10:16.19 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
10:16.19 | *** topic/#asterisk is Asterisk: The Open Source PBX || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 http://www.cluecon.com || Astricon 2005 in Anaheim Oct 12-14 http://www.astricon.net/2005/ - Speakers wanted |
10:18.02 | *** join/#asterisk dacleric (~dacleric@p5482B6D9.dip0.t-ipconnect.de) |
10:19.46 | emrah | Anyone here uses AreskiCC please? I just have a question about the AGI script. I'm having like an error when I launch the script. |
10:21.16 | florz | gres: try copying it to some other variable first, from the dialplan |
10:22.56 | emrah | Anyone use AreskiCC here? |
10:23.25 | *** join/#asterisk amir (~amir@195.226.9.186) |
10:23.39 | gres | florz, ok. thks |
10:30.18 | nDuff | emrah, maybe, maybe not -- but if you ask your question and wait around, you have a better chance of getting it answered than not asking until someone tells you they'll help. |
10:32.25 | Kraven | is there someone who can help me with bintec rcapi? |
10:35.12 | clive- | kraven what is that |
10:37.32 | Kraven | Capi over Ethernet |
10:40.56 | gres | florz, before execute my agi script, i do SetVariable(CALLDST=${EXTEN}). Then i recive in my script CALLDST! But i cat't still recive DIALSTATUS in my script after execute DIAL SIP/2760|30|tr... |
10:41.19 | ManxPower | You can't get automagically set variables in AGIs |
10:41.48 | ManxPower | Do a SetVar(MY_EXTEN=${EXTEN}) before calling your AGI and access ${MY_EXTEN}. |
10:42.37 | gres | ManxPower, I understand, thanks. |
10:42.39 | ManxPower | And if you do a Dial from within your AGI I know of no way to access things that are set by Dial, like DISALSTATUS. |
10:42.46 | ManxPower | That's why I NEVER EVER use Dial from inside an AGI |
10:43.20 | gres | :) |
10:44.22 | *** part/#asterisk fourcheeze (~rich@westbury.doilywood.org.uk) |
10:47.08 | *** join/#asterisk jonathh (~asd@host217-46-145-65.in-addr.btopenworld.com) |
10:54.35 | *** join/#asterisk d00gster (~doughant@toronto-HSE-ppp4334604.sympatico.ca) |
10:57.30 | tzafrir | emrah, maybe nobody, but ask your question anyway, for the record |
10:59.38 | Jas_Williams | Kraven: is this what you are talking about http://lists.digium.com/pipermail/asterisk-users/2005-June/112893.html |
11:01.49 | *** part/#asterisk d00gster (~doughant@toronto-HSE-ppp4334604.sympatico.ca) |
11:02.06 | joerg | any idea how I can detect wether the phone is picked up when I do auto-dial-out with asterisk? |
11:02.39 | joerg | I'm using sipgate.net for pstn connectivity |
11:02.59 | ManxPower | joerg: In what way? |
11:03.08 | joerg | http://www.voip-info.org/tiki-print.php?page=Asterisk+auto-dial+out |
11:03.11 | ManxPower | DIALSTATUS will give you that status of the Dial |
11:03.30 | joerg | it's exactly what is written at "please note" |
11:03.39 | ManxPower | This doesn't apply to dialing out via ANALOG FXO ports, of course. |
11:04.01 | ManxPower | joerg: you mean this " If you are using POTS (Plain Old Telephone System) lines attached to a channel ban....." |
11:04.07 | joerg | yes |
11:04.17 | joerg | I dial out via a sip to pots gateway |
11:04.17 | ManxPower | Are you using analog ports? |
11:04.42 | ManxPower | an ITSP like sipgate should be using PRI ports and so it's not an issue. |
11:08.32 | joerg | mhh... |
11:08.37 | joerg | so what can I do? :) |
11:08.45 | joerg | maybe iptables blocks sth? |
11:08.52 | ManxPower | joerg: nothing. You should be getting the correct DIALSTATUS back. |
11:09.27 | ManxPower | Unless there's a problem with .call files, which can be the case. |
11:09.49 | ManxPower | What IS DIALSTATUS set to when a call ends? |
11:11.04 | joerg | how do I check that? |
11:11.33 | ManxPower | in your dialplan. |
11:11.57 | ManxPower | You need to test this WITHOUT using .call files first. |
11:13.02 | pa | cant i have asterisk more verbose than vvv? |
11:13.08 | ManxPower | pa: yes |
11:13.25 | ManxPower | you can have up to 255 v's and up to 255 d's |
11:13.34 | pa | umh.. i used -vvvvvvvc but the error is the same |
11:13.41 | joerg | ManxPower: I have a dialplan for dialing out manually from my voip phone |
11:13.46 | joerg | ManxPower: that works very well |
11:14.00 | ManxPower | joerg: and what is DIALSTATUS when you do it that way? |
11:15.09 | ManxPower | You MAY have to use a Local/ channel in your .call file to make the call be processed via the Dialplan. Maybe not. |
11:16.14 | ManxPower | http://voip-info.org/tiki-index.php?page=Asterisk%20local%20channels |
11:17.11 | *** join/#asterisk gonzo- (~gonzo@gif.lesnik.portaone.com) |
11:25.21 | emrah | Please, anyone here uses AreskiCC |
11:26.29 | joerg | ManxPower: its the same problem |
11:26.44 | joerg | ManxPower: when I call my voip phone it works of course |
11:27.34 | *** join/#asterisk postel (~zz@postel.user) |
11:28.51 | *** join/#asterisk X-Rob (~rob@dsl-202-173-151-24.qld.westnet.com.au) |
11:29.16 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
11:29.32 | puzzled | morning |
11:31.50 | florz | puzzled: I guess any of the more recent ones should do, you just have to fix the zaptel include path version manually. |
11:32.25 | puzzled | florz: thanks :) |
11:35.51 | *** join/#asterisk loick (~loick@82.236.197.96) |
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11:38.54 | darkskiez | can I do integer math easily? |
11:40.38 | X-Rob | yes. 1+1 = 2 |
11:40.40 | X-Rob | that's pretty easy |
11:40.44 | darkskiez | in a dialplan |
11:41.03 | darkskiez | Math(var=1+1), results in var=2.00000 |
11:47.17 | darkskiez | so ? |
11:47.39 | ManxPower | darkskiez: try SetVar(VAR=$[1 + 1]) |
11:47.55 | darkskiez | oh ta, cheers |
11:47.59 | ManxPower | notice the spaces around + |
11:48.22 | darkskiez | and can local channels call an extension that matches a pattern, i'm getting errors that they cant be found |
11:48.36 | ManxPower | it should be able to |
11:48.45 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
11:52.31 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
11:53.07 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
11:55.43 | *** join/#asterisk tomtom_ (~root@83-217-70-161.reverse.realroot.be) |
11:57.15 | darkskiez | dont think it was liking alpha in the extension |
11:59.01 | ManxPower | Should not be a problem. |
11:59.10 | ManxPower | I use alpha extensions with Local/ and .call files. |
11:59.24 | ManxPower | Well, at least non-number extensions. |
11:59.25 | darkskiez | i just created an infinite loop, oopps |
11:59.34 | ManxPower | darkskiez: That would be a problem. 8-) |
12:01.10 | *** join/#asterisk fenlander (~neils@82.152.81.57) |
12:01.33 | darkskiez | 953 active channel(s) |
12:01.33 | darkskiez | 477 active call(s) |
12:01.34 | darkskiez | arse |
12:07.48 | *** join/#asterisk lathos42 (~lathos42@65-42-27-66.dowdingindustries.com) |
12:12.12 | *** join/#asterisk daork (~daork@don.bhnb.net.nz) |
12:12.36 | daork | can i force a digium FXO port to stay offhook? |
12:16.24 | daork | we've got 8 lines in a hunt group, and we only know where 7 of them physically are, so we're going to plug them all in, and get them offhook and then call the hunt group number |
12:17.40 | ManxPower | daork: Not really. |
12:18.08 | ManxPower | short each of the analog lines to make them busy |
12:18.37 | lathos42 | daork: Does the 8th line have its own phone number? I know our Analog lines all do. |
12:19.02 | ManxPower | You can, of course, also just dialout from each of the lines and call somthing that supports callerid |
12:19.08 | ManxPower | then you can find each line. |
12:20.07 | ManxPower | analog POTS lines always have their own phone number |
12:21.13 | *** join/#asterisk pnviking (~pnlarsson@c83-248-2-153.bredband.comhem.se) |
12:21.35 | *** join/#asterisk |nix (~inix@cm11.gamma116.maxonline.com.sg) |
12:25.36 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
12:25.44 | So3kris | knows someone if the x100p uses 2 drivers i for the voip and 1 for the modem |
12:27.15 | *** join/#asterisk QbY (~QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
12:27.57 | *** join/#asterisk virterm (~virterm@shiva.kanatek.com) |
12:28.13 | QbY | in what order does asterisk read extensions.conf? does it read extensions.conf and insert my includes into it in the order where they are 'included'? |
12:28.27 | QbY | or does it follow extensions.conf first, then look into the others? |
12:28.28 | X-Rob | QbY yes |
12:28.35 | X-Rob | order of inclusion |
12:28.41 | QbY | k |
12:29.29 | daork | ManxPower: actually, these dont. sorta |
12:29.43 | daork | they all do, but the missing line's numgber is the one that hunts across the tohers |
12:29.46 | daork | others* |
12:32.03 | *** join/#asterisk |dennis| (~dennis@200.32.215.82) |
12:32.17 | X-Rob | bah |
12:32.22 | X-Rob | cvs is down. *grump* |
12:34.19 | *** part/#asterisk daork (~daork@don.bhnb.net.nz) |
12:35.38 | QbY | http://pastebin.ca/18446 |
12:36.34 | QbY | I am trying to get all of the calls to this number into a special automated attendant (so the caller doesn't have to go through the first).. but all of the calls are being answered by the main attendant.. and its ignoring my extensions_custom --- the playback(abandon) was put in simply to test to see if its working.. http://pastebin.ca/18446 |
12:38.53 | X-Rob | Are you actually using AMP? |
12:39.02 | QbY | yes.. b |
12:39.04 | X-Rob | or is it just the remnanants of an AMP system? |
12:39.20 | QbY | more the remnants.. because i'm doing the custom stuff by hand |
12:39.32 | QbY | amp can't do (or i haven't found a way to) what i need done |
12:39.47 | X-Rob | well, if you were using AMP, you'd go to DID and set that number to go to the digitial rececptionist |
12:40.09 | *** part/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985436.sympatico.ca) |
12:40.10 | QbY | DID's won't do it for some reason.. |
12:40.15 | X-Rob | yes it will |
12:40.23 | X-Rob | that's what they're for. |
12:41.07 | QbY | son of a bitch |
12:41.18 | QbY | yesterday they wouldn't.. and some guy had me doing it this way |
12:41.36 | X-Rob | ask the guy doing the AMP documentation next time (eg, me 8) |
12:41.43 | QbY | hehe |
12:41.48 | X-Rob | and you're not even _on_ #amportal |
12:42.07 | QbY | figured it was more of an asterisk question |
12:42.32 | ManxPower | When using any WebGUI no question is an Asterisk question. |
12:43.18 | QbY | yes sir |
12:50.00 | *** join/#asterisk iCEBrkr (icebrkr@24.129.130.158) |
12:54.31 | *** join/#asterisk wigyori (wigyori@azigazi.hu) |
12:54.31 | *** join/#asterisk crash3m (crash3m@crash3m.user) [NETSPLIT VICTIM] |
12:54.31 | *** join/#asterisk Brumle (~brumle@brumle.com) [NETSPLIT VICTIM] |
12:54.31 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) [NETSPLIT VICTIM] |
12:54.31 | *** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM] |
12:54.33 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
12:55.04 | [TK]D-Fender | Just read some news on SineApps.... anyone got a decent link for info on the new AEL scripting feature comin up? |
12:57.45 | joerg | ManxPower: hey, are you there? :) |
13:04.15 | darkskiez | Yay, my cool recursive hunt macro script trick seems to work, huzzah |
13:06.16 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
13:08.32 | JunK-U | tk: REAME.ael ? |
13:12.48 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
13:14.45 | tuxinator_linuxM | watch out, its a staffer ;-) |
13:16.00 | [TK]D-Fender | JunK-U : Where would I find that? In a new download of CVS-HEAD? |
13:16.08 | *** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
13:16.21 | *** join/#asterisk |nix (~inix@cm11.gamma116.maxonline.com.sg) |
13:16.53 | *** part/#asterisk pooh_ (~pooh_@cust.15.241.adsl.cistron.nl) |
13:17.13 | JunK-U | ya |
13:18.40 | Darwin35 | what is asterisk? what does it do? what are the requirements to run it? where can I get it ? how much does it cost ? how much is support ? |
13:19.36 | ManxPower | [TK]D-Fender: Well you are not going to find AEL in 1.0.x |
13:20.13 | pnviking | A pbx, connection phones together, linux (win/mac), www.asterisk.org, 0$, How much do you want to pay? |
13:20.48 | [TK]D-Fender | ManxPower : Ok, where should I go for a heads-up on the new features since they don't seem to be in the Wiki so far? |
13:21.04 | ManxPower | [TK]D-Fender: README.upgrade? |
13:21.15 | ManxPower | and of course Changelog if it's been updated. |
13:21.23 | ManxPower | [TK]D-Fender: the correct place is the asterisk-cvs mailinglist. |
13:21.40 | ManxPower | Anyone using CVS-HEAD that is not on the asterisk-cvs mailing list is an idiot. |
13:22.16 | *** join/#asterisk lehel (~Lehel@82.79.20.17) |
13:22.24 | felipex | anybody using chan_bluetooth or miax ? |
13:23.30 | tzanger | miax? What's that? |
13:23.33 | *** join/#asterisk PCadach (~paul@212.19.157.154) |
13:23.39 | lehel | hello |
13:24.27 | |nix | i'm the biggest idiot |
13:24.27 | |nix | hahah |
13:25.56 | *** join/#asterisk Katty (~angela@68.112.15.110) |
13:26.18 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
13:26.29 | Hmmhesays | har har |
13:28.57 | Katty | mew |
13:29.05 | Darwin35 | my embedded board should be here today |
13:29.33 | puzzled | tzanger: http://www.voip-info.org/wiki-Mobile+IAX |
13:30.00 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
13:30.00 | *** mode/#asterisk [+o anthm] by ChanServ |
13:30.35 | *** join/#asterisk fitzel (~flint@p50861D74.dip0.t-ipconnect.de) |
13:30.41 | fitzel | Moi |
13:31.51 | Ahrimanes | mmmust.. hhhave.. caffeine..? |
13:32.19 | *** join/#asterisk _-Jon-_ (jon@CPE00112f6dfbee-CM00003989406c.cpe.net.cable.rogers.com) |
13:32.23 | _-Jon-_ | good morning everyone |
13:32.46 | Hmmhesays | BRAINS |
13:33.22 | CoaxD | BRAINS!@#$ |
13:33.24 | Hmmhesays | well, coffee will do I guess |
13:33.28 | Ahrimanes | hehe |
13:33.30 | _-Jon-_ | I'm wondering if anyone can assist me in linking 2 asterisk boxes together via iax. I'm getting confused with the config file between type=peer, type=friend, type=user, etc |
13:33.41 | iCEBrkr | CoaxD: Yea, you gotz none. |
13:33.44 | CoaxD | that was one fo the cheesiest movie series' i've ever seen |
13:33.44 | *** join/#asterisk loick (~loick@per92-7-82-236-197-96.fbx.proxad.net) |
13:33.51 | CoaxD | Icebrkr: not this morning I don't, you're absolutely right |
13:34.00 | iCEBrkr | CoaxD: hehe |
13:34.00 | Darwin35 | read asterisk/configs/sip..conf.sample |
13:34.17 | iCEBrkr | IAX != SIP |
13:34.18 | Darwin35 | it explains them |
13:34.27 | Darwin35 | its the same in both |
13:34.43 | _-Jon-_ | Alright I'll give that a read |
13:34.49 | fitzel | Anyone here with some practical experience for a softphone on a windows mobile PDA with wifi/wlan? |
13:34.58 | CoaxD | gawd, my data entry company that i work for - has a stupid server problem that isn't allowing me to download my batch headers. Can't do my work without it. I'm getting annoyed. |
13:35.46 | iCEBrkr | _-Jon-_: I used type=peer in my 'master' asterisk server in my iax.conf |
13:35.48 | CoaxD | ooh, i had the batch anyway. i can do my work now. woo! |
13:35.55 | Darwin35 | I hate mondays |
13:36.13 | fitzel | Darwin, they will pass, as every monday. |
13:36.57 | Darwin35 | could not get comfy in bed |
13:37.08 | Darwin35 | or on the couch or on the floor |
13:37.10 | iCEBrkr | Darwin35: Drink a few beers before bed :D |
13:37.17 | *** join/#asterisk tengulre (~tengulre@219.145.57.79) |
13:37.50 | Hmmhesays | :) morning Katty |
13:38.03 | fitzel | is there an iax-client available for windows mobile? or only sip-stuff? |
13:38.12 | Katty | Hmmhesays: i'm all groggy :< |
13:38.20 | Darwin35 | iaxcomm |
13:38.30 | Hmmhesays | Katty: why is that? |
13:38.40 | Katty | Hmmhesays: uhmm, cause my caffeine iv is not installed? |
13:39.03 | Hmmhesays | ahh I see, if it makes you feel any better I'm all sore |
13:39.18 | Katty | :< |
13:39.28 | Katty | i'm a little sore in sunburnt fashion |
13:39.43 | Hmmhesays | I'm a bit sunburnt and sore from wakeboarding |
13:39.50 | Katty | i was swimming |
13:39.59 | Katty | and dunking $person under teh water |
13:40.08 | Hmmhesays | I did a bit of that too |
13:40.14 | Hmmhesays | lake or pool? |
13:40.18 | Katty | pool |
13:40.24 | Katty | i don't swim well enough to be in a lake, i don't think |
13:40.40 | Darwin35 | firefly |
13:40.45 | Katty | not without another human floaty device |
13:40.45 | Darwin35 | does iax also |
13:41.03 | Hmmhesays | ahh I see |
13:41.06 | fitzel | iaxcomm on a pda? |
13:41.19 | Katty | iaxcomm isn't a j2me app |
13:41.26 | Katty | but i hope to write one (= |
13:41.58 | Katty | speaking of such, my new beep beep phone comes in today :> |
13:42.03 | Katty | twisted[asteria]: and then i shall bug you! |
13:42.05 | Katty | twisted[asteria]: BEEP BEEP |
13:42.31 | Katty | DarthClue: you'll be happy to know i'll set my little 128 pixel wallpaper to the cluecon logo (= |
13:42.46 | *** join/#asterisk astoria (~haydenth@66.235.201.217) |
13:42.47 | Katty | Hmmhesays: besides lack of coverage? ;) |
13:42.52 | astoria | Good Morning all. |
13:42.55 | Hmmhesays | haha nope |
13:43.07 | Katty | Hmmhesays: and the nextel phone support? |
13:43.17 | Hmmhesays | never talked to their support |
13:43.18 | Katty | Hmmhesays: and their insane notion you should pay for upload software? |
13:43.30 | Katty | Hmmhesays: whyfor grumbling? |
13:43.43 | Hmmhesays | cause I got no beep beep phone |
13:43.49 | Katty | aww. |
13:44.12 | Katty | is there nextel coverage out there? |
13:44.19 | Katty | or are you waiting for your contract to end? |
13:44.21 | fitzel | I am looking to get the ipaq H6340, is it worth it? Can it be used to phone via WLAN? |
13:44.32 | Hmmhesays | hrm, they do have coverage up here now |
13:44.39 | Hmmhesays | they didn't last time I looked |
13:44.45 | Katty | fitzel: you should probably visit the website the ipaq is posted on and read the specs :) |
13:44.51 | Katty | Hmmhesays: :> |
13:44.54 | astoria | Everytime I hear a nextel phone in a public place I want to vomit. |
13:45.01 | Katty | astoria: you would |
13:45.11 | Katty | astoria: lots of people don't turn the privacy setting on |
13:45.12 | fitzel | I know the specs and the website, but some first-hand experience is more interesting. |
13:45.23 | Katty | astoria: and so it blairs like a walky talky |
13:45.43 | Hmmhesays | heh, bad plans though geez |
13:45.52 | fitzel | it HAS a WLAN and there are some sip-clients available. But I am curious, how far the wlan can reach. |
13:45.55 | Katty | fitzel: you might try going to a retailer and asking (= |
13:46.11 | fitzel | I am living in countryside |
13:46.13 | Katty | k |
13:46.16 | mut | probly a few hundred feet like any other generic wlan device |
13:46.24 | astoria | I hear those damn things go off in class all the time.. It annoys me like no other. |
13:46.28 | pa | can i search somehow asterisk-users archive? |
13:46.32 | mut | like 100ft |
13:46.38 | astoria | pa: google site:digium.com |
13:46.59 | pa | oh, ok |
13:47.15 | Katty | astoria: it's not just nextel phones |
13:47.25 | Katty | astoria: there are all sorts of people who leave their phone on when it's not appropriate |
13:47.33 | Katty | astoria: and with all other providers, there is a vibrate option (= |
13:47.49 | astoria | Katty: i know, but nextel users are especially inconsiderate. |
13:48.06 | astoria | Katty: or i just recognize their ring more. |
13:48.19 | pa | here my problem: http://lists.digium.com/pipermail/asterisk-users/2005-July/118413.html |
13:49.29 | Katty | astoria: well, i'm quite considerate |
13:49.44 | *** join/#asterisk oej (~oej@apollo.webway.se) |
13:49.50 | Katty | astoria: i can't imagine why nextel people would be less considerate than other people |
13:50.01 | Katty | astoria: they're all the unwashed public, afterall |
13:50.11 | tuxinator_linuxM | astoria: I was in court when one of the jurors phones did that walkie talkie thing, dead silent afterwords |
13:50.52 | *** join/#asterisk zotz (~zotz@24.231.36.100) |
13:50.53 | tuxinator_linuxM | serious, whats worse, your phone ringing, or your phone talking really loud |
13:51.06 | *** join/#asterisk lters_ (~lters@eg1.ekn.com) |
13:51.14 | tuxinator_linuxM | lathos42: what * in the middle ;-) |
13:51.17 | Nugget | hopefully never. |
13:51.30 | Katty | pif: ..? |
13:51.52 | tuxinator_linuxM | may, it only takes a second to connect, how much faster do you need it? |
13:52.00 | Katty | lathos42: probably awhile yet, nextel is going to change their name to sprint i think |
13:52.05 | Katty | lathos42: and that hasn't happened yet |
13:52.05 | astoria | It's way worse when your sitting in class and you hear the nextel beeps followed by a, "HEY. YOU THERE??" |
13:52.16 | Katty | astoria: now /that/ is impolite |
13:52.28 | tuxinator_linuxM | I concure |
13:52.32 | Katty | astoria: i don't talk to someone if they don't answer their page |
13:52.34 | astoria | I've seen people answer int he middle of lectures! |
13:52.35 | tuxinator_linuxM | concure, he he |
13:52.49 | *** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
13:52.55 | Hmmhesays | its in my sight, and the timing is right, for taking a bow, into the now |
13:52.56 | lathos42 | astoria: I dont want a Nextel for exactly that reason.. I know alot of people that expect you to respond immediately if they direct connect you |
13:52.57 | tuxinator_linuxM | I think it is time for bed |
13:52.58 | astoria | ANd they'll just start talking... |
13:53.25 | tuxinator_linuxM | Hey tiger, come over to my cube.... |
13:53.37 | Katty | everyone has rude family members |
13:53.42 | Katty | my family has several |
13:53.48 | tzanger | Katty: :-) |
13:54.15 | Katty | tzanger: mew (= |
13:54.42 | tzanger | how are you Katty |
13:54.59 | pa | here this guy seems to have my same problem: |
13:55.01 | pa | http://lists.digium.com/pipermail/asterisk-users/2004-June/051684.html |
13:55.01 | lathos42 | Of course, then again, some of those same people call my phone over and over again until I answer or they get tired of hearing my voicemail greeting |
13:55.01 | Katty | tzanger: sunburnt and groggy |
13:55.07 | pa | no answers :( |
13:55.14 | tuxinator_linuxM | lathos42: You too ;-) |
13:55.20 | tzanger | Katty: that's not good |
13:55.28 | Katty | tzanger: i'll get over it (= |
13:56.15 | pa | umh.. isdn voice support into kernel? |
13:56.19 | pa | i didnt noticed it.. |
13:56.23 | pa | let me check |
13:56.44 | ManxPower | lathos42: nobody knows my cell phone number. They MUST call my extension and dial 0 if they get my voicemail if they have something urgent. |
13:57.02 | *** join/#asterisk willim_M (~icechat5@62.231.36.101) |
13:57.13 | pa | OK, prolly my fault |
13:57.19 | *** join/#asterisk skeffling (~Andrew_He@andrew.1ec.aaisp.net.uk) |
13:57.20 | pa | that should be the problem |
13:57.21 | ManxPower | If a call comes into my cell phone and I know it was a direct call I'll usually answer the call, scream "YOU HAVE THE WRONG NUMBER!" and then hangup. |
13:57.27 | Darwin35 | I love my dial the weather |
13:57.33 | willim_M | what ports do i need to open for iax |
13:57.44 | Darwin35 | look in the iax.conf |
13:57.47 | Darwin35 | it tells you |
13:57.55 | willim_M | thanks |
13:57.56 | astoria | read the wiki, there is a page about firewalls. |
13:58.04 | lathos42 | ManxPower: None of my coworkers have my cell phone number, and I plan to keep it that way |
13:58.07 | Katty | willim_M: i'll get you my port, hold on |
13:58.20 | astoria | Why are you afraid of co-workers calling your cell? |
13:58.25 | astoria | I have a direct extension for my cellphone. |
13:58.32 | astoria | I WANT them to call me. |
13:58.33 | Darwin35 | people read the wikie and the conf files before asking stupid questions that are already covered |
13:58.49 | Katty | willim_M: mines on 4569, and udp (= |
13:58.55 | skeffling | Hello, we've just starting seeing loads of these in the asterisk CLI/log file.... |
13:58.57 | skeffling | Jul 25 14:45:34 ERROR[14205]: utils.c:509 tvfix: warning negative timestamp -198466.-574000 |
13:58.58 | skeffling | and I've no idea what the problem is! |
13:59.07 | astoria | Katty: you're too nice. |
13:59.14 | Hmmhesays | woo hoo hoo |
13:59.19 | willim_M | i already opened port 4569 and it didnt work |
13:59.26 | Katty | willim_M: then it's not a port problem |
13:59.27 | doolph | anyone good with gnugk? |
13:59.33 | Katty | astoria: and you are too annoying |
13:59.38 | Katty | astoria: m'kay? |
13:59.38 | astoria | Katty: ha ha, am i? |
13:59.49 | *** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net) |
13:59.54 | yaaar | word |
13:59.55 | Katty | astoria: well nothing here would work if Hmmhesays always told me to go read the wiki (= |
14:00.11 | Hmmhesays | heh |
14:00.15 | astoria | Katty: i was just kidding, but i know what you mean. |
14:00.21 | Katty | k |
14:00.59 | Katty | anthm: which reminds me i need to redo my dial plan to define the callerid for 6 and 2 lines |
14:01.00 | lathos42 | Alot of my coworkers use their personal cell phones for business use, even though the company will happily buy you a cell phone if you need one |
14:01.12 | Katty | anthm: so maybe later you can look at it and tell me what i've goofed up after i give it a try (= |
14:01.20 | yaaar | Hmmhesays: hope you have your astrolabe |
14:01.24 | Katty | yay for anthm making my dial plan pretty |
14:01.29 | anthm | ok |
14:01.37 | astoria | lathos42: i prefer to use my personal cell phone because the amount of minutes that work uses up is much less than my personal minute use. |
14:02.04 | anthm | define the callerid on who? |
14:02.05 | astoria | lathos42: it's a hassle to deal with that kind of stuff if the office is paying for it. |
14:02.18 | Hmmhesays | this is like walking through a forest with a blindfold on |
14:02.22 | joerg | is anyone reachable via sip? |
14:02.27 | joerg | would like to try out sth |
14:02.47 | lathos42 | astoria: Yeah, my last job they just reimbursed me for part of my personal cell phone.. Which did have the advantage of not needing to carry two phones |
14:02.47 | joerg | my automated callback doesn't work |
14:04.13 | Hmmhesays | why not? |
14:04.16 | lathos42 | Hmmhesays: Thanks for giving me flashbacks to when I tried to find information on the Shiva LANRover on Intel's site.. I just now stopped having nightmares about it |
14:04.31 | Hmmhesays | lathos42: No Problem |
14:04.48 | lehel | phrrr.. i cannot make an IAX call.. friday i could, today not anymore ??? |
14:06.14 | lehel | Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL |
14:06.28 | joerg | Hmmhesays: are u talking to me? :) |
14:06.41 | Hmmhesays | yeah what is wrong with your callback deal? |
14:07.15 | joerg | there must be sth. wrong with sip |
14:07.38 | joerg | it starts the musiconhold app when I answer the phone |
14:07.44 | joerg | but I don't hear anything |
14:08.05 | Hmmhesays | are you using a callfile or the manager to originate the call |
14:08.18 | joerg | callfile |
14:08.34 | *** join/#asterisk kimosabe (~nat@dsl-200-67-12-220.prod-empresarial.com.mx) |
14:08.53 | joerg | the callfile calls Local/200@wakeup |
14:09.03 | joerg | and that calls a sip phone |
14:09.31 | Hmmhesays | are you in a nat environment? |
14:09.39 | joerg | nope |
14:09.52 | Hmmhesays | can you call into a moh extension with that sip phone? |
14:10.02 | joerg | the confusing thing is, that it works, when I initiate a call manually with my sip hone to outside |
14:10.44 | Hmmhesays | <Hmmhesays> can you call into a moh extension with that sip phone? |
14:11.06 | joerg | Hmmhesays: let me explain you my setup :) |
14:11.12 | Hmmhesays | no |
14:11.15 | Hmmhesays | just answer my question |
14:11.17 | joerg | ok |
14:11.18 | joerg | yes |
14:11.34 | joerg | the sip phone is actually a phone on pstn |
14:11.54 | Hmmhesays | same phone you are originating the call to? |
14:11.59 | astoria | Can SpanDSP do ECM? |
14:12.13 | joerg | yes |
14:12.21 | astoria | Where do I enable it? |
14:12.27 | joerg | sipgate.net acts as gateway |
14:12.29 | *** join/#asterisk brookshire (~matt@207.111.174.1) |
14:12.58 | joerg | from the phone, I can call my voip number and hear my asterisk playing the greeting |
14:13.04 | eldu | is there a way to force a codec per telco number within the same sip trunk ? |
14:13.23 | joerg | but the other way round it doesn't work |
14:14.14 | Hmmhesays | argh, "click here for a non commercial license" *click* rinse repeat |
14:14.33 | Hmmhesays | joerg: that is a problem |
14:14.40 | *** join/#asterisk Cadu20 (~Cadu20@200.102.53.174) |
14:14.56 | lathos42 | astoria: From what i'm seeing, I dont think it does |
14:15.03 | Cadu20 | How could I see the version/flavor of my G723 codecs? |
14:15.07 | joerg | Hmmhesays: placing calls from my voip phone via asterisk to the outisde world works. |
14:15.18 | ManxPower | Cadu20: you don't have G723.1 codecs |
14:15.19 | astoria | lathos42: thats too bad. Maybe soon, when they finish the T.30 stuff. |
14:15.22 | Cadu20 | Or how do I see more informations about a installed codec? |
14:15.30 | joerg | Hmmhesays: that is exactly the same thing the callback should do |
14:15.40 | ManxPower | Cadu20: "show codecs" and "show translations" |
14:15.49 | Hmmhesays | except you are originating the call from asterisk itself |
14:16.00 | *** join/#asterisk Goshen (~Goshen@67-40-107-29.slkc.qwest.net) |
14:16.30 | Cadu20 | ManxPower, but show codecs only gives me G723.1.. the gw operator tells me they got g723.1r6... I want to know if I´m compatible. |
14:16.35 | *** join/#asterisk jr352k ([U2FsdGVkX@pcp03618028pcs.univde01.de.comcast.net) |
14:16.50 | Darwin35 | then you did not set your codecs in the conf files |
14:16.55 | joerg | I have set verbose to 9 |
14:17.03 | joerg | it detects, when the line is answered |
14:17.04 | ManxPower | Cadu20: "show codecs" does not show the INSTALLED codecs, it just lists all codecs |
14:17.13 | joerg | and starts the moh app |
14:17.15 | Cadu20 | But i have installed G723. |
14:17.23 | *** join/#asterisk syle2 (~blah@wnpgmb06dc1-167-98.dynamic.mts.net) |
14:17.39 | Cadu20 | That no comercial license. |
14:17.43 | ManxPower | Cadu20: Don't tell us that. It's illegal and you are breaking patent and copyright law, as well as Intel's own license. |
14:17.54 | lathos42 | astoria: I really like the idea of SpanDSP, but from what i've heard, I dont dare try to sell it to my company as part of our Asterisk system |
14:18.06 | Cadu20 | Thats not what the folks told me. |
14:18.16 | astoria | lathos42: I'm going live with it right now on a real-life implementation.. It's actually not bad. It gets a bad rep. |
14:18.22 | Cadu20 | Tell you that I have installed a piece of software is ilegal too? |
14:18.40 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
14:18.50 | syle2 | 3rd rule of business: don't get caught |
14:18.51 | ManxPower | Cadu20: I can tell you that you can fly. That doesn't make it true. |
14:18.52 | astoria | lathos42: I wish it had ECM, because that really comes in handy, but otherwise, it's okay. |
14:19.09 | Cadu20 | Exactly |
14:19.17 | ManxPower | Cadu20: nobody here can help you with G723.1 |
14:19.19 | lathos42 | astoria: What kind of success rate are you seeing with it? |
14:19.22 | ManxPower | Since none of us use it. |
14:19.26 | Cadu20 | So.. how do I get more information about installed codecs? |
14:19.41 | ManxPower | Cadu20: The source code is the only real docs for htat. |
14:19.42 | ManxPower | that |
14:19.44 | astoria | lathos42: But the software is free, so I don't have a whole lot of room to complain. So far, about 80-90%. Way better than using a SIP adapter on a fax machine. |
14:19.47 | syle2 | unfortunately ignorance is not an excuse in the law hehe |
14:19.57 | ManxPower | Here is the licensing priceing info for G723.1 direct from the patent holder's web site: http://www.dspg.com/technology/LicensePricing.html |
14:20.15 | astoria | lathos42: sometimes, it just mangles a page here and there, but thats because of noise on the line and the lack of ECM. |
14:20.17 | Cadu20 | But they have download for the code? I didn´t found it. |
14:20.31 | astoria | lathos42: once they get ECM working okay, I would say it's ready for prime time. |
14:20.59 | Cadu20 | All right, i pay. |
14:21.04 | Cadu20 | This is NOT the problem. |
14:21.12 | Cadu20 | I JUST wanted to know, how the f* a |
14:21.21 | Cadu20 | II diplay more information about a codec? |
14:21.29 | lathos42 | astoria: That's cool.. I'd like to give everyone their own Fax DID, but I doubt the company will front the money for a T1 Fax Card and another server to run Hylafax on |
14:21.51 | astoria | lathos42: plus you gotta pay for a bunch of pots lines too, spandsp lets me run it through a PRI. |
14:22.14 | mishehu | pri pri pri |
14:22.18 | syle2 | when it says you pay per channel, this mean if you have a PRI line you pay for 23 lines |
14:22.30 | Cadu20 | ? No one? Codec information? |
14:22.37 | astoria | syle2: yeah, a PRI is 23 (technically 24) channels. |
14:22.38 | *** join/#asterisk teapot (~tandrews@mail.grok.org.za) |
14:22.49 | mishehu | Cadu20: http://www.voip-info.org ? |
14:22.50 | syle2 | yeah but last one is used for data |
14:22.51 | newl | Unless you're talking about an E1 PRI ;) |
14:22.56 | ManxPower | Cadu20: Um, the patent holders will refuse to license G723.1 to you unless you comit to a LARGE purchase. |
14:23.07 | astoria | syle2: the last one is used for signalling. Yeah, I'm talking about T1 PRI |
14:23.12 | lathos42 | astoria: I'm hoping to continue my sales pitch for Asterisk on our Company president now that she's back from Vacation |
14:23.15 | syle2 | how many in a E1 not familiar with europe, prob something like 27 hehe |
14:23.29 | Cadu20 | ManxPower, if you don´t want to help, ok. I take that. But please, DONT DISTORT WHAT IM SAYING. |
14:23.45 | astoria | lathos42: if you want to see how spandsp works on a fax, drop me a msg, and i'll give you my fax number and email you a copy of what it generates. |
14:23.46 | newl | 30 plus data and sync. |
14:23.48 | Cadu20 | By the way, thanks a lot. |
14:23.50 | Cadu20 | My god... |
14:24.15 | syle2 | anyone running SER? |
14:24.23 | mishehu | ManxPower: what's so great about g723.1 that the patent holders are so cocky? |
14:24.35 | ManxPower | Cadu20: Use G729 if you MUST use a patented codec. The G729 patent holders licensed it to Digium and Digium sells per channel licenses for G729 |
14:24.41 | ManxPower | mishehu: nothing 8-) |
14:24.44 | fearnor | patent holders are cocky, period |
14:24.50 | lathos42 | astoria: It might be interesting to see how it handles a fax from our crappy Laserjet 3100 :) |
14:25.04 | mishehu | I donno, the g729 holders are somewhat reasonable. |
14:25.08 | So3kris | gewoon die kernel qosq |
14:25.13 | pif | bic! |
14:25.19 | *** join/#asterisk grimse (~grimse@p5481D4AE.dip.t-dialin.net) |
14:25.36 | mishehu | Irq 233: no one cared! <--- that's the kernel panic of the day... |
14:25.39 | *** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com) |
14:25.40 | eKo1 | Why not use GSM or iLBC or some other non-patented codec. |
14:25.46 | Cadu20 | ManxPower, i GOT g729 |
14:25.46 | syle2 | i run only hp printers now, never a problem since, owww i hated brother, 5 colors drying up is expensive |
14:25.57 | Cadu20 | ManxPower, as I told, paying for it is NOT the problem |
14:26.29 | ManxPower | Cadu20: See http://lists.digium.com/pipermail/asterisk-users/2004-September/064099.html |
14:26.39 | ManxPower | Cadu20: There is no legal way to use G723.1 with Asterisk. |
14:26.47 | syle2 | does noone run SER at all? |
14:26.57 | fearnor | i run ser |
14:27.02 | syle2 | k |
14:27.03 | syle2 | PM |
14:27.30 | *** join/#asterisk Mike (~mike@201.135.48.172) |
14:27.35 | teapot | hohum |
14:27.54 | *** join/#asterisk BoNaDs (~theplot@ool-44c53748.dyn.optonline.net) |
14:28.13 | Mike | anyone having trouble with recent cvs and voicemail? |
14:28.23 | teapot | Is the bristuff ever going to make it into CVS ? |
14:28.31 | BoNaDs | to checkout cvs head, do i checkout -r vCVS-HEAD? |
14:28.31 | *** join/#asterisk coppice (~chatzilla@30.195.17.210.dyn.pacific.net.hk) |
14:28.53 | Mike | hey steve |
14:29.02 | ManxPower | This kind of sucks. Digium won't replace an old TDM400P (the one without the power connector) if it's older than 2 years. |
14:29.23 | ManxPower | teapot: BRIStuff will never be in Asterisk |
14:29.31 | BoNaDs | what is the power connector for? |
14:29.36 | teapot | huh ? |
14:29.40 | BoNaDs | always wondered why it had one |
14:29.43 | ManxPower | teapot: BRIStuff will never be in Asterisk |
14:29.45 | teapot | why is that ManxPower ? |
14:30.01 | ManxPower | BoNaDs: because many motherboards cannot supply enough voltage over the PCI bus to make the card work |
14:30.02 | BoNaDs | doesnt the board power from the pci bus? |
14:30.02 | Cadu20 | Ok, thank folks. |
14:30.17 | BoNaDs | ah so if i power the board while on the bus it wont hurt anything? |
14:30.20 | ManxPower | teapot: because BRIstuff contains stuff from people that will not disclaim the code to Digium. |
14:30.32 | teapot | pah! |
14:30.35 | ManxPower | BoNaDs: Well it wont work if you have FXS modules, that's all. |
14:30.48 | BoNaDs | but i wont fry the board if i plug the power into it? |
14:31.10 | ManxPower | BoNaDs: Huh? If you don't plug power into the board the FXS modules won't work. |
14:31.33 | BoNaDs | right which i have none |
14:31.35 | BoNaDs | only fxo |
14:31.52 | fearnor | manx: have you played with bristuff? |
14:31.52 | ManxPower | on boards without a power connector the FXS ports may or maynot work depending on your motherboard. Mostly they will work for a while then fail. |
14:31.58 | ManxPower | BoNaDs: then it should not matter. |
14:31.59 | BoNaDs | but i am having some wierd shit which i think would make sense to be relared to being underpowered |
14:32.04 | ManxPower | fearnor: since I don't have a PRI....no. |
14:32.12 | BoNaDs | gonna give it a shot |
14:32.15 | BoNaDs | brb |
14:32.27 | ManxPower | BoNaDs: just be sure to do it when the system is powered off. |
14:32.45 | fearnor | i'm thinking that doing BRI to the isdn phones seems to be a very cheap solution for non-ghetto phones |
14:32.46 | ManxPower | fearnor: um...since I don't have a BRI...no. |
14:32.59 | *** join/#asterisk mkrufky (~mk@68.160.103.77) |
14:33.08 | Katty | i sure could use a new recipe |
14:33.16 | Katty | i don't suppose anyone would like to volunteer their favorite |
14:33.19 | *** join/#asterisk emp (~emp@70.57.239.37) |
14:33.44 | ManxPower | Katty: procmail? |
14:34.10 | Katty | ManxPower: you have a recipe for procmail? :P |
14:34.25 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
14:34.26 | Katty | ManxPower: i was thinking pasta, actually (= |
14:34.52 | Ariel_ | morning everyone |
14:35.01 | Ariel_ | hello Katty |
14:35.10 | Katty | Ariel_: moo'rning (= |
14:35.19 | BoNaDs | being that i am cvs-dumbass, how do i get CVS-HEAD rather than v1-0 as shown on the website instructions |
14:35.27 | ManxPower | Katty: My best source for pasta is Walmart 8-) |
14:35.28 | BoNaDs | vCVS-HEAD? |
14:35.36 | Katty | ManxPower: i see. |
14:35.52 | Katty | Ariel_: recipe? |
14:35.59 | ManxPower | BoNaDs: don't specify -r v1-0 on your CVS command line |
14:36.03 | BoNaDs | ah |
14:36.13 | BoNaDs | danke |
14:36.16 | mishehu | I much prefer maildrop over procmail |
14:36.17 | *** join/#asterisk Malthus (~admin@port0043-aas-adsl.cwjamaica.com) |
14:36.23 | *** part/#asterisk Malthus (~admin@port0043-aas-adsl.cwjamaica.com) |
14:37.01 | Katty | maildrop cookies! |
14:37.22 | Katty | there should /so/ be an asterisk cook book in engineering layout |
14:43.31 | *** join/#asterisk Blackvel (~blackvel@dsl-084-057-124-066.arcor-ip.net) |
14:47.03 | syle2 | crap whats LCR stand for again |
14:47.21 | skeffling | Least cost Routing |
14:47.48 | *** join/#asterisk thieums (~darkmind@bea75-1-82-234-122-35.fbx.proxad.net) |
14:48.00 | coppice | Inductor, capacitor, resistor |
14:48.24 | thieums | do you know where I can find a running oh323 modules for 1.0.9 ? |
14:48.48 | *** join/#asterisk darkskiez (~darkskiez@194.247.78.146) |
14:48.54 | *** join/#asterisk Defraz (~t0tal@67.130.216.13) |
14:48.55 | Hmmhesays | h323 ouch |
14:49.03 | thieums | yes I know |
14:49.06 | thieums | but i need it |
14:49.14 | Hmmhesays | all of the versions of oh323 or on the inaccessnetworks site |
14:49.19 | Hmmhesays | *are on even |
14:49.26 | thieums | fu**ing customers |
14:49.37 | *** join/#asterisk greg_work (~greg@d221-73-237.commercial.cgocable.net) |
14:49.47 | darkskiez | ManxPower: you were wondering about the recursive script I was writing : http://voip-info.org/tiki-index.php?page=Asterisk+power+hunt |
14:49.52 | Hmmhesays | you might try ooh323 with cvs-head |
14:49.59 | Hmmhesays | it seems to work alright |
14:50.00 | *** join/#asterisk ChkDigit (~mike@static65-87-228-18.regina.accesscomm.ca) |
14:50.06 | izo | anybody has 7940 or 7960 |
14:50.07 | thieums | ok I'll try it |
14:50.08 | Hmmhesays | not dependant on openh323 and pwlib |
14:50.08 | ManxPower | darkskiez: I've written recursive dialplan macros before 8-) |
14:50.10 | thieums | thanks |
14:50.45 | darkskiez | ManxPower: oh? perhaps recursive is the wrong word for this, multithreaded? :) |
14:51.06 | ManxPower | darkskiez: My macro calls itself when needed. |
14:51.26 | darkskiez | ManxPower: whats it do? |
14:51.48 | *** join/#asterisk SwK[Work] (~SwK@border0hsv.asterisksgi.com) |
14:51.50 | SwK[Work] | j #redbull |
14:51.59 | anthm | hmm, nobody every mentions chan_woomera =D |
14:52.08 | darkskiez | woom what? |
14:52.11 | *** join/#asterisk file[laptop] (~file[lapt@mctnnbsah25-142166093154.nb.aliant.net) |
14:52.17 | SwK[Work] | *yawn8 |
14:52.17 | *** join/#asterisk pussfeller (~todd@216.223.173.189) |
14:52.41 | coppice | i saw someone mention chan_woomera just a few lines ago |
14:53.22 | astoria | What is chan_woomera? |
14:53.28 | ManxPower | coppice: I have the same issues with chan_woomera as I do with chan_oh323. I.e. I feel that the basic design is flawed. |
14:53.45 | ManxPower | astoria: Yet Another Asterisk H323 Driver. |
14:53.49 | astoria | coppice: You said the other day that you have thousands of users using SpanDSP? Doesn't the lack of ECM cause some problems? |
14:54.01 | astoria | ManxPower: thanks, i'm a h323 virgin thankfully. |
14:54.08 | anthm | ooh neet, can I hear how the design is flawed ? |
14:54.08 | *** part/#asterisk QbY (~QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
14:54.23 | syle2 | how do PRI's work for calling numbers, lets say you want to forward a DID to a sip server...do you do exten => 2223334444,1, blah as well as exten => 3334444 , for local calls as well as 12223334444 for long distance calls or does the PRI recognise all calls as one format no matter how it is dialed? |
14:54.45 | ManxPower | anthm: by handleing all the RTP stuff inside the channel instead of letting Asterisk's RTP stack handle it. |
14:55.09 | izo | anthm: chan woomera support VAD by any chance ? |
14:55.13 | astoria | syle2: i'm not sure what you're saying exactly. ALl your calls are on zap channels. |
14:55.18 | anthm | hmm you ever hear that bit about assume |
14:55.48 | syle2 | yes but how do you forward just a specific DID? |
14:55.54 | anthm | chan_woomera is only 20k compiled as a matter of fact so I bet there is no rtp stack |
14:56.07 | syle2 | as far as i know the DID can be on any of the 23 channels |
14:56.07 | astoria | syle2: you look at the dnis data ( the called id ) |
14:56.11 | ManxPower | syle2: DIDs come into Asterisk as extensions |
14:56.17 | astoria | syle2: your provider will forward the dialed numbers to you |
14:56.22 | coppice | astoria: why should the lack of ECM be a problem. Most FAX machines lack ECM |
14:56.23 | ManxPower | anthm: at 20k it could not even handle H323 by itself. |
14:56.31 | anthm | ding ding |
14:56.59 | syle2 | ok then read back what i wrote above |
14:57.03 | anthm | chan_woomera is a thin client to an external process where you runn the voip engine in it's own process on any platform you choose |
14:57.14 | anthm | an another box if you wish |
14:57.18 | izo | hey coppice i started using your spandsp last week really cool stuff !!! |
14:57.31 | ManxPower | syle2: It all depends on how the telco delivers the dialed number. 3 digits? 4 digits? 10 digits? |
14:57.35 | astoria | coppice: ECM is a nice thing to have, most people I've dealt with have ECM lines. |
14:57.43 | anthm | and asterisk only blisfully sees a natice slin audio channel and has no worries about integration |
14:57.46 | astoria | coppice: You have a new spandsp? yay! |
14:57.47 | ManxPower | anthm: And chan_woomera communicated with the rest of Asterisk using SLIN. |
14:58.09 | ManxPower | Still seems rather limited. More like chan_woomera is really a channel to access external woomera gateways. |
14:58.29 | yaaar | hey guys, i'm trying to setup cdr_mysql right now, and when i reload asterisk i get an error saying 'could not connect to mysql database asterisk on localhost.' ...but i can connect and view the tables in that database just fine with the username and password from cdr_mysql.conf. |
14:58.38 | yaaar | anybody know what i might have screwed up? |
14:58.44 | syle2 | manxpower: is it possible they could use more than 1 format? |
14:58.55 | ManxPower | I can see the advantages of pretty much ripping all audio processing out of Asterisk, but I still think its a flawed design |
14:58.55 | syle2 | for longdistance, local etc |
14:58.57 | *** join/#asterisk [TK]D-Fender (~joe@216.252.67.4) |
14:59.09 | anthm | well since woomera will soon support sip iax and h323 you will soon get 3 protocols to work on 1 tiny simplistic channel driver |
14:59.10 | izo | yaar: you either use local or TCP/IP socket |
14:59.16 | coppice | ECM shouldn't be that important these days. even without it most FAXes are clean. it was pretty important before the mass deployment of digital exchanges, but just as they got ECM speced that mass deployment ook place :-) |
14:59.20 | ManxPower | syle2: that would be unusual in my experience, but I've only dealt with a telco that ASKS us what format we want the dialed number to arrive in. |
14:59.26 | anthm | and btw, I also wrote the skeleton of the objective sys one |
14:59.39 | fearnor | ecm doesn't help against frame slips, and without frame slips, there's no problem ;) |
14:59.49 | anthm | so i am not pluging anything I just want ppl to test it |
14:59.51 | yaaar | izo: not sure i follow.... |
14:59.56 | astoria | coppice: cool, i've got some confidence in spandsp now. I just went live with a deployment this morning! |
15:00.18 | ManxPower | I still need to "backport" spanDSP to 1.0.x |
15:00.21 | *** join/#asterisk illek (~mike@ip68-227-104-152.ok.ok.cox.net) |
15:00.37 | ManxPower | last I saw it was a few changes to CLID handleing and that was it. |
15:00.51 | coppice | ManxPower: a man too mean to even *know* about non-free codecs :-) |
15:00.52 | ManxPower | anthm: so what would be the point of using Asterisk then? |
15:00.53 | izo | yaaar: you can connect to mysql on two ways via local socket /var/run/mysqld/mysqld.sock or port 3306 they both hmight have different permissions |
15:01.11 | ManxPower | coppice: Yeah! Viva la Free Codecs! |
15:01.12 | coppice | ManxPower: why would spandsp need backporting? |
15:01.28 | ManxPower | coppice: CVS-HEAD's callerid structures are slightly different than 1.0.x's |
15:01.29 | *** join/#asterisk denisgalvao (~Denis@200.146.0.254) |
15:01.34 | anthm | for all the asteriskness |
15:01.37 | denisgalvao | Hi all. |
15:01.50 | izo | ManxPower: i belive copppice has two versions on his FTP |
15:01.54 | denisgalvao | I need some help on Asterisk + Unicall.... |
15:01.58 | izo | ManxPower: for both 1.0.x and 1.1x |
15:02.02 | anthm | providing native interface to voip protocols is not asterisk's strongest offering |
15:02.11 | ManxPower | coppice: I of course mean "SpanDSP+rxfax+txfax" when I say "SpanDSP". |
15:02.16 | yaaar | izo: it gives me the same error either way. |
15:02.21 | fitzel | Anyone here with some practical experience for a softphone on a PDA with wifi/wlan in every-day use? |
15:02.40 | izo | yaaar : well then you have something wrong with your mysql setup |
15:03.02 | ManxPower | izo: *nod* The old "not compatable with Brother or Cannon fax machines" version for 1.0.x and the "works with prectically all fax machines" for CVS-HEAD. 8-) Unless he released a 1.0.x version with his updates in the past couple of months. |
15:03.25 | izo | yaaar : rty connecting frmo command line like mysql -u user -h localhost -p database |
15:03.33 | anthm | with the example of h323, most channel drivers based on openh323 have issues cos the threading model is not compatable so by letting the h323 run in it's own process the way it wants it makes compatability easier |
15:03.34 | izo | ManxPower : really ? |
15:03.38 | ManxPower | Hmm...I'll bet I can use the new SpanDSP with the old rxfax/txfax.... |
15:03.47 | yaaar | izo: yeah like i said connecting with mysql -u asteriskuser -p works just fine |
15:03.49 | *** join/#asterisk montag___ (~montag@host187-252.pool8175.interbusiness.it) |
15:03.49 | ManxPower | izo: what version are you running with 1.0.9? |
15:04.01 | izo | yaaar : notice -h parameter for localhost |
15:04.15 | izo | ManxPower yep |
15:04.24 | *** join/#asterisk joerg (~joerg@cl-666.ham-01.de.sixxs.net) |
15:04.46 | montag___ | hi, when a call a sip extension with a .call file asterisk don't wait that remote extensions answer, but forward immediately the call...only with sip....with IAX or ZAP all it's working....any tip ? |
15:04.59 | coppice | ManxPower: what's wrong with the latest rxfax and txfax? |
15:05.12 | *** join/#asterisk kshumard (~kenny_@207.111.174.1) |
15:05.19 | yaaar | izo: yeah, still works fine with 'mysql -u asteriskuser -h localhost -p asteriskcdrdb' |
15:05.20 | ManxPower | coppice: give me 10 mins and I'll tell you. What's the URL of a working location to get the stuff? |
15:05.59 | ManxPower | coppice: I've not bothered to report it since I pretty much get shot down anytime I report any problem with 1.0.x |
15:06.03 | ManxPower | to ANYONE. |
15:06.45 | izo | yaaar : how do you specify mysql database in cdr_mysql.conf ? |
15:06.57 | anthm | the whole notion of 1.0.x to begin with is a problem |
15:07.02 | coppice | I only use 1.0.x. support for 1.1.x in my software is a bit weak. they kept breaking things, so I decided to wait until the dust settles. 1.2.5 sounds a likely time for that :-) |
15:07.08 | yaaar | izo: dbname=asteriskcdrdb |
15:07.09 | ManxPower | See what I mean? |
15:07.27 | izo | yaaar : enter whoe thing into pastebin.com |
15:07.34 | yaaar | k |
15:07.51 | ManxPower | coppice: you restructured your site! |
15:07.59 | coppice | no |
15:08.17 | ManxPower | coppice: last time I looked there was NO 1.0.x directory for rxfax/txfax |
15:08.47 | denisgalvao | coppice: Could you help on an Unicall related problem? |
15:08.50 | *** join/#asterisk _DAW (daw@67.128.57.2) |
15:08.56 | yaaar | izo: http://pastebin.com/320288 |
15:09.01 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
15:09.14 | coppice | drwxr-sr-x 2 root root 4096 Jun 12 21:29 asterisk-1.0.x |
15:09.16 | coppice | drwxr-sr-x 2 root root 4096 Jun 12 21:32 asterisk-1.1.x |
15:09.18 | coppice | -rw-r--r-- 1 root root 1445 May 11 22:31 README |
15:09.19 | coppice | -rw-rw-r-- 1 root root 1288267 May 11 22:25 spandsp-0.0.2pre18.tar.gz |
15:09.20 | ManxPower | coppice: But CVS-HEAD is perfect! It works! It's more stable than 1.0.x! |
15:09.34 | yaaar | izo: hehe |
15:09.38 | yaaar | izo: never mind |
15:09.52 | coppice | so why are you interested in 1.0.x? |
15:09.52 | astoria | coppice: where can I obtain these spandsp+rxfax+txfax changes you've made? |
15:09.52 | izo | yeah |
15:10.01 | ManxPower | ftp://ftp.soft-switch.org/pub/spandsp/spandsp-0.0.2pre18/ |
15:10.04 | astoria | Thanks! |
15:10.11 | yaaar | izo: only when i typed it into the pastebin did i go 'gee, wonder if those single quotes really belong there' |
15:10.23 | izo | yaaar :-) |
15:10.44 | ManxPower | coppice: I run 1.0.x |
15:10.56 | ManxPower | I'm repeating the drivel everyone else seems to be saying about CVS-HEAD |
15:11.02 | coppice | I think the 1.0.x and 1.1.x stuff for fax is OK. There are some issues with the 1.1.x stuff for unicall |
15:11.16 | *** join/#asterisk |dennis| (~dennis@200.32.215.82) |
15:11.22 | anthm | asterisk is way too unstable as a whole to be developing sects towards imaginary version neumbers |
15:11.41 | denisgalvao | coppice: Yes. Im runnig 1.1.x rxfax and txfax without prpoblem, but Unicall... |
15:11.47 | bkw_ | ManxPower, drivel? |
15:12.15 | bkw_ | why 1.0.x days are numbered in the double to single digits |
15:12.19 | ManxPower | The award says "Against all odds, still a friend of 1.0.x" |
15:12.20 | izo | ManxPower : whats the award/prize meadl or something or just handshake ? |
15:13.03 | ManxPower | izo: a small sculpture made from old Digium USB FXS adapters. |
15:13.17 | bkw_ | CVS-Stable is not the answer! |
15:13.40 | ManxPower | Everyone knows what MY idea of "the answer" is. |
15:13.43 | anthm | it's really rather silly to argue about the code from the outside perspective =D |
15:14.16 | Katty | less arguing, more hugging. |
15:14.19 | anthm | especially since it's the same code to start with |
15:14.28 | MikeJ[Laptop] | sigh... |
15:14.32 | coppice | well if asterisk of any sort is the answer, it was a bloody strange question |
15:14.41 | ManxPower | coppice: LOL! |
15:14.44 | MikeJ[Laptop] | bitch less, work more... |
15:14.47 | MikeJ[Laptop] | :P |
15:15.01 | ManxPower | coppice: keep the award, it's already a collectors item. |
15:15.18 | bkw_ | ManxPower, you the award fairy? |
15:15.30 | anthm | the number of bytes typed bitching about stuff probably shadows the asterisk code base on a daily basis |
15:15.41 | izo | :-) |
15:15.41 | bkw_ | hahahahahahahahah |
15:16.12 | coppice | well, there are far more people moaning than actually capable of doing anything useful |
15:16.26 | izo | that is actually true |
15:16.34 | bkw_ | ya think? |
15:16.43 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
15:17.03 | izo | and there is plenty ppl doing stuff for commercial use not bothering to participate |
15:17.04 | anthm | imagine if everyone typed thier rants into articles and migrated them into docs |
15:17.54 | MikeJ[Laptop] | heh.. anthm, yeah right.. like thats going to happen |
15:18.03 | *** join/#asterisk ArkyLady (ArkyLady@adsl-67-64-6-10.dsl.ltrkar.swbell.net) |
15:18.12 | Katty | anthm: i'd explode. |
15:18.42 | Hmmhesays | heh, sounds like fun |
15:18.53 | coppice | * needs VAD pretty badly. I wish I had the time to do the DSP bit, then I'm sure Steve Kann would get the rest done. |
15:18.55 | MikeJ[Laptop] | bitch less, work more... that means you ManxPower, you need a cvshead test box... |
15:19.16 | anthm | the truth is, most people who get to stable follow the same path of "I wish I could ____" |
15:19.27 | *** join/#asterisk switch (~switch@61.206.115.5) |
15:19.36 | Katty | twisted[asteria]: beep beep? |
15:19.39 | ManxPower | MikeJ[Laptop]: I need many things. |
15:19.46 | twisted[asteria] | Katty, meh... let me wake up |
15:19.50 | anthm | and that path has already been carved by the ppl who went and added that same functionality |
15:19.53 | Katty | twisted[asteria]: oh, i wasn't going to |
15:19.57 | twisted[asteria] | Katty, oh ;P |
15:19.58 | Katty | twisted[asteria]: was just saying hi (= |
15:19.59 | anthm | hence CVS head |
15:20.01 | twisted[asteria] | Katty, hi :) |
15:20.06 | ManxPower | Maybe I'll set up two CVs-HEAD boxes when I get my two TE110Ps in for testing. |
15:20.09 | *** join/#asterisk fugitivo (~ajf@201.255.104.144) |
15:20.10 | fugitivo | hello |
15:20.18 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-80-62-239.red.bezeqint.net) |
15:20.24 | astoria | ManxPower: bling bling |
15:20.31 | anthm | so when a guy says "I want foo" and someone says get CVS HEAD it |
15:20.41 | ManxPower | anthm: I do that sometimes. |
15:20.43 | PoWeRKiLL | Hi |
15:20.50 | anthm | is because someone already added that and it's the only way to get it |
15:20.59 | PoWeRKiLL | Someone have an idea how to log cdr to 2 mysql server ? |
15:21.00 | anthm | unless you backport |
15:21.14 | ManxPower | I've paid for at least 1 packport 8-) |
15:21.19 | twisted[asteria] | PoWeRKiLL, uhm, that's an oldie, and has quite a few steps involved... i reccommend the wiki |
15:21.26 | Hmmhesays | wiki wiki asterisk-addons wiki wiki |
15:21.29 | anthm | and if you backport everything you like in HEAD to stable you get a clone of HEAD |
15:21.33 | *** join/#asterisk jsharp (~jsharp@65.88.255.132) |
15:21.36 | twisted[asteria] | anthm, yup |
15:21.49 | twisted[asteria] | anthm, how you doin btw? |
15:21.51 | ManxPower | Except it would not change every few days. |
15:21.57 | anthm | which proves that stable is indeed not the answer |
15:22.01 | twisted[asteria] | ManxPower, s/days/hours |
15:22.07 | anthm | that is my theorm |
15:22.15 | anthm | twisted, not bad you? |
15:22.29 | jsharp | Odd SIP problem. When I call into Asterisk 1.0.7 with an ATA-186, I get a fast busy on the ATA and this on the Asterisk console: chan_sip.c:7192 handle_request: Ignoring too old packet packet 1 (expecting >= 2) |
15:22.46 | twisted[asteria] | anthm, not bad, staying busy |
15:22.51 | ManxPower | jsharp: any reason you are not using 1.0.9? |
15:23.01 | jsharp | Other than I haven't downloaded and installed it, no. |
15:23.08 | twisted[asteria] | PoWeRKiLL, DO NOT /msg W/o ASKING. that's bad form. I gave you what help i'm going to give |
15:23.13 | twisted[asteria] | PoWeRKiLL, you /msg me again, i will kick you. |
15:23.22 | ManxPower | jsharp: I don't recall any specific change that woule fix that, but..... |
15:23.33 | anthm | twisted, did you see app_rss I thought you'd like it being a cepstral fan |
15:23.45 | lehel | tcpdump -i any port 4569 < this is correct? |
15:23.51 | jsharp | I am running it across a 600ms satellite link, so I figure that probably has something to do with it. |
15:23.55 | *** join/#asterisk _omer (dfsdf@203.215.180.254) |
15:23.55 | twisted[asteria] | anthm, no, i have not yet... is it on pbxfreeware? |
15:24.02 | PoWeRKiLL | twisted sorry |
15:24.13 | ManxPower | anthm: I do admit to drooling over AEL. 8-) |
15:24.23 | tzafrir | lehel, does that dump both udp and tcp? |
15:24.25 | twisted[asteria] | PoWeRKiLL, s'ok. that's why I warned rather than immediately taking action |
15:24.32 | anthm | yah |
15:24.37 | sudhir492 | anyone using x-lite with asterisk |
15:24.39 | coppice | ManxPower: I drool over head :-) |
15:24.51 | lehel | tzafrir: yes.. both, but i got no response |
15:24.57 | tzafrir | lehel, I also tend to add '-n' to avoid delays from name resolution |
15:24.58 | lehel | it's strange |
15:25.13 | anthm | well it appears 1.2 is going to be done which is more like 1.0 if you ask me cos the 1.0 was not really ready |
15:25.15 | ManxPower | coppice: realtime would be nice, the new jitterbuffer would be nice, auto priorities would be nice, etc, but the thing that is REALLY nice is AEL |
15:25.20 | PoWeRKiLL | I find a way to do it using cdr_mysql and odbc but in this install i can't use odbc |
15:25.28 | anthm | and it will actually be able to keep you busy for a while |
15:25.34 | lehel | last friday worked well the connection between.. and today.. it;s not working the IAX2 |
15:25.39 | anthm | but the same thing is gonna happen eventualyl |
15:26.01 | sudhir492 | I am not able to get that work. In particular, what should I set in domain/realm? |
15:26.09 | coppice | ManxPpower: the jitterbuffer is what most people will notice, and the other thing mahy people need badly is VAD/DTX |
15:26.16 | tzafrir | lehel, not working=? |
15:26.20 | lehel | it is no difference ... no response. could be firewall problem? |
15:26.43 | lehel | not- working = doesn't get (ping) response |
15:26.46 | tzafrir | if it were a firewall on your system, you'd still see packets with tcpdump |
15:27.01 | PoWeRKiLL | is there another way ? |
15:27.39 | tzafrir | lehel, a lower level test is using netcat: echo test |nc -u asterisk_hostname 4569 |
15:28.21 | lters_ | PoWeRKiLL, write a script that watches your local cdr file and sends the records anywhere u want it to. |
15:28.48 | *** join/#asterisk santiago (~santiago@63.245.86.188) |
15:28.55 | izo | ManxPower: what is AEL ? |
15:29.01 | PoWeRKiLL | lters_ I want it to be via mysql |
15:29.10 | lehel | tzafrir: forward host lookup failed: Host name lookup failure : Resource temporarily unavailable |
15:29.13 | ManxPower | coppice: on the links I care about, there really isn't much jitter. i.e. Point-to-Point T-1 w/QoS or local LAN. |
15:29.15 | PoWeRKiLL | AEL = Asterisk Entreprise Linux :D |
15:29.24 | izo | :-)) |
15:29.31 | izo | what about Asterisk Business Edition ? |
15:29.32 | coppice | ManxPower: so selfish :-) |
15:29.35 | ManxPower | izo: it's a real programming language for extensions.conf. It's included with CVS-HEAD. |
15:29.49 | coppice | Asterisk End Life |
15:29.56 | izo | ManxPower : really ? damn how come I didnt notice |
15:30.06 | izo | ManxPower : what about overhead ? |
15:30.09 | ManxPower | izo: because you were not reading the asterisk-cvs mailing list. |
15:30.11 | _omer | How many calls at a time a pentium 4 machine with 1 GB ram could afford?????? |
15:30.20 | izo | ManxPower: i am but not all the changes :-P |
15:30.22 | ManxPower | izo: all it does is translate the "code" into regular extensions.conf stuff |
15:30.49 | ManxPower | there is also a javascript and perl extensions.conf support on pbxfreeware |
15:30.54 | _omer | How many calls (SIP - No Hardware) at a time a pentium 4 machine with 1 GB ram could afford?????? |
15:31.00 | lters_ | PoWeRKiLL, that is fine, make your script do mysql |
15:31.44 | Qwell | _omer: repeating your question over and over isn't the best way to get help |
15:31.59 | brenda | Does anyone know how long ast_play_and_wait() waits? |
15:32.02 | InfraRed | _omer: check the wiki , under dimentioning server |
15:32.27 | izo | ManxPower:damn i'm really backwards with this stuff |
15:33.19 | PoWeRKiLL | lters_ what do you mean what master.csv and do mysql ? |
15:34.21 | anthm | brenda, probably till the file is done playing |
15:34.30 | *** join/#asterisk gniretar (~mark@198.173.197.15) |
15:34.45 | gniretar | he guys |
15:34.53 | Qwell | she girls |
15:35.04 | gniretar | i'm having a little trouble with my Aastra SIP phones |
15:35.17 | gniretar | they arnt storing Caller ID info. |
15:35.21 | gniretar | and girls ;-) |
15:35.52 | anthm | brenda, according to the source, it would be till you hanngup, press any digit, or the file is done playing |
15:35.55 | gniretar | i need to be able to flip through previous calls and i cant. Does anyone know of something on the Asterisk side that would help me with this? |
15:36.25 | brenda | anthm: interesting... maybe I should add a timeout |
15:37.02 | anthm | brenda, what's the goal ? |
15:37.17 | eldu | is there a way to force a specific codec per telco number within the same sip trunk ? |
15:37.58 | brenda | anthm: background a file, wait X seconds |
15:38.27 | anthm | then cancel the background ? |
15:39.14 | anthm | with any intention of collection digits ? |
15:39.27 | anthm | collecting digits |
15:39.31 | brenda | the timeout doesn't start until the file is done |
15:39.43 | brenda | yes collect digits |
15:40.00 | anthm | inside your app right ? |
15:40.01 | brenda | unless of course the timeout happens |
15:40.03 | brenda | yeah |
15:40.04 | anthm | C code? |
15:40.09 | brenda | yup |
15:40.32 | anthm | seconds or microsec ? |
15:40.37 | brenda | either is fine |
15:40.40 | *** join/#asterisk jimmybob46 (~jim@81.5.154.235) |
15:40.42 | anthm | or millisec |
15:40.50 | Qwell | picoseconds |
15:40.58 | jr352k | anthm: are you collecting digits in the dial plan? or using agi? |
15:41.01 | jimmybob46 | hello all |
15:41.06 | gniretar | hi |
15:41.13 | anthm | ? when |
15:41.33 | jimmybob46 | is this the main asterisk help channel? or have i gone wrong? |
15:41.57 | brenda | I'm the one collecting digits |
15:42.18 | jr352k | ohh! branda! then the question is for you |
15:42.21 | Qwell | brenda: I have a 6 you can add to your collection |
15:42.23 | brenda | lol yeah |
15:42.24 | tzafrir | jimmybob46, it is. About you going wrong? I have no idea |
15:42.51 | jimmybob46 | lol |
15:42.53 | jimmybob46 | nice one |
15:43.07 | jimmybob46 | thanks tzafrir |
15:43.19 | brenda | I'm playing my hand with apps so I can actually help with the asterisk code too |
15:43.55 | jimmybob46 | I got a capi problem, I am behind a pbx, and have to dial 9 to get an isdn line |
15:44.33 | jimmybob46 | I cant work out where to do this.. :( |
15:45.13 | *** join/#asterisk akrall (user@201.128.92.42) |
15:45.40 | tzafrir | jimmybob46, where exactly is your problem? can you make calls inside the PBX? |
15:46.06 | akrall | What kind of phones are the panasonic KX-T7730 and KX-T7030X ones? digital or plain old analog? will they work on asterisk (of course, using an ATA)? |
15:46.20 | tzafrir | does the isdn/capi/whatever channel work? |
15:46.32 | jimmybob46 | my asterisk pbx is fine, chan_capi is working fcpci is loaded, but i get Reason 0x3481 |
15:46.47 | jimmybob46 | how can i determine if the capi channel works? |
15:48.27 | jimmybob46 | i can sip call inside the pbx to other sip numbers, its dialing out over capi that does not go anywhere |
15:49.41 | jimmybob46 | I apologise for my ignorance, I have spent 5 days trying to figure this out. |
15:49.53 | *** join/#asterisk _gigi_ (gigi@jabber.szczecin.pl) |
15:50.04 | _gigi_ | Hello. |
15:50.57 | jsharp | Well, foo. Upgrading to 1.0.9 didn't solve my SIP problem. |
15:51.30 | _gigi_ | im looking for some software for analize RTP streams (delay, jitter, delta, and loss), someone know something ? :) |
15:52.56 | lters_ | PoWeRKiLL, are u wanting to send the cdr's to 2 servers? |
15:53.42 | Darwin35 | hmm |
15:53.48 | Darwin35 | what phone to get |
15:54.16 | InfraRed | cisco 7960 |
15:55.30 | Darwin35 | have a office full of those |
15:55.37 | Darwin35 | this is for home use |
15:55.38 | *** join/#asterisk fugitivo (~ajf@168-226-245-129.mrse.com.ar) |
15:55.44 | Darwin35 | x401 I think |
15:56.08 | *** join/#asterisk Gunnar (~gunnar@62.97.243.70) |
15:56.53 | *** join/#asterisk Gunnar (~gunnar@62.97.243.70) |
15:57.00 | jimmybob46 | I am in the UK, is there anyone that knows about any capi specific problems? BT usually cause a few problems.. |
15:57.08 | gordonjcp | BT always causes problems |
15:57.16 | gordonjcp | even if you're not doing anything complicated |
15:57.17 | anthm | brenda, off the top of my head, something like this http://66.250.68.190/eg/play_timed.c |
15:57.23 | anthm | didnt try it tho |
15:57.50 | puzzled | jimmybob46: does capiinfo output a lot of info indicating that your card is active? |
15:58.21 | jimmybob46 | capi info says lots |
15:58.51 | puzzled | jimmybob46: ok, and have you correctly loaded chan_capi.so in /etc/asterisk/modules.conf? |
15:59.03 | *** join/#asterisk BoNaDs (~theplot@ool-44c53748.dyn.optonline.net) |
15:59.17 | BoNaDs | does "echotraining" when enabled, determine and override the setting for "echocancel"? |
15:59.40 | *** join/#asterisk Andrezo (~www@217.129.208.124) |
16:00.04 | jimmybob46 | Yes, I have made sure that I have that in my modukles. asterisk -vvvvvvgc shows me lots of info about capi loading |
16:01.21 | Darwin35 | ok cool |
16:01.22 | puzzled | jimmybob46: ok, did you configure /etc/asterisk/capi.conf? |
16:01.42 | Darwin35 | the x401 is the netweb 401 renamed and they are going to replace my broken oone |
16:01.52 | jimmybob46 | I have configured capi.conf yes. |
16:01.58 | Hmmhesays | I love it when people call wanting general knowledge on voip |
16:02.38 | puzzled | jimmybob46: if all that is ok then I guess it must be something in the dialplan. are you using chan_capi-cm-0.5.4? |
16:02.44 | _gigi_ | im looking for some software for analize RTP streams (delay, jitter, delta, and loss), someone know something ? :) |
16:03.04 | jimmybob46 | I am using the chan_capi from sourceforge (o.54?) |
16:03.19 | puzzled | yes latest is 0.5.4 |
16:04.26 | *** join/#asterisk ArkyLady (ArkyLady@adsl-67-64-6-10.dsl.ltrkar.swbell.net) |
16:04.41 | jimmybob46 | I would paste my extensions.conf in here, but dont want to upset anyone :) |
16:04.50 | ManxPower | ~pastebin |
16:04.50 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
16:05.10 | ManxPower | ~mailinglist |
16:05.10 | jbot | [mailinglist] Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
16:05.11 | ManxPower | ~docs |
16:05.11 | jbot | i guess docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
16:05.29 | *** topic/#asterisk by drumkilla -> Asterisk: The Open Source PBX || Asterisk 1.2 Janitors Needed! http://dev.asteriskdocs.org || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 http://www.cluecon.com ||Astricon 2005 in Anaheim Oct 12-14 http://www.astricon.net/2005/ - Speakers wanted |
16:05.48 | *** topic/#asterisk by drumkilla -> Asterisk: The Open Source PBX || Asterisk 1.2 Janitors Needed! http://dev.asteriskdocs.org || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 http://www.cluecon.com || Astricon 2005 in Anaheim Oct 12-14 http://www.astricon.net/2005/ - Speakers wanted |
16:05.50 | *** join/#asterisk Gunnar (~gunnar@62.97.243.70) |
16:05.52 | akrall | What kind of phones are the panasonic KX-T7730 and KX-T7030X ones? digital or plain old analog? will they work on asterisk (of course, using an ATA)? |
16:06.54 | ManxPower | akrall: plug one of the phone into an analog POTS line. If you can use the phone then it will work with Asterisk |
16:07.09 | Qwell | if it starts smoking, you probably can't |
16:07.25 | syle2 | do you use codecs on zap channels? |
16:07.26 | akrall | jajaja not very scientific :) |
16:07.43 | Qwell | akrall: That is a very scientific method |
16:08.40 | _omer | does Asterisk Support Dual Processors??? fully utilize both processors??? |
16:08.56 | akrall | Qwell: :) |
16:09.01 | *** join/#asterisk diegodi (~diegoluig@host-84-222-36-3.cust-adsl.tiscali.it) |
16:09.04 | Qwell | _omer: sure |
16:09.23 | _omer | Qwell: sure for both questions? :) |
16:09.37 | Qwell | no to the first, yes to the second |
16:09.53 | jimmybob46 | well, please could u check this simple config : http://pastebin.ca/18457 |
16:10.02 | *** join/#asterisk sloPP (~slepp@S0106000f663692da.ed.shawcable.net) |
16:10.07 | diegodi | Hi all!! |
16:10.14 | jsharp | Buh. |
16:10.25 | sloPP | anyone know the difference between National 1 and National 2? |
16:10.36 | essobi | about 10 dollars |
16:10.45 | _omer | Qwell : :D |
16:12.31 | jimmybob46 | what do u reckon puzzled? |
16:15.13 | *** join/#asterisk agave-txlink (phanop@216.81.47.201) |
16:15.21 | *** part/#asterisk akrall (user@201.128.92.42) |
16:15.30 | agave-txlink | is it possible to do caller ID matching in realtime? ex: exten => _NXXNXXXXXX/9725551212 ? |
16:17.17 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
16:17.21 | diegodi | someone can explain me how to modify the stripping 2 to 1 in the command {EXTEN:2} |
16:17.25 | *** join/#asterisk nitram (foo@superblob.com) |
16:17.54 | Qwell | diegodi: by replacing the 2 with a 1? |
16:18.19 | diegodi | yes, every time I save the configuration asterisk make this to 2!!! |
16:18.28 | Qwell | asterisk doesn't change your configs |
16:18.39 | diegodi | so...? |
16:18.47 | Qwell | so figure out what is |
16:19.18 | *** join/#asterisk Beave (~beave@vistech.org) |
16:19.39 | diegodi | I don't know, I'm using asterisk@home and the command CAPI/XXXXXXB$OUTNAME} |
16:19.43 | *** join/#asterisk joshpbx (~joshpbx@83.27.103.159) |
16:19.47 | Qwell | There you go. |
16:19.52 | Qwell | so amp is probably changing it on you |
16:19.53 | *** join/#asterisk SteveL (~stephen@smtp.burlesonisd.net) |
16:19.56 | Qwell | Thats why we don't use amp. ;] |
16:20.35 | diegodi | thanks very much!!! |
16:23.08 | puzzled | jimmybob46: let me have a look |
16:23.14 | SteveL | What do I need to do to access cvs HEAD? I keep getting this error: cvs login: authorization failed: server cvs.digium.com rejected access to /usr/cvsroot for user anoncvs |
16:23.40 | SteveL | I'm using anoncvs for password. |
16:23.56 | Qwell | password is blank, right? I forget, its been a while |
16:24.10 | DarthClue | anoncvs is the correct pw...let me try it |
16:24.13 | Qwell | oh |
16:24.23 | SteveL | blank password won't work either |
16:24.45 | *** part/#asterisk diegodi (~diegoluig@host-84-222-36-3.cust-adsl.tiscali.it) |
16:25.01 | MikeJ[Laptop] | SteveL, I just tried and it is working. |
16:25.03 | *** join/#asterisk grimse (~grimse@p5481D4AE.dip.t-dialin.net) |
16:25.16 | MikeJ[Laptop] | tripple check your CVSROOT |
16:25.28 | MikeJ[Laptop] | and make sure your not screwing up the pwd |
16:25.58 | puzzled | jimmybob46: the only difference is that I use "|" so e.g. exten => _00XXXXXXXXX,2,Dial(CAPI/contr1/b${EXTEN:1}|45|r) |
16:26.14 | DarthClue | works here too. follow the directions at http://www.darthclue.org/categories/3-Chalkboard-Examples |
16:26.56 | SteveL | STRANGE |
16:26.59 | *** join/#asterisk drooth (~drooth@ip68-111-235-172.sd.sd.cox.net) |
16:27.06 | SteveL | it's working now doing the exact same thing I was doing before |
16:27.42 | DarthClue | SteveL: sometimes it just takes a little evil to make things work. |
16:27.51 | SteveL | haha |
16:27.51 | SteveL | true |
16:28.07 | *** join/#asterisk Dave-- (~a@choloconsultancy.plus.com) |
16:28.32 | Dave-- | evening |
16:29.26 | Dave-- | dont suppose theres anyone awake that can tell me why.... |
16:29.26 | Dave-- | exten => _0.,1,Dial(Zap/1/${EXTEN},10) |
16:29.26 | Dave-- | exten => _0.,2,NoOp(HANGUPCAUSE is ${HANGUPCAUSE}) |
16:29.36 | Dave-- | the second line is never got to |
16:29.50 | Qwell | Dave--: it probably jumps to 101 on failure |
16:29.57 | Qwell | or, 102? I forget. |
16:30.15 | florz | Dave--: Or to h upon hangup. |
16:30.21 | Dave-- | lol |
16:30.50 | Dave-- | so what should my next line be to catch it? |
16:31.05 | MikeJ[Laptop] | Dave--. what version? |
16:31.09 | florz | Dave--: what exactly do you wanna catch? |
16:31.27 | ManxPower | Dial will jump to n+1 on busy if there is no n+101 |
16:31.37 | Dave-- | Asterisk 1.0.9-BRIstuffed-0.2.0-RC8h |
16:31.52 | Dave-- | i need to catch hangup on busy |
16:31.58 | Dave-- | so i can redial |
16:32.15 | MikeJ[Laptop] | see ManxPower's comment |
16:32.46 | Dave-- | therefore it should be getting to the second line, correct? |
16:34.25 | *** join/#asterisk DagMoller (~DagMoller@4b83b8828b0c682f.session.tor) |
16:35.30 | DagMoller | Have iaxtel problems? i'm from brasil, and canot connect to iaxtel |
16:35.44 | brenda | anthm: wow thanks! |
16:36.47 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-129-44-144-219.buff.east.verizon.net) |
16:36.55 | florz | Dave--: I'd say so, yeah. |
16:37.34 | Dave-- | thanks |
16:38.34 | florz | Dave--: So, what was it? |
16:38.40 | Dave-- | ive no idea |
16:38.51 | florz | But it works no? |
16:38.52 | Dave-- | i cant get it to go anywhere beyond |
16:38.52 | florz | +w |
16:38.52 | BoNaDs | anyone here used fxotune? |
16:38.53 | Darwin35 | man hop-op has made a sip/wifi phone priced at 39.95 but are unwilling to sell to the general public |
16:39.08 | Darwin35 | they only want to sell to providers of voip service |
16:39.16 | Dave-- | would they be willing to see to a call centre? |
16:39.26 | Darwin35 | called the hop-on 1515 |
16:39.27 | *** join/#asterisk Goshen (~Goshen@67-40-107-29.slkc.qwest.net) |
16:39.48 | InfraRed | http://www.mosnews.com/news/2005/07/25/spammerdead.shtml |
16:39.50 | InfraRed | coo |
16:39.52 | Darwin35 | http://www.wifi-cell.com/ |
16:40.42 | Darwin35 | it just pisses me off all these companies now making good products for voip but unwilling to sell to the general user |
16:40.59 | *** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1) |
16:41.10 | Darwin35 | and in nothing less then a bulk of 1000 units at a time to caompanies |
16:41.11 | *** join/#asterisk Derkommissar (~alberto@66.64.215.6.nw.nuvox.net) |
16:42.04 | brenda | Darwin35: supporting end users is expensive |
16:42.25 | Dave-- | florz - any idea why it might not be getting there? |
16:42.38 | ManxPower | We have way cool product, but you can't actually buy it! |
16:42.58 | Qwell | vaporware? |
16:42.59 | Darwin35 | heck I would offer to do support for the units |
16:43.05 | mkrufky | How come channel topic says "Asterisk 1.2 Janitors needed" ... aren't we up to 1.09 or 1.10 or something |
16:43.19 | Qwell | mkrufky: 1.2 > 1.0.x |
16:43.20 | mkrufky | (i'm still using 1.07 on my server) |
16:43.29 | *** join/#asterisk mjmac (~mjmac@mjmac.active.supporter.pdpc) |
16:43.31 | mkrufky | aha... when did 1.2 come out? |
16:43.40 | Darwin35 | it has not yet |
16:43.52 | Darwin35 | there is a planned code freeze |
16:43.53 | mkrufky | oh, okay |
16:43.58 | Darwin35 | then a move to 1.2 |
16:44.03 | florz | Dave--: Nope, no clue. |
16:44.07 | mkrufky | ah, thats good news |
16:44.27 | florz | Dave--: It's a BRI line? |
16:44.37 | Dave-- | BRI line? |
16:44.43 | file[desk] | Cresl1n: Mattttttt |
16:44.48 | florz | Dave--: ISDN |
16:44.55 | Dave-- | nope, analogue line |
16:45.04 | *** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com) |
16:45.11 | Dave-- | thru a Wildcard X100P |
16:45.31 | florz | Dave--: Ah, OK, maybe that's causing that the busy condition isn't correctly recognized? |
16:45.52 | Dave-- | well i was thinking that |
16:45.58 | Darwin35 | I just want to see usable products on the market |
16:46.11 | florz | Dave--: I've got no experience with analog lines and asterisk, but it's quite likely to cause problems, I guess :-) |
16:46.12 | Dave-- | and ive been thru all the gubbins in indicators.conf |
16:46.34 | Dave-- | im wondering if BT's "You can press 5 to use ringback" over the top of the engaged tone |
16:46.43 | Dave-- | might be fucking with the signalling |
16:47.26 | Darwin35 | whoi is it that owns atcomm |
16:47.31 | Darwin35 | I forget his name |
16:48.55 | *** join/#asterisk coppice (~pocketirc@30.195.17.210.dyn.pacific.net.hk) |
16:50.03 | *** join/#asterisk tarvid (~tarvid@68-67-192-202.chvlva.adelphia.net) |
16:53.56 | *** join/#asterisk marc_in_lux (~gergesm@cable-83.217.135.132.coditel.net) |
16:54.09 | *** join/#asterisk mut (~animenodv@65.111.201.79) |
16:54.13 | marc_in_lux | good evening |
16:54.17 | tarvid | looking for an 800 provider |
16:54.38 | tarvid | early eveing in luxembourg? |
16:54.41 | mut | local telco? |
16:54.47 | Darwin35 | astralink |
16:54.50 | marc_in_lux | tarvid, 7 pm... yes :-) |
16:55.03 | *** join/#asterisk zoa (~k@213.91.216.136) |
16:55.10 | *** part/#asterisk DagMoller (~DagMoller@4b83b8828b0c682f.session.tor) |
16:55.15 | marc_in_lux | looking for a beginners introduction into dialplans. Replicating those festures found in SOHO PBX's. |
16:55.28 | marc_in_lux | like transferring calls, parking them, picking up from another location etc... |
16:55.58 | DarthClue | marc_in_lux: start with the wiki. |
16:56.02 | emrah | marc_in_lux: What would you like to do? |
16:56.08 | *** join/#asterisk loick (~loick@APuteaux-151-1-49-217.w82-124.abo.wanadoo.fr) |
16:56.30 | emrah | Do you want to be able to receive a call to a given extension, park it and take it in another station? |
16:56.38 | marc_in_lux | emrah, for example |
16:56.53 | marc_in_lux | or, doing a 3-way conference with two internal sip phones and an external call |
16:57.07 | emrah | Ok... |
16:57.17 | marc_in_lux | or, routing all external call to one extension, but allowing others to pick up those calls on request |
16:57.30 | Darwin35 | read the wiki |
16:57.34 | emrah | Let me talk with you in a private window without disturbing here |
16:57.38 | Darwin35 | the wiki has all your answers |
16:57.52 | Qwell | emrah: best to answer these questions in public, so everybody benefits |
16:57.53 | Darwin35 | the wiki is the book of the goods |
16:57.54 | marc_in_lux | Darwin35, it sure may. But you're new in telephony, it's hard to digest. |
16:58.15 | emrah | ok |
16:58.16 | DarthClue | marc_in_lux: start with the wiki, you'll be much happier later. |
16:58.33 | Darwin35 | then sit your butt down and write a sterisk-for-dummies book |
16:58.46 | emrah | marc_in_lux: It's good to read the wiki. It's perfectly explained. www.voip-info.org |
16:58.53 | marc_in_lux | Darwin35, DarthClue - happy to document where I'm getting. |
16:58.58 | Qwell | I don't know about perfectly... ;] |
16:59.09 | emrah | You have also a good example files. |
16:59.18 | emrah | Just make them with the appropriate cmd |
16:59.24 | marc_in_lux | I'll restudy the wiki. |
16:59.27 | [TK]D-Fender | DarthClue : We as a community really should make a much "fuller" sample dialplan with comments to give people as a place to start for understanding how things work. Going right to the Wiki to break apart every command can be frustrating to newbs... |
16:59.45 | marc_in_lux | [TK]D-Fender, thanks :-) |
16:59.50 | astoria | It really does help to have a working extensions.conf to work from. |
17:00.00 | astoria | But once you pick it up, the wiki organization is purrrfect |
17:00.14 | marc_in_lux | the problem for newbies in telecoms is that most concepts are not really understood. I can use my little home pbx. |
17:00.21 | [TK]D-Fender | astoria : I agree, but we didn't do that groundwork yet. |
17:00.24 | Darwin35 | I have posted a few on pastebin.ca |
17:00.29 | lathos42 | wiki good |
17:00.31 | Darwin35 | goo lok for darwin35 |
17:00.36 | astoria | Yeah, thats true. I am working on a degree in telecom, so I guess it was easier. |
17:00.40 | marc_in_lux | but I have a tough time following how a 3 way conf. should go in extensions.conf |
17:00.41 | Darwin35 | its got loads of extensions |
17:00.45 | [TK]D-Fender | marc_in_lux : Feel frr to PM me for the time being. |
17:01.04 | Qwell | marc_in_lux: meetme |
17:01.06 | marc_in_lux | [TK]D-Fender, thanks. |
17:01.06 | [TK]D-Fender | free* |
17:01.12 | Darwin35 | but I an still adding functions as I can |
17:01.27 | *** join/#asterisk brettnem (~brettnem@207.90.232.34) |
17:01.51 | *** join/#asterisk denisgalvao (~Denis@200.146.0.254) |
17:02.01 | [TK]D-Fender | marc_in_lux : 3-way calling is often nothing more that being able to accept another call while on one already and using the IP phones "conference feature". That isn't even in the dialplan. |
17:03.30 | marc_in_lux | Darwin35, found lots of stuff from you. will now go and study that |
17:04.00 | denisgalvao | coppice: How may I get the unicall logs? They will go to the same files of Asterisk logs? |
17:04.14 | coppice | yes |
17:04.48 | denisgalvao | coppice: Could you help on a problem? |
17:04.59 | coppice | ok |
17:05.33 | denisgalvao | coppice: Jul 25 09:50:07 WARNING[3200] channel.c: Exception flag set on 'UniCall/13-1', but no exception handler |
17:06.14 | denisgalvao | coppice: Im running FC2 with * HEAD and unicall0.0.3pre3 |
17:06.34 | denisgalvao | coppice: CPU 100% |
17:06.40 | coppice | i saw your email about that i have to check |
17:06.55 | denisgalvao | coppice: Hmmm.. Ok. |
17:07.39 | *** join/#asterisk xcore (~xcore@200.175.93.58.tbprof.gvt.net.br) |
17:07.42 | denisgalvao | coppice: I can test your unicall with a lot of E1 PABX. |
17:08.28 | xcore | people, anybody here know how to eliminate the echo on asterisk+IAX... i have echo when i speak, i listen my voice |
17:09.17 | *** join/#asterisk denisgalvao (~Denis@200.146.0.254) |
17:09.32 | denisgalvao | coppice: Sorry... I lost my conn. |
17:10.35 | denisgalvao | coppice: I was writing that Im on a telephony company, so we have a lot of equipments to test it out. |
17:11.29 | *** join/#asterisk craziman2 (~donnie@boromir.apid.com) |
17:12.11 | denisgalvao | How may get mor information(logs) from UNICALL? |
17:12.31 | craziman2 | does anyone know why I can't get anything when I do a cvs checkout -r v1-0_stable zaptel |
17:14.16 | *** join/#asterisk craziman2 (~donnie@boromir.apid.com) |
17:16.15 | *** join/#asterisk craziman2 (~donnie@63.238.52.21) |
17:16.17 | coppice | add the log control statement to unical.conf |
17:16.35 | *** part/#asterisk craziman2 (~donnie@63.238.52.21) |
17:16.45 | denisgalvao | loglevel=255 is the higher level? |
17:17.14 | coppice | yes |
17:17.20 | ronn | hi guys.. i'm lookin for polish and hungarian DIDs |
17:17.29 | anthm | brenda, did that code work? |
17:17.40 | denisgalvao | coppice: Ok. |
17:17.41 | Qwell | anthm: fn != filename |
17:17.41 | Qwell | p |
17:17.43 | *** join/#asterisk craziman2 (~donnie@63.238.52.21) |
17:17.44 | Qwell | :p rather |
17:18.24 | anthm | doh! |
17:18.27 | denisgalvao | coppice: Why I can call from PABX to the Asterisk but I cannot call from Asterisk to PABX? |
17:18.34 | craziman2 | Sorry to ask again.. but I dc'ed... any one have any idea why I can't pull anything with, "cvs checkout -r v1-0_stable zaptel" |
17:18.51 | denisgalvao | coppice: From Asterisk to PABX I got the 100% CPU usage. |
17:19.09 | joshpbx | craziman2: u cant download nighly snapshots? |
17:19.50 | denisgalvao | craziman2: try it: cvs checkout -r v1-0 |
17:19.59 | craziman2 | Trying to setup a semi production system... so I figured the stable would be better. |
17:20.02 | denisgalvao | without _stable |
17:20.07 | craziman2 | k |
17:20.21 | denisgalvao | v1-0 will poitn you to the satble. |
17:20.54 | craziman2 | I get "cvs [server aborted]: no such tag -v1-0" |
17:21.21 | denisgalvao | export CVS_RSH= |
17:21.22 | denisgalvao | export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot |
17:21.31 | denisgalvao | cvs login (the password is: anoncvs) |
17:21.37 | MikeJ[Laptop] | cvs co -r v1-0 asterisk |
17:21.39 | denisgalvao | cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons |
17:22.53 | craziman2 | seams to be working... thanks |
17:24.31 | *** join/#asterisk joerg (~joerg@p54889B45.dip0.t-ipconnect.de) |
17:25.23 | Darwin35 | when are sip phones going to come down to the cost of a normal home phone |
17:26.18 | *** join/#asterisk mithro (~tim@83-169-171-16-dynip.superkabel.de) |
17:26.33 | *** join/#asterisk jsharp (~foo@65.90.64.82) |
17:26.46 | Qwell | Darwin35: sometime in 2036 I'd imagine |
17:28.05 | *** part/#asterisk xcore (~xcore@200.175.93.58.tbprof.gvt.net.br) |
17:30.00 | MikeJ[Laptop] | Darwin35, check tomorrow.. maybe by then.. if not, lather, rinse, repeat. |
17:30.19 | *** join/#asterisk rashid (~rashid@63.133.150.3) |
17:30.38 | Qwell | I wouldn't mind paying $10 at Radio Shack for a SIP phone |
17:30.52 | rashid | neither would i |
17:31.04 | *** join/#asterisk oej (~oej@213.204.186.40) |
17:31.16 | Qwell | oej: evening |
17:31.25 | oej | Evening |
17:31.30 | oej | ~drumilla |
17:32.04 | oej | ~drumkilla |
17:32.04 | jbot | drumkilla is, like, Russell Bryant, the Asterisk release branch maintainer <russelb@clemson.edu> |
17:33.01 | astoria | ok? |
17:35.12 | zoa | hey ho oej |
17:35.22 | Andrezo | where can i get the list of new features on 1.2? the changelog on cvs, says nothing more than a few lines |
17:35.27 | oej | Hej zoa! Dinner time. Brb |
17:35.30 | anthm | ok |
17:35.31 | anthm | http://66.250.68.190/eg/play_timed.diff |
17:35.36 | zoa | hey ho anthm |
17:35.49 | anthm | http://66.250.68.190/eg/app_testit.c |
17:35.51 | anthm | hey zoa |
17:35.52 | brenda | NEAT! |
17:36.05 | anthm | testit(demo-congrats|5) |
17:36.35 | Qwell | anthm: would be a nice head feature |
17:36.37 | rashid | so, I have an, hmm, interesting problem. We have a 24 port sip gateway at one location that registers to an asterisk box in our data center, calls then go out to the PSTN. When you call certain PSTN numbers the called party can not hear the caller. Other numbers work fine and once in awhile even the affected number work fine |
17:37.01 | brenda | you're tellin me |
17:37.12 | rashid | i didn't believe it until I played with it for myself |
17:37.26 | *** join/#asterisk Barmal (~info@c-24-30-75-206.hsd1.ga.comcast.net) |
17:38.07 | rashid | because my cellphone, my bosses number and our consultants numbers all work fine. My girl friend's office doesn't, neither does her cell phone, my co workers house number doesn't work. |
17:38.34 | brenda | anthm: your skills make me jealous |
17:38.41 | rashid | its insanity |
17:38.55 | Qwell | brenda: he has "other skills" too, I hear |
17:39.03 | rashid | my boss and I sat in the pbx room for 5 hours on friday trying to figure it out |
17:39.16 | ManxPower | rashid: remove the "r" command to the Dial line. |
17:39.17 | brenda | Qwell: lol... as in what? |
17:39.30 | Qwell | brenda: dunno |
17:39.34 | essobi | Anyone played with the new H323 drivers? |
17:40.48 | astoria | ha ha. that would suck if you spend all that time just to find out it was one letter! |
17:41.21 | rashid | we're not using the 'r' command |
17:42.22 | anthm | do you need to use it from the dialplan or just in C ? |
17:43.01 | craziman2 | loader.c:440 load_modules: Loading module app_realtime.so failed! |
17:43.01 | craziman2 | [root@smokey lib]# Ouch ... error while writing audio data: : Broken pipe |
17:43.01 | craziman2 | loader.c:440 load_modules: Loading module app_realtime.so failed! |
17:43.01 | craziman2 | [root@smokey lib]# Ouch ... error while writing audio data: : Broken pipe |
17:43.01 | craziman2 | trying to go from latest CVS to 1.0... now I get this message when I try an asteris -cvvv "loader.c:440 load_modules: Loading module app_realtime.so failed! |
17:43.03 | craziman2 | [root@smokey lib]# Ouch ... error while writing audio data: : Broken pipe |
17:43.05 | craziman2 | " |
17:43.26 | *** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
17:43.38 | Qwell | craziman2: remove all your old modules |
17:44.19 | joshpbx | and include`s too |
17:44.36 | ManxPower | craziman2: thats a mpg123 error |
17:44.42 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
17:44.59 | rashid | yup, not using it anywhere :-/ |
17:45.00 | ManxPower | the Ouch is, the other one is not. |
17:45.15 | ManxPower | craziman2: run "make install" and READ THE MESSAGE AT THE END |
17:45.22 | Blackthorn | Hello. About every 5-7 days my sip and disa users experance poor voice quality and I can reboot the server and everything clears up. How can/should I trouble shoot this issue? |
17:45.32 | rashid | not using any options in any of the Dial commands actually |
17:45.38 | ManxPower | ya know the message that talks about what you have to do do downgrade to 1.0.x |
17:45.53 | craziman2 | where do I find the include files to remove? |
17:45.53 | rashid | Blackthorn, do you have to reboot the box or just restart asterisk |
17:46.02 | ManxPower | craziman2: run "make install" and READ THE MESSAGE AT THE END |
17:46.21 | *** join/#asterisk dan2_ (~foobar@dan2.professional.supporter.pdpc) |
17:47.03 | rashid | Blackthorn, we had a similar issue, ended up just making a cronjob that ran 'restart when convenient' at 3am each day. Its a hack solution, but it worked |
17:47.20 | ManxPower | the part where it says " ** NOTE FOR DOWNGRADING FROM CVS HEAD ** " |
17:47.33 | ManxPower | Blackthorn: using analog cards? |
17:48.22 | *** join/#asterisk file (~jcolp@mctnnbsah25-142166093154.nb.aliant.net) |
17:48.49 | Blackthorn | rashid: i have to reboot the box. manixpower: no just running one 4 port pri card. |
17:49.02 | rashid | any other ideas for the random no voice situation? |
17:49.07 | *** join/#asterisk DagMoller (~DagMoller@4b83b8828b0c682f.session.tor) |
17:49.15 | Barmal | what application is beeing used for voice Forbidden |
17:49.15 | Barmal | You don't have permission to access /admin/ on this server. |
17:49.15 | Barmal | -------------------------------------------------------------------------------- |
17:49.15 | Barmal | Apache/1.3.33 Server at amp.eiktel.com Port 8080 |
17:49.18 | rashid | any commands we _should_ be using in the Dial command? |
17:49.23 | Barmal | SORRY |
17:49.26 | *** join/#asterisk wshs (screwy@69-168-246-252.bflony.adelphia.net) |
17:49.56 | ManxPower | rashid: connect a SIP phone directly to the SIP/PSTN gateway. If the problem still occurs, contact the gateway vendor |
17:50.02 | craziman2 | okay... thanks for the help guys... sorry I ask ?? when I should have RTFM... I appreciate the help. |
17:50.22 | ManxPower | craziman2: Did you read the README in the zaptel directory too? |
17:50.49 | craziman2 | uh... I am now :) |
17:51.23 | ManxPower | and the stuff in the doc/ directory of Asterisk? |
17:51.33 | rashid | ManxPower, so here's the other part of the story. At this location we have another 24 port sip gateway, that registers to a local asterisk box, and then passes the call off to the datacenter to go out over the same connection to the PSTN |
17:51.38 | rashid | everything works fine here |
17:51.40 | rashid | however |
17:52.01 | ManxPower | rashid: same vendor/make/model? |
17:52.02 | rashid | if we change the sip gateway to register directly to the datacenter asterisk, the problem rears its ugly head once again |
17:52.04 | rashid | yes |
17:52.12 | Barmal | what application is best to use for voice activated auto attendant? |
17:52.24 | ManxPower | Barmal: I can't think of any. |
17:52.56 | Barmal | ManxPower: but there is one made it should be.... |
17:53.16 | craziman2 | One more question... everything is working now I get these messages for my IAX Trunks... "chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?) |
17:53.16 | craziman2 | " |
17:53.38 | craziman2 | the reading I have done on this show that it's a stable for latest head issue. |
17:53.48 | rashid | its a real head scratcher |
17:54.04 | craziman2 | Is there a way to talk IAX between a 1.0 box and a Latest CVS? |
17:54.04 | ManxPower | Barmal: Why? There's only one open source voice recognition project that I'm aware of and it sucks. |
17:54.35 | ManxPower | craziman2: turn off trunking and jitterbuffer on the cvs-head box |
17:54.55 | Barmal | manxpower, whats the name of it? |
17:55.06 | rashid | we've been completely unable to replicate the problem reliably |
17:55.09 | ManxPower | Barmal: Sphinx |
17:55.44 | Katty | is firefly any good? |
17:55.48 | Katty | or is there a better one? |
17:55.57 | zoa | idefisk!!!! |
17:55.58 | zoa | :) |
17:56.35 | fugitivo | firefly has a problem with cpu usage |
17:56.40 | lathos42 | oops, wrong Firefly :D |
17:56.56 | fugitivo | idefisk seems to work well, but i don't like the interface |
17:58.33 | Corydon-w | Speaking of speech recognition, ever been overhearing a conversation, and only discover about 5 minutes in that they're actually speaking English? |
17:59.15 | zoa | fugitivo, what dont you like about the interface ? |
17:59.34 | Corydon-w | That's Sphinx, only it's more picky |
18:00.10 | doolph | mmm |
18:00.18 | doolph | behind nat is a problem eh |
18:01.31 | Katty | nat=yes |
18:02.44 | *** join/#asterisk Juxt (~Juxt@64.135.20.202) |
18:02.52 | Juxt | good day |
18:03.02 | Juxt | does anyone have any experience with chanspy? |
18:03.49 | Juxt | i need to know where chanspy writes it's temporary file |
18:04.04 | craziman2 | Have any of you seen sip 'lock up' using the latest cvs? I see this thing where sip show peers doesn't list anything... then when I restart asterisk everything is there? |
18:04.06 | Corydon-w | Hah, temporary file? |
18:04.17 | Corydon-w | What do you think this is, Windows? |
18:04.20 | Juxt | lol |
18:04.29 | Juxt | well i believe that chanspy does write something |
18:04.32 | Juxt | some sort of a stream file |
18:04.38 | Juxt | cause i've seen it before |
18:04.38 | *** part/#asterisk DagMoller (~DagMoller@4b83b8828b0c682f.session.tor) |
18:04.38 | *** join/#asterisk perlmonky (~perlmonky@pix.benchmark-systems.com) |
18:04.40 | essobi | uhhh. |
18:04.40 | Corydon-w | No, it writes to a socket |
18:04.44 | essobi | No ddude. |
18:04.49 | essobi | No files. |
18:04.57 | Juxt | ok i am wrong then |
18:05.12 | Juxt | then is there a reason why my system studders sometimes while monitoring a channel via chanspy? |
18:05.12 | essobi | yes |
18:05.18 | essobi | yes you are |
18:05.24 | essobi | umm. |
18:05.37 | essobi | too slow to multiplex/mixing/encoding? |
18:05.41 | ManxPower | craziman2: at least in 1.0.x if you do a "stop when convienent" or "stop gracefully" you can't do anything at the console except exit and reconnect |
18:05.46 | essobi | possibly a bug |
18:06.06 | Juxt | the system is a pretty powerful athlon and sometimes it studders even when there's only 1 channel going |
18:06.46 | essobi | then it's a bug or it's hardware. |
18:06.56 | essobi | what's "stuttering" ? |
18:07.14 | Juxt | well the sound starts crackling |
18:08.37 | rashid | oh |
18:08.47 | rashid | and the problem only occurs with outgoing calls |
18:08.56 | Katty | oh noes! Hmmhesays fell off teh interweb! |
18:13.18 | Katty | hmm |
18:13.28 | Katty | i don't understand this, in asterisk, zap show channels shows i have 4 channels. |
18:13.36 | Katty | but in zapata.conf i said channels is 1-8 |
18:13.51 | ManxPower | Katty: put your zapata.conf on pastebin |
18:15.45 | *** join/#asterisk lehel (~Lehel@82.79.20.17) |
18:15.52 | lehel | hello again |
18:16.07 | *** join/#asterisk _deg_ (~deg@200.146.0.254) |
18:16.18 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net) |
18:16.23 | lehel | requested format = ilbc, ?? "ilbc" <?? what's this? |
18:16.42 | lehel | i set up ulaw,gsm |
18:17.00 | BoNaDs | argh! |
18:17.02 | lehel | wherefrom is coming this "ilbc" ? |
18:17.03 | rashid | perhaps the requested codec from the sip client? |
18:17.04 | *** join/#asterisk bsd3 (~bsd@203.134.192.221) |
18:17.15 | lehel | it's IAX2 |
18:17.19 | tzafrir_laptop | ilbc is a codec. format_ilbc.so or something similar |
18:17.21 | rashid | or IAX2 client |
18:17.23 | BoNaDs | anyone successfully get fxotune to work? |
18:17.26 | *** join/#asterisk pa (~Paolo@pa.user) |
18:17.47 | lehel | and why it is requsting in this format?.. i never specified it |
18:17.51 | BoNaDs | i stop asterisk, but it skips my fxo devices on my tdm04b cards |
18:17.57 | *** part/#asterisk Juxt (~Juxt@64.135.20.202) |
18:17.58 | rashid | maybe the client defaults to that |
18:18.01 | *** part/#asterisk bsd3 (~bsd@203.134.192.221) |
18:18.26 | tzafrir_laptop | what is fxotune? |
18:18.44 | rashid | in any case, if you disallow ilbc in your asterisk configuration the codec actually being used will be whatever you allowed |
18:19.01 | BoNaDs | fxotune turns the hardware echo cans on in the tdm 400 cards |
18:19.39 | tzafrir_laptop | BoNaDs, I wish those small utilities were simply integrated into ztcfg |
18:20.00 | BoNaDs | heh i'll settle for just wishing that it would work |
18:20.01 | BoNaDs | =P |
18:20.08 | lehel | rashid: i found ilbc in my sip.conf.. but i'm not using SIP |
18:20.10 | tzafrir_laptop | simply one binary that has all the ioctls |
18:20.35 | lehel | howto stop requesting "ilbc"? |
18:20.38 | ManxPower | anyone know of a web/domain hosting company that will host e-mail and allow access via imap |
18:20.38 | BoNaDs | its not detecting my FXO devices when clearly it should |
18:20.48 | joshpbx | disallow this codec? |
18:20.58 | lehel | yes |
18:21.00 | tzafrir_laptop | disallow=all, allow=certain codecs |
18:21.14 | lehel | in my iax.conf |
18:21.21 | tzafrir_laptop | BoNaDs, strace will show you where it looks |
18:21.43 | BoNaDs | what is strace? |
18:22.08 | yaaar | BoNaDs: it's a stack tracer |
18:22.14 | joshpbx | tzafrir_laptop: but he can find it in default iax conf.. but if somone are lazy to read all conf.. ;/ |
18:22.22 | yaaar | helps you track down what a crashing program is doing |
18:22.32 | lehel | ppl i have: diasallow=all, and allow=ulaw&gsm |
18:22.38 | BoNaDs | oh its not exactly crashing |
18:22.40 | BoNaDs | =P |
18:22.46 | BoNaDs | its just not detecting my boards |
18:22.51 | BoNaDs | asterisk sees them fine however |
18:23.10 | BoNaDs | instructions say to stop asterisk before running it which i did |
18:23.42 | ManxPower | BoNaDs: did you do a "ps -ax | grep asterisk" to make SURE asterisk is not running? |
18:24.07 | BoNaDs | yep |
18:24.12 | joshpbx | lehel: then paste u call to pastebin. maybe gw support only ilbc. |
18:24.30 | BoNaDs | Skipping non-TDM / non-FXO |
18:24.38 | BoNaDs | fxotune outputs that 8 times |
18:25.01 | Katty | anthm: www.copi-rite.com/zapata.txt <- did i goof my channels up at the bottom for that second card somehow? |
18:25.06 | BoNaDs | which i presume is once for each of thr 4 fxo resources on my 2 TDM04b cards |
18:25.12 | ManxPower | BoNaDs: you have a 4-port analog card with RED modules? |
18:25.28 | BoNaDs | i have 2 brand new digium TDM04b cards |
18:25.39 | ManxPower | with red modules? |
18:25.43 | BoNaDs | yes |
18:27.00 | ManxPower | and you can dial out of them with Asterisk? |
18:27.00 | BoNaDs | yes |
18:27.00 | ManxPower | no idea |
18:27.00 | BoNaDs | me either =P |
18:27.00 | ManxPower | You didn't find anything helpful when you searched the mailinglist archives? |
18:27.00 | ManxPower | ~mailinglist |
18:27.00 | jbot | it has been said that mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
18:27.08 | Qwell | ManxPower: how much volume? |
18:27.14 | tzafrir_laptop | strace is s system-calls trace. nothing to do with the stack |
18:27.17 | Qwell | for the web/domain hosting |
18:28.19 | *** join/#asterisk hex_ffff (~0x3e44d@67-41-182-243.slkc.qwest.net) |
18:29.02 | lehel | here is my call: http://pastebin.ca/18471 |
18:29.02 | lehel | joshpbx: the call |
18:29.02 | tzafrir_laptop | jbot, strace is a system-calls tracer. spits out a useful trace to stderr. To trace the command 'cmd with params' use: 'strace cmd with params' |
18:29.03 | jbot | ...but strace is already something else... |
18:29.06 | tzafrir_laptop | ~strace |
18:29.06 | jbot | somebody said strace was now in a .ipk in the bleeding feed. |
18:29.12 | Qwell | heh |
18:29.24 | *** part/#asterisk hex_ffff (~0x3e44d@67-41-182-243.slkc.qwest.net) |
18:29.34 | tzafrir_laptop | jbot, forget strace |
18:29.44 | lehel | how's my call ppl? |
18:29.57 | tzafrir_laptop | jbot, no strace |
18:30.13 | Qwell | tzafrir_laptop: no, strace is x, should work too |
18:30.27 | tzafrir_laptop | jbot, no, strace is a system-calls tracer. spits out a useful trace to stderr. To trace the command 'cmd with params' use: 'strace cmd with params' |
18:30.27 | jbot | okay, tzafrir_laptop |
18:30.28 | joshpbx | lehel: u use stable or cvs heads? |
18:30.45 | *** join/#asterisk Recursion (~0x3e44d@67-41-182-243.slkc.qwest.net) |
18:30.46 | lehel | CVS HEAD |
18:30.54 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E386.versanet.de) |
18:30.56 | Nebukadneza | hi |
18:31.20 | Nebukadneza | is there a way to execute a system command (like "sh /some/path/to/a/shellscript.sh") with asterisk? |
18:31.24 | Nebukadneza | without a agi? |
18:31.27 | Qwell | system |
18:31.29 | lehel | joshpbx: mention: i can make the call in the other way |
18:31.34 | Qwell | show application system |
18:31.42 | tzafrir_laptop | Nebukadneza, help application system |
18:31.52 | tzafrir_laptop | bah, show |
18:32.05 | Nebukadneza | hmm ... okay - ill check |
18:32.06 | Nebukadneza | thanks |
18:32.09 | joshpbx | lehel: find in u default conf and read about codecpriority |
18:32.13 | *** join/#asterisk jarrod (anon@juniperyour.net) |
18:32.13 | BoNaDs | bbib |
18:32.15 | *** part/#asterisk BoNaDs (~theplot@ool-44c53748.dyn.optonline.net) |
18:33.23 | pa | I can formerly announce: asterisk + i4l is now working :-) |
18:33.25 | joshpbx | try with host, if u really dont have any allow=ilbc in u iax conf it can work. |
18:33.46 | Qwell | joshpbx: unless there is an allow=all |
18:34.04 | joshpbx | Qwell: true. |
18:35.04 | jarrod | when using sipusers (realtime config) should i be able to see my sip users via 'sip show users' that are located in sql? |
18:35.25 | pa | yeah, i4l has a bit of echo.. not the best quality i think.. can be lot better with other drivers? |
18:35.56 | *** join/#asterisk meppl (~mephisto@87.193.4.139) |
18:35.57 | *** join/#asterisk bscheller (~none@corpwall.gsi-kc.com) |
18:36.14 | ManxPower | pa: That depends on a lot of things, but other drivers might be better |
18:36.30 | zoa | is this a chan_i4l ? |
18:36.48 | pa | yes, ithink |
18:36.52 | pa | w8 1 moment |
18:37.07 | pa | chan_modem_i4l.so |
18:37.34 | jarrod | does anyone suspect t38 support being built into asterisk for faxing? |
18:37.39 | jarrod | that seems to be the only thing lacking |
18:38.48 | *** join/#asterisk jaimeco (~chatzilla@216.230.138.103) |
18:38.58 | Delta34 | does anybody know what can be causing this msg |
18:39.07 | brettnem | dream on for t38 |
18:39.11 | Delta34 | <PROTECTED> |
18:39.11 | Delta34 | <PROTECTED> |
18:39.20 | jarrod | brett: what do you do for faxing? |
18:39.24 | ronn | i'm using Playtones(ring) in my dialplan but the ringing do not stop when i start dialing extensiions ... any idea? |
18:39.25 | Nebukadneza | hm |
18:39.30 | brettnem | Delta34: perhaps a malformed request |
18:39.38 | brettnem | jarrod. G.711 and a whole lot of faith. |
18:39.50 | mut | anyone know of any scripts/programs to analyze CDR records, stored in a database.. |
18:39.53 | jaimeco | somebody know where I can download a good manual to configurate the sip extensions |
18:40.02 | jarrod | bah i think im going to spring for the cisco voice t1 card to use t38 |
18:40.04 | Delta34 | i get it when i call cisco iphone to iphone sip to sip |
18:40.10 | Nebukadneza | exten => 6265494,1,System(sh /root/stable_01_ast&) << this should execute a script that runs for ... lets say 10 minutes, but proceeds in the asterisk dialplan, right (becaus of the &) |
18:40.20 | brettnem | jarrod: running t.38 through asterisk will crush it |
18:40.52 | jarrod | yea i am just going to use a cisco then |
18:41.12 | brettnem | jarrod: you mean CCM? |
18:41.32 | jarrod | a 7206 with a PA-VXB-2TE1 and t.38 to a t.38 compatible ATA |
18:41.34 | brettnem | I haven't found a good faxing solution yet.. however fax over g.711 doesn't work too bad if you control the network |
18:41.59 | brettnem | jarrod: t.38 implementation vary wildly.. no guarentee that it'll work |
18:42.22 | jarrod | i have seen it work |
18:42.30 | jarrod | usually to another cisco |
18:42.37 | brettnem | well cisco to cisco should work |
18:42.37 | jarrod | im going to experiment with various ATA devices |
18:42.55 | brettnem | let us know how it goes.. I've only seen trouble with t.38 |
18:43.06 | ronn | i'm using Playtones(ring) in my dialplan but the ringing do not stop when i start dialing extensiions ... any idea? |
18:43.07 | joshpbx | somone maybe have idea, why i hear voice ok, but when i record even local it`s very slowly.. when i play it 2 time fasters it`s ok.. |
18:43.19 | ManxPower | ronn: you need StopTones |
18:43.19 | brettnem | ronn, why are you using playtones(ring) ?? |
18:43.24 | jarrod | right now i just forward the inbound faxes out another pri channel to the POTS line installed for the fax heh |
18:43.38 | *** join/#asterisk harryvv (~none@S0106006097af532d.vs.shawcable.net) |
18:43.54 | brettnem | jarrod: embarassingly enough, that's about all I could come up with now for an alternative too.. |
18:44.14 | ronn | brettnem: i wanted to give a dialtone as soon as some one enters an extensions and allow user dial different exten |
18:44.31 | brettnem | ronn, use DISA for that |
18:44.32 | lehel | i can;t imagine.. why is this ilbc codec is active?? i give now disallow=ilbc.. still requested format=ilbc |
18:44.55 | brettnem | lehel: I phone can still request it even if it's disallowed |
18:45.00 | brettnem | er I=a |
18:45.12 | brettnem | ronn: really -> show application DISA |
18:45.26 | ronn | brettnem : DISA .. is that a command? |
18:45.51 | jarrod | this realtime config is pimp++ with the ability to load extensions and sip/iax users into sql |
18:46.06 | brettnem | ronn: it's a dialplan application.. like "playtones" is.. from the CLI type: show application DISA.. it plays dialtone, authenticates (or not) and then connects to an extension in a given priority.. |
18:46.31 | ronn | thanks brettnem, i got it now. |
18:46.45 | *** join/#asterisk darkskiez (~darkskiez@host-84-9-85-42.bulldogdsl.com) |
18:47.02 | brettnem | ronn: It does exactaly what you are talking about.. keep in mind the security problems.. it's meant to give EXTERNAL users INTERNAL dialtone.. but you can do lots with it.. |
18:47.27 | Delvar | jarrod: loading extensions.conf into mysql isnt so good... but the other stuff saves time++ makes it so much easyer for us to handle customers |
18:47.55 | Delta34 | i did a debug and i found this Warning: 399 Bad Request - 'Malformed/Missing FROM: field' |
18:48.13 | Beirdo | hmm |
18:48.14 | *** join/#asterisk frogy (~edmund@cm222-167-86-25.hkcable.com.hk) |
18:48.26 | Beirdo | I might just change to using DISA |
18:48.47 | brettnem | Beirdo: what are you doing now? |
18:48.50 | Beirdo | right now I do the same thing with an Authenticate and then go to the internal context and let them dial |
18:49.01 | brettnem | Delta34: yep |
18:49.04 | *** join/#asterisk CONTRABANDA (~M@213.131.37.202) |
18:49.10 | CONTRABANDA | Hi all |
18:49.11 | Beirdo | pretty much the same effect though |
18:49.20 | Beirdo | but DISA is the "right" way |
18:49.24 | Delta34 | where do you define the FROM: field in a sip conversation |
18:49.27 | brettnem | Beirdo: do you get 2nd dialtone that way>? |
18:49.38 | brettnem | Delta34: what are you dialing from?? what is the UA? |
18:49.49 | CONTRABANDA | How can i get h323 termination calls end then send it to various gateways ? |
18:49.53 | Beirdo | not a dialtone, but I make it say "please enter the extension you wish to dial" |
18:49.54 | Delta34 | cisco 7960 to cisco 7960 |
18:50.09 | brettnem | ah |
18:50.11 | Delta34 | sip to sip local net |
18:50.17 | Beirdo | which can include external calls |
18:50.24 | *** part/#asterisk perlmonky (~perlmonky@pix.benchmark-systems.com) |
18:50.26 | brettnem | Delta34: it's one of the "name" fields when you setup the SIP stuff |
18:50.35 | Delta34 | in sip.conf? |
18:50.49 | bscheller | Has anyone had problems with an asterisk box with Linux 2.6 kernel freezing after running ztcfg? |
18:50.51 | ManxPower | Well THIS isn't good. |
18:51.15 | ManxPower | Prelim indications are that exten => fax does not work inside a macro. Anyone else have this problem? |
18:51.17 | *** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
18:51.32 | obsidian-studios | greetings, what would cause the CALLERID var to be the value specified in a zap channel? I have * set to email me caller id info just after a zap channel is answered. Most times I get the CID/ANI info of the caller via the CALLERID var. However sometimes it returns the zap channels caller id info? |
18:51.37 | Beirdo | never tried, ManxPower |
18:52.02 | ManxPower | obsidian-studios: It didn't get the callerid info then |
18:52.18 | ManxPower | you want usecallerid=yes and callerid=asreceived |
18:52.51 | obsidian-studios | ok, I have another setup using a TDM400p that always returns the zap caller id info in voicemails and etc. I have never been able to get it to get the ANI/CID info. |
18:52.59 | brettnem | bscheller: are you using hdlc data or anything fancy like that? |
18:53.06 | harryvv | This is bad, when calling in on my ivr the calling party gets the typicall ivr to call one of two extentions but then thay get a all circuits are busy please call again. No one is using the phones at the time. What would most likly cause this. |
18:53.25 | brettnem | harryvv: show us the CLI output |
18:53.33 | ManxPower | harryvv: you have been at this long enough to know that we need a pastebin of the cli output to help you |
18:53.47 | harryvv | sure |
18:53.49 | harryvv | i know |
18:53.52 | bscheller | No, I am using pretty standard stuff. The hardware is a T410P. No other cards. Only fix has been to power cycle the machine to recover |
18:54.23 | jarrod | i just need a way for all of the servers in this asterisk environment to read from the same extensions |
18:54.30 | jarrod | the realtime seems like the best way |
18:55.05 | bscheller | brettnem: sorry TE410P |
18:55.10 | brettnem | jarrod: dundi can do some of that too |
18:55.40 | brettnem | bscheller: I had some problems like that with doing hdlc data on asterisk.. never got it to work right.. I'd call digium about it.. |
18:55.43 | obsidian-studios | ManxPower: thanks for the info, we will see how it goes |
18:55.57 | *** join/#asterisk dacrazyz (~www@217.129.208.124) |
18:56.09 | brettnem | they'll want to know about a card locking up a box |
18:56.25 | bscheller | brettnem: thanks. I had seen some weird stuff, but this one tops it. I will give them a call |
18:56.25 | Nebukadneza | hm ... i need to test some sipgate testing ... if anyone of you guys is using sipgate ... could you try calling 6265494 please? :P |
18:56.35 | jarrod | im lookin more to where if one server dies the other has an exact copy of its extensions and not where it queries |
18:56.55 | jarrod | or does 'lookups' on another server and their extensions differ |
18:57.06 | brettnem | ah |
18:57.12 | *** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
18:57.16 | brettnem | yeah.. trying to come up with a good solution for that myself as well.. |
18:57.26 | jarrod | sql is the best way ive found |
18:57.35 | brettnem | unfortunately, I've had way too many deadlocks with all of the asterisk database components.. so I don't trust them |
18:58.14 | _deg_ | Anyone using Unicall with Asterisk? |
18:58.42 | *** part/#asterisk thal (~thalunil@walledcity.de) |
18:58.50 | brettnem | jarrod: maybe you'd have better luck ditching asterisk and going to SER? :) |
18:58.55 | greg_work | jarrod: how about lowtech, like cron + scp |
18:58.59 | greg_work | or rsync |
18:59.23 | brettnem | yeah.. I wrote a rsync + cron utility to centrlize my configs |
18:59.59 | jarrod | that is lame |
18:59.59 | brettnem | hmm what is that config pushing utility out there.. I forget it's name.. very powerful package.. |
19:00.06 | Delta34 | I think the problem is I am getting three "Name" 4088" in the from field, not sure how thats getting inserted into the sip conversation |
19:00.23 | Delta34 | three " in the sip From field |
19:00.29 | brettnem | jarrod: having asterisk hang calls because the network connectivity to the database dies is lame. |
19:00.36 | ManxPower | Delta34: by the SIP device usually. |
19:00.54 | ManxPower | Or you have quotes in your sip.conf, which is bad. |
19:01.02 | brettnem | yeah that's gotta be your phone, I'd think |
19:01.04 | jarrod | if its on the same LAN |
19:01.04 | *** part/#asterisk lehel (~Lehel@82.79.20.17) |
19:01.07 | jarrod | im not worried about that |
19:01.25 | ManxPower | I suspect I'll have to post another bounty to get the fax problem fixed |
19:01.32 | Delta34 | ohh my caller id has "username" extension |
19:01.32 | *** join/#asterisk Ash (~aaron@outofband.org) |
19:01.36 | *** join/#asterisk craziman2 (~donnie@boromir.apid.com) |
19:01.37 | Delta34 | i should remove the " right? |
19:01.43 | brettnem | yes |
19:01.57 | ManxPower | always remove quotes from sip.conf |
19:02.02 | bkw_ | Russia's Biggest Spammer Brutally Murdered <-- see what spamming gets your ass |
19:02.10 | *** join/#asterisk pa (~Paolo@pa.user) |
19:02.13 | brettnem | wow |
19:02.16 | bkw_ | quotes? |
19:02.26 | Ash | has anybody seen this error on debian-stable before? libgcc_s.so.1 must be installed for pthread_cancel to work |
19:02.27 | brettnem | yeah |
19:02.29 | ManxPower | bkw_: Have you ever had an exten => fax inside a macro? |
19:02.30 | brettnem | know any good quotes? |
19:02.38 | bkw_ | ManxPower, I think it works in cvs-head |
19:02.42 | Ash | the only problem is that libgcc_s is installed and ld knows about it |
19:02.53 | bkw_ | callerid="username" <number> <-- is the righ tway |
19:02.54 | ManxPower | bkw_: I think it doesn't in 1.0.9 |
19:02.59 | bkw_ | and is what the callerid parsing is looking for |
19:03.01 | _deg_ | ManxPower, I will do the same bounty to Unicall |
19:03.02 | blitzrage | anyone have an idea why AGI(myscript.agi,1234) would pass the value in HEAD, but not in 1.0.9 ? I just get "" in the same script when calling from stable, as opposed to 1234 in HEAD. |
19:03.19 | ManxPower | bkw_: callerid= doesn't care about quotes. |
19:03.31 | ManxPower | And having them will cause problems with at least some SIP clients. |
19:03.39 | CONTRABANDA | How can i get h323 termination calls end then send it to cisco access server ? |
19:03.42 | brettnem | oh this ought to be good |
19:03.44 | blitzrage | and I think Asterisk puts the quotes around the string for you (you'll get ""name"" if you add quotes) |
19:04.16 | jr352k | hi there!! |
19:04.46 | _deg_ | #asteriskbrasil |
19:04.55 | jr352k | has anyone worked w/ adit 600 CMG card + asterisk? |
19:05.03 | Delta34 | Yes it worked =) thxs manxpower and brettnem |
19:05.11 | Delta34 | using 1.0.9 |
19:05.45 | brettnem | jr352k: I have.. |
19:06.59 | bkw_ | snprintf(buf, bufsiz, "\"%s\" <%s>", name, num); |
19:06.59 | ManxPower | Delta34: bkw_ is USUALLY technically correct. |
19:06.59 | ManxPower | Delta34: but as you can see in this case..... |
19:07.00 | blitzrage | lol |
19:07.05 | brettnem | hmm |
19:07.10 | Delta34 | it must of changed when i upgraded to 1.0.9, was working on 1.0.2 |
19:08.17 | bkw_ | ManxPower, ast_callerid_parse I think isn't as picky about it.. but you should still quote it to stop it from trying to figure out exactly what you ment |
19:09.10 | harryvv | ManxPower take a look at this. have both cli output and the extentions below it. I have come to relize for some odd reason when the calling party calls in on the zap hears the ivr to press one of two extentions that asterisk instead dial up the main phone number the calling party is calling on. I have not made any changes to really reflect this. http://pastebin.ca/18476 |
19:11.18 | ManxPower | bkw_: if you quote it the quotes are passed to the destination SIP device and as Delta34 has seen, some of them don't like that |
19:12.25 | harryvv | manx, ever see a bug where a extention is pressed and it dials the main zap dial un number? |
19:12.34 | ManxPower | harryvv: no. |
19:12.36 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
19:12.43 | harryvv | Thats what this is doing. |
19:12.49 | ManxPower | Since Asterisk doesn't even know what the main zap number is |
19:12.49 | harryvv | never done this before. |
19:13.21 | anthm | i'd say the thing setting callerid should nuke quotes as a policy I recall making a dequote func in app.c |
19:13.29 | anthm | for the gotoif stuff |
19:13.38 | *** join/#asterisk dant (~dan@81-86-69-213.dsl.pipex.com) |
19:13.38 | ManxPower | <PROTECTED> |
19:13.50 | ManxPower | harryvv: the SIP client has call forwarding turned on |
19:13.57 | harryvv | ahhh |
19:14.02 | harryvv | mabey wife did that |
19:14.47 | harryvv | so asterisk does not automaticly send that to the phone company then |
19:15.05 | Nugget | probley not. |
19:15.52 | anthm | http://66.250.68.190/eg/tmp.c |
19:16.06 | harryvv | its a feature my wife wants |
19:16.26 | anthm | data = ast_strip_quoted(data, "\"", "\""); |
19:17.00 | anthm | you can just do that on the name portion and it will strip whitespace and stay inside quotes if the exist and eliminate them |
19:19.07 | ManxPower | Or you can just not use quotes in sip.conf, iax.conf, zapata.conf, etc |
19:19.36 | harryvv | I guess the only way around my case is to route it out to iax.cc or another zap line. |
19:22.55 | anthm | kinda like you can just opt to not hit tab in the cli cos you never know when it will crash your box |
19:23.18 | ManxPower | Exactly! |
19:23.18 | *** join/#asterisk Darwin35 (~darwin@ip70-186-117-198.ma.dl.cox.net) |
19:25.50 | essobi | heh |
19:25.50 | anthm | or always say disallow=all cos it makes one feel they worked around a codec bug => |
19:25.50 | *** join/#asterisk focks (~cbruender@nsc66.147.95-93.newsouth.net) |
19:25.50 | ManxPower | Granted it should be fixed in the source code, but that doesn't help anyone not running CVS-HEAD |
19:25.50 | Darwin35 | ManX did it I saw him |
19:25.50 | ManxPower | anthm: I do that too. |
19:25.50 | focks | what would cause sporatic echo on a PRI Zap channel? Using Polycom Soundpoint 501 |
19:25.50 | jarrod | brett: do you use ser? |
19:25.50 | anthm | well I suggested that code cos it could be easily patched into stable |
19:25.50 | ManxPower | these days I do allow=all in [general] then disallow=all and allow=myhappycodec in each sip.conf section |
19:25.50 | anthm | it's like a 10 line func in utils.c |
19:25.50 | Dovid | hello all |
19:25.50 | ManxPower | focks: lower your txgain on the pri. the exho is caused by the remote analog line |
19:25.50 | Dovid | anyone know of any software for phone spoofing ? |
19:25.50 | anthm | when you get to 1.2 you will be in for a treat cos I reworked all the codec stuff to work right |
19:25.50 | focks | ManxPower, that's in zaptel.conf right? |
19:25.51 | harryvv | dovid, you mean cid spoofing? |
19:25.51 | ManxPower | anthm: Hmm? How do? Does allow=all no longer say Asterisk supports G723.1 or G729 (without a license)? |
19:25.51 | *** join/#asterisk ai-a[afk] (~gandalfii@81.168.0.204) |
19:25.51 | Dovid | harryvv: yes |
19:25.51 | anthm | allow=all means whatever you said in general now |
19:25.51 | ManxPower | Dovid: there is no such application. I'm sorry. |
19:25.59 | Dovid | kk |
19:26.02 | harryvv | I suspect register with a iax service that allows you to set your own cid. |
19:26.06 | ManxPower | anthm: Um, it seems to me that that happens in 1.0.x |
19:26.09 | Dovid | cause i know people that did it with asterisk |
19:26.20 | Nebukadneza | exten => 6265494,1,System(sh /root/stable_01_ast&) |
19:26.20 | anthm | so you setup general the way you like it and it's inherited to each peer |
19:26.20 | Dovid | but i want the caller to set it when they call |
19:26.23 | Nebukadneza | hm .. why doesnt this work |
19:26.24 | anthm | maybe that made it in stable |
19:26.27 | Nebukadneza | it seems its just not executed |
19:26.30 | anthm | i get lost on the timeline |
19:26.43 | tzafrir_laptop | Nebukadneza, * runs as root? bad |
19:26.50 | Nebukadneza | i know :P |
19:26.57 | Nebukadneza | this is going to change soon ... |
19:27.06 | anthm | if show codecs has this |
19:27.08 | anthm | <PROTECTED> |
19:27.17 | anthm | it the top its got my patch |
19:27.25 | Nebukadneza | but still ... why isnt this working? |
19:27.27 | Dovid | ManxPower: I just found an AGI for it |
19:27.57 | ManxPower | anthm: the entire reason I have allow=all in [general] is because disallow=all in [general] not allow ANY codecs in each sip.conf stanza |
19:29.45 | anthm | with the new way, if you allow=a,b,c,d instead of all it preserves the order |
19:29.50 | *** part/#asterisk dan2_ (~foobar@dan2.professional.supporter.pdpc) |
19:30.32 | anthm | it works better in iax than sip cos mark vetoed the bit that enforced the order based on the config in favor of the stupid sip way |
19:30.56 | Delta34 | in the sip show peers, how do u get the status code to "monitored" mine shows as Unmonitored |
19:30.57 | anthm | but in iax you have a ton of control over codec negotiation now |
19:32.29 | ManxPower | Delta34: qualify=yes in each sip.conf entry |
19:32.36 | *** join/#asterisk R3DB0x (nobody@66.142.28.36) |
19:33.19 | Delta34 | i have that set in global settings for sip.conf |
19:33.27 | Delta34 | doesnt it apply to all sip clients? |
19:36.56 | ManxPower | Delta34: it might in cvs-head, but not in 1.0.x |
19:37.38 | twisted[asteria] | anyone know if zapata can successfully be read from realtime? |
19:38.22 | ManxPower | twisted: it doesn't really matter since chan_zap.so has to be unloaded and reloaded anyway |
19:38.47 | zoa | does somebody know what esf is exactly ? |
19:38.57 | zoa | extended super frame, but what does it do ? |
19:39.02 | zoa | and how does it relate to asterisk ? |
19:39.19 | anthm | isnt that one of the pri choices ? |
19:39.22 | twisted[asteria] | zoa, it's t1 framing |
19:39.30 | ManxPower | zoa: ESF is what lets you have 64k clear channel on T-1 |
19:39.40 | anthm | yah that's it , t1 the physical layer option |
19:39.41 | twisted[asteria] | ManxPower, no it doesnt, ninny, I made chan_zap reloadable last year |
19:39.47 | twisted[asteria] | reload chan_zap.so |
19:40.00 | twisted[asteria] | it will reload everything except the signalling |
19:40.03 | ManxPower | twisted: |
19:40.05 | ManxPower | pbx-1*CLI> reload chan_zap |
19:40.05 | twisted[asteria] | and modify the settings accordingly |
19:40.05 | ManxPower | pbx-1*CLI> |
19:40.06 | anthm | its the one you pick between d4 and esf it's usually on channel banks as one or the other |
19:40.12 | ManxPower | twisted: nifty. |
19:40.16 | twisted[asteria] | chan_zap.so |
19:40.23 | twisted[asteria] | don't forget the .so |
19:40.31 | ManxPower | pbx-1*CLI> reload chan_zap.so |
19:40.32 | ManxPower | pbx-1*CLI> |
19:40.32 | jarrod | anyone running a multiple server environment? |
19:40.35 | ManxPower | nothing. |
19:40.38 | twisted[asteria] | ManxPower, heh... what version? |
19:40.40 | jarrod | with multiple entry points and config sharing |
19:40.41 | *** join/#asterisk pfn (~pfnguyen@netblock-66-245-252-239.dslextreme.com) |
19:40.42 | ManxPower | Does it reload with no output? |
19:40.44 | MikeJ[Laptop] | that's silly that it needs the .so.. silly |
19:40.45 | ManxPower | twisted: 1.0.x |
19:40.46 | twisted[asteria] | no, it gives output |
19:40.52 | twisted[asteria] | but remember, realtime isn't in stable |
19:40.56 | twisted[asteria] | and that may not be either |
19:41.00 | ManxPower | Specifically 1.0.9 |
19:41.05 | ManxPower | twisted: *nod* |
19:41.11 | zoa | is it like a specific version of AMI and B8ZS ? |
19:42.50 | Katty | twisted[asteria]: WAKE UP |
19:43.12 | twisted[asteria] | Katty, i'm awake |
19:43.14 | twisted[asteria] | barely |
19:43.15 | twisted[asteria] | heh |
19:43.20 | MikeJ[Laptop] | :) |
19:43.41 | Katty | eww :< |
19:43.44 | Katty | i got 8ish |
19:43.46 | twisted[asteria] | yeah, it sucks |
19:43.52 | twisted[asteria] | i usually get anywhere from 6-8 |
19:43.57 | twisted[asteria] | but tonight i feel will be a 12 hour night ;) |
19:44.49 | Katty | yum. |
19:45.18 | Katty | i'll spend it for you |
19:45.20 | file | I got a new geek toy today but I can't play with it until work is over :( makes me sad |
19:45.21 | Katty | so you feel important, etc. |
19:45.28 | Katty | file: :< |
19:45.45 | Katty | file: :> |
19:45.52 | file | haha |
19:46.04 | twisted[asteria] | Katty, hmm? spend what? |
19:46.16 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
19:46.22 | Katty | twisted[asteria]: your 12 hour night's worth of pay :> |
19:46.40 | twisted[asteria] | oh, i meant 12 hour night's worth of sleep :P |
19:46.51 | Corydon-w | file: yeah, I think the boss would look poorly upon playing with a dildo at the office... |
19:46.52 | Katty | oh |
19:46.55 | Katty | welllllll |
19:46.56 | twisted[asteria] | i'm salaried, i get paid the same regardless |
19:46.57 | file | haha |
19:46.57 | Katty | nevermind then! |
19:47.01 | Katty | i'll go spend my own money! |
19:47.04 | Katty | and yours too |
19:47.07 | MikeJ[Laptop] | Corydon-w, bad! |
19:47.09 | jarrod | does a module need to be loaded for extconfig.conf to be parsed |
19:47.12 | jarrod | or is that default |
19:47.25 | Katty | Corydon-w: dirty. |
19:47.32 | *** join/#asterisk neonet2006 (~icechat5@iphost-64-56-140-28.wpg.wiband.net) |
19:47.36 | neonet2006 | hello |
19:47.44 | twisted[asteria] | Katty, if you can find a way to spend my money, heh, you'd deserve it |
19:47.46 | neonet2006 | how can set the npi and ton in asterisk |
19:47.49 | Corydon-w | Katty: not with proper soap and water cleaning, it won't be... |
19:48.05 | pfn | aw shit, I deleted my old box and deleted my asterisk diffs, boooo |
19:48.24 | Corydon-w | Now you know why most of us submit patches... |
19:48.25 | Katty | twisted[asteria]: i shall attempt sleeve tugging, and then the pout. |
19:48.31 | Katty | twisted[asteria]: if that does not work, then i shall whine. |
19:48.38 | Katty | twisted[asteria]: and if that does not work, then i shall tickle |
19:48.43 | Corydon-w | It's not because we feel like contributing; it's so we don't lose them |
19:48.48 | Corydon-w | ;-) |
19:48.51 | Katty | twisted[asteria]: there's a whole list of tricks ;) |
19:49.26 | neonet2006 | how can change the ton and npi in asterisk? |
19:49.34 | twisted[asteria] | Katty, hehe.. sleeve tugging and pouting usually only gets evil looks and the occasional head patting |
19:49.37 | Corydon-w | Katty: you forgot finding twisted wallet while he's sleeping... |
19:49.42 | twisted[asteria] | whining usually makes me turn on the ipod |
19:49.47 | twisted[asteria] | and tickling usually makes me tickle back |
19:50.03 | Corydon-w | Ooooh... I can get twisted to tickle me? |
19:50.12 | Katty | twisted[asteria]: what about The Whimper(tm)? |
19:50.13 | twisted[asteria] | Corydon-w, uhm... no |
19:50.30 | twisted[asteria] | Katty, lol... i haven't had that one tried on me yet |
19:50.34 | Katty | twisted[asteria]: k |
19:51.53 | neonet2006 | ?any help? |
19:53.00 | Katty | my jewish blood is quite handy at times (= |
19:53.09 | Katty | and bloody annoying at others |
19:53.37 | Corydon-w | neonet2006: pridialplan in zapata.conf |
19:54.22 | focks | I've got Polycom SoundPoint 501's setup to auto-answer for 4 digit internal extensions. Problem is, when I perform a transfer (blind or supervised) it doesn't distinguish between a transfer and dialing normal internal extension to intercom. How can I work around this? |
19:54.28 | Corydon-w | That specifies the outgoing TON |
19:54.28 | *** join/#asterisk Jearil (~Jearil@67.151.27.214) |
19:54.36 | twisted[asteria] | Katty, you're jewish? |
19:54.52 | focks | aside from assigning a second extension like 8XXXX for intercom and leaving XXXX for normal dialing |
19:54.59 | neonet2006 | is there a way to force the ton and npi in the extensions.conf |
19:55.06 | neonet2006 | for specific dial plans |
19:55.17 | Katty | twisted[asteria]: about 45% german, 45% irish, and 10% jewish |
19:55.22 | Corydon-w | neonet2006: why would you want to? |
19:55.23 | twisted[asteria] | hehe word |
19:55.24 | *** part/#asterisk bscheller (~none@corpwall.gsi-kc.com) |
19:55.25 | tzanger | Katty: rearrange some over to mine |
19:55.25 | *** part/#asterisk illek (~mike@ip68-227-104-152.ok.ok.cox.net) |
19:55.29 | Katty | twisted[asteria]: you could call me aggressively thrifty (= |
19:55.46 | Katty | my savings account is fluffy |
19:55.49 | Katty | but i have no credit |
19:55.51 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
19:55.52 | twisted[asteria] | Katty, i'm 25% irish, 25% scottish, 25% english, and 25% cherokee indian |
19:55.56 | ManxPower | zoa: http://en.wikipedia.org/wiki/Digital_Signal_1 |
19:56.02 | Katty | soon i shall have credit card and then i shall aquire credit |
19:56.06 | neonet2006 | i would like to set specific caller ID Number with proper TON and NPI to be terminated on another gateway |
19:56.21 | Katty | twisted[asteria]: k (= |
19:56.29 | Katty | my hair has streaks of red in it |
19:56.36 | Katty | i guess that's called auburn |
19:56.36 | file | Katty rocks my world |
19:56.53 | Katty | ..! |
19:56.57 | twisted[asteria] | Katty, hehe, yeah, my hair is like a dirty red, and my facial hair is red. |
19:57.05 | Katty | twisted[asteria]: post gifs :> |
19:57.06 | twisted[asteria] | whoa |
19:57.09 | file | death by ticklking? |
19:57.10 | twisted[asteria] | Katty, there are some already |
19:57.11 | Darwin35 | my replacement and new x401 phones are on thier way |
19:57.11 | file | er tickling? |
19:57.14 | Katty | twisted[asteria]: post url |
19:57.16 | Darwin35 | 2 phone |
19:57.17 | twisted[asteria] | not gifs, but jpegs |
19:57.19 | Katty | twisted[asteria]: k |
19:57.20 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
19:57.27 | twisted[asteria] | Katty, http://gallery.indigent-networks.com |
19:57.29 | twisted[asteria] | somewhere in there.. |
19:57.31 | Katty | file: death by SOY |
19:57.34 | Katty | twisted[asteria]: k |
19:57.37 | file | ooh |
19:57.55 | ManxPower | zoa: or http://www.dcbnet.com/notes/9611t1.html |
19:58.07 | izo | anybody using ata 186 ? |
19:58.17 | Katty | twisted[asteria]: in friends? |
19:58.35 | Darwin35 | now to get the firmware src code |
19:58.41 | neonet2006 | ?? |
19:58.51 | twisted[asteria] | Katty, nah, that's mostly friends of mine. look in VON for one or two |
19:58.55 | Katty | twisted[asteria]: k |
19:58.56 | twisted[asteria] | i know there's a good one of me and drumkilla |
19:59.22 | pa | I have a debian dist, and i changed in /etc/defaults/asterisk RUNASTERISK from no to yes, but if i issue "/etc/init.d/asterisk start", it does not start... :-/ |
19:59.25 | Corydon-w | neonet2006: pridialplan=dynamic |
19:59.50 | neonet2006 | and how do i set the NPI and TON in extensions.conf |
20:00.05 | pa | i can still launch it by hand.. |
20:00.27 | Katty | twisted[asteria]: oooh, pretty female at VON :> |
20:00.42 | pa | i could add it to inittab, but i think that letting start it to rc is better |
20:00.45 | *** join/#asterisk darkskiez (~mhb@host-84-9-85-42.bulldogdsl.com) |
20:00.51 | twisted[asteria] | Katty, hmm? |
20:00.51 | ManxPower | Corydon-w: What version of Asterisk supports pridialplan=dynamic? |
20:01.45 | Corydon-w | ManxPower: dunno... it's been in there a while |
20:01.50 | neonet2006 | and how do i set the NPI and TON in extensions.conf |
20:01.51 | ManxPower | pa: can you start Asterisk with "asterisk -cvvv"? |
20:02.18 | Qwell | hmm |
20:02.19 | Corydon-w | Somebody's switch wouldn't support setting pridialplan=unknown, so he went ahead and made it work |
20:02.26 | *** part/#asterisk Nebukadneza (~daddel9@i3ED6E386.versanet.de) |
20:02.32 | ManxPower | Corydon-w: Well it's not documented in zapata.conf.sample in 1.0.9 |
20:02.41 | Qwell | forget this whole "bootup services" stuff |
20:03.01 | Corydon-w | ManxPower: Dunno... I don't run 109 |
20:03.15 | Corydon-w | ManxPower: I don't use it, either |
20:03.30 | Katty | twisted[asteria]: neat sunglasses |
20:03.46 | ManxPower | Then it's prolly not in 1.0.x |
20:04.16 | pa | ManxPower: yes |
20:04.27 | twisted[asteria] | Katty, thx |
20:04.36 | Corydon-w | ManxPower: yep, it's in HEAD only |
20:04.38 | pa | ManxPower: asterisk -vvvc works great |
20:04.39 | neonet2006 | so no doc for this feature? |
20:04.48 | Corydon-w | ManxPower: ain't the source grand? |
20:06.34 | ManxPower | neonet2006: what version of Asterisk are you running |
20:06.44 | Corydon-w | neonet2006: who needs documentation when you have the source? |
20:07.15 | Darwin35 | coool |
20:07.29 | Darwin35 | I will have 2 new phones this rocks |
20:07.45 | neonet2006 | 1.0.8 |
20:07.58 | neonet2006 | i still need documentation..... |
20:08.11 | neonet2006 | a project with docs....means nothing |
20:08.12 | Darwin35 | for what |
20:08.15 | ManxPower | neonet2006: I strongly doubt that feature is supported in 1.0.8 |
20:08.24 | ManxPower | neonet2006: You're not running BRIStuff, are you? |
20:08.26 | Corydon-w | ManxPower: in fact, it is not |
20:08.34 | neonet2006 | no I am not running it |
20:09.05 | Corydon-w | Forget support... it's not even in 108 |
20:09.09 | neonet2006 | what I am trying to do is set the ton and npi when I force the dial plan to show specific callerID number |
20:09.16 | ManxPower | Apparently Wait(1) will trigger a faxdetect if it's run |
20:09.18 | neonet2006 | SetCallerID() |
20:09.23 | ManxPower | (and a fax machine is calling) |
20:09.38 | neonet2006 | is there a way to do it in the dial plan (extensions.conf) |
20:09.55 | *** join/#asterisk T-Squared (~ted@hidden.serreyn.com) |
20:09.58 | ManxPower | neonet2006: 1) Why? and 2) I don't think you can do what you want to do using the version of Asterisk you are using. |
20:10.42 | Corydon-w | neonet2006: no, there is no way to do that with the version you're using |
20:11.08 | neonet2006 | I am trying to terminate on a gateway (hooked up to a PRI) the PRI provider needs to see specific ton and npi |
20:11.12 | neonet2006 | national/isdn |
20:11.20 | neonet2006 | what version should I get? |
20:11.35 | ManxPower | neonet2006: What happens if you just set pridialplan=unknown? |
20:11.40 | zoa | what is npi and ton ? |
20:11.57 | ManxPower | zoa: You saw my ESF links? |
20:12.20 | neonet2006 | i am not originating the calls on the zaptel cards |
20:12.27 | neonet2006 | stricly SIP |
20:12.30 | ManxPower | neonet2006: %90 of people's TON problems is that they set prodialplan= something other than "unknown" |
20:12.31 | zoa | yeah thanks |
20:12.53 | ManxPower | neonet2006: You can't send that information over SIP as far as I know. |
20:13.09 | neonet2006 | that what I think as well |
20:13.31 | neonet2006 | so I need to be using zaptel hardware to get it working or what? |
20:14.00 | neonet2006 | the info as far as i know is related from one gateway to the other? no SIP intervention? |
20:16.16 | neonet2006 | how would transmitt that info, then? |
20:16.24 | Goshen | Nufone.net crapping out again? |
20:16.24 | zoa | twisted, do you have a link explaining that ESF makes sure you can use 64kbit ? |
20:16.47 | Goshen | the voice is really choppy again, like it was when they were having problems last week.... |
20:16.48 | *** join/#asterisk PakiPenguin (uppal@202.61.58.73) |
20:17.05 | *** join/#asterisk __kop__ (~kop@71-35-174-95.tukw.qwest.net) |
20:17.17 | DarthClue | Goshen: if it's a tollfree, come to asterlink. |
20:17.46 | Goshen | url? |
20:17.50 | *** join/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca) |
20:17.59 | DarthClue | http://www.asterlink.net |
20:18.02 | TripleFFF2sdf | hey.. |
20:18.21 | Goshen | Darthclue: not resolving |
20:18.24 | DarthClue | Goshen: if you're interested, msg me and we can get you on the express route. |
20:18.37 | pa | i think i will let asterisk start from inittab, so i can redirect it to a virtual terminal |
20:18.49 | Goshen | DarthClue: asterlink.com? |
20:18.52 | DarthClue | Goshen: sorry, need more caffiene, asterlink.com |
20:19.12 | neonet2006 | how would transmitt that info, then? |
20:19.13 | darkskiez | pa, safe_asterisk script does all that for you |
20:24.37 | pa | darkskiez: but i cant start it from rc script :-( |
20:24.40 | neonet2006 | is there a way to see what npi and ton is trasmitted from asteriks |
20:24.48 | neonet2006 | how do i enable the log |
20:25.00 | neonet2006 | what log should i enable |
20:25.01 | TripleFFF2sdf | ./usr/lib64/gcc-lib/x86_64-suse-linux/3.3.3/../../../../x86_64-suse-linux/bin/ld: cannot find -lssl |
20:25.01 | TripleFFF2sdf | collect2: ld returned 1 exit status |
20:25.06 | TripleFFF2sdf | hmm |
20:25.25 | file | you don't have the library |
20:26.07 | *** join/#asterisk L|NUX (linux@202.5.146.154) |
20:26.19 | TripleFFF2sdf | yes itsthere |
20:27.05 | ManxPower | TripleFFF2sdf: in /usr/lib ? |
20:27.06 | *** join/#asterisk craziman2 (~donnie@boromir.apid.com) |
20:27.13 | TripleFFF2sdf | y |
20:27.19 | TripleFFF2sdf | added |
20:27.20 | TripleFFF2sdf | -ldl -lpthread -lncurses -lm -lresolv -L /usr/local/lib -L /usr/lib -L/usr/pkg/lib -lssl |
20:27.20 | ManxPower | or somewhere else that's listed in /etc/ld.so.conf |
20:27.36 | ManxPower | can't have a space after -L I don't thin |
20:27.40 | TripleFFF2sdf | hmm |
20:28.08 | TripleFFF2sdf | fixed that |
20:28.10 | TripleFFF2sdf | but still |
20:28.17 | file | it can't find it... |
20:28.25 | TripleFFF2sdf | now.. b1-ast/libdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv -L/usr/local/lib -L/usr/lib -L/usr/pkg/lib -lssl |
20:28.38 | kFuQ | <PROTECTED> |
20:28.59 | ManxPower | TripleFFF2sdf: well put the directories in /etc/ld.so.conf and rerun ldconfig -v then you don't have to put all that extra shit in the makefile |
20:32.44 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
20:33.01 | *** join/#asterisk BleedingMe (~BleedingM@ppp-69-238-171-80.dsl.scrm01.pacbell.net) |
20:33.56 | TripleFFF2sdf | still no go |
20:33.57 | TripleFFF2sdf | oh |
20:33.59 | TripleFFF2sdf | hmm |
20:34.42 | BleedingMe | does anyone know anything about www.iax.cc / www.sixtel.net ? |
20:34.50 | TripleFFF2sdf | would suse compile as ifeq (${OSARCH},SunOS) ? |
20:35.06 | ManxPower | TripleFFF2sdf: If it does, it would be a major bug in Suse |
20:35.43 | *** join/#asterisk loick (~loick@APuteaux-151-1-20-68.w82-124.abo.wanadoo.fr) |
20:36.02 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
20:36.46 | TripleFFF2sdf | ahah |
20:40.02 | TripleFFF2sdf | still not |
20:41.28 | file | there's a readme for suse... |
20:41.42 | file | might yield some info |
20:42.31 | TripleFFF2sdf | k |
20:42.38 | TripleFFF2sdf | ./usr/local/ssl/lib/ is where it is |
20:42.44 | TripleFFF2sdf | and thats in ld.conf |
20:42.52 | TripleFFF2sdf | ,res_crypto.c:556: warning: implicit declaration of function `SSL_library_init' |
20:42.52 | TripleFFF2sdf | res_crypto.c:557: warning: implicit declaration of function `ERR_load_crypto_strings' |
20:42.52 | TripleFFF2sdf | make[1]: *** [res_crypto.o] Error 1 |
20:42.52 | TripleFFF2sdf | make[1]: Leaving directory `/usr/local/src/asterisk/res' |
20:42.53 | TripleFFF2sdf | now |
20:42.54 | TripleFFF2sdf | lol |
20:43.02 | TripleFFF2sdf | man this IS not a plug and play |
20:43.04 | TripleFFF2sdf | ;l) |
20:43.54 | *** join/#asterisk wunderkin (~kev@12-215-218-160.client.mchsi.com) |
20:43.58 | mishehu | TripleFFF2sdf: I put my monopoly money on the problem being something to do with includes. |
20:45.04 | TripleFFF2sdf | lol yeah |
20:45.17 | TripleFFF2sdf | thing is. |
20:45.29 | TripleFFF2sdf | is i add that dir to libs of the ifeq (${OSARCH},SunOS) then i goest a bi further but dumps |
20:48.09 | *** join/#asterisk jsmith (~jsmith@smithfam.dsl.xmission.com) |
20:52.29 | TripleFFF2sdf | hmm if i addd -L/usr/lib |
20:52.37 | TripleFFF2sdf | ./usr/lib64/gcc-lib/x86_64-suse-linux/3.3.3/../../../../x86_64-suse-linux/bin/ld: skipping incompatible /usr/lib/libdl.so when searching for -ldl |
20:52.40 | TripleFFF2sdf | i get these |
20:52.43 | TripleFFF2sdf | 64 |
20:52.52 | TripleFFF2sdf | so .. i trie /use/lib64 then no luck at all |
20:53.14 | RaYmAn-Bx | how is adding libraries to the command line going to help with problems finding declarations that has to do with includes? :> |
20:53.25 | TripleFFF2sdf | not sure |
20:53.27 | TripleFFF2sdf | lol |
20:53.39 | TripleFFF2sdf | also its says while searching for searching for -lm |
20:53.43 | TripleFFF2sdf | well |
20:53.49 | TripleFFF2sdf | im at a dead end then |
20:53.52 | TripleFFF2sdf | any dsuggestio ? |
20:54.55 | RaYmAn-Bx | Do you have a library called "libssl.so" in any of the paths you pass? Like, check that it's actually there and stuff |
20:55.32 | TripleFFF2sdf | <PROTECTED> |
20:55.52 | TripleFFF2sdf | <PROTECTED> |
20:55.58 | puzzled | anyone use MMX optimizations enabled in zconfig.h in zaptel (stable)? positive difference? |
20:56.07 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
20:56.27 | zoa | dunno if there is a difference, but it doesnt seem to crash :) |
20:56.34 | puzzled | hehe |
20:56.46 | puzzled | zoa: so you usually have it enabled on prodction boxes? |
20:57.29 | jsmith | puzzled: I've been using it on production boxes for years. |
20:57.52 | *** join/#asterisk asterisk99 (~chatzilla@modemcable111.209-131-66.mc.videotron.ca) |
20:57.54 | puzzled | jsmith: ok thanks |
20:57.56 | file[laptop] | oh no... IT'S JARED SMITH! |
20:58.01 | RaYmAn-Bx | TripleFFF2sdf: was that an answer or? Because it doesn't really give any information whatsoever... |
20:58.25 | jsmith | file[laptop]: That would be me -- scary, isn't it? |
20:58.30 | file[laptop] | jsmith: quite |
20:58.39 | mut | there a way to turn off verbosity no the cli at all? |
20:58.42 | mut | set verbose 0 |
20:58.45 | jsmith | Good thing they're not putting my picture on the book... |
20:58.50 | mut | and i still see warnings when i reload the config |
20:58.53 | file[laptop] | jsmith: it would scare people away |
20:58.54 | asterisk99 | anyone know how to specify specific days of week in GotoIfTime (e.g. Mon+Sat+Sun) ???? |
21:02.08 | Meaty | GotoIfTime(*|sat-mon|*|*?context,extension,priority) |
21:02.13 | Meaty | asterisk99 ^^ |
21:02.27 | Meaty | or Maybe GotoIfTime(*|sat-sun-mon|*|*?context,extension,priority) |
21:02.38 | Meaty | dunno |
21:02.46 | Meaty | test it ! |
21:02.59 | jsmith | asterisk99: If I remember correctly, it's mon,tue,wed |
21:03.09 | ManxPower | It uses the same syntax as include => |
21:03.11 | *** part/#asterisk mkrufky (~mk@68.160.103.77) |
21:03.49 | asterisk99 | Meatty: sat-sun-mon is invalid |
21:03.57 | asterisk99 | Meaty: sat-sun-mon is invalid |
21:04.09 | Recursion | jsmith: You're West's cousin, right? |
21:04.11 | Meaty | sat-mon ? |
21:04.21 | asterisk99 | jsmith: can't use commas either |
21:04.32 | Meaty | ? |
21:04.43 | ManxPower | perhaps it's 0123 for sun,mon,tue |
21:04.44 | Meaty | yes |
21:04.47 | Meaty | \, |
21:04.55 | jsmith | Recursion: Man, I'm famous! |
21:04.56 | ManxPower | Or you can look up the syntax for include => |
21:04.59 | jsmith | Recursion: Yes, I am. |
21:05.09 | Recursion | jsmith: soon to be even more famous! |
21:05.20 | Meaty | try sat-mon |
21:05.41 | jsmith | Recursion: Exactly! |
21:06.06 | ManxPower | day-day is the correct formaty |
21:06.30 | jsmith | ManxPower: So if I want Monday, Friday, and Saturday, how do I do that? |
21:06.42 | ManxPower | jsmith: Did you look it up? |
21:06.47 | jsmith | Not yet... |
21:06.54 | ManxPower | Perhaps now is the time to do so. |
21:06.58 | darkskiez | jsmith: you could always do multiple lines of gotoiftime |
21:07.00 | asterisk99 | jsmith: it'll do range, but nothhing else |
21:07.17 | *** join/#asterisk criptos (~criptos@201.138.231.189) |
21:07.28 | asterisk99 | jsmith: yeah.... looks like I have to tetest 1 day at a time |
21:07.35 | criptos | anyone using snom190 phones? I want to know how, from asterisk, specify the ring tone to use... |
21:07.38 | jsmith | I was afraid of that... |
21:07.41 | Delta34 | so i removed the " " in the callerid field under sip.conf but now my direct to voicemail buton is not working now, i guess it is looking for "Username" 4088 but its getting Username 4088 instead |
21:07.56 | Delta34 | the voicemail button now is asking for extension then pwd |
21:08.02 | Delta34 | rather then just asking for pwd |
21:08.09 | ManxPower | Delta34: that's a callerid problem. |
21:08.18 | ManxPower | What is the Voicemailmain line show on the CLI? |
21:09.12 | Delta34 | exten => 4199,1,VoicemailMain(${CALLERIDNUM}@default) |
21:09.20 | *** join/#asterisk santiago (~santiago@63.245.86.188) |
21:09.37 | ManxPower | No, from the CLI! And your voicemail button should be dialing 4199 |
21:09.56 | *** join/#asterisk zotz (~zotz@24.231.36.100) |
21:10.01 | ManxPower | i.e. AFTER CALLERIDNUM is replaced with the correct value during the dialplan processing |
21:10.03 | Delta34 | whats the command in CLI |
21:10.24 | darkskiez | wouldnt be hard to rewrite the get_dow function in pbx.c to take comma seperated days. |
21:10.26 | ManxPower | no, reproduce the problem. You'll see in the CLI the voicemailmain being run |
21:10.48 | Delta34 | ohhh one sec |
21:10.53 | ManxPower | darkskiez: Since Asterisk replaces "," with "|" internally, it's a lot harder than you think. |
21:11.29 | darkskiez | ManxPower: err, fuck. |
21:11.43 | ManxPower | Whenever I need to concatinate things I do it using & |
21:11.48 | ManxPower | Like Dial and others do. |
21:11.53 | Delta34 | Executing VoiceMailMain("SIP/4088-c09f", "Jackie Lau 4088@default") in new stack |
21:12.12 | ManxPower | Delta34: Egads! It should be 4088 right? |
21:12.21 | ManxPower | What have you done do your callerid? |
21:12.29 | ManxPower | Delta34: paste the sip.conf line for callerid= |
21:12.44 | darkskiez | wouldnt be hard to rewrite the get_dow function in pbx.c to take & seperated days. |
21:12.47 | darkskiez | :) |
21:12.50 | Delta34 | callerid=Jackie Lau 4088 |
21:12.53 | ManxPower | darkskiez: yeah |
21:12.56 | Delta34 | i removed " " |
21:13.08 | ManxPower | Delta34: callerid=Jackie Chan <4088> |
21:13.10 | ManxPower | Notice the <> |
21:13.17 | ManxPower | whatever is in <> should be CALLERIDNUM |
21:13.49 | ManxPower | It's callerid=calleridname <calleridnum> |
21:15.18 | jr352k | guys is there any way on the dial plan to limit the number of digits a user enters in order to get them capture: ex dial your 8 digits pin number |
21:15.36 | Hmmhesays | when did chanspy make it back into head? |
21:15.48 | ManxPower | Hmmhesays: Dunno. Was news to me too a few weeks ago |
21:15.56 | anthm | when did it leave ? |
21:16.15 | Hmmhesays | hrm...i'm pretty sure it wasn't in there for awhile |
21:16.27 | Delta34 | cool thxs manxpower |
21:16.28 | anthm | it's been there since it was added |
21:16.34 | Delta34 | was a syntax error on my part |
21:16.37 | Hmmhesays | hrm, I must be losing my mind |
21:16.43 | Hmmhesays | which is completely possible |
21:16.58 | anthm | revision 1.1 |
21:16.58 | anthm | date: 2005/03/24 01:19:02; author: markster; state: Exp; |
21:16.58 | anthm | Add chanspy (bug #3836) |
21:17.35 | Hmmhesays | hasn't chanspy been around a lot longer than that? |
21:17.52 | anthm | not in CVS |
21:18.01 | anthm | on my pc it's been there for a year |
21:18.27 | anthm | you'll want 2005/07/20 or better for the best version of it |
21:18.29 | ManxPower | Hmmhesays: I recommend doing a "show applications" ever few months 8-) |
21:19.18 | Hmmhesays | of course |
21:19.27 | anthm | there is also app_muxmon on deck which is a call recorder that uses the chanspy hooks |
21:20.44 | *** join/#asterisk caramb1 (~alfons@c-f8ef70d5.09-237-73746f34.cust.bredbandsbolaget.se) |
21:21.35 | ManxPower | I really like being a consultant. |
21:22.04 | ManxPower | I just called up my largest customer, asked if anything was happening this afternoon, he said there isn't, so I said I was taking the rest of the afternoon off. |
21:22.39 | caramb1 | I read that using mpg123 is not recommended. So I wonder, is there a gstreamer module for asterisk? That would be far more versatile than mpg123. |
21:22.50 | ManxPower | caramb1: where did you read THAT? |
21:22.52 | file[laptop] | mpg321 is not recommended |
21:22.55 | file[laptop] | mpg123 IS recommended |
21:22.56 | puzzled | you can use madplay too |
21:23.03 | file[laptop] | 0.59r to be exact |
21:23.09 | file[laptop] | or you can use native audio if you really want... |
21:23.12 | file[laptop] | or something else... |
21:23.19 | ManxPower | file: You sure are getting smart in your old age. |
21:23.32 | file[laptop] | haha |
21:23.47 | puzzled | -O6 optimization in asterisk. do people leave that in or tone it down to something less aggressive? |
21:23.59 | SwK[Work] | most people leave it |
21:24.01 | puzzled | (in stable) |
21:24.08 | ManxPower | puzzled: I never touch the makefile, except to enable the option to make G726 work with SIPura |
21:24.14 | Nugget | that's nutty. I'd never noticed the -O6 |
21:24.22 | emrah | Hello again everyone |
21:24.26 | file[laptop] | make thing go fasta! |
21:24.35 | SwK[Work] | Nugget: doesnt d.net use like -O99? heh |
21:24.45 | puzzled | Nugget: line 66 of the Makefile |
21:24.52 | Nugget | no, d.net doesn't use compiler optimization |
21:24.53 | caramb1 | ManxPower: At http://www.mpg123.de/ it says: It is highly recommended to not use the source code you can download from this site |
21:24.57 | Nugget | dnetc is -by--hand |
21:24.58 | Nugget | :) |
21:25.01 | Nugget | er, --by-hand |
21:25.04 | SwK[Work] | thats like -funroll-loops |
21:25.16 | Qwell | caramb1: from the asterisk source dir, do make mpg123 |
21:25.17 | SwK[Work] | see www.funroll-loops.org for definitions |
21:25.19 | ManxPower | caramb1: ignore that. |
21:25.22 | Nugget | is -O6 even valid? |
21:25.25 | SwK[Work] | yes |
21:25.28 | caramb1 | Qwell: Aha! Thanks. |
21:25.35 | emrah | I'm having a problem with AreskiCC. The script is just like doing nothing. May I past a part of the error message, or a link to pastebin? |
21:25.48 | tzafrir_laptop | ~pastebin |
21:25.48 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
21:26.03 | ManxPower | caramb1: if you can install mpg123 0.59r from a package do that. If not do a "make mpg123" in the Asterisk source directory before running "make install" |
21:26.09 | tzafrir_laptop | paste your stuff there, paste link here |
21:26.15 | Nugget | what nutball gentoo-addled mouth breather put -O6 in the Makefile? |
21:26.22 | puzzled | lol |
21:26.25 | emrah | Ok |
21:26.29 | ManxPower | What's wrong with -O6? |
21:26.43 | eKo1 | isn't that for optimizing? |
21:26.44 | puzzled | back at 0.7 or so I always changed it back to -O2 or -O3 |
21:27.03 | tzafrir_laptop | Nugget, I think it is taken from a kernel tree build command |
21:27.05 | ManxPower | eKo1: -Ox is optimization level |
21:27.12 | Nugget | -O6 is totally a "this one goes up to eleven" act. |
21:27.27 | Nugget | hell, -Os is proably the best bet for most cases |
21:28.39 | ManxPower | Nugget: What problems might -O6 cause? |
21:28.52 | emrah | there is the link |
21:28.54 | emrah | http://pastebin.ca/18489 |
21:29.06 | puzzled | ManxPower: in gcc4 all hell will break loose and you will not have the afternoon off :) |
21:29.07 | *** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
21:29.16 | Nugget | -O7 is where the gcc documentation says to never tread, so presumably someone saw that note and decided that -O6 must be the best choice. |
21:29.18 | emrah | No logs |
21:29.20 | Nugget | but it often isn't |
21:29.21 | emrah | anything |
21:29.26 | Ayano | Any of the asterlink guys around? |
21:30.04 | bkw_ | yes |
21:30.06 | puzzled | Nugget: what do you think is a good value? -O2, -O3? never heard of -Os. I have gcc 3.4.3 |
21:30.06 | DarthClue | yes |
21:30.08 | bkw_ | Ayano, whas up |
21:30.19 | bkw_ | -06 baby |
21:30.21 | Nugget | lots of code runs faster at -O2 or -O3 than at -O6. it's not like a big "code speed" volume knob |
21:30.22 | bkw_ | er O |
21:30.29 | bkw_ | haha |
21:30.37 | bkw_ | ya it might make smallercode |
21:30.38 | Ayano | I wanna go to cluecon, do you know if there are any rooms still available? |
21:30.38 | bkw_ | but not faster |
21:30.44 | bkw_ | Ayano, yes |
21:30.46 | DarthClue | Ayano, yes |
21:30.51 | MikeJ[Laptop] | echo |
21:30.54 | bkw_ | need to get you registered and paid ASAP |
21:30.59 | MikeJ[Laptop] | echo |
21:31.00 | Ayano | What do I need to do? |
21:31.14 | DarthClue | Ayano: go to cluecon.com and register |
21:31.15 | file[laptop] | go to http://www.cluecon.com/ and register |
21:31.17 | emrah | Please, anyone can help me? |
21:31.21 | MikeJ[Laptop] | hehe |
21:31.32 | file[laptop] | I have a beep, someone beeped me... |
21:31.35 | Nugget | -Os optimizes for size, which can reduce cache pressure and in many cases is a much bigger win than trying to optimize for speed. |
21:31.42 | Beirdo | BEEP! |
21:31.50 | Beirdo | it wasn't me though |
21:32.15 | ManxPower | +zaptel 1.0.9.1 |
21:32.16 | ManxPower | + -- continue fxo operation after the magical 25 days |
21:32.21 | Nugget | heh |
21:32.26 | *** join/#asterisk stkn (nobody@stkn-active-pdpc.developer.gentoo) |
21:32.38 | iCEBrkr | STFU LILO |
21:32.39 | ManxPower | Nugget: Don't we only care about speed? |
21:32.53 | emrah | Is there anyone wh could help me? |
21:33.05 | Nugget | -O is about specific optimizations, not speed. Sometimes they translate to speed, sometimes they harm speed. |
21:33.11 | DarthClue | emrah: be patient, someone will eventually come along who understands areskicc |
21:33.18 | emrah | Thanks |
21:33.44 | puzzled | like areski |
21:33.48 | iCEBrkr | What the heck is areskicc? LOL |
21:33.56 | iCEBrkr | The calling card thing? |
21:33.58 | puzzled | calling card app |
21:34.03 | *** join/#asterisk drooth (~drooth@ip68-111-235-172.sd.sd.cox.net) |
21:34.11 | Hmmhesays | I had all kinds of trouble with areskicc |
21:34.36 | ManxPower | I am SO glad I don't have to bill for calls. |
21:34.49 | iCEBrkr | I wrote my own CC thing before he came out with a dedicated project on it. I was just tinkering. |
21:34.55 | Hmmhesays | post paid isn't so bad |
21:35.05 | asterisk99 | why not bill based on the info in thhe CDRs? |
21:35.12 | iCEBrkr | ...and to be honest, I'm not sure mine worked 100% |
21:35.50 | asterisk99 | you gottum da number called... youo gottum the billsecs... you gottum the source... what more could you want? |
21:36.04 | iCEBrkr | asterisk99: It's not about what to bill on, it's how to do all the IVR stuff. |
21:36.10 | ManxPower | asterisk99: the account information 8-) |
21:36.18 | asterisk99 | thilly me!!!! |
21:36.42 | ManxPower | I just let my LD carrier bill for the calls based on the account code they enter in |
21:36.56 | ManxPower | Granted, this is in an enterprise enviroment |
21:37.16 | ManxPower | I'll be the first IAX2 provider that does something similar will make a fortune. |
21:37.48 | Darwin35 | ? |
21:38.11 | fugitivo | can i use a fax machine with the tdm400? |
21:38.13 | *** join/#asterisk Habakuk (~andy@64.1.15.130.ptr.us.xo.net) |
21:38.57 | Darwin35 | re think that question and think what ports are on the tdm400 |
21:39.31 | *** join/#asterisk IRCMonkey0815 (~asterisk-@dsl-084-056-187-232.arcor-ip.net) |
21:39.40 | fugitivo | my question is "does fax work with the tdm400 or it just receives noise"? |
21:39.47 | emrah | No-one can help me? |
21:39.49 | Delta34 | when using the Queue cmd, is there a way of actually ringing the queue instead of playing the moh for the enduser |
21:40.19 | *** join/#asterisk Robot_ (~Robot_@84.47.4.242) |
21:40.28 | ManxPower | Delta34: "show application queue" |
21:40.37 | *** join/#asterisk Trifixion (~trif@c-24-23-130-6.hsd1.ca.comcast.net) |
21:40.41 | *** join/#asterisk justin_e (~justin_e@c-67-180-16-102.hsd1.ca.comcast.net) |
21:40.51 | ManxPower | Delta34: pay SPECIAL attention to the "r" option of queue |
21:40.52 | Trifixion | Question for the Asterlink crew - how do I connect to Asterlink using SIP instead of IAX? |
21:40.58 | Delta34 | cool thxs |
21:41.20 | DarthClue | Trifixion: msg me your userid and i'll get file right on that. |
21:41.24 | denon | Trifixion: you're probably better off asking asterlink support |
21:41.41 | denon | or him |
21:41.54 | IRCMonkey0815 | hi all, i've problems with my TE411P (Zaptel-1.0.9), when overlapdial=yes * get killed when someone calls on my 3 digit extension, can anyone help me ? |
21:42.53 | emrah | anyone still? |
21:43.01 | ManxPower | IRCMonkey0815: turn off overlapdial |
21:43.47 | IRCMonkey0815 | thats not really an option! because only phone which calls with complete Number will be able to call |
21:44.00 | rashid | still haven't figure out my silence problem |
21:44.03 | ManxPower | IRCMonkey0815: You don't need overlapdial for that. |
21:44.11 | rashid | might have to give up and just put a local asterisk box at that site |
21:44.24 | ManxPower | IRCMonkey0815: Are you using PRI or E&M for your DIDs |
21:44.56 | justin_e | Hey all, I just installed Asterisk CVS-v1-0-07/25/05-11:15:51, when I try to dial from my 7960 to Ast Demo Ext I get no audio, if unload the wct1xxp and try again it works fine, ideas? |
21:44.59 | Trifixion | whats overlapdial? |
21:45.04 | IRCMonkey0815 | hmm, really. But if not incoming calls are being rejected after the first number, |
21:45.23 | ManxPower | IRCMonkey0815: Are you using PRI or E&M for your DIDs |
21:45.51 | Nugget | I suspect the choice of -O6 was from someone who stumbled across the old docs which warn you to never use O7 or higher and the person figured that O6 must be optimal. |
21:45.57 | IRCMonkey0815 | ManxPower: which zapata or zaptel option are you talking about |
21:46.06 | Trifixion | lol O6 doesn't do any more than O3. |
21:46.20 | ManxPower | IRCMonkey0815: I cannot help you any further until I know what signalling the line is set for. |
21:46.49 | tzafrir_laptop | Nugget, check how modules are being compiled in kernel 2.4. IIRC it is with -O6 as well |
21:47.03 | IRCMonkey0815 | ok what i know - German E1, ccs,hdb3,crc4 |
21:47.04 | Nugget | well, that's linux for you. :) |
21:47.30 | ManxPower | IRCMonkey0815: in /etc/asterisk/zapata.conf what is the signaling= line? |
21:47.34 | *** join/#asterisk hound (~AirSaniti@1feaf9de96afd7da.session.tor) |
21:47.53 | *** join/#asterisk DannyF (~dannyf@h194n1fls32o865.telia.com) |
21:48.03 | ManxPower | Not knowing the signaling of a line is like not knowing your default route or netmask |
21:48.05 | IRCMonkey0815 | ManxPower: euroisdn |
21:48.10 | Katty | oh |
21:48.13 | Katty | Jul 25 16:46:27 WARNING[879]: chan_iax2.c:7357 socket_read: Received mini frame before first full voice frame |
21:48.16 | Katty | what's that mean? |
21:48.21 | ManxPower | IRCMonkey0815: I've NEVER gotten overlapdial working with a USA PRI. |
21:48.24 | *** join/#asterisk bonez41 (~aint@c-67-166-77-14.hsd1.ut.comcast.net) |
21:48.27 | Nugget | so, I'm relieved to learn that the ridiculous -O6 in the asterisk makefile hasn't been producing shit code for us all, but I'm dismayed to learn that it's there to begin with. :) |
21:48.30 | ManxPower | Katty: harmless |
21:48.35 | Katty | ManxPower: uhm |
21:48.39 | Katty | ManxPower: let me rephrase that |
21:48.42 | Katty | ManxPower: WHAT DOES IT MEAN |
21:48.46 | Katty | ManxPower: m'kay? (+ |
21:48.46 | Katty | (= |
21:48.57 | ManxPower | IRCMonkey0815: make SURE immediate=no and overlapdial=np |
21:49.05 | Qwell | it means a mini frame was received before the first full voice frame was |
21:49.10 | Katty | i see. |
21:49.10 | Qwell | but, what do I know? |
21:49.17 | IRCMonkey0815 | ManxPower: is configured definetly |
21:49.31 | ManxPower | Katty: it means that Asterisk got a miniframe (which can NEVER be the first frame) before it got the first full frame for a call. |
21:49.43 | Katty | ManxPower: and what causes this? |
21:49.53 | ManxPower | Katty: random routing oddities |
21:49.58 | Katty | k |
21:50.07 | ManxPower | Katty: out of order packets are not all that unusual. |
21:50.39 | Katty | ManxPower: thank you (= |
21:51.55 | tzafrir_laptop | so why warn about this? |
21:52.11 | IRCMonkey0815 | ManxPower: with that config: calls to 'XXXX0' ok! calls to 'XXXX11' will be cut to 'XXXX1' |
21:52.45 | IRCMonkey0815 | ManxPower: but only in slow dial |
21:53.26 | *** join/#asterisk Robot_ (~Robot_@84.47.4.242) |
21:54.19 | emrah | Please, I'm desesperatting with this strange problem. |
21:54.34 | Trifixion | emrah - whats the problem? |
21:54.51 | Trifixion | if you want asterisk to run faster do the -mcpu=686 stuff. |
21:54.56 | Trifixion | not -O99999 |
21:55.08 | *** join/#asterisk d00gster (~doughant@toronto-HSE-ppp4334604.sympatico.ca) |
21:55.12 | Qwell | Trifixion: can-tehl-pself-itis? |
21:55.25 | *** part/#asterisk d00gster (~doughant@toronto-HSE-ppp4334604.sympatico.ca) |
21:55.28 | Qwell | thel rather, heh |
21:55.30 | doolph | anyone can help me with gnugk? |
21:55.38 | doolph | or where to get that kind of help?= |
21:56.17 | Trifixion | doolph - i'm not ashamed to admit that i'm very intimate with gnugk. |
21:56.25 | puowvip | I'm starting a company and you are all hired. |
21:56.28 | Trifixion | in fact, i use gnugk source code as erotic fiction every now and again. |
21:57.27 | doolph | Trifixion I didnt understand you lol |
21:57.38 | Trifixion | i'm saying that gnugk gets my rocks off! |
21:57.40 | Trifixion | got it? |
21:57.41 | puowvip | #asterisk <Trifixion> in fact, i use gnugk source code as erotic |
21:57.41 | puowvip | +fiction every now and again.#asterisk <Trifixion> in fact, i use gnugk source code as erotic |
21:57.58 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
21:58.16 | doolph | it means that you are good or bad |
21:58.24 | Trifixion | i'm badgood/goodbad. |
21:58.32 | IRCMonkey0815 | ManxPower: any hints |
21:59.13 | puowvip | Trifixion: you're a zebra-hat? |
21:59.29 | Trifixion | yup |
21:59.36 | ManxPower | IRCMonkey0815: those two things are all I had to set to make my PRI work, other than the obvious things |
22:00.03 | doolph | Trifixion do you have time to me? |
22:00.12 | Trifixion | yup |
22:00.35 | opus_ | hey |
22:00.35 | IRCMonkey0815 | ManxPower: hmm! what is your option at pridialplan "unknown" ? |
22:00.41 | opus_ | hmmtl |
22:01.15 | *** join/#asterisk Oryn (oryn@falcore.fsck.tv) |
22:01.58 | Oryn | anyone know if its possable to use ogg vorbis for voicemail attachments? |
22:02.06 | Trifixion | http://www.crapville.com/media_videos12/treat_her_right.wmv |
22:02.08 | Trifixion | (fyi) |
22:03.17 | Nugget | is there a techical reason to prefer vorbis, or is it just a political desire because vorbis makes your shit smell like roses? |
22:03.25 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
22:03.50 | Nugget | "RMS refuses to leave voicemail until we stop using GSM" :) |
22:04.08 | *** join/#asterisk hypa7ia (~leigh@33051397cc8359ec.session.tor) |
22:04.24 | *** join/#asterisk Coriantum (~asdfkle@67-41-182-243.slkc.qwest.net) |
22:04.42 | Oryn | yeah its easyer to play than wav files, for some reason the player that my email client uses wont play wav files |
22:04.55 | Oryn | plus I'm hoping they will be smaller |
22:05.24 | Oryn | Nugget: mp3 would be ok I guess |
22:05.25 | puowvip | eh |
22:05.33 | Coriantum | I just started using AEL, could someone recommend a website I could look at for some documentation? Other then the Readme. |
22:06.15 | jsmith | Coriantum: If you find something good, jump in to #asterisk-doc and let us know, please... |
22:06.25 | *** join/#asterisk Milligan (~stephen@208.189.212.200) |
22:07.06 | Coriantum | If I cant find something then I will start writing my own |
22:07.22 | Darwin35 | AEL is new not well documented yet |
22:07.27 | Darwin35 | its only in head |
22:07.33 | Coriantum | I noticed |
22:07.34 | Darwin35 | if you use it document it |
22:07.43 | Darwin35 | then put it in a wiki |
22:07.51 | jsmith | And in #asterisk-docs, please... |
22:08.18 | Trifixion | did you guys check out that URL i posted? |
22:08.20 | Trifixion | it's hilarrrrrious. |
22:08.22 | Darwin35 | I take the .ael must be asterisk-extensions-logic language |
22:08.34 | Coriantum | Will do |
22:08.48 | *** join/#asterisk pdracevich (~bob@210.54.249.228) |
22:09.16 | pdracevich | is the Asterisk CVS server down? |
22:09.57 | Darwin35 | it should not be |
22:10.29 | Oryn | so is ogg a dirty word in here? |
22:10.29 | pdracevich | tryed downloadin from it. It timed out at the PW stage tried to ping it no luck as well |
22:10.52 | file | working for me |
22:10.57 | *** join/#asterisk gast (~riemensc@83-169-155-92-dynip.superkabel.de) |
22:11.04 | gast | guten abend |
22:11.24 | pdracevich | *sigh* must be my ISP |
22:12.14 | gast | ich habe mal ne frage bezüglich asterisk mit zaphfc |
22:12.38 | gast | bei mir klappt zwar die eingehende telfonie aber nicht die ausgehende |
22:13.01 | Nugget | Wir sind alle Kuhe. |
22:13.09 | eKo1 | lol |
22:13.23 | gast | ich erhalte immer die fehlermeldung extensions in context ´default´ from ´meine gesetzte msn´ does not exists rejeting call on channel 0/2, span 1 |
22:13.45 | pdracevich | exit |
22:13.52 | gast | kennt jemand von euch die fehlermeldung und wie kann man ihn beheben? |
22:14.04 | eKo1 | bitte frag auf englisch weil die meisten hier verstehen dich nicht. |
22:14.17 | gast | spreche leider nicht so gut english |
22:14.39 | gast | könnte es aber mal schnell durch babelfish senden oder meinst du das bringt nichts |
22:14.57 | eKo1 | versuch es mal |
22:15.08 | florz | gast: Wenn Du hier fuer Belustigung sorgen willst ... =:-) |
22:15.17 | Nugget | deine english ist besser dan meine deutcsh. |
22:15.36 | Nugget | und ich have zwei jahre deutschklasse! |
22:15.36 | gast | sorry, jeder kann sich mal vertippen |
22:16.16 | gast | da ich mir ja schon den spot auf mich gezogen habe, kann mir jemand schildern wie man das problem löst? |
22:16.37 | gast | asterisk mit hfc karte -> soll sich am sip anmeldungen |
22:16.38 | florz | gast: Was Dein Problem angeht: Naja, Du hast halt im Dialplan nicht spezifiziert, was passieren soll, wenn die betreffende MSN angewaehlt wird. |
22:16.50 | gast | das telefon klingelt |
22:17.04 | gast | kann ja nur nicht abgehend telefonieren, das macht mich ja so verrückt |
22:17.21 | *** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-7-21.w81-248.abo.wanadoo.fr) |
22:18.01 | florz | gast: Aehm, oder so - halt die Nummer, die Du waehlst, ist im Dialplan nicht spezifiziert. |
22:18.09 | tzafrir_laptop | gast, isn't there any #asterisk-de or something? |
22:19.03 | gast | [general] |
22:19.03 | gast | static = yes |
22:19.03 | gast | writeprotect=no |
22:19.03 | gast | [globals] |
22:19.03 | gast | IAXINFO=guest ; IAXtel username/password |
22:19.03 | gast | [default] |
22:19.05 | gast | include=>calls |
22:19.07 | gast | [calls] |
22:19.09 | gast | exten => 1015396,1,Dial(Zap/2/30690116,60tT) |
22:19.11 | gast | exten => 1015396,2,Hangup |
22:19.13 | gast | ;exten => s,1,NoOP |
22:19.15 | gast | exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate,60,tT) |
22:19.17 | gast | exten => _9.,2,Congestion |
22:19.19 | gast | exten => _9.,3,Busy |
22:19.21 | tzafrir_laptop | ~pastebin |
22:19.21 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
22:19.21 | gast | exten => _9.,4,Hangup |
22:19.23 | gast | die 30690116 sollte meine msn sein |
22:19.27 | puzzled | gast: please use pastebin.com |
22:23.01 | Darwin35 | ~jbot sex |
22:23.01 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
22:24.42 | eKo1 | hahaha |
22:25.44 | Nugget | that command line is fundamentally flawed. |
22:25.49 | Nugget | they should be &&'s, not ;'s |
22:26.09 | Nugget | you sure as hell wouldn't want to proceed to strip if the cd failed! |
22:26.34 | *** join/#asterisk Ahewes (~rsb@209.133.58.210.aaph.com) |
22:26.48 | Ayano | Alright, I'm signed up to get a clue at cluecon! |
22:27.59 | *** join/#asterisk pfn (~pfnguyen@netblock-66-245-252-239.dslextreme.com) |
22:28.22 | niZon | does nufone allow you to set the outgoing callerid? |
22:28.39 | DarthClue | Asterlink does. |
22:28.54 | Nugget | asterlink rocks my socks. |
22:30.02 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
22:30.18 | niZon | they seem a little coming soon.. |
22:30.29 | DarthClue | niZon: who? asterlink? |
22:30.52 | niZon | yeah |
22:30.53 | niZon | lol |
22:31.00 | Ayano | These guys at asterlink worry me. They are too on top of thier game! They are too good! :) |
22:31.01 | Ayano | lol |
22:31.22 | niZon | lol |
22:31.31 | niZon | can they do paypal? |
22:31.33 | DarthClue | niZon: we aren't coming soon. we are here. unfortunately, i think the web guys went on vacation and forgot to finish the site. |
22:31.40 | niZon | lol |
22:31.43 | DarthClue | niZon: if that's what you use, i'm sure we can. |
22:32.05 | niZon | US50/Canada toll free? prepaid? |
22:32.07 | DarthClue | niZon: might cost a little more though since we would have to pay a fee to paypal. |
22:32.28 | DarthClue | US48 right now, we are working towards Canada, etc. yes, 2c prepaid. |
22:32.46 | niZon | aw i want canada |
22:34.09 | *** join/#asterisk focks (~cbruender@nsc66.147.95-93.newsouth.net) |
22:34.34 | file | Canada is expensive |
22:34.52 | Ayano | A lot of taxes right? |
22:36.21 | justin_e | Hey all, I'm running CVS-v1-0-07/25/05-11:15:51, SIP->External Sip Provider works, but SIP->AST (Playback) doesn't send audio (tethereal shows no outbound audio). If unload the wct1xxp and try again it works fine. Ideas? |
22:36.21 | niZon | hm |
22:36.21 | file[laptop] | no, it's just expensive... |
22:38.00 | niZon | link2voip claims they can do it for 3 cents per min |
22:38.27 | niZon | problem is, they have almost nonexistant support and they don't seem to actually have DIDs |
22:38.49 | file[laptop] | all I can tell you is that for people who have their own PRIs and circuits, it's expensive |
22:39.15 | syle2 | thats why you hookup to tandum switches file |
22:39.26 | syle2 | PRI is wasteful money hehe |
22:39.37 | file[laptop] | PRI was just in my head |
22:39.48 | file[laptop] | doesn't mean I have a PRI or something :P |
22:41.43 | *** join/#asterisk danalien (~danalien@danalien.user) |
22:49.12 | focks | on an incoming context, why won't s,1,... match any incoming trunk? I see an error about no matching extension in [incoming] but i didn't think s cared about the 4 digits coming in off the PRI |
22:50.01 | *** join/#asterisk meppl (~mephisto@87.193.6.7) |
22:50.27 | file[laptop] | s is when asterisk doesn't know what number was dialed |
22:51.01 | focks | file[laptop], so if it does know the extension, i have to explicitly use it? |
22:51.22 | *** join/#asterisk daniel101 (~daniel101@dsl15-088.express.oricom.ca) |
22:51.43 | focks | ie, instead of s i'd use like _XXXX to match any digits for the [incoming] context |
22:52.10 | file[laptop] | exactly! |
22:52.41 | focks | file[laptop], hmm, wonder why this example only listed s and nothing about actual extensions |
22:53.29 | focks | oh well |
22:54.05 | file[laptop] | bkw! |
22:54.11 | *** part/#asterisk justin_e (~justin_e@c-67-180-16-102.hsd1.ca.comcast.net) |
22:56.58 | syle2 | i;m having a problem where can barely hear the caller on the pap2-na |
22:57.38 | *** join/#asterisk kslater (~kslater@24.svnf1.xdsl.nauticom.net) |
22:57.40 | syle2 | ulaw from termination provider to me-> tried ulaw to pap2-na device and g729 without any result changes |
22:57.41 | eKo1 | turn the volume up |
22:57.45 | syle2 | any ideas what else i could try |
22:57.54 | syle2 | turn volume up where |
22:58.05 | eKo1 | on your phone |
22:58.06 | *** join/#asterisk iswm (iswm@iswm.user) |
22:58.24 | syle2 | there is no volume on the phone |
22:59.00 | eKo1 | sucks to be you then |
22:59.22 | syle2 | sucks to be every use that buys a 20 dollar phone at walmart you mean |
22:59.27 | syle2 | thats pretty much everyone |
23:00.20 | syle2 | no personally i have the latest and greatest, but thats not the point, its the enduser i;m talking about :) |
23:01.20 | eKo1 | Ditto. |
23:02.39 | *** join/#asterisk btm (~b1ueemu@70-33-140-162.agstme.adelphia.net) |
23:03.00 | *** join/#asterisk justin_e (~justin_e@c-67-180-16-102.hsd1.ca.comcast.net) |
23:03.55 | *** join/#asterisk colinm_ (~colol@VDSL-130-13-9-155.PHNX.QWEST.NET) |
23:03.59 | *** part/#asterisk justin_e (~justin_e@c-67-180-16-102.hsd1.ca.comcast.net) |
23:05.08 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
23:09.38 | timecop | hm nice |
23:09.47 | timecop | ever since i changed fwd to IP address i dont see it failing to register anymore |
23:09.53 | timecop | DNS IN ASTERISK SUCKS. |
23:10.56 | opus_ | Jul 25 16:07:26 WARNING[2252]: rtp.c:950 ast_rtp_settos: Unable to set TOS to 184 |
23:11.47 | puzzled | try tos=0x18 |
23:11.57 | *** join/#asterisk Romik_ (~romik_@1.fix.netvision.net.il) |
23:12.12 | *** join/#asterisk stock (~stock@stokkie.demon.nl) |
23:12.23 | Romik_ | somebody can advice what DEAD PACKAGE means? iax2 show stats |
23:12.25 | Romik_ | <PROTECTED> |
23:12.27 | Romik_ | --------------------- |
23:12.29 | Romik_ | Outstanding frames: 6 (0 ingress, 6 outgress) |
23:12.31 | Romik_ | Packets in transmit queue: 6 dead, 1 final, 6 total |
23:12.41 | Romik_ | packets i rather. |
23:15.12 | opus_ | is there a problem with SER where the enduser hangs up and SER never sends the hangup back to asterisk? |
23:17.14 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
23:18.23 | *** join/#asterisk craziman2 (~donnie@boromir.apid.com) |
23:18.58 | Ahewes | anyone else getting a lot of errors trying to compile chan_zap.c in cvs head? |
23:20.26 | Ahewes | ooops, my bad. libri.h no such file or directory. bonehead mistake. |
23:21.52 | *** part/#asterisk Ash (~aaron@outofband.org) |
23:22.52 | kslater | trying to find an answer on E911 for a server for use at home |
23:23.40 | niZon | get a cheap basic landline and an FXO interface |
23:23.50 | Nugget | does "trying to find an answer" ever actually involve, you know, asking a question? |
23:23.54 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
23:24.04 | niZon | or if you're cheap, get a big red phone and plug it in to the landline |
23:24.06 | Nugget | that's what most of us do. |
23:25.02 | newl | or just don't provide it..you're not a carrier. |
23:26.13 | kslater | ok. so since it's a home system and I have a TDM with both FXO and FXS modules, I should be fine, eh? |
23:26.49 | kslater | Nugget: I suppose you're correct, I should have asked for a pointer at least. |
23:26.59 | gordonjcp | kslater: what's the problem? |
23:27.15 | newl | Sure. Though if you're concerned with E911 service, you should be speaking with your VoIP provider instead. :) |
23:27.27 | *** part/#asterisk caramb1 (~alfons@c-f8ef70d5.09-237-73746f34.cust.bredbandsbolaget.se) |
23:28.06 | kslater | no problem yet. I intend to be my own voip provider |
23:28.24 | kslater | so my concern is that people in my house still have access to 911 service |
23:28.46 | gordonjcp | kslater: ? |
23:29.18 | gordonjcp | kslater: do you have a landline coming in at all? |
23:29.34 | kslater | yes |
23:29.51 | gordonjcp | great, plug an el-cheapo POTS phone in |
23:29.58 | gordonjcp | job done |
23:30.14 | gordonjcp | or use your mobile |
23:30.23 | gordonjcp | no sense hunting around for a landline in an emergency |
23:30.26 | gambolputty | switch => Realtime isn't working for me. Failed for anyone else? |
23:30.34 | kslater | so they'd have to use the POTS phone for 911, couldn't use any phone connected to * for 911? |
23:30.46 | colinm_ | kslater: They could use a phone connected to *, so long as there's power. |
23:30.49 | gordonjcp | kslater: yes, if it dialled out over the normal line |
23:31.12 | kslater | ok. so if I lose power, then they'd need to the landline |
23:31.13 | Nugget | if you buy a sipura to hook up the pots line to asterisk, then you can just route 911 out over the pots line |
23:31.21 | Nugget | buy a UPS. they're like $99 |
23:31.30 | *** join/#asterisk tessier (~treed@146.82.146.22) |
23:31.39 | kslater | Nugget: have a honkin' ups already on the server with * |
23:31.46 | gordonjcp | or, as I say, just don't worry about it |
23:31.50 | kslater | also have a TDM400 card |
23:31.51 | tessier | Anyone know the trick to configuring a Handytone-486? I plug in my analog phone, press the button, but don't hear an IVR. |
23:31.53 | gordonjcp | use your mobile |
23:32.00 | Nugget | so what's the problem? |
23:32.06 | Nugget | sounds like you know all the answers already |
23:32.27 | kslater | just trying to be sure I understand. I don't know much as it turns out. that's why I'm asking here. |
23:33.40 | kslater | I suppose having all cordless phones it's no different in a power failure situation. |
23:34.53 | opus_ | ever tried calling 911 from a cellphone? 9 out of 10 times it won't wor |
23:34.54 | opus_ | wor |
23:34.57 | opus_ | work |
23:35.01 | tessier | opus_: Why not? |
23:35.05 | opus_ | dunno |
23:35.07 | gordonjcp | opus_: wtf? |
23:35.13 | Nugget | i've never had a problem with 911 on a mobile. |
23:35.28 | opus_ | how many times have you called 911? |
23:35.32 | Nugget | three or four times |
23:35.38 | gordonjcp | no problems either time |
23:36.04 | gordonjcp | I wouldn't even consider using a landline |
23:36.12 | Nugget | once to report a car crash and three times to report drunk drivers. no problems at all |
23:36.42 | syle2 | power failure at home? |
23:36.52 | gordonjcp | I don't think I've even had a landline for about 10 years |
23:36.56 | gordonjcp | maybe 8 |
23:36.56 | kslater | gordonjcp: wouldn't consider a landline? |
23:36.58 | syle2 | just install a solar power panel |
23:36.59 | kslater | ah |
23:36.59 | gordonjcp | nope |
23:37.05 | gordonjcp | kslater: not these days |
23:37.10 | kslater | got it |
23:37.11 | syle2 | run all your electricity in your house off that :) |
23:37.12 | Delta34 | anybody know much about the tranfer cmd |
23:37.12 | gordonjcp | kslater: maybe if I was getting ADSL |
23:37.14 | kslater | single? |
23:37.36 | gordonjcp | single what? |
23:37.38 | Delta34 | if someone is calling thru zap channel to sip line |
23:37.52 | kslater | gordonjcp: are you single? |
23:38.00 | Delta34 | can i transfer the call back out the zap channel to external number |
23:38.05 | gordonjcp | kslater: no |
23:38.15 | gordonjcp | kslater: why? and what kind of question is that anyway? |
23:38.36 | kslater | gordonjcp: too many ppl in this house for cellular to be affordable |
23:38.44 | kslater | that's why I asked. |
23:39.04 | gordonjcp | kslater: it's practically free, my bill is about £15 a month |
23:39.20 | gordonjcp | I don't think i could even get a line for that, never mind call anyone |
23:39.36 | syle2 | kslater just clone your smartcard |
23:39.38 | gordonjcp | I'm not even sure it's as much as that, tbh |
23:39.44 | kslater | got it. different situation here. sorry if I offended you |
23:40.14 | opus_ | yeah, i need a sim card cloner |
23:40.20 | opus_ | can i borrow yours syle2? |
23:40.21 | gordonjcp | kslater: put it this way, the direct debit for my phone bill isn't enough to show up on a monthly bank statement |
23:40.25 | opus_ | does it work off the air? |
23:40.43 | gordonjcp | and I use my phone a lot |
23:41.53 | kslater | ok. you guys answered my questions. time to push forward with the asterisk project. I have 4 kids that could use it.. |
23:43.59 | Sedorox | too bad there isn't a iden kit to connect to asterisk :p |
23:45.37 | *** join/#asterisk wolfson (~hehe@68-187-187-034.dhcp.mant.nc.charter.com) |
23:45.57 | puzzled | asterisk on slashdot: http://joybubbles.telephreak.org/papers/vpa/ |
23:48.12 | PyroSteve | hey guys |
23:48.49 | PyroSteve | i just implentmented concurrent call limits in my asterisk install by using the SetGroup stuff |
23:48.57 | PyroSteve | seems like its work great |
23:49.22 | PyroSteve | but here is an a question that I guess could be answered by expermintation ... |
23:49.43 | PyroSteve | when a call is added to a group .. then the call is finished ... |
23:49.58 | PyroSteve | does the call simply get removed from the group |
23:50.27 | Sedorox | 0_o |
23:50.50 | PyroSteve | is my keyboard working ? |
23:50.56 | Sedorox | no.. it isn't |
23:50.57 | Sedorox | :p |
23:50.59 | Katty | there are 4 T in 1/4 C, right? |
23:51.03 | Sedorox | I duno the answer.. sorry |
23:51.14 | Sedorox | -_- |
23:51.19 | Katty | WHAT |
23:51.29 | Katty | have you insaned? |
23:51.41 | Sedorox | I've been insane |
23:51.46 | Katty | oh |
23:51.50 | Katty | well that explains everything |
23:51.51 | Sedorox | lol |
23:51.53 | Katty | anyone else know? |
23:51.54 | Sedorox | yup |
23:52.24 | gordonjcp | four what in a quarter what? |
23:52.39 | Katty | nevermind ;) |
23:52.42 | Sedorox | lol |
23:52.59 | *** join/#asterisk hypa7ia ([U2FsdGVkX@a7ac4e237de46e4f.session.tor) |
23:53.58 | twisted | wheeee. |
23:54.08 | Katty | BEEP BEEP |
23:54.16 | twisted | beep beep :) |
23:54.19 | Katty | twisted: don't call, i'm working out |
23:54.28 | twisted | you're working out on irc? |
23:54.28 | Sedorox | backuping up? |
23:54.29 | Katty | i shall not interrupt teh reverse wrist calls. |
23:54.39 | Katty | well im sorta between sets |
23:54.45 | twisted | ahh |
23:54.56 | *** join/#asterisk litage_ (~nick@ws01.5749.dsl.winshop.com.au) |
23:55.02 | twisted | i consider IRC a workout sometimes |
23:55.21 | Qwell | twisted: Want a real workout? |
23:55.26 | Sedorox | Hmmm |
23:55.29 | Katty | i hear sex is a good workout |
23:55.32 | Sedorox | 0_o |
23:55.37 | twisted | Qwell, uhm, if I wanted that, I'd ask Katty if i could join her set |
23:55.37 | Katty | someday i'll find out (= |
23:55.39 | Sedorox | say... Katty wanna work out with me |
23:55.40 | Sedorox | ? |
23:55.41 | Sedorox | :p |
23:55.41 | Qwell | put documentation in one room, a shitload of cables in front of you, and make sure you have a computer that you can't get to the back of |
23:55.45 | Katty | Sedorox: uhh, no |
23:55.57 | twisted | but my workout today has been staying awake |
23:56.15 | Qwell | (in my case, a shitload is 1) |
23:56.26 | twisted | Qwell, like the size of a volkswagen/ |
23:56.43 | Qwell | twisted: 1/8" stereo cable :p |
23:56.54 | twisted | Qwell, ah. close enough. |
23:57.10 | Qwell | I had a space about 11" wide, 3' deep, trying to plug shit in blind... |
23:57.19 | twisted | Qwell, welcome to racks |
23:57.20 | Sedorox | Katty: darn :( |
23:57.23 | Qwell | twisted: indeed |
23:57.43 | Katty | Sedorox: sorry, i'm not a casual sex type person |
23:57.47 | Sedorox | thats the best.. plugging stuff in blind |
23:57.51 | Katty | Sedorox: in fact, i'm still one of the innocents (= |
23:57.53 | twisted | don't harass Katty, she's a sweetheart |
23:57.55 | Katty | and prefer to keep it that way |
23:57.59 | Sedorox | Katty: hey.. so I am... |
23:58.00 | Sedorox | :/ |
23:58.02 | Katty | k |
23:58.05 | Delta34 | anybody know if a 2.4 or 2.6 kernel better for asterisk? or really doesn't matter |
23:58.13 | Sedorox | I use 2.6 |
23:58.25 | opus_ | 2.6 |
23:58.45 | syle2 | what the HELL is wrong with casual sex!!! |
23:58.47 | opus_ | 2.4 requires a timing device, |
23:58.58 | Sedorox | no one said there was |
23:58.59 | opus_ | well, you can use the dummy one. |
23:59.00 | niZon | anyone used asterisk_php? |
23:59.01 | Katty | syle2: i didn't say anything was. i just said i didn't have it |
23:59.02 | *** part/#asterisk hound (~AirSaniti@1feaf9de96afd7da.session.tor) |
23:59.03 | file | casual sex where? |
23:59.04 | Sedorox | we're just stating that we're not into it |
23:59.04 | Sedorox | :p |
23:59.07 | Sedorox | lol |
23:59.12 | twisted | file, your hand |
23:59.15 | Katty | file: over there, in the corner |
23:59.15 | Sedorox | file: see #causual_sex |
23:59.18 | Delta34 | doesnt digium card have a timer device? |
23:59.23 | Sedorox | sp |
23:59.28 | Sedorox | Delta34: yes |
23:59.31 | twisted | Katty, ew.. I don't wanna clean that up... |
23:59.41 | gambolputty | I compiled the newest CVS *, and suddenly my phones won't register. http://pastebin.com/320657 could someone take a look? |