.ca and maybe someone can help you. |
| 17:57.28 | ManxPower | ~pb |
| 17:57.29 | 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 |
| 17:58.40 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 17:58.42 | MrTelephone | qwell |
| 17:58.45 | MrTelephone | you there |
| 17:58.47 | MrTelephone | http://bugs.digium.com/view.php?id=11245 |
| 18:01.48 | *** join/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2) |
| 18:01.50 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 18:02.42 | *** part/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 18:03.13 | scooby2 | anyone seen a gui interface of any kind that just lets you add users/agents and add them to queues? freepbx is too controlling |
| 18:03.25 | scooby2 | other wise i think i might write something |
| 18:03.41 | tripps | any iax2 gurus here who can help me debug some logs? looking to play with jitter buffefr and other settings to try and help with call quality issues. particularly get my arms around various RR_xxx verbose information, etc. |
| 18:05.01 | jhb | ManxPower, http://pastebin.ca/802820 |
| 18:05.36 | jhb | the funny thing is that if I call the extension 887 directly, which does the same thing, everything works fine... |
| 18:12.52 | ManxPower | tripps: the first question is WHAAT DIRECTION is the bad call quality? |
| 18:15.09 | coppice | downwards |
| 18:17.29 | *** join/#asterisk eliel (n=eliel@201.148.150.53) |
| 18:19.17 | tripps | ManxPower: that's what I'm trying to work through - I think it may be both but defininitely inbound audio to where the endpoints are in this case |
| 18:19.32 | tripps | ManxPower: note the trunk is between two * servers |
| 18:20.29 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 18:20.42 | *** part/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 18:21.14 | ManxPower | tripps: Are you using IAX2 trunking or are using the word trunk because you don't know any better>? |
| 18:22.39 | ManxPower | So you have SIP Phone <-> Asterisk <-> Internet <-> Asterisk <-> SIP/PRI gateway <-> Telco |
| 18:22.41 | *** join/#asterisk bantu (n=Miranda@p54A32CB7.dip0.t-ipconnect.de) |
| 18:23.43 | tripps | ManxPower: iax2 trunking. yes you're correct. the internet component is where the trunk is |
| 18:23.57 | ManxPower | ~trunk |
| 18:23.58 | jbot | somebody said trunk was is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 18:24.25 | tripps | right. i do have trunking enabled on the iax2 config |
| 18:24.58 | *** join/#asterisk RoyK (n=roy@ip-135-16-149-91.dialup.ice.no) |
| 18:25.21 | ManxPower | Does setting trunk=no on both sides fix the problem? |
| 18:25.26 | ManxPower | (do a reload after, of course) |
| 18:27.09 | tripps | ManxPower: haven't tried that - the call quality issues seem to happen even when there is only one call - would that affect that scenario? |
| 18:27.22 | ManxPower | trunk=yes is active even with one call |
| 18:31.26 | ManxPower | tripps: there are MANY possible causes of call quality issues that have nothing to do with jitter. |
| 18:32.17 | coppice | yeah, like being a stinky PSTN call, without the benefit of a wideband codec :-) |
| 18:32.45 | tripps | ManxPower: agreed - it seems though the dropped voice is only inbound and typically starts several minutes into a call |
| 18:32.56 | *** join/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 18:37.35 | tripps | ManxPower: what should i look for in the iax2 debug logs that would help me isolate things or look for tell-tale signs? |
| 18:40.09 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 18:41.59 | *** join/#asterisk los415 (n=los415@cpe-76-171-125-207.socal.res.rr.com) |
| 18:43.09 | *** join/#asterisk CVirus (n=GoD@196.205.192.131) |
| 18:44.40 | *** join/#asterisk JimmyDee (n=James@97-86-161-253.static.stls.mo.charter.com) |
| 18:45.43 | JimmyDee | can anyone explain why I would get a busy signal dialing my sip dialup number when asterisk is running? |
| 18:49.01 | polerin | JimmyDee: did it register? are you seeing any indication of an incoming call on the asterisk console on verbosity > 5 ? |
| 18:51.41 | ManxPower | tripps: I already told you what you should try. |
| 18:54.50 | tripps | ManxPower: tried that made it ten times worse |
| 18:55.14 | ManxPower | tripps: That would be expected if you only changed the setting on one side, not both sides. |
| 18:55.29 | *** join/#asterisk mog (i=mog@nat/digium/x-13ac97c3f0d35a0a) |
| 18:55.29 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:55.35 | tripps | ManxPower: did it both sides |
| 18:55.41 | ManxPower | tripps: weird. |
| 18:55.45 | Navion | Any ideas why console/dsp should go busy? Paging can't connect to it anymore. |
| 18:56.01 | hmmhesays | can anyone recommend me a decent snmp manager for linux or windows? I need to collect traps from a gateway |
| 18:56.13 | ManxPower | Unfortunatly it took so long to get this far that I now have to leave to go run errands. Best of luck. |
| 18:56.24 | tripps | ManxPower: thanks |
| 19:00.16 | *** join/#asterisk enalert (n=trelane@2001:4830:150c:0:20d:61ff:fe31:a58) |
| 19:01.56 | hmmhesays | anyone? |
| 19:02.17 | *** part/#asterisk ussrback (n=MAX@80.92.183.84) |
| 19:05.42 | tripps | i want to set up an inbound route to simply call and listen to hold music. what is the easiest dialplan to accomplish this? |
| 19:09.40 | enalert | tripps, s,1,answer() s,2,Musiconhold(context) iirc |
| 19:10.50 | blitzrage | probably don't even need the Answer() as the MusicOnHold() application will probably answer the line (if it's like Playback or Background) |
| 19:11.00 | blitzrage | (not that Answer() will hurt anything) |
| 19:11.34 | tripps | enalert blitzrage: thanks! |
| 19:18.09 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:21.09 | *** join/#asterisk techie (n=techie@adsl-76-214-21-156.dsl.lsan03.sbcglobal.net) |
| 19:23.32 | lirakis | is there a way to correlate ext with an agent after they have logged in? .. like is it stored in a DB variable some where? |
| 19:24.13 | lirakis | some times FOP doesnt show an agent logged in.. but the extesion rings from a queue any way... and i want to be able to figure out what agent IS logged into that extension |
| 19:28.26 | polerin | mmph, anyone have suggetsions for which route to look into for php/asterisk scripting? |
| 19:28.35 | *** join/#asterisk afrosheen (n=cj@207.71.49.137) |
| 19:29.06 | afrosheen | does anyone have experience with polycoms and blind transfers |
| 19:30.25 | afrosheen | we want to have calls transferred straight to voicemail sometimes, but after hitting the blind soft key and the vm extension (*200 for example) the star breaks out of Blind mode on the phone |
| 19:38.16 | afrosheen | sure is dead here today |
| 19:38.54 | outtolunc | yep yep |
| 19:39.59 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:41.31 | *** join/#asterisk nirz (n=nir@bzq-79-182-159-80.red.bezeqint.net) |
| 19:42.06 | Yourname`` | So, all I do to make sure gsm is used is set it in sip.conf? That's IT? |
| 19:43.32 | [TK]D-Fender | Yourname``: disallow=all allow=gsm |
| 19:43.36 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:43.43 | [hC] | hurrghh.. how come the FOP forgets about which agents are logged in when i restart it.. |
| 19:44.59 | Yourname`` | And hello to you too, [TK]D-Fender :) |
| 19:45.11 | Yourname`` | How is this chilly but normal weather treating you? |
| 19:45.19 | [TK]D-Fender | #@%$%#@$%#$@5 cold |
| 19:46.27 | *** join/#asterisk RoyK (n=roy@ip-135-16-149-91.dialup.ice.no) |
| 19:46.28 | Yourname`` | haha |
| 19:46.36 | Yourname`` | It's kind of nice here in TO today. |
| 19:57.49 | afrosheen | ok I fixed it, had to add a |*xxx to my digitmap |
| 20:01.11 | *** join/#asterisk nirz (n=nir@bzq-79-183-111-228.red.bezeqint.net) |
| 20:01.30 | *** join/#asterisk saftsack (n=saftsack@pD9E0402C.dip.t-dialin.net) |
| 20:03.56 | [hC] | jeez I wish CLI had a filter (like tcpdump) to specify a pattern in which to match for watching calls. src <tech/ext>, dst <tech/ext> etc.. |
| 20:04.05 | [hC] | on high volume systems that would help so much |
| 20:04.30 | [hC] | how many times ive asked someone to try doing something only to confuse a result with another call in progress |
| 20:05.41 | _x86_ | [hC]: i'm with ya :) |
| 20:06.12 | jer | [TK]D-Fender, thanks, disabling the usb port entirely fixed my earlier problem, whereas disabling it in software did not |
| 20:07.52 | blitzrage | [hC]: I agree... although everyone says it should be filtered either outside of the CLI, or some other such biznez |
| 20:08.14 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 20:08.46 | [hC] | blitzrage: yeah... It seems that the only thing suited for doing this would be accessible via manager. |
| 20:09.13 | [hC] | blitzrage: cause even if you filter the log files, you have to do a lot of work to match the thread pid in the log entry as well, yada yada.. |
| 20:09.21 | *** join/#asterisk trippss (n=ss@72.20.150.196) |
| 20:09.23 | blitzrage | indeed... |
| 20:11.55 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 20:12.36 | *** join/#asterisk RoyKa (n=roy@ip-167-16-149-91.dialup.ice.no) |
| 20:12.52 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 20:12.53 | teknoprep | hey all |
| 20:12.55 | *** part/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 20:13.00 | teknoprep | i can't get MWI to work on more then one phone |
| 20:13.15 | Yourname`` | [TK]D-Fender: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '318ddb5f5f534329339075f40622c7c8@64.xx.xx.xx'. Giving up. -> Can't get any google on this :( |
| 20:13.26 | trippss | is there a good resource explaining the various IAX2 debug codes, RR_PKTS, etc.? |
| 20:13.28 | teknoprep | i have a series of phones that don't have there own MailBox |
| 20:13.42 | [TK]D-Fender | jer: You're welcome |
| 20:13.44 | teknoprep | so i put them on the gernerla mail box of the entire office... |
| 20:13.47 | [TK]D-Fender | Yourname``: No idea. |
| 20:13.55 | teknoprep | but i can only get one phone to see when there is a msg waiting |
| 20:14.09 | Yourname`` | [TK]D-Fender: Are you being SERIOUS?! |
| 20:14.11 | [TK]D-Fender | teknoprep: PASTEBIN <------ |
| 20:14.22 | Yourname`` | Shit, this is topic materal. Defender said "No idea" to something! |
| 20:14.23 | blitzrage | do you have the mailbox=100@default for each of the phones in sip.conf? |
| 20:14.34 | blitzrage | (where 100@default is replacable) |
| 20:16.25 | Yourname`` | Is there anyway where I can learn how to really understand sip debugs? |
| 20:16.38 | *** part/#asterisk mog (i=mog@nat/digium/x-13ac97c3f0d35a0a) |
| 20:16.40 | Yourname`` | What do I look for, etc, what does it mean, etc? |
| 20:16.42 | SwK | Yourname``, the RFCs heh |
| 20:16.46 | Yourname`` | SIP/2.0 404 Not Found -> Stuff like that. |
| 20:16.56 | SwK | Yourname``, unfortunately thats the best place |
| 20:17.06 | Yourname`` | Makes sense. |
| 20:17.20 | Yourname`` | I'm seeing tons of SIP/2.0 404 Not Found in the debug. |
| 20:18.12 | SwK | 404 means it cant find a number |
| 20:18.20 | SwK | just like a 404 on a webserver |
| 20:18.48 | SwK | in fact a lotta the SIP codes are equiv or similar in meaning to the http response codes |
| 20:19.27 | Yourname`` | Ah. |
| 20:19.31 | Yourname`` | So what would this mean? -> http://pastebin.ca/802980 |
| 20:20.16 | SwK | ok... any one of the ops wanna chime in on what #asterisk-biz is keyed all of a suffed |
| 20:20.39 | SwK | well the .111 box is telling you it cant find s as a valid destination |
| 20:21.03 | SwK | the matching invite could be helpful to see what the invite URI is |
| 20:21.31 | Yourname`` | Ok, the copying pasting of all this is the culprit. Gimme one min, please. |
| 20:22.12 | bkw_ | SwK: I can't join #asterisk-biz either |
| 20:22.38 | Yourname`` | There's some stuff totally messed up, I'd think, heh, one min |
| 20:22.54 | nestAr | HAX! |
| 20:23.24 | SwK | yeah I'm wondering why I was kicked and the channel set to +k.... |
| 20:23.41 | bkw_ | SwK: no clue |
| 20:24.10 | Yourname`` | Olden days they used to call that a "channel takeover" |
| 20:24.11 | Yourname`` | heh |
| 20:24.20 | bkw_ | you cant do that on freenode |
| 20:24.21 | nestAr | they still do, on efnet |
| 20:24.22 | russellb | it was a sillyness channel will just a small group of digium folks, but we started killing off the channel today |
| 20:24.25 | nestAr | but not so much here. |
| 20:24.43 | bkw_ | russellb: reason being? |
| 20:24.56 | russellb | to move to digium jabber |
| 20:25.05 | SwK | it was not a sillyness channel... i have used it many times for actual business as its the closet thing on irc to the biz list |
| 20:25.18 | russellb | SwK: what? |
| 20:25.36 | russellb | there has _never_ been anyone in there in the past long time i have been in there other than a few digium folks talking about nothing |
| 20:25.38 | SwK | then you should also close #asstricks and all the other off colour sillyness channels as well |
| 20:25.45 | russellb | no ... those are fine |
| 20:26.26 | russellb | this has nothing to do with anyone else ... it's just that we tell digium people to keep internal chats on an internal server |
| 20:26.29 | russellb | it's as simple as that |
| 20:27.12 | russellb | file: you can fix up #asterisk-biz so it can be used by others |
| 20:27.18 | Qwell | russellb: already done |
| 20:27.21 | russellb | ok, good |
| 20:27.40 | russellb | so there, it's open and free for use |
| 20:27.42 | russellb | sorry for the downtime. |
| 20:28.56 | Deeewayne | I haven't seen a business conversation in -biz in 8 months |
| 20:29.05 | russellb | Deeewayne: exactly ... |
| 20:29.11 | Qwell | 8 months? |
| 20:29.24 | Deeewayne | it was file and I for a while |
| 20:29.28 | file | Deeewayne was there longest, I followed him in |
| 20:29.41 | file | after talking to him on the phone once about it :D |
| 20:29.52 | Deeewayne | I think I actually did see people using it once or three times when I was doing chan_dialogic |
| 20:29.56 | Yourname`` | SwK: I pasted it to you |
| 20:30.15 | Yourname`` | Pastebin for problems: http://pastebin.ca/802997 |
| 20:31.11 | Yourname`` | I even get these errors in there when calls are going on: |
| 20:31.12 | Yourname`` | [Nov 29 16:44:35] WARNING[32380]: translate.c:163 framein: no samples for ulawtolin |
| 20:31.12 | Yourname`` | [Nov 29 16:40:39] WARNING[26711]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '4a5042636c589e7c02c4f1906f1ba6e8@64.34.139.74'. Giving up. |
| 20:31.17 | SwK | yeah IU'm on the phone |
| 20:31.35 | Yourname`` | k |
| 20:32.59 | *** join/#asterisk pdd_ (n=pdd@static-71-189-209-74.lsanca.dsl-w.verizon.net) |
| 20:33.50 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 20:35.50 | Yourname`` | What's that command to see what codec is in use on a given call? |
| 20:36.08 | blitzrage | show channel <channel> |
| 20:36.32 | Yourname`` | Thank you leif |
| 20:38.02 | *** join/#asterisk kkn088 (n=kikoun@77.204.31.201) |
| 20:38.38 | Yourname`` | Ok, so there's two channels.. one call. One channel is directed to the phone, and the other is directed to the provider. I want to use GSM from phone -> asterisk -> phone, so I should be looking at the channel of the phone, correct? |
| 20:38.48 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 20:39.39 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:39.52 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 20:41.14 | MrTelephone | russell b where are the digium internal chat channels? |
| 20:41.26 | Qwell | MrTelephone: internal |
| 20:41.34 | MrTelephone | send me the proxy |
| 20:41.39 | Qwell | what proxy? |
| 20:41.42 | MrTelephone | hey qwell i patched that crap |
| 20:41.52 | MrTelephone | but it should check realm and nonce |
| 20:42.00 | *** join/#asterisk Tonak (i=Tonak@216.54.19.104) |
| 20:42.06 | MrTelephone | so should I make a whole new sub called get_header_auth |
| 20:42.07 | MrTelephone | or something |
| 20:42.13 | Qwell | dunno |
| 20:42.15 | Yourname`` | Great, now I get this new error: translate.c:163 framein: no samples for gsmtolin |
| 20:42.22 | Qwell | MrTelephone: ask oej on the bug |
| 20:42.27 | Tonak | hey could someone help me with something SOO simple??? |
| 20:42.29 | MrTelephone | did you see my queer little loop code?\ |
| 20:42.34 | MrTelephone | alright |
| 20:42.35 | Tonak | will be a really quick question |
| 20:42.38 | Qwell | ~ask |
| 20:42.39 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:42.57 | Tonak | I have a trixbox and 2 phones |
| 20:43.09 | Qwell | stop right there |
| 20:43.10 | Qwell | ~trixbox |
| 20:43.11 | 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 |
| 20:43.12 | Tonak | i just want to be have the phone to be able to call each other |
| 20:43.14 | Yourname`` | Someone help a brotha out on this one, please -> translate.c:163 framein: no samples for gsmtolin |
| 20:43.24 | Yourname`` | LOLOL @ 'stop right there' |
| 20:43.53 | Tonak | well i can delete trixbox |
| 20:43.56 | MrTelephone | wish i could but i tried to stay away from gsm |
| 20:43.59 | Tonak | im not using any of its features |
| 20:44.04 | Tonak | i just want to be able to call |
| 20:44.37 | Yourname`` | MrTelephone: Client is using 4-5 agents softphones on a DSL connection. He hears too much echo/static on his calls and we were at ulaw then. So trying gsm now. :( |
| 20:45.09 | MrTelephone | echo and static between two sip endpoints? |
| 20:45.39 | Yourname`` | MrTelephone: No, between softphone and PSTN endpoint. |
| 20:45.41 | Tonak | Qwell: could u help me out please? i got asterisk, and 2 phones all 3 hooked up to a switch box, just want the phones to be able to call each other |
| 20:46.04 | MrTelephone | yourname, pri or analog? |
| 20:46.15 | Tonak | Qwell: got the phones to register with asterisk, but they give me a busy signal when i type something |
| 20:46.24 | Qwell | Tonak: that's trixbox doing that |
| 20:46.33 | MrTelephone | tonak, set static ips on the phones and static usernames then call the phone using uri |
| 20:46.36 | Qwell | they love to mangle configs. we cannot help you, if you're using their configs |
| 20:46.57 | _Sam-- | "im not using their configs" |
| 20:47.00 | Tonak | ok |
| 20:47.03 | Tonak | if i were to start over |
| 20:47.05 | Yourname`` | MrTelephone: SIP trunks.. |
| 20:47.08 | Tonak | should i use CentOS? |
| 20:47.19 | MrTelephone | don't say sip trunks its a swear word in here |
| 20:47.20 | MrTelephone | lol |
| 20:47.26 | Yourname`` | lol |
| 20:47.37 | [hC] | what has happened to this channel... |
| 20:47.41 | [hC] | it reads like AOL |
| 20:47.41 | teknoprep | if a sip trunk isn't a sip trunk then what is it? |
| 20:47.49 | Qwell | ~trunk |
| 20:47.50 | jbot | it has been said that trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use ... |
| 20:47.50 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust441.winn.cable.ntl.com) |
| 20:47.59 | MrTelephone | tonak i run a cable tv comp, internet comp and phone comp by myself i can't give you dedicated help :( |
| 20:48.26 | Tonak | I can erase al the configs |
| 20:48.27 | [hC] | Haha |
| 20:48.39 | Tonak | will that help? |
| 20:48.45 | Yourname`` | MrTelephone: Well, in that case, it's not PRI, neither Analog ;) |
| 20:49.00 | teknoprep | in telecom, a trunk is a "single voice channel between two pieces of switching equipment." ???? <<<----- isn't that what your SIP "trunk" does ? |
| 20:49.00 | teknoprep | IncCalls.wav |
| 20:49.00 | teknoprep | gabcast-channel.gsm |
| 20:49.00 | teknoprep | Closed.wav |
| 20:49.00 | teknoprep | nv-greeting.gsm |
| 20:49.02 | teknoprep | ClosedREC.wav |
| 20:49.06 | teknoprep | gabcast-episode.gsm |
| 20:49.08 | teknoprep | reminder4.gsm |
| 20:49.10 | teknoprep | nv-menu.gsm |
| 20:49.12 | teknoprep | NoAnswer.wav |
| 20:49.14 | teknoprep | Lunch.wav |
| 20:49.16 | teknoprep | reminder1.gsm |
| 20:49.18 | *** kick/#asterisk [teknoprep!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 20:49.22 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 20:49.24 | teknoprep | reminder3.gsm |
| 20:49.26 | teknoprep | wow |
| 20:49.27 | Yourname`` | lol |
| 20:49.28 | teknoprep | i am really sorry |
| 20:49.34 | Yourname`` | Right click, teknoprep? |
| 20:49.37 | Qwell | damn, I was hoping you were gonna have something like... |
| 20:49.39 | bkw_ | teknoprep: you just cause trouble everywhere dont ya? |
| 20:49.39 | teknoprep | i have a mac |
| 20:49.49 | Qwell | I don't know - maybe a nasty-nurses/ dir |
| 20:50.07 | MrTelephone | yeah its hard to diagnose problems when your using another sip provider for pstn |
| 20:50.35 | MrTelephone | sip to sip sound good? |
| 20:50.44 | Tonak | Question: will making the extensions.conf file empty hurt?? |
| 20:50.56 | Yourname`` | MrTelephone: This is the setup. His softphones connect to our asterisk server over IP. Our asterisk server connects to a provider like broadvoice over IP. |
| 20:50.59 | MrTelephone | telephone crap is hard to deal with.. someone says they have a static call but they forgot they called a cell phone customer going through a tunnel |
| 20:51.00 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:52.07 | Yourname`` | Other than that error, my pastebin full of idiots: http://pastebin.ca/802997 |
| 20:52.42 | lirakis | teknoprep: its not a trunk.. its a peer |
| 20:52.58 | lirakis | teknoprep: wash your brain free of the trixbox verbiage |
| 20:53.45 | MrTelephone | if you have echo then its in the rtp and not the sip messages |
| 20:53.47 | MrTelephone | i know that much |
| 20:53.56 | *** join/#asterisk g_sus (n=blitzrag@CPE000024c29e38-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 20:53.58 | MrTelephone | echo cancellation can't work properly with jitter |
| 20:54.13 | MrTelephone | test your link from your asterisk box to your clients |
| 20:54.19 | MrTelephone | then from yoru asterisk box to the pstn |
| 20:54.25 | *** part/#asterisk g_sus (n=blitzrag@CPE000024c29e38-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 20:54.26 | Yourname`` | I did, 64-60ms. |
| 20:54.35 | Yourname`` | Then 74ms from asterisk to the provider. |
| 20:54.45 | MrTelephone | do you have a softphone or something on your laptop connected to your asterisk box? |
| 20:55.21 | SwK | Yourname``, those are just 404s on options pinging |
| 20:55.23 | SwK | nothing to worry about |
| 20:55.29 | *** join/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 20:55.36 | anarcat | hello |
| 20:56.24 | anarcat | i have those two extensions in my dialplan, and wonder why the first one never gets used: |
| 20:56.27 | Yourname`` | SwK: Oh, ok. What about errors like this: [Nov 29 16:44:35] WARNING[32380]: translate.c:163 framein: no samples for gsmtolin |
| 20:56.31 | *** part/#asterisk unbeatabl (n=haha@outbound.thoughtconvergence.com) |
| 20:56.44 | anarcat | exten => _901133.,1,... |
| 20:56.53 | anarcat | exten => _9011.,1,... |
| 20:57.03 | anarcat | why can't i override the second wildcard somehow? |
| 20:57.41 | MrTelephone | i wish cisco would have used sip for everything.. then it would be a solid protocol |
| 20:58.00 | Qwell | MrTelephone: you're joking, right? |
| 20:58.40 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 20:58.42 | MrTelephone | well becuase they would MAKE it RFC |
| 20:58.54 | MrTelephone | then it would be deployed everywhere and everyone would just use their implementation |
| 20:58.58 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:59.03 | MrTelephone | and it has to work because there crap is severely overpriced |
| 20:59.04 | MrTelephone | hah |
| 20:59.25 | MrTelephone | they'd RFC their own sip hah |
| 20:59.31 | MrTelephone | sorry im doing drugs or something |
| 20:59.33 | MrTelephone | don't mind me |
| 20:59.33 | Yourname`` | Stupid RTP stuff. I hear a slight echo now, MrTelephone:( |
| 20:59.53 | MrTelephone | from where to where do you hear echo\/ |
| 21:00.25 | Yourname`` | From the cellphone -> softphone |
| 21:00.25 | Yourname`` | The cellphone guy hears his own voice in the echo, not bad.. VERY slight. |
| 21:01.17 | Tonak | Anyone: Is there a way to dial a phone ext from Asterisk CLI |
| 21:01.58 | Yourname`` | Tonak: help originate |
| 21:02.35 | *** part/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 21:02.58 | MrTelephone | cellphone -> pstn provider -> asterisk -> your softphone? |
| 21:03.01 | MrTelephone | is that what you mean |
| 21:03.15 | lesouvage | I have a strange problem with a sangoma A102 Dual E1 card. It's loosing connection with the telco (or telco is loosing connection with the card) now and then (almost once a day). See http://www.pastebin.be/7137 for logging. Any suggestion is more then welcome. |
| 21:03.39 | MrTelephone | lexvouvage, interesting because I got 36 carrier drops in 112 days |
| 21:03.39 | Yourname`` | MrTelephone: More like cellphone -> pstn -> my voip provider -> asterisk -> softphone |
| 21:03.44 | MrTelephone | yeah |
| 21:03.48 | MrTelephone | i mean how do you fix that |
| 21:03.54 | MrTelephone | your adding a lot of hops there |
| 21:04.05 | MrTelephone | cellphones suck to begin with |
| 21:04.14 | muiro | question: I'm getting a strange message, but only when calling through a normal phone through a sip service into my asterisk box. I was wondering if someone could shed some light on this: http://pastebin.com/m6d7a12d2 Please not that this happens only when calling through the sip trunk. A soft phone does not generate this notice. |
| 21:04.19 | MrTelephone | then you add another 100ms + jitter from a voip provider |
| 21:04.33 | MrTelephone | you should try another voip provider |
| 21:04.42 | Yourname`` | MrTelephone: Why so? |
| 21:05.02 | lesouvage | After unplugging the line, wait a couple of second, plug it back in and the problem is solved for one or two days. Unfortunately the customer doesn't accept this as a solution :-( |
| 21:05.20 | MrTelephone | lesouvage, upgrade to the latest firmware |
| 21:05.31 | MrTelephone | lesouvage, try a new cat 5 cable |
| 21:05.47 | MrTelephone | and replease your telcos demarc maybe the pins on it or shit? |
| 21:05.54 | MrTelephone | replace |
| 21:07.22 | MrTelephone | how long is your cable to the sangoma card |
| 21:07.31 | lesouvage | MrTlephone: The telco is actually now testing the line (during the weekend) to find any reason on their site for this strange behaviour. We already have updated the firmware, the wanpipe driver, asterisk, zaptel. |
| 21:07.32 | MrTelephone | email sangoma they are good at troubleshooting |
| 21:07.58 | MrTelephone | you'll have to do a loopback test with your card to verify if the ports are functioning correctly |
| 21:10.29 | MrTelephone | lesouvage, check for master/slave timing settings.. check for irq misses, make sure there are no electric motors in the vicinity of your cabling |
| 21:10.35 | MrTelephone | if so use sheilded cables |
| 21:10.44 | MrTelephone | proper grounding |
| 21:11.04 | lesouvage | MrTelphone: another Sangoma card is on the way. What do you mean with telcos demarc? The loopback tet is a good idea. I should have done this during the weekend while the system is disconnected because the telco is testing. |
| 21:11.32 | De_Mon | its dmark |
| 21:11.37 | Qwell | demarc |
| 21:11.46 | Qwell | demarcation point |
| 21:11.53 | *** join/#asterisk RoyK (n=roy@ip-167-16-149-91.dialup.ice.no) |
| 21:12.18 | De_Mon | I ment dmarc |
| 21:12.25 | MrTelephone | i get the odd carrier drop but its so minute im not worrying about it at this point |
| 21:12.27 | Qwell | no, it's demarc :) |
| 21:12.38 | De_Mon | no e's! |
| 21:12.44 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 21:12.45 | outtolunc | hehe |
| 21:12.47 | Qwell | yes, e |
| 21:12.52 | MrTelephone | i called it a demarc because im not sure what you call their cpe equipment |
| 21:13.02 | Qwell | MrTelephone: CPE |
| 21:13.05 | De_Mon | http://en.wikipedia.org/wiki/Demarcation_point |
| 21:13.08 | MrTelephone | hahah |
| 21:13.11 | MrTelephone | ok whatever then |
| 21:13.22 | jameswf | ~cpe |
| 21:13.23 | jbot | cpe is, like, Customer Premises Equipment. Telephone devices such as handsets and PBXs located at the customer.s site that interface with the public network. It includes equipment such as modems, terminals and routers supplied by the telephone company, installed at customer sites and connected to the telephone network. |
| 21:13.23 | MrTelephone | i can't keep track of all this crap :P\ |
| 21:13.30 | Qwell | "A demarcation point is also referred to as the demarc" |
| 21:13.33 | nestAr | let's have a semantic debate |
| 21:13.35 | Qwell | De_Mon: from your own link. |
| 21:13.40 | jameswf | ~dmarc |
| 21:13.45 | Qwell | ~demarc |
| 21:13.46 | nestAr | excellent use of our time |
| 21:13.46 | jameswf | ~dmark |
| 21:13.47 | nestAr | ;) |
| 21:13.52 | jameswf | bleh |
| 21:13.55 | Qwell | ~demarcation point |
| 21:13.59 | Qwell | stupid bot |
| 21:14.06 | jameswf | ~dmarcation |
| 21:14.11 | jameswf | ~demarcation |
| 21:14.15 | MrTelephone | demarc woulb be a synonymn for pop |
| 21:14.18 | [hC] | if you spell it dmark or dmarc you're spelling it wrong on your own man. if you look at WHY its spelled dEmarc, it makes perfect sense. |
| 21:14.20 | jameswf | ~stupid bot |
| 21:14.21 | jbot | Stupid human. |
| 21:14.21 | De_Mon | Qwell nice try how about the whole sentance |
| 21:14.37 | Qwell | De_Mon: you can't correct somebody using "demarc" if it's first in the list. |
| 21:14.51 | lesouvage | MrTelephony: what can happen with electric moters close to the system/cable |
| 21:15.41 | jameswf | nana nana boo boo |
| 21:15.43 | MrTelephone | induction. heard a few stories about this happening with ip network cabling |
| 21:15.51 | Yourname`` | Is there anyway I can check how much bandwidth is being used by a call in asterisk? |
| 21:15.57 | MrTelephone | it scrambles the electrons runnin on the cable |
| 21:16.02 | De_Mon | in a call? |
| 21:16.17 | De_Mon | Yourname`` get the ipaddress and use your standard network traffic monitor |
| 21:16.32 | De_Mon | ip/port if necessary |
| 21:16.48 | muiro | question: I'm getting a strange message, but only when calling through a normal phone through a sip service into my asterisk box. I was wondering if someone could shed some light on this: http://pastebin.com/m6d7a12d2 Please not that this happens only when calling through the sip trunk. A soft phone does not generate this notice. |
| 21:17.04 | muiro | this issue seems to be causing an issue further in the dialplan |
| 21:17.04 | Yourname`` | De_Mon: I mean the server is the only thing that's 'here', the client is elsewhere. |
| 21:17.16 | MrTelephone | telcos should have a cpe testing tool for t1s as well. im sure if you called in a trouble shoot they will test it |
| 21:17.18 | *** join/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 21:17.23 | Yourname`` | De_Mon: I want to see the bandwidth just on that call. |
| 21:17.25 | lesouvage | Is the demarcation point where the infrastructure of the telco ends and the private infrastructure starts. The walloutlet for the cable. |
| 21:17.35 | MrTelephone | yeah |
| 21:17.56 | jameswf | ~vote |
| 21:17.57 | MrTelephone | my t1 demarc looks like a circuit board with the same lights you'd find on a modem |
| 21:18.02 | MrTelephone | or cpe |
| 21:18.04 | De_Mon | Yourname`` yes each call uses a specific ip and port (we're talking about sip/iax right -- i mean who cares abound bandwidth on the other technologies) |
| 21:18.24 | Yourname`` | De_Mon: SIP, yes. |
| 21:18.59 | MrTelephone | we have a fiber -> copper box that terminates into this t1 dsu card.. then from there it goes rj45 to the sangoma |
| 21:19.06 | MrTelephone | i suppose everyone elses is different |
| 21:19.06 | De_Mon | Yourname`` asterisk doesnt have that information use your standard network monitoring tools after you find out from asterisk what ip/port it is using |
| 21:19.17 | *** join/#asterisk tzafrir_laptop (n=tzafrir@89.130.6.136) |
| 21:19.22 | MrTelephone | alright gotta jet for a few |
| 21:19.51 | lirakis | is there any way to see what extension is associated with an agent? |
| 21:19.58 | Yourname`` | ok |
| 21:20.08 | jameswf | ~agent |
| 21:20.08 | jbot | Newsflex has a free-agent like interface, but without requiring Gnome or KDE, you can find it at freshmeat |
| 21:20.40 | muiro | can anyone help me figure out what this message means? I only get this issue whan I'm using a standard phone to call into my asterisk box through a sip service (voicemeup). [Nov 30 16:17:39] NOTICE[8491]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms? |
| 21:21.12 | lesouvage | MrTelephone: The tip on the mixed electronics in the cable could be usefull. The cable is passing lots of other servers and other electronics. A proper shielded cable and avoiding the cable passing to close to other machines might be the answer. |
| 21:22.13 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 21:22.37 | jameswf | ~dmarc |
| 21:22.38 | jbot | well, dmarc is the demarcation point is the point at which the telephone company network ends and connects with the wiring at the customer premises. A demarcation point is also referred to as the demarc, DMARC, MPOE, or minimum point of entry. |
| 21:22.44 | *** join/#asterisk tzafrir_laptop (n=tzafrir@89.130.6.136) |
| 21:23.00 | jameswf | yay |
| 21:23.56 | MrTelephone | ~pop |
| 21:23.57 | jbot | pop is probably (Point Of Presence) This is a local telephone number through which you can access your ISP. The largest national ISPs have POPs all over the country. |
| 21:24.13 | jameswf | ~soda |
| 21:24.14 | jbot | Soda is a yummy carbonated drink, or the main form of subsistence for programmers, or actually pop |
| 21:24.48 | jameswf | ~jolt |
| 21:24.49 | jbot | i heard jolt is the most caffeine in a holdable bottle., or http://www.wetplanet.com/, or something that Servowire drinks alot :) |
| 21:24.49 | De_Mon | james you have way too much time on your hands |
| 21:24.49 | jameswf | yeah |
| 21:24.49 | jameswf | ~botsnack |
| 21:24.49 | jbot | jameswf: :) |
| 21:24.49 | Nugget | http://macnugget.org/photos/misc/pophell |
| 21:25.03 | MrTelephone | it pisses me off when a software designer says there is nothing wrong with their sip implementation while in production the brand name products like polycom and cisco work fine |
| 21:25.07 | MrTelephone | ~cocksuckers |
| 21:25.08 | De_Mon | ~jameswf |
| 21:25.09 | jbot | somebody said jameswf was he has way to much time on his hands |
| 21:25.36 | jameswf | sip is inharently flawed use IAX2 |
| 21:25.43 | De_Mon | iax2 sucks use sip |
| 21:26.07 | MrTelephone | how many computers have you come across that won't boot up with a usb mass storage device plugged in? |
| 21:26.08 | *** join/#asterisk santor_ (n=xidarian@pool-71-180-154-74.tampfl.fios.verizon.net) |
| 21:26.11 | jameswf | sip doesnt nat properly |
| 21:26.22 | De_Mon | MrTelephone um, a lot actually |
| 21:26.24 | jameswf | ~jameswf |
| 21:26.25 | jbot | you are, like, he has way to much time on his hands, or a GOD |
| 21:26.25 | MrTelephone | why? |
| 21:26.27 | Nugget | that's NATs fault, mostly. NAT blows goats. :) |
| 21:26.36 | De_Mon | sip is fine we just need IPV6 already |
| 21:26.42 | lesouvage | Endless hours spend because a cable is passing some electronics that shouldn't be there in the first place, what a shame. |
| 21:26.48 | MrTelephone | why doesn't sip nat properly |
| 21:26.57 | santor_ | does anyone here connect to teliax for termination |
| 21:27.05 | jameswf | um poor design |
| 21:27.07 | santor_ | yea |
| 21:27.10 | Nugget | sip header contain ip information which can confuse the ever living shit out of devices if their IP isn't what they think it is. |
| 21:27.11 | De_Mon | mememe I do |
| 21:27.13 | MrTelephone | as long as the client uses symetric ports |
| 21:27.23 | santor_ | IAX is better design but not a good option till everyone adopts it |
| 21:27.35 | MrTelephone | this is what I would like to see.. a proxy that can tell if two clients are behind the same nat |
| 21:27.44 | MrTelephone | I think openser can probably do it |
| 21:27.48 | santor_ | and someone makes a good IAX router that handles calls like openser does for sip |
| 21:27.51 | De_Mon | yeah find me a non asterisk device that does iax and I might consider it |
| 21:28.20 | santor_ | De_Mon: there are a few, but there not common yet |
| 21:28.20 | MrTelephone | make tunnels using cisco gear or linux boxes/routers |
| 21:28.22 | Qwell | De_Mon: iaxclient |
| 21:28.30 | Nugget | When I travelled a lot I would run asterisk on my laptop so that I could use a softphone talking to the local asterisk which then talked IAX to my real asterisk server. |
| 21:28.30 | MrTelephone | if you really have problems you should tunnel |
| 21:28.35 | jameswf | technicaly my pc running mozphone is a non asterisk device |
| 21:28.36 | De_Mon | since when did software count as a device? |
| 21:28.37 | santor_ | put your pbx on the edge of your network |
| 21:28.39 | [hC] | oh this is fun... seemingly exten => h will get picked up cross-context when a call is hung up |
| 21:28.42 | [hC] | :| |
| 21:28.44 | Nugget | it was the only way to navigate the broken, multilayer NAT you often encounter at hotels |
| 21:28.54 | Nugget | now I just never leave the house. Much simpler solution. :) |
| 21:28.58 | De_Mon | iax polycoms anyone? |
| 21:29.09 | MrTelephone | our pbx is network edge and the sip clients are private ip's that the edge can see but public can't |
| 21:29.22 | trippss | is polycom thinking about it? |
| 21:29.40 | santor_ | not sure about polycom those are good phones, but there are a few phones that support iax |
| 21:29.41 | jameswf | we were considering a sip to nat gateway, plug any sip device to side a and iax poops out side b |
| 21:30.01 | jameswf | jbot: eat poop |
| 21:30.02 | jbot | ACTION eats poop and falls over dead |
| 21:30.15 | De_Mon | most other voip software doesn't talk iax either so you gotta covert anwho ala ccm and ms exchange are the ones I'm thinking of at the moment |
| 21:30.37 | santor_ | we just need sip 3.0 |
| 21:30.38 | lesouvage | jameswf: you could use asterisk for that I guess |
| 21:30.45 | muiro | anyone have any idea what this might mean: [Nov 30 16:17:39] NOTICE[8491]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms? |
| 21:30.51 | santor_ | if they called IAX sip 3.0 everyone would support it |
| 21:30.54 | De_Mon | I went to a bad site in IE and now my computer has some fscking spyware i can't remove~ |
| 21:30.55 | trippss | jameswf: were you going to do trunking on the IAX2 side? would be interesting to "mux" those SIP signaling sessions through a single IAX trunk |
| 21:31.00 | jameswf | yes but thinking lower level lika an ata style device |
| 21:31.15 | De_Mon | need to delete locked files i think, since delete-on-boot isn't cutting it :( |
| 21:31.33 | De_Mon | i cant decide if i hate windows more or less than ie rightnow |
| 21:31.45 | jameswf | should hack a pap2 to do IAX |
| 21:31.50 | santor_ | De_Mon: why use either |
| 21:32.05 | trippss | is there an rfc out for iax? |
| 21:32.06 | [hC] | jameswf: why not use an iaxy? |
| 21:32.16 | De_Mon | santor_ playing games mostly |
| 21:32.37 | jameswf | ~windows |
| 21:32.38 | jbot | i guess windows is a 32 bit hack on a 16 bit operating system, originally designed for an 8 bit CPU, with a 4 bit system bus, made by a 2 bit company that can't stand 1 bit of competition... or the World of Warcraft bootloader, or the most important collection of bugs |
| 21:32.40 | lesouvage | muiro: google on "ast_sched_add_variable: Scheduled event in 0 ms?" and lots of links show up. |
| 21:32.57 | santor_ | De_Mon: dual boot, and use wine, i play hl on wine, i'd play hl2 but all i have is a laptop an dit has intel video |
| 21:32.59 | *** join/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 21:33.29 | santor_ | wow works on wine |
| 21:33.53 | santor_ | hl2, hl and alot of other games |
| 21:34.18 | santor_ | but if you have a top of the line gfx card that may not have full 3d support in linux |
| 21:34.45 | De_Mon | I am dual booting one computer is running widows and the other is linux, ssh rocks. its windows and IE that I have a problem with. |
| 21:35.18 | santor_ | do you have vista? |
| 21:35.45 | trippss | ~vista |
| 21:35.46 | jbot | You're not serious, are you? |
| 21:35.51 | De_Mon | yes, but not for playing games :P |
| 21:36.31 | trippss | i finally banished windows from my desk. running ubuntu 7 and wine+ie6 when i absolutely need it. so far a good combo |
| 21:37.17 | santor_ | my laptop came wiht vista and i rarely boot into vista |
| 21:37.46 | santor_ | it doesn't support the intel 945 gfx so it won't even play old games |
| 21:38.08 | santor_ | but ubuntu supports opengl on that car |
| 21:38.12 | santor_ | *card |
| 21:38.18 | santor_ | er gfx chip |
| 21:38.33 | santor_ | so i just have ot be happly wiht hl 1.6 |
| 21:38.55 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 21:39.19 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:40.16 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 21:40.25 | tripps | now running mirc in wine :) |
| 21:40.35 | *** join/#asterisk bcnl (n=mike@S010600131078957c.vc.shawcable.net) |
| 21:40.42 | tripps | pidgin seems to have a hard time staying connected . . . |
| 21:41.25 | bcnl | in Asterisk 1.2.X is there an easy way to make an extension ring *all* registered SIP devices on the server? |
| 21:41.51 | De_Mon | tripps ew |
| 21:41.57 | bcnl | kind of a easy way to make '0' ring everyone and not make me change it everytime I MAC users? |
| 21:42.15 | santor_ | does anyone connect a voip provider called teliax? i'm having trouble with them and connecting asterisk |
| 21:42.32 | De_Mon | santor_ I'm connected fine right now |
| 21:42.33 | tripps | ~teliax |
| 21:42.34 | jbot | well, teliax is an ITSP that sucks less than most. |
| 21:42.53 | santor_ | well, i'm using for pstn termination in a test lab |
| 21:42.59 | santor_ | just to test things |
| 21:43.06 | santor_ | so call quality doesn't matter |
| 21:43.11 | De_Mon | well, you did something wrong becuase its working for me |
| 21:43.16 | santor_ | ~teliax |
| 21:43.17 | jbot | methinks teliax is an ITSP that sucks less than most. |
| 21:43.25 | MrTelephone | who wants to buy some expensive minutes of me :( |
| 21:43.52 | santor_ | De_Mon, are you connecting with sip or IAX |
| 21:43.59 | De_Mon | mmmm |
| 21:44.27 | santor_ | mmmm never heard of that protocal |
| 21:44.45 | De_Mon | iax |
| 21:44.55 | De_Mon | i was checking |
| 21:45.07 | De_Mon | and now I must reboot, hopefully windows will still be working :) |
| 21:45.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:45.24 | jameswf | why does em wink suck ass |
| 21:45.35 | De_Mon | em wink? |
| 21:45.36 | jameswf | ~wink |
| 21:45.37 | jbot | ACTION winks at jameswf |
| 21:45.41 | De_Mon | wink is cool |
| 21:45.47 | De_Mon | duno about 'em' wink tho |
| 21:46.10 | lesouvage | bcnl: see http://www.asteriskguru.com/tutorials/autotest_peers.html |
| 21:46.40 | santor_ | does anyone here connect to teliax via SIP??? |
| 21:48.47 | santor_ | what does it mean when you get Failed to authenticate on INVITE when you try to make a call to a sip channel that is a sip provider |
| 21:49.12 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 21:50.41 | *** join/#asterisk usam (n=usam@202.91.18.194) |
| 21:51.44 | De_Mon | it means your username and password are not correct |
| 21:51.48 | jameswf | can I increase the winkback wait time |
| 21:51.53 | *** join/#asterisk mog (i=mog@nat/digium/x-d498326c6a926b8a) |
| 21:51.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:51.54 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:56.02 | De_Mon | jameswf what are you talking about? |
| 21:56.17 | De_Mon | doesnt sound like http://www.debugmode.com/wink/ |
| 21:57.05 | De_Mon | okay I think my computer is malware free atlast one more reboot for good measure (stupid windows and its reboots!) |
| 21:57.31 | MrTelephone | fta sat receivers suck |
| 21:58.57 | MrTelephone | i got like 10 motorola cable modems i need to sell |
| 21:59.01 | MrTelephone | anyone interested ? |
| 22:01.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@89.130.6.136) |
| 22:03.03 | usam | TTHE ROOM with TONES |
| 22:03.06 | usam | ;) |
| 22:03.23 | usam | kram online lately? |
| 22:03.48 | De_Mon | I threw mine away |
| 22:03.51 | usam | isint /seen username ? |
| 22:03.54 | *** join/#asterisk dushantch (n=chatzill@adsl-ppp-1180.yubc.net) |
| 22:05.40 | hmmhesays | ~seen |
| 22:07.01 | tripps | ~seen trippss |
| 22:07.04 | jbot | trippss is currently on #asterisk (1h 57m 43s). Has said a total of 6 messages. Is idling for 30m 33s, last said: 'i finally banished windows from my desk. running ubuntu 7 and wine+ie6 when i absolutely need it. so far a good combo'. |
| 22:07.04 | jameswf | ~simpson |
| 22:07.27 | tripps | ~seen kram |
| 22:07.28 | jbot | kram <n=kram@cm56-214-79.liwest.at> was last seen on IRC in channel #debian, 89d 1h 55m 32s ago, saying: 'hey guys, my debian decided to show me parts of gnome and apt outout in half geran since a few weeks, can anyone tell me how to make it speak english eagain?'. |
| 22:07.42 | Qwell | ^ not kram |
| 22:07.53 | tripps | usam: so kram hasn't been around in a while :) |
| 22:07.59 | Qwell | 27 weeks |
| 22:08.19 | jameswf | ~peepee |
| 22:08.19 | jbot | ACTION AAAAAAAAAHHHHHHHHHHHHHHH!!!!!!!!!!! Oh boy, that feels gooooooooooooood!!!! Anybody got the latest issue of PlayBot or Hustlerbot Magazine or something? |
| 22:08.28 | usam | ~ghostrider |
| 22:08.35 | usam | lol |
| 22:08.36 | usam | hjrhehhe |
| 22:08.48 | tripps | ~seen jbot |
| 22:08.50 | jbot | jbot is currently on #asterisk-doc (4d 32m 32s) ##t42 (4d 32m 32s) #how (4d 32m 32s) #ol (4d 32m 32s) #flyspray (4d 32m 32s) #asterisk-dev (4d 32m 32s) #asterisk (4d 32m 32s) #byumug (4d 32m 32s) #orkut (4d 32m 32s) #nslu2-linux (4d 32m 32s) ##ducleague (4d 32m 32s) #storm (4d 32m 32s) #uphpu ... |
| 22:09.18 | tripps | hope I just didn't start an infinite loop :) |
| 22:09.22 | usam | i thought ot was going a "come back' for him after i wrote some kind of a letter than we will be joing the community again |
| 22:09.49 | Qwell | usam: English? |
| 22:11.58 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 22:13.12 | jameswf | ~translate en es the cheese is old and moldy, where's the bathroom |
| 22:13.46 | muiro | question: having an issue with asterisk thinking extensions have been dialed that haven't. The situation is this: let's say the user dials extension 2, and in that extension there is a Goto to another context. The next time that Background() or WaitExten() is used, the last dialed digit is entered seems to stick around. Could this be caused by lag from my sip trunk? Is there any command I could issue to cleare out asterisk's dtmf memory or |
| 22:14.25 | usam | Qwell: after i saw, sorry |
| 22:14.46 | usam | s/ot/he |
| 22:14.57 | usam | FAT FINGER + on drug |
| 22:17.59 | *** join/#asterisk moprilo (n=jjohn@201.198.78.23) |
| 22:18.25 | moprilo | hi guys, has anyone heard of a company voicetrading.com? sounds good, but jmm something says scam to me |
| 22:19.00 | De_Mon | moprilo no |
| 22:19.03 | santor_ | ~translate en es wtf |
| 22:19.15 | *** join/#asterisk RoyK (n=roy@91.149.31.108) |
| 22:19.28 | santor_ | a translator bot |
| 22:19.55 | santor_ | jbot is fun |
| 22:20.09 | De_Mon | play with him in #asterisk-doc |
| 22:20.37 | *** join/#asterisk RoyK (n=roy@91.149.31.108) |
| 22:23.39 | moprilo | De_Mon: no what?.. they good? |
| 22:24.41 | De_Mon | no anyone has not heard of them |
| 22:24.50 | moprilo | ohh ok .. |
| 22:25.20 | *** join/#asterisk RoyK (n=roy@91.149.31.108) |
| 22:26.34 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:27.23 | muiro | ok, I'll ask my question again. I seem to be getting some kind of lag when calling asterisk through a sip trunk. At least I think it's lag. Here's what's happening: Let's say after dialing extension 2, the dialplan moves to another context. I usually stick a Wait(1) in there. Immediately afterward I use Background(), and for whatever reason asterisk seems to detect a DTMF tone, and it is ALWAYS the tone that was last dialed. This seems to |
| 22:29.19 | [TK]D-Fender | muiro, Next time have your failed call's CLI output at verbose 10 in a pastebin, along with your dialplan |
| 22:29.31 | *** join/#asterisk ruffyen (n=chatzill@bigmail2.bodnargroup.com) |
| 22:30.05 | ruffyen | what is a really good frontend gui for asterisk...? |
| 22:30.18 | bcnl | 13:46 < lesouvage> bcnl: see <deleted> |
| 22:30.36 | bcnl | errr |
| 22:30.36 | bcnl | 13:46 < lesouvage> bcnl: see http://www.asteriskguru.com/tutorials/autotest_peers.html |
| 22:30.52 | ruffyen | lol |
| 22:30.55 | bcnl | I was really hoping for something within asterisk to dial all registerd SIP devices |
| 22:30.56 | ruffyen | thanks |
| 22:31.09 | ruffyen | oh wait |
| 22:31.11 | ruffyen | lol |
| 22:31.16 | ruffyen | thats not for me |
| 22:31.27 | bcnl | is there anything in the realtime database that you can throw a call to? |
| 22:31.35 | bcnl | dial(SIP/*) |
| 22:31.56 | bcnl | ruffyen: what part of asterisk do you want a gui for and what platform/OS do you want the GUI on? |
| 22:32.00 | [TK]D-Fender | bcnl, what exactly are you looking to do? |
| 22:32.19 | bcnl | [TK]D-Fender: well I've got about 35 users, and they move/leave/change often |
| 22:32.24 | muiro | woah, this is strange too |
| 22:32.27 | ruffyen | well a web based version would be nice....and most of the people using the system are on windows |
| 22:32.29 | bcnl | so I'd like to have an extension I call that rings everyone |
| 22:32.49 | bcnl | and if I go away, or things move, it sucks to have to keep updating the one massive Dial() command |
| 22:33.04 | [TK]D-Fender | bcnl, Ok, make an AGI that dumps out "sip show peers" and builds a dial string. |
| 22:33.30 | muiro | here's a different strange behavior: every time I use SayDigits() to say a "6", this happens: |
| 22:33.37 | muiro | http://pastebin.com/m30566305 |
| 22:33.38 | [TK]D-Fender | bcnl, And no, there are no wildcards for dialing like that |
| 22:33.40 | bcnl | [TK]D-Fender: yea lesouvage gave me a link to a script that makes a bunch of .call files |
| 22:33.50 | bcnl | [TK]D-Fender: thanks, that's all I needed to know |
| 22:34.03 | bcnl | I wanted to know if I had to hack something myself, or if there was a *correct* way to do it |
| 22:34.19 | bcnl | I'll do AGI over making call files |
| 22:34.28 | [TK]D-Fender | bcnl, call files? um.... I'm not sure how thats going to work for you... |
| 22:34.42 | bcnl | http://www.asteriskguru.com/tutorials/autotest_peers.html |
| 22:34.46 | bcnl | is what someone else pasted |
| 22:35.04 | bcnl | like I said, AGI makes more sense I just wanted to make sure I wasn't about to hack something together that already existed in * |
| 22:35.20 | [TK]D-Fender | bcnl, you just want to "dial everyone". Call Files will start MULTIPLE callouts and is used to start a call from * to BOTH ends, not as a way to take a call on * and send IT to multiple places |
| 22:36.06 | bcnl | yea I've never used .call files, I'll just have something that parses 'asterisk -rx "SIP show peers"' to build the dial() on the fly |
| 22:36.16 | [TK]D-Fender | bcnl, however, multi-dial WILL have another uissue quite likely : string length limitation. |
| 22:36.26 | bcnl | curses! |
| 22:36.43 | bcnl | 35 peers, SIP/XXX and SIP/#XXX |
| 22:36.46 | [TK]D-Fender | bcnl, You'll need to keep an eye that your overall string doesn't overflow. |
| 22:36.55 | bcnl | 210 chars? |
| 22:37.01 | bcnl | give or take |
| 22:37.04 | bcnl | say 250 to be safe |
| 22:37.08 | bcnl | what's my limit? |
| 22:37.08 | [TK]D-Fender | bcnl, Someone else ehre modded the source to allow a SICK length. |
| 22:37.23 | [TK]D-Fender | bcnl, not sure on the limit |
| 22:37.24 | bcnl | to me 250 is "reasonable" |
| 22:37.28 | ManxPower | [TK]D-Fender: That must have been 1.4x or later |
| 22:37.44 | [TK]D-Fender | ManxPower, Yup. [hC] I think..... |
| 22:37.47 | ManxPower | 1.2 is closer to 253 chars for the application options |
| 22:37.56 | [TK]D-Fender | Yay, ANOTHER Firefox release! |
| 22:37.56 | [hC] | What now? |
| 22:38.18 | [TK]D-Fender | [hC], were you the one who modded the * string length for paging? |
| 22:38.18 | Nugget | [Nov 30 16:38:06] WARNING[3841]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 1: Red Alarm |
| 22:38.19 | Nugget | [Nov 30 16:38:06] WARNING[3841]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 1 |
| 22:38.24 | bcnl | well if there are any devs in here, Dial(SIP/*) would be sweet :> |
| 22:38.30 | [hC] | [TK]D-Fender: it sounds like something i would do, its been so long now i totally forget. hah. |
| 22:38.31 | Nugget | ^ I'm getting a lot of this, followed immediately by the alarms clearing |
| 22:38.35 | [TK]D-Fender | bcnl, NOT happening :) |
| 22:38.36 | Nugget | nothing has changed. any clues? |
| 22:38.50 | [TK]D-Fender | [hC], You wouldn't have forgotten... |
| 22:38.51 | Nugget | happening every 5 minutes or so |
| 22:38.57 | bcnl | [TK]D-Fender: techinical, or security issue? |
| 22:39.22 | [TK]D-Fender | bcnl, "psycho config oversight" :) |
| 22:39.52 | bcnl | hrmm |
| 22:40.01 | bcnl | is there a way I could auto add everyone to a queue? |
| 22:40.15 | bcnl | that'd achive the same result |
| 22:40.28 | [hC] | [TK]D-Fender: youd be surprised. :) What is the maximum length currently? |
| 22:40.35 | ManxPower | Nugget: RED alarms are almost always a physical failure. |
| 22:40.57 | [TK]D-Fender | bcnl, umm.... maybe. that'd be SICK though. |
| 22:41.01 | ManxPower | i.e. cable issue, repeater issue, etc. |
| 22:41.04 | Nugget | *nod* thanks |
| 22:41.08 | bcnl | heh sick in a good way? |
| 22:41.14 | Mavvie | that reminds me I still have to find a way to disable a physical trunk within the console. |
| 22:41.54 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 22:43.51 | bcnl | [TK]D-Fender: heh sorry to be a pain, but I'm curious... is that SICK in a good way, or a bad thing to do |
| 22:44.14 | [TK]D-Fender | bcnl, queue = ... shudder.. |
| 22:45.37 | bcnl | hrmm well I'm sort of ignorant on doing things like this |
| 22:45.43 | bcnl | I was hoping for a "right" way to do it |
| 22:46.09 | tripps | ~polycom |
| 22:46.09 | jbot | somebody said polycom was the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 22:46.12 | bcnl | that fell within the normal operating parameters of asterisk |
| 22:46.31 | muiro | [TK]D-Fender: http://pastebin.com/m7ee8cdbd |
| 22:46.35 | *** part/#asterisk techie (n=techie@adsl-76-214-21-156.dsl.lsan03.sbcglobal.net) |
| 22:47.02 | [TK]D-Fender | bcnl, well.. this was as I said.. a little unforseen. |
| 22:47.11 | tripps | ~phones |
| 22:47.12 | jbot | well, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like ... |
| 22:47.29 | bcnl | reall, I would have though someone else would have wanted a catchall/ringall application |
| 22:48.13 | bcnl | well maybe I'll look at automagically making a queue with every SIP device that's registered |
| 22:48.34 | bcnl | if I grow to 50+ devices, apparently I'll run into issues with string length so this might head that off at the pass |
| 22:49.33 | tripps | ~grandstream |
| 22:49.34 | jbot | well, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 22:50.31 | muiro | [TK]D-Fender: I realize there's a spelling mistake in my dialplan. It shouldn't be effecting what is happening. |
| 22:51.05 | bcnl | ~aastra |
| 22:51.36 | bcnl | great, I got a pile of them and the bot won't even comment on them |
| 22:53.02 | *** join/#asterisk RoyK (n=roy@91.149.31.108) |
| 22:53.46 | muiro | [TK]D-Fender: changed that final Background() into a Playback() seems to clear it up, but this seems indicative of some other problem so I want to get to the bottom of it |
| 23:07.27 | *** join/#asterisk CVirus (n=GoD@196.205.192.131) |
| 23:11.12 | *** join/#asterisk mog (i=mog@nat/digium/x-0dc49e22996d9de8) |
| 23:11.12 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:17.01 | *** join/#asterisk ruffyen (n=paul@68.113.105.2) |
| 23:26.20 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 23:26.31 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.11) |
| 23:31.37 | *** part/#asterisk rnovotny22 (n=rnovotny@70.13.57.232) |
| 23:32.04 | Navion | Anyone know what can make console/dsp busy? |
| 23:36.10 | [TK]D-Fender | Navion, "show channels" |
| 23:43.10 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 23:43.29 | ThatKidKel | Other than monitor, what dial options could I send that would prevent asterisk from trying a re-invite? |
| 23:46.26 | JunK-Y | canreinvite=no ? |
| 23:53.13 | *** join/#asterisk RoyK (n=roy@91.149.31.108) |
| 23:54.58 | hmmhesays | can you tell the polycom which server you want to watch a particular buddy on when you have multiple registrations? |