| 00:04.02 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 00:07.08 | delmar | hi everyone. I have a bit of a curly little problem here that started a while ago and has gotten worse.... Incoming calls are just crap.... all chopped up and simply don't work, but outgoing calls work perfectly.. crystal clear. the path for calls in or out is... Polycom501 -- sip/g729 -- Asterisk on local lan --Iax/g729---- (through router ST608WL-Nat) --- Asterisk on public IP ----- SIP/alaw to Provider. |
| 00:08.31 | delmar | the codecs are exactly the same for a call in or out, and the first things I tried were replacing codecs, removing jitterbuffer and such with no luck. |
| 00:08.56 | delmar | outbound sounds great. inbound doesn't work.. |
| 00:10.00 | delmar | the next thing I want to do is throw away the ST608WL.. (omg u have no idea how much they suck)... and put my Linux box back in place as the main router and see if that cures it but I was wondering if anyone in here might have any suggestions or ideas. |
| 00:10.39 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 00:11.21 | d3wayne | delmar: it can't hurt to take the variable out of the equation |
| 00:11.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:12.54 | *** join/#asterisk dreamydevon (n=dreamyde@adsl-71-141-108-31.dsl.snfc21.pacbell.net) |
| 00:13.02 | dreamydevon | hello |
| 00:14.00 | dreamydevon | I was wondering if there are any asterisk gurus in her who know anything about the new Asterisk appliance |
| 00:14.02 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 00:15.05 | dreamydevon | anyone in here? |
| 00:15.48 | d3wayne | dreamydevon: what do you want to know ? |
| 00:16.06 | generalhan | dreamydevon: i see almost 300 people in here ! |
| 00:16.31 | dreamydevon | well... couple of things... first off just got one last week, pretty easy to set up, the only thing is I am having some issues with i guess some of the more advanced functionality |
| 00:16.48 | dreamydevon | for example I have three POT lines coming in to the box |
| 00:17.01 | dreamydevon | and want to use those for outbound calling in a roll over type set up |
| 00:17.17 | dreamydevon | right now we can only use a single line, others get the busy signal |
| 00:18.09 | pat2man | I seem to be having issues with agent penalties in general, anyone know of any good documentation? |
| 00:18.10 | *** join/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 00:18.11 | d3wayne | have you tried calling support ? |
| 00:18.25 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 00:18.57 | nny | 2 questions 1.) how do I check which zaptel modules are loaded on a box and 2.) anyone know why fxotune craps the bed with "could not fill input buffer" |
| 00:19.11 | dreamydevon | no havent tried that... been trying to figure it out on my own :-) |
| 00:19.32 | dreamydevon | and it seems that the appliance boxes dont support some basic commands either, like scp and rpm |
| 00:20.11 | d3wayne | your not going to get rpm, I think scp works |
| 00:20.28 | dreamydevon | so you cant install packages on it at all? |
| 00:20.30 | dreamydevon | like webmin? |
| 00:20.33 | delmar | d3wayne, yeah. im just gonna have to start changing stuff 1 by 1 until I can't do anything else. |
| 00:21.18 | delmar | i just can't understand why it would be so bad on incoming calls but not outgoing. |
| 00:21.41 | nny | dreamydevon: IMHO we use a 1ghz via box with the tdm400 (with fxo and fsx variations) and [insert your distro here] ... I have nothing against the appliance, but if you got the knack, and want the extra goodies, the blackfin based stuff is not all that customizable |
| 00:22.01 | d3wayne | dreamydevon: review your extensions.conf, you most likely have a configuration issue |
| 00:22.35 | *** join/#asterisk hyphenex (n=scott@60-241-154-60.tpgi.com.au) |
| 00:22.45 | dreamydevon | so they cripple the appliance so you cant add in stuff eh? hmmm |
| 00:23.31 | d3wayne | no, the appliance is not crippled so you can't add stuff to it |
| 00:23.56 | nny | dreamydevon: not crippled, just designed with a purpose |
| 00:23.56 | d3wayne | you may not be part of the market segment that it is targeted at |
| 00:24.02 | dreamydevon | hmmm |
| 00:24.17 | dreamydevon | i guess my vision of it was wrong then... |
| 00:24.44 | d3wayne | you may be envisioning stronger, more dense appliances |
| 00:24.49 | nny | dreamydevon: the blackfin proc and whatnot need specialized environments to run in |
| 00:25.38 | nny | i have a blackfin based box here (not comparing to appliance) and they are still woriing on getting pthreads fixed and callerid working.. otherwise the box runs out of resources :) |
| 00:25.52 | dreamydevon | hmmm |
| 00:26.33 | dreamydevon | so installing other stuff on it might be a bad idea... but it seems more limited then it was "advertised" to me as being... like of of the simplest stuff doesnt seem to be in there |
| 00:27.03 | d3wayne | you need to break your goals down into smaller segmets. The density your are attempting is nowhere near the limitations of the appliance |
| 00:27.20 | d3wayne | you can't be adding additional packages to it, so first thing is to draw a line there |
| 00:27.52 | d3wayne | the problems you are having w/ your 3-4 channels is a configuration issue, not an appliance limitation |
| 00:28.39 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 00:28.54 | dreamydevon | well yes that i know |
| 00:29.02 | nny | so anyone know how to check which version of zaptel module are loaded? |
| 00:29.18 | d3wayne | the gui provides you with access to all the configuration files, so you can always manually modify your configuration if the GUI doesn't support it yet |
| 00:29.30 | dreamydevon | well thats what i was looking at |
| 00:29.58 | d3wayne | I think its in the advanced tab, but let me check, it may take a couple minutes |
| 00:30.03 | dreamydevon | but was having a hard time understanding exactly what it was doing.... im not an asterisk guru, but ive been messing with it enough i understand a few things about it |
| 00:30.36 | d3wayne | one thing, after changes you need to activate your changes and save |
| 00:30.46 | d3wayne | its 2 different steps |
| 00:31.08 | dreamydevon | yeah i figured that out... kind of an annoyance :-/ |
| 00:31.18 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:31.51 | *** join/#asterisk crimethinker (i=ircuser@legacy.diamond.org) |
| 00:33.01 | *** join/#asterisk lyoungz (n=Larry@ool-4576723a.dyn.optonline.net) |
| 00:34.58 | d3wayne | dreamydevon: I don't know if its the fastest way to do it, but click <options>, then <advanced> |
| 00:35.16 | d3wayne | then <file editor> |
| 00:37.22 | delmar | OK the problem is IAX |
| 00:37.47 | delmar | outbound calls via IAX are working great. incoming via IAX don't, but SIP works fine. |
| 00:38.41 | dreamydevon | ok |
| 00:38.50 | shido6 | nat? |
| 00:38.56 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.181) |
| 00:39.01 | delmar | nat yes |
| 00:39.02 | shido6 | are you behind nat? |
| 00:39.45 | nny | so I am guessing that the Failure in fxotune is because it is expecting to break the line and then get silence, and something is failing |
| 00:39.45 | d3wayne | dreamydevon: did you find the files ? |
| 00:39.52 | nny | can someone help me figure out what? |
| 00:40.08 | nny | I have tried it manually, and it worked fine, not to mention it worked last week etc |
| 00:40.22 | d3wayne | nyy: what card are you using ? |
| 00:40.39 | nny | d3wayne: tdm400p with 2 fxo modules |
| 00:40.52 | dreamydevon | the file that I am looking at is the zapata.conf |
| 00:41.11 | dreamydevon | that is what controls the roll over if a line is busy correct? line grouping |
| 00:41.19 | dreamydevon | i mean trunk groups |
| 00:41.48 | delmar | So why would a call coming in via IAX sound really really terrible and choppy but the SIP call sounds fine... and the call path for IAX or SIP uses the exact same codecs.. so in theory its not a codec issue |
| 00:41.52 | unixdog | vicidial aka the perdictive dialer from hell |
| 00:42.10 | d3wayne | dreamydevon: you might want ring groups |
| 00:42.23 | dreamydevon | thats inbound |
| 00:42.30 | dreamydevon | im concerned with outbound at the moment |
| 00:42.32 | d3wayne | nyy: 1 sec |
| 00:42.34 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 00:42.38 | nny | d3wayne: np |
| 00:42.40 | dreamydevon | like if one line is using line one |
| 00:42.44 | dreamydevon | or |
| 00:42.47 | dreamydevon | thats confusing |
| 00:42.57 | dreamydevon | you have a person they pick up their phone |
| 00:43.00 | dreamydevon | they use line one to call out |
| 00:43.07 | nny | d3wayne: FYI tried to give it a full number to break the line, and it didn't even ring the number.. gonna try to find a splitter and listen in |
| 00:43.12 | dreamydevon | another person picks up their phone, the sytem puts them on to line 2 |
| 00:43.17 | dreamydevon | because line one is busy |
| 00:43.50 | d3wayne | 1 minute, I have to quiet a 1 year old |
| 00:45.13 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 00:47.07 | dreamydevon | k |
| 00:47.51 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 00:53.38 | d3wayne | dreamydevon: you can use the DIALSTATUS variable like the [macro-stdexten] |
| 00:54.05 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 00:54.08 | d3wayne | brb |
| 00:54.19 | dreamydevon | not farmiliar with DIALSTATUS, is there a good guide for that online somewhere? |
| 00:54.20 | dreamydevon | k |
| 00:55.50 | nny | d3wayne: this is looking like signaling? seems when * trys to dial 4 (as in -i 4) it returns a beep beep beep noise |
| 00:56.02 | nny | when i do it from a normal phone it is fine |
| 00:56.40 | nny | local ip uses a Centrex solution.. I use ks right now |
| 00:57.37 | nny | strangely, it only breaks when trying to do an fxotune.. if i dial out or use blind forward etc it works fine |
| 00:58.55 | docelmo | dreamydevon what do you want to know about DIALSTATUS? |
| 00:59.45 | dreamydevon | can I use it with a dialplan to have asterisk route outgoing calls thru a different trunk if the one line it is set to is busy? |
| 00:59.53 | docelmo | yes |
| 00:59.56 | docelmo | as well as AGI |
| 01:00.21 | nny | quick q, does it matter which line i use to do an fxo tune, say for example, I have another line here that is POTS and not using the telco centrex |
| 01:00.54 | docelmo | you would do something like a goto(x-${DIALSTAUS) and it would automaticly route to the priority x-noanswer or x-answer etc.. |
| 01:01.40 | docelmo | yes.. you can use it. FXO is FXO as long as it has a dialtone on it. |
| 01:02.05 | docelmo | techincly you could use a TDM FXO off of a PAP2's FXS port |
| 01:02.30 | nny | k gonna just try that |
| 01:02.44 | nny | this local centrx stuff if becoming more of a hinderance than anything |
| 01:03.39 | docelmo | if you are putting your POTS line into an asterisk box you dont need a centrex option |
| 01:03.47 | nny | well sigh |
| 01:03.51 | logyati | how does asterisk work with nat? |
| 01:04.01 | nny | docelmo: the local telco has a centrex they use to provide service |
| 01:04.13 | docelmo | I havent played with 1.4 but I know 1.2 sucks w/o a public IP |
| 01:04.30 | docelmo | nny you dont need centrex.. Just dialtone |
| 01:04.34 | nny | docelmo: good news: it offers things like flash hook transfer (transfer call AND free up line instantly) which is great for smaller systems bad news: it does stuff like this |
| 01:04.43 | nny | docelmo: gotcha |
| 01:04.50 | nny | docelmo: and a new telco |
| 01:04.51 | docelmo | all of the "centrex" features you are paying for are provided with the power of asterisk |
| 01:05.00 | *** join/#asterisk dreamydevon (n=dreamyde@adsl-71-141-108-31.dsl.snfc21.pacbell.net) |
| 01:05.05 | dreamydevon | ok im back |
| 01:05.30 | docelmo | nny yep.. hold, 3way, (20way if you wish), etc.. |
| 01:05.40 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-3e551c6573be1b95) |
| 01:05.46 | docelmo | Asterisk is a full Class5 switching platform |
| 01:05.55 | nny | docelmo: yeah true, except that dialtone has to come from somewhere.. in theory I can have a call come in to asterisk, and it uses a macro to send the call back out to a cell phone etc. and it actually fress the line up instantly for the next call.. saves all kinds of dough on lines |
| 01:06.08 | docelmo | and if you do it right you can make it a fault tollerant class4 switch with LCR also |
| 01:06.16 | nny | docelmo: obviously at a price as far as the differences from POTS though |
| 01:06.56 | docelmo | nny no.. 1 line for in and 1 line for out.. Now if you went sip you can usually find providers that offer say 5 calls on 1 DID at a time |
| 01:07.02 | docelmo | then have unlimited outbound |
| 01:07.07 | nny | docelmo: nope, 1 line in and back out :) |
| 01:07.26 | nny | docelmo: thats why I use their centrex, and only why |
| 01:07.27 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 01:07.37 | docelmo | nny you could use your home broadband connect to do all kinds of fun things |
| 01:07.38 | nny | docelmo: and* it only uses the line for 5 seconds at max |
| 01:07.41 | dreamydevon | so dialstatus? |
| 01:07.53 | docelmo | what about it? |
| 01:08.03 | docelmo | ohh you bailed before seeing my answer.. |
| 01:08.10 | dreamydevon | yes sorry |
| 01:08.15 | docelmo | you would do something like a goto(x-${DIALSTAUS) and it would automaticly route to the priority x-noanswer or x-answer etc.. |
| 01:08.16 | dreamydevon | connection cut out |
| 01:08.21 | nny | docelmo: yeah true.. been looking at sip, but the broadband here is not all that great.. thats "in development" and subject to testing, (as far as SIP as a medium goes) |
| 01:08.43 | nny | docelmo: but yeah.. using the old pots line for the fxotune for now |
| 01:09.21 | docelmo | nny where on gods green earth are you? |
| 01:10.05 | nny | docelmo: Hilton head island, SC.. we have Time Warner and a local co called Hargray |
| 01:10.45 | nny | docelmo: the telco offers a metro ethernet option (1/2 price t1 kind of) but other than that.. flakiness is enough to where a old fashioned POTS or similar is required |
| 01:10.48 | docelmo | btw.. if anyone is wondering I have over 4 years asterisk experience about 150 deployments from single pbx to 20+ gateway class 4 solution. Im also dCAP 1.0.x, 1.2.x, 1.4.x |
| 01:11.15 | dreamydevon | docelmo: where are you located? |
| 01:11.29 | docelmo | I would get the metro ethernet for internet access. If you can justify it get a PRI solution |
| 01:11.40 | dreamydevon | east or west coast? |
| 01:11.44 | docelmo | I am in Newark, DE about 50 miles south of central philly |
| 01:11.50 | dreamydevon | ah |
| 01:11.57 | nny | docelmo: yeah our larger clients (price break at 15 lines) use pri |
| 01:12.05 | nny | docelmo: we are developing boxes for small business |
| 01:12.08 | nny | soho etc |
| 01:12.16 | nny | docelmo: using 2 lines |
| 01:12.26 | nny | lots of local businesses are in and out of office a lot |
| 01:12.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:12.41 | docelmo | ahh.. I have made asterisk systems that weigh less than a pound and are diskless and fanless |
| 01:12.45 | nny | so the idea that 2 lines can server 1 phone + numerous cell phones is pretty good |
| 01:13.09 | nny | docelmo: good stuff good stuff |
| 01:13.39 | wishes | docelmo: oh so are you a guru? |
| 01:13.46 | dreamydevon | so am i right in looking at the zapata.conf? |
| 01:13.51 | wishes | :D |
| 01:13.54 | docelmo | Well technically when the call comes in POTS you have access to send the call out one more time if a 2 port FXO system OR 2 incoming calls if in a hunt group or someone calls the 2nd line |
| 01:14.13 | hyphenex | would the Cisco 7931g (non SIP phone) work with asterisk even though it's not on the list of supported devices? |
| 01:14.26 | docelmo | wishes I dont know about a GURU but I was chosen to speak at astricon again this year.. I had to however cancel due to family issues. |
| 01:14.33 | jm|laptop | !goaix |
| 01:14.40 | docelmo | dreamydevon for what? |
| 01:14.51 | dreamydevon | to set up the roll over |
| 01:15.22 | wishes | docelmo: so can you tell me how i can get wengophone with video to work nicely with record() on 1.2 (no i cant upgrade without mega major hassles) |
| 01:15.22 | docelmo | for calling to the PSTN yes. You setup groups in the /etc/zapata.conf I belive I would have to look at the file |
| 01:15.28 | jm|laptop | ~goaix |
| 01:15.35 | wishes | ive been trying to figure a work around for a day or so |
| 01:15.43 | docelmo | hyphenex yes if it uses skinny |
| 01:16.00 | *** join/#asterisk mangolian (n=Frossty@CPE00c049e0d0b4-CM00111ae2bb20.cpe.net.cable.rogers.com) |
| 01:16.03 | dreamydevon | [trunkgroups] |
| 01:16.03 | dreamydevon | <PROTECTED> |
| 01:16.03 | dreamydevon | <PROTECTED> |
| 01:16.04 | dreamydevon | <PROTECTED> |
| 01:16.04 | dreamydevon | <PROTECTED> |
| 01:16.04 | dreamydevon | <PROTECTED> |
| 01:16.07 | dreamydevon | something like that |
| 01:16.12 | hyphenex | docelmo: I've heard it does not work with dkinny. What about the other SCCP driver? |
| 01:16.13 | docelmo | jm|laptop with all due respect to Mark Spencer which I know fairly well.. IAX SUCKS! :) |
| 01:16.17 | dreamydevon | trunk groups i believe is what i want to set up |
| 01:16.23 | jm|laptop | docelmo: :O |
| 01:16.38 | jm|laptop | docelmo: I've Googled it now; it wasn't even really compatible with my country |
| 01:16.44 | jm|laptop | moreover: I quite rate IAX :/ |
| 01:16.57 | wishes | docelmo: http://pastebin.ca/709590 |
| 01:16.58 | jm|laptop | mind you I don't really have NAT problems |
| 01:17.11 | docelmo | wengophone video will not work on 1.2.. 1.2 doesnt support h.264 video |
| 01:17.14 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 01:17.21 | wishes | actually it does work fine |
| 01:17.39 | hyphenex | wishes: the Cisco 7931g? |
| 01:17.41 | wishes | its the record() im having issues with, i specifically set it to use wav and it still gives me the above error |
| 01:17.43 | docelmo | really.. did you get a custom driver for 264 somewhere? I couldnt get it to work |
| 01:17.56 | docelmo | lemme look at the pastebin |
| 01:18.00 | wishes | i didnt set this system up, i got landed with it :/ |
| 01:18.41 | docelmo | file.c:243 ast_writestream: Unable to translate to format h263, <--- This is VIDEO |
| 01:18.44 | wishes | but yeah we've done video to video calls fine before, they all work. Id love to upgrade to 1.4 for functionality but i would have to spend a weekend doing it and have some sort of emergancy 'if the shit hits the fan revert' button :D |
| 01:19.07 | wishes | docelmo: yeah i know thats video, i dont want video on the recorded messages - basicly the record is recording a custom voice message |
| 01:19.20 | docelmo | wishes I would say when the call is up check your codecs.. sip show channels and make sure there is a path. You also need you make sure the media is coming thru the asterisk box otherwise record will not work |
| 01:19.59 | *** join/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 01:20.18 | docelmo | check your voice codecs by using sip show channels for the channels up. |
| 01:20.22 | docelmo | This will let you know. |
| 01:20.23 | wishes | the function works with the fugly grandstream |
| 01:20.35 | denon | anyone awake from NZ? |
| 01:20.41 | wishes | denon: aye |
| 01:20.45 | docelmo | Also make sure your RTP is coming thru the asterisk box |
| 01:20.46 | wishes | its the middle of the day |
| 01:21.05 | docelmo | wishes if you dont mind I could log in and give you a quick hand.. seems like a 2 second fix.. |
| 01:21.14 | docelmo | well 2 second diagnosis |
| 01:21.59 | nny | mwah? |
| 01:22.06 | nny | no fxotune says no modules present |
| 01:22.08 | nny | -_- |
| 01:22.13 | wishes | docelmo: its behind a firewall |
| 01:22.13 | nny | oh nm |
| 01:22.16 | nny | asterisk |
| 01:22.22 | dreamydevon | docelmo: is trunk groups the method for in a dialplan having a phone roll over to subsequent lines if one is busy? |
| 01:22.44 | wishes | and yeah it works fine for any normal phone, just not the softphone/wengo |
| 01:22.49 | nny | oh sweet god shoot me |
| 01:22.57 | nny | Tuning module /dev/zap/1 |
| 01:22.58 | nny | Could not fill input buffer - got -1 bytes, expected 4000 bytes |
| 01:22.58 | nny | Failure! |
| 01:23.02 | nny | why! |
| 01:23.12 | nny | I can pick up the line and do a tune just fine |
| 01:23.18 | docelmo | dreamydevon yes.. you would do a dial(ZAP/g1/${EXTEN}) the g1 represents Group 1 |
| 01:23.33 | nny | well * I can dial a 4* |
| 01:23.47 | dreamydevon | but how do i create group 1? is there a simple guide online you could point me 2? |
| 01:25.26 | nny | dreamydevon: have you seen "the book"? |
| 01:26.22 | nny | docelmo: so any thoughts as to why I can't fxotune this box, both on a POTS and on the whole centrex based dial tone thing? |
| 01:27.02 | wishes | docelmo: see privmsg pastebin |
| 01:27.34 | nny | docelmo: it picks up.. pauses for a sec, dials "4" (as -i 4 says it should) and then just goes Breeep Breep Breep (technical term).. If i pick up a regular handset on a splitter and try the same thing it works fine |
| 01:27.36 | dreamydevon | no what is the book |
| 01:27.47 | nny | anyone know how to make the bot to the thing with the book? |
| 01:27.56 | nny | dreamydevon: i have one here.. let me find clicky |
| 01:28.01 | dreamydevon | ok |
| 01:28.02 | dreamydevon | thanks |
| 01:28.08 | docelmo | nny check dtmf |
| 01:28.53 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 01:29.00 | nny | dreamydevon: http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip |
| 01:29.15 | nny | dreamydevon: I am just starting out myself, that book helped a metric ton |
| 01:29.28 | nny | I got the dead tree version for ease of reading |
| 01:29.37 | nny | docelmo: as far as relaxed/ yes or no? |
| 01:29.54 | docelmo | tone or pulse |
| 01:30.40 | docelmo | ??????????????? anyone know how to make the bot to the thing with the book? |
| 01:30.40 | nny | docelmo: sorry, which config file would that be under for zaptel? |
| 01:30.46 | docelmo | what do u mean? |
| 01:30.55 | nny | docelmo: lol was trying to get the link for the asterisk book |
| 01:30.57 | nny | !book |
| 01:30.59 | nny | dunno |
| 01:32.00 | docelmo | ahh. holding daughter.. so its gonna take me a few to respond.. |
| 01:32.06 | nny | np |
| 01:32.15 | docelmo | I believe the one in /etc |
| 01:32.29 | rob0 | Is there a way at the console to see the caller ID of an active call? Seems like that is only logged after it hangs up. |
| 01:32.34 | docelmo | I know Zaptel to look at it. I dont know it well enough to spit it out |
| 01:32.45 | rob0 | oh nm I see it now |
| 01:32.47 | nny | docelmo: yeah thats zaptel.conf... nothing related to dtmf in there |
| 01:33.09 | rob0 | wasn't in "core show channels" but it is in "sip show channels". |
| 01:33.10 | docelmo | rob0 yes add a priority with NoOp(${CALLERID(all)} |
| 01:34.14 | docelmo | nny like i said I would need to see the zap config and play with it. I dont know it well enough to just say X is the answer |
| 01:34.28 | wishes | how old is the daughter? |
| 01:35.19 | docelmo | 3mos |
| 01:35.25 | wishes | aww the cute stage :D |
| 01:35.33 | rob0 | nice |
| 01:35.36 | wishes | my youngest daughter is at the 'OMG PONIES!' stage |
| 01:35.38 | docelmo | yes and no |
| 01:35.42 | wishes | shes 10 |
| 01:36.07 | rob0 | I have a 9-y-o and 2 grown stepdaughters. |
| 01:36.25 | wishes | docelmo: so what do i do to fix my problems - or give me something to google :) |
| 01:36.32 | docelmo | she's cute in the morning when she first wakes up and when she is sleeping and just waking up from sleep. Other than that is holy terror when she is hungry |
| 01:36.53 | wishes | docelmo: just wait till she hits teenhood |
| 01:38.20 | nny | so what is the difference between zapata.conf and zaptel.conf? |
| 01:38.22 | docelmo | sigh.. I dont want to even think about her dating.. |
| 01:38.43 | docelmo | zapata is the hardware interface.. and zaptel is the software interface |
| 01:38.55 | wishes | docelmo: sports! get her into sports! when she hits teenhood it solves a whole lot of problems |
| 01:39.17 | wishes | my oldest would rather be cycling or hockey or softball etc than looking at boys |
| 01:39.18 | docelmo | tell that to my wife.. |
| 01:39.49 | docelmo | I dont kow.. but lemme try and find a solution for one of you guys.. I have a project I was working on I am going to try and get back to |
| 01:40.02 | wishes | thanks for the help btw, i appretiate it |
| 01:40.20 | wishes | im almost debating rebuilding this from scratch because the other 2 admins left it undocumented huge mess |
| 01:40.39 | docelmo | wishes where are you? |
| 01:40.59 | docelmo | I do consult.. :) And Im probably the best back for your buk considering what most charge.. :) |
| 01:41.06 | wishes | New Zealand :D |
| 01:41.17 | wishes | nah im the network manager (as of a month ago) |
| 01:41.24 | wishes | so its my job to do |
| 01:41.32 | wishes | just trying to sort the mess the last couple of guys left |
| 01:42.07 | docelmo | ahh.. well I would say upgrade to 1.4 it should solve about 50% of your problems the other 50% is add canreinvite=no to your SIP peers |
| 01:42.16 | wishes | id never touched asterisk before a month back, now im pretty fluent |
| 01:42.50 | docelmo | wishes I thought that too 4 years ago.. hehe |
| 01:43.08 | docelmo | asterisk == tool kit.. I tell most clients the limit of asterisk is their imagination |
| 01:43.33 | wishes | well ive already re-written most of the previous stuff |
| 01:43.40 | docelmo | brb |
| 01:43.42 | wishes | now its running out of a database |
| 01:43.43 | docelmo | bottle time.. |
| 01:44.06 | traxx | hi. anybody know how i can show missed calls in asterisk? |
| 01:44.11 | wishes | hooked up with the intranet so users can set their status (busy, in meeting, away sick and custom recorded) via internet etc |
| 01:44.33 | wishes | and eventually in my spare time im gonna write in a D&D style easter egg :D |
| 01:44.41 | *** part/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 01:45.13 | wishes | "you are in a large field facing north ... in front of you there is a castle with an old man ... press 1 to speak to the man, press 2 to enter the castle ..' |
| 01:45.24 | wishes | festival+mysql+funtimes :D |
| 01:45.35 | wishes | anyway, thats for another day/time :) |
| 01:45.46 | wishes | nothing like amusement for clients |
| 01:46.55 | mangolian | how do i add DTMF in the logger.conf ?? |
| 01:47.15 | docelmo | DTMF? |
| 01:47.40 | docelmo | what do you mean add it to logger.conf? |
| 01:47.46 | docelmo | you cant put dtmf in logger.conf |
| 01:48.12 | docelmo | wishes did you get what I said to fix yer problem? |
| 01:48.38 | mangolian | i wanted to c wat kind of DTMF signals r comning in |
| 01:48.50 | mangolian | some of the guys here recommended it |
| 01:49.55 | docelmo | what do you mean see? The only thing I can tell you is wireshark.. thats the simplest way to get the most for your information unless its inband. |
| 01:49.57 | mangolian | i can't recive any DTMF signal from an external netwrok (WAN) |
| 01:50.15 | docelmo | RFC2833 or SIP INFO? |
| 01:50.35 | mangolian | well tried them all but RFC2833 |
| 01:51.08 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 01:51.34 | wishes | docelmo: dunno , just checking now |
| 01:51.37 | wishes | heh :D |
| 01:51.49 | wishes | feck sakes |
| 01:52.41 | mangolian | works fine when the account connects from internal network but when the same account from an external network the DTMF doesn't get recnized =[ |
| 01:53.38 | wishes | docelmo: nope, didnt help |
| 01:53.43 | docelmo | mangolian do you have problems with audio? |
| 01:53.51 | docelmo | wishes you tried asterisk 1.4 already? |
| 01:54.06 | docelmo | wishes also if you could get me a quick SDP that would be cool.. |
| 01:54.08 | mangolian | had problem installing 1.4 |
| 01:54.21 | mangolian | connection is perfect just the DTMF |
| 01:54.38 | docelmo | mangolian make sure you set DTMFTYPE=rfc2833 for both sides |
| 01:54.48 | docelmo | rfc2833 is the simplest of all dtmf types |
| 01:54.53 | docelmo | err dtmfmode |
| 01:55.00 | mangolian | yup so i heard |
| 01:55.15 | wishes | docelmo: nah, but the careinvite |
| 01:55.16 | wishes | http://pastebin.ca/709892 |
| 01:55.20 | wishes | :( |
| 01:55.26 | *** join/#asterisk Entr4nced (n=Entr4nce@cpe-76-190-132-60.neo.res.rr.com) |
| 01:56.13 | mangolian | its weird cause the same account works internaly but not externaly(WAN) |
| 01:56.39 | wishes | meh, hometime for me anyway |
| 01:56.39 | docelmo | mangolian is your asterisk behind a nat? |
| 01:56.50 | wishes | thanks for helping docelmo :) |
| 01:56.58 | docelmo | all good.. Sorry couldnt help more |
| 01:57.04 | mangolian | if nat means router than yes |
| 01:57.06 | docelmo | hold on tho.. lemme look at the pastebin |
| 01:57.06 | wishes | and yeah i think ill look into starting over again with the darned thing |
| 01:57.17 | docelmo | no nat.. router doesnt mean anything |
| 01:57.37 | wishes | it created 2 files , 1 .wav and 1 .h263 |
| 01:57.50 | docelmo | wishes I ment the SDP which is the SIP debug |
| 01:58.45 | docelmo | the invite |
| 02:01.06 | mangolian | im not sure abt the NAT part but i have tried turing it on and off...didn't help |
| 02:08.16 | *** join/#asterisk Entr4nced (n=Entr4nce@cpe-76-190-132-60.neo.res.rr.com) |
| 02:20.55 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 02:23.31 | *** join/#asterisk holiday42 (n=me@71-210-73-208.farg.qwest.net) |
| 02:26.00 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 02:27.38 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 02:27.38 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 02:34.40 | *** join/#asterisk TJNII (n=TJNII@71-32-163-67.desm.qwest.net) |
| 02:35.17 | TJNII | Has anyone played with a SoundPoint IP 400? I think I've picked up a dud, and I'm stabbing in the dark. |
| 02:39.10 | *** join/#asterisk blmm (n=rehn@adsl-76-225-158-248.dsl.pltn13.sbcglobal.net) |
| 02:41.22 | blmm | How do I format the caller ID number for an international number when passing it to a US cell phone (T mobile)? I tried some things like 011xxyyyyyyy, but most formats are just rejected and show as "private number". |
| 02:44.13 | Corydon76-dig | blmm: +CCnumber |
| 02:45.25 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:45.28 | blmm | Tried that, doesn't work. |
| 02:46.06 | Corydon76-dig | Call your service provider and ask, then |
| 02:46.22 | Corydon76-dig | Are you able to send ANY callerid? |
| 02:46.44 | Corydon76-dig | Call your rep and ask them to remove the filter |
| 02:46.52 | blmm | Yes. Without the "+", the first number gets overwritten by a 1, for instance. |
| 02:47.46 | blmm | I somehow doubt they would change their system for me... |
| 02:47.50 | Corydon76-dig | Might also be that tmobile is rejecting it |
| 02:48.00 | *** join/#asterisk FireMac (n=firemac@CPE000e08ea3707-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 02:48.23 | Corydon76-dig | blmm: you can have them remove the filter. Just explain what you're trying to do and they'll remove it |
| 02:48.51 | Corydon76-dig | They're only trying to prevent false cid... if you show them that you're in good faith, they'll remove it |
| 02:48.52 | blmm | Oh, you mean my outgoing SIP provider. I was assuming T-mobile did this. |
| 02:49.11 | Corydon76-dig | Oh, it's not PRI? |
| 02:49.20 | Corydon76-dig | Get a PRI, they work better |
| 02:49.39 | Corydon76-dig | SIP providers tend not to do more than how they're initially set up |
| 02:49.44 | blmm | I see... |
| 02:49.52 | Corydon76-dig | unless you're buying a ton of minutes from them |
| 02:50.34 | Corydon76-dig | and even then, it's a matter of what they know how to do |
| 02:51.11 | Corydon76-dig | I know firsthand that you can send cid "911" to tmobile and they'll accept it and display it on the phone handset |
| 02:51.18 | blmm | That's what I thought. Well, I'd have to find out who is mangling the CID first; my provider or T-mobile. I'll compare to what another phone says. |
| 02:51.23 | Corydon76-dig | but then again, I'm using a PRI |
| 02:51.51 | FireMac | hi, |
| 02:52.11 | blmm | OK. I'm using Vitelity. |
| 02:52.31 | Corydon76-dig | In fact, I have unintentionally sent my internal 3-digit extension across multiple telcos from Colorado to Tennessee and it showed up on our PRI in TN |
| 02:52.47 | Corydon76-dig | So it can be done |
| 02:53.06 | Corydon76-dig | Congressional interference notwithstanding |
| 02:53.11 | FireMac | i just talked to a tech support from my voip provider to find out if they use sip or iax. the guy said they use sip |
| 02:53.18 | blmm | Yes, that would be a pain... |
| 02:53.31 | FireMac | aside from knowing it is sip, what other info should i know to setup my sip.conf? |
| 02:53.59 | Corydon76-dig | I prefer when my lawmakers are locked in gridlock and doing nothing at all |
| 02:54.40 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 02:55.41 | blmm | They are pretty close though. Unless you happen to take out an inappropriate ad... |
| 02:56.57 | blmm | My vitelity log shows all CID:s as sent, so I still suspect T mobile. I'll try sending 911. Or maybe 711 to be less suspicious. |
| 02:58.09 | blmm | That did not come through either. |
| 03:00.42 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 03:01.13 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) |
| 03:01.21 | *** join/#asterisk ManxPower (n=manxpowe@ip68-107-250-148.no.no.cox.net) |
| 03:03.19 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:03.19 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:03.25 | J4k3 | caller ID != ANI |
| 03:03.35 | J4k3 | afaik its perfectly ok to BS CID |
| 03:03.43 | J4k3 | but trying to falsify ANI data is a no-no |
| 03:05.13 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:06.09 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 03:11.18 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.4.194.revip2.asianet.co.th) |
| 03:12.10 | HaMYaI | does it exist a way to combine sip static + realtime users? |
| 03:12.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:14.40 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:26.11 | delmar | So would anyone like to hazard a guess or offer any suggestions or ideas as to why calls going out over IAX are fine but calls received via IAX are utter crap and all chopped as if there is congestion, while calls received along the same route with same codecs but using SIP.. are perfectly clear and fine. ? |
| 03:41.55 | *** join/#asterisk TJNII (n=TJNII@67-41-106-46.desm.qwest.net) |
| 03:50.29 | *** join/#asterisk techie (n=techie@adsl-69-232-119-5.dsl.frs2ca.pacbell.net) |
| 03:52.30 | *** join/#asterisk bmg505 (n=leon@196.209.176.38) |
| 03:53.31 | *** part/#asterisk techie (n=techie@adsl-69-232-119-5.dsl.frs2ca.pacbell.net) |
| 03:58.58 | mrtelephone | delmar? |
| 03:59.27 | delmar | mrtelephone, yes? |
| 04:04.53 | *** join/#asterisk OrNix (n=ornix@gw.nsib.ru) |
| 04:15.22 | J4k3 | cisco 186's any good? |
| 04:17.03 | Strom_M | ancient junk |
| 04:21.14 | *** join/#asterisk seebs (n=seebs@mail.seebs.net) |
| 04:21.47 | seebs | Hey, I have a kubuntu box with asterisk. It behaves INSANELY weirdly after asterisk has been up for a while. Any hints on likely mistakes I could have made configuring * or the zt drivers? |
| 04:22.05 | seebs | # date; sleep 1; date |
| 04:22.06 | seebs | Mon Sep 24 23:12:54 CDT 2007 |
| 04:22.06 | seebs | Mon Sep 24 23:13:30 CDT 2007 |
| 04:22.11 | seebs | <-- that is not good. |
| 04:23.46 | Strom_M | your first mistake is running Asterisk on a box that is also running x windows |
| 04:24.05 | seebs | Okay, I could shut X down, no skin off my back. |
| 04:24.29 | Strom_M | servers do not need x windows :) |
| 04:24.48 | seebs | No, they don't. |
| 04:28.25 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) |
| 04:30.24 | seebs | I'll reboot with X disabled (and some other junk that was on by default, which I don't need) and see how it looks then. |
| 04:31.42 | seebs | The other obvious question is whether the zaptel driver (assuming a digium card in the system) is expected to have any noticable impact on timers. |
| 04:32.49 | *** join/#asterisk vitaminmoo (n=vitaminm@70.58.177.109) |
| 04:40.22 | tzafrir_home | seebs, please be more specific |
| 04:40.33 | tzafrir_home | What do you expect to happen? What happens? |
| 04:41.00 | seebs | What I expect to happen: System behaves "normally" -- 'sleep 1' takes barely over a second, commands run quickly, and so on. |
| 04:41.08 | tzafrir_home | Shutting down X does *not* take a reboot. Especially not on Debians, where the *dms are merely services |
| 04:41.22 | seebs | What actually happens: 'sleep 1' takes 36 seconds according to the clock. Anything that does any kind of delay delays about 30x as long as expected. |
| 04:41.45 | seebs | And I know it doesn't need a reboot, but I wanted to be sure that there weren't lingering problems caused by X mucking about with my AGP card. |
| 04:41.48 | tzafrir_home | Is the CPU busy at that time? |
| 04:41.57 | tzafrir_home | Asterisk very busy? |
| 04:42.18 | seebs | No discernable load. Asterisk not even doing anything (well, nothing active; presumably it's watching the lines in case they are picked up). |
| 04:42.24 | tzafrir_home | What display adapter do you have? |
| 04:42.27 | seebs | System load on the order of 0.01. |
| 04:42.35 | seebs | It's an old AGP Radeon 7000 with no functioning IRQ. |
| 04:42.36 | OrNix | tell me please, when does number begin with " _ " (for example _XXX) and when without _ in extensions.conf ? (sorry for my poor english) |
| 04:42.57 | tzafrir_home | I think its drivers are farly good |
| 04:43.14 | seebs | The machine seems "normal" immediately after being brought up, but then after maybe one call goes through, starts being very very odd and delayed. |
| 04:43.35 | tzafrir_home | (as opposed to the sucky VIA I have at work, that manages to kill my display once in a few weeks) |
| 04:43.53 | J4k3 | seebs: I learned something real quick with asterisk... don't run it on garbage, or expect a lot of weird hassles. |
| 04:44.04 | tzafrir_home | 20 seconds is a lot of time. |
| 04:44.24 | J4k3 | and why-oh-why-oh-why are you running X on your * box? |
| 04:44.28 | seebs | Well, the machine itself should be okay. It's an IBM desktop, no sucky parts except the video card, and now that X is disabled, it should just be staying in text mode. |
| 04:44.35 | tzafrir_home | seebs, ps aux | grep D |
| 04:44.37 | seebs | Card's a digium ... er. wcfx..s, I think? 4-port. |
| 04:44.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:44.58 | tzafrir_home | do you see processes in state D? (as opposed to S or R)? |
| 04:45.02 | seebs | (Will be a bit, the machine's still got its 30x delay thing going, and is gradually rebooting, but every "sleep 1" in that process takes about 30 seconds.) |
| 04:45.49 | tzafrir_home | another thing, just to eliminate problems: run 'date' , wait a second, and run it another time |
| 04:46.13 | seebs | That produced expected results -- reported time moved by 0-2 seconds depending on typing speed. |
| 04:46.16 | tzafrir_home | Does it really show exactly one second past? |
| 04:46.45 | tzafrir_home | (if the system clock s not OK, you'll probably also eventually get very long console beeps) |
| 04:46.52 | seebs | Well, roughly. If I type it really fast and start early in a second, I get 0; if I start really late in a second, I get 2. But yes, normally it's just a second. |
| 04:47.11 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:47.13 | tzafrir_home | So the system clock seems OK |
| 04:47.21 | seebs | Yeah. But 'sleep 1' takes 36 seconds. |
| 04:47.30 | tzafrir_home | strace sleep 1 |
| 04:47.34 | tzafrir_home | see where it hangs |
| 04:47.54 | seebs | Good thought. I'll see what happens when the system comes back. (I have normally been giving it around 15 minutes for the reboot cycle...) |
| 04:48.00 | Frogzoo | seebs: you thrashing? |
| 04:48.17 | seebs | Pretty sure not. Machine's "only" got I think 512MB, P4 2.26GHz or something. |
| 04:48.28 | seebs | And I'm just ssh'd in, nothing else running but *. |
| 04:48.36 | Frogzoo | how much swap? |
| 04:48.45 | tzafrir_home | 15 minutes for a reboot cycle??? What does it do ? |
| 04:49.08 | seebs | I forget how much swap, but none in use. |
| 04:49.32 | Frogzoo | seebs: you checked /var/log/messages - for failing disk errors for example? |
| 04:49.34 | seebs | tza: Every shutdown script tends to do some kind of 'do this, give it a second, then do that' routine. We never notice because they go so fast. |
| 04:49.53 | seebs | Yes, no warnings or anything. Also, fresh after boot, it's fine. |
| 04:50.15 | tzafrir_home | A shutdown is normally faster than the startup, as there's nothing to set up |
| 04:50.21 | Frogzoo | so you have a memory leak? |
| 04:50.30 | seebs | Yeah, but there's a lot of kill, then wait, then kill -9. |
| 04:50.35 | tzafrir_home | 15 minutes mean "something is terribly wrong" |
| 04:50.40 | seebs | I don't think it's a memory leak -- if it were, the system wouldn't have tons of free memory. |
| 04:50.42 | seebs | Well, yes. |
| 04:51.04 | seebs | Something is terribly wrong, which is that, at some point after boot, every attempt by the system to use timers delays for about 30+ times as long as normal. |
| 04:51.21 | Frogzoo | tons of free memory on linux as atypical of course |
| 04:51.23 | tzafrir_home | BTW: check /var/log/kern.log for suspicous kernel messages |
| 04:51.32 | seebs | I will check that, too. |
| 04:51.37 | tzafrir_home | e.g: disk problems, as suggested before |
| 04:52.04 | seebs | I don't think it's that unusual for a barely-loaded box to have virtually no load. I was running this on a P3 550 previously; I only "upgraded" because I wanted a smaller and quieter chassis. |
| 04:52.33 | Frogzoo | I would focus on the 'sleep takes 30 seconds' issue |
| 04:52.39 | tzafrir_home | seebs, what kernel version is that, exactly? What version of (k)ubuntu? |
| 04:53.53 | tzafrir_home | If you suspect a specific system call misbehaving, then strace is your friend to trace it |
| 04:55.44 | seebs | 7.0.4 (kubuntu). 2.6.20.16, I think. |
| 04:56.03 | seebs | Hmm. |
| 04:56.09 | seebs | Well, okay. After a reboot, date's fine. |
| 04:56.12 | *** join/#asterisk Putzz (n=me@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 04:56.38 | seebs | Something that occurred to me, was to disable all the APM features in the BIOS, because a system trying to power-save can EASILY break timer stuff. |
| 04:57.26 | seebs | So now it's working fine... If it breaks again, I have some ideas for things to check on. |
| 04:57.42 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 05:00.23 | seebs | I should see a tad over 1k interrupts per second on wctdm, right? |
| 05:00.40 | seebs | So, 1008-1013 per second should be okay? |
| 05:02.30 | seebs | BTW, this reminds me that I have another question. I have * set up just to record all incoming and outgoing calls, which it does by having any extension dialed trigger monitoring. |
| 05:03.12 | seebs | This has the odd side effect that, if I dial something DURING a call, long after it's started, * starts a new monitoring item with the "phone number" I dialed (usually someone's voicemail hell). |
| 05:03.52 | seebs | e.g.: |
| 05:03.54 | seebs | [house2] |
| 05:03.54 | seebs | ; setup for when a house line picks up Line 2 |
| 05:03.54 | seebs | exten => _[1-9#*]., 1, NoOp(Line 2 outbound to ${EXTEN}) |
| 05:03.54 | seebs | exten => _[1-9#*]., 2, Set(CALLFILENAME=${TIMESTAMP}-OUT-6792-${EXTEN}) |
| 05:03.54 | seebs | exten => _[1-9#*]., 3, MixMonitor(${CALLFILENAME}.wav|b) |
| 05:03.55 | seebs | exten => _[1-9#*]., 4, Dial(ZAP/4/${EXTEN}, 60, r) |
| 05:04.23 | seebs | Is there a way to make these apply only once when I pick up, and then to just have the tones recorded along with the rest of the call until an actual disconnect? |
| 05:04.45 | Strom_M | first off: take the spaces out after the commas |
| 05:05.04 | Strom_M | second: take that stupid "r" flag off of Dial() |
| 05:05.09 | seebs | Okay. |
| 05:05.25 | seebs | Does that mean },60,), or just },60) |
| 05:05.33 | Strom_M | the latter |
| 05:06.02 | seebs | What's the r do, anyway? This is, as you might have guessed, all cargo-cult. |
| 05:06.35 | Strom_M | it completely fucks up any call progress information you may receive |
| 05:06.40 | seebs | Ahh. |
| 05:07.00 | Strom_M | there is a use for it, but it's only rarely necessary |
| 05:07.11 | seebs | The whole thing exists only because I hate junk faxers and the like, and decided to start calling them and recording their stuff. |
| 05:07.39 | seebs | So, was that why it was spawning new calls whenever I pressed a button, or is that a separate thing? |
| 05:08.16 | Strom_M | no |
| 05:08.43 | Strom_M | i dont know why it was doing that; i'd need more info |
| 05:09.10 | Strom_M | you also probably want to take the 60 second timeout off your Dial() string as well |
| 05:10.29 | seebs | Hmm. I wonder if that's it -- it could be that, once that timeout was up, asterisk was interpreting button presses as new calls. |
| 05:10.40 | [TK]D-Fender | you should also probably HANGUP after your dial. maybe its falling back afterwares causing those calls.. |
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| 05:12.29 | seebs | I'm a bit confused by that. I would assume that hangup would break the connection, no? |
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| 05:13.53 | [TK]D-Fender | seebs, no. |
| 05:13.57 | seebs | Egh, I just realized how late it is. I might be back another time to seek more sa |
| 05:14.05 | seebs | ... Oh? Hmm. Maybe I should fix this. |
| 05:14.12 | [TK]D-Fender | seebs, if the next priority executes its because the OTHER SIDE hungup. |
| 05:14.26 | seebs | Ahh. |
| 05:14.28 | [TK]D-Fender | seebs, So you want to make sue that when its over, its OVER |
| 05:14.33 | seebs | Oh, I get it! |
| 05:15.01 | seebs | So, the Dial command runs until the conversation ends... And only when it's done running, the next command would execute. |
| 05:15.36 | [TK]D-Fender | seebs, actually... even then generally not... |
| 05:15.41 | seebs | Hey.... |
| 05:15.47 | seebs | exten => _[1-9#*].,1,NoOp(Line 2 outbound to ${EXTEN}) |
| 05:15.53 | seebs | Should that be 0-9#*? |
| 05:16.03 | [TK]D-Fender | seebs, but it never hurts to do a nice explicit hangup. if the dial FIALS, then it might fall through |
| 05:16.46 | [TK]D-Fender | seebs, do you WANT a number to be able to start with 0? |
| 05:17.12 | seebs | Yes. The goal is to take ANYTHING that I dial, and simply record that I dialed it, monitor the call, and pass it on to the telco. |
| 05:17.39 | seebs | Think of this as elephant-gun-for-a-mosquito style overkill for replacing one of those radio shack phone line recorders. :) |
| 05:20.18 | seebs | Anyway, I have to sleep. Thanks much for all the help! |
| 05:25.05 | HaMYaI | I use the Dial(Sip/..) and during the call, the recipient press any digit on the key pad |
| 05:25.17 | HaMYaI | the call will just hangup |
| 05:26.17 | HaMYaI | how do I know what type of sip message being sent to asterisk? |
| 05:26.56 | Putzz | well you could start by gives us your dial command |
| 05:27.05 | Putzz | *giving |
| 05:28.31 | HaMYaI | $dialstr="SIP/2112|60|L(2700000:60000:30000)m"; |
| 05:28.32 | HaMYaI | $tgi->exec("Dial $dialstr"); |
| 05:28.56 | HaMYaI | Putzz: I am using agi |
| 05:30.22 | *** part/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 05:30.39 | HaMYaI | Putzz: though the system hangs up the call. I really want to know if I can simulate the same message as it seems to produce correct disconect supervision |
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| 06:07.02 | hyphenex | Just wondering if anybody knows a thing about VPN's. but just a quick question. but do I need NAT-T if my server is behind my routers DMZ? |
| 06:13.25 | Yourname`` | ../usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file or directory -> Why does this happen during installation exactly?!!! |
| 06:13.29 | Yourname`` | Gosh! |
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| 06:20.43 | dlynes_laptop | Yourname``: because 'make' failed? |
| 06:21.15 | dlynes_laptop | Yourname``: if make failed, you shouldn't be trying to run 'make install' |
| 06:27.13 | Yourname`` | Yeah, I know.. but make is never intuitive. |
| 06:27.20 | Yourname`` | And never tells what can be done to fix it, lol |
| 06:28.10 | Yourname`` | dlynes_laptop: This is the error --> http://pastebin.ca/710083 |
| 06:37.17 | dlynes_laptop | Yourname``: like i said...stop running make install |
| 06:37.35 | dlynes_laptop | Yourname``: run make....you had an error in that...'make install' WILL NOT RUN UNTIL MAKE COMPLETES SUCCESSFULLY |
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| 06:44.50 | sergee | what is the status of SS7 (SIGTRAN) in asterisk? |
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| 06:52.41 | |Vulture| | Anyone here try using the MYSQL cmd with asterisk-addons-1.4.2? I am trying to get a stored procedure to work, it calls it but then the ${fetchid} stays null |
| 06:53.26 | awk | can anyone advise with call parking and not being able to get hints? |
| 06:53.32 | awk | hence, inverted parking.. |
| 06:53.57 | awk | so you can't tell you phone to monitor a parkcade :) haha, you can select the parked call |
| 06:54.05 | awk | but no light is on to indicate there is a call parking |
| 06:54.10 | |Vulture| | http://www.pastebin.ca/710105 |
| 06:54.15 | awk | and I believe it all to do with BLF |
| 06:54.32 | |Vulture| | thats the script I am using |
| 06:55.03 | |Vulture| | lookupagent() returns 2 field phone and loan_number |
| 06:56.45 | awk | http://www.pastebin.ca/710106 |
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| 06:58.07 | awk | thats a 1.4 problem |
| 06:58.36 | awk | the other problem now is with 1.2.24 where the lights are inverted I know this is a bug with digium that they say they wont fix. |
| 06:58.52 | awk | what happens there is that the lights are allays on, even when a call is not parked |
| 06:59.21 | awk | thing is I never upgraded this client I kept him on 1.2.22 due to the bug I know, some how its happend on 1.2.22 aswell? any sugestions |
| 07:02.20 | *** part/#asterisk swombat (n=KDan@87-194-122-30.bethere.co.uk) |
| 07:03.47 | Aurs | have anyone here tried thomson st2030 with asterisk? |
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| 07:09.36 | awk | WOW! asterisk 1.4 is soooo fucking bugy |
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| 07:11.53 | Yourname`` | How come, awk? |
| 07:12.03 | Yourname`` | Oh, nvmd. |
| 07:14.51 | awk | everything about it, to many iax2 connections I just gets dumps of cli errors, |
| 07:14.53 | awk | ocket_read: Out of idle IAX2 threads for I/O, pausing |
| 07:15.14 | awk | try do anything with 1.4 and it just breaks |
| 07:15.34 | awk | its fine, i' |
| 07:15.44 | awk | ll just disable all my iax2 connections, who needs iax? |
| 07:15.59 | awk | who needs call parking? i'll just disable that, hints are totally fucked with 1.4 |
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| 07:21.02 | arekm | anyone with zhone z-plex 10 crap? I have problems with transfer hanging the phone. A calls to B, B transfers to C, before C answers A disconnects, C then answers and disconnects few seconds later. Now B has phone hangt - hears silence, cannot dial anywhere (no dialtone anyway) |
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| 07:37.41 | dlynes_laptop | awk: so I'm not the only one having issues with hints on 1.4? |
| 07:38.31 | dlynes_laptop | awk: are you having issues with watchers failing to show up in show hints, hints showing idle, when they should be showing busy, ...? |
| 07:38.41 | awk | dlynes_laptop heh, its messed !I have tried every trick in the book to get it working.. changed friends to peers, call-limit, you name it, ive tried it |
| 07:38.56 | awk | dlynes on some hints yes |
| 07:39.00 | dlynes_laptop | awk: yeah...it seems to be buggy as all hell |
| 07:39.12 | awk | on others I have the watchers, but no status what so ever |
| 07:39.15 | dlynes_laptop | awk: and my zaptel hints didn't show with having any watchers for about 8 hours after I had set thme up |
| 07:39.28 | awk | so how can I have a phone monitor an exten if it cant see if its active, etc? |
| 07:39.35 | dlynes_laptop | awk: and then all of a sudden, out of the blue, zaptel hints started working |
| 07:39.37 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
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| 07:39.44 | dlynes_laptop | good morning, russellb |
| 07:40.00 | awk | well thats the kind of issues ive had with 1.2 now, all of a sudden the new bugs in 1.2.24 have found their way in 1.2.22 |
| 07:40.07 | dlynes_laptop | awk: that's the same issue we're running into |
| 07:40.09 | awk | so parking lights are inverted |
| 07:40.25 | dlynes_laptop | awk: it wasn't an issue with 1.2 that i've found so far |
| 07:40.41 | awk | iesh! its really an issue., and digium say they not fixing parking in 1.2.24 they say it works in 1.4 |
| 07:40.42 | dlynes_laptop | awk: only with 1.4 |
| 07:40.53 | awk | thing is it was working untill they screwed up 1.2.24 as it worked in 1.2.23 |
| 07:41.18 | awk | so i would feel they should fix the mess they made, because i can sure as hell tell you not there is no way im putting these clients on 1.4 as its as buggy as fuck |
| 07:41.53 | dlynes_laptop | awk: it's working in 1.2.18, if that's any help |
| 07:42.06 | awk | BLF is just a balls up |
| 07:42.36 | dlynes_laptop | awk: anyways...1.2.18 isn't much help to me, because this particular customer i'm running into this problem with, needs 1.4 so that they can monitor parked calls |
| 07:42.49 | awk | 1.2.18? hah, that has so many sip vulnerablities it would be a joke to install it :) |
| 07:43.42 | dlynes_laptop | awk: SLA is screwed up, too...you can barge into the calls from another phone that never originated an existing call |
| 07:44.31 | awk | ;_; |
| 07:44.42 | dlynes_laptop | awk: from what I understand, my boss will be getting me to open a ticket with Digium paid support tomorrow on the BLF issue |
| 07:44.54 | awk | under 1.4? |
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| 07:44.56 | dlynes_laptop | awk: he may or may not be pursuing the SLA issue...I think he's undecided on that |
| 07:45.01 | dlynes_laptop | awk: correct |
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| 07:45.22 | awk | well thats nice :) hopefully in a few weeks we will have it resolved! |
| 07:45.28 | awk | hah :) i needed this working last week |
| 07:45.48 | dlynes_laptop | awk: well, if he goes ahead with it, I don't think he'll want to wait a few weeks |
| 07:45.59 | russellb | dlynes_laptop: good morning |
| 07:46.10 | awk | i wish I had time to audit and patch my own stuff |
| 07:46.16 | awk | hopefully from next year I have more time |
| 07:46.53 | dlynes_laptop | awk: well, unfortunately, i don't have the time to learn the hint code, or how to read sip debug logs to understand how all this stuff works, or I'd fix it myself |
| 07:47.19 | russellb | we accept bug reports :) |
| 07:47.40 | awk | i have to fix tng's billing system now, it is so way off its scary |
| 07:47.42 | dlynes_laptop | russellb: already submitted one to bugs.digium.com and the asterisk-users mailing list |
| 07:47.58 | russellb | dlynes_laptop: what is the bug id |
| 07:48.12 | awk | russellb why is it that you guys messed up 1.2.24 with the inverted lights for parking.. then wont fix it? |
| 07:48.22 | dlynes_laptop | russellb: I'll be submitting one to the asterisk forum tonight, too...assuming I ever get my password mailed back to me (told it I forgot my password over two hours ago) |
| 07:48.23 | russellb | awk: i have absolutely no idea what you are talking about |
| 07:48.24 | awk | knowing the code the way you goes do it what is so hard to set a 1 to a 0? |
| 07:49.02 | dlynes_laptop | russellb: 10818 |
| 07:49.08 | russellb | awk: and i can assure you that it is far more complicated than "set a 1 to a 0" |
| 07:49.11 | awk | russellb call parking, the lights are allways on, when a call is not on park. then when a call goes on park the light/lights go off |
| 07:49.13 | *** join/#asterisk ussrback (n=MAX@80.241.177.19) |
| 07:49.34 | awk | should be the other way arond, no lights on, call goes on park a light comes on |
| 07:49.43 | *** part/#asterisk ussrback (n=MAX@80.241.177.19) |
| 07:49.52 | dlynes_laptop | awk: fwiw, 1.4.12 when it comes out, it'll allow you to have a parked call return to someone other than the parker, too |
| 07:50.03 | arekm | Grandstream BT101 - is this phone working fine with * (/me needs something reliable) |
| 07:50.13 | dlynes_laptop | arekm: yes, it is |
| 07:50.24 | dlynes_laptop | arekm: it's a piece of crap, but it does work well with asterisk |
| 07:50.35 | awk | dylnes_laptop yes I have allot of excitment for 1.4 |
| 07:50.38 | arekm | dlynes_laptop: why piece of crap then? |
| 07:50.45 | dlynes_laptop | arekm: really poor sound quality |
| 07:50.54 | dlynes_laptop | arekm: not to mention, super ugly |
| 07:50.55 | awk | im going to start using it for our personal box soon so I can get these bugs posted and help get it stable asap |
| 07:51.31 | russellb | dlynes_laptop: your report is very vague, but initially sounds like a phone issue. I'd have to detail what additional information we need ... but it's the middle of the night for me right now |
| 07:51.48 | dlynes_laptop | russellb: yeah, I know...you're on EDT |
| 07:51.48 | arekm | dlynes_laptop: is there any other phone but with good sound quality and comparable price? |
| 07:52.09 | dlynes_laptop | arekm: when you're looking at that price point, you're not going to get great quality |
| 07:52.35 | awk | well i worked out our last issue with rfc compliance issue with power and cisco phones, cisco stated it was impossible for them to work on these POE switches from HP, but with a bit of a cable swop here and there it works, and cisco couldnt believe it.. so something good from today I guess.. |
| 07:52.38 | russellb | awk: yes, please report any problems you have ... i consider 1.4 very stable and usable |
| 07:53.03 | dlynes_laptop | arekm: the linksys phones are slightly higher price, from what I understand better voice quality (but I've never used it, so I can't comment personally), and also from what I understand a few problems with the software (also can't comment personally) |
| 07:53.15 | arekm | dlynes_laptop: don't need features. just simple phones that could replace analog phones |
| 07:54.01 | arekm | currently using analog panasonics with zhone zplex channel bank and zhone is huge piece of crap |
| 07:54.30 | dlynes_laptop | arekm: also, if you don't care about functionality, or looks, the Azatel/GNet phones are fairly good voice quality, have multiple line appearances, and support most common codecs |
| 07:54.52 | dlynes_laptop | arekm: and they're a bit more expensive than the Grandstreams, and still cheaper than the Linksys phones |
| 07:55.34 | dlynes_laptop | arekm: The biggest problem with them, is that the company that makes them is no longer releasing any new firmwares |
| 07:56.11 | dlynes_laptop | russellb: a hint showing the SIP node as idle, even when it's busy is a phone issue, not an asterisk issue? |
| 07:56.27 | dlynes_laptop | russellb: asterisk monitors whether the phone is busy from the phone, not from its own code? |
| 07:56.52 | *** join/#asterisk hyphenex (n=scott@60-241-154-60.tpgi.com.au) |
| 07:57.46 | dlynes_laptop | russellb: I can see you thinking the watchers not being listed or not showing up for eight hours as being a phone issue, but the incorrect status of the channel I would think is something asterisk would get the status of by viewing its own internal state engine |
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| 07:59.00 | russellb | dlynes_laptop: like i said, your report is vague ... some of it sounds like the phone not subscribing at all, some of it sounds like it is incorrect state information. I can't really tell without some more informationl. |
| 08:00.50 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 08:01.43 | nettie | hi guys, on a production box which worked flawless for months I'm getting this error message: |
| 08:01.44 | nettie | Sep 25 09:44:08 WARNING[5355]: chan_zap.c:9393 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner. |
| 08:01.55 | nettie | when a someone calls |
| 08:02.29 | nettie | I'm running asterisk 1.2.17 bristuffed with 0.3.0-PRE-1y-e |
| 08:02.55 | nettie | using a Junghannes dualBRI |
| 08:03.07 | *** join/#asterisk billybongo (n=rich@82.153.23.79) |
| 08:03.16 | russellb | nettie: you are using an ancient version ... |
| 08:03.23 | russellb | later versions have fixes for that type of issue |
| 08:03.27 | nettie | ahhh |
| 08:03.55 | billybongo | slightly OT, but what routers do people recommend for an asterisk/openser cluster ? |
| 08:03.57 | nettie | great then, you know.. when something works like a chamop you dont really think to update it :) |
| 08:04.02 | nettie | thanx russellb though |
| 08:04.14 | russellb | nettie: yep, understandable |
| 08:04.36 | russellb | just letting you know that i know there are related fixes in later versions, so i would consider upgrading |
| 08:04.46 | russellb | (and furthermore, upgrading to a supported release series, 1.4) |
| 08:05.20 | nettie | that's great then I'll restart and then manage to update it. |
| 08:05.22 | nettie | uhmm |
| 08:05.35 | nettie | 1.4 works fine junghannes duoBRI ? |
| 08:05.45 | russellb | don't ask me. |
| 08:05.47 | nettie | ehehe |
| 08:05.53 | nettie | that's the point |
| 08:05.56 | russellb | heh |
| 08:06.41 | nettie | I went 1.2 because it was supported .. have to check now.. thanx a lot anyway for the hints.. |
| 08:07.45 | russellb | np |
| 08:08.16 | awk | russellb: ok. can you explain this, in 1.4 I have no hints for parked calls, I can only see watches, but it never says if a call is active/idle, etc, just unavilable, 703@default-local : Local/703@parkedcall State:Unavailable Watchers 1 ; extensions.conf exten => 703,hint,Local/703@parkedcalls ; include to exteions-apps.conf exten => 703,1,ParkedCall(703) |
| 08:08.25 | awk | any advice would be great |
| 08:09.22 | russellb | in 1.4, that is not how you should set up a hint for parked calls |
| 08:09.43 | awk | oh, ok let me read up then |
| 08:09.43 | russellb | it should be something like .... exten => 703,hint,Parking:703 |
| 08:09.50 | russellb | let me check |
| 08:10.39 | russellb | i'm going to spam the channel because i'm tired and lazy, but this is from extensions.conf.sample. ... |
| 08:10.47 | awk | ok |
| 08:11.08 | russellb | ;If you want to subscribe to the status of a parking space, this is |
| 08:11.08 | russellb | ;how you do it. Subscribe to extension 6600 in sip, and you will see |
| 08:11.08 | russellb | ;the status of the first parking lot with this extensions' help |
| 08:11.09 | russellb | ;exten => 6600,hint,park:701@parkedcalls |
| 08:11.41 | awk | I see thanks |
| 08:11.45 | awk | over looked that |
| 08:13.00 | russellb | no problem, doesn't appear to be very well documented |
| 08:14.01 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 08:14.11 | awk | ye and why I trusted it worked that way too is that I have a ScopServ box, and can see they have not coded things correctly to structure hints in the correct way for parking |
| 08:14.27 | awk | anyway, my self to blame |
| 08:14.30 | awk | let me fix this |
| 08:14.31 | russellb | ScopServ? |
| 08:14.34 | russellb | another gui? |
| 08:14.56 | awk | yes, they massive, the best GUI you could ever have worked with, every single thing to the T is covered in it. |
| 08:14.56 | russellb | ah, sure enough |
| 08:14.59 | awk | www.scopserv.com |
| 08:15.40 | awk | right down to zaptel configuration, mailing through high load, pri down, firwall intergration, billing system |
| 08:16.51 | awk | promiscious mode scanners :) |
| 08:17.31 | awk | vpn, radious, rsa keys! anyway, thanks again |
| 08:17.41 | awk | err radius |
| 08:18.07 | russellb | you're welcome |
| 08:19.19 | awk | well i have you :) you have not had major problems with DTMF? it feels like training is set on, as I have to punch in a digit around 3/4 times before it takes it |
| 08:19.30 | awk | as does traing if set to 300ms then after 3 seconds it kicks in |
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| 08:19.35 | russellb | i hate DTMF |
| 08:19.38 | awk | thats another major reason for not using 1.4 |
| 08:19.39 | awk | :) |
| 08:19.56 | russellb | another developer and I have rewritten all kinds of dtmf support crap in the 1.4 code multiple times ... |
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| 08:19.59 | russellb | it should be working very well |
| 08:20.11 | russellb | but the problems depend on what type of dtmf you're talking about ... |
| 08:20.11 | awk | not in 1.4.11 :) |
| 08:20.23 | awk | I dont have relaxed dtmf on |
| 08:20.25 | awk | I dont have training on |
| 08:20.42 | awk | dtmfmode=rfc2833 |
| 08:20.53 | russellb | ok, with rfc2833 relaxdtmf has no effect |
| 08:21.05 | russellb | that only is related to inband dtmf processing (processing DTMF sent as audio) |
| 08:21.19 | awk | ahh I see |
| 08:21.26 | awk | is that the same in 1.2? |
| 08:21.29 | awk | or just as of 1.4? |
| 08:21.44 | russellb | rfc2833 support in 1.4 is vastly different than 1.2 |
| 08:21.51 | russellb | but the same statement applies, those settings are not relevant |
| 08:22.02 | russellb | i would be surprised if there was an rfc2833 problem in the latest code |
| 08:22.08 | russellb | you can also try the 1.4 branch directly ... |
| 08:22.10 | awk | 4.11? |
| 08:22.21 | russellb | svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 |
| 08:22.24 | awk | yes there is a problem I have to hit it a good couple times before it takes it |
| 08:22.45 | russellb | well, i would start by verifying that the phone is actually sending rfc2833 dtmf |
| 08:22.50 | russellb | which you can see with "rtp debug |
| 08:23.11 | awk | if I stick a mirror image box of 1.2 in |
| 08:23.16 | awk | it works with the phones |
| 08:23.22 | awk | it sends the dtmf after a peroid of time |
| 08:23.24 | awk | then I see 1 1 1 |
| 08:23.45 | russellb | well, if you're having trouble with that in 1.4, i'd be happy to look at it |
| 08:24.01 | awk | ok what do you want from me? |
| 08:24.08 | russellb | basically, set up an extension that does something simple like Read() |
| 08:24.14 | russellb | turn on rtp debug |
| 08:24.21 | russellb | and do a quick test, saving all of the console output |
| 08:24.29 | russellb | then post it to bugs.digium.com |
| 08:24.30 | awk | ok |
| 08:24.38 | russellb | i'll have to look at it tomorrow or something .. |
| 08:24.46 | russellb | but that's what i (or whoever) would need to see |
| 08:24.56 | awk | so why is it that you hate dtmf :) |
| 08:24.57 | awk | haha |
| 08:25.14 | russellb | for a sanity check, you could try to get someone here to help you figure out if the phone is actually sending rfc2833 dtmf |
| 08:25.23 | russellb | but it should be fairly obvious in the rtp debug output |
| 08:25.37 | awk | ye |
| 08:25.41 | russellb | dtmf is just irritating because it seems like it should be so easy |
| 08:25.44 | awk | anyway ill get that bug report sent |
| 08:25.53 | awk | russellb isn't it ? :) |
| 08:25.59 | russellb | but then we have to support 5 different ways of sending and receiving it across different technology types |
| 08:26.05 | awk | haha |
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| 08:26.14 | awk | ye I can imagine that must crap |
| 08:26.22 | russellb | and preserving the length of digits across DTMF types and channel types added another big layer of complexity in 1.4 |
| 08:27.03 | awk | well hopefully its something small with regards to what we experiencing |
| 08:27.14 | awk | take care, im off |
| 08:27.19 | awk | will send that bug report through |
| 08:27.23 | russellb | k. |
| 08:27.33 | russellb | take it easy .. |
| 08:29.17 | yang | I am wondering about these free enum numbers e164.org - Those numbers are binded to USA calling number ? |
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| 08:40.36 | grEvenX | hi guys |
| 08:41.08 | grEvenX | does anyone of you know how to check how many agents in a given queue are on a queue-call ? |
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| 08:55.20 | grEvenX | s/agents/queue member/ |
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| 09:40.11 | Zefk | Hi. How can I restart zaptel? /etc/init.d/zaptel [start|stop|status] command returns the prompt. Thx |
| 09:40.45 | ghatak | Hi, my in comming calls cut off after exactly 20 secs if they are not answered. What controls this? I want to increase this to 45 secs |
| 09:53.02 | arekm | is "loopstart + forward disconnect" == "kewlstart" ? |
| 09:53.39 | ghatak | arekm: was that for me ? |
| 09:54.36 | arekm | ghatak: no, for those who know the answer |
| 09:54.57 | ghatak | arekm: sorry |
| 09:55.48 | arekm | ghatak: Dial() has timeout option btw. No idea if default==20 |
| 09:57.53 | ghatak | arekm: ok thanks, let me check |
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| 10:06.15 | traxx | hi. anybody know how i can show missed calls in asterisk? |
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| 10:34.25 | disposable | does anybody here use voiptalk.org's IAX trunk? |
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| 10:44.10 | joobie | hey guys.. anyone know of a good / cheap voip phone for asterisk? |
| 10:46.10 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 10:46.28 | yang | Is there a special setting that needs to be apÄplied to asterisk configs to be able to receive calls over SIP? I can only do it to local extensions |
| 10:50.34 | mvanbaak | joobie: most of the time cheap voip phones are not the good ones |
| 10:53.32 | joobie | ahh |
| 10:53.36 | joobie | what about a good voip phone |
| 10:54.04 | arekm | so what are cheapest but still good? ;) |
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| 11:03.34 | Nugget | polycom. |
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| 11:08.49 | joobie | Nugget, got a particular model in mind? |
| 11:16.01 | Nugget | ip320 or 330 are good phones. |
| 11:17.13 | mvanbaak | snom phones are not bad either |
| 11:17.29 | mvanbaak | ~phones |
| 11:17.29 | jbot | well, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like ... |
| 11:17.56 | joobie | what does it mean by "full duplex speakerphone" |
| 11:17.57 | joobie | ? |
| 11:18.09 | Nugget | it means you can be talking and listening at the same time. |
| 11:18.41 | mvanbaak | means you have a speakerphone with 2way audio |
| 11:18.57 | mvanbaak | the cheaper grandsuck phones have a oneway speakerphone |
| 11:19.16 | mvanbaak | means you can only hear the other side through the speaker, you have to talk in the handset |
| 11:19.23 | joobie | as in it can only be doing the mic or speaker at any given time, not both? |
| 11:19.27 | joobie | ahhh |
| 11:19.34 | Nugget | It's hard to make a speakerphone that doesn't echo (because your microphone picks up the sound coming from the speaker) |
| 11:19.44 | mvanbaak | uhhuh |
| 11:19.49 | Nugget | so the cheap phones dodge the issue by turning off your mic when the speaker is making noise |
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| 11:19.55 | joobie | i see |
| 11:20.00 | mvanbaak | I know the cisco 7960 has a very good speakerphone |
| 11:20.29 | joobie | is there a cheap store in AU to buy these phones? |
| 11:20.32 | joobie | the polycom's? |
| 11:20.51 | joobie | like something well known in the channel? |
| 11:22.05 | Nugget | apnp 'oow umop apisdn ue w,i |
| 11:22.11 | Nugget | ^ australian moo. |
| 11:22.17 | joobie | heh |
| 11:22.23 | MicW | i'd like to run asterisk on a vmware server. the problem is that i cannot access the isdn card (pci) from there. is it possible to use external isdn devices accessed by tcp/ip or can i run a minimal asterisk on a differen (non-virtual) server that handles the isdn calls (inbound and outbound) and gives the virtualized asterisk access to them? |
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| 11:35.22 | joobie | guys |
| 11:35.30 | joobie | can someone give me a hand |
| 11:35.36 | joobie | help explain this whole sip thing |
| 11:36.12 | joobie | i wana use voip over my cable internet connection.. but have it so i have a telephoen number people dial.. it comes to my asterisk box.. then rings my voip phone |
| 11:37.02 | joobie | not sure what componetns i need |
| 11:37.09 | joobie | i got the voip phone now.. the polycom 330 |
| 11:39.46 | JT | polycom make much better speakerphones than ciscos |
| 11:39.57 | JT | joobie: cheaper to buy from the US than AU |
| 11:40.05 | joobie | ahh serious |
| 11:40.14 | joobie | you from AU JT? |
| 11:40.20 | JT | yes |
| 11:40.24 | joobie | cool |
| 11:40.36 | joobie | where is a cheap place int he US to buy it form? |
| 11:40.39 | joobie | im not in any rush.. |
| 11:40.46 | joobie | ps. where in AU r u from? |
| 11:40.53 | JT | telephonydepot.com |
| 11:41.01 | JT | but shipping could be better |
| 11:41.03 | JT | sydney |
| 11:41.13 | JT | if you must buy locally, try Westan |
| 11:41.25 | joobie | ta |
| 11:42.17 | joobie | thanks JT |
| 11:42.19 | joobie | im frmo VIC |
| 11:42.26 | joobie | will buy on the net tho |
| 11:42.34 | joobie | no rush, can wait out the US delivery time |
| 11:42.40 | joobie | JT.. what else do i need tho man? |
| 11:42.56 | joobie | I have cable in my home.. i wanted to get rid of my fixed landline and use voip over the cable.. |
| 11:43.28 | joobie | basically want to have a telephone number people can dial on standard telephones, have it come through my asterisk box and visa versa.. so i can dial out voip or dial out to standard pots phones too.. |
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| 11:49.32 | JT | joobie: sure, that will work |
| 11:51.24 | joobie | JT, do you know of any cheap providers i can tie into to give me that telephone number? |
| 11:51.42 | joobie | it's really to make local / mobile telephone calls.. not really any interstate / overseas calls |
| 11:55.49 | JT | there's plenty |
| 11:56.49 | joobie | JT, know of a cheap and reliable one? |
| 11:56.53 | joobie | i dont know any :/ new to this |
| 11:57.11 | joobie | after one that is not going to go away tomorrow.. something cheap and reliable |
| 11:57.18 | joobie | reliable being the bigger factor than cheap |
| 11:58.38 | JT | cheap cheap cheap, i get it... |
| 11:58.55 | Teln1100A | Try Teliax |
| 11:59.24 | JT | Teln1100A: useless suggestion |
| 11:59.45 | JT | joobie: maybe pennytel |
| 12:01.02 | joobie | ive heard of pennytel before |
| 12:01.09 | *** join/#asterisk Martin79 (n=martin@cpc4-leed12-0-0-cust360.leed.cable.ntl.com) |
| 12:01.19 | joobie | the name rings a bell |
| 12:01.46 | joobie | JT.. so do they give you a telephoe number people can dial.. and when they dial, it converts the voice call into a SIP singal sent to your asterisk box? |
| 12:02.00 | Martin79 | hi guys, i'm having some trouble with my b410p if anyone can help please? |
| 12:02.04 | JT | right |
| 12:02.16 | JT | you get a DID if you pay for one |
| 12:02.23 | joobie | JT.. btu they are US based |
| 12:02.32 | JT | nope |
| 12:02.36 | joobie | does this mean the data needs to come from the US? bit of latency delay aint it? |
| 12:02.45 | joobie | oh |
| 12:02.52 | joobie | pennytel.com right? |
| 12:02.56 | JT | their sbc/sip proxy is in Global Switch or Equinix Sydney |
| 12:02.59 | JT | right |
| 12:03.11 | JT | not sure where on earth you got evidence they were US base |
| 12:03.41 | joobie | ahh i just based it on their site being a .com |
| 12:04.10 | JT | yes and there have never been australian businesses using .coms before... |
| 12:04.27 | joobie | heh point taken;P |
| 12:04.43 | joobie | so JT.. is there a voip card u recommend? |
| 12:04.51 | JT | voip card? |
| 12:04.53 | joobie | i see the polycom 330 has a PoE feature.. |
| 12:05.06 | joobie | yea, something i can hook up my polycom phone to.. and tie into this DID feature of pennytel |
| 12:05.31 | JT | ethernet... |
| 12:06.12 | Martin79 | anyone able to help me out with some chan_misdn issues please? |
| 12:07.07 | joobie | JT.. know of a decent model? |
| 12:08.01 | JT | joobie: of what? |
| 12:08.57 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:09.46 | joobie | JT, voip card.. |
| 12:09.59 | joobie | JT, I will get that polycom 330 phone.. need a card that support PoE to hook it up to |
| 12:10.02 | joobie | something of decent quality |
| 12:10.12 | JT | dude |
| 12:10.13 | Martin79 | get a poe injector? |
| 12:10.16 | JT | it's ethernet |
| 12:10.22 | JT | no "voip card" |
| 12:10.29 | joobie | oh |
| 12:11.11 | Martin79 | if you only have one phone you want an injector, if it doesn't have a seperate power supply for the phone, if you have multiple phones look at a poe switch |
| 12:11.22 | JT | joobie: "voice over INTERNET PROTOCOL" |
| 12:11.47 | joobie | arent there hardware cards that do extra things like echo cancellation, etc |
| 12:12.02 | JT | for pstn lines, yes |
| 12:12.05 | JT | not for voip |
| 12:12.08 | joobie | ahhh |
| 12:12.13 | joobie | i think im getting confused with them |
| 12:12.19 | joobie | i saw all these cards taht interface with PRI/BRI/POST |
| 12:12.21 | joobie | err POTS |
| 12:12.32 | joobie | with all these neat features.. was thinking i needed a card for the LAN too |
| 12:12.59 | joobie | what's the dealwith the POE injectors? |
| 12:13.03 | Martin79 | just need a phone or sip client ;) |
| 12:13.12 | joobie | Is there a ethernet card that has poe native? or must use an injector?? |
| 12:13.26 | joobie | nice:P |
| 12:13.34 | Martin79 | you don't plug your phone into an ethernet card |
| 12:14.23 | joobie | oh? |
| 12:14.32 | joobie | you plug it into a PoE enabled switch instead? |
| 12:14.39 | Martin79 | you plug into a switch/hub/router to connect your phone to the internet |
| 12:14.55 | joobie | ahh |
| 12:15.03 | joobie | but i want to connect it to an asterisk server |
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| 12:15.15 | joobie | so i can do some neat stuff like voicemail, etc through the asterisk server |
| 12:15.23 | joobie | so will need to feed the ethernet back into the server |
| 12:15.35 | [TK]D-Fender | joobie: it will connect to *.... through your LAN <------ |
| 12:15.45 | joobie | i was thinking if this was the case, given it's just the one phone, i dont need to buy a switch/hub.. can just plug it direct to a NIC on the asterisk server |
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| 12:15.51 | [TK]D-Fender | joobie: It may as well be a PC shaped like a PHONE |
| 12:15.55 | joobie | ahhh |
| 12:15.56 | joobie | I see |
| 12:16.11 | [TK]D-Fender | joobie: yeah you can, but why add more NIC's to manage to your system? |
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| 12:16.37 | joobie | im just thinking it'll be a cheaper option to achieve PoE for 1 phone.. rather than buying a switch with numerous ports supporting PoE that i won't use |
| 12:16.45 | joobie | i just need PoE for the one phone.. maybe 2 tops |
| 12:16.51 | [TK]D-Fender | joobie: the wiring advantage to VoIP hard-phones is you can put them ANYWHERE. Why physically TIE them to your box? You're undoing their advantage |
| 12:16.52 | Martin79 | buy a cheap switch and use the psu that comes with your phone, no need to over complicate everything? |
| 12:17.05 | joobie | I see |
| 12:17.18 | [TK]D-Fender | joobie: a single-port injector IS the cheaper way and I have NEVER heard of a PC NIC that can do PoE |
| 12:17.46 | joobie | [TK]D-Fender, can you recommend a decent / cheap single-port injector? |
| 12:17.54 | Martin79 | google? |
| 12:17.58 | joobie | Martin79, i'm trying to minimize the wires |
| 12:18.02 | [TK]D-Fender | joobie: For Polycom320/330 you can get either an injector, or buy a standard-style poewr brink |
| 12:18.23 | [TK]D-Fender | joobie: Only way to do that is using a PoE switch. Everything else ADDS wires |
| 12:19.02 | joobie | ahh |
| 12:19.07 | joobie | even an injector? |
| 12:19.11 | *** join/#asterisk rapid (n=rapid@unaffiliated/rapid) |
| 12:19.15 | [TK]D-Fender | joobie: Of course |
| 12:19.20 | Martin79 | injector would be tidier on the desk side, just more wires at the switch |
| 12:19.32 | rapid | Hi |
| 12:19.50 | Martin79 | hi |
| 12:20.03 | [TK]D-Fender | joobie: picture this Ethernet +Power --> injector ---> Phone . You're adding a "nugget" in-line with your phone that ITSELF needs a wall-wart |
| 12:20.14 | joobie | Martin79, sounds like a goer |
| 12:20.23 | joobie | can you recommend an injector? come across heaps in google |
| 12:20.46 | Martin79 | nope, never used one, made a custom one for my ap on the roof :P |
| 12:20.47 | [TK]D-Fender | Martin79: Yes if the switch is far enough away it'll make the DESK look clean, but the overall solution won't have less wiring |
| 12:21.04 | joobie | yea |
| 12:21.23 | [TK]D-Fender | joobie: Do you have a switch at the desk you want to place your phone at? |
| 12:21.55 | joobie | [TK]D-Fender, not at the desk.. but the ethernet will run to a switch |
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| 12:22.08 | joobie | Martin79, nice |
| 12:22.35 | [TK]D-Fender | joobie: Ok, will you be plugging the phone into its own jack, and the PC to its own as well? Or are you intending on putting the phone BETWEEN your PC & switch? |
| 12:23.44 | joobie | would prefer to put the phone between the pc and switch.. though duno if the switch in the polycom phone is smart enough to drop the power on the siwtched port going to the computer |
| 12:24.19 | [TK]D-Fender | joobie: Yes it is.... thats common sense |
| 12:24.26 | Martin79 | i wouldn't expect they're allowed to sell somthing that's gonna fry your nic ;) |
| 12:24.42 | joobie | hehe |
| 12:24.46 | [TK]D-Fender | joobie: How many phones are you looking at buying? |
| 12:24.51 | joobie | 2 |
| 12:25.07 | [TK]D-Fender | joobie: Any real expectations for expansion? |
| 12:25.33 | joobie | not in the short-term.. but would like to be aware of any limitations to expansion |
| 12:25.51 | joobie | i was thinking perhaps to have the ability to receive numerous calls on the one DID |
| 12:26.18 | joobie | not sure if that is possible.. ie. say you dial the DID number.. it goes through to phone #1.. if someone else dials, though it's in use.. it goes through to phone #2.. if someone else dials again, it goes through to voicemail... |
| 12:26.25 | [TK]D-Fender | joobie: Wrong topic |
| 12:26.30 | joobie | etc.. so numerous people can dial the one DID .. not sure if it's possible |
| 12:26.32 | [TK]D-Fender | joobie: We're not even talking about * yet |
| 12:26.32 | joobie | ahh k |
| 12:26.40 | joobie | ahh |
| 12:26.57 | [TK]D-Fender | joobie: You know what... I'm thinking you may as well get the standard power brick for your phone and be done with it. |
| 12:27.11 | joobie | yer |
| 12:27.12 | joobie | me too |
| 12:27.13 | joobie | :P |
| 12:27.17 | joobie | maybe a good start |
| 12:27.21 | joobie | and look into PoE down the track |
| 12:27.27 | [TK]D-Fender | joobie: after #2 |
| 12:27.41 | joobie | http://www.powersystemsdirect.com/Aastra_Telecom/Venture_IP_480i_Power_Injector_PoE_1540.php |
| 12:27.46 | joobie | check that out |
| 12:27.46 | arekm | was there any bug with queues where * would call to the members all the time and member would hear nothing after answering + couldn't hang up? |
| 12:27.49 | joobie | only $40 |
| 12:28.05 | yang | I am trying to work with billing, but I cannot dial out - http://paste.debian.net/37949 |
| 12:28.11 | [TK]D-Fender | joobie: the normal power brick is just under $20 <0 |
| 12:28.37 | [TK]D-Fender | yang: a2billing is NOT supported here |
| 12:28.48 | yang | [TK]D-Fender: where is their support channel? |
| 12:29.16 | [TK]D-Fender | yang: I'm not sure they HAVE one. Check out their home page to see what resources they offer. |
| 12:29.24 | yang | only forum |
| 12:30.06 | [TK]D-Fender | yang: Then there's your answer |
| 12:30.17 | Martin79 | I'm have some help with * and chan_misdn not compiling when installing my b410p, if anyone can help? |
| 12:30.35 | creativx | damn |
| 12:30.47 | joobie | ill get the power brick i think |
| 12:30.51 | joobie | move to PoE downt he track |
| 12:30.54 | [TK]D-Fender | Martin79 : Call up Digium support, its free. |
| 12:31.01 | creativx | is there any way to override the most annoying feature of the polycom phone... that it dont want to remember what volume setting you have on the headset |
| 12:31.09 | [TK]D-Fender | joobie: Injectors are a waste if you're thinking of expanding really. |
| 12:31.15 | joobie | [TK]D-Fender, on the other topic .. can taht be done, by getting one DID and have it route numerous calls to your phone? |
| 12:31.19 | [TK]D-Fender | creativx: Yes, in sip.cfg |
| 12:31.33 | joobie | [TK]D-Fender, ya.. i can see how it would become costly / messy with injectors (numerous ones) |
| 12:31.40 | creativx | [TK]D-Fender: thank god.. heh |
| 12:31.57 | [TK]D-Fender | joobie: Yes, but you need to clarify how many calls your PROVIDER is going to allow. |
| 12:32.15 | joobie | ahh |
| 12:32.29 | [TK]D-Fender | creativx: You can set each of the 3 volume settings as "sticky" in there independantly |
| 12:32.33 | joobie | [TK]D-Fender, what about pennytel? I saw their home plan.. thought it was good enough |
| 12:32.39 | joobie | but it doesnt meniotn how many calls the provider will allow |
| 12:32.52 | creativx | [TK]D-Fender: yeah i see.. im googling how to right now |
| 12:32.53 | [TK]D-Fender | joobie: Don't know them.... here's something to consider : |
| 12:32.54 | [TK]D-Fender | ~itsp |
| 12:32.55 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower |
| 12:33.15 | joobie | ahah |
| 12:33.49 | joobie | what's the difference between Teliax and Pennytel? |
| 12:34.25 | [TK]D-Fender | joobie: call quality, customer service, flexibility, probably cost. go LOOK |
| 12:35.22 | joobie | Cheers, lookin |
| 12:37.19 | JT | the fact that teliax is useless for joobie |
| 12:38.02 | joobie | ya |
| 12:38.07 | joobie | just noticed.. they are only for US |
| 12:38.12 | JT | joobie: umm, what do you mean how many calls it would allow? |
| 12:38.41 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:38.45 | *** part/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 12:39.04 | joobie | JT.. like if i get 2 polycom phones and one DID number (say the personal phone plan with pennytel).. just wondering if they give me 9849 3847 for example.. someone dials it so it's busy.. one pohone it taken up.. but wondering if someone else dials, if i can make the 2nd phone ring |
| 12:39.24 | JT | yes |
| 12:39.28 | joobie | ie. keep reusing the same telephone number.. or is it like standard POTS where if the line is busy, it's busy |
| 12:39.44 | joobie | JT, that comes down to my asterisk setup ya? |
| 12:39.52 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 12:40.10 | *** join/#asterisk DarylVOIP (n=daryl@host-24-225-239-34.patmedia.net) |
| 12:40.13 | [TK]D-Fender | joobie: that depends on your PROVIDER |
| 12:40.18 | JT | yes they can easily send multiple calls over sip |
| 12:40.21 | JT | destined for the same DID |
| 12:40.27 | joobie | I see |
| 12:40.35 | [TK]D-Fender | joobie: As its delivered over SIP (a VoIP protocol), you're limited by your provider, and your BANDWIDTH |
| 12:41.15 | joobie | JT, will the personal phone plan suffice with pennytel? I can't see any mention of linking their service to asterisk -- very hesitant to subscribe |
| 12:41.23 | joobie | i see [TK]D-Fender |
| 12:41.31 | JT | yes it will work fine |
| 12:41.34 | JT | it's sip |
| 12:41.43 | JT | the fact you use asterisk is irrelevant |
| 12:41.49 | joobie | I see |
| 12:41.54 | joobie | one last question JT:P |
| 12:42.46 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:42.54 | joobie | with these DID numbers.. i want a number that i can move with me.. ie if i move houses it comes with me.. etc.. something permanent.. im thinking can i actually buy the number myself so i own it for life? or is it safe to presume if i buy it through pennytel paying them the $5.00/month they want, that it will be with me for life? |
| 12:43.15 | lirakis | joobie: you are not buying the DID ... |
| 12:43.22 | lirakis | joobie: you are getting them to point it to you |
| 12:43.50 | lirakis | joobie: buying DID's is a highly regulated (lots of paperwork) and complex deal... you have to track them for life etc. |
| 12:43.58 | joobie | ahh |
| 12:44.07 | joobie | but what is stopping these guys reallocating the DID |
| 12:44.18 | lirakis | joobie: they wont as long as you keep paying them |
| 12:44.19 | JT | customer anger |
| 12:44.23 | JT | why would they do that |
| 12:44.25 | joobie | i mean i'm planning on subscribing to the 'personal plan' on pennytel.. but use it for business purposes |
| 12:44.37 | joobie | im thinking twice about this in case they play with the number.. and it affects my business |
| 12:44.37 | JT | as long as they exist, and you have an account, you continue using the did |
| 12:44.46 | joobie | i see |
| 12:44.47 | lirakis | joobie: what JT said |
| 12:44.48 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:44.50 | JT | you cannot port the number though |
| 12:45.06 | JT | joobie: if you want real reliability, get a PSTN connection |
| 12:45.09 | joobie | so if pennytel go under.. my number goes with them? |
| 12:45.21 | JT | yes |
| 12:45.25 | joobie | i see |
| 12:46.37 | joobie | well |
| 12:46.58 | joobie | thanks heaps JT, lirakis, Martin n [TK]D-Fender |
| 12:47.00 | creativx | how cheap ass can you be if you are running a business |
| 12:47.01 | creativx | :) |
| 12:47.11 | joobie | have filled my brain with voip / asterisk knowledge |
| 12:47.19 | joobie | appreciate it |
| 12:47.45 | *** join/#asterisk ManxPower (n=manxpowe@ip68-107-250-148.no.no.cox.net) |
| 12:48.43 | lirakis | creativx: .. you have no idea.. lol .. ive seen some really really really cheap bastards customers at the old company i worked at |
| 12:48.46 | [TK]D-Fender | creativx: Like Einstein said, only 2 things are infinite : The univers, and human stupidity. He's not so certain on the former though.... |
| 12:49.13 | creativx | lol indeed |
| 12:49.30 | creativx | i'd rather focus my energy on making enough profit to atleast afford ... phones |
| 12:49.58 | *** join/#asterisk etfonhomey (n=chatzill@12.169.248.226) |
| 12:51.05 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 12:51.18 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 12:52.30 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:53.52 | [TK]D-Fender | creativx: That sounds like a good plan... |
| 12:54.12 | [TK]D-Fender | creativx: Though honestly I could be happy with an ATA & decent cordless phone. |
| 12:56.27 | creativx | hehe |
| 12:58.02 | creativx | you know what would make me happy |
| 12:58.17 | creativx | if i could find a way to limit the sip softphones to 1 call |
| 12:58.18 | coppice | someone special to call? |
| 12:58.26 | creativx | without having to patch asterisk |
| 12:58.31 | creativx | or upgrading to 1.4 |
| 12:58.39 | creativx | heheh coppice.. naa |
| 12:59.21 | Martin79 | a large bat for instances then somone places more than 1 call, they'll only do it once ;) |
| 12:59.35 | etfonhomey | [TK]D-Fender What kinds of things do you use the idle display for on the Polycom's? |
| 12:59.37 | creativx | the problem is inbound :) |
| 12:59.44 | creativx | people may call out as much as they like |
| 12:59.53 | creativx | but torturing people with more than 1 inbound call at the time is.. torturing them |
| 13:00.01 | [TK]D-Fender | etfonhomey: Live call-queue stats & phone status |
| 13:00.08 | [TK]D-Fender | etfonhomey: And of course corporate logos |
| 13:00.21 | creativx | how do you collect the qstats [TK]D-Fender? |
| 13:00.28 | [TK]D-Fender | creativx: AMI |
| 13:00.49 | [TK]D-Fender | creativx: Check if they're on a call before sending another to them then |
| 13:01.01 | creativx | [TK]D-Fender: cant check that when using queue |
| 13:01.13 | creativx | atleast as far as I know.. |
| 13:01.30 | [TK]D-Fender | creativx: yes, you CAN <- |
| 13:01.34 | creativx | I CAN!!!1 |
| 13:01.48 | creativx | please enlighten me |
| 13:01.58 | [TK]D-Fender | creativx: Reconsider how you're calling them. |
| 13:01.59 | creativx | as i have not thought this properly through, how i would solve that |
| 13:02.01 | *** join/#asterisk razu (n=razu@razu.data.ee) |
| 13:02.51 | creativx | hmm |
| 13:03.05 | creativx | i am thinking |
| 13:03.08 | [TK]D-Fender | creativx: how do you call an agent currently? |
| 13:03.26 | creativx | dynamically add queue members using sip/extnumber |
| 13:03.33 | creativx | using AMI |
| 13:03.45 | creativx | hmm hmm |
| 13:03.58 | creativx | thanks [TK]D-Fender :) i think i might have gotten an idea there |
| 13:03.59 | creativx | hehe |
| 13:04.34 | creativx | sip/extnum would naturally mean extension => extnum being called.. and there i could check if the user is busy |
| 13:05.13 | Martin79 | if you used agents, would that do it automatically? |
| 13:05.41 | [TK]D-Fender | Martin79: define "agents" and the answer will announce itself |
| 13:06.39 | etfonhomey | [TK]D-Fender You just create an XHTML web page and set mb.idleDisplay.home and mb.ideDisplay.refresh to look at it? Then you continously update the page data? |
| 13:06.46 | Martin79 | as in using the agents.conf and having users sign on, i've never used it so i don't know |
| 13:06.58 | creativx | already reading up on it tk.. hehe |
| 13:07.09 | [TK]D-Fender | etfonhomey: yup |
| 13:07.27 | [TK]D-Fender | Martin79: Go get some experience then |
| 13:07.32 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:08.50 | Martin79 | personally i prefer to ask somone with experiance and take 30 sec for a yes/no answer than spend more pratting about *shrugs* |
| 13:09.52 | [TK]D-Fender | Martin79: Asking is fine, but you should take a little time to get your hands dirty a bit... |
| 13:10.05 | creativx | getting pointed in the right direction by asking is sweet |
| 13:10.06 | [TK]D-Fender | Martin79: its not like finding out costs your anything but a little time :) |
| 13:10.22 | Martin79 | time is money ;) |
| 13:10.27 | creativx | like you just did now [TK]D-Fender... though it seems it might mean i have to replan .. alot=) hehe |
| 13:10.28 | [TK]D-Fender | creativx: Here's your freebie then : chan_local <- |
| 13:11.45 | creativx | hehe |
| 13:16.26 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 13:17.37 | *** join/#asterisk ^JimmyRidge^ (i=nobody@EV-ESR-74-215-26-69.fuse.net) |
| 13:19.41 | *** join/#asterisk yannj_fr (n=yannj@chilli.esiee.fr) |
| 13:20.06 | ^JimmyRidge^ | any1 familiar with asterisk and FWD IAX? seems to me the iax2.fwdnet.net's iaw port is closed |
| 13:20.59 | *** join/#asterisk hank (n=hank@netwichtig.de) |
| 13:21.00 | hank | hi |
| 13:21.27 | ^JimmyRidge^ | whois jimmyridge |
| 13:21.33 | ^JimmyRidge^ | whoopz heh |
| 13:21.44 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.37.205) |
| 13:22.22 | hank | i want to try the ajam interface. logging in via manager?action=login works fine, but when opening rawman?action=status i always get 'permission denied'. what could be wrong? |
| 13:22.46 | rob0 | Jim, IWFM AFAICT (I don't really use FWD, but I still have it set up.) |
| 13:22.58 | *** join/#asterisk guillote_GNU (n=bancaria@host35.201-253-17.telecom.net.ar) |
| 13:23.11 | rob0 | I set up FWD a long time ago, and the SIP side was what was broken. |
| 13:23.42 | [TK]D-Fender | hank: what RIGHTS does that account have? |
| 13:23.46 | hank | deny = 0.0.0.0/0.0.0.0 and permit = 192.168.240.117/255.255.255.0 what would i have to set for permit = to allow access from every host? |
| 13:24.07 | hank | [TK]D-Fender: read = system, call, log, verbose, command, agent, user |
| 13:24.10 | hank | [TK]D-Fender: same for write. |
| 13:24.34 | [TK]D-Fender | hank: is that hos correct? |
| 13:24.39 | [TK]D-Fender | hank: Whats on that IP? |
| 13:25.38 | hank | [TK]D-Fender: i guess so. asterisk server is running in a virtualbox so i am not sure about networking. thats why i want to allow access from every host. no, that would not be a security issue since its only for testing and i am in a net where i trust everyone not to tamper with a test installation. |
| 13:25.48 | *** join/#asterisk dmcmorris_ (n=don@adsl-160-0-165.asm.bellsouth.net) |
| 13:27.18 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 13:27.23 | *** join/#asterisk d1mas (n=chatzill@host-99.dataart.net) |
| 13:28.10 | *** join/#asterisk nethium (i=nethium@216.129.53.236) |
| 13:28.24 | d1mas | hello. Question - why asterisk does not create hint when it registers sip/iax realtime friend? If that friend is described by users.conf, the hint is created... |
| 13:29.19 | nethium | hi asterix newby here... basic question, if I wanted to programmitcally bridge two callers, should I use asterix or find something else? I.e program calls person 1, program calls person 2 then connects them together. |
| 13:30.28 | hank | [TK]D-Fender: i just checked: ip is correct. |
| 13:30.42 | davevg-btwtech | nethium: you can do that through a call file or manager a originate command |
| 13:30.48 | d1mas | I do not think "caller" is proper word here... |
| 13:31.45 | nethium | so I can programatically generate a call file and submit that somewhere to a folder on the server, and the server will initiate the two calls? |
| 13:32.37 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:32.41 | Katty | herro |
| 13:33.19 | davevg-btwtech | nethium: yes http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 13:33.54 | nethium | Davevg thanks for the info this is perfect |
| 13:33.54 | hank | [TK]D-Fender: any idea? |
| 13:34.01 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 13:34.01 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:35.19 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:35.19 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:35.26 | *** join/#asterisk gardo (n=gardo@121.97.249.68) |
| 13:39.01 | [TK]D-Fender | hank: corrent for WHAT device? |
| 13:39.36 | [TK]D-Fender | Katty: Mew. |
| 13:39.47 | Katty | [TK]D-Fender: hewwo. |
| 13:40.23 | hank | [TK]D-Fender: i ran 'tshark dst 192.168.240.126' on the debian in virtualbox. 192.168.240.126 is debians ip. then i connected to 192.168.240.126:8088 from my workstation and it returned 192.168.240.117 as the source. so the ip in the config is correct. (or i am wrong) |
| 13:40.32 | Katty | [TK]D-Fender: tomorrow i get to take my earings out. |
| 13:40.37 | [TK]D-Fender | d1mas: Because users.conf is a flaming piece of transh and its parser GENERATES dialplan. |
| 13:40.42 | Katty | [TK]D-Fender: they'll have been pierced for exactly 6 weeks (= |
| 13:40.50 | rob0 | ouch |
| 13:41.04 | [TK]D-Fender | d1mas: And jsut because you have a sip realtime entry implies NOTHING about how your dialplan operates. |
| 13:41.07 | Katty | file :> |
| 13:41.25 | Katty | almongst other things... |
| 13:41.27 | [TK]D-Fender | Katty: You need pierings like a hole in the head... |
| 13:41.30 | Katty | i've been thinking about moving to vegas. |
| 13:41.33 | [TK]D-Fender | Katty: err.... wait a sec.... |
| 13:41.39 | Katty | [TK]D-Fender: ;) |
| 13:42.05 | Katty | i don't think i know anyone in vegas :/ |
| 13:42.31 | JT | [TK]D-Fender: sort of like how people need bad spelling ;) |
| 13:43.00 | creativx | heheh |
| 13:43.04 | [TK]D-Fender | JT : My spelling is perfect, my TYPING considerably less so ;) |
| 13:43.33 | *** join/#asterisk dmcmorris_ (n=don@adsl-154-39-83.asm.bellsouth.net) |
| 13:43.44 | JT | sure ;) |
| 13:44.41 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 13:45.36 | Katty | [TK]D-Fender: do you know anyone in vegas? |
| 13:45.55 | [TK]D-Fender | Katty: No, and remember.... what happens in Vegas STAYS in Vegas :) |
| 13:46.10 | nethium | davevg-btwtech - One quick question, is there any logical difference between connecting the outbound callback to an internal extension as these example callfiles show and connecting it to another outside session? |
| 13:46.11 | *** join/#asterisk gardo (n=gardo@121.97.249.68) |
| 13:46.16 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 13:46.17 | Corydon76-dig | What happens in Vegas stays on Youtube |
| 13:46.19 | Katty | [TK]D-Fender: >.< |
| 13:46.39 | Katty | well i can't stay in southern missouri |
| 13:46.42 | Katty | it's slowly eating my SOUL |
| 13:47.10 | [TK]D-Fender | Katty: Become a Canuckian and you shall be SAVED! |
| 13:47.16 | Katty | joy. |
| 13:47.29 | hank | [TK]D-Fender: assuming my manager.conf is correct what else could cause this? |
| 13:47.35 | ManxPower | Hell, in 50 years canada will be warm enough to live in. |
| 13:47.53 | davevg-btwtech | nethium: it calls the Channel parameter 1st and when that answers connects it to the Context/Extension/Priority |
| 13:48.20 | [TK]D-Fender | hank: Now why would I go and do a crazy thing like that? :p |
| 13:48.51 | [TK]D-Fender | ManxPower: And many southern states are warm enough to be HELL already :p |
| 13:49.00 | [TK]D-Fender | ManxPower: ..... pwned |
| 13:49.04 | [TK]D-Fender | :D |
| 13:50.24 | hank | [TK]D-Fender: 'g' i dunno... i'll nopaste it... |
| 13:50.44 | [TK]D-Fender | hank: And that STILL won't answer anything of use... |
| 13:51.09 | hank | [TK]D-Fender: hm ok... so how do i debug? :( |
| 13:52.06 | [TK]D-Fender | hank: You never said WHOSE IP that was. And being from HTTP, I'm not sure if the connection should be considered as coming from your CLIENT, or your SERVER <--- |
| 13:52.22 | [TK]D-Fender | hank: I would allow your entire subnet access right now for sanity's sake |
| 13:53.00 | hank | [TK]D-Fender: 192.168.240.117 is my workstation and 192.167.240.126 is the virtualbox running on my workstation. |
| 13:53.13 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 13:53.20 | hank | [TK]D-Fender: yes, thats what i was asking for about 10 minutes ago. what would i have to use for deny and permit? |
| 13:53.20 | [TK]D-Fender | hank: Allow EVERYTHING for now |
| 13:53.28 | *** join/#asterisk s34n (n=chatzill@ip-206-159-190-125.mvdsl.com) |
| 13:53.53 | [TK]D-Fender | hank: permit would be something like 192.168.0.0 / 255.255.0.0 |
| 13:54.00 | [TK]D-Fender | hank: deny looked fine |
| 13:54.13 | s34n | I'm trying to get the register=> command right to connect * to my itsp. |
| 13:54.34 | s34n | I've tried all different versions of username@domain:password |
| 13:54.35 | hank | [TK]D-Fender: still, permission denied |
| 13:54.57 | s34n | username:password:authname |
| 13:55.46 | s34n | but nothing seems to work. I get timeouts or I get "Forbidden - wrong password" |
| 13:56.03 | [TK]D-Fender | s34n: register => user:pass@host/extentodial |
| 13:56.03 | s34n | but my phones can register to the same itsp no problem. |
| 13:56.30 | [TK]D-Fender | s34n: because your phones are sending the right info in the right places. Your registers were improperly formatted. |
| 13:56.37 | hank | [TK]D-Fender: argh |
| 13:56.47 | s34n | [TK]D-Fender: The /extentodial is optional and defaults to s, right? |
| 13:56.59 | [TK]D-Fender | s34n: Yes, though i highly recommend it. |
| 13:57.04 | hank | [TK]D-Fender: whitespace between permissions caused the problem |
| 13:57.16 | [TK]D-Fender | Katty: Weebles may wobble but they won't fall down! |
| 13:57.20 | hank | like: 'agent, user'. 'agent,user' seems to work. |
| 13:59.17 | hank | but i am a bit surprised about that webinterface... |
| 13:59.27 | hank | theres not really much you can do with it, is there? |
| 14:00.03 | [TK]D-Fender | hank: its AMI... I presume you can do whatever AMI lets you do... |
| 14:00.25 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 14:00.25 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 14:00.38 | hank | [TK]D-Fender: AMI? |
| 14:00.44 | ^JimmyRidge^ | ugh im retarded |
| 14:00.52 | ^JimmyRidge^ | its udp |
| 14:00.55 | [TK]D-Fender | hank: Manager = AMI <- |
| 14:01.02 | ^JimmyRidge^ | nmap's default scan =tcp only |
| 14:01.06 | ^JimmyRidge^ | :P |
| 14:01.12 | ^JimmyRidge^ | its open im just stupid |
| 14:01.14 | [TK]D-Fender | ^JimmyRidge^: Excellent... the first step is admitting you have a problem.... |
| 14:01.18 | s34n | [TK]D-Fender: I have tried user:pass@host/extentodial. It tells me wrong password |
| 14:01.18 | ^JimmyRidge^ | hah |
| 14:01.34 | s34n | [TK]D-Fender: and yes, I have rechecked my password 20 times |
| 14:01.35 | [TK]D-Fender | s34n: something is wrong then. |
| 14:01.44 | [TK]D-Fender | s34n: and its most likely YOU |
| 14:01.58 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 14:01.58 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 14:02.04 | hank | [TK]D-Fender: ah ok... yes but why is it called 'ajam'? atm i cannot find anything that reminds me in the least to ajax... |
| 14:02.14 | Katty | [TK]D-Fender: i'm all wobbly today :< |
| 14:02.17 | Katty | [TK]D-Fender: lot on my mind. |
| 14:02.23 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:02.23 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:02.44 | Katty | [TK]D-Fender: they had a ferret at the humane society (which i ended up not getting), but I went out to see the doggies... and started crying :/ |
| 14:02.46 | s34n | [TK]D-Fender: I certainly agree |
| 14:02.57 | s34n | [TK]D-Fender: But I can't track it down |
| 14:03.18 | ^JimmyRidge^ | you kno how asterisks likes to register with hosts? |
| 14:03.26 | *** join/#asterisk redondos (n=redondos@unaffiliated/redondos) |
| 14:03.32 | ^JimmyRidge^ | is there a option to tell it how often to do that |
| 14:03.45 | s34n | [TK]D-Fender: I think my eyes and fingers are all crossed the right ways |
| 14:04.02 | ^JimmyRidge^ | right now its retying every 15/30 sec |
| 14:04.42 | s34n | ^JimmyRidge^: 15-30, or 1/2? |
| 14:05.07 | ^JimmyRidge^ | i dunno but its register interval is really fast |
| 14:05.29 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 14:05.48 | s34n | ^JimmyRidge^: "between 15 and 30 seconds" or "every half second" ? |
| 14:05.57 | ^JimmyRidge^ | 15sec |
| 14:06.01 | [TK]D-Fender | hank: You are getting live access to AMI over HTTP, of course thats AJAX |
| 14:06.18 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 14:06.34 | [TK]D-Fender | ^JimmyRidge^: likely your provider telling * that's how often to re-reg |
| 14:06.34 | s34n | ^JimmyRidge^: successful each time? or failing each time? |
| 14:06.40 | Defraz | Nick Defraz |
| 14:06.43 | ^JimmyRidge^ | fail |
| 14:06.55 | s34n | ^JimmyRidge^: that is your sip.conf |
| 14:07.00 | ^JimmyRidge^ | they havent added my credentials to the server ilbeit |
| 14:07.07 | ^JimmyRidge^ | its iax |
| 14:07.17 | *** join/#asterisk Op3r (n=Op3r@121.97.211.37) |
| 14:07.53 | hank | [TK]D-Fender: i guess i expected something more. like a nice interface. or at least something where you can click. just calling some url and see wether there was a positive or a negative result was definitely not what i expected |
| 14:08.05 | s34n | ^JimmyRidge^: ok. but the retry interval is probably set in your conf |
| 14:08.21 | ^JimmyRidge^ | not in mine ;) |
| 14:08.27 | ^JimmyRidge^ | bareboned |
| 14:08.51 | ManxPower | Asterisk is not a PBX. It is a toolkit that lets you build a pbx. |
| 14:09.04 | s34n | ^JimmyRidge^: then it is using the default. I think the sip retry default is 20 seconds. Not sure about iax |
| 14:09.30 | *** join/#asterisk Geert (i=geert@irssi/staff/geert) |
| 14:09.30 | *** join/#asterisk medicACS (i=medic@d149-67-142-10.clv.wideopenwest.com) |
| 14:09.33 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:09.33 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:09.44 | ^JimmyRidge^ | minregexpire = and or rtautoclear= |
| 14:09.53 | Geert | Can somebody give me the difference between, allowed_not_screened, allowed_passed_screen and allowed_failed_screen |
| 14:10.00 | ^JimmyRidge^ | but whokno's if that aplies to a failed attempt |
| 14:10.55 | ^JimmyRidge^ | i guess i should just chill and wait and keep enhancing asterisks in the meantime |
| 14:11.14 | ^JimmyRidge^ | im learning, been tweaking with it allnight |
| 14:11.32 | medicACS | g'day.. anyone successfully get SLA working in here that can spare a moment? |
| 14:12.33 | De_Mon | Geert where do yous ee them? |
| 14:13.12 | ^JimmyRidge^ | any1 kno a good tutorial that talks about asterisk that talks about festival and option menus? |
| 14:13.26 | ^JimmyRidge^ | wow thats a terrible sentance |
| 14:13.48 | Geert | De_Mon: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerPres & http://www.voip-info.org/wiki-Asterisk+cmd+CallingPres |
| 14:13.54 | Geert | it's to force outbound CLI |
| 14:14.02 | [TK]D-Fender | hank: Oh... go get FOP then :p |
| 14:14.16 | Geert | But I want to know the differences between them |
| 14:14.38 | ^JimmyRidge^ | ah its called IVR interactive voice response |
| 14:14.49 | [TK]D-Fender | medicACS: There is no SLA in *, on a dirty and high-useless hack that pretends to be. |
| 14:15.27 | [TK]D-Fender | ^JimmyRidge^: nothing to know really.... |
| 14:15.28 | hank | [TK]D-Fender: o_O yeah well thats flash... what is ajam for then? |
| 14:15.46 | [TK]D-Fender | hank: For you to have HTTP access for testing. |
| 14:16.02 | hank | oh ok |
| 14:16.11 | hank | i thought it was meant to become something like fop |
| 14:16.14 | ^JimmyRidge^ | i just wanna play with stuff besides the basic call routing tables |
| 14:16.18 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 14:16.38 | medicACS | tnks Fender so this here http://asterisk.org/node/48342 is junk and will not work? |
| 14:16.47 | hank | probably i was just confused by 'ajax'. it sounds sooo 'web2.0ish' :-p |
| 14:18.38 | hank | but where is javascript used? |
| 14:19.02 | s34n | [TK]D-Fender: register => user:pass@host/extentodial then I can add a [host] section with more config, right? |
| 14:19.32 | s34n | [TK]D-Fender: like default context, set it as a peer, etc. |
| 14:19.43 | [TK]D-Fender | s34n: that is setting up a PEER. That is aCOMPLETELY seperate part of this process. |
| 14:20.18 | [TK]D-Fender | medicACS: Oh, it will WORK, its just a question of what it technically DOES <---- |
| 14:20.54 | medicACS | ah... my problem is I cannot get my polycoms to subscribe to the right lines... |
| 14:21.02 | [TK]D-Fender | ^JimmyRidge^: What is "besides", and what is "BASIC call routing tables" mean anyways? |
| 14:21.17 | ^JimmyRidge^ | i dunno |
| 14:21.36 | [TK]D-Fender | medicACS: that SLA is a 2-bit hack using Presence & MeetMe. |
| 14:21.47 | [TK]D-Fender | medicACS: What do you intend to use this for? |
| 14:21.58 | Daviey | Anybody going to "Asterisk and VoIP in |
| 14:22.00 | Daviey | enterprise" |
| 14:22.07 | Daviey | in belgium ? |
| 14:22.16 | [TK]D-Fender | Daviey: Never heard of that city before.... |
| 14:22.19 | s34n | [TK]D-Fender: the register will use the domain from [general] right? |
| 14:22.21 | medicACS | I have a pizza joint that wants to keep key system like functionality |
| 14:22.27 | [TK]D-Fender | Daviey: darn... you added a COUNTRY to that! |
| 14:22.35 | [TK]D-Fender | s34n: NO. |
| 14:22.49 | [TK]D-Fender | s34n: the register line does EVERYTHING based on THAT LINE <- |
| 14:22.58 | [TK]D-Fender | s34n: it doesn't look at ANYTHING |
| 14:23.00 | *** join/#asterisk bweschke (n=bweschke@204.13.233.213) |
| 14:23.07 | Daviey | [TK]D-Fender: http://www.profoss.eu/events/ |
| 14:23.17 | *** join/#asterisk SickTear (i=SickTear@89.137.56.184) |
| 14:23.39 | [TK]D-Fender | medicACS: you'll need high end phones with LOTS of mappable buttons and VERY few lines for that... |
| 14:23.46 | s34n | [TK]D-Fender: but I can register => user@domain:password@host ? |
| 14:23.49 | [TK]D-Fender | medicACS: They should simply learn to cope with parking... |
| 14:24.09 | [TK]D-Fender | s34n: Yeah I think that format is valid... |
| 14:24.25 | [TK]D-Fender | s34n: if that domain is part of their user system |
| 14:24.53 | medicACS | stubborn.. so park is not an option.. yea i have 4 POTS lines and a 6line polycom 601 so I am ok hardware wise... just got get the darn subscriptions working |
| 14:25.29 | Geert | Daviey: url? |
| 14:25.34 | *** join/#asterisk svenl (n=svenl@power.plz.fr) |
| 14:25.55 | Daviey | Geert: ^ |
| 14:26.03 | Daviey | 15:22 < Daviey> [TK]D-Fender: http://www.profoss.eu/events/ |
| 14:26.21 | Geert | yes I'm going to that |
| 14:26.59 | Daviey | Geert: I can't decide... :S |
| 14:27.53 | [TK]D-Fender | Daviey: That looks like a real beginners conference.... |
| 14:28.24 | Geert | why not? it's not expensive |
| 14:28.31 | Daviey | [TK]D-Fender: Yeah, not a -dev one for sure - but still could be fun :) |
| 14:28.34 | Geert | tho I'm not really convinced it will be really interesting |
| 14:28.56 | [TK]D-Fender | Geert: That'd depend on your idea of "interesting" :) |
| 14:29.03 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 14:29.18 | Daviey | Geert: What country are you in? |
| 14:29.25 | Geert | Daviey: Belgium, Brussels ;) |
| 14:29.30 | Daviey | bah |
| 14:30.08 | Daviey | Incidently, the return flight is only 10mins for me (timezone) :) |
| 14:31.30 | Geert | .co.uk is doable :) |
| 14:32.35 | Geert | and I can give you 30% discount on the entrance fee if you're coming :p |
| 14:34.43 | Daviey | Geert: using the code on asterisk-biz? |
| 14:35.02 | variable_office | i am trying to do make menuselect with *1.4 but odbc voicemail storage is XXX; both things it requires have (E) next to them? what does this mean? |
| 14:35.14 | Geert | no, we sponsor the event :p |
| 14:36.51 | Daviey | Geert: Making it 203 + VAT? |
| 14:37.11 | *** join/#asterisk caguy (i=caguy@124-169-19-67.dyn.iinet.net.au) |
| 14:38.25 | *** part/#asterisk hank (n=hank@netwichtig.de) |
| 14:38.58 | [TK]D-Fender | VAT is a horrible misnomer... |
| 14:39.08 | [TK]D-Fender | it doesn't add VALUE, it adds COST |
| 14:39.41 | *** part/#asterisk ManxPower (n=manxpowe@ip68-107-250-148.no.no.cox.net) |
| 14:39.51 | Daviey | VALUE as in tax based on the VALUE of product :) |
| 14:40.19 | syzygyBSD | well then it should still be cost, value is a subjective term |
| 14:40.29 | Geert | Daviey: indeed, but I don't think you have to pay VAT since you're in .co.uk |
| 14:40.48 | Daviey | Blimey O'reily!!! 21% VAT |
| 14:40.58 | [TK]D-Fender | Daviey: Thats not valued ADDED, thats value BASED :p |
| 14:41.52 | Daviey | TAX that is based on the VALUE and is ADDED to the total |
| 14:42.14 | syzygyBSD | Daviey: wouldn't it be TVA then? |
| 14:42.42 | Corydon76-dig | Tennessee Valley Authority? |
| 14:43.04 | Corydon76-dig | That would be my power supplier |
| 14:43.10 | lirakis | is there anyway to make it so my cdr's dont fork when using queues? .. or rather.. how/what do you do reporting with when you have split cdr's? |
| 14:43.32 | Corydon76-dig | Digium's power supplier, too |
| 14:44.45 | syzygyBSD | Corydon76-dig: not in white county are you? |
| 14:44.55 | syzygyBSD | might be the wrong state |
| 14:45.02 | Corydon76-dig | Uh, no, Davidson |
| 14:45.56 | Corydon76-dig | White county is east of here |
| 14:46.32 | _x86_ | <PROTECTED> |
| 14:46.41 | _x86_ | I have that in the CLI when I do: sip show settings |
| 14:46.52 | _x86_ | how do i enable SIP realtime? |
| 14:47.10 | Corydon76-dig | _x86_: extconfig.conf |
| 14:47.29 | _x86_ | I've setup res_mysql.conf, realtime mysql status shows that it is connected to the database, and extconfig.conf is setup |
| 14:47.29 | Corydon76-dig | See doc/realtime.txt |
| 14:47.36 | _x86_ | yeah, been there, done that ;) |
| 14:47.41 | _x86_ | not working ;) |
| 14:47.54 | Corydon76-dig | did you reload after changing it? |
| 14:48.17 | _x86_ | i'm fairly certain... |
| 14:48.19 | _x86_ | I will again |
| 14:48.25 | _x86_ | reload sip? |
| 14:48.32 | _x86_ | reload chan_sip.so |
| 14:48.34 | arekm | has anyone seen asterisk ghosts? http://pastebin.com/m556ef69e the phone is connected, onhook, and... it's calling somehow |
| 14:49.00 | _x86_ | <PROTECTED> |
| 14:49.01 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:49.02 | _x86_ | did not help |
| 14:49.08 | arekm | I soft hungup it and it starts dialing again immediately |
| 14:49.16 | Corydon76-dig | reload extconfig |
| 14:49.19 | *** join/#asterisk vader-- (n=me@c-71-226-197-0.hsd1.nj.comcast.net) |
| 14:49.21 | vader-- | hello |
| 14:49.52 | _x86_ | Corydon76-dig: i did a full restart, same results |
| 14:50.25 | Corydon76-dig | _x86_: okay, first things first. What line do you have in extconfig.conf for sippeers? |
| 14:50.35 | _x86_ | sipusers => mysql,asterisk,sip_friends |
| 14:50.36 | _x86_ | sippeers => mysql,asterisk,sip_friends |
| 14:50.59 | _x86_ | like i said, from the CLI, I can do realtime mysql status and it connects to the database just fine |
| 14:51.05 | *** join/#asterisk masus (n=tet@88.248.73.2) |
| 14:51.14 | _x86_ | *CLI> realtime mysql status |
| 14:51.15 | _x86_ | Connected to asterisk@127.0.0.1, port 3306 with username asterisk for 1 minutes, 58 seconds |
| 14:51.30 | masus | "Sep 25 13:49:05 WARNING[2469] loader.c: /usr/lib/asterisk/modules/func_odbc.so: undefined symbol: SQLGetDiagField" |
| 14:51.41 | masus | can anybody help with func_odbc +asterisk 1.2 |
| 14:51.54 | Corydon76-dig | _x86_: and the table sip_friends exists in the asterisk database? |
| 14:52.01 | _x86_ | yessir |
| 14:52.21 | tzafrir | masus, hmmm, non-matching version of unixodbc? |
| 14:52.25 | Corydon76-dig | _x86_: does the mysql user 'asterisk' have permission to select from the table? |
| 14:52.42 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:52.48 | masus | tzafrir : i'll see one mom. |
| 14:53.26 | _x86_ | http://pastebin.ca/710432 |
| 14:53.31 | _x86_ | Corydon76-dig: check that out |
| 14:53.58 | Corydon76-dig | _x86_: okay, what makes you think it's not working? |
| 14:54.17 | _x86_ | polycom phones trying to connect get rejected with 404 not found |
| 14:54.49 | Corydon76-dig | _x86_: core set debug 5 and try again |
| 14:55.07 | _x86_ | http://pastebin.ca/710436 |
| 14:55.23 | _x86_ | Corydon76-dig: heh, running 1.2 here ;) |
| 14:55.40 | masus | tzafrir: unixODBC 2.2.11 |
| 14:55.46 | Corydon76-dig | Fine, set debug 5, ensure debug is enabled for console in logger.conf |
| 14:56.20 | [TK]D-Fender | _x86_: pastebin the SIP debug shoing the failure |
| 14:56.20 | _x86_ | http://pastebin.ca/710437 |
| 14:56.24 | vader-- | anyone inh ere running 7940G phones? |
| 14:56.31 | _x86_ | [TK]D-Fender: check that URL |
| 14:56.39 | vader-- | im wondering if it's possible to update the firmware on just one phone? |
| 14:56.51 | tzafrir | masus, asterisk is self-built, right? and vs. that exact version of unixodbc? |
| 14:57.16 | masus | yes self built |
| 14:57.17 | Corydon76-dig | _x86_: core debug, not sip debug |
| 14:57.25 | [TK]D-Fender | _x86_: please use ASTERISK's CLI sip debug |
| 14:57.25 | _x86_ | i did both |
| 14:57.27 | masus | but i do not which version tu use |
| 14:57.31 | _x86_ | [TK]D-Fender: i am? |
| 14:57.41 | tzafrir | hmmm.... I figure others around here may know func_odbc better than me... |
| 14:57.55 | [TK]D-Fender | _x86_: I think your headers are a little wierd is all... |
| 14:57.55 | Corydon76-dig | _x86_: then you either haven't set debug to 5 or you haven't enabled debug in logger.conf |
| 14:58.17 | _x86_ | http://pastebin.ca/710440 |
| 14:58.29 | _x86_ | Corydon76-dig: i did here |
| 14:58.35 | _x86_ | (had to reload the logger) |
| 14:58.46 | masus | tzafrir:thanks |
| 14:59.16 | tzafrir | masus, what distro is it? |
| 14:59.28 | masus | debian etch |
| 14:59.31 | masus | asterisk 1.2 |
| 14:59.33 | tzafrir | is unixodbc from a distro package? |
| 14:59.38 | masus | yes |
| 14:59.41 | *** join/#asterisk kmchen (n=kmchen@gar13-4-82-240-99-84.fbx.proxad.net) |
| 14:59.50 | kmchen | hi |
| 14:59.51 | masus | i'll try from source now |
| 14:59.55 | *** join/#asterisk demiv (n=demiv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 15:00.02 | _x86_ | set debug 5 doesn't really offer any insight |
| 15:00.04 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 15:00.26 | tzafrir | hmmm... the deb builds fine vs. that version |
| 15:00.32 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 15:00.34 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 15:00.41 | tzafrir | try installing it in a debootstrap... |
| 15:00.54 | [TK]D-Fender | _x86_: pastebin your sip peers dump. |
| 15:00.55 | _x86_ | [2007-09-25 10:00:39] NOTICE[16126]: chan_sip.c:11575 handle_request_register: Registration from '<sip:7095@10.1.1.249>' failed for '10.1.1.183' - ACL error (permit/deny) |
| 15:01.00 | *** join/#asterisk dlynes_home (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 15:01.04 | _x86_ | eh? |
| 15:01.21 | Corydon76-dig | _x86_: give me remote access and I'll look at it |
| 15:01.33 | _x86_ | *CLI> sip peers dump |
| 15:01.34 | _x86_ | No such command 'sip peers' (type 'help' for help) |
| 15:01.37 | kmchen | Does some body would help a newbie to configure an asterisk/freephonie.net/X-lite under Debian ? |
| 15:01.57 | [TK]D-Fender | kmchen: ... |
| 15:02.00 | [TK]D-Fender | ~book |
| 15:02.01 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 15:02.04 | [TK]D-Fender | ~wikis |
| 15:02.04 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 15:02.37 | [TK]D-Fender | kmchen: http://www.jeremy-mcnamara.com/index.php/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 15:02.47 | tzafrir | kmchen, which Debian exactly? |
| 15:03.15 | tzafrir | ah, ok |
| 15:03.18 | tzafrir | misread it |
| 15:03.27 | masus | tzafrir: after .configure new unixodbc from source i get this error -> error: cannot find correct Qt headers! |
| 15:03.34 | kmchen | I did read http://www.aocomputing.net/AsteriskTFOT.pdf and configured my asterisk. I can receive calls and communicate in local but no way to get out since several days ! |
| 15:03.59 | tzafrir | masus, installing it from source will probably only increase the noise level |
| 15:04.09 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:04.19 | tzafrir | for starters, let's verify that the right version is used |
| 15:04.47 | tzafrir | ldd /usr/lib/asterisk/modules/func_odbc.so |
| 15:04.53 | kmchen | tzafir: sarge |
| 15:05.02 | tzafrir | does it show the right unixodbc ? |
| 15:05.09 | variable_office | i am trying to do make menuselect with *1.4 but odbc voicemail storage is XXX; both things it requires have (E) next to them? does that mean i have wrong unixodbc version? |
| 15:05.33 | tzafrir | or maybe need to re-run ./configure ? |
| 15:05.50 | masus | have found it "--enable-gui=no" |
| 15:08.23 | kmchen | I have already successfully installed every packages and sip and extensions.conf but I can't get authenticated by my provider (freephonie.net) |
| 15:08.24 | tzafrir | masus, asterisk 1.2 builds fine on my Etch with that package |
| 15:09.55 | kmchen | perhaps a NAT problem. Does somebody understand something about it ? |
| 15:10.48 | variable_office | what version of unixodbc does odbc voicemail storage require? |
| 15:11.49 | pjz | anyone have recommendations on an asterisk appliance (besides THE asterisk appliance from digium) ? |
| 15:11.57 | *** join/#asterisk masus (n=tet@88.248.73.2) |
| 15:12.41 | arekm | what does "bpviol" value in zap show status means? |
| 15:13.02 | Corydon76-dig | bipolar violation |
| 15:13.16 | pjz | no one? everyone builds their own boxes? |
| 15:13.23 | [TK]D-Fender | kmchen: Show us SIP debug of your failure in a PASTEBIN |
| 15:13.25 | [TK]D-Fender | ~pb |
| 15:13.26 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:13.27 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 15:13.30 | arekm | Corydon76-dig: and what's that? |
| 15:13.51 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:13.57 | Strom_M | arekm: it's a T1 line coding error |
| 15:14.15 | Strom_M | arekm: read this |
| 15:14.15 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-187-88.dsl.irvnca.pacbell.net) |
| 15:14.16 | Strom_M | http://www.stromcarlson.com/docs/basics/t1svcfund.pdf |
| 15:14.20 | UnixDog | morning |
| 15:14.32 | masus | tzafrir: do u mean installing unixodbc ? |
| 15:14.36 | arekm | http://pastebin.com/m4cee7375 I assume 5k of these is very bad |
| 15:14.50 | Strom_M | arekm: yes, extremely |
| 15:15.11 | Strom_M | arekm: are you using AMI or B8ZS? |
| 15:16.10 | kmchen | ok I can try that. |
| 15:16.15 | arekm | Strom_M: b8zs, these two t1 are connected to infamous zhone z-plex channel banks |
| 15:16.37 | lirakis | how do I associate the two CDR's that are created when the CDR forks after a call in a queue is answered? |
| 15:16.44 | Strom_M | arekm: read the document I just linked you to |
| 15:17.00 | arekm | Strom_M: reading it already |
| 15:17.21 | *** join/#asterisk iam8up|lpy (n=iam8up@rrcs-74-218-88-130.central.biz.rr.com) |
| 15:17.24 | lirakis | .. i get a CDR .. of basically the ringtime... and then i get another CDR of the call time with the agent.... but i have no way to associate them b/c the calldate of the agent talk time is stamped when they hang up |
| 15:18.27 | kmchen | OK http://pastebin.com/m619f3310 |
| 15:18.36 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 15:18.59 | *** join/#asterisk hfb (n=hfb@pool-71-106-223-11.lsanca.dsl-w.verizon.net) |
| 15:19.03 | kmchen | Can you see it ? |
| 15:19.13 | iam8up|lpy | is anyone familiar with the cisco 7940s/7960s with sip firmware? i just got a lot o fthem and what i'm normally able to do is hold the # key while i plug in the power cord and it gives me the factory default sequence; with some of these phones it boots normally, others it does as it's supposed to |
| 15:19.43 | [TK]D-Fender | kmchen: Sep 24 10:14:01 NOTICE[5802]: chan_sip.c:11131 handle_request_register: Registration from 'freephonie <sip:0954XXXXXX@freephonie.net>' failed for '192.168.0.10' - Username/auth name mismatch |
| 15:19.46 | iam8up|lpy | what i need to do is change the tftp server, but i've no idea how to do so if i can't factory default it; the tftp server seems to be configured by hand |
| 15:20.00 | [TK]D-Fender | kmchen: Your register has the wrong info or isn't formatted properly |
| 15:21.00 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 15:21.19 | [TK]D-Fender | :O |
| 15:21.56 | kmchen | Yes I understood that but can't figure out why ? |
| 15:22.11 | *** join/#asterisk boch (n=fran@190.48.208.209) |
| 15:22.12 | pjz | no one? everyone builds their own boxes? I just need soething small to handle like 4 FXO and 4FXS and then the usual voicemail, etc. for a couple dozen internal SIP phones. maybe internet VOIP providers in the future. Anyone? |
| 15:22.17 | kmchen | Do you wanna see my conf files |
| 15:23.21 | defsdoor | is there a way to add a 0 prefix to the incoming cli ? |
| 15:23.43 | UnixDog | morning ... ok |
| 15:23.44 | boch | do you know what is the 101 number in this SDP line: a=fmtp:101 0-15 |
| 15:23.45 | Strom_M | pjz: the consensus, as far as I'm aware, is either 'build it yourself" or "go with the digium appliance" |
| 15:23.57 | pjz | Strom_M: the digium appliance is giving me problems. |
| 15:24.03 | Strom_M | o rly |
| 15:24.09 | pjz | Strom_M: yah rly |
| 15:24.15 | Strom_M | what kind of problems |
| 15:24.18 | pjz | Strom_M: not being able to save its config thru a reboot |
| 15:24.28 | Strom_M | pjz: have you called support? |
| 15:24.32 | defsdoor | pjz: I've installed a couple of trixboxes |
| 15:24.35 | pjz | Strom_M: I have. they're due to call me back today. |
| 15:25.09 | [TK]D-Fender | kmchen: PM me your "register" line |
| 15:25.32 | dlynes_laptop | defsdoor: how about Set(CALLERID(num)=0${CALLERID(num)})? |
| 15:26.18 | UnixDog | I want to grab the dialed did nomner that comes into my box . be it exten => x.,1, on the inbound. I need to grab te dialed numbe and if ithe first 2 numbers match shove it back out a trunk |
| 15:26.35 | UnixDog | nomner/number |
| 15:27.25 | *** join/#asterisk thansen|laptop (n=thansen@128.187.146.199) |
| 15:27.37 | Strom_M | UnixDog: that's easy |
| 15:27.40 | UnixDog | I forget the context is it ${DID} |
| 15:27.40 | *** join/#asterisk ta^3 (n=tacvbo@65.116.224.30) |
| 15:27.42 | Strom_M | assuming you have DNIS service |
| 15:28.00 | arekm | Strom_M: hm, in this channel bank I have option to set ds1-mode to DSX or CSU and only csu is mentioned in that doc - what's DSX? |
| 15:29.38 | Strom_M | if you have two ports on a channel bank, one pointing at the network and one pointing at CPE, the side pointing at CPE is a DSX-1 |
| 15:30.27 | *** join/#asterisk Runlvl (n=juan@190.2.40.57) |
| 15:30.27 | UnixDog | exten = s,n,Set(${EXTEN}=${DID}) I thinl this is right |
| 15:30.31 | Runlvl | Hello |
| 15:30.47 | UnixDog | gawd I hate laptop keyboards |
| 15:30.59 | Strom_M | UnixDog: no and no. |
| 15:31.01 | Runlvl | I have a problem with meetme - the agi background |
| 15:31.08 | Runlvl | any one can help me ? |
| 15:31.30 | Strom_M | Runlvl: |
| 15:31.32 | Strom_M | ~ask |
| 15:31.33 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:31.36 | variable_office | is there a prerequisite list somewhere so ic an tell why i cant install unixodbc |
| 15:31.42 | variable_office | voicemail straoge |
| 15:31.56 | UnixDog | ok storm can oyu point me in the right dir |
| 15:32.04 | Strom_M | who's "storm"? |
| 15:32.12 | UnixDog | Strom |
| 15:32.18 | [TK]D-Fender | UnixDog>exten = s,n,Set(${EXTEN}=${DID}) I thinl this is right <--- no this is completely wrong. |
| 15:32.19 | Runlvl | Strom_C, when i run the script in meetme |b the script works.. but the zaptel kick me |
| 15:32.41 | Runlvl | Hungup 'Zap/pseudo-379051020' |
| 15:32.43 | [TK]D-Fender | UnixDog: For 2 separate reasons. |
| 15:32.44 | *** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.41) |
| 15:32.54 | Runlvl | i have a zaptel card |
| 15:32.59 | UnixDog | ok |
| 15:33.03 | UnixDog | listening |
| 15:33.09 | Strom_M | UnixDog: if you're matching on DNIS in your inbound context, then EXTEN will already contain the DNIS string which matched the pattern |
| 15:33.30 | l0verb0y | hey has anyone used adhearsion? |
| 15:33.32 | arekm | Strom_M: the second port is to do kind of pass-thru channel bank to connect some additional equipment? |
| 15:33.35 | [TK]D-Fender | UnixDog: first in that case ${EXTEN} = "s", so its WOTHLESS to you. |
| 15:33.36 | Strom_M | exten => _X.,1,NoOp(The DNIS says: ${EXTEN}) |
| 15:33.56 | Strom_M | arekm: yes |
| 15:33.57 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 15:34.02 | [TK]D-Fender | UnixDog: Second you can't SET ${EXTEN} its READ ONLY |
| 15:34.17 | *** join/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 15:34.32 | [TK]D-Fender | UnixDog: And next, you don't put ${} around the variable you want to set. |
| 15:35.30 | UnixDog | ok |
| 15:35.36 | UnixDog | brb |
| 15:36.16 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:36.41 | arekm | Strom_M: and when connecting to asterisk I should rather set CSU mode on channel bank right? |
| 15:37.04 | arekm | not DSX |
| 15:38.32 | UnixDog | [inbound] exten => 52.,1,Dial(SIP/VD-OUT,00${EXTEN},,) |
| 15:39.00 | *** join/#asterisk Entr4nced (n=Entr4nce@67-129-213-39.dia.static.qwest.net) |
| 15:39.11 | Strom_M | arekm: assuming that the channel bank is farther from the center of the network than asterisk, then yes |
| 15:39.19 | Strom_M | UnixDog: why the extra commas on the end of Dial()? |
| 15:39.21 | UnixDog | but this is not working to force it to dial back out . all it shows when it dials is the 52. |
| 15:39.39 | Strom_M | UnixDog: because you forgot your underscore at the beginning to indicate pattern matching |
| 15:39.57 | UnixDog | ahh ok |
| 15:40.00 | arekm | Strom_M: yes, it's telco -- asterisk -- channel bank -- analog phones. Oh, there is "other" mode, too (whatever that means in this channel bank) |
| 15:40.44 | UnixDog | [inbound] exten => _52.,1,Dial(SIP/VD-OUT,00${EXTEN},,) then |
| 15:41.19 | Strom_M | UnixDog: i hope those are actually on two separate lines in extensions.conf |
| 15:41.26 | Strom_M | and take the extra commas off the end of Dial() |
| 15:41.30 | UnixDog | they are |
| 15:41.48 | *** join/#asterisk Runlvl (n=juan@190.2.40.57) |
| 15:42.11 | Runlvl | When asterisk run the agi background |
| 15:42.23 | Runlvl | the zaptel hangup me |
| 15:42.35 | Runlvl | Launched AGI Script /usr/share/asterisk/agi-bin/conf-background.agi |
| 15:42.35 | Runlvl | <PROTECTED> |
| 15:42.35 | Runlvl | <PROTECTED> |
| 15:43.07 | *** part/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 15:43.28 | Runlvl | any one can help me? |
| 15:43.53 | Runlvl | guys? |
| 15:45.34 | UnixDog | ok that worked thansk |
| 15:45.38 | UnixDog | thanks |
| 15:45.55 | UnixDog | the calls are now being dialed backout |
| 15:46.05 | *** part/#asterisk masus (n=tet@88.248.73.2) |
| 15:46.19 | s34n | [TK]D-Fender: It turns out that changing the expire time fixed the problem |
| 15:46.36 | Runlvl | Strom_C, can you help me please? |
| 15:46.39 | *** join/#asterisk coppice (n=chatzill@79.193.17.210.dyn.pacific.net.hk) |
| 15:46.53 | s34n | [TK]D-Fender: * wanted 120, my itsp wanted 3600 |
| 15:46.55 | Strom_M | Runlvl: I wouldn't know where to start. |
| 15:47.30 | s34n | [TK]D-Fender: and I guess they don't auto-negotiate, so the registration failed |
| 15:47.31 | Runlvl | I made an agi for meetme | b |
| 15:47.40 | Runlvl | and when asterisk run this agi |
| 15:47.49 | Runlvl | the zaptel hang up me |
| 15:47.50 | Strom_M | Runlvl: yes, you said all that already |
| 15:47.54 | Runlvl | :( |
| 15:47.57 | Strom_M | i can't help you. |
| 15:48.03 | Runlvl | google can't help me neither |
| 15:48.27 | Strom_M | well...be patient then and eventually someone in here will know the answer |
| 15:48.49 | s34n | [TK]D-Fender: when you add a /extentodial to the register=>, how does it know which context? |
| 15:51.23 | kmchen | TK: Did you see the pastebin ? |
| 15:52.20 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:52.48 | [TK]D-Fender | s34n: it will match on your peer if possible, otherwise it takes it from [general] |
| 15:53.12 | [TK]D-Fender | kmchen: link it here and someone else can continue I won't be able to concentrate on this. |
| 15:55.22 | s34n | [TK]D-Fender: how do I set the peer match? |
| 15:55.46 | s34n | [TK]D-Fender: [host] ? |
| 15:56.09 | variable_office | is make menuselect broken? |
| 15:57.09 | kmchen | OK: my debug on a call from the LAN http://pastebin.com/d42cf7eb3 |
| 15:58.01 | [TK]D-Fender | s34n: make your peer, have the auth credientails match and hope your itsp is sending you AUTHED calls |
| 15:58.34 | kmchen | Does somebody can have a look ? |
| 15:59.08 | s34n | [TK]D-Fender: I make the peer by creating a [host] entry in sip.conf where host matches the host in the register cmd? |
| 15:59.31 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:01.24 | outtolunc | Runlvl: for determining if agi's will work properly you need to check it's permissions, make sure it is in the agi-bin dir, enable 'agi debug' and *make sure you answer() the call. |
| 16:01.43 | outtolunc | (and reload after making changes to one0 |
| 16:01.44 | outtolunc | ) |
| 16:01.53 | s34n | variable_office: broken in what way? |
| 16:02.01 | kmchen | I have already successfully installed every packages and sip and extensions.conf but I can't get authenticated by my provider (freephonie.net). Can somebody help ? |
| 16:02.17 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:02.34 | Runlvl | http://forums.digium.com/viewtopic.php?t=18226 |
| 16:02.39 | s34n | kmchen: do you have a register => command in your sip.conf? |
| 16:02.41 | tzafrir | kmchen, can you successfully register? |
| 16:02.42 | variable_office | s34n, i am trying to install odbc voicemail storage but it has XXX in it so i cant install it, but i have unixodbc installed |
| 16:03.10 | kmchen | s34n yes and sip show registry shows my asterisk definition |
| 16:03.23 | s34n | variable_office: menuselect is probably no the problem. |
| 16:03.33 | tzafrir | variable_office, obviously asterisk thinks it isn't installed |
| 16:03.44 | s34n | kmchen: does it show it registered? or unregistered? |
| 16:03.46 | outtolunc | runlvl: you need to 'answer' the call |
| 16:03.49 | tzafrir | menuselect knows it isn't, because this is the result of the last configure |
| 16:03.50 | kmchen | I receive calls and it works in internal. But cans' call out from a peer |
| 16:04.11 | s34n | variable_office: does it need headers or some such? |
| 16:04.26 | variable_office | dont think so, its the same for res_odbc |
| 16:04.50 | kmchen | keynux*CLI> sip show registry |
| 16:04.51 | kmchen | Host Username Refresh State |
| 16:04.51 | kmchen | freephonie.net:5060 0954951395 1785 Registered |
| 16:04.53 | variable_office | but i installed the exact same packages as i did for *1.2 and that worked(no menuselect though) |
| 16:05.00 | outtolunc | Runlvl: put $AGI->answer() above the 'while' |
| 16:05.11 | variable_office | iam using pgsql realtime current just fine, but that wont cut it for voicemail storage |
| 16:05.16 | s34n | kmchen: ok, then your statement that you can't get authenticated by your provider seems inaccurate |
| 16:05.21 | Runlvl | outtolunc, i try this |
| 16:05.24 | Runlvl | outtolunc, ok |
| 16:05.29 | styelz | kmachen: i had a similar issue, i solved by removing my localnet lines |
| 16:06.47 | kmchen | styelz: I need my localnetlines |
| 16:06.56 | variable_office | s34n, does that mean anything does asterisk just look in the wrong place in 1.4 and not 1.2 |
| 16:06.57 | variable_office | ? |
| 16:07.03 | styelz | if you remove them does it let you dial out though |
| 16:07.15 | Runlvl | outtolunc, nothing happen |
| 16:07.23 | styelz | just curious |
| 16:07.24 | kmchen | s34n How to configure It correctly ? |
| 16:07.24 | *** part/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 16:07.55 | outtolunc | Runlvl: then make sure you 'installed' asterisk-perl (since you are on asterisk 1.2 make sure you install the 0.08 version) |
| 16:08.05 | kmchen | styelz: I think I don't understand what you mean |
| 16:08.34 | outtolunc | http://asterisk.gnuinter.net/files/asterisk-perl-0.08.tar.gz |
| 16:09.31 | kmchen | styelz: What do you mean by remove your local lines |
| 16:10.02 | kmchen | Styelz: Remove the channels In sip.conf ? |
| 16:10.29 | styelz | nah localnet= lines. |
| 16:10.41 | variable_office | tzafrir, how can i tell were asterisk is looking for unixodbc? |
| 16:10.43 | styelz | if you had any |
| 16:11.02 | kmchen | styelz: is it a sip.conf statement ? |
| 16:11.14 | styelz | yea |
| 16:11.30 | kmchen | I don't have such statements |
| 16:11.33 | s34n | kmchen: If you can take incoming calls, your registration is ok |
| 16:11.35 | tzafrir | outtolunc, Runlvl , actually the latest asterisk-perl there is 0.10 . And it is also available in CPAN . |
| 16:11.39 | styelz | ok |
| 16:11.47 | kmchen | s34n I think so |
| 16:11.48 | tzafrir | So why get ver. 0.08? |
| 16:11.49 | s34n | kmchen: sounds like you have dialplan issues |
| 16:11.52 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 16:12.05 | s34n | kmchen: not sending outgoing calls to the right place |
| 16:12.10 | styelz | its failing auth on the reg |
| 16:12.10 | outtolunc | tzafrir: it isn't asterisk 1.2 'happy' |
| 16:12.22 | kmchen | Do you wan't to see my dialplan (5lines) |
| 16:12.25 | tzafrir | outtolunc, can you be more specific? |
| 16:12.36 | outtolunc | meaning, it was mod'd for 1.4 compat |
| 16:12.54 | tzafrir | 0.10 maybe. Surely not 0.09 |
| 16:12.54 | styelz | Sep 25 17:32:48 NOTICE[9569]: chan_sip.c:11131 handle_request_register: Registration from 'freephonie <sip:0954951395@freephonie.net>' failed for '192.168.0.10' - Username/auth name mismatch |
| 16:12.55 | outtolunc | the 0.08 works just fine with asterisk 1.2 |
| 16:13.11 | s34n | kmchen: you can pastebin it |
| 16:13.12 | tzafrir | outtolunc, but can you be more specific? |
| 16:13.36 | kmchen | OK. I give you the url in a mn. |
| 16:13.43 | outtolunc | there are changes in the originate, callerid, and more |
| 16:14.05 | outtolunc | if you want i can diff it for you <G> |
| 16:14.15 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:14.38 | s34n | [TK]D-Fender: I set a a sip.conf section where [sectionname] matches the host on the register command |
| 16:14.39 | *** join/#asterisk runlvl (n=juan@190.2.40.57) |
| 16:15.21 | s34n | [TK]D-Fender: I set the context there |
| 16:16.02 | [TK]D-Fender | s34n: it matches on the USERNAME, you shouldn't use [hostname] as the name of your peer |
| 16:16.32 | *** join/#asterisk Egonis (n=root@acid.auricnet.ca) |
| 16:16.52 | kmchen | s34: http://pastebin.com/dcf46f0c |
| 16:17.09 | Egonis | I just installed a Tormenta2 T1/E1 Card, and want to do a quick configuration in zaptel.conf -- I have never done this before, is there a quick *BASIC* howto in using this? |
| 16:17.21 | styelz | kmchen: what about this pastebin http://pastebin.com/d42cf7eb3 |
| 16:17.30 | styelz | it says it is failing Auth |
| 16:17.31 | tzafrir | Egonis, genzaptelconf |
| 16:17.33 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 16:17.33 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 16:17.34 | tzafrir | :-( |
| 16:17.45 | Egonis | tzafrir: is that an application? |
| 16:17.57 | tzafrir | xpp/utils/genzaptelconf |
| 16:18.07 | Egonis | tzafrir: just ran it, good god I wish I knew about that. :P |
| 16:18.11 | Egonis | tzafrir: Thank you! |
| 16:18.23 | tzafrir | should generate a generally working config. That is: it will certainly pass ztcfg |
| 16:18.50 | tzafrir | There's a limit to the amount of guesswork that can be done to T1 |
| 16:19.12 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 16:19.35 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 16:19.51 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 16:20.12 | kmchen | styelz: that was my debug on a call out from LAN that shows I can't get registered with my xLite on the LAN |
| 16:20.17 | Strom_M | a Tormenta2? |
| 16:20.24 | Strom_M | where'd you get that thing? |
| 16:20.48 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
| 16:20.54 | kmchen | This one shows my extensions.conf http://pastebin.com/dcf46f0c |
| 16:21.33 | Egonis | tzafrir: Is there an automatic configuration app for zaptel.conf? ;) |
| 16:21.51 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:22.21 | styelz | Egonis: genzaptelconf |
| 16:22.37 | styelz | Egonis: genzaptelconf -c <county code> |
| 16:22.55 | Egonis | styelz: niccce. <country code> would be '1' for canada? i.e. the numeric country code? |
| 16:23.04 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:23.07 | styelz | no us, au, ca |
| 16:23.13 | Egonis | styelz: nevermind, got it. Thank you! |
| 16:23.41 | Egonis | styelz: Anything for zapata? ;) |
| 16:23.52 | kmchen | s34: Did you find the pastebin ? |
| 16:24.07 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:24.14 | *** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:24.44 | *** part/#asterisk Egonis (n=root@acid.auricnet.ca) |
| 16:25.20 | styelz | Egonis: didn't it make a zapata-auto.conf |
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| 16:30.54 | styelz | McLovin |
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| 16:33.46 | citats | outtolunc: whats the problem with asterisk-perl 0.10 and asterisk 1.2? |
| 16:34.42 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:34.42 | outtolunc | probably config issues, but there was a change in how it looks at callerid (i think taken out and replaced with vars) and originate, i think you added vars... |
| 16:34.50 | outtolunc | so they aren't 'problems' |
| 16:34.50 | kmchen | s34, styelz and TK: Thanks for your help. I found out: My pass were note the same between the register and the freephonie-out channel... 5 days of turn around !!! |
| 16:35.30 | citats | outtolunc: i use it on 1.2 and 1.4 and have not had any issues |
| 16:35.30 | outtolunc | but since the person in this case 'stated' asterisk 1.2 and what what 1.2.xx i pointed that person to 0.08 'just to be sure' |
| 16:35.32 | s34n | [TK]D-Fender: thx. that works |
| 16:36.21 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 16:36.49 | styelz | kmchen: hehe nice |
| 16:37.14 | tru_`z24 | Anyone able to look at my config and tell me why asterisk doesn't pick up on the demo? http://rafb.net/p/ZB9NGJ53.html |
| 16:37.51 | mocker | Anyone in the developer training at astricon? |
| 16:38.04 | *** join/#asterisk xipi (n=oliver@212.17.244.236) |
| 16:38.07 | xipi | hi |
| 16:38.37 | davevg-btwtech | /me raises hand |
| 16:38.49 | xipi | i would like to receive calls from voip and send the caller straight to an IVR |
| 16:39.34 | xipi | when i do this, asterisk claims "everyone is busy/congested" |
| 16:39.44 | xipi | what am i doing wrong? |
| 16:40.12 | xipi | does anyone know of a good tutorial? |
| 16:40.15 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:41.01 | Deeewayne | ~thebook |
| 16:41.01 | jbot | hmm... thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:41.19 | xipi | ok |
| 16:44.21 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:45.32 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:45.35 | [TK]D-Fender | tru_`z24: how about showing us your DIALPLAN |
| 16:45.45 | tru_`z24 | its the default demo one |
| 16:45.49 | tru_`z24 | context=demo |
| 16:46.03 | [TK]D-Fender | tru_`z24: SHOW US. |
| 16:46.08 | tru_`z24 | k |
| 16:46.11 | [TK]D-Fender | tru_`z24: and that is not a dialplan. |
| 16:47.02 | [TK]D-Fender | xipi: pastebin the CLI output of a failed call at verbose 10 & SIP debug enabled |
| 16:47.05 | [TK]D-Fender | ~pb |
| 16:47.05 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:47.06 | [TK]D-Fender | ~~~~~~~~~~~~~~~~~~~ |
| 16:48.02 | tru_`z24 | [TK]D-Fender: if the demo context isn't a dial plan what is. Do you want to see my extensions.conf file? |
| 16:48.20 | [TK]D-Fender | tru_`z24: dialpe = extensions.conf |
| 16:48.23 | [TK]D-Fender | dialplan* |
| 16:48.24 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 16:48.32 | *** join/#asterisk putnopvut (n=putnopvu@65.116.224.30) |
| 16:48.34 | tru_`z24 | k |
| 16:48.37 | variable_office | what does the (E) next to the Depends on: items mean in make menuselect? |
| 16:52.02 | tru_`z24 | [TK]D-Fender: here is the extensions.conf file: http://rafb.net/p/BvPDmb84.html |
| 16:52.52 | [TK]D-Fender | tru_`z24: what do you see when a call comes in? |
| 16:53.01 | tru_`z24 | nothing at all |
| 16:53.06 | [TK]D-Fender | tru_`z24: What card? |
| 16:53.06 | tru_`z24 | I have a phone hooked up |
| 16:53.09 | tru_`z24 | x100p |
| 16:53.14 | tru_`z24 | and it rings |
| 16:53.19 | tru_`z24 | and asterisk does nothing |
| 16:53.21 | [TK]D-Fender | tru_`z24: waht do you mean "a phone hooked up"? |
| 16:53.31 | [TK]D-Fender | tru_`z24: Ok, so the JACK is OK |
| 16:53.41 | tru_`z24 | its an analog card, and has a line in, and a line out for a phone to access the pots line |
| 16:53.47 | [TK]D-Fender | tru_`z24: you 100% sure you're on the right jack on the card itself? |
| 16:54.01 | tru_`z24 | yes |
| 16:54.08 | tru_`z24 | Otherwise i wouldn't be able to use the phone |
| 16:54.12 | [TK]D-Fender | tru_`z24: enable zap debug, debug 10, verbose 10, and try another call... |
| 16:54.12 | tru_`z24 | It doesn't work backwards |
| 16:56.40 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 16:57.24 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:00.32 | xipi | [TK]D-Fender: http://pastebin.org/3307 |
| 17:00.59 | [TK]D-Fender | xipi: perhaps you didn't hear me... I said with SIP DEBUG ENABLED |
| 17:02.34 | Daviey | <PROTECTED> |
| 17:02.46 | *** part/#asterisk kmchen (n=kmchen@gar13-4-82-240-99-84.fbx.proxad.net) |
| 17:03.21 | [TK]D-Fender | Daviey: I hold "shift" for occasions like that :) |
| 17:04.05 | Daviey | heh |
| 17:04.57 | tzafrir | setxkbmap -option ctrl:nocaps |
| 17:05.18 | tzafrir | That's how it should be |
| 17:06.13 | *** join/#asterisk zm23 (n=chatzill@zaara.cuit.columbia.edu) |
| 17:06.48 | [TK]D-Fender | tzafrirYes, that will be very effective on this WinXP machine :) |
| 17:06.52 | xipi | [TK]D-Fender: are you sure? there are pages and pages. what info do you need exactly? |
| 17:07.10 | [TK]D-Fender | xipi: Yes, I want those pages and pages. I asked for them. Twice. |
| 17:07.24 | xipi | [TK]D-Fender: ok. just a moment |
| 17:08.05 | [TK]D-Fender | xipi: Thats like asking a prosecutor to see the murder weapon and he answers "its in this box with all the other evidence, do you really want to see it?" |
| 17:08.23 | zm23 | hello all. how can i edit/change sip headers in asterisk? I want to rewrite the contact header |
| 17:08.29 | mvanbaak | lol |
| 17:08.44 | mvanbaak | "nah, dont bother. If it's that much work...." |
| 17:10.11 | zm23 | i have not been able to find any information online. is it possible? |
| 17:10.51 | zm23 | to edit sip headers |
| 17:13.04 | xipi | [TK]D-Fender: http://pastebin.org/3308 |
| 17:13.23 | twisted | yay |
| 17:13.47 | [TK]D-Fender | REGISTER sip:proxy01.sipphone.com SIP/2.0 |
| 17:13.48 | xipi | [TK]D-Fender: i just didn't want to bombard you with any unnessecary stuff |
| 17:13.52 | [TK]D-Fender | SIP/2.0 401 Unauthorized |
| 17:14.53 | xipi | the line in question is the carpo line |
| 17:15.24 | [TK]D-Fender | xipi: So your registers are being refused under bad user/pass, and you are noteven ATTPTING to dial in there! |
| 17:15.59 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:16.38 | xipi | [TK]D-Fender: sorry, but could you repeat that in plain english, please? i am not a professional yet |
| 17:17.30 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 17:18.38 | dlynes_home | xipi: you've got the wrong username and/or password in your sip.conf file |
| 17:18.48 | dlynes_home | xipi: or in your sip phone/ata/gateway and/or both |
| 17:19.07 | xipi | dlynes_home: strange, that i can call out then |
| 17:19.14 | xipi | confused |
| 17:19.29 | dlynes_home | xipi: your sip device probably doesn't need you to authenticate to dial out |
| 17:19.52 | dlynes_home | xipi: but your asterisk box needs your sip device to authenticate when it's transferring calls to asterisk |
| 17:20.15 | xipi | dlynes_home: is this possible? this would mean, i could abuse any line to make "free" calls? |
| 17:20.28 | xipi | dlynes_home: hard to believe |
| 17:20.38 | dlynes_home | xipi: no...your sip device is probably using a username and password to connect to your ITSP |
| 17:21.02 | xipi | dlynes_home: no. my sip device only connects to the asterisk |
| 17:21.13 | dlynes_home | xipi: so where's your phone line then? is it analog? |
| 17:21.25 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:21.57 | xipi | i have an isdn line. but i am using voip for outbound calls |
| 17:22.19 | dlynes_home | xipi: so what's making the outbound calls? your sip device, or asterisk? |
| 17:22.32 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:22.33 | xipi | dlynes_home: asterisk |
| 17:22.49 | dlynes_home | xipi: so the sip device is a sip phone then? not an ata or a gateway? |
| 17:23.40 | xipi | dlynes_home: i have a phone, connected to an ata, which connects with an external asterisk |
| 17:23.57 | dlynes_home | xipi: ok...and do you have a static ip? |
| 17:24.08 | *** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org) |
| 17:24.57 | xipi | dlynes_home: the server - running the asterisk - has a static ip |
| 17:25.29 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 17:25.34 | dlynes_home | xipi: ok, and the asterisk box and your firewall are one and the same, correct? |
| 17:25.38 | rhombus | I'm having Polycom 501 digit map grief |
| 17:25.54 | rhombus | I can't get the phone to take a leading asterisk, as in |
| 17:25.58 | rhombus | *4 |
| 17:26.23 | dlynes_home | rhombus: if it's anything like the Aastra, you need to specify that it can take those in your dialplan |
| 17:26.26 | xipi | dlynes_home: what is the point, you are trying to get to? |
| 17:26.36 | hmmhesays | theres someone I haven't seen in awhile |
| 17:26.44 | dlynes_home | xipi: whether the asterisk box is owned by you and on the same lan as the ata, or if it's at a colo |
| 17:26.47 | *** join/#asterisk roberted (n=roberted@c-76-17-219-22.hsd1.mn.comcast.net) |
| 17:26.52 | Katty | :> |
| 17:27.10 | hmmhesays | how are you? |
| 17:27.18 | Katty | mostly..uhh... |
| 17:27.19 | Katty | hmm |
| 17:27.21 | Entr4nced | wow, you can feel the love in this channel :) |
| 17:27.22 | hmmhesays | and what is the proper way of reading stdin in C |
| 17:27.23 | xipi | dlynes_home: the ata is local, while the asterisk is remote on a vserver |
| 17:27.26 | rhombus | well, it's not an Aastra -- the Polycom's use a digit map |
| 17:27.27 | Katty | hmmhesays: i think i'm okay. |
| 17:27.31 | rhombus | which is in an XML config |
| 17:27.36 | s34n | [TK]D-Fender: and now it doesn't work |
| 17:27.37 | hmmhesays | just think, something bad going on er what? |
| 17:27.42 | rhombus | the thing I don't get is that this same digit map works on an IP 650 |
| 17:27.52 | Katty | hmmhesays: i'm considering moving to vegas or dallas. |
| 17:28.04 | dlynes_home | xipi: ok, and is the asterisk box administered by you, or is it just your voip provider that you know how to connect to, and just happen to know that it's running asterisk? |
| 17:28.06 | Katty | hmmhesays: lot going on right now, in that whole, thinking process |
| 17:28.06 | s34n | [TK]D-Fender: the * box is sending back a 404 not found |
| 17:28.15 | Katty | Entr4nced: sorta :P |
| 17:28.26 | hmmhesays | i'd pick dallas over vegas |
| 17:28.31 | xipi | dlynes_home: i am running the box. i am the admin |
| 17:28.32 | Katty | why? |
| 17:28.34 | hmmhesays | i mean.... vegas is fun but, I wouldn't want to live there |
| 17:28.49 | s34n | [TK]D-Fender: I can see the invite come into * with the proper extension |
| 17:28.54 | hmmhesays | how far is dallas from the gulf? |
| 17:28.56 | Katty | dallas is closer to mom. |
| 17:29.04 | Katty | no clue. |
| 17:29.09 | hmmhesays | why are you considering moving at all? |
| 17:29.11 | Katty | 611 miles from STL tho :P |
| 17:29.17 | Katty | because southern missouri white trash is eating my soul. |
| 17:29.20 | dlynes_home | xipi: Ok, then when the ata dials out to your asterisk box, it needs to authenticate, but when you send calls to the Ata, no authentication is needed, right? |
| 17:29.29 | hmmhesays | haha good reason |
| 17:29.42 | rhombus | Any Polycom Soundpoint users out there? |
| 17:29.52 | hmmhesays | I wouldn't move to dallas then |
| 17:29.53 | rhombus | [TK]D-Fender? |
| 17:29.59 | hmmhesays | plenty of redneckery there |
| 17:30.01 | rhombus | I wouldn't move to Dallas either |
| 17:30.01 | xipi | dlynes_home: oh. yes |
| 17:30.03 | Katty | :/ |
| 17:30.11 | Katty | i need a warm city. |
| 17:30.15 | dlynes_home | xipi: you need to set up your ATA to use authentication then |
| 17:30.15 | rhombus | if you want to get away from white trash you need to move to northern California or urban Oregon |
| 17:30.23 | Katty | nope, too cold. |
| 17:30.39 | rhombus | If that's too cold for you, you haven't experienced cold. |
| 17:30.40 | Katty | i think the desert would help my allergy problems |
| 17:30.45 | dlynes_home | xipi: if you're using a Sipura 2000/2002/2100/2102, or a PAP2, there's plenty of tutorials on forums.voxilla.org that tell you how to do it |
| 17:30.50 | rhombus | cold weather is good for allergies |
| 17:30.53 | rhombus | :) |
| 17:31.00 | hmmhesays | Katty: get a cost of living chart for sure |
| 17:31.06 | dlynes_home | xipi: just do a search through the sipura and linksys user groups on there to find some information on the subject |
| 17:31.20 | Katty | hmmhesays: okies. |
| 17:31.27 | xipi | dlynes_home: that's not the point right now. |
| 17:31.32 | dlynes_home | xipi: assuming you're using one of those devices and not a Grandstream ATA-486 or something like that |
| 17:31.53 | dlynes_home | xipi: i thought that was the whole point? that one end needed authentication, and one didn't, and you thought that was insecure? |
| 17:32.23 | xipi | dlynes_home: i was trying to send inbound calls to an aa |
| 17:32.30 | dlynes_home | xipi: aa? |
| 17:32.44 | xipi | dlynes_home: automated attendant |
| 17:32.51 | xipi | dlynes_home: ivr |
| 17:32.55 | dlynes_home | xipi: ok... |
| 17:33.03 | dlynes_home | xipi: but that has nothing to do with the ATA |
| 17:33.23 | hmmhesays | is there an easy way in C to clear stdin after a getchar()? |
| 17:33.31 | xipi | dlynes_home: of course not. |
| 17:33.36 | dlynes_home | hmmhesays: fflush(stdin) |
| 17:33.43 | xipi | dlynes_home: sorry, if i expressed myself wrong |
| 17:33.48 | hmmhesays | I read you can't flush an input stream |
| 17:34.03 | hmmhesays | and its not working for me |
| 17:34.05 | dlynes_home | hmmhesays: one sec...lemme verify that...i'm pretty sure you can |
| 17:34.36 | hmmhesays | i'm brushing up on my C stuffs |
| 17:35.35 | dlynes_home | xipi: so what exactly is it that you're having trouble with, with the ivr, then? |
| 17:37.15 | dlynes_home | hmmhesays: yeah...you're right |
| 17:37.34 | xipi | an inbound call hits the asterisk and is supposed to go to a voice menu. but there is this "all lines busy" error message |
| 17:37.35 | hmmhesays | dlynes_home: which leaves me with a dilema |
| 17:37.38 | *** join/#asterisk rpm (n=russell@75.153.43.151) |
| 17:37.45 | rpm | does canada have an operator toll-free number? basically it does source dn routing of your number to the correct operator group? |
| 17:38.05 | rhombus | rpm: can you elaborate? |
| 17:38.06 | dlynes_home | hmmhesays: Just do a loop on getchar(), until it returns with EOF |
| 17:38.33 | xipi | dlynes_home: i am afraid, we'll have to leave it for later. i need to rush out. otherwise my wife will get mad at me |
| 17:38.39 | rpm | can i dial like 1-800-555-1212 lets say, and it route me to the correct operator based upon the number i am calling. |
| 17:38.43 | hmmhesays | yeah I'll have to do that |
| 17:38.45 | xipi | dlynes_home: hope to see you later |
| 17:38.55 | hmmhesays | this C book is completely wrong |
| 17:39.30 | hmmhesays | too bad there is no chomp() like in perl |
| 17:42.43 | *** join/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 17:42.49 | supjigator | Hi! |
| 17:45.32 | hmmhesays | hello |
| 17:45.45 | arekm | what's impedance line for EU analog phones? I found 750 ohms + 270 ohms ( European harmonized, France Telecom & Telefonica) just why two separate impedances? |
| 17:45.51 | *** part/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 17:45.53 | *** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org) |
| 17:45.59 | rhombus | rpm: it works the same everywhere |
| 17:46.13 | rhombus | that's NANP convention |
| 17:47.05 | supjigator | Are there any good solutions for virtual hosting asterisk like with tenants? |
| 17:47.20 | hmmhesays | buy a dedicated server |
| 17:47.23 | hmmhesays | they're cheap |
| 17:47.51 | hmmhesays | you can get a p4 with 500gb of transfer for 60/month |
| 17:48.39 | dlynes_home | hmmhesays: what does your book tell you? |
| 17:48.45 | supjigator | I have a large SIP gateway and want to sell virtual pbx's for the 1-5 extension market. |
| 17:48.51 | hmmhesays | tells me to use fflush() on stdin |
| 17:48.54 | dlynes_home | hmmhesays: another thing you can do too, is fgets(stdin)/gets |
| 17:49.20 | dlynes_home | hmmhesays: well, I think fflush(stdin) will work in Windows, DOS and OS/2....just not POSIX |
| 17:49.43 | hmmhesays | actually it tells me to use fpurge(stdin); on the *nix platforms |
| 17:49.45 | dlynes_home | hmmhesays: I'm pretty sure I used it for that back in my DOS and OS/2 days |
| 17:49.58 | dlynes_home | hmmhesays: fpurge? never heard of that function call |
| 17:50.04 | hmmhesays | yeah, this buck sucks |
| 17:50.23 | hmmhesays | my spelling as well |
| 17:51.30 | dlynes_home | hmmhesays: ah...seems to be a bsd/solaris only call, but yet it still appears in my linux headers and linux man pages |
| 17:51.57 | dlynes_home | hmmhesays: it's non-portable, however; getchar() is portable |
| 17:52.24 | vader-- | is there a way to upgrade 1 cisco phone's firmware without having to upgrade everyone's? |
| 17:52.35 | vader-- | my phones pull their firmware from the asterisk server |
| 17:52.48 | vader-- | i just want to test out the new firmware on my phone |
| 17:53.14 | _x86_ | vader--: don't they ask for MACADDRESS.cfg like the polycom phones do? |
| 17:53.51 | hmmhesays | that they do |
| 17:53.55 | _x86_ | vader--: another option is setting up an additional VLAN for testing, and have the TFTP server on that VLAN only serve the newer firmware |
| 17:54.11 | _x86_ | put the phone on the same VLAN and voila ;) |
| 17:57.20 | *** join/#asterisk BugKhaM (i=HaMYaI@125-25-207-36.adsl.totbb.net) |
| 17:57.57 | *** join/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es) |
| 17:58.22 | BugKhaM | apart from cmd Congestion? can we simulate any other status like 'BUSY', 'CANCEL'? |
| 17:58.36 | BockBilbo | good evening |
| 17:58.46 | BugKhaM | without using cmd Dial |
| 18:00.22 | BockBilbo | Is it possible to access an user vm name wav file from extensions.conf? |
| 18:00.32 | *** join/#asterisk tripps (n=ss@c-76-31-153-101.hsd1.tx.comcast.net) |
| 18:01.02 | hmmhesays | BugKhaM: with Busy() maybe? |
| 18:01.10 | *** join/#asterisk Cresl1n (n=matt@65.116.224.30) |
| 18:01.11 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 18:01.20 | hmmhesays | BockBilbo: yes |
| 18:01.41 | BockBilbo | hmmhesays, i know there is an option called ${VM_NAME} |
| 18:01.49 | BockBilbo | but can't access to it directly |
| 18:02.10 | BockBilbo | im looking at voip-info.org but with no answer yet |
| 18:03.02 | hmmhesays | what are you looking to accomplish exactly |
| 18:04.47 | BockBilbo | well, i just want the voicemail to say "you've called "+VM_USER_NAME (if exist) or "you've called"+theCalledNumber(using the digits voice files) |
| 18:05.28 | hmmhesays | app_voicemail does that by default |
| 18:05.38 | hmmhesays | if no greeting exists it reads the voice mail box number |
| 18:05.43 | BockBilbo | right |
| 18:05.53 | BockBilbo | but i don't want to use a personal greeting |
| 18:05.56 | hmmhesays | isn't that what you are looking for? |
| 18:06.00 | BockBilbo | nop |
| 18:06.02 | BockBilbo | one sec |
| 18:06.05 | BockBilbo | let me show you |
| 18:06.53 | BugKhaM | hmmhesays: Busy seems to work, anything else? |
| 18:07.07 | hmmhesays | BugKhaM: show applications on the cli |
| 18:07.20 | BockBilbo | http://pastebin.com/d23df5a9f |
| 18:07.53 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 18:08.06 | BockBilbo | right before 7,3, im wishing to check if the user has configured his VM_NAME and if so, i would play it instead of playing the whole number |
| 18:08.08 | hmmhesays | BockBilbo: that is what you want to accomplish? first of all use Saydigits |
| 18:08.22 | BockBilbo | ok |
| 18:08.56 | BockBilbo | yes, i was thinking on accomplishing it that way unless you tell me any better way |
| 18:09.13 | hmmhesays | BockBilbo: you need to STAT for a greeting in the voicemail directory |
| 18:09.54 | hmmhesays | if it returns true then you can play it, if false SayDigits(5552331234); etc |
| 18:10.09 | hmmhesays | oh boy monopoly is back at mcdonalds |
| 18:10.10 | hmmhesays | time for me to get fat@! |
| 18:10.39 | *** join/#asterisk hfb (n=hfb@pool-71-106-223-11.lsanca.dsl-w.verizon.net) |
| 18:10.40 | BockBilbo | hehe |
| 18:10.40 | BockBilbo | xD |
| 18:10.45 | BockBilbo | thanks hmmhesays |
| 18:10.54 | hmmhesays | show function STAT on the cli |
| 18:11.08 | *** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca) |
| 18:11.14 | hmmhesays | I use something similar for my fscking awesome privacy manager |
| 18:11.25 | BockBilbo | :) |
| 18:11.29 | BockBilbo | and |
| 18:11.37 | BockBilbo | what would be the best way to play the file? |
| 18:11.39 | DeeJayTwo | Hi... we upgraded from 1.4.3 to 1.4.11 and when dialing a user extension, we hear a ring tone with a lot of noise... |
| 18:11.40 | BockBilbo | PlayBack ? |
| 18:11.45 | DeeJayTwo | what could explain this? |
| 18:11.50 | *** join/#asterisk ta^3 (n=tacvbo@65.116.224.30) |
| 18:12.14 | hmmhesays | depends on if you want to caller to be able to skip the greeting and go straight to the beep or not |
| 18:12.30 | BockBilbo | mmm |
| 18:12.54 | BockBilbo | is there a way to skip everything by dialing a number and make a "secret" phone call |
| 18:13.22 | hmmhesays | http://www.youtube.com/watch?v=rFxwOLbImcU <-- easily my favorite youtube video today |
| 18:13.36 | BockBilbo | lets say that by default the VM starts, and that if the caller press 0, instead of leaving a message, a call is made to a certain SIP user |
| 18:13.48 | hmmhesays | of course |
| 18:13.56 | BockBilbo | what function should i look on? |
| 18:14.05 | hmmhesays | func pay me money |
| 18:14.10 | BockBilbo | xDDDD |
| 18:14.37 | BockBilbo | alrite :/ |
| 18:14.41 | BockBilbo | thanks a lot anyway |
| 18:14.42 | BockBilbo | :) |
| 18:14.42 | hmmhesays | background, read, |
| 18:14.56 | hmmhesays | you're looking for user input there are many ways to gather it |
| 18:15.22 | s34n | when I send out a REGISTER request I am getting back an ICMP that says the port is unreachable |
| 18:15.30 | BockBilbo | thank you very much :) Ill google about it |
| 18:15.55 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 18:15.58 | generalhan | hey all ! |
| 18:16.02 | hmmhesays | asterisk cli is as good a place as any to get information |
| 18:16.06 | hmmhesays | generalhan |
| 18:16.12 | generalhan | hmmhesays ! |
| 18:16.27 | hmmhesays | i will post this again, because you missed it |
| 18:16.29 | hmmhesays | http://www.youtube.com/watch?v=rFxwOLbImcU |
| 18:16.39 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 18:16.51 | BockBilbo | thanks hmmhesays ill look on the cli |
| 18:17.40 | generalhan | hmmhesays: HAHAHA |
| 18:17.50 | hmmhesays | LOL |
| 18:17.53 | hmmhesays | yeah, that made my day |
| 18:17.57 | hmmhesays | BockBilbo: np |
| 18:19.53 | generalhan | i got some advice here yesterday to use call-limit in sip.conf as a way to disable the callwaiting on a phone that has, by default, 3 lines. that worked perfectly for all calls comming into that phone from the outside. but anyone from in here can still transfer calls to that phone. any ideas on how to stop this from happening ? |
| 18:22.22 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 18:22.25 | hmmhesays | even if you have call-limit=1 ? |
| 18:22.36 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 18:22.36 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 18:22.43 | generalhan | hmmhesays: yea |
| 18:23.03 | hmmhesays | how are you transfering to that phone? |
| 18:23.38 | generalhan | i have call-limit=1, if that person is on the phone and a call comes from the outside it goes RIGHT to voicemail. but if they are on a phone and someone from the inside blind transfers a call to that phone it rings right through |
| 18:23.54 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 18:24.51 | dijungal | hello... if i need to implement agents accepting inbound calls from a queue which is better to use static or dynamic agents? |
| 18:26.58 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 18:27.44 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:27.57 | AndrewGearhart | afternoon folks. Does anybody know-of/use a web based application similar to HudLite from Fonality? |
| 18:28.12 | jameswf | can anyone telll me how one touch monitor generates its unigue id |
| 18:28.17 | jameswf | *unique |
| 18:28.20 | hmmhesays | generalhan: need to see some cli output goodness |
| 18:28.32 | hmmhesays | theres always FOP |
| 18:28.39 | hmmhesays | or write something of your own |
| 18:28.56 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 18:28.59 | generalhan | hmmhesays: ill see what i can generate ... AFTER lunch ! lol. im starving ! |
| 18:29.10 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 18:29.11 | outtolunc | jameswf it used to be the uniqueid reported with - vice . |
| 18:29.12 | hmmhesays | yeah I'm drinking a rockstar right now being hungry |
| 18:29.17 | hmmhesays | thank god its taco tuesday |
| 18:29.21 | generalhan | hahaha |
| 18:29.27 | rob0 | TT? Where? |
| 18:29.28 | generalhan | im doing taco tuesday too ! |
| 18:29.37 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 18:29.42 | hmmhesays | rob0: taco johns man |
| 18:29.46 | hmmhesays | are they nation wide? |
| 18:29.51 | deeperror | not here |
| 18:30.03 | generalhan | hmmhesays: nah, taco tuesday is at the Tavern on Mill here !!! |
| 18:30.14 | hmmhesays | haha : i see |
| 18:30.17 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:30.17 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:30.22 | generalhan | ok brb |
| 18:30.32 | rob0 | Unknown host www.tacojohns.com |
| 18:30.57 | hmmhesays | what? |
| 18:30.58 | hmmhesays | http://www.tacojohns.com/ |
| 18:31.02 | hmmhesays | works fine |
| 18:31.11 | rob0 | hmmm, I says. |
| 18:31.35 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 18:32.40 | rob0 | www.tacojohns.com. 5860 IN A 199.236.122.39 |
| 18:33.10 | rob0 | DNS not working on laptop. |
| 18:34.02 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 18:34.30 | deeperror | any clues why i would have different results dialing out on a sip peer when using zap vs sip client? It seems i have authentication issues when using my zap phones and when using sip phones it works fine |
| 18:34.51 | jameswf | is onetouch record asterisk or freepbx |
| 18:35.24 | lirakis | can any one reccomend a cdr reporting package? .. or do you all write your own stuff? .. ive used the old "asterisk call data analyzer" .. but its .. got no real reporting in it |
| 18:35.34 | [TK]D-Fender | jameswf: Asterisk |
| 18:35.53 | [TK]D-Fender | jameswf: Whose configuration that sets this is made by FreePBX but you can do just as well yourself. |
| 18:36.04 | lirakis | particularly for a queue based system.. im still having problems trying to figure out how to report accurate call volumes etc because of the forked cdr's |
| 18:36.26 | jameswf | From looking it appears to call monitor but I am not sure where |
| 18:37.17 | jameswf | I am of the opinion ARI sucks |
| 18:37.22 | davevg-btwtech | lirakis: If it is queue stats you are after, check out the queue_log |
| 18:37.46 | [TK]D-Fender | jameswf: features.conf "automo" <--- and "show application dial" <- to show you which Dial options you need to set WHO can initiate recording |
| 18:38.01 | *** join/#asterisk logyati (n=logyati@201.51.217.230) |
| 18:38.04 | logyati | hello guys |
| 18:38.20 | logyati | im studying voip |
| 18:38.28 | logyati | i know there is openser and asterisk |
| 18:38.38 | lirakis | davevg-btwtech: its not queue stats.. its general cdr and call reporting |
| 18:38.39 | logyati | but why would i use asterisk instead openser for sip? |
| 18:38.47 | Strom_M | hmmhesays: http://www.wittandwisdom.com/home/2003/01/taco_tuesday.html |
| 18:38.49 | logyati | im asking your opinion |
| 18:39.03 | jameswf | Thats not the issue, the issue is it generates a unique id that differs greatly in size and name from what cdr grabs |
| 18:39.10 | lirakis | logyati: asterisk is "class 5" .. openser is more for "class 4" |
| 18:39.15 | Strom_M | asterisk is a PBX, openser is a SIP proxy |
| 18:39.21 | jameswf | ARI doesnt understand the files |
| 18:40.25 | logyati | hmmm |
| 18:40.25 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:40.41 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 18:41.06 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 18:41.15 | logyati | well i use asterisk for sip here |
| 18:41.22 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 18:41.27 | lirakis | whoops |
| 18:41.35 | logyati | i couldnt find anything i do with openser that i couldnt do with asterisk |
| 18:41.40 | dijungal | which is better to use agentlogin or agentcallbacklogin ? |
| 18:42.18 | arekm | signal-type:----> fxsloopstart fxsgroundstart fxsloopstartfd fxsgroundstartautomatic fxsgroundstartimmediate fxsdnloopstart fxsdnloopstartfd fxsdngroundst |
| 18:42.20 | logyati | if i can call using sip, see videoconferences and exchange messages |
| 18:42.25 | *** join/#asterisk Rants (i=Rants@ool-182f4d05.dyn.optonline.net) |
| 18:42.26 | logyati | why would i use openser? |
| 18:42.28 | [TK]D-Fender | logyati: SER is a PROXY and has other application servers and functionality as well. SER is good for load balancing, LCR, billing, etc. * is not. * is better in larger cases as a mere IVR / VM / Application server and leaving managing SIP users to SER which is far more robust |
| 18:42.34 | arekm | fxsloopstartfd == loopstart with forward disconnect == kewlstart I assume |
| 18:42.44 | arekm | but what's the rest? fxsdnloopstart for example? |
| 18:42.50 | logyati | hmmm |
| 18:42.52 | arekm | who has an idea, speak :) |
| 18:43.01 | Rants | I'm looking for a good unlimited DID->sip with multiple channels, anyone have any ideas for a good provider? |
| 18:43.01 | fetcher | <PROTECTED> |
| 18:43.17 | fetcher | what would cause "Not Acceptable Here"? Codec mismatch? |
| 18:43.19 | logyati | [TK]D-Fender, you mean, * cant do accounting |
| 18:43.40 | [TK]D-Fender | logyati: No, I'm just saying it sucks for it :) |
| 18:43.40 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 18:43.56 | [TK]D-Fender | fetcher: Yes |
| 18:44.21 | Rants | anyone have any info? |
| 18:44.21 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 18:44.23 | [TK]D-Fender | fetcher: If you want a better idea, enable sip debug and pastebin a call |
| 18:44.39 | logyati | [TK]D-Fender, hmmm, now i understood... but the accounting magic in openser is done by radius... does * has radius integration? |
| 18:44.45 | jmls | could someone look at a dialplan for me (http://www.pastebin.ca/712312) and set me straight on the questions I pose there ? |
| 18:44.52 | jmls | I would appreciate it |
| 18:44.56 | [TK]D-Fender | loyI think ABE does, but not the public version. |
| 18:45.37 | fetcher | [TK]D-Fender: can sip debug be restricted to show only a single SIP endpoint? It's otherwise hard to separate one call out of the stream on a busy system |
| 18:46.25 | [TK]D-Fender | jmls: [Why CONGESTION ? Q931 gives 21 as AST_CAUSE_CALL_REJECTED, AST_CAUSE_NORMAL_CIRCUIT_CONGESTION IS 34] <- 34 is used in some places to actually report that the DESTINATION phone # is "busy" . |
| 18:48.14 | fetcher | do some Polycom phones ship without G.729 support? |
| 18:49.06 | [TK]D-Fender | fetcher: No, they all have it |
| 18:50.03 | fetcher | [TK]D-Fender: yeah, that's what I thought. Yet the "Not Acceptable Here" issue was resolved when I changed 'allow=g729' to 'allow=all' in sip.conf. Strange. |
| 18:50.12 | *** join/#asterisk saftsack (n=saftsack@pD9E05A91.dip.t-dialin.net) |
| 18:50.13 | jmls | [TK]D-Fender: I have looped my 2 channels on the PRI card so the "some places" is actually my own, and I am setting the PRI_CAUSE to be 21 |
| 18:50.33 | jmls | I dial out on span 1, come in on span 2 |
| 18:50.36 | [TK]D-Fender | fetcher: And do you HAVE G.729 licenses available on your *? And do the codec on BOTH sides of your call match? |
| 18:51.11 | [TK]D-Fender | jmls: Ok, I've never tried to cheat the reporting back.... |
| 18:52.01 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:52.04 | jmls | I was just trying to simulate "real world" calls where I get some busy, some no answer, and some out of order. It's part of a general testing script for asterisk that I am trying to write |
| 18:52.06 | mcab | fetcher: what model of Polycom? IP4000s support G.729, but ship with it disabled by default (IIRC) |
| 18:54.26 | fetcher | mcab: it's an IP 501. I haven't physically seen it, though... a customer picked one up on his own, but set it to pull a known-good config from my FTP server. Could something misconfigured in the boot-parameters screen kill G.729? |
| 18:54.38 | hmmhesays | i'm liking eyebeam |
| 18:55.21 | mcab | fetcher: no, you can only change the codec prefs via the config files |
| 18:55.28 | *** join/#asterisk teknoprep (n=eangstad@unaffiliated/teknoprep) |
| 18:55.30 | fetcher | thought as much |
| 18:55.30 | teknoprep | does anyone know an application that shows caller id on the computer screen when an incomming call in inbound? |
| 18:57.38 | Yourname`` | Sep 25 14:52:44 WARNING[3064892304]: chan_sip.c:12253 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"6198145226" <sip:a4322sybvag2Epb0@64.34.139.76>;tag=as4f81c8ea' |
| 18:57.55 | Yourname`` | Trying to use a callfile, and this is what the error is.. |
| 18:57.55 | fetcher | teknoprep: FOP (Flash Operator Panel), although it would be kinda clunky in that role. |
| 18:58.03 | Yourname`` | I checked the sip.conf and passwords are correct! |
| 18:58.05 | teknoprep | yeah i was thinking of a windows app |
| 18:58.35 | teknoprep | that would check the manager |
| 18:58.40 | *** join/#asterisk jsaunders (n=super@S0106006008145635.vs.shawcable.net) |
| 18:58.45 | [TK]D-Fender | Yourname``: its formatted wrong somehow. |
| 18:59.03 | [TK]D-Fender | fetcher: pastebin a call attempt with SIP debug enabled and pastebin their entries from sip.conf |
| 18:59.04 | jsaunders | Question: How do you determine average number of channels at any given point in time, ever? |
| 18:59.16 | Yourname`` | [TK]D-Fender: You mean the callfile is formatted wrong? Meaning dosformat or what? |
| 18:59.30 | Strom_M | jsaunders: you mean besides using the Erlang Formula? |
| 18:59.33 | [TK]D-Fender | jsaunders: at a point in time isn't an agerave, its a COUNT, and you'd do that via "show channels" |
| 18:59.43 | [TK]D-Fender | average* |
| 18:59.45 | [TK]D-Fender | aklshdsfjhdsklfdg |
| 18:59.55 | jsaunders | Yeah, I figured I was wrapping my brain around it wrong. |
| 19:00.01 | [TK]D-Fender | Yourname``: Meaning your CHANNEL line. |
| 19:00.03 | jsaunders | Erlang you say, let me have a look at that. |
| 19:00.14 | fetcher | jsaunders: could write a perl script to parse through /var/log/asterisk/cdr-csv/Master.csv ... |
| 19:00.33 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:00.34 | Yourname`` | [TK]D-Fender: channel: SIP/11231231234@gafachi this is h ow I have it |
| 19:01.08 | jsaunders | And what... generate a giant column (inc on call started, dec on call stopped) then average? |
| 19:01.14 | [TK]D-Fender | Yourname``: Sure doesn't match your error... of course this doesn't tell me that its not the OTHER end of your call-file thats makeiong the bad call.. |
| 19:01.36 | Yourname`` | [TK]D-Fender: Let me PB it. |
| 19:02.12 | fetcher | jsaunders: yup. A bit tedious, though. Probably best to convert the timestamps to an integer timeval() (unix time) first |
| 19:02.32 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 19:02.38 | fetcher | timeval -> time_t |
| 19:02.52 | jsaunders | Good thinkin'. :D |
| 19:03.12 | *** join/#asterisk Seldon75 (n=chatzill@69.77.161.2) |
| 19:03.25 | Yourname`` | [TK]D-Fender: http://pastebin.ca/713153 |
| 19:03.56 | Seldon75 | Hi all, we have a Digium TDM2400 card currently, and we want to install a Sangoma card instead. WHat will be involved to install this (other than physically switching the cards)? |
| 19:05.17 | jmls | coffee ? ? Pah. Need beer now |
| 19:05.20 | [TK]D-Fender | Yourname``: And your sip.conf? |
| 19:05.24 | fetcher | jsaunders: if you only want to track activity from now forward, another option is plotting channels-in-use with something like mrtg |
| 19:05.31 | wishes | jmls: well its too early for beer here |
| 19:05.39 | wishes | just past 7am |
| 19:05.41 | jmls | it's late for beer here ;) |
| 19:05.43 | [TK]D-Fender | Seldon75: Setting up the wanpipe drivers and init script. Not a big deal. |
| 19:05.50 | wishes | date? |
| 19:06.00 | jmls | 20:05 |
| 19:06.01 | Seldon75 | [TK]D-Fender: thanks! any references I can use? |
| 19:06.07 | fetcher | have a cron job do 'asterisk -rx "show channels"' and extract the count every minute, or every 5 min |
| 19:06.08 | wishes | oh thats not too late for beer! |
| 19:06.19 | jmls | uh huh |
| 19:06.24 | [TK]D-Fender | Seldon75: its all on Sangoma's WIKI |
| 19:06.30 | Seldon75 | coolio |
| 19:07.05 | fetcher | and once enough data accumulates, mrtg will automatically compute the average over a week / month / year |
| 19:07.05 | jmls | jeez that's a lot to put on the label :) |
| 19:07.32 | jmls | oh, forgot the "Brewers Since 1876" |
| 19:08.00 | Yourname`` | [TK]D-Fender: http://pastebin.ca/713426 |
| 19:08.35 | wishes | jmls: ever seen the labels on guinness cans? |
| 19:08.42 | wishes | now those are funny :D |
| 19:09.01 | rob0 | No taco Tuesday for me .... sigh |
| 19:09.07 | Strom_M | rob0: I've got you beat |
| 19:09.12 | [TK]D-Fender | Yourname``: ok, thats not working so great, verify your peer manually. |
| 19:09.17 | Strom_M | 235.32 miles! |
| 19:10.16 | jmls | wishes: heh |
| 19:10.19 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:10.22 | rob0 | Home is near Flo., Ala, nearest one's in Murray, KY. |
| 19:10.34 | Yourname`` | [TK]D-Fender: Ok. |
| 19:10.42 | rob0 | ~130 miles |
| 19:10.49 | jmls | wishes: nope - I only ever drink Guiness when I'm in Ireland, and in a country pub. |
| 19:11.19 | Yourname`` | Another question, I upgrade 1.2 to 1.4, and now I'm h aving one way audio? LOL.. called party can hear agents, but agents cannot hear called party |
| 19:11.21 | deeperror | anyone have a link to the list of changes between 1.2 and 1.4? |
| 19:11.40 | [TK]D-Fender | deeperror: read all the docs that come with 1.4 tarball. |
| 19:12.12 | wishes | jmls: well,... if that was me id never drink it :/ |
| 19:12.56 | wishes | im just trying to figure out why the previous admin compiled asterisk rather than using the debs, and what he compiled in (addons, flags etc) so i can upgrade |
| 19:13.15 | wishes | would there be any major reason a debian package wouldnt do what a compiled version would ? |
| 19:13.44 | Yourname`` | Any help please? |
| 19:13.49 | [TK]D-Fender | wishes: Because the Debian packages are triassic or jurassic (I forget which) |
| 19:14.06 | [TK]D-Fender | Yourname``: You didn't show me your MANUAL test.... |
| 19:14.07 | jm|home | :) |
| 19:14.20 | jm|home | wishes: for trassic read 'stabeish' |
| 19:14.33 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 19:14.54 | wishes | cant be worse than working with a compiled version of 1.2 with *^%*( addons/flags/etc |
| 19:15.13 | jm|home | no addons here, either |
| 19:15.31 | jm|home | v1.4 on sid/lenny on a Soekris box |
| 19:16.17 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:16.44 | wishes | mmm actually, i lie, i just found a breif mention in the docs why it was compiled |
| 19:17.02 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:17.09 | jm|home | sure. The changlog should explain things |
| 19:17.24 | wishes | "The ZAP-HFC cards use the zapata interface in asterisk (as do the digium isdn cards) however, your version of asterisk as to be patched to use them. " |
| 19:17.31 | wishes | bingo |
| 19:17.43 | wishes | "Unfortunately the above patches are unlikely to ever make it into Asterisk, as Digium require you to sign over the copyright for the code to them, and kapejod (the Junghanns developer) is unwilling to do that." |
| 19:17.45 | jm|home | tada.wav |
| 19:17.49 | wishes | :D |
| 19:18.09 | jm|home | Soft Cell -- Tainted Kernel.mp3 |
| 19:18.24 | wishes | it also says (how out of date is this) that buying the nicer hardware wasnt cost-effective option until we had 8 lines or more |
| 19:18.44 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 19:23.15 | dijungal | will the agents module be deprecated in asterisk 1.4 ? |
| 19:24.01 | syzygyBSD | dijungal: it isn't a "will" question anymore. It is a is it deprecated |
| 19:24.38 | syzygyBSD | I don't use 1.4 so I don't really know, but since it is the standard now.. |
| 19:25.37 | dijungal | is it? |
| 19:25.43 | dijungal | k |
| 19:26.36 | dijungal | i'm using 1.2, and i'm wondering if I upgrade what will happen to my current Queue -> agents -> channel configuration |
| 19:26.58 | wishes | hah i love funny documentation "user.support (also permission to write flags). To give read permissions to jamesbond, do the following:" then has a nice long example giving jamesbond permissions |
| 19:28.28 | rob0 | chmod 007 ~jamesbond |
| 19:28.42 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:28.46 | jm|home | rob0: lol :) |
| 19:28.58 | jm|home | he is somewhat private, like that |
| 19:29.03 | dijungal | lol |
| 19:29.12 | rob0 | He's never $HOME |
| 19:29.19 | rob0 | it's a trap |
| 19:31.12 | *** join/#asterisk guillote_GNU (n=bancaria@host35.201-253-17.telecom.net.ar) |
| 19:32.08 | wishes | haha |
| 19:34.13 | *** join/#asterisk BockBilbo (n=Unai@eu85-84-62-227.clientes.euskaltel.es) |
| 19:34.35 | deeperror | why when dialing out would i get 407 authentication failure and then on my second dial attempt it works 100% of the time? |
| 19:35.24 | *** join/#asterisk dfriend (i=dfriend@69.89.174.2) |
| 19:35.33 | BockBilbo | one last question... when doing Playback(/var/spool/asterisk/voicemail/buzondevoz/user/greet.gsm) on a dialplan for playing an user greet.gsm file, i get an error saying that the file doesnt exist in any format... which is false, cause the file does exist... am i doing something wrong? |
| 19:36.19 | *** join/#asterisk soulfreshner (n=D@dsl-243-22-02.telkomadsl.co.za) |
| 19:36.39 | *** join/#asterisk rudholm (i=rudholmm@nat/yahoo/x-5c3b210caae55255) |
| 19:37.10 | soulfreshner | I just set up an asterisk server for a client, and he has some serious echo when trying to phone on a pstn line |
| 19:37.42 | soulfreshner | I've tried compiling zaptel with the different echo cancellation options, but it doesn't seem to work that nicely at all |
| 19:37.57 | rudholm | you're using an analog FXO port |
| 19:38.01 | rudholm | ? |
| 19:38.12 | soulfreshner | rudholm, yes - unfortunately |
| 19:38.17 | rudholm | yeah, they suck, I fear |
| 19:38.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:38.21 | dijungal | what does this mean "iax_poke_peer: Still have a callno..." |
| 19:38.31 | [TK]D-Fender | BockBilbo: Yes, you are not allowed to specify the EXTENSION in your filename for Playback / background, etc... |
| 19:38.32 | rudholm | my TDM400 was useless, the TDM800 is significantly better, but still sucks. |
| 19:38.38 | *** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com) |
| 19:38.46 | [TK]D-Fender | Sangoma A200d FTW |
| 19:39.00 | soulfreshner | rudholm, so there is not much of an option other than replaceing the card? |
| 19:39.32 | BockBilbo | great [TK]D-Fender |
| 19:39.35 | BockBilbo | thanks! |
| 19:40.15 | soulfreshner | [TK]D-Fender, is that what you are using (A200d)? |
| 19:40.21 | *** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com) |
| 19:40.23 | [TK]D-Fender | soulfreshner: fOR ANALOG, YES |
| 19:40.49 | [TK]D-Fender | soulfreshner: All but 2 installs worldwide I have been involved with necessitated replacement with those... |
| 19:40.52 | soulfreshner | [TK]D-Fender, and it handles the analog nicely? |
| 19:40.57 | dijungal | rudholm: i'm getting allot of HDLC T1 errors on the TE410P |
| 19:41.14 | [TK]D-Fender | soulfreshner: perfectly. No echo, crackle, gains issues, etc |
| 19:41.15 | soulfreshner | handles echo nicely I mean... |
| 19:41.19 | soulfreshner | cool |
| 19:41.48 | soulfreshner | you're in canada, though right? pstn the same as here in SA? |
| 19:41.59 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-187-88.dsl.irvnca.pacbell.net) |
| 19:42.02 | [TK]D-Fender | soulfreshner: AFIAK |
| 19:42.12 | UnixDog | ok I need input |
| 19:42.15 | soulfreshner | I mean - if it works there, it should work here... ah - nice |
| 19:42.21 | UnixDog | http://pastebin.ca/713905 |
| 19:43.36 | [TK]D-Fender | UnixDog: -- Executing [_X.@mexico:1] Dial("SIP/64.192.196.12-b760e480", "VD-OUT|00_X.||") in new stack <---- I really don't know what this is going to take... YOUR DIAL HAS NO BLOODY TECH IN IT!! |
| 19:43.52 | UnixDog | its making it to the dial point but its not dialing |
| 19:44.11 | [TK]D-Fender | UnixDog: Correct, because it has no TECH IN IT. |
| 19:44.38 | [TK]D-Fender | UnixDog: You have completly failed to grasp the concept of channels in Asterisk. |
| 19:44.53 | [TK]D-Fender | ~book |
| 19:44.54 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 19:44.55 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:44.56 | Corydon76-dig | He probably meant ${VD-OUT} |
| 19:45.05 | Corydon76-dig | and just forget to deref the variable |
| 19:45.22 | UnixDog | I forgot the SIP/ |
| 19:45.30 | UnixDog | my bad I just saw that |
| 19:46.00 | [TK]D-Fender | UnixDog: And your parameters, # to dial, etc are also all bad in there |
| 19:46.10 | [TK]D-Fender | UnixDog: "Show application dial" <------------- |
| 19:47.48 | nDuff | I've got two * boxen with a T1 line between them, Sangoma cards on both ends. One is running pri_cpe, the other pri_net. They can initiate calls to each other, but audio is only moving in one direction. What should I look into? |
| 19:48.17 | [TK]D-Fender | nDuff: what are you using to actually place the call? |
| 19:48.33 | [TK]D-Fender | nDuff: you need to debug each endpoint independantly |
| 19:48.37 | tzanger | interesting |
| 19:48.46 | tzanger | I am putting 0xaa on the line, and zaptel's showing 0xed |
| 19:48.48 | nDuff | [TK]D-Fender: most recently, just "originate Zap/g1/5128747665 application Read testvar" |
| 19:48.51 | tzanger | I wonder if it's a slin->ulaw thing |
| 19:49.11 | nDuff | [TK]D-Fender: ...reports no input received, even if I'm punching buttons on the phone. |
| 19:49.24 | [TK]D-Fender | nDuff: that assumes OSS is OK.. Thats a crappy test |
| 19:49.38 | [TK]D-Fender | nDuff: Get "real", k? :) |
| 19:49.41 | nDuff | [TK]D-Fender: the real client here is iaxmodem; enabling its support for recording the audio, it's very clear what's going on. |
| 19:50.17 | nDuff | [TK]D-Fender: OSS? eh? You're not talking about the sound library and kernel modules, are you? what do those have to do with this? |
| 19:50.25 | [TK]D-Fender | nDuff: Trying to make some sort of Fax multiplexer/etc? |
| 19:50.40 | [TK]D-Fender | nDuff: Originate = CONSOLE = soundcard on PC. |
| 19:50.46 | nDuff | [TK]D-Fender: yup. already works great on one server, but I'm trying to build a second one (and use the first, which is a working production system, to test it) |
| 19:50.58 | [TK]D-Fender | nDuff: so as I said.. a SHITTY test :) |
| 19:51.05 | [TK]D-Fender | nDuff: Try harder... |
| 19:51.09 | nDuff | [TK]D-Fender: eh? no, Originate doesn't use the sound card in that usage. |
| 19:51.11 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:51.43 | [TK]D-Fender | nDuff: Really? Whats on the other side of the call then? |
| 19:51.49 | nDuff | [TK]D-Fender: I can call "originate SIP/7665 application Read testvar" on the other side (production server, has SIP phones on it) and it does exactly what I expect, and it has no sound card whatsoever. |
| 19:52.05 | nDuff | [TK]D-Fender: a phone. 5128747665 goes to SIP/7665 on the remote end. |
| 19:52.42 | nDuff | [TK]D-Fender: and as I said, I have actual wav files from iaxmodem describing what it's sending and receiving on each end of the conversation. |
| 19:52.44 | [TK]D-Fender | nDuff: try using a real sip phone on each side for a test. then test each to its server. |
| 19:53.13 | [TK]D-Fender | nDuff: then phone >* > PRI > * in each direction. |
| 19:53.31 | *** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com) |
| 19:55.30 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:57.58 | soulfreshner | is voice cancellation still necessary for BRI cards phoning analog lines? |
| 19:58.09 | soulfreshner | or does the pstn handle that? |
| 19:59.01 | [TK]D-Fender | soulfreshner: Some telcos might do it, not sure for BRI |
| 19:59.47 | logyati | is there a way to integrate asterisk with ldap? i mean fot authentication |
| 20:00.45 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 20:00.53 | *** join/#asterisk jmikeharvey (n=jmikehar@px1.xfoneusa.com) |
| 20:01.24 | jmikeharvey | We have Asterisk working with SIP, but I could use some help with mgcp.conf |
| 20:01.32 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 20:02.14 | [TK]D-Fender | jmikeharvey: Thats a virtually dead topic here.... |
| 20:02.37 | jmikeharvey | I am almost scared to ask, but why? |
| 20:04.22 | *** join/#asterisk icel (n=icel@63.78.162.35) |
| 20:05.58 | [TK]D-Fender | jmikeharvey: Because thats like asking about biplane rotary engines... sure they exist, but who's SEEN ONE in 50 years? |
| 20:06.39 | [TK]D-Fender | jmikeharvey: MGCP = dumb device, and making * do all the work to manae phones is kinda nuts :) Its nearly a proprietary concept |
| 20:07.07 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:07.10 | jmikeharvey | Fender: Well, Mazda still uses the rotary engine, Mazda 3 or 6 it is one of htem :) |
| 20:08.48 | [TK]D-Fender | jmikeharvey: That's why I added the BIPLANE part :P |
| 20:09.03 | jmikeharvey | Fender: The reason I ask is it was going to be a way to convert customers who are using the Adtran Total Access 6xx series over to VoIP since they choose not to put SIP on these unites only on the 9xx series |
| 20:09.09 | [TK]D-Fender | jmikeharvey: And that'd be the RX-7 :) |
| 20:09.29 | jmikeharvey | Fender: Thank you, it has been awhile but I knew Mazda had one :) |
| 20:09.42 | hmmhesays | mazda still has one |
| 20:09.45 | hmmhesays | in the rx-8 |
| 20:09.50 | icel | where are * modules installed by default? |
| 20:10.25 | [TK]D-Fender | jmikeharvey: well I suppose using MGCP for analog VS buying a T1 card to connect them would be cost effective... I'd jsut start testing, and check the mailing lists if I were you. |
| 20:10.52 | [TK]D-Fender | icel: on most distros : /usr/lib/asterisk/modules |
| 20:10.53 | jmikeharvey | Fender: URL? |
| 20:11.05 | [TK]D-Fender | jmikeharvey: www.asterisk.rog |
| 20:11.07 | icel | [TK]D-Fender: thx |
| 20:11.45 | jmikeharvey | Fender: Thanks |
| 20:12.16 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 20:12.34 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 20:15.33 | jameswf | does asterisk get its time from somewhere other than the system time |
| 20:16.23 | soulfreshner | jameswf, that's a good question - coz my phones are all out by 8 hours |
| 20:16.30 | soulfreshner | I just never cared :P |
| 20:16.51 | *** part/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:16.58 | *** join/#asterisk shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com) |
| 20:17.07 | [TK]D-Fender | perhaps your phones need to know their TIME ZONE |
| 20:17.08 | syzygyBSD | aww what a shame, only 1.2.13 is standard for etch? |
| 20:17.13 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 20:17.17 | [TK]D-Fender | syzygyBSD: ARCHAIC. |
| 20:17.20 | syzygyBSD | guess I will have to compile from source |
| 20:17.33 | [TK]D-Fender | syzygyBSD: Reality bids you welcome! |
| 20:17.47 | syzygyBSD | its been a long time! how are you? |
| 20:17.53 | jameswf | the ARI time stamp and the system time vary by 5 hours |
| 20:18.05 | soulfreshner | [TK]D-Fender, more than likely - but I've got a lot of timepieces in the house - and the phones are just for shitz and giggles |
| 20:18.14 | soulfreshner | so I never cared ;) |
| 20:18.35 | jameswf | *ARI vs CDR |
| 20:19.05 | soulfreshner | jameswf, like [TK]D-Fender said - it's probably in your phone setup - you need to check the time zone settings |
| 20:19.06 | jameswf | bleh i need sleep |
| 20:19.13 | jameswf | umm no |
| 20:20.02 | jameswf | cdr says time is now call monitor says call happened in 5 hours |
| 20:20.05 | jameswf | no phones |
| 20:21.03 | [TK]D-Fender | maybe ARI sucks.. |
| 20:21.08 | Yourname`` | [TK]D-Fender: That error from long ago is back! Extension 12 calls extension 30 and ext 12 presses # it says "Transfer".. I don't want that to happen! I removed Tt from the Dial string of ext 12.. and it still happens. :( |
| 20:21.25 | nDuff | [TK]D-Fender: phone's hooked up. Behavior is consistent with what I previously reported. |
| 20:21.36 | [TK]D-Fender | Yourname``: You obviously missed one. |
| 20:21.36 | *** join/#asterisk putnopvut (n=putnopvu@65.116.224.30) |
| 20:21.56 | Yourname`` | [TK]D-Fender: Ok, just noticed. 12 has no Tt, but 30 does. |
| 20:22.06 | Yourname`` | [TK]D-Fender: Does it need to be gone on both of them? |
| 20:22.15 | syzygyBSD | ya, you have to remove it from the extension that is getting dialed |
| 20:22.18 | [TK]D-Fender | Yourname``: DUH <- |
| 20:22.48 | [TK]D-Fender | Yourname``: if that Dial gets called, guess what... DTMF transer = ok/fine/sure |
| 20:22.53 | Yourname`` | lol |
| 20:22.59 | Yourname`` | Thanks guys |
| 20:23.02 | Yourname`` | Lemme try now |
| 20:24.41 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 20:25.49 | [TK]D-Fender | ok, GTG, back in a few hours |
| 20:27.08 | Zefk | Hi. I have a TE412P and a PRI line connected to port 1. I can receive calls but I cannot dial over this trunk. I have got the message: app_dial.c:741 wait_for_answer: Unable to forward voice or dtmf. The same line connected to Avaya is working fine in inbound and outbound calls. Any hints? Thx |
| 20:27.46 | syzygyBSD | wow, when did digium make all the links go through a redirect page before I can download? Sucks for being able to copy a link to wget the file on a server |
| 20:27.59 | hmmhesays | the actual link is in a query string |
| 20:28.37 | bkruse | syzygyBSD: haha, I have that problem everyday. lol |
| 20:28.53 | cpm | at least sourceforge has the decency to offer up a direct link so you can wget the file |
| 20:29.11 | syzygyBSD | ya, I can pull it out, but it is so much easier if it actually links to the file |
| 20:29.47 | syzygyBSD | cpm: only after you have selected your mirror though |
| 20:29.54 | nDuff | grrr. |
| 20:30.41 | cpm | syzygyBSD, true. |
| 20:31.05 | syzygyBSD | I always hate when I forget and just get the html file of the list of mirrors... |
| 20:35.18 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 20:35.23 | *** join/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es) |
| 20:36.10 | _ShrikE | hehe.. dont feel alone! |
| 20:38.52 | syzygyBSD | but I am alone. So very, very alone. |
| 20:41.04 | Elwell | Q - is there a history of asterisk somewhere? from 1999-present? |
| 20:42.30 | Strom_M | you mean besides the changelogs? |
| 20:42.38 | tru_`z24 | how come the old zapata library is not used in asterisk anymore? |
| 20:43.31 | Elwell | Strom_M: problem with changelogs is they don't have dates... |
| 20:43.32 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 20:43.42 | Elwell | at least not the ones imported into SVN |
| 20:43.58 | Elwell | (hmm, the ftp site has em tho) |
| 20:44.19 | lirakis | ok all .. talk to you tomorrow |
| 20:44.21 | lirakis | night |
| 20:44.33 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:46.35 | deeperror | any suggestions on a reliable high volume sip termination provider? |
| 20:46.46 | *** join/#asterisk RazaMetaL (n=razameta@200.93.220.27) |
| 20:46.59 | RazaMetaL | hi |
| 20:47.51 | AndrewGearhart | deeperror: no experience with them... but I've been talking with bandwidth.com |
| 20:48.22 | dfriend | I have a stand alone * box for voicemail server. Each extension belongs to their own context which is the same as their extension #. |
| 20:48.32 | dfriend | When a user is directed to VoicemailMain it prompts for mailbox # then password. The server always responds invalid login. |
| 20:48.48 | dfriend | I am sure it is because * is not looking in the default context. Anyone know how to suggest VoicemailMain to look in a different context after getting the mailbox number? |
| 20:49.00 | dfriend | Is there anyway * can fall-through and see if the mailbox number is in a different context than the default? |
| 20:49.16 | bkruse | mailbox = context in the user definition |
| 20:49.37 | xipi | help |
| 20:49.54 | xipi | <PROTECTED> |
| 20:50.02 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 20:50.03 | bkruse | ~pb |
| 20:50.04 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:50.33 | putnopvut | A pastabin is a bucket of noodles. |
| 20:50.45 | dfriend | Actually, there is a different * server which the user connects to. That server forwards voicemail requests to this stand alone voicemail server. |
| 20:52.17 | *** join/#asterisk ta^3 (n=tacvbo@65.116.224.30) |
| 20:54.46 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 20:55.00 | RazaMetaL | I've a problem.. please help |
| 20:55.45 | *** join/#asterisk korihor (n=humberto@190.75.38.113) |
| 20:56.37 | RazaMetaL | I've one e1 span and one analog tdmfxs card |
| 20:56.58 | RazaMetaL | when I reboot the module order changes |
| 20:57.19 | RazaMetaL | so after every reboot mi cat /proc/zaptel/* i sdifferent |
| 20:57.26 | bkruse | RazaMetaL: you can give the cards a hard pci idea with the circle switch on the board. |
| 20:58.36 | syzygyBSD | there is no way to multithread make is there? |
| 20:58.36 | RazaMetaL | ? on the mother boards ? |
| 20:58.37 | tzafrir_home | RazaMetaL, what distro is it? |
| 20:58.50 | Katty | bored. |
| 20:58.52 | RazaMetaL | hi tzafrir , is Ubuntu feisty server |
| 20:58.53 | Katty | so very bored. |
| 20:59.03 | RazaMetaL | Katty, me too :) |
| 20:59.10 | tzafrir_home | put the modules in the proper order in /etc/modules |
| 20:59.15 | Katty | i was play some warcraft. |
| 20:59.23 | Katty | but this annoy alliance warrior cept camping my corpse |
| 20:59.41 | RazaMetaL | I've the following order |
| 20:59.43 | RazaMetaL | lp |
| 20:59.47 | RazaMetaL | wcte11xp |
| 20:59.53 | RazaMetaL | and finalyy wctdm |
| 20:59.56 | tzafrir_home | RazaMetaL, modules listed in /etc/modules get probed before the automatic bus scan |
| 21:00.17 | RazaMetaL | tzafrir, yes... I know and I've listed |
| 21:00.24 | tzafrir_home | lp ? zaptel over paraller port? |
| 21:00.28 | tzafrir_home | :-) |
| 21:00.40 | RazaMetaL | but seems like ubuntu load de modules in a different order every time |
| 21:00.48 | syzygyBSD | Katty: find / -iname \*c -exec mv {} {}c \; |
| 21:00.52 | syzygyBSD | then see if you can fix it |
| 21:00.55 | RazaMetaL | sometimes /proc/zaptel/1 is assigned to my e1 card |
| 21:00.56 | Katty | uhh |
| 21:00.57 | Katty | no |
| 21:01.00 | Katty | kthx |
| 21:01.07 | RazaMetaL | and /proc/zaptel/2 is assigned to the analog card |
| 21:01.24 | RazaMetaL | and sometimes zaptel/1 is assigned to analog and zaptel/2 to e1 |
| 21:01.28 | syzygyBSD | it won't break everything... |
| 21:01.29 | tzafrir_home | This is after a cold boot, or after restarting zaptel? |
| 21:01.38 | Katty | i'm not /that/ bored |
| 21:01.39 | *** part/#asterisk SickTear (i=SickTear@89.137.56.184) |
| 21:02.12 | RazaMetaL | tzafrir_home, after a cold boot |
| 21:02.29 | syzygyBSD | I mean there is a command that would fix the one i gave you... |
| 21:02.39 | syzygyBSD | even as a mental exercise it is great fun! |
| 21:02.47 | RazaMetaL | I i run genzaptelconf -sdvM the correct order is stablished |
| 21:02.55 | tzafrir_home | RazaMetaL, hmmm , whell hceck /etc/rcS.d |
| 21:03.35 | syzygyBSD | find / -iname \*cc -exec mv {} `sed s/cc$/c/` \; |
| 21:03.38 | syzygyBSD | I think* |
| 21:04.09 | syzygyBSD | oops, forgot to give sed the filename, but meh, I hope nobody really ran the first command |
| 21:04.24 | tzafrir_home | syzygyBSD, I suppose you tested this on your system first, right? |
| 21:04.44 | RazaMetaL | let me try adding genzaptelconf -sdvM at the boot time ... |
| 21:04.46 | syzygyBSD | uhhh... of course i did |
| 21:05.00 | tzafrir_home | RazaMetaL, that is way wrong |
| 21:05.21 | tzafrir_home | is there any restart of zaptel in your standard boot? |
| 21:05.26 | waltj | How can one connect to channel by the channel ID (${CHANNEL})? |
| 21:05.50 | RazaMetaL | tzafrir_home, let me check it... |
| 21:07.04 | RazaMetaL | tzafrir_home, I've one zaptel file at /etc/init.d |
| 21:07.34 | tzafrir_home | Right. But the start target there won't unload modules |
| 21:08.29 | RazaMetaL | i've another zaptel file at /etc/default/zaptel |
| 21:08.42 | RazaMetaL | i've copied it from the internet |
| 21:08.55 | *** join/#asterisk FunnyManVA (n=funnyman@65.116.224.30) |
| 21:16.03 | syzygyBSD | what do I need to do to make zttool compile? |
| 21:16.13 | bkruse | libnewt |
| 21:17.34 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:17.58 | syzygyBSD | thanks! |
| 21:19.20 | bkruse | np |
| 21:19.24 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.167.16) |
| 21:19.46 | syzygyBSD | wow, really, thanks. I don't know if I could have guessed what issue someone was having with a compile with what I gave you. Spot on chap |
| 21:20.54 | *** join/#asterisk masus (n=masus@78.162.8.36) |
| 21:21.21 | masus | hi all , i get errrors after "make" -> http://pastebin.com/m5fa0728e... |
| 21:21.32 | masus | thanks |
| 21:22.06 | masus | http://pastebin.com/m5fa0728e |
| 21:22.27 | syzygyBSD | my guess is you have the wrong version of mysql headers |
| 21:22.36 | syzygyBSD | are you upgrading? |
| 21:22.49 | masus | i have install func_odbc |
| 21:22.58 | syzygyBSD | oops, mssql, not mysql |
| 21:23.01 | masus | and make clean and install |
| 21:23.18 | masus | mssql headers ? |
| 21:23.41 | *** join/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 21:24.09 | masus | do u know how to fix it |
| 21:24.39 | nny | so about to recompile zaptel. Running Asterisk 1.4.5 and zaptel 1.4.3 |
| 21:25.01 | syzygyBSD | nny: why do you have to recompile? |
| 21:25.31 | masus | for activate func_odbc |
| 21:25.33 | nny | having issues where trying to fxotune results in a Failure! due to something crapping the bed. I listened in with a splitter, and it dials 4, and then blasts out a serious of "beeps" (like a line being off the hook) |
| 21:25.58 | masus | ah sorry |
| 21:26.02 | nny | everything else works AFAIK.. although the echo is a bit nasty atm |
| 21:26.40 | nny | if I dial 4 with a POTS phone, it doesn't do the same thing |
| 21:26.42 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:27.08 | nny | couldn't find any good leads in mailing lists, back alleys, etc. so I figured recompile of zaptel would be the first best step.. although I have to say.. it worked when I did it two weeks ago |
| 21:27.45 | nny | maybe strace could help figure it out? |
| 21:30.14 | nny | Tuning module /dev/zap/1 |
| 21:30.14 | nny | Could not fill input buffer - got -1 bytes, expected 4000 bytes |
| 21:30.14 | nny | Failure! |
| 21:30.21 | nny | is the error for what it's worth |
| 21:33.41 | syzygyBSD | does spandsp 0.0.3 work yet? |
| 21:41.12 | nDuff | syzygyBSD: I always use the one bundled with iaxmodem, so I couldn't comment on version numbers |
| 21:41.35 | syzygyBSD | I haven't tried it with 1.4 yet, so I don't know which version will work |
| 21:41.37 | nDuff | syzygyBSD: that said, the 20070619 snapshot (which iaxmodem 0.3.1 uses) works well. |
| 21:41.49 | nDuff | syzygyBSD: are you referring to app_{tx,rx}fax? |
| 21:41.53 | syzygyBSD | yup |
| 21:41.55 | nDuff | syzygyBSD: I don't exactly recommend those. |
| 21:42.11 | syzygyBSD | they have worked well for me in the past, but what do you use? |
| 21:42.18 | nDuff | iaxmodem+hylafax |
| 21:43.41 | *** join/#asterisk entelechy (n=chatzill@mail.beanproducts.com) |
| 21:43.43 | syzygyBSD | hmm, sounds like more points of failure... how has it been for you? |
| 21:44.10 | entelechy | Greetings and salutations! |
| 21:44.11 | syzygyBSD | do you ever get the annoying 1/3 of a page issues I sometimes see in rxfax? |
| 21:44.50 | entelechy | I am wondering: is there a way to get Cisco 7940's to have a different ringtone on each line? |
| 21:45.30 | syzygyBSD | it has been a while since I set one of those up (I boxed up my test phone) but I believe so |
| 21:45.31 | entelechy | The Grandstream GXP-2000 supports this. I can't believe the Cisco which costs nearly 3X as much doesn't support this. |
| 21:45.32 | nny | mwahaha recompile fixed.. something.. yay done |
| 21:45.53 | masus | "cdr_tds.c:350: error: too many arguments to function âtds_alloc_contextâ" |
| 21:45.55 | entelechy | syzygyBSD: well if there is a way, it probably involves reverting the phone from SIP firmare to SCCP at minimum |
| 21:46.00 | syzygyBSD | entelechy: how are you configuring it |
| 21:46.14 | entelechy | syzygyBSD: SIP via tftp. |
| 21:46.20 | entelechy | 8.6 firmware |
| 21:46.43 | syzygyBSD | k, let me hop into a server with a bunch of files and see if I can find the option |
| 21:46.59 | entelechy | syzygyBSD: thank you! I've been googling this for *days* and can't find this tidbit. |
| 21:50.28 | *** join/#asterisk codefreeze (n=steve_mu@65.116.224.30) |
| 21:50.28 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 21:53.52 | syzygyBSD | entelechy: nope, not luck in my files, did you check http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00805caf30.shtml ? |
| 21:54.44 | *** join/#asterisk mangolian (n=Frossty@CPE00c049e0d0b4-CM00111ae2bb20.cpe.net.cable.rogers.com) |
| 21:54.54 | entelechy | thanks, checking |
| 21:55.02 | syzygyBSD | ie, you need a RingList.xml file on your tftp server |
| 21:55.23 | entelechy | yes |
| 21:55.25 | entelechy | thing is |
| 21:55.29 | entelechy | that just updates the available rings |
| 21:55.38 | entelechy | it has nothing to do with configuring rings on a per-line basis |
| 21:55.54 | syzygyBSD | right... would be a phone setting I think |
| 21:55.59 | entelechy | in the SIP 8.x firmware |
| 21:56.03 | entelechy | there is no setting on a per-line basis |
| 21:56.56 | entelechy | e.g. "ring type" is a main menu option on the phones config menu. there do not appear to be any corresponding sub-menu options in the SIP configuration entries |
| 21:57.17 | entelechy | choosing the menu option "ring type" lists the available ring tones, but sets this globally. |
| 21:58.07 | entelechy | i was hoping maybe if i reverted the phone to SCCP firmware maybe there'd be an option there |
| 21:58.50 | nny | are you supposed to have to recompile asterisk after recompiling zaptel??? |
| 21:59.12 | syzygyBSD | nny: depends, did you change the version of zaptel? |
| 21:59.26 | nny | syzygyBSD: nope |
| 21:59.30 | syzygyBSD | then no |
| 21:59.54 | syzygyBSD | you shouldn't even have to recompile zaptel, but who knows, maybe just reinstalling it would have worked |
| 21:59.55 | BockBilbo | EXTEN is the variable used for the current extension.... is there any variable that returns the initially called extension? or should i use a variable to know that information? |
| 22:00.09 | nny | syzygyBSD: I would have thought so too |
| 22:01.02 | nny | it;s weird ztmonitor shows incoming tx/r x etc. but asterisk never picks up,... guess trying a recompile can't kill anyone |
| 22:01.12 | nny | it worked right before I recompiled zaptel -_- |
| 22:02.06 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:02.11 | syzygyBSD | nny: ahh, have you reloaded the modules? |
| 22:02.24 | syzygyBSD | and restarted asterisk? |
| 22:02.26 | nny | syzygyBSD: yah |
| 22:02.31 | nny | yup |
| 22:03.08 | syzygyBSD | umm, did you chant the sacred rite with a candle under the full moon with one arm over your head? |
| 22:04.11 | mvanbaak | lol |
| 22:05.26 | *** join/#asterisk el_critter (n=chatzill@190.74.96.121) |
| 22:05.49 | el_critter | hi |
| 22:06.04 | *** join/#asterisk Rtx (i=xtr@bas1-quebec09-1178040637.dsl.bell.ca) |
| 22:06.30 | el_critter | is there some asterisk "glossary" where I can look for asterisk variables used in config files? I don't want to bother you with newbie questions all the time |
| 22:06.54 | _ShrikE | like core show variables? |
| 22:06.54 | Rtx | hi someone know why zaptel recongnize a openvox 400p with one fxo module as fxs card ? |
| 22:06.58 | el_critter | I mean... I don't know what a "guest call" is or what "overlap dialing support" is |
| 22:07.17 | syzygyBSD | ~book |
| 22:07.18 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 22:08.05 | el_critter | thx! |
| 22:08.52 | Rtx | someone ? |
| 22:12.53 | *** part/#asterisk dfriend (i=dfriend@69.89.174.2) |
| 22:15.45 | generalhan | can some one help me figure out why CHANISAVAIL seems think that this one user's phone is always off hook ? 'sip show channels' doesnt show the phone as being used yet my dial plan just wont dial this user (just started happening about 20 minutes ago): http://generalhan.pastebin.ca/714062 |
| 22:15.53 | nny | ok so wtfbbq |
| 22:16.18 | nny | reloaded/ compiled zaptel modules and now i get nothing after starting asterisk, even after recompiling asterisk |
| 22:16.25 | Deeewayne | nny: how'd you do last night w/ your analog problem ? |
| 22:16.41 | nny | Deeewayne: recompiled zaptel and got fxotune working |
| 22:16.45 | nny | now asterisk won't answer the line |
| 22:17.30 | Deeewayne | what does ztcfg -vv output ? |
| 22:18.06 | nny | should i post it here? kind of a lot |
| 22:18.08 | syzygyBSD | generalhan: what does sip show peers say for his line? |
| 22:18.16 | Deeewayne | pastebin |
| 22:18.18 | nny | basic nutshell, Channel 01 and 03 FXK Kewlstart |
| 22:18.22 | nny | k |
| 22:18.44 | nny | http://pastebin.com/m6a22467d |
| 22:18.45 | generalhan | syzygyBSD: shows his IP addy and port number like any other user ! |
| 22:19.13 | Rtx | Hi i own an openvox A400P01 thats have 1 fxo port on it. But when i do a genzaptelconf it recongnize it like an fxs it is normal ? |
| 22:20.00 | generalhan | syzygyBSD: isnt that weird ? this user can make calls out just fine, but CahIsAvail seems to want to push to VM instead of ring the phone |
| 22:20.49 | saftsack | hi is someone here ready for a test call? |
| 22:20.53 | Deeewayne | nny: can you post your /etc/zaptel.conf and /etc/asterisk/zapata.conf too ? |
| 22:21.04 | nny | Deeewayne: yes |
| 22:21.23 | syzygyBSD | generalhan: well, why do you even need the ChanIsAvail step at all? |
| 22:21.33 | Rtx | no one can answer :( ? |
| 22:21.44 | syzygyBSD | can you dial the phone without that step? |
| 22:21.57 | generalhan | syzygyBSD: hmm good question. lets see ! |
| 22:23.07 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.136.194) |
| 22:23.32 | syzygyBSD | rtx, are you sure you are right? and that it isn't the type of line comming in you are looking at? |
| 22:23.56 | Rtx | yes im sure thats i have an fxo card |
| 22:24.30 | nny | Deeewayne: http://pastebin.com/m4862d53e |
| 22:24.36 | syzygyBSD | what are you looking at that tells you it is fxs, pastebin it |
| 22:24.46 | Rtx | but ztcfg return me this: Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 22:25.53 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-57-98.ph.ph.cox.net) |
| 22:26.01 | generalhan | syzygyBSD: Hmmm, i have call-limit=1 set in sip.conf ... but this user is NOT on the phone right now and still the CLI shows "NOTICE[27154]: chan_sip.c:2275 update_call_counter: Call to peer '7106' rejected due to usage limit of 1" when i bypass the ChanIsAvail |
| 22:26.09 | syzygyBSD | Rtx: have you tested to see if it works? |
| 22:26.30 | Rtx | no because it detect it as a fxs |
| 22:26.46 | syzygyBSD | Rtx: try |
| 22:26.48 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
| 22:27.03 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:27.14 | Rtx | i runed: trixbox pro, trixbox 2.2.4 and finally trixbox (openvox moded) 2.2.4 |
| 22:27.28 | syzygyBSD | lol... no wonder... |
| 22:27.55 | nny | Rtx: under ztcfg I think it is supposed to report what signalling it uses, aka FXS, as FXO uses FXS signalling and visa versa |
| 22:28.02 | nny | Rtx: I'd test it right quick |
| 22:28.06 | generalhan | so i guess i need to figure out why call-limit thinks there is a call on that line, when there really isnt :( |
| 22:28.26 | syzygyBSD | you will get better trixbox support in their channel, but my first suggestion is to test it |
| 22:28.56 | Rtx | syzygyBSD: k gonna try thank you |
| 22:29.03 | Deeewayne | nny: I don't know if it matters, but you might want to remove the spaces around the equal sign in zapata.conf for signalling and zaptel.conf loadzone |
| 22:29.17 | nny | Deeewayne: yah just noticed those will do |
| 22:29.45 | Deeewayne | also, not sure but you may want to reload your drivers too |
| 22:29.59 | nny | Deeewayne: have tried one reboot already |
| 22:30.08 | Deeewayne | I mean after removing the spaces |
| 22:30.27 | nny | heh oops ok doign that now |
| 22:30.40 | Rtx | ah cruel world :\ buyed a more expensive card to save headhatch but same to be more hard than a cheap x100p heh :\ |
| 22:31.17 | Yourname`` | Sep 25 14:28:52 WARNING[29795]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 22:31.19 | Yourname`` | whats this? |
| 22:31.33 | syzygyBSD | looks like a warning, you better listen |
| 22:31.46 | *** join/#asterisk svensk_neutrino (n=tze@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 22:31.49 | svensk_neutrino | hi |
| 22:31.51 | svensk_neutrino | anyone home |
| 22:32.01 | syzygyBSD | nope, I'm at work |
| 22:32.02 | svensk_neutrino | when i do sip show registty |
| 22:32.03 | Ritzerisk | hey hey does anyone if i can install vicidial through the asterisk console |
| 22:32.18 | svensk_neutrino | i get two entries of my box registered with same Ip address |
| 22:32.34 | nny | Deeewayne: sonofa |
| 22:32.35 | svensk_neutrino | registered to same proxy i mean but listed twice when i do sip show registry |
| 22:32.52 | nny | Deeewayne: well i guess that was it.. funny thing is I have had those conf files for a while -_- |
| 22:32.54 | Deeewayne | nny: got it ? O.o |
| 22:33.00 | nny | Deeewayne: magic :) |
| 22:33.13 | nny | Deeewayne: thank you |
| 22:33.14 | Deeewayne | spaces could get you in the dialplan too |
| 22:33.26 | Deeewayne | no problem and good luck! |
| 22:33.30 | syzygyBSD | svensk_neutrino: if you ask a question on only one line it is much easier to read and answer |
| 22:33.45 | nny | Deeewayne: now to test that echo again |
| 22:33.53 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 22:33.57 | Deeewayne | ~thebook |
| 22:33.57 | jbot | somebody said thebook was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:34.01 | *** join/#asterisk gremzoid (n=gremzoid@d58-111-170-15.rdl5.qld.optusnet.com.au) |
| 22:34.04 | Deeewayne | lots of great examples in the book |
| 22:34.05 | nny | anyone have an opinion on the Digium Echo Software vs. open source etc?? |
| 22:34.11 | Deeewayne | Answer() -> Echo() |
| 22:34.13 | nny | Deeewayne: got Version 1 here |
| 22:34.33 | generalhan | syzygyBSD: when i run 'sip show inuse' it shows this user has 1 call and 1 limit, even though the user is not on the phone. have you heard of this happening before, or maybe a way to fix this ? |
| 22:34.57 | *** join/#asterisk seebs (n=seebs@mail.seebs.net) |
| 22:35.01 | nny | Deeewayne: this system has been running since march with the help of mercestes... but I have been learning the various troubleshooting stuff etc in hopes of building more |
| 22:35.04 | svensk_neutrino | syzygyBSD:i am sorry about that. |
| 22:35.07 | nny | Deeewayne: need v2 of book :) |
| 22:35.20 | syzygyBSD | generalhan: ya, I have had that problem on old versions of asterisk. solution is to restart |
| 22:35.41 | Deeewayne | me too |
| 22:35.43 | seebs | Well, my mysterious problem isn't fixed. Synopsis: kubuntu (with X off) + asterisk, after a few hours of uptime, the system becomes virtually unusable due to HUGE lags. |
| 22:36.11 | generalhan | syzygyBSD: hmmm ... that is NOT an option on this production system. so the only way for this to reflect the correct call/limit is to restart asterisk ? |
| 22:36.14 | seebs | Specifically, any and all timer operations take MUCH longer than they should. 'date; sleep 1; date' will show a lag of about a minute, and will actually take about a minute. Or maybe just 30 seconds. |
| 22:36.28 | syzygyBSD | generalhan: possibly just rebooting his phone... |
| 22:36.30 | seebs | System's fine fresh after a boot, but goes south "later". |
| 22:36.37 | syzygyBSD | it might fix it when it reregisters |
| 22:36.40 | generalhan | syzygyBSD: nah, triued that already ! |
| 22:37.17 | svensk_neutrino | On asterisk cli i do, 'sip show registry' and it shows 1.an old proxy that i once put in sip.conf but not any more as unregistered 2.another proxy that i current have in my sip.conf listed to be registered but twice (2 enteries). can someone explain why old proxy is still listed even if i do reload and the new proxy is listed twice. I am using asterisk realtime static. |
| 22:37.18 | generalhan | and im using 1.2.23, so not too old of a version :( |
| 22:37.28 | syzygyBSD | I have had queues get stuck like that too, on production systems. Had to create new queues to fix it in the interm |
| 22:37.52 | seebs | Does the zaptel driver normally affect host system timing? |
| 22:38.21 | generalhan | hmm, wonder if i comment out the call-limit line reload, and then add it back again |
| 22:38.23 | syzygyBSD | svensk_neutrino: do you have it in your sip.conf as a friend entry and also a register => |
| 22:38.45 | nny | anyone know what happened tp the ftp site for HPEC? |
| 22:38.46 | Ritzerisk | anyone messed with vicidial |
| 22:38.53 | nny | I have a license key, but the links they emailed me are dead |
| 22:39.05 | nny | did an fxotune, played with rx and tx, yet the echo remians |
| 22:39.08 | nny | remains |
| 22:39.16 | nny | gonna try HPEC to see if it is better |
| 22:40.03 | nny | vm found it |
| 22:40.06 | nny | nm* |
| 22:40.11 | nny | http://ftp2.digium.com/pub/telephony/hpec/32-bit |
| 22:40.45 | svensk_neutrino | syzygyBSD: I have one register=> and another entry for it as peer that i refer to in the dialplan for dialling out through it. |
| 22:40.47 | syzygyBSD | generalhan: it doesn't fix the issue but you might look through this thread for ideas of how to prevent this in the future http://lists.digium.com/pipermail/asterisk-users/2007-February/180496.html |
| 22:41.06 | Deeewayne | nny try downloads.digium.com, not ftp.digium.com |
| 22:41.13 | syzygyBSD | svensk_neutrino: you don't need a register if you have a friend entry for it |
| 22:41.19 | syzygyBSD | that is why you have two |
| 22:41.41 | svensk_neutrino | syzygyBSD:its type=peer not friend |
| 22:42.06 | syzygyBSD | it is the same issue |
| 22:42.35 | syzygyBSD | I haven't found a need ever for register => |
| 22:42.44 | svensk_neutrino | syzygyBSD:okay but then why is the old proxy still lingering in the sip show register cache |
| 22:43.15 | gremzoid | have you restarted? |
| 22:43.18 | gremzoid | :P |
| 22:43.23 | syzygyBSD | because it hasn't timed out yet? |
| 22:43.47 | svensk_neutrino | its been 15minutes now |
| 22:43.56 | Ritzerisk | whats a good predictive dialer with asterisk |
| 22:44.04 | svensk_neutrino | and that proxy didnt even exist. |
| 22:44.36 | svensk_neutrino | syzygyBSD:that proxy is not my friend, i dont expect them to call me ever. |
| 22:44.49 | JT | nny: get hardware echo cancellation if you're serious about removing echo |
| 22:45.17 | syzygyBSD | svensk_neutrino: then why do you need to register with it? |
| 22:46.51 | JT | syzygyBSD: the silence is deafening ;) |
| 22:48.27 | *** join/#asterisk putnopvut (n=putnopvu@65.116.224.30) |
| 22:48.42 | nny | JT lol no kidding, I have to sacrifice a goat to use HPEC |
| 22:48.43 | syzygyBSD | ya, logic hurts the brain sometimes. I hope someone is there to help if needed |
| 22:48.46 | nny | someone peep this readme |
| 22:48.51 | nny | http://ftp2.digium.com/pub/telephony/hpec/README |
| 22:49.18 | nny | I have the tdm400p.. not a T1... I hear there are some new 8 port cards that support hardware echo cancel.. but not this one |
| 22:49.32 | *** join/#asterisk dan__t (i=dan@neener.neener.org) |
| 22:49.40 | dan__t | HI. |
| 22:49.42 | JT | nny: yes the 8 and 24 port support it i believe |
| 22:49.48 | dan__t | Is there an application to speak the caller ID number back to you? |
| 22:49.52 | JT | nny: how many fxo lines do you have? |
| 22:50.12 | syzygyBSD | dan__t: you mean like SayDigits(${CALLERID}) |
| 22:50.17 | dan__t | That'll work. |
| 22:50.20 | dan__t | Thansk. |
| 22:50.23 | dan__t | Thanks, too. |
| 22:50.31 | nny | JT tweo |
| 22:50.32 | nny | two* |
| 22:50.37 | dan__t | How about speaking the actual textual description of the caller ID data? |
| 22:50.42 | nny | JT lol they license HPEC to the MAC address |
| 22:50.48 | syzygyBSD | dan__t: festival |
| 22:50.48 | JT | ${CALLERID(num)} |
| 22:50.48 | nny | yeah cause that never fails -__ |
| 22:51.03 | JT | ${CALLERID} is not really correct |
| 22:51.08 | dan__t | Yeah, I could do it with Festival, I was just hoping there was already an application to do ti. |
| 22:51.10 | dan__t | it, rather. |
| 22:51.29 | nny | JT tried fxotune in 1.4 zaptel.. got better, although still audible until echo training kicks in |
| 22:51.49 | JT | nny: i've already said the only way to get rid of it if you're serious is hardware |
| 22:52.04 | JT | i'm not going to entertain screwing with software |
| 22:52.05 | JT | :) |
| 22:52.14 | nny | JT figured as much,... working on systems for SOHO... most business around here have 2-3 lines .. (yeah i understand completely) |
| 22:52.28 | generalhan | syzygyBSD: umm. so i decided to just take the call-limit off of this user's phone. took that line out, did a 'reload' in the CLI and it still show that they have a limit of 1 ? |
| 22:52.29 | syzygyBSD | wetwear is so much more fun to screw with |
| 22:52.45 | nny | JT is there any 2-4 FXO cards that do echo cancel in hardware.. I'd buy it |
| 22:53.09 | nny | JT guess I could sell the 8 port with echo cancel and just say the extra ports are for "upgrading" |
| 22:53.30 | syzygyBSD | generalhan: have they reregistered yet? |
| 22:53.51 | generalhan | even when i reboot the phone it doesnt re-register ? |
| 22:53.57 | syzygyBSD | it should |
| 22:54.08 | generalhan | how can i for it to UNregister ? |
| 22:54.16 | generalhan | is there a CLI command that will allow me to kill a registration ? |
| 22:54.28 | generalhan | then i can reboot that phone and make sure that it starts a new registration |
| 22:54.39 | syzygyBSD | uh.. other than "stop now" ? |
| 22:54.48 | generalhan | syzygyBSD: yes, other than that ! |
| 22:55.03 | Ritzerisk | whats a good predictive dialer with asterisk |
| 22:55.13 | syzygyBSD | not that I have found |
| 22:55.19 | syzygyBSD | Ritzerisk: write your own |
| 22:55.38 | Ritzerisk | haha tricky i wish i can |
| 22:55.38 | generalhan | syzygyBSD: maybe if i put the registration period on the phone for like 100 rather than 3000 ? |
| 22:56.16 | Ritzerisk | i heard of vicidial and gnudialer but im not to sure on the installation procedures |
| 22:56.17 | syzygyBSD | generalhan: na, since there is already a call in for that sip line it probably wont.... |
| 22:57.11 | syzygyBSD | generalhan: any chance that use is in a queue? |
| 22:57.15 | syzygyBSD | user* |
| 22:57.23 | *** part/#asterisk korihor (n=humberto@190.75.38.113) |
| 22:57.26 | generalhan | no, no queues hit this phone |
| 22:57.46 | syzygyBSD | any local channels? |
| 22:58.18 | generalhan | um, not sure what you mean |
| 22:58.20 | *** join/#asterisk famx (n=fem@user-0cevbvh.cable.mindspring.com) |
| 22:58.25 | famx | hi |
| 22:58.30 | svensk_neutrino | thanks bsd dude ;) |
| 22:58.37 | JT | nny: you dont have to buy cards fully populated |
| 22:59.48 | famx | running eyebeam here and having that clicking sound static sound time to time...especially when I am on phnoe and I am opening web browser or any other application |
| 23:00.11 | famx | any know what might be causing that ? |
| 23:00.11 | Mavvie | you found out where the problem lies :-) |
| 23:00.25 | famx | :( |
| 23:00.45 | famx | trying but so far no luck was hoping that might help |
| 23:01.02 | nny | sigh |
| 23:01.05 | nny | man F digium |
| 23:01.15 | nny | installed HPEC... and now it doesn't work -_- |
| 23:01.31 | Mavvie | get faster disks, have less programs open (i.e. less memory in use), get a faster computer etc etc etc. |
| 23:01.38 | rob0 | No manual entry for F |
| 23:01.38 | rob0 | No manual entry for digium |
| 23:01.58 | davevg-btwtech | famx: try and change the priority in task manager for the eyebeam app? so it is higher than the other apps? |
| 23:01.59 | nny | lol! |
| 23:02.04 | Corydon76-dig | No manual entry for digium in section F |
| 23:02.05 | nny | thanks sir bot |
| 23:02.09 | famx | my hd index in vista is 5.9 |
| 23:02.12 | famx | memory 5.5 |
| 23:02.17 | famx | cpu 4.5 |
| 23:02.19 | nny | is there other echo settings that screw with HPEC? |
| 23:02.37 | Corydon76-dig | nny: no, HPEC overrides everything |
| 23:02.49 | famx | davevg-btwtech: ok let me try that |
| 23:03.00 | nny | Corydon76-dig: it picks up, but I can't hear othe rline and visa versa.. console shows normal |
| 23:03.23 | Corydon76-dig | nny: using what protocol? |
| 23:03.49 | Mavvie | http://bugs.digium.com/view.php?id=9511 <- for fuck sake... |
| 23:03.53 | nny | Corydon76-dig: commented all echo related stuff in zapata.conf |
| 23:03.58 | Mavvie | just because you don't have the problem doesn't mean that it isn't there. |
| 23:04.00 | nny | Corydon76-dig: works now |
| 23:04.09 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 23:04.09 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 23:04.17 | famx | davevg-btwtech: where excatly I set that ? |
| 23:04.17 | nny | Corydon76-dig: I had echocancel=512, is that too much for a 1 GHZ with 1 GB of ram? |
| 23:04.30 | Corydon76-dig | nny: HPEC uses 1024, I think |
| 23:04.41 | davevg-btwtech | farmx: right click on the task and change the priority? |
| 23:04.42 | Corydon76-dig | Hardcoded, you can't change it |
| 23:04.49 | nny | Corydon76-dig: well commenting out all stuff worked well |
| 23:05.04 | nny | Corydon76-dig: their readme says otherwise regarding taps... maybe they changed it? |
| 23:05.18 | Corydon76-dig | nny: I don't know offhand, actually |
| 23:05.32 | nny | Corydon76-dig: no problem.. it works now.. and echo is less than ever |
| 23:05.36 | nny | so yay stuff |
| 23:06.49 | Deeewayne | nny: you had 'echocancelwhenbridged=no' in your configuration file...maybe this is causing problems |
| 23:06.55 | Mavvie | Is there a Digium developer in Australia? |
| 23:07.13 | nny | Deeewayne: i did... is that a problem for HPEC? |
| 23:07.46 | nny | Deeewayne: figured that when two phones where using SIP to communicate, echo cancel would just consume CPU |
| 23:09.07 | nny | Deeewayne: nah I had echo training on too.. and relaxedDTMF=yes... 86'd those and it works |
| 23:09.09 | nny | cool cool |
| 23:09.14 | nny | thanks all for the help and advice |
| 23:09.16 | nny | time for beer |
| 23:09.24 | nny | drink one to #asterisk |
| 23:10.49 | Corydon76-dig | Drink one to the Digium guys who helped you |
| 23:14.47 | *** join/#asterisk lukus (n=luke@202.172.122.210) |
| 23:14.48 | nny | lol ty that must mean you all... ^^' |
| 23:15.22 | nny | will do... cheers. ty again |
| 23:15.24 | *** part/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 23:15.40 | gremzoid | sounds like a happy chap :P |
| 23:19.42 | famx | anyone know what might causing clicking/static in eyebeam |
| 23:20.55 | lukus | famx: well, that could be a wide range of issues, it could be your asterisk setup, or it could even be your sound setup. You will have to share more information. |
| 23:21.29 | *** part/#asterisk gremzoid (n=gremzoid@d58-111-170-15.rdl5.qld.optusnet.com.au) |
| 23:21.48 | famx | that clicking/static increase when I am opening new application |
| 23:22.33 | famx | and hear less maybe every 3 seconds when running eyebeam alone |
| 23:22.38 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 23:25.48 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.135) |
| 23:29.17 | *** join/#asterisk osiris (n=osiris@c-71-205-35-230.hsd1.mi.comcast.net) |
| 23:34.11 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 23:36.58 | ManxPower | I've got you now, my pretty! |
| 23:37.52 | famx | lukus: it is not the asterisk installed different sip soft and no clicking |
| 23:38.05 | s34n | I've got my itsp registration working again again, but incoming calls are getting rejected |
| 23:38.25 | s34n | I would think that my peer is misconfigured, but... |
| 23:38.38 | s34n | it was working just fine before, and |
| 23:38.40 | *** join/#asterisk sunsmasher (n=Beamer@66.251.47.154) |
| 23:38.44 | famx | now if i could only register with my asterisk with latest eyebeam 1.5 |
| 23:38.53 | famx | for some reason it does not let me |
| 23:38.56 | s34n | it matches the sip invite that I see coming in |
| 23:40.25 | RazaMetaL | hi, can any 1 help me with one small ubuntu issue? is related with the zaptel load at start up |
| 23:40.32 | s34n | incoming calls would get a 404 Not Found if the peer was misconfigured, or if the receiving extension was misconfigured |
| 23:40.39 | s34n | Any other suspects? |
| 23:40.45 | *** join/#asterisk metfan2007 (n=metfan20@189.135.156.38) |
| 23:41.16 | s34n | RazaMetaL: you'll have to give more info than that |
| 23:41.51 | RazaMetaL | s34n, thank you s34n ... i've the /etc/init.d/zaptel at the runlevel 2 |
| 23:42.12 | RazaMetaL | but asterisk starts and is not loading the zaptel configuration |
| 23:42.23 | s34n | RazaMetaL: what runlevel are you in? |
| 23:42.32 | s34n | RazaMetaL: 3? |
| 23:42.33 | RazaMetaL | i need to run asterisk -r -x "module load chan_zap.so" |
| 23:42.37 | RazaMetaL | s34n, 2 |
| 23:43.16 | RazaMetaL | s34n, my distro is ubuntu, release 7.04 for server |
| 23:43.23 | RazaMetaL | does not have X installed on it |
| 23:43.26 | metfan2007 | hi all, is there a way to sotre in the CDR data what agent answer a call in a queue in a inbound campaign??? |
| 23:44.06 | s34n | RazaMetaL: are you sure runlevel 2? no networking? |
| 23:44.40 | RazaMetaL | s34n, yes... ubuntu have runlevel 2 with networking |
| 23:44.43 | RazaMetaL | let me see |
| 23:45.05 | CCFL_Man2 | anyone seen rudholm? |
| 23:45.11 | CCFL_Man2 | rudholm!!!! |
| 23:45.18 | rudholm | CCFL_Man2: yes |
| 23:45.52 | CCFL_Man2 | rudholm: how dare you go away from irc for a long period of time and tend to your personal life! :P |
| 23:46.11 | rudholm | haha |
| 23:46.34 | rudholm | CCFL_Man2: well, sometimes I seem to lose my connection to freenode and it takes me a few days to think about reconnecting |
| 23:46.48 | s34n | RazaMetaL: http://en.wikipedia.org/wiki/Runlevel#Ubuntu_Linux |
| 23:46.49 | CCFL_Man2 | rudholm: i heard you took an old WE500 and made it wireless and battery powered |
| 23:46.54 | metfan2007 | any idea on how to store that??? :D Pls! |
| 23:46.59 | CCFL_Man2 | rudholm: ahh |
| 23:47.24 | RazaMetaL | s34n, thnkz.. let me check it |
| 23:47.46 | CCFL_Man2 | rudholm: i wanted to do the same to a northern electric pyramid phone |
| 23:47.50 | rudholm | CCFL_Man2: back in the old days, I had a commercial cordless telephone that was a 500 with a huge telescoping antenna coming out the back. it wasn't something I built, though. |
| 23:48.22 | CCFL_Man2 | rudholm: i heard you made some phone wireless voip and battewry powered |
| 23:48.27 | rudholm | CCFL_Man2: the easiest way I would think, would be to take a cordless and retrofit it into a 2500 |
| 23:48.48 | J4k3 | and my voip provider doesn't support ilbc. |
| 23:49.06 | J4k3 | I'd rather have crappy audio all the time, than crappy audio 7/10ths of the time :P |
| 23:49.18 | CCFL_Man2 | rudholm: you used a digium iaxy or something like that |
| 23:49.18 | rudholm | CCFL_Man2: that does sound like something I *would* do, but I haven't been soldering too much lately. |
| 23:49.39 | CCFL_Man2 | Strom_M told me that you did :P |
| 23:50.00 | rudholm | well, a 500, an IAXy, a wireless ethernet bridge, and some gel-cells would be all you'd need. |
| 23:50.09 | rudholm | since the IAXy supports pulse |
| 23:50.25 | CCFL_Man2 | kick ass |
| 23:50.40 | ManxPower | it might support pulse, but it does not support DNS |
| 23:50.48 | CCFL_Man2 | i'd never want to hard the integrity of the design |
| 23:51.04 | rudholm | CCFL_Man2: there's probably enough room in the 500 for the IAXy and a small wifi bridge |
| 23:51.19 | rudholm | ManxPower: yeah, I know. it's a dumb little thing. |
| 23:51.28 | RazaMetaL | s34n, sorry, is runlevel 3 |
| 23:51.40 | RazaMetaL | s34n, the output of runlevel is N 3 |
| 23:52.45 | *** join/#asterisk Corydon76-dig (i=indigo@pdpc/supporter/sustaining/Corydon76-home) |
| 23:52.45 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 23:52.47 | CCFL_Man2 | rudholm: Strom_M told me you build something like that and had the phone on a table, people sat around the table, the phone rang and one person answered it and gave the phone to you, you grabbed the phone and went into the other room while the people around the table gave you a priceless look |
| 23:53.41 | CCFL_Man2 | walked away with an old phone while people were dumbfounded as to how you did it |
| 23:54.20 | CCFL_Man2 | this is what Strom_M told me, granted the story is probably so twisted with my foggy memory it's nothing like what he told me :P |
| 23:54.24 | *** join/#asterisk bintut (n=bintut@cm47.gamma178.maxonline.com.sg) |
| 23:54.27 | bintut | hello all.. |
| 23:54.51 | bintut | how can i check the last caller from my asterisk logs? |
| 23:55.18 | ManxPower | bintut: I have no idea, but the info is in /var/log/asterisk somwhere |
| 23:55.21 | rudholm | CCFL_Man2: well, now I think I'll have to do this. |
| 23:55.25 | bintut | i mean, the call i received from the pots line |
| 23:56.47 | ManxPower | bintut: asterisk logs ALL calls. |
| 23:57.45 | bintut | ManxPower: i got a /var/log/asterisk/cdr.db but i think it's an sqlite db format |
| 23:58.20 | bintut | ManxPower: i received a call but the caller just hangup the call and i want to know who's the caller |
| 23:59.06 | ManxPower | bintut: any thing else in /var/log/asterisk? |
| 23:59.23 | ManxPower | specifically cdr-csv |
| 23:59.37 | bintut | # ls |
| 23:59.38 | bintut | cdr-csv cdr-custom cdr.db event_log messages queue_log |
| 23:59.55 | ManxPower | did you look in cdr-csv? |