| 00:00.23 | tzafrir_home | I'm trying to figure why I can't manipulate the dialplan... |
| 00:01.25 | saftsack | tzafrir_home, is this a gui for the webbrowser? |
| 00:01.38 | tzafrir_home | yes |
| 00:02.19 | tzafrir_home | Runs from the integrated httpd that was added recently. |
| 00:02.29 | tzafrir_home | Appears to be rewriting configuration |
| 00:02.31 | saftsack | oh ok so asterisk gui is a delivered original webinterface? |
| 00:04.15 | tzafrir_home | well, time to try to figure out javascript code... |
| 00:04.59 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 00:05.03 | saftsack | js -> :( |
| 00:06.39 | *** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 00:07.08 | hoobastooba | i have mentioned this issue before and i have tried a few things that have been suggested, but i am not getting anywhere... |
| 00:07.24 | hoobastooba | i have 5 asterisk servers 4 working very well and one not. |
| 00:07.55 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 00:07.55 | hoobastooba | they are all installed relatively the same, but the one that is not working is acting up when queued calls come in |
| 00:08.07 | kink0 | I am trying to do Dial() to a PSTN phone, and then capture the pressed keys on that phone, any idea ? |
| 00:08.19 | hoobastooba | some calls come in and the queue member cannot hear the caller, but the caller can here the queue member. |
| 00:08.38 | hoobastooba | i have verified all of the hardware settings, irq hyperthreading and such. |
| 00:09.37 | hoobastooba | and have reinstalled everything... still happening. can anyone help? |
| 00:15.06 | *** join/#asterisk MoineauMort (n=pipop@196.203.43.49) |
| 00:15.28 | MoineauMort | hello evry body |
| 00:15.51 | saftsack | hi |
| 00:16.40 | MoineauMort | please i need help can u help me ? |
| 00:17.48 | Strom_C | ~data |
| 00:17.57 | jbot | Don't Ask To Ask. Just ASK |
| 00:17.57 | MoineauMort | im delphi developper |
| 00:18.20 | *** join/#asterisk anthonyl (n=anthonyl@ip70-185-8-160.ma.dl.cox.net) |
| 00:18.33 | MoineauMort | and i cant find iax library for delphi |
| 00:18.51 | R3PTII3 | is there anybody here that knows how to make the asterisk configuration to work with www.nufone.net ? i will pay for a little service if someone can help me |
| 00:18.59 | MoineauMort | have you any idea ? |
| 00:21.30 | *** join/#asterisk Cyon (n=Cyon@cyons.net) |
| 00:23.44 | MoineauMort | please can u help me ? |
| 00:24.08 | Cyon | MoineauMort: With? |
| 00:24.31 | MoineauMort | <PROTECTED> |
| 00:25.30 | Cyon | Going to be beyond me...but you just need to find a lib to include? |
| 00:25.58 | MoineauMort | yes |
| 00:26.12 | *** join/#asterisk KuJaX (n=one@c-67-164-204-41.hsd1.ut.comcast.net) |
| 00:26.13 | MoineauMort | i need to develop a iax softphone |
| 00:26.18 | Cyon | MoineauMort: I assume you reviewed: http://www.voip-info.org/wiki/view/IAXClient |
| 00:26.20 | MoineauMort | with delphi |
| 00:26.39 | MoineauMort | yes but this lib is poure c++ |
| 00:27.28 | anthonyl | can't you load functions exported from shared libraries into delphi somehow? |
| 00:27.56 | Cyon | I'd have thought so to... |
| 00:27.57 | MoineauMort | yes but callback function caused problem |
| 00:28.15 | anthonyl | dlopen (linux) or loadlibrary (win32) something like that should be in delphi |
| 00:28.16 | anthonyl | oh |
| 00:28.31 | *** join/#asterisk alrac (n=carla@12.169.163.241) |
| 00:28.49 | anthonyl | i'm not to informed on delphi, but what kind of error are you getting? |
| 00:29.09 | MoineauMort | my application crashed |
| 00:29.20 | MoineauMort | with error memory fault |
| 00:29.45 | anthonyl | did you check it out with a debugging so you should find the exact point? |
| 00:29.53 | anthonyl | s/debugging/debugger |
| 00:30.44 | MoineauMort | :( |
| 00:30.54 | MoineauMort | this my 5 day |
| 00:30.58 | MoineauMort | for debuging |
| 00:31.01 | MoineauMort | :( |
| 00:31.07 | MoineauMort | and the ocx |
| 00:31.19 | MoineauMort | is not integrable with delphi |
| 00:31.21 | MoineauMort | :( |
| 00:31.35 | Cyon | Why are you using delphi anyway ;-) |
| 00:32.01 | anthonyl | an ocx is just a com object it should be. |
| 00:32.25 | MoineauMort | have you any ocx compatible |
| 00:32.27 | MoineauMort | :) |
| 00:32.39 | MoineauMort | delphi because i metrised him |
| 00:32.42 | MoineauMort | :) |
| 00:35.16 | florz | istn |
| 00:35.22 | florz | gnah |
| 00:38.03 | saftsack | florz, hi are you the hfc patch florz? |
| 00:38.30 | Cyon | Ah well, anyone able to ponder an iax2 issue I'm having on 1.2.13 |
| 00:40.19 | florz | saftsack: yep |
| 00:41.09 | saftsack | florz, :) i tested it one time and it helped :) i ask you as programmer. are the hfc cards good cards from the ground up? |
| 00:42.49 | florz | saftsack: Well, there are several different HFC chipsets - as far as the HFC-S PCI A is concerned, the answer probably would be no ;-) |
| 00:43.48 | saftsack | the hfc chipsets which just do the signalling and do not affect the voice quality (do they???) differ in their quality? |
| 00:44.16 | florz | saftsack: Well, not in the "voice signal quality", obviously :-) |
| 00:44.59 | saftsack | but there is for example the hfc-mini for example too. i thought that an isdn implementation from one manufactor is the same in every chip |
| 00:45.27 | florz | saftsack: Well, the ISDN implementation isn't that much of a problem. It's more the PCI side :-) |
| 00:45.53 | kink0 | I am trying to do Dial() to a PSTN phone, and then capture the pressed keys on that phone, any idea ? |
| 00:46.40 | saftsack | ok ... what is if you have one card in nt and one in te mode? then the isdn hasnt to do anything but to give routing informations if you would connect the cards with cables directly, or? |
| 00:46.54 | florz | saftsack: Like, if it doesn't get PCI access within a few microseconds, it causes an effective xrun, because all the buffering is done in the host's memory, for that's the cheapest way to do things |
| 00:47.54 | saftsack | so this is the reason for echo? |
| 00:48.09 | saftsack | i didnt have see any hfc-a chipset card without having echo troubles |
| 00:48.29 | florz | saftsack: Or the fact that counters that are written to by the card and counters that are written to by the host are within one cache line, which basically makes it impossible to write drivers that work on platforms where cache coherence is not guaranteed by the hardware ... |
| 00:50.09 | saftsack | so you say it is impossible with a normal pc to get an echoless configuration? |
| 00:51.02 | florz | saftsack: ISDN never causes any echo in the voice channel on the transmission path |
| 00:51.57 | saftsack | but if i call somebody who has an analog telephone .... |
| 00:52.08 | florz | saftsack: as far as connecting the cards directly is concerned, there AFAIK is no driver currently that makes use of the chip's capability to move b channel data over a PCM bus |
| 00:52.27 | florz | saftsack: Yeah, then that analog telephone does cause echo |
| 00:52.42 | saftsack | so the only choice to do that is using a hfc4s chipset? |
| 00:53.11 | saftsack | w/ hardware bridgin? |
| 00:53.14 | florz | saftsack: Or more exactly it's the hybrid at the CO end of that analog line |
| 00:53.54 | saftsack | yes but if i plug a telephone directly to the ntba and place a call to anywhere there never is echo |
| 00:54.07 | saftsack | if i call over a te, nt hfc-s a card combination there is echo everytime |
| 00:55.21 | florz | saftsack: That depends on what your intention is - do you wanna actually get rid of the echo or just make it so that it won't be noticed by the user? |
| 00:55.34 | florz | saftsack: Sure, there is echo. You just don't notice. |
| 00:55.43 | saftsack | hrhr :) the echo made me rid :) |
| 00:56.22 | saftsack | the only device with isdn which hasnt echo until now was a patton gateway with echocancelling and a hfc-s mini chipset |
| 00:57.23 | florz | saftsack: The latency of the echo is in the millisecond-range, which is why you don't notice it (your brain simply doesn't distinguish between the echo and the direct sound waves of yoru speech) |
| 00:58.11 | florz | saftsack: Which is why you either have to keep latency that low or you have to actually get rid of the echo (usually using echo cancelling) |
| 00:58.14 | saftsack | yes but why do i experience echo with pci cards? |
| 00:59.11 | florz | saftsack: Well, obviously it must be because of the latency between the to interfaces :-) |
| 00:59.27 | florz | saftsack: which obviously can be kept low using hardware bridging |
| 01:00.13 | florz | saftsack: However, I never had major problems even with software briding between two HFC-S PCI A cards |
| 01:00.32 | saftsack | hmm but why doesnt everyone has problems but me and some other people? |
| 01:00.55 | florz | saftsack: that's a good question :-) |
| 01:01.10 | *** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 01:01.57 | saftsack | florz, there are too people in general. the one who have echo and arent able to get it away and the others who havent any echo. so the conclusion is, that some people are annoyed from little changes in the sound and the others not, or what? |
| 01:02.37 | *** join/#asterisk grantm (n=grantm@207.88.78.2) |
| 01:02.49 | florz | saftsack: I guess it's more like something in your asterisk box causes major delays :-) |
| 01:02.50 | hoobastooba | I came across the error: "Got SIP response 500 "Internal Server Error" back from 10.0.2.119" which is extension 3716. I cannot find in the wiki what this error means. |
| 01:03.39 | hoobastooba | saftsack: look at your irqs. |
| 01:03.51 | hoobastooba | do a top and see what your iowait is, and do sar and see what it has been |
| 01:03.53 | saftsack | i tested it in more than 2 different computer |
| 01:03.53 | florz | saftsack: Do you have an idea how much latency there is between the original and the echo? |
| 01:03.53 | saftsack | s |
| 01:03.58 | *** join/#asterisk quellhorst (n=pro@unaffiliated/rend) |
| 01:04.11 | hoobastooba | i had this issue before and it was an issue with irqs |
| 01:04.14 | saftsack | florz, short. it is more a background voice |
| 01:04.30 | florz | saftsack: Well, "short" is pretty relative :-) |
| 01:04.45 | florz | saftsack: is it, like, 10ms, 100ms, 500ms, 1s, 2s, 5s? |
| 01:04.48 | quellhorst | i have a remote server where i get 19ms pings from, i have a broadband connect here, should that be fast enough to setup asterisk remotely and have a sip phone here? |
| 01:05.07 | hoobastooba | saftsack: what is your iowait reporting when you have this issue? |
| 01:05.36 | saftsack | hoobastooba, i havent got an actual built up. my last tests were 3 months ago |
| 01:05.39 | saftsack | florz, it was about 50ms |
| 01:05.50 | saftsack | but i will build up a new builtup tomorrow |
| 01:05.55 | saftsack | but one theoretical question |
| 01:06.17 | hoobastooba | is this on a t1? |
| 01:06.28 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:06.45 | hoobastooba | sorry, i jumped in after you gave the details probably |
| 01:06.48 | saftsack | if i call home (from the office) both BRI lines there should be no echo implicided by the line if there are just isdn telephones, or? |
| 01:07.42 | florz | saftsack: well, there could be, if the sound get back to the microphone in the handset |
| 01:07.50 | florz | saftsack: But usually, there shouldn't |
| 01:08.28 | saftsack | yes i mean if there will be now poor audio coupling |
| 01:09.04 | saftsack | with this setup it would be possible to test if the * computer creates the echo or not |
| 01:09.06 | florz | saftsack: As I said, within the ISDN transmission path, there is basically nothing that could cause echo on the voice channel |
| 01:09.18 | saftsack | yes this is what i want to test |
| 01:09.51 | quellhorst | ok, from the talk here, 19ms pings to asterisk isnt bad? |
| 01:09.59 | quellhorst | for a server in another city. |
| 01:10.07 | saftsack | but why are there big ECs for T1,E1 cards if isdn signalling doesnt implicit echo? i mean echo from an analog phone from the other site is filterred by the telco, or? |
| 01:10.28 | hoobastooba | anyone tell me what it means if i get Got SIP response 500 "Internal Server Error" back from 10.0.2.119 |
| 01:10.33 | florz | quellhorst: If you mean by that 19ms RTT between the two ends of the call, that shouldn't be a problem |
| 01:11.22 | kink0 | saftsack, always there some audio feedback on the headset from speaker to microphone, and also in the lines due to parasit capacitances in the wires |
| 01:11.33 | florz | saftsack: Nope, it's not, for the very reason that it isn't noticed by anyone anyway because of the small latency |
| 01:11.55 | kink0 | saftsack, that is the reason always there somo class of EC on the telco side, even if you dial from ISDN to ISDN |
| 01:12.04 | Cyon | Just about to head off; no chance anyone is free to consider an issue with iax2 between two 1.2.13 servers; which only existed once both were upgraded, only one server upgraded and there is no issue... |
| 01:12.35 | saftsack | so you mean that it isnt possible to telephone without echo with different partners without having a good ec? |
| 01:12.43 | Cyon | Obviously not excluding human error...just can't figure out how |
| 01:13.17 | Cyon | Ah well, I'll try again another time then |
| 01:14.01 | *** join/#asterisk jaike (i=jaike@58.69.31.44) |
| 01:14.32 | florz | saftsack: You basically do need EC only for either really long distance calls or when using any more-than-PSTN-latency medium, basically. |
| 01:15.04 | saftsack | PSTN-latency medium what is that? |
| 01:15.09 | jaike | is there any advantage of running asterisk on 64bit OS, like fedora 5-64bit? |
| 01:15.35 | florz | saftsack: "more-than-PSTN-latency medium" -> any medium with latency greater than on the PSTN |
| 01:15.54 | saftsack | ok |
| 01:16.25 | saftsack | so if you just place calls to the pstn you need no ec? |
| 01:17.31 | *** join/#asterisk steveaj (n=steve@62.55.147.53) |
| 01:20.00 | florz | saftsack: I mean, just consider that if you take the speed of light as the speed at which the signal travels on the PSTN (which isn't quite correct, of course), it could travel a whole 150 km, that is to a point at 75 km distance and back, in the same time it takes for the sound wave from your mouth to reach your ear |
| 01:20.39 | saftsack | ok this sounds logical |
| 01:21.17 | saftsack | but another question for comparing. does an elmeg tk (this 10 years old big boxes which have 4 s0 modules for example) have echocancelling chips? |
| 01:21.17 | florz | saftsack: So, before it accumulated 10 ms of RTT, it's once accross .de and back :-) |
| 01:22.04 | saftsack | which is the top latency border when humans can hearing echo? |
| 01:23.15 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:23.21 | *** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 01:23.59 | hoobastooba | so has anyone else experienced where the incoming calls are not audible to the agent, but the person on the outside who called can hear everything just fine? |
| 01:24.01 | florz | saftsack: it depends on how loud it is. And being able to hear it actually isn't that much of a problem, either. Only if it's to far from the original, it will stop you speaking and stuff :-) - I mean, you usually do hear yourself while speaking, don't you? :-) |
| 01:24.03 | kink0 | saftsack, that is the reason always there somo class of EC on the telco side, even if you dial from ISDN to ISDN |
| 01:24.07 | kink0 | sorry... |
| 01:24.10 | kink0 | I am trying to do Dial() to a PSTN phone, and then capture the pressed keys on that phone, any idea ? |
| 01:26.23 | florz | kink0: So, in spain you do have EC for every voice channel there is? |
| 01:26.54 | kink0 | florz, yes, we have. Echoes is less noticiable when no long distance and/or very low latency |
| 01:27.04 | saftsack | kink0, so i think that the next EC in the pstn isnt more far than 75km away from me |
| 01:27.20 | saftsack | or what does t-com do in relation to ec, florz ? |
| 01:27.22 | kink0 | but for all calls, gsm net to isdn, isdn to rtb, isdn to voIP... etc is required some class of EC |
| 01:28.07 | kink0 | saftsack, the question is not the distance in kilometers, is the line quality and latency |
| 01:28.07 | saftsack | doesnt the telco do this? |
| 01:28.08 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 01:28.13 | kink0 | saftsack, some times you call from PSTN to PSTN and get a terrible echo |
| 01:28.14 | florz | saftsack: I don't know their policies on this, but they certainly don't do EC on every voice channel - not even for analog lines. |
| 01:28.51 | saftsack | we have an plain old elmeg here and i never had echo before. does the elmeg do echocancelling? |
| 01:30.59 | florz | kink0: so, what exactly does cause latency, if not distance? In the PSTN, that is ... |
| 01:31.38 | florz | GSM and VoIP over ADSL and the like are a different matter, obviously |
| 01:31.46 | kink0 | florz, actually mostly phone circuits are digital, even if you use analog phones |
| 01:32.15 | kink0 | these digital phones circuits does compression, and so, that can cause also delays |
| 01:32.22 | florz | kink0: You mean circuit as in within-the-phone or circuit as in voice channel in the network? |
| 01:32.36 | kink0 | the delay on the line is not very important, because the electrons goes near light speed |
| 01:32.54 | kink0 | circuits on the phone network for compressions and for routing |
| 01:34.16 | florz | kink0: Well, no, the switching employed in the PSTN doesn't have much latency. And the "compression" that's used (G.711a and G.711µ) are a matter of a few nanoseconds to do, with no interdependence of samples. |
| 01:34.47 | florz | kink0: Plus, obviously, the number of switching "hops", also depends on the distance. |
| 01:35.23 | saftsack | florz, do you have a tip for me for a good and cheap ec circuit? |
| 01:35.23 | kink0 | florz I agree, with same latency on the swithes , more distance is more latency |
| 01:35.44 | florz | saftsack: zaptel software EC? |
| 01:35.47 | hoobastooba | quick clearification.... |
| 01:35.53 | kink0 | but considere they use multiple switching circuits, so may be you get less latency in a long distance than in a local call |
| 01:35.55 | hoobastooba | set in sip.conf |
| 01:35.55 | hoobastooba | nat=yes |
| 01:35.56 | hoobastooba | externip="your-public-ip" |
| 01:35.56 | hoobastooba | localnet="internal-network-address"/"internal-subnet-mask" |
| 01:35.56 | hoobastooba | internal-network-address should be the network address like so: |
| 01:35.56 | hoobastooba | localnet=192.168.1.0/255.255.255.0 |
| 01:35.59 | saftsack | florz, the zaptel software ec is a joke ;) |
| 01:36.05 | kink0 | because they chose low latency equipments for long distance |
| 01:36.06 | hoobastooba | local net is the servers address? |
| 01:36.12 | saftsack | isnt there something for the pcm bus for the hfc chipset? |
| 01:36.13 | *** part/#asterisk jaike (i=jaike@58.69.31.44) |
| 01:36.36 | florz | saftsack: No, it works pretty well actually. Just try out different algorithms than the one compiled in by default :-) |
| 01:36.56 | hoobastooba | but if i am not using nat in between my phone and my server, i would not require this, right? |
| 01:37.19 | saftsack | but if i call me home and deactivate the microphone in the phone i shouldnt get echo without an ec, right? |
| 01:37.53 | florz | saftsack: Correct. At least no audible echo :-) |
| 01:38.04 | saftsack | ok then i will test this tomorrow |
| 01:38.08 | saftsack | thank you i go to bed now because its late |
| 01:38.08 | saftsack | gn8 |
| 01:38.18 | florz | indeed, I'm off, too :-) |
| 01:38.21 | saftsack | ok |
| 01:38.23 | saftsack | cYa |
| 01:39.53 | florz | kink0: Well, but instead of the switching latency you will have the light-speed latency, then :-) |
| 01:40.03 | benjk | florz, saftsack seems to be the only one on the planet to have these issues with his BRIstuff setup |
| 01:40.33 | *** join/#asterisk kilobit2001 (n=locid@206-248-176-76.dsl.teksavvy.com) |
| 01:40.44 | kink0 | florz, hehehe, but when you insert an EC into a light speed circuit, the speed goes down again :) |
| 01:41.01 | kilobit2001 | whats ec? |
| 01:41.28 | kink0 | echo cancelers |
| 01:42.42 | benjk | kink0, speed != latency |
| 01:42.49 | florz | benjk: Well, they are at least relatively seldom, yeah. I have heard of such problems once or twice already, but never of any final result as to what the root of the problem was, so *shrug* |
| 01:43.08 | benjk | florz, saftsack has been complaining for at least a year |
| 01:43.43 | benjk | and from his comments I get the impression that he is cheap cheap cheap cheap cheap |
| 01:43.44 | kink0 | well.. I have waste a week trying it, appears to be impossible ... to get DTMF for a call orginated in Asterisk to some phone |
| 01:43.56 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:44.13 | benjk | so I can't avoid thinking that this cheapness has something to do with his echo problem |
| 01:45.31 | florz | benjk: Maybe it was always him? =:-) - well, that is to say: I wasn't really aware of him yet, even though I think I've seen his nick at times ... |
| 01:48.17 | kink0 | nobody know the way to do Dial() to a phone and then capture the pressed keys on it ? |
| 01:49.27 | florz | benjk: @speed/latency: Erm, yeah, obviously speed != latency, but I was just saying that even if you do have faster (as in lower-latency) switches on long distance, you then do have latency because of the distance ... |
| 01:50.07 | benjk | sure latencies add up |
| 01:54.10 | florz | What I still do wonder is how he found out the latency of the voice channel of 50 ms ... =:-) |
| 01:55.36 | kink0 | well time for sleep , good night all |
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| 02:22.01 | R3PTII3 | !seen kink0 |
| 02:25.36 | *** join/#asterisk DasTech (n=DasTech@ppp-71-128-113-209.dsl.irvnca.pacbell.net) |
| 02:25.42 | DasTech | ok hello |
| 02:25.48 | DasTech | need input |
| 02:25.57 | DasTech | got a error I have never seen |
| 02:26.01 | DasTech | Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 200.71.63.179 |
| 02:27.21 | DasTech | the voip gate way unit registers but cant call internaly the get 5 sec dial tone thenit goes busy |
| 02:27.39 | DasTech | and I cant dial them it goes right to vm |
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| 02:54.55 | hellop2 | lo |
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| 04:21.37 | Nivex | It didn't escape the >, this should have been "Why IAX2 > SIP": http://pastebin.ca/225784 |
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| 04:36.50 | callsign | does anyone here use any cisco phones? |
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| 05:27.16 | voipster99 | Good morning everyone ;;; I have got a prob with my asterisk (trixbox) ;;; all a sudden MySQL fails to restart ;;; I have upload g729 and g723 (intel) codecs last night ;; could that be the prob? |
| 05:28.21 | voipster99 | How can I manually start mysql in asterisk? I tried: asterisk -r and then reload |
| 05:28.24 | voipster99 | but still the problem exists |
| 05:28.27 | voipster99 | any ideas guys? |
| 05:29.19 | voipster99 | this is the msg that I get when trying to load FreePBX : [nativecode=Lost connection to MySQL server during query] ** mysql://asteriskuser:amp109@localhost/asterisk |
| 05:29.43 | voipster99 | Are you guys real ppl or bots? |
| 05:32.41 | voipster99 | ??????????????? |
| 05:32.46 | *** join/#asterisk Tond (i=Tond@CPE0015f208f890-CM0010954a5ffa.cpe.net.cable.rogers.com) |
| 05:32.52 | voipster99 | come on |
| 05:33.03 | voipster99 | someone gotta know how to manually restart mysql in asterisk |
| 05:34.41 | Tond | is there a way to change the dialed number parameter in a call when I am sending the call out? meaning if someone had dialed 201 and now I am forwading their call to another number, i want the receiving party to see 444 as the dialed number as suppose to the actual 201 that was dialed |
| 05:35.23 | voipster99 | i dont think any experts answer here |
| 05:36.18 | Tond | u want to restart mysql? |
| 05:36.30 | Tond | but isn't mysql a seperate applicatoin than Asterisk? |
| 05:36.54 | voipster99 | well, it fails when my asterisk starts |
| 05:37.08 | voipster99 | so i am wondering if there is a command within CLI or on linux |
| 05:37.15 | voipster99 | that will get mysql restart |
| 05:37.23 | voipster99 | because FreePBX doesn't function without it |
| 05:41.14 | Tond | i think u need to look at why MySQL is not starting up |
| 05:41.50 | *** part/#asterisk voipster99 (n=voipster@bas12-toronto63-1088794816.dsl.bell.ca) |
| 05:42.06 | Tond | i am not sure if it ha any tied to ur Asterisk or not. but most likely u need to troubleshoot and find out why it is failing to start. read the logs and try to trace the issue |
| 05:51.58 | super_froggy | how to build a dial pattern for my area..? my country in indonesia (+62) area (22) |
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| 06:26.25 | Tond | how do I change the data that is being sent to the CDR database for the field "channel"? |
| 06:26.46 | Tond | I want that field to show the account code information that I am sending instead of the channel name |
| 06:27.21 | Tond | I am right now using 1,Set(Channel="12345") |
| 06:27.27 | Tond | but it is not working... |
| 06:27.35 | docelmo | In the cdr_addon_mysql? |
| 06:27.59 | docelmo | Wow.. Your new to this.. You didnt bother to check the WIKI or anything did you? |
| 06:28.07 | Tond | hrm.. I cna't do it in the extentions file? because for every different number i have a different account code |
| 06:28.22 | Tond | I actually did... Didn't find much |
| 06:28.38 | Tond | I found out how to set the account code and chnage caller id |
| 06:28.39 | Tond | lol |
| 06:28.44 | docelmo | When the account code is set its set in the DB |
| 06:29.08 | docelmo | There is a field called account code |
| 06:29.22 | Tond | I know that, but I want that data to be stored in th src or channel field in the db instead |
| 06:29.23 | Juggie | Tond, you need to do Set(CDR(value)=blah) |
| 06:29.31 | Juggie | where value is some part of the cdr you want to set. |
| 06:29.32 | docelmo | the information is put there. You cant change the channel data by setting ${Channel} |
| 06:29.33 | Juggie | see the wiki |
| 06:29.52 | Tond | oh perfect.. thnaks a lot |
| 06:30.03 | docelmo | That doesnt work for cdr_addon_mysql |
| 06:30.07 | docelmo | Jug.. |
| 06:30.21 | docelmo | I see what you guys are doing down stairs now.. |
| 06:30.28 | Juggie | i'm not downstairs |
| 06:30.41 | Juggie | i just came up 5minutes ago, i need to get shit ready i have to leave @ 4am |
| 06:31.02 | Tond | so doc, what do u recommend then? |
| 06:31.02 | Juggie | are you sure that doesnt work for cdr mysql, it should. |
| 06:31.04 | docelmo | ahh.. I feel ya.. have a safe flight.. I am leaving butt crack of dawn also |
| 06:31.05 | Juggie | its all reading the same data. |
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| 06:31.25 | Juggie | yeah, same to you man. my flight is 6:30am |
| 06:31.27 | Juggie | and its international |
| 06:31.44 | docelmo | Well its pulling information from the channel CDR structure. If the data is changed it may work but I dont believe it does.. I just hacked the hell out of the C code.. |
| 06:32.01 | docelmo | I maybe back down in a bit.. |
| 06:32.13 | docelmo | I dont know for sure.. Who else is down there right now? |
| 06:32.19 | Juggie | i think almost everyone left |
| 06:32.27 | Juggie | the hottie is still there, and maybe a few people with her |
| 06:32.38 | docelmo | ya the alchol nazi's probably showed up |
| 06:32.45 | Juggie | actually they did earlier |
| 06:32.49 | docelmo | haha |
| 06:32.51 | docelmo | they suck |
| 06:32.52 | Juggie | and damin told them we bought the beer from the hotel |
| 06:32.54 | Juggie | and they went away |
| 06:33.00 | docelmo | hah |
| 06:33.07 | docelmo | I love that guy |
| 06:33.11 | Juggie | no one is inbed in this hotel |
| 06:33.16 | docelmo | I am on the 9th floor right now |
| 06:33.23 | Juggie | i hear people running around on the floor above me, below me, etc. |
| 06:33.38 | docelmo | Well we are geeks having fun its the last day for christ sake |
| 06:33.53 | Juggie | too bad they closed the code zone up on us |
| 06:34.14 | docelmo | bitches.. :P |
| 06:34.32 | Juggie | the matrix is on TNT |
| 06:34.40 | docelmo | I dont know.. Hopefully next year we will have absout control of that room next year |
| 06:34.51 | docelmo | I may flip it on.. I am enjoying trance right now |
| 06:34.57 | Juggie | yeah, not having total control sucks |
| 06:34.59 | docelmo | Trance and Mt. Dew.. |
| 06:35.02 | Juggie | this county is fucked |
| 06:35.06 | docelmo | hehe |
| 06:35.07 | Juggie | gotta have a license to drink and shit |
| 06:35.09 | Juggie | whats that about |
| 06:35.15 | Juggie | join a club to drink |
| 06:35.25 | docelmo | Dont have to have a license.. Just proof of age |
| 06:35.27 | Tond | for some reason Set(CDR(clid)=Foo Fighters) |
| 06:35.30 | Tond | doesn't work |
| 06:35.41 | Juggie | what the hell is clid? |
| 06:35.46 | docelmo | I dont know |
| 06:35.50 | Tond | caller id |
| 06:35.56 | docelmo | is that the header name? |
| 06:35.58 | Tond | I am following the wiki |
| 06:36.05 | docelmo | you cant just create shit out of thin air |
| 06:36.07 | Tond | ya.. the variable name |
| 06:36.42 | docelmo | that being the case make it.. "NAME" <123456> |
| 06:36.51 | Tond | i also tried exten => 444,1,Set(CDR(src})=blah) and that didn't work |
| 06:37.00 | Tond | ok, i;ll try that |
| 06:37.02 | Juggie | tond, try w/ just regular cdr |
| 06:37.04 | Juggie | see if thast works |
| 06:37.30 | Tond | what do u mean? |
| 06:37.56 | Juggie | disable cdr_addon_mysql |
| 06:38.01 | Juggie | so it just writes to the flatfile |
| 06:38.08 | Juggie | and see if it works then |
| 06:38.13 | docelmo | flat files fucking suck |
| 06:38.27 | Juggie | oh i know, but i thought you said CDR() didnt work w/ addon_mysql |
| 06:38.35 | Juggie | i cant confirm or deny as i've never used it |
| 06:38.41 | Juggie | so i'm just suggesting you can try that. |
| 06:38.48 | Tond | Ok, just wanted to check and see if i am using the command right |
| 06:38.55 | Tond | ok, thnaks guys |
| 06:39.20 | Juggie | come to think of it |
| 06:39.24 | Juggie | CDR may be only read only |
| 06:39.26 | docelmo | Tond are you trying to dump to mysql right? |
| 06:41.05 | Tond | yes |
| 06:41.24 | docelmo | You have to modify the C code directly.. |
| 06:41.39 | Tond | I already have the CDR wirrting to MySQL and I can pass the accountcode to get sotred there but none of the other fields i can change |
| 06:41.46 | Tond | ah i see |
| 06:42.12 | docelmo | You can not modify CDR shit from the dialplan for mysql_addon_cdr |
| 06:42.27 | docelmo | Trust me I know this for sure.. I know this application very well |
| 06:42.45 | Tond | :) |
| 06:42.54 | Tond | Ok, i'll take your word for it.. ;) |
| 06:42.59 | Juggie | All fields except userfield and account code are read only! |
| 06:42.59 | Juggie | by Eric Lyons on Wednesday 05 of April, 2006 [15:05:32] |
| 06:42.59 | Juggie | Turns out you can't use Set(CDR(<name>)=value) for anything but userfield and accountcode. |
| 06:42.59 | Juggie | These fields are read-only. |
| 06:43.39 | Tond | i see... |
| 06:43.51 | docelmo | So your modifing the code directly.. |
| 06:44.00 | Tond | well thanks anyways for all your help Juggie and docelmo |
| 06:44.08 | docelmo | for a nominal fee I will do it for you |
| 06:44.09 | docelmo | :) |
| 06:44.19 | Tond | ;) |
| 06:44.29 | Tond | I may take ya up on that actually... |
| 06:44.48 | Juggie | i'm curious why you need to modify anything besides the obvious fields |
| 06:44.59 | Tond | I am going to need some modifications done soon, i don't mind doing with people who knwo their stuff and are also willing to help.. ;) |
| 06:45.04 | docelmo | I dont know.. But if you want it done email info@molten.us |
| 06:45.27 | docelmo | Tell them you want to talk to Brian Fertig and I will get in touch with you |
| 06:45.28 | Juggie | tond, why do you need to modify anything besides accountcode or userfield? |
| 06:46.12 | Tond | well, i currently have an interface that shows and can sreach based on some fields that doens't include account code. so if i wanted to pull up a customer's logs and minutes i wont be able to do it. |
| 06:46.29 | Tond | unless i do the search based on accountcode |
| 06:46.33 | Tond | or modify the interface |
| 06:46.35 | Juggie | well, what fields? |
| 06:46.39 | Tond | or do it directly from MySQL |
| 06:47.04 | Tond | time, channel, source, clid, dst |
| 06:47.06 | docelmo | What's your interface coded in? |
| 06:47.11 | Juggie | tond, this is a CDR application that requires the data to be a perticular way? |
| 06:47.20 | Tond | it is php, so it shouldn't be very hard to chnage actually |
| 06:47.31 | docelmo | I can do that in my sleep |
| 06:47.31 | Juggie | tond, you could do that, or i would suggest creating a View |
| 06:47.34 | Juggie | in sql |
| 06:47.48 | Juggie | such that the app gets the data how it expects it |
| 06:48.20 | docelmo | or I can change the query in like < 2 seconds |
| 06:48.26 | Tond | all it is really is an interface |
| 06:48.41 | Tond | lol.. I know.. baby stuff.. |
| 06:48.56 | Tond | maybe it's a good challange for me to get into PHP |
| 06:48.57 | Tond | ha ha |
| 06:49.10 | Tond | but the * source code i wouldn't ever touch |
| 06:49.10 | Tond | lol |
| 06:50.07 | docelmo | asterisk's code is a bitch.. but once you kick it a few thousand times it gets easier |
| 06:50.18 | Tond | ha ha ha |
| 06:50.40 | Juggie | i wish we could spy on the lobby from our rooms like last yuear |
| 06:50.42 | Juggie | *year |
| 06:50.44 | Juggie | that was cool |
| 06:50.47 | docelmo | Im not a expert but I am good enough to get around and get shit done |
| 06:50.54 | docelmo | hehe ya |
| 06:51.12 | docelmo | if you noticed last year all of the numbers for the floors on the elevators disappeared.. :) |
| 06:51.20 | docelmo | Take a wild guess where they went.. :P |
| 06:51.43 | Juggie | haha really? |
| 06:51.45 | Juggie | what happened |
| 06:54.14 | Juggie | heh damn internet |
| 06:54.25 | Juggie | btw, whats up with the phone next to the toilet |
| 06:54.35 | Juggie | and even so, who really wants to touch that thing if it rings |
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| 07:24.11 | stephane_ | jour |
| 07:24.30 | EyeCue | de |
| 07:24.37 | EyeCue | soup |
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| 09:25.59 | L|NUX | !seen mog |
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| 10:17.19 | tdawgpharaoh | My context has this, but my extension does not ring, am I missing something? exten => 41225105016,6,Dial(SIP/2102) |
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| 10:27.49 | tdawgpharaoh | Fixed it, it was extension error |
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| 10:46.06 | Nixis | hi all |
| 10:46.30 | Nixis | could anybody help me ? |
| 10:47.12 | Nixis | where do i rent a dadicated server which is compatible with asterisk i know it's debian but is there a hosting company that sell hosting along with asterisk installation |
| 10:47.14 | Nixis | ?? |
| 10:50.41 | Nixis | anybody here ? |
| 10:50.46 | Nixis | Qwell[], |
| 10:50.49 | Nixis | file, |
| 10:50.51 | Nixis | De_Mon, |
| 10:50.55 | Nixis | denon, |
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| 13:30.06 | file | so... |
| 13:30.09 | file | I can't make it |
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| 13:31.03 | Nixis | file, |
| 13:31.18 | Nixis | you're expert in asterisk ? |
| 13:31.35 | file | some might say that but right now I'm trying to find a way to get home |
| 13:32.28 | file | my flight was cancelled, and no other flights on the same airline can make it for my connection |
| 13:32.32 | file | a connection that only goes once a day |
| 13:32.53 | Nixis | lol :) |
| 13:33.26 | Nixis | ok as an experet do you know any hosting company provide dedicated asterisk server ? |
| 13:33.28 | file | so I will either 1. Be taking another airline and running around, 2. Spending another night in Dallas or 3. Spending a night in Newark |
| 13:33.32 | file | Nixis: nope |
| 13:34.25 | Nixis | well am a telecom consultant and i want to explore asterisk and get my server online do you know anyone could help me install it |
| 13:34.46 | Nixis | i can rent a dedicated server instantly .. |
| 13:34.52 | file | lots of consultants are available, lots are listed on voip-info.org |
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| 13:41.36 | file | looks like I'm in Dallas for another night |
| 13:42.24 | pigpen2 | Could be worse..you could be Houston. |
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| 13:42.50 | file | I think the Astricon crew are still around... |
| 13:42.58 | file | and I need to get my room for another night |
| 13:43.54 | file | I'm on hold with front desk now |
| 13:44.07 | pigpen2 | I am in San Antonio....not in a Hotel though....in my Lazy Boy..... |
| 13:44.44 | file | yay, done |
| 13:45.10 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
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| 13:50.33 | benjk | file, Newark sounds good |
| 13:51.14 | benjk | from there to downtown Manhattan, Ben Benson's steak house, then home next day ;) |
| 14:00.38 | file | okay... hotel is booked for another room, flight is rebooked for tomorrow, internet will run out tonight but I can buy it again... I think I'm set besides food |
| 14:00.54 | file | another room? another night.. |
| 14:02.07 | Nixis | lol file |
| 14:02.12 | Nixis | what is your main job file |
| 14:02.43 | file | I'm a Software Developer for Digium, I've been at Astricon the past week... and would like to fly home but the world hates me |
| 14:03.51 | Nixis | good .. i was working With NexTone & Centile a french IP pbx |
| 14:03.54 | Nixis | i belive you know it :D |
| 14:04.28 | *** join/#asterisk roxy_ (n=roxy_@203.249.97-84.rev.gaoland.net) |
| 14:04.32 | Nixis | how old are you file ? |
| 14:04.57 | roxy_ | can asterisk be use as a server for videoconferencing ? |
| 14:05.53 | roxy_ | or is the question not even relevant ? |
| 14:06.03 | Nixis | well as far as i know roxy_ that there is already efforts to do that |
| 14:06.14 | Nixis | but not yet released offecially |
| 14:06.59 | roxy_ | Nixis: but that would be part of asterisk to do that sort of job ? do you know an open source project that does ? |
| 14:07.47 | roxy_ | I have to implement video conferencing at work and I am just looking for the different option atm. |
| 14:07.48 | Nixis | no no open source in my mind do that :D |
| 14:08.23 | Nixis | cisco avaia :d |
| 14:08.25 | Nixis | :D |
| 14:08.41 | Nixis | roxy_,: will pay a lot of money |
| 14:09.58 | *** join/#asterisk AnAnt (n=anant@62.139.225.218) |
| 14:10.10 | *** part/#asterisk AnAnt (n=anant@62.139.225.218) |
| 14:10.57 | roxy_ | we could pay but I still need to know pro/con compare to asterisk. this page: http://www.voip-info.org/wiki-Asterisk+video made me think that video-conf could be available. |
| 14:15.48 | *** join/#asterisk jeffik (n=Jeff@CABLE-206-188-86-227.cia.com) |
| 14:17.18 | *** part/#asterisk jeffik (n=Jeff@CABLE-206-188-86-227.cia.com) |
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| 14:19.32 | JunK-Y | file: use the force. |
| 14:20.41 | Nixis | lol |
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| 14:32.40 | callsign | anyone here use cisco phones? |
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| 14:41.36 | pifiu | morning everyone |
| 14:42.11 | *** join/#asterisk Arnar (n=arnarb@landi.oddi.is) |
| 14:43.28 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 14:43.55 | robin_sz | hey ... I just completed a guranteed fix for the display blanking problems on the GXP2000 |
| 14:44.27 | robin_sz | all you have to do is: unplug the network cable and connect it to a Snom instead! |
| 14:45.32 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 14:45.39 | MoutaPT | Would it be possible to encrypt extensions.conf in a way only asterisk would be able to read it? |
| 14:47.14 | [TK]D-Fender | MoutaPT : Write a patch to the parser |
| 14:47.31 | [TK]D-Fender | MoutaPT : and remove the "show dialplan" ability |
| 14:48.07 | MoutaPT | yeah i know the show dialpan will expose everyting :) |
| 14:48.38 | MoutaPT | but at first approach my worry is not to expose extensions.conf for a common user managing this system |
| 14:48.48 | MoutaPT | i must say i'm not encrypt expert... |
| 14:50.13 | [TK]D-Fender | MoutaPT : Don't have to be for this. you jsut need to add a single parse line as the line is read in. |
| 14:51.39 | MoutaPT | Sorry if i'm not getting it, i want to prevent a root user to edit this file, like using vi |
| 14:52.37 | [TK]D-Fender | MoutaPT : just scrable the file and add a decode line to the extensiosn.conf parser |
| 14:52.51 | MoutaPT | oh cool |
| 14:52.54 | MoutaPT | now i got it |
| 14:52.55 | tzafrir_laptop | MoutaPT, if asterisk can read it, then root can. No point obfuscating it |
| 14:53.46 | [TK]D-Fender | tzafrir : Just because root can access the file doesn't imply you can't encode it so its not humanly readable |
| 14:53.49 | tzafrir_laptop | BTW: will that user be able to run 'asterisk -rx "show dialplan"' ? |
| 14:53.57 | pifiu | fender, let me ask you something. in my current iax.conf i seem to be using ulaw for in between IAX machines and for incoming and outgoing calls. However I dont have anything enabled saying disallow=all and allow=ulaw. It seems to be picking it automatically? In my new iax.conf I dont have anything either and it seems to be using ulaw in between IAX machines which is fine, but for incoming and outgoing its using GSM |
| 14:54.07 | pifiu | any idea what i could try to force ulaw all across? |
| 14:54.09 | [TK]D-Fender | tzafrir : not after you're done mangling that part of the code :) |
| 14:54.34 | MoutaPT | so if you are providing a demo server, and you want to protect acess to the services your deploy on extensions.conf what would be the best? |
| 14:54.47 | [TK]D-Fender | pifiu : Just set it like normal yourself |
| 14:55.02 | MoutaPT | i know i could exec to get the extensions_my.conf on a remote server on reload |
| 14:55.09 | tzafrir_laptop | what are you trying to protect against? What access level will those users have? |
| 14:55.19 | pifiu | well i tried setting disallow=all and allow=ulaw but then the IAX provider gives me an error saying cannot negotiate codec and doesnt let the call trhough |
| 14:55.24 | MoutaPT | probably would be a root user |
| 14:55.33 | MoutaPT | but as this this a demo |
| 14:55.56 | MoutaPT | i want to prevent the copy of deployed config on extensions.conf, at least i've been asked about this:) |
| 14:55.57 | tzafrir_laptop | pifiu, so it seems that your provider does not support ulaw |
| 14:56.07 | tzafrir_laptop | allow another codec to your provider |
| 14:56.13 | pifiu | whats weird is that with the current setup i dont have that line and its using ulaw, but then i put thew new setup and it uses GSM. so i know for sure the provider does use ulaw |
| 14:56.31 | pifiu | tzafrir but i am currently using ulaw with my provider |
| 14:56.36 | pifiu | all i am changing is the iax.conf |
| 14:56.52 | pifiu | and for some magical reason it wants to use gsm now |
| 14:57.03 | callsign | anyone here have any cisco phones? |
| 14:57.21 | tzafrir_laptop | MoutaPT, again, obfuscating extensions.conf is normally pointless if the user can see the result of Asterisk parsing it. |
| 14:58.42 | *** join/#asterisk R3PTII3 (i=reptile@86.127.95.247) |
| 14:59.05 | MoutaPT | yeah you r right, i'm wondering if i need to get into AGI script... |
| 15:00.40 | [TK]D-Fender | tzafrir_laptop : Well a quick change disables "show dialplan", and just staring at CLI won't give you proper priorities. They'd waste an inordinate mount of time trying to rebuild it. |
| 15:00.50 | MoutaPT | tzafrir_laptop are you used with #exec ? |
| 15:00.57 | MoutaPT | on extensions.conf |
| 15:01.47 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:02.02 | MoutaPT | hi oej! |
| 15:02.21 | oej | hello |
| 15:02.55 | MoutaPT | what's the nome of parser module .c ?:) |
| 15:03.01 | MoutaPT | nome=name |
| 15:06.23 | roxy_ | this page: http://www.voip-info.org/wiki-Asterisk+video make me think that video-conferencing is possible with asterisk.( like 4 people speaking/seeing each other). Am I wrong ? |
| 15:07.49 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.176) |
| 15:09.53 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:14.10 | tzafrir_laptop | MoutaPT, but then again, why give them root in the first place? |
| 15:14.42 | tzafrir_laptop | roxy_, From what I understand: it is not yet possible |
| 15:16.04 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 15:17.01 | MoutaPT | tzafrir, isn't a customer, is a different relationship where if you don't give root, you are already not well seen |
| 15:17.08 | MoutaPT | sorry for my english |
| 15:17.17 | MoutaPT | got what i try to say? |
| 15:17.25 | roxy_ | tzafrir_laptop: ok, thanks |
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| 15:28.59 | callsign | anyone here have any cisco phones? |
| 15:30.00 | tzafrir_laptop | callsign, not me. But ask your question anyway |
| 15:30.56 | tzafrir_laptop | MoutaPT, anyway the config parsing is in pbx/pbx_config.c or something similar |
| 15:31.06 | tzafrir_laptop | that is: pbx.c |
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| 15:32.40 | callsign | im looking for someone that has the cisco cfgmt utility |
| 15:43.40 | Nixis | file, you there ? |
| 15:44.07 | file | sort of, I was downstairs saying goodbye to people who are leaving the hotel |
| 15:44.20 | Nixis | lol |
| 15:44.25 | Nixis | do you have msn ? |
| 15:44.37 | file | yes, but I don't give it out to random people |
| 15:45.26 | Nixis | youre right |
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| 15:45.36 | *** part/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 15:45.41 | file | I'm crazy like that |
| 15:45.41 | Nixis | i just feel irc like street |
| 15:45.41 | *** join/#asterisk yel (n=yassine@xdsl-87-78-35-99.netcologne.de) |
| 15:45.48 | Nixis | chatting :D |
| 15:45.48 | yel | hi everyone |
| 15:46.35 | Nixis | do you do a freelance job file ? |
| 15:46.50 | yel | can i set my asterisk@home to use my modem/fon to call over the net ? |
| 15:47.40 | Nixis | am talking about a project |
| 15:47.45 | file | Nixis: no I do not |
| 15:48.33 | Nixis | ok then i have to it manually :( |
| 15:49.48 | Nixis | i want to setup asterisk box for large amount of subscribers starting from 1000 |
| 15:50.24 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 15:50.30 | *** join/#asterisk blitz[airport] (n=blitzrag@m815f36d0.tmodns.net) |
| 15:50.34 | Nixis | with automated subscription .. for example efonica |
| 15:50.41 | Nixis | with me file ? |
| 15:51.22 | Nixis | but i tried to know if asterik support clustring or not and how to that |
| 15:51.27 | Nixis | could you guide me ? |
| 15:52.08 | blitz[airport] | you can cluster asterisk -- it's not trivial |
| 15:52.31 | blitz[airport] | it's not difficult, but you need to know several parts of asterisk, such as using DUNDi |
| 15:52.54 | Nixis | i asked before blitz[airport] .. but i couldn't get exact answer some say yes some say no |
| 15:53.08 | Nixis | some say you set master box and then childs |
| 15:53.13 | blitz[airport] | you can cluster asterisk -- it depends what you mean by clustering, and what you are trying to accomplish |
| 15:53.27 | blitz[airport] | nah, you distribute with DUNDi and have no central box |
| 15:53.37 | *** join/#asterisk sniffe (n=sniffe@ti211110a081-8052.bb.online.no) |
| 15:53.52 | Nixis | am trying to open a free calling service like efonica |
| 15:54.24 | blitz[airport] | never heard of it |
| 15:56.33 | Nixis | www.efonica.com |
| 15:56.40 | Druken | blitz[airport]: dundi works for outbound, but how would it work for incoming? |
| 15:57.33 | blitz[airport] | Druken: what do you mean? DUNDi is used to find locations and pull information -- its not a VoIP protocol |
| 15:57.47 | Druken | i know... |
| 15:58.20 | blitz[airport] | you're question doesn't make sense to me... |
| 15:58.21 | Druken | it's ment to find routes for termination is it not? |
| 15:58.23 | blitz[airport] | yes |
| 15:58.40 | Druken | so how does it help with clustering for origination ? |
| 15:58.41 | blitz[airport] | dundi lookup extension@mapping |
| 15:58.59 | blitz[airport] | Druken: it doesn't unless you have a cluster of boxes that you send calls to for origination |
| 15:59.06 | sniffe | A <SIP> Asterisk <IAX2> Asterisk <ZAP> B. Now, if A calls B, and B transfers to another Zap channel (or whatever), the channel hungs up. |
| 15:59.20 | blitz[airport] | I just send my calls to a cluster of softswitches |
| 15:59.38 | sniffe | Anyone who can help me out a little? |
| 16:00.27 | blitz[airport] | Nixis: oh, you mean to want to start an ITSP |
| 16:00.29 | Nixis | blitz[airport], cluster of softswitches .... which type you use which is compatible with asterisk |
| 16:00.44 | blitz[airport] | Nixis: any softswitch that can talk SIP is compatible |
| 16:00.59 | Nixis | exactly blitz[airport] |
| 16:01.19 | Nixis | i want to build and ITSP based on Asterisk |
| 16:01.50 | Nixis | i belive it will be a very good case to asterisk :) |
| 16:01.53 | pifiu | what exactly does trunk frequency do? |
| 16:03.27 | Nixis | blitz[airport], : i just need some one to guide me .. specially for large amount of subscribers i dont know which codec should i use .. how to automate subsctibtion on asterisk |
| 16:03.50 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 16:03.52 | blitz[airport] | Nixis: you have a lot of research to do and a lot to learn -- running an ITSP on Asterisk is not trivial |
| 16:04.23 | blitz[airport] | no one will tell you how to do it -- anyone who is doing it well is doing it based on their own experience and testing |
| 16:04.37 | Nixis | blitz[airport], i know .. i was working with NexTone us .. one of the major Softswitches in us .. used by skype .. google .. etc |
| 16:05.10 | blitz[airport] | so to answer your question, the codec you should use is the one that works in the situations you have done testing on, such as with SIPp or some other tools |
| 16:05.23 | Nixis | blitz[airport], i need only head lines and i will start the resarch :) |
| 16:05.53 | blitz[airport] | go start with g.729 |
| 16:06.10 | pifiu | if you dont specify a codec to use in iax.conf how does asterisk decide what to use? |
| 16:06.15 | pifiu | based on what? |
| 16:06.23 | blitz[airport] | pifiu: what is in the [general] section? |
| 16:06.29 | blitz[airport] | that'll be the default |
| 16:06.34 | pifiu | and if it doesnt have anything |
| 16:07.01 | blitz[airport] | depends if the default is to allow everything or not -- why would you not specify? |
| 16:07.04 | pifiu | i only have bindport, language, jitter buffer related stuff, tos, and mailboxdetail |
| 16:07.18 | pifiu | im curious, because right now i dont have it specified, and its using ulaw |
| 16:07.22 | pifiu | i am wondering what makes it decide |
| 16:07.29 | Nixis | blitz[airport], yeah i know about this but am still researching where i can licencsing it unlimited |
| 16:07.35 | blitz[airport] | just the default order in the source code |
| 16:07.44 | pifiu | how would i know waht that is? |
| 16:07.49 | blitz[airport] | Nixis: you can't license it unlimited -- its $10 a license |
| 16:07.56 | blitz[airport] | pifiu: you'd have to read the C code |
| 16:08.01 | pifiu | lol ok |
| 16:08.08 | pifiu | blitz because did you see what was happening to me? |
| 16:08.17 | pifiu | let me retype it |
| 16:08.56 | pifiu | well i tried setting disallow=all and allow=ulaw but then the IAX provider gives me an error saying cannot negotiate codec and doesnt let the call trhough |
| 16:09.40 | pifiu | whats weird is that with the current setup i dont have that line saying which codec to use and its using ulaw, but then i put thew new setup and it uses GSM even though again I am not specifying a codec. so i know for sure the provider does use ulaw since I was using it before |
| 16:10.15 | blitz[airport] | not sure, I don't use IAX2, I use SIP for all my termination, and I'd then tell you to look at the SDP headers |
| 16:10.18 | pifiu | so the provider accepts ulaw on my current setup, without specifying to use it, but on my new setup which is just a cleaned up iax.conf i dont have it specified either and its using gsm |
| 16:11.00 | pifiu | then i tried forcing ulaw and it gave me an error saying cannot negotiate codec, which esentially means they dont support ulaw, but i was just using it! |
| 16:11.41 | Nixis | thanks blitz[airport] :) |
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| 16:18.10 | Druken | blitz[airport]: how long till your flight? |
| 16:18.27 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 16:19.14 | Druken | hungover? |
| 16:19.15 | Druken | hehe |
| 16:19.37 | file | nah |
| 16:19.42 | file | I only had one drink lastnight |
| 16:19.55 | Druken | how big was said one drink? |
| 16:20.01 | file | not that big |
| 16:20.04 | Druken | :) |
| 16:20.16 | *** part/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 16:20.45 | Druken | close the window... won't be so cold :) |
| 16:22.46 | coppice | if you open the window, won't the air con be ineffective? |
| 16:23.28 | Druken | uhmm... it's fall? who said anything about air cond.?? hehehe damn furnace is on here.... |
| 16:23.48 | coppice | the air con is on here |
| 16:24.06 | Druken | coppice: you in the server room? hehehehe |
| 16:24.34 | coppice | just because its fall doesn't mean you are living near the polar bears |
| 16:24.42 | sniffe | Well, I am... :P |
| 16:24.48 | Druken | hehehe |
| 16:25.26 | coppice | its 12:30AM, but still rather warm |
| 16:27.20 | *** join/#asterisk le_nec (n=lldf@99.Red-217-127-136.staticIP.rima-tde.net) |
| 16:27.29 | le_nec | hi |
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| 16:33.43 | le_nec | i'm a newbaby with asterik, i have test trixbox and xorcom, but i'm interested in install asterisk over debian, my question it's if there are a asterisk repository for debian. |
| 16:33.45 | le_nec | thanks |
| 16:35.43 | blitz[airport] | Druken: leaves in another hour and 10 mins |
| 16:38.46 | *** join/#asterisk spunz_ (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 16:38.54 | Druken | you there early or just a stop over? |
| 16:39.15 | blitz[airport] | here way too early, but had to drop Lisa off for her flight |
| 16:39.27 | Druken | ahh... |
| 16:39.39 | le_nec | anybody can help me? |
| 16:40.12 | Druken | le_nec: if someone can help, i'm sure they will... |
| 16:40.23 | blitz[airport] | le_nec: compile asterisk, don't use packages |
| 16:41.07 | le_nec | ok |
| 16:41.56 | *** join/#asterisk lorinc (n=ang@caracas-0905.adsl.interware.hu) |
| 16:42.39 | le_nec | work fine over debian? i read that more people use CentOS, but i prefer debian |
| 16:42.53 | blitz[airport] | I prefer CentOS, but Asterisk doesnt' care what you use |
| 16:44.15 | le_nec | ok |
| 16:44.32 | *** join/#asterisk cian (n=cian@cian.ws) |
| 16:46.15 | le_nec | now i was running xorcom, but i prefer install a clear system, xorcom have more aplications that i don't use |
| 16:46.23 | le_nec | sorry for my english |
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| 16:56.27 | pifiu | how can i diagnose why gsm codec acts like shit? lol |
| 16:56.31 | pifiu | it misses words sometimes |
| 16:56.35 | pifiu | and jitters sometimes too |
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| 17:03.21 | le_nec | bye |
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| 17:45.20 | *** join/#asterisk Math` (n=privmath@modemcable238.43-82-70.mc.videotron.ca) |
| 17:52.33 | *** join/#asterisk ne0phas3 (n=xterm@59.160.164.97) |
| 17:53.18 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:53.32 | ne0phas3 | asterisk suxxx |
| 17:53.37 | ne0phas3 | no security |
| 17:53.39 | ne0phas3 | :) |
| 17:53.43 | ne0phas3 | anyone thr |
| 17:54.58 | anthonyl | i am. |
| 17:55.04 | florz | I am not |
| 17:55.05 | ne0phas3 | yo |
| 17:55.08 | ne0phas3 | :)) |
| 17:55.21 | ne0phas3 | asterisk 1.2.10 suxxx |
| 17:56.05 | sniffe | Informative statement. |
| 17:56.16 | anthonyl | maybe you should try a more recent release of 1.2 |
| 17:56.29 | ne0phas3 | as it said that "outstanding AUTHREQs waiting for replies" that doesn't work |
| 17:56.35 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:56.39 | ne0phas3 | even the 1.4.beta thing |
| 17:56.47 | ne0phas3 | its doesn't work ... |
| 17:56.49 | *** join/#asterisk paolob_ (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:57.01 | ne0phas3 | ne oen can spoof ip and make rougue calls |
| 17:57.10 | anthonyl | ne0phas3, have you put the problems you have been having in a report on the bug-tracker? |
| 17:57.55 | ne0phas3 | how do i do that |
| 17:58.01 | ne0phas3 | i donn know anything abt that |
| 17:58.07 | anthonyl | you do it at bugs.digium.com .. |
| 17:58.27 | ne0phas3 | but then as per my company rules i can't disclose nething to anyoen |
| 17:58.30 | anthonyl | bug submissions about problems in asterisk are always a great help to making things better ;) |
| 17:58.38 | ne0phas3 | and i am new recruit |
| 17:58.48 | anthonyl | hotmail maybe.. |
| 17:58.48 | ne0phas3 | but thr are som much craps in asterisk |
| 17:58.49 | ne0phas3 | :(( |
| 17:59.39 | ne0phas3 | yes will try that ... |
| 17:59.46 | ne0phas3 | but wanted my name on that .... |
| 17:59.52 | ne0phas3 | but i can't use my name |
| 17:59.55 | ne0phas3 | ne idea ?? |
| 17:59.56 | *** join/#asterisk ScurvyDawg (n=scurvyda@S0106000d883f28a0.gv.shawcable.net) |
| 18:00.34 | anthonyl | well you don't have to use your real name on bugs.digium.com to contribute a bug report if you don't want to. |
| 18:06.18 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.190) |
| 18:16.06 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:36.06 | *** join/#asterisk cian (n=cian@cian.ws) |
| 18:36.59 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 18:37.40 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 18:41.57 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:42.00 | PakiPenguin | hello everyone |
| 18:42.56 | ne0phas3 | hello paki |
| 18:48.14 | *** join/#asterisk jimbo- (i=jhio8838@112.sub-75-195-79.myvzw.com) |
| 18:48.31 | jimbo- | does anyone know where I can find the Asterisk realtime queries? |
| 18:49.28 | jimbo- | the actual SQL statements |
| 18:49.47 | *** join/#asterisk soylentgreen (n=fgast@bb1-fe0.only640k.org) |
| 18:50.16 | jimbo- | anyone? |
| 18:51.05 | jimbo- | hello |
| 18:53.26 | ne0phas3 | hi jimbo |
| 18:53.37 | *** join/#asterisk file2 (n=IrcNet@out.clearnet.com) |
| 18:53.37 | *** mode/#asterisk [+o file2] by ChanServ |
| 18:55.57 | pifiu | has anyone ever heard of a command in linux called "npr" or something like that to do a traceroute |
| 18:56.06 | pifiu | like an enhanced traceroute |
| 19:10.24 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:21.08 | *** join/#asterisk effenberg (n=jone@pD9E9D874.dip.t-dialin.net) |
| 19:23.46 | *** join/#asterisk jgoo (n=249fe70a@athedsl-91654.otenet.gr) |
| 19:23.52 | jgoo | hey guys, an OT Q here |
| 19:24.02 | jgoo | any people here AVphiles? |
| 19:24.29 | jgoo | I am setting up a wireless scart to scart based connection from my computer (and thus, asterisk PBX) to my home cinema |
| 19:25.20 | jgoo | I am installing a SIP client on my WIFI'd PDA, I can use it as a handset, or, go 'speaker phone' =] |
| 19:25.31 | jgoo | also, I can watch family guy torrents on my cinema setup... :p |
| 19:25.55 | jgoo | BUT, I think I need a scart adparter (male/female scart, not component) but I cannot find online, dunno if they exist. there, that is all :p |
| 19:41.12 | *** join/#asterisk prttp (i=achi@140.Red-83-38-109.dynamicIP.rima-tde.net) |
| 19:45.19 | *** join/#asterisk edguy3 (n=edguy@host-208-115-200-88.patmedia.net) |
| 20:06.50 | pifiu | should trunk be enabled in type=users or type=peers? |
| 20:06.51 | pifiu | or both? |
| 20:10.32 | *** join/#asterisk jakehow (n=jakehow@user-0cev4b1.cable.mindspring.com) |
| 20:10.59 | jakehow | anyone have experience with Linksys spa941? |
| 20:20.44 | *** join/#asterisk Wazb^ (n=wazb@199.243.74.220) |
| 20:21.48 | *** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net) |
| 20:22.58 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 20:24.35 | Wazb^ | hi |
| 20:25.45 | Wazb^ | i have configured Asteisk with As5300, but some how asterisk is not getting DTMF from As5300. I set AUTO in SIP configuration in * |
| 20:25.59 | Wazb^ | any idea how can i solve this problem? |
| 20:26.07 | Wazb^ | please? |
| 20:28.26 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 20:35.33 | *** join/#asterisk [F] (n=f@pool-70-110-28-229.washdc.fios.verizon.net) |
| 20:35.54 | [F] | Hi guys! |
| 20:36.07 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
| 20:36.12 | kink0 | hello |
| 20:37.17 | kink0 | again my ethernal question, is possible in any way execute Dial() and while the channel is up execute other functions like Read() ? |
| 20:37.59 | *** join/#asterisk Tili (n=tili@202.133.67.65) |
| 20:40.44 | *** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-167-184.columbus.res.rr.com) |
| 20:40.47 | pifiu | how good is g.726 in terms of performance/bandwidth and compatability with *? |
| 20:42.30 | *** join/#asterisk tdawgpharaoh (n=chatzill@196.205.196.1) |
| 20:52.01 | kink0 | I need to do something like Answer() + Background() + Read() + Goto... |
| 20:52.10 | kink0 | but instead Answer, Dial() |
| 20:52.25 | kink0 | the same but originating the call from Asterisk, any sugestions ? |
| 20:52.29 | [Airwolf] | kink0, why would you need to do that ? |
| 20:53.04 | kink0 | [Airwolf], several applications, first I need to originate calls because I Can reach the PSTN at low cost, then do a conference |
| 20:53.37 | kink0 | also, to offer messages awaiting in mailbox to users, originating a call from Asterisk to her/his phones and offering a menu |
| 20:54.10 | kink0 | also, I can after doing this class of call-back, offer tone and transfer call to other extensions |
| 20:54.13 | [Airwolf] | So basicly, what you want is to call local to your * box and then do a conference of a international call ? |
| 20:54.28 | [Airwolf] | or |
| 20:54.34 | [Airwolf] | And other things |
| 20:54.40 | kink0 | [Airwolf], hmmm not exactly, is some class of DISA, but where all calls are originated in my Asterisk |
| 20:55.25 | kink0 | so Asterisk will not receives any call, Asterisk must call first party, sends dialtone and once have gotten the second party phone ussing dtmf, do a transfer. |
| 20:55.39 | [Airwolf] | hmm ok, can |
| 20:55.44 | [Airwolf] | t help you there |
| 20:55.50 | [Airwolf] | sorry |
| 20:55.56 | kink0 | :( |
| 20:56.13 | kink0 | I have waste about a week trying it, but not susccessfull yet |
| 20:58.38 | *** join/#asterisk pder (n=pder@cpe-69-133-88-135.twmi.res.rr.com) |
| 20:59.55 | *** part/#asterisk pder (n=pder@cpe-69-133-88-135.twmi.res.rr.com) |
| 21:07.59 | *** join/#asterisk lorinc (n=ang@caracas-4045.adsl.interware.hu) |
| 21:09.13 | *** join/#asterisk Arnar (n=arnarb@landi.oddi.is) |
| 21:09.58 | kink0 | well, anyway to execute Dial() and then other function , of course no DeadAGI, but when the call is up. |
| 21:15.49 | *** join/#asterisk cian (n=cian@cian.ws) |
| 21:16.38 | *** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il) |
| 21:16.48 | *** join/#asterisk fjean5 (i=fjean5@modemcable131.131-37-24.mc.videotron.ca) |
| 21:17.06 | fjean5 | hi guys |
| 21:18.40 | fjean5 | tell me, what exactly has been added to 1.4 regarding XMPP capability, peering with a jabber server ? |
| 21:18.56 | pifiu | is there any way to see if trunking is actually working? |
| 21:21.42 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 21:23.57 | fjean5 | pifiu: if you have trunk=yes on both sides and you are able to make a call, then it should be ok |
| 21:29.48 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 21:36.06 | *** join/#asterisk lennard (i=lennard@lennardk2.student.ipv6.utwente.nl) |
| 21:36.12 | lennard | whoa, large channel |
| 21:36.19 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 21:36.40 | *** part/#asterisk fjean5 (i=fjean5@modemcable131.131-37-24.mc.videotron.ca) |
| 21:37.20 | Dovid | lol |
| 21:39.18 | mitcheloc | size is relative ;) |
| 21:40.12 | Dovid | but not much talk. sooooo boring |
| 21:41.22 | lennard | shall I try then? |
| 21:41.30 | lennard | why the hell was chan_modem removed? |
| 21:41.43 | mitcheloc | replaced by something better? |
| 21:42.04 | lennard | thus rendering any support for *all* passive cards gone |
| 21:42.27 | lennard | unless I magicly missed some mISDN drivers ofcourse |
| 21:42.39 | lennard | besides, it sortof still works |
| 21:43.25 | Dovid | dont know - never used it - what was it used for ? |
| 21:43.41 | lennard | isdn cards, for one :) |
| 21:43.51 | mitcheloc | hola Dovid, still using snap? |
| 21:43.55 | lennard | I got mine to sortof work |
| 21:44.02 | lennard | I can make a first call to asterisk |
| 21:44.24 | lennard | then something breaks, cause the second doesn't work untill kill -9 and a restart, but hey, it works! :P |
| 21:44.27 | Dovid | yup. - not as much |
| 21:44.47 | Dovid | i am outa the country now - i did tell afew people about it - get any new signups ? |
| 21:45.02 | Dovid | also did u talk about it on the biz list ? i know lots of people would want it |
| 21:45.06 | mitcheloc | every day :) |
| 21:45.50 | Dovid | good |
| 21:46.17 | Dovid | did u create a proxy yet ? |
| 21:46.34 | mitcheloc | working on it |
| 21:46.42 | Dovid | ;) |
| 21:49.29 | Dovid | mitcheloc: what time is it by u guys in the US now ? |
| 21:52.26 | mitcheloc | 2:50 here i think |
| 21:53.43 | Dovid | makes sense 2350 here |
| 21:54.09 | mitcheloc | i'm not much for keeping ontop of what time of day it is, or much less what day of the week, and sometimes what year it is |
| 21:54.18 | mitcheloc | time is relative |
| 21:54.24 | Dovid | hehe. i used to be the same when i worked in IT |
| 21:57.08 | *** join/#asterisk alerios (n=alerios@190.24.99.75) |
| 22:02.40 | yassine | what can be the reason for this ? Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) should i create the file manually ? |
| 22:03.28 | *** join/#asterisk Powerkill (n=Powerkil@84.205.154.179) |
| 22:03.39 | Powerkill | hi |
| 22:03.50 | Powerkill | who have a nokia e61 ? |
| 22:04.53 | Dovid | not me. i heard it isnt too good |
| 22:05.26 | Powerkill | I like it ! It's very good I even manage to register it to asterisk |
| 22:05.42 | Powerkill | now I'm working to register it to SER but I have strange behaviour |
| 22:10.09 | *** join/#asterisk _omer (n=omer@lhr-mp-dig-p11-185.brain.net.pk) |
| 22:10.12 | _omer | hi |
| 22:10.19 | _omer | have a prob |
| 22:10.50 | PakiPenguin | _omer, shoot |
| 22:10.51 | PakiPenguin | :) |
| 22:11.13 | Dovid | which is ? |
| 22:11.26 | _omer | Getting call over a DID number....and then Dialing out through a SIP Carrier...when call is connected I hear the sound for 1 or 2 seconds and then silence....Asterisk is not behind NAT... |
| 22:12.10 | _omer | how do I check what makes the sound pause ... |
| 22:13.38 | Dovid | never had that b4 :( |
| 22:14.09 | Dovid | did u check each did individually to see if its a problem with one of the DID's ? |
| 22:14.49 | _omer | problem is with 7 or 10 calls.. |
| 22:14.53 | _omer | 3 calls work fine.. |
| 22:15.05 | _omer | 7 out of 10* |
| 22:15.08 | Dovid | thru the same DID's ? |
| 22:15.15 | _omer | yep |
| 22:15.15 | Dovid | anything come up in the CLI ? |
| 22:15.21 | _omer | nope |
| 22:15.37 | Dovid | hmm |
| 22:15.45 | Dovid | any firewalls ? |
| 22:16.00 | _omer | nope |
| 22:16.01 | Dovid | or crappy switches ? |
| 22:16.09 | _omer | nope |
| 22:16.38 | Dovid | i heave seen some switches do some stupid shit. simply switching some switches can fix te issues |
| 22:16.57 | _omer | its a dedicated server ..hosted at dellhost.com |
| 22:17.33 | Dovid | ah |
| 22:17.47 | Dovid | try other providers it can be a provider issue - who are u using ? |
| 22:19.28 | _omer | I think its a provider's issue |
| 22:20.08 | Dovid | who's the provider ? |
| 22:20.24 | Dovid | broadvoice ? |
| 22:23.32 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 22:23.32 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:23.50 | _omer | voxbone ... |
| 22:23.54 | _omer | DID provider |
| 22:24.02 | Dovid | and for outbound ? |
| 22:24.08 | _omer | iconnecthere |
| 22:24.18 | Dovid | voxbone has been food for me (although I dont use em a lot) |
| 22:24.30 | Dovid | try a diffrent outbound provider and see what happens |
| 22:24.36 | _omer | is there any other DID provider? |
| 22:24.44 | _omer | a GOOD ONE!!! |
| 22:24.57 | _omer | DIDX.ORG SUCKSSS!!!! |
| 22:24.57 | Dovid | lol |
| 22:25.01 | Dovid | teliax.com |
| 22:25.15 | Dovid | i also use voipjet.com but thier customer service sux |
| 22:26.30 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 22:26.38 | _omer | do they provide DIDs with multiple channels? |
| 22:26.57 | Dovid | well voipjet is outbound only |
| 22:27.07 | Dovid | and teliax u can get out only or both |
| 22:27.26 | Dovid | keep the voxbone and try a diff provider for outbound and see what happens |
| 22:27.46 | MoutaPT | Hi is there someone used with OpenSER+Asterisk solutions? i'm wondering if i should cross calls through asterisk for CDRs or should use something like CDRtool project for CDR on openser? |
| 22:27.52 | _omer | both dont provide dids |
| 22:28.17 | PakiPenguin | _omer, which country dids do u need ? |
| 22:28.51 | _omer | usa,singapore and pakistan |
| 22:29.01 | PakiPenguin | pakistan , i can look into _omer |
| 22:29.05 | Dovid | _omer: for not just get outbound from a new proider and see what happens |
| 22:29.37 | _omer | yep |
| 22:29.42 | _omer | I tried telasip.. |
| 22:29.47 | _omer | same problem.. |
| 22:29.53 | Dovid | hmm |
| 22:29.58 | Dovid | then sounds like a voxbone issue |
| 22:30.06 | _omer | yeah I think so |
| 22:30.08 | Dovid | try teliax - they have been good to me |
| 22:30.17 | _omer | for outbound? |
| 22:30.37 | Dovid | for outbound |
| 22:30.56 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 22:30.59 | Dovid | and if there are no changes then it is voxbone OR the originating provider |
| 22:31.07 | Dovid | voxbone is just a clearing house |
| 22:32.12 | Dovid | yes file: the room is borin tonight |
| 22:33.46 | *** join/#asterisk alerios (n=alerios@190.24.99.75) |
| 22:33.51 | Powerkill | omer ata mehapes misparim ? |
| 22:34.06 | _omer | wat? |
| 22:34.19 | Dovid | Powerkill is speakin hebrew |
| 22:34.20 | PakiPenguin | Powerkill, whats that :p |
| 22:34.32 | Dovid | he asked if ur lookin for numbers |
| 22:34.42 | Powerkill | :) |
| 22:34.57 | Powerkill | sorry I mistake between both of you :) |
| 22:35.01 | Dovid | oper is in PK not Il |
| 22:35.11 | Dovid | IL* |
| 22:35.14 | PakiPenguin | haha i am in pk too :) |
| 22:35.54 | Dovid | Powerkill: ata bi'yisrael ? |
| 22:37.53 | mitcheloc | *ouch* |
| 22:38.42 | *** join/#asterisk R3PTII3 (i=reptile@86.127.17.101) |
| 22:39.01 | R3PTII3 | !seen kink0 |
| 22:40.22 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 22:41.12 | Powerkill | Dovid achav lo aval ani israeli |
| 22:41.20 | Dovid | ah ok |
| 22:41.35 | Dovid | ani amrickai - aval ani bi'yisrael achshav |
| 22:43.18 | Powerkill | ani careka betsarfat |
| 22:43.35 | Dovid | ah - avodah shamah ? |
| 22:45.29 | Powerkill | ken |
| 22:45.48 | Dovid | ah - bi'hatzlachah |
| 22:46.03 | Dovid | avodah bi'asterisk ? |
| 22:46.21 | Dovid | ani michapes avodah ph bi'yisrael im asterisk |
| 22:46.50 | *** join/#asterisk knarfly (n=jdean@c-65-34-177-3.hsd1.fl.comcast.net) |
| 22:49.35 | Dovid | Powerkill: yesh lichah chevrah poh ? |
| 22:53.10 | Powerkill | ken |
| 22:55.57 | Dovid | eizeh ? |
| 22:56.01 | R3PTII3 | !seen kink0 |
| 22:56.12 | Dovid | ani michapes avodah achsha - karegah bi'yishivah |
| 22:56.49 | Dovid | ~seen kink0 |
| 22:57.04 | jbot | kink0 is currently on #asterisk (2h 20m 57s). Has said a total of 13 messages. Is idling for 1h 47m 6s, last said: 'well, anyway to execute Dial() and then other function , of course no DeadAGI, but when the call is up.'. |
| 22:57.04 | R3PTII3 | ~seen kink0 |
| 22:57.07 | jbot | kink0 is currently on #asterisk (2h 21m). Has said a total of 13 messages. Is idling for 1h 47m 9s, last said: 'well, anyway to execute Dial() and then other function , of course no DeadAGI, but when the call is up.'. |
| 23:00.45 | kink0 | jbot: arghhhhhhhhhhhhhhhhhhhhhhhhhhhhh |
| 23:00.52 | kink0 | xDDDDDDD |
| 23:00.59 | R3PTII3 | :) |
| 23:01.12 | kink0 | hey people !!! listen everybody !! I FOUND the way !!! after a week :) |
| 23:01.25 | Dovid | the way to ? |
| 23:01.38 | kink0 | I can execute commands after a Dial ( originating calls from my Asterisk, and then other commands ) |
| 23:02.01 | kink0 | Dovid: I have been worked in do Answer() + Read() + Dial () ... and so... |
| 23:02.13 | Dovid | huh ? |
| 23:02.16 | kink0 | that works fine when calls arrives to my Asterisk, |
| 23:02.34 | Dovid | u figured out how to execute when ur on the phone ? |
| 23:02.42 | kink0 | but never when calls was originated in my ASterisk , because Dial() stops in itselft until hangup. |
| 23:02.55 | Dovid | cool |
| 23:03.01 | Dovid | how did u do it ? |
| 23:03.02 | kink0 | Dovid, yes , while I am on phone call, but when my Asterisk originates the call |
| 23:03.20 | kink0 | is very simple, just use M in your Dial() call |
| 23:03.31 | kink0 | then execute the rest of commands inside a macro |
| 23:03.33 | Dovid | i goto look it up |
| 23:03.43 | Dovid | ah ok. yes i did that |
| 23:04.19 | kink0 | I had waste about a week seeking how to detect pressed keys ( i.e. Read() ) when I originate a call from Asterisk to PSTN |
| 23:04.22 | Dovid | i used it for when a call came in - put it on hold and then dialed some one else - asked them if they wanted the call - if they wanted it, the macro bridgem them if not it sent them to VM |
| 23:04.39 | Dovid | it happens - u bang ur head for a week |
| 23:04.50 | Dovid | did u try asking ur question on the users list ? |
| 23:05.03 | kink0 | yes, I had not tryid with macros for that until tonight, but appears the only way to pass next priorities after a Dial |
| 23:05.15 | kink0 | Dovid, really no, I just asked it here |
| 23:05.34 | Dovid | kink0: the list is a good place for when ur stuck. it has helped me b4 |
| 23:06.39 | kink0 | Dovid, I know, I search for a lot of things in the list, but never I asked on it |
| 23:06.56 | Dovid | u goto start some time - its waste not to use such a good resource |
| 23:07.01 | knarfly | :-) |
| 23:07.09 | knarfly | 8-) |
| 23:07.22 | Dovid | it takes time to learn and then when u know u help others |
| 23:07.32 | kink0 | I agree |
| 23:07.50 | Dovid | i one time had some one help me with a centOS bug in the centos room for 1 1/2 hours for free |
| 23:07.59 | Dovid | that inspired me to sit here and help those that i can |
| 23:08.53 | kink0 | yes, I help for free when I know the answer |
| 23:09.57 | Dovid | :-) |
| 23:10.53 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 23:13.23 | Powerkill | Dovid give me you contact details in Private chat |
| 23:14.45 | Dovid | PowerKill: sent |
| 23:14.50 | Dovid | ~book |
| 23:14.57 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:17.08 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 23:19.19 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 23:20.56 | PakiPenguin | hmmms |
| 23:28.09 | knarfly | ~book |
| 23:28.10 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:30.50 | Dovid | is port forwarding needed for IAX ? |
| 23:31.21 | knarfly | Dovid: if your * server is behind your firewall, yes |
| 23:31.40 | Dovid | okies |
| 23:36.14 | *** join/#asterisk jakehow (n=jakehow@user-0cev4b1.cable.mindspring.com) |
| 23:37.31 | jakehow | anyone have experience with SPA941's? |
| 23:37.58 | Dovid | only 841's |
| 23:38.00 | Dovid | whats the problem ? |
| 23:38.55 | jakehow | Dovid: i have some users who said they could terminate 4 calls on them with their previous provider |
| 23:39.03 | jakehow | even they they have 2line versions |
| 23:39.55 | jakehow | I can do CW at the switch level to get 4 calls on there but is it possible to spread them accross the 4 buttons on the phone? |
| 23:41.24 | hads | You can assign accounts to the buttons, but if you want one account for each you'd need the 4 line version. |
| 23:41.28 | Dovid | CW = ? |
| 23:41.32 | Strom_C | you can have each line appearance on two buttons |
| 23:42.13 | [TK]D-Fender | jakehow : Each button supports 1 call. You can mix and math them according to how many reg licenses you hav (2/4) |
| 23:42.32 | jakehow | what is the difference between an appearance and a call? |
| 23:42.40 | jakehow | i have 2 line license version |
| 23:42.46 | [TK]D-Fender | jakehow : Meaning if you only need 1 reg, you can juggle 4 calls w/o buying the extra reg's |
| 23:42.52 | [TK]D-Fender | jakehow : One & the same |
| 23:43.21 | jakehow | [TK]D-Fender: how do i juggle 4 calls accross all 4 buttons w/ the 2 line version is what im getting at |
| 23:43.29 | hads | You don't. |
| 23:43.30 | [TK]D-Fender | jakehow : Some people to take "appearance" to imply "registration" however |
| 23:43.32 | jakehow | [TK]D-Fender: docs say each line supports 2 appearances |
| 23:43.57 | [TK]D-Fender | jakehow : "2 lines" in the case of the 941 means you can have 2 distinct registrations (different servers). |
| 23:44.27 | hads | Terminology confusion |
| 23:44.44 | hads | There are too many names for the same things. |
| 23:44.46 | [TK]D-Fender | jakehow : Witht aht in mind you can use it for 2 lines of 2 calls each (CW blinks through the next key for each. 1&2 for the 1st, 3&4 for the 2nd |
| 23:45.25 | jakehow | [TK]D-Fender: it will move it onto the next button automatically? |
| 23:45.32 | hads | Yes |
| 23:45.44 | kink0 | tones from Mx are differents , right ? I am unable to Read() it from DTMF |
| 23:45.54 | jakehow | so what happens at the switch level? |
| 23:46.13 | hads | Whatever the switch wants? |
| 23:46.54 | jakehow | hads: do i set button 2 to LIne 1 private, or LIne 1 shared? |
| 23:47.45 | [TK]D-Fender | jakehow : Yes. |
| 23:47.57 | [TK]D-Fender | jakehow : Forget everything but the phone. |
| 23:48.09 | [TK]D-Fender | jakehow : the phone is what controls everything. |
| 23:48.12 | jakehow | [TK]D-Fender: ok |
| 23:48.32 | jakehow | [TK]D-Fender: right now i have these guys in a hunt group |
| 23:48.40 | [TK]D-Fender | jakehow : for your buttons, you want "private" for them. |
| 23:48.44 | jakehow | k |
| 23:49.39 | [TK]D-Fender | jakehow : Thats not a term you should use for that. "Hunt group" refers to a systems whereby someone calling the "primary" # of your busines will loko for the first available line amongst a pool of line to ring. |
| 23:50.17 | [TK]D-Fender | jakehow : Also misappropriated by FreePBX for just ringing multiple phones SIMULTANEOUSLY. |
| 23:50.19 | jakehow | [TK]D-Fender: calls terminate on an 800 number and then all 8 of their phones ring at the same time |
| 23:50.34 | jakehow | they want each individual to be able to pick up up to 4 of these calls |
| 23:51.15 | hads | I can't do four things at once. |
| 23:51.21 | jakehow | right now it only does 2, probably because the 1st line isnt ringing doing CW once there is a live conversation on that phone |
| 23:51.32 | [TK]D-Fender | jakehow : then I might suggest you use 1 "line" using all 4 buttons in "private" mode. That way each subsequent call will just ring through the next available line-key untill they are all full. |
| 23:51.59 | [TK]D-Fender | jakehow : Well it should never "ring", just give a "beep" (maybe a few), and flash on the key |
| 23:52.14 | jakehow | ok |
| 23:52.29 | [TK]D-Fender | jakehow : You neevr want your current caller to know that you're ignoring someone else or feel they are being interrupted even if you don't answer |
| 23:53.00 | jakehow | [TK]D-Fender: they will hear the beep or a little pause in this setup? |
| 23:54.34 | [TK]D-Fender | jakehow : Its typically transparent to the caller esp if they are talking. |
| 23:54.50 | jakehow | ok |
| 23:55.23 | jakehow | so you can have 3 CW calls trunked onto one active line? |
| 23:59.06 | [TK]D-Fender | jakehow : A sloppily worded, but accurate assesment. |
| 23:59.20 | jakehow | [TK]D-Fender: hah.. ok |
| 23:59.28 | jakehow | are there any good references for this sort of info? |
| 23:59.43 | [TK]D-Fender | jakehow : If you wanted real call handling abilities, get a Polycom instead. |
| 23:59.59 | [TK]D-Fender | jakehow : What info in particular? |