| 00:00.55 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 00:01.01 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 00:05.32 | *** join/#asterisk sacitec (n=tobi@189.149.99.172) |
| 00:05.46 | sacitec | hello, good afternoon |
| 00:06.06 | sacitec | what do you recommend for knowing the state of a channel, if it's bussy or not ? |
| 00:06.07 | sacitec | <PROTECTED> |
| 00:11.31 | De_Mon | hehehe |
| 00:11.32 | De_Mon | http://www.saynotocrack.com/index.php/2007/10/28/extreme-hangman/ |
| 00:11.57 | *** part/#asterisk shtoom (n=godson@59.93.114.32) |
| 00:14.18 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-d74f48d27827c383) |
| 00:16.19 | *** join/#asterisk marl (n=marl@78.144.49.39) |
| 00:16.59 | marl | ok can anyone tell me how to disable the 'please leave your name' option within the followme application? only thing i can find is an enable option! |
| 00:21.11 | rhombus | Any Aastra users out there? |
| 00:21.17 | rhombus | I can't get my 480i to register |
| 00:26.05 | PaulAviles | hey guys ,can you use meetme without having any zaptel card? |
| 00:27.22 | flenders | PaulAviles: yes, but you need ztdummy module loaded |
| 00:27.59 | PaulAviles | flenders: thx, does it need to be loaded before you compile or just eh fact that you compile it is enough? |
| 00:28.39 | PaulAviles | yeap ,is compiling it now.... |
| 00:29.14 | flenders | PaulAviles: you need to compile zaptel first |
| 00:29.23 | flenders | PaulAviles: then you compile asterisk |
| 00:29.33 | flenders | PaulAviles: and before you start asterisk, you load ztdummy |
| 00:29.58 | PaulAviles | yeah, I did that, but I guess you also need to have the modules loaded or does not generate the app_meetme.o file. I think is working now |
| 00:30.20 | PaulAviles | lets. see.... |
| 00:30.26 | flenders | good luck |
| 00:35.15 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 00:36.04 | *** join/#asterisk izaak (n=izaak@modemcable132.248-130-66.mc.videotron.ca) |
| 00:38.17 | *** join/#asterisk izaak (n=izaak@modemcable132.248-130-66.mc.videotron.ca) |
| 00:40.00 | PaulAviles | flenders: unable to open pseudo channel -trying device? |
| 00:40.45 | flenders | PaulAviles: where are you getting those? |
| 00:41.09 | PaulAviles | on full when I try do dial into a conferece room |
| 00:41.15 | *** join/#asterisk jakamo (n=gmc@ip24-252-121-125.no.no.cox.net) |
| 00:43.51 | russellb | do you have ztdummy loaded? |
| 00:45.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:45.24 | PaulAviles | yes is zt is loaded |
| 00:46.46 | PaulAviles | what I get in the log when I dial the conference number is no channel type registered for zap |
| 00:48.21 | PaulAviles | then app_meetme.c: Unable to open pseudo channel - trying device |
| 00:48.30 | PaulAviles | and app_meetme.c: Unable to open pseudo device |
| 00:51.24 | PaulAviles | since I have no zaptel hardware, how is the zaptel.conf supposed to look like? |
| 00:55.03 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:56.16 | dlynes_laptop | PaulAviles: empty |
| 00:56.55 | *** join/#asterisk techie (n=techie@adsl-76-214-18-225.dsl.lsan03.sbcglobal.net) |
| 00:57.00 | dlynes_laptop | PaulAviles: pastebin the output of 'lsmodules | grep zaptel' |
| 00:58.15 | PaulAviles | http://pastebin.ca/754475 |
| 00:58.37 | PaulAviles | zaptel.conf should be empty? |
| 01:03.04 | PaulAviles | 'etc/zaptel.conf' is all empty |
| 01:03.09 | PaulAviles | nothing is enabled |
| 01:08.37 | PaulAviles | anyone? |
| 01:10.23 | PaulAviles | I also get in the message "that is not a valid conference number, please try again" and it hangs up |
| 01:12.17 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 01:12.43 | voiper1 | is it possible to have 2 x tdm2460b and then add a tdm03b? |
| 01:12.55 | rpm | by default in the asterisk dialplan if i have an "h" extension, it will always be processed after a call has hung up right? |
| 01:24.50 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
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| 01:44.41 | gallitzin | Any recommended business-class IAX2 capable hardware phones? |
| 01:45.24 | Alowishus | there are very very few IAX phones |
| 01:45.32 | Alowishus | just go with SIP |
| 01:45.36 | gallitzin | Yeah, that's why I'm asking. :) |
| 01:45.43 | Alowishus | heh |
| 01:46.51 | Alowishus | any particular reason you want an IAX hard phone? |
| 01:47.27 | gallitzin | 1 man business without full network control and traveling clients. |
| 01:48.00 | tzanger | gallitzin: I've had *zero* trouble with polycom phones behind nat so long as asterisk was not |
| 01:48.18 | tzanger | I've even got an install with three ip501s behind a standard wrt54 and regular dsl |
| 01:48.18 | *** join/#asterisk BBHoss (n=hoss@146.229.180.198) |
| 01:48.23 | tzanger | zero, and I mean zero, problems |
| 01:48.48 | gallitzin | Often installing Asterisk behind NAT also. :) |
| 01:48.57 | tzanger | gallitzin: I avoid it at all costs |
| 01:49.36 | *** part/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 01:50.34 | J4k3 | hrm, on an asterisk setup, normally, is there any reason to have normal r/w activity to anywhere except /var? |
| 01:50.54 | J4k3 | I'm seriously considering using a USB flash device for boot |
| 01:50.57 | tzanger | offhand I can't think of it |
| 01:51.01 | J4k3 | normally RO |
| 01:51.01 | tzanger | er can't think of any |
| 01:51.14 | tzanger | try it and see where it complains |
| 01:51.17 | Alowishus | config file changes in /etc/asterisk? |
| 01:51.34 | J4k3 | Alowishus: this setup will *very* rarely |
| 01:51.50 | J4k3 | I've been using trixbox for like 6 months and changed settings like 3 times once it was all working |
| 01:52.11 | Alowishus | well yeah by proper *nix standards, then /var and /home are your only real r/w candidates, and /home is pretty much out of the equation with Asterisk |
| 01:52.20 | Alowishus | well and /tmp |
| 01:52.23 | tzanger | I find myself getting very aggravated with trixbox/freepbx |
| 01:52.34 | J4k3 | tzanger: me too, its a piece of shit |
| 01:52.49 | voiper1 | is it possible to have 2 x tdm2460b and then add a tdm03b? |
| 01:52.52 | tzanger | "let's keep config in mysql, except for voicemail, and then use astdb to patch things where needed, but let's not document it all that well" |
| 01:53.17 | tzanger | "oh, and let's add a dozen more config files that are included but are half-generated from db" |
| 01:53.36 | tzanger | I can't fault them too much for it, I mean it's a bit of a pig's ear to try and get it all under one system |
| 01:53.46 | tzanger | but DAMN if I can't get it all to fit in my head |
| 01:54.25 | J4k3 | I haven't done that much research into * directly yet |
| 01:54.49 | rhombus | Any Aastra users out there? |
| 01:54.59 | J4k3 | I realize how crappy trixbox is, and its already saved me a LOT of money, and it works amazingly well on some crappy hardware. |
| 01:55.18 | rhombus | I have discovered that my 480i will not register if I give it a numerical SIP authentication name |
| 01:55.39 | rhombus | I mean, if I give an ALPHABETICAL authentication name :) |
| 01:55.40 | *** join/#asterisk linxroute (n=VietPhon@222.252.108.5) |
| 01:55.44 | rhombus | it registers fine with a number |
| 01:55.50 | rhombus | seems like strange behaviour to me. |
| 01:55.50 | linxroute | hi there |
| 01:56.25 | BBHoss | rhombus: i have all of the aastras using numbers as SIP auth |
| 01:56.37 | rhombus | yeah, I think I am going to have to do that |
| 01:56.59 | rhombus | The Polycom phones don't care, and that's all I've used up until now |
| 01:57.18 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:57.18 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 01:57.24 | rhombus | BBHoss: what firmware are you using? |
| 01:58.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 01:58.07 | linxroute | have anyone used Soundstation 330/320 |
| 01:58.12 | linxroute | Polycom i meantt |
| 01:58.25 | rhombus | linxroute: you mean soundpoint IP :) |
| 01:58.27 | J4k3 | wow, this isn't any good at all |
| 01:58.32 | rhombus | "Polycom Soundpoint IP 330" |
| 01:58.33 | J4k3 | my old trixbox box keeps lagging to the ethernet |
| 01:58.36 | J4k3 | by about 80ms |
| 01:58.38 | linxroute | yeah |
| 01:58.40 | linxroute | sorry |
| 01:58.42 | rhombus | :) |
| 01:58.44 | J4k3 | for 3-20 seconds, regularly |
| 01:58.54 | rhombus | No, I haven't. Only 501 and 650 models |
| 01:58.56 | J4k3 | crap, I'm on the same switch and subnet as it, too. |
| 01:58.57 | linxroute | any good , rhombus |
| 01:58.59 | linxroute | oh |
| 01:59.10 | rhombus | linxroute: well, the Polycom phones are generally good and have a good reputation |
| 01:59.19 | linxroute | they have presence in the phone |
| 01:59.22 | linxroute | which is nice |
| 01:59.30 | rhombus | and Polycom finally fixed their brain-dead PoE implementation in the x30 series phones |
| 02:00.03 | BBHoss | rhombus: not sure |
| 02:00.29 | rhombus | BBHoss: do you tftp provision your Aastra phones? |
| 02:00.29 | linxroute | we have around 15 cisco 1972G , convert those phone to sip just really a pain in ... |
| 02:00.46 | BBHoss | heh cisco firmware for SIP usually blows anyways |
| 02:01.03 | rhombus | BBHoss: and if so, can you fire me a sample cfg file? Aastra doesn't provide any in their firmware packages. |
| 02:01.14 | linxroute | my 7940 after updae to 806 firmware |
| 02:01.20 | rhombus | linxroute: get the Polycom phones |
| 02:01.21 | linxroute | the voice became like a robot |
| 02:01.27 | rhombus | linxroute: you will not regret it |
| 02:01.28 | BBHoss | i am just using the web interface |
| 02:01.40 | rhombus | blast. Okay. |
| 02:02.08 | *** join/#asterisk Entr4nced (n=IMG001@cpe-76-190-132-60.neo.res.rr.com) |
| 02:02.25 | linxroute | soudpoint 320 would be very nice |
| 02:02.39 | linxroute | i'm trying to find a resller in my country |
| 02:03.15 | rhombus | linxroute: where are you? |
| 02:03.18 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 02:03.28 | linxroute | oh far and way |
| 02:03.30 | linxroute | from vietnam |
| 02:03.32 | linxroute | :) |
| 02:03.45 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 02:03.57 | rhombus | linxroute: hmn -- have you checked the Polycom website? |
| 02:03.59 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:04.06 | rhombus | linxroute: You might be able to get them from a supplier in Singapore |
| 02:04.19 | linxroute | yeap but over here they sell it so expensive |
| 02:04.24 | linxroute | like 190$ |
| 02:04.38 | rhombus | linxroute: that's not much more expensive than they are here, actually |
| 02:04.44 | rhombus | linxroute: you could try finding some on Ebay |
| 02:04.50 | linxroute | really |
| 02:04.56 | linxroute | i found some on the web |
| 02:05.03 | linxroute | digiumcars.com |
| 02:05.07 | rhombus | linxroute: We bought all the 501 phones for our office on Ebay |
| 02:05.08 | linxroute | they sell it for around 110 |
| 02:05.18 | linxroute | digiumcards.com |
| 02:05.32 | rhombus | linxroute: but then you have to ship the phones to Vietnam :) |
| 02:05.37 | linxroute | yeahh |
| 02:05.39 | rhombus | and then you will pay the difference in shipping |
| 02:05.44 | rhombus | so it's not as cheap as it looks |
| 02:06.00 | rhombus | and if you have problems with them, it is nightmare to do a return |
| 02:06.08 | linxroute | they claim that they would ship it free |
| 02:06.11 | *** join/#asterisk BBHoss (n=hoss@146.229.180.198) |
| 02:07.38 | linxroute | ebay is really good, i bought some cisco powerinline switch there, ws-c3524-pwr-xl |
| 02:08.25 | BBHoss | how much |
| 02:08.29 | rhombus | You should be able to find Polycom phones very cheap in Vietnam, because you are close to where they are manufactured |
| 02:08.33 | linxroute | 280$ |
| 02:08.55 | linxroute | i belive they are manufacured in thailand |
| 02:09.54 | J4k3 | well, vietnam<->thailand is prolly cheap tarriff too |
| 02:10.03 | linxroute | BBHoss , ws-c3524-pwr only support old cisco phones |
| 02:10.12 | linxroute | it's not really poe |
| 02:10.19 | BBHoss | yeah |
| 02:10.29 | J4k3 | last time I checked, it cost about $2700 to ship a very large intermodal crate from sg to california, US |
| 02:10.39 | J4k3 | $2700 divided across a whole crapload of phones = not much |
| 02:10.46 | J4k3 | its the import tarriffs that kill |
| 02:11.32 | linxroute | for an Soundstation 2 Conference |
| 02:11.55 | linxroute | we can get it around 590$ , but ip phones are expensive , dont know why |
| 02:12.11 | linxroute | maybe this too lilte order |
| 02:13.02 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:27.51 | rhombus | Any Aastra users who are doing TFTP provisioning? |
| 02:29.28 | linxroute | oh rhombus , may i ask you , does polycom phone comes with dialplan function ? |
| 02:29.49 | rhombus | linxroute: that depends on what you mean by that :) |
| 02:29.58 | linxroute | well like in cisco phones |
| 02:30.07 | linxroute | we have a xml file |
| 02:30.39 | linxroute | for the fone which define how the fone should wait |
| 02:30.47 | linxroute | after we press the number |
| 02:31.08 | linxroute | otherwise there's matching number |
| 02:31.14 | linxroute | the fone dial immedialty |
| 02:31.22 | linxroute | instead of pressing # |
| 02:31.26 | linxroute | or Dial |
| 02:31.28 | linxroute | .. to call |
| 02:31.29 | rhombus | linxroute: in the Polycom phones this is called the digit map |
| 02:31.32 | rhombus | and yes, it does support that |
| 02:31.38 | linxroute | kewl |
| 02:31.39 | linxroute | thanks |
| 02:31.43 | rhombus | The Polycom phones are every bit as good as the Cisco phones |
| 02:31.48 | JT | all sip devices have dialplans |
| 02:31.52 | JT | unelss they really suck balls |
| 02:32.04 | JT | polycom phones are better than cisco phones |
| 02:32.05 | rhombus | some things do suck balls, JT |
| 02:32.13 | linxroute | hahaha |
| 02:32.30 | rhombus | The Polycom phones, however, do not. |
| 02:32.52 | rhombus | Disclaimer: i am not employed by Polycom, i only use their phones -- I don't like how they don't make current firmware available to end users |
| 02:33.06 | linxroute | is away |
| 02:33.57 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:37.59 | *** join/#asterisk techie (n=techie@adsl-76-214-18-225.dsl.lsan03.sbcglobal.net) |
| 02:40.13 | De_Mon | rhombus you forgot to complain about the horrible boot time |
| 02:41.00 | rhombus | Oh yeah, thanks -- and they take 14 years to boot |
| 02:41.37 | rhombus | (I am told the x30/x20 series phones are better that way, but I've never tried one) |
| 02:41.45 | rhombus | De_Mon: what sort of phones do you use? |
| 02:42.22 | De_Mon | don't remember 320? |
| 02:42.38 | De_Mon | isnt there a cli command to see the useragent of a phone? |
| 02:42.54 | rhombus | you have to send it a SIP notify, I think |
| 02:43.06 | rhombus | failing that, you can do sip debug ip <ip_address> |
| 02:43.15 | rhombus | or sip debug peer <peername> |
| 02:43.33 | rhombus | and you'll see the useragent headers in the SIP packet |
| 02:44.25 | De_Mon | User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.1.0037 |
| 02:44.35 | De_Mon | yeah thats where i've seen it, debug messages ;) |
| 02:44.49 | rhombus | how long does your 320 take to boot? |
| 02:45.05 | De_Mon | 14 seconds sounds about right, maybe as few as 1 |
| 02:45.07 | De_Mon | err 10 |
| 02:45.22 | rhombus | De_Mon: well, that's a massive improvement, I'd say |
| 02:45.25 | De_Mon | I try not to reboot it often ;) |
| 02:45.31 | rhombus | that's faster than the Aastra phones I'm testing now |
| 02:45.41 | rhombus | and MUCH faster than the 501s we actually use here |
| 02:45.53 | rhombus | which take about 45 seconds to 1 minute to reboot |
| 02:46.02 | JT | the ip301 is the slowest by far |
| 02:46.06 | JT | imho |
| 02:46.12 | JT | slower then 501 |
| 02:46.18 | JT | and the 430 is faster than the 501 |
| 02:46.20 | rhombus | not really surprising, doesn't it have the weakest processor? |
| 02:46.27 | JT | probably |
| 02:46.30 | De_Mon | the linksys sipura 941's boot lightning fast |
| 02:46.44 | rhombus | De_Mon: but they are ugly. |
| 02:46.46 | rhombus | :) |
| 02:46.48 | De_Mon | I haven't really decided which one I like more |
| 02:46.54 | De_Mon | naaa the sipura's arnt that bad |
| 02:47.49 | JT | grandstream and snom take the ugly cake |
| 02:48.07 | De_Mon | the Aastra I used was a pita |
| 02:48.10 | rhombus | JT: I think some would beg to differ with you on the Snom |
| 02:48.39 | De_Mon | come to think of it, I'm not sure why specifically we ditched it |
| 02:48.47 | JT | rhombus: what, the insane people? |
| 02:48.59 | JT | everyone i've spoken to about phones agrees the snoms are ugly |
| 02:49.02 | JT | horribly designed |
| 02:49.04 | rhombus | the European phones are generally more elegant -- somebody actually thinks about the design, and they actually hire industrial designers :) |
| 02:49.06 | De_Mon | oh thats right, it stoped registering with * after we upgraded to 1.4 and I said f'it |
| 02:49.15 | JT | who cares what they hire |
| 02:49.18 | JT | or where they are from |
| 02:49.21 | rhombus | JT: I think that's a cultural difference |
| 02:49.24 | JT | germans can still make ugly shit |
| 02:49.38 | JT | not really, bmws and mercedes cars look nice, snoms do not |
| 02:49.41 | JT | they are ugly |
| 02:49.43 | rhombus | Or Americans can still have no taste. It doesn't seem to have hurt them so far. |
| 02:49.44 | JT | have an awful UI |
| 02:49.50 | JT | i'm not american |
| 02:50.04 | JT | the screens on snoms are archaic |
| 02:50.05 | tzanger | you can't possibly tell me snoms look good |
| 02:50.13 | JT | tzanger: some people really are crazy |
| 02:50.22 | tzanger | I wanted to like snom |
| 02:50.35 | tzanger | but put it beside a polycom and it's no-contest |
| 02:50.42 | tzanger | and the polycoms aren't even all that hot |
| 02:51.00 | rhombus | people have their prejudices |
| 02:51.05 | rhombus | the Europeans think the Polycom phones are ugly |
| 02:51.15 | JT | not most that i've spoken to |
| 02:51.20 | JT | you must be imagining things |
| 02:51.30 | rhombus | The ones I have spoken to do :) |
| 02:51.32 | De_Mon | or talking to different europeans |
| 02:51.34 | rhombus | (this discussion is silly) |
| 02:51.40 | JT | lots of crazy friends |
| 02:51.50 | tzanger | yeah not that I>'ve spoken to either |
| 02:52.02 | [hC] | im not a huge fan of the polycom style |
| 02:52.04 | rhombus | I can tell you guys talk to a lot of Europeans |
| 02:52.12 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 02:52.19 | tzanger | rhombus: I live with a VERY european woman |
| 02:52.31 | rhombus | Polish doesn't count |
| 02:52.43 | tzanger | the type who will quite literally take two purses.. one that is function but won't be seen by anyone, and one to match her outfit |
| 02:52.45 | *** join/#asterisk LakeSolon (n=blake@12-202-201-70.client.mchsi.com) |
| 02:52.47 | tzanger | rhombus: hahaha no not polish |
| 02:53.03 | rhombus | Your woman like phones? |
| 02:53.17 | tzanger | one who, when I came to her house the first time, had her *slippers* match her sweater |
| 02:53.22 | JT | i have a snom 360 sitting here staring at me |
| 02:53.28 | JT | must escape the ugly gaze |
| 02:53.37 | rhombus | That has nothing to do with European and everything to do with woman |
| 02:53.46 | tzanger | this woman takes the concept of form over function to the extreme |
| 02:54.37 | BBHoss | JT: what do you think about the new snom m3? |
| 02:54.43 | JT | haven't seen it |
| 02:54.50 | BBHoss | http://www.snom.com/en/snom_m3.html |
| 02:54.53 | rhombus | It is very sexy |
| 02:55.06 | grimsy | ooh, that is nice |
| 02:55.08 | BBHoss | Innovative both inside and out: snom worked with a renowned designer, known for his work in luxury car phone installations, such as Maybach, Porsche and BMW, to create the snom m3’s modern, sophisticated look. |
| 02:55.10 | rhombus | a damn sight nicer looking than the Kirk or Aastra DECT units |
| 02:55.17 | BBHoss | yes |
| 02:55.22 | tzanger | that ain't bad |
| 02:55.30 | tzanger | too bad it won't have NA DECT |
| 02:55.39 | tzanger | the aastra dect aren't pretty tis true |
| 02:55.40 | BBHoss | yeah they need to start selling them |
| 02:55.43 | rhombus | tzanger: I can tell you talk to a lot of Europeans |
| 02:55.43 | BBHoss | NA DECT? |
| 02:55.49 | JT | looks ok, but it's cordless |
| 02:55.57 | rhombus | tzanger: the Snom M3 does NAm DECT |
| 02:56.01 | JT | rhombus: why do you keep repeating that? |
| 02:56.05 | tzanger | rhombus: really |
| 02:56.10 | rhombus | tzanger: yes, really |
| 02:56.31 | rhombus | Snom's been doing the NAm dog and pony show circuit |
| 02:56.36 | tzanger | BBHoss: north america has different dect frequencies |
| 02:56.49 | rhombus | I've seen all the new sets and the M3 |
| 02:56.57 | rhombus | and they are nice looking phones |
| 02:57.05 | rhombus | they don't feel cheap like much of the competition |
| 02:57.06 | BBHoss | i though it was ~1.9ghz everywhere? |
| 02:57.14 | JT | no. |
| 02:57.19 | rhombus | BBHoss: it is, but the subdivision is different |
| 02:57.25 | tzanger | now the million dollar question |
| 02:57.27 | rhombus | the channels are different |
| 02:57.28 | tzanger | does the m3 have bluetooth |
| 02:57.37 | rhombus | tzanger: i don't know, and I don't care |
| 02:57.48 | [hC] | that snom m3 dect phone will suck |
| 02:57.49 | tzanger | no, of course not |
| 02:58.01 | tzanger | no fucking wifi or dect phone (NA DECT) has bluetooth |
| 02:58.03 | [hC] | its cheap which is nice, but i held it and used it... it feels like a toy. |
| 02:58.14 | BBHoss | too bad |
| 02:58.19 | BBHoss | how much $$$? |
| 02:58.21 | JT | tzanger: err, some dect phones DO have bluetooth |
| 02:58.36 | tzanger | JT: note I said NA DECT |
| 02:58.45 | JT | oh ok |
| 02:58.46 | tzanger | very few, and I mean abysmally few, DECT phones have BT |
| 02:59.13 | [hC] | the aastra dect is not glitzy and shiny plastic with jumper-sized buttons like the snom |
| 02:59.13 | rhombus | Why does it matter? Why does everything have to have Bluetooth? BT is a PITA -- half the time it doesn't work |
| 02:59.20 | rhombus | and it guzzles juice like nobody's business |
| 02:59.22 | [hC] | but they function well, feel good in your hand, etc |
| 02:59.26 | BBHoss | dunno why noone has developed a GOOD DECT phone for USA |
| 02:59.34 | tzanger | rhombus: bluetooth is the bee's knees when it comes to connectivity, IMO |
| 02:59.36 | [hC] | I dont see anything wrong with aastra's dect phone. |
| 02:59.38 | tzanger | it just works by and large |
| 02:59.49 | rhombus | tzanger: well, my experience has been unpleasant |
| 02:59.56 | tzanger | why would you want to have a cordless phone with a goddamned wire tether to your head |
| 03:00.03 | BBHoss | the only problem with aastra is that you have to jump through hoops to buy them |
| 03:00.05 | tzanger | battery life and having a second unit with a battery yes it's a downside |
| 03:00.12 | tzanger | but for 8h use if I can get that I'm happy |
| 03:00.14 | rhombus | bluetooth mice and keyboards that conk out and take forever to get reconnected |
| 03:00.15 | tzanger | BBHoss: yes |
| 03:00.23 | phix | hey |
| 03:00.27 | rhombus | and cellphones that crash when you activate bluetooth |
| 03:00.32 | BBHoss | and you are required to buy a STARTER KIT!! |
| 03:00.35 | [hC] | BBHoss: ah, sorta.. i had to jump through hoops to be a reseller. and at that the hoops were easy as hell. |
| 03:00.44 | [hC] | you're talking about becoming a reseller. |
| 03:00.45 | tzanger | rhombus: the only ones that do that are the treos in my experience |
| 03:00.53 | tzanger | BBHoss: yep that too |
| 03:00.54 | [hC] | I have sold dect solutions to people who had to do nothing but buy them. |
| 03:01.06 | rhombus | my homorola both crashes and won't receive SMS when bluetooth is on |
| 03:01.10 | phix | I am having issues understanding how the transfering calls work |
| 03:01.11 | tzanger | I went to their training, have to buy the starter kit to be a reseller now |
| 03:01.13 | [hC] | -i- had to buy a starter kit, and take qualification tests.. |
| 03:01.15 | tzanger | I'm impressed by and large |
| 03:01.18 | BBHoss | anywhere i can buy some to play with? |
| 03:01.19 | [hC] | I think thats good practice. |
| 03:01.23 | [hC] | i sell them. |
| 03:01.35 | phix | I want to do a blind transfer to a particular extension |
| 03:01.39 | [hC] | I have a demo network set up in my office with two panels and two phones and so far, i have nothing to bitch about. |
| 03:01.59 | BBHoss | panels? |
| 03:02.35 | phix | I have include => featuremap or does it have to be #include features.conf ? |
| 03:02.48 | JT | BBHoss: operator panels |
| 03:02.56 | phix | JT: hi |
| 03:03.10 | JT | hi |
| 03:03.16 | [hC] | panels = the base stations |
| 03:03.21 | BBHoss | thats what i tohugh |
| 03:03.28 | phix | JT: you know how to setup call transfering right? :) |
| 03:03.29 | [hC] | they look like small panel antennas |
| 03:03.29 | BBHoss | thought |
| 03:03.37 | BBHoss | yeah i have been to training too |
| 03:03.47 | BBHoss | just didnt buy the starter kit yet |
| 03:04.02 | rhombus | Are you guys all in Ont? |
| 03:04.09 | phix | who? |
| 03:04.12 | tzanger | I am |
| 03:04.18 | phix | Ont? |
| 03:04.19 | BBHoss | im in Huntsville, AL |
| 03:04.20 | tzanger | BBHoss: where are you at |
| 03:04.22 | tzanger | ah |
| 03:04.29 | rhombus | Where can I do the Aastra training? |
| 03:04.55 | tzanger | rhombus: in ontario? |
| 03:05.07 | BBHoss | its basically you listen to some webinars and read some powerpoints |
| 03:05.16 | tzanger | BBHoss: I went onsite |
| 03:05.18 | rhombus | BBHoss has done it, and he's in Huntsville |
| 03:05.28 | rhombus | I'm in Alberta... |
| 03:05.29 | BBHoss | i had a brief tutorial in Dallas |
| 03:05.32 | tzanger | rhombus: ah |
| 03:05.36 | tzanger | just call your aastra rep |
| 03:05.44 | rhombus | Is it worth doing? |
| 03:06.02 | tzanger | rhombus: I learned a lot, and I've done long-haul wifi links |
| 03:06.03 | rhombus | I expect I'll be using them more. They have a workhorse quality to them |
| 03:06.14 | tzanger | yes |
| 03:06.20 | tzanger | I don't like their deskphones all that much |
| 03:06.27 | phix | JT: ? |
| 03:10.46 | [hC] | rhombus: i am in vancouver bc. i can sell you aastra stuff |
| 03:10.50 | [hC] | i am certified |
| 03:11.12 | rhombus | I don't need a supplier, I have one -- I'm curious about the training, though. |
| 03:11.41 | [hC] | oh. |
| 03:11.49 | [hC] | yeah just contact aastra, they'll hook you up |
| 03:12.43 | voiper1 | <PROTECTED> |
| 03:13.51 | phix | do I need to define the blindxfer number in extensions.conf? |
| 03:19.11 | rhombus | Well, I think I've pretty much confirmed it |
| 03:20.34 | rhombus | wait, I was wrong |
| 03:20.36 | rhombus | it's working |
| 03:20.40 | rhombus | IT'S A MIRACLE |
| 03:21.26 | phix | hello |
| 03:23.34 | rhombus | phix: hello. |
| 03:23.40 | rhombus | phix: how are you this fine evening? |
| 03:24.31 | phix | trying to get call transfering working |
| 03:25.06 | phix | rhombus: you know anything about it? |
| 03:25.24 | rhombus | I transfer calls sometimes, yes. What specifically are you struggling with? |
| 03:25.59 | phix | transfering calls :) |
| 03:26.09 | rhombus | can you be more specific? there's lots of ways to transfer calls |
| 03:26.10 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 03:26.29 | phix | I include featuremap and parkedcalls, I set blinxfr to *2 and attrxfr to #2 |
| 03:26.40 | phix | while in a call none work |
| 03:26.50 | phix | I am calling out of my VoIP trunk |
| 03:27.00 | phix | does this need to have tT defined in Dial()? |
| 03:27.11 | phix | or only in my outgoing SIP user context? |
| 03:28.05 | phix | do I need to define *2 and #2 in my extension.conf? or just include => featuremap? |
| 03:28.12 | *** join/#asterisk chinku (n=chinku@c-68-37-156-186.hsd1.nj.comcast.net) |
| 03:28.25 | [hC] | you need it in Dial() for the blindxfer and attendedxfer to work |
| 03:29.08 | rhombus | phix: what kind of phone are you using? |
| 03:29.13 | phix | softphone |
| 03:29.17 | phix | X-Lite |
| 03:29.25 | rhombus | phix: does it not support transfer natively? |
| 03:29.29 | fujin_ | why not use the built in transfer functionality? |
| 03:29.35 | rhombus | exactly my point |
| 03:29.41 | fujin_ | yucky @ blindxfer etc. |
| 03:29.48 | phix | [hC]: so my outgoing context needs the tT in dial? |
| 03:29.54 | fujin_ | native SIP reinvite transfers are much better |
| 03:29.59 | fujin_ | well, work; :) |
| 03:30.00 | rhombus | the features.conf stuff is mostly for dumb phones that don't have the features integrated, say a Zap phone |
| 03:30.00 | phix | fujin_: because that is not available in free vers |
| 03:30.25 | phix | I will need to support zap phone in the future :) |
| 03:30.29 | phix | so either way I would like this to work |
| 03:30.38 | [hC] | phix: that allows the caller and callee to both be able to xfer. you probably want to limit it to just t |
| 03:30.40 | [hC] | i think. |
| 03:30.56 | [hC] | but yeah, builtin to phone stuff works better in alot of casess |
| 03:31.01 | phix | yep, show application dial tells me t ;) |
| 03:31.12 | rhombus | that's because SIP doesn't transmit keypresses the way an analog phone does |
| 03:31.24 | rhombus | it's out-of-band |
| 03:31.34 | rhombus | so you can't expect the phone to behave the way a regular one does |
| 03:31.37 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 03:31.38 | phix | [hC]: what sip phones do you recommend for this feature? I was thinking on getting a linksys, although I cant find one with a XML/HTML browser inbuilt |
| 03:31.56 | rhombus | and that's why features.conf based transfer functions are a royal pain to maintain, because every phone has to be handled differently |
| 03:32.00 | rhombus | the Aastras are a case in point |
| 03:32.03 | fujin_ | rhombus: not always out-of-band. |
| 03:32.11 | fujin_ | I run all of my hardphones DTMF inband |
| 03:32.13 | rhombus | Generally out of band. |
| 03:32.14 | fujin_ | easier to manage |
| 03:32.17 | phix | rhombus: ah ok :) so is there another way to get SIP to bind xfer? |
| 03:32.26 | phix | without using invite that is :) |
| 03:32.31 | rhombus | Well, the integrated SIP function keys :) |
| 03:32.43 | phix | rhombus: ssshh, besides that :) using features.conf |
| 03:32.44 | fujin_ | change your dtmf mode to inband, then the blindxfer stuff in asterisk will recognize it |
| 03:32.48 | rhombus | You could try doing what fujin does and set the phone to do inband DTMF |
| 03:32.55 | phix | or perhaps you would recommend a free SIP softphone? |
| 03:33.00 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:33.03 | fujin_ | Ekiga |
| 03:33.05 | rhombus | have you tried Zoiper? It does SIP now. |
| 03:33.10 | phix | rhombus: hmmm |
| 03:33.11 | fujin_ | or htat |
| 03:33.17 | phix | rhombus: it uses rfc2something or other atm |
| 03:33.20 | [hC] | phix: polycom or linksys is good. |
| 03:33.20 | fujin_ | or buy a hardphone? ;] |
| 03:33.30 | [hC] | phix: or aastra. |
| 03:33.36 | phix | [hC]: they have minibrowsers/ |
| 03:33.47 | phix | or support SOAP? :) |
| 03:33.59 | [hC] | depends how much you can pay |
| 03:34.03 | fujin_ | dude, they're hardphones |
| 03:34.04 | fujin_ | -_- |
| 03:34.11 | [hC] | polycom and aasta both do |
| 03:34.12 | phix | my VoIP provider offers an API for its web based system, so centralised address book etc |
| 03:34.16 | [hC] | aastra has GREAT xml support |
| 03:34.22 | [hC] | the 57i or 480i are good choices. |
| 03:34.25 | [hC] | depending on the style you want. |
| 03:34.25 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:34.26 | phix | [hC]: how are they on the budget? |
| 03:34.32 | fujin_ | ~cheap |
| 03:34.32 | jbot | extra, extra, read all about it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 03:34.40 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 03:34.44 | [hC] | gotta run.. look up aastra 480i, 57i, and polycom ip550 |
| 03:34.48 | phix | hehe ok |
| 03:34.48 | phix | thnx |
| 03:36.24 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.248) |
| 03:36.48 | rhombus | fujin_: thanks -- I'm going to remember that! |
| 03:36.54 | rhombus | ~cheap |
| 03:36.55 | jbot | somebody said cheap was a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 03:36.58 | phix | ok it is T I want, not t |
| 03:36.58 | fujin_ | ;] |
| 03:37.10 | phix | hehe |
| 03:37.29 | phix | $298 seems ok to me :) I just didnt want to pay $400+ for a SIP phone |
| 03:38.13 | phix | $268 even |
| 03:38.27 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:39.01 | rhombus | phix: I bought the 501s we are using in our office for 110 each on Ebay |
| 03:39.30 | rhombus | and these Aastra 480i's were 120 each, new |
| 03:39.39 | rhombus | A good SIP phone doesn't have to be exorbitant |
| 03:39.56 | JT | that's not really an option for a corporation or professional rollout though |
| 03:40.15 | rhombus | JT: what's not? |
| 03:40.24 | neax | good afternoon, ladies and gentlemen |
| 03:40.55 | JT | rhombus: buying surplus phones on ebay |
| 03:41.16 | rhombus | JT: Well, as a general guideline, I'd agree -- but it's hardly a hard rule |
| 03:41.34 | rhombus | JT: Sometimes you can get hundreds of sets in excellent condition |
| 03:41.53 | rhombus | JT: and you'd be surprised at how many corporate clients are okay with that |
| 03:44.46 | *** part/#asterisk xeraa (n=philipp@M950P018.adsl.highway.telekom.at) |
| 03:45.52 | JT | rhombus: it really depends how big or stingey they are |
| 03:47.51 | rhombus | Anyway, phix clearly has a budget pressure problem |
| 03:48.05 | rhombus | so I'm giving him advice that fits phix's circumstances |
| 03:48.18 | rhombus | or her, as the case may be :P |
| 03:48.23 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:52.57 | phix | rhombus: USD or AUD? |
| 03:53.13 | rhombus | phix: USD |
| 03:53.16 | rhombus | phix: are you in Oz? |
| 03:53.36 | rhombus | phix: actually, it was CAD, but now 1 CAD = 1.05 USD :P |
| 03:53.46 | phix | yes |
| 03:54.00 | phix | haha, the USD is shit now :) |
| 03:54.13 | phix | either that of everyone else has got better :) |
| 03:54.48 | phix | 1 AUD buys 0.99 USD I think, well last time I checked anyway |
| 03:55.25 | fujin_ | 1nzd=77.5c usd |
| 03:55.27 | fujin_ | stil pretty shit here |
| 03:55.28 | phix | although have retail stores in the US put their price up because of this? |
| 03:55.46 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-69-66.dsl.irvnca.pacbell.net) |
| 03:56.02 | phix | fujin_: :) That's because all you have to export is sheep :P |
| 03:56.14 | fujin_ | Oh wait |
| 03:56.41 | phix | All building supplies need to be imported, there are no mines in NZ right? |
| 03:56.50 | fujin_ | I don't know. |
| 03:56.57 | fujin_ | I get money, I buy shit |
| 03:57.02 | fujin_ | care factor is low where it comes from |
| 03:57.35 | phix | Well yeah the care factor will remain low until you want to buy a house :) |
| 03:57.58 | fujin_ | Not really |
| 03:58.02 | fujin_ | I'll buy a house, and be done with it |
| 03:58.12 | fujin_ | Seriously, I get money |
| 03:58.17 | JT | phix: what are you talking about? there are trees in new zealand |
| 03:58.21 | fujin_ | Indeed. |
| 03:59.04 | phix | yeah I supose :) |
| 03:59.14 | phix | JT: Trees yes, Steel no |
| 03:59.24 | phix | well mines no even |
| 03:59.27 | fujin_ | last time I checked, we don't build houses with steel |
| 03:59.28 | phix | or am I just being ignorant? |
| 03:59.35 | phix | any way :) back to asterisk |
| 03:59.39 | JT | probably, china imports a lot of steel too |
| 03:59.46 | phix | changing dtmfmode to inband didn't work |
| 03:59.50 | phix | no transfer love yet :( |
| 03:59.52 | fujin_ | That's *way* below my abstraction layer. |
| 04:00.04 | fujin_ | phix: did you set the dtmf mode in your sip.conf, too? and in the client? |
| 04:00.13 | puzzled | JT: afaik China consumes by far the most steel in the world |
| 04:00.27 | JT | yes |
| 04:00.28 | phix | fujin_: in sip.conf it is under [general] |
| 04:00.36 | JT | they are pushing up the price massively |
| 04:05.19 | rhombus | you build your house with the materials you have on hand |
| 04:05.24 | rhombus | if you have rocks, you build with rocks |
| 04:05.28 | rhombus | if you have wood, you build with wood |
| 04:05.36 | rhombus | if you have cement, you build with cement |
| 04:06.04 | rhombus | China consumes steel because they are selling it to the rich countries |
| 04:06.18 | rhombus | I love how China gets blamed for everything -- who is buying all their shit? |
| 04:07.19 | rhombus | Everybody gladly piles into Walmart because everything there is fire sale cheap, then they bitch at the gas pump and fail to make the connection |
| 04:07.28 | rhombus | you pay one way or you pay another -- consumption costs |
| 04:08.37 | JT | china are also building massive citites where there was nothing |
| 04:08.39 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 04:08.42 | JT | but yes a lot goes to export |
| 04:18.37 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 04:19.03 | MrTelephone | anyone see a good dialplan for blocking incoming ani's |
| 04:25.02 | WilliamK | are you trying to route your inlaws to /dev/null for christmas? |
| 04:27.52 | MrTelephone | yeah |
| 04:28.11 | MrTelephone | hahah |
| 04:28.37 | MrTelephone | couldn't find any incoming call block examples |
| 04:28.40 | WilliamK | naughty TelephoneMan - Saint Nick see that you get lots of coal in your stockings for christmas! |
| 04:30.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:32.54 | BBHoss | exten => 5555555,1,Answer ; can't Playtones unless we answer first |
| 04:33.11 | BBHoss | exten => 5555555,2,Playtones(congestion) |
| 04:33.25 | BBHoss | exten => 5555555,3,Congestion |
| 04:33.34 | BBHoss | replace the 5s with the number you want |
| 04:34.17 | BBHoss | put that in extensions.conf |
| 04:37.21 | TJNII | That doesn't work for incoming calls. |
| 04:37.33 | phix | so any other ideas? |
| 04:37.43 | TJNII | agi, probably |
| 04:37.45 | riddlebox | WilliamK, there are some good ideas on the wiki |
| 04:37.50 | TJNII | Since it passes the caller ID |
| 04:38.42 | JT | who said you can't Playtones unless you answer first |
| 04:38.52 | BBHoss | wiki |
| 04:39.40 | JT | probably wrong |
| 04:39.42 | JT | early media |
| 04:39.50 | JT | depends on channel type obviously |
| 04:41.01 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:41.18 | riddlebox | oops sorry WilliamK I meant that for MrTelephone |
| 04:42.09 | JT | you shouldn't need to Answer to Playtones |
| 04:43.33 | MrTelephone | yeah i'll have to build the configs then for block for individual accounts |
| 04:43.54 | MrTelephone | minus a for |
| 04:44.02 | MrTelephone | thanks for the idea |
| 04:46.09 | *** part/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 04:47.55 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 04:48.01 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 04:50.54 | phix | JT: isn't using Answer a good habbit to get into? |
| 04:51.05 | phix | or so "the book" tells me |
| 04:52.52 | JT | only when a call needs answering |
| 04:53.02 | JT | if you plan to hang it up, answering is not a good habit |
| 04:53.26 | JT | as then the far end will get charged for the call, presuming you're using some form of digital signalling |
| 04:53.29 | JT | like pri or sip |
| 04:55.41 | phix | ok |
| 04:56.21 | phix | JT: ummm I still need transfering help :) any other ideas you can suggest? |
| 05:00.36 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 05:02.25 | phix | hmmmmmmmmmmmmmmmm |
| 05:10.47 | TJNII | JT: Good point about the charges. I wouldn't have thought of that. |
| 05:12.15 | *** join/#asterisk chinku (n=chinku@c-68-37-156-186.hsd1.nj.comcast.net) |
| 05:21.10 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 05:22.16 | TrentCreek | OT: Whats a good Assembler compiler for Linux? |
| 05:22.31 | jql | you looking beyond the GNU toolchain? |
| 05:22.50 | TrentCreek | no |
| 05:23.02 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 05:23.13 | TrentCreek | I am using MASM, but god it sucks..dammed errors when they should not be happening |
| 05:23.52 | jql | well, nobody works very hard on assemblers (except nasm. use nasm) |
| 05:25.20 | TrentCreek | groovy. What is the compile command? So I can check to see if it is on one system I am on...if not I have to start another one up and install it on there |
| 05:25.39 | jql | umm... |
| 05:25.47 | jql | nasm -o foo.o foo.asm |
| 05:26.04 | TrentCreek | thought so..thanks..its not on there |
| 05:26.41 | jql | google says there's an introduction to nasm for masm users |
| 05:27.32 | Sweeper | TrentCreek: writing zapata drivers for your c64 modem? :P |
| 05:27.32 | TrentCreek | MASN gets so screwed up when you start making changes to the program and starts generating errors. That is sad when even the examples on the MSDN web site do not even work on it. |
| 05:27.40 | TrentCreek | no |
| 05:27.45 | TrentCreek | for my VIC20 |
| 05:27.50 | Sweeper | nice |
| 05:28.16 | jql | a computer as old as I am... ooooookay |
| 05:28.29 | Sweeper | it is feeding time, so i type slow :3 |
| 05:28.33 | TrentCreek | ahh..a young whipper snapper |
| 05:29.04 | Sweeper | that thing is 4 years older than I am :D |
| 05:29.12 | jql | yeah, we damn 80s babies ruined the world (while shaking his fist at the 90s babies) |
| 05:30.59 | TrentCreek | yeah...made the internet gekless |
| 05:32.05 | linxroute | can somebody please explain for me in this case 0:1 is mean what - GotoIf($["${CALLERID(num):0:1}" = "1"]?5:6); - |
| 05:33.10 | *** join/#asterisk jer_ (n=jtregunn@unaffiliated/jer) |
| 05:35.50 | TrentCreek | I have nothing at this time because I have not gotten that far |
| 05:36.07 | TrentCreek | Have you looked at the sample sin the booK |
| 05:36.12 | Shaun2222 | anybody know if the polycom's can display a browser page while the phone is in use? |
| 05:36.33 | TrentCreek | jql: isnt th '.' in a loop part of assembly? |
| 05:36.35 | Shaun2222 | while on a call i want a page to be pulled and show information. |
| 05:36.55 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 05:37.39 | TrentCreek | okay then do it |
| 05:38.22 | TrentCreek | i know nothing about those phones or whatever they are |
| 05:40.38 | linxroute | i worked it out |
| 05:40.42 | linxroute | thanks anyway |
| 05:47.26 | TrentCreek | okay glad you got it |
| 05:47.37 | TrentCreek | i will be using that thingy soon |
| 05:47.50 | TrentCreek | what model of polycom? |
| 05:47.59 | linxroute | no it's not polycom |
| 05:48.03 | linxroute | it's cisco 7940G |
| 05:48.20 | linxroute | we had this stupid CallerID |
| 05:48.22 | TrentCreek | sorry..I mean't to ShaunOftheDead2222 |
| 05:48.27 | linxroute | okay |
| 05:49.22 | TrentCreek | while u in Vietnam...get me a wife |
| 05:49.23 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 05:49.24 | TrentCreek | ;-) |
| 05:49.45 | linxroute | oh sure |
| 05:50.09 | linxroute | plenty of them |
| 05:50.17 | linxroute | young and fresh |
| 05:50.41 | TrentCreek | I should have been in Vietnam war..but I was only a baby....I would have been in Heaven with all those Vietnamese gals |
| 05:51.30 | linxroute | so you r from the states |
| 05:51.34 | TrentCreek | yes |
| 05:51.46 | phix | hmmm does asterisk (or sip for that matter) support shared lines? |
| 05:51.55 | linxroute | SLA ? |
| 05:51.57 | linxroute | you meant |
| 05:52.06 | TrentCreek | I read about a guy who kept visiting Vietnam back in the 90s, so they kicked him out and raised the age of ceonsent |
| 05:52.07 | linxroute | shared line appearance |
| 05:52.14 | TrentCreek | no...US Army |
| 05:52.26 | jql | heh, sip has like 100 drafts and rfcs. a couple of them are about shared lines. :) |
| 05:52.44 | phix | linxroute: yeah :) I will google that, unless yuo know a link to a howto on it :) |
| 05:53.04 | linxroute | just google "asterisk SLA" |
| 05:53.07 | linxroute | :) |
| 05:53.15 | phix | yep |
| 05:53.59 | phix | linxroute: you know much about getting bind or attneded transfers (from features.conf) working on a SIP softphone? |
| 05:54.08 | linxroute | for sure asterisk 1.4 support SLA |
| 05:54.12 | linxroute | yeah |
| 05:54.18 | linxroute | you can config it in.. |
| 05:54.21 | linxroute | features.conf |
| 05:54.48 | TrentCreek | jql: isnt th '.' in a loop part of assembly? |
| 05:54.59 | linxroute | blindtransfer |
| 05:55.10 | linxroute | most of the sip ( hard phone ) |
| 05:55.14 | linxroute | already support it |
| 05:55.22 | jql | assembly only consists of opcodes. the rest is syntax |
| 05:55.34 | linxroute | even cheap 20$ chinese knockoff sip phone |
| 05:55.42 | linxroute | still does blindxfer |
| 05:56.06 | TrentCreek | You know what I mean, right? I saw some loops had 2 '.' in line in a loop |
| 05:56.20 | TrentCreek | when I did not include I got an error saying ...shit..cant remember |
| 05:56.33 | jql | they're trying to word-align the loop boundary, methinks |
| 05:56.35 | linxroute | phix inorder to get transfer |
| 05:56.59 | linxroute | you need to include DYNAMIC_FEATURES=>blindxfer |
| 05:57.03 | TrentCreek | ohhh... |
| 05:57.14 | linxroute | in globals |
| 05:57.18 | linxroute | extensions.conf |
| 05:57.59 | TrentCreek | I am just trying to figure out how to INC the character 041h (i.e.'A') |
| 05:58.34 | linxroute | something to do with the firmware ? |
| 05:58.48 | TrentCreek | kinuxroute: I had a good time with a Vietnamese gal back in 2003...she had the softest skin of any woman I ever been with |
| 05:58.56 | linxroute | really |
| 05:59.00 | linxroute | vietnamese american |
| 05:59.03 | linxroute | or .. |
| 05:59.07 | linxroute | vietnam gal |
| 05:59.21 | linxroute | plenty of them over here man |
| 05:59.27 | TrentCreek | no..she was just in the country on a visa to make money..if you know what I mena ;-) |
| 05:59.30 | TrentCreek | *mean |
| 05:59.37 | linxroute | espcially in saigon |
| 05:59.38 | linxroute | oh |
| 05:59.39 | linxroute | sure |
| 05:59.45 | linxroute | lotta them do that |
| 06:00.29 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 06:00.31 | tengulre | hi,all |
| 06:00.42 | TrentCreek | My friend's mother is Vietnamese..but he don't like vietnamese...he claims to be white |
| 06:00.48 | tengulre | where have open source sip phone code? |
| 06:00.48 | phix | linxroute: oohhh |
| 06:00.54 | phix | linxroute: thank you :) |
| 06:01.05 | linxroute | u r welcome phix |
| 06:01.41 | TrentCreek | tengulre...explain a bit better |
| 06:02.12 | phix | linxroute: I need to do that to enable other features in features.conf? |
| 06:02.18 | TrentCreek | linuxroute: His mother brought all her family to the US over the years and I went crazy for his cousin |
| 06:02.33 | linxroute | yes |
| 06:02.43 | linxroute | you need to remove the ";" |
| 06:03.07 | linxroute | it's under the [featuremap] |
| 06:03.10 | linxroute | i think |
| 06:04.21 | tengulre | TrentCreek: I need a open source sip phone, that support g711 codec and have sip protocol, rtp ..., can compile with VC++ |
| 06:04.35 | linxroute | opensource firmware |
| 06:04.42 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-69-66.dsl.irvnca.pacbell.net) |
| 06:04.45 | linxroute | ah |
| 06:04.48 | linxroute | softphone |
| 06:05.03 | linxroute | pletty of them out there |
| 06:05.05 | linxroute | sjphone |
| 06:05.10 | linxroute | for instance |
| 06:05.24 | TrentCreek | you dont need opensource..just go buy a box and you will have those codecs |
| 06:05.26 | phix | linxroute: w000t! blind transfers worked nicely! ok do I add anything else to DYMANIC_FEATURES=> for attended transfers? |
| 06:05.40 | phix | linxroute: you are extremely helpful :) |
| 06:06.08 | linxroute | no |
| 06:06.14 | linxroute | for attend x fer |
| 06:06.24 | linxroute | just remove the sign";" |
| 06:06.27 | linxroute | in atxfer |
| 06:06.35 | linxroute | if i remmber correctly |
| 06:06.49 | linxroute | u r welcome anytime, phix |
| 06:06.54 | phix | ok well atxfer isn't working :) and in featuremap it is uncommented |
| 06:07.27 | linxroute | *2 ? |
| 06:07.37 | linxroute | did you reload asterisk ? |
| 06:08.48 | linxroute | mine works just fine |
| 06:08.52 | linxroute | i just test it |
| 06:09.01 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 06:10.03 | linxroute | is away |
| 06:10.06 | phix | I did a reload in console |
| 06:10.07 | phix | aaww |
| 06:10.39 | linxroute | what's your phone ? |
| 06:10.40 | phix | linxroute: I need tT in Dial() too right? |
| 06:10.46 | phix | xlite |
| 06:10.50 | phix | softphone |
| 06:11.03 | phix | I need to find a better softphone prog, preferably a free one |
| 06:11.18 | TrentCreek | The books lists some |
| 06:11.24 | TrentCreek | but softphones suxs |
| 06:11.29 | TrentCreek | ~book |
| 06:11.30 | 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 |
| 06:11.39 | linxroute | no you dont need to add tT |
| 06:11.41 | linxroute | just let me check |
| 06:12.04 | phix | ok I see the problem |
| 06:12.08 | phix | I need to type it in very fast |
| 06:12.12 | phix | need to set the timeout |
| 06:12.23 | TrentCreek | I have never been happy with the sound waulity of software phones |
| 06:12.28 | TrentCreek | *quality |
| 06:12.37 | phix | yeah |
| 06:12.51 | phix | I am going to buy an ip phone |
| 06:13.06 | phix | looking for one that support coreless phones |
| 06:13.11 | TrentCreek | I got a Linksys delivered for $55 |
| 06:13.23 | phix | Aastra 57I CT IP Phone with dect handset |
| 06:13.24 | TrentCreek | that is really a sipura2000 |
| 06:13.35 | phix | Is what I am looking at, but they are like $570 AUD |
| 06:13.50 | phix | TrentCreek: nice, where at? |
| 06:14.09 | TrentCreek | ATA is better because you get to connect any phone you want and overall costs are cheaper |
| 06:14.35 | TrentCreek | i think the name of it is..... |
| 06:14.37 | TrentCreek | let me see |
| 06:15.33 | phix | with an ATA it routes calls between a landline and POTS phones right? |
| 06:15.51 | phix | and they connect to asterisks fine? |
| 06:16.06 | phix | or would I be better with a tdm2400 |
| 06:16.15 | TrentCreek | not all of them. you have to buy one that has that ability |
| 06:16.36 | phix | and will default to ring through asterisk? |
| 06:16.37 | TrentCreek | I have to pull my email up to find the name of company |
| 06:16.42 | phix | hehe ok :) |
| 06:16.45 | phix | I will wait |
| 06:17.59 | phix | I mean I would like asterisk to decide whether the call is forwarded back to ATA to dial out via landline or not |
| 06:18.10 | phix | as ATAs are usually stupid :) |
| 06:18.12 | TrentCreek | This video shows you (downloadable) how to connect ATA to Asterisk and POTS |
| 06:18.14 | TrentCreek | http://revision3.com/systm/asterisk/ |
| 06:18.24 | TrentCreek | I had mine connected in moments |
| 06:20.59 | *** join/#asterisk chinku (n=chinku@c-68-37-156-186.hsd1.nj.comcast.net) |
| 06:22.05 | TrentCreek | The place is called Telephony Depot, but it is in the US |
| 06:22.26 | TrentCreek | They did have the loswest prices I could find online |
| 06:23.54 | TrentCreek | with the US $ shrinking to all time lows, I am sure the cost would benefit you |
| 06:26.51 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:29.15 | tengulre | anybody successful running SS7 with asterisk? |
| 06:29.54 | Shaun2222 | any solutions for ztdummy on a Xen vps? |
| 06:31.07 | JT | why not just use a conferencing app that doesn't rely on ztdummy |
| 06:31.13 | TrentCreek | never heard of it |
| 06:33.10 | JT | app_conference |
| 06:33.12 | Shaun2222 | JT: what else is there? |
| 06:33.22 | JT | for asterisk, app_conference |
| 06:33.29 | JT | as app_meetme is zaptel conferencing |
| 06:33.58 | TrentCreek | what about "meet_me?" |
| 06:34.33 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 06:36.39 | phix | interesting video |
| 06:36.59 | *** join/#asterisk alpha232 (i=alpha232@198-144-143-60.dyn.megabroadband.net) |
| 06:37.18 | TrentCreek | i think it is a good one. It had me connected in moments after seeing it |
| 06:37.38 | TrentCreek | People say "gawd its old" but it still the same principal |
| 06:40.58 | Shaun2222 | god that video has that fag from that stupid show. |
| 06:41.08 | Shaun2222 | i want to kick that guy in the throat. |
| 06:41.44 | alpha232 | *sigh* |
| 06:41.54 | alpha232 | why is telephony hardware so expensive? |
| 06:43.43 | TrentCreek | it does not have to be if you know what to look for |
| 06:44.07 | *** join/#asterisk beasty_ (n=jdecoste@about/apple/macbook/beasty) |
| 06:44.09 | TrentCreek | ShaunOfTheDead: what stupid show? |
| 06:44.11 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:44.20 | beasty_ | can anyone tell me what a stun server is ? |
| 06:44.31 | alpha232 | TrentCreek: suggestions? |
| 06:44.34 | Shaun2222 | TrentCreek: well he was on that show screen savers which wasnt too terrible.. |
| 06:44.45 | Shaun2222 | but he has some stupid show he puts on youtube.. |
| 06:44.49 | TrentCreek | oh....Thanks to G TV they killed it |
| 06:44.49 | alpha232 | beasty_: it allows UDP over NAT |
| 06:44.51 | Shaun2222 | it's retarded. |
| 06:44.57 | [hC] | Shaun2222: what show? |
| 06:45.03 | alpha232 | beasty_: http://www.voip-info.org/wiki-STUN |
| 06:45.04 | TrentCreek | What are you looking for alpha??? |
| 06:45.05 | beasty_ | alpha232: is it hard to setup ? |
| 06:45.10 | beasty_ | oh ok thanks alpha232 |
| 06:45.47 | Shaun2222 | i cant remember the name of the other shows |
| 06:45.52 | Shaun2222 | one of them was a game kiddie show. |
| 06:46.17 | Shaun2222 | the other was retarded and a wanabee hacker show. |
| 06:46.24 | [hC] | Shaun2222: who are you talking about over all here? |
| 06:46.46 | tengulre | anybody successful running SS7 in asterisk ? |
| 06:46.57 | Shaun2222 | Kevin Rose |
| 06:47.03 | TrentCreek | http://revision3.com/systm/asterisk/ |
| 06:47.19 | TrentCreek | He is talking about this video and the guy in it |
| 06:47.22 | alpha232 | shit ss7?! thats heavydutyshit |
| 06:47.39 | TrentCreek | Intel's Super Socket 7 ;-) |
| 06:47.48 | alpha232 | OHHHH |
| 06:47.58 | TrentCreek | no..I don't know |
| 06:47.58 | alpha232 | i thought we were talking switching |
| 06:48.04 | [hC] | he does diggnation, and did systm |
| 06:48.06 | TrentCreek | yes he is |
| 06:48.11 | [hC] | alpha232: he is. |
| 06:48.12 | alpha232 | TrentCreek: wiseass |
| 06:48.24 | alpha232 | ss7 is da bomb :) |
| 06:49.29 | alpha232 | oob signaling is where it's at, i can't believe the US still has analog |
| 06:49.30 | tengulre | alpha232: why ? |
| 06:50.05 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 06:50.13 | alpha232 | tengulre: sorry I am a big fan of ISDN |
| 06:50.23 | alpha232 | tengulre: I use it at home for my voice service... |
| 06:50.24 | TrentCreek | People just dont realise that the US is the second most country with ..what do you call it |
| 06:50.36 | tengulre | I want building a medium call centre, have 4E1 and 240 agents, in our city most of using SS7, |
| 06:50.39 | TrentCreek | 2 least population density |
| 06:50.48 | TrentCreek | Austrialia is First |
| 06:50.59 | alpha232 | tengulre: and your telco will give you an SS7 tie in? |
| 06:51.01 | tengulre | service provider only provide SS7 protocol. |
| 06:51.02 | tengulre | so.. |
| 06:51.03 | TrentCreek | imagine the cost of trying to update everything every year? |
| 06:51.10 | tengulre | alpha232:yeah. |
| 06:51.12 | alpha232 | tengulre: shit you're kidding, damn lucky |
| 06:51.17 | TrentCreek | with everyone so spread out? |
| 06:51.18 | tengulre | ;( |
| 06:51.29 | alpha232 | tengulre: where are you located? |
| 06:51.37 | tengulre | Xi'an of china. |
| 06:52.07 | alpha232 | tengulre: http://www.voip-info.org/tiki-index.php?page=Asterisk+SS7 |
| 06:52.21 | tengulre | does the asterisk support 240 agents in a single linux box? |
| 06:52.32 | TrentCreek | tengalre...send me a Chinese wife |
| 06:52.37 | alpha232 | tengulre: what kind of agents |
| 06:52.53 | TrentCreek | Mister Anderson s |
| 06:52.57 | alpha232 | tengulre: ACD? UCD? Single Stations, analog, ip? |
| 06:53.06 | TrentCreek | I mean Agent Smith |
| 06:54.07 | tengulre | alpha232: 4E1--->asterisk--->Queue-->iax2 client. |
| 06:54.18 | tengulre | is it possible? |
| 06:54.45 | tengulre | all iax2 clients in a same lan. |
| 06:54.55 | alpha232 | tengulre: it's somewhat more granular than that |
| 06:55.23 | alpha232 | are all 240 agents going to be on the same splits? |
| 06:55.31 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-70-240-164-157.dsl.hstntx.swbell.net) |
| 06:55.42 | alpha232 | tengulre: and will you have enough bandwidth |
| 06:56.25 | tengulre | alpha232: you are point switch in lan? I using 1000/100M switch. |
| 06:57.23 | alpha232 | tengulre: is this a dedicated lan, vlan over existing lan or shared lan |
| 06:58.05 | tengulre | alpha232: not dedicated lan, in a existing lan. |
| 06:58.13 | alpha232 | good luck |
| 06:58.21 | tengulre | ;( |
| 06:58.37 | tengulre | now it's only a plan. |
| 06:58.49 | tengulre | I want get answer here. |
| 06:59.16 | alpha232 | tengulre: if you through enough money at it, sure it will work |
| 06:59.22 | alpha232 | but if you're required to do SS7 |
| 06:59.30 | tengulre | I only building 4 agents before. so.. I aferid. |
| 06:59.31 | alpha232 | you might just buy a real switch |
| 06:59.33 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:59.37 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:59.43 | tengulre | oh! |
| 07:00.00 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-34-39.bchsia.telus.net) |
| 07:00.14 | tengulre | alpha232: you are right, but to buy real swtich, that too dear. |
| 07:01.48 | alpha232 | tengulre: are you going to be using HSL for your signaling? |
| 07:02.28 | alpha232 | plus 240 agents? |
| 07:02.32 | tengulre | High Speed Link? |
| 07:02.35 | alpha232 | 4 E1's won't do that |
| 07:02.58 | alpha232 | you'll run out of trunks |
| 07:03.09 | tengulre | how many lines supported in asterisk , limited? |
| 07:03.17 | tengulre | ;( |
| 07:03.35 | tengulre | I known your mean, cry!!!..... ;( |
| 07:03.40 | alpha232 | my point is 240 agents at one time... |
| 07:04.17 | alpha232 | and how many channels, each E1 is 32 channels and with SS7 you need a dedicated signaling channel |
| 07:04.31 | alpha232 | so you'll only have 124 channels |
| 07:04.33 | TrentCreek | The number of channels you can handle depend on many factors. |
| 07:05.03 | tengulre | 4 lines signaling, and other running audio. |
| 07:05.12 | tengulre | s/other/othres. |
| 07:05.13 | alpha232 | TrentCreek: but i'm just talking about having 240 agents but only 4 E1's |
| 07:05.34 | TrentCreek | If the calls are already in digital format, meaning no AD/DA conversion, then the next thing is how fast your network connection is |
| 07:05.35 | *** join/#asterisk Buhntz (i=Boones@port-212-202-42-6.dynamic.qsc.de) |
| 07:06.20 | alpha232 | TrentCreek: just transcoding |
| 07:06.22 | tengulre | 4E1 only incoming interface. 240 agents running iaxclient. |
| 07:06.38 | tengulre | all agents using gsm codec. |
| 07:07.59 | alpha232 | TrentCreek: what souces for hardware would you suggest, say if i wanted to build a home asterisk as a hobby |
| 07:08.19 | alpha232 | tengulre: there are alot of things to consider |
| 07:08.33 | alpha232 | tengulre: you're sharing an existing network, what is it's topology |
| 07:08.57 | alpha232 | tengulre: are there any existing bottlenecks, does everything run Gigabit from one end to the other |
| 07:08.59 | TrentCreek | where you located? |
| 07:09.04 | alpha232 | TrentCreek: he's in china |
| 07:09.31 | alpha232 | TrentCreek: china has become one of the #1 technology countries because all the US mfgs outsource to china |
| 07:09.43 | TrentCreek | isn't chinese goodies there cheap? |
| 07:09.47 | alpha232 | TrentCreek: so they just do a few extra production runs and can sell within the country |
| 07:09.51 | alpha232 | yep |
| 07:09.59 | TrentCreek | then buy stuff there |
| 07:10.17 | TrentCreek | Get the AMD Turon 64 x2 |
| 07:10.29 | alpha232 | TrentCreek: he is panicing because he's only rolled out a 4 agent system before |
| 07:10.49 | TrentCreek | its cheaper and you will get better performance using 64 bit linux |
| 07:10.52 | tengulre | anybody know how to press testing in asterisk. |
| 07:11.21 | TrentCreek | and with 2=4 GB of RAm you should be able to handle about 1000 callers |
| 07:11.30 | tengulre | TrentCreek: alpah232 is right. |
| 07:11.31 | TrentCreek | if you got a fast connection |
| 07:12.09 | tengulre | TrentCreek: you means is add RAM can add callers? |
| 07:12.20 | TrentCreek | no |
| 07:12.53 | TrentCreek | at a certain point speed of memory and computer Bus come into play |
| 07:13.12 | TrentCreek | Mempry bus is the weak link in a computer |
| 07:13.12 | *** join/#asterisk socken23 (n=socken@ip-213-189-154-029.fix.magnet.ch) |
| 07:13.18 | TrentCreek | *Memory |
| 07:14.26 | TrentCreek | But don't worry about it..I got a P4 with 2 GB of RAM and it can handle 250 callers with ease |
| 07:15.21 | tengulre | Result, building 4E1/SS7 and 240 agents in asterisk is dangerous, is right? |
| 07:15.23 | TrentCreek | The problem comes if Asterisk has to convert Analog to Digital or the other way arounf |
| 07:15.51 | TrentCreek | no...you will not have any problem |
| 07:15.51 | tengulre | TrentCreek: which channel are you using? SIP?IAX2? |
| 07:16.03 | TrentCreek | It depends |
| 07:16.06 | TrentCreek | I use both |
| 07:16.23 | TrentCreek | my outgoing channel uses IAX2 and my incoming uses SIP |
| 07:17.19 | tengulre | 250 callers working at one time? |
| 07:17.23 | alpha232 | thats interesting, any reason why? |
| 07:17.53 | TrentCreek | sure..it can handle that many callers., but my net connectiion cannot |
| 07:18.06 | TrentCreek | alpha: because It is the services I am using |
| 07:18.30 | alpha232 | ah |
| 07:18.30 | TrentCreek | Incoming DIDs use SIP and my ATAs use SIP and outgoing trunks use IAX2 |
| 07:18.57 | alpha232 | TrentCreek: i'd love to roll out voip at the hotel i work at but ehh too much work lol |
| 07:19.33 | TrentCreek | not really..it's easy |
| 07:19.50 | alpha232 | TrentCreek: only cat3 going to the guest rooms |
| 07:20.01 | TrentCreek | that's all u need. |
| 07:20.01 | alpha232 | and typically only one pair is wired |
| 07:20.25 | TrentCreek | First call only uses 84KB and next ones use 64KB |
| 07:21.21 | alpha232 | TrentCreek: well right now my main concern is getting some hardware of my own to play with |
| 07:21.31 | alpha232 | TrentCreek: then I can take it in, hook it up and go "voila" |
| 07:21.56 | TrentCreek | Not diffucult..after I saw that video I realized how easy it is to setup |
| 07:21.59 | alpha232 | i hate people call it voype |
| 07:22.08 | alpha232 | not difficult |
| 07:22.15 | alpha232 | but expensive hardware lol |
| 07:22.31 | k31th | Morning |
| 07:22.41 | TrentCreek | no it is not |
| 07:22.49 | k31th | 7:22 here |
| 07:23.02 | alpha232 | 3:22 here |
| 07:23.02 | k31th | cat3 ? |
| 07:23.09 | alpha232 | k31th: yep |
| 07:23.14 | alpha232 | k31th: voice lines thats it |
| 07:23.19 | TrentCreek | I used a cheap computer (P4) I out together and bought a cheap ATA and i was in business |
| 07:23.26 | TrentCreek | only cost me about $300 US |
| 07:23.36 | k31th | 10mbps ? |
| 07:23.46 | k31th | ah |
| 07:23.47 | alpha232 | k31th: analog, 1 pair |
| 07:23.54 | k31th | of coarse |
| 07:23.58 | tengulre | TrentCreek : which sip gateway are you using? |
| 07:24.17 | TrentCreek | the Asterisk one |
| 07:24.35 | TrentCreek | is that what you mean? |
| 07:24.51 | tengulre | TrentCreek: it have not PSTN interface? |
| 07:25.25 | tengulre | 15:25 here. |
| 07:25.30 | TrentCreek | Mine does not, but you can buy a ATA with them, which a LOT cheaper than that Digium stuff |
| 07:25.51 | TrentCreek | and the ATA does the DA conversion leaving the computer free from that |
| 07:25.56 | tengulre | ok. |
| 07:26.39 | TrentCreek | I got a Linksys PAP2 (2 line) delivered for only $55 US |
| 07:27.07 | TrentCreek | compared to Digium used stuff would cost 4 times that |
| 07:27.11 | alpha232 | i need a cheep bri interface |
| 07:27.35 | TrentCreek | no..you just need a LAN connection |
| 07:27.43 | TrentCreek | and broadband |
| 07:28.10 | alpha232 | ummm no I need a BRI interface lol |
| 07:28.18 | alpha232 | thats how I get my POTS |
| 07:30.17 | TrentCreek | you dont need that |
| 07:30.25 | TrentCreek | just order a cheap DID |
| 07:30.37 | TrentCreek | and have a number from anywhere in the world |
| 07:31.23 | alpha232 | *cough* |
| 07:31.45 | alpha232 | i'm not looking to cut off from the telco |
| 07:36.19 | ManxPower | alpha232: good move. |
| 07:36.23 | ManxPower | where are you located? |
| 07:36.56 | alpha232 | New England |
| 07:37.05 | alpha232 | It took me 2 years to get ISDN working right here lol |
| 07:37.15 | alpha232 | i'm not getting rid of it lol |
| 07:37.37 | alpha232 | I just want to get a dual line pbx in the house |
| 07:37.37 | TrentCreek | ISDN? yuck |
| 07:37.39 | k31th | alpha232: pri / bri ? |
| 07:37.45 | alpha232 | k31th: bri |
| 07:37.53 | alpha232 | TrentCreek: do you know anything about ISDN? |
| 07:37.56 | k31th | alpha232: interesting... What card did you use? |
| 07:38.02 | TrentCreek | yes. |
| 07:38.02 | alpha232 | k31th: i havn't yet lol :) |
| 07:38.13 | alpha232 | TrentCreek: name one thing wrong with ISDN |
| 07:38.21 | TrentCreek | I found out that I was better off with cheaper iDSL |
| 07:38.21 | phix | alpha232: $$$$$$$$$$$$$$$$$$ |
| 07:38.25 | k31th | alpha232: A client of mine is after bri |
| 07:38.37 | alpha232 | TrentCreek: ummm I'm not using it for DATA |
| 07:38.43 | TrentCreek | But that was when I lived in the woods |
| 07:38.44 | k31th | ISDN is the most reliable... |
| 07:38.46 | alpha232 | I use ISDN for my POTS |
| 07:38.55 | TrentCreek | just use a DID |
| 07:38.55 | alpha232 | setup and breakdown is amazing |
| 07:39.01 | ManxPower | alpha232: you have a long road ahead of you. There is no good support for USA ISDN BRI, only EuroISDN BRI |
| 07:39.09 | alpha232 | k31th: is it me or does TrentCreek throw around acronyms |
| 07:39.11 | TrentCreek | why would anyone pay so much $$$$$$$$$$$$$$ for ISDN? |
| 07:39.38 | k31th | for a reliable phone system ? |
| 07:39.41 | ManxPower | TrentCreek: There ARE places where a 2B+D BRI is about the same as two analog lines. |
| 07:39.49 | alpha232 | actually less :) |
| 07:40.15 | k31th | BRI 2 channel is the same price as two analog lines here. |
| 07:40.24 | alpha232 | actually i save :) |
| 07:40.26 | ManxPower | alpha232: honestly, you're screwed. |
| 07:40.28 | alpha232 | 1 port charge, 2 lines |
| 07:40.30 | TrentCreek | Just use a DID..its a LOT cheaper |
| 07:40.38 | TrentCreek | Direct Inward Dialing - A method in which incoming calls are routed directly to endpoints on the LAN, without operator intervention |
| 07:40.41 | alpha232 | ManxPower: well I was looking at getting a cisco BRI with POTS on the rear |
| 07:40.45 | alpha232 | ManxPower: but it doesn't do voip |
| 07:41.27 | TrentCreek | It does do VOIP if you send it with data |
| 07:41.38 | ManxPower | alpha232: that makes you have to learn the ins and outs of Cisco. You might be able to get a Cisco BRI Voice card and use the Cisco for routing and ISDN/SIP conversion. |
| 07:42.06 | alpha232 | ManxPower: yeah but most of the cheap ones with a BRI and POTS the pots are FXS |
| 07:42.25 | k31th | hang on, your going to use isdn for voice right? |
| 07:42.29 | ManxPower | plug them into an analog card on the asterisk box. |
| 07:42.48 | alpha232 | ManxPower: yeah but i want the power of call setup/breakdown/CLASS etc |
| 07:43.13 | ManxPower | Just make SURE the BRI/POTS box supports disconnect supervision, and any other sort of supervision would be handy. |
| 07:43.28 | alpha232 | ManxPower: answer/disconnect is key |
| 07:44.03 | ManxPower | alpha232: I've been on this channel for at least 4 years and I have never heard of anyone getting USA ISDN BRI to work with Asterisk. In fact I don't know if I've ever heard of someone trying. |
| 07:44.14 | TrentCreek | so just use ATA with FXO and FXS |
| 07:44.19 | ManxPower | Oh, yes, someone was trying with the Digium BRI card. |
| 07:44.35 | alpha232 | ManxPower: if i could get it at a reasonable price, I would do it |
| 07:44.39 | TrentCreek | yeah because it makes no sense to pay too much for that stuff ;-) |
| 07:44.54 | TrentCreek | People use Asterisk to SAVE money..not hand it out ;-) |
| 07:45.14 | ManxPower | TrentCreek: People use Asterisk for many different reasons. |
| 07:45.59 | TrentCreek | yes and I am sure the first one is about saving a few bucks |
| 07:46.27 | alpha232 | ManxPower: it seems the digiums are all quad bri lol |
| 07:46.44 | alpha232 | if i was gonna do quad bri, i'd save the money and do PRI with 8 channels lol |
| 07:46.57 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:47.11 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 07:48.34 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:50.51 | alpha232 | ManxPower: so anyways :) I guess since the hardware isn't available, thats the issue |
| 07:54.48 | TrentCreek | not even on eBay |
| 07:55.32 | TrentCreek | ? |
| 07:55.48 | J4k3 | unluckily most of you don't have internet access good enough to handle voip nicely |
| 07:55.53 | J4k3 | :) |
| 07:56.32 | TrentCreek | Well..I am lucky enough that my ISP upgraded their system and for a small extra $10 a month I now have 10MB down and 1.5MB up |
| 07:56.54 | TrentCreek | good enough for 30 callers |
| 07:56.57 | agx | J4k3, get g729 and put a zeroshell box between router and LAN with a simple QOS script that prioritize SIP proxy/server IP |
| 07:57.49 | alpha232 | pffft |
| 07:58.25 | J4k3 | agx: nice plan except, at least in the USA, there are "broadband" providers that still suck too bad to pass sip |
| 07:59.17 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 07:59.43 | J4k3 | your provider needs to not have interest in making your voip experience suck |
| 08:00.17 | alpha232 | ManxPower: my favorite thing about ISDN is that I can answer a call before the caller even hears it ring, and no need to wait for the first ring for caller id info |
| 08:00.50 | TrentCreek | I use Road Runner from Time Warner. I maxed the up speed and then made a call and it sounded fine |
| 08:01.46 | agx | J4k3, they provide a line that never has a minimum bandwidth? |
| 08:02.21 | J4k3 | agx: rarely at a rate thats reasonably affordable |
| 08:02.38 | J4k3 | SLAs are harder to get these days |
| 08:03.28 | agx | J4k3, that's scary O.o i tought Italy was the most worst country about broadband, standard xDSL is 128/640 with only 32k guaranteeded (so basically you can use skype or 1 SIP call) |
| 08:06.51 | J4k3 | in the USA practically all consumer bandwidth is sold "up to xx kbps" and there is no "down to xx kbps... its assumed 0. |
| 08:06.54 | J4k3 | its horrid. |
| 08:07.40 | alpha232 | J4k3: of course, because it's all about overselling |
| 08:07.50 | alpha232 | unless you own the pipe, you get no SLA |
| 08:07.52 | J4k3 | yep. |
| 08:08.26 | alpha232 | if I buy a T1 from bell, and pay corkies internet for a T1 port on their network.. then I can get Guaranteed T1 to their network |
| 08:08.29 | alpha232 | after that, who knwos |
| 08:08.38 | J4k3 | yep |
| 08:09.02 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:09.56 | phix | supples |
| 08:10.17 | alpha232 | pft |
| 08:12.21 | *** part/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 08:15.04 | TrentCreek | huh? |
| 08:16.17 | alpha232 | TrentCreek: Kevin Rose, the interviewer in that video |
| 08:16.35 | TrentCreek | you better watch Torchwood if you want that and see John Barrowman. |
| 08:17.25 | alpha232 | uhh huh ok |
| 08:18.02 | TrentCreek | ;-) |
| 08:19.11 | TrentCreek | Barrowman is best known on British television for his acting and his presenting work on theatre. His most prominent television role has been the Captain Jack Harkness in Doctor Who and its spin-off series Torchwood. Barrowman has been with his partner, British architect Scott Gill, since 1993 when they met during a production of Rope at the Chichester Festival Theatre.[17] The couple have homes in London and the Bay Area of Cardiff |
| 08:19.30 | TrentCreek | LOL |
| 08:20.14 | alpha232 | ok and? |
| 08:20.19 | alpha232 | your point is what? |
| 08:20.24 | alpha232 | are you saying i'm a lesbian? |
| 08:20.40 | TrentCreek | Maybe ol John would be your type more than Kevin Rose ;-p |
| 08:22.09 | TrentCreek | gosh I need to learn that rather than --- |
| 08:23.02 | tzafrir | geekboy? or greekboys? (from Lesbos) |
| 08:23.21 | alpha232 | GEEK not Greek |
| 08:23.42 | TrentCreek | no trouts |
| 08:25.00 | Sweeper | take it to #asstricks imo |
| 08:25.13 | alpha232 | naw, not really big into anal |
| 08:25.33 | alpha232 | i'm more concerned with getting BRI under * so I can toy with it |
| 08:27.36 | TrentCreek | lol |
| 08:36.23 | agx | Anyone has a working version of app_txfax.c ? my version keep locking/unlocking the channel list making everything freeze. |
| 08:46.08 | TrentCreek | alpha..why not just get a ISDN modem and plug in from LAN? |
| 08:47.46 | alpha232 | TrentCreek: one last time... i'm not using it for data |
| 08:49.10 | TrentCreek | yes I know... |
| 08:49.19 | alpha232 | why not just get a ISDN modem and plug in from LAN? |
| 08:49.27 | alpha232 | idiot |
| 08:51.18 | *** join/#asterisk OzyWebMaster (n=ozywebma@adsl123-108-66-11.dyn.koalatelecom.com.au) |
| 08:51.27 | OzyWebMaster | Hello All |
| 08:51.56 | OzyWebMaster | Can anyone point me in the direction of some DOCS on using integrating my VisualBasic application with ManagerAPI |
| 08:52.11 | OzyWebMaster | I only found a small reference to an old library with no work done in over 12mths |
| 08:52.23 | JT | TrentCreek: isdn modem, why would you do that? |
| 08:53.14 | k31th | you guys done much with blacklisting? |
| 08:55.42 | TrentCreek | ol Alpha seems to have problems and I suggested using a DID on the ISDN line |
| 08:56.16 | TrentCreek | of course there is the option of buying a ATA with a FXS and FXO on it |
| 08:57.11 | alpha232 | JT: either Trent fails to understand what ISDN is or just likes acronyms |
| 08:57.21 | JT | and how would an ISDN modem help at all, TrentCreek ? |
| 08:57.43 | *** join/#asterisk blq (n=Bl@dslb-088-067-042-033.pools.arcor-ip.net) |
| 08:57.45 | TrentCreek | By connecting via data |
| 08:57.49 | JT | how would an ATA help at all? |
| 08:57.58 | alpha232 | and how would connecting via data help? |
| 08:57.59 | JT | you're just talking crap now |
| 08:58.13 | TrentCreek | oh? |
| 08:58.22 | alpha232 | JT like the ircname, got any pfy's to farm out for organ donation? |
| 08:58.41 | TrentCreek | then what the fuck is a FXO and FXS for? |
| 08:58.52 | TrentCreek | which an ATA can have both |
| 08:58.53 | alpha232 | TrentCreek: analog POTS you stupid fuck *cough* |
| 08:59.26 | JT | alpha232: nah sorry |
| 08:59.29 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:59.34 | alpha232 | I'm talking 2B1Q not POTS |
| 08:59.46 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-34-39.bchsia.telus.net) |
| 08:59.51 | JT | TrentCreek: WHAT DOES AN ATA HAVE TO DO WITH ISDN? |
| 08:59.56 | alpha232 | JT: thats ok, i feel a brain donation to TrentCreek would be a waste |
| 09:00.31 | TrentCreek | Then how the hell wold you plug in a plain ol telephone into an ISDN voice line????? |
| 09:01.11 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 09:01.27 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:01.39 | JT | TrentCreek: you are an idiot, please stop misleading channels on ISDN when you clearly don't know the first thing about it |
| 09:01.46 | JT | s/channels/the channel/ |
| 09:02.04 | tzafrir | alpha232, do you have any ISDN card? |
| 09:02.05 | TrentCreek | hey JT *PLONK********************** |
| 09:02.22 | JT | i have yet you see a statement from you with regards to ISDN that is not COMPLETELY WRONG, TrentCreek |
| 09:02.31 | JT | killfiles are for pussies |
| 09:02.32 | alpha232 | tzafrir: at the moment I am shopping around for one, however I almost had a heartattack at one price tag. |
| 09:02.34 | JT | and people who are wrong |
| 09:02.52 | alpha232 | tzafrir: I won't be doing anything though using a TA and ATA, i want straight digital |
| 09:02.54 | tzafrir | alpha232, where are you at? |
| 09:03.13 | alpha232 | tzafrir: US unfortunately as I am an avid supporter of ISDN |
| 09:03.18 | TrentCreek | kill files are for dick heads like you who insult without reason..see ya fucking loser |
| 09:03.24 | JT | yes |
| 09:03.26 | tzafrir | alpha232, I figure you should try ebay |
| 09:03.27 | JT | you really did kill file |
| 09:03.34 | JT | finny how you could still read what i said |
| 09:03.40 | alpha232 | tzafrir: thats the next stop, i've been reading up on "supported" BRI cards |
| 09:04.03 | JT | most publicity whores who do public ignore announces lie about doing the ignore |
| 09:04.25 | tzafrir | hmmm... the US... not sure what it actually takes to get ISDN BRI to work there |
| 09:04.39 | JT | tzafrir: NI2 signalling i think |
| 09:04.53 | alpha232 | NI1 |
| 09:05.08 | JT | TrentCreek: yes no reason, except you getting things utterly wrong and refusing to research |
| 09:05.26 | alpha232 | I am hoping to avoid putting in an NT1 and looking for a card that has an onboard NT1 (ie provides a U interface) |
| 09:06.23 | alpha232 | tzafrir: re your /msg, it seems there are a fair number of people who do but ISDN in the US sucks ass |
| 09:06.36 | JT | don't know of any bri cards that have an onboard NT1, alpha232 |
| 09:07.15 | alpha232 | JT: most don't say :/ but i eicon has a Diva board for US with NT1 built in |
| 09:07.25 | alpha232 | but i saw the price tag and nearly shit myself |
| 09:07.31 | JT | hah |
| 09:07.36 | alpha232 | 700-900 |
| 09:07.44 | alpha232 | I've seen T1 cards cheaper |
| 09:07.49 | JT | and who knows if it works with asterisk |
| 09:08.09 | alpha232 | JT: http://www.asteriskguru.com/tutorials/bri.html |
| 09:08.20 | alpha232 | tzafrir: fyi on BRI http://www.asteriskguru.com/tutorials/bri.html |
| 09:08.22 | *** join/#asterisk arcanine (n=saxon_m2@203.82.44.179) |
| 09:08.39 | JT | i've heard a total of zero people who have bri working with asterisk in the usa, alpha232 |
| 09:08.45 | JT | and i don't need a bri tutorial :) |
| 09:09.01 | alpha232 | JT: well it lists some known working bri cards |
| 09:09.29 | JT | the configuration (NI/NI2) needs to be supported, not just the card |
| 09:09.55 | alpha232 | true |
| 09:10.39 | alpha232 | right now I have a TA with 2 pots but i want to replace it, the box is over 10 years old, i got it when I worked at motorola |
| 09:10.44 | alpha232 | they had to custom order it lol |
| 09:10.50 | JT | heh |
| 09:10.52 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 09:11.07 | alpha232 | I started at Motorola when I was 17 and they didn't know how old i was LOL |
| 09:11.29 | alpha232 | then some people found out it wsa my birthday and we had a little party, someone asked how old i was, said 18... and the room went quiet |
| 09:11.46 | JT | why |
| 09:12.15 | alpha232 | lol because I had just come over from an engineering department where I worked for 3 years |
| 09:12.26 | alpha232 | lol I started working for an ISP when iwas 15, |
| 09:12.36 | JT | heh ok |
| 09:13.00 | alpha232 | yeah so they knew i was young, but not THAT young |
| 09:13.10 | alpha232 | anyways I bought a buttsurfer pro isdn modem |
| 09:13.13 | JT | did HR know? |
| 09:13.23 | alpha232 | i was contract so thats a No ;) |
| 09:13.29 | JT | ah |
| 09:14.14 | tzafrir | alpha232, that astriskguru page is rather obsolete |
| 09:14.22 | JT | asteriskguru always is |
| 09:14.26 | tzafrir | chan_capi is no longer part of bristuff |
| 09:14.28 | JT | EVERYTHING is out of date there |
| 09:14.33 | tzafrir | bristuff is not just for zaphfc |
| 09:14.38 | alpha232 | my point is though... |
| 09:14.58 | alpha232 | it may be old but i don't see card support regressing in * |
| 09:15.06 | alpha232 | there is alot of old stuff still supported |
| 09:15.18 | JT | but it does |
| 09:15.25 | JT | if there's proprietary drivers |
| 09:15.30 | JT | they might not be in 1.4 |
| 09:15.34 | alpha232 | hrrm |
| 09:15.36 | tzafrir | When I get search hits to asteriskguru they are usually to comments on an article, and often to someone asking an irrelevant question or so |
| 09:16.10 | alpha232 | well my ultimate goal in the end is to have 1BRI and 1GSM for my telco side |
| 09:16.34 | JT | easy way to acheive that goal is to move to somewhere with euroisdn ;) |
| 09:16.40 | alpha232 | lol |
| 09:16.53 | alpha232 | JT yeah, the US sucks for ISDN & GSM |
| 09:17.12 | JT | true |
| 09:17.57 | alpha232 | I think I might be crazy to do it |
| 09:18.23 | alpha232 | but it would be cool having my CO lines and having a GSM line |
| 09:18.41 | alpha232 | esp with free nights/weekends and mobile2mobile |
| 09:18.56 | alpha232 | plus that opens up text messaging options and all sorts of stuff |
| 09:19.20 | Uatec | hi |
| 09:19.32 | Uatec | is it possible for me to find out the user agent of a sip peer from the CLI? |
| 09:19.32 | *** join/#asterisk Ludanek (n=IceChat7@88.103.127.254) |
| 09:19.52 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 09:20.58 | JT | Uatec: sip debug |
| 09:21.02 | Ludanek | hi, can somebody help me with registering G729 [astlinux] ? Can I paste my problem here ? |
| 09:21.37 | Ludanek | then... |
| 09:21.41 | Ludanek | **: verbosity is enabled. |
| 09:21.41 | Ludanek | **: checking to see if you are running 'register' as 'root' ......OK |
| 09:21.41 | Ludanek | **: initializing copy protect subsystem ......OK |
| 09:21.41 | Ludanek | **: trying to determine hostid ......OK (host: '39:c0:78:3f:0e:e2:b6:ca:84:92:aa:65:d1:83:3b:77:76:a5:f1:55') |
| 09:21.41 | Ludanek | **: checking to see if you are running the 'PoundKey' distribution ......NO (but it's not a problem) |
| 09:21.43 | Ludanek | **: contacting digium ....**: FAILED to contact 'https://register.digium.com/register.php' |
| 09:21.53 | Uatec | JT, that's a bummer, the only way? |
| 09:22.00 | Uatec | asterisk doesn't remember this information? |
| 09:22.15 | Uatec | the problem is, i don't know which one of my 20 client devices is the one i'm looking for |
| 09:22.17 | Uatec | :( |
| 09:22.47 | brookshire | Ludanek: if you email support@digium.com i believe they can manually generate a certificate for you |
| 09:23.55 | alpha232 | is that the commercial product? |
| 09:24.36 | brookshire | g729 has to be.. patent restrictions |
| 09:25.04 | Uatec | WTF? I'm getting "timeout::callback: Registering with timeout of 0 ms" in my phone logs |
| 09:25.11 | Uatec | immediatley followed by a timeout message |
| 09:25.50 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 09:28.08 | linxroute | you free |
| 09:28.11 | linxroute | g729 |
| 09:28.23 | linxroute | if you are not selling it |
| 09:28.34 | JT | you free? |
| 09:28.43 | alpha232 | JT: i'm determined to get the bri side working |
| 09:28.45 | brookshire | you mean.. commerically using it |
| 09:28.53 | linxroute | use |
| 09:28.54 | linxroute | sorry |
| 09:29.13 | alpha232 | i was just curious as to why it needed to register |
| 09:29.16 | linxroute | when |
| 09:29.23 | linxroute | actually it's free |
| 09:29.36 | JT | no, it's not |
| 09:29.44 | JT | stolen source code does not make it free to use |
| 09:29.55 | linxroute | which one is stolen ? |
| 09:30.09 | JT | the stupid "free" one that everyone keeps refering to |
| 09:30.31 | linxroute | well |
| 09:30.43 | linxroute | people already buy cards from digium |
| 09:30.55 | JT | it does not matter |
| 09:31.00 | linxroute | http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
| 09:31.02 | JT | digium do not own the g.729 patent rights |
| 09:31.04 | JT | and the |
| 09:31.06 | JT | " |
| 09:31.12 | JT | free" one is intel |
| 09:31.16 | JT | ... |
| 09:31.20 | linxroute | well you can go on and argues about it for days |
| 09:31.31 | JT | but i won't, it's extremely simple |
| 09:31.31 | linxroute | buy an intel commmercial license |
| 09:31.41 | JT | this is the WRONG channel to be promoting software piracy |
| 09:31.51 | linxroute | ? |
| 09:31.59 | linxroute | piracy ? |
| 09:32.13 | linxroute | he also posted it to intel development forum |
| 09:32.22 | linxroute | there's nothing piracy here |
| 09:32.24 | linxroute | well |
| 09:34.09 | Chris-NB | hi |
| 09:34.31 | Chris-NB | anyone knows how many peers can be in one meetme room? |
| 09:34.46 | brookshire | how many do you need? |
| 09:35.34 | Chris-NB | approx. 10~15 |
| 09:35.41 | brookshire | it will do that easy |
| 09:35.47 | Chris-NB | ok |
| 09:35.51 | brookshire | on a p4 |
| 09:35.56 | Chris-NB | do you know if there is a limitation? |
| 09:36.07 | brookshire | cpu is the only limitation i know of |
| 09:36.17 | JT | zap timing |
| 09:36.17 | brookshire | and i believe it's in the 100s |
| 09:36.21 | Chris-NB | ok |
| 09:36.25 | *** join/#asterisk BeeBuu (n=chatzill@219.132.188.27) |
| 09:36.36 | Chris-NB | it's a 1 Xeon Dualcore @ 3GHz |
| 09:37.46 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 09:38.24 | BeeBuu | exten => s,1,ChanIsAvail(Zap/2&Zap/1) |
| 09:39.01 | BeeBuu | if i got 120 lines... |
| 09:39.29 | BeeBuu | exten => s,1,ChanIsAvail(Zap/120&Zap/119...&Zap/1)? |
| 09:39.45 | linxroute | no |
| 09:39.53 | linxroute | you define into group |
| 09:40.01 | linxroute | those zap channel |
| 09:40.15 | linxroute | group 1 (1- 30) |
| 09:40.34 | *** join/#asterisk saftsack (n=saftsack@pD9E05C60.dip.t-dialin.net) |
| 09:40.56 | BeeBuu | i want to check which line is available,any suggestion? |
| 09:41.51 | linxroute | 120 lines |
| 09:41.56 | linxroute | mean 4 E1s |
| 09:42.04 | BeeBuu | yes. |
| 09:42.16 | BeeBuu | you are so smart! |
| 09:42.28 | JT | BeeBuu: you're crazy, use zap groups |
| 09:42.57 | JT | look up zap groups |
| 09:43.11 | BeeBuu | i want to check which line is available,any suggestion? |
| 09:43.19 | JT | and greenhand, that doesn't make sense in most western countries btw, at least not the way i think you want it to |
| 09:43.31 | JT | BeeBuu: no need, zap groups check line availability for you |
| 09:43.40 | tzafrir | Dial(Zap/g1) |
| 09:43.45 | tzafrir | that's it |
| 09:43.52 | linxroute | exten => s,1,ChanIsAvail(Zap/G2&Zap/G1) |
| 09:44.00 | JT | you don't really think that everyone with dozens of lines make complicated dialplans to work out if a zap channel is free? :) |
| 09:44.01 | linxroute | in zapata.conf |
| 09:44.15 | linxroute | under [channels] |
| 09:44.20 | linxroute | group=.. |
| 09:44.27 | linxroute | 1 2 3.. N |
| 09:44.29 | JT | linxroute: no |
| 09:44.34 | alpha232 | JT: is there a way to specify if it is terminal or circular? |
| 09:44.37 | JT | linxroute: just group ALL channels into a group |
| 09:44.42 | BeeBuu | exten => s,1,ChanIsAvail(Zap/G2&Zap/G1),what is the return? |
| 09:45.04 | linxroute | he has like 4 E1s |
| 09:45.06 | JT | alpha232: g/G/r/R |
| 09:45.09 | JT | linxroute: YES |
| 09:45.12 | *** join/#asterisk gallitzin (n=gallitzi@121-79-23-247-dsl.ispone.net.au) |
| 09:45.14 | JT | linxroute: put them in the one zao group |
| 09:45.17 | JT | zap |
| 09:45.28 | JT | if he wants to be able to use all of them from the one dial command |
| 09:45.29 | linxroute | yeap |
| 09:45.33 | linxroute | JT is right |
| 09:45.33 | alpha232 | JT: you actually understood the question? :D |
| 09:45.50 | linxroute | put it in to 1 :) |
| 09:46.02 | JT | alpha232: you can get zap to call sequentually up or down, and round robin up or down |
| 09:46.28 | linxroute | returns you with which group is avai |
| 09:46.34 | alpha232 | JT: nice... why do I have the feeling you've worked in telecom |
| 09:47.16 | JT | i'm not as much of an expert in telecommunications as i'd really like to be |
| 09:47.26 | BeeBuu | thanks JT,btw: "green hand " not make sense? |
| 09:47.46 | JT | BeeBuu: nup, "newbie" perhaps |
| 09:47.59 | linxroute | green horn , perhaps |
| 09:48.01 | linxroute | :) |
| 09:48.16 | alpha232 | JT: ah, i've suffered a bit with it and currently support my hotels mitel switch *ugh* |
| 09:48.21 | BeeBuu | linxroute:only return which group? |
| 09:48.31 | JT | alpha232: ooh, pass that knowledge to me |
| 09:48.41 | JT | if i knew mitel i could be making so much $$$ |
| 09:48.51 | alpha232 | JT: yeah but most of it is working your ass off |
| 09:48.52 | JT | there's a desperate shortage of mitel engineers here |
| 09:48.54 | linxroute | group that's not busy |
| 09:48.55 | JT | actually |
| 09:49.01 | JT | i think i mean broadsoft |
| 09:49.06 | alpha232 | lol |
| 09:52.07 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 09:52.55 | arcanine | why i get chanunavail error? and most of the time i cant pass through to call out and i get this message |
| 09:54.10 | linxroute | with SIP ? |
| 09:54.17 | linxroute | arcaine |
| 09:54.59 | alpha232 | JT: http://www.beronet.com/content/view/91/28/lang,en/ this gets me all excited |
| 09:55.14 | arcanine | SIP |
| 09:56.22 | JT | junghanns do them too |
| 09:56.27 | JT | they cost more than bri cards |
| 09:57.10 | alpha232 | i'm just thinking it would be cool ;) |
| 09:57.51 | JT | chan_mobile is more promising for lower budgets |
| 09:58.13 | alpha232 | but that requires a phone |
| 09:58.35 | JT | and? |
| 09:59.03 | alpha232 | it's cool, really it is |
| 10:04.24 | *** join/#asterisk KremenX (i=kremen@195.168.3.32) |
| 10:05.16 | alpha232 | JT: https://shop.beronet.com/product_info.php/products_id/304 thats what I need |
| 10:05.22 | alpha232 | woof! |
| 10:07.36 | linxroute | gsm sip gateway |
| 10:07.50 | JT | heh |
| 10:07.50 | linxroute | dual sim and LCR |
| 10:08.43 | linxroute | they use this kind of device alot in my country to provide |
| 10:08.49 | linxroute | long distance termination |
| 10:08.57 | linxroute | around 450$ |
| 10:09.19 | JT | the big ones can do many channels |
| 10:09.23 | JT | like an E1 or so worth |
| 10:09.32 | linxroute | yeap |
| 10:09.50 | linxroute | the chinese and taiwanese they make alot this type |
| 10:10.57 | alpha232 | linxroute: I want to see that type of thing take over here in the US |
| 10:11.06 | alpha232 | put all those roll over and free nights and weekends to use |
| 10:11.16 | JT | telcos also cancel the sims like crazy on those |
| 10:11.32 | JT | they royally screw up cell density provisioning |
| 10:11.42 | JT | and also lose margin for the telco |
| 10:12.01 | linxroute | yeap |
| 10:12.04 | alpha232 | pigfuckers 359Euro |
| 10:12.17 | linxroute | it's also illegal in my country |
| 10:12.26 | linxroute | but people uses it anyway |
| 10:12.26 | alpha232 | JT: i'm talking about for my personal use |
| 10:12.29 | linxroute | you can go here |
| 10:12.34 | linxroute | to buy it |
| 10:13.22 | alpha232 | OMG !!!! |
| 10:13.34 | alpha232 | Single Port HFC S0 $25Euro |
| 10:14.08 | alpha232 | and since it's software based... it should support whatever the software drivers support |
| 10:15.02 | JT | yes, you can get them cheaper on ebay |
| 10:15.13 | JT | there is no software to make them work with american BRI |
| 10:16.15 | alpha232 | argggh |
| 10:16.30 | alpha232 | should i just buy a fritzcard? |
| 10:18.22 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-16-252.lns10.syd7.internode.on.net) |
| 10:20.02 | linxroute | SC-385 GSM VoIP Gateway (2SIM, 2LAN) for GSM <-> SIP VoIP Conversion |
| 10:22.49 | alpha232 | interesting |
| 10:23.49 | alpha232 | linxroute: how about this one... |
| 10:23.53 | alpha232 | http://www.suncomm.com.tw/Product_detail.asp?P_ID=6008296&F_ID=1000344 |
| 10:23.55 | alpha232 | it does it all |
| 10:24.06 | alpha232 | 1LAN, 1FXS+1PSTN, SMS sending & receiving; |
| 10:24.15 | alpha232 | plus 1SIM |
| 10:24.26 | alpha232 | so you have GSM, FXO and FXS all in one box |
| 10:24.29 | linxroute | yeap |
| 10:24.56 | linxroute | and you can provide long distance termination service |
| 10:25.01 | linxroute | :) |
| 10:25.33 | alpha232 | linxroute: not what I need :D lol |
| 10:25.44 | linxroute | well |
| 10:25.47 | linxroute | 1PSTN |
| 10:25.50 | linxroute | does not mean |
| 10:25.51 | alpha232 | linxroute: i'm looking at making a unified PBX switch |
| 10:25.52 | linxroute | FXS |
| 10:26.00 | alpha232 | 1PSTN means FXO |
| 10:26.00 | linxroute | FXO |
| 10:26.03 | linxroute | sorry |
| 10:26.12 | linxroute | it's "Life line" port |
| 10:26.20 | alpha232 | hrrm |
| 10:26.29 | linxroute | means you can decice to call over Ip or switch to pstn |
| 10:26.48 | linxroute | i also got confused about this bit be4 |
| 10:27.12 | alpha232 | that seems vague |
| 10:27.19 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-16-252.lns10.syd7.internode.on.net) |
| 10:27.21 | linxroute | those chinese guys |
| 10:27.23 | alpha232 | because this box would be ideal |
| 10:27.43 | linxroute | always use "life line" or PSTN |
| 10:27.46 | linxroute | not FXO |
| 10:28.15 | linxroute | it's turned out to be you can only make a call from the FXS port to that PSTN line |
| 10:28.29 | alpha232 | not from sip |
| 10:28.33 | alpha232 | sonabitch |
| 10:30.11 | alpha232 | i'm disappointed now |
| 10:31.23 | linxroute | http://star-net.en.alibaba.com/product/0/50080293/SVX8016_voip_pbx.html |
| 10:31.30 | linxroute | have all your needs |
| 10:33.26 | alpha232 | except it doesn't run * :D |
| 10:36.54 | alpha232 | linxroute: i love the idea of using GSM for pay phones |
| 10:44.05 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 10:46.00 | JT | POTS not PSTN btw, PSTN refers to the whole network and there are a variety of ways to connect to the PSTN, one of them being POTS |
| 10:46.50 | alpha232 | JT: we're talking about the terms used by the mfgs |
| 10:47.39 | JT | heh |
| 10:49.41 | alpha232 | 06:48 -!- TrentCreek [i=GeekBoy@cpe-70-117-207-168.rgv.res.rr.com] has quit [Read error: 110 (Connection timed out)] |
| 10:49.45 | alpha232 | thank god |
| 10:52.57 | linxroute | oh starnet |
| 10:53.00 | linxroute | is asterisk |
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| 10:57.27 | badcfe | hello. i see with "manager show command originate" that i can pass channel variables. what format should be used? |
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| 11:18.53 | BeeBuu | <PROTECTED> |
| 11:20.17 | BeeBuu | <PROTECTED> |
| 11:20.19 | BeeBuu | <PROTECTED> |
| 11:20.21 | BeeBuu | <PROTECTED> |
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| 11:27.24 | tzafrir | <PROTECTED> |
| 11:27.34 | tzafrir | or G2, or r2 or R2 |
| 11:28.25 | ai-a | Anyone know of a c++ / php / whatever scripting shell (wrapper, not console shell) that wraps the manager tcp/ip stream for development... save me writing something.. if not, i will write the project myself. |
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| 11:29.14 | *** part/#asterisk aikanaro79 (n=boo@89-180-62-19.net.novis.pt) |
| 11:31.11 | BeeBuu | thnaks,all,c u. |
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| 12:05.09 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 12:29.51 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:33.49 | anv1 | is it possible to compile "chan_mobile" on 1.4? |
| 12:38.17 | *** join/#asterisk linxroute (n=VietPhon@222.252.108.5) |
| 12:39.42 | agx | anv1, i think that the version for 1.4 is called chan_blueetooth |
| 12:39.46 | phix | sooo, should I use t or T with Dial()? |
| 12:42.08 | [TK]D-Fender | phix: No, you should use decent phones with dedicated transfer functionality. |
| 12:43.04 | linxroute | still on attended transfer , phix :) |
| 12:44.01 | destructure | ai-a: are you looking for an api to interface with ami, or are you looking for a proxy? |
| 12:44.27 | destructure | or are you looking for an actual shell, which is a neat idea |
| 12:44.57 | destructure | that would be an interesting use for fuse, actually |
| 12:44.59 | destructure | ls /channels |
| 12:45.02 | destructure | I love it! |
| 12:45.09 | anv1 | agx: I've downloaded svn version of 1.4, I configure, make menuconfig but I can't find this channel... |
| 12:45.13 | ai-a | an api mainly in c++ to have a thread handling the events and keeping a track of the pbx. so my front app can query these quite quickly and disaply monitor information. |
| 12:45.27 | phix | [TK]D-Fender: oh |
| 12:45.40 | destructure | ah yes. I use rami, which is satisfactory, but not great, but it's for ruby, not c++ |
| 12:45.49 | ai-a | destructure: you could write a device that is mounted in /pbx :) |
| 12:46.34 | anv1 | I also checked in asterisk-addons-1.4 |
| 12:46.35 | phix | [TK]D-Fender: ok so, t means the called party can transfer calls, meaning if somebody on a PSTN line calls me they can transfer me to a local extension on my asterisl box? |
| 12:46.35 | destructure | yeah, exactly ls /pbx/channels/iax; rm /pbx/meetme/1001/<tab> |
| 12:46.46 | ai-a | yep :) |
| 12:46.47 | phix | or does it mean that if someone calls me, the nI can transfer them |
| 12:47.16 | ai-a | I'll check out rami, however I am still not sure which language i want to use. But ruby would never be my first choice. |
| 12:47.31 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 12:47.42 | phix | [TK]D-Fender: ? |
| 12:48.34 | [TK]D-Fender | phix: t & T are self explanatory. One ltes the CALLER transfer, the other the CALLEE |
| 12:48.47 | destructure | ai-a: I take it you've looked at the wiki? asterisk manager examples has a c++ one. C++ has gone the way of the lightsaber though, hasn't it? |
| 12:49.09 | ai-a | destructure: my choice is LOLcode http://lolcode.com/ |
| 12:49.20 | destructure | o rly? |
| 12:49.26 | russellb | ya rly. |
| 12:49.34 | linxroute | is there any something out there similar to switchbox |
| 12:49.36 | ai-a | just started wondering about it. not researching yet. get destracated with web development. |
| 12:50.12 | destructure | maybe check out rami then. and switch your web dev to rails while at it |
| 12:50.29 | ai-a | Yuck!. |
| 12:50.29 | tzafrir | ai-a, there's an existing php wrapper for the manager. I forgot its name |
| 12:50.35 | tzafrir | Not sure about C++ |
| 12:50.46 | ai-a | tzafrir got the examples pages. will have a look... |
| 12:51.17 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 12:51.32 | ai-a | next question is.. manager is bindaddr 127.0.0.1, anyway to connect to the manager using openssh when its locked down to the localhost ? |
| 12:51.37 | phix | [TK]D-Fender: so does that mean any body who calls me can transfer me? cause I dont want that |
| 12:51.50 | destructure | of course! check out remote port forwarding |
| 12:52.01 | [TK]D-Fender | phix: means OMG read the instructions to figure out WHICH IS WHICH. |
| 12:52.08 | phix | [TK]D-Fender: I did |
| 12:52.10 | phix | I am confused |
| 12:52.32 | [TK]D-Fender | phix: t & T are self explanatory. One lets the CALLER transfer, the other the CALLEE <---- what don't you get here? |
| 12:52.45 | destructure | ai-i:ssh -R |
| 12:53.09 | destructure | you could also use -D for socks, but a simple port forward will do in this case |
| 12:54.19 | phix | [TK]D-Fender: so if I call someone I am the caller? and if some one else out side of my asterisk box calls me then they are the caller? |
| 12:54.37 | linxroute | :) |
| 12:54.55 | [TK]D-Fender | phix: U R SMRT |
| 12:55.03 | phix | I mean I want to transfer both calls I make and calls I receive but I dont want people outside of my asterisk box to be able to transfer me |
| 12:55.23 | destructure | actually, I got it backwards. I always do. it's a local forward |
| 12:55.39 | destructure | you want the local port to go to the remote server |
| 12:56.31 | rantsh | hi people |
| 12:57.11 | linxroute | way to go, phix mate |
| 12:57.39 | rantsh | since yesterday I've been trying to make my queue record calls through mixmonitor, per [tk]d-fender suggestion I installed sox and it still doesn't work |
| 12:58.12 | rantsh | any suggestion anyone? it is recording BUT it creates 2 files in and out |
| 12:58.14 | linxroute | if attended x fer does not workout for you try , set maximum time for ring then move to an option to leave a voicemail or go to the operator |
| 12:59.06 | linxroute | and the best one just like TK said |
| 12:59.14 | linxroute | buy an Ip phone |
| 12:59.46 | destructure | rantsh: why would you want them mixed in real time? it's a waste of cpu |
| 13:01.07 | rantsh | destructure: isn't it better than having a cpu spike on a post-call mix ? ? ? |
| 13:01.18 | destructure | when will you listen to the audio? |
| 13:01.25 | *** join/#asterisk nightrid3r (n=nnscript@d54C0303C.access.telenet.be) |
| 13:02.08 | [TK]D-Fender | destructure: it isn't mixed in real-time |
| 13:02.42 | [TK]D-Fender | destructure: always post, and it is better the way it is since its done only once and cleans up after itself. |
| 13:02.43 | rantsh | destructure if a customer has a complain about the service, |
| 13:02.58 | ai-a | rantsh: have a script running that joins the audios together reading from a queue, with a nice of 19, then your a happy chappy. |
| 13:03.34 | ai-a | post-call inserts into a db. |
| 13:03.39 | rantsh | yeah, but... doesn't it defeat the purpouse? mixmonitor is supposed to be mor efficient |
| 13:03.39 | destructure | I see. I should have looked at the summary first. Still, I think I'd do it when the customer complains, on another box |
| 13:04.02 | ai-a | rantsh: then use it :) |
| 13:04.07 | rantsh | I am |
| 13:04.28 | ai-a | destructure: ssh -L works... thanks. |
| 13:04.36 | rantsh | but it's not mixing the audio, the weirdest thing is that on my test box it works just fine |
| 13:04.52 | *** join/#asterisk bantu (n=Miranda@p54A31294.dip0.t-ipconnect.de) |
| 13:05.03 | nightrid3r | i'm looking for an asterisk cd that allows an uber dumbfuck (tm) to manage an asteriskbox |
| 13:05.54 | ai-a | nightrid3r: AsteriskNow. |
| 13:06.01 | ai-a | FreePBX |
| 13:06.10 | ai-a | there are existing channels on this erver for these products too. |
| 13:06.20 | destructure | [TK]D-Fender: mixmonitor says "Record a call and mix the audio during the recording". Am I missing something? |
| 13:06.30 | ai-a | However, the 'dumbfuck' will get fucked up as they prevent a lot of stuff. |
| 13:06.44 | linxroute | does asterisknow or freepbx compareable to switchvox ? |
| 13:06.48 | linxroute | just wondering |
| 13:06.51 | ai-a | nightrid3r: get the asterisk book (its free online). and it will teach you everything you require. |
| 13:06.57 | destructure | ai-i: great. for added win, put the local forward in ~/.ssh/config |
| 13:07.01 | linxroute | have anyone used switchvox |
| 13:07.19 | _x86_ | morning |
| 13:07.39 | tzafrir | what do you mean by "compatible to switchvox"? |
| 13:07.41 | ai-a | destructure: well, i willb e adding openssh to my project and fowarding it internally i think. requesting user login from the project to gain access to the box's manager port. |
| 13:08.16 | linxroute | sorry |
| 13:08.19 | linxroute | compareable |
| 13:08.25 | linxroute | my english is bad |
| 13:08.27 | tzafrir | why would you want to use a non-free software, or strive to be compatible with it? why get yourself into licensing overhead? |
| 13:08.46 | [TK]D-Fender | destructure: I could be mistaken, but it was as I described last I checked. Go verify I guess. |
| 13:08.58 | linxroute | i meant in term of it's features |
| 13:08.58 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 13:09.14 | [TK]D-Fender | linxroute: go LOOK for yourself |
| 13:09.39 | destructure | ai-a:it's a reasonable approach. I wouldn't want people randomly connecting to ami |
| 13:09.55 | linxroute | i'haven't got a chance to use switchvox |
| 13:10.08 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 13:10.21 | linxroute | so just asking if anyone happens to used it |
| 13:10.40 | [TK]D-Fender | linxroute: People using GUI's don't tend to stick around here. |
| 13:10.53 | linxroute | i don't |
| 13:10.55 | linxroute | too |
| 13:10.59 | [TK]D-Fender | linxroute: Except the few FreePBX schmucks who just don't get it... |
| 13:15.14 | Mw3 | does somebody around here use pri with digium hardware? |
| 13:15.33 | destructure | a little |
| 13:15.57 | [TK]D-Fender | Mw3: Now try being speicific with your questions... |
| 13:16.27 | destructure | heh |
| 13:16.36 | Mw3 | [TK]D-Fender: i would be curious about the uptime of the system and the number below the IRQ row in zap show status |
| 13:16.51 | destructure | corollary of "don't ask to ask" |
| 13:17.28 | [TK]D-Fender | destructure: For a relative newcomer you are remarkably astute. |
| 13:17.41 | s34n | is anyone here connecting to a Metaswitch? |
| 13:17.54 | coppice | If you need to talk to someone, don't. Hesitate to call us with any problems you have. |
| 13:18.21 | destructure | [TK]D-Fender: thank you. it's the power of lurking |
| 13:18.52 | [TK]D-Fender | coppice: Ah the power of punctuation... |
| 13:19.37 | destructure | Mw3: of the two boxes I have in prod, one has been up 53 days with no trouble, the other is a constant mess. That seems to be the way it goes |
| 13:19.50 | destructure | I think the problem is on the remote end |
| 13:20.20 | destructure | TE405P's |
| 13:21.51 | *** join/#asterisk oej (n=olle@64.251.114.2) |
| 13:23.42 | Mw3 | i have a te110p which was working prefectly. now the system has 4 days uptime and the irq loss count is at 15972 |
| 13:23.54 | Mw3 | and i have trouble with incoming faxes |
| 13:24.27 | [TK]D-Fender | Mw3: what ver of * & zaptel? pastebin the following : "dmesg" , and "cat /proc/interrupts" |
| 13:24.28 | [TK]D-Fender | ~ |
| 13:24.30 | [TK]D-Fender | pb |
| 13:24.31 | [TK]D-Fender | ~pb |
| 13:24.32 | jbot | A Pastebin is a place to paste your stuff 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 |
| 13:24.33 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 13:24.55 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 13:25.39 | Mw3 | [TK]D-Fender: ok, a moment. there is no shared irq by the way. the asterisk is 1.2.9.1 and the zap is 1.2.6 |
| 13:26.14 | [TK]D-Fender | Mw3: there have been major changes to zaptel since that time as far as interrupts handling is concerned. |
| 13:26.26 | Mw3 | shall i try upgrading? |
| 13:26.36 | [TK]D-Fender | Mw3: Of course the version numbers are disturbingly familiar.... |
| 13:26.45 | [TK]D-Fender | Mw3: I would highly recommend it. |
| 13:27.16 | Corydon76-dig | 'tis always a good idea to upgrade to the latest, to ensure that the bug has not already been fixed |
| 13:27.53 | Mw3 | [TK]D-Fender: ok im going to try it. can hyperthreading be harmful? the old server had 2 cpu with ht disabled, the new has only 1 cpu but with ht enabled |
| 13:28.03 | *** join/#asterisk Darthclue (n=e054502@165.111.2.149) |
| 13:29.07 | coppice | hyperthreading tends to slow things down, by increasing the amount of cache thrashing |
| 13:29.56 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:29.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:34.08 | *** join/#asterisk fenlander (n=fenlande@80.169.36.194) |
| 13:34.09 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:34.26 | Mw3 | thanks. then ill try disabling ht and upgrading zaptel and asterisk |
| 13:34.29 | *** join/#asterisk saftsack (n=saftsack@IP-213157005203.dialin.heagmedianet.de) |
| 13:35.48 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 13:36.36 | tzanger | I've got some asterisk call centre questions... never done it before, can anyone answer a few questions? |
| 13:37.10 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 13:37.51 | tzanger | I understand that an agent can be a member of multiple queues, and that agents can see which queue an incoming call is for through distinctive ring or whatever (polycom phones) |
| 13:37.56 | Mimmus | hi, as Asterisk doesn't support SIPoverTCP, can I use OpenSER for TCP <--> UDP |
| 13:37.58 | Mimmus | +conversion? |
| 13:38.17 | tzanger | I also am pretty sure that asterisk can provide queue-specific on-hold music |
| 13:38.38 | [TK]D-Fender | tzanger: every queue can have its own MWI class. |
| 13:39.08 | tzanger | [TK]D-Fender: ok so that's no problem for asterisk then |
| 13:39.21 | tzanger | is there such a thing as a "queue" voicemail box? |
| 13:39.28 | [TK]D-Fender | tzanger: As for identifying calls, the only real way is to mangle CID, or if using Local to call your "agents" you could ass a distintive ring |
| 13:40.02 | anv1 | anybody knows how to compile chan_mobile / chan_cellphone on asterisk 1.4? |
| 13:40.05 | tzanger | [TK]D-Fender: through local/ could I also not push a message to the polycom phone to show it up in a specific way on their phone? |
| 13:40.13 | [TK]D-Fender | tzanger: No, You can have the Queue leave based on lack of agent availability, or on timeout at which point the next line of dialplan executes |
| 13:40.26 | [TK]D-Fender | tzanger: Or use the 'exitcontext" where you can provider a 1-digit IVR |
| 13:40.29 | anv1 | chan_cellphone seems to not exists anymore, there is chan_mobile, but I can't compile it. |
| 13:40.33 | tzanger | [TK]D-Fender: ok, but I can have the polycom phone watch multiple "extensions" for voicemail |
| 13:40.46 | [TK]D-Fender | tzanger: in Local you can do whatever you can imagine doing already |
| 13:40.56 | [TK]D-Fender | tzanger: Indeed. |
| 13:40.58 | tzanger | [TK]D-Fender: *nods* |
| 13:41.26 | [TK]D-Fender | tzanger: I have my Polycoms Idle browser giving live stats on 2 queues, and 4 agents every 10s |
| 13:43.42 | tzanger | nice |
| 13:47.40 | [TK]D-Fender | anv1: http://svn.digium.com/view/asterisk-addons/trunk/doc/chan_mobile.txt?view=co |
| 13:48.08 | [TK]D-Fender | anv1: And as you'll see here there have been numerous issues with compiling it : http://www.google.ca/search?hl=en&q=chan_mobile+asterisk+compile&btnG=Google+Search&meta= |
| 13:48.21 | [TK]D-Fender | anv1: Make sure you follow the first guide, then rever to the second link |
| 13:49.15 | anv1 | [TK]D-Fender: ok, I've tried some patches but will follow your instructions, thanks. |
| 13:52.19 | *** join/#asterisk Mazzachre (n=mr@195.74.76.71) |
| 13:52.54 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 13:52.58 | Mw3 | another issue. we have a linksys pap2. its power led turned red (it was blue) and the ethernet led is constantly on (blue) even with unplugged ethernet cable. i tried to change the power adapter, but nothing changed. any idea? |
| 13:53.38 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 13:54.17 | *** join/#asterisk saftsack (n=saftsack@pD9E05C60.dip.t-dialin.net) |
| 13:55.26 | *** join/#asterisk gardo (n=gardo@121.97.251.60) |
| 13:55.29 | *** join/#asterisk oej (n=olle@64.251.114.2) |
| 13:55.51 | Mazzachre | We have an inbound IVR system for out support in asterisk, but we would like to draw out statistics from it like what people are telling the IVR and how long each consulent is on the phone and how many people give up (and hang up) and how many end up on the voicemail (If they sit and listen to our music for 1hour) Is asterisk capable of that?? Or should I convert to a FS for that application??? |
| 13:55.55 | [TK]D-Fender | Mw3: try the factory reset switch |
| 13:56.34 | [TK]D-Fender | Mazzachre: If you're using queues then you already have a full log |
| 13:56.54 | [TK]D-Fender | Mazzachre: You only need to tie the log of IVR choies to that. |
| 13:57.09 | [TK]D-Fender | Mazzachre: Which is something you'll have to script yourself into your dialplan. |
| 13:57.13 | tzanger | [TK]D-Fender: if an agents answers a queue call (so caller has queue-specific moh, etc.) and then transfers the call outside the queue, which gets transfered to someone else, transfered to someone else again, etc... does that MOH class "stick" so long as it doesn't get focibly changed in the dialplan? |
| 13:58.58 | Mazzachre | [TK]D-Fender: Can I get asterisk to tell it to a socket/app somewhere instead of having to parse its logfiles in a loop??? |
| 14:00.35 | [TK]D-Fender | tzanger: That follows the transferer |
| 14:01.05 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.227) |
| 14:01.06 | [TK]D-Fender | Mazzachre: Not sure if AMI is "end-to-end" enough for that. probably not viable |
| 14:01.23 | tzanger | ... follows the *transferer* ? |
| 14:01.25 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 14:01.30 | tzanger | the one who did the transfer? |
| 14:01.33 | tmccrary | Does anyone have recommendations for a good FXO gateway w/8-12 ports? |
| 14:02.34 | tmccrary | I've tried two so far and both are not very good: Audiocodes and Grandstream |
| 14:02.46 | [TK]D-Fender | tzanger: I answer the call and I initiate a tranfer, they get MY Moh |
| 14:03.09 | [TK]D-Fender | tmccrary: What happened with the AudioCodes? |
| 14:03.17 | Mw3 | [TK]D-Fender: can you tell me where is it on the pap2? |
| 14:03.18 | *** join/#asterisk RockHound (n=rockhoun@85.183.138.242) |
| 14:03.27 | [TK]D-Fender | tmccrary: Meditrix has some, but I might be iffy on that. Their FXS is nice though. |
| 14:03.35 | tzanger | [TK]D-Fender: can that be overridden in the dialplan? i.e. set some var to the queue moh class, then set the moh class before dialing? |
| 14:03.41 | destructure | Mazzachre: you could use ami events to calculate that, for example, using nextexten, but you'd have to do the calculations yourself. you could use cdr or something if it's not already in use |
| 14:03.41 | [TK]D-Fender | Mw3: No, thats why it comes with a manual :) |
| 14:03.44 | coppice | audiocodes stuff gets a least one thumbs up |
| 14:03.56 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 14:03.58 | [TK]D-Fender | tzanger: I was thinking of that, but not sure |
| 14:04.13 | Mw3 | [TK]D-Fender: i dissassembled the device, looking at the pcb, but there is no god damned switch on this :( |
| 14:04.30 | [TK]D-Fender | tmccrary: Can always try Patton. Heard little, but at least largely good. |
| 14:04.32 | tmccrary | D-Fender: Strange sounds like distortion, intermittant cut outs and more. The grandstream FXO products has done much better, and these were tested at multiple locations, LA, somewhere in texas, chicago and detroit |
| 14:04.36 | tzanger | wtf wy would I be getting CDR not posted for simple CDR? |
| 14:04.44 | tzanger | [Oct 30 10:04:33] NOTICE[864]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/benshaw-081ba9c8' not posted |
| 14:04.50 | tmccrary | but the grandstream has its own quality "quirks" |
| 14:04.58 | [TK]D-Fender | ~gs |
| 14:04.59 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 14:05.00 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 14:05.02 | coppice | mediatrix stuff is seriously buggy |
| 14:05.18 | [TK]D-Fender | coppice: Their PRI gateway is a SHITTY Cisco wannabe |
| 14:05.30 | [TK]D-Fender | coppice: Cryptic PITA |
| 14:05.35 | tmccrary | yeah, Grandstream stuff is cheap... but it is still much better than Audiocodes |
| 14:05.43 | coppice | mediatrix T.38 is the weirdest I have found |
| 14:05.52 | tzanger | coppice: heh |
| 14:05.53 | RockHound | hi everyone ... if I get really high values from zttest, what could this be causing? using ztdummy which is using RTC. |
| 14:05.55 | tmccrary | If grandstream is cheap junk, audiocodes is garbage :) |
| 14:05.58 | [TK]D-Fender | coppice: WITHOUT the tzatziki |
| 14:06.01 | coppice | audiocodes is very good stuff. what problems have you had? |
| 14:06.10 | tzanger | mediatrix with tzatziki? |
| 14:06.20 | tmccrary | Strange sounds like distortion, intermittant cut outs and more. these were tested at multiple locations, LA, somewhere in texas, chicago and detroit |
| 14:06.29 | *** join/#asterisk flujan (n=flujan@200.160.115.20) |
| 14:06.32 | tmccrary | all had the same problem an we tested with 4 different units |
| 14:06.38 | tmccrary | I have a stack of those bastards on my shelf |
| 14:06.41 | [TK]D-Fender | tzanger: Acronym humour.... jus let it slide :) |
| 14:06.43 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 14:06.44 | tmccrary | and Audiocodes tech support was no help |
| 14:06.52 | [TK]D-Fender | Katty: Mew. |
| 14:07.07 | Katty | tuesdays suck :< |
| 14:07.08 | Katty | llane is down. |
| 14:07.17 | coppice | I've never used the audiocodes boxes, but I've used several things with the audiocodes silicon and software. they do a much better job that most people |
| 14:07.23 | Katty | [TK]D-Fender: herro. |
| 14:07.23 | flujan | hi guys... I am having a serious problem with asterisk... I am dialing through a sip trunk... When the end point ( a telephone on the PSTN ) hang ups asterisk keeps the connection with it. |
| 14:07.32 | flujan | someone here experiencing this issue? |
| 14:07.33 | tmccrary | coppice: that's great, but their FXO products are garbage |
| 14:08.22 | flujan | It is keeping the line of the end point busy... It is a damned behavior to a call-center... :( |
| 14:08.55 | tzanger | flujan: analog connection to PSTN? |
| 14:09.19 | flujan | tzanger: both, I have a sip trunk and a E1 link. both shows the same behavior... :( |
| 14:09.36 | tzanger | is your E1 CAS or CCS? |
| 14:10.05 | coppice | tmccrary: well, the other boxes just use the audiocodes designs without change. functionally they should be the same. dunno about the build quality, but they doesn't appear to be your complaint |
| 14:10.25 | tzanger | flujan: i.e. is it PRI or just channel aggregation |
| 14:10.30 | RockHound | --- Results after 75 passes --- |
| 14:10.30 | RockHound | Best: 0.000 -- Worst: -207.051 -- Average: -205.123491, Difference: 405.123492 |
| 14:10.54 | flujan | PRI ISDN the trunk |
| 14:10.56 | flujan | OPS |
| 14:10.59 | flujan | the E1 Link |
| 14:11.11 | tzanger | flujan: you are not getting disconnect supervision with E1/PRI?? |
| 14:11.12 | flujan | and a sip trunk just behave like a normal phone... |
| 14:11.24 | tzanger | flujan: sounds like a switch problem on the other side, or a config mismatch |
| 14:11.32 | tzanger | flujan: a PRI debug might be helpful in this case |
| 14:11.43 | flujan | ok and about the sip trunk? |
| 14:11.51 | tmccrary | coppice: The fact remains the same, we tested Audiocodes in multiple locations, using mulitple gateways and got the same poor results: Sound problems, intermittant cut outs in audio and more. Personally, I think Audiocodes is some kind of scam, like some out found a bunch of old audio hardware in a dumpster and rigged it up to "kind of" work with voip :) |
| 14:12.00 | flujan | My softphone calls to asterisk... the extension leads to a agi script that place the call... |
| 14:12.47 | tmccrary | So far, digium hardware has been the only reliable FXO equipment we've put in use |
| 14:12.49 | agx | tmccrary, i always supposed every telephone device around has some audiocodec chip in it... |
| 14:13.08 | tmccrary | agx: k |
| 14:13.12 | flujan | guys, here is my dialplan... http://pastie.caboo.se/112318 |
| 14:13.19 | coppice | tmccrary: huh? the digium T1/E1 cards are OK, but the TDM cards are terrible |
| 14:13.39 | tmccrary | We've used both and both have worked consistantly, that's always a plus :) |
| 14:13.41 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:13.46 | tmccrary | Can't say the same for Audiocodes |
| 14:13.50 | [TK]D-Fender | flujan: lol, how completely USELESS |
| 14:13.53 | *** part/#asterisk Rawplayer (n=kevin@cp108757-a.landg1.lb.home.nl) |
| 14:13.54 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:13.58 | [TK]D-Fender | flujan: says NOTHING |
| 14:14.11 | flujan | [TK]D-Fender: ? |
| 14:14.26 | flujan | this script just dials for me... so I do not need to create dialplans... |
| 14:14.27 | flujan | :D |
| 14:14.28 | [TK]D-Fender | flujan: You cll an AGI and after that? Who knows. |
| 14:14.56 | flujan | cll? |
| 14:15.19 | flujan | the AGI makes asterisk to dial to a phone number |
| 14:15.19 | Qwell | call |
| 14:15.34 | *** join/#asterisk littleball (n=littleba@bb220-255-16-159.singnet.com.sg) |
| 14:15.50 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 14:16.09 | *** join/#asterisk Op3r (n=edwin@125.212.117.210) |
| 14:16.17 | flujan | well, it calls a agi that DIAL to the end point IF that extension have permission to dial to this kind of number. i.e.: cell phones and stuff. |
| 14:16.19 | littleball | hello, it seems that the installation of 1.4.* is a bit different thn 1.2.*. any document available? |
| 14:16.53 | Darthclue | flujan, you'll need to pastebin the agi so that we know what it is trying to do if you want help. |
| 14:17.40 | tzanger | when it comes to CRM integration with asterisk, what's generally used these days? I've used snapanumber but they seem to have fallen away now |
| 14:17.45 | tzanger | I see HUD and HUDLite |
| 14:17.45 | littleball | hi, in my new installation of 1.4.*, why PRI SHOW is not available on the CLI console? |
| 14:18.11 | tzanger | littleball: did you build libpri before zaptel and asterisk? |
| 14:19.01 | littleball | yes. tzanger |
| 14:19.24 | littleball | this is my first tiem to compile 1.4. |
| 14:19.28 | tzanger | hmm |
| 14:19.33 | Qwell | zaptel and libpri 1.4? |
| 14:19.49 | flujan | guys, here is the output produced by the agi script http://pastie.caboo.se/112322 |
| 14:19.54 | littleball | Qwell, yes |
| 14:19.59 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 14:20.17 | Qwell | and you re-ran the configure script afterwards? |
| 14:20.36 | littleball | Qwell, under zaptel? |
| 14:20.40 | Qwell | asterisk |
| 14:20.44 | littleball | yes. |
| 14:20.52 | littleball | i did run make config |
| 14:20.53 | Qwell | and did it find zaptel and libpri? |
| 14:21.20 | *** part/#asterisk Mazzachre (n=mr@195.74.76.71) |
| 14:21.26 | littleball | under asterisk, the make config does not display anything |
| 14:21.33 | littleball | which is not like under asterisk 1.2 |
| 14:21.40 | Qwell | make config != configure script |
| 14:21.48 | agx | tzanger, HUD is not a CRM (customer relationship management) but a CTI (Client Telephone Interface) |
| 14:22.02 | Qwell | <rant>Damn you HP, I want my laptop back</rant> |
| 14:22.15 | Qwell | s/rant/whine/g |
| 14:22.18 | littleball | Qwell, oh, yes |
| 14:22.20 | tzanger | agx: no I realize that, what I'm saying is what CRM integration software do people use with asterisk |
| 14:22.25 | littleball | i did ./configure before make |
| 14:22.26 | tzanger | HUD is a CRM integration tool |
| 14:22.32 | Qwell | after installing zaptel and libpri? |
| 14:22.37 | littleball | yes |
| 14:22.38 | agx | tzanger, well i've a very bad opinion about Sugar and VTiger |
| 14:22.48 | littleball | i make libpri, then zaptel , then asterisk |
| 14:23.09 | littleball | maybe i can re-compile everythign again. can? |
| 14:23.17 | agx | tzanger, you want to check OutCall or Asterisk Desktop Manager if you want to pop-up a web page on incoming calls |
| 14:23.42 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:26.11 | tzanger | agx: ahh okay |
| 14:26.49 | littleball | Qwell, if i re-compile everything again, will it work? Do i need to uninstall anything first? |
| 14:29.46 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 14:30.09 | Katty | someone has been in my office... |
| 14:30.20 | Katty | and they went through the software... |
| 14:30.27 | _x86_ | agx: know of one that supports MacOS X? |
| 14:30.29 | cpm | find anything good? |
| 14:30.34 | Darthclue | It wasn't me! |
| 14:30.42 | _x86_ | tzanger: note: dont google "outcall"... i just made that mistake ;) |
| 14:30.45 | _x86_ | NSFW heh |
| 14:31.56 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 14:32.37 | RockHound | anyone in here who is willing to help with my timing problem? |
| 14:32.58 | ai-a | RockHound: replace the timing belt. |
| 14:33.09 | ai-a | needs doing every 60,000 miles. |
| 14:33.24 | RockHound | LOL |
| 14:33.39 | RockHound | well .. to be more exact, I have a ztdummy which is going nuts |
| 14:33.43 | Darthclue | and where might a find a minute to mile converter? |
| 14:33.54 | Katty | oh, look |
| 14:33.56 | Katty | it's Darthclue |
| 14:33.57 | RockHound | zttest gives me ~200% bad |
| 14:34.07 | RockHound | 0 good |
| 14:34.09 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:34.22 | Darthclue | where? |
| 14:34.45 | Katty | hiding behind the muffin table. |
| 14:34.56 | Darthclue | oh, ok |
| 14:35.54 | s34n | someone brought muffins? |
| 14:36.16 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 14:36.18 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:36.22 | Katty | s34n: no. |
| 14:36.25 | *** join/#asterisk Cresl1n_ (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 14:36.25 | *** mode/#asterisk [+o Cresl1n_] by ChanServ |
| 14:37.28 | CunningPike | Katty: Right back atcha! |
| 14:37.36 | CunningPike | Katty: How you doin'? |
| 14:37.38 | CunningPike | Long time |
| 14:37.46 | Katty | ^_- |
| 14:37.49 | Katty | i was here all last week... |
| 14:37.54 | Katty | i uhh, i think. |
| 14:38.00 | Katty | CunningPike: i'm good. |
| 14:38.12 | Katty | oh ah. |
| 14:38.20 | Katty | you lurker you. |
| 14:38.29 | CunningPike | ;) |
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| 14:52.23 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 14:53.47 | funxion | anyone with familiar with the flash operator panel? |
| 14:54.06 | destructure | ick |
| 14:54.55 | funxion | destructure do youhave other suggestions? |
| 14:55.00 | *** join/#asterisk BBHoss (n=hoss@proxy-srv.uah.edu) |
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| 14:58.18 | destructure | funxion: what do you need it for? |
| 14:58.27 | [TK]D-Fender | funxion: What exact features are you looking for? |
| 14:59.06 | funxion | I need it for a receptionist to see who's online transfer calls etc.. |
| 14:59.30 | funxion | I have a st2030 with the expansion module but have had nothing but problems with it |
| 15:00.11 | funxion | I'm going to replace the phone and use some sort of app to get the functionality going |
| 15:00.20 | [TK]D-Fender | funxion: How many people to watch? |
| 15:00.25 | funxion | 60-70 |
| 15:00.31 | *** join/#asterisk BBHoss (n=hoss@proxy-srv.uah.edu) |
| 15:00.44 | funxion | no conf or queues |
| 15:00.50 | [TK]D-Fender | funxion: May be worth considering the Aastra 57i + LCD attendant module |
| 15:00.59 | funxion | very basic |
| 15:01.01 | funxion | hmmm |
| 15:01.02 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 15:01.09 | *** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 15:01.16 | funxion | I need to get the functionality working today |
| 15:01.22 | funxion | the pbx is in paris |
| 15:01.39 | [TK]D-Fender | funxion: Oh... |
| 15:01.49 | funxion | the Aastra is an option but not at the moment |
| 15:01.55 | [TK]D-Fender | funxion: Might be best to work up with FOP |
| 15:02.06 | funxion | thats what I was thinking |
| 15:02.11 | destructure | if you are just monitoring, it should light up ok |
| 15:02.35 | funxion | I checked out the asteriskguru switchboard which is good but has too much functionality |
| 15:02.51 | funxion | I like FOP because I can control the access I give the operator |
| 15:03.13 | funxion | [TK]D-Fender are you familiar with FOP |
| 15:03.24 | fiXXXerMet | My company is considering replacing our current phone solution with asterisk. Is there a comparison somewhere of asterisk/trixbox/buying asterisk from digium/etc? |
| 15:03.40 | [TK]D-Fender | funxion: Only passingly. Have never installed or really used it myself. |
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| 15:03.43 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 15:03.45 | funxion | o |
| 15:03.52 | [TK]D-Fender | fiXXXerMet: Not really. |
| 15:03.55 | funxion | limited support |
| 15:04.01 | denon | fiXXXerMet: don't bother with trixbox .. and if you have to ask, you're probably best off getting Business Edition from digium |
| 15:04.15 | [TK]D-Fender | fiXXXerMet: What do you want to DO? |
| 15:05.02 | fiXXXerMet | For starters, we want to setup a call conference center, and eventually replace our entire system (which has voicemail, etc) |
| 15:05.15 | BBHoss | how many phones |
| 15:05.24 | BBHoss | and lines going out |
| 15:05.31 | BBHoss | maximum |
| 15:05.41 | fiXXXerMet | 12-16 lines, we have 40-50 phones |
| 15:05.49 | BBHoss | hmm |
| 15:05.54 | fiXXXerMet | We'd need 12 for the conference center |
| 15:06.12 | [TK]D-Fender | fiXXXerMet: Looking to do some custom stuff? |
| 15:06.15 | *** join/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 15:06.23 | [TK]D-Fender | fiXXXerMet: Got Linux capable staff? |
| 15:06.45 | mrtelephone | what causes sip show inuse to show active channels when there is none active? |
| 15:06.51 | fiXXXerMet | I do linux administration but as far as programming, not really. |
| 15:07.07 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 15:07.28 | BBHoss | do you use ip phones now? or are you using an old nortel system or similar |
| 15:07.33 | littleball | <PROTECTED> |
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| 15:07.46 | fiXXXerMet | We'll probably replace all of our phones |
| 15:07.59 | BBHoss | do you have a PRI? |
| 15:08.00 | [TK]D-Fender | fiXXXerMet: then Straight * may be for you. |
| 15:08.10 | BBHoss | defiitely |
| 15:08.16 | BBHoss | definitely * |
| 15:08.20 | fiXXXerMet | PRI? |
| 15:08.24 | BBHoss | t1? |
| 15:08.24 | *** part/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 15:08.53 | fiXXXerMet | Yes |
| 15:08.58 | BBHoss | ok |
| 15:08.58 | fiXXXerMet | 1 for data and 1 for voice |
| 15:09.02 | BBHoss | hmm |
| 15:09.38 | BBHoss | you could attempt VoIP on the t1, if the data utilization is not high, but if you want to keep them separate, you will need a t1 card |
| 15:10.40 | BBHoss | consolidating the t1s would save you a lot of money |
| 15:10.48 | l2trace99 | has anyone set an * cluster ? |
| 15:10.57 | fiXXXerMet | We need the bandwidth |
| 15:11.04 | *** join/#asterisk shawdog22 (n=shaw@pacman.oaklandcorp.com) |
| 15:12.05 | BBHoss | well you will need a t1 card then to interface with your t1 for voice |
| 15:12.57 | BBHoss | a TE120P if you want a digium card |
| 15:13.08 | BBHoss | http://www.digium.com/en/products/digital/te120p.php |
| 15:13.59 | BBHoss | but when you get into voice cards, you get into zaptel, which is another animal by itself |
| 15:14.23 | mrtelephone | noone else has problems with sip inuse not decrementing? |
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| 15:14.27 | shawdog22 | I'm having an issue with getting a AT&T PRI working with a Sangoma card. The PRI has 8 channels, card seems to recognize the setup, but Asterisk fails to make calls on it. |
| 15:15.07 | funxion | if I have 2 phones registered as the same extension shouldnt they both ring? |
| 15:15.07 | bakermd | Does anyone use Asterisk as a least cost routing engine? |
| 15:16.05 | BBHoss | you can do LCR with asterisk, i'm not sure i would use it as an 'engine' per say |
| 15:16.26 | Strom_M | shawdog22: show me your zaptel.conf and zapata.conf |
| 15:16.29 | Strom_M | pastebin |
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| 15:17.22 | *** part/#asterisk littleball (n=littleba@bb220-255-16-159.singnet.com.sg) |
| 15:17.29 | bakermd | We are processing about 1M minutes per month and have about 6000 route entries in our current switch - can an asterisk box do that? |
| 15:17.52 | funxion | I would use SER for that before I'd use aterisk |
| 15:17.52 | Qwell | bakermd: sure, 1M minutes is about a T1 12/7 IIRC |
| 15:18.33 | Qwell | oh, no, it's 24/7 |
| 15:18.40 | shawdog22 | #Sangoma A102 port 1 [slot:8 bus:0 span: 1] |
| 15:18.40 | shawdog22 | span=1,0,0,esf,b8zs |
| 15:18.40 | shawdog22 | bchan=1-23 |
| 15:18.40 | shawdog22 | dchan=24 |
| 15:18.40 | shawdog22 | #Sangoma A102 port 2 [slot:8 bus:0 span: 2] |
| 15:18.40 | shawdog22 | span=2,0,0,esf,b8zs |
| 15:18.42 | shawdog22 | bchan=25-31 |
| 15:18.44 | shawdog22 | dchan=48 |
| 15:18.46 | shawdog22 | ;Inbound - Works |
| 15:18.48 | shawdog22 | ;Sangoma A102 port 1 [slot:8 bus:0 span: 1] |
| 15:18.49 | Strom_M | NO NO NO |
| 15:18.50 | shawdog22 | switchtype=national |
| 15:18.52 | shawdog22 | context=oakland-incoming |
| 15:18.53 | Strom_M | ah fuck |
| 15:18.53 | *** kick/#asterisk [shawdog22!i=north@pdpc/sponsor/digium/Qwell] by Qwell (don't paste) |
| 15:19.15 | Qwell | I love the use of multiple ways to comment there |
| 15:19.16 | *** join/#asterisk shawdog22 (n=shaw@pacman.oaklandcorp.com) |
| 15:19.21 | Qwell | ~pb |
| 15:19.21 | jbot | A Pastebin is a place to paste your stuff 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 |
| 15:19.23 | shawdog22 | Okay how do I do it then. |
| 15:19.46 | Qwell | bakermd: but yeah, 1M minutes/month isn't a big deal |
| 15:20.18 | *** join/#asterisk sriramnrn (n=chatzill@203.200.201.52) |
| 15:20.38 | shawdog22 | Just paste you the link to pastebin? |
| 15:20.41 | [TK]D-Fender | shawdog22: If your PRI has 8 channels, why are you configure ALL 23? |
| 15:20.42 | *** join/#asterisk BBHoss (n=hoss@146.229.181.183) |
| 15:20.44 | Qwell | paste the link here |
| 15:20.50 | shawdog22 | http://pastebin.com/d7615e193 |
| 15:21.09 | [TK]D-Fender | shawdog22: or is that your 2nd port? |
| 15:21.33 | [TK]D-Fender | shawdog22: pastebin your zapata.conf and the CLI output of a good incoming call and a bad outgoing one. |
| 15:21.53 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 15:21.58 | dijungal | hi |
| 15:22.04 | shawdog22 | I have 2 PRI. First one works, it used for inbound tollfree. Second is VOIP for outbound calls. The second has 8 channels. |
| 15:22.15 | dijungal | is there anyway i can set a limit on the amount of calls on a iax trunk? |
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| 15:22.59 | dijungal | is there anyway i can set a limit on the amount of calls in a queue? |
| 15:23.21 | mrtelephone | if call-limit worked properly that would be great |
| 15:23.48 | dijungal | i need to be able to set a limit on the amount of calls going into a queue from a trunk |
| 15:24.00 | dijungal | so i have to either do that on the queue or the trunk |
| 15:24.10 | dijungal | preferably the trunk |
| 15:25.01 | bakermd | Qwell: Isn't a big deal for * or for SER? (Thanks!) |
| 15:25.11 | shawdog22 | [TK]D-Fender: http://pastebin.com/d57e2428b |
| 15:25.23 | bakermd | The * box I am using is dual processor 3GHz with 4 GB ram |
| 15:25.36 | Qwell | bakermd: asterisk could handle it fine |
| 15:26.03 | [TK]D-Fender | shawdog22: and the rest? |
| 15:26.22 | bakermd | Qwell: What should I be searching for to get instructions to implement a robust LCR table |
| 15:26.29 | [TK]D-Fender | shawdog22: I see it : -- Executing Dial("SIP/123-08a969d8", "Zap/g1/15555555555||T") in new stack |
| 15:26.51 | [TK]D-Fender | shawdog22: You are dialing group "1" but you don't HAVE a group "1", only a "0" <------ |
| 15:27.09 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:27.19 | [TK]D-Fender | shawdog22: and I get the impression you wanted your 2nd PRI to be group "1" |
| 15:27.46 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 15:27.46 | [TK]D-Fender | shawdog22: And your 2nd PRI does not have the word "VoIP" assocuiated to to. PRI is PRI. |
| 15:27.51 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:28.08 | [TK]D-Fender | shawdog22: It would be advisable for you to consolidate your telco resources as well. A PRI for only INBOUND is ridiculous. |
| 15:29.01 | davevg-btwtech | dijungal, the combination of GROUP, GROUP_COUNT, and GotoIf in the dialplan may be able to get you what you want |
| 15:29.33 | shawdog22 | Sorry that didn't group 0 didn't get changed back to group 1 from my other testing. |
| 15:29.51 | dijungal | ok |
| 15:29.55 | [TK]D-Fender | shawdog22: Well there's the problem. Go fix it. |
| 15:30.27 | dijungal | davevg-btwtech: problem is... a client is sending calls to me to handle.. so i don't have access to their IVR |
| 15:30.41 | dijungal | so i need to be able to stop calls on my end |
| 15:30.59 | davevg-btwtech | dijungal, tag all calls coming in from them with the same group |
| 15:31.27 | shawdog22 | [TK]D-Fender: I change it and get... http://pastebin.com/d13ef4c3f |
| 15:32.02 | [TK]D-Fender | shawdog22: How is that different? |
| 15:33.01 | shawdog22 | <[TK]D-Fender>: I get a "Channel 0/1, span 2 got hangup request, cause 31" and the channel closes. |
| 15:33.04 | [TK]D-Fender | shawdog22: You STILL have group=0 in zapata, and dialing G1 <- |
| 15:33.19 | shawdog22 | I changed it on the setup, just not the pastebin |
| 15:33.22 | funxion | TK when registering two phones under the same sip user shouldnt they both ring? |
| 15:33.37 | [TK]D-Fender | shawdog22: show me something CURRENT |
| 15:33.49 | shawdog22 | http://pastebin.com/d6f4d0568 |
| 15:33.53 | [TK]D-Fender | funxion: no, that is something you CAN'T do |
| 15:34.00 | funxion | damn |
| 15:34.50 | flujan | guys, on asterisk I limit the number of logged peers? |
| 15:35.08 | rpm | Grrr. I hate dealing with call routing problems to NPA 876. They're routing is always broken. |
| 15:35.22 | flujan | for instance the peer SIP/40000 can log just one time... If someone grab another phone and try to use the 40000 I deny the logging. |
| 15:37.00 | *** join/#asterisk sasch (n=info@host117-234-static.4-79-b.business.telecomitalia.it) |
| 15:37.01 | bakermd | I have been corrected - I am not talking about LCR, I am talking about number prefix based routing with multiple route choices per entry |
| 15:37.36 | shawdog22 | TK: http://pastebin.com/d6f4d0568 |
| 15:37.59 | [TK]D-Fender | shawdog22: Ok, well now we're getting somewhere. |
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| 15:38.47 | [TK]D-Fender | shawdog22: You should probably have your "pridialplan" set in zapata, as well as a few other odd things. Go wiki up the config and play around. |
| 15:38.49 | [TK]D-Fender | \~wikis |
| 15:38.52 | [TK]D-Fender | ~wikis |
| 15:38.52 | jbot | wikis is probably http://www.voip-info.org |
| 15:40.08 | dijungal | ok.. is there anyway to limit calls going into a queue? |
| 15:40.59 | [TK]D-Fender | dijungal: The GROUP way taht was suggested is the proper way |
| 15:41.12 | dijungal | k |
| 15:41.24 | dijungal | thnk u |
| 15:41.45 | *** join/#asterisk r3zon8 (n=hg@64.80.101.162) |
| 15:42.04 | dijungal | what's the difference between groups and queues? |
| 15:42.37 | *** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210) |
| 15:43.02 | [TK]D-Fender | dijungal: They have nothing to do with each other |
| 15:43.28 | dijungal | hmmm.. i'm trying to determine if i should put my agents in a group instead of a queue |
| 15:43.37 | [TK]D-Fender | dijungal: You want to limit calls in your dialplan from reaching a certain point (the queue). Do a group set & check to DECIDE whether to allw them in. |
| 15:43.55 | dijungal | k |
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| 15:47.10 | puzzled | hi |
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| 16:01.08 | tzanger | { 0xdd84, 0xae84, 0xae84, 0xdd84, 0x387c, 0x517c, 0x517c, 0x227c } |
| 16:01.22 | [TK]D-Fender | tzanger: ou812? |
| 16:01.23 | mrtelephone | does anyone here use rtp timeout? |
| 16:01.24 | tzanger | does that look about right for a slinear milliwatt signal (0dbm) ? |
| 16:01.32 | mrtelephone | turns out the modems are not always sending a BYE message |
| 16:01.42 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 16:01.49 | agx | Can i disable echo cancellation when dialing out on zap via Dial() as i already can do with mISDN ? |
| 16:01.52 | *** join/#asterisk techie (n=techie@adsl-76-214-18-225.dsl.lsan03.sbcglobal.net) |
| 16:02.57 | tzafrir | analog zap? |
| 16:03.08 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:03.18 | agx | tzafrir, y |
| 16:03.50 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 16:03.52 | tzafrir | echo cancelling should only be enabled by chan_zap once you're done dialing |
| 16:04.11 | tzanger | coppice: do you have the "true" 1003Hz or whatever slinear or ulaw pattern for a test tone? IIRC the digium milliwatt() app is a little fudged to make it fit nicely |
| 16:04.12 | agx | Dial(Zap/g0m/123456) will work? |
| 16:04.32 | tzafrir | what is the 'm'? |
| 16:04.50 | Strom_M | tzafrir: i think I have it in one of my ITU documents |
| 16:04.59 | agx | tzafrir, dunno, should i backport ZapdisableEc from bristuff as well? |
| 16:05.01 | Strom_M | tzafrir: lemme go see if I can hunt it down for you |
| 16:05.06 | tzanger | Strom_M: you mean me? |
| 16:05.12 | Strom_M | er |
| 16:05.13 | Strom_M | yes |
| 16:05.17 | Strom_M | it's early here still :) |
| 16:05.25 | BCOBB | Can any one tell me where I can find out how to call functions from a res module from the dialplan? or can any one tell me how the realtime function comunicates with res_config_mysql. Or could someone tell me that I am completly wrong on how this works and could point me in the write direction. |
| 16:05.35 | Strom_M | tza___r |
| 16:05.54 | coppice | tzanger: the usual test tones are 1000Hz (a repeating pattern of 8 samples) and 1004Hz (which will go through a much wider range of sample values) |
| 16:06.47 | *** part/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 16:07.10 | agx | BCOBB, you need to register an application or a function to use it from dialplan |
| 16:08.09 | BCOBB | ok |
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| 16:08.31 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:08.35 | tzafrir | agx, you can check with 'zap show channel NNN' |
| 16:08.47 | BCOBB | ok I know how to get a function and an app to work on the dialplan |
| 16:08.56 | tzafrir | or enable debug logging |
| 16:09.03 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 16:09.33 | agx | tzafrir, its not what i need :) i want to send faxes out via analog i want to disable EC on that channel before TxFax |
| 16:09.57 | tzafrir | agx, faxes should be disabled by the fax tone |
| 16:10.01 | agx | tzafrir, on mISDN i can use extra options after Dial on outgoing calls or misdn_set_options for incoming calls |
| 16:10.39 | tzafrir | If the fax tone does not disable the EC it is a bug that should get fixed |
| 16:10.44 | agx | tzafrir, enforcing it will not hurt :-P |
| 16:10.47 | badcfe | lame question -- is it necessary to end the h extention with a priority doing the Hangup application? |
| 16:11.08 | agx | badcfe, uhm, i suppose its already hanged up |
| 16:11.25 | tzafrir | agx, have you actually encountered a problem here? |
| 16:11.28 | tzanger | coppice: I have the 1000Hz one, do you have the 1004Hz one? |
| 16:11.36 | badcfe | the thing i do in the h is Macro wich does a UserEvent. seems that sometimes i dont get the event in the mapi .... |
| 16:12.16 | badcfe | may an explisit Hangup be good to flush stuff like pending events and maybe cdr related stuff? |
| 16:12.19 | agx | tzafrir, no on small faxes but with 30/50 pages faxes it can lead to |
| 16:12.20 | BCOBB | agx, I need to write a module that will maintain a sql connection and then from the dial plan pass queries to it. So I thought that I would need to write a res_module to maintain the sql connection. Would a res module beable to do something tike this? |
| 16:12.29 | coppice | tzanger: huh? its just a tone. only the 1000Hz option is expected to follow any rigid pattern |
| 16:12.41 | agx | BCOBB, you can use AGI and make a perl/shell script to the job |
| 16:13.23 | tzanger | coppice: oh, I thought the 1004Hz one was designed to not fit inside a 2^n buffer nicely |
| 16:14.25 | Strom_M | tzanger: nope...here's what telcordia has to say about 1000Hz DMW: |
| 16:14.28 | Strom_M | Digital Milliwatt: An important concept relating to signals in digital facilities is Digital Milliwatt (DMW). A DMW is a digital representation of a 0-dBm, 1000-Hz (analog) sine wave. A representation consisting of eight 8-bit words has been adopted by the ITU-T (formerly the CCITT) for a DMW that is encoded according to the ? = 255 encoding law. |
| 16:14.28 | Strom_M | The DMW is not recommended for transmission over DS1 facilities, because the bit pattern duplicates the framing sequence of D-type channel banks. Under certain conditions, the bank can frame on the DMW sequence, causing frame misalignment without any accompanying alarm condition. However, the DMW can be used internally in digital switching systems, because these systems do not operate at the DS1 rate. |
| 16:14.29 | Strom_M | In addition, it is not advisable to inject a 1000 Hz signal into a DS1 system because of the harmonic relation to the 8000 Hz sampling rate. Interference signals can result in erroneous loss measurements when these frequencies are mixed. |
| 16:14.48 | BCOBB | agx, wouldn't that create a new connection for every channel? |
| 16:15.10 | BCOBB | or every call |
| 16:15.20 | Strom_M | tzanger: i'm still looking for the actual recommended pattern :) |
| 16:15.50 | Katty | herro geekables. |
| 16:16.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:17.11 | BCOBB | agx. because I would like one connection when asterisk startes that every channel could use and not have to create thier own. Unless something already does this. in which case could you tell me what it is. |
| 16:17.46 | mrtelephone | is this a valid regex ? ("^807[229|822|826|825|868]...." |
| 16:17.48 | tzanger | weird, if I modify coppice's sliptest app to spit out the slinear 8-word milliwatt pattern, it sounds "higher" than milliwatt from astersik |
| 16:18.04 | tzanger | it should be playing out at the same rate |
| 16:18.29 | mrtelephone | I think I should take out the [ ] |
| 16:20.14 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 16:21.11 | agx | BCOBB, AGI create a connection for every channel ; FAGI keep the conneection open (like an application server) |
| 16:21.53 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:22.05 | hi365 | antone using call files to scedual a call in the future? |
| 16:22.26 | *** join/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 16:24.07 | BCOBB | agx, thanks I will look into it |
| 16:24.10 | *** join/#asterisk saftsack (n=saftsack@pD9E05C60.dip.t-dialin.net) |
| 16:24.18 | agx | hi365, change the callfile timestamp with the desidered scheduled date, or create a scheduler that create the file when needed |
| 16:24.38 | agx | BCOBB, if you go for FAGI post me an example back plz :-P |
| 16:24.44 | Alan_Hicks | Hey, can anyone provide me with a link to an example of congestion detection when using Dial? |
| 16:24.49 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:24.58 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:25.00 | hi365 | agx: im trying to use the timestamp method, but after i mv the call file the timestamp gets reset! |
| 16:25.05 | Alan_Hicks | Say all my telco lines are in use and I try to place an outbound call. |
| 16:25.24 | BCOBB | agx, ok I will |
| 16:25.37 | Alan_Hicks | I'd appreciate an example before I go delving into this head-first. |
| 16:26.34 | hi365 | agx: does it work for you? |
| 16:27.03 | agx | hi365, uhm yeah but i'm on some old 1.2 version |
| 16:28.36 | *** join/#asterisk linxroute (i=linxrout@117.0.17.61) |
| 16:29.25 | linxroute | . |
| 16:30.20 | *** part/#asterisk techie (n=techie@adsl-76-214-18-225.dsl.lsan03.sbcglobal.net) |
| 16:30.21 | Strom_M | linxroute: do I look like SMTP to you or something? |
| 16:30.34 | linxroute | ? |
| 16:30.47 | Strom_M | sigh |
| 16:30.54 | Strom_M | the joke would take too long to explain |
| 16:31.02 | linxroute | oh |
| 16:31.25 | linxroute | just dont know it |
| 16:34.20 | agx | hi365, didn't worked for you??? http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out#HowtoscheduleaCallintheFuture |
| 16:34.20 | flujan | guys do you already use a xcap or webdav server to store your sofphones/phones contacts? |
| 16:35.14 | *** join/#asterisk rpyne (n=rpyne@69.77.169.14) |
| 16:35.26 | rpyne | hello everyone |
| 16:35.38 | hi365 | agx: no. the timestamp get changed after the mv (im notusing /var/spool/asterisk/tmp - could that be why?) |
| 16:37.50 | agx | hi365, uhm, are you moving it from diffrente filesystems? |
| 16:38.02 | rpyne | I am currently working on asterisk realtime setup. I have the sip, voicemail and extensions working okay. But I'm not seeing any support for meetme.conf setup. |
| 16:38.37 | rpyne | Anyone has any ideas on the meetme set up, or if its available at all. |
| 16:39.02 | hi365 | agx: no form /home to /v/s/a/outgoing |
| 16:39.11 | hi365 | on the same physical drive |
| 16:39.14 | agx | hi365, strange it works for me |
| 16:39.24 | agx | hi365, same partition? |
| 16:39.27 | hi365 | from where to where? |
| 16:39.28 | hi365 | yup |
| 16:39.40 | agx | hi365, pastebin the script plz :) |
| 16:39.50 | rpm | what is the point of DeadOGI if channel variables are not persistant after a call has hung-up? |
| 16:40.15 | agx | rpm, ROLFMAO |
| 16:40.41 | hi365 | agx: it proprietary ;-) |
| 16:40.44 | hi365 | http://pastebin.ca/755292 |
| 16:41.19 | agx | hi365 try inverting chown and touch :) |
| 16:43.27 | hi365 | agx: didnt help. the call gets played imediatly |
| 16:43.50 | mrtelephone | did I miss something funny? |
| 16:44.09 | hi365 | ags: the touch seems fine. its the mv thats messing it up. form what directory did you move the file? |
| 16:45.52 | agx | hi365, /tmp |
| 16:46.35 | hi365 | agx: same problem :( |
| 16:47.18 | agx | hi365, verify the date of the file before it move (man ls, i don't remember how to print the 3 timestamps now *g*) |
| 16:47.49 | hi365 | agx: did you try my script? does it work for you? |
| 16:48.38 | agx | hi365, its same as mine, try "ls --fule-time callfile.berfore.its.moved" |
| 16:48.45 | agx | ops; --full-time |
| 16:48.48 | *** join/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 16:50.29 | hi365 | agx: [root@asterisk1 tmp]# ls --full-time 16528.call |
| 16:50.29 | hi365 | -rw-r--r-- 1 asterisk asterisk 270 2007-10-30 04:26:00.000000000 +0200 16528.call |
| 16:50.55 | agx | hi365, so? is it correct? in future? |
| 16:51.10 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:51.17 | hi365 | yes. but after the move it gets reset! |
| 16:51.30 | dandre | hello, |
| 16:51.50 | agx | hi365, uhm, turn off *; try to move and see if it reset; are you using a GNU mv or its busybox? |
| 16:52.03 | hi365 | centos (GNU?) |
| 16:52.54 | hi365 | agx: i think i see the error. the time being set is am. it should be pm |
| 16:53.23 | hi365 | hmm, now for a command line to convert am -> pm? |
| 16:53.24 | mrtelephone | what is the proper method for having a failover asterisk box.. duplicate sip accounts on each.. but how do you handle incoming calls... $EXTEN/sipbox1 and if it errors try $EXTEN/sipbox2? |
| 16:53.40 | dandre | I have 2 asterisk box connected with a sip account. I cant dial extensions from each server to the other. The problem is that I don't get the CID from one of them. What sould I do? |
| 16:54.54 | agx | hi365, the problem is in your $sunset variable then |
| 16:55.01 | hi365 | yup |
| 16:55.13 | hi365 | i guess ill add the seconds to epoch to convert it |
| 16:55.41 | agx | hi365 check do a better parsing of the HTTP output :) |
| 16:55.56 | hi365 | how so? (is the pm there?) |
| 16:56.19 | agx | hi365, postcard and beers welcome now :) send them to #agx-beerware-asterisk-support or #agx-postcardware-asterisk-support |
| 16:56.36 | hi365 | agx: ok. it works. thanks |
| 16:56.55 | hi365 | s/to/2/ |
| 16:58.22 | tzanger | uh |
| 16:58.41 | tzanger | if I have a sip ulaw rtp stream, should I not see the ulaw data clearly in a packet dump? |
| 16:59.02 | tzanger | I can see it in the tdmoe stream, but the rtp stream does not show it |
| 16:59.57 | moa_ | simple question: Is there a way to setup first in first out for roughly 12 queues globally? |
| 17:01.33 | Alan_Hicks | Hey guys. My telco has this incredibly stupid system that offers to continue trying a phone call for 30 minutes if the number you dial is busy (for a huge fee of course). Since their automated operator picks up the line, am I to assume that there's no way to get asterisk to handle these "busy" signals? |
| 17:02.04 | Alan_Hicks | Telco is Bellsouth/AT&T. |
| 17:03.30 | *** join/#asterisk techie (n=techie@adsl-76-214-18-225.dsl.lsan03.sbcglobal.net) |
| 17:04.37 | Strom_M | Alan_Hicks: the solution is simple |
| 17:04.51 | Strom_M | don't use call progress detection on analog circuits |
| 17:04.52 | Alan_Hicks | Enlighten me, please. |
| 17:05.07 | Alan_Hicks | Call progress detection? I've never heard of this before. |
| 17:05.24 | Strom_M | Alan_Hicks: what kind of entrance facilities does your asterisk system have> |
| 17:05.25 | Strom_M | ? |
| 17:05.39 | Alan_Hicks | Analogue POTS lines. |
| 17:05.45 | Strom_M | exactly |
| 17:05.55 | dandre | I have 2 asterisk box connected with a sip account. I cant dial extensions from each server to the other but I get 'asterisk' as CID instead of the extension number of the phone. I don't know where to look for this |
| 17:06.05 | Strom_M | so don't do call progress detection - leave that to your calling party |
| 17:06.06 | Alan_Hicks | I'm afraid I don't understand what you're getting at. |
| 17:06.19 | Strom_M | Alan_Hicks: perhaps you should read this document |
| 17:06.21 | Strom_M | ~101 |
| 17:06.22 | jbot | extra, extra, read all about it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 17:06.55 | Alan_Hicks | Strom_M: I'll give it a look, but I think you're misunderstanding. I _am_ the calling party. |
| 17:07.03 | Strom_M | Alan_Hicks: I understand completely |
| 17:07.07 | Alan_Hicks | ok |
| 17:07.16 | Strom_M | let the human do the call progress detection |
| 17:07.26 | Strom_M | don't tell asterisk to do it |
| 17:08.20 | Alan_Hicks | That's simple enough, but what if I really want asterisk to do it? Just for argument's sake. |
| 17:08.29 | Strom_M | if you need full pass-through of call progress and answer supervision to the station equipment, then don't use analog circuits |
| 17:08.32 | Strom_M | use ISDN or VoIP |
| 17:08.46 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.227) |
| 17:08.48 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 17:09.16 | tzanger | is there an application which will play dialtone while waiting for an extension like waitexten |
| 17:09.20 | tzanger | or is it just a passwordless disa? |
| 17:09.43 | Strom_M | tzanger: I think you can do playtones followed by waitexten... |
| 17:10.51 | *** join/#asterisk bkruse_home (n=kruz@76.73.154.117) |
| 17:10.52 | moa_ | Strom_M: since your answering questions, got a sec for mine? |
| 17:11.16 | Strom_M | moa_: what's your question? |
| 17:11.34 | moa_ | I need a way to do first in first out globally, with roughly 12 queues. |
| 17:11.56 | Strom_M | all queues in asterisk are FIFO unless you do fancy things like weighted priority queueing |
| 17:12.22 | agx | Strom_M, look cool :) |
| 17:12.30 | Strom_M | agx: ? |
| 17:12.34 | agx | the PDF |
| 17:12.39 | Strom_M | agx: ah |
| 17:12.40 | Strom_M | thanks |
| 17:12.43 | moa_ | Yes, but different calls come into different queues, are not fifo |
| 17:12.59 | Strom_M | moa_: ok, so explain exactly what you want to happen |
| 17:13.10 | Strom_M | you want calls to cascade through 12 different queues? |
| 17:13.32 | moa_ | Ok, call a into queueA, call b into queueB, call c into queueC, etc |
| 17:14.11 | moa_ | I need fifo globally, they are being answered not fifo, I end up having calls stuck in queue for 5-10 minutes |
| 17:14.12 | dijungal | is there a way to accept a call from a iax trunk then send a channelunavailable signal up the line so the call can be processed on the other end of the trunk ? |
| 17:14.30 | moa_ | If that makes sense... |
| 17:14.34 | Strom_M | moa_: so if call B comes in after call A, you want call A always answered first? |
| 17:14.40 | moa_ | Yes |
| 17:14.49 | Strom_M | why not just put them all in the same queue? |
| 17:14.59 | moa_ | Can't do that for stats reason |
| 17:15.13 | Strom_M | what "stats reason" is this? |
| 17:15.21 | moa_ | We've got 10 different telcos that all need their own stats. |
| 17:15.57 | Strom_M | ok.....? |
| 17:16.00 | moa_ | I can't generate individual stats if they are all in their own queue. |
| 17:16.16 | Strom_M | but stats for what? why do you have 10 telcos? what are you doing? |
| 17:16.21 | moa_ | Tech support |
| 17:16.31 | moa_ | DSL, email, dialup |
| 17:16.34 | Strom_M | your answers are irritatingly short. |
| 17:17.41 | Strom_M | if you give me a complete picture of what you're trying to accomplish, then I can help you out. |
| 17:19.02 | moa_ | I apologize, let me see if I can clarify, We're a ISP that does tech support for several different telcos. Each one, doesn't need to know anyone else's stats, call volume, etc. I have 12 queues, each one for a different telco. I understand that individual queues are FIFO, but I need FIFO for the asterisk system per say. |
| 17:19.31 | Strom_M | ugh...it's "per se" :) |
| 17:19.36 | moa_ | lol |
| 17:19.40 | moa_ | yea yeah |
| 17:19.45 | Strom_M | "per say" makes zero sense |
| 17:19.55 | moa_ | s/per say/per se |
| 17:19.57 | moa_ | that help? |
| 17:20.34 | Strom_M | are the answering endpoints all members of all queues? |
| 17:20.45 | moa_ | Yes they are |
| 17:21.53 | Strom_M | what's going wrong with the current setup? does it appear that one queue has priority over another? |
| 17:22.43 | moa_ | Yes, if I put a call in each queue, in order queue a thorugh d. They get answered randomly. |
| 17:23.14 | Strom_M | what distribution strategy are you using? |
| 17:23.34 | moa_ | least recent, if that's what your asking |
| 17:23.51 | Strom_M | yes |
| 17:24.22 | Strom_M | so ok...let me think about this for a sec |
| 17:25.38 | Strom_M | AFAIK, there is no way, at present, to unify a number of disparate queues into one "group" |
| 17:26.19 | moa_ | I was afraid of that |
| 17:26.20 | Strom_M | you might be able to do it if you can have calls be in two queues at once, but I don't know if that's possibe |
| 17:26.37 | Strom_M | possible |
| 17:28.28 | agx | there is no way i can change CDR(src) and CDR(dst) with my own values? :-( |
| 17:28.29 | Strom_M | yeah, it appears that, at least in asterisk 1.4, that's not possible |
| 17:28.51 | Woifi1988 | hi |
| 17:28.52 | Strom_M | moa_: you may wish to talk to one of the developers |
| 17:29.26 | Woifi1988 | can someone help me plz: can't asterisk use gsm out of the box? |
| 17:29.27 | moa_ | Ok, I might just do that |
| 17:29.32 | Corydon76-dig | agx: yes, just change CALLERID(ani) and Goto(extenyouwant,1) |
| 17:29.45 | Woifi1988 | i get this error when i try to phone over gsm |
| 17:29.48 | Woifi1988 | No compatible codecs, not accepting this offer! |
| 17:30.01 | Strom_M | Woifi1988: the other endpoint has to support GSM as well |
| 17:30.09 | moa_ | Strom_M: I appreciate your time and effort. I may play around with nesting queues and see if I can't get it to work that way. |
| 17:30.20 | Corydon76-dig | agx: those CDR variables DIRECTLY reflect those values |
| 17:30.40 | Woifi1988 | Storm_M: Bothe devices are x-lite phones and should support gsm |
| 17:30.52 | Strom_M | Woifi1988: my handle is not storm :) |
| 17:31.07 | Woifi1988 | sorry ;-> |
| 17:31.11 | Strom_M | Woifi1988: you've got to turn on gsm support on asterisk for that specific sip peer |
| 17:31.25 | Corydon76-dig | You mean you don't look like Halle Berry in a skintight suit? |
| 17:31.27 | Strom_M | as well as tell your softphone to use gsm... |
| 17:31.40 | Strom_M | Corydon76-dig: hang on, let me grow some tits |
| 17:32.05 | Corydon76-dig | Strom_M: and get a deep tan? |
| 17:32.07 | Woifi1988 | Strom_M: i tried to disallow all codecs in [general] an turned in [context1] alaw on and in[context2] ulaw |
| 17:32.23 | Strom_M | Woifi1988: neither alaw nor ulaw are gsm |
| 17:32.29 | Woifi1988 | yes |
| 17:32.30 | Woifi1988 | but |
| 17:32.39 | codefreeze | agx: I wonder, if you could mod the callerID info; then do a cdr_reset... hmmm... gotta look in the source... isn't there a cdr_reset builtin app? |
| 17:32.48 | Woifi1988 | when i called one another the call hung up! |
| 17:33.03 | Strom_M | Woifi1988: and what is your softphone set to support? |
| 17:33.30 | Woifi1988 | i didn't found any configuration in x-lite for codecs |
| 17:33.47 | Woifi1988 | doesn't x-lite decide this automatically? |
| 17:33.52 | agx | codefreeze,Corydon76-dig, the only chance is to patch cdr_XYZ.c to allow storing other extra CDR vars (like cdr_custom seems to do)... |
| 17:33.57 | agx | gtg, bye :-P |
| 17:34.24 | Strom_M | Woifi1988: beats me; i don't use x-lite |
| 17:35.30 | Woifi1988 | Strom_M: Okay, but the error message appears normally when the phones don't support the codec? |
| 17:35.40 | Strom_M | Woifi1988: yes |
| 17:36.01 | Woifi1988 | Strom_M: do I need ztdummy for codec conversation? |
| 17:36.01 | Corydon76-dig | Great, ask a question, then immediately sign off |
| 17:36.26 | Strom_M | Woifi1988: no |
| 17:36.51 | Woifi1988 | do you know a softphe which supports different codecs? |
| 17:37.04 | dijungal | A call comes in from a iax trunk, i set the call to a group before i send it to a queue, I do a group_count to determine > 2, how do i send the call back or refuse the call on the trunk? |
| 17:37.07 | [TK]D-Fender | Woifi1988: X-Lite supports all the common free ones |
| 17:37.14 | dijungal | i do not want to hang up the call |
| 17:37.32 | Woifi1988 | where can i change the codecs? |
| 17:37.33 | [TK]D-Fender | Woifi1988: Show us your attempt failing at verbose 10, SIP DEBUG enabled |
| 17:37.34 | [TK]D-Fender | ~pb |
| 17:37.35 | jbot | A Pastebin is a place to paste your stuff 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 |
| 17:37.36 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 17:37.58 | [TK]D-Fender | Woifi1988: Their menu isn't Raw-Cat Science. Seriously, jsut look. |
| 17:37.59 | Woifi1988 | [T |
| 17:38.44 | Woifi1988 | i can't show you this outputs! I have only ssh access and no sip from outside |
| 17:38.52 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 17:39.08 | [TK]D-Fender | Woifi1988: All I want to SEE can be acquired through SSH. |
| 17:39.31 | [TK]D-Fender | Woifi1988: Now go place another call and PASTEBIN IT along with the matching configs. |
| 17:39.33 | Woifi1988 | how can i enable sip when nobody phones? |
| 17:39.45 | Woifi1988 | i cant play a call now |
| 17:39.52 | Woifi1988 | i have only access to port 22 |
| 17:40.06 | [TK]D-Fender | Woifi1988: Well I guess its going to be pretty hard to tell you where you went wrong, now isn't it? |
| 17:40.14 | dijungal | how do i reject a call on a trunk without hanging it up? |
| 17:40.16 | [TK]D-Fender | Woifi1988: pastebin your configs at least |
| 17:40.44 | Woifi1988 | yes i know but i thougth that you may tell me that i need ztdummy because i know that i have done something wrong when installing it |
| 17:40.52 | Woifi1988 | okay i'll pots my configs |
| 17:40.59 | [TK]D-Fender | Woifi1988: that has NOTHING to do with codec translation. |
| 17:41.48 | Woifi1988 | http://pastebin.ca/755377 |
| 17:41.52 | Woifi1988 | is the sip.conf |
| 17:42.24 | *** join/#asterisk Shmattie (n=Shmattie@cpe-75-179-191-147.woh.res.rr.com) |
| 17:42.44 | Woifi1988 | extensions.conf: http://pastebin.ca/755378 |
| 17:43.44 | Shmattie | Anyone know where I can get a list of all the IMAPFLAGS options? |
| 17:43.59 | [TK]D-Fender | Woifi1988: Most of your entries you don't even SPECIFY codecs. Its a mess. Put disallow=all, and the the SPECIFIC allow(s) for EACH. |
| 17:44.59 | Woifi1988 | yes because i have just uncomment these... i just use [10] an [20] |
| 17:45.33 | [TK]D-Fender | Woifi1988: Then fix at least THEM accordingly. |
| 17:46.11 | fetcher | Can a Handytone HT-496 do G.729 on both FXS ports at once? |
| 17:46.28 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 17:46.37 | Woifi1988 | okay! i see that ulaw and alaw is enabled in x-lite |
| 17:46.42 | fetcher | I know the PAP2 can only handle compression for a single call... looking for an alternative |
| 17:46.49 | tzafrir | current news at voip-info: What's up with [ http://forums.digium.com/|Digium] & [ http://www.asterisk.org/|Asterisk] Websites? Hacked? I guess their all at [ http://www.von.com/2007/boston/web/|VON]. |
| 17:47.44 | fetcher | J4k3: I'd take iLBC+IAX2 :) |
| 17:48.12 | coppice | oooh, iLBC :-\ |
| 17:48.15 | *** join/#asterisk michael-i (n=michael-@Wbdf3.w.pppool.de) |
| 17:48.32 | J4k3 | "is that voice or an electric carving knife?" |
| 17:48.56 | [TK]D-Fender | LPC10 : Domo Arigato! |
| 17:49.12 | Woifi1988 | [TK]D-Fender : should it work now? alaw and ulaw are enabled in x-lite and this is my sip.conf http://pastebin.ca/755387 |
| 17:49.36 | fetcher | J4k3: you notice a lot of artifacts? It sounds decent to me, even with a different lossy codec at the far end (G.729 or cellular) |
| 17:50.09 | J4k3 | fetcher: it was pretty rough, but it may have been the softphone itself. |
| 17:50.15 | rantsh | am I missing something for my queue to mixmonitor calls successfully? I'm using asterisk 1.2.24 |
| 17:50.23 | J4k3 | I was using a softphone + bt earpiece on a laptop with a USB<->ethernet IRQ sharing issue |
| 17:50.23 | coppice | the problem with iLBC is not that it is bad. its that its design is dumb |
| 17:50.27 | J4k3 | so it chunks data no matter what. |
| 17:50.28 | [TK]D-Fender | Woifi1988: nicer. So if those a2 accounts are the ones being called you should be fine. |
| 17:50.29 | fetcher | higher CPU use seems to be the main limitation... |
| 17:50.37 | rantsh | this is my queues context and sox/soxmix -h output http://pastebin.com/d37004c0a |
| 17:50.51 | J4k3 | xv6700 should be here thursday or friday, woo! |
| 17:50.59 | fetcher | J4k3: bluetooth headsets do their own compression, I think. Probably ADPCM (like DECT)? |
| 17:51.06 | Woifi1988 | [TK]D-Fender: okay i'll tell you tommorow if it had worked |
| 17:51.10 | *** join/#asterisk jimmysolis (n=jimmy@190.41.82.1) |
| 17:51.12 | [TK]D-Fender | J4k3: Don't forget to get the baked WM6 ROM for it :) |
| 17:51.23 | Woifi1988 | [TK]D-Fender: is gsm a free codec? |
| 17:51.23 | [TK]D-Fender | J4k3: BIG improvement |
| 17:51.30 | [TK]D-Fender | Woifi1988: GSM610 is |
| 17:51.52 | coppice | GSM 06.10 to be more accurate |
| 17:52.11 | Woifi1988 | is this codec already included? |
| 17:52.14 | Woifi1988 | in asterisk= |
| 17:52.23 | tzafrir | yes |
| 17:52.30 | [TK]D-Fender | J4k3: I have inside info that Ma Bell here is announcing the HTC Touch (regular & Duo) next month, and possiby others. |
| 17:52.43 | [TK]D-Fender | Woifi1988: This is the GSm that * uses |
| 17:53.08 | [TK]D-Fender | J4k3: I really like the HTC Touch and may be able to live without the WiFi.... |
| 17:53.17 | coppice | the existing HTC Touch or the new one? |
| 17:53.21 | jimmysolis | hey guys, what machine is recommended to 60 calls concurrents with 1 card TE205 and codec g711?? |
| 17:53.24 | tzafrir | Woifi1988, so are ilbc and speex, if you want to stress your CPU |
| 17:53.38 | [TK]D-Fender | coppice: The is a pure non-keypad on, and the new slider (the Duo) |
| 17:53.45 | [TK]D-Fender | coppice: We are supposed to be getting both. |
| 17:53.51 | [TK]D-Fender | one* |
| 17:54.13 | Woifi1988 | and whats when i want to use a gsm gateway? |
| 17:54.15 | [TK]D-Fender | jimmysolis: I'd say a decent P4 would be more than fine. |
| 17:54.20 | Woifi1988 | can asterisk handle this? |
| 17:54.24 | [TK]D-Fender | Woifi1988: Depends on that gateway |
| 17:54.25 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:54.32 | Woifi1988 | it's a normal mobile phone |
| 17:54.34 | coppice | yeah, but the touch we've had for 6 months or the one that;s is supposed to be out soon with UMTS? |
| 17:54.39 | Woifi1988 | with a gsm modem |
| 17:54.41 | [TK]D-Fender | Woifi1988: This term "gateway" implies NOTHING about how it connects to *. |
| 17:54.45 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:55.06 | [TK]D-Fender | coppice: Dunno the fine points, esp on Data. I have no intention of using it with Telco data services. |
| 17:55.32 | Woifi1988 | [TK]D-Fender: with a gsm modem?! |
| 17:55.36 | [TK]D-Fender | coppice: more like for quicky camera (very nice!), music, PDA use |
| 17:55.55 | [TK]D-Fender | Woifi1988: DEPENDS. Holy crap get the manual of whatever device you're looking at and READ IT! |
| 17:55.58 | coppice | the one that has been aound since the spring is nice. same old issue with HTC phoines, though. people love the hardware and hate the software |
| 17:56.43 | tzanger | which phone's this? |
| 17:56.51 | Woifi1988 | okay i am new to asterisk sry... |
| 17:56.53 | [TK]D-Fender | coppice: the Touch has that nifty secodary interface over WM6 which is cute. I'd really like to see a Croix or Prada around here though... but my provider is CDMA and all the cool stuff is GSM only. |
| 17:57.09 | [TK]D-Fender | coppice: Much like the Neo1973 |
| 17:57.12 | coppice | that's like polishing a turd |
| 17:57.18 | jimmysolis | in the future maybe increase the channels with a p4 is better? |
| 17:57.25 | [TK]D-Fender | coppice: Mighty fine polish :p |
| 17:57.32 | coppice | that interface is pretty neat, but underneath its still MS |
| 17:57.48 | [TK]D-Fender | coppice: Yeah, imperfect solutions for an imperfect world... |
| 17:57.52 | tzanger | I'm hoping to grab one of the new openmokos |
| 17:57.56 | tzanger | although I hate how they look |
| 17:58.01 | tzanger | at least the internals I can play with |
| 17:58.03 | coppice | why HTC don't ditch it is a puzzle to most people. there must be some leverage MS has over HTC |
| 17:58.06 | [TK]D-Fender | tzanger: I like it personally... |
| 17:58.09 | russellb | tzanger: wait for the next hw rev ... |
| 17:58.28 | [TK]D-Fender | tzanger: beautiful screen though I COULD do with squared edges. |
| 17:58.36 | coppice | the openmoko is just another WinCE box that bombed |
| 17:58.43 | [TK]D-Fender | coppice: Leverage? Duh :) |
| 17:59.01 | [TK]D-Fender | coppice: this is M$! |
| 17:59.11 | tzanger | yes it's teh edges |
| 17:59.22 | tzanger | russellb: which hardware rev |
| 17:59.33 | russellb | tzanger: of the openmoko phone |
| 17:59.46 | tzanger | russellb: yes, which, I'm waiting for the commercial release not just the dev release |
| 17:59.50 | russellb | there will be a next rev coming out in the next few months, so i'm saying you should wait for that :) |
| 17:59.51 | russellb | a ... |
| 17:59.56 | russellb | s/a/ah/ |
| 18:00.03 | coppice | but MS is the biggest thing holding HTC back. MS must have an investment in them, or some erally screwed up long term contract. without HTC WinCE would be dead by now |
| 18:01.04 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:01.24 | [TK]D-Fender | coppice: Plenty of other players stuck with them like HP. I figured they'd be #1 on that hit-list |
| 18:02.07 | coppice | HP is nowhere. only HTC has ever got a WinCE box to really sell |
| 18:02.15 | *** join/#asterisk shtoom (n=godson@59.93.119.70) |
| 18:02.35 | [TK]D-Fender | coppice: Funny I thought the iPaq practically made the W$ PDA market. |
| 18:03.08 | coppice | iPaq never really hit any serious volume. then HP took over and killed it completely |
| 18:03.11 | [TK]D-Fender | coppice: Sure in the cell market HTC owns, but thats another matter. |
| 18:03.27 | tzanger | dammit I need some repeating data source that does not fit nicely in 8 bytes |
| 18:03.33 | [TK]D-Fender | coppice: Hrm, well I'll take that little bit of info home with me then :) |
| 18:03.54 | coppice | tzanger: try a 9 byte pattern, then |
| 18:04.22 | coppice | the current ipaqs are really HP jounadas, and sell about as well as they did |
| 18:04.23 | tzanger | coppice: yeah I need something that is easy to hear for corruption in |
| 18:04.26 | tzanger | milliwatt is good for that |
| 18:04.45 | coppice | tzanger: try a donald rumsey speechh |
| 18:04.51 | tzanger | coppice: hahaha |
| 18:05.38 | [TK]D-Fender | coppice: Rumsey is a 2 bit idiot.... not sure if thats enough :p |
| 18:05.43 | tzafrir | I saw an app_mwanalyze (milliwatt analyzer) or something similar. Anybody used it? |
| 18:05.56 | Shmattie | Anyone know where I can get a list of all the IMAPFLAGS options? |
| 18:06.31 | coppice | tzafrir: is that the thing where someone tried to do a clumsy FFT analysis on a looped back tone? |
| 18:07.04 | tzafrir | I just saw it existed. |
| 18:07.11 | tzafrir | Never got to test it |
| 18:13.07 | tzanger | what good would fft do on milliwatt, you don't get frequency distortion as a general rule in voip |
| 18:13.14 | tzanger | you get jitter and delay |
| 18:14.02 | coppice | tzanger: who said it made sense? |
| 18:14.13 | tzanger | coppice: indeed |
| 18:14.31 | rantsh | is there an error in this line? (it's from the sample in queues.conf) Set(MONITOR_EXEC=mv /var/spool/asterisk/monitor/^{MONITOR_FILENAME} /tmp/^{MONITOR_FILENAME}) |
| 18:14.55 | coppice | send AWGN, and cross correlate the echo.... that sounds familiar :-) |
| 18:15.04 | tzanger | coppice: yeah who did that ... hmm |
| 18:15.16 | tzanger | someone who obviously knew what he was doing |
| 18:15.37 | tzanger | only problem with awgn is it's awfully hard to try and locate corruption :-) |
| 18:16.07 | coppice | its actually pretty easy, unless the corruption only affects the last bit |
| 18:16.08 | tzanger | I am getting interesting mixing effects here |
| 18:16.22 | tzanger | I have some good hard trance in the background and listening to milliwatt |
| 18:16.36 | tzanger | as I turn my head, I get different sounds depending on where my head's turned |
| 18:18.09 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:18.34 | *** join/#asterisk ghento (n=ghento@64.180.85.230) |
| 18:20.19 | tzafrir | tzanger, ok, so a more complicated pattern could be used to test jitter and delay. |
| 18:21.22 | tzanger | well awgn wiull do that |
| 18:21.34 | tzanger | corruption's more diffuclt since there's so much data, which is why I'm sending a known pattern |
| 18:21.58 | tzafrir | you mean sliptest? |
| 18:23.04 | tzafrir | I so far have not been lucky "calibrating" it to make it print anything useful enough for me |
| 18:23.35 | tzafrir | And it doesn't check a complete line |
| 18:23.58 | dijungal | how do I refuse a call form a iax2 trunk without hanging up the call? |
| 18:24.29 | *** join/#asterisk tripps (n=ss@72-20-150-196.dhcp.cmts1.phonoscope.net) |
| 18:25.12 | tzafrir | tzanger, so in short, it appears noone has yet written anything better than the existing app_test that is in Asterisk |
| 18:25.41 | Strom_M | dijungal: step 1: don't answer it |
| 18:25.45 | Strom_M | dijungal: step 2: send congestion or similar condition |
| 18:25.57 | dijungal | ohooo |
| 18:26.09 | dijungal | but it has to be answered in other to send congestion or busy |
| 18:26.20 | dijungal | *in order |
| 18:30.25 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:31.09 | J4k3 | woo woo |
| 18:32.30 | dijungal | ahhh it worked... thanks Storm_M... i guess u can't believe everything you read |
| 18:32.44 | hmmhesays | lord I was born a rambling man |
| 18:34.28 | hmmhesays | haha |
| 18:34.52 | Darthclue | i expect clear and concise information from you. you know better. |
| 18:36.35 | hmmhesays | nice |
| 18:37.50 | *** join/#asterisk Bl0w_M0nk (n=ak@66-168-56-207.dhcp.mdsn.wi.charter.com) |
| 18:38.38 | *** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com) |
| 18:44.20 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 18:45.28 | *** join/#asterisk tripps (n=ss@72-20-150-196.dhcp.cmts1.phonoscope.net) |
| 18:45.47 | cpm | rambling is a crime that will land you at gitmo |
| 18:46.10 | *** join/#asterisk asterisk3700 (n=silver19@adsl-153-178-73.mia.bellsouth.net) |
| 18:46.11 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-171947493d7ca3c8) |
| 18:46.11 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 18:46.14 | asterisk3700 | Hello |
| 18:46.55 | asterisk3700 | Is there anyone here having issues with the parkcall(one step parking) feature? |
| 18:47.12 | Strom_M | what issue are you having? |
| 18:47.50 | asterisk3700 | I can park the call, but when I un-park, I cannot park it anymore... |
| 18:48.09 | sevard | I wish there was a browser that was a lightweight as dillo but was totally javascript compliant with a DOM error console like firefox |
| 18:48.13 | sevard | :( |
| 18:48.23 | *** part/#asterisk Darthclue (n=e054502@165.111.2.149) |
| 18:48.28 | *** join/#asterisk Darthclue (n=e054502@165.111.2.149) |
| 18:48.30 | asterisk3700 | this is only happening when I use the one step parking feature... |
| 18:49.09 | asterisk3700 | I think it's a bug in the parkcall feature.... |
| 18:50.06 | asterisk3700 | Strom_M-Are you having the same issue? |
| 18:50.27 | Strom_M | asterisk3700: no, i don't use that feature |
| 18:50.33 | file | the code was not written to do it, therefore it does not work as you expect |
| 18:50.33 | asterisk3700 | I see... |
| 18:50.47 | file | dialing options are not inherited when you pick the call back up |
| 18:51.09 | asterisk3700 | Is there any way to get around it? |
| 18:51.47 | Qwell | closed |
| 18:52.02 | file | you can use chan_local to add dialing options when you pick the call back up, Local/701@parkedcalls would pick up the parked call on 701 |
| 18:52.04 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 18:52.36 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 18:52.45 | *** join/#asterisk galeras (n=Martin@201.244.246.21) |
| 18:53.27 | asterisk3700 | Ok, so I would have to create extensions to pick it each individual parking extension right? |
| 18:53.28 | [TK]D-Fender | asterisk3700: The problem is HOW you are retreiving it. |
| 18:53.43 | hmmhesays | when its time to leave here I hope you understand, |
| 18:53.49 | *** join/#asterisk Op3r (n=edwin@125.212.117.210) |
| 18:55.19 | [TK]D-Fender | asterisk3700: Because when you pik it up through the automatically generated Extens (701+) those ONLY call ParkedCall(XXX) so you won't have any DTMF means of parking again |
| 18:55.34 | file | you can just use a pattern match instead of including the parkedcalls context... that gives you the ability to use all the features accessible from Dial, and you could also add in a "There is no call parked" prompt if you wanted as well |
| 18:55.35 | [TK]D-Fender | asterisk3700: There is a way around this is you use your brain just a little :) |
| 18:56.08 | asterisk3700 | ok, thank you for your help |
| 18:56.15 | [TK]D-Fender | asterisk3700: And add bonus' like file alluded :) |
| 18:56.16 | asterisk3700 | I think I can get it done |
| 19:00.39 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:01.01 | *** join/#asterisk jcims (n=jcims@cpe-24-26-148-163.columbus.res.rr.com) |
| 19:03.55 | jcims | is anyone doing a volunteer-based call center that i could look to for some 'lessons learned'? |
| 19:04.11 | jcims | as in staffed by volunteers |
| 19:04.55 | [TK]D-Fender | jcims: Install cameras. They tend to walk off with stuff :) |
| 19:06.16 | jcims | good point, but i'm thinking more of a 'distributed call center'...where the workers call in with a 'forwarding number' and the calls that come in are forwarded as appropriate |
| 19:07.08 | jcims | so most folks would be at home or whatnot. while i'd love to get a bootstrap on the technology, i think it's fairly straightforward. i'm more thinking of the squishy people issues that would be included. |
| 19:07.32 | jcims | i want to set up a 'homework help' line, which creates even more concerns about minors speaking with adults.. |
| 19:08.34 | Shaun2222 | [TK]D-Fender: you said with the polycom's microbrowser i could have it show a page while the phone was in use... how can i do this... the mb.idle displays when the phone is not in use and goes away when on a call. |
| 19:10.25 | [TK]D-Fender | Shaun2222: If you're already stuck on a call, why are you worring about what else is waiting? |
| 19:11.13 | Shaun2222 | [TK]D-Fender: not really stuck on a call, but it's nice to know so people can wrap it up. |
| 19:11.43 | [TK]D-Fender | Shaun2222: Well You could always go through the Services screen manually if you wanted to check in-call... |
| 19:12.41 | Shaun2222 | [TK]D-Fender: ya but i would like them to just be able to glance over at it and see.. |
| 19:13.03 | [TK]D-Fender | Shaun2222: Or you could run a custom DeviceState updating daemon and set presence for it. |
| 19:13.42 | *** join/#asterisk devonmeyers (n=devonmey@c-71-198-211-71.hsd1.ca.comcast.net) |
| 19:13.44 | Shaun2222 | hmm, how does that work? |
| 19:14.07 | devonmeyers | I need some help with remote provisioning of a polycom phone |
| 19:15.09 | *** part/#asterisk jcims (n=jcims@cpe-24-26-148-163.columbus.res.rr.com) |
| 19:16.02 | devonmeyers | anyone available to help? |
| 19:16.09 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 19:16.12 | Shaun2222 | ask your q |
| 19:16.15 | [TK]D-Fender | Shaun2222: You write a program to poll your queues and when there are calls lined up toggle a custom devicestate HINT on/off |
| 19:16.17 | blitzrage | do you have a specific question? |
| 19:16.23 | [TK]D-Fender | ~ask |
| 19:16.23 | jbot | 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. |
| 19:16.25 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 19:17.22 | devonmeyers | yes, i have a polycom 550 at a remote office, and i am trying to get it to auto provision with the Asterisk business edition server, the server has an external IP that I put in to the polycom, but it doesnt seem to be getting the files... |
| 19:17.28 | *** join/#asterisk jjasper (n=jjasper@h-66-112-162-129.connactivity.com) |
| 19:17.31 | [T]ank | I am still trying to fix an issue on my asterisk server. Have been at it for a week and getting no where. Here are the details: http://pastebin.ca/664665?srch=uniqueid. it says that sox failed, however the files are recorded and converted and stored like I expect. |
| 19:17.38 | [T]ank | the CLI> is getting flooded with the error |
| 19:17.50 | [T]ank | I have verified that Nice and Sox are installed and up to date. |
| 19:17.59 | *** join/#asterisk techie (n=techie@adsl-76-214-18-225.dsl.lsan03.sbcglobal.net) |
| 19:17.59 | [T]ank | and in the path of root |
| 19:18.29 | rantsh | anyone has had a queue where you wanted to either move the recording or change (set) a custom name to the file? |
| 19:18.33 | Shaun2222 | [TK]D-Fender: ok, the writing the program to poll queues is simple enough... the device state hint i'm alittle confused on. |
| 19:18.48 | [T]ank | rantsh: I do that |
| 19:18.49 | [T]ank | what can I help with |
| 19:19.16 | [TK]D-Fender | Shaun2222: Go lookup the Custom DeviceState patch for 1.4 on asterisk.org |
| 19:19.31 | jjasper | Hi, I am implementing Asterisk on my WAN of 12 offices, AT&T said they can do QOS but is asking me if Asterisk can tag the packets so the routers can identify it for QOS - is this possible? |
| 19:19.42 | rantsh | [t]ank I'm using asterisk 1.2.24 and there are 2 things I haven't been able to do |
| 19:20.05 | devonmeyers | any thoughts? |
| 19:20.13 | dijungal | Storm_M: apparently sending the busy tone up the IAX trunk drops the call |
| 19:20.24 | rantsh | [T]ank 1) couldn't use mixmonitor instead had to use monitor-join=yes |
| 19:20.41 | Shaun2222 | devonmeyers: how is the phone suppose to get the files? |
| 19:21.16 | rantsh | [T]ank I've tried to use the example lines in the sample queues.conf but I'm not able to establish either MONITOR_FILENAME or MONITOR_EXEC |
| 19:21.23 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 19:21.25 | [T]ank | <PROTECTED> |
| 19:21.33 | [T]ank | here is how i do it: |
| 19:21.34 | rantsh | [T]ank can you give me any pointers on how to do this? |
| 19:21.35 | devonmeyers | well when the phones are on the intranet its litterally plug and play, i just set the MAC in the asterisk system and it created the config files and the correct phone finds it |
| 19:21.39 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:22.12 | dijungal | i use mixmonitor |
| 19:22.22 | devonmeyers | but now that I have a phone that is NOT within the intranet i still want it to be able to auto provision like that, but I dont know if its possible or what settings i need to put in to the phone so that it can find the server from the outside of the intranet |
| 19:22.25 | [T]ank | http://pastebin.ca/755530 |
| 19:22.47 | Shaun2222 | devonmeyers: i'm goign to guess this asterisk system has a dhcp server handing off a boot p server |
| 19:23.01 | dijungal | i use mixmonitor but i can't get the agent names on my file names.. :( |
| 19:23.04 | Shaun2222 | devonmeyers: and in the remote location thats probably not the case. |
| 19:23.05 | devonmeyers | yes |
| 19:23.12 | [T]ank | you can either you monitor-join=yes, but if you want to change the path of where to save it to you would need to use monitor in the extensions.conf |
| 19:23.15 | rantsh | [t]ank I'm checking it right now |
| 19:23.28 | Shaun2222 | devonmeyers: you need to configure the phone settings by hand then.. |
| 19:23.37 | devonmeyers | no its not, the remote phone does have a static IP tho, and i have a static IP for the asterisk server |
| 19:23.49 | devonmeyers | manual set up is the only solution? |
| 19:23.57 | dijungal | [t]ank how do i make a custom name for the recordings... like number-agent-date-time.wav ? |
| 19:24.01 | Shaun2222 | devonmeyers: yes but it's not pulling from your dhcp server because it's remote.. |
| 19:24.18 | Shaun2222 | devonmeyers: what i do... is setup my phones to log into a ftp server and download the configs.. |
| 19:24.18 | rantsh | [T]ank the way you do it, don't you record customers while they're on MOH? |
| 19:24.26 | devonmeyers | ok thats what i was looking for |
| 19:24.30 | Shaun2222 | so each phone i have to configure a ftp server/user/pass and thats it pretty much |
| 19:24.33 | devonmeyers | but ftp not tftp? |
| 19:24.41 | dijungal | [t]ank: how do i make a custom name for the recordings... like number-agent-date-time.wav ? |
| 19:24.43 | Shaun2222 | you can do tftp |
| 19:24.49 | Shaun2222 | i just dont like it because it's open.. |
| 19:24.53 | Shaun2222 | i like authentication |
| 19:24.56 | [T]ank | dijungal: Set(CALLFILENAME=${SOME_VARIABLE_HERE}) |
| 19:24.57 | rantsh | dijungal what version of asterisk do you use? 1.2.x? how did you make it work? |
| 19:25.11 | dijungal | 1.2.19 |
| 19:25.14 | devonmeyers | yeah... well i dont believe that ABE has FTP capability, if it does i cant seem to find it |
| 19:25.32 | dijungal | right now i cannot get the agent variable working |
| 19:25.34 | [TK]D-Fender | devonmeyers: If it works locally, thena ll you need to do is set a fixed PROVISIONING SERVER IP address. |
| 19:25.44 | dijungal | i can get the callerid-date-time.wav |
| 19:25.45 | [TK]D-Fender | devonmeyers: Instead of using Option 66 to pick up the server. |
| 19:25.59 | dijungal | but i need the agent code in the name also |
| 19:26.16 | *** join/#asterisk ThatKidKel (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 19:26.22 | [T]ank | you can set a vaiable in your agents.conf and call if from the extensions.conf |
| 19:26.25 | devonmeyers | [TK]D-Fender: ok, thats what im looking for, is there a good guide online or some resources on how to set that up? |
| 19:26.37 | ThatKidKel | True or False. Asterisk is capable of video conferencing? |
| 19:26.38 | rantsh | dijungal how are you doing your recordings? how did you make mixmonitor work? i installed sox and nothing |
| 19:26.40 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 19:26.53 | [T]ank | rantsh: show us what you are trying. |
| 19:27.00 | rantsh | sure |
| 19:27.06 | [T]ank | that may be easier for us to help you |
| 19:27.38 | Shaun2222 | devonmeyers: well if you name the cofnigs with the MAC prefix chances of sombody finding your configs are probably minimal... |
| 19:27.56 | dijungal | rantsh: exten => s,n,SetGlobalVar(MON_FILENAME=${recordings}${ARG3}-${ARG2}-${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}${MEMBERINTERFACE}) |
| 19:28.13 | Shaun2222 | devonmeyers: but agian i just use ftp because i can... i would use sftp but polycom's are broken... |
| 19:28.18 | devonmeyers | Shaun2222: in regards to tftp you mean? |
| 19:28.18 | dijungal | and then i use: exten => s,n,MixMonitor(${MON_FILENAME}.wav,b) |
| 19:28.25 | Shaun2222 | devonmeyers: right. |
| 19:28.37 | rpyne | hello everyone |
| 19:28.42 | rpyne | I am currently working on asterisk realtime setup. I have the sip, voicemail and extensions working okay. But I'm not seeing any support for meetme.conf setup. |
| 19:28.45 | rpyne | Anyone has any ideas on the meetme set up, or if its available at all. |
| 19:28.53 | rantsh | dijungal [t]ank http://pastebin.com/d12624174 |
| 19:28.55 | dijungal | rantsh: but i'm not getting the memeberinterface to work |
| 19:29.51 | rantsh | all i really want is to move my calls to a different folder (if possible) and set name as agent-date.gsm/wav |
| 19:29.52 | [T]ank | rantsh: and... how do you come to the assumption that the recording is not working? |
| 19:30.01 | dijungal | u can remove line 26 - 28 |
| 19:30.04 | rantsh | recording is workin |
| 19:30.16 | [T]ank | you just need to move them... got it. |
| 19:30.21 | rantsh | what doesn't work is the renaming |
| 19:30.26 | [T]ank | can you add path to your file name? |
| 19:30.29 | devonmeyers | Shaun2222: ok... i believe tftp is what the phones use internally... so here is what I tried then, on the phone i went in and gave it a static IP, and set that static IP for DMZ (to test) and then changed under the server menu the server type to TrvialFTP, and then the server address to my astersk's external IP, left user and password settings alone |
| 19:30.39 | devonmeyers | and it doesnt seem to pull down the config files |
| 19:30.42 | [T]ank | ie... Set(MONITOR_FILENAME=/dev/shm/foo) |
| 19:31.09 | rantsh | the sample file says you can |
| 19:31.14 | [TK]D-Fender | devonmeyers: You set the provisioning method in the BootROM directly. reboot your phone and enter Setup directly on it |
| 19:31.35 | [T]ank | have you tried it? what was your experience? |
| 19:31.57 | rantsh | but I'm trying to set the recording name as foo (keeping it simple to begin with) and it doesn't work |
| 19:32.07 | [T]ank | try it my way |
| 19:32.08 | rantsh | I keep getting file names like |
| 19:32.11 | *** join/#asterisk agx (n=badpengu@81-174-47-191.dynamic.ngi.it) |
| 19:32.21 | devonmeyers | [TK]D-Fender: thats what I believe that I did, when the system starts to boot i enter the set up on the phones interface to what i mentioned earlier, with no success |
| 19:32.21 | [T]ank | dont record using the queue. |
| 19:32.28 | rantsh | 1193771812.0.gsm |
| 19:32.30 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583408.dsl.bell.ca) |
| 19:32.40 | [T]ank | do it in the extensions.conf |
| 19:32.49 | rantsh | [t]ank |
| 19:33.00 | [T]ank | both are appropriate, but to give you a different way to try it... let me build an example for you. |
| 19:33.09 | rantsh | [t]ank does it record while people is listening to MOH? |
| 19:33.33 | [T]ank | not till it is delivered to an agent would it record. |
| 19:33.54 | dijungal | [T]ank: so u'r saying set the CALLFILENAME variable to the filename plus path and then call the monitor command? |
| 19:34.16 | rantsh | let me try that and see so start monitor and then call Queue command in extensions.conf? |
| 19:34.19 | [TK]D-Fender | devonmeyers: Even if it doesn't contact the boot server it should still start with whatever config was on it prior. Perhaps its not getting a gateway to your provisioning server.... |
| 19:34.48 | devonmeyers | [TK]D-Fender: there was no config on it prior |
| 19:34.58 | [TK]D-Fender | devonmeyers: OH, that'd tdo it... |
| 19:35.23 | [TK]D-Fender | devonmeyers: Make sure your networking is setup right. test from a PC on that same side to mAKE SURE THAT'S ok |
| 19:35.28 | [T]ank | rantsh: http://pastebin.ca/755546 |
| 19:35.46 | [T]ank | dijungal: see my pastbin link |
| 19:35.52 | dijungal | k |
| 19:36.08 | [T]ank | so when the queue dials an extension, it calls this macro. |
| 19:36.15 | jjasper | anyone know how to tag asterisk packets for quality of service ? |
| 19:36.26 | [T]ank | this is how i keep it from recording hold music and how i can make it so that I can name the file as the agents extension. |
| 19:36.35 | [T]ank | if i wanted |
| 19:36.39 | *** join/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 19:36.45 | Shaun2222 | jjasper: http://www.voip-info.org/wiki-Asterisk+QoS <-- may have som einfo for you |
| 19:36.57 | dijungal | let me try that |
| 19:37.01 | dijungal | hold :) |
| 19:37.05 | devonmeyers | [TK]D-Fender: brand new phone, well i can reach the asterisk web config from my PC that is on the same network behind the same firewall and all... i dont know if TFTP is working from the outside, or ftp... but i can connect in to the server... oh and softphones work outside... because i have one on my pc that works fine |
| 19:37.11 | devonmeyers | its really just this auto provisioning issue |
| 19:37.23 | [T]ank | dijungal: you can do it a number of ways... that is how I do it. |
| 19:37.24 | jjasper | thank you Shaun2222 |
| 19:37.42 | nny | any reason why i would be continuously losing modules/ cards (tdm08) on 2 boxes with the same hardware? |
| 19:37.46 | Shaun2222 | jjasper: i didnt read it really but it's talking about asterisk and QoS so my guess is it may help |
| 19:37.48 | [TK]D-Fender | devonmeyers: You'll need to go test the specific method used |
| 19:37.50 | dijungal | i just wanna get it working.. whatever way possible.. :) |
| 19:38.12 | dijungal | how do i get the agent extension ?? ${EXTEN} ? |
| 19:38.28 | nny | I have had to replace numerous FXO modules lost to unknown circumstances, at 2 locations, but on identical hardware (msi based VIA chipsets) |
| 19:38.53 | Shaun2222 | dijungal: if i recall correctly i had to query the asterisk db manually to get that.. |
| 19:38.56 | devonmeyers | [TK]D-Fender: what do you mean by specific method used? I believe that the phones provision internally via TFTP but i dont know, like i said its plug and play, with the other phones I just unbox them and plug them in |
| 19:38.57 | nny | i get a busy signal on one right now.. using scientific method to determine if it's the module or the card itself |
| 19:39.05 | dijungal | ooooh |
| 19:39.15 | Shaun2222 | let me check my old code |
| 19:39.33 | [TK]D-Fender | devonmeyers: See one problem is that you BELIEVE things instead of KNOWING them. Get a clue, then come back :) |
| 19:40.49 | devonmeyers | [TK]D-Fender: well thanks for your helpful advice so far... but I have tried to figure out what it is using, ABE is not as open as I had hoped so I dont have a lot of options and even less feedback as to what is happening behind the scenes |
| 19:40.53 | [T]ank | dijungal: you can do ${EXTEN} for sure |
| 19:41.11 | devonmeyers | [TK]D-Fender: I dont even know where the provisioning files are kept on the server |
| 19:41.26 | [T]ank | or you can create a variable in the agent.conf file and call that variable as well. |
| 19:42.05 | [TK]D-Fender | devonmeyers: And the reason you should have bought ABE : Digium support. Go call them up.... |
| 19:42.15 | devonmeyers | $200 a call |
| 19:42.27 | [TK]D-Fender | devonmeyers: So few of us here have ever touched ABE (staff excluded) |
| 19:42.31 | devonmeyers | only offer email support, with about a 48hour turn around |
| 19:42.40 | [TK]D-Fender | devonmeyers: You mean ABE isn't SUPPORTED? What did you BUY then? |
| 19:42.53 | devonmeyers | email support and a GUI basically |
| 19:42.55 | russellb | provisioning phones isn't Asterisk support ... |
| 19:42.57 | [TK]D-Fender | devonmeyers: Oh well... that blows. Get hacking on your * box then,. |
| 19:43.09 | outtolunc | hehe |
| 19:43.11 | devonmeyers | russellb: thats what i was wondering |
| 19:43.11 | [TK]D-Fender | russellb: its part of the products function, no? |
| 19:43.31 | [TK]D-Fender | russellb: to at least inform on the where's and how's. |
| 19:43.45 | devonmeyers | russellb: i dont want to pay $200 for them to point their finger at polycom and say "its their deal" |
| 19:44.00 | Shaun2222 | cant find old code.. |
| 19:44.21 | Shaun2222 | i cant remember if it was agent->extn i had to write that code for or exten->agent |
| 19:45.10 | k31th | Main cause of jitter is bandwidth ? |
| 19:49.39 | BBHoss | jitter is kindof hard to pinpoint |
| 19:49.59 | BBHoss | it deals with many different connections and the variation in bandwidth and latency |
| 19:50.02 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:50.08 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 19:50.59 | *** join/#asterisk Flauto (n=zhao@71.194.141.225) |
| 19:51.05 | Flauto | hi all |
| 19:51.47 | Flauto | for voicemail, is it possible to copy one message to multi-users's mailboxes |
| 19:52.07 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:52.07 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:52.32 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.248) |
| 19:52.43 | k31th | BBHoss: seems to be happening every 60 seconds |
| 19:52.57 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-f8f37ce9d6229ff7) |
| 19:53.08 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584839.dsl.bell.ca) |
| 19:53.22 | dijungal | [T]ank: tried it... the problem is it records every dial attempt to the agents also |
| 19:53.52 | Shaun2222 | Flauto: i think so, i though i remember reading about that some somewhere... probably http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf |
| 19:54.04 | Shaun2222 | Flauto: look for voicemail broadcasts i think thats what you want. maybe not. |
| 19:54.12 | dijungal | so it if tries 10 agents before it finaly get's 1 agent to accept the call there will be 11 records.. one of which is the actual recording i want |
| 19:54.17 | [T]ank | dijungal: are the agents just taking one call at a time? |
| 19:54.33 | BBHoss | k31th: its probably not jitter if it happens exactly every 60 seconds +-5second |
| 19:54.36 | [T]ank | and they are being presented calls for everything? |
| 19:54.44 | BBHoss | it could be bad hardware |
| 19:54.50 | dijungal | yes.... the queue distributes the call to the next available agent.. but to do that it dials the extensions to see if they're available |
| 19:55.07 | agx | i wonder if with all the provisioning problem outside there is a room for a dedicated product for it |
| 19:55.09 | [T]ank | tell me how you add the agents to the queues. agents.conf? or dynamically? |
| 19:55.24 | dijungal | which brings up a good point.. i can probably do a chanisavail() before i run the dial command ... hmmm |
| 19:55.30 | [T]ank | correct |
| 19:55.33 | dijungal | agents.conf |
| 19:55.34 | [T]ank | what I was just going to say |
| 19:55.39 | [TK]D-Fender | Flauto: No. |
| 19:55.59 | [T]ank | or... in queues.conf in the queue, do ringinuse=no |
| 19:56.01 | Flauto | hmm.. |
| 19:56.14 | [T]ank | not sure what version that was made available though. you are on an old 1.2, right? |
| 19:56.21 | dijungal | ringinguse ... hmm.. isn't that for 1.4 |
| 19:56.29 | dijungal | i'm on 1.2.19 |
| 19:56.41 | [T]ank | checking to see when it was made avaialble |
| 19:57.03 | *** join/#asterisk Tourinho (n=gtourinh@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 19:57.05 | *** join/#asterisk tripps (n=ss@72-20-150-196.dhcp.cmts1.phonoscope.net) |
| 19:57.20 | Tourinho | hello there, does anyone know how to receive a PIN (DTMF) using agi+perl? |
| 19:57.23 | [T]ank | you are correct: "This is a feature in /trunk that was introduced post 1.2 release. As with all new features, it will not be backported to 1.2." |
| 19:57.36 | Tourinho | wait_for_dtmf receive just the first dtmf |
| 19:58.04 | *** join/#asterisk oej (n=olle@64.251.114.2) |
| 19:58.09 | dijungal | [T]ank: so i would have to do a chanisavail() |
| 19:58.13 | [T]ank | dijungal: any reason not to upgrade to 1.4? |
| 19:58.23 | dijungal | if not available i should do what??? pass a busy tone or something? |
| 19:58.29 | [T]ank | busy |
| 19:58.30 | dijungal | 1.2 is stable |
| 19:58.44 | dijungal | 1.4 is development |
| 19:59.30 | [T]ank | dijungal: Just from my own observation and from the "for what its worth category" I am running a 250 seat call center with some very complex dialplan structure on 1.4 very successfully. |
| 20:00.03 | dijungal | hmmm... |
| 20:00.07 | dijungal | i would love to go to 1.4 |
| 20:00.16 | [T]ank | tell me about your environment and use |
| 20:00.19 | agx | [T]ank, ... but ... if you have 1.2 + your own patch takes time to port personal patches to 1.4 and test them |
| 20:00.22 | dijungal | if it was up to me.. i would have done a make clean, make, make install already :) |
| 20:00.31 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 20:00.34 | [T]ank | agx: that is true |
| 20:00.45 | k31th | BBHoss: it's an IAX trunk to a provider |
| 20:01.00 | [T]ank | every persons situation is different. That is why I asked him why he did not upgrade and stated that in my own use I have been satisfied. |
| 20:01.14 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 20:01.24 | fujin_ | 1.4 working perfect here, also in a callcentre |
| 20:01.29 | fujin_ | only 50~ hardphones |
| 20:01.36 | dijungal | [T]ank: i have 1 asterisk box taking inbound calls from 2 T1s using the Te410P, sending them to queue and distributing to about 25 agents |
| 20:01.38 | agx | [T]ank, i'm going to migrate to 1.4 in 3/4 weeks i'm finishing porting apps :) |
| 20:01.39 | fujin_ | but I do some freaky shit with AEL/app_addon_sql_mysql |
| 20:02.04 | dijungal | [T]ank are you taking only inbounds at your center or are you doing outbounds also? |
| 20:02.19 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 20:02.42 | [T]ank | I am doing inbound, outbound, fax, predictive dialer, and conference bridging. I have 3 DS3s with PRI plus IAX2 and SIP providers |
| 20:02.53 | agx | Well since you have callcenters, are you using some decent FXS-SIP gateway? tryed bugstream and they sigfault X times each day... awfull experience |
| 20:03.28 | [T]ank | agx: does not do IAX2, but take a look at openSER for your SIP gateway needs |
| 20:03.32 | file | dijungal: 1.4 is not development, it's a release version just like 1.2 |
| 20:03.46 | dijungal | k |
| 20:04.18 | [T]ank | dijungal: there are a lot of improvements for call centers in 1.4. I think you would be pleased with your efforts to make the upgrade. |
| 20:04.35 | [T]ank | Do a test setup and check out the few differences in dialplan however. |
| 20:04.36 | Shaun2222 | [T]ank: what sip/iax providers you using? |
| 20:04.39 | [T]ank | There are a few. |
| 20:05.01 | [T]ank | SIP: Global Crossing and Qwest. IAX2: Binfone |
| 20:05.05 | dijungal | [T]ank if someone could point me to a good reason to upgrade then i would be glad to convince my superiors |
| 20:05.38 | [T]ank | dijungal: It will be based on your own needs. |
| 20:05.45 | dijungal | [T]ank: Qwest, VNSL |
| 20:05.58 | *** join/#asterisk mugawuki (n=admin_ae@extranet.lehighgas.com) |
| 20:06.07 | [T]ank | but there are new features, like autopause and ringinuse for your queues. |
| 20:06.18 | [T]ank | more stable IAX2 |
| 20:06.22 | [T]ank | there are many reasons |
| 20:06.33 | *** part/#asterisk Tourinho (n=gtourinh@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 20:06.47 | *** part/#asterisk jjasper (n=jjasper@h-66-112-162-129.connactivity.com) |
| 20:06.48 | Alan_Hicks | Howdy folks. I've got my test asterisk server up with a TDM400P connected to the POTS, and a single SIP phone. When a call is placed to my phone number, Asterisk answers the call and dials my SIP phone. However, there's about a three second delay between when Asterisk answers, and when it dials my internal SIP phone. |
| 20:06.52 | [T]ank | there are many cons as well... you have to make that list for yourself. |
| 20:07.06 | Alan_Hicks | Here's my [incoming] context from extensions.conf: http://pastebin.com/m6f12ed3e |
| 20:07.21 | Alan_Hicks | Is there anything I can do to cut down this delay? |
| 20:07.40 | mugawuki | Does anybody know of any decent standalone endpoint managers for SIP phones (specifically polycom)? I.E. similar to TrixBox ep manager, but without the rest of TrixBox? |
| 20:08.10 | dijungal | [T]ank: do you do outbound on asterisk also? |
| 20:08.12 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 20:08.38 | Alan_Hicks | Note: There's no perceptible delay if I place a call from my SIP phone to an outside line such as my cell phone. |
| 20:08.51 | [T]ank | dijungal: yes |
| 20:10.15 | [TK]D-Fender | Alan_Hicks: Delay would be due to * waiting for CALLER ID |
| 20:10.17 | tzafrir | Alan_Hicks, caller ID detection |
| 20:11.06 | Alan_Hicks | Ah! That makes sense. I have caller ID on my agenda of "Things To Figure Out". |
| 20:11.34 | Strom_M | Alan_Hicks: did you read that document I pointed you at this morning? |
| 20:11.51 | Alan_Hicks | Strom_M: Not yet. I've been busy reading further into the * book. |
| 20:12.09 | Alan_Hicks | Lots of reading to do, so little time. |
| 20:12.20 | Strom_M | Alan_Hicks: read telephony 101 /first/ so that your time reading the asterisk book is more useful |
| 20:13.43 | Alan_Hicks | ~101 |
| 20:13.44 | jbot | i heard 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 20:13.46 | *** part/#asterisk myiagy (n=myiagy@189.4.79.137) |
| 20:19.07 | *** part/#asterisk ThatKidKel (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 20:19.14 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 20:19.48 | Sci_05 | ok who broke asterisk.org? |
| 20:20.22 | outtolunc | i believe it is getting a makeover |
| 20:20.50 | Sci_05 | ahh, any place I can get their downloads? |
| 20:20.50 | outtolunc | new shoes and a hip tshirt |
| 20:20.55 | *** join/#asterisk rhombus (n=rhombus@d75-154-93-37.abhsia.telus.net) |
| 20:21.06 | outtolunc | various |
| 20:21.38 | tzafrir | Sci_05, download what exactly? |
| 20:22.03 | Sci_05 | I am needing to get the lastest asterisk package and zaptel. Updating a system |
| 20:22.05 | agx | Strom_M, is ASDI a thing available only on US analog telephones? |
| 20:22.23 | Alan_Hicks | Strom_M: Is there any section of 101 you recommend I read? Most of it just seems like stuff I already know. |
| 20:22.27 | tzafrir | is downloads.digium.com up? |
| 20:22.28 | Strom_M | agx: i don't know much about ADSI |
| 20:22.42 | Strom_M | Alan_Hicks: lazy lazy lazy |
| 20:23.23 | Sci_05 | yup, I should be good from here, thaks tzafrir |
| 20:23.42 | [T]ank | anyone looked at my question regarding nice and soxmix? |
| 20:23.46 | [T]ank | posted a while ago. |
| 20:23.58 | [T]ank | can repost if anyone thinks they could help |
| 20:24.04 | agx | [T]ank, yup |
| 20:24.09 | Alan_Hicks | Strom_M: No seriously. I've worked with the PSTN before, so a lot of this is just rehashing what I already know. |
| 20:24.27 | Strom_M | Alan_Hicks: and yet you weren't familiar with call progress information |
| 20:24.39 | [T]ank | repost: |
| 20:24.39 | [T]ank | (01:13:56 PM) [T]ank: I am still trying to fix an issue on my asterisk server. Have been at it for a week and getting no where. Here are the details: http://pastebin.ca/664665?srch=uniqueid. it says that sox failed, however the files are recorded and converted and stored like I expect. |
| 20:24.49 | Strom_M | Alan_Hicks: skim over the parts you're familiar with, and then read the ones that aren't familiar |
| 20:25.13 | Alan_Hicks | That's what I planned, but I was hoping you might point me direct to the meat. Thanks anyway. |
| 20:25.18 | billybongo | anyone storing voicemail on a replicated filesystem? |
| 20:25.22 | Strom_M | i dont know what you do and don't know |
| 20:25.31 | Strom_M | so therefore, i can't really guide you :) |
| 20:25.57 | Alan_Hicks | BTW, searching that doc for "call progress" turns up nothing. |
| 20:25.58 | agx | [T]ank, you using a compact flash? i noticed that i had the same problem today... |
| 20:26.11 | outtolunc | that * looks to be on the wrong side of the string |
| 20:26.12 | [T]ank | no. hard drive |
| 20:26.20 | mugawuki | billybongo: I'm not, but I've been wanting to accomplish something like that.. I was planning on trying the voicemail odbc storage, and replicate using mysql circular replication |
| 20:27.18 | mocker | Does anyone know if the grandstream's support provisioning through something like an option-66 FTP server? |
| 20:27.34 | outtolunc | tank: meaning the file....-in/out.whatevers are attempting to be rm -rf'd and failed |
| 20:28.01 | billybongo | mugawuki: I've not heard of circular replication - I've just reached a dead end with using mysql-cluster |
| 20:28.13 | billybongo | mugawuki: how does circular replication work? |
| 20:28.38 | mugawuki | billybongo: it's just poor-mans multimaster replication, it doesn't use their clustering stuff |
| 20:28.50 | agx | mocker, yes it does |
| 20:28.53 | mugawuki | billybongo: basically, it's master-slave replication, but in a loop |
| 20:28.57 | billybongo | mugawuki: any idea how fast it is? |
| 20:29.42 | mugawuki | billybongo: I've used it on asterisk realtime configs so far, but only on a local lan.. my next step (once i've ironed out the vm storage issue) is trying it at some of our slower wan sites |
| 20:29.57 | mugawuki | billybongo: on the lan it's practically instantateous |
| 20:30.13 | errr | The time of my voicemails is an hour off. The date/time on my server is correct (using date) but when I am left a voicemail the time is an hour off (asterisk 1.2.24) |
| 20:30.31 | mugawuki | billybongo: I have a link that gives a decent explanation of the circular replication.. let me see if I can find it |
| 20:31.23 | billybongo | I don't see how it would work if a node failed |
| 20:32.25 | billybongo | mugawuki: I was wondering if coda was stable/fast enough |
| 20:32.39 | mugawuki | billybongo: one of the shortcomings is that you have to intervene in the event of a node failure |
| 20:32.47 | mugawuki | billybongo: although nothing should be lost |
| 20:32.52 | *** join/#asterisk tripps (n=ss@72-20-150-196.dhcp.cmts1.phonoscope.net) |
| 20:33.28 | mugawuki | billybongo: you just tell the node downstream of the failed node that the node upstream of the failed node is its new master |
| 20:33.38 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 20:33.40 | tripps | anyone successfully register and provision a cisci 7970 with *? |
| 20:34.08 | mugawuki | billybongo: the logs should make sure that any changes during the node outage get replicated like nothing ever happened |
| 20:34.15 | mugawuki | billybongo: http://capttofu.livejournal.com/1752.html |
| 20:34.18 | billybongo | ok, well that's interesting |
| 20:35.13 | mugawuki | billybongo: I thought about trying coda/intermezzo/etc but I figured I'd give the mysql thing a shot first, since I already got it working for the rest of the ARA stuff |
| 20:35.54 | billybongo | do you have realtime working with mysql-cluster? |
| 20:35.58 | mugawuki | billybongo: that guys howto might have some gaps in it, but between that and the mysql reference, you should be abel to get it going |
| 20:36.16 | mugawuki | billybongo: i havent messed with mysql-cluster at all.. I don't know much about it, to be honest |
| 20:36.28 | mugawuki | billybongo: what's the problem you're having with it? |
| 20:36.55 | billybongo | well it's ideal except for: it needs to hold all the live data in memory and it doesn't cope with blobs |
| 20:37.05 | billybongo | so great for your realtime sip users etc |
| 20:37.07 | mugawuki | billybongo: I see |
| 20:37.16 | billybongo | not so great for storing voicemail |
| 20:38.13 | mugawuki | billybongo: I was definitely a bit concerned about performance with the voicemail odbc storage and replication, but with how small the voicemail blobs are, I'm hoping it can cope |
| 20:38.23 | mugawuki | billybongo: plus our vm volume is relatively low here |
| 20:39.08 | tripps | ~7970 |
| 20:39.27 | tripps | jbot: bestow some magic upon me pls :) |
| 20:40.43 | Strom_M | ~cohujibuggle |
| 20:40.44 | jbot | methinks cohujibuggle is gublgubbglggugglbuglgbugblgbgbgbgbglbglgbulgblugbgubgublgbglulllbgbb |
| 20:41.05 | tripps | mmm must have been asleep |
| 20:41.07 | tripps | :) |
| 20:41.22 | billybongo | mugawuki: yeah, the blobs themselves are small. I was also thinking about using it for ivr |
| 20:43.18 | mugawuki | mugawuki: my hope is to use it to keep config in sync across * machines at several sites, and let their sip phones pick the best proxy with dns srv.. hopefully that'll give us some fault tolerance |
| 20:43.38 | mugawuki | billybongo: that was for you actually |
| 20:43.43 | mugawuki | billybongo: lol |
| 20:43.49 | billybongo | mugawuki: yeah, that's a bit like what I'm doing except all in one place |
| 20:44.10 | billybongo | I'm surprised there isn't more info out there about this |
| 20:45.07 | mugawuki | billybongo: yeah, that's what I was thinking.. if I get this working half decent, I'll probably throw together a tutorial to save others the pain :) |
| 20:45.19 | billybongo | alternatively I could use nfs with regular backups and a failover plan, but that sounds like too much effort |
| 20:45.31 | mugawuki | billybongo, I tried that first |
| 20:45.53 | *** join/#asterisk karleeto (n=karl@207.191.91.242) |
| 20:45.57 | mugawuki | billybongo: it sets up real easy |
| 20:46.27 | mugawuki | billybongo: but i spent too much time figuring out a good way to failover in case of a wan link outage |
| 20:46.34 | billybongo | still a single point of failure and I'm trying to avoid those |
| 20:46.46 | mugawuki | billybongo: so I gave it up.. although it may be I just don't know enough about that |
| 20:46.57 | billybongo | mugawuki: how are you planning to load balance your sql connections? |
| 20:48.21 | Bl0w_M0nk | mugawuki if you do can ya throw the tutorial my way?? plz :-)) |
| 20:48.37 | billybongo | mugawuki: did you ever look into postgresql/slony ? |
| 20:49.18 | mugawuki | billybongo: I wasn't really, at least not in the beginning.. our volume doesn't really justify it.. the only db update events we'll have on a daily basis are the voicemails and sip registrations, etc and the rest should be local selects that I'm hoping are well-indexed |
| 20:50.24 | mugawuki | Bl0w_M0nk: no problem.. if I ever get it together, you're welcome to it! |
| 20:50.35 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:50.38 | mugawuki | mugawuki: no.. should I be looking into that? |
| 20:51.56 | mugawuki | billybongo: talking to myself again |
| 20:52.36 | billybongo | mugawuki: it seems to be a similar thing for postgres, I haven't looked at it properly yet |
| 20:52.51 | mugawuki | billybongo: I see |
| 20:52.52 | billybongo | mugawuki: how do you solve the problem of auto incremented ids? |
| 20:53.14 | billybongo | two simultaneous writes are going to produce identical and conflicting ids |
| 20:53.19 | mugawuki | billybongo: when you do circular replication, you set an offset |
| 20:53.20 | Bl0w_M0nk | thanks :-)) |
| 20:53.38 | mugawuki | billybongo: i.e. you set a maximum number of servers in the replication group |
| 20:53.44 | mugawuki | billybongo: could be 100 |
| 20:53.50 | billybongo | mugawuki: clever |
| 20:53.53 | billybongo | that makes sense |
| 20:53.55 | mugawuki | billybongo: then you set a different offset for each server |
| 20:54.04 | billybongo | so one server is always ending 100 the next 101 etc |
| 20:54.11 | mugawuki | billybongo: yet |
| 20:54.15 | mugawuki | billybongo: yep |
| 20:54.29 | billybongo | ok, I'm going to have to look into this |
| 20:54.33 | billybongo | it should work for voicemail |
| 20:54.51 | billybongo | and for config I'll use mysql-cluster because that seems very fast |
| 20:54.59 | billybongo | although for me that's openser |
| 20:55.16 | mugawuki | billybongo: gotcha |
| 20:55.30 | mugawuki | billybongo: good luck with that.. let me know how it turns out |
| 20:55.56 | billybongo | mugawuki: what happens in the case of a network split? |
| 20:56.07 | billybongo | does it all sort out once reconnected? |
| 20:56.10 | mugawuki | yes |
| 20:56.14 | billybongo | and what happens to resolve conflicts? |
| 20:56.33 | billybongo | if the same entry changes in both places during a split? |
| 20:56.38 | mugawuki | when you join a machine to the replication group, it syncs to its masters log |
| 20:56.50 | mugawuki | billybongo: so when the network splits |
| 20:56.59 | mugawuki | billybongo: the master log stops incrementing |
| 20:57.04 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:57.25 | mugawuki | billybongo: when connectivity comes back, it knows which spot in the master log to pick up replication |
| 20:57.50 | *** join/#asterisk blq (n=Bl@dslb-088-067-029-030.pools.arcor-ip.net) |
| 20:57.52 | *** join/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 20:57.56 | billybongo | ok but if it changes in two places they will try to change each other on reconnection |
| 20:58.01 | nny | any aastra guys here? |
| 20:58.22 | nny | I remember there being an issue with the firmware name in the zip file from aastra (heh) not being correct |
| 20:58.28 | mugawuki | billybongo: well now I'm making assumptions, but I think each machine watches it's master log independently |
| 20:58.34 | nny | trying to remember what* it is supposed to be :) |
| 20:58.48 | billybongo | mugawuki: but lets say you have 2 machines |
| 20:58.55 | rhombus | nny: i'm using aastra right now -- what's the problem? |
| 20:58.59 | mugawuki | billybongo: right |
| 20:59.02 | billybongo | and both have the same record changed |
| 20:59.07 | billybongo | they are each the master for the other |
| 20:59.15 | billybongo | so they will push down the change |
| 20:59.27 | mocker | ~grandstream |
| 20:59.28 | jbot | somebody said grandstream was the Yugo of VoIP hardware. Run. Run away now. |
| 20:59.43 | *** join/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net) |
| 20:59.46 | tripps | ~cisco |
| 20:59.46 | 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! |
| 20:59.47 | nny | rhombus: no problem per se.. trying to upgrade to latest firmware.. i distinctly remember the original phone i had needing me to change the name of the firmware file |
| 20:59.58 | billybongo | mugawuki: I think this is something that mysql-cluster gets much better |
| 21:00.04 | nny | right now it is "480i CT.st" |
| 21:00.09 | mugawuki | billybongo: I like to hope that's unlikely but I understand what you're saying.. I just don't know what happens in a case like that.. you'd have to try it |
| 21:00.12 | nny | which i remember is wrong :( |
| 21:00.15 | vAd0r | can someone tell me if i can use my cisco vg200 to call out on my home phone? |
| 21:00.22 | rpyne | exit |
| 21:00.50 | [T]ank | is there a bot for linksys? |
| 21:00.50 | [T]ank | ~linksys |
| 21:00.51 | jbot | i guess linksys is a tool of satan |
| 21:00.51 | [T]ank | :-( |
| 21:00.51 | billybongo | mugawuki: right I'll give it a go |
| 21:00.51 | rhombus | nny: hmn... well, i used the firmware file as packaged, and it worked fine |
| 21:00.51 | rhombus | the one you have to be careful of is the case on the cfg files :) |
| 21:00.51 | billybongo | billybongo: possibly better to use 3 servers rather than 2 to start with |
| 21:00.53 | nny | rhombus: hmm this was from aastra tech support, let me see what I can figure out |
| 21:00.53 | billybongo | mugawuki: ^^ |
| 21:01.03 | mugawuki | billybongo: yes, that's what I used in my test lab |
| 21:01.32 | [T]ank | so i have sold my sole... been happy about it ;-) |
| 21:01.45 | mugawuki | billybongo: again.. I think if you have an extended outage.. you'll want to bridge the gap over the failed link/machine |
| 21:02.02 | mugawuki | billybongo: to reduce the possibility of a conflict like the one you mentioned |
| 21:02.32 | mugawuki | billybongo: but it's just something you have to setup and beat up on for a while before you'll be able to trust it |
| 21:04.08 | agx | with 1.4 can i have 2 phones sharing the same [peer] section in sip.conf ? |
| 21:04.46 | mugawuki | agx: good question, I was also wondering that |
| 21:05.13 | [TK]D-Fender | No |
| 21:05.22 | mugawuki | bummer |
| 21:05.34 | *** join/#asterisk bantu (n=Miranda@p54A31294.dip0.t-ipconnect.de) |
| 21:06.46 | tzanger | there's an app that will take an RTP stream and decode it to wav... what was it called? |
| 21:07.06 | [TK]D-Fender | tzanger, I'm sure theres a Wireshark plugin for that. |
| 21:07.22 | tzanger | wireshark will decode it as rtp |
| 21:07.27 | tzanger | but the payload doesn't make sense to me |
| 21:07.32 | tzanger | i.e. it's a ulaw payload of miliwatt |
| 21:07.42 | tzanger | 20ms packetization by default |
| 21:07.49 | errr | The time of my voicemails is an hour off. The date/time on my server is correct (using date) but when I am left a voicemail the time is an hour off (asterisk 1.2.24) any ideas why? |
| 21:07.55 | tzanger | but I can't find my ulaw stream in there |
| 21:08.14 | agx | tzanger, wireshark does not decoded it? is the payload the correct size? |
| 21:08.17 | [TK]D-Fender | errr, check your timezones in voicemail.conf |
| 21:08.58 | tzanger | agx: yes |
| 21:09.08 | tzanger | it doesn't make a wav from it but it shows sequence, etc. |
| 21:09.16 | tzanger | but the payload, as I said, does not seem to contain my ulaw data |
| 21:10.17 | agx | tzanger, is it from misdn? i will start checking a working payload and the broken one for size of each packet |
| 21:10.49 | [hC] | can anyone think of a reason why i might get one way audio on an incoming Zap call, to a sip device on the lan?? |
| 21:12.47 | [TK]D-Fender | [hC], what Zap interface exactly? |
| 21:12.50 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:12.54 | tzanger | no, it's jsut a sip ulaw rtp stream |
| 21:12.57 | generalhan | hey all ! |
| 21:13.01 | *** join/#asterisk jetlagmk2 (n=jetlag@70.17.52.96) |
| 21:13.08 | tzanger | the payload is fine, I just can't "see" my payload data in it |
| 21:13.16 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 21:13.16 | tzanger | i.e. milliwatt is a very specific data pattern |
| 21:13.22 | tzanger | I should be able to easily spot it |
| 21:13.24 | tzanger | I think |
| 21:13.27 | generalhan | is there a way to logon an Agent via the CLI? like Agent logoff does for logging an Agent out ? |
| 21:13.57 | errr | [TK]D-Fender: so I need a tz=<zone> and the <zone> needs to be a valid zone from /usr/share/zoneinfo/ ? |
| 21:14.14 | [TK]D-Fender | errr, makes sense |
| 21:14.34 | ReDNeQ | sup |
| 21:14.36 | vAd0r | can someone recommed to me what card to buy to hook up my 1 home phone line from att |
| 21:14.45 | errr | [TK]D-Fender: so would I need America/Chicago or just Chicago |
| 21:14.58 | [TK]D-Fender | errr, Get WIKI-ing for samples |
| 21:15.01 | [hC] | [TK]D-Fender: Im not sure what you mean specifically, but its a Sangoma A200d FXO incoming call, ringing via SIP to a polycom and the customer was saying they got one way audio, and I dont see how. |
| 21:15.21 | [TK]D-Fender | [hC], Do you get 1-way to a soft-phone against the same reg? |
| 21:15.36 | [TK]D-Fender | [hC], is it only 1-way with Zap? |
| 21:15.55 | [hC] | [TK]D-Fender: er, havent tried. only one way with zap, when their internet connection is down. normally they trunk in via IAX and thats fine. |
| 21:16.07 | [hC] | [TK]D-Fender: the situation is, internet is down, and calls come in via zap in a failover setting. |
| 21:16.40 | *** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net) |
| 21:17.00 | [TK]D-Fender | [hC], the Otasic EC make have locked up and is buffering all the audio. heck that |
| 21:17.05 | errr | [TK]D-Fender: tz=san-diego is all I found in the wiki and I dont have a /usr/share/zoneinfo/America/san-diego or a /usr/share/zoneinfo/san-diego and wouldnt the default tz be what ever /etc/localtime is? |
| 21:17.07 | jayk- | is there a way i can dial a number from the asterisk console and forward the call to a sip phone? |
| 21:17.23 | [TK]D-Fender | errr, faith-- go check, and try stuff |
| 21:17.38 | [TK]D-Fender | jayk-, make a call file |
| 21:17.55 | jayk- | TK: how do i do that? |
| 21:17.57 | errr | [TK]D-Fender: what about the default zone stuff, am I right to think that what ever is in /etc/localtime |
| 21:18.00 | [TK]D-Fender | jayk-, Here : |
| 21:18.02 | [TK]D-Fender | ~wikis |
| 21:18.02 | jbot | hmm... wikis is http://www.voip-info.org |
| 21:18.09 | [TK]D-Fender | jayk-, go look up "call files" |
| 21:18.18 | jayk- | ok |
| 21:18.19 | jayk- | thanks |
| 21:18.20 | jayk- | got it |
| 21:20.16 | [hC] | [TK]D-Fender: if that had happened (apparently it doesnt ALWAYS happen) what would cause it and what would i do to fix it? new drivers? |
| 21:20.26 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 21:20.57 | [TK]D-Fender | [hC], An idea. There is a quick fix I don't recall off-hand. Call them about it |
| 21:21.17 | [hC] | [TK]D-Fender: will do. thanks. |
| 21:23.20 | errr | [TK]D-Fender: I have tred both tz=America/Chicago and tz=Chicago and done a reload each time I changed but the time is still an hour off, any other suggestions? |
| 21:23.33 | atomicd | join #centos |
| 21:23.34 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 21:23.46 | atomicd | oops... |
| 21:23.47 | MacWinner | if i'm running asterisk inside a VM on a co-lo box, can i configure a second IP address on the VM and set the MAC address to the same as the normal box? |
| 21:23.47 | hohum | which asterisk variable corresponds to the request URI (not the To:) |
| 21:23.55 | MacWinner | i have a second IP assigned by the co-lo |
| 21:25.21 | nny | noob q, where do sip auth messages show? |
| 21:25.35 | Alan_Hicks | Strom_M: Ok, I've read it all save what I already understood and what was just marketing stuff, but I still have no idea what you mean by "call progress information |
| 21:25.38 | Alan_Hicks | 20:24 < [T]ank> repost: |
| 21:25.45 | Alan_Hicks | Oops. :^) |
| 21:26.07 | hohum | which asterisk variable corresponds to the request URI (not the To:) |
| 21:26.16 | [TK]D-Fender | MacWinner, wrong channel for that question. |
| 21:26.58 | MacWinner | [TK]D-Fender: sorry, second question was going to be whether the bridge vs NAT makes a difference to Asterisk running in the VM if I'm using IAX2 trunking |
| 21:27.39 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-69-66.dsl.irvnca.pacbell.net) |
| 21:28.04 | mitcheloc | is anyone familiar with this? "chan_iax2.c:6877 socket_process: Rejecting packet from '61.220.69.66.4569' that is flagged as a full frame but is too short" -- trying to dial from one asterisk system to another via IAX2 |
| 21:28.44 | [hC] | i dont know if this could help anyone, but I seem to have found an issue in cdr_addon_mysql.so even up to current date. if you are using this, and your internet goes down, it can/will cause your phones to hang. if asterisk cannot write to the cdr record, the sip stack seems t ohang until it times out. |
| 21:29.30 | rhombus | I can't make calls out of my partial PRI. Inbound calls work. Does anybody want to look at my CLI output? |
| 21:30.05 | rhombus | http://pastebin.ca/755682 |
| 21:30.36 | Strom_M | rhombus: uh |
| 21:30.36 | rhombus | be warned, this is FreePBX -- but I'm pretty sure the problem is with the PRI setup. |
| 21:31.19 | rhombus | Strom_M: you were saying? :) |
| 21:31.23 | Strom_M | rhombus: are you sure you're only supposed to be sending seven digits? |
| 21:31.38 | Strom_M | and are you sure you're supposed to be hunting from channel 1 upwards? |
| 21:31.41 | rhombus | Strom_M: no, not at all. I'm a PRI virgin. Maybe it's rejecting the call because of that |
| 21:31.55 | Strom_M | rhombus: call the telco and find out what they want |
| 21:31.58 | rhombus | Strom_M: no, not sure about that either |
| 21:32.19 | rhombus | Strom_M: how do I set the hunt orientation? |
| 21:32.30 | Strom_M | with asterisk it's easy |
| 21:32.35 | Strom_M | fuck if i know how to do it in freepbx |
| 21:32.38 | rhombus | It would be the same here |
| 21:32.50 | Strom_M | probably involves jumping on your head and baking a cake |
| 21:32.54 | rhombus | It's in zapata.conf, right? FreePBX doesn't touch that |
| 21:32.59 | Strom_M | nope |
| 21:33.05 | Strom_M | it's in the dial() command |
| 21:33.20 | rhombus | ah, okay, then I have an idea where I need to do it |
| 21:33.32 | generalhan | anyone in here know of a good site, other than the wiki, for teaching how to use the Manager API? maybe that has some good sample scripts ? |
| 21:33.58 | Katty | how do i show the directory application at the CLI? |
| 21:34.09 | Katty | i'm dumb and not seeing it.. |
| 21:34.17 | Katty | is it called something else? |
| 21:34.22 | Strom_M | Katty: core show application directory |
| 21:34.27 | *** join/#asterisk saftsack (n=saftsack@pD9E05C60.dip.t-dialin.net) |
| 21:34.37 | Katty | thanks |
| 21:34.47 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 21:34.55 | Katty | my brain dun work anymore today |
| 21:35.44 | [TK]D-Fender | rhombus, Cause No. 28 - invalid number format (address incomplete). <--- |
| 21:35.52 | rhombus | thanks |
| 21:35.56 | rhombus | so they want 10 digits |
| 21:36.00 | rhombus | makes sense |
| 21:36.03 | [TK]D-Fender | rhombus, -- Executing Dial("SIP/100-0817a760", "ZAP/g0/3380140|300|") in new stack <--- ist guess... maybe your provider requires 10 digit dialing |
| 21:36.15 | rhombus | now to figure out how to send 10 :) |
| 21:36.22 | [TK]D-Fender | rhombus, ........ |
| 21:36.25 | [TK]D-Fender | #freepbx |
| 21:36.27 | [TK]D-Fender | :p |
| 21:36.43 | rhombus | [TK]D-Fender: thanks, I've already been there :) |
| 21:36.57 | [TK]D-Fender | rhombus, Then it should be a familiar trip BACK :) |
| 21:37.22 | rhombus | [TK]D-Fender: be nice |
| 21:37.22 | Strom_M | i wonder what goes on in #freepbx: "ISDN??? IS THAT LIKE ADSL???" |
| 21:37.37 | nny | Strom_M: it isn't.. |
| 21:37.43 | nny | ?? |
| 21:38.42 | nny | wow you guys would like the panama system we are working on... holy moly |
| 21:38.47 | *** join/#asterisk pat2man (n=pat2man@ip67-90-247-203.z247-90-67.customer.algx.net) |
| 21:38.47 | rhombus | I'll go where I need to go to get my questions answered |
| 21:38.48 | [TK]D-Fender | Strom_M, No, ADSL is like ADSI because they share 3 letters! |
| 21:39.12 | Strom_M | i thought ADSL was closer to salad |
| 21:39.13 | nny | someone sold our client a 10k system with 8 snom phones and it is a total POS |
| 21:39.34 | [TK]D-Fender | Strom_M, tossed with a little balsamic vinagrette |
| 21:39.38 | Strom_M | mmmmmmm |
| 21:39.39 | rhombus | no, ADSI is closer to salad |
| 21:39.49 | Strom_M | rhombus: that's more of a caesar |
| 21:39.56 | nny | gir,... aastra configs are a bastard |
| 21:40.28 | nny | hey whats your username for SIP? ok.. whats the pass: ok.. (in the mac.cfg file) and i get <sip:No%20User@192.168.100.15:5060>' failed |
| 21:40.33 | nny | hmm |
| 21:40.59 | nny | i remember the aastra tech telling me "well, sometimes the config file and the phones automagic web crap collide |
| 21:41.19 | Strom_M | nny: that is quality engineerification |
| 21:41.23 | nny | oh yes |
| 21:41.42 | nny | not to mention BY DEFAULT the firmware for 1.41 aastra absolutely kills the phone when you hit the left key |
| 21:41.52 | rhombus | Polycoms are no less confusing |
| 21:42.19 | nny | http://www.youtube.com/watch?v=tBs4Y41r3O8 |
| 21:42.22 | nny | heh |
| 21:42.44 | nny | i set up 10 polycoms yesterday in 2 hours |
| 21:42.54 | nny | i have spent 1 and 1/2 hour so far with ther aastra |
| 21:43.04 | nny | (doing this for cordless, mainly) |
| 21:43.08 | nny | i want kirsk damnit |
| 21:43.10 | nny | kirks* |
| 21:44.07 | s34n | nny: I hear nokia makes a decent cordless ;) |
| 21:44.19 | nny | o rly? |
| 21:44.37 | nny | 802.X or proprietary pairing? |
| 21:44.46 | s34n | and they come free with a 2 year contract |
| 21:44.57 | nny | (trying to avoid wireless 802.11 etc. atm) |
| 21:45.18 | fujin_ | dect cordless and analogue to sip translators |
| 21:45.40 | nny | hrrm i hear bad things about the analog-sip stuff |
| 21:45.48 | nny | may just be the troll under my desk though |
| 21:45.50 | nny | :) |
| 21:46.19 | fujin_ | analogue<->sip > 802.11 sip |
| 21:46.28 | nny | yah true that |
| 21:46.44 | nny | the aastra (have one out in the wild) works great once you sacrifice the proper small animals |
| 21:46.55 | fujin_ | ha |
| 21:47.10 | *** join/#asterisk el_critter (n=chatzill@190.74.96.121) |
| 21:47.29 | nny | although they don't let you ring the base station and handset at the same time :( |
| 21:47.38 | nny | well actually *if one is in use |
| 21:47.38 | fujin_ | ;\ |
| 21:47.51 | nny | meh who needs a phone that rings anyways |
| 21:47.58 | fujin_ | indeeda |
| 21:48.04 | fujin_ | all of the phones down here in the NOC have flashy lights |
| 21:48.06 | fujin_ | disabled the ringers |
| 21:48.21 | fujin_ | it'd be carnage, ringing phones.. ;\ |
| 21:48.26 | nny | heh disabled or "can't hear them" |
| 21:48.39 | fujin_ | lowest volume |
| 21:48.57 | nny | meh gonna have to get aastra tech support on the phone again.. must.. document.. small animal sacrifice |
| 21:49.31 | [hC] | nny: i have asked aastra for the same thing recently. |
| 21:49.43 | [hC] | you're talking about the CT here yes? |
| 21:49.45 | s34n | My ITSP is giving me a 500 error on INVITE :( |
| 21:49.53 | Darthclue | flashy lights? That is so last year. You should get the mind interface. It let's you see who is calling before they even call and even offers the ability to autoforward to the monkeys. |
| 21:49.55 | nny | [hC]: yeah i distinctly remember that the (yes) firmware name is different than what is in the zip file |
| 21:50.10 | nny | [hC]: and the sip auth stuff is confusing |
| 21:50.14 | nny | getting auth errors here |
| 21:50.27 | *** join/#asterisk analyysi (n=ayrjola@cs181173201.pp.htv.fi) |
| 21:50.35 | nny | <PROTECTED> |
| 21:50.40 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 21:50.46 | *** join/#asterisk anthm (n=anthm@mb70736d0.tmodns.net) |
| 21:50.46 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:50.47 | [hC] | thats odd. |
| 21:50.49 | nny | saying it is blank, but my mac.cfg has the line1 user/ pass setup |
| 21:51.07 | nny | OTICE[2369]: chan_sip.c:14861 handle_request_register: Registration from 'sip:101@192.168.100.15:5060' failed |
| 21:51.19 | [hC] | you could pastebin your config and i can look |
| 21:51.28 | [hC] | or you can check for whitespace at the end of the line too, that may be screwing you up |
| 21:51.33 | nny | k let me check |
| 21:51.43 | [hC] | but pastebin.ca your mac file and ill look for you |
| 21:51.47 | nny | k |
| 21:54.40 | nny | one sec, think i had some white space on my config.. getting tissue |
| 21:55.09 | nny | [hC]: do you know what the 480 firmware filename is supposed to be verbatim |
| 21:55.29 | [hC] | 480i or 480iCT? |
| 21:55.36 | generalhan | nny: the ones that i DLed are just called 4801.st |
| 21:55.47 | [hC] | its either 480i.st or 480iCT.st i believe |
| 21:55.56 | generalhan | and 4801 CT.st |
| 21:56.04 | generalhan | the CT has a space |
| 21:56.11 | nny | 1.42 is "480i CT.st" |
| 21:56.12 | [hC] | really??? |
| 21:56.15 | nny | out of the zip file |
| 21:56.17 | [hC] | thats stupid. |
| 21:56.21 | nny | but the firmware doesn't upload on boot |
| 21:56.30 | nny | and i remember that the issue was the filename |
| 21:56.31 | generalhan | yea i didnt like the space either |
| 21:56.31 | *** part/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 21:56.31 | [hC] | do you know what the phone requests from the tftp/ftp/http server? |
| 21:56.38 | nny | i have another server, but i need to get access to it |
| 21:56.51 | nny | [hC]: i need to figure that out somewhow |
| 21:56.53 | nny | somehow* |
| 21:56.58 | generalhan | i didnt have to change anything i DLed it, unzipped it to my tftp dir and BAM, done |
| 21:57.09 | nny | generalhan: you sure you are running it? |
| 21:57.16 | generalhan | haha ! yes |
| 21:57.20 | nny | generalhan: k |
| 21:57.37 | generalhan | i have a 480i and a 480iCT and both have the latest firmware and are working ! |
| 21:57.42 | generalhan | brb cig break |
| 22:00.38 | nny | [hC]: http://pastebin.com/mdee37ca |
| 22:00.45 | nny | [hC]: not very much in it |
| 22:01.43 | [hC] | no whitespace? and aastra.cfg specifies the register server and proxy server |
| 22:03.22 | nny | [hC]: yes they are one and the same |
| 22:03.46 | nny | [hC]: gonna tell the phone to hit factory defaults and redownload it's stuff |
| 22:03.51 | [hC] | hm, should work then. in the phones menus can you get the firmware version and verify you are on a new(ish) version? |
| 22:03.56 | nny | nope |
| 22:03.58 | [hC] | yeah factory resetting them is necessary sometimes |
| 22:03.58 | nny | on old version |
| 22:03.59 | [hC] | for some reason |
| 22:04.03 | nny | it won't get the new one |
| 22:04.03 | [hC] | and find a way to update it |
| 22:04.06 | nny | even though it is in there |
| 22:04.12 | nny | yah name issue methinks |
| 22:05.38 | nny | wish i had ssh to the other box i have |
| 22:06.15 | nny | will try some variants |
| 22:06.24 | nny | maybe upper/lowercase issue or space |
| 22:06.26 | Alan_Hicks | I should know how to do this, but for the life of me I can't remember. If I have two analogue lines connected to FXO modules and I want to place an outbound call on whichever one is available, can't I specify that as something like Zap/1&Zap/2? |
| 22:06.56 | Alan_Hicks | Say: Dial(Zap/1&Zap/2/${EXTEN:1}) |
| 22:07.14 | [hC] | Alan_Hicks: thats why zaptel has groups |
| 22:07.22 | [hC] | in zapata.conf you specify group=0 or group =1 or whatever |
| 22:07.26 | Alan_Hicks | [hC]: Thanks. |
| 22:07.31 | [hC] | then when you dial you do Dial(Zap/g1/number) |
| 22:07.44 | karleeto | we did a 10 phone job a week ago, and and hooked straight from the soundcard to their old overhead paging system so they could continue to use it. i did this according to some directions on voip-info.org, pretty simple, works great, however, it seems that one employee is complaing that on his phone (ONLY HIS PHONE) he will hit the PAGE (i programmed the 3rd line key to be the "Page" button), button and it will just automatically be playing music-on-hold o |
| 22:07.46 | [hC] | the g1, g0, g2, whatever you put, indicates the group and uses whatever is available. |
| 22:07.50 | [hC] | sequentially. |
| 22:08.13 | Alan_Hicks | So just group=n for some context. |
| 22:08.33 | [hC] | yeah, then all the channel= lines below that will become children of group=n |
| 22:08.49 | [hC] | if you do like.. group=1 channel=1-3 group=2 channel=4-8 |
| 22:08.56 | [hC] | then g1 = channels 1-3, group2 = channels 4-8 |
| 22:08.58 | nny | anyone ever made a hybrid system that uses sip lines (like broadvoice) as extra channels? |
| 22:09.22 | [hC] | nny: as far as i know mitel also does that. |
| 22:10.15 | nny | [hC]: heh yeah for 100k plus an unborn child |
| 22:10.54 | nny | [hC]: i figured it would be trivial to setup an asterisk box with FXO etc. on some ports, and then some sip lines for extra outbound stuff at cheaper rates |
| 22:11.13 | nny | broavoice is 25-30 bucks a month unlimited international per trunk.. we have local only here for 50 a month |
| 22:11.23 | Alan_Hicks | [hC]: Thanks again. I know that Zap/1&Zap/2 syntax is good somewhere. Must be in zapata.conf. |
| 22:12.02 | [hC] | Alan_Hicks: You can use multiple lines like that, its most useful for ringing sip devices. Zaptel just provides a cleaner way to do it, even though technically that would work. |
| 22:12.09 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 22:12.11 | [hC] | Alan_Hicks: they just happened to have designed groups for that purpose |
| 22:12.51 | *** join/#asterisk geparkt (n=geparkt@lx10-hrz.uni-duisburg.de) |
| 22:12.55 | Alan_Hicks | [hC]: When I tried that Zap/1&Zap/2 syntax in my dialplan, I got an error to the console. Let me see if I can drag it up. |
| 22:13.37 | geparkt | Anyone using CAPI with asterisk-gui? |
| 22:13.50 | nny | ha now this aastra won't boot.. have to let it sit for a minute :\ |
| 22:15.31 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 22:16.01 | agx | nny, read the aastra official doc, if you turned it off during firmware upgrade it is now on a recovery subnet; also run arpwatch on that subnet, it helps a lot |
| 22:16.02 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 22:16.02 | Alan_Hicks | [hC]: http://pastebin.com/md4d1179 |
| 22:16.36 | Alan_Hicks | Asterisk would just hang up when I used that syntax. I gotta be doing something wrong there. |
| 22:18.05 | [hC] | Alan_Hicks: well first of all, DO use groups, it is the 'right' way to do it. your way could work, but the dial command would have to read Dial(Zap/1/number&Zap/2/number) and would actually dial out over both lines if both were free, so its not the way to go. |
| 22:18.24 | [hC] | you want OUTBOUND_TRUNK= to read Zap/g1 |
| 22:18.32 | [hC] | and put group=1 before your channel line in zapata.conf |
| 22:18.34 | nny | agx: nah it is fine now.. heh i wish i could get the firmware to upgrade |
| 22:18.40 | Alan_Hicks | [hC]: Yeah, when I did that the way you said everything was fine. |
| 22:18.48 | nny | rar |
| 22:19.05 | agx | nny, you doing something wrong, even my uber-noobish employer was able to configure that aastra phones :))) |
| 22:19.07 | nny | and it won't get the proper auth info from the mac.cfg file |
| 22:19.12 | nny | agx: hah! |
| 22:19.14 | Alan_Hicks | I just didn't understand why it didn't work when I tried it the prior way. |
| 22:19.28 | nny | agx: he prolly used the web interface |
| 22:19.34 | Alan_Hicks | Even then, shouldn't asterisk dial out both lines? All it was doing was hanging up. |
| 22:19.42 | [hC] | Alan_Hicks: because the syntax was wrong. and it would not do what you wanted it to do, even if it did work. |
| 22:19.52 | nny | agx: i had aastra tech support tell me that the firmware for the 480i CT is named wrong, and the auth issue is still a mystery |
| 22:20.07 | [hC] | Alan_Hicks: your syntax was making it actually execute: Dial(Zap/1&Zap/2/123456) where 123456 is the number |
| 22:20.20 | [hC] | it would have had to read Zap/1/123456&Zap/2/123456 |
| 22:20.22 | Alan_Hicks | [hC]: Just for my own curiosity, what is the proper syntax if I didn't use a group? |
| 22:20.28 | Alan_Hicks | Ah I see. |
| 22:20.45 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 22:20.46 | nny | heh under the menu --SIP SETTINGS, User Name: BLANK, even though in mac.cfg it has the proper info |
| 22:21.00 | nny | ironically the display name is right |
| 22:21.17 | nny | auth name is right |
| 22:21.18 | nny | hmm |
| 22:21.27 | nny | 'No User <sip:No%20User@192.168.100.15:5060>' |
| 22:21.47 | agx | nny, cannot be of help now, 23.21, no access to office files :) |
| 22:21.53 | nny | np np |
| 22:21.57 | [hC] | nny: figure out upgrading the firmware first. you may save yourself a lot of time. do you not have a machine you can install a web, ftp or tftp server on and put the file there? |
| 22:21.57 | nny | just moaning and bitching out loud |
| 22:22.06 | *** join/#asterisk anujsingh (n=anuj@59.94.132.105) |
| 22:22.08 | nny | it should work here |
| 22:22.56 | nny | i gets all other info from ftp |
| 22:23.00 | nny | just not the firmware file |
| 22:23.10 | [hC] | what does it ask the ftp server for, filename wise? |
| 22:23.16 | [hC] | it may be asking for firmware.st, you could try renaming to that |
| 22:23.29 | [hC] | thats what it used to be called. |
| 22:24.00 | nny | [hC]: not sure what it asks for, using vsftpd.. need to figure out where the damn log file is.. not in /var/log atm |
| 22:24.37 | [hC] | maybe /var/log/xferlog |
| 22:24.58 | nny | ha |
| 22:25.06 | nny | "480i Cordless.st" |
| 22:25.18 | nny | wheres that guy who swears it is right out of the zip file |
| 22:25.54 | nny | ty on the log location.. forgot that one |
| 22:26.15 | nny | like why the holy hell would you not name it right in the zip file.. and then not document it |
| 22:26.21 | Alan_Hicks | vsftpd logs to /var/log/vsftpd by default, but it has different logging options and can output a wuftpd style log file at xferlog. |
| 22:26.38 | Alan_Hicks | pastebin your vsftpd.conf file. |
| 22:26.51 | nny | Alan_Hicks: yah was looking for vsftpd.conf, it is using xferlog, that is fine for my needs |
| 22:27.04 | nny | woohoo new firmware.. by by failure.st |
| 22:27.10 | nny | bye bye* |
| 22:27.32 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583970.dsl.bell.ca) |
| 22:27.38 | nny | er *looking for log/vfstpd* sorry |
| 22:27.45 | nny | too much crack today |
| 22:28.05 | Alan_Hicks | FWIW, the vsftpd.log file is muh more easily parsed by human eyes to me. |
| 22:28.24 | nny | gotcha |
| 22:28.34 | anujsingh | I am using ztdummy as i dont have any extra hardware device related to asterisk. how to check load balancing , i created 35 users , manually i can dial ,which playbacks a 25 seconds voice, dont have so many pc's . |
| 22:28.49 | Alan_Hicks | xferlog options exists because there are lots of tools out there that can automatically parse that style. |
| 22:29.05 | anujsingh | using astgui . |
| 22:29.22 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 22:29.36 | *** join/#asterisk mastermanx (n=hfh@cpe-76-175-136-37.socal.res.rr.com) |
| 22:29.43 | mastermanx | any one ever tried this |
| 22:29.48 | mastermanx | i seen the video of brain |
| 22:29.55 | mastermanx | krow |
| 22:30.00 | nny | anujsingh: astertest looks good for load testing |
| 22:30.02 | hohum | <PROTECTED> |
| 22:30.03 | mastermanx | i was thinking of doing this |
| 22:30.41 | hohum | if so how do I get it to stop |
| 22:31.00 | nny | [hC]: seems that the 'No User <sip:No%20User@192.168.100.15:5060>' is the issue |
| 22:31.31 | nny | [hC]: trying to find more info, at verbosity 5 ATM |
| 22:31.59 | anujsingh | what about clustering of asterisk ? |
| 22:32.42 | nny | i would think the PCI interrupt latency of multiple clusters may be an issue, but thats just me talking far out of my ass |
| 22:32.49 | anujsingh | i used drbd and heartbeat . dont know if any other better option . |
| 22:33.20 | mastermanx | hey i was thinking of doing the asterik home thing that brain krow been doing i was wondering is it worth doing? |
| 22:33.54 | anujsingh | scalability and high availability of asterisk., any suggestion for clustering ? |
| 22:34.04 | nny | meh pebkac |
| 22:34.16 | nny | user name vs. username in mac.cfg |
| 22:34.20 | nny | bran.. dying |
| 22:34.26 | mastermanx | hey i was thinking of doing the asterik home thing that brain krow been doing i was wondering is it worth doing? |
| 22:34.47 | nny | mastermanx: not wsure what you are referring to |
| 22:34.56 | mastermanx | the one brain krow |
| 22:34.57 | mastermanx | idd |
| 22:35.06 | mastermanx | brian |
| 22:35.20 | mastermanx | doing his own pbx system buying the tdm4000 |
| 22:35.20 | mastermanx | thing |
| 22:35.40 | nightrid3r | mastermanx url? |
| 22:35.40 | *** join/#asterisk gaba (n=gaba@c-67-188-214-186.hsd1.ca.comcast.net) |
| 22:35.54 | mastermanx | Hack Your Home Telephone with Asterisk |
| 22:35.55 | mastermanx | this |
| 22:36.17 | nny | [hC]: yay |
| 22:36.21 | nny | [hC]: works now |
| 22:36.42 | [hC] | cool. what was it? |
| 22:36.52 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 22:37.05 | *** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 22:37.05 | nny | [hC]: in mac.cfg had username vs user name |
| 22:37.10 | [hC] | ahhhh |
| 22:37.12 | [hC] | doh |
| 22:37.18 | [hC] | sorry i didnt catch that. |
| 22:37.21 | nny | lol hah no worries me neither |
| 22:37.27 | nny | [hC]: and firmware is up to date now |
| 22:41.40 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 22:43.22 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 22:46.33 | nny | trying to figure out why the speeddial for Voicemail adds a # to the end of the extension.. |
| 22:46.38 | nny | on the aastra |
| 22:47.48 | tripps | anyone see "error verifying config info" messages on 7970 phones? i can't seem to get the xml file to take |
| 22:48.04 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 22:48.34 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-171947493d7ca3c8) |
| 22:49.38 | *** join/#asterisk galeras (n=Martin@201.244.246.21) |
| 22:49.42 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 22:52.06 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584146.dsl.bell.ca) |
| 22:54.26 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 22:56.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:57.46 | *** join/#asterisk dr0ck (i=dr0ck@nat/digium/x-92855d54e2950c8b) |
| 22:58.36 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 23:01.12 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 23:06.00 | *** join/#asterisk techie (n=techie@76.214.18.225) |
| 23:10.27 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 23:11.27 | hohum | Where in my system am I supposed to put an AGI script!? |
| 23:11.48 | Qwell | /var/lib/asterisk/agi-bin/, by default |
| 23:12.06 | hohum | there, or how about in $PATH somewhere? WIll that work? |
| 23:18.40 | TJNII | Hmmm... My * server is behind a nat, the router of which has a static IP. I'd like to have a couple phones out in the wilds of the intartubes connect in. Is there a "send this IP" option that can be used on a per-phone basis? |
| 23:19.14 | JT | you need to portforward sip and rtp from the gw to asterisk and user externip= and localnet= |
| 23:19.30 | el_critter | Hi, do I have to apply a patch to Asterisk's Makefile in order to be able to use spandsp to send/receive fax? |
| 23:20.00 | TJNII | JT: I got the port forwarding done, and was looking for the options you gave me. Thanks! |
| 23:20.09 | JT | ~sipnat |
| 23:20.10 | 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 |
| 23:22.03 | TJNII | hmmmm.... that says I need STUN |
| 23:22.29 | JT | what does? |
| 23:22.50 | TJNII | The wiki on AAsterisk SIP NAT Solutions |
| 23:23.06 | TJNII | I'm looking at situation 4 |
| 23:23.45 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:25.23 | JT | forget the wiki |
| 23:25.26 | JT | first link. |
| 23:26.01 | TJNII | mmkay |
| 23:26.49 | TJNII | I know clients behind nats usually don't have too much trouble, and with the static IP I think I should be able to make my * server like its not behind NAT. |
| 23:28.20 | JT | yes people do it all the time |
| 23:32.35 | *** join/#asterisk fnordus (n=dnall@S01060050044bbf28.vc.shawcable.net) |
| 23:34.19 | TJNII | I think I've got it set up. Now I just need my friend to get home so I can go over there and do an off-site test... |
| 23:35.17 | *** part/#asterisk galeras (n=Martin@201.244.246.21) |
| 23:38.48 | *** join/#asterisk trippss (n=ss@72-20-150-196.dhcp.cmts1.phonoscope.net) |
| 23:39.52 | *** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com) |
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| 23:53.31 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 23:58.10 | trippss | so i'm having a bitch of a time getting cisco 7970 phones to provision/register. Apparently this is a very common situation. some people have cracked it. Anyone here? |
| 23:59.32 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |