| 00:00.00 | mvanbaak | lol |
| 00:00.06 | hax | seanbright: do they 'sound' a particular way? or, how are you supposed to go about knowing what's the issue? |
| 00:00.10 | mvanbaak | I'll have to take my laptop |
| 00:00.26 | mvanbaak | going to talk to olle about this multiparking stuff I took over from him |
| 00:00.28 | seanbright | hax: you should ask on irc in #asteri... oh wait... crap |
| 00:01.00 | hax | seanbright: yeah, i'm currently trying that, but i'm not sure if they're really going to be able to help... though i think it's probably a fairly common question |
| 00:01.07 | mvanbaak | I'm redoing the 'multiple parking contexts' branch he once did |
| 00:01.12 | mvanbaak | but he lost interest |
| 00:01.19 | mvanbaak | and I want to get it merged |
| 00:01.19 | seanbright | hax: well i feel like i've let you down. |
| 00:01.25 | seanbright | hax: i'm just going to go... |
| 00:01.36 | mvanbaak | so I'm updating his branch. but it's lagging for 8 months |
| 00:01.39 | hax | seanbright: heh |
| 00:01.40 | puzzled | mvanbaak: saw the branch msgs on the ml |
| 00:01.49 | mvanbaak | basically it means I have to do it all over again |
| 00:02.02 | mvanbaak | might be a good thing to sit down with olle and talk it over |
| 00:02.12 | mvanbaak | can be on thursday, or on the fosdem |
| 00:02.23 | mvanbaak | we both will be there |
| 00:02.37 | nvrpunk | is it notransfer=no or transfer=no |
| 00:02.51 | nvrpunk | to disable native iax transfering |
| 00:02.52 | mvanbaak | nvrpunk: IAX ? |
| 00:02.56 | nvrpunk | yeah |
| 00:03.17 | puzzled | iirc notransfer=yes |
| 00:03.29 | mvanbaak | transfer=no |
| 00:03.29 | puzzled | but you can find it in the sample configs |
| 00:03.34 | mvanbaak | in trunk |
| 00:03.49 | mvanbaak | ;transfer=no ; Disable IAX native transfer |
| 00:03.49 | mvanbaak | ;transfer=mediaonly ; When doing IAX native transfers, transfer |
| 00:03.49 | mvanbaak | <PROTECTED> |
| 00:04.13 | hax | seanbright: it's just odd because it seems to work fine, but it's like there's little dropouts, almost like when a cell phone doesn't sound perfect |
| 00:04.17 | mvanbaak | nvrpunk: look in the asterisk sources dir: configs/iax.conf.sample |
| 00:04.26 | hax | seanbright: i'm thinking maybe it's the codec or something, but i don't see any statistics or anything i could use to see what's going wrong |
| 00:04.49 | mvanbaak | 5% battery power left ;) |
| 00:05.23 | hax | seanbright: actually, maybe it's just my microphone |
| 00:05.32 | mvanbaak | meh, my nagios is sending me emails |
| 00:06.11 | mvanbaak | 'thinkpad warning - battery power is below 5%' |
| 00:06.11 | mvanbaak | latero all |
| 00:06.11 | puzzled | night |
| 00:09.15 | *** join/#asterisk ecto (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 00:21.22 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 00:21.52 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 00:22.50 | *** join/#asterisk Docfxit (i=ExUser@netblock-208-127-208-174.dslextreme.com) |
| 00:25.31 | *** join/#asterisk Sniffadog (n=cameron@CPE-121-223-233-73.static.vic.bigpond.net.au) |
| 00:25.38 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 00:25.39 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 00:25.51 | Sniffadog | Hi! |
| 00:26.43 | nvrpunk | so question, we have SIP phones going out an IAX2 to a pstn gateway (junction networks) and we are getting 50packets per second per call |
| 00:26.54 | nvrpunk | on the tx |
| 00:26.55 | Sniffadog | anyone able to answer basic asterisk questions? |
| 00:27.03 | nvrpunk | and then 150 pps on the rx |
| 00:27.17 | nvrpunk | if it was trunking shouldnt the calls combine into the same packet? |
| 00:27.22 | LiNeTuX | Sniffadog: sometimes :) |
| 00:27.26 | nvrpunk | and thus only 50 pps for two calls? |
| 00:36.32 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 00:45.38 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 00:47.06 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 00:47.24 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-759019249004e23d) |
| 00:48.39 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:56.31 | *** join/#asterisk MaliutaWrk (n=nikolai@203.201.152.211) |
| 01:01.03 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 01:01.44 | *** join/#asterisk vrtk (n=bruno@189.70.254.86) |
| 01:03.39 | jameswf-home | pong |
| 01:09.09 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:09.09 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 01:15.44 | *** join/#asterisk egypcio (n=vinicius@unaffiliated/egypcio) |
| 01:18.46 | ManxPower | perhaps trunking is enabled in one direction only |
| 01:26.53 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.181.66) |
| 01:41.43 | Docfxit | I just connected a new install to phone lines. If I try calling out I get a dial tone. I'm calling 91805xxxxxxx or 9xxxxxxx If I try calling in I get the voice menu. I can call in from the outside to an extension. |
| 01:42.07 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 01:42.07 | Docfxit | I think my dialing rules are set up correctly. |
| 01:42.28 | Docfxit | I have re-booted all the phones. |
| 01:43.17 | Docfxit | They are Polycom phones. The display has black phones that are solid black. |
| 01:44.13 | *** join/#asterisk Kumbang (n=kumbang@125.163.83.153) |
| 01:49.43 | *** part/#asterisk Docfxit (i=ExUser@netblock-208-127-208-174.dslextreme.com) |
| 01:51.07 | *** join/#asterisk Docfxit (i=ExUser@netblock-208-127-208-174.dslextreme.com) |
| 01:51.33 | Docfxit | <PROTECTED> |
| 01:51.33 | Docfxit | [5:42pm] <Docfxit> I think my dialing rules are set up correctly. |
| 01:51.33 | Docfxit | [5:42pm] <Docfxit> I have re-booted all the phones. |
| 01:51.33 | Docfxit | [5:43pm] <Docfxit> They are Polycom phones. The display has black phones that are solid black. |
| 01:52.56 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-69-154.lns10.syd6.internode.on.net) |
| 02:02.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:07.43 | Wayhigh | that dang mouse somehow got out of the glue trap.. |
| 02:07.52 | Wayhigh | it must have only got one paw in it |
| 02:08.39 | Wayhigh | I've made a decent trap for it now.. glue traps funnelling the mouse to the other live catch trap |
| 02:09.52 | LiNeTuX | Wayhigh: just make one big 120V metal platform. It steps on it, volia'! Cooked mouse! |
| 02:11.06 | Wayhigh | that's a decent idea.. wonder if I can find a pressure switch light enough so I don't have to have the mouse cooker running all the time. |
| 02:11.22 | Mw3 | actually a 230V one would be better :D |
| 02:12.16 | Wayhigh | yeah but that'd certainly fry anything it touched |
| 02:12.26 | Wayhigh | including people.. |
| 02:12.40 | LiNeTuX | Wayhigh: or even better, put in on an angle so when it crossed the threshold for the bait... switch... zappo! |
| 02:12.41 | Wayhigh | there's a good chance you're coming away from a 120v shock but not a 240v |
| 02:13.16 | Wayhigh | i'm just praying there's nothing bigger than a mouse in my basement.. |
| 02:13.27 | Wayhigh | cause whatever got stuck on the trap drug it about 1.5 feet |
| 02:13.41 | LiNeTuX | When the lights go dim... |
| 02:17.28 | nvrpunk | ive got whacked by 240 three phase |
| 02:17.30 | nvrpunk | it hurts |
| 02:17.33 | nvrpunk | but not deadly |
| 02:17.51 | nvrpunk | you feel for about 1 hour after :/ |
| 02:18.10 | drmessano | All 3 phases? |
| 02:18.13 | drmessano | or 1 phase? |
| 02:18.23 | LiNeTuX | i don't think people have 3 hands :) |
| 02:18.23 | nvrpunk | think just one or two |
| 02:18.32 | drmessano | If it was more than one, you wouldnt be here |
| 02:18.32 | nvrpunk | I was holding the metal parts of my multimeter :p |
| 02:18.33 | obnauticus | lol |
| 02:18.41 | obnauticus | i could attach 3 phases to my penis |
| 02:18.44 | obnauticus | at the same time |
| 02:18.46 | obnauticus | with some sort of machine |
| 02:18.52 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:18.55 | tengulre | hi,all |
| 02:19.00 | nvrpunk | it's the amps that get you |
| 02:19.01 | nvrpunk | :p |
| 02:19.10 | obnauticus | ya high voltage just burns you really badly. |
| 02:19.15 | obnauticus | a quarter of an amp can kill you |
| 02:19.16 | billytwowilly | smells like burnt hotdogs.. |
| 02:19.25 | nvrpunk | 1/4 amp across the heart |
| 02:19.26 | drmessano | one phase, you're a highly resistive path to ground, 2 or more and you become a low resistant path between phases, and you end up being the fuse |
| 02:19.31 | drmessano | So 2 phases, gone |
| 02:19.39 | nvrpunk | 1 amp inderectly is survivable |
| 02:20.11 | drmessano | So Im guessing you touched 1 phase :) |
| 02:20.17 | obnauticus | man |
| 02:20.20 | nvrpunk | drmessano, didn;t put much thought about it, I just know it hurt |
| 02:20.21 | obnauticus | i need a 3 phase generator |
| 02:20.24 | obnauticus | i can't run big PDU'sb |
| 02:20.28 | obnauticus | because i don't have a 3 phase |
| 02:20.28 | obnauticus | "\ |
| 02:21.13 | drmessano | 3 phase is fun |
| 02:21.20 | drmessano | Espeically when you single phase |
| 02:21.21 | obnauticus | how do you get your 3 phase? |
| 02:21.34 | drmessano | Off the pole lol |
| 02:21.43 | obnauticus | how much do they make you pay for that |
| 02:22.04 | drmessano | No clue.. Cant remember the last power bill I had to deal with |
| 02:22.14 | obnauticus | it's not that expensive though is it |
| 02:22.15 | drmessano | No more than the basic price of power per kw |
| 02:22.21 | nvrpunk | obnauticus, it was equipment off the army I got whacked with |
| 02:22.22 | nvrpunk | :P |
| 02:22.25 | Wayhigh | obnauticus: the problem is.. the voltage taking the shortest path to the ground may go through your heart which causes it to stop beating.. |
| 02:22.44 | drmessano | 30ma is all it takes |
| 02:22.44 | Wayhigh | so yeah.. high voltage does burn.. but it's the path it travels to ground that is the real problem |
| 02:23.13 | drmessano | Bumping 240 with your hand is no biggie |
| 02:23.16 | drmessano | Sucks, but you live |
| 02:23.30 | drmessano | grabbing something else when you're doing it.. BYE BYE |
| 02:23.32 | Wayhigh | dude.. have you ever seen what happens to people that grab high voltage lines? |
| 02:23.39 | drmessano | Yes |
| 02:23.47 | Wayhigh | the arc blows out the backside of their leg usually.. it's pretty nasty looking |
| 02:23.49 | Qwell | he IS a dr afterall |
| 02:23.54 | drmessano | I saw a pic of a guy that blew himself up stealing copper |
| 02:24.06 | drmessano | heh |
| 02:24.14 | obnauticus | pix? |
| 02:24.18 | drmessano | Some guy was stealing copper from... a substation |
| 02:24.24 | Wayhigh | man.. if you're stealing live copper you so deserve what you get |
| 02:24.33 | drmessano | and somewhere I have a real graphic pic of his 3 or 4 pieces |
| 02:25.04 | nvrpunk | that were left? |
| 02:25.15 | nvrpunk | i saw a picture of a marine who bit a blasting cap and lived :( |
| 02:25.18 | nvrpunk | pretty nasty |
| 02:25.23 | Wayhigh | that's about as good as siphoning gas using a shopvac |
| 02:27.06 | nvrpunk | http://www.rockymountainnews.com/drmn/local/article/0,1299,DRMN_15_4835565,00.html |
| 02:28.54 | drmessano | I'll crack my laptop open later and put that pic online.. its apparently not on my personal machine |
| 02:30.40 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:30.40 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:32.31 | *** join/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 02:34.43 | drmessano | Family Guy.. good stuff |
| 02:34.55 | Wayhigh | wonder if I can find someone with a lot of co2 I can pump into my basement.. |
| 02:35.06 | drmessano | heh |
| 02:35.22 | Wayhigh | remove family from home.. pump co2 into basement/house.. no more live mice |
| 02:35.39 | drmessano | Put a couple 18 inch JBL speakers down there |
| 02:35.43 | Wayhigh | hahaha |
| 02:35.45 | drmessano | 500 watt crown AMP |
| 02:35.49 | Docfxit | I just connected a new install to phone lines. If I try calling out I get a dial tone. I'm calling 91805xxxxxxx or 9xxxxxxx If I try calling in I get the voice menu. I can call in from the outside to an extension. |
| 02:35.56 | drmessano | Generate a 20,000 cycle tone |
| 02:36.00 | drmessano | and wait |
| 02:36.09 | Docfxit | I have re-booted all the phones. |
| 02:36.18 | Wayhigh | just figuring I'll drive it/them out? |
| 02:36.23 | Docfxit | They are Polycom phones. The display has black phones that are solid black. |
| 02:36.25 | drmessano | oh yes |
| 02:36.46 | Docfxit | My dialing rules are set up correctly. |
| 02:37.16 | drmessano | Once you get about 16,000, humans lose out, and the rats start getting it |
| 02:37.24 | drmessano | 20,000 is a good number |
| 02:37.41 | Docfxit | Any ideas why I only get a dial tone when calling out? |
| 02:39.12 | *** join/#asterisk LiuYan (n=LiuYan_f@211.154.128.135) |
| 02:44.09 | drmessano | HOLY CRAP |
| 02:44.13 | drmessano | http://www.compoundsecurity.co.uk/teenage_control_products.html |
| 02:44.18 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 02:44.35 | Qwell | where have you been? |
| 02:44.53 | drmessano | Who |
| 02:45.08 | Qwell | you |
| 02:45.16 | drmessano | What ever do you mean? |
| 02:45.19 | Wayhigh | some places are finding those teenage control products illegal to use |
| 02:45.47 | Wayhigh | and personally.. I hate them because I may be 35 but I can hear the anti-mosquito racket and it really is VERY irritated to my ears |
| 02:45.51 | drmessano | I guess in a cave.. didnt know someone made use of the technology like that... an outdoor mounted teenager deterrent.. |
| 02:46.47 | drmessano | I was trying to find some data on hearing range loss with age.. and I found that |
| 02:46.56 | *** join/#asterisk pkunkra (n=chris@cpe-74-73-10-32.nyc.res.rr.com) |
| 02:47.02 | Wayhigh | some teenagers near my place were shocked when they found out that I could hear their ringtones |
| 02:47.09 | *** part/#asterisk LiuYan (n=LiuYan_f@211.154.128.135) |
| 02:47.34 | Wayhigh | we were on the metro and I asked them to shut off their ringtones cause it was disorienting |
| 02:48.07 | drmessano | We tried that at work last year with some decent speakers and Adobe Audition generating the tones.. I think my limit was 15,000.. and the 20 yr olds were up near 18,000 |
| 02:48.31 | Wayhigh | it's strange.. I used to have a 40% hearing loss.. |
| 02:48.53 | Wayhigh | now I'm wondering if there's just some lower tones I have a hard time hearing but can thear the upper ones more clearly |
| 02:48.59 | drmessano | My hearing sucks.. too much rock music as a teenager |
| 02:49.03 | Wayhigh | s/thear/hear/ |
| 02:49.17 | drmessano | Its possible |
| 02:52.55 | LiNeTuX | This Knight Rider sucks. It's nothing more than a Ford & MS commercial. |
| 02:54.29 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 02:57.07 | *** join/#asterisk Telek (n=rimnar@018.120-113-64.ftth.swbr.surewest.net) |
| 02:57.55 | Telek | Hey, can someone point me to a tutorial on setting up asterisk with a WRTP54G for the sip lines? |
| 02:58.24 | *** join/#asterisk Docfxit (i=ExUser@netblock-208-127-208-174.dslextreme.com) |
| 02:58.42 | Docfxit | I just connected a new install to phone lines. If I try calling out I get a dial tone. I'm calling 91805xxxxxxx or 9xxxxxxx If I try calling in I get the voice menu. I can call in from the outside to an extension. |
| 02:58.59 | Docfxit | I have re-booted all the phones. |
| 02:59.14 | Docfxit | They are Polycom phones. The display has black phones that are solid black. |
| 02:59.23 | Telek | I had one set up at some point in the past, but don't have the config files handy to figure out what I'm doing wrong. It's registering fine, and it has some communication with the asterisk server (since pressing # gets me a message about 'thank you for using asterisk, blah blah blah), but actually dialing of extensions hangs for about 10 seconds then throws a busy signal. |
| 02:59.25 | Docfxit | My dialing rules are set up correctly. |
| 02:59.36 | Docfxit | Any ideas why I only get a dial tone when calling out? |
| 03:09.47 | drmessano | WTF |
| 03:09.52 | drmessano | This isnt Knight Rider |
| 03:09.56 | scooby2 | nope |
| 03:09.59 | drmessano | This is "The OC Rider" |
| 03:10.09 | scooby2 | lolz |
| 03:10.27 | drmessano | Who the eff is MIKE? |
| 03:10.30 | drmessano | MIKE? |
| 03:10.54 | drmessano | Screw this.. I want Hasselhoff.. I want treating women like sex objects |
| 03:11.48 | drmessano | Solar powered and energy efficient? HA.. BURN SOME GAS KITT, LIKE ITS 1984 |
| 03:13.21 | drmessano | Next they'll bring Airwolf back, and it will be a Hovercraft |
| 03:14.46 | scooby2 | a stealth solarpowered huey |
| 03:15.24 | drmessano | Welcome to 2008.. KITT: Powered by Windows Mobile |
| 03:15.28 | *** join/#asterisk tomierna (n=newbie@pool-72-77-231-220.tampfl.fios.verizon.net) |
| 03:15.40 | LiNeTuX | drmessano: that's why it can't outrun the POS other Fords |
| 03:16.39 | drmessano | "Michael, I can't arm the missiles right now, I have to reboot to install this windows update" |
| 03:17.05 | LiNeTuX | "But KITT!" "I'm sorry Mike. There's 32 critical updates - just this hour." |
| 03:17.15 | tomierna | Hi. Trying to set up a digium AA-50 for the first time. It's not publishing DHCP as promised. Anyone know of any tricks? |
| 03:17.22 | drmessano | HA |
| 03:18.11 | LiNeTuX | KITT: "Oh dear. I seem to have been 0wn34d. Michael, what does that mean?" |
| 03:20.40 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 03:21.18 | Nugget | Windows and Ford -- seems like a perfect match to me. |
| 03:22.08 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:22.08 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:23.55 | Nugget | then again, if KITT ran Linux Michael would never get out of the garage. He'd just be surrounded by wrenches and poring over the FUEL INJECTION HOWTO trying to recolve a conflict between the KDE ECU and and the gnome-maf-0.02b package. |
| 03:24.30 | Nugget | and of course the limited slip differential package is abandonware writtedn by some 19 year old liberal arts student while he was high. |
| 03:27.29 | styelz | and then ya die |
| 03:28.13 | *** join/#asterisk Faithful (n=Faithful@ppp246-13.static.internode.on.net) |
| 03:30.33 | *** join/#asterisk jbigbee (n=jbigbee@24.96.154.69) |
| 03:30.45 | *** part/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 03:31.58 | drmessano | Nugget: ROFL |
| 03:32.31 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 03:34.07 | russellb | sooooooo ... |
| 03:34.19 | russellb | someone start fighting or something, i'm bored |
| 03:34.36 | jbigbee | ha, what's going on russel? |
| 03:34.55 | russellb | waiting on a delayed flight ... |
| 03:36.06 | russellb | ~hug file |
| 03:36.06 | jbot | ACTION gets a running start and tackle-hugs file |
| 03:36.09 | LiNeTuX | Gaaahhh! You dirty brat! Look what you've done! I'm MELTING! MELTING! |
| 03:36.12 | file | eeeeeeep |
| 03:36.17 | MrTelephone | russell |
| 03:36.23 | russellb | MrTelephone |
| 03:36.28 | MrTelephone | when is the next asricon |
| 03:36.28 | jbigbee | russellb, so I've decided to pick up a challenge. I'm going to run/swim/cycle in a triathlon. Think Digium would sponsor me? |
| 03:36.31 | MrTelephone | astricon |
| 03:36.36 | russellb | MrTelephone: astricon.net :) |
| 03:36.41 | MrTelephone | are you going? |
| 03:36.46 | russellb | jbigbee: i ... have no idea |
| 03:36.50 | russellb | MrTelephone: of course :) |
| 03:37.02 | MrTelephone | russellb, some kids rooted my server with that new vmsplice exploit |
| 03:37.04 | file | jbigbee: I would totally give you currency |
| 03:37.30 | jbigbee | file, there are qualifying events in canada. I'd be sure to use it |
| 03:37.40 | russellb | MrTelephone: should i know what you're referring to? :) |
| 03:37.50 | russellb | Nugget: oh god, that's terrible |
| 03:38.03 | jbigbee | I can see it, Ironman triathlon 2008 won by Digium Sales Rep. |
| 03:38.37 | drmessano | OH WOW |
| 03:38.40 | file | Nugget: you probably mean X101P |
| 03:38.46 | scooby2 | MrTelephone: why did they have local system access? |
| 03:38.51 | drmessano | X101P 4 LIFE |
| 03:38.51 | jbigbee | Nugget, I get a lot of requests for that card |
| 03:39.03 | MrTelephone | a mistake |
| 03:39.47 | drmessano | If you install 8 X101P cards, you end up with a box with 8 X101Ps... :/ |
| 03:39.57 | Docfxit | tomierna » I'm no expert. But I believe for DHCP you need to have another box that has a DHCP server. |
| 03:40.13 | drmessano | wow |
| 03:40.21 | drmessano | Let me get this right |
| 03:40.27 | russellb | drmessano: did you just say 8 == 8 ? |
| 03:40.32 | drmessano | They can't turn KITT back on because he's exploitable |
| 03:40.32 | russellb | that's brilliant |
| 03:40.58 | jbigbee | it's insane |
| 03:41.12 | drmessano | No, I said the equivalent of.. If someone gives you 6 blocks of SPAM for christmas, 6 people hated you enough to give you SPAM for christmas |
| 03:41.25 | Docfxit | I just connected a new install to phone lines. If I try calling out I get a dial tone. I'm calling 91805xxxxxxx or 9xxxxxxx If I try calling in I get the voice menu. I can call in from the outside to an extension. |
| 03:41.31 | russellb | drmessano: oic. |
| 03:41.34 | Docfxit | I have re-booted all the phones. |
| 03:41.41 | Nugget | to paraphrase jwz... |
| 03:41.44 | Docfxit | They are Polycom phones. The display has black phones that are solid black. |
| 03:41.55 | Docfxit | My dialing rules are set up correctly. |
| 03:41.55 | Nugget | "Some people, when faced with a problem, say to themselves 'I know! I'll use an X100P" |
| 03:41.58 | drmessano | HA |
| 03:42.00 | Nugget | "Now they have TWO problems." |
| 03:42.00 | drmessano | YES |
| 03:42.05 | Docfxit | Any ideas why I only get a dial tone when calling out? |
| 03:42.15 | drmessano | They turned KITT back on, and quickly enabled the Windows Firewall |
| 03:42.17 | drmessano | AWESOME! |
| 03:42.23 | russellb | Docfxit: because your extension that dials out is probably wrong ... |
| 03:42.27 | russellb | Docfxit: pastebin it |
| 03:42.29 | russellb | ~pb |
| 03:42.29 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:42.47 | russellb | ~kitt |
| 03:43.02 | Docfxit | russellb» What should I pastebin? |
| 03:43.12 | russellb | the extension that handles dialing out that isn't working |
| 03:43.12 | MrTelephone | docfxit, change your dtmfmode in sip.conf to rfc2855 or whatever it is |
| 03:43.24 | LiNeTuX | THAT was the windows firewall |
| 03:43.26 | drmessano | jbot: KITT is the Knight Industries Two Thousand, the car of the future |
| 03:43.27 | jbot | drmessano: okay |
| 03:43.29 | drmessano | hah |
| 03:43.33 | LiNeTuX | Effective, no? |
| 03:43.54 | drmessano | What they did was more like pfsense |
| 03:44.01 | LiNeTuX | heh |
| 03:44.07 | Docfxit | russellb» All extensions are the same. Only dial tone when dialing out. |
| 03:44.20 | MrTelephone | anyone know where to get a cisco as5300 with 2 t1s with 60 dsps for under 5 thousand :( |
| 03:44.50 | MrTelephone | docfxit, your not breaking tone if your not sending a digit |
| 03:45.05 | drmessano | Actually, if you want to be technically correct, his father is Michael Long, not Michael Knight |
| 03:45.13 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 03:47.36 | Docfxit | MrTelephone » My Sip.conf is the same when I had all phones plugged into an AsteriskNow setup. I installed Asterisk into Ubuntu with the same conf files . |
| 03:48.10 | Docfxit | MrTelephone » I don't see dtmfmode in sip.conf. I'll look at the samples. |
| 03:50.22 | MrTelephone | ok |
| 03:50.31 | Telek | Huh |
| 03:51.19 | MrTelephone | paste some log output into pastebin of an outgoing call |
| 03:51.26 | *** join/#asterisk AndyGraybeal (n=andy@node220.36.251.72.1dial.com) |
| 03:51.41 | Docfxit | Okay |
| 03:51.51 | *** join/#asterisk oneeyedelf1 (n=knic@c-71-227-234-32.hsd1.or.comcast.net) |
| 03:52.01 | Telek | Okay, I have asterisk setup fine and the router configured to connect, and capable of using the demo extensions, however they do no seem to be configuring properly for dialing amongst themselves... any thoughts? |
| 03:54.42 | Docfxit | MrTelephone » How can I find a log of an outgoing call. The Asterisk log doesn't say much. |
| 03:55.11 | MrTelephone | asterisk -rvvvvv |
| 03:56.46 | MrTelephone | it worked in asterisknow? |
| 03:59.09 | *** join/#asterisk FuturePrimitive (n=stephenm@c-24-7-186-196.hsd1.ca.comcast.net) |
| 04:00.11 | FuturePrimitive | Good evening. I am trying to get a census on which Asterisk distribution seems to be the most popular. So far, PIAF and Elastix seems to be the top picks? Any other great recommendations? |
| 04:00.49 | FuturePrimitive | Also, I am looking for which distribution works best with 1.6 |
| 04:00.55 | FuturePrimitive | I need the TCP features. |
| 04:01.27 | MrTelephone | why do you need tcp |
| 04:02.15 | FuturePrimitive | I have a gateway server between my other pbx that only works using SIP TCP. I have tried to work with OpenSER and SipX, but I would rather everything in one box. |
| 04:02.44 | MrTelephone | nice |
| 04:02.44 | russellb | i don't think there are any distributions that support 1.6 yet |
| 04:02.48 | russellb | you will have to install that yourself |
| 04:02.48 | FuturePrimitive | I tried the TCP patch, but it seemed really buggy. |
| 04:02.50 | MrTelephone | what brand pbx? |
| 04:03.06 | russellb | the patch isn't the same code that went into 1.6 ... |
| 04:03.09 | Docfxit | MrTelephone » http://www.pastebin.ca/908132 Example of outgoing call log. |
| 04:03.37 | FuturePrimitive | yeah, I heard they completely rewrote the TCP part for 1.6. |
| 04:03.37 | FuturePrimitive | Thats why I was interested. |
| 04:04.08 | FuturePrimitive | Now dont flame me here, but the other PBX is OCS. |
| 04:04.55 | MrTelephone | docfxit, it looks like it should work |
| 04:05.04 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 04:05.24 | Docfxit | I can't figure out why it won't. |
| 04:05.44 | Docfxit | I have the same config files from the other box when it did work. |
| 04:06.04 | Docfxit | I have the same card from the other box. |
| 04:06.29 | Docfxit | I do get incoming calls just fine. |
| 04:06.35 | MrTelephone | and they sound ok? |
| 04:06.43 | Docfxit | yes. |
| 04:06.46 | FuturePrimitive | So, my best bet for 1.6 is to just install it on top of say CentOS 5.1 and then load up FreePBX 2.4? |
| 04:07.09 | FuturePrimitive | Anyone have experience with 1.6 yet? |
| 04:07.09 | MrTelephone | can you setup an extension so that it dials ZAP/g1 without the number |
| 04:07.11 | MrTelephone | ? |
| 04:07.26 | Docfxit | How? |
| 04:07.28 | MrTelephone | setup extension 9 so it dials ZAP/g1 |
| 04:07.42 | MrTelephone | do you know how to do that? |
| 04:07.48 | Docfxit | no. |
| 04:08.25 | Docfxit | I have a GUI installed. |
| 04:08.33 | MrTelephone | ohh |
| 04:08.58 | Docfxit | That may make it easier. |
| 04:09.04 | MrTelephone | can you snoop on the call with an analog handset to see if its even sending the digits? |
| 04:10.17 | Docfxit | I have an analog handset. I'd have to figure out which line it's going out on. I'll try. |
| 04:10.45 | MrTelephone | your telco might not be recognizing dtmf or something |
| 04:13.08 | *** join/#asterisk Faithful (n=Faithful@ppp246-13.static.internode.on.net) |
| 04:13.51 | FuturePrimitive | Has anyone had any issues installing 1.6? |
| 04:14.08 | Docfxit | The same phone lines/connection are in use now as with the AsteriskNow box that worked just fine. |
| 04:14.31 | MrTelephone | is the zapata.conf the same too? |
| 04:14.36 | MrTelephone | the rx/tx gains? |
| 04:15.21 | MrTelephone | i dont want to send you on a wild goose chase though |
| 04:18.42 | Docfxit | yes. it's the same |
| 04:19.02 | Docfxit | I'll gook at the gains. |
| 04:20.02 | Docfxit | The gains are both set a 5. The same as on the other box. |
| 04:20.37 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-112-81.knology.net) |
| 04:21.24 | *** join/#asterisk HelloWorl (n=jer5345@wsip-68-110-218-186.ks.ok.cox.net) |
| 04:22.44 | HelloWorl | i'm trying to set up my asterisk for dialing out pstn via tdm400p |
| 04:22.59 | Docfxit | I connected an analog phone up to every line while dialing out. I only heard dial tone. no tones. |
| 04:23.17 | HelloWorl | not having very good success.. |
| 04:24.01 | Docfxit | MrTelephone »Go ahead and send me on a wild goose chase. I'm desperate to get this up and running tonight. |
| 04:24.37 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 04:25.09 | Docfxit | Could it be it's not recognizing the card eventhough the GUI says it is? |
| 04:26.43 | Docfxit | Do I need a dtmfmode line in sip.conf. even though it worked without it in AsteriskNow? |
| 04:32.37 | ManxPower | Docfxit: stop comparing with Asterisknow |
| 04:32.48 | *** join/#asterisk Telek (n=rimnar@018.120-113-64.ftth.swbr.surewest.net) |
| 04:33.00 | Docfxit | I currently don't have dtmfmode = rfc2833 in my sip.conf Should I put it in? |
| 04:33.19 | ManxPower | Docfxit: Does it work without it? |
| 04:33.39 | Docfxit | ManxPower >> not now. |
| 04:33.51 | Docfxit | It did before. |
| 04:34.05 | Docfxit | Not with this build. |
| 04:34.32 | Docfxit | I'm in the U.S. Is that the correct tone? |
| 04:34.54 | Docfxit | It's the default in the sample file. |
| 04:40.17 | Docfxit | dtmfmode = rfc2833 I put it in the sip.conf, activated changes. It didn't help. |
| 04:45.06 | cmantito | hmm |
| 04:48.28 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 04:49.37 | drmessano | I have a question |
| 04:49.43 | drmessano | Im missing something stupid |
| 04:50.06 | drmessano | Wait |
| 04:50.07 | drmessano | nm |
| 04:50.27 | styelz | hehe |
| 04:51.29 | styelz | pebkac |
| 04:58.03 | *** part/#asterisk FuturePrimitive (n=stephenm@c-24-7-186-196.hsd1.ca.comcast.net) |
| 05:01.47 | *** join/#asterisk Benny_132 (n=benmarti@CPE-61-9-142-76.static.vic.bigpond.net.au) |
| 05:04.59 | Benny_132 | Hi all im having a problem which is i have 2 cards a digium wildcard TE405P E1 card and a Sangoma A200 analog card, the problem is that the analog card is coming in on random zap channels instead of the range 32-35 which i have set E1 uses 1-15, D-channel 16 than 17-21 |
| 05:07.24 | JT | Benny_132: E1 uses 1-15 for B chans, and 17-31 |
| 05:09.29 | Benny_132 | thanks i'll make that change we only have 10 lines anyway, the E1 works perfect its jsut the analog lines when when u ring in come in on like Zap/1-1 instead of Zap/32-1 like it says in the zaptel and zapata conf files |
| 05:10.27 | cmantito | yawns |
| 05:10.35 | cmantito | so I've decided to make a phone based IRC client XD |
| 05:11.09 | styelz | heh |
| 05:11.26 | cmantito | not like, mobile based, there's lot of those, voice based, using ast |
| 05:11.28 | cmantito | ^^ |
| 05:12.05 | cmantito | I've already got it connecting to servers and reading off chatter in the room |
| 05:12.10 | cmantito | now I need to design an input method |
| 05:13.06 | styelz | can you change channels and join channels with voice commands |
| 05:13.52 | cmantito | not yet |
| 05:14.02 | cmantito | I can't afford voice recognition :P |
| 05:14.08 | cmantito | so I'm gonna have to do something with keypad input |
| 05:14.41 | styelz | oh fun |
| 05:14.56 | cmantito | I haven't found any open source voice recogs yet anyway :P |
| 05:19.08 | mihinomenest | of course not, they're too profitable. |
| 05:19.24 | *** join/#asterisk AndyGraybeal (n=andy@node220.36.251.72.1dial.com) |
| 05:26.15 | styelz | you could set it up to controll a botnet remotely i guess... |
| 05:26.38 | sweeper | 23:14 < cmantito> I haven't found any open source voice recogs yet anyway :P |
| 05:26.41 | sweeper | sphinx |
| 05:27.00 | styelz | its not very good though is it |
| 05:27.04 | cmantito | I was looking at that but from what I ready it didn't work with asterisk cause of .. something. |
| 05:27.07 | styelz | didnt work well on zork |
| 05:27.13 | styelz | at least |
| 05:27.32 | sweeper | naw, doesn't work well, like mihinomenest said, good voice recog makes too much money |
| 05:27.35 | styelz | zoip |
| 05:28.18 | sweeper | I dislike even good voice recognition anyways |
| 05:28.31 | cmantito | I don't need it to work well, does it work at all with ast then? |
| 05:29.01 | cmantito | "It is fairly easy to integrate Asterisk with Sphinx, the only trouble is that you |
| 05:29.01 | cmantito | need to have an Acoustic Model (AM) for 8KHz, which are not (yet) readily available." |
| 05:29.09 | cmantito | are they available at this point? |
| 05:30.43 | sweeper | probably not, see the next note |
| 05:30.53 | drmessano | I've almost got grandcentral pwn3d |
| 05:32.14 | cmantito | damn |
| 05:43.22 | drmessano | HAHA |
| 05:43.24 | drmessano | Guys |
| 05:43.27 | drmessano | I have great news |
| 05:43.37 | cmantito | ? |
| 05:43.43 | drmessano | Chris Pirillo figured out how to get Fax over VOIP working.. |
| 05:43.44 | drmessano | I mean |
| 05:43.50 | drmessano | All the brainpower in here |
| 05:43.54 | drmessano | Screw you guys.. |
| 05:43.59 | drmessano | Chris Pirillo got it |
| 05:44.01 | drmessano | http://chris.pirillo.com/2008/02/17/how-to-fax-over-voip-on-the-internet/ |
| 05:44.39 | cmantito | doesn't look any different than all the other things I've read :P |
| 05:45.04 | drmessano | Looks like the same lame ass, half ass attempts at making work that everyone else has tried |
| 05:45.05 | drmessano | But het |
| 05:45.07 | drmessano | hey* |
| 05:45.18 | drmessano | ITS CHRIS PIRILLO! THE LOCKERGNOME! |
| 05:45.23 | drmessano | It's DONE |
| 05:46.05 | drmessano | "When all else fails, call your VOIP provider." |
| 05:46.19 | drmessano | Yes, i'm sure they would like to know that Fax over VOIP doesn't work |
| 05:46.34 | drmessano | Because, you know, they don't already.. |
| 05:47.23 | drmessano | This is why I love twitter.. It's like a troll search engine with chat built in |
| 06:02.58 | HelloWorl | anyone up to helping get my cisco 7960 working with asterisk? (pbx iaf) |
| 06:03.48 | HelloWorl | i was able to upgrade the phone to SIP 8.8 successfully..so i'm like a quarter of the way there...i think |
| 06:09.16 | *** join/#asterisk MaliutaWrk (n=nikolai@203.201.152.211) |
| 06:18.16 | jameswf-home | ~troll |
| 06:18.17 | jbot | troll is probably a race on some muds; a guy under a bridge: an annoying robot; a port scanner, or somebody faking being clueless to be shown the One Linux Way so he can argue against it for his psychology thesis on linux advocates, or an employee of trolltech, or at http://www.kuro5hin.org/story/2001/7/27/51233/2979 or ... |
| 06:18.40 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:19.25 | jameswf-home | ~haxors |
| 06:22.20 | J4k3 | ~h4x |
| 06:22.35 | J4k3 | jbot: h4x is the core coding style for Linux and all related projects |
| 06:22.36 | jbot | J4k3: okay |
| 06:22.47 | drmessano | Oh god |
| 06:22.56 | J4k3 | haha |
| 06:23.04 | drmessano | Stallone is going to resurrect his character from "Cliffhanger" for another movie |
| 06:23.07 | jameswf-home | ~omfg |
| 06:23.08 | jbot | i heard omfg is "oh my fluffy gerbil" |
| 06:23.13 | drmessano | HA |
| 06:23.32 | jameswf-home | my gerbil is declawed |
| 06:23.54 | jameswf-home | ~britney |
| 06:24.09 | jameswf-home | ~idnms |
| 06:24.09 | jbot | Why would a Wookiee, an eight-foot tall Wookiee, want to live on Endor, with a bunch of two-foot tall Ewoks? That does not make sense! But more important, you have to ask yourself: What does this have to do with this channel? Nothing |
| 06:24.20 | drmessano | ha |
| 06:24.24 | drmessano | ~idk |
| 06:24.31 | drmessano | ~drmessano |
| 06:24.31 | jbot | you are probably the leading cause of censorship in #asterisk, or a jbot junky in training |
| 06:25.59 | jameswf-home | ~42 |
| 06:26.00 | jbot | from memory, 42 is the answer to life the universe and everything, see also http://en.wikipedia.org/wiki/the_answer_to_life,_the_universe,_and_everything |
| 06:26.04 | jameswf-home | ~88 |
| 06:26.05 | jbot | from memory, 88 is two fat ladies |
| 06:26.12 | drmessano | ~73 |
| 06:26.21 | jameswf-home | ~98 |
| 06:26.27 | jameswf-home | ~69 |
| 06:26.28 | jbot | hmm... 69 is something you must learn to know more about.. looks the same if you turn it upside down |
| 06:26.52 | drmessano | ~73 |
| 06:26.53 | jbot | from memory, 73 is Ham Radio speak for "10-4, over and out" |
| 06:26.55 | drmessano | I rule |
| 06:27.15 | jameswf-home | ~stupid human |
| 06:27.35 | drmessano | ~HappyClownPhone |
| 06:27.37 | jameswf-home | ~id10t |
| 06:27.37 | jbot | it has been said that id10t is a chair to keyboard user interface error. |
| 06:27.39 | drmessano | ~HappyClownPBX |
| 06:27.40 | jbot | [happyclownpbx] currently in closed beta, is close to 7GB in size, and it pwns |
| 06:28.00 | jameswf-home | ~centpbx |
| 06:28.01 | drmessano | I need to fix that |
| 06:28.12 | jameswf-home | ~centpbx is dead |
| 06:28.12 | jbot | jameswf-home: okay |
| 06:28.16 | jameswf-home | ~centpbx |
| 06:28.16 | jbot | from memory, centpbx is dead |
| 06:28.53 | jameswf-home | ~adminsparadise is dead |
| 06:28.53 | jbot | jameswf-home: okay |
| 06:29.12 | jameswf-home | ~pbxinaflash |
| 06:29.12 | jbot | from memory, pbxinaflash is Ward Mundy's toy assembled by joe roper visit pbxinaflash.org or #pbxinaflash |
| 06:29.22 | drmessano | ~HappyClownPBX |
| 06:29.22 | jbot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, and it pwns |
| 06:30.11 | jameswf-home | I hear its just another crap trixbox clone |
| 06:30.38 | drmessano | I heard it has more bloat |
| 06:30.42 | drmessano | Not sure I believe that |
| 06:32.28 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:34.39 | jameswf-home | lets go to our judges |
| 06:34.46 | jameswf-home | ~paula |
| 06:34.46 | jbot | I dunno I am on the fence |
| 06:34.53 | jameswf-home | ~randy |
| 06:34.53 | jbot | for me it was just kina aight dawg yeah just aight |
| 06:34.58 | drmessano | ha |
| 06:34.59 | jameswf-home | ~simon |
| 06:35.00 | jbot | That was utterly and completely mind numbingly painful I would rather debug windows |
| 06:35.08 | drmessano | HAHHA |
| 06:35.43 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 06:36.07 | HelloWorl | i'm trying to set up cisco 7960 for pbx iaf...any suggestions? |
| 06:36.15 | jameswf-home | ~cisco |
| 06:36.15 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 06:36.20 | HelloWorl | haha |
| 06:37.41 | HelloWorl | well...i managed to get sip 8.8 loaded...what to do now...i have tftpboot with SIPDefault.cnf...when i call asterisk...i get fast busy |
| 06:41.26 | HelloWorl | no helpy? |
| 06:41.34 | HelloWorl | well i tried...thx |
| 06:43.20 | drmessano | I guess I am gonna go to bed.. big day tomorrow |
| 06:46.48 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 06:47.11 | nvrpunk | how do I make it so a certain set of phones dont route out? |
| 06:47.19 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 06:50.15 | SwK | nvrpunk, read up on contexts |
| 06:50.16 | *** join/#asterisk ixx (i=foobar@cpe-70-112-149-89.austin.res.rr.com) |
| 07:09.04 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 07:09.10 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.57) |
| 07:12.05 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 07:12.08 | loompek | morning |
| 07:12.20 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:12.37 | loompek | anyone of you guys ever successfully registered sysmaster tornado m20 to your asterisk server? |
| 07:13.15 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:14.20 | *** join/#asterisk steliosk (n=Stelios@athedsl-291405.home.otenet.gr) |
| 07:20.00 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.103.65) |
| 07:20.32 | g0mb0 | hi, I'm looking for good open source h323<->sip signalling proxy |
| 07:20.50 | g0mb0 | are there such software? |
| 07:31.18 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-10fe7bb5d212cbf6) |
| 07:41.25 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 07:44.01 | nvrpunk | can you have 1 user with 2 contexts? |
| 07:48.47 | sweeper | anyone around know much about mrtg? I want to know how to put info from two devices on one graph...and there's no #mrtg here :P |
| 07:57.51 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:01.48 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 08:02.58 | *** join/#asterisk RealUser19633469 (n=adib@dxb-b14859.alshamil.net.ae) |
| 08:03.06 | RealUser19633469 | hi |
| 08:03.16 | RealUser19633469 | i'm having a problem with adding destination of DID in a2billing, i need to see if someone knows how to fix it |
| 08:03.26 | RealUser19633469 | anyone alive? |
| 08:04.50 | tzafrir | RealUser19633469, maybe. But you also provided practically 0 information about the problem |
| 08:05.15 | tzafrir | Unless the fact that you use a2billing is the problem |
| 08:06.08 | RealUser19633469 | the problem is that i want to forward the DID to an iax user |
| 08:06.23 | RealUser19633469 | how shall i type the iax address? |
| 08:08.32 | RealUser19633469 | tzafrir, can pvt u |
| 08:08.48 | tzafrir | It won't really help |
| 08:08.55 | tzafrir | I don't know a2billing |
| 08:09.16 | RealUser19633469 | u know how does iax exntsion looks like? |
| 08:09.30 | RealUser19633469 | i mean for sip it will be sipid@domainname or ip |
| 08:09.36 | RealUser19633469 | how does iax looks like? |
| 08:10.29 | nvrpunk | RealUser19633469, is the last leg of the DID voip? |
| 08:10.37 | nvrpunk | coming into you |
| 08:10.45 | RealUser19633469 | yes |
| 08:10.55 | RealUser19633469 | it is coming i can see it in asterisk -vvvvr |
| 08:10.57 | RealUser19633469 | Enter the phone number you wish to call, or the VoIP client to reach. (ie: 347894999 or SIP/jeremy@182.212.1.45). If the call is VoIP, the VoIP_Call must be set to yes. |
| 08:11.02 | nvrpunk | RealUser19633469, you have to contact them then. Its tricky |
| 08:11.19 | nvrpunk | already did reasearch on using them for our billing |
| 08:11.20 | RealUser19633469 | to contact who nvrpunk |
| 08:11.22 | nvrpunk | it can be done |
| 08:11.25 | nvrpunk | a2billing |
| 08:11.29 | nvrpunk | their support |
| 08:11.30 | nvrpunk | :P |
| 08:11.36 | RealUser19633469 | i did everything , there is nothing about it online man |
| 08:11.44 | nvrpunk | yeah i know |
| 08:11.45 | nvrpunk | :/ |
| 08:11.59 | nvrpunk | we are getting a custom build of a2billing from them |
| 08:12.22 | nvrpunk | they already informed us DID inward with last leg voip is tricky |
| 08:12.23 | nvrpunk | to setup |
| 08:12.38 | nvrpunk | -inward as DID obviously states that |
| 08:13.11 | RealUser19633469 | from who man |
| 08:13.22 | nvrpunk | once we have our custom build I am sure i would be able to answer the question |
| 08:13.24 | nvrpunk | the devs |
| 08:13.37 | nvrpunk | anyhow, contact them off their site |
| 08:13.39 | nvrpunk | and ask |
| 08:14.01 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 08:15.07 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 08:20.13 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:23.55 | *** join/#asterisk duckz (n=duckz@81.180.102.217) |
| 08:27.29 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 08:30.26 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 08:32.10 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 08:39.17 | *** join/#asterisk slavon_net (n=slavon@slavon.bigtelecom.ru) |
| 08:44.33 | *** join/#asterisk qdk_ (n=qdk@85.235.253.139) |
| 08:50.06 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-52f73bbd5a6d9f57) |
| 08:51.32 | *** join/#asterisk glen2 (n=glen@87-194-2-134.bethere.co.uk) |
| 08:55.38 | *** join/#asterisk Sajjad_Ali_Musht (n=Sajjad_A@octroi.enst-bretagne.fr) |
| 08:58.09 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 09:04.10 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 09:08.49 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:14.14 | cmantito | http://www.voip-info.org/wiki/view/Sphinx <-- the AGI script on that page, is that called using AGI() or Perl()? |
| 09:14.26 | cmantito | ...correction, the Perl example AGI script. |
| 09:15.07 | cmantito | I know that question should really answer itself, but a one word answer would be appreciated ;) |
| 09:16.12 | *** join/#asterisk gazzerh (n=garryh@79.173.139.34) |
| 09:18.05 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 09:23.40 | *** join/#asterisk af_ (n=getsmart@88-149-240-219.dynamic.ngi.it) |
| 09:23.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:36.54 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 09:48.05 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-04e8b6010dce35a8) |
| 09:49.44 | *** join/#asterisk nirz (n=nir@mail2.tikalnetworks.com) |
| 09:52.30 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:53.49 | cmantito | anyone happen to know if res_perl works under ast 1.4? |
| 10:00.26 | RoyK | cmantito: prolly portable |
| 10:00.52 | cmantito | well I can't get it to compile on it's own, and afaict, it's suposed to be part of asterisk-addons, but I don't see it in there -_- |
| 10:01.25 | cmantito | so I was just looking for any information anyone might have on trying to make it work lol :P |
| 10:13.46 | tzafrir | cmantito, AFAIK, it doesn't |
| 10:14.00 | cmantito | -_- |
| 10:14.01 | cmantito | to think |
| 10:14.06 | cmantito | I just recompiled perl for this. |
| 10:14.11 | cmantito | ah well |
| 10:14.17 | cmantito | next hopeful solution. |
| 10:15.18 | cmantito | thanks tzafrir |
| 10:16.18 | loompek | so i guess you had no success with tornado m20 :S |
| 10:51.47 | *** join/#asterisk ahbritto (n=guest@adsl-69-104-3-183.dsl.pltn13.pacbell.net) |
| 10:52.16 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 10:54.33 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 10:57.58 | *** join/#asterisk bakerboi (n=bakerboi@dsl-203-113-237-8.SA.netspace.net.au) |
| 11:01.31 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 11:02.42 | bakerboi | Hi... can anyone please help me with a NAT problem? |
| 11:05.26 | *** join/#asterisk fiddur (i=fiddur@78.82.198.246) |
| 11:05.47 | fiddur | Hello. I just installed asterisk-gui from svn trunk. It starts well, but the "Service Providers"-part in cfgbasic and the same in setup/install just throws me back to the first page in that context (that is, cfgbasic.html or setup/install.html). It says "Loading" for a short while, and then loads start page again... There's no output in the console... Where can I start debugging? |
| 11:05.50 | *** join/#asterisk quigon (n=matias@32.59.64.130) |
| 11:06.22 | fiddur | oh sorry, missed that there was an #asterisk-gui channel. |
| 11:07.40 | *** join/#asterisk quigon (n=matias@32.59.64.129) |
| 11:15.35 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:29.32 | *** join/#asterisk jks (n=jks@0x503e4c12.abnxx10.adsl-dhcp.tele.dk) |
| 11:29.42 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 11:30.43 | tzafrir | fiddur, not that there are some many people on it :-( |
| 11:38.49 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 11:38.56 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 11:42.12 | *** join/#asterisk davidcsi (n=davidcsi@180.Red-213-97-249.staticIP.rima-tde.net) |
| 11:43.23 | davidcsi | anyone knows what "asterisk[4085]: rc_avpair_new: unknown attribute 1490026597" is? i've only got iax with trunking running on that box |
| 11:43.59 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:47.03 | ruied | can I use the set(callerID()) in sip.conf? I have two fax machines and would like to set the outbubound number XXX for fax X and outbound number YYY for fax Y. I'm trying to do with GotoIF() is there a better method? |
| 11:48.05 | ruied | not outbound, the fax public number |
| 11:48.43 | loompek | [Feb 18 12:47:59] WARNING[13061]: chan_sip.c:12621 handle_response: Remote host can't match request NOTIFY to call '63dc0b42005a482a0cd1e36170c0e70b@1.2.3.4'. Giving up. |
| 11:48.59 | loompek | i've got quite a few of this messages |
| 11:49.03 | loompek | because of tornado m20 |
| 11:49.24 | loompek | anyone ever heard of it? or even configured it |
| 11:50.23 | jks | anyone knows how to force a channel "on hold" from asterisk, instead of from the SIP phone? |
| 11:51.17 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 11:51.57 | *** join/#asterisk Aurs (n=Ove_Aurs@85.80-203-122.nextgentel.com) |
| 11:57.02 | davidcsi | jks, i think you can park it.. |
| 11:58.02 | davidcsi | jks exten => 6000,1,Answer |
| 11:58.03 | davidcsi | <PROTECTED> |
| 11:59.34 | *** join/#asterisk Avatar42 (n=chris@p57B12823.dip0.t-ipconnect.de) |
| 12:05.19 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 12:09.03 | loompek | that won't place the call 'on hold' |
| 12:10.01 | yangvnc | Why does error framein: no samples for ulawtoalaw happen ? |
| 12:10.07 | yangvnc | kje si loompek |
| 12:10.28 | *** part/#asterisk zamba (i=marius@sveigde.hih.no) |
| 12:16.27 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 12:19.07 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 12:21.37 | *** join/#asterisk byte_slave (n=chatzill@213.63.2.234) |
| 12:21.50 | byte_slave | hello! |
| 12:23.57 | byte_slave | i want to have a wepage with internal extension, customers numbers, etc and i would like to do soemting that when i clicked in a contact asterisk dial it. I would have a first page login, for me to know who wants call a specific contact and after that establish the connection automatically |
| 12:24.09 | byte_slave | any ideas how to do this? |
| 12:24.59 | jks | davidcsi, well, that is for a "new call" or how you would phrase it |
| 12:25.11 | jks | davidcsi, I'm looking for something that can take an existing conversation and put it on hold |
| 12:25.38 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 12:25.39 | jks | davidcsi, (and obviously be able to resume the conversation a while later) |
| 12:43.38 | davidcsi | anyone knows what "asterisk[4085]: rc_avpair_new: unknown attribute 1490026597" is? |
| 12:47.23 | *** join/#asterisk nebojsajsimic (n=nebojsaj@cable-89-216-16-106.static.sbb.co.yu) |
| 12:47.26 | nebojsajsimic | hi all |
| 12:47.41 | nebojsajsimic | can someone use php agi??? |
| 12:48.01 | nebojsajsimic | does someone use php agi>??? |
| 12:48.26 | nebojsajsimic | i need some explanation for exec_dial |
| 12:50.33 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 12:51.24 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 12:52.13 | nebojsajsimic | when i answer call i don't get res 'ANSWER' |
| 12:54.09 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 12:59.05 | *** join/#asterisk Stingy1 (n=tbaa@80.120.42.206) |
| 12:59.08 | Stingy1 | hi |
| 12:59.34 | Stingy1 | i try to complie app_asr.c with asterisk 1.4.18 |
| 13:00.04 | Stingy1 | but i get some errors. has anybody else try this? |
| 13:02.24 | *** join/#asterisk Victor_Yure (n=Victor_Y@unaffiliated/victoryure/x-837844) |
| 13:04.50 | *** join/#asterisk demlak (i=demlak@schwarz-pUnK.de) |
| 13:04.54 | *** part/#asterisk demlak (i=demlak@schwarz-pUnK.de) |
| 13:07.21 | *** part/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 13:10.50 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 13:11.05 | defswork | byte_slave: asterisk call files would be the easiest way to go |
| 13:12.01 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 13:17.55 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:19.03 | *** join/#asterisk miltux (n=miltux@62.1.19.195) |
| 13:21.52 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 13:23.00 | lirakis | morning |
| 13:25.37 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.232.128) |
| 13:26.53 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:28.28 | *** join/#asterisk saint_mon (n=gag@124.104.47.115) |
| 13:30.07 | saint_mon | is it possible to call from iax2(zoiper) to sip(xlite) on the same box with vmware(centos,asterisk) in it? |
| 13:31.28 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:34.24 | *** join/#asterisk Skamakazi (n=jogi@license.querix.com) |
| 13:36.11 | Skamakazi | Hello, does anyone have any experience with the grandstream gxp-2000 phones? Im getting a weird problem where even though the GMT offset is set to 0 in the config file, the phone is setting itself to be GMT-12 |
| 13:36.21 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 13:39.43 | *** part/#asterisk Skamakazi (n=jogi@license.querix.com) |
| 13:41.36 | *** join/#asterisk Duplicator (n=Duplicat@proxy.fmpa.com) |
| 13:44.15 | nebojsajsimic | can someone help me with phpagi |
| 13:44.16 | nebojsajsimic | ??? |
| 13:53.01 | *** join/#asterisk geminidomino (n=ciro@65.41.157.192) |
| 13:53.48 | geminidomino | Stumped.. Anyone have any experience on what might cause this: WARNING[9023] app_dial.c: Unable to create channel of type 'ZAP' (cause 0 - Unknown) |
| 13:55.41 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 13:56.08 | tzafrir | geminidomino, that's a very generic error message |
| 13:56.12 | tzafrir | Anything before it? |
| 13:56.47 | byte_slave | desfwork, i ear about taht, i'll give it a shot and see if it applies to my needs, thanks |
| 13:57.02 | byte_slave | defswork, i ear about taht, i'll give it a shot and see if it applies to my needs, thanks |
| 13:57.02 | geminidomino | tzafrir: No, that's what makes it so odd. The first warning preceeding it is just a dialplan glitch reporting. |
| 13:57.07 | x86 | geminidomino: do 'zap show channels' in CLI |
| 13:57.21 | x86 | make sure zaptel is loaded |
| 13:57.23 | geminidomino | x86: It is |
| 13:57.38 | x86 | geminidomino: pastebin the output of zap show channels |
| 13:57.45 | x86 | http://pastebin.ca/ |
| 13:58.08 | geminidomino | http://pastebin.com/d4b2d8918 |
| 14:02.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:02.46 | x86 | ok, now show us your Dial statement |
| 14:04.07 | *** join/#asterisk ddunavant (n=David@pool-96-231-71-126.washdc.east.verizon.net) |
| 14:04.54 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 14:06.54 | *** join/#asterisk jbigbee (n=jbigbee@216.207.245.1) |
| 14:08.47 | geminidomino | the log message? |
| 14:10.02 | geminidomino | <PROTECTED> |
| 14:10.07 | geminidomino | (phone number replaced) |
| 14:10.34 | x86 | yeah no wonder |
| 14:10.37 | x86 | fix the Dial ;) |
| 14:10.46 | x86 | also, pastebin both zaptel.conf and zapata.conf |
| 14:11.00 | geminidomino | what's wrong with the Dial? |
| 14:11.13 | x86 | show me the Dial from the dialplan |
| 14:11.20 | anonymouz666 | wow. d-fender changed his nickname to x86 ;) |
| 14:11.38 | x86 | anonymouz666: no, I'm just the protege ;) |
| 14:16.27 | *** join/#asterisk saftsack (n=oliver@p54A7E440.dip.t-dialin.net) |
| 14:16.38 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 14:17.31 | geminidomino | I can't find it. So a broken Dialplan can cause a "cause 0", or are you just messing with me? |
| 14:17.35 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 14:26.20 | *** join/#asterisk jhb (n=joerg@81-5-139-2.dsl.eclipse.net.uk) |
| 14:27.30 | jhb | hi *. I have an agi-script that uses Dial(Sip/123&SIP/456). How can I catch the case that nobody answers the call (I would like to send an xml-rpc request) |
| 14:27.54 | lirakis | geminidomino: your dial is messed up |
| 14:28.14 | lirakis | geminidomino: you should use & to seperate multiple peers |
| 14:28.56 | lirakis | geminidomino: unless thats .. not your actual dial statement |
| 14:29.10 | geminidomino | that's what's showing up in the log |
| 14:29.26 | geminidomino | I'm still trying to untangle this #$! dialplan to find the actual |
| 14:31.15 | *** join/#asterisk ming_zym (n=ming_zym@220.181.54.7) |
| 14:31.42 | lirakis | geminidomino: sounds like youve got other issues... i mean .. if you cant find the Dial call |
| 14:32.25 | geminidomino | Oh, I've got issues aplenty. I just didn't want to hose my dialplan if it turned out to be a hardware problem. |
| 14:33.22 | lirakis | geminidomino: (shrug) dont know.. but its unlikely |
| 14:33.27 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:33.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:33.40 | *** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 14:33.50 | Zeeek | anyone using an AA50? |
| 14:34.04 | geminidomino | All right. Guess I'll have to tear it down and start again then. Thanks |
| 14:35.47 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:38.55 | nebojsajsimic | How to catch answer in php agi dial ??? |
| 14:41.16 | nebojsajsimic | exec_dial in php agi don't send back Answer just -1 when call end any idea for this prob |
| 14:41.18 | nebojsajsimic | ?? |
| 14:43.52 | x86 | geminidomino: just search your dial plan for Dial statements... grep is your friend ;) |
| 14:44.07 | x86 | geminidomino: grep -r 'Dial' /etc/asterisk/extensions.conf |
| 14:44.09 | geminidomino | x86: There's a lot of them. |
| 14:44.22 | geminidomino | So I was trying to backtrack to find out which one it was |
| 14:44.26 | x86 | ok, limit it further by dials out of the g2 group |
| 14:44.58 | geminidomino | if it's a dialplan issue, then you guys can't help me anyways. :) |
| 14:45.44 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:45.46 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 14:45.49 | x86 | geminidomino: no? |
| 14:45.59 | x86 | heh |
| 14:46.10 | x86 | sounds like you do not desire help ;) |
| 14:46.11 | geminidomino | x86: Nope. That puts it into FPBX clusterfsck territory. |
| 14:46.14 | lirakis | x86: i dont think i want to see his dial blan |
| 14:46.16 | lirakis | *plan |
| 14:46.27 | lirakis | geminidomino: ahh .. well your in the wrong channel friend |
| 14:46.28 | lirakis | ;) |
| 14:46.29 | geminidomino | Not at all. I just don't desire to be flamed for not reading the /topic :) |
| 14:46.50 | geminidomino | lirakis: I know. That's why I wasn't asking about the dplan here. |
| 14:48.57 | *** join/#asterisk seanbright (n=elixer@65.207.74.18) |
| 14:50.18 | SteveTotaro | looks like a tough room this morning |
| 14:51.20 | SteveTotaro | riddlebox around? |
| 14:51.21 | x86 | SteveTotaro: never a dull moment ;) |
| 14:53.24 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
| 14:56.39 | patrick-- | Hey, my asterisk suddently stopped during the day and i dont have a clue why. |
| 14:56.44 | patrick-- | can someone tell me how to debug? |
| 14:57.42 | deeperror | <PROTECTED> |
| 14:57.43 | lirakis | patrick--: look at your log files /var/log/asterisk/ |
| 14:57.50 | patrick-- | messages doesnt show anything |
| 15:01.01 | *** part/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 15:01.11 | patrick-- | where would i have to look? |
| 15:01.26 | saftsack | is there a channel for electronic purposes? nand-gatters and microcontrollers? |
| 15:01.28 | deeperror | <PROTECTED> |
| 15:01.38 | deeperror | may have dumps in there |
| 15:02.10 | patrick-- | mhh |
| 15:02.23 | patrick-- | nah dumps is nothing |
| 15:03.29 | *** join/#asterisk james4765 (n=james476@office.neteasyinc.com) |
| 15:04.08 | patrick-- | [Feb 18 16:01:23] WARNING[24143] chan_misdn.c: Could not create channel on port:1 with extensions: |
| 15:04.13 | patrick-- | whats that all about? |
| 15:05.39 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 15:06.43 | *** join/#asterisk Docfxit (i=ExUser@netblock-208-127-208-174.dslextreme.com) |
| 15:07.58 | james4765 | I'm having a bit of a problem with my TDM400 after installing Trixbox 2.4 - it's getting a ring signal but no sound is going through |
| 15:09.11 | Qwell | ~trixbox |
| 15:09.16 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 15:09.17 | james4765 | I've got a handset connected in parallel with the line, and there's no problem with the signal |
| 15:09.28 | james4765 | ah |
| 15:09.55 | ruied | I have a voipcheap account for outgoing calls, I would like also to receive the incoming calls from voipcheap also. My problem is: a person that tries to add my pbx's voipcheap account into the friends list, needs to be accepted by the other party (my pbx account). Is there any option so a person that adds my pbx voipcheap account be automattically added into the his/her friends list without the other party confirmation? |
| 15:10.30 | *** part/#asterisk james4765 (n=james476@office.neteasyinc.com) |
| 15:10.41 | ruied | the other party confirmation = my pbx account confirmation... |
| 15:11.23 | *** join/#asterisk ManxPower (n=manxpowe@127.sub-75-201-207.myvzw.com) |
| 15:11.37 | patrick-- | can anyone tell me why my asterisk keeps crashing? |
| 15:11.45 | patrick-- | i cant see anything from the log files |
| 15:12.04 | ManxPower | how often does it crash? |
| 15:13.40 | ManxPower | Well if you don't want help... |
| 15:14.50 | x86 | anyone know of a java applet softphone? preferrably IAX? |
| 15:15.11 | x86 | i've checked out jiaxclient, but it only supports windows and linux... no MacOS X support :( |
| 15:15.23 | MrTelephone | ~Manxpower |
| 15:15.24 | jbot | i heard manxpower is Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. |
| 15:15.26 | x86 | njiax seems to be just a library with no applet front-end |
| 15:15.58 | MrTelephone | ~MrTelephone |
| 15:16.12 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-185-62.dsl.tul2ok.sbcglobal.net) |
| 15:16.24 | jbot | holds MrTelephone to the floor and spanks him with a cat-o-nine-tails, courtesy of royk |
| 15:18.09 | ManxPower | jbot is well trained. |
| 15:19.04 | SteveTotaro | ~stevetotaro |
| 15:19.05 | jbot | you are probably an IRC nub |
| 15:20.26 | patrick-- | ManxPower: sorry... it crashes when theres much telephony going on |
| 15:21.11 | *** join/#asterisk `paul (n=aldee@125.252.68.126) |
| 15:22.32 | `paul | im trying to configure an audiocode mp124 to work with asterisk and i get a "determine_firstline_parts: Bad request protocol asterisk SIP/2.0" warning. what seems to be the prob? pls help... |
| 15:22.36 | *** join/#asterisk riddlebox (n=riddlebo@75-128-170-26.static.stls.mo.charter.com) |
| 15:24.07 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:24.27 | SteveTotaro | riddlebox |
| 15:24.58 | riddlebox | hey |
| 15:26.45 | nebojsajsimic | ok can somone help please with php dial and result |
| 15:27.11 | nebojsajsimic | i get status when line hung-up |
| 15:27.19 | riddlebox | can any Sangoma AFTA101, card handle a pri? I only see something about the 8 port cards and PRI? |
| 15:27.45 | riddlebox | nebojsajsimic,what script? |
| 15:28.02 | nebojsajsimic | i make php script |
| 15:28.03 | ManxPower | patrick--: then you need to read backtrace.txt in the asterisk source code. |
| 15:28.24 | nebojsajsimic | when i do $agi->exec_dial(.........) |
| 15:28.36 | nebojsajsimic | i get result -1 |
| 15:29.01 | nebojsajsimic | $zovi = $agi->exec_dial(SIP,$broj); |
| 15:29.05 | ManxPower | nebojsajsimic: the results are not for AGI, they are for C modules. |
| 15:29.21 | ManxPower | you would need to check the value of DIALSTATUS, etc. |
| 15:29.45 | nebojsajsimic | i try but php stop and wait for call end |
| 15:29.53 | ManxPower | and shouldn't it be Dial(SIP/$broj) |
| 15:30.09 | nebojsajsimic | i get same result |
| 15:30.28 | ManxPower | Yes, your AGI script will stop running until the call ends. That is why it is not a good idea ro execute Dial inside an AGI script. |
| 15:31.03 | nebojsajsimic | is some way to fih this or just to make calls from conf??? |
| 15:31.08 | nebojsajsimic | *fix |
| 15:31.45 | nebojsajsimic | ??? |
| 15:31.46 | ManxPower | nebojsajsimic: I designed my scripts so I did not have to do Dial from AGI. I split my AGI into two parts, the in the dialplan I run the first AGI, then the Dial, then the 2nd AGI |
| 15:32.15 | ManxPower | Obviously everything still stops until the call ends. |
| 15:32.22 | nebojsajsimic | ok i can make on that way i just ask is some way to make in one agi |
| 15:32.36 | x86 | anyone know of a java applet softphone? preferrably IAX? |
| 15:32.50 | nebojsajsimic | because i have more vars to pass fromm first to second agi |
| 15:32.52 | ManxPower | the only other option is to use .call files, but you still can't do the Dial within the AGI |
| 15:33.15 | ManxPower | nebojsajsimic: you can easily set the VARs in the first AGI to be read by the 2nd AGI |
| 15:33.37 | nebojsajsimic | can you show some example?? plz |
| 15:34.05 | ManxPower | nebojsajsimic: you do not know how to set vars in the php-asterisk library? |
| 15:34.53 | nebojsajsimic | i think not :( |
| 15:34.54 | ManxPower | this is in Perl: $AGI->set_variable("VOICEMAIL_GROUP", "$group_list"); |
| 15:34.59 | nebojsajsimic | ok |
| 15:35.30 | ManxPower | the variable VOICEMAIL_GROUP will be set in the dialplan for all other apps/agis that are part of that call |
| 15:35.50 | ManxPower | I can't tell you how to do it in PHP |
| 15:36.40 | nebojsajsimic | ok thx |
| 15:37.44 | SteveTotaro | anyone test http://moziax.mozdev.org/ ? |
| 15:39.14 | defswork | SteveTotaro: no - but I will now - sounds great |
| 15:39.44 | SteveTotaro | i want to test it on a thinclient |
| 15:39.59 | SteveTotaro | that would be awesome if it works well |
| 15:41.21 | defswork | wonder if it will work on an eee pc |
| 15:41.49 | ManxPower | if it worked well everyone would be using it. |
| 15:42.38 | SteveTotaro | maybe they are on the down low or maybe it has not gained attention yet |
| 15:42.48 | SteveTotaro | i just learned of it a week or two ago |
| 15:43.41 | ber___ | skype has a plugin which gives you a click to call on all ph#s in IE |
| 15:44.00 | ber___ | phone calls are pretty cheap so it doesnt bother me its skype versus my own system |
| 15:44.06 | SteveTotaro | but this is iax2 |
| 15:44.18 | SteveTotaro | can you use speex with it? |
| 15:44.37 | ber___ | whats teh benefit of IAX2 or SPEEX over standard skype? |
| 15:44.49 | ber___ | i can see this maybe being useful for dialing stuff which isnt standard nanpa #s |
| 15:44.56 | ber___ | like say you had a large company intranet with extensions |
| 15:44.59 | SteveTotaro | how about in 3rd world countries where port blocking is a reality for VoIP |
| 15:45.00 | ber___ | all web based |
| 15:45.08 | defswork | skype is evil |
| 15:45.26 | ber___ | skype works for me, who knows if they are shunting off my data to the nsa :) |
| 15:45.28 | SteveTotaro | this isn't the skype channel |
| 15:45.51 | ber___ | no all im saying is that what that app tries to do users can already get in some form |
| 15:45.54 | SteveTotaro | ~skype |
| 15:45.55 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 15:46.32 | SteveTotaro | yes, that argument can be made for most things, including asterisk |
| 15:46.35 | SteveTotaro | and skype |
| 15:47.02 | ber___ | well anytime you have a new app users will move to it if it can do something new or significantly better than existing |
| 15:47.26 | ber___ | so one thing the skype piece cant do is things that require fancy dialplans, its good for normal direct dials only |
| 15:47.46 | ber___ | for cheapness free versus $.02 a minute doesnt matter that much to me |
| 15:48.03 | SteveTotaro | oh so i cannot connect to a repeater in afghanistan? |
| 15:48.10 | ber___ | hehheh |
| 15:48.11 | SteveTotaro | vi app_rpt with skype? |
| 15:48.19 | ber___ | broadband2camel |
| 15:48.22 | *** join/#asterisk MrChimpy (n=tony@212.158.8.163) |
| 15:48.54 | defswork | what do you guys use to route calls out over GSM ? |
| 15:49.02 | agx | isn't google talk working exactly like skype? you create an account, install software and it works ? (+ you can integrate with asterisk and others softswitch? ) |
| 15:49.09 | SteveTotaro | ~gsm |
| 15:49.10 | jbot | it has been said that gsm is a codec, operating at approx 13kbps up/down. |
| 15:49.40 | defswork | SteveTotaro: I mean route calls out via GSM operator (SIM dialler) |
| 15:49.40 | agx | not true, GSM has a real bw of 32/38 knps |
| 15:49.45 | SteveTotaro | ~goog411 |
| 15:49.46 | jbot | Google has a free 411 service call 1-800-goog-411 |
| 15:50.14 | SteveTotaro | you can use chan_mobile |
| 15:50.31 | SteveTotaro | turn bluetooth cells into FXOs |
| 15:50.48 | x86 | anyone know of a java applet softphone? preferrably IAX? |
| 15:50.50 | x86 | ;) |
| 15:50.59 | SteveTotaro | jiax |
| 15:51.15 | x86 | only works with windows or linux |
| 15:51.27 | x86 | need something that works on macosx as well |
| 15:51.53 | x86 | hmm, can be flash, doesn't have to be java I guess |
| 15:52.15 | *** join/#asterisk andrew` (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 15:52.22 | SteveTotaro | you can compile it so why can't you compile it to work with mac? |
| 15:52.42 | x86 | because it has os-specific libs? |
| 15:53.03 | SteveTotaro | dunno i am not a mac guy |
| 15:53.07 | x86 | jiaxc_linux_x86.jar and jiaxc_windows_x86.jar |
| 15:53.35 | SteveTotaro | i thought mac was using x86 hardware now, shows how much i know |
| 15:53.38 | *** join/#asterisk jhb (n=joerg@81-5-139-2.dsl.eclipse.net.uk) |
| 15:55.23 | *** part/#asterisk Aurs (n=Ove_Aurs@85.80-203-122.nextgentel.com) |
| 15:58.13 | *** join/#asterisk flujan_ (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 16:02.10 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 16:02.13 | x86 | SteveTotaro: they are? ;) |
| 16:02.20 | x86 | SteveTotaro: but still not a windows box? :) |
| 16:02.46 | SteveTotaro | i saw one running an instance of xp or maybe it was vista |
| 16:07.38 | agx | wich FW do you guys uses with GXP 2000 ? I found comfortable with 1.1.2.27 but the new model comes with 1.1.4.x or 1.1.5.x and there is no way to rollback... |
| 16:08.37 | x86 | SteveTotaro: my mac dual-boots MacOS and Vista |
| 16:08.54 | x86 | SteveTotaro: but the point is, my customers may be running MacOS, and I need to support that |
| 16:09.50 | ber___ | can u tell them to use the windows emulator? |
| 16:10.07 | x86 | you know of a windows emulator that runs on a Mac? |
| 16:10.14 | ber___ | http://www.jackenhack.com/jackeniax/ |
| 16:10.19 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 16:10.36 | *** join/#asterisk Enf0rc3r (n=prophet@enforcer.kicks-ass.org) |
| 16:10.49 | Enf0rc3r | howdy |
| 16:11.08 | x86 | ber___: no, I want a web-based (java or flash) SIP or IAX phone so I can do click-to-call from my website to my asterisk server |
| 16:11.40 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 16:12.53 | SteveTotaro | muxaur or whatever sells what you need |
| 16:13.01 | SteveTotaro | not sure if it works on a mac though |
| 16:13.14 | x86 | it does, but I was looking for free ;) |
| 16:13.35 | SteveTotaro | how many instances do you need? |
| 16:13.37 | ber___ | if they use mac they should be used to paying for things :) |
| 16:13.56 | x86 | ber___: most of the software I use on my Mac is free.... |
| 16:14.10 | x86 | ber___: MacOS X probably has just as much free software as any other OS |
| 16:14.24 | ber___ | i meant the fact th hardware costs 2x as much |
| 16:14.30 | x86 | true |
| 16:14.48 | ber___ | its good, was looking at powerobook or pc laptop, i just didnt want to spend the extra cash |
| 16:14.51 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 16:14.52 | ber___ | many peopel i know have switched to mac |
| 16:14.56 | SteveTotaro | is it just one, your mac? |
| 16:15.05 | SteveTotaro | blech |
| 16:16.41 | ber___ | you can do click2call from a website to asterisk with other options than java or flash |
| 16:16.49 | ber___ | there were some freeware php implementations of it |
| 16:17.09 | x86 | ber___: since powerbooks have been discontinued for some time now, you can probably get one very cheap |
| 16:17.24 | ber___ | whatever their highest end notebook is |
| 16:17.26 | x86 | ber___: php can handle client-side audio? |
| 16:17.29 | ber___ | i dont know the nomenclature |
| 16:17.44 | x86 | last I knew, php was server-side, not client-side |
| 16:17.49 | ber___ | oh you want to do it to voip? i was thinking it could click2call to a phone |
| 16:18.06 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:18.07 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:18.07 | x86 | that's click2callback |
| 16:18.11 | x86 | completely different |
| 16:18.35 | SteveTotaro | just pony up and pay |
| 16:18.48 | ber___ | yeah all the time you invested in talkign about this crap you could have bee making money |
| 16:18.50 | SteveTotaro | instead of asking the same question over and over ;) |
| 16:19.04 | ber___ | but you are right, click2call i never messed with |
| 16:19.09 | ber___ | just that callback app |
| 16:19.22 | x86 | SteveTotaro: I believe I'm discussing potential options, not asking over and over ;) |
| 16:19.35 | ber___ | i def odnt knwo the answer |
| 16:19.42 | ber___ | does google come up with anything good? |
| 16:19.47 | x86 | not really ;) |
| 16:19.50 | ber___ | yuck |
| 16:19.54 | ber___ | if google cant find it |
| 16:20.02 | ber___ | ... |
| 16:20.05 | x86 | I found something called TringMe |
| 16:20.36 | x86 | which is free and flash-based, but when the call is placed into asterisk, and I pick up my phone, no audio is ever exchanged |
| 16:20.48 | ber___ | do you haev source to it? |
| 16:21.06 | ber___ | you can just debug the flash |
| 16:21.35 | ber___ | and run protocol sniffer on yoru side make sure RTP is bridged end2end |
| 16:21.57 | ber___ | no audio is normally an IP reachability thing |
| 16:22.18 | ber___ | but i dont know the implementation library this flash app uses |
| 16:24.00 | x86 | i think it's because I'm going out the pix, and coming back in via 1:1 static NAT |
| 16:24.12 | x86 | so I think I need someone on the outside of my network to try calling in |
| 16:25.50 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:27.03 | SteveTotaro | (10:36:07 AM) x86: anyone know of a java applet softphone? preferrably IAX? |
| 16:27.33 | x86 | SteveTotaro: first couple times I was asking for java, then I was asking for flash... see the difference? :) |
| 16:27.49 | SteveTotaro | (10:17:55 AM) x86: anyone know of a java applet softphone? preferrably IAX? |
| 16:27.57 | x86 | uh huh ;) |
| 16:28.24 | SteveTotaro | (10:00:09 AM) x86: anyone know of a java applet softphone? preferrably IAX? |
| 16:28.50 | x86 | yay! steve can effectively copy+paste! |
| 16:28.50 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 16:28.55 | x86 | hehe |
| 16:29.40 | *** join/#asterisk twisted (n=root@pdpc/supporter/active/twisted) |
| 16:29.40 | *** mode/#asterisk [+o twisted] by ChanServ |
| 16:29.58 | SteveTotaro | ~pb |
| 16:29.58 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:30.04 | ber___ | yeah it doesnt sound like an issue in the app sounds like a network issue |
| 16:30.33 | SteveTotaro | ~cisco |
| 16:30.34 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 16:30.50 | SteveTotaro | they also have some really good networking certs |
| 16:32.32 | SteveTotaro | i have yet to meet a ccie that didn't know networking |
| 16:33.30 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 16:33.35 | *** join/#asterisk geek_cl (n=lletelie@200.75.18.211) |
| 16:35.44 | SteveTotaro | anyone know of a webbased iax or sip phone that can diagnose my PIX and NAT settings for free? |
| 16:36.21 | SteveTotaro | anyone know of a webbased iax or sip phone in Java that can diagnose my PIX and NAT settings for free? |
| 16:36.32 | SteveTotaro | anyone know of a webbased iax or sip phone in flash that can diagnose my PIX and NAT settings for free? |
| 16:36.48 | anonymouz666 | SteveTotaro? |
| 16:36.49 | ber___ | yuou are mean |
| 16:37.06 | SteveTotaro | i am funny! |
| 16:37.29 | SteveTotaro | x86 is a good sport |
| 16:37.40 | ber___ | anyways x86 can you unprotect yoru server from your firewall for your testing |
| 16:37.52 | ber___ | then test and if the app is working ok start fixing th eissue |
| 16:38.17 | ber___ | 1:1 map external IP to server |
| 16:38.25 | x86 | I have 1:1 map |
| 16:38.45 | x86 | I guess I can do access list myacl permit ip any host blah |
| 16:38.54 | ber___ | yeah any type of filtering remove |
| 16:39.08 | ber___ | sometimes people inadvertantly filter some of the RTP possible ports |
| 16:40.30 | x86 | yeah I know |
| 16:40.50 | x86 | and with SIP, it's a PITA because the RTP port range can be 2000+ ports |
| 16:41.03 | x86 | that's why my original question was preferring IAX ;) |
| 16:41.14 | x86 | since it uses a single port for both signalling as well as RTP |
| 16:41.30 | SteveTotaro | try it with a linux or windows box first |
| 16:41.35 | SteveTotaro | then mess around with mac |
| 16:42.17 | x86 | I've got a friend external to my LAN who is testing on windows |
| 16:42.51 | x86 | but since it's flash, all of the hardware is abstracted anyway, and there is no need for os-specific stuff for connecting to audio devices |
| 16:42.58 | x86 | (unlike Java) |
| 16:43.17 | SteveTotaro | anyone know of a webbased iax or sip phone in java or flash that can diagnose my PIX and NAT settings for free? oh and run on a mac.... |
| 16:43.22 | SteveTotaro | lol |
| 16:43.41 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:43.54 | SteveTotaro | i say simplify as much as possible first and then make it more complex if needed |
| 16:44.24 | x86 | it is simple |
| 16:44.29 | x86 | very simple ;) |
| 16:44.36 | x86 | at least with flash |
| 16:44.48 | anonymouz666 | SteveTotaro: Take a look at JIAXCLIENT and please stop repeating yourself. it's annoying. |
| 16:44.57 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 16:45.05 | x86 | hahaha |
| 16:45.18 | x86 | anonymouz666++ |
| 16:45.29 | SteveTotaro | lmao |
| 16:45.45 | SteveTotaro | i am not repeating myself, i changed my question each time' |
| 16:45.54 | SteveTotaro | see the difference? |
| 16:46.24 | file | now now... everyone play nice |
| 16:46.45 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:46.47 | *** join/#asterisk qdk_ (n=qdk@195.242.194.42) |
| 16:47.01 | lirakis | thinking about doing hvm with xen |
| 16:47.47 | *** join/#asterisk admin0 (n=admin@116.90.228.34) |
| 16:48.40 | *** join/#asterisk robeph (n=robf@router.asteriasgi.com) |
| 16:48.42 | *** join/#asterisk seanbright (n=elixer@65.207.74.18) |
| 16:49.37 | *** part/#asterisk robeph (n=robf@router.asteriasgi.com) |
| 16:49.57 | Enf0rc3r | SteveTotaro |
| 16:50.00 | *** join/#asterisk Dovid (n=Dovid@bzq-79-180-16-229.red.bezeqint.net) |
| 16:50.04 | Enf0rc3r | sup homie? |
| 16:50.18 | SteveTotaro | not much here |
| 16:50.23 | Dovid | is there any way of seein if a digium card is physically installed on a box with out installing zaptel |
| 16:50.24 | SteveTotaro | just messin around |
| 16:50.25 | Enf0rc3r | this is my nagios box |
| 16:50.27 | Enf0rc3r | ModelPentium 75 - 200 |
| 16:50.27 | Enf0rc3r | Chip MHz165.96 MHz |
| 16:50.32 | SteveTotaro | nice |
| 16:50.34 | Enf0rc3r | IDE Deviceshda: QUANTUM BIGFOOT_CY2160A (Capacity: 1.97 GB) |
| 16:50.36 | Enf0rc3r | lol |
| 16:50.36 | ber___ | http://www.macshareware.com/review/zoiper_free_iax_and_sip_softphone |
| 16:50.38 | Enf0rc3r | old skewl |
| 16:51.04 | ber___ | dovid, 'lspci' |
| 16:51.08 | ber___ | or check out dmesg |
| 16:51.21 | SteveTotaro | nice ber |
| 16:51.38 | ber___ | lspci is normally what i use if i have a driver issue |
| 16:51.45 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 16:52.22 | apocn | Hello, when an agent is talking to a client (from the queue) and another call comes in, he's softphone starts ringing (even tho he's talking to another client)... |
| 16:52.25 | apocn | how can I prevent this? |
| 16:52.54 | SteveTotaro | apocn, turn off call waiting on the softphone |
| 16:52.57 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-48-245.pskn.east.verizon.net) |
| 16:53.59 | apocn | SteveTotaro, thanks |
| 16:54.11 | SteveTotaro | de nada |
| 16:55.33 | apocn | :] |
| 17:01.07 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:01.22 | fiddur | Has anyone here implemented any kind of "skills based routing" for queues in asterisk? e.g what is described here: http://www.voip-info.org/wiki/index.php?page=PBX+Skill+Based+Routing |
| 17:01.59 | coppice | dumbass avoidance routing |
| 17:02.07 | fiddur | ...the example there is a bit easy though;the point is combined skills... |
| 17:03.10 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:03.22 | SteveTotaro | skills based routing on who is about to make commission, don't route to them anymore |
| 17:03.37 | SteveTotaro | i bet alot of bosses would like that |
| 17:03.55 | fiddur | hehe |
| 17:04.07 | *** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 17:04.20 | SteveTotaro | route to the next best closer who is further from hitting his numbers |
| 17:05.16 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:05.49 | coppice | oh, you mean dumbass seeking routing |
| 17:06.08 | SteveTotaro | no, you want it to go to a good closer |
| 17:06.13 | fiddur | But seriously... a manager is supposed to update everyones skills in e.g. e-mail-support and norwegian language... and a call matching these two criteria should be routed to the one best suited to serve him/her in that language... |
| 17:06.17 | SteveTotaro | but you want to avoid commissions |
| 17:06.39 | coppice | the good closers hit their numbers, so your requirements are in serious conflict |
| 17:06.53 | SteveTotaro | not if you run a tight ship |
| 17:07.02 | fiddur | ...if it isn't done allready, I guess I will add it, and some gui for it... |
| 17:07.06 | SteveTotaro | dangle the carrot on a string |
| 17:07.10 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:07.19 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 17:07.44 | SteveTotaro | fiddur, you could automate that with a customer survey, if they will take the time to answer |
| 17:08.07 | drmessano | Gotta love a job interview |
| 17:08.28 | SteveTotaro | giving or getting interviewed? |
| 17:08.29 | fiddur | SteveTotaro: That would be a call center karma system then? :D |
| 17:08.32 | drmessano | getting |
| 17:09.01 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:09.02 | *** join/#asterisk cjk (n=ldidelot@d90-129-43-59.cust.tele2.lu) |
| 17:09.04 | SteveTotaro | fiddur, why not? it is certainly worth a try |
| 17:09.23 | SteveTotaro | customer feedback is more valuable than some supervisor's opinion |
| 17:09.44 | clyrrad | If using func_odbc.conf to select data, and your select returns multiple rows, how does asterisk receive it? Does it come back as one variable thats comma seperated? Or does it return ARRAY or something similiar that we walk though with while loop? |
| 17:09.49 | SteveTotaro | did the dr get a jobby job? |
| 17:09.52 | fiddur | SteveTotaro: Well, it's worth a look... but that's AFTER I've gotten the routing up in first hand! |
| 17:09.59 | drmessano | Dunno.. |
| 17:10.03 | MrTelephone | where |
| 17:10.09 | MrTelephone | for who |
| 17:10.12 | drmessano | Im pessimistic |
| 17:10.28 | SteveTotaro | that's no good for interviews |
| 17:10.32 | drmessano | Some IT company that does helpdesk and onsite |
| 17:10.48 | SteveTotaro | you have to walk in there knowing you nailed it before you even say word one |
| 17:10.53 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:11.01 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
| 17:11.10 | MrTelephone | nice |
| 17:11.20 | drmessano | WellWell |
| 17:11.22 | SteveTotaro | fiddur, do you have metrics for agents stored somewhere else? |
| 17:11.31 | drmessano | I walked in with a good attitude |
| 17:11.33 | SteveTotaro | some kind of crm |
| 17:11.40 | drmessano | and I showed them I knew a lot more than they expected |
| 17:11.59 | clyrrad | anyone done this? |
| 17:12.00 | drmessano | But we got into a back and forth over enabling the windows zero wireless config wizard |
| 17:12.03 | SteveTotaro | don't show them you know more than expected, doh! |
| 17:12.08 | Datax | Hi all, I have a Cisco 7961G with the SIP41.8-3-1S firmware but I can't get it to register with asterisk |
| 17:12.18 | drmessano | I think it can be done from double clicking the tray icon |
| 17:12.24 | drmessano | and hitting a button |
| 17:12.25 | Datax | the asterisk server keeps saying that the registration failed |
| 17:12.32 | SteveTotaro | services.msc |
| 17:12.45 | drmessano | Thats what he said.. "this is how you need to do it" |
| 17:12.48 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:12.49 | clyrrad | Datax: doule check the username / password |
| 17:12.57 | drmessano | I at least told him how to get to the service in CP |
| 17:13.07 | drmessano | But I dont think I would walk a user through that |
| 17:13.20 | Datax | clyrrad: I agree that that looks to be the problem and I have checked. I'm wondering if I'm not just messing up in the XML file that the cisco phone downloads |
| 17:13.22 | SteveTotaro | start run cmd services.msc |
| 17:13.32 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 17:14.00 | drmessano | net start wzcsvc |
| 17:14.02 | clyrrad | Datax: like I said double check the username and password as well as the registration context |
| 17:14.20 | Datax | ok |
| 17:14.40 | clyrrad | do any of you know the answer to my Database question? |
| 17:14.45 | mort_gib | drmessano: Only loosers ask for stupid stuff like that |
| 17:14.53 | SteveTotaro | and losers too |
| 17:14.54 | fiddur | SteveTotaro: no, not yet.. I can put it where I want it; it'll start out with only 18 people, and I think there'll be 4 skill areas plus 3 languages.. for starters. |
| 17:15.04 | mort_gib | :-) Yes, very right |
| 17:15.38 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 17:15.39 | SteveTotaro | are you going to use agent penalty for skills based? |
| 17:15.46 | SteveTotaro | that's what i did |
| 17:16.22 | fiddur | SteveTotaro: What does that mean? |
| 17:16.49 | SteveTotaro | you can give an agent a penalty from 1 to 100 |
| 17:16.57 | SteveTotaro | i believe 1 is the default |
| 17:17.17 | SteveTotaro | so you just use the AMI to update their penalties on the fly |
| 17:17.38 | SteveTotaro | you may have to patch AMI for that, i am not sure, it has been a while |
| 17:19.11 | fiddur | Hmm... you mean using penalty to make the normal queue-operations to prioritize the better skilled? |
| 17:19.47 | SteveTotaro | yes, along with the penalty you create creative use of queues and cascading queues |
| 17:19.48 | lirakis | SteveTotaro: why dont you just use a different queue? |
| 17:20.05 | SteveTotaro | how many different queues are we talking about |
| 17:20.09 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 17:20.14 | SteveTotaro | let's say you have a spanish queue |
| 17:20.23 | SteveTotaro | with 100 agents |
| 17:20.32 | lirakis | SteveTotaro: lin queue mac queue win queue ... then send to the properly trained group |
| 17:20.55 | lirakis | SteveTotaro:... or have agents be members of more than 1 queue |
| 17:20.59 | SteveTotaro | and in that group how many members |
| 17:21.09 | lirakis | SteveTotaro: i dont know.. thats up to you |
| 17:21.10 | lirakis | lol |
| 17:21.15 | SteveTotaro | how do you rank the members? |
| 17:21.21 | SteveTotaro | with agent penalty |
| 17:21.31 | lirakis | SteveTotaro: im talking about NOT using penalties |
| 17:21.45 | SteveTotaro | it all comes down to agent penalty with the fine grain |
| 17:22.06 | fiddur | SteveTotaro: I have thought of auto-creating queues from the skills-db... There would be at least 12 main queues with a prio set by penalties then... |
| 17:22.19 | lirakis | SteveTotaro: depending on your operations... i mean if you offer a product for 3 os's .. then you have 3 queues |
| 17:22.29 | lirakis | SteveTotaro: thats a simple operation... |
| 17:22.35 | SteveTotaro | ewww queueprio killed my live call center |
| 17:22.40 | lirakis | SteveTotaro: but getting into penalties can be sticky business |
| 17:23.10 | fiddur | But is it so hard to hack in another queue-operations mode so that it's worth creating queues from outside? |
| 17:23.21 | *** join/#asterisk Docfxit (i=ExUser@netblock-208-127-208-174.dslextreme.com) |
| 17:23.32 | SteveTotaro | not really, penalties are easy |
| 17:23.49 | fiddur | It would be very easy to make a program that can see who is most suited of all available agents right now.. |
| 17:23.51 | SteveTotaro | easy to move people around |
| 17:24.18 | SteveTotaro | do what you want, i am just telling you what i know works well |
| 17:24.19 | lirakis | SteveTotaro: i guess my point is... it is probably best to first analyze your operation, and design your queues accordingly, instead of just slapping penalties all over to "control" a generic queue |
| 17:24.58 | SteveTotaro | nobody said slapping penalties all over a generic queue except you |
| 17:25.22 | SteveTotaro | (12:05:00 PM) irc: yes, along with the penalty you create creative use of queues and cascading queues |
| 17:25.35 | lirakis | SteveTotaro: guess i got to the conv. late. |
| 17:25.54 | lirakis | SteveTotaro: 20-25 min late |
| 17:25.59 | lirakis | ;) |
| 17:26.14 | fiddur | No not one generic queue; on queue per relevant combination of skills... |
| 17:26.37 | *** join/#asterisk Docfxit (i=ExUser@netblock-208-127-208-174.dslextreme.com) |
| 17:27.20 | SteveTotaro | and my suggestion is then you can get to finer grain by using penalties |
| 17:28.55 | *** part/#asterisk geminidomino (n=ciro@65.41.157.192) |
| 17:30.00 | fiddur | SteveTotaro: yes, that's a good solution that I guess will work well for the need. I sitll consider writing a new queue-handling-method though; I have to place the code that generates those lists somewhere, and it could as well be done from the asterisk queue and then get the agents skills in the same gui that sets up the agents themselves. |
| 17:30.23 | fiddur | but I thank you for your suggestions! |
| 17:30.38 | SteveTotaro | no problem |
| 17:30.47 | SteveTotaro | you are talking about a 1 megapixel camera |
| 17:31.09 | SteveTotaro | i am looking at an 8 megapixel that you can blow up to super fine granularity |
| 17:32.11 | SteveTotaro | something you could not get unless each agent in a group of one hundred agents unless they all had their own queue |
| 17:35.07 | *** join/#asterisk DarWin_vcch (i=red@207.191.183.43) |
| 17:36.34 | fiddur | SteveTotaro: I think the result will be the same... if the queue-method considers the skills the same way a queue-constructor with penalty would, the result is the same... it's just the gui and the implementation that differs; the same agent would be rung in both cases |
| 17:38.05 | fiddur | Mind you I am quite new to asterisk, and don't know how the queue methods are coded... but I'm not new to coding or messing around with open sources :) |
| 17:48.24 | jks | anyone knows how to force a channel "on hold" from asterisk, instead of from the SIP phone? |
| 17:50.07 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 17:51.33 | ManxPower | jks: you might be able to via the Manager Interface |
| 17:54.04 | *** join/#asterisk mmmToop (n=michaelt@dsl-243-217-82.telkomadsl.co.za) |
| 17:55.34 | *** join/#asterisk seanbright (n=chatzill@m4c5f36d0.tmodns.net) |
| 17:55.57 | jks | ManxPower, that's what I'm trying to do - but I'm not sure how :-| |
| 17:56.41 | jks | ManxPower, I don't see any manager actions that indicates a hold possibility... but it might be something that must be done through a combination of things or similar |
| 17:56.58 | *** join/#asterisk cli4me (n=shizm@cpe-071-070-229-009.nc.res.rr.com) |
| 17:58.44 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 18:03.50 | ManxPower | jks: If you can't do it via manager, then you can't do it. |
| 18:04.13 | ManxPower | why do you want to place a call on hold not using the phone and what are you trying to accomplish? |
| 18:04.20 | drmessano | hacking |
| 18:05.50 | jks | ManxPower, hmm, well, I don't know if it is possible from the manager - I'm just not able to figure out how to do it |
| 18:05.58 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 18:06.13 | jks | ManxPower, the reason for putting the call on hold not using the phone is that I have a SIP headset |
| 18:06.24 | clyrrad | Can anyone tell me how to handle multiple ROWS with REALTIME? IE... How do you get the next row from the result set? Currently I have it working but it only every returns the first hit, how do you itterate the whole result set? |
| 18:06.30 | jks | ManxPower, so there are no keys or buttons for placing the call on hold on the phone itself |
| 18:06.43 | ManxPower | then your softphone sucks |
| 18:06.53 | jks | ManxPower, it's not a softphone. |
| 18:07.03 | ManxPower | then what is it? |
| 18:07.12 | ManxPower | Does it require a PC to work? |
| 18:07.13 | jks | ManxPower, a SIP phone (hardware) |
| 18:07.17 | jks | nope, doesn't require a PC to work |
| 18:07.28 | ManxPower | Thenyour SIP phone sucks. Every single phone I've seen has a hold button |
| 18:07.33 | jks | ManxPower, I've written an app that gives me access to the "missing" features such as DTMF dialing |
| 18:07.39 | jks | ManxPower, it's not a "phone" phone |
| 18:07.43 | jks | ManxPower, it's a wireless headset |
| 18:07.44 | Qwell | your phone can't dial? |
| 18:07.57 | clyrrad | all the examples I find on google and voip-info demonstrate returning 1 row, but what if there are more than one row? If anyone can help I would apprecate it :) :) |
| 18:08.07 | jks | ManxPower, so it has button for taking the call, volume up and down... that sort of thing.... no hold button |
| 18:08.14 | ManxPower | jks: Well you need some form of phone to access Asterisk. Either a softphone or a hardphone. |
| 18:08.22 | jks | ManxPower, it is a hard phone |
| 18:08.25 | cli4me | is there another config, similar to callprogress= that will allow me to try and pass call status to the PBX behind my * box? |
| 18:08.32 | jks | ManxPower, it's just a desk phone with a gazillion buttons |
| 18:08.48 | ManxPower | jks: One might think that if you had bought a decent phone in the first place, you would not be wasting your time. |
| 18:08.54 | jks | ManxPower, it's a small thing you wear on a ear-clip... there's a limitation to how many buttons will fit there ;-) |
| 18:09.04 | jks | ManxPower, I think you're completely missing the point |
| 18:09.23 | Qwell | BT headset > SIP phone > Asterisk |
| 18:09.23 | ManxPower | jks: No, you are missing the point. The CLIENT is supposed to be the device that puts a call on hold. |
| 18:09.24 | jks | ManxPower, I'm really fond of this phone... I'm not wasting my time... I just want to implement a hold feature |
| 18:09.42 | jks | ManxPower, yes, but in this case the client cannot do it, so I want to initiate it from the server |
| 18:09.50 | ManxPower | jks: Then I guess you had better start hacking chan_sip.c then. |
| 18:09.59 | ManxPower | or res_manager.c |
| 18:10.01 | jks | ManxPower, Okay, I might have to resort to that then |
| 18:10.47 | jks | Qwell, I did that before, but I like the fact that I don't have a phone occupying space on my desk... (bit weird, I know) |
| 18:11.02 | clyrrad | can anyone help me out with my REALTIME question? |
| 18:11.13 | Qwell | clyrrad: use func_odbc |
| 18:11.33 | clyrrad | Qwell: how do I handle itterating multiple rows though? |
| 18:12.19 | Qwell | func_odbc in 1.6 can do multirow |
| 18:12.34 | clyrrad | Qwell: how do I do it with 1.4? |
| 18:12.36 | ManxPower | Qwell: so the answer is "you can't do it in any released version of Asterisk"? |
| 18:12.40 | Qwell | Corydon76-dig: is there a 1.4 backport for that? |
| 18:13.37 | clyrrad | REALTIME works fine, but only seems to get the first row :P |
| 18:15.57 | clyrrad | I am trying to make a realtime DB table where extensions can opt in and opt out of the paging group, but for it to work need to be able to handle mulitiple rows being returned.... if current versions dont support multi-rows, have you guys done anyting similar? If so how did you work around this limiation? |
| 18:16.33 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 18:20.40 | *** join/#asterisk Strom_C (n=strom@ip68-104-88-203.lv.lv.cox.net) |
| 18:24.18 | *** join/#asterisk nirz (n=nir@192.115.113.28) |
| 18:24.43 | clyrrad | where'd everyone go? LOL |
| 18:33.38 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 18:34.11 | *** join/#asterisk Tuari (n=Tuari@cpe-76-183-79-199.tx.res.rr.com) |
| 18:34.48 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 18:34.58 | *** join/#asterisk PseudoNim (n=pseudo@modemcable131.6-57-74.mc.videotron.ca) |
| 18:35.06 | PseudoNim | hey all |
| 18:35.47 | PseudoNim | i'm trying to set up a callback/dialout gateway... but i can't quite wrap my head around the Authenticate() command. is it possible to have it so that if i enter one password it goes to one context (in which it asks me for a callback #) and if i enter another password it would give me DISA? |
| 18:36.27 | *** join/#asterisk guillote_GNU (n=guillote@host157.201-253-55.telecom.net.ar) |
| 18:37.21 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-48-245.pskn.east.verizon.net) |
| 18:54.16 | *** join/#asterisk ManxPower (n=manxpowe@127.sub-75-201-207.myvzw.com) |
| 18:54.18 | *** join/#asterisk Docfxit (i=ExUser@netblock-208-127-208-174.dslextreme.com) |
| 18:54.53 | Docfxit | Where can I find instructions on how to setup a message box? |
| 18:55.13 | ManxPower | Docfxit: You mean voicemail? |
| 18:55.30 | Docfxit | yes |
| 18:55.47 | ManxPower | voicemail.conf |
| 18:55.52 | ManxPower | you should know this already |
| 18:56.17 | Docfxit | I'm looking for a manual to record a greeting. |
| 18:56.34 | Docfxit | How to record a greeting. |
| 18:56.49 | JerJer | login to the voicemail main app and listen to the prompts |
| 18:56.57 | ManxPower | log into voicemail, option 0, then follow the options |
| 18:57.21 | ManxPower | But your question was not how to record a greeting, but was how do you set up a voicemailbox. |
| 18:57.47 | Docfxit | When I listen to the prompt she doesn't say how to end the greeting. |
| 18:58.15 | Docfxit | Hanging up the phone wipes out the greeting you just recorded. |
| 18:58.28 | ManxPower | you press #, IIRC |
| 18:58.40 | Docfxit | Great. Thanks. |
| 18:58.43 | ManxPower | and I believe it does tell you that |
| 18:59.47 | Docfxit | I didn't find that. Thanks. |
| 19:09.06 | *** join/#asterisk jdg (n=jdg@203.185.183.44) |
| 19:10.24 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 19:11.24 | Dovid | hi. |
| 19:12.39 | Dovid | I am using n+101 if my primary route does not work. If my primary route does not respond (at all) it takes about 30 seconds before asterisk rolls over and tries the n+101 extension. When running a sip trace asterisk seems to send the invites over and over. Is it possible to loer this time ? (Yes I alredy checked the wiki and I didn't find anything there). |
| 19:14.15 | ManxPower | Dovid: Don't use +101 |
| 19:14.41 | ManxPower | That feature may have been removed from 1.4 already. If not, it should be removed in 1.6 |
| 19:15.05 | ManxPower | Dovid: Asterisk will keep trying. |
| 19:15.27 | ManxPower | Use qualify= or a low registration length |
| 19:15.27 | Dovid | ManxPower: Anyway I can set it to stop after X amount of seconds ? |
| 19:15.43 | ManxPower | Dovid: I don't believe you can. |
| 19:15.55 | Dovid | time to start a bounty |
| 19:15.57 | *** join/#asterisk SteveTotaro (n=root@c-71-206-46-139.hsd1.md.comcast.net) |
| 19:16.03 | ManxPower | there might be some settings in sip.conf.sample, but I assume you looked at that before taking up our time. |
| 19:16.13 | Dovid | yes i did Manx |
| 19:16.13 | shido6 | anyone know where I can get something similar to this : http://www.smartdesks.com/monitor-lift-popup-motorized-monitor-arm.asp |
| 19:16.37 | shido6 | im building a training facility for asterisk |
| 19:17.02 | drmessano | Dude |
| 19:17.11 | drmessano | I got a perfect assignment |
| 19:17.27 | SteveTotaro | fix my pc |
| 19:17.30 | drmessano | 1. Install Linux |
| 19:17.35 | drmessano | 2. Install Asterisk |
| 19:17.39 | drmessano | 3. ??????? |
| 19:17.42 | drmessano | 4. Callcenter |
| 19:17.48 | drmessano | Worth 100 points |
| 19:18.02 | drmessano | Est time: 1 Hour |
| 19:18.15 | seanbright | 1. Steal Underpants |
| 19:18.16 | seanbright | 2. ??? |
| 19:18.19 | seanbright | 3. Profit! |
| 19:18.29 | drmessano | Get cracka-lackin.. I want some queues, bitches |
| 19:18.57 | SteveTotaro | "finance" computers for a $2000 profit |
| 19:19.06 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 19:19.45 | drmessano | Better yet |
| 19:20.00 | drmessano | Do it the way americans schools do |
| 19:20.21 | drmessano | Make them type in all the source code and then compile it.. They should know Asterisk inside and out by then |
| 19:20.25 | drmessano | ..right? |
| 19:20.40 | drmessano | Oh wait, that doesn't work |
| 19:20.50 | *** join/#asterisk simbol76ss (n=simbol@87.10.235.12) |
| 19:21.11 | *** part/#asterisk mmmToop (n=michaelt@dsl-243-217-82.telkomadsl.co.za) |
| 19:23.27 | simbol76ss | Hi chat!!! |
| 19:23.43 | simbol76ss | AsteriskB ootCamp is a good Course??? |
| 19:24.30 | Dovid | ManxPower; The qualify seemed to do it |
| 19:24.46 | Dovid | it "poked" the primary IP which failed of course and rolled right over |
| 19:27.03 | ManxPower | Expect that device to randomly be unreachable. sip qualify will take the device offline if even one packet is missed. |
| 19:28.21 | *** join/#asterisk redax (i=redax@r6.hu) |
| 19:28.23 | redax | hi, |
| 19:29.47 | SteveTotaro | Asterisk boot camp is a paper mill!!! |
| 19:30.25 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 19:30.29 | Katty | hai. |
| 19:30.47 | SteveTotaro | i think it retries six or eight times before becoming unreachable |
| 19:31.56 | redax | anybody using mISDN here? |
| 19:32.46 | *** join/#asterisk jbigbee (n=jbigbee@216.207.245.1) |
| 19:32.55 | ManxPower | SteveTotaro: IAX2 qualify has smoothing. SIP does not, as of 1.4, IIRC |
| 19:33.55 | Katty | hewwoes? :< |
| 19:34.33 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 19:36.08 | _ShrikE | hello |
| 19:36.36 | Katty | so unusually quiet. |
| 19:37.22 | defsdoor | you can help solve my problem if you want :) |
| 19:37.26 | simbol76ss | . |
| 19:38.09 | defsdoor | I'm currently manually clearing calls whenever it surfaces |
| 19:38.09 | Katty | i has wanpipe problems. |
| 19:38.09 | Katty | more like, i'm the problem. |
| 19:38.20 | defsdoor | I have asterisk still spewing call streams at phones when the phone doesn't think it has a call problems |
| 19:38.34 | Katty | mad. |
| 19:39.01 | tclark | so i am clearing out some t1 gear any in a real compant need a smokin deal on a sangoma 104 w/hw echo canlel pm if you want |
| 19:39.07 | Katty | i don't think i've ever had that problem with asterisk before. |
| 19:39.07 | defsdoor | I'm clinging on to a hope that is related to one line not hanging up |
| 19:39.27 | defsdoor | Katty: me neither - on 2 other similar installs |
| 19:39.29 | Katty | we have hangup problems sometimes.. |
| 19:39.36 | Katty | but they were on ole analog tdm cards |
| 19:39.39 | defsdoor | tclark: what sort of price ? |
| 19:39.45 | tclark | pm me |
| 19:40.05 | Katty | where's mister fender? |
| 19:40.14 | _ShrikE | hes been mia lately |
| 19:40.33 | *** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 19:41.10 | lirakis | Katty: not working today |
| 19:43.01 | Katty | oh :< |
| 19:43.03 | Katty | is he ill? |
| 19:43.21 | anonymouz666 | Katty is back! |
| 19:43.23 | lirakis | Katty: dont think so... it is a holidy here |
| 19:43.27 | lirakis | (shrug) |
| 19:43.57 | SteveTotaro | and then it tries again a minute later which can be changed in the source code |
| 19:56.28 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 19:57.28 | grandpapadot | On Aastra 480i CT phones, anyone having issues using a hostname in the config file for the proxy addresses? A hostname in TFTP seems to work, but the phone seems to flake out *sometimes* using anything but an IP address in the config files. At first, we thought it was a DNS issue, but it's definitely with the phone. Latest firmware. |
| 19:58.19 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 20:08.42 | *** join/#asterisk [hC] (n=hardcore@216.251.157.146) |
| 20:12.46 | *** join/#asterisk juanant (n=chatzill@190.156.245.114) |
| 20:12.54 | juanant | hi people |
| 20:12.59 | juanant | i need help |
| 20:13.09 | juanant | http://pastebin.com/m33d75f58 |
| 20:13.40 | juanant | it playback with full path but not with only the name |
| 20:13.50 | juanant | can anybody helpmy?? |
| 20:14.13 | juanant | seanbright ARE YOU HERE???? |
| 20:14.23 | juanant | hi??? |
| 20:14.29 | seanbright | juanant: no, i'm not here. |
| 20:14.35 | juanant | jajaja |
| 20:14.54 | juanant | you whent the last day |
| 20:15.00 | juanant | i am here again |
| 20:15.22 | seanbright | juanant: you don't need the /var/lib/asterisk/sounds/ part |
| 20:15.34 | seanbright | err, nevermind |
| 20:15.41 | ManxPower | juanant: looks like you are using a pre-built package. |
| 20:15.54 | juanant | yep i am using fedoras package |
| 20:16.08 | juanant | sorry, fedora packages |
| 20:16.15 | seanbright | juanant: run this at a command prompt: ls -al /var/lib/asterisk/sounds/demo-congrats.gsm |
| 20:16.17 | ManxPower | juanant: then you will have to ask the fedora packager where they set the default location for Asterisk sounds |
| 20:16.28 | juanant | ManxPower what i should do???? |
| 20:16.34 | ManxPower | They must have changed the defaults from /var/lib/asterisk/sounds |
| 20:16.51 | ManxPower | juanant: either remove the package and install from source, or contact the package maintaner for support. |
| 20:16.54 | juanant | mmm interesting.... |
| 20:17.29 | mvanbaak | or look in /etc/asterisk.conf |
| 20:17.42 | seanbright | or maybe he doesn't have the GSM sound files installed... |
| 20:18.07 | mvanbaak | astvarlibdir |
| 20:18.35 | juanant | [directories] |
| 20:18.37 | juanant | astetcdir => /etc/asterisk |
| 20:18.38 | juanant | astmoddir => /usr/lib/asterisk/modules |
| 20:18.40 | juanant | astvarlibdir => /var/lib/asterisk |
| 20:18.43 | mvanbaak | ok |
| 20:18.46 | mvanbaak | that one is correct |
| 20:18.51 | seanbright | juanant: run this at a command prompt: ls -al /var/lib/asterisk/sounds/demo-congrats.gsm |
| 20:19.07 | juanant | i installed wav and gsm |
| 20:19.13 | seanbright | awesome... |
| 20:19.15 | seanbright | juanant: run this at a command prompt: ls -al /var/lib/asterisk/sounds/demo-congrats.gsm |
| 20:19.19 | *** join/#asterisk CrashSys (n=kumba@216-199-37-76.tpa.fdn.com) |
| 20:19.26 | mvanbaak | lol |
| 20:19.35 | mvanbaak | how long are you going to repeat that seanbright ;) |
| 20:19.44 | juanant | whait i will run it.... |
| 20:19.45 | seanbright | until he does it? |
| 20:19.47 | seanbright | :) |
| 20:20.12 | seanbright | i prefer the eliminate-the-obvious form of problem solving |
| 20:20.20 | seanbright | rather than the random stab in the dark form |
| 20:20.20 | ManxPower | seanbright: I usually give them three chances, then stop helping them. |
| 20:20.50 | CrashSys | I am running Asterisk v.2.3 and want to know how to make the IVR menu change when they press 3 |
| 20:20.54 | mvanbaak | meh: http://www.xkcd.com |
| 20:21.00 | juanant | -rw-r--r-- 1 asterisk asterisk 64746 nov 20 17:26 /var/lib/asterisk/sounds/demo-congrats.gsm |
| 20:21.03 | juanant | this is |
| 20:21.06 | CrashSys | It's also not connecting to vonage |
| 20:21.14 | CrashSys | and it's virtualized |
| 20:21.41 | seanbright | juanant: now this: ps -ef | grep asterisk |
| 20:21.41 | mvanbaak | CrashSys: maybe that's because asterisk 2.3 is not out yet. We are in the process of getting 1.6 out |
| 20:21.52 | seanbright | juanant: and paste the output into a pastebin |
| 20:21.53 | seanbright | ~pb |
| 20:21.54 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:21.56 | *** join/#asterisk ph0ne (n=ph0ne@64.56.225.123) |
| 20:22.57 | ManxPower | CrashSys: you are a moron. There is no such thing as Asteriwk 2.3 |
| 20:23.01 | ManxPower | or Asterisk 2.3 |
| 20:23.02 | *** join/#asterisk tristanbob (n=tristanr@oalug/member/tristanbob) |
| 20:23.37 | seanbright | ManxPower: that's kinda harsh, no? |
| 20:23.46 | CrashSys | Yes, I know... but people do e-mail me about Asterisk v.2.3 (trixbox) |
| 20:23.48 | ManxPower | seanbright: not if you'd talked to him before. |
| 20:23.54 | juanant | asterisk 20823 1 0 14:42 ? 00:00:02 /usr/sbin/asterisk -U asterisk -G asterisk -C /etc/asterisk/asterisk.conf |
| 20:23.55 | juanant | root 21528 18627 0 15:21 pts/1 00:00:00 grep asterisk |
| 20:23.56 | seanbright | ManxPower: good point |
| 20:24.14 | mvanbaak | ~trixbox |
| 20:24.15 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 20:24.21 | ManxPower | CrashSys: There is no such thing as Asterisk 2.3. Therefore you should be asking on some other channel. |
| 20:24.33 | CrashSys | Manx: Twas a joke mang... |
| 20:24.43 | juanant | thats all |
| 20:24.57 | ManxPower | Ah, so you are just trying to waste all our time. |
| 20:24.59 | seanbright | juanant: hrmmm |
| 20:25.04 | CrashSys | Virtualized trixbox connecting to vonage sounds kind of funny to me... |
| 20:25.06 | juanant | what??? |
| 20:25.08 | CrashSys | Manx: Pretty much |
| 20:25.38 | ManxPower | <PROTECTED> |
| 20:25.44 | mvanbaak | what's funny about virtualized asterisk ? |
| 20:25.53 | juanant | i also installed asterisk-java and its working well but with the full path too |
| 20:26.05 | CrashSys | :( |
| 20:26.06 | seanbright | juanant: does it work when you specify the absolute path? |
| 20:26.11 | juanant | YES |
| 20:26.18 | seanbright | juanant: oh |
| 20:26.26 | seanbright | juanant: problem solved then. use the absolute path. |
| 20:26.31 | juanant | jajajajaja |
| 20:26.46 | juanant | but i need to use sayAlpha |
| 20:26.49 | juanant | and sayDigits too |
| 20:27.01 | ManxPower | juanant: Have you confirmed that SayDigits and Sayalpha does not work? |
| 20:27.10 | juanant | taht functions cant use absolute path |
| 20:27.13 | juanant | yes |
| 20:27.16 | juanant | the doesnt work |
| 20:27.33 | ManxPower | juanant: then I guess YOU SHOULD TALK TO THE DAMN PACKAGE BUILDER OR UNINSTALL ASTERISK AND NISTALL FROM SOURCE. |
| 20:27.37 | ManxPower | this is not racket science. |
| 20:27.38 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-170-218.dsl.tul2ok.sbcglobal.net) |
| 20:28.08 | seanbright | juanant: or rocket science either. |
| 20:28.59 | juanant | maxpower homer rocket science??? |
| 20:29.22 | *** join/#asterisk thomas_newbie__ (n=thomas@CPE0014bf493235-CM00140493ede8.cpe.net.cable.rogers.com) |
| 20:29.27 | juanant | ok tanks a lot maxpower |
| 20:29.36 | seanbright | juanant: that is probably a humrous pop culture reference in south america |
| 20:30.16 | juanant | ore you kidding? |
| 20:30.29 | seanbright | jajajajajajaja |
| 20:31.00 | Nugget | You don't have to be a rocket surgeon to install asterisk. |
| 20:31.26 | juanant | oohh! |
| 20:31.38 | juanant | i am only 15 sorry |
| 20:32.00 | CrashSys | What version of Asterisk are you using? |
| 20:32.12 | juanant | ok i will try using gcc and unistalling the rpms |
| 20:32.21 | mvanbaak | CrashSys: must be v 2.3 |
| 20:32.23 | juanant | thanks bye sean bye maxpower |
| 20:32.30 | seanbright | juanant: adios |
| 20:37.02 | *** part/#asterisk redax (i=redax@r6.hu) |
| 20:37.20 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 20:46.42 | *** part/#asterisk simbol76ss (n=simbol@87.10.235.12) |
| 20:50.25 | *** join/#asterisk kjs (n=kjs@mx1.vm.bytemark.co.uk) |
| 20:55.08 | x86 | seanbright: obviously! |
| 20:55.15 | x86 | <-- chingon |
| 20:55.36 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:55.50 | seanbright | x86: that you are... that you are |
| 20:55.52 | nny_1 | woohoo |
| 20:56.05 | nny_1 | gotta 7940 en route from ebay for testing |
| 20:56.19 | x86 | nny_1: sorry to hear that |
| 20:56.30 | nny_1 | hehe |
| 20:56.32 | x86 | Cisco phones suck |
| 20:56.35 | x86 | all of them |
| 20:56.41 | nny_1 | yeah i have been reading |
| 20:56.44 | x86 | Polycom phones are really what you want |
| 20:56.47 | nny_1 | indeed |
| 20:56.55 | nny_1 | that's what I have now and sell |
| 20:57.47 | nny_1 | but i bought a cisco to have a first hand experience with it.. We have "cisco" advertised, I don't recommend them, but people are inclined to recognize the name, sadly |
| 20:58.02 | anonymouz666 | Asterisk just support sending SIP MESSAGE method while a dialog is established. Does that make sense for you? :-) I couldn't resist. |
| 20:58.22 | nny_1 | ~cisco |
| 20:58.22 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 20:58.28 | nny_1 | lol |
| 20:58.30 | nny_1 | nice |
| 20:59.37 | nny_1 | I am sure I will have plenty to bitch about come next week |
| 21:00.09 | nny_1 | till then, I am off to buy an yugo and get a high maintenance replacement GF for the "experience" :P later |
| 21:00.31 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 21:10.08 | *** join/#asterisk nitrus^ (n=nitrus@cpe-76-166-248-27.socal.res.rr.com) |
| 21:10.36 | nitrus^ | has anyone here had a problem where they experience a loud screetching noise in the middle of a conversation if the conversation is over a certain length? |
| 21:10.51 | *** join/#asterisk glwgoes (n=guilherm@201.67.242.157) |
| 21:11.41 | seanbright | only everytime i call my inlaws! |
| 21:11.42 | seanbright | ba dum dum |
| 21:11.48 | nitrus^ | haha |
| 21:12.11 | *** join/#asterisk iamhrh (n=iamhrh@office.amsvans.com) |
| 21:13.04 | iamhrh | is anyone here familiar with the cisco 7961? specifically, how to set up things in the skinny.conf file to get the thing working properly. I'm able to place and recieve calls with is, but can't get hinting and speed dials to work |
| 21:13.33 | Qwell | iamhrh: 1.4? |
| 21:13.38 | iamhrh | yeah |
| 21:13.40 | mvanbaak | iamhrh: 1.4 or 1.6 ? |
| 21:13.50 | iamhrh | 1.4 |
| 21:13.56 | Qwell | hinting is a speeddial line, like speeddial => 1234,Bob,exten@context |
| 21:13.58 | Qwell | or something |
| 21:14.08 | Qwell | erm, no |
| 21:14.12 | Qwell | exten@context,Name |
| 21:14.25 | x86 | what does hinting do? |
| 21:14.31 | iamhrh | so in extensions.conf, i should have: |
| 21:14.31 | Qwell | hints... |
| 21:14.40 | iamhrh | exten => 1000,hint,SIP/1000 |
| 21:14.42 | x86 | nice description... almost ;) |
| 21:14.50 | Qwell | exten => 6006,hint,Skinny/6006@7961 |
| 21:14.56 | Qwell | That's what I have to hint my skinny phone |
| 21:15.07 | Qwell | then in skinny.conf, for another phone to monitor that one, I have |
| 21:15.07 | x86 | what does hinting do? |
| 21:15.17 | iamhrh | atm I'm tring to show the status of some sip devices on the skinny device |
| 21:15.18 | Qwell | speeddial => 6006@hints,qwell |
| 21:15.26 | mvanbaak | yup |
| 21:15.30 | mvanbaak | that's how it works |
| 21:15.38 | mvanbaak | speeddial => 6002@hints,Livingroom |
| 21:15.58 | mvanbaak | and in extensions.conf: |
| 21:16.14 | mvanbaak | [hints] |
| 21:16.16 | mvanbaak | exten => 6002,hint,Skinny/6002@livingroom |
| 21:16.53 | iamhrh | is there a better way to reload the skinny config than restarting asterisk (i'm sure there is, but didn't see it listed in the under the help from the cli) |
| 21:16.59 | *** join/#asterisk angom (n=angom@200.79.141.128.dsl.dyn.telnor.net) |
| 21:17.13 | Qwell | I don't remember whether we fixed reloads for chan_skinny or not |
| 21:17.24 | iamhrh | heh, yeah i don't think so |
| 21:17.34 | iamhrh | cause even issuing a "reload" doesn't do it |
| 21:17.40 | iamhrh | have to stop gracefully |
| 21:17.41 | iamhrh | and restart |
| 21:17.46 | x86 | Qwell: are hints just so you can subscribe and do stuff like SLA and so forth? |
| 21:17.54 | iamhrh | and skinny reload diesn't do anything |
| 21:17.59 | mvanbaak | Qwell: no |
| 21:18.00 | iamhrh | its for like busy lamps and whatnot |
| 21:18.06 | mvanbaak | you cannot reload skinny |
| 21:18.11 | mvanbaak | you have to unload and load it |
| 21:18.15 | mvanbaak | even in trunk |
| 21:18.15 | x86 | iamhrh: yeah, SLA ;) |
| 21:18.32 | x86 | iamhrh: shared line appearances |
| 21:18.41 | iamhrh | mvabaak: what commands would i use to do that? |
| 21:18.44 | iamhrh | x8x: thanks |
| 21:19.20 | mvanbaak | module unload chan_skinny.so |
| 21:19.20 | iamhrh | module unload skinny? |
| 21:19.26 | iamhrh | ah ok thanks |
| 21:19.26 | mvanbaak | module load chan_skinny.so |
| 21:19.29 | *** join/#asterisk angryuser (n=nononon@df01t2-212-194-39-102.d4.club-internet.fr) |
| 21:19.41 | mvanbaak | x86: hints is for busy lamps |
| 21:20.01 | Qwell | busy lamps != "SLA" |
| 21:20.22 | *** join/#asterisk BBHoss (n=hoss@c-71-207-173-38.hsd1.al.comcast.net) |
| 21:21.03 | x86 | Qwell: what's the difference? |
| 21:21.14 | ManxPower | x86: you can make calls on SLA lines |
| 21:21.21 | x86 | Qwell: isn't SLA just a busy lamp everyone who is subscribed to that extension can see? |
| 21:21.27 | Qwell | BLF is just lights. SLA is something completely different. |
| 21:21.37 | x86 | hmm |
| 21:21.41 | ManxPower | Well, SLA seems to be BLF + somethnig else. |
| 21:21.54 | Qwell | ManxPower: don't even really *need* BLF |
| 21:22.03 | ManxPower | But as I understand it, hints are used in both BLF and SLA |
| 21:22.28 | angryuser | SLA shared lines, old way |
| 21:22.33 | ManxPower | Qwell: So the user just has to press the line appearance to see if it's available or not? |
| 21:22.50 | ManxPower | SLA without busy lamps seems pretty useless. |
| 21:24.18 | ManxPower | angryuser: Users sure do love their key systems |
| 21:24.34 | iamhrh | ok, now I'm seeing "chan_skinny.c: 1295 find_subchannel_by_instance_reference: could not find subchannel with reference '0' on 'wtf' |
| 21:24.34 | iamhrh | my device registration looks like this: |
| 21:24.34 | iamhrh | [wtf] |
| 21:24.34 | iamhrh | device=SEP001795B0D93A |
| 21:24.34 | iamhrh | callerid="Test 7961" <1004> |
| 21:24.36 | iamhrh | context=phones |
| 21:24.38 | iamhrh | line=>1004 |
| 21:24.40 | iamhrh | speeddial=>1000@phones,Extension 1000 |
| 21:24.42 | iamhrh | speeddial=>1001@phones,Extension 1001 |
| 21:24.44 | iamhrh | speeddial=>1002@phones,Extension 1002 |
| 21:24.46 | iamhrh | speeddial=>1003@phones,Extension 1003 |
| 21:24.48 | iamhrh | (sorry for the wall of text) |
| 21:25.05 | ManxPower | iamhrh: next time use pastebin.ca |
| 21:25.09 | ManxPower | ~pb |
| 21:25.10 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:25.14 | mvanbaak | iamhrh: I'm getting those find_subchannel_by_instance_reference too all the time |
| 21:25.19 | mvanbaak | ignore it |
| 21:25.22 | mvanbaak | that's what I do |
| 21:25.26 | angryuser | no, before there were no busy lamps at all so, so secretary receved a call from A, A asked talk to B, secretary call B and "there is someonne who whant to talk you on Line X" B hang up, press Line X |
| 21:25.43 | iamhrh | heh, well , i would but the buttons still don't work :-( |
| 21:26.11 | ManxPower | angryuser: that's how things work NOW in Asterisk |
| 21:26.25 | iamhrh | oh bloody hell, i used the wrong context. |
| 21:26.33 | iamhrh | shoulda been 'internal' |
| 21:27.16 | mvanbaak | :) |
| 21:27.29 | angryuser | <ManxPower> well after Sla was integrated, yes, but before, no, only with xfer, and xfer on hang up + pers scripts + parking of course |
| 21:29.08 | ManxPower | This is exactly how all my users do it now: secretary receved a call from A, A asked talk to B, secretary call B and "there is someonne who whant to talk you on Line X" B hang up, press Line X |
| 21:29.21 | ManxPower | iamhrh: either stop flooding the channel or we will ban you |
| 21:29.36 | angryuser | <ManxPower> so you are yousing sla? |
| 21:29.39 | ManxPower | sorry, my scrooback buffer was messed up. |
| 21:29.52 | ManxPower | angryuser: no, we are not using SLA |
| 21:30.10 | ManxPower | Ah. sorry, I miss read what you said. |
| 21:30.14 | anonymouz666 | Asterisk is now AUDIO-LESS! |
| 21:30.15 | iamhrh | ouch, flooding the channel? sorry man :-( |
| 21:30.19 | anonymouz666 | :) |
| 21:30.33 | ManxPower | my users do this: secretary receved a call from A, A asked talk to B, secretary call B and "there is someonne who whant to talk you on Line X" A hangs up. |
| 21:30.36 | ManxPower | there, that's better |
| 21:30.50 | angryuser | <ManxPower> ok so we understand each other ;) |
| 21:31.21 | ManxPower | It saves a step or two |
| 21:31.56 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 21:32.05 | angryuser | <ManxPower> yes standard good way, but some old folks dont like that, so Sla appeared ;) |
| 21:32.28 | Katty | anyone have experience with wanpipe? |
| 21:33.05 | mvanbaak | anonymouz666: actually, it's pretty cool that oej fixed that |
| 21:33.44 | angryuser | ha someone heard about a book of asterisk 1.4 by Wintermeyer(aprox) ? |
| 21:33.52 | clyrrad | ManxPower: You still here? |
| 21:34.08 | angryuser | or Wintermayer |
| 21:34.12 | clyrrad | ManxPower: found another alternate solution to the multi row not being supported until 1.6 |
| 21:34.27 | *** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org) |
| 21:35.03 | sweeper | http://www.geeks.com/details.asp?InvtId=8835Y11-R <-- got my new asterisk boxen :D |
| 21:35.27 | angryuser | this one, i am tempted tu buy, but need some feedback coz it is not cheap http://www.amazon.fr/Asterisk-Telephony-Solutions-Installing-Customizing/dp/0321525663/ref=pd_bbs_sr_11?ie=UTF8&s=english-books&qid=1203370474&sr=8-11 |
| 21:35.50 | *** join/#asterisk umay (n=chris@71-208-162-10.hlrn.qwest.net) |
| 21:36.47 | iamhrh | ok, still not having any luck with this speed dial business. here are the relevant (i think, please tell me if there is more you need to see) configuration lines: http://pastebin.ca/909017 |
| 21:37.49 | ManxPower | iamhrh: Are you sure that the SCCP channel you are using even supports speeddials? |
| 21:38.18 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-69-154.lns10.syd6.internode.on.net) |
| 21:38.19 | iamhrh | its a cisco 7961, has 6 lighted keys, and the text is showing up next to the keys |
| 21:38.21 | *** join/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net) |
| 21:38.31 | iamhrh | i'm pretty sure it does, but i've been wrong before |
| 21:38.37 | angryuser | and tell me what is speeddial for ;) like you press 40 it dials 4030238493492439 really fast ? |
| 21:40.08 | sweeper | angryuser: honestly, I really really doubt it's worth 100 e-bucks |
| 21:40.08 | sweeper | just take a pdf of tfot to a print shop :P |
| 21:40.21 | angryuser | <sweeper> yes i was impressed by 600 paged |
| 21:40.24 | angryuser | *pages |
| 21:42.40 | angryuser | but there is a lack of info in the "book" about * inteconnection dundi, mass deploy, lack of agi knowledge, load balancing, failover, more examples also needed. |
| 21:42.43 | BBHoss | angryuser, its a translated book so i dunno |
| 21:44.20 | ManxPower | iamhrh: See: http://lists.digium.com/pipermail/asterisk-users/2003-December/024362.html |
| 21:44.33 | ManxPower | that is an old message |
| 21:46.06 | lirakis | later all |
| 21:46.26 | *** part/#asterisk lirakis (i=lirakis@pr0tected.us) |
| 21:46.38 | iamhrh | yeah - are you saying that the chan_skinny doesn't support the speedials? |
| 21:47.21 | ManxPower | iamhrh: This one is from 2007: http://lists.digium.com/pipermail/asterisk-users/2007-April/183897.html |
| 21:47.37 | ManxPower | iamhrh: I am saying read the links and make up your own mind. |
| 21:47.58 | iamhrh | i'm reading it right now man, i've been at this for like 6 hours |
| 21:48.15 | Qwell | iamhrh: you can try removing the #ifdef's, but it'll probably crash |
| 21:48.31 | ManxPower | iamhrh: very few people use SCCP/Skinny with Asterisk |
| 21:49.24 | mvanbaak | dont remove the ifdefs |
| 21:49.25 | iamhrh | that's what I'm starting to see, I've been investigating it while I wait for my cisco login to get updated so I can get the sip firmware |
| 21:49.28 | mvanbaak | it will crash |
| 21:49.29 | mvanbaak | I tried it |
| 21:49.35 | Qwell | heh |
| 21:49.38 | angryuser | when you read changelos, it does not changed a lot also from 1.2-1.6 beta, some changes, but not a big progression |
| 21:49.44 | Qwell | mvanbaak: it's probably a trivial fix, if you're bored |
| 21:49.46 | angryuser | *changelogs |
| 21:49.50 | mvanbaak | yeah |
| 21:50.05 | Katty | anyone have experience with wanpipe? |
| 21:50.19 | ManxPower | Katty: Yes. |
| 21:50.48 | mvanbaak | Qwell: is there a way to test skinny with a softphone ? |
| 21:50.57 | drmessano | Hmmmm |
| 21:51.02 | Qwell | there's a skinny softphone out there, but it sucks |
| 21:51.07 | J4k3 | heh, wanpipe |
| 21:51.12 | angryuser | <Katty> sangoma's cards? |
| 21:51.13 | Qwell | plus the cisco one, which is non-free |
| 21:51.13 | drmessano | hey, skinny |
| 21:51.18 | J4k3 | another company thinking you can make a decnet router of a peecee? *vomit* |
| 21:51.28 | J4k3 | been there/done that/got the card, got the invoice, got the hassles, got the cisco. |
| 21:51.29 | mvanbaak | Qwell: I dont feel like taking my skinny phones to fosdem :) |
| 21:51.35 | J4k3 | err |
| 21:51.38 | drmessano | skinny is the way of the future |
| 21:51.40 | J4k3 | decent, not decnet ;) |
| 21:51.41 | Qwell | mvanbaak: build a robot |
| 21:51.42 | Katty | angryuser: yes. (= |
| 21:51.48 | mvanbaak | lol Qwell |
| 21:51.49 | Qwell | actually.. |
| 21:51.54 | Katty | angryuser: having some compiling errors. |
| 21:51.54 | Qwell | you could probably telnet into it :p |
| 21:51.54 | Nugget | telnet is eeeeeeevil! |
| 21:52.00 | Katty | Nugget: you are. |
| 21:52.10 | J4k3 | ~nugget |
| 21:52.10 | jbot | well, nugget is & |
| 21:52.22 | J4k3 | ~& |
| 21:52.23 | jbot | it has been said that & is AND |
| 21:52.32 | Qwell | ~AND |
| 21:52.33 | jbot | it has been said that and is a binary operation, when 1 is returned ONLY when both operands are true |
| 21:52.38 | mvanbaak | Qwell: not into the 7905 |
| 21:52.41 | Qwell | mvanbaak: ahh |
| 21:52.48 | *** join/#asterisk Jake[work] (n=Jake@pool-71-175-117-161.phlapa.east.verizon.net) |
| 21:52.54 | mvanbaak | my 7960 is rebooting |
| 21:53.12 | mvanbaak | because the speeddial #ifdef is there for a reason ;) |
| 21:53.13 | angryuser | <Katty> not a clue |
| 21:53.55 | Nivex | J4k3: 3v1l |
| 21:54.02 | Katty | ManxPower: from the http://wiki.sangoma.com/wanpipe-linux-drivers page I have downloaded wanpipe-3.2.3.tgz for my a101d (echo) card. |
| 21:54.40 | J4k3 | YOUR SERVER IS ALREADY A ROUTER!!! OMG! |
| 21:54.48 | J4k3 | your server is already a piss-poor router! |
| 21:55.05 | Katty | ManxPower: around the section of selecting the drivers for my card, i get a little confused. |
| 21:55.30 | ManxPower | pick the TDM (asterisk) only option |
| 21:55.31 | J4k3 | Katty: terminate your circuits to a box of cracker jacks, you'll be better off. |
| 21:55.47 | BBHoss | Katty, what error is it giving |
| 21:55.54 | J4k3 | or, are these lame cards also being used as T1 interfaces for asterisk? *shudder* |
| 21:55.54 | mvanbaak | J4k3: the A101d actually is a very nice card |
| 21:56.15 | ManxPower | J4k3: what Katty is doing has nothing to do with routing, regardless of the driver names |
| 21:56.20 | J4k3 | mvanbaak: well, it might be for asterisk connectivity. I've seen a lot of 'serial interface routing cards' over the years, and they all universally sucked. |
| 21:56.40 | ManxPower | J4k3: She has a Sangoma, which does not universally suck. |
| 21:56.56 | mvanbaak | J4k3: since this is #asterisk we assume it's for asterisk connectivity |
| 21:57.10 | Katty | ManxPower: "TDM Voice (zaptel) support, correct? not the one with wan protocol support? |
| 21:57.18 | mvanbaak | Katty: indeed |
| 21:57.26 | ManxPower | Katty: ""TDM Voice (zaptel) support" |
| 21:57.58 | Katty | k, it's doin its thing |
| 21:58.02 | ManxPower | unless you are doing something incredibly stupid like want to use the box as an IP router as well. In that case, you are beyond even my help. |
| 21:58.05 | mvanbaak | hhmm, the cisco's annoy me |
| 21:58.12 | Katty | nope. |
| 21:58.16 | Katty | just want this card to work. |
| 21:58.19 | ManxPower | good |
| 21:58.20 | mvanbaak | but I dont feel like fixing it right now |
| 21:58.21 | J4k3 | mvanbaak: I'd still have a really hard time buying from some manufacturer that says a line like this... "So your server can become the router it was designed to be, routing IP over multi-megabit per second links with the reliability, ease of use and security inherent in a no-box solution." |
| 21:58.27 | defsdoor | Katty: I have 2 sites with A101D and they "just work" (tm) |
| 21:58.55 | Katty | ManxPower: i'm not smrt enough to do anything too fun (= |
| 21:59.18 | Katty | ManxPower: it's asking to visually confirm driver compilation... |
| 21:59.22 | Katty | ManxPower: i'm unsure how to do that. |
| 21:59.28 | Jake[work] | just take the defaults |
| 21:59.29 | ManxPower | Katty: does it look like it worked? |
| 21:59.40 | Katty | ManxPower: i didn't see any errors. |
| 21:59.46 | Katty | ManxPower: but i'm not sure what to actually check |
| 21:59.55 | Jake[work] | it would show errors |
| 21:59.56 | ManxPower | then assume it worked |
| 22:00.15 | Katty | ManxPower: and enabling startup scripts is good, yes? |
| 22:01.09 | ManxPower | yes |
| 22:01.48 | Katty | is there anything else i need to do? |
| 22:01.53 | *** join/#asterisk enzo (n=enzo@extranet.source-rh.com) |
| 22:01.55 | Katty | like make samples..or...some other thing |
| 22:01.58 | enzo | hi |
| 22:02.15 | defsdoor | Katty: it's actually on the sangoma wiki |
| 22:02.37 | enzo | i' seen links on skype supported by asterisk (or sip gateway, or skype channel), do you know if this is already possible ? |
| 22:02.49 | *** join/#asterisk AppleBoy (n=AppleBoy@about/cooking/nakedchef/apple/tarts) |
| 22:02.53 | AppleBoy | file: you there? |
| 22:03.01 | BBHoss | ~skype |
| 22:03.02 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 22:03.16 | mvanbaak | what jbot said |
| 22:03.22 | enzo | ok |
| 22:03.24 | enzo | snif |
| 22:03.46 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 22:03.53 | Katty | [hC]: (= |
| 22:04.49 | [hC] | Hello Katty, how are you? :) |
| 22:04.58 | Katty | [hC]: confused. first wanpipe install |
| 22:05.13 | ManxPower | Katty: all any of us did was follow the Sangoma wiki |
| 22:05.19 | [hC] | Katty: ahh! I can help you if you need some help |
| 22:05.20 | Katty | ManxPower: do you have a link? |
| 22:05.21 | [hC] | most liekly. |
| 22:05.23 | [hC] | likely* |
| 22:05.27 | Katty | probably, hc. |
| 22:05.34 | [hC] | Where are you stuck? |
| 22:05.43 | defsdoor | http://wiki.sangoma.com/ |
| 22:05.58 | *** join/#asterisk glen2 (n=glen@87.194.2.134) |
| 22:09.52 | *** join/#asterisk sacitec (n=tobi@189.149.133.58) |
| 22:10.05 | *** part/#asterisk sacitec (n=tobi@189.149.133.58) |
| 22:11.36 | [hC] | Anyone here use snom phones very much? specifically the 320/370? |
| 22:11.52 | BBHoss | i have used the 320 extensively, and the 370 somewhat |
| 22:12.06 | x86 | anyone ever use FOP? |
| 22:12.29 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta2 (2008/01/28), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.9 (2008/02/18), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 22:13.14 | file | AppleBoy: erm? |
| 22:13.34 | [hC] | BBHoss: so quick question.. the 320 has kinda smushy rubber buttons.. is the 370 more hard-plastic buttons? it looks to be in the pictures. |
| 22:13.35 | x86 | i'm having some issues getting FOP to monitor a range of zap channels... worked fine for my first span (also specified as a range), but when I go to specify the second range it doesn't seem to work |
| 22:14.05 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk, -Addons, Libpri 1.6.0-beta3 (2008/02/18), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.9 (2008/02/18), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 22:14.16 | BBHoss | [hC], the early-rev snom phones have the squishy buttons, but they told me that they had a new rev where they were all the hard ones |
| 22:14.25 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 22:14.32 | [hC] | BBHoss: yeah, im glad they did that. i dont like squishy buttons. |
| 22:14.47 | [hC] | BBHoss: i think they also have new material ont he handset that is more grippy on the inside |
| 22:15.14 | *** join/#asterisk iamthelostboy (n=nathan@125-236-212-46.adsl.xtra.co.nz) |
| 22:15.57 | iamthelostboy | i have a small problem with my queue |
| 22:17.05 | BBHoss | [hC], yeah last expo i went to, there were people bitching about the color of the phone, it was too blue. I didn't think it was a big deal |
| 22:17.23 | [hC] | BBHoss: meh. big deal. its so dark it really doesnt matter. |
| 22:17.36 | Katty | file: mew? |
| 22:17.48 | *** join/#asterisk RoyK (n=roy@ip-103-19-149-91.dialup.ice.no) |
| 22:17.49 | iamthelostboy | its quite simple, in that the queue phones several sip accounts whenever a call comes into it... when one of the phones answers, it should stop ringing the rest, which mostly is what happens, though sometimes some of the sip phones keep ringing, and when answered show 'call ended' |
| 22:17.56 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:19.51 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 22:20.24 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 22:21.32 | *** join/#asterisk sacitec (n=tobi@189.149.133.58) |
| 22:22.21 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 22:32.51 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 22:33.13 | *** join/#asterisk nirz (n=nir@84.229.92.126) |
| 22:36.24 | *** join/#asterisk vk4akp (n=vk4akp@c210-49-238-7.ipswc1.qld.optusnet.com.au) |
| 22:36.42 | vk4akp | HI everyone. |
| 22:37.12 | vk4akp | I've just installed Asterisk 1.4.11 and set up an outbound SIP account. |
| 22:37.17 | vk4akp | I dial in and get the demo stuff. |
| 22:37.21 | vk4akp | I'm so happy!. :) |
| 22:38.02 | vk4akp | I'm wondering if someone can walk me through a quick setup to add a conference extension PSE? |
| 22:38.04 | endre | cool |
| 22:39.12 | murdmath | [h |
| 22:39.13 | murdmath | [hC] The snom 320's now come with hard buttons |
| 22:39.18 | Jake[work] | vk4akp: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe - is this what you're looking for? |
| 22:39.43 | vk4akp | TNX, I'll have a look now. |
| 22:40.10 | Katty | i like the speakerphone meetme. |
| 22:40.16 | Katty | or.. i guess they call that intercom. |
| 22:40.29 | Katty | all you techy people and your techy words. |
| 22:40.50 | murdmath | [hC]: The sidecare also has the hard buttons now. |
| 22:41.06 | [hC] | nice.. |
| 22:41.11 | *** part/#asterisk AppleBoy (n=AppleBoy@about/cooking/nakedchef/apple/tarts) |
| 22:41.36 | murdmath | [hC]: But the handsets on the 300, 320, & 370 all seem to be the same. |
| 22:41.43 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 22:42.02 | [hC] | Ive just started getting my hands ont he snom's ... I like them quite a bit so far... i seem to recall some people not being so impressed.. but so far i like them.. |
| 22:42.19 | murdmath | [hC]: I took two 320's one with the soft buttons and one with the hard buttons and they are not interchangeable. |
| 22:42.37 | murdmath | [hC]: I have about 100 in service right now. |
| 22:43.02 | [hC] | murdmath: any beefs with the line? |
| 22:43.32 | J4k3 | [hC]: everything is a matter of perspective. I still like my grandsuck budgetnone 101's |
| 22:43.50 | J4k3 | yes, I know they're inferior to every other phone on the market |
| 22:44.13 | murdmath | [hC]: always test new firmware before deploying... The way they tilt the screen is a bit cheap. |
| 22:44.13 | [hC] | i put my budgetone in the bathroom |
| 22:44.14 | J4k3 | ... and they're half the price (or less) of the nearest competitor... and they still work beter than any POTS line in the whole f'n county :P |
| 22:44.18 | [hC] | i use it as bathroom music moh |
| 22:44.27 | J4k3 | [hC]: haha, not a bad idea |
| 22:44.54 | J4k3 | [hC]: I use one of mine as a wifi phone |
| 22:45.03 | [hC] | J4k3: .... ? |
| 22:45.23 | J4k3 | lunchkit + battery + wifi bridge + gsbt01 |
| 22:45.26 | J4k3 | er gsbt101 |
| 22:45.29 | murdmath | [hC] The 370 has a nice screen compared to the rest. |
| 22:45.43 | [hC] | murdmath: yeah it looked that way. |
| 22:45.52 | murdmath | [hC]: Expensive though. |
| 22:46.03 | J4k3 | it works, which is more than I can say for the 4 wifi 'handsets' I used. |
| 22:46.11 | J4k3 | I'd rather like to move to a wifi-to-dect setup |
| 22:46.24 | J4k3 | the SBCs I'm using support USB |
| 22:46.26 | ManxPower | The only thing worse than a Grandstream is a WiFi SIP phone. |
| 22:46.27 | J4k3 | err USB2 |
| 22:46.41 | J4k3 | so I could easily add a USB DECT base of some sort |
| 22:46.55 | murdmath | <J4k3>: I want to try snom's m3 |
| 22:47.03 | J4k3 | ManxPower: well, I'd say the grandstreams are several orders of magnitude better than any wifi handset I've touched :) |
| 22:48.59 | sweeper | J4k3: so grandsream must have hired an entirely new crew to do their wireless stuff? |
| 22:49.26 | sweeper | because grandstream wired handsets are orders of magnitude worse than any other wired handset I've touched :P |
| 22:50.20 | *** join/#asterisk Enron (n=foo@216.70.173.176) |
| 22:50.35 | Enron | Hi anyone here use cisco ip phones? |
| 22:50.55 | ManxPower | Enron: not nearly as many as use Polycom phones |
| 22:51.25 | Enron | heres our problem cisco 7912 phones time are off by +1 hour, but the newer models all have correct time |
| 22:51.27 | Jake[work] | i udr s 7960 |
| 22:51.37 | Jake[work] | use a |
| 22:51.38 | Jake[work] | haha |
| 22:51.48 | drmessano | Enron? |
| 22:51.53 | Enron | the phone itself doesn't have a time setting i'm thinking it gets time from our asterisk server |
| 22:51.53 | drmessano | ENRON would use Cisco |
| 22:51.57 | Enron | lol |
| 22:52.02 | drmessano | No wonder they're out of business |
| 22:52.15 | drmessano | Enron: WUT STOCK? |
| 22:52.18 | ManxPower | Enron: the time on the phone is NOT set by Asterisk. |
| 22:52.38 | ManxPower | The phone needs to be configured with the right timezone and NTP server |
| 22:52.59 | Enron | it downloads the config from the tftp server right? |
| 22:53.13 | Enron | are the config for each phone same or do each have their own? |
| 22:53.53 | *** join/#asterisk hfb (n=hfb@pool-72-67-142-193.lsanca.dsl-w.verizon.net) |
| 22:54.11 | ManxPower | Enron: I think you need to go to Cisco |
| 22:54.15 | ManxPower | s site for some docs |
| 22:54.15 | Enron | not sure how half the phones are correct and the others arn't |
| 22:54.28 | Enron | thought it was an asterisks thing sorry :) |
| 22:54.29 | ManxPower | I am assuming you are using SIP phones, not SCCP/Skinny. |
| 22:54.29 | Enron | ty |
| 22:54.34 | Enron | yea sip |
| 22:59.47 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 23:01.39 | J4k3 | sweeper: they don't make wifi handsets |
| 23:06.00 | drmessano | Does this mean Enron is back in business? |
| 23:09.57 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 23:11.29 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:19.03 | sweeper | J4k3: oh, gotcha |
| 23:19.23 | sweeper | well, I've got a utstarcom 1000, it's rpetty basic, but it's pretty trouble-free |
| 23:19.50 | *** join/#asterisk DJF5 (n=DJF5@84-105-201-37.cable.quicknet.nl) |
| 23:21.13 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 23:22.21 | *** join/#asterisk anthm (n=anthm@mb40736d0.tmodns.net) |
| 23:22.21 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:30.31 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:36.18 | *** join/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2) |
| 23:38.24 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:43.27 | *** join/#asterisk PepOSX (n=angeldav@190.72.133.19) |
| 23:49.01 | *** part/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net) |
| 23:51.35 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |