| 00:00.55 | v4mp | bkw_, any clue to my problem ? |
| 00:01.01 | infinity1 | bkw_: i don't think i can, because it just says playback "record/myfile" |
| 00:01.09 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 00:01.23 | infinity1 | which would be /usr/share/asterisk/sounds/record/myfile |
| 00:01.38 | bkw_ | v4mp: no clue dude... I don't use any of the latest Asterisk releases |
| 00:01.50 | bkw_ | infinity1: look in asterisk.conf in /etc/asterisk |
| 00:02.07 | infinity1 | bkw_: yea. i looked in that. there isn't a record/sound path |
| 00:02.10 | bkw_ | plus I don't agree with direct to DB stuff in a Telephony process |
| 00:02.16 | bkw_ | infinity1: their should be |
| 00:02.52 | *** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320) |
| 00:02.54 | bkw_ | thats right they key off something else in there |
| 00:03.14 | bkw_ | its /var/lib/asterisk/sounds |
| 00:03.32 | bkw_ | then if you do playback(blah.wav) it should find it in /var/lib/asterisk/sounds/blah.wav |
| 00:03.36 | infinity1 | the closest thing i can find is astvarlibdir => /var/lib/asterisk |
| 00:03.36 | bkw_ | or thats how it used to work |
| 00:03.51 | infinity1 | bkw_: i'm using the debian asterisk |
| 00:03.54 | v4mp | :@ |
| 00:03.57 | bkw_ | the only thing I can think is your AGI resets it or uses absolute paths |
| 00:04.00 | infinity1 | apparently it begs to difffer :) |
| 00:04.11 | bkw_ | infinity1: distro mans nothing really |
| 00:04.27 | infinity1 | bkw_: i don't see where the path is set to /usr/share/... |
| 00:04.33 | bkw_ | show me the agi script |
| 00:04.37 | bkw_ | bet you it does somewhere in there |
| 00:04.43 | infinity1 | # grep 'usr/share' * |
| 00:04.43 | infinity1 | asterisk.conf:astagidir => /usr/share/asterisk/agi-bin |
| 00:04.44 | infinity1 | voicemail.conf:; Look in /usr/share/zoneinfo/ for names of timezones. |
| 00:04.55 | bkw_ | maybe agi does that |
| 00:04.56 | bkw_ | which is weird |
| 00:04.57 | infinity1 | its extensions.conf |
| 00:05.07 | *** join/#asterisk mog (n=mog@74.95.48.254) |
| 00:05.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:05.17 | bkw_ | I would say thats a bug if what you're saying is right |
| 00:05.18 | infinity1 | exten = s,2,background(record/thankYouForCalling) |
| 00:05.18 | infinity1 | exten = s,3,background(record/test1) |
| 00:05.28 | bkw_ | and record is where? |
| 00:05.30 | mog | bkw_, ! |
| 00:05.34 | mog | how goes |
| 00:05.44 | bkw_ | mog: reported two bugs with Asterisk today |
| 00:05.51 | bkw_ | and a third one I need someone to confirm |
| 00:05.59 | infinity1 | well, it asterisk-gui saves to /var/lib/asterisk/sounds/record |
| 00:06.04 | mog | fun |
| 00:06.11 | bkw_ | mog: its a really nasty codec one in SIP |
| 00:06.16 | bkw_ | I |
| 00:06.21 | mog | eep |
| 00:06.25 | bkw_ | i'm sure its just a flag thats not reset on re-invite |
| 00:06.39 | bkw_ | because when your peer says 200 OK GSM ONLY DUDE.. and you start sending it ulaw... thats broken |
| 00:06.44 | v4mp | what about |
| 00:06.45 | v4mp | <PROTECTED> |
| 00:06.52 | mog | agreed |
| 00:07.18 | bkw_ | we had someone say audio out enum was bad.. and I went to tracking it down and found the issue |
| 00:07.23 | bkw_ | pesky re-invite |
| 00:07.39 | bkw_ | which reminds me |
| 00:07.41 | bkw_ | let me pm you |
| 00:07.44 | jeev | mog |
| 00:08.36 | v4mp | any clues to that problem ? |
| 00:08.47 | bkw_ | v4mp: you're like all over the radar dude |
| 00:09.01 | v4mp | in what sense ? |
| 00:09.02 | *** join/#asterisk c4t3l (n=root@74.95.210.124) |
| 00:09.10 | bkw_ | well didn't you just ask about a MySQL issue? |
| 00:09.13 | bkw_ | then this zap issue |
| 00:09.20 | v4mp | yes |
| 00:09.25 | v4mp | got both issue's |
| 00:09.31 | drmessano | because when your peer says 200 OK GSM ONLY DUDE.. and you start sending it ulaw... thats broken <-- Have heard of that elsewhere.. Thought it would have been reported |
| 00:09.49 | bkw_ | drmessano: I reported it with traces today |
| 00:09.52 | c4t3l | ls |
| 00:09.55 | c4t3l | oops |
| 00:09.59 | v4mp | well u didn't know about 1 so thought u might know about other least then if u did could try fix 1 problem fo rthe time being |
| 00:10.01 | bkw_ | talked with oej about it this week at astricon |
| 00:10.04 | mog | yes jeev |
| 00:10.11 | jeev | like my new acronym |
| 00:10.22 | bkw_ | v4mp: the latter is the fact your are specifying the signalling before any channels are |
| 00:10.31 | drmessano | Cool |
| 00:10.32 | bkw_ | v4mp: the error is pretty clear on what to do... |
| 00:10.46 | bkw_ | drmessano: it was a bitch to nail down but I finally found a scenario that I could trace down. |
| 00:10.48 | v4mp | i dont understand it i just followed a guide |
| 00:11.28 | v4mp | i didn't tell it an order |
| 00:12.38 | bkw_ | well open up zapata.conf and double check |
| 00:13.39 | v4mp | its empty because the previous 1 threw out more errors :/ |
| 00:14.07 | *** part/#asterisk v4mp (n=Gary@82.118.111.250) |
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| 00:14.36 | v4mp | i followed a guide that said to remove it and restart it and it will write another |
| 00:15.04 | v4mp | because config wasn't right or sumthing so i done so its wrote a blank 1 tho :/ |
| 00:19.31 | v4mp | ok so a problem i thought was fixed isn't |
| 00:19.51 | v4mp | any clue on this then |
| 00:19.52 | v4mp | ERROR[5680]: chan_zap.c:7209 mkintf: Unable to open channel 1: No such device or address |
| 00:19.52 | v4mp | here = 0, tmp->channel = 1, channel = 1 |
| 00:22.23 | maxxim | can i use zaptel's echi cancelation for using external gateway? |
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| 00:33.19 | v4mp | any clue why it would be complaining about signalling before channel when the signalling is specified before the channel |
| 00:39.50 | [gnubie] | waves to all |
| 00:40.08 | [gnubie] | i'm having a weird problem here |
| 00:41.01 | v4mp | aren't we all :( |
| 00:41.56 | [gnubie] | placing an outbound call to an analog telephone via pots from a sip phone gives me a one way audio.. |
| 00:43.11 | [gnubie] | you can take a look at the scenarios i did at http://paste.debian.net/18044/ |
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| 01:20.49 | v4mp | does anyone even know how to deal with my problems ? because google isn't finding me anything useful either |
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| 02:29.03 | RypPn | [TK]D-Fender: cheers for the push earlier, solved the challenge with app_notify and gastify :) |
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| 02:41.06 | Yourname` | Hi, I'm following this for recordings: http://www.trixbox.org/forums/trixbox-forums/trixbox-projects/converting-recordings-mp3-during-idle-times -> however, I am doing it to quite a few recordings. Even ls doesn't seem to work. Can anyone suggestion a way for me to get this to work please? pb -> http://pastebin.ca/1212004 |
| 02:41.47 | Yourname` | Sorry, ls actually works. |
| 02:42.05 | Yourname` | There's 11178 recordings in there, and doing the make mp3 is giving that error as outlined in the pastebin. |
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| 03:05.58 | luke-jr | any idea why Asterisk stopped emailing voicemails a few weeks ago? |
| 03:07.29 | jaytee | no, this is the first I've heard of it. I didn't even know it was supposed to. |
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| 04:17.58 | genji1981 | Hello all. Okay, My asterisk on Openwrt (A 125 mhz 4mb flash, 8 mb ram) box works fantastically. Only problem ive got is.. how do i get asterisk to put a call on hold, then dial a number, starting a new call (so theres 2 calls) and when it answers, dial #? oh.. i tried D(ww#) in the dial string, it didn't work. |
| 04:19.17 | *** join/#asterisk jtodd (i=enlh0398@ns2.loligo.com) |
| 04:33.43 | Qwell | I need an electrician! :p |
| 04:33.51 | Qwell | stupid house wiring... |
| 04:35.16 | Qwell | if somebody knows a bit about house wiring, and wouldn't mind lending a hand with some advice, I'd be quite grateful.. |
| 04:35.35 | genji1981 | Hmm? |
| 04:36.32 | Qwell | genji1981: is that on zap with 1 fxo or something? |
| 04:36.40 | Qwell | I wouldn't really expect it to work |
| 04:36.46 | genji1981 | Qwell: nope. Sip stuff |
| 04:37.16 | Qwell | not really sure how you'd be putting a call on hold from the dialplan |
| 04:37.43 | genji1981 | Qwell: voip provider provides "hold call, dial *55 then at the voice, press #, and the call goes to voicemail" |
| 04:38.34 | Qwell | yuck.. |
| 04:39.10 | Qwell | I'd really like to kick whoever mucked with the wiring here...ugh. stupid, stupid, stupid |
| 04:39.21 | genji1981 | So, no ideas, Qwell? |
| 04:39.29 | Qwell | genji1981: no, sorry :( |
| 04:43.22 | *** join/#asterisk [gnubie] (n=gnubie@cm141.omega112.maxonline.com.sg) |
| 04:43.28 | [gnubie] | waves |
| 04:44.06 | [gnubie] | anyone here can point me to the right url where i can read on how to connect 2 asterisk boxes through an encrypted iax2? |
| 04:44.22 | Qwell | [gnubie]: it's a standard iax2 setup, with encryption=yes |
| 04:44.42 | Qwell | erm, no |
| 04:45.02 | [gnubie] | Qwell: that's all? i mean, i don't need to create certificates or similar to that? |
| 04:45.11 | Qwell | hang on, heh |
| 04:45.52 | Qwell | I'm thinking of something else |
| 04:45.57 | mog | no it does it based on pass |
| 04:46.02 | mog | all you need is enc=yes |
| 04:46.37 | Qwell | wait, yeah, that is right |
| 04:46.46 | [gnubie] | what about the media part? |
| 04:46.58 | mog | media is encrypted too |
| 04:47.05 | Qwell | mog: that isn't in the default config O.o |
| 04:47.13 | Qwell | I had to look in chan_iax2.c |
| 04:47.20 | mog | only first new frame is unencrypted |
| 04:47.25 | mog | its encrypted=yes |
| 04:47.31 | mog | i was just too lazy to type it all |
| 04:47.37 | [gnubie] | i am running 1.4.21.2 by the way |
| 04:47.47 | Qwell | mog: I mean, encryption= isn't in the config |
| 04:47.53 | mog | really? |
| 04:47.54 | Qwell | yeah |
| 04:48.04 | [gnubie] | so, simply adding that line "encrypted=yes" on my iax.conf will do the trick already? |
| 04:48.10 | Qwell | [gnubie]: encryption=yes |
| 04:48.25 | Qwell | mog: weird, eh? |
| 04:48.25 | [gnubie] | i mean, yes it's encryption=yes |
| 04:48.33 | mog | very Qwell |
| 04:48.35 | [gnubie] | is that all? |
| 04:48.45 | Qwell | oh, it is in trunk |
| 04:48.48 | Qwell | [gnubie]: that's all |
| 04:49.01 | Qwell | ; Enable IAX2 encryption. The default is no. |
| 04:49.01 | Qwell | ; |
| 04:49.02 | Qwell | ; encryption = yes |
| 04:49.04 | Qwell | heh |
| 04:49.12 | mog | groovy |
| 04:49.15 | [gnubie] | ok.. |
| 04:49.36 | [gnubie] | i never thought that how easy to encrypt an iax2 channel |
| 04:49.45 | mog | so easy its number 1 |
| 04:49.58 | [gnubie] | thanks guys.. ;) |
| 04:52.13 | mog | encryption = hotness |
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| 05:31.30 | drmessano | omg |
| 05:31.34 | drmessano | IAX2 encryption |
| 05:49.50 | RypPn | su - |
| 05:49.56 | RypPn | :) |
| 05:53.01 | drmessano | Just reading up on the Skype business |
| 05:56.10 | Qwell | drmessano: yum update |
| 05:57.30 | drmessano | ok |
| 05:57.47 | Qwell | not much reason, but yeah :p |
| 05:57.59 | Qwell | do you already have 2.5? |
| 05:58.48 | drmessano | um |
| 05:59.15 | drmessano | no |
| 05:59.23 | Qwell | update should get it, and fix perms |
| 05:59.49 | drmessano | Only pulled OS based updates |
| 05:59.54 | Qwell | O.o |
| 06:00.06 | Qwell | yum clean headers? |
| 06:00.17 | Qwell | no, headers should be there already |
| 06:00.28 | Qwell | *shrug* |
| 06:00.35 | genji1981 | voip provider provides "hold call, dial *55 then at the voice, press #, and the call goes to voicemail"... how do i make use of this in Asterisk? is there a way to dial without bridging the current call to the new dial? as in, the call hasn't left the asterisk box yet, due to being inside a menu. |
| 06:00.52 | drmessano | repo is there |
| 06:01.03 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 06:01.03 | *** mode/#asterisk [+o russellb] by ChanServ |
| 06:01.06 | Qwell | asterisk-current? |
| 06:01.32 | drmessano | Saw the address when I did an update, lemme check the .repo |
| 06:01.40 | Qwell | yum list repos I think |
| 06:01.47 | Qwell | erm, no |
| 06:01.52 | Qwell | yum repolist? |
| 06:02.05 | genji1981 | oh.. by the way, version Asterisk 1.4.21.2 |
| 06:02.18 | russellb | Qwell: are you letting him use the new hotness? |
| 06:02.23 | drmessano | yeah all good as far as being there |
| 06:02.24 | Qwell | he's had it for weeks |
| 06:02.26 | russellb | oh. |
| 06:02.28 | russellb | :) |
| 06:02.29 | Qwell | but he's a newb and hasn't tested it yet :P |
| 06:02.32 | russellb | lame |
| 06:02.33 | drmessano | lol |
| 06:02.41 | Qwell | something about "priorities" |
| 06:02.42 | Qwell | pfft |
| 06:02.45 | drmessano | lol |
| 06:03.00 | genji1981 | no ones got answers to do this? |
| 06:03.03 | drmessano | and the box I had for testing was some lame 3GHZ box with 256MB Ram, ordered ram for it |
| 06:03.29 | russellb | priorities are whack |
| 06:03.41 | drmessano | I didnt know 256MB DDR2 even existed |
| 06:03.48 | drmessano | "why?" |
| 06:03.49 | Qwell | drmessano: that seems a little silly |
| 06:03.58 | drmessano | very |
| 06:03.59 | genji1981 | russellb: oh... is it worth me reposting my question, so you can look at it? |
| 06:04.18 | russellb | genji1981: probably not to be honest |
| 06:04.20 | russellb | My brain is fried |
| 06:04.26 | drmessano | It was probably designed for Linux |
| 06:04.46 | drmessano | You know.. fast process and no RAM "because *nix doesn't need it" |
| 06:04.47 | russellb | I'm just here to be social.. |
| 06:04.55 | Qwell | russellb: fix my wiring, kthx |
| 06:05.02 | russellb | stares at Qwell |
| 06:05.02 | drmessano | Whats wrong with your wiring? |
| 06:05.10 | Qwell | drmessano: frankly, it's fucked. |
| 06:05.25 | Qwell | I went on a long diatribe about it elsewhere :p |
| 06:05.30 | drmessano | lol |
| 06:05.40 | drmessano | I scrolled up and saw you mention it |
| 06:05.41 | file | throws a volleyball to russellb |
| 06:05.42 | drmessano | No details tho |
| 06:05.44 | Qwell | I have a switch that turns on (ONLY ON?!?!) a light outside |
| 06:05.50 | drmessano | hmm |
| 06:06.14 | Qwell | it's a mess, heh |
| 06:06.18 | russellb | ic |
| 06:06.22 | drmessano | You flip it down, doesnt go off? |
| 06:06.28 | russellb | bumps the ball to Qwell |
| 06:06.38 | Qwell | drmessano: nope. it's one of those automatic lights that turn on when you walk/drive by |
| 06:06.44 | Qwell | flipping the switch triggers it to turn on |
| 06:06.48 | drmessano | Oh I see |
| 06:06.55 | Qwell | it makes no sense whatsoever |
| 06:07.21 | Qwell | what's weird, is that's the switch that controls the fan in my living room |
| 06:07.28 | russellb | ..... |
| 06:07.31 | Qwell | (and it's a 3-way...) |
| 06:07.38 | file | Qwell: huh |
| 06:07.46 | Qwell | file: now you see the need for crayons! |
| 06:07.55 | Qwell | it's gonna take me like an hour to diagram it |
| 06:07.56 | file | yes, crayons will solve the problem |
| 06:08.02 | drmessano | Sounds like where I used to live.. They had a floodlight on the corner of the house which was controlled by a switch in a double gang box on the wall in the bedroom nearest that corner, next to the overhead light switch |
| 06:08.05 | drmessano | Laziness |
| 06:08.20 | Qwell | but... the outside light has a switch of it's own! |
| 06:08.24 | Qwell | which DOES work |
| 06:08.47 | drmessano | Ah |
| 06:09.23 | genji1981 | Strange thing here too. There is a switch by the front door, has a safety-against-accidental-switching flip cover on it.. the switch doesn't seem to do anything, anywhere. |
| 06:09.24 | drmessano | So that switch, which also runs the fan in the living room, puts power to the switch for the outside light |
| 06:09.37 | Qwell | drmessano: apparently |
| 06:09.49 | Qwell | but, like I said, turning that switch off has no affect |
| 06:09.54 | file | Qwell: perhaps whomever did the wiring... was on drugs |
| 06:09.59 | Qwell | file: or worse |
| 06:10.08 | Qwell | they very clearly had no idea what they were doing |
| 06:10.12 | file | worse would be they were a Comcast employee |
| 06:10.52 | Qwell | file: don't even get me started on the switch next to it |
| 06:11.05 | Qwell | (it's hardwired always-on) |
| 06:11.12 | Qwell | why? I have no idea! |
| 06:11.23 | Qwell | but I'll have to turn off the breaker to fix it :( |
| 06:11.56 | drmessano | Qwell: Sounds like they wired the fan switch wrong, which is not uncommon... probably when the fan was put in.. The wire for the outside light switch should passively jump across the back of that switch, but instead is inline with it. |
| 06:12.20 | Qwell | but that doesn't explain why flipping the switch turns it on, and only on |
| 06:12.27 | Qwell | and the fan...works |
| 06:14.06 | drmessano | How does the light eventually go off? |
| 06:14.16 | Qwell | timed motion |
| 06:14.36 | Qwell | 4 minutes or whatever |
| 06:14.42 | drmessano | Does the switch turn the fan off? |
| 06:14.45 | Qwell | yep |
| 06:15.10 | drmessano | Does the light work without you using the switch? |
| 06:15.19 | drmessano | by motion. |
| 06:15.26 | Qwell | yep, it's "on" with that switch in either position |
| 06:15.34 | drmessano | Ah |
| 06:15.38 | drmessano | I see |
| 06:15.56 | Qwell | so unless it loses power very briefly when the switch is in the middle (undefined?) position... |
| 06:16.07 | Qwell | I don't know. even then, I can't explain it, since it's a 3-way switch |
| 06:16.18 | Qwell | 3-ways have 2 hot wires coming out of them |
| 06:16.30 | Qwell | so unless it's pulling off of both hot wires... |
| 06:16.41 | Qwell | I don't know. my head hurts though |
| 06:16.42 | drmessano | Yes that is exactly what is happening |
| 06:16.58 | Qwell | but it can't, because then it would be connecting the hot wires, making the fan always on |
| 06:16.59 | drmessano | When you change positions, it loses power for a split second |
| 06:17.04 | TJNII | That's the only reasonable explination.... |
| 06:17.11 | drmessano | So its sees power off, then on |
| 06:17.20 | drmessano | Which triggers it to auto-on |
| 06:17.31 | drmessano | No, the fan is wired differently in the switch |
| 06:17.56 | Qwell | but then which hot wire is it pulling from? |
| 06:18.11 | Qwell | neutral/common is obvious - that'd just steal from the white wire |
| 06:18.23 | drmessano | It probably has blue/red coming from the fan |
| 06:19.30 | drmessano | Most fans I have seen lately are 4-wire |
| 06:19.36 | Qwell | 4 wire? |
| 06:19.45 | drmessano | White/Black/Blue/Red |
| 06:19.48 | Qwell | that would be 2 in 2 out, with fan at source |
| 06:19.49 | Qwell | no? |
| 06:20.02 | Qwell | wait, wait, what's this blue nonsense? |
| 06:20.13 | Qwell | tell me blue controls the light in the fan |
| 06:20.18 | drmessano | yes |
| 06:20.19 | Qwell | (because that doesn't work) |
| 06:20.37 | Qwell | wtf! |
| 06:20.38 | drmessano | Well |
| 06:21.22 | Qwell | another variable is the last thing I need |
| 06:21.57 | Qwell | but I guess it sort of makes sense if this mysterious blue wire isn't connected to the light on the fan |
| 06:22.09 | drmessano | Hang on |
| 06:22.13 | drmessano | Yeah |
| 06:22.51 | drmessano | Black/Red/White control the fan variably.. with a know it should be continously variable, otherwise 3 way with a switch |
| 06:23.05 | Qwell | with a know? knob? |
| 06:23.13 | drmessano | err knob |
| 06:23.19 | Qwell | no knob |
| 06:23.23 | drmessano | know? wtf.. sleepy |
| 06:23.34 | drmessano | blue / white makes the light circuit |
| 06:24.11 | *** join/#asterisk Segnale007 (n=Segnale0@host112-254-dynamic.32-79-r.retail.telecomitalia.it) |
| 06:24.46 | drmessano | Black/White are the fan power, and red is the wiper on the potentiometer inside |
| 06:25.13 | Qwell | and here I thought it would just be 2 wires from the second switch box... |
| 06:25.52 | drmessano | Sounds to me like someone just botched a ceiling fan job |
| 06:25.56 | drmessano | Typical |
| 06:25.58 | Qwell | among other things, heh |
| 06:26.20 | drmessano | I spent 3 hours at a friends helping him put a ceiling fan in... 2 hours was electrical lecture |
| 06:26.24 | Qwell | the switch next to the "funky" 3-way has black and red (?!) wired to terminal 1, leaving terminal 2 empty (ie; always on) |
| 06:26.35 | drmessano | ROFL |
| 06:26.44 | drmessano | Red is from your fan |
| 06:26.46 | Qwell | (white is common - where the heck is red coming from?) |
| 06:26.58 | denon | heh heh |
| 06:27.00 | Qwell | or, so that would be the knob? |
| 06:27.18 | denon | just tell him he doesnt need the red |
| 06:27.22 | denon | and go home early |
| 06:27.23 | drmessano | This three way switch has a center position or not? |
| 06:27.33 | Qwell | no, 2 switches |
| 06:27.44 | drmessano | wait wai |
| 06:27.46 | drmessano | wait |
| 06:27.52 | drmessano | 2 single throw switches? |
| 06:28.05 | drmessano | Just side-by-side |
| 06:28.06 | Qwell | 2 standard 3-way switches |
| 06:28.09 | drmessano | oh |
| 06:28.18 | Qwell | those control the fan |
| 06:28.22 | drmessano | LOL |
| 06:28.27 | drmessano | wow |
| 06:28.35 | Qwell | yeah, no kidding |
| 06:28.41 | drmessano | One should control the fan |
| 06:28.44 | drmessano | Period |
| 06:28.47 | Qwell | mind pm? this is getting a bit spammy here, heh |
| 06:29.04 | drmessano | Hang on, do you control your Skype with it? |
| 06:29.11 | drmessano | Then its on-topic |
| 06:52.13 | *** join/#asterisk matsk (n=Mats@host-90-235-15-91.mobileonline.telia.com) |
| 06:52.40 | *** join/#asterisk MindTheGap_ (n=MindTheG@189.59.202.44) |
| 07:01.43 | jeev | obama for life |
| 07:11.13 | *** join/#asterisk [netman] (n=netman@58.Red-83-50-80.dynamicIP.rima-tde.net) |
| 07:14.46 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 07:21.02 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 07:27.05 | drmessano | I'm trying to decide if allergy season is horrible or if my allergy meds aren't working |
| 07:27.14 | drmessano | I'm leaning towards #2 |
| 07:30.06 | drmessano | So this Skype stuff sounds to me like the magic is all done server-side |
| 07:31.07 | drmessano | That the channel driver for asterisk is a gateway into the "cloud" (god I hate that term) |
| 07:39.32 | *** part/#asterisk matsk (n=Mats@host-90-235-15-91.mobileonline.telia.com) |
| 07:46.27 | *** join/#asterisk Tond (n=t@CPE001a70b2449d-CM00194747ae5e.cpe.net.cable.rogers.com) |
| 07:47.11 | Tond | Hi, anyone here familiar with nagios and Asterisk. I want to link them together |
| 07:47.30 | bkw_ | Tond: so you can get called and bugged when shit dies? |
| 07:48.29 | drmessano | lol |
| 07:48.34 | drmessano | That would piss me off |
| 07:48.47 | Tond | no, so it can check on ast and make sure it is up, but that will be the second part |
| 07:49.03 | Tond | (to make it call me) |
| 07:49.40 | Tond | i have found a plugin check_asterisk, but nowhere it says how to install or use it with nagios. And it is my first try and it is very poorly documented |
| 07:49.46 | drmessano | hang on |
| 07:50.06 | drmessano | So you want nagios to have asterisk call you when asterisk is down o.o |
| 07:50.28 | Tond | no |
| 07:50.47 | Tond | I want nagios to monitor Ast as a service and if other srevices but ast goes down to clal me |
| 07:51.25 | drmessano | What do you want it to do when Asterisk goes down? |
| 07:51.37 | Tond | e-mail me |
| 07:52.08 | Tond | it is mostly for me to check and see the usage, number of conenctions, etc |
| 07:52.33 | Tond | my asterisk doesn't go down often. the only thing that made it crash was the briidging between H323 and SIP |
| 07:53.25 | drmessano | You could always run a cron job to restart asterisk if it's down |
| 07:53.45 | Tond | ya, i have done that |
| 07:53.47 | drmessano | I guess cron jobs or scheduled tasks to restart failed apps aren't cool anymore |
| 07:53.58 | drmessano | I'm so old school :( |
| 07:54.01 | Tond | :) |
| 07:54.40 | Tond | I just want to monitor channel usage numbers, and stuff like that. Plus in some cases asterisk has ben up and running, yet not responsive |
| 07:55.05 | Tond | i had to kill and start it again. And nagios can do IAX and SIP pings |
| 07:55.14 | Tond | well iax and sip checks |
| 07:55.34 | drmessano | if app_pr0n == down_as_hell [ send drmessano@ohnoesitsdown.com O HAI, Restart | ./asterisk_zomgstartnow ] |
| 07:55.56 | Tond | but only if i learn how to use that check_asterisk.pl module... |
| 07:56.33 | drmessano | nagios can't just monitor a socket? |
| 07:57.12 | drmessano | if 5060 = who_wut then goto zomgwtfpanic ? |
| 07:57.33 | Tond | well there is a perl scrip that ties into it, and i ran the perl schipt and using the manager it connected and checked the iax and SIP, etc |
| 07:58.08 | drmessano | ah |
| 07:58.13 | Tond | hrm.. that can work too i guess, but i would need to know how to code that. I dont know any coding |
| 07:58.15 | Tond | :( |
| 07:58.28 | drmessano | nagios doesn't port monitor? |
| 07:58.47 | Tond | i dont know.. i first heard it's name 3 hours ago |
| 07:58.47 | Tond | lol |
| 07:59.09 | drmessano | Oh I see |
| 07:59.21 | Tond | :) |
| 07:59.24 | drmessano | ~nowwhat |
| 07:59.25 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 08:04.59 | *** join/#asterisk af_ (n=getsmart@88-149-241-240.dynamic.ngi.it) |
| 08:05.34 | genji1981 | okay, how do i get a phone, on another net connection, to be a type=friend with my asterisk box behind my router? |
| 08:05.47 | drmessano | Open ports |
| 08:05.55 | drmessano | set up NAT settings |
| 08:06.05 | drmessano | ~sipnat |
| 08:06.05 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 08:06.10 | genji1981 | 5060-5061 plus the ones in rtp.conf right? |
| 08:06.15 | drmessano | No |
| 08:06.18 | drmessano | 5060 and rtp.conf |
| 08:06.45 | genji1981 | k. not working. |
| 08:06.57 | drmessano | and you made the other edits? |
| 08:07.34 | drmessano | localnet, externhost/externip, nat=yes |
| 08:07.39 | drmessano | right? |
| 08:07.39 | genji1981 | ah... hmm. |
| 08:22.24 | *** join/#asterisk grEvenX (n=even@c9A7F8BC3.dhcp.bluecom.no) |
| 08:23.48 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:36.20 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 08:38.03 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-18-172.revip2.asianet.co.th) |
| 08:39.44 | genji1981 | okay, my friend's managed to log into my asterisk box.. but he can't ring out. i can ring his extension, he can hear me, but i can't hear him. Whats going on? |
| 08:40.18 | Tond | genji> sounds like nat issue |
| 08:40.37 | genji1981 | he needs to forward ports towards himself? |
| 08:41.01 | Tond | depedning on what he uses, with cisco phones u ahve to |
| 08:41.14 | genji1981 | hes using SJphone |
| 08:41.16 | Tond | but with most other u dont have to and make sure u ahve nat=yes |
| 08:41.28 | Tond | in ur config check and see if u have nat turned on for him |
| 08:41.40 | genji1981 | yup. nat=yes is there. |
| 08:42.10 | Tond | so u can call him but he can't call u? |
| 08:42.19 | genji1981 | he can't call out, full stop |
| 08:42.22 | Tond | what's the error? |
| 08:42.34 | Tond | do u see his call data coming to u? |
| 08:42.35 | drmessano | Sounds like dialplan on the phone end |
| 08:42.42 | Tond | ya |
| 08:43.05 | Tond | i was just gonna say, look at the context he is being forwrded to |
| 08:43.44 | genji1981 | its a direct copy of one of my other friend entry's. except for the additional 'nat=yes' line |
| 08:44.20 | Tond | so when he makes a call, if u do a debug on ast, do u see the sip messages coming to you? |
| 08:45.02 | genji1981 | nope. suggesting hes not registered? yet , sip show peers, has him there.. and i can ring him. |
| 08:45.49 | Tond | hrm.. if he is behind a router ask him ro reboot the router, exit and open the sjphone again and register |
| 08:45.58 | Tond | my router does that with my cisco phone |
| 08:46.12 | Tond | it's a wierd problem i haven't sat down to figure out |
| 08:48.45 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 08:48.52 | genji1981 | weirdness.. |
| 08:49.07 | genji1981 | sip show peers, Nat column.... N where his line is. |
| 08:49.13 | genji1981 | shouldn't it be Y? |
| 08:50.01 | [gnubie] | brb |
| 08:59.03 | *** join/#asterisk salzh (n=root@58.247.193.78) |
| 08:59.48 | *** join/#asterisk [gnubie] (n=gnubie@cm141.omega112.maxonline.com.sg) |
| 09:02.01 | [gnubie] | hhmmm.. i'm always getting a "denied access to register peer" and rejected: 'Registration Refused' error when trying to connect 2 asterisk-1.4.21.2 servers |
| 09:05.43 | *** join/#asterisk tkbeat (n=tk@p54B94DE8.dip.t-dialin.net) |
| 09:09.22 | *** join/#asterisk zydoon (n=zydoon@41.225.153.114) |
| 09:09.45 | *** part/#asterisk zydoon (n=zydoon@41.225.153.114) |
| 09:18.12 | [gnubie] | i'm always getting this message: CAUSE : Registration Refused |
| 09:18.12 | [gnubie] | <PROTECTED> |
| 09:24.55 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
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| 09:32.30 | [gnubie] | anyone here can help me about iax2 trunking between 2 asterisk servers? |
| 09:32.49 | [gnubie] | i'm always getting a registration refused error.. :( |
| 09:35.11 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 09:39.00 | [gnubie] | i changed from fqdn to public ip address of my register line and it was able to register.. why is this so? i mean, why is it that iax2 doesn't work with fqdn? |
| 09:39.13 | [gnubie] | both boxes uses a dynamic dns so i don't have a permanent ip address.. |
| 09:39.42 | [gnubie] | how can i connect these 2 boxes permanently using fqdn instead of an ip address? |
| 09:50.27 | *** join/#asterisk mateo_au (n=chatzill@12.144.159.231) |
| 09:55.53 | *** join/#asterisk colulu (n=jg@61.141.158.178) |
| 09:56.07 | colulu | Does anyone know if Sangoma 101 card supports asterisk ss7 |
| 09:56.22 | colulu | I have difficulty setting it up. |
| 10:03.11 | bkw_ | colulu: it should have no problem in zaptel mode... |
| 10:08.22 | colulu | bkw_: zapcfg can detect the card |
| 10:08.32 | bkw_ | ztcfg? |
| 10:08.38 | colulu | but within asterisk, it can't recognize the zap channel |
| 10:08.41 | colulu | yes ztcfg |
| 10:08.52 | bkw_ | colulu: why not call sangoma up for support? |
| 10:08.56 | bkw_ | i'm sure its something simple |
| 10:09.11 | bkw_ | colulu: how much is your time worth? |
| 10:11.26 | tzafrir_laptop | ztcfg is not a diagnotic tool . what do you see on cat /proc/zaptel/* |
| 10:15.00 | colulu | bkw_: I don't have much time left to get ss7 to work |
| 10:16.25 | *** join/#asterisk oilinki (n=oil@124.120.18.172) |
| 10:17.25 | colulu | tzafrir: I see 31 channels with "....RED" |
| 10:33.47 | tzafrir_laptop | colulu, "red" means you have no lower-layer link to the remote side |
| 10:34.17 | tzafrir_laptop | likewise, we have no lower-layer link to colulu |
| 10:38.52 | eric2 | on a fresh machine, I installed libpri, wanpipe, zaptel, asterisk-1.2.21.2... then rebooted and the machine doesn't come back to life.... any suggestions? |
| 10:39.03 | eric2 | asterisk 1.4.x I mean |
| 10:46.55 | *** join/#asterisk somjuk (n=chatzill@125.26.211.137.adsl.dynamic.totbb.net) |
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| 11:29.28 | *** join/#asterisk growltiger (n=growltig@ip70-179-54-235.sd.sd.cox.net) |
| 11:43.39 | v4mp | any clue why it would be complaining about signalling before channel when the signalling is specified before the channel ? |
| 11:44.37 | v4mp | this being the error |
| 11:48.50 | v4mp | this is the problem i have |
| 11:50.11 | v4mp | more details and zapata.conf at http://pastebin.com/m621a817e |
| 12:13.01 | *** join/#asterisk shinao1 (n=shinao1@193.238.30.36) |
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| 12:53.23 | *** join/#asterisk stephank (n=urk@82-197-207-120.dsl.cambrium.nl) |
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| 13:21.35 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 13:28.16 | [gnubie] | anyone in here? |
| 13:40.02 | v4mp | nope :/ |
| 13:44.24 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:50.59 | tzafrir_laptop | v4mp, what is the output of: cat /proc/zaptel/* |
| 13:53.21 | v4mp | Span 1: ZTDUMMY/1 "ZTDUMMY/1 (source: RTC) 1" (MASTER) |
| 13:56.44 | *** join/#asterisk MrNaz (n=naz@ppp59-167-150-186.lns4.mel6.internode.on.net) |
| 13:58.43 | v4mp | tzafrir_laptop, that should work ok seen as i dont have any hardware for it ? |
| 14:00.27 | tzafrir_laptop | v4mp, so aparantly the driver wasn't loaded or did not detect the device |
| 14:00.37 | tzafrir_laptop | what do you see on: zaptel_hardware |
| 14:01.19 | v4mp | where would i see that ? |
| 14:02.52 | v4mp | i dont have any hardware purposely for zaptel but from my understanding it can be setup and used without said hardware and just use software |
| 14:08.27 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 14:10.52 | *** join/#asterisk af_ (n=getsmart@88-149-241-240.dynamic.ngi.it) |
| 14:17.42 | [gnubie] | i'm trying to make an inter-asterisk setup via iax2 and based on the 'iax2 show registry' on both servers, they're both registered.. my problem is i cannot place a call.. my /etc/asterisk/iax.conf for both servers are at http://paste.debian.net/18069/ |
| 14:19.34 | tzafrir_laptop | v4mp, your zapata.conf is set for zaptel hardware |
| 14:20.33 | [gnubie] | i'm always getting a call rejected error: No such context/extension |
| 14:22.09 | v4mp | tzafrir_laptop, is it possible to change that? |
| 14:22.57 | tzafrir_laptop | change what? your zapata.conf? |
| 14:23.08 | tzafrir_laptop | A text editor is known to do the trick |
| 14:23.38 | v4mp | no i mean what do i need to change for it to not use hardware |
| 14:24.16 | v4mp | what i pasted on the pastebin link is all thats in the config i still have the origional but made a basic 1 to test with to try fix 1 problem before i added the rest back |
| 14:27.01 | tzafrir_laptop | If you don't have any zaptel hardware you can simply do without zapata.conf |
| 14:27.12 | tzafrir_laptop | because you don't need chan_zap |
| 14:28.28 | v4mp | oh so just delete it ? |
| 14:30.44 | v4mp | why does asterisk do this when it starts |
| 14:30.52 | v4mp | <PROTECTED> |
| 14:41.40 | Juxt | i just installed a fresh 1.4.x asterisk and there's no meetme app. what am i missing? |
| 14:43.21 | v4mp | ;o |
| 14:56.15 | x86 | the bank, the bank, the bank is on fire... we dont need no bailout let the motherfcker burn |
| 15:11.34 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 15:12.42 | v4mp | how would i remove support of zaptel ? |
| 15:14.50 | *** join/#asterisk leif[astricon] (n=Leif@63-255-123-231.ip.mcleodusa.net) |
| 15:15.51 | [TK]D-Fender | v4mp: rm the .so's |
| 15:16.12 | *** join/#asterisk simNIX__ (n=simNIX@82-204-21-111.dsl.bbeyond.nl) |
| 15:16.12 | v4mp | where are hey located ? |
| 15:17.21 | [TK]D-Fender | v4mp: thats what "find" is for. |
| 15:17.36 | v4mp | i can find .o and .c's |
| 15:17.49 | [TK]D-Fender | and .ko's |
| 15:18.41 | v4mp | would it be safe to just uninstall zaptel ? |
| 15:20.23 | [TK]D-Fender | v4mp: Depends how I guess. Why are you looking to do this? |
| 15:20.50 | v4mp | because i dont actually have hardware to go with it im using a standard server |
| 15:21.02 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 15:21.25 | [TK]D-Fender | v4mp: does it pose some kind of problem? It's used for more than just hardware you know... |
| 15:21.45 | [TK]D-Fender | v4mp: Without it you get no MeetMe, Page, or IAX2 trunking |
| 15:21.52 | v4mp | hmm |
| 15:22.01 | v4mp | well the problem i have is this |
| 15:22.55 | v4mp | chan_zap.c:10524 build_channels: Signalling must be specified before any channels are. |
| 15:24.00 | [TK]D-Fender | v4mp: thats a warning, nothing youneed to care about. |
| 15:24.32 | v4mp | [TK]D-Fender, well |
| 15:24.58 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 15:25.01 | leif[astricon] | hello from AstriDevCon! |
| 15:25.11 | leif[astricon] | I don't want to know your name! I just want... |
| 15:25.18 | [TK]D-Fender | leif[astricon]: ! ! ! |
| 15:25.35 | leif[astricon] | [TK]D-Fender: oh hells ya... especially with the girls beach volleyball tournament going on beside the pool |
| 15:25.47 | leif[astricon] | and the giant pool party they are throwing at the hotel today |
| 15:25.51 | v4mp | <PROTECTED> |
| 15:26.01 | leif[astricon] | signalling=<something> |
| 15:26.03 | [TK]D-Fender | v4mp: Doesn't matter. |
| 15:26.19 | leif[astricon] | channel => 1-23 |
| 15:26.22 | v4mp | leif[astricon], i have already done that |
| 15:26.28 | leif[astricon] | shrugs |
| 15:26.32 | v4mp | channel=1 |
| 15:26.35 | [TK]D-Fender | v4mp: unless your system is CRASHING I fail to see how this matters. |
| 15:26.42 | [TK]D-Fender | leif[astricon]: Just said he didn't have any hardware |
| 15:26.46 | leif[astricon] | goes to do some work on CLI aliases |
| 15:27.00 | leif[astricon] | [TK]D-Fender: oh... then the problem is he is loading something other than ztdummy |
| 15:27.08 | leif[astricon] | as you know |
| 15:27.13 | [TK]D-Fender | leif[astricon]: Actually, I don't see a problem at all... |
| 15:27.32 | leif[astricon] | true, but the ERROR would be annoying... so fix it by just loading the appropriate driver and nothing else |
| 15:27.51 | leif[astricon] | if using redhat based system, modify /etc/sysconfig/zaptel to control what loads, then do, "service zaptel restart" |
| 15:28.12 | [TK]D-Fender | leif[astricon]: If I got concerned with every little warning I'd be on kiling sprees DAILY ;) |
| 15:28.38 | v4mp | leif[astricon], but it will load ztdummy |
| 15:28.41 | v4mp | then says this |
| 15:28.42 | [TK]D-Fender | v4mp: So again, what is an actual PROBLEM that is causes? |
| 15:29.21 | v4mp | hmm ok doesn't seem to be complaining like it was |
| 15:29.31 | v4mp | ok if it can be ignored then it can stay there |
| 15:29.51 | [TK]D-Fender | v4mp: Good. |
| 15:30.08 | *** join/#asterisk Juxt (n=denis@adsl-233-30-165.mia.bellsouth.net) |
| 15:30.25 | Juxt | which variable stores the peer name? i want to pass it automatically to VoiceMailMain |
| 15:30.44 | v4mp | [Sep 27 16:29:59] NOTICE[9535]: chan_sip.c:15236 handle_request_register: Registration from '"Gary Syms"<sip:XX@XX>' failed for 'X.X.X.X' - No matching peer found |
| 15:30.50 | v4mp | how would i fix this |
| 15:30.56 | v4mp | the account is in users.conf |
| 15:31.50 | leif[astricon] | Juxt: probably something in the CHANNEL() function |
| 15:33.00 | Juxt | but if my peer is defined as [700] |
| 15:33.01 | Juxt | type=friend |
| 15:33.01 | Juxt | username=700 i should be able to exact the username no? |
| 15:33.50 | *** join/#asterisk simNIX (n=simNIX@82-204-21-111.dsl.bbeyond.nl) |
| 15:33.57 | [TK]D-Fender | Juxt: "SetVar=THEVMBOX=700" , "${THEVMBOX}" |
| 15:34.28 | Juxt | that's easy to do if there was 1 mailbox, i have 70 |
| 15:34.41 | Juxt | that's why a variable would be useful |
| 15:35.25 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 15:35.40 | [TK]D-Fender | Juxt: You missed it.. you put this IN the peer entry for each |
| 15:35.48 | Juxt | oh |
| 15:35.57 | *** join/#asterisk jtodd (i=ubqsnx47@ns2.loligo.com) |
| 15:36.20 | [TK]D-Fender | Juxt: and that var gets sett when the device creates a channel |
| 15:37.18 | Juxt | hmm i could also use accountcode |
| 15:37.59 | [TK]D-Fender | Juxt: Set a custom one as not to tie yourself to CDR related things you'll want to use later |
| 15:39.03 | v4mp | is this correct |
| 15:39.03 | v4mp | http://pastebin.com/m3b5cba42 |
| 15:39.22 | *** join/#asterisk stephank (n=urk@82-197-207-120.dsl.cambrium.nl) |
| 15:39.40 | *** part/#asterisk leif[astricon] (n=Leif@63-255-123-231.ip.mcleodusa.net) |
| 15:40.56 | Juxt | i didnt know you could set vars right in peer, weird |
| 15:42.26 | *** join/#asterisk freakazoid0223 (n=mattc@pool-68-238-180-106.phil.east.verizon.net) |
| 15:46.08 | *** join/#asterisk moy (n=moy@65.200.11.7) |
| 15:46.37 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 15:48.33 | [TK]D-Fender | v4mp: Why are you setting "zapchan" for a SIP entry? |
| 15:49.01 | Juxt | @D-Fender - i added that setvar to the peer and it's not getting set when a channel is cerated |
| 15:49.04 | Juxt | you sure that's possible? |
| 15:49.21 | [TK]D-Fender | Juxt: Yes |
| 15:49.22 | v4mp | [TK]D-Fender, cuz was setting it up to try with sip via voiptalk.org |
| 15:49.43 | [TK]D-Fender | v4mp: Again why ZAPCHAN? It si nto a ZAP channel. |
| 15:50.18 | Juxt | @D-Fender: does this look right? http://pastebin.com/m2ff4d18d |
| 15:50.34 | v4mp | [TK]D-Fender, i dont know because i only managed to find a guide to setup asterisk and said to setup zaptel first |
| 15:50.48 | v4mp | and the users.conf had it there alread i didn't know any diff |
| 15:51.25 | [TK]D-Fender | v4mp: You need to learn to separate the bits that aren't pertinent to your needs and not blindly follow instructions. |
| 15:51.51 | [TK]D-Fender | Juxt: Silly. |
| 15:51.57 | [TK]D-Fender | Juxt: You don't put the QUOTES |
| 15:52.21 | [TK]D-Fender | Juxt: thats because I showed you 2 things back to bac so you can see all the stuff in the middle so it doesn't blur in your eyes |
| 15:52.26 | v4mp | would have helped if i was able to find stuff or peple that could help but i couldn't so had to do what i could find out how to then go from there as shouldn't be that much extra work to get rid/not use things i dont need for time being |
| 15:52.43 | Juxt | ha! thanks it worked |
| 15:52.53 | [TK]D-Fender | Juxt: I showed you MAKING the assignment, and the other USING it (in the dialplan) |
| 15:53.09 | Juxt | yeah i figured, pardon the ignorance |
| 15:53.16 | [TK]D-Fender | v4mp: Read what the paramaters are for before using them next time |
| 15:53.21 | *** join/#asterisk ManxPower (n=manxpowe@159.sub-75-202-18.myvzw.com) |
| 15:54.52 | v4mp | [TK]D-Fender, thats 1 problem i have if i do reading i cant taken it in some things if its very little reading i can... but most the time best chance i got of getting things going is trial and error and maybe a lil help with maybe a lil reading |
| 15:56.10 | [TK]D-Fender | v4mp: be sure to read all of the sample configs and look at how they are used and described. You won't see ZAP things in SIP sections, etc... |
| 15:56.26 | v4mp | [TK]D-Fender, so how would i fix the registration problem |
| 15:57.05 | *** join/#asterisk Porks (i=Porks@unaffiliated/porks) |
| 15:58.27 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 16:00.13 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 16:00.27 | [TK]D-Fender | v4mp: pastebin your actual entry, and the actual SIP DEBUG from CLI and maybe I can point something out for you. |
| 16:02.52 | [TK]D-Fender | ~pb |
| 16:02.52 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:06.27 | v4mp | [TK]D-Fender, the entry i added for the account ? |
| 16:06.32 | v4mp | on users.conf ? |
| 16:06.36 | [TK]D-Fender | v4mp: clearly |
| 16:06.37 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 16:06.41 | v4mp | because i already pasted that here |
| 16:06.42 | *** part/#asterisk Porks (i=Porks@unaffiliated/porks) |
| 16:06.42 | [TK]D-Fender | v4mp: masking only the password |
| 16:06.53 | v4mp | http://pastebin.com/m3b5cba42 |
| 16:09.05 | [TK]D-Fender | v4mp: You have commented out its entry. |
| 16:09.48 | v4mp | http://pastebin.com/m78215a40 |
| 16:10.14 | v4mp | the [6000] part ? because its same with or without that commented |
| 16:10.27 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:11.01 | v4mp | should it even be [6000] ? |
| 16:11.26 | [TK]D-Fender | v4mp: To: "Gary Syms"<sip:V4mpire@gv-hosting.co.uk> <- this is not an "e-mail" |
| 16:11.46 | [TK]D-Fender | v4mp: its looking for an entry named "V4mpire" |
| 16:12.00 | v4mp | so it should be [V4mpire] ? |
| 16:12.26 | v4mp | it must be getting the gv-hosting.co.uk from the email part on users.conf then |
| 16:13.26 | [TK]D-Fender | v4mp: Go read, go try. You don't seem to be following the parallels drawn by every sample out there. |
| 16:13.51 | v4mp | i just used what was already there |
| 16:14.02 | v4mp | i just changed it to my own parts |
| 16:16.27 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 16:18.47 | *** join/#asterisk jtodd (i=yxukhesl@ns2.loligo.com) |
| 16:18.53 | *** part/#asterisk Juxt (n=denis@adsl-233-30-165.mia.bellsouth.net) |
| 16:19.26 | v4mp | [TK]D-Fender, just looked up sample users.conf's and all same as mine |
| 16:20.01 | [TK]D-Fender | v4mp: and what do you see in "sip show peers"? |
| 16:20.39 | v4mp | Name/username Host Dyn Nat ACL Port Status Realtime |
| 16:20.39 | v4mp | voiptalk/84455707 77.240.48.94 5060 Unmonitored |
| 16:20.39 | v4mp | 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline |
| 16:21.01 | *** part/#asterisk disposable (i=disposab@blackhole.sk) |
| 16:21.05 | [TK]D-Fender | v4mp: so yuo can now see what I said before, you commented out your entry, thus you have no entry at all |
| 16:21.45 | v4mp | if u mean the ;[6000] .. i have already said i tryed it with and without it commented same error |
| 16:22.09 | [TK]D-Fender | v4mp: And what you showed me is wrong. |
| 16:22.19 | v4mp | whats wrong with it ? |
| 16:22.20 | [TK]D-Fender | v4mp: fix it and lok at your peers again |
| 16:22.26 | [TK]D-Fender | v4mp: COMMENTED OUT! |
| 16:22.30 | v4mp | ... |
| 16:22.35 | [TK]D-Fender | ]'v4AKA you have no entry! |
| 16:22.43 | v4mp | i've tryed both ways |
| 16:22.53 | v4mp | but im now trying anywa |
| 16:23.03 | [TK]D-Fender | v4mp: FIX IT and show me the NEW list of SIP PEERS |
| 16:23.21 | v4mp | hmm i see |
| 16:23.36 | v4mp | but shouldn't the account V4mpire be a user |
| 16:23.50 | [TK]D-Fender | v4mp: I'm waiting. |
| 16:24.14 | v4mp | Name/username Host Dyn Nat ACL Port Status Realtime |
| 16:24.14 | v4mp | voiptalk/84455707 77.240.48.94 5060 Unmonitored |
| 16:24.14 | v4mp | 6000 (Unspecified) 5060 Unmonitored |
| 16:24.30 | [TK]D-Fender | v4mp: Good, you now at least have DEFINED a user |
| 16:24.43 | [TK]D-Fender | v4mp: Now you can go about making it the right user |
| 16:25.24 | v4mp | how would i set it for name and username then i can only work out how i would set the username or wont name matter ? |
| 16:25.49 | [TK]D-Fender | v4mp: [6000] IS the username. |
| 16:26.02 | v4mp | aye |
| 16:26.08 | [TK]D-Fender | v4mp: And you have somehow figured that "V4ampire" should match it. It shouldn't, and clearly doesn't |
| 16:27.54 | v4mp | well i've sorted that now |
| 16:28.08 | v4mp | same notice but now error of trying to register :S |
| 16:29.03 | *** join/#asterisk stephank (n=urk@82-197-207-120.dsl.cambrium.nl) |
| 16:29.29 | [gnubie] | i'm lost.. anyone can point me here to the right url where i can read more about connecting 2 asterisk boxes via iax2? |
| 16:30.01 | [gnubie] | checking my setup with 'iax2 show registry' on both asterisk boxes tells me that they're registered already |
| 16:30.44 | *** part/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 16:30.58 | [gnubie] | but when trying to place a call, calling from A to B asterisk boxes, tells me that there is "No such context/extension" |
| 16:32.31 | [gnubie] | my call is always rejected.. |
| 16:34.14 | [gnubie] | i'm suspecting that the problem is on my extensions.conf |
| 16:34.19 | v4mp | [TK]D-Fender, i would use xlite1 setting on sip.conf seen as thats the cliet im using correct ? |
| 16:35.20 | [TK]D-Fender | v4mp: SIP doesn't CARE what client you're using. |
| 16:38.15 | *** join/#asterisk Shotygun (n=thorn@82.166.246.66) |
| 16:40.31 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 16:40.31 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:40.53 | v4mp | thats nice |
| 16:40.54 | v4mp | Segmentation fault |
| 16:40.55 | v4mp | :( |
| 16:41.28 | *** join/#asterisk Chris-NB (n=chris@213162066151.public.t-mobile.at) |
| 16:42.05 | *** join/#asterisk moy (n=moy@63-255-104-71.ip.mcleodusa.net) |
| 16:46.56 | v4mp | any clue why when u try dial a number it seg faults ? |
| 16:47.32 | [TK]D-Fender | v4mp: Because of a bug. What version are you running? |
| 16:48.05 | v4mp | 1.4.21.2 |
| 16:48.30 | [TK]D-Fender | v4mp: Try another |
| 16:48.52 | v4mp | -_- |
| 16:49.10 | v4mp | deffo a bug and not sumthing i may have configured wrong ? |
| 16:57.07 | _ShrikE | v4mp: configuration issues should not cause asterisk to segfault |
| 16:57.19 | v4mp | ahh ok >.< |
| 16:57.54 | v4mp | no wonde rnot many people touch asterisk and would prefer to pay for a service cuz its such a pain and no good unless u got the patience lol |
| 16:58.09 | drmessano | wut? |
| 16:58.28 | drmessano | Yeah, nobody touches asterisk man |
| 16:58.38 | drmessano | Who has the time to read and do it right |
| 16:58.50 | v4mp | not nobody |
| 17:01.20 | *** join/#asterisk oej (n=olle@63-255-123-231.ip.mcleodusa.net) |
| 17:02.07 | _ShrikE | v4mp: I put about 50,000 calls per day though my asterisk servers adn I wouldn't trade it for any commercial solution |
| 17:02.27 | jblack | I'm having a minor problem with ${EPOCH} not matching the Calldate in the cdr. I'm looking at wiki-Asterisk+variables, and I don't see a variable that represents the calldate. |
| 17:02.29 | [gnubie] | [Sep 28 00:59:29] WARNING[27584]: chan_iax2.c:7775 socket_process: Call rejected by 11.22.33.44: No such context/extension |
| 17:02.41 | v4mp | o_O |
| 17:03.06 | drmessano | "no wonde rnot many people touch asterisk.." <-- a LOT of people are very successful learning and setting up asterisk. I don't see how you can make a baseless statement like that when you're obviosuly knew to asterisk and have no idea how many people use or have problems with Asterisk. |
| 17:03.15 | drmessano | new* |
| 17:03.23 | drmessano | It |
| 17:03.37 | jblack | Usually the calldate and ${EPOCH} are the same, but sometimes they're off by a second. DOes anyone know of a variable that stores the calldate? |
| 17:03.56 | *** join/#asterisk oej_ (n=olle@63-255-123-210.ip.mcleodusa.net) |
| 17:04.39 | drmessano | It's unfortunate you're having problems, but everyone struggles at something new. Somehow this horribly difficult application has had hundreds of thousands of downloads and users over the years, so apparently "many people" isn't as "many" as you seem to think. |
| 17:05.10 | jblack | v4mp: Yeah, it's not easy software to learn how to use. |
| 17:05.14 | v4mp | whats the easiest option to change the version without redoing everything from scratch ? |
| 17:05.39 | drmessano | I don't think you would last 10 minutes in #apache struggling with a problem, and then throwing out a "no wonder nobody uses Apache, it's too hard". You would be banned or laughed at, at the very least. |
| 17:05.49 | _ShrikE | v4mp: just download a different version, recompile, and go...... |
| 17:05.51 | jblack | v4mp: Are you trying to upgrade asterisk? |
| 17:06.12 | _ShrikE | I think he is trying to downgrade |
| 17:06.15 | jblack | drmessano: Yah, they're harsh. |
| 17:06.15 | v4mp | jblack, well the version i have seg faults |
| 17:06.48 | v4mp | apache i can do fine |
| 17:07.13 | jblack | I'm not sure why you're not answering my question. |
| 17:07.24 | drmessano | jblack: That, and the claim of "many people not touching it because it's too hard" wouldn't go very well for the #1 httpd on the web. I'd say a statement about asterisk is nearly as baseless and silly. |
| 17:07.48 | drmessano | But hey, when all else fails, call the app crap |
| 17:07.59 | [gnubie] | i'm always getting a call rejected error: No such context/extension |
| 17:08.01 | jblack | Not many people do double entry accounting. |
| 17:08.09 | [gnubie] | i'm lost.. anyone can point me here to the right url where i can read more about connecting 2 asterisk boxes via iax2? |
| 17:08.26 | jblack | Sometimes, the domain of a problem requires sophisticated hardware. * is one such example. |
| 17:08.38 | jblack | pardon, requires sophisticated solutions. |
| 17:09.24 | [gnubie] | 'iax2 show registry' on both asterisk boxes tells me that they're both registered to each other but placing a call from one asterisk box to another tells me that no such context/extension |
| 17:09.26 | v4mp | so which is best version to change to ? |
| 17:09.29 | jblack | There's not necessarily anything wrong here. Asterisk may be too hard for him. Doesn't make him a bad person. They're just not a good match together, and he can go work on something that matches his experience levels. |
| 17:10.20 | jblack | v4mp: No version of asterisk is any easier or harder than another. If you found 1.x.x too hard to learn, you'll find 1.y.y too hard to learn too. |
| 17:11.43 | v4mp | jblack, im not saying 1 is easier than another i would prefer a version that hopefully is less likely to seg fault |
| 17:12.17 | jblack | oh man. I'm gonna need to futz up these recordings with ${UNIQUEID}, aren't I..... seems odd to me that there's no ${CALLDATEEPOCH} variable. |
| 17:12.52 | *** join/#asterisk tvirus (i=TheVirus@c-68-54-165-28.hsd1.md.comcast.net) |
| 17:12.59 | jblack | v4mp: If you can tell me what exactly it's segfaulting over, maybe I can make a suggetsion. I can't do that with "it's broken, cause it segfaults" |
| 17:13.05 | v4mp | hmm i just found a guide that mentions asterisk-sounds not haven't that wouldn't cause a seg fault would it ? |
| 17:13.32 | seanbright | v4mp: no |
| 17:13.33 | _ShrikE | shouldn't |
| 17:13.56 | v4mp | ok so which version would u recommend then ? |
| 17:14.06 | jblack | perhaps if you turn on various debugging things, watch your log, compile * with debugging symbols, enable coring and gdb the core.. you can tell us where it's bombing out. |
| 17:14.18 | seanbright | v4mp: what version are you having problems with? |
| 17:14.37 | jblack | I don't have a suggestion, because "it's segfaulting" isn't specific enough to steer you. |
| 17:14.45 | v4mp | 1.4.21.2 |
| 17:14.52 | jblack | that's like saying "My car doesn't start" |
| 17:15.01 | v4mp | jblack, when u try call a number it seg faults |
| 17:15.35 | jblack | There you go. "My car doesn't start on cold days" is a step in the right direction. |
| 17:15.41 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 17:15.46 | v4mp | tho i dunno if that bit is setup right but from what u said that wouldn't cause a seg fault so dunno |
| 17:15.46 | seanbright | v4mp: you could try a couple things. downgrade to 1.4.21.1 or lower and see if that works. |
| 17:15.55 | seanbright | v4mp: or you could try the latest 1.4 from SVN |
| 17:15.55 | v4mp | jblack, i had already mentioned that tho |
| 17:16.14 | v4mp | hmm |
| 17:16.49 | seanbright | v4mp: svn co http://svn.digium.com/svn/asterisk/branches/1.4 |
| 17:16.49 | jblack | Ok. see my prior advice about debugging. If you can provide a function, that'll help. |
| 17:16.51 | seanbright | if you wanted the latest and greatest. |
| 17:16.59 | jblack | and see if you can't get debugging quite verbose. |
| 17:17.25 | jblack | perhaps asterisk -dcvvvvvvvvvvvvvvv |
| 17:17.38 | seanbright | but jblack makes a good point, a gdb backtrace with debugging symbols enabled would be the most helpful in terms of getting things fixed for you. |
| 17:17.55 | jblack | perhaps asterisk -dfcvvvvvvvvvvvvvvv |
| 17:18.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:18.23 | jblack | If you check the asterisk man page, you can go through the list of options of various flags you can give. |
| 17:18.29 | seanbright | when in doubt, add another 'v' |
| 17:18.33 | jeev | ahhhhhhh |
| 17:18.36 | v4mp | i didn't see much but will check |
| 17:19.02 | jblack | By the way, these are the sorts of skills a typical system administrator should have. |
| 17:19.03 | jeev | you guys better vote obama or asterisk will change it's name to jeev |
| 17:19.50 | jblack | v4mp: from what you've shown so far, you lack the appropriate skillset. Asterisk is probably not for you. |
| 17:21.23 | v4mp | jblack, wont learn until u try tbh |
| 17:21.45 | jblack | Learn your core skillset with something easier. :) |
| 17:22.57 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 17:22.57 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:23.59 | v4mp | jblack, i can do many other things but no point learning sumthing im not interested in and not going to use |
| 17:24.06 | v4mp | http://pastebin.com/m330cc92d is the debug |
| 17:24.20 | *** join/#asterisk Dystemper (n=mark@ip72-198-100-29.ok.ok.cox.net) |
| 17:24.49 | *** join/#asterisk growltiger_ (n=growltig@ip70-179-54-235.sd.sd.cox.net) |
| 17:28.28 | *** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net) |
| 17:29.22 | Qwell | newegg should accept thanks instead of money for products |
| 17:34.55 | jblack | qwell: Mind a couple quick questions about uniqueid? |
| 17:35.05 | Qwell | sure |
| 17:35.39 | jblack | It looks like the current format of uniqueid is asterisk-PID-CALLDATEEPOCH . |
| 17:35.45 | jblack | Would that be about right? |
| 17:36.24 | Qwell | uniqueid? |
| 17:36.42 | jblack | ${UNIQUEID} is a channel variable that provides a unique id for the call. |
| 17:37.06 | Qwell | there's also an incrementing number at the end |
| 17:37.09 | jblack | A typical example of one would be "asterisk-3907-1218652439.8826" |
| 17:37.20 | Qwell | see line 809 of channel.c in branch 1.4 |
| 17:37.38 | Qwell | it uses system name if available, otherwise it doesn't |
| 17:37.42 | jblack | I suppose I should check the sources myself. sorry. |
| 17:37.46 | v4mp | so thats the end of my help ? |
| 17:38.05 | Qwell | I don't see where pid is being added to it. |
| 17:38.24 | jblack | I'll see what it is from the source. it's that epoch I'm interested in. |
| 17:38.38 | Qwell | yes, it's just time |
| 17:38.47 | Qwell | (long)time(NULL) |
| 17:39.32 | jblack | And the other of the question... there's an inherent promise that uniqueid is unique. Is the uniqueid format pretty stable, as in unlikely that I'll get myself in trouble by taking the calldate epoch out of uniqueid? |
| 17:40.02 | Qwell | I suspect it could change at any time |
| 17:40.03 | jblack | I.E. you guys aren't likely to arbitrarily swithch from using the epoch to something rand() based, right? |
| 17:40.19 | jblack | Ok. Darn. |
| 17:40.40 | Qwell | it may be unlikely, but.. |
| 17:41.02 | jblack | Yeah. Getcha. I could get away with it for six months, or six years, but I takes my chances. |
| 17:42.00 | jblack | makes sure http://www.voip-info.org/wiki-Asterisk+variables is up to date that there's no ${CALLDATEEPOCH} variable and that there's no bug filed for it |
| 17:42.29 | *** join/#asterisk growltiger (n=growltig@ip70-179-54-235.sd.sd.cox.net) |
| 17:44.37 | jblack | Thanks much qwell |
| 17:44.50 | jblack | v4mp: I've given you all the help I can. |
| 17:45.15 | jblack | v4mp: If * is segfaulting on you, you're gonna have to debug it to know what's wrong, so that you know what to avoid. |
| 17:45.16 | [gnubie] | hello.. |
| 17:45.21 | [gnubie] | anybody can help me here? |
| 17:45.47 | jblack | Well, they have to be "here", and here (as in not asleep) |
| 17:46.07 | jblack | And they may not be able to help you if you start asking for trixbox help. |
| 17:46.08 | v4mp | jblack, i pasted a link to the debug because im not sure about much cuz i haven't learnt much about * yet |
| 17:47.26 | [gnubie] | i'm having a problem with an inter-asterisk trunking using iax2.. both servers are registered to each other but i always get a "no such context/extension" error everytime i place a call |
| 17:47.36 | jblack | v4mp: Now you can try the things I suggested. |
| 17:48.09 | v4mp | what to try a diff version jblack ? |
| 17:48.18 | jblack | basically, get * running with debugging symbols, get * to dump a core, and backtrace the segfault. |
| 17:48.21 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 17:48.35 | v4mp | :/ |
| 17:48.56 | v4mp | thats whats on the link which is what happens when it seg faults |
| 17:49.03 | [TK]D-Fender | v4mp: Just install a different version. |
| 17:49.27 | jblack | he wants to know which one. |
| 17:49.40 | [TK]D-Fender | v4mp: You aren't even comfortable configuring * properly so I would forget about trying to trace segfaults and serious programming errors. |
| 17:49.56 | *** join/#asterisk oej (n=olle@63-255-123-210.ip.mcleodusa.net) |
| 17:50.00 | [TK]D-Fender | v4mp: try the RC in the topc. Then try the version prior. |
| 17:50.15 | jblack | I don't want to give him a version to downgrade to, because whatever is triggering the segfault in the version he's using could be present in earlier versions too. |
| 17:51.16 | [TK]D-Fender | jblack: You're right so he should spend an indeterminate amount of time tracking DEV bugs instead... |
| 17:51.21 | jblack | maybe it's some simple permissions thing, or maybe he's loading a module that uses a mismatched library. |
| 17:52.08 | jblack | It's not my job to decide how badly he wants an accurate answer to his question. :P |
| 17:53.09 | jblack | Any normal person would just try several versions to avoid that effort, but he's here, asking for a specific version to avoid a problem. :) |
| 17:54.28 | v4mp | jblack, reason being i asked for a specific version was incase from what i pasted when the seg fault happens could have been fixed on other versions but if it doesn't help to track it for anyone then i can just try any versions |
| 17:55.29 | jblack | v4mp: Ok. Then let's be siple about it. You have a nonspecific version, so there's no way to specifically avoid it. As [TK]D-Fender suggested, just try other versions until you're happy. |
| 17:55.59 | jblack | pardon, you have a nonspecific problem. |
| 17:56.11 | v4mp | ahh ok |
| 17:56.45 | jblack | Do me a favor? |
| 17:58.28 | jblack | never mind |
| 17:59.25 | v4mp | what was u going to say ? |
| 17:59.41 | jblack | Whine less next time. |
| 17:59.58 | v4mp | i didn't whine :o |
| 18:00.47 | jblack | Fair enough. Do me a different favor? |
| 18:00.54 | [gnubie] | hello..? |
| 18:00.57 | jaytee | R.I.P. Paul Newman |
| 18:01.05 | v4mp | whats that ? stop asking so many questions ? :p |
| 18:01.09 | jblack | [gnubie]: Can't you see Im busy larting someone? Hold up.. geeze. |
| 18:01.38 | jblack | v4mp: Modify the way you talk so that I'm less inclined to think I'm hearing whining. :) |
| 18:02.18 | jblack | [gnubie]: Turn on verbose and debug, and see where it's trying to send the calls. |
| 18:02.41 | jblack | You probably need to set the context in the iax2 accounts in iax2.conf, or you're trying to go to s@... etc |
| 18:03.43 | jaytee | and if it's an IAX trunk and you're following the book the examples have the contexts ass-backwards |
| 18:03.59 | v4mp | sorry but i have trouble wording things how others prefer always have done i tend to word it how i undestand so then hopefully the reply keeps on track for me to understand and not get lost which i sumtimes do not realising what was said was quite simple lol <<< probz sounds weird |
| 18:04.04 | [gnubie] | my /etc/asterisk/iax.conf for both servers are at http://paste.debian.net/18069/ |
| 18:05.15 | jblack | v4mp: Ok. You're being open and polite, but I read what you just said as "whine whine whine". |
| 18:05.32 | v4mp | pfft lol |
| 18:05.52 | jaytee | sounded like whining to me |
| 18:06.08 | v4mp | u guys just have it in for me dont u :p |
| 18:06.23 | jblack | Yay! asterix redux |
| 18:06.37 | v4mp | redux ? |
| 18:07.00 | jblack | Nah. Just come into the channel with the understanding that people are busy with their own problems, and they take care of yours only by putting asside theirs. :) |
| 18:07.07 | jblack | [gnubie]: ok |
| 18:07.22 | jaytee | [gnubie], you have the inbound_trunks context on server A and server B, correct? |
| 18:07.29 | [gnubie] | yes |
| 18:07.41 | [gnubie] | they both are registered to each other |
| 18:07.49 | jaytee | as it should be, yes |
| 18:08.06 | jblack | try insecure=invite on both. |
| 18:08.18 | jblack | And can we safely assume that you really, really, did iax2 reload these two configs? |
| 18:08.20 | jaytee | or port,invite |
| 18:08.42 | jblack | If he's failing auth, he'll go to the default context, which is unset on both of these. |
| 18:08.50 | jaytee | ah |
| 18:09.19 | jblack | [gnubie]: Understand what I just said, or do you need it unpacked some? |
| 18:09.29 | [TK]D-Fender | [gnubie]>but when trying to place a call, calling from A to B asterisk boxes, tells me that there is "No such context/extension" |
| 18:09.33 | [gnubie] | jblack: where? on the [general] context of the iax.conf ? |
| 18:09.46 | [TK]D-Fender | FFS its a context & exten error. Look at all this running around liek headless chickens |
| 18:10.14 | [TK]D-Fender | Look at the DEBUG to see the number coming in, the peer matched, and the CONTEXT its looking in |
| 18:10.19 | jblack | [gnubie]: tk is right. Just paste a traced call, please. |
| 18:10.31 | v4mp | hopefully this version works |
| 18:10.32 | [gnubie] | ok |
| 18:10.47 | jaytee | yeah! cuz like I'm all grief shattered over Paul Newman and this concentrating thing just isn't working for me |
| 18:11.03 | [TK]D-Fender | goes and looks for a new slad dressing... |
| 18:11.08 | [TK]D-Fender | salad* |
| 18:11.12 | jaytee | ol |
| 18:11.13 | jaytee | lol |
| 18:11.14 | jblack | [TK]D-Fender: Can I cheer you up with a beer... or maple syrup, or a hockey game, or something? |
| 18:12.10 | [TK]D-Fender | jblack: I don't particularly like beer, can't be bothered to watch sports, but I'll take you up on the syrup :) |
| 18:12.23 | jblack | hmmmm. air guitar, air hockey.. think about it |
| 18:12.34 | jblack | Ok! I'll see what I can do |
| 18:12.58 | v4mp | this should be fine shouldn't it seen as it was from v1.4.* anyway |
| 18:13.00 | v4mp | Your Asterisk modules directory, located at |
| 18:13.00 | v4mp | <PROTECTED> |
| 18:13.24 | jblack | v4mp: Stop there. Use pastebin.ca please. |
| 18:13.53 | jblack | um, yeah, make sure you have the asterisk modules. I bet that could cause a segfault. |
| 18:14.40 | [TK]D-Fender | jblack: More like real guitar, and no hockey. |
| 18:14.44 | v4mp | well the modules were from the previous version which was 1.4.* i've just used svn for 1.4 |
| 18:15.16 | jblack | Make sure everything thing matches everyting else, versioning wise. |
| 18:16.00 | v4mp | is res_config_mysql not used on newest version ? as its saying it wont be initialized shouldn't cause a problem seen as tho its only a warning |
| 18:16.03 | jaytee | See, this is another sign of your tragic space dementia, all paranoid and crotchety. Breaks the heart. |
| 18:16.46 | *** join/#asterisk troubled_ (n=troubled@pdpc/supporter/sustaining/troubled) |
| 18:16.57 | jblack | Some day, I'm gonna lose it. I'm going to strap bleach all over myself, and take over a dye factory. |
| 18:17.32 | jblack | "This is NOT COLORSAFE, so back off!" |
| 18:17.37 | jaytee | lol |
| 18:18.42 | v4mp | jaytee, no its because i would prefer to have it setup nice and not have chance of more problems i reckon its better to ask early on rather than get furthe in then not realise then get wound up to realise it was just a silly thing of sumthing that wasn't loaded at the beginning :) |
| 18:19.23 | jblack | Now you understand why I wasn't willing to recommend version hopping, and instead track down the problme? |
| 18:19.37 | jaytee | well, you could always use Gparted and reinstall everything from scratch |
| 18:19.54 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 18:20.21 | [gnubie] | it's here already => http://paste.debian.net/18089/ |
| 18:20.54 | jblack | [gnubie]: Not enough debugging. set verbose 9, set debug 9 and rerun, please |
| 18:21.05 | jblack | Oh, geeze. nevermind |
| 18:21.11 | v4mp | jblack, see i asked how i could find it but i was just told to debug which i did and yet i dont understand it so it wouldn't have helped me |
| 18:21.13 | jblack | takes off the football helmet |
| 18:21.14 | [gnubie] | jblack: i actually set to 10 |
| 18:21.29 | jblack | [gnubie]: I wasn't thinking |
| 18:21.29 | [gnubie] | jblack: 'core set verbose 10' |
| 18:21.33 | jblack | You're trying to go to "80211@family" |
| 18:21.47 | jblack | Never mind. gah |
| 18:21.58 | jblack | I need you to track the call on the _other_ side. |
| 18:22.24 | jblack | You're debugging the calling side, I need you to debug the called side. |
| 18:22.42 | [gnubie] | ok |
| 18:25.03 | [gnubie] | this is what i got from the called side: |
| 18:25.24 | jblack | drumrolls |
| 18:26.13 | jblack | feels his lips getting tired |
| 18:26.32 | v4mp | nearly asked a silly question but the problem lies my end |
| 18:26.36 | [gnubie] | [Sep 28 02:26:01] NOTICE[3067] chan_iax2.c: Rejected connect attempt from 1.2.3.4, request '211@peering_first' does not exist |
| 18:27.22 | jblack | Uhhhhh. |
| 18:27.59 | jblack | So, http://paste.debian.net/18069/ isn't your iax2 configs |
| 18:29.12 | v4mp | works :D |
| 18:29.50 | [gnubie] | jblack: i think i got it already |
| 18:30.11 | [gnubie] | i just "included" the other contexts where to call |
| 18:30.35 | jblack | marks [gnubie] down for "lies about whether he's reloaded his configs" |
| 18:31.42 | jblack | Aww man! Japan is going to do 1 gigabit fiber! |
| 18:31.44 | [gnubie] | jblack: i reload.. |
| 18:34.00 | v4mp | anyone got a link to a page to get more of an undrstanding how to setup extensions ? |
| 18:34.07 | jblack | ~book |
| 18:34.08 | jbot | i guess book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 18:35.17 | jaytee | America 2030: "Hey, I heard on the tech news channel that Japan is going 1TB fiber!" "Damn! Wish we could get some of that! Hand me that 100MB Cat5 cable over there, would ya?" |
| 18:36.36 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 18:36.40 | jblack | Go free market. See how well it works for housing, mortgages, credit card loans, payday lenders, and food safety. Doing wonders for an internet around you. |
| 18:37.00 | jaytee | I want MY bailout package please!! |
| 18:37.25 | jblack | You? Do you drive a porsche? |
| 18:37.27 | jaytee | I can screw up just as good as any of those Wall Street guys! |
| 18:37.42 | jaytee | no, I drive an old beat up Subaru |
| 18:37.56 | jblack | Silly rabbit. Bailouts are for rich folk. |
| 18:38.25 | jblack | If you need a place, use the subaru |
| 18:38.51 | jblack | Cause they don't want you on the street, keying their mercedes. |
| 18:40.33 | v4mp | application for voicemail would be exten => name or num,priority,VoiceMail() ? |
| 18:41.08 | jaytee | I heard there was this thing called a book ......... |
| 18:41.11 | jblack | v4mp: Read the (#*@& book |
| 18:41.20 | v4mp | i did |
| 18:41.27 | jblack | Read it again. |
| 18:41.38 | v4mp | . |
| 18:41.39 | jblack | And this time, pay attention to the "word" things in it. |
| 18:41.40 | jaytee | no! you didn't! or you wouldn't be asking that. try the index! |
| 18:42.05 | v4mp | jaytee, i looked at the extensions part on how to do extensions |
| 18:42.31 | jaytee | good, now go back and read some more, it gets real fun when you get to AMI and AGI |
| 18:43.11 | jblack | v4mp: We're only going tell you the things taht are in the book, because the book _HAS THE ANSWERS_ |
| 18:43.59 | jblack | If you want someone to read the book to you, I'm available for a fair cost. If you want someone to read it to you for free, ask mommie to read you a bedtime story. |
| 18:44.21 | v4mp | well even an extension setup in extensions.conf for voicemail doesn't even work |
| 18:44.53 | jaytee | did you setup voicemail? |
| 18:45.48 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:46.22 | [gnubie] | thanks jblack and [TK]D-Fender |
| 18:46.31 | [gnubie] | now, i can rest already.. :) |
| 18:46.34 | [gnubie] | waves |
| 18:46.46 | [gnubie] | gtg now.. |
| 18:46.50 | [gnubie] | thanks once again.. |
| 18:48.07 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 18:49.37 | *** join/#asterisk scardinal (n=supreme@90.184.100.170) |
| 18:50.22 | *** join/#asterisk LND (n=lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk) |
| 18:52.55 | *** part/#asterisk LND (n=lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk) |
| 18:57.25 | v4mp | ok i've followed the instructions in book and it still says extension is not found |
| 18:58.40 | *** join/#asterisk dlewis (i=4578ad9f@about/security/staff/dlewis) |
| 18:58.51 | dlewis | anyone know about this: http://www.xeeta.com/ |
| 19:09.00 | *** join/#asterisk oej (n=olle@63-255-123-231.ip.mcleodusa.net) |
| 19:10.07 | *** join/#asterisk scardinal (n=supreme@90.184.100.170) |
| 19:11.50 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:13.25 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 19:19.59 | AndyMillar | are there any guidelines on what the best "gui" for asterisk is |
| 19:22.08 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 19:22.50 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 19:23.12 | [TK]D-Fender | AndyMillar: Whichever satisfies more of you needs with the least number of things you'd rather not live with |
| 19:23.37 | [TK]D-Fender | AndyMillar: Whats better, a Ferrari or a minivan? |
| 19:23.57 | [TK]D-Fender | AndyMillar: And at what price? |
| 19:24.53 | adr3nalin3 | Is there a way to get the ring group name to display on the phone instead of just the incoming number? |
| 19:25.41 | adr3nalin3 | I need a way to identify calls from different numbers going to the same person. |
| 19:25.53 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:32.04 | *** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk) |
| 19:33.06 | maxxim | hi. where can i find the whole list of usable functions for aster 1.6 ? like SayNumber, SayDigits,etc... |
| 19:38.36 | *** join/#asterisk SteveTotaro (n=Administ@pool-141-157-95-245.balt.east.verizon.net) |
| 19:42.29 | Qwell | maxxim: those aren't functions |
| 19:49.02 | [TK]D-Fender | maxxim: "core show applications" , "core show functions |
| 19:49.09 | [TK]D-Fender | adr3nalin3: Changes the callerID |
| 19:51.06 | AndyMillar | [TK]D-Fender: any reccomendations? i've only seen freepbx before |
| 19:51.20 | AndyMillar | [TK]D-Fender: and price isn't much of an issue |
| 19:51.45 | [TK]D-Fender | AndyMillar: Switchvox, ScopServ, Thirdlane, Asterisk-GUI. DOZENS out there. |
| 19:52.58 | AndyMillar | [TK]D-Fender: used many? |
| 19:53.33 | *** join/#asterisk l8router (n=me@d122-109-92-165.sbr12.nsw.optusnet.com.au) |
| 19:56.11 | [TK]D-Fender | AndyMillar: Only ScopServ with any frequency. |
| 19:57.14 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 19:58.16 | v4mp | ok i folloed chapter 4 on the book and skipped the zaptel section seen as im not using it atm |
| 19:58.20 | v4mp | and its saying |
| 19:58.33 | v4mp | NOTICE[31778]: chan_sip.c:14413 handle_request_invite: Call from 'V4mpire' to extension '500' rejected because extension not found. |
| 19:58.44 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 19:58.57 | v4mp | any clues what i could have somehow done wrong for this to happen ? |
| 19:59.23 | AndyMillar | [TK]D-Fender: easy to use? |
| 19:59.49 | *** join/#asterisk slingr (i=santas@will.one.day.hack-the-pla.net) |
| 20:00.55 | slingr | hey all |
| 20:01.05 | [TK]D-Fender | v4mp: Means exactly what is says. You have no extension to match 500 in the context that it is being looked for in. |
| 20:01.22 | [TK]D-Fender | v4mp: this isn't a "clue", this is the 100% answer |
| 20:01.32 | slingr | anyone recommend a place to research pci cards for asterisk? |
| 20:01.36 | v4mp | [TK]D-Fender, but i followed te book i will show the said configs |
| 20:01.38 | [TK]D-Fender | AndyMillar: pretty much |
| 20:02.12 | AndyMillar | [TK]D-Fender: what's it like for installing etc, i need ti for a ~80 user install and i want it to "just work(tm)" |
| 20:02.43 | [TK]D-Fender | AndyMillar: they have a comprehensive install guide, etc. www.scopserv.com . Check it out, call them up, etc |
| 20:03.07 | AndyMillar | i shall do :) |
| 20:03.51 | slingr | herro? |
| 20:03.52 | slingr | lol |
| 20:05.01 | adr3nalin3 | Can anybody tell me what I might be able to do to fix this? http://pastebin.com/m64a678a5. It says I have no d-channel. It is a PRI and the switch type is NI2. |
| 20:05.08 | drmessano | Interesting |
| 20:05.29 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 20:05.30 | adr3nalin3 | My PRI sometimes comes online for about 5 minutes that either goes into alarm or just doesn't work. busy signal |
| 20:05.48 | drmessano | I had loaded up a Voxee account way back when, with $5 in termination.. used like a couple minutes of it |
| 20:05.53 | drmessano | Then they went AWOL |
| 20:06.22 | drmessano | Went back there the other day, and they relaunched.. I coverted my account to the "new system" and have $4.95 in termination sitting there |
| 20:06.23 | drmessano | lol |
| 20:06.51 | adr3nalin3 | This is my zaptel.conf....http://pastebin.com/m3ba79980 |
| 20:07.57 | slingr | can anyone point me in the right direction for a decent FXO card to start out with? I've used asterisk before, but it was all IP-based and my company wants me to experiment for possible implemenation here (4 phones) |
| 20:08.56 | adr3nalin3 | digium.com |
| 20:09.35 | adr3nalin3 | AndyMillar: |
| 20:09.38 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:09.42 | adr3nalin3 | oops wrong window |
| 20:10.27 | v4mp | righ ok i got the test dial working |
| 20:10.42 | v4mp | but its not echoing whats said when it should |
| 20:14.16 | drmessano | http://www.slash7.com/pages/vampires <-- Ironic |
| 20:15.24 | [TK]D-Fender | slingr: How many lines? And what kind of phones? |
| 20:17.10 | slingr | [TK]D-Fender> to be honest... i'm a little lost... i've used asterisk with a LES.net before and some IP phones, but this is my first journey into the analog world. It will be two phone lines (one voice, 1 fax) and around 4 extensions. I can purchase new phones if needed (would think IP phones would be easier ?) |
| 20:18.01 | [TK]D-Fender | slingr: Everything depends on your wiring, budget, what kind of use, etc. You looking at ditching VoIP for PSTN and jsut going analog? Just 1 channel? |
| 20:18.16 | [TK]D-Fender | slingr: Is this business or personal? |
| 20:19.25 | slingr | well, i'm looking to experiment first. i have a extra line at home that I would like to run asterisk with, |
| 20:19.58 | slingr | well to start.. its personal... depending on how familiar i get with it I may bring it over into my business... I've been playing with asterisk for awhile, but its all be on the voip end |
| 20:21.39 | slingr | i.e. before i just had a phone number and ip termination though les.net ... so i didn't need a pci card for my server |
| 20:21.41 | *** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk) |
| 20:21.48 | maxxim | exten => s,4,Background(transfer) |
| 20:21.56 | maxxim | exten => t,1,Goto(s,4) |
| 20:21.59 | slingr | now that I'm going to be bringing two lines into it, i'm not sure what would be best |
| 20:22.35 | maxxim | why it didnt work? when music finish, i enter into status is 'UNKNOWN' |
| 20:24.43 | v4mp | ok i have the test call setup to echo what is said but its not echoing |
| 20:24.53 | v4mp | this is the extensions.conf and sip.conf http://pastebin.com/m5bcbd85e |
| 20:26.42 | [TK]D-Fender | slingr: Were you dissatisfied with les.net? |
| 20:26.52 | *** join/#asterisk oej (n=olle@63-255-123-231.ip.mcleodusa.net) |
| 20:27.23 | [TK]D-Fender | maxxim: becasue "t" is not "s". it does not magically jump through the alphabet. |
| 20:27.45 | [TK]D-Fender | maxxim: and these extens have special purposes. Go read up on your dialplan basics again. |
| 20:27.47 | [TK]D-Fender | ~book |
| 20:27.48 | jbot | i heard book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 20:27.49 | [TK]D-Fender | ^^^^^^^^ |
| 20:28.53 | *** join/#asterisk shine (n=stroll@lamantin.achamo.net) |
| 20:29.40 | maxxim | [TK]D-Fender> i want that if no extension is entered during playing 'transfer' music, to make it to repeat using Goto... is it not right? |
| 20:29.53 | maxxim | t -> timeout (as i read) |
| 20:29.56 | *** join/#asterisk oej_ (n=olle@63-255-123-210.ip.mcleodusa.net) |
| 20:30.01 | maxxim | may be i'm not doing it right |
| 20:30.07 | maxxim | tell me pls |
| 20:31.51 | [TK]D-Fender | maxxim: if you are making this as an IVR, then you need to have set "autofallthrough=no" Where appropriate |
| 20:32.42 | maxxim | [TK]D-Fender> yep, as an IVR, let me try, thanks |
| 20:34.14 | [TK]D-Fender | maxxim: Google it up.... you'll see where |
| 20:34.34 | riddlebox | hrmm I guess snom's website is having problems, I cannot get to the firmware downloads |
| 20:35.14 | slingr | woop |
| 20:35.16 | slingr | lost connection |
| 20:35.29 | slingr | dunno if you responded to me TK |
| 20:36.56 | maxxim | [TK]D-Fender> cool! it works... but it waits 3 sec after that music finish. Where can i set this timeout, just to be a goto right after the music finished |
| 20:36.59 | maxxim | thanks [TK]D-Fender!!! |
| 20:37.01 | *** join/#asterisk ZefK (n=ZefK@wsc-fo.b.astral.ro) |
| 20:37.35 | [TK]D-Fender | maxxim: "core show function TIMEOUT" |
| 20:38.24 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 20:39.11 | maxxim | [TK]D-Fender> you know so much :) |
| 20:39.15 | slingr | :) |
| 20:39.32 | [TK]D-Fender | slingr: Were you dissatisfied with les.net? |
| 20:39.36 | [TK]D-Fender | slingr: Is this business or personal? |
| 20:39.45 | slingr | right now its personal |
| 20:39.46 | [TK]D-Fender | slingr: Everything depends on your wiring, budget, what kind of use, etc. You looking at ditching VoIP for PSTN and jsut going analog? Just 1 channel? |
| 20:39.49 | slingr | i wasn't dissatisfiend |
| 20:39.52 | slingr | ahh |
| 20:39.59 | slingr | you are copying and pasting |
| 20:40.03 | slingr | thinks |
| 20:40.06 | [TK]D-Fender | slingr: almsost |
| 20:40.16 | slingr | ok |
| 20:40.21 | slingr | so basically what i'm looking for |
| 20:40.28 | *** join/#asterisk jtodd (i=jxeg2b7g@ns2.loligo.com) |
| 20:40.33 | [TK]D-Fender | slingr: What reason would you pay for hardware rather than use les.net? |
| 20:40.41 | slingr | bandwidth |
| 20:41.01 | slingr | my internet connection isn't the greatest |
| 20:41.07 | [TK]D-Fender | slingr: Has this proven an issue? |
| 20:41.22 | slingr | choppy audio quality |
| 20:41.28 | slingr | even with proper codecs |
| 20:41.49 | [TK]D-Fender | slingr: Very well. What do you currently posess by way of phones? |
| 20:42.23 | [TK]D-Fender | slingr: And by chance have you tested with at least 1 other provider or are you dead certain about the unreliability of your connection? |
| 20:42.31 | slingr | couple of ip phones |
| 20:42.45 | [TK]D-Fender | slingr: What models offhand? |
| 20:43.34 | slingr | lemme check |
| 20:46.04 | slingr | i have 2 Linksys SPA921's and 2 Linksys SPA1001 adapters |
| 20:46.37 | [TK]D-Fender | slingr: Ok, for home I'd recommend the SPA-3102 |
| 20:46.38 | drmessano | wow |
| 20:46.44 | drmessano | Someoen actually bought a SPA-1001 |
| 20:47.01 | slingr | lol |
| 20:47.05 | slingr | it was awhile ago |
| 20:47.34 | drmessano | 1 port FXS that costs more than the 2 port FXS, but who's counting.. |
| 20:47.43 | slingr | again.. awhile ago |
| 20:47.45 | slingr | :P |
| 20:47.58 | drmessano | Has always been that way :P |
| 20:48.16 | slingr | [TK]D-Fender> i've tested with voicenetwork.ca too |
| 20:48.18 | slingr | also choppy |
| 20:48.27 | maxxim | [TK]D-Fender> give me a hint/example how to build a number for the destination, by collecting the digits that the clint will press, and the last digit to be # |
| 20:48.46 | [TK]D-Fender | maxxim: "core show application read" |
| 20:49.14 | [TK]D-Fender | maxxim: "core show applicaitons" <- I HIGHLY recommend you look at the full list of basic commands available to you and read each in detail |
| 20:49.40 | maxxim | [TK]D-Fender> k, thanks!!! |
| 20:55.57 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 20:55.57 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:58.24 | slingr | i've tried les.net and voicenetwork.ca and both had audio issues |
| 21:00.21 | [TK]D-Fender | slingr: We'll you've now got my recommendation for home use. For business use I would recommend either a Sangoma A200d or Digium TDM410 series card with HWEC |
| 21:05.08 | slingr | googles HWEC |
| 21:06.24 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:06.42 | drmessano | slingr: What are the specs of your asterisk box? |
| 21:06.42 | v4mp | ok i've worked through the guide sip.conf for people to login and the connection to the voip service provider and added the relevent extensions for the provide and also a test extension 1 of the 2 internals work the 1 that works is meant to echo whats said but it does't |
| 21:07.07 | slingr | p4 2.4ghz 1gb ram 40gb hd |
| 21:07.08 | v4mp | nor is the incoming calls from provider being received on * but it connected to privder |
| 21:07.29 | drmessano | slingr: Running in a VM or with other apps? |
| 21:07.44 | *** join/#asterisk c4t3l (n=root@74.95.210.124) |
| 21:08.04 | [TK]D-Fender | slingr: Hardware Echo Cancellation |
| 21:08.08 | *** part/#asterisk c4t3l (n=root@74.95.210.124) |
| 21:09.04 | slingr | drmessano... solely dedicated to pbx |
| 21:09.05 | slingr | no vm |
| 21:10.14 | slingr | i've run a couple different configs... tried trixbox.... my last config was just debian 4.0 minimal then i installed asterisk and freepbx |
| 21:10.58 | v4mp | this is my sip conf and extensions conf of which i cant see whats wrong because the echo doesn't work nor does incoming calls http://pastebin.com/m362486ee |
| 21:13.29 | slingr | so SPA-3102 for home use eh |
| 21:14.17 | *** join/#asterisk Chris-NB (n=chris@213162066129.public.t-mobile.at) |
| 21:21.13 | *** join/#asterisk ManxPower (n=manxpowe@68-186-179-205.dhcp.leds.al.charter.com) |
| 21:27.34 | *** join/#asterisk John_Clay (n=J@unaffiliated/johnclay) |
| 21:27.44 | John_Clay | Anyone around for a few questions? |
| 21:27.58 | ManxPower | ~ask |
| 21:27.59 | jbot | i guess ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:28.17 | John_Clay | Fair enough. |
| 21:28.28 | John_Clay | I'm looking to put Asterisk on a Mac mini running 10.4 Server. |
| 21:28.52 | John_Clay | I found a binary install for it, but was wondering if it would work with an Apple USB Modem, or if I should get another phone adapter. |
| 21:29.21 | ManxPower | oh, don't expect and card or adapter to work, only SIP stuff. |
| 21:29.28 | John_Clay | The end goal for this is to serve as an answering machine, perhaps with future expansion to a full VOIP setup, instead of POTS. |
| 21:29.28 | ManxPower | well voip stuff, at least. |
| 21:30.18 | ManxPower | Your only option on a MAC is full voip setup. |
| 21:30.20 | tvirus | If someone is in a queue, is it possible for them to dial an extension while waiting? |
| 21:30.40 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 21:30.40 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:30.48 | John_Clay | ManxPower: would it be any different on a Linux-based system? |
| 21:30.54 | ManxPower | tvirus: only as documented. |
| 21:31.06 | ManxPower | On linux you have a wide variety of cards available. |
| 21:31.23 | ManxPower | Linux/x86 is the officially supported platform |
| 21:31.53 | tzafrir_laptop | huh? |
| 21:31.54 | tvirus | ManxPower: Where can I find that documentation? I can't find anything on how to interact while in a queue. |
| 21:32.13 | ManxPower | I believe someone did quite a bit of work on FreBSD for card drivers. |
| 21:33.02 | John_Clay | does the device have to be specifically compatible with Asterisk, or will any voice/fax device recognized by the OS work? |
| 21:34.20 | ManxPower | John_Clay: no voice modems, no fax modems, you need one of the supported interface cards. |
| 21:34.27 | ManxPower | (or you can use SIP) |
| 21:34.34 | John_Clay | Ah. Thanks. |
| 21:35.08 | tzafrir_laptop | ~fxsfxo |
| 21:35.09 | jbot | [~fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 21:35.42 | ManxPower | tvirus: maybe I was thinking of a different application |
| 21:35.56 | tzafrir_laptop | Technically a modem has all the hardware needed to function as a FXO adapter |
| 21:36.19 | John_Clay | I know there are apps for OS X that will use the Apple adapter, but they're pretty crap. |
| 21:36.25 | tzafrir_laptop | There is the "tiny" question of writing the software to drive that card |
| 21:36.28 | John_Clay | and the modem hardware supports voice, CID and a few other things. |
| 21:36.30 | ManxPower | So you are welcome to write a kernel driver for any device you want. |
| 21:36.56 | ManxPower | John_Clay: voice modems have large buffers -- work great for IVR, not so good with two way vouce. |
| 21:36.59 | ManxPower | and voice too! |
| 21:37.15 | John_Clay | hm, right. |
| 21:37.22 | John_Clay | That'd be suitable for an answering machine though. |
| 21:37.28 | ManxPower | tzafrir_laptop: As I understand it, only winmodems expose the low level interface to the hardware |
| 21:37.38 | ManxPower | But Asterisk is not an answering machine. |
| 21:37.50 | John_Clay | well, it can function as one based on what I've read |
| 21:37.52 | John_Clay | and heard |
| 21:38.00 | slingr | <[TK]D-Fender> thanks for the help |
| 21:38.02 | John_Clay | Rather overkill though. |
| 21:38.16 | ManxPower | Sure it can. |
| 21:38.46 | ManxPower | I station wagon can function as a pickup truck, but it's not recommended. |
| 21:39.01 | ManxPower | s/I/A/ |
| 21:39.18 | John_Clay | Well, yeah. |
| 21:39.45 | tvirus | ManxPower: I think I need to do Queue(900|tT|||15) and then exten = t,1,<stuff goes here>, correct? |
| 21:39.48 | tzafrir_laptop | ManxPower, most modems nowadays are software modems |
| 21:40.34 | ManxPower | tzafrir_laptop: But not random modems sitting in random computers, which is what people want to use. |
| 21:41.12 | ManxPower | tvirus: try it ans see. |
| 21:41.21 | tzafrir_laptop | And Asterisk is about host-side processing. So mocking at a modem for being a "winmodem" here is a bit strange |
| 21:41.22 | John_Clay | According to wiki, the Apple one works as an FXO |
| 21:41.41 | John_Clay | Perhaps I'll try it and see what happens :P |
| 21:43.21 | tvirus | I did, didn't work. |
| 21:43.22 | tvirus | Damnit. |
| 21:44.10 | *** join/#asterisk genji1981 (n=g@ip-118-90-122-1.xdsl.xnet.co.nz) |
| 21:44.54 | genji1981 | <PROTECTED> |
| 21:45.21 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 21:46.57 | genji1981 | okay, its also independant of extensions. Any extension can divert to voicemail, any other extensions call. found out just now.. it doesn't even need to be held. |
| 21:48.15 | *** join/#asterisk smth (n=chatzill@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 21:49.28 | *** part/#asterisk smth (n=chatzill@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 21:50.38 | [TK]D-Fender | genji1981: How & why does this call get "held"? What is dialing "*55" if not a NEW channel? |
| 21:50.54 | [TK]D-Fender | genji1981: Your description lacks important details |
| 21:52.12 | genji1981 | [TK]D-Fender: ignore the 'held' part. it just means that *55 just needs to be dialed out, on a new channel, without bridging it to any current call. |
| 21:52.28 | *** part/#asterisk John_Clay (n=J@unaffiliated/johnclay) |
| 21:52.54 | [TK]D-Fender | genji1981: And what is supposed to trigger this action in the first place? |
| 21:53.58 | genji1981 | [TK]D-Fender: a WaitforExten timeout, i believe. |
| 21:54.54 | genji1981 | "If you know the extension you require, please enter that now, otherwise, please hold." |
| 21:55.44 | genji1981 | the wait timesout.. and on a new channel, *55 is dialed.. upon answering, SendDTMF # is done (Dial D(#) doesn't seem to work.) |
| 21:55.50 | [TK]D-Fender | genji1981: Ok, so they sit in the IVR, don't choose anything. You want * to call out ALONE to this other number. What happens next for the caller? Then this outbound call.. what happens to them? |
| 21:57.21 | genji1981 | The caller gets stolen by the VOIP provider, on my end, it just hangs up. And after the new channel does the *55 and then #, it should hang up after a few seconds. |
| 21:58.16 | [TK]D-Fender | genji1981: IF YOU TRIGGER THE OUTCALL IN "T", THE CALLER WOULD STILL BE IN YOUR SYSTEM. aRE YOU WANTING * TO HANG UP ON THEM FROM THERE? |
| 22:00.40 | genji1981 | After * dials *55 # on a new channel, the caller will be taken by my voip provider, and asterisk will just see a hangup. |
| 22:02.05 | genji1981 | question is.. can * call numbers, alone? |
| 22:02.47 | genji1981 | Or do i need a scriptable SIP client/softphone/ |
| 22:02.48 | genji1981 | ? |
| 22:03.05 | [TK]D-Fender | genji1981: So your provider will KNOW to take the other channel back forcefully? |
| 22:03.16 | genji1981 | nods. "Yes. its a feature." |
| 22:03.27 | genji1981 | They're running asterisk too, i think. |
| 22:03.33 | [TK]D-Fender | genji1981: Extremely unusual to see one channel indicate the hijacking of anything |
| 22:03.55 | [TK]D-Fender | genji1981: So for that just make a call-file ro AMI originate to initiate this new call. Look those both upon the WIKI |
| 22:04.37 | genji1981 | and as for the sending of DTMF.... why doesn't Dial D(#) work? |
| 22:05.03 | v4mp | this is my sip conf and extensions conf of which i cant see whats wrong because the echo doesn't work nor does incoming calls http://pastebin.com/m362486ee anything i've done wrong ? |
| 22:07.05 | [TK]D-Fender | genji1981: Vou've shown me nothing and Im not psychic. take your pick. |
| 22:07.13 | [TK]D-Fender | ~sipnat |
| 22:07.14 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:07.17 | *** join/#asterisk John_Clay (n=J@unaffiliated/johnclay) |
| 22:07.17 | [TK]D-Fender | v4mp: ^^^^ |
| 22:07.35 | genji1981 | <PROTECTED> |
| 22:07.35 | genji1981 | <PROTECTED> |
| 22:07.35 | genji1981 | <PROTECTED> |
| 22:07.41 | John_Clay | ManxPower: What do you make of this? http://www.sunrisetel.net/software/asterisk/HOWTOs/SPA3K-FXO-and-Asterisk.shtml |
| 22:07.55 | genji1981 | so.... alls good.. but called party doesn't hear the #. |
| 22:08.30 | v4mp | [TK]D-Fender, the server isn't behind a NAT |
| 22:11.18 | [TK]D-Fender | genji1981: Its clearly sending the DTMF |
| 22:13.26 | *** join/#asterisk linuxtuxie (n=kku@91.178.106.89) |
| 22:14.54 | linuxtuxie | Hello all, Does someone know if I can couple the following box to an asterisk system ? http://www.udtech-usa.com/pdfs/bosch/VIPX1Single-cha_DataSheet_enUS_E2142694155.pdf |
| 22:15.31 | *** join/#asterisk niros (n=nir@89-138-11-112.bb.netvision.net.il) |
| 22:15.43 | genji1981 | [TK]D-Fender: but called party isn't hearing it. Why? does it need to be set to reinvite=no? |
| 22:16.40 | linuxtuxie | I want to use it for a video door phone |
| 22:17.23 | [TK]D-Fender | genji1981: genji1981> -- Sending DTMF '#' to the called party. |
| 22:17.44 | [TK]D-Fender | genji1981: Yes, it IS being sent. Either the mode is wrong, or it is sent too fast. use M() in the latter case |
| 22:22.28 | *** join/#asterisk tkbeat (n=tk@g229164173.adsl.alicedsl.de) |
| 22:22.58 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 22:27.34 | *** part/#asterisk John_Clay (n=J@unaffiliated/johnclay) |
| 22:30.51 | Sedorox | I almost feel embarrassed to ask this, but does anyone know where there is a decent changelog/feature change list of changes since 1.2? |
| 22:37.33 | *** join/#asterisk _Krieger_ (n=warsword@ip233.15.airbites.od.ua) |
| 22:37.40 | _Krieger_ | is it possible to use zaptel somehow having disabled kernel modules load support? |
| 22:37.59 | *** join/#asterisk bradleyprice86 (n=Compaq_O@adsl-63-194-8.bhm.bellsouth.net) |
| 22:42.15 | v4mp | [TK]D-Fender, what have i done wrong because the server is on a dedi |
| 22:42.19 | *** join/#asterisk tkbeat (n=tk@p54B94DE8.dip.t-dialin.net) |
| 22:46.50 | tzafrir_laptop | Sedorox, UPGRADE.txt or something |
| 22:53.31 | ManxPower | _Krieger_: I don't think you can do that. |
| 23:22.31 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:22.31 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 23:24.47 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177581609.dsl.bell.ca) |
| 23:27.09 | *** part/#asterisk linuxtuxie (n=kku@91.178.106.89) |
| 23:44.11 | v4mp | right guys i think it might be receiving the concomig call but complaining about a noisey line then dropping it |
| 23:44.12 | v4mp | http://pastebin.com/m610fe4f0 |
| 23:44.32 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:47.44 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-211-254-206.dsl.pltn13.sbcglobal.net) |
| 23:49.28 | [TK]D-Fender | v4mp: those are only "register'"'s Whats the actual problem? |
| 23:49.39 | *** part/#asterisk bradleyprice86 (n=Compaq_O@adsl-63-194-8.bhm.bellsouth.net) |
| 23:50.03 | v4mp | ahh well the server isn't receiving the call |
| 23:50.21 | v4mp | [TK]D-Fender, the noisey line being part of the register too ? |
| 23:51.09 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 23:54.24 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 23:54.41 | [TK]D-Fender | v4mp: I see no call attempt at all |
| 23:57.34 | v4mp | [TK]D-Fender, this is what i dont get because the connection to the provider is active |
| 23:59.40 | [TK]D-Fender | v4mp: check your firewall |