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00:30.00 | xp_prg | what does asterisk gateway mean, what is a gateway too exactly? |
00:30.42 | shmaltz | ~wiki gateway |
00:31.06 | shmaltz | ~wiki Gateway (telecommunications) |
00:33.39 | shmaltz | ~quiet |
00:33.40 | jbot | ACTION ok, ok, I will be quiet. When you start making sense, that is. |
00:34.31 | *** join/#asterisk angryuser (i=nononon@df01t2-212-195-117-162.d4.club-internet.fr) |
00:36.34 | justdave | anyone know if the directories defined in /etc/asterisk.conf are available as variables in dialplans? |
00:40.02 | [hC] | If asterisk accepts a g729 call, and ends up going to someones voicemail, do i need a g729 license/codec for the message to be left in .g729 format? |
00:42.16 | shmaltz | justdave, I don't think so, but for what purpose? |
00:42.35 | justdave | it'd be a one-time setting, I can put it in [globals] easily enough I suppose :) |
00:42.40 | shmaltz | [hC], you need it just for accepting it |
00:43.01 | justdave | I'm running a System() command to move an audio file on the filesystem after a user records it |
00:43.29 | shmaltz | justdave, you could use the app_record and specify where it's stored |
00:43.52 | justdave | I am specifying where it's stored. :) In /tmp so it doesn't overwrite an existing file |
00:44.08 | justdave | after the user okays that they like what they recorded, then I want to move it overtop of the original |
00:44.23 | justdave | but that means figuring out where Asterisk is looking for it, and I hate hard-coding pathnames in the dialplan |
00:44.28 | shmaltz | justdave, http://pastebin.ca/894519 |
00:44.56 | shmaltz | oh, so I guess globals will do |
00:46.23 | justdave | Ubuntu uses /usr/lib/asterisk/sounds, RHEL uses /var/lib/asterisk/sounds, want to be able to use the same script on both boxes |
00:47.22 | justdave | s/RHEL/source install/ |
00:47.32 | justdave | hah, nice bot |
00:47.41 | *** join/#asterisk micander (n=Michael_@Full-Service-Travel-1157986.cust-rtr.pacbell.net) |
00:50.10 | [hC] | shmaltz: no you dont. you dont need the codec to do passthru, just the format. |
00:50.31 | shmaltz | [hC], then you do need it for VM |
00:51.08 | [TK]D-Fender | justdave, make a constant yourself for each popular option and uncomment the one applicable |
00:52.49 | justdave | he only needs it for voicemail if the user doesn't have a g.729-capable phone when he checks it |
00:53.02 | [hC] | the user does. |
00:53.05 | justdave | only need the license to transcaode it I believe |
00:53.14 | [hC] | I asked if app_voicemail would save in native g729 |
00:53.18 | [hC] | and i think it does. |
00:53.38 | sbingner | right |
00:53.48 | sbingner | to justdave |
00:53.50 | *** join/#asterisk weazahl_ (n=jeremy@adsl-68-90-168-7.dsl.ksc2mo.swbell.net) |
00:54.07 | [hC] | yes that is correct, I just didnt know if app_voicemail transcoded it to g729 to save it or not |
00:54.08 | [hC] | I'll just check., |
00:54.09 | sbingner | you need to codec to play the prompts to the user on g729, unless perhaps if you have them all recorded in g729 already |
00:56.00 | drmessano | oh crap |
00:56.11 | drmessano | s/oh crap/gee whiz/ |
00:56.56 | weazahl_ | what a day, DSL went down for the entire city last night. 9pm to 5am. i was in withdraw... then this morning i had to deal with the aftermath. of course, 1 of the 2 sites that didnt comeback up was my shop. |
00:57.06 | sbingner | s/g729/evil proprietary expensive format/ |
00:57.17 | sbingner | wth it only did it once? |
00:57.37 | sbingner | s/it/test/g |
00:57.42 | weazahl_ | so we had one analogue line to run the store on this morning. fun |
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01:25.10 | husimon | so how much echo cancellation do you folks usually deploy? |
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01:29.49 | [TK]D-Fender | husimon, at leasty thirty. |
01:31.53 | husimon | [TK]D-Fender, What if you have hardware with EC built in, does it just automatically try and remove echo? Or is that dependent on what hardware you have. |
01:32.12 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
01:32.20 | [TK]D-Fender | husimon, if it detects echo it will activate and suppress it |
01:33.22 | Kobaz | hmm, this isn't good |
01:33.28 | husimon | [TK]D-Fender, hmm yeah I have EC hardware but still a slight bit of echol, so I'm thinking about enabling EC in *. I could turn off the hardware EC and only do *, or try both. Which do you think is the better route? |
01:33.32 | Kobaz | latest asterisk stable seg faults on startup |
01:34.13 | [TK]D-Fender | husimon, perhaps you should tell us what you have, show us how you are using it (and that it is in use), and show some care in the question you ask. |
01:34.40 | husimon | [TK]D-Fender, well the hardware appears to have no way to controlling the actual EC, except for on off, that's why I didn't give you further information. |
01:35.05 | husimon | i'm not using any EC in * yet. |
01:35.09 | [TK]D-Fender | husimon, including the MAKE and MODEL. Yes, very informative... |
01:35.41 | husimon | [TK]D-Fender, sure i'll tell you but you won't like it ;) redfone fonebridge2-ec 2 port pri. Uses tdmoe (yes I know you hate this) |
01:35.43 | b11d | Kobaz.. works for me. |
01:35.50 | Kobaz | yeah |
01:35.51 | Kobaz | heh |
01:35.52 | Kobaz | that's good |
01:35.59 | b11d | aye :) |
01:36.23 | [TK]D-Fender | husimon, yeah... |
01:36.26 | [TK]D-Fender | ~wglwat |
01:36.26 | jbot | it has been said that wglwat is well, good luck with all that |
01:36.47 | [TK]D-Fender | Crap-tastic |
01:37.10 | b11d | husimon.. where are you getting echo anyways? just between your voip phones and the PRI? |
01:37.27 | husimon | echo between voip phones and outside numbers |
01:37.33 | husimon | not major |
01:37.37 | husimon | but sometimes I can hear bits of myself talking |
01:37.38 | Kobaz | what's weird |
01:37.46 | Kobaz | is if i spawn asterisk as a daemon, it doesn't seg fault |
01:37.52 | Kobaz | if i start it in the foreground, it crashes |
01:37.58 | b11d | and when you attach to it afterwards? |
01:38.01 | husimon | kobaz what does the log say? |
01:38.21 | b11d | husimon.. you're just crazy.. its all in your head. |
01:38.22 | Kobaz | some undefined symbols for the queue and meetme modules |
01:38.25 | husimon | blld laugh |
01:38.31 | husimon | blld i've had users mention it too |
01:38.35 | b11d | heh |
01:38.53 | b11d | do you have echotraining enabled in your zapata.conf ? |
01:39.05 | Kobaz | hmm |
01:39.08 | Kobaz | i think so |
01:39.24 | b11d | im asking husimon.. not you Kobaz :) |
01:39.34 | Kobaz | oh |
01:39.36 | b11d | sorry |
01:39.39 | [TK]D-Fender | b11d, shouldn't have to. |
01:39.50 | b11d | does it segfault when you attach to the daemon Kobaz? |
01:39.51 | husimon | yeah i don't blld |
01:39.59 | b11d | hmm.. you enabled it for me TK :) |
01:40.07 | b11d | husimon.. try it? |
01:41.01 | husimon | is that for * ec or hardware ec |
01:41.07 | husimon | atm I only have hardware ec |
01:41.14 | [TK]D-Fender | b11d, yes, but in his case those vars can't control the remote TDMoE interface. it delivers pre-processed frames. |
01:41.14 | husimon | enabled that is. |
01:41.20 | b11d | oh :) |
01:41.32 | husimon | i'm emailing the company to askt hem |
01:41.33 | husimon | ask them |
01:50.04 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
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01:52.34 | b11d | ttyl al |
01:52.35 | b11d | all |
01:55.45 | lunaphyte | i guess the dialplan is not applied to dialed extensions in a linear fashion, from top to bottom? |
01:56.46 | husimon | i think so |
01:56.51 | husimon | except for the priorities |
01:56.54 | husimon | which it goes by number |
01:58.15 | lunaphyte | oh, oops. looks like you're right. i had a line unplugged. |
01:59.35 | lunaphyte | so one could craft a dialplan with overlapping patterns to try 1 action first and then another if not successful? |
02:00.51 | husimon | what do you men? |
02:00.56 | husimon | try one outbound trunk then another? |
02:01.06 | lunaphyte | yeah, more or less. |
02:01.33 | lunaphyte | remember my questions from yesterday, about a long list of prefixes? |
02:02.25 | lunaphyte | the point with all of that is that not all of the prefixes in my area code are considered "local" |
02:02.43 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
02:03.10 | lunaphyte | so my goal is to send all calls to truly local pots prefixes out the pots line, and then send any other prefixes out through my itsp. |
02:03.34 | lunaphyte | and, at the same time, not require 10 digit dialing for those non-local prefixes. |
02:03.51 | husimon | so shouldn't that be a simple matter of listing all the local calls as patterns |
02:05.15 | lunaphyte | yeah, i was able to do that, with your guys' help - but, to match the remainder means i have to use a pattern that overlaps with the local patterns. |
02:05.26 | lunaphyte | at least, that was the only way i could figure to do it. |
02:05.40 | *** join/#asterisk spyder12345 (n=kyle@rrcs-67-78-17-78.se.biz.rr.com) |
02:06.11 | *** join/#asterisk HeXeD (n=hex@87-194-8-43.bethere.co.uk) |
02:06.36 | lunaphyte | so i was a little concerned that when a local pattern was dialed, it might get picked up by the "catch-all" pattern and get sent out through the itsp instead of pots. |
02:06.51 | spyder12345 | With version 1.6 how can I play a sip call using tcp? |
02:07.33 | lunaphyte | it seems to be working, though i wonder if there might be a better approach. |
02:08.03 | husimon | lunaphyte, how come there is overlap? |
02:08.11 | husimon | shouldn't the area code decide if it is local or not |
02:08.52 | lunaphyte | the idea was to eliminate the need to dial the area code. |
02:09.33 | husimon | lunaphyte, so just send all 7 digit numbers to your pots |
02:09.54 | husimon | then create patterns that handle the local area codes and send those to the pots |
02:10.13 | husimon | i think I understand your question now |
02:10.20 | husimon | because after those 2 rules you require a catch all |
02:10.23 | phix | ok I am here again to try and get my ports working |
02:10.23 | husimon | right? |
02:10.39 | phix | so I will try to compile zaptel and the zaptel modules from source and see how that goes |
02:10.44 | phix | any other ideas? |
02:11.35 | [TK]D-Fender | lunaphyte : there is, stop using catch-alls |
02:12.02 | phix | [TK]D-Fender: hey! :) |
02:12.06 | husimon | [TK]D-Fender, how do you then tell all area codes besides a few to go through a given trunk? |
02:12.09 | [TK]D-Fender | lunaphyte : or use one that checks if it would match a specific series you'd route differently and branch off as needed |
02:12.19 | phix | [TK]D-Fender: How are you with TDM400p's? |
02:12.23 | lunaphyte | husimon: if i send all 7 digit numbers to pots, some will have errors, because even though they're in the same area code, certain prefixes are not local and require 1 + the area code be included. |
02:12.28 | [TK]D-Fender | phix, in what way? |
02:12.51 | husimon | lunaphyte, that's your users problem for not dialing in the correct way then. |
02:12.54 | [TK]D-Fender | lunaphyte : ok, whats the problem with this? |
02:14.00 | phix | [TK]D-Fender: I am trying to figure out why I can only get 2 out of 3 FXS working |
02:14.10 | *** join/#asterisk jwh (n=jwh@kelley.ber.rewt.org.uk) |
02:14.29 | phix | [TK]D-Fender: working as in I have a dial tone, the third one sounds like zaptel or asterisk isn't running (no dial tone, amps everything sent to it) |
02:14.59 | [TK]D-Fender | phix, be very careful because the ports in the back aren't in the order you might think |
02:15.01 | phix | the zaptel module (wctdm) picks up all three ports |
02:15.08 | lunaphyte | let me back up a bit and try to state my goal. |
02:15.39 | [TK]D-Fender | lunaphyte : Go deal with "1NXXXXXX" |
02:15.50 | phix | [TK]D-Fender: oh ok, when looking towards the back plane, right to left is ports 1, 2, 3, 4? |
02:16.16 | phix | or looking left to right, 4, 3, 2, 1 even |
02:16.16 | phix | :) |
02:18.28 | spyder12345 | Does anyone know with version 1.6 when dialing via sip how you specify tcp instead of udp? |
02:18.56 | [hC] | phix: try one. if it doesnt work, try the other. |
02:18.56 | [hC] | :) |
02:20.16 | *** join/#asterisk jpeeler (n=jpeeler@adsl-249-75-145.hsv.bellsouth.net) |
02:22.42 | lunaphyte | in my area code, prefixes are split into 2 groups - local and "short-distance" (or whatever it's called). local calls can be dialed with 7 digits, but short-distance calls must require 10 digit dialing. my pots line has only local service on it, so no short-distance or long-distance calls can be made on it |
02:22.58 | *** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
02:23.10 | phix | [hC]: :) only two work out of three! |
02:23.37 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
02:23.39 | lunaphyte | additionally, it would be great to not have to remember or lookup, or do trial and error each time a calls is made, which prefixes require 10 digit dialing and which don't. |
02:24.46 | [TK]D-Fender | phix, the modules on the card are something like 4,1,2,3 IIRC |
02:25.02 | husimon | lunaphyte, i dunno seems like you can't do that because the numbers will overlap |
02:25.22 | husimon | lunaphyte, the solution is probably to just require people use 10 digit for all numbers. |
02:25.55 | lunaphyte | ultimately, if that's the case, that's ok. i'm just experimenting w/ * for the time being. |
02:26.11 | husimon | tk might have better ideas on how to solve that |
02:26.23 | husimon | that was only my first reaction |
02:30.09 | lmadsen | lunaphyte: 'phyte' is what my old bbs was called |
02:30.21 | lmadsen | actually... it was 'phyte!' |
02:30.21 | lmadsen | heh |
02:30.36 | lunaphyte | funny :) |
02:30.54 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
02:31.27 | lmadsen | one of my biggest regrets was formatting the HD of that computer without backing up the bbs |
02:31.49 | lmadsen | it was on purpose because I'd had enough of the bbs, but would be awesome to have a copy of it for nostalgic value |
02:32.11 | lunaphyte | such is hindsight... |
02:32.22 | lmadsen | indeed |
02:33.03 | husimon | had enough of the bbs? what did it do to you? |
02:33.04 | husimon | heh |
02:33.30 | lmadsen | nothing... I upgraded to internet :) |
02:33.34 | lmadsen | the internets! |
02:33.39 | lmadsen | a series of tubes |
02:35.15 | lunaphyte | you never know, you could have had the next isca. |
02:35.22 | alrs | lmadsen: there are still g-files around that list me as a cosysop of a board that hasn't existed for 18 years |
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02:37.00 | lmadsen | alrs: :D |
02:37.07 | lmadsen | ya... the internet is a funny thing |
02:37.24 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
02:37.29 | SwK | anyone know a good source for refurb'd polycoms? |
02:37.37 | ZaVoid | nope :( |
02:37.38 | lunaphyte | me! |
02:38.00 | lunaphyte | oh, you probably mean phones... :( |
02:38.04 | SwK | yeah |
02:38.07 | SwK | i do |
02:38.21 | lmadsen | that business exists already? |
02:38.28 | lunaphyte | darn, i've got a viewstation i'd like to unload. |
02:38.37 | lmadsen | last time I heard about refurb'd phones, it was for a meridian system |
02:38.58 | lmadsen | anyone have an idea what I can use my Zaurus SL-5500 for? :) |
02:39.13 | [TK]D-Fender | paper-weight ;) |
02:39.16 | lmadsen | aye |
02:40.53 | outtolunc | probably be useful as a remote control |
02:41.52 | outtolunc | send it to me i will find a use for it <G> |
02:41.53 | lunaphyte | if you had something fairly lightweight nearby that you wanted to damage without having to get up, it could be handy for that. |
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02:46.25 | *** join/#asterisk NoRemorse (n=fred@eth2459.vic.adsl.internode.on.net) |
02:46.38 | NoRemorse | hi all |
02:46.39 | NoRemorse | can anyone tell me why if I directly edit sip.conf in trixbox and add a peer, it does not show up under the FreePBX |
02:47.05 | lmadsen | NoRemorse: see #freepbx for support |
02:47.11 | ZaVoid | because trixbox loads everything into a db i believe |
02:47.15 | ZaVoid | and what lmadsen said |
02:47.21 | ZaVoid | hey lmadsen question for you |
02:47.28 | outtolunc | shouldn't you be editing sip_additional/custom.conf something like thta.. and asking on #trixbox? |
02:47.28 | lmadsen | or #trixbox |
02:47.43 | ZaVoid | exten => s,n,Set(CHANNEL(language)=en) <--- can i replace "en" with a variable i pull from my db via agi? |
02:47.51 | lmadsen | ZaVoid: of course |
02:47.53 | kyron | lmadsen, and you could probably have ran your BBS off a telnet port like my buddy Synoptic does ;) |
02:47.53 | Nugget | telnet is eeeeeeevil! |
02:47.58 | *** join/#asterisk b1ch0 (n=ralabiso@static-200-105-209-46.acelerate.net) |
02:48.14 | ZaVoid | so exten => s,n,Set(CHANNEL(language)=${ARG3}) |
02:48.17 | ZaVoid | is that right? |
02:48.19 | lunaphyte | yay telnet! |
02:48.51 | ZaVoid | or am i formatting that wrong? |
02:48.53 | kyron | Nugget, nooooooooooo, it's fun for password sniffing ;) |
02:48.56 | lmadsen | exten => s,n,Set(CHANNEL(language)=${ODBC_SQL(SELECT language FROM my_users WHERE username = '${ARG1}')}) |
02:49.15 | kyron | I prooved my point about using hubs and not switches at my first internship ;) |
02:49.24 | ZaVoid | yeah but i'll already have the variable from a sql query i did earlier in the dialplan |
02:49.36 | kyron | lmadsen, "why does it take so long for the calls to be answered" |
02:49.47 | lmadsen | ZaVoid: format is correct -- or replace it with another function where you get data from. ${DB(family/var)}) could be another place |
02:49.52 | NoRemorse | sip show peers in asterisk console lists the peer tho, |
02:49.55 | ZaVoid | ok thanks man |
02:50.17 | ZaVoid | of course it does NoRemorse you've added it to the asterisk |
02:50.21 | [TK]D-Fender | NoRemorse, FreePBX does not READ * config files, it generates (obliterates) them. |
02:50.27 | lmadsen | kyron: collisions are hawt |
02:50.41 | [TK]D-Fender | NoRemorse, and... |
02:50.42 | [TK]D-Fender | ~freepbx |
02:50.43 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
02:50.45 | NoRemorse | interesting, i wonder why a2billing uses flat files instead of writing into the database :( |
02:51.06 | NoRemorse | ok thanks |
02:51.09 | *** part/#asterisk NoRemorse (n=fred@eth2459.vic.adsl.internode.on.net) |
02:51.53 | *** join/#asterisk PepOSX (n=angeldav@190.72.146.71) |
02:52.51 | lmadsen | I also heard trixbox has the worlds largest asterisk community... so i always wonder why trixbox users keep coming in here asking for help |
02:53.56 | [TK]D-Fender | lmadsen, Same reason Lisa-Marie Presley & Michael Jackson broke up... irreconcilable similarities ;) |
02:54.18 | kyron | lmadsen, LOL |
02:54.30 | kyron | lmadsen, sniffing used to be sooo esy |
02:54.50 | ZaVoid | lmadsen: so in my php i'm doing $agi->set_variable("LANG", intval($row['lang'])); so i can just Set(CHANNEL(language)=${LANG}) makes sense right? |
02:55.16 | ZaVoid | gonna try it in a few mins.. just wanna make sure i got the theory down... think i do |
02:55.16 | lmadsen | yes |
02:55.21 | ZaVoid | cool thanks |
02:59.31 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
03:05.52 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
03:05.52 | *** mode/#asterisk [+o russellb] by ChanServ |
03:11.19 | ZaVoid | whos bot records everything for ircarchive.info ? |
03:11.47 | lmadsen | jbot |
03:11.58 | lmadsen | jbot: who owns you? |
03:11.59 | jbot | TimRiker does |
03:14.29 | lmadsen | ok... so I've a customer who needs me to answer a fax from a zap channel, then dial out another zap channel to a fax server. Once the fax server answers, I need to play some DTMF to the fax server. I see the D() option in Dial(), but it says, "Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged"....so I wonder if that will work or not.... anyone try the D() flag before? |
03:15.15 | [TK]D-Fender | lmadsen, That'd work. |
03:15.21 | lmadsen | oh ya? ok swet |
03:15.23 | lmadsen | sweet* :) |
03:15.47 | lmadsen | I'm just a bit confused by what "but before the call gets bridged" |
03:15.50 | lmadsen | means |
03:16.21 | ZaVoid | maybe the rtp he means? |
03:16.22 | [TK]D-Fender | lmadsen, means you pass on DTMF before the adio is bridged from the calling side to the called (you don't hear the DTMF or anything till after its done) |
03:16.34 | lmadsen | ahhhhhhhh |
03:16.38 | lmadsen | gokie |
03:16.54 | lmadsen | cute quit msg :) |
03:19.43 | JunK-Y | http://www.monopolyworldvote.com/en_US/world/leaders , wow montreal is #1! |
03:20.27 | russellb | heh, jpeeler must be trying to set up his tdm400p at home |
03:20.51 | lmadsen | :) |
03:26.36 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
03:26.51 | lmadsen | anyone wanna try faxing something to me? :) |
03:27.10 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
03:28.52 | lmadsen | :D |
03:29.01 | lmadsen | this is actually a customers fax line, so no worries :) |
03:29.19 | _ShrikE | I test with http://www.interpage.net/sub-wwwfax.html |
03:29.34 | _ShrikE | not sure about calling canada though |
03:30.11 | outtolunc | http://www.tpc.int/ |
03:30.12 | lmadsen | this is a US number |
03:30.24 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-48-245.pskn.east.verizon.net) |
03:30.27 | lmadsen | _ShrikE: oh hawt... thx! |
03:32.00 | *** join/#asterisk weazahl_ (n=jeremy@adsl-68-90-168-7.dsl.ksc2mo.swbell.net) |
03:33.14 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
03:34.48 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
03:36.17 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
03:37.35 | *** join/#asterisk AJayMN (i=AJaymn@71-82-218-158.dhcp.mdsn.wi.charter.com) |
03:38.53 | eric2 | anyone use callweaver? |
03:38.56 | eric2 | for faxing |
03:41.35 | ZaVoid | why do people still fax? :( |
03:42.17 | justdave | I just scan to a pdf and email it to the recipient :) |
03:42.56 | ZaVoid | remember that company visonx or somthing like that.. that back in mid 90's made personal scanners(single sheet) that sat behind your keyboard? |
03:42.57 | coppice | although it is stupid from a technical point of view, faxing has various legal implications |
03:42.59 | *** part/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
03:43.22 | _ShrikE | eric2: I have used callweaver with t38gateway with some pretty good success. |
03:44.12 | eric2 | I need faxing for my customers |
03:44.20 | eric2 | ya, that's right, legacy faxing |
03:44.27 | ZaVoid | oh snap they still around: http://orders.visioneer.com/category.jsp?category=MOBILE |
03:44.34 | ZaVoid | althought they cost more now then in the 90's |
03:44.35 | eric2 | _ShrikE did you use a 3rd party for faxing? |
03:44.50 | eric2 | got any recommendations? |
03:45.13 | riddlebox | what sites do you guys buy your phones from? |
03:45.28 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
03:45.28 | ZaVoid | yeah PAPERPORT http://support.visioneer.com/products/mobile/EOL/Sheetfed/default.asp |
03:45.33 | ZaVoid | lets talk oldschool :) |
03:45.35 | _ShrikE | just get the latest spandsp and callweaver and give it a try. I am using it with an audiocodes mp-114. |
03:45.58 | eric2 | riddlebox - gentek.com |
03:46.19 | ZaVoid | i buy mine form abptech |
03:47.33 | ZaVoid | anyone else use this new program called skitch? im in love with it |
03:47.44 | eric2 | what does it do? |
03:48.03 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
03:48.19 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
03:48.51 | ZaVoid | lets you take a snapshot of anyhwere on your screen and lets you easily mark it up.. type on it.. circle stuff etc etc etc then quick upload to a server of yours and gives you the link |
03:48.55 | L|NUX | Hello every one |
03:48.59 | L|NUX | can some one tell me how can i brodcast livestream on sip |
03:49.22 | eric2 | L|NUX - that's a good question |
03:49.48 | L|NUX | eric2: thanks but there should be a good answer for that question :) |
03:50.10 | husimon | yeah I really don't understand why faxing is around still |
03:50.17 | [TK]D-Fender | L|NUX, SIP is not a "broadcast" medium |
03:50.29 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-112-81.knology.net) |
03:50.36 | [TK]D-Fender | husimon, in a word : luddites |
03:50.41 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
03:50.41 | *** mode/#asterisk [+o russellb] by ChanServ |
03:50.58 | husimon | laugh |
03:51.13 | eric2 | faxing is used for legal docs |
03:51.20 | eric2 | main reason as far as I know |
03:51.20 | L|NUX | [TK]D-Fender: i can understand |
03:51.20 | husimon | it's not like you couldn't scan them |
03:51.29 | eric2 | people are old skool |
03:51.36 | L|NUX | [TK]D-Fender : but can we play mms:// stream using some program on sip ? |
03:51.45 | husimon | LINUX EH!? |
03:51.59 | L|NUX | [TK]D-Fender : so that when some one call to number they can listen stream ? |
03:52.22 | coppice | Sarbanes Oxley has created a resurgence of faxing |
03:53.12 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
03:53.12 | *** mode/#asterisk [+o russellb] by ChanServ |
03:54.14 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
03:55.18 | phix | hi |
03:55.30 | [TK]D-Fender | L|NUX, use something like rawplayer for MoH and tune to a broadcast |
03:56.24 | eric2 | callweaver is only used if you have additional hardware... no? |
03:56.32 | riddlebox | [TK]D-Fender, do you recomend using a TDM808B for a system that will have 7 lines? or should I get two 4 port cards? |
03:57.01 | [TK]D-Fender | riddlebox, partial PRI too pricey? |
03:57.11 | riddlebox | [TK]D-Fender, yeah |
03:57.23 | riddlebox | and the company doesnt want to use it for some reason |
03:57.40 | [TK]D-Fender | riddlebox, Only justifiable one is money. |
03:57.45 | *** join/#asterisk adjohn (n=adjohn@219.106.248.145) |
03:57.54 | riddlebox | ohh yeah they need to use a union company to do it and AT&T is the only union one around |
03:57.59 | eric2 | what's the competitive price on a PRI these days? |
03:58.10 | riddlebox | eric2, it depends on your area |
03:58.29 | eric2 | ah |
03:58.39 | *** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
03:59.32 | methods | why when i type a number it keeps changing it to sip:<number>@mysipserver |
03:59.38 | methods | ??? i just want it to call a land line |
04:00.22 | [TK]D-Fender | methods, "it"> |
04:00.30 | [TK]D-Fender | ? |
04:00.36 | methods | i tried 2 software app's |
04:00.38 | methods | both do the same thing |
04:00.59 | [TK]D-Fender | methods, so, whats the actual impact of the fact it visually reformats the URI like that? |
04:01.15 | methods | well i want it to call a land line |
04:01.22 | methods | is it in fact doing that ? |
04:01.38 | [TK]D-Fender | methods, You haven't shown us anything and we're not psychic. |
04:01.48 | [TK]D-Fender | methods, PASTEBIN is your friend. |
04:01.54 | [TK]D-Fender | methods, and so is SIP DEBUG |
04:02.00 | methods | dude what in world do you want me to pastebin ? |
04:02.12 | methods | i'm not running the asterisk box |
04:02.15 | methods | i'm connecting to it |
04:03.27 | [TK]D-Fender | methods, if we can't see the SIP communication then we can't tell whats happening now can we? |
04:03.45 | [TK]D-Fender | methods, and if you don't have access to the server then you're really up a creek |
04:04.08 | [TK]D-Fender | methods, it could be misconfigured and I guess you'll never know for sure how or why. |
04:05.24 | methods | Received from: 66.55.150.197:5060 |
04:05.24 | methods | SIP/2.0 404 Not Found |
04:05.24 | methods | Via: SIP/2.0/UDP |
04:05.38 | phix | [TK]D-Fender: any ideas? |
04:05.48 | [TK]D-Fender | methods, pastebin the whole thing please. Many things can say 404 |
04:06.03 | [TK]D-Fender | phix, on? |
04:06.05 | methods | the number i tried to dial was 404 |
04:06.11 | methods | it looked like it had my password |
04:06.23 | phix | [TK]D-Fender: my TDM400p issue |
04:06.29 | phix | [TK]D-Fender: I have tried all ports |
04:06.46 | phix | (I even moved one from the thrid position (the one that wasn't working) to the 4th |
04:06.47 | [TK]D-Fender | phix, pastebin your configs & dmesg |
04:07.16 | phix | http://rafb.net/p/y1l9ih85.html |
04:07.25 | phix | I will do dmesg now |
04:08.02 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
04:08.03 | phix | actually it is 6 lines, I could probably prv msg it to you if you like |
04:08.37 | [TK]D-Fender | phix, nope |
04:08.49 | phix | hmmm ok |
04:10.01 | phix | http://rafb.net/p/HBBIhG95.html |
04:10.35 | AJayMN | is there a way to raise the volume level of g729? i have the phones cranked up but its so quiet compaired to u711 |
04:11.18 | Robba | Hi guys, for some reason unknown to me, my boss wants me to configure our asterisk server to have to dial 0 to get to an outside line, what would be the most simple way to do this? |
04:12.45 | phix | [TK]D-Fender: everything look in order? I didn't forget any important directives? |
04:13.26 | [TK]D-Fender | AJayMN, Nope. |
04:13.39 | [TK]D-Fender | phix, looks fine. you've tried all ports an only 2 work? |
04:17.47 | *** join/#asterisk asr33 (n=asr33@dsl-207-112-72-48.tor.primus.ca) |
04:22.32 | Kobaz | heh |
04:22.44 | *** join/#asterisk shmaltz (n=mybox@mail2.dmaven.com) |
04:22.44 | Kobaz | [TK]D-Fender: you're always here :P |
04:23.24 | phix | [TK]D-Fender: correct |
04:24.19 | [TK]D-Fender | phix, Ok, tell you what, SWAP your modules in place to see if you have a dead one. |
04:25.52 | lmadsen | file: you still fail |
04:27.36 | [TK]D-Fender | Robba, put a 0 prefix on the appropriate extensions in your dialplan. |
04:27.49 | *** join/#asterisk micander (n=Michael_@ip70-181-134-119.sd.sd.cox.net) |
04:29.10 | *** join/#asterisk SteveTotaro (n=Elizabet@c-69-243-124-5.hsd1.md.comcast.net) |
04:30.36 | SteveTotaro | is Tzafrir around ? |
04:31.28 | SteveTotaro | anyone know bristuff? |
04:32.55 | *** join/#asterisk LakeSolon (n=blake@12-202-198-20.client.mchsi.com) |
04:34.15 | *** join/#asterisk jetlagmk2 (i=jetlag@70.17.48.245) |
04:37.54 | phix | [TK]D-Fender: ok |
04:40.40 | *** join/#asterisk ManxPower (n=manxpowe@251.sub-70-223-214.myvzw.com) |
04:46.04 | SteveTotaro | nobody can help with bristuff? |
04:46.54 | *** join/#asterisk erojasv (n=erojasv@201.240.80.159) |
04:51.19 | [TK]D-Fender | SteveTotaro, Europe is closd right now, come back in a few hours |
04:54.58 | drmessano | lol |
04:55.21 | drmessano | Fiber being cut... IT'S THE FINAL COUNTDOWN |
04:56.49 | JT | i wonder who's cutting all the submarine fibre in the middle east |
04:56.56 | [TK]D-Fender | drmessano, about a month or so ago I was inspired to play "Carrie" on piano, learning from the memory of having heard it last a long time ago. |
04:57.10 | drmessano | niice |
04:57.41 | drmessano | According to The Register, it's 3 Fiber cuts |
04:57.47 | drmessano | 4th was a power fail |
04:58.08 | drmessano | and 2 of them are a few km apart at most, likely from the same event |
04:58.15 | drmessano | Iran didn't lose connectivity |
04:58.25 | JT | it's up to 5 fibre cuts |
04:58.36 | drmessano | and with traffic being rerouted, egypt is almost back up |
04:58.37 | drmessano | no |
04:58.39 | drmessano | 3 cuts |
04:58.52 | *** join/#asterisk asr33 (n=asr33@dsl-207-112-48-78.tor.primus.ca) |
04:59.18 | drmessano | http://www.theregister.co.uk/2008/02/07/cut_underseas_cable_conspiracies/ |
04:59.20 | drmessano | There you go |
04:59.55 | husimon | drmessano, i saw 5 |
05:00.03 | drmessano | Nope |
05:00.05 | drmessano | Read the link |
05:00.20 | drmessano | Reports were blown out of proportion and just outright wrong in a few cases |
05:00.33 | husimon | still pretty unsettling |
05:00.57 | drmessano | Coincidence it looks like |
05:01.06 | drmessano | 2 of the cables were from one single event |
05:02.10 | husimon | yeah |
05:02.18 | husimon | crazy coicidence |
05:02.42 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
05:03.12 | drmessano | According to Becket, there's nothing unusual about the number of outages. There are about 100 cut cables every year, enough to keep a fleet of 25 cable repair ships fully occupied. |
05:03.16 | husimon | yeah i saw that |
05:03.17 | drmessano | That says it all, really |
05:03.58 | husimon | reading the stories that said 5 in two weeks in the that area did make me think wtf though |
05:04.01 | *** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au) |
05:04.05 | drmessano | Well |
05:04.09 | husimon | obviously that wasn't the case |
05:04.12 | drmessano | Welcome to the liberal fox news media |
05:04.19 | drmessano | I saw the same things |
05:04.39 | husimon | fox news can jump off a cliff |
05:04.41 | husimon | i hate them so much |
05:05.09 | husimon | i'd really like to see another major network run a story on all the fucked up shit they do |
05:05.13 | husimon | instead of it just being on internet sites |
05:05.33 | drmessano | I remember the plane that had a stuck landing gear above LAX.. They were flying in circles to burn the fuel out so they could crash land with a near empty tank.... |
05:05.51 | drmessano | CNN: Plane with 130 on board in trouble over LAX |
05:06.01 | husimon | the fox news headlines are so messed up |
05:06.04 | drmessano | FOX: 130 prepare to die in horrific LAX airline disaster |
05:06.08 | JT | liberal fox news media? |
05:06.11 | JT | who's joking? |
05:06.20 | husimon | right wing is more like it but... |
05:06.23 | JT | fox news is like a commedy channel |
05:06.37 | husimon | i prefer to watch colbert and the daily show then any news channel |
05:06.38 | JT | i think we only get it on cable here so people can laugh at it |
05:06.53 | husimon | fox news is their #1 content provider |
05:07.34 | husimon | i hate fox news just about as much as I hate ann coulter |
05:07.36 | drmessano | Fox news represents more of modern news media than CNN does.. it's all about sensationalism |
05:07.38 | husimon | omfg that bitch needs to die |
05:07.54 | drmessano | CNN has it's problems too.. |
05:07.58 | drmessano | But Fox.. yeah |
05:08.59 | JT | fox takes a very republican right wing standpoint |
05:09.12 | JT | and it's definitely sensationalist rubbish |
05:10.23 | *** join/#asterisk AndyGraybeal (n=andy@node254.38.251.72.1dial.com) |
05:12.10 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
05:13.01 | drmessano | I really think Fox completely fakes their viewpoints |
05:14.06 | drmessano | I think they do it as an attack on the rest of the media, purely for ratings and entertainment value.. Like "Not Necessarily The News" did on HBO in the 80s lol |
05:14.21 | [TK]D-Fender | drmessano, no, I'm sure they are that #&$^ed up an actually believe that garbage |
05:14.52 | drmessano | That's also impossible to comprehend though |
05:15.04 | drmessano | lol |
05:15.12 | [TK]D-Fender | drmessano, "Never underestimate the power the stupid people in large groups" |
05:15.20 | drmessano | True |
05:16.04 | coppice | all news is faked all the time, as a matter of policy. if they only lied some of the time, you'd easily tell the difference, so they distort everything |
05:16.28 | drmessano | It's impossible to report actual news and make consistant ratings from it |
05:16.33 | drmessano | No doubt |
05:16.55 | coppice | if you've ever been involved in something that got into the news, you'll know the reporting beared only a passing resemblence to what you know actually happened |
05:18.00 | drmessano | I was involved in a disaster situation a few years back that made national news.. it was fascinating what angles they were looking at in questions I was asked |
05:18.45 | drmessano | I remember one 30 minute phone interview I had with a reporter where I don't think i've ever had to try harder to carefully pick out my words |
05:19.15 | drmessano | I came to truly realize why people are so vague and generic at times.. heh |
05:21.19 | coppice | "An inmate at the local asylum got into the laundry, raped all the women, and escaped" |
05:21.20 | coppice | "That's awful. Those poor women. We need a headline that can really capture their plight........ Got it - 'Nut screws washers, and bolts'" |
05:21.30 | drmessano | ROFLL |
05:23.13 | [TK]D-Fender | coppice, comedy GOLD |
05:24.35 | drmessano | If you really want true lulz, read Digg |
05:24.42 | *** join/#asterisk SomethingISOdd (n=TestMast@S010600a0d1757bfb.cg.shawcable.net) |
05:24.44 | drmessano | Now that... is good stuff |
05:24.52 | SomethingISOdd | hello all anyone here use h323 with asterisk |
05:25.07 | drmessano | "Ron Paul endorses Asterisk 1.6!!!!" |
05:28.24 | coppice | Asterisk 1.6. Cindy Crawford's choice |
05:28.53 | drmessano | "Digium defiantly releases Asterisk 1.6 beta" |
05:30.02 | *** part/#asterisk asr33 (n=asr33@dsl-207-112-48-78.tor.primus.ca) |
05:30.17 | drmessano | The trick is, you have to be able to base it in fact.. at some level.. because all nutjobs have a traceable level of fact, however warped it may be |
05:32.54 | drmessano | "Microsoft admits severe flaws in Windows 2000, releases Service Pack 4" |
05:33.10 | drmessano | Uh.. yeah.. ok |
05:33.23 | drmessano | I suppose.. ok.. yep |
05:33.27 | jwh | there is a severe flaw in win2k |
05:33.31 | jwh | they EoL'd it |
05:33.38 | jwh | thats the flaw :P |
05:33.40 | drmessano | lol |
05:35.16 | drmessano | I remember thinking how cool it was 10 years ago that we had an AP Newswire in our buildings |
05:35.28 | drmessano | Now AP is absolutely boring |
05:35.33 | drmessano | Too much "fact" |
05:35.52 | jwh | lol |
05:38.23 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
05:40.02 | BBHoss | anyone using 1.6 with dundi? I seem to be having some trouble peering with someone on 1.4.17 |
05:40.22 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-216-147-93.nsw.bigpond.net.au) |
05:40.35 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
05:41.11 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-74-65-38-18.rochester.res.rr.com) |
05:41.46 | GameGamer43 | has anyone ever seen asterisk pass a call from zap to the phone, but when the phone picks up asterisk doesnt acknowledge it and continues to ring |
05:44.02 | *** join/#asterisk tristanbob_ (n=tristanr@oalug/member/tristanbob) |
05:44.05 | [TK]D-Fender | GameGamer43, what is the "phone"? |
05:44.24 | [TK]D-Fender | GameGamer43, and its be a good idea to pastebin the entire attempt with all appropriate debugs enabled |
05:45.24 | JT | SteveTotaro: i know bristuff |
05:46.02 | GameGamer43 | its not my issue neccessarily, but using cisco phone and tried a softphone |
05:46.02 | GameGamer43 | using tdm400p |
05:46.02 | GameGamer43 | with 4 fxo |
05:46.36 | *** join/#asterisk Malkut (n=Malkut@rrcs-76-79-244-73.west.biz.rr.com) |
05:46.44 | Malkut | i am here |
05:46.46 | GameGamer43 | this issue I described is Malkut's issue |
05:47.08 | Malkut | and what a bad issue it is |
05:47.54 | GameGamer43 | Malkut: please pastebin the cli output of the issue |
05:48.04 | Malkut | which portion would u like? |
05:48.36 | GameGamer43 | from the call coming in to the disconnect |
05:48.43 | Malkut | http://pastebin.ca/index.php |
05:49.09 | GameGamer43 | need your pastebin link |
05:49.23 | Malkut | GameGamer43, http://pastebin.ca/894730 |
05:50.50 | Malkut | you see it pickup the call and do this and that when the call is not really picked up |
05:50.51 | *** join/#asterisk ravichandran (n=Ravi@ip72-206-113-190.om.om.cox.net) |
05:51.32 | ravichandran | any helpful soul around here |
05:51.41 | Malkut | ravichandran, get in line :) |
05:52.03 | ravichandran | I am facing a one way audio problem with my workflow and have been crying without help :( |
05:52.13 | GameGamer43 | what way |
05:52.15 | ravichandran | one way audi with asterisk |
05:52.18 | ravichandran | audio |
05:52.21 | ravichandran | here is the call flow |
05:53.06 | ravichandran | the call comes into the PSTN and from that it hits a switch called CopperCom that has an IPM blade on it. This switch is behind the firewall. The SIP trun is inside the IPM blade. |
05:53.10 | ravichandran | trun=trunk |
05:53.28 | ravichandran | the call then goes to Jasomie which is on the public IP (session border controller) |
05:53.46 | ravichandran | from here the call gets to Cisco Pix firewall and then into our asterisk server. |
05:54.03 | drmessano | wow |
05:54.10 | ravichandran | Our asterisk server plays back an IVR. when the user presses 5 it makes an outbound call using the outbound context |
05:54.20 | GameGamer43 | theres more |
05:54.23 | JT | yeah cisco pix tend to screw sip |
05:54.26 | JT | junk |
05:54.28 | ravichandran | :) |
05:54.33 | drmessano | Yep |
05:54.52 | J4k3 | *bing* |
05:54.53 | J4k3 | ;) |
05:55.07 | J4k3 | (j/k) |
05:55.09 | ravichandran | how do I disable that pix for the time being to make sure that everything goes through that firewall into the server |
05:55.26 | drmessano | We've almost completely done away with Cisco VPN stuff |
05:55.29 | drmessano | Thank goodness |
05:55.31 | GameGamer43 | the simple solution is a bonfire |
05:55.40 | GameGamer43 | just throw the cisco pix in |
05:55.49 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
05:55.58 | [TK]D-Fender | PIX = real trouble for SIP |
05:56.13 | ravichandran | so how do I proceed from here |
05:56.27 | coppice | SIP = real trouble for everyone |
05:56.39 | BBHoss | hey anyone here using the DUNDi-test network or using DUNDi period? There aren't any good docs for it |
05:56.57 | GameGamer43 | [TK]D-Fender: http://pastebin.ca/894730 is the pastebin you asked for and all questions should go to Malkut |
05:57.15 | GameGamer43 | BBHoss: I found one good doc for dundi some time back, let me see if I can dig it up |
05:57.31 | Malkut | <<<< |
05:57.45 | ravichandran | can I disable the cisco pix for right now to just act like a dummy guy for right now |
05:58.03 | BBHoss | GameGamer43, thanks, i need a working doc because I believe i'm experiencing a bug in 1.6 that needs to be reported |
05:58.31 | ravichandran | if I can any leading tips would be appreciated |
05:58.42 | GameGamer43 | BBHoss: what is your setup and are you trying load balancing or no |
05:58.55 | [TK]D-Fender | GameGamer43, [s@macro-dial:10] Dial("Zap/1-1", "SIP/200|15|tr") in new stack <- "r" = evil |
05:58.56 | [TK]D-Fender | and |
05:58.58 | [TK]D-Fender | ~freepbx |
05:58.59 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
05:59.01 | BBHoss | no, i'm trying to join a peering network, but i can't get it to work |
05:59.22 | ravichandran | drmessano are u there |
05:59.23 | GameGamer43 | BBHoss: you look at this http://blog.thegoldfish.net/dundi-tutorial-for-asteriskhome/ |
05:59.25 | drmessano | yes |
05:59.34 | Malkut | GameGamer43, :( |
05:59.45 | GameGamer43 | BBHoss: that one of the better guide I've ever found |
05:59.56 | GameGamer43 | Malkut: i didnt forget about your issue yet |
05:59.58 | GameGamer43 | lol |
06:00.47 | Robba | does anyone know if its possible to remove the need to press dial on a linksys SPA-941 to make a call? |
06:01.06 | drmessano | Proper dialplan |
06:01.16 | Robba | proper? |
06:01.17 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
06:01.42 | drmessano | Whats your dialplan on the phones now? |
06:02.35 | Robba | 10 and 8 digit dialling, and now that you mention it, it doesn't seem to dial after typing the 10 digits.... hmmm back to the drawing board |
06:03.07 | drmessano | Why not just load something generic? |
06:03.12 | drmessano | Let Asterisk do the rest |
06:05.44 | [TK]D-Fender | only reason to need send is a bad phone dialplan |
06:07.25 | drmessano | (*x.|x.) |
06:07.36 | drmessano | and let Asterisk do the real work |
06:14.24 | *** join/#asterisk AJayMN (n=mypocket@76.201.155.119) |
06:19.25 | AJayMN | I upgraded to 1.2.26.1 last night and now my wakeup and weather isnt working anymore.. what could have been broke? |
06:21.49 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
06:21.54 | [TK]D-Fender | AJayMN, Any number of things, and none of those are ASTERISK. |
06:22.06 | AJayMN | ok |
06:22.20 | AJayMN | figured someone else may have had the same issue at some point |
06:23.32 | [TK]D-Fender | AJayMN, With no details of any kind. Not likely. |
06:23.44 | drmessano | AJayMN.. you're asking in the wrong place :) |
06:24.42 | [TK]D-Fender | ok, I'm off, later all |
06:24.50 | drmessano | later |
06:28.32 | Malkut | lol |
06:28.33 | Malkut | thnx |
06:30.59 | *** part/#asterisk AJayMN (n=mypocket@76.201.155.119) |
06:37.28 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
06:39.14 | BBHoss | ~seen bkw |
06:39.18 | jbot | bkw <n=bkw@h-235-0.A189.cust.bahnhof.se> was last seen on IRC in channel #debian, 10h 21m 42s ago, saying: 'valdyn: apt-cache search apache2 ssl suggest several packages. But which one installes the modules'. |
06:49.05 | AndyGraybeal | anyone famillar with a ron paul phone call project on the web.. i swear it was either here or #freeswitch ... but they don't think so |
06:49.13 | AndyGraybeal | or somewhere else.. another offshoot of asterisk |
06:49.30 | BBHoss | callweaver? |
06:49.40 | AndyGraybeal | i don't think so |
06:50.05 | J4k3 | ~ron paul |
06:50.06 | jbot | ZOMG RONPAULAPPLEUBUNTU FOR PRESIDENT |
06:51.33 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
06:51.34 | drmessano | Maharishi University of Management |
06:51.43 | drmessano | I bet they don't teach telecom |
06:51.49 | drmessano | chan_om |
06:54.26 | drmessano | Ron Paul? |
06:54.41 | drmessano | Ron Paul phone call project... what is that? |
06:55.25 | J4k3 | 00:50 < J4k3> ~ron paul |
06:55.25 | J4k3 | 00:50 < jbot> ZOMG RONPAULAPPLEUBUNTU FOR PRESIDENT |
06:55.40 | J4k3 | phone spam politics |
06:55.44 | drmessano | Oh |
06:56.05 | drmessano | That ZOMG line is mine, btw.. I have trademarks on it now |
06:56.12 | drmessano | Matter of fact, consider yourself sued |
06:56.22 | J4k3 | ron paul should pony up some cash and hire indians to do it |
06:56.26 | drmessano | lol |
06:56.27 | drmessano | HAH |
06:56.38 | drmessano | HALO, MY NAME IS.... JOE.. |
06:56.44 | J4k3 | I am from new york! |
06:57.21 | drmessano | Ron Paul has Digg.. who cares if they're all too young to vote |
06:57.46 | J4k3 | wtf is digg |
06:57.49 | J4k3 | is it like facebook? |
06:57.55 | drmessano | OMG |
06:57.58 | J4k3 | something else for fake-nerds to do? |
06:58.00 | drmessano | Digg.com is ummm |
06:58.09 | drmessano | It's a "news" site |
06:58.14 | J4k3 | ahh |
06:58.16 | J4k3 | its one of those |
06:58.18 | drmessano | It's like an unmoderated Fark |
06:58.20 | J4k3 | I hate the internet |
06:58.21 | J4k3 | I just sell it |
06:58.35 | J4k3 | I don't get high off my own supply |
06:58.42 | drmessano | Everything is "BREAKING NEWS: ZOMGGGGG" |
06:58.56 | J4k3 | and then you sue them for ZOMG abuse? |
06:59.29 | *** join/#asterisk steliosk (n=Stelios@85.75.198.88) |
06:59.29 | drmessano | "NEWS FLASH: APPLE TO ADD AN EXTRA 1GB TO MACAIR 2 !!!!ELEVEN!!!" |
06:59.46 | drmessano | Thats pretty close to a Digg headline |
07:00.09 | J4k3 | haha damn |
07:00.14 | J4k3 | thats some lame people |
07:00.20 | drmessano | It's hilarious |
07:00.22 | J4k3 | but not suprising, the internet lacks :| |
07:00.32 | drmessano | Apple fanboys, ron paul fanboys, linux fanboys |
07:00.48 | J4k3 | theres all these places with info, but nobody has any real info |
07:01.09 | J4k3 | an internet-load of reporters and nobody out making news |
07:01.10 | drmessano | "BREAKING NEWS: UBUNTU TO ADD UPDATED VERSION OF NANO TO 9.09 BUSTY BEAVER!!!!!!!!ONES!!!1111!!!2!!" |
07:01.24 | J4k3 | busty beaver... now *thats* my release. |
07:01.41 | coppice | you like 'em furry? |
07:01.46 | drmessano | lol |
07:02.05 | J4k3 | beggers can't be choosers. |
07:02.38 | *** join/#asterisk IPGHOST (i=IPGHOST@202.142.153.55) |
07:02.41 | drmessano | I should start a Digg-like Asterisk news site |
07:03.15 | J4k3 | ZOMG CHAN_SPOOGE GETS GISM CONNECTIVITY TO ASTERISQE |
07:03.40 | drmessano | "BREAKING NEWS: AT&T ADMITS ASTERISK IS HURTING THEIR BUSINESS!!!!" |
07:03.57 | sbingner | lol |
07:03.59 | drmessano | Link to some article about AT&T embrasing VoIP |
07:04.03 | J4k3 | BREAKING NEWS: Level 3 still sucks! |
07:04.28 | drmessano | "AVAYA LOSES MILLIONS DUE TO SUCCESS OF ASTERISK" |
07:04.37 | J4k3 | AVAYA GOES AWAYA |
07:04.45 | drmessano | No, those arent lame enough... hmmm |
07:05.01 | denon | breaking news: avaya is running asterisk on their new units |
07:05.04 | drmessano | Ha, got it.. |
07:05.13 | BBHoss | ABE at that :) |
07:05.16 | denon | (and not releasing source) |
07:05.24 | denon | <rumors/> |
07:06.02 | drmessano | "1283 REASONS TO UPGRADE FROM ASTERISK 1.2 TO 1.^ (BEFORE IT'S TOO LATE!!!)!!!!" |
07:06.09 | drmessano | 1.6* |
07:06.27 | drmessano | There has to be some drama |
07:06.36 | J4k3 | BREAKING NEWS: UPGRADE TO ASTERISK 1.6 FOR Y10K COMPLIANCE |
07:06.39 | denon | 1) we'll be taking the upgrade script off the servers tomorrow! |
07:06.44 | drmessano | lol |
07:06.52 | Corydon76-dig | Uh, all it takes for Avaya to lose millions (in sales) is for them to fail to get a single contract |
07:06.55 | denon | upgrade tonight or lose all your settings! |
07:07.23 | Corydon76-dig | That's less about other people coming on the market than it is about how massively expensive Avaya equipment is |
07:07.38 | denon | of course, small countries run their telcos on avaya.. |
07:08.00 | drmessano | "BREAKING NEWS: IRAN TO START USING ASTERISK" |
07:08.25 | denon | Dial(IAX2/iran/terrorist-queue) |
07:08.28 | J4k3 | http://www.whoopis.com/howtos/telco-basics/beirut-phone-wiring.jpg |
07:08.34 | coppice | the US army in Iraq is using Asterisk |
07:08.36 | J4k3 | ^^ awaiting asterisk upgrade |
07:08.47 | J4k3 | also, need GPS locator for cat |
07:08.54 | drmessano | lol |
07:09.11 | drmessano | cat /pees/interrupts |
07:09.28 | J4k3 | if I was a cat and saw that phone box |
07:09.31 | J4k3 | I'd spray it, twice. |
07:10.14 | drmessano | The one thing you'll find about Digg is that Apple does NOTHING wrong |
07:10.22 | denon | coppice: I set up some asterisk gear in iraq for soldiers to call their families at home |
07:10.33 | drmessano | The headline may not support it, but the comments will |
07:10.39 | denon | just a simple bridge back to some some LD trunks we did for free on the US side |
07:10.53 | denon | on a guy's hardware over there |
07:10.54 | drmessano | denon: that's awesome |
07:11.00 | coppice | denon: they are doing more with it now, it seems |
07:11.10 | J4k3 | nice, asterisk is good stuff. |
07:11.12 | J4k3 | flexible, easy, etc. |
07:11.16 | denon | that was a very long time ago, I dont think its in use anymore, 'cause those divisions came home |
07:11.31 | denon | I should disable it so nobody abuses the free LD heh |
07:11.32 | J4k3 | hell, trixbox can be configured by the average shop mechanic. |
07:11.41 | drmessano | ahmadinejad loves Asterisk |
07:11.44 | denon | J4k3: trixbox is really only good for a shop mechanic :) |
07:11.49 | drmessano | lol |
07:11.53 | coppice | denon: well, someone is setting up something right now, needing MFC/R2 interconnect |
07:12.06 | drmessano | My cat loves Trixbox.. she can set one up in an hour |
07:12.09 | denon | ah, yeah, doubt that's my rig heh |
07:12.12 | J4k3 | denon: exactly, but worse than avaya?! :D |
07:12.23 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
07:12.25 | J4k3 | my cat knows what to use trixbox for |
07:12.32 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
07:12.37 | J4k3 | *scratch scratch scratch* |
07:12.40 | denon | trixbox works fine for some stuff .. just don't forget to change the litter |
07:12.47 | drmessano | lol |
07:13.19 | drmessano | echo 127.0.0.1 *.fonality.com >> /etc/hosts |
07:13.56 | denon | yeah, the nice thing about trixbox is that it's a truly open system .. all your data, system details, usage info, etc .. totally open for fonality to do whatever they want with it |
07:14.06 | drmessano | lol |
07:14.08 | drmessano | Yep |
07:14.22 | drmessano | "Anonymous usage data" |
07:14.34 | denon | those threads are entertaining |
07:14.42 | denon | at the beginning, they claim the processes don't exist |
07:14.53 | denon | then they claim it's not personal data |
07:15.03 | denon | then they claim it's optional, and well-documented |
07:15.06 | drmessano | I was called a TROLL.. that wasn't entertaining to me :( |
07:15.32 | denon | J4k3: two grids, 2 LP or diesel backup gens, ASCO transfers |
07:15.32 | drmessano | "We could have hidden it much better, but we decided not to" |
07:15.36 | denon | yeah |
07:15.46 | drmessano | That was my argument |
07:15.46 | J4k3 | theres two ways I see to build it... high capacity storage, or low capacity storage with redundant generation... leaning toward the second. |
07:15.55 | J4k3 | denon: not possible here. |
07:15.58 | drmessano | "We could have hidden it much better, but we decided not to" and "We didn't think anyone would be upset" |
07:16.02 | denon | J4k3: you know, LP is *really* nice for long-term storage |
07:16.10 | J4k3 | yeah, we've got LP onsite already |
07:16.12 | denon | engines burn clean, fuel doesn't get goofy |
07:16.14 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
07:16.19 | drmessano | If they didnt think anyone would be upset, they never would have NEEDED to discuss hiding it |
07:16.20 | J4k3 | I keep 100 gallons onhand at all times |
07:16.23 | denon | and it's easy to keep a week or more onsite |
07:16.27 | denon | add a 0 :) |
07:16.30 | J4k3 | what I love about LP on smaller engines is the fact you *can't* flood it |
07:17.02 | denon | nod |
07:17.06 | J4k3 | I mean you can, but its not a wet flood, cranking it over a couple turns with the throttle open will clear it |
07:17.12 | denon | you wouldn't know of such things, but it also runs pretty well in cold weather |
07:17.21 | J4k3 | yeah |
07:17.23 | denon | though our gen gear is indoors anyway |
07:17.33 | J4k3 | can't always trust indoors to be warm. |
07:17.45 | denon | well, we keep oil warm, as well as battery |
07:17.57 | drmessano | We've had a few problems with the tanks when they get below about 40% |
07:18.20 | denon | drmessano: yeah, some engines require a certain column inch volume .. |
07:18.37 | denon | depends on your tank size, type, etc |
07:18.38 | drmessano | Yeah.. It was a pain in the ass during the last few ice storms |
07:18.47 | denon | you have an upright tank? |
07:18.54 | drmessano | no |
07:19.08 | denon | your gen burn vapor or liquid? |
07:19.28 | J4k3 | I'm thinking I could probably live with about 5kW of generation capacity |
07:19.41 | drmessano | I'm not the expert.. but that was the issue.. I believe it burns vapor |
07:19.52 | drmessano | and we were talking about putting something in line to change that |
07:20.04 | J4k3 | the lowest temps we see here are -5F, should I worry? |
07:20.04 | denon | well, vapor isn't bad .. |
07:20.09 | denon | no |
07:20.23 | denon | vapor's not bad, but you need a large enough tank |
07:20.27 | drmessano | yes |
07:20.35 | drmessano | I remember that |
07:20.40 | sergee | any gprof experts here? |
07:20.42 | denon | it's not so much the size, but the amount of airspace |
07:20.57 | drmessano | We got two much larger tanks for next time |
07:21.21 | drmessano | Biggest issue was keeping them topped off |
07:21.25 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
07:21.27 | denon | 1,000 gal tanks are usually a good match for a couple decent gens to run a long time |
07:21.49 | J4k3 | debating 12V or 24V system... 12V is nice due to the quantity of cheap alternators available |
07:22.08 | denon | J4k3: why not just pump 480 or 240 or whatever into your mains? |
07:22.14 | J4k3 | thinking of maybe a small 4 cylinder car engine set up on a sled.. I've helped build something similar before |
07:22.19 | J4k3 | denon: cost, lack of need. |
07:22.20 | denon | sure, DC is more efficient, but.. |
07:22.28 | denon | nice to have your workstation when the lights go out :) |
07:22.31 | J4k3 | denon: well, it'll get inverted. |
07:22.38 | J4k3 | my workstation is a laptop |
07:22.48 | J4k3 | the only thing I miss is my desk lamp... and I plug it into the inverter now ;) |
07:22.53 | denon | hehe |
07:23.01 | denon | you and Edison would get along well |
07:23.18 | denon | that darn Westinghouse and his silly A/C nonsense |
07:23.19 | BBHoss | anyone here using adhearsion? |
07:23.23 | J4k3 | no |
07:23.40 | J4k3 | I appreciate AC a lot :) |
07:23.48 | J4k3 | DC scares the holy crap out of me, thats why I refuse to play with 48v |
07:23.56 | J4k3 | 24v scares me pretty well |
07:23.56 | denon | hehe |
07:24.09 | denon | eh, why? |
07:24.12 | J4k3 | I've been hit with everything 12v had to offer... it stung like hell but I suffered no damage |
07:24.26 | J4k3 | electricution by DC is nasty |
07:24.30 | J4k3 | you jam up and won't let go |
07:24.32 | J4k3 | AC slings you out |
07:24.44 | denon | I'm pretty sure the electric chairs runs on AC .. |
07:24.51 | J4k3 | yeah |
07:24.55 | J4k3 | stepped up |
07:25.02 | denon | course you're a little constrained in that. . |
07:25.13 | J4k3 | texas's first electric chair ran on a generic home pole pig wired backwards for step-up |
07:25.32 | denon | did you know that Edison was lobbying the govt to call that process "getting Westinghoused"? |
07:25.54 | drmessano | Bastard killed an elephant |
07:25.55 | denon | 'cause he was trying to get the public to fear A/C |
07:26.48 | J4k3 | but yeah, the human body can sink a lot of current/heat |
07:26.51 | J4k3 | before it fails |
07:26.57 | J4k3 | so yeah, the electric chair has to be pretty brutal |
07:27.14 | J4k3 | and most electric chairs are pretty darned current limited |
07:27.26 | J4k3 | they wanna make sure it looks good. |
07:27.26 | denon | The Electric Chair: Like Cisco Call Manager, with a seatbelt! |
07:27.27 | J4k3 | :| |
07:27.30 | denon | there's a topic .. |
07:27.37 | J4k3 | haha owned |
07:27.56 | sergee | "adterisk and electric chair" - friday 7/8 central on ABC |
07:28.08 | sergee | s/adterisk/asterisk/ |
07:28.11 | drmessano | "BREAKING NEWS: ZOMG, CISCO KILLS KITTENS!!!!!!" |
07:28.16 | denon | yeah, going from generators to electric chairs, perhaps we're a little off topic |
07:28.31 | denon | but, cisco does kill kittens |
07:28.35 | denon | I read about it in a press release |
07:28.39 | sergee | guys, please , enlight me |
07:28.40 | J4k3 | denon: see, my idea is run it off an inverter |
07:29.07 | drmessano | "Cisco Call Manager is powered by the tiny hearts of slaughtered kittens :(((((" |
07:29.10 | sergee | what is "central US time zone" - what is the offset from GMT |
07:29.15 | J4k3 | sit a relay with time delay between the mains and my gear... basically let it fall onto the inverter immediately (standby UPS style) but delay (or wait for me to do it manually) switch back to mains |
07:29.20 | denon | sergee: -6 |
07:29.22 | denon | depending on DST |
07:29.42 | J4k3 | likely a 5k sustained (8-10k peak) inverter |
07:29.58 | denon | J4k3: why not just toss an asco in? |
07:30.13 | J4k3 | asco? |
07:30.16 | drmessano | Onan FTW |
07:30.20 | denon | yeah, brand of transfer switch |
07:30.21 | sergee | denon: thanks and one more question, how to read times 8/9, 7/8 ? |
07:30.27 | J4k3 | onan costs a lot, but their stuff is nice |
07:30.27 | denon | it's UL listed |
07:30.32 | J4k3 | more than I want to spend |
07:30.34 | denon | sergee: 9, 8 central .. the 9 is eastern |
07:30.37 | drmessano | We use all Onan.. good stuff |
07:30.51 | denon | er .. eastern? I dunno, some timezone |
07:30.51 | J4k3 | denon: ahh... a proper transfer switch would work... but the idea is there'd always be AC available from the inverter from battery |
07:31.05 | J4k3 | as soon as theres an AC outage event, crank the motor and start generating DC |
07:31.08 | sergee | denon: so 9/8c == 21:00 east coast, 20:00 central, am i right? |
07:31.15 | denon | somethin like that yeah |
07:31.16 | J4k3 | so you get nice clean 'electronically generated' AC |
07:31.24 | J4k3 | instead of whatever a cheap-ish generator feels like generating |
07:31.53 | denon | J4k3: well, I dunno, good gens usually have the engine and generator tuned pretty well to eachother |
07:32.04 | denon | I see a pretty consistent 59hz over a long time |
07:32.11 | denon | 59.8-60 or something |
07:32.33 | denon | biggest problem I see, is when people underpower their gen |
07:32.36 | denon | and the engine pulls hard |
07:32.42 | denon | a poorly designed gen will just slow down |
07:32.47 | denon | (and obviously cycles go with it) |
07:32.49 | J4k3 | well, I don't want to spend $5k on a $1k problem |
07:32.50 | J4k3 | yeah |
07:33.08 | denon | sounds like you want lots of batteries from the telco |
07:33.11 | J4k3 | the problem I've seen is when the generators themselves start failing... good units catch it and unload... bad units destroy stuff |
07:33.32 | J4k3 | thats what happened with my last generator...controller went nuts after about 3 hours, ate a few dozen psus. |
07:33.33 | denon | yeah |
07:33.39 | denon | ug |
07:33.46 | denon | UPSs didn't catch? |
07:34.09 | J4k3 | they tried |
07:34.17 | denon | on, off, on, off |
07:34.19 | J4k3 | they didn't fail, oddly enough... they ate hardware behind them |
07:34.23 | J4k3 | IDidn't trust them anymore, though |
07:34.24 | J4k3 | yeah |
07:34.35 | denon | gotta set the thresholds really tight |
07:34.36 | denon | sucks |
07:34.47 | denon | gotta force em to just go hard to battery if things aren't perfect |
07:34.55 | J4k3 | yeah |
07:35.09 | J4k3 | but what I'm thinking is... if I run a 12V system off battery |
07:35.15 | J4k3 | I can charge it off pretty much anything |
07:35.28 | J4k3 | currently I use a car in case of long term outage. it works. |
07:35.33 | J4k3 | short term we just run off battery |
07:35.53 | denon | you can rip the inverter/charger out of a UPS.. |
07:36.02 | J4k3 | yeah |
07:36.13 | denon | with adequate cooling, I'm told those things will push a lot more than they rate em at |
07:36.16 | J4k3 | I could get a decent sized UPS and throw a lot of battery behind it |
07:36.30 | J4k3 | you can usually run them a lot longer than their rating |
07:36.33 | *** join/#asterisk ik_5 (n=ik@85.64.203.142.dynamic.barak-online.net) |
07:36.39 | J4k3 | a lot of cheap ones overheat due to lack of heatsink area |
07:36.42 | denon | if you cool em yeah |
07:36.44 | J4k3 | if you over-battery them |
07:36.51 | J4k3 | but yeah, if you use non-crap UPSes and fans, no problem |
07:36.51 | denon | and of course, make sure you vent your hydrogen ;) |
07:37.03 | denon | assuming you're probably going to use truck batts or something |
07:37.10 | drmessano | This sounds like a job for a honda generator and an APC power conditioner :) |
07:37.42 | denon | drmessano: or a symmetra, and a pallet of old batteries from his local scrap yard :) |
07:37.44 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
07:37.51 | drmessano | heh |
07:37.57 | J4k3 | thats why god made tractor supply company.... for cheap plastic truck tool boxes (Battery boxes) and 2 gauge stranded heavy duty 'welding wire' by the foot |
07:38.13 | denon | hehehe |
07:38.38 | J4k3 | you gotta go crazy with the wire when you're running inverters off 12V |
07:39.02 | adeel | would an echo canceller running on an FXO port cause problems with faxing? and if so, would setting the faxdetection options in zapata help? |
07:39.19 | denon | adeel: most people disable echo can on native bridge fax stuff |
07:40.00 | adeel | denon, alright, i'll try making sure that the echo cancel on bridged setting is properly set |
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07:40.46 | adeel | hmmm, it's already set to no |
07:41.35 | denon | take a look at the channel while it's up |
07:41.44 | denon | during a fax or whatever |
07:41.56 | adeel | using ztmonitor? |
07:42.02 | denon | nah, just in the console |
07:42.25 | adeel | well the output seems to be normal, i can see the zap/8 channel connect to zap/g0 and the call being passed |
07:42.32 | adeel | but for some reason, transmission fails |
07:43.07 | denon | are you looking at: core show channel zap/123-1 ? |
07:43.31 | adeel | well, i'm running * 1.2 still on this box |
07:43.43 | denon | well, sans the core |
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07:44.31 | adeel | which output should i be looking for...Echo Cancellation? |
07:45.55 | denon | it'll show if it's on or not during the call |
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07:46.58 | denon | you'll see something like Echo Cancellation: 128 taps, Currently Off |
07:47.23 | denon | that's realtime, during the call |
07:47.29 | adeel | yeah, i see that...but to make things more complicated, i'm using a commerical echo canceller (octasic softecho suite) because the built-in routines weren't cutting it |
07:47.37 | denon | Ive gotta bail, see you guys later |
07:47.42 | adeel | thanks, see ya |
07:47.49 | denon | ah, yeah, you'd need to check with them then, I dunno anything about it |
07:47.58 | denon | you might also play with digium's HPEC |
07:48.14 | denon | it's free on cards currently under a year old or something |
07:50.09 | denon | http://www.digium.com/en/products/software/hpec.php |
07:50.11 | denon | anyway |
07:50.12 | denon | audios |
07:56.04 | ik_5 | hello, I have an asterisk box wich is a blackbox for me, that is I do not have access to the internals, it has few fxo and fxs, and I'm connected using ethernet (iax/sip), I wish to send a fax using the machine with the iax/sip throu the fxo, what components on the client side (linux btw) i require to have in order to achive the sending of a fax ? |
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08:00.12 | JT | ik_5: it can't be done reliably like that |
08:01.29 | ik_5 | JT, what does it mean ? |
08:01.43 | DavisGr | ik_5 are U have t.38 support also ? |
08:02.27 | ik_5 | DavisGr, no |
08:03.07 | DavisGr | poor, but this is good aplication http://news.asteriskguru.com/2/9220/2008/1/9/Zoiper_softphone_now_supports_T.38_Faxing |
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08:05.28 | DavisGr | try to find some fax over sip software / or buy some Linksys ATA fxs to ethernet then U can use your faxmachine but if U have large network then U will lose some fax's |
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08:06.58 | ik_5 | thanks |
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08:13.08 | JT | ik_5: i mean it probably won't work |
08:13.12 | JT | especially with analogue cards |
08:14.37 | ik_5 | JT so i must have T1 support as well ? |
08:14.53 | JT | ideally |
08:15.00 | JT | and faxing isn't that reliable in asterisk anyway |
08:15.06 | JT | and faxing isn't reliable over VoIP |
08:15.14 | Frogzoo | anyone like to recommend a linux sip client? |
08:15.32 | ik_5 | Frogzoo, twinkle |
08:15.37 | Frogzoo | ik_5: thanks |
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08:24.59 | ik_5 | is there a way to use hylafax and iaxmodem to send the fax over iax to the fxo ? |
08:25.56 | DavisGr | it's depend how to make your system - if U will use hylafax then U can send fax over fxo usin hylafax print driver or using emails |
08:27.07 | ik_5 | the probelm is, that i can't install anything on the asterisk with the fxo... |
08:27.58 | ik_5 | so i must do things on the client side |
08:28.22 | DavisGr | If U cant install anything then U can try to use Linksys ATA - but it fill work if U will not have large network from ATA to pbx |
08:29.02 | DavisGr | but then U will need sip accaunt/extension for this ATA |
08:29.31 | ik_5 | it's very small network, i can even make it almost direct connection ... |
08:29.46 | DavisGr | then it will work |
08:30.32 | DavisGr | I have some cases where is one swich behind and its working |
08:31.14 | DavisGr | The moust important is network load if U have Data & VoIP network the same |
08:31.44 | DavisGr | dont Use p2p software like torents dc++ etc. |
08:32.13 | ik_5 | this is a dedecated network, so no one will use it :) |
08:32.34 | DavisGr | then fine it will work! |
08:32.57 | ik_5 | thanks, i'll look at the ata |
08:35.38 | DavisGr | Linksys SPA-1001 1xFXS , Linksys PAT2T - 2xFXS but for them U will need use the fax machine ofcourse! |
08:35.47 | JT | DavisGr: is it that hard to type "you"? ;) |
08:36.05 | J4k3 | joo |
08:36.29 | JT | also, saying it WILL work is a big too optimistic |
08:36.34 | JT | it "might" work |
08:37.33 | DavisGr | JT bouth end of last year clients from one voip commpany there was such cases and its work |
08:38.13 | DavisGr | For faxs I prefer callweaver becouse he support T.38 |
08:38.42 | JT | yes, but you definitely can't guarantee it will work for him, especially if he;s using analogue digium cards |
08:40.07 | DavisGr | where is problem width analog digium cards - there in market is also copies of digium and they also work! |
08:41.34 | DavisGr | JT for 100% I cant guarantee but I can say I had solutions where in that way working! |
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08:49.05 | JT | DavisGr: digium analogue cards are not good at faxing |
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08:52.12 | scooby2 | Whats the best way to debug why zaptel or the te412p module is kernel panic'ing centos? mentions smp.c |
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08:58.14 | DavisGr | JT what is the best card for analog lines |
08:58.42 | DavisGr | scooby2 witch kernel you have |
08:58.56 | JT | DavisGr: a real fax machine or fax modem |
09:00.00 | DavisGr | OK :) but if you need handle inbound and outbound calls via asterisk |
09:00.17 | DavisGr | with the same line |
09:00.20 | JT | do what sensible people do |
09:00.33 | JT | and avoid it unless you have digital or a high end card |
09:01.27 | J4k3 | efax! |
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09:02.00 | DavisGr | wi JT |
09:02.45 | JT | ? |
09:03.33 | J4k3 | vi JT |
09:03.47 | ik_5 | Jt, so maybe you'll have a different solution for the problem, I have to update 30 people with information regarding things they bought, or things they requested to know, and they only use faxes... how would you computerize this without having a modemfax ? |
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09:05.37 | JT | ik_5: buying an analogue line is probably the easiest |
09:06.08 | ik_5 | JT, at this time I'm trying to use only things (in hardwre) I already have |
09:06.17 | DavisGr | JT it's depend of situation where are you! |
09:07.44 | DavisGr | <PROTECTED> |
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09:08.34 | ik_5 | DavisGr, it's a flash based pbx so any change can destroy things... |
09:09.00 | DavisGr | to bad |
09:09.39 | ik_5 | yes, the problem is that I do not have a lot of control over things... other wise it would have been less chanlnging and "too easy" ;) |
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09:16.03 | Frogzoo | does fax on analogue digium always have problems? |
09:16.22 | JT | mainly on the single and quad port card |
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09:17.01 | Frogzoo | JT: hardware software or driver issues? |
09:17.17 | the_5th_wheel | Hi. Can anyone reccommend me a billing system, wich just log wich calls are made from wich sip user? |
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09:18.24 | scooby2 | DavisGr: centos 5.1 - 2.6.18-53.1.6.el5 |
09:21.45 | FlatFoot | morning all |
09:21.47 | DavisGr | scooby2 I was the problem last summer width enabled smp on slackware width 2.6.16-20 kernels when smp switching off then things start work but on latest kernel versions I havent problems |
09:21.59 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-2c66359af2644e4e) |
09:22.21 | FlatFoot | can anyone help with iax2 one way breakdown probs ? http://www.pastebin.ca/894824 shows a bit of the debug |
09:23.00 | FlatFoot | i have got jitterbuffer turned on and it is a trunk |
09:23.42 | FlatFoot | both ends are running 1.4.17 |
09:25.29 | JT | Frogzoo: hardware |
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09:27.51 | jeremy_g | oej, oej!! |
09:27.54 | Frogzoo | JT: ugh, that kinda sux |
09:28.17 | oej | jeremy_g: Jeremy, jeremy! |
09:28.22 | jeremy_g | :) |
09:28.47 | jeremy_g | I have to put asterisk in front of this IMS (CSCF) with a big operator |
09:29.21 | jeremy_g | I am making a feasibility if asterisk would be usable in this case. I am in sweden ;) as you know. |
09:29.42 | Sajjad_Ali_Musht | anyone knows "how to capture the QoS parameters from SIP, ZAP and CAPI channels for completed calls" |
09:31.20 | jeremy_g | oej: Would the IMS specific headers like P-Asserted Identifity, Visited Network Id, "reg event" confuse asterisk? |
09:32.34 | scooby2 | DavisGr: thx |
09:33.01 | scooby2 | can anyone point me to a 1.4 IVR example? trying to upgrade from 1.2 and it is not going well. |
09:33.32 | ik_5 | DavisGr, JT, thank you for the help, bye |
09:34.24 | oej | jeremy-g: Not confuse, just be ignored |
09:40.21 | synthetiq | where can i find the text2wave application? |
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09:46.43 | mvanbaak | synthetiq: festival |
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09:52.23 | yang | I wonder has anyone been using munin plugins for graphing asterisk processes? |
09:54.23 | matlads | hello |
09:54.31 | Frogzoo | yang: cacti is nicer than munin or mrtg |
09:54.39 | Frogzoo | imo |
09:54.44 | matlads | one question, can I do a SET VARIABLE in a hangup script? |
09:55.54 | yang | no, cacti is a pain to setup |
09:56.11 | yang | snmp!"#$WR |
09:56.55 | XnOSX | have any probs with asterisk and kernel 2.6.24-1-686? |
09:57.29 | jeremy_g | oej:it would ignore them even if these are additional headers within a proper sip message? |
09:58.24 | jeremy_g | oej:you have read the TS 24.229 right? You know that these are just really sip headers within the regular SIP methods like INVITE, BYE etc. |
09:58.44 | jeremy_g | oej:so would an additional header like above cause the * to ignore the whole properly formatted SIP? |
09:59.34 | jeremy_g | yang:cacti is not a pain to setup, we run it here and its a breeze |
09:59.41 | jeremy_g | Frogzoo:yup |
10:04.37 | jeremy_g | oej:reply me |
10:04.51 | oej | Was offline... |
10:05.15 | oej | No, we just ignore the headers |
10:05.21 | oej | I can't remember TS 24.229 - URL? |
10:05.36 | jeremy_g | http://www.3gpp.org/ftp/Specs/html-info/24229.htm |
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10:08.49 | oej | jeremy_g: If you need any work done to make asterisk more compliant with this, you know where to find me :-) |
10:09.34 | nixguy | any nice webgui tools for monitoring asterisk calls n stuff with? |
10:12.13 | jeremy_g | oej:gimme ur contact and i dont even know if there is a need for more work to be done. |
10:13.36 | jeremy_g | oej:my understanding is that * wud still process the invites coming in from cscf no matter what additional headers are there, both wud happily talk sip as long as cscf does not expect * to talk those additional headers. |
10:14.38 | tzafrir_home | XnOSX, you'll probably need zaptel from svn at this point |
10:14.42 | mvanbaak | jeremy_g: you can add extra headers from the dialplan |
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10:19.56 | jeremy_g | mvanbaak:yup i know |
10:21.58 | oej | jeremy_g Yes, that's right |
10:23.39 | XnOSX | hey friends! what is the best kernel for asterisk |
10:23.40 | XnOSX | ? |
10:23.48 | XnOSX | 2.6.18? |
10:24.09 | jeremy_g | :D |
10:24.19 | jeremy_g | Col. Alfred Douglas |
10:27.11 | mvanbaak | XnOSX: any 2.6 that you are comfortable with |
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10:32.28 | L|NUX | hello |
10:32.47 | L|NUX | can some one help me with MoH mms stream ? |
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11:07.13 | sergey | Hi. Is callcentr on * can "speak" number into line and/or how mach time till answers by op ? |
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11:22.01 | Sweeper | anyone know how i can see some logs on the aastra 9133i? |
11:23.24 | Sweeper | having some issues getting it to register, and don't have access to the sip server |
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11:41.44 | synthetiq | where can i find the text2wave application? |
11:42.23 | synthetiq | because app_festival is built but i cant find the script or binary for text2wave |
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11:53.28 | tnt_ | Hi everyone. I have an issue with a B410p. Just to be sure, the leds must be green even without asterisk launched, just mISDN is enough to sync right ? |
11:53.53 | ArchSSM | tnt_: That is correct. |
11:54.11 | tnt_ | ArchSSM: And if it's red ... what could that be ? |
11:54.36 | ArchSSM | then it's out of sync. probably a driver issue |
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12:08.13 | phix | hi |
12:08.16 | phix | lets all help me |
12:08.21 | phix | now please k thnx |
12:08.25 | phix | or whaen ever |
12:08.29 | phix | prefererably npow |
12:08.31 | phix | that would be great |
12:08.44 | ArchSSM | If you have a question, just ask. |
12:12.14 | tzafrir | jbot, tell phix about ask |
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12:20.07 | synthetiq | where can i find the text2wave application? |
12:20.30 | synthetiq | because it is not built by default.. |
12:21.04 | uwe | hello, im having problems with g729 sound quality ... ive setup a client to use g729 ... the recipient hears the voice very well, but the caller hears the recipient with very bad voice quality ! and i have no idea how to debug or tune g729 ... if it possible at all, another thing, i suppose translating from g729 to other codecs goes always through slin , so it doesnt matter if i translate from gsm to g729, or is there a better combination ? |
12:22.18 | uwe | its g729 <=> asterisk <=>gsm<=>landline .... landline hears g729 well, but g729 doesnt hear landline well |
12:26.25 | J4k3 | I do not believe g729 and gsm are complimentary |
12:26.31 | J4k3 | I think the end result is very awful audio |
12:27.31 | J4k3 | they'd be better off running asterisk on their end transcoding from gsm to g711, if their equipment doesn't natively support gsm and doesn't have the bandwidth for g711 |
12:27.54 | J4k3 | (I'm sure it supports g711 if it supports g729) |
12:30.19 | SteveTotaro | anyone know bristuff that can help? |
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12:30.25 | SteveTotaro | with qozap? |
12:31.04 | ArchSSM | SteveTotaro: Ask, and we'll answer as best as possible |
12:31.16 | uwe | J4k3, ... sorry, i dont understand you very well, i can change the gsm encoding to something else, but not g729 |
12:31.35 | uwe | what do you suggest to set it to ... or do you mean it doesnt make a diffrence |
12:31.47 | tnt_ | ArchSSM: And what kind of driver issue could do that ? Try another version or ? |
12:32.39 | ArchSSM | tnt_: That is very hard to say. Do a pastie on a pastiesite on the appropriate logs, and comment on what driver you're using |
12:35.20 | tnt_ | ArchSSM: http://pastebin.com/m65ff81df |
12:35.34 | tnt_ | And I'm using mISDN-1_1_7_2 |
12:35.50 | ArchSSM | what card was it? |
12:36.04 | JT | uwe: compressed codec, to another compressed codec, don't be surprised if the end result is not very good, is what he's saying |
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12:37.25 | tnt_ | ArchSSM: b410p |
12:37.34 | tnt_ | used in Belgium ... (if that matters :) |
12:38.29 | *** part/#asterisk Sweeper (i=sweeper@66.221.78.1) |
12:38.44 | ArchSSM | shouldnt :) |
12:39.24 | ArchSSM | well... at least try to turn on debug and se if you get anymore information |
12:40.05 | tnt_ | What value to use for debug ? just 1 ? |
12:41.38 | ArchSSM | yep |
12:44.26 | uwe | JT, what codecs are not compressed, what is the raw format ? ulaw and alaw ? |
12:44.39 | uwe | g711 ? |
12:46.09 | RoyK | everything's compressed somehow |
12:46.30 | RoyK | g.711a/u is 13/14bit 'compressed' to 8bit |
12:46.33 | uwe | hmm ... |
12:47.00 | RoyK | but g.711a is the closest to 'raw' (i think) |
12:47.06 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
12:47.10 | aiksa[LV] | howdy everyone |
12:47.33 | aiksa[LV] | anyone comming to asterisk con in NL? |
12:47.50 | tnt_ | ArchSSM: http://pastebin.com/m2cfd0f68 That doesn't really speak to me ... :) |
12:48.49 | ArchSSM | Same here really |
12:49.43 | aiksa[LV] | I had another question in mind, is it possible to automatically move to the next dialplan priority a caller who had been waiting in queue for a given amount of time? |
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12:56.01 | mosty | aiksa[LV], i believe there's a queue option to specify that |
13:00.09 | Frogzoo | usual pbx's allow you to dial zero to get an 'outside line' - how best to do this in the dialplan? |
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13:01.12 | mosty | Frogzoo, it's simpler to just use local extensions which aren't confused with regular telephone numbers, like 3 digit local extensions. then you don't need any of that crap |
13:01.12 | cpm | interesting, all the pbxs I've worked with are dial-9 |
13:01.23 | Spyder12345 | anyone in here know anything about dialing sip via tcp in 1.6? |
13:02.51 | Frogzoo | thanks mosty, that helps |
13:03.10 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.57) |
13:03.56 | *** join/#asterisk PepOSX (n=angeldav@190.79.246.105) |
13:05.26 | nebojsajsimic | hi all |
13:05.52 | nebojsajsimic | <PROTECTED> |
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13:11.41 | mosty | is there an asterisk management interface event for failed authentication of sip clients? |
13:12.10 | nebojsajsimic | no |
13:12.20 | nebojsajsimic | it normal work from exten |
13:12.27 | *** join/#asterisk lirakis (i=lirakis@66.252.24.133) |
13:12.45 | mosty | nebojsajsimic, huh? |
13:14.18 | nebojsajsimic | i dont use autentication all is blank |
13:14.24 | nebojsajsimic | it is local system |
13:14.57 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
13:15.38 | mosty | nebojsajsimic, i wasn't answering your question about AGI, i was asking about AMI |
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13:16.29 | MikeBest | hello |
13:16.57 | nebojsajsimic | hi |
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13:17.26 | MikeBest | anyone experinced in asterisk? |
13:17.55 | nebojsajsimic | mosty: i have configured GSM GW in sip |
13:18.05 | ArchSSM | MikeBest: Probably quite a few here. Just ask if you have a question |
13:18.08 | nebojsajsimic | sip.conf |
13:18.22 | MikeBest | i am a total newbie |
13:18.28 | nebojsajsimic | me tooo |
13:18.31 | MikeBest | even asking questions is difficult |
13:18.33 | nebojsajsimic | but i try-ing |
13:18.35 | nebojsajsimic | :_ |
13:18.44 | ArchSSM | MikeBest: In that case, you should read up a bit first. |
13:18.48 | MikeBest | i just have a project in my mind |
13:18.52 | aiksa[LV] | mosty, the only cooresponding option seems to be servicelevel, but I am not sure if it will disconnect user fro queue |
13:18.58 | MikeBest | and looking for some advice |
13:19.13 | aiksa[LV] | thats seems like an option for statistics only |
13:19.24 | ArchSSM | MikeBest: Sure.. try us |
13:19.41 | mosty | aiksa[LV], i want sip client authentication errors |
13:20.01 | mosty | nebojsajsimic, what are you trying to do with agi? |
13:20.13 | nebojsajsimic | just to make a call |
13:20.16 | aiksa[LV] | mosty: - i was referring to that queue issue |
13:20.51 | nebojsajsimic | Dial(SIP/number@GW,r,15) |
13:21.11 | aiksa[LV] | anyone else, has any ideas? I really would not like to keep those timers in some kind of °service connected through AMI |
13:21.24 | mosty | aiksa[LV], there is a timeout option to the queue command, have you tried that? |
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13:22.13 | aiksa[LV] | mosty: nope, its not what i want |
13:22.30 | aiksa[LV] | timeout looks at how long agents phone have rung |
13:22.31 | atis_work | nebojsajsimic: you should also not write("Dial") directly, but give AGI commad "EXEC Dial..." |
13:22.45 | mosty | aiksa[LV], what are you trying to do? |
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13:22.58 | aiksa[LV] | ok, here comes explanation: |
13:23.42 | atis_work | nebojsajsimic: please check http://www.voip-info.org/wiki/view/exec |
13:23.43 | tuxfoo | http://www.voip-info.org/wiki/view/Asterisk+monitoring |
13:23.58 | nebojsajsimic | Thanks!!!! |
13:24.06 | tuxfoo | see if that gets you what you want or gets you started in the right direction |
13:24.15 | atis_work | nebojsajsimic: i think also Dial options should be separated with space from Dial |
13:24.17 | aiksa[LV] | if a user joined the queue, but for lets say 5 min. he hadnt reach any of the agents (too long could be one of reasons) - he gots thrown out of the queue and proceeds to next priority |
13:25.52 | nebojsajsimic | Feb 7 14:24:06 WARNING[13425]: res_agi.c:1115 handle_exec: Could not find application i get this |
13:25.56 | aiksa[LV] | mosty: that was the funcionality I wanted to achieve |
13:26.31 | mosty | aiksa[LV], the timeout in the Queue command does that, doesn't it? |
13:27.26 | aiksa[LV] | I guess - not, it counts seconds while a user had an interaction with agenty, but agent never picked up |
13:27.35 | lirakis | morning |
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13:27.42 | *** mode/#asterisk [+o lmadsen] by ChanServ |
13:27.42 | nebojsajsimic | morning |
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13:28.27 | mosty | aiksa[LV], "'timeout' sets the time in seconds that a call will wait in the queue before it is routed to the next priority in the dialplan" - isn't that what you want? |
13:30.15 | aiksa[LV] | wow exactly |
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13:30.39 | aiksa[LV] | mosty: but asterisk book SE gives different explanation for timeout parameter in queues.conf |
13:30.49 | aiksa[LV] | or this is not a queues.conf parameter? |
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13:30.59 | MaartenB | hello everyone |
13:31.05 | mosty | aiksa[LV], i'm talking about an option the the Queue application, not the setting in queues.conf |
13:31.34 | MaartenB | I would like to have a queue systems but without the trouble of agents logging on and off, I just want to have phones available for pickup by using the DND button on my phone, is that possible? |
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13:32.03 | aiksa[LV] | mosty: Oh, i see it now. Stupid - me. I was thinking you referred to a config file. |
13:32.06 | aiksa[LV] | sorry |
13:32.21 | mosty | aiksa[LV], i did originally |
13:32.46 | atis_work | nebojsajsimic: try issuing "agi debug" in CLI to see what you're sending |
13:33.11 | aiksa[LV] | MarteenB - can you modify what DND button does? |
13:33.13 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
13:33.14 | ZaVoid | morning |
13:33.25 | aiksa[LV] | what extension to ring upon press? |
13:33.48 | aiksa[LV] | mosty - and stupid me, never to look at aparemeters which Queue() takes .. :P |
13:34.11 | hmmhesays | it is morning |
13:34.13 | lmadsen | morning |
13:34.38 | hmmhesays | I'm about to crack my cell phone open and modify the battery case to use AAA batteries |
13:35.04 | cpm | just kludge up a power adapter, it's easier |
13:35.11 | cpm | and you can use D cells |
13:35.44 | hmmhesays | I can make triple aaa's look fancy and the mah rating is still 3x what the original battery is |
13:36.56 | hmmhesays | and I don't have an extra power supply to cut up |
13:37.10 | hmmhesays | AND the e815 psu connector sucks big time |
13:37.15 | eric2 | anyone ever use attractel's t.38 faxing software for use with asterisk? |
13:37.19 | cpm | yeah, they do suck |
13:37.55 | hmmhesays | I have a month left before I get a new phone, I this one eats batteries |
13:38.12 | mvanbaak | ugh, I HATE it when I cant get it working |
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13:39.32 | yang | I would like to enable parked call the *8 string takes the call...I have these values, but it doesnt work well yet - http://openpaste.org/en/5008/ |
13:39.59 | lmadsen | anyone know if the D() flag to Dial() accepts 'w' as a wait pattern? |
13:40.06 | coppice | eric2: a few people seem to have tried it |
13:40.25 | hmmhesays | yang what are you asking? |
13:40.25 | eric2 | coppice: is it the way to go? |
13:40.37 | lmadsen | yang: did you enable tT (pls check to see what they mean with 'core show application dial')? |
13:40.40 | hmmhesays | I could use a service manual fo rthis phone |
13:40.54 | hmmhesays | lmadsen, kK is the 1 touch parking flag |
13:40.56 | yang | http://openpaste.org/en/5009/ |
13:41.09 | coppice | eric2: well, the people I know who tried it abandoned it. I don't know their reasons, though |
13:41.13 | lmadsen | hmmhesays: ahhh right... stupid Justin for waking me up early |
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13:41.18 | eric2 | interesting |
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13:41.37 | eric2 | would be good to find out the reasons for abandonment |
13:42.11 | coppice | they might have just been they were too tight to pay at the end of the evaluation period, or it might be troublesome. I have no clear idea |
13:42.13 | yang | lmadsen: i need to attach tT to each DIal string at the end |
13:42.43 | tuxfoo | does anyone know how to get the wildcard masks to work on an inbound zap line? I can get an exact match to work, but the wildcard masks don't seem to work. I want to intercept 800 numbers - I have s/8XXXXXXXXXX as my mask but it ignors it. - thanks |
13:42.57 | lmadsen | yang: i was wrong.. it's 'kK' |
13:42.58 | lmadsen | <PROTECTED> |
13:42.58 | lmadsen | <PROTECTED> |
13:43.11 | coppice | eric2: it does appear they offer evaluations, though, so you could try it |
13:43.20 | mosty | tuxfoo, patterns start with _ |
13:43.41 | eric2 | price is a tad expensive, the euro is high |
13:43.55 | tuxfoo | I tried that as well and it did not work s/_8XXXXXXXXXX |
13:44.06 | coppice | i've never actually seen a price for it |
13:44.06 | mosty | tuxfoo, what is the s/ in there? |
13:44.09 | eric2 | looks like they offer their licensing on a per channel basis |
13:44.10 | lmadsen | I paid 19 pounds for goosync and it's well worth it :) |
13:44.19 | yang | lmadsen: I have rt enabled (no T) http://openpaste.org/en/5010/ |
13:44.28 | yang | lmadsen: ah ok ! |
13:44.33 | yang | adding k there |
13:44.56 | coppice | eric2: I do the same with my T.38 software - $0 per channel :-) |
13:45.10 | eric2 | what's your setup like? |
13:45.16 | lmadsen | coppice: you should really up the price |
13:45.33 | eric2 | are you even using t.38? |
13:45.51 | coppice | lmadsen: you mean like a 10% increase, or something |
13:45.59 | tuxfoo | s/1234567890 works as an exact match, but the wild cards don't. I really don't want to add every number,so I thought I could just mask it. But it is not working that way |
13:46.04 | eric2 | 10% of 0 is still a bad number :) |
13:46.05 | lmadsen | coppice: heck... 200%! |
13:46.19 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
13:46.35 | lmadsen | shhhh.... stop talking about him |
13:46.52 | eric2 | oh great, another cannuk |
13:47.03 | mosty | tuxfoo, are you trying to match s and 1234567890 extensions in that example? |
13:47.19 | lmadsen | mosty: no, he is trying to patch a CID with that |
13:47.54 | yang | lmadsen: http://openpaste.org/en/5011/ |
13:47.56 | lmadsen | tuxfoo: s/_8XXNXXXXXX |
13:48.15 | lmadsen | tuxfoo: although I haven't actually tried to see if it works |
13:48.54 | mosty | tuxfoo, you could use GotoIf and a regex expression if that doesn't work |
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13:49.23 | hmmhesays | there seems to be an extra terminal on this e815 battery |
13:49.36 | hmmhesays | I have ground, <something>, +, + |
13:50.44 | hmmhesays | I guess a trip to radioshack is in order |
13:51.35 | tuxfoo | s/ is my start so certain number get a menu, while others just go striaght through. That part is working. For, example, s/123456789 is my cell phone. When it calls in to my system I get a menu to administer the system, while a unmasked nmber just enters the system and rings my wife's phone. I would like to take it a step further and mask numbers to other extensions or ZAPTELLER them without having to know every 800 number. |
13:52.03 | lmadsen | tuxfoo: did you try the pattern match in the CID match or not? |
13:52.16 | yang | lmadsen: its not working (quite) yet |
13:52.31 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
13:52.50 | tuxfoo | no I did not try that since an exact match seems to work fine, I just assumed I could pattern match. I will try that |
13:52.59 | yang | i get nothing to pick up error |
13:55.26 | yang | Do I have to apply http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups to be able to pickup the phone with *8 ? |
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14:01.11 | mosty | modules.conf |
14:01.32 | mosty | does "show features" in the console show anything? |
14:01.32 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
14:03.07 | yang | mosty: no modules called feature in /usr/lib/asterisk/modules/ |
14:03.30 | mosty | yang, no res_features.so? |
14:03.51 | *** join/#asterisk AndyGraybeal (n=andy@node50.34.251.72.1dial.com) |
14:04.10 | yang | mosty: http://openpaste.org/en/5012/ |
14:06.59 | yang | mosty: and here are my extensions.conf http://openpaste.org/en/5010/ |
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14:11.41 | plik | yang: features.conf only gets loaded at start - not reload |
14:12.31 | b1ch0 | <PROTECTED> |
14:12.56 | [TK]D-Fender | b1ch0: "core show function CHANNEL" |
14:13.51 | plik | good morning [TK]D-Fender |
14:14.07 | [TK]D-Fender | mornin' |
14:14.19 | yang | hi TK ! |
14:14.40 | yang | plik: yeah i tried with restart too |
14:15.31 | [TK]D-Fender | yang: [2008-02-07 14:47:04] NOTICE[3252]: chan_sip.c:13957 handle_request_invite: Nothing to pick up for 4dcf5b624542e8e3@10.105.2.64 <-- I'm betting you didn't even set your call-groups. |
14:16.04 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
14:16.18 | [TK]D-Fender | yang: Go show us your sip.conf for those 2 phones. |
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14:18.08 | yang | [TK]D-Fender: yeah i havent , however there arent any Callgroups in features.conf, like i saw on http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups |
14:19.20 | [TK]D-Fender | yang: thats because those are the VALUES you have to set in SIP>CONF <-- |
14:19.38 | [TK]D-Fender | yang: they will not appear in features.conf, but they are USED by it |
14:20.22 | mintee | is there a way to write DB functions in the asterisk console? |
14:21.09 | [TK]D-Fender | mintee: which DB, and what do you mean by "write functions"? |
14:22.32 | yang | [TK]D-Fender: http://openpaste.org/en/5013/ |
14:22.50 | synthetiq | for asterisk agi, the only application that works is exec and set_extension, but nothing else works, get_variable, noop, etc .... anyone have a clue why? |
14:22.57 | yang | http://openpaste.org/en/5012/ & extensions.conf http://openpaste.org/en/5010/ |
14:23.16 | yang | [TK]D-Fender: I just printed Asterisk book from page 50-80 |
14:23.20 | _gm | synthetiq: can you show your code |
14:24.04 | [TK]D-Fender | yang: You did not set callgroups & pickupgroups for ANY of your phones. |
14:24.17 | [TK]D-Fender | yang: those values you saw belong in your peer entries |
14:24.54 | [TK]D-Fender | yang: otherwise they won't know which calls theyr even permitted to "pick up". Otherwise you could have people randomly hijacking other peoples calls. |
14:25.32 | synthetiq | my $digit = $AGI->wait_for_digit(60); $AGI->set_extension($digit); |
14:25.40 | synthetiq | plain and simple |
14:25.40 | yang | [TK]D-Fender: so I must apply Callgroup=1 to every sip entry |
14:25.54 | [TK]D-Fender | yang: BOTH. |
14:26.01 | yang | ok |
14:26.20 | yang | for the pickupgroup i assume i need to add 60-80 |
14:26.24 | yang | into each sip |
14:26.35 | synthetiq | or i try to do noop on variable using : $AGI->exec('read','TARGET_USER||1|||60'); $AGI->noop('\${TARGET_USER}'); |
14:27.02 | synthetiq | nasicly i want to do processing on a dtmf digit |
14:27.06 | synthetiq | basicly |
14:27.11 | synthetiq | im using Asterisk::AGI |
14:27.14 | [TK]D-Fender | yang: no. |
14:27.33 | *** part/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
14:27.45 | synthetiq | i also tryied the STDIN method of WAIT FOR DIGIT and no luck |
14:27.52 | [TK]D-Fender | yang: go read the page on those 2 values again, you are failing to understand the basics of this. |
14:28.09 | synthetiq | actually the script exits when waiting for a dtmf digit..... |
14:28.23 | synthetiq | so the only thing that works is $AGI->exec('read','TARGET_USER||1|||60'); |
14:30.11 | yang | [TK]D-Fender: each sip needs a different Callgroup like Callgroup 1 for ext. 60 and Callgroup 2 for ext. 61, but pickupgroup is then 1-20 for all SIPS |
14:30.17 | yang | as i understood? |
14:37.06 | yang | ok seems i got it right now:) |
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14:39.05 | yang | [TK]D-Fender: do you happen to know the "trick" about Grandstream BLF lights ? I have enabled the Subscribecontext=BLF and I can see the green lights and i can also dial those extensions by pressing the lights, but i dont see the occupied light (red) |
14:39.43 | mosty | yang, do you have sip hints in your dialplan? |
14:40.36 | yang | sure i do |
14:40.44 | yang | they are in BLF context |
14:41.35 | yang | exten => 60,hint,SIP/60 ; Jozi |
14:41.35 | yang | exten => 61,hint,SIP/61 ; Fax |
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14:43.39 | yang | We ran out of toner right in the middle of printing the Asterisk manual :/ |
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14:47.05 | mintee | [TK]D-Fender, sorry, I had to walk away. I mean I just wanna query some things from the asterisk DB from the cli, and not have to write it in an extension for testing... |
14:47.07 | [TK]D-Fender | yang: I seriously doubt you need more that 3-4 pickupgroups/callgroups in your system |
14:47.22 | [TK]D-Fender | mintee: "help database" |
14:47.44 | mintee | oh, thanks |
14:47.55 | *** join/#asterisk drako (n=thinkpad@nelug/coreteam/luisjose) |
14:48.32 | drako | having bad times with ooh323 |
14:48.38 | *** join/#asterisk funxion (n=x@63.214.236.169) |
14:48.48 | [TK]D-Fender | yang: BLF could be an incorrectly configured phone. |
14:49.18 | yang | [TK]D-Fender: it works with a vlines version of asterisk...when we plug the phones there, and it doesnt work with my asterisk configuration |
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14:49.53 | drako | http://pastie.caboo.se/148665 |
14:50.19 | [TK]D-Fender | yang: I'd have to see the whole mess, but please try to tackle 1 problem at a time. |
14:50.44 | yang | [TK]D-Fender: ok , next week then |
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14:52.28 | yang | thanks ! |
14:53.09 | *** join/#asterisk unenough (n=jfds@CBL217-132-95-18.bb.netvision.net.il) |
14:54.03 | unenough | how hard is it to implement a module that interacts with audio in real time? meaning it should receive the decoded audio stream and be able to write to it as well |
14:54.27 | zeeesh | missing module will anybody guide the name "agi.pm" "Can't locate Asterisk/AGI.pm in @INC (@INC contains: /"??????? |
14:54.31 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
14:54.48 | mosty | zeeesh, did you install that perl module? |
14:55.58 | [TK]D-Fender | oh wow, awesome breaking news!!!! 2008-02-07 - AsteriskWin32 0.66 released build from asterisk 1.2.26.2, with X100P support. |
14:56.13 | drako | Had to drop call because I couldn't make SIP/101-081c6590 compatible with OOH323/ |
14:56.27 | [TK]D-Fender | Now Windows shmucks can use WinModems and asstricks! |
14:56.31 | zeeesh | i hv installed more than 15 different perl modules ... but i am unable to find by the name of "AGI.PM" so from where can i get |
14:57.08 | [TK]D-Fender | unenough: Go look at an * app that does this already and see |
14:57.59 | unenough | [TK]D-Fender, if I knew which one to look at... |
14:58.37 | [TK]D-Fender | unenough: Pick any that listen, and any that play back. Look for the voice changer patch as well (GOOGLE) |
14:58.44 | mosty | zeeesh, try #perl since that problem really has nothing to do with asterisk |
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14:58.56 | the_5th_wheel | does anyone kbnow if there are something like a ATA avaliable, but with isdn lines? |
15:00.13 | [TK]D-Fender | the_5th_wheel: I've heard of BRI channel banks once or twice, but this idea is virtually unmentioned in here. |
15:02.01 | *** join/#asterisk gdiebel (n=gregd@adsl-69-217-146-185.dsl.mdsnwi.ameritech.net) |
15:04.05 | the_5th_wheel | What are my options then to connect an analouge PBX to an asterisk pbx? |
15:04.35 | [TK]D-Fender | the_5th_wheel: Anything else clearly. VoIP, PRI, Analog. |
15:05.20 | mosty | the_5th_wheel, what kind of interfaces does the old pbx have? |
15:05.48 | drako | the_5th_wheel, b410p ? |
15:05.53 | drako | tdm b410p |
15:06.12 | the_5th_wheel | Im trying to connect various of our offices to the asterisk system. It varies, some have analuge trunks, some have bri trunks, some have PRI trunks avaliable |
15:06.22 | [TK]D-Fender | Ah, you could use a BRI card for interfacing with *, but you mentioned ATA. |
15:06.46 | mosty | the_5th_wheel, you can get pci/pci express PRI, BRI and TDM cards |
15:06.53 | [TK]D-Fender | the_5th_wheel: For which there are a number of vendors |
15:07.13 | the_5th_wheel | So i can connect a PRI card directly to the bri trunk of a pbx? |
15:07.31 | mosty | no |
15:07.39 | [TK]D-Fender | the_BRI to BRI |
15:09.11 | the_5th_wheel | Im not following. |
15:09.21 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
15:09.25 | the_5th_wheel | How is that going to help me connect their PBX to the asterisk server? |
15:09.57 | drako | any idea, [Feb 7 10:05:30] WARNING[3234]: app_dial.c:1685 dial_exec_full: Had to drop call because I couldn't make SIP/101-081b7bc8 compatible with OOH323/89.149.164.34-3de3 |
15:10.08 | *** join/#asterisk gkirk (n=gkirk@S0106000fcbb46719.su.shawcable.net) |
15:10.10 | *** part/#asterisk gkirk (n=gkirk@S0106000fcbb46719.su.shawcable.net) |
15:10.17 | *** join/#asterisk bantu (n=bantu@p54A32BE6.dip0.t-ipconnect.de) |
15:10.18 | [TK]D-Fender | the_5th_wheel: If they have BRI port, you can put a BRI card into your * server and connect them. |
15:10.34 | mintee | database put\get at the cli doesn't deal with the mysql database does it? |
15:10.55 | *** join/#asterisk joshkidd (n=josh@adsl-068-209-028-087.sip.asm.bellsouth.net) |
15:10.57 | the_5th_wheel | [TK]D-Fender: then i would need to go over telecom infrastructure |
15:10.58 | [TK]D-Fender | drako: Here's an idea, show us the entire call attempt with SIP and H323 debug so we can actually see the CAUSE, not just the warning. |
15:11.04 | [TK]D-Fender | drako: PASTEBIN is your friend |
15:11.10 | the_5th_wheel | [TK]D-Fender: the idea is to route things over the internet |
15:11.12 | mosty | mintee, no. try func_odbc |
15:11.40 | [TK]D-Fender | the_5th_wheel: You are jsut not getting it. You need to connect the 2 BOXES together. Telling the PBX to send calls to * to deal with is another matter. |
15:11.59 | mintee | mosty, that's not the problem really, i'm just curious as to what kinda DB it's dealing with them. |
15:12.16 | [TK]D-Fender | the_5th_wheel: they need to TALK first. then what you can tell your PBX to do to send calls over the connection you now have with * is another matter. |
15:12.20 | mosty | mintee, berkeley db, version 1 or something i think |
15:12.30 | [TK]D-Fender | mintee: No, it does not |
15:12.49 | [TK]D-Fender | mintee: those commands are for AstDB only. |
15:13.15 | [TK]D-Fender | mintee: If you want to manipulate MySQL, use "mysql" at *nix DCLI like the rest of the universe |
15:14.09 | the_5th_wheel | aso you are saying i put a PC into the small office, with one of these BRI cards, and have the PC route the calls to asterisk, but how do i connect to the standard PBX, what do i conect it to? |
15:15.14 | mosty | the_5th_wheel, an asterisk box with a BRI card can connect to another pbx that has BRI ports. i don't understand what else you are asking |
15:15.14 | [TK]D-Fender | the_5th_wheel: what is this EXTRA PC you're talking about? |
15:15.47 | [TK]D-Fender | the_5th_wheel: you put the card into your ASTERISK server. You plug the ports on this card into your existing PBX. Whats so hard to understand here? |
15:15.59 | [TK]D-Fender | the_5th_wheel: there is no second PC. |
15:16.24 | *** join/#asterisk envoy (i=bri@173.sub-75-221-226.myvzw.com) |
15:16.30 | tuxfoo | Trunk the PBX to * - use what ever interface you want |
15:16.57 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
15:17.33 | tuxfoo | route calls between systems via the trunk |
15:17.39 | the_5th_wheel | ok, that makes sence.So the b410p will act like the teleco Exchange? |
15:18.14 | [TK]D-Fender | the_5th_wheel : or as a station depending on what kind of port on your PBX you want to plug it into and how you want it to work. |
15:18.25 | envoy | We are using asterisk for our office and for some reason when we try to connect to a conference bridge and enter out conference number the remote PBX doesn't reconize all the digits. I've adjusted the tx/rx gain in zapata.conf (but all positive values) could that be the issue |
15:18.25 | mintee | [TK]D-Fender, thanks for the info... oh, and your well known low-blow too. |
15:18.41 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
15:19.16 | [TK]D-Fender | mintee: Wasn't a low blow... that was a "use the right tool for the right job" :) Someone else would have gone "ouch" if I really threw something at you. |
15:19.25 | *** join/#asterisk DarWin_vcch (n=daryl@vcchgate.vcch01.springfield.tn.us.vcch.net) |
15:19.29 | mosty | envoy, analogue telephone line between the two asterisk machines? |
15:20.19 | mintee | whatever... it only seems that with you there is a 1-to-1 ratio on answer-insult. Just saying.... not everyone jumped on the asterisk wagon the day it came out. |
15:20.21 | envoy | analog lines on one asterisk machine (our office) going to customer PBX's |
15:20.21 | [TK]D-Fender | mosty: Umm.. think about how that sounds... |
15:20.52 | mosty | [TK]D-Fender, i'm asking, not suggesting |
15:20.57 | [TK]D-Fender | mintee: s'ok alls good. just try not to use a spoon as a crowbar and life will be good :) |
15:21.07 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
15:21.12 | mintee | :P |
15:21.14 | [TK]D-Fender | envoy: You get some of the digits? |
15:21.15 | mosty | envoy, what dtmfmode settings do you have on your sip clients? |
15:21.24 | envoy | [TK]D-Fender, yup |
15:21.51 | mosty | envoy, i think there's a relaxdtmf option in zapata.conf, look at the docs on that and see if it's worth trying |
15:21.58 | [TK]D-Fender | envoy: Playing with gains is one thing. try this as well "relaxdtmf=yes" for your channel definition. You'll have to reload Zap or *. |
15:22.28 | ZaVoid | relaxdtmf? whas that do? |
15:22.30 | envoy | [TK]D-Fender, dtmfmode is rfc2833 |
15:22.43 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
15:22.57 | [TK]D-Fender | envoy: No need for that yet... you jsut said they are cannected over analog. DTMFMODE does not factor into this at all. |
15:23.20 | [TK]D-Fender | ZaVoid: that loosens the constraints on the detection routine when pickup seems spotty |
15:23.21 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
15:23.30 | [TK]D-Fender | envoy: Also, what ver of * & zaptel are you using? |
15:23.44 | ZaVoid | really... |
15:23.49 | ZaVoid | i'm gonna go look at that |
15:23.49 | [TK]D-Fender | ZaVoid: yup |
15:24.28 | mosty | tzafrir, where is that patch of yours for a "live" asterisk test install? |
15:24.29 | drako | how is the webpage with the codecs compiled for all plataforms |
15:24.35 | drako | and cpu types |
15:24.57 | envoy | [TK]D-Fender, * 1.2.18 and zapata I'm not sure |
15:24.59 | *** join/#asterisk nybbled (n=nybbled@about/apple/performa/nybble) |
15:25.18 | envoy | zaptel rather |
15:25.27 | [TK]D-Fender | envoy: I would highly recommend upgrading to 1.4. Zaptel's DTMF detection has undergone MAJOR improvements. |
15:26.08 | funxion | Anyone know why I would get dropped calls on a T1 PRI with * v1.2.9.1? I'm receiving ": Didn't get a frame from channel: Zap/1-1" in debug... |
15:26.16 | envoy | [TK]D-Fender, I keep meaning to |
15:26.50 | [TK]D-Fender | funxion: Check to see if you are getting HDLC aborts, PCI master warnings, of frame slips/losses |
15:27.07 | envoy | [TK]D-Fender, any ideas aside from upgrading? |
15:27.08 | funxion | I'm not getting any T1 errors oneither side |
15:27.16 | tzafrir | mosty, in a bug report: |
15:27.27 | [TK]D-Fender | envoy: Since it is directly pertinent to your current problem you should perhaps revisit your piority to do so. |
15:27.36 | envoy | [TK]D-Fender, or is this a know issue in 1.2? |
15:27.46 | [TK]D-Fender | envoy: Just gains & "relaxdtmf". If that doesn't do it, then you know whats next. |
15:27.54 | envoy | [TK]D-Fender, thanks |
15:28.03 | [TK]D-Fender | envoy: Its well known that they did major changes for 1.4 with a reason. |
15:28.25 | [TK]D-Fender | funxion: is it sharing an interrupts by any chance? |
15:28.32 | funxion | not that I know of |
15:28.37 | funxion | but its possible |
15:28.41 | funxion | lemme look |
15:28.44 | [TK]D-Fender | funxion: Go verify |
15:29.57 | funxion | doesnt seem like it |
15:30.00 | funxion | its IRQ 169 |
15:30.09 | funxion | never seen a ## that high before |
15:30.12 | funxion | qweird |
15:31.27 | [TK]D-Fender | funxion: That's probably fine |
15:32.06 | ZaVoid | oh this is relaxdtmf is only for zapta cards? |
15:32.24 | funxion | weird |
15:32.31 | hmmhesays | 4 hours till my soup is done |
15:32.46 | hmmhesays | 5 quarts of vegetable roast |
15:32.48 | [TK]D-Fender | ZaVoid: I'm not 100% sure.... take a look on the WIKI. |
15:32.53 | funxion | I just turned on pri debug for the 2 spans and have a test call up waqiting for it to drop |
15:32.56 | ZaVoid | i did.. that whats it sounds like |
15:33.14 | ZaVoid | oh wait.. Note: Asterisk 1.4 now also has the relaxdtmf= setting available in sip.conf. |
15:33.16 | ZaVoid | hmm |
15:33.16 | hmmhesays | [TK]D-Fender, you ever use a keeley looper? |
15:33.29 | [TK]D-Fender | hmmhesays: Never heard of the term. |
15:34.05 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
15:34.16 | funxion | isnt that for music? |
15:34.21 | hmmhesays | hardware loop you can put your effect on so you don't have any signal degredation when you aren't using them |
15:35.38 | [TK]D-Fender | hmmhesays: I've seen them on amps (this is really an A/B at that level). Personally I have no need. I only run my GT-8, and have no need of any external effects :) |
15:36.05 | [TK]D-Fender | hmmhesays: Guitar > Wireless > Receiver > GT-8 > Amp |
15:36.07 | *** join/#asterisk patrick-- (n=patrick@devnull.biz) |
15:36.17 | patrick-- | whats the difference between TE and NT mode? |
15:36.24 | hmmhesays | [TK]D-Fender, yeah I don't run a gt-8 |
15:36.29 | jeremy_g | I have installed asterisk 1.4. But now i want to install asterisk 1.6. How do i uninstall asterisk 1.4? I want to remove whatever files were created as a result of ../configure and make install ..everything back to the way it was? |
15:36.38 | hmmhesays | I have a hardware effet loop in my peavey XXL |
15:36.42 | hmmhesays | but not on my marshall |
15:36.51 | the_5th_wheel | Would i be able to connect an ISDN phone to a B410P card? |
15:37.09 | [TK]D-Fender | the_5th_wheel: Yes |
15:37.23 | the_5th_wheel | Ok, Cool. Now everything is clear :-D |
15:37.42 | *** join/#asterisk shasta (i=shasta@bluzg.slackware.pl) |
15:38.08 | *** join/#asterisk Spyder12345 (n=bob@169.139.217.48) |
15:38.40 | envoy | [TK]D-Fender, Changing my gains fixed the issue. I had both postive gains and when I lowered them it fixed it. Any idea on the reason for this |
15:39.08 | [TK]D-Fender | envoy: Probably to high and was distorting |
15:39.11 | envoy | [TK]D-Fender, I would have thought this would be an issue it I was putting negative gains, but... |
15:39.45 | Spyder12345 | anonyone familiar with tcp dialing via sip in 1.6 or know where I can find some info on this? |
15:39.51 | patrick-- | anyone? |
15:40.15 | [TK]D-Fender | envoy: Like on old guitar amps where you up the gain on the pre-amp till it distorts and use the master volume to lower. Welcome to rock&roll :) |
15:40.42 | [TK]D-Fender | Spyder12345: Odds are it'll use whatever protocol you specified for your peer. |
15:40.50 | tzanger | [TK]D-Fender: I'm running into two different PBXes now where the PCM data coming in seems awfully 'hot' |
15:41.02 | tzanger | if I transmit a ulaw file to the PCM out it sounds fine |
15:41.03 | jeremy_g | how can i undo what ./configure && make install does to my file system? |
15:41.06 | [TK]D-Fender | tzanger: Yup, I've seen a few PRIS like that before. |
15:41.14 | tzanger | I don't know if I'm missing a pad on the input or not |
15:41.31 | tzanger | [TK]D-Fender: this isn't PRI, this is an actual PBX that I've hijacked the TDM bus from :-) |
15:41.49 | [TK]D-Fender | tzanger: O RLY? Do tell... |
15:42.10 | Spyder12345 | hmm.. well I am not having much luck with that.. is the peer command to enable it transport=tcp? |
15:42.53 | [TK]D-Fender | Spyder12345: You might want to actually follow Mantis, use Google, read docs, etc.... that setting sounds like what I would expect.... |
15:42.55 | tzanger | I have to play around a little more to see if the the PCM encoder's misconfigured, I've misconfigured a gain stage before it or what |
15:43.05 | ZaVoid | meh i can't find any explanation anywhere what relaxdtmf actually does |
15:43.32 | jeremy_g | make uninstall-all |
15:43.34 | jeremy_g | silly me |
15:43.56 | Spyder12345 | well none of those places are very useful since I was told this is a new feature only in 1.6. I have already done that with no working results. |
15:44.14 | ZaVoid | oo 1.6 does tcp sip? |
15:44.37 | [TK]D-Fender | ZaVoid: Yup |
15:44.41 | ZaVoid | nifty |
15:45.09 | Spyder12345 | yes it actually listens on both udp and tcp.. But finding good documentation on its use is hard to find. |
15:45.52 | ZaVoid | still can't find anything on relaxdtmf |
15:46.11 | [TK]D-Fender | ZaVoid: relaxdtmf = picture * "scoring" a tone its receiving and trying to say if I hit 90% accuracy then its OK. Then relaxed it'll say... you know... 80% isn't too bad now. |
15:46.44 | ZaVoid | no i understand that fender.. i'm just look somewhere to see how much it does etc etc |
15:47.04 | ZaVoid | brb |
15:48.08 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
15:48.51 | tzanger | ZaVoid: it's in dsp.c in asterisk's source |
15:50.32 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
15:52.27 | BBHoss | someone check this bug out if they have a chance http://bugs.digium.com/view.php?id=11946 |
15:53.00 | tzafrir | tzanger, hmm... pointing to dsp.c is a very cruel variation of 'use the source, Luke' |
15:53.07 | ZaVoid | thanks tzanger |
15:56.16 | ZaVoid | hey tzanger i do a more dsp.c | grep relaxdtmf and i get nothing |
15:56.42 | jeremy_g | what is that wiki wesbite where people normally search for asterisk info |
15:56.45 | jeremy_g | voip-info? |
15:56.49 | ZaVoid | ahh case senseitive |
15:56.51 | ZaVoid | i found it |
15:57.02 | ZaVoid | <PROTECTED> |
15:57.22 | nebojsajsimic | how to mark hungup event on any extension ??? |
15:57.39 | nebojsajsimic | can it be something like h,1 ..... |
15:57.40 | nebojsajsimic | ??? |
15:57.42 | nebojsajsimic | or not |
15:59.40 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
16:01.19 | ZaVoid | meh relaxdtmf dind't work very well for digits entered quickly |
16:01.34 | drmessano | so enter them slower |
16:01.43 | ZaVoid | yeah i know that |
16:01.52 | ZaVoid | but not everyone will... |
16:02.15 | drmessano | Success is only achieved through repeated failure |
16:02.24 | drmessano | They will learn...... eventually |
16:03.13 | ZaVoid | yeah you'd think |
16:03.29 | *** join/#asterisk ManxPower (n=manxpowe@175.sub-75-203-57.myvzw.com) |
16:07.04 | *** join/#asterisk _gm (n=gmustafa@58.27.175.222) |
16:08.00 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
16:09.03 | *** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
16:12.27 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
16:12.27 | *** mode/#asterisk [+o russellb] by ChanServ |
16:16.41 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
16:18.40 | *** join/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
16:19.05 | *** join/#asterisk zoid99 (n=chris@router.asteriasgi.com) |
16:21.52 | nebojsajsimic | work if you enable the "*" hangup |
16:22.05 | nebojsajsimic | how to enable * hungup |
16:23.46 | *** join/#asterisk micander (n=Michael_@Full-Service-Travel-1157986.cust-rtr.pacbell.net) |
16:24.16 | *** join/#asterisk GrumpManAtWork (n=meanderi@pool-72-78-33-219.phlapa.east.verizon.net) |
16:24.47 | GrumpManAtWork | anyone using new xml syntax for notifies on polycom phones ? |
16:27.17 | agx | when i transfer using features.conf is there a channel variable with the number of who started the transfer?? |
16:28.29 | *** join/#asterisk fl1p (n=fl1p@port-83-236-208-172.static.qsc.de) |
16:28.39 | *** join/#asterisk lakesolon (n=blake@63.231.182.86) |
16:28.55 | [TK]D-Fender | GrumpManAtWork: what "notify"? Which SIP? |
16:28.57 | ManxPower | nebojsajsimic: "core show application dial" |
16:29.23 | ManxPower | agx: if there is it would be documented in channelvariables.txt in the Asterisk source code. |
16:29.32 | *** join/#asterisk Bourrelle (n=Bourrell@132.207.156.100) |
16:30.06 | LakeSolon | Anyone know how to disable UPnP in Asterisk? (part of TrixBox) |
16:30.10 | Bourrelle | One quick question, if I close a video session with asterisk, does asterisk will send me an ack for the session ending ? |
16:30.33 | hmmhesays | seriously what could possibly cause a guy to shoot his wife in front of kids |
16:30.35 | ManxPower | LakeSolon: UPnP is not part of Asterisk Ask on the correct channel like *gasp* #trixbox |
16:30.39 | Bourrelle | ill trying to make the sequence diagram |
16:31.02 | ManxPower | Bourrelle: that might be a question for #asterisk-dev |
16:31.10 | Bourrelle | thx |
16:32.21 | LakeSolon | ManxPower: well I didn't think it was part of asterisk either, but UPnP is an application-level proto, so it's gotta be asterisk doing it. Either that or it's traversing my NAT by *magic* =p |
16:33.05 | LakeSolon | ManxPower: I just mentioned it's a trixbox install if that might be a clue as to an unusual default config. |
16:33.55 | ManxPower | LakeSolon: ALL of Trixbox uses unusual configs |
16:33.57 | ManxPower | ~trixbox |
16:33.58 | 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 |
16:34.48 | ManxPower | LakeSolon: Huh? Asterisk does not support UPnP. Asterisk has it's own SIP specific NAT features. |
16:36.20 | mintee | at the CLI, `database show` shows no sign of the following key, however this code http://pastebin.ca/895147 goes straight to 1000. |
16:36.44 | mintee | am i missing something? |
16:36.48 | [TK]D-Fender | LakeSolon: That is third-party stuff that has nothing to do with *. Go use their support resources. |
16:37.23 | agx | ManxPower, unluckly that files does not that that DIALEDPEERNUMBER and NAME are availble only on blindtransfer while attended trasnfer is a bit broken :-P |
16:37.24 | [TK]D-Fender | mintee: pastebin your database dump |
16:37.44 | ZaVoid | what debug would i need to do capture the DTMF.. using rfc2833.... an rtp debug? |
16:38.16 | [TK]D-Fender | agx: thats because an attended transfer is a new call where the audio is passed off. a blind transfer alerts the server to take the audio immediately and attach it to the new invite. |
16:38.36 | mintee | [TK]D-Fender http://pastebin.ca/895150 |
16:38.41 | agx | [TK]D-Fender, i see but there is a way to know who started the atxfer? |
16:39.05 | ManxPower | agx: attended transfer is considered a 3-way call. |
16:39.08 | agx | [TK]D-Fender, i tried DumpChan() and its quite empty |
16:39.17 | mintee | unless it's picking up the /AMPUSER/2957/voicemail, which doesn't match the DID |
16:39.18 | ManxPower | so, whoever the call came from is who did the transfer |
16:39.27 | [TK]D-Fender | mintee: I see... |
16:39.39 | [TK]D-Fender | mintee: exten => s,1,GotoIf(DB_EXISTS(voicemail/${DNID})?1000:2) <-- because you have fogotten how to EVALUATE a function. |
16:39.39 | ManxPower | mintee: Why do you think we can help with a system that was originally setup for AMP? |
16:39.54 | [TK]D-Fender | ManxPower: nope, obvious dialplan error |
16:40.09 | ManxPower | [TK]D-Fender: You're so smrt! |
16:40.09 | funxion | [TK]D-Fender can you take a look at http://www.pastebin.ca/895151 its the debug from a dropped call |
16:40.28 | agx | ManxPower, true but the callerid (strange) is from the original group :-) not the phone that answered the call [ i call 100,Dial(SIP/A&SIP/B), SIP/A atxfer to SIP/C |
16:41.16 | ManxPower | Just to be pedantic, you transfer to an EXTENSION, not a device. |
16:41.34 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
16:42.17 | ManxPower | funxion: "Cause: Wrong message (98)," Looks to me like a signalling issue. |
16:42.44 | funxion | but Im not getting any errors on the circuit |
16:42.52 | ManxPower | yes you are, I just posted it. |
16:43.11 | ManxPower | What IS your signaling set to? |
16:43.13 | [TK]D-Fender | ManxPower: O S I M! |
16:43.13 | mintee | [TK]D-Fender, do explain... from GotoIf(condition?label1[:label2]) where label1 = true and label2 = false.... as well as DB_EXISTS(<family>/<key>) ... I'm not sure how GotoIf(DB_EXISTS(voicemail/${DNID})?1000:2) isn't working. It was working about 15 minutes ago, until I put the Set() function in there |
16:43.26 | [TK]D-Fender | mintee: ${} <------- |
16:43.40 | [TK]D-Fender | mintee: You are not evaluating your function. |
16:43.44 | funxion | pri_net switchtype=5ess |
16:43.57 | agx | ManxPower, yes only variable i have is TRANSFERERNAME=SIP/A-083d42b8 ... unless patching res_features.c i see no way of knowing who answered the group call.... |
16:43.58 | ManxPower | funxion: and does it start working if you try setting it to national? |
16:44.12 | ManxPower | agx: You are looking for CALLERID |
16:44.24 | ManxPower | as you are not really doing a transfer, you are doing a THREE_WAY CALL. |
16:44.44 | ManxPower | attended transfers are basically 3-way calls where the person that does the 3-way call drops out of the call |
16:44.46 | agx | ManxPower, caller id is set to "100" that is the extension that make Dial(SIP/A&SIP/B) |
16:44.48 | funxion | ManxPower its not that it doesnt work, it dropps calls randomly |
16:45.07 | ManxPower | funxion: are you going to argue with me or are you going to try my suggestions? |
16:45.18 | funxion | since I've changed the timing source and updated the zaptel version its gotten a little better |
16:45.22 | funxion | going to change it |
16:45.29 | funxion | jsut giving more info |
16:45.32 | ManxPower | 5ess and national are VERY, VERY similar. In fact they are interchangable if you want random problems. |
16:45.42 | funxion | lol |
16:45.59 | funxion | national would be NI2 in a nortel? |
16:46.16 | mintee | [TK]D-Fender, thanks... I got it. Not sure i understand why, but i'll figure it out. |
16:46.27 | ManxPower | no, national is National ISDN 2, it is a vendor agnostic protocol. |
16:46.53 | agx | ManxPower, the issue is only when using features.conf inband dtmf; with SIP phones with the transfer button (snom, junkstream, etc.) i don't have this problem the callerid is correct |
16:47.28 | [TK]D-Fender | mintee: Because its a function that you have to evaluate. "NoOp(CALLERID(num))" winn NOT show you the caller ID, but rather "CALLERID(num)". "NoOp(${CALLERID(num)}) will evaulate the functionand return the result |
16:47.30 | cpm | poppr0n! |
16:47.35 | cpm | cornpr0n |
16:47.38 | [TK]D-Fender | mintee: this is dialplan 101 |
16:48.16 | ManxPower | mintee: you might want to learn how to use the dalplan 8-) |
16:48.50 | mintee | yes, i'm working on asterisk 101.. :D |
16:48.54 | mintee | :P |
16:49.38 | [TK]D-Fender | mintee: reason it came back as "true" for your gotoif, is because it lokos to see if everything before the "?" comes back as "0". if it does, then its false. ANY other value is considered "true" |
16:49.42 | jameswf | anyone know of any modern nethdlc docs |
16:50.00 | [TK]D-Fender | mintee: Including your unprocessed attemtp to call a function which is jsut a pile of text. |
16:50.13 | mintee | gotcha, so to eval, it must be within ${}... |
16:50.25 | [TK]D-Fender | mintee: Correct. |
16:50.32 | *** join/#asterisk madgeek (n=madgeek@i2router.fi.edu) |
16:50.40 | [TK]D-Fender | mintee: the same way you reference a variable |
16:50.57 | mintee | thanks... yeah, i grasped the idea at first, with the ${NDID} but didn't realize the original function. |
16:51.01 | mintee | exactully |
16:51.35 | [TK]D-Fender | abslutly |
16:51.36 | [TK]D-Fender | :) |
16:51.37 | ManxPower | NDND? |
16:51.57 | *** part/#asterisk madgeek (n=madgeek@i2router.fi.edu) |
16:52.08 | [TK]D-Fender | ${DNID} |
16:52.16 | outtolunc | ${TYPO} |
16:52.28 | mintee | ${FAIL} |
16:52.45 | outtolunc | ${KILL} |
16:53.00 | [TK]D-Fender | ${STFUKPLZTHXBIBI} |
16:53.02 | jameswf | ~nethdlc |
16:58.55 | *** join/#asterisk barsik76 (n=barsik3@mail.sigmagroup.com) |
16:58.58 | *** part/#asterisk joshkidd (n=josh@adsl-068-209-028-087.sip.asm.bellsouth.net) |
17:00.00 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
17:00.10 | *** join/#asterisk freezey (n=freezey@gw.mypublisher.com) |
17:00.13 | barsik76 | hi guys. |
17:00.30 | [T]ank | what is the preferred monitor type in queues.conf? |
17:00.34 | freezey | [TK]D-Fender: got that visio diagram done.. finished it yesterday |
17:00.35 | [T]ank | mixmonitor or monitor |
17:00.40 | barsik76 | i'm trying to setup an asterisk pbx but running into a problem with dnd. |
17:00.54 | barsik76 | it's not visibile on the fop. |
17:01.19 | barsik76 | it seems like polycom integrated a new feature in their 2.2 firmware that allows server based dnd. |
17:02.02 | barsik76 | does anyone know anything about voIpProt.SIP.serverFeatureControl.dnd ? |
17:02.20 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
17:03.54 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
17:08.44 | [T]ank | reading through the docs and other web pages I am not finding the pros and cons to monitor vs mixmonitor. can anyone give me their opinion on it? |
17:11.33 | [TK]D-Fender | freezey: \o/ |
17:12.01 | [TK]D-Fender | barsik76: * is not able to process that yet |
17:12.16 | funxion | has anyone done a asterisk cluster? |
17:12.37 | [TK]D-Fender | barsik76: I had a nifty idea on how to implement it, but it hasn't been done to date |
17:14.23 | freezey | [TK]D-Fender: so how should we get this over to you? so you could take a look |
17:14.34 | [TK]D-Fender | freezey: imgshack, etc |
17:14.44 | freezey | k |
17:14.46 | SteveTotaro | why isn't bristuff part of the asterisk source? |
17:14.47 | barsik76 | [TK]D-Fender Sorry. |
17:14.49 | freezey | 1 sec |
17:15.12 | [TK]D-Fender | SteveTotaro: because its authors didn't or couldn't dislaim it to Digium |
17:15.16 | [TK]D-Fender | disclaim. |
17:15.26 | barsik76 | [TK]D-Fender: are there any plans regarding implementation in * 1.6? |
17:15.37 | [TK]D-Fender | barsik76: I already answered that. |
17:16.04 | *** join/#asterisk AndyGraybeal (n=andy@node191.34.251.72.1dial.com) |
17:16.19 | SteveTotaro | i find it hard to believe they would not disclaim patches |
17:16.40 | [TK]D-Fender | SteveTotaro: Maybe they use other GPL code they have no rights to <- |
17:16.52 | SteveTotaro | yeah, i suppose |
17:16.58 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
17:17.01 | SteveTotaro | crappy GPL |
17:17.34 | SteveTotaro | free as in beer I say! |
17:18.04 | Qwell | doesn't bristuffed cause a lot of issues as well?.. |
17:18.18 | barsik76 | [TK]D-Fender: I'm new to asterisk I'm not sure what is Mantis |
17:18.47 | [TK]D-Fender | barsik76: the bug tracker that tracks new pattches, etc |
17:19.08 | Qwell | barsik76: bugs.digium.com |
17:20.52 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
17:21.20 | SteveTotaro | i don't know but i just spent the last 24 hours fighting with bristuffed |
17:21.38 | SteveTotaro | problem was the upgraded kernel |
17:21.57 | SteveTotaro | system clock was jumping all over the place |
17:22.28 | SteveTotaro | tzafrir is my hero ;) |
17:22.33 | *** part/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
17:22.50 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
17:25.58 | *** join/#asterisk esaym (n=user@72.183.198.134) |
17:26.41 | bsdwarrior | tkd-fender Im really stuck with setting the userfield for cdr in a perl script. can you point me in the right direction |
17:28.16 | [TK]D-Fender | bsdwarrior: set it before you do your outbound dial for your first leg of the call. |
17:32.15 | *** join/#asterisk RoyK (n=roy@ip-107-15-149-91.dialup.ice.no) |
17:32.19 | patrick-- | Hey i keep getting this error: |
17:32.21 | patrick-- | Got EVENT_FACILITY but we don't have a ch! |
17:32.30 | patrick-- | how can i fix it? |
17:32.41 | patrick-- | its a chan_misdn error |
17:32.42 | bsdwarrior | tkd-fender so send a separate action command and set the variable ? |
17:33.16 | [TK]D-Fender | bsdwarrior: You have to set the FUNCTION, which means you have to do it in the dialplan. |
17:33.38 | outtolunc | just add 'Variable: __VAR=VALUE' to the bottom of the rest of the originate |
17:33.39 | [TK]D-Fender | bsdwarrior: CDR is not a "variable". That tells you you have to be executing dialplan code to set it. Think on that |
17:34.04 | outtolunc | then you can set the cdr(yadda) to it on the way back in |
17:35.30 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
17:38.26 | patrick-- | Hey i keep getting this error: |
17:38.28 | patrick-- | Got EVENT_FACILITY but we don't have a ch! |
17:39.00 | [TK]D-Fender | patrick--: And we only heard you ask that 5 minutes ago... |
17:39.07 | patrick-- | sorry |
17:41.04 | patrick-- | google wont give me no answer |
17:41.13 | outtolunc | This may come from a call we don't know nothing about, so we ignore it. |
17:41.47 | patrick-- | mhh, well its to an extension mentioned on the dialplan |
17:42.29 | outtolunc | having an exten in a dialplan does not a channel make... until you start a pbx event on it |
17:43.01 | patrick-- | outtolunc: how do i "make" a channel? i though my misdn was setup correctly |
17:43.20 | b11d|bbl | Did the behaviour of Comedian Mail change at all between 1.2 and 1.4? |
17:43.55 | outtolunc | either you are are attempting to do something (like a hint/etc) on a channel that is not created yet, or has already gone bye bye |
17:44.06 | outtolunc | would be my guess |
17:45.30 | b11d | I am finally moving my PBX to 1.4 and would really like to know if my end users will notice any difference in their voicemail. I dont currently see any changes, but would like to know for certain. |
17:45.43 | b11d | changes.txt doenst list anything that I can see.. |
17:45.51 | barsik76 | [TK]D-Fender: i don't seem to be able to find any information regarding serverFeatureControl.dnd on bugs.digium.com |
17:45.52 | outtolunc | actually this is a facility event.. so it would be a network issue |
17:45.54 | b11d | err upgrade.txt |
17:46.06 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
17:46.08 | barsik76 | [TK]D-Fender: is it under some different name? |
17:48.48 | [TK]D-Fender | barsik76: I told you it doesn't EXIST. It is not supported yet. You can stop looking. |
17:49.08 | RoyK | how does asterisk compare to this? http://www.museumoflondon.org.uk/piclib/images/%5CMID%5C0330001612_5mb.jpg :D |
17:49.41 | drmessano-LT | Asterisk takes the people out of the loop... the weak link :) |
17:50.25 | RoyK | hehe |
17:50.37 | b11d | anyone? any changes to the voicemail? |
17:50.43 | b11d | I will assume none.. |
17:54.20 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
17:55.47 | *** join/#asterisk rcahilig (n=test@203.115.187.98) |
17:56.19 | [TK]D-Fender | b11d: nothing really. The recordings are a different picth because of their rebuild |
17:56.36 | b11d | as long as the behaviour didnt change, thats all I care about.. Thanks TK |
17:57.07 | [TK]D-Fender | b11d: No, functions the same. |
17:57.09 | rcahilig | Hi, How do I install PHPAGI, I cannot find any tutorial on google installing PHPAGI |
17:57.24 | [TK]D-Fender | rcahilig: ... |
17:57.26 | [TK]D-Fender | ~book |
17:57.27 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
17:58.36 | [TK]D-Fender | rcahilig: http://phpagi.sourceforge.net/ |
17:59.02 | *** join/#asterisk klictel (n=klictel@atelka.info) |
18:00.23 | b11d | while im thinking of it, is there any way to enforce a voicemail password policy? I'd like users to have to change their VM passwords every now and again.. |
18:00.59 | [TK]D-Fender | b11d: You could do this with excessive dialplan scripting, but nothing convenient. |
18:02.48 | b11d | ah.. its probably overkill anyways. |
18:04.09 | *** join/#asterisk worgil (i=worgil@88.252.187.193) |
18:07.04 | *** join/#asterisk Docfxit (n=none@ip-64-32-143-214.lax.megapath.net) |
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18:14.08 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
18:15.48 | *** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
18:17.40 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
18:21.50 | *** join/#asterisk na0mi (n=kieran@mail.datadream.co.uk) |
18:22.58 | jeremy_g | whats this -lwrap library that asterisk 1.6 needs? |
18:24.18 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
18:24.28 | na0mi | Hi I just installed asterisk last night and am very happy with it - however something strange has happened - I was able to recieve externel calls through my voip provider but now I get this message and the call clears - but I swear I never changed a thing - any ideas? http://pastebin.ca/895294 |
18:27.47 | keith4 | na0mi: codec negotiation problem? |
18:28.05 | jeremy_g | do i need to repeat myself to get answered? |
18:28.13 | keith4 | what codecs are you accepting? |
18:28.15 | patrick-- | How do i create a channel for my HFC cards to be able to communicate at? |
18:28.22 | keith4 | jeremy_g: no, but you might have to be patient |
18:28.23 | na0mi | keith4: have allow all in and it did work |
18:28.32 | *** join/#asterisk guillote_GNU (n=guillote@host63.201-253-22.telecom.net.ar) |
18:28.50 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
18:29.21 | barsik76 | [TK]D-Fender: is it possible to use presence to view whether the user's phone on dnd? |
18:29.22 | keith4 | jeremy_g: presumably it's lib wrap ... |
18:29.29 | keith4 | TCP wrapper library |
18:29.46 | keith4 | are you asking a trick question? |
18:30.32 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
18:31.50 | [TK]D-Fender | barsik76: No |
18:31.51 | na0mi | lol - just got an email from my voip provider they have problems their end - guess its that |
18:33.51 | patrick-- | can anyone tell me how to setup channels with misdn? |
18:34.39 | barsik76 | [TK]D-Fender: is there a way to view user's dnd status with polycom phones without using the *78/*79? |
18:35.20 | barsik76 | [TK]D-Fender: i would like the phones to dispay if it's on dnd or not... |
18:38.42 | *** join/#asterisk moellerdk (n=chatzill@4806ds1-hl.0.fullrate.dk) |
18:40.12 | jameswf | jbot: tell patrick-- about wikis |
18:40.17 | hmmhesays | doesn't dnd change the subscription status? |
18:41.04 | plik | na0mi: Hello :) |
18:41.13 | na0mi | plik: Hi |
18:42.00 | [TK]D-Fender | barsik76: No. |
18:42.13 | [TK]D-Fender | hmmhesays: No. |
18:42.44 | hmmhesays | hmm it should |
18:44.36 | [TK]D-Fender | hmmhesays: No, DND is a transparent thing that only causes a reject and with a reason code based on provisioning. |
18:44.37 | kyron | ~book |
18:44.38 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
18:44.43 | kyron | Synoptic, ^^^ |
18:44.48 | kyron | Synoptic, read chapter 5 |
18:45.05 | hmmhesays | gotcha |
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18:58.26 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
19:00.33 | *** join/#asterisk murdmath (n=vircuser@17.sub-70-193-177.myvzw.com) |
19:07.27 | *** join/#asterisk servergod (n=servergo@70.97.159.120) |
19:07.42 | servergod | hello all!!! |
19:09.19 | BBHoss | sup dog! |
19:10.11 | servergod | trying to set up asterisk as a B2BUA. have 1 trunk going voip to cisco gateway to pstn, and another is another asterisk. |
19:11.06 | *** join/#asterisk fl1p (n=fl1p@ip-90-186-3-96.web.vodafone.de) |
19:11.18 | servergod | i need for asterisk to send call to B2BUA, then have that one run it through the outbound routes, then out the other sip trunk |
19:11.35 | servergod | but the B2B |
19:11.37 | [TK]D-Fender | servergod: what is this other B2BUA? |
19:11.41 | servergod | asterisk |
19:11.54 | servergod | sends the calls to s|1 |
19:12.05 | [TK]D-Fender | servergod: You are thowing terms an conenction info around without making much sense. Draw a picture. |
19:12.18 | servergod | k brb |
19:15.02 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
19:15.14 | drmessano-LT | Can I integrate * and OU812? |
19:16.54 | [TK]D-Fender | drmessano-LT: Sure.... eat'em & smile :p |
19:17.22 | [TK]D-Fender | drmessano-LT: Let me know when you've found the reference :) |
19:17.41 | drmessano-LT | hmmm |
19:18.10 | drmessano-LT | The eat'em and smile isn't obvious to me lol |
19:18.26 | [TK]D-Fender | drmessano-LT: its both together, and I've dropped other clues... |
19:18.56 | drmessano-LT | Well, if it's more than just a vague Van Halen reference, i'm lost |
19:20.44 | [TK]D-Fender | drmessano-LT: When David Lee Roth left VH in 1985 to go solo his first release was named in revenge "Eat'em And Smile". When Sammy Hagar filled his place with VH, they responded in kind with the title of their next rease "OU812" |
19:20.52 | [TK]D-Fender | drmessano-LT: Mutual snub :) |
19:21.24 | drmessano-LT | Ahhh... Little trivia I had no clue about.. cool |
19:22.11 | [TK]D-Fender | drmessano-LT: Another useful tidbit for you.... |
19:22.30 | [TK]D-Fender | drmessano-LT: I used to play with a VH cover band, and did DLR's solo stuff too... |
19:22.34 | *** join/#asterisk southtel (n=southtel@68-114-17-226.dhcp.gwnt.ga.charter.com) |
19:22.42 | drmessano-LT | Oh cool |
19:23.35 | drmessano-LT | I've actually had to forcibly boycott Van Halen |
19:24.15 | drmessano-LT | Would you believe that on TWO occasions, I have had two seperate cars break down on me while listening to "Running with the Devil" |
19:24.49 | [TK]D-Fender | drmessano-LT: You should boycott cars. Twice they've broken down on you while being on the road! |
19:25.00 | [TK]D-Fender | drmessano-LT: Go bike-boy! |
19:25.53 | drmessano-LT | LOL |
19:25.59 | drmessano-LT | Well, they were shit cars |
19:26.20 | drmessano-LT | I learned that a car payment is a reasonable tax to keep on the road |
19:26.34 | drmessano-LT | But I couldn |
19:26.42 | drmessano-LT | But I couldn't ignore that coincidence |
19:26.51 | drmessano-LT | Eddie was speaking to me |
19:26.54 | drmessano-LT | Channeling me |
19:27.12 | drmessano-LT | KILL KILL KILL, HE SLEWED WITH HIS AXE |
19:27.12 | [TK]D-Fender | I just lost a car this week (1998 Taurus SW shitbox), and replaced with a very nice 1998 Chevy Malibu (soptless) |
19:27.27 | *** join/#asterisk emmix-devin (n=devinsai@c-68-51-54-72.hsd1.ar.comcast.net) |
19:27.34 | [TK]D-Fender | drmessano-LT: if you want "Eddie" to channel.... thats what Iron Maiden is for :P |
19:27.42 | drmessano-LT | heh |
19:27.47 | J4k3 | doh-mestic |
19:27.48 | emmix-devin | any here used mediatrix 1124 boxes |
19:27.49 | [TK]D-Fender | Spotless even. |
19:27.57 | [TK]D-Fender | emmix-devin: I have, as have others. |
19:28.18 | emmix-devin | how was the quality of the box |
19:28.54 | [TK]D-Fender | emmix-devin: Decent, friendly to use, typical features. How many ports do you need? |
19:28.58 | drmessano-LT | So [TK]D-Fender, I MUST ask.. |
19:29.21 | emmix-devin | 160 ports |
19:29.31 | drmessano-LT | Are you OK with Van Hagar, or did Van Halen die when DLR left? |
19:29.31 | [TK]D-Fender | emmix-devin: Ok, not a bad idea. |
19:29.52 | [TK]D-Fender | drmessano-LT: Mixed. Wasn't the same, but Gary Cherone KILLED VH. |
19:30.09 | [TK]D-Fender | drmessano-LT: DLR si supposed to be working on a comeback album with them right now. |
19:30.27 | [TK]D-Fender | Gary Cherone = good with "Extreme".... but not VH |
19:30.38 | drmessano-LT | I agree.. I liked Hagar.. it wasn't AT ALL the same, but still good.. Gary Cherone just.... |
19:30.45 | [TK]D-Fender | WRONG!!! |
19:30.51 | drmessano-LT | Too much Cocaine... |
19:30.51 | [TK]D-Fender | lol |
19:31.04 | emmix-devin | so you think mediatrix is the way to go for that many analog ports, we have used Xorcom and Channel banks, but have not been happy with them |
19:31.05 | drmessano-LT | Rotted their brains on that decision |
19:31.09 | tnt_ | #`Has anyone the guide for the 3.0.0 fw for Polycom phones ? |
19:31.18 | [TK]D-Fender | tnt_: www.polycom.com |
19:31.42 | [TK]D-Fender | emmix-devin: Yeah, for that kind of density, sure |
19:31.54 | [TK]D-Fender | emmix-devin: Zaptel FXS = ASS |
19:31.57 | drmessano-LT | Forget THEIR decision, what made Gary Cherone think he could rock out, Van Halen style? |
19:32.13 | [TK]D-Fender | emmix-devin: This way you can have redundency and reduced overhead. |
19:32.19 | drmessano-LT | A; More drugs |
19:32.27 | tnt_ | [TK]D-Fender: nope ... The latest version of their site is 2.2 ... I saw the 3.0 existed only a brief moment when they put the fw by mistake ... |
19:32.29 | J4k3 | I'm not a big sammy hagar fan, but I'd kill david lee roth given we were in the same room |
19:32.34 | emmix-devin | thanks |
19:32.35 | [TK]D-Fender | E; Even more drugs still... |
19:32.51 | [TK]D-Fender | tnt_: the admin guide is on their site. |
19:33.09 | scooby2 | any idea how to make zaptel not cause a kernel panic on smp kernels with one cpu? |
19:34.51 | tnt_ | scooby2: not using a smp kernel ? |
19:35.19 | scooby2 | almost all linux distributions come with smp kernels these days |
19:36.04 | ManxPower | scooby2: you have some OTHER issue. Zpatel does not kernel panic just because you have an SMP kernel on a UP system |
19:36.33 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
19:36.34 | tzafrir | scooby2, hmmm... can you reproduce such a kernel panic? what kernel? |
19:36.46 | tzafrir | what hardware involved? what version of Zaptel? |
19:36.50 | scooby2 | 2.6.18-53.1.6.el5 |
19:37.08 | scooby2 | zaptel 1.4.8 |
19:37.11 | scooby2 | Dell 1850 |
19:37.23 | servergod | got it |
19:37.27 | ArchSSM | any analog/bri/pri zaptel cards? |
19:37.27 | scooby2 | 2.8ghz xeon w/ 1gb ram |
19:37.39 | scooby2 | te212p |
19:37.55 | servergod | k here is the pic i drawded http://foshizzlenet.net/myspaceimg/b2bua.png |
19:37.57 | ArchSSM | and it reboots when you load the module? ... or? |
19:38.11 | scooby2 | after about 20-25 calls it kernel panics |
19:38.41 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
19:38.43 | scooby2 | trying to move from sangoma to this new card |
19:39.15 | J4k3 | its an old dell, the machine is probably just getting tired. |
19:39.25 | [TK]D-Fender | servergod: Why 2 * boxes? |
19:40.47 | scooby2 | i can confirm it on 2 identical machines |
19:41.01 | servergod | want to use the b2bua as a main gateway for all of our devices. We have about 6 customers with 2651xm that feed pri to vendor pbx's. |
19:41.31 | servergod | each of those need full dial plans. it would just be easier to send all calls to the b2bua, |
19:41.52 | [TK]D-Fender | servergod: Ok, so the 2811 does what exactly? |
19:42.47 | Qwell | servergod: drawded? |
19:43.02 | BadHorsie | can i call ChanSpy several times over the same channel at the same time? |
19:44.04 | tzafrir | scooby2, any chance you have a trace from that panic? |
19:44.42 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088937109.dsl.bell.ca) |
19:45.07 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
19:45.40 | [TK]D-Fender | BadHorsie: I think there may be an opening for you at AT&T |
19:45.55 | drmessano-LT | ROFL |
19:45.59 | drmessano-LT | pwn3d |
19:46.16 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
19:46.17 | *** join/#asterisk maszlo (n=reckenro@65.223.240.146) |
19:46.33 | jameswf | I haer AT&T voted dfor ron paul |
19:46.52 | jameswf | stupid korean keyboard |
19:46.55 | maszlo | i am having problems getting the callerID to get sent along when having a line setup on call foward |
19:47.03 | maszlo | The fowarded call carrys the callerid of an internal number, what can i do to make it the actual caller? |
19:47.36 | drmessano-LT | AT&T voted for Ron Paul |
19:47.41 | drmessano-LT | So they're the "one" |
19:47.44 | jameswf | maszlo: what is your trunk type |
19:47.59 | maszlo | sip |
19:48.20 | maszlo | its coming from a pri |
19:48.22 | jameswf | some sip providers and ron paul do not allow caller ID spoofing |
19:48.42 | maszlo | lol |
19:49.02 | J4k3 | ~ron paul |
19:49.03 | jbot | ZOMG RONPAULAPPLEUBUNTU FOR PRESIDENT |
19:49.04 | BadHorsie | [TK]D-Fender, sorry for trying to learn asterisk :P |
19:49.20 | jameswf | BadHorsie: is forgiven |
19:49.28 | jameswf | now run away |
19:49.31 | jameswf | :) |
19:49.34 | hmmhesays | atlantis flight looks good |
19:49.58 | [TK]D-Fender | BadHorsie: Just that you're trying to seriously gang-rape your callers. Like... WTF? |
19:50.34 | maszlo | where should i start to figure out if it is the provider to blame? its verizon, we were able to make calls on the pri before we even had the numbers ported, w/o a callerid |
19:50.37 | J4k3 | good sip providers don't care, since they know they're used for termination for numbers that they don't own. |
19:51.04 | maszlo | i think that it is something in the configureation, i am just new with this system |
19:51.12 | maszlo | glad it makes calls ;) |
19:51.38 | jameswf | For a while I was calling my brother with his cell phone... I tried calling his sellphone and got in to his voice mail t-mobile rocks |
19:52.20 | maszlo | i think most cell phones are like that |
19:52.50 | maszlo | or were you spoofing the caller id to get into his voicemail? |
19:52.53 | [TK]D-Fender | one piece of happy news today.... looking like Mitt's dropping out of the race... |
19:53.07 | drmessano-LT | [TK]D-Fender: How can repeatedly ChanSpy on a caller, record their convo, and then e-mail it to our CEO and CC our personal GMAIL inboxes? |
19:53.20 | drmessano-LT | YAY |
19:53.27 | [TK]D-Fender | drmessano-LT: Forgot the cherry on top! |
19:53.46 | drmessano-LT | oh |
19:54.03 | drmessano-LT | [TK]D-Fender: How can we repeatedly ChanSpy on a caller, record their convo, and then e-mail it to our CEO and CC our personal GMAIL inboxes AND post it on YouTube? |
19:54.24 | drmessano-LT | Shit, left out MySpace |
19:54.27 | [TK]D-Fender | drmessano-LT: And no, not e-mailing... those dinosaurs force other concessions like warranting automatic "burning" of the recodings on LP and snail-mailing them. |
19:54.36 | drmessano-LT | ROFL |
19:54.48 | [TK]D-Fender | now THAT pwns |
19:55.02 | drmessano-LT | yes, yes it does |
19:55.32 | drmessano-LT | "IIIIII can't lie.... it's FOURTY FIIIIIVE" |
19:56.07 | J4k3 | [TK]D-Fender still listens to wax tube recordings. |
19:56.28 | drmessano-LT | I want one of those Laser LP players.. |
19:56.35 | J4k3 | me too |
19:56.53 | J4k3 | I recently had opportunity to get a couple albums (same album, two copies) ripped with one |
19:57.04 | maszlo | can i get a direction to where i should look to straighten the callerid, actaully they are all getting the same number, all 15 phones make calls with the same outbound callerid, so i guess that means that spoofing is allowed |
19:57.11 | J4k3 | trying to make a high quality copy of an album whos sources got destroyed 25+ years ago |
19:57.20 | drmessano-LT | Ah |
19:57.54 | *** join/#asterisk CVirus (n=GoD@82.201.174.232) |
19:57.57 | drmessano-LT | We still have turntable in one of our production rooms here.. I haven't put a stylus on it in years lol |
19:58.18 | drmessano-LT | I'm willing to bet the motor doesn't work |
19:58.22 | J4k3 | haha. a stereo isn't a stereo without a turntable. ;) |
19:58.39 | J4k3 | I *hate* CDs... they're a fucking high dollar ripoff of lousy quality and horrible longevity |
19:59.05 | drmessano-LT | Purple Haze on LP.. it just get's NO better |
19:59.12 | drmessano-LT | gets* |
19:59.54 | drmessano-LT | I listen to the Beatles on CD and there's no comparison to the LP's I grew up with |
20:01.13 | J4k3 | well, lots of music just doesn't sound right otherwise... for example heart - dreamboat anne - magic man... it sounds completely different off a digital source because you're *supposed* to hear the tonearm resonance |
20:02.09 | J4k3 | it may not have been engineered to sound that way, but thats the way everyone got to hear it for like 15 years... then CDs come along and change it. |
20:02.09 | maszlo | record players are nice, but headphones are a key part of the equation as well |
20:02.18 | drmessano-LT | Yes.. and listening to any of the Doors CDs is the same way... |
20:02.18 | J4k3 | but I'll be the first to say 33.3 was too damned slow |
20:02.29 | J4k3 | what this world needed was more 45 rpm 2-disk releases |
20:02.50 | *** join/#asterisk javar (n=javar@69.79.134.24) |
20:02.55 | maszlo | i got a pair a akg muffs last month best set i have ever owned |
20:02.56 | J4k3 | yeah... gotta have a good set of headphones for any musical enjoyment |
20:03.02 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-03c2ea82100cd452) |
20:03.38 | J4k3 | I got a pair of cheapish sennheiser closed-back |
20:03.40 | b1ch0 | drmessano, are you there ??? need help again !!!! |
20:03.41 | J4k3 | pro 280s I think |
20:03.47 | drmessano-LT | I am here |
20:03.58 | b1ch0 | how can i change default language (en) to es ? |
20:04.07 | J4k3 | they're not perfect, but they effectively sound better than my previous pair (HD600's) due to not letting in all the background noise. |
20:04.28 | maszlo | they noise canceling or something? |
20:04.35 | J4k3 | nah, just fully closed-back |
20:04.49 | J4k3 | about 30 db of natural rejection |
20:05.00 | drmessano-LT | language=es |
20:05.33 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
20:05.49 | maszlo | yeah its all about that akg k240 |
20:05.56 | b1ch0 | ok, but where do i change it ? iax.conf, sip.conf , zapata.conf |
20:06.56 | drmessano-LT | sip.conf at the very least |
20:07.01 | drmessano-LT | Not sure about the others |
20:07.10 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
20:08.39 | *** join/#asterisk Beirdo_ (n=gjhurlbu@unaffiliated/beirdo) |
20:10.29 | *** part/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
20:10.49 | maszlo | callerid?? can someone tell me where to adjust this? |
20:10.55 | *** 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 1.4.8 (2008/01/14), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
20:11.09 | drmessano-LT | YAY |
20:11.20 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
20:12.06 | drmessano-LT | Website hasn't been updated |
20:12.14 | drmessano-LT | or I am in cache hell |
20:14.02 | *** join/#asterisk servergod (n=servergo@70.97.159.120) |
20:15.14 | jameswf | 1.4.18 is release?? why dont i see that on the lists |
20:15.26 | Qwell | because you haven't looked |
20:15.48 | Qwell | (that, and they probably haven't been sent out yet. give the man a minute) |
20:15.57 | jameswf | no NOW :) |
20:16.08 | Qwell | k/ickban jameswf nub |
20:16.09 | Qwell | erm |
20:16.33 | Qwell | :D |
20:16.47 | drmessano-LT | It's about time.. What happened to monday night??? I WANT MY MONEY BACK |
20:17.49 | jameswf | oooh they are offering refunds now.... |
20:17.57 | J4k3 | "the boys in brown" takes on a whole new meaning. |
20:18.11 | *** join/#asterisk LeBowlingAlley (n=derek@71.16.158.170) |
20:18.36 | LeBowlingAlley | has anyone seen a problem of a Parked Call answering an incoming call? |
20:18.47 | Qwell | LeBowlingAlley: you're gonna need to be a little more specific |
20:18.52 | drmessano-LT | Now that 1.4.18 is out, HappyClownPBX is getting a refresh |
20:19.14 | drmessano-LT | honk! |
20:19.23 | jameswf | I realy need to start work on xobxirt |
20:19.36 | drmessano-LT | yes you do |
20:19.50 | drmessano-LT | Before I make HappyClownPBX public |
20:20.08 | LeBowlingAlley | A call over a zap channel came in and rang, and then eventually connected to a call that was in the parking lot |
20:25.55 | LeBowlingAlley | Qwell: http://pastebin.ca/895421 |
20:26.17 | LeBowlingAlley | The parked call answering the incoming call over a zap channel is at the end |
20:26.23 | *** join/#asterisk sx|lappy (n=sxpert@home.riquer.fr) |
20:28.09 | [TK]D-Fender | LeBowlingAlley: Yay, one of your callee's forwarded their phone to "71" picking up a parked call. Go fix their PHONE. |
20:28.31 | Qwell | or fix the user |
20:28.39 | LeBowlingAlley | Is that the 209 user? |
20:29.46 | maszlo | where do you set an outbound callerid? |
20:29.52 | *** join/#asterisk jhiver (i=jhiver@164-242.206-83.static-ip.oleane.fr) |
20:29.57 | jhiver | hi guys |
20:30.14 | jhiver | do you know a way to batch convert a bunch of .wav files to .g729 format? |
20:30.35 | Qwell | jhiver: use asterisk's convert CLI command |
20:30.35 | [TK]D-Fender | LeBowlingAlley: Feb 7 15:10:36 VERBOSE[20561] logger.c: -- Got SIP response 302 "Moved Temporarily" back from 192.168.4.109 |
20:30.45 | [TK]D-Fender | LeBowlingAlley: Feb 7 15:10:36 VERBOSE[9866] logger.c: -- Now forwarding Zap/5-1 to 'Local/71@from-internal' (thanks to SIP/209-09146bf8) |
20:30.49 | [TK]D-Fender | LeBowlingAlley: What do YOU think? |
20:30.57 | Qwell | and a simple for loop with asterisk -rx - assuming you have g729 codec licenses |
20:30.58 | LeBowlingAlley | :D Thanks. |
20:31.00 | *** join/#asterisk beighto (n=chatzill@12.176.156.130) |
20:31.25 | [TK]D-Fender | jhiver: Digium's site has a converter as well. |
20:31.36 | Qwell | [TK]D-Fender: really? |
20:31.36 | [TK]D-Fender | jhiver: depending how many you need to do. |
20:31.42 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
20:31.43 | [TK]D-Fender | Qwell: Used to... |
20:31.45 | *** join/#asterisk RoyKa (n=roy@ip-77-15-149-91.dialup.ice.no) |
20:31.50 | jhiver | about 150 |
20:31.54 | Qwell | I know we did at one point, but I thought that was for prompts from thevoice |
20:32.32 | *** join/#asterisk servergod (n=servergo@70.97.159.120) |
20:33.01 | Qwell | jhiver: do you have g729 licenses for asterisk? |
20:33.32 | jhiver | so first i batch convert to 8khz using sox, then i write a shell script to use asterisk -rx 'file convert foo.wav foo.g729' |
20:33.43 | jhiver | well i can always buy ONE |
20:33.51 | jhiver | that won't kill me |
20:33.54 | J4k3 | you need g729 licenses, then you gotta find g729 codecs that actually work :P |
20:34.57 | hmmhesays | mccain is staying pretty cool at his speech |
20:35.46 | jhiver | beside it looks like i have the g729 hx0rd codec on the box, oops |
20:35.50 | jhiver | which is crazy since i don't transcode at all on this box :) |
20:36.09 | [TK]D-Fender | hmmhesays: Yah, while C&L just said that if he makes it to office he'll make Cheney look like GHANDI |
20:36.36 | [TK]D-Fender | hmmhesays: Mccain = militant. |
20:36.45 | J4k3 | mccain = old gay turd. |
20:37.10 | jhiver | so will asterisk's 1.4 file convert utility be smart enough to convert to 8khz or do i need to do it first? |
20:38.15 | hmmhesays | [TK]D-Fender, militant not so much |
20:38.24 | hmmhesays | we're fscked if hillarybot3000 gets in |
20:39.00 | jhiver | aah politics :( |
20:39.01 | [TK]D-Fender | hmmhesays: This is Mr. "in Iraq for the next 100 years, and other wars to come" |
20:39.09 | *** join/#asterisk lgc (n=lgc@189.146.43.17) |
20:39.36 | hmmhesays | taking that statement way out of context |
20:39.39 | jhiver | here in france we have been fscked for years, and sarkozy is just the latest fuckage in a long line of fucked up presidents :) |
20:39.49 | lgc | Hi. What do I need to use Asterisk in order to send faxes from my computer? |
20:40.01 | hmmhesays | use callweaver |
20:40.03 | [TK]D-Fender | hmmhesays: Yes, Hillary is bad, so is every republican except Ron Paul who if he doesn't pick up his act FAST had better run as a Libertarian Party candidate for the final |
20:40.22 | hmmhesays | Ron Paul I would vote for if I knew he was going to get in |
20:40.36 | hmmhesays | you know obama talks a big game but he'll never follow through |
20:40.44 | J4k3 | Ron Paul is the sucker candidate |
20:41.02 | lgc | hmmhesays, was the callweaver answer for me? |
20:41.02 | J4k3 | he's the person the republicans are using to get people with an IQ over 80 to vote republican. |
20:41.07 | hmmhesays | lgc, yes |
20:41.10 | [TK]D-Fender | hmmhesays: He will probably not get anywhere as a Republican, but he can make a place on the Ballot that counts, but he's so conservative noone knows who he is! |
20:41.19 | J4k3 | he has no chance, nor does his party ever plan to give him a chance, to do anything useful |
20:41.31 | lgc | hmmhesays, let me check callweaver... |
20:41.51 | hmmhesays | [TK]D-Fender, he is. I think that that mccain will be the best candidate, sad but true. He is a fiscal conservative which is what we need right now |
20:42.00 | [TK]D-Fender | J4k3: Latter sure, former depends... there is a small chance. But if Barack makes the Dem side I could live with that. |
20:42.19 | hmmhesays | we're just as fscked if obama gets in |
20:42.19 | J4k3 | [TK]D-Fender: hopefully he will.. I don't think I can vote for hillary clinton. |
20:42.30 | hmmhesays | We don't need more government we dont' need more social programs |
20:42.41 | J4k3 | we don't need iraq |
20:42.43 | hmmhesays | we need less social programs, free market health care |
20:42.50 | [TK]D-Fender | hmmhesays: How can you say fiscal when the dumbass is ready for war with the world and to maintain forces in Iraq for the nexyt CENTURY? THERE'S your econimic sink-hole |
20:42.51 | J4k3 | social programs don't cost the US money, it rolls right back in |
20:42.54 | jameswf | if you hate babies and wanna kill all small animals vote hillary |
20:43.00 | J4k3 | assuming your social programs don't involve buying a lot of foreign items |
20:43.07 | [TK]D-Fender | J4k3: No, she is as bad as the worst of them. |
20:43.20 | hmmhesays | the democrats have a great vision, but they are fiscal morons |
20:43.32 | hmmhesays | in general |
20:43.39 | [TK]D-Fender | hmmhesays: Except for Kucinich who dropped out... |
20:43.42 | J4k3 | hmmhesays: yes and no... step 1 to fisical stability - don't start an empire you can't maintain. |
20:43.47 | jameswf | I heard hilliry likes to kick small kids with down syndrome |
20:43.57 | hmmhesays | I heard hillary is the man in bed with bill |
20:43.59 | [TK]D-Fender | hmmhesays: Barack isn't too bad. by pulling out of Iraq he'd have a lot mroe to work with |
20:44.07 | hmmhesays | we can't just pull out of iraq |
20:44.15 | hmmhesays | it needs to be calculated with set goals |
20:44.15 | J4k3 | sure we can |
20:44.17 | jameswf | real men dont pull out |
20:44.23 | J4k3 | we ran up in there without any plans or justification |
20:44.26 | J4k3 | we can leave just the same |
20:44.41 | J4k3 | real men get done and go to bed |
20:44.41 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
20:44.59 | jameswf | we still hvent puled out of any place we have fought a war except vietnam and look what happened there |
20:45.18 | [TK]D-Fender | jameswf: Never really pulled out.... still have permanent bases... |
20:45.19 | drmessano-LT | If McCain wins, we're going back to Vietnam to finish them off! |
20:45.22 | J4k3 | jameswf: vietnam? happened? we just fucked over those that sided with us... and that should be a good lesson to the rest of the world. |
20:45.29 | hmmhesays | drmessano-LT, bs |
20:45.39 | husimon | in zt monitor what should the levels be, about 50%? |
20:45.48 | jameswf | lol you think bush jr had a vendetta |
20:45.54 | J4k3 | we're as bad as the french, except we have more corporate evilness built in. |
20:45.57 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
20:45.57 | drmessano-LT | Yep.. and we didnt go into Iraq for Oil and "He tried to kill my dad" |
20:46.08 | drmessano-LT | whatever |
20:46.23 | J4k3 | "he tried to kill my dad"... and we thanked him for it and was saddened that he didn't get the job done. |
20:46.25 | drmessano-LT | McCain is gonna send us back to Vietnam, like a friggin Chuck Norris movie |
20:46.32 | J4k3 | too bad nobody kicked barbara in the stomach a few dozen times :P |
20:46.52 | drmessano-LT | MISSING IN ACTION 6: HERE COMES THE MARINES |
20:46.56 | jameswf | you know the iraq war is about oil but if you own a car or anything delivered anywhere by a vehicle stfu cause it doesnt go without oil |
20:46.57 | husimon | jameswf, I think hating babies and small animals is the least of our worries right now :P |
20:47.19 | J4k3 | jameswf: it ran on much cheaper oil before the war. |
20:47.25 | drmessano-LT | True |
20:47.26 | J4k3 | jameswf: I didn't need that war to keep BUYING oil |
20:47.52 | J4k3 | and that oil was being pumped much cheaper by saddam via france, than it is the US contractors. |
20:47.58 | husimon | so the war wsas to what, protect oil prices? |
20:48.03 | drmessano-LT | We went over there to librate them of their existance |
20:48.03 | J4k3 | hence why oil went from $23 to $105 a barrel. |
20:48.06 | husimon | or to protect us oil p |
20:48.11 | J4k3 | husimon: keep oil prices high |
20:48.19 | J4k3 | husimon: same as the war on drugs... the war on drugs is to keep the profits high. |
20:48.19 | jameswf | well if we got rid of all this deplomicy crap and leveled iraq and made it a giant golf course for the US oil would be like a buck a barrel |
20:48.27 | husimon | J4k3, ah ok i was going to say no fucking way the war was to make it lower |
20:48.31 | J4k3 | husimon: if you could buy cocaine in stores, it'd cost about as much as talcum powder. |
20:49.01 | husimon | J4k3, i think the manufacturing process might be little more then talcum powder :P |
20:49.09 | J4k3 | and you wouldn't have crackheads kicking in your grandmother's door to rob her for the last few cents off her pension check. |
20:49.20 | husimon | but I get your point |
20:49.23 | J4k3 | husimon: not really, and its that cheap columbian labor |
20:49.24 | J4k3 | ;) |
20:49.32 | husimon | you still need gas |
20:49.36 | jameswf | I would like to see all the trupes pulled out of iraq and dropped in ti vietnam but the truth is no matter the president we will probably stay.... even billary wont commit ti a timeline |
20:49.42 | husimon | for cocaine, so maybe this war was a war on drugs |
20:49.45 | husimon | two for one |
20:49.49 | J4k3 | haha |
20:49.56 | J4k3 | gas for cocaine? just ether. |
20:50.06 | J4k3 | well, and you gotta cook it down |
20:50.08 | jameswf | damn i cant type... damn korean kyboard |
20:50.09 | *** join/#asterisk pkunkra (n=chris@cpe-72-229-148-29.nyc.res.rr.com) |
20:50.26 | drmessano-LT | John McCain "Elect me, and I will win the Vietnam War... for the POWs, for Chuck Norris, and for Rambo!" |
20:50.42 | Qwell | drmessano: Chuck is with Huckabee |
20:50.47 | Qwell | (seriously) |
20:50.53 | drmessano-LT | Oh thats right |
20:50.58 | drmessano-LT | He's devout and all |
20:51.05 | jameswf | god i saw the new rambo and holy crap there is a movie that should have never happned |
20:51.07 | drmessano-LT | WWJVF |
20:51.07 | J4k3 | yeah, huckabee is the only guy thats a big enough asshole to keep him around. |
20:51.10 | Qwell | they were here in Huntsville last week :p |
20:51.11 | drmessano-LT | Who Would Jesus Vote For? |
20:51.19 | J4k3 | man, huckabee is a piece of shit to speak to in real life. total prick. |
20:51.41 | drmessano-LT | So McCain is gonna go back for the POW's and John Rambo |
20:51.47 | drmessano-LT | Never forget! |
20:51.50 | J4k3 | maybe we can turn mccain back into a POW |
20:51.56 | J4k3 | could we, like, give him back? |
20:52.04 | jameswf | one thing you can say about billary is she doesnt have a prick.... or can you |
20:52.11 | J4k3 | well |
20:52.20 | J4k3 | the only thing I can say about hillary is you don't have any fear of her sucking any corporate dick |
20:52.24 | J4k3 | she won't even suck her husband's |
20:52.31 | Qwell | #politics |
20:52.33 | Qwell | kthx |
20:52.36 | drmessano-LT | Well, with Billary, the white house will always be fully staffed by women |
20:52.50 | J4k3 | topless women ftw |
20:52.56 | drmessano-LT | boobies |
20:52.58 | J4k3 | well, as long as hillary keeps hers on |
20:53.04 | jameswf | ~boobies |
20:53.04 | jbot | well, boobies is (.)(.) |
20:53.05 | J4k3 | that'll turns my smile upside-down. |
20:53.19 | *** part/#asterisk rcahilig (n=test@203.115.187.98) |
20:53.54 | husimon | how can I tell what zaptel channel my call is being made on? |
20:54.22 | drmessano-LT | Not that I enjoy installing XP all that much.. but I can't wait for SP3.. Damn 3 hours of updating from SP2 level |
20:54.22 | *** part/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
20:54.25 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
20:54.25 | *** mode/#asterisk [+o Qwell] by ChanServ |
20:54.25 | husimon | zap show channels? |
20:54.37 | husimon | drmessano, slipstreaming is your friend |
20:54.44 | drmessano-LT | I know |
20:54.54 | *** join/#asterisk Wangster (n=Wangster@host-253.epicnet.ca) |
20:54.55 | drmessano-LT | But my slipstream disk is very custm |
20:54.57 | drmessano-LT | custom |
20:54.58 | husimon | as well as never using windows again |
20:55.05 | husimon | that's also your friend |
20:55.11 | drmessano-LT | and I needed a more generic install for this machine |
20:55.13 | Wangster | Why the heck does my asterisk 1.4 keep over-writing my voicemail.conf file??! How do i stop it? |
20:55.24 | husimon | http://goodbye-microsoft.com/ |
20:55.34 | husimon | that's my favorites site to tell peole to goto :) |
20:55.36 | drmessano-LT | Well, we can argue that crap all day long.. you wanna port these apps to Linux? lol |
20:55.44 | husimon | drmessano, what apps? |
20:56.04 | drmessano-LT | Custom stuff we use at work... So drop the rhetoric.. I get it :) |
20:56.28 | husimon | drmessano, yeah I'm not completely anti windows, I know sometimes you just have to use it. |
20:56.29 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
20:56.33 | husimon | drmessano, I like to avoid it is all |
20:56.38 | drmessano-LT | Ditto |
20:56.44 | drmessano-LT | But sometimes you just can't |
20:56.55 | husimon | drmessano, when I want to crank out some games, windows xp is on my dual boot. |
20:56.59 | husimon | ;) |
20:57.06 | *** part/#asterisk gdiebel (n=gregd@adsl-69-217-146-185.dsl.mdsnwi.ameritech.net) |
20:57.18 | husimon | in os x I have parallels with windows, that's just about the perfect marriage of two os' |
20:57.22 | drmessano-LT | I was thinking this morning how much I would prefer this running on Linux |
20:57.30 | drmessano-LT | But :( |
20:57.33 | husimon | drmessano, you could always try wine |
20:57.41 | *** part/#asterisk RoyKa (n=roy@ip-77-15-149-91.dialup.ice.no) |
20:57.49 | husimon | it might be worth a few minutes of trying it |
20:57.58 | drmessano-LT | Not worth the hassle for a single tasked machine |
20:58.10 | drmessano-LT | May as well install XP and forget it |
20:58.38 | jameswf | I got Photoshop running in wine... almost cried.. |
20:59.00 | husimon | jameswf, hehe |
20:59.18 | *** join/#asterisk sx|lappy (n=sxpert@home.riquer.fr) |
20:59.19 | drmessano-LT | Besides, as with a lot of specialized apps, this one would end up with some croak in wine anyway |
20:59.24 | drmessano-LT | Thats always the luck |
20:59.31 | husimon | yeah |
20:59.36 | husimon | you are not bad off with windows xp |
20:59.41 | husimon | windows vista can suck my ballzzzz |
20:59.42 | drmessano-LT | or my 4 port serial card would burp |
20:59.45 | drmessano-LT | or some crap |
20:59.45 | jameswf | drmessano-LT: vmware.......... |
21:00.01 | husimon | jameswf, yeah, but you still gotta run windows :) |
21:00.09 | husimon | doesn't help you not install all the patches and sps |
21:00.12 | jameswf | i have like 8 installs from the comfort of my desktop |
21:00.13 | drmessano-LT | Yeah, still not worth it.. this machine is getting one app and then being shoved in a closet |
21:00.27 | husimon | more complexity to break |
21:01.01 | jameswf | the last windows box I touched ran windows 3.1 |
21:01.11 | jameswf | of course that was this morning |
21:01.13 | pkunkra | does anyone use teliax here? |
21:01.13 | husimon | jameswf, hehe that was my first windows box |
21:01.16 | drmessano-LT | I could always throw AsteriskWin on there lol |
21:01.22 | husimon | course before that the box had dos |
21:01.28 | husimon | asteriskwin? |
21:01.31 | pkunkra | i use them right now but i'm getting dropped calls. |
21:01.33 | drmessano-LT | Don't ask |
21:01.47 | jameswf | I sent an email to asteriskwin32 still havent heard back |
21:01.50 | pkunkra | trying to figure out if the issue is on their end or mine. |
21:01.59 | drmessano-LT | jameswf: three months ago? lol |
21:02.01 | husimon | how can I use asterisk to listen in on calls in the network? |
21:02.09 | husimon | like dial a special number and bridge into a call muted. |
21:02.15 | drmessano-LT | Ask AT&T |
21:02.22 | jameswf | ~chanspy |
21:02.22 | jbot | chanspy is probably an application that adds the ability to spy on any bridged call, this includes VoIP only calls where ZapScan/ZapBarge couldn't this can. As of october 19 2004, ChanSpy is not included in the standard Asterisk distribution or the development CVS tree. |
21:02.37 | *** join/#asterisk cpjosh (n=josh@cp120.cardplayer.com) |
21:02.38 | J4k3 | ~at&t |
21:02.38 | jbot | it has been said that at&t is the devil. |
21:02.40 | J4k3 | ~att |
21:02.45 | husimon | so much power... :) |
21:02.52 | jameswf | ~devil |
21:02.53 | jbot | They say if you play a Windows CD backwards you hear satanic messages. What's even more scary is if you play it forward, it installs Windows. |
21:03.06 | husimon | it's neat to watch a convo in ztmonitor |
21:03.07 | pkunkra | hahah |
21:03.09 | husimon | watching it go back and forth |
21:03.19 | drmessano-LT | Jesus.. I am NEVER going to use the browser on this box.. WHY make me upgrade to IE7 |
21:03.28 | drmessano-LT | Damn it all to hell! |
21:03.31 | husimon | drmessano, you can remove it |
21:03.32 | denon | nothing's making you upgrade |
21:03.36 | denon | just click No |
21:03.39 | denon | and it won't ask again |
21:03.39 | drmessano-LT | I know |
21:03.40 | husimon | drmessano, i remove ie7 on all my xp boxes |
21:04.20 | drmessano-LT | Actually, the date of forced upgrade is coming |
21:04.25 | drmessano-LT | Mar 1 maybe |
21:04.55 | drmessano-LT | No |
21:04.57 | drmessano-LT | Feb 12th |
21:04.58 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
21:04.58 | *** mode/#asterisk [+o bkruse] by ChanServ |
21:05.22 | *** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
21:05.36 | pkunkra | i believe windows update is making you use ie7 |
21:05.41 | *** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
21:05.41 | J4k3 | whats scary about microsoft is every time you upgrade you end up with MORE bugs |
21:05.45 | J4k3 | correct one, add five. |
21:05.58 | cpjosh | Can anyone give me some advice on this? I have a working asterisk install and can dial in and out, however when dialing out on a second line i get a 'line is busy' and the following error in the log: app_dial.c: Unable to create channel of type 'ZAP' (cause 0 - Unknown) |
21:06.00 | [TK]D-Fender | J4k3: Just like COBOL |
21:06.11 | cpjosh | I have a zaptel trunk setup and it has 24 channels... |
21:06.25 | pkunkra | ie6 has a whole slew of its own bugs. |
21:06.27 | drmessano-LT | J4k3, thats interesting math... how do you explain the 117 to 1 ratio with Vista? lol |
21:07.10 | J4k3 | drmessano-LT: I explain vista like this... I sit there for a few minutes, quietly, looking good as I always do... then I crap my pants violently and fall over and start convulsing |
21:07.20 | husimon | cpjosh, pastebin your extensions.conf |
21:07.20 | drmessano-LT | HA |
21:07.23 | J4k3 | then I make sure to take as much of everyone's time as possible in the process of cleaning myself back up |
21:07.29 | husimon | cpjosh, are you using a group for your trunk, or a single channel |
21:07.52 | hmmhesays | hmm do dynamic features work in meetme? |
21:08.03 | husimon | cpjosh, for instance on my outgoing patterns I use Zap/g1 |
21:08.05 | drmessano-LT | "I'm really not into Pokemon" |
21:09.21 | cpjosh | husimon: my outbound pattern is ZAP/1 |
21:09.35 | drmessano-LT | Wow |
21:09.42 | husimon | cpjosh, I don't know much about this but I'd say try ZAP/g1 |
21:09.48 | drmessano-LT | So the Dell Trixbox Pro system won't be named Trixbox afterall |
21:10.00 | J4k3 | Dell DudeBox |
21:10.08 | drmessano-LT | "Fonality VoIP Phone System" |
21:10.18 | J4k3 | where all your calls get sent through to the Indian Voice Translation Service |
21:10.19 | cpjosh | husimon: oh you think im referencing a channel instead of a channel group? |
21:10.23 | husimon | cpjosh, yes |
21:10.27 | jameswf | ~fonality |
21:10.30 | cpjosh | husimon: Ah! |
21:10.32 | J4k3 | so your organization can go from professional to twinkie-salesman overnight. |
21:10.33 | drmessano-LT | That's damn interesting |
21:10.35 | cpjosh | husimon: thank you very much. |
21:10.49 | drmessano-LT | I was wondering if Dell was going to REALLY market something with a name like Trixbox |
21:10.53 | denon | I do not know dell dudebox, please kindly hold while I look up your question in our kindly knowedgebase articles sir |
21:10.54 | drmessano-LT | I guess I know the answer |
21:11.10 | *** part/#asterisk Synoptic (i=Synoptic@modemcable034.152-81-70.mc.videotron.ca) |
21:11.15 | jameswf | ~fonality is <reply> Fonality is hiring no Asterisk knowledge needed or, I just installed asterisk now what? |
21:11.17 | jbot | okay, jameswf |
21:11.17 | Qwell | drmessano: Fonality is calling it "foncore" |
21:11.34 | J4k3 | drmessano-LT: Dell has cornered the market with a whole line of PCs that anyone with a clue refers to as "shitbox" |
21:11.40 | *** join/#asterisk sx|lappy (n=sxpert@home.riquer.fr) |
21:11.49 | husimon | J4k3, true but they are cheap |
21:11.51 | jameswf | ~dell |
21:11.52 | jbot | Dude! Are you getting a Dell?, or stupid for only selling PCs with Vista, or might be cooler if they start pre-packaging OpenOffice per the OpenOffice.org request sent to the CEO |
21:11.54 | husimon | J4k3, throw a 3 year on them and who cares |
21:11.59 | J4k3 | husimon: crap hardware is never cheap |
21:12.17 | J4k3 | downtime > warranty |
21:12.18 | husimon | J4k3, I agree for certain applications they are a bad idea |
21:12.25 | denon | J4k3: funny you should say that, knowing our involvement in openwrt |
21:12.34 | drmessano-LT | Qwell: Is foncore the name of the butchered Asterisk, or the whole technology? |
21:12.37 | b11d | . |
21:12.39 | husimon | J4k3, but for peoples workstations and general purpose crap machines they are fine, especially if you keep on in reserve and image your installs |
21:12.42 | J4k3 | I mean, if you're buying an office load of crap for useless employees... buy them some old 1st gen imacs used and save yourself a lot of money |
21:13.10 | husimon | keep one in reserver* |
21:13.14 | husimon | reserve. |
21:13.24 | daven | or don't and spare yourself the headache of repairing the drives in those fuckers |
21:13.26 | J4k3 | denon: thats the part that kaloz pissed me off the other day. I've ran openwrt on a LOT of hardware, and madwifi on even more hardware than that. |
21:13.49 | husimon | I have to say dells case design has gotten a lot better. |
21:13.55 | J4k3 | denon: thats when I concluded that openwrt was never going anywhere and left. |
21:14.07 | J4k3 | husimon: considering a cardboard box was better than their old cases.... |
21:14.12 | husimon | i hate clamshell |
21:14.16 | husimon | is what I mean to say |
21:14.18 | J4k3 | I hate tool-less. |
21:14.23 | husimon | why? |
21:14.27 | drmessano-LT | "Dude, I got a PBX" |
21:14.32 | husimon | lol drmessano |
21:14.33 | J4k3 | because it generally causes two things |
21:14.35 | J4k3 | A> idiots in PCs |
21:14.41 | J4k3 | B> having to beat up the box to get in the SOB |
21:14.56 | husimon | J4k3, I guess I don't have that problem because my users don't open the boxes. |
21:15.00 | J4k3 | dell's newish microatx chassis requires a beating to get inside |
21:15.03 | trixbox-fanboy | I need help |
21:15.07 | drmessano-LT | Thats NOT far from SOME of the Trixbox users.. I wont say all, or most, but some are just.. WINDOZ4LIFE |
21:15.17 | husimon | J4k3, I have to admit once, with an ibm case I couldnt' figure out how to get it open :P |
21:15.24 | husimon | J4k3, fucking secret release lever |
21:15.25 | drmessano-LT | yes, trixbox-fanboy, wut can I helpz you wit? |
21:15.28 | J4k3 | heh drmessano-LT, you've heard about the put-asterisk-on-win32 project right? |
21:15.40 | drmessano-LT | I've seen AsteriskWin32 |
21:15.49 | husimon | trixbox-fanboy, windows and trixbox and asterisk oh my! |
21:15.50 | trixbox-fanboy | I ran yum update now zaptel is broked |
21:15.57 | J4k3 | omfg, windows can't even bring up the start menu in a reasonable period of time... how the hell is it going to perform in a network situation with devices that notice a 20ms jitter. |
21:16.12 | drmessano-LT | foncore will fix All that |
21:16.21 | drmessano-LT | foncore will save the world |
21:16.42 | husimon | foncore vs chuck norris |
21:16.53 | husimon | first round chuck will be in here crying |
21:16.55 | drmessano-LT | Dellnality PhonesHome PBXcellent Phone System will rock balls |
21:17.21 | drmessano-LT | I CAN HAZ AKERISK? |
21:17.38 | outtolunc | no its MINE |
21:17.40 | outtolunc | all mine |
21:17.42 | J4k3 | GrandDell DellTone 101 |
21:17.48 | drmessano-LT | Oh god |
21:17.49 | husimon | You remember that old t-shirt, reading your emails for fun and profit? |
21:17.53 | drmessano-LT | yes |
21:17.58 | husimon | I want an asterisk t-shirt that says "listening into your calls for fun and profit" |
21:18.22 | drmessano-LT | "You know that thing your wife does? I heard you and her talking about it." |
21:18.28 | husimon | and on the front have a man entry for chanspy |
21:18.34 | husimon | or the back |
21:18.47 | husimon | LAUGH |
21:18.51 | husimon | big white letters |
21:18.54 | drmessano-LT | lol |
21:18.56 | husimon | <front> CHANSPY |
21:19.03 | husimon | <back> Listening to your calls for fun and profit |
21:19.04 | husimon | lmao |
21:19.07 | J4k3 | ASTERISK CAT LISTENS WHILE YOU PHONESEXOR |
21:19.15 | drmessano-LT | I would NOT trust a Fonality system, and here Dell is selling them.. WTF.. |
21:19.17 | husimon | that's such a perfect shirt for asterisk geeks |
21:19.20 | drmessano-LT | Does that mean I can be rich too? |
21:19.30 | J4k3 | drmessano-LT: maybe you shouldn't trust dell? |
21:19.35 | drmessano-LT | ASTERISK CAT <----- HAHAHAHAH |
21:19.48 | husimon | i don't get the cat bit? |
21:19.57 | drmessano-LT | Google for "ceiling cat" |
21:20.09 | husimon | ah |
21:20.19 | J4k3 | http://rudd-o.com/wp-content/uploads/images/funny/Ceiling_cat_is_watching_you_masturbate.jpg |
21:20.55 | husimon | people have a retarded fascination with cats |
21:20.58 | husimon | lolcats ... |
21:21.05 | J4k3 | lolicats |
21:21.05 | drmessano-LT | LOL |
21:21.06 | [TK]D-Fender | heading home, BBIAB |
21:21.09 | J4k3 | pre-teen pussy |
21:21.33 | *** join/#asterisk sbingner (n=john@pdpc/supporter/sustaining/sbingner) |
21:21.33 | husimon | J4k3, your preference? |
21:21.41 | J4k3 | husimon: nah, my cats are ooooold |
21:21.42 | drmessano-LT | Asterisk cat is in ur foncore, stealing your SIPs |
21:21.44 | jameswf-home | irc doesnt like the nick changes |
21:21.56 | tobias | can folks here recommend a particular VOIP provider for a small asterisk setup? |
21:22.08 | husimon | "I'm in your asterisk stealing your sips |
21:22.10 | husimon | " |
21:22.20 | tobias | I've been using VoicePulse but a network issue between our server and theirs has made it practically unusable |
21:22.35 | husimon | hmm |
21:22.39 | husimon | i need to make that asterisk shirt |
21:22.44 | husimon | where can I make custom t-shirts online |
21:22.55 | Qwell | husimon: cafepress |
21:23.02 | J4k3 | tobias: I love that. my itsp changed internet connections and things weren't right for 2 months after |
21:23.10 | Qwell | I assume they're still around |
21:24.36 | drmessano-LT | they are |
21:25.02 | husimon | what the crap |
21:25.05 | husimon | i can't customize the back of the shirt |
21:25.12 | Qwell | try harder |
21:25.24 | husimon | oh I will |
21:25.37 | tobias | J4k3: yeah. i think it was working ok for awhile. traceroutes to voicepulse look great from my other servers across the country too. |
21:25.56 | jameswf | is asterisk compattible with web 2.0 |
21:26.14 | tobias | though really, voicepulse should have servers in more than one location. |
21:26.17 | Corydon76-vcch | I dunno, are you compatible with blue cars? |
21:26.46 | *** join/#asterisk servergod (n=servergo@70.97.159.120) |
21:27.05 | tobias | Corydon76-vcch: your question still makes more sense than his :p |
21:28.19 | servergod | hi all |
21:28.52 | husimon | Qwell, nope can't figure out a way to get stuff on the back |
21:28.55 | J4k3 | tobias: yeah, I suspect the problem is all these itsp's are too small potatos to do it right |
21:29.13 | maszlo | i can not figure out this caller id problem at all. i have debugged from asterisk, viewed the full log and i can not see where this callerid is coming from. it shows the correct callerid, until it hits my cell phone, then it changes |
21:29.27 | J4k3 | you've got the big companies that won't speak to you with less than 1M/minutes/month or more, then you've got everyone else who's scratching over that 8th of a cent per minute profit they're making. |
21:29.36 | husimon | it appears you can't print on the back of black shirts |
21:29.42 | husimon | how stupid is that |
21:29.54 | J4k3 | asterisk crosses me as more of a blue shirt kind of subject. |
21:30.17 | husimon | chanspy doesnt |
21:30.47 | maszlo | does anyone have experience with running asterisk on an verizon pri? |
21:31.02 | servergod | yes |
21:31.13 | maszlo | can you set the callerid? |
21:31.25 | *** part/#asterisk javar (n=javar@69.79.134.24) |
21:31.32 | mvanbaak | "I drive a blue Toyota Prius" |
21:31.41 | jameswf | gayyyyyyyyyyyyyyyyyyyy |
21:31.51 | servergod | as long as it matches a DID in their CNAM database |
21:32.00 | J4k3 | "I ran over a blue toyota prius in the driveway, oops" |
21:32.01 | mvanbaak | "do you know what sound that makes when you drive by ?" |
21:32.10 | mvanbaak | "Iiiiiiiiiiiiiiiiiiiiiiiiiiiiiii'm gay" |
21:32.15 | J4k3 | I keep a thick layer of mud on the bottom of my truck so I can drive over economy cars and not scratch the paint. |
21:32.16 | maszlo | do i have to enter the number without a name? |
21:32.40 | jameswf | I took my wife to go see duhnum live funny stuff |
21:33.11 | mvanbaak | SILENCE! I KIIIIIL YOU " |
21:33.29 | *** join/#asterisk ShaunWing (n=chatzill@dsl-243-73-60.telkomadsl.co.za) |
21:33.36 | servergod | Verizon uses the DID on the sip:nxxnxxxxx@foo.com to do an auth and a lookup. no custom id is accepted or used, since they do a dip on the CNAM database |
21:33.43 | servergod | op |
21:33.45 | servergod | pri |
21:33.45 | ShaunWing | Help asterisk -r does not let me reconnect |
21:33.55 | servergod | sorry, pri |
21:34.07 | mvanbaak | ShaunWing: ps ax | grep asterisk |
21:34.29 | ShaunWing | <PROTECTED> |
21:34.31 | ShaunWing | <PROTECTED> |
21:34.40 | servergod | If you want a custom name on a DID, you have to submit it to verizon |
21:35.11 | maszlo | i just want it to get correct number on outbound calls |
21:35.26 | mvanbaak | ShaunWing: what's the error |
21:35.43 | servergod | on a specific trunk? or just for specific extensions? |
21:35.56 | maszlo | have been playing with this all day with no luck, "sip:nxxnxxxxx@foo.com" can you explain this |
21:36.07 | maszlo | all line show the same outbound callerid |
21:36.10 | servergod | sorry, that is for sip not a pri |
21:36.23 | maszlo | what is a automated system, not even a calling number |
21:36.27 | *** join/#asterisk UnixDog (n=unixdog@ppp-71-128-114-104.dsl.irvnca.pacbell.net) |
21:36.30 | servergod | using freepbx or just cle? |
21:36.37 | servergod | *cli |
21:36.41 | maszlo | freepbx |
21:36.44 | ShaunWing | [root@localhost asterisk]# asterisk -r |
21:36.45 | Qwell | #freepbx |
21:36.45 | ShaunWing | Asterisk 1.4.14, Copyright (C) 1999 - 2007 Digium, Inc. and others. |
21:36.47 | ShaunWing | certain conditions. Type 'core show license' for details. |
21:36.48 | ShaunWing | ========================================================================= |
21:36.50 | ShaunWing | HANGS HERE |
21:37.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
21:37.12 | mvanbaak | ShaunWing: and asterisk is working ok ? |
21:37.16 | mvanbaak | or not working at all |
21:37.18 | ShaunWing | yes |
21:37.35 | *** join/#asterisk Docfxit (i=ExUser@netblock-208-127-208-174.dslextreme.com) |
21:37.39 | ShaunWing | I'm happy to just kill it but how and then restart it |
21:37.40 | servergod | edit the trunk, and under general outbound caller id |
21:37.47 | ShaunWing | Don't want to reboot my server |
21:37.56 | *** part/#asterisk UnixDog (n=unixdog@ppp-71-128-114-104.dsl.irvnca.pacbell.net) |
21:38.19 | servergod | ShaunWing |
21:38.31 | ShaunWing | yes |
21:38.34 | mvanbaak | ShaunWing: killall -9 asterisk |
21:39.08 | jameswf | killall -9 silly_beavers |
21:39.16 | maszlo | my current outbound callerid is blank in the trunk |
21:39.58 | servergod | set it there, and do the reload and call, see if the CID chages |
21:40.09 | ShaunWing | Tx that worked |
21:40.20 | maszlo | wont that make it for all lines on that trunk? |
21:40.54 | servergod | on a specific trunk? or just for specific extensions? That is why i asked that |
21:40.55 | Qwell | ~freepbx |
21:40.56 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
21:41.16 | drmessano-LT | Oh god |
21:41.23 | drmessano-LT | "Hacking Trixbox" |
21:41.27 | drmessano-LT | I can just see it now |
21:41.30 | mvanbaak | ~closedpbx |
21:41.33 | maszlo | i have 12 extensions on one trunk, if i set the callerid on the trunk wont they all have that as the callerid? |
21:41.39 | servergod | how can i share a trunk on *1.4 if i send a sip call to it from ccm? |
21:41.41 | drmessano-LT | "I hacked my trixbox and removed all the GUI stuff.. how cool is that?" |
21:41.42 | Qwell | maszlo: there is no such thing as a trunk |
21:42.01 | kyron | drmessano-LT, maaan...you're like so 1337 |
21:42.13 | maszlo | Qwell: what do you mean? |
21:42.17 | mvanbaak | drmessano-LT: svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk |
21:42.18 | mvanbaak | ;) |
21:42.26 | drmessano-LT | "Yeah, I stripped so much out of it, it comes up in freakin DOS" |
21:42.32 | *** join/#asterisk riksta (n=rick@rhamnett.plus.com) |
21:42.37 | Qwell | ~siptrunk |
21:42.37 | jbot | There is nothing special about a SIP trunk in the protocol like there is in the case of IAX2, for example. You set up a SIP trunk like a regular peer in sip.conf. |
21:42.47 | Qwell | maszlo: go to #freepbx, we cannot help you with that here |
21:43.22 | drmessano-LT | All bashing aside |
21:43.23 | mvanbaak | we should update jbot |
21:43.24 | kyron | drmessano-LT, I have to use a _keyboard_ and there is NOTHING to click!!!....talk about retarded! |
21:43.31 | riksta | Hi, I'm originating a channel and I want to record separate CDRs for both legs, for billing purposes. I'm using the ForkCDR command but it does not work as expected, the fork makes the first leg's bill seconds 0. What is the proper way to log both legs correctly? |
21:43.32 | kyron | pun |
21:43.44 | mvanbaak | freepbx is a virus that will render you asterisk unusable in most cases |
21:43.46 | drmessano-LT | Why would you want to install a trixbox system, and THEN use it to experiment with |
21:44.11 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
21:44.11 | *** mode/#asterisk [+o lmadsen] by ChanServ |
21:44.19 | lmadsen | ANSWEREDTIME is in seconds I presume.. ? |
21:44.19 | mvanbaak | lmadsen !!!!!!!!!!!1 |
21:44.25 | lmadsen | mvanbaak: !!!!!!1one! |
21:44.32 | mvanbaak | lol |
21:44.44 | Qwell | drmessano: speaking of book titles... I love what Kerry named his book. |
21:44.44 | mvanbaak | they still did not merge it :( |
21:44.54 | Qwell | "trixbox made easy"...as if it were ever difficult |
21:44.55 | lmadsen | those bastards! |
21:45.01 | drmessano-LT | Yeah, no crap |
21:45.08 | mvanbaak | lmadsen: I can merge it....... |
21:45.15 | lmadsen | mvanbaak: mwahahaha |
21:45.45 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
21:45.50 | mvanbaak | would be funny, my 3rd commit to -trunk being 50K of changes |
21:45.51 | drmessano-LT | Well, it takes a real tool to take something as easy as Trixbox, make it out to be hard, and then write a 500 page book on it |
21:46.00 | *** join/#asterisk CrashSys (n=kumba@t1.databalance.com) |
21:46.21 | drmessano-LT | If I wrote a book on trixbox, I could read it in one sitting on the shitter |
21:46.25 | plik | there's another book called "trixbox without tears"... which seems unlikely to me |
21:46.41 | *** join/#asterisk atis_work (n=atis_wor@81.198.164.2) |
21:46.44 | Robba | i'd have to agree with that |
21:46.57 | drmessano-LT | My CentOS + Asterisk + FreePBX guide is called "Asterisk without Tricks" lol |
21:47.04 | plik | ha |
21:47.05 | ShaunWing | By the way.. Anyone registered Asterisk to Net2Phone? |
21:47.08 | Robba | lol |
21:47.49 | drmessano-LT | I was damn proud of that name |
21:47.56 | mvanbaak | asterisk without tears => vim /etc/asterisk/* |
21:48.13 | *** join/#asterisk chavigny (n=nrp@c-67-171-147-26.hsd1.or.comcast.net) |
21:49.37 | chavigny | Anyone have and sugguestions, Im looking for a carrier that can sell me some minutes and a few dids in an areacode of my choice/plus toll free, i looked at gzlink but its not that good |
21:50.02 | plik | ~itsp-us |
21:50.18 | drmessano-LT | 168 pages.. Chapter 1: Click |
21:50.19 | plik | ~itsplist-us |
21:50.19 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com |
21:50.26 | drmessano-LT | Chapter 2: Unclick |
21:51.00 | *** join/#asterisk roxlu (n=Roxlu@84-107-142-180.dsl.quicknet.nl) |
21:51.02 | roxlu | hi there |
21:51.09 | Robba | anyone have any idea as to why i can't use (CallerID(num)=number) |
21:51.15 | roxlu | is it possible to see what 'calls' are currently being made? |
21:51.16 | mvanbaak | Chapter 3: get lost in the spaghetti code in your dialplan |
21:51.33 | Robba | it just keeps saying module not registered |
21:51.52 | Robba | but if i run module load func_callerid.so |
21:52.06 | mvanbaak | Robba: pastebin the dialplan lines |
21:52.14 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
21:52.18 | mvanbaak | ~pb |
21:52.19 | 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 |
21:52.24 | mvanbaak | ~tk |
21:52.25 | jbot | ACTION snipes $herlo with a straw and rolled up piece of paper |
21:53.06 | mvanbaak | you all have to start worry now, [TK]D-Fender is here |
21:53.16 | mvanbaak | ;) |
21:53.19 | [TK]D-Fender | :O |
21:53.44 | mvanbaak | =4 |
21:54.09 | Robba | http://rafb.net/p/aLzd9M84.html |
21:54.10 | ShaunWing | Say Any idea why there is no speech in either direction when use Quintum registered to Asterisk. Quintum on its own works fine |
21:56.03 | mvanbaak | Robba: what version of asterisk |
21:56.10 | Robba | 1.4.17 |
21:57.21 | mvanbaak | Robba: try in all caps |
21:57.30 | Robba | ok |
21:57.42 | *** join/#asterisk djweis (n=djweis@67.55.197.226) |
21:57.56 | mvanbaak | exten => s,1,Set(CALLERID(num)=0398982277) |
21:57.58 | ShaunWing | Say no one heard of Quintum? |
21:59.04 | Robba | ok i did that now its calling out from a random number from our 100 number range |
21:59.07 | [TK]D-Fender | Robba, and like I told you before "_s" is NOT a pattern. it must be just "s" |
21:59.17 | [TK]D-Fender | Robba, Go fix your contexts |
21:59.42 | [TK]D-Fender | Robba, 58 Polo |
22:00.00 | *** join/#asterisk nybbled (n=nybbled@about/apple/performa/nybble) |
22:00.02 | [TK]D-Fender | Robba, is this the same broken thing I gave you the play-by-play on already? |
22:00.08 | Robba | no |
22:00.30 | Robba | i have changed from setcallerid to callerid(num)= |
22:01.32 | [TK]D-Fender | Robba, you still seem to have a lot of the stuff I commented on already unfixed. Also you have "congestion" following a Macro call you never come back from. wastage. |
22:01.43 | *** part/#asterisk cpjosh (n=josh@cp120.cardplayer.com) |
22:03.14 | mvanbaak | waste -> wastega |
22:03.14 | Robba | so i shouldn't have those lines at all? |
22:03.25 | mvanbaak | final fantasy ftw ! |
22:05.48 | [TK]D-Fender | Robba, Clearly not |
22:07.41 | J4k3 | hmm does voicepulse filter callerid? |
22:08.02 | J4k3 | er toward |
22:08.47 | [TK]D-Fender | J4k3, Not to my awareness |
22:09.05 | twisted | oh holy crap |
22:09.14 | twisted | svnfs <3 |
22:09.41 | Robba | http://rafb.net/p/SXgGdz57.html |
22:10.42 | twisted | you know you have too much stuff open when it takes 5 minutes to get to the point the system will restart |
22:11.13 | [TK]D-Fender | Robba, [awc-phones] and [incoming] are very redundant and you may have missed 1 entry |
22:12.49 | Robba | i used [incoming] because that was the context asterisk was looking for, for the extension 7600 |
22:12.53 | [TK]D-Fender | Robba, exten => 101,1,VoiceMailMain(${CALLERID}@${CONTEXT}) <-- after everything we went through, STILL not right |
22:13.27 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
22:13.59 | *** part/#asterisk maszlo (n=reckenro@65.223.240.146) |
22:14.44 | Robba | [TK] i'm still not sure what that is supposed to be |
22:14.50 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
22:16.23 | husimon | [TK]D-Fender, do you have any idea what would cause intermittant echo? Do you think the hardware echo cancellation is trying to cancel but fails once in a while? |
22:16.54 | husimon | [TK]D-Fender, if you remember from yesterday i'm running a phonebridge2 with hardware ec. |
22:17.00 | [TK]D-Fender | Robba, "core show function CALLERID" and go reread chapter 5. |
22:17.15 | [TK]D-Fender | husimon, yes, entirely possible that it just sucks |
22:17.21 | husimon | [TK]D-Fender, i'm thinking that too |
22:17.34 | [TK]D-Fender | husimon, Don't say you weren't warned |
22:17.37 | husimon | [TK]D-Fender, but then again, the number I had a problem with has a really long run between the phone and the pbx |
22:17.44 | husimon | [TK]D-Fender, we have historically had problems with that office |
22:18.15 | *** join/#asterisk twisted (n=root@pdpc/supporter/active/twisted) |
22:18.15 | *** mode/#asterisk [+o twisted] by ChanServ |
22:18.16 | twisted | oops. |
22:18.21 | twisted | helps not to kill the term window :P |
22:18.32 | husimon | [TK]D-Fender, the asterisk box and this hardware is a huge improvement over the old setup. I can actually use the speaker phones now, before calling that office would completely echo out and breakup while using a speaker phone. But other lines were fine. |
22:23.37 | *** part/#asterisk roxlu (n=Roxlu@84-107-142-180.dsl.quicknet.nl) |
22:23.39 | husimon | hehe i made the t-shirt http://kolea.ifa.hawaii.edu/~nhuisman/front.jpg http://kolea.ifa.hawaii.edu/~nhuisman/back.jpg |
22:25.11 | twisted | might i suggest getting digium's permission to use the asterisk logo? |
22:25.19 | husimon | twisted, yeah i'm not actually making it... |
22:25.20 | husimon | heh |
22:25.24 | twisted | aww |
22:25.26 | husimon | probably just do Asterisk * |
22:25.31 | husimon | or a big * |
22:26.15 | husimon | twisted, it's pretty far on the geek scale of things :P |
22:26.33 | [TK]D-Fender | or a bull's eye with a hole punched out & fake blood-stain around it |
22:26.50 | husimon | hehe |
22:27.12 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
22:27.26 | plik | tshirts must be a hot topic just now... this just got posted in another channel http://www.themishmash.com/2008/02/10-actual-t-shi.html |
22:27.26 | [TK]D-Fender | slightly off-center so as to look plausible |
22:27.31 | x86 | I've got an old legacy Toshiba phone system that I want to connect with Asterisk over a T1 interface |
22:27.35 | lmadsen | grrr... I'm getting errors when I place a call that my ODBC connection is down, but 1) it isn't, and 2) I'm setup to write CDR records to the DB.... so I've no idea why this is happening... *sight* |
22:27.38 | lmadsen | -t |
22:27.41 | x86 | the Toshiba is setup already using the T1 interface to the PSTN |
22:27.51 | x86 | it's using E&M Wink to the PSTN |
22:27.54 | husimon | plig tshirthell.com |
22:27.57 | husimon | plik |
22:28.11 | Juggie | lmadsen, cdr_odbc or cdr_adaptive_odbc? |
22:28.17 | lmadsen | Juggie: shouldn't be... |
22:28.20 | x86 | I was wondering if I could just set E&M Wink on the interface on the Asterisk side, and it would work correctly? |
22:28.20 | plik | been there, done that... so have those guys |
22:28.26 | x86 | do I need a T1 crossover cable or something? |
22:28.31 | lmadsen | Juggie: maybe just having the module loaded does it though |
22:28.41 | lmadsen | let me try that |
22:28.57 | Juggie | lmadsen, do you want to write cdr's w/ odbc? |
22:29.03 | lmadsen | Juggie: no, I don't |
22:29.13 | lmadsen | not yet anyways... but I didn't configure it to write them |
22:29.18 | Juggie | then noload both those modules. |
22:29.30 | Juggie | some old config you have setup is making one of them load |
22:30.38 | lmadsen | Juggie: same BS |
22:30.45 | lmadsen | Juggie: this is a new config |
22:30.48 | husimon | wow I can't find any asterisk t-shirts |
22:31.03 | *** join/#asterisk angryuser (i=nononon@df01t2-212-194-235-109.d4.club-internet.fr) |
22:31.08 | lmadsen | it should not be trying to connect to the DB.... |
22:31.14 | lmadsen | there's no reason for it to |
22:31.19 | husimon | beside of course from digium |
22:31.27 | Juggie | lmadsen, 'module show like cdr' |
22:31.28 | husimon | with their butt ugly orange shirt |
22:31.29 | lmadsen | oh well, will have to fix this later... conference call with another client |
22:31.42 | Juggie | see which of them is loaded (via use count) |
22:31.50 | drmessano-LT | I have an Asterisk shirt ;) |
22:31.55 | lmadsen | cdr_manager.so Asterisk Manager Interface CDR Backend 0 |
22:31.55 | lmadsen | func_cdr.so CDR dialplan function 0 |
22:31.55 | lmadsen | cdr_csv.so Comma Separated Values CDR Backend 0 |
22:31.55 | lmadsen | cdr_custom.so Customizable Comma Separated Values CDR 0 |
22:31.59 | husimon | drmessano, did you check out the one I made? |
22:32.04 | drmessano-LT | No |
22:32.08 | husimon | http://kolea.ifa.hawaii.edu/~nhuisman/front.jpg http://kolea.ifa.hawaii.edu/~nhuisman/back.jpg |
22:32.28 | drmessano-LT | HAHAHHAHAHHAH |
22:32.38 | husimon | :) |
22:32.47 | Juggie | lmadsen, paste bin the errors? |
22:32.57 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
22:33.03 | husimon | i'm super tempted to cafepress that |
22:33.10 | drmessano-LT | My day is done.. be back on from home later |
22:33.15 | husimon | with a non-copyright digum logo. |
22:33.18 | lmadsen | basically does this... but there is no reason to be connecting to the database for any reason at this point.... |
22:33.19 | lmadsen | [Feb 7 17:30:23] WARNING[9023]: res_odbc.c:103 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42000: [FreeTDS][SQL Server]Error converting data type varchar to bigint. (66) |
22:33.19 | lmadsen | [Feb 7 17:30:23] WARNING[9023]: res_odbc.c:111 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a reconnect... |
22:33.27 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-216-147-93.nsw.bigpond.net.au) |
22:33.40 | lmadsen | cdr_odbc.conf is commented out |
22:33.55 | lmadsen | 1.4, so no cdr_adaptive_odbc to evenbe loaded |
22:34.31 | Juggie | lmadsen, cdr_odbc doesnt use res_odbc anyways i dont think |
22:34.36 | Juggie | unless its been upgraded recentally. |
22:35.16 | *** join/#asterisk codejunky (n=jan@codejunky.org) |
22:35.31 | Juggie | weird.... i would have to see the box to know more but odd. |
22:35.43 | lmadsen | ya... very odd... |
22:35.46 | lmadsen | probably a bug... |
22:35.55 | lmadsen | because there really is no reason to be trying to write a CDR to the DB |
22:35.56 | husimon | what happens with hardware and software echo cancellation are used together? FUBAR? |
22:35.58 | Juggie | possible |
22:36.05 | Juggie | because when a module refuses to load |
22:36.08 | Juggie | it just refuses |
22:36.11 | Juggie | but it doesnt really not load |
22:36.21 | lmadsen | the funniest thing is I'm trying to make something NOT work :) |
22:36.37 | Juggie | did you do a noload on cdr_adaptive_odbc and restart? |
22:36.45 | husimon | in modules.conf |
22:36.45 | lmadsen | cdr_adaptive_odbc is not in 1.4 |
22:37.02 | lmadsen | but I added it to the modules.conf in a noload anyways |
22:37.07 | lmadsen | and I did 'restart now' |
22:37.10 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
22:38.10 | Juggie | lmadsen, that is odd, because i dont think cdr_odbc uses res_odbc |
22:38.24 | lmadsen | ya.... then this is even more confusing |
22:38.45 | Juggie | doing an update of my 1.4 code now to look |
22:38.51 | lmadsen | Dial(SIP/${EXTEN}@${PROVIDER}) should nto cause a DB connection at all |
22:39.07 | lmadsen | does it after the Dial(), and after the Hangup() |
22:39.30 | Juggie | ya, in 1.4 cdr_odbc definitally does not use res_odbc so far as i can tell |
22:39.37 | Juggie | it would have a call to SQLPrepareAndExecute |
22:39.40 | Juggie | but does not. |
22:40.14 | lirakis | later all |
22:40.21 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
22:40.34 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
22:40.53 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
22:41.03 | CrazyTux | Hey guys, how do I allow a bigger inbox |
22:41.09 | CrazyTux | For voicemail? |
22:41.26 | Juggie | lmadsen, ya, odd, no call. has to be something else it cant be cdr_odbc |
22:41.34 | lmadsen | ya... |
22:41.50 | Juggie | good luck, i'm gone :) |
22:41.52 | lmadsen | lates |
22:42.09 | Juggie | odbc voicemail? |
22:42.21 | CrazyTux | Juggie, yea mysql |
22:42.32 | Juggie | CrazyTux, was talking to lmadsen. |
22:42.37 | CrazyTux | Juggie, ah |
22:42.47 | Juggie | lmadsen, try disabeling the freetds connection, which ever module is using it should complani. |
22:42.49 | lmadsen | Juggie: ya... but no call to Voicemail() |
22:42.52 | Juggie | *complain. |
22:42.58 | lmadsen | I'll figure something out |
22:43.04 | Juggie | lmadsen, doesnt matter. |
22:43.08 | lmadsen | lates |
22:43.16 | Juggie | it would still get called every minute to check mwi, |
22:43.18 | grandpapadot | Hello all. On Aastra 480i CT phones, if I have 3 lines provisioned, will they all use g729a (if so configured in the sip peer) or will only the first line use g729a and the rest fallback to ulaw (like some other phones and/or ata's?) |
22:44.04 | lmadsen | grandpapadot: asterisk would have no concept of the separate lines if they are all the same peer configuration |
22:44.30 | CrazyTux | Does anyone here know if I can increase the size of a specific mailbox in asterisk? |
22:44.34 | grandpapadot | @lmadsen: 3 differnet peers. My question was really on the phones codec implementation/support, will it do 3 simultaneous g729a calls. |
22:45.17 | grandpapadot | CrazyTux: I had an email here someone that was saying how to increase the size.. oh, wait, that was for my penis, nevermind. |
22:46.32 | *** join/#asterisk AndyGraybeal (n=andy@node191.34.251.72.1dial.com) |
22:46.32 | *** join/#asterisk servergod (n=servergo@70.97.159.120) |
22:47.06 | CrazyTux | grandpapadot, haha. |
22:47.28 | lmadsen | grandpapadot: not sure... depends on the phone I guess |
22:49.09 | servergod | have 1 *1.4 and a cisco ccm. have sip trunk from asterisk to voip provider. CCM uses pri. want to send LD from ccm to asterisk. have trunk from ccm to * when call ld i get * saying that extension dont exist. How can i route it out? |
22:50.03 | husimon | what the hell, if I make a constant sound in my phone the other persons voice cuts out |
22:50.25 | grandpapadot | husimon: The nerve! |
22:50.37 | husimon | no say i call an IVR somewhere |
22:50.41 | husimon | then hum a note |
22:50.48 | husimon | their voice all but cuts out and i get bits and pieces |
22:50.57 | husimon | wonder what is causing that |
22:51.11 | grandpapadot | husimon: Yea, I hear ya. More information might be helpful to get a good answer from someone. |
22:51.25 | husimon | not sure what information to give |
22:51.35 | grandpapadot | husimon: What phone? How is asterisk configured? Analog, SIP, PRI, etc? |
22:52.04 | husimon | cisco 7940, normal asterisk configuration, no echo cancelation or tx/rx adjustments, sip, over a pri. |
22:52.21 | husimon | grandpapadot, are you on an asterisk system right now? |
22:52.28 | husimon | i'm curious, call 1800polycom |
22:52.29 | grandpapadot | Yes, 1.2.26 |
22:52.30 | husimon | and try humming |
22:52.37 | husimon | like sing a note and see if you can hear the prompter |
22:52.47 | ManxPower | Turn off VAD/CNG on the phone |
22:52.58 | husimon | ManxPower, what are those? |
22:53.01 | grandpapadot | lol |
22:53.08 | ManxPower | If you can only hear the far end when you make noise -- then it's prolly a VAd/CNG issue |
22:53.20 | husimon | ManxPower, it's not that I can only hear them when I make noise |
22:53.24 | grandpapadot | ManxPower: Call 1-800-Polycom, guy's trolling |
22:53.26 | husimon | it's that when I make a constant noise they cut out |
22:53.36 | husimon | pfft screw that it's just the first number I thought of |
22:53.49 | ManxPower | husimon: those are options you should disable on the phone. I don't know how or where those options are, as I understand it Cisco phones do NOT default to VAD/CNG. |
22:53.54 | grandpapadot | It's a sex line... |
22:54.08 | grandpapadot | funny, though, that 1-800-polycom is a sex line |
22:54.11 | ManxPower | Oh, so when YOU send something, the far end cuts out? |
22:54.23 | husimon | ManxPower, yeah not normal conversation |
22:54.28 | husimon | but if I hum a constant note |
22:54.50 | husimon | their side cuts in and out, now I haven't tried this with a real person yet just noticed it while trying to test for echo |
22:54.50 | grandpapadot | husimon: That's just noise cancellation on the phone probably. |
22:54.55 | husimon | grandpapadot, ah |
22:57.20 | beighto | I am having a serious problem with an Asterisk system I installed. It seems that about 50% of the non-local calls (800 or long distance) don't go through. There are 3 pots lines hooked up and when dialed out upon say "you must dial a 1 before completing this call" from the local phone company. This happens at random and not on any 1 particular line out of the 3. I have it dialing a 1 in... |
22:57.21 | beighto | ...the dialplan and just to make sure I even added a + to the front with no change. I switched from inband to RFC2833 with no change. I cranked up the tx gain to 10 with no change. Any ideas? |
22:58.43 | husimon | beighto, sounds like your dialplan is wrong |
22:58.54 | husimon | beighto, or one of you 3 lines is different then the other two |
22:59.17 | beighto | husimon: I have checked it over and over. The console shows the 1 being dialed and all 3 pots lines do the same thing at random. |
22:59.29 | husimon | beighto, random for the same number |
22:59.38 | beighto | husimon: yes |
22:59.53 | husimon | beighto, you should probably pastebin your extensions.conf |
22:59.56 | husimon | so we can look at it |
23:00.05 | beighto | okay, just a minute |
23:00.44 | husimon | also a sanity check would be to plug a phone into that analog line and make the same calls and see what happens |
23:02.59 | J4k3 | report: low |
23:03.07 | husimon | hehe |
23:03.25 | J4k3 | what I'm getting tired of is my damned cellphone |
23:03.46 | J4k3 | call my *, dtmf works fine for the ivr... get to a human and the echo canceler goes 180 degrees out of whack |
23:03.50 | J4k3 | they can BARELY hear me |
23:04.05 | J4k3 | but if I beat on the dtmf a bit, it'll work again |
23:04.06 | husimon | i'm probably just going to leave the users to deal with the small bit of echo we have |
23:04.10 | husimon | it's only to one office |
23:04.19 | husimon | where their stupid loop is wayyyyy too long |
23:04.28 | J4k3 | but I suspect its my cellphone... the other cellphone here (same provider/technology) doesn't do it |
23:04.28 | husimon | local loop i mean |
23:05.00 | J4k3 | husimon: you could always get the telco to install some PAIRGAIN hardware |
23:05.02 | J4k3 | *screams* |
23:05.04 | beighto | husimon: http://pastebin.com/d489db2ee |
23:05.28 | ManxPower | husimon: HPEC is your friend if you do not have many channels and a fast system |
23:05.50 | *** join/#asterisk PepOSX (n=angeldav@190.72.132.46) |
23:06.10 | husimon | ManxPower, hpec is software or is that the ec built into digiums card? |
23:06.43 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
23:06.47 | husimon | laugh they sell it for $10, why not just make it free |
23:07.01 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
23:07.01 | *** mode/#asterisk [+o anthm] by ChanServ |
23:07.05 | grandpapadot | Using static realtime in 1.2.2x, what will initiate a musiconhold class reload that's stored in the db? A reload doesn't seem to do it. |
23:07.13 | grandpapadot | Restarting asterisk does, however. |
23:08.26 | grandpapadot | Also, is there a way to 'list' loaded musiconhold classes from the cli? |
23:08.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
23:08.35 | husimon | beighto, so with that config you dial a given number, it works. Then dial it again and it doesn't work? |
23:08.46 | beighto | husimon: correct |
23:09.02 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
23:09.34 | grandpapadot | ugh, nevermind, lol |
23:09.34 | pkunkra | alright, i'm trying to figure something out. my callers call my pbx located in my datacenter as normal, the go into the queue and it rings my extension, as soon as i pick up, it disconnects. |
23:09.37 | husimon | what happens if you remove one of your patterns and only provide the _1N..... |
23:10.04 | husimon | pkunkra, i'd go into the cli and set verbose to 9 and watch |
23:10.05 | pkunkra | there are no errors in the logs anywhere. |
23:10.12 | pkunkra | ok |
23:11.07 | husimon | does anyone know how I can get a telco test number to adjust my tx and rx gain? |
23:11.09 | beighto | husimon: I would image that wouldn't make any difference. I only put the _NXX pattern there just in case they are too lazy to dial the 1. |
23:11.19 | pkunkra | problem is, it is intermittent. |
23:11.21 | husimon | beighto, shrug it's worth a quick try. |
23:11.33 | husimon | pkunkra, keep trying until it happens. |
23:11.49 | husimon | pkunkra, use your cell |
23:12.30 | pkunkra | alright. i'll give it a try |
23:12.32 | husimon | beighto, Try plugging in a real analog phone to the first line and see if it's fine. |
23:12.47 | beighto | husimon: Yes, analog phones work fine |
23:13.02 | beighto | husimon: I even had a different phone company run 4 new lines |
23:13.16 | beighto | 3 new lines 1 was for faxing |
23:13.32 | husimon | you've tested all right? |
23:13.39 | husimon | all three |
23:13.41 | husimon | i mean |
23:13.49 | beighto | yes, I plugged a fax machine into all the lines and tested them |
23:14.08 | beighto | that was my version of an analog phone at the time |
23:14.20 | husimon | I mean plug in the analog phone and dial a number you know has problems and make sure each line works with it. |
23:15.02 | beighto | yes |
23:15.32 | beighto | did that the first day |
23:15.37 | husimon | I might suggest putting some NOOP statements in there to echo ${EXTEN} |
23:15.46 | husimon | to see what your * is actually getting |
23:15.54 | beighto | husimon: I am using presence too, but I wouldn't imagine that would effect it either |
23:16.25 | beighto | the SIP debug shows the 1 being dialed as well... |
23:17.04 | beighto | I haven't done a NOOP, but the dial statement clearly shows the whole number |
23:17.13 | husimon | true |
23:17.23 | beighto | It is driving me crazy! |
23:17.47 | husimon | beighto, i'm fairly new to * so that's about all I got. |
23:17.56 | beighto | bummer |
23:18.11 | husimon | beighto, you should ask [TK]D-Fender sometime, make sure to provide lots of logs. He is pretty much the local expert. |
23:18.28 | beighto | I am wondering if maybe the polarity is reversed on the lines, but I think that shows up on the console |
23:19.28 | beighto | [TK]D-Fender: Are you there? Care to chime in? |
23:19.42 | *** join/#asterisk pkunkra (n=chris@cpe-72-229-148-29.nyc.res.rr.com) |
23:19.51 | pkunkra | argh. |
23:19.55 | phix | [TK]D-Fender: hey, regarding echos and feedback from ATA, I still experience it, the changes made to gain settings have only seem to come into effect when going through the landline, not SIP. |
23:20.14 | beighto | husimon: Thanks for trying |
23:20.15 | husimon | phix you might be able to adjust the tx and rx for the phones |
23:20.18 | husimon | np |
23:20.50 | ManxPower | phix: that would be because you can't have echo on a all VoIP call (at least traditional echo, you can still get feedback between the mic/speaker that would be echoy" |
23:21.08 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
23:21.27 | ManxPower | And in fact, you can't have loss on an all VoIP call, so there is no reason to change the audio gain |
23:22.39 | husimon | ManxPower, so where exactly does HPEC sit? inside asterisk? or outside or? |
23:23.06 | ManxPower | husimon: It's software. |
23:23.09 | Qwell | husimon: in zaptel |
23:23.13 | husimon | ah |
23:23.20 | husimon | does it require digium hardware? |
23:23.24 | Qwell | nope |
23:23.31 | husimon | cause I have tdmoe |
23:23.48 | ManxPower | it DOES require zaptel. I doubt it will do you any good with TDMoE |
23:23.58 | ManxPower | You might have to talk to Digium |
23:24.05 | husimon | yeah probably doubtful |
23:24.14 | husimon | i do have a digium card I could test it with |
23:25.00 | ManxPower | You could use a Sangoma or other zaptel compat boards as well |
23:25.42 | husimon | does anyone know where I can read how to use the milliwatt application to do tx rx testing? |
23:26.00 | Qwell | show application milliwatt? |
23:26.42 | husimon | yeah i see how to use it in a dialplan, just wondering if there were any tricks |
23:26.50 | lmadsen | ummm.... it plays a tone... |
23:26.53 | lmadsen | not much else to figure out |
23:26.58 | *** join/#asterisk thedonvaughn (i=jayson@unaffiliated/printk) |
23:27.26 | husimon | yeah so then what do you adjust your tx/rx to, |
23:27.46 | husimon | the wiki says 100% |
23:28.06 | husimon | for the phone telco test number |
23:28.09 | ManxPower | the wiki is wrong half the time |
23:28.14 | ManxPower | you need ztmonitor I believe |
23:28.32 | husimon | i'm not even sure using milliwatt on the local side gives me anything |
23:29.16 | pkunkra | ok. so i've called my phone about ten times now. can't reproduce. but it seems to only happen with calls from other parts of the U.S. |
23:29.42 | husimon | pkunkra, i guess the best I can say is next time it happens go back into your cli and find that time and call |
23:29.51 | husimon | pkunkra, might want to output your cli to a text log file |
23:31.05 | pkunkra | husimon, now, here's the catch, a customer called me and i had the problem. then i called with my call, no issue. i then had the customer call back and it was the same issue. this all happened in about five minutes. |
23:31.20 | pkunkra | husiman, how do i output it to a text log? |
23:31.26 | pkunkra | that'd be useful. |
23:31.30 | husimon | pkunkra, i'm looking that up now |
23:32.16 | pkunkra | ok |
23:33.01 | ManxPower | /etc/asterisk/logger.conf |
23:35.38 | husimon | ManxPower, i see that it lets you specify levels of logging to different files |
23:35.49 | husimon | but how do you actually get the cli output in a log? |
23:37.49 | pkunkra | i suppose i could stuff it inside a screen or a script |
23:38.18 | husimon | it seems like you should be able to output it a log file |
23:39.25 | pkunkra | i would think so. |
23:39.29 | pkunkra | that's an important feature |
23:39.32 | husimon | either i'm missing something |
23:39.39 | husimon | or ... |
23:39.56 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
23:40.54 | pkunkra | none of the options in logger.conf seem to be a reproduction of the cli output. |
23:41.06 | pkunkra | it does dump some stuff in /var/log/asterisk/* |
23:41.14 | pkunkra | but it isn't all that useful |
23:42.19 | husimon | anyone care to comment? |
23:43.16 | pkunkra | ok. i have a hunch. |
23:43.20 | pkunkra | let me try something. |
23:43.39 | pkunkra | i'm going to start up a slew of traffic |
23:43.39 | pkunkra | bittorrent, etc. |
23:43.48 | pkunkra | and see if that might make the problem reappear again. |
23:44.27 | pkunkra | problem seems to happen in the middle of the day. busiest times, like 2-3pm. |
23:46.29 | husimon | pkunkra, oh you have a sip trunk? |
23:47.23 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
23:48.22 | pkunkra | it is sip. not sure if its a trunk or not. |
23:49.15 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
23:49.31 | pkunkra | googled. |
23:49.46 | pkunkra | it does run over the same wires as the regular internet traffic. |
23:49.59 | pkunkra | i just did a test call. lots of jitter and delay but no dropping. |
23:49.59 | Qwell | ~siptrunk |
23:50.00 | jbot | There is nothing special about a SIP trunk in the protocol like there is in the case of IAX2, for example. You set up a SIP trunk like a regular peer in sip.conf. |
23:50.11 | pkunkra | hah |
23:50.26 | pkunkra | its a regular sip peer then. |
23:50.26 | Qwell | meh, somebody changed that |
23:50.55 | pkunkra | ( i did have an IAX2 trunk before. it sucked and i switched it to sip ) |
23:51.16 | *** part/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
23:51.24 | pkunkra | seems sip is much better at handling jittery networks than iax2 is |
23:52.19 | pkunkra | but i do use an external provider to terminate the phone lines. the do a DID into my pbx in the datacenter. |
23:52.27 | pkunkra | the pbx then calls my extension. |
23:52.37 | husimon | obviously there must be some distinction between a sip peer that is a phone and that is a provider, so what do you use in conversation when you are talking about a sip peer that is your provider? "sip provider" ? |
23:53.13 | jameswf | husimon: the distinction is context |
23:53.16 | husimon | obviously you are trunking your traffic across that peer |
23:53.36 | husimon | seems like you should be able to use the word trunk. |
23:53.39 | *** join/#asterisk Ad-Hoc (n=nimbus@88.218.87.95) |
23:54.05 | jameswf | I got junk in my trunk hey |
23:54.08 | husimon | or is it just nope, use peer always and be smart enough to figure it out. |
23:54.12 | pkunkra | hah |
23:55.12 | husimon | jameswf, do you know of a way to log the messages on the cli to a file? the logger.conf doesn't really seem to do it. unless i'm totally stupid and don't understand the logger.conf (which is highly possible) |
23:55.38 | jameswf | yes add verbose |
23:56.18 | husimon | so "file = options,options,verbose" |
23:56.18 | pkunkra | i'm thinking the issue could be with one of two points in the network. either my cable router in my home office. or the provider i buy the voip from |
23:56.23 | husimon | err => |
23:57.14 | husimon | k jameswf thanks |
23:57.17 | husimon | pkunkra, you catch that? |
23:57.27 | pkunkra | i think i might be able to figure it out |
23:57.33 | pkunkra | looking in logger.conf now |
23:57.53 | *** join/#asterisk mchou (n=mchou@c-71-198-127-234.hsd1.ca.comcast.net) |
23:58.13 | husimon | btw i'm sure glad that was documented.... |
23:58.29 | mchou | anyone here happen to use linksys wrtp54g? |
23:58.44 | mchou | I mean the ATA portion |
23:58.47 | husimon | gonna add that to the wiki |
23:58.49 | husimon | verbose = cli.... |