| 00:00.14 | Yosam | :( no |
| 00:01.00 | hsv-al | well we know the night is over when you know who idles for 1 or more hours |
| 00:01.09 | hsv-al | | [TK]D-Fender ³ idle / 2h 36m 17s |
| 00:01.17 | hsv-al | :) |
| 00:03.17 | paul0 | tzafrir, V92 modems |
| 00:05.01 | Yosam | [May 27 17:04:50] ERROR[15529]: res_speech_lumenvox.c:421 lumenvox_new: No SRE server is available for processing speech |
| 00:06.20 | Yosam | but i have it installed |
| 00:10.17 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 00:11.42 | drmessano | paul0: the best answer is "no" |
| 00:12.04 | paul0 | drehlecom, how sad :/ |
| 00:12.25 | drmessano | There are a few VERY VERY specific obscure chipsets that are basically the same as cheap ass X100P cards |
| 00:12.46 | drmessano | Even if you did find one, you would still have basically a cheap ass X100P card |
| 00:13.04 | coppice | I find it interesting that so many people ask for drivers for these things, yet nobody bothers to cook them up :-) |
| 00:13.23 | drmessano | It's a waste of time, IMO |
| 00:13.26 | drmessano | Get a real card |
| 00:13.33 | drmessano | Chinese clones are down to $50 now |
| 00:14.02 | coppice | they work fine with the right software. even USB ones, which let you do demos with notebooks |
| 00:14.07 | drmessano | If you want to be cheap, at least you can get a cheap imitation of something decent |
| 00:14.10 | CCFL_Man2 | drmessano: a db and a half less attentuationon my adsl line!!@!@# |
| 00:14.32 | drmessano | a db and a half is so meaningless |
| 00:14.45 | drmessano | That's an ant fart |
| 00:15.54 | coppice | you can't measure things like that. an ant fart consumes a specific amount of energy. 1.5dB is a relative amount of energy |
| 00:16.22 | drmessano | Well |
| 00:16.46 | drmessano | A proper db measurement would specify teh source |
| 00:16.53 | drmessano | Such as dbi |
| 00:16.59 | florz | drmessano: nope |
| 00:17.08 | drmessano | Yes |
| 00:17.14 | coppice | nope |
| 00:17.18 | florz | drmessano: only if you want to have an absolute value |
| 00:17.31 | drmessano | A db measurement is useless without a reference |
| 00:17.33 | florz | drmessano: db alone is perfectly valid for factors |
| 00:17.38 | coppice | he said 1.5dB less. that is a proper statement |
| 00:17.56 | *** join/#asterisk smps (n=maher@193.170.53.51) |
| 00:18.08 | drmessano | 1.5db less than before gives no validity to the original measurement |
| 00:18.19 | drmessano | The original measurement is still in question |
| 00:18.35 | CCFL_Man2 | i wonder if ant farts stink? |
| 00:18.45 | drmessano | Typical |
| 00:19.07 | drmessano | He's an expert on everything anyway |
| 00:19.10 | drmessano | Ahem |
| 00:19.14 | florz | drmessano: well, actually you kindof need two measurements to find that 1.5 dB |
| 00:19.21 | drmessano | Indeed |
| 00:19.29 | CCFL_Man2 | drmessano: ios said i had 1.5db less attenuation |
| 00:19.36 | *** join/#asterisk x_or (n=cdawson@68.178.75.89) |
| 00:20.01 | drmessano | florz: Have you ever ventured into the world of antenna and audio measurements? |
| 00:20.21 | drmessano | It's quite fascinating how "relative" a "relative" measurement is |
| 00:20.38 | drmessano | I could claim a damn piece of string has 20db gain |
| 00:21.05 | CCFL_Man2 | 20dBi gain? |
| 00:21.09 | drmessano | No |
| 00:21.15 | drmessano | Youre being too correct :) |
| 00:21.20 | drmessano | 20db.... |
| 00:21.22 | CCFL_Man2 | heh |
| 00:21.31 | drmessano | Now, 20dbi.. you got something |
| 00:21.33 | Yosam | :(( |
| 00:21.37 | florz | drmessano: probably not - I guess it's a bit difficult for a string to produce energy ;-) |
| 00:22.01 | drmessano | Actually, a wet piece of string can be a great radiator |
| 00:22.05 | drmessano | Well |
| 00:22.11 | drmessano | GreatER than not wet |
| 00:22.33 | drmessano | Audio products are the same |
| 00:22.34 | florz | so, you mean, like, 20 dBd(ry)? =:-) |
| 00:22.37 | drmessano | 120db speakers |
| 00:22.57 | drmessano | heh |
| 00:23.11 | CCFL_Man2 | in reference to an rms watt? |
| 00:23.14 | hsv-al | LOL |
| 00:23.25 | hsv-al | http://youtube.com/watch?v=QNNl_uWmQXE |
| 00:23.28 | drmessano | I had a guy show up at my house once with a 30db gain CB antenna |
| 00:24.07 | drmessano | It was a basically a copper pipe, interrupted with a thick copper coil in the middle, a big ass capacitor tapped to the coil, and a teflon insulator |
| 00:24.12 | drmessano | He paid $200 for it |
| 00:24.33 | drmessano | Went back and checked and it was something on the order of 1.2 dbi |
| 00:24.47 | drmessano | Never did find out what it was 20db better than |
| 00:24.52 | drmessano | I guess a dummy load |
| 00:24.57 | florz | *g* |
| 00:25.10 | florz | 30 dBwp? |
| 00:25.37 | drmessano | But yeah.. I always ask when I get a db measurement, especially when compared to a previous measurement, what the reference is |
| 00:25.47 | drmessano | db is pretty useless otherwise |
| 00:25.58 | Yosam | Lumenvox users? |
| 00:26.10 | drmessano | Yosam: Is this jeopardy? |
| 00:26.13 | drmessano | Hang on |
| 00:26.28 | drmessano | A: Who is having a hard time getting help? |
| 00:26.33 | drmessano | Lumenvox users! |
| 00:26.36 | drmessano | Err |
| 00:26.37 | drmessano | Sorry |
| 00:26.43 | CCFL_Man2 | o rly |
| 00:27.01 | drmessano | Actually, I can't help you.. I know nothing about lumenvox |
| 00:27.25 | drmessano | But if your asterisk install is still borked from it, I suggest you back up and punt, and get your box back online first |
| 00:28.07 | florz | drmessano: Well, for example a cable having an attenuation of n dB/m doesn't really need any reference to be a useful value ... |
| 00:28.35 | drmessano | Ha.. it already does |
| 00:28.42 | CCFL_Man2 | i should set up one of my lines to plar to drmessano's phone |
| 00:28.46 | Yosam | i have no problem my box runs ok! |
| 00:28.56 | Yosam | but i hate this i purchased a fucking product and cant get support |
| 00:29.03 | drmessano | A cable with x db loss per foot or meter has a loss over that signal traveling through freespace |
| 00:29.05 | Yosam | i mailed them |
| 00:29.10 | Yosam | they replied with nonsense |
| 00:29.22 | drmessano | The reference is freespace |
| 00:30.05 | drmessano | In which case you're talking about the signal injected into the cable, with a loss at the square of the distance |
| 00:30.10 | drmessano | Which is easily calculated |
| 00:30.14 | CCFL_Man2 | i thought the reference was the input signal power |
| 00:30.16 | florz | drmessano: uh? not more, like, no distance? |
| 00:30.34 | *** part/#asterisk x_or (n=cdawson@68.178.75.89) |
| 00:31.15 | drmessano | The loss is based on the signal radiated from an isotropic radiator into freespace, for MOST cable |
| 00:31.21 | florz | drmessano: so, yes, when you want to make use of the cable, you have to multiply that factor with your input power to find out whether enough signal is left at the end |
| 00:32.07 | drmessano | Sure.. but again, the db measurement isn't based on a non-existant reference.. in this case, it's freespace |
| 00:32.34 | CCFL_Man2 | freespace is nonexistant |
| 00:33.00 | drmessano | Sure it is |
| 00:33.31 | drmessano | It's made up of gases.. the ones that keep outer space from falling on you :) |
| 00:33.32 | lmadsen | free radicals! |
| 00:36.20 | CCFL_Man2 | free bitches! |
| 00:36.31 | hsv-al | FRS.com drinks |
| 00:36.35 | hsv-al | Free Radical Scavenger :) |
| 00:36.39 | drmessano | Free Kevin! |
| 00:36.41 | drmessano | No |
| 00:36.46 | drmessano | Put Kevin back in jail |
| 00:38.36 | *** join/#asterisk drdrain (n=drdrain@cpe-066-057-105-080.nc.res.rr.com) |
| 00:38.37 | lmadsen | puts drmessano in jail |
| 00:38.45 | drmessano | lol |
| 00:39.03 | drmessano | isn't a big fan of Kevin Mitnick |
| 00:39.23 | lmadsen | isn't a big fan of drmessano |
| 00:39.32 | drmessano | :( |
| 00:39.46 | lmadsen | don't worry.. it's not just you ... l dislike everyone equally |
| 00:39.50 | drmessano | If I had feelings... that, that would hurt |
| 00:40.07 | florz | drmessano: erm, I don't quite get what you want to say - a cable has a gain of -n dB/m (n being a positive number) from "focussing" the signal inside the cable versus isotropic radiation? |
| 00:40.51 | florz | (as in power per area) |
| 00:41.20 | drmessano | First off, there's no gain in cable |
| 00:41.24 | drmessano | It's all loss |
| 00:41.27 | drdrain | Oh lord ... compared to discussions of isotropic radiation ... |
| 00:41.29 | florz | well, sure there is |
| 00:41.44 | drdrain | My question is gonna really seem foolish |
| 00:41.45 | florz | yeah, and positive is just negative gain |
| 00:41.46 | drmessano | Uh... no |
| 00:41.51 | florz | erm |
| 00:41.54 | florz | yeah, and positive loss is just negative gain |
| 00:43.15 | drmessano | Cable is a loss compared to passing that signal through freespace, which is lossy as well.. The loss calculated on a given piece of cable is the difference in loss between the signal in the cable and the signal radiating in freespace from an isotropic radiator |
| 00:44.10 | drmessano | If I measure the signal at one end of a 300 foot piece of cable, vs taking a freespace relative signal measurement, you should see a difference equal to the loss of the cable |
| 00:44.42 | florz | well, measuring exactly _what_ in the radiation case? |
| 00:45.53 | florz | total radiation power passing through the spherical surface at 300 feet distance from the radiator? |
| 00:46.37 | drmessano | In the case of RF, the relative signal strength of the signal in microvolts or millivolts |
| 00:46.37 | florz | radiation power passing through the same surface as the cable's cross-sectional area at 300 feet distance from the radiator? |
| 00:48.07 | drmessano | you're talking about distance from the transmitter |
| 00:48.41 | drmessano | The isotropic radiator simply changes the transmission medium from copper into freespace |
| 00:49.19 | drmessano | The cable, of course, is passing the same signal through a copper path, but one that is no less a resistor as freespace is to the radiated signal |
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| 00:51.48 | drmessano | If you look at waveguide, waveguide is nothing more than a 0 gain radiator into a focused freespace area |
| 00:52.00 | hsv-al | and the partial subway glangule quadroplizes the synthetic diaboloid phree blasphemy radiaii |
| 00:52.12 | florz | hsv-al: nope, it does not! |
| 00:52.23 | drmessano | hsv-al: You missed a decimal place |
| 00:52.48 | drdrain | Tech Adapt High Cant |
| 00:52.53 | florz | drmessano: well, I still don't see what powers exactly you are comparing |
| 00:53.13 | drdrain | The Omnissiah will be pleased |
| 00:53.20 | drmessano | Radiated energy |
| 00:53.26 | drmessano | in volts |
| 00:53.46 | florz | drmessano: well, volts isn't quite a unit of energy!? |
| 00:54.13 | drmessano | No it's not.. That's a bit oversimplified |
| 00:56.06 | drmessano | 30 watts from a transmitter passed through 1000 feet of cable vs measured 1000 feet from a zero gain antenna vs 1000 feet down a waveguide will all yield comparisons that are directly attributable to the measured properties of the respective medium |
| 00:56.58 | drmessano | When you're talking about RF a difference in 4x the power is 6db |
| 00:57.12 | drmessano | or 6dBi |
| 00:57.31 | florz | drmessano: I mean, basically: either you are comparing the output from the cable to the radiation power passing through some spherical surface around the radiator - in which case, assuming that free space doesn't attenuate, obviously free space will have more energy left - but then the fact that the radiator is isotropic doesn't matter |
| 00:57.52 | florz | (or rather s/energy/power/) |
| 00:58.15 | drmessano | The fact that the radiatior is isotropic matters completely |
| 00:59.23 | florz | alternatively, it does matter, then you are measuring the power through a certain area at a certain distance - and then I can hardly imagine how a usual cable should have lower power per area than an isotropic radiator at some non-small distance |
| 01:00.09 | drmessano | The power is measured at a single point from the isotropic radiator, for 1.. and 2, the radiating of the power into freespace with 0 gain is the only way you can compare the radiation properties of the freespace and the cable effectively.. You're not gonna put an amplifer in the cable, so why put gain in the radiator? |
| 01:01.02 | florz | how ya mean? |
| 01:01.04 | florz | I mean |
| 01:01.21 | florz | a cable obviously has a pretty high dBi value |
| 01:01.23 | drmessano | Air is far less resistive to RF than copper is.. |
| 01:02.45 | drmessano | If I pass 40,000 watts into a piece of 6 inch flexible copper line, I might have half the power left at 500ft |
| 01:03.08 | drmessano | But if I radiate that 40,000 watts directly to an isotropic radiator, I can get 20 miles from that signal |
| 01:03.13 | florz | yeah, which would be a 3 dB attenuation, exactly |
| 01:03.47 | florz | but still that would be many, many dBis of gain |
| 01:04.04 | florz | if you really want to apply that concept there ;-) |
| 01:04.07 | drmessano | There is NO gain in cable.. it is all a loss |
| 01:04.11 | [TK]D-Fender | edibrac, nothing wrong with your DB idea. |
| 01:04.24 | [TK]D-Fender | hsv-al, And I've got martial arts twice a week.... |
| 01:04.55 | florz | drmessano: well, sure is there a gain |
| 01:05.14 | florz | drmessano: or to put it in another way: can there be gain in an antenna? |
| 01:05.21 | drmessano | Yes |
| 01:05.25 | drmessano | But not in a cable |
| 01:05.30 | drmessano | Cable does not have any gain |
| 01:05.34 | florz | drmessano: now, how can there be gain in an antenna? |
| 01:06.00 | drmessano | It has to do with element spacing, element stacking |
| 01:06.03 | florz | I mean, it can't produce energy on its own, right? ;-) |
| 01:06.37 | florz | well, yeah, but how can it be that we call what's happening there "gain" even though it doesn't produce any energy? |
| 01:06.54 | *** join/#asterisk rpm (n=rpm@S010600111155e117.cg.shawcable.net) |
| 01:06.56 | lanning | focusing |
| 01:07.15 | florz | exactly ;-) |
| 01:07.30 | florz | now, what does a cable do? |
| 01:07.31 | rpm | will a tdm400p with an fxs module, if i don't plug the 12v into the card work as a fxo module? |
| 01:07.36 | drmessano | Well, then your logic is flawed |
| 01:07.39 | lanning | a very high gain antenna is focused on a specific point (instead of broadcasting everywhere) |
| 01:07.46 | jjshoe | rpm what? |
| 01:07.55 | drmessano | Because I am focusing all my energy in one direction in a cable end up with LESS power |
| 01:07.59 | drmessano | That is a LOSS |
| 01:07.59 | jjshoe | rpm fxs ports do not work without 12v. not plugging in 12v. means nothing will happen. |
| 01:08.17 | florz | drmessano: nope, I guess you are simply mixing up things |
| 01:08.18 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 01:08.33 | lanning | right, cables and junctions are measured in quantity of loss. |
| 01:08.34 | drmessano | I spent 20 years working in RF.. im pretty sure I am not confused :) |
| 01:08.35 | florz | drmessano: the one thing is that due to ohmic resistance you are losing some energy to heat |
| 01:08.44 | drmessano | Yes |
| 01:08.44 | *** join/#asterisk Sky-Knight (n=Sky-Knig@ip68-2-137-221.ph.ph.cox.net) |
| 01:08.46 | drmessano | Lots of it |
| 01:08.54 | *** part/#asterisk Sky-Knight (n=Sky-Knig@ip68-2-137-221.ph.ph.cox.net) |
| 01:08.56 | rpm | jjshoe; i don't want it to work as fxs. i want to recieve fxs signalling from my telco.. i was just wondering if it was just a 5v/12v difference between the modules.. |
| 01:09.05 | hsv-al | I bet dremessano and florz failed statics |
| 01:09.06 | rpm | it's just some old hardware i have here |
| 01:09.06 | hsv-al | and dynamics |
| 01:09.07 | jjshoe | rpm no. |
| 01:09.07 | hsv-al | nuff said |
| 01:09.24 | drmessano | Why do you say that? |
| 01:09.29 | hsv-al | ;-] |
| 01:10.24 | drmessano | You must work for a cable manufacturer if you're gonna tell me you got cable with gain :) |
| 01:10.27 | florz | drmessano: the other thing is that you focus the propagation of the power into a specific direction - which is expressed as dBi, gain versus propagation in all directions. Which, when applied analogously to a cable, will usually result in a gain, no? |
| 01:11.01 | drmessano | But I am measuring dbi at ONE point |
| 01:11.06 | drmessano | and cable goes to ONE point |
| 01:11.07 | drmessano | So no |
| 01:11.25 | jbeez | shorter the cable, the better |
| 01:11.38 | drmessano | jbeez: Thats NOT what she said |
| 01:11.40 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 01:11.45 | lanning | cable is always measured in loss. |
| 01:11.58 | lanning | you ALWAYS loss signal in a cable |
| 01:12.00 | florz | drmessano: So, no, you obviously don't get any more power out of a cable than you put into it - but you concentrate it into much smaller area than in case of an isotropic radiator |
| 01:12.24 | drmessano | florz: But you're not measuring dBi in a sphere |
| 01:12.27 | drmessano | You |
| 01:12.32 | drmessano | You measure it at ONE point |
| 01:12.43 | jbeez | thats what I said |
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| 01:12.48 | jbeez | im not quoting anyone |
| 01:12.52 | florz | drmessano: well, yeah, exactly |
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| 01:13.35 | lanning | cables are point to point (not radiators) so you always want the least amount of loss (which usually costs more for better quality conductors/shielding...) |
| 01:13.44 | florz | drmessano: and in that single point at the end of a cable, the power density is much higher than at the same point relative to an isotropic radiator |
| 01:13.50 | lanning | so it is a balance of cost and quality. |
| 01:14.07 | drdrain | I have a call inbound over a ZAP channel. |
| 01:14.29 | drdrain | I route it back out to a cell phone over an IAX2 truck |
| 01:14.50 | jjshoe | drdrain does that truck go vroom? :D |
| 01:14.54 | lanning | starts the truck and drives off. |
| 01:15.03 | drdrain | sorry trunk |
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| 01:15.24 | jbeez | who lotta dr's in here |
| 01:16.08 | hsv-al | id rather pop out pde's all day then listen to more of this back/forth |
| 01:16.09 | hsv-al | heh |
| 01:16.15 | drdrain | The cell phone has the call volume it receives substantially attenuated |
| 01:16.46 | lanning | check the gains in the zap driver |
| 01:17.13 | drdrain | That's the thing\ |
| 01:17.45 | drdrain | The Zap channels are tuned fine when they terminate on a SIP extension inside the PBX |
| 01:18.00 | florz | drmessano: so the one thing is comparing power density per area, or rather per angle in the usual case, the other one comparison of total power left after using a certain kind of transport |
| 01:18.00 | drdrain | TX and RX sound great |
| 01:18.17 | drmessano | florz: I see what you are saying.. Yes, the power density is focused down the cable.. But of course, the measurement is not based on the power fed into the isotropic radiator.. obviously there would be a huge disparity with the 360 degree radiation pattern.. I'm not quite sure an isotropic radiator is ever really used as a reference for anything other than a textbook 0 db gain point |
| 01:18.20 | lanning | the iax trunk goes where? |
| 01:18.45 | drdrain | IAX goes out to the PSTN |
| 01:18.56 | drdrain | VOIPStreet is the ITSP |
| 01:19.17 | lanning | any other calls get the attenuation? |
| 01:19.22 | drdrain | Nope |
| 01:19.51 | drdrain | Calls that come in on one channel of the IAX trunk and route back out are fine |
| 01:20.12 | drdrain | Back out over another channel of the IAX that is |
| 01:20.24 | lanning | calls coming in the trunk and routed to a SIP extension are fine? |
| 01:20.30 | drdrain | Yep |
| 01:20.32 | *** join/#asterisk ZX81 (n=matt@202.55.97.173) |
| 01:20.37 | ZX81 | hi all |
| 01:21.18 | ZX81 | I have an AEX800 card with 8FXO, dmesg says 4 of the ports are not installed (3,4,7,8) any ideas? |
| 01:21.22 | florz | drmessano: and for the comparison of total power left the directionality of any radiator doesn't matter, as the power passing through the full cross-sectional area of the medium's "end" is considered anyway |
| 01:21.26 | drdrain | Only other wierdness is a little echo when calls come in IAX and are routed out a ZAP channel |
| 01:22.34 | drmessano | florz: Indeed. All that matters is the 0db gain, which is difficult to achieve in theory |
| 01:23.03 | drdrain | The analog card is a Rhino R4FXO card with onboard EC |
| 01:23.06 | drmessano | florz: you can screw up and get 1 db gain from a light bulb |
| 01:23.46 | florz | drmessano: so, in case of cable losses, you specify simply the factor of power lost versus a hypothetical 0-loss connection - which in the easiest case would be moving the parts to be connected directly next to each other. Well, if you ignore connector loss and stuff at least ... |
| 01:23.55 | drdrain | Hey ZX81. What about lszaptel output? |
| 01:25.17 | drmessano | florz: and nice impedence bumps in things like PL-259s lol |
| 01:26.40 | florz | drmessano: well, yeah, with the usual inaccuracies of the real, non-digital, world ;-) |
| 01:27.17 | drmessano | florz: Nothing like a TDR to show how perfect the world of cabling and connectors REALLY are |
| 01:33.30 | drmessano | Hmm.. I thought vtiger CRM had all the asterisk crap SugarCRM does |
| 01:34.33 | ZX81 | all 8 in use |
| 01:35.04 | drmessano | all 8 work? |
| 01:35.07 | ZX81 | dmesg still says port 3,4,7,8 not in use - even though I just took the card out and moved the modules |
| 01:35.22 | drmessano | oh |
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| 01:41.06 | deltaray2 | Hi. If I have two POTS lines on my asterisk server, is there some trick to making it so that an outbound call uses whichever trunk is avaiable? The example dialplans use a specific trunk like 'exten => _9NXXXXXX,1,Dial(Zap/1/${EXTEN:1})' |
| 01:43.15 | ZX81 | says they are not in use because they are not in use - the card has 4 tdm400p fxo modules in it - man, flew 1500Km for this install |
| 01:43.30 | drdrain | That sux |
| 01:45.09 | [TK]D-Fender | deltaray2, set "group=1" for each of them in zapata.conf and use Dial(Zap/g1/${EXTEN:1}) |
| 01:45.23 | deltaray2 | Ah, that's what I was looking for. Thanks. |
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| 01:48.31 | deltaray2 | Cool that worked. |
| 01:48.40 | hsv-al | d-fender is still up |
| 01:48.41 | hsv-al | heh |
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| 01:57.26 | *** join/#asterisk Sky-Knight (n=Sky-Knig@ip68-2-137-221.ph.ph.cox.net) |
| 01:57.36 | *** part/#asterisk Sky-Knight (n=Sky-Knig@ip68-2-137-221.ph.ph.cox.net) |
| 01:58.34 | docelmo | ARGH! Qwest is a pain in the ass! |
| 01:58.47 | drmessano | Yes, they are |
| 01:59.02 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 01:59.30 | docelmo | All Im trying to do is convert my NFAS PRI Group to individual PRI's |
| 01:59.42 | docelmo | cause NFAS SUCKS! |
| 02:04.18 | jaytee | even with 10 PRI's using one D channel with the equipment costs it doesn't seem like it would save anything and only give you 9 addtional channels. |
| 02:07.06 | deltaray2 | Has anyone heard of any project to make a pool of asterisk servers in communities that use people's phone lines so that you can call to a local POTS line in that area without toll? I haven't heard of anything, but someone just suggested something like that on the local LUG list. |
| 02:08.44 | lanning | um, DUNDI |
| 02:09.10 | *** join/#asterisk BeeBuu (n=beebuu@59.38.99.48) |
| 02:09.43 | lanning | er.... DUNDi |
| 02:10.53 | *** join/#asterisk FarrisG (n=jrush@gateway.wiquest.com) |
| 02:11.02 | FarrisG | Are there any tips/howtos out there for easily changing config options on multiple grandstream phones? Thinking of using curl. I have about 150 phones that I need to change ONE option on. |
| 02:13.58 | mosty | if they don't support automatic provisioning then curl + your favorite scripting language would be your best bet |
| 02:16.17 | deltaray2 | lanning: But isn't DUNDi only for finding other asterisk servers to dial to? What I mean is people donating their POTS line so that you can use their asterisk server to dial out locally even though you are normally long distance. |
| 02:16.25 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 02:16.52 | lmadsen | deltaray2: there were several projects that did that, but they are all gone now -- the legal liabilities were too great |
| 02:17.47 | deltaray2 | lmadsen: Legal liabilities? You mean like being liable for what someone does during the call or harrassment or was it to do with violations of phone company TOSes? |
| 02:17.55 | lmadsen | yes |
| 02:18.01 | deltaray2 | :-) |
| 02:18.02 | lmadsen | harrassement etc... |
| 02:18.02 | deltaray2 | both? |
| 02:18.15 | lmadsen | someone using your line to place anonymous calls for harrassement |
| 02:18.17 | deltaray2 | That makes sense. |
| 02:18.18 | *** part/#asterisk FarrisG (n=jrush@gateway.wiquest.com) |
| 02:18.48 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-e1c145b1c8260d25) |
| 02:19.03 | lmadsen | you can search for a system called FWDout (formerly bellster) |
| 02:20.33 | deltaray2 | Thanks |
| 02:28.09 | *** join/#asterisk excAliBuR (n=sales@207.134.8.33) |
| 02:28.27 | excAliBuR | a simple question... can asterisk send voicemail to email? |
| 02:28.39 | mosty | yes it can |
| 02:28.46 | rob0 | by default it does just that |
| 02:28.53 | excAliBuR | nice |
| 02:28.55 | excAliBuR | :} |
| 02:29.02 | rob0 | (but you need a functioning MTA with sendmail binary) |
| 02:29.26 | excAliBuR | ohhh.... i didn't set up any smtp stuff |
| 02:29.47 | rob0 | a "null client" might be easier to set up |
| 02:29.59 | jaytee | I've got my * calls routing to Exchange Unified Messaging instead of * voicemail |
| 02:30.04 | rob0 | ssmtp, nullmailer or the like |
| 02:30.34 | jaytee | msmtp will do for debian based systems. |
| 02:30.48 | excAliBuR | ubuntu is what i have :) |
| 02:31.12 | jaytee | i just setup msmtp on Hardy this past weekend |
| 02:31.43 | excAliBuR | i'm apt getting it now |
| 02:31.44 | excAliBuR | :D |
| 02:32.58 | jaytee | I had a project to build a web kiosk that was locked down so the user could only go to the default site and use that or otherwise logout or shutdown and I used msmtp with mutt in bash scripts to email the hostname and outside IP address of the system to a central account so I could track their addresses for ssh remote support. |
| 02:35.48 | denon | jaytee: sounds like a job for dynamic dns |
| 02:36.21 | jaytee | yeah, but I just wanted something quick and dirty to get the job done. |
| 02:37.27 | jaytee | so I just used the automation portion of whatismyip.com to pull a .asp file with the address in it and slap that together with the local hostname and bang! out the door. |
| 02:38.53 | rob0 | Heirloom mailx (find it on freshmeat) would take the place of both mutt and msmtp in that. |
| 02:39.39 | rob0 | It's a mailx-compatible thing with SMTP client support built-in. |
| 02:39.42 | jaytee | rob0, for me? or for excAliBuR ? |
| 02:43.41 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 02:45.05 | *** join/#asterisk pigpen2 (n=pigpen@fw.seamans.cc) |
| 02:47.01 | *** join/#asterisk Yosam (i=GuyOCana@S0106001c109e98db.no.shawcable.net) |
| 02:47.02 | pigpen2 | Hi all, quick question. The setting in the sip.conf, in previous years, sip would only bind to one ip address, one interface. Can it bind to multiple addresses, on seperate interfaces now? |
| 02:47.15 | lmadsen | it can bind to all, or 1 |
| 02:47.39 | lmadsen | bindaddr=0.0.0.0 or bindaddr=192.168.1.2 (for example) |
| 02:48.10 | pigpen2 | yeah, that was there a long time ago, when it would only bind to the ip addresses on a single interface. |
| 02:48.16 | lmadsen | still the same |
| 02:48.32 | pigpen2 | So, multiple IP addresses (as it has been) = yes |
| 02:48.44 | pigpen2 | IP addresses on seperate interfaces = no. |
| 02:49.02 | pigpen2 | Unlike the IAX, which will bind to all ip addresses on all interfaces. |
| 02:49.42 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 02:52.28 | Yosam | anyone experienced with lumenvox!? |
| 02:53.17 | *** join/#asterisk alancio (n=Alancio@190.75.3.207) |
| 02:53.53 | deltaray2 | Is there any kind of plugin or script for Asterisk that will dial an external voicemail system like SBC's voicemail and check the messages there and return them back to a voicemail box locally? |
| 02:54.13 | lmadsen | that is the kind of thing you develop yourself |
| 02:54.23 | deltaray2 | ok |
| 02:54.31 | deltaray2 | But have people done this? |
| 02:54.34 | deltaray2 | Is it possible? |
| 02:54.36 | lmadsen | possibly |
| 02:55.28 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 02:58.31 | lanning | you need voice recognition to handle the prompts. |
| 02:59.25 | *** join/#asterisk Shazzamy (n=evies980@d206-75-75-10.abhsia.telus.net) |
| 02:59.41 | Shazzamy | hello |
| 02:59.55 | Yosam | I need help with lumenvox |
| 03:03.58 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.05 | deltaray2 | lanning: Well how about the signal that you get when you pickup a normal POTS phone and it lets you know that you have voicemail. is there anyway for Asterisk to just interpret this and notify someone through e-mail. That would be enough for me. |
| 03:08.52 | *** join/#asterisk WindBack (n=jorge@host14.190-31-68.telecom.net.ar) |
| 03:08.57 | jaytee | deltaray2, not that I'm aware of for *. I don't think the zaptel drivers are programmed to detect stutter dial tone. |
| 03:09.48 | *** join/#asterisk yojimbo-san (n=CheethJ@120.89.81.19) |
| 03:10.56 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 03:10.56 | deltaray2 | Ok, so that's what its called. Do you know if the stutter dial tone can cause any problems for Asterisk? |
| 03:11.03 | deltaray2 | like when you try to dial out? |
| 03:11.10 | jaytee | no, it doesn't |
| 03:11.56 | *** join/#asterisk schmooze (n=schmooze@rrcs-67-52-210-195.west.biz.rr.com) |
| 03:12.04 | jaytee | I used fxo cards to bridge analog extensions on a Nortel Option 11c to SIP voip phones on * and they used Nortel's CallPilot voicemail which gives stutter dial tone on analog lines. |
| 03:12.31 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 03:12.36 | jaytee | but the SIP phones won't hear stutter dial tone though. |
| 03:12.47 | jaytee | they'll just get regular dial tone |
| 03:17.10 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 03:24.18 | alancio | hi people, how can I know from a CDR if the call was answered, busy, etc? |
| 03:24.43 | jaytee | it's one of the default fields called Disposition |
| 03:25.12 | alancio | but it always shows ANSWERED, regardless of what happened with the call |
| 03:25.38 | jaytee | really? doesn't on my system |
| 03:26.08 | jaytee | it either shows ANSWERED, BUSY or NO ANSWER |
| 03:26.23 | alancio | oh I should say that I'm using mysql cdr logging |
| 03:26.32 | alancio | could that be the reason? |
| 03:26.41 | mosty | alancio, that should not matter |
| 03:27.06 | mosty | what kind of calls are you making to test this? |
| 03:27.23 | alancio | I called my cell phone, and didn't answer it |
| 03:27.37 | alancio | it shows ANSWERED |
| 03:28.07 | jaytee | alancio, I'm using cdr_mysql too |
| 03:28.40 | alancio | do you have anything special in cdr.conf? |
| 03:29.05 | jaytee | are you using the Answer function in * to auto answer every call and then hand it off? |
| 03:29.53 | alancio | no, this call was a call from * to POTS |
| 03:31.03 | alancio | I have some calls that show FAILED on the CSV logs, before switching to mysql |
| 03:31.04 | jaytee | might be different then on an analog POTS line, I'm using SIP-SIP or SIP-PRI only. |
| 03:31.44 | alancio | ok I'll have to test a little more |
| 03:33.07 | mosty | alancio, how are you initiating the call? and what type of connection to the POTS do you have? |
| 03:34.01 | alancio | I use the Dial application, I have a zaptel card |
| 03:34.34 | [TK]D-Fender | alancio, All of your calls are considered "answered" because Zaptel by default does not do progress detection on analog channels. |
| 03:35.03 | [TK]D-Fender | alancio, If you notice upon dial you can see immediately in CLI that it says zap/XX answered" |
| 03:35.28 | alancio | oh, that sounds logical, and hard to solve |
| 03:35.42 | [TK]D-Fender | alancio, You can change this by setting "callprogress=yes", but thats also synonymous with "disconnect my calls at random=yes" |
| 03:36.09 | alancio | really? is that a bug with asterisk? |
| 03:36.25 | mosty | it's more of a lack of a feature than a bug |
| 03:37.56 | [TK]D-Fender | alancio, its analog don't ask much |
| 03:38.02 | [TK]D-Fender | alancio, want real progress, get a PRI |
| 03:38.24 | alancio | I don't know if that is even possible :( |
| 03:39.10 | alancio | a PRI is like a T1 or E1 from the phone company? |
| 03:39.23 | jaytee | yep |
| 03:39.38 | alancio | then I can't |
| 03:39.41 | jaytee | T1 is 24 channels with 1 for out of band signalling |
| 03:39.50 | jaytee | when using it for PRI |
| 03:40.02 | jaytee | so you get 23 for voice |
| 03:40.06 | alancio | ok |
| 03:41.06 | alancio | I'll just give callprogress=yes a try |
| 03:44.50 | jaytee | good luck! |
| 03:45.05 | alancio | thanks |
| 03:47.34 | jaytee | nite everyone |
| 03:47.40 | alancio | nite |
| 03:47.41 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 03:51.25 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-f6773d867e5862d8) |
| 03:51.46 | pigpen2 | [TK]D-Fender, long time. Question, can sip on asterisk bind to two interfaces yet? (both with their own ip address) |
| 03:53.26 | [TK]D-Fender | pigpen2, All or one. Your choice. |
| 03:53.36 | pigpen2 | interfaces or ip addresses. |
| 03:53.43 | [TK]D-Fender | pigpen2, same thing |
| 03:53.58 | pigpen2 | earlier ver's would only do multiple IP addresses on a single interface. |
| 03:54.17 | [TK]D-Fender | pigpen2, ummm.. nope |
| 03:54.31 | pigpen2 | Well, I must have had my head up my newbe ass at the time. |
| 03:54.32 | mosty | pigpen2, asterisk 1.2 can bind to all interfaces, i think 1.0 could also |
| 03:55.28 | rob0 | My * has SIP clients on the internal interface and connects to SIP servers via external interface. |
| 03:55.28 | pigpen2 | well, good. I am being forced to make sip available on an outside interface on an embeded linux box acting as a firewall. |
| 03:56.07 | pigpen2 | I am sure things will get goofy, due to the cheap bastards won't pony up for a static on the outside. |
| 03:56.59 | pigpen2 | Glad to hear sip acts like IAX does. |
| 03:57.01 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 03:57.07 | rob0 | Sic semper cheapskatemus. |
| 03:57.35 | pigpen2 | yeah. When things get weird, and they ask what will fix it, I will say, "Magic" |
| 03:58.06 | pigpen2 | Anyway, thanks for the quick confirmation. |
| 03:58.24 | pigpen2 | Maybe I can find more time to hang out in this channel. It has been about 6 months. |
| 03:58.41 | rob0 | Change of a dynamic IP will require a sip reload on the peer. |
| 03:58.53 | pigpen2 | yeah. Magic. |
| 03:59.01 | rob0 | (Dynamic DNS isn't enough, by itself.) |
| 03:59.21 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:59.37 | pigpen2 | yeah, I would rather have a static on the head end, then run a vpn using x509 certs via the remote. |
| 04:00.09 | pigpen2 | stkn_, welcome gentoo guy. |
| 04:05.25 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 04:23.38 | *** part/#asterisk Shazzamy (n=evies980@d206-75-75-10.abhsia.telus.net) |
| 04:32.36 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-17dca8aff70196ab) |
| 04:35.36 | *** join/#asterisk philip76 (n=philip@202.21.177.3) |
| 04:53.26 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 04:55.34 | yojimbo-san | I'm having problems getting a snom300 phone to send DTMF to my ISP's Asterisk voicemailbox; I can dial from the phone manually just fine, but if I try to set up a function key to call sip:*98w012345 (i.e. 012345 is the mailbox account name) I don't hear any times, and neither does the voicemail, it just times out. Can someone help supply me with a clue please? |
| 04:56.20 | Strom_L | yojimbo-san: that's not how sip works ;) |
| 04:56.47 | yojimbo-san | well, I forgot to say ;user=dialstring |
| 04:56.53 | yojimbo-san | does that help? |
| 04:56.54 | Strom_L | and anyway, don't leave your voicemail as a security hole; just enter the password when you call |
| 04:57.07 | Strom_L | auto-password is a stupid stupid stupid stupid stupid idea |
| 04:57.18 | yojimbo-san | that's not the password, that's the account name |
| 04:57.20 | yojimbo-san | :-) |
| 04:58.04 | Strom_L | erm, yeah |
| 04:58.12 | drmessano | *97? |
| 04:58.17 | Strom_L | sorry, ive had my head in a credit card procesing script all day |
| 04:58.24 | yojimbo-san | yep, that works for my current line :-) |
| 04:58.43 | yojimbo-san | but *98 will allow me to select a different mailbox, if I understand it riight |
| 05:07.54 | *** join/#asterisk Trifixxx (n=Mike@c-67-169-21-227.hsd1.ca.comcast.net) |
| 05:08.03 | Trifixxx | hey there d-fender! |
| 05:10.19 | [TK]D-Fender | yojimbo-san, there is no way to dial in with a sip phone that I know of that will also pass on DTMF. You should make a dedicated extension that will call voicemailmain without ever asking for your password |
| 05:11.07 | Trifixxx | what? no hello? |
| 05:11.11 | Trifixxx | cmon buddy |
| 05:11.19 | [TK]D-Fender | Trifixxx, hello |
| 05:11.27 | yojimbo-san | That's a shame. I also wanted to access a corporate phone system via an audio bridge, and that requires DTMF. It would have been nice to be able to do that on one keypress ... :-( |
| 05:11.42 | Trifixxx | did you find a good dialplan yet? |
| 05:11.44 | [TK]D-Fender | yojimbo-san, how is it connected? |
| 05:11.59 | [TK]D-Fender | Trifixxx, how many angels can dance on the head of a pin? |
| 05:12.11 | yojimbo-san | Public PSTN provides an access number, it says "hello" then you dial the 7-digit corporate extension |
| 05:12.21 | [TK]D-Fender | Trifixxx, and I gave you your sample last night. |
| 05:12.30 | [TK]D-Fender | yojimbo-san, and how do you get to the PSTN? |
| 05:12.48 | yojimbo-san | oh, that's just a case of dialling a public phone number |
| 05:12.57 | Trifixxx | oh yeah. my sample. |
| 05:13.22 | Trifixxx | i ws thinking about it, d-fender, and i actually think the right answer is "the asterisk dialplan is clumsy and inherently creates bad code." |
| 05:13.26 | Trifixxx | is that the right answer? |
| 05:13.29 | [TK]D-Fender | yojimbo-san, I just asked how you get to the PSTN. What piece of hardware or service do you use for this? |
| 05:13.37 | Trifixxx | i mean, goto and gosub are so 1985 Commodore 64 basic. |
| 05:13.39 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 05:13.56 | yojimbo-san | [TK]D-Fender: sorry, I misunderstood. a SIP phone (snom300) signed in to the ISP's asterisk service |
| 05:14.21 | [TK]D-Fender | Trifixxx, it certainly isn't structured programming, but "bad code" is so much worse based on who's writing. |
| 05:14.35 | [TK]D-Fender | yojimbo-san, Do you control that *? |
| 05:14.51 | yojimbo-san | [TK]D-Fender: nope, but they are helpful people if I know what to ask for |
| 05:15.05 | [TK]D-Fender | yojimbo-san, ok, how do THEY get to the PSTN? |
| 05:15.15 | yojimbo-san | [TK]D-Fender: unknown |
| 05:15.41 | Trifixxx | ok. |
| 05:15.47 | [TK]D-Fender | yojimbo-san, well you can try to have them use the D() Dial parameter to pass on digits. |
| 05:16.34 | yojimbo-san | [TK]D-Fender: thanks, that's something for me to look up then :-) |
| 05:17.47 | [TK]D-Fender | yojimbo-san, this may or may not work depending on how they terminate. if they go through 3rd party SIP termination that treats the call as "answered the moment it is placed then this will probably not work. but go ask. |
| 05:18.16 | [TK]D-Fender | yojimbo-san, and be sure to ask them to set up a pattern for you to use so you can pass on the extra digits in your dialstring. They would break it out when you dial |
| 05:18.32 | [TK]D-Fender | yojimbo-san, so that you could chage your password and just update the speed-dial on your phone. |
| 05:19.04 | yojimbo-san | [TK]D-Fender: OK, thanks. I'll try! |
| 05:19.34 | [TK]D-Fender | yojimbo-san, you're welcome. |
| 05:25.38 | *** join/#asterisk xenonex (n=xenonex@89.218.237.83) |
| 05:38.46 | *** join/#asterisk Frogzoo (n=Frogzoo@121.217.196.136) |
| 05:46.07 | [TK]D-Fender | ok, checkout time, back later. |
| 05:46.31 | *** part/#asterisk Trifixxx (n=Mike@c-67-169-21-227.hsd1.ca.comcast.net) |
| 05:51.30 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:06.36 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 06:11.26 | *** join/#asterisk nobesnickr (n=pmccaffr@ip72-201-157-30.ph.ph.cox.net) |
| 06:12.40 | nobesnickr | i am having an issue where no sound is transmitted to one of my cisco 7940's when called from another sip phone on the network, i have tried basic debugging but nothing has pointed me in the right direction. Does anyone have any ideas i can try? |
| 06:13.31 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:13.41 | nobesnickr | anyone? |
| 06:14.16 | mosty | tethereal |
| 06:14.22 | mosty | is there NAT involved? |
| 06:15.40 | nobesnickr | there is but the phone is registering on the asterisk box (hosted external from this network) and i the phone works perfect when calling outside lines through IAX |
| 06:16.04 | nobesnickr | and the users i am trying to call can hear my voice perfectly |
| 06:16.33 | mosty | IAX is not affected by NAT like SIP is |
| 06:16.53 | nobesnickr | yea i know, i wish there was a iax firware for these phones but sadly not |
| 06:16.56 | mosty | look up asterisk + sip + nat on the voip-info wiki, it's a very common problem |
| 06:17.24 | nobesnickr | i will do just that, thank you for your help :) |
| 06:19.01 | JT | ~sipnat |
| 06:19.02 | 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 |
| 06:19.06 | JT | look at the first url too |
| 06:19.27 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 06:25.13 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
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| 07:03.33 | nobesnickr | can anyone point me in the right direction please, i am trying to find the script or at least an idea of how to script my website to place a call when a user puts in their phone number, id like the system to connect their phone to mine |
| 07:04.32 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 07:05.32 | ikevin | you need to make a script who use socket for connecting to asterisk and launching the call and redirect it to a real line |
| 07:06.18 | nobesnickr | do you know what that would be called so i can research it by any chance? |
| 07:10.43 | *** join/#asterisk slyzhnyak (i=slyzhnya@82.200.254.104) |
| 07:10.56 | slyzhnyak | hi all! |
| 07:12.04 | slyzhnyak | someone knows where i can download prebuild debian etch package for Asterisk 1.6? |
| 07:12.07 | *** join/#asterisk af_ (n=getsmart@88-149-230-31.dynamic.ngi.it) |
| 07:15.45 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) [NETSPLIT VICTIM] |
| 07:15.45 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) [NETSPLIT VICTIM] |
| 07:15.45 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) [NETSPLIT VICTIM] |
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| 07:15.46 | *** join/#asterisk CCFL_Man2 (i=8de858c7@71.241.96.159) [NETSPLIT VICTIM] |
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| 07:15.46 | *** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net) [NETSPLIT VICTIM] |
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| 08:09.39 | the_5th_wheel | is there anyone in the states who can please test a number for me? Its a toll free one, and im being told by people they cant get thru |
| 08:12.31 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
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| 08:24.01 | Shotygun | Hi. Can anyone tell me in generally for which purposes asterisk needs zaptel's timing? Is it required for moh, queues or chanspy? |
| 08:24.42 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:24.58 | liri | is it possible inside a conference to support feature codes? |
| 08:32.08 | *** part/#asterisk nreinartz (n=chatzill@ntmail1.datus.com) |
| 08:39.56 | mvanbaak | Shotygun: conferencing and sla(which uses meetme in the background) |
| 08:41.18 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 08:53.47 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-246-236.balt.east.verizon.net) |
| 08:54.11 | SteveTotaro | oh what a beautiful morning, oh what a beautiful day.... |
| 08:55.12 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 08:59.17 | linuxmaniac | slyzhnyak: there is no such thing. |
| 08:59.47 | linuxmaniac | no one is working on it on Debian Voip team |
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| 09:07.52 | tzafrir | if anybody wants to, it would be nice |
| 09:08.07 | tzafrir | (work on an asterisk 1.6 deb package) |
| 09:08.43 | *** join/#asterisk af_ (n=getsmart@88-149-230-31.dynamic.ngi.it) |
| 09:25.30 | *** join/#asterisk sysadmin-lb22 (n=asdf@mail.splendor.net) |
| 09:25.59 | sysadmin-lb22 | hi ..I installed ztdummy and loaded it after that I reinstalled asterisk..however there is no meetme module any help ? |
| 09:28.37 | awk | hmm, span = 1,2,0 what is the 0 part used for? |
| 09:30.27 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 09:31.50 | tzafrir | sysadmin-lb22, module load app_meetme.so |
| 09:31.59 | tzafrir | does this do anything? |
| 09:32.08 | tzafrir | (in the asterisk CLI) |
| 09:33.00 | sysadmin-lb22 | tzafrir did that it executed and did not throw an error |
| 09:33.02 | awk | does the 0 have anything to do with signal strength? |
| 09:33.10 | sysadmin-lb22 | however when I did shwo applications after that there was not meetme either |
| 09:33.25 | awk | is your verbosity high enough? |
| 09:33.29 | awk | to get an warning message? |
| 09:34.05 | tzafrir | logger show channels |
| 09:34.09 | tzafrir | (to tell) |
| 09:34.28 | tzafrir | loading error is an "error" |
| 09:35.14 | sysadmin-lb22 | awk, verbosity is 4 |
| 09:35.28 | sysadmin-lb22 | tzafrir /var/log/asterisk/full File Enabled - Debug Verbose Warning Notice Error |
| 09:35.59 | tzafrir | that's too noisy for you to notice a simple error message :-) |
| 09:36.11 | tzafrir | What about "console"? |
| 09:37.05 | sysadmin-lb22 | tzafrir no errors in console |
| 09:37.21 | sysadmin-lb22 | tzafrir I just tried |
| 09:37.27 | sysadmin-lb22 | module load asdfadfadfa |
| 09:37.31 | sysadmin-lb22 | and it did not throw an error either |
| 09:37.40 | sysadmin-lb22 | so I think I should increse verbosity ? |
| 09:38.33 | tzafrir | hmm.. it is actually a "warning" |
| 09:38.46 | tzafrir | [080528-123817] WARNING[25905]: loader.c:665 load_resource: Module 'blabla' could not be loaded. |
| 09:39.00 | sysadmin-lb22 | tzafrir I just searched I dont have app_meetme.so on my ssytem |
| 09:39.20 | sysadmin-lb22 | all i have is app_meetme.c in /usr/src/asterisk/apps/ |
| 09:39.22 | awk | I was right it is LBO |
| 09:39.24 | tzafrir | <PROTECTED> |
| 09:39.52 | sysadmin-lb22 | how comes the meet me app is not being compiled ? |
| 09:39.58 | sysadmin-lb22 | I have ztdummy loaded and working |
| 09:40.08 | tzafrir | sysadmin-lb22, what about /usr/lib/asterisk/modules ? |
| 09:40.58 | *** join/#asterisk wordzilla (n=me@122.109.125.252) |
| 09:40.59 | sysadmin-lb22 | tzafrir no app_meet there |
| 09:42.17 | sysadmin-lb22 | I have downloaded the svn sources for 1.4 |
| 09:42.56 | tzafrir | sysadmin-lb22, so you need to rebuild asterisk . Probably re-run ./configure so it will pick up the presence of zaptel in the system |
| 09:43.20 | sysadmin-lb22 | tzafrir I thnk you are right ..I did not run ./configure after I installed zaptel |
| 09:43.39 | sysadmin-lb22 | tzafrir let me try that.."and of course you are right :p" |
| 09:43.49 | *** join/#asterisk fiddur (n=fiddur@78.82.252.60) |
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| 09:49.17 | *** part/#asterisk fgl (n=p@42.Red-88-1-215.dynamicIP.rima-tde.net) |
| 09:56.26 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 09:56.45 | *** join/#asterisk ikey (n=ikey@81.199.192.11) |
| 09:56.49 | iceyp | hey guys... is there a timeout for register => and if so how can this be extended as I'm having issues registering to a vendor far away |
| 09:56.59 | iceyp | they can see me as registered and I dont see it as registered my end |
| 09:57.02 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 09:57.18 | ikey | hi |
| 09:57.29 | ikey | i have a problem with sip can any one help |
| 09:57.31 | ikey | WARNING[1116941120]: Maximum retries exceeded on call 9519b2ec-c0da277a-4c6af309@10.10.0.4 for seqno 1 (Res |
| 09:57.32 | ikey | ponse) |
| 09:57.33 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 10:01.19 | viraptor | hey, does anyone know about an issue in * that causes recording second on or later Dial() to fail? produces only 190 bytes of wav... I can't find anything in tracker/google, but maybe I missed it |
| 10:01.25 | viraptor | ? |
| 10:06.04 | iceyp | I continue to get SIP/2.0 408 Request Timeout |
| 10:06.48 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:09.23 | *** join/#asterisk redax (i=redax@r6.hu) |
| 10:09.25 | redax | hi, |
| 10:11.01 | redax | I have a TE220 card here, and seems like it's in T1 mode, and not in E1 |
| 10:11.05 | redax | where can I configure that? |
| 10:14.51 | tzafrir | redax, you can set it in a jumper (IIRC) and/or in the module parameter t1e1override |
| 10:16.48 | redax | tzafrir: found |
| 10:17.02 | redax | I'm fool got the jumper in the documentation :) |
| 10:17.06 | redax | thanks |
| 10:20.42 | *** join/#asterisk [hC] (n=hardcore@S0106001346a4b813.vc.shawcable.net) |
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| 11:39.42 | *** join/#asterisk Vec (n=Vec@host-87-74-7-57.dslgb.com) |
| 11:40.36 | Vec | Hi, Whats the diffirence between, Dial(SIP/siptrunk/55555), and Dial(SIP/5555@siptrunk) ? |
| 11:43.15 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 11:43.18 | r0land | hello all |
| 11:44.48 | r0land | could someone help me with asterisk not able to transfer sip extensions plz! http://www.pastebin.ca/1031983 |
| 11:46.53 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 11:46.57 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 11:50.33 | slyzhnyak | r0land: sip.conf? |
| 11:50.42 | r0land | slyzhnyak k lemme pastebin it |
| 11:52.22 | r0land | slyzhnyak http://pastebin.ca/1031991 |
| 11:52.37 | *** join/#asterisk disposable (i=disposab@blackhole.sk) |
| 11:55.55 | Vec | Anyonw know what the diffirence is to what I asked earlier ? |
| 11:56.13 | disposable | what's the difference between playing .sln and .alaw files cpuworkloadwise? |
| 11:56.48 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
| 11:57.25 | ghenry | How do you pronounce IAX again? |
| 11:57.30 | ghenry | Not I A X |
| 11:57.32 | ghenry | ? |
| 11:57.50 | ghenry | BTW |
| 11:58.02 | r0land | slyzhnyak any advice? |
| 11:58.23 | ghenry | Anyone knwo of how to connect a VOIP trunk from Dubai, as UAE block all voip |
| 11:58.36 | slyzhnyak | exten => 120,n,Goto(spa,${EXTEN}192.168.0.111:5061,1) |
| 11:58.48 | slyzhnyak | what does it mean? |
| 11:59.25 | r0land | it means tht for any extension punched in earlier in "waitexten" tht comes from 120.. to go to context spa and dial the extension |
| 12:00.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:00.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:01.00 | r0land | slyzhnyak its not the right way to do it ? |
| 12:04.58 | slyzhnyak | not |
| 12:05.08 | r0land | any advice how to fix it ? |
| 12:05.29 | slyzhnyak | DISA application |
| 12:06.02 | slyzhnyak | http://www.voip-info.org/wiki-Asterisk+cmd+DISA |
| 12:07.29 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:07.29 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:10.04 | slyzhnyak | r0land you can try exten => _X.,1,Goto(spa,${EXTEN},1) |
| 12:10.48 | r0land | slyzhnyak thts to b added in sPA context |
| 12:10.50 | r0land | or sipura-line |
| 12:11.49 | slyzhnyak | WaitExten reexecutes current context, 120,n,Goto... never executes |
| 12:12.24 | slyzhnyak | sipura-line |
| 12:13.06 | slyzhnyak | when WaitExten finished sipura-line context reexecuted with new extension |
| 12:13.16 | *** join/#asterisk eXistenZ (i=pectic@unaffiliated/existenz) |
| 12:13.16 | slyzhnyak | but you have only 120 in it |
| 12:13.52 | slyzhnyak | sorry for my english |
| 12:13.59 | slyzhnyak | do you understand me? |
| 12:15.16 | r0land | slyzhnyak no |
| 12:15.19 | r0land | slyzhnyak wht do u mean |
| 12:16.16 | slyzhnyak | http://pastebin.ca/1032007 |
| 12:19.20 | *** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk) |
| 12:19.26 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:19.31 | slyzhnyak | r0land try this http://pastebin.ca/1032016 |
| 12:19.36 | r0land | k |
| 12:20.29 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 12:25.43 | *** join/#asterisk ukdolphin (n=ukdolphi@host86-175-56-89.wlms-broadband.com) |
| 12:26.21 | r0land | slyzhnyak http://pastebin.ca/1032024 |
| 12:26.28 | r0land | it gave me an error.. |
| 12:27.20 | ukdolphin | hi all, I have just made a new install of asterisk 1.4.20 on an openvz node. I have so far used the manager interface to setup a sip user. The Sip phone can register ok, but everytime i try to call anything like the echo test the system dies with a segfault but no clue as to where. |
| 12:27.23 | *** join/#asterisk jack_sparo (n=eddy@91.73.203.98) |
| 12:27.27 | ukdolphin | does anyone have any ideas? |
| 12:27.41 | jack_sparo | looking for zap patched to detect dialtone, anyone has any idea about it? |
| 12:27.50 | jack_sparo | ukdolphin repeat ur question plz |
| 12:28.01 | ukdolphin | I have just made a new install of asterisk 1.4.20 on an openvz node. I have so far used the manager interface to setup a sip user. The Sip phone can register ok, but everytime i try to call anything like the echo test the system dies with a segfault but no clue as to where. |
| 12:28.05 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:29.47 | [TK]D-Fender | ukdolphin: right off the bat I'd say "upgrade" |
| 12:30.03 | slyzhnyak | r0land just remove DigitTimeout |
| 12:30.12 | ukdolphin | i meant 1.4.20.1 download yesturday |
| 12:30.16 | r0land | slyzhnyak k |
| 12:30.26 | slyzhnyak | just comment it |
| 12:31.03 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 12:31.44 | slyzhnyak | PCadach was here very long time ago? |
| 12:32.12 | jack_sparo | looking for zap patched to detect dialtone, anyone has any idea about it? |
| 12:32.23 | lmadsen | ukdolphin: only way to find out is to get a backtrace with DONT_OPTIMIZE enabled, and to probably open a bug with the backtrace (NOT THE COREDUMP FILE) with a very descriptive description |
| 12:33.02 | ukdolphin | lmadsen: not sure how to? |
| 12:33.12 | lmadsen | ukdolphin: what distro? |
| 12:33.32 | lmadsen | 1) install gdb 2) read backtrace.txt in the 'doc' directory of your asterisk source |
| 12:33.55 | jack_sparo | lmadsen, any idea dude about zap patch? |
| 12:34.01 | r0land | slyzhnyak didnt work either.. |
| 12:34.02 | lmadsen | jack_sparo: I didn't answer, so no. |
| 12:34.10 | ukdolphin | centos 4.6 onto off openvz |
| 12:34.35 | jack_sparo | sorry dude |
| 12:34.39 | jack_sparo | lmadsen :) |
| 12:34.57 | *** join/#asterisk cjk (n=cjk@vodsl-11071.vo.lu) |
| 12:36.27 | ukdolphin | would 1.6 be any better? |
| 12:37.47 | slyzhnyak | tell me what are u doing and want? |
| 12:37.53 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:40.20 | Nobbie | hi, i'm having problems using dialparties.agi and have narrowed it down to the parts which run "database get". when dialing an extension, it can sometimes takes up to 4 seconds to complete a "database get" command. |
| 12:40.54 | *** join/#asterisk ikevin (n=kevin@kevin.linux-fr.net) |
| 12:41.01 | Nobbie | time -p /var/lib/asterisk/astdb > /dev/null completes in under a second |
| 12:41.12 | Nobbie | astdb is 400KB |
| 12:41.45 | Nobbie | what can i do to optimize the use of astdb ? |
| 12:41.48 | [TK]D-Fender | Nobbie: ... |
| 12:41.50 | [TK]D-Fender | ~freepbx |
| 12:41.51 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 12:42.34 | tzafrir | nah. a 400kb berkeley db should not be a performance issue |
| 12:42.34 | Nobbie | ok, but i've narrowed it down to "database get" which is an AGI command and hence more related to Asterisk surely ? |
| 12:43.47 | tzafrir | any chance your system is in bad lack of memory, and occasionally needs to swap in some memory to answer your 'database' command? |
| 12:44.32 | tzafrir | jack_sparo, what patch do you mean? |
| 12:45.17 | tzafrir | on the FXO? |
| 12:46.13 | tzafrir | Steve Davis (of the UK) wrote such patch , right? |
| 12:46.55 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 12:47.31 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 12:49.29 | Nobbie | tzafrir: not a chance, the server has 3GB of RAM and is dedicated to Asterisk |
| 12:51.24 | tzafrir | jack_sparo, http://bugs.digium.com/view.php?id=12382 ? |
| 12:51.58 | tzafrir | (And the name is Steve Davies, and not what I miswrote above) |
| 12:54.22 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
| 12:56.55 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 12:57.18 | hsv-al | hello - are we all looking forward to another long & glorious day on irc? :) |
| 12:57.31 | pawel | no |
| 12:58.15 | hsv-al | d-fender is actually up early |
| 12:58.19 | hsv-al | so I know the day has started |
| 12:58.24 | hsv-al | idle / 16m 19s |
| 13:02.01 | hsv-al | people are still waking up |
| 13:02.09 | hsv-al | it's as if the time is 3am now |
| 13:03.40 | cjk | hi, i would to change the callerid on pickups and transfers on the phoen doing the pickup or getting the attended transfer. i know asterisk doesnt work like this. but is there any hack that i could do? |
| 13:03.55 | *** join/#asterisk UnixDog (n=UnixDog@254.69.118.70.cfl.res.rr.com) |
| 13:04.02 | *** join/#asterisk s0lid (n=s0lid@121.97.140.100) |
| 13:04.16 | UnixDog | when is the major flaw in asterisk going to be fixed ? |
| 13:04.35 | hsv-al | ? |
| 13:04.55 | [TK]D-Fender | UnixDog: Care to narrow that down a bit, we have so many :p |
| 13:04.56 | UnixDog | if I have a exten 1000 on my box and my friend has a exten 1000 on his system and you do a sip uri dial it fails |
| 13:05.21 | [TK]D-Fender | UnixDog: make the names unique |
| 13:05.43 | UnixDog | well with most gui setups you cant |
| 13:05.50 | UnixDog | thats the issue |
| 13:06.14 | UnixDog | point and fact freepbx |
| 13:06.19 | [TK]D-Fender | UnixDog: That is most definitely not our problem... |
| 13:06.34 | UnixDog | well it is |
| 13:06.44 | UnixDog | because its a sip uri issue |
| 13:06.44 | [TK]D-Fender | UnixDog: take their inflexibility up with them. |
| 13:07.20 | [TK]D-Fender | UnixDog: Feel free to try and support a patch or one of the now 1/2 dozen chan_sip replacements out there |
| 13:07.58 | UnixDog | chan_sip replacements ? |
| 13:08.07 | Nobbie | Unix: in fact, all vendors might, including Cisco will tell you to use unique extensions |
| 13:08.46 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 13:08.52 | drmessano | I dont understand |
| 13:08.57 | hsv-al | here we go again |
| 13:09.05 | drmessano | If I have an exten 1000, I cant dial 1000@something? |
| 13:09.05 | Nobbie | Unix: as a workaround, you can use speed dials |
| 13:09.07 | hsv-al | more "drmessano", more "d-fender |
| 13:09.10 | hsv-al | another day :) . . . . |
| 13:09.45 | drmessano | howdy |
| 13:09.56 | hsv-al | so your not some fool using "dr" for irc attention I see |
| 13:10.03 | hsv-al | the conversation yesterday seemed to prolonged for a complex lie |
| 13:10.20 | drmessano | heh |
| 13:10.29 | drmessano | ~drmessano |
| 13:10.30 | jbot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily |
| 13:10.33 | hsv-al | After watching you talk with forz yesterday |
| 13:10.42 | hsv-al | Id rather pop out partial differentail equations all day |
| 13:10.45 | hsv-al | then listen to radiation talk |
| 13:11.06 | drmessano | RF is a pain in the ass, plain and simple |
| 13:11.39 | hsv-al | well, I guess if people have that level of fundamental understanding |
| 13:11.53 | hsv-al | it can bleed over into solving issues, possibly relating to their work w/ this software. |
| 13:13.26 | drmessano | My first area of expertise was RF engineering.. have a piece of wallpaper for 2 years of tech school for electronics and RF.. then got into IT... |
| 13:13.43 | hsv-al | I'm wondering if I should use an alpha value of .01, and give a 99% prediction interval that drmessanos talk was not factual or not? :) |
| 13:13.47 | hsv-al | *laugh* |
| 13:14.44 | drmessano | I always had a soft spot for telco (how many people do you know bought a butt set because they just wanted one?) and asterisk make it possible to do some neat stuff with telco and a PC.. So there I went.. |
| 13:14.53 | Katty | morning |
| 13:15.10 | hsv-al | hello katty did you grind last night? |
| 13:15.14 | hsv-al | and i mean pve grind :) |
| 13:15.15 | drmessano | After that it became obsessive compulsiveness lol |
| 13:15.58 | hsv-al | well, I was just kicking back watching you two. Im pursuing ccvp cert, but I'm trying to break away from cisco |
| 13:16.33 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 13:16.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:17.23 | drmessano | CCVP is good if you want a job babysitting CCM in large environments.. |
| 13:17.50 | hsv-al | well, thats what I was thinking |
| 13:17.52 | Katty | hsv-al: yeah. badlands. |
| 13:18.08 | hsv-al | Theres a few TS jobs in town that are paying 130k for degree + ccvp + secret clearance |
| 13:18.21 | drmessano | IMO, it's more of a tax Cisco puts on vendors who want to sell their products |
| 13:18.22 | hsv-al | working on a siprnet |
| 13:18.44 | drmessano | that's cool |
| 13:18.47 | *** join/#asterisk Mark17 (n=mark@vnc.tt.streamservice.nl) |
| 13:19.23 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 13:19.33 | hsv-al | its sort of ridiculous, but some of these contract jobs are paying graduates in CPE/CS, 95, 100k doing half ass unix admin work |
| 13:19.46 | *** join/#asterisk ice_croft (n=nolan@85.172.54.214) |
| 13:19.47 | hsv-al | because the government wants skilled workers, nothing more to it |
| 13:20.09 | drmessano | yeah |
| 13:21.55 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:22.21 | Katty | man. i am just... leaking anger from my ears this morning |
| 13:22.26 | Katty | moody! angry!!! |
| 13:22.36 | Katty | next i'll be craving pickles and ice cream, the way my luck is going >.< |
| 13:22.55 | pawel | :> |
| 13:24.20 | *** join/#asterisk francogwapito (n=chatzill@203.82.47.128) |
| 13:24.50 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 13:24.55 | lirakis_work | hrm.. |
| 13:25.06 | lirakis_work | im trying to get everything but the last char from a variable |
| 13:25.08 | lirakis_work | on asterisk 1.2 |
| 13:25.15 | lirakis_work | so i dont think the len() function exists |
| 13:25.16 | drmessano | Are you pregnant? |
| 13:25.29 | lirakis_work | drmessano: err.. i hope not |
| 13:25.36 | drmessano | lol |
| 13:26.04 | lirakis_work | ive tried ${EXTEN:0:-1} .. but no luck .. the crux is that i dont know the length of the extension |
| 13:26.38 | *** join/#asterisk anthooooooooo (n=AnthonyC@LAubervilliers-153-52-32-150.w217-128.abo.wanadoo.fr) |
| 13:26.40 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 13:26.42 | anthooooooooo | hello |
| 13:27.08 | [TK]D-Fender | lirit does. |
| 13:27.19 | [TK]D-Fender | lirakis_work: Yes, that function exists. |
| 13:27.26 | [TK]D-Fender | lirakis_work: look harder next time. |
| 13:27.43 | lirakis_work | [TK]D-Fender: lhmm .. okay .. i looked in the 1st ed atoft |
| 13:27.47 | lirakis_work | maybe i missed it |
| 13:28.04 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 13:28.08 | [TK]D-Fender | lirakis_work: Next time go look in CLI |
| 13:28.46 | *** join/#asterisk robevans (n=robevans@OL6-231.fibertel.com.ar) |
| 13:29.10 | anthooooooooo | We work with Asterisk 1.4.19. We have some problems: Sometimes When I call, We have a sound in "background". For example, We can hear the voice of my voicemail in the same time of my call. Why have I this problem? My phone is an Aastra 55i |
| 13:29.11 | lirakis_work | [TK]D-Fender: I also tried verbose(${EXTEN:0:[LEN(${EXTEN)-1]) .. but didnt get that to work either |
| 13:29.40 | anthooooooooo | thanks for your help |
| 13:29.56 | [TK]D-Fender | lirakis_work: Of course not. Thats no way to call a function. |
| 13:30.43 | lirakis_work | [TK]D-Fender: uhh.. okay.. |
| 13:31.04 | [TK]D-Fender | lirakis_work: your brackets and braces are so mismatched in there its almost funny. |
| 13:31.27 | lirakis_work | yeah i saw that |
| 13:31.35 | lirakis_work | .. that was just typed in here .. i didnt copy paste |
| 13:32.02 | lirakis_work | [TK]D-Fender: but given that the brackets & braces are matched... i should be able to call the len function and get the substring like that correct? |
| 13:32.16 | [TK]D-Fender | lirakis_work: yes, very easy. |
| 13:32.19 | *** join/#asterisk go|dfish (i=goldfish@losers.yore.ma) |
| 13:33.12 | lirakis_work | exten => _X.,n,verbose( ${EXTEN:0: [ LEN( ${EXTEN} )-1 ] } ) |
| 13:33.18 | lirakis_work | it hink that looks good but ill try it out |
| 13:33.25 | go|dfish | Hey guys, just wondering if anybody has a Digium 410p card? There was a power outage at work, and the status lights ont he card are no longer operational, outgoing calls are also failing. I'm just wondering if the status lights are not on at all, whether it means the card itself is busted? |
| 13:33.54 | [TK]D-Fender | lirakis_work: [ <- is used for expressions, not function calls. |
| 13:33.55 | go|dfish | Nothing has changed in the config of the machine, modules are loaded, etc. |
| 13:34.08 | [TK]D-Fender | lirakis_work: Go read the book again. |
| 13:34.27 | go|dfish | The card's manual mentions this channel, so I thought i'd give a try asking here... |
| 13:34.44 | [TK]D-Fender | go|dfish: they only light up once the driver is initialized IIRC. Go make sure zaptel is loaded. |
| 13:34.44 | lirakis_work | [TK]D-Fender: so i dont need the brackets to subtract the result of the function |
| 13:34.48 | *** join/#asterisk amaache (n=maache76@41.221.26.84) |
| 13:35.17 | [TK]D-Fender | lirakis_work: Ah, well you do need it for the math bit, but then again its $[] not [] |
| 13:35.22 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 13:35.28 | [TK]D-Fender | lirakis_work: and you aren't referencing your function at all. |
| 13:35.44 | lirakis_work | [TK]D-Fender: referencing my function? |
| 13:36.07 | [TK]D-Fender | lirakis_work: Go look at samples of how functions are called. |
| 13:36.09 | *** join/#asterisk _gm (n=gmustafa@static-host119-30-120-210.link.net.pk) |
| 13:36.22 | lirakis_work | oy... |
| 13:36.25 | go|dfish | [TK]D-Fender: Hm, zaptel appears to be loaded, ztcfg -vvv says 0 channels configured, though. |
| 13:36.51 | [TK]D-Fender | go|dfish: modprobe its driver, double check its config files, pastebin everything |
| 13:36.53 | [TK]D-Fender | ~pb |
| 13:36.54 | 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 |
| 13:36.54 | Nobbie | argh whyyyyyyy...... |
| 13:36.55 | [TK]D-Fender | ^^^^^^^^^^ |
| 13:37.09 | Nobbie | 15seconds to do a simple "database get" |
| 13:37.27 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:38.01 | lirakis_work | [TK]D-Fender: can i use that len() function as a parameter of another function? .. im looking at pages of function calls.. and i dont see what your getting at |
| 13:38.25 | jack_sparo | looking for zap patch to detect dialtone, anyone has any idea about it? my asterisk is not dropping the channels when i hangup the phone |
| 13:38.29 | [TK]D-Fender | lirakis_work: You don't seem to know how to call a function at all... |
| 13:39.13 | [TK]D-Fender | lirakis_work: and yes you can nest functions. |
| 13:40.07 | go|dfish | [TK]D-Fender: The cards driver is loaded 'wct4xxp', It was working up until yesterday after a power cut, the configurations haven't changed. Hence why I thought it was the card itself. I didn't set up this system, I inherited so I'll have to do some reading before I'll know what to pastebin and check configs :-) |
| 13:40.09 | Nobbie | ah, CPU is 34% waiting. does that indicate deadlock avoidance/detection problem ? |
| 13:41.24 | drmessano | I just confirmed the SIP URI issue.. thats ugly |
| 13:41.45 | r0land | hey all |
| 13:42.10 | r0land | anyone ever faced a prob in asterisk tht it doesnt find the sip extension in sip.conf even though its there? http://www.pastebin.ca/1031983 |
| 13:42.29 | Nobbie | r0land: after doing a "asterisk -rx 'reload' " |
| 13:42.31 | Nobbie | >? |
| 13:42.48 | r0land | Nobbie let me try |
| 13:42.49 | lirakis_work | [TK]D-Fender: so ... i still have no idea what im doing wrong .. can you be any more specific? .. |
| 13:43.23 | lirakis_work | [TK]D-Fender: ive tried changing to exten => _X.,n,verbose( ${EXTEN:0:$[${LEN( ${EXTEN} )}-1]} ) |
| 13:43.56 | [TK]D-Fender | lirakis_work: shitespace = BAD <----- |
| 13:44.00 | [TK]D-Fender | \whitespace* |
| 13:44.24 | lirakis_work | [TK]D-Fender: exten => _X.,n,verbose(${EXTEN:0:$[${LEN(${EXTEN})}-1]}) |
| 13:44.46 | r0land | Nobbie yes even |
| 13:44.58 | lirakis_work | [TK]D-Fender: was it just adding the ${ } around the len() call that you were talkuing about? |
| 13:45.13 | [TK]D-Fender | lirakis_work: Concerning referencing a function, yes. |
| 13:46.10 | lirakis_work | [TK]D-Fender: that would have been a lot clearer .. " your missing ${ } around your len() call ... in otherwords.. you arent referencing the function" |
| 13:46.26 | lirakis_work | [TK]D-Fender: thanks for your help though |
| 13:47.03 | [TK]D-Fender | lirakis_work: Sorry, I had to let you come to this yourself or it might not stick. |
| 13:47.05 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:47.17 | lirakis_work | [TK]D-Fender: fair enough .. |
| 13:47.18 | r0land | [TK]D-Fender hello again |
| 13:47.23 | anthooooooooo | I use Asterisk 1.4.19. We have some problems: Sometimes When I call, We have a noise in "background". For example, I can hear the voice of my voicemail in the same time of my call. It seem that my call take a channel that is not close. Is it possible? |
| 13:47.36 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.207.130) |
| 13:48.00 | [TK]D-Fender | r0land: there is no pattern match for 10 in THAT CONTEXT. |
| 13:48.14 | r0land | [TK]D-Fender hmm ok ill figure it out |
| 13:48.15 | r0land | thanks :) |
| 13:49.00 | [TK]D-Fender | r0land: where do YOU see an exten that will match '10' in [sipura-line] ? |
| 13:49.24 | r0land | exten => 1X.,n,Goto(spa,${EXTEN}@192.168.0.111:5061,1) |
| 13:49.29 | r0land | " 1X." |
| 13:49.57 | [TK]D-Fender | r0land: pastebin it all |
| 13:50.00 | r0land | k |
| 13:50.05 | UnixDog | 1x. would = any number dialed with a 1 |
| 13:50.17 | [TK]D-Fender | UnixDog: Not that line.... |
| 13:50.41 | [TK]D-Fender | and what the hell is that bastardized goto with a URI in it? |
| 13:51.13 | [TK]D-Fender | I swear people pull every random bit of syntax they can find and mash it together for no good reason. |
| 13:51.14 | r0land | [TK]D-Fender http://www.pastebin.ca/1032091 |
| 13:51.56 | [TK]D-Fender | r0land: Your broken attempt at an IVR does not look at [spa] for extens to dial <- |
| 13:52.09 | hsv-al | d-fender, I have a 12pack of sugar-free redbull, and "5 hour energy" |
| 13:52.12 | hsv-al | which do you want? |
| 13:52.32 | r0land | [TK]D-Fender exten => 1X.,n,Goto(spa,${EXTEN}@192.168.0.111:5061,1)<<== shouldnt this do the trick ? |
| 13:52.33 | [TK]D-Fender | r0land: and THIS... what is a URI doing in the middle of a GOTO? exten => 1X.,n,Goto(spa,${EXTEN}@192.168.0.111:5061,1) ; Goto the correct extension |
| 13:52.48 | [TK]D-Fender | r0land: and next, that isn't even a PATTERN. Go look at whats clearly missing. |
| 13:52.49 | r0land | "uri" = ? |
| 13:53.06 | [TK]D-Fender | r0land: why is there an IP ADDRESS in a Goto? |
| 13:53.12 | r0land | hmm |
| 13:53.14 | *** join/#asterisk xenonex (n=xenonex@89.218.237.83) |
| 13:53.28 | r0land | k done |
| 13:53.36 | r0land | but still how can i direct it to spa! |
| 13:53.50 | [TK]D-Fender | r0land: include <---- |
| 13:53.58 | r0land | oh |
| 13:54.06 | [TK]D-Fender | r0land: Go read the book on dialplan basics again. You seem to have fallen completely off the wagon. |
| 13:54.07 | r0land | freaking hell if thts the solution im gonna kill myself |
| 13:54.20 | hsv-al | r0land, dont fret |
| 13:54.24 | hsv-al | accept it for what it is, become a clear |
| 13:54.31 | hsv-al | Dynetics, by L.Ron Hubbard |
| 13:54.36 | [TK]D-Fender | r0land: And whats at that IP? |
| 13:54.37 | r0land | [TK]D-Fender i kinda got the hang of it for a while back... ive set up everything now im lost again :( |
| 13:54.54 | r0land | [TK]D-Fender the other asterisk |
| 13:55.52 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:56.35 | hsv-al | r0land, think back to 15 years ago |
| 13:56.37 | jaytee | is using SLA the way to go if I have 3 users and want 1 of the phones that's mulitline to have the other 2 extensions ring there also? |
| 13:57.26 | [TK]D-Fender | jaytee: no. |
| 13:57.53 | [TK]D-Fender | jaytee: setup a separate registration and dial both devices for that other users extens. |
| 13:58.23 | jaytee | [TK]D-Fender, that's what I was thinking as an alternative to setting up SLA |
| 13:58.42 | jaytee | [TK]D-Fender, thanks for confirming my suspicions. |
| 13:58.46 | *** join/#asterisk kamh (n=q@host-81-190-236-85.wroclaw.mm.pl) |
| 13:59.16 | [TK]D-Fender | jaytee: *'s fake SLA was only made with LINES in mind, |
| 13:59.33 | russellb | it works for extensions, as well. |
| 13:59.43 | russellb | it's just dialplan. |
| 13:59.55 | jaytee | that's kind of the impression I got from "the book" and a couple articles on digium's site |
| 13:59.57 | r0land | [TK]D-Fender thanks tht did the trickj :) |
| 14:00.05 | russellb | i need to document how to do it ... |
| 14:00.16 | r0land | [TK]D-Fender thanks for ur help :) |
| 14:01.23 | [TK]D-Fender | r0land: Just because I'm feeling generous : http://www.pastebin.ca/1032101 |
| 14:01.58 | [TK]D-Fender | jaytee: actually... you don't need to make a separate reg at all unless you want to know that the call was for them. |
| 14:02.15 | [TK]D-Fender | jaytee: but there is all sorts of other dialplan trickery you can do even on a single reg. |
| 14:02.29 | r0land | [TK]D-Fender thanks :) everything works perfectly.. now need to wrap my mind around voicemail.. and ill finaly b done |
| 14:02.32 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 14:03.14 | Katty | dumdedum |
| 14:04.17 | *** join/#asterisk bogphanny (n=Miranda@ppp121-45-58-153.lns11.adl2.internode.on.net) |
| 14:04.21 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:04.28 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:04.51 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 14:08.18 | Katty | today is annoying. i should just play my pally and ignore work |
| 14:12.19 | bogphanny | I have recently setup an Asterisk service (self-confessed n00b), whereby a caller calls in using our SIP trunk, lands in the IVR, then chooses an option to be transfered to an external number using the same SIP trunk. This currently works fine, though my Asterisk service remains connected as a "middle-man" for the duration of the call. Is there a way that I can initiate a call transfer, so my Asterisk box is then removed |
| 14:13.54 | tzafrir | go|dfish, it means you have nothing in /etc/zaptel.conf . compare to /proc/zaptel/* |
| 14:14.20 | tzafrir | (oops, was looking back a bit) |
| 14:14.53 | pigpen | bogphanny, yeah, probably have your sip provider forward the call. |
| 14:15.01 | pigpen | rather than hitting your pbx |
| 14:15.39 | bogphanny | problem is that the external number is chosen based on the options selected in the IVR of the PBX |
| 14:15.47 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 14:16.47 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 14:17.19 | Vec | Please help, can't figure out the diffirence between Dial(SIP/trunk/5555) and Dial(SIP/5555@trunk) ? |
| 14:17.32 | disposable | i've uploaded my custom sounds to /var/lib/asterisk/sounds/custom/ but asterisk says file does not exist in any format. i tried /usr/local/share/asterisk/sounds/ but same result. what am i missing? (it's on debian, ast 1.4.19.1, files are in sln and wav 8000hz,16bit) |
| 14:17.34 | *** join/#asterisk s0lid (n=s0lid@125.60.135.68) |
| 14:17.47 | go|dfish | tzafrir: Thanks for your respsone, yeah, /proc/zaptel/1/ exists, and zaptel.conf appears to have everything commented out. The really weird thing is everything was working up until yesterday, when there was a power cut. That's why my initial thought was the card itself is fried. |
| 14:18.00 | Vec | disposable : try putting them in /var/lib/asterisk/sounds/ |
| 14:18.25 | tzafrir | you need something in zaptel.conf at boot time, not for normal operation |
| 14:18.29 | disposable | Vec: didn't help |
| 14:18.30 | Vec | disposable : also if they in custom address them as custom/soundname |
| 14:19.02 | Vec | disposable : check there permissions chmod 655 them ? |
| 14:20.53 | disposable | Vec: didn't help. it must be something obvious. do i need to enable something in config files? |
| 14:22.08 | tzafrir | go|dfish, you probably need to unrem some lines or recreate them... |
| 14:22.26 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 14:24.05 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 14:24.37 | *** join/#asterisk railsmunky (n=nick@collaboration.capuk.org) |
| 14:24.51 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:25.13 | go|dfish | tzafrir: Hm, apparantly this card doesn't use the zaptel driver, but rather the misdn one. Bah, sorry, this is rather new to me. |
| 14:25.15 | railsmunky | Howdy people. Can anyone help me, i'm trying to get keypresses from a caller whilst in a queue. Is this possible? |
| 14:25.16 | railsmunky | http://pastebin.com/m2eb17b84 |
| 14:25.25 | disposable | Vec: i've now got them in /var/lib/asterisk/sounds, in /var/lib/asterisk/sounds/custom and in /usr/local/share/asterisk/sounds/, yet it still can't see them :( |
| 14:25.34 | tzafrir | go|dfish, what card is it? |
| 14:25.34 | railsmunky | that's the relevent config i have at the moment |
| 14:25.53 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 14:25.59 | [TK]D-Fender | railsmunky: Yes, it is. set the "context=" in your queue definition. |
| 14:26.28 | UnixDog | ok setting fromdomain= is not working on extensions |
| 14:26.30 | go|dfish | tzafrir: Digium b410p |
| 14:26.30 | Katty | i'm am SO grumpy today |
| 14:26.45 | the_5th_wheel | is there anyone in the states who can please test a number for me? Its a toll free one, and im being told by people they cant get thru. |
| 14:27.07 | railsmunky | context of the queue ie context=clu or context of the main extension? |
| 14:27.18 | slyzhnyak | r0land _1X. |
| 14:27.29 | UnixDog | this sip uri issue is a major issue |
| 14:27.35 | [TK]D-Fender | railsmunky: the context whose single digit eaxtens you want the queue to be able to exit to |
| 14:27.36 | UnixDog | its killing me |
| 14:28.04 | [TK]D-Fender | UnixDog: If your GUI was smarter you could have gotten out of this. |
| 14:28.17 | UnixDog | its not my gui |
| 14:28.22 | railsmunky | [TK]D-Fender: aha okeokde. Great thanks |
| 14:28.39 | [TK]D-Fender | UnixDog: You said it wouldn't let you pick non-numeric peer names... |
| 14:28.47 | UnixDog | and I just get clients bitching because things are broken |
| 14:28.52 | slyzhnyak | how stable is SS7 suppeort in asterisk? |
| 14:28.53 | *** join/#asterisk FreedomBI (n=freedomb@mn01.freedombi.com) |
| 14:29.06 | UnixDog | thats freepbx and its not my gui |
| 14:29.34 | [TK]D-Fender | UnixDog: well rename them out of the way. If you can't, oh well... |
| 14:29.45 | drmessano | Alphanumeric isn't the issue, it's the workaround |
| 14:30.08 | drmessano | If dialing 1000@something didn't dial the internal 1000 extension, you wouldn't need one |
| 14:30.25 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
| 14:31.10 | *** join/#asterisk golumn (n=golumn@201.220.132.138) |
| 14:31.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:32.23 | golumn | Hi guys I want to record the conversations on a channel. Right now what I am doing it Record(myfile.wav,3,0,q) and the make a Dial(). it creates the file but it don't record anything. I also try with Monitor. It works fine the only problem is that it dive the file in and out, and I want a single one |
| 14:33.54 | [TK]D-Fender | golumn: Record is for you to record a sound file for the active channel. Monitor is for recording CALLS. |
| 14:34.16 | [TK]D-Fender | golumn: "core show application monitor" <- read the instructions. |
| 14:34.28 | *** join/#asterisk mkoch (n=koch@228-177.static.ew.hu) |
| 14:34.30 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 14:35.20 | mkoch | hi |
| 14:35.23 | Katty | runs around murdering. |
| 14:35.55 | seanbright | that seems excessive |
| 14:36.10 | Katty | i'm feeling excessively angry. |
| 14:36.13 | [TK]D-Fender | seanbright: Yeah I like being able to do that from the comfort of my armchair |
| 14:36.26 | golumn | thanks |
| 14:36.28 | Corydon76-dig | sends Katty to St Petersburg, FL |
| 14:36.30 | [TK]D-Fender | links Google Maps into ICBM launch control. |
| 14:36.36 | pawel | Katty: too much coffee |
| 14:37.16 | seanbright | Katty: ok, i'll bite... why are you excessively angry? |
| 14:37.35 | jbeez | whats in st petersburg FL? |
| 14:37.39 | Nobbie | would it be a bad idea to run the asterisk process on linux with nice value of -3 to give it higher priority then default ? |
| 14:38.05 | jbeez | Hello Mr Bright |
| 14:38.06 | Corydon76-dig | jbeez: someone on the lists who has been pissing on open wounds |
| 14:38.12 | seanbright | jbeez: hello |
| 14:38.21 | mkoch | i know, that everybody hates newbies with questions, but may i ask some help? :) |
| 14:38.30 | [TK]D-Fender | ~ask |
| 14:38.31 | jbot | somebody said ask was 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. |
| 14:38.32 | seanbright | mkoch: ask away |
| 14:38.55 | jbeez | Corydon76-dig: which lists? sorry I'm not familiar, and what are they doing? just being total dickwads? |
| 14:40.07 | UnixDog | ok back to this issue |
| 14:42.42 | mkoch | ok. so I want to use asterisk in a project. I have 10 student pc's, and 10 instructor pc's. Every student has an own conference room, and any other student or instruktor can join. I have to do it all with Linux and command line scripts (driven by an own GUI) and backup every conversation to a backup server. Is it possible with Asterisk? |
| 14:42.50 | Vec | disposable : what about an installed one like tt-monkeys |
| 14:43.07 | liri | is it possible inside a conference to support feature codes? like inside a conference room (meetme) to dial *200 and it will dial extension 200? |
| 14:44.00 | Vec | Please help, can't figure out the diffirence between Dial(SIP/trunk/5555) and Dial(SIP/5555@trunk) ? |
| 14:44.48 | lirakis_work | is there a way to capture the return result of a system() call ? i think that SYSTEMSTATUS only contains a FAILURE or SUCCESS value .. but not the discrete returned value |
| 14:45.06 | seanbright | mkoch: yes, it is. |
| 14:46.03 | mkoch | seanbright: do i need anything else than asterisk? |
| 14:46.21 | seanbright | mkoch: phones of some kind |
| 14:46.43 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 14:46.49 | iratik | Can anyone help real quick ... when i comment out line 4... the caller is directed to 000... if i uncomment it ... the caller gets a busy. .. what am i doing wrong ? (can't pull the output from asterisk -r ... too many other things going on in there) ... http://pastie.caboo.se/204722 |
| 14:46.51 | mkoch | only softphones, PCs with headsets |
| 14:46.52 | go|dfish | [TK]D-Fender, tzafrir -- thanks for your responses. Have a good day. |
| 14:46.58 | *** part/#asterisk go|dfish (i=goldfish@losers.yore.ma) |
| 14:47.14 | [TK]D-Fender | mkoch: that'll do |
| 14:47.42 | seanbright | lirakis_work: nope. just SYSTEMSTATUS. |
| 14:47.56 | [TK]D-Fender | iratik: exten => 999,n,Goto('from-internal',000,1) |
| 14:47.59 | [TK]D-Fender | iratik: no quote |
| 14:48.00 | mkoch | can asterisk automatically make audio backup files from the conversations? |
| 14:48.16 | lirakis_work | seanbright: damn .. how about some thing like set(myvar=${system(mycommand)} |
| 14:48.16 | seanbright | mkoch: yup. see the MixMonitor dialplan app for details. |
| 14:48.17 | lirakis_work | ?? |
| 14:48.34 | [TK]D-Fender | lirakis_work: you have quotes around your context name. |
| 14:48.39 | seanbright | lirakis_work: System is a dialplan app, it doesn't return a value |
| 14:48.53 | iratik | [TK]D-Fender: that part is actually working |
| 14:48.58 | iratik | i'm having problems with the curl part |
| 14:49.08 | lirakis_work | [TK]D-Fender: ..?? |
| 14:49.13 | [TK]D-Fender | iratik: you sure CURL even exists in your version? |
| 14:49.27 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 14:49.28 | seanbright | lirakis_work: he was talking to iratik, not you. |
| 14:49.43 | iratik | 1.4.19 ? |
| 14:49.47 | [TK]D-Fender | lirakis_work: nope |
| 14:49.48 | lirakis_work | seanbright: hrgm... i guess ill have to figure out another way of doing this .. im trying to avoid agi |
| 14:49.54 | [TK]D-Fender | bad aim, sorry |
| 14:49.59 | [TK]D-Fender | asdhlasjhda |
| 14:50.15 | seanbright | lirakis_work: that might be your only option. short of modifying asterisk. |
| 14:50.19 | *** join/#asterisk ManxPower (n=manxpowe@111.sub-70-221-101.myvzw.com) |
| 14:50.41 | [TK]D-Fender | iratik: there is no curl in 1.4 |
| 14:50.42 | mkoch | thanks and i tried the console dial command, it works fine on the same pc, where the asterisk runs. but from the other PCs how can i call from command line? is there any command-line softphone whitch i can use with MeetMe? |
| 14:50.44 | lirakis_work | seanbright: ehheh.. already been that route with openser .. looking for some thing without so many nastybits |
| 14:50.48 | golumn | I am simply courius. Does asterisk has an application for sending emails()? |
| 14:51.00 | seanbright | [TK]D-Fender: yes there is |
| 14:51.01 | iratik | [TK]D-Fender: so ... have to use system(curl ...) ? |
| 14:51.08 | [TK]D-Fender | mkoch: make, but you're better off with an X app |
| 14:51.31 | [TK]D-Fender | seanbright: aocomputing*CLI> show application curl |
| 14:51.32 | [TK]D-Fender | Your application(s) is (are) not registered |
| 14:52.08 | seanbright | core show function curl |
| 14:52.08 | [TK]D-Fender | seanbright: not as an application is doesn't ;) |
| 14:52.13 | *** join/#asterisk CVirus (n=GoD@196.205.192.192) |
| 14:52.14 | seanbright | [TK]D-Fender: too true. |
| 14:52.27 | [TK]D-Fender | seanbright: Details will kill you. |
| 14:52.32 | *** join/#asterisk amaache (n=maache76@41.221.16.91) |
| 14:52.41 | seanbright | [TK]D-Fender: bullets are more likely |
| 14:53.02 | Corydon76-dig | [TK]D-Fender: there is so a CURL in 1.4 |
| 14:53.12 | seanbright | we've just been over that |
| 14:53.16 | seanbright | ^^^^ |
| 14:53.21 | [TK]D-Fender | Corydon76-dig: indeed. |
| 14:53.32 | [TK]D-Fender | [TK]D-Fender>seanbright: not as an application is doesn't ;) |
| 14:53.43 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 14:53.46 | [TK]D-Fender | guess the keyword in there... |
| 14:53.59 | mkoch | [TK]D-Fender: i need a command line one, i have to use an own GUI with video stream, simulations and so, and a self GUI control panel whitch controls everything, including the conferences |
| 14:54.08 | seanbright | <[TK]D-Fender> iratik: there is no curl in 1.4 |
| 14:54.10 | *** join/#asterisk svenna_ (n=svenna@p548D1685.dip0.t-ipconnect.de) |
| 14:54.17 | seanbright | find the ambiguity in there... |
| 14:54.29 | [TK]D-Fender | mkoch: go check out the WIKI for one then. I have heard of one, but don't recall the name |
| 14:54.31 | [TK]D-Fender | ~wikis |
| 14:54.32 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 14:54.46 | [TK]D-Fender | seanbright: yes, clarified a line or so later. |
| 14:54.55 | seanbright | [TK]D-Fender: indeed. |
| 14:55.00 | seanbright | [TK]D-Fender: don't let it happen again. |
| 14:55.06 | ManxPower | how about "core show function CURL" |
| 14:55.06 | seanbright | :P |
| 14:55.11 | seanbright | ... |
| 14:55.12 | seanbright | sigh |
| 14:55.15 | hsv-al | Asterisk Predictable HTTP Manager Session ID Security Bypass Vulnerability |
| 14:55.24 | hsv-al | AN 1.0.2 not affected by this? |
| 14:55.34 | russellb | it probably is |
| 14:55.47 | russellb | the advisory should tell you. |
| 14:55.51 | ManxPower | seanbright: functions are CASE SENSITIVE |
| 14:56.02 | ManxPower | curl is the the function, CURL is. |
| 14:56.12 | ManxPower | ..er.. curl is NOT the function, CURL is. |
| 14:56.17 | mkoch | [TK]D-Fender: thanks, but the truth is that i tried to google some command-line softphone, but i can't find any... |
| 14:56.19 | [TK]D-Fender | iratik: And if you actually looked at your CLI output you'd ahve seen the application was not valid. Go read the application & function listings for a change. |
| 14:56.29 | hsv-al | found it, rb, some alerts say 1.0.2 is, some say it isn't |
| 14:56.30 | [TK]D-Fender | mkoch: There is a list on the WIKI |
| 14:56.31 | seanbright | ManxPower: yes, you're right. way to add zero value to this discussion. |
| 14:56.32 | hsv-al | is there a final say? |
| 14:56.34 | seanbright | was wrong |
| 14:56.41 | iratik | [TK]D-Fender: like i said.. could read the CLI output... too much activity |
| 14:56.53 | lirakis_work | any one have any other ideas on how to get data from an external system call without using agi? .. it would be so simple if i could just get the return data some how ... :\ i need no other agi functionality |
| 14:56.58 | Corydon76-dig | iratik: core set verbose off |
| 14:57.10 | *** join/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 14:57.10 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 14:57.18 | [TK]D-Fender | lirakis_work: "external system call.... could you be more vague? |
| 14:57.20 | iratik | thats what i was looking for |
| 14:57.51 | Corydon76-dig | iratik: and: core set verbose 31337 |
| 14:57.52 | [TK]D-Fender | iratik: uhh... yeah sure. |
| 14:58.38 | [TK]D-Fender | iratik: text search for curl in your output.... not Raw-Cat Science |
| 14:58.38 | hsv-al | 1.0.2 isnt affected. |
| 14:58.38 | hsv-al | this guy has no life that found it |
| 14:58.38 | iratik | What do you mean |
| 14:58.39 | hsv-al | he gets jollies by bypassing mpls boundaries |
| 14:58.39 | iratik | i wish there was something i could do cat somefile | grep curkl |
| 14:58.42 | iratik | err.. grep curl |
| 14:58.47 | iratik | but there is no somefile to do that to |
| 14:58.55 | ManxPower | iratik: AGI is the best and perhaps only workable method to do what you want |
| 14:58.56 | iratik | all i can do is go into asterisk -r and watch |
| 14:58.59 | [TK]D-Fender | iratik: You're wondering why it bombed with that line not commented out, you should have looked for it being called and see like 2 lines below what the error wasw |
| 14:58.59 | seanbright | iratik: /var/log/asterisk/full? |
| 14:59.18 | iratik | wow! |
| 14:59.28 | *** part/#asterisk ukdolphin (n=ukdolphi@host86-175-56-89.wlms-broadband.com) |
| 14:59.29 | iratik | there is a file i can do that too.. .that would have helped so much all these months |
| 15:00.15 | iratik | i actually wrote an expect script to watch a screen instance that was running asterisk -r and take snapshots at certain intervals to try to get the same effect as an output log file |
| 15:00.15 | ManxPower | iratik: if you paid more attention to the channel you would have learned about it months ago |
| 15:00.48 | iratik | i gave up on this channel months ago, only come here when i'm really in a rut. ... started reading the asterisk bible... haven't been in here nearly as much |
| 15:00.55 | mkoch | [TK]D-Fender: thanks, i found the list. is it a SIP softphone what i need? (i want to use it with meetme) |
| 15:01.12 | ManxPower | *nod* So you only come here to get help, not provide help or learn. Pretty typical. |
| 15:01.17 | [TK]D-Fender | mkoch: Thats a protocol * supports. Should be fine |
| 15:01.24 | iratik | everytime i came here.. people would be like "duhh... why didn't you know that, its pretty obvious to me" .... |
| 15:01.31 | seanbright | iratik: if you make that line -> exten => 999,n,Set(FOO=${CURL(http://192.168.1.110/MissedCall.php?cid=${REALCALLERIDNUM})}) |
| 15:01.35 | seanbright | iratik: that should work |
| 15:01.40 | iratik | I already got it working |
| 15:01.45 | seanbright | iratik: ah, good. |
| 15:01.46 | ManxPower | iratik: that did not tell you something like "read the book, the mailing list, or the wiki"? |
| 15:02.09 | *** join/#asterisk Nasra (n=maxshipp@190.166.71.39) |
| 15:02.21 | ManxPower | I'm VERY surprised The Book makes no mention of the Asterisk log files. |
| 15:02.29 | iratik | exten => 999,n,System(curl http://192.168.1.110/MissedCall.php?cid=${REALCALLERIDNUM}) |
| 15:02.37 | seanbright | iratik: that will work too |
| 15:02.45 | iratik | ManxPower: it probably does ... but hey .. its a big book |
| 15:02.56 | seanbright | iratik: you can configure what goes into the logs with /etc/asterisk/logger.conf |
| 15:03.04 | mkoch | now i'm leaving, thanks a lot! |
| 15:03.08 | seanbright | iratik: and create new ones, etc. |
| 15:03.08 | mkoch | bye |
| 15:03.10 | lirakis_work | [TK]D-Fender: a shell script .. can i "call" a shell script in some manner (system()) or otherwise, that will allow me to get a return variable .. or some how access data that is generated in that shell script |
| 15:03.16 | ManxPower | iratik: And yet, you did not read the single best source of information about Asterisk. No wonder people sad mean things to you. |
| 15:03.27 | lirakis_work | [TK]D-Fender: without using agi |
| 15:03.29 | jsmith | ManxPower: Yeah, it's a subject we just never got around to covering |
| 15:03.33 | MikeJ | heh |
| 15:03.38 | iratik | i did look there.. just didn't know where to look |
| 15:03.46 | ManxPower | jsmith: there's a a lot of that in the book. |
| 15:03.47 | MikeJ | hands jsmith a chapter on logging |
| 15:03.47 | iratik | people are under the impression that i didn't try to find the answer on my own |
| 15:03.54 | MikeJ | I think you dropped that |
| 15:03.55 | MikeJ | :P |
| 15:04.04 | Corydon76-dig | jsmith: when are we starting on the 3rd edition? |
| 15:04.20 | jsmith | ManxPower: We accept patches ;-) But yes, seriously, there's a lot of material that needs to be covered, but hasn't been yet |
| 15:04.42 | Corydon76-dig | jsmith: and will we surpass the Bat book this time? ;-) |
| 15:04.43 | jsmith | Corydon76-dig: Technically, several months ago. But nothing major has been done yet fora 3rd edition |
| 15:04.50 | [TK]D-Fender | lirakis_work: And whats wrong with AGI? Just about every hack around for it will be extremely limited and take more work to set up. |
| 15:04.58 | jsmith | Corydon76-dig: Now you're asking for "Asterisk: The Definitive Guide", right? |
| 15:05.24 | *** join/#asterisk railsmunky (n=nick@82-70-72-101.dsl.in-addr.zen.co.uk) |
| 15:05.57 | ManxPower | iratik: I think the problem is you just suck royally at using search engines. |
| 15:05.59 | lirakis_work | [TK]D-Fender: nothing is wrong with it.. the system is already running with a shell script .. (call to ticket) and i want to add a part that reads back ticket numbers that are generated.. so i wanted to avoid having to redo the system as an agi script, if at all possible |
| 15:06.11 | lirakis_work | i guess its agi though |
| 15:06.19 | seanbright | ManxPower: ok, he gets it. what say we move on? |
| 15:06.23 | drmessano | Wiw |
| 15:06.29 | drmessano | Wow too |
| 15:06.37 | ManxPower | lirakis_work: you can EASILY convert your app to AGI with very little coding. |
| 15:06.41 | [TK]D-Fender | \o/ |
| 15:06.45 | ManxPower | seanbright: he does NOT get it. |
| 15:07.20 | *** join/#asterisk ctooley (n=ctooley@209.33.106.165) |
| 15:07.27 | seanbright | ManxPower: and you feel compelled to chastise him in a public channel for the last 10 minutes because...? |
| 15:07.29 | lirakis_work | ManxPower: .. yeah .. thats what im starting on .. i just wanted to avoid it if possible .. and i thought that it would be reasonable to expect system() to return the actual return value of the call |
| 15:07.40 | seanbright | ManxPower: his question has been asked and answered, move on. |
| 15:08.01 | hsv-al | d-fender |
| 15:08.09 | ManxPower | seanbright: because if he can't find something as basic as the asterisk log files after searching for months something is WRONG. |
| 15:08.11 | hsv-al | is there a generic extension mechanism with iax2? |
| 15:08.29 | ManxPower | Either he is a moron or the information is very hard to find. I'll assume he's not a moron for now. |
| 15:08.54 | pigpen | Speaking of moron's, I am here. |
| 15:08.57 | pigpen | :) |
| 15:09.07 | ManxPower | hsv-al: nothing special for IAX2, just the standard pattern match stuff |
| 15:09.38 | hsv-al | so how are new features worked in? |
| 15:09.56 | ManxPower | hsv-al: you mean PROTCOL stuff, not EXTENSIONS stuff. |
| 15:10.12 | pigpen | Quick Question: Every month or so, I have had my Queue stop ringing the queue members. It just puts the call into eternal queue. Any words of wisdom? |
| 15:10.18 | ManxPower | hsv-al: you would have to ask on #asterisk-dev, but I don't know of any way to extend IAX2 except perhaps via IEs |
| 15:10.30 | anthooooooooo | I use Asterisk 1.4.19. We have some problems: Sometimes When I call, We have a noise in "background". For example, I can hear the voice of my voicemail in the same time of my call. It seem that my call take a channel that is not close. Is it possible? |
| 15:11.17 | pigpen | anthooooooooo, is it possible you are getting feed back via your receiver? |
| 15:12.31 | ManxPower | Customer E-mail: HELP! I deleted and e-mail account a couple of days ago that needs to be restored! Me: Perhaps you should contact the person in charge of backups for your e-mail system. |
| 15:12.33 | banzaika | anybody knows when digium plans to put curl support back ? |
| 15:12.36 | iratik | ManxPower: see! "i suck royally" ... i mean... what the heck was i supposed to search for .. i searched for "asterisk curl" ... i landed on http://www.voip-info.org/wiki/index.php?page=Asterisk+func+curl ... that page doesn't mention anything about not being supported in any version of asterisk .... Okay.. so i tried it.. it didn't work... I looked at the CLI and there was too much output ... so i commented the line out.. then it work |
| 15:12.36 | iratik | ed... so i came here... where did i go wrong? |
| 15:12.47 | ManxPower | iratik: "asterisk log" |
| 15:12.52 | Corydon76-dig | banzaika: what are you talking about? |
| 15:13.11 | Corydon76-dig | banzaika: it's a dialplan function. It's there. |
| 15:13.20 | ManxPower | iratik: the Wiki is the last place to look for information because it contains so much wrong informatin |
| 15:13.37 | iratik | what does that have to do with curl .... ? yeah curl wasn't working .... but i guess the problem is that i had accepted for months now that asterisk -r was the only way to monitor asterisk output |
| 15:13.47 | banzaika | <Corydon76-dig> not in mine :( |
| 15:13.57 | ManxPower | banzaika: "core show function CURL" CASE SENSITIVE. Did you not read the upgrade.txt file that comes with Asterisk? |
| 15:14.11 | Corydon76-dig | banzaika: then you probably don't have the right libcurl packages installed |
| 15:14.16 | iratik | thanks for the help guys btw |
| 15:14.18 | banzaika | <Corydon76-dig> Asterisk autotag_for_sx00i (sx00i 1.1.0.1) built by doug @ aa50dev on a i686 running Linux on 2008-02-08 19:18:59 UTC |
| 15:14.23 | ManxPower | iratik: nothing I'm saying in any way relates to Curl |
| 15:14.35 | banzaika | i have an AA50 |
| 15:14.36 | Corydon76-dig | banzaika: the AA50 will never support curl |
| 15:14.46 | banzaika | bahhh |
| 15:14.50 | Corydon76-dig | There's just not enough memory |
| 15:14.50 | ManxPower | banzaika: AA50 is not really supported here |
| 15:15.18 | sysadmin-lb22 | hi just installed asterisk but not ilbc something I missed here ? |
| 15:15.21 | banzaika | i guess i have to go to digium for this |
| 15:15.26 | sysadmin-lb22 | I can see it in the show codecs but it wont work |
| 15:15.29 | Corydon76-dig | Yes, you do |
| 15:15.37 | ManxPower | sysadmin-lb22: "core show translations" |
| 15:15.40 | Corydon76-dig | The AA60, however, will support curl |
| 15:16.01 | Corydon76-dig | but that's a completely different box |
| 15:16.10 | banzaika | free upgrade ? |
| 15:16.11 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:16.12 | banzaika | lol |
| 15:16.19 | ManxPower | What do you mean my embeded closed system I bought does not support all the features of the open source Asterisk???????????? |
| 15:16.28 | sysadmin-lb22 | ManxPower, no such command |
| 15:16.38 | ManxPower | sysadmin-lb22: what version of Asterisk? |
| 15:16.54 | sysadmin-lb22 | ManxPower, 1.4 |
| 15:17.00 | ManxPower | sysadmin-lb22: it may be "core show translation" |
| 15:17.13 | sysadmin-lb22 | ManxPower, ok got the matrix |
| 15:17.23 | ManxPower | show codecs only shows the codec numbers, not what codecs asterisk can use |
| 15:17.24 | sysadmin-lb22 | ManxPower, ilbc is all - |
| 15:17.36 | ManxPower | sysadmin-lb22: then the libs were not found when you built Asterisk |
| 15:18.10 | sysadmin-lb22 | ManxPower, I did configure and make all |
| 15:18.15 | sysadmin-lb22 | ManxPower, what have I missed here ? |
| 15:18.41 | sysadmin-lb22 | ManxPower, 2008-04-02 - iLBC no longer included with Asterisk source (1.4.19 and 1.6); run script contrib/scripts/get_ilbc_source.sh to install (may have to run menuselect/menuselect and go to Codecs section to select iLBC) |
| 15:18.47 | ManxPower | sysadmin-lb22: maybe you missed installing the iLIBC libraries for your distro. |
| 15:19.34 | sysadmin-lb22 | ManxPower, you mean some devel files ? |
| 15:19.55 | ManxPower | sysadmin-lb22: You did not see this message when you did a "show codecs" "Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration." |
| 15:20.06 | ManxPower | sysadmin-lb22: YES! YES! |
| 15:20.21 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 15:20.24 | r0land | hello all |
| 15:20.32 | anthooooooooo | pigpen: No. For example, I call *43 (echo test) and I hangup my call , after I call Harry. When I speak with Harry, we hear the echo test |
| 15:20.58 | r0land | could some1 help me out with voicemail please! i've setup voicemail.conf though im having trouble letting asterisk to transfer the call to voicemail after lets say 8 seconds of the user not answering his sipphone |
| 15:21.38 | jsmith | r0land: Typically, it goes something like this: |
| 15:21.52 | jsmith | exten => 123,1,Dial(SIP/joe,8) ; dial SIP/joe for 8 seconds |
| 15:22.00 | r0land | ah ok |
| 15:22.02 | r0land | thanks :) |
| 15:22.13 | r0land | but when the 8 seconds are done |
| 15:22.15 | r0land | wht happens! |
| 15:22.19 | jsmith | exten => 123,n,Voicemail(123@blah) ; send the call to mailbox 123 in the [blah] context |
| 15:22.21 | r0land | how does it know tht it should go to the voicemail |
| 15:22.22 | r0land | ah |
| 15:22.24 | r0land | lol |
| 15:22.26 | r0land | sorry for tht |
| 15:22.34 | jsmith | It goes on to the next priority in the dialplan |
| 15:22.38 | r0land | yep got it |
| 15:22.40 | r0land | thank you |
| 15:22.47 | jsmith | If you want to know *why* it didn't work, look at the ${DIALSTATUS} variable |
| 15:22.49 | ManxPower | r0land: you might have seen an example if you look at extensions.conf.sample in the Asterisk source. |
| 15:23.10 | r0land | true i never thought of tht.. thanks ManxPower |
| 15:23.22 | r0land | thanks u guys :) ill give it a try |
| 15:23.40 | ManxPower | r0land: your single BEST source of docs is in the Asterisk source tree. |
| 15:23.45 | *** join/#asterisk ctooley (n=ctooley@209.33.106.165) |
| 15:24.42 | ManxPower | the configs and doc (maybe docs) directories |
| 15:24.44 | anthooooooooo | My prblem is a following: For example, I call *43 (echo test) and I hangup my call , after I call Harry. When I speak with Harry, we hear the echo test. I seem that the first channel is not close. Why? Do you have any ideas about it? |
| 15:25.20 | ManxPower | anthooooooooo: your problem is incredibly hard to diagnose and fix. I guess nobody here wants to spends several yours helping you. |
| 15:26.00 | ManxPower | MY guess is the problem will not go away until you replace the motherboard of the system with another make/model |
| 15:27.53 | *** join/#asterisk fatcop (n=223343@ppp121-44-119-201.lns10.syd6.internode.on.net) |
| 15:27.56 | r0land | jsmith, ManxPower is this about right? http://www.pastebin.ca/1032187 |
| 15:28.01 | anthooooooooo | ManxPower. When I call echo test (*43) I hear the call.After, I close the echo test. After if I call someone, during the conversation, I still hear the echo test .... |
| 15:28.14 | ManxPower | Would someone kick this guy? |
| 15:28.28 | ManxPower | r0land: did you look at extensions.conf.sample? |
| 15:28.34 | r0land | ManxPower dont hae it |
| 15:28.35 | MikeJ | kicks anthooooooooo |
| 15:28.39 | MikeJ | why? |
| 15:28.44 | ManxPower | r0land: then GET it |
| 15:28.54 | ManxPower | MikeJ: repeating the same question over and over |
| 15:29.27 | MikeJ | anthooooooooo: take a look at the sip trace... |
| 15:29.32 | r0land | ManxPower how can tht b done! if i may ask |
| 15:29.57 | ManxPower | r0land: download the Asterisk source again, unpack it, look at the documenation and sample config files. |
| 15:30.17 | MikeJ | my guess is the bye is not getting back to asterisk.. but the phone thinks its hung up... |
| 15:30.18 | ManxPower | You must have done this once if you have Asterisk installed. |
| 15:30.37 | anthooooooooo | MikeJ, Where can I put the sip trace? |
| 15:30.45 | MikeJ | on your screen... |
| 15:30.50 | drmessano | In the USB port |
| 15:30.50 | MikeJ | then look at it.. |
| 15:30.58 | MikeJ | along with the debug on the asterisk side.. |
| 15:31.01 | drmessano | or the game port |
| 15:31.06 | drmessano | ok, no |
| 15:31.14 | MikeJ | to see how asterisk is handling (or not getting) the bye |
| 15:31.22 | ManxPower | drmessano: there is no need to be an asshole and confuse an already VERY confused user. |
| 15:31.48 | drmessano | Well, I did say "ok, no" |
| 15:31.56 | hsv-al | 3 bottles of red bull + coffee + skoal dip |
| 15:31.58 | hsv-al | = wired hell |
| 15:32.16 | *** join/#asterisk s0lid (n=s0lid@125.60.135.68) |
| 15:32.25 | ManxPower | hands hsv-al some meth and says "Here, this should help with the jitters" |
| 15:32.53 | hsv-al | it masks hunger manxpower |
| 15:33.02 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 15:33.25 | Qwell | so then don't eat? sounds like one of those problems that work themselves out |
| 15:33.30 | hsv-al | heh |
| 15:33.31 | drmessano | lol |
| 15:34.13 | hsv-al | usually when im buzzed like this, im more friendly. A homeless g uy came up to me, and asked for some dip |
| 15:34.17 | hsv-al | one of the many homeless vets in town |
| 15:34.24 | hsv-al | and i actually listened to his story for 5 min, felt bad for him |
| 15:34.32 | [TK]D-Fender | hsv-al: Masks hunger? Take some weed too to equalize it out ;) |
| 15:35.17 | hsv-al | d-fender, drugs arent an option. :) |
| 15:36.00 | ManxPower | hsv-al: drugs are ALWAYS an option -- just maybe not a GOOD option |
| 15:36.41 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 15:36.58 | hsv-al | I havent dipped in about 5 years, so when I had some today it felt like I was high |
| 15:37.28 | *** join/#asterisk asdx (n=diego@adsl-129-35.click.com.py) |
| 15:38.27 | asdx | hi, does asterisk do forward of signaling, eg if asterisk receives BUSY, INVALID, etc, can it transfer that to another SIP clients |
| 15:38.44 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 15:40.07 | rob0 | You can use extensions based on ${DIALSTATUS} |
| 15:41.34 | *** join/#asterisk LuisTorres (n=chatzill@bl9-251-53.dsl.telepac.pt) |
| 15:41.48 | asdx | ok thanks |
| 15:42.58 | fatcop | hi. I am not getting any RTP sounds thru for my IVR. The CLI trace shows it playing the right file. I can listen to that file via same phone via other means. Its just IVR. |
| 15:43.07 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:43.37 | *** join/#asterisk SanityIO__ (n=SanityIO@77.242.106.224) |
| 15:43.49 | ManxPower | fatcop: stop asterisk, rmmod ztdummy, start Asterisk |
| 15:44.28 | fatcop | hmmmm .. k will try that |
| 15:45.26 | fatcop | "ERROR: Module ztdummy does not exist in /proc/modules/" |
| 15:46.09 | *** join/#asterisk SanityIO__ (n=SanityIO@77.242.106.224) |
| 15:46.29 | jsmith | fatcop: Do you happen to have an unconfigured T1 card in the box? |
| 15:46.32 | fatcop | there is no RTP data being sent for that IVR annoucement. Whereas I can see RTP packets in the debug for other stuff |
| 15:46.45 | fatcop | no just basic PC with SIP extensions |
| 15:47.11 | [TK]D-Fender | fatcop: I suggest you actually show us the CLI output of this call at high verbose |
| 15:47.13 | [TK]D-Fender | ~pb |
| 15:47.14 | 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 |
| 15:47.15 | [TK]D-Fender | ^^^^^^^^^^^ |
| 15:47.29 | *** join/#asterisk RoyK (n=roy@ip-67-200-241-92.dialup.nmt.net) |
| 15:47.32 | fatcop | <PROTECTED> |
| 15:48.01 | *** join/#asterisk SanityIO__ (n=SanityIO@77.242.106.224) |
| 15:48.30 | fatcop | at that point it waits the 8 secs then goes onto the ringing of the reception extension |
| 15:48.51 | [TK]D-Fender | fatcop: and you hear nothing? |
| 15:49.20 | fatcop | nups. verified by no "To:" rtp packets |
| 15:49.32 | [TK]D-Fender | fatcop: what kind of file is it? |
| 15:49.37 | fatcop | wav |
| 15:49.55 | [TK]D-Fender | fatcop: go verify its characteristics |
| 15:50.19 | fatcop | i can hear it back thru other menus (using same phone) .. just no thru IVR |
| 15:50.29 | [TK]D-Fender | fatcop: Also background a known good asterisk-provided sound file as well |
| 15:50.37 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:50.39 | [TK]D-Fender | fatcop: show us that as well then |
| 15:50.50 | fatcop | k |
| 15:51.53 | fatcop | err actually, i didn't listen to it back since I recorded it. Now I listen to the wav thru the PC .. so its not the same. |
| 15:52.14 | jameswf-home | heh http://theybannedme.com/ |
| 15:52.29 | Qwell | this should be good |
| 15:52.35 | fatcop | k will try and drop in a wav that shld work then I guess |
| 15:52.41 | [TK]D-Fender | fatcop: Congratulations. |
| 15:52.47 | lirakis_work | yargh!! i cant get the stupid freakin variable i set from before i spawned the agi .. grumble .. this is why i didnt want to do agi... some little thing is going to take 2 hours to figure out |
| 15:53.13 | ManxPower | lirakis_work: What language? |
| 15:53.21 | jameswf-home | lirakis_work: did you try wait() |
| 15:53.24 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:53.37 | lirakis_work | ManxPower: php .. using phpagi |
| 15:53.41 | lirakis_work | jameswf-home: i did |
| 15:53.46 | ManxPower | lirakis_work: OK, That should work. |
| 15:53.52 | fatcop | well it did play back on this phone when i recorded it.. but after it got moved to diff folder that wasn't possible anymore .. but will try another file |
| 15:54.06 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 15:54.14 | jameswf-home | sounds like permissions |
| 15:54.20 | ManxPower | lirakis_work: show us the one like where you are trying to get the variable value |
| 15:54.26 | ManxPower | line, not like |
| 15:54.40 | lirakis_work | pastbinning |
| 15:54.54 | ManxPower | you don't really need to pastebin for only ONE line |
| 15:55.05 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:57.17 | ManxPower | waits |
| 15:57.18 | lirakis_work | ManxPower jameswf-home: http://rafb.net/p/Eyw5NE18.html |
| 15:58.03 | ManxPower | lirakis_work: you are so great at following directions |
| 15:58.41 | lirakis_work | ManxPower: what? |
| 15:59.00 | ManxPower | lirakis_work: I said paste the ONE LINE. |
| 15:59.25 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:59.42 | *** join/#asterisk Vec (n=Vec@host-87-74-7-57.dslgb.com) |
| 15:59.52 | lirakis_work | ManxPower: i was already working on the paste before you said that (shrug) .. but the paste is verbose (though not overly) .. and more useful than a single line .. i would assime |
| 15:59.54 | lirakis_work | *assume |
| 16:00.07 | ManxPower | lirakis_work: are you parsing or reading stdin to clear out all the stuff Asterisk sends before trying to get variables (as talked about on the voip-info wiki page) ? |
| 16:00.47 | lirakis_work | ManxPower: i was under the impression that the phpagi class took care of all of that and stored it into $request array |
| 16:00.47 | [TK]D-Fender | lirakis_work: What part isn't working? |
| 16:00.48 | ManxPower | lirakis_work: Yes, but I'm helping you debug your script. I get paid for that stuff. I'm helping you with the specific issue of not being able to get a variable |
| 16:01.03 | Vec | Please help, can't figure out the diffirence between Dial(SIP/trunk/5555) and Dial(SIP/5555@trunk) ? |
| 16:01.16 | ManxPower | [TK]D-Fender: I think he is not reading STDIN to get all the stuff Asterisk sends at AGI startup, so he can't do any other AGI stuff. |
| 16:01.25 | lirakis_work | [TK]D-Fender: .. it always returns 1 for the CUSTID .. which is clearly set as some thing else |
| 16:01.50 | [TK]D-Fender | lirakis_work: thats the RESULT CODE showing it succeeded, not the VALU. |
| 16:02.09 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:02.11 | ManxPower | Vec: one dials the extension "trunk" using the sip peer [5555], the other dials the extension 555 using the sip peer [trunk] |
| 16:02.51 | ManxPower | Vec: use the 2nd format |
| 16:03.20 | lirakis_work | [TK]D-Fender: maybe im looking at the wrong array field |
| 16:03.27 | fatcop | no luck really. Tried a few files. |
| 16:03.27 | Vec | ManxPower : thanks but why ? |
| 16:03.28 | [TK]D-Fender | lirakis_work: really!? |
| 16:03.59 | hsv-al | where do i make the relationship between funcs I created and sql commands they need to execute? |
| 16:04.19 | hsv-al | with funk_odbc dialplans |
| 16:04.20 | ManxPower | Vec: because that is the format almost every piece of Asterisk documentation uses. |
| 16:04.35 | hsv-al | is there a configuration file? |
| 16:04.44 | russellb | func_odbc.conf, believe it or not! |
| 16:04.47 | lirakis_work | <PROTECTED> |
| 16:04.49 | Vec | ManxPower : ok so 5555@trunk. |
| 16:04.50 | Nasra | any1 know of a good website for Asterisk in spanish....it's so hard to dind these days.... |
| 16:04.55 | Nasra | thanks |
| 16:04.56 | lirakis_work | <PROTECTED> |
| 16:04.57 | lirakis_work | ;) |
| 16:05.00 | lirakis_work | thanks guys |
| 16:05.16 | ManxPower | lirakis_work: you might want to spend a few mins reading the phpagi docs |
| 16:05.16 | hsv-al | rolls eyes |
| 16:05.19 | [TK]D-Fender | lirakis_work: http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI.html#methodget_variable |
| 16:05.32 | lirakis_work | ManxPower: i had them open .. i just read too fast |
| 16:05.39 | ManxPower | [TK]D-Fender is an awesome google proxy |
| 16:06.15 | [TK]D-Fender | I somehow think we are a large part of the problem. |
| 16:06.56 | ManxPower | [TK]D-Fender: we could leave and see what happens over a couple of seeks. |
| 16:06.59 | ManxPower | weeks too |
| 16:07.16 | [TK]D-Fender | We enable people to do everything half-assed and not apy attention to anything they do. Thus they get to remain lazy douchebags while we like doormats look it all up for them. |
| 16:07.19 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 16:07.23 | hsv-al | russel thx, didnt even know it existed |
| 16:07.24 | hsv-al | heh |
| 16:07.46 | ManxPower | [TK]D-Fender: who is this "we". I hope I don't usually do stuff for people. |
| 16:08.08 | [TK]D-Fender | ManxPower: Yeah you still do. Not as much as others, but you still do... |
| 16:08.16 | ManxPower | [TK]D-Fender: I shall work on that. |
| 16:08.35 | [TK]D-Fender | ManxPower: You are still a while away from full BOFH glory. |
| 16:08.53 | ManxPower | [TK]D-Fender: BOFHhood is a journey, not a destination 8-)_ |
| 16:09.30 | *** join/#asterisk grEvenX (n=even@pc107-102.ktv.no) |
| 16:10.21 | [TK]D-Fender | ManxPower: so STOP RUNNING :p |
| 16:10.30 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 16:11.01 | deeperror | anyone use a te412p or similar? |
| 16:11.11 | ManxPower | ~ask |
| 16:11.11 | jbot | ask is probably 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. |
| 16:12.11 | deeperror | would like to compare the te412 with the rhino r4t1 |
| 16:12.20 | fatcop | [TK]D-Fender: no luck with the IVR |
| 16:12.33 | [TK]D-Fender | fatcop: show me a situation that works, and one that doesn't |
| 16:12.46 | ManxPower | deeperror: The Rhino is not very popular so not many people will be able to help you with it. |
| 16:12.54 | ManxPower | why not consider Sangoma? |
| 16:13.09 | [TK]D-Fender | fatcop: and confirm its charateristics. |
| 16:13.37 | deeperror | i've got 4 rhino channelbanks at this time just need a t1 card that will work and maintain a stable system that has drivers that work etc |
| 16:13.42 | ManxPower | I didn't say "use Sangoma", just consider sangoma |
| 16:13.59 | Kobaz | who here is a nortel genius? :) |
| 16:14.13 | *** join/#asterisk railsmunky (n=nick@collaboration.capuk.org) |
| 16:14.14 | deeperror | ManxPower: never even looked into them |
| 16:14.14 | [TK]D-Fender | Kobaz: Depends on your basis for qualification. |
| 16:14.18 | fatcop | wot do you mean |
| 16:14.25 | fatcop | characterists ? |
| 16:14.31 | [TK]D-Fender | Kobaz: I could be an expert on MICS, and know NOTHING about Option11, etc |
| 16:14.36 | fatcop | the wav file ? |
| 16:14.38 | Kobaz | ah true |
| 16:14.41 | [TK]D-Fender | fatcop: precise file format details |
| 16:14.52 | deeperror | Like the A104? |
| 16:14.55 | Kobaz | we have an NT1R20 line card, we're trying to configure some ports on it as fxs |
| 16:14.58 | [TK]D-Fender | deeperror: indeed |
| 16:14.59 | fatcop | i know its 8000Mhz PCM 16bit |
| 16:15.00 | ManxPower | deeperror: They are good cards, good support. Personally, I think Sangoma was the reason Digium made such significant improvements to their cards over the past couple of years. |
| 16:15.13 | drmessano | 800HZ |
| 16:15.14 | ManxPower | deeperror: we use all A102ds |
| 16:15.15 | drmessano | 8000HZ |
| 16:15.19 | fatcop | y |
| 16:15.23 | drmessano | Not 8000MHZ |
| 16:15.29 | deeperror | standard zaptel drivers? |
| 16:15.35 | Kobaz | all our nortel contact people are missing |
| 16:15.40 | [TK]D-Fender | fatcop: go show me one where it works, one where it doesn't |
| 16:15.54 | ManxPower | deeperror: the sangoma drivers present a zaptel interface |
| 16:15.58 | [TK]D-Fender | fatcop: And play 2 files back-to-back, 1 * provided, one yours |
| 16:16.04 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:16.14 | deeperror | ok reading the specs on them now |
| 16:16.21 | ManxPower | deeperror: the Sangoma is slightly more complex to build the drivers for. |
| 16:16.22 | fatcop | k will try |
| 16:16.36 | mort_gib | ManxPower: not that bad |
| 16:16.39 | deeperror | i'm just having a horrible time with the rhino r4t1...i ran 3x r1t1's in the same maching with no issues but the r4 doesn't work |
| 16:17.00 | Qwell | deeperror: the other option, if you like supporting the company behind Asterisk, is Digium, of course |
| 16:17.03 | ManxPower | deeperror: Digium and Sangoma are both very popular for use with Asterisk -- community support should be good regardless of which one you pick |
| 16:17.19 | deeperror | well i've been talking with digium on their card |
| 16:17.33 | deeperror | the reason i went with rhino was cost....but we all know the old adage |
| 16:17.41 | Qwell | ~ygwypf |
| 16:17.41 | jbot | ygwypf is, like, You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 16:17.43 | Kobaz | rhino... blah |
| 16:17.52 | Qwell | jbot: jbot++ |
| 16:17.58 | Kobaz | deeperror: yes they are cheaper, their drivers are terrible |
| 16:18.23 | ManxPower | mort_gib: more complicated that "make install" like with the Digium drivers |
| 16:18.59 | deeperror | well my idea was to setup this analog stuff and move into 100% sip stations....but since IT and business development move opposite each other were now a year in and they are wanting to just stick with analog cause it's been working so well...but they don't realize scailing sucks |
| 16:19.01 | mort_gib | ManxPower: But the DO make all changes to extensions.conf required, including zaptel.conf |
| 16:19.15 | ManxPower | We switched to ALL Sangoma back when Digium used the original Zaptel design. They were pretty horrible in some specific situations. |
| 16:19.17 | fatcop | [TK]D-Fender: thx. Bit burn out with this stuff 2nite. Will try and do as you said. Not sure how with my setup.. but will find a way. cheers. |
| 16:19.37 | [TK]D-Fender | fatcop: You're welcome |
| 16:19.48 | Qwell | ManxPower: You should give Digium hardware another try. Just sayin'.. |
| 16:19.53 | ManxPower | If we started using Asterisk today -- with all the massive improvements to Digium cards in the past few years - I doubt we would have switched. |
| 16:20.15 | deeperror | with that said i'm going to go with someone that will let me test it out first haha |
| 16:20.19 | ManxPower | Qwell: Does Digium want to buy the T400P we have and the 8 or so A102Ds? |
| 16:20.33 | Qwell | ManxPower: I don't mean replace existing installs |
| 16:20.34 | hsv-al | heh, after looking at all of this |
| 16:20.39 | ManxPower | We moved to Sangoma -- no reason to change now. |
| 16:20.39 | hsv-al | why would anyone not use dialplans other then |
| 16:20.44 | hsv-al | the func_odbc usage |
| 16:20.48 | Qwell | but, yes. if you buy new hardware, and it doesn't work out - we will absolutely take it back |
| 16:20.50 | ManxPower | Qwell: we like to have a minimum of different hardware |
| 16:20.51 | [TK]D-Fender | deeperror: Better off with SIP gateways anyways and forgetting channel-backs, and T1 cards |
| 16:21.05 | deeperror | another question how about dsp and multi core processors? any issues with them and ec? |
| 16:21.23 | Qwell | deeperror: nah |
| 16:21.25 | deeperror | the reason we have to use analog is due to the custom nature of our CRM |
| 16:21.27 | *** join/#asterisk gharz (n=garry@dxb-as72281.alshamil.net.ae) |
| 16:21.31 | *** join/#asterisk mirrorcolor (n=iunixan@196.218.222.116) |
| 16:21.35 | ManxPower | We are committed to Cisco routers and Switches, Polycom Phones, PRIs, and Sangoma |
| 16:21.48 | Qwell | ManxPower: so, you're saying...you're stuck with them? :) |
| 16:21.49 | hsv-al | manxpower, what do your pri's hook into? 5 series? |
| 16:21.56 | deeperror | and to change the CRM to a SIP based solution will take more time since no one is really behind the move... |
| 16:22.01 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:22.05 | ManxPower | Qwell: unless we have a VERY good reason to change. |
| 16:22.25 | ManxPower | hsv-al: We hook the PRIs into Sangoma 102Ds |
| 16:22.26 | Qwell | ManxPower: You yourself just moments ago gave a reason |
| 16:22.34 | ManxPower | Qwell: I did? |
| 16:22.36 | Qwell | <ManxPower> If we started using Asterisk today -- with all the massive improvements to Digium cards in the past few years - I doubt we would have switched. |
| 16:23.09 | ManxPower | Qwell: I don't see that as a reason to switch. |
| 16:23.36 | ManxPower | twice the number of spare cards needed, different cards in different servers, have to remember which is which. |
| 16:24.09 | hsv-al | manxpower, so your using a cisco gateway, and can see the caller id? |
| 16:24.21 | ManxPower | hsv-al: we do no voip on cisco stuff |
| 16:24.27 | hsv-al | ahh |
| 16:24.35 | ManxPower | we use Cisco for what its fairly good at -- moving packets |
| 16:24.35 | hsv-al | was gonna say voice service voip |
| 16:24.40 | hsv-al | signaling forward unconditional |
| 16:24.40 | hsv-al | :) |
| 16:24.54 | hsv-al | heh |
| 16:25.01 | deeperror | on phone with sangoma |
| 16:25.32 | ManxPower | Qwell: It took me having to spenf $1,200 of my own money to solve a customer problem before I switched away from Digium -- it will take something similar to move away from them. |
| 16:25.41 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 16:26.08 | [TK]D-Fender | deeperror: If you don't have the CB's yet, just forget them and buy SIP gateways |
| 16:26.17 | deeperror | have all the CB's |
| 16:26.19 | deeperror | in production |
| 16:26.20 | ManxPower | I disagree with [TK]D-Fender |
| 16:26.33 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:26.36 | *** join/#asterisk smash- (i=smash@prettysquishy.com) |
| 16:27.16 | [TK]D-Fender | Gor general voice use CB's offer little over SIP gateways, and have many more Gotchas |
| 16:27.29 | [TK]D-Fender | For* |
| 16:28.02 | ManxPower | [TK]D-Fender: And I say the same thing about SIP gateways -- many more gotchas and they offer little over a channel bank. |
| 16:28.16 | Kobaz | do de do |
| 16:28.30 | Kobaz | so are there any nortel people around? |
| 16:28.36 | ManxPower | Kobaz: no. |
| 16:28.39 | Kobaz | heh |
| 16:28.43 | Kobaz | figured |
| 16:28.53 | ManxPower | Maybe I'm just spoiled by Adtran channel banks, but they Just Work. |
| 16:29.14 | smash- | Hello, how do i set my callid in sip.conf |
| 16:29.35 | ManxPower | smash-: you should look at sip.conf.sample to see a good, valid, callerid setting |
| 16:29.36 | deeperror | well we have software that is old school |
| 16:29.51 | deeperror | can't upgrade it because it's 100% hacked and customized to the operation/business model |
| 16:30.12 | deeperror | so we use channel banks and modems dial out |
| 16:30.38 | deeperror | pretty lame (i'm aware) so working to force them into a sip crm like aheeva or something |
| 16:30.50 | Kobaz | sip crm? |
| 16:30.51 | [TK]D-Fender | smash-: Pardon? |
| 16:31.25 | deeperror | something that will work with sip and not just com port and tapi devices |
| 16:32.02 | Kobaz | ah |
| 16:34.30 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 16:34.37 | kannan | hello all |
| 16:37.06 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 16:40.35 | *** join/#asterisk rcahilig (i=ca4e4bf5@gateway/web/ajax/mibbit.com/x-96b4eb6d32d6b0d0) |
| 16:41.03 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 16:43.09 | *** join/#asterisk railsmunky (n=nick@collaboration.capuk.org) |
| 16:43.37 | railsmunky | is there a way to set a dummy value or something like that in an dialplan? |
| 16:43.48 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:43.49 | railsmunky | eg exten => s,5,Background(${IFTIME(9:00-17:00|mon-fri|*|*?out_of_hours_message:)}) |
| 16:44.26 | Nobbie | where does * keep track of callwaiting for each extension ? |
| 16:44.34 | *** join/#asterisk s0lid (n=s0lid@125.60.135.68) |
| 16:44.41 | [TK]D-Fender | railsmunky: silence/1 |
| 16:44.52 | [TK]D-Fender | Nobbie: what call-waiting? |
| 16:44.57 | [TK]D-Fender | Nobbie: what device? |
| 16:45.06 | Nobbie | SIP device |
| 16:45.22 | [TK]D-Fender | Nobbie: they don't |
| 16:45.34 | [TK]D-Fender | Nobbie: Call-waiting is a user percieved feature. |
| 16:45.47 | Nobbie | voip-info.org lists Call waiting and DND as Asterisk features |
| 16:45.54 | railsmunky | [TK]D-Fender: Brilliant thanks! I'm learning :) |
| 16:45.56 | smash- | [TK]D-Fender : callerid user name <extnumber> |
| 16:46.21 | smash- | just sets an internal extension aswell as setting outgoing caller id number? |
| 16:46.40 | [TK]D-Fender | smash-: callerid="name" <12345> |
| 16:47.14 | railsmunky | just got to get the flippin callerid working now on outbound calls |
| 16:47.17 | smash- | [TK]D-Fender: yes exactly. forgot to put the " and <> |
| 16:48.21 | [TK]D-Fender | railsmunky: calls go out via what? |
| 16:48.35 | railsmunky | [TK]D-Fender: UK BT PRI |
| 16:49.25 | [TK]D-Fender | railsmunky: Ok, well you should be able to set it direct int he dialplan with the CALLERID function. |
| 16:49.57 | railsmunky | [TK]D-Fender:i've tried exten => _907XXXXXXXXX,1,Set(CALLERID(all)=blah}) for example with no joy |
| 16:51.28 | [TK]D-Fender | railsmunky: just set the #, and you have a trailing bad } there |
| 16:52.12 | Nobbie | [TK]D-Fender: what you mean is that CW is setup on the SIP endpoint ? |
| 16:53.19 | [TK]D-Fender | Nobno, what I mean is that call-waiting is a concept that exists only in your head. The phone "beeps". The phone can also not run 2 completely independent calls. Each call to * is 100% independent from the other. the phone simply places 1 on hold to take the other. |
| 16:54.07 | Nobbie | [TK]D-Fender: but the feature has to be enabled somewhere for the PBX to know not to send you a 2nd call if you're busy with another? |
| 16:54.20 | [TK]D-Fender | Nobbie: No it doesn't |
| 16:54.50 | *** part/#asterisk UnixDog (n=UnixDog@254.69.118.70.cfl.res.rr.com) |
| 16:54.52 | Nobbie | huh ? |
| 16:54.53 | [TK]D-Fender | Nobbie: Deciding whether or not to pass a call to a device that may already be on a call is up to YOU to determine in yuor DIALPLAN. |
| 16:55.29 | railsmunky | [TK]D-Fender: great - it's the simplest things! Also need to strip the leading 0 for BT to be happy days with it. Thanks! |
| 16:55.31 | Nobbie | ok, so other then using astdb, which is something i'm having major problems with, how would i store a per user CW value to make that decision in my dialplan ? |
| 16:56.05 | [TK]D-Fender | Nobbie: you can use anything you can retreive via the dialplan. |
| 16:56.27 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
| 16:56.38 | [TK]D-Fender | Nobbie: You could also try to limit *'s ability to send multiple channels per-se via "call-limit", etc, but thats iify. |
| 16:56.53 | [TK]D-Fender | Nobbie: And I'm not sure if thats 1.4+ only or not. |
| 16:57.10 | [TK]D-Fender | Nobbie: You could try those too, but that'd be fixed in your peer setup. |
| 16:57.23 | [TK]D-Fender | Nobbie: go read the parameter list on the WIKI for sip.conf |
| 16:57.32 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:57.59 | Nobbie | what i really need, is to find out why getting info from astdb takes up to 10 seconds |
| 16:58.37 | ManxPower | Most phones let you disable call waiting |
| 16:58.44 | Nobbie | and astdb is only 400KB, on a high end HP DL380 with 3GB RAM |
| 16:59.07 | Nobbie | even with astdb stored in /dev/shm it takes up to 10 seconds |
| 16:59.27 | Qwell | Nobbie: how are you measuring that? |
| 16:59.32 | ManxPower | Nobbie:then you have something WRONG. It's not supposed to take that long. |
| 16:59.35 | Qwell | that seems, as you can imagine, extremely excessive |
| 16:59.54 | Nobbie | QWell: getting system microtime before and after "database get" is called via AGI |
| 17:00.11 | Nobbie | it's intermittently a problem. |
| 17:00.18 | ManxPower | BTW, you want AstDB on non-volatile storage, as Asterisk stores lots of information in there. |
| 17:00.36 | ManxPower | SIP registration info, for example. |
| 17:00.37 | Nobbie | yes, i copy from SHM to disk every few minutes, and at * startup, copy from disk to shm |
| 17:00.45 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 17:00.49 | ManxPower | Nobbie: that is a VERY bad idea |
| 17:01.02 | Nobbie | it's a temporary attempt at fixing a problem, will undo it when i find a proper solution |
| 17:01.20 | Nobbie | it seems to alleviate the problem slightly |
| 17:01.27 | ManxPower | Nobbie: I would expect database corruption doing it your way |
| 17:01.46 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-82-206.vif.net) |
| 17:01.56 | Nobbie | using db1_dump185 to dump the DB |
| 17:02.11 | ManxPower | Ah, so you are NOT just copying it. |
| 17:02.30 | Nobbie | database corruption is the least of my worries right now. i have 350 angry users shouting becuase it takes more then 10 seconds to dial in internal extension |
| 17:02.35 | ManxPower | I assume db1_dump185 is safe for use on a live active database? |
| 17:02.47 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 17:03.13 | ManxPower | Nobbie: the only time I've ever seen that it was an issue on the phone dialplan or Asterisk dialplan |
| 17:04.07 | ManxPower | Nobbie: do you have the same delay if you use the dialplan function/application for getting stuff from the DB? |
| 17:04.32 | Nobbie | mmmmm, but i'm timing how long the "database get" takes by getting start_time before calling AGI->database_get(), and getting end_time directly after it returns, and that can take up to 10 seconds |
| 17:05.07 | ManxPower | Nobbie: mmmm, but that's not what I suggested you try |
| 17:05.19 | Nobbie | what did you suggest ? |
| 17:05.28 | ManxPower | (12:04:07 PM) ManxPower: Nobbie: do you have the same delay if you use the dialplan function/application for getting stuff from the DB? |
| 17:05.34 | ManxPower | dialplan = extensions.conf |
| 17:05.39 | Nobbie | from dialplan vs AGI ? |
| 17:05.41 | ManxPower | "show applications" and "show functions" |
| 17:05.46 | ManxPower | Nobbie: CORRECT! |
| 17:06.01 | Nobbie | even though AGI is called from DialPlan ? |
| 17:06.10 | ManxPower | Nobbie: What you are experiencing is NOT normal. |
| 17:06.16 | Nobbie | tell me about it |
| 17:06.35 | [TK]D-Fender | Nobbie: Always a good thing to confirm that it is indeed that slow via direct dialplan. |
| 17:06.46 | *** part/#asterisk fiddur (n=fiddur@78.82.252.60) |
| 17:06.59 | ManxPower | I'm hoping to be able to stick around to find out what the result of your test is, but I have a pretty busy day. |
| 17:07.05 | Nobbie | ok,that's a good idea. will try it |
| 17:07.09 | ManxPower | So if you make it fast.... |
| 17:08.03 | Nobbie | Manx: won't be, users have gone home, load on box is reduced and the problem then occurs less frequently. if you want, msg me an email address i can send my result to |
| 17:08.24 | ManxPower | Nobbie: No. I only do offchannel consulting for a fee |
| 17:08.42 | Nobbie | you asked, i offered |
| 17:08.53 | ManxPower | Nobbie: did you mention that it mostly happened under load or did I miss that? |
| 17:09.37 | Nobbie | too soon to say, only got proper stats on how long it takes to return from database get during peak office hours |
| 17:09.57 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 17:09.59 | Nobbie | i did mention it's intermittent |
| 17:10.22 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 17:10.44 | Qwell | Nobbie: how many active calls? |
| 17:10.45 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 17:10.49 | ManxPower | Nobbie: try deleting the file if it does not contain information you need. I believe asterisk will recreate it if it is missing. Might want to just move the file where else rather then delete it |
| 17:11.34 | [TK]D-Fender | Nobbie: dumpt the keys in it. Then install * elsewhere (or jsut grab a stock astdb) Then copy the keys you need back. |
| 17:11.44 | [TK]D-Fender | Nobbie: See if that helps. |
| 17:11.56 | Nobbie | QWell: 60 concurrent SIP Channels, most of them are direct from other PSTN gateway to SIP handsets registered to * |
| 17:12.24 | Qwell | I think ManxPower's idea of deleting/moving it out of the way to test is a good one |
| 17:12.34 | [TK]D-Fender | Nobbie: You also said this was running in a VM as well, right? |
| 17:12.38 | Nobbie | nope |
| 17:12.56 | [TK]D-Fender | Nobbie: Ok, must have mixed that comment up earlier. |
| 17:13.09 | *** part/#asterisk ManxPower (n=manxpowe@111.sub-70-221-101.myvzw.com) |
| 17:13.57 | Nobbie | maybe if i #define DETECT_DEADLOCKS the logs could point to the problem ? |
| 17:14.43 | Nobbie | and recompile of course |
| 17:15.51 | deeperror | anyone have some documentation or links on the installation of a sangoma a104d? |
| 17:16.18 | Nobbie | will #define DETECT_DEADLOCKS have a performance degradation ? |
| 17:16.21 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 17:17.45 | jjshoe | deeperror sandgoma's wiki is excellent. |
| 17:18.07 | deeperror | jjshoe: thanks there now |
| 17:19.12 | asdx | shit, my co-worker is a f******** moron, he keeps saying to use some gui crap |
| 17:19.27 | asdx | i should consider another job |
| 17:19.41 | Nobbie | asdx: which u could be my coworker then |
| 17:22.01 | [TK]D-Fender | asdx: You mean the part where they kept asking you to undertake activities that could get you thrown in jail wasn't incentive enough? |
| 17:24.01 | asdx | [TK]D-Fender: nah, i'm doing other things now |
| 17:24.31 | asdx | [TK]D-Fender: trying to make things (non asterisk related) on a xen environment, but i think the environment has some quotas on space issues and they keep saying "use a gui" |
| 17:24.34 | asdx | :S |
| 17:25.14 | *** join/#asterisk yosam (i=B@89.149.214.120) |
| 17:25.18 | yosam | <PROTECTED> |
| 17:25.20 | asdx | virtualization sucks |
| 17:25.35 | JT | virtualisation rocks |
| 17:28.07 | Qwell | actualization is better |
| 17:28.34 | jaytee | people kept telling me to "get a life!" because I worked to much but I was always too busy so I got a "Second Life" and while I was in there I got so bored I started my own Second Life business. Pretty soon I was so busy other avatars were telling me to "get a Third Life!" |
| 17:29.05 | Qwell | jaytee: somebody needs to create an MMO inside of Second Life |
| 17:29.14 | jaytee | Qwell :-) |
| 17:30.03 | jaytee | I prefer actualization to virtualization although virtualization has lots of merit you can't kick the box when it misbehaves. |
| 17:30.10 | jsmith | Qwell: Or an MVNO? |
| 17:30.19 | Qwell | oh dear |
| 17:30.35 | Qwell | jsmith: that would be...interesting |
| 17:30.46 | Qwell | pay 2c/min to talk on your cellphone in second life |
| 17:31.03 | *** join/#asterisk Braxus (n=braxus@netblock-68-183-228-84.dslextreme.com) |
| 17:31.41 | jsmith | Qwell: Wouldn't that be lindens/min, though? |
| 17:31.47 | Qwell | whichever |
| 17:33.48 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 17:33.49 | *** join/#asterisk nny_2 (n=Scott@64.20.130.209.dyn-e-pool3.pool.hargray.net) |
| 17:34.03 | nny_2 | ~mp3 |
| 17:34.04 | jbot | well, mp3 is (MPEG-1 layer 3) This is a compression standard for music. It enables you to fit over 100 full length songs on a single CD with almost no loss of quality. You can find MP3 players and MP3 files on the Web--you just have to look. The music industry is unhappy about MP3 files being swapped around and has shut down some sites that distribute them.. MIME type - audio/mpeg |
| 17:34.14 | nny_2 | :\ |
| 17:34.28 | nny_2 | man voip-info 's wiki is thrashed in the moh mp3 stuff |
| 17:34.42 | hsv-al | i noticed that whole site in general |
| 17:34.46 | hsv-al | the last week has been weird, wtf |
| 17:35.11 | nny_2 | so as far as i understand it, mpg123 is no longer needed and has been replaced by format_mp3 |
| 17:35.14 | nny_2 | heh yeah |
| 17:35.43 | nny_2 | is working on having an extension play a stream from an mp3 source, yet not be the default music on hold |
| 17:36.11 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 17:36.15 | nny_2 | and noticing the intarwebs be full of useful info, both old and new |
| 17:38.47 | yosam | lumenvox really sucks! |
| 17:38.50 | flush | yo |
| 17:39.03 | flush | exten => _9NXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1},,mwW) why did i put this.. what is {EXTEN:1} ? |
| 17:39.16 | flush | sorry im a newb |
| 17:39.18 | nny_2 | so [classes] (new line) stream => blah foo is the right way to add this to moh or the old way |
| 17:39.30 | nny_2 | damn wiki is spattered with version conflicts :) |
| 17:39.45 | Nobbie | flush: it's a substring. you're stripping the 9 from the front of the dialed number before dialing it |
| 17:39.53 | [TK]D-Fender | flush: that represents the # you dialed minus the leading "9" |
| 17:40.10 | [TK]D-Fender | nny_2: What do yuo want to do exactly? |
| 17:40.16 | flush | ohhh kk i remember now.. thanks a lot |
| 17:40.23 | Katty | ohai |
| 17:40.27 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:40.34 | Katty | i can haz orange sodie?! |
| 17:40.58 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 17:41.02 | nny_2 | [TK]D-Fender: i have created a custom extension that does a SetMusicOnHold,stream and trying to define that MOH class in musiconhold.conf to be an mp3 stream |
| 17:41.35 | nny_2 | even though mp3 is iscky |
| 17:41.37 | nny_2 | icky |
| 17:41.53 | [TK]D-Fender | nny_2: So you want to stream for MoH? |
| 17:42.30 | flush | hey a friend of mine asked me if i could forward my incoming call to my skype account if it doesnt answer.. can this be done ? |
| 17:43.24 | kannan | to gtalk, it could be done , i think |
| 17:43.37 | flush | ill google hunt |
| 17:43.46 | *** join/#asterisk hardwire (n=hardwire@rdbk-15777.mtaonline.net) |
| 17:43.50 | kannan | ~skype |
| 17:43.50 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 17:43.51 | hardwire | sup meh homies |
| 17:44.02 | asdx | what do you do when others want to impose some tool you don't want to use at work? |
| 17:44.06 | nny_2 | [TK]D-Fender: yeah |
| 17:44.10 | asdx | impose on you |
| 17:44.15 | hardwire | asdx: duct tape the tool to the wall of shame. |
| 17:44.22 | hardwire | human tool or other tool? |
| 17:44.27 | hardwire | I've met both |
| 17:44.44 | [TK]D-Fender | nny_2: Go look at some rawplayer samples for that. |
| 17:44.52 | nny_2 | [TK]D-Fender: i have mpg123 59r compiled |
| 17:45.07 | nny_2 | [TK]D-Fender: rawplayer, ill check it out ty |
| 17:45.35 | [TK]D-Fender | hardwire: There are numerous "tools" here already... |
| 17:45.45 | hardwire | *here* ? |
| 17:46.36 | hardwire | I have a funny PSTN question for y'all this fine mornin |
| 17:46.50 | hardwire | I have a single line going into the house with DSL on it. |
| 17:47.00 | hardwire | the house wiring goes to my office first, then the rest of the house |
| 17:47.36 | hardwire | off of the office port the dsl connects fine, but plugging a phone (+filter) into that port offers no dialtone |
| 17:47.42 | hardwire | just a bit of white noise |
| 17:47.49 | hardwire | other phones in the house work fine. |
| 17:47.55 | hardwire | am I insane y/n ? |
| 17:48.47 | *** join/#asterisk francogwapito (n=chatzill@125.252.90.5) |
| 17:49.43 | [TK]D-Fender | hardwire: and with no filter? Perhaps the filter is bad? |
| 17:49.52 | nny_2 | [TK]D-Fender: do you know if rawplayer is useful for streamed mp3s from shoutcast etc too? All the info so far seems to suggest it is for local files, but i am still reading |
| 17:49.57 | hardwire | even w/o a filter it's funky |
| 17:50.13 | hardwire | it's like I have two lines, even though I don't |
| 17:50.14 | [TK]D-Fender | nny_2: Never used it personally, just trying to send you somewhere that might help./ |
| 17:50.44 | nny_2 | [TK]D-Fender: kk thanks... i am going to find some way to post a "as of 1.4.X this worked for me on voip-info, cause right now it is missing :) |
| 17:51.20 | *** join/#asterisk jamuse (n=josh@bzq-219-135-48.static.bezeqint.net) |
| 17:51.41 | nny_2 | [TK]D-Fender: seems mpg123 has problems with memory leaks etc |
| 17:51.50 | nny_2 | so rawplayer seems like a good start |
| 17:52.02 | *** join/#asterisk Nasra (n=maxshipp@190.166.71.39) |
| 17:52.32 | jamuse | Can someone help me with the following error msg: Call from 'X' to extension 's' rejected because extension not found |
| 17:52.55 | nny_2 | urp ok streamplayer is the tool now |
| 17:53.09 | nny_2 | man.. you could write a whole book on changes that happen in asterisk heh |
| 17:53.30 | _ShrikE | jamuse: that error is pretty explanatory. The extension s cannot be found in the context you are calling it in. |
| 17:53.31 | russellb | it would never be finished |
| 17:53.36 | russellb | we can easily change 50 to 100 things a day |
| 17:53.38 | jamuse | I dont think I have an extention 's', this started after I upgraded to SVN-branch-1.4-r116466 |
| 17:54.30 | jamuse | sorry for the newb question but I dont think I have an extention s in extentions.conf |
| 17:55.00 | _ShrikE | jamuse: that is what it is telling you. No extension s |
| 17:55.09 | [TK]D-Fender | jamuse: Of course you don't have an "s" extension, thats what its complaining about! |
| 17:55.11 | kannan | hwo to decide between svn checkout of asterisk or to dload the tar balls and get asterisk? |
| 17:55.29 | kannan | how , i meant |
| 17:55.45 | [TK]D-Fender | ~8ball Should kannan use SVN to install Asterisk instead of the source tarballs? |
| 17:55.46 | jbot | Are you smoking crack? |
| 17:55.53 | [TK]D-Fender | kannan: You heard the bot! |
| 17:55.54 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 17:56.04 | Qwell | [TK]D-Fender: are you suggesting he smoke crack? |
| 17:56.08 | [TK]D-Fender | kannan: Tarball full release only! |
| 17:56.09 | jamuse | ahh ok, whats the extention s, I want to foward calls to my iaxy, so I have the following: exten => ${FWDNUMBER},1,Dial(IAX2/iaxy@iaxy/s) |
| 17:56.13 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:56.17 | jamuse | where would the extention s come in? |
| 17:56.22 | kannan | [TK]D-Fender , thanks |
| 17:56.23 | [TK]D-Fender | Qwell: No, jbot is questioning that he already is ;) |
| 17:56.34 | jsmith | kannan: Use the tarballs, unless you have a specific reason not to |
| 17:56.38 | [TK]D-Fender | jamuse: thats where you inbound call is trying to land. |
| 17:56.43 | Qwell | jamuse: I don't think you can use a variable for the exten like that |
| 17:56.56 | Qwell | unless the extension is literally "${FWDNUMBER}" |
| 17:57.04 | jamuse | strange this worked in the 1.2 branch fine |
| 17:57.12 | Qwell | is it a global or something? |
| 17:57.16 | jamuse | yup |
| 17:57.20 | Qwell | that might work |
| 17:57.26 | [TK]D-Fender | jamuse: Thing is that when you did your "register =>" in sip.conf for FWD, you didn't tell them what exten to send your inbound calls to, so * told them to use "s". |
| 17:57.29 | Qwell | but, to answer your question.. |
| 17:57.36 | Qwell | s is used when no extension is given |
| 17:57.37 | [TK]D-Fender | Qwell: thats a constant <- |
| 17:57.39 | kannan | jsmith, ok, i was going thru the doc for asterisk2billing, they had given to build from svn, but i thought it is better to use the tarballs only |
| 17:57.56 | jsmith | kannan: Yes, I recommend the tarballs |
| 17:57.56 | jamuse | cool thanks |
| 17:58.11 | kannan | jsmith , of ATFOT ? , |
| 17:58.23 | [TK]D-Fender | jamuse: if you add "/1234" for whatever your FWD # is to the end of your Register statement, the incoming call will land on that instead of "S" |
| 17:59.17 | jsmith | kannan: Guilty as charged... but promise you won't hold it against me! |
| 17:59.18 | jamuse | sec, I dont understand I want in incoming FWD to get forwarded to my iaxy |
| 17:59.42 | kannan | jsmith, gr8 ! to meet you :) |
| 17:59.43 | jamuse | so how does added my FWD to the end of the register statement help? |
| 18:00.37 | jsmith | jamuse: The purpose of the register statements is to say "Hey FWD, I'm over here. When you get a call for me, send it to this IP address and port." |
| 18:00.49 | Qwell | (and extension, if given) |
| 18:01.02 | jsmith | jamuse: Adding the extension to the end of the register statement is like saying "Hey, and by the way, send it to extension 1234, instead of this other extension" |
| 18:01.45 | jamuse | ok so if my iaxy is called iaxy, I could just add /iaxy right? |
| 18:02.30 | *** join/#asterisk Tourinho (n=tourinho@201.37.118.16) |
| 18:03.24 | Tourinho | hello guys.. which one of g729 codecs do you recommend for a Xeon 2.33 CPU? codec_g729a_v34_pentium4m.tar.gz ? |
| 18:03.26 | [TK]D-Fender | jamuse: Well * doesn't register to your IAXY... its the other way around... |
| 18:03.52 | Qwell | Tourinho: 32-bit install? |
| 18:04.31 | *** join/#asterisk ix33 (n=ix@206.222.13.162) |
| 18:05.03 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:05.10 | Tourinho | Qwell: 32 bits |
| 18:05.22 | Qwell | probably 4m then, yeah |
| 18:05.34 | Qwell | i686 will definitely work though |
| 18:05.44 | Tourinho | Qwell: thank you |
| 18:06.39 | *** join/#asterisk fas3r (n=chatzill@haz95-2-82-243-75-212.fbx.proxad.net) |
| 18:06.46 | fas3r | hello everybody |
| 18:06.54 | fas3r | someuse 7942 ? |
| 18:07.45 | fas3r | just to know if it's possible to edit directly on the phone ip parameter or it's necessary i use TFTP file for conf it ... |
| 18:12.26 | jamuse | I have a section in extentions.conf that forwards incoming calls from FWD coming in on IAX to my iaxy, when I cant connt to FWD via IAX I want to use SIP instead, I added a default context in sip.conf to use the relevant context in extentions.conf but I'm still getting message about the 's' extention, any ideas? |
| 18:12.28 | tzafrir | jameswf-home, regarding Zambia, I'm not sure what "UK colony" means. AFAIK UK, India, Israel, UAE, Kwait, Jordan are far from having the same telephony signalling |
| 18:13.05 | tzafrir | And UAE and Kwait were British up until 1971, IIRC |
| 18:14.26 | *** join/#asterisk dFence (n=chatzill@p5496A2E2.dip0.t-ipconnect.de) |
| 18:15.13 | dFence | hey guys.. anyone of you familiar with a Siemens-HiCom PBX? (not asterisk-related but didn't know any other channel somehow telephony-related...) |
| 18:16.59 | *** join/#asterisk drzed (n=drzed@synflood.homelinux.org) |
| 18:20.36 | drzed | hi there! |
| 18:21.00 | drzed | what could be the problem if signaling calls (via sip) works |
| 18:21.41 | *** join/#asterisk zeniffty2002 (n=zeniffty@mail.revenueworx.com) |
| 18:21.42 | drzed | but rtp does not? |
| 18:21.56 | spokra | firewall? |
| 18:22.38 | drzed | yes there is a firewall in front of asterisk that is configrued to accept all outgoing connections |
| 18:23.18 | spokra | what about the incoming? |
| 18:23.58 | drzed | and additionally allows 10000 10100 incomming |
| 18:24.09 | Strom_L | udp, or tcp? |
| 18:24.16 | drzed | udp |
| 18:24.22 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 18:24.26 | Strom_L | and have you made sure to restrict all sip to just those ports? |
| 18:24.29 | Qwell | and rtp.conf is set to use only those ports? |
| 18:24.42 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-82-206.vif.net) |
| 18:26.33 | drzed | yess rtp conf looks like: rtpstart=10000 \n rtpend=10100 |
| 18:26.42 | nny_2 | is there a way to set WaitMusicOnHold to infinity? |
| 18:26.58 | nny_2 | like exten => 400,n,WaitMusicOnHold() ? |
| 18:27.08 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:27.31 | drzed | Strom_L: why would i need to restrict sip to those ports |
| 18:27.34 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 18:28.26 | [TK]D-Fender | nny_2: that app is indefinite |
| 18:29.19 | nny_2 | [TK]D-Fender: k ty.. fwiw i got it working with mpg123, going to add a responce to voip info |
| 18:29.23 | nny_2 | response |
| 18:29.28 | *** join/#asterisk CVirus (n=GoD@196.205.192.192) |
| 18:29.33 | [TK]D-Fender | drzed: read up : |
| 18:29.35 | [TK]D-Fender | ~sipnat |
| 18:29.35 | 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 |
| 18:29.40 | [TK]D-Fender | ^^^^^^^^^ |
| 18:29.51 | nny_2 | it is kind of already there, but the clarification for what 1.2 needed to define classes in moh and 1.4 uses is tossed back and forth |
| 18:30.10 | nny_2 | anyone who has dealt with classes before should be able to pick out the good parts, but for noobs like me :) |
| 18:31.03 | *** part/#asterisk CVirus (n=GoD@196.205.192.192) |
| 18:31.37 | nny_2 | res_musiconhold.c:609 moh1_exec: WaitMusicOnHold requires an argument (number of seconds to wait) |
| 18:31.43 | drzed | [TK]D-Fender: there is no nat involed in my problem |
| 18:31.44 | nny_2 | heheh 6000000000000000000000000000000000 |
| 18:31.48 | nny_2 | should ifx it!!1 |
| 18:31.52 | nny_2 | ;) |
| 18:32.01 | drzed | [TK]D-Fender: there is just a firewall script running on the asterisk srv |
| 18:32.21 | nny_2 | reading usage notes in console |
| 18:33.39 | nny_2 | everything seems to point to a duration |
| 18:35.29 | ix33 | wow i have a working t1 now. |
| 18:36.05 | nny_2 | ix33: yay, i am waiting for the onsite install :\ |
| 18:36.15 | drzed | do i also need udp for rtp? |
| 18:36.29 | ix33 | i have trial-and-errored my way into passing calls this morning. what an accomplishment. |
| 18:37.20 | ix33 | does anything special need to be set in the zap configs to take advantage of the hardware echo cancellation module on this digium t1 card? |
| 18:40.51 | Strom_L | ix33: set echocancel=yes in zapata.conf for the relevant channels |
| 18:43.57 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 18:45.06 | drzed | Strom_L: do i need tcp for RTP? |
| 18:45.06 | Katty | 61MB isn't very big in terms of database size is it? |
| 18:45.17 | Katty | it shouldn't cause a cap size |
| 18:45.19 | Katty | right? |
| 18:45.26 | Strom_L | drzed: no |
| 18:46.32 | fas3r | someone know where it's possbile to find cisco IOS for Ip Phone (7942) ( not on cisco.com ) my account was expired :s |
| 18:46.43 | ix33 | Strom_L: thanks |
| 18:46.50 | *** join/#asterisk MmixX (i=mmixx@Linux.outboxexpress.com.ph) |
| 18:46.53 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 18:47.34 | ix33 | what is the effect of setting LBO too high? |
| 18:48.50 | fas3r | or if it's possible someone download it for me :p |
| 18:49.04 | Strom_L | ix33: too-high gain on your T1 card |
| 18:49.31 | ix33 | Strom_L: can that physically hurt it, or is this an audio quality issue? |
| 18:49.57 | Strom_L | ix33: um, it's digital |
| 18:50.16 | Strom_L | just set the LBO correctly :) |
| 18:50.34 | ix33 | centurytel can't tell me the distance, so i'm guessing. |
| 18:50.43 | Strom_L | is there a smartjack on your prem> |
| 18:50.44 | Strom_L | ? |
| 18:50.48 | ix33 | s/can't/won't |
| 18:51.09 | ix33 | i don't know, i haven't physically been there yet (will be there tonight) |
| 18:51.21 | Strom_L | facepalms |
| 18:54.09 | drmessano | wow |
| 18:54.11 | drmessano | Yeah |
| 18:54.27 | *** join/#asterisk kimo_sabe (n=nick@zappa.azrackspace.net) |
| 18:54.50 | ix33 | why? i've been doing everything remotely till now |
| 18:55.06 | kimo_sabe | zaptel/zapata isn't disabled the echo can for my IAXModems, what could I be doing wrong? |
| 18:55.39 | *** join/#asterisk reallost1 (n=reallost@72.169.24.231) |
| 18:55.42 | *** join/#asterisk excAliBuR (n=sales@207.134.8.33) |
| 18:55.58 | excAliBuR | where can i tell asterisk what mail thing i'm using? |
| 18:56.06 | Qwell | excAliBuR: mail thing? |
| 18:56.07 | excAliBuR | i use exim |
| 18:56.18 | excAliBuR | to send voicemail to email |
| 18:56.23 | [TK]D-Fender | excAliBuR: voicemail.conf |
| 18:56.33 | Qwell | it doesn't matter. pretty much every MTA is sendmail compatible, and has a sendmail binary |
| 18:56.42 | Qwell | (MTA = mail thing) |
| 18:56.44 | excAliBuR | is there a howto ? |
| 18:56.58 | [TK]D-Fender | excAliBuR: read the sample config and bring some IQ along. |
| 18:56.58 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-184-133.dsl.stlsmo.sbcglobal.net) |
| 18:57.12 | excAliBuR | my voicemail.conf only has 1 line in it :( |
| 18:57.25 | [TK]D-Fender | excAliBuR: read the SAMPLE config and bring some IQ along. |
| 18:57.31 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:57.41 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:58.05 | LemensTS | when im in an "s" extension, and need to give the optin of the user to press 1, 2, 3, 4, etc, is the only way to do that by: exten => 1,1,blah; exten => 2,1,blah; exten => 3,1,blah; ? |
| 18:58.39 | [TK]D-Fender | LemensTS: no, but running an IVR off it is popular. You could always use Read if you wanted to. |
| 18:58.42 | ix33 | there is an NIU box with an adtran h2tur card in it? |
| 19:00.08 | Strom_L | ix33: that sounds right |
| 19:00.21 | Strom_L | ix33: you want the distance from that to the t1 card |
| 19:00.55 | ix33 | so, like, the length of the patch cable the guy used? |
| 19:00.58 | ix33 | easy |
| 19:00.59 | *** part/#asterisk ctooley (n=ctooley@209.33.106.165) |
| 19:01.11 | ix33 | even he should be able to tell me that... |
| 19:01.43 | Strom_L | well, be specific...make sure you account for the full length of wire between the two |
| 19:02.02 | Strom_L | there may be more than just the patch cable ;) |
| 19:02.12 | ix33 | good point. |
| 19:03.11 | drzed | ok found one firewall problem |
| 19:03.16 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:03.36 | drzed | but now the other guy can hear me but i cannot hear him |
| 19:04.39 | [TK]D-Fender | drzed: And where are "you" relative to you * server, and where is "he" relative to it as well? |
| 19:06.15 | ix33 | Strom_L: thanks, it sounds like he just grabbed a huge length of cat5 he had laying around |
| 19:06.23 | ix33 | we'll address that tonight... |
| 19:06.40 | nny_2 | man there seems a way to turn this did stack into a variable based setup |
| 19:06.46 | drzed | he is connected via iax to the * |
| 19:06.49 | nny_2 | the numbers are XXX-XX10, 11 etc |
| 19:06.55 | nny_2 | and the extensions are 10, 11 etc |
| 19:06.56 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:07.20 | vector | ix33, tie him up with it ;) see how he likes being inconvenienced by an unnecessarily huge bundle of cable |
| 19:07.30 | drzed | i am behind a nat and connect to a public sip service (using stun stuff) |
| 19:07.53 | nny_2 | so rather than a list of 20 exten=> _XXXXX10,1,Goto etc. I should be able to pull the 2 digits out.. and make that a variable right? |
| 19:08.05 | [TK]D-Fender | drzed: please draw a full picture including this newly introduced "sip service". |
| 19:08.27 | [TK]D-Fender | nny_2: You need to read up on your variable basics... |
| 19:08.35 | drzed | ok, mom (its a little wired) ... |
| 19:08.40 | [TK]D-Fender | nny_2: that IS a variable... guess which one.. |
| 19:08.47 | *** part/#asterisk LemensTS (i=CustomGT@adsl-70-238-184-133.dsl.stlsmo.sbcglobal.net) |
| 19:09.13 | *** join/#asterisk reallost1 (n=reallost@72.169.24.231) |
| 19:09.41 | reallost1 | I'm running 1.6-beta9 and I've got asterisk processes using 100% cpu |
| 19:09.46 | nny_2 | extension |
| 19:09.49 | nny_2 | ahhh |
| 19:09.52 | vector | nny_2, dial(${EXTEN:5}) or something of that nature off the top of my (dusty) head |
| 19:09.53 | reallost1 | Anyone around who can help me debug it? |
| 19:10.03 | nny_2 | ha vector was just gonna type that |
| 19:10.23 | nny_2 | vector: ty though, [TK]D-Fender 's kick helped |
| 19:10.53 | vector | I would have suggested to follow his lead anyway since I really was not sure about exact syntax |
| 19:11.34 | nny_2 | vector: nah your suggestion is perfect |
| 19:11.52 | nny_2 | well changing it to fit my plan but yeah |
| 19:12.34 | [TK]D-Fender | winds up for another swing |
| 19:13.02 | *** join/#asterisk metfan2007 (n=jc@201.103.142.225) |
| 19:14.10 | metfan2007 | hi all |
| 19:14.44 | *** join/#asterisk los415 (n=los415@sfca-office.corp.race.com) |
| 19:16.12 | metfan2007 | I have a problem installing a Polycom 330 phone and Asterisk, when I try to call without pressing "Line 1" or "Line 2" everything is Ok, but if I take line pressing "Line 1" and Dial a number I receive a "chan_sip.c:13815 handle_request_invite: Failed to authenticate user "Juan Carlos Huerta" <sip:1102@192.168.2.99>;tag=31BD6741-D850EFEE" message in Asterisk |
| 19:16.16 | metfan2007 | any idea? |
| 19:18.05 | ix33 | sounds like a phone.cfg issue |
| 19:18.21 | spokra | if you press line two and dial does it work? |
| 19:18.40 | ix33 | is it actually using both lines (like, two different SIP accounts?) |
| 19:18.44 | metfan2007 | no, only if I dial the number directly |
| 19:18.58 | metfan2007 | is the same account, only 1 Asterisk server |
| 19:19.11 | Kobaz | does "flash" in zapata.conf actually do anything? |
| 19:19.22 | Kobaz | i'm trying to change the flash duration |
| 19:19.25 | ix33 | you can have 2 on the same server (2 different auth'd extnsions) |
| 19:19.52 | spokra | I;d double check the phone config. are you using the web interface? |
| 19:20.00 | ix33 | but anyway, xmllint the file containing your reg information to make sure there's no typos or something |
| 19:20.28 | ix33 | answer spokra 1st, since i assumed you were doing file based configs |
| 19:20.28 | metfan2007 | Yes, I only use web interface, no XML |
| 19:20.31 | spokra | if it works at all you have the auth setup right somewhere in the phone.. probably just not correct on both lines |
| 19:20.34 | ix33 | ok never mind |
| 19:20.53 | ix33 | silly me i thought everyone wrestled with the file configs... |
| 19:20.58 | ix33 | on polycom |
| 19:22.15 | [TK]D-Fender | people who configure Poloycom phones by anything other than provisioning should be dragged out and shot. |
| 19:22.17 | reallost1 | Anyone here I can show a backtrace to? |
| 19:22.18 | [TK]D-Fender | Polycom* |
| 19:22.49 | metfan2007 | ???? mmm |
| 19:23.00 | Kobaz | do de do |
| 19:23.03 | Kobaz | kicks flash() |
| 19:23.39 | metfan2007 | so? |
| 19:23.40 | metfan2007 | :( |
| 19:25.40 | los415 | lol the polycom web interface on there phones suck |
| 19:25.41 | Kobaz | kicks flash() repeatidly |
| 19:25.57 | Kobaz | los415: it does, you can only change one section and then you have to reboot |
| 19:26.05 | los415 | yup |
| 19:26.17 | los415 | i dont know if they still do it either |
| 19:26.23 | Strom_L | los415: almost as bad as your lack of knowledge as to whether to use "there" "their" or "they're" :) |
| 19:26.26 | los415 | where u reboot and it takes like 5 min for the web interface to start working |
| 19:26.43 | los415 | strom i didnt know we where in english class |
| 19:26.45 | spokra | that is true.. but to learn and setup the first one it works then when you want to do 50 you know what to do in the xml file! |
| 19:26.46 | los415 | sorry |
| 19:27.12 | Kobaz | errrr, god damn hook flash |
| 19:27.41 | reallost1 | http://pastebin.ca/1032432 |
| 19:29.15 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 19:29.24 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 19:29.28 | hmmhesays | what up folks |
| 19:29.41 | nny_2 | vector: exten => _84398750xx,1,Goto(transfer,${EXTEN:8},1) |
| 19:29.45 | nny_2 | is what i ended up with |
| 19:29.50 | nny_2 | seems to work well |
| 19:30.10 | nny_2 | transfer has options for call routing, multi ringing etc in it, hence the use instead of dial |
| 19:30.35 | [TK]D-Fender | nny_2: Yippy-kai-yay! |
| 19:30.43 | seanbright | reallost1: what version of asterisk? |
| 19:30.57 | reallost1 | Asterisk 1.6.0-Beta9 |
| 19:31.04 | seanbright | ah |
| 19:31.14 | *** part/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 19:31.19 | seanbright | reallost1: if you haven't already, i would submit an issue on mantis |
| 19:31.29 | reallost1 | k, will do. |
| 19:31.38 | nny_2 | [TK]D-Fender: heh .. funny you quote bruce willis, did anyone see the last die hard, or as I call it "Bruce Willis doesn't understand this whole computer thing" |
| 19:31.59 | nny_2 | at least his character |
| 19:32.29 | nny_2 | I, (Bruce Willis) kick ass.. you (annoying mac guy) act like a little bitch... |
| 19:33.01 | [TK]D-Fender | nny_2: Notice how the Mac in there is a useless whiny douchebag? Who says movies are unrealistic? ;) |
| 19:33.14 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:33.22 | nny_2 | LOL |
| 19:33.24 | nny_2 | er heh |
| 19:33.32 | nny_2 | http://penny-arcade.com/comic/2007/7/16/brains-with-urgent-appointments/ |
| 19:37.48 | hubguruJR | Hi All, I'm testing 1.6.0-beta, going well |
| 19:39.55 | Kobaz | does "flash" in zapata.conf actually do anything? i'm trying to change thr flash duration |
| 19:40.09 | [TK]D-Fender | Kobaz: Yes, it does. |
| 19:40.27 | [TK]D-Fender | Kobaz: And that'd be Flash() in extensions.conf, not Zapata.conf |
| 19:43.00 | Kobaz | zapata.conf |
| 19:43.03 | Kobaz | flash = xxx |
| 19:43.19 | Kobaz | in the docs it says it controls the duration of the flash |
| 19:44.42 | Kobaz | i dont seem to get any difference between 50ms or 1000ms |
| 19:45.13 | Kobaz | we do have flash hook capability on this pbx tha we're getting some fxs's from |
| 19:45.43 | Kobaz | on the pbx it's range is 45ms to 750ms, i've tried just about everything in between |
| 19:46.48 | *** join/#asterisk mintee (n=mintone@72-165-177-94.dia.static.qwest.net) |
| 19:46.58 | [TK]D-Fender | Kobaz: defaults to 500ms IIRC. |
| 19:47.04 | hubguruJR | comments in extensions.conf, [;-some comment] ok, [;--some comment] breaks parsing of file, [;---some comment] ok, there is something about 2 dashes that asterisk doesn't like? |
| 19:47.07 | *** join/#asterisk [hC] (n=hardcore@S0106001346a4b813.vc.shawcable.net) |
| 19:47.13 | Kobaz | yeah |
| 19:47.16 | [TK]D-Fender | Kobaz: this is indeed the first time I've ever heard anyone try to tweak that |
| 19:48.23 | mintee | ~book |
| 19:48.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:48.25 | [TK]D-Fender | hubguruJR: pastebin your complete sample and its associated "show dialplan" output. |
| 19:48.44 | florz | hubguruJR: yes, you can make multi-line comments somehow with ;-- ... --; or such |
| 19:49.07 | Kobaz | [TK]D-Fender: mm |
| 19:49.49 | hubguruJR | in extensions.conf: ;--testing comments , cli output: lab11*CLI> dialplan reload |
| 19:49.50 | hubguruJR | <PROTECTED> |
| 19:49.50 | hubguruJR | [May 28 14:43:17] WARNING[29502]: config.c:1392 config_text_file_load: Unterminated comment detected beginning on line 10 |
| 19:49.50 | hubguruJR | <PROTECTED> |
| 19:49.50 | hubguruJR | [May 28 14:43:17] WARNING[29502]: pbx.c:5492 ast_merge_contexts_and_delete: Requested contexts didn't get merged |
| 19:49.58 | smash- | [TK]D-Fender Hey should this file be the only place i am setting the callerid info? > http://pastebin.com/m366f0efb = part of sip.conf |
| 19:50.37 | [TK]D-Fender | smash-: looks fine |
| 19:50.56 | smash- | =/ |
| 19:51.21 | smash- | iss there anything in zaptel i would have to configure to make those numbers available to be asigned? |
| 19:51.50 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584441.dsl.bell.ca) |
| 19:51.55 | [TK]D-Fender | hubguruJR: well florz here seems to be aware of a multi-line comment syntax, so I'd take his lead on this |
| 19:52.15 | [TK]D-Fender | smash-: you need to look at whats actually happening in your call... |
| 19:52.23 | smash- | my pbx wont let me set callid through sip.conf it apears. I'm trying to find a secondary source it is setting callerid from. not much luck so far |
| 19:52.27 | smash- | call trace it, ok |
| 19:53.59 | Kobaz | [TK]D-Fender: i do a flash(), then senddtmf(), i get the tones back to the calling party |
| 19:54.03 | hubguruJR | thanks [TK]D-Fender |
| 19:54.21 | Kobaz | when they should be going to the pbx on the other end of the fxo |
| 19:54.50 | [TK]D-Fender | Kobaz: I'd pay close attention as to who is getting what there. I'm guessing the wrong party. pastebin exactly what you're doing now. |
| 19:55.30 | hubguruJR | florz, 1 or 3 dashes works fine, 2 screws up. |
| 19:56.31 | hubguruJR | so prefixing ; in front of a comment isn't a sure thing |
| 19:57.04 | Kobaz | http://pastebin.com/m344f97ac |
| 19:57.12 | drmessano | I'm so sick of hearing about iLbc |
| 19:57.19 | hubguruJR | somehow the parser is recognizing ;-- and wants to interpret it into something useful instead of ignoring it |
| 19:57.21 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:57.30 | drmessano | could someone PLEASE.. PLEASE post a sticky "We dont' hate iLbc, REALLY" |
| 19:58.32 | Kobaz | [TK]D-Fender: it's been pastebinned... |
| 19:58.46 | [TK]D-Fender | Kobaz: .... link.... |
| 19:59.07 | Kobaz | http://pastebin.com/m344f97ac |
| 19:59.16 | Kobaz | dankee :) |
| 20:00.11 | Kobaz | let me simplify it |
| 20:00.45 | florz | hubguruJR: well, then something like that must be the sequence, I suppose :-) |
| 20:01.11 | florz | hubguruJR: have fun reading the source if you want to know it exactly - I don't think that's documented anywhere ... |
| 20:01.36 | hubguruJR | yea, right, not bedtime yet |
| 20:01.47 | hubguruJR | still have work to do |
| 20:01.58 | florz | hubguruJR: but with asterisk parsing something, generally nothing is a sure thing ... |
| 20:02.39 | hubguruJR | thinking of posting to the user or dev list |
| 20:02.52 | hubguruJR | I bet someone else has noticed this |
| 20:03.16 | florz | hubguruJR: well, at least the developers should =:-) |
| 20:03.29 | florz | hubguruJR: but I wonder what that will help you?! |
| 20:04.10 | hubguruJR | they do fine |
| 20:04.17 | hubguruJR | help when they can |
| 20:04.47 | florz | well, yeah, but I don't really see what your problem is!? |
| 20:04.50 | Kobaz | now i'm getting device or resource busy |
| 20:05.01 | Kobaz | [May 28 16:04:40] WARNING[5553]: app_flash.c:97 flash_exec: Unable to flash channel Zap/3-1: Device or resource busy |
| 20:05.19 | Kobaz | i'm trying to just do a quick, answer() and then flash(), with a wait in between, hmm |
| 20:05.28 | [TK]D-Fender | Kobaz: whats actually going on. You see to call out with auto-answer. Whatexactly happens from there? |
| 20:05.28 | Kobaz | i was getting this before and adding a wait() did the trick, but that's not doing it now |
| 20:05.34 | hubguruJR | for instance, I use comments ALLOT in all files, for others that my need to come behind me |
| 20:06.00 | nny_2 | if someone is dialing out from a sip client, what is variable descibes the extension? (i.e. 10) |
| 20:06.01 | [TK]D-Fender | hubguruJR: this seem easy enough to avoid... |
| 20:06.04 | hubguruJR | comments usually start with ;----------some comment |
| 20:06.11 | nny_2 | looking at variables list, and SIPCALLID looks close |
| 20:06.14 | Kobaz | [TK]D-Fender: i answer the call, drop it into a queue, i use the ami to move the call to the zapTransfer context which does a flash() on the line and then sends digits |
| 20:06.17 | [TK]D-Fender | nny_2: .... ${EXTEN} <- |
| 20:06.31 | smash- | w0w |
| 20:06.33 | nny_2 | oh.. eh i thought that would be the number they dialed |
| 20:06.35 | smash- | tk ur a genius |
| 20:06.37 | hubguruJR | the other day, i happened to leave a ;--some comment, and it broke the installation |
| 20:07.02 | smash- | i see in call trace, execute application: (SetCIDNum) |
| 20:07.07 | florz | hubguruJR: well, I'd change to a different format then!? |
| 20:07.08 | hubguruJR | i didn't notice till the next day, when the customer notified me that a feature was not working |
| 20:07.16 | nny_2 | i asked the question wrong, my fault |
| 20:07.17 | smash- | now i need to find a way to manipulate this |
| 20:07.49 | hubguruJR | this was a big deal for me so it's on my radar to find a fix |
| 20:07.57 | Kobaz | anyways, i simplified it to this: http://pastebin.com/m45963b58 |
| 20:09.04 | Kobaz | oh, bah, i have a syntax error |
| 20:09.13 | [TK]D-Fender | Kobaz: I don't see your "Wait" being executed in there, do you? |
| 20:09.27 | [TK]D-Fender | Kobaz: I have a good idea why :) |
| 20:09.48 | [TK]D-Fender | Kobaz: Not all syntax errors are equal.... |
| 20:09.57 | Kobaz | okay, so i fixed the -> |
| 20:10.04 | [TK]D-Fender | Kobaz: If this one wre, it wouldn't be an error :p |
| 20:10.10 | [TK]D-Fender | were* |
| 20:10.15 | smash- | [TK]D-Fender : this is my call trace http://pastebin.com/m5c8ab62b , i see it executing AGI script but i dont know where that is, and u can see it setting the callerid to 3609891290 instead of the value placed in sip.conf |
| 20:10.46 | Kobaz | so i fixed it: and now: http://pastebin.com/m55aacf8c |
| 20:10.47 | smash- | [TK]D-Fender: let me rephraze that idk what or where agi script is. |
| 20:10.50 | [TK]D-Fender | smash-: "thats nice" |
| 20:10.59 | [TK]D-Fender | smash-: This is your server, get a clue. |
| 20:11.02 | nny_2 | ha SIPCALLID was fun |
| 20:11.08 | nny_2 | CALLERID(num)=8439875000503b3f52-727ddaf7@10.0.0.103" |
| 20:11.45 | Kobaz | [TK]D-Fender: which would leave one to believe it was successfull, but it wasn't |
| 20:12.12 | [TK]D-Fender | Kobaz: I never said that it would be, but if I'm going to judge something I'd like to start from a sane place. |
| 20:12.27 | Kobaz | yeap |
| 20:12.43 | NovceGuru | drmessano: what about the g.719 codec |
| 20:12.48 | NovceGuru | for the polycom "hd voice" krap |
| 20:12.55 | Kobaz | what other info would you like me to dig up? |
| 20:14.01 | [TK]D-Fender | Kobaz: I don't see the wait kicking in so the line hasn't been siezed long enough for a flash to be effective. |
| 20:14.26 | nny_2 | hmmph |
| 20:14.36 | Kobaz | the wait kicks in |
| 20:14.47 | [TK]D-Fender | Kobaz: not in the last ver you showed me. |
| 20:14.55 | Kobaz | <PROTECTED> |
| 20:15.07 | [TK]D-Fender | Kobaz: strike that.. missed the new PB |
| 20:15.43 | nny_2 | having a hard time with defining dialed number extension vs. sip client extension in terminology forms, as they are both extensions to asterisk. This is probably due to the sharp blow to my head from hitting it with my palm the last time I asked such a question. |
| 20:15.44 | [TK]D-Fender | Kobaz: What is supposed to have happened? |
| 20:16.02 | Kobaz | we're supposed to now get another line to be able to dial a number and transfer |
| 20:16.06 | Kobaz | i can do it with a regular phone |
| 20:16.10 | Kobaz | but not with asterisk |
| 20:16.35 | Kobaz | regular phone, plug it in, someone calls, hit flash, dial a number, hang up, call gets transfered over just fine |
| 20:16.43 | [TK]D-Fender | Kobaz: here's an idea : SendDTMF each digit with a .5s wait between each, and a wait(10) at the END. |
| 20:16.47 | nny_2 | Kobaz: I have that here |
| 20:17.05 | nny_2 | Kobaz: I have a DP set up with that in place would it help if i posted it? |
| 20:17.15 | Kobaz | nny_2: sure |
| 20:17.43 | Kobaz | [TK]D-Fender: yeah i've done that, it doesnt matter what the delay in dtmf is because it's not sending it to the right place |
| 20:17.47 | [TK]D-Fender | Kobaz: it could also be taht * sends the DTMF too fast for the other system to get it all. |
| 20:17.55 | nny_2 | Kobaz: http://pastebin.com/m6bbf4740 |
| 20:18.06 | Kobaz | i hear the dtmf on the calling party side |
| 20:18.12 | nny_2 | the ZAP cnotext is so if my system uses a SIP channel it doesn't try the same thing |
| 20:18.23 | nny_2 | mind you this is with my telco's centrex, yours may vary |
| 20:18.24 | Kobaz | yeah |
| 20:18.41 | nny_2 | has exten => s,2,Goto(s-${CHANNEL:0:3},1) previous |
| 20:18.43 | Kobaz | nny_2: yeah, see, i have exactly the same thing |
| 20:18.57 | nny_2 | kk lol one of these days i'll giev out usegul info :D |
| 20:19.01 | nny_2 | useful* |
| 20:19.04 | Kobaz | heh |
| 20:19.19 | nny_2 | sorry to interrupt |
| 20:19.42 | Kobaz | [TK]D-Fender: person a calls phone b (b = asterisk), b does a flash, and dtmf, but person a hears dtmf |
| 20:20.35 | Kobaz | it's like, the flash hook isn't flashy enough |
| 20:20.44 | Kobaz | which is why i've been playing with the delay |
| 20:20.56 | Kobaz | flash hook duration |
| 20:21.41 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 20:22.05 | Kobaz | i can set flash to 5000 |
| 20:22.20 | Kobaz | and it does seem like it wait 5 seconds |
| 20:22.52 | Kobaz | but the thing is, if you're on hook for that long, i would assume the remote pbx would hang it up due to detecting a hangup |
| 20:23.10 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 20:23.27 | Kobaz | if i hold the hook for more than 3 seconds, the line will hang up, but not with asterisk doing it |
| 20:24.37 | spokra | Kobaz: have you tried pulling one of your trunks putting a POTS phone on it and calling someone.. doing a flash and seeing if you can 3-way!! the line might not even have the capability |
| 20:24.44 | Kobaz | yeah, that all works fine |
| 20:24.58 | nny_2 | anyone got a magic way to get a sip user ID intoa variable? |
| 20:25.17 | spokra | in zapada what kind of line do you define it as |
| 20:25.29 | spokra | loop start kewl start etc.. |
| 20:25.39 | Kobaz | fxs_ks |
| 20:26.18 | spokra | and it happens on more then one line |
| 20:26.37 | [TK]D-Fender | nny_2: ${CHANNEL} + CUT |
| 20:26.43 | Kobaz | spokra: yeah |
| 20:26.44 | Katty | HAI FENDER |
| 20:26.54 | nny_2 | [TK]D-Fender: ok will try |
| 20:26.55 | Katty | [TK]D-Fender: i just got a present :> |
| 20:27.02 | [TK]D-Fender | Katty: :O |
| 20:27.06 | Katty | [TK]D-Fender: GUESS! |
| 20:27.16 | nny_2 | >< |
| 20:27.20 | spokra | kobaz: how about a butset.. bridge across the line in monitor.. do you hear the hook flash? |
| 20:27.25 | Kobaz | yeah |
| 20:27.43 | hmmhesays | Holy carp its Katty |
| 20:27.53 | Katty | hmmhesays: shhhh |
| 20:27.57 | Katty | hmmhesays: no one knows i'm here! |
| 20:28.02 | hmmhesays | ducks |
| 20:28.03 | spokra | strange.. all a hook flash is, is an open for 500ms |
| 20:28.04 | Kobaz | somehow the astrerisk hook flash is different from a single line set hook flash |
| 20:28.08 | Kobaz | yeah i know |
| 20:28.13 | Katty | hugs hmmhesays |
| 20:28.21 | Katty | hmmhesays: any new lady friends? :> |
| 20:28.42 | hmmhesays | Katty, no, I almost made a no no with an ex lady friend though |
| 20:28.49 | hmmhesays | Now I'm staying away from the chicka's for awhile |
| 20:28.57 | Katty | oh |
| 20:29.02 | Katty | but... |
| 20:29.14 | Katty | pre-designated contracts... |
| 20:29.15 | Katty | and stuff |
| 20:29.19 | Katty | no? |
| 20:29.24 | Kobaz | spokra: so yeah, this is kinda rought |
| 20:29.51 | spokra | get sip trunks and ditch you analog!! ROFLOL!! |
| 20:29.58 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 20:30.02 | Assid | heya |
| 20:30.04 | [TK]D-Fender | ok, heading home, BBIAB |
| 20:30.15 | Assid | is it possible to run asterisk in a VPS environment? |
| 20:30.30 | hmmhesays | you can |
| 20:30.57 | spokra | would you be able to use conf or music on hold? |
| 20:31.04 | Assid | hmmhesays: and it will support meetme using ztdummy ? |
| 20:31.08 | spokra | you would not have the hardware timer would you |
| 20:31.10 | Katty | Assid: yes. |
| 20:31.19 | Assid | sweet |
| 20:31.35 | Assid | so moh.. meetme.. all should work |
| 20:31.36 | Assid | kool |
| 20:31.41 | Assid | thanks |
| 20:31.42 | spokra | kool.. I was thinking of using a vsp also.. |
| 20:34.18 | nny_2 | SIP/12-08f11f28 into ${CHANNEL:4:6} should be 12 right? |
| 20:36.41 | Assid | Katty: works within openvz by chance? |
| 20:36.50 | *** join/#asterisk CVirus (n=GoD@196.205.192.192) |
| 20:37.34 | Assid | i guess i better play and check |
| 20:38.44 | nny_2 | i get Set("SIP/12-08f11f28", "CALLERID(num)=8439875012-08") |
| 20:38.53 | nny_2 | wonder why the -08 are still there |
| 20:39.03 | nny_2 | is/are |
| 20:39.15 | *** join/#asterisk SamuraiDio (n=diovani@201.41.41.235) |
| 20:39.17 | SamuraiDio | hi |
| 20:39.56 | SamuraiDio | how do asterisk knows it a call is inside the network (sip-sip) or to the outside? |
| 20:40.05 | SamuraiDio | ...if* a call... |
| 20:40.16 | nny_2 | based on what channel you tell it to use |
| 20:40.39 | Katty | Assid: baroo? |
| 20:40.43 | Katty | Assid: i think i missed something. |
| 20:41.02 | ix33 | i saved my company $10,000 on a pbx and all i got was a lousy jar of peanut m&m's. |
| 20:41.07 | nny_2 | lol |
| 20:41.10 | Katty | Assid: 'openvz' does not parse. |
| 20:41.14 | Katty | Assid: pls to try again. |
| 20:41.20 | Assid | hehe.. k |
| 20:41.27 | nny_2 | ix33 sell them to other companies and buy many many jars |
| 20:41.35 | SamuraiDio | i found it |
| 20:42.05 | Assid | Katty: planning to put up a ubuntu box.. load up openvz on it.. so asterisk has its own private little thing.. and i can work with the addl resources for other tasks.. without having 1 system bother the next one |
| 20:42.22 | nny_2 | any variable ninjas here know why a ${CHANNEL:4:6} returns "12-08"? |
| 20:42.37 | *** join/#asterisk jdjurici (n=jdjurici@78-1-137-66.adsl.net.t-com.hr) |
| 20:42.42 | nny_2 | full CHANNEL = SIP/12-08f11f28 |
| 20:42.45 | Katty | Assid: ahhhh. |
| 20:42.48 | jdjurici | io |
| 20:42.50 | Katty | Assid: i haz no idea. |
| 20:42.54 | jdjurici | how are you folks? |
| 20:42.56 | Katty | Assid: but i hope it works for you (= |
| 20:43.04 | Assid | hehe.. thanks |
| 20:44.20 | Kobaz | spokra: so, heh... i'm very stuck |
| 20:44.23 | jdjurici | anyone having hint on what could cause problems between h323 and mgcp, that would cause asterisk to not send rtp to mgcp gateway.... |
| 20:44.48 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 20:46.02 | jdjurici | asterisk is version Asterisk 1.4.19 |
| 20:47.28 | spokra | kobaz: what digiam card are you using? or is it a knock off? |
| 20:47.50 | spokra | what version of astersik |
| 20:48.42 | Kobaz | sangoma |
| 20:48.52 | Kobaz | i've tried different asterisk versions... 1.4.14, 1.4.18, etc |
| 20:49.19 | spokra | what about the drivers for the card |
| 20:49.58 | Kobaz | latest |
| 20:50.03 | fas3r | someone have sip ios for 7942G ( version 8 ) ? i don't find it and i don't know how to get Special Access File it's not possible to download it with traditionnal account ... |
| 20:50.44 | spokra | zaptel-1.4.X ? |
| 20:50.49 | Kobaz | yeah 1.4 |
| 20:51.50 | spokra | don;t know.. I know hook flash works on digiam hardware I've done it |
| 20:54.07 | Kobaz | yeah |
| 20:54.12 | Kobaz | i know it works on sangoma hardware too |
| 20:54.18 | Kobaz | since i was doing it this morning |
| 20:54.25 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 20:54.28 | Katty | [TK]D-Fender: get out. |
| 20:57.11 | [TK]D-Fender | Katty, ....PARDON? |
| 20:57.20 | Qwell | [TK]D-Fender: She said... |
| 20:57.56 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:58.08 | Katty | get out. |
| 20:58.23 | kannan | bye all |
| 21:02.12 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 21:04.47 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 21:05.53 | *** join/#asterisk sniper_voip (n=michofr@62.84.81.170) |
| 21:05.56 | grandpapadot | Greetings, Might Baud Warriors! |
| 21:06.18 | iratik | Just googled this and had less than 10 results total! ... ."SIOD ERROR: unbound variable : tts_textasterisk" when executing command "Festival(mary had a little lamb)" |
| 21:06.33 | sniper_voip | hi all..I need to know please how I can configure the asterisk server to send calls to a gateway directly and not redistered as endpoint |
| 21:07.24 | *** join/#asterisk osiris (n=osiris@c-71-205-9-42.hsd1.mi.comcast.net) |
| 21:07.49 | NovceGuru | what up grandpapadot |
| 21:08.04 | grandpapadot | 'sup G |
| 21:08.51 | NovceGuru | Not much, getting through the day |
| 21:09.21 | grandpapadot | Same here... It's been a busy^100 month... |
| 21:09.32 | *** join/#asterisk mgdm_ (n=michael@serenity.mgdm.net) |
| 21:09.35 | grandpapadot | First time I've been in channel since Sat. |
| 21:11.05 | fas3r | where it's possible to find ios sip for ip phone cisco ? |
| 21:11.20 | grandpapadot | NovcGuru: Hey check out these videos we're putting online, we're still waiting on the audio from Allison so there's no public link to them yet: http://ironvoice.com/tv |
| 21:11.41 | Katty | [TK]D-Fender: i still wuv you. |
| 21:11.50 | grandpapadot | Sup, Katty. |
| 21:12.01 | Katty | grandpapadot: sky |
| 21:12.20 | grandpapadot | Katty: I haven't seen it in what seems like weeks, lol, so I wouldn't know |
| 21:12.29 | Katty | grandpapadot: aww :< |
| 21:12.34 | Katty | grandpapadot: you need a mini vacation, and a hike! |
| 21:12.46 | Katty | grandpapadot: fruits, veggies, water, and sunshine! |
| 21:12.54 | NovceGuru | grandpapadot: nice |
| 21:12.57 | grandpapadot | Katty: I did manage to break out for a run last night, but it was like 10:00p. |
| 21:13.05 | Katty | :< |
| 21:13.09 | Katty | maybe in alaska... |
| 21:17.12 | iratik | can someone help me with festival? when using "festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n" in /etc/asterisk/festival.conf ... i get an error in the festival server log ... ."SIOD ERROR: unbound variable : tts_textasterisk" when executing command in the dial plan "Festival(mary had a little lamb)" |
| 21:17.14 | iratik | any ideas? |
| 21:18.38 | fas3r | ok thank all |
| 21:18.47 | nny_2 | [TK]D-Fender: SIP/12-08f11f28 into ${CHANNEL:4:6} should be 12 right? |
| 21:19.15 | [TK]D-Fender | nny_2, nope |
| 21:19.45 | nny_2 | er 4:5 ? |
| 21:20.03 | [TK]D-Fender | nny_2, nope. Go read the chapter on variable usage again. |
| 21:20.07 | spokra | iratik did you modify the festival config file.. to accept the new command. |
| 21:20.13 | iratik | yes |
| 21:20.15 | iratik | need a paste? |
| 21:20.35 | russellb | it would be easiest to use CUT() for that, IMO |
| 21:20.36 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 21:20.41 | spokra | not /etc/asterick.festival.conf the festivial conf file in usr/share .... |
| 21:21.10 | iratik | no conf files i can see in /usr/share |
| 21:22.07 | [TK]D-Fender | nny_2, ... Page 140 |
| 21:22.14 | nny_2 | k |
| 21:22.22 | iratik | bingo .. found something that i might be doing wrong |
| 21:22.38 | spokra | /usr//share/festival/festival.scm |
| 21:22.50 | spokra | there are changes you need to make in there |
| 21:23.47 | nny_2 | Ah |
| 21:23.49 | nny_2 | >< |
| 21:23.57 | nny_2 | X is the number of digits to return |
| 21:24.08 | nny_2 | 4:2 |
| 21:24.11 | iratik | spokra: can you give me a pastie of your tts_textasterisk definition? |
| 21:24.17 | [TK]D-Fender | nny_2, close, but not quite |
| 21:24.19 | nny_2 | the answer to everything |
| 21:24.19 | iratik | the one on http://www.voip-info.org/wiki/index.php?page=Asterisk+festival+installation is janky |
| 21:24.23 | nny_2 | er |
| 21:24.29 | nny_2 | 4:! because 0 is counted? |
| 21:24.32 | nny_2 | 4:1 |
| 21:24.48 | [TK]D-Fender | nny_2, look at what you wrote and read that paragraph again |
| 21:25.14 | nny_2 | kk |
| 21:25.35 | spokra | iratik.. add this to the end of the scm file http://pastebin.com/m286c1110 |
| 21:25.57 | iratik | thanks |
| 21:26.00 | iratik | restart festival? |
| 21:26.11 | spokra | yes |
| 21:26.19 | hmmhesays | fscking voipjet is not sending back a proper busy indication |
| 21:26.26 | nny_2 | so :4 would be 12-08f11f28 but :4:2 wouldn't be 12 or was the leading : the error you saw? |
| 21:27.33 | [TK]D-Fender | nny_2, No, there error was you saying "x" is the number of digits to return. It was "y" |
| 21:28.18 | outtolunc | pos:offset |
| 21:28.27 | nny_2 | ahh ok |
| 21:28.34 | [TK]D-Fender | offset:length |
| 21:28.38 | outtolunc | nods |
| 21:28.40 | nny_2 | k thanks for the pointers |
| 21:29.02 | outtolunc | <- almost no sleep last night brain-fried |
| 21:29.19 | *** join/#asterisk adr3nalin3 (n=REDGLAZE@asa.redglaze.com) |
| 21:29.29 | nny_2 | heh sad thing is i orinally mis-read/ screwed it up as pos1:pos2 and all mys variables that used it so far still worked n the same manner that rainman manages to wipe his own ass |
| 21:29.47 | iratik | it worked! |
| 21:29.53 | nny_2 | pos1: pos2 stupid smiley BS |
| 21:29.54 | Corydon76-dig | outtolunc: deep-fried brains? |
| 21:30.10 | [TK]D-Fender | graaaarrrrgh!!! |
| 21:30.16 | outtolunc | almost as deep fried as the -dev list <G> |
| 21:30.33 | *** part/#asterisk yojimbo-san (n=CheethJ@120.89.81.19) |
| 21:31.00 | outtolunc | notes sorry, shouldn't bring that up |
| 21:31.02 | spokra | iratik.. now you tell me why it will not work as a callout!! ROLFOL |
| 21:31.18 | iratik | callout? |
| 21:31.21 | iratik | what do you mean? |
| 21:32.16 | spokra | you can create a callout file and make asterisk call you.. and start at a context extension.. it gives an error when you do that!! but not when you call it as an extension from another extension |
| 21:32.35 | iratik | something tells me i know the answer to this |
| 21:32.48 | iratik | switch the roles |
| 21:32.52 | iratik | see if it does the same thing |
| 21:34.34 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 21:36.12 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:36.15 | *** part/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 21:39.01 | *** join/#asterisk golumn (n=golumn@201.220.132.138) |
| 21:41.30 | golumn | I want to replace a meridian system with an asterisk. I have a couple meridian M7100 telephones, which card will I need that recognize does phones? |
| 21:42.44 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 21:44.03 | [TK]D-Fender | golumn, There is no card, but there are expensive gateways. It'd be much more cost effective to replace them entirely |
| 21:45.09 | Katty | hugs [TK]D-Fender |
| 21:45.10 | golumn | so I rather buy new phones |
| 21:47.44 | golumn | [TK]D-Fender, one more thing. I have only try asterisk with SIP lines. I want to connect some normal lines for incoming calls. Where can I find documentation for that |
| 21:48.23 | [TK]D-Fender | Depends how you want "noarmal lines" to come into to *. |
| 21:48.52 | [TK]D-Fender | golumn, most popularly you'd use a PCI type card which is Zaptel compatible |
| 21:49.09 | [TK]D-Fender | golumn, there are plenty of guides out ther based on the card you choose. |
| 21:49.23 | [TK]D-Fender | golumn, and most of this stuff is in the BOOK. |
| 21:50.09 | golumn | thanks. Right now there is a meridian pbx, and the idea is to replace that. Will get some info of the book |
| 21:51.23 | fas3r | i need sip ios for 7942G ... someone ? |
| 21:56.27 | CCFL_Man2 | fas3r: it uses it's own firmware, doesn't use ios |
| 21:57.00 | CCFL_Man2 | you can buy an $8 phone smartnet contract and get access to everything |
| 21:57.09 | fas3r | CCFL_Man2: this : http://tools.cisco.com/support/downloads/go/ImageList.x?relVer=8.3(4)_SR1&mdfid=281346593&sftType=Session%20Initiation%20Protocol%20(SIP)%20Software&optPlat=&nodecount=2&edesignator=null&modelName=Cisco%20Unified%20IP%20Phone%207942G&treeMdfId=278875240&treeName=Voice%20and%20Unified%20Communications&modifmdfid=null&imname=null&hybrid=Y&imst=N&lr=Y |
| 21:57.11 | fas3r | ? |
| 21:57.33 | fas3r | i need to convert my ip phone to sip from sccp |
| 21:57.48 | fas3r | i need to download this no ? |
| 21:58.11 | jaytee | is it just me or do Polycom phones seem to take forever to reboot |
| 21:58.20 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 21:58.47 | fas3r | or it's possible to use sccp directly ... |
| 21:59.02 | fas3r | sorry i start asterisk .. :s |
| 21:59.19 | fas3r | i had read the better is to convert it |
| 21:59.21 | [TK]D-Fender | jaytee, about 2 mins |
| 21:59.35 | SplasPood | jaytee: newer ones seem a bunch faster |
| 21:59.43 | SplasPood | but maybe its all in my head |
| 22:00.42 | fas3r | CCFL_Man2: ? |
| 22:00.56 | jaytee | ok, guess I have to setup provisioning then cuz mine take about 10 minutes or so |
| 22:02.57 | fas3r | ok thanks ;) |
| 22:06.21 | *** join/#asterisk anthm (n=anthm@mb80736d0.tmodns.net) |
| 22:07.03 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 22:08.40 | fas3r | CCFL_Man2: can you explain a little please |
| 22:12.18 | *** join/#asterisk edibrac (n=edibrac3@75.149.50.41) |
| 22:13.05 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 22:14.12 | CCFL_Man2 | fas3r: you are new to this, aren't you? |
| 22:14.22 | fas3r | yes |
| 22:15.37 | fas3r | i'm new |
| 22:16.41 | drmessano | ~cisco |
| 22:16.41 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!, or <reply>Cisco phones are expensive crap which should be avoided with extreme prejudice |
| 22:17.04 | CCFL_Man2 | cisco has sccp or sip firmware for their phones |
| 22:17.44 | CCFL_Man2 | with an $8 phone smartnet contract from cdw, you can download anything from cisco |
| 22:18.24 | fas3r | CCFL_Man2: i have two cisco |
| 22:18.40 | fas3r | one with cable and one wifi |
| 22:18.49 | fas3r | it's not the problem to buy it :) |
| 22:18.59 | *** join/#asterisk RoyK (n=Roy@91.149.38.225) |
| 22:19.00 | CCFL_Man2 | the wifi one does not have any sip firmware for it |
| 22:19.21 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 22:19.50 | fas3r | yes but a ios exist to convert sccp to sip |
| 22:20.52 | drmessano | Not for the wifi one |
| 22:22.12 | CCFL_Man2 | not for the 7920/21 |
| 22:22.27 | CCFL_Man2 | and the phones don't run ios |
| 22:22.41 | fas3r | ha yes that's right |
| 22:22.51 | fas3r | sorry i have just look for 7942G |
| 22:23.00 | fas3r | but i need CCIE access :) |
| 22:23.07 | fas3r | i need to wait tomorrow |
| 22:23.40 | edibrac | wait, i didn't need to $8 contract to download the ..7.x firmware - that was last week |
| 22:23.56 | edibrac | but the 8.x i got from voip-wiki |
| 22:24.07 | fas3r | edibrac: but for the 8.... yes |
| 22:24.13 | edibrac | ah |
| 22:24.34 | fas3r | edibrac: it is on voip-wiki ? |
| 22:24.42 | fas3r | the 8. ? |
| 22:24.47 | edibrac | yeah |
| 22:24.52 | fas3r | erf .. |
| 22:25.02 | edibrac | though i guess, you never know.. there could be some evil modification to it |
| 22:25.34 | fas3r | CCFL_Man2: and it's not possible to use skinny for the 7921 ? |
| 22:26.15 | jaytee | [TK]D-Fender, is this the polycom provisioning tutorial you recommended some time ago? http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+7#7224HowtouseProvisioningCentralBootServe |
| 22:26.27 | [TK]D-Fender | jaytee, looks about right |
| 22:27.36 | jaytee | something munged my Firefox bookmarks about 2 weeks ago and I've been trying to recreate them. I've managed to complete a bunch of goals so my next one is setting up provisioning. |
| 22:28.15 | jaytee | gonna grab some chow, be back later |
| 22:34.43 | stephbul | hello, I want to learn how to generate debian package for asterisk. Do you know a good tutorial about debian/rules for asterisk? |
| 22:40.40 | *** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl) |
| 22:45.30 | *** join/#asterisk unspin (n=unspin@24.82.161.85) |
| 22:48.57 | *** join/#asterisk frieze (n=frieze@pool-71-251-13-242.nycmny.fios.verizon.net) |
| 22:52.33 | puzzled | tzafrir: can I use your 1.4/bristuff-current.tar.gz with plain zaptel 1.4.9.2 for use with a Junghanns OctoBRI? |
| 22:53.35 | tzafrir | puzzled, I have a newer version of that in testing right now |
| 22:53.53 | puzzled | tzafrir: how stable is it? |
| 22:54.02 | tzafrir | but that version should be able to use the octobri, sure |
| 22:54.49 | puzzled | tzafrir: ok, with plain zaptel 1.4.9.2 or do I need to grab your zaptel-xpp? |
| 22:55.26 | znoG | hey, just wondering .. is it possible to check whether an extension is defined if I'm using RealTime? |
| 22:55.37 | znoG | hopefully via some application call |
| 22:57.27 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 22:59.22 | [TK]D-Fender | znoG, its in a DB, I'm sure you can figure out how to query it.... |
| 23:00.29 | znoG | [TK]D-Fender: regardless of the realtime module I'm using? (in my case, ldap) |
| 23:01.29 | [TK]D-Fender | znoG, Do you think *'s core is the only thing that can read LDAP? Keep thinking... |
| 23:01.30 | znoG | [TK]D-Fender: oh, i see what you mean. I'm using LDAP. If I was using a DB, then it's probably a lot easier. |
| 23:01.43 | [TK]D-Fender | znoG, LDAP is a DB |
| 23:01.50 | tzafrir | puzzled, if a specific version has a 'xpp.r<something>' zaptel it is a standard zaptel tarball with a lsightly newer xpp subdir |
| 23:02.03 | znoG | [TK]D-Fender: yes, are you implying I could call an AGI script or something to do it? |
| 23:02.13 | puzzled | tzafrir: ah right. thanks |
| 23:02.24 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:02.40 | [TK]D-Fender | znoG, Why not? |
| 23:02.54 | znoG | [TK]D-Fender: i thought it was a bit overkill but it's not a bad idea. |
| 23:03.02 | frieze | Is beta9 markedly more stable than beta8? I can't seem to get 8 to actually start correctly without tripping over its pid file |
| 23:03.20 | frieze | thinking I should just backup my config files and start over with 1.4 |
| 23:07.06 | frieze | is there an uninstall in the makefile? |
| 23:07.10 | [TK]D-Fender | znoG, what led you to choose LDAP as your realtime backend? |
| 23:08.09 | [hC] | it actually seems quite fitting. |
| 23:11.37 | *** join/#asterisk _MrSeb_ (n=SebaX@87.253.113.240) |
| 23:11.41 | _MrSeb_ | Hi to all |
| 23:12.24 | _MrSeb_ | Someone can say to me how to change the ip that appear in contact info when I'm in debug mode? |
| 23:13.09 | frieze | okay, enough talking to myself. Just installed asterisk 1.4 and when I run /etc/init.d/asterisk I get "Starting Asterisk PBX: Unable to open pid file /'var/run/asterisk.pid': Permission denied asterisk. |
| 23:13.16 | frieze | anyone have any idea what would cause this? |
| 23:15.24 | drmessano | Denied permissions |
| 23:16.20 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 23:16.22 | frieze | right... |
| 23:16.25 | frieze | got that far |
| 23:16.35 | drmessano | Unable to open pid file /'var/run/asterisk.pid': Permission denied asterisk |
| 23:16.47 | hsv-al | hello all, are we all looking forward to another long & glorious night of irc addiction? :) |
| 23:16.59 | hardwire | drmessano: what dist? |
| 23:17.04 | frieze | more trying to understand how asterisk might try to create that pid |
| 23:17.10 | hardwire | oh |
| 23:17.10 | hardwire | haha |
| 23:17.11 | znoG | [TK]D-Fender: well, I already use LDAP to store the employee's accounts .. it seemed logical (except for the fact that 1.6 is beta) |
| 23:17.14 | hardwire | hides |
| 23:17.19 | frieze | ubuntu 64 |
| 23:17.24 | znoG | [TK]D-Fender: but it seems to be working well enough for now |
| 23:17.31 | drmessano | asterisk can't create /var/run/asterisk.pid |
| 23:17.31 | hardwire | frieze: installed asterisk 1.6 from source? |
| 23:17.32 | frieze | though I just got sidetracked into reinstalling zaptel |
| 23:17.47 | frieze | which apparently was renamed yesterday or so |
| 23:17.59 | drmessano | It was? |
| 23:18.01 | frieze | thought it might be a beta bug and so downgraded |
| 23:18.02 | hardwire | yar |
| 23:18.03 | hsv-al | drmessano, thats because the RF is radiation out an isotropic gauge, causing the denial of pid access |
| 23:18.12 | frieze | you'd think so |
| 23:18.13 | hsv-al | w/ sprinkles of neutrino electro radiation |
| 23:18.19 | frieze | the first thing I did was build a faraday cage |
| 23:18.29 | hardwire | frieze: asterisk isn't running as a user that has access to do anything |
| 23:18.32 | frieze | and then put it in a vault deep below an old lead mine |
| 23:18.33 | hardwire | frieze: here's what happened |
| 23:18.38 | hardwire | 1.) you installed asterisk yourself |
| 23:18.43 | hardwire | 2.) you ran it as root, not as asterisk |
| 23:18.51 | hardwire | 3.) it left a bunch of poop on your filesystem |
| 23:18.59 | drmessano | WTF |
| 23:19.01 | frieze | hardwire: that seems quite likely |
| 23:19.03 | drmessano | No |
| 23:19.06 | hsv-al | heh |
| 23:19.10 | frieze | so where do I go from here |
| 23:19.10 | drmessano | STFU dude |
| 23:19.19 | hardwire | 4.) when you try to run it as asterisk it can't do what you want because it can't overwrite the files root created |
| 23:19.27 | hardwire | drmessano: you must be heard? no other opinion valid? |
| 23:19.29 | drmessano | Change the permissions so asterisk can create the pid |
| 23:19.35 | hardwire | been there, done that, took a chill pill |
| 23:19.46 | drmessano | hardwire: You have no idea what youre talking about, apparently |
| 23:19.48 | hardwire | drmessano: teach a man to fish or beat the fish on the head for him? |
| 23:20.28 | *** join/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 23:20.46 | drmessano | hardwire: Your opinion doesn't matter if you are talking out your and not giving him the help he needs, no |
| 23:20.49 | hardwire | frieze: have fun, I'll let the good Dr take it from here - however I'm probably 98% correct |
| 23:21.06 | _MrSeb_ | Someone can say to me how to change the ip that appear in contact info when I'm in debug mode? I've a problem with NAT and the pachet go out with incorrect identification... |
| 23:21.08 | frieze | hardwire: okay now there's a /var/run/asterisk directory and it belongs to asterisk.asterisk |
| 23:21.13 | frieze | same deal |
| 23:21.21 | hardwire | frieze: but the files inside may not |
| 23:21.33 | hardwire | chmod asterisk.asterisk -Rv /var/run/asterisk |
| 23:21.44 | frieze | bermanmk@elwood:/etc/asterisk$ ls -al /var/run/asterisk/ |
| 23:21.44 | frieze | total 0 |
| 23:21.44 | frieze | drwxr-xr-x 2 asterisk asterisk 40 2008-05-28 19:20 . |
| 23:21.44 | frieze | drwxr-xr-x 15 root root 580 2008-05-28 19:20 .. |
| 23:21.50 | drmessano | chown -R asterisk:asterisk /var/run/asterisk |
| 23:21.52 | deeperror | I see ztdummy in lsmod, i've created rooms in meetme.conf, but when calling meetme() in extensions.conf i get no application meetme for extension....what am I missing here? |
| 23:22.01 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 23:22.02 | hardwire | drmessano: day late and a buck short. |
| 23:22.03 | frieze | seems like asterisk should be able to write to it |
| 23:22.18 | hardwire | frieze: it's an empty dir? |
| 23:22.22 | drmessano | hardwire: Yeah, your answer was still wrong |
| 23:22.26 | frieze | yes |
| 23:22.33 | hardwire | drmessano: it probably was very wrong. |
| 23:22.35 | frieze | actually wasn't one there |
| 23:22.37 | frieze | had to make it |
| 23:22.45 | hardwire | frieze: did you use make install? |
| 23:22.50 | frieze | yes |
| 23:22.58 | hardwire | wild man |
| 23:23.00 | drmessano | hardwire: I'll just sit here and continue to let you troll him until an OP steps in |
| 23:23.11 | hardwire | drmessano: an op will stop me from helping? |
| 23:23.19 | drmessano | Oh, thats.. what youre.. ok |
| 23:23.22 | fas3r | the last version of asterisk include sccp or i need to patch it ? |
| 23:23.38 | hardwire | drmessano: I don't get it man.. I *am* being helpful. |
| 23:23.44 | hardwire | you're .. competing? |
| 23:23.51 | drmessano | ha.. ok |
| 23:23.52 | frieze | umm... |
| 23:24.02 | hardwire | everybody welcome to #asterisk, where people fight to freely help you. |
| 23:24.13 | frieze | as fun as this all is, I'd more like to have an answer than resolve a pissing match |
| 23:24.21 | hardwire | I'd like you to have one too. |
| 23:24.30 | frieze | well then two of us are in agreement |
| 23:24.39 | hardwire | frieze: 1.6.0-beta? |
| 23:24.54 | frieze | 1.4.whateversonthewebsite |
| 23:24.54 | deeperror | how do i get the application meetme to work with asterisk? It doesn't seem to be available? |
| 23:25.17 | frieze | I uninstalled 1.6 when it was having this problem and installed 1.4 from source |
| 23:25.18 | hardwire | frieze: on ubuntu 8.04? |
| 23:25.23 | frieze | yes |
| 23:25.44 | hardwire | if you're feeling up to it, install the ubuntu packages and forget about installing from source |
| 23:26.04 | hardwire | otherwise, I have a feeling a lot of files asterisk needs to access are owned by root |
| 23:26.25 | hardwire | I've done that a lot, compile asterisk, install it, run asterisk -cv... and then curse |
| 23:26.50 | drmessano | Most asterisk installs are installed BY ROOT, you then go back and change the permissions, and BAM it works |
| 23:26.57 | frieze | right |
| 23:27.00 | drmessano | You're stating the obvious as being the problem |
| 23:27.11 | frieze | can't see how a non su would install it |
| 23:27.20 | frieze | just trying to figure out what to unfuck |
| 23:27.38 | frieze | not to put to fine a point on it |
| 23:27.49 | drmessano | chown -R asterisk:asterisk /var/run/asterisk and chown -R asterisk:asterisk /var/log/asterisk |
| 23:28.08 | deeperror | how would one compile meetme? |
| 23:28.27 | frieze | incidentally should I install the newly renamed zaptel first? |
| 23:28.28 | hardwire | deeperror: what linux dist and what asterisk version? |
| 23:28.36 | deeperror | centos 1.4.19 |
| 23:28.43 | hardwire | from source? |
| 23:28.46 | deeperror | yes |
| 23:28.55 | drmessano | frieze: Go for it |
| 23:28.56 | hardwire | deeperror got the zaptel modules installed? |
| 23:28.56 | deeperror | ztdummy is listed on lsmod |
| 23:29.02 | hardwire | wow, you're on it |
| 23:29.17 | hardwire | deeperror: do you see a meetme app in your asterisk install dirs? |
| 23:29.21 | deeperror | yea i've been reading a while seems like it should just work but it says no application meetme for extension |
| 23:29.31 | deeperror | what dir ? |
| 23:29.37 | deeperror | i see meetme in /var/spool/asterisk/meetme |
| 23:29.47 | hardwire | that's where it holds temporary data |
| 23:29.55 | deeperror | the source is there in /usr/src/asterisk |
| 23:30.00 | deeperror | is there something to add in modules.... |
| 23:30.21 | deeperror | has load => app_meetme.so |
| 23:30.22 | hardwire | find /var/lib/asterisk | grep meetme |
| 23:30.32 | hardwire | I think that's where apps/chans go, right? |
| 23:30.51 | deeperror | nothing |
| 23:31.00 | hardwire | you sir, broke it. |
| 23:31.09 | hardwire | :) |
| 23:31.16 | deeperror | ha i have a way in doing that? |
| 23:31.18 | hardwire | how did you install from source? |
| 23:31.25 | *** join/#asterisk budol (i=budol@202.124.138.72) |
| 23:31.32 | deeperror | download, make, make install |
| 23:31.38 | hardwire | no configure? |
| 23:31.40 | deeperror | probably |
| 23:31.43 | deeperror | damn |
| 23:31.52 | deeperror | i bet i didn't configure after making ztdummy |
| 23:32.14 | hardwire | libpri first, then zaptel, then asterisk |
| 23:32.22 | hardwire | that's right, right? |
| 23:32.26 | deeperror | is libpri required? |
| 23:32.49 | deeperror | i havent installed that in a while don't think i need it for what i've been running |
| 23:34.55 | deeperror | doh...seems like it may have been the configure |
| 23:35.13 | deeperror | still making but i see meetme compiling |
| 23:37.07 | *** join/#asterisk x_or (n=cdawson@68.178.72.172) |
| 23:37.33 | hardwire | I don't really know, maybe it's not required for simple zt modules. |
| 23:37.33 | hardwire | but chan_zap may require it, and chan_zap may be needed to install zap pseudo channels |
| 23:37.38 | hardwire | deeperror, a self titled album |
| 23:38.12 | deeperror | yea that was it |
| 23:38.21 | *** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk) |
| 23:39.12 | deeperror | yea its working now i installed zaptel later and didnt configure just did the make clean, make make install blah |
| 23:40.21 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 23:40.27 | hardwire | deeperror: please rate my assistance with the channel operators if you found my free and devoted service to be adequate. |
| 23:40.37 | hardwire | come again. |
| 23:40.54 | lanning | you want fries with that? |
| 23:41.07 | hardwire | I had Carl Jr's criss cut fries today |
| 23:41.10 | hardwire | I love those little things. |
| 23:41.21 | NovceGuru | I haven't been to a carl jrs yet |
| 23:41.23 | lanning | :) |
| 23:41.39 | NovceGuru | tried jack in the box for the first time a few weeks ago, A+ fast food right there |
| 23:43.22 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 23:43.30 | frieze | ooh! I go to SF on friday....in n out time |
| 23:43.42 | deeperror | A+++ fast payment will do business with again! |
| 23:44.11 | budol | may i ask a question here about vicidial? |
| 23:44.14 | frieze | forgot all about that |
| 23:47.10 | drmessano | Instead of office chair, package contained bobcat |
| 23:47.19 | drmessano | would not buy again |
| 23:47.38 | drmessano | Best. Ebay. Feedback. Ever. |
| 23:48.14 | *** part/#asterisk x_or (n=cdawson@68.178.72.172) |
| 23:48.34 | deeperror | wasn't there some user that had just a list of crazy feedbacks like that? |
| 23:48.44 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 23:48.46 | drmessano | I dunno |
| 23:49.01 | drmessano | That's from XKCD... the funniest part of the internet |
| 23:51.02 | drmessano | --> //do not crash(); |
| 23:51.10 | drmessano | crap |
| 23:51.16 | drmessano | --> //do_not_crash(); |
| 23:51.30 | coppice | I wonder what EBay Nigeria is like? :-) |
| 23:51.44 | drmessano | Gentoo: Vulnerable to Flattery |
| 23:51.47 | deeperror | http://feedback.ebay.com/ws/eBayISAPI.dll?ViewFeedback2&userid=andy46477&ftab=FeedbackLeftForOthers&page=1&frompage=-1&memberid=andy46477&iid=-1&de=off&items=25 |
| 23:55.25 | disposable | can somebody share their asterisk server's iptables rules with me please? basically a /etc/network/interfaces file from a debian system is what i need. |
| 23:56.00 | fas3r | how to nat sccp with iptables ? |
| 23:56.28 | hardwire | fas3r: haha.. hahahaha |
| 23:56.36 | hardwire | that sounds less fun than the least fun thing I can think of. |
| 23:57.24 | deeperror | disposable: i just deny all allow what is needed |
| 23:57.53 | drmessano | fas3r: You're going to NAT that SCCP phone? |
| 23:58.06 | hardwire | you have an SCCP phone? |
| 23:58.12 | fas3r | 2 |
| 23:58.43 | drmessano | fas3r: Sell them on eBay, get some polycoms, and NAT until you can't NAT anymore |
| 23:58.45 | disposable | deeperror: i know nothing about iptables. i need to enable sip, rtp, rtcp, iax2, ssh, ntp, https. that's it. as to how to do it is beyond me. |
| 23:58.54 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 23:59.12 | drmessano | disposable: Allow everything if the box is firewalled |
| 23:59.23 | drmessano | disposable: Is your internal network really a problem? |
| 23:59.40 | deeperror | i use a rules script with a list of iptable commands |
| 23:59.46 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:59.46 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:59.48 | deeperror | i then just edit that and run it when i need to make changes |