| 00:00.38 | JT | F2Knight: i take it 9999999999999 is above the extension number on the phone? |
| 00:00.47 | JT | if it is it should ring once |
| 00:02.07 | F2Knight | well in that example the -e100-9999999999999999999999 tries every extension in that range. you can also pass a port option to hit multiple ports, |
| 00:02.22 | F2Knight | and yes in the test it does ring it until you pick it up |
| 00:03.53 | F2Knight | part of what I am seeing is that the calls will light up multiple times.. because it is getting more then one attempt. this makes sense because the phone will light up the other lines as roll overs to support multiple inbound calls. |
| 00:04.01 | JT | but it only rings once not zillions of times? assuming you pick up |
| 00:04.17 | F2Knight | usually its about 5 calls that are comming in at once so my 4 lines light up i hang them all up and get one more |
| 00:04.26 | JT | hrm |
| 00:04.30 | F2Knight | assuming i pick up yes |
| 00:04.37 | JT | and since the phone only has 1 identity |
| 00:04.42 | JT | curious |
| 00:04.57 | F2Knight | but it will ring for about 20 - 30 seconds before giving up |
| 00:05.32 | F2Knight | just rand a default test.. .and ,,,, |
| 00:05.40 | F2Knight | still ringinging |
| 00:05.44 | F2Knight | ahh 1 min |
| 00:06.07 | F2Knight | so it will ring the phone for 1 min by default.. (using sip vicious tools) |
| 00:07.15 | p3nguin | My phone's name is not related to the extension number, so I'll test in a bit. |
| 00:07.32 | F2Knight | just changed the attack streing.. |
| 00:07.37 | F2Knight | string* |
| 00:07.58 | F2Knight | ./svwar.py -m INVITE ipaddress |
| 00:08.09 | F2Knight | same thing.. didnt even need to pass an extension range. |
| 00:08.44 | JT | be curious what the packet trace shows, but it could be a big one |
| 00:08.45 | citywok | were you getting scanned and curious what was going on, or are you just playing? |
| 00:09.12 | F2Knight | i just attacked my own phone with the sv tools .. |
| 00:09.22 | F2Knight | it replicated the issue just as I have been seeing. |
| 00:09.41 | p3nguin | From outside the NAT, this should never be a problem. |
| 00:09.43 | citywok | there was a talk at astricon about sending invites to phones trying to get them to initiate long distance calls or something |
| 00:10.32 | F2Knight | and doing this on the cli ./svwar.py -v --port=5060 -m INVITE 192.168.0.201;./svwar.py -v --port=5060 -m INVITE 192.168.0.201;./svwar.py -v --port=5060 -m INVITE 192.168.0.201;./svwar.py -v --port=5060 -m INVITE 192.168.0.201;./svwar.py -v --port=5060 -m INVITE 192.168.0.201 made all the lines light up as I expected |
| 00:10.51 | F2Knight | and it is a pain to end all the calls when that happens |
| 00:11.01 | p3nguin | Go outside the NAT and try it. |
| 00:11.09 | citywok | i'd just reboot the phone but then again it's a polycom and that takes a couple years |
| 00:11.24 | Naikrovek | ... |
| 00:11.27 | F2Knight | citywok, I wonder if this is what is happening .. people still trying to do this attack |
| 00:11.27 | Naikrovek | 45 seconds? |
| 00:11.45 | citywok | my ip 650 doesn't boot in 45 seconds |
| 00:11.46 | citywok | lol |
| 00:12.01 | p3nguin | Not everyone has a provisioning server to hand out files when the phone wants them. |
| 00:12.19 | F2Knight | my GXP2110 boots in less then 20 seconds.. about 13 - 15 seconds. |
| 00:12.44 | citywok | hmm i should time all my phones |
| 00:12.57 | F2Knight | p3nguin, yes this needs to be preformed out side the nat.. and that is where I have not been able to replicate.. |
| 00:13.19 | JT | F2Knight: you need a dodgier router! |
| 00:13.29 | F2Knight | so I am not sure if they have a modified script that is taking advantage of some hole or what. |
| 00:13.30 | JT | get some model numbers off affected customers |
| 00:13.58 | F2Knight | they are using there ITSP supplied equipment |
| 00:14.10 | F2Knight | at&t/ comcast etc |
| 00:14.12 | JT | itsp? that's you isnt it? |
| 00:14.27 | *** join/#asterisk slidesinger-lt (~jtatum@173-161-172-121-Philadelphia.hfc.comcastbusiness.net) |
| 00:14.29 | F2Knight | sorry ment ISP |
| 00:14.32 | JT | ah |
| 00:14.36 | F2Knight | so use to typing ITSP lol |
| 00:14.42 | JT | well yeah try behind one of those devices |
| 00:14.58 | SeRi | p3nguin, you avail? |
| 00:15.08 | p3nguin | If you're lucky. |
| 00:15.24 | SeRi | lol. |
| 00:16.20 | SeRi | I am getting some weird issues with voip.ms where a fax like tone keeps answering the calls. Have you ever had an issue like that with voip.me? |
| 00:16.35 | p3nguin | Stop calling a fax number. |
| 00:16.43 | F2Knight | lol |
| 00:16.43 | SeRi | incoming and outgoing inside voipms |
| 00:16.45 | SeRi | I am not |
| 00:16.48 | SeRi | I am calling my house. |
| 00:16.52 | SeRi | and my brother |
| 00:17.12 | JT | don't whistle like a fax handshake |
| 00:17.17 | p3nguin | I've never used the voipms internal extensions. Is that what you're talking about? |
| 00:17.25 | SeRi | Yes. |
| 00:17.36 | p3nguin | I've never had a reason to use them. |
| 00:18.12 | SeRi | It was a test.... so I guess not big deal. I report it but wanted to see if you ever had that issue |
| 00:18.35 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 00:18.40 | citywok | Aastra 6739i 1minute, 6757i 30 seconds; but both were POE booted and they boot 5 seconds quicker on wall power. CloudTc Glass phone running android about a minute |
| 00:19.44 | citywok | my local provisioning server isn't running so the ip650... well. yea :P |
| 00:19.51 | p3nguin | How are you dialing the internal extension? |
| 00:20.13 | SeRi | just the ext |
| 00:20.45 | SeRi | both phones reg to voip.ms (Soft Phones) over cell. |
| 00:21.23 | p3nguin | Do I need to rephrase the question? |
| 00:23.19 | SeRi | I dont know what you are looking for. I register to voip.ms with sipdroid and than I dial the ext associated with the account. |
| 00:23.25 | *** join/#asterisk cerberus_za (~coert@8ta-151-72-107.telkomadsl.co.za) |
| 00:24.41 | SeRi | was that you? |
| 00:25.07 | SeRi | I got dead air. |
| 00:25.52 | SeRi | p3nguin, ^^ |
| 00:26.34 | p3nguin | Okay, registering directly to voipms and dialing the extension makes sense to me. |
| 00:26.43 | p3nguin | I just wanted to know HOW you were dialing it. |
| 00:27.01 | p3nguin | Dial it through your Asterisk box and see how that works. |
| 00:27.42 | SeRi | It wont work. |
| 00:29.02 | SeRi | p3nguin, jump in the conf. |
| 00:30.45 | Jupe | geeze. you'd think i could google this and not find an answer to every question BUT this simple question. I can't remember the name of the file you use on apache to set the hostname in debian linux. |
| 00:30.54 | Jupe | not with virtual hosts, btw. |
| 00:31.37 | p3nguin | What do you mean it won't work? It will only not work if you do it wrong. |
| 00:31.57 | Jupe | there's a file you have to create... it's like, /etc/apache2/hostname.conf or something |
| 00:32.34 | p3nguin | You don't set a systems host name via the web server. |
| 00:32.46 | SeRi | p3nguin, I guess I did it wrong when I tried it would give me a fast busy tone. Right now I cant try it because my brother is off line. |
| 00:32.54 | p3nguin | Are you just trying to set the canonical name for the web server? |
| 00:32.58 | Jupe | yeah |
| 00:33.03 | p3nguin | It's in httpd.conf |
| 00:33.05 | SeRi | httpd.conf |
| 00:33.35 | p3nguin | ServerName |
| 00:33.54 | Jupe | thanks. |
| 00:34.43 | Jupe | i used to always go back to the same tutorial at a certain VPS hosting provider to copy and paste the commands to install mysql, apache2 and php on Debian, and then rackspace swallowed them up |
| 00:34.59 | SeRi | robot |
| 00:35.08 | SeRi | copypasta |
| 00:35.11 | ChannelZ | see how weak these GUIs make you? :P |
| 00:35.11 | SeRi | :) |
| 00:35.28 | Jupe | guis are good for... adobe photoshop =p |
| 00:35.33 | Jupe | and that's it :D |
| 00:37.00 | JT | Enhance 224 to 176. Enhance, stop. Move in, stop. |
| 00:37.07 | JT | don't need GUI for photoshop ;) |
| 00:37.36 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 00:39.41 | citywok | Jupe: apt-get install apache2 phpmyadmin |
| 00:39.47 | citywok | that should cover pretty much everything you want :P |
| 00:46.27 | *** join/#asterisk Jupe (~rez@fl-71-55-208-129.dhcp.embarqhsd.net) |
| 00:47.26 | SeRi | p3nguin, does arch core give you ethernet modules on a fresh install? |
| 00:47.53 | p3nguin | Yes. |
| 00:48.06 | SeRi | ok Thanks. |
| 00:50.51 | SeRi | installing it now. |
| 00:51.01 | p3nguin | ^5 |
| 00:51.27 | SeRi | :) I finally got the time. |
| 00:51.33 | Jupe | it's servername.conf |
| 00:52.07 | Jupe | I found that old tutorial i always c/p from |
| 00:52.39 | Jupe | interestingly enough, it told me it couldn't determine my hostname when i tried to symlink to /var/www |
| 00:54.29 | SeRi | is updating hes vibrant to a new r0m and installing Arch. something is bound to go wrong. |
| 00:55.10 | p3nguin | What kind of a jacked up distro breaks httpd.conf into pieces as small as servername.conf? |
| 00:55.18 | p3nguin | Let me guess, Ubuntu. |
| 00:55.24 | Jupe | no. i use Debian |
| 00:55.39 | Jupe | lol Ubuntu |
| 00:55.41 | p3nguin | Debian does that? |
| 00:55.56 | Jupe | yes. |
| 00:56.07 | SeRi | Jupe, what version of Debian? |
| 00:57.11 | Jupe | 6.0.3 |
| 00:57.43 | SeRi | are you using vhost? |
| 00:57.46 | Jupe | "squeeze" v 3 |
| 00:57.49 | Jupe | nope |
| 00:58.14 | SeRi | If you are not using vhost than it shouldnt split it that way.... |
| 00:58.32 | SeRi | I remember something about vhost been in seperate files like newserver.conf |
| 00:58.43 | SeRi | and so forth and so on for vhosts |
| 00:58.49 | WIMPy | So there's potential for freepbx :-) |
| 00:59.00 | SeRi | not for the main .conf file |
| 00:59.16 | SeRi | canonical names are set in the general area |
| 00:59.21 | Jupe | what? lol. i don't know much about vhosts with apache. i've never really used them |
| 00:59.29 | SeRi | Thats just weird from Debian |
| 00:59.35 | Jupe | plain ole hostname is in /etc/apache2/hostname.conf |
| 01:00.18 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 01:00.32 | Jupe | next time i have to set up apache2, i'll be leet and do 'sudo cat /etc/debian_version >> /etc/apache2/hostname.conf' =p |
| 01:00.49 | SeRi | WIMPy, I have notice that most people who have issues and mention ubuntu along those lines there is a freepbx install. |
| 01:01.25 | Jupe | someone's been trying to convince me to use FreeSwitch and not asterisk |
| 01:01.32 | Jupe | I'm like, well, i'll set up a freeswitch some other time |
| 01:01.39 | Jupe | I wanna do Asterisk first. lol |
| 01:02.19 | Jupe | one of the reasons why is this: http://www.projectmf.org/patches.html |
| 01:02.24 | Jupe | Bluebox patch :D |
| 01:03.04 | WIMPy | That also exists in LCR. |
| 01:04.21 | SeRi | fuck my monitor just died. |
| 01:04.32 | SeRi | now I have to go dig in the attic. |
| 01:07.24 | p3nguin | (1900.31) <Jupe> next time i have to set up apache2, i'll be leet and do 'sudo cat /etc/debian_version >> /etc/apache2/hostname.conf' =p <---- except that this command will not work, because the sudo won't pass through the redirection, and your regular user can't write to /etc/apache2/hostname.conf. :/ |
| 01:08.33 | p3nguin | But you could use sudo -i to get a root shell first, and then cat /etc/debian_version >> /etc/apache2/hostname.conf |
| 01:09.09 | SeRi | p3nguin, moving my install to a netbook for now. my damn monitor just died. |
| 01:09.29 | Jupe | spank you spanky helper =p |
| 01:09.48 | p3nguin | Netbook: Asterisk Edition? |
| 01:10.07 | SeRi | rofl. probably. |
| 01:10.20 | p3nguin | Asterisk: Netbook Edition? |
| 01:10.24 | SeRi | Its an Old HP NetBook I have. |
| 01:10.58 | SeRi | I installed aptosid long time ago. I am sure arch will work |
| 01:12.45 | Jupe | p3nguin i could also do 'cat /etc/version >> ~/blah | sudo mv ~/blah /etc/apache2/hostname.conf' =p |
| 01:14.18 | *** part/#asterisk daemonn (~Govna202@mrdreamer.com) |
| 01:19.10 | SeRi | pv is awesome |
| 01:19.57 | Jupe | speaking of tutorials, i'm looking for a good one on how to set up a home asterisk box, in case anyone has any recommendations. i was told by someone on #telephreak on 2600 IRC to check voip-info.org, so that's what i'm doing |
| 01:20.50 | SeRi | think Jupe is a robot-copypasta |
| 01:20.55 | SeRi | :) |
| 01:21.04 | Jupe | lol robot copypasta |
| 01:21.13 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 01:21.29 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 01:21.59 | Jupe | i copypasta whenever possible, because my hypothalamus wants me to forget everything important, like what i had for breakfast yesterday |
| 01:22.03 | Jupe | =p |
| 01:22.32 | SeRi | do you understand what you copypasta? |
| 01:22.44 | Jupe | and where the apache hostname conf file is at, and all of that good stuff. can't live without a web server. |
| 01:23.04 | Jupe | sometimes. i try to at least understand that what i'm copying and pasting is the right stuff, even if i don't understand all of it word-for-word |
| 01:23.15 | Jupe | i try to understand what all of it means. i just don't always remember. |
| 01:23.45 | Jupe | sometimes i'll figure out what it is I need to run, and then put it in a shell script for re-use later. |
| 01:24.54 | Jupe | so, in other words... [20:22] <SeRi> do you understand what you copypasta? <- what? :) |
| 01:26.00 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:26.09 | JT | Jupe: not /etc/hostname? |
| 01:26.44 | Jupe | JT no. |
| 01:26.53 | SeRi | moves to electro bugy and raps copypasta-robot |
| 01:27.01 | Jupe | that's a different hostname file. that's for the linux system hostname. |
| 01:27.15 | Jupe | seri: lol. copypast dubstep eh? |
| 01:27.37 | JT | Jupe: i just use vhosts for everything in apache generally |
| 01:27.41 | SeRi | thinks dubstep is c00l |
| 01:27.49 | SeRi | JT, +1 |
| 01:28.20 | Jupe | Jt i've never really used vhosts. i started reading about vhosts, and I was like, what? lol |
| 01:28.31 | Jupe | so i figured I would move on until i found a reason to use vhosts |
| 01:29.46 | Jupe | i'm pretty much just trying to figure out where and how i need to modify my asterisk conf files to get things they way i want them now :) |
| 01:29.54 | Jupe | this is my first asterisk PBX setup :D |
| 01:30.08 | Naikrovek | i always use virtualization unless i have a very good reason not to |
| 01:30.10 | SeRi | raps robot robot.... copypasta-robot w00t! |
| 01:30.31 | JT | Jupe: most websites are vhosts these days :P |
| 01:30.47 | JT | ServerName is the attribute you want i believe |
| 01:31.14 | SeRi | p3nguin, you in? |
| 01:31.33 | Jupe | what would you use apache vhosts for? |
| 01:31.48 | JT | more than one domain name per IP |
| 01:31.54 | JT | there's not many IPv4 IPs anymore |
| 01:32.26 | Jupe | ah i see. that's cool |
| 01:32.51 | SeRi | you can run as many sites as your server can handle and keep the seperate from each other |
| 01:33.12 | SeRi | vhost p2wn3s your Debian skills |
| 01:33.15 | Jupe | interesting. cool. |
| 01:33.29 | Jupe | i've never used more than one domain name per IP |
| 01:33.58 | SeRi | looks like people dont liek to seed arch :/ |
| 01:34.10 | SeRi | like* |
| 01:34.17 | SeRi | let me try the bot thing |
| 01:34.21 | Jupe | /\/0 17 p\/\//\/$ j00/2 |-|4><0/2 7y91/\/6 |
| 01:34.23 | Jupe | =p |
| 01:34.41 | Jupe | <- 0ldsk00l |
| 01:34.41 | SeRi | holy mother of bat man! is a h4x0r! |
| 01:34.44 | Jupe | ph33r. |
| 01:34.47 | Jupe | lol |
| 01:35.28 | SeRi | wow can I be your friend? |
| 01:35.55 | Jupe | lol |
| 01:35.59 | SeRi | :) |
| 01:36.19 | Jupe | NO! i'm too cool! I hang out with the "in" crowd. we stuff people in lockers all day |
| 01:36.44 | SeRi | :( |
| 01:37.16 | SeRi | p3nguin, when you get a chance msg me. Please. Thanks. |
| 01:50.25 | *** join/#asterisk nighty- (~nighty@69-165-220-105.dsl.teksavvy.com) |
| 01:51.41 | *** join/#asterisk hfb (~hfb@cpe-98-151-249-95.socal.res.rr.com) |
| 01:52.08 | *** part/#asterisk tessier (~treed@kernel-panic/copilotco) |
| 01:53.50 | *** join/#asterisk Jupe (~rez@fl-71-55-208-24.dhcp.embarqhsd.net) |
| 02:00.07 | autofsckk | good night, i have a warning on my * chan_sip.c:3351 __sip_xmit: sip_xmit of 0xb6e1bcf8 (len 390) to (null) returned -1: Invalid argument |
| 02:00.39 | autofsckk | there are a lot of warnings :S and an error ERROR[2083]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo("vsp.sskoip.com", "(null)", ...): Temporary failure in name resolution |
| 02:02.59 | autofsckk | well it seems that it cant register with my provider :S |
| 02:04.30 | SeRi | autofsckk, is dns broken in your system? |
| 02:04.48 | SeRi | autofsckk, can you ping vsp.sskoip.com? |
| 02:04.54 | SeRi | from your asterisk server |
| 02:06.53 | Jupe | autofsckk maybe you broke the internet. i did that once. |
| 02:06.55 | SeRi | do you have srvlookup=yes? |
| 02:07.24 | autofsckk | sorry i move from my desktop |
| 02:07.27 | SeRi | Jupe, you have 3lit3 h4x0r skiils. Its expected. |
| 02:07.42 | autofsckk | i think my provider banned my ip with fail2ban :S |
| 02:07.55 | SeRi | autofsckk, how did you determine that? |
| 02:09.14 | autofsckk | its connected now |
| 02:09.21 | Jupe | yeah. |
| 02:09.25 | autofsckk | SeRi: because it happened 2 days ago |
| 02:09.29 | Jupe | me and the l0pht took down the internet in 45 minutes |
| 02:09.29 | autofsckk | thanks for your help |
| 02:10.24 | p3nguin | #! |
| 02:13.36 | SeRi | autofsckk, lower your retry's |
| 02:13.50 | SeRi | p3nguin, you avail? |
| 02:14.15 | Jupe | crunchbang |
| 02:14.39 | autofsckk | SeRi: where do i change that? asterisk.conf? |
| 02:14.42 | SeRi | I just wanted to know where was you ast pkg located in the repo's... |
| 02:14.49 | SeRi | p3nguin, ^^ |
| 02:15.20 | autofsckk | p3nguin: thanks for helping me with the spa3102, it is working great now, i follow your advice and didnt upgrade the firmware |
| 02:18.03 | p3nguin | seri: What? |
| 02:18.15 | p3nguin | Oh, my asterisk? |
| 02:18.19 | SeRi | autofsckk, I am not sure I think is registerattempts= andregistertimeout= under sip.conf |
| 02:18.29 | SeRi | p3nguin, Yes I have Arch going |
| 02:18.40 | p3nguin | mkdir -p .build/asterisk |
| 02:18.47 | p3nguin | cd $_ |
| 02:19.01 | SeRi | ok |
| 02:19.42 | autofsckk | ok SeRi thanks |
| 02:19.47 | *** join/#asterisk TJNII (~TJNII@tjnii.com) |
| 02:20.13 | p3nguin | curl -O ftp://24.171.71.250/pub/linux/arch/asterisk/asterisk-1.8.7.1-1.src.tar.gz |
| 02:20.47 | p3nguin | tar xf asterisk-1.8.7.1-1.src.tar.gz |
| 02:20.51 | p3nguin | vim PKGBUILD |
| 02:20.58 | p3nguin | Look over the file. |
| 02:21.14 | SeRi | p3nguin, one sec. |
| 02:21.49 | p3nguin | crap |
| 02:22.11 | p3nguin | I made a mistake. |
| 02:22.19 | p3nguin | There's an asterisk directory in the tarball. |
| 02:22.43 | SeRi | hold on. my system craped out. looks like the hdd is not taking it :( |
| 02:22.51 | p3nguin | rm -fr asterisk |
| 02:22.57 | p3nguin | mv asterisk-1.8.7.1-1.src.tar.gz .. |
| 02:23.03 | p3nguin | cd .. |
| 02:23.05 | p3nguin | tar xf asterisk-1.8.7.1-1.src.tar.gz |
| 02:23.11 | p3nguin | THEN vim PKGBUILD |
| 02:23.14 | p3nguin | :/ |
| 02:23.21 | p3nguin | Wait, no... |
| 02:23.26 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:23.27 | p3nguin | I'm still telling you wrong. |
| 02:23.37 | p3nguin | you have to cd asterisk after tar. |
| 02:23.44 | p3nguin | Then you can vim the PKGBUILD |
| 02:23.53 | p3nguin | needs more beerz |
| 02:24.08 | Jupe | i'm tryin to just get a basic extension set up, dial into my asterisk box and mess with stuff. it's a lot to learn for a newbie :D |
| 02:24.47 | p3nguin | seri: Do I need to take it from the top, or will you fix what I forgot? |
| 02:25.05 | p3nguin | jupe: Extensions don't dial in. Extensions are run when phones dial in. |
| 02:25.11 | Jupe | i know |
| 02:25.16 | SeRi | Ill fix it. Thanks for the info. I just need it the build. |
| 02:25.19 | SeRi | Thanks p3nguin |
| 02:25.27 | LiuYan | ~monitor |
| 02:25.27 | infobot | rumour has it, monitor is A device for viewing the output from a computer, traditionally a much more precise TV set. |
| 02:25.46 | Jupe | i'm familiar with the concept of a PBX extention. i've never set up asterisk before |
| 02:26.07 | SeRi | p3nguin, you take donations? Ill buy you a beer right now via pay pal ;) |
| 02:26.23 | p3nguin | Once you look over the PKGBUILD, you'll probably see a couple things you want to deal with in there. Edit accordingly. When done, save and exit. Then run makepkg. |
| 02:26.32 | LiuYan | ~mixmonitor |
| 02:26.37 | SeRi | p3nguin, got it! |
| 02:26.51 | p3nguin | I'm guessing you probably need to take care of some dependencies, though. |
| 02:26.55 | Jupe | there's a dude i know who's handle is penguin, and he does a lot of cool stuff with Asterisk. |
| 02:27.09 | SeRi | p3nguin, I am sure. |
| 02:27.23 | p3nguin | makepkg should tell you what you're missing. |
| 02:27.25 | SeRi | Ill have to bring a monitor from work. the netbook didnt survive |
| 02:27.28 | Jupe | i'm not sure where to begin. lol |
| 02:27.32 | p3nguin | It died, too? |
| 02:27.41 | SeRi | The hdd is having issues during part |
| 02:27.59 | Jupe | kind of wish i wasn't running this on a VM, to tell you the truth |
| 02:28.06 | p3nguin | You want to know what *I'm* having issues with? |
| 02:28.26 | SeRi | p3nguin, I dare to ask? :) |
| 02:28.32 | Jupe | p3nguin women? |
| 02:28.40 | Jupe | or was that answer too easy? |
| 02:29.25 | Jupe | infobot: die |
| 02:29.25 | infobot | ACTION takes two shots to the head and crumples to the ground, lifeless. |
| 02:29.26 | p3nguin | I wanted to use a mic on my iPod touch 3rd gen. I borrowed the mic from my wife's 2nd gen, which she uses regularly. When I connect it to my 3rd gen, it does not work -- peck, peck peck, peck peck, peck, peck peck. Broken. |
| 02:29.31 | Jupe | hehe |
| 02:29.49 | SeRi | o! that sucks nuts! |
| 02:30.03 | p3nguin | So I got another 3rd gen to replace what I thought was a broken iPod. |
| 02:30.10 | p3nguin | This new one does the exact same thing! |
| 02:30.14 | p3nguin | What gives?! |
| 02:30.27 | SeRi | iPod. |
| 02:30.28 | Jupe | broken or incompatible mic? |
| 02:30.39 | SeRi | j/k |
| 02:30.55 | Jupe | incompatible mic, probably |
| 02:30.56 | p3nguin | How could a working mic compatible with a 2nd gen be a broken incompatible mic on a 3rd gen? |
| 02:31.20 | Jupe | i don't know. i don't have an iphone, so i've never got to show anyone my broken iphone =p |
| 02:31.25 | p3nguin | It should be the same jack. |
| 02:31.40 | Jupe | i like the keypad on the iphone |
| 02:31.46 | Jupe | QWERTY ftw |
| 02:32.06 | SeRi | p3nguin, I can probably help. call the conf. I am working on the Arch setup so I am not paying attention to the mon. |
| 02:33.31 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-hkfjhkgbblupkfya) |
| 02:42.28 | SeRi | p3nguin, looks like I revivedthe hdd. is all going again. |
| 02:42.36 | SeRi | waiting fir it to finish building |
| 02:42.40 | SeRi | for* |
| 02:43.43 | SeRi | so far likes pacman |
| 02:44.52 | Jupe | i like playing PacMan on Nestopia |
| 02:45.19 | Jupe | PacMan, Tetris, Super Mario Bros 1 - 3, Friday the 13th, Snake Rattle 'n Roll |
| 02:46.16 | SeRi | face palm. |
| 02:48.02 | autofsckk | SeRi: install yaourt or clyde |
| 02:48.44 | p3nguin | s/yaourt or clyde/cower/ |
| 02:49.12 | SeRi | lmao |
| 02:49.18 | p3nguin | or packer, if you'd prefer it. |
| 02:49.48 | SeRi | autofsckk, you use arch? |
| 02:49.54 | autofsckk | yes i do |
| 02:50.06 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 02:51.19 | SeRi | so far pacman suits my needs |
| 02:51.33 | autofsckk | SeRi: what do you use? |
| 02:51.34 | p3nguin | It will until you need things from AUR. |
| 02:51.50 | p3nguin | Then you'll be asking how to get stuff from AUR... and the answer is cower or packer. |
| 02:51.55 | SeRi | I am a slack guy |
| 02:52.23 | SeRi | p3nguin, I see |
| 02:54.36 | *** join/#asterisk slidesinger-lt (~jtatum@173-161-172-121-Philadelphia.hfc.comcastbusiness.net) |
| 02:55.18 | p3nguin | This is really irritating. I can't even use my square reader. |
| 02:56.55 | SeRi | well shit I didnt install dhclient... LOL |
| 02:57.20 | autofsckk | dhcpcd on asterisk |
| 02:57.43 | p3nguin | YOu don't use dhclient on arch. |
| 02:58.01 | SeRi | well tahts different |
| 02:58.13 | p3nguin | Use dhcpcd, but don't use it directly. Configure the network in /etc/rc.conf and then use /etc/rc.d/network start |
| 02:58.19 | p3nguin | or rc.d start network? |
| 02:58.34 | SeRi | rc.d network start |
| 02:58.47 | p3nguin | I was thinking it was backward for some reason. |
| 02:58.51 | p3nguin | I don't use it, so I don't know. |
| 02:59.17 | SeRi | cool. well its online. I am going to move it to my other network to start the configuration on it. |
| 02:59.27 | autofsckk | you can configure it on rc.conf, its pretty easy |
| 03:00.09 | SeRi | I am sure. |
| 03:01.04 | autofsckk | SeRi: http://wiki.archlinux.org/ its the best wiki i have seen so far |
| 03:01.35 | p3nguin | And no, dhcpcd doesn't really work any differently for the end user... dhcpcd eth0. |
| 03:01.51 | p3nguin | Oh, goody... my square reader does work on the replacement iPod. |
| 03:02.29 | p3nguin | So maybe the mic really isn't compatible with a 3rd gen. How weird. |
| 03:02.40 | SeRi | dhclient is just a wrapper I know. |
| 03:02.43 | SeRi | ;) |
| 03:03.17 | p3nguin | It is? |
| 03:03.21 | SeRi | Yes. |
| 03:03.25 | p3nguin | For what? |
| 03:03.54 | SeRi | well fuck I was wrong. it is not is a static bin |
| 03:03.56 | p3nguin | This is the first I'm hearing about it being a wrapper for anything. |
| 03:04.06 | p3nguin | I'm just saying YOU DON"T NEED IT. |
| 03:04.10 | p3nguin | There's no reason to install it. |
| 03:04.18 | p3nguin | YOu have dhcpcd by default. Use it. |
| 03:04.35 | SeRi | I guess I got it confused. for some reason I thought in another sys dhclient was a wrapper.... never mind |
| 03:04.41 | SeRi | Yea I saw that is not need it |
| 03:04.50 | SeRi | I have my network going |
| 03:06.26 | autofsckk | i have never used slack |
| 03:07.06 | SeRi | not too far from arch in a way.... |
| 03:07.42 | SeRi | things seem a bit simpler as far as pkg's go. |
| 03:08.15 | p3nguin | Wait until you get into AUR. |
| 03:08.22 | autofsckk | is it rolling release too? |
| 03:08.29 | p3nguin | Slackware isn't. |
| 03:08.30 | autofsckk | yes, aur is great |
| 03:08.47 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 03:09.30 | SeRi | autofsckk, no they dont do like ubuntu etc.... |
| 03:10.14 | p3nguin | Scheduled releases? |
| 03:10.24 | SeRi | only stable release and patch fixes. you are in charge of your own pakg's and the maintenance behind it. including your kernel |
| 03:10.28 | SeRi | p3nguin, Yes |
| 03:11.11 | SeRi | 12.1, 13.1, 13.37 <---- that one was odd |
| 03:11.37 | SeRi | an example ^^ |
| 03:11.54 | SeRi | or maybe I got it backwards |
| 03:11.55 | SeRi | lol |
| 03:12.37 | SeRi | p3nguin, AUR? |
| 03:12.43 | autofsckk | this install is from 2009 i think, i changed my hd from 160 to 250, changed from ext4 to xfs on some partitions, and made an lvm too :S |
| 03:12.57 | p3nguin | Arch User Repository |
| 03:13.09 | SeRi | Yea I just saw that at the wiki |
| 03:13.24 | p3nguin | It's where all of the unsupported, user contributed stuff is. |
| 03:13.45 | p3nguin | If someone else didn't already have an asterisk in there, I would have mine in AUR. |
| 03:13.46 | SeRi | cool |
| 03:14.55 | autofsckk | SeRi: if you liked pacman, you're gonna love yaourt |
| 03:15.21 | p3nguin | s/yaourt/cower/ |
| 03:15.30 | p3nguin | We don't use yaourt anymore. |
| 03:15.58 | SeRi | I am sure I am going to like them all since automated pkg system in slackware is not huge and very obscure |
| 03:15.59 | autofsckk | why not? |
| 03:16.10 | p3nguin | Last I knew, it was broken anyway. |
| 03:17.12 | autofsckk | yaourt is working with no flaws, the one that was broken was clyde, but i think i saw that is fixed now |
| 03:25.44 | SeRi | Man this is dependency resolution is awesome! Is like yum :) |
| 03:26.53 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:30.33 | SeRi | asterisk is installing |
| 03:30.37 | SeRi | fail2ban installing |
| 03:30.47 | SeRi | iptables is going |
| 03:30.49 | p3nguin | How are you installing asterisk? |
| 03:31.01 | SeRi | makepkg |
| 03:31.07 | p3nguin | That doesn't install it. |
| 03:31.15 | SeRi | You know what i mean |
| 03:31.18 | SeRi | it builds the pkg |
| 03:31.19 | p3nguin | That simply makes the package. |
| 03:31.33 | p3nguin | But you can use makepkg -i to make and install. |
| 03:31.44 | SeRi | cool. Thanks. |
| 03:31.53 | p3nguin | or the way I usually do, makepkg; pacman -U my-new-pkg |
| 03:32.16 | autofsckk | makepkg -si |
| 03:32.57 | SeRi | I am sure ill have to rebuild it again for what ever reason... LOL |
| 03:33.36 | SeRi | so far the build is going |
| 03:33.43 | SeRi | ah! |
| 03:33.44 | p3nguin | Did you edit the pkgbuild? |
| 03:33.54 | SeRi | yes I just add it mp3 |
| 03:34.20 | p3nguin | I have another source package with a patch for mp3, too, but I haven't uploaded it. |
| 03:34.37 | p3nguin | I don't know if you saw I have patches included. |
| 03:34.47 | SeRi | ah! :) that would be useful :) |
| 03:34.58 | SeRi | yes I did. and I can see them been download it |
| 03:37.36 | SeRi | This is a Dual Core Atom/2GB RAM/60GB HDD/netbook so I am sure it will run ok. |
| 03:38.37 | SeRi | mhhhhhh I got an idea for a modd. |
| 03:38.43 | p3nguin | I run arch with asterisk on an 800MHz CPU, 256MB RAM, 4G flash system, so I'm sure you'll be fine. |
| 03:39.01 | SeRi | nice. |
| 03:39.20 | autofsckk | i run it on my asus 900ha and runs great |
| 03:39.21 | SeRi | Thats some nice specs for an embedded system :/ |
| 03:40.08 | p3nguin | But I also run Arch on my Core 2 Quad, 8G RAM, 2x 2TB system as well. |
| 03:40.24 | SeRi | p3nguin, simmer down killer! |
| 03:40.26 | SeRi | LOL |
| 03:40.41 | p3nguin | That's my desktop. |
| 03:40.47 | SeRi | nice specs |
| 03:41.16 | p3nguin | Core 2 Quad Q9550 |
| 03:41.25 | SeRi | my office: http://www.dslreports.com/forum/r26216477-Home-Office |
| 03:41.48 | SeRi | upgrades: http://www.dslreports.com/forum/r26284766-Network-Upgrade |
| 03:42.15 | SeRi | Ill be redoing my network this week end... I am hoping at least |
| 03:42.23 | SeRi | *cabling* |
| 03:43.00 | SeRi | Those are old pics so my network has grown now. |
| 03:45.04 | SeRi | is still "making" asterisk |
| 03:45.36 | SeRi | I am going to leave this system as is. it will be only for asterisk and some asterisk devel. |
| 03:46.09 | SeRi | once I get use to it I might move to my desktop... dont see that happening any time soon though.... |
| 03:46.11 | carrar | You need a bigger UPS |
| 03:46.22 | SeRi | carrar, I did uograded it my ups |
| 03:46.25 | SeRi | 1300AVR |
| 03:46.33 | SeRi | not in the pics :) |
| 03:46.49 | SeRi | haven updated the pics with the new stuff |
| 03:46.57 | *** join/#asterisk master_of_master (~master_of@p57B52F27.dip.t-dialin.net) |
| 03:47.08 | SeRi | <SeRi> Those are old pics so my network has grown now. <------ havent updated it for a while :) |
| 03:48.12 | p3nguin | I have only one thing to say after looking at the pics and reading the posts. |
| 03:48.56 | p3nguin | its use, not it's use. "it is" use does not make any sense. |
| 03:49.25 | SeRi | really? Thats it? |
| 03:49.30 | p3nguin | That is all. |
| 03:49.44 | SeRi | uffff man I thought you where ready to burn me. |
| 03:49.51 | SeRi | is happey |
| 03:49.51 | p3nguin | I'm really irritated about people thinking it's means its. |
| 03:49.52 | *** join/#asterisk freakazoid0223 (~matt@pool-173-49-209-91.phlapa.fios.verizon.net) |
| 03:50.10 | p3nguin | it's means IT fucking IS, not something belongs to it. |
| 03:50.40 | SeRi | well English is not my native language. I didnt even know it was different. I apologize. |
| 03:50.48 | p3nguin | Oh? |
| 03:51.10 | SeRi | seriously I didnt know :( |
| 03:51.34 | p3nguin | its = possessive pronoun to show it owns something. E.g., its paint; its ears |
| 03:51.46 | SeRi | I got burned before for some fucked up typos that even makes me laugh LOL |
| 03:51.56 | SeRi | AH! |
| 03:52.05 | SeRi | well thank you p3nguin! |
| 03:52.13 | p3nguin | it's = contraction for "it is" or "it has." E.g., It's a wonderful day in the neighborhood. |
| 03:52.47 | p3nguin | or... It's going to be a shitty day in the neighborhood! |
| 03:53.07 | [TK]D-Fender | p3nguin: I'll be keeping an eye out for news reports of a grammar-nazi with a high-powered rifle on a water tower overlooking a middle-school ;) |
| 03:53.32 | p3nguin | You know I'll make the TV news. |
| 03:53.34 | SeRi | ah. wow. well thanks. now I know. like These and This and There and Their they all almost sound the same and all ways get me in trouble |
| 03:54.02 | [TK]D-Fender | p3nguin: http://www.youtube.com/watch?v=OonDPGwAyfQ |
| 03:55.33 | SeRi | lmao @ p3nguin |
| 03:55.44 | SeRi | I mean [TK]D-Fender |
| 03:55.54 | p3nguin | /: |
| 03:56.17 | SeRi | I hope you like jelapanos. maybe you dont shoot me. |
| 03:56.30 | SeRi | I got plenty in my backyard. :P |
| 03:56.32 | p3nguin | I do like them. |
| 03:56.55 | p3nguin | I like them stuffed with cream cheese, wrapped in bacon, and baked. |
| 03:57.26 | SeRi | come to my BBQ and you will have the best ones you would ever have eaten! |
| 03:58.24 | p3nguin | I also like habaneros, but I typically fire roast them and put them into my chili. |
| 03:58.28 | SeRi | my wife loves when I make them the exact same way you describe. the trick is to snap the top and take out "some" seeds but not all. the cheese will take care of the hot.... |
| 03:59.04 | p3nguin | So don't slice them in half? |
| 03:59.21 | SeRi | It has a nice blend of fucking hot and still eatable... the cheese kicks in and simmers your fucked in fire mouth |
| 03:59.37 | SeRi | p3nguin, first mistake. dont cut them in half if your going to BBQ them |
| 04:00.08 | SeRi | fucking in fire* |
| 04:00.53 | SeRi | we have habaneros and jelapanos verdes |
| 04:01.21 | p3nguin | I'm not quite sure what that means. |
| 04:01.43 | p3nguin | I thought that means green. |
| 04:01.43 | SeRi | the problem with cutting them in half if you bbq theme the cheese will fall off if you try to cook them well |
| 04:01.58 | SeRi | green jalapenos* |
| 04:02.35 | SeRi | You could also se foil to get around it. and let the bacon juice do the rest.... but is not the same |
| 04:02.43 | SeRi | let the* |
| 04:02.49 | SeRi | use* |
| 04:03.42 | p3nguin | I like my habaneros red or orange, but I'm good with green jalapenos. |
| 04:04.32 | SeRi | :) |
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| 04:08.21 | SeRi | pacman -U asterisk-1.8.7.1-1-i686.pkg.tar.xz |
| 04:08.59 | LiuYan | ~curl proxy |
| 04:09.10 | SeRi | Mem in use 40MB |
| 04:09.15 | SeRi | nice :/ |
| 04:09.16 | SeRi | :) |
| 04:09.30 | LiuYan | ~proxy |
| 04:09.30 | infobot | [proxy] This is commonly a form of Internet security. You can use a proxy or proxy server to pass data between your internal network and the Internet. A machine on your network sends a request to the proxy. The proxy sends the request to a server on the Internet. Thus, it stands in for the computer on your network. The server on the Internet never knows that the request is coming from anywhere but the proxy. Thus 100 machines on your network could all ... |
| 04:13.21 | SeRi | p3nguin, all done :) |
| 04:13.53 | SeRi | taking a small brake. |
| 04:19.42 | SeRi | [root@archaista ~]# |
| 04:20.22 | SeRi | its beautiful only 40MB of ram. |
| 04:21.02 | SeRi | carrar, http://www.cyberpowersystems.com/products/ups-systems/intelligent-lcd-ups/cp1350avrlcd.html |
| 04:21.12 | SeRi | Thats the new ups |
| 04:21.17 | SeRi | well not so new. |
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| 05:03.10 | ChannelZ | "screen is simulated" |
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| 05:10.24 | SeRi | http://americancensorship.org/ |
| 05:24.08 | Kobaz | hmm |
| 05:24.25 | Kobaz | seems like you can't query channel ringing status via CHANNEL |
| 05:26.20 | SeRi | p3nguin, you avail? |
| 05:30.17 | p3nguin | la la la la la |
| 05:30.53 | SeRi | lol |
| 05:31.17 | SeRi | is it possible to turn on and off recording via keys? |
| 05:31.49 | SeRi | key press* |
| 05:32.29 | SeRi | I couldn't find any reference on that while the conference is going. |
| 05:32.56 | p3nguin | Yes and no. |
| 05:33.07 | p3nguin | It depends on what recording you are doing. |
| 05:33.27 | p3nguin | I don't think you can stop MixMonitor() mid-call. |
| 05:33.49 | SeRi | ah. I see. that was the question |
| 05:34.06 | SeRi | Mhhhhhhh interesting. |
| 05:38.54 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 05:50.09 | SeRi | Thanks and g/n |
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| 06:07.48 | SeRi | well for got about my cell doing the r0m update :/ |
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| 06:19.22 | LiuYan | ~tts |
| 06:19.23 | infobot | methinks tts is time to sleep, or text to speech |
| 06:19.51 | LiuYan | ~espeak |
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| 07:25.50 | schmidts | good morning |
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| 07:47.26 | wdoekes2 | morning |
| 07:48.17 | tick | morning |
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| 07:57.18 | schmidts | i have a question to you guys about fraud. i have read in the FCFA Fraud Report for 2011 that around 3% of Fraud terminates to Austria (+43) has anyone of you ever seen a fraud to this destination? |
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| 07:57.59 | schmidts | btw this means austria is on the 6.place of destinations, first one is cuba (what else) :D |
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| 08:44.35 | OldSmurf | I am looking for a way to test the maximum capacity on my meetme installation. What tools are available that can help me with this? |
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| 08:45.26 | tuxx- | OldSmurf: sipp maybe |
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| 08:55.46 | irroot | handy man bible .... if it moves and it should not tape it ... if it does not move and it should hit it with hammer |
| 08:55.53 | irroot | morning folks |
| 08:57.01 | coppice | Th gospel according to duct tape makers |
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| 09:50.02 | Sakuranbo | hello, I need some help |
| 09:50.54 | Sakuranbo | I had a user who activated the call forwarding function on the phoneset and we are in seperate geographical offices connected by 4M MPLS |
| 09:51.36 | Sakuranbo | we havent implemented any voice compression over the WAN link |
| 09:51.59 | Sakuranbo | but whoever calls the far end, the ringtone is distorted |
| 09:52.23 | Sakuranbo | and on the asterisk console I have the following error |
| 09:53.19 | Sakuranbo | " Dropping incompatible voice frame on Local/992307620@default-209f,2 of format slin since our native form at has changed to ilbc " |
| 09:54.19 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 09:55.29 | FlashDeluxe | hi! i got a problem, if i want to start asterisk i get an error chan_capi.c:7834 cc_init_capi: CAPI not installed, chan_capi disabled! i start asterisk as root, so there shouldn`t be a permission problem, does anybody got a hint for me? i am using asterisk 1.6 with current chan_capi |
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| 10:10.53 | tompaw | Morning guys |
| 10:11.54 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:13.02 | tompaw | I have a small problem with my DP: http://pastecode.com/eP - I have a queue+confbridge configuration which puts an agent on a call within a conference room. The only problem is - that agent is not properly marked as "busy", and queue doesn't connect following calls to another agent. |
| 10:13.20 | tompaw | I have therefore added a manual agent pause (line 27 in the pastebin). |
| 10:13.34 | tompaw | The only question is - how can I capture the agent hanging up and unpause him? |
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| 10:30.39 | schmidts | just want to put a little attention to this: http://www.avaaz.org/en/save_the_internet_d/ take 1 minute to read and sign it, its worth doing it! |
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| 10:31.59 | irroot | tompaw use custom device states in your case save you many headaches |
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| 11:13.27 | jkroon | hi guys, what can be done re the asthostid keep changing on some systems? |
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| 11:34.41 | dandate2 | im having an issue with a pap2 voip gateway; it won't recognize the blind transfer code whereas other voip adapters will? |
| 11:36.45 | dandate2 | we set the In-Call Asterisk Blind Transfer to ## but when i dial that through a pap2 nothing happens |
| 11:37.14 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 11:39.03 | Chainsaw | dandate2: And how are you sending DTMF to Asterisk for this PAP2 SIP peer? |
| 11:39.27 | *** join/#asterisk irroot (~gregory@dsl-185-122-74.dynamic.wa.co.za) |
| 11:42.59 | dandate2 | dtmf mode says auto |
| 11:43.39 | jkroon | and what is the ATA set to? try using rfc2833. |
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| 11:48.28 | dandate2 | on the asterisk server the extentions are set to rfc2833 |
| 11:49.48 | jkroon | dandate2, then force it on the ata too |
| 11:50.22 | dandate2 | shoot i just dont see any option for that in the pap2 setting |
| 11:50.54 | dandate2 | RFC 2543 Call Hold: |
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| 11:53.02 | *** join/#asterisk cfchris6 (~nihilus@unaffiliated/cfchris6) |
| 11:53.32 | jkroon | no, it has to be there. sure of it. |
| 11:55.14 | cfchris6 | hm, I am running asterik 1.8 on debian here. somehow, the originate command does not exist. Has it been renamed/moved recently? (details at http://pastebin.com/2efQXfLD) |
| 11:55.22 | dandate2 | would that be under SIP |
| 11:55.23 | dandate2 | Provisioning |
| 11:55.23 | dandate2 | Regional |
| 11:55.23 | dandate2 | L1 ? |
| 11:55.28 | jkroon | channel originate |
| 11:56.12 | cfchris6 | jkroon: m( thanks |
| 11:56.37 | jkroon | check cli_aliasses.conf iirc. originate should still be aliassed there. |
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| 12:03.42 | dandate2 | i compared the non-functioning pap2 to a working one, only difference in settings i could find is that the non-functioning one didnt have all the Vertical Service Activation Codes populated |
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| 12:20.16 | dandate2 | ok so i made a bit of a breakthrough, it seems incall blind transfer works for the user receiving a call, but not if he had dialed out. is this a known issue? |
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| 12:29.51 | *** join/#asterisk DanFromUK (~DanFromUK@2.27.1.180) |
| 12:31.06 | DanFromUK | Hi. Currently, when i make a call to an IVR, from an asterisk SIP Peer, if I need to press #, the local asterisk says "transfer" and starts a blind transfer process. Is there any way to remove that? |
| 12:34.50 | dandate2 | need to edit your feature codes then and change transfer to ## or the likes |
| 12:34.57 | kaldemar | DanFromUK: sure, remove option T from your Dial or Queue command. or change the blindxfer value in features.conf to be something else than #. |
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| 12:42.56 | tompaw | Hm... is there a way to list confbridge conferences? |
| 12:42.57 | DanFromUK | ah, i didnt realise blindxfer defaults to #. I just commented it out, when in fact, i have to change it completely. |
| 12:43.16 | DanFromUK | any way to reload features.conf without a full reload? |
| 12:43.58 | DanFromUK | got it |
| 12:44.00 | DanFromUK | thanks all |
| 12:46.56 | kaldemar | tompaw: confbridge <TAB><TAB> |
| 13:03.38 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:09.45 | tuxx- | hi guys, what does the realtime sipfriend 'allow' field expect? I got 'all' in that field, but when i try to set up a call to asterisk i get the following: chan_sip.c:8897 process_sdp: No compatible codecs, not accepting this offer! |
| 13:09.54 | tuxx- | this is the line from the sip invite: Capabilities: us - 0x0 (nothing), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) |
| 13:10.11 | tuxx- | asterisk does not accept ANY codec it seems |
| 13:14.49 | tuxx- | hm weird, even when i put 'alaw;ulaw;gsm' in the allow field, and 'all' in disallow, it still says i have no codecs: Codecs : 0x0 (nothing) |
| 13:14.50 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:14.53 | tuxx- | >_> |
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| 13:25.59 | jkroon | is there a function for DAHDI similar to SIPPEER ? |
| 13:26.13 | irroot | dandate2 you need to see the the t/T options to Dial |
| 13:26.41 | irroot | jkroon in what way |
| 13:26.42 | jkroon | specifically I want to be able to retrieve the accountcode and some channel variables. |
| 13:27.19 | irroot | jkroon the CDR / CHANNEL function will be the place to go what channel variables ?? |
| 13:27.19 | *** join/#asterisk lcat (~lcat@187.45.254.21) |
| 13:27.20 | jkroon | ie, something like DAHDICHAN(1,accountcode), similar to what I would do SIPPEER(jkroon,accountcode) |
| 13:27.54 | dandate2 | right now its set to tr |
| 13:27.56 | irroot | jkroon ah ok so its for random channels nope dont think so |
| 13:27.59 | dandate2 | should i change it to tTr |
| 13:28.48 | irroot | dandate2 tT will allow anyone to transfer in or out this is not "safe" you ideally want it set appropriately |
| 13:29.01 | irroot | so only "trusted" parties can do this |
| 13:29.30 | dandate2 | we use a different feature code than # for transfer |
| 13:29.40 | jkroon | irroot, yes, unfortunately. |
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| 13:31.17 | jkroon | ok crap, that means I need to jump a few loops here :( |
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| 13:43.33 | irroot | jkroon that sort of thing i do by func_odbc into the base of config that lands up in config files |
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| 13:46.26 | jkroon | irroot, i know, it's one of those _should_ cases, but the rest of the design isn't done that way at the moment, so it's kinda hard to start doing it now. |
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| 13:46.51 | irroot | always is .... |
| 13:47.02 | jkroon | although, this might be an "easy" use case in this case, if I could write queries that recursively traverses tables ... |
| 13:49.34 | tompaw | kaldemar: I don't have a "confbridge" command in my CLI :/ |
| 13:49.43 | tompaw | dunno why, its the latest 1.8.x |
| 13:50.38 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
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| 14:05.28 | dandre | Hello |
| 14:07.38 | *** join/#asterisk BuenGenio (~Gene@152.101.3.33) |
| 14:07.47 | dandre | I have difficulties to get MWI working. I have some pending voicemail messages (shown by voicemail show users). I have set my phone to subscribe to mwi but this doesn't show any indication of mwi sugnal. |
| 14:07.59 | dandre | sip sow mwi shows nothing |
| 14:08.31 | dandre | sip shows notifications show a mwi registration |
| 14:08.57 | dandre | what can I do to see where my mistake is? |
| 14:14.30 | jkroon | do you have a mailbox parameter set up in the SIP peers ? |
| 14:19.34 | irroot | bacon4leif sharing is caring :P |
| 14:19.43 | bacon4leif | irroot: heck ya :) |
| 14:20.14 | [TK]D-Fender | na na na na na |
| 14:20.43 | dandre | jkroon: no but sip show peer 53 shows this: |
| 14:20.45 | dandre | <PROTECTED> |
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| 14:26.02 | jkroon | you have appropriate hints set up? |
| 14:26.36 | dandre | yes |
| 14:26.44 | dandre | I have a warning: |
| 14:26.53 | dandre | WARNING[18258]: chan_sip.c:18552 handle_response: Remote host can't match request NOTIFY to call '3b4a-c0a80101-d-2@192.168.0.143'. Giving up. |
| 14:27.03 | dandre | what does that means? |
| 14:27.37 | jkroon | i'm not the best person to be asked that question, but as I understand it it means that there is an issue matching up the subscribes and the notifies. |
| 14:29.00 | p3nguin | A SIP peer named 53? Horrid! |
| 14:29.11 | dandre | why? |
| 14:29.31 | p3nguin | How is that unique and significant to the device? |
| 14:29.48 | p3nguin | I bet your extension to reach it is also 53. |
| 14:29.57 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-bbvdqogkyuexbdwl) |
| 14:30.05 | p3nguin | ~devicenames |
| 14:30.05 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 14:30.15 | dandre | yes |
| 14:35.46 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:37.58 | [TK]D-Fender | <jkroon> do you have a mailbox parameter set up in the SIP peers ? <dandre> jkroon: no but sip show peer 53 shows this: |
| 14:38.07 | [TK]D-Fender | dandre, Fix your peer. |
| 14:41.30 | jkroon | can VALID_EXTEN deal with labels for piorities? eg, VALID_EXTEN(randomcontext,${EXTEN},${var}) where var=asdf type of thing? |
| 14:42.09 | [TK]D-Fender | sounds like a safe bet |
| 14:42.45 | jkroon | interesting ... because I can't seem to get it working :( |
| 14:43.18 | p3nguin | Show us evidence of a problem. |
| 14:43.51 | dandre | I have put mailbox=53@default but same result |
| 14:44.57 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 14:44.57 | *** mode/#asterisk [+o file] by ChanServ |
| 14:45.02 | jkroon | http://pastebin.com/65F7V162 |
| 14:45.13 | jkroon | p3nguin, i'd be very happy to be wrong in this case. |
| 14:45.14 | p3nguin | And you remembered to run "sip reload" after saving that change? |
| 14:47.26 | tompaw | Is there a way to record a ConfBridge directly? |
| 14:47.27 | p3nguin | You have dahdi in the VALID_EXTEN, but DAHDI as the label. |
| 14:49.39 | jkroon | p3nguin, crap, will recheck that quick thanks. |
| 14:49.59 | jkroon | i know that was one of the crazy ideas I tested. |
| 14:52.31 | dandre | in sip show subscriptions I have this: |
| 14:52.32 | dandre | 192.168.0.143 53 110958-c0a80101 -- <none> mwi 53,53@defa 003600 |
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| 14:55.28 | jkroon | p3nguin, http://pastebin.com/ihh6PyNY |
| 14:56.37 | Katty | shivers. |
| 14:57.19 | jaytee | wraps a blanket around Katty and hands her a hot toddy |
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| 15:00.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:03.07 | jkroon | p3nguin, [TK]D-Fender - anything? I'm at a loss, it should work (and if I allow it to drop through the Hangup() at line 4 it does ... |
| 15:03.22 | grharry | Hey all, I've installed asterisk 1.8.xx from the debian asterisk repo and trying to setup freepbx with it however I am having a difficulty to get asterisk manager to start and freepbx script "retrive_conf" fails ... with "Unable to connect to manager localhost:5038" |
| 15:04.42 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
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| 15:09.33 | [TK]D-Fender | jkroon, show your latest version... |
| 15:09.50 | jkroon | http://pastebin.com/ihh6PyNY |
| 15:09.57 | *** join/#asterisk bchia (~Adium@nat/digium/x-rrqkaatlcreiagqp) |
| 15:13.17 | [TK]D-Fender | jkroon, Looks legit... PB the app instructions |
| 15:13.18 | [TK]D-Fender | and |
| 15:13.27 | [TK]D-Fender | [DAHDI] 5. Set(chn=${CUT(CUT(CHANNEL(name),/,2),-,1)}) [pbx_config] <--$ error in 2nd cut |
| 15:13.46 | [TK]D-Fender | Shouldn't actually interfere though |
| 15:13.53 | [TK]D-Fender | but should get fixed |
| 15:17.11 | *** join/#asterisk nanoha-sama (~nanoha-sa@curriegrad2004-3-pt.tunnel.tserv14.sea1.ipv6.he.net) |
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| 15:23.37 | jkroon | [TK]D-Fender, funny enough, that line does work. |
| 15:24.49 | [TK]D-Fender | jkroon, I think I see why.. because CUT evals s the 1st parm which is normally the var name, not the evaluation of it |
| 15:24.56 | *** part/#asterisk grharry (~harry@ppp-94-65-231-247.home.otenet.gr) |
| 15:24.56 | [TK]D-Fender | Just looks damn freaky |
| 15:25.18 | jkroon | [TK]D-Fender, DIALPLAN_EXISTS seems to work. |
| 15:25.42 | jkroon | [TK]D-Fender, so you suggest just splitting it out into two steps? |
| 15:26.28 | [TK]D-Fender | What did you do to fix the DIALPLAN_EXISTS issue? |
| 15:26.43 | jkroon | jus swapped VALID_EXTEN for DIALPLAN_EXISTS |
| 15:26.49 | [TK]D-Fender | oops... |
| 15:26.56 | [TK]D-Fender | yeah, just noticed the reversal |
| 15:27.27 | jkroon | forgot about VALID_EXTEN being deprecated, figured i'd take a chance and see. |
| 15:27.27 | [TK]D-Fender | <PROTECTED> |
| 15:27.36 | jkroon | yes |
| 15:28.01 | [TK]D-Fender | jkroon, No, what you're doing is perfectly valid. Just stating the oddities that are *'s parsing engine |
| 15:28.44 | jkroon | ok, i'm missing the oddity, i'm aware that CUT takes the first param and evals it, that's why I didn't ${CUT()} the inner one ... |
| 15:29.40 | r0m|u | I just laugh at the silence in #elastix |
| 15:29.57 | jkroon | hmm, although, it could possibly have grabbed the first , for the inner CUT as the delimeter for the outer CUT come to think about it. |
| 15:29.59 | bacon4leif | [TK]D-Fender: ya, that ${CUT(CUT(... stuff is valid, and you got it (you pass it the variable name, not the evaluation) |
| 15:30.18 | jkroon | freaky but valid :p |
| 15:30.28 | bacon4leif | yes, that looks like an example I probably wrote :) |
| 15:30.36 | [TK]D-Fender | bacon4leif, Yeah I always knew the var part... just created a momentary mental separation where function calls are concerned |
| 15:30.41 | bacon4leif | :D |
| 15:31.19 | r0m|u | [TK]D-Fender, hola |
| 15:32.36 | jkroon | is glad he haven't yet had to look at the asterisk parser code - can't be pretty. |
| 15:33.32 | r0m|u | p3nguin, I haven't heard anything from you and the package.... Did you ever get it? |
| 15:34.17 | [TK]D-Fender | jkroon, I wrote a type-sensitive language while in college and bored 18 years ago |
| 15:37.45 | jkroon | [TK]D-Fender, tha's a while back. last time I was bored was 5 years ago whilst working on network card firmware ... splicing engines etc ... |
| 15:37.51 | jkroon | glad those days are over. |
| 15:39.27 | [TK]D-Fender | jkroon, I've reached new levels of existentialism lately. What do you do when you see n need for new "toys" and run out of things to care to do? |
| 15:40.56 | *** join/#asterisk Tim_Toady (~fuzzy@188.4.11.28.dsl.dyn.forthnet.gr) |
| 15:41.28 | jkroon | [TK]D-Fender, you end up existing. not a good place to be. |
| 15:41.33 | jkroon | find something to do. |
| 15:41.56 | [TK]D-Fender | jkroon, Seriously motivationally challenged. That's the issue. |
| 15:42.26 | [TK]D-Fender | Thankfully I am quite capable on running on "empty" but its dragging on... |
| 15:42.34 | jkroon | it always is. and finding motivation is HARD |
| 15:42.41 | *** join/#asterisk DanFromUK (~DanFromUK@2.27.1.180) |
| 15:43.59 | DanFromUK | Hi, I have a question. If call-limit is set to 1, and the limit is exceeded during a Dial command, ${DIALSTATUS} is set to "CHANUNAVAIL". However, if the SIP Peer is simply offline, ${DIALSTATUS is also CHANUNAVAIL. |
| 15:44.32 | DanFromUK | Is there any way to find out whether the dial was unsuccessful due to call-limit or sip peer offline? |
| 15:44.58 | Kobaz | call-limit is depricated |
| 15:45.18 | DanFromUK | in 1.4 call-limit is required in order to get call queuing to work |
| 15:45.22 | Kobaz | ah 1.4 |
| 15:45.26 | Kobaz | that's okay then |
| 15:45.53 | DanFromUK | waiting as long as possible before switching production to 1.8 |
| 15:45.59 | bacon4leif | (in 1.6.2+ it's called callcounter=yes fyi) |
| 15:46.15 | p3nguin | r0m|u: Not as of yesterday. |
| 15:47.08 | *** join/#asterisk asilva (~andre@2801:88:1000:2::12) |
| 15:47.35 | r0m|u | p3nguin, stand by. heading to the university post office right now. by the way arch is as clean as any OS can get. Thanks for the recommendation. |
| 15:47.49 | asilva | Hello all, how do you guys refer to a Conventional Telephony System(PABX and Analog phones and sutff) ? |
| 15:47.56 | tuxx- | ello, short question, is it possible to make an ivr when using realtime dialplans ? |
| 15:48.05 | Qwell | asilva: We call them Sue. |
| 15:48.11 | asilva | Sue ?! |
| 15:48.14 | Qwell | Sue. |
| 15:48.17 | [TK]D-Fender | asilva, You mean non-*? Dead-end junk :p |
| 15:48.26 | Qwell | and if anybody corrects me, I will cut them. |
| 15:48.26 | asilva | [TK]D-Fender, yes |
| 15:48.28 | asilva | Qwell, l0l |
| 15:48.36 | [TK]D-Fender | "Toaster" is pretty common as well |
| 15:48.48 | p3nguin | jkroon: Did you get it to work yet? |
| 15:48.50 | Qwell | this conversation could be quite humorous. asilva: proceed with "Sue", and we'll help you out. |
| 15:49.11 | [TK]D-Fender | s/we/Qwell |
| 15:49.14 | asilva | i'm writing something about in english and here we say "Conventional Telephony System" and i'm looking for the right term in english |
| 15:49.22 | [TK]D-Fender | "We" will be busy bleeding ;) |
| 15:49.28 | asilva | l0l |
| 15:49.49 | Qwell | asilva: people call it lots of different things. Any of the terms you've used would work fine. |
| 15:49.54 | [TK]D-Fender | Well ... "they" anyway. I'm far to well trained in the sharp and pointy.... |
| 15:49.58 | tuxx- | asilva: POTS - Plain Old Telephone System ? :) |
| 15:50.06 | *** join/#asterisk navaismo (~navaismo@189.180.239.217) |
| 15:50.09 | p3nguin | r0m|u: Wouldn't it be easier to simply enter in the tracking number on the web site? |
| 15:50.11 | asilva | tuxx-, that sounds about right! |
| 15:50.14 | asilva | l0l |
| 15:50.22 | tuxx- | :P |
| 15:50.23 | tuxx- | http://en.wikipedia.org/wiki/Plain_old_telephone_service |
| 15:50.24 | tuxx- | oh its |
| 15:50.31 | tuxx- | telephone service, not system :) |
| 15:51.36 | p3nguin | jkroon: I have an idea... but if you fixed it, I don't need to say it. |
| 15:51.51 | DanFromUK | how can i get the status of a peer during the dialplan? I cant seem to find the variable |
| 15:52.31 | asilva | there is IAXPeer Function and SIPPEER |
| 15:52.32 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 15:52.40 | [TK]D-Fender | DanFromUK, "core show function DEVICE_STATE" |
| 15:53.35 | DanFromUK | [TK]D-Fender, Its not available in 1.4 |
| 15:53.54 | [TK]D-Fender | ~devstate |
| 15:53.54 | infobot | func_devstate is a module included with Asterisk 1.6 and above. A 1.4 backport is available here: http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/func_devstate.c |
| 15:53.55 | mandla | Is there a way of assigning phone users phone usage codes?? |
| 15:53.57 | [TK]D-Fender | ^^^ |
| 15:54.12 | [TK]D-Fender | DanFromUK, And you didn't tell us you took off with a PBX from the Smithsonian :p |
| 15:54.35 | mandla | To dial before making a call, so that i can trace who used the phone. |
| 15:54.45 | [TK]D-Fender | mandla, It's your dialplan... shove in checks wherever you'd like. |
| 15:55.00 | p3nguin | mandla: CDR(accountcode) |
| 15:55.36 | [TK]D-Fender | just shove the accountcode right in the peer <- |
| 15:55.37 | DanFromUK | [TK]D-Fender, is 1.8 ready for production use? |
| 15:55.41 | p3nguin | and accountcode= in sip.conf |
| 15:55.47 | bacon4leif | if anyone wants to test Asterisk 10.0.0-rc2 that would be great, because pending anything major, I'm releasing it on Friday |
| 15:56.23 | bacon4leif | DanFromUK: I've deployed several Asterisk 1.8 servers over the last few months. Works fine. |
| 15:56.33 | [TK]D-Fender | DanFromUK, Yes |
| 15:56.37 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 15:56.45 | mandla | [TK]D-Fender, p3nguin, thanx. |
| 15:56.52 | [TK]D-Fender | DanFromUK, Pretty much any branch in full-release is. |
| 15:57.04 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:57.14 | bacon4leif | everything is production ready with enough verification testing |
| 15:57.42 | bacon4leif | it totally depends on what you're using... if you're going to deploy chan_sccp with a lot of app_sms, then no, it's probably not production ready |
| 15:57.47 | jkroon | p3nguin, perhaps say it anyway? |
| 15:58.14 | jkroon | but yes, I did fix it, swapping VALID_EXTEN for DIALPLAN_EXISTS sorted out the issue |
| 15:58.19 | DanFromUK | Ok, i'll look into testing 1.8. thanks |
| 15:58.28 | p3nguin | GotoIf(${VALID_EXTEN(callorigination,${EXTEN},${CHANNEL(channeltype)})}? should have been GotoIf($[${VALID_EXTEN(callorigination,${EXTEN},${CHANNEL(channeltype)})}]? |
| 15:58.42 | bacon4leif | plus the fact it's the only LTS branch receiving support makes it a good version to start with |
| 15:58.54 | p3nguin | I don't know if that mistake is enough to make it not work, though. |
| 15:59.15 | bacon4leif | I always wrap my GotoIf() condition checks in $[ ] |
| 15:59.22 | bacon4leif | makes sure you get a 0 or 1 |
| 15:59.31 | jkroon | p3nguin, what difference would the $[] make in this case though? VALID_EXTEN returns a "0" or "1", which is what GotoIf expects. |
| 15:59.47 | p3nguin | *shrug* |
| 16:00.01 | jkroon | but yes, for general it's not a bad idea. |
| 16:00.02 | [TK]D-Fender | jkroon, it wouldn't |
| 16:02.15 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 16:06.06 | *** join/#asterisk lcat (~lcat@187.45.254.21) |
| 16:10.45 | asilva | which function i can use to search a char in a variavel !? |
| 16:11.14 | *** join/#asterisk irroot (~gregory@197.174.62.37) |
| 16:11.42 | [TK]D-Fender | asilva, not standard means for this AFAIK. You would have to spawn a call to "core show channel:" dump that and parse out the var. Likely in AGI |
| 16:12.03 | asilva | i see |
| 16:12.32 | p3nguin | There's always REGEX() |
| 16:12.42 | irroot | Walter Sisulu University Phase 1 Switchover to asterisk has been done |
| 16:12.45 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 16:12.47 | wcselby | o/ |
| 16:12.54 | irroot | evening folks |
| 16:13.39 | [TK]D-Fender | <PROTECTED> |
| 16:14.30 | _Corey_ | irroot: Your project? (Congrats if so) |
| 16:15.20 | r0m|u | p3nguin, got it. package was still sitting in the pick up room. some fucktard moved it off the pick up table. Taking the damn package to my local usps instead of the university mail. I should of done that from the beginning know what has happen in the past. sorry. |
| 16:15.23 | irroot | yeah colab between 2 companies thx |
| 16:15.30 | r0m|u | wcselby, whats going on |
| 16:15.42 | wcselby | not much |
| 16:15.45 | wcselby | wassup with you? |
| 16:16.07 | r0m|u | nothing much. happy that all my comcast issues are "gone" |
| 16:16.11 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 16:16.13 | r0m|u | for now. |
| 16:16.42 | r0m|u | haven't had an issue for a few days now "2" all times records |
| 16:16.47 | wcselby | lol |
| 16:17.43 | r0m|u | other than that waiting to strangle who ever keeps fucking around with the mail room here at work. university mail just sucks |
| 16:18.03 | wcselby | lol |
| 16:20.04 | *** join/#asterisk chazzam (~chazz@173-24-236-90.client.mchsi.com) |
| 16:21.19 | wcselby | i'm tired |
| 16:22.20 | r0m|u | just curious... does any body here used elastix? |
| 16:22.28 | wcselby | i have used it in the past |
| 16:22.47 | r0m|u | how was your experience? |
| 16:23.07 | wcselby | it was a red trixbox |
| 16:23.17 | wcselby | :) |
| 16:23.19 | r0m|u | lol |
| 16:24.18 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 16:24.57 | r0m|u | My brother is just bent for gui stuff and he is using it.... I found it so heavy and buggy.... their community support just plain suck. |
| 16:25.09 | eppigy | 8[] |
| 16:25.14 | wcselby | they do have a certification path thought |
| 16:25.27 | wcselby | though* |
| 16:25.39 | wcselby | so apparently there's someone out there that knows how to support it |
| 16:25.39 | r0m|u | holy crap. on what? on how to clean their build? |
| 16:25.50 | wcselby | i dunno, never looked into it |
| 16:25.59 | wcselby | i got my dcap and stuck with that :) |
| 16:25.59 | r0m|u | :) |
| 16:26.07 | r0m|u | cool |
| 16:28.30 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-fbqtzkeopsketdet) |
| 16:28.34 | r0m|u | wcselby, you got your cert online? |
| 16:28.46 | Qwell | r0m|u: dcap is done in person |
| 16:28.47 | wcselby | i got it at astricon in dc in 2010 |
| 16:28.58 | r0m|u | ah I see. |
| 16:29.12 | r0m|u | like rhce. |
| 16:29.22 | _Corey_ | You can do a dCAA online, I believe |
| 16:29.46 | r0m|u | I see. |
| 16:30.01 | Qwell | _Corey_: you sure about that? |
| 16:30.10 | _Corey_ | Yeah, it's done through the portal |
| 16:30.18 | _Corey_ | one of my guys did it a couple weeks ago |
| 16:30.25 | Qwell | hmm |
| 16:30.32 | _Corey_ | Brainshark I think |
| 16:30.45 | Qwell | yeah you're right |
| 16:31.38 | r0m|u | 2k for the fast start. |
| 16:35.49 | r0m|u | Mhhhh I wonder if work would send me to train.... It does not relate to my job though :( |
| 16:37.29 | shido6 | <PROTECTED> |
| 16:37.45 | eppigy | books homie |
| 16:39.07 | p3nguin | r0m|u: If he insists on using something other than plain Asterisk, AsteriskNOW is the answer. |
| 16:39.28 | diegoCronos | hi guys |
| 16:39.31 | r0m|u | AsteriskNOW? |
| 16:39.57 | r0m|u | is that the digium spin of asterisk gui? |
| 16:40.03 | Qwell | no |
| 16:40.09 | Qwell | well, sort of, but no. |
| 16:40.10 | p3nguin | No, because that wouldn't make sense. |
| 16:40.24 | r0m|u | I dont make sense most of the time lol |
| 16:40.37 | Qwell | AsteriskNOW is a Linux distribution. It includes Asterisk and your choice of: FreePBX, Asterisk-GUI, no GUI. |
| 16:40.41 | shido6 | you're here, so you'll save dollars. :) |
| 16:40.44 | p3nguin | AsteriskNOW is a CentOS-based Asterisk + FreePBX or Asterisk GUI distribution. |
| 16:40.54 | r0m|u | ah! |
| 16:40.58 | r0m|u | nice. |
| 16:41.06 | r0m|u | is looking at it now. |
| 16:41.42 | blizzow | When I do a queue show 602 for my queue, I have some registered members that are completely screwed up. For example: |
| 16:41.42 | blizzow | SIP/ (619) 659-7144 (dynamic) (Unknown) has taken no calls yet |
| 16:41.47 | Qwell | r0m|u: The guy who wrote and maintains it is a seriously cool guy. |
| 16:41.58 | Katty | hi Qwell |
| 16:42.00 | p3nguin | For those who can't seem to install their own distro and Asterisk, there's the no GUI option. This is the option I recommend. |
| 16:42.06 | blizzow | I tried queue remove member SIP/ (619) 659-7144 from 602 and that didn't work. |
| 16:42.17 | blizzow | How do I remove a member with spaces and special characters? |
| 16:42.20 | Qwell | Katty: ohai |
| 16:42.38 | r0m|u | p3nguin, Thanks for the recommendation! |
| 16:42.42 | Qwell | p3nguin: I'm rather fond of that option myself. |
| 16:43.24 | r0m|u | Qwell, Thats good to know. unlike elastix (no bashing intended) which is just lost.... |
| 16:43.46 | Qwell | I have no comment on other distributions (except trixbox, which I can bash all day long). |
| 16:44.05 | r0m|u | lol |
| 16:45.02 | *** join/#asterisk hfb (~hfb@pool-98-112-242-158.lsanca.dsl-w.verizon.net) |
| 16:45.06 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:45.41 | wcselby | blizzow try escaping the special characters with a backslash "\" |
| 16:46.09 | wcselby | about the whole online certification thing, that seems really odd to me to be able to take a cert online. all certs i've ever taken you have to do in person |
| 16:46.10 | Qwell | blizzow: You queue members name has spaces? how the heck did that happen? O.o |
| 16:46.17 | wcselby | but I guess times, they's a changin' |
| 16:46.41 | [TK]D-Fender | Qwell, He didn't look at what he passed AQM |
| 16:46.56 | Qwell | [TK]D-Fender: I'm surprised it was allowed |
| 16:46.58 | blizzow | Qwell: we use queuemetrics to log into queues and I think a couple of agents were confused or not paying attention when they put their login info into queuemetrics. |
| 16:47.33 | [TK]D-Fender | Qwell, you can add anything. remeber the preload issues with chan_local? type doesn't need to be valid even.. it'll egt added.. and well crippled |
| 16:47.54 | Qwell | yeah but think of the children |
| 16:49.05 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 16:49.38 | saisoma | Hi guys. Is there a way to force an existing call to go on hold from the dialplan? |
| 16:50.04 | Qwell | saisoma: depends on your definition of "on hold" |
| 16:50.16 | Qwell | You can certainly start music on hold |
| 16:50.53 | saisoma | Qwell: I have built an emergency announcement system that works very well, except that if someone is on the phone, they do not get the call |
| 16:51.20 | saisoma | Qwell: I don't want to force a hangup (bad for those ont he phone with emergency responders), but putting the call on hold would be ok i think |
| 16:51.49 | Qwell | a phone "on hold" really doesn't mean anything special. It's just getting no audio. |
| 16:51.59 | saisoma | Qwell: I'm using polycom 560's and 330's and AutoAnswer, just FYI. |
| 16:52.04 | [TK]D-Fender | saisoma, How are you announcing to the others? |
| 16:52.17 | [TK]D-Fender | saisoma, then spawn a ChanSpy w/e whisper |
| 16:52.19 | [TK]D-Fender | ^^^ |
| 16:52.29 | Qwell | yes, that would be a vastly superior method |
| 16:52.32 | saisoma | [TK]D-Fender: gotcha. good idea |
| 16:52.42 | saisoma | [TK]D-Fender: great idea. Going to try that now. thanks guys! |
| 16:57.21 | irroot | must remember to not allow spaces in the member interface in the fixups im busy with |
| 17:00.29 | p3nguin | While you're at it, fix up the Arguments section of core show application ConfBridge. It says option s give you a menu when pressing '*' but it's really '#' |
| 17:01.17 | Qwell | p3nguin: what version? |
| 17:01.23 | p3nguin | 1.8.7.1 |
| 17:01.33 | wcselby | whole new confbridge in 10 |
| 17:01.46 | p3nguin | Doesn't make it any more accurate in 1.8 |
| 17:04.18 | Qwell | p3nguin: It says # in r345545 |
| 17:04.32 | p3nguin | Does that mean someone changed it after the release? |
| 17:04.41 | Qwell | maybe |
| 17:04.53 | p3nguin | I don't know what revision the release is. |
| 17:04.54 | Qwell | r345545 | qwell | 2011-11-17 11:04:05 -0600 (Thu, 17 Nov 2011) | 6 lines |
| 17:04.55 | Qwell | Fix documentation of 's' option. |
| 17:04.55 | Qwell | The menu key is #, not *. |
| 17:04.55 | Qwell | Reported by p3nguin on #asterisk. |
| 17:05.04 | p3nguin | hmm |
| 17:05.10 | wcselby | that's from this morning |
| 17:05.10 | Qwell | :p |
| 17:05.14 | Qwell | that's from right now |
| 17:05.19 | wcselby | oh lol |
| 17:05.19 | p3nguin | That's from NOW |
| 17:07.08 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 17:07.26 | r0m|u | can two asterisk server coexist with each other? |
| 17:07.31 | Qwell | has the power to make people wrong |
| 17:07.35 | Qwell | r0m|u: sure, why not? |
| 17:07.39 | irroot | Qwell go fix "NODATE" that should be NODATA in include/asterisk/astobj2.h :P |
| 17:07.42 | wcselby | Qwell yeah, on the same box even |
| 17:07.46 | wcselby | sorry not Qwell |
| 17:07.50 | wcselby | i meant that for r0m|u |
| 17:08.01 | r0m|u | Qwell, I am trying to create a fail over server |
| 17:08.07 | *** join/#asterisk edge (~edge@97-64-216-2.client.mchsi.com) |
| 17:08.21 | p3nguin | Don't do it on the same network segment. |
| 17:08.23 | r0m|u | I setup Arch and I want to move it as my primary system and move my alix as back up |
| 17:08.27 | Qwell | irroot: I see no such thing |
| 17:08.29 | irroot | or main/astobj2.c .... |
| 17:08.42 | irroot | is going down in a few minutes |
| 17:08.44 | Qwell | pfft, comments. Nobody cares about comments. |
| 17:08.50 | p3nguin | cares |
| 17:08.51 | Qwell | ps: you have commit. Fix it yourself. :p |
| 17:09.05 | irroot | lol that is why i left it ...... |
| 17:09.06 | irroot | ill sort it out soon |
| 17:09.11 | edge | Can Asterisk (paired with the right phone) park a call to a button on the phone, and then pick it up. I'm looking into doing a Asterisk deployment but i need to match our current phone systems features for ease of use. Currently if we hit hold on our phone systems it puts the call on a "line" and all our phones have those 4 lines. and we can then pickup that line with a button on the phone |
| 17:09.35 | r0m|u | p3nguin, put it in a different block? |
| 17:09.41 | Qwell | edge: You could map hints on your phone to park extens if your phone supports that. |
| 17:09.48 | bacon4leif | ya that |
| 17:10.14 | edge | Qwell, what kinds of phones would support that? |
| 17:10.16 | p3nguin | The point of failover is to not have a single point of failure. Using the same computer system on the same network for a failover doesn't do much good. |
| 17:10.32 | edge | Qwell, i havn't picked the phones to test yet. I have some Cisco 7961s laying around here |
| 17:10.43 | Qwell | edge: throw them away IMO |
| 17:11.04 | edge | Qwell, i don't want to use them, they are overly complicated. I just can't decided on a phone, but i know that park feature is a huge one. |
| 17:11.05 | wcselby | edge Qwell is biased against Cisco phones |
| 17:11.25 | wcselby | edge some of the cisco 5xx phones will probably support what you want, or some of the aastra phones |
| 17:11.40 | wcselby | even some polycoms |
| 17:11.53 | wcselby | it all basically comes down to phones with enough buttons |
| 17:12.18 | r0m|u | p3nguin, I understand. I have two systems. I have three different subnets and for the sake of redundancy ill keep one sync with the other. |
| 17:12.18 | edge | wcselby, and these would allow a user to press "hold" and it would park it to a line, and then allow them to say "you have a call on button 3" instead of line? |
| 17:12.22 | p3nguin | edge: If you use chan_sccp-b, you'll have a Park softkey, which will park the call and tell you what extension it is parked on. If you have a speeddial for, say, 701, then you could pick up the call you parked on 701. |
| 17:12.31 | wcselby | i've seen what you're asking for done with the cisco 5xx phones and aastra, and in theory you should be able to do it with some of the polycoms and others by other manufacturers |
| 17:12.35 | p3nguin | Or just dial 701. |
| 17:13.49 | wcselby | it all comes down to being able to map custom extensions to custom hints and then setup speed dials to those extensions on the phone's buttons |
| 17:14.26 | Qwell | honestly, it's a pretty basic phone feature |
| 17:14.57 | r0m|u | I have my polycom map with a softkey to park a call and another one to retrive a call |
| 17:15.04 | wcselby | Qwell I also left out the monitoring of those extenions in order to turn on a light next to the button (or light up the button, etc). i'm distracted |
| 17:15.11 | r0m|u | from parking* |
| 17:15.15 | Qwell | wcselby: I think that's implied |
| 17:15.28 | wcselby | Qwell good 'cause i'm distracted |
| 17:15.34 | wcselby | in case I didn't mention that |
| 17:15.34 | Qwell | I suppose some phones might use icons for that though. whatever |
| 17:16.53 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 17:17.04 | edge | So picking up a cisco 504G , i could have them hit hold , it would auto park to one of the 4 or so 700-705 extensions, then it would light up (differently) that extension button's light or softbutton? |
| 17:17.20 | Qwell | edge: not hold, no |
| 17:17.32 | edge | Qwell, a different button then? |
| 17:17.34 | Qwell | hold != park |
| 17:18.04 | edge | Qwell, our current nortel does that, holding parks it to anybody can pick up the flashing light on all the sets, i'm kind of trying to duplicate that function |
| 17:18.20 | *** part/#asterisk ShaunR (~ShaunR@freenode/sponsor/NDChost.com) |
| 17:18.28 | [TK]D-Fender | edge, Let go the the NorHell reflex.... |
| 17:18.37 | p3nguin | If you want to park, park; don't use hold. |
| 17:18.45 | p3nguin | If you want to hold, hold; don't park. |
| 17:18.51 | wcselby | edge you'd more likely do a transfer to one of those monitored extensions, which would actually be a parking lot |
| 17:20.49 | edge | i'm concerneed my users put calls "down" to give off to another user (after calling that user) and they forget which line it is on NOW, i can't imagine if they have to remember a 3 digit number |
| 17:21.08 | edge | if it left something up like a button and flashed that button then these guys/gals could figure that out |
| 17:21.28 | p3nguin | If you use the call parking feature, it will return to the caller who parked it after a configured timeout value. |
| 17:21.35 | Qwell | edge: we aren't saying it's not possible - you're just using the wrong term. |
| 17:21.44 | Qwell | hold = I can pick this call back up when I'm ready. |
| 17:21.50 | Qwell | park = anyone can pick this call up |
| 17:21.51 | p3nguin | And if you want to send a call to another user, use the transfer button. |
| 17:22.24 | edge | Qwell, i do mean park. |
| 17:22.40 | Qwell | I know you do. |
| 17:23.01 | p3nguin | "Please hold." <transfer key> <friend's extension> (ringing) (answer) "I have a call for you." "Okay, transfer it." <transfer key> |
| 17:23.19 | edge | p3nguin, normally, we park the call, call upstairs to the intended party and say "you have a call on line 2 , its Joe bob", and then the upstairs party would pickup line 2 and say "hi joe, how are you" |
| 17:23.33 | edge | p3nguin, Oh? thats how that works? |
| 17:23.36 | p3nguin | Time to learn how to use transfer. |
| 17:23.51 | edge | p3nguin, yes i think transfer would work great!, |
| 17:24.15 | eppigy | HOLLA |
| 17:24.16 | edge | p3nguin, what happens if friends extention doesn't pickup or doesn't want it? i can just not hit the key? and get the call back? |
| 17:24.37 | p3nguin | Just end the call and press resume... because the caller was placed on hold when you hit transfer. |
| 17:24.41 | [TK]D-Fender | edge, It never left |
| 17:24.53 | wcselby | edge what I'm saying is, you transfer the call to the parking lot by clicking the transfer button on the phone, and then hit the monitored button on the phone. the call goes into the parking lot, the custom hint that ll of your phones are monitoring will cause that line to blink on all of the phones. then when someone clicks that button, it dials into the parking lot and you're good. just like the way it is now, except instead o |
| 17:24.53 | wcselby | f "hold" button it's a "transfer - button" button |
| 17:24.58 | [TK]D-Fender | edge, Attended Transfer <- |
| 17:25.42 | p3nguin | A blind transfer sends the call to the other exten blindly. You have no way to know what happens when you do this. With an attended transfer, you control what happens. |
| 17:25.51 | edge | wcselby, that will also work |
| 17:26.25 | edge | [TK]D-Fender, in attended transfers, does the person i'm planning on transfering hear my converstation with the entended party? |
| 17:26.28 | wcselby | the way I described is a lot more difficult to setup than just using the built-in phones "transfer" feature, but in the end it's less for your end-users to learn |
| 17:26.37 | wcselby | edge no |
| 17:26.41 | [TK]D-Fender | edge, No |
| 17:26.47 | p3nguin | It's not a conference call. |
| 17:27.02 | edge | wcselby, i like that way, what phone would you use, i'll buy 4 and get it to work, if i can i'll deploy it that way, if i can't then i'll have to figure something else out |
| 17:27.07 | [TK]D-Fender | You could choose to make it one if you chose though |
| 17:27.11 | wcselby | an attended transfer works like this - you answer, realize someone else needs to talk to the person, you say "please hold while I transfer you to jim", you hit the transfer button on your phone, the call goes on hold and you're presented a new dialtone. |
| 17:27.16 | p3nguin | When you press the transfer key the first time, your caller is put on hold and you are given a new "line" for calling the other exten. |
| 17:27.32 | wcselby | you then call jim, explain to him what's going on, then you click the "transfer" button again, and the original call is connected to jim |
| 17:27.36 | edge | [TK]D-Fender, i dont' want it a conference, sometimes our other employees don't want to be bothered and i don't want the customer to hear that "don't bother me right now" attitude lol |
| 17:27.53 | p3nguin | When the other side answers, you talk to the person, then press transfer a second time to transfer that on-hold call to this new extension. |
| 17:28.26 | p3nguin | But if the other party does not answer or it goes to voice mail, don't press the transfer key (don't complete the transfer). |
| 17:28.32 | edge | p3nguin, does my SIP phones then need to be configured with two exentions for two lines? |
| 17:28.52 | [TK]D-Fender | edge, I've never seen a ophone that can't do this |
| 17:28.57 | p3nguin | When you end that new call, just hit resume to get back to the caller you put on hold when you originally hit transfer. |
| 17:28.58 | miztic | they should do multiple lines with the same extension |
| 17:28.58 | wcselby | edge msot don't |
| 17:29.03 | wcselby | most* |
| 17:29.24 | p3nguin | A single-line phone can do transfers. |
| 17:29.35 | miztic | sometimes it's a config option on how many lines it can do at the same time, but the default is usually pretty high like 4 simultaneous calls |
| 17:29.56 | *** join/#asterisk kfife (~wircer@kfife.com) |
| 17:30.02 | wcselby | so edge, I'm thinking you can do this with the cisco 5xx series phones (the ones with lots of buttons), but pretty much any phone that will support presence and has plenty of buttons should be able to do this |
| 17:30.03 | [TK]D-Fender | edge, Next important concept for you : completely disassociate a "call" from a "line" |
| 17:30.26 | p3nguin | I've never hit the limit on calls on a single line on my 7960. |
| 17:30.48 | edge | [TK]D-Fender, i think that is the most confusing part for me |
| 17:31.13 | [TK]D-Fender | edge, A call is a call is a call. The closest thing to a "line" is a registration for a specific identity. |
| 17:31.30 | [TK]D-Fender | edge, Typically a Phone has only one identity and can handle multiple simultaneous calls. |
| 17:31.55 | [TK]D-Fender | edge, On "key systems" like Norstar, etc phones would have buttons directly tied into physical lines. This is sooo 1980 |
| 17:32.33 | [TK]D-Fender | edge, To just abuot ever phone, having multiple buttons to represent individual calls has no assoiation to any particular resource like "intercom" or "line" keys on a key system |
| 17:35.09 | edge | [TK]D-Fender, while being so 1980, thats what my users would like to see, but i like the idea of attended transfers better, or blinking a monitored button because they can handle that. A lot of old people here, don't like to change. but this nortel system is about to die, its like ... 20+ years old almost |
| 17:35.27 | [TK]D-Fender | edge, Looking at a decrepit Polycom IP 300. Ancient model with 2 line keys. This means it could support up to 2 completely separate identities. Now if you only needed 1 identity (99% of phone users out there) you can actually handle up to *5* calls per keys |
| 17:36.52 | miztic | we used to have a key system here with complete neophytes using them, they got used to the voip way pretty quick, only 1 guy has a decent reason for having a key system on his desk, he's got two cisco phones and is regularly on multiple calls at the same time |
| 17:37.41 | [TK]D-Fender | edge, I ditched my Norstar 8x24 in 2005 for * + Polycom phones. Thank God... |
| 17:38.10 | edge | [TK]D-Fender, i want to do the same , i just need to make sure i understand how transfers are going to work, and that i pick the right phones |
| 17:38.20 | [TK]D-Fender | edge, Polycom > All |
| 17:38.24 | Naikrovek | ^^^ |
| 17:39.00 | edge | If I use the transfer call button, in attended mode, and the friend i'm sending the call to its ready at that moment to recieve the call , would i have to wait a minute or so before hitting transfer? or can i then elect to park the call |
| 17:39.39 | r0m|u | nom nom nom pupusa's. delicious! |
| 17:40.15 | wcselby | [TK]D-Fender they're not good for trying to replicate key systems, but they are very good otherwise |
| 17:40.26 | wcselby | they're a nice evolution of office phone though |
| 17:40.33 | wcselby | if you can get your users to adjust |
| 17:40.38 | [TK]D-Fender | edge, Wait a sec ...[Transfer] 5000 ...... (answered) Hey you wanna take this guy I got on the line here? OK [Transfer] |
| 17:40.43 | p3nguin | Once you choose to not transfer to the new extension, and you end that part of the call, you hit resume to take back the call you put on hold. |
| 17:40.55 | p3nguin | At that time, you can park it or tranfer it to someone else. |
| 17:41.24 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-auvzmqurqjvddsla) |
| 17:41.51 | wcselby | one thing i never liked about polycoms (about the only gripe I have really) is having to use all my buttons in sequential order. i can't just assign an extension to the last button on a side car and leave six blanks in front of it, for example |
| 17:42.22 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 17:42.35 | [TK]D-Fender | wcselby, Well. you can sort of cheat.. I added a "-" entry to space mine out for my receptionist. |
| 17:43.01 | [TK]D-Fender | but that isn't really "blank" |
| 17:43.13 | wcselby | [TK]D-Fender yeah I suppose, and that's a good idea, but I'd rather just have the abillity to assign any button at any time |
| 17:43.27 | [TK]D-Fender | wcselby, Yeah, it might eb nice... |
| 17:43.31 | wcselby | but like I said, that's my only gripe with polycoms |
| 17:43.44 | wcselby | and I still recommend them over any other phone when my client's ask my opinion |
| 17:43.49 | [TK]D-Fender | Aastra's do this well.. but way to many physical caveats for me to stomach |
| 17:44.17 | wcselby | aastra's do it, the new cisco 5xx is supposed to do it, and I think the snom or yealink guys said you coudl do that on their phones when I talked to them at astricon |
| 17:44.18 | [TK]D-Fender | One of my CSRs has their top'o'the'line colour touch-screen ones.... it is purrrty |
| 17:45.02 | [TK]D-Fender | But... the buttons are still rubber shit and general call handling... blah :p |
| 17:46.13 | edge | [TK]D-Fender, so Polycom 321/331 would work great for this, or should i go bigger, saving money is cool with me. |
| 17:46.30 | r0m|u | any opinions on yealink? |
| 17:46.30 | [TK]D-Fender | edge, Those will do. |
| 17:46.40 | Naikrovek | i have 321s all over the place. receptionist has a 650 w/2 sidecars |
| 17:46.42 | Naikrovek | everyone is happy |
| 17:46.46 | Naikrovek | even those who like to dial 9 |
| 17:46.48 | Naikrovek | .... idiots |
| 17:46.50 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:46.51 | r0m|u | lol |
| 17:47.06 | wcselby | Naikrovek lol |
| 17:47.18 | Naikrovek | i have a few 450s around as well. managers get PISSED when they don't get a "better" phone... |
| 17:47.24 | wcselby | Naikrovek you use freepbx still? |
| 17:47.29 | Naikrovek | for some, yes. |
| 17:47.33 | wcselby | ahh |
| 17:47.48 | [TK]D-Fender | Naikrovek, I "down"graded from an IP 600 to an IP 335. I like the smale profile and backlight |
| 17:47.52 | Naikrovek | trying to get a new asterisk system in here, but i'm not sure i'm of the caliber to maintain it |
| 17:47.54 | [TK]D-Fender | small* |
| 17:48.01 | Naikrovek | yeah the 335s are nice. |
| 17:48.08 | Naikrovek | i'm going to start ordering those for all new non-manager phones |
| 17:48.24 | wcselby | Naikrovek why would you be able to handle it? |
| 17:48.49 | Naikrovek | new system will be a three-server vanilla asterisk solution that leifmadsen and I have put together. |
| 17:48.57 | Naikrovek | i don't know vanilla asterisk very well at all |
| 17:49.05 | [TK]D-Fender | OMGWTFBBQ! |
| 17:49.07 | Naikrovek | i can learn it, yes, but i don't have a lot of time |
| 17:49.29 | Naikrovek | it is a time concern more than anything else. |
| 17:49.44 | Naikrovek | certainly not a capability concern. /gloat |
| 17:50.12 | wcselby | lol |
| 17:50.24 | Naikrovek | so yeah i have a basic quote from him, i know what servers I need and where they need to go, leif is going to set most of it up once I get the software in place then he'll leave instructions for me to maintain |
| 17:50.27 | Naikrovek | that's the plan |
| 17:50.39 | Naikrovek | waiting on management for approval for new machines and leif's time. |
| 17:50.48 | wcselby | nice |
| 17:50.51 | [TK]D-Fender | Naikrovek, How big a setup? |
| 17:51.02 | Naikrovek | three servers, two in india, on in illinois |
| 17:51.10 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:51.10 | wcselby | is it distributed call center setup? |
| 17:51.16 | [TK]D-Fender | Was referring more to complexity of deployment |
| 17:51.17 | Naikrovek | eh will wind up being about 450 endpoints by the time i'm done. |
| 17:51.21 | jaytee | considering it's leif I wouldn't be too concerned about being able to support it. probably won't require much on your end and he's usually available |
| 17:51.22 | Naikrovek | not complex |
| 17:51.37 | Naikrovek | just three servers all connected to each other, static extensions, basic |
| 17:51.48 | Naikrovek | not a call center |
| 17:51.53 | Naikrovek | just a lot of people that need to talk on the phone |
| 17:51.55 | wcselby | are you doing like an xmpp distributed call presence feature or something? |
| 17:51.57 | wcselby | gotcha |
| 17:52.00 | wcselby | that's not bad |
| 17:52.18 | Naikrovek | no, no xmpp, just basic phone stuff. |
| 17:52.39 | wcselby | i hate it when my bluetooth mouse decides it doesn't want to work |
| 17:52.43 | Naikrovek | also waiting on management to decide if people will move between offices, and if they need to keep their extension number if they do. |
| 17:52.59 | Naikrovek | dynamic extensions would be nice, but i'm not sure it's needed |
| 17:53.06 | *** join/#asterisk xpot-mobile (~james@dhcp69.emcb.utah.edu) |
| 17:53.16 | [TK]D-Fender | All pretty basic |
| 17:53.19 | p3nguin | People should always keep their extension numbers, unless you have the same numbers in every office. |
| 17:53.28 | p3nguin | ~devicenames |
| 17:53.29 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 17:54.07 | r0m|u | p3nguin, your math was correct the other day. (me paying for CC) it came out to $3.24 or somewhere around there for voip.ms |
| 17:54.08 | Naikrovek | i agree with you, p3nguin, but people don't seem to move that much, and if they do, they take on a whole new role with new people to call. not once has anyone asked me for this, and people have moved around a lot. |
| 17:54.39 | r0m|u | p3nguin, so I am ditching CC and using voip.ms all the way. |
| 17:54.53 | p3nguin | It doesn't matter what the role is in relation to an extension, unless the role dictates the extensions. |
| 17:54.57 | Naikrovek | probably been 50 moves since I started here (in two offices) and not once has anyone asked if they could keep their extension |
| 17:55.06 | Naikrovek | p3nguin: role does not dictate extension |
| 17:55.33 | p3nguin | peons, 1000s; managers, 3000s |
| 17:55.39 | Naikrovek | but with polycom sip 4 i won't have to worry about it. it abstracts user from device all by itself. |
| 17:55.54 | p3nguin | That might be a problem. |
| 17:55.56 | Naikrovek | sit down, log into phone, you see YOUR extension, then you log out when you leave |
| 17:56.00 | p3nguin | Devices should never be tied to a person. |
| 17:56.16 | Naikrovek | i agree, sip 4.0 allows it natively |
| 17:56.17 | p3nguin | Auto-hot-desking? |
| 17:56.20 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:56.22 | Naikrovek | yes |
| 17:56.25 | p3nguin | That's fancy. |
| 17:56.40 | eppigy | oh you fancy huh |
| 17:56.42 | p3nguin | Saves for some configuration, I suppose. |
| 17:56.52 | Naikrovek | the polycom sip 4 config separates device from user, if you configure it. |
| 17:57.21 | Naikrovek | so I could sit at a different desk each day, and keep my extension only by logging into and out of the phone where I'm sitting |
| 17:57.26 | Naikrovek | zero asterisk configuration |
| 17:57.37 | Naikrovek | at least that's what the documentation says |
| 17:57.42 | Naikrovek | i don't have the firmware yet. :( |
| 17:57.49 | p3nguin | When I deploy a system, no matter how large or small, the extensions make the association in the astdb. If a person moves offices and phones, I just change the device associated to the extension in the db and go on with my day. |
| 17:58.23 | r0m|u | http://www.facebook.com/group.php?gid=2231796597 |
| 17:58.35 | p3nguin | This allows for the person to use the phone without logging in or out. It's his phone until otherwise instructed. |
| 17:58.55 | r0m|u | p3nguin, +1 |
| 17:59.04 | r0m|u | p3nguin, I like that idea! |
| 17:59.06 | vader-- | Anyone have a good source for a TDM400P with 2FXO and 2FXs? Trying to stay around $100 for it |
| 17:59.09 | wcselby | btw edge, here's an example of using custom blf lamps and extensions to do stuff (a la what I was talking about with park). this example is for configuring a night mode, and monitoring it, but it's the same basic principle --> http://pastebin.com/Ka6d57mG |
| 18:00.10 | [TK]D-Fender | vader--, 3rd shelf ... right between the unicorns and leprechauns ... |
| 18:00.16 | vader-- | :-) |
| 18:00.18 | vader-- | sweet |
| 18:00.21 | r0m|u | HAHAHAHA |
| 18:00.21 | wcselby | this is just the asterisk configuration, btw. the phone is then configured separately |
| 18:01.43 | wcselby | afk |
| 18:02.16 | [TK]D-Fender | trickiest bit of hot-esking = MWI |
| 18:07.50 | *** part/#asterisk pietro (~pietro@88-149-226-183.dynamic.ngi.it) |
| 18:09.33 | Naikrovek | ... whoa |
| 18:09.45 | Naikrovek | just got told that my company is moving OUT OF the building it owns. |
| 18:09.52 | Naikrovek | this is straight up wtf territory |
| 18:14.26 | wcselby | lol |
| 18:14.36 | wcselby | i heard that's what enron did too |
| 18:14.49 | wcselby | not that's that is what's in store for you or anything |
| 18:14.57 | wcselby | just, you know.... |
| 18:14.59 | wcselby | :) |
| 18:15.10 | r0m|u | Naikrovek, You in the US? |
| 18:15.16 | Naikrovek | I am. |
| 18:15.36 | r0m|u | I see. |
| 18:15.39 | Naikrovek | lol we're not that big, only about 15 people. the other company will stay in this building. |
| 18:15.56 | Naikrovek | turns out that renovation of unused space will cost FAR more than leasing a new office for a decade. |
| 18:16.06 | Naikrovek | so, we move. |
| 18:16.06 | r0m|u | Naikrovek, I beet they are renting out the rest of the space |
| 18:16.30 | r0m|u | ah I see. |
| 18:16.33 | Naikrovek | no, there are two companies in this building. both are short on space. one moves out, frees space for the other company. both companies are owned by the same dude and managed as one. |
| 18:16.47 | [TK]D-Fender | "No, YOU move out..." |
| 18:16.53 | r0m|u | lol |
| 18:16.54 | Naikrovek | the company without strong ties to this facility will stay |
| 18:17.02 | Naikrovek | (this means I'm moving) |
| 18:17.05 | r0m|u | most side with revenue wins |
| 18:17.09 | r0m|u | :) |
| 18:17.14 | *** join/#asterisk kikohnl (~kotis@ext-dip-171.hnl.cdsinc.com) |
| 18:17.23 | Naikrovek | which is seriously messed up because this is a 48k sq ft facility |
| 18:17.33 | r0m|u | :/ |
| 18:17.35 | Naikrovek | the smaller company will be moving |
| 18:17.53 | wcselby | lol |
| 18:18.04 | wcselby | Naikrovek so do you have to move all of your servers and stuff too? |
| 18:18.06 | wcselby | or will those stay? |
| 18:18.20 | Naikrovek | most of it will stay, with very small new infrastructure being put up at the new place |
| 18:18.34 | Naikrovek | the company that's staying is who needs most of what's here now |
| 18:18.54 | Naikrovek | the company I work for will create a new small infrastructure to serve the local office; core infrastructure will remain here. |
| 18:19.02 | r0m|u | [TK]D-Fender, Polycom phones know what files to put don base on the mac.cfg file? so many configs can exist together? |
| 18:19.16 | Naikrovek | polycoms are smart |
| 18:19.18 | Naikrovek | ish |
| 18:19.29 | Naikrovek | the phone knows which ones it needs |
| 18:19.44 | [TK]D-Fender | <mac>.cfg says what it needs |
| 18:20.17 | r0m|u | ok thanks. just curious. |
| 18:21.07 | *** join/#asterisk ruffle (51bb4263@gateway/web/freenode/ip.81.187.66.99) |
| 18:23.25 | wcselby | aight |
| 18:23.33 | wcselby | alright* |
| 18:23.35 | wcselby | i can't type today |
| 18:23.40 | r0m|u | lol |
| 18:23.49 | wcselby | i'm heading to our datacenter, i'll see you folks next week :) |
| 18:23.55 | r0m|u | cya! |
| 18:23.57 | r0m|u | take care! |
| 18:24.17 | ruffle | Shaun Ruffell? You on here? See you're on my Asterisk box. |
| 18:24.36 | Qwell | ruffle: he doesn't come to #asterisk |
| 18:24.51 | ruffle | Ah. He asked if I was on Freenode... so I assumed. |
| 18:24.51 | Qwell | try messaging him - sruffell |
| 18:25.40 | ruffle | Righty Ho. Ta. Ummmm.. I don't really use IRC... Off I go to find out how. |
| 18:27.10 | *** join/#asterisk b0ot (~Jinxed---@147.177.57.101) |
| 18:29.04 | bacon4leif | <PROTECTED> |
| 18:31.08 | b0ot | in my extensions.conf file how would I add a line that would send the digits dialed to a specific ip address... for example I want to send any call that is 1XXX to 10.1.1.1 lets say would it be exten => 1***,1,Dial(SIP/@10.1.1.1,20) ? |
| 18:31.19 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 18:31.28 | b0ot | I can't find the variable for the number dialed |
| 18:31.35 | b0ot | to add before the @ sign |
| 18:32.32 | bchia | b0ot "exten => _1XXX,1,Dial(SIP/@10.1.1.1,20/${EXTEN})" is what yr looking for? |
| 18:32.38 | b0ot | yep |
| 18:32.40 | b0ot | thanks |
| 18:32.43 | bchia | np |
| 18:33.42 | bchia | actually that synatax is off - the time out comes later Dial(teck/resource/digits,timeout) |
| 18:34.09 | b0ot | ? |
| 18:34.29 | bchia | "exten => _1XXX,1,Dial(SIP/@10.1.1.1,20/${EXTEN})" is wrong |
| 18:34.49 | bchia | "exten => _1XXX,1,Dial(SIP/10.1.1.1/${EXTEN},20) is more correct |
| 18:34.56 | b0ot | exten => _1XXX,1,Dial(SIP/{EXTEN}@10.1.1.1,20) |
| 18:35.10 | b0ot | exten => _1XXX,1,Dial(SIP/${EXTEN}@10.1.1.1,20) |
| 18:35.42 | bchia | you can do it that way too |
| 18:36.18 | b0ot | alright thanks again bchia |
| 18:39.42 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 18:48.25 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:48.53 | *** part/#asterisk kfife (~wircer@kfife.com) |
| 18:49.16 | b0ot | bchia, Is that considered a trunk? |
| 18:49.45 | bchia | right, I was thinking you were dialing out a trunk and sending digits out the trunk |
| 18:50.53 | r0m|u | p3nguin, is G722 that much superior? |
| 18:51.06 | r0m|u | in terms of quality |
| 18:51.12 | p3nguin | Sure. |
| 18:51.26 | p3nguin | But you have to use it end to end, or it's worthless. |
| 18:52.05 | r0m|u | ah. ok. so must be digital all the way across. correct? |
| 18:52.57 | r0m|u | sip phone----astersik----provider----asterisk----sip phone? |
| 18:54.08 | p3nguin | We're not talking about digital, we're talking about VoIP. |
| 18:54.59 | p3nguin | And your provider isn't going to support G.722, so you just broke the whole end to end thing I told you about. |
| 18:55.09 | r0m|u | well voip is digital? |
| 18:55.23 | p3nguin | No. |
| 18:55.28 | p3nguin | VoIP is VoIP. |
| 18:55.42 | p3nguin | Digital, in the terms of telephony is something completely different. |
| 18:55.54 | p3nguin | See also: Panasonic PBX |
| 18:56.02 | r0m|u | ah. I see. voip.ms does not support G722 even thought they tell you to enable it only if your system supprots it? |
| 18:56.20 | r0m|u | Ok got it! |
| 18:56.43 | p3nguin | End to end, meaning your phone will use g722, your asterisk will support g722, and the other phone you're calling which is connected to your asterisk supports g722. |
| 18:57.02 | r0m|u | wow so calling voip digital is a miss concept. I learn something new today :) |
| 18:57.11 | p3nguin | Or if you peer with another asterisk, that asterisk must support g722 and any phone you call on that system has to support g722. |
| 18:57.19 | r0m|u | go it. |
| 18:57.51 | p3nguin | At any point in the line of communication, if the codec is a lesser quality, your overall call quality will be reduced to that lower quality codec. |
| 18:58.06 | [TK]D-Fender | b0ot, Better to make a peer than to shov IP's PW's, etc into the dilaplan. |
| 18:58.58 | p3nguin | So you can set up your phone and asterisk to use g722, and any call you make to asterisk will be g722. |
| 18:59.23 | p3nguin | Set up another phone on asterisk to use g722, and call it from your phone... it'll be g722 end to end. |
| 18:59.36 | p3nguin | But call your ITSP, and the call will be reduced to ulaw at best. |
| 18:59.47 | r0m|u | p3nguin, ok. I see. I am experimenting with codes from one asterisk to another. |
| 19:00.08 | p3nguin | If you can peer them with g722, go for it. |
| 19:00.49 | r0m|u | awesome. me and my brother are finally connected from asterisk to asterisk. |
| 19:01.23 | r0m|u | I am looking for a phone for him. he wants a cisco phone |
| 19:01.27 | p3nguin | If you both support g722 on your handests, use g722. You'll hear a difference. |
| 19:01.31 | p3nguin | handsets |
| 19:01.47 | r0m|u | awesome. |
| 19:01.53 | p3nguin | I like my 7960G. |
| 19:02.04 | r0m|u | noted. |
| 19:02.09 | p3nguin | But if you want to use SIP, consider a SIP phone. |
| 19:03.32 | r0m|u | got it. |
| 19:03.34 | r0m|u | brb |
| 19:04.02 | p3nguin | The newer line of Cisco phones is SIP. |
| 19:05.55 | carrar | They should use some propriatary protocol |
| 19:13.30 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 19:13.36 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
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| 19:17.33 | *** join/#asterisk clintc (~clintc@n128-227-125-126.xlate.ufl.edu) |
| 19:22.51 | p3nguin | I'm having a problem with a lot of delay after entering a dtmf choice in a menu. If I change 4 to _4! am I asking for trouble? It makes it accept 4 instantly instead of waiting for a timeout. |
| 19:23.23 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-oyhzqoqopeclvneu) |
| 19:23.32 | [TK]D-Fender | it won't wait unless there is something longer that could match |
| 19:24.09 | [TK]D-Fender | if you have 4000 in there it will wait. If you dial 41 it will stop immediately on the 1 |
| 19:24.41 | [TK]D-Fender | "As soon as there is no otehr number you could be dialing ; accept" |
| 19:26.42 | vader-- | legit or not legit: http://www.ebay.com/itm/TDM800P-asterisk-card-4FXO-4FXS-a800p-TDM400P-/120769413945?pt=LH_DefaultDomain_0&hash=item1c1e6b0339 |
| 19:26.43 | vader-- | ? |
| 19:28.13 | p3nguin | It waits because there could be more to match. But I don't want it to wait, I want it to be immediate if 4 is entered. Is _4! safe to use where someone could dial more than just 4? |
| 19:29.18 | *** join/#asterisk jaybee_ (~quassel@114.31.212.161) |
| 19:29.38 | [TK]D-Fender | "where someone could dial more than just 4?" = you can't just accept 4 instantly |
| 19:29.59 | jaybee_ | Hi all. Is there any way to send DTMF to the person who initiated the call? SendDTMF sends it to the recipient |
| 19:30.28 | [TK]D-Fender | jaybee_, When? How? |
| 19:32.00 | jaybee_ | Ok, so I call an extension from my phone. I want it to make DTMF noises to me |
| 19:32.59 | jaybee_ | I just realised my question makes little sense. Let me think about this, and come back to you :) |
| 19:33.20 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 19:36.26 | p3nguin | I believe the reason there is a long delay is because I have includes. |
| 19:36.48 | p3nguin | In an included context, 4xx would be possible to dial. |
| 19:37.06 | p3nguin | But I don't really NEED 4xx to be reachable. |
| 19:37.19 | p3nguin | So I'd love for the 4 to be accepted straight away. |
| 19:37.34 | p3nguin | I've done it with _4! but I'm looking for caveats. |
| 19:38.00 | *** join/#asterisk jjlee (~john@cpc3-nmal4-0-0-cust1414.croy.cable.virginmedia.com) |
| 19:38.24 | p3nguin | But now that I think about it, I think I'm going to restructure the menu to get away from the includes. I'll make it so that if you want to get to those included extensions, you'll first enter another key to get to another menu level. |
| 19:39.05 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:40.13 | lordvadr | Can we discuss SSRC and RFC2833 for a little bit? |
| 19:41.18 | lordvadr | Asterisk 1.8 changes the SSRC for each DTMF event. I'm finding conflicting statements about whether this is correct, acceptable, or completely incorrect. Can anybody shed any light on this? |
| 19:43.28 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 19:43.35 | blizzow | If I do a queue show command, and I see one of the lines in a "ringing" state, does that mean it's an inbound ring or the user is making an outbound call and it's "ringing"? |
| 19:44.19 | p3nguin | I think I found a bug in Directory(). |
| 19:44.38 | [TK]D-Fender | Should be calling the agent |
| 19:45.19 | p3nguin | I press * in the directory, and it says invalid extension. The console says Invalid extension '*' in context 'internal', but pressing * in the directory is supposed to jump to 'a' as per core show application Directory. |
| 19:45.40 | p3nguin | And extension 'a' does exist in internal, so that's not the cause. |
| 19:47.19 | *** part/#asterisk clintc (~clintc@n128-227-125-126.xlate.ufl.edu) |
| 19:48.43 | [TK]D-Fender | Didn't think Directory used the escapes like Vicemail does |
| 19:49.33 | mjordan | p3nguin: which version? |
| 19:50.33 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 19:51.10 | p3nguin | 1.8.7.1 |
| 19:51.32 | *** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net) |
| 19:51.49 | p3nguin | Let me confirm it is trying to run * instead of a. |
| 19:54.12 | p3nguin | Disregard. It seems to work as expected. Not sure... maybe I forgot a dialplan reload or something. |
| 19:54.20 | p3nguin | Bug averted. |
| 19:54.25 | mjordan | :-) |
| 19:55.12 | p3nguin | I don't think that's it, though, because I have always had 'a' in internal, and the console verbose output clearly said it was trying to find * in internal. |
| 19:56.04 | p3nguin | Ah, I found it! |
| 19:56.54 | p3nguin | I pressed 0 in the directory. I do not allow that, so there is no 0 in internal. i in internal says invalid extension and then has a WaitExten to wait for a valid one. I pressed * while in WaitExten rather than while in Directory. |
| 19:57.15 | mjordan | ah |
| 19:57.43 | p3nguin | Calling Directory fresh and entering * took me to 'a' in internal as it should have. |
| 19:59.58 | tzanger | I've already asked the so-called "experts" I know ( :-) ) ... does anyone know of anything even remotely multi-vendor for auto provisioning sip devices? a kind of bonjour or upnp kind of system for advertising voip termination? |
| 20:07.27 | [TK]D-Fender | tzanger, FreePBX's EPM is multi-vendor IIRC, and it doesn't sound like a stretch to have it scan a network. |
| 20:08.47 | *** join/#asterisk francisvgarcia (~francis.g@190.80.239.124) |
| 20:13.33 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-217-233.inter.net.il) |
| 20:15.32 | tzanger | [TK]D-Fender: hmm, I'll check out EPM and see where it leads me |
| 20:15.34 | tzanger | thanks |
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| 20:45.17 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 20:51.54 | jeffspeff | how do i set the moh for when people are actually on hold? i have the moh class defined in the different dial()'s... however, when you dial an exten you hear the moh instead of ringing (like it's supposed to), but after the call is answered and the person puts the line/channel on hold, it goes to the [default] moh context instead of what's defined in the dial(). |
| 20:53.07 | [TK]D-Fender | jeffspeff, because its the calls of the person they are talking to. |
| 20:53.24 | [TK]D-Fender | the calling channel doesn't crontrol the hold. The one who initiates the hold does |
| 20:53.55 | jeffspeff | [TK]D-Fender, how do i change that? |
| 20:54.03 | [TK]D-Fender | You don't |
| 20:54.10 | [TK]D-Fender | (the behaviour) |
| 20:54.21 | [TK]D-Fender | yuo can go ahead and change your extensions MoH though |
| 20:56.15 | jeffspeff | ok, so just so we are clear... if you call me right now, you'll hear [custom_moh_context] while my phone is "ringing". after i answer the call and put you on hold, it will play you the [default] moh from my * box. and there's no way to change it from [default] |
| 20:56.45 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
| 21:02.02 | [TK]D-Fender | Stop attaching classnames on thins |
| 21:02.11 | [TK]D-Fender | the HOLDER's MoH class is what gets used |
| 21:03.15 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
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| 21:10.15 | vader-- | hmmm i wish flowroute.com had another payment method... Paying through amazon is kinda weird |
| 21:11.37 | *** join/#asterisk garymc (~chatzilla@host31-53-157-167.range31-53.btcentralplus.com) |
| 21:21.41 | p3nguin | Pay them with your credit card or PayPal. |
| 21:24.10 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 21:24.35 | cj | have any of you trunked google voice calls with 1.8? |
| 21:25.09 | p3nguin | Trunked? No. Used Google Voice with 1.8? Yes. |
| 21:27.39 | *** join/#asterisk beccara (~beccara@mail.ubergroup.co.nz) |
| 21:29.20 | p3nguin | Is fail2ban really so stupid that it insists on banning my local network devices, even though I have ignoreip for my correct localnet? |
| 21:29.43 | tompaw | Hi, what's the current approach to dynamic peers/friends (preferably in pgsql/mysql)? |
| 21:29.49 | tompaw | Still have to recompile * to enable? |
| 21:33.18 | tompaw | http://www.voip-info.org/wiki/view/Asterisk+sip+mysql+peers << this is dated 2005 |
| 21:35.51 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 21:36.32 | Qwell | and this is why we suggest not using voip-info anymore. |
| 21:36.55 | tompaw | ok, what do you suggest using? |
| 21:37.07 | tompaw | the official wiki? |
| 21:37.23 | Qwell | That would be a good start. |
| 21:37.24 | p3nguin | Is there a way to have asterisk use a phone's auth name rather than its user name for registrations? For example, I'd like the user name to be the extension number, but the auth name is the MAC address. This config on the phone causes asterisk to reject due to no matching peer. |
| 21:37.50 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:38.17 | tompaw | Qwell: ok, so this: https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration ? |
| 21:39.33 | tompaw | ok, sounds like an up-to-date stuff :P |
| 21:40.22 | cj | p3nguin: that should be good enough to start :) |
| 21:40.31 | r0m|u | p3nguin, how can I enable asterisk to spill out everything to a log file? |
| 21:41.07 | p3nguin | cj: There's a wiki page for it. It's somewhere on wiki.asterisk.org. |
| 21:41.20 | p3nguin | r0m|u: see logger.conf |
| 21:41.27 | r0m|u | Thanks p3nguin |
| 21:48.20 | *** join/#asterisk tenex (~tenex@unaffiliated/tenex) |
| 21:53.41 | r0m|u | p3nguin, I got a gui going for my brother with arch... but is sort of buggy so I abandon it. Looks like is going to be freepbx for him. |
| 21:54.25 | r0m|u | I was trying to keep it minimal but with gui you have to pay the price :P |
| 21:55.20 | p3nguin | "a gui" |
| 21:55.44 | r0m|u | sorry. more specific. an asteric webui |
| 21:55.49 | r0m|u | asterisk* |
| 21:55.53 | p3nguin | There's a FreePBX package for Arch, too. |
| 21:55.59 | r0m|u | really? |
| 21:56.09 | p3nguin | Or you could dump Arch altogether and use AsteriskNOW. |
| 21:56.44 | r0m|u | well I was trying to keep it very low on mem foot print. I like the fact that arch uses only 40Mb |
| 21:57.29 | r0m|u | since I am the one who is doing it for him I rather stic with the same thing I am using |
| 22:02.19 | r0m|u | ill keep at it with arch. he is not going to touch it any way. |
| 22:02.38 | tompaw | A couple of questions about ARA - I want to use realtime sipusers. 1) Will the existing users from sip.conf be ignored or added to the realtime ones? 2) Are all columns in a table needed? 3) Can a table have EXTRA columns - not related to ARA? |
| 22:04.21 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:06.31 | tompaw | This is like the worst-documented feature ever :D I'm gonna try and create a Django object for sipfriends. |
| 22:09.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:13.34 | p3nguin | r0m|u: I guess you've discovered why I'm using Arch for my tiny Asterisk system. |
| 22:19.23 | tompaw | Why is a configuration for res_config_pgsql in res_pgsql.conf? |
| 22:19.32 | tompaw | for mysql there is a _config_ in the file name |
| 22:20.27 | tompaw | Also, it would be cool if it was possible to pick an IP which * should use when connecting to pgsql. |
| 22:20.33 | p3nguin | It's probably something similar to how chan_sip is configured in sip.conf, but chan_dahdi is configured in chan_dahdi.conf. |
| 22:37.05 | tompaw | How do I debug ARA? Have set verbosity to 99, no errors. |
| 22:37.11 | tompaw | Can't tell if it even tries to touch my database. |
| 22:37.23 | tompaw | Yet registration doesn't work. |
| 22:37.39 | *** join/#asterisk mario-goulart (~user@67.205.85.241) |
| 22:40.39 | *** join/#asterisk coppice (~chatzilla@14.63.101.80) |
| 22:46.18 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-eiaypxgavrfraslu) |
| 22:47.33 | tompaw | No, seriously, how do I debug it? Not a single error message anywhere, yet core show configuration shows that ARA iss enabled. |
| 22:48.44 | [TK]D-Fender | I'd start by looking at all of this.... |
| 22:49.50 | tompaw | Realtime mapping for 'sipusers' found to engine 'pgsql', but the engine is not available |
| 22:49.53 | tompaw | lol |
| 22:50.43 | [TK]D-Fender | That looks like a pretty clear message |
| 22:51.10 | tompaw | But I had to trigger it with my clever digging into "realtime" cli command ;) |
| 22:56.52 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 22:58.45 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 23:02.43 | tompaw | Hm... now when I do "realtime load sipusers name 771" it shows me the stuff from the database. |
| 23:03.54 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 23:04.21 | tompaw | But it doesn't seem to want to actually use it for SIP authentication: http://pastecode.com/eW |
| 23:10.16 | [TK]D-Fender | You should have a type field. |
| 23:10.58 | tompaw | OK |
| 23:14.29 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 23:19.45 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 23:20.57 | blizzow | Once in a while one of my users will complain that they dial out to a customer and hear no ring tone. I called our PRI provider, they monitored the circuit for two hours and saw nothing. Is there something I can look for in asterisk that will tell me WTF is going on? |
| 23:21.33 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-bbvdqogkyuexbdwl) |
| 23:21.34 | blizzow | A user who is suffering the problem usually gets the symptom for a few minutes, and then it goes away. |
| 23:22.04 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-tonuxdwzmhcdvibv) |
| 23:25.16 | p3nguin | What does this mean? |
| 23:25.22 | p3nguin | WARNING[32072]: chan_sip.c:6433 sip_write: Can't send 10 type frames with SIP write |
| 23:33.21 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 23:34.53 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ffwidwuoezgkibbo) |
| 23:39.22 | tompaw | [TK]D-Fender: working fine, thanks! |
| 23:39.59 | [TK]D-Fender | tompaw: You're welcome |
| 23:40.33 | p3nguin | It floods the console when calling from one phone. I'd really like to stop it. |
| 23:40.58 | tompaw | I wonder if anyone has any suggestions on recording confbridge calls. |
| 23:52.03 | *** join/#asterisk Jupe (~rez@fl-71-55-208-24.dhcp.embarqhsd.net) |