| 00:00.00 | Trionnis | welcome ;) |
| 00:02.31 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:03.51 | *** join/#asterisk gek (n=maxx@gekkonaut.com) |
| 00:04.12 | gek | sixtel can suck it. for real. i've been trying to add international calling for 3 days now and they're completely ignoring me |
| 00:04.17 | *** join/#asterisk RoyK (n=roy@cm-80.111.22.187.chello.no) |
| 00:04.35 | opus_ | hello |
| 00:04.38 | gek | can anyone recommend a company for termination that actually does something? |
| 00:04.39 | *** join/#asterisk dano (n=dano@212.115.50.7) |
| 00:04.43 | opus_ | haha |
| 00:04.55 | Trionnis | I use sipmedia |
| 00:05.02 | Trionnis | no complaints really |
| 00:05.17 | Trionnis | few hiccups every now and then, but nothing serious |
| 00:05.18 | Sedorox | douns like you went with link2voip |
| 00:05.18 | Sedorox | :p |
| 00:05.26 | gek | they do iax obviously, right? |
| 00:05.26 | Sedorox | sounds* |
| 00:05.36 | Sedorox | teliax I've heard good things about... |
| 00:05.37 | Trionnis | actually no, they don'yt |
| 00:05.40 | Trionnis | don't* |
| 00:05.51 | Trionnis | if you're wanting outbound only, check voipjet |
| 00:06.12 | Trionnis | <- digium fanboy, obviously |
| 00:06.15 | Trionnis | ;) |
| 00:06.19 | Sedorox | lol |
| 00:06.41 | opus_ | some of the asterisk bounties are impossible |
| 00:06.54 | Sedorox | Hmm |
| 00:07.04 | Trionnis | no, no, no |
| 00:07.15 | Trionnis | *anything* is possible.... at zombo.com! |
| 00:07.19 | Trionnis | *ducks* |
| 00:08.04 | gek | i am looking at voxee.com and gafachi.com |
| 00:08.16 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@m010e36d0.tmodns.net) |
| 00:08.17 | opus_ | anything is possible, but if you don't describe it nobody will know what the fuck you are talking about |
| 00:08.41 | Trionnis | but.... you're a programmer... you should already know these things! |
| 00:08.43 | Trionnis | sheesh! |
| 00:08.50 | malverian[work] | Yay :) |
| 00:08.54 | malverian[work] | Snom autoanswer works perfectly. |
| 00:09.17 | blitzmere | I am a real newbie, but does anyone know how I can fix this --->> NOTICE[2376]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us' <---- |
| 00:09.31 | Trionnis | google it |
| 00:09.42 | Trionnis | I saw that a while back, don't remember what the fix was though |
| 00:09.45 | opus_ | language=us is missing in your conf |
| 00:09.50 | opus_ | is my guess |
| 00:09.53 | blitzmere | ok thanks |
| 00:10.24 | Trionnis | so..... festival, anyone? |
| 00:10.33 | Trionnis | new voices? |
| 00:10.37 | Trionnis | please? :) |
| 00:11.10 | opus_ | we're hiring a mexican in a box to do our tts |
| 00:11.15 | opus_ | but its currently own spanish |
| 00:11.21 | Trionnis | ... |
| 00:12.01 | opus_ | they're working on a patch |
| 00:12.36 | tzanger | opus_: HAHAHAHAHAHAHAHA |
| 00:14.14 | Trionnis | *crickets chirping* |
| 00:14.22 | bkw__ | so when is someone going to fix this issue with chan_iax2? |
| 00:16.02 | Cybertoy | what issue? |
| 00:17.59 | tzanger | 'Yes, firefox is indeed greater than women. Can women block pops up for you? No. Can Firefox show you naked women? Yes.' |
| 00:23.44 | gek | ok. so. if i do a Dial() from an incoming IAX call, to forward a call, with teh same provider.. does the traffic stop hitting my asterisk box? |
| 00:24.23 | Ariel_ | Hmmhesays, we are using mysql database |
| 00:27.19 | tuxinator_linuxM | tzanger: I like the named women my wife shows me. |
| 00:27.26 | tuxinator_linuxM | naked women |
| 00:27.40 | tzanger | tuxinator_linuxM: :-) yes when I was married that was always the best |
| 00:28.26 | Ariel_ | wow I just won the super lotto.... spam emails suck |
| 00:28.27 | tuxinator_linuxM | Enjoy it while I can, huh ;-) |
| 00:28.54 | Trionnis | hell, my wife looks at more nekkid women than I do |
| 00:28.56 | tuxinator_linuxM | Ariel_: You need Mozilla Thunderbird |
| 00:29.17 | Ariel_ | tuxinator_linuxM, no I nee gmail to remove the spam better. |
| 00:29.39 | tuxinator_linuxM | Trionnis: I suppose you will tell us more about this |
| 00:29.45 | Trionnis | of course, I'm about 90% certain she's a lesbian that just married me for money |
| 00:29.52 | Trionnis | but hey, whatever :) |
| 00:29.58 | *** join/#asterisk PBXtech (i=nik@14.sub-70-218-13.myvzw.com) |
| 00:30.02 | tuxinator_linuxM | Trionnis: How lucky can you get |
| 00:30.19 | Trionnis | I don't "get lucky", that's why I suspect she's a lesbian |
| 00:30.28 | Trionnis | *rimshot* |
| 00:30.33 | Ariel_ | opus_, your having a mexican do your spanish sound files or translation in text. |
| 00:30.34 | PBXtech | does spandsp have issues with 1.2b1? |
| 00:31.15 | tuxinator_linuxM | Ariel_: someone put my on a porn mailing list many many years ago. Thunderbird only lets about 1 out of a couple thousand through. |
| 00:31.27 | Ariel_ | PBXtech, spandsp has lots of issues. You might have to fix the patch file for it. But I have not installed it yet on 1.2b1 |
| 00:31.50 | [hC] | is there a way to make it so that when i run out of g729 licenses, it fails over to using another codec instead of just passing into dead air?? |
| 00:32.00 | tzanger | Ariel_: ? spandsp's been pretty damn stable for me in the last rev |
| 00:32.07 | PBXtech | the patch worked just fine. rxfax is ok, just not receiving to a .tif file |
| 00:32.11 | tzanger | there was a /0 bug that coppice corrected but aside from that it's been damn good |
| 00:32.32 | puowvip | Efforts to stop the levee break at the 17th Street Canal have ended unsuccessfully and the water is expected to soon overwhelm the pumps in that area, allowing water to pour into the east bank of Metairie and Orleans to an expected height of 12-15 feet. |
| 00:32.39 | tzanger | [hC]: not at this time, you'd have to try some refcounting and try dialing thorugh another type=peer entry that doesn't have g729 enabled |
| 00:32.51 | tzanger | puowvip: that's insane |
| 00:32.57 | Ariel_ | spandsp pre19 has been working fine for me but I use it on stable |
| 00:33.03 | tzanger | has anyone seen manxpower here since katrina hit? |
| 00:33.12 | Trionnis | yeah, new orleans is pretty well farked |
| 00:33.14 | Trionnis | :( |
| 00:33.18 | tzanger | I imagine he's safe and sound but still |
| 00:33.46 | [hC] | tzanger: yeah, the problem is i dont just use g729 license in one place.. i guess i could write an AGI or something that utilized show modules |
| 00:33.46 | [hC] | Secondly, is there a way to have asterisk boost up the volume on the wav files that voicemail emails out? people complain they're quiet. |
| 00:33.50 | tzanger | <jewish mother>You'd think that he'd have the decency to call or somehow get word out that he's alright, lord knows we don't worry enough about him already</jewish mother> |
| 00:34.09 | tzanger | [hC]: exactly, it's a problem. I believe it's on the list though |
| 00:34.26 | opus_ | hC - tell them they don't listen |
| 00:34.27 | tuxinator_linuxM | tzanger: not yet |
| 00:34.47 | tuxinator_linuxM | ~seen manxpower |
| 00:34.50 | jbot | manxpower <n=eric@dsl-209-16-67-160.datasync.com> was last seen on IRC in channel #asterisk, 2d 5h 7m 16s ago, saying: 'ride is here!'. |
| 00:35.08 | Trionnis | well, I guess he got out |
| 00:35.11 | Trionnis | hopefully |
| 00:35.11 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 00:35.12 | Trionnis | :-/ |
| 00:35.47 | tzanger | there's a neat article on /. about the predetermined worst-case challenges |
| 00:35.48 | opus_ | connect the dots. lalalalalot |
| 00:35.51 | tzanger | in new orleans for this kin dof thing |
| 00:36.02 | [hC] | Is this gotoif syntax right? |
| 00:36.02 | [hC] | exten => s,1,GotoIf($"[${ARG5}" = ""]?macro-doDial,s,4) |
| 00:36.08 | opus_ | nope |
| 00:36.21 | [hC] | ive had people tell me to take the "" out of around [${ARG5}] but that seems to always return 1 |
| 00:36.26 | [hC] | even if its false |
| 00:36.28 | blitzmere | ok i cannot seem to figure this out. I have tested this again and again. its playing music like is on hold. perminateltly...any ideas |
| 00:36.28 | opus_ | s,1,Goto($[${ARG5} = ""]?macro-dodial|s|4) |
| 00:36.42 | opus_ | or |
| 00:37.02 | MRH2 | hi i am getting a warning warning warning message after i make asterisk cvs 2day" /usr/....modules/ contains modules not installed by this version of asterisk" |
| 00:37.12 | SwK_ | s,1,Goto($["${ARG5}x" = "x"]?foo) |
| 00:37.19 | [hC] | opus_: thats what i had (except ,s,4 at the end) and the test kept failing |
| 00:37.21 | tzanger | MRH2: so rm -rf /usr/lib/asterisk/ and reinstall |
| 00:37.27 | opus_ | s,1,GotoIf($[${LEN(${ARG5})} = 0]?order_pizza) |
| 00:37.28 | MRH2 | do i just delete the module directory and re do it |
| 00:37.36 | tzanger | it's saying you've got old shit in there, or custom shit that isn't part of the normal install |
| 00:37.42 | tzanger | I get it with the g729 codec, it's normal |
| 00:37.54 | tzanger | but as apps change and get renamed you'll get that if you odn't keep it clean between installs |
| 00:38.00 | MRH2 | ah ok thanks that what i thought |
| 00:38.36 | MRH2 | i have app_curl.so and codec_ilibc.so |
| 00:38.41 | opus_ | i spent like 16 hours trying to figure that out. you're price: 15 seconds |
| 00:39.00 | [hC] | interesting. |
| 00:39.17 | [hC] | I put the quotes back in and it works |
| 00:39.22 | [hC] | fuckin gotoif's. |
| 00:39.32 | MRH2 | cheers tzanger |
| 00:39.43 | [hC] | it wants GotoIf($["${ARG5}" = ""]?foo) |
| 00:39.56 | [hC] | not GotoIf($[${ARG5} = ""]?foo) |
| 00:40.16 | opus_ | use ${LEN(${ARG5})} |
| 00:40.18 | SwK_ | [hC]: if you have this: ,GotoIf($"[ in your dial plan that " is not supposed to go between the $ and [ |
| 00:40.22 | opus_ | LEN is for length |
| 00:40.42 | [hC] | swik: i know. |
| 00:40.48 | [hC] | Ill switch to len later |
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| 00:46.42 | [hC] | so apparently |
| 00:46.42 | [hC] | <PROTECTED> |
| 00:46.44 | [hC] | is valid |
| 00:46.53 | [hC] | with the quotes after the first ${ |
| 00:47.07 | [hC] | did that change recently or something, cause cvs head wants the quotes there |
| 00:47.33 | Ariel_ | [hC], I have always used the " |
| 00:47.46 | Rowter | for what are the tdm400 power plugs_ |
| 00:47.47 | Rowter | ? |
| 00:47.51 | Sedorox | lilo: you seem to do a lot of work with the v6 servers :p |
| 00:47.57 | [hC] | <PROTECTED> |
| 00:48.01 | newmember | how far of a cat3 run can I have off of a FXO port? can it go 500ft? |
| 00:48.04 | Rowter | tdm04B power plugs. |
| 00:48.12 | *** join/#asterisk hellagony (n=egutierr@200.121.219.198) |
| 00:48.38 | Sedorox | newmember: fxo or fxs? |
| 00:48.47 | newmember | FXO |
| 00:48.52 | Sedorox | if from the wall to the fxo... I don't think it really matters.. especially if its coming from the telco.. |
| 00:49.43 | newmember | I have an FXO interface on an * in the bottom of building in the server room, and I am on the 5th floor |
| 00:50.11 | newmember | not sure if there is enough power in those cards |
| 00:50.20 | Sedorox | you mean fxs.. then... so you want to have the phone connected to asterisk...? |
| 00:50.21 | tzanger | newmember: it's fine |
| 00:50.23 | newmember | maybe I run CAT6 |
| 00:50.39 | tzanger | 500' is nothing compared to the distance (kilofeet) that it has to go to some homes |
| 00:50.44 | newmember | for a fax machine |
| 00:51.09 | tzanger | newmember: all you need to worry about is crosstalk/interference. it's all based on a 20mA loop |
| 00:51.13 | newmember | pardon a small pbx |
| 00:51.19 | Sedorox | I would think its fine.. but dun listen to me.. listen to tzanger and he says its fine :p |
| 00:51.27 | newmember | thanks |
| 00:51.35 | tzanger | hahah |
| 00:51.57 | tzanger | FXS is a different matter, those cards aren't really designed to source tons of power and long runs means greater voltage drop to maintain the loop current |
| 00:52.05 | tzanger | might work but really they're not designed for longhaul |
| 00:52.38 | newmember | tzanger: thats what I thought, guess it cant hurt to try |
| 00:53.02 | newmember | maybe I can find a spare pair in the riser room on the BIX |
| 00:53.23 | Sedorox | lol |
| 00:53.26 | Sedorox | bix? |
| 00:53.46 | Sedorox | never heard that term |
| 00:53.49 | Ariel_ | newmember, you don't need cat5 or cat6 it's not going to change things |
| 00:53.57 | tzanger | bix blocks |
| 00:54.01 | tzanger | also known as 110 blocks IIRC |
| 00:54.04 | Sedorox | oooo ok |
| 00:54.07 | Ariel_ | newmember, it's the voltage amp lost that your going to run into. |
| 00:54.11 | *** join/#asterisk litage (n=nick@203.201.98.169) |
| 00:54.15 | Sedorox | punchdown blocks |
| 00:54.17 | tzanger | Ariel_: huh? |
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| 00:56.41 | Ariel_ | tzanger, the phone cable I have used for the tdm is a normal phone cable up to about 1000 feet. He does not need to go to a cat6 for 500 feet |
| 00:57.03 | tzanger | unless he's got some major interference issues |
| 00:57.05 | tzanger | but yeah should be just fine |
| 00:57.26 | tzanger | (interference with other wires running parallel to that 500ft run) |
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| 00:58.14 | newmember | thanks guys |
| 00:58.14 | bkw__ | eureka |
| 00:58.16 | *** join/#asterisk fugitivo (n=ajf@201.255.102.212) |
| 00:58.18 | newmember | sorry just on the phone |
| 00:58.19 | bkw__ | I found the fix for chan_iax2 |
| 00:58.27 | bkw__ | just don't use it! |
| 00:58.29 | bkw__ | :P |
| 00:58.32 | Sedorox | I'm hoping what I got on ebay is a FXS channel bank... tho I'm not sure |
| 00:58.41 | Sedorox | will be interesting to use that combined with the dslam I got ;-) |
| 00:58.43 | [hC] | bkw???? |
| 00:58.46 | [hC] | What fix! |
| 00:58.49 | [hC] | oh |
| 00:58.49 | SwK_ | hah |
| 00:58.54 | [hC] | you tricked me |
| 00:58.56 | [hC] | Wow |
| 00:59.06 | [hC] | ive been at the computer way too fucking long |
| 00:59.19 | [hC] | haha |
| 00:59.49 | Weezey | bkw_: what's the problem with iax2? |
| 00:59.51 | Sedorox | I wish I could : |
| 00:59.54 | Sedorox | <PROTECTED> |
| 00:59.59 | fugitivo | Weezey: please, don't ask |
| 01:00.06 | SwK_ | ok i'm, about to be pissed |
| 01:00.32 | SwK_ | someone is f'n with the Feed for "Dog" at the cable company |
| 01:00.39 | zippp | Weezey, implementation of iax2, not iax2 |
| 01:01.30 | *** join/#asterisk flashnet (n=root@200.61.65.203) |
| 01:01.34 | Ariel_ | iax2, so no good to use for many channels.... |
| 01:01.51 | blitzmere | well I am running out of options |
| 01:02.03 | fugitivo | i'm out of food |
| 01:02.10 | Ariel_ | bkw_, could the problem be with the trunk command in how it can put more then one call with the same ethernet overhead from the first call? |
| 01:02.13 | Weezey | so, iax2 is bad? Is that why my call quality with it licks? |
| 01:02.24 | *** join/#asterisk Brumle (n=brumle@brumle.com) [NETSPLIT VICTIM] |
| 01:02.36 | fugitivo | Weezey: no, it's not bad |
| 01:02.39 | SwK_ | Weezey: what he is saying is there is a bug in the chan_iax2 |
| 01:02.52 | SwK_ | IAX2 the protocol isnt the problem |
| 01:02.59 | Weezey | fugitivo: ahh, so *'s iax2 licks |
| 01:03.07 | [hC] | Haha |
| 01:03.14 | [hC] | Okay now i dont feel so bad. |
| 01:03.15 | blitzmere | SwK_ |
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| 01:04.12 | SwK_ | blitzmere |
| 01:04.21 | blitzmere | pm? |
| 01:04.35 | SwK_ | ask me in channel |
| 01:04.42 | SwK_ | it helps everyone |
| 01:04.55 | blitzmere | just i think I know you |
| 01:05.38 | Weezey | hrmm, what's the proper way to dial out a SIP peer? I keep getting circuit-busy when I'm registered and everything. |
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| 01:07.21 | blitzmere | well when I make a call on this system, the local rings as it its on hold, and when the other side answers the stupid elevator music is playing. how can I turn that off? any know |
| 01:07.22 | SwK_ | netsplits rule |
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| 01:07.39 | bkw__ | I really gotta get marks attention longer than two seconds to look at this IAX bug |
| 01:07.44 | SwK_ | blitzmere: what? |
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| 01:07.50 | SwK_ | sounds liek you gotta screwed config |
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| 01:08.23 | blitzmere | i know, but don't know where its screwed |
| 01:08.27 | Ariel_ | bkw__, do you think it's the turnk command in iax2 that is part of the problem? |
| 01:08.33 | bkw__ | no |
| 01:08.39 | bkw__ | because we don't have machines doing trunking |
| 01:08.57 | Ariel_ | I was just wondering ok |
| 01:09.16 | *** part/#asterisk newmember (n=newmembe@dsl-lkbn-66-18-211-34-cgy.nucleus.com) |
| 01:10.39 | opus_ | so, will there be a newer 1.2 beta release soon? |
| 01:10.40 | blitzmere | has anyone had this problem before |
| 01:11.03 | SwK_ | blitzmere: so paste your config to pastebin.ca and ask someone to look at it |
| 01:11.12 | opus_ | blitzmere are you using the "r" in Dial() ? |
| 01:11.45 | opus_ | the "r" option even. |
| 01:12.00 | opus_ | r == stands for root of all dial bugs |
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| 01:12.16 | SwK_ | opus_: kpf posted that there will be a beta2 out pretty soon |
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| 01:12.37 | opus_ | net splits today hmm.. |
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| 01:13.13 | redG | <PROTECTED> |
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| 01:13.51 | asteriskDOTbz | <PROTECTED> |
| 01:14.29 | Ariel_ | wow hi all echo echo echo.... |
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| 01:15.23 | tzanger | Ariel_: you can't fool us. you don't generate echo in an all-digital path :-) |
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| 01:17.05 | Ariel_ | tzanger, no but the split is doing a good job for me as it is. |
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| 01:17.43 | opus_ | somebody rediscovered nuke.c |
| 01:18.05 | tzanger | heh |
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| 01:25.27 | blitzmere | GRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRR |
| 01:25.45 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 01:26.50 | Ariel_ | can't these splits happen at 2 or 3 am |
| 01:26.53 | Ariel_ | argh |
| 01:26.59 | Hmmhesays | heh |
| 01:27.18 | blitzmere | i have looked over all my configs, and still have same freaking issue |
| 01:27.22 | DarthClue | split, split, split, split, split, split, split, split, split, split, split, split, split, split, split!!!!!!!!!!111111111111 |
| 01:28.47 | DarthClue | blitzmere: can you restate the issue so that it is a bit clearer? |
| 01:28.47 | Sedorox | <PROTECTED> |
| 01:28.47 | Sedorox | :p |
| 01:28.52 | Ariel_ | falalalala ohhh |
| 01:29.01 | *** join/#asterisk r0d3nt (i=anonymou@tinfoilhat.net) |
| 01:29.01 | blitzmere | well when I make a call on this system, the local rings as it its on hold, and when the other side answers the stupid elevator music is playing. how can I turn that off? any know |
| 01:29.06 | *** part/#asterisk Cybertoy (n=Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 01:29.47 | DarthClue | blitzmere: so you want to disable the hold music? |
| 01:29.53 | blitzmere | yes |
| 01:30.16 | Ariel_ | blitzmere, there has to be something in the extensions.conf you have setup. |
| 01:30.28 | DarthClue | blitzmere: wiki. but it should be as simple as changing a single line to turn of music on hold. |
| 01:30.28 | blitzmere | i will check it again |
| 01:31.45 | blitzmere | exten => 3001,1,Dial(SIP/3001,30,Ttm) |
| 01:31.45 | blitzmere | exten => 3001,2,Playback(invalid) |
| 01:31.45 | blitzmere | exten => 3001,3,Hangup() |
| 01:31.45 | blitzmere | exten => 3001,102,Voicemail(3001) |
| 01:31.45 | blitzmere | exten => 3001,103,Hangup() |
| 01:31.48 | Ariel_ | anyone heard from manxpower I am seen the news with peoples just flotting down the river |
| 01:32.02 | Ariel_ | (SIP/3001,30,Ttm) |
| 01:32.04 | blitzmere | is there something wrong with that |
| 01:32.04 | Ariel_ | remove the m |
| 01:32.10 | blitzmere | ok |
| 01:32.18 | Ariel_ | told you about an hour ago |
| 01:32.39 | DarthClue | Ariel_: i heard that he went inland but i don't know where. i was gone all weekend so i missed his departure. |
| 01:33.00 | *** join/#asterisk vetter (i=vetter@adsl-68-91-7-225.dsl.tulsok.swbell.net) |
| 01:33.13 | vetter | hey blitzrage, you around? |
| 01:33.17 | *** join/#asterisk ryansc_ (n=ryansc@c-67-164-188-180.hsd1.co.comcast.net) |
| 01:33.18 | Ariel_ | yes he said he was going up into missippii |
| 01:33.49 | DarthClue | yeah, should have gone to florida |
| 01:34.07 | Ariel_ | new orleans is now getting flooded as well. |
| 01:34.17 | Ariel_ | water coming in via the drains |
| 01:34.46 | *** join/#asterisk alk (n=tony@12-23-137-226.dhcp.dlth.mn.charter.com) |
| 01:35.13 | Ariel_ | DarthClue, ha... really now |
| 01:35.47 | DarthClue | Ariel_: i have one get within a few blocks to a mile every year. I've gotten used to them. |
| 01:36.19 | *** join/#asterisk alexis101 (n=a@toronto-HSE-ppp4329037.sympatico.ca) |
| 01:36.23 | Ariel_ | DarthClue, well I did not get used to them besides I lived in a pit hole of a place |
| 01:36.43 | DarthClue | i'm contemplating moving, just haven't decided where yet. |
| 01:36.54 | *** join/#asterisk jaike (n=jaike@203.131.137.76) |
| 01:37.15 | alexis101 | hello |
| 01:37.19 | Ariel_ | DarthClue, just don't come down here there are enough folks from the north arriving every day here. |
| 01:37.51 | DarthClue | Ariel_: where is here? i would probably go north, i prefer the cold, but then again, someplace more moderate would be even better. |
| 01:38.01 | alexis101 | anyone can point me a tutorial or something like this on how to use the realtime architecture on 1.2.0 |
| 01:38.09 | Ariel_ | as far south you can get and still be in the US |
| 01:38.17 | vetter | ok going pstn > * > softphone, which is the best codec? I have no zaphardware in the box, when I make a call sip to sip on the same lan, all sounds fine, but when calling pstn to * to sipit is barely audible |
| 01:38.20 | Ariel_ | ~docs |
| 01:38.24 | jbot | well, 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 |
| 01:38.41 | DarthClue | Ariel_: um, yeah, i'm not much in the mood for south texas nor south florida. |
| 01:38.44 | Ariel_ | ulaw |
| 01:38.58 | opus_ | you can move west |
| 01:39.11 | Ariel_ | west it's soon going to fall off the map |
| 01:39.13 | opus_ | we got volcanos and earthquakes. |
| 01:39.25 | opus_ | and apparently terrorist training camps |
| 01:39.30 | DarthClue | opus_: with my luck. the moment i set foot in california, the big one will wipe it off the map. |
| 01:39.34 | *** join/#asterisk lyroy (n=lyroy@modemcable117.123-202-24.mc.videotron.ca) |
| 01:39.42 | opus_ | don't move to california |
| 01:39.50 | opus_ | taxes are to high |
| 01:39.59 | opus_ | i want to move to idaho |
| 01:40.02 | DarthClue | i did like colorado when i was there a few years ago. |
| 01:40.15 | Ariel_ | hummm little island in the middle of the pacific maybe |
| 01:40.23 | Hmmhesays | with some hotties |
| 01:40.38 | opus_ | i have the beach front property in arizona in the next 10 years |
| 01:40.52 | lyroy | Does some know why I get this error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 01:40.53 | Hmmhesays | never, eVer forget the hotties |
| 01:41.38 | Hmmhesays | now if I just used asterisk for registering peers and i have say 5 boxes is there a way I can know which box that peer is currently registered to? |
| 01:41.53 | Hmmhesays | if the config on all of them is identical |
| 01:41.53 | vetter | i know that 729 requires a liscense, how is the quality tho? |
| 01:42.07 | Hmmhesays | g729 is decent quality, like talking on a cell phone |
| 01:42.14 | *** join/#asterisk Chuji (i=Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) |
| 01:43.10 | opus_ | people say it sounds great |
| 01:43.56 | Hmmhesays | sorry i'm a spoiled american |
| 01:44.34 | *** join/#asterisk Umaro (n=umaro@209.140.74.64) |
| 01:44.36 | Hmmhesays | anyone using cmd ldapget in production? |
| 01:44.50 | Umaro | Hey guys.. can anyone remind me what to put in ztdummy.h to enable the RTC support? |
| 01:44.56 | Umaro | bkw_: you around? |
| 01:45.33 | hellop | Ariel I live in Hilo. |
| 01:45.39 | hellop | Where U live? |
| 01:46.02 | hellop | South Point? |
| 01:46.26 | Ariel_ | hellop, I live in Miami, Floirda. I just want to go away to an Island out of the reach of hurricanes and everything. |
| 01:47.07 | hellop | Ariel, Actually I live in the Southern Most point in US, it's on big island of Hawaii.. |
| 01:47.29 | hellop | And, living on an island would predispose you to hurricanes I would think.. |
| 01:48.32 | Ariel_ | hellop, ok you got me... hummm I mean continental US But You get lots of Typhons there anyway |
| 01:50.14 | *** join/#asterisk methos (n=lot@S0106000f6694b86f.ed.shawcable.net) |
| 01:51.46 | *** join/#asterisk Moc_ (n=mochouin@modemcable135.109-70-69.mc.videotron.ca) |
| 01:52.23 | Moc_ | 24dbi antenna + a 300mw wireless card aint bad.. |
| 01:52.44 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 01:53.53 | *** join/#asterisk tabo2 (n=acura0@S010600119503044b.vc.shawcable.net) |
| 01:58.02 | *** join/#asterisk \etc\bin (n=squirrel@202.57.81.146) |
| 01:58.37 | \etc\bin | can 2 or more clients behind a nat connect to an asterisk server? |
| 01:59.23 | Weezey | vetter: I think g729 is a little worse than a cell, but it seems to depend who you're connecting to with it. |
| 02:00.06 | Weezey | I can go from my wifi phone to my asterisk box to my friend's wifi phone all with g729 (really just the phones) and it's perfect. |
| 02:00.25 | *** join/#asterisk Kumbang (n=unknown@167.205.24.5) |
| 02:00.27 | Weezey | Other calls with voip provider sound just okay and other times it sounds like balls |
| 02:00.28 | greekman | hi all |
| 02:00.56 | tabo2 | can I remove all outbounding routing |
| 02:01.05 | tabo2 | I don't want any restrictions |
| 02:01.07 | Weezey | \etc\bin: sure, I have 4 phones behind the same natted IP and they all connect to the outside * box |
| 02:01.12 | greekman | anyone get low zttest scores without irq sharing? |
| 02:01.32 | tabo2 | . |
| 02:02.43 | \etc\bin | Weezey, how did you do it? whats ur firewall? |
| 02:02.59 | tabo2 | weezey can u remove all outbounding routing |
| 02:03.03 | tabo2 | i don't want any restrictions |
| 02:03.05 | \etc\bin | im using iptables, cant make it work |
| 02:03.14 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-45.west.biz.rr.com) |
| 02:03.16 | Weezey | FreeBSD box, but it's all in the device |
| 02:03.25 | *** join/#asterisk wulfy814 (n=lorentz@dsl093-061-214.pit1.dsl.speakeasy.net) |
| 02:03.37 | Weezey | as long as your device does keepalives you're set |
| 02:03.41 | \etc\bin | Weezey, is possible with my setup? |
| 02:03.44 | Weezey | sure |
| 02:04.19 | \etc\bin | Weezey, any links on howto? |
| 02:04.39 | Weezey | tabo2: I have clients that end up connecting on port 60003 or whatever because 5060 is routed elsewhere. It all did it automatically too. |
| 02:04.48 | \etc\bin | Weezey, my colleagues in here wants to have it, but i cant do it to do with more than 1 clients |
| 02:06.03 | Weezey | well, you could set his on port 5061 just to get around it right away |
| 02:06.24 | Weezey | that's what sipura hardware does if it has more than one extension in the device. |
| 02:06.39 | \etc\bin | 1st - 5060, 2: 5061, 2:5062? |
| 02:07.12 | \etc\bin | how bout the rtp ports? |
| 02:07.23 | Weezey | same. |
| 02:07.34 | Weezey | wait... |
| 02:07.46 | Weezey | not same, you know, I've never forwarded the RTP ports |
| 02:07.50 | Weezey | I just allow them through |
| 02:08.01 | Weezey | that's the beauty of FreeBSD's natd. |
| 02:08.34 | \etc\bin | i thought it would be possible with my linux iptables? |
| 02:09.08 | Weezey | are you specifically forwarding the ports now, or are you actually using a nat daemon? |
| 02:09.14 | Weezey | Have you looked at: http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules |
| 02:09.54 | \etc\bin | forwarding... |
| 02:11.47 | greekman | anyone having issues with low zttest scores? |
| 02:12.17 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 02:12.31 | Weezey | I've always had a nat engine of some kind to do the work, that way it's always dynamic. Whether it's a FreeBSD machine or a Linksys router. |
| 02:12.55 | Weezey | greekman: Best: 100.000000 -- Worst: 99.975586 -- Average: 99.990175 |
| 02:13.08 | Weezey | what's zttest do anyway? |
| 02:13.35 | greekman | i'm not sharing any irqs and my card won't go beyond 99.975586 |
| 02:14.44 | *** join/#asterisk brookshire (n=matt@digium.com) |
| 02:14.52 | greekman | weezy what's the front side bus spped of your motherboard? |
| 02:14.59 | greekman | speed |
| 02:15.46 | *** join/#asterisk Kumbang (n=unknown@167.205.24.5) |
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| 02:20.23 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 02:22.40 | greekman | can anyone help with low zttest scores? |
| 02:23.21 | *** join/#asterisk adker (n=adker@67-136-213-101.dsl1.glv.ny.frontiernet.net) |
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| 02:27.38 | *** join/#asterisk criptos (n=criptos@201.135.121.4) |
| 02:27.55 | criptos | does anyone have used a telextreme service with asterisk? |
| 02:31.13 | criptos | ? |
| 02:31.16 | criptos | hi! :) |
| 02:31.39 | wulfy814 | ok, I'm trying to configure a "night mode" scenario that users can set |
| 02:31.53 | *** join/#asterisk Blake0PS (n=blake@blakeops.com) |
| 02:32.01 | wulfy814 | I've got extension 2000 , Set, ("NIGHTIME=1") |
| 02:32.06 | *** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com) |
| 02:32.17 | wulfy814 | and extension 2001, Set, ("NIGHTTIME=0) |
| 02:32.26 | Blake0PS | Are there any IAX softphones that have a voicemail indicator? |
| 02:32.48 | wulfy814 | no I'm thinking that I will use a gotoif in my PRI-IN context (where all pstn calls end up) |
| 02:33.11 | wulfy814 | to send them to context [night-mode] if the NIGHTIME=1 |
| 02:33.17 | wulfy814 | does that sound reasonable? |
| 02:35.11 | wunderkin | if your hours will always be the same, use gotoiftime |
| 02:36.18 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
| 02:37.03 | *** join/#asterisk Jzaalae (n=sk@216-220-249-60.midmaine.com) |
| 02:37.13 | wulfy814 | wunderkin: hours will not be the same |
| 02:37.17 | wulfy814 | hence the variable |
| 02:37.26 | wulfy814 | I don't really understand the gotoif command |
| 02:38.09 | greekman | anyone having issues with low zttest scores? |
| 02:39.13 | *** join/#asterisk flynux_ (i=b0lm38t@cl-8.bru-01.be.sixxs.net) |
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| 02:39.38 | wulfy814 | exten => s,1,GotoIf($NIGHTTIME=1?night-mode?mainaa) |
| 02:40.59 | criptos | check the sintax wulfy ant README.variables from your doc dir from your * source code.. |
| 02:41.03 | criptos | there is an example... |
| 02:41.41 | *** join/#asterisk j0sephi` (i=j0e@2001:6f8:99e:243:0:0:0:32) |
| 02:43.10 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 02:43.19 | kshumard_home | greekman, what kind of scores are you seeing? |
| 02:43.42 | *** part/#asterisk criptos (n=criptos@201.135.121.4) |
| 02:44.00 | *** join/#asterisk websae (i=websae@207-118-143-134.dyn.centurytel.net) |
| 02:44.01 | wulfy814 | ctiptos: syntax I believe :-) and I'm having trouble understanding the example |
| 02:44.26 | wulfy814 | damn, I meant criptos, kinda funny considering I was correcting your spelling |
| 02:45.22 | *** join/#asterisk mhnoyes_ (n=mhnoyes@user-2ivfi52.dialup.mindspring.com) |
| 02:45.46 | wulfy814 | ok - so it should be: exten = s,1,GotoIF($[${NIGHTTIME} = 1]?night-mode:default) |
| 02:46.50 | wulfy814 | the example isexten => s,5,GotoIf($[${varc} = 6]?99|1:s|6) |
| 02:46.59 | wulfy814 | what are the | 's for ? |
| 02:47.14 | wulfy814 | exten => 1,2,gotoif(condition?label1:label2) |
| 02:47.21 | wulfy814 | no pipes in it |
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| 02:48.51 | *** part/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg) |
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| 02:52.13 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 02:52.18 | criptos | whatis the sintax to search inside the digim list at google? |
| 02:52.39 | shmaltz | I get unresolved symboles wcusb.o when doing make install in /usr/src/zaptel |
| 02:52.52 | shmaltz | I don't have USB support loaded, do i need it? |
| 02:52.53 | criptos | dont wory shmaltz.. |
| 02:53.18 | criptos | nope.. that why there are unresolved symbols at wcusb.,.. becose u dont have usb support |
| 02:53.19 | shmaltz | criptos, this is just complaining like a lady, and I can just igonore it? |
| 02:54.01 | shmaltz | I understand that part, but will my wcte11xp module work fine? |
| 02:54.22 | shmaltz | even withou the usb support and the error that make gives me? |
| 02:54.31 | criptos | yeap... :) you just can ignore the module.. |
| 02:54.55 | criptos | doesn´t matter... yout wctdm.o module will work fine. I have the same issue, and the ctdm work ok |
| 02:55.07 | criptos | Any one is using telextreme service with asterisk? |
| 02:55.17 | shmaltz | thanks, criptos |
| 02:55.24 | drumkilla | criptos: "site:lists.digium.com asterisk is cool" |
| 02:56.36 | *** part/#asterisk criptos (n=criptos@201.135.121.4) |
| 02:58.14 | twisted | aiyee |
| 02:58.42 | *** join/#asterisk tugalone (n=tugalone@pcp0010318441pcs.avenel01.nj.comcast.net) |
| 02:58.56 | twisted | every time i see kopete i think kopecktate (sp?) |
| 02:59.20 | vetter | is it possible to offer diffrent MoH files for diffrenent tenants or call queues? |
| 02:59.38 | hardwire | pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel |
| 02:59.40 | hardwire | mo to the fo |
| 02:59.42 | hardwire | what would that mean.. other than its getting unplugged |
| 02:59.51 | hardwire | yes |
| 02:59.54 | twisted | today is backwards day |
| 03:00.22 | twisted | that means that an HDLC abort message has been recieved on the d channel |
| 03:00.46 | kshumard_home | vetter, Yeah, you can specify the music on hold class when you call MusicOnHold from the dialplan |
| 03:01.14 | hardwire | twisted: yeeh yeh |
| 03:01.17 | vetter | fantastic, thank you |
| 03:01.29 | hardwire | asteriskguru thinks that it may or may not be an interrupt issue |
| 03:01.34 | hardwire | where some device on the system took too long to sync |
| 03:02.01 | hardwire | but its on its own irq.. and the drives are doing nothing |
| 03:02.20 | kshumard_home | hardwire, How frequently does it happen? |
| 03:02.30 | hardwire | today.. its happened 5 time sin the last 10 minutes |
| 03:02.46 | hardwire | it kills every channel on the t100p card and then it suddenly goes up |
| 03:03.02 | hardwire | I am nowhere near the card or the server its in.. so I have no idea if somebody is just monkeying around |
| 03:03.11 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 03:03.34 | *** join/#asterisk RoyKa (n=roy@cm-80.111.22.187.chello.no) |
| 03:03.37 | kshumard_home | Are you positive the timing is set right on the card? Sometimes that can cause it... |
| 03:03.44 | kshumard_home | but you should have seen this before now if that were the case |
| 03:04.29 | hardwire | I just reboot the machine.. its working again |
| 03:04.42 | hardwire | but this is very strange behaviour |
| 03:04.52 | hardwire | nothing has changes for 10 days.. then poof weird errors |
| 03:05.08 | drray | what motherboard? |
| 03:05.19 | hardwire | its a gigabyte board.. |
| 03:05.23 | hardwire | I don't have the specifics |
| 03:05.34 | hardwire | but its worked nicely for over a month now.. |
| 03:07.11 | spackle | hardwire: I have similar issues with mine. Takes a power cycle, and it can be coaxed back. |
| 03:07.27 | tabo2 | spackel |
| 03:07.30 | tabo2 | its connected! |
| 03:07.34 | tabo2 | and in service |
| 03:07.37 | tabo2 | but when I call out |
| 03:07.41 | spackle | tabo, what did it take? |
| 03:07.42 | tabo2 | I get a fast busy signale |
| 03:07.47 | tabo2 | you are right |
| 03:07.56 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
| 03:08.00 | tabo2 | i have to type in my proxy ip + sip ip and username and pw |
| 03:08.05 | tabo2 | they have to match to that of sip.conf |
| 03:08.14 | spackle | tabo, who are you calling? Did you call extension 500? |
| 03:08.28 | hardwire | hmm |
| 03:08.29 | tabo2 | the phone is connected to asterisk, but when i call a number i get a fast busy signal |
| 03:08.40 | hardwire | iax to iax with both as friends is definatly not working |
| 03:08.42 | tabo2 | i think something wrong with my trunk |
| 03:08.53 | spackle | now you are into dial plans and contexts, tabo |
| 03:09.18 | tabo2 | which files |
| 03:09.22 | *** join/#asterisk Moc_ (n=mochouin@229-198-0-72-ppp.3menatwork.com) |
| 03:09.24 | *** join/#asterisk spootnick (n=julio@CPE-147-10-168-100.nsw.bigpond.net.au) |
| 03:09.26 | *** join/#asterisk amp (i=amp@66-192-64-146.gen.twtelecom.net) |
| 03:09.39 | spackle | tabo: extensions.conf - or are you using asterisk@home? |
| 03:09.43 | tabo2 | outbound routing and trunks? |
| 03:09.45 | tabo2 | yes sir |
| 03:10.04 | tabo2 | i just have |
| 03:10.06 | spackle | tabo, not familiar with it. |
| 03:10.12 | *** join/#asterisk diego_br (n=diegovie@200-208-62-23-mns.cpe.vivax.com.br) |
| 03:10.15 | tabo2 | exten => 200,1,Dial(SIP/phone1,20) |
| 03:10.47 | spootnick | does the Record (asterisk 1.2x) application has to be preceeded by Answer() ? |
| 03:12.09 | *** join/#asterisk PiruchoOSX (n=Mauricio@c-67-169-18-21.hsd1.ca.comcast.net) |
| 03:13.37 | PiruchoOSX | it is any GUI to manage asterisk PBX open source or something like that any help is welcome |
| 03:14.00 | PiruchoOSX | i am not very good with cli |
| 03:14.05 | fugitivo | AMP |
| 03:14.29 | PiruchoOSX | $$$$ how much |
| 03:14.37 | fugitivo | nothing |
| 03:14.41 | twisted | uhh |
| 03:14.52 | spootnick | won't help if you cannot get it installed. the installation does require some CLI |
| 03:15.01 | spootnick | lots of, ihmo |
| 03:15.10 | twisted | learn linux |
| 03:15.24 | twisted | THEN use linux stuff |
| 03:15.26 | PiruchoOSX | i im plaing with third lane now but i do not have 300 to pay |
| 03:15.39 | twisted | learn linux |
| 03:15.41 | twisted | THEN use linux stuff |
| 03:15.47 | PiruchoOSX | i am working on it |
| 03:15.54 | Ariel_ | PiruchoOSX, have you installed anything yet? |
| 03:16.02 | spootnick | PiruchoOSX: got mac? |
| 03:16.09 | twisted | er lol |
| 03:16.10 | PiruchoOSX | only asterisk 1.0.9 |
| 03:16.10 | twisted | mac |
| 03:16.22 | spootnick | twisted: apple, not the fruit |
| 03:16.25 | PiruchoOSX | running on fedora 4 |
| 03:16.34 | twisted | spootnick, uh, no shit sherlock |
| 03:16.37 | twisted | sh: line 1: unam: command not found |
| 03:16.38 | *** join/#asterisk RoyKa (n=roy@cm-80.111.22.187.chello.no) |
| 03:16.39 | twisted | Darwin Acantha.local 8.2.0 Darwin Kernel Version 8.2.0: Fri Jun 24 17:46:54 PDT 2005; root:xnu-792.2.4.obj~3/RELEASE_PPC Power Macintosh powerpc |
| 03:16.49 | spootnick | =) |
| 03:16.50 | Ariel_ | PiruchoOSX, if you want to get this going faster look at installing Asterisk@home it comes with everything presetup on an ISO |
| 03:17.06 | Ariel_ | AMP gui and everything you will need to get started |
| 03:17.08 | PiruchoOSX | ok thx |
| 03:17.09 | tabo2 | go get asterisk@homoe |
| 03:17.13 | tabo2 | it comes with CENTOS |
| 03:17.17 | twisted | oh lord. |
| 03:17.22 | tabo2 | once its installed |
| 03:17.33 | websae | asterisk@home isn't too bad |
| 03:17.36 | Ariel_ | PiruchoOSX, http://asteriskathome.sourceforge.net/ |
| 03:17.36 | tabo2 | u can type 192.168.0.40 and use the GUI |
| 03:17.42 | websae | but learn to use backend as well |
| 03:17.44 | tabo2 | or 192.168.0.160 |
| 03:17.46 | twisted | oh lord. |
| 03:17.48 | tabo2 | CLI is ok... |
| 03:17.49 | websae | not just GUI |
| 03:17.51 | tabo2 | just use vi alot |
| 03:18.09 | fugitivo | i don't see what is difficult with the CLI |
| 03:18.13 | PiruchoOSX | well i just load fedora a week ago |
| 03:18.20 | tabo2 | fedora is great |
| 03:18.24 | twisted | fedora sucks |
| 03:18.28 | PiruchoOSX | and i am not very good |
| 03:18.28 | fugitivo | well |
| 03:18.30 | tabo2 | but I had compatibility problems with my digium card =( |
| 03:18.33 | PiruchoOSX | hahhaha |
| 03:18.38 | tabo2 | asterisk@home solved all my problems |
| 03:18.40 | Ariel_ | fedora is oik but it's basicly red hats test bed |
| 03:18.41 | PiruchoOSX | it is good |
| 03:18.53 | tabo2 | twisted: did u try very 4?? |
| 03:18.55 | PiruchoOSX | i want to believe that |
| 03:18.57 | twisted | Tabo, yes |
| 03:19.03 | tabo2 | its very nice |
| 03:19.06 | twisted | tabo2, it blows. |
| 03:19.07 | Ariel_ | PiruchoOSX, save your self time and money use asterisk@home |
| 03:19.08 | tabo2 | but i don't like firefox |
| 03:19.32 | twisted | yeah, use *@home so you don't have to learn anything |
| 03:19.42 | twisted | it's much better that way |
| 03:19.46 | tabo2 | as a webdesigner... stuff looks different in explorer vs firefox |
| 03:19.51 | Ariel_ | twisted, you still learn allot from it |
| 03:19.54 | spootnick | the guys is trying to get something to work... geez |
| 03:20.01 | twisted | tabo2, yes, it does.. IE does things all broken like |
| 03:20.14 | twisted | MS has their own verison of W3C |
| 03:20.32 | PiruchoOSX | ok thx i will try asterisk@home |
| 03:20.41 | blitzrage | can the dialplan deal with ABCD DTMF? |
| 03:20.51 | Ariel_ | can't understand people when you can get started and working on a system that you can learn and not loose a user. |
| 03:20.55 | blitzrage | twisted: me too |
| 03:20.55 | tabo2 | I hate this call failed + busy signal |
| 03:20.57 | tabo2 | any ideas? |
| 03:21.35 | twisted | Ariel_, because the lazier you get, you don't learn from it. |
| 03:21.58 | twisted | you just get the mindset of "oh, it works now, cool." and are done with it |
| 03:22.04 | Ariel_ | twisted, well not all of us can learn from the cli from the beginning. |
| 03:22.24 | twisted | sure ya can |
| 03:22.43 | twisted | there are 100x more resources to learn with nowadays than there was when I started 2 years ago |
| 03:22.50 | Ariel_ | twisted, funny. I use asterisk, linux, and lots of different things with it. But I see if as it's a way to get started faster. But it still leads you to learning |
| 03:23.11 | twisted | Ariel_, that wasn't directed at you, that was contextual |
| 03:23.23 | Ariel_ | twisted, yes your right. When I started with asterisk 3 years ago it's was still on .5 and there was no wiki on it. |
| 03:25.04 | spackle | yeah, and all the pris ran on dirt roads, not copper. |
| 03:25.28 | *** join/#asterisk Moc__ (n=mochouin@229-198-0-72-ppp.3menatwork.com) |
| 03:25.35 | distortio | mmm pri's |
| 03:26.09 | spackle | Hi Moc! |
| 03:26.14 | tabo2 | spackle I think my voip provider is not good |
| 03:26.26 | Moc__ | tabo2 who is it ? |
| 03:26.26 | tabo2 | since I have digium 400tdm |
| 03:26.30 | tabo2 | can I just use PSTN |
| 03:26.31 | Moc__ | hi spackle |
| 03:26.39 | tabo2 | I got two ports |
| 03:26.43 | tabo2 | hooked up alread |
| 03:26.52 | Ariel_ | tabo2, more info is needed as to why your getting the error |
| 03:27.38 | tabo2 | well phone is hooked up |
| 03:27.43 | tabo2 | as there is no error light |
| 03:27.49 | tabo2 | and no service message is gone |
| 03:27.50 | blitzrage | anyone ever handle the ABCD DTMF digits from within the dialplan? (I know SendDTMF can send ABCD.... just need to recieve) |
| 03:27.54 | twisted | tabo2, depends on which boards are on the tdm400p |
| 03:27.57 | tabo2 | when i dial a number, it just goes fast busy |
| 03:28.01 | tabo2 | and says calls failed |
| 03:28.04 | spackle | tabo, cann you call voicemail or anything? |
| 03:28.08 | twisted | blitzrage, i believe it will... have you tried it? |
| 03:28.27 | tabo2 | twisted: fxsks - 1 |
| 03:28.32 | tabo2 | and fxsks =4 |
| 03:28.32 | twisted | no |
| 03:28.34 | twisted | no |
| 03:28.39 | tabo2 | 100m |
| 03:28.51 | tabo2 | spacke: how do u chekc voicemail? |
| 03:28.57 | twisted | what daughter boards are on the card? |
| 03:29.00 | tabo2 | *98? |
| 03:29.05 | Ariel_ | green or red |
| 03:29.10 | Ariel_ | love them colors |
| 03:29.31 | twisted | ugh. |
| 03:29.33 | Moc__ | they should have made a orange board... |
| 03:29.38 | Ariel_ | blue |
| 03:29.43 | twisted | this is too low level. |
| 03:29.51 | Ariel_ | rofl |
| 03:29.54 | blitzrage | twisted: how... with _A,1,NoOp(-- 'A' DTMF tone! |
| 03:29.57 | blitzrage | ) |
| 03:30.11 | twisted | blitzrage, you do realize that _ is only used for pattern matching |
| 03:30.25 | twisted | but yes, basically |
| 03:30.35 | twisted | _A will still work, but it's unnecessary |
| 03:30.42 | *** join/#asterisk _mwoodj_ (n=mwoodj@24.96.145.218) |
| 03:31.07 | fugitivo | green |
| 03:31.29 | spackle | tabo: you might need to add a 98 extension that looks the same as *98. |
| 03:34.44 | tabo2 | let me check |
| 03:35.09 | shmaltz | which one is the single span digium card when using lspci? |
| 03:35.11 | shmaltz | is it: |
| 03:35.11 | tabo2 | LOL |
| 03:35.12 | shmaltz | Jens Schoenfeld Intel 537 |
| 03:35.14 | shmaltz | ? |
| 03:35.20 | tabo2 | i don't have *98 in my extensions.conf |
| 03:35.36 | Ariel_ | Tabo2 yes you do |
| 03:35.41 | Ariel_ | wait |
| 03:37.06 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) [NETSPLIT VICTIM] |
| 03:38.15 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 03:38.42 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 03:41.09 | pygrammer | no one's here |
| 03:41.12 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
| 03:41.13 | pygrammer | i'm alone |
| 03:41.24 | glm2k | shmaltz: depends on the card i guess. my te110p is a Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 03:41.25 | spootnick | why would Monitor() not run? is it supposed to be used before Dial() ? |
| 03:41.39 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:42.04 | glm2k | pygrammer: this world is inhabited by lots of denizens...all quiet at this time :) |
| 03:42.14 | harryvv | http://seattletimes.nwsource.com/html/businesstechnology/2002460194_webvoip30.html |
| 03:42.25 | harryvv | microsoft has aquired a voip carrier |
| 03:42.49 | harryvv | a company that makes voip software |
| 03:43.15 | glm2k | uhoh. we're in trouble >:) |
| 03:43.16 | pygrammer | glm2k: it's a perfect time to be up and about |
| 03:43.25 | glm2k | pygrammer: i agree hehe |
| 03:43.48 | harryvv | glm2k, good book to buy on mysql and use it to learn with a laptop at work? |
| 03:45.22 | glm2k | the thing about books, and don't get me wrong, i love the o'reilly books...is they get old really fast. |
| 03:45.33 | harryvv | you mean stale |
| 03:45.39 | harryvv | dry reading |
| 03:45.43 | harryvv | ? |
| 03:45.46 | glm2k | better download the mysql manual and get a good mysql site |
| 03:46.02 | glm2k | i mean books get tied to certain version |
| 03:46.05 | harryvv | yea, it all needs to be on laptop as i have no net connection at work |
| 03:46.07 | drray | the o'reilly books are a good starting point |
| 03:46.13 | glm2k | on that, i agree |
| 03:46.36 | drray | I'm going to buy the asterisk o'reilly book just to see what I don't know |
| 03:46.45 | *** join/#asterisk serg_b (n=sergey@9i.ru) [NETSPLIT VICTIM] |
| 03:46.59 | glm2k | i hope it's based on the latest 2/3 versions |
| 03:47.03 | spackle | drray : the asterisk book or the switching to VOIP book? |
| 03:47.10 | drray | asterisk |
| 03:47.11 | glm2k | like 1.0.7-1.0.9 |
| 03:47.30 | drray | the one that is due out any day now |
| 03:48.02 | spackle | drray: switching is asterisk-heavy too, could have used a chapter about echo though |
| 03:48.08 | gek | i need a good provider for australia mobile calls |
| 03:48.13 | gek | they all suck |
| 03:48.16 | harryvv | switching? |
| 03:48.21 | gek | termination |
| 03:48.32 | harryvv | heavy in the explaination of the dial plan? |
| 03:48.34 | spackle | "switching to VOIP" Oreilly book. |
| 03:48.38 | glm2k | gek: i think that was mentioned on the mailing list |
| 03:48.43 | gek | voipjet is no good..sixtel is no good.. |
| 03:48.52 | gek | hm |
| 03:48.53 | harryvv | i used sixtel alot |
| 03:48.53 | gek | k |
| 03:48.56 | harryvv | has some problems |
| 03:49.02 | harryvv | mostly it works |
| 03:49.28 | *** join/#asterisk file (n=jcolp@mctnnbsa30w-156034035160.nb.aliant.net) |
| 03:49.38 | gek | i'm about to set up a gsm-voip connection in my apartment if i cannot find a decent carrier for australia mobiles |
| 03:50.11 | jaike | harry: sixtel has no support whatsoever |
| 03:50.17 | *** join/#asterisk marv (n=ilovekim@pcp01529782pcs.huntsv01.al.comcast.net) [NETSPLIT VICTIM] |
| 03:50.20 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-40.cybersurf.com) [NETSPLIT VICTIM] |
| 03:50.26 | harryvv | i know |
| 03:50.30 | *** join/#asterisk _mwoodj_ (n=mwoodj@24.96.145.218) [NETSPLIT VICTIM] |
| 03:50.31 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
| 03:50.42 | gek | i called sixtel's support once |
| 03:50.45 | gek | sounded like teh guy just woke up |
| 03:50.49 | gek | and he wrote down what i needed |
| 03:50.51 | gek | and didn't do it |
| 03:50.56 | Insanity5 | jaike - I beginning to think nufone is the same way :( |
| 03:51.27 | Nugget | heh |
| 03:51.30 | jaike | were also using txlink...good support |
| 03:51.38 | jaike | but having DID problems |
| 03:51.41 | Insanity5 | gek - How do you do a gsm - voip? |
| 03:51.50 | *** join/#asterisk bmg505 (n=leon@rndf-146-55-157.telkomadsl.co.za) |
| 03:51.55 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 03:52.36 | Insanity5 | jaike - How do you sign up for txlink anyways? :) |
| 03:52.45 | Insanity5 | JerJer- You around? |
| 03:54.20 | *** join/#asterisk julia (n=philippe@sunset-209-91-146-2.vianet.ca) |
| 03:54.48 | jaike | 8668954650 <-- txlink |
| 03:55.00 | jaike | theyre pretty decent..and cheap |
| 03:55.08 | IronHelix | gek- if you are looking to set up a gsm picocell and terminate its calls on voip, you are looking at a difficult situation |
| 03:55.11 | gek | Insanity5: http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html |
| 03:55.14 | gek | thinking of getting one of these |
| 03:55.19 | gek | i just can't get quality calls |
| 03:55.24 | IronHelix | ahh |
| 03:55.26 | gek | with anyone |
| 03:55.39 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 03:55.41 | harryvv | geek, what time of day making all these calls? |
| 03:55.45 | Insanity5 | gek - expensive :) |
| 03:55.57 | IronHelix | why use gsm- why not ip based termination? |
| 03:55.59 | Insanity5 | gek - All that will do is interface to a cell provider, right? |
| 03:56.04 | IronHelix | insanity- yeah |
| 03:56.24 | Insanity5 | Can't you get land line service cheaper? |
| 03:56.29 | gek | Insanity5: yeah. would buy a sim card and throw it in there. |
| 03:56.38 | IronHelix | or voip service cheaper? |
| 03:56.50 | IronHelix | i dunno about .au but over here SIP minutes are much cheaper than GSM |
| 03:56.55 | Insanity5 | IronHelix - Well I assume he's exhausted that option -- grin |
| 03:56.57 | gek | i cannot find any voip termination providers that give quality service to australia mobiles |
| 03:57.06 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
| 03:57.06 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) [NETSPLIT VICTIM] |
| 03:57.06 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-40.cybersurf.com) [NETSPLIT VICTIM] |
| 03:57.07 | *** join/#asterisk marv (n=ilovekim@pcp01529782pcs.huntsv01.al.comcast.net) [NETSPLIT VICTIM] |
| 03:57.13 | IronHelix | hehe |
| 03:57.19 | gek | one option may be to forward a ground line to one and call that from asterisk |
| 03:57.20 | gek | i dunno |
| 03:57.25 | IronHelix | :| that sucks |
| 03:57.37 | gek | so that is why i am considering going on my own |
| 03:57.45 | glm2k | anyone know how much a voiceblue box costs? |
| 03:57.55 | websae | what is voiceblue? |
| 03:58.03 | IronHelix | gsm sip gateway |
| 03:58.06 | Insanity5 | glm2k - that gsm box? It's somewhere in there. like 600-1000 |
| 03:58.08 | Insanity5 | + |
| 03:58.19 | Insanity5 | They sell it as an "engineering sample". |
| 03:58.26 | Insanity5 | It's not really made for retail, i'm guessing. |
| 03:59.01 | gek | my work uses a gsm card in their pbx to avoid ground-to-mobile charges |
| 03:59.22 | *** join/#asterisk E|nyPRI_ (i=les@S0106000bdb97681e.wp.shawcable.net) |
| 03:59.33 | E|nyPRI_ | Hi. |
| 03:59.47 | Insanity5 | gek - I guess ground <-> mobile is billed backwards in your country :) |
| 04:58.33 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 04:58.33 | *** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Speakers wanted! || A core dump is your computer's way of saying "Here's what's on my mind, what's on yours?" |
| 04:58.41 | shmaltz | E|nyPRI_, I'm not even running asterisk yet |
| 04:58.49 | jaike | had problems with vbr mp3s for musiconhold |
| 04:59.04 | shmaltz | it does 100% as well as 99.998.... |
| 04:59.09 | jaike | also make sure if your using mp3s to remove id3s |
| 04:59.43 | E|nyPRI_ | so, you can insmod the modules, and thats ok? |
| 05:00.10 | Rowter | jaike, where are the ids?... there are only the files mmh.. |
| 05:00.33 | *** join/#asterisk malverian (n=malveria@adsl-065-005-207-210.sip.gnv.bellsouth.net) |
| 05:01.46 | shmaltz | E|nyPRI_, correct |
| 05:01.53 | blitzrage | wow I'm dumb sometimes :) |
| 05:01.53 | E|nyPRI_ | and then ztcfg hangs? |
| 05:01.59 | E|nyPRI_ | whats -s do.. i always use ztcfg -vv |
| 05:02.05 | shmaltz | nope, ztcfg still works |
| 05:02.18 | E|nyPRI_ | ok, so... then whats the problem again? |
| 05:02.30 | shmaltz | yeah, this is where the trouble starts when using either -s (s is for shutdown) or rmmod it hangs |
| 05:02.43 | E|nyPRI_ | what kinda zaptel card |
| 05:02.56 | *** join/#asterisk Tabo (n=acura0@S010600119503044b.vc.shawcable.net) |
| 05:02.56 | shmaltz | TE110p |
| 05:03.07 | E|nyPRI_ | try moving it a slot over? or to different slots? |
| 05:03.29 | shmaltz | E|nyPRI_, don that, no luck |
| 05:03.29 | *** join/#asterisk Beccara (n=Beccara@219.89.208.80) |
| 05:03.38 | E|nyPRI_ | try another machine? |
| 05:05.19 | E|nyPRI_ | anyone got any max tnt ? (parts or complete units) |
| 05:06.03 | shmaltz | E|nyPRI_, it works on another machine |
| 05:06.30 | E|nyPRI_ | same OS on the other machine? |
| 05:06.38 | shmaltz | I'm going to try 1-0-7 |
| 05:06.48 | shmaltz | yeah, but different verison zap drivers |
| 05:06.58 | shmaltz | and I'm goint to try HEAD as well |
| 05:07.01 | shmaltz | c ya |
| 05:07.03 | shmaltz | gtg now |
| 05:07.10 | shmaltz | thanks E|nyPRI_ |
| 05:07.14 | MikeJ[Laptop] | 1.0.7? |
| 05:07.18 | shmaltz | yep |
| 05:07.21 | MikeJ[Laptop] | sigh |
| 05:07.21 | shmaltz | why? |
| 05:07.26 | MikeJ[Laptop] | old |
| 05:07.33 | shmaltz | so is 1.0.9 |
| 05:07.36 | shmaltz | HEAD is new |
| 05:07.39 | MikeJ[Laptop] | well.. kinda |
| 05:07.40 | shmaltz | heh |
| 05:07.45 | MikeJ[Laptop] | but yes |
| 05:07.56 | shmaltz | if it works for me it works |
| 05:08.00 | MikeJ[Laptop] | 1.2 will be soon |
| 05:08.11 | MikeJ[Laptop] | well.. rather 1.0.9 than 7 |
| 05:08.17 | shmaltz | anyhow, anybody seen this: |
| 05:08.19 | shmaltz | www.voip-info.org/wiki-asterisk+cmd+vmauthenticate |
| 05:08.38 | shmaltz | I like this one, its only available in HEAD |
| 05:08.44 | MikeJ[Laptop] | ok |
| 05:08.45 | shmaltz | but is very usefull |
| 05:08.49 | MikeJ[Laptop] | user 1.2 beta 1 |
| 05:08.51 | MikeJ[Laptop] | :) |
| 05:08.56 | MikeJ[Laptop] | use |
| 05:10.18 | tzafrir_laptop | there's a mail forwarding loop on asterisk-users . Seems to that looped messages are ones sent to subscriber deepak.dhiman@uf4.net |
| 05:10.40 | tzafrir_laptop | msg me for full headers |
| 05:14.00 | *** join/#asterisk pbxbart__ (n=pbxbart@ip-80-226-231-239.vodafone-net.de) |
| 05:14.06 | *** part/#asterisk pbxbart__ (n=pbxbart@ip-80-226-231-239.vodafone-net.de) |
| 05:15.27 | *** join/#asterisk spootnick (n=julio@CPE-147-10-168-100.nsw.bigpond.net.au) |
| 05:15.55 | spootnick | can i receive fax through asterisk with no more than a fax modem? |
| 05:16.08 | spootnick | (i have the feeling that the answer is no) |
| 05:19.26 | *** join/#asterisk IOscanner (n=IOscanne@c-67-166-249-43.hsd1.tx.comcast.net) |
| 05:20.08 | IOscanner | any anyone give a an idea of what I am doing wrong. I am trying to setup a 4 port FXS card for 4 analog phones |
| 05:20.30 | *** join/#asterisk blessen (n=Bless@140.99.23.26) |
| 05:20.44 | blessen | HI |
| 05:20.46 | IOscanner | I set signalling=fxo_ks |
| 05:20.51 | blessen | anyone heard of junctionnetworks |
| 05:20.53 | IOscanner | group=1 |
| 05:21.07 | blessen | are they good or bad..i wish to use them for connecting my asterisk server |
| 05:21.09 | IOscanner | then callerid="name" <100> |
| 05:21.17 | IOscanner | channel => 2 |
| 05:21.20 | *** join/#asterisk DarthClue (n=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 05:21.24 | IOscanner | I don't get any thing on the line |
| 05:21.27 | *** join/#asterisk aYCa\ (n=YuzNumar@server.ivinskis.kursenai.lm.lt) |
| 05:21.33 | IOscanner | am I missing something |
| 05:21.59 | blessen | anyone heard of junctionnetworks.......are they good or bad..i wish to use them for connecting my asterisk server |
| 05:22.24 | IOscanner | I have never heard of them |
| 05:26.48 | blessen | how is broadvoice....i tried to call them..but didnt even pickup the phone..just kept me waiting...:-( |
| 05:27.05 | blessen | they seems to be supporting asterisk server |
| 05:27.19 | blessen | i mean allowing asterisk to connect to their device |
| 05:28.19 | IOscanner | customer service sucks |
| 05:28.26 | niZon | bleh |
| 05:28.37 | IOscanner | they have had a week outage not too long ago |
| 05:28.37 | niZon | how do i properly start festival in daemon mode? |
| 05:28.43 | IOscanner | they are okay for the price |
| 05:28.47 | IOscanner | not the best |
| 05:28.53 | blessen | so who are the best |
| 05:28.59 | blessen | and the ones i can relay on |
| 05:29.07 | IOscanner | I will let you know when I find one...lol |
| 05:29.19 | blessen | ;-) |
| 05:29.45 | IOscanner | it is either bad customer service or bad service I have found. Once they have your money they don't care about you. Or they charge per min |
| 05:30.42 | IOscanner | So no one on here has setup an FXS line? |
| 05:31.03 | niZon | anyone? |
| 05:31.13 | tzafrir_laptop | what's the problem? |
| 05:31.24 | IOscanner | going once...lol |
| 05:31.55 | IOscanner | I can't seem to get it fxs working. I have a TDM400P with 4 FXS modules |
| 05:32.21 | IOscanner | I have ztcfg telling me I have them setup correct. I have one FXO port and 4 FXS ports |
| 05:32.26 | IOscanner | all is good there |
| 05:32.35 | tzafrir_laptop | niZon, on debian it is probably /etc/init.d/festival start . But you first have to edit /etc/init.d/festival and rem-out the 'exit 0' in the beginning |
| 05:32.55 | IOscanner | I setup he zapata.conf, but I get no signaling from the card |
| 05:33.23 | tzafrir_laptop | IOscanner, are the relevant modules loaded? lsmod | grep zaptel |
| 05:33.51 | tzafrir_laptop | Also: what OS? What * ver? |
| 05:34.09 | IOscanner | yes zaptel wcfxs and wcfxo |
| 05:34.14 | IOscanner | I checked that |
| 05:34.28 | IOscanner | I know the FXO card works |
| 05:34.35 | IOscanner | I can dial in |
| 05:34.51 | IOscanner | 1.0-08 asterisk |
| 05:34.53 | IOscanner | debian |
| 05:35.41 | IOscanner | Ah had chan_zap.so set to noload forgot. Turnned it off for testing |
| 05:36.17 | tzafrir_laptop | IOscanner, if it is a really new card, you may need zaptel 1.0.9 . zaptel 1.0.9 is basically 1.0.8 with extra definitions for identifying newer TDM cards |
| 05:37.03 | *** join/#asterisk grimse (n=grimse@p5481D7A7.dip.t-dialin.net) |
| 05:37.04 | tzafrir_laptop | IOscanner, how do you know that the cards work? they work on your system? |
| 05:37.41 | IOscanner | No it is an old card I have had it for 1 - 2 years |
| 05:38.11 | IOscanner | I know the FXO card works. I have never setup the TDM card on this box. someone else used them for a bit to play and it worked |
| 05:38.16 | IOscanner | so the card should work |
| 05:38.33 | IOscanner | Now I get audio data error |
| 05:40.28 | tzafrir_laptop | ~genzaptelconf |
| 05:40.29 | jbot | it has been said that genzaptelconf is a script to general zaptel.conf and zapata.conf snippet for TDM cards, availble from http://tzafrir.org.il/genzaptelconf or from the Rapid zaptel packaged. Ask tzafrir about it. ignore warning about missing ast-cmd. |
| 05:41.12 | tzafrir_laptop | IOscanner, do both cards show on /proc/zaptel/* ? |
| 05:41.20 | IOscanner | yep |
| 05:41.34 | IOscanner | I think it is an error in my zapata.conf |
| 05:41.47 | IOscanner | I am not sure how to configure for FXS. I have only setup FXO in asterisk |
| 05:42.02 | blitzrage | gah - why does everything try to authenticate with the guest user if I uncomment it, but if I comment it out, the far end authenticates correctly |
| 05:42.04 | IOscanner | I thought I would push fax lines on the 4 ports |
| 05:42.43 | distortio | IO: i could be wrong but i remember reading somewhere today that asterisk treats fxo/fxs cards the same |
| 05:43.00 | distortio | i dont use them, so i just scanned past it. |
| 05:43.34 | IOscanner | well I will try to use the genzaptelconf and see how that goes |
| 05:43.42 | IOscanner | I am testing with it now |
| 05:43.47 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 05:43.51 | IOscanner | I am not on a production box so I can play for once |
| 05:43.59 | IOscanner | I miss playing |
| 05:44.26 | opus_ | i sware i saw an experimental t.38 patch for asterisk a few days ago but i can't find it |
| 05:44.30 | opus_ | anyone see it? |
| 05:44.42 | IOscanner | really |
| 05:44.43 | IOscanner | cool |
| 05:45.16 | distortio | is there a way to configure asterisk so that it does not proxy media/rtp? |
| 05:45.42 | djin_ib | opus_: You mean http://www.ionidea.ua/oss/asterisk/? |
| 05:45.48 | distortio | so that the orig and destination endpoints transmit media direct, and the signaling only goes through asterisk? |
| 05:46.10 | opus_ | yeah! |
| 05:46.43 | opus_ | distortio have you tried it? |
| 05:46.47 | opus_ | anyone here try it? |
| 05:47.22 | opus_ | oh fuck. its for asterisk 1.0.7 |
| 05:47.48 | opus_ | that means i'll have to run two asterisk processes on my box. one for cvs head and one for 1.0.7 |
| 05:49.27 | distortio | opus, if there is a way to route media direct between endpoints that will also work (assuming you arent originating from a tdm card) |
| 05:50.06 | opus_ | canreinvite=yes ? |
| 05:51.26 | *** join/#asterisk Rowters (n=SilverDr@dsl-201-129-89-96.prod-infinitum.com.mx) |
| 05:51.33 | *** join/#asterisk sime_ (n=sime@c211-30-187-243.rivrw1.nsw.optusnet.com.au) |
| 05:51.59 | opus_ | so now all I need is a t.38 ATA adapter |
| 05:53.33 | bkw__ | WOOOOOOOHOOOOOOOOO RPID works |
| 05:53.37 | *** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3682410.sympatico.ca) |
| 05:53.51 | IOscanner | What could cause asterisk to do with Ouch ... error while writing audio data: : Broken pipe |
| 05:54.03 | bkw__ | thats not asterisk |
| 05:54.04 | bkw__ | thats mpg123 |
| 05:54.08 | bkw__ | when the pipe breaks |
| 05:54.09 | IOscanner | Yes T.38 mmmmmmmm |
| 05:54.16 | opus_ | i don't understand why people use the term TDM. Isn't everything TDM |
| 05:54.26 | bkw__ | opus_, no |
| 05:54.33 | IOscanner | What would cause it... |
| 05:54.35 | opus_ | my eye ball is TDM |
| 05:55.16 | IOscanner | it failes just after saying unable to register channel 1-2 |
| 05:55.30 | distortio | yes, but can you send phone calls over your eyeball? |
| 05:55.32 | opus_ | does every codec divides things by time and multiplexes them through a network? |
| 05:55.47 | *** join/#asterisk Defraz_ (n=t0tal@24-119-12-238.cpe.cableone.net) |
| 05:55.55 | opus_ | then, wouldn't G.711 be TDM... wtf |
| 05:56.48 | serg_b | opus_: multeplexes with what? |
| 05:56.59 | SarahEmm | anyone here running the xten pocketpc softphone? |
| 05:57.06 | opus_ | a network card |
| 05:57.10 | opus_ | that also does TDM |
| 05:57.14 | serg_b | :) |
| 05:57.45 | SarahEmm | sure |
| 05:57.50 | SarahEmm | but your regular POTS line isn't TDM |
| 05:57.50 | IOscanner | doh found it someone put a typo in the zapata.conf |
| 05:57.51 | serg_b | network is packet |
| 05:57.52 | SarahEmm | one call at a time |
| 05:57.53 | SarahEmm | nothing's muxed |
| 05:58.01 | distortio | What distinguishes coarse time-division multiplexing from packet switching is that the time-slots are pre-allocated to the channels, rather than arbitrated on a per-time slot basis. |
| 05:58.21 | serg_b | exactly |
| 05:58.54 | opus_ | TDM by itself doesn't even make sense if you say that then. |
| 05:59.21 | IOscanner | perfect |
| 05:59.40 | IOscanner | genzaptelconf work wow that is fast and easy |
| 06:00.07 | distortio | well now your debating telecom acronyms |
| 06:00.10 | af_ | sometimes a playback does not emit any sounds |
| 06:00.10 | opus_ | distortio i stand corrected. |
| 06:00.12 | *** join/#asterisk puowvip (i=ircuser@thegrey.diamond.org) |
| 06:00.17 | af_ | what can be? |
| 06:01.13 | opus_ | superdome is going to be evaucated. |
| 06:01.34 | opus_ | they're all moving to the .. thunderdome |
| 06:01.35 | opus_ | haha |
| 06:01.55 | distortio | for what? |
| 06:02.28 | distortio | ahh, huricane katrina |
| 06:02.57 | distortio | stupid huricane |
| 06:05.11 | *** join/#asterisk drmc_ (i=drmc@64.95.190.37) |
| 06:06.12 | *** join/#asterisk Delta34 (n=delta34o@198.87.24.253) |
| 06:10.26 | *** join/#asterisk gdsm (n=gdsm@e1-1.ns500-1.ts.milt.as9105.net) |
| 06:10.38 | *** join/#asterisk razu (n=razu@fw.voicenet.ee) |
| 06:12.09 | *** join/#asterisk comadreja (n=comadrej@120.Red-81-36-28.pooles.rima-tde.net) |
| 06:13.05 | Juggie | what the hell is the thunderdome |
| 06:16.19 | *** join/#asterisk kajtzu (n=kajtzu@shell1.fi.basen.net) |
| 06:16.28 | Juggie | but yes, New orleans is getting killed |
| 06:17.13 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
| 06:19.17 | cced | hi |
| 06:20.35 | *** join/#asterisk NoRemorse (n=bah@202.161.68.6) |
| 06:20.52 | *** join/#asterisk Rowter (n=SilverDr@dsl-201-129-89-96.prod-infinitum.com.mx) |
| 06:21.01 | NoRemorse | hi all, does anyone know of a decent sip softphone client for java2 or symbian phones? |
| 06:21.26 | *** join/#asterisk grimse (n=grimse@p5481D7A7.dip.t-dialin.net) |
| 06:21.33 | cced | who clone Tormenta2 ? |
| 06:22.29 | *** join/#asterisk asteriskph (n=yakumo@203.87.204.126) |
| 06:22.31 | SarahEmm | cced: who makes clones you mean? or what? |
| 06:22.41 | SarahEmm | sivana: you here? |
| 06:22.41 | cced | how to join maillist? |
| 06:22.56 | SarahEmm | ~mailinglist |
| 06:22.57 | jbot | somebody said mailinglist was Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 06:23.11 | cced | SarahEmm :clone Tormenta2 card .4 port E1 |
| 06:23.18 | SarahEmm | i know what it is |
| 06:23.20 | SarahEmm | what about it? |
| 06:23.38 | cced | Who have clone the card himself? |
| 06:23.57 | SarahEmm | oh |
| 06:24.02 | SarahEmm | cced: here's how to join the lists btw. http://www.asterisk.org/support/#mailing_list |
| 06:24.17 | cced | it puzzle me .it is too difficult to make that card.... |
| 06:24.37 | cced | o thanks a lot:) |
| 06:25.29 | SarahEmm | why does it have to be that specific card cced? |
| 06:26.47 | pygrammer | jesus getting wx/wxPython/wxWidgets to work is a PITA |
| 06:26.48 | cced | why does it have to be that specific card cced? // what means ? O:) |
| 06:27.24 | cced | do you have made that card? Tormenta2 card or E400p |
| 06:29.31 | cced | :) |
| 06:31.15 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
| 06:32.50 | SarahEmm | cced: err... no. i mean, why not buy a digium or sangoma card? |
| 06:33.10 | drray | I have a govarion card which is a tor2 clone of the zapata card |
| 06:33.59 | *** join/#asterisk Fraeggl (n=Fraeggl@rkom.r-kom.de) |
| 06:34.13 | cced | SarahEmm : do it .and can study it ... |
| 06:34.39 | cced | SarahEmm : I like to do it yourself. |
| 06:35.17 | cced | drray : does your card work well?do it work with E1 line or T1 lines? |
| 06:46.12 | *** join/#asterisk \etc\bin (n=squirrel@202.57.81.146) |
| 06:46.27 | \etc\bin | anyone tried calling a talkpro.net user on asterisk box? |
| 06:46.43 | \etc\bin | what is the access code for talkpro.net voip users? |
| 06:46.57 | drray | cced - it works fine, I use it with t1 lines, and channel banks |
| 06:47.17 | drray | there is no echo DSP stuff on it |
| 06:48.26 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 06:48.41 | cced | drray : O:) does it test under heavy load? |
| 06:50.52 | drray | cced - I'm driving a hotel with it, so at most I have 11 lines in use at once |
| 06:50.54 | SarahEmm | cced: err... but buying a premade clone isn't doing it yourself any more than buying a digium card tho |
| 06:53.20 | blitzrage | evening all |
| 06:53.27 | SarahEmm | hihi |
| 06:54.06 | pygrammer | it's the morning |
| 06:54.08 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 06:54.09 | pygrammer | 2:54 AM |
| 06:57.20 | drray | we've been very happy with it, we originally bought a 1 port digium card, but when it came time to add another port, we went with the govarion card since we use so little in actual resources |
| 06:57.35 | *** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it) |
| 07:03.04 | *** join/#asterisk dev2005 (n=dev2003@222.33.36.205) |
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| 07:10.32 | *** join/#asterisk marcvw (i=marcvw@lit75-2-82-225-244-138.fbx.proxad.net) |
| 07:10.36 | marcvw | . |
| 07:13.08 | pygrammer | . |
| 07:13.41 | SarahEmm | @ |
| 07:18.18 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:19.46 | blitzrage | ok, bedtime now that my dialplan works |
| 07:19.48 | blitzrage | night! |
| 07:20.14 | SarahEmm | nini |
| 07:20.22 | blitzrage | SarahEmm: night! |
| 07:24.08 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
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| 07:24.31 | *** join/#asterisk jero (n=sflphone@savoirfairelinux.net) |
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| 07:24.31 | *** join/#asterisk mogorman (n=mogorman@digium.com) |
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| 07:25.13 | *** join/#asterisk trig (n=jb@xob.neospire.net) |
| 07:25.31 | santoshr | hello people... i am a call coming in from a fxo box, but asterisk down not understand tht the call has been disconnected by the person callin |
| 07:26.11 | SarahEmm | santoshr: what card in the * box? |
| 07:26.24 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
| 07:26.47 | *** join/#asterisk Corndawg (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
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| 07:27.28 | *** join/#asterisk colombus (n=colombus@193.190.210.151) |
| 07:28.18 | Umaro | any sangoma guys here? :) |
| 07:28.52 | cced | who use E400P? |
| 07:29.45 | *** part/#asterisk colombus (n=colombus@193.190.210.151) |
| 07:30.01 | santoshr | what exactly does wht card mean.. please pardon me because i am kind of a newbie |
| 07:30.40 | santoshr | its a voip gateway.. which has telephone lines comin in.. and a wan connection |
| 07:33.04 | *** join/#asterisk crich1999 (n=crich@p54BFA88A.dip0.t-ipconnect.de) |
| 07:33.06 | *** join/#asterisk flashnet (n=root@200.61.65.203) |
| 07:33.26 | crich1999 | marcvw, hey how are you |
| 07:33.45 | asteriskph | santoshr: what voip gateways? |
| 07:33.56 | asteriskph | santoshr: brand and model? |
| 07:35.03 | santoshr | 4 port fxo.. welltech |
| 07:35.30 | asteriskph | hmm so well tech has fxs port, and ethernet port right? |
| 07:36.02 | asteriskph | your gw has fxs port and ethernet? |
| 07:36.56 | santoshr | its got 4 telephone line ports and wan port |
| 07:36.56 | *** join/#asterisk crich1999 (n=crich@p54BFA88A.dip0.t-ipconnect.de) |
| 07:37.00 | *** part/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
| 07:37.44 | *** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it) |
| 07:39.38 | santoshr | it has 4 fxo and a wan |
| 07:44.02 | *** join/#asterisk MmmmToop (n=chatzill@196-11-241-43.mtn.co.za) |
| 07:44.44 | *** join/#asterisk marcvw (i=marcvw@lit75-2-82-225-244-138.fbx.proxad.net) |
| 07:45.34 | *** join/#asterisk Nix (n=Nix@81.214.255.57) |
| 07:47.36 | santoshr | the call is sent to asterisk by a fxo gateway .. suppose the cal the exten dialed makes h323 call to a fxs instrument which is registered with some gatekeeper. |
| 07:48.02 | *** join/#asterisk Gronker__ (n=Gronker2@adsl-220-86-7.ags.bellsouth.net) |
| 07:49.00 | santoshr | now when reciever is hung up on the fxo instrument.. the asterisk just keeps dialing and then times out. the ringing and exits stating no answer. |
| 07:49.20 | santoshr | my question is how to make him listen to the disengage by the calling party |
| 07:50.43 | *** join/#asterisk marcvw (i=marcvw@lit75-2-82-225-244-138.fbx.proxad.net) |
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| 07:53.08 | *** join/#asterisk NoRemorse (n=bah@202.161.68.6) |
| 07:54.08 | NoRemorse | hi all, I am trying to implement wakeupcalls with AGI but when the call time is verified it is being converted to a different timezone, I have my local unix time set to what the local time is (AEST) and tz=aest in voicemail.conf, have I missed anything please? |
| 07:54.14 | *** join/#asterisk fenlander (n=neils@82.152.81.57) |
| 07:54.17 | Netgeeks | Any Realtime buffs here? |
| 07:57.09 | *** part/#asterisk \etc\bin (n=squirrel@202.57.81.146) |
| 07:57.49 | *** part/#asterisk zoo (i=nobody@ip-176-16.travedsl.de) |
| 07:57.51 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 07:57.52 | wasim | NoRemorse: i doubt voicemail tz has anything to do with AGI ... |
| 07:58.38 | *** join/#asterisk razu (n=razu@fw.voicenet.ee) |
| 07:58.50 | *** part/#asterisk marcvw (i=marcvw@lit75-2-82-225-244-138.fbx.proxad.net) |
| 07:58.53 | *** join/#asterisk sigmounte (n=sigmount@213.251.156.186) |
| 07:59.01 | Faithful | Help... my VoIP sounds like a jackhammer at the other end all of a sudden & I have shutdown & restarted everything |
| 08:01.17 | wasim | Faithful: did you tell the other end to not construct sidewalks during calls? |
| 08:01.22 | Faithful | Hmmm it is only one phone doing it. |
| 08:01.52 | Faithful | and it is connected to a different switch than the others so it must be that. |
| 08:02.27 | Netgeeks | you awake twisted? |
| 08:05.56 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:06.39 | santoshr | people.. |
| 08:06.48 | SarahEmm | ostriches... |
| 08:07.05 | Netgeeks | *sigh* can I start shooting asterisk devs? |
| 08:07.12 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 08:08.45 | tuxinator_linuxM | SarahEmm: I see you |
| 08:09.11 | Netgeeks | know anything about realtime stuff Sarah? |
| 08:09.22 | SarahEmm | nope! |
| 08:09.27 | SarahEmm | i don't use realtime |
| 08:10.29 | Netgeeks | ah, it's just too bad developers can't also be implementors... things might actually work... *sigh* |
| 08:10.57 | *** join/#asterisk secure75 (n=mic@p549A0CAD.dip0.t-ipconnect.de) |
| 08:12.26 | santoshr | my question is how to make asterisk listen to the disengage by the fxo box sending th call to asterisk |
| 08:13.31 | wasim | santoshr: depends on the line signalling, it can if you have far-end-disconnect-supervision |
| 08:14.35 | santoshr | far-end-disconnect-supervision ...... what is this ? |
| 08:14.44 | santoshr | how to supervise. |
| 08:14.53 | wasim | santoshr: it lets you know when the far end has hung up the phone |
| 08:15.36 | santoshr | wasim: how do u implement disconnect-supervision |
| 08:15.47 | *** join/#asterisk argos73 (n=mike@65-85-207-101.client.dsl.net) |
| 08:15.48 | wasim | santoshr: i don't, fxo sucks |
| 08:16.53 | santoshr | wasim: i get the point tht u dont. but how does one implement.. ;) i mean.. can u guide me to some docs where in someone may have implemented such a thing or something ? |
| 08:17.39 | *** part/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3682410.sympatico.ca) |
| 08:17.45 | *** join/#asterisk sime_ (n=sime@c211-30-187-243.rivrw1.nsw.optusnet.com.au) |
| 08:17.45 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) [NETSPLIT VICTIM] |
| 08:17.45 | *** join/#asterisk juice (n=juice@mo-67-77-177-215.dyn.sprint-hsd.net) [NETSPLIT VICTIM] |
| 08:17.45 | *** join/#asterisk jero (n=sflphone@savoirfairelinux.net) [NETSPLIT VICTIM] |
| 08:17.45 | *** join/#asterisk newlook_ (n=newlook@203-59-168-152.perm.iinet.net.au) [NETSPLIT VICTIM] |
| 08:17.45 | *** join/#asterisk mogorman (n=mogorman@digium.com) [NETSPLIT VICTIM] |
| 08:17.45 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) [NETSPLIT VICTIM] |
| 08:17.45 | *** join/#asterisk ptiggerdine__ (n=ptiggerd@203-217-87-135.dyn.iinet.net.au) [NETSPLIT VICTIM] |
| 08:17.45 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) [NETSPLIT VICTIM] |
| 08:17.45 | *** join/#asterisk ptiggerdine (n=ptiggerd@c210-49-98-194.rochd1.qld.optusnet.com.au) [NETSPLIT VICTIM] |
| 08:17.46 | wasim | santoshr: well, _ks is one way, but that depends on your exchange |
| 08:20.20 | tuxinator_linuxM | Any of you familiar with California area codes? |
| 08:21.20 | *** join/#asterisk NoRemorse (n=bah@202.161.68.6) |
| 08:21.43 | NoRemorse | hi, has anyone here managed to get a ptt client connected to asterisk? |
| 08:21.53 | santoshr | what is _ks |
| 08:22.34 | NoRemorse | my phone has a ptt client that ties to auth with my * server when I put in it's ip address as the server details, but when I enter in the correct username and password details for the coresponding sip user I get an auth failure in asterisk logs |
| 08:24.10 | pygrammer | what time is it for all you all? |
| 08:24.37 | NoRemorse | 6:30pm |
| 08:24.45 | pygrammer | wow |
| 08:24.49 | pygrammer | you in aus? |
| 08:24.49 | santoshr | 13:45 in india |
| 08:25.16 | pygrammer | santoshr: your minutes are terribly off |
| 08:25.27 | pygrammer | NoRemorse: you in aussieland? |
| 08:25.59 | santoshr | pygrammer: terribly off to with repect to wht |
| 08:26.04 | NoRemorse | yes au |
| 08:26.26 | pygrammer | santoshr: the correct time ... |
| 08:26.32 | gordonjcp | ishikawa:~$ date |
| 08:26.33 | gordonjcp | Wed Aug 31 09:26:27 BST 2005 |
| 08:26.39 | pygrammer | ^ |
| 08:26.43 | pygrammer | goddamn stupid newegg is down |
| 08:27.01 | pygrammer | it's 4:30 AM here |
| 08:27.11 | pygrammer | 4:27 more accurately |
| 08:27.12 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 08:27.19 | santoshr | <PROTECTED> |
| 08:27.26 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 08:27.31 | gordonjcp | santoshr: your clock is 20 minutes fast |
| 08:28.02 | pygrammer | anyone listen to podcasts here? |
| 08:28.50 | gordonjcp | ooh that's odd |
| 08:29.10 | *** join/#asterisk zwu (n=me@62.218.44.178) |
| 08:29.16 | *** join/#asterisk Ja[M]eS (i=james@202.133.207.16) |
| 08:29.18 | santoshr | i am talkin abt abt +5.30 GMT |
| 08:29.19 | zwu | hi all |
| 08:29.26 | Ja[M]eS | hello all |
| 08:29.35 | zwu | got a stupig problem with suse 8.2 and the Mysql database |
| 08:29.36 | gordonjcp | santoshr: yeah, just reading about that |
| 08:29.44 | gordonjcp | and Nepal uses +5:45 |
| 08:29.52 | gordonjcp | ok |
| 08:29.56 | Ja[M]eS | hi zwu |
| 08:29.56 | gordonjcp | your clock is 10 mins slow |
| 08:30.14 | santoshr | guys u have ne idea abt far-end-disconnect-supervision.. |
| 08:30.16 | *** part/#asterisk secure75 (n=mic@p549A0CAD.dip0.t-ipconnect.de) |
| 08:30.22 | santoshr | how can it be implemented with fxo boxes |
| 08:30.49 | pygrammer | wait, so minutes vary across countries??? |
| 08:30.51 | pygrammer | that's news to me! |
| 08:31.19 | santoshr | welcome to the real world pygrammer |
| 08:31.20 | santoshr | ;) |
| 08:31.31 | pygrammer | ?? |
| 08:31.44 | pygrammer | i guess i wasn't in the real world because i didn't know minutes varied across the world :) |
| 08:33.08 | gordonjcp | pygrammer: .au has time zones that vary by 1/4 hours |
| 08:33.19 | santoshr | actually time is always supposed to be with respect to something.. me tellin 13:47 doesnt make sense at all |
| 08:33.32 | gordonjcp | you should all be using GMT, dammit |
| 08:33.39 | Ja[M]eS | im GMT |
| 08:33.56 | santoshr | exactly |
| 08:34.01 | pygrammer | gordonjcp: hm |
| 08:34.26 | pygrammer | k, good night all |
| 08:34.28 | *** join/#asterisk Jas_Williams (n=Jason@host86-130-0-82.range86-130.btcentralplus.com) |
| 08:35.32 | asteriskph | santoshr: check the fxs box disconnect suppervision |
| 08:35.42 | asteriskph | look for silence based disconnecy |
| 08:36.13 | NoRemorse | has anyone seen a symbian or java2 sip client for gsm phones? |
| 08:36.52 | santoshr | k lemme check thanks |
| 08:37.46 | *** join/#asterisk zoa (n=kkk@83.228.8.96) |
| 08:37.53 | santoshr | no but asteriskph it is a fxo box.. i mean the call is originating from the pstn > fxo > asterisk > fxs |
| 08:38.04 | asteriskph | ahhh |
| 08:38.16 | asteriskph | look for some config there.. |
| 08:38.20 | asteriskph | i have that also |
| 08:38.39 | asteriskph | it has answer and disconnect supervision... |
| 08:38.45 | santoshr | there is a Detect silence voice(FXO): |
| 08:39.04 | asteriskph | its enable? |
| 08:39.18 | santoshr | ya |
| 08:39.24 | asteriskph | ahhh |
| 08:39.33 | asteriskph | asterisk > fxs? |
| 08:39.36 | santoshr | but thr aint no disconnect |
| 08:39.42 | asteriskph | via digim card? |
| 08:39.47 | santoshr | dial (H323/ ) |
| 08:39.49 | santoshr | not h323 |
| 08:39.53 | asteriskph | sip? |
| 08:39.58 | santoshr | h323 |
| 08:40.21 | asteriskph | protocol is not the problem.. |
| 08:40.44 | asteriskph | asterisk detect if fxs is answered? |
| 08:40.47 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 08:40.48 | santoshr | ya agred.. but how should asterisk know tht the calling party has disconnected the call |
| 08:41.11 | santoshr | if i dont pick its times out the ringing.. stating no answer and exit 1 |
| 08:42.09 | *** join/#asterisk Aze` (n=aze@85-18-136-114.ip.fastwebnet.it) |
| 08:42.13 | asteriskph | hmmm but if you pickup... do you hear each other... |
| 08:42.26 | *** join/#asterisk r0m (n=SysOp@bl8-28-13.dsl.telepac.pt) |
| 08:42.36 | asteriskph | do u hear the other party... |
| 08:42.38 | santoshr | ya because the call has been disconnected by the callin party |
| 08:42.39 | r0m | good morning |
| 08:42.47 | asteriskph | morning too... |
| 08:42.49 | santoshr | if not dissconnected then i head |
| 08:42.58 | santoshr | i hear i mean sorry |
| 08:43.15 | asteriskph | hmmm |
| 08:43.49 | santoshr | ideally i do hear the voice cross over but .. the issu is if the callin party disconnects his phone..before the asterisk >fxs picks up.. it keeps ringing.. |
| 08:44.23 | asteriskph | im analysing... |
| 08:44.47 | santoshr | me waiting for the analysis report.. ;) |
| 08:45.47 | asteriskph | is your pstn provides reverse polarity signal? |
| 08:46.53 | santoshr | by pstn u mean pstn telephone provider or the fxo box. |
| 08:50.19 | santoshr | asteriskph: ?? |
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| 08:51.17 | *** part/#asterisk Gronker__ (n=Gronker2@adsl-220-86-7.ags.bellsouth.net) |
| 08:52.21 | santoshr | no asteriskph tell me wht r u thinkin.. ok so we get polarity reversal. when the call disconnects.. wht will tht polarity reversal do ? |
| 08:54.42 | asteriskph | wait.. |
| 08:55.09 | santoshr | ok.. |
| 08:55.11 | asteriskph | polarity reverse... send connect and disconnect signal to switch.... |
| 08:55.36 | asteriskph | you must modify the zapata.conf |
| 08:56.28 | *** join/#asterisk meppl (n=mephisto@p54AAC883.dip.t-dialin.net) |
| 08:56.32 | santoshr | no no.. let me get this correctly |
| 08:56.57 | tuxinator_linuxM | Night guys |
| 08:57.28 | santoshr | the callin party dc's and then becasue of the polariy reversal the fxo sends a disconnect request to asterisk |
| 08:57.31 | santoshr | correct |
| 08:57.39 | *** join/#asterisk zed_pda (n=zed_pda@du213-130-141-32.as15444.net) |
| 08:57.45 | santoshr | how does zapata.conf come into picture |
| 08:59.37 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
| 08:59.50 | cced | who use e400p card ?thanks |
| 09:01.43 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
| 09:04.16 | wasim | cced: i do, you're welcome |
| 09:04.24 | *** join/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg) |
| 09:04.28 | *** part/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg) |
| 09:06.47 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 09:07.41 | shadebob | hi, I have a small problem with API Manager. I want to make a zap destroy channel x but I have a msq permission denied. In manager.conf I have the "command" |
| 09:10.57 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 09:11.46 | faa_ | is it possible to answer incoming call in xlite form command line? |
| 09:11.57 | *** join/#asterisk Doofmannsgehilfe (n=chatzill@dsl-084-060-017-224.arcor-ip.net) |
| 09:13.51 | *** part/#asterisk jaike (n=jaike@203.131.137.76) |
| 09:18.25 | *** join/#asterisk djin_ib (n=djin_ib@217-19-19-230.dsl.cambrium.nl) |
| 09:19.15 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 09:19.20 | *** join/#asterisk mayank (n=mayank@210-210-81-203.lan.sify.net) |
| 09:19.37 | wasim | faa_: you mean from *CLI? |
| 09:20.21 | *** join/#asterisk basty (n=basty@212.218.65.232) |
| 09:20.23 | mayank | hey .. has anyone got the rate-engine addon to working ? (TROLLPHONE) |
| 09:20.23 | basty | Hi |
| 09:20.53 | faa_ | wasim no. from windows command line. |
| 09:21.02 | faa_ | wasim to call out i am running x-lite.exe <number> |
| 09:21.08 | faa_ | but how to answer an incomaing call |
| 09:21.26 | basty | I would like to know, if it is possible to setup Asterisk only for internal calls. Means if somebody calls "1234" the phonecall goes tho the Asterisk. If somebody calls "+49 2121 545112" - Asterisk should forward that to a external Voip-Gateway. |
| 09:21.27 | zwu | hmm what to do when the values in ./install_AMP are erroneous |
| 09:21.56 | darkskiez | basty: yes |
| 09:22.04 | mayank | basty: you can manage this thru the dialplan |
| 09:22.09 | basty | ah |
| 09:22.14 | *** join/#asterisk jonathh (n=asd@host217-46-145-65.in-addr.btopenworld.com) |
| 09:22.29 | darkskiez | basty: sounds like a typical setup to be honest. |
| 09:22.29 | basty | and the authentification goes tho the "sip.conf" with register ? |
| 09:22.46 | mayank | basty: thats right |
| 09:23.13 | zwu | anyone can hlep with AMP setup? |
| 09:23.20 | basty | dark: heh - all informations I read in the internet was like setting up an Asterisk with Digi-Cards and stuff. |
| 09:23.37 | santoshr | asteriskph: ???? |
| 09:23.39 | basty | mayank: Ah cool - are there any more informations on any webpage ? |
| 09:23.50 | mayank | anyone can help with the meaning of rate-engine's table columns ???? |
| 09:24.29 | shadebob | Anyone can help me with Action: Command with API manager? I have permission denied on all command |
| 09:24.48 | faa_ | shadebob did you configure manager.conf and reload? |
| 09:24.49 | mayank | basty: IMO, just reading about how to write a sip.conf and extensions.conf shd get you going. The best resources are ofcourse voip-info.org and the *docs |
| 09:25.05 | basty | okay..thanks much, mayank |
| 09:25.33 | basty | bye! |
| 09:25.37 | mayank | basty: it will aslo help <but not required> to see debug mesgs on ur softphone and *CLI ... so you know if register is happening , calls are going et cetera |
| 09:25.49 | shadebob | faa_ : read and write to system, call, log, verbose, command, agent, user... I m connected via telnet originate work but command no :( |
| 09:26.19 | mayank | REQD HELP: anyone can help me with the meaning of rate-engine's table columns ???? what some of the columns mean ? |
| 09:27.28 | faa_ | shadebob that was some answet or what? |
| 09:28.04 | shadebob | faa_ : Action: command |
| 09:28.04 | shadebob | command: show dialplan |
| 09:28.04 | shadebob | Response: Error |
| 09:28.04 | shadebob | Message: Permission denied |
| 09:28.41 | faa_ | shadebob show dialplan is cli command, no manager. |
| 09:29.17 | shadebob | faa_ but action command send a CLI request |
| 09:29.35 | faa_ | really? |
| 09:30.09 | shadebob | http://www.voip-info.org/tiki-index.php?page=Asterisk+Manager+API+Action+Command |
| 09:31.43 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:31.56 | faa_ | what you received whe you sent ACTION: COMMAND\ncommand: Show Channels |
| 09:32.12 | santoshr | asteriskph: we were discussing something abt polarity reversal. |
| 09:32.51 | shadebob | The same error : Permission denied |
| 09:33.16 | faa_ | shadebob pastr on some paste service your manager.conf |
| 09:34.54 | shadebob | faa_ : private window |
| 09:35.45 | zwu | HATE |
| 09:35.51 | zwu | HATE² |
| 09:36.13 | *** join/#asterisk Romik (n=romik_@212.143.5.146) |
| 09:36.14 | zwu | damn you install_AMP |
| 09:36.21 | zwu | put in wrong ip |
| 09:36.25 | zwu | now nothing works |
| 09:36.38 | *** join/#asterisk [Airwolf] (n=airwolf@hq.netgamez.nl) |
| 09:36.40 | zwu | anyone knows how to correct this? |
| 09:37.12 | Netgeeks | mayank: did you find someone to help you with rate_engine? |
| 09:37.45 | *** join/#asterisk burmese986419 (n=cheebuu@213.165.224.55) |
| 09:38.54 | *** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 09:39.25 | Netgeeks | Hello X-Rob |
| 09:40.12 | *** join/#asterisk secure75 (n=mic@ppp-82-135-72-130.mnet-online.de) |
| 09:43.44 | *** join/#asterisk damned (n=damned@prior.lanck.net) |
| 09:43.50 | damned | good time all |
| 09:45.06 | mayank | netgeeks: no |
| 09:45.50 | mayank | netgeeks: can you help :) ? |
| 09:45.55 | *** join/#asterisk zoa (n=kkk@83.228.8.96) |
| 09:51.24 | Netgeeks | yeah, I can help, mayank, what do you need to know? |
| 09:51.38 | shadebob | faa_ : ok I see in manager.c directive all |
| 09:51.51 | shadebob | so read and write pass to "all" and now it work |
| 09:51.57 | mayank | netgeeks: TIA, in the egress table ... peer corresponds to ??? |
| 09:52.08 | X-Rob | Bah |
| 09:52.12 | X-Rob | Bloody visual studio |
| 09:52.14 | X-Rob | *rebooting* |
| 09:52.22 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 09:53.10 | mayank | netgeeks: also, in egress ... what will be the understanding of _substitute_? ..... |
| 09:53.10 | mayank | it will really help if there is some documentation for the tables .... |
| 09:53.32 | Netgeeks | mayank: the egress table peer would be the peer statment... lets say you are trying to route to Zap/g1 for calls.... Then in egress for that route you would have technology = Zap and peer=g1 |
| 09:53.55 | Netgeeks | or if you wanted to send to SIP/vonage it would be technology = sip, peer = vonage |
| 09:54.38 | Netgeeks | pattern and substitute go together.... substitute is a perl regular expressions field where the pattern is provided as $1 |
| 09:55.02 | Netgeeks | so if you didn't want to change the number dialed, you'd set pattern to (.*) and substitute to $1 |
| 09:55.20 | Netgeeks | but if you wanted to say add 011 in front of the number dialed, you'd set substitute to 011$1 |
| 09:55.41 | mayank | netgeeks: got the peer thing |
| 09:56.24 | mayank | pattern/substitute in that case is effectively to change the number to be dialled? but in what case will we need this ? |
| 09:57.16 | *** join/#asterisk insomni_ (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 09:57.50 | mayank | ok, I understand there can be situations of the sort ... |
| 09:58.07 | Netgeeks | um, well, lets say you are smart and develop your rate tables in standard format of <country code><area code><city code><number>, but one of your providers wants you to prefix any calls for an 'international' number with 00, but another provider doesn't |
| 09:58.21 | Netgeeks | this way, you can change the appearance of the dialed number based on the provider |
| 09:58.31 | mayank | yes, right |
| 09:59.37 | Netgeeks | you know that the trollphone guys haven't updated the code since like july of last year, right? |
| 09:59.56 | mayank | in the rate table, wats the meaning of: iso, extra, firstperiod (is itt the minimum time for which the call is charged?), periods (is it the pulse?) trialcost? |
| 10:00.15 | Netgeeks | iso, dunno, I never use it... same with extra |
| 10:00.30 | Netgeeks | first period is for setting a different time frame for the first period |
| 10:00.40 | mayank | I will be writing a similar code myself and wanted to understand trollphone's way of doing things too. |
| 10:01.05 | Netgeeks | say you have a provider who says they bill you 0.25 for the first minute, and 0.20 per minute for all following minutes in 6 second increments |
| 10:01.22 | mayank | my coding is also 80% done, just wanted to check if some other guys have done things very diferently or if they hav more features |
| 10:01.40 | mayank | ok |
| 10:01.45 | Netgeeks | this allows you to set the first period to 60 (seconds) and set first period rate to 0.25 and then let the rest be 6 second periods charged at 0,20 per minute |
| 10:02.48 | mayank | ok. got this too |
| 10:02.53 | mayank | netgeeks, is there a doc for the rate-engine tables? or any other doc for rate engine in general, apart from their README ? |
| 10:03.03 | Netgeeks | you writing a LCR system? |
| 10:03.10 | mayank | yes |
| 10:03.16 | Netgeeks | nope, as far as I know, just thier readme |
| 10:03.24 | *** join/#asterisk Ganlron (n=Ganlron@omega.csolve.net) |
| 10:03.25 | Netgeeks | Ah nice, will it be open source? |
| 10:03.38 | mayank | yes, surely |
| 10:04.20 | Netgeeks | Very nice, I've been looking for something to replace rate_engine |
| 10:04.28 | mayank | its only a week's effort into it till now, so will take some time tough before i m set to upload ... and there will be no CDR :( in the first release |
| 10:04.52 | *** join/#asterisk Gerd (n=gmueller@zola.netways.de) |
| 10:05.00 | mayank | I'll be glad if you find it good enuf to replace rate-engine |
| 10:05.26 | *** join/#asterisk surfdue (n=surfdue@user-0c6t1g9.cable.mindspring.com) |
| 10:05.36 | mayank | thanks for all the answers n ur time |
| 10:05.49 | *** join/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 10:05.57 | Netgeeks | You are welcome, and good luck! |
| 10:06.18 | Akelavlk | Hello, has anybody experience with new asterisk version 1.2.0 beta 1? |
| 10:06.39 | Akelavlk | I got error "loader.c:314 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create" |
| 10:06.45 | *** join/#asterisk djin_ib (n=djin_ib@217-19-19-230.dsl.cambrium.nl) |
| 10:08.25 | Gerd | Hi all, how can I configure asterisk to let me dial something like this "sip://any.internet-address.com" |
| 10:09.23 | mayank | gerd: write a small application/agi script that will set for you a dialstring |
| 10:09.50 | *** join/#asterisk Ganlron (n=Ganlron@omega.csolve.net) |
| 10:09.54 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 10:10.27 | *** join/#asterisk Dybdahl (n=Lars@oelsted.dybdahl.dk) |
| 10:10.30 | Gerd | hm, no easier (direct) way? |
| 10:12.29 | mayank | depends on how all will you form the 'any.internet-address.com' .. can you elaborate ? |
| 10:14.58 | *** join/#asterisk Ganlron (n=Ganlron@omega.csolve.net) |
| 10:15.28 | Gerd | example: http://www.junghanns.net/en/contact.html (look at VoIP) |
| 10:20.45 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 10:21.11 | mayank | gerd: let me understand this: if for a given called extension, the call goes thru (is SUCCESSFUL) with iax://... then there's no need to try the 2nd DS with sip://... |
| 10:21.12 | mayank | but if the call to first voip DS fails, then try 2nd. if that fails then 3rd ... is this what you want ? |
| 10:22.16 | Gerd | I just want to call them over sip :-) |
| 10:22.42 | *** join/#asterisk sycofly (n=syco@sycofly.com) |
| 10:23.49 | mayank | oh. just define the sip address in sip.conf and use it in extensions.conf. ur done |
| 10:24.00 | *** part/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 10:24.05 | puzzled | morning all |
| 10:24.17 | mayank | you mus be puzzled :D |
| 10:27.10 | Gerd | mayank: thank you. I thought there might be a general way |
| 10:28.21 | *** join/#asterisk vlrk (n=vlrk@59.93.65.120) |
| 10:28.32 | vlrk | hello friends |
| 10:29.58 | serg_b | Gerd: maybe you need srvlookups=yes in sip.conf ? |
| 10:32.36 | *** join/#asterisk djin_ib (n=djin_ib@217-19-19-230.dsl.cambrium.nl) |
| 10:38.12 | *** join/#asterisk surfdue_ (n=surfdue@user-0c6t1g9.cable.mindspring.com) |
| 10:38.55 | *** join/#asterisk FedericoAlves (n=sales@c-67-191-30-58.hsd1.fl.comcast.net) |
| 10:39.35 | FedericoAlves | hello |
| 10:39.57 | FedericoAlves | anybody here wants to make a few bucks fixin my asterisk? |
| 10:40.07 | *** join/#asterisk Henguei (n=HYPOTEK@196.203.249.73) |
| 10:40.20 | Henguei | hello every body |
| 10:41.12 | Jas_Williams | FedericoAlves: Waht is the problem with it ? |
| 10:41.30 | Henguei | i install new sounds in var/lib/asterisk |
| 10:41.47 | Henguei | and i want to listen to it |
| 10:41.57 | jontow | Henguei: so do it :) |
| 10:42.12 | Henguei | but it s impossible using xmms |
| 10:42.47 | *** join/#asterisk Thumann (n=Thumann@217.157.30.66) |
| 10:42.50 | Thumann | hi all :) |
| 10:44.04 | Henguei | can play listen to it with another reader ?? |
| 10:44.15 | jontow | install sox |
| 10:44.24 | jontow | it includes the 'play' command that'll let you |
| 10:44.38 | Thumann | True/false: Only one ip can be registered to one sip account (using two phones for one account) |
| 10:44.46 | jontow | alternately, just make a floating extension that you plugin a soundfile to and dial it |
| 10:44.59 | jontow | and remove it when you're done |
| 10:45.35 | Jas_Williams | Henguei: Create a dial plan to play the sounds exten 1234,1,Playback(/var/lib/asterisk/sound_to_play) |
| 10:45.37 | Henguei | jontow asterisk use which reader to play sounds ? |
| 10:45.46 | *** join/#asterisk DannyF (n=dannyf@c-6d4fe353.24-0099-74657210.cust.bredbandsbolaget.se) |
| 10:45.57 | jontow | its native code.. it doesn't have a program to listen to them, it just supports the GSM codec. |
| 10:45.58 | Thumann | i'm trying to make a stationary and a cell phone ring at the same time.. but only the one who registers last gets the call.. |
| 10:46.45 | Henguei | OK Jas_Williams i ll try it |
| 10:46.46 | jontow | thumann; Dial(SIP/station1&Zap/G1/18885551212|20) ? |
| 10:47.16 | jontow | Thumann: honestly.. i've got the same thing going here.. |
| 10:47.45 | jontow | 1 DID rings my home, my old apartment (which is now 'camp'), my cell, my gf's cell, and a few VoIP phones in various locations |
| 10:48.06 | jontow | it's got a full PRI to do it with :) |
| 10:49.45 | damned | can anybody suggest what do i need to do if i whant my asterisk call to the list of numbers and play an audio file? |
| 10:50.43 | FedericoAlves | jas? |
| 10:51.08 | FedericoAlves | can you analize some core dumps and fix chan_h323? |
| 10:52.15 | *** part/#asterisk sivana (n=sivana@mixdown.ca) |
| 10:52.29 | Thumann | jontow: hmm.. i'm not sure i follow you.. that extension line you wrote rings 2 diff. numbers right? |
| 10:52.37 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 10:52.42 | puzzled | damned: if you use HEAD I think the Dial application supports playing a file |
| 10:53.23 | Thumann | damned: making phone spam are we? :D |
| 10:53.43 | Thumann | damned: dial a list of numbers and play a recorder message.. sounds.. icky.. ;> |
| 10:53.53 | Thumann | *recorded |
| 10:54.06 | damned | Thumann, no, that would be ballance announcing |
| 10:54.12 | damned | nothing criminal |
| 10:54.43 | *** join/#asterisk Beccara (n=Beccara@60-234-136-147.bitstream.orcon.net.nz) |
| 10:54.44 | Thumann | :) |
| 10:54.46 | *** join/#asterisk surfdue (n=surfdue@user-0c6t1g9.cable.mindspring.com) |
| 10:55.46 | damned | actually i'm new to asterisk, but i think it can originate calls |
| 10:57.30 | *** join/#asterisk RoyK (n=roy@cm-80.111.22.187.chello.no) |
| 10:59.50 | serg_b | damned: read wiki about * call files |
| 11:03.00 | *** part/#asterisk mayank (n=mayank@210-210-81-203.lan.sify.net) |
| 11:03.48 | Thumann | jontow: are you there ? :) |
| 11:06.17 | sivana | tzanger: ping |
| 11:08.10 | jontow | yes |
| 11:08.15 | jontow | and yes. |
| 11:09.17 | Jas_Williams | FedericoAlves: Not at that level you need to ask on the developers list chan_h323 works fine for me are you sure you are running correct versions of open h323 and have the enviroment correctly loaded |
| 11:09.55 | Henguei | jontow having sox can i listen to sounds of asterisk *.gsm *.ogg *.spx |
| 11:10.23 | jontow | Henguei: gsm yes, spx maybe, ogg .. doubtful -- i think you'll need to install vorbis-tools and use ogg123 from that. |
| 11:10.57 | Henguei | ok jontow thks |
| 11:11.15 | jontow | np |
| 11:11.27 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 11:11.58 | Thumann | jontow: well.. my problem isn't dialing 2 numbers at once.. it's making 2 sip phones work with the same number.. |
| 11:12.16 | jontow | a shared extension.. yes.. i think the phones have to support that, but im not sure.. |
| 11:12.20 | jontow | let me think for a couple seconds :) |
| 11:12.24 | Thumann | =) |
| 11:13.38 | jontow | honestly--i had this same problem with a client once, and never really managed to make it work |
| 11:13.51 | Jas_Williams | Thumann: Asterisk does not support the registration of two phones to the same number the last one to register gets the calls |
| 11:14.03 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 11:14.11 | Thumann | jontow: a solution could be to make a queue for every extension.. and add the the numbers too it.. like.. extension 701 and 801 |
| 11:14.17 | serg_b | Jas_Williams: why do you need two phones with same number ? |
| 11:14.27 | Thumann | Jas_Williams_ yes.. so i've found out... |
| 11:14.59 | Jas_Williams | Thumann: the only way round that is to have an other number that rings both phones or use SER |
| 11:15.14 | Thumann | SER? |
| 11:15.17 | jontow | in my experience, the problem is with line presence |
| 11:15.21 | Druken | uhmm.... put a pipe inbetween to call two phones with the same extension |
| 11:15.31 | Jas_Williams | SIP EXPRESS ROUTER |
| 11:15.46 | jontow | thats not really an acceptable solution for a lot of companies.. |
| 11:15.59 | Jas_Williams | works in partnership with asterisk to manage registration of SIP endpoints |
| 11:16.02 | Thumann | Druken: huh? |
| 11:16.32 | Thumann | Jas_Williams: so it's a piece of hardware/software? |
| 11:16.36 | Druken | Thumann: never mind.. i came in in the middle... disreguard |
| 11:16.50 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 11:17.16 | Thumann | Druken: hehe |
| 11:17.41 | Jas_Williams | Thumann: Software, A lot of carriers use it to manage the call control of sip devices, however it doe not do media conversion or anything with the voice stream |
| 11:17.59 | Thumann | http://www.iptel.org/ser/ <-- ? |
| 11:18.21 | Jas_Williams | Thumann: Yep |
| 11:18.34 | Thumann | so the sip phones register 2 the SER .. and the ser registers to the asterisk? |
| 11:18.40 | Thumann | *to the ser.. |
| 11:19.29 | Jas_Williams | Thumann: Yes, Asterisk dial plan delivers any calls to a range of numbers to SER for onward routing of the sip call like sending a call via sip to a carrier |
| 11:20.34 | Jas_Williams | Thumann: This is the way FWD and other carrires work, * controlls the PSTN lines and IAX, SER controlls the SIP devices |
| 11:21.08 | Thumann | i see.. |
| 11:22.22 | Jas_Williams | Thumann: http://www.voip-info.org/wiki-Asterisk+at+large |
| 11:22.24 | Thumann | soo.. lets say.. we have extension 101 which is James Bond.. james bond has 2 sip phones.. one registers to the ser with number 101 and the other with 102.. some setup is made at the ser.. that when it recieves 101.. it should dial both.. ? |
| 11:22.35 | *** part/#asterisk secure75 (n=mic@ppp-82-135-72-130.mnet-online.de) |
| 11:22.48 | Jas_Williams | Thumann: You can do that with asterisk |
| 11:22.53 | Thumann | exactly |
| 11:23.01 | Thumann | so again.. ?! :D |
| 11:23.23 | RoyK | Thumann: do you know SER any good? |
| 11:23.49 | Thumann | RoyK: not at all.. didn't know anything about it.. |
| 11:24.01 | Thumann | RoyK: just trying to figure out how it could help me.. |
| 11:24.08 | Jas_Williams | Thumann: say james bond has two phones one at home and one in his car both phone number 007 then both phones can register at the same time to SER and SER will call both phones when asterisk calls SER extension 007 |
| 11:25.16 | Thumann | ahh |
| 11:25.18 | Thumann | figures.. |
| 11:25.40 | Thumann | the ser doesn't 'cut off' the first one.. when the second one registers.. |
| 11:26.15 | Jas_Williams | SER can support thousands of sip phones, and upto 5 registers with the same number AFAIK |
| 11:26.23 | Thumann | cool |
| 11:26.32 | Thumann | and can it play the james bond theme? |
| 11:26.35 | Thumann | ;> |
| 11:26.52 | Jas_Williams | Nah you need asterisk for that :) SER cannot do audio |
| 11:27.05 | RoyK | Jas_Williams: do you know how to setup SER to just proxy all SIP REGISTERs? |
| 11:27.24 | Thumann | Jas_Williams: hehe |
| 11:27.43 | Jas_Williams | RoyK: No sorry I just know the theory I have not setup SER as I have no need of it .. |
| 11:28.00 | Thumann | hmm.. i'll try it out.. |
| 11:28.19 | *** join/#asterisk brc_ (n=brc@pdpc/supporter/basic/brc) |
| 11:28.32 | Thumann | Jas_Williams: thanks :) |
| 11:32.01 | drray | so I have a side question, I'm trying to write a ncurses based program to monitor asterisk, and I'd like to let people just telnet to it like a mud, is that possible? |
| 11:33.27 | hugo-v6 | drray: yep. but if u ask such question, im not sure it will be easy for you ;) |
| 11:33.59 | jontow | drray; keep in mind, MUDs don't usually acquire an ncurses-based interface.. |
| 11:34.08 | crash3m | drray: why not just setup nagios and point them to a web page? |
| 11:34.21 | drray | well, do I run it as a shell and let people log in, or do I have to use perl to emulate a telnet session? |
| 11:35.05 | drray | because I'm playing with perl/ncurses and I want to make a top like asterisk panel |
| 11:35.12 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 11:35.21 | jontow | i'd say you best use it as an after-login executable |
| 11:35.31 | jontow | and if you want a shell that runs only that.. then write a wrapper that allows nothing else |
| 11:36.57 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 11:36.57 | jontow | but, use the conveniences that the OS provided to you.. |
| 11:38.10 | Druken | jesus christ... 1.20 a litre for gas in toronto today.... |
| 11:38.23 | Druken | fucken rip me a new asshole a milewide |
| 11:39.34 | *** join/#asterisk brc__ (n=brc@pdpc/supporter/basic/brc) |
| 11:41.10 | *** join/#asterisk JamesDotCom (i=jamesdot@sweep.bur.st) |
| 11:44.20 | jontow | :/ |
| 11:44.35 | *** join/#asterisk r0m (n=SysOp@bl8-28-13.dsl.telepac.pt) |
| 11:45.16 | jontow | drray; on a good note.. i *really* like that idea.. |
| 11:45.35 | *** join/#asterisk Zaw (i=zaw@unaffiliated/zaw) |
| 11:48.45 | r0m | yellowz |
| 11:48.46 | vlrk | had any body successfull in the mrtg+asterisk till now ? |
| 11:49.44 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 11:49.50 | drray | which part, the top part or the telnet mud part? |
| 11:49.51 | drray | :) |
| 11:50.09 | *** join/#asterisk Malthus (n=dumbdumb@port0091-abr-s-adsl.cwjamaica.com) |
| 11:51.32 | *** join/#asterisk mariahh (n=lyhdiah@85-48-102-21.mad1.adsl.uni2.es) |
| 11:53.37 | jontow | the top part |
| 11:53.46 | jontow | i'd probably rewrite it if it wasn't a command i could run at the shell |
| 11:54.18 | jontow | btw, gonna use the manager interface to write it? |
| 11:55.27 | *** part/#asterisk mariahh (n=lyhdiah@85-48-102-21.mad1.adsl.uni2.es) |
| 11:58.20 | drray | I'm not sure |
| 11:58.35 | drray | I really want it to be a CLI version of Flash Operator Panel |
| 11:58.52 | jontow | yep, well.. that uses the manager interface -- so you just answered your own question |
| 11:59.13 | jontow | go download the operator panel, and take a look at the proxy they've written in perl to minimize load.. you're gonna need it |
| 11:59.24 | drray | yeah |
| 11:59.41 | *** join/#asterisk Ogun (n=johangri@h236n2fls34o865.telia.com) |
| 12:00.47 | drray | for starters it's just going to monitor the 11 outbound zap lines I have |
| 12:01.18 | drray | although I have a romantic fancy of making the phone number go bold when the line is ringing |
| 12:01.30 | jontow | :D |
| 12:01.35 | *** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com) |
| 12:02.36 | *** join/#asterisk MeTaLLiCa (i=YoNetici@62.162.14.37) |
| 12:03.03 | drray | what I want is a line 206-555-1212 00:00:00 < 300 TEST USER 912065551212 |
| 12:03.11 | drray | or |
| 12:03.36 | drray | 206-555-1212 00:00:00 > 300 TEST USER CALLERID UNAVAILABLE |
| 12:03.38 | drray | etc |
| 12:04.35 | Ogun | Is it possible to build 1.2.0 beta with bristuff? |
| 12:04.37 | drray | but I'm still dinking around with ncurses |
| 12:04.40 | *** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net) |
| 12:04.43 | puzzled | Ogun: don't think so |
| 12:04.58 | puzzled | drray: did you look at astmanproxy? |
| 12:05.29 | puzzled | Ogun: check the download dir at junghanns.net. maybe he updated it to work with HEAD again |
| 12:06.14 | Ogun | Or is it possible to use NT mode with chan_mISDN with a HFC card? IIRC chan_mISDN is in HEAD nowadays |
| 12:06.51 | puzzled | Ogun: yes you can use NT mode with chan_misdn but afaik it doesn't live in head. it's a separate module you need to install |
| 12:06.58 | *** join/#asterisk ZX81 (n=ZX81@222-153-80-230.jetstream.xtra.co.nz) |
| 12:07.18 | Ogun | puzzled: Thanks. I'll have a look around and see what I can find. |
| 12:07.28 | *** part/#asterisk ZX81 (n=ZX81@222-153-80-230.jetstream.xtra.co.nz) |
| 12:09.31 | *** join/#asterisk jeffik (n=Jeff@toronto-HSE-ppp3985822.sympatico.ca) |
| 12:21.17 | *** join/#asterisk _cleric_ (n=dacleric@p5482ACB5.dip0.t-ipconnect.de) |
| 12:22.24 | *** join/#asterisk Jugie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 12:22.34 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:26.00 | *** join/#asterisk olivier_ (n=olivier_@obs92-4-82-239-116-113.fbx.proxad.net) |
| 12:29.28 | *** join/#asterisk mupf3l (n=chatzill@dsl-084-056-225-034.arcor-ip.net) |
| 12:29.33 | mupf3l | hi |
| 12:29.37 | mupf3l | anyone here? |
| 12:30.16 | *** join/#asterisk fafnir (n=hello@tdds-gw.Moscow.gldn.net) |
| 12:31.38 | mupf3l | crash seems not to be a bot... ;-) |
| 12:31.57 | mupf3l | I am new to asterisk, does anyone have 5 min for me? |
| 12:32.27 | Nivex | ~data |
| 12:32.29 | jbot | Don't Ask To Ask. Just ASK |
| 12:32.29 | gordonjcp | well that all kind of depends |
| 12:32.39 | gordonjcp | indeed |
| 12:32.45 | gordonjcp | well said, that bot |
| 12:33.08 | mupf3l | ok, thats fair. well, I live in germany and just got my national flatrate |
| 12:33.26 | mupf3l | i have been using sip for a while |
| 12:34.02 | mupf3l | and would like to be able to use my sipphone for national calls through a isdn gateway |
| 12:34.24 | mupf3l | I think I can make it with asterisk, am I right? |
| 12:36.21 | *** join/#asterisk W|NGNUT (n=wingnut-@207.80-203-25.nextgentel.com) |
| 12:36.39 | mupf3l | right? |
| 12:37.04 | gordonjcp | yeah |
| 12:38.03 | mupf3l | cool. next question - what version to use. my suse offers me 1.0.6, there is a 1.2.0 beta on the site. |
| 12:39.02 | tzanger | um |
| 12:39.09 | tzanger | I've got a mail forwarding loop happenning here |
| 12:39.11 | tzanger | with -users |
| 12:39.15 | tzanger | what changed |
| 12:40.07 | *** join/#asterisk Samoied (n=samoied@popeye.opens.com.br) |
| 12:42.00 | RoyK | hmm |
| 12:42.10 | RoyK | can i use extconfig/realtime with pgsql without odbc? |
| 12:47.58 | *** join/#asterisk tclark (n=TC@S0106000c413a1c61.gv.shawcable.net) |
| 12:48.38 | *** join/#asterisk newl (n=newlook@203-59-168-152.perm.iinet.net.au) |
| 12:49.25 | mupf3l | the sip getting started is quite hard to understand for a newbie - could somebody help me to setup a sip2pstn gateway? |
| 12:51.33 | RoyK | mupf3l: just create sip accounts in sip.conf, assign a context to where they should end up, add that in extensions.conf and from there, Dial(Zap/..) or something |
| 12:51.44 | RoyK | (that was the asterisk howto in one sentence) |
| 12:51.45 | zwu | so lets try it once again |
| 12:51.58 | hugo-v6 | mupf3l: and on voip-info there are good docs to work with. |
| 12:52.02 | zwu | anyone here who can help me with the AMP install on SUSE8.2 ? |
| 12:52.13 | RoyK | AMP? |
| 12:52.15 | *** join/#asterisk Mavvie (i=edwin@252-131-222-203.rev.techex.net.au) |
| 12:52.22 | zwu | Asterisk Management Portal |
| 12:52.24 | mupf3l | just found that (voip-info) |
| 12:52.33 | hugo-v6 | zwu: they got a great setup-manual. read it and tdo the stuff told in it and it works |
| 12:52.41 | zwu | i know |
| 12:52.45 | zwu | im doint it with that |
| 12:52.56 | hugo-v6 | then u did something wrong. |
| 12:53.08 | hugo-v6 | zwu: btw: this is #asterisk not #amp |
| 12:53.08 | zwu | i only fear one of the last steps where i dont know wether it can be redone or deleted |
| 12:53.14 | hugo-v6 | aeh #amportal |
| 12:53.21 | zwu | there is a #amp too?!?!?! |
| 12:53.29 | zwu | really? |
| 12:53.32 | zwu | kewl |
| 12:53.41 | zwu | im sorry gotta go there |
| 12:53.42 | zwu | ;)) |
| 12:53.46 | hugo-v6 | well |
| 12:53.49 | zwu | <PROTECTED> |
| 12:54.03 | zwu | damn you keyboard |
| 12:56.38 | *** join/#asterisk ymorin (n=ymorin@savoirfairelinux.net) |
| 12:57.35 | *** join/#asterisk Praktikant01 (n=lars@dsl-084-059-129-197.arcor-ip.net) |
| 12:57.59 | Praktikant01 | hi, how can i transfer a established call to an other extension ?? |
| 12:59.02 | serg_b | 3-way call? |
| 12:59.07 | *** join/#asterisk likwid-- (n=likwid@nc-69-69-189-124.dyn.sprint-hsd.net) |
| 12:59.12 | Praktikant01 | i just want to be able to press #1 for example during a call and the incomming call would be moved in to a meetme .. |
| 12:59.39 | serg_b | features.conf and options tT for Dial cmd. |
| 13:00.12 | zwu | hmm @ hugo-v6 : its pretty dead in that channel |
| 13:00.46 | zwu | but ether way: its more of a linux specific question i think i have ;) |
| 13:00.49 | Praktikant01 | hm, i just find parking in that config-file, do jou have an example ?? |
| 13:00.58 | hugo-v6 | zwu: well. there is nothing i can do about it |
| 13:01.03 | zwu | i know |
| 13:01.10 | serg_b | [featuremap] |
| 13:01.10 | serg_b | blindxfer => *1 ; Blind transfer |
| 13:01.10 | serg_b | disconnect => *2 ; Disconnect |
| 13:01.10 | serg_b | ;automon => *1 ; One Touch Record |
| 13:01.10 | serg_b | atxfer => # ; Attended transfer |
| 13:01.23 | mupf3l | does anyone here use eyebeam? |
| 13:01.25 | Praktikant01 | serg_b: thx |
| 13:01.29 | zwu | but maybe you can help me anyway ? |
| 13:01.34 | Samoied | mupf3l: I |
| 13:01.41 | *** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg) |
| 13:01.42 | hugo-v6 | zwu: ask and well see |
| 13:01.48 | zwu | ok |
| 13:01.55 | *** join/#asterisk apardo (n=apardo@62-15-42-157.inversas.jazztel.es) |
| 13:02.01 | zwu | the Amp needs a webserver |
| 13:02.08 | mupf3l | samoied: what must I setup to make eyebeam register at asterisk? |
| 13:02.16 | zwu | and i think i got the apache on asterisk installed |
| 13:02.28 | serg_b | Praktikant01: or type: show features in cli |
| 13:02.37 | hugo-v6 | wzu: apache on astersik? |
| 13:02.42 | zwu | how do i find out where the "default" paths to the apache files are |
| 13:02.52 | Praktikant01 | serg_b: thx |
| 13:02.53 | zwu | like www/html |
| 13:03.01 | zwu | and /www/cgi-bin |
| 13:03.12 | Samoied | mupf3l: set username, password and server :) |
| 13:03.21 | zwu | or which other paths its using |
| 13:03.30 | mupf3l | zwu: have a look in /etc/apache(2)/httpd.conf and search for document root... |
| 13:03.41 | hugo-v6 | zwu: in your apache.conf |
| 13:03.44 | zwu | k will do ( that went fast9 |
| 13:04.08 | *** join/#asterisk popvoxdave (i=user@dave2.toad.net) |
| 13:04.21 | hugo-v6 | well yes net apcahe, but httpd :p |
| 13:04.32 | hugo-v6 | s/et/ot/ |
| 13:05.43 | bjohnson | I think there is an amp help channel |
| 13:06.17 | hugo-v6 | bjohnson: already told trhat |
| 13:06.36 | mupf3l | why can't I post private messages? aargh! |
| 13:06.45 | mupf3l | @samo: I have display name, username, pwd, auth name, domain |
| 13:06.55 | hugo-v6 | mupf3l: because ure not registered. |
| 13:07.22 | mupf3l | i tried to register mupf3l, it says its already... |
| 13:07.23 | hugo-v6 | mupf3l: new feature since spambot-actions increased. |
| 13:07.36 | mupf3l | could it be that I was not the first? |
| 13:07.37 | hugo-v6 | mupf3l: well tehn it is |
| 13:07.52 | mupf3l | hugo: but its probably not me... |
| 13:09.27 | mupf4l | ok, now I am registered |
| 13:10.02 | Praktikant01 | serg_b: hm, it ignores if i press *1 during a call |
| 13:10.49 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 13:12.17 | serg_b | Praktikant01: first, open cli, set verbose at least 3 and see what happens. there is two params in features.conf: featuredigittimeout and transferdigittimeout |
| 13:12.32 | serg_b | that controls how * detects features |
| 13:13.13 | serg_b | by default, featuredigittimeout=500, that means that you must press *1 within 500ms to activate feature |
| 13:13.35 | serg_b | try to set it to at least 1000 |
| 13:13.55 | gek | MUTT |
| 13:13.58 | gek | oops |
| 13:14.04 | gek | WRONG WINDOW |
| 13:14.39 | Praktikant01 | serg_b: thx |
| 13:15.07 | zwu | how do i start the mysqldaemon in suse? |
| 13:16.09 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 13:16.42 | Praktikant01 | serg_b: it isn't registered in asterisk cli that i try o anything, do i hav to disbale 3way calling ore overlabdial ?? |
| 13:17.04 | lathos42 | Does $2400 seem high to anyone else for a 2-line, 6 extension phone system for a Church? |
| 13:17.50 | tzanger | lathos42: what's it all include |
| 13:17.55 | tzanger | sounds like a typical CICS |
| 13:18.01 | tzanger | (norstar CICS) |
| 13:19.12 | *** join/#asterisk r0m (n=SysOp@bl8-28-13.dsl.telepac.pt) |
| 13:19.48 | MikeJ[Laptop] | norstar.. blah |
| 13:19.48 | *** join/#asterisk Guggemand (i=irc@80.198.131.49) |
| 13:19.50 | serg_b | Praktikant01: does module res_features loaded ? what says: show features ? |
| 13:20.37 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net) |
| 13:24.06 | *** join/#asterisk Hmmhesays (n=Hmmm@66.173.103.107) |
| 13:24.17 | Hmmhesays | morning |
| 13:24.19 | lathos42 | tzanger: I think its an Avaya Partner system with 12 voicemail boxes.. nothing fancy |
| 13:25.35 | Praktikant01 | serg_b: hm the comman is not available, but if i try load res-features it fails, il have a look if it is missing in xorcom |
| 13:26.16 | mupf4l | aaargh - the extension.conf is very pretty. ufff! is there a magic quote for "use capi" ??? |
| 13:26.37 | tzanger | lathos42: well that's about right then I suppose -- it's a proprietary closed system. You could use a pair of TDM cards and an old P3 to do the same thing |
| 13:27.14 | Praktikant01 | serg_b: ah res_features.so ;-), already exists, so it is available |
| 13:27.47 | *** join/#asterisk MeinNick2 (n=username@195.46.47.202.static.cablesurf.de) |
| 13:28.13 | serg_b | Praktikant01: what "show features" command says ? |
| 13:28.20 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:28.22 | lathos42 | tzanger: That's what I thought.. I told them that they need to at least shop around, and that I would let them know what I could do with * |
| 13:28.31 | tzanger | yup |
| 13:28.38 | bowman | heya. if I try to dial stuff like foo@bar.net from kiax, asterisk says: NOTICE[21141]: chan_iax2.c:6629 socket_read: Rejected connect attempt from 127.0.0.1, who was trying to reach 'foo@bar.net' - why? |
| 13:28.44 | Praktikant01 | serg_b: i just found it t not T ..., thx for jour great help |
| 13:29.09 | Ariel_ | morning everyone |
| 13:29.27 | Hmmhesays | Ariel_ I got to thinking about what you said last night about asterisk and ser |
| 13:29.48 | *** join/#asterisk foxb (n=foxb@Toronto-HSE-ppp3729961.sympatico.ca) |
| 13:29.52 | Ariel_ | Hmmhesays, ok |
| 13:30.54 | Hmmhesays | having all the peers register to a ser box makes complete sense. that way I can use a single dialplan distributed across multiple asterisk boxes, instead of having registrations and the dialplan distributed across multiple asterisk boxes |
| 13:31.35 | Ariel_ | Hmmhesays, now you got it |
| 13:31.43 | serg_b | Hmmhesays: why not to use dialplan switches ? |
| 13:32.02 | bowman | I can dial only pure numeric extensions :( (see above question) |
| 13:32.35 | bjohnson | lathos42: unless you find someone experienced with *, a voip system to do that will be about the same cost (but will be much more expandable) |
| 13:33.02 | foxb | I have problems running Festival with * can sombody help me? |
| 13:33.17 | *** join/#asterisk drbrown (n=chatzill@63.238.117.40) |
| 13:33.33 | tzanger | bjohnson: screw the voip stuff, just use the regular phones at this point |
| 13:33.37 | lathos42 | bjohnson: Well, i'm hopefully going to deploy * here at my day job, so I've been working with it for a couple of months now |
| 13:33.37 | Hmmhesays | 1 dual proc xeon box should be able to handle 4k registrations |
| 13:33.49 | tzanger | you can always fleabay the cards and go with SIP phones if htey relaly want to |
| 13:34.43 | lathos42 | tzanger: Well, they have an Avaya Phone system right now (i'm not sure what they have exactly) so i'd probably have to quote them new phones anyways |
| 13:34.48 | Hmmhesays | now I just have to look at the LDAP woes |
| 13:35.09 | nick125 | it seems that my asterisk 1.2 box will only say the 'thank you' message after i press a button again (in the queue) |
| 13:35.21 | tzanger | Hmmhesays: eh? |
| 13:35.25 | foxb | about festival in festival log I have accepted from localhost and disconnected after that how I fix this |
| 13:36.14 | tzanger | lathos42: I would throw a pair of TDM cards in a PC and thow it in the basement and see what they think. |
| 13:36.22 | Hmmhesays | tzanger, this company is using ldap right now and they want something that they can integrate, basically for user auth |
| 13:36.24 | tzanger | what's that cost, under $800? |
| 13:36.30 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe201.t-net.net.ve) |
| 13:36.40 | tzanger | actually that's not including phones . |
| 13:36.44 | tzanger | nice |
| 13:37.01 | Hmmhesays | in the dialplan i'd like to have something grab the ldap key, and route based upon that |
| 13:37.11 | tzanger | I wish I could get my head wrapped around LDAP better, I really think it's an amazing tech. I keep designing the schemas I think I'm gonna use and hten I go crazy |
| 13:37.30 | Hmmhesays | have you used ldapget in asterisk? |
| 13:37.40 | tzanger | Hmmhesays: no |
| 13:38.02 | epablo | Is there something I have to do so asterisk will handle varius IP's? |
| 13:38.09 | *** join/#asterisk daryl (n=darylj@swspool-102.introspect.net) |
| 13:38.27 | daryl | Hey all....just broke something and I'm not sure what I did..... |
| 13:38.40 | tzanger | well when you figure it out, undo it. |
| 13:38.52 | epablo | I just added an aditional IP + dns name to a server.. And some clients can't reg using it |
| 13:38.58 | daryl | Things were working, and after a reboot all of my zap channels load, then unload, and asterisk exits without an wrror. |
| 13:39.04 | daryl | error, that is.... |
| 13:39.10 | daryl | and yeah....I undid my changes. |
| 13:39.24 | daryl | I'm just looking for some suggestions on where to start diagnosing this one. |
| 13:39.55 | mupf4l | ok, as far as I understood, I would like to have a _0X. dial rule that uses my capi isdn card to make the call. the application is probably Dial - am I right? |
| 13:40.04 | daryl | Actually...it registers 23 PIR channels, and the first 2 of my TDM400B channels...then dies. |
| 13:40.57 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 13:41.05 | *** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 13:41.33 | daryl | Damn AMP. Nevermind. I'm an idiot. It changed my last 2 zap channels back to 3 and 4....and they're 99 and 100 with the new PRI card in. |
| 13:41.41 | daryl | Guess I'll have to figure out that automagic later. |
| 13:42.49 | nick125 | is there a way to customize what a queue does (such as the messages it says and such)? |
| 13:43.18 | drbrown | is it possible to detect if a SIP phone is on another line??? I am setting up an intercom system, and want to setup auto answer |
| 13:43.53 | *** part/#asterisk daryl (n=darylj@swspool-102.introspect.net) |
| 13:43.59 | puzzled | drbrown: I guess if you set call-limit to 1 then you will get a busy back. this is in head |
| 13:45.27 | Delvar | anyone know the UK hangup tone frequency? |
| 13:46.29 | lathos42 | Hmm.. I need to send out some emails to see if anyone will give me a good deal on 70 IP-501s |
| 13:46.38 | *** join/#asterisk mayank (n=mayank@210-210-81-203.lan.sify.net) |
| 13:46.52 | puzzled | Delvar: look in indications.conf. iirc there is a link to the ITU docs |
| 13:47.02 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 13:47.05 | Delvar | thanks ill take alook |
| 13:47.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:48.29 | *** join/#asterisk kshumard_home (n=ksh@pcp08979908pcs.huntsv01.al.comcast.net) |
| 13:48.45 | bowman | can I do pattern matching for extensions starting with a letter? |
| 13:48.50 | bowman | like [a-z]? |
| 13:49.03 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
| 13:50.06 | RoyK | bowman: why?? |
| 13:52.02 | bowman | RoyK: because asterisk doesn't like my dial commands ;) I have a _.,1,Dial(SIP/${EXTEN},20) which is not very intelligent but should work imho, if I dial "foo@bar.net" in kiax I get " NOTICE[21141]: chan_iax2.c:6629 socket_read: Rejected connect attempt from 127.0.0.1, who was trying to reach 'foo@bar.net'" |
| 13:52.24 | *** join/#asterisk Fivex (n=JordiH@80-102-195-208.bcn1.dialup.uni2.es) |
| 13:52.31 | bowman | which does NOT appear if I dial pure numbers |
| 13:52.40 | cypromis | _. ? |
| 13:53.02 | *** join/#asterisk Moc (n=mochouin@h66-201-214-109.gtconnect.net) |
| 13:53.03 | cypromis | so you are dialing on timeout and hangup as well ? |
| 13:53.23 | cypromis | sounds like endless call fun |
| 13:53.28 | bowman | cypromis: sure. it's only a test extension that will be removed :-) |
| 13:53.28 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
| 13:53.34 | lehel | hello |
| 13:54.53 | *** join/#asterisk coppice (n=chatzill@167.203.17.210.dyn.pacific.net.hk) |
| 13:55.00 | drbrown | puzzled: is it possible to change a variable when a call is disconected |
| 13:57.11 | puzzled | dunno |
| 13:57.45 | mupf4l | how do I test if my isdn card is working properly? isdn.log is not logging incoming calls... |
| 13:57.53 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 13:59.05 | Mimmus | hi, any idea why I'm not able to do attended call transfer? |
| 13:59.09 | bjohnson | drbrown: I think you can do that if you have a hangup extension |
| 14:01.02 | bowman | hm. obviously, either asterisk or kiax does something bad. I dial SIP/foo@bar.net, asterisk gets "SIP" as extension. I dial foo@bar.net, Asterisk tries to call foo at context bar.net instead of domain/host bar.net. any hint? |
| 14:02.37 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
| 14:03.44 | *** join/#asterisk bagohell (i=bagohell@209.151.138.178) |
| 14:04.30 | epablo | Has anyone done a SER-asterisk setup? I need to make config for a couple 100.000 users and read that this is the best way to do it |
| 14:04.38 | Hmmhesays | haha |
| 14:04.44 | Hmmhesays | i'm starting one right now |
| 14:05.11 | epablo | Hmmhesays: any pointers? a good link? |
| 14:05.21 | bagohell | i have a question about using asterisk as a PSTN pbx replacement. Do you need to have a FXS card for every phone on the system, or can 1 FXS card handle multiple phones? |
| 14:05.28 | Hmmhesays | well read the SER administration guide on iptel.org to start |
| 14:05.51 | *** join/#asterisk dtwilson (n=dave@host217-36-121-129.in-addr.btopenworld.com) |
| 14:06.07 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-181.sw.biz.rr.com) |
| 14:06.44 | darkskiez | SER brings me out in a cold sweat |
| 14:06.48 | Hmmhesays | once you have a good idea of how SER operates then things will start to click |
| 14:06.58 | bjohnson | bagohell: 1 fxs can handle multiple phones but then they will share the line (like a typically home phne system) |
| 14:07.05 | fulgas | i'm using 3 |
| 14:07.46 | bagohell | does that limit the ability to intercom a specific phone on the system? |
| 14:07.54 | bjohnson | bagohell: yes |
| 14:08.16 | *** join/#asterisk PiruchoOSX (n=Mauricio@c-67-169-18-21.hsd1.ca.comcast.net) |
| 14:08.19 | epablo | Hmmhesays: What I want to make sure is that the clients get their creds checked when dialing to PSTN. In this case I FWD the call to couple of * servers. |
| 14:08.58 | bjohnson | bagohell: typically not aceptable in an office setting |
| 14:09.08 | Hmmhesays | you are right on the money epablo |
| 14:09.09 | bagohell | bjohnson: thanks... but one more question, just so i get the hardware I need. If i wanted to be able to intercom each phone seperatly on the system, i would need a different fxs card for each phone |
| 14:09.14 | *** join/#asterisk Har|ock (n=himeko@S01060040ca128fc3.ed.shawcable.net) |
| 14:09.24 | bjohnson | bagohell: yes |
| 14:09.37 | *** join/#asterisk oej (n=oej@digium.com) |
| 14:09.46 | bjohnson | bagohell: or you could also mix in voip phones that wouldn't need fxs |
| 14:10.22 | epablo | Hmmhesays: How do I do that? the auth? |
| 14:10.24 | bjohnson | bagohell: if you're spending $100 USD or more on phones anyway, you should consider just buying voip phones |
| 14:10.43 | Hmmhesays | epablo there are a lot of ways you can do that |
| 14:11.10 | epablo | Hmmhesays: would you mind sharing a link ;) ? |
| 14:11.14 | bjohnson | bagohell: of course cabling may influence that too |
| 14:11.53 | Hmmhesays | there is no real link to do that, take a look at some of the dialplan apps on www.voip-info.org epablo |
| 14:12.04 | bagohell | in a voip system, can the phones all be voip, and the phone connection still be PSTN, or do you need a voip phone connection? |
| 14:12.38 | Hmmhesays | bagohell, yes you can have the phones be voip with pstn hop off |
| 14:12.51 | epablo | Hmmhesays: ok.. I'll take a look.. thanks. Another thing that is on my mind is the load balancing, how are you managing that? |
| 14:13.08 | Mimmus | I'm not able to do attended call transfer (only unattended) between PRI channels (on different trunks)? |
| 14:13.14 | Mimmus | why? |
| 14:13.58 | *** join/#asterisk konrads (n=konrads@out.ctkom.lv) |
| 14:14.08 | konrads | Hello |
| 14:14.15 | konrads | I am running asterisk-1.0.1 |
| 14:14.23 | konrads | wanted to add chan_misdn |
| 14:14.24 | Mimmus | konrads: upgrade! |
| 14:14.29 | konrads | Mimmus: that a requirement? |
| 14:14.37 | Hmmhesays | well what i'm looking at doing right now is using SER for user registration, and having multiple asterisk boxes for the dialplan, ivr, etc |
| 14:14.40 | bjohnson | bagohell: many systems (most small ones) are a combination of voip + analog |
| 14:14.45 | Mimmus | konrads: no but a general suggestion |
| 14:14.46 | bagohell | is it recommened to have the voip network seperate from your data network? |
| 14:15.02 | konrads | basically, asterisk loads the module and exits |
| 14:15.09 | konrads | unknown port(1) type 0x00000000 |
| 14:15.22 | bjohnson | bagohell: yes .. but in practice it depends on user data bandwidth usage |
| 14:15.27 | konrads | ah my bad |
| 14:16.07 | *** join/#asterisk jac]Z[oby (n=me@193.83.248.26) |
| 14:16.48 | konrads | okay, i am lost |
| 14:16.53 | konrads | how do I configure mISDN channels? |
| 14:16.55 | bjohnson | bagohell: did you say how big of a system you are doing? |
| 14:17.21 | bagohell | so in a voip/pstn configuration, the hardware required would be a FXO card, and 2 network cards (one for the voip, and one for the data), plus the asterisk system? |
| 14:18.14 | bagohell | no i never mentioned it... I am working on a test system, to see if its a viable solution to provide to my customers |
| 14:18.24 | *** join/#asterisk DrWho (n=MIKE@mike-new.tc3net.com) |
| 14:18.56 | Mimmus | konrads: I tried only with chan_CAPI |
| 14:19.08 | konrads | Mimmus: HFC-PCI doesn't work with just capi |
| 14:19.25 | konrads | i have no clue, tbh |
| 14:19.39 | Mimmus | konrads: sorry, no help. I had an Eicon DIVA 4 BRI |
| 14:20.14 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 14:20.22 | konrads | well, can you give me an overview of configuration of channels? |
| 14:20.30 | konrads | i managed to get sip and iax2 working |
| 14:20.34 | konrads | but isdn is a first |
| 14:20.53 | *** join/#asterisk Aleksandra (n=zheki24@P-3.184.EUnet.yu) |
| 14:21.35 | Mimmus | konrads: well, only some remembrances... now I'm using a PRI |
| 14:22.37 | *** join/#asterisk nel (n=nel@199.75.113.6) |
| 14:23.07 | nel | hello0 |
| 14:23.24 | Mimmus | konrads: no, I'm not able to find old configs, sorry... |
| 14:23.27 | *** part/#asterisk Aleksandra (n=zheki24@P-3.184.EUnet.yu) |
| 14:23.58 | nel | I'm having problems with a customer with echo and TDM400 digium cards |
| 14:24.06 | bagohell | can anyone point me to a website with some hardware hook-up diagrams? |
| 14:24.18 | nel | anybody has solved the problem? |
| 14:24.25 | nel | the echo goes away after 5 seconds |
| 14:24.43 | bjohnson | bagohell: break it down more into smaller pieces .. the handsets can be softphones, an analog phone with a fxs adapter, or a voip phone. Each option has pros and cons and each has a range of features that can be used) |
| 14:24.55 | Mimmus | nel: try to play with echotraining |
| 14:25.02 | konrads | How does capi and misdn relate even? |
| 14:25.06 | spackle | ne1: the joy of echo. |
| 14:25.09 | brad_mssw | running asterisk 1.0.9 ... have an issue with SIP transfers on my UTStarcom F1000, though it works perfectly with a Sipura 841. Basically, it seems that the UTStarcom attempts to establish a second connection to the desired extension to transfer to, when you tell it to complete the transfer, so the person that called ends up getting transferred to voicemail b/c the party the call is being transferred to is already on the line ... Bad Transfer (F1000 |
| 14:25.09 | brad_mssw | ) SIP Debug: http://pastebin.com/351000 Good Transfer (SPA841) SIP Debug: http://pastebin.com/350998 |
| 14:25.31 | brad_mssw | can anyone look at those dumps and tell me if it's asterisk or UTStarcom? |
| 14:25.39 | *** join/#asterisk Goshen (n=Goshen@67-40-107-29.slkc.qwest.net) |
| 14:25.46 | bjohnson | bagohell: on the pstn side you can have a digital T1, ISDN if in Europe, a fxo for connection to a pstn line, or even hardware that connects you to the cellular system |
| 14:25.54 | konrads | mISDN: INTERNAL ERROR in /root/asterisk-1.0.1/channels/chan_misdn-0.1.1/lib/mISDN/drivers/isdn/hardware/mISDN/stack.c:596 |
| 14:25.58 | konrads | this clearly isn't nice ? |
| 14:26.12 | *** join/#asterisk Fddayan (n=fddayan@66.240.80.130) |
| 14:26.13 | *** part/#asterisk FedericoAlves (n=sales@c-67-191-30-58.hsd1.fl.comcast.net) |
| 14:26.22 | bjohnson | bagohell: basically you can hook it to damn near anything you want and make a hybrid system that operates as one cohesive phone system |
| 14:26.36 | nel | Mimmus: is there anything else that can be set on echo training besides echotraining=yes ?? |
| 14:26.46 | nel | I'm using ztmonitor and I can see the echo all the time |
| 14:26.55 | Fddayan | somebody knows how to instal cepstral (a TTS) in asterisk ? |
| 14:27.17 | bjohnson | bagohell: they don't even have to be all at the same site. I haven't even mentioned all the options you have when you start looking at using the internet or other network |
| 14:27.57 | Mimmus | nel: you can use echotraining, echocancel and echocancelwhenbridged |
| 14:28.11 | nel | Mimmus: thanks, I'm using all those options |
| 14:28.37 | Mimmus | nel: echocancel=yes/no or echocancel=a power of two from 32 to 256 |
| 14:28.56 | *** join/#asterisk apardo (n=apardo@90.Red-83-46-197.pooles.rima-tde.net) |
| 14:29.13 | bjohnson | bagohell: cheapest startup system is a softphone and one of the free voip services .. then move to a prepaid system to test outgoing, then get a did to test incoming, then get a fxs or voip phone to get away from the softphone (that nobody likes as much as a handset), then get a fxo if you have a pstn line |
| 14:30.00 | Mimmus | nel: you seems to be in the case described in echotraining explanation: "In some cases, the echo canceller doesn't train quickly enough and there |
| 14:30.01 | Mimmus | ; is echo at the beginning of the call." |
| 14:31.36 | *** join/#asterisk comadreja (n=comadrej@4.Red-83-50-110.pooles.rima-tde.net) |
| 14:32.43 | *** join/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer) |
| 14:34.16 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 14:34.49 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
| 14:37.10 | RoyK | anyone here that knows if it's possible to block/disable a sip peer/user/friend without removing it entirely? |
| 14:37.44 | jontow | put them in a context that does nothing? |
| 14:38.18 | *** join/#asterisk SwK[Work] (n=SwK@border0hsv.asterisksgi.com) |
| 14:38.35 | *** join/#asterisk Nuxi (n=nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 14:38.59 | Nuxi | Any recommendations for 911 providers? |
| 14:39.40 | SwK[Work] | call 911 on your phone and ask them who they use |
| 14:39.44 | JerJer | 911 |
| 14:39.46 | *** join/#asterisk wzl (n=Miranda@82-169-62-42-mx.xdsl.tiscali.nl) |
| 14:39.57 | wzl | good afternoon 4all in europe! |
| 14:40.09 | SwK[Work] | WE SURRENDER |
| 14:40.12 | gordonjcp | hello wzl |
| 14:40.18 | SwK[Work] | oh i'm not in france.... |
| 14:40.20 | JerJer | don't beat me, masta |
| 14:40.40 | wzl | ive a question for radius icw asterisk ... is anyone familiar with radius? |
| 14:40.52 | JerJer | no |
| 14:40.59 | spackle | wzl: I can spell it. |
| 14:41.14 | Nuxi | I'm assuming somebody has support for 911 for Asterisk. |
| 14:41.31 | gordonjcp | why can't you just call out to 911? |
| 14:41.33 | jontow | wellll, to get the volume of a cylinder, one would use the radius.. (pi*(radius^2)) .. does that help? :) |
| 14:41.39 | JerJer | sure - put a TE411P and a PRI in asterisk and make it dial 911 |
| 14:41.47 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 14:42.07 | JerJer | if you do it repeatedly you will have a representative of your local 911 provider show up very quickly |
| 14:42.42 | MikeJ[Laptop] | I can do 911 no problem |
| 14:42.46 | MikeJ[Laptop] | right through my pri's |
| 14:42.47 | Nuxi | we need somebody that can do 911 across the country. |
| 14:43.01 | gordonjcp | who the hell has got time to find a landline anyway, if it's an emergency |
| 14:43.08 | gordonjcp | just use your mobile |
| 14:43.15 | Mimmus | JerJer: in Italy there is not a 911 analogous |
| 14:43.18 | Goshen | most people have cell phones strapped to their hip |
| 14:43.22 | SwK[Work] | nuxi: did you even try google? like "asterisk 911" or check the asterisk mailing lists? |
| 14:43.28 | jontow | goshen; or their head. |
| 14:43.38 | nel | I'm wondering if I use PRIs instead of analog lines, if I'm going to have the same problems with echo??? |
| 14:43.38 | *** join/#asterisk n4y (n=frodo7@host-ip237-209.crowley.pl) |
| 14:43.52 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc032.dialup.mindspring.com) |
| 14:44.03 | Nuxi | SwK[Work], yes, I just wanted recommendations. I can look at lot's of pretty pages, but recommendations are nice to get past the pretty pages. |
| 14:44.04 | spackle | ne1: I have problems with echo on PRI. |
| 14:44.09 | Mimmus | nel: theoretically no, but it seems that I have this! |
| 14:44.24 | nel | spackle: How come? I thought the problem was only with analog lines |
| 14:44.28 | nel | nooooooo :( |
| 14:44.46 | Nuxi | experienced users are better than salespersons. |
| 14:44.48 | spackle | ne1: it's wherever 2 line phones meet digital. |
| 14:44.55 | Mimmus | I tried to disable echocancel but can hear my voce very well! |
| 14:45.18 | spackle | er, one pair lines meet digital. |
| 14:45.25 | nel | so there is no way out of echo |
| 14:45.33 | RoyK | argh. why isn't there a native psql driver for realtime????? |
| 14:45.39 | nel | how does vonage work or the other companies, they don't have echo |
| 14:45.44 | Mimmus | echocancel, sw or hw (with newest Digium cards) |
| 14:45.45 | spackle | ne1: tuning, echo cancellers, time and experience. |
| 14:46.27 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-206.rockynet.com) |
| 14:46.47 | nel | spackle: how do you tune the cards? I have a tdm400 I was playing around with txgain rxgain, echocanceller=yes, echotraining=yes, everything that said echo in front yes :) the thing is that incoming calls get more echo than outgoing calls |
| 14:46.48 | spackle | ne1: sangoma is going to introduce an echo cancelling card. |
| 14:46.49 | *** join/#asterisk criptos (n=criptos@201.135.121.4) |
| 14:47.29 | nel | so the operator is complaining that every call that she responds she hears the annoying echo even if its for 2 3 seconds, sometimes is better sometimes worst on the same channel isn't that weird? |
| 14:47.36 | JerJer | spackle: big deal |
| 14:47.44 | JerJer | Digium already has one that kicks serious ass |
| 14:47.47 | coppice | ne1: all echo cancellers are not created equal, but some are more unequal than others |
| 14:47.47 | spackle | ne1: I'm not sure yet, still in the throes of it. |
| 14:47.58 | criptos | What channel bank is good to be used with * ? |
| 14:48.08 | spackle | JerJer, I hope so I'm going to get one in the next week (upgrade my existing) |
| 14:48.09 | JerJer | criptos: Adtran TA750 |
| 14:48.15 | coppice | JerJer: why are so many people complaining about it, then? |
| 14:48.15 | Mimmus | nel: echotraining=yes but echotraining=<a number> too can help (imho) |
| 14:48.30 | criptos | I will need to recive from 10 to 30 ptsn analog incomming lines, and NO, I cannot use a pri/e1, they´re analog... |
| 14:48.30 | nel | Mimmus: hmm gonna try that thanks |
| 14:49.00 | spackle | ciptos: or adtran TA850 |
| 14:49.00 | JerJer | criptos: you will save money using a digital circuit for any more than 11 lines |
| 14:49.00 | Mimmus | what's mean "kicks serious ass"? (I'm italian) |
| 14:49.33 | spackle | mimmus: works better than all the rest. (and beats a donkey in the butt) |
| 14:49.35 | JerJer | Mimmus: it eats lots more spaghetti than without echo |
| 14:49.38 | *** join/#asterisk sams2100 (n=sams@pcp424683pcs.naugus01.ga.comcast.net) |
| 14:49.49 | nel | is there any tools that help you tune the echo besides ztmonitor?? |
| 14:49.49 | JerJer | echo can |
| 14:49.51 | criptos | JerJer and aproximately pricing? for a new one? And jerjer, I said i cannot use a pri/e1/or any other digital signalling :( |
| 14:50.02 | *** join/#asterisk Chuji (i=Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) |
| 14:50.14 | JerJer | criptos: why? smack your telco around - they will save money as well |
| 14:50.22 | Mimmus | spackle, JerJer: well, this channel is a good place to learn 'true' english! |
| 14:50.30 | spackle | criptos: don't get a new one, the market is flooded with used ones, you can get two used ones for what you pay for a new one |
| 14:50.54 | coppice | why are there so many used ones around? |
| 14:51.00 | JerJer | and unlike many other companies, Adtran stands behind their hardware, no matter how you acquired it |
| 14:51.09 | JerJer | coppice: fire sales |
| 14:51.19 | Mimmus | also "beats a donkey in the butt" is a beautiful expression |
| 14:51.24 | spackle | coppice: telcom companies out of business (norvergence) |
| 14:51.35 | JerJer | Focal |
| 14:51.48 | JerJer | McLeod |
| 14:52.01 | konrads | yay! capi show says i have 2 B channels |
| 14:52.06 | spackle | JerJer: so you are happy with the new digium echo cancellers? |
| 14:52.09 | sams2100 | hey guys, can anyone try to explain the DBGet cmd to me? I guess I just dont understand how you can get something from a DB using only the table name (family) and the key (is key the column?) |
| 14:52.11 | konrads | how do i see if something reaches something? |
| 14:52.18 | konrads | e.g. if a call is there? |
| 14:52.29 | *** join/#asterisk tangel (i=tangel@64.135.81.8) |
| 14:52.36 | JerJer | konrads: place a call to one of the msn's on that circuit |
| 14:52.52 | criptos | what is the difference beween the adtran ta850 and ta750? |
| 14:52.55 | JerJer | spackle: it was a totally seemless upgrade here - yet processor usage dropped significantly |
| 14:53.00 | JerJer | an 8 |
| 14:53.06 | blitzrage | morning y'all |
| 14:53.07 | konrads | JerJer: and what output should i see on console? |
| 14:53.15 | JerJer | konrads: duno - don't use bri |
| 14:53.29 | criptos | spackle at ebay? |
| 14:53.36 | spackle | jerjer : did you have echo problems before the switch (my nophone calls always sound great to me - but I'm not on the analog end) |
| 14:53.37 | coppice | the most interesting things about sangoma's EC is the name of the chip maker - octasic. I've never seen a better demonstration that the world is running out of good new company names :-) |
| 14:53.50 | konrads | JerJer: and what do you use? |
| 14:53.52 | spackle | criptos: you bet. search for adtran 850 or 750 |
| 14:53.54 | criptos | JerJer: they are from diferrent providers, and no one is providing more than 5 lines... |
| 14:54.10 | criptos | whic is better spackle or what are the diferences? |
| 14:54.24 | JerJer | spackle: none - however now the host cpu is not doing echo can |
| 14:54.30 | JerJer | there is hardware assist for it |
| 14:54.38 | JerJer | which == less host cpu usage |
| 14:55.00 | JerJer | <PROTECTED> |
| 14:55.10 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 14:55.18 | JerJer | and that's with 87 calls right now |
| 14:55.22 | spackle | criptos: the 850 can be managed over net, which is handy. The 750 requires vt100 and is very "dumb" |
| 14:55.23 | JerJer | versus |
| 14:55.43 | JerJer | <PROTECTED> |
| 14:55.51 | spackle | jerjer, what are you using for echo can then? telllabs? |
| 14:55.52 | JerJer | with 38 calls - no hardware echo can |
| 14:56.00 | JerJer | spackle: TE411P |
| 14:56.29 | *** join/#asterisk brc_ (n=brc@pdpc/supporter/basic/brc) |
| 14:56.38 | *** join/#asterisk Darwin35 (n=kvirc@ip70-179-214-245.dl.dl.cox.net) |
| 14:57.13 | spackle | ciptos: yeah, remind me how to do that. |
| 14:57.50 | criptos | use /msg user to msg a specific user. |
| 14:58.14 | criptos | So, for what I see, there is a chasis, actually the cureent channel bank and modules.. wich can be fxo or fxs, rigth? |
| 14:58.17 | Darwin35 | but ask the user before msging them |
| 15:00.03 | dtwilson | I got call recipients complaining of poor quality/low volume sound when being called from our * box via an x101p - anything I should look at for tweaking this? |
| 15:00.08 | *** join/#asterisk asterisk_newbie (n=root@12-203-248-191.client.insightBB.com) |
| 15:00.12 | *** join/#asterisk supaigtr (n=yurplsl@152.53.16.10) |
| 15:00.17 | supaigtr | Sup ppl. |
| 15:00.43 | Darwin35 | the sun the moon you |
| 15:00.59 | asterisk_newbie | im a real newbie to asterisk could someone help me i just setup with someone to do my orig and term and i have to set a sip prefix in asterisk dialstream |
| 15:01.01 | bkw_ | blitzrage, yo |
| 15:01.19 | razu | can any1 tell me, what could cause this error >>>>> can any1 help m |
| 15:01.21 | razu | sry |
| 15:01.40 | razu | this error >>>> ug 31 17:51:47 NOTICE[2385]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/vicitest/3 of format slin since our native format has changed to ulaw |
| 15:01.45 | bkw_ | that is not an error |
| 15:01.47 | bkw_ | its an NOTICE |
| 15:01.52 | razu | ok |
| 15:02.05 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 15:02.13 | bkw_ | its dropping the sln frame because the format has changed to ulaw |
| 15:02.23 | bkw_ | I suspect the queue had an slin frame still in it when the change took place |
| 15:02.34 | JerJer | how is slin getting queued in the first place? |
| 15:02.39 | razu | but i'm having a problem with vicidial ... it drops IP phone calls, but doesnd drop cellular calls ... |
| 15:02.50 | asterisk_newbie | im a real newbie to asterisk could someone help me i just setup with someone to do my orig and term and i have to set a sip prefix in asterisk dialstream |
| 15:03.09 | bkw_ | well we transcode via sln |
| 15:03.10 | bkw_ | he's using stable |
| 15:03.10 | bkw_ | I can tell |
| 15:03.24 | bkw_ | that might be his whole problem :P |
| 15:03.41 | razu | that i'm using stable asterisk ? |
| 15:03.54 | bkw_ | ya |
| 15:04.06 | razu | hmm |
| 15:04.36 | razu | so what version should i use ? |
| 15:04.39 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 15:05.26 | razu | i'm havin 1.0.7 in my box right now |
| 15:06.38 | bkw_ | well stable is at 1.0.9 |
| 15:07.47 | JerJer | razu: not -stable |
| 15:08.01 | razu | hmm |
| 15:08.02 | JerJer | perhaps the new 1.2 rc1 that is out there |
| 15:08.55 | razu | well ... i'll try |
| 15:10.43 | supaigtr | Anyone have troubleshooting tips for sudden lost call in one direction? CVS HEAD? |
| 15:10.56 | tzanger | how do yo ulose a call in only one direction |
| 15:11.09 | supaigtr | No audio in one direction. |
| 15:11.24 | supaigtr | They can hear you but you can't hear them. |
| 15:11.35 | JerJer | iax? |
| 15:11.48 | JerJer | firewall? |
| 15:11.50 | JerJer | nat? |
| 15:11.50 | supaigtr | Seems like its both. |
| 15:11.52 | wzl | me3! |
| 15:12.11 | supaigtr | Well I have phones doing it connected direct to the box and over firewall. |
| 15:12.27 | supaigtr | chan_sip.c:1311 __sip_ack: Stopping retransmission . I see alot of these. |
| 15:12.34 | supaigtr | Is that normal? |
| 15:12.50 | *** join/#asterisk Lee__ (n=lee@ip67-153-86-42.z86-153-67.customer.algx.net) |
| 15:13.30 | lathos42 | supaigtr: I had a problem with my Polycom IP-501 talking to my Sipura SPA-3000 with one way audio.. turning off reinvites for the Sipura took care of the problem |
| 15:14.01 | *** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg) |
| 15:14.10 | supaigtr | Hmm. Didn't they just change some invite stuff. Wasn't haveing the problem until a day or so ago with head. |
| 15:14.25 | Lee__ | is echo over a SIP trunk impossible? It would have to come from a PSTN gateway, right? |
| 15:14.48 | supaigtr | Anyone know what this is? chan_sip.c:1311 __sip_ack: Stopping retransmission |
| 15:17.20 | supaigtr | Anyone have procedures to get out of head and to stable? I've had problems with spandsp etc when moving to head and back. Is there something I need to remove before compile install of stable code? |
| 15:17.23 | oej | ~seen twisted |
| 15:17.26 | jbot | twisted is currently on #asterisk |
| 15:18.06 | brad_mssw | when does SetGroup() decrement it's count ?? |
| 15:18.38 | brad_mssw | it just messed up on one extension for some reason. .. |
| 15:18.39 | criptos | when the call leaves the extension which setgroup is being set |
| 15:18.42 | Umaro | any sangoma guys here? |
| 15:19.12 | brad_mssw | hmm, it usually works, but it got 'locked' on one ext for some reason ... |
| 15:19.24 | JerJer | Umaro: no, buy DIgium |
| 15:19.35 | *** join/#asterisk DrWho (n=MIKE@mike-new.tc3net.com) |
| 15:19.58 | brad_mssw | CheckGroup(1) kept returning saying the line was occupied, but it was not |
| 15:20.17 | *** join/#asterisk azrishahril (n=azrishah@60.50.199.102) |
| 15:20.35 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 15:23.09 | Lee__ | supaigtr: yes, uninstall HEAD, remove everything in /usr/lib/asterisk/modules and install stable. |
| 15:23.39 | tzanger | Lee__: eww |
| 15:23.43 | Lee__ | you have a better idea? |
| 15:23.52 | *** join/#asterisk clyrrad (n=ddd@CPE000c4120d329-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 15:23.54 | tzanger | yes, stay with head, it's fine |
| 15:24.12 | tzanger | but ofr his specific problem either back up head to when it was working or yes, go to stable |
| 15:24.18 | Lee__ | I was responding to supaigtr's question, which was to /not/ stay with head but move to stable. |
| 15:24.19 | tzanger | but I think he'll be into a lot of other fun stuff |
| 15:24.30 | tzanger | Lee__: I know |
| 15:25.10 | Lee__ | so do you have a better idea for how to downgrade? cause that way worked well for me when I needed to do it. |
| 15:25.25 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe201.t-net.net.ve) |
| 15:25.37 | coppice | move to stable when there's no room at the inn |
| 15:25.38 | tzanger | cvs -D '1 day ago' up asterisk? |
| 15:25.42 | tzanger | anyway |
| 15:25.43 | tzanger | gotta go |
| 15:26.19 | tuxinator_linuxM | Should we start findmanxpower.com ?, hope he is okay |
| 15:26.52 | tuxinator_linuxM | Been two and half days since he has been on |
| 15:27.10 | *** join/#asterisk djspeck (n=root@ewersbach.net) |
| 15:27.23 | Hmmhesays | what would you guys recommend I use for a network file system |
| 15:27.24 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
| 15:27.34 | mutilator | is there any way to set the hook flash shorter? |
| 15:27.36 | Hmmhesays | I need to pull a single config to about 6 asterisk boxes |
| 15:27.36 | Lee__ | Hmmhesays: NFS |
| 15:27.54 | tuxinator_linuxM | Hmmhesays: Not doing realtime? |
| 15:27.57 | spackle | rsync |
| 15:28.02 | Hmmhesays | no realtime |
| 15:28.12 | Lee__ | yeah, rsync if you don't need random access. |
| 15:28.14 | Hmmhesays | using asterisk for the dialplan only |
| 15:28.19 | *** join/#asterisk Derkommissar (n=alberto@fl-southhub-u1-c6-0a-30.miamfl.adelphia.net) |
| 15:28.26 | JerJer | Hmmhesays: tftp :P |
| 15:28.29 | Derkommissar | anyone here ever worked with quimtums ? |
| 15:28.30 | Lee__ | Asterisk can share dialplans. |
| 15:28.31 | Hmmhesays | SER will be taking care of user registrations |
| 15:28.32 | nel | has anybody player around with the new asterisk1.2 beta? |
| 15:28.33 | tuxinator_linuxM | floppy transfer, just kidding |
| 15:28.35 | nel | played |
| 15:29.15 | *** join/#asterisk djspeck (n=root@ewersbach.net) |
| 15:29.15 | Lee__ | Hmmhesays: I believe it's the switch => option in extensions.conf |
| 15:29.17 | tuxinator_linuxM | Derkommissar: Qumtums? Are they little creatures? |
| 15:29.41 | djspeck | hello guys, i have a problem with loading my hfc_pci card with mISDN: |
| 15:29.41 | Derkommissar | ;) |
| 15:29.42 | JerJer | Lee__: what about sip users and peers? |
| 15:29.44 | Derkommissar | yea |
| 15:29.53 | djspeck | when i try to load an error occours: |
| 15:29.57 | Lee__ | JerJer: not sure. |
| 15:30.00 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 15:30.01 | drbrown | is there any way to detect whether a hint is active or not???? This would be for the purpose of call redirection |
| 15:30.05 | djspeck | mISDN: Could not register |
| 15:30.25 | *** join/#asterisk Ganlron (n=Ganlron@omega.csolve.net) |
| 15:30.31 | djspeck | (mISND: Could not register devices) |
| 15:31.13 | djspeck | WARNING: error inserting mISDN_core... ... Input/Output error |
| 15:31.37 | djspeck | can someone help...?? |
| 15:32.23 | nel | do you think using asterisk 1.2 and the new zaptel drivers will help me on my quest against echo??:) |
| 15:32.48 | Lee__ | nel: probably not. though reading about the cause of echo might. |
| 15:33.00 | Lee__ | check voip-info.org |
| 15:33.24 | nel | Lee_: I have read and tried everything I was wondering if there are any new tuning options |
| 15:33.47 | coppice | the voip-info.org stuff on echo isn't really very enlightening |
| 15:33.47 | *** join/#asterisk hellagony (n=egutierr@200.121.129.178) |
| 15:34.24 | djspeck | does someone know something about loading hfc-cards with misdn |
| 15:34.26 | *** join/#asterisk PakiPenguin (n=uppal@unaffiliated/pakipenguin) |
| 15:34.29 | djspeck | ?? |
| 15:35.41 | coppice | most people haven't a clue what that page really means |
| 15:36.16 | Hmmhesays | most people don't have a clue period |
| 15:36.33 | Hmmhesays | in reference to the general world population |
| 15:37.30 | *** join/#asterisk Deep6 (n=DEEP6@208.38.35.162) |
| 15:37.36 | Deep6 | anyone here? |
| 15:37.43 | crash3m | no, go away :P |
| 15:37.49 | Deep6 | hhaha fantatstic |
| 15:38.06 | djspeck | is there somehone who can help me with my hfc-card, i tried the whole day |
| 15:38.07 | Lee__ | coppice: perhaps those people should get that clue by studying it. |
| 15:38.31 | *** join/#asterisk aminorex (n=tony@12-23-137-226.dhcp.dlth.mn.charter.com) |
| 15:38.41 | Deep6 | anyone knowledgeable on IVR stuff within asterisk? |
| 15:38.41 | Lee__ | not having a clue is a pretty lame excuse for not getting one. |
| 15:39.06 | coppice | Lee___ most people will not acquire a clue from that page |
| 15:39.08 | Lee__ | Deep6: there's an example IVR menu in extensions.conf |
| 15:39.18 | *** join/#asterisk Henguei (n=smoka@196.203.249.73) |
| 15:39.26 | jeffik | Deep6: what do you need? |
| 15:39.32 | Deep6 | Lee__, is it possible to take user input for a scenario like say Internet banking type applications |
| 15:39.32 | Wonka | djspeck: modprobe hfcpci protocol=0x12 layermask=0x3 debug=0x0 |
| 15:39.40 | Wonka | djspeck: and then use chan_misdn |
| 15:39.48 | Deep6 | jeffik, I'm wondering if I can do time cards via IVR in asterisk |
| 15:39.52 | coppice | in fact I don't think anyone will. it only makes sense if you understand the subject before you read it |
| 15:39.57 | Lee__ | coppice: oh well. it was very illustrative to me. I guess having a clue is relative. |
| 15:40.29 | Deep6 | ie what's your employee # user enters 4123, hours worked for Monday of this week 7, Tuesday of this week etc |
| 15:40.37 | Lee__ | Deep6: yes, but you will have to write your own AGI application to process your time card data. There's AGI interfaces for Perl, PHP and C. Perhaps others I don't know of. |
| 15:40.50 | Deep6 | or 7 # 5 for 7.5 for instance |
| 15:40.53 | djspeck | Wonka: same error as before |
| 15:40.54 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:41.13 | coppice | Lee__ it certainly is relative. lots of people who come here seem to muddle up the terms read and understand |
| 15:41.14 | Wonka | djspeck: can't help you, then, it's too long ago... |
| 15:41.18 | djspeck | Wonka: mISDN: Could not register devices |
| 15:41.19 | Lee__ | coppice: so you're saying that you would like to understand how to eliminate echo without first understanding the subject? |
| 15:41.39 | Deep6 | Lee__, this is sort of a brain wave of mine, I'm really not that knowledgeable on asterisk at all so it's greek to me...but I've been looking at a something that I can try to get some time playing with asterisk for |
| 15:41.58 | spackle | Lee__ I'd like to drive a car without understanding how a road is made or the internal combusion engine |
| 15:42.05 | Wonka | ~nopaste |
| 15:42.06 | jbot | it has been said that nopaste is http://nopaste.snit.ch/ or http://pastebin.com/ |
| 15:42.17 | Lee__ | Deep6: you will have to write a custom application. If you can't do it you should find someone who can. |
| 15:42.18 | Henguei | hello i have always the same probleme i can't listen to the sounds of asterisk even though i install sox and vorbis |
| 15:42.27 | Henguei | what should i do |
| 15:42.29 | Lee__ | spackle: you and I are different. |
| 15:42.33 | coppice | I'm saying that page wasn't written for the average person to get a solid enough grip on the subject to solve their problems. |
| 15:42.43 | Hmmhesays | anyone using g726 with asterisk? |
| 15:42.45 | Delvar | hay, if i wer to plug a non poe device into a poe switch what would happen? |
| 15:42.47 | Deep6 | Lee__, I likely can....I just need to know it's possible from the get go to begin investigating |
| 15:42.51 | Lee__ | spackle: although I'd expect that if you designed your own car you might want to know how an engine works first. |
| 15:43.02 | spackle | lee__ on the otherhand we both use *nix and do care about the source of echo. |
| 15:43.08 | djspeck | modeprobe hfcpci doesn't work, anyone an idea?? |
| 15:43.12 | Deep6 | if you say it's possible then I can find my path to get there |
| 15:43.12 | Lee__ | Deep6: yes, you can parse any data from an external application and pass it to Asterisk via AGI scripts. |
| 15:43.29 | coppice | i dunno who wrote it, but it reads like it was written by someone very pissed off |
| 15:43.44 | Lee__ | wiki attack! edit that shit! |
| 15:44.04 | *** join/#asterisk blitzmere (i=blitzmer@CPE000129fe5661-CM000f212f9067.cpe.net.cable.rogers.com) |
| 15:44.09 | Deep6 | well forgive my ignorance, but I want to essentially use asterisk as the capturing mechanism and I want to export data entered into it into a db |
| 15:44.12 | spackle | lee__ car I design would be electric ;-) (or run on alchohol) |
| 15:44.25 | Henguei | any help |
| 15:44.41 | Deep6 | or is AGI how you make the prompting and define the bounds checking? |
| 15:44.54 | Umaro | anyone here know how to work with fftw? |
| 15:44.58 | blitzmere | Can anyonw give me direction on where I could find a good supplier for pri's |
| 15:45.03 | coppice | spackle: will the driver run on alcohol too? |
| 15:45.05 | djspeck | there must be someone who can help me with loading my hfc driver..... |
| 15:45.22 | spackle | oh yeah. |
| 15:45.23 | Lee__ | Deep6: you have to write some kind of script to capture your data and put it somewhere, be that from a phone to a database or from a application to an IVR menu. |
| 15:45.27 | wzl | exit |
| 15:45.38 | wzl | oeps wrong window :-S |
| 15:45.38 | Lee__ | AGI is the glue between Asterisk and other programs. |
| 15:45.46 | Deep6 | Lee__, ok, do you have any recommended reading? |
| 15:45.50 | Deep6 | ok |
| 15:45.56 | Lee__ | Deep6: you know Perl or PHP? |
| 15:46.02 | Deep6 | python and php |
| 15:46.06 | Deep6 | my perl is weak |
| 15:46.12 | Deep6 | I made the change to php |
| 15:46.16 | Deep6 | er python rather |
| 15:46.49 | Lee__ | excellent. I don't know if there's a python AGI interface but probably is. Check the wiki or Google for the core PHP AGI scripts. |
| 15:46.54 | blitzrage | I prefer PHP |
| 15:46.57 | *** join/#asterisk Assid (n=assid@203.115.64.59) |
| 15:47.12 | blitzrage | someone was also working on res_php too |
| 15:47.14 | blitzrage | but not used it yet |
| 15:47.19 | cypromis | I prefer sleep |
| 15:47.30 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 15:47.38 | Assid | heya |
| 15:47.45 | cypromis | but python at least fullfills the readability requirements for reusable code |
| 15:47.46 | djspeck | someone kind with the error: mISDN: could not register devices?? |
| 15:47.46 | cypromis | hehe |
| 15:48.14 | blitzrage | I think python codes looks extremely ugly |
| 15:48.34 | bkw_ | blitzrage, I agree... but some would shoot ya for saying that |
| 15:48.36 | drray | I think there are just people who like perl, and then people who don't |
| 15:48.43 | nel | I'm trying fxotune |
| 15:48.45 | bkw_ | I LUV perl |
| 15:48.50 | nel | has anybody had luck with that ?? |
| 15:48.53 | spackle | blitzrage: it's not ugly, it's just designed that way. |
| 15:48.57 | drray | I've listened to otherwise rational people just rip into perl |
| 15:49.05 | drray | I <3 perl |
| 15:49.14 | bkw_ | drray, I hear ya... Perl is nice |
| 15:49.18 | blitzrage | I've used Perl... didn't *love* it... but its good for scripting |
| 15:49.19 | file | I feel like I went out drinking lastnight |
| 15:49.20 | bkw_ | file dear where were you today? |
| 15:49.23 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 15:49.23 | file | but I'm pretty sure I didn't |
| 15:49.28 | Assid | this is weird as hell.. these guys in the office are complaining of bad quality.. but when i call in/out.. it seems to be fine |
| 15:49.33 | Hmmhesays | yeah were is file, i have another n00b question about ser |
| 15:49.50 | file | bkw_: I drifted in/out of consciousness since 8AM my time |
| 15:50.03 | file | I should be back to my normal self come tomorrow, unless I'm sick |
| 15:50.05 | drray | I've been playing with ncurses/perl all night |
| 15:50.07 | blitzrage | bkw_: yah well, people can find lots of reasons to shoot me - I stand by my statement of Python being really ugly :) |
| 15:50.11 | Deep6 | python is readable |
| 15:50.15 | file | Hmmhesays: ask |
| 15:50.17 | file | blitzmere: !!! |
| 15:50.21 | Deep6 | perl you need a rosetta stone to decipher |
| 15:50.29 | cypromis | paint it blue |
| 15:50.38 | bkw_ | perl is easy to understand |
| 15:50.45 | coppice | Hey, I live in the perl delta :-) |
| 15:50.45 | bkw_ | but it depends on who wrote it |
| 15:50.48 | cypromis | I wasn't talking about your houses interior design but about readability of code |
| 15:50.49 | blitzrage | holy war! |
| 15:50.54 | Delvar | hmm how would i get a bug on bugs.digium.com re opend? |
| 15:50.57 | bkw_ | hahahaha |
| 15:50.59 | Hmmhesays | say I have a dual proc xeon box, 3ghz or so. will ser have any problems with handling 4k registrations? |
| 15:51.03 | bkw_ | Delvar, bug numbah? |
| 15:51.09 | Delvar | 2863 |
| 15:51.15 | Hmmhesays | i'm trying to do some network layout here |
| 15:51.22 | drray | $1,$2,$3,$4, I declare a holy war |
| 15:51.28 | blitzrage | Hmmhesays: you'll find out once you try it :) Registrations don't require much power though |
| 15:51.35 | file | Hmmhesays: against what, db? |
| 15:51.41 | Hmmhesays | file yeah |
| 15:51.44 | bkw_ | Delvar, done |
| 15:51.46 | bkw_ | NEXT!!! |
| 15:51.47 | cypromis | Hmmhesays: with 8k of ram ? |
| 15:51.48 | Delvar | thnaks! |
| 15:51.53 | Delvar | :x |
| 15:51.56 | Hmmhesays | I can't just 'try it' I need to have an idea if its going to work or not |
| 15:51.59 | file | Hmmhesays: I think you'll be fine |
| 15:52.18 | file | 4k at the same time? or spread out? |
| 15:52.33 | Hmmhesays | well 4k re-registering every 60 seconds or so |
| 15:52.34 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 15:52.42 | cypromis | all at the same time ? |
| 15:52.48 | Deep6 | Lee__, this sound accurate? |
| 15:52.50 | file | yeah, all at the same time is the key question |
| 15:52.50 | Deep6 | Python works fine with AGI; the only proviso is that after writing a message to Asterisk (via stdout) or to the Asterisk console (via stderr) you have to flush the corresponding file. Failing to flush stdout will cause the script to come to a grinding halt while Python waits for the buffer to fill and Asterisk waits for a command. A minimal AGI script in Python looks like this: |
| 15:52.54 | spackle | don't call him file, you call him Mister Ser! |
| 15:52.57 | bkw_ | asterisk can't do that |
| 15:52.59 | bkw_ | it would blow up |
| 15:53.00 | bkw_ | haha |
| 15:53.03 | *** join/#asterisk axscode (n=Netgeker@203.131.153.34) |
| 15:53.11 | cypromis | mist the ser |
| 15:53.19 | bkw_ | I know dear |
| 15:53.21 | bkw_ | I know he's talking SER |
| 15:53.27 | axscode | guyz.. anyoen happens to know any SDK or any sourcecode for SIP Phone... just simple call and rcv call.. thats all.. |
| 15:53.38 | Lee__ | Deep6: yeah. I think that's an issue with PHP too. But it's something you write once and just include in all your scripts. |
| 15:53.40 | Hmmhesays | well pretty close to the same time |
| 15:53.43 | *** join/#asterisk generalhan (i=general_@63.133.146.82) |
| 15:53.57 | *** join/#asterisk trimi` (i=Pharrel@62.162.242.16) |
| 15:54.19 | cypromis | axscode: try OPAL or resiprocate |
| 15:54.19 | djspeck | HELP: driver does not load (HFC) :: mISDN: Could not register devices - error inserting mISDN_core: Input Output error |
| 15:54.24 | cypromis | and google is your friend |
| 15:54.25 | *** join/#asterisk sig-couch (n=sigterm@cpe-66-91-126-101.hawaii.res.rr.com) |
| 15:54.25 | file | Hmmhesays: I think you'll be fine |
| 15:54.27 | axscode | thankx cypromis.. |
| 15:54.40 | axscode | i already tried google.. but no good... |
| 15:54.44 | file | Hmmhesays: what db? |
| 15:54.54 | Hmmhesays | file: haven't decided yet |
| 15:55.11 | Hmmhesays | can you do registration against a flat text file? |
| 15:55.34 | spackle | holy war, up next, posgres v. mysql |
| 15:55.39 | file | you can do a combination memory+database |
| 15:55.42 | blitzrage | postgres > mysql |
| 15:56.16 | Hmmhesays | if I could get away without a database that would be even better |
| 15:56.24 | Pj | yeah but postgres = slug |
| 15:56.44 | Pj | I think googlefs is the ultimate solution. |
| 15:56.49 | file | Hmmhesays: well you can do it as a combination like I said, so it keeps it in memory and writes periodically... |
| 15:56.52 | cypromis | 42 |
| 15:57.04 | file | so it's not a huge load on the db |
| 15:57.15 | blitzrage | bah - 42 is never the answer |
| 15:57.39 | Hmmhesays | danke file, one day I will not have so many n00b questions about ser |
| 15:57.53 | file | Hmmhesays: :) |
| 15:57.55 | *** join/#asterisk Connor- (n=billy@198-144-174-5.knx.tn.nxs.net) |
| 15:57.55 | Pj | well, except if the question is what's the decimal ascii value of '*' |
| 15:58.00 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 15:58.10 | blitzrage | Hmmhesays: guarenteed I've asked file more questions than you :) |
| 15:58.11 | Lee__ | djspeck: perhaps you should do some more debugging before repeating yourself so much. |
| 15:58.33 | file | blitzrage owes me many beers |
| 15:58.34 | Lee__ | Hmmhesays: there's also a #ser channel on freenode |
| 15:58.40 | *** part/#asterisk Malthus (n=dumbdumb@port0091-abr-s-adsl.cwjamaica.com) |
| 15:58.58 | coppice | or what do you get if you multiply 6 by 9 in base 13 |
| 15:59.23 | Hmmhesays | with 12 people in it |
| 15:59.37 | Nuxi | 40 |
| 15:59.53 | blitzmere | file, i am looking for a good company that can offer me inbound access to my asterisk |
| 16:00.08 | Hmmhesays | there are many out there |
| 16:00.11 | MikeJ[Laptop] | coppice, 42 |
| 16:00.20 | MikeJ[Laptop] | why, 42 is the answer |
| 16:00.35 | MikeJ[Laptop] | blitzrage, it appears you are wrong |
| 16:00.51 | Nuxi | 40 |
| 16:00.56 | Pj | coppice: nobody will ever understand us :' |
| 16:00.59 | MikeJ[Laptop] | Nuxi, try again |
| 16:01.14 | MikeJ[Laptop] | 52 base 10 is 40 base 13 |
| 16:01.22 | Hmmhesays | now is asterisk going to explode if I send a call from to ser, and ser sends it back to asterisk? |
| 16:01.27 | Nuxi | 6 * 9 = 54 in base 10 |
| 16:01.43 | MikeJ[Laptop] | yes |
| 16:01.45 | MikeJ[Laptop] | correct |
| 16:01.47 | Nuxi | hmmm pesky math tables. |
| 16:01.49 | MikeJ[Laptop] | heh |
| 16:02.04 | blitzrage | MikeJ[Laptop]: I'm never wrong (except for when I am) |
| 16:02.09 | Nuxi | base conversion is much simpler than arithmetic. |
| 16:02.09 | file | Hmmhesays: yes. |
| 16:02.18 | file | Hmmhesays: same callid, it'll FrEaK oUt |
| 16:02.53 | coppice | the first person who pointed out to me that 6 x 9 = 42 in base 13 was a little girl about 8. I never figured out if she actually understood what she was saying, or if she just read it. |
| 16:03.00 | Hmmhesays | so I have to define did's in asterisk that need to go out the same T1 as it came in on |
| 16:03.24 | file | cypromis: poke |
| 16:03.38 | Hmmhesays | and not define the route in ser |
| 16:03.41 | *** join/#asterisk meppl (n=mephisto@p54AAF414.dip.t-dialin.net) |
| 16:04.05 | bkw_ | I need to find Andy |
| 16:04.14 | bkw_ | cypromis, where is ScardyCat |
| 16:04.16 | Pj | Nuxi: you must suck at arithmetics then... ;) |
| 16:04.58 | cypromis | in the netherlands |
| 16:06.34 | *** join/#asterisk piruchoOSXXX (n=Mauricio@adsl-69-233-188-132.dsl.pltn13.pacbell.net) |
| 16:07.38 | *** join/#asterisk Defraz (n=t0tal@65.124.150.49) |
| 16:08.55 | lehel | j |
| 16:09.04 | generalhan | anyone know a good site to get some .pcm or .raw ringtones ? |
| 16:09.55 | axscode | is there resiprocate for windows? |
| 16:10.28 | Assid | okay can someone help me with a nufone connection |
| 16:10.31 | JerJer | dd count=8192 if=/dev/urandom of=screech.pcm |
| 16:10.36 | Assid | people complaining of "choppy" reception |
| 16:10.42 | Assid | JerJer!!! my man! |
| 16:10.44 | JerJer | Assid: traceroute |
| 16:10.46 | JerJer | ping |
| 16:11.55 | Godsey | how long before microsoft competes head to head w/ vonage? :) |
| 16:12.05 | JerJer | 20 months |
| 16:12.29 | JerJer | who wants to buy a VoIP provider? |
| 16:12.37 | Godsey | ms just did? |
| 16:12.59 | JerJer | who wants to buy another VoIP provider? |
| 16:13.01 | *** join/#asterisk piruchoOSXXX (n=Mauricio@adsl-69-233-188-132.dsl.pltn13.pacbell.net) |
| 16:13.09 | JerJer | i'd take borgsoft money |
| 16:13.10 | Lee__ | sure $50. |
| 16:13.16 | Godsey | :) |
| 16:13.28 | Lee__ | who'd m$ buy? |
| 16:13.29 | JerJer | Lee__: perhaps with a g behind it |
| 16:13.33 | JerJer | teleo |
| 16:13.44 | Godsey | I don't see how much they paid tho |
| 16:14.06 | spackle | teleo run on Windows? |
| 16:14.22 | Godsey | it's going to be part of messenger |
| 16:14.30 | *** join/#asterisk jedirl (n=g844988t@154.Red-217-127-168.pooles.rima-tde.net) |
| 16:14.32 | jedirl | Hello |
| 16:14.37 | *** join/#asterisk fugitivo (n=ajf@201.255.102.212) |
| 16:14.38 | fugitivo | hello |
| 16:14.43 | coppice | MS doesn't have that good a track record of success in these things, but this one might be enough like a browser for the same strategy to win |
| 16:14.52 | criptos | <PROTECTED> |
| 16:15.09 | *** join/#asterisk drumkilla_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:15.09 | *** mode/#asterisk [+o drumkilla_] by ChanServ |
| 16:15.14 | criptos | And as more markets you arrive, more dificult to know where revenew is.. |
| 16:15.28 | file | hi drumkilla |
| 16:15.28 | *** join/#asterisk riksta (n=rick@213.121.151.210) |
| 16:15.31 | Godsey | criptos: sometimes |
| 16:15.33 | jedirl | I'm having problems with outgoing calls with asterisk |
| 16:15.40 | Godsey | ms may become just another index fund :) |
| 16:15.50 | fugitivo | jedirl: what kind of problems? |
| 16:15.54 | jedirl | the call is placed but it doesn't connect to the extension I want it to connect, or at least it doesn't do what the extension says it should do |
| 16:15.54 | drumkilla_ | file!!! |
| 16:16.01 | fugitivo | the times of MS are over |
| 16:16.14 | Godsey | fugitivo: maybe, maybe not. |
| 16:16.17 | Godsey | they have a shit load of money |
| 16:16.29 | criptos | Godsay: maybe.. but again, their shit load of money is finite.. |
| 16:16.29 | Godsey | so if they get something that works, there not out |
| 16:16.32 | fugitivo | today, money is nothing |
| 16:16.38 | jedirl | any hint? |
| 16:16.48 | coppice | don't be so sure. they have a lot of resources to keep them going however many times they screw up new things |
| 16:17.02 | dtwilson | what's a decent quality - but inexpensive SIP handset? Have finally had enough of these crappy grandstreams here |
| 16:17.31 | Godsey | I have pap2-na @ home |
| 16:17.37 | Godsey | I wish I'd gone w/ a normal sip phone now :) |
| 16:17.42 | JerJer | Microsoft is going to do nothing more than to further solidify the future of VoIP |
| 16:18.06 | coppice | anyone know what teleo did? |
| 16:18.07 | spackle | dtwilson: polycom 300/1 maybe sipura 841 |
| 16:18.15 | dtwilson | spackle thanks |
| 16:18.18 | *** join/#asterisk roulduke (i=xlbecvv0@p508D1CAD.dip0.t-ipconnect.de) |
| 16:18.18 | fugitivo | Godsey: why? i like the pap2-na |
| 16:18.44 | Godsey | you don't get nifty transfer buttons :) |
| 16:18.49 | coppice | JerJer: VoIP doesn't need solidifying. big businesses have already given up using anything else for new installations |
| 16:19.40 | JerJer | then why does the PSTN still exist today? |
| 16:19.55 | criptos | that's rigth, what is needed is a massification of devices, and a convergence of techology, cellullars & voip phones.. at the same device... |
| 16:20.24 | coppice | I said new installations |
| 16:20.33 | fugitivo | JerJer: because things doesn |
| 16:20.33 | dtwilson | criptos: tis already happening |
| 16:20.36 | Godsey | JerJer: well if $$ wasn't an issue, it probably wouldn't :) |
| 16:20.39 | fugitivo | JerJer: because things doesn't day one day to another |
| 16:20.44 | Godsey | fiber or ethernet to every home and be done w/ it |
| 16:20.51 | *** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
| 16:20.57 | Godsey | I'll have verizon fiber next month :) |
| 16:21.07 | coppice | FTTH is still expensive to do |
| 16:21.12 | JerJer | coppice: we just did a new school district |
| 16:21.20 | JerJer | they had PRIs brought in |
| 16:21.23 | pauldy | I've got a nice big cable sittig less that 20 feet form me andthey can tell me when we will have service |
| 16:21.33 | JerJer | plus fiber will be here before winter |
| 16:22.04 | pauldy | cant tell me |
| 16:22.09 | Godsey | coppice: I don't know what verizon's fios service costs them to bring to the house, but w/ no contract and under $40/mo I don't care :) |
| 16:22.09 | coppice | JerJer: schools are so out of date, I'm surprised they even used ISDN :-) |
| 16:22.16 | Godsey | 30meg down/10meg up |
| 16:22.22 | Hmmhesays | pauldy: get a railroad spike and jam it through there |
| 16:22.32 | Hmmhesays | tell them you are trying to extend their service area |
| 16:22.36 | JerJer | local government funded the project |
| 16:22.39 | DrWho17 | coppice: not here, we serve a district of 54 schools all interconnected via fiber |
| 16:22.45 | JerJer | perhaps some county funds too - don't really know |
| 16:22.56 | DrWho17 | their own *fiber* they had run |
| 16:23.03 | JerJer | eek |
| 16:23.05 | Godsey | the county I'm in now owns a huge fiber network |
| 16:23.15 | Godsey | all city/county/state buildings are on it |
| 16:23.15 | pauldy | yea that wouldn't cause me any pain |
| 16:23.24 | *** join/#asterisk Exstatic (i=exstatic@static-71-116-196-11.lsanca.dsl-w.verizon.net) |
| 16:23.25 | Godsey | but I'm moving next month |
| 16:23.36 | JerJer | DrWho17: we did wireless between schools - then fiber/telco to central location |
| 16:23.45 | coppice | I wenr in a shiny new university that has blackboards last week. that was like a step back in time |
| 16:24.08 | drumkilla_ | my school has blackboards ... |
| 16:24.11 | drumkilla_ | :) |
| 16:24.12 | DrWho17 | yea, I'm not a wireless fan, we just provided them bandwidth |
| 16:24.14 | spackle | coppice, you mean black as in chalk? |
| 16:24.29 | drumkilla_ | chalkboards, indeed |
| 16:24.30 | Har|ock | crazy |
| 16:24.35 | drumkilla_ | and professors still use overhead projectors |
| 16:24.38 | coppice | yes, black as in not whiteboards |
| 16:24.52 | spackle | somebody did the ROI on chalk v whiteboard pens |
| 16:25.11 | Har|ock | so many ppl are allergic to chalk dusk though these days |
| 16:25.17 | DrWho17 | whiteboard pens require more skilled staff |
| 16:25.37 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe201.t-net.net.ve) |
| 16:25.39 | spackle | kids huffing whiteboard pens |
| 16:25.58 | drumkilla_ | file: you wish |
| 16:26.00 | spackle | snorting optic fiber |
| 16:26.06 | *** join/#asterisk point (i=1000@213.27.44.55) |
| 16:26.12 | file | drumkilla_: pfft |
| 16:26.18 | file | drumkilla_: don't flatter thyself! |
| 16:26.48 | *** join/#asterisk usam (n=alx@203.156.39.9) |
| 16:26.54 | coppice | I think snorting fibre might cause severe bleeding |
| 16:26.56 | epablo | How do I return a certain disconnect code when I hang up a OH323 call. For example I wan't to return 1 when someone call an unasiggned number.. |
| 16:28.25 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:28.27 | usam | hello, .. any know problem with AMD CPU with asterisk? just upgraded my system AMD and using a AMD optimzed kernel .. |
| 16:28.45 | drray | my amd server runs fine with asterisk |
| 16:28.52 | harryvv | usam, ran asterisk on my dual opteron cpu motherboard with one cpu |
| 16:28.58 | coppice | AMD 32 and 64 bit both work OK |
| 16:29.03 | harryvv | but |
| 16:29.08 | spackle | I smell gentoo |
| 16:29.08 | usam | hm.. |
| 16:29.22 | harryvv | the amd 64 with linux 64 has some shortcommings. |
| 16:29.29 | usam | now my sip & mgcp phone pickup the each other .. |
| 16:29.32 | harryvv | lack of apps built in 64 for one reason. |
| 16:29.44 | fugitivo | harryvv: wrong |
| 16:29.44 | *** join/#asterisk max}{ (n=asd@82.79.20.17) |
| 16:29.45 | Netgeeks | Yep, I have Asterisk running on a dual core amd system with sustained 200+ calls |
| 16:29.51 | fugitivo | harryvv: perfect support for amd64 |
| 16:30.07 | harryvv | fugitivo right! thay do...but not all apps are built for amd64 ! |
| 16:30.14 | coppice | FC4 works nicely with my AMD64 X2 box |
| 16:30.18 | fugitivo | harryvv: well, just build them |
| 16:30.29 | harryvv | fug..im not a develloper |
| 16:30.33 | harryvv | developer |
| 16:30.43 | fugitivo | harryvv: you don't need to develop anything |
| 16:30.44 | Netgeeks | what apps do you need to run besides asterisk? |
| 16:30.56 | spackle | developers, developers, developers, developers |
| 16:31.00 | fugitivo | harryvv: 64bits is compatible with 32bits apps, get the source, compile it and done |
| 16:31.03 | coppice | only annoyance is the way they name the 32 and 64 library directories. it upsets a number of makefiles |
| 16:31.06 | usam | you guys are running optimized kernel or MMX optimzed on amd? |
| 16:31.07 | drray | build one unix app you are not a developer, blow one guy, you are a cocksucker for life |
| 16:31.11 | harryvv | netgeeks this was a test box ...so it was more then used for asterisk. |
| 16:31.12 | jedirl | Is it possible to place a call and make asterisk do a Playback in that channel once it's picked up in the remote end? |
| 16:31.56 | coppice | usam: are you talking about 32 bit Athlons or 64 bit? |
| 16:32.03 | Netgeeks | Harry: ah. Only apps I'm running on my AMD64 box is Asterisk, MySQL, and HTTP |
| 16:32.11 | jedirl | I can dial to the extension I've created |
| 16:32.14 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:32.17 | jedirl | but I can't connect a outgoing call to that extension |
| 16:32.25 | harryvv | netgeeks, thats fine..i run more then tat on my test box. |
| 16:32.35 | usam | coppice: runnig sempron here .. my dual xeon was RIP 2 days ago :(... |
| 16:32.36 | spackle | jedirl: context issues? |
| 16:32.45 | jedirl | spackle: I got it in the correct context |
| 16:32.48 | harryvv | netgeeks..anyway the intention of this opteron was to run high end windows 3d applications anyway. |
| 16:32.54 | coppice | semperon 32 or 64? |
| 16:32.57 | *** part/#asterisk Henguei (n=smoka@196.203.249.73) |
| 16:33.04 | usam | coppice: 32 |
| 16:33.26 | *** join/#asterisk Defraz (n=t0tal@tim.ibccom.net) |
| 16:33.31 | coppice | I used to use an athlon kernel in the 2.4 days with *. worked OK |
| 16:33.35 | Netgeeks | harryvv: oh, windows support of AMD64 is.....lacking |
| 16:33.43 | usam | now my phones wont pickup the call, .... even when you lift the handset, nothing :( |
| 16:33.50 | jedirl | Netgeeks: there's a Windows XP 64 |
| 16:34.00 | fugitivo | harryvv: now, there are not too many apps for win 64bits |
| 16:34.07 | Netgeeks | jedirl: yes, yes there is, have you played with it? |
| 16:34.08 | harryvv | netgeeks...i did not buy it for the 64 bit capability. |
| 16:34.18 | fugitivo | harryvv: you have full linux support for 64bits, but not full windows support |
| 16:34.20 | file | I run XP Pro 64-bit on here |
| 16:34.21 | jedirl | nope, all my 64bit machines run either solaris or linux |
| 16:34.21 | jedirl | :D |
| 16:34.25 | spackle | NetGeeks: is it horrible? |
| 16:34.34 | usam | i think i will try to downgrade the asterisk ... it was working before ... |
| 16:34.50 | jedirl | how can be that difficult to make a call connect to an extensions |
| 16:34.55 | harryvv | fug. There is no full 64 bit capability for a test workstation/server. |
| 16:35.11 | coppice | win98 was mostly 16 bit code. I wonder how long it will take them to get a pure 64 bit windows system |
| 16:35.18 | Netgeeks | spackle: yeah, I thought I'd give it a try since I needed to update my systems BIOS firmware and I didn't have a floppy and the only otehr app was windows based |
| 16:35.26 | fugitivo | harryvv: linux or win? |
| 16:35.27 | Netgeeks | so I downloaded XP pro 64 bit |
| 16:35.32 | harryvv | Also, I dont need access to a kazillion megs of memory. |
| 16:35.51 | drray | as long as the cpu has a 32bit emulation layer, you'll never see 64 bit windows |
| 16:35.55 | Netgeeks | Needless to say, it took me nearly a day to tweak the hardware enought to get windows to be stable enough to run for 10 minutes to flash the bios |
| 16:35.59 | fugitivo | harryvv: if you've them, use them :) |
| 16:36.02 | file | I hate people who hog the CPU on my devel box |
| 16:36.17 | Nix | :-( |
| 16:36.28 | file | root 17694 35.0 58.8 399036 298564 pts/1 R Aug08 11893:39 |
| 16:36.30 | harryvv | I perfer stability and access to all linux apps be it mundane or server based. |
| 16:36.43 | drray | jeebus |
| 16:36.59 | fugitivo | harryvv: 64bit linux is stable |
| 16:37.09 | harryvv | I perfer stability and access to all linux apps be it mundane or server based. |
| 16:37.12 | *** join/#asterisk razu (n=razu@ip192.cab63.mus.starman.ee) |
| 16:37.18 | fugitivo | harryvv: i have dual opterons servers and my notebook is amd64 too |
| 16:37.22 | coppice | 64 bit linux is about 10 years old |
| 16:37.22 | harryvv | That does not state it was unstable..its just a comment. |
| 16:37.40 | spackle | one os under linux, with stability and access for all |
| 16:37.44 | fugitivo | coppice is right |
| 16:37.48 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe201.t-net.net.ve) |
| 16:37.56 | jedirl | well |
| 16:38.03 | jedirl | NT 4 also ran in 64bit processors |
| 16:38.07 | drray | alphas |
| 16:38.08 | coppice | it ran on alphas in full 64 bits at a very early stage |
| 16:38.09 | jedirl | and MIPS |
| 16:38.10 | fugitivo | x86_64 linux is really old |
| 16:38.17 | coppice | NT was only 32 bits |
| 16:38.21 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
| 16:38.25 | harryvv | spacke, reciting the pledge of Alligence to linux? :) |
| 16:38.35 | Nix | fugitivo: no. x86_64 linux is only about 3 years old |
| 16:38.45 | fugitivo | Nix: isn't that old? :) |
| 16:38.58 | Nix | but 64bit Linux (alpha and Sparc etc) is 10 years old or so |
| 16:39.06 | Nix | 3 years? no.. |
| 16:39.12 | fugitivo | Nix: i think it's older than 3 years |
| 16:39.16 | drray | wasn't solaris 64 bit? |
| 16:39.22 | harryvv | I need to beef up on my linux skills sorry to say that it is somewhat lacking. Dont use them every day. |
| 16:39.26 | jedirl | even solaris 10 isn't full-64bit |
| 16:39.28 | coppice | I don't think the sparc version is so old, but the alpha one is. |
| 16:39.36 | *** join/#asterisk adrenaline (n=mike@h-68-164-41-130.sttnwaho.covad.net) |
| 16:39.40 | harryvv | what about SCO? |
| 16:39.40 | Nix | how long has the Opteron and Athlon64 existed for??? |
| 16:39.46 | fugitivo | one thing is x86_64 and another thing is 64bit |
| 16:39.50 | harryvv | nix three years |
| 16:39.56 | Nix | exactly |
| 16:39.57 | fugitivo | x86_64 is compatible with 32bit |
| 16:40.08 | drray | the cpu or the kernel? |
| 16:40.09 | coppice | NT was only ever 32 bits. Its was sad to see only half our alphas being used :-\ |
| 16:40.10 | Nix | no it is not |
| 16:40.36 | harryvv | Im thinking of getting LPI cirtified |
| 16:40.48 | criptos | LPI critified? |
| 16:40.49 | coppice | Nix: what is not? |
| 16:40.51 | criptos | :) |
| 16:40.55 | harryvv | google it |
| 16:41.03 | drray | LPI crucified |
| 16:41.15 | bkw_ | I SWEAR asterisk has a root exploit |
| 16:41.27 | *** join/#asterisk DrJolo (n=chatzill@host-ip58-113.crowley.pl) |
| 16:41.32 | criptos | You are going to be critized by LPI becose the way you solve your problems, isn´t the way they "certify" how to solve the problem? |
| 16:41.36 | axscode | whats the default sip proxy port where * listen? |
| 16:41.36 | bkw_ | I SWEAR I SWEAR I SWEAR it has a root exploit |
| 16:41.45 | coppice | everything has a root exploit, only some have been discovered |
| 16:41.53 | bkw_ | Box runs nothing but sshd (latest verison) and asterisk on sip and iax |
| 16:42.08 | bkw_ | and it gets owned by some stupid movie kiddies |
| 16:42.15 | criptos | asterisk have a lot of insecure string handling all over the code... |
| 16:42.20 | coppice | sshd has had its share of exploits |
| 16:42.23 | bkw_ | I swear I'm gonna call the mpaa myself and say HERE go get these fuckers |
| 16:42.25 | fugitivo | bkw_: that doesn't mean that asterisk has a root exploit |
| 16:42.26 | bkw_ | coppice, yes it has |
| 16:42.29 | harryvv | drray, who is a customer going to hire..somone one that says thay know linux or somone that says thay are cirtified? Customers want assurance and reasurance. Unfortunaly how techies look at it, cirtification sells. |
| 16:42.34 | bkw_ | I trust sshd more than asterisk |
| 16:42.44 | bkw_ | and I think you can agree on that one |
| 16:42.46 | criptos | But I suppose, it must be this way upcon velocity... |
| 16:42.52 | drray | well, harryvv I start by spelling certified right on your resume ;) |
| 16:43.03 | coppice | you means * is *worse* than sshd? ooh, bad! |
| 16:43.05 | fugitivo | bkw_: yes, but... |
| 16:43.10 | criptos | drrray lol |
| 16:43.11 | drray | and I mean that in the nicest way possible |
| 16:43.26 | file | bkw_: what about manager? |
| 16:43.26 | groogs | bkw_: do you run asterisk as root? |
| 16:43.35 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 16:43.37 | bkw_ | manager is blocked from access but from our management IP's |
| 16:43.42 | fugitivo | bkw_: what OS? |
| 16:43.59 | *** part/#asterisk axscode (n=Netgeker@203.131.153.34) |
| 16:43.59 | bkw_ | custome distro |
| 16:43.59 | fugitivo | bkw_: huh? |
| 16:43.59 | file | custom distro of Linux |
| 16:44.03 | fugitivo | ok |
| 16:44.15 | fugitivo | from scratch? |
| 16:44.25 | *** join/#asterisk popvoxdave (i=user@dave2.toad.net) |
| 16:44.25 | JerJer | it is not hard |
| 16:44.55 | drray | I hope harryvv know I was just teasing |
| 16:44.57 | JerJer | i had a working linux envrionment in maybe 2 hours of thought and another 2 or 3 of work |
| 16:45.00 | drray | er, knows |
| 16:45.19 | harryvv | drray, I only had 3 hours of sleap last night. |
| 16:45.23 | DrWho17 | harryvv: certification, our M$ certified folks are barely cutting it providing tech support |
| 16:45.36 | DrWho17 | tier 1 support |
| 16:45.45 | *** join/#asterisk bbjunkie_ (n=bbjunkie@i-83-67-48-18.freedom2surf.net) |
| 16:45.53 | harryvv | DrWho17, yea but thay are working. Tier 1? mmmm you mean break/fix? |
| 16:45.54 | bkw_ | custom linux distro takes some guess work out of a few things when you have specific needs |
| 16:46.02 | *** join/#asterisk docubot (n=docubot@65.221.143.2) |
| 16:46.16 | fugitivo | bkw_: what kernel? |
| 16:46.18 | Nix | coppice: x86_64 can run i386 binaries but it is not "compatible" |
| 16:46.21 | drray | tier 1 means front line phone monkey |
| 16:46.24 | DrWho17 | harryvv: we don't let them touch anything important, and hand them flowcharts to troubleshoot dialup/email problems all day |
| 16:46.28 | spackle | bkw_: specific needs - like security? |
| 16:46.32 | bkw_ | fugitivo, 2.6.9 |
| 16:46.40 | Netgeeks | Any Asterisk SIP implementation Guru's want to take a look at this and see if you can tell me where I am screwed up? http://pastebin.ca/21743 |
| 16:46.44 | Nix | x86_64 binaries cannot run on 386 machines.. |
| 16:47.02 | bkw_ | Netgeeks, Asterisk? SIP Implementation? Don't ya mean SIP Nightmare? |
| 16:47.40 | coppice | nobody said that. they said the x86_64 linux machines are compatible with the 32 bit binaries |
| 16:47.40 | harryvv | DrWho17 so you work for a isp then? |
| 16:47.44 | *** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net) |
| 16:47.48 | Netgeeks | bkw: at the moment, yes, nightmare, as it seems both IAX and sip are borken in different places to the degree that I can't get something simple to work using either |
| 16:47.55 | file | ugh my stomach hurts |
| 16:48.07 | drray | some 64 bit cpu's have 32 bit emulation layers |
| 16:48.10 | bkw_ | Netgeeks, at this point SIP is less broken on the audio quality side |
| 16:48.13 | harryvv | my cpu does |
| 16:48.19 | DrWho17 | harryw: ja |
| 16:48.27 | Netgeeks | bkw: I can't get that far.... |
| 16:48.32 | harryvv | drwho17 your 17 years of age? |
| 16:48.33 | Netgeeks | in this specific example |
| 16:48.41 | DrWho17 | harryw: no |
| 16:48.44 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 16:48.44 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:48.48 | drray | I think it's his 17th regneration |
| 16:48.53 | DrWho17 | correct |
| 16:48.54 | Netgeeks | in all appearances it looks like the canreinvite=no is being ignored |
| 16:49.04 | drray | or there were 16 other losers using the nick |
| 16:49.07 | drray | ;) |
| 16:49.10 | Netgeeks | anthm! just the chap I need to speak with.... <evil grin> |
| 16:49.21 | bkw_ | anthm, RUN |
| 16:49.24 | bkw_ | RUN |
| 16:49.28 | bkw_ | haha |
| 16:49.29 | *** join/#asterisk pbxbart__ (n=pbxbart@ip-80-226-228-191.vodafone-net.de) |
| 16:50.29 | drray | I'm torn, I never much cared for DrWho until Eccleston turned up |
| 16:50.55 | r0m | good afternoon |
| 16:51.20 | harryvv | Im torn if I should continue into this voip venture based on this article on voip and its compitition. |
| 16:51.46 | harryvv | all these carriers are nickel and dimming each other to death. |
| 16:51.52 | Hmmhesays | of course they are |
| 16:51.59 | Hmmhesays | that is the nature of business |
| 16:52.03 | drray | harryvv - just don't get in to bed with just one |
| 16:52.23 | Netgeeks | Thats been going on for 20 years at least.... since the de-reg of Ma Bell |
| 16:52.49 | groogs | i was going to refer someone to an ISP i used to use.. but apparently they're shutting down |
| 16:53.01 | harryvv | The article was fairly long. It said if your only primary bussiness is voip, then you may go out of bussiness if there is no other work outside of voip to keep the bussiness alive. |
| 16:53.12 | drray | I've been tempted to try and network with other asterisk users some use of my Seattle PSTN for exchange with their other areas |
| 16:53.51 | harryvv | drray, your in seattle? |
| 16:53.51 | drray | yes |
| 16:53.51 | *** join/#asterisk fugitivo (n=ajf@201.255.102.212) |
| 16:53.51 | harryvv | drrray, so you mean share pstn lines |
| 16:54.00 | groogs | theres a note on their site about how another isp in the area is partly owned by sprint, and due to dereg stuff, they have direct access to the CO's and can put DSLAMs right on customer lines. basically, they could offer "naked dsl" at retail for less than anyone else could even buy it wholesale from bell |
| 16:54.28 | harryvv | drray, look at the gas prices down at one of your gas stationes lately? |
| 16:54.29 | groogs | so the owner was effectively giving up (istop.com) |
| 16:54.31 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 16:54.33 | drray | harryvv - I was thinking more along the lines of trading minutes with someone in the bay area |
| 16:54.35 | Netgeeks | The people who own the layer 1 network will always have the advantage |
| 16:54.45 | drray | harryvv - I don't own a car, so no ;) |
| 16:54.56 | harryvv | drray, okay good |
| 16:55.08 | drray | gas could be $9 a gallon for all I care |
| 16:55.20 | harryvv | GAS has gone up 20 cents this morning..a 20% leap from yesterday. What a fricken shock. |
| 16:55.38 | harryvv | BOMB opec...those criminals. |
| 16:55.45 | drray | compared to what it cost in the 50's compared to what you earn now, gas is cheaper now |
| 16:55.50 | fugitivo | sessionsaver for firefox is great |
| 16:56.09 | groogs | harryvv: hm, it was down a couple cents here this morning |
| 16:56.14 | Netgeeks | drray: nah, you don't gas to go that high, it would impact you, all your other costs would go up becuse nothing anymore is made locally, everything is shipped in/out |
| 16:56.16 | drray | harryvv - gas is expensive because of futures trading |
| 16:56.28 | harryvv | drray, no its not...vancouver just got over a month long truckers strike that shut down all the ports because of diesel going sky high. |
| 16:56.30 | drray | not because of slouch in demand |
| 16:56.42 | drray | er, slouch of supply |
| 16:59.21 | *** join/#asterisk mrkz (n=marko@BSN-77-54-92.dsl.siol.net) |
| 16:59.26 | drray | I need to look into dundi |
| 17:00.21 | generalhan | can anyone help me out with my cisco 7960 ? i at least have it in a SIP mode, but it just wont dial out or anything. i gave it the U/N and P/W for an exten that i have tested with another phone and it just wont dial out, even though i get a dial tone when i pick up the phone. anyone have any ideas ? |
| 17:00.25 | coppice | In china there is a gas shortage. the government controls retail prices, but can't control the raw material price. the refiners are looking for every litre they sell. they stopped refining :-) |
| 17:00.41 | coppice | s/looking/loosing |
| 17:00.54 | harryvv | coppice yea...china is also driving up the price of fuel. |
| 17:01.05 | coppice | and india |
| 17:01.12 | usam | so .... PROBLEM WITH asterisk 1.0.9 ... downgraded to 1.0.5 everything is working again .. |
| 17:01.26 | drray | futures speculators are driving up the price of fuel right now, same thing that occured during the rolling blackouts ... |
| 17:01.51 | coppice | but india and china are real problems driving it up. |
| 17:02.15 | spackle | yeah, U.S. isn't to blame. |
| 17:02.34 | harryvv | its a serios problem here |
| 17:02.38 | JerJer | blame canada |
| 17:02.47 | harryvv | at over 4 dollars a gallon cdn people are getting pissed. |
| 17:03.02 | harryvv | A suburban takes 100-115 dollars to fill its tank. |
| 17:03.07 | JerJer | i can hear them up there at night sharpening their ice skates |
| 17:03.09 | Wonka | we pay 1.30 EUR per liter |
| 17:03.09 | coppice | spackle: the US being a problem is a given. :-) |
| 17:03.16 | jedirl | we pay 1.00 EUR |
| 17:03.17 | Wonka | for super |
| 17:03.25 | spackle | just making sure ;-) |
| 17:03.26 | Wonka | coppice: ack |
| 17:03.26 | harryvv | wonka, guess what..canada is now 1.20 per liter this morning. |
| 17:03.49 | Wonka | a week ago, it was 1.25EUR/l for super... |
| 17:03.55 | drray | a car is a waste of resources, 90% of the energy produced does not even go to turn the wheels, it's wasted as heat |
| 17:04.10 | Wonka | 2 months ago, 1.20EUR/l super was expensive |
| 17:04.27 | drray | compared to what you paid in the 50's gas is plenty cheap |
| 17:04.30 | coppice | the energy used to make the car is actually a large part of the life cycle energy |
| 17:04.36 | *** join/#asterisk razu (n=razu@ip192.cab63.mus.starman.ee) |
| 17:04.38 | fugitivo | i hate admins spam |
| 17:05.08 | Wonka | and this time, it wasn't the government rising the taxes, but raw oil price rising, then oil corps not lowering prices when it fell, but sacking extra cash, and then again rising prices... |
| 17:05.26 | drray | and the futures market being speculated on |
| 17:05.26 | fugitivo | Wonka: you live in argentina? |
| 17:05.32 | Wonka | fugitivo: germany |
| 17:05.41 | fugitivo | Wonka: lol |
| 17:05.48 | jedirl | blame turkey |
| 17:05.51 | harryvv | drray, not today...you lived a better life in the 1950s over all. Today people depend on there cars more then in the 1950s. |
| 17:05.52 | fugitivo | Wonka: same here |
| 17:05.53 | jedirl | :)) |
| 17:06.33 | Wonka | when we still hat the DM, prices rose and fell in 0.01 or 0.02 DM steps, 0.03 DM they didn't dare |
| 17:06.48 | coppice | short sighted governments have changed cars from luxuries into necessities in many places. it doesn't have to be that way |
| 17:07.05 | Har|ock | it's only going to get worse |
| 17:07.09 | Wonka | now, with 1 EUR == 1.95 DM, prices rise in 0.03EUR steps and fall, if ever, in 0.01EUR steps, one might think... |
| 17:07.14 | drray | seattle won't even build the monorail, even though it has passed 4 times via public vote.. |
| 17:07.20 | spackle | coppice and short-sighted cities and communities too. |
| 17:07.34 | harryvv | drray, ever traveled on the skytrain in vancouver? |
| 17:07.41 | jedirl | at least in germany you have great public transport, right Wonka ? |
| 17:07.44 | drray | harryvv - yes, every time I am up there |
| 17:07.45 | Wonka | LOL |
| 17:07.50 | Wonka | in the big cities maybe |
| 17:07.50 | coppice | I live in a place which is quite affluent, and has few cars. it can happen |
| 17:08.02 | drray | coppice, where? |
| 17:08.05 | harryvv | drray, do you want the monorail then? i love the sky train. |
| 17:08.08 | coppice | HK |
| 17:08.10 | Wonka | i need _at_ _least_ 75mins from home to university, which is 15km on the road |
| 17:08.14 | Wonka | with public transport |
| 17:08.15 | drray | I want a monorail |
| 17:08.20 | jedirl | hehe |
| 17:08.25 | Har|ock | it's really silly to be using up crude oil as a fuel when it is the raw material of pretty much all synthetic material we use |
| 17:08.32 | drray | I'm pissed that I have to buy a car |
| 17:08.36 | spackle | Monorail, hrmph, that's more of a Shlebyville idea |
| 17:08.40 | criptos | I want a personal helicopter :) But, well, a segway could do the trick :) |
| 17:08.46 | harryvv | If a city was built around a monorail/skytrain it would never be so expensive to build. |
| 17:09.04 | Wonka | they would make better schedules, then it would only be like 45 or 50 min, but "duck! a customer!" |
| 17:09.17 | coppice | put the trains underground |
| 17:09.28 | jedirl | i hate trains |
| 17:09.29 | *** join/#asterisk Defraz_ (n=t0tal@tim.ibccom.net) |
| 17:09.30 | Faithful | Hey guys, I need some advice... I have 2 identical server boxes and I want one to be a failover. What would the best method of achieving this? I was thinking rsyncing one box to the other. |
| 17:09.43 | spackle | rsync |
| 17:09.47 | drray | rsync |
| 17:09.49 | criptos | use mysql |
| 17:10.06 | jedirl | Faithful: spend a few bucks and get acronis trueimage |
| 17:10.07 | spackle | I hate phones |
| 17:10.11 | criptos | for keep registration at least with redundancy... |
| 17:10.13 | *** join/#asterisk diego_br (n=liebert@200.208.241.178) |
| 17:10.17 | criptos | I hate voip... |
| 17:10.23 | *** join/#asterisk pussfeller (n=todd@12.150.129.171) |
| 17:10.28 | drray | I <3 zap |
| 17:10.35 | spackle | lol |
| 17:11.00 | drray | I've had more fun the past 1.5 years with asterisk and the channel banks |
| 17:11.29 | spackle | "we love the channel banks, the channel banks are our friends" |
| 17:11.36 | SwK[Work] | appearently the British Royal household's phone number is (+44) (0)20 7930 4832... its amazing what you can find on google |
| 17:11.52 | *** join/#asterisk hellagony (n=egutierr@200.121.129.178) |
| 17:12.02 | criptos | DIALWAR to the Royal Households!!!! |
| 17:12.22 | SwK[Work] | i'm still trying to find a number for the kremlin |
| 17:12.24 | drray | ask if prince albert ever had fergie in a can |
| 17:12.28 | drray | er, the can |
| 17:12.31 | drray | doh |
| 17:12.35 | drray | I pooched that |
| 17:12.43 | *** part/#asterisk pbxbart__ (n=pbxbart@ip-80-226-228-191.vodafone-net.de) |
| 17:15.11 | Netgeeks | grrrrrrrrrrrrrrrrrr |
| 17:15.21 | Wonka | grrrrrrrrrrrrrrrrrrrr |
| 17:15.30 | *** join/#asterisk Twister (i=Twister@24-179-91-088.dhcp.chtn.wv.charter.com) |
| 17:16.07 | Netgeeks | "not a dev question, go talk to someone on #asterisk to help you figure out authentication".... |
| 17:16.16 | Twister | can i use bri on a te110p? |
| 17:16.56 | drray | go read the wiki |
| 17:17.08 | drray | not you twister |
| 17:17.15 | drray | I was replying to Netgeeks |
| 17:17.29 | Twister | hehe, ya read my mind |
| 17:17.38 | Netgeeks | drray: heh, yeah, thats another one l love to see people say |
| 17:17.39 | drray | I need some sleep |
| 17:17.43 | bkw_ | JunK-Y, yo |
| 17:17.51 | PupenoL | What softphone for Linux do you recomend that can iteroperate nicely with Asterisk (kphone does, but it crashes) ? |
| 17:18.01 | jedirl | linphone? |
| 17:18.03 | drray | I like moziax |
| 17:18.08 | fugitivo | PupenoL: kphone is a headache, it's a pity |
| 17:18.23 | fugitivo | PupenoL: somebody was saying something about kiax, i dind't try it |
| 17:18.36 | Wonka | xten xlite looks nice and works, but it's not OSS :( |
| 17:18.53 | PupenoL | fugitivo: the developer didn't have much of a clue how to make a KDE program (it breaks lot's of KDE rules). |
| 17:19.01 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:19.12 | fugitivo | PupenoL: kphone or kiax? |
| 17:19.21 | PupenoL | fugitivo: kphone. |
| 17:19.26 | *** join/#asterisk piruchoOSX (n=Mauricio@adsl-69-233-188-132.dsl.pltn13.pacbell.net) |
| 17:19.34 | fugitivo | yes, i know |
| 17:20.07 | PupenoL | I'll try kiax. |
| 17:20.23 | drray | moziax is a plugin for mozilla |
| 17:20.41 | fugitivo | drray: i couldn't make it work |
| 17:20.48 | drray | really? |
| 17:20.59 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
| 17:20.59 | coppice | do these things all use iaxclient, or are there any other IAX libraries? |
| 17:21.12 | fugitivo | drray: yep, i didn't invest time on it, but it didn't work at the first try |
| 17:21.20 | *** join/#asterisk gooagle (n=goldenol@ns2.xoasisnetworks.com) |
| 17:21.25 | drray | fugitivo - I don't blame you |
| 17:21.37 | gooagle | anyone used the varion cards? |
| 17:21.38 | drray | fugitivo - I spent way too much time getting the cisco working |
| 17:21.45 | fugitivo | drray: lol |
| 17:21.45 | drray | gooagle - i do |
| 17:21.47 | JerJer | gooagle: no, don't |
| 17:21.49 | harryvv | ddray, what router? |
| 17:21.54 | drray | what router? |
| 17:22.07 | harryvv | yes, what was it a 3600 series? |
| 17:22.08 | JerJer | good rephrase |
| 17:22.09 | gooagle | drray: ? no good? |
| 17:22.09 | PupenoL | What *SIP* softphone for Linux do you recomend that can iteroperate nicely with Asterisk (kphone does, but it crashes) ? |
| 17:22.25 | harryvv | Pup, xlite |
| 17:22.26 | drray | gooagle - it's a tor2 card.. it's fine |
| 17:22.32 | JerJer | PupenoL: Asterisk |
| 17:22.56 | drray | gooagle - it's the card digium was selling 3-4 years ago, it does not have echo canceling on it |
| 17:23.08 | gooagle | similar to the t400p? |
| 17:23.14 | PupenoL | harryvv: thanks, but I am not interested in non-free software. |
| 17:23.19 | drray | I think it predates that |
| 17:23.39 | drray | but I use it in a production for a small hotel pbx |
| 17:23.55 | fugitivo | PupenoL: xlite is free, but not opensource |
| 17:24.15 | gooagle | are you using it for PSTN or channel bank applications? |
| 17:24.20 | PupenoL | fugitivo: I mean free as in freedom or free speech (not as in gratis). |
| 17:24.22 | gooagle | I am worried about the echo issue |
| 17:24.22 | drray | gooagle, yes |
| 17:24.36 | drray | if you are worried about echo, get the digium card |
| 17:24.52 | drray | the new ones have echo cancel on the card (I've not used them) |
| 17:25.18 | drray | our shop is a govarion card with zhone channel banks |
| 17:25.36 | fugitivo | PupenoL: then try kiax or linphone |
| 17:25.39 | drray | which I swear to god I will get rid of this fiscal year |
| 17:25.40 | JerJer | drray: oh god you are just asking for trouble |
| 17:25.44 | drray | the channel banks |
| 17:25.54 | drray | jerjer - trouble I have had |
| 17:26.04 | JerJer | yeah no wonder |
| 17:26.06 | drray | I'm getting adit 600's this year |
| 17:26.09 | PupenoL | fugitivo: is kiax a sip phone ? I've tried linphone, it didn't work. |
| 17:26.14 | drray | but the zplexs work |
| 17:26.16 | fugitivo | PupenoL: iax |
| 17:26.17 | JerJer | and get a real T-1 card |
| 17:26.19 | gooagle | JerJer... not a fan of the Varion's? |
| 17:26.26 | fugitivo | PupenoL: k-iax :) |
| 17:26.29 | JerJer | gooagle: no - they are inferior |
| 17:26.34 | drray | jerjer - at most I have 8 calls on it at a time |
| 17:26.38 | drray | I average 4 calls |
| 17:26.41 | PupenoL | fugitivo: I need a sip phone. |
| 17:26.46 | drray | right now there are zero calls on it |
| 17:27.03 | fugitivo | PupenoL: i'm out of options |
| 17:27.25 | PupenoL | I got linphone to register, but I fail to make or answer calls with it (it rings thoug). |
| 17:27.34 | *** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net) |
| 17:27.42 | Rav1974 | hello hello :) |
| 17:27.46 | drray | jerjer - my first card was a digium 1 port, I simply could not justify the 1500 dollars for a fourport in our setup |
| 17:27.57 | coppice | gooagle: the govarian cards are as good or bad as the digium versions were - but there is a good reason why digium replaced them :-) |
| 17:28.29 | gooagle | Does anyone find the Sangoma cards to be as good as the Digium ones? |
| 17:28.42 | drray | gooagle - if in doubt get the better cards.. I kind of new what I was doing, and was willing to eat the loss if the govarion/zplex did not work out |
| 17:28.48 | drray | er, knew |
| 17:29.10 | drray | I also did not come in here bitching about the zplex needing to be reset three times in one weekend :D |
| 17:30.24 | DarthClue | gooagle: sangoma cards are equal to if not better than digium cards |
| 17:30.59 | drray | I think the govarion cards are a viable option if you are willing to work with them |
| 17:31.07 | coppice | goagle: people like the sangoma cards, but they complain the drivers are a pain to set up |
| 17:31.10 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 17:31.11 | *** join/#asterisk trimi (i=Pharrel@62.162.242.213) |
| 17:31.23 | drray | if money was no option I'd do digium cards |
| 17:31.24 | PupenoL | it seem kphone is my best option then. |
| 17:31.41 | drray | since they give so much back to asterisk |
| 17:31.44 | DarthClue | coppice: that is about to change |
| 17:31.47 | ender | is there a lot of config file syntax change between 1.0 and 1.2 ? |
| 17:31.55 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 17:32.11 | shadebob | hi, I have a problem with my * box. Sip phone don't register to asterisk... everything is ok for the network, phone config and sip.conf... I user cvs v1-0 |
| 17:32.31 | drray | do the phones work? |
| 17:32.37 | shadebob | yes |
| 17:32.44 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
| 17:32.48 | shadebob | 2 budgetone - 2 swissvoice |
| 17:32.53 | fugitivo | shadebob: what does the CLI says? |
| 17:32.56 | shadebob | xlite don't register |
| 17:33.12 | shadebob | fugitivo : Nothing. sip show peers show unmonitered |
| 17:33.30 | fugitivo | shadebob: start the CLI with -vvvvvvvvvvvvvv and check the messages |
| 17:33.33 | drray | I was speaking of what mark has done |
| 17:33.42 | drray | given back was a poor choice |
| 17:33.53 | shadebob | fugitivo : no error. chan_sip load correctly |
| 17:33.59 | *** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net) |
| 17:34.18 | fugitivo | shadebob: you don't see the phone trying to register? |
| 17:34.27 | shadebob | an unload chan_sip.so and a load chan_sip.so work |
| 17:34.34 | DarthClue | got a list? i know what others have given, but very little seems to have come out of digium or mark recently |
| 17:34.34 | shadebob | fugitivo : yes |
| 17:34.55 | fugitivo | shadebob: yes what? |
| 17:34.57 | *** join/#asterisk Brumle (n=brumle@brumle.com) |
| 17:35.01 | drray | sorry, not getting sucked into it |
| 17:35.08 | coppice | people have to put a lot into asterisk because of the dumb licencing. It would be a far smaller project if it were GPL |
| 17:35.17 | shadebob | fugitivo : no registration request |
| 17:35.46 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 17:35.53 | drray | I think TDMoE is amazing |
| 17:36.05 | DarthClue | i'm serious. but i guess that is what happens when someone asks for a list, one can't be produced or simply isn't. |
| 17:36.12 | fugitivo | shadebob: then your phone is not trying to register to your asterisk box |
| 17:36.23 | DrWho17 | DarthClue: t.38 |
| 17:36.38 | shadebob | fugitivo ; why? |
| 17:38.44 | DarthClue | DrWho17: what about it? |
| 17:39.15 | fugitivo | shadebob: i don't know why |
| 17:39.32 | fugitivo | shadebob: did you setup the ip address of your asterisk box in your phone? |
| 17:39.44 | PupenoL | coppice: isn't it GPL ? |
| 17:39.49 | shadebob | fugitivo : it's my xxx * install and I don't known the pb |
| 17:39.58 | DarthClue | PupenoL: nope |
| 17:40.16 | Hmmhesays | ldap ldap ldap |
| 17:40.17 | drray | it's the same license as apache, unless I'm mistaken |
| 17:40.30 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 17:40.49 | *** join/#asterisk Ganlron (n=Ganlron@omega.csolve.net) |
| 17:41.50 | DarthClue | drray: not quite. take a closer look. |
| 17:42.08 | BrianR___ | aparently the speed dial buttons on the GXP-2000 don't work for transfers. That's lame. |
| 17:42.22 | *** join/#asterisk ScaredyCat (n=ScaredyC@84.119.131.232) |
| 17:43.07 | coppice | PupenoL: GPL plus something that means they can use anyone else's GPL code. Hence, they have a half baked RTP, a half-baked SIP, etc. instead of absorbing the good stuff |
| 17:43.30 | pauldy | BrianR___, have you tried the new 1.0.1.12 firmware? |
| 17:43.32 | harryvv | so who can vouce for race.com and its sip reliability? |
| 17:43.45 | PupenoL | coppice: you mean that code inside Asterisk can't use third party GPL libraries ? |
| 17:43.45 | ScaredyCat | kroq |
| 17:44.18 | coppice | PupenoL: yep. its licenced under the NIH licence |
| 17:44.49 | *** join/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca) |
| 17:44.51 | TripleFFF2sdf | 05:27:22 WARNING[25478]: config.c:893 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 17:44.53 | TripleFFF2sdf | on 1.2 |
| 17:44.55 | drray | I thought that it meant that digium, could license a copy of asterisk under a different license if they wanted.. the same way apache does it |
| 17:44.57 | TripleFFF2sdf | anyidea ? |
| 17:45.31 | drray | the same way that all gpl licensed authors can do it |
| 17:45.42 | mutilator | anyone know where i can get a 'crappy' 1U server? |
| 17:45.56 | TripleFFF2sdf | crapstuff.com |
| 17:45.57 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 17:45.57 | harryvv | a case or a server |
| 17:45.59 | ScaredyCat | I have one, it's really really creap |
| 17:46.03 | ScaredyCat | crap |
| 17:46.06 | mutilator | well |
| 17:46.12 | drray | 1u cases are cheap at newegg |
| 17:46.14 | ScaredyCat | I throw rocks at it |
| 17:46.14 | mutilator | server |
| 17:46.25 | mutilator | case/mobo anyway |
| 17:46.31 | DarthClue | drray: the license automatically assigns copyright to digium aka mark and voilates the gpl in doing so. therefore, using any other libraries that are gpl would taint the asterisk license and as such can't be used with anything included in asterisk by default. |
| 17:46.47 | coppice | drray: they can only do that if they do everything themselves. they cannot make use of anything anyone else does |
| 17:46.52 | ScaredyCat | yes.. tainted gpl.. |
| 17:46.52 | mutilator | i just need a dummy station to go from 1-4 24 port channel banks and send out via iax to another destination |
| 17:47.02 | TripleFFF2sdf | i to enable engine mysql on 1.2 ? |
| 17:47.04 | mutilator | so it doesn't have to be a super computer |
| 17:47.16 | mutilator | even a .5U would do |
| 17:47.18 | drray | n new egg |
| 17:47.23 | DarthClue | yeah i know, as if it wasn't tainted to begin with, but that is the basic gist as i understand it |
| 17:47.53 | ScaredyCat | try buying a license... |
| 17:47.57 | mutilator | i need something with older stuff |
| 17:48.03 | mutilator | i probly don't even need 1ghz to do this.. |
| 17:48.23 | mutilator | and thats like as low as they go |
| 17:48.24 | *** join/#asterisk brookshire (n=matt@digium.com) |
| 17:48.24 | ScaredyCat | ... digium took all the paperwork and never got back to me dispite numerous calls |
| 17:48.36 | coppice | buying a licence is not the core issue. the real issue is all the half baked modules |
| 17:48.38 | DarthClue | ScaredyCat: for? oh, well, i have no need for such a thing. |
| 17:48.57 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 17:49.10 | ScaredyCat | it is for me... I want to sell modules per server to a customer... |
| 17:49.16 | malverian[work] | Hmm... for some reason Dial(...,...,rTt) does not generate a ring sound to the calling party. |
| 17:49.31 | ScaredyCat | if I ave to give them the source on the first sale they won't be buying it a second time... |
| 17:50.20 | mutilator | anyone bought anything like that before? |
| 17:50.32 | coppice | ScaredyCat: sounds like you really aren't an open source kinda guy :-) |
| 17:51.21 | ScaredyCat | I am coppice - but if a telco is going to use it to run their operation they can fecking pay me properly or pay per server... |
| 17:52.22 | ScaredyCat | I tried to do it properly but digium just wouldn;t even talk about it... |
| 17:52.27 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 17:52.56 | ScaredyCat | or lost interest |
| 17:53.01 | ScaredyCat | I dunno which... |
| 17:53.05 | coppice | we have commercial pricing for * |
| 17:53.08 | pauldy | if it truely is a module and does not make direct use of digium code then where is the problem |
| 17:53.24 | ScaredyCat | because it links against * code |
| 17:53.40 | pauldy | is it staticly linked? |
| 17:54.05 | ScaredyCat | no |
| 17:54.12 | coppice | it doesn't matter if its static or dynamic |
| 17:54.17 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 17:54.26 | ScaredyCat | but the gpl doesn;t care |
| 17:55.23 | pauldy | well ianal but it sure seems that would be ok to produce code that calls another apps functions without worry about the procurement of that software |
| 17:55.27 | PupenoL | coppice: I can't find anything that says that Asterisk is not GPL, where is it ? |
| 17:55.51 | *** join/#asterisk jpayne (n=jpayne@baconhouse.sackheads.org) |
| 17:56.06 | *** join/#asterisk slak- (i=slak@ircdz.com) |
| 17:56.06 | ScaredyCat | PupenoL: it's dual licensed... but digium break the gpl, but the gpl allows for that.. |
| 17:56.13 | slak- | hi |
| 17:56.15 | slak- | Aug 31 15:25:13 WARNING[21225]: pbx.c:2325 __ast_pbx_run: Invalid extension '#', but no rule 'i' in context 'incoming' |
| 17:56.19 | *** join/#asterisk darkskiez (n=darkskie@host86-133-147-187.range86-133.btcentralplus.com) |
| 17:56.33 | slak- | whats i? also, if a user enters an invalid extension how can i make it so he/she does not have to call back |
| 17:56.39 | DaPrivateer | heh |
| 17:56.41 | DaPrivateer | i is invalid |
| 17:56.42 | ScaredyCat | pauldy: well, technically I could keep my gob shut, but I wanted to do it right |
| 17:56.44 | slak- | rather restart the menu |
| 17:56.57 | DaPrivateer | you would do something like exten => i,1,Goto(incoming,s,1) |
| 17:56.59 | PupenoL | ScaredyCat: dual license it's ok, Asterisk doesn't break GPL because GPL doesn't forbid you te license code you own under a different license. |
| 17:57.15 | ScaredyCat | no, that's not why it breaks it |
| 17:57.26 | generalhan | can anyone helpe me out with an error that im getting while a phone is trying to log into the asterisk system ?? |
| 17:57.31 | PupenoL | ScaredyCat: but coppice claimed that Asterisk was released under a modiffied GPL, and I can't find that. |
| 17:57.33 | ScaredyCat | but digium CAN break it under the GPL |
| 17:57.42 | DaPrivateer | slak- then when they put in an invlaid extension, they would go back to priority 1 of exten s in context incoming |
| 17:57.48 | generalhan | NOTICE[3453]: chan_sip.c:7733 handle_request: Registration from '<sip:7333@192.168.0.46;user=phone>' failed for '192.168.0.59' |
| 17:57.51 | generalhan | what is that all about ? |
| 17:58.11 | coppice | PupenoL: I didn't say modified. I said not pure. its that lack of purity that makes the code suck |
| 17:58.24 | harryvv | Darth, considering my situation why do you think my asterisk is rejecting sip calls outside of registration? |
| 17:58.28 | Nix | PupenoL: it is. it has exceptions for openssl and openh323 |
| 17:58.33 | Nix | that means modified gpl |
| 17:58.50 | pauldy | ScaredyCat, just seems odd to me that there could be issue with that and not an issue that would invalidate the entire license from digium because they link against libc |
| 17:59.04 | coppice | oh, yeah. that's true. and the exception for g.729 goes further |
| 17:59.29 | Nix | yep |
| 17:59.37 | generalhan | anyone seen that error before ? |
| 17:59.49 | ScaredyCat | pauldy: errm... mmmm |
| 17:59.50 | coppice | those can be worked around, though. you can still use G.729 and be pure GPL if you really want to |
| 17:59.51 | Hmmhesays | anyone got an audiocodes fxs box near them? |
| 18:00.01 | Hmmhesays | I need to know if you can change the rtp ports |
| 18:00.04 | harryvv | darthclue, recall when you helped me with my case last time? |
| 18:00.06 | Nix | the whole g729 things makes asterisk incompatible with any other gpl software |
| 18:00.29 | Nix | technically the openssl and openh323 exceptions would also if the other gpl software doesnt have the same exceptions |
| 18:01.33 | DarthClue | harryvv: sorry, kinda busy at the moment. if you want, pm me in about 5 hours and i'll give it a gander |
| 18:01.43 | Nix | which means that you can never legally use asterisk with mysql |
| 18:01.56 | Nix | ;-) |
| 18:01.56 | eKo1 | and who would want to anyways |
| 18:01.59 | Nix | sure |
| 18:02.05 | Nix | postgresql is much better ;-) |
| 18:02.17 | coppice | what is the issue with openssl? |
| 18:02.19 | harryvv | Darth okay |
| 18:02.21 | Nugget | the gpl is designed to be difficult to integrate with non-gpl code. It's by design. |
| 18:02.26 | ScaredyCat | isn't that why the mysql stuff is spearate? |
| 18:02.44 | harryvv | Some kind of asterisk book for small bussiness has/is out. Anyone look into that before and make comments on it? |
| 18:02.59 | Nix | ScaredyCat: yes. but that is not enough |
| 18:03.06 | generalhan | can anyone please help me out with this error ... i REALLY need to get this phone up and running |
| 18:03.06 | generalhan | NOTICE[3453]: chan_sip.c:7733 handle_request: Registration from '<sip:7333@192.168.0.46;user=phone>' failed for '192.168.0.59' |
| 18:03.10 | ScaredyCat | if it's the signate one it was out of dat when it came out harryvv |
| 18:03.16 | ScaredyCat | date |
| 18:03.26 | PupenoL | If Asterisk is GPL, why can't it be linked against other GPL code, like MySQL ? |
| 18:03.29 | Nix | as soon as YOU take the mysql module code and the asterisk code and link it to the mysql libs YOU are breaking the GPL that mysql is under because Asterisk uses a modified GPL |
| 18:03.35 | Uther_P | generalhan: wrong username and/or secret |
| 18:03.48 | ScaredyCat | generalhan are you using 192.168.0.59 as the account in fip.conf? |
| 18:03.49 | pauldy | actually according to the precompiled version I have it links against a lot more than just libc: linux-gate.so.1 libdl.so.2 libpthread.so.0 libncurses.so.5 libm.so.6 libresolv.so.2 libssl.so.5 libc.so.6 lib/ld-linux.so.2 libgssapi_krb5.so.2 libkrb5.so.3 libcom_err.so.2 libk5crypto.so.3 libcrypto.so.5 libz.so.1 libkrb5support.so.0 |
| 18:03.50 | ScaredyCat | sip |
| 18:04.08 | generalhan | Uther_P: are you sure ? its all set up correctly |
| 18:04.13 | Nix | PupenoL: Asterisk is not GPL. it is GPL + exceptions |
| 18:04.19 | *** part/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 18:04.19 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 18:04.22 | coppice | Nix: the source for * only says GPL. as long as you aren't using SSL, openh323 or the other stuff its OK |
| 18:04.23 | Uther_P | if it was all set up correctly then it would work |
| 18:04.32 | Nix | therefore any other GPL code that u link against must ALSO have the same exceptions |
| 18:04.34 | generalhan | lol |
| 18:04.34 | Nugget | mysql is only gpl for noncommercial distribution. |
| 18:04.39 | generalhan | i meant the password and username |
| 18:04.43 | Uther_P | generalhan: pastebin your sip entry for that phone |
| 18:05.01 | Nix | do you need to ask the Mysql guys to grant an exception for digium's g729 codec and openh323 |
| 18:05.07 | Nix | then you would be legally able to use mysql |
| 18:05.08 | DrWho17 | mysql was removed because of mysql licensing changes |
| 18:05.18 | Nugget | or, rather, since digium uses a slightly tweaked gpl with their disclaim copyright requirements, it doesn't mix with mysql |
| 18:05.49 | pauldy | what about the mysql client access libs? I thought they had a diff license |
| 18:05.51 | Nix | disallow=all |
| 18:05.52 | Nix | allow=speex |
| 18:05.55 | Uther_P | ~pastebin |
| 18:05.57 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 18:05.57 | slak- | dudes VoicemailMain quit working adter upgrading to 1.2 and nothing shows up on console to suggest a config problem |
| 18:05.58 | slak- | can someone help? |
| 18:06.01 | Nix | oops ;-) |
| 18:06.02 | PupenoL | What are the exeptions exactly ? |
| 18:06.10 | pauldy | if not they hould then all is well in the world |
| 18:06.11 | pauldy | hehe |
| 18:06.15 | DrWho17 | pauldy: mysql has changed them back, the linux distro's are including it again |
| 18:06.42 | Nugget | digium would still run into problems trying to sell asterisk under commercial licensing. |
| 18:06.42 | Nix | coppice: so you are either breaking the GPL license when u use openssl or when you use mysql.. which is it? |
| 18:06.49 | Twister | can i use bri on a te110p? |
| 18:06.50 | pauldy | oi I'm going to have to get a law degree to navigate these trecherous waters |
| 18:07.00 | ScaredyCat | lol |
| 18:07.06 | anthm | problems as in corefiles aplenty? |
| 18:07.11 | slak- | dudes all my applications such as VM and dat_time are broken what do i check? |
| 18:07.15 | slak- | 1.2 upgrade broke them |
| 18:07.24 | Nugget | both digium and mysql ab are dual-licensing the code, which is the crux of the problem. |
| 18:07.25 | slak- | exten => 500,1,VoicemailMain(${CALLERIDNUM}) |
| 18:07.25 | slak- | exten => 500,2,Hangup |
| 18:07.29 | jeffik | anybody usig ASTPP? |
| 18:07.39 | coppice | Nix: If I use a bunch of source which says GPL, with MySQL and without openssl, what is the problem? |
| 18:07.51 | Nix | Nugget: the dual license is NOT the problem.. |
| 18:07.53 | Nix | it is the exceptions |
| 18:07.54 | PupenoL | Nugget: not really, if it is gpl, it is cpl, period. |
| 18:08.07 | Nugget | but the exceptions are mutually incompatible. |
| 18:08.10 | Nix | coppice: the license would still be incompatible.. |
| 18:08.15 | PupenoL | er, gpl |
| 18:08.24 | *** part/#asterisk criptos (n=criptos@201.135.121.4) |
| 18:08.25 | Nix | whether the code is actually linked to or not doesnt matter |
| 18:08.39 | slak- | uh can someone respond please ;) |
| 18:08.40 | Nix | that is my understanding |
| 18:08.42 | coppice | Nix: nonsense. its GPL code under those circumstances |
| 18:08.44 | slak- | dudes all my applications such as VM and dat_time are broken what do i check? |
| 18:08.46 | Nix | it is a legal thing. not a technical thing |
| 18:09.00 | ScaredyCat | roll back slak- |
| 18:09.06 | slak- | why? |
| 18:09.12 | Nix | coppice: so it magically changes fomr gpl to modified gpl depending on your build environment?? |
| 18:09.13 | slak- | im sure its just something simple |
| 18:09.16 | ScaredyCat | because it broke ur stuff |
| 18:09.28 | slak- | i get a fast busy signal when dialing VoicemailMain |
| 18:09.28 | slak- | wtf |
| 18:09.33 | *** join/#asterisk roulduke_ (i=clpxmace@p508D3022.dip0.t-ipconnect.de) |
| 18:09.39 | coppice | Nix: consider ghostscript for comparison |
| 18:10.01 | *** join/#asterisk mhnoyes_ (n=mhnoyes@user-2ivfnj3.dialup.mindspring.com) |
| 18:10.06 | ScaredyCat | /moviecheck |
| 18:10.09 | slak- | exten => 500,1,VoicemailMain(${CALLERIDNUM}) |
| 18:10.14 | slak- | whats wrong with thatr |
| 18:10.22 | Rav1974 | guys, I wanted to test out g729, so I found the intel drivers for non commercial use. how do these compare to the digium drivers? |
| 18:10.43 | ScaredyCat | check the calleridnum variable changes slak- |
| 18:11.39 | DrWho17 | mysql support used to be in asterisk it was moved to addons when mysql changed their license |
| 18:11.39 | slak- | nope its the context :LD |
| 18:11.44 | coppice | Rav1974 what is your measure of comparison? |
| 18:12.04 | fugitivo | slak-: what is the value of ${CALLERIDNUM} ? |
| 18:12.17 | ScaredyCat | that makes it user error then slak- :) |
| 18:12.39 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 18:13.15 | *** join/#asterisk neonkl (i=neon@210.195.36.37) |
| 18:13.59 | ScaredyCat | movie time...! |
| 18:14.02 | *** part/#asterisk ScaredyCat (n=ScaredyC@84.119.131.232) |
| 18:14.49 | Rav1974 | coppice: what is the difference |
| 18:15.22 | coppice | difference in what? price? legality? speed? |
| 18:15.38 | Nugget | how is openssl a problem for asterisk licensing, though? can't openssl be relicensed however you want? |
| 18:15.54 | Rav1974 | coppice: speed basically |
| 18:16.09 | *** join/#asterisk Nix (n=Nix@81.214.255.57) |
| 18:16.23 | *** join/#asterisk FuzzyCat (n=ScaredyC@84.119.131.232) |
| 18:16.47 | slak- | encrypting *? |
| 18:16.49 | bjohnson | likely no speed difference |
| 18:17.05 | brookshire | *** NewOrleans has quit IRC (Excess Flood) |
| 18:17.07 | brookshire | :( |
| 18:17.22 | coppice | all G.729 codecs are pretty resource intensive. Last time we tried they speed of the Digium one wasn't that much difference. The Digium one has changed since then, though. |
| 18:18.37 | *** part/#asterisk mrkz (n=marko@BSN-77-54-92.dsl.siol.net) |
| 18:18.44 | Rav1974 | coppice: thanks |
| 18:21.37 | *** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net) |
| 18:22.54 | *** join/#asterisk meppl (n=mephisto@p54AAF5F3.dip.t-dialin.net) |
| 18:24.11 | *** part/#asterisk mayank (n=mayank@210-210-81-203.lan.sify.net) |
| 18:25.46 | *** join/#asterisk eMZed (n=marko@BSN-77-54-92.dsl.siol.net) |
| 18:30.16 | *** join/#asterisk neonet2006 (n=icechat5@sonic-wpg.wiband.net) |
| 18:30.40 | neonet2006 | how can you limit how many inbound calls you want through one specific context in asterisk |
| 18:31.36 | neonet2006 | how can you limit how many inbound calls you want through one specific context in asterisk |
| 18:32.00 | *** join/#asterisk FuzzyCat (n=ScaredyC@84.119.131.232) |
| 18:32.16 | Netgeeks | SetGroup and CheckGroup |
| 18:32.30 | FuzzyCat | ! |
| 18:32.41 | FuzzyCat | not in 1.2 |
| 18:33.48 | Netgeeks | Set(GROUP=<blah>) and hrm... whats the new form of checkgroup |
| 18:33.50 | neonet2006 | what is SetGroup/CheckGroup? |
| 18:34.29 | *** join/#asterisk DagMoller (n=DagMolle@tor/session/x-27e38a24d7e8f166) |
| 18:35.44 | syle | be nice if there was a calleridname and calleridnumber in sip.conf and iax.conf instead of just callerid |
| 18:35.47 | FuzzyCat | allows u to limit calls on a number... |
| 18:36.13 | FuzzyCat | or for example route to a different queue/exten if there are too many ppl in a queue |
| 18:36.56 | FuzzyCat | syle, just use: callerid=Fred Smith <123456789> |
| 18:37.10 | syle | yeah that is what i have been doing |
| 18:37.28 | syle | but they do have set(calleridnumber) etc in dialplan |
| 18:37.52 | syle | which i;ve never found a use for |
| 18:38.05 | syle | sip.conf and iax.conf are much better places to put them |
| 18:38.51 | syle | maybe a user for it in dialplan would be something like a calling card application |
| 18:38.52 | FuzzyCat | well, I use them... when I get a call come in it looksup a local table to see if it can find a match then goes to the global table ifnot |
| 18:40.22 | *** join/#asterisk tsume (n=tsume@unaffiliated/tsume) |
| 18:42.35 | *** join/#asterisk Assid (n=assid@203.115.64.59) |
| 18:42.41 | Assid | oh man.. this just isnt my day |
| 18:43.25 | *** join/#asterisk caryon (n=caryon@195.227.103.35) |
| 18:44.05 | *** join/#asterisk darkskiez (n=darkskie@host86-133-147-187.range86-133.btcentralplus.com) |
| 18:44.30 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-45.west.biz.rr.com) |
| 18:46.06 | generalhan | anyone here worked with the Cisco 7960 with asterisk ??? |
| 18:46.50 | *** join/#asterisk Ashis (i=ashis_co@61.2.79.50) |
| 18:46.58 | Ashis | Hi.... :-) |
| 18:47.45 | supaigtr | Anyone working on provisioning system for poly 300-601 and snom 360's? |
| 18:48.24 | Ashis | I have amajor Question ? What the hell is Asterisk :-| |
| 18:48.49 | glm2k | Ash: * <- this? |
| 18:48.56 | supaigtr | VOIP phone system, among other things. |
| 18:49.10 | glm2k | er Ashis, sorry |
| 18:49.15 | JerJer | ascii symbol 42 |
| 18:49.52 | glm2k | Ashis: the key below "0" on your phone keypad |
| 18:50.08 | Ashis | Well......... :-( |
| 18:50.15 | Ashis | Ok I got it |
| 18:50.18 | glm2k | Ashis: also the key above "8" on your keyboard |
| 18:50.21 | glm2k | hehe |
| 18:50.53 | Ashis | What is it doing in the IRC ??? :-@ |
| 18:51.25 | glm2k | read the topic. the website will explain everything |
| 18:53.09 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
| 18:53.33 | FuzzyCat | it's a linux based, open source pbx |
| 18:53.34 | FuzzyCat | supporting many protocols |
| 18:53.44 | MattH | Hi.. I was running a 1.0.7 system... with a 4 port analog digium card FXO... I ugpraded to cvs-head... compiled zaptel.. libpri and asterisk from cvs-head... everything was fine... however when I rebooted the zaptel card is no longer detected or loaded.. only ztdummy |
| 18:53.44 | Rav1974 | glm2k: its above the 9 key on the number pad too :) |
| 18:53.45 | MattH | any thoughts? |
| 18:53.46 | FuzzyCat | Ashis I have amajor Question ? What the hell is Asterisk :-| |
| 18:53.46 | Ashis | Ok..... But It tells Nothin...at the start page :'( |
| 18:53.48 | FuzzyCat | FuzzyCat it's a linux based, open source pbx |
| 18:53.50 | FuzzyCat | FuzzyCat supporting many protocols |
| 18:54.02 | glm2k | Rav1974: lol. good call. i missed that. |
| 18:54.06 | fugitivo | MattH: what module are you using? |
| 18:54.10 | FuzzyCat | http://www.aterisk.org |
| 18:54.13 | FuzzyCat | http://www.asterisk.org |
| 18:54.27 | Ashis | pbx..... Phone Exchange thing.... ??? |
| 18:54.32 | glm2k | aye |
| 18:54.42 | FuzzyCat | telephone system Ashis |
| 18:54.44 | Ashis | Thanks :-D |
| 18:55.02 | MattH | fugitivo, : wxco? I'm drawing a blank actually as to what it was using before |
| 18:55.50 | MattH | it is a WCTDM/0 "Wildcard TDM400P REV e/f" |
| 18:56.38 | fugitivo | MattH: wcfxo? |
| 18:56.47 | MattH | yeah that was it :P |
| 18:57.42 | MattH | and a modprobe wcfxo comes up /lib/modules/2.4.21-27.0.1.EL/misc/wcfxo.o : init_module No such device |
| 18:57.49 | *** part/#asterisk Ashis (i=ashis_co@61.2.79.50) |
| 18:58.03 | fugitivo | MattH: try wcfxs |
| 18:58.33 | MattH | well that's just sitting there.. no output yet |
| 18:59.15 | fugitivo | wait, cvs? i think it uses wctdm |
| 18:59.58 | MattH | oh really |
| 19:00.14 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 19:00.17 | MattH | hrmm that one also is just sitting there doing nothing =\ |
| 19:01.18 | fugitivo | dmesg shows nothing? |
| 19:01.25 | MattH | nope |
| 19:01.52 | MattH | last item is usbserial.c: USB serial support registered for Generic |
| 19:01.59 | *** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-80.z142-153-67.customer.algx.net) |
| 19:02.16 | fugitivo | are you using kernel 2.4 ? |
| 19:02.37 | MattH | yes |
| 19:05.58 | *** join/#asterisk pc4 (n=pc@209.151.52.81) |
| 19:08.19 | MattH | hrmm |
| 19:08.21 | MattH | if I run zaptel start |
| 19:08.23 | MattH | it loaded fine |
| 19:08.50 | MattH | what the heck |
| 19:09.03 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe201.t-net.net.ve) |
| 19:09.23 | slak- | UGH |
| 19:09.29 | *** join/#asterisk todd1 (n=toddf@ns0.fries.net) |
| 19:09.32 | slak- | QoS is confusing and a pain in the ass |
| 19:09.54 | MattH | supaigtr, seriously? |
| 19:10.00 | epablo | Hi people.. Anyone know how to send a certain H323 disconnect cause when I want to reject a call comming in via OH323? |
| 19:10.04 | *** part/#asterisk slak- (i=slak@ircdz.com) |
| 19:10.11 | supaigtr | Yes. about 20 ppl rolling around the shell station. |
| 19:10.40 | MattH | supaigtr, are you in indiana or someplace where they are rationing it? |
| 19:10.44 | supaigtr | NC |
| 19:11.02 | bzbw | anyone can shoot me a quick sample dialplan to make multiple phone taking the same call and whichever phone picks up the call will silence the other phone(ring)? |
| 19:11.10 | distortio | epablo: congestion cause 34? or something else? |
| 19:11.10 | supaigtr | We'ved called into SC, GA, and VA nobody has any left or the line is too big to get in now. |
| 19:11.18 | gordonjcp | heh |
| 19:11.25 | gordonjcp | petrol rationing is so 1970s |
| 19:11.57 | supaigtr | Theres not rationing when theres none to ration. There are over 30 station in this town and they are all out. |
| 19:12.07 | Assid | this sucks.. there is tons of bandwith |
| 19:12.14 | Assid | and the quality of voice is starting to get yucky |
| 19:12.17 | supaigtr | Glad I can ride my bike to work. UPS and DHL just told us they didn't expect to be delivering tommorow. |
| 19:12.31 | MattH | supaigtr, yeah I can ride to work too (thank God).. where are you? |
| 19:12.32 | gordonjcp | supaigtr: we had supply problems a few years ago when protesters blockaded refineries |
| 19:12.32 | *** join/#asterisk Craziman2 (n=Craziman@boromir.apid.com) |
| 19:12.40 | supaigtr | Western NC |
| 19:12.41 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 19:12.43 | epablo | distortio: something else. I really wan't to give 1 when someone dials wrong |
| 19:12.50 | *** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 19:12.54 | supaigtr | Downhill to work. Gotta work to get back up. |
| 19:12.59 | gordonjcp | supaigtr: so I filled up the car with 98 octane leaded, which no-one else wanted to buy, and filled up the van with veg oil |
| 19:13.22 | supaigtr | gordonjcp: you got one of those stills? |
| 19:14.25 | gordonjcp | supaigtr: nope, just straight veg oil |
| 19:14.35 | epablo | distortio: I send back 19, with just doing Hangup, but I need to give more detail |
| 19:14.36 | gordonjcp | I used to run one of the Citroëns on it too, but I sold it |
| 19:14.42 | *** join/#asterisk ManxPower (i=eric@66-90-222-45.dyn.grandenetworks.net) |
| 19:14.43 | gordonjcp | this was a Mercedes van |
| 19:15.00 | ManxPower | Yes, I am alive. |
| 19:15.10 | gordonjcp | hey ManxPower |
| 19:15.17 | generalhan | can some one help me out with this error please ? |
| 19:15.17 | generalhan | <PROTECTED> |
| 19:15.24 | supaigtr | Cool. I used to have a buddy that would fill up my 250 with it. WOrked great just smelt like I was a frydaddy going down the road. |
| 19:15.36 | gordonjcp | supaigtr: hmmm, you shouldn't get a smell off it at all |
| 19:15.48 | *** join/#asterisk PakiPenguin (n=uppal@unaffiliated/pakipenguin) |
| 19:15.51 | PakiPenguin | hello there |
| 19:15.57 | gordonjcp | I only got a faint "greasy food" smell when the engine was very cold |
| 19:16.23 | supaigtr | it was used/processed oil. The docs said it was normal. Yea its was when it was cold in the morning. It had lots more power and seems to get about 2mpg better miliage. |
| 19:16.58 | PakiPenguin | i'd like to know , how do i setup failover , like i have call going onto one provider , if all the channels at that gateway are busy , i'd like it to send call to the second gateway , how do i do that? |
| 19:17.10 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 19:17.29 | gordonjcp | supaigtr: yeah, I ran my Cit on waste veg oil, but all I did was strain the lumps out ;-) |
| 19:17.33 | supaigtr | pakipenguin: There is an example in the wiki on that. I think it uses vars and qualify. |
| 19:17.49 | gordonjcp | supaigtr: that's actually probably cleaner than most mineral diesel that you buy in the US |
| 19:18.02 | gordonjcp | European cars have to be heavily modified to even run on it |
| 19:18.23 | PakiPenguin | supaigtr: any pointers / search query? i've been tyring to find it on the wiki , all i could find was AppLCDial |
| 19:18.38 | supaigtr | gordonjcp: It was very clean compared to pump fuel. I really liked it but my buddy got tired of doing all the work. He needed a larger capacity system so he could produce it in volume. |
| 19:18.46 | gordonjcp | supaigtr: I didn't notice a difference in power, but I was running at a frankly unwise amount of boost |
| 19:19.28 | supaigtr | :) . I haven't tried it since I put on the 5" exhast and downpipe. Sounds like a jet going down the road. |
| 19:19.49 | gordonjcp | heh |
| 19:19.51 | dalabera | has anyone has succesfully configure an agi script to control the user hangup with the setcallback function?? |
| 19:20.16 | gordonjcp | this was a stock exhaust on a Citroen CX 25DTR, but the boost opened out to 22psi |
| 19:20.24 | gordonjcp | and the excess fuelling adjusted to suit |
| 19:20.34 | supaigtr | 22psi? Whoa |
| 19:20.37 | *** join/#asterisk mrfrenzy (i=mrfrenzy@1-1-9-15a.hka.j.bostream.se) |
| 19:21.24 | gordonjcp | yes, you read that right |
| 19:21.33 | gordonjcp | I backed it off to around 18, 19 or so |
| 19:21.39 | mrfrenzy | Hi. Can you point me to a list of X100P clones that are confirmed to work with asterisk? |
| 19:21.53 | gordonjcp | 'cos the excess fuel diaphram kept sticking open and the engine would run away |
| 19:22.21 | crash3m | gordonjcp: heh atleast yours ran away, mine only opens partially |
| 19:22.30 | gordonjcp | crash3m: sticky? |
| 19:22.38 | PakiPenguin | mrfrenzy: md3200 intel works , but get the real card , its neat! |
| 19:22.39 | crash3m | yeah, ancient diaphram |
| 19:22.40 | gordonjcp | ah |
| 19:22.43 | gordonjcp | what pump? |
| 19:22.56 | gordonjcp | using straight veg oil in anything other than Bosch is asking for it... |
| 19:22.58 | crash3m | just a carb on a motorcycle, no real pump other than the vacuum |
| 19:23.11 | *** join/#asterisk L|NUX (n=linux@202.5.145.14) |
| 19:23.26 | gordonjcp | crash3m: ah, this is the thing on a diesel injector pump that ramps up the fuelling as boost pressure increases |
| 19:23.36 | gordonjcp | you know how diesel engines don't have a mixture, as such? |
| 19:23.43 | gordonjcp | just always use excess air? |
| 19:23.45 | crash3m | yeah |
| 19:23.50 | crash3m | well, most always use excess air |
| 19:24.04 | crash3m | I've seen more than one diesal belch black smoke for miles |
| 19:24.08 | gordonjcp | yeah |
| 19:24.21 | gordonjcp | over-rich, possible blocked air filter |
| 19:24.30 | gordonjcp | or pratted about with by a monkey on crack |
| 19:24.36 | crash3m | heh |
| 19:24.51 | gordonjcp | I figured 220bhp at 3,200rpm from a 2.5 litre four-pot was good enough... |
| 19:25.08 | crash3m | sometimes its just given more gas than it can handle...was usually the case when driving an 800ci tractor with a turbo |
| 19:25.28 | crash3m | by 'gas' I just mean fuel |
| 19:26.18 | gordonjcp | crash3m: the turbo starts spinning on this at about 1200rpm |
| 19:26.25 | pygrammer | hello all :) |
| 19:26.35 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:26.40 | gordonjcp | by 1800rpm you've hit peak torque, about 300lb/ft |
| 19:26.44 | syle | i have a callcenter locally here that wants me to hookup 8 phones with VOIP, i keep thinking over and over again what is best way to do this... |
| 19:26.47 | gordonjcp | hey pygrammer |
| 19:26.54 | gordonjcp | syle: asterisk of course |
| 19:27.00 | pygrammer | gordonjcp: :) |
| 19:27.04 | pygrammer | haha, no duh :D |
| 19:27.11 | syle | i beleive she currently has 8 analog lines |
| 19:27.42 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe201.t-net.net.ve) |
| 19:27.49 | syle | i;d hate to charge her full cost for a T1 |
| 19:27.55 | syle | for only 8 people |
| 19:28.03 | trimi | err syke |
| 19:28.06 | trimi | err syle |
| 19:28.09 | trimi | she has a callcenter |
| 19:28.14 | syle | yeah |
| 19:28.15 | trimi | she dont need FXO |
| 19:28.20 | syle | only dials locally right now |
| 19:28.35 | trimi | ohhhh localy thats bad :) |
| 19:28.42 | trimi | i thought VOIP |
| 19:28.51 | syle | well i beleive she is switching completely to calling LD |
| 19:29.00 | trimi | where is she from ? |
| 19:29.07 | syle | canada |
| 19:29.12 | trimi | thats easy then |
| 19:29.25 | *** join/#asterisk werkenbass (i=werkenba@pc-14-79-86-200.cm.vtr.net) |
| 19:29.27 | trimi | get some unlimited plan to canada from VOIP providers |
| 19:29.36 | werkenbass | hola |
| 19:29.39 | trimi | install asterisk, get a switch and 8 IP phones |
| 19:29.46 | syle | from canada, calling the US |
| 19:29.54 | trimi | same thing |
| 19:29.59 | trimi | even better |
| 19:30.02 | werkenbass | hola |
| 19:30.04 | syle | whats cheaper 8 ip phones or 2 TDM400's |
| 19:30.09 | trimi | for 20@ you get unlimited calling to canada/us |
| 19:30.22 | trimi | check on google :) |
| 19:30.38 | syle | well my worry is about the T1 connection |
| 19:30.48 | trimi | but you dont need it |
| 19:30.52 | werkenbass | :) |
| 19:30.54 | werkenbass | :( |
| 19:30.56 | werkenbass | :'( |
| 19:31.04 | syle | how you gonna get 8 users out on the internet |
| 19:31.39 | werkenbass | :s |
| 19:31.39 | trimi | well 1st of all 8 rarely will make simultanious calls |
| 19:31.41 | werkenbass | :S |
| 19:31.48 | syle | telemarketing |
| 19:31.48 | *** part/#asterisk werkenbass (i=werkenba@pc-14-79-86-200.cm.vtr.net) |
| 19:31.49 | groogs | uhoh, i think my laptop p/s just actually died .. |
| 19:31.50 | *** join/#asterisk TheCops (n=mdb@206-248-136-187.dsl.teksavvy.com) |
| 19:31.50 | syle | of course lol |
| 19:31.50 | trimi | just get a 1 Mbit internet connection |
| 19:31.51 | TheCops | Hi |
| 19:32.00 | trimi | just get a 1 Mbit internet connection |
| 19:32.04 | trimi | and use GSM codec |
| 19:32.07 | trimi | or G729 |
| 19:32.13 | trimi | <PROTECTED> |
| 19:32.33 | trimi | in canada you can get 4 Mbit connection for like 60$ |
| 19:32.40 | TheCops | I just bought 2 snom 320, I'm trying to make working the MWI support, I see asterisk sending a SIP packet with "Voicemail 0/0 and no message" and the phone think that it have a message waiting...someone could help me ? |
| 19:32.43 | syle | outgoing? |
| 19:32.48 | Hmmhesays | hmm how hard would it to be to make a module for ser that returns $i+1 each time an invite comes in? |
| 19:32.53 | syle | i beleive thats just incomming |
| 19:33.20 | syle | i;ll have a talk with local providers |
| 19:33.38 | syle | see what is available for some decent outgoing bw per month |
| 19:33.51 | Assid | hrmm |
| 19:34.29 | groogs | trimi: i think its about to get a lot cheaper too |
| 19:34.41 | *** join/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net) |
| 19:34.42 | Assid | bandwith |
| 19:34.48 | Assid | man.. i wish it gets cheaper here |
| 19:34.49 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 19:34.50 | syle | i also know a local CLEC here so thats an option as well |
| 19:34.50 | *** join/#asterisk zed_home (n=knoppix@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk) |
| 19:34.51 | Assid | we pay quite a bit |
| 19:35.04 | Assid | 140kbps is like $30 |
| 19:35.09 | file[laptop] | Hmmhesays: depends |
| 19:35.16 | lathos42 | I'm hoping to renegotiate our T1 when our contract comes up.. we're paying $799/mo for it |
| 19:35.36 | file[laptop] | Hmmhesays: do you want it unique overall, or per forked process? |
| 19:35.36 | groogs | trimi: cybersurf.com apparently sells 5mbps for $25 and 7mbps for $35 (though i think its about $5 more if you don't get their long-distance bundle or something) |
| 19:36.06 | trimi | <groogs> thats 2 cheap |
| 19:36.15 | trimi | i pay 70$ for 384Kbps :( |
| 19:36.17 | glm2k | lathos42: what kinda T1 is that? |
| 19:36.19 | syle | i;m in winnipeg so i;m stuck to sprint, mts and global unfortunately |
| 19:36.21 | trimi | kbps |
| 19:36.42 | lathos42 | glm2k: Its a 1.5MB Frame Relay connection to SBC |
| 19:36.46 | trimi | im gonna cry now :( |
| 19:37.00 | lathos42 | err mb :) |
| 19:37.03 | glm2k | 800 bucks. wow. |
| 19:37.24 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 19:37.25 | lathos42 | We are a bit "rural" in relation to the CO |
| 19:37.28 | Assid | damn.. this router sucks |
| 19:37.40 | groogs | trimi: yikes. my connection (cable) is 10mbps down/1mbps up with 3 IPs (seperately capped) for $90/mo |
| 19:37.42 | groogs | cdn$ |
| 19:38.06 | *** part/#asterisk Craziman2 (n=Craziman@boromir.apid.com) |
| 19:38.15 | glm2k | groogs: bloody hell, that's cheap |
| 19:38.17 | trimi | <groogs> i hate cable |
| 19:38.18 | TheCops | I just bought 2 snom 320, I'm trying to make working the MWI support, I see asterisk sending a SIP packet with "Voicemail 0/0 and no message" and the phone think that it have a message waiting...someone could help me ? |
| 19:38.21 | groogs | and it's pretty close to that too, i've had three downloads going (each on different ips) at like 600kb/s each |
| 19:38.28 | trimi | DSL its better |
| 19:38.46 | glm2k | lathos42: sounds suspiciously like yahoo >:) |
| 19:38.54 | groogs | i didn't have much choice. no DSL here (despite sharing a parking lot with the local CO..) |
| 19:38.55 | Assid | does anyone have a FVS318 netgear router? |
| 19:39.04 | trimi | <groogs> what are you using you internet connection for |
| 19:39.07 | Assid | i cant find any documentation to allow traceroute/icmp packets |
| 19:39.07 | lathos42 | glm2k: How'd ya know? :) |
| 19:39.09 | trimi | are you using it to earn |
| 19:39.11 | Assid | ping works tho |
| 19:39.12 | trimi | or just for fun |
| 19:39.37 | groogs | and fiber was a tiny bit too expensive (compared to just having a 'regular' connection plus colo for a server) |
| 19:40.00 | lathos42 | They're much better than the local cable company, who wants to charge $29.95 for 512k down, 128k up |
| 19:40.09 | groogs | trimi: internet access for the company, voip for extra call capacity, and some backend and internal hosted stuff |
| 19:40.35 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 19:40.48 | trimi | groogs: nice, do you have your own VOIP company ? |
| 19:41.18 | trimi | Btw does any1 know good iac providers, i like iax workz gr8 under my nat, SIP sucks, any1 got a list of providers ? |
| 19:41.36 | glm2k | lathos42: it's a pretty good deal imho |
| 19:41.43 | groogs | no, we just use voip when our POTS lines are all used up (last number on our hunt group goes to our voip DID number) |
| 19:42.00 | groogs | i WAS using voip for all long-distance calls, but it wasnt reliable enough |
| 19:42.00 | Nugget | I c u r in irc n askn bout t iax 4 u'r srv n i thnk thz gd foak esh aekl. |
| 19:42.10 | trimi | Btw does any1 know good iac providers, i like iax workz gr8 under my nat, SIP sucks, any1 got a list of providers ? |
| 19:42.30 | Hmmhesays | SIP is fine under nat if your router isn't a pile of shiat |
| 19:42.38 | lathos42 | Nugget: That made my brain hurt |
| 19:42.44 | glm2k | jeez Nugget i feel like my cellphone has just taken over my screen. |
| 19:42.49 | trimi | <Hmmhesays> i use internet connection sharing no router |
| 19:43.01 | Hmmhesays | that blows |
| 19:43.13 | Hmmhesays | trimi: don't blame that on sip |
| 19:43.16 | Nugget | I think your internet connection is dropping characters, trimi. |
| 19:43.18 | file[laptop] | Hmmhesays: did thy see my responses to thy question of SER |
| 19:43.28 | Hmmhesays | no, I shall scroll up |
| 19:43.35 | trimi | i think i will install keiro firewall |
| 19:43.38 | nel | is it possible to use asterisk voicemail with an old PBX system like centrex? |
| 19:43.39 | trimi | they support SIP |
| 19:43.53 | groogs | trimi: what do you mean no router? you can't share it without a router |
| 19:44.07 | drray | groogs, his windows pc is acting like a router |
| 19:44.07 | groogs | trimi: or are you using windows "internet connection sharing" service ? |
| 19:44.11 | groogs | oh good god |
| 19:44.19 | groogs | here |
| 19:44.23 | groogs | you'll feel better |
| 19:44.28 | trimi | hehehe |
| 19:44.36 | groogs | seriously |
| 19:44.41 | trimi | dont blame me, things costs 2 much in my country |
| 19:44.42 | groogs | go spend $30 on a router |
| 19:44.47 | groogs | oh.. |
| 19:44.50 | trimi | its like 150$ here |
| 19:45.00 | groogs | where? |
| 19:45.03 | trimi | and the avarage payment its 140$ |
| 19:45.07 | trimi | Macedonia |
| 19:45.41 | trimi | when i bought a ciscko 775 ISDN router i payed 670$ 7years ago |
| 19:45.42 | *** join/#asterisk paulc (n=Paul@216.187.75.190.novuscom.net) |
| 19:45.54 | Hmmhesays | file[laptop], well what i'm looking to do is just basic round robin inviting to a group of asterisk boxes, if ($i==1){forward(host1)} elseif($i==2){forward(host2)} |
| 19:46.30 | file[laptop] | Hmmhesays: oh then per forked process is fine |
| 19:46.48 | Hmmhesays | basically a poor mans way of distributing invites |
| 19:47.23 | file | easy as pie |
| 19:47.44 | Hmmhesays | my C skills are bad though, so maybe you could give me a couple pointers |
| 19:48.08 | file | use an existing module, a basic one... and build off of that |
| 19:48.19 | Hmmhesays | yeah I just started looking at print |
| 19:48.37 | file | options is a basic one too |
| 19:48.42 | file | you can rip out most of it for a skeleton module |
| 19:49.02 | lesouvage | I want to add the dial prefix 9 to the callerid so I can use the number of the caller to call back. I tried SetVar(CALLERIDNUM=9+${CALLERIDNUM}) and other options. Can somebody please help me with this? |
| 19:49.21 | file | lesouvage: there's an application... called setcallerid |
| 19:49.30 | file | so you would do SetCallerID(9${CALLERIDNUM}) |
| 19:49.44 | lesouvage | file: thanks alot |
| 19:49.50 | Hmmhesays | options.c? |
| 19:49.55 | file | yeah |
| 19:50.11 | Hmmhesays | hrm I don't seem to have it in the source |
| 19:50.20 | file | it's a module |
| 19:50.20 | Assid | hrmm |
| 19:50.26 | file | it's in modules/mod_options |
| 19:50.27 | blitzrage | Set(CALLERID(num)=9${CALLERIDNUM}) should work... (untested) |
| 19:50.35 | Assid | can someone provide bandwith to my isp? |
| 19:50.40 | blitzrage | file: SetCallerID is deprecated! :D |
| 19:50.43 | file | yeah that's for 1.2... with functions (what blitzrage said) |
| 19:50.53 | file | blitzrage: hush you :P |
| 19:50.53 | Hmmhesays | yeah its not there |
| 19:50.57 | blitzrage | file: lol |
| 19:51.14 | file | Hmmhesays: I'll write you up a quick module if you buy me a beer |
| 19:51.16 | file | :P |
| 19:51.36 | Hmmhesays | this is stable off of iptels cvs though |
| 19:51.41 | Hmmhesays | if that makes any difference |
| 19:51.47 | file | what version? |
| 19:52.40 | tsume | hmm |
| 19:52.58 | Hmmhesays | version: ser 0.8.14-6 (i386/linux) |
| 19:53.03 | file | old |
| 19:53.06 | file | :) |
| 19:53.15 | Hmmhesays | yeah whatever stable was |
| 19:53.18 | Hmmhesays | what do you recommend |
| 19:53.25 | file | I usually grab a CVS snapshot |
| 19:53.39 | file | but I can whip something up for what you want |
| 19:54.01 | Hmmhesays | I can just go grab that options.c off the cvs and look at it |
| 19:54.07 | file | well sorta kinda yeah |
| 19:54.08 | lesouvage | file: it's working, thanks again. |
| 19:54.10 | file | or grab another basic module |
| 19:54.14 | Hmmhesays | unless you want to show me something more straight to the point |
| 19:54.22 | tsume | I just rebooted my friend's asterisk server, and started safe_asterisk. The phones aren't getting configs, and the operation panel isn't logging in. However all the polycom phones seem to be in a logged in mode since they can make ans recieve phone calls. I get a "cant get boot configuration error" on the polycoms, and the snom panel wont even login, its in a forbidden mode. Anyone want to give me a clue? |
| 19:54.34 | BrianR___ | Yay. I found a reproducable way of crashing the polycom ip500... |
| 19:54.37 | file | Hmmhesays: well I'm looking for another example |
| 19:54.40 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
| 19:54.46 | BrianR___ | Aparently it can't cope if an incoming call arrives at exactly the same time you're placing one. |
| 19:54.54 | tsume | server was working yesterday, but I rebooted because I thought the phone companies problem wasn't theirs at first :) |
| 19:55.00 | tsume | did I not start up something? |
| 19:55.12 | file | Hmmhesays: you're making me want to whip up a module now, DAMN YOU! |
| 19:55.14 | paulc | tsume: is the FTP server running |
| 19:55.21 | opus_ | you know what would be cool in asterisk... if there was an 'alias' function , for example alias SetCallerId { SET(CALLERID(${ARG1})=9${CALLERIDNUM}) |
| 19:55.43 | Hmmhesays | file: oh you know you want to, lol |
| 19:55.47 | tsume | paulc: out of curiosity, is asterisk the ftp server, or is there usually another? |
| 19:55.50 | file | Hmmhesays: bah |
| 19:55.55 | file | Hmmhesays: want it for 0.8.14? |
| 19:56.00 | tsume | paulc: its a udp based ftp right? |
| 19:56.08 | tsume | or is it a tcp one? |
| 19:56.15 | paulc | tsume: In most of my setups, asterisk and the FTP are on the same box. Check in the config on the Polycom which server it's pointing to for FTP? |
| 19:56.18 | file | Hmmhesays: if so, give me a snapshot and I'll write against that :) |
| 19:56.20 | Hmmhesays | i suppose there are large differences between that and the latest snapshot eh? |
| 19:56.27 | file | Hmmhesays: well it may be compatible |
| 19:56.29 | Hmmhesays | naw, I'll just upgrade |
| 19:56.32 | file | k |
| 19:56.33 | angler | paulc, whats up! |
| 19:56.35 | file | I'll grab the latest CVS |
| 19:56.36 | paulc | angler! :D |
| 19:56.44 | paulc | what up fool? :) |
| 19:56.46 | tsume | paulc: the ftp should be on the same box. My friend brags :) I'm still learning asterisk. THought I could handle his customer |
| 19:57.00 | angler | paulc, about to im you |
| 19:57.42 | tsume | paulc: this is stupid aster newbie question. What ftp do you use? |
| 19:57.57 | dudes | file - you busy? |
| 19:57.58 | paulc | tsume: fuck knows.. whatever was installed with my distro.. vsftpd maybe? |
| 19:58.03 | paulc | file's never busy ;-) |
| 19:58.07 | file | dudes: slightly |
| 19:58.10 | tsume | paulc: aha. okay. |
| 19:58.15 | file | what'cha need? |
| 19:58.20 | file | paulc: PFFT |
| 19:58.25 | paulc | file: you know it to be true :-p |
| 19:58.29 | tsume | paulc: I didn't know if it was a tcp ftp.I thought it would have used udp ftp |
| 19:58.33 | blitzmere | ok i have asterisk behind a 3com wifi router. it itself is behind the NAT. I have given it DMZ host. but cannot connect to it from ouside my router. does anyone know what I need to do to fix this |
| 19:58.35 | dudes | Still fight that proxy stuff ... so I grabbed some sip debug |
| 19:58.56 | paulc | blitzmere: the recommended way is not to put asterisk behind NAT ;) |
| 19:59.12 | tsume | erm |
| 19:59.18 | tsume | I think it does use tftp |
| 19:59.22 | blitzmere | ok i will see what i can do |
| 19:59.23 | tsume | and vsftpd |
| 19:59.42 | tsume | and neither are running, wth |
| 19:59.43 | mutilator | is there any way to change what cdr logs for src and dst in the dialplan? |
| 20:00.02 | paulc | tsume: Polycoms use FTP not TFTP.. by default.. |
| 20:00.09 | tsume | paulc: okay |
| 20:00.14 | paulc | tsume: what distribution? |
| 20:00.14 | *** join/#asterisk flashnet (n=root@200.61.65.203) |
| 20:00.15 | tsume | oh. heh |
| 20:00.19 | paulc | chkconfig --list | grep ftp |
| 20:00.27 | tsume | paulc: well knoppix. my friend used xinetd :P |
| 20:00.45 | tsume | paulc: both are on :( |
| 20:00.47 | tsume | oh dear |
| 20:01.03 | hardwire | blah |
| 20:01.06 | hardwire | anybody own a snom 360? |
| 20:01.10 | hardwire | speakerphone is so.. odd |
| 20:01.29 | tsume | oh.. shit.. |
| 20:01.35 | tsume | I changed the root password |
| 20:01.46 | paulc | tsume: can you FTP to the box from a PC? |
| 20:01.48 | tsume | why the hell did he make the phones login as root? :) |
| 20:01.55 | tsume | paulc: I just looked at the logs |
| 20:02.05 | tsume | paulc: he had all the phones log in as user root |
| 20:02.10 | tsume | paulc: I changed the root password |
| 20:02.15 | tsume | argh :P |
| 20:02.22 | paulc | a) your FTP shouldn't allow root logins.. b) your phones should definitely not be connecting as root.. c) your config's a mess! |
| 20:02.25 | lesouvage | hardwire: I have two snom 200 phones and I have the same problem. |
| 20:02.33 | tsume | paulc: my friend is an idiot |
| 20:02.50 | hardwire | lesouvage: fading speakerphone? |
| 20:02.50 | tsume | paulc: I'm a BSD devel, I know what security is, he doesn't |
| 20:02.54 | hardwire | I think their echocan is faulty |
| 20:02.55 | tsume | *growls* |
| 20:03.01 | lesouvage | hardwire: yes |
| 20:03.06 | hardwire | it would be nice to disable it |
| 20:03.19 | hardwire | or have it train itself on the room every once in a while |
| 20:03.30 | *** join/#asterisk jsandnes (i=jsandnes@blupp.no) |
| 20:03.42 | jsandnes | Hey! i got this strange error msg. Aug 31 21:56:34 NOTICE[114695]: chan_sip.c:6935 handle_request: Failed to authenticate user <sip:95723958@213.160.242.135:5060>;tag=84f2a0d5-13c4-431627aa-90ddeca-4b25 |
| 20:03.52 | jsandnes | this happends if any number tries to dial the SIP number i have |
| 20:03.53 | jsandnes | any ideas? |
| 20:03.57 | Delta34 | should you start a context with an Answer always? |
| 20:04.02 | tsume | paulc: thank you very much |
| 20:04.09 | tsume | paulc: now I'm kinda pissed at friend :) |
| 20:04.30 | tsume | paulc: time to install ruby, listen on ftp port, and see password :P |
| 20:04.53 | paulc | I'll leave you to play ;-) |
| 20:05.10 | jsandnes | anyone? |
| 20:05.16 | tsume | paulc: I'm just really pissed he set all of them to use root:P |
| 20:05.23 | tsume | paulc: I'm going to beat him senseless |
| 20:05.44 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
| 20:05.44 | Hmmhesays | jsandes: asterisk is dropping the call because it is coming from a source that is sending an invite without credentials |
| 20:05.55 | bjohnson | jsandnes: means the authentication isn't set up right |
| 20:06.12 | bjohnson | Delta34: no |
| 20:06.14 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
| 20:06.34 | *** join/#asterisk surfdue (n=surfdue@user-0c6t1g9.cable.mindspring.com) |
| 20:06.34 | bjohnson | tsume: use one of the phones |
| 20:06.38 | jsandnes | bjohnson: well, i cant see anything wrong with the register line |
| 20:06.42 | jsandnes | ..? |
| 20:06.46 | bjohnson | tsume: a kind of poetic justice |
| 20:06.56 | tsume | bjohnson: thats what I'm goin going to do |
| 20:07.02 | tsume | bjohnson: he has the phones passworded |
| 20:07.03 | bjohnson | jsandnes: that isn't from the register line |
| 20:07.13 | nel | is it possible to use asterisk as a voicemail system for an old pbx? |
| 20:07.13 | tsume | bjohnson: probably with the same password :) |
| 20:07.17 | mutilator | mm did anyone answe that |
| 20:07.18 | bjohnson | nel: yes |
| 20:07.20 | jsandnes | bjohnson: from where then? |
| 20:07.26 | nel | bjohnson: how? |
| 20:07.28 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
| 20:07.29 | Delta34 | i am still trying to figure out why I am getting this warning, WARNING[29115]: chan_sip.c:701 retrans_pkt: Maximum retries exceeded on call |
| 20:07.34 | tsume | ohhhh |
| 20:07.34 | mutilator | can you change cdr at all? specifically i want to change what cdr logs as the dst |
| 20:07.55 | *** join/#asterisk srt (n=nobody@gw0-cgn.reucon.net) |
| 20:07.59 | tsume | when is astricon? |
| 20:08.05 | tsume | and am I still able to register? |
| 20:08.06 | hardwire | too soon |
| 20:08.06 | bjohnson | mutilator: you can add certain notes |
| 20:08.13 | nel | bjohnson: I don't see how asterisk is going to know from a forwarded call to a zap channel the extension you are trying to leave voicemail to |
| 20:08.29 | bjohnson | nel: start by reading. it isn't a plug and play system |
| 20:08.47 | bjohnson | jsandnes: you said they were calling in |
| 20:09.00 | nel | bjohnson: is there any other place besides voip-info.org that you recommend? |
| 20:09.03 | file | Hmmhesays: check privmsg |
| 20:09.15 | jsandnes | insecure=very did it |
| 20:09.48 | bjohnson | nel: school of hard knocks |
| 20:10.18 | bjohnson | nel: you put the * in front of the pbx. * acts as the routing agent |
| 20:10.41 | nel | bjohnson : unfortunately I can't do that, it's a centrex system |
| 20:11.15 | bjohnson | not familiar with centrex |
| 20:11.22 | mutilator | bjohnson: other than that though? |
| 20:11.26 | bjohnson | are the phones analog? |
| 20:11.31 | mutilator | you can't actaully change a field? |
| 20:11.53 | bjohnson | mutilator: ascii manipulation with bash script on cron job? |
| 20:12.07 | mutilator | heh |
| 20:12.15 | bjohnson | mutilator: you can change some easily, others require code rewrites and recompile |
| 20:12.32 | mutilator | i need to change the dst |
| 20:12.52 | bjohnson | mutilator: most people doing it with reporting software instead of trying to change the cdr |
| 20:13.00 | Darwin35 | ? |
| 20:13.02 | mutilator | just lookin for a simple solution other than recoding a bunch of billing and log reviews crap |
| 20:13.13 | Darwin35 | opic |
| 20:13.22 | Darwin35 | ~topic |
| 20:13.45 | bjohnson | looking for speakers? |
| 20:13.48 | Darwin35 | ~ãstricon |
| 20:14.16 | Darwin35 | go to the web site in the topic |
| 20:14.19 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 20:14.29 | bjohnson | does one have to be interesting or knowledgeable? |
| 20:14.32 | Darwin35 | grrr this new kbd sucks |
| 20:14.52 | *** join/#asterisk L|NUX (n=linux@202.5.146.14) |
| 20:15.55 | DaPrivateer | hrm. anyone here use FWD? |
| 20:16.14 | bjohnson | for testing. it is recommended for all newbies |
| 20:17.00 | DaPrivateer | i am trying to use it via IAX for outbound 800 calling |
| 20:17.14 | DaPrivateer | it was working but now im getting "Call not approved" |
| 20:18.20 | FuzzyCat | fwd seems broke atm |
| 20:18.45 | DaPrivateer | ok |
| 20:18.51 | DaPrivateer | that makes me feel a little better |
| 20:19.02 | DaPrivateer | was worried my account got turned off |
| 20:20.00 | FuzzyCat | I'vebeen trying to use it all day... |
| 20:20.39 | DaPrivateer | curious, its the only provider i can find that does toll free calling in US for free (and uses IAX2). are you aware of any otheres? |
| 20:21.31 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 20:21.34 | tsume | who does toll free calling for free? |
| 20:21.41 | drray | payphone |
| 20:21.45 | FuzzyCat | mostppl |
| 20:21.46 | tsume | oh, heh |
| 20:22.00 | DaPrivateer | FWD does |
| 20:22.02 | Hmmhesays | its toll free to the caller, not the callee |
| 20:22.11 | DaPrivateer | well ya |
| 20:22.16 | *** join/#asterisk xxi (i=foobar@cpe-24-27-44-163.austin.res.rr.com) |
| 20:22.17 | DaPrivateer | i mean outgoing toll free via VoIP |
| 20:22.26 | Hmmhesays | fwd |
| 20:22.43 | DaPrivateer | the question was any others besides fwd |
| 20:22.57 | Hmmhesays | haha |
| 20:23.01 | Hmmhesays | well I wasn't watching |
| 20:23.52 | nel | does anybody know a command to see debug info from zap channels |
| 20:24.22 | eKo1 | is there a pri on those channels? |
| 20:24.37 | nel | just analog lines |
| 20:24.59 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfi6j.dialup.mindspring.com) |
| 20:25.00 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 20:25.04 | nel | centrex lines connected to zaptel card tdm400 I would like to see debug info i.e if a called was forwarded what was the original receiver of the call |
| 20:25.05 | eKo1 | Nope, not that I know of. |
| 20:27.02 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 20:30.13 | toddf | anyone know if iLBC video is supported in asterisk? |
| 20:34.18 | *** join/#asterisk TheCopss (n=mdb@206-248-136-187.dsl.teksavvy.com) |
| 20:34.29 | TheCopss | Hi |
| 20:34.33 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
| 20:34.49 | *** join/#asterisk [hC] (n=hardcore@8.10.2.40) |
| 20:35.03 | TheCopss | Someone have snom product ? I'm currently put the phone in DND mode and when I'm calling, it tell that is busy. Someone can help me ? |
| 20:35.50 | [hC] | I have an x100p clone in my box here, and a meetme room on that same box. Some people were in it last night, and were all complaining about audio cut-out. I believe all at the same time. Would this most likely be attributed to the fact that the timing source on that card is crappy? or..? |
| 20:36.11 | [hC] | Its an Ambient MD3200 based PCI modem |
| 20:36.59 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 20:37.20 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 20:37.31 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 20:37.34 | hugo-v6 | TheCopss: ill hopefully get mine ip 190 in two days |
| 20:38.08 | hugo-v6 | TheCopss: are u happe with it? (leds) call forwarding etc. |
| 20:38.40 | syle | hmm are you able to run a T1 line from some place in your city directly to your house |
| 20:39.01 | *** join/#asterisk audela (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk) |
| 20:40.03 | TheCopss | hugo-v6 ho yeah this is a fucking good phone |
| 20:40.12 | Hmmhesays | is fwd accepting 800 numbers for anyone else? |
| 20:40.14 | TheCopss | the best one I ever seen with asterisk |
| 20:41.00 | toddf | hmmmesays: 800 #'s check, 1800callatt is my test # ;-) |
| 20:41.20 | *** join/#asterisk Veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 20:41.23 | Hmmhesays | i just call 1800biteme2 thats att customer service |
| 20:41.25 | toddf | anybody ever play with Motorola Ojo & asterisk? |
| 20:41.56 | hugo-v6 | TheCopss: thats wat i wnated to hear :)=) |
| 20:41.58 | *** join/#asterisk voiper (n=unix@212.36.29.150) |
| 20:42.08 | DaPrivateer | heh |
| 20:42.14 | DaPrivateer | its rejecting everything from me right now |
| 20:42.18 | DaPrivateer | even in network calling |
| 20:42.42 | voiper | how i can add users in my asterisk |
| 20:42.44 | *** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:42.44 | *** mode/#asterisk [+o drumkilla_laptop] by ChanServ |
| 20:42.50 | Hmmhesays | echotest is working |
| 20:42.56 | voiper | sorry .. my englishh is bad :) |
| 20:43.05 | Hmmhesays | i'm 255718 |
| 20:43.27 | FuzzyCat | wow,that's old |
| 20:43.39 | Hmmhesays | not as old as my personal one |
| 20:43.45 | FuzzyCat | I betyouhave a walking frame |
| 20:43.52 | hugo-v6 | voiper: exactly u dont add users, but sip/iax phones. in in sip/extensions conf |
| 20:45.00 | *** part/#asterisk voiper (n=unix@212.36.29.150) |
| 20:45.06 | *** join/#asterisk voiper (n=unix@212.36.29.150) |
| 20:45.26 | Hmmhesays | 254084 is my personal number |
| 20:45.27 | *** join/#asterisk flashnet (n=root@200.61.65.203) |
| 20:45.41 | FuzzyCat | prrf 6 digits - pfft |
| 20:46.18 | voiper | ehooo |
| 20:46.23 | hugo-v6 | "mine is shorter than yours" contest? |
| 20:46.23 | ender | Can you use 501 firmware files for 301 phones? |
| 20:46.28 | ender | is it the same stuff? |
| 20:46.31 | hugo-v6 | ender: nope |
| 20:46.36 | ender | oh damn. |
| 20:46.41 | ender | I only found a source for 501 files. |
| 20:46.44 | *** join/#asterisk CoffeeIV (n=CoffeeIV@mail.airlinksystems.com) |
| 20:46.46 | voiper | hugo-v6 |
| 20:46.50 | voiper | friend |
| 20:47.06 | hugo-v6 | ender: afaik they have this shit on ther website, but really dont know |
| 20:47.11 | voiper | help me |
| 20:47.14 | ender | hugo-v6: is there a source on freedomphones for the 301s? |
| 20:47.16 | Chotaire | bbl |
| 20:47.17 | *** part/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 20:47.23 | hugo-v6 | voiper: with what? |
| 20:47.34 | hugo-v6 | ah ic |
| 20:47.45 | voiper | how add user(telephone number) in server |
| 20:47.47 | Hmmhesays | why does everyone on the phone look so happy, usually when i'm on the phone i'm not smiling |
| 20:47.52 | hugo-v6 | voiper: pls read the damn good documentation at voi-info.org |
| 20:47.59 | hugo-v6 | voip-info.org |
| 20:48.06 | hugo-v6 | ~docs |
| 20:48.07 | jbot | well, 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 |
| 20:48.12 | voiper | i read but a need help |
| 20:48.16 | voiper | little help |
| 20:48.28 | hugo-v6 | voiper: then ask a question |
| 20:48.33 | *** join/#asterisk sams (n=sams@pcp424683pcs.naugus01.ga.comcast.net) |
| 20:49.00 | voiper | ! abort add agi answer autoanswer database debug dial dont |
| 20:49.00 | voiper | dump extensions hangup help iax2 include init load local logger |
| 20:49.00 | voiper | mgcp no reload remove restart save send set show sip |
| 20:49.00 | voiper | skinny soft stop transfer unload |
| 20:49.00 | sams | anyone have mailboxes working with realtime? |
| 20:49.03 | hugo-v6 | argl |
| 20:49.06 | voiper | soory :( |
| 20:49.07 | pygrammer | has anyone used budgetones? |
| 20:49.08 | pygrammer | like the 101 or something |
| 20:49.09 | djin_ib | ~seen zoa |
| 20:49.38 | jbot | zoa <n=k@213.91.216.136> was last seen on IRC in channel #asterisk, 3d 7h 50m 34s ago, saying: ':)'. |
| 20:49.38 | djin_ib | !seen zoa |
| 20:49.38 | djin_ib | mm |
| 20:49.38 | voiper | with ??? |
| 20:49.46 | djin_ib | ah :) |
| 20:49.48 | voiper | with mikrotik |
| 20:50.02 | hugo-v6 | voiper: i dont get it. whats ur problem? |
| 20:50.02 | generalhan | pygrammer: i am using 15 of the new Aastra 9112i and they are cheap but stil rather nice for the pice |
| 20:50.17 | hugo-v6 | generalhan: dont know them, got a url? |
| 20:50.25 | voiper | it.s work but with asterisk , a don't know how use |
| 20:50.26 | CoffeeIV | pygrammer: I have used a Budgettone 100 |
| 20:50.27 | generalhan | go to voipsupply.com |
| 20:50.54 | voiper | i have cisco ata186 |
| 20:50.59 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 20:51.07 | voiper | config with sip client |
| 20:51.15 | generalhan | i NEED help with my Cisco 7960 please tell me some one has one up and running on asterisk !!!??? |
| 20:51.22 | hugo-v6 | voiper: dunno this device |
| 20:51.31 | CoffeeIV | I am trying to make it so that when someone calls a certain one of my incoming DIDs, it goes to a specific context |
| 20:51.46 | hugo-v6 | generalhan: iirc on voip-info there is a document about using it with asterisk |
| 20:51.49 | voiper | how add user in asterisk for cisco |
| 20:51.53 | sams | so no one (that is reading the channel right now) has voicemail working with realtime in *? |
| 20:51.54 | CoffeeIV | it works if that context is a local extension |
| 20:52.03 | generalhan | iirc ? |
| 20:52.15 | hugo-v6 | voiper: if its a sip-phone, then in sip.conf. read about it on voip-info.org |
| 20:52.19 | voiper | line 1 and line 2 on my cisco is 8555 and 8888 |
| 20:52.20 | CoffeeIV | but not if it is the auto-attendant I am trying to set up |
| 20:52.23 | djin_ib | voiper, read docs first |
| 20:52.23 | hugo-v6 | if i remmeber correctly |
| 20:52.28 | djin_ib | ~docs |
| 20:52.30 | jbot | extra, extra, read all about it, 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 |
| 20:52.38 | generalhan | hugo: i have been though a million different docs and i have followed them, but still no go i got the phone running but i just cant get it to register with the asterisk server |
| 20:52.40 | hugo-v6 | djin_ib: told him already :/ |
| 20:53.01 | djin_ib | yeah, in the same minute ;) |
| 20:53.04 | paulc | generalhan: what's up with your 7960? |
| 20:53.05 | hugo-v6 | generalhan: then i guess the setup on the phone isnt correct |
| 20:53.19 | generalhan | <PROTECTED> |
| 20:53.25 | generalhan | thats my problem with the phone |
| 20:53.26 | generalhan | lol |
| 20:53.41 | paulc | sounds like bad credentials - either on the phone or in sip.conf |
| 20:53.50 | *** join/#asterisk Berkey (n=berkey@las-cust-208.57.3.251.mpowercom.net) |
| 20:53.50 | paulc | do you have a [7333] entry in sip.conf? |
| 20:53.55 | generalhan | yep |
| 20:54.12 | paulc | configuring your phone from the TUI or from a config file? |
| 20:54.13 | generalhan | and i made it soooooo simple i made the name and the password and the context all 7333 so i couldnt mess it up and it still wont work |
| 20:54.48 | Berkey | I want some opinions. What is the best all around bussiness phone for use with an asterisk server? |
| 20:54.54 | paulc | Polycom IP600 |
| 20:55.00 | paulc | or the 500/501's not bad either |
| 20:55.04 | paulc | beats the cisco hands down |
| 20:55.06 | paulc | but that's just me :) |
| 20:55.16 | paulc | generalhan: you got user ID + auth ID set on the cisco? |
| 20:55.17 | Hmmhesays | crisco ain't bad |
| 20:55.39 | generalhan | paulc:yep |
| 20:55.43 | Berkey | Will the cisco's pretty lcd ever be put to good use by asterisk? |
| 20:55.45 | CoffeeIV | I have a "Goto(aa_default)" that makes this error in the log: "pbx.c: No such label 'aa_default' in extension '5126873302' in context 'from-trunk'" followed by "pbx.c: Priority 'aa_default' must be a number > 0, or valid label". what does that mean ? |
| 20:55.53 | generalhan | paulc: can i PM you |
| 20:56.02 | paulc | Berkey: I've got them set up to show ACD queue stats when you go in to services |
| 20:56.04 | paulc | generalhan sure |
| 20:56.23 | paulc | CoffeeIV: add a ",s,1" or similar to that |
| 20:56.28 | paulc | unless aa_default really IS a priority |
| 20:56.39 | paulc | as opposed to a context and/or extension |
| 20:56.46 | CoffeeIV | paulc: aa_default is a context -- I'll try "s,1" |
| 20:56.57 | hugo-v6 | well... /me goes a walk with his dog now. |
| 20:57.00 | hugo-v6 | l8r pals |
| 20:57.23 | Berkey | I got a snom 190 to test but it feels very cheap. I thought germans made better stuff. |
| 20:58.20 | CoffeeIV | paulc: that was exactly right -- thanks a million |
| 20:58.31 | pygrammer | sorry CoffeeIV and generalhan, was afk |
| 20:58.34 | hugo-v6 | Berkey: i get mine in two days. but i know the phones as isdn-phones. and they rock. |
| 20:58.35 | pygrammer | CoffeeIV: how is the budgetone 100 |
| 20:58.40 | pygrammer | generalhan: how much is the aastra |
| 20:58.48 | Assid | i need to find a hacked g729 |
| 20:58.50 | generalhan | $130 |
| 20:59.08 | Berkey | What do you think of the polycom IP 300's as basic set for the lobby? |
| 20:59.15 | Berkey | Will they hold up? |
| 20:59.17 | CoffeeIV | pygrammer: OK, I don't use it much. It was cheap and it worked |
| 20:59.18 | paulc | Good enough for use as a basic no frlls phone, yeah :) |
| 20:59.27 | pygrammer | the grandstream gxp-2000 looks ok too |
| 20:59.34 | Berkey | Cool thanks for the feedback |
| 20:59.34 | pygrammer | CoffeeIV: ah, what do you use primarily then? |
| 20:59.37 | paulc | the GXP-2000 is kinda tacky |
| 20:59.50 | BrianR___ | I just got a gxp-2000 here. |
| 20:59.50 | JerJer | Assid: just pay Digium |
| 20:59.55 | JerJer | very trivial |
| 21:00.04 | BrianR___ | A few things work _better_ on the GXP-2000 than the polycom ip500... |
| 21:00.07 | generalhan | paulc: did you get my message ? |
| 21:00.12 | pygrammer | paulc: really |
| 21:00.16 | pygrammer | BrianR___: that's interesting |
| 21:00.21 | pygrammer | i was considering the ip501 |
| 21:00.23 | pygrammer | but didn't want to spend $190 |
| 21:00.40 | pygrammer | the gxp-2000 looks ok |
| 21:00.40 | CoffeeIV | pygrammer: oh, my friend did try to hook up a plantronics wireless headset to it -- the kind where you hit a button and a little robot lifts the handset -- and it didn't work, there was just silence |
| 21:00.50 | BrianR___ | pygrammer: The GXP-2000's poe situation is better than the IP501's. Also, the display is backlit. |
| 21:00.51 | pygrammer | lol |
| 21:00.59 | pygrammer | BrianR___: yeah |
| 21:00.59 | paulc | generalhan: sure - did you get my reply? |
| 21:01.06 | pygrammer | k bbs |
| 21:01.12 | pygrammer | thanks all for your suggestions, will talk about it later |
| 21:01.14 | generalhan | lol |
| 21:01.15 | generalhan | nope |
| 21:01.28 | ender | has anybody gotten the sntp stuff working on the IP501s? |
| 21:01.41 | BrianR___ | GXP hangs on wall without a special bracket |
| 21:03.07 | generalhan | paulc: ok so for some reason im not getting your replies ... |
| 21:03.07 | BrianR___ | GXP has early dial |
| 21:03.08 | BrianR___ | GXP has GSM codec.. |
| 21:03.10 | BrianR___ | etc. etc. |
| 21:04.06 | paulc | <generalhan> ok so youre trying to help me out ... have you gotten this to work ? |
| 21:04.06 | paulc | <paulc> sure - I've got a bunch of 7960 and 7940s working with asterisk |
| 21:04.56 | generalhan | paulc: i just dont understand why it wont autherize. the phone is up, its showing the line # i programmed it with. yet im looking in my console and its just repeating that eroor over and over again |
| 21:04.56 | groogs | GXP-2000 vs SPA-841 ? |
| 21:05.55 | paulc | is it behind NAT? |
| 21:06.05 | sams | anyone know why my realtime voicemail might not be working even when I have a voicemail record in the voicemail table and have my extconfig.conf setup for it? |
| 21:06.28 | generalhan | paulc:no |
| 21:07.20 | generalhan | paulc: i can do a pastebin and show you what i have if you can tell me how to copy and paste from PuTTY lol cause i cant figure it out |
| 21:07.47 | sams | generalhan, just highlight and its automatically copied |
| 21:07.49 | paulc | generalhan: press CTRL-L to clear the screen.. then clear the scroll back buffer in the putty menu.. cat the config file.. then select all |
| 21:08.12 | ender | hugo-v6: I had to get the firmware emailed to me from voip-supply. I wonder what the license is on putting these up somewhere for download. |
| 21:08.20 | ender | like how does freedomphones get away w/ it? |
| 21:10.49 | Godsey | generalhan: copy from putty is just select |
| 21:11.13 | bzbw | anyone know * support SIP forking? |
| 21:11.30 | *** join/#asterisk Cresl1n (n=Cresl1n@digium.com) |
| 21:11.36 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa30w-156034035160.nb.aliant.net) |
| 21:11.44 | generalhan | paulc: http://pastebin.ca/21768 |
| 21:11.56 | generalhan | that is my SIP<mac address>.conf |
| 21:12.07 | generalhan | .cnf rather |
| 21:12.21 | hugo-v6 | ender: dunno also. havent to do anything with those polycom. because teh ip 300 sucks and i think im in love with snoms ;) |
| 21:12.54 | hugo-v6 | so. now im on a walk with my dog. really. l8r. |
| 21:13.39 | paulc | generalhan: try removing the quotes around everything? can't think if I did or not but it rings a bell.. |
| 21:13.47 | paulc | plus show me your sip.conf for that one peer |
| 21:13.53 | generalhan | ok |
| 21:16.03 | generalhan | http://pastebin.ca/21769 |
| 21:16.21 | generalhan | thats the sip.conf for 1 that works with a different phone and the cisco one that doesnt work |
| 21:18.55 | sams | file[laptop], got a min to help me with a realtime voicemail question? (you only one that seems to know realtime stuff) |
| 21:19.22 | file[laptop] | I know realtime stuff? uh oh |
| 21:19.35 | sams | well you answered one of my previous realtime questions :) |
| 21:19.39 | file[laptop] | did I? |
| 21:19.46 | file[laptop] | if you say so I guess I did |
| 21:19.48 | sams | ya |
| 21:20.13 | zedkatuf2 | Newb Q: is G723 = GSM or no? |
| 21:20.27 | JerJer | um no |
| 21:20.36 | *** join/#asterisk Defraz (n=t0tal@tim.ibccom.net) |
| 21:20.43 | *** join/#asterisk Tabo (n=acura0@s142-179-67-118.bc.hsia.telus.net) |
| 21:21.45 | sams | file[laptop], so anyway, got voicemail.conf in a conf table in mysql without any mailboxes, just some global settings, and I have a voicemail table for the realtime mailboxes with an entry for my extension... but I cant seem to get it working at all, it doesnt even record voicemails or let me log into the voicemailmain |
| 21:21.50 | L|NUX | can some one help me with dundi ? |
| 21:21.53 | *** join/#asterisk GaryH (n=root@gromit.garysoft.co.uk) |
| 21:22.01 | L|NUX | ? |
| 21:22.08 | generalhan | paulc: unless you found something in sip.conf i dont know what to do, i took all the quotes out and still no go same error |
| 21:22.34 | L|NUX | www*CLI> dundi lookup 17806691185 |
| 21:22.35 | L|NUX | DUNDi lookup returned no results. |
| 21:22.35 | L|NUX | DUNDi lookup completed in 0 ms |
| 21:22.37 | L|NUX | :( |
| 21:22.37 | Piranha- | anyone know of a good voip provider they'd recommend? |
| 21:22.41 | sams | file[laptop], any ideas as to why that might be happening? :) |
| 21:22.46 | generalhan | VoicePulse |
| 21:22.48 | paulc | generalhan: got distracted |
| 21:22.49 | L|NUX | Piranha- : nufone |
| 21:22.51 | tzanger | Piranha-: nufone has my vote. asterlink and unlimitel.ca too |
| 21:22.58 | paulc | I'd lose the regexten maybe.. and the username.. keep it as simple as possible |
| 21:23.10 | Piranha- | broadvoice ive been using recently (got a free month) and really dont like it |
| 21:23.18 | tzanger | Stay away from sixtel |
| 21:23.23 | tzanger | which is also iax.cc |
| 21:23.54 | generalhan | paulc: if i lose the username how does the phone registar with asterisk |
| 21:24.03 | paulc | using [7333] - the name of the peer |
| 21:24.03 | zedkatuf2 | Pirhana: sipgate.co.uk good so far |
| 21:24.16 | paulc | I'd also try registering with a different phone, or using x-lite.. |
| 21:24.20 | generalhan | ohh it will register with the context name >? |
| 21:24.24 | paulc | did you try stripping the quotes out of the cisco config file? |
| 21:24.29 | generalhan | yep |
| 21:24.54 | Umaro | ok guys, what if I wanted to use uhci for ztdummy on 2.6? |
| 21:24.58 | Umaro | instead of RTC |
| 21:25.00 | paulc | I'll brb - on a call |
| 21:25.04 | paulc | msg me if you need more |
| 21:25.18 | L|NUX | any one helpfull with dundi ? |
| 21:25.19 | L|NUX | :( |
| 21:26.03 | *** join/#asterisk werkenbass (n=werkenba@pc-14-79-86-200.cm.vtr.net) |
| 21:26.09 | file[laptop] | sams: not really |
| 21:26.16 | werkenbass | Hello |
| 21:26.51 | *** part/#asterisk werkenbass (n=werkenba@pc-14-79-86-200.cm.vtr.net) |
| 21:27.34 | sams | file[laptop]: well that's no help |
| 21:28.18 | Piranha- | does voicepulse allow byod? |
| 21:28.29 | sams | Piranha: kinda |
| 21:28.57 | Piranha- | I know one provider (inter.net) dont.. but you can get the box, unlock it, and get the login/password from it |
| 21:28.59 | sams | Piranha: you can get the voicepulse Connect! service which is SIP/IAX and BYOD, buts per minute charges on outgoing calls |
| 21:29.02 | zedkatuf2 | newb Q: (OK, I'm being lazy :P): allow=g723 in sip.conf is ok, yes? (ie codec is written like this) |
| 21:29.02 | Piranha- | then use it on asterisk |
| 21:30.03 | Piranha- | but they use the azacall box which is retarded |
| 21:30.12 | Piranha- | huge, featureless, and ugly |
| 21:30.12 | sams | Piranha: you can order the standard SIP monthly service from voicepulse and pay for the device, and still use asterisk with the login info. pretty sure they will provide you with the info instead of having to get it off the box |
| 21:30.40 | Piranha- | yeh but I got sipuras right now |
| 21:30.47 | Piranha- | no need for a device |
| 21:30.56 | file | zedkatuf: g723.1 |
| 21:31.06 | sams | voicepulse will make you pay for the device... unless you get their voicepulse connect service |
| 21:31.08 | *** join/#asterisk iris_ (n=ewex@c-65-96-5-8.hsd1.ma.comcast.net) |
| 21:31.17 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net) |
| 21:31.22 | Piranha- | ah |
| 21:31.23 | *** join/#asterisk joe (n=nnnjsaue@ip66-107-33-195.z33-107-66.customer.algx.net) |
| 21:31.41 | *** join/#asterisk tbone (n=bdixon@adsl-158-23-124.asm.bellsouth.net) |
| 21:31.42 | sams | Piranha: I have their connect service right now and its pretty good |
| 21:31.49 | zedkatuf2 | file: tnx...it's not case-sensitive is it? |
| 21:31.58 | sams | not sure if its the best value or not, but the latency and sound quality is great |
| 21:31.59 | Piranha- | well im up in canada.. dont know how the delay is |
| 21:32.07 | file | zedkatuf2: lowercase it... |
| 21:32.08 | Piranha- | braodvoice's isnt very good |
| 21:32.10 | iris_ | anyone mind helping out a newbie? http://forums.digium.com/viewtopic.php?t=1137 |
| 21:32.15 | Piranha- | always interrupting people |
| 21:32.15 | sams | yeah I wasn't happy with broadvoice |
| 21:32.18 | tzanger | digium has forums??! |
| 21:32.36 | zedkatuf2 | file: ta.....I'll hopefully ask some stimulating questions at some point down the road :P |
| 21:32.41 | iris_ | piranha broadvoice tends to have alot of packetloss inside the gnaps network |
| 21:32.43 | file | zedkatuf2: hehe |
| 21:32.49 | sams | Piranha: try the connect service it only costs $20 to start and you never pay another penny until you use up your first minutes of your $20 |
| 21:33.03 | iris_ | and when i called then on it they didnt know what packetloss was nor did any of the engineers care to call me back |
| 21:33.10 | Piranha- | hahah |
| 21:33.18 | Piranha- | I was talking to this one guy there who uses asterisk |
| 21:33.26 | Nugget | web forums are the cesspools of the internet. |
| 21:33.26 | *** join/#asterisk Defraz (n=t0tal@tim.ibccom.net) |
| 21:33.28 | Piranha- | when I was trying to get it working... |
| 21:33.31 | iris_ | not trying to badmouth them, just relaying my experience. |
| 21:33.52 | Piranha- | yea |
| 21:33.55 | iris_ | nugget i figured it would be better then rehashing my problems here ;) |
| 21:34.13 | sams | their support was pretty good for me, but their service sucks |
| 21:34.23 | iris_ | man using IRC again is odd, it has to be at least 7 years since i was last on |
| 21:34.28 | sams | broadvoice has really bad latency during peak times |
| 21:34.38 | sams | same here iris :) |
| 21:34.40 | Piranha- | yeah I notice that |
| 21:35.11 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 21:35.18 | iris_ | and hell i know the owner of GNAPS/Broadvoice and i still say there service sucks. |
| 21:36.45 | sivana | Broavoice = Global NAPs? |
| 21:36.56 | iris_ | yes |
| 21:36.58 | syzygyBSD | I am trying to setup voicemail to run a script when a new voicemail is saved using externnotify in voicemail.conf. Is there a way to identify the call that left the voicemail? |
| 21:37.06 | iris_ | same owner and BV uses GNAPs network |
| 21:37.14 | Delta34 | how do u end a sip channel manually? |
| 21:37.17 | Piranha- | the nice thing about broadvoice though, is the included free country calling ... |
| 21:37.21 | sivana | i c |
| 21:38.14 | syzygyBSD | it says the 3 arguments that are passed are context, extension, and number of voicemail. I was hoping extension was from the context not the boxnumber, but alas it is not |
| 21:38.49 | syzygyBSD | Delta34: does soft hangup work? |
| 21:39.11 | *** join/#asterisk Katty (n=katrina@68.112.15.110) |
| 21:39.14 | CoffeeIV | What's a good way to record a greeting for * if I am not using AMP ? |
| 21:39.23 | Delta34 | nope softhang only shows active channels |
| 21:39.32 | Delta34 | when i do sip show channels |
| 21:39.36 | tzanger | CoffeeIV: exten => **1,1,Record(filename.gsm) |
| 21:39.45 | Delta34 | i have a bunch of sip channels still going |
| 21:39.48 | blitzrage | I'm trying to determine if theres something I need to change in my php config files to allow arguments passed to a PHP script from the console to actually contain something in $argv[x] ? I keep getting blank values.... |
| 21:39.53 | CoffeeIV | tzanger: thanks |
| 21:40.22 | tzanger | blitzrage: yeah, don't use PHP. :-p |
| 21:40.22 | blitzrage | same script on another box echo's back the arguments to me... but this one box that I need the script to run on, just returns blanks |
| 21:40.27 | blitzrage | tzanger: not an option :) |
| 21:40.36 | blitzrage | tzanger: oh... and you're a smartass |
| 21:40.38 | blitzrage | :) |
| 21:41.12 | tzanger | blitzrage: :-) |
| 21:41.41 | Umaro | can anyone tell me what I need to do to get ztdummy on 2.6 using uhci instead of rtc? |
| 21:41.51 | blitzrage | damn this is annoying... the one box I need the script to run on, and I can't get it to accept arguments |
| 21:45.16 | sams | Umaro: not sure if its possible... why cant you use the rtc? |
| 21:46.54 | Umaro | sams: ia64 |
| 21:48.10 | sams | rtc doesn't with with 64 bit cpu's? |
| 21:48.20 | sams | work* with |
| 21:48.28 | iris_ | an suggestions on settion up call/transfer confirmation? |
| 21:49.00 | Umaro | sams: apparently not |
| 21:49.03 | groogs | blitzrage: what does php -v say? |
| 21:49.10 | blitzrage | damn you PHP and your lack of register_globals! |
| 21:49.15 | blitzrage | groogs: I got it now :) |
| 21:49.27 | sams | iris, I saw something for it yesterday, let me see if I can find the url |
| 21:49.41 | iris_ | sams i appreciate it. |
| 21:49.46 | blitzrage | groogs: version 4.4.0 :) |
| 21:53.13 | tzanger | any word on maxpower yet? |
| 21:53.56 | syzygyBSD | in extensions.conf how do you catch a -1 return value and process based on it? |
| 21:54.08 | sams | iris_: http://www.voip-info.org/wiki-Asterisk+Tips+follow+me |
| 21:54.34 | sams | look at the dial(${TRUNK}c/...,20,r) cmd |
| 21:54.49 | [hC] | k |
| 21:55.17 | groogs | blitzrage: CLI version always populates $argv IIRC |
| 21:55.33 | blitzrage | groogs: yah, seemed I had register_globals turned off |
| 21:55.38 | groogs | blitzrage: CGI version, which a lot of people run for some stupid reason doesn't and requires register_globals |
| 21:55.42 | sams | iris_: the c in there causes the remote end to press the # key to accept the call, or else it will continue on and give the call back to your phone, or maybe voicemail, whatever you want |
| 21:55.45 | blitzrage | groogs: could have swore I checked that previously |
| 21:55.56 | iris_ | sams, i saw that before and later found something saying that was for ZAP interfaces only....? |
| 21:55.57 | groogs | i think theres a couple distros that use the cgi version instead of cli at /usr/bin/php |
| 21:55.59 | JerJer | famous last words |
| 21:56.14 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
| 21:56.18 | blitzrage | JerJer: ? |
| 21:56.21 | JerJer | gee sarge, I swore i put the pin back in |
| 21:56.23 | sams | iris_: not sure, I have not actually done it |
| 21:56.27 | blitzrage | JerJer: lol |
| 21:56.34 | iris_ | sams, ok thanks ill give it a try again |
| 21:58.38 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 21:59.23 | JerJer | Reminder: When the sun goes down tonight, step outside and look west. Do you see a pair of lights beaming through the rosy glow of sunset? That would be Venus and Jupiter: the two bright planets are converging for a beautiful close encounter. They'll be side-by-side for the next few days. |
| 21:59.32 | bzbw | hi, anyone know how to make 2 sip phone register to * with the same account, so that when a call comes in, both of them will ring? |
| 21:59.34 | JerJer | --spaceweather.com |
| 21:59.44 | toddf | bzbw: you don't want that |
| 21:59.50 | toddf | just say in the extension to dial |
| 21:59.53 | blitzrage | JerJer: schweeet - I'm going to take my camera and tripod out tonight then |
| 22:00.00 | toddf | Dial(SIP/phone1&SIP/phone2) |
| 22:00.00 | bzbw | toddf: why? |
| 22:00.16 | toddf | because it is so much more flexible to register two separate extensions |
| 22:00.23 | toddf | ;-) |
| 22:00.24 | JerJer | blitzrage: i took some black and white of the metor shower a couple weeks ago |
| 22:00.30 | blitzrage | JerJer: tasty |
| 22:00.31 | jeffik | bzbw: use call group for that |
| 22:00.34 | blitzrage | JerJer: have them online? |
| 22:00.36 | toddf | you can keep track of which extension made which calls |
| 22:00.46 | toddf | bzbw: listen to jeff, I've not setup call groups yet .. |
| 22:00.55 | groogs | JerJer: you mean beaming through the reminans of Hurricane Katrina ? :p |
| 22:00.55 | blitzrage | I took some cool pictures while out biking yesterday |
| 22:01.04 | JerJer | got two nice shots on an extended exposure shot |
| 22:01.07 | bzbw | thanks. but can you silence the rest of phone when one phone pick up the call? |
| 22:01.20 | jeffik | bzbw: i mean ring gorups |
| 22:01.31 | toddf | bzbw: why would you want that? ;-) |
| 22:01.36 | JerJer | i centered on the north star and focused out to infinity - then bracketed the exposures. |
| 22:01.47 | blitzrage | JerJer: I need to go take some cool pictures out at my parents house which is in the middle of no where - you can see so many stars since there's so little city light where they live |
| 22:01.49 | jeffik | bzbw: no all the phones will ring until one is answered or one goes to voice mail |
| 22:02.07 | JerJer | don't have a flatbed scanner that works, so no onlineage - need to get an all-in-one crap thing i guess :( |
| 22:02.21 | JerJer | blitzrage: it is like that where i live |
| 22:02.40 | JerJer | i head up the road about 5 miles to nothing but overgrown farmland |
| 22:02.44 | bzbw | jeffik: we have a team of 5 need to support some important customer, and the call has to be pick up when it comes in, but if one of them pick up, it show allow other user to continue their work. |
| 22:02.56 | blitzrage | JerJer: yah, thats sweet |
| 22:03.05 | blitzrage | JerJer: can get some good pictures that way |
| 22:03.15 | bzbw | I thought this is a SIP feature called "forking" |
| 22:03.28 | blitzrage | sweet - my externpass script worked first try! |
| 22:03.32 | hardwire | your a busy little beee |
| 22:04.00 | bzbw | jeffik: can call group to ring all the members in the group? |
| 22:06.42 | groogs | bzbw: that is what Dial(SIP123&SIP/234) does |
| 22:07.08 | groogs | think about it, why would it continue to ring when someone answers? |
| 22:07.08 | bzbw | toddf and groogs: thanks! |
| 22:07.33 | groogs | though, that could be fun i guess. send them to a context that does Background(ha-ha-no-one-is-there) |
| 22:09.46 | generalhan | anyone here successfully made a cisco 7960 work with asterisk ?? |
| 22:10.09 | eKo1 | many |
| 22:10.24 | generalhan | eKo1: youve made many work ? |
| 22:10.26 | blitzrage | lots of times |
| 22:10.33 | toddf | many people have made it work lots of times |
| 22:10.37 | toddf | flash it with sip, be happy |
| 22:10.40 | generalhan | i need some serious help |
| 22:10.45 | eKo1 | i'm not one of those many |
| 22:10.55 | generalhan | i have already flashed with the SIP configs. but i cant get it to register with asterisk |
| 22:11.15 | generalhan | i keep getting a handle_request error |
| 22:11.31 | generalhan | no one can figure out why |
| 22:11.34 | ender | hugo-v6: it seems that the 501 and 301 bootrom and sip roms are the same. |
| 22:18.47 | Katty | ~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~hmm |
| 22:18.58 | eKo1 | ehh |
| 22:19.01 | Katty | oooh, hiccup. |
| 22:19.40 | generalhan | some one please help me with my stupid 7960 ! lol. why wont it work ! |
| 22:19.44 | generalhan | this is killing me |
| 22:19.53 | *** join/#asterisk jeffgus_ (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 22:19.55 | *** join/#asterisk yartelecom (n=no-email@62.33.182.162) |
| 22:19.56 | eKo1 | then throw it away |
| 22:20.08 | eKo1 | that last thing i need is someone dying from cisco hardware |
| 22:20.08 | Uberbot | Because it's Cisco? |
| 22:20.25 | eKo1 | ...again |
| 22:20.27 | generalhan | EKo1: when i buy 15 phones at $300/ea i cant just throw them away |
| 22:20.50 | eKo1 | ebay? |
| 22:20.51 | Uberbot | You can give them to me. I'll dispose of them. :-D |
| 22:20.58 | Uberbot | What are the symptoms? |
| 22:20.59 | generalhan | knock cisco all you want ... when these phones were SCCP they ROCKED ! i just need them to rock in SIP too |
| 22:21.13 | generalhan | they wont register with asterisk |
| 22:21.23 | generalhan | chan_sip.c:7733 handle_request: Registration from '<sip:guy@192.168.0.46;user=phone>' failed for '192.168.0.59' |
| 22:21.56 | Uberbot | Are they set for host=dynamic in sip.conf? |
| 22:22.01 | generalhan | yep |
| 22:22.05 | Uberbot | My Pingtel had the problem. |
| 22:22.13 | Uberbot | And the username/password is right.....? |
| 22:22.16 | generalhan | yep |
| 22:22.27 | CoffeeIV | I want to set up some call groups -- I'm having trouble finding example on voip-info.org. Does anyone have a link ? |
| 22:22.27 | generalhan | i made it SUPER easy so that i couldnt mess it up |
| 22:22.46 | Uberbot | What is "guy?" |
| 22:22.51 | generalhan | the name |
| 22:22.59 | generalhan | the SIP context |
| 22:23.05 | Uberbot | Ok. |
| 22:23.21 | Uberbot | And in sip.conf, that phone lands in the guy context? |
| 22:23.28 | generalhan | yes |
| 22:23.34 | Piranha- | is packet8 byod? |
| 22:23.34 | generalhan | are you familar with the cisco config files ? |
| 22:23.43 | Uberbot | Yes. |
| 22:23.47 | generalhan | ill show you the paste bin if that will help you |
| 22:23.55 | toddf | Piranha: p8 is not byod |
| 22:24.04 | Uberbot | Do you have a softphone you can test with? |
| 22:24.24 | toddf | voxee.com for outbound calling is ... I like it .. I use inbound libretel.com .. but there are many others for DID and outbound ... |
| 22:24.25 | generalhan | yea the softphone logs right in |
| 22:24.43 | Uberbot | Ok, can you use the softphone and call the Cisco directly? |
| 22:24.49 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 22:24.51 | generalhan | nope |
| 22:24.51 | Uberbot | Without using * |
| 22:25.17 | generalhan | i can use the softphone to log into the asterisk server using the phone sip stuff in the guy context |
| 22:25.22 | generalhan | but the 7960 cant |
| 22:25.35 | Uberbot | ...and this has NEVER worked, right? |
| 22:25.41 | generalhan | NEVER |
| 22:25.57 | generalhan | its the oddest thing though |
| 22:25.57 | Uberbot | Have you turned sip debugging on for that IP? |
| 22:26.02 | generalhan | no |
| 22:26.06 | generalhan | how do i go about doing that |
| 22:26.09 | Uberbot | I'd try that. |
| 22:26.19 | Uberbot | sip debug ip 192.168.1.1..... |
| 22:26.29 | generalhan | take a look at http://pastebin.ca/21768 and see if maybe you can tell something is off |
| 22:26.39 | Uberbot | Then reboot the cisco. |
| 22:26.43 | Uberbot | Looking.... |
| 22:27.42 | *** join/#asterisk eivindtr (n=eivindtr@062016241059.customer.alfanett.no) |
| 22:27.44 | Uberbot | I don't seem to be able to load that page. |
| 22:27.48 | generalhan | crap |
| 22:27.49 | ender | Can you boost the audio of polycom phones serverside? |
| 22:28.10 | hardwire | ender: using AGI you could. |
| 22:28.19 | hardwire | EAGI |
| 22:28.19 | Uberbot | I'd get on the * console and look at the debug info..... |
| 22:28.34 | generalhan | the phone is rebooting now |
| 22:28.44 | hardwire | ender: it would be neat to have volume normalization ;) |
| 22:28.48 | Uberbot | So you see it try to register? |
| 22:28.58 | generalhan | not yet |
| 22:29.07 | generalhan | the phone takes FOREVER to reboot now that its SIP |
| 22:29.31 | generalhan | try this one http://pastebin.ca/21775 |
| 22:29.31 | Delta34 | does someone know how to have * answer call, then take a 10 digit dtmf then make another call from that 10 digit dtmf? |
| 22:29.48 | Katty | ... |
| 22:29.48 | hardwire | can you make another call? |
| 22:29.49 | Uberbot | I just took a 7960 home from work... to test with... :-D |
| 22:29.55 | Katty | i see. |
| 22:29.59 | hardwire | Uberbot: yay :) |
| 22:30.05 | pfn | anyone want to give me a screen from a dead 7960? |
| 22:30.05 | Uberbot | Oh? |
| 22:30.06 | hardwire | I keep taking phones home as well |
| 22:30.08 | pfn | my lcd is broken :) |
| 22:30.12 | hardwire | they keep taying however. |
| 22:30.53 | Uberbot | Wow, I thought the Pingtel took long to boot! |
| 22:31.07 | generalhan | lol |
| 22:31.14 | bzbw | Hi, can someone tell me how to CVS check 1.20 beta(i hate to use ftp)? |
| 22:31.20 | generalhan | well it just booted and i dont know what to make of this debug code |
| 22:31.26 | hardwire | yay |
| 22:31.34 | hardwire | going to get a dedicated 512kbps sync sat link soon. |
| 22:31.42 | hardwire | that should make my life wonderfull |
| 22:31.45 | Umaro | anyone here running asterisk on ia64? |
| 22:31.51 | Uberbot | Toward the top, you will see the response from the * server for the registration request..... |
| 22:31.53 | generalhan | Uberbot: ill pastebin it right now maybe you can tell me whats going on |
| 22:31.53 | hardwire | the advent of asterisk 1.0.2 with juitter+trunk will make it much better as well |
| 22:32.04 | generalhan | http://pastebin.ca/21777 |
| 22:32.09 | hardwire | Umaro: 8000hz on a 64 bit system |
| 22:32.15 | hardwire | just seems wrong |
| 22:32.17 | Uberbot | I got the previous one.... looking at both. |
| 22:32.20 | generalhan | k |
| 22:32.29 | generalhan | i hope you can make something of it cause im WAY confused |
| 22:32.31 | Umaro | hardwire: lol |
| 22:32.52 | bzbw | I don't want to cvs checkout the most latest firmware but 1.20BETA, anyone know the command? |
| 22:32.58 | *** join/#asterisk dsalama (n=dsalama@200.62.6.122) |
| 22:33.58 | *** part/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca) |
| 22:34.15 | Uberbot | Your username, password, and displayname are all the same 4 digit number.... |
| 22:34.27 | generalhan | yep |
| 22:34.32 | generalhan | i told you that i made it SUPER easy |
| 22:34.42 | Uberbot | What do you have for it in sip.conf? |
| 22:34.50 | Uberbot | Yes, you did. :-D |
| 22:34.55 | generalhan | lol |
| 22:34.58 | generalhan | same |
| 22:35.30 | Uberbot | SIP/2.0 403 Forbidden |
| 22:35.57 | Uberbot | 7xxx@192.168.0.46;user=phone>;tag=as52db2f27 |
| 22:36.04 | generalhan | what does that mean ? |
| 22:36.10 | Uberbot | Curious where the user=phone is coming from.... |
| 22:36.15 | generalhan | me too |
| 22:36.34 | generalhan | and it also says something in there about not being able to find the default context ... i have NEVER called on a default context |
| 22:36.58 | Uberbot | From the cisco config: user_info: phone |
| 22:37.26 | Uberbot | Change that to your extension number....? |
| 22:37.27 | generalhan | WHA ...... |
| 22:38.00 | Uberbot | Can you paste the paragraph from your sip.conf? |
| 22:39.34 | generalhan | yea |
| 22:40.05 | Uberbot | Is it working? |
| 22:40.06 | generalhan | uh ohj |
| 22:40.08 | generalhan | no |
| 22:40.17 | generalhan | but i gave you the wrong config file at first |
| 22:40.23 | Uberbot | uh oh doesn't sound good. |
| 22:40.24 | generalhan | so its gonna look like my sip.conf is wrong |
| 22:40.36 | Uberbot | Doh! :-O |
| 22:40.38 | generalhan | ill give you the sip.conf now and then re paste the correct config filel |
| 22:40.48 | Uberbot | K. |
| 22:41.46 | generalhan | http://pastebin.ca/21779 |
| 22:42.18 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-59-175.cybersurf.com) |
| 22:42.29 | pygrammer | <PROTECTED> |
| 22:42.33 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 22:42.44 | pygrammer | no! |
| 22:42.44 | Delta34 | is it possible to send dtmf and store it as a variable? |
| 22:43.05 | pygrammer | so the gxp-2000 is ok? |
| 22:43.13 | pygrammer | it's cheap as hell |
| 22:43.50 | bjohnson | Delta34: yes |
| 22:43.56 | Uberbot | ... and the correct Cisco config? |
| 22:44.08 | bjohnson | Delta34: but not directly |
| 22:44.25 | Delta34 | can you give me an example of how to do it |
| 22:45.01 | generalhan | http://pastebin.ca/21782 |
| 22:45.33 | Delta34 | I tried SendDTMF(Setvar=XXX) |
| 22:45.43 | generalhan | Uberbot: thats the old one AGAIN. sorry just imagine that the "cisco" is really "7333" |
| 22:45.59 | Uberbot | Was just going to mention that one..... |
| 22:46.04 | generalhan | lol |
| 22:46.15 | generalhan | but thats the only thing that isnt right so if you see something else |
| 22:46.32 | Uberbot | Do you have a 7333 entry in extensions.conf in the "internal" context? |
| 22:46.45 | *** part/#asterisk Cresl1n (n=Cresl1n@digium.com) |
| 22:46.50 | generalhan | yep |
| 22:47.59 | pygrammer | nothing about the gxp-2000 anyone? |
| 22:48.09 | bjohnson | make an ivr and have each exten set a variable |
| 22:48.52 | tsume | *ugh* |
| 22:49.21 | Darwin35 | asterisk is for chumps |
| 22:49.24 | Darwin35 | lol |
| 22:49.25 | tsume | okay, there is an extension which I have on the system 101. and it wont stop saying there is a person on the phone, even when I disconnect the set! :( |
| 22:49.26 | Darwin35 | joking |
| 22:49.31 | generalhan | Uberbot: isnt it just the weirdest freaking thing ever it all looks like it should be working |
| 22:49.32 | tsume | any ideas? :) |
| 22:49.45 | Darwin35 | thats because the phone did not register |
| 22:49.53 | Uberbot | Yes, this is weird. I'm stumped. |
| 22:50.06 | tsume | Darwin35: gah, damn thing booted up though |
| 22:50.07 | Uberbot | Reflash the image? |
| 22:50.09 | tsume | Darwin35: snom220 |
| 22:50.16 | Darwin35 | if the phone does not register it goues righ to the busy vm msg |
| 22:50.27 | tsume | I see.. |
| 22:50.30 | surfdue | guys |
| 22:50.30 | generalhan | downgrade then back up again ? |
| 22:50.31 | surfdue | read this |
| 22:50.31 | surfdue | http://www.i-hacked.com/content/view/168/67/ |
| 22:50.34 | tsume | hmm |
| 22:50.35 | surfdue | its very interesting |
| 22:50.37 | Darwin35 | it might boot but is it showing it registerd |
| 22:50.44 | surfdue | turn any wireless phone into a broadband phone for free |
| 22:50.45 | surfdue | :) |
| 22:50.53 | tsume | I guess I need to log in to the snom interface |
| 22:50.56 | *** join/#asterisk JunK-Y_ (n=junky@MTL-HSE-ppp197086.qc.sympatico.ca) |
| 22:51.01 | tsume | I hope my friend placed in the right passwd |
| 22:51.50 | Uberbot | generalhan, that's what I'm thinking.... But I admit I'm grasping for anything at this point. |
| 22:52.07 | CoffeeIV | how does one go about setting up a "callgroup" or "ring group" that is round-robin versus random ? Is an app needed for that ? |
| 22:52.20 | CoffeeIV | hmm, am I confusing "group" and "queue" ? |
| 22:52.29 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 22:52.33 | Uberbot | BRB. |
| 22:53.50 | Darwin35 | tsume is your box behind a firewall/router |
| 22:54.03 | Darwin35 | if so do you have the asterisk box on the dmz port |
| 22:55.13 | *** join/#asterisk clyrrad (n=ddd@CPE000c4120d329-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 22:56.16 | clyrrad | Is there anyting special you need to do to allow a fax machine to dial out and send faxes? Currently it can receive faxes, and it can also dial out, but for some reason the receving fax machine does not connect with the one sending (the one connected to asterisk) what am I missing? |
| 22:56.17 | *** join/#asterisk bucky (n=bucky@pdpc/supporter/active/bucky) |
| 22:56.40 | file | clyrrad: faxing over VoIP isn't exactly the best |
| 22:56.48 | clyrrad | how come? |
| 22:56.56 | Nugget | faxing over voip is tremendously unreliable. |
| 22:57.04 | file | you should go Google that... it's a long discussed subject |
| 22:57.09 | *** part/#asterisk CoffeeIV (n=CoffeeIV@mail.airlinksystems.com) |
| 22:57.31 | adrenaline | anybody have asterisk successfully running on slackware? |
| 22:57.49 | Nugget | I do. |
| 22:57.51 | clyrrad | ok, will do.... are they any work arounds that you have implemented and had working? Or did you have to stick with your phone company for fax line? |
| 22:58.03 | file | clyrrad: it's pretty much |
| 22:58.06 | adrenaline | So your saying theres a chance? |
| 22:58.07 | file | "if it works, good for you" |
| 22:58.13 | file | otherwise, "too bad..." |
| 22:58.15 | opus_ | fax over ultrawideband works for vonage |
| 22:58.44 | opus_ | i'm going to implement it for asterisk soon. |
| 22:59.13 | opus_ | its a 2 pizza difficultality level job |
| 22:59.13 | file | it's the jitter and latency that gets you |
| 22:59.15 | clyrrad | LOL... well we are half way there it can recieve faxes, just cant transmit them :P |
| 22:59.30 | adrenaline | opus_, I like your style |
| 22:59.46 | clyrrad | perhaps faxing over SIP instead of IAX2 will solve the problem |
| 22:59.53 | tsume | oh damn it |
| 23:00.01 | tsume | the phone looks like it logs on |
| 23:00.07 | tsume | has X in the lcd |
| 23:00.10 | tsume | damn snom |
| 23:00.17 | opus_ | clyrrad - what was your method of getting super mario brothers to work when the NES light was flashing, curious |
| 23:00.46 | clyrrad | clean the sucker and try again |
| 23:00.55 | opus_ | some people blew really hard on the cartiage. |
| 23:01.12 | clyrrad | qtip worked well to clean em |
| 23:01.26 | *** join/#asterisk RoyK (n=roy@cm-80.111.22.187.chello.no) |
| 23:02.01 | opus_ | fax over ip is like playing that old super mario brothers cartiage in a sense:) |
| 23:02.04 | brettnem | hey all.. anyone know of an open source circuit/network inventory system? :) |
| 23:03.00 | *** join/#asterisk cinix (n=beecards@tor/session/x-10bf946326f61265) |
| 23:03.20 | clyrrad | opus_ yah... just hate to keep my phone line with my regular phone company want to get away from paying 25 bucks a month |
| 23:04.01 | *** join/#asterisk burton27 (i=mimx@w201.ljudmila.org) |
| 23:05.26 | *** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:05.26 | *** mode/#asterisk [+o drumkilla_laptop] by ChanServ |
| 23:05.42 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:06.19 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
| 23:06.37 | Umaro | What version of zaptel started using rtc for ztdummy? |
| 23:06.56 | *** join/#asterisk Johaan (n=johaan@ip70-190-168-159.ph.ph.cox.net) |
| 23:06.57 | opus_ | umaro i think like a year ago, with make linux26 ? |
| 23:07.20 | Umaro | opus_: well, basically what I want is to compile ztdummy for 2.6, but use the usb |
| 23:07.25 | blitzrage | Umaro: since 2.6.0 kernel |
| 23:07.41 | blitzrage | Umaro: I'm not sure if it works like that anymore... |
| 23:07.57 | blitzrage | Umaro: in terms of on 2.6 - not sure if you can tell it to use USB... never had anyone ask that before :) |
| 23:08.10 | bkw_ | oooooooooooooooooklahoma where the wind comes sweeping down the plains |
| 23:08.16 | opus_ | umaro why? |
| 23:08.20 | *** join/#asterisk mcn_ (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 23:08.24 | blitzrage | bkw_: yer odd |
| 23:08.24 | bjohnson | I wanna go west, into the sunshine |
| 23:08.28 | blitzrage | :) |
| 23:08.34 | bkw_ | blitzrage, yes very odd |
| 23:08.34 | bjohnson | where a man can be a man and still be free |
| 23:08.40 | JunK-Y | u wanna go WEST, u sure of that? :P |
| 23:08.41 | Umaro | opus_: rtc isn't available on a ia64 |
| 23:08.47 | bkw_ | bjohnson, you mean San Francisco? |
| 23:08.54 | opus_ | Umaro then you'll have bigger problems |
| 23:08.55 | bkw_ | JunK-Y, I commited that patch |
| 23:09.00 | *** join/#asterisk TheCops (n=mdb@206-248-136-187.dsl.teksavvy.com) |
| 23:09.04 | opus_ | Umaro replace the motherboard:) |
| 23:09.08 | Umaro | opus_: how's that? |
| 23:09.18 | Umaro | opus_: this is an SGI altix 350 |
| 23:09.37 | bkw_ | JunK-Y, can you double check it fer me? |
| 23:09.52 | JunK-Y | yep |
| 23:10.05 | pygrammer | any suggestions again? i know the budgetone was said to be ok by some people here and the gxp-2000 as well, and that other brand that starts with an a |
| 23:10.15 | bkw_ | I see one bug |
| 23:10.18 | opus_ | Umaro I think other programs like musiconhold help apps would have a hard time syncing with everything |
| 23:10.31 | DarthClue | pygrammer: get a polycom ip300, you'll be happier |
| 23:10.37 | Umaro | opus_: hmm.. |
| 23:10.44 | DarthClue | with a 501 if you can afford it. |
| 23:10.49 | *** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:10.49 | *** mode/#asterisk [+o drumkilla_laptop] by ChanServ |
| 23:10.57 | pygrammer | DarthClue: i don't want to spend $190 on a hardphone right now |
| 23:11.01 | pygrammer | so the 501 is definitely out |
| 23:11.05 | file[laptop] | drumkilla_laptop: eep |
| 23:11.16 | pygrammer | and on voipsupply, there is no "asterisk tested" logo for the ip300 |
| 23:11.18 | opus_ | Umaro try a 2.4 kernel? the problem is the asterisk make files are very tightly integrated with the OS and its hard to muck with them, esp when there is new versions of asterisk ever day |
| 23:11.20 | DarthClue | 501 is only 172 |
| 23:11.34 | drumkilla_laptop | file[laptop]: !!! |
| 23:11.39 | file[laptop] | drumkilla_laptop: !!! |
| 23:11.47 | DarthClue | and the ip300/301 does work with asterisk. i think anthm even has one. |
| 23:11.49 | pygrammer | how about the aastra 9112i? heard that was ok |
| 23:11.49 | file[laptop] | drumkilla_laptop: still at teh gym? |
| 23:12.03 | pygrammer | DarthClue: who is anthm? |
| 23:12.10 | Umaro | opus_: 2.4 kernel.. hmm.. ok |
| 23:12.15 | pygrammer | and where is the 501 $172 |
| 23:12.17 | pygrammer | shipped ... |
| 23:12.20 | RoyK | http://karlsbakk.net/nicolas.html |
| 23:12.22 | pygrammer | i found it $190 shipped |
| 23:12.38 | drumkilla_laptop | file[laptop]: yeah ... waiting on my friend. I hope he shows up :/ |
| 23:12.44 | file[laptop] | drumkilla_laptop: cooool |
| 23:12.59 | drumkilla_laptop | but I have wireless here, so I'm just working on a school programming project. |
| 23:13.00 | DarthClue | well, 190 is probably about right with shipping. ask file who anthm is, he knows him better, or even bkw. |
| 23:13.01 | pygrammer | it's just a lot of money to spend right now -- i just want something decent on which to test my asterisk system and talk on the phone occasionally |
| 23:13.14 | pygrammer | but something that looks decent, if that's possible -- the budgetone looks shitty |
| 23:13.37 | Umaro | opus_: let me think of how to do that properly, lol |
| 23:14.13 | opus_ | Umaro ask the dev list. and if anybody saids ask the user list, start a flame war |
| 23:14.27 | opus_ | j/k |
| 23:15.17 | tsume | mmm |
| 23:15.21 | tsume | this damn snom |
| 23:15.33 | tsume | curious, how can I find out why this snom isn't registering? |
| 23:15.44 | pygrammer | DarthClue: ok |
| 23:15.50 | pygrammer | tsume: what is snom? i keep hearing it being mentioend |
| 23:15.52 | pygrammer | *mentioned |
| 23:15.53 | tsume | I see in asterisk -r it it trying to agent-add, but then fails with non-zero |
| 23:16.00 | tsume | pygrammer: this type of phone |
| 23:16.07 | tsume | pygrammer: a receptionist phone |
| 23:16.07 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 23:16.29 | Umaro | Anyone want to sell me a TE410P? I would order one from digium but I need it mailed overnight tommorow |
| 23:16.37 | tsume | right now its worthless unless it will register |
| 23:16.39 | tsume | it used to |
| 23:16.44 | pygrammer | oh, i see it now |
| 23:16.47 | pygrammer | the 190 or the 360? |
| 23:16.59 | tsume | 220 |
| 23:17.31 | tsume | right now this just sucks |
| 23:17.33 | pygrammer | tsume: don't see it on voipsupply |
| 23:17.39 | tsume | pygrammer: discontinuted |
| 23:17.47 | sams | anyone know a way to get the voicemail notification light working using realtime? (supposedly * doesn't support the voicemail light with realtime) |
| 23:17.49 | pygrammer | tsume: ah |
| 23:17.57 | tsume | the 360 is sold in replacement |
| 23:18.02 | tsume | larger, more buttons |
| 23:18.11 | tsume | I'm about to cry however :) |
| 23:18.16 | pygrammer | lol |
| 23:18.25 | pygrammer | we all have times like that |
| 23:18.27 | tsume | I didn't set these up |
| 23:18.33 | tsume | how does registration work? |
| 23:18.34 | pygrammer | like yesterday, trying to get wxPython 2.6 with debian testing |
| 23:18.37 | tsume | any passwords? |
| 23:18.55 | pygrammer | tsume: reg for what? |
| 23:18.56 | pygrammer | asterisk? |
| 23:19.03 | *** join/#asterisk Pj (n=pj@fernande.happycoders.org) |
| 23:19.03 | tsume | agent registration |
| 23:19.05 | pygrammer | passwords for what? |
| 23:19.06 | pygrammer | oh |
| 23:19.11 | pygrammer | depends how you have it set up... |
| 23:19.18 | tsume | There has to be _SOMETHING_ wrong |
| 23:19.27 | pygrammer | is the uniden uip 200 ok? |
| 23:19.42 | pygrammer | back-ordered -- never mined |
| 23:19.43 | pygrammer | *mind |
| 23:19.45 | pygrammer | want to get something soon |
| 23:20.01 | bkw_ | I've been waiting all my life... |
| 23:20.06 | bkw_ | to feel what i'm feelin now |
| 23:20.13 | pygrammer | the Aastra 9112i SIP IP Phone |
| 23:20.13 | pygrammer | w/ Speakerphone looks ok -- $139.95 |
| 23:20.32 | DarthClue | pygrammer: link? |
| 23:20.57 | pygrammer | there's always the sipura-841, though i've heard it's not very good |
| 23:20.59 | pygrammer | DarthClue: h/o |
| 23:21.18 | pygrammer | DarthClue: http://www.voipsupply.com/product_info.php?cPath=95_106&products_id=1009 |
| 23:21.18 | tsume | Aug 31 15:02:54 asterisk[31564]: Failed attempt from 127.0.0.1 |
| 23:21.25 | tsume | what the hell does this mean? :P |
| 23:21.26 | pygrammer | tsume: look in agents.conf |
| 23:21.34 | tsume | failed attempt to what? |
| 23:21.36 | pygrammer | how do you have your agent configured? |
| 23:21.44 | pygrammer | tsume: lol, no idea |
| 23:21.45 | tsume | no idea, I didn't set this up :( |
| 23:21.53 | pygrammer | tsume: uh, can't you login and check? |
| 23:22.03 | pygrammer | agents.conf |
| 23:22.14 | tsume | agent => 101,101,Reception 1 |
| 23:22.22 | DarthClue | pygrammer: i think you'll be happier with an ip300/301 |
| 23:22.25 | pygrammer | password is 101 then, right? |
| 23:22.31 | tsume | hmm |
| 23:22.38 | tsume | setting password on snom.. hmm |
| 23:22.40 | pygrammer | or have i forgotten my agents.conf |
| 23:22.55 | pygrammer | DarthClue: do you have one? what do you use? |
| 23:23.00 | tsume | I'm going to shoot myself if this is what happened |
| 23:23.32 | DarthClue | pygrammer: i use an ip501. take a look here for an ip301 ... http://www.tritechcoa.com/product/b-7V-234.html |
| 23:24.14 | *** join/#asterisk Goshen (n=Goshen@67-40-107-29.slkc.qwest.net) |
| 23:24.20 | hugo-v6 | the 300/1 looks like a toy for kids |
| 23:24.33 | hugo-v6 | and its expensive |
| 23:24.41 | tsume | I'm going to cry |
| 23:24.43 | *** part/#asterisk roche (n=roche@obiwan.inalambrica.net) |
| 23:24.46 | DarthClue | hugo-v6: what would you recommend? |
| 23:24.53 | tsume | that stupid password window appeared and I punched in garbage |
| 23:24.54 | pygrammer | hugo-v6: what he said ^ |
| 23:24.57 | tsume | now I can't get it back :( |
| 23:25.05 | hugo-v6 | DarthClue: i decided for snom 190 |
| 23:25.18 | pygrammer | DarthClue: that's not a terrible price, but it just has like no features |
| 23:25.31 | pygrammer | http://voip-info.org/tiki-index.php?page=480i+CT+Cordless |
| 23:25.34 | DarthClue | pygrammer: what kind of features? |
| 23:25.34 | pygrammer | that's awesome :) |
| 23:25.42 | pygrammer | DarthClue: just like the 7960G -- everything you can name :D |
| 23:25.53 | pygrammer | speakerphone, directory, XML, etc. |
| 23:26.02 | DarthClue | hugo-v6: um...you realize that the snom 190 is $200 right? |
| 23:26.30 | hugo-v6 | DarthClue: dunno, get it for about 97euros :) |
| 23:26.32 | pygrammer | hugo-v6: that's expensive for me right now -- again, review my needs if you're going to suggest something |
| 23:26.36 | DarthClue | pygrammer: don't expect those features on a good phone for a price less than that of an ip501 then. |
| 23:26.46 | *** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 23:26.51 | hugo-v6 | i dont know who and for what price u buy it |
| 23:26.52 | pygrammer | DarthClue: well, i don't need them -- i was just being irrational for my budget |
| 23:26.56 | *** join/#asterisk kereemcaam (i=KELEBEK@81.212.220.171) |
| 23:27.04 | *** join/#asterisk jeffgus_ (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 23:27.05 | pygrammer | i just *like* them :) |
| 23:27.09 | hugo-v6 | if u want something cheap get a grandstream |
| 23:27.22 | pygrammer | i've already spent about $400 today and don't feel like spending hundreds on an ip phone atm |
| 23:27.34 | Vco | ummm...someone have a good link for jabber notification other than the "heres a config file and no instructions" from the wiki? |
| 23:27.37 | pygrammer | hugo-v6: those look like shit and the cheap ones are only one line |
| 23:27.38 | DarthClue | pygrammer: start with an ip301 and go from there. that will get you something to test with and something that you'll be happy to keep. then get an ip501 or ip600 later when you can afford it. |
| 23:28.00 | hugo-v6 | but then a ip 300 from polycom looks just as bad as the grandstream |
| 23:28.01 | DarthClue | grandstreams are shit, they are enough to test with, but not worth keeping. |
| 23:28.01 | Ariel_ | hello everyone |
| 23:28.11 | Vco | get a grandstream 386 |
| 23:28.25 | DarthClue | the polycoms are worth keeping and actually look alot better than the grandstreams |
| 23:28.33 | generalhan | can some one take a look at my sip.conf and tell me why the top one works with X-Lite and the bottom one doesnt ? ~~~~~~~~ http://pastebin.ca/21784 |
| 23:28.37 | Vco | aroudn the same price as a bt-100 and you can run 2 regular phones off it.... |
| 23:29.03 | DarthClue | they even look better than the snom in my opinion, but then again, i've got a polycom sitting right here that looks very, very nice |
| 23:29.03 | Vco | which makes it a little handy for testing.. |
| 23:29.09 | pygrammer | Vco: i would prefer not to get an ATA ... |
| 23:29.12 | pygrammer | if that's what it is |
| 23:29.21 | Vco | it is... |
| 23:29.39 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 23:29.39 | *** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net) |
| 23:30.57 | generalhan | anyone have a second to look at my sip.conf for me ? |
| 23:30.58 | pygrammer | DarthClue: so you like your 501? |
| 23:30.59 | generalhan | please |
| 23:31.31 | hugo-v6 | DarthClue: not the ip 300/1 or u got a damn bad taste :> |
| 23:31.33 | pygrammer | DarthClue: oh, also, no speakerphone on the ip301 |
| 23:31.34 | DarthClue | pygrammer: yes, it has many of the features of a more expensive cisco and didn't hurt to get it. |
| 23:32.01 | DarthClue | pygrammer: like i said, don't expect certain features on a good phone for a price less than the ip501. |
| 23:32.22 | RoyK | otmar: http://karlsbakk.net/nicolas.html |
| 23:32.22 | *** join/#asterisk one2 (n=knoppix@cpe-66-87-4-181.ut.sprintbbd.net) |
| 23:32.26 | pygrammer | DarthClue: :-/ |
| 23:32.35 | DarthClue | hugo-v6: um, well, i can say that the snom looks like a cheap toy, much like the grandstreams. but that is just my opinion. |
| 23:32.40 | *** join/#asterisk DukeOfURL (n=chatzill@67-41-211-223.brbn.qwest.net) |
| 23:33.00 | pygrammer | maybe an ATA would be better, but i think it'd be cool just to have an IP phone and go from there -- there are some problems with ATAs that i've heard about, and an ip phone just seems to make more sense |
| 23:33.03 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 23:33.03 | *** join/#asterisk darkskiez (n=darkskie@host86-133-147-187.range86-133.btcentralplus.com) |
| 23:33.06 | *** part/#asterisk kereemcaam (i=KELEBEK@81.212.220.171) |
| 23:33.15 | Vco | well..really... |
| 23:33.19 | DarthClue | pygrammer: if you just want to test, get a ip301 and test. once you decide to keep going, get a 501 and worry about all those features |
| 23:33.31 | Vco | if you're looking at a budget phone |
| 23:33.40 | Vco | there not a large margin of difference there.. |
| 23:34.11 | hugo-v6 | DarthClue: not seriously. |
| 23:34.36 | pygrammer | DarthClue: then i'll have spent $330 |
| 23:34.42 | pygrammer | *sigh* life sucks sometimes :) |
| 23:35.10 | Connor__ | okay, wtf.. I can't use my normal Connor_ because someone else is logged in as me. and I cann't send private message unless I'm registered.. |
| 23:35.15 | hugo-v6 | look at the grandstream and the snom. snom looks clean and not like a toy. well if u like toys for kids... then grandstream and polycom 300/1 are right fro u. my opinion |
| 23:36.15 | DukeOfURL | What is the best HTML composer in open-source land???? |
| 23:36.20 | *** join/#asterisk Connor__ (n=billy@198-144-165-65.knx.tn.nxs.net) |
| 23:36.39 | hugo-v6 | vim |
| 23:36.47 | DukeOfURL | hehe |
| 23:37.08 | tsume | finally |
| 23:37.17 | DukeOfURL | that's what I'm doing now--is there an easier way? |
| 23:37.22 | Vco | pfft....vi is for pussy's |
| 23:37.33 | DukeOfURL | oh heck! an emacs guy |
| 23:37.44 | Vco | echo "#your htlms stuff here" > index.html |
| 23:37.46 | hugo-v6 | if u need a own os. then get emacs |
| 23:37.48 | hugo-v6 | *sigh* |
| 23:38.00 | Vco | hell.....html even |
| 23:38.08 | DukeOfURL | yep, what did i say |
| 23:38.13 | DukeOfURL | now it's a religious discussion |
| 23:38.58 | hugo-v6 | beside of the os thing emacs conatins mac. a boo word |
| 23:39.11 | Vco | boo-urns? |
| 23:39.30 | generalhan | i was saying boo-urns |
| 23:39.33 | generalhan | hahaha |
| 23:40.25 | pygrammer | so, does everyone agree that the ip301 is good for a starter phone? |
| 23:40.33 | pygrammer | how is sound quality, echo, etc.? |
| 23:40.44 | pygrammer | does it sound like a land-line (provided you're using the right codec)? |
| 23:40.59 | *** join/#asterisk Connor__ (n=billy@198-144-165-65.knx.tn.nxs.net) |
| 23:41.05 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 23:41.47 | hugo-v6 | pygrammer: everyone says polycom is ok. (i hate it because here its more expensive than snoms and it looks bad) but if i get too much money ill buy one for myself to tell u all that it doesnt only look bad ;)) |
| 23:42.02 | Vco | dude...if you live where i do , you DON"T want it to sounds like the land line..... |
| 23:42.25 | hugo-v6 | vco: .ua? ;)) *run* |
| 23:42.29 | pygrammer | Vco: haha |
| 23:42.32 | DukeOfURL | BTW, IP301 doesn't have a mic for the speakerphone |
| 23:42.51 | hugo-v6 | DukeOfURL: well, how u talk then? |
| 23:42.55 | pygrammer | well, i know it sounds *better* than a land-line on sip-to-sip if you're using ulaw/alaw/g.729, but what about calling the PSTN? |
| 23:43.02 | pygrammer | DukeOfURL: so there *is* a speaker? |
| 23:43.03 | pygrammer | but no mic? |
| 23:43.05 | DukeOfURL | you pick up the handset |
| 23:43.09 | pygrammer | so it's just for listening to messages, etc.? |
| 23:43.28 | hugo-v6 | aaw... thats gay |
| 23:43.34 | DukeOfURL | yes, and people that listen to vm on their speaker phone should be taken out and shot |
| 23:43.46 | DukeOfURL | no bugger disruption in an office |
| 23:43.54 | DukeOfURL | no bigger |
| 23:44.12 | Vco | especially when it's their encologist (sp) |
| 23:44.16 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 23:44.26 | pygrammer | what's an encologist? |
| 23:44.29 | DukeOfURL | you mean cancer doc? |
| 23:44.29 | pygrammer | oncologist? |
| 23:44.33 | *** join/#asterisk SwK (n=SwK@12-219-144-126.client.mchsi.com) |
| 23:44.34 | pygrammer | that's a cancer doctor |
| 23:44.51 | DukeOfURL | the good news is that you have 30 days to live.... |
| 23:44.55 | pygrammer | "Hey, this is Dr. Smith calling, just letting you know that you're going to die in 3 weeks! Thanks, have a great day! Bye." |
| 23:44.58 | *** join/#asterisk _cleric_ (n=dacleric@p5482ACB5.dip0.t-ipconnect.de) |
| 23:45.04 | DukeOfURL | the bad news is that i got the lab results a month ago |
| 23:45.10 | pygrammer | "SHUT THAT FUCKING SPEAKERPHONE OFF!" |
| 23:45.22 | DukeOfURL | right |
| 23:45.22 | Vco | thats not the one i'm thinking of.. |
| 23:45.25 | hugo-v6 | hrrrrr |
| 23:45.40 | pygrammer | so -- DukeOfURL do you recommend the ip301? |
| 23:45.53 | DukeOfURL | go for the ip501 |
| 23:45.56 | pygrammer | bah |
| 23:45.58 | hugo-v6 | vco: did u mean some green greenpeace franatic? |
| 23:46.11 | DukeOfURL | if your client is cost concious, show him a Sipura 851 |
| 23:46.15 | Connor__ | Well, it said it killed them.. but.. they're still logged in. |
| 23:46.15 | Vco | Endoscopy kinda thing |
| 23:46.16 | pygrammer | too expensive -- already spent almost $400 today and don't want to spend too much, especially if i don't use it much or lose interest... |
| 23:46.21 | pygrammer | DukeOfURL: this is for myself |
| 23:46.23 | pygrammer | i don't even have a hardhpone |
| 23:46.26 | pygrammer | *phone |
| 23:46.30 | *** join/#asterisk shido6 (n=shido@d57-87-253.home.cgocable.net) |
| 23:46.32 | Connor__ | mis-chan |
| 23:46.43 | hugo-v6 | the 500 looks a lot better than teh 300 |
| 23:46.56 | pygrammer | agreed, but it's more expensive |
| 23:47.08 | hugo-v6 | pygrammer: u loose intrest in telephony? |
| 23:47.18 | hugo-v6 | pygrammer: well yes |
| 23:47.29 | pygrammer | hugo-v6: not yet, but i may at some point, as it's not a business thing for me -- yet |
| 23:47.35 | hugo-v6 | teh 500 is a phone i would buy... |
| 23:47.53 | *** part/#asterisk Connor__ (n=billy@198-144-165-65.knx.tn.nxs.net) |
| 23:47.58 | hugo-v6 | pygrammer: but u wont get a cheaper and better telephony/fax-solution than asterisk |
| 23:47.58 | DukeOfURL | people that use "u" instead of "you" should definitely be taken out behind the barn and shot |
| 23:48.08 | hugo-v6 | DukeOfURL: go away |
| 23:48.30 | tsume | I'm gonna cry :P |
| 23:48.34 | DukeOfURL | it may have been appropriate in the days of 300baud modems, not not today. |
| 23:48.34 | tsume | *:( |
| 23:48.35 | Vco | or fisted by an nba player |
| 23:48.38 | tsume | damn phone |
| 23:49.01 | *** join/#asterisk newl (n=newlook@203-59-168-152.perm.iinet.net.au) |
| 23:49.01 | pygrammer | hugo-v6: i understand -- but i don't *rely* on asterisk, i just do it for fun |
| 23:49.09 | pygrammer | so i'm saying, if i got a phone, it would be used for fun at this point, not business |
| 23:49.11 | tsume | nooooo |
| 23:49.19 | tsume | "Status: forbidden" |
| 23:49.21 | hugo-v6 | DukeOfURL: it saves bits and is shorter to type. if u dont like it write an alias to let it rewrite. otherwise. go away. |
| 23:49.22 | tsume | damn you asterisk |
| 23:49.24 | pygrammer | hence why a 501 is probably a bit over what i want to -- and rationally SHOULD -- spend |
| 23:49.43 | DukeOfURL | go away? is that an order, sir? |
| 23:49.47 | *** join/#asterisk Connor_ (n=billy@198-144-165-65.knx.tn.nxs.net) |
| 23:49.56 | hugo-v6 | pygrammer: me too... but i havent found a better solution ... pbx4linux would be a option... yes... |
| 23:50.06 | DukeOfURL | save bits of what? |
| 23:50.09 | hugo-v6 | DukeOfURL: take it like u want. |
| 23:50.12 | Vco | thats why i mentioned an ATA kinda thing...you get the ip phone deal out of it (x2) an dyou can use it with a cordless or whatever... |
| 23:50.21 | Vco | and they're cheap |
| 23:50.22 | hugo-v6 | DukeOfURL: bandwith |
| 23:50.50 | DukeOfURL | so you're on a 300baud modem? waiting for each and every bit to come across? |
| 23:51.13 | Vco | i orderd one just to use with one line registered to seperate asterisk boxes.. |
| 23:51.20 | newsmafia | #oracle |
| 23:51.25 | DukeOfURL | you pay your ISP by the bit, i assume... |
| 23:51.27 | pygrammer | Vco: true, but what do you mean ip phone deal out of it? it *isn't* an ip phone, that's the whole point. and my dad has this old nortel phone, but i don't know if it still works or if it would work well with the ATA, i.e. caller id, call waiting, transfer, speakerphone, EVERYTHING i want to work |
| 23:51.28 | newsmafia | oops |
| 23:51.40 | pygrammer | DukeOfURL: $0.000000000000000000000000000000000000000000000001/bit |
| 23:51.45 | Vco | ii get called id etc etc etc... |
| 23:51.52 | Vco | i can flash and transfer |
| 23:52.06 | Vco | like.. |
| 23:52.31 | Vco | you really don't get much else out of an acutal sip phone that you can't do.. |
| 23:53.13 | Vco | fuck i need to learn to type |
| 23:53.22 | one2 | sip and nat sounds like cludge |
| 23:54.00 | hugo-v6 | well it gets late... one of a few last cigarettes now. |
| 23:58.05 | ender | so I had a sip phone that I re-issued to a new user. Unfortunately calls for the old user keep going to the new user because the IP addres sis the same (dhcp bound somewhat to mac). Is there any way to clear the cache in asterisk on what phone has what IP? |
| 23:58.10 | ender | basically flush the sip registration? |
| 23:59.06 | *** join/#asterisk _cleric_ (n=dacleric@p5482ACB5.dip0.t-ipconnect.de) |