| 00:00.27 | carrar | takes app_voicemail_plain.so and throws it on the ground! |
| 00:07.06 | *** join/#asterisk Trepinon (n=nemus@lbuttars.dsl.xmission.com) |
| 00:08.09 | Trepinon | could someone help me with about how I could connect three asterisk systems togather? |
| 00:08.21 | lmadsen | Trepinon: please elaborate |
| 00:08.23 | drmessano | Screws |
| 00:08.28 | lmadsen | drmessano++ |
| 00:08.40 | lmadsen | ~drmessano |
| 00:08.41 | infobot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily |
| 00:08.46 | *** join/#asterisk pyite_mac (n=dschreib@unaffiliated/pyite) |
| 00:09.02 | lmadsen | uh huh |
| 00:09.52 | hardwire | heh |
| 00:09.54 | Trepinon | basically I would like to connect three 1.4 asterisk system together and having dificulities with iax2 |
| 00:10.42 | Trepinon | incoming siptrunk --> system1 -> system2 -> system3 -> extension |
| 00:11.30 | Trepinon | type=friend |
| 00:11.30 | Trepinon | context=from-pstn |
| 00:11.41 | Trepinon | these two options in iax trunking I don't understand very well |
| 00:11.56 | Trepinon | all I really want to do is route calls to different systems. |
| 00:12.25 | Trepinon | so ideally system1 -> system2 goes to system 3 or system4 depending on which number is dialed. |
| 00:12.33 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:12.51 | Trepinon | I can get two systems to talk to each other ,but the third one always gives me trouble. |
| 00:14.49 | *** join/#asterisk jtodd (i=fn1y88n9@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 00:14.49 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 00:15.02 | carrar | drmessano, wood screws? |
| 00:18.03 | drmessano | Sure |
| 00:18.09 | drmessano | I prefer metal ones |
| 00:18.13 | drmessano | But to each his own |
| 00:18.49 | carrar | Do they need to be matching? |
| 00:19.11 | *** join/#asterisk DrZeus (n=chatzill@201.226.170.106) |
| 00:19.14 | carrar | And do I need one on each side? |
| 00:19.50 | carrar | Not sure why I can't connect them together with a sound card |
| 00:20.23 | carrar | Should be able to do this in 30 MINS!! |
| 00:23.35 | *** join/#asterisk path (i=path@server1.bshellz.net) |
| 00:25.23 | JafoJ | or your pizzas' free?! |
| 00:27.00 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 00:31.22 | Trepinon | I guess that maybe no one here has ever tried to setup multiple systems? Does anyone know any good tutorials or something I could follow? I basicly just want to route calls and have no idea how it really should be done |
| 00:32.22 | KavanS | Trepinon, google and voip-info are your friends....try to read heavily, and then ask directed questions here |
| 00:32.51 | KavanS | Trepinon, that's how I've acheived good luck...if those are not options, I would suggest going to www.digium.org and purchasing professional support services from the experts ;-) |
| 00:33.44 | Trepinon | I might have to do that i've been googling and I just don't think I am asking the right questions. |
| 00:34.24 | lmadsen | your questions are too general -- what is actually the problem? Use pastebin to show configurations and the issue that is actually happening |
| 00:34.44 | Trepinon | the main questions I have is how do you create an IAX2 trunk to mulitple systems from one system |
| 00:34.54 | lmadsen | just create multiple peers |
| 00:34.57 | Trepinon | the third system never picks up the call |
| 00:35.11 | lmadsen | that's the error that you need to show -- we can't read your console from here |
| 00:35.16 | Trepinon | would I create an exte for those peers? |
| 00:35.32 | lmadsen | you would create a peer connection in iax.conf for each server you want to connect to |
| 00:35.34 | Trepinon | i've beem using the type=friend |
| 00:35.45 | lmadsen | then you can Dial(IAX2/server3/555) for example |
| 00:35.54 | lmadsen | type=friend is fine |
| 00:36.24 | Trepinon | allow=ulaw |
| 00:36.24 | Trepinon | type=friend |
| 00:36.24 | Trepinon | context=from-pstn |
| 00:36.24 | Trepinon | nat=no |
| 00:36.24 | Trepinon | insecure=very |
| 00:36.24 | Trepinon | host=server2 |
| 00:36.26 | Trepinon | dtmfmode=rfc2833 |
| 00:36.33 | *** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 00:36.34 | lmadsen | ~pb |
| 00:36.35 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 00:36.40 | lmadsen | don't flood the channel |
| 00:36.51 | Trepinon | sorry |
| 00:37.01 | lmadsen | don't be sorry, just don't do it :) |
| 00:37.07 | Trepinon | gotcha |
| 00:48.21 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 00:48.21 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:48.40 | DrZeus | hi all. Question: im trying to use the 'Originate' command to generate a call between two softphones(SIP); when the 'sender' picks up the call to get transfered to the receipt the call hangs up |
| 00:49.15 | drmessano | and? |
| 00:49.30 | DrZeus | the call doesn't get connected |
| 00:49.34 | DrZeus | don't know why |
| 00:50.01 | DrZeus | as soon as the sender picks up the call, receives a 'goodbye' voice message, and the call goes down |
| 00:50.19 | drmessano | Who wants first guess? |
| 00:50.38 | DrZeus | im trying to emulate what the flash operator does in the switchboard |
| 00:51.27 | DrZeus | that you can create a call between two phones, using the web board |
| 00:53.37 | joat | DrZeus: have you added a delay to the script that sends the originate? a recent patch to AMI requires that the script hang around until the command is actually executed. if the script disconnects before the "originate" is executed, it is "forgotten" |
| 00:55.11 | DrZeus | today i thought about some loop while the call goes on |
| 00:55.41 | drmessano | Why dont you start off by showing us your command and maybe some debug |
| 00:55.43 | DrZeus | question: when the receipt answers the call, then the 'sender' will be connected? |
| 00:55.51 | drmessano | I mean, we could guess all freakin day |
| 00:55.59 | drmessano | Maybe its a bad hard drive |
| 00:56.01 | DrZeus | drmessano: ok, pastebin? |
| 00:56.06 | DrZeus | or just paste it here |
| 00:56.30 | drmessano | ZIP it and send it to russellâs gmail |
| 00:56.31 | drmessano | Yes, pastebin |
| 00:56.41 | drmessano | Paste it here? Are you kidding? Weâll murder you |
| 00:57.04 | DrZeus | hehehe, hold on |
| 00:57.19 | DrZeus | im using the asterisk-php-api, but let me get the console command |
| 00:57.46 | *** join/#asterisk coppice (n=chatzill@148.162.17.210.dyn.pacific.net.hk) |
| 00:58.05 | *** join/#asterisk psykon (n=psilikon@140-1.35-65.tampabay.res.rr.com) |
| 00:58.42 | DrZeus | http://pastebin.com/d5eac4081\ |
| 00:59.34 | DrZeus | that's how the AMI receives it |
| 00:59.55 | DrZeus | i mean, it receives the actual values of the variables; but that's the format |
| 01:02.34 | DrZeus | what do you think? |
| 01:05.07 | DrZeus | i added an infinite loop to check, but the call just keeps falling over and over |
| 01:05.13 | DrZeus | and redialing again |
| 01:12.28 | *** join/#asterisk simplydrew (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
| 01:13.07 | DrZeus | im reading something about using local channels |
| 01:13.26 | DrZeus | to get connected 2 phones with the script |
| 01:13.29 | DrZeus | ever used that? |
| 01:17.31 | russellb | don't need local channels for that. |
| 01:23.55 | DrZeus | russellb: really? |
| 01:24.03 | russellb | yup ... |
| 01:24.18 | *** join/#asterisk ardnat (n=ardnat@user-0ce2kkt.cable.mindspring.com) |
| 01:24.21 | ardnat | hey |
| 01:24.25 | DrZeus | im trying to use originate, but the calls keep falling down\ |
| 01:24.45 | russellb | printf "Channel: SIP/phoneA\nApplication: Dial\nData: SIP/phoneB" > /var/spool/asterisk/outgoing/silly.call |
| 01:24.49 | russellb | something like that. |
| 01:25.01 | russellb | yeah, originate should work, using the same type of format. |
| 01:25.37 | DrZeus | im sending the commands straight to the AMI console |
| 01:25.42 | DrZeus | but with no luck |
| 01:26.43 | ardnat | hey russell what is your view on if phone companys keep logs of cdr's |
| 01:28.13 | florz | ... then that's called the european union |
| 01:28.27 | ardnat | lol.. |
| 01:28.32 | ardnat | no but seriously |
| 01:28.44 | ardnat | do they keep records |
| 01:29.02 | DrZeus | brb |
| 01:29.24 | florz | in the EU? yes, they have to, by law |
| 01:29.34 | ardnat | ah but what about in the US? |
| 01:29.43 | florz | I don't know :-) |
| 01:29.58 | florz | I guess the NSA does that for them anyhow ;-) |
| 01:30.16 | ardnat | yeah |
| 01:30.20 | ardnat | im not sure |
| 01:30.25 | ardnat | just woundering :P |
| 01:30.49 | ardnat | thats kinda crepy if they do, never though they would |
| 01:31.37 | florz | it's kinda creepy that politicians over here don't seem to grasp what data retention really means ... |
| 01:32.07 | ardnat | :P |
| 01:32.15 | ardnat | you in the EU? |
| 01:32.49 | florz | especially, given that it was actually explicitly _forbidden_ before that law was passed, as far as germany is concerned, if you didn't need the data for billing purposes |
| 01:32.53 | florz | yeah |
| 01:33.17 | ardnat | yeah |
| 01:33.24 | ardnat | kewl |
| 01:33.39 | ardnat | no but even if they loged the data |
| 01:33.49 | ardnat | if you changed your ani and cid they coudnt track you right? |
| 01:34.06 | russellb | pretty much every phone company does ... that's how they do billing |
| 01:34.18 | ardnat | hmm |
| 01:34.39 | ardnat | ah well i mean could some other party trace you |
| 01:34.45 | ardnat | the desitionation of your call |
| 01:34.53 | ardnat | trace you back to their company |
| 01:34.54 | florz | well, if you don't have a flat rate, they don't really need CDRs for billing |
| 01:35.04 | florz | erm, -don't, one of them |
| 01:35.29 | ardnat | dont one of them? |
| 01:35.37 | florz | well, if you have a flat rate, they don't really need CDRs for billing |
| 01:36.50 | ardnat | yeah |
| 01:36.52 | ardnat | but if you do |
| 01:36.58 | ardnat | could they track you |
| 01:37.14 | florz | depends on what you mean by "tracking", I guess |
| 01:37.23 | ardnat | trace the call to you |
| 01:38.00 | ardnat | how does a call get terminated? like sip->server->ptsn->destitnaion |
| 01:38.01 | florz | In principle, "they" always can. After all, that call doesn't move from one place to another by magic. |
| 01:38.21 | ardnat | well i mean haveing the technolgy they have today |
| 01:38.29 | *** join/#asterisk BeeBuu (n=beebuu@121.9.232.26) |
| 01:38.35 | ardnat | the operations set in place with standered calls |
| 01:38.48 | ardnat | not delibertly trying to find where a call if from |
| 01:41.47 | ardnat | so? |
| 01:42.46 | ardnat | yes? lol |
| 01:43.10 | *** join/#asterisk Kumbang (n=kumbang@125.163.83.153) |
| 01:48.03 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 01:48.25 | *** join/#asterisk e4 (n=e4@cpe-76-84-81-72.neb.res.rr.com) |
| 01:55.51 | drmessano | ardnat: Put down that copy of 2600, pl33z |
| 01:56.01 | ardnat | LMAO! |
| 01:56.23 | ardnat | no i mean the real reason i want to know is to try some social engearing stuff out |
| 01:56.27 | ardnat | *legal :P |
| 01:56.47 | russellb | social engearing? |
| 01:56.47 | drmessano | Are you implying 2600 isnât legal? |
| 01:56.52 | ardnat | but i dont want teh custer support reps to like track my my number back |
| 01:57.02 | ardnat | lol |
| 01:57.05 | russellb | facepalms |
| 01:57.08 | ardnat | lamo |
| 01:57.18 | carrar | You wanna uber secret? |
| 01:57.22 | russellb | plz to be helping me be l33t h4x0r kthxbye |
| 01:57.22 | carrar | wanna be |
| 01:57.24 | ardnat | yeah basicly |
| 01:57.34 | ardnat | xD @ russell |
| 01:57.47 | ardnat | nah i just want to try it out |
| 01:58.00 | ardnat | i can already do pen. testing on computers |
| 01:58.13 | ardnat | so i want to see how i can do using voice heh |
| 01:58.25 | drmessano | Heâs been blowing into his softphone with his captain crunch whistle all day and still it wont even register, nevermind m4d trunk h4x |
| 01:58.35 | ardnat | lamo |
| 01:58.38 | carrar | SipAddHeader(X-31337: enable) |
| 01:58.44 | carrar | add that |
| 01:58.48 | ardnat | someone acculey made a system that emulates the old phone system |
| 01:58.53 | ardnat | and you can box it and stuff |
| 01:58.54 | drmessano | Pwn() |
| 01:58.55 | russellb | NOWAI |
| 01:59.04 | ardnat | o god this is what i started |
| 01:59.22 | ardnat | meh im 14 i wasnt arround durrin the days of craption crunch |
| 01:59.46 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 02:00.29 | ardnat | PING |
| 02:00.37 | drmessano | SYN |
| 02:00.45 | carrar | FIN |
| 02:00.49 | ardnat | noo its PONG |
| 02:00.51 | ardnat | ugh |
| 02:00.51 | ardnat | lol |
| 02:01.06 | russellb | ~enter |
| 02:01.07 | infobot | the enter key is not a substitute for punctuation. Use a period '.', exclamation mark '!', question mark '?', comma ',', colon ':', semi-colon ';' emdash '--', or ellipsis '...' instead. |
| 02:01.07 | carrar | h |
| 02:01.19 | *** join/#asterisk jtodd (i=rbw3mnza@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 02:01.19 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 02:01.28 | ardnat | MY CAPS KEY IS LOUD |
| 02:01.29 | drmessano | ardnat: Actually, we adults refer to it as âICMPâ or âTEH MAD WIN95 PORT 135 H4X: |
| 02:01.34 | drmessano | Google it |
| 02:01.50 | ardnat | mk xD |
| 02:01.59 | drmessano | You could be pwning Windows 95 machines RIGHT NOW |
| 02:02.02 | ardnat | did you heard abnout the windows 7 vunl. |
| 02:02.10 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 02:02.10 | ardnat | is it the ping buffer overflow? |
| 02:02.18 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 02:02.25 | russellb | this is not #zomg31337 |
| 02:02.30 | ardnat | lamo |
| 02:02.55 | shmaltz | is there a known problem with DTMF when using chan_local? |
| 02:03.08 | russellb | shmaltz: no, it should be passed through. |
| 02:03.24 | shmaltz | russellb, using 1.2.2x overhere |
| 02:03.27 | *** join/#asterisk ketema (n=ketema@ketema.net) |
| 02:03.32 | russellb | shmaltz: oh dear. |
| 02:03.43 | shmaltz | wonders what next |
| 02:03.51 | drmessano | russellb: i can pwn teh SIP STACK with a coathanger+m4d sk1llz |
| 02:04.02 | ardnat | .. |
| 02:04.07 | russellb | drmessano: NOWAI |
| 02:04.15 | drmessano | cookes up a m4d STACK of 2dozun ROFFLES |
| 02:04.24 | ardnat | mmm.. |
| 02:04.41 | russellb | drmessano: i got so many sploitz man, you got nothin on me |
| 02:05.06 | carrar | Asterisk is really a distributed spam server |
| 02:05.16 | carrar | Just also does voip |
| 02:05.30 | drmessano | russellb: it would be m4d h4x if u cud get liek root into teh Akerisk SVN and liek stuf |
| 02:05.39 | ardnat | xD |
| 02:05.46 | ardnat | puts on my robe and wizard hat |
| 02:06.00 | russellb | drmessano: I actually do have root on that box <3 |
| 02:06.08 | ardnat | really.. |
| 02:06.17 | drmessano | russellb: Cuz like teh open s0res sofwarez use like code |
| 02:06.21 | ardnat | :O report him! |
| 02:06.28 | drmessano | russellb: unliek the close sores |
| 02:06.45 | russellb | YAH DUDE |
| 02:06.49 | ardnat | wholey crap got to do my homwork |
| 02:06.51 | ardnat | adios |
| 02:06.54 | russellb | bye! |
| 02:06.55 | drmessano | leik windoz has no code, but liek astrik, you can leik see it |
| 02:06.57 | carrar | hhaha |
| 02:07.11 | russellb | okay that was fun. |
| 02:07.14 | drmessano | Yep |
| 02:07.28 | drmessano | straightens his bowtie and his clown nose |
| 02:07.47 | drmessano | Back to seerius bisnes |
| 02:08.10 | russellb | yeah, serious stuff like ... |
| 02:08.18 | russellb | have you seen my release schedule post from today? |
| 02:08.31 | drmessano | No, I do believe I have missed that |
| 02:08.41 | shmaltz | russellb, other than upgrading do you know if there is a problem with chan_local on 1.2.x? |
| 02:08.52 | russellb | drmessano: it was on -dev, like 6 hours ago ... http://lists.digium.com/pipermail/asterisk-dev/2009-October/040082.html |
| 02:08.52 | drmessano | OMFG no |
| 02:09.01 | hardwire | drmessano: get with the times. |
| 02:09.14 | russellb | shmaltz: my in-brain bug database cache doesn't go back that far |
| 02:09.21 | shmaltz | :( |
| 02:09.27 | drmessano | Before I go further |
| 02:09.27 | hardwire | lul.. Asterisk LTS |
| 02:09.29 | drmessano | RUSSELL |
| 02:09.36 | drmessano | I see LTS.. please tell me.. |
| 02:09.36 | shmaltz | ~beer for russellb |
| 02:09.37 | infobot | ACTION deftly decants a fine La Maudite for for russellb |
| 02:09.37 | drmessano | Please |
| 02:09.41 | drmessano | No animal names |
| 02:09.43 | drmessano | PLEASE |
| 02:09.45 | russellb | lol! |
| 02:09.50 | russellb | no silly animal names. |
| 02:09.52 | russellb | just numbers. |
| 02:09.54 | Nivex | hehehe |
| 02:09.55 | hardwire | and puppets |
| 02:10.12 | hardwire | russellb: it'll be nice to see LTS like numbers in Debian |
| 02:10.17 | drmessano | ASTERISK 1.6.4 FRISKY FERRET |
| 02:10.27 | russellb | instead, we will name releases after teletubbies |
| 02:10.30 | hardwire | as in Asterisk 1.6.4.LTS and that's that |
| 02:10.31 | drmessano | HAHAH |
| 02:10.41 | drmessano | Asterisk 1.6.4 Dipsy Doodle? |
| 02:10.42 | Nivex | russellb: <achmed>Silence! I kill you!</achmend> |
| 02:10.57 | russellb | :-D |
| 02:11.12 | hardwire | it's a big milestone to get to that point, so yay for digium |
| 02:11.21 | russellb | hardwire: yay for the asterisk project |
| 02:11.53 | shmaltz | russelb, the reason I'm using chan_local is becuase I'm trying to utilize dynamic features could that be the problem? |
| 02:11.57 | drmessano | Lets name them after animals in the format <adjective most associated with said animal> <animal> |
| 02:12.14 | Nivex | Ginormous Giraffe? |
| 02:12.21 | russellb | shmaltz: it ... could be maybe, you could answer that yourself by disabling the dynamic features :-) |
| 02:12.44 | shmaltz | :( |
| 02:12.47 | drmessano | âIâm running stinky.. i just upgraded from dopey and my inbound calls fail.. help?â |
| 02:13.05 | russellb | no no, how about we give it a name that is offensive to some group of people |
| 02:13.16 | shmaltz | russellb, like? |
| 02:13.21 | drmessano | Like Asterisk 1.6.4 drmessano ? |
| 02:13.29 | drmessano | O.o |
| 02:13.31 | shmaltz | :- |
| 02:13.43 | russellb | asterisk 1.8 LTS, "stupid redneck" |
| 02:13.48 | russellb | except something funnier than that |
| 02:13.48 | drmessano | HAHA |
| 02:13.51 | shmaltz | what is LTS |
| 02:13.54 | shmaltz | ~LTS |
| 02:13.54 | infobot | hmm... lts is Leaning Toothpick Syndrome. |
| 02:14.00 | hardwire | however.. this makes my opinion of "make it future ready" a little weird |
| 02:14.00 | hardwire | I was going to go with 1.6 as the stable |
| 02:14.00 | hardwire | now I can just focus on dealing with 1.4 stuff and say the product supports asterisk 1.4 LTS |
| 02:14.00 | hardwire | heh |
| 02:14.01 | hardwire | which isn't really what I want |
| 02:14.03 | hardwire | just talkin |
| 02:14.21 | ian6 | mmm La Maudite. |
| 02:14.22 | russellb | hardwire: 1.4 won't be around for forever. |
| 02:14.33 | russellb | 1.8 LTS will be the new hotness! |
| 02:14.39 | russellb | ... come Q2 of next year or something |
| 02:15.28 | shmaltz | russellb, hot in what way? |
| 02:15.28 | hardwire | 2011 |
| 02:15.34 | hardwire | that's forever in my book |
| 02:15.46 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 02:16.06 | hardwire | moy: I know your cable modems mac address |
| 02:16.10 | drmessano | Actually, russellb, I stand up and applaud a freakin schedule, finally, thank <diety>, like many other vendors and projects have moved to. |
| 02:16.17 | jaytee | 1.8 LTS? |
| 02:16.17 | hardwire | it's not every day I can say that to a stranger |
| 02:16.28 | russellb | drmessano: *nods* |
| 02:16.44 | hardwire | and apparently your firewalls mac |
| 02:16.59 | jaytee | I got SIP/TCP working with Exchange on my RHEL 5 64 bit box today |
| 02:17.06 | russellb | wheeeeee |
| 02:17.11 | shmaltz | jaytee wowowowowowowowowo |
| 02:17.27 | drmessano | seriously.. a way to damn kill off shit for the sake of progress, with a STATIC warning, so youâre no longer bound to pussyfooting around retiring old releases |
| 02:17.28 | jaytee | with SIP refer working so I can use the autoattendant |
| 02:17.29 | russellb | shmaltz: you too could experience such glorious features if you didn't use an ancient version :-p |
| 02:17.32 | hardwire | jaytee: what? |
| 02:17.37 | drmessano | This is damn good |
| 02:17.54 | drmessano | âBut... but.. I love 1.2â |
| 02:18.07 | drmessano | I loved Windows 98SE.. 11 years ago |
| 02:18.08 | jaytee | hardwire, what? |
| 02:18.12 | shmaltz | glorious featues like what? |
| 02:18.15 | hardwire | jaytee: what? |
| 02:18.21 | drmessano | CHAN MOBILE |
| 02:18.22 | jaytee | hardwire, yes |
| 02:18.45 | russellb | like ... svn cat http://svn.digium.com/svn/asterisk/branches/1.4/CHANGES && http://svn.digium.com/svn/asterisk/trunk/CHANGES |
| 02:18.50 | drmessano | I also like how they added the app_stability module in to 1.6 |
| 02:18.52 | hardwire | jaytee: are you screwing with us? |
| 02:18.58 | russellb | i forgot a svn cat on the second URL, whatever. |
| 02:19.06 | drmessano | Much better than app_nocrash in 1.4 |
| 02:19.16 | jaytee | hardwire, i've been testing 1.6.1.7-rc1 for eventually upgrading my 1.4 servers and eliminate sipX as a proxy to Exchange UM |
| 02:19.26 | hardwire | What's Exchange UM? |
| 02:19.31 | drmessano | 1.6.1.x completely rocks |
| 02:19.32 | hardwire | Thats quite possibly why I'm confused |
| 02:19.34 | russellb | jaytee: RC2 is out :-) |
| 02:19.39 | jaytee | Microsoft Exchange Unified Messaging |
| 02:19.48 | hardwire | ah. |
| 02:19.54 | jaytee | I use it instead of Asterisk's voicemail system |
| 02:19.58 | hardwire | and Communicator can use it to do evil? |
| 02:20.10 | drmessano | I had slung tons of monkey poo at 1.6.1.x and it just laughs at me |
| 02:20.15 | moy | hardwire: awesome |
| 02:20.22 | hardwire | moy: I know right |
| 02:20.37 | drmessano | Exchange UM doesnt use Communicator |
| 02:20.40 | jaytee | I haven't messed with Communicator and Asterisk yet but from what I've dug into I think I'd need MS Mediation server in the mix |
| 02:20.41 | drmessano | Different product |
| 02:21.02 | hardwire | drmessano: so Exchange UM is just a SIP endpoint somewhere that you can forward voicemail services too? |
| 02:21.03 | jaytee | Communicator is like a softphone on steroids |
| 02:21.14 | drmessano | Exchange UM _IS_ the voicemail service |
| 02:21.27 | hardwire | jaytee: si.. didn't understand how exchange would use SIP.. I get it now |
| 02:21.35 | drmessano | and stores the messages in IMAP along with Outlook 2007 integration goodness |
| 02:21.45 | hardwire | goodness eh? |
| 02:22.00 | jaytee | Exchange UM gives you voicemail that also puts a wma file in your Outlook inbox, reads your email to you over the phone, lets you check and change calendar appointments. |
| 02:22.04 | drmessano | Youre not big on the sarcasm, are you? |
| 02:22.19 | jaytee | all the cool stuff that drmessano does for free on Google Apps with Asterisk |
| 02:22.24 | hardwire | I've had a day full of suck today drmessano |
| 02:22.24 | drmessano | heh |
| 02:22.30 | hardwire | my today day |
| 02:22.30 | hardwire | yes |
| 02:22.35 | drmessano | loves teh google apps hackin |
| 02:22.49 | hardwire | gdata master eh? |
| 02:22.55 | russellb | drmessano: you should really blog about how you have that stuff set up if you really are doing that |
| 02:23.00 | jaytee | drmessano, it's looking like it's shaping up to be a major player |
| 02:23.09 | russellb | I would even be interested in that kind of thing as a guest blog post on *.org |
| 02:23.25 | hardwire | I feel like I'm missing out on all the fun stuff |
| 02:23.36 | russellb | jaytee: also, a blog post about a successful setup with Exchange UM would be an awesome post, too |
| 02:23.45 | jaytee | I still owe Rod Montgomery a white paper on Nortel Meridian and Asterisk integration |
| 02:24.02 | russellb | that's what happens when you start talking about things going well :-) |
| 02:24.04 | hardwire | those are hard to write |
| 02:24.08 | drmessano | I guess I need to get it structured... I either keep really good, detailed instructions, or completely cryptic notes, unreadable to most. |
| 02:24.24 | hardwire | or you co-develop |
| 02:24.45 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 02:24.49 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 02:24.53 | jaytee | russellb, y'know what I think is interesting? I deviated from one of the blog examples I used as a reference and I can get calls to UM without having canreinvite=no and my phones set to use udp as a transport. |
| 02:24.53 | file | wobbles |
| 02:25.09 | russellb | file: joshhhhhhhhhh |
| 02:25.17 | hardwire | don't call him names. |
| 02:25.21 | jaytee | asterisk is smart enough to know to negotiate the calls and let reinvites happen when and where they need to. |
| 02:25.23 | russellb | everyone say happy yesterday birthday to file ! |
| 02:25.30 | file | hardwire: shaneeeeeee |
| 02:25.36 | hardwire | no.. shaaaaaaane |
| 02:25.40 | drmessano | jaytee: That wasnt so great on 1.6.0.x.. glad itâs working better now |
| 02:25.44 | hardwire | shaneeeeee is so.. odd |
| 02:25.48 | jaytee | happy birthday Josh |
| 02:26.04 | hardwire | jaytee: whats it do differently than 1.6.0.x? |
| 02:26.07 | hardwire | .2 |
| 02:26.21 | hardwire | needs to catch up on stuffs 4 sure. |
| 02:26.23 | jaytee | drmessano, yeah, I had qualms about having everything locking my media through asterisk on every call |
| 02:26.36 | drmessano | The SIP TCP stuff seems to be much better on 1.6.1.x |
| 02:26.54 | ZPertee | does anyone know why ~thebook gives me a link that isn't downloadable? How can I get AsteriskTFOT? |
| 02:26.59 | jaytee | voicemail and PSTN calls are one thing but I didn't want the extra burden of all the sip to sip extension calls going through the server too |
| 02:27.00 | drmessano | ~book |
| 02:27.01 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:27.13 | drmessano | Works fine for me |
| 02:27.34 | hardwire | SIP TCP will be interesting to test in a while |
| 02:27.45 | hardwire | I haven't gotten to test it yet.. but the spa942 supposedly supports it |
| 02:27.48 | drmessano | ZPertee: Click on THE BOOK |
| 02:27.56 | drmessano | Hyperlink FAIL |
| 02:28.00 | jaytee | ZPertee, what drmessano said! |
| 02:28.08 | russellb | it's about to get a lot more robust once a patch that is up for review right now makes it in |
| 02:28.21 | drmessano | russellb: I saw that |
| 02:28.31 | hardwire | russellb: SIP/TCP? |
| 02:28.33 | ZPertee | hey hey got it. thanks you guys are the best. it just took a while to load on my connection. |
| 02:28.35 | jaytee | oh, no! another patch? that'll probably break what I have working :-( |
| 02:28.46 | drmessano | heh |
| 02:28.49 | russellb | jaytee: it won't change anything protocol wise |
| 02:28.59 | hardwire | Adding a method to limit packet size? |
| 02:29.00 | hardwire | :P |
| 02:29.00 | russellb | just robustness against connection problems |
| 02:29.04 | jaytee | russellb, if it does you owe me lunch! |
| 02:29.10 | russellb | jaytee: heh, k! |
| 02:29.15 | drmessano | By that, he means, a koozie |
| 02:29.25 | russellb | how about ... an asterisk pen |
| 02:29.28 | hardwire | interviewed with a company that uses and still processes H.323 video |
| 02:29.33 | hardwire | processes/professes |
| 02:29.40 | hardwire | they are slowly moving to SIP |
| 02:29.42 | jaytee | ooooh! an asterisk pen! |
| 02:29.45 | jaytee | drools |
| 02:30.10 | file | bacon for all! |
| 02:30.18 | russellb | file: turkey bacon? |
| 02:30.33 | file | no. |
| 02:30.37 | russellb | ewak |
| 02:30.42 | russellb | s/ewak/weak/ |
| 02:30.48 | hardwire | haha |
| 02:30.52 | jaytee | mmmm, bacon |
| 02:30.54 | hardwire | s/haha/wtfover/ |
| 02:31.13 | hardwire | I love bacon |
| 02:31.35 | drmessano | I would love to have a highly flammable Digium Asterisk koozie signed by that cute Digium employee with the blue shirt and the red hair in all the flickr photos |
| 02:31.35 | hardwire | s/I \(.*\) bacon/My life is full of \1/ |
| 02:31.35 | russellb | hardwire: you going to astricon again? |
| 02:31.42 | drmessano | Just sayinâ |
| 02:31.45 | russellb | drmessano: the who? |
| 02:31.48 | file | blue shirt and red hair... |
| 02:31.52 | hardwire | russellb: sadly.. no.. You won't have me rubbing on your leg again this year. |
| 02:31.57 | russellb | O.O |
| 02:31.59 | jaytee | hardwire, http://xkcd.com/418/ |
| 02:32.19 | hardwire | jaytee: it's what growing up is all about. |
| 02:32.28 | jaytee | hehe |
| 02:32.38 | russellb | drmessano: link? |
| 02:32.44 | jaytee | yeah, link? |
| 02:32.51 | drmessano | Umm |
| 02:33.04 | russellb | i knew you were full of lies. |
| 02:33.19 | hardwire | drmessano: boy or girl? |
| 02:33.32 | drmessano | Girl, I dont play for the Mets, bro |
| 02:33.39 | hardwire | Bra. |
| 02:33.44 | jaytee | not a turdpusher |
| 02:34.09 | hardwire | really? in this channel? you've got some brass ones jaytee. |
| 02:34.10 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 02:34.17 | drmessano | http://www.flickr.com/photos/roblith/1908106042/in/pool-digium |
| 02:34.30 | file | OH |
| 02:34.32 | drmessano | Maybe its more brown.. |
| 02:34.39 | jaytee | auburn |
| 02:34.56 | hardwire | I'm still back on "wtf is a koozie?" |
| 02:35.08 | jaytee | it keeps your can of beer cold |
| 02:35.27 | hardwire | has a bottle |
| 02:35.31 | drmessano | The lighting is bad in that pic.. Iâll gladly pay for anything professionally made, such as Glamour Shots in the mall.. I am not picky |
| 02:35.32 | hardwire | it's very tall |
| 02:35.40 | hardwire | and full of a special spire cider that I am nomming |
| 02:35.43 | hardwire | koozieless |
| 02:36.04 | russellb | drmessano: I am so linking her to this log. |
| 02:36.10 | hardwire | drmessano: that's the most interesting stalking I've ever heard of. |
| 02:36.16 | drmessano | LOL |
| 02:36.23 | jaytee | where was Asterisk World / VON 2007 ? |
| 02:36.30 | hardwire | sends glamour shots coupons to the neighbor girl. |
| 02:36.35 | hardwire | waits |
| 02:36.52 | russellb | drmessano: she is married, and her husband could probably crush you, fyi |
| 02:36.53 | russellb | :-p |
| 02:37.01 | hardwire | aww.. squishy. |
| 02:37.03 | file | probably? |
| 02:37.05 | jaytee | South Boston? my old stomping grounds!!!! Southie represent!!!!!!! |
| 02:37.13 | drmessano | The first part doesnât bother me.. the second, a little |
| 02:37.13 | file | I don't think there's any question of it. |
| 02:37.32 | file | you *do not* want to get in a fight with him. |
| 02:37.45 | drmessano | lol |
| 02:38.10 | russellb | file knows, *points to file's scar from when his arm was reattached* |
| 02:38.46 | drmessano | This is the second time my dreams have been crushed by Digium.. the first time was when I found out I couldnât plug a SIP into the wall |
| 02:38.48 | drmessano | :( |
| 02:38.53 | russellb | lol |
| 02:39.04 | file | no SIP cable? |
| 02:39.08 | hardwire | heh |
| 02:39.12 | jaytee | I had to exchange my cable box for a new one cuz the old one crapped out after a power failure the other night. I got the new one hooked up and there was still hundreds of channels with nothing worth watching |
| 02:39.15 | drmessano | âBut we have this card...â <-- Screw you guys |
| 02:40.17 | drmessano | jaytee: Funny, a friend of mine in Toledo lost his USB wireless NIC thanks to a power failure the other night |
| 02:40.28 | russellb | i wonder if i set up a web store selling premium SIP cables, if i'd sell any ... |
| 02:40.55 | *** join/#asterisk rizwan (n=u2006231@121.52.144.100) |
| 02:40.58 | jaytee | drmessano, wireless USB nic? weird |
| 02:41.00 | drmessano | I suspect EMP, or that guy that wants to crush me for googling his chick under the pretense of âlooking for cool asterisk picsâ |
| 02:41.10 | drmessano | :( |
| 02:41.13 | jaytee | lol |
| 02:41.40 | drmessano | This is like the worst issue of 2600 ever |
| 02:42.59 | jaytee | somewhere buried in one of the boxes of crap marked misc. that I've been lugging around the country for the past 30 years everytime I moved is my Cap'n Crunch whistle. |
| 02:43.27 | russellb | that guy came to astricon once. |
| 02:43.38 | russellb | 2006. |
| 02:44.10 | drmessano | Lemme guess, he twisted a trixbox appliance into a pretzel |
| 02:44.13 | file | rotates russellb |
| 02:44.15 | drmessano | runs off |
| 02:44.56 | jaytee | John Draper went to Astricon? |
| 02:46.25 | drmessano | Oh, that guy |
| 02:46.40 | drmessano | Sorry, updating my Will |
| 02:47.17 | drmessano | bequeaths all his farmville animals to jaytee |
| 02:47.59 | russellb | jaytee: *nods* |
| 02:48.07 | jaytee | I, Jaytee, being of questionable sound mind spent every goddamn cent I had before I died and had the foresight to run my credit cards up to the limit on gambling and hookers in the days leading up to my death. So long Suckas |
| 02:48.14 | drmessano | russellb: Can I get still get the autographed koozie, before the whole âsmashing in of my faceâ? |
| 02:50.15 | russellb | drmessano: bold |
| 02:50.46 | drmessano | russellb: I know a âyesâ when I hear it.. THANK YA |
| 02:56.36 | carrar | russell, can your premium SIP cables convert g711 to g729? |
| 02:56.37 | [TK]D-Fender | carrar: Monster (tm) |
| 02:56.52 | jaytee | premium SIP cables LOL |
| 02:57.04 | drmessano | googles the $500 denon cat5 cable |
| 02:57.16 | hardwire | heh |
| 02:57.32 | hardwire | is that anything like the $400 100mbit card? |
| 02:57.50 | russellb | it's like monster audio cables ... but for VoIP |
| 02:58.01 | carrar | I heard someplace that premium SIP cables are 5 pair inside |
| 02:58.19 | drmessano | Have you ever heard about the monster cables and the coathangers? |
| 02:58.19 | russellb | there's an extra pair which helps cancel RTP echo |
| 02:58.21 | *** join/#asterisk rizwan (n=u2006231@121.52.144.100) |
| 02:58.23 | hardwire | http://www.bigfootnetworks.com/killer-nic-k1/ |
| 02:58.39 | carrar | like balanced microphone cable |
| 02:58.50 | carrar | 180 degrees outof phase |
| 02:58.51 | hardwire | now.. if anybody were to slap asterisk on one of those NIC.. they could sell it as a "TRANSCODING AWESOME CARD OF DOOM" |
| 02:59.02 | jaytee | it's the Unobtanium cladding that make them so much better |
| 02:59.03 | hardwire | you could literally use LACP to determine if a "SIP CABLE" is being used. |
| 02:59.04 | hardwire | hehe |
| 02:59.46 | carrar | gold plated RJ45 jacks |
| 03:00.00 | drmessano | Some magazine or website took some audiophiles and had them listen to what they claimed were new high dollar monster cables, but were actually unbent coathangers |
| 03:00.30 | hardwire | drmessano: $500 coathangers? |
| 03:00.34 | hardwire | it makes a difference |
| 03:00.35 | drmessano | They had no clue.. and were going on and on about how great the quality was |
| 03:00.47 | russellb | hardwire: it is so sad to think that people buy the killer nic ... |
| 03:01.08 | hardwire | drmessano: like.. if I were hanging out of a window and had a $500 audio cable to hold on too.. I could trust it |
| 03:01.12 | hardwire | same with a $500 coat hanger. |
| 03:01.27 | hardwire | its bound to be made out of melted astroid fragments. |
| 03:01.32 | carrar | but that nic card is painted black |
| 03:01.37 | carrar | so it has to be better |
| 03:01.51 | hardwire | russellb: I like the idea... but not for what it's sold for. |
| 03:01.51 | drmessano | I like cables made from melted unicorn horns.. they resonate so well |
| 03:01.53 | jaytee | the gold plated RJ-45 connectors eliminate near end crosstalk and the Unobtanium cladding reduces and corona effect. |
| 03:02.25 | hardwire | russellb: having 4 of those in a machine for half the price would make an interesting cluster for some very specific applications. |
| 03:02.44 | drmessano | I dont buy NICâs that donât have wammy bars and turkish cymbals |
| 03:02.45 | file | the killer nic is rather interesting though, and it is cool that it runs Linux |
| 03:03.09 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 03:03.11 | hardwire | jaytee: not to mention the quality does not suffer due to changing lagrange points. |
| 03:03.15 | carrar | and SIP cables have ferroids to help prevent aurora borealis resonate feedback |
| 03:03.49 | hardwire | carrar: as well as the electrostatic discharge of a thousand cats rubbing themselves backwards. |
| 03:03.52 | KavanS | 500$ cat 5 cable? |
| 03:03.53 | KavanS | wtf? |
| 03:03.55 | KavanS | seriouslY/ |
| 03:04.07 | drmessano | Zildjian NICS and Cisco guitar amps are where itâs at |
| 03:04.46 | hardwire | http://www.usa.denon.com/ProductDetails/3429.asp |
| 03:05.29 | drmessano | Cisco guitar amps âWe go to 11!â (with purchase of the 0501-11 AMP11 interface card and TAC agreement) |
| 03:05.44 | jaytee | Zildjian had a cymbal factory in Norwell, MA when I was in high school. I lived in the next town. |
| 03:06.06 | hardwire | because it was so noisy? |
| 03:06.12 | russellb | has 6 Zildjian cymbals. |
| 03:06.16 | jaytee | the brothers that took over the company after Avedis died ruined the company |
| 03:06.30 | jaytee | his sons |
| 03:06.52 | hardwire | would like to hear the sound of a thousand cymbals hitting the street at once. |
| 03:06.53 | carrar | I can do a mean hihat with my mouth |
| 03:07.06 | hardwire | carrar: got youtube? |
| 03:07.13 | hardwire | I need to record my "face wave" |
| 03:07.19 | hardwire | maybe I should do that real quick |
| 03:07.57 | carrar | I've heard rumors about this whole video era |
| 03:08.15 | jaytee | time for some sleep, nite all |
| 03:08.27 | carrar | sleep!@#? |
| 03:08.29 | carrar | wuss |
| 03:08.47 | hardwire | removes self from internet and watches some TV |
| 03:09.21 | carrar | I'm bridging two continents at the moment |
| 03:09.32 | carrar | Thats the POWER of Asterisk |
| 03:10.36 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.225) |
| 03:10.38 | carrar | actually 3 |
| 03:11.30 | hardwire | carrar: I want in |
| 03:12.14 | carrar | no 2 |
| 03:12.26 | carrar | Japan and India are the same continent |
| 03:12.39 | hardwire | ish |
| 03:14.09 | drmessano | Asia needs to be notated as As;ia. |
| 03:14.15 | drmessano | No |
| 03:14.19 | drmessano | Asia needs to be notated as Asi;a. |
| 03:14.29 | hardwire | go back to your wine. |
| 03:14.31 | drmessano | Itâs like a continent with a pause |
| 03:14.44 | carrar | ^^) |
| 03:14.49 | drmessano | LAAAAAAAAAAAAAAAAAAAAAAAANNNNNNNNNNN...........D |
| 03:14.55 | hardwire | drmessano: ASIwwwA |
| 03:15.30 | drmessano | ASI.. *tick*... *tick*... A |
| 03:15.44 | drmessano | Blink, you miss it |
| 03:15.59 | hardwire | bushman ticks? |
| 03:16.20 | drmessano | No, I think its just head lice |
| 03:16.24 | drmessano | Oh, you mean.. |
| 03:17.28 | hardwire | Oh I do. |
| 03:17.30 | hardwire | mean. |
| 03:19.12 | hardwire | is pretty mean |
| 03:19.18 | hardwire | damnit.. what to do with the rest of my night. |
| 03:19.43 | MaliutaLap | hardwire: make my life betterer |
| 03:19.43 | *** part/#asterisk ChannelZ (i=channelz@burner.com) |
| 03:26.43 | hardwire | MaliutaLap: what do you need dear? |
| 03:28.12 | MaliutaLap | hardwire: new right hip, some reliable contract work ... |
| 03:28.27 | MaliutaLap | gets off bus |
| 03:28.44 | hardwire | haha.. biteme too |
| 03:37.23 | fofware | hello, one question the only way to store voicemessages in MySql is using unixodbc, or I can set mysq direct? |
| 03:37.47 | hardwire | you can call a custom command from voicemails app |
| 03:38.33 | fofware | thanks hardwire, is over asterisk 1.6 my question |
| 03:38.44 | hardwire | what? |
| 03:38.51 | hardwire | I'm drinking, speak up. |
| 03:40.09 | fofware | about that question you answer me I forget tell I'm on Asterisk 1.6 |
| 03:40.41 | hardwire | ok. thankfully that doesn't change my answer. |
| 03:40.54 | fofware | I'm drinking too, enjoy it |
| 03:41.18 | hardwire | look at the externnotify options |
| 03:41.36 | hardwire | here's an example |
| 03:41.38 | hardwire | http://mikecathey.com/code/vmnotify/ |
| 03:42.57 | fofware | thanks hardwire |
| 03:47.28 | *** join/#asterisk Trepinon (n=nemus@lbuttars.dsl.xmission.com) |
| 03:48.42 | drmessano | [23:39] <hardwire> I'm drinking, speak up. <-- Win |
| 03:50.47 | psykon | i'm about to re-build * so I can try out cepstral but since that will take awhile on the pIII it's running on. What does the trial include and not include? If it is going to be severely crippled I don't want to waste the time building * again. |
| 03:52.40 | psykon | But I really want to use those voices. Diane in particular. |
| 04:01.58 | *** join/#asterisk dustybin (n=dustybin@thinkdebian.org) |
| 04:09.53 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 04:13.05 | *** join/#asterisk Milad (n=milad@unaffiliated/slackark) |
| 04:34.50 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-wddygksbzcvbufes) |
| 04:40.44 | ZPertee | what is the best way to get cepstral voices working with asterisk |
| 04:46.06 | *** join/#asterisk Trepinon (n=nemus@lbuttars.dsl.xmission.com) |
| 04:46.37 | Trepinon | I am following the asteriskdocs freebox and the command dialplan reload is not showing up all I have is show |
| 04:46.49 | Trepinon | did I not compile asterisk right or did the command change? |
| 04:47.13 | Trepinon | I am following the freebook* from asteriskdocs |
| 04:48.50 | ZPertee | Trepinon, what version of asterisk are you using? |
| 04:48.57 | Trepinon | 1.4.25 |
| 04:49.00 | Trepinon | .1 |
| 04:49.38 | ZPertee | I'm not a great asterisk guru but I am running 1.4.26.2 and it works for me anyhow |
| 04:49.53 | Trepinon | hmmm |
| 04:50.38 | kaldemar | did you compile it yourself? do you have configuration files under /etc/asterisk ? |
| 04:50.50 | Trepinon | yes and yes |
| 04:51.29 | kaldemar | why are you using an outdated version, btw? |
| 04:51.42 | Trepinon | i've had issues with 1.4.26 |
| 04:51.51 | ZPertee | such as? |
| 04:52.01 | Trepinon | not picking up on IAX2 trunks |
| 04:52.20 | ZPertee | ok thanks. I'll keep an eye out for that in mine |
| 04:52.32 | Trepinon | I normaly do everything with freepbx |
| 04:52.45 | Trepinon | so now I am trying to figure out how to run and install asterisk with out it |
| 04:53.04 | kaldemar | are your configuration files generated by freepbx? |
| 04:53.09 | Trepinon | no |
| 04:53.15 | Trepinon | clean install |
| 04:53.23 | Trepinon | i did the make samples command |
| 04:53.25 | Trepinon | but thats about it |
| 04:57.42 | Trepinon | hmm so I guess it was because my extensions.conf file was named incorrectly. |
| 04:57.58 | Trepinon | I had it as extension.conf instead of extensions.conf |
| 04:59.23 | kaldemar | how did that happen with make samples? |
| 04:59.34 | Trepinon | I renamed the file |
| 04:59.41 | Trepinon | ID10T error |
| 05:00.00 | *** join/#asterisk maragani (n=rahul909@122.169.130.130) |
| 05:04.57 | *** join/#asterisk maragani (n=rahul909@122.169.130.130) |
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| 05:15.07 | fofware | hi, one more question I don't understand why I get this error http://pastebin.ca/1603340 some one can help me? |
| 05:17.26 | *** join/#asterisk Tim_Toady (n=moi@adsl10-199.kln.forthnet.gr) |
| 05:19.44 | p3nguin | No entry in voicemail config file for '2001' |
| 05:20.01 | p3nguin | Should be self-explanitory. |
| 05:20.39 | p3nguin | And as you can see, it's a warning, not an error. [Oct 8 02:08:12] WARNING |
| 05:21.15 | kaldemar | "2001@default),u" <-- that looks strange. show the extension. |
| 05:21.38 | *** join/#asterisk renzoe (n=clc@213.132.40.2) |
| 05:23.36 | p3nguin | If you're satisfied that the voicemail.conf is configured correctly for that exten, then I'm curious if you didn't reload after you configured it. |
| 05:24.52 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 05:26.45 | renzoe | guys how can i configure the channels? i tried to do dahdi show channels but nonw showed up |
| 05:27.58 | *** join/#asterisk Get_The_Fish (n=IceChat7@c-24-8-50-199.hsd1.co.comcast.net) |
| 05:29.00 | Get_The_Fish | hello all- dbget is only for the berkeley astdb, correct? I cant apply that to an odbc database, that would be a seperate dialplan app right? |
| 05:29.48 | fofware | p3nguin: but I did defined 2001 in voicemail.conf like you can see in the paste |
| 05:29.57 | kaldemar | Get_The_Fish: right |
| 05:30.08 | kaldemar | fofware: are you going to show the extension? |
| 05:30.17 | Get_The_Fish | k thanks... |
| 05:31.19 | p3nguin | fofware: Note the part where I asked if you reloaded after you defined it. |
| 05:31.21 | fofware | kaldemar: yes giveme min |
| 05:31.23 | Get_The_Fish | anyone have a link to a good explanation on the astdb family/key system- cant find one that makes it click |
| 05:32.28 | p3nguin | Well, I guess I didn't really ask... fofware, did you remember to reload after you set up the voicemail.conf? There, now I have asked. |
| 05:33.02 | fofware | p3nguin: yes I did reload and try restarting asterisk too |
| 05:33.32 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202) |
| 05:33.53 | fofware | kaldemar: http://pastebin.ca/1603374 |
| 05:34.44 | kaldemar | fofware: Voicemail(${MACRO_EXTEN}@default),u) <-- you have an extra ) in there. in every voicemail app. |
| 05:35.02 | kaldemar | asterisk is trying to find 2001 in default), not default. |
| 05:35.07 | fofware | uyyy yes |
| 05:37.41 | fofware | kaldemar: yes, thank I guess i need go to sleep |
| 05:37.49 | p3nguin | I wonder how many people would change their context to match the mistake rather than fixing all the mistakes. |
| 05:38.03 | fofware | now it's working |
| 05:38.50 | fofware | p3nguin: thanks to you too |
| 05:39.26 | p3nguin | Glad you got it working. |
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| 06:24.35 | frek818 | Hello |
| 06:25.37 | frek818 | I'm using asterisk 1.6 and I'm trying to work with DIALSTATUS |
| 06:27.16 | frek818 | I have this in my dialplan exten => s,n,Dial(sip/1001&sip/1002&sip/1003,5000,m) |
| 06:27.17 | frek818 | exten => s,n,NoOp("Dial Status ${DIALSTATUS}") |
| 06:27.45 | frek818 | But it never execute NoOp after the dial. Does anyone have any ideas? |
| 06:28.00 | Ad-Hoc | hi |
| 06:28.32 | frek818 | hello |
| 06:29.30 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 06:30.56 | kaldemar | frek818: define never |
| 06:31.48 | frek818 | If I answer the call and hang it up or if I press DND the NoOp step doesn't get executed. |
| 06:32.13 | kaldemar | it's not supposed to execute the next priority if you answer the call. |
| 06:32.29 | frek818 | What about if I DND it? |
| 06:33.15 | kaldemar | i don't know how you're dnd works. but, if you use g option in dial, it will continue the dialplan execution when the destination channel hangs up. |
| 06:33.19 | frek818 | What would you suggest that I do if I want the NoOp to execute regardless |
| 06:33.42 | frek818 | Ok, thank. I'll try that. |
| 06:35.18 | frek818 | That worked! Thank you. |
| 06:37.12 | Milad | I have diguim card which use module : wcte12xp, I have a problem when calling out through E1 line , calls end after 4 secs , |
| 06:37.13 | Milad | I check it and find that after call established , zaptel send connect signal to other side but other side did not respond to that request |
| 06:37.34 | *** join/#asterisk xrmx__ (n=rm@host27-26-dynamic.24-79-r.retail.telecomitalia.it) |
| 06:37.42 | Milad | do you know any config or even any place in source code which I could fix this issue ? |
| 06:59.21 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
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| 07:30.24 | *** join/#asterisk renzoe (n=clc@213.132.40.2) |
| 07:30.48 | renzoe | guys. any one of you using a DV-31 phone? how can i configure it to work with asterisk |
| 07:30.50 | renzoe | ? |
| 07:33.58 | sfire | renzoe, not here (just figured you would like a response)... are you using SIP or IAX (it appears to support both) |
| 07:36.02 | renzoe | yes but i cannot checnge anything from the menu |
| 07:36.34 | renzoe | its working from x-lite |
| 07:36.43 | sfire | does it have a separate admin section ?? (the phone) |
| 07:36.55 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:37.28 | *** join/#asterisk smtx (n=smtx@p50998557.dip0.t-ipconnect.de) |
| 07:45.34 | renzoe | sfire, hmm i dont think so |
| 07:45.51 | renzoe | when i press menu, it only configures ips |
| 07:46.15 | sfire | there is more to it than that somewhere |
| 07:47.34 | renzoe | hmm probably i am asking the supplier for manuals |
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| 09:10.09 | *** join/#asterisk slowhand (n=slowhand@opensuse/member/medwin) |
| 09:16.00 | slowhand | hello, I have TDM410P and Asterisk 1.6.1.5 with dahdi-linux 2.2.0. I use opensuse rpm package. I believe I have a problem similar to https://issues.asterisk.org/view.php?id=15619 and https://issues.asterisk.org/view.php?id=9096. |
| 09:16.09 | slowhand | can someone here give me a light? |
| 09:20.26 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 09:22.28 | kaldemar | slowhand: the first one is about a gui generating improper configuration in dialplan, and the second one is about caller id signaling with an analog card. what exactly is your issue? |
| 09:24.28 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:25.18 | xrmx__ | is possible to disable the voicemail per user? |
| 09:25.33 | slowhand | kaldemar: according to the 15619 the workaround should work, but after I follow to change the extensions.conf and users.conf the problem remain. (after booting) The line only work after a call from outside. |
| 09:26.46 | kaldemar | xrmx__: yes, just don't configure it. |
| 09:27.11 | slowhand | kaldemar: about 9096, I still cannot manage to receive the call because |
| 09:28.35 | slowhand | kaldemar: sorry, because DTMFCID timed out waiting for ring. I live in Indonesia. |
| 09:28.36 | kaldemar | slowhand: 15619 is a pure gui issue, so you're in the wrong place asking about that. there's #asterisk-gui for it. |
| 09:28.58 | slowhand | kaldemar: ah... ok |
| 09:29.44 | tzafrir_laptop | slowhand, what is the actual problem you have? |
| 09:30.06 | slowhand | kaldemar: sorry I'm new to asterisk |
| 09:30.50 | kaldemar | slowhand: don't worry |
| 09:31.11 | tzafrir_laptop | slowhand, (the problem you have, not a problem that appears to be similar to it) |
| 09:33.49 | kaldemar | there are cid parameters for chan_dahdi, you might want to try those out if you're having a problem with cid. http://svn.digium.com/svn/asterisk/tags/1.6.1.5/configs/chan_dahdi.conf.sample |
| 09:34.33 | slowhand | tzafrir_laptop: ok, I cannot make outside call. After one incoming call (even I cannot receive it) then I can make outsdide call. |
| 09:35.28 | slowhand | about the incoming call there is a message "DTMFCID timed out waiting for ring" |
| 09:35.37 | tzafrir_laptop | try the following workaround: |
| 09:35.55 | tzafrir_laptop | set signalling to 'ls' instead of 'ks' |
| 09:35.56 | slowhand | kaldemar: will try |
| 09:36.23 | tzafrir_laptop | that is: in /etc/dahdi/system.conf use fxsls=... |
| 09:36.49 | tzafrir_laptop | in chan_dahdi.conf just use signalling=auto ; which will owrk in both ways |
| 09:37.36 | slowhand | tzafrir_laptop: just a moment let me try it |
| 09:43.05 | slowhand | tzafrir_laptop: should i reboot or just reload? |
| 09:43.37 | tzafrir_laptop | dahdi_cfg |
| 09:43.45 | slowhand | tzafrir_laptop: reload, still no change |
| 09:43.52 | tzafrir_laptop | and then in asterisk: dahdi restart |
| 09:44.59 | slowhand | I already reboot it :( |
| 09:48.29 | tzafrir_laptop | What is the output of lsdahdi ? |
| 09:51.55 | slowhand | WCTDM/0"Wildcard TDM410P Board1: (MASTER) |
| 09:52.26 | slowhand | change ls to ks doesn't work |
| 09:52.35 | slowhand | sorry ks to ls |
| 09:52.56 | tzafrir_laptop | what do you mean by "doesn't work"? |
| 09:53.15 | tzafrir_laptop | what happens? |
| 09:53.36 | slowhand | now the message is No channel type regoster for 'DAHDI' |
| 09:54.07 | slowhand | Unable to create channel of type 'DAHDI' |
| 09:54.10 | tzafrir_laptop | that's the output of lsdahdi? No channels at all? |
| 09:54.33 | tzafrir_laptop | If not, can you pastebin it? |
| 09:55.13 | slowhand | ok just a minute |
| 09:58.11 | slowhand | http://pastebin.be/21298, for lsdahdi. Sorry for my English |
| 09:58.44 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
| 09:59.03 | *** join/#asterisk renzoe (n=clc@213.132.40.2) |
| 09:59.30 | renzoe | guys. help. xlite gives out 503 error. asterisk is wirking fine 30 mins ago |
| 09:59.41 | renzoe | but after i restart the machine it does that everytime |
| 10:00.49 | kaldemar | renzoe: show a sip debug of a call attempt |
| 10:01.06 | renzoe | how can i do that? in asterisk cli? |
| 10:02.13 | kaldemar | "sip set debug on" in cli |
| 10:04.06 | renzoe | there's no ON |
| 10:04.12 | renzoe | i only did sip set debug |
| 10:04.12 | tzafrir_laptop | slowhand, hmm... http://pastebin.be/21298 not found |
| 10:04.17 | kaldemar | renzoe: what version of asterisk are yu using? |
| 10:04.24 | renzoe | then it says sip debugging enabled |
| 10:04.38 | renzoe | asterisknow 1.5 but based on asterisk 1.4 |
| 10:04.55 | *** part/#asterisk BeeBuu (n=beebuu@121.9.232.26) |
| 10:05.15 | renzoe | ok i tried to connect but there's no logs generated |
| 10:05.29 | kaldemar | well if it says sip debuggin is enabled then we're good to go. give also command "core set verbose 10" and then make the call and pastebin all output. |
| 10:05.49 | tzafrir_laptop | slowhand, is that what you get? "SHOW_NOT_FOUND"? |
| 10:06.09 | renzoe | kaldemar, x-lite cannot make a call because it stops at registering |
| 10:06.25 | renzoe | Registration Error: Service Unavailable" |
| 10:06.40 | kaldemar | renzoe: if there is no output in asterisk's cli, then your x-lite isn't sending messages to asterisk. |
| 10:06.50 | renzoe | hmm |
| 10:07.16 | renzoe | even 3cx cannot connect |
| 10:07.37 | renzoe | server unreachable |
| 10:07.48 | renzoe | but its only in our local network |
| 10:08.48 | kaldemar | you have a connection problem. prove that asterisk is listening to a sip port with "netstat -nlp". |
| 10:09.36 | kaldemar | tzafrir_laptop: that's an error message from pastebin.be :) |
| 10:09.49 | slowhand | tzafrir_laptop: my mistake. It should 21296. hmm I need to change my glass |
| 10:10.26 | renzoe | local address 0.0.0.0:5060 5038 4569 |
| 10:10.53 | renzoe | all are udp except 5038 |
| 10:11.11 | renzoe | i also tried to port forward it to the asterisk ip but no avail |
| 10:11.15 | tzafrir_laptop | slowhand, a glass of coffee? :-) |
| 10:11.33 | kaldemar | 5038 is the manager interface port, it uses tcp. well, you have a network problem. |
| 10:11.35 | *** join/#asterisk knobo (n=user@fcnoos-fw03.freecode.no) |
| 10:11.56 | renzoe | hmmm |
| 10:12.14 | kaldemar | renzoe: if it's on the same LAN, how is port forwarding in the mix? |
| 10:12.14 | slowhand | tzafrir_laptop: he..he... already 3 |
| 10:12.14 | AeroCloud | Anyone know if I setup and IAX2 peer for another server to connect, how I can track how many minutes they have used without additional 3rd party software |
| 10:12.48 | renzoe | no, i already disabled PF |
| 10:13.59 | renzoe | i really dont get why this is happening. |
| 10:14.06 | renzoe | it just worked a while ago |
| 10:14.36 | renzoe | already tried a different pc but still same results |
| 10:15.15 | renzoe | even my visual dialplan cannot ssh to it |
| 10:15.17 | kaldemar | what ip address does your asterisk box have? what ip address does your client send its packets to? |
| 10:15.44 | renzoe | 192.168.1.2XX 192.168.1.1XX |
| 10:16.19 | renzoe | you think its a firewall issue? |
| 10:16.25 | kaldemar | 1. why are you masking private addresses? 2. do those look the same to you? |
| 10:18.13 | renzoe | wmy asterisk is 1.252 and the client that has x-lite has 1.121 |
| 10:18.48 | kaldemar | i wasn't asking the client's ip address, but the address the client is sending it's packets to. |
| 10:19.18 | kaldemar | i.e. where does the client think asterisk is |
| 10:20.26 | renzoe | in the settings i specified the IP address to the asterisk box |
| 10:21.10 | kaldemar | and it is the same address your asterisk box has now? |
| 10:21.29 | renzoe | yes didnt changed it |
| 10:21.37 | kaldemar | is there a firewall on your asterisk box? |
| 10:21.40 | renzoe | the thing is i can ssh to my box |
| 10:21.55 | renzoe | its default se linux is disabled |
| 10:22.22 | renzoe | i havent added anything since i installed asteriskNow |
| 10:22.41 | kaldemar | what does iptables say? |
| 10:23.29 | renzoe | did iptables --list and there's no entry in Chain input,output and forward |
| 10:24.53 | renzoe | hmm do i need to input our subnet in iptables to accept? |
| 10:25.14 | kaldemar | if the policy is ACCEPT, then no. |
| 10:25.46 | renzoe | so its fine the iptables is empty |
| 10:26.41 | renzoe | hmm... |
| 10:29.02 | *** join/#asterisk ast-thecode (n=asterisk@scalise.csata.tno.it) |
| 10:31.33 | renzoe | kaldemar, now i tried to login using my ip phone |
| 10:31.38 | renzoe | and asterisk gives me this: |
| 10:31.38 | renzoe | Scheduling destruction of SIP dialog '145962321224244-566151425643@192.168.1.251' in 32000 ms (Method: REGISTER) |
| 10:33.57 | kaldemar | turn on the sip debug, set verbosity to 10 and see what happens |
| 10:36.10 | renzoe | http://pastebin.com/pastebin.php?dl=m627164b1 |
| 10:37.02 | kaldemar | there's nothing in sip configuration that matches your register request. |
| 10:37.49 | renzoe | you mean the extension? |
| 10:37.58 | renzoe | but i just created that extension 200 |
| 10:38.05 | kaldemar | no. a peer. |
| 10:38.18 | kaldemar | are you using the GUI to congfigure asterisk? |
| 10:38.22 | renzoe | yes |
| 10:38.45 | kaldemar | then #asterisknow or #freepbx is the place to ask these questions. |
| 10:39.03 | renzoe | but can i just do it on CLI |
| 10:39.18 | *** join/#asterisk nicola_pav (n=chatzill@83.244.78.241) |
| 10:39.39 | nicola_pav | hello. trying to create an IVR in arabic |
| 10:39.45 | nicola_pav | i write an agi perl script |
| 10:39.54 | nicola_pav | when it comes to SayNumber |
| 10:40.06 | nicola_pav | it still pronounces it in english |
| 10:40.27 | nicola_pav | what should i do in order to say the numbers in arabic |
| 10:40.38 | nicola_pav | i already have the numbers recorded in arabic |
| 10:40.51 | kaldemar | renzoe: and then the gui bashes your hand-made configurations if you put them in the wrong place. go consult someone who uses the same gui as you. |
| 10:41.52 | renzoe | i see thanks |
| 10:44.53 | kaldemar | nicola_pav: set language in the dialplan with func LANGUAGE |
| 10:45.16 | nicola_pav | kaldemar: i arleady did |
| 10:45.35 | kaldemar | and the files are in the right place? |
| 10:45.40 | nicola_pav | still the function SayNumber in the script speaks in english |
| 10:45.52 | nicola_pav | yes |
| 10:46.09 | nicola_pav | under /var/lib/asterisk/sounds/ar |
| 10:46.18 | nicola_pav | that;s the right folder, right? |
| 10:46.22 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 10:46.43 | tzafrir_laptop | slowhand, hmm... you have KS signalling. What do you have in /etc/dahdi/system.conf ? |
| 10:46.50 | nicola_pav | coz other messages like enterpin and welcome messages i put work |
| 10:49.03 | *** join/#asterisk Dovid[Laptop] (n=annon@213.8.121.90) |
| 10:49.10 | kaldemar | nicola_pav: iirc, the number files should be in a separate dir under the language dir |
| 10:49.41 | nicola_pav | i tried both |
| 10:49.43 | nicola_pav | under /ar |
| 10:49.51 | nicola_pav | and under /ar/digits |
| 10:50.37 | nicola_pav | kaldemar: does indications.conf paly role in this? |
| 10:51.47 | kaldemar | nicola_pav: it should not. channel language and indication courtry are two separate things |
| 10:52.01 | nicola_pav | ok |
| 10:52.14 | nicola_pav | kaldemar: any hint? |
| 10:52.31 | nicola_pav | i am trying now the function SayNumber in a dial plan for testing |
| 10:52.46 | nicola_pav | in my dial plan |
| 10:52.59 | nicola_pav | first line is Set(LANGUAGE(()=ar) |
| 10:53.01 | kaldemar | i'd turn on debug in logger.conf and see if it gives something useful |
| 10:53.17 | nicola_pav | in second line SayNumber(56789) |
| 10:53.18 | kaldemar | is that line a copy-paste from the dialplan? |
| 10:53.29 | nicola_pav | not a copy paste\ |
| 10:53.51 | nicola_pav | i just wanted to make sure that SayNumber works in a dial plan |
| 10:53.53 | kaldemar | if you show configurations, it's best to use a pastebin and copy-paste the conf. |
| 10:53.55 | nicola_pav | it does not |
| 10:59.17 | nicola_pav | kaldemar: i tried norweigian "no" |
| 10:59.28 | nicola_pav | it did not work either |
| 10:59.43 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 11:00.22 | tzafrir_laptop | nicola_pav, SayNumber takes more that having the sounds for the digits. |
| 11:00.52 | tzafrir_laptop | What version of Asterisk do you use? Does it have support in the code for arabic numbers? |
| 11:01.18 | nicola_pav | tzafrir_laptop: 1.2.22 |
| 11:01.32 | tzafrir_laptop | Do you speak Arabic? |
| 11:01.37 | nicola_pav | yes |
| 11:01.47 | nicola_pav | i speak arabic |
| 11:02.03 | tzafrir_laptop | Try listening to the resulting number. It should sound funny |
| 11:02.29 | nicola_pav | it says the number in english |
| 11:02.33 | tzafrir_laptop | It should sound as if some American tourist is trying to speak Arabic :-) |
| 11:02.39 | nicola_pav | :) |
| 11:03.03 | nicola_pav | i understand coz the numbers are pronounced differently |
| 11:03.09 | nicola_pav | but even though |
| 11:03.20 | nicola_pav | it speaks the numbers in english |
| 11:03.21 | kaldemar | ah, is there even language support in the same manner in 1.2? |
| 11:03.34 | tzafrir_laptop | for 1.2, try SetLanguage(ar) |
| 11:03.49 | tzafrir_laptop | or: Set(LANGUAGE=ar) |
| 11:03.58 | tzafrir_laptop | I'm not sure exactly |
| 11:04.21 | nicola_pav | i have the tried the second so far |
| 11:04.21 | tzafrir_laptop | or set: language=ar ; in the channel driver config (e.g. sip.conf) |
| 11:04.23 | kaldemar | 1.2 should have the language function, SetLanguage was deprecated when 1.2 came out. |
| 11:05.45 | nicola_pav | SetLangauge() is deprecated as kaldemar said |
| 11:07.45 | kaldemar | chances are that 1.2 is looking for the digit files in /var/lib/asterisk/sounds/digits/<language> |
| 11:08.10 | nicola_pav | i will try that |
| 11:10.13 | nicola_pav | kaldemar: it worked |
| 11:10.25 | *** join/#asterisk jozza (n=chatzill@89-212-75-91.static.t-2.net) |
| 11:10.30 | nicola_pav | it started to say numbers in arabic |
| 11:10.39 | nicola_pav | but as tzafrir_laptop said |
| 11:10.42 | nicola_pav | it sounds funny |
| 11:10.53 | nicola_pav | coz 21 in english is twenty one |
| 11:11.09 | nicola_pav | in arabic it will say "1" and after that "20" |
| 11:11.14 | nicola_pav | any hint how to solve that? |
| 11:12.10 | kaldemar | record your own digits and use them in some fancy manner. :) |
| 11:14.32 | jozza | hi i have aproblem with audio with chan_h323. On connect it says remoteipaddress for audio is 127.0.0.1 |
| 11:16.02 | jozza | which cannot be right |
| 11:16.55 | jozza | any ideas what could be the problem? |
| 11:26.16 | tzafrir_laptop | nicola_pav, the way to solve this it to provide a language-specific function in say.c |
| 11:26.33 | tzafrir_laptop | As you can see, say.c is rather messy |
| 11:46.00 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 11:51.12 | *** join/#asterisk Methose (n=Methose@38.101.237.250) |
| 11:53.04 | *** join/#asterisk TSM (n=the_soft@fw-lon1.wenn.com) |
| 11:54.00 | *** join/#asterisk bn-7bc (n=bjarne-i@mac.lan.noare-1.holmedal.net) |
| 11:54.42 | TSM | ive restarted my asterisk 1.4 box after putting in a sangoma A200 card and its hanging on 'Waiting for /dev/dahdi' |
| 11:55.31 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:55.31 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:04.41 | slowhand | tzafrir_laptop: ok, now I change all to fxsls. lsdahdi http://pastebin.be/21297, /etc/dahdi/system.conf http://pastebin.be/21298, /etc/asterisk/chan_dahdi.conf http://pastebin.be/21299, /etc/asterisk/users.conf http://pastebin.be/21300 |
| 12:11.43 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:12.50 | Chainsaw | TSM: I'd expect wanpipe instead of DAHDI to drive a Sangoma. |
| 12:12.56 | Chainsaw | TSM: Are you sure you're following the right guide? |
| 12:13.24 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 12:17.25 | TSM | Chainsaw: i just put the card in and restarted the machine |
| 12:17.43 | TSM | Chainsaw: wanpipe starts just befor dahdi |
| 12:17.56 | Chainsaw | TSM: Question remains, why are you trying to operate DAHDI drivers if you have no relevant hardware? |
| 12:19.03 | TSM | Chainsaw: this i dont know, sangoma wiki does talk about dahdi stuff, i just did not do anything, i ran the setup-sangoma util and restarted the computer |
| 12:19.35 | TSM | Chainsaw: i have both a A101 and A200 in the same computer |
| 12:19.43 | Chainsaw | TSM: So, what guide are you following to get these going? |
| 12:21.17 | TSM | Chainsaw: i just use the util in my elastix distro that is setup-sangoma, it creates the wanpipe config automaticly |
| 12:21.35 | Chainsaw | TSM: Yes. What guide that you read talks about DAHDI? |
| 12:22.04 | TSM | Chainsaw: this one, http://wiki.sangoma.com/wanpipe-asterisk-configure , but i did not follow it |
| 12:22.43 | Chainsaw | TSM: Did you miss out dahdi_cfg then? |
| 12:22.52 | Chainsaw | TSM: It is waiting for a DAHDI channel to become available, which isn't happening. |
| 12:23.26 | TSM | Chainsaw: mabey the setup did not work, ile run it again |
| 12:26.06 | tzafrir_laptop | slowhand, hmm.... setting them all up in users.conf is a bit, well, creative |
| 12:26.22 | tzafrir_laptop | It works, indeed. But it is not as intended :-) |
| 12:26.41 | garymc | yo... I accidently unplugged my Polycom phone earlier, now I plugged it back in an hour ago, but the time and date has not corrected itself? Anyone know why this would be? |
| 12:27.06 | Chainsaw | garymc: Does that handset have NTP or SNTP support? |
| 12:27.29 | garymc | hmmm that im not sure. I recently bought them, they are Ip330 phones |
| 12:27.35 | tzafrir_laptop | Move that as-is to [channels] in chan_dahdi.conf , or use the newer 'dahdihcan' syntax to have a cleaner config |
| 12:27.37 | slowhand | tzafrir_laptop: as I said I'm new to asterisk :-) |
| 12:27.54 | Chainsaw | garymc: They'd need to have support for either protocol in order to update their clock automatically. |
| 12:28.04 | garymc | ive had them working. but now cos i unplugged it and plugged back in it hasnt corrected, normaly they correct themselves |
| 12:28.07 | Chainsaw | garymc: It's unlikely that they don't have it, but it could be that it needs configuration (which you may not have done yet). |
| 12:28.23 | garymc | i thought i had, as i got it to work before |
| 12:28.31 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 12:28.36 | Chainsaw | Mind doing a grep -i ntp on your config then? |
| 12:28.54 | garymc | ok |
| 12:29.06 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 12:29.38 | *** join/#asterisk robl^laptop (n=robl@m445336d0.tmodns.net) |
| 12:30.19 | garymc | do i just type that in the command cli in root? |
| 12:30.25 | *** join/#asterisk e4 (n=e4@cpe-76-84-81-72.neb.res.rr.com) |
| 12:30.29 | garymc | cos i done that now nothing happening |
| 12:30.54 | Chainsaw | garymc: You have to tell it where to look. grep -i ntp /etc/gary/phone-system/handsets/blah.txt |
| 12:31.10 | Chainsaw | I don't know where your handset configs are. You do. |
| 12:32.12 | *** join/#asterisk cyberfab007 (n=cyberfab@CPE001b11cf4f69-CM0014f85c3ada.cpe.net.cable.rogers.com) |
| 12:32.14 | *** part/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net) |
| 12:32.19 | *** join/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net) |
| 12:32.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 12:32.21 | garymc | they are built into the phone i think |
| 12:32.32 | garymc | I configured them by going to their ip address |
| 12:32.50 | *** join/#asterisk mythicalbox (n=mythical@rrcs-64-183-110-250.west.biz.rr.com) |
| 12:32.52 | Chainsaw | Alright. Try and have a look for (S)NTP settings there? |
| 12:34.13 | garymc | yes the sntp server is set to the asterisk box ip |
| 12:34.20 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 12:34.28 | Chainsaw | garymc: And you're running an actual SNTP server there? |
| 12:34.49 | *** join/#asterisk oej (n=olle@1x-193-157-197-57.uio.no) |
| 12:34.53 | garymc | well it points to my Linux server which is running the sntp so i suppose iam as one phone is working fine |
| 12:34.56 | Chainsaw | garymc: (Asterisk does not offer one, you'd have to run something in addition to Asterisk itself) |
| 12:35.06 | garymc | yes iasm |
| 12:35.43 | Chainsaw | garymc: Interesting behaviour. Is there a 'sync now' button available? |
| 12:35.59 | garymc | no |
| 12:35.59 | garymc | hold on im checking my notes |
| 12:36.42 | garymc | ok theres an ntp.comf configured on my linux server |
| 12:36.47 | garymc | .conf |
| 12:36.54 | *** join/#asterisk renzoe (n=clc@213.132.40.2) |
| 12:37.02 | garymc | i installed NTPD on my linux box |
| 12:37.13 | Chainsaw | But NTP != SNTP. Want another hostname/IP where SNTP is guaranteed to be running? |
| 12:37.35 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 12:38.46 | kaldemar | the sntp setting for polycom phones is for ntp. |
| 12:39.14 | Chainsaw | kaldemar: Confusing, as those protocols are not the same. I'll leave it with you then. |
| 12:39.25 | garymc | well i got it all working fine, it was all working fine. I turnt one phone off it its time and date is set wrong and flashing |
| 12:39.42 | garymc | i think you helped me get this working in the first place kaldemar |
| 12:39.42 | kaldemar | is the ntpd running? |
| 12:39.51 | garymc | how do i test |
| 12:39.54 | garymc | i think it is |
| 12:39.55 | slowhand | tzafrir_laptop: done, but I still got the warning 'DTMFCID timed out waiting for ring' for incoming call |
| 12:40.09 | kaldemar | garymc: see with ps ax that the process is there |
| 12:40.17 | Chainsaw | slowhand: Is this a Digium TDM400 attached to a UK BT line? |
| 12:40.24 | tzafrir_laptop | slowhand, is the problem the outgoing call or the caller ID? |
| 12:41.29 | garymc | i cant see it. |
| 12:41.48 | kaldemar | garymc: then you need to start the server. |
| 12:42.00 | garymc | When i first did this, my asterisk box never had internet connection. But my ltsp one did so we got the time from the ltsp |
| 12:42.26 | garymc | ?? dont know if that makes a differnce. But my asterisk box now has its own internet connection |
| 12:42.47 | slowhand | tzafrir_laptop: I cannot receive any call, and it is in Bali :-) |
| 12:43.00 | kaldemar | it sure as hell makes a difference if the phones tries to get time from your asterisk box and there's no time server running on the box. |
| 12:43.25 | garymc | ok well maybe there is a time server running i just cant see it |
| 12:43.38 | Chainsaw | kaldemar: I have an SNTP/NTP server in the same country as garymc (UK) if you require it for testing. |
| 12:43.40 | *** join/#asterisk _ShrikE (n=_ShrikE@74.185.215.60) |
| 12:43.54 | kaldemar | Chainsaw: no thanks, i don't. |
| 12:44.19 | slowhand | Chainsaw: TDM410P connect to Indonesia Telkom line |
| 12:44.25 | kaldemar | garymc: what does ps ax | grep ntpd show you? |
| 12:44.29 | garymc | i just typed ps ax in the CLI and ill pastebin my output if thats ok |
| 12:44.35 | garymc | ok hold on ill try |
| 12:44.41 | kaldemar | yes, pastebin that |
| 12:45.16 | Chainsaw | slowhand: Okay. It suggests your caller ID configuration is not set up correctly. (It is waiting for DTMF caller ID tones and never gets any) |
| 12:45.19 | garymc | 9217 pts/1 S+ 0:00 grep ntpd |
| 12:45.20 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:ddb8:ca88:f49b:1b0f) |
| 12:45.30 | *** join/#asterisk oej (n=olle@1x-193-157-197-57.uio.no) |
| 12:45.32 | kaldemar | garymc: so start the server. |
| 12:45.33 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:46.08 | garymc | sorry Kaldemar, i dont understand start the server? u mean NTPD server? or my LTSP server? |
| 12:46.31 | slowhand | Chainsaw: any hint how I correct it? |
| 12:47.49 | Chainsaw | slowhand: I'm trying to find out what caller ID standards are in use in Indonesia. No luck so far. |
| 12:48.19 | slowhand | Chainsaw : I believe it should be the same with nl |
| 12:48.44 | kaldemar | garymc: the ntp server |
| 12:48.54 | garymc | ok how |
| 12:49.00 | garymc | ntp start? |
| 12:49.21 | kaldemar | what distro? |
| 12:49.51 | garymc | AsteriskNOW i think its Centos |
| 12:49.51 | *** join/#asterisk PoWeRKiLL (n=powerkil@bzq-79-181-102-215.red.bezeqint.net) |
| 12:49.59 | tzafrir_laptop | search for a package with 'ntp' in its name? |
| 12:50.09 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:50.17 | tzafrir_laptop | (yeah, I know there's also chrony) |
| 12:50.26 | garymc | ive found ntpd |
| 12:50.36 | garymc | just dont know how to start, restart or stop it |
| 12:50.42 | kaldemar | /etc/init.d/ntpd start maybe |
| 12:50.53 | tzafrir_laptop | hmmmm /etc/init.d/ntp<tab> |
| 12:50.56 | garymc | its ok ive fixed it |
| 12:51.16 | garymc | i changed the ip in my polycom time settings to my ltsp server ip and it works now |
| 12:51.21 | [TK]D-Fender | service ntpd start <- centos |
| 12:51.35 | garymc | ahh thats the one TK |
| 12:51.55 | Chainsaw | slowhand: It helps to be sure in cases like these. The good thing about caller ID standards is that there are so many to choose from. |
| 12:52.31 | garymc | yeah so why doesnt it start when reboot. How do i make it start on reboot? |
| 12:53.11 | [TK]D-Fender | garymc: chkconfig ntpd on |
| 12:53.28 | [TK]D-Fender | garymc: go learn how to administer your distro... #centos |
| 12:53.38 | garymc | Thanks TK |
| 12:53.41 | garymc | thats works now |
| 12:55.38 | *** join/#asterisk hat_panda (n=pete-joh@triton.dsv.su.se) |
| 12:56.01 | garymc | Thanks guys cheers |
| 12:56.41 | renzoe | guys |
| 12:56.45 | *** join/#asterisk qu1ckkkk (n=leonja51@dsl-240-218-251.telkomadsl.co.za) |
| 12:56.47 | renzoe | i did some netstat -nlp |
| 12:57.17 | renzoe | and in the line of 5060, under the "State" column, there is no "Listen" |
| 12:57.26 | renzoe | will that affect the connection? |
| 12:58.08 | hat_panda | Hello, Im trying to setup realtime together with mysql for handling user settings. Its working for inbound registrations, but not for outbound registrations |
| 12:58.14 | *** join/#asterisk klochan (i=kvirc@195.222.93.238) |
| 12:58.49 | hat_panda | anyone using realtime + some database for outbound registrations? |
| 12:58.55 | *** part/#asterisk klochan (i=kvirc@195.222.93.238) |
| 13:03.23 | hat_panda | yes?no? |
| 13:04.38 | hat_panda | E.g. Inbound calls are matched properly against data in db, but no registrations are sent from asterisk to other regstrars |
| 13:07.22 | [TK]D-Fender | hat_panda: I see no mechanism to store registration info in the DB.... |
| 13:07.35 | [TK]D-Fender | hat_panda: And has never been present to my awareness |
| 13:08.28 | *** join/#asterisk Baylink2 (n=jra@cerberus.vicimarketing.com) |
| 13:08.45 | Baylink2 | G'morning |
| 13:09.09 | Chainsaw | Afternoon Baylink2. |
| 13:09.30 | Baylink2 | Yay, timezones. :-) |
| 13:12.09 | Katty | :> |
| 13:12.15 | Katty | is wearing Shiny Birthday Present. |
| 13:12.35 | Katty | it SPARKLES :> |
| 13:13.13 | [TK]D-Fender | Katty: Happy birthday |
| 13:13.23 | Katty | thanks. |
| 13:14.06 | jaytee | is it made of crystalized carbon? |
| 13:15.04 | Katty | no, there are no diamonds on it. |
| 13:15.07 | Katty | it's a gemstone band. |
| 13:15.11 | *** join/#asterisk Tim_Toady (n=moi@adsl10-199.kln.forthnet.gr) |
| 13:15.24 | Katty | multicolored gemstones, in fact. |
| 13:15.30 | robl^laptop | Happy BD, Katty! |
| 13:15.37 | Baylink2 | Spiffy. HBD |
| 13:15.44 | Katty | robl^laptop: thank you (= |
| 13:16.48 | robl^laptop | Katty: did it come in a clear plastic bubble like you would find in on eof those vending machines next to the bubble gum and jawbreaker dispensers? *ducks* |
| 13:17.22 | jaytee | "The Claw has chosen!" |
| 13:17.23 | robl^laptop | j/k. ;-) Hope the day goes well for you. |
| 13:18.02 | [TK]D-Fender | jaytee: To inifinity... and beyond! |
| 13:18.09 | jaytee | :-) |
| 13:18.54 | jaytee | I found one of those 3 eyed aliens on the ground one day. Some child must have lost it. If you squeeze it the eyes bug out. |
| 13:19.36 | jaytee | did ya know that they had a Buzz Lightyear on the ISS for about the last year? He came home on the last shuttle trip. |
| 13:19.47 | Katty | robl^laptop: no, it was in a black velvet ring case. |
| 13:20.00 | jaytee | "He went to Jared!" |
| 13:20.18 | Katty | who's Jared? |
| 13:20.31 | jaytee | a national jewelry store chain |
| 13:20.53 | Katty | oh. hmm. i don't think there's one around here. |
| 13:20.58 | jaytee | at least they have them here and on the west coast and the tv and radio commercials I've heard even back in New England. |
| 13:21.11 | eppigy | hello |
| 13:21.20 | jaytee | you are dave! |
| 13:21.51 | Pan3D | good morning astripeople |
| 13:22.18 | Katty | this particular ring is from JCPenny |
| 13:22.35 | Katty | it's not all /that/ fancy (= |
| 13:22.46 | *** join/#asterisk Chesther (n=cam2@cam2-mac.cit.cornell.edu) |
| 13:24.21 | DigitalFlux-AFK | Hey guys |
| 13:24.34 | DigitalFlux-AFK | i am trying to use the Asterisk Real Time arch with Mysql |
| 13:24.39 | Pan3D | you are not AFK |
| 13:24.42 | Pan3D | lies |
| 13:25.31 | DigitalFlux-AFK | i did the steps at http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 13:25.31 | DigitalFlux | :) |
| 13:25.35 | DigitalFlux | however when i add a user to the database |
| 13:25.35 | Pan3D | :) |
| 13:25.42 | DigitalFlux | asterisk still doesn't see it |
| 13:25.49 | DigitalFlux | connecting to it using the phone i get |
| 13:25.57 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 13:25.57 | DigitalFlux | No matching peer found |
| 13:26.05 | DigitalFlux | any hints ? |
| 13:26.28 | [TK]D-Fender | DigitalFlux: Show us something useful. |
| 13:26.59 | DigitalFlux | [TK]D-Fender: The database table structure ? |
| 13:27.13 | DigitalFlux | <PROTECTED> |
| 13:27.27 | DigitalFlux | i get this when i open the asterisk CLI |
| 13:27.36 | DigitalFlux | when means that it is up, now ? |
| 13:27.40 | DigitalFlux | no * ? |
| 13:27.47 | hat_panda | [TK]D-Fender: okay thx, maybe such mechanism does not exist |
| 13:27.56 | [TK]D-Fender | DigitalFlux: We don't see your tables, SIP DEBUG for the failed call, nothing... |
| 13:28.42 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:29.39 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 13:29.46 | ManxPower-work | ~answers |
| 13:29.47 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:30.15 | *** join/#asterisk tgunr (n=tgunr@polymicro.net) |
| 13:31.11 | jaytee | ~snide_remarks |
| 13:31.18 | jaytee | darn! |
| 13:31.41 | Katty | jaytee: http://www.whitefirejewelry.com/Heart-Promise-Ring-WF00812_medium.jpg |
| 13:31.45 | Pan3D | oooh, I like the additon of the docs part |
| 13:32.18 | jaytee | Katty, Shiny! |
| 13:32.35 | Katty | jaytee: http://www.mdcdiamonds.com/images/ProductImages/ES140HS.jpg <- that one's better. |
| 13:32.43 | Katty | jaytee: crystalized carbon ring. |
| 13:33.56 | Katty | http://applesofgold.com/Merchant2/jewelry/gemstone-jewelry/JGEMRG-8C.jpg <- that looks like the one ryan got me |
| 13:34.08 | Katty | except the gemstones go all the way around the band. not just the top |
| 13:34.29 | jaytee | some of the higher quality diamonds today come with a laser etched id number on them that can only be seen under a microscope |
| 13:35.34 | *** join/#asterisk benneton (n=benneton@adsl-29-211.teol.net) |
| 13:35.36 | *** part/#asterisk benneton (n=benneton@adsl-29-211.teol.net) |
| 13:36.00 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 13:36.24 | jaytee | alot of companies today are selling lab created diamonds or moissanite and calling it diamond |
| 13:36.58 | jaytee | caveat emptor, I always say.......well, not always but often |
| 13:36.59 | Katty | if i was going to get a diamond, i'd want a certified NonBlood diamond |
| 13:37.05 | jaytee | yep |
| 13:37.11 | Katty | but really, Moissanite or even CZ would be fine with me |
| 13:37.34 | Katty | i can't tell the difference anyway |
| 13:37.42 | jaytee | so your the kind of person with low self-esteem that would settle for less? |
| 13:38.03 | jaytee | Stand up, girl!!! Demand the best!!! |
| 13:38.06 | Katty | i just don't see spending 6k on an engagement ring |
| 13:38.12 | Katty | i'd rather have a plasma ion rocket |
| 13:38.14 | jaytee | true |
| 13:38.23 | jaytee | especially in this economy |
| 13:38.32 | Katty | oh and i figured out the gforce effect of it too |
| 13:38.48 | ManxPower-work | Don't get married. |
| 13:39.08 | Kobaz | i'm having a pretty bad t1 problem http://kobaz.net/misc/1255008818.16.WAV |
| 13:39.20 | jaytee | I hate seeing young couples who end up throwing a Great Gatsby or Kennedy style wedding for 20K and then can't come up with a down payment on a house. |
| 13:39.38 | jaytee | actually 20K is cheap nowadays |
| 13:39.42 | Kobaz | every so often, on a recurring basis, we'll get a complete cutout of audio... and then it will go back to normal |
| 13:39.50 | Katty | jaytee: i'm all for a Justice of the Peace wedding |
| 13:39.56 | Kobaz | and it's happening on every call |
| 13:40.06 | Katty | but, i'm gonna have me a dress. |
| 13:40.25 | n3hxs | That beats going without one Katty |
| 13:40.28 | *** join/#asterisk jamicque (n=jam@80.50.125.74) |
| 13:40.31 | Kobaz | there's no frame slips, no crc errors, nothing in the logs to indicate a problem |
| 13:40.31 | ManxPower-work | I got married for under $600 |
| 13:40.59 | Kobaz | i didn't get married for $0 |
| 13:41.08 | jamicque | Hi, any body uses Digium's fax for asterisk? I have a question ho to make Recieve fax app force to send reinvite with t38 |
| 13:41.35 | Kobaz | anyways... can someone who'se good with diagnosing tdm problems, check out my recording |
| 13:41.38 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:41.40 | *** join/#asterisk voipmonk (n=voipmonk@74-132-202-71.dhcp.insightbb.com) |
| 13:41.41 | ManxPower-work | jamicque: why don't you contact Digium. You paid for the product and apparently nobody here uses it. |
| 13:42.18 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 13:42.20 | Kobaz | basically it's nothing special... but there's a complete audio cutout very consistantly on every call for about 1/2 a second |
| 13:42.49 | ManxPower-work | Kobaz: I suspect latency caused by disk access |
| 13:43.12 | Kobaz | it's not just in the recording |
| 13:43.26 | Kobaz | like, i was sitting here, in the room... and one of the people on the phone said "it just dropped out" |
| 13:43.30 | Kobaz | so i checked out the recording |
| 13:43.32 | n3hxs | Kobaz, how full is the disk. |
| 13:43.39 | Kobaz | 75 gigs free |
| 13:43.44 | Kobaz | out of 80 |
| 13:43.46 | n3hxs | CPU load? |
| 13:43.50 | Kobaz | nodda |
| 13:43.54 | Kobaz | like 5-10% at the most |
| 13:44.01 | ManxPower-work | Kobaz: What is the interrupt latency when the problem happens? That is what I am referring to. |
| 13:44.07 | Kobaz | ah |
| 13:44.08 | Baylink2 | Did you roll back your asterisk, Kobaz? |
| 13:44.19 | Kobaz | Baylink2: yeah, and the problem got worse, they said |
| 13:44.23 | Baylink2 | Hmmm. |
| 13:44.24 | Kobaz | which makes no sense |
| 13:44.43 | Kobaz | ManxPower-work: how do you check inturrupt latency |
| 13:44.45 | n3hxs | Drive might have bad spots which would slow down access. |
| 13:44.58 | ManxPower-work | Many on-board controllers lock interrupts for incredible amount of time. Do a google of your motherboard model + audio. The same IRQ latency issues that affect Digium cards are the same issues that sometimes cause TDM problems |
| 13:45.10 | Naikrovek | could be just a couple dropped packets, doesn't have to be system relateed |
| 13:45.12 | ManxPower-work | Kobaz: heck if I know, I just wanted you to stop dismissing disk issues as a problem. |
| 13:45.14 | Naikrovek | related* |
| 13:45.14 | Katty | so there's one fatal flaw with using a plasma ion rocket to get us to mars. |
| 13:45.16 | Kobaz | this is a sangoma card |
| 13:45.22 | ManxPower-work | Naikrovek: it's TDM, there's no "packets" |
| 13:45.25 | Naikrovek | ah |
| 13:45.27 | Naikrovek | missed that |
| 13:45.30 | Katty | power requirements. |
| 13:45.32 | Kobaz | ManxPower-work: hmm |
| 13:46.01 | ManxPower-work | Kobaz: Are the endpoint phones IP or TDM? |
| 13:46.13 | Kobaz | tdm |
| 13:46.18 | Katty | the energy requirements to keep a ship holding people accelerated at 52k mph, is far more than a solar panel can provide. |
| 13:46.31 | ManxPower-work | so you have two tdm cards? one T-1 and one analog? |
| 13:46.36 | Kobaz | pstn <-t1-> asterisk <-t1-> pbx <-> digital phones |
| 13:46.46 | Baylink2 | Ok. I heard the discontinuity at :20, but on the *recording*, there was no matching gap; it just clicked. There was lost audio in that period, then? |
| 13:46.58 | Kobaz | Baylink2: yeah, loss of audio |
| 13:47.07 | Kobaz | there's a gap when i listen to the recording |
| 13:47.09 | ManxPower-work | span=1,1,whatever <-- telco side and span=2,0,whatever <-- pbx side? |
| 13:47.20 | Naikrovek | yeah i hear a small, maybe quarter second gap |
| 13:47.30 | Baylink2 | There's a gap, but it's only about 100ms, to me. |
| 13:47.37 | Naikrovek | yeah it's small |
| 13:47.38 | Kobaz | telco is cpe, recieving clock... pbx side is network, picking clock from the network side |
| 13:47.39 | Naikrovek | but still there |
| 13:47.50 | Kobaz | yeah, it's on every call, at some point |
| 13:47.50 | ManxPower-work | Kobaz: you have them reversed |
| 13:47.58 | Baylink2 | ManxPower-work: yeah, we checked his Sangoma and zaptel setups yesterday. |
| 13:48.00 | ManxPower-work | you should RECEIVING t-1 sync timing from the telco |
| 13:48.07 | ManxPower-work | and SENDING it to the PBX |
| 13:48.08 | Kobaz | ManxPower-work: yeah, that's what i said |
| 13:48.20 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:48.26 | *** join/#asterisk LGW (n=large@xdslho073.osnanet.de) |
| 13:48.27 | ManxPower-work | "telco is cpe, recieving clock." <-- exact opposite of what I said |
| 13:48.31 | Baylink2 | It's not quite what you said. :-) |
| 13:48.47 | LGW | hi there :) |
| 13:48.55 | Kobaz | the port going to the telco from asterisk, is provisioned as cpe, recieving clock |
| 13:49.01 | Baylink2 | But what you said yesterday was that the * box was slave/CPE on the PRI side, and master/NET on the PBX side. |
| 13:49.10 | Kobaz | Baylink2: exactly |
| 13:49.18 | LGW | I want to set up a "small" (from the Asterisk Book definition) Asterisk system for our office |
| 13:49.29 | Baylink2 | Anyway. What OS and kernel version, and did you compile the kernel yourself? |
| 13:49.37 | Kobaz | we're all saying the same thing, three different ways |
| 13:49.38 | Baylink2 | This does sound like a latency problem. |
| 13:49.58 | Baylink2 | LGW: good for you. :-) |
| 13:50.05 | ManxPower-work | Kobaz: PASTE just the two span lines |
| 13:50.06 | Kobaz | debian with 2.6.23.14 |
| 13:50.08 | [TK]D-Fender | LGW: You have our official permission |
| 13:50.11 | Kobaz | ManxPower-work: okay |
| 13:50.11 | LGW | we want use VoIP Clients (up to 10), and carrier connection will be via ISDN |
| 13:50.14 | LGW | thing is :) |
| 13:50.25 | Kobaz | span=1,0,0,esf,b8zs |
| 13:50.28 | Kobaz | span=2,0,0,esf,b8zs |
| 13:50.40 | LGW | will a 1GHz CPU (Athlon 1600+) be sufficient for this? |
| 13:50.50 | Baylink2 | LGW: PRI? |
| 13:51.03 | Kobaz | Baylink2: i rebooted the box, checked cables, etc |
| 13:51.07 | Naikrovek | LGW: probably enough, yes |
| 13:51.09 | [TK]D-Fender | LGW: Sure |
| 13:51.11 | Baylink2 | Kobaz: just those 2 spans? |
| 13:51.18 | LGW | Baylink2, what? |
| 13:51.19 | Kobaz | Baylink2: i did see in dmesg, not sure what time... a fatal error 0x10 from the sangoma card |
| 13:51.24 | Kobaz | Baylink2: yeah, just two |
| 13:51.25 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:51.29 | ManxPower-work | Kobaz: make span=1,1,0,esf,b8zs |
| 13:51.34 | Baylink2 | Then you're not mastering the PBX. |
| 13:51.36 | ManxPower-work | you are getting frame slips. |
| 13:51.38 | Baylink2 | What he said |
| 13:51.49 | Kobaz | ManxPower-work: it's a sangoma card, the clocking,etc is in the wanpipe config |
| 13:51.52 | Baylink2 | LGW: what kind of ISDN? |
| 13:52.02 | Baylink2 | Yeah, Kobaz, but Asterisk has to agree. |
| 13:52.05 | ManxPower-work | Kobaz: then it can't hurt to change it then. |
| 13:52.09 | Kobaz | there are no frame slips in the counters either |
| 13:52.12 | Kobaz | ManxPower-work: k, i'll try that |
| 13:52.14 | Baylink2 | Nice one. :-) |
| 13:52.20 | Kobaz | i've never had any problems with having it 0,0,0 |
| 13:52.30 | Baylink2 | You're having problems. :-) |
| 13:52.32 | Kobaz | we have a box that's been going for two years configured the same way |
| 13:52.36 | Kobaz | well, yeah, on this box |
| 13:52.37 | Kobaz | heh |
| 13:53.25 | LGW | Baylink2, standard german ISDN BRI |
| 13:53.30 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 13:53.32 | Baylink2 | LGW: have you ever done any Asterisk before? And are you doing this for production or practice? |
| 13:53.37 | Baylink2 | LGW: Ah. |
| 13:53.42 | Kobaz | port 1: Bit Errors (CRC6/Ft/Fs) : 0 |
| 13:53.45 | Kobaz | port 2: Bit Errors (CRC6/Ft/Fs) : 0 |
| 13:53.55 | Baylink2 | Yeah, yeah, we know. Change it anyway. :-) |
| 13:53.55 | LGW | Baylink2, well, I set up a Asterisk box about... 6 years ago |
| 13:54.01 | Kobaz | yeah, waiting for people to get off the phone |
| 13:54.27 | LGW | Baylink2, for VoIP routing. I just HOPE that documentation has improved a bit ;) |
| 13:54.35 | Kobaz | LGW: a little |
| 13:54.36 | LGW | but it looks very promising. |
| 13:54.39 | Baylink2 | LGW: Ok. Opinions will differ, but I have my nomex longjohns on (I kept them from my Usenet days). I'd try installing PBXinaFlash 1.4, first, if I were you. |
| 13:55.15 | LGW | Baylink2, thanks for the advice |
| 13:55.19 | Kobaz | people need to make less calls damnit |
| 13:55.21 | LGW | looks interesting |
| 13:55.24 | Baylink2 | I'm running it on an office system, in production, with 11 phones and a PRI, and it's been just fine to work with; very tractable. |
| 13:56.12 | Baylink2 | If you want to do weird stuff you will need to suss it a bit; if you want to do *really* weird stuff, you will probably have to work around it. But if you just want to install a PBX, Bob's yer uncle. |
| 13:56.40 | Baylink2 | Now: I'm running it on a Core2Quad Supermicro with 4GB of RAM... :-) |
| 13:57.04 | Baylink2 | Kobaz: yes, Asterisk's lack of implementation of the IBM 'VARY OFF' command regularly annoys me. |
| 13:57.20 | Baylink2 | (Apache would call it "restart graceful") |
| 13:57.29 | leifmadsen | Baylink2: you can make that yourself with cli_aliases |
| 13:57.39 | Baylink2 | Not on everything... |
| 13:57.43 | LGW | currently, we use a traditional key telephone system, wich will be kept for "production"? |
| 13:57.57 | Baylink2 | Sometimes I want to vary off *one specific thing*, but not the whole box. |
| 13:58.14 | leifmadsen | Baylink2: you should still be able to make multiple aliases |
| 13:58.16 | ManxPower-work | LGW: asterisk is a PBX, not a key system. They work very differently |
| 13:58.23 | Baylink2 | Well, my point was merely that experimentation and production don't always mix, unless you're really good, LGW. |
| 13:58.46 | Baylink2 | leifmadsen; let me remember a specific example; I'll get back to you. |
| 13:59.42 | LGW | ManxPower-work, maybe I did get mixed up with technical terms here ;) |
| 14:00.00 | DigitalFlux | what entries should i input in the sip mysql table in order to create a new sip user ? |
| 14:00.06 | *** join/#asterisk gain_ (n=gain@dynamic-adsl-94-36-232-231.clienti.tiscali.it) |
| 14:00.20 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:00.20 | LGW | I just know that Asterisk will be able to route ISDN calls to our new VoIP phones - so it does what I need ;) |
| 14:00.27 | gain_ | hi all |
| 14:01.25 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.15) |
| 14:01.44 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:01.44 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:02.21 | Baylink2 | Morning, gain_ (he, said, throwing himself manfully on the grenade) |
| 14:02.22 | LGW | thing is, our current production hardware will be kept for production, so there's little risk breaking everything. |
| 14:02.35 | Baylink2 | Cool |
| 14:03.09 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 14:03.12 | LGW | We could either buy a new auerswald TK for about 1500¤, or try Asterisk. And as it worked back in 2002, I'm pretty sure I can make it work again :) |
| 14:03.43 | Kobaz | http://pastebin.ca/1604202 there's my pri debug |
| 14:04.05 | Kobaz | that's the call that i pasted with the dropout |
| 14:04.15 | gain_ | hi Baylink2 |
| 14:05.34 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 14:05.51 | LGW | OK, thanks. I'll try that old 1GHz machine, then :) |
| 14:06.32 | LGW | If it gets to slow when more and more people switch from the current to the new system, I can upgrade the hardware anyway. |
| 14:06.37 | MaliutaLap | [da-house]\n exten => MaliutaLap |
| 14:07.07 | LGW | I just wanted to make sure that it's not totally stupid to try it with the old server in the first place. Time is money, you know *eg* |
| 14:07.19 | LGW | but it should be great for training, so... |
| 14:07.47 | Kobaz | i'm seeing about 1000 inturupts a second on the sangoma card |
| 14:08.17 | ManxPower-work | that's what you should see |
| 14:08.21 | Kobaz | k |
| 14:08.44 | LGW | thanks for your suggestions. l8er... |
| 14:09.23 | ManxPower-work | Kobaz: you didn't do something stupid like set busydetect=yes or callprogress=yes, did you? |
| 14:09.37 | Kobaz | nope |
| 14:11.03 | Kobaz | http://pastebin.ca/1604217 |
| 14:11.19 | Kobaz | probably could get rid of some options, but that's what was generated by wancfg |
| 14:11.39 | Baylink2 | LGW: I will note: Part of the price you pay for using PIAF/FPBX is that it's more code, to insulate you from the bare metal; that does take some CPU. Make sure you have enough RAM to avoid swapping. 1G should be ok; 2G be great if you can pull it off. |
| 14:12.02 | ManxPower-work | GUIs are just so hard to use. |
| 14:12.26 | *** join/#asterisk coppice (n=chatzill@148.162.17.210.dyn.pacific.net.hk) |
| 14:12.41 | Baylink2 | Kobaz: I'd be surprised if a PRI debug showed anything about this problem, honestly. |
| 14:12.47 | Kobaz | yeah |
| 14:12.48 | Kobaz | me too |
| 14:12.49 | Baylink2 | That's mostly call setup |
| 14:12.52 | Kobaz | yeap |
| 14:13.01 | Baylink2 | MaxPower-work: Depends on your skillset. |
| 14:13.08 | Kobaz | but i don't know enough about q931 to know if i saw a problem or not |
| 14:13.14 | Baylink2 | What I like about FPBX is it's not WYSIAYG; you can go behind its back. |
| 14:13.45 | Baylink2 | Kobaz: note that you can con the Sangoma drivers into dumping the q931 to a file, and that Wireshark can read it. |
| 14:14.01 | ManxPower-work | Kobaz: AUDIO is not Q.931 |
| 14:14.29 | Kobaz | ManxPower-work: yeah.. audio is gonna be ulaw |
| 14:14.38 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:14.49 | *** join/#asterisk nephilinturbo (n=victorsa@189.28.1.2) |
| 14:15.42 | Kobaz | ManxPower-work: with the dahdi config changed to 1,1,0 ... still getting cutouts |
| 14:15.55 | Kobaz | ManxPower-work: we had the provider check their logs/counters/etc yesterday |
| 14:16.04 | Baylink2 | And you posted your wanpipe configs yesterday, and they looked ok. |
| 14:16.09 | Kobaz | they monitored for about 6 hours, and told us they found nothing, everything was clean |
| 14:16.17 | Baylink2 | I didn't hear whether you built your own kernel? |
| 14:16.23 | Kobaz | oh... yeah. i did |
| 14:16.26 | Kobaz | i always do |
| 14:16.35 | Kobaz | running the same kernel on bunches of other boxen without issue |
| 14:16.48 | Baylink2 | Hmmm. Same motherboards? |
| 14:17.01 | Kobaz | this box no |
| 14:17.04 | Kobaz | this is the odd one out |
| 14:17.05 | Kobaz | heh |
| 14:17.17 | Baylink2 | Well there you go then. :-) |
| 14:17.19 | Baylink2 | Multicore? |
| 14:17.20 | Kobaz | what's weird... is that this just started happening recently |
| 14:17.22 | Kobaz | yeah |
| 14:17.32 | Baylink2 | # watch -d 'cat /proc/interrupts' |
| 14:17.37 | Kobaz | yeah |
| 14:17.58 | Kobaz | everything is going to cpu0 pretty much |
| 14:17.58 | *** join/#asterisk ast-thecode (n=asterisk@scalise.csata.tno.it) |
| 14:18.03 | Baylink2 | Changing your CPU affinity could help |
| 14:18.10 | ast-thecode | anyone of you knows (even if commercial) T38 implmentation for asterisk? |
| 14:18.27 | Kobaz | actually |
| 14:18.29 | Kobaz | you know |
| 14:18.41 | Kobaz | all the inturupts are hitting cpu1 now |
| 14:18.45 | Baylink2 | Yes, yes I do. |
| 14:18.46 | Baylink2 | :-) |
| 14:18.46 | ManxPower-work | ast-thecode: 1.4 has very limited T.38 support, 1.6 is better from what I've heard |
| 14:19.10 | Kobaz | maybe every time the process switches cpus... it drops out |
| 14:19.18 | leifmadsen | ast-thecode: there is a T.38 server patch in the issues.asterisk.org site |
| 14:19.27 | ManxPower-work | ast-thecode: most T.38 products seem to not be compatible with each other. That's one of the reasons so few people use it. |
| 14:19.27 | leifmadsen | passthrough t.38 should work fine |
| 14:20.06 | ManxPower-work | I work for a CLEC. Guess how they handle fax for clients? |
| 14:20.06 | Baylink2 | Lee Howard on the Hylafax list is pretty up on t38, IIRC. |
| 14:20.07 | Kobaz | oh wait |
| 14:20.11 | coppice | ManxPower-work: there is a SIP Forum working group on that topic |
| 14:20.15 | Kobaz | Baylink2: i was looking at a different box |
| 14:20.25 | Baylink2 | Good job you didn't type "reboot" |
| 14:20.27 | Kobaz | it's still on cpu0 |
| 14:20.29 | Kobaz | hah |
| 14:20.34 | Kobaz | i have a reboot helper |
| 14:20.50 | Baylink2 | Do you know what to poke to spread the CPU affinity? |
| 14:20.59 | ManxPower-work | We install POTS lines for fax and then NEVER HAVE TO WORRY ABOUT IT AGAIN. |
| 14:20.59 | Kobaz | # reboot |
| 14:21.00 | Kobaz | note: this is foobar-1, do you really want to reboot |
| 14:21.26 | Kobaz | yeah it's taskset |
| 14:23.34 | Baylink2 | Hee. To both of those messages. :-) |
| 14:25.38 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-aiasoizttoqxjxkc) |
| 14:25.54 | nephilinturbo | hi guys !! |
| 14:26.09 | nephilinturbo | Someone to help me about transfers ? |
| 14:26.25 | tzafrir_laptop | transfers nephilinturbo to ##linux |
| 14:26.35 | ManxPower-work | nephilinturbo: To transfer you press the Transfer button on your phone. |
| 14:26.55 | Kobaz | hmmmmmmmmmmmmmmmm |
| 14:27.09 | Kobaz | i'm listening to another recording, that was complained about |
| 14:27.19 | nephilinturbo | Guys .. Im configuring a Asterisk , and i m trying to configure transfers |
| 14:27.32 | tzafrir_laptop | What phones? |
| 14:27.37 | Kobaz | and i don't see any dropouts in the time the person said they heard one |
| 14:27.39 | ManxPower-work | nephilinturbo: there are at least 4 different transfers in Asterisk. |
| 14:27.42 | nephilinturbo | in files features.conf , sip.conf etc |
| 14:27.44 | Kobaz | i'm thinking it's the other pbx |
| 14:27.59 | nephilinturbo | I m following many tutorials but never run |
| 14:28.00 | Kobaz | tzafrir_laptop: me? siemens |
| 14:28.02 | ManxPower-work | nephilinturbo: features.conf is for "DTMF transfers" |
| 14:28.07 | Baylink2 | Have you confirmed the *PBX* isn't trying to clock-master the line? |
| 14:28.16 | ManxPower-work | why not use the built in transfer features on your phone? |
| 14:28.26 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:28.26 | nephilinturbo | Sip phones - SIEMENS EUROSET 3005 |
| 14:28.30 | Kobaz | Baylink2: yeah... according to the configs it isn't |
| 14:28.48 | Baylink2 | Do you know if it was rebooted to load said configs? :-) |
| 14:28.57 | *** part/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 14:28.59 | Kobaz | oh yeah, rebooted it about 10 minutes ago |
| 14:29.38 | nephilinturbo | Baylink2: I want to use transfer in Asterisk cause i have here in my job users that use SoftPhone for example |
| 14:29.46 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:29.46 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:31.49 | *** join/#asterisk mazpe (n=mazpe@ec2-75-101-211-169.compute-1.amazonaws.com) |
| 14:33.04 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 14:34.01 | nephilinturbo | Someone knows how i configure my asterisk to make transfers ? in features.conf ? |
| 14:34.24 | nephilinturbo | look my confs : |
| 14:34.26 | nephilinturbo | [featuremap] |
| 14:34.27 | nephilinturbo | ;blindxfer => #2 |
| 14:34.27 | nephilinturbo | atxfer => #2 |
| 14:34.27 | nephilinturbo | automon=*1 |
| 14:34.27 | nephilinturbo | disconnect=** |
| 14:34.31 | Naikrovek | ack |
| 14:34.33 | Naikrovek | ~pb |
| 14:34.34 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 14:34.48 | Naikrovek | nephilinturbo: ^^^^^ |
| 14:34.59 | nephilinturbo | sorry |
| 14:35.12 | Naikrovek | it's cool, just don't do it twice |
| 14:35.58 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 14:36.19 | nephilinturbo | When i insert #2 for exaample in a call . Nothing happens |
| 14:36.41 | *** join/#asterisk moy (n=moy@74.12.134.3) |
| 14:36.59 | Naikrovek | nephilinturbo: hold tight, be patient. someone will respond soon |
| 14:37.15 | mort_gib | nephilinturbo: Your on the wrong track.... Your phones will dictate how transfers are done |
| 14:38.21 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 14:38.37 | IBC_jkenney | has anyone here used queue metrics |
| 14:38.47 | nephilinturbo | mort_gib: Dont understand. I know that i can make transfers using button flash for example in my phones |
| 14:38.50 | IBC_jkenney | <===== is new to tomcat |
| 14:38.52 | ZPertee | trying to install app_swift on centos5. Following error when trying to load via CLI. Error loading module 'app_swift.so': libswift.so.4: cannot open shared object file: No such file or directory |
| 14:39.03 | ZPertee | any ideas? |
| 14:39.34 | nephilinturbo | mort_gib: but i have many users that have only softphones .. So i need make it in asterisk |
| 14:39.36 | mort_gib | nephilinturbo: There you go, you rely on your phone for transferring |
| 14:40.04 | mort_gib | Softphones have their own way of transferring (I'm sure) |
| 14:40.45 | nephilinturbo | mort_gib: I know But if Asterisk Manual tells me that its possible, so i want to try |
| 14:41.00 | [TK]D-Fender | [10:37]<mort_gib>nephilinturbo: Your on the wrong track.... Your phones will dictate how transfers are done <- not necessarily |
| 14:41.07 | *** join/#asterisk voipmonk (n=voipmonk@74-132-202-71.dhcp.insightbb.com) |
| 14:41.09 | *** join/#asterisk icyValk77 (n=icyValk7@213.129.64.4) |
| 14:41.14 | [TK]D-Fender | nephilinturbo: Show us the CALL ATTEMPT |
| 14:41.54 | *** join/#asterisk friartuck (n=pmccary@66.162.90.56) |
| 14:43.43 | Kobaz | details schmetails |
| 14:45.49 | mort_gib | [TK]D-Fender: You are of course right, but in reality some phones want it their way! |
| 14:45.54 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:46.44 | [TK]D-Fender | mort_gib: And his circumstances call for it for well known reasons... |
| 14:47.52 | mort_gib | I suppose so... I never even tried a Softphone... -Are there any good ones out there?? |
| 14:48.11 | [TK]D-Fender | mort_gib: Depends on your definition of "good" |
| 14:48.36 | mort_gib | :-) Client are not calling me all the time on their mobiles! |
| 14:49.04 | mort_gib | because the %favorite-swearword% softphone is not working |
| 14:49.54 | [TK]D-Fender | mort_gib: They pretty much all "work". that isn't the issue |
| 14:50.15 | mort_gib | Then what is?? |
| 14:50.32 | *** join/#asterisk galeras (n=galeras@190.27.246.226) |
| 14:51.58 | [TK]D-Fender | mort_gib: His cheap shit one don't have a transfer feature. X-Lite for example holds this and other features back for the PAID version instead of the FREE one |
| 14:52.40 | MaliutaLap | should be able to get around that by parking lot stuff in the dialplan though |
| 14:52.41 | mort_gib | [TK]D-Fender: Well, I might have some kind of understanding for their attitude... |
| 14:52.43 | *** join/#asterisk korcan (n=korcan@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 14:52.57 | Kobaz | ManxPower-work: i've switched some people to ip phones off the asterisk box (instead of digital phones on the other pbx)... i'll try and see if it's the other pbx that's causing the problems |
| 14:53.22 | Kobaz | so that leaves only one t1 in the picture... to the pstn |
| 14:53.44 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:53.52 | [TK]D-Fender | mort_gib: I didn't say all do this. Other work fine for what they are, but softphones are hated because of having your head tied to your PC and having to use an attention-stealing interface to manipulate/place calls. |
| 14:54.20 | mort_gib | [TK]D-Fender: Besides I haven't gone down that road because I see too many crappy M$ machines that are not running very well |
| 14:54.24 | Kobaz | and pc microphones and speakers generally suck for making calls |
| 14:54.29 | Kobaz | unless you're on a nice headset |
| 14:55.00 | mort_gib | AV starts a scan and you can't call ANYONE |
| 14:55.05 | Kobaz | hah |
| 14:55.27 | mort_gib | cracke-blib-blob-click-click |
| 14:56.34 | nephilinturbo | [TK]D-Fender: Sorry man ..My boss calls me ... How i can show you the CALL ATTEMPT ? |
| 14:56.43 | [TK]D-Fender | nephilinturbo: * CLI of course. |
| 14:58.21 | nephilinturbo | [TK]D-Fender: Dont understand .. Im making calls right. Talking. But i need transfer calls with a config in Asterisk |
| 14:58.41 | [TK]D-Fender | nephilinturbo: pastebin the CALL |
| 14:59.05 | nephilinturbo | like the line atxfer => #2 .. But when i input #2 on the phone during a call, nothing hapens . Nothing |
| 14:59.14 | *** part/#asterisk smtx (n=smtx@p50998557.dip0.t-ipconnect.de) |
| 15:00.22 | *** join/#asterisk blkry (n=chatzill@96.37.27.72) |
| 15:00.53 | [TK]D-Fender | nephilinturbo: pastebin the CALL <-------- |
| 15:03.04 | nephilinturbo | [TK]D-Fender: Here : http://pastebin.com/m4851e1a0 |
| 15:03.25 | *** join/#asterisk tiav (n=tiav@fw.sj.tdf-pmm.net) |
| 15:03.39 | [TK]D-Fender | nephilinturbo: "core show application dial" <- now read the instructions |
| 15:04.16 | [TK]D-Fender | mort_gib: And yes i was already certain this is what the issue was 20 minutes ago |
| 15:05.04 | mort_gib | [TK]D-Fender: -I know "[TK]D-Fender Rules" :-) |
| 15:06.07 | tzafrir_laptop | the rules of the channel :-) |
| 15:06.15 | nephilinturbo | [TK]D-Fender: sorry man ..But what can i find it ? |
| 15:06.29 | [TK]D-Fender | nephilinturbo: I jsut gave you the CLI command to get the instructions. |
| 15:06.58 | *** join/#asterisk SBD (n=sbd@96-24-106-20.stc.clearwire-dns.net) |
| 15:07.29 | nephilinturbo | [TK]D-Fender: ty man ..will read now |
| 15:07.41 | SBD | Anyone use VoipInnovations as a sip provider? |
| 15:08.20 | [TK]D-Fender | SBD: may as well be "No Name" around here |
| 15:08.56 | nephilinturbo | [TK]D-Fender: I also make all these instructions man |
| 15:09.12 | SBD | Oh? |
| 15:11.46 | *** join/#asterisk anonymouz666 (n=anonymou@187-28-37-118.poolip.RJO.embratel.net.br) |
| 15:16.20 | Katty | I FEEL SO UNSURE |
| 15:16.34 | Katty | AS I TAKE YOUR HAND AND LEAD YOU TO... THE... DANCE FLOOR |
| 15:17.16 | *** join/#asterisk Da-Geek (n=Da-Geek@62.189.17.99) |
| 15:18.23 | voipmonk | then the dance floor caves in... |
| 15:18.25 | Nugget | dances with Katty |
| 15:18.29 | voipmonk | now im never dancing again |
| 15:21.10 | ast-thecode | <[TK]D-Fender> : Hi, finally I did what I was searching for.... with dialplan ad agi perl ... :) |
| 15:21.14 | Baylink2 | Ah yes, GM. One of my favorite karaoke songs. |
| 15:21.16 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 15:21.26 | Baylink2 | No one's ever heard it, so it doesn't matter if you screw it up. |
| 15:21.29 | Kobaz | we've taken the other pbx out of the possabilities of the source of problems.... ip phone on asterisk box... to t1 out to pstn... and i have another recording... www.kobaz.net/misc/2.wav |
| 15:21.33 | [TK]D-Fender | ast-thecode: And never needed AGI for any of it |
| 15:22.00 | Kobaz | and this isn't just a dropout.. these some weird pre and post stuff going on |
| 15:22.11 | ast-thecode | <[TK]D-Fender>: maybe you didn't understand well what I want to do.... |
| 15:22.14 | Kobaz | if you look at the audio in a sound editor it's a weird looking wave |
| 15:22.32 | ast-thecode | <PROTECTED> |
| 15:22.33 | [TK]D-Fender | astYes you wanted ti to do a "call me back when they can answer" |
| 15:22.39 | ast-thecode | not-help |
| 15:23.07 | superbeef | If i see this Got a FRAME_CONTROL (-1) frame on channel IAX2/ and the call dropped... that would indicate that? |
| 15:23.23 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 15:23.25 | ast-thecode | <[TK]D-Fender> : yes, choosing of doing it by the pressing of a number |
| 15:23.47 | Kobaz | Baylink2: so it's either. the computer itself, the t1 card, or the provider |
| 15:23.55 | Baylink2 | Kobaz: that *really* sounds like unsynchronized clocking to me. |
| 15:23.58 | [TK]D-Fender | ast-thecode: Yes, and it did not require any AGI or AMI to do. |
| 15:24.13 | Kobaz | Baylink2: okay |
| 15:24.27 | Baylink2 | Lemme put it in my editor for a second |
| 15:29.10 | Baylink2 | Kobaz: Ok; that's much what I expected to see. |
| 15:29.37 | Kobaz | wouldn;t you get frame slips from something like that |
| 15:29.40 | ast-thecode | <[TK]D-Fender>: YOU don't need them to do this... I need AGI to do it .... :) |
| 15:30.03 | [TK]D-Fender | ast-thecode: If don't need it, then you don't need ti. What you chose to do.. that is another matter. |
| 15:30.13 | [TK]D-Fender | I* |
| 15:30.51 | ast-thecode | <[TK]D-Fender>: okay okay... |
| 15:30.55 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 15:30.57 | anonymouz666 | [TK]D-Fender: do you use func_volume stuff? |
| 15:31.13 | [TK]D-Fender | anonymouz666: Nope... I tend to use sources that normalize properly :) |
| 15:31.42 | *** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) |
| 15:32.23 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 15:32.43 | ast-thecode | Anyone of you can explain why my BLF phones work with incoming calls but not with outgoing calls? |
| 15:32.44 | Kobaz | Baylink2: so if it's clocking... how would i debug it |
| 15:33.14 | ZPertee | can anyone help simplify the process of getting app_swift working in asterisk 1.4. keep getting error: Error loading module 'app_swift.so': libswift.so.4: cannot open shared object file: No such file or directory. I can see that many others have it as well but can't seem to find solution |
| 15:33.15 | Baylink2 | Hmmm. Now you take me into deep water. :-) You don't have a PRI analyzer, I assume? |
| 15:33.21 | Kobaz | hah |
| 15:33.22 | Kobaz | no |
| 15:33.32 | Kobaz | i'll just run to the corner store and get one... |
| 15:33.43 | Baylink2 | I do, but it has a battery charging problem. (When you have 45 PRIs, it's worth it) |
| 15:34.14 | Baylink2 | Have you reinstalled the very latest Sangoma drivers and re run the config script from scratch? |
| 15:34.18 | anonymouz666 | I am about to see a person who uses the on the fly volume setting offered by func_volume, can't put that to work on the latest 1.4 |
| 15:34.20 | Baylink2 | Cause that's probably nexy. |
| 15:34.39 | anonymouz666 | just the dialplan function works |
| 15:34.59 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 15:35.13 | Kobaz | Baylink2: i don;t have the latest latest, bit i've got 3.5.4 in there |
| 15:35.38 | Baylink2 | HWEC on that card? |
| 15:37.14 | *** join/#asterisk korcan (n=korcan@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 15:38.22 | *** join/#asterisk gain_ (n=gain@dynamic-adsl-94-36-249-15.clienti.tiscali.it) |
| 15:41.12 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 15:42.22 | *** join/#asterisk cyberfab007 (n=cyberfab@CPE001b11cf4f69-CM0014f85c3ada.cpe.net.cable.rogers.com) |
| 15:42.24 | Kobaz | Baylink2: yeah |
| 15:42.52 | Baylink2 | Is it enabled? |
| 15:43.05 | Baylink2 | Check your dmesg logs and make sure it's actually coming on. |
| 15:43.37 | Kobaz | dahdi_echocan_mg2: Registered echo canceler 'MG2' |
| 15:44.24 | Baylink2 | Registered sure, but turned on? |
| 15:45.08 | Kobaz | looking |
| 15:45.45 | Baylink2 | I'm thinking that you may be falling back to SWEC, and bumping your driver load |
| 15:47.06 | Kobaz | what's the differece between hardhdlc and dchan |
| 15:47.28 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 15:47.30 | Kobaz | some of the older drivers auto configured dahdi.conf to use hardhdlc... and the new drivers use dchan |
| 15:52.52 | Katty | dances with Nugget |
| 15:53.04 | [TK]D-Fender | ... |
| 15:53.06 | Kobaz | any idea? |
| 15:53.07 | [TK]D-Fender | telnet |
| 15:53.07 | Nugget | telnet is eeeeeeevil! |
| 15:53.08 | Kobaz | heh |
| 15:53.14 | [TK]D-Fender | makes Nugget dance... |
| 15:57.32 | Nugget | heh |
| 15:57.40 | *** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net) |
| 15:58.33 | Naikrovek | howcome they don't make beds shaped like race cars, except for adults |
| 15:58.42 | Naikrovek | i know a lot of people who would buy that |
| 15:59.08 | Nugget | I just sleep in my race car. |
| 16:00.19 | Naikrovek | king size bunk bed race cars |
| 16:00.25 | Naikrovek | hell, king size bunk beds alone |
| 16:01.05 | Nugget | http://macnugget.org/photos/2007gt3rs/DSC_2816 |
| 16:01.11 | Nugget | the seats recline. what more do you need? |
| 16:01.15 | Katty | ^_- |
| 16:01.23 | Nugget | hush katty ; |
| 16:01.25 | Katty | k |
| 16:04.10 | Naikrovek | telnet isn't evil, just insecure. and with ssh-agent, ssh is easier to use once set up |
| 16:04.30 | Naikrovek | loves ssh-agent |
| 16:04.35 | Naikrovek | pageant |
| 16:04.45 | n3hxs | A halloween car! |
| 16:05.14 | n3hxs | goes back to sleep. |
| 16:05.58 | Nugget | very festive |
| 16:06.27 | Nugget | and ssh-agent rocks :) |
| 16:08.30 | Katty | you rock. |
| 16:11.57 | tzafrir_laptop | Kobaz, the D channel traffic is encoded as HDLC |
| 16:12.00 | tzafrir_laptop | ~hdlc |
| 16:12.00 | infobot | extra, extra, read all about it, hdlc is High-level Data Link Control |
| 16:13.17 | *** part/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 16:13.49 | tzafrir_laptop | Zaptel/DAHDI knows how to decode it. 'dchan' (a.k.a fcshdlc) means DAHDI decodes the HDLC on the D-channel |
| 16:15.05 | tzafrir_laptop | In many cases the hardware can decode HDLC. hardhdlc asks the driver to use such hardware HDLC decoding |
| 16:15.05 | tzafrir_laptop | (decodic/encoding) |
| 16:15.24 | Naikrovek | i kinda wish i had some scabs to pick on. |
| 16:15.27 | Naikrovek | wonder if i'm ocd |
| 16:15.31 | xrmx__ | does anybody know if with a recent bootrom / sip application polycom ip330 have g722 support? |
| 16:15.44 | Naikrovek | xrmx__: depends on the phone |
| 16:15.52 | Naikrovek | xrmx__: for the 320/321/330/331, no |
| 16:16.19 | Naikrovek | polycom rebranded G722 as "HDVoice" and is only available on higher-end phones |
| 16:16.58 | xrmx__ | Naikrovek, ok, so it's just stupid marketing choices :) |
| 16:17.24 | Naikrovek | xrmx__: well G711 gives all the quality i'd ever need, but i do understand that other people may have different views than my own |
| 16:17.40 | xrmx__ | Naikrovek, g722 with polycom ip650 is spectacular |
| 16:18.16 | Naikrovek | yeah, if I ever get nicer phones I'll be able to experience that but not now |
| 16:18.28 | Naikrovek | Latest Polycom SIP firmware does add iLBC though |
| 16:18.33 | Naikrovek | ~ilbc |
| 16:18.34 | infobot | i guess ilbc is at http://www.ilbcfreeware.org |
| 16:18.40 | Naikrovek | pfft. |
| 16:18.57 | Naikrovek | correct answer I guess, but you could be more descriptive |
| 16:19.35 | xrmx__ | Naikrovek, thanks for the answer |
| 16:19.47 | Naikrovek | xrmx__: no problem |
| 16:23.41 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 16:29.11 | *** join/#asterisk jeremyb (i=d8fe4bf3@gateway/web/freenode/x-wkdrwbrxbrtdzxlb) |
| 16:32.04 | p3nguin | haha http://www.dailyherald.com/story/?id=327123&src= |
| 16:32.11 | p3nguin | oops, wrong window |
| 16:36.05 | nephilinturbo | Guys ... Im trying to config a Asterisk, to make transfers by pressing a DTMF buttons on phone, configured in features.conf . But i followed many instructiond and doesnt run ..Any idea ? |
| 16:38.06 | Chainsaw | nephilinturbo: Please get us the relevant part of the dial plan on pastebin. |
| 16:38.07 | Chainsaw | ~pb |
| 16:38.07 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 16:38.18 | Chainsaw | nephilinturbo: Also, what version of Asterisk is it? |
| 16:39.28 | p3nguin | You're simply wanting to use parking and do so with DTMF? |
| 16:39.44 | p3nguin | You probably didn't includd the parking context in your dialplan. |
| 16:39.51 | nephilinturbo | Chainsaw: Asterisk 1.4.18.1 |
| 16:39.57 | p3nguin | Once you show us, maybe we can figure that out. |
| 16:39.59 | *** join/#asterisk rps2 (n=rick@adsl-99-74-144-118.dsl.lsan03.sbcglobal.net) |
| 16:40.16 | nephilinturbo | p3nguin: example |
| 16:40.26 | *** join/#asterisk dysinger (n=tim@cpe-75-85-135-191.hawaii.res.rr.com) |
| 16:40.44 | p3nguin | Are you asking me how to include it? |
| 16:41.09 | *** part/#asterisk dysinger (n=tim@cpe-75-85-135-191.hawaii.res.rr.com) |
| 16:41.23 | nephilinturbo | p3nguin: Look the line in my extension.conf - exten => 3001,1,Dial(SIP/${EXTEN},20,tT) |
| 16:41.23 | nephilinturbo | p3nguin: are you talking about this ? |
| 16:41.29 | p3nguin | In the context for your phones, use "include => parkedcalls", assuming your parked calls context is "parkedcalls" |
| 16:41.47 | p3nguin | The tT will just allow both sides of the call to do transfers. |
| 16:41.53 | rps2 | (raises hand) I'm back. |
| 16:42.00 | p3nguin | Thanks for letting us know. |
| 16:42.19 | rps2 | Different question this time. |
| 16:42.50 | nephilinturbo | p3nguin: Also have this line in my extension.conf - include => parkedcalls |
| 16:43.06 | [TK]D-Fender | nephilinturbo: PASTEBIN <------- |
| 16:43.16 | p3nguin | nephilinturbo: It must be in a context that your phone is using. Some random location might not work. |
| 16:44.24 | rps2 | I'm getting no audio now, e.g. if I call into a conference room, the normal announcement isn't sent out. |
| 16:44.25 | p3nguin | Well, my SPA3102 arrived. I hope it's not too difficult to get it going. |
| 16:44.44 | nephilinturbo | p3nguin: Can you look one of my conf files, like features.conf to try see what im doing wrong please ? |
| 16:45.04 | p3nguin | nephilinturbo: Three of us are waiting on you to put your files in pastebin. |
| 16:45.17 | p3nguin | Until you do that, you're not going to get complete answers. |
| 16:45.21 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 16:45.57 | nephilinturbo | my features.conf - http://pastebin.com/m62396ff2 |
| 16:46.57 | nephilinturbo | my extensions.conf - http://pastebin.com/m4acedbb2 |
| 16:47.07 | p3nguin | I do have one concern about the SPA3102. If the power goes out, does the PSTN still work through the device? |
| 16:47.27 | ariel_ | yes |
| 16:47.34 | nephilinturbo | my sip.conf - http://pastebin.com/m70d3133f |
| 16:49.41 | nephilinturbo | p3nguin: this line - atxfer => #2 must abble the phones do make transfers by pressing #2 during a call. Right ? |
| 16:50.47 | p3nguin | nephilinturbo: Heck, I don't know. I have those lines commented out in my features.conf, and I just use the # key to start a transfer. If I want to park a call, I dial # 700. |
| 16:51.14 | [TK]D-Fender | nephilinturbo: where is the CALL? |
| 16:51.44 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 16:52.29 | nephilinturbo | [TK]D-Fender: I pastebin to you ..That time .. i used asterisk -r and maked a call between two extensions here and press #2 but nothing happens .. |
| 16:52.46 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 16:52.54 | nephilinturbo | [TK]D-Fender: Sould i pastebin another thing ? Please help me .. |
| 16:53.01 | [TK]D-Fender | nephilinturbo: No you just showed us 3 config files. |
| 16:53.13 | [TK]D-Fender | nephilinturbo: Where is the CLI output of this NEW failed call & transfer attempt? |
| 16:53.23 | nephilinturbo | p3nguin: I know . But i think transfer its not park . Are different things right ? |
| 16:53.37 | p3nguin | You use transfer to park calls. |
| 16:53.44 | p3nguin | You transfer them into the parking lot. |
| 16:54.06 | [TK]D-Fender | nephilinturbo: You Transfer to the parking extension specified in features.conf. And by that... you should be doing an ATTENDED transfer, not a BLIND transfer |
| 16:54.11 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 16:54.18 | p3nguin | So are you having troubles with parking calls, or with transferring everything/anything? |
| 16:54.48 | nephilinturbo | [TK]D-Fender: I sent to you 2 hours ago. But i ll do this now . But cant see any failed in CLI output for a call cause nothing happens .. :) |
| 16:54.57 | p3nguin | *2700 |
| 16:55.02 | p3nguin | Did you try that? |
| 16:55.14 | [TK]D-Fender | nephilinturbo: Your one from 2 hours ag o was WRONG, and you've done changes and I will never trust that it looks at all the same |
| 16:56.51 | p3nguin | Is a full restart required for changes to features.conf? |
| 16:57.17 | [TK]D-Fender | p3nguin: reload the module |
| 16:57.41 | nephilinturbo | [TK]D-Fender: Look . A Call between two extensions in my Asterisk. 3001 and 3002 . I talk to other phone normaly , but when i press #2 nothing hapens, no erros, no log, no output .. - http://pastebin.com/m43564a80 |
| 16:57.44 | p3nguin | Ah, yes. res_features.so |
| 16:58.03 | p3nguin | You didn't complete the transfer if you only dialed #2. |
| 16:58.19 | p3nguin | Dial #2, followed by the exten you you want transfer TO. |
| 16:58.29 | p3nguin | #2700, like I said, should sent to the parking exten. |
| 16:58.36 | nephilinturbo | p3nguin: *2700 ? No couse its not in my confs .. Where did you see this ? |
| 16:58.49 | p3nguin | Well, *2, since you used a * in the conf instead of #. |
| 16:59.06 | [TK]D-Fender | nephilinturbo: -- Executing Dial("SIP/3002-b7d05118", "SIP/3001|60") <---- you are not ALLOWING TRANSFERS |
| 16:59.08 | raden_work | morning Katty |
| 16:59.33 | p3nguin | I don't know; you've confused me so much, I don't know if you're using stars or pounds. |
| 16:59.57 | p3nguin | If you configured atxfer to use #2, then dial #2700 to transfer into parking. |
| 17:00.28 | p3nguin | Or #2234 to transfer to exten 234. |
| 17:00.57 | p3nguin | I did not uncomment nor change that setting in features.conf, and I only have to press # to start a transfer. |
| 17:03.36 | p3nguin | I know why I thought you were using *2. That's the default commented setting. You've uncommented and changed it to #2. |
| 17:03.48 | rps2 | Anyone ready to help me? Don't want to bust in on debugging neph's trouble. |
| 17:04.37 | [TK]D-Fender | rps2: Show complete debug for the failed call. |
| 17:04.58 | [TK]D-Fender | rps2: And some usable detail would help |
| 17:06.06 | rps2 | The log output is in http://pastebin.ca/1604638. I'm using Ekiga as a SIP client. When I call extension 6300 (a conference room), I get no audio. |
| 17:06.07 | *** join/#asterisk mumtazah (n=mumtazah@203.82.91.101) |
| 17:06.20 | nephilinturbo | p3nguin: ahhh . i understand what u are saying. But first> when i press # to transfer it sould make a signal like sound off like or music transfer . But not .. Also i tried put this : #23002 . #2 to get transfer and 3002 its extension . Whitout sucess |
| 17:07.05 | p3nguin | nephilinturbo: Did you use tT options in your dialplan? |
| 17:07.59 | [TK]D-Fender | nephilinturbo: You have not permitted transfers. Your attempts to do so are pointless. |
| 17:07.59 | nephilinturbo | p3nguin: when you say dialplan i are talking about file extensions.conf ? |
| 17:08.14 | [TK]D-Fender | [12:59]<[TK]D-Fender>nephilinturbo: -- Executing Dial("SIP/3002-b7d05118", "SIP/3001|60") <---- you are not ALLOWING TRANSFERS |
| 17:08.17 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 17:08.24 | [TK]D-Fender | WAKE UP TIME |
| 17:08.40 | rps2 | I might also mention that I'm using the GUI to configure this. Asterisk/1.6.2.0-rc3, GUI SVN-branch-2.0-r4980 |
| 17:08.50 | rps2 | On Fedora 11. |
| 17:09.18 | [TK]D-Fender | rps2: verbose 10, SIP DEBUG enabled please |
| 17:10.17 | nephilinturbo | [TK]D-Fender: Man .. I think you dont understand my problem.. Executing Dial("SIP/3002-b7d05118", "SIP/3001|60") - This is the call that im making between two extensions to test .. |
| 17:10.20 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 17:10.38 | [TK]D-Fender | nephilinturbo: Yes, and you cannot transfer to aprking fron that call |
| 17:10.48 | [TK]D-Fender | nephilinturbo: Yes, and you cannot transfer to parking from that call |
| 17:11.01 | nephilinturbo | [TK]D-Fender: I cant send to you outputs from transfers, cause i dont have ! when i press # to transfer , nothing happens |
| 17:11.08 | *** join/#asterisk errotan (n=errotan@62.201.122.189) |
| 17:11.28 | [TK]D-Fender | nephilinturbo: You don't have output because * is IGNORING YUO because you didn't set the OPTION YOU REQUIRE |
| 17:12.21 | nephilinturbo | [TK]D-Fender: humm .. What about this ? - atxfer => #2 . Its the option not ? |
| 17:12.21 | raden_work | anyone have any idea how far from a businesses main place of business a non compete will hold up ? |
| 17:12.30 | [TK]D-Fender | nephilinturbo: IRRELEVANT |
| 17:12.34 | Gugge | nephilinturbo: core show application dial |
| 17:12.39 | Corydon76-dig | raden_work: talk to a lawyer |
| 17:12.50 | [TK]D-Fender | nephilinturbo: Your DIAL doesn't not PERMIT transfers so what you have in features.conf is meaningless |
| 17:13.13 | nephilinturbo | [TK]D-Fender: Hummm .. What sould i do them ? |
| 17:13.14 | Corydon76-dig | raden_work: in the opinion of some legal professionals, a non-compete is unenforceable |
| 17:13.52 | [TK]D-Fender | nephilinturbo: You've already been handed the answer to this. Go read the instructions for it again. |
| 17:13.54 | raden_work | Corydon76-dig, hard to find one within 100 miles that this company doesnt have contact with |
| 17:13.54 | Corydon76-dig | but you need to talk to a lawyer if you're planning on violating some agreement like that |
| 17:14.00 | Naikrovek | raden_work: they're enforcable in illinois. and the distance is defined in the contract. if it isn't, it's zero |
| 17:14.16 | nephilinturbo | [TK]D-Fender: In my extensions.conf i set the options tT in dialplan |
| 17:14.23 | p3nguin | zero distance, or zero limit? |
| 17:14.25 | nephilinturbo | [TK]D-Fender: Its not right ? |
| 17:14.25 | raden_work | Naikrovek, the distance definition in the contract is the entire USA |
| 17:14.32 | [TK]D-Fender | nephilinturbo: And in your CLI output you clearly DON'T have it |
| 17:14.37 | raden_work | its freakin BS |
| 17:14.39 | Naikrovek | raden_work: that's probably an illegal contract |
| 17:14.44 | rps2 | Fender: http://pastebin.ca/1604684 |
| 17:14.52 | rps2 | That what you need? |
| 17:14.53 | Naikrovek | i'm back to Corydon76-dig's response - see a lawyer |
| 17:15.05 | [TK]D-Fender | [13:08]<[TK]D-Fender>[12:59] <[TK]D-Fender> nephilinturbo: -- Executing Dial("SIP/3002-b7d05118", "SIP/3001|60") <---- you are not ALLOWING TRANSFERS |
| 17:15.15 | raden_work | Naikrovek, will fax to my attorney out of state im just sick of all the BS here |
| 17:15.17 | [TK]D-Fender | nephilinturbo: another 15 minutes of shoving the proof right in your face. |
| 17:15.28 | [TK]D-Fender | nephilinturbo: No "Tt" in there |
| 17:15.40 | [TK]D-Fender | nephilinturbo: Look at the bloody evidence already... |
| 17:15.43 | Naikrovek | raden_work: find a job before you leave that one, unless you need a few months worth of vacation |
| 17:15.59 | raden_work | Naikrovek, I have my own business I sub contract |
| 17:16.10 | raden_work | Naikrovek, I have enough put away to live a year with 0 income |
| 17:16.17 | Naikrovek | raden_work: golden |
| 17:16.29 | raden_work | this place is crumbling apart |
| 17:16.34 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:16.42 | Naikrovek | raden_work: if your business is contracted to work there, then non-compete agreements are probably illegal |
| 17:16.56 | raden_work | I just lost $150,000 a year account becuase the owner of the company was lazy and didnt send out a $3,4000 credit check within 2 weeks |
| 17:17.02 | Naikrovek | one business can contract for multiple clients |
| 17:17.24 | p3nguin | nephilinturbo: I uncommneted the atxfer => *2 in features.conf, and now I can use # or *2 to begin the transfer. |
| 17:17.45 | Naikrovek | raden_work: that would be another reason to call a lawyer |
| 17:17.57 | raden_work | Naikrovek, well my view on this is I was offered emplyment then i signed a non compete as an employee and im still billing them after 3 years |
| 17:18.07 | rps2 | Naikrovek, Non-competes are unenforceable anyway, legal or not...at least here in California (Land of the Lawyers) |
| 17:18.29 | nephilinturbo | [TK]D-Fender: How you know , reading my output , that im not ALLOW to transfer ? |
| 17:18.30 | Naikrovek | rps2: in illinois they are most definitely enforcable. he's in wisconsin tho |
| 17:18.31 | rps2 | (where lawyers outnumber doctors 6 to 1) |
| 17:18.52 | Gugge | nephilinturbo: the output shows you dial command, without the Tt options |
| 17:18.59 | raden_work | Naikrovek, looking up state statues |
| 17:19.24 | raden_work | and then there the 1620 hours of overtime that ive worked in 4 years at base rate |
| 17:19.31 | [TK]D-Fender | [13:15]<[TK]D-Fender>[13:08] <[TK]D-Fender> [12:59] <[TK]D-Fender> nephilinturbo: -- Executing Dial("SIP/3002-b7d05118", "SIP/3001|60") <---- you are not ALLOWING TRANSFERS |
| 17:20.17 | rps2 | [TK]D-Fender: did you see my post? I know you're busy, just wanted to check. |
| 17:20.21 | [TK]D-Fender | nephilinturbo: Naikrovek ther is no "tT in there" |
| 17:20.32 | [TK]D-Fender | rps2: dump your firewall and test again |
| 17:20.43 | ariel_ | it's the case of no t or T used..... |
| 17:21.06 | rps2 | Well, I'll be damned! |
| 17:21.09 | [TK]D-Fender | I've only said it 10 &#$ing times pasting the culprit |
| 17:21.18 | ariel_ | yes I know |
| 17:21.33 | [TK]D-Fender | rps2: You need to not block 5060 & 10000-20000 UDP typically |
| 17:21.47 | nephilinturbo | [TK]D-Fender: humm ..you are right .. I think its cause im using this extensions 3001 , 3002 registered in another aplication called MOR |
| 17:21.57 | rps2 | I never thought of the firewall screwing that up. I have 5060 unblocked. Didn't think of the higher ones. |
| 17:22.20 | [TK]D-Fender | nephilinturbo: I think you are not paying attention to what you're doing or even looking at what I am putting right in front of you. |
| 17:22.26 | nephilinturbo | [TK]D-Fender: I think its couse this they are not following my dialplans |
| 17:22.45 | [TK]D-Fender | rps2: 5060 is the SIP control channel. that only sets up & handles the call. Voice is carried on RTP |
| 17:22.55 | rps2 | Got it. |
| 17:23.06 | rps2 | <---- FARKING MORON! |
| 17:23.26 | nephilinturbo | [TK]D-Fender: Sorry man .. Just tells me what i have to do. Couse im not understanding |
| 17:24.09 | [TK]D-Fender | nephilinturbo: -- Goto (mor_local,3001,9) <-- you aren't even showing us the dialplan where this context is DEFINED |
| 17:24.24 | [TK]D-Fender | rps2: No, you seem relative far from that state.... |
| 17:24.32 | nephilinturbo | [TK]D-Fender: You can look my dialplan in pastebin extension.conf and tells me what sould i do please |
| 17:24.47 | [TK]D-Fender | nephilinturbo: it isn't IN the dialplan you showed us. |
| 17:25.03 | [TK]D-Fender | nephilinturbo: because there is an INCLUDED FILE you somehow felt wasn't important to LOOK AT |
| 17:25.53 | [TK]D-Fender | nephilinturbo: 75. #include extensions_mor.conf |
| 17:25.55 | nephilinturbo | [TK]D-Fender: haaaa .. its truth ..sorry a lot .. i ll look here .. |
| 17:25.57 | [TK]D-Fender | nephilinturbo: HMMMMMMMMMMMMM |
| 17:26.58 | fofware | hello every buddies |
| 17:27.32 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 17:27.55 | ariel_ | says hello, while listening to One Night in Bangkok.... |
| 17:29.27 | *** join/#asterisk pyite_mac_ (n=dschreib@unaffiliated/pyite) |
| 17:29.32 | *** join/#asterisk galeras (n=galeras@190.27.246.226) |
| 17:30.30 | galeras | hello |
| 17:32.25 | ChannelZ | flashes his moobs |
| 17:33.20 | [TK]D-Fender | flash mobs |
| 17:34.31 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 17:34.43 | Naikrovek | moobs his flashes. |
| 17:37.06 | Naikrovek | SSHs his telnets |
| 17:37.35 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 17:39.01 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 17:39.46 | Katty | hmm. |
| 17:39.48 | Katty | dr appt soon. |
| 17:39.57 | Katty | wish me luck (= |
| 17:41.13 | Naikrovek | good luck |
| 17:42.44 | Naikrovek | my wife is getting a hysterectomy at the end of this month |
| 17:42.44 | Naikrovek | she is so flippin' stoked |
| 17:44.35 | KavanS | heh |
| 17:44.39 | KavanS | TMI |
| 17:49.14 | raden_work | Naikrovek, youd be so proud of me |
| 17:49.26 | Naikrovek | raden_work: i would? explain, please |
| 17:49.32 | raden_work | VirtualBox |
| 17:49.56 | raden_work | Installed |
| 17:50.01 | Naikrovek | nice |
| 17:50.08 | Naikrovek | took you long enough |
| 17:50.19 | raden_work | lol didnt think itd be so freakin easy |
| 17:50.19 | Naikrovek | :) |
| 17:50.25 | Naikrovek | i know |
| 17:50.39 | Naikrovek | you were gearing up for a fight with the computer i think |
| 17:50.45 | Naikrovek | such is the life of a linux user |
| 17:50.52 | raden_work | LOL |
| 17:51.11 | raden_work | i have to gear up for fights with windows more than linux i know for most people complete opposite |
| 17:51.18 | rps2 | [TK]D-Fender: Thanks! Things seem fine now. I really appreciate it! I'll leave you to assist the other people with problems while I deal with my wanker of a boss. Thanks again! |
| 17:51.29 | [TK]D-Fender | rps2: You're welcome |
| 17:51.32 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:ddb8:ca88:f49b:1b0f) |
| 17:51.32 | Naikrovek | yeah for me it was the opposite. I got so tired of fighting with linux on various things |
| 17:51.34 | raden_work | Naikrovek, well i have service in rockford ill catchya later |
| 17:51.39 | Naikrovek | wow haven't seen anyone thank you for a while. nice |
| 17:51.45 | Naikrovek | raden_work: laters |
| 17:51.52 | rps2 | Hey, I was raised properly! |
| 17:52.05 | rps2 | Later gang, and thanks again for the help! |
| 17:52.13 | rps2 | KZZZCHT! |
| 17:52.45 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 17:52.48 | p3nguin | Today, I fight with Windows Vista 64-bit. The networking on it is wonky. Running BartPE on the same machine (via CD), networking is fine. So the hardware is good. |
| 17:53.50 | Naikrovek | what's the network issue you're seeing |
| 17:53.51 | p3nguin | Oh, and I have to find some time to play with the SPA3102 and get that all working transparently. |
| 17:55.15 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 17:56.31 | p3nguin | naikrovek: Browsing sites, it's hit and miss if the pages load or not. At first, I thought it was probably a slow/bad DNS server causing lookup latency, but that was not the case. I decided to ping addresses not on the local machine just to see if there was lag or something. Pinging the DHCP server or the gateway or anything, might have 20 good pings, then destination host unreachable a couple times, then a ping time of maybe ... |
| 17:56.37 | p3nguin | ... 3000ms, then back to normal for a few pings. |
| 17:56.46 | Naikrovek | ew |
| 17:56.54 | guax | hello, what if i need to record and transfer that A calls B, A transfer to C, B talks to C, i need to record all that B heard |
| 17:58.45 | p3nguin | naikrovek: Since it's hit and miss on pings, I started trouble shooting the modem and the NIC and the usual stuff that could cause problems. A new modem didn't help, so I tried a different computer -- no problems. I put a different NIC in the "bad" computer, and the problem persisted. I tried BartPE on the "bad" computer, and no problems on pings there. That narrows it down to Vista64 having a problem. |
| 17:58.52 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:ddb8:ca88:f49b:1b0f) |
| 17:59.34 | Naikrovek | or some network magic that vista is trying to do with the card. |
| 17:59.36 | Naikrovek | or a driver |
| 17:59.36 | p3nguin | naikrovek: I reinstalled the Dell NIC driver, but that didn't help. I tried running the Startup Repair (thinking it would be like the repair of XP), but it failed, saying that it could not automatically fix the problem. |
| 17:59.54 | Naikrovek | k so not the driver, probably |
| 18:00.14 | p3nguin | Today, I am going to try another Vista64 installation on that machine. |
| 18:00.24 | Naikrovek | yeah |
| 18:00.51 | carrar | ewe |
| 18:01.10 | p3nguin | The lady told me she had already reinstalled Vista with the help of Dell phone support, so that makes me think there's something wrong with Vista in general. |
| 18:01.11 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 18:01.15 | DelphiWorld | hi all! |
| 18:01.16 | DelphiWorld | anyone have a gigaset phone? |
| 18:01.33 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 18:02.55 | Chainsaw | DelphiWorld: Yes, I use a C450 IP myself. |
| 18:02.57 | p3nguin | I can't imagine they would have used the actual reinstall DVD that shipped with the computer to install it at the factory. That's the one she used, and it's still having the problem. I intend to find someone else that has a 64-bit Vista DVD to eliminate yet another piece of this unusual puzzle. |
| 18:03.02 | Chainsaw | DelphiWorld: We have a few deployed in colo sites. |
| 18:03.24 | DelphiWorld | Chainsaw: could you help me if you can? |
| 18:03.31 | Chainsaw | DelphiWorld: Sure, what's up? |
| 18:03.43 | DelphiWorld | Chainsaw: just about dialplan... |
| 18:03.50 | DelphiWorld | Chainsaw: i have 3 difere connection into it |
| 18:03.51 | p3nguin | If Vista is the problem in general, she's willing to opt for the XP downgrade. |
| 18:03.58 | DelphiWorld | Chainsaw: how to chouse one to dial troug it? |
| 18:04.05 | Naikrovek | p3nguin: just install win 7 and end your problems :) |
| 18:04.16 | *** join/#asterisk PinkFreud (n=WhyNot@75-147-87-197-Philadelphia.hfc.comcastbusiness.net) |
| 18:04.41 | p3nguin | naikrovek: She doesn't have a license for 7. She bought the computer with Vista, and she would have to trade that in for the XP downgrade. |
| 18:04.49 | Naikrovek | ah |
| 18:04.53 | PinkFreud | hi folks. I have a bit of a headscratcher. Getting the infamous 'Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection.' error. |
| 18:04.56 | Naikrovek | how old is the computer |
| 18:04.57 | Chainsaw | DelphiWorld: I only have 1 SIP connection and 1 analog line to mine. |
| 18:05.14 | Chainsaw | DelphiWorld: It defaults to SIP, when I want analog I hold down the green button for ~3 secs. |
| 18:05.18 | p3nguin | naikrovek: She bought it 8/25/09.] |
| 18:05.20 | DelphiWorld | Chainsaw: could you find with me please;) |
| 18:05.27 | Naikrovek | PinkFreud: that's an infamous error? i never heard of it before. that doesn't mean much though. |
| 18:05.31 | PinkFreud | we're using the wct4xxp module. asterisk previously worked on here before a reboot, and now refuses to start. |
| 18:05.32 | p3nguin | naikrovek: aka, brand spankin' new. |
| 18:05.38 | Naikrovek | p3nguin: pretty sure she's eligible for a free win7 upgrade |
| 18:05.45 | PinkFreud | Naikrovek: google it. It's infamous. :) |
| 18:05.57 | p3nguin | naikrovek: Oh. Sounds like something I should check into. |
| 18:06.09 | Naikrovek | p3nguin: stand by, i'm way ahead of you :) |
| 18:06.21 | PinkFreud | teh google knows all, except why I'm receiving this error with actual channel hardware installed. |
| 18:06.25 | Chainsaw | DelphiWorld: I'm not near mine. Please ask specific questions. I don't know how to differentiate between multiple SIP profiles on one handset. I simply haven't used it that way. |
| 18:06.32 | Naikrovek | p3nguin: http://www.microsoft.com/windows/buy/offers/upgrade.aspx |
| 18:06.58 | DelphiWorld | Chainsaw: np |
| 18:07.10 | DelphiWorld | Chainsaw: i have GIGASET C470IP |
| 18:07.48 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp158-113.adsl.forthnet.gr) |
| 18:08.16 | PinkFreud | any ideas why asterisk throws that error and refuses to start? |
| 18:08.57 | Naikrovek | PinkFreud: don't think there's a whole lot of DAHDi expertise in here. If there is, someone please correct me. Maybe you could call Digium |
| 18:08.59 | p3nguin | naikrovek: https://win7.dell.com/Eligibility.aspx Looks like we're probably eligible. |
| 18:09.08 | *** part/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 18:09.14 | Naikrovek | p3nguin: yeah i was thinking you probably were. |
| 18:09.15 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 18:09.16 | [TK]D-Fender | PinkFreud: What hardware are you using with it? apstebin their configs. run "dahdi_cfg -vvvv" then restart * manually and confirm that it still occurs. |
| 18:09.17 | [TK]D-Fender | ~pb |
| 18:09.18 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 18:09.20 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 18:09.25 | Naikrovek | p3nguin: win7 is teh bombz |
| 18:09.51 | p3nguin | I don't know if she's up for the upgrade or not. She was willing to do the downgrade, though. |
| 18:10.29 | PinkFreud | interesting. not seeing dahdi_cfg on here. |
| 18:13.25 | PinkFreud | huh. can't find a debian package that contains that util. |
| 18:14.00 | PinkFreud | oddly, asterisk *did* run before the reboot |
| 18:14.07 | [TK]D-Fender | PinkFreud: And how DID you install * and its related bits exactly? |
| 18:14.27 | PinkFreud | compiled asterisk from source. |
| 18:14.42 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 18:15.04 | PinkFreud | this was earlier this year. got it running, and I haven't had the time to play with this box since. |
| 18:15.14 | [TK]D-Fender | PinkFreud: And what about DAHDI? |
| 18:15.28 | ZPertee | does anyone know anything about how to wire voip for your whole house? I understand asterisk, fxo/fxs, etc. but am not sure how I would hook it up to existing wiring |
| 18:15.41 | PinkFreud | that's a very good question. I know that we have channel hardware, but it's not connected to anything yet, and therefore not configured within asterisk. |
| 18:16.03 | [TK]D-Fender | ZPertee: You home's wiring is 1 giant FXS loop |
| 18:16.07 | ian6 | ZPertee: you could just wire it for ethernet and get phones that like that ;) |
| 18:16.46 | [TK]D-Fender | ZPertee: So disconnect from the telco at demark, and plug an ATA into a jack and your whole home wiring pools into the ATA |
| 18:17.02 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.11 | ZPertee | [TK]D-Fender, sounds easier than I thought really |
| 18:17.15 | ZPertee | :-) |
| 18:17.33 | PinkFreud | the thing that has me scratching my head is that asterisk previously ran, albeit only with local phones. |
| 18:17.57 | [TK]D-Fender | PinkFreud: And you haven't answered 2 of my questions yet |
| 18:18.10 | PinkFreud | the only thing that changed on here is a reboot. asterisk should still be configured the same way as before the reboot, so in theory, it should start. |
| 18:19.33 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp3-139.adsl.forthnet.gr) |
| 18:19.33 | PinkFreud | [TK]D-Fender: As I stated, I'm unsure how dahdi was installed. It looks like I'm using a debian package for the dahdi modules, and I have debian's dahdi-linux installed, but this does not appear to include dahdi_cfg. |
| 18:20.18 | [TK]D-Fender | PinkFreud: dahdi-linux != dahdi-tools |
| 18:20.22 | [TK]D-Fender | PinkFreud: Go look |
| 18:21.08 | [TK]D-Fender | PinkFreud: And because part is packaged who knows if the kernel modules match for Dahdi -> Kernel, and that Dahdi has followed *'s chan_dahdi |
| 18:21.09 | PinkFreud | Indeed I did. apt-cache search dahdi returns three packages: dahdi-linux, dahdi-source, and dahdi-modules-2.6.26-2-amd64 |
| 18:21.49 | PinkFreud | again, these packaes were installed well prior to the reboot, and asterisk *was* working fine. |
| 18:21.58 | Naikrovek | i like debian and i love apt-get but packages for a phone system are bad |
| 18:22.01 | PinkFreud | packages, too |
| 18:22.27 | Naikrovek | so, next time you do this, compile from source if you want to remove any package headaches. |
| 18:22.28 | PinkFreud | [TK]D-Fender: the modules were compiled for this kernel, yes. |
| 18:23.12 | tzafrir_laptop | PinkFreud, dahdi_cfg is part of the package 'dahdi' |
| 18:23.19 | tzafrir_laptop | Do you use dahdi or zaptel? |
| 18:23.32 | PinkFreud | there is no debian package 'dahdi' |
| 18:24.06 | tzafrir_laptop | dahdi comes from the source package dahdi-tools . dahdi-linux, dahdi-source: from the source package dahdi-linux |
| 18:24.07 | PinkFreud | tzafrir_laptop: looks like dahdi, but let me confirm. |
| 18:24.35 | tzafrir_laptop | How have you installed it? |
| 18:24.38 | PinkFreud | tzafrir_laptop: that's just it - I don't see dahdi-tools |
| 18:25.06 | tzafrir_laptop | What version of Debian? Any extra soruces? |
| 18:25.11 | PinkFreud | apt-cache search dahdi-tools returns nothing, and apt-cache search dahdi returns the above three packages |
| 18:25.15 | PinkFreud | lenny. |
| 18:25.32 | PinkFreud | nope. just using debian's main, contrib, and non-free repos |
| 18:25.33 | tzafrir_laptop | dahdi is not included in Lenny |
| 18:25.43 | PinkFreud | er, actually, we *do* have the squeeze repos enabled. |
| 18:25.51 | *** part/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 18:25.54 | PinkFreud | interesting. forgot about that. |
| 18:26.09 | PinkFreud | ahh, and sid. |
| 18:26.21 | PinkFreud | sighs. |
| 18:26.37 | tzafrir_laptop | I figure dahdi was not pulled as it depends on a newer libc6 |
| 18:26.42 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:26.49 | PinkFreud | ahh hah. just ran apt-get update. now dahdi appears. |
| 18:26.54 | PinkFreud | still no dahdi-tools, though |
| 18:27.14 | [TK]D-Fender | kicks packaged * bits in the nads |
| 18:27.16 | PinkFreud | that explains a missing package, but it still doesn't explain why asterisk previously worked. |
| 18:28.02 | tzafrir_laptop | http://packages.debian.org/search?keywords=dahdi |
| 18:28.38 | tzafrir_laptop | What version of Asterisk? |
| 18:28.52 | PinkFreud | 1.6.0.9 |
| 18:29.10 | tzafrir_laptop | built with dahdi support? |
| 18:29.27 | PinkFreud | I'm assuming so, given the error. Let me check. |
| 18:29.38 | tzafrir_laptop | what about libtonezone? |
| 18:29.47 | tzafrir_laptop | which do you have? |
| 18:30.10 | PinkFreud | <PROTECTED> |
| 18:30.19 | PinkFreud | heh, yep, built with dahdi support. |
| 18:30.52 | PinkFreud | ii libtonezone1 1:1.4.11~dfsg-3 tonezone library (runtime) |
| 18:30.54 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 18:32.57 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 18:33.04 | tzafrir_laptop | that's from zaptel. I suspect it didn't actually build with dahdi support |
| 18:33.13 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 18:33.32 | PinkFreud | interesting. |
| 18:33.40 | PinkFreud | so, how did asterisk previously work? |
| 18:34.04 | ChannelZ | All of that magic fairy dust I kept farting out |
| 18:34.40 | tzafrir_laptop | Not sure. But I keep everything as packages with proper build dependencies to avoid such issues |
| 18:34.47 | PinkFreud | ChannelZ: ahhh, THAT explains it! |
| 18:34.55 | PinkFreud | ChannelZ: ... can I have some more, please? |
| 18:35.09 | PinkFreud | ahhhhh, this MIGHT explain it. |
| 18:35.19 | ChannelZ | sorry I ran out |
| 18:35.23 | tzafrir_laptop | http://updates.xorcom.com/rapid/ |
| 18:35.46 | PinkFreud | ChannelZ's excess gas issues notwithstanding, it appears that unloading wct4xxp and dahdi lets asterisk start normally |
| 18:35.56 | citywok | can somebody eplease translate this error for me? it's causing registration to fail and read as unreachable: http://pastebin.com/d703b2d9f |
| 18:36.20 | PinkFreud | I wonder if those modules were simply not loaded before the reboot. |
| 18:36.50 | PinkFreud | if so, it means I have some work ahead to get dahdi to behave. bleh. |
| 18:37.06 | [TK]D-Fender | citywok: that has nothing to to with reg's failing. |
| 18:37.15 | fofware | sorry guys, but anyone can tellme what mean this [Oct 8 15:35:05] ERROR[6793]: chan_iax2.c:4529 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 82.25.95.128 in the calltokenignore list or setting user trine requirecalltoken=no |
| 18:37.18 | [TK]D-Fender | citywok: that is jsut a qualify packet from a peer entry |
| 18:37.35 | citywok | [TK]D-Fender: it's the only communication from asterisk to the peer whe nit attempts to register |
| 18:37.48 | citywok | they go back and forth with that same set of packets over and over, and it reads unreachable |
| 18:38.06 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 18:38.50 | [TK]D-Fender | citywok: I see none fo what you've described. No reg attempt, no answer to it, no configs, no peer status, no * registry dump. |
| 18:40.01 | citywok | i've got wireshark running on the peer, from the time i start asterisk the only communication to the peer is those 2 packets |
| 18:40.18 | citywok | on port 5060 anyways, and i'm assuming the reg attempt happens on 5060 |
| 18:41.28 | Naikrovek | PinkFreud: do you use DAHDi? |
| 18:41.39 | Naikrovek | PinkFreud: oh yeah of course you do |
| 18:41.45 | Naikrovek | PinkFreud: nevermoind |
| 18:41.53 | PinkFreud | apparently, not right now |
| 18:42.15 | Naikrovek | well you have phone line hardware? POTS lines going into your server? |
| 18:43.01 | PinkFreud | I have the hardware, but nothing's connected to it yet. |
| 18:43.09 | Naikrovek | ah |
| 18:43.57 | PinkFreud | looks like wct4xxp and dahdi had loaded, and due to dahdi not being configured, asterisk was bombing on start. |
| 18:44.18 | [TK]D-Fender | citywok: wireshark = irrelevant. Look at actual * SIP DEBUG |
| 18:45.24 | PinkFreud | anyway, [TK]D-Fender and tzafrir_laptop - thank you. |
| 18:45.46 | eppigy | hello |
| 18:45.49 | eppigy | i am dave |
| 18:45.50 | PinkFreud | I'll stop back if I have further questions, but at least * is running now. dahdi promises to be fun to configure, but that's further down the road. |
| 18:47.38 | citywok | [TK]D-Fender: the only info in sip debug talking to that peer is the exact same thing that came out of the wireshark dump |
| 18:48.10 | [TK]D-Fender | citywok: And no reg attempt to show or any of the other bits that matter. |
| 18:49.07 | citywok | correct, no other chatter between them. * receives method not allowed message, and then waits 30 seconds and sends the same exact request again. |
| 18:49.33 | [TK]D-Fender | citywok: Which is only the qualify. Where si the rest of what I asked for? |
| 18:52.13 | citywok | here's config & show peer http://pastebin.com/d1bc1c85c |
| 18:52.33 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-65-139.link.net.pk) |
| 18:54.06 | citywok | i'm compiling 1.6.0 right now to try against that because the guy on the microsoft UC team said he got it all working in 1.6.0, so to find out if it's a 1.6.1.6 issue i'm going back |
| 18:54.52 | [TK]D-Fender | citywok: Well I see no failed call, no failed reg (you put a fixed host ANYWAY, so unless * regs to THEM.... well) . Given you've put a fixed hos for a local LAN device I see no point in even bothering with the qualify |
| 18:55.17 | [TK]D-Fender | citywok: Problem is you have nothing to show and your description is pretty hollow |
| 18:56.11 | citywok | If i do not qualify, it works just fine, so i'm curious why the qualify fails. We've been diagnosing every little failure because when a call is placed from asterisk to the peer, and the peer answers, i think the RTP packet never gets back to asterisk , so it just says ringing forever until it times out. |
| 18:56.34 | citywok | if i call from the peer to asterisk, everything works perfectly fine. |
| 18:56.41 | [TK]D-Fender | citywok: They don't support it anyway and there is no need or point to it |
| 18:57.00 | [TK]D-Fender | citywok: And this has nothing to do with RTP |
| 18:57.19 | citywok | who doesn't support what? |
| 18:57.33 | [TK]D-Fender | citywok: your OCS doesn't support OPTIONS |
| 18:58.21 | citywok | yea, the qualify has nothing to do with RTP. but it's intriguing that it fails |
| 18:58.21 | [TK]D-Fender | citywok: Just like the response says and there is no point in doint it |
| 18:58.22 | citywok | yea, the guy on the UC team is investigating that since it works in 1.6.0. in 1.6.0 did that request not happen? |
| 18:58.22 | [TK]D-Fender | citywok: it doesn't technically "fail". its gets an answer, and thats all it should care about |
| 18:58.22 | [TK]D-Fender | citywok: If even |
| 18:58.23 | [TK]D-Fender | citywok: Neurosing over this is a waste of time |
| 18:58.38 | *** join/#asterisk galeras (n=galeras@190.144.60.186) |
| 18:58.39 | citywok | alrighty then, good to know i can leave it off and not worry about it. |
| 18:59.17 | Naikrovek | what's UC in this conversation |
| 18:59.38 | citywok | microsoft's unified communications team (the developers of OCS) |
| 19:00.08 | *** join/#asterisk Wangster_ (n=johnlang@host-253.epicnet.ca) |
| 19:00.59 | *** join/#asterisk lozarythmic (n=lpraties@e1-1.ns500-1.ts.milt.as9105.net) |
| 19:01.07 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 19:01.07 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:01.23 | Wangster_ | Hey all. I'm trying to set asterisk's umask so it creates group writable voicemail files. Google tells me that it should be done at runtime, but how the heck do you do that? |
| 19:02.50 | Naikrovek | Wangster_: i believe all you gotta do is change the permissions on the directory '.' - they'll get inherited by the files. ... I think. |
| 19:04.48 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 19:05.13 | Wangster | Naikrovek, negative.... :( Just tried. |
| 19:05.32 | *** join/#asterisk rps2 (n=rick@adsl-99-74-144-118.dsl.lsan03.sbcglobal.net) |
| 19:05.33 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 19:05.36 | Naikrovek | [TK]D-Fender: possible to change the sound a polycom phone makes when it's paged? I want to change the sound |
| 19:06.01 | rps2 | :back to bother you yet again: |
| 19:07.42 | rps2 | Not able to call out via our FXO ports. |
| 19:10.45 | rps2 | I get a "No channel type registered for DAHDI" error in the logs when I try. |
| 19:11.41 | Katty | well. |
| 19:11.45 | Katty | that was a fun doctor visit. |
| 19:12.42 | [TK]D-Fender | Naikrovek: Of course... thats what the Alert Info class is for |
| 19:12.48 | mchou | Katty: swine flu? |
| 19:13.12 | Wangster | Man, I can't believe this umask thing is that hard. Why would that not be a setting in an asterisk config file? |
| 19:13.52 | citywok | [TK]D-Fender: on an interesting note, it registers just fine in 1.6.0.1 |
| 19:14.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:14.30 | Katty | mchou: not exactly |
| 19:14.35 | [TK]D-Fender | citywok: Well you can't seem to even show me a failure... not sure what you're expecting at this point... |
| 19:14.42 | Katty | jaytee: you will be happy to know my blood pressure is going back down now that i'm off everything. |
| 19:14.55 | citywok | i get nothing other than those 2 packets going back and forth, sip debug also shows nothing more. |
| 19:15.01 | Katty | jaytee: 120/76 down from 131/85 |
| 19:15.13 | Kobaz | Baylink2: it's definitly the box or the t1 card, or asterisk.... |
| 19:15.24 | Kobaz | Baylink2: we have the old pbx plugged directly into the wall... no dropouts |
| 19:15.42 | [TK]D-Fender | rps2: chan_dahdi is clearly not loaded |
| 19:15.47 | Katty | jaytee: however they're putting me on yet another SSRI, to access if a different drug will work or if I do indeed have seratonin syndrome. |
| 19:16.03 | rps2 | My doctor always tells me to keep drinking coffee and smoking otherwise I'd be dead. My BP is like 100/60. |
| 19:16.10 | Katty | jaytee: can't say that i'm looking forward to the possiblity of my body going into shock again. |
| 19:17.04 | [TK]D-Fender | Katty: "Just Say No" |
| 19:17.18 | Katty | if i have seratonin syndrome, i need to know. |
| 19:17.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.29 | *** join/#asterisk levity (i=canuck@unaffiliated/canuck) |
| 19:17.53 | Katty | i've also been assigned to a therapist to help with my health spaz attacks. |
| 19:18.09 | citywok | something changed from 1.6.0.1 to 1.6.1.6 that broke it, but 1.6.0.1 didn't solve my main problem, so whtaever i'm leaving qualify off. someday somebody else will probably stumble across this bug too. |
| 19:18.24 | Katty | i guess that means i need therapy :< |
| 19:18.47 | Naikrovek | therapy is good |
| 19:18.51 | Naikrovek | good good good |
| 19:18.55 | Naikrovek | for everyone |
| 19:19.14 | [TK]D-Fender | Katty: "Serotonin syndrome is a potentially life-threatening adverse drug reaction that may occur following therapeutic drug use, inadvertent interactions between drugs, overdose of particular drugs, or the recreational use of certain drugs" <- means you get it BECAUSE of taking drugs, so just STOP. |
| 19:20.04 | rps2 | [TK]D-Fender: Ok, I'll restart and watch it closely. |
| 19:21.07 | dustybin | Katty: stay away from 'E' |
| 19:21.13 | Katty | e? |
| 19:21.13 | dustybin | hugs Katty |
| 19:21.30 | Katty | i just need to perminantly move into a hospital. |
| 19:21.35 | Katty | then i'd probably be perfectly fine! |
| 19:21.46 | dustybin | http://en.wikipedia.org/wiki/MDMA |
| 19:21.47 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:22.03 | Katty | oh that |
| 19:22.14 | Katty | i've never taken that before. |
| 19:22.21 | dustybin | aye good, just as well |
| 19:22.28 | [TK]D-Fender | Katty: My grandfather was perfectly fine UNTIL he ended up in a hopital. He banged himself in a fall. once in he when into congestive heart failure and then upon scaping out of that caught C-difficile |
| 19:22.53 | dustybin | jeeeze |
| 19:23.00 | Katty | dustybin: they had me on 20mg of prozac. |
| 19:23.18 | Katty | dustybin: they still don't know if that's what caused it. my bloodwork came back fine, tho my blood pressure was a bit high. |
| 19:23.26 | [TK]D-Fender | Katty: Katty And anti-depressants are on the list of CAUSES of seratonin syndrome. |
| 19:23.29 | dustybin | Katty: exercise is the best medicine |
| 19:23.30 | Katty | dustybin: my doctor said it was a Panic Attack |
| 19:23.32 | *** join/#asterisk [netman] (n=netman@80.31.167.130) |
| 19:23.50 | dustybin | good food, exercise, and irc |
| 19:23.57 | Katty | [TK]D-Fender: which means if SSRIs cause seratonin syndrome in me, i need to know. |
| 19:23.59 | dustybin | (well, maybe not irc) |
| 19:24.09 | Katty | i want to know. |
| 19:24.24 | [TK]D-Fender | Katty: And WORRYING about having seratonin syndrome leads to your taking anti-depressants which can LEAD to actually causing that problem. Youa re inventing your own problems. |
| 19:24.40 | Katty | yes, i know i need therapy |
| 19:25.01 | [TK]D-Fender | Katty: Told you before, and there is large evidence backing this.... GTFO or Missouri <- |
| 19:25.07 | [TK]D-Fender | of* |
| 19:25.15 | Katty | pats [TK]D-Fender |
| 19:25.21 | Katty | it's very nice of you to care so much. |
| 19:25.47 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 19:25.49 | dustybin | Katty: i bet if you did a 10 mile run, when you get back, you would no longer feel down |
| 19:26.16 | [TK]D-Fender | dustybin: Fair bet |
| 19:26.24 | mchou | screw running |
| 19:26.24 | dustybin | aye |
| 19:26.29 | mchou | bike RIDE |
| 19:26.33 | mchou | :) |
| 19:26.34 | Katty | i like running |
| 19:26.37 | Katty | it's enjoyable |
| 19:26.40 | dustybin | another cause of depression, is too much sleep |
| 19:26.41 | Katty | and often i take the pup with me. |
| 19:26.45 | [TK]D-Fender | Katty: http://www.youtube.com/watch?v=rHXXTCc-IVg |
| 19:26.59 | dustybin | people who sleep far too much are more likely to feel down |
| 19:27.11 | mchou | dustybin: I thought it was the other way around |
| 19:27.16 | robl^laptop | and feeling down makes you want to sleep more.. |
| 19:27.17 | dustybin | nope its not |
| 19:27.19 | citywok | dustybin: i sleep 9 hours a day and i feel great :) |
| 19:27.20 | Katty | i sleep a lot. |
| 19:27.30 | Katty | 8 hours :P |
| 19:27.31 | robl^laptop | its one of those self-feeding cycles. |
| 19:27.35 | Katty | preferably 10 |
| 19:28.02 | dustybin | one needs to exercise mind and body, eat good food |
| 19:28.42 | mchou | hmm FOOF...... |
| 19:28.47 | mchou | FOOD* |
| 19:28.51 | mchou | Doh! |
| 19:30.31 | [TK]D-Fender | "Ask your doctor if getting off your ass is right for you!" |
| 19:30.42 | mchou | dustybin: 10 mile run is a bit extreme |
| 19:31.04 | mchou | that's virtually half a marathon |
| 19:31.30 | dustybin | mchou: sure, not that far, but a good jog will wake your brain up |
| 19:31.42 | mchou | 6 miles should be more than sufficient :) |
| 19:31.43 | Katty | i run for 30min |
| 19:31.49 | Katty | bu ti'm not sure how far that really is |
| 19:32.02 | Katty | probably 4 or 5 miles, round trip |
| 19:32.07 | mchou | Katty: 5 miles max :) |
| 19:33.23 | dustybin | i got the very latest version of zoneminder 1.24.2 working with my logitech webcam, it took me a while to sus this out |
| 19:33.27 | dustybin | the problem was down to ffmpeg |
| 19:34.31 | rps2 | Hmmm, chan_dahdi.so wasn't built for some odd reason. |
| 19:35.16 | [TK]D-Fender | rps2: #1 is installing DAHDI after * |
| 19:35.57 | *** join/#asterisk scalex000 (n=chatzill@158puntacana02.codetel.net.do) |
| 19:35.58 | lozarythmic | dustybin: grats! i spent ages a while ago trying to get the same setup working |
| 19:36.10 | lozarythmic | if you've done it i might try again :) |
| 19:37.19 | Katty | scheduled to visit ENT on monday. sigh. |
| 19:37.35 | dustybin | lozarythmic: i compiled ffmpeg from latest svn, and that solved the problem |
| 19:37.42 | Katty | they're going to tell me i'm fine. and what i'm hearing is just normal ear noise. |
| 19:38.33 | Katty | that i'm just oversensitive and listening for it now :< |
| 19:38.44 | dustybin | Katty: what star sign are you? |
| 19:38.58 | Katty | star sign? ^_- |
| 19:39.05 | dustybin | birth sign? |
| 19:39.11 | [TK]D-Fender | Octagon.. as in "stop" :p |
| 19:39.17 | dustybin | :P |
| 19:39.20 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 19:39.32 | Katty | libra, i think |
| 19:39.49 | Katty | are you going to tell me my horoscope?! |
| 19:40.11 | dustybin | asterisk + zoneminder + heyu X10 + mythtv all have some kind of communication with eachother |
| 19:40.18 | dustybin | Katty: nope |
| 19:40.47 | rps2 | [TK]D-Fender: Not sure what you mean. I rebuilt dahdi and reinstalled it. |
| 19:40.53 | Katty | dustybin: why did you want to know then |
| 19:41.06 | [TK]D-Fender | rps2: Yes, and you need to recompile * AGAIN to include DAHDI support |
| 19:41.12 | dustybin | doesnt matter.. haha this is a strange freenode channel |
| 19:41.31 | rps2 | Ah. |
| 19:42.08 | Katty | mmmmmmmk. |
| 19:43.37 | *** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de) |
| 19:44.48 | rps2 | Still not building dahdi, TK. |
| 19:46.52 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 19:48.09 | [TK]D-Fender | rps2: tash the folder and reextract from scratch |
| 19:48.14 | [TK]D-Fender | trash* |
| 19:48.26 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 19:48.29 | DelphiWorld | Chainsaw: re |
| 19:48.38 | DelphiWorld | Chainsaw: do you have your GIGASET now? |
| 19:48.44 | Chainsaw | DelphiWorld: No, I'm still at work. |
| 19:48.52 | DelphiWorld | Chainsaw: ok, np |
| 19:49.00 | DelphiWorld | Chainsaw: could you dial my gigaset later? |
| 19:49.47 | *** join/#asterisk came0 (n=came0@rrcs-71-42-25-233.se.biz.rr.com) |
| 19:49.54 | rps2 | Doing the ./configure now. |
| 19:50.05 | DelphiWorld | rps2enjoi;) |
| 19:50.24 | rps2 | Doing "make" |
| 19:51.20 | *** join/#asterisk ssokol (n=fn-javac@ip70-189-74-33.ok.ok.cox.net) |
| 19:51.34 | Chainsaw | DelphiWorld: Sure, once I'm at home and get some time to myself. Got to assemble some furniture first. |
| 19:53.31 | *** join/#asterisk traxx (n=traxx@91-64-130-76-dynip.superkabel.de) |
| 19:56.33 | DelphiWorld | Chainsaw: thanks;) |
| 19:56.37 | DelphiWorld | Chainsaw: please remind me |
| 19:56.41 | citywok | [TK]D-Fender: i spent 2 days troubleshooting all of this, and i just comiled 1.6.0.1 and had issues, went to 1.6.0.8 and everythign works PERFECTLY. |
| 19:56.55 | citywok | in 1.6.0.1 in the FROM header, it's sending two quotes isnetad of one. |
| 19:57.15 | citywok | they fixed that for 8, and call control syncs up, registration works, everything is great. seems as though the 1.6.1 tree broke something somewhere |
| 19:57.18 | DelphiWorld | anyone havze H.323 up? |
| 19:57.55 | [TK]D-Fender | citywok: Well you didn't show a single relevant SIP communication. Not sure what you're expecting.... |
| 19:58.14 | DelphiWorld | any H.323 endpoint to dial it? |
| 19:59.21 | traxx | anybody know how to clear/delete a channel listed with ``sip show channels'' without having to restart asterisk? |
| 20:00.12 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:00.34 | DelphiWorld | hi tzafrir_laptop |
| 20:01.35 | Joel | traxx, soft hangup |
| 20:01.42 | rps2 | TK: Blew the dahdi and asterisk build directories away, then untarballed, rebuilt and installed dahdi. |
| 20:01.51 | DelphiWorld | Joel: hi, are you my friend in MSN? |
| 20:02.04 | rps2 | Untarballed asterisk, ./configure, make and chan_dahdi.so still isn't built. |
| 20:02.22 | Qwell | rps2: pastebin your config.log file |
| 20:02.23 | rps2 | This is RC3, BTW. |
| 20:02.40 | Joel | DelphiWorld, ? |
| 20:02.56 | [TK]D-Fender | rps2: And you should have done "make menuconfig" which would show you why |
| 20:03.00 | DelphiWorld | Joel: are you joel, i have joel in my MSN... |
| 20:03.27 | Joel | DelphiWorld, http://www.gpeters.com/names/baby-names.php?name=Joel |
| 20:03.52 | DelphiWorld | Joel: see my message in msn;) |
| 20:04.13 | Joel | DelphiWorld, what's msn? |
| 20:04.44 | DelphiWorld | Joel: SQLSRVX86@hotmail.com |
| 20:04.47 | DelphiWorld | Joel: i'm sur |
| 20:05.39 | Joel | DelphiWorld, no idea wtf you are on about. |
| 20:06.44 | Joel | I wish freenode had +g like efnet. |
| 20:07.03 | rps2 | Ah, depends on res_smdi and dahdi(-something (covered by the EXIT button) |
| 20:08.49 | *** part/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 20:10.03 | traxx | Joel: thanks. |
| 20:10.39 | Joel | traxx, np |
| 20:18.55 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:18.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:21.01 | rps2 | TK: looks like res_smdi would be built, but it's dependent on dahdi (which is installed) and tonezone (and I have no farking idea what that is). |
| 20:21.28 | rps2 | Ok, I'm obviously a boob on this. |
| 20:22.50 | [TK]D-Fender | checkout time, later all |
| 20:24.08 | *** join/#asterisk ShaunR (n=Shaun@staff.ndchost.com) |
| 20:30.00 | Naikrovek | O. M. F. G. (sorry but this is a nice util for me) http://technet.microsoft.com/en-us/sysinternals/ee656415.aspx |
| 20:30.10 | Naikrovek | maybe someone else here could use that |
| 20:30.37 | rps2 | Ok, why can't I find out what this tonezone thing is? |
| 20:30.44 | Naikrovek | timezone thing? |
| 20:31.21 | rps2 | No, tonezone. Asterisk won't build the chan_dahdi.so thing without dahdi, res_smdi and tonezone. |
| 20:31.35 | Naikrovek | what OS |
| 20:31.39 | Naikrovek | what distro, rather |
| 20:31.42 | rps2 | Fedora 11 |
| 20:32.22 | rps2 | dahdi is installed, res_smdi appears to be built by asterisk's make, but I can't find out what the devil tonezone is. |
| 20:32.39 | Naikrovek | tzselect? |
| 20:33.21 | p3nguin | Holy cow. This SPA-3102 has so many settings, I don't know how to configure it to work with my Asterisk on a remote network. |
| 20:33.38 | Naikrovek | rps2: what does 'ls -l /etc/localtime show |
| 20:33.50 | Naikrovek | rps2: what does 'ls -l /etc/localtime' show (quote correction) |
| 20:34.51 | rps2 | [root@asterisk dahdi]# ls -l /etc/localtime |
| 20:34.52 | rps2 | -rw-r--r--. 1 root root 2819 2009-10-07 10:08 /etc/localtime |
| 20:35.14 | bmoraca | p3nguin: what are you trying to have it do? |
| 20:35.14 | rps2 | I don't think that's it. |
| 20:35.35 | Naikrovek | okay, does 'ls -l /usr/share/zoneinfo/ show timezones? |
| 20:35.38 | Naikrovek | damn quotes |
| 20:36.07 | Naikrovek | rps2: what timezone are you in |
| 20:36.07 | p3nguin | bmoraca: I got an SPA-3102 and I have no idea how to configure it. My asterisk box is in a remote location. The guide that comes with the device SUCKS. |
| 20:36.08 | rps2 | I don't think it's a timezone thing, Naikrovek. |
| 20:36.32 | rps2 | It is "tonezone", NOT "timezone" |
| 20:36.44 | bmoraca | p3nguin: that didn't answer my question. |
| 20:36.49 | *** join/#asterisk ZPertee (n=ZPertee@cpe-71-79-176-153.neo.res.rr.com) |
| 20:36.53 | rps2 | Seems to be something from Zaptel, I'm using Dahdi. |
| 20:36.55 | Naikrovek | aaaahhh tonezone |
| 20:36.57 | Naikrovek | hah |
| 20:37.00 | Naikrovek | no idea on that one |
| 20:37.12 | bmoraca | p3nguin: are you just trying to use it as a remote FXS port or are you trying to use it as a remote analog trunk port (FXO)? |
| 20:37.30 | jaytee | rps, where are you located? |
| 20:37.38 | rps2 | California. |
| 20:37.45 | jaytee | tonezone=us |
| 20:37.47 | ZPertee | I have kind of an odd question. If I open up my demarc device and I am wired for 2 lines how do I know which is which. We used to have 2 lines of service but then canceled the one. that is the one i want to use |
| 20:37.58 | p3nguin | bmoraca: I'm trying to configure the SPA-3102 to work with my Asterisk box. It's going to be using FXS for a phone and FXO to the wall jack, but Ethernet to register to Asterisk. |
| 20:38.03 | jaytee | and that's set in /etc/dahdi/system.conf |
| 20:38.48 | *** join/#asterisk d4rkstar (n=bruno@ip-42-150.sn1.eutelia.it) |
| 20:39.01 | bmoraca | p3nguin: "work with my Asterisk box" tells me absolutely nothing about HOW you want it to WORK. i can't help you configure it until you tell me how you want it to be configured. |
| 20:39.09 | p3nguin | bmoraca: The purpose of purchasing it was to retain landline service, but add SIP capabilities to this phone. |
| 20:39.14 | rps2 | Ok, there was "loadzone = us" and "defaultzone = us" but no "tonezone = us". |
| 20:39.16 | rps2 | That's it? |
| 20:39.50 | bmoraca | p3nguin: that's better. what exactly is it that is giving you problems? |
| 20:39.54 | Naikrovek | ZPertee: is there a cable coming out of it? |
| 20:39.57 | jaytee | sorry, loadzone not tonezone. that should set the correct zone for your tone indications |
| 20:40.01 | SuPrSluG | p3guin: just go to the lines and enter the proxy + username + password. that's all. you don't need to configure anything else |
| 20:40.29 | jaytee | rps2 so you don't need to change that then |
| 20:40.31 | ZPertee | Naikrovek, yes. that's probably the active one? |
| 20:40.50 | Naikrovek | ZPertee: well how many wires are in that cable |
| 20:41.07 | p3nguin | bmoraca: Quite frankly, all the settings are overwhelming. I just need to put in my asterisk SIP credentials, but I don't really know where it all goes. |
| 20:41.31 | *** join/#asterisk qu1ck (n=leonja51@dsl-240-245-33.telkomadsl.co.za) |
| 20:41.35 | SuPrSluG | p3guin: just go to the lines and enter the proxy + username + password. that's all. you don't need to configure anything else |
| 20:41.39 | rps2 | jaytee: Yeah, I didn't think so, but it still won't build the beblistered chan_dahdi.so file. |
| 20:41.45 | qu1ck | him i need help to get accountcode written to mysql cdr |
| 20:41.45 | ZPertee | Naikrovek, I'm going to get myself in trouble. I am not at my house right now, so I'm not entirely sure of all of the specifics. I'll have to check it out better |
| 20:42.15 | p3nguin | I put in my asterisk server's name in the proxy field, then the user ID (phone exten number) and the password. |
| 20:42.20 | Naikrovek | ZPertee: well phone cords typically can carry between 2 and 4 phone lines |
| 20:42.32 | *** join/#asterisk [jmc] (n=John@93-45-193-132.ip103.fastwebnet.it) |
| 20:42.41 | p3nguin | I left outbound proxy empty and made sure the box was marked No. |
| 20:42.43 | ZPertee | Naikrovek, ok |
| 20:42.45 | qu1ck | is there something other than accountcode=whatever in iax.conf that needs to be set? |
| 20:42.58 | bmoraca | p3nguin: have you logged in to the web interface yet? if you have, click on the "admin" link on the upper right side of the page. from there, most of it is pretty straight forward as far as configuring the Line (FXS port). the other port (FXO), depending on how you want it to behave, is a little different, and i can help you with that at that time...for now, though, just configure the FXS... |
| 20:43.00 | bmoraca | ...port to get it working as a SIP peer to your asterisk box. |
| 20:43.09 | *** join/#asterisk |Rain| (i=rain@ev.il.net) |
| 20:43.30 | Naikrovek | ZPertee: two wires per phone line. the center pair is line 1, the pair outside that is line 2 ... the outside wires are line 4. so if you look at a cable with 6 wires, the wires belong to lines 3,2,1,1,2,3 |
| 20:43.31 | ZPertee | Naikrovek, can I plug a phone into the demarc telephone jack? |
| 20:43.36 | bmoraca | p3nguin: i've found it's easier to configure those in "Advanced" mode. is it having trouble registering with asterisk? if it is, that might be a NAT issue, which you can correct in Advanced mode. |
| 20:43.46 | Naikrovek | ZPertee: if what i'm thinking is correct, yes |
| 20:44.50 | p3nguin | bmoraca: I put in the info and saved. When running sip debug, I see 401 Unauthorized. |
| 20:44.55 | qu1ck | hi. i need help to get accountcode written to mysql cdr? |
| 20:45.17 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:45.27 | ZPertee | Naikrovek, ok. I'll check it out more. thanks for your help! :-) |
| 20:45.39 | Naikrovek | ZPertee: don't thank me until i'm proven correct |
| 20:45.47 | ZPertee | Naikrovek, :-) |
| 20:45.48 | Naikrovek | ZPertee: but you're welcome |
| 20:46.15 | bmoraca | p3nguin: are you absolutely positive that you've entered in the correct authentication information? |
| 20:46.44 | p3nguin | bmoraca: Yep. I even went back over them thinking I could have typoed the password. |
| 20:47.27 | bmoraca | p3nguin: you mentioned they're on different networks...do you have any denys or anything in your SIP peer? |
| 20:47.49 | bmoraca | p3nguin: additionally, are they connected by a VPN or are they going over the general internet? |
| 20:48.23 | jaytee | rps2, what dahdi tarball did you try to build from? |
| 20:49.36 | p3nguin | bmoraca: I have been using a softphone from a Windows computer on this LAN. It's been working fine. I turned off the softphone and connected an Ethernet cable from the Ethernet switch to the Internet port on the 3102. SIP traffic goes over the public internet. |
| 20:50.05 | bmoraca | p3nguin: what is the local subnet on each? |
| 20:52.10 | bmoraca | p3nguin: also, try enable "Make Call Without Reg" and try to place a call. if you still get 401, then you've got an authentication issue |
| 20:52.11 | p3nguin | bmoraca: The asterisk server is on the public internet, the LAN where the Linksys device operates is 172.16.0.0. |
| 20:52.17 | p3nguin | http://pastebin.com/d28eed4ac |
| 20:52.44 | |Rain| | so I'm looking at RFC3842 (SIP MWI) and I think the value that's used (global.vmexten in sip.conf) isn't correct (my interpretation is that Message-Account should reflect the mailbox number, not the VM extension) |
| 20:52.52 | |Rain| | anyone know anything about that? |
| 20:52.53 | bmoraca | p3nguin: and you don't have any localnets in your [general] or any deny/allows in your peers in sip.conf? |
| 20:53.26 | bmoraca | p3nguin: that's a register not an invite...try to place a call. |
| 20:53.38 | p3nguin | bmoraca: Nah, since it's on a public IP address, I didn't define those things (since they are for NAT). |
| 20:54.05 | bmoraca | p3nguin: not always...they can also be used to restrict who can or cannot register with the server. |
| 20:56.16 | p3nguin | The dialplan is also going to be a problem, so I can't make a call. |
| 20:56.20 | SuPrSluG | p3guin: try nat=yes under the perr entry |
| 20:56.26 | SuPrSluG | peer |
| 20:56.56 | bmoraca | p3nguin: what's wrong with the same context you used with the softphone? |
| 20:57.12 | p3nguin | bmoraca: The 3102 has a dialplan built in. |
| 20:57.32 | p3nguin | I don't understand it, so I don't really know what's what. |
| 20:57.35 | bmoraca | p3nguin: the default settings for that will let you dial pretty much anything. |
| 20:57.51 | jaytee | rps2, did you install dahdi-tools? that's where you get libtonezone.so from |
| 20:57.53 | [jmc] | hi |
| 20:57.57 | p3nguin | Let me try to capture a call in debug. |
| 20:58.30 | rps2 | Ok, this is bloody ridiculous. |
| 20:58.42 | [jmc] | what happens_ |
| 20:58.43 | [jmc] | ? |
| 20:58.46 | jaytee | rps2, did you install dahdi from a tarball or a package? |
| 20:58.53 | rps2 | tarball. |
| 20:58.58 | hardwire | what did you call me? |
| 20:59.47 | p3nguin | I can't. There's way too much shit flying past to get the debug of a call. |
| 20:59.52 | jaytee | for the tonezone you need to install either dahdi-complete-2.2.0.2+2.2.0 or if you just installed the dahdi drivers tarball you need to install the dahdi-tools tarball as well |
| 20:59.54 | rps2 | Yes, I installed dahdi-tools. It's in /usr/lib64 |
| 21:00.20 | jaytee | what distro? |
| 21:00.28 | rps2 | Fedora 11 |
| 21:00.53 | carrar | Can't I just click GO and everything works? |
| 21:00.59 | rps2 | And libtonezone is actually in the dahdi-tools-libs RPM |
| 21:01.32 | jaytee | rps2, I'm running it on RHEL 5 64 bit and on my system it's in /usr/lib not /usr/lib64 |
| 21:01.39 | p3nguin | Well, maybe I can. Just going to take a minute or so. |
| 21:02.01 | rps2 | Are you running a 64-bit OS? |
| 21:02.15 | jaytee | rps2, does it successfully build the dahdi driver modules? |
| 21:02.16 | rps2 | And I did a "./configure --with-tonezone=/usr/lib64" |
| 21:02.30 | [jmc] | is there anybody here that succeeded in connecting Asterisk with Google Talk? |
| 21:02.32 | rps2 | Yes, the drivers are built and the hardware checks out. |
| 21:02.37 | [jmc] | I can't get it to work, it seems |
| 21:02.53 | jaytee | are you set on using Fedora 11? |
| 21:03.48 | [jmc] | I'm using this jabber.conf file: http://www.nopaste.com/p/aeG2jj4Nl |
| 21:04.36 | [jmc] | and this gtalk.conf file: http://www.nopaste.com/p/a1NZyodRjb |
| 21:04.42 | p3nguin | bmoraca: Any chance this captures any of what you wanted? http://pastebin.com/d4aed30d1 |
| 21:05.06 | rps2 | The Fedora RPM of asterisk worked fine, except when you tried to use the directory function, then it would get a general protection crash on one of the libraries. |
| 21:05.19 | rps2 | Someone said, "use the tarball", so I'm trying. |
| 21:05.33 | hardwire | try furballs next |
| 21:05.34 | rps2 | This is the RC3 tarball. Perhaps it's screwed up? |
| 21:05.50 | hardwire | what url? |
| 21:06.23 | rps2 | The link from the home page. |
| 21:06.39 | bmoraca | p3nguin: Retransmitting #1 (no NAT) to 172.16.197.244:5060: <===== well there's your problem. |
| 21:06.42 | rps2 | 1.6.2.0-rc3 |
| 21:07.24 | p3nguin | bmoraca: But a softphone on this same network works fine. I can't see how that's a problem. |
| 21:07.36 | hardwire | rps2: when you installed the other tools.. specifically the dahdi tools.. did you use a prefix? |
| 21:08.00 | bmoraca | p3nguin: how can that NOT be a problem? Your asterisk server on a PUBLIC network has no route to 172.16.197.244. |
| 21:08.07 | hardwire | if not.. no need to specify a prefix.. if tonezone isn't being found you have more problems then defining where it is. |
| 21:08.13 | hardwire | found/detected |
| 21:08.17 | bmoraca | p3nguin: enable nat=yes on the peer in sip.conf and try again |
| 21:08.25 | p3nguin | okay |
| 21:08.27 | rps2 | No, no prefix. |
| 21:08.39 | rps2 | I used the RPM. |
| 21:08.48 | hardwire | did you install the dev RPM? |
| 21:09.12 | rps2 | Now, THAT I didn't. Hmmm. |
| 21:09.21 | hardwire | suggests removing libpri/libss7/dahdi-etc.. RPM packages and building them directly |
| 21:10.16 | p3nguin | bmoraca: Although I feel it's ridiculous that a softphone in parallel works with nat=no, I was able to make a call after changing it to nat=yes. |
| 21:10.24 | rps2 | Ok, devel installed, trying configure again. |
| 21:10.32 | hardwire | it WILL find it. |
| 21:10.35 | hardwire | :P |
| 21:10.53 | hardwire | hopefully it won't respond with "Tonezone too old: Seriously dude you're killing me here.." |
| 21:11.05 | p3nguin | bmoraca: Someone was just saying a couple days ago, too, that clients behind NAT very rarely require nat=yes on the server. Odd. |
| 21:11.06 | Kobaz | http://pastebin.ca/1605184 |
| 21:11.11 | DigitalFlux | I need to implement an H323 - SIP proxy |
| 21:11.18 | DigitalFlux | Can asterisk do that for me ? |
| 21:11.21 | hardwire | DigitalFlux: sure |
| 21:11.27 | bmoraca | p3nguin: a softphone behaves differently. NAT can be configured from both sides...i.e. the UA can take care of formulating the packets to behave with NAT or the server can read the packets in a particular manner which is consistent with NAT |
| 21:11.31 | Kobaz | i'm seeing this on my audiocodes fxo gateway, when recieving a call from the fxo line |
| 21:11.49 | DigitalFlux | hardwire: can you help me with more details or resources ? |
| 21:11.56 | Kobaz | asterisk is sending the audiocodes something it doesn't like |
| 21:12.07 | hardwire | DigitalFlux: Whats your linux experience like? |
| 21:12.17 | rps2 | Well, I'll be gobsmacked! |
| 21:12.22 | rps2 | Look at that! |
| 21:12.29 | bmoraca | p3nguin: i wouldn't say that. nat=yes causes Asterisk to ignore the source address reported in the SIP messages and instead use the address it RECEIVED the packet from |
| 21:12.32 | hardwire | ^5's rps2 |
| 21:12.45 | rps2 | It should build chan_dahdi.so now. Let's see if it's fibbing to me. |
| 21:13.05 | hardwire | watches it complain about libssl being gone |
| 21:13.09 | hardwire | err libss7 |
| 21:13.10 | bmoraca | p3nguin: as stated, some SIP UAs are smart enough to realize that they're behind a NAT and can discover what the outside IP is...some people run them through a transparent proxy or ALG, as well. |
| 21:14.13 | [TK]D-Fender | DigitalFlux: Seting an H323 entry to receive calls from and send calls to and another for SIP |
| 21:14.16 | bmoraca | p3nguin: in general, though, it's ALWAYS best to be as explicit as possible in your server configuration and not rely on tricks like transparent proxies and ALGs, which may or may not work. |
| 21:15.10 | p3nguin | bmoraca: Okay, sounds reasonable. |
| 21:15.54 | Kobaz | http://pastebin.ca/1605184 <- i'm having some issues with recieving calls on an audiocodes gateway via an fxo port |
| 21:16.10 | p3nguin | bmoraca: The next thing is the dialplan on the device. I need PSTN to take almost all calls by default, and only send calls to asterisk that start with #. e.g., 545-3000 will use PSTN, but #500 will use SIP. |
| 21:16.34 | rps2 | It built! |
| 21:16.37 | [jmc] | guys |
| 21:16.46 | rps2 | <---- completely red faced. |
| 21:16.49 | [jmc] | how can I compute the need for an Asterisk server? |
| 21:17.14 | bmoraca | p3nguin: my experience with the SPA-3102s is to have asterisk handle all call control with them. i configure the FXO port to behave as a trunk in Asterisk and then use Asterisk to handle where calls should go. |
| 21:17.19 | [jmc] | I mean, if I want to make sure the hardware on which Asterisk runs on is good enough to take on a certain number of calls |
| 21:17.38 | [jmc] | how can I know what will I be needing? |
| 21:18.22 | bmoraca | p3nguin: i would recommend reading this document thoroughly: http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf |
| 21:18.37 | p3nguin | I have it on screen right now. |
| 21:20.07 | bmoraca | pretty much everything you need is in that document. are calls coming in on that analog line going to go anywhere other than the phone attached to the SPA-3102? |
| 21:20.22 | carrar | jmc, expirence |
| 21:20.52 | [jmc] | d'oh :D |
| 21:20.57 | [jmc] | ok |
| 21:21.00 | rps2 | It works! Thanks, jaytee, hardware, TK, all you chaps! Ok, I'm back online. I shall flog myself unmercifully with limp linguini for wasting your time. |
| 21:21.38 | bmoraca | [jmc]: go big or go home...Core i7 with RAID 6 15K RPM SAS hard drives and 8gb RAM for everything |
| 21:21.51 | [TK]D-Fender | rps2: Al dente..... |
| 21:23.32 | citywok | is it possible to add a queue member that is a trunk? SIP/OCS_TRUNK/1593 for example. i've tried a few ways and been unsucesful |
| 21:23.48 | carrar | jmc, http://www.supermicro.com/products/motherboard/Xeon1333/5100/X7DCU.cfm |
| 21:23.51 | bmoraca | citywok: use a local channel |
| 21:23.52 | carrar | that will do you right |
| 21:24.12 | *** join/#asterisk degrade (n=degrade@unaffiliated/degrade) |
| 21:24.26 | [TK]D-Fender | citywok: Yes |
| 21:24.29 | p3nguin | bmoraca: Why does the PSTN Line configuration page accept proxy settings? Do people use IP over PSTN somehow? |
| 21:24.35 | bmoraca | carrar: those are old-style... |
| 21:24.45 | [TK]D-Fender | citywok: And there is no such thing as a "trunk" |
| 21:24.47 | carrar | and work great |
| 21:24.48 | citywok | bmoraca: thank you, i knew there had to be a way |
| 21:25.27 | bmoraca | p3nguin: no...you can configure it to register as a SIP peer to an asterisk server and then have asterisk place calls out of and accept calls into it |
| 21:25.43 | degrade | Does anybody that will go to astricon? |
| 21:25.49 | [TK]D-Fender | citywok: And there is no technical need for a Local channel |
| 21:25.50 | carrar | I was |
| 21:25.53 | Naikrovek | there are a few in here that are going i think |
| 21:25.54 | carrar | not anymore |
| 21:26.09 | citywok | [TK]D-Fender: how would you do it? |
| 21:26.10 | carrar | can't risk getting swine flu from you craazy people |
| 21:26.14 | p3nguin | bmoraca: Isn't that what the Internet port is for? |
| 21:26.26 | bmoraca | p3nguin: er...no... |
| 21:26.42 | bmoraca | p3nguin: the "internet" port is just a WAN port...you can use that device as a NAT router if you want to |
| 21:27.09 | carrar | too hot in AZ anyways |
| 21:27.22 | bmoraca | p3nguin: the PSTN line is just an FXO port, and you can configure that FXO port as a SIP peer and use it to place and receive calls to and from the PSTN |
| 21:27.24 | [TK]D-Fender | citywok: Depends how you want to have them. |
| 21:27.48 | citywok | the local channel route works, it dials straight in to my OCS server so i can have microsoft communicator be a member of an asterisk queue |
| 21:27.55 | [TK]D-Fender | citywok: Of course the real problem is that you keep saying things "don't work" and never show failure |
| 21:28.06 | *** part/#asterisk robl^laptop (n=robl@m445336d0.tmodns.net) |
| 21:29.25 | p3nguin | bmoraca: Oh, so I could dial the landline phone number from some other arbitrary landline phone, which would reach the PSTN jack on the device, then it would route the call over the ethernet to the asterisk server? |
| 21:30.10 | [jmc] | another quick question |
| 21:30.13 | bmoraca | p3nguin: if that's how you want to configure it, yes. that's how I use them for my hosted PBX customers...some of them have off-site extension lines for hospitals... |
| 21:30.28 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:30.32 | [jmc] | do mailboxes in VoiceMail need to have the same name of a SIP user? |
| 21:30.41 | carrar | no |
| 21:30.43 | [jmc] | I think "no" |
| 21:30.48 | ariel_ | no |
| 21:30.57 | p3nguin | bmoraca: This device obviously does way more than I needed. I just wanted an ATA that supported PSTN fallback. |
| 21:31.00 | [jmc] | ok |
| 21:31.09 | [jmc] | it says me wrong login |
| 21:31.20 | [jmc] | even if I enter the correct data |
| 21:31.26 | bmoraca | p3nguin: it will do that...it doesn't HAVE to be used the way I use it |
| 21:31.56 | ariel_ | it's that time, time to go home... g'nite folks |
| 21:33.18 | *** join/#asterisk Jumpie (i=n3rdz@c-76-100-241-4.hsd1.md.comcast.net) |
| 21:33.19 | Jumpie | sup guys |
| 21:33.20 | Jumpie | :) |
| 21:33.44 | Jumpie | quick q....curious if there is a way to have some kinda 'heartbeat' check at quick glance for specific servers |
| 21:34.10 | Jumpie | so like lets say i have 10 customers with a pbx..and instead of doing a test call, or trying to log into the box, etc..some kinda sip version of whatsupgold, etc for a quick check maybve in some kinda console that a server health is good |
| 21:38.03 | [jmc] | would this voicemail.conf be enough? http://www.nopaste.com/p/a7Q2bd0jbb |
| 21:38.31 | [jmc] | just one mailbox for testing purposes |
| 21:38.52 | *** join/#asterisk rizwan (n=u2006231@121.52.144.100) |
| 21:41.36 | *** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 21:42.08 | [TK]D-Fender | [jmc]: You have no mail settings from [general] so forget the e-mail... |
| 21:42.17 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 21:42.37 | [TK]D-Fender | [jmc]: Then there is the obvious question : Does it work? |
| 21:42.45 | [jmc] | [TK]D-Fender, I don't mind the email right now |
| 21:42.51 | [jmc] | no, that's the point |
| 21:43.03 | [jmc] | I thought it was simple enough: mailbox â 1, password â 159753 |
| 21:43.13 | [jmc] | but it says the following: |
| 21:43.17 | ManxPower-work | make in dahdi-linux-complete errors out. It's downloading the file "auth.asp?url=http:%2F%2Fdownloads.digium.com%2Fpub%2Ftelephony%2Ffirmware%2Freleases%2Fdahdi-fwload-vpmadt032-1.17.0.tar.gz" |
| 21:43.30 | [jmc] | http://www.nopaste.com/p/aqVT1usK9 |
| 21:43.41 | [jmc] | telling me "login incorrect" |
| 21:43.47 | geneticx | Hi everyone. if I already have a digium card TDM400P and would like to add another one for more ports can I do that easily? |
| 21:45.27 | ManxPower-work | [jmc]: need your voicemail.conf pastebin |
| 21:45.34 | ManxPower-work | geneticx: no problem, add the card, be done. |
| 21:45.51 | [jmc] | ManxPower-work, posted it before: http://www.nopaste.com/p/a7Q2bd0jbb |
| 21:45.51 | [TK]D-Fender | [jmc]: - Incorrect password '159753' for user '1' (context = ivr) <---------- HMMMMMMMMMMMMMM |
| 21:46.06 | [TK]D-Fender | [jmc]: Maybe there is something that stands out here |
| 21:46.08 | [jmc] | "user"? |
| 21:46.14 | ManxPower-work | Does anyone know why dahdi dails to install the firmware? |
| 21:46.16 | [TK]D-Fender | [jmc]: Kinda like what shelf to look in... |
| 21:46.35 | [TK]D-Fender | [jmc]: No, not "shelf"... thats not right........... |
| 21:46.36 | ManxPower-work | [jmc]: you have the mailbox in the [default] context. You are trying to log into that mailbox in the [ivr] context. |
| 21:46.56 | ManxPower-work | default != ive: FAIL |
| 21:47.02 | [jmc] | d'oh |
| 21:47.09 | [jmc] | -.-' |
| 21:47.17 | [jmc] | stabs himself with a knife |
| 21:47.28 | *** join/#asterisk telecos (n=sergio@198.167.219.87.dynamic.jazztel.es) |
| 21:47.44 | ManxPower-work | you're also missing the [general] section and settings. |
| 21:47.47 | [jmc] | thanks! |
| 21:48.42 | ManxPower-work | http://pastebin.ca/1605276 |
| 21:49.05 | ManxPower-work | It looks like digium's site is hosed for non-browser downloads |
| 21:49.28 | [jmc] | god it works great now :) |
| 21:49.31 | [jmc] | really thanks guys |
| 21:53.08 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 21:53.57 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 21:53.57 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:55.02 | drmessano | is running 1.6.4 LTS - Punky Platypus |
| 21:55.56 | russellb | hax |
| 21:55.56 | [TK]D-Fender | is waiting for Quagmire Quail to sink his call performance.... |
| 21:56.16 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:57.57 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 21:58.26 | drmessano | What about bones of the body? |
| 21:58.48 | drmessano | Like 1.6.4 LTS DISTAL PHALANGE |
| 21:59.18 | scalex000 | TK: I getting a notice, failed to decrypt, this is a bad news or I don't need to pay attention. |
| 21:59.19 | ManxPower-work | russellb: |
| 21:59.32 | ManxPower-work | russellb: Do you know if Digium's FTP server listens on any ports in addition to 80? |
| 21:59.49 | ManxPower-work | not FTP server. downloads.digium.com |
| 22:00.01 | carrar | you mean web server? |
| 22:00.11 | ManxPower-work | I mean downloads.digium.com |
| 22:00.20 | drmessano | upgrades from lumbar vertebrae to intermediate cuneiform |
| 22:00.53 | carrar | downloads.digium.com:80 is Apache/2.2.8 running on Ubuntu |
| 22:01.22 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 22:02.08 | ManxPower-work | carrar: Any my question is "is it running on any other ports". It looks like the answer is "no" |
| 22:02.21 | carrar | scan it |
| 22:02.26 | hardwire | heh |
| 22:02.38 | ManxPower-work | nevermind then |
| 22:02.41 | carrar | heh |
| 22:02.41 | drmessano | SSH in and check the apache config |
| 22:02.45 | carrar | yeah! |
| 22:02.56 | drmessano | User: root, password: trixbox |
| 22:03.00 | carrar | haha |
| 22:03.21 | carrar | maybe it's a cobalt cube |
| 22:03.37 | carrar | guess it wouldn't be running Ubuntu |
| 22:03.54 | russellb | ManxPower-work: 8080 maybe? |
| 22:04.02 | carrar | no |
| 22:04.05 | carrar | I tried that |
| 22:04.07 | ManxPower-work | russellb: tried. |
| 22:04.10 | russellb | 8000 ? |
| 22:04.14 | russellb | other than that, i dunno |
| 22:04.22 | carrar | 8675309 |
| 22:04.31 | drmessano | Port 9 doesn't work either, but I think i woke someone up |
| 22:04.38 | p3nguin | Should the Line LED be lit on the SPA-3102 when there is a phone wire running from the device to the wall jack? I know the wire is good, because I can receive PSTN calls on the phone connected to the device. |
| 22:04.42 | ManxPower-work | russellb: Customer IT guy is gone, nobody told us they had a frickin' McAfee web filter between our asterisk and the rest of the world. |
| 22:05.14 | carrar | thats good times |
| 22:05.15 | drmessano | p3nguin: No |
| 22:06.40 | carrar | have them upgrade ssh on that box too |
| 22:07.13 | carrar | it's nice that it's open to the world |
| 22:07.17 | geneticx | ManxPower-work: nice, Thanks. |
| 22:07.44 | carrar | Digium hiring a security person? |
| 22:09.21 | russellb | on what, downloads? |
| 22:09.26 | carrar | yeah |
| 22:09.37 | russellb | thanks for pointing that out in a public chat room, jerk |
| 22:09.38 | carrar | China owns you |
| 22:09.38 | russellb | :-p |
| 22:09.43 | *** kick/#asterisk [carrar!n=russellb@asterisk/digium-open-source-team-lead/russellb] by russellb (russellb) |
| 22:10.01 | ManxPower-work | how about hiring a person to make chan_iax2.so not automatically match the last entry in iax.conf for all unauthed incoming calls? |
| 22:10.18 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 22:10.21 | carrar | woah |
| 22:10.26 | carrar | kicked my russle |
| 22:10.28 | carrar | by |
| 22:10.31 | russellb | ;-) |
| 22:10.39 | carrar | heh |
| 22:12.01 | Jumpie | anybody know of a way i can quick glance the server health status of all my customers? |
| 22:12.09 | Jumpie | pinging an ip is obviously not good enough...needs to check sip |
| 22:12.16 | Jumpie | maybe even make some kinda 'status extension' |
| 22:12.17 | shido6 | what do u mean by "health" ? |
| 22:12.27 | carrar | sip show peers ? |
| 22:12.35 | Jumpie | no..hh |
| 22:12.37 | Jumpie | lol |
| 22:12.37 | carrar | with qualify turned on? |
| 22:12.39 | Jumpie | i meant |
| 22:12.45 | Jumpie | totaly different servers/companies |
| 22:12.52 | Jumpie | instead of having to log into each box, make test calls, etc |
| 22:13.02 | shido6 | not like MOSS |
| 22:13.04 | Jumpie | i wanna birds eye view of all my customers, and basically that the server is acting correctly |
| 22:13.06 | shido6 | mos scores? |
| 22:13.10 | shido6 | call quality? |
| 22:13.10 | carrar | write a NRPE module for nagios? |
| 22:13.21 | ManxPower-work | Jumpie: sipsak |
| 22:13.42 | Jumpie | interesting |
| 22:13.54 | Jumpie | i wonder if i need to create this test extension for every customer |
| 22:13.57 | Jumpie | im gonna look into that more |
| 22:14.04 | carrar | Jumpie, using NRPE you could fire off a script on each box and it would report back via NRPE to nagios |
| 22:14.09 | carrar | and nagios would alert you |
| 22:14.28 | carrar | could do it with SNMP too |
| 22:14.38 | Jumpie | wasnt sure off hand of a particular snmp string that'd show sip stats |
| 22:14.39 | Jumpie | :D |
| 22:14.42 | drmessano | SNMP is teh bomb |
| 22:15.03 | Jumpie | i have an hp openview cert...but itsl ike 6 years old..havent done alot of snmp stuff since like 03 |
| 22:15.04 | Jumpie | hehe |
| 22:15.21 | carrar | monitoring logs files helps also |
| 22:15.23 | Jumpie | carrar..i could somewhat keep it all grouped and deliver results to a web page |
| 22:15.24 | carrar | I do both |
| 22:15.25 | Jumpie | or custom app |
| 22:15.45 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 22:15.48 | carrar | just use existing monitoring tools |
| 22:15.49 | Jumpie | carrar i mean..i clearly know how to check each server..i was just hopin more for a very general view of just 'yes this pbx is passing calls, etc" for everybody |
| 22:16.13 | Jumpie | i've had a memory leak a few times |
| 22:16.13 | Jumpie | strange bug |
| 22:16.13 | Jumpie | outbound calls work..and internal calls work...you can log into the box |
| 22:16.14 | drmessano | Check out some of the GUIs.. Maybe you can rip out one of the status pages from FreePBX, trixbox, or the like.. |
| 22:16.17 | Jumpie | but will not route inbound |
| 22:16.20 | Jumpie | so a simple ping wouldnt work |
| 22:16.30 | Jumpie | drmessano..hmm thast cool too..i could then tweak what i wanna see |
| 22:16.36 | drmessano | Yep |
| 22:17.03 | carrar | Best fix that issue as thats not normal |
| 22:20.13 | *** join/#asterisk garymc (n=garymc@host86-165-105-126.range86-165.btcentralplus.com) |
| 22:25.59 | *** join/#asterisk Zerocross (n=AS@d51A4FAD6.access.telenet.be) |
| 22:26.07 | Zerocross | hi ppl |
| 22:27.10 | carrar | hi!! |
| 22:27.16 | Zerocross | Have a problem running my embedded asterisk box .. where shoud i place the register string to let my PBX handle incoming calls from the provider ? |
| 22:27.43 | Zerocross | normaly in sip.conf isn't it ? |
| 22:27.52 | carrar | if it's sip yes |
| 22:28.08 | Zerocross | doesn't work .. |
| 22:28.22 | Zerocross | tried already |
| 22:28.38 | *** join/#asterisk KingDavidNYC (n=Chris123@pool-96-224-162-219.nycmny.east.verizon.net) |
| 22:28.39 | ManxPower-work | in sip.conf and the end of [general] section. |
| 22:28.40 | [TK]D-Fender | Cause yeah... you can just shove parameters and directives anywhere you wnt in sip.conf.... |
| 22:28.45 | ManxPower-work | You do not put them in the [device] parts |
| 22:28.47 | KingDavidNYC | Hello Everybody!!! |
| 22:29.00 | carrar | heh |
| 22:29.14 | Zerocross | the strange thing is there is a users.conf file where all parameters are stored |
| 22:29.19 | *** join/#asterisk dysinger (n=tim@cpe-75-85-135-191.hawaii.res.rr.com) |
| 22:29.38 | bmoraca | ~users.conf |
| 22:29.39 | infobot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 22:29.54 | Zerocross | but tried there also .. my firewall is ok because tried with another (self made ast box and it worked) |
| 22:30.01 | ManxPower-work | bmoraca: and it doesn't even make toast! |
| 22:30.24 | ManxPower-work | Zerocross: for the most part you won't get much support for users.conf here. Very few people use it. |
| 22:30.28 | bmoraca | ManxPower-work: put it on a prescott-based Intel box and you could probably boil soup...maybe... |
| 22:30.34 | ManxPower-work | Lots of GUIs use it, but no people. |
| 22:30.53 | bmoraca | the only GUI that i know of that uses it is the one from AsteriskNOW |
| 22:31.00 | [TK]D-Fender | s/lots/one/ |
| 22:31.01 | carrar | biscotti-based Intel box? |
| 22:31.07 | ManxPower-work | bmoraca: I sit corrected. |
| 22:31.10 | carrar | mmm toast |
| 22:31.26 | Zerocross | well as i said before worked woth the complete selfmade PBX perfectly (under sip.conf ) .. etc |
| 22:31.57 | carrar | Sounds like it's just not possible then |
| 22:32.22 | Zerocross | no graphical / webinterface / gui sh*t |
| 22:33.06 | KingDavidNYC | Hello guys, I have a question: How come if I telnet into port 5060 the server rejects me, and however, when I use a softphone, I can connect?? |
| 22:33.07 | Zerocross | yah right ... a voip pbx without possibility for incoming calls ? |
| 22:33.35 | carrar | KingDavidNYC, because telnet is TCP |
| 22:33.36 | Zerocross | nah ... |
| 22:33.40 | carrar | voip is UDP |
| 22:33.48 | KingDavidNYC | wow |
| 22:33.57 | KingDavidNYC | awsome |
| 22:34.14 | KingDavidNYC | carrar: thank you |
| 22:34.24 | Zerocross | outgoing calls no problem |
| 22:34.34 | [jmc] | regarding the voicemail.conf file |
| 22:34.42 | Zerocross | just the registration for incoming calls |
| 22:35.03 | [jmc] | what is "pager_email"? if I want to send notifications to mobile phones or something like that? |
| 22:35.18 | carrar | sends to from length message via Email |
| 22:35.26 | carrar | like to a cell phone |
| 22:35.42 | Zerocross | unsing register=myID:myPassword@voipprovider ... this worked on the other box |
| 22:35.43 | [jmc] | ok, thanks |
| 22:35.48 | ManxPower-work | I should put a bounty on a feature for Asterisk that listens on 5060/TCP and when something connects justs prints some explaining message and disconnect. |
| 22:36.21 | ManxPower-work | [jmc]: pager_e-mail is a much shorter message and no attachments |
| 22:36.37 | [jmc] | good :) |
| 22:36.51 | [TK]D-Fender | ManxPower-work: Umm.. why? |
| 22:37.16 | ManxPower-work | [TK]D-Fender: one more common question eliminated |
| 22:37.30 | [TK]D-Fender | ManxPower-work: With a wrong answer... |
| 22:37.34 | carrar | then there would no use for the cluebat! |
| 22:37.48 | [TK]D-Fender | [18:33]<carrar>voip is UDP <- like this dangerously generalized answer |
| 22:37.54 | carrar | hahah |
| 22:37.55 | carrar | I know |
| 22:38.01 | ManxPower-work | [TK]D-Fender: "SIP uses UDP, not TCP. You can't use telnet for testing SIP." |
| 22:38.05 | [TK]D-Fender | I should swat you BOTH. |
| 22:38.20 | carrar | didn't think he was using tcp sip |
| 22:38.21 | [TK]D-Fender | [18:38]<ManxPower-work>[TK]D-Fender: "SIP uses UDP, not TCP. You can't use telnet for testing SIP." <- Inaccurate |
| 22:38.34 | [jmc] | why? |
| 22:38.49 | carrar | I made a calculated guess |
| 22:38.51 | carrar | and I was right |
| 22:38.54 | [jmc] | I thought most "media" streams used UDP only |
| 22:39.01 | ManxPower-work | [TK]D-Fender: Any why is that, Person That Doesn't Seem To Mind The Use Of The Term "SIP Trunk"? |
| 22:39.04 | carrar | going with that |
| 22:39.12 | [TK]D-Fender | [jmc]: SIP != media. Thats the point |
| 22:39.26 | carrar | but 5060 is SIP |
| 22:39.28 | [jmc] | hmm, right |
| 22:39.45 | [jmc] | but it should carry something :D |
| 22:39.54 | [jmc] | generally, media, am I wrong? |
| 22:39.59 | [jmc] | audio/video |
| 22:40.18 | ManxPower-work | SIP *can* use TCP, but nothing out there requires it except for the Microsoft SIP Server. |
| 22:40.19 | [TK]D-Fender | [jmc]: No, SIP is a call setup/teardown protocol |
| 22:40.36 | [jmc] | ok |
| 22:40.36 | [TK]D-Fender | ManxPower-work: I'm sure other apps and hardware require it... |
| 22:40.36 | ManxPower-work | SIP uses RTP for audio. |
| 22:40.44 | ManxPower-work | [TK]D-Fender: which ones? |
| 22:40.48 | [jmc] | right |
| 22:41.05 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:41.05 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 22:41.14 | [TK]D-Fender | ManxPower-work: My answer was abstract o the theory that MS isn't likely the ONLY one to require TCP for SIP |
| 22:41.48 | [TK]D-Fender | ManxPower-work: While I don't have a specific sample its more than reasonable to assume that some other authoer was equally as dumb :) |
| 22:41.55 | carrar | next MSFT will use TCP for the media |
| 22:41.59 | carrar | heh |
| 22:42.03 | ManxPower-work | SIP *can* use TCP, but nothing that's been released requires it except for the Microsoft SIP Server. |
| 22:42.23 | ManxPower-work | 8-) |
| 22:43.12 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 22:43.32 | [TK]D-Fender | ManxPower-work: That has come to our attention here anyway. |
| 22:44.42 | bmoraca | i think ResponsePoint uses TCP, but i could be wrong. also, CallManager can use SIP over TCP... |
| 22:44.53 | bmoraca | as can MAX TNTs... |
| 22:45.05 | bmoraca | of course, none of that is really useful in the scope of Asterisk...so the point is moot |
| 22:45.37 | [TK]D-Fender | bmoraca: The key factor we were referring to was the exclusive requirement of TCP SIP only and no UDP |
| 22:46.17 | *** join/#asterisk came0 (n=came0@webmail.flyairorlando.com) |
| 22:46.26 | bmoraca | as far as REQUIRING SIP over TCP...yes, I think only Microsoft's OCS was stupid enough to not include UDP |
| 22:47.27 | *** join/#asterisk jelly-bean1 (n=jelly@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 22:47.40 | jelly-bean1 | how do i configure the time on my polycom phones/ |
| 22:47.40 | carrar | MAX TNT heaters http://pics.osburn.com/photo/6437/original |
| 22:47.43 | carrar | heh |
| 22:48.51 | bmoraca | nice |
| 22:49.41 | carrar | jelly, tell it to sync to a SNTP server |
| 22:49.42 | [TK]D-Fender | jelly-bean1: lookup "sntp" in your sip.cfg |
| 22:50.04 | jelly-bean1 | [TK]D-Fender: found it. that was what we suspected. but which attribute then? shouldn't there be one for offset? |
| 22:50.11 | jelly-bean1 | and can we specify an NTP server to pull from? |
| 22:50.15 | [TK]D-Fender | jelly-bean1: There is |
| 22:50.34 | [TK]D-Fender | jelly-bean1: If the fields weren't obvious enough (they are), thats what the admin guide is for |
| 22:51.01 | jelly-bean1 | kthnx |
| 22:52.02 | jelly-bean1 | 337 pgs wow |
| 22:52.26 | jelly-bean1 | SNTP server address, SNTP GMT offset |
| 22:52.30 | [TK]D-Fender | Ctrl - F = 2 keys |
| 22:52.45 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 22:52.57 | p3nguin | Okay, using the SPA-3102, how can I get ALL calls to go out over the PSTN, except for calls dialed with # on the front (such as #500)? |
| 22:54.10 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 22:54.49 | *** join/#asterisk Micc (n=Micc@c-98-225-59-171.hsd1.wa.comcast.net) |
| 22:55.07 | Micc | Is there anything that will dial a cell phone and tell me if voicemail picked up? |
| 22:55.43 | carrar | ask the cell phone user to press a key |
| 22:56.20 | path | Dial(SIP/trunk/cellphone) |
| 22:56.27 | path | e.g. |
| 22:56.38 | shido6 | careful what you do with it, Micc - there is a hefty fine out there now |
| 22:57.20 | [TK]D-Fender | Micc: "core show application amd" |
| 22:58.21 | Micc | I think amd might work. |
| 23:05.02 | [jmc] | what does amd do? |
| 23:05.14 | Qwell | [jmc]: <[TK]D-Fender> Micc: "core show application amd" |
| 23:05.14 | carrar | reads peoples minds |
| 23:05.16 | [jmc] | let's check |
| 23:05.28 | [jmc] | Qwell, yes I was picking up the terminal :D |
| 23:06.36 | [jmc] | I don't have it O.o |
| 23:07.45 | *** join/#asterisk siregghead (n=siregghe@216.47.212.112) |
| 23:07.59 | siregghead | is there a log of dialed numbers in asterisk? |
| 23:08.16 | [TK]D-Fender | siregghead: lookup "CDR" in.... THE BOOK |
| 23:08.18 | [TK]D-Fender | ~book |
| 23:08.19 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 23:08.21 | [TK]D-Fender | ~cdr |
| 23:08.21 | infobot | i heard cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 23:08.28 | siregghead | thanks |
| 23:21.13 | [jmc] | hmm, one quick question |
| 23:21.23 | [jmc] | if I want to record a custom message |
| 23:21.40 | [jmc] | say I record it in a waveform file |
| 23:21.54 | [jmc] | and I want to use it in my Asterisk prompts |
| 23:22.10 | [jmc] | do I have to convert it into every format I'm using? |
| 23:22.33 | [jmc] | for example, mysound.gsm, mysound.ulaw, mysound.alaw, ... |
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| 23:28.56 | shido6 | no |
| 23:29.33 | shido6 | core show application Playback |
| 23:29.40 | shido6 | core show application Background |
| 23:34.19 | [jmc] | yes, I know |
| 23:34.25 | [jmc] | it can play different formats |
| 23:34.43 | [jmc] | but recording them in all the formats I'll enable in my channels |
| 23:34.53 | [jmc] | would consume less CPU time, right? |
| 23:37.46 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 23:39.16 | [jmc] | never mind |
| 23:39.27 | [jmc] | it's already too late, time to go to bed |
| 23:39.42 | [jmc] | goodbye, and thanks to you all for helping :) |
| 23:39.45 | leifmadsen | [jmc]: yes, it would |
| 23:39.51 | [jmc] | :) |
| 23:39.52 | leifmadsen | but you could use something like sox |
| 23:39.55 | [jmc] | good |
| 23:40.02 | leifmadsen | record to wav, then convert with a script |
| 23:40.13 | [jmc] | leifmadsen, that's exactly what I'm doing right now :D |
| 23:40.18 | leifmadsen | enjoy :) |
| 23:40.26 | [jmc] | :) |
| 23:40.28 | [jmc] | bye! |
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| 23:53.59 | Jumpie | heh |
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