| 00:04.59 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 00:05.31 | [hC] | Anyone familiar with how to set polycom phones to increased verbosity for logging? |
| 00:10.34 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 00:10.50 | TrentCreek | who was touting their hosting services a while back? |
| 00:13.32 | [hC] | I think its possible that polycom has the worst website and customer support that i have EVER seen. |
| 00:13.34 | [hC] | ....ever. |
| 00:16.48 | govtcheez | other than the guy who said that a polycom should work fine with a sandberg if i just open the polycom ports, i can't really say anything about them |
| 00:24.03 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 00:24.13 | *** join/#asterisk simprix (n=simprix@c-24-11-30-113.hsd1.mi.comcast.net) |
| 00:24.58 | drmessano | ~polycommunist |
| 00:24.59 | jbot | A polycommunist is someone who believes Polycom phones can do no wrong.. that Polycom's are so over and above anything else, that what you are using is surely crap, and the mere fact you mention another brand name is of great insult to the channel, the community, and the world. They may also be getting a 10% kickback. |
| 00:26.11 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:26.11 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:27.48 | alrs | guilty |
| 00:29.08 | [hC] | I'm really not that impressed with polycom's garbage firmware. |
| 00:30.05 | alrs | hang on to the zultys |
| 00:30.34 | alrs | shoretel ip100 |
| 00:30.36 | mcab | [hC]: I think it's the log.level.change.$blah parameters |
| 00:30.47 | [hC] | mcab: yeah it is, im just not sure of which to change, and to which value |
| 00:30.52 | [hC] | I think i mightve got it though. |
| 00:31.02 | mcab | [hC]: heh, depends what's going wrong |
| 00:31.10 | [hC] | phone is locking up and rebooting itself |
| 00:31.17 | JT | TrentCreek: i doubt it was me touting them, but i do have hosting services |
| 00:31.27 | [hC] | I think it has -something- to do with BLF, but ive reduced the thing down as far as i can. other sites dont do this. i just dont get it. |
| 00:32.37 | mcab | [hC]: I'm guessing the logs don't show anything useful? I think newer firmwares will do a task and stack dump in some cases |
| 00:32.43 | [hC] | aastra is so close to being a hands down winner. they only have a few dislikes left. |
| 00:33.05 | [hC] | mcab: that would be great if the phone was up to catch it! :) the default logging had nothing useful of course, no.. I think ive got it now though, log level to 0 for like 5 items |
| 00:33.15 | [hC] | mcab: the files are huge now. next time it reboots i should have a good idea of why |
| 00:33.31 | mcab | [hC]: set log.render.level to 0 as well |
| 00:33.34 | [hC] | no thanks to polycom closing the last 3 tickets ive sent in about it, with no response, going back to 2007 |
| 00:33.55 | mcab | although, watch out, you may start loading the phones down with all that logging :-7 |
| 00:34.18 | [hC] | heh yeah, i mean... if the phone dies due to too much logging going on... wtf. |
| 00:34.19 | [hC] | heh |
| 00:35.51 | [hC] | hmm. sweet, onto the next polycom issue.. re organizing someone's sidecars with >60 names listed. in alphabetical order. |
| 00:35.53 | [hC] | ... :| |
| 00:40.07 | *** join/#asterisk Entranced (n=Entrance@191.23.119.70.cfl.res.rr.com) |
| 00:43.27 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 00:46.09 | govtcheez | anybody in here work with cross-connect telephone wire? |
| 00:46.40 | govtcheez | like on frames connected to a pbx? |
| 00:46.54 | *** join/#asterisk Morrocco (n=ivan@189.182.55.161) |
| 00:48.01 | Morrocco | Hi Guys, does anybody have a TimeClock, to track employees when they come in to the office and leave? is there such a thing for asterisk? |
| 00:48.21 | *** part/#asterisk Entranced (n=Entrance@191.23.119.70.cfl.res.rr.com) |
| 00:48.40 | jblack | Not built in. |
| 00:49.05 | jblack | It's not too hard to build yourself, if you dump your cdr data into an sql database |
| 00:49.44 | iceyp | hey guys, I have a problem with DTMF... I have a linksys spa 2102 configured with g729 and inband , DTMF seems to work to external locations just not to the PABX... for instance feature codes such as monitor or blind transfer dont seem to work, and if i call the pabx IVR , dtmf doesnt work locally |
| 00:50.09 | jblack | That's if you want automatic tracking. If you want manual tracking, it's as simple as writing a hundred line script and a dozen lines in a dialplan. |
| 00:50.42 | jblack | iceyp: Try setting rfc2833 as your dtmf protocol in your sip.conf and iax.conf |
| 00:51.04 | JT | govtcheez: |
| 00:51.07 | JT | ~ask |
| 00:51.08 | jbot | well, ask is 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. |
| 00:51.48 | govtcheez | yes? |
| 00:51.49 | iceyp | jblack this is what I have already |
| 00:52.01 | iceyp | if i use inband in sip.conf then its all distorted |
| 00:52.11 | jblack | That's my only advice, to use dtmf2833 |
| 00:52.12 | iceyp | if i dont sent inband on the ATA then its also all distorted |
| 00:52.19 | JT | govtcheez: if you have a question, ask it |
| 00:52.28 | govtcheez | i did |
| 00:52.42 | JT | no, you asked to ask |
| 00:52.48 | JT | that wasn't your real question |
| 00:52.48 | govtcheez | uh |
| 00:52.50 | govtcheez | no i didn't |
| 00:53.03 | JT | govtcheez: i assume you have a question about patching |
| 00:53.05 | *** join/#asterisk stevie_ramjet (n=putnopvu@c-71-228-178-34.hsd1.al.comcast.net) |
| 00:53.05 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
| 00:53.25 | govtcheez | no i don't |
| 00:53.41 | jblack | watches with amusement |
| 00:53.43 | JT | then what's your question? |
| 00:53.47 | govtcheez | see above |
| 00:54.26 | JT | ok, let's say the answer is "yes" |
| 00:54.26 | jblack | Yes is an excellent answer. Full of promise |
| 00:54.37 | govtcheez | is the answer "yes" or are we just being hypothetical? |
| 00:55.03 | JT | i do sometimes work with patching phone lines |
| 00:55.11 | JT | i don't know why you are being so obtuse |
| 00:55.19 | JT | i was hoping i could help |
| 00:55.23 | JT | but nevermind i guess |
| 00:55.26 | govtcheez | i don't know why you're being such a helpdesk hardass |
| 00:55.38 | govtcheez | i just wondered |
| 00:55.38 | JT | i beg your pardon? |
| 00:55.43 | JT | this is not helpdesk |
| 00:55.46 | JT | sorry to say |
| 00:55.56 | JT | no-one is paid to be here |
| 00:56.20 | govtcheez | you're trying to squirt a question out of me so it might as well be |
| 00:56.56 | JT | so you were just asking a pointless question for your own amusement then? i don't get it |
| 00:57.06 | govtcheez | not pointless at all |
| 00:57.20 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:57.20 | govtcheez | it's a fine question |
| 00:57.40 | jblack | I get it. |
| 00:57.50 | jblack | "Government Cheese". As in free handouts. |
| 00:57.55 | JT | what was the purpose of asking it if you have nothing else to ask about it? |
| 00:59.14 | govtcheez | i don't guess there was one |
| 00:59.15 | *** part/#asterisk govtcheez (i=govtchee@pluto.lunarshells.com) |
| 01:00.23 | iceyp | how do i extend the feature timeout, i.e. I have to press *2 to be able to do a transfer, but i have to press the sequence within .5seconds or something |
| 01:00.47 | iceyp | I want to be able to press *, find the 2 key and press 2, takes about a second, not half a second |
| 01:00.53 | stevie_ramjet | iceyp, featuredigittimeout in features.conf |
| 01:01.13 | iceyp | thanks bud |
| 01:01.18 | stevie_ramjet | The default is a really stupid value, like you said. It's like half a second. |
| 01:01.59 | lmadsen | ya... I wish that was changes to something like 2000ms |
| 01:02.04 | *** join/#asterisk Entranced (n=Entrance@191.23.119.70.cfl.res.rr.com) |
| 01:02.11 | *** part/#asterisk Entranced (n=Entrance@191.23.119.70.cfl.res.rr.com) |
| 01:06.47 | iceyp | thanks guys :) |
| 01:14.42 | *** join/#asterisk VoiceCX (n=VoiceCX@216.10.136.139) |
| 01:15.40 | VoiceCX | does anyone know of a good resource for instructions on putting together a fanless system with something like Damn Small Linux, Asterisk, and FreePBX or maybe Tiny TrixBox |
| 01:15.43 | *** join/#asterisk Gwayne (n=Gwayne@cm196.kappa245.maxonline.com.sg) |
| 01:16.55 | Morrocco | does anybody use :Crystal Clear Recording Interface and know if its a good software? I need to log the call logs and also play calls recordings |
| 01:19.09 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 01:25.11 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 01:28.25 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 01:29.10 | jblack | morrocco: Hmmmm. |
| 01:30.06 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 01:41.55 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-24-227.dhcp.embarqhsd.net) |
| 01:44.52 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:48.10 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:49.05 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 01:51.18 | *** join/#asterisk pcrack (n=pcack@202.164.176.106) |
| 01:51.50 | pcrack | guys how much is the fee for setuping an asterisk? |
| 01:53.11 | jblack | Nothing, if you do it yourself. |
| 01:53.32 | jblack | Anywhere from fifty to a few hundred bucks if you get help, depending upon your experience level |
| 01:53.58 | pcrack | i mean..someone wants to pay me to setup them an asterisk, but i dont know how much will i ask them |
| 01:54.23 | jblack | calculate how many hours it will take you, then decide how many dollars an hour you want to get paid. |
| 01:54.51 | pcrack | ic... |
| 01:55.31 | jaytee | then add 10% |
| 01:56.32 | pcrack | ic if it is an basic asterisk feature only? |
| 01:56.54 | jblack | then instead, you calculate how many hours it will take you, then decide how many dollars an hour you want to get paid. |
| 01:57.13 | jblack | pokes jaytee |
| 01:58.34 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 01:59.19 | jaytee | what? |
| 01:59.52 | jblack | you're supposed to say "then add 10%" |
| 02:00.09 | jaytee | oh, sorry. the pasta turned out nasty so I got up to throw it out. |
| 02:00.16 | jaytee | "then add 10%!" |
| 02:00.58 | jblack | =) |
| 02:01.58 | *** join/#asterisk digitalirony (n=eric@216.207.245.1) |
| 02:01.59 | jaytee | and of course if it requires you to drive more than 3 miles you should add on a fuel surcharge too. After all with gas prices the way the are and everyone getting screwed you don't want to miss out on being able to do some of the screwing instead of being the guy who always gets screwed, right? |
| 02:02.06 | digitalirony | Hello |
| 02:02.14 | digitalirony | Digium Tech Support here |
| 02:02.16 | jaytee | hello |
| 02:02.22 | jaytee | you are? |
| 02:02.25 | digitalirony | aye |
| 02:02.30 | jaytee | matey |
| 02:02.48 | jaytee | Digium rocks!!!!!! |
| 02:02.49 | digitalirony | The only one on staff atm actually |
| 02:02.58 | jblack | Dont' forget your wheat surcharge too; hard work makes for empty stomaches, and it's not your fault they're making you hungry. |
| 02:03.13 | digitalirony | sup corydon, file, and anyone else i might know |
| 02:03.16 | voxter | digitalirony: be glad you came in at 7pm and not 2pm, you'd have been bombarded. |
| 02:03.25 | digitalirony | lol |
| 02:03.30 | jaytee | wheat surcharge? oh!!! yeah! I forgot......Hefewiezen FTW!!!! |
| 02:03.48 | jaytee | 2pm? was deion here again? |
| 02:03.49 | digitalirony | voxter: i knows you? |
| 02:04.00 | voxter | digitalirony: maybe? |
| 02:04.08 | digitalirony | do you work with me? |
| 02:04.16 | voxter | digitalirony: i normally go by [hC] and my name's dayton. |
| 02:04.20 | voxter | digitalirony: at digium? haha no. |
| 02:04.21 | digitalirony | did you mean bombarded here or where im at |
| 02:04.27 | digitalirony | oh |
| 02:04.39 | voxter | bombarded here, in irc, yes. |
| 02:04.41 | digitalirony | im running out of e-mail cases :P |
| 02:04.49 | digitalirony | and no one calls this later |
| 02:05.18 | digitalirony | heh |
| 02:05.38 | digitalirony | so wheres all the ops |
| 02:06.26 | voxter | everyone is pretty quiet in here this late, aside of course from the foreign guys who start to pipe up |
| 02:06.35 | digitalirony | ahh |
| 02:06.44 | digitalirony | thats all I ever talk to anymore |
| 02:06.51 | digitalirony | through e-mail too |
| 02:06.55 | digitalirony | they refuse to call me |
| 02:07.02 | jaytee | some nights it's still pretty active in here at this time |
| 02:07.11 | digitalirony | so it takes like 3 weeks to get something fixed |
| 02:07.17 | voxter | Im trying to think of how I can make you answer some questions for me while you're here, but i cant come up with anything.. :P |
| 02:07.27 | digitalirony | :P |
| 02:07.30 | lmadsen | who said op? |
| 02:07.31 | lmadsen | :) |
| 02:07.34 | digitalirony | want to know how to fix phantom rings? |
| 02:07.41 | digitalirony | lief |
| 02:07.46 | digitalirony | whats up |
| 02:07.47 | lmadsen | s/ie/ei |
| 02:07.51 | voxter | that depends, what do you mean by phantom rings? :) |
| 02:07.52 | lmadsen | nada much :) |
| 02:08.08 | digitalirony | calls comming into the asterisk system |
| 02:08.17 | digitalirony | that never originated from an actual end point |
| 02:08.18 | jaytee | the phone! she a ring but a nobody dere! |
| 02:08.27 | voxter | digitalirony: haha. never seen that one. |
| 02:08.38 | digitalirony | lmadsen: never met you but i know you :P |
| 02:08.42 | digitalirony | really? |
| 02:08.42 | voxter | digitalirony: then again i do almost zero analog installs. |
| 02:08.47 | digitalirony | i get calls for that ALL the time |
| 02:08.54 | voxter | digitalirony: so, whats the fix? :) |
| 02:08.57 | digitalirony | easy |
| 02:09.04 | lmadsen | I get lots of calls from poorly configured asterisk systems |
| 02:09.13 | Corydon76-dig | digitalirony: many times, it's the telephone company testing the line |
| 02:09.49 | digitalirony | <PROTECTED> |
| 02:10.03 | Corydon76-dig | Asterisk sees the initial part of the test and thinks the line is being raised, even though ring voltage never comes across |
| 02:10.06 | jaytee | Corydon76-dig, you must have a pretty pro-active phone company then. AT&T here in Indy would just wait for you to call them and complain. |
| 02:10.12 | digitalirony | Corydon76-dig: sometimes....but when they get it every 10 min everyday its not |
| 02:10.35 | Corydon76-dig | jaytee: It's the phone switch doing it, not AT&T specifically |
| 02:10.45 | digitalirony | alls i know is |
| 02:10.57 | digitalirony | the Default Ring_Debounce is kinda low |
| 02:11.08 | jaytee | ah, so as long as the schlubs at AT&T don't actually have to put the donut down and hit a button then they'll test the line? |
| 02:11.15 | digitalirony | im not a programmer....but a good feature would be to make it dynamic |
| 02:11.16 | digitalirony | lol |
| 02:11.31 | voxter | digitalirony: maybe you could tell me why when i have chan_zap.so unloaded, my 60 polycom phones all join a Page() meetme on time, but when it is loaded, random phones never hear the page. Asterisk 1.2 |
| 02:12.01 | digitalirony | voxter: Asterisk 1.2 |
| 02:12.20 | voxter | digitalirony: This is fixed in 1.4 you say? as in, known problem? |
| 02:12.20 | digitalirony | lol |
| 02:12.35 | digitalirony | i don't know I never messed with 1.2 enough to know what was changed yet |
| 02:12.43 | lmadsen | I really like 1.4 |
| 02:12.46 | digitalirony | yeah |
| 02:12.52 | lmadsen | been using it for a long time now |
| 02:13.00 | voxter | I havent upgraded people yet, because their 1.2 installs 'just work' for the most part. |
| 02:13.09 | digitalirony | well |
| 02:13.15 | digitalirony | they might JUST work |
| 02:13.17 | lmadsen | if you know 1.4 well, 1.2 upgrades are NOT that hard |
| 02:13.29 | digitalirony | BUT they still have security issues that aren't tech a problem |
| 02:13.32 | lmadsen | you just need to know how to change applications to functions for the most part |
| 02:13.41 | voxter | im actually planning an upgrade to not only 1.4 but our new gui management thing too. woo! |
| 02:13.45 | voxter | reinstalling the OS |
| 02:13.53 | digitalirony | vortex: drop the gui |
| 02:14.14 | digitalirony | vortex: from what i have seen the gui's make much more problems than they fix |
| 02:14.14 | voxter | digitalirony: no thanks! Ive edited config files on over 100 pbxs by hand for far too long sir! |
| 02:14.20 | Corydon76-dig | If you don't use anything that generates a deprecation warning in 1.2, then you should have no problems upgrading to 1.4 |
| 02:14.30 | lmadsen | guis allow other people to manage the add/move/changes and leaves you to do real development. People who say, "don't use the gui" love busy work |
| 02:14.40 | lmadsen | Corydon76-dig: bingo |
| 02:14.42 | voxter | lmadsen: amen brother. |
| 02:14.44 | lmadsen | and hi :) |
| 02:14.59 | Corydon76-dig | hugs lmadsen |
| 02:15.04 | voxter | lmadsen: i would like to get on to more interesting work than "can you change the name of extension XXX" |
| 02:15.06 | voxter | UGH. |
| 02:15.08 | digitalirony | lmadsen: maybe....but i get so many cases of people using a gui and its the gui thats messing up the system |
| 02:15.14 | voxter | lmadsen: and hi :) (its dayton) |
| 02:15.25 | lmadsen | asterisk gui isn't that bad... a few things I don't like, but all in all, it's a pretty decent system (once you learn it's little quirks, like anything else). The learning curve was very low. |
| 02:15.41 | lmadsen | voxter: [hC]! |
| 02:15.44 | voxter | digitalirony: classic case of people using a gui improperly. think of the damage they would have caused if they tried to do it by hand. |
| 02:15.47 | jaytee | I'd like to have a gui for MAC stuff but the specifics of my installation don't allow for that I think. my dialplan is pretty customized and I don't think *Now or trixbox could handle it. |
| 02:15.48 | digitalirony | lmadsen: the gui is very easy. |
| 02:16.01 | digitalirony | lmadsen but it doesn't let you do things you can do with .conf |
| 02:16.06 | voxter | ok time to go home. |
| 02:16.09 | voxter | take it easy fellas. |
| 02:16.13 | lmadsen | digitalirony: I agree. But it doesn't get in the way like some other GUi systems |
| 02:16.16 | jaytee | nite voxter |
| 02:16.17 | lmadsen | peas |
| 02:16.22 | lmadsen | he's on the west coast :) |
| 02:16.26 | Corydon76-dig | jaytee: It's a shame you haven't figured out func_odbc yet |
| 02:16.28 | lmadsen | it's only 7pm there |
| 02:16.34 | lmadsen | func_odbc ftw |
| 02:16.47 | lmadsen | it's the new hawtness |
| 02:16.48 | digitalirony | well Asterisk gui isn't that bad....its the trixbox and the pbx in a flash shit |
| 02:16.50 | Corydon76-dig | jaytee: it makes it very easy to design your own add/change/delete GUI |
| 02:16.52 | digitalirony | i hat that stuff |
| 02:17.03 | digitalirony | trixbox doesn't even come with gcc installed on it |
| 02:17.10 | digitalirony | and it uses ancient asterisk |
| 02:17.14 | lmadsen | digitalirony: also agree -- for a gui, it's not very intrusive |
| 02:17.43 | lmadsen | digitalirony: also another advantage -- build your own system under it |
| 02:17.49 | jaytee | does it also allow for customization of extensions.conf ? because that trixbox crap wants to manage it all for itself. |
| 02:17.53 | lmadsen | yes |
| 02:18.01 | lmadsen | you just need to know which macro's it expects to use |
| 02:18.08 | digitalirony | lmadsen: build what system under what? lol |
| 02:18.09 | lmadsen | and from there, you know how to step around it quite easily |
| 02:18.20 | Corydon76-dig | jaytee: the purpose of func_odbc is to let you separate data from logic |
| 02:18.32 | digitalirony | Lmadsen: you mean build my own asterisk system under AsteriskNow? |
| 02:18.33 | lmadsen | digitalirony: build your asterisk system + gui on top of your custom installed distro -- I prefer CentOS |
| 02:18.56 | Corydon76-dig | So you have some fairly complex (but static) logic in the dialplan and all the stuff that varies or changes gets put into a database |
| 02:18.57 | lmadsen | I use the asterisknow gui on top of my own installed asterisk system (for certain customers) |
| 02:18.59 | digitalirony | <---debian |
| 02:19.02 | jaytee | I prefer CentOS 5 too |
| 02:19.24 | jaytee | I'm running RHEL 5 64bit though at the moment but my test server is CentOS 5 |
| 02:19.36 | digitalirony | yeah thats what i hate about trixbox is that sometimes when you change something in a .conf it doesn't do anything |
| 02:19.40 | lmadsen | I say use what you know best, in case anyone wanted to start a boring distro discussion |
| 02:19.44 | Corydon76-dig | I wrote func_odbc so that I could query data from a customer's MS SQL Server database |
| 02:19.51 | lmadsen | Corydon76-dig: it works very well for that |
| 02:20.05 | Corydon76-dig | lmadsen: works even better in 1.4 |
| 02:20.18 | lmadsen | I use (in order of use, not necessarily of preference), is MySQL, MSSQL and PostgreSQL |
| 02:20.25 | Corydon76-dig | When I was writing for that customer's system, it was pre-1.2 |
| 02:20.28 | lmadsen | Corydon76-dig: works hawt in 1.4 -- and I like the backport |
| 02:20.39 | lmadsen | Corydon76-dig: wow, I didn't realize it had such a long history |
| 02:20.43 | lmadsen | no wonder it works so f'n well |
| 02:20.44 | jaytee | Corydon76-dig, so can I use func_odbc to take data from mysql cdr and export it to a MSSQL database? |
| 02:20.44 | digitalirony | sos what kind of crap we going to have to go through when 1.6 comes out |
| 02:20.51 | digitalirony | im sure the lines here are going to blow up |
| 02:21.01 | Corydon76-dig | Yep, I wrote it just after the 1.2 feature freeze went into effect |
| 02:21.33 | digitalirony | Cordydon76-dig: have you seen putman and treys' cdr curl? |
| 02:21.38 | lmadsen | Corydon76-dig: I have people ask me if func_odbc has an effect on the asterisk call quality or number of calls, and in my testing, it has zero effect |
| 02:21.40 | digitalirony | trey got it to work |
| 02:21.54 | Corydon76-dig | digitalirony: not yet |
| 02:22.06 | Corydon76-dig | but that's great |
| 02:22.38 | hackeron | just out of interest, if not going with an asterisk solution, how much is a typical 6 ISDN line and 24 FXS port PBX cost? |
| 02:22.40 | digitalirony | Corydon76-dig: it's not bad, could use some polish though. the web page is kind of jumbled and hard to read |
| 02:22.53 | lmadsen | jaytee: that's not really a good usage of it... because you would just export the data from mysql into a table in the mssql database, then have asterisk continue writing via res_odbc and cdr_odbc.conf into the mssql database |
| 02:23.02 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:23.02 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:23.03 | Corydon76-dig | digitalirony: Heh, are they still using my sample page? |
| 02:23.16 | Corydon76-dig | digitalirony: I wrote that as a quick & dirty test platform |
| 02:23.24 | digitalirony | Corydon76-dig: not sure....but it doesn't look very good |
| 02:23.36 | _ShrikE | lmadsen: I run thousands of func_odbc queries daily through my dialplans and have never seen any quality issues either. |
| 02:23.40 | lmadsen | jaytee: a better use would be to lookup what device extension 100 connects to, what voicemail box was associated with extension 100, who was associated with the phone, etc.... |
| 02:23.58 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 02:24.24 | digitalirony | Corydon76-dig: I have been learning some more about perl, but i think im going to drop that to learn php...i have more application for it |
| 02:24.35 | lmadsen | for example, I used func_odbc to write a hot-desking feature which allowed a *very* dynamic asterisk system |
| 02:24.56 | lmadsen | from that I've learned to build a lot of dynamics into systems |
| 02:25.14 | lmadsen | digitalirony: php rocks |
| 02:25.25 | lmadsen | I did perl once... dropped it in favour of php |
| 02:25.26 | jaytee | lmadsen, ah yeah I can see that would be handy. I'll probably start to roll my own gui using it in a few months for the MAC stuff but I don't have voicemail boxes on my * server. I'm routing to Exchange UM for voicemail. |
| 02:25.39 | digitalirony | lmadsen: yeah but im not a programmer...im just a wannabe |
| 02:25.40 | digitalirony | :P |
| 02:25.44 | lmadsen | jaytee: that's the best part about using func_odbc :) |
| 02:25.51 | km2 | digitalirony, once you know one well, you pretty much know the other. i was adamant about perl for a long while, but once i needed to use php, it was almost automatic |
| 02:25.56 | lmadsen | digitalirony: I'm not a programmer either... I'm more of a scripter |
| 02:26.07 | digitalirony | I really need to take some formal classes or soemthing on it so i can have some projects to work on and keep what i learn |
| 02:26.12 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 02:26.13 | lmadsen | I write just enough code to do what I need... everything else is in the DB and .conf files |
| 02:26.25 | Corydon76-dig | What I don't write in C, I tend to write in Perl. |
| 02:26.43 | digitalirony | Cordon76-dig: i figured that |
| 02:26.47 | digitalirony | you like perl :P |
| 02:27.21 | Corydon76-dig | What scared me a few years ago was when I wanted to write a quick little program to check an algorithm, I coded it in C, not in Perl |
| 02:27.41 | digitalirony | I just need to learn how to think in programming i think |
| 02:27.50 | Corydon76-dig | That's right, a quick and dirty program in C... |
| 02:27.55 | Corydon76-dig | and it worked the first time |
| 02:28.19 | digitalirony | like i told you before....i can learn syntax all day long...but until i get in place where i have projects and other people to work with on it....i just won't get anywhere |
| 02:28.48 | Corydon76-dig | Right, it's inspiration that you need |
| 02:28.53 | digitalirony | exactly |
| 02:28.55 | digitalirony | and motivation |
| 02:29.08 | digitalirony | and a good mentor |
| 02:29.10 | lmadsen | * Dialplan functions such as IF(), EXISTS(), ISNULL() |
| 02:29.10 | lmadsen | * Applications GotoIf(), Exec(), ExecIf(), Set(), While() |
| 02:29.15 | lmadsen | grrr |
| 02:29.19 | digitalirony | <PROTECTED> |
| 02:29.34 | digitalirony | heh |
| 02:29.35 | lmadsen | i was writing up my "must knows" list |
| 02:29.35 | Corydon76-dig | Motivation comes from the enjoyment you get from others finding what you've written to be useful |
| 02:29.46 | digitalirony | yeah |
| 02:29.51 | digitalirony | and they don't |
| 02:29.57 | digitalirony | because they don't see it as neat |
| 02:29.58 | digitalirony | lol |
| 02:30.07 | *** join/#asterisk hsv-al (n=sdfsdf@user-24-214-126-81.knology.net) |
| 02:30.10 | hsv-al | . |
| 02:30.12 | lmadsen | Corydon76-dig: I find practically everything you make to be useful and reliable |
| 02:30.15 | digitalirony | Non tech people see some words on the screen and say wow....what is that.... |
| 02:30.25 | jaytee | I need to figure out using ISNULL() so I can trap calls that have CallerID blocked. |
| 02:30.26 | Corydon76-dig | You don't know how awesome it is to find someone really enthusiatically recommending a tool you've written, and not realize that they're talking to the author... |
| 02:30.45 | digitalirony | I can imagine |
| 02:30.51 | digitalirony | IF only |
| 02:31.07 | lmadsen | hrmmmmm |
| 02:31.09 | digitalirony | I just know i don't want to do tech support forever |
| 02:31.16 | digitalirony | I want to go up the third floow |
| 02:31.26 | lmadsen | maybe I'm making these cookbook recipes too complex in my head... |
| 02:31.28 | digitalirony | *floor |
| 02:31.28 | Corydon76-dig | Third floor is sales and marketing |
| 02:31.36 | jaytee | I think it's cool that I can be reading from * TFOT and have the book open on the desk next to me while I'm here in chat talking with one of the authors :-) |
| 02:31.41 | digitalirony | yours on the third floow |
| 02:31.42 | digitalirony | floor |
| 02:31.50 | Corydon76-dig | No, I'm on the second floor |
| 02:31.53 | digitalirony | oh |
| 02:31.54 | digitalirony | see |
| 02:31.57 | digitalirony | im bad with numbers |
| 02:32.03 | lmadsen | I'm on the 16th floor... soon to be 34th floor |
| 02:32.05 | digitalirony | but better with words :P |
| 02:32.23 | Corydon76-dig | Bad with numbers is not a great sign |
| 02:32.24 | digitalirony | but still not good |
| 02:32.34 | jaytee | then you should become a math teacher because your talents as an English teacher would be wasted |
| 02:32.34 | digitalirony | oh im good at math |
| 02:32.41 | digitalirony | just not numbers |
| 02:32.56 | jaytee | your handle suits you then! |
| 02:33.01 | digitalirony | yep |
| 02:33.09 | Corydon76-dig | I gave Mark 3 wooden puzzles, each in the shape of a cube for his 27th birthday... |
| 02:33.10 | digitalirony | thats where i got it |
| 02:33.18 | lmadsen | Corydon76-dig: that sounds cool |
| 02:33.22 | Corydon76-dig | and that is a math joke |
| 02:33.52 | digitalirony | i get it |
| 02:33.54 | digitalirony | see |
| 02:34.10 | digitalirony | 3^3 |
| 02:34.15 | Corydon76-dig | Yep |
| 02:34.20 | digitalirony | i know math |
| 02:34.22 | lmadsen | clever :) |
| 02:34.25 | lmadsen | I suck at math |
| 02:34.28 | lmadsen | I'm good at logic though |
| 02:34.31 | digitalirony | i suck at numbers |
| 02:34.33 | Corydon76-dig | lmadsen: Mark groaned |
| 02:34.39 | digitalirony | like 2+2 = 5 |
| 02:34.50 | russellb | runs in circles and then runs away |
| 02:34.51 | digitalirony | but i can do algebra and and stuff that is solving |
| 02:34.52 | Corydon76-dig | (for extremely large values of 2) |
| 02:36.32 | d00gster | gents, how can I ask * to anchor the media? |
| 02:37.03 | russellb | what do you mean? |
| 02:37.26 | russellb | asterisk has no anchors to drop |
| 02:37.40 | d00gster | nice |
| 02:37.56 | Corydon76-dig | russellb: I think he means, preventing the media from being natively bridged |
| 02:38.17 | d00gster | media has to go through * |
| 02:38.21 | russellb | perhaps, but i didn't feel like guessing, i'd rather just get a straight question :) |
| 02:38.26 | d00gster | not directly between phones |
| 02:38.33 | russellb | for SIP phones? |
| 02:38.38 | d00gster | yes |
| 02:38.41 | russellb | canreinvite=no in sip.conf |
| 02:40.51 | d00gster | ok |
| 02:41.26 | *** join/#asterisk erojasv (n=erojasv@190.43.14.49) |
| 02:46.21 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
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| 02:47.05 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:52.44 | hsv-al | digitalirony |
| 02:52.53 | hsv-al | you need to learn how to pop out PDE's all day long |
| 02:52.59 | hsv-al | before you can use asterisk |
| 02:53.11 | hsv-al | partial differential equations is necessary , in order to do dialplans correctly |
| 02:53.14 | jaytee | what's a PDE? |
| 02:53.27 | jaytee | never mind |
| 02:53.35 | jaytee | just looked at the last line |
| 02:55.15 | russellb | lies! |
| 02:55.17 | JT | you need differential equations? |
| 02:55.40 | russellb | you have to take the 2nd derivative of the dialed extension to understand how asterisk does pattern matching |
| 02:55.43 | jaytee | I think anyone who uses PDE's in their dialplan is overthinking the solution |
| 03:00.08 | *** join/#asterisk bkruse (n=bkruse@phoenixii.total-vision.net) |
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| 03:00.57 | jaytee | now I can't get that song out of my head |
| 03:03.12 | *** join/#asterisk jetlagmk2 (i=jetlag@70.17.48.245) |
| 03:03.50 | *** part/#asterisk pcrack (n=pcack@202.164.176.106) |
| 03:06.46 | raytruz` | LOL |
| 03:08.17 | Strom_M | what song? |
| 03:08.42 | jaytee | "Anchors Away" |
| 03:08.59 | jaytee | you know, the old Navy tune |
| 03:09.29 | jaytee | <d00gster> gents, how can I ask * to anchor the media? |
| 03:09.49 | jaytee | <russellb> asterisk has no anchors to drop |
| 03:10.20 | Strom_M | ah |
| 03:10.27 | jaytee | damn earworms |
| 03:10.43 | jaytee | they say to get rid of it you have to sing the whole song to the end. |
| 03:11.12 | lanning | is that like getting rid of the hiccups? |
| 03:11.29 | jaytee | probably about as reliable a cure, yeah |
| 03:11.50 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 03:12.14 | Strom_C | my connection has been flaking out all day |
| 03:12.35 | jaytee | best sure fire cure for hiccups I've ever found was a tablespoon of sugar with lemon juice and a dash of bitters gulped down real quick. |
| 03:12.58 | Strom_C | that makes no sense |
| 03:13.26 | jaytee | actually it does, it helps release the air trapped in the diaphragm |
| 03:13.42 | digitalirony | Anyone here know why i wouldn't be able to upload a file via ftp to a folder that the ftp user is owner of? |
| 03:13.44 | hsv-al | !qu:303:[jaytee]-I think anyone who uses PDE's in their dialplan is overthinking the solution |
| 03:13.48 | digitalirony | and chmoded to 755 |
| 03:14.06 | Strom_C | isnt 755 rwxr-xr-x? |
| 03:14.15 | hackeron | anyone? if not going with an asterisk solution, how much is a typical 4 PRI and 24 FXS port PBX cost? |
| 03:14.23 | digitalirony | heh |
| 03:14.27 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 03:14.29 | digitalirony | i mean 744 |
| 03:14.36 | Strom_C | hackeron: why 4 PRIs and only 24 FXS ports? |
| 03:14.44 | digitalirony | either way i just looked and its actually 777 |
| 03:14.46 | Strom_C | 92 channels for 24 phones? |
| 03:15.04 | hackeron | Strom_C: yeah, call center |
| 03:15.25 | Strom_C | ook then...call centers usually require specialized PBXes and that can get pricey really fast |
| 03:16.13 | hackeron | Strom_C: well, asterisk does everything :) - just curious how much those old fashioned PBXes are - say for a fairly cheap one, what sort of price range? |
| 03:16.24 | hackeron | Strom_C: $10k? more? |
| 03:16.25 | jaytee | you could get all that from Nortel for about 80K with the ACD package, licensing and the phones or you could roll your own for less than 20K with Asterisk. |
| 03:16.40 | Strom_C | hackeron: depending on how fancy you get, anywhere from $40k-$150k |
| 03:16.49 | hackeron | wow, that's a lot |
| 03:16.56 | Strom_C | that's just a rough rough estimate though |
| 03:17.25 | Strom_C | I hope you've run your extended erlang B formula and definitively know that you need all four of those PRIs :) |
| 03:17.38 | Strom_C | or was that erlang C |
| 03:17.40 | jaytee | yeah, Nortel's M3904 ACD phones are usually about 400 to 500 apiece through your "friendly neighborhood VAR" |
| 03:17.43 | Strom_C | looks again |
| 03:17.46 | hackeron | looking at asterisk hardware the PRI card is $2k, the 24 FXS card is another $2k and the intel xeon powerful enough is what, another 2k-3k -- how on earth would you get from $7k to 80, lol |
| 03:18.01 | hsv-al | what i want more then anything is a sip client that works 100% |
| 03:18.05 | hsv-al | on a blackberry 8703 |
| 03:18.14 | JT | hackeron: you forgot phones |
| 03:18.17 | JT | hackeron: and a built solution |
| 03:18.37 | hsv-al | no one knows why there arent sip clients on blackberrys yet |
| 03:18.47 | hsv-al | its as if they cant interface to the blackberrys hardware, for the audio in/out |
| 03:18.49 | Strom_C | hackeron: also, you seriously don't want to run a call center on analog phones |
| 03:19.14 | jaytee | hackeron, just to get callerid on an analog set on a Nortel Meridian system you need a 2500.00 Class Modem card in your PBX. The commercial big solutions cost big bucks (overpriced from the get go) |
| 03:19.22 | JT | hackeron: but yeah, a commercial traditional PBX will cost way more than a typical asterisk install |
| 03:19.42 | jaytee | yep, way, way more |
| 03:19.46 | Strom_C | ok, thats right, erlang B and erlang C :) |
| 03:20.06 | hackeron | jaytee: hmm, you get the callerid on the PRI card and can't asterisk pass it on to the internal FXS ports? |
| 03:20.25 | JT | hackeron: we're talking about 2 different things here |
| 03:20.33 | JT | hackeron: forget about FXS ports for a callcentre |
| 03:20.36 | *** join/#asterisk plla (n=h@200.31.103.86) |
| 03:20.37 | JT | use sip phones |
| 03:20.42 | jaytee | hackeron, only on their digital sets by default, the analog phones require sending the CallerID in the audio stream as tones. |
| 03:21.31 | hackeron | JT: well, this is a friend's office and he already invested into analog phones but realised POTS and a cheap PBX is not a good idea so wants to replace the PBX but keep the analog phones - is that a bad idea? |
| 03:21.36 | jaytee | yeah, use SIP with Asterisk and a Digium 4 port PRI card and you'll save a ton of dough and have more flexibility and power. |
| 03:21.47 | Strom_C | hackeron: sell the analog phones |
| 03:21.49 | JT | hackeron: yes it's not a good long term solution |
| 03:21.50 | Strom_C | seriously |
| 03:21.56 | jaytee | yeah, sell the phones |
| 03:22.03 | hackeron | lol, ok |
| 03:22.15 | hackeron | those polycom IP650 phones sure look nice |
| 03:22.38 | jaytee | the SIP phones will have access to all of *'s features while the analog phones will be limited. |
| 03:23.00 | Strom_C | hackeron: if you want solid, professional advice on this project (which I'm guessing you might need), I do run a telecom consulting business :) |
| 03:23.11 | hackeron | yes, I'll try to convince him to get sip phones |
| 03:23.40 | hackeron | Strom_C: haha, a bit of shameless self promotion, I like it :) |
| 03:23.48 | hackeron | where are you based? |
| 03:24.00 | plla | Hello, I am having a problem with zapata configuration, apparently I am doing something wrong with the gains, debug yells at me with "chan_zap.c:1575 set_actual_rxgain: Failed to read gains: Invalid argument" |
| 03:24.02 | plla | http://pastebin.ca/1045681 |
| 03:24.08 | Strom_C | I'm in Los Angeles, but this sort of thing isn't really all that dependent on me being there in person |
| 03:24.21 | hackeron | Strom_C: that's true |
| 03:24.22 | plla | I would appreciate some pointers here. |
| 03:24.30 | Strom_C | plla: lemme look |
| 03:24.34 | jaytee | I'm using IP330s at work and just ordered a IP501 and a IP550 |
| 03:25.05 | hackeron | well, for the hardware, I'm thinking just any xeon 1U, the digium 4 port pri card with echo cancelling and polycom IP650 phones - is this more or less on the right track? |
| 03:25.17 | Strom_C | IP650 phones? for a call center? |
| 03:25.18 | Strom_C | overkill |
| 03:25.32 | hackeron | 501? |
| 03:25.32 | Strom_C | but yes, the other stuff is good |
| 03:25.41 | Strom_C | hackeron: 430 is what I'd recommend |
| 03:26.01 | Strom_C | headset port, two line appearances, roomy display |
| 03:26.36 | hackeron | I have a 430 on my desk, I guess it's OK |
| 03:26.57 | jaytee | the sound quality of the Polycoms is by far the best I've tested. |
| 03:27.23 | Strom_C | plla: that looks fine to me, actually |
| 03:27.36 | hackeron | yeah, I played with snom320, useless speakerphone, average sound quality but I couldn't find a wireless headset for any other phone |
| 03:27.37 | Strom_C | but since you're not actually setting gain from default anyway, i wouldnt worry about it |
| 03:27.50 | hackeron | on the polycoms, the wireless headset couldn't hang up or answer the call |
| 03:27.50 | Strom_C | hackeron: wireless headsets are going to be a disaster in the call center |
| 03:27.56 | Strom_C | don't use them |
| 03:28.01 | plla | Should I ignore the debug output? |
| 03:28.16 | jaytee | yeah, wireless headsets would be a support nightmare |
| 03:28.28 | plla | The warning appears even when setting the gains to other values than 0.0. |
| 03:28.40 | jaytee | plla, looks fine to me too. What kind of card is this? |
| 03:28.45 | Strom_C | plla: you've got a PRI. you shouldn't be setting gains anyway |
| 03:28.53 | hackeron | Strom_C: well, it's a call center ish, they get a lot of small calls that are in the queue but there are a couple of office phones and a receptionist |
| 03:29.10 | plla | jaytee: TE110P |
| 03:29.31 | hackeron | Strom_C: btw, what about a normal core 2 quad on one of the better asus boards or should I really get a xeon? |
| 03:29.35 | Strom_C | hackeron: well, ok, you can splurge on the office phones and the receptionist phone, but we're talking about the actual agents' sets here |
| 03:29.46 | plla | No gains? I have worked with other PRIs and for the echo canceler to work I always balance the gains. |
| 03:29.51 | jaytee | try commenting out the rxgain and txgain statements and do a restart |
| 03:30.04 | plla | Though this time that debug output is coming out. |
| 03:30.27 | Strom_C | hackeron: you're probably OK with that, but I would need to do a more thorough traffic analysis to be able to tell you with any certainty. |
| 03:30.45 | jaytee | I'm using the TE212P and I didn't need to adjust the gains on either T1. |
| 03:30.50 | hackeron | Strom_C: hmm, traffic analysis? |
| 03:31.05 | plla | jaytee: Commented out. Same warning. |
| 03:31.39 | Strom_C | hackeron: yes. presumably you know how many peak hour erlangs you can handle... |
| 03:31.48 | plla | I am using Asterisk 1.4.20.1 |
| 03:32.12 | hackeron | Strom_C: hmm, I see :) |
| 03:32.51 | hackeron | Strom_C: ok, I'll speak to my friend, I think he'll probably just say forget it and stick with his analog crap, lol - but hopefully I'll convince him |
| 03:33.06 | *** join/#asterisk isamar (i=1000@voice.maxirede.net) |
| 03:33.10 | isamar | hi folks |
| 03:33.23 | Strom_C | for regular office pbxes, traffic analysis is something you can usually just fudge, but for a call center, you really really want to run all the formulae and make sure you know exactly where you stand with regard to call traffic and hold times and so on |
| 03:33.35 | isamar | looking for a packet loss monitoring tool.... |
| 03:33.40 | isamar | anybody can recommend one? |
| 03:34.04 | hackeron | Strom_C: so hardware looks like it's going to be £7k it seems, how much would your part of setting it up, traffic analysis, etc cost roughly? -- to give him some idea :) |
| 03:34.22 | Strom_C | hackeron: let's discuss that in privmsg |
| 03:34.30 | hackeron | sure :) |
| 03:35.00 | isamar | having quality problems to my ITSP and I need to identify who in middle is messing up.... |
| 03:35.06 | *** join/#asterisk [cfdisk] (n=cfdisk@68-116-156-85.dhcp.ftwo.tx.charter.com) |
| 03:35.08 | jaytee | isamar, have you tried wireshark? |
| 03:35.36 | *** join/#asterisk NovceGuru (n=NovceGur@oh-65-40-70-180.sta.embarqhsd.net) |
| 03:35.52 | isamar | jaytee: I know wireshark .. but AFAIK it's only capture packets... |
| 03:36.09 | jaytee | yeah, but you can filter to only capture SIP and RTP |
| 03:36.16 | isamar | jaytee: I need to figure out which router/hop in the middle is messing me up |
| 03:36.35 | isamar | jaytee: and alert to me... |
| 03:36.50 | jaytee | ah, I see. Yeah, shark wouldn't help you there |
| 03:37.02 | isamar | jaytee: having too much trouble with rtp packet loss f*cking the call quality |
| 03:38.52 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:39.07 | *** join/#asterisk ManxPower (n=manxpowe@221.sub-70-223-226.myvzw.com) |
| 03:41.07 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 03:41.15 | isamar | anybody using nProbe? |
| 03:42.29 | *** join/#asterisk iphonecan (n=Wildman@68.148.0.227) |
| 03:42.51 | *** part/#asterisk iphonecan (n=Wildman@68.148.0.227) |
| 03:42.58 | *** join/#asterisk iphonecan (n=Wildman@68.148.0.227) |
| 03:45.26 | iphonecan | does anybody have experience with 57i CT |
| 03:46.23 | *** join/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 03:48.36 | plla | Hello, can there be two calls with the same CDR(uniqueid) ? |
| 03:49.21 | plla | I am trying to record every call so I would like a unique identifier. |
| 03:49.50 | plla | Can I use CDR(uniqueid) ? |
| 03:50.25 | plla | For the filenames, I mean. |
| 03:51.57 | Strom_C | how about using the epoch instead? |
| 03:52.10 | *** part/#asterisk iphonecan (n=Wildman@68.148.0.227) |
| 03:52.41 | Strom_C | something like caller ID and epoch would have an extremely low probability of duplication |
| 03:52.55 | plla | I usually use this: ${CDR(uniqueid)}-${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)} |
| 03:53.20 | Strom_C | that's fine |
| 03:53.21 | plla | Though I wanted to get rid of the EPOCH since cdr adaptive odbc is a blessing. |
| 03:53.33 | Strom_C | why? |
| 03:54.23 | plla | The filename is going into the cdr, it's going to make it easier web reports with audio files. |
| 03:54.40 | Strom_C | ah, right |
| 03:55.01 | plla | userfield was a pain to manage. |
| 03:55.02 | Strom_C | why not just put the filename into a custom field in the CDR? |
| 03:55.16 | plla | I am using a database. |
| 03:55.33 | Strom_C | so? |
| 03:55.59 | plla | Err, the only custom field you have when using cdr_pgsql or cdr_mysql is userfield |
| 03:56.12 | plla | cdr_odbc too |
| 03:56.29 | jaytee | hahaha, http://i29.tinypic.com/9fv5nq.jpg |
| 03:56.31 | plla | so cdr_adaptive_odbc is going to be handy for me. |
| 03:56.57 | jaytee | it's good to know there are other cynics in the world besides myself. |
| 03:57.52 | Strom_C | plla: it may be worth figuring out how uniqueid is calculated. I don't know offhand myself |
| 03:58.28 | Corydon76-dig | It's the epoch timestamp, plus a monotonically incrementing integer |
| 03:59.02 | plla | Can it be a good choice for a unique filename? |
| 03:59.34 | Corydon76-dig | For a single server, yes |
| 04:00.03 | *** join/#asterisk [cfdisk] (n=cfdisk@68-116-156-85.dhcp.ftwo.tx.charter.com) |
| 04:00.08 | Corydon76-dig | with multiple servers, there's still a possibility of duplicates |
| 04:01.02 | plla | Thanks, that's what I wanted to know. |
| 04:01.52 | Corydon76-dig | Glad to hear you like cdr_adaptive_odbc |
| 04:04.46 | *** join/#asterisk bkruse (n=bkruse@phoenixii.total-vision.net) |
| 04:04.46 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 04:05.40 | jaytee | damn it's late |
| 04:05.45 | jaytee | nite all |
| 04:05.52 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
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| 04:31.52 | codestr0m | Is there a working and maintained stun server floating around? I've checked the voip wiki and nothing there unfortunately |
| 04:34.49 | Strom_C | there are plenty |
| 04:47.57 | rabelais | does asterisk send out the notify message for messages waiting only once? or does it repeat it, and if so, how often does it repeat it? (assuming no new messages have arrived, etc) |
| 04:48.30 | rabelais | I know it sends it on every register as well, exclude that instance as well |
| 04:59.26 | denon | man, why do so few companies make silent PoE switches |
| 05:00.18 | JT | apparently switchmode PSUs of any decent size need non passive cooling |
| 05:00.31 | denon | nod |
| 05:00.50 | denon | it seems they can do 8 ports quietly |
| 05:00.55 | JT | i've seen a silent PoE switch, but it was 8 ports with only 4 * PoE |
| 05:00.57 | JT | yeah |
| 05:01.23 | denon | Im really looking for about 16 gig-e ports, 1 or more SFPs, and yeah, I guess at least 8 of them PoE |
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| 05:01.25 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 05:01.34 | denon | sucker is going to leave a few feet from my head at home |
| 05:02.37 | denon | looks like adding sfp to a silent poe unit is kinda hard too |
| 05:02.55 | JT | get 2 switches? ;) |
| 05:03.22 | denon | yeah, I might have to, but that's a waste of my prime network board realestate |
| 05:03.35 | denon | all my main gear is getting wall-mounted right in my new office, instead of hidden away |
| 05:03.42 | denon | plenty of airflow, easy to glance at, etc |
| 05:11.34 | *** join/#asterisk Der-Tim (n=tkorves@irc.tim.korv.es) |
| 05:12.10 | Der-Tim | hi there |
| 05:15.28 | Der-Tim | does anyone has a linksys pap2(t) connected to an asterisk server? got a problem that the calls are established, but there's no voice connection... both sides can't hear each other... there's no natting between pap2 and asterisk, just a direct connection.. i tried g711u and a as codec, but with no luck... any ideas? |
| 05:16.43 | Strom_C | Der-Tim: i've got a linksys ATA |
| 05:16.52 | Strom_C | never had a problem like that though |
| 05:17.04 | Strom_C | pastebin your sip.conf, mask the passwords |
| 05:17.17 | Strom_C | ~pb |
| 05:17.18 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 05:17.59 | denon | heh, so I was looking at this netgear switch .. |
| 05:18.03 | denon | specs call it fan-less |
| 05:18.17 | denon | but I see one reviewer who bought 2 .. his second one had a fan |
| 05:18.23 | denon | he was thrilled, since it ran so much cooler |
| 05:18.32 | denon | can't please em all |
| 05:22.52 | Der-Tim | Strom_C: http://pastebin.com/d27d3bcef |
| 05:23.07 | Strom_C | damn my client...brb |
| 05:23.23 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 05:24.08 | Strom_C | ok, back |
| 05:24.20 | Strom_C | pastebin is telling me that's not a valid url |
| 05:24.26 | Strom_C | please check it |
| 05:24.40 | Der-Tim | Strom_C: sorry, had to delete it once... ;-) missed to mark password |
| 05:24.44 | Der-Tim | Strom_C: http://pastebin.com/d4e040755 |
| 05:25.07 | Strom_C | why do you have "nat=yes" if the device isn't behind nat? |
| 05:25.09 | Der-Tim | Strom_C: that's the only relevant regarding that ata device |
| 05:25.22 | Der-Tim | Strom_C: good question, haven't seen... ;-) |
| 05:25.35 | Strom_C | you did write this config yourself, right? |
| 05:26.30 | Der-Tim | Strom_C: no. it's (shame on me) freepbx... but as this issue is (imho) server sided, i choosed to ask here at #asterisk |
| 05:26.38 | Strom_C | sigh |
| 05:27.34 | Der-Tim | Strom_C: very strange for me is, that another sip device with the same config (only cid differs) works just fine... |
| 05:28.11 | Strom_C | well, just try that |
| 05:28.20 | Strom_C | is that other sip device also a linksys ATA? |
| 05:28.34 | Der-Tim | no, it's an siemens c450ip dect phone |
| 05:28.50 | Strom_C | well, see, then that's not quite the same |
| 05:28.56 | Strom_C | try nat=no and see if that fixes it |
| 05:30.00 | Der-Tim | Strom_C: but what would the device try, if nat is enabled but not needed? |
| 05:30.08 | Strom_C | ... |
| 05:30.14 | Strom_C | set nat=no in sip.conf |
| 05:30.17 | Strom_C | reload sip on asterisk |
| 05:30.19 | Strom_C | try a call |
| 05:30.32 | Strom_C | gogogo |
| 05:31.21 | Der-Tim | Strom_C: i changed it, but have to wait till my dad can try it (i'm in office right now)... |
| 05:31.48 | Strom_C | ...so you mean you're asking for troubleshooting help and you don't even have the equipment in front of you to test? |
| 05:32.01 | Strom_C | draws a circle on the wall and prepares to bang his head into it |
| 05:34.53 | *** join/#asterisk nicox (n=nicox@212-183-40-93.adsl.highway.telekom.at) |
| 05:35.39 | Der-Tim | Strom_C: why not? i'm struggling with this since yesterday... and all configuration can be done via vpn... but i don't have hardware access... and this thing needs to run asap, because it's an emergency-dialing device for my dad, who can't call the paramedics, if he needs them... so he only has to push a button and he will get help... but the emergency center needs to hear him or speek to him... so this fixing is more than urgent for him... sorry... |
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| 05:46.23 | Strom_C | what amuses me about all this is that you demand reliability, and yet you're running freepbx |
| 05:46.28 | Strom_C | and/or trixbox |
| 05:47.27 | *** join/#asterisk nicox_ (n=nicox@213-33-6-109.adsl.highway.telekom.at) |
| 05:48.40 | nicox_ | hi |
| 05:48.48 | Strom_C | hello |
| 05:48.59 | nicox_ | did anyone tried php-fast-agi? |
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| 06:43.59 | codestr0m | anyone know how to reset a Cisco 7960 with sip firmware.. (I've done it a long time ago), but the directions at http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00800941bb.shtml#topic1 don't see correct and haven't worked thus far.. I want to change the pw..so I can update the tftp server |
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| 06:51.23 | truent | anyone use gizmoproject+asterisk? |
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| 07:12.59 | codestr0m | nobody awake that knows how to reset sip firmware v6 on a cisco 7960.. (I'm trying to avoid the mess of custom network + weird dhcp settings + tftp) |
| 07:13.31 | Strom_C | what are you trying to "reset" exactly |
| 07:13.31 | Strom_C | ? |
| 07:18.16 | JT | codestr0m: mess, as in you don't want to use tftp provisioning? |
| 07:20.24 | *** join/#asterisk af_ (n=getsmart@88-149-230-57.dynamic.ngi.it) |
| 07:21.11 | codestr0m | JT: tftp provisioning is fine, but the old tftp server is a non-local ip I don't have access to anymore.. so I'll have to configure the laptop as a dhcp server with that ip as a tftp. (all offline) I know there's some funky way to just dial the reset command |
| 07:21.35 | Strom_C | codestr0m: or you could just punch the tftp server's IP into the phone |
| 07:21.40 | codestr0m | Strom_C: my phone is locked and I need to reset the pw so I can update some settings |
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| 07:28.23 | codestr0m | ok. well. going to try this alternative route.. thanks guys. later |
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| 07:31.31 | Strom_M | bllllllllllllllllllllllllllllllllllllllllllll |
| 07:34.08 | ThoMe | Strom_M: pscht |
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| 07:35.07 | Strom_M | buhhhhh |
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| 08:00.41 | pputman | Regarding dtmf cid, this asterisk system is properly detecting the dtmf, however it looks like it's doing it a little late. the debug errors say it did not detecct caller id, and then I see the callerid start to come through in the DTMF logs (and hear it being dialed). I've tried toggling cidstart=ring and polarity, I've tried sendcalleridafter=1 and 2, still no luck. Any ideas? |
| 08:00.49 | pputman | This system is in amsterdam, btw. |
| 08:01.17 | Strom_M | what's actually happening on the line? |
| 08:01.25 | pputman | I'll show you the pastebin. |
| 08:01.27 | Strom_M | no |
| 08:01.34 | pputman | Oh you mean the voltage. |
| 08:01.36 | Strom_M | i want to know what's actually happening on the phone line |
| 08:01.43 | Strom_M | not the voltage |
| 08:01.45 | Strom_M | the timing |
| 08:01.53 | Strom_M | what events happen in what order? |
| 08:02.35 | pputman | Strom_M, Well, first the caller id fails (according to asterisk), and then we start sending the dtmf. |
| 08:02.46 | pputman | (for caller id dtmf) |
| 08:03.03 | Strom_M | ...on the line |
| 08:03.05 | Strom_M | not in asterisk |
| 08:03.15 | Strom_M | i don't care about what asterisk is doing at this point |
| 08:03.25 | pputman | Strom_M, I'm not sure how I would determine that |
| 08:04.03 | pputman | If you're asking what I'm hearing when I call it, I hear us dialing dtmf. |
| 08:04.13 | Strom_M | well, what's the ring cadence? at what point does the switch send you caller ID info? |
| 08:04.33 | pputman | it's sending it immediately, and then I hear a ring, and it answers. |
| 08:04.48 | Strom_M | so it sends DTMF /before/ the first ring? |
| 08:04.52 | pputman | yes |
| 08:05.04 | Strom_M | does it reverse polarity before the DTMF? |
| 08:05.24 | pputman | yes |
| 08:05.33 | Strom_M | what kind of FXO card are you using? |
| 08:05.56 | pputman | He's got an 800 |
| 08:06.00 | pputman | tdm800 |
| 08:06.16 | Strom_M | "he"? |
| 08:06.27 | pputman | Strom_M, yeah, working the night shift |
| 08:06.32 | Strom_M | am I debugging by proxy, or are you in front of the system? |
| 08:06.44 | pputman | Strom_M, I'm logged onto it, yes. |
| 08:06.48 | Strom_M | ok |
| 08:06.51 | pputman | I'm not physically in front of it, no |
| 08:06.57 | Strom_M | pastebin zapata.conf and zaptel.conf |
| 08:12.53 | pputman | http://pastebin.com/m6c7bb6ac |
| 08:13.14 | Strom_M | oh jesus christ on a stick |
| 08:13.23 | Strom_M | you're not using freepbx/trixbox are you? |
| 08:13.38 | pputman | Strom_M, btw, this is a trixbox system, but with open source zaptel. If the configuration is okay on zaptel I'm going to look towards upgrading his asterisk to open source. |
| 08:13.50 | pputman | non trixbox |
| 08:13.59 | Strom_M | sigh |
| 08:14.06 | pputman | hey not my choice =) |
| 08:14.56 | Strom_M | there is so much unnecessary shit in this file |
| 08:15.03 | Strom_M | so so so so so so so much |
| 08:15.19 | *** join/#asterisk oej (n=olle@bkkb-gw.voop.net) |
| 08:16.31 | pputman | Strom_M, and I already took out the duplicate channel entries in additional |
| 08:17.53 | Strom_M | yeah, i'm not awake enough to deal with all this crao |
| 08:17.53 | pputman | http://pastebin.com/m6f3f6629 this could also be helpful |
| 08:17.57 | Strom_M | crap |
| 08:18.22 | pputman | alright, thanks anyways |
| 08:18.26 | JT | freepbx dialplans are pretty much unreadable |
| 08:18.32 | JT | they also randomly call some AGIs |
| 08:18.42 | pputman | yeah, but that isn't the dialplan, as much as the zaptel debug |
| 08:20.25 | pputman | I'm guessing the problem is probably related to trixbox, I'm going to talk this guy into moving over. |
| 08:25.00 | hi365 | Google seems to think that random is "lacking any definite plan or order or purpose". I wouldnt really say that about FreePBX's agi's |
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| 08:26.03 | loompek | morning |
| 08:27.06 | JT | hi365: then they're definitely random, they were implemented where completely not in order or having a necessary purpose |
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| 08:29.31 | hi365 | JT: personaly I dont like the strain that the agi's add to the system either, but I dont think that the agi's do anything that can be done from the dialplan |
| 08:30.23 | Strom_M | select * from wangs where length > 5.75 sort by stoned, gay desc limit 186000; |
| 08:30.51 | JT | hi365: pretty sure that whatever the freepbx AGIs do can also be acheived in dialplan |
| 08:30.53 | pputman | ... wow |
| 08:31.43 | pputman | the agis are only half of it too. the gotos and tons of different context/macros jumbled around everywhere is bad enough. |
| 08:32.58 | hi365 | JT: It would be great to have someone on board that can shape up the agi's, as they dont scale very nicley. From what i've seen thought - most of it cannot be done from the dialplan |
| 08:33.43 | JT | hi365: name something |
| 08:34.05 | hi365 | pputman: the dialplans are definatly long. very long. I guess that being that most users only see the web gui, it doesnt bother most people (untill they end up here asking for help!) |
| 08:34.32 | hi365 | JT: regex's, arrays for starters |
| 08:36.15 | JT | what sort of regexs? |
| 08:36.44 | hi365 | how good are you with php? |
| 08:37.14 | JT | i was hoping for a simple answer |
| 08:37.18 | JT | like what do they do |
| 08:37.38 | JT | and i don't really see why they'd need arrays, freepbx does nothing that complex |
| 08:39.03 | hi365 | while i dont agree with this script, it may serve as an example. http://freepbx.org/trac/browser/modules/branches/2.4/core/agi-bin/recordingcheck |
| 08:39.23 | JT | what about dialpeers.agi |
| 08:39.59 | hi365 | i actualy tried to replace it with dialplan, and i might still do it, but then well be back to pputman's poit - very long dialplans (i needed about 15-20 lines for this script) |
| 08:40.57 | hi365 | dialparties is 700 lines of php. even if it where in dialplan it would burn anyones eyes out when they see it on pastebin (and it will probably neeed about 500 lines of dialplan - if its even posible to convert!) |
| 08:41.14 | hi365 | http://freepbx.org/trac/browser/modules/branches/2.4/core/agi-bin/dialparties.agi <----- i would love to see this in dialplan |
| 08:42.40 | pputman | hi365, well mind you I wouldn't know how to clean it up, but it just seems the trixbox dial plan is very unorganized from a reader's perspective. Not just long, but too many goto's, includes, etc... |
| 08:43.11 | hi365 | connot easly defend that point |
| 08:43.20 | JT | most of those lines are comments and php punctuation |
| 08:43.39 | JT | and it could be easily done in way less lines in dialplan if they were designed properly |
| 08:44.51 | hi365 | your probably right that MOST of it could be done in dialplan (i havnt checked). but what about the rest that cant? |
| 08:45.22 | hi365 | from a dev's point of view - if your allready running the script - might as well do someore stuff there |
| 08:45.59 | JT | name something it does that can't be done in dialplan |
| 08:46.06 | JT | don't get me wrong, AGI has its uses |
| 08:46.11 | JT | this is just not one of them |
| 08:46.35 | hi365 | again, i havnt reviwed the script, but by glancing at the links i posted some things done seem simple |
| 08:46.42 | hi365 | like if, else |
| 08:46.47 | hi365 | case |
| 08:47.01 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 08:47.09 | hi365 | try this in dialplan: $fields = explode(':',$sippheader,2); |
| 08:47.09 | hi365 | 90 debug("Setting sipheader ".$fields[0].": ".$fields[1], 4); |
| 08:48.22 | JT | you can manipulate sip headers and write debug messages in dialplan too |
| 08:50.30 | hi365 | check out the follow me stuff - how many line of dialplan would that take? and you WILL end up with some sort of string manipulation that just cant be done in asterisk |
| 08:50.59 | JT | dialplan can read databases, still not seeing the problem |
| 08:51.32 | JT | this is really basic stuff |
| 08:51.41 | hi365 | so if your allready writing a script in a language your comfertable with, might as well do some more stuff |
| 08:51.46 | JT | we're talking about switching things on and off |
| 08:51.53 | JT | maybe taking a number |
| 08:52.15 | JT | i fail to see how that relates to freepbx being poorly structured |
| 08:52.48 | hi365 | again: I havnt reviewed the script. but if its totaly doable in dialplan I would definatly give it a stab some time, as i persoanly much rather long (and cpu efeciant) dialplan that agi's |
| 08:54.12 | hi365 | JT: would you like to help out migrating it? |
| 08:54.14 | JT | i don't understand why FastAGI isn't used at a minimum |
| 08:55.02 | JT | using plain AGI only makes sense for quick and dirty tests |
| 08:55.02 | hi365 | how does fast differ? |
| 08:55.04 | JT | not really, i don't have a vested interest in freepbx atm |
| 08:55.10 | JT | it daemonises the interpreter |
| 08:55.29 | JT | and does not spawn a new interpreter for every single call of every single agi in every call |
| 08:55.34 | JT | similar to FastCGI really |
| 08:56.43 | hi365 | i will open a feature request for that |
| 09:00.14 | *** join/#asterisk Aaton (i=yazz@ghost.230volts.net) |
| 09:02.52 | hi365 | JT: http://www.freepbx.org/trac/ticket/2844 |
| 09:03.42 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 09:04.53 | JT | cool |
| 09:10.24 | Strom_M | macsbug |
| 09:11.52 | hi365 | does XXX in make menuselect mean that the option is selected or that its unavalible? |
| 09:12.29 | Strom_M | unavailable |
| 09:12.52 | hi365 | hmm, what is mysqlclient then? (Depends on: mysqlclient(E)) |
| 09:15.09 | pputman | hi365, I would think it just requires the Mysql database client. |
| 09:15.29 | hi365 | rpm-qa shows mysql-5.0.22-2.1.0.1, so that seems to be installed |
| 09:15.46 | pputman | hi365, did you do a ./configure afterwards? |
| 09:15.58 | hi365 | after i installed mysql? yes |
| 09:16.09 | digitalirony | heh when in make menuselect press "i" for a neat little easteregg |
| 09:16.09 | pputman | hrm no clue |
| 09:16.26 | hi365 | checking for mysql_config... /usr/bin/mysql_config |
| 09:16.36 | hi365 | checking for mysql_init in -lmysqlclient... no |
| 09:16.45 | *** join/#asterisk ikevin (n=Informat@ANancy-256-1-145-123.w90-33.abo.wanadoo.fr) |
| 09:16.51 | ikevin | hello |
| 09:17.07 | ikevin | i've a little problem while configuring moh |
| 09:17.15 | hi365 | digitalirony: sweet |
| 09:17.28 | digitalirony | heh |
| 09:17.40 | digitalirony | su |
| 09:17.42 | ikevin | i define new class in musiconhold.conf and while i use hold function in a call i've a message: get_mohbyname: Music on Hold class 'default' not found |
| 09:17.58 | ikevin | and i never use the default class in my configs files |
| 09:18.07 | ikevin | anyone know this problem? |
| 09:22.55 | *** part/#asterisk digitalirony (n=eric@216.207.245.1) |
| 09:25.12 | hi365 | checking for mysql_init in -lmysqlclient... no <----- why would this be no? |
| 09:30.08 | *** part/#asterisk Aaton (i=yazz@ghost.230volts.net) |
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| 09:33.46 | hi365 | was missing mysql-devel |
| 09:34.53 | JT | hi365: msql is not the mysql database client |
| 09:35.00 | JT | it is the server |
| 09:35.26 | hi365 | the server was in allready installed, installing mysql-devel is what cleared up the issue |
| 09:35.51 | JT | the dependency wasn't for the server... |
| 09:36.12 | hi365 | right |
| 09:36.20 | JT | it was for the client |
| 09:36.33 | hi365 | seems like the client was there as well |
| 09:37.08 | hi365 | (although it got updated...) |
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| 09:48.58 | ThoMe | JT: hi. is it posible with asterisk OR on elseif ? |
| 09:49.11 | *** join/#asterisk af_ (n=getsmart@88-149-230-57.dynamic.ngi.it) |
| 09:49.18 | ThoMe | ,GotoIf($["${sipid}" = "8" OR $["${sipid}" = "11"]?6:8) |
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| 10:02.21 | JT | ThoMe: don't think that's legal |
| 10:02.47 | ThoMe | joerg no, i mean in the IF a AND or OR ? |
| 10:02.49 | ThoMe | aeh JT |
| 10:03.03 | JT | right |
| 10:03.18 | ThoMe | hm |
| 10:03.21 | ThoMe | i need two line? |
| 10:03.22 | ThoMe | s |
| 10:05.28 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 10:05.31 | JT | http://www.voip-info.org/wiki/index.php?page=Asterisk+Expressions |
| 10:05.41 | JT | the documentation is always pretty helpful to read |
| 10:06.08 | ThoMe | JT: muy bien, gracias |
| 10:09.30 | *** join/#asterisk RoyK (n=roy@ip-80-5-149-91.dialup.ice.no) |
| 10:09.34 | ThoMe | JT: exten => 2128293,n,GotoIf($["${sipid}" = "8" | "${sipid}" = "11"]?sipid-eingabe:spy) |
| 10:09.38 | ThoMe | is it correcto or? |
| 10:10.36 | JT | ThoMe: why don't you try it? |
| 10:10.57 | ThoMe | JT: sorry, i am stupid. work now! |
| 10:11.09 | JT | cool |
| 10:11.50 | ThoMe | JT: sorry. |
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| 10:45.49 | codestr0m | For those I'm not boring with my questions.. I can't set a stun server on this Cisco 7960.. can anyone give feedback on 1) chan_sccp for both quality/reliability and 2) does SCCP work better with nat.. I can register and call fine, but inbound isn't working and I don't have control over the router to do port forwarding.. suggestions? |
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| 10:58.25 | ThoMe | hö, why this? |
| 10:58.26 | ThoMe | Jun 12 12:58:01 WARNING[16728]: pbx.c:1720 pbx_extension_helper: No application 'MeetMe' for extension (eingehend, 6094728, 4) |
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| 10:59.36 | DonAlex | ThoMe: That app_meetme is not loaded? |
| 10:59.44 | ThoMe | DonAlex: how i can load this? |
| 10:59.47 | DonAlex | try module load app_meetme |
| 11:00.11 | DonAlex | ThoMe: From the CLI of course.. :) |
| 11:00.34 | s0ck | when asterisk crashes |
| 11:00.38 | ThoMe | Unable to load module app_meetme |
| 11:00.39 | ThoMe | Jun 12 13:00:20 WARNING[16791]: loader.c:326 __load_resource: /usr/lib/asterisk/modules/app_meetme: cannot open shared object file: No such file or directory |
| 11:00.40 | s0ck | where is the best place to find a log... |
| 11:00.45 | ThoMe | DonAlex: you mean this? |
| 11:00.52 | DonAlex | ThoMe: Means the module is missing. |
| 11:01.01 | ThoMe | oh, why this? |
| 11:01.05 | s0ck | i've moved my pbx onto another pstn line and i tried an outbound call and (never seen this before) it locked asterisk |
| 11:01.15 | s0ck | with some mumbo jumbo on screen |
| 11:01.18 | DonAlex | ThoMe: should be in /usr/lib/asterisk/modules |
| 11:01.41 | DonAlex | ThoMe: if it is not and you compile from source then I suggest you did not enable it's build. |
| 11:02.08 | pputman | s0ck, what type of card? |
| 11:02.09 | DonAlex | ThoMe: If you are using a distro I'd check to see what package that is included in |
| 11:02.11 | ThoMe | servetux:/usr/src/isdn/mittwoch/asterisk-1.2.24/apps# ls |grep meet |
| 11:02.11 | ThoMe | app_meetme.c |
| 11:02.17 | ThoMe | is it posible onl.y this compile? |
| 11:03.14 | DonAlex | ThoMe: Not sure to be honest... I am not using 1.2 asterisk.. and have not for a long time.. cannot recall what the build process is like.. |
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| 11:03.22 | DonAlex | Anone else here have a clue though ? |
| 11:03.48 | s0ck | pputman: aex805e (new tdm analogue card) |
| 11:04.21 | pputman | s0ck, what version of asterisk and zaptel? |
| 11:04.52 | ThoMe | why is meetmet not compile if i run "make" ? |
| 11:05.44 | s0ck | Asterisk 1.4.19-1 |
| 11:05.57 | s0ck | Zaptel Version: 1.4.9.2 |
| 11:06.53 | pputman | s0ck, I would compile both the latest builds, and if you have any issues after that email support@digium.com please. |
| 11:07.41 | pputman | but as far as how to tell why asterisk is crashing, you'd probably have to compile it with debugging and get a core dump. |
| 11:08.11 | s0ck | sigh, i only changed the line :| |
| 11:08.24 | s0ck | digium did help me set gain on the other line, might it cause an issue on the new line? |
| 11:08.57 | s0ck | and the only other thing which has changed is my voice prompts, i doubt that would do it... |
| 11:09.00 | pputman | s0ck, anything's possible, there could be a bug with gains. |
| 11:09.12 | s0ck | i will take the gain off the driver and retry now |
| 11:09.26 | pputman | however it shouldn't crash the system |
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| 11:18.25 | s0ck | vfs_ioctl / sys_ioctl |
| 11:18.35 | s0ck | another page full of random looking numbers and calls |
| 11:18.53 | viraptor | hi - could someone tell me what's the current status of ztdummy and related stuff? do I need to think about it on ast-1.4.11 & kernel-2.6.18 if I want conferences to work properly? or should I just load rtc? |
| 11:20.51 | *** part/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 11:21.26 | s0ck | hmm |
| 11:21.29 | s0ck | it aint the line |
| 11:21.46 | s0ck | does it with it unplugged lol |
| 11:21.50 | s0ck | how bizarre |
| 11:22.19 | s0ck | i updated asterisk and zaptel yesterday and voice prompts |
| 11:22.22 | s0ck | musta broke it |
| 11:23.22 | pputman | s0ck, definitely sounds like a zaptel bug. |
| 11:23.30 | pputman | I would get 1.4.11 |
| 11:24.00 | ThoMe | how i can show me all modules |
| 11:24.04 | ThoMe | which loadet on asterisk? |
| 11:24.40 | pputman | from the asterisk cli: show modules |
| 11:26.28 | ThoMe | ok |
| 11:26.28 | ThoMe | thx |
| 11:27.51 | s0ck | ok, thanks pputman |
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| 11:49.40 | flush | yo |
| 11:49.46 | flush | how would i record incoming calls automatically |
| 11:49.59 | flush | i have set up asterisk so when i press 9 before number it records the call, but for incomming calls i dont know.. any idea |
| 11:50.06 | Maliuta | record as in audio record? or CDR record |
| 11:50.28 | s0ck | pputman: 1.4.11 = worked a charm :) |
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| 11:54.39 | pputman | s0ck, awesome. |
| 11:55.02 | flush | Maliuta record like in .wav |
| 11:55.47 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
| 11:56.10 | WildPikachu | is it easy instead of dialing simultaneously out of two zap channels to dial the second if the first one is congested? |
| 11:57.17 | Maliuta | flush: look at monitor() in the book |
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| 11:57.39 | pputman | WildPikachu, yes, you can assign in your zapata.conf a group, like group=1, and then in your dial string, you will just exten => s,1,Dial(zap/g1/18885555555). That way it will dial out the first available channel. |
| 11:57.56 | WildPikachu | yea, i got two groups sorry :) |
| 11:58.09 | WildPikachu | g0 and g1 .... if it dials on g0 and its congested i want it to fall back to g1 |
| 11:58.37 | pputman | WildPikachu, you'd have to write dialplan that checked if group 1 was availabe with chanisavail maybe? |
| 11:58.54 | WildPikachu | ah, thanks ... i was wondering if there was a smart trick ;) |
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| 12:50.47 | jaytee | morning all |
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| 13:05.31 | jack_sparo | hi when i upgraded from asterisk 1.2 to asterisk 1.4 zap stopped working, any idea please? |
| 13:05.44 | lmadsen | did you upgrade zap as well? |
| 13:05.48 | mvanbaak | did you also upgrade zaptel ? |
| 13:05.51 | mvanbaak | lol lmadsen |
| 13:05.59 | yang | if I defined sip as user, i cannot see the lag response , unlike in peer, is there a special command for it? |
| 13:06.11 | lmadsen | mvanbaak: :) |
| 13:06.15 | mvanbaak | yang: qualify=yes |
| 13:06.15 | yang | hi lmadsen |
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| 13:06.22 | jack_sparo | when im trying to upgrade zap it is giving me kernal erros |
| 13:06.25 | yang | mvanbaak: I do have this in all |
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| 13:06.54 | nicox | hi |
| 13:06.55 | [TK]D-Fender | jack_sparo: pastebin is your friend.... make sure you've got the proper headers for the kernel you're running |
| 13:07.07 | nicox | did anyone tried fastagi-php? |
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| 13:07.24 | yang | mvanbaak: just changed all phones from type=friend to type=user and lost the status section |
| 13:07.47 | jack_sparo | lmadsen what u think dude? |
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| 13:09.25 | jack_sparo | you do not appear appear to have the sources for the 2.6.9-34.0.1.ELsmp kernal installed |
| 13:10.16 | [TK]D-Fender | jack_sparo: "yum install kernel*" |
| 13:10.24 | lmadsen | yum install kernel-devel |
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| 13:12.35 | jack_sparo | i already tried it |
| 13:13.01 | jack_sparo | it installs it |
| 13:13.01 | mvanbaak | yang: I see. why did you do that ? |
| 13:13.03 | mvanbaak | hey russellb |
| 13:13.13 | jack_sparo | and when i try to recompile zaptel it gives me the same error |
| 13:13.28 | lmadsen | jack_sparo: unless that is an older kernel that you're runnign than for what is actually out -- you'll need to specify the correct kernel-devel package for the kernel |
| 13:13.39 | russellb | waves |
| 13:13.51 | lmadsen | like: yum install kernel-devel-2.6.9-34.0.1.ELsmp (or whatever the format should be) |
| 13:14.00 | lmadsen | waves at russellb |
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| 13:16.10 | mike-ekim | can someone help me troubleshoot CDR, it is not writing to tables, all teh configuration with usernames and passwords are correct |
| 13:16.16 | mike-ekim | any recommendation or debug tool that I can run? |
| 13:17.12 | lmadsen | mike-ekim: usually when that happens, it's a table configuration problem -- you'll want to check the logs of your DB in order to determine what Asterisk is writing, and what the DB doesn't like |
| 13:17.28 | lmadsen | especially if you're doing it via ODBC - you won't be able to debug that from the asterisk side |
| 13:18.15 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.140.193) |
| 13:18.19 | mike-ekim | but no changes have occured |
| 13:18.29 | mike-ekim | is there any file in asterisk configuration that contains information of how it writes to the tables? |
| 13:18.33 | lmadsen | that is untrue :) |
| 13:18.35 | mike-ekim | cause that is only thing I can think of that was changed |
| 13:18.44 | lmadsen | I didn't understand that you had this working before |
| 13:18.52 | mike-ekim | hehe yeah |
| 13:18.58 | mike-ekim | only things that have changed, are configuration files in /etc/asterisk |
| 13:19.24 | lmadsen | mike-ekim: the source code contains *how* it writes to the tables... |
| 13:19.58 | lmadsen | runs off to do some work for customers |
| 13:20.49 | yang | mvanbaak: I was readng the manual wanted to sorted the things correct, I am changing now the phones back to type=friend |
| 13:23.11 | *** part/#asterisk codestr0m (n=asura@76.74.174.194) |
| 13:27.16 | mike-ekim | oh |
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| 13:56.58 | dieno | exten => _1NXXNXXXXXX,1,Set(TIMEOUT(absolute)=5) exten => _1NXXNXXXXXX,n,Dial,IAX2/rapidvox/${EXTEN} can any please tell me why is that absolutetimeout not working i mean where am i mistaking |
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| 13:59.03 | jack_sparo | hi |
| 13:59.17 | dieno | helo jack_sparo |
| 13:59.31 | [TK]D-Fender | dieno: You want to make that call last for 5 seconds? |
| 13:59.43 | dieno | yup |
| 13:59.58 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 13:59.59 | [TK]D-Fender | dieno: So you call, they answer, and you can talk forever? |
| 14:00.01 | jack_sparo | when i call in the zap channels are not opening, the ivr is not starting, |
| 14:00.01 | jack_sparo | how can i check what is wrong\ |
| 14:00.19 | [TK]D-Fender | jack_sparo: You didn't describe WHAT was wrong. |
| 14:01.06 | dandre | hello |
| 14:01.23 | jack_sparo | when i call in, zap channels are not starting |
| 14:01.26 | jack_sparo | not answering the call |
| 14:01.28 | [TK]D-Fender | jack_sparo: pastebin the CLI output of a failed attempt at verbose 10, and include your zapata.conf and extensions.congf |
| 14:02.05 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 14:02.13 | dandre | I have a strange behaviour: on an inbound call on a zap channel, the call is answer after the dialparty hungup. |
| 14:02.37 | DavidR2008 | is it possible to kill IAX2 channels? |
| 14:02.38 | DavidR2008 | I issue a "core show channels" and I see this line: |
| 14:02.40 | DavidR2008 | Channel Location State Application(Data) |
| 14:02.41 | DavidR2008 | IAX2/192.168.0.92:45 R800488720018395@inc Up Festival(Running in developmen |
| 14:02.43 | DavidR2008 | this channel has been stuck in Festival for almost 18 hours, I tried a "soft hangup IAX2/192.168.0.92:45" but I got: |
| 14:02.44 | DavidR2008 | IAX2/192.168.0.92:45 is not a known channel |
| 14:02.50 | [TK]D-Fender | dandre: Lack of hangup detection, and answered before * knows the line has stopped "ringing" (caught in between rings) |
| 14:03.14 | [TK]D-Fender | DavidR2008: "soft hangup [channel]" |
| 14:03.38 | [TK]D-Fender | DavidR2008: the channel you see there is truncated, and not the full channel name in all ccases. |
| 14:03.48 | dandre | the call isn't answer until the party hungup |
| 14:03.50 | [TK]D-Fender | DavidR2008: to fit the column layout. use "core show channels concise" |
| 14:04.12 | [TK]D-Fender | dandre: Still sounds like a race condition to me. |
| 14:04.23 | dieno | [TK]D-Fender i am looking to do something like that but using something like that exten => _1NXXNXXXXXX,n,Dial(IAX2/rapidvox/${EXTEN},30,L(5000)) because if we use L(5000) it implements call limit after second party receives and in case of absolute it starts when 1st party receives and waiting for 2nd party to receive or drop but limit will goes on as absolute implement |
| 14:04.39 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 14:04.40 | dandre | what do you mean? |
| 14:05.08 | [TK]D-Fender | dandre: your issue is between hangup detection and timing for ringing, etc. |
| 14:05.22 | DavidR2008 | [TK]D-Fender: thx! that soved the soft hangup error, but the channel didn't die. |
| 14:05.38 | [TK]D-Fender | DavidR2008: Try killing each end. |
| 14:06.07 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:06.10 | [TK]D-Fender | dieno: Sorry, you are becoming hard to follow. Try rewording what is not working, and how exactly it is that you WANT it to work. |
| 14:06.12 | *** join/#asterisk nicox (n=nicox@213-33-6-168.adsl.highway.telekom.at) |
| 14:06.38 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 14:06.48 | DavidR2008 | The other end doesn't exist, which is why I have a problem. |
| 14:06.57 | dieno | ok need to implement call limit using AbsoluteTimeout :D for Outbound |
| 14:07.07 | [TK]D-Fender | DavidR2008: Hrm, so soft hangup tried to kill it and jsut failed? |
| 14:07.23 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 14:07.33 | [TK]D-Fender | dieno: pastebin a call attempt at verbose 10 and an Absolute limit set. |
| 14:07.38 | [TK]D-Fender | ~pb |
| 14:07.39 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:07.40 | [TK]D-Fender | ^^^^^^^^^ |
| 14:07.41 | Titanous | is there any way to change the SIP useragent for one peer? |
| 14:07.47 | dieno | ok koool |
| 14:07.49 | dieno | let me |
| 14:07.54 | [TK]D-Fender | Titanous: Nope. |
| 14:08.01 | DavidR2008 | [TK]D-Fender: well it said: Requested Hangup on channel 'IAX2/192.168.0.92:4569-3' but it's still there |
| 14:08.31 | [TK]D-Fender | DavidR2008: Yeah, sometimes things just hang. Can't really advise any further on this beyond suggesting restarting * |
| 14:09.04 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 14:09.48 | jack_sparo | [TK]D-Fender |
| 14:09.55 | jack_sparo | http://www.pastebin.ca/1046004 |
| 14:10.00 | jack_sparo | <PROTECTED> |
| 14:10.02 | DavidR2008 | [TK]D-Fender: I got it from the other direction, I could see that it was stuck in festival so I ps -e found the festival pid and killed that, then the channel went away. |
| 14:10.15 | DavidR2008 | thx for the help! |
| 14:10.16 | [TK]D-Fender | DavidR2008: :) |
| 14:10.35 | [TK]D-Fender | DavidR2008: Dirty, and I feel just a little slow for not having immediately come to that idea myself... |
| 14:10.43 | jack_sparo | [TK]D-Fender dude? |
| 14:11.00 | DavidR2008 | [TK]D-Fender: sometimes you have to play dirty ;-) |
| 14:11.08 | [TK]D-Fender | jack_sparo: FreePBX is NOT supported in here. |
| 14:11.28 | *** join/#asterisk viperdude (n=jon@195.74.96.122) |
| 14:11.29 | jack_sparo | regardless freepbx dude |
| 14:11.32 | [TK]D-Fender | DavidR2008: Totally. |
| 14:11.35 | jack_sparo | i have zap problems |
| 14:11.42 | viperdude | hi guys |
| 14:11.48 | viperdude | any ideas what causes chan_sip.c:1921 retrans_pkt: Maximum retries exceeded on transmission |
| 14:11.53 | viperdude | errors? |
| 14:12.10 | [TK]D-Fender | jack_sparo: You've shown no debug, no configs, or much else. |
| 14:12.31 | [TK]D-Fender | viperdude: * trying until it gives up. typically a networking issue |
| 14:12.32 | jack_sparo | but when i call in |
| 14:12.34 | jack_sparo | nothing happens |
| 14:12.50 | jack_sparo | how can i check if zaptel is running or not? |
| 14:13.00 | [TK]D-Fender | jack_sparo: "zap debug" <- |
| 14:13.01 | viperdude | [TK]D-Fender: this happens as soon as the called party answers |
| 14:13.17 | [TK]D-Fender | viperdude: probably a reinvite type issue |
| 14:13.22 | jack_sparo | No such command 'zap debug' (type 'help' for help) |
| 14:13.27 | [TK]D-Fender | viperdude: NAT involved I'd bet |
| 14:13.28 | viperdude | but there is no latency or dropped packets and only happens with Funkwerk IP50 phones |
| 14:13.39 | [TK]D-Fender | jack-turn up core debug to 10 and retry |
| 14:13.42 | viperdude | no NAT it is a PWAN |
| 14:13.54 | [TK]D-Fender | viperdude: what PWAN? thats a new one for me... |
| 14:14.01 | jack_sparo | it is set verbose 10 |
| 14:14.06 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:14.07 | viperdude | Private WAN |
| 14:14.17 | [TK]D-Fender | jack_sparo: DEBUG 10, not VERBOSE 10. |
| 14:14.25 | jack_sparo | how |
| 14:14.30 | [TK]D-Fender | viperdude: same subnet? |
| 14:14.39 | viperdude | no diffrent subnets |
| 14:14.39 | *** join/#asterisk kombi (n=kombi@port-87-234-216-47.static.qsc.de) |
| 14:14.40 | [TK]D-Fender | jack_sparo: "set debug 10" <- |
| 14:14.50 | [TK]D-Fender | viperdude: Check your localnet clause. |
| 14:15.19 | viperdude | phones 10.220.11.0/24 asterisk 192.168.1.0/24 |
| 14:15.33 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff-emeritus/madkiss) |
| 14:15.35 | Madkiss | hi all. |
| 14:15.40 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:15.54 | [TK]D-Fender | jack_sparo: And provide your configs. |
| 14:16.18 | kombi | how do I know the gateway accepted me after an iax2 reload? |
| 14:16.21 | Madkiss | just a simple question; if somebody is calling me, and i am calling at that moment, i hear that somebody is calling by the knocking ... |
| 14:16.31 | [TK]D-Fender | kombi: "iax show registry" |
| 14:16.35 | [TK]D-Fender | kombi: "iax2 show registry" |
| 14:16.36 | viperdude | [TK]D-Fender: wjhat should i set localnet to? the 10.220.11.0 or 192.168.1.0 ? |
| 14:16.38 | Madkiss | what i want to achieve is that if I don't answer that knocking within five seconds, the caller should be redirected to another queue |
| 14:16.43 | kombi | thanks fender! |
| 14:16.50 | [TK]D-Fender | viperdude: Each on their own line. |
| 14:16.57 | viperdude | aha ok |
| 14:17.06 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:17.28 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.134) |
| 14:18.06 | [TK]D-Fender | Madkiss: Is htis first call coming from a Queue? |
| 14:18.36 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:18.37 | Madkiss | [TK]D-Fender: what do you mean exactly? |
| 14:18.53 | jaytee | wonders how [TK]D-Fender manages to make a living since he spends so much time in here helping people for free. |
| 14:18.56 | [TK]D-Fender | Madkiss: Hos is this first call reaching you? What is "knocking"? |
| 14:19.09 | jack_sparo | [TK]D-Fender not working |
| 14:19.26 | [TK]D-Fender | jack_sparo: And you haven't provided any of the other things I've requested. |
| 14:19.34 | Madkiss | [TK]D-Fender: I heare that "knock"-sound which tells me that there is somebody trying to call me while I am having a call myself |
| 14:19.55 | [TK]D-Fender | Madkiss: you mean your phone's |
| 14:20.03 | [TK]D-Fender | "call waiting" indicator? |
| 14:20.08 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:20.09 | Madkiss | eeeeexactly. |
| 14:20.27 | [TK]D-Fender | Madkiss: So the call is arriving via a normal "extension" in your dialplan? |
| 14:20.30 | Madkiss | i want to configure asterisk so that if I don't answer the waiting call within 5 seconds, the caller is redirected to another group |
| 14:20.34 | Madkiss | [TK]D-Fender: yes. |
| 14:21.19 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:21.31 | *** join/#asterisk xnosx (n=xnosx@212.145.172.127) |
| 14:21.41 | [TK]D-Fender | Madkiss: Then before you dial your phone, use "Chanisavail to see if you are on the phone. If so, dial with 5 sec timeout, then move on toa Queue, etc. If not, dial with normal timeout and send to wherever else you'd want it to do if unanswered. |
| 14:22.03 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:22.12 | Madkiss | [TK]D-Fender: that is neat. |
| 14:23.17 | *** join/#asterisk PodMan99a (n=PodMan99@78-86-189-73.zone2.bethere.co.uk) |
| 14:23.36 | kombi | How do I debug an iax connection? Can one test one's credentials somehow? |
| 14:23.50 | PodMan99a | hey all im gettiong Got SIP response 500 "Internal Server Error" back from ..... from my asterisk server... two polycom 430's asterisk is in remote location so using nat |
| 14:24.06 | PodMan99a | any ideas? |
| 14:24.47 | [TK]D-Fender | PodMan99a: You can ignore those. Polycom's tend to spit those out after having transfered a call or something-or-other. Its harmless. They do tend to clear up after a while though. |
| 14:25.21 | jack_sparo | asterisk1*CLI> zap show channels |
| 14:25.23 | PodMan99a | yea problem is when i answer the call cli shows ive answered and hold music stops but the call is still saying ringing on my phone |
| 14:25.23 | jack_sparo | No such command 'zap show' (type 'help' for help) |
| 14:25.25 | jack_sparo | sorry |
| 14:26.34 | PodMan99a | nat and sip suck or is it me? |
| 14:26.45 | *** join/#asterisk mosty (n=mosty@60-241-198-194.static.tpgi.com.au) |
| 14:27.01 | [TK]D-Fender | jack_sparo: "load chan_zap.so" |
| 14:27.11 | *** join/#asterisk Dio_ (n=dima@77-109-24-97.dynamic.peoplenet.ua) |
| 14:27.28 | [TK]D-Fender | PodMan99a: if SIP is involved, you do have to set up * to accomodate : |
| 14:27.30 | [TK]D-Fender | ~sipnat |
| 14:27.30 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:27.32 | [TK]D-Fender | ^^^^^^^^^^ |
| 14:27.42 | jack_sparo | http://www.pastebin.ca/1046018 |
| 14:27.57 | PodMan99a | umm... thanks [TK]D-Fender will investigate |
| 14:28.21 | Dio_ | hello, is there murf around? |
| 14:28.40 | [TK]D-Fender | jack_sparo: that doesn't tell me that you've even initialzed zaptel or configured your channels... |
| 14:29.47 | fskrotzki | ~book |
| 14:29.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:29.51 | Dio_ | or may be anybody who is in contact with him? |
| 14:30.14 | Madkiss | [TK]D-Fender: do you have an example for the isavail-stuff? |
| 14:30.29 | [TK]D-Fender | Madkiss: "core show application ChanIsAvail" |
| 14:31.39 | Madkiss | [TK]D-Fender: this doesn't perfectly help me. hm |
| 14:32.07 | [TK]D-Fender | Madkiss: How so? Read the instructions It will tell you what it returns so you can see if you're on a call already. |
| 14:32.53 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 14:33.11 | anonymouz666 | [TK]D-Fender: do you know if SPA3102 can switch the FXS to FXO line through a code? Just pickup the FXS dial a code and then get the tone |
| 14:33.32 | mosty | anonymouz666, that device has a dialplan of sorts |
| 14:33.46 | mosty | i'm not sure how powerful it is |
| 14:33.48 | [TK]D-Fender | anonymouz666: Think so. |
| 14:33.57 | [TK]D-Fender | anonymouz666: Its pretty powerful. |
| 14:39.36 | James|TCC | afternoon guys :) |
| 14:39.41 | *** part/#asterisk Dio_ (n=dima@77-109-24-97.dynamic.peoplenet.ua) |
| 14:39.57 | anonymouz666 | yeah, this is important becase you get a call in FXO line and don't need to forward to asterisk box. Just forward to the local FXSs. It has the bad sides (when the call don't going through *, like monitor etc) and at least you don't miss the call if the internet goes down. |
| 14:41.13 | *** join/#asterisk ^shark_ (n=^shark_@41.222.2.65) |
| 14:42.29 | ^shark_ | is it possible to connect my asterisk box to an analog telephone line or telephone phone, and what else would i need? |
| 14:42.34 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:43.48 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 14:45.48 | flush | ^shark_ you need a fxo module to plug in the wall line and fxs module(s) to plug your normal phones in |
| 14:47.41 | kombi | voovox -> I'm doing the simplest test: try connecting with x-lite and it keeps giving me 404, what is wrong? |
| 14:48.15 | *** join/#asterisk BBHoss (n=hoss@c-68-62-175-86.hsd1.al.comcast.net) |
| 14:49.02 | mosty | "user not found" |
| 14:49.27 | ^shark_ | flush: thanks mate. |
| 14:49.42 | kombi | mosty: log says "unknown host".. |
| 14:50.08 | flush | ^shark_ np.. i think a good card is TDM400P |
| 14:50.23 | flush | you can have 4 modules on it, like 1 phone line with 3 phones or 2 phone lines with 2 phones and stuff |
| 14:50.51 | mosty | kombi, can you ping the host? |
| 14:51.02 | kombi | like a charm.. |
| 14:51.32 | mosty | run tshark on the asterisk box, verify that you can see the incoming sip packets |
| 14:51.58 | BBHoss | kombi: i'm late to the party, whats the problem? |
| 14:52.30 | kombi | can't connect to voovox, neither with * or softphone, keep getting 404s |
| 14:52.52 | BBHoss | got a sip trace? |
| 14:53.15 | kombi | BBHoss: what's that? |
| 14:53.23 | BBHoss | guess not :) |
| 14:53.42 | BBHoss | the output of trying to call or connect when "sip debug" is enabled |
| 14:53.58 | kombi | oh, ok, I try that |
| 14:54.08 | BBHoss | also the console logs |
| 14:54.14 | BBHoss | pastebin them |
| 14:55.47 | James|TCC | how do i configure IMAP_STORAGE, it says it relies on imap_tk, which i cant find |
| 14:56.13 | mosty | you need to download and compile the imap package from washington.edu |
| 14:56.14 | James|TCC | is that a part of asterisk, or an additional app? |
| 14:56.19 | James|TCC | ok thanks |
| 14:56.32 | mosty | i think it's mentioned in the readme file in the asterisk source |
| 14:57.14 | flush | yo any ways to have nfs setup on asterisk so i can share my recorded calls over the lan |
| 14:57.17 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:57.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:57.17 | mosty | it's compiled into asterisk statically, so you need to point the configure script at the dir you compiled in |
| 14:57.49 | mosty | flush, you can just setup nfs or samba for the correct location, asterisk won't know or care |
| 14:58.34 | *** join/#asterisk xnosx (n=xnosx@212.145.172.127) |
| 14:58.35 | James|TCC | the word 'imap' doesnt appear in the lastest readme mosty :( |
| 14:59.33 | mosty | then look in ./configure --help, and google for washington.edu imap |
| 14:59.36 | flush | hrm.. it has to have a nfsd running so it can share files |
| 14:59.37 | flush | no ?? |
| 15:00.07 | PodMan99a | [TK]D-Fender, thanks for the SIP tip works BRILL NOW!! thanks |
| 15:00.14 | mosty | flush, yeah but that's completely separate to asterisk. you just tell nfsd which directory to share |
| 15:00.22 | flush | yar kk |
| 15:00.36 | flush | cause i did the mistake to install asterisk@home.. i dont know if its missing stuff so i can compile nfsd |
| 15:00.43 | flush | sorry im a newb |
| 15:01.00 | mosty | i have no idea what you're talking about now |
| 15:01.00 | *** join/#asterisk philippel (n=p_lindhe@pool-71-164-18-224.sttlwa.fios.verizon.net) |
| 15:04.15 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:06.08 | *** join/#asterisk ManxPower (n=manxpowe@169.sub-75-200-179.myvzw.com) |
| 15:07.14 | *** join/#asterisk nephfl (n=none@wsip-70-168-186-225.ga.at.cox.net) |
| 15:07.24 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 15:07.36 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 15:07.53 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 15:07.57 | nephfl | hello, im using vtwhite and my ivr dies after a few seconds it says it hung up...im not sure how to resolve it..can someone help me out? |
| 15:10.36 | *** join/#asterisk mintee (n=mintone@75.150.132.150) |
| 15:10.42 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:10.53 | ManxPower | Never heard of vtwhite |
| 15:11.14 | nephfl | vtwhite is the wholesale provider for viatalk |
| 15:11.21 | mintee | what's currently proper? Goto(someexten,s,1) or Goto(someexten|s|1) |
| 15:12.06 | kombi | this is driving me nuts, there is only host, user and passwd. Is voovox trash? |
| 15:12.46 | kombi | mintee: no more pipes allowed in 1.6 |
| 15:13.00 | mintee | kombi, cool, thanks |
| 15:15.39 | kombi | how do you telnet into sip? |
| 15:16.54 | BBHoss | heh, its UDP, you'd need to use netcat for that |
| 15:18.36 | *** join/#asterisk defswork (n=andy@mx1.3gcomms.co.uk) |
| 15:20.56 | *** join/#asterisk pLr (n=pLr@unaffiliated/plr) |
| 15:22.32 | *** join/#asterisk grEvenX (n=even@1mldj72.ip.ssc.net) |
| 15:23.50 | *** join/#asterisk mike-ekim (n=mike@adsl-072-151-207-108.sip.mia.bellsouth.net) |
| 15:23.57 | mike-ekim | what does it mean to: 1 - Set your RxCodec to 3 |
| 15:23.57 | mike-ekim | 2 - Set your TxCodec to 3 |
| 15:23.58 | mike-ekim | 3 - Set your LBRCodec to 3 |
| 15:23.58 | mike-ekim | 4 - Set your AudioMode to 0x00140014 |
| 15:24.23 | mike-ekim | what changes need to be made to sip.conf for this?? |
| 15:25.18 | BBHoss | what is txcodec3? |
| 15:25.37 | kombi | which port does sip authenticate on? |
| 15:26.04 | BBHoss | 5060 |
| 15:26.17 | BBHoss | UDP |
| 15:26.18 | *** join/#asterisk dlynes (n=chatzill@209.52.60.113) |
| 15:26.23 | BBHoss | telnet will not work |
| 15:26.59 | dlynes | Anyone know what the error 'check_auth: username mismatch, have <4923049>, digest has <something_else>' means? |
| 15:28.01 | dlynes | And then 'handle_request_invite: Failed to authenticate user "Caller ID Name" <sip:calleridnum@domain.com;tag=3098423' |
| 15:28.43 | BBHoss | well, obviously, the auth is failing |
| 15:28.58 | BBHoss | thats weird looking though |
| 15:29.57 | dlynes | b b Hoss: I was looking for something other than the blatantly obvious...I know the authentication is failing....I just don't know why |
| 15:30.58 | dlynes | I can call into the phone; I just can't dial out on it...and I'm thinking the caller id is being interpreted as teh username for some reason |
| 15:31.05 | BBHoss | hmm |
| 15:31.30 | BBHoss | have you tried a basic dial command from the cli, bypassing extensions.conf? |
| 15:31.43 | dlynes | huh? |
| 15:31.49 | ManxPower | Do you have more than one phone behind a NAT? |
| 15:32.08 | dlynes | ManxPower: yes....I've got several mediatrix boxes connected to an asterisk box |
| 15:32.13 | BBHoss | like "originate SIP/12565551212@myprovider application playback tt-monkeys" |
| 15:32.21 | ManxPower | BBHoss: the CLI dial command only works if you have a sound card installed and all the Asterisk libs are istalled |
| 15:32.22 | dlynes | ManxPower: then i've got the asterisk box placing sip calls to another asterisk box that's not behind a firewall |
| 15:32.30 | BBHoss | i mean originate |
| 15:32.45 | ManxPower | dlynes: maybe if you pasted the ACTUAL error? |
| 15:34.26 | dlynes | [Jun 12 08:35:09] WARNING[14092]: chan_sip.c:8377 check_auth: username mismatch, have <7783730263>, digest has <hamlets_ws> |
| 15:35.04 | BBHoss | dlynes: have you tried the originate command? |
| 15:35.22 | dlynes | [Jun 12 08:35:09] NOTICE[14092]: chan_sip.c:13815 handle_request_invite: Failed to authenticate user "Guest Suite 2" <sip:7783730263@domain.com;tag=as25c4c5ed |
| 15:35.49 | iratik | permission to ask a n00b question? I've got 2 sip trunks setup with two other providers, just signed up with vtwhite and voipinvite... on both of them I don't get full duplex audio, its only one way. I would usually assume that must be the firewall, but then why would the other 2 sip trunking providers be working fine, rtp ports are open ... I can't find my asterisk bible, and either way... what is a good process for debugging this? ... |
| 15:35.49 | iratik | core set verbose gives way too much info ... I can paste the sip show peer information .... any ideas ? |
| 15:36.10 | dlynes | b b hoss: no; I don't have alsa or oss libs installed, a speaker, a microphone, or any of that stuff; it's all running on a headless box |
| 15:36.24 | BBHoss | ~book |
| 15:36.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:36.38 | ManxPower | dlynes: is the incoming call from another Asterisk server, or from some other device? |
| 15:36.47 | dlynes | ManxPower: from another asterisk box |
| 15:37.01 | BBHoss | dlynes you don't need that for originate |
| 15:37.10 | dlynes | ManxPower: it's dialing SIP/peername/${EXTEN} |
| 15:37.12 | ManxPower | dlynes: you need a fromuser= on the remote box, set it the same to whatever this box is expecting |
| 15:37.18 | beek | iratik: Typical NAT problem |
| 15:37.23 | beek | ~book |
| 15:37.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:37.52 | ManxPower | dlynes: Yup, that's what you need to do. without fromuser= the sending asterisk will use the username in the callerid info, and since that info varies with each call..... |
| 15:38.44 | dlynes | ManxPower: beautiful...that's exactly what i needed |
| 15:38.48 | dlynes | ManxPower: thanks, man |
| 15:39.27 | dlynes | ManxPower: I'm used to doing pure analog installs with sip phones |
| 15:39.40 | dlynes | ManxPower: but now i'm building care homes with all analog extensions and no analog lines |
| 15:39.48 | BBHoss | heh |
| 15:39.58 | dlynes | ManxPower: everyything's all sip trunks |
| 15:40.17 | ManxPower | dlynes: InterAsterisk using SIP is only slightly more comlicated than using IAX2 |
| 15:40.24 | ManxPower | There is no such thing as a SIP trunk |
| 15:40.50 | BBHoss | dlynes: BTW you don't need ALSA or OSS to use originate, I have a headless debian server in a colo that it bare-bones, and i run originate all the time |
| 15:40.59 | BBHoss | dial, yes, originate, no |
| 15:41.20 | ManxPower | BBHoss: 1.4 was the first version with a CLI originate command |
| 15:41.27 | BBHoss | yep |
| 15:42.11 | BBHoss | is dlynes using 1.2? |
| 15:42.30 | dlynes | ManxPower: sip trunk as in simulating a PRI using SIP...they're basically running a dedicated, leased network to the telco |
| 15:42.33 | ManxPower | dlynes: feel free to send me large sums of money via PayPal to eric@fnords.org |
| 15:42.59 | ManxPower | dlynes: Here we call those sip peers |
| 15:43.28 | *** join/#asterisk tobias (n=tobias@cpe-069-134-205-184.nc.res.rr.com) |
| 15:43.34 | dlynes | BBHoss: no...using 1.4; we could probably get by with 1.2, for that matter...not using blf or anything like that |
| 15:44.03 | dlynes | ManxPower: yeah...that's what I call them too...I just got accustomed to calling them sip trunks because that's what this particular provider calls them |
| 15:44.07 | BBHoss | dlynes: just thought i missed something since ManxPower said something about 1.4 being first for originate |
| 15:44.39 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:44.40 | ManxPower | dlynes: the provder could call them "Zebras" and be just as technically accurate as calling them "SIP Trunks". |
| 15:44.43 | ManxPower | ~trunk |
| 15:44.43 | jbot | trunk is, like, a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 15:44.43 | dlynes | BBHoss: that's probably because there's a lot of asterisk boxes out there with large installs that sys admins refuse to downgrade to 1.4 |
| 15:44.45 | BBHoss | originate can help alot, you can pin down where your problems are a bit easier, rule out your config |
| 15:44.56 | BBHoss | dlynes downgrade? |
| 15:45.01 | iratik | beek: Thanks for the link to the book, but the book didn't help me any ... I searched for all references to NAT and Network Address Translation, there are two instances , on p190 and p185 .. they just describe what nat is and why it is a problem for sip ..... I don't even have an actual router ... there is a firewall.. but no NAT involved |
| 15:45.18 | dlynes | BBHoss: I've found in general 1.4 is a hell of a lot more unstable than 1.2 |
| 15:45.27 | ManxPower | BBHoss: Many people, myself included, think 1.4 is less stable than 1.2 |
| 15:45.41 | dlynes | BBHoss: I've only been using 1.4 because 1.2 didn't have the features I needed |
| 15:45.49 | iratik | there is a router on our network, but the PBX .. isn't using it as a gateway ... its using a direct connection to our T1 gateway... (which is a cisco router) ... but its not doing NAT |
| 15:46.04 | BBHoss | i haven't had any trouble with 1.4, never used 1.2 except on shitbox, but i don't use zaptel. often zaptel is usually the culprit for bad bugs |
| 15:46.07 | dlynes | BBHoss: but with all these mediatrix installs, I don't really need 1.4 |
| 15:46.26 | dlynes | BBHoss: no...lots of stuff in 1.4 is buggy...not just zaptel |
| 15:46.48 | jblack | I'm still getting complaints about "noise" from the pri. It doesn't like noise to me, but more like dropped packets. I have some representative calls at http://linuxguru.net/~jblack/calls/ . I would love some useful suggestions |
| 15:46.57 | ManxPower | ~ecfo |
| 15:46.58 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly realises its a crappy design based on a half baked 20 year old apps note. |
| 15:47.11 | ManxPower | jblack: ~ecfo was for you |
| 15:47.13 | BBHoss | dlynes i must just be lucky then |
| 15:47.15 | flush | yo got question |
| 15:47.20 | jblack | reads |
| 15:47.36 | ManxPower | jblack: could easily be a sync source issue. Do you have a 1 in the end field of any of your spans? |
| 15:47.46 | jblack | checking |
| 15:47.47 | dlynes | BBHoss: do you use shared line appearance? blf? have any large installs? have a lot of sip phones? |
| 15:48.02 | ManxPower | dlynes: BLF works in 1.2, we use it |
| 15:48.03 | Qwell | dlynes: have you reported any bugs? |
| 15:48.13 | jblack | span=1,1,0,esf,b8zs |
| 15:48.16 | BBHoss | i dont use SLA or BLF, not really any big installs (like over 20 phones) |
| 15:48.29 | dlynes | Qwell: yes |
| 15:48.39 | jblack | And misconfiguration of the rhino r1t1 is very possible. It was the first time I set one up, and I didn't know what I was doing, and I couldn't find good docs. |
| 15:48.41 | ManxPower | Qwell: you know that all the easy to diagnose bugs were fixed long ago, the only ones left are the really hard ones to diagnose -- like race conditions and the like |
| 15:48.43 | dlynes | Qwell: I've reported lots of bugs...most of them have been fixed |
| 15:48.45 | flush | when i call my mom i have to pass through a central so it doesnt cost me annything, i have this in extensions.conf; exten => 1,1,Dial(${OUTBOUNDTRUNK}/1234567890mmm1234567890,,mwW) |
| 15:49.05 | dlynes | Qwell: the only ones that haven't been fixed are ones I haven't reported because I haven't been able to figure out what's causing it |
| 15:49.06 | flush | the thing is, when it calls the central at first, it has music on hold, but whne central bounces me to the other phone number i hear rigning instead of music |
| 15:49.08 | ManxPower | flush: "m" is not a dial char |
| 15:49.10 | flush | whats the matter? |
| 15:49.14 | *** join/#asterisk km2 (n=x@c-24-23-255-173.hsd1.ca.comcast.net) |
| 15:49.20 | flush | its to wait i think i have it there.. |
| 15:49.24 | jblack | I'll throw my zaptel and zapata confs on pastebin. |
| 15:49.28 | flush | i have to wait for the tone before i compose the other number |
| 15:49.31 | dlynes | Qwell: and I pretty much know that if I can't replicate the problem, the developers probably won't be able to fix the problem |
| 15:49.47 | ManxPower | flush: the far end is answering the call and sending you ringing audio |
| 15:50.00 | ManxPower | flush: since "m" does not wait...... |
| 15:50.01 | flush | so theres nothing i can do? |
| 15:50.08 | flush | what do i use to wait? cause it actually works.. |
| 15:50.10 | ManxPower | Pehaps you are looking for the "w" dial char, which pauses for .5 seconds |
| 15:50.15 | flush | i want a like 1 sec wait |
| 15:50.24 | ManxPower | then use two of them |
| 15:50.24 | BBHoss | iratik: so you're using the T1 for phone calls, or is it a data t1? |
| 15:50.25 | flush | oh |
| 15:50.28 | flush | thanks a lot |
| 15:50.34 | iratik | BBHoss: data T1 |
| 15:50.37 | jblack | zaptel.conf and zaptel.conf to go with (http://linuxguru.net~jblack/calls) http://pastebin.com/m6fcdfa70 |
| 15:50.46 | ManxPower | flush: in fact, there is nothing you can do, the far end answers (to get the auth code, I assume) |
| 15:51.16 | BBHoss | iratik: so all the computers behing the cisco router have public ips? |
| 15:51.18 | flush | hrm, on the asterisk Dial options page i see this; w: Allow the called user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon) |
| 15:51.36 | flush | you say its a .5 sec delay?, why they say its about recording options there? |
| 15:51.50 | ManxPower | flush: that stuff is all last on the dial line, those are OPTIONS. "w" in the dial STRING is a WAIT |
| 15:51.51 | iratik | BBHoss: there is only one computer behind the router ... or more accurately .. one computer that uses the router as a gateway ...... our PBX has a public ip yes. |
| 15:52.06 | flush | ohh |
| 15:52.07 | jblack | Based on what you said a few minutes ago, it's probably worth mentioning that the r1t1 module (which is supposed to have echo cancellation built in) is being run without any options. I believe there's an echo cancellation option "ec=1" that I'm not using, as the module wasn't loading wtih it |
| 15:52.08 | flush | k thanks |
| 15:52.08 | iratik | We have a separate internet connection for non-VOIP traffic through a separate router |
| 15:52.21 | ManxPower | i.e. Dial(Zap/g1/5551212wwww1234,,w) The first w's are waits, the w at the end is for other stuff |
| 15:52.49 | BBHoss | iratik: so no ports are being blocked? no iptables running somewhere forgotten? |
| 15:52.53 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 15:52.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:52.56 | flush | nice |
| 15:53.00 | iratik | BBHoss: yes there is an iptables configuration |
| 15:53.07 | iratik | on the pbx.. after all. its a public ip |
| 15:53.18 | BBHoss | i find firewalls nearly useless |
| 15:53.26 | flush | ManxPower do you know where i can find those options within the dial command, since on the page i dont see anything else but commands at the end |
| 15:53.47 | BBHoss | best use is access control, so only certain ip ranges can access services |
| 15:53.47 | iratik | but if it were the firewall, wouldn't all my sip traffic be non-functioning... instead of just with certain providers ? there is no IP-address based restrictions on the chain rules |
| 15:53.59 | ManxPower | flush: I have never seen them documented in the offical docs |
| 15:54.07 | iratik | BBHoss: I like that too |
| 15:54.08 | jblack | ~zaptel |
| 15:54.09 | jbot | rumour has it, zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. a phone card company |
| 15:54.12 | dlynes | Qwell: I've even submitted code to fix deficiencies in the asterisk code, as well |
| 15:54.25 | flush | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 15:54.27 | iratik | BBHoss: disable the firewall and see if that fixes it? |
| 15:54.32 | dlynes | Qwell: but that's only been for new behaviour; not bug fixes |
| 15:54.34 | BBHoss | iratik: why dont you try disabling iptables temporarily and see if it works, if it does, then you know where your problem lies |
| 15:54.48 | Qwell | saying "1.4 is buggy" does not help the problem |
| 15:54.56 | ManxPower | flush: the Wiki is filled with incorrect, outdated, and just plain wrong information. You do not use it as a referance for Asterisk apps. |
| 15:55.09 | flush | copy |
| 15:55.10 | ManxPower | you use "core show application X" as the reference for Asterisk docs |
| 15:56.26 | jaytee | 1.4 buggy? runs real stable for me! |
| 15:56.44 | ManxPower | jaytee: It runs real stable for many people. |
| 15:56.56 | ManxPower | It runs real unstable for many other people. |
| 15:57.10 | *** join/#asterisk shido6 (n=shido6@74-130-224-188.dhcp.insightbb.com) |
| 15:57.10 | jaytee | think it might be the people then |
| 15:57.24 | ManxPower | All these n.n.1 releases makes me have very little faith in 1.4 |
| 15:57.51 | ManxPower | Well, at least in the 1.4 release process. |
| 15:58.14 | ManxPower | jaytee: do you use Queues? |
| 15:58.15 | BBHoss | iratik: precisely |
| 15:58.16 | spokra | try 1.6.beta9.. ROFLOL |
| 15:58.41 | ManxPower | jaytee: How about lots of call recording. |
| 15:58.44 | iratik | BBHoss: the firewall wasn't the problem.. i'm calling the T1 people to see if their managed router might be blocking it |
| 15:58.49 | jaytee | ManxPower, not at the moment but I will be by the middle of next year when I migrate my Nortel ACD's over |
| 15:59.00 | *** join/#asterisk resin0008 (n=resin000@7.218.204.68.cfl.res.rr.com) |
| 15:59.05 | BBHoss | iratik: small possibility |
| 15:59.28 | jaytee | ManxPower, and virtually no call recording other than the names of people entering a MeetMe |
| 15:59.33 | ManxPower | jaytee: play around with both and see how stable 1.4 is for you. Give it a try -- it may work great it may not, but don't assume just because the features you are currently using do not cause issues, and features you start using will ace the same. |
| 15:59.45 | BBHoss | iratik: did you say that it worked with one sip ITSP but not with the other? |
| 15:59.48 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 15:59.50 | resin0008 | hey guys, I still need help with hints if anyone is willing |
| 15:59.54 | iratik | BBHoss: yep |
| 16:00.15 | jaytee | ManxPower, yeah, I model everything first on a test server before I move it to the live box. |
| 16:00.19 | BBHoss | what is the full sip trace you get from the provider that's not working |
| 16:00.27 | iratik | BBHoss: how do I do that? |
| 16:00.36 | iratik | BBHoss: should i search the bible for that? |
| 16:00.47 | BBHoss | iratik: go to the cli, type "sip debug" then try to make a call |
| 16:00.55 | BBHoss | copy/paste the contents to pastebin |
| 16:00.56 | ManxPower | jaytee: Many of the issues I've seen people on this channel have only happen when the system is under load -- not found during testing. |
| 16:01.12 | iratik | BBHoss: is there a way to limit that to communications to/from a specific host? |
| 16:01.14 | ManxPower | And honestly, thats why they have not been fixed -- the only systems that have the problems are the ones in production |
| 16:01.15 | BBHoss | sip no debug turns it off |
| 16:01.32 | BBHoss | iratik: yeah, you can do sip debug peer $PEERNAME |
| 16:01.38 | iratik | BBHoss: found the doc for sip debug |
| 16:01.44 | ManxPower | That's why I think Digium's corporate PBX should run the current Asterisk release. |
| 16:01.59 | jaytee | ManxPower, do you think in most of those cases it's the hardware that can't handle the load or software bugs that only show up under load. |
| 16:02.23 | ManxPower | Digium says upgrade! upgrade! Test! Test! Report bugs! Report bugs! Like upgrading is something simple like breathing. |
| 16:03.09 | ManxPower | jaytee: I think most of the issues are software bugs that only show up under load. |
| 16:03.39 | jaytee | I think we need a solid load testing suite of tools that test queues, recordings, high call volumes etc. |
| 16:04.00 | ManxPower | When you have a hundred angry users breathing down your neck and you want to do is get them to leave you alone, not spend three days diagnosing a bug while the company has a crashing PBX. |
| 16:04.19 | iratik | BBHoss: getting pastie |
| 16:04.42 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 16:05.01 | iratik | BBHoss: http://www.pastie.org/213762 |
| 16:05.18 | ManxPower | Our PBX did not become stable until we stopped upgrading every time a new version comes out. |
| 16:05.33 | jaytee | ManxPower, I hear ya. Everyone takes their phone for granted until it doesn't work and then they expect it to be fixed yesterday. |
| 16:05.36 | ManxPower | We found a version that is stable for our usage and stick with it. |
| 16:05.50 | ManxPower | jaytee: I think that is what Digium does not understand. |
| 16:05.56 | DonAlex | Afternoon peeps... |
| 16:06.14 | BBHoss | iratik: so voipinvite is the one that isnt working? |
| 16:06.28 | ManxPower | jaytee: Somehow I think things would work a lot differently if the Digium developers managed the Digium PBX -- not the Digium IT department. |
| 16:06.56 | iratik | BBHoss: yep |
| 16:07.19 | jaytee | that's why I'm sticking with 1.4.15 for now. I've seen lots of people with horror stories about 1.4.18 and 1.4.19 in here. "Everything worked great before! Now I can't do "X"! Help!" |
| 16:07.46 | ManxPower | jaytee: ever notice these people are almost always running production systems and are in a panic? |
| 16:07.56 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 16:08.52 | jaytee | ManxPower, sounds like my boss. When one of our productions systems or the PBX is down all he knows how to do is hang over your shoulder asking stupid questions and acting nervous. As an IT Director the man is totally useless in a crisis. |
| 16:09.09 | BBHoss | iratik: does it not work at all or just half-audio? |
| 16:09.19 | DonAlex | jaytee: Well I am not ;) But I am still curious why asterisk is still using 97% of one cpu when it is idle.. ;) |
| 16:09.19 | iratik | BBHoss: half audio |
| 16:09.38 | DonAlex | anyone know why this seems to occur ? http://pastebin.com/m7e8ebdd9 |
| 16:09.40 | BBHoss | iratik: try putting nat=no in the voipinvite peer config |
| 16:09.59 | iratik | and rerun the trace? |
| 16:10.06 | jblack | i'm missing /etc/modprobe.d/zaptel. Would someone mind pastebinning theirs? |
| 16:10.17 | BBHoss | iratik: yeah, also set debug to 10 |
| 16:10.20 | jaytee | DonAlex, Asterisk is using the 97%? or something else |
| 16:10.35 | DonAlex | for the record using Asterisk SVN-trunk-r121716 on Linux 2.6.22-3-vserver-686 |
| 16:10.48 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 16:11.03 | DonAlex | jaytee: Yes. .it is using 97-99% of one CPU with that error.. |
| 16:11.20 | DonAlex | ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, 0xbf8c8d98) = -1 ENOTTY (Inappropriate ioctl for device) |
| 16:11.20 | DonAlex | write(1, "\0", 1) = 1 |
| 16:11.20 | DonAlex | write(1, "*CLI> ", 6) |
| 16:11.21 | ManxPower | jblack: go to the zaptel source directory, type "make config" |
| 16:11.27 | ManxPower | DonAlex: DO NOT FLOOD THE CHANNEL! |
| 16:11.28 | DonAlex | that is from the strace on the process. |
| 16:11.30 | ManxPower | ~pb |
| 16:11.30 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:11.50 | DonAlex | ManxPower: Tht was suppose to be just one line.. I do apologise.. |
| 16:12.00 | ManxPower | jblack: sorry, /etc/modprobe, nevermind. I was referring to /etc/rc.d/init.d/zapte.l |
| 16:12.17 | DonAlex | ManxPower: The pastbin is just that reinterated anyway. |
| 16:12.23 | jblack | No problem. |
| 16:12.49 | Strom_C | "reinterated" -- god, i don't know whether I love or hate the retarded gibberish that passes for English half the time in this channel |
| 16:13.40 | DonAlex | Strom_C: Awww come on now.. have you ever stopped to spellcheck your IRC chatter.. ? Some of us have been up 18 hours already ;) |
| 16:13.44 | jaytee | Strom_C, I take it you didn't vote for Bush since you seem to have a dislike for retarded gibberish :-) |
| 16:13.51 | DonAlex | Strom_C: y'all know what I meant.. |
| 16:13.54 | jblack | Right now I'm trying to use the hardware echo canceller on the card, to see if that abates the problem. However, I think (but am not sure) that zaptel is loading the r1t1 module with ec=1 as specified in /etc/modules. |
| 16:13.56 | BBHoss | lolz |
| 16:13.57 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 16:14.12 | jblack | I think I'm supposed to do it through /etc/modules.d/zaptel, which I don't have. |
| 16:14.23 | ManxPower | jblack: I suspect that is a Rhino question |
| 16:14.23 | Strom_C | DonAlex: if you can't even take the time to spell correctly, then perhaps now is not the time to be troubleshooting your PBX, since you're obviously too tired to be noticing details |
| 16:14.27 | Nasra | hello crowd ...question, has the TD400P being replaced for a TD410? |
| 16:14.33 | Strom_C | Nasra: yes |
| 16:14.37 | Nasra | thanks |
| 16:14.45 | iratik | BBHoss: thanks for helping me,.. sorry its taking a second .. getting it now |
| 16:14.45 | ManxPower | Nasra: perhaps you mean the TDM400P and TDM410P? |
| 16:14.55 | iratik | was verifying the ip |
| 16:14.55 | Nasra | yes |
| 16:15.03 | Qwell | no P on410 |
| 16:15.03 | BBHoss | iratik: np, taking a break from web shit |
| 16:15.05 | jblack | Yeah. It may be. It naively feels more like a zaptel question at the moment; how to load a module with options. |
| 16:15.21 | ManxPower | Nasra: In telecom one letter can mean the difference between getting a problem fixed and wasting days |
| 16:15.29 | km2 | is it normal to occasionally see this in the CLI: "-- B-channel 0/1 successfully restarted on span 1" (one for each line; in my case 23 for our PRI)? |
| 16:15.33 | Nasra | oh lol |
| 16:15.35 | ManxPower | jblack: naw, that's a distro question |
| 16:15.35 | Strom_C | km2: yes |
| 16:16.12 | km2 | Strom_C, thank you. do you know what's going on there? i assume it's fine but i'm a little curious |
| 16:16.22 | DonAlex | ManxPower: Oh now that is charming.. And who stole your pillow this morning huh? It is not like I am not giving as much intel as I can on the subject. And many eyes bugs shallow yada yada... |
| 16:16.36 | Nasra | ManxPower needed to buy a PCI Card for communications.....told has been changed/ upgraded to TDM410 |
| 16:16.49 | Strom_C | km2: asterisk is restarting the idle B channels on your PRI. It's perfectly normal. |
| 16:17.44 | DonAlex | ManxPower: Oh now that is charming.. And who stole your pillow this morning huh? It is not like I am not giving as much intel as I can on the subject. And many eyes bugs shallow yada yada... |
| 16:17.53 | nephfl | i cant figure out why my calls are dropping |
| 16:17.57 | DonAlex | Whops |
| 16:18.01 | DonAlex | lagged out.. sorry. |
| 16:18.17 | iratik | BBHoss: http://www.pastie.org/213767 |
| 16:18.39 | nephfl | im just getting "exited non-zero" |
| 16:18.48 | iratik | BBHoss: Why am I seeing that 70.248.216.14 address in there... thats an old IP for this server.. but i'm still seeing it .. why is that in the trace.. could it be causing a problem? |
| 16:19.07 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:19.14 | Strom_C | nephfl: PSTN calls? or are they internal calls? |
| 16:19.29 | BBHoss | iratik: which way is the audio, is the server not recieving audio or not sending? |
| 16:19.44 | nephfl | vtwhite sip trunk |
| 16:19.45 | Nasra | ManxPower: I seen the card for $175 ( TDM410P ) at Telephonydepot....just about to set my system up and running... |
| 16:20.07 | BBHoss | iratik: btw i'm gonna gloss over the fact you're using freepbx |
| 16:20.12 | iratik | BBHoss: going from inside the network to outside way only... can't hear ringing or pickup or anything on the inside of the network |
| 16:20.21 | iratik | BBHoss: much appreciated |
| 16:20.34 | Strom_C | nephfl: have you done a SIP debug? |
| 16:20.43 | BBHoss | but the person answering can hear you? |
| 16:20.48 | iratik | BBHoss: yes |
| 16:20.52 | BBHoss | k |
| 16:21.14 | iratik | BBHoss: does that 70.248.216.14 address have anything to do with the problem... what if the trunk provider is sending packets back to the wrong address? |
| 16:21.21 | BBHoss | iratik: what files do you have in /etc/asterisk? do you see a sip_nat.cfg? |
| 16:21.45 | iratik | BBHoss: !! |
| 16:21.49 | BBHoss | im pretty sure i know |
| 16:21.52 | BBHoss | old extern_ip |
| 16:22.09 | Qwell | if he does have sip_nat.cfg, he should leave |
| 16:22.31 | iratik | Qwell: i don't have sip_nat.cfg then |
| 16:22.39 | BBHoss | i don't think we're interrupting anything here |
| 16:23.10 | BBHoss | iratik: is that what it was? |
| 16:23.28 | iratik | BBHoss: i should have done cd /etc/asterisk && grep -R 70.248.216 . |
| 16:23.34 | BBHoss | heh |
| 16:23.35 | iratik | and i would have found out where it was |
| 16:23.56 | nephfl | sip debug doesnt show anything |
| 16:24.07 | BBHoss | yeah that first packet in the debug is what you are SENDING to the ITSP |
| 16:24.24 | BBHoss | dunno how the other was working, maybe a register string or something |
| 16:24.50 | Strom_C | nephfl: does this only happen with that one provider? have you tried other ITSPs to see if the problem remains? |
| 16:27.07 | iratik | BBHoss: let me see if that fixes everything ... thanks btw |
| 16:27.17 | BBHoss | iratik: good luck |
| 16:28.58 | James|TCC | how do i strip digits from an extension number, eg for outbound calls i need to strip the 9 from the start |
| 16:29.12 | BBHoss | EXTEN:1 |
| 16:29.28 | James|TCC | so exten => _9X.,1,Dial(Zap/G1/${EXTEN:1}) |
| 16:29.36 | BBHoss | should do it |
| 16:29.40 | James|TCC | cool thanks |
| 16:29.52 | James|TCC | dunno if my outbound lines are working, but might find out now :) |
| 16:30.11 | BBHoss | heh |
| 16:30.15 | BBHoss | always fun |
| 16:30.53 | James|TCC | hmm |
| 16:30.55 | James|TCC | <PROTECTED> |
| 16:30.59 | James|TCC | what have i missed lol |
| 16:31.10 | James|TCC | ive installed suse, |
| 16:31.44 | James|TCC | installed zaptel, then asterisk and asterisk-addons |
| 16:32.01 | James|TCC | configured zaptel.conf, zapata.conf, |
| 16:32.09 | lmadsen | after compiling and installing zaptel, you ran ./configure then reinstalled asterisk right? |
| 16:32.12 | James|TCC | put my phones (which work) in sip.conf |
| 16:32.21 | James|TCC | i installed zaptel first lmadsen |
| 16:32.28 | lmadsen | asterisk will only build the chan_zap + other modules if zaptel is actually instaleld |
| 16:32.31 | lmadsen | installed* |
| 16:32.37 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.21 (2008/06/12) Asterisk 1.2.29 (2008/06/03), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.4 (2008/...) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 16:32.46 | James|TCC | i followed the digium guide, which says do it first :) |
| 16:32.50 | lmadsen | w00t 1.4.21 |
| 16:32.57 | James|TCC | asterisk followed |
| 16:32.58 | lmadsen | goes to lunch |
| 16:33.48 | James|TCC | should i have something like "zap show stuff" in the cli if ive done it right? |
| 16:33.53 | James|TCC | cause there isnt atm :( |
| 16:35.52 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 16:36.23 | Strom_C | James|TCC: go back to the asterisk source directory, rerun ./configure, and then run make menuselect and look at whether it found zaptel |
| 16:36.40 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 16:36.54 | nephfl | I use this provider with other boxes hosted in a different location |
| 16:37.07 | Strom_C | nephfl: that's not what I'm asking |
| 16:37.19 | Strom_C | nephfl: does THIS box work with OTHER providers? |
| 16:37.32 | nephfl | i dont have another provider to try |
| 16:37.43 | Strom_C | well, pick one and throw five dollars at it |
| 16:37.59 | nephfl | true.. i have a teliax account with one customer |
| 16:38.01 | jaytee | got 2 Digium TDM04B cards I can sell, each with 4 FXO modules only used for 4 months before switching to PRI cards. Anyone interested? |
| 16:38.42 | James|TCC | <PROTECTED> |
| 16:38.59 | James|TCC | Depends on: res_smdi(M), zaptel_vldtmf(E), zaptel(E), tonezone( |
| 16:39.03 | spokra | hehehe anyone need a single span digiam T1 card? |
| 16:39.11 | Strom_C | James|TCC: ok...recompile asterisk |
| 16:39.13 | James|TCC | is the (E) error, or something else? |
| 16:39.27 | Strom_C | James|TCC: actually |
| 16:39.28 | James|TCC | (M) i assume is module? |
| 16:39.29 | Strom_C | wait |
| 16:39.52 | Strom_C | is chan_zap loading properly when asterisk starts, or is it spitting out an error? |
| 16:39.58 | James|TCC | lemme check |
| 16:39.59 | James|TCC | hang on |
| 16:42.13 | James|TCC | ahha |
| 16:42.14 | James|TCC | [Jun 12 17:40:17] WARNING[20457]: chan_zap.c:897 zt_open: Unable to open '/dev/zap/channel': No such file or directory |
| 16:42.40 | BBHoss | James|TCC: sounds like the driver isnt loaded |
| 16:42.54 | Strom_C | you did configure and load zaptel before starting asterisk, right? |
| 16:43.11 | James|TCC | is there an installation wlakthrough anywhere? |
| 16:43.32 | BBHoss | James|TCC: is it a Digium card? |
| 16:43.41 | James|TCC | i followed the instructions on http://www.digium.com/en/docs/TDM800P/800series_quickstart.pdf |
| 16:43.46 | James|TCC | yeah tdm800p |
| 16:44.06 | James|TCC | 1-4 are unused fxs, 5-8 are fxo (with lines to 5 and 6 atm) |
| 16:44.07 | BBHoss | call digium then, they will walk you through installation for free |
| 16:44.15 | James|TCC | ahha sounds like a plan :P |
| 16:44.31 | James|TCC | from a blank install? |
| 16:44.33 | BBHoss | sounds like the drivers aren't being 'modprobed' |
| 16:44.45 | BBHoss | installation support is free, never done it |
| 16:44.49 | James|TCC | ie, if i remove all traces of asterisk |
| 16:44.53 | James|TCC | ok, i'll let you know :P |
| 16:44.54 | *** join/#asterisk tobias (n=tobias@cpe-069-134-205-184.nc.res.rr.com) |
| 16:45.04 | BBHoss | i doubt they'll make you start over |
| 16:45.10 | Strom_C | James|TCC: just call it with the system the way it is |
| 16:45.16 | Strom_C | they'll help you out |
| 16:45.25 | Strom_C | they've done this once or twice before, I think ;) |
| 16:45.32 | BBHoss | the ubuntu 8.04 buil of asterisk isnt bad, apt-get install zaptel asterisk asterisk-addons :) |
| 16:46.17 | loompek | should the context in voicemail.conf be the same as the users context? |
| 16:46.25 | loompek | ... in sip.conf |
| 16:46.27 | BBHoss | no |
| 16:46.31 | BBHoss | separate |
| 16:46.51 | Strom_C | loompek: voicemail context is a completely independent thing from dialplan context |
| 16:46.57 | loompek | yeah.. that's what i thought... |
| 16:47.04 | loompek | but the strangest thing is... |
| 16:49.25 | *** join/#asterisk deeperror (n=deeperro@adsl-76-226-146-19.dsl.sfldmi.sbcglobal.net) |
| 16:52.06 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 16:52.46 | loompek | [Jun 12 18:37:03] NOTICE[29103]: chan_sip.c:15092 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 5 |
| 16:52.58 | loompek | even though 5 is a plain ol valid mailbox |
| 16:53.30 | loompek | nicely defined in [default] in voicemail.conf |
| 16:53.53 | Strom_C | pastebin voicemail.conf and sip.conf |
| 16:54.30 | *** join/#asterisk exvito (n=exvito@mail.colours.pt) |
| 16:55.42 | *** part/#asterisk exvito (n=exvito@mail.colours.pt) |
| 16:55.42 | loompek | http://pastebin.com/m971c37a |
| 16:56.07 | Strom_C | where's the mailbox= line in sip.conf? |
| 16:57.40 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 16:57.50 | jeremy_g | anyone home |
| 16:58.00 | Strom_C | ALL DEAD HERE |
| 16:59.03 | *** join/#asterisk flynux_ (n=flynux@2a01:38:0:0:0:0:0:1) |
| 17:00.41 | loompek | oh... thaaaaat mailbox= line :D |
| 17:00.47 | x86 | lol... voipsupply.com is getting a DoS attack |
| 17:01.00 | x86 | now it's back up |
| 17:01.27 | BBHoss | anyone here have a snom m3? |
| 17:03.03 | BBHoss | after a few hours, it just stops transmitting DTMF, period. I've tried inband, rfc2833, and even SIP INFO, all fail after an apparently random amount of time (cannot reproduce on demand) |
| 17:03.20 | rootlogin | when i dial "0" in this dialplan http://pastebin.com/d25e285b5 there is a delay of about 5s until the extension gets executed ... other extensions work fine .. any ideas ? |
| 17:03.44 | JT | rootlogin: how is the phone connected? |
| 17:04.00 | rootlogin | a SIP-phone on the network |
| 17:05.01 | rootlogin | a hardphone SPA901 .. i can call from the same phone .. and the time changes |
| 17:05.38 | rootlogin | there is also no log output so asterisk is waiting for something ? |
| 17:06.00 | raytruz` | root, its waiting for you to dial a longer number |
| 17:06.55 | Strom_C | rootlogin: modify the dialplan on the /phone/ |
| 17:07.01 | JT | rootlogin: it's a phone dialplan issue |
| 17:07.03 | *** join/#asterisk ikevin (n=kevin@www.icedslash.org) |
| 17:07.15 | ikevin | back |
| 17:07.24 | rootlogin | hmm .. so to avoid that i would have to change the other extensions ? like _[123456789]X ? |
| 17:07.32 | Strom_C | no no no no no and no |
| 17:07.35 | ikevin | does anyone know a good howto about extensions under mysql? |
| 17:07.35 | Strom_C | rootlogin: modify the dialplan on the /phone/ |
| 17:07.37 | rootlogin | hehe :) |
| 17:07.58 | rootlogin | the phones config ? .. i had one issue there before .. hmm |
| 17:08.32 | rootlogin | the phone .. k ill check that .. maybe it does something diff on a single 0 |
| 17:12.39 | *** part/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 17:13.14 | *** join/#asterisk exvito (n=exvito@89-180-10-119.net.novis.pt) |
| 17:13.59 | James|TCC | just removed/recompiled the lot, and its now working |
| 17:14.23 | James|TCC | the fact i had fxoks and fxsks the wrong way round probably also didnt help much :P |
| 17:14.27 | Strom_C | heheh |
| 17:14.51 | James|TCC | now to get the other lines wired in :) |
| 17:15.09 | ManxPower | IP Phones collect ALL the digits of the number BEFORE even connecting to Asterisk |
| 17:15.26 | exvito | hi all... i'm not sure 100%, but i think i lost my "regextens" after "sip prune realtime user/peer all" -- those same "regextens" wouldn't show again on the dialplan context after the sip phones re-registered (only after phone reboot) -- any experiences ? |
| 17:16.07 | exvito | (my fix: reload chan_sip) |
| 17:18.11 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 17:18.24 | exothermc | So who fired off that 1.4.21 release email? |
| 17:19.25 | exothermc | cause they screwed up the change log link , and the package isn't "available for immediate download from the Digium downloads site." |
| 17:19.32 | Strom_C | it was all his fault |
| 17:19.34 | Strom_C | points |
| 17:20.01 | ManxPower | exothermc: It NEVER is. The MARKETING department updates the web site and you know marketing people are a little slow |
| 17:20.32 | exvito | ...I guess some other times, they are too quick... ;) |
| 17:20.37 | ManxPower | It's been like that ever since they removed the Digium FTP server. |
| 17:21.38 | ManxPower | exothermc: It's never been an issue for me, since I wait to see what issues are in a new release before trying it, and I stick to 1.2 |
| 17:22.07 | anonymouz666 | ManxPower: do you still run the 1.2? |
| 17:22.20 | ManxPower | anonymouz666: Yes on all of my servers. |
| 17:22.24 | *** part/#asterisk exvito (n=exvito@89-180-10-119.net.novis.pt) |
| 17:22.27 | errr | same here |
| 17:22.29 | ManxPower | MAND people do |
| 17:22.45 | ManxPower | and MANY people do |
| 17:23.10 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 17:23.14 | anonymouz666 | ManxPower: are you happy with many deadlocks in 1.2 version? many of them was fixed by 1.4 release. |
| 17:23.26 | ManxPower | anonymouz666: we never have deadlocks |
| 17:23.54 | ManxPower | 2 or 3 times a year we have an issue that might be a deadlock. |
| 17:24.18 | ManxPower | anonymouz666: For one thing my customer won't fund an upgrade of 5 servers just because there is a new release. |
| 17:24.39 | ManxPower | They already whine about the maint costs of Asterisk being much higher than their Nortel system |
| 17:24.43 | anonymouz666 | SIP REFER in chan_sip.c (1.2) causes deadlocks, there are also many situations involving the chan_local that causes deadlock... |
| 17:25.02 | ManxPower | anonymouz666: and yet there are still people that can't use 1.4 because it doesn't work for them |
| 17:25.26 | ManxPower | And the maint costs of Asterisk ARE much higher than any commercial system -- even if you just count the cost of upgrades. |
| 17:26.02 | coppice | and the nortel PBX will run for 25 years |
| 17:26.43 | ManxPower | Testing a new release, finding and fixing bugs, making our existing configs work with the upgrade, then downgrading because of something breaking, in all US$5,000 at least |
| 17:27.09 | anonymouz666 | coppice: it's more than my life! |
| 17:27.19 | ManxPower | and my customer refuses to pat $5,000 every month or two for their phone systems |
| 17:27.53 | coppice | people are still using SL1 line cards from the 1970s in meridian switches. they are fully compatible, and still reliable |
| 17:28.00 | ManxPower | coppice: exactly. This upgrade every month or two stuff is crap |
| 17:28.34 | coppice | well, why are you doing it? do they want new stuff, or it just upgrade for the sake of it? |
| 17:28.44 | *** join/#asterisk nicox_ (n=nicox@213-33-14-110.adsl.highway.telekom.at) |
| 17:29.14 | anonymouz666 | ManxPower: If you run 1.2 just fine so there's no reason to upgrade. I consider that a miracle. I just upgrade every box because there's no way to live with deadlocks in basic services. |
| 17:29.16 | ManxPower | coppice: I'm not doing it. I'm whining about everyone telling me that is what I should be doing. |
| 17:29.27 | errr | I dont even update my system at my house as often as releases come out |
| 17:29.36 | ManxPower | We are on 1.2.24 across all systems in the company (currently 5 or 6) |
| 17:30.10 | errr | we have all our servers at work on 1.2.22 |
| 17:30.24 | ManxPower | With the possible exception of the single Asterisk server that is accessable from the internet. |
| 17:30.39 | coppice | well, to some extent you have to upgrade when you touch anything IPish. you need to apply the security updates at least. I doubt any 1970s SL1s still around have ever had a security update |
| 17:31.11 | ManxPower | coppice: We don't worry too much about security updates for the systems with no internet connectivity -- which is most of our PBXs |
| 17:31.51 | *** join/#asterisk cyberdeath (n=cyberdea@67.131.149.209) |
| 17:31.53 | anonymouz666 | If I remember well there's a fun stuff with SIP register before 1.2.26 release. |
| 17:32.23 | ManxPower | anonymouz666: by "fun stuff" do you mean security issues? |
| 17:32.31 | anonymouz666 | yes |
| 17:32.49 | anonymouz666 | you crash the PBX and drop all calls just sending a SIP Register. |
| 17:32.56 | ManxPower | And exactly how will that be exploited if the system cannot connect to the internet? |
| 17:33.12 | anonymouz666 | well, any user can do that. |
| 17:33.20 | ManxPower | You've not met my users. |
| 17:33.31 | errr | if the user is smart enough to do that they generally wrk in IT |
| 17:34.13 | ManxPower | My users are technophobic realestate agents that would rather go to the country club or to play golf than read a single page of instructions. |
| 17:34.40 | errr | same here only I deal with insurance agents |
| 17:34.40 | anonymouz666 | heh |
| 17:34.42 | hsv-al | manxpower |
| 17:34.47 | ManxPower | And if a user knows enough to craft a SIP register packet to crash the system then I want them working for ME |
| 17:35.02 | hsv-al | what I want more then ANYTHING is a native software sip client, or software iax client for Iphone, or Blackberries |
| 17:35.09 | hsv-al | that can utilize all the audio hardware, once that happens |
| 17:35.19 | hsv-al | i ditch all phone plans, and just use bb's or iphone for data service :) |
| 17:35.39 | ManxPower | One (typical) user told me that she doesn't use txt messages on her cell phone because "it's too complicated" |
| 17:35.49 | ManxPower | hsv-al: best of luck with that. |
| 17:35.55 | hsv-al | none exist |
| 17:36.06 | hsv-al | it would be great, why dont we have any yet? |
| 17:36.08 | hsv-al | its ridiculous |
| 17:36.12 | ManxPower | When was the last time you looked at the latency and jitter of an iPhone or BB data connection? |
| 17:36.28 | hsv-al | well, google: IM+ for skype on Blackberry 8703e |
| 17:36.35 | hsv-al | it works, a 3rd party skype client, but no sip yet |
| 17:36.39 | hsv-al | quality was pretty good too |
| 17:36.43 | ManxPower | Just ping me and you can see the latency and jitter of a "3G" cell data connection |
| 17:36.44 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 17:36.47 | Strom | INTERACTIVE CLICKABLE SHOCKWAVE CYBERTOONS!?!?!?!?!? |
| 17:36.52 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:36.58 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 17:37.09 | hsv-al | manxpower, one of the RIM devs posted on blackberryforums.com, the reason there isnt a sip client yet |
| 17:37.16 | hsv-al | is because J2ME has some issues with sip stack |
| 17:37.21 | hsv-al | stuff that is out of my realm of knowledge |
| 17:37.30 | hsv-al | <PROTECTED> |
| 17:37.35 | hsv-al | not latency issues, who knows |
| 17:37.38 | ManxPower | But that was not my question |
| 17:38.48 | ikevin | does anyone know a good howto about extensions under mysql? |
| 17:41.44 | *** join/#asterisk MrNaz (n=naz@ppp121-44-231-163.lns2.mel4.internode.on.net) |
| 17:47.42 | *** join/#asterisk mags2 (n=egray@ampulex.whoi.edu) |
| 17:47.57 | *** join/#asterisk BBHoss (n=hoss@c-68-62-175-86.hsd1.al.comcast.net) |
| 17:48.14 | mags2 | is there any documentation for switch => besides random stuff on voip-info? |
| 17:49.29 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 17:49.49 | *** join/#asterisk xnosx (n=xnosx@212.145.172.127) |
| 17:50.03 | BBHoss | PSA: Avoid Snom M3 like the plague |
| 17:50.47 | _ShrikE | BBHoss: why? other than busted blind transfer, it works ok for me. |
| 17:52.23 | BBHoss | _ShrikE: my DTMF mysteriously stops working after about a day or two, and every week the thing will start saying error 903 which mean that the number of concurrent calls is maxed, when there is nobody on the phone. Not to mention the total lack of docs and support for them. |
| 17:52.38 | _ShrikE | wow |
| 17:52.44 | BBHoss | yeah |
| 17:52.50 | BBHoss | are you a home user? |
| 17:52.52 | _ShrikE | I dont heavily use mine, but have never seen anything like that |
| 17:53.06 | _ShrikE | business user |
| 17:53.25 | BBHoss | yeah these phones are being used in a business, along with polycom 500s |
| 17:53.35 | BBHoss | the pcoms work fine of course |
| 17:53.50 | _ShrikE | I just upgraded the firmware hoping it would fix the blind xfer issue, but no joy. |
| 17:54.08 | _ShrikE | I have never had much love for snom though so im not that suprised. |
| 17:54.08 | BBHoss | yeah that sucks too, but its manageable |
| 17:54.22 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 17:54.56 | BBHoss | Also absolutely no mention of how to do GAP |
| 18:02.25 | *** join/#asterisk nvrpunk (n=zomgobli@c-71-228-142-33.hsd1.ga.comcast.net) |
| 18:02.45 | nvrpunk | will the 1.4.7 addons work with 1.4.19.1 ? |
| 18:03.08 | nvrpunk | just confused about the addons naming convention compared to the main |
| 18:03.33 | russellb | just use the latest version with the latest version :) |
| 18:04.19 | nvrpunk | ok |
| 18:04.27 | nvrpunk | addons has the cdr correct? |
| 18:04.33 | nvrpunk | looking at outdated howtos :) |
| 18:04.41 | Qwell | "the cdr"? |
| 18:04.45 | nvrpunk | cdr module |
| 18:04.47 | nvrpunk | yes |
| 18:04.54 | Qwell | it has several cdr modules, but not all of them, no |
| 18:05.01 | nvrpunk | mysql |
| 18:05.05 | nick125 | Asterisk has a CDR module or two with it, and I think addons has the mysql CDR module |
| 18:05.06 | nvrpunk | sorry for being vague |
| 18:05.30 | nvrpunk | ok |
| 18:05.45 | nvrpunk | my apologies, i knew what i was talking about in my head :D |
| 18:07.04 | *** join/#asterisk RoyK (n=roy@ip-88-4-149-91.dialup.ice.no) |
| 18:15.48 | *** join/#asterisk Ward1983 (n=ward@91.178.169.228) |
| 18:22.28 | nvrpunk | does realtime support need a clock source? |
| 18:23.01 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 18:23.16 | nvrpunk | i havent read up on it yet so that may be a blatantly dumb question |
| 18:23.39 | Strom | i don't see why it would |
| 18:23.43 | *** join/#asterisk talntid (n=erict@66.208.251.170) |
| 18:23.59 | nvrpunk | well the timing on the trunks do |
| 18:24.02 | nvrpunk | hence my asking |
| 18:24.02 | nvrpunk | :) |
| 18:24.09 | nvrpunk | granted neither are related |
| 18:24.10 | nvrpunk | heh |
| 18:26.44 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:28.43 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:32.11 | *** join/#asterisk mmcgrath (n=mmcgrath@67-207-142-96.slicehost.net) |
| 18:32.16 | *** part/#asterisk mmcgrath (n=mmcgrath@67-207-142-96.slicehost.net) |
| 18:36.17 | MrNaz | is wctdm included in the zaptel package, or do i need to install it separately? |
| 18:36.24 | Strom | included in zaptel |
| 18:36.32 | MrNaz | great |
| 18:36.34 | MrNaz | thanks |
| 18:36.49 | MrNaz | for an asterisk system, do i need a working sound card in the server? |
| 18:37.02 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:37.28 | nick125 | MrNaz: You shouldn't. |
| 18:37.34 | *** join/#asterisk gego (n=gego@host-091-097-124-225.ewe-ip-backbone.de) |
| 18:38.04 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:38.12 | JT | none of my asterisk boxes have working soundcards |
| 18:38.39 | MrNaz | great |
| 18:38.44 | MrNaz | didnt think so, but thought i'd check |
| 18:38.49 | Corydon76-dig | All of mine do, but that's because the sound card comes on the motherboard |
| 18:39.53 | outtolunc | only my dev boxes do |
| 18:42.43 | *** join/#asterisk lzhang (n=lzhang@rrcs-67-78-33-170.sw.biz.rr.com) |
| 18:43.36 | lzhang | hi, I'm getting one way jitter on a SIP phone to SIP phone call... is there some way I can check if the jitter buffer is on in 1.4? |
| 18:46.20 | mags2 | for some reason switch is passing on 's' as the extension instead of the actual extension, anyone? |
| 18:46.25 | MrNaz | g729 is proprietary right? and needs to be transcoded when talking to open protocols right? |
| 18:48.50 | Strom | MrNaz: it's patent-encumbered, not proprietary |
| 18:49.01 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 18:52.24 | MrNaz | aah |
| 18:52.42 | *** join/#asterisk mike-ekim (n=mike@adsl-072-151-207-108.sip.mia.bellsouth.net) |
| 18:52.52 | mike-ekim | when I do sip set debug peer peername, it tells me |
| 18:52.55 | *** join/#asterisk Netlynx (n=Jan@lugwv/member/Netlynx) |
| 18:52.59 | mike-ekim | Unable to get IP address of peer peername |
| 18:53.11 | mike-ekim | why is that? I am having hard time getting this operator context to register |
| 18:53.15 | mike-ekim | it was working perfectly fine before |
| 18:53.31 | *** part/#asterisk gego (n=gego@host-091-097-124-225.ewe-ip-backbone.de) |
| 18:54.06 | cpm | proprietary adj : protected by trademark or patent or copyright; made or produced or distributed by one having exclusive rights; |
| 18:54.18 | cpm | does this *not* apply to g729? |
| 18:54.30 | *** join/#asterisk CVirus (n=GoD@62.135.96.108) |
| 18:54.36 | cpm | guess the rights are not exclusive, , , or, are they? |
| 18:54.40 | cpm | gets really confused |
| 18:58.21 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:59.22 | jaytee | does Polycom make a single line phone or is their lowest end model 2 line? |
| 19:00.00 | Strom | it's two-line, but you don't have to provision the second appearance to be a separate extension number |
| 19:00.14 | Strom | hell, you don't have to provision it at all if you don't want to :) |
| 19:00.14 | jaytee | thanks, Strom |
| 19:00.27 | jaytee | what happened to your _c |
| 19:00.36 | nick125 | Strom: Yeah, if you need a nice flashy paperweight, no need to provision it at all. |
| 19:01.00 | Strom | nick125: don't be a smartass -- i was talking about the second line appearance |
| 19:01.28 | Strom | jaytee: "Strom" hadn't identified for 37 weeks, so I requested the nick from the network |
| 19:02.33 | nick125 | Strom: Aw :) |
| 19:07.19 | jaytee | Strom, cool! I did the same thing a little bit ago cuz I've had this nick for years on Blitzed but had to add my age for Freenode. The guy never showed up for 120 days so I snagged it. |
| 19:07.47 | jaytee | Strom, do you have anything to do with the Strom Carlson website? |
| 19:08.02 | Strom | I am Strom Carlson |
| 19:08.41 | jaytee | so that's your page then |
| 19:08.42 | Qwell | Strom: confusing without the _X |
| 19:08.44 | DavidR2008 | Does anyone know if the meetme app changed from 1.4.0 to 1.4.20? |
| 19:08.51 | Qwell | DavidR2008: of course it has |
| 19:09.17 | Strom | Qwell: interactive clickable shockwave cybertoons |
| 19:09.20 | jaytee | I just showed my boss MeetMe and Page. He's all excited now. :-) |
| 19:10.28 | russellb | DavidR2008: i can tell you exactly how many times it has changed ... one sec :) |
| 19:10.41 | *** join/#asterisk grandpapadot (n=anonymou@mail.heavylogic.com) |
| 19:10.46 | Strom | jaytee: yes, that's my page ;0 |
| 19:10.47 | Strom | er :) |
| 19:10.58 | DavidR2008 | I used to create dynamic conferences on the fly using just using meetme(*number*) that doesn't seem to work in 1.4.20 and I went and read the docs and it seems to say I should be able to do this: meetme(*number*,d) but that doesn't work either |
| 19:11.05 | russellb | $ ./changes_since asterisk 1.4.0 apps/app_meetme.c ... Changes since asterisk Version 1.4.0/apps/app_meetme.c - svn revision 48926 ... 64 |
| 19:11.07 | russellb | 64 times :) |
| 19:11.14 | Qwell | russellb: that's it? |
| 19:11.18 | russellb | nods |
| 19:11.21 | grandpapadot | Hi all. Is there a way to do an extensions.conf dialplan entry for just toll-free? i.e., _8XXNXXXXXX but to get 877,888,866,800 in one line? |
| 19:11.28 | Qwell | Strom: gladstone..nice |
| 19:11.45 | russellb | Qwell: 2372 changes overall to 1.4 since 1.4.0 |
| 19:11.57 | DavidR2008 | grandpapadot: _8XXNXXXXXX doesn't work? |
| 19:12.02 | Strom | grandpapadot: no |
| 19:12.07 | russellb | DavidR2008: it matches more than you want |
| 19:12.09 | grandpapadot | Yea, but also for 843, 801, etc |
| 19:12.14 | Strom | DavidR2008: that'll also catch things like 808, 801 |
| 19:12.20 | russellb | the best thing to do is make them separate lines ... |
| 19:12.24 | russellb | grandpapadot: 843 <3 |
| 19:12.29 | grandpapadot | Can you do something like _8[00|66|77|88]NXXXXXX ??? |
| 19:12.32 | Strom | and unless hawaii and salt lake city are toll free now... |
| 19:12.36 | seanbright | grandpapadot: nope |
| 19:12.46 | grandpapadot | Thanks. |
| 19:12.57 | Qwell | can't forget 855 |
| 19:12.59 | jaytee | grandpapadot, you'd have to "stack" the pattern masks in a context |
| 19:13.16 | Strom | Qwell: 855 isn't active yet, AfAIK |
| 19:13.19 | Qwell | soon |
| 19:13.22 | Strom | fucking shift key |
| 19:13.35 | seanbright | i thought all of the 8[double digits] were reserved |
| 19:13.36 | Strom | it's been "soon" for eight years or something now |
| 19:13.40 | Qwell | seanbright: they are |
| 19:14.02 | Qwell | well, no |
| 19:14.03 | Qwell | not 811 |
| 19:14.07 | russellb | not 843 |
| 19:14.08 | russellb | not 803 |
| 19:14.09 | Qwell | that will never be a tollfree |
| 19:14.10 | russellb | not a bunch of them |
| 19:14.15 | grandpapadot | Thanks, all. |
| 19:14.18 | Qwell | russellb: 8XX where X = X. newb |
| 19:14.23 | seanbright | 8[2 repeating digits] |
| 19:14.24 | DavidR2008 | any suggestion on how to do the dynamic meetme without having to have rooms in meetme.conf? |
| 19:14.35 | *** kick/#asterisk [Qwell!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (stfu nub) |
| 19:14.43 | Strom | DavidR2008: um...use the dynamic option? :) |
| 19:14.45 | nick125 | DavidR2008: There's a flag for dynamic meetme |
| 19:14.48 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 19:14.49 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:14.55 | Qwell | if I could spell... |
| 19:15.20 | Qwell | * Channel #asteirsk created on Thu Jun 12 14:14:40 2008 |
| 19:15.21 | Qwell | >.< |
| 19:15.33 | DavidR2008 | I think that's what I tried i.e. Meetme(123,d) |
| 19:15.35 | DavidR2008 | if so it doesn't work |
| 19:15.59 | Strom | ASTEIRSK |
| 19:16.12 | Strom | Qwell: so are you going to the asteirsk convention in Phoneix this year? |
| 19:16.21 | jaytee | lol |
| 19:16.26 | Qwell | asteiricon |
| 19:16.55 | Qwell | Strom: WELL, I'm speaking. So...hopefully. |
| 19:16.55 | nick125 | DavidR2008: You don't need to specify a room number..MeetMe(|d) |
| 19:17.17 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 19:17.56 | *** part/#asterisk angom (n=angom@201.170.65.143) |
| 19:18.32 | DavidR2008 | I had been using it to create rooms dynamically that I could send different people to. this is what I was actually doing: |
| 19:18.34 | DavidR2008 | exten => _X.,n,Authenticate(/etc/asterisk/meetme.pw|a) |
| 19:18.36 | DavidR2008 | exten => _X.,n,MeetMe(${CDR(accountcode)}) |
| 19:18.38 | DavidR2008 | and using the passwords to create rooms |
| 19:18.59 | DavidR2008 | so I'm trying to keep that functionality |
| 19:20.16 | DavidR2008 | Reading the docs, I think I can do what I want simply by creating the rooms, it didn't work before because you could enter any number and it would create it dynamically, but I think that was a bug (reading voip-info) |
| 19:23.15 | DavidR2008 | that did it, thx! |
| 19:27.42 | *** join/#asterisk Dr-Linux|home (n=Nothing@221.132.117.17) |
| 19:28.49 | mags2 | hm apparently switch only works correctly if you don't use it via an included macro |
| 19:28.49 | Dr-Linux|home | I'm recording calls and i wanna save recorded audio file with the name of Agent ID .. so what variable shold i use? |
| 19:31.59 | ManxPower | Dr-Linux|home: All channel variables are listed in channelvariables.txt |
| 19:32.00 | ManxPower | Read it |
| 19:33.06 | ManxPower | Gawd, I wish I could find different job. |
| 19:33.52 | Dr-Linux|home | asterisk billing is a big heck |
| 19:38.57 | *** join/#asterisk ajricoveri (n=ajricove@190.37.169.212) |
| 19:40.53 | x86 | ManxPower: you still working at that mancamp? |
| 19:41.10 | raytruz` | to get the calling party phone, its ${CALLERID} right? |
| 19:41.49 | ManxPower | x86: I *NEVER* worked at "that man camp" |
| 19:42.20 | ManxPower | My*job* is primary consultant for a $600mil/year real estate company that is getting ready to fire all of IT. |
| 19:43.07 | Dr-Linux|home | :O |
| 19:43.21 | ManxPower | (and all their consultants) In the early fall. Until then the other departments are running amok, buying stuff without consulting IT at all and then telling us to "make it work" |
| 19:44.03 | ManxPower | In the past month they have bought an electronic sign and a card access system, both of which need internet access, neither of which can be configured to use a proxy. |
| 19:44.16 | x86 | ManxPower: oh man that's horrible |
| 19:44.29 | ManxPower | And IT was not consulted during the decision making process. |
| 19:44.31 | x86 | ManxPower: oh you just volunteer at the mancamp? for free rent or whatever? |
| 19:44.39 | ManxPower | x86: Now you see why I want a different job. |
| 19:44.44 | x86 | yeah no kidding |
| 19:44.51 | x86 | you _need_ a different job, not just want ;) |
| 19:44.54 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:44.55 | ManxPower | x86: I ran a small ISP/CableCo/Telco, all on my own, all as a hobby. |
| 19:45.09 | x86 | ManxPower: ran as in past tense? |
| 19:45.18 | ManxPower | Correct. |
| 19:45.25 | ManxPower | There were ......logistical.....issue. |
| 19:46.24 | ManxPower | Where "logistical" means "one of the owners (the crazy one) created a situation where it was impossible to continue with the project" |
| 19:46.54 | x86 | ah |
| 19:47.15 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-8-rrdg-esr-2.dynamic.isadsl.co.za) |
| 19:47.24 | x86 | so no more inet/cable/phone for the mancamp userbase? |
| 19:48.22 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 19:49.34 | MrNaz | in ubuntu hardy, do you still need to get the xaptel driver source, build the driver and install it manually? |
| 19:49.43 | MrNaz | zaptel* |
| 19:50.20 | MrNaz | that seems like a pretty long winded procedure |
| 19:51.57 | ManxPower | x86: if you want internet access at the camp you have to bring a cell internet card |
| 19:51.59 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
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| 19:52.21 | *** join/#asterisk MoreAllLess (n=jackjust@cpe-76-169-252-172.socal.res.rr.com) |
| 19:53.00 | MrNaz | ok |
| 19:53.39 | MrNaz | i have asterisk running, and have gotten the asterisk cli using asterisk -rvvv (now sure what i can do here yet, but anyway)... how do i verify that the zaptel package is working, and the card drivers are functional ? |
| 19:55.00 | jblack | You take up relgion and pray to $DIETY |
| 19:55.13 | Qwell | Invalid use of NULL. |
| 19:55.22 | Dr-Linux|home | Qwell: ) |
| 19:55.34 | jblack | yeah. Nice one qwell |
| 19:55.35 | Dr-Linux|home | Qwell: any good news about Cisco 7935? :) |
| 20:02.24 | MrNaz | jblack an amusing, but unhelpful answer |
| 20:03.22 | jblack | mrnaz: I know. I have a pri here, and other than the original ztcfg, how well it works has been out of my hands. |
| 20:03.39 | ManxPower | MrNaz: you verify it's working by making calls. You can run ztcfg -vvv and make sure it has not errors |
| 20:04.14 | MrNaz | ManxPower i do get an error... it says "Unable to open master device /deve/zap/ctl |
| 20:04.23 | ManxPower | then it's not loaded |
| 20:04.30 | MrNaz | i assume that means the driver hasnt been installed |
| 20:04.47 | MrNaz | ManxPower all i've done so far (ubuntu 8.04) is installed zaptel |
| 20:04.51 | ManxPower | no, it means the driver is not LOADED INTO MEMORY, it means nothing about it being installed or not |
| 20:05.02 | MrNaz | does the zaptel package include the drives, or do i need to build the kernel driver myself? |
| 20:05.46 | ManxPower | MrNaz: in the zaptel source do a "make config" then "service zaptel start" then "chkconfig zaptel on". Of course if you are running a Debin based distro all the commands except "make config" will be different. |
| 20:06.04 | MrNaz | i saw a 2 year old ubuntu how-to which told me to get the zaptel-source package, bulid a kernel module and install it myself |
| 20:06.17 | ManxPower | MrNaz: this is STANDARD LINUX stuff and has nothing to do with zaptel |
| 20:06.37 | ManxPower | once you do a "make install" then it's up to you to make your OS load the driver on boot. |
| 20:06.41 | ManxPower | What card do you have anyway? |
| 20:06.50 | MrNaz | digium tdm410 |
| 20:06.57 | MrNaz | 404 configuration |
| 20:07.01 | MrNaz | 404B |
| 20:07.14 | ManxPower | I hope you used the latest zaptel source. |
| 20:07.22 | ManxPower | older versions of zaptel do not have support for that card. |
| 20:07.28 | MrNaz | i just apt-got it 5 minutes ago, i assume its the latest |
| 20:07.45 | Qwell | from where did you get it? |
| 20:07.48 | ManxPower | MrNaz: you sure are naive |
| 20:07.51 | MrNaz | i would hope the repos keep that up to date |
| 20:07.57 | MrNaz | doh |
| 20:08.07 | ManxPower | If you build from a package then you should not be here you should be talking to the package maintainer, this channel is for building from source. |
| 20:08.10 | MrNaz | Qwell from the ubuntu repo |
| 20:09.32 | MrNaz | the version in the repo that i got is 1.4.10 |
| 20:13.18 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 20:13.26 | exothermc | Where can I find some good docs on asterisk |
| 20:15.17 | MrNaz | exothermc there's a really good oreilly ebook you can get from www.asterisk.org |
| 20:15.36 | bbryant | ~book |
| 20:15.37 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:15.59 | MrNaz | bbryant i believe all oreilly ebooks are also available in print |
| 20:16.13 | bbryant | they are |
| 20:16.34 | exothermc | Ya that is great for a basic install, but is there nothing besides the source code for a complete reference? |
| 20:16.51 | tzafrir_laptop | MrNaz, aptitude install zaptel zaptel-source; m-a a-i zaptel |
| 20:17.00 | Strom | exothermc: there's a complete reference in the back |
| 20:17.57 | MrNaz | tzafrir_laptop yeap... that's what the howto said... i'm in the process of doing that now |
| 20:17.59 | MrNaz | thanks |
| 20:19.21 | exothermc | searches oreilly book for "busy-level" with no results |
| 20:19.35 | exothermc | define complete. |
| 20:19.56 | Strom | well, instead of being snarky about it, why not just ask your question? |
| 20:20.11 | exothermc | I did |
| 20:20.32 | MrNaz | exothermc just because grepping for a string finds nothing does not mean that the answer to your question isnt there |
| 20:20.35 | Strom | ...I mean the question you're trying to answer by looking in the documentation |
| 20:20.37 | MrNaz | grep != search |
| 20:21.40 | exothermc | documentation helps me understand different possibilities of the software, and what functions I could implement, I'm sure this is not the forum for such a broad question as that. |
| 20:22.27 | Strom | really now. |
| 20:22.31 | Strom | try it. |
| 20:22.34 | Strom | surprise yourself. |
| 20:23.04 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 20:24.26 | exothermc | Ok what are all the configuration parameters of of sip.conf that aren't in the book, and how do they function? |
| 20:24.26 | x86 | snarky hehehe |
| 20:24.52 | *** join/#asterisk hsv-al` (n=hsval@66.0.46.210) |
| 20:24.58 | Strom | exothermc: what are you actually trying to accomplish? |
| 20:25.08 | exothermc | Strom: knowledge |
| 20:25.11 | Strom | surely you had a task in mind when you grepped for "busy-level" |
| 20:25.49 | exothermc | Not really I stumbled upon limitonpeer which led me to busy-level which I couldn't find information for. |
| 20:26.26 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 20:26.26 | *** mode/#asterisk [+o mog] by ChanServ |
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| 20:26.41 | exothermc | at the time I was looking at presence type functionality, but I highly doubt the busy-level has to do with that. |
| 20:27.17 | Strom | quick google search reveals its function |
| 20:29.56 | exothermc | Strom: Man you are thick. Did I ever ask what busy-level did? No simply me finding it outside of any other reference leads me to believe there maybe other items like it which I may find useful and someone else who wrote the documentation wasn't interested in. I was simply asking is there a comprehensive source. To which I have to drag you over several lines of chat to get a 'No' by implication. |
| 20:30.27 | hsv-al` | heh |
| 20:31.10 | hsv-al | :) |
| 20:31.14 | exothermc | You must work in sales for a company, can't just give the straight no we don't have that or I don't know. It has to be the tell me your story so I can find a solution for you. |
| 20:31.42 | Corydon76-dig | exothermc: chill. He's doing his best to help |
| 20:31.45 | outtolunc | feels the love |
| 20:32.18 | outtolunc | maybe if you searched 'call-limit' you would have better results <G> |
| 20:32.31 | MrNaz | exothermc i'm not an asterisk user (yet) but i can tell an ass when i see one... and when i look at your attitude here all i can see is a giant pair of buttocks |
| 20:32.51 | Strom | exothermc: actually, before you go insulting me, perhaps you should realize that the vast majority of people who come into this channel looking for help are actually barking up the wrong tree entirely and are asking a question that is off the mark from what they actually want to know...so, from experience, just answering vague general questions like that is usually unhelpful. |
| 20:33.36 | Qwell | also, considering that busy-limit doesn't actually exist... |
| 20:33.46 | Qwell | rather, busy-level |
| 20:34.00 | *** join/#asterisk wideser (n=wideser@viper.office2-ww.wideideas.net) |
| 20:35.38 | putnopvut | Qwell: close, it's busylevel in sip.conf |
| 20:35.52 | MrNaz | tzafrir_laptop ok i've done the m-a, but ztcfg still reports drivers not being loaded... how do i load the driver? (and yes i know this is linux stuff not really asterisk so i really appreciate your help here) |
| 20:35.53 | Qwell | that isn't what he searched for. :) |
| 20:35.58 | putnopvut | Qwell: right. |
| 20:36.22 | putnopvut | Also, that book is written as a reference for Asterisk 1.4, and busylevel is an option that is only in trunk/1.6.0, so that's why it isn't there. |
| 20:36.43 | *** part/#asterisk ddunavant (n=David@75.145.240.14) |
| 20:36.49 | putnopvut | As far as documentation for options in sip.conf, the sip.conf.sample file in the configs/ directory is probably your best bet, exothermc. |
| 20:37.01 | exothermc | putnopvut: ok thanks that is helpful. |
| 20:37.40 | putnopvut | The same goes for pretty much all of the conf files. |
| 20:37.46 | tzafrir_laptop | MrNaz, what is the output of: lsmod | grep ^zaptel |
| 20:38.20 | Strom | but seeing as how useless the asterisk book is to exothermc, i think we should take the documentation quill out of lmadsen's hand and give him some inside-out underwear and shove him back in the dicklicking room already !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 20:38.37 | lmadsen | o.O |
| 20:38.51 | Strom | heheheh |
| 20:39.00 | MrNaz | tzafrir_laptop : zaptel 200324 0 |
| 20:39.05 | wideser | anyone using AEL? I'm trying to do a regex extraction and it isn't working using NoOp($[ "${CHANNEL}" : "\[^/\]+/(.+)\[-\]\[^-\]+" ]) |
| 20:39.57 | wideser | That yields "" |
| 20:40.51 | wideser | if written in extensions.conf I get the real sip user |
| 20:41.11 | tzafrir_laptop | MrNaz, what card do you have? |
| 20:41.47 | outtolunc | why aren't you using the REGEX function? |
| 20:42.52 | MrNaz | tzafrir_laptop tdm410 |
| 20:43.11 | wideser | duuh I du no. missed that it exists :) |
| 20:43.24 | outtolunc | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+regex |
| 20:43.28 | tzafrir_laptop | modprobe wctdm2400xxp |
| 20:44.36 | Strom | no on |
| 20:44.39 | Strom | er, no no |
| 20:44.43 | wideser | but shouldn't that work anyway? |
| 20:44.47 | Strom | modprobe wctdm24xxp |
| 20:46.07 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 20:47.16 | wideser | ahhh. I see why I haven't been using REGEX function. I need the sub string that matches in side the () section. not just 1 if there was a match. |
| 20:48.04 | MrNaz | $ modprobe wctdm2400xxp returns: "FATAL: module not found" while modprobe wctdm24xxp returns nothing |
| 20:48.21 | wideser | according to doc at http://www.voip-info.org/wiki/index.php?page=Asterisk+func+regex |
| 20:48.24 | Qwell | the latter is good |
| 20:48.52 | MrNaz | Qwell ok... does that mean the driver is loaded? |
| 20:48.59 | Qwell | yes |
| 20:49.08 | MrNaz | oh wow |
| 20:49.13 | MrNaz | ztcfg is doing something different |
| 20:49.22 | MrNaz | gets excited and starts throwing his popcorn around |
| 20:49.28 | Strom | is it making that quesadilla I ordered? |
| 20:49.39 | Strom | because ztcfg makes a damn good quesadilla |
| 20:49.48 | MrNaz | bloody hell... its 6:30am and i've spent the last 4 hours researching asterisk and trying to get all this working... i'm a sad, sad man |
| 20:49.52 | Qwell | Strom: You know what I miss? |
| 20:49.57 | Strom | Qwell: taco trucks? |
| 20:49.59 | Qwell | yes |
| 20:50.07 | Qwell | you sir, are good :p |
| 20:50.38 | Strom | taco trucks are the best |
| 20:50.41 | Qwell | that, and the corn guy |
| 20:50.50 | Qwell | or the tamale guy |
| 20:50.53 | MrNaz | Qwell ok zaptel -vvv tells me "One channel to configure" shouldnt that be 4 seeing as i have 4 fxo modules? or do i have to state that in the .conf ? |
| 20:50.55 | Strom | if and when you come back down here for a visit, i'll take you to my favorite one |
| 20:50.59 | Strom | OMG YES, the tamale guy |
| 20:51.03 | Qwell | your favorite taco truck? |
| 20:51.04 | exothermc | can't imagine living somewhere that didn't have taco trucks |
| 20:51.08 | Strom | yes |
| 20:51.10 | Qwell | exothermc: it's sad. |
| 20:51.14 | *** join/#asterisk s0lid (n=s0lid@124.106.140.114) |
| 20:51.28 | *** join/#asterisk aliver (n=aliver@ip-216-17-160-99.rev.frii.com) |
| 20:51.45 | MrNaz | does zaptel have a web site or docs i can read so i dont have to bug you guys for hand holding ? |
| 20:51.47 | aliver | Does asterisk 1.4 come with fax detection or do I have to use that nvfaxdetect thing? |
| 20:52.11 | aliver | I didn't see any mention of it in the docs. |
| 20:55.48 | keith4 | aliver: zap does the faxdetection |
| 20:55.48 | keith4 | MrNaz: you need something that's not in the wiki? |
| 20:55.59 | James|TCC | hey keith4 |
| 20:56.19 | James|TCC | has a working Asterisk now (not NOW) :P |
| 20:56.53 | Strom | is it now AsteriskNOW now? |
| 20:57.03 | James|TCC | and i think its safe to say zap groups dont exist in asterisknow, as i had them working after about 5 minutes |
| 20:57.06 | James|TCC | no :P |
| 20:57.07 | Strom | i think it's already ready already |
| 20:57.08 | keith4 | James|TCC: congratulations. doesn't it feel better? |
| 20:57.12 | James|TCC | vast difference aint it lol |
| 20:57.27 | exothermc | What are the ways to deal with hung channels, for instance if a call is on hold then the user is disconnected from the network? I know sip session timers are one way of handling that, but if they aren't supported on the end points what are the next bets? |
| 20:57.33 | James|TCC | the next problem we have is the handsets lol |
| 20:57.51 | lmadsen | exothermc: handle the rtp, and setup the rtptimeout option in sip.conf |
| 20:58.02 | MrNaz | keith4 i didnt know there was a wiki for zaptel |
| 20:58.06 | James|TCC | the phones we have a 4 line flexor 500's, if i just pick up and dial, it works, but how do i get the line buttons assigned to specific lines? |
| 20:58.21 | *** join/#asterisk talntid (n=erict@66.208.251.170) |
| 20:58.21 | Strom | i've never heard of "flexor 500" |
| 20:58.25 | Strom | is it a sip phone? |
| 20:58.42 | James|TCC | atm if i pres line 1 for example, it just says "calls not possible" |
| 20:58.48 | exothermc | lmadsen: iirc that wouldn't effect the on hold scenario since no rtp would be expected to be passed. |
| 20:58.49 | keith4 | MrNaz: this might be a good starting page: http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf |
| 20:58.49 | James|TCC | theyre camrivox phones |
| 20:58.51 | James|TCC | and yeah sip |
| 20:59.07 | MrNaz | keith4 thanks |
| 20:59.12 | James|TCC | internal calls, and direct dialled outgoing ones work |
| 20:59.25 | keith4 | MrNaz: the See Also section should lead you where you need to go, too |
| 20:59.31 | Strom | James|TCC: fff, i have no idea how to provision those...but presumably you have to do that |
| 20:59.32 | tzafrir_laptop | MrNaz, a simpler starting point: run zapconf or genzaptelconf . For analog cards it's all you need |
| 20:59.36 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:59.54 | tzafrir_laptop | almost |
| 20:59.57 | James|TCC | i had to log into the phone and configure its sip account |
| 20:59.58 | MrNaz | tzafrir_laptop nice.... thanks |
| 21:00.02 | James|TCC | and its logged on etc |
| 21:00.20 | aliver | keith4 When you say zap does fax autodetection, does that include just running the ztdummy module without zaptel hardware? I'm using a pure software SIP-trunking Asterisk box with no Zaptel hardware. |
| 21:00.35 | keith4 | ahahahaaa.... good luck with that |
| 21:00.50 | ManxPower | aliver: no, it means fax detection in zap only works for zap channels |
| 21:01.06 | ManxPower | aliver: there is an Asterisk app called NVFaxDetect that works on zap and non-zap stuff. |
| 21:01.08 | *** join/#asterisk Sky-Knight (n=Sky-Knig@ip68-2-137-221.ph.ph.cox.net) |
| 21:01.13 | aliver | keith4 funny you should say that, cause the fax detection we have now in 1.2 works fine with the nv hack. |
| 21:01.18 | *** part/#asterisk Sky-Knight (n=Sky-Knig@ip68-2-137-221.ph.ph.cox.net) |
| 21:01.29 | ManxPower | Granted, if you are trying to send FaxOverVoiceOverIP you are crazy and should be put down for your own good. |
| 21:01.52 | ManxPower | keith4: we use NVFaxDetect for ALL our fax detection |
| 21:02.00 | aliver | ManxPower that's what I'm using now, and yes, just for receiving. But it's in 1.2 and I'm being told to upgrade to 1.4. |
| 21:02.15 | Strom | ManxPower: you forgot to add the Magic Brownie Transport Layer and Ass to Balls Transfer Protocol into the mix |
| 21:02.19 | MrNaz | tzafrir_laptop genzaptelconf generated the zaptel.conf file, but zapconf spits the error: No default file at /usr/share.... |
| 21:02.19 | ManxPower | aliver: MANY people still run 1.2 |
| 21:02.51 | lmadsen | no one runs 1.2 |
| 21:02.57 | lmadsen | except those who do |
| 21:03.01 | outtolunc | haha |
| 21:03.03 | aliver | ManxPower I just wanted to make sure that the NVfaxdetect stuff hadn't been integrated with 1.4 |
| 21:03.05 | *** join/#asterisk rupa (i=rupa@gw.rupa.com) |
| 21:03.23 | aliver | ManxPower Well, I'm being _told_ (by my boss) to upgrade to 1.4. |
| 21:03.31 | ManxPower | Best of luck with that. |
| 21:03.37 | aliver | 'cause, you know, newer and shinier is always better. |
| 21:03.46 | aliver | Ugh. yeah, thanks. |
| 21:03.47 | ManxPower | make SURE you read all the upgrade.txts in the 1.4 doc directory |
| 21:03.52 | outtolunc | hands lmadsen another fortune cookie to read <G> |
| 21:04.00 | ManxPower | aliver: in Asterisk shinier usually means "undiscovered bugs" |
| 21:04.05 | rupa | arggh, I am so upset with linksys/cisco. They refuse to RMA my linksys ATA because they require me to go through "My reseller". Heck if I know who that was, I bought it online somewhere and don't have the receipt. Any tips on how to deal with linksys/cisco? |
| 21:04.26 | aliver | ManxPower done that. Fortunately, I've also tested what I've got so far on the new 1.4-based box and it seems to work fine 'cept for faxdetect. |
| 21:04.28 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 21:04.28 | ManxPower | I would seriously consider quitting if I had to upgrade all our 1.2 servers to 1.4 |
| 21:04.50 | ManxPower | aliver: Based on what I've seen on the channel, most of the remaining bugs in 1.4 only happen under load. |
| 21:04.54 | aliver | ManxPower as for newer not always being better, trust me, I concur. Not my call. |
| 21:05.02 | Qwell | Strom: Where's QLA? |
| 21:05.14 | Qwell | is that new? did I miss a memo? |
| 21:05.17 | aliver | ManxPower Well, that's good to know, at least. |
| 21:05.31 | ManxPower | aliver: MANY people run 1.4 with no issues, many don't. |
| 21:05.36 | lmadsen | likes 1.4 a lot |
| 21:05.50 | ManxPower | lmadsen: done much MixMonitor/Monitor or Queues? |
| 21:05.54 | lmadsen | yep |
| 21:05.55 | lmadsen | both |
| 21:06.05 | ManxPower | Those seem to be the ones people have problems with in 1.4. |
| 21:06.06 | lmadsen | and ChanSpy() too |
| 21:06.07 | aliver | I'm sure my 1.4 will break just after I give it enough time that it'll be a real PITA to move over. |
| 21:06.21 | ManxPower | At least we are getting fewer and fewer "I upgraded and X broke" reports on the channel |
| 21:06.26 | lmadsen | I use Monitor() w/ Queue() + ChanSpy() |
| 21:06.29 | lmadsen | all at the same time :) |
| 21:06.39 | ManxPower | lmadsen: You are TRYING to have deadlocks aren't you? |
| 21:06.58 | lmadsen | personally, PBXs should not be upgraded -- new installs should be performed on a separate box, and then the box swapped out |
| 21:07.14 | aliver | My boss is also crying for an Asterisk GUI. Should I kill him or is there something I can hand him that's going to work well enough to shut him up? |
| 21:07.20 | ManxPower | lmadsen: I usually just use a new HD and save the old one for 6 months or so. |
| 21:07.34 | ManxPower | aliver: kill him. |
| 21:07.35 | lmadsen | ManxPower: that could work too -- I never upgrade the production box |
| 21:07.44 | Strom | aliver: why does he want a GUI? |
| 21:07.48 | ManxPower | aliver: All GUIS for Asterisk totally take over all dialplan and config files |
| 21:07.49 | lmadsen | aliver: use the Asterisk GUI |
| 21:07.56 | lmadsen | Strom: because managing the system is useful |
| 21:07.58 | aliver | ManxPower That's my thought. Ammo is easier to get, even if it has gone up. |
| 21:08.13 | aliver | Strom he's a windoze freak. |
| 21:08.14 | lmadsen | ManxPower: asterisk-gui does not so much -- I've used it recently, and it's pretty easy to work with |
| 21:08.25 | *** part/#asterisk rupa (i=rupa@gw.rupa.com) |
| 21:08.32 | Strom | lmadsen: most of the time, the answer is "because Mr. Foo wants the secretary to administer the PBX" |
| 21:08.36 | ManxPower | lmadsen: *nod* I'm pretty skeptical of AsteriskGUI, but it does seem to play nicer with customizations |
| 21:08.55 | aliver | lmadsen asterisk-gui == the one digium sells? |
| 21:09.02 | lmadsen | ManxPower: I was skeptical too, but once you learn how it works with the dialplan, the learning curve is pretty easy |
| 21:09.04 | ManxPower | And obviously the one thing you DON'T want is a secretary manging a PBX |
| 21:09.09 | lmadsen | aliver: does not sell -- gives away |
| 21:09.14 | lmadsen | see #asterisk-gui |
| 21:09.18 | ManxPower | lmadsen: on my systems at least I am sure it would not work. |
| 21:09.28 | lmadsen | you have to build a system to use it, but ya |
| 21:09.39 | ManxPower | My dialplan design: For each extension in extensions.conf, set channel variables, then run a macro |
| 21:09.45 | aliver | lmadsen it's a custom "applicance" / distro right? |
| 21:10.01 | lmadsen | aliver: it runs on a distro/appliance -- the GUI is just a GUI... you install it on any OS you want |
| 21:10.07 | Strom | MACSBUG |
| 21:10.08 | lmadsen | with any version of Asterisk (1.4 based) that you want |
| 21:10.21 | aliver | lmadsen interesting. I might have to check that out. |
| 21:10.36 | lmadsen | I had to use it for a customer, and I didn't like it at first, but that's just because I had a bias against GUIs, and I have come to like it |
| 21:10.38 | lmadsen | and that is saying a lot |
| 21:10.49 | lmadsen | it needs a bit of work, but for basic administration stuff, it helps a lot |
| 21:11.03 | lmadsen | and I have not found that it steps on my stuff |
| 21:11.14 | James|TCC | ok, so how do you provision a standard sip phone? |
| 21:11.17 | aliver | lmadsen I normally hate GUIs but it'd be nice to have something to say "Here is some pretty HTML you can click around on and think you are managing things. Now shut up and go away." |
| 21:11.23 | James|TCC | are there any howto's etc around? |
| 21:11.40 | ManxPower | James|TCC: There is no such thing as a "standard SIP phone" |
| 21:11.49 | lmadsen | in my systems, I define devices in sip.conf as MAC addresses, then associates extension numbers with users, and users with the device (so that makes things dynamic since I'm using func_odbc and a database to make it all dynamic) |
| 21:11.56 | ManxPower | Phone provisioning is not a PBX function. |
| 21:12.12 | ManxPower | lmadsen: that's pretty much what I do. |
| 21:12.19 | lmadsen | yep |
| 21:12.36 | lmadsen | devices should *not* be configured as an extension |
| 21:12.48 | lmadsen | i.e. your phone should not register as '205' |
| 21:13.39 | ManxPower | lmadsen: we add -a -b -c, etc to the MAC for each line appearance. |
| 21:14.00 | lmadsen | that works too |
| 21:14.16 | mvanbaak | we use a combination of vpbx id, user id and username |
| 21:14.39 | lmadsen | for another customer we registered as username#vpbx_id |
| 21:14.46 | lmadsen | also worked well |
| 21:15.02 | mvanbaak | vpbx_id-username-user_id |
| 21:15.13 | mvanbaak | so they can have two johns |
| 21:16.12 | keith4 | ~pb |
| 21:16.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:16.16 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 21:16.41 | mvanbaak | works great here |
| 21:16.43 | keith4 | (needed to steal that, to paste in another channl [needs a bot!]) |
| 21:16.59 | Strom | keith4: you can also PM jbot |
| 21:17.09 | Qwell | jbot_: tell keith about pb |
| 21:17.14 | Qwell | maybe |
| 21:17.22 | keith4 | Strom: yah, i tried... but I didn't realize it was jbot_ instead of jbot |
| 21:17.23 | Strom | <PROTECTED> |
| 21:17.25 | ManxPower | I like the mac based solution because "what is your extension" could be answered by "any one of the 5 phones exteison 5412 rings on, whereas "what are the numbers beginning with 0004 on the white sticker on the bottom of the phone?" only ever has one answer |
| 21:17.29 | Strom | <PROTECTED> |
| 21:17.50 | Qwell | ManxPower: an incorrect one, I'm betting. |
| 21:17.59 | ManxPower | Qwell: almost never |
| 21:18.04 | Qwell | "That code doesn't exist in our system.. RE-READ it.." |
| 21:18.19 | Strom | whatever happened to all that hoo-ha about shared line appearances anyway? |
| 21:18.26 | Strom | does it work now |
| 21:18.50 | ManxPower | All Polycom phones have a big white sticker on the bottom of the phone with the MAC on it. |
| 21:18.50 | ManxPower | Strom: I think almost nobody cares. |
| 21:19.00 | Strom | i remember everyone being totally up in arms about it two years ago |
| 21:19.17 | ManxPower | They try Asterisk, whine about lack of Shared Line Appearance, then continue working with Asterisk and realize just how silly SLA is and why they ever thought they needed it in the first place |
| 21:19.40 | ManxPower | repeat the cycle as new users start using Asterisk |
| 21:20.47 | mog | heh |
| 21:23.03 | *** join/#asterisk ac1djazz (i=acidjazz@fc.24.5646.static.theplanet.com) |
| 21:24.37 | ac1djazz | anyone ever do any work/reserach on detecting if you ahve reached someone s vboicemail on an outgoing asterisk call? maybe detecting the beep? |
| 21:24.56 | ManxPower | ac1djazz: You like app_AMD? |
| 21:24.57 | _ShrikE | ac1djazz: core show application amd |
| 21:25.19 | ManxPower | ac1djazz: You might consider doing a "core show applications" once in a while in the Asterisk CLI |
| 21:25.39 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:27.34 | ac1djazz | awesome :) |
| 21:34.14 | ac1djazz | when using AMD to detect a voicemail can i simultaneousely play something? like as a background? |
| 21:34.56 | ac1djazz | or stream a file? |
| 21:40.36 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:41.13 | *** join/#asterisk pjz (n=pj@zachs.place.org) |
| 21:41.50 | pjz | anyone have experience with the digitmaps on a polycom 330 using sip v2.1.2.0049 ? |
| 21:42.12 | pjz | mine's acting like I've got a [67]xxx in it.. but I don't! |
| 21:46.56 | Strom | show me your digitmap |
| 21:47.05 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:47.10 | pjz | well, just to test I changed it to x.T |
| 21:47.55 | pjz | but it was [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[5]xxT |
| 21:48.15 | *** join/#asterisk xand (n=xand@82-71-12-170.dsl.in-addr.zen.co.uk) |
| 21:49.02 | Strom | ok |
| 21:49.05 | pjz | but if I picked up the line and dialed 6xxx or 7xxx it would immeidately dialtone |
| 21:49.28 | Strom | where are you setting the digitmap? in the config file? |
| 21:49.32 | pjz | yeah |
| 21:49.34 | jaytee | ah, the fun of messing with the digitmap in Polycoms |
| 21:49.48 | Strom | pastebin the config file |
| 21:52.05 | lmadsen | don't forget to make sure the timeout options are also matching up |
| 21:52.55 | pjz | http://pastebin.com/m58f4c5c6 |
| 21:53.23 | Strom | that's the entire config file? |
| 21:53.35 | pjz | lmadsen: that's my custom.cfg; there's a whole default sip.cfg too |
| 21:53.42 | pjz | er, that was to Strom |
| 21:53.54 | Strom | we look so alike, lmadsen |
| 21:54.00 | lmadsen | cries |
| 21:54.00 | pjz | lmadsen: honestly, I'm okay with the default imteout of 3 that polycom says |
| 21:54.08 | Strom | pjz: show the default sip.cfg too |
| 21:54.12 | Strom | pastebin the whole thing |
| 21:54.19 | pjz | nah, I was just halfway thorugh typing to lmadsen when Strom asked that q |
| 21:55.05 | *** join/#asterisk infinity1 (i=brendon@saleen.netcal.com) |
| 21:55.37 | infinity1 | hey. i'm using IAX and i'm having intermittent call completion when terminating internationally. is there something i can adjust to help the reliability ? i'm using 1.2.13 |
| 21:55.37 | pjz | you want me to pastebin all 555 lines? eep |
| 21:55.39 | *** join/#asterisk makkksimal (n=makkksim@e177210144.adsl.alicedsl.de) |
| 21:56.05 | pjz | is there a pastebin that takes file uploads? |
| 21:56.47 | Strom | well, just do me a favor and look at sip.cfg and make sure there's no digitmapping in that one |
| 21:57.10 | pjz | well, there is, but custom.cfg is supposed to override it |
| 21:57.26 | pjz | and I've also taken it out of custom.cfg before and changed it in sip.cfg instead |
| 21:58.00 | Strom | ... |
| 21:58.08 | Strom | take it out of sip.cfg, would you? |
| 21:58.29 | pjz | okay |
| 22:00.11 | pjz | http://pastebin.ca/1046472 is the whole sip.cfg |
| 22:00.18 | *** join/#asterisk makkksimal (n=makkksim@e177210144.adsl.alicedsl.de) |
| 22:00.23 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:00.30 | pjz | with the digitmap still in, though I've taken it out and am rebooting the phone and trying that now |
| 22:02.19 | pjz | the bizarre behaviour is that you pick up the line, and it says 'Enter number:' like normal, then you dial [6-7]xxx and it immeidately gives me a dialtone and says 'Enter more digits:' |
| 22:02.49 | pjz | taking the digitmap out of sip.cfg didn't change anything |
| 22:03.10 | Strom | try completely resetting the phone |
| 22:03.15 | Strom | sometimes the polycoms can lose their mind |
| 22:03.37 | pjz | completely resetting how? |
| 22:03.45 | pjz | I've been doing reboots |
| 22:03.52 | pjz | and even have tried coldboots |
| 22:04.52 | Strom | menu 3 2 456 1 4 1 yes |
| 22:05.37 | infinity1 | is there a place where people rate sip/iax termination companies? |
| 22:05.37 | pjz | okay, trying that |
| 22:05.44 | pjz | 'Reset Local Config' ? |
| 22:05.45 | Qwell | ~itsp-list |
| 22:05.52 | Qwell | ~itsp-list-us |
| 22:06.00 | Qwell | jbot_: ... |
| 22:06.05 | infinity1 | ? |
| 22:06.11 | Qwell | stupid bot |
| 22:06.16 | Strom | yes |
| 22:06.21 | Qwell | ~itsplist |
| 22:06.22 | infinity1 | heh |
| 22:06.24 | Qwell | ~itsp |
| 22:06.25 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 22:06.41 | infinity1 | ~itsplist-us |
| 22:06.42 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 22:07.12 | infinity1 | voipjet didn't make the list eh |
| 22:07.39 | outtolunc | voipjet just sent out a no dialer/call center email like yesterday |
| 22:07.54 | infinity1 | anyone use les.net? |
| 22:08.08 | infinity1 | outtolunc: yea. i saw that. whats with that? they don't want business? |
| 22:08.19 | outtolunc | probably got overrun |
| 22:08.27 | ManxPower | Maybe voiojet does "unlimited" service? |
| 22:08.40 | infinity1 | ManxPower: no. i don't think so |
| 22:08.56 | infinity1 | outtolunc: you mean too much variation in volume? |
| 22:09.27 | outtolunc | basically, meaning the call centers were eating all the trunks and the other users complained |
| 22:09.37 | outtolunc | ' i assume ' |
| 22:10.21 | infinity1 | i'm trying to terminate calls in china, and voipjet is intermittent |
| 22:10.32 | infinity1 | sometimes they go through, some times they dont |
| 22:10.36 | infinity1 | not sure what the issue is |
| 22:10.55 | infinity1 | outtolunc: sounds possible |
| 22:11.17 | outtolunc | i haven't sent calls over voipjet since the funds i had disappeared and my login didnt' work, and every attempt i've made to get off their mailing list has failed <G> |
| 22:12.08 | outtolunc | i feel like a ghost, and the volume of ML's is so low i seen no need to 550 them |
| 22:12.10 | infinity1 | outtolunc: for some reason, using iax on my asterisk box gets VERY Poor quality, but iax + voipjet for my friend works perfect |
| 22:12.12 | infinity1 | strange. |
| 22:12.34 | infinity1 | i never figured it out. I just switched to SIP and have had no issues |
| 22:13.14 | outtolunc | maybe they prioritize the sip traffic an not the iax traffic |
| 22:13.41 | infinity1 | outtolunc: voipjet doesn't support sip |
| 22:14.07 | outtolunc | (it has been like 2-3 years since i've used them) |
| 22:14.24 | infinity1 | outtolunc: i have $60 sitting in my account not being used. argh |
| 22:14.33 | outtolunc | fun fun |
| 22:14.51 | outtolunc | i think i had like 19.20 when mine disappeared |
| 22:15.04 | infinity1 | yea. stupid $20 minimum |
| 22:15.15 | infinity1 | which makes some sense, but its annoying :) |
| 22:16.07 | outtolunc | is it friday yet? <G> |
| 22:16.22 | infinity1 | heh |
| 22:18.39 | outtolunc | hahah, storage compression outfit just called and asked why so little (when i stated i only maintain about 100gig of data online) |
| 22:20.20 | outtolunc | it was cute .. sheesh |
| 22:22.37 | cyberdeath | Hi. Is anyone familiar with the Asterisk AA50 Appliance (yeah, the crappy hardware version)? |
| 22:22.54 | ManxPower | cyberdeath: you must contact Digium for support for that product. |
| 22:22.55 | mvanbaak | outtolunc: it _IS_ friday here |
| 22:23.01 | mvanbaak | for 22 minutes already ! |
| 22:23.07 | outtolunc | sweet! |
| 22:23.18 | mvanbaak | outtolunc: it's friday the 13th ! |
| 22:23.37 | mvanbaak | same as 5 years ago |
| 22:23.47 | jblack | bah. Mondays also come early for you, and everyone knows mondays are worse than fridays. |
| 22:23.48 | mvanbaak | the day me and mrsmafkees got married |
| 22:23.56 | jblack | So really, you're screwed. :) |
| 22:23.59 | outtolunc | nice.. then i should probably do that midnight showing of the hulk eh <G> |
| 22:25.10 | mvanbaak | jblack: nah. we got together on friday 13th, my dad is born on friday 13th, my parents got engaged on friday 13th, my grandparents got married on friday 13th, and my grandgrandmother was also born on friday 13th |
| 22:25.26 | mvanbaak | so when I decided I wanted to marry mrsmafkees the date was clear. |
| 22:25.30 | tzanger | he's a witch! BURN HIM!! |
| 22:25.42 | jblack | I could do with a good burning. |
| 22:26.09 | ManxPower | defends mvanbaak with mugwort and lobelia! |
| 22:26.20 | *** join/#asterisk clive- (n=pirch@dsl-242-151-13.telkomadsl.co.za) |
| 22:26.41 | mvanbaak | ;) |
| 22:26.42 | *** part/#asterisk clive- (n=pirch@dsl-242-151-13.telkomadsl.co.za) |
| 22:32.16 | *** part/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 22:42.07 | *** part/#asterisk KenLee (n=k3leland@bg-fw2out.monmouth.com) |
| 22:51.58 | *** join/#asterisk jmacz (n=jmacz@190.158.236.60) |
| 22:59.02 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 22:59.14 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:02.01 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 23:10.55 | *** join/#asterisk PaulQ (n=PaulQ@dc2.ssh.dimenoc.com) |
| 23:11.06 | PaulQ | Easy question, Was AgentLogoff removed from the manager API? |
| 23:11.11 | PaulQ | I cant seem to use it in 1.4 |
| 23:13.13 | PaulQ | channels/chan_agent.c:manager_event(EVENT_FLAG_AGENT, "Agentlogoff", |
| 23:13.17 | PaulQ | Seems to be there |
| 23:14.07 | PaulQ | It registers as "Agentlogoff" but unregisters as AgentLogoff |
| 23:14.34 | *** join/#asterisk raz (n=y@unaffiliated/raz) |
| 23:14.40 | raz | hi gusy |
| 23:15.22 | raz | anyone know a simple tutorial for newbies? i'd like to set up a SIP answering machine (or even small pbx). |
| 23:15.27 | Qwell | ~book |
| 23:15.28 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:15.31 | Qwell | raz: that |
| 23:15.36 | Qwell | it's the best you'll find |
| 23:15.44 | raz | ah ok thx |
| 23:15.49 | Qwell | (read: it's THAT good) |
| 23:15.55 | raz | hehe ok ok |
| 23:16.01 | raz | i'll start reading :) |
| 23:16.12 | PaulQ | Ok so Agentlogoff is missing from the manager API |
| 23:16.30 | PaulQ | Oh |
| 23:16.41 | PaulQ | yea |
| 23:16.42 | PaulQ | hmm |
| 23:16.53 | jaytee | reading it opens your mind, you'll renounce your current religion and embrace Asteriskism |
| 23:17.07 | Qwell | which is a lack of religion. |
| 23:17.09 | PaulQ | That's a bit extreme |
| 23:17.17 | Qwell | PaulQ: again - it's THAT good. |
| 23:17.39 | PaulQ | Does it tell me where my missing Agentlogoff went when it does the manager_register in chan_agent.c |
| 23:17.43 | PaulQ | But it never makes it in? |
| 23:17.48 | PaulQ | Or am I doing something just silly |
| 23:19.40 | PaulQ | Yeah, this is annoying |
| 23:20.11 | PaulQ | Am I suppose to use QueueRemove? |
| 23:20.24 | Qwell | PaulQ: nope.. |
| 23:20.29 | Qwell | not for an agent |
| 23:21.22 | PaulQ | But Agentlogoff does not exist |
| 23:21.23 | Qwell | PaulQ: this 1.4? |
| 23:21.40 | PaulQ | Asterisk 1.4.19.1 built by root @ voip on a x86_64 running Linux on 2008-05-06 16:25:58 UTC |
| 23:21.53 | Qwell | ast_manager_register2("AgentLogoff", EVENT_FLAG_AGENT, action_agent_logoff, "Sets an agent as no longer logged in", mandescr_agent_logoff); |
| 23:21.57 | Qwell | it's certainly there... |
| 23:22.01 | PaulQ | Yea I see it there also |
| 23:22.03 | Qwell | is it showing up in manager show commands? |
| 23:22.20 | PaulQ | Negative |
| 23:22.28 | Qwell | do agents or agentcallbacklogin? |
| 23:22.40 | PaulQ | Also negative, hence my concern |
| 23:22.47 | Qwell | is chan_agent.so loaded? |
| 23:23.00 | PaulQ | chan_agent.so Agent Proxy Channel 0 |
| 23:23.04 | PaulQ | Ah. |
| 23:23.41 | Qwell | if it's showing up, it's loaded |
| 23:23.47 | PaulQ | use count: 0 |
| 23:23.52 | Qwell | module unload chan_agent.so |
| 23:23.53 | Qwell | module load chan_agent.so |
| 23:23.56 | Qwell | any errors/warnings? |
| 23:24.24 | PaulQ | It's my mistake. |
| 23:24.32 | PaulQ | I use AddQueueMember |
| 23:24.41 | PaulQ | Thats not a 'Agent' |
| 23:24.46 | Qwell | there you go |
| 23:25.04 | PaulQ | my mistake completely. |
| 23:25.10 | Qwell | it happens |
| 23:25.29 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:25.31 | PaulQ | So loop thru all queues and call QueueRemove |
| 23:25.54 | Qwell | I think if you're trying to remove an agent from all queues, you can omit the queue name |
| 23:25.57 | Qwell | erm |
| 23:26.01 | Qwell | s/agent/queue member/ |
| 23:26.03 | PaulQ | It errored out |
| 23:26.16 | PaulQ | Message: Need 'Queue' and 'Interface' parameters. |
| 23:26.20 | PaulQ | I thought the same |
| 23:28.08 | Qwell | ahh, it's pause that allows that, I guess |
| 23:28.31 | *** join/#asterisk plla (n=h@200.31.103.86) |
| 23:29.04 | plla | Hello, how do I prevent Asterisk from trying to use g729 to stream files when using it as pass through. |
| 23:29.08 | plla | ? |
| 23:29.20 | Qwell | You can't. That's what passthrough means. |
| 23:29.25 | Qwell | all audio MUST be g.729 |
| 23:30.09 | plla | I am allowing other codecs, can't it pick another codec for local audio and change to g729 when calling to the provider? |
| 23:30.13 | *** join/#asterisk Segnale007 (n=Segnale0@asy-tiv-ppp302.bmts.com) |
| 23:30.21 | Qwell | not if the call is setup as g729 |
| 23:31.06 | Qwell | doing so would require transcoding, which would no longer be passthrough |
| 23:31.16 | plla | hmm, so the only solution is to create those g729 audio files. |
| 23:31.25 | Qwell | correct |
| 23:31.36 | Qwell | or get licenses for g729, so you can transcodew |
| 23:31.38 | Qwell | -w |
| 23:31.41 | plla | Ok, I will do that, thanks. |
| 23:32.01 | Qwell | all of the standard Asterisk prompts are also distributed as g.729. Check `make menuselect` |
| 23:32.04 | plla | g729 is quite expensive hardware wise, I am working with little resources. |
| 23:33.43 | jaytee | what is so great about g729? |
| 23:34.18 | plla | Provider seems to like the compression rate and gives me no other choice. |
| 23:34.32 | *** join/#asterisk [cfdisk] (n=cfdisk@68-116-156-85.dhcp.ftwo.tx.charter.com) |
| 23:34.37 | jaytee | a SIP provider? |
| 23:36.27 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@67.8.117.171) |
| 23:40.51 | plla | Yep. |
| 23:41.47 | plla | I wonder if it's right to use the "free" g729 codec to transform the files. |
| 23:42.54 | drmessano | Don't you mean "illegal" g729? |
| 23:42.57 | _ShrikE | right meaning legal? |
| 23:43.03 | drmessano | Since, there's nothing free about it |
| 23:43.13 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 23:44.25 | *** join/#asterisk coppice (n=chatzill@240.166.17.210.dyn.pacific.net.hk) |
| 23:44.44 | plla | ¯\(º__o)/¯ |
| 23:45.33 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:45.37 | plla | I would not be using it to transcode in realtime... |
| 23:47.17 | d-k-t | I hope a pcie transcoder card comes along soon |
| 23:50.32 | *** join/#asterisk Segnale007 (n=Segnale0@asy-tiv-ppp302.bmts.com) |
| 23:50.42 | ManxPower | plla: contact voiceage and ask them |
| 23:51.29 | ManxPower | voiceage.com, I think |
| 23:51.42 | coppice | I think if you do the transcoding in imaginary time, the licencing gets complex :-\ |
| 23:53.37 | tzanger | coppice: hahaha |
| 23:54.11 | outtolunc | looks for imaginary friends while we are at it <G> |
| 23:54.20 | d-k-t | coppice, should be free as with all imagination |
| 23:54.29 | coppice | tzanger: usually trying a maths joke here gets some response about being a retard |
| 23:54.32 | d-k-t | coppice, unless you're in China I guess |
| 23:54.39 | d-k-t | coppice, but that's a different time of non-free |
| 23:54.47 | coppice | I am in China... sort of |
| 23:54.53 | d-k-t | yeah? |
| 23:54.56 | tzanger | coppice: indeed, but I particuarly liked that one |
| 23:55.08 | d-k-t | ahh hk |
| 23:55.14 | d-k-t | hangzhou here |
| 23:55.46 | coppice | I should be in hangzhou, but the visa system has gone crazy because of the olympics. I live in HK |
| 23:56.16 | d-k-t | I've heard it's now somewhat more difficult to get mainland visas in HK |
| 23:57.17 | coppice | I usually get annual ones. this week all they could offer is a one entry visa, and they need an airline ticket and hotel booking before the application. this has thrown my schedule |
| 23:57.39 | d-k-t | HK used to be the fallback for people coming here if they needed to extend their stay beyond 30 days, night out in HK then fly back with a 6 month multi-entry visa with no 30 day limitation |
| 23:58.25 | d-k-t | what are you coming to hangzhou for? |
| 23:58.28 | coppice | the visa I get always have the 30 day stay limitation (F visa), but that is not an issue for me |
| 23:59.29 | d-k-t | it's easier with a residence permit :) |
| 23:59.38 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 23:59.58 | coppice | anyway this is really stupid. treating business travellers like the olympics tourists is ridiculuous |
