00:00.30 | Dr-Linux | i'm using soft clients "sip" i wanna avail transfer call feature, is this feature comes with softphone or i need to do something in asterisk configuration? |
00:04.16 | FuriousGeorge | chanisavail else priority n+101 |
00:04.33 | FuriousGeorge | that what you mean? |
00:08.42 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
00:10.05 | *** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-11-9.w81-248.abo.wanadoo.fr) |
00:12.17 | justinu | Dr-Linux: eyebeam supports transfer |
00:12.28 | justinu | Dr-Linux: xlite doesn't, iirc |
00:13.05 | file[laptop] | marketing tactic cause it's free |
00:13.33 | justinu | that's what I figured |
00:13.46 | fugitivo | use # |
00:14.18 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
00:16.26 | harryvv | anyone ever test asterisk in single board computers before? |
00:16.44 | fugitivo | hm? |
00:17.20 | file[laptop] | just Google it. |
00:17.53 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
00:18.36 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
00:19.01 | harryvv | no not just google it. |
00:19.21 | file[laptop] | then I'll give you a name, Kristian Kielhofner |
00:19.22 | *** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-80.z142-153-67.customer.algx.net) |
00:19.24 | harryvv | asking for personal experaince is a bit different. and i really dont think its been done. |
00:19.58 | file[laptop] | it has been done. |
00:20.21 | file[laptop] | in fact, Google "asterisk single board computers" and look at the first thing that comes up :) |
00:20.31 | file[laptop] | it's an article on linuxdevices.com with Kristian aboot it |
00:21.23 | hardwire | crazy crazy |
00:21.25 | hardwire | little file |
00:21.32 | *** part/#asterisk CoffeeIV_ (n=CoffeeIV@mail.airlinksystems.com) |
00:21.56 | hardwire | the gumstix scares me |
00:22.01 | hardwire | but I think it would be cool for a phone platform |
00:22.17 | file[laptop] | hehe |
00:22.22 | Vco | nexcom has ok stuff |
00:22.50 | harryvv | if your talking about the gumstix that was not what i was refering to :) |
00:23.02 | hardwire | doing some filtering.. and creating a better percieved quality |
00:23.10 | hardwire | sample at 32khz |
00:23.20 | file[laptop] | harryvv: soekris boards are popular too |
00:23.24 | harryvv | thin client that fits in a 10baset wall box. |
00:23.27 | harryvv | http://www.windowsfordevices.com/news/NS3139003780.html |
00:23.29 | hardwire | harryvv: I use WRAP's |
00:23.30 | harryvv | od |
00:23.32 | harryvv | odd |
00:23.33 | harryvv | :) |
00:23.36 | harryvv | wraps? |
00:23.59 | hardwire | ok |
00:24.04 | Funar | anyone want to see what happens to a TNT when it meets a 12-gauge up close and personal? |
00:24.04 | hardwire | I am in love with that device now |
00:24.10 | hardwire | harryvv: you suck |
00:24.13 | hardwire | now I have to get one |
00:24.15 | harryvv | hehehe |
00:24.19 | file[laptop] | Funar: ...SURE! |
00:24.24 | Funar | lol |
00:24.57 | harryvv | kinda interesting..imagine the look on somones face if thay see peripheral connections in a wall outlet box. |
00:24.59 | MRH2 | drumkilla patch on bug 5630 works great thanks |
00:25.12 | Funar | i'm about ready to give up on this.. |
00:25.20 | drumkilla | MRH2: cool, just note in the bug :) |
00:26.27 | marcus2 | anyone else running asterisk on a wrt54gs-class device? |
00:26.33 | mog_work | i do |
00:26.49 | file[laptop] | I run Asterisk on my toaster. |
00:27.08 | *** join/#asterisk cia-dabar (i=bosas@82.135.166.46) |
00:27.13 | file[laptop] | which just happens to be a highly fast undercooled machine, naturally |
00:27.19 | mog_work | i still havent gotten that crappy linksys sip device to work |
00:27.23 | tzafrir_laptop | file[laptop], is your laptop your toaster? |
00:27.33 | drumkilla | mog_work: nub. |
00:27.39 | file[laptop] | tzafrir_laptop: nope |
00:27.50 | mog_work | its not my fault drumkilla |
00:27.57 | drumkilla | :-p |
00:28.08 | *** part/#asterisk cia-dabar (i=bosas@82.135.166.46) |
00:28.25 | mog_work | linksys admit it didnt work as they say it did |
00:28.42 | Vco | such as "One of our Customer Service employees has already tryed to telephonically reach you. As our employee did not manage to reach you, this email has been sent to your notice. |
00:28.43 | Vco | " |
00:28.47 | Vco | i mean....really now... |
00:28.58 | file[laptop] | telephonically? |
00:29.01 | file[laptop] | telepathically! |
00:29.01 | marcus2 | i have a linksys pap2 working with asterisk |
00:29.03 | marcus2 | it works great |
00:29.17 | marcus2 | took me like 5 minutes to unlock the vonage crap, and another 2 minutes to set it up |
00:29.29 | Funar | good to know.. i ordered a PAP2-NA to test with. |
00:29.37 | mog_work | but my pap2-na doesnt talk on the lan |
00:29.38 | Vco | i just bought a PAP2 that wasnt' locked... |
00:29.39 | mog_work | only on the wan |
00:29.45 | mog_work | thus it is pretty useless to me |
00:29.48 | file[laptop] | mog_work: that makes me sad |
00:29.55 | marcus2 | huh? |
00:29.56 | mog_work | i know |
00:30.03 | marcus2 | "doesn't talk on the lan" ? |
00:30.08 | mog_work | you cant make it talk to a local address marcus2 |
00:30.14 | mog_work | only to a far end sip provider |
00:30.22 | mog_work | so i cant have it talk to my asterisk box at home |
00:30.25 | Funar | hmm.. wrong netmask? |
00:30.27 | mog_work | which makes me sad |
00:30.34 | file[laptop] | he's using the router... |
00:30.35 | marcus2 | well, i will experiment with that tonight |
00:30.38 | drumkilla | so don't use it as a router |
00:30.38 | mog_work | i think its just hard locked to the wan port |
00:30.39 | Funar | oh, that one |
00:30.42 | file[laptop] | although you could make it loop back, but that would be interesting |
00:30.42 | *** join/#asterisk JunK-Y (n=junky@69.156.171.57) |
00:30.45 | mog_work | well then whats the point |
00:30.47 | mog_work | i have an iaxy |
00:30.49 | marcus2 | since i'm setting up asterisk at home, finally |
00:30.57 | marcus2 | uhm wait, you're using a pap2, or something else? |
00:30.57 | drumkilla | mog_work: to have another device? |
00:31.00 | file[laptop] | JunK-Y: you two left me all alone in here! 'tsk 'tsk |
00:31.29 | file[laptop] | Star Academie! |
00:31.49 | harryvv | Is there a way for a phone to dial my extention if its picked up automaticly? Was thinking of putting a phone in the outside of the house where if its picked up it will ring the extentions inside. |
00:31.50 | mog_work | why do i need three phones for my little apt. drumkilla |
00:32.02 | drumkilla | mog_work: fun! |
00:32.10 | Vco | sheer laziness? |
00:32.16 | Vco | lazyness |
00:32.17 | test34 | asterisk need root by default ? |
00:32.18 | Vco | whatever |
00:32.25 | drumkilla | r0000t! |
00:32.39 | marcus2 | no it doesn't |
00:32.41 | Funar | harryvv- i have a callbox at my office that does that. you push a button on it and it dials a number. |
00:32.47 | harryvv | i have two seperate extentions so my wife does not pick up my calls |
00:32.57 | harryvv | funar, where can i get one? |
00:33.16 | file[laptop] | Junky ScArEs me |
00:34.02 | *** join/#asterisk danzig (n=nicolas@130.226.169.135) |
00:34.03 | *** join/#asterisk bwzb (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net) |
00:34.28 | fugitivo | http://www.nslu2-linux.org/wiki/Unslung/Asterisk |
00:34.36 | bwzb | hi, any one had issue on installing TDM400P card? |
00:34.44 | bwzb | I got following error: |
00:34.46 | bwzb | zttool |
00:34.46 | bwzb | Unable to open /dev/zap/ctl: No such file or directory |
00:34.52 | Funar | i don't seem to have the bookmark here at home. i found it on google tho. try "outdoor call box" or something similar. |
00:34.57 | fugitivo | bwzb: ls -la /dev/zap/ctl |
00:35.09 | drumkilla | bwzb: /usr/src/zaptel/README.udev |
00:35.23 | bwzb | fugitivo: ls: /dev/zap/ctl: No such file or directory |
00:35.29 | Funar | very simple device.. just a speakerphone with a set of dip switches to assign a phone number to dial.. single button on the font grill face. |
00:35.31 | fugitivo | bwzb: that's your problem |
00:36.12 | bwzb | fugitivo: I have the same 2 cards installed in another machine which is working fine |
00:36.31 | fugitivo | bwzb: did you create udev rules correctly? |
00:36.33 | drumkilla | bwzb: README.udev |
00:37.02 | bwzb | drumkilla: k, I read it now and see what's the issue. |
00:37.10 | drumkilla | cool |
00:37.54 | fugitivo | anyone tried this? http://www.nslu2-linux.org/wiki/Unslung/Asterisk |
00:40.49 | marc324 | how do you update the aliases table in SER in realtime? |
00:41.22 | Vco | see if i had a buncha money...it would be fun to get a bunch of these.. http://www.vikingelectronics.com/products/view_product.php?pid=79 and put them around town |
00:41.36 | Vco | just put "DO NOT PUSH THIS BUTTON" on them... |
00:41.38 | *** join/#asterisk wolfson (n=ggggg@usr-kdh-208-6-58-26.beachlink.com) |
00:41.46 | Vco | but have it connect to a meetme if they do... |
00:44.14 | MRH2 | off topic - anyone know a good noncisco, cisco router / IOS site - (other than http://www.routergod.com/agentsmith/ of course) |
00:44.43 | masked | could someone explain to me why my 'iax2 friend's registration is restricted to 60 seconds? |
00:48.55 | masked | how can i change this? Nov 8 11:49:44 NOTICE[3894]: chan_iax2.c:5650 update_registry: Restricting registration for peer 'masked' to 60 seconds (requested 300) ??? |
00:52.22 | enemy | anyone here run e1`s or t1`s even in conjunction with asterisk? |
00:52.37 | mog_work | lots of people enemy |
00:52.39 | mog_work | whats up |
00:53.28 | marc324 | ne1 knows ser here? |
00:53.38 | enemy | mog_work: still having some issues managing to native bridge a call from one of the e1`s over to the other. Im thinking it could be some setup within the zaptel.conf or the zapata.conf. I`m running a dual e1 card with TE/NT. |
00:54.26 | Pegger | anyone have acess to cisco firmware? |
00:55.15 | mog_work | k what seems to be the problem |
00:57.09 | enemy | first of all, I get some errors, No D-channels available! Using Primary on channel anyway 28! ... I guess this is a bad error? |
00:57.39 | enemy | the zttool shows no errors :( |
00:58.53 | pooh_ | enemy: afaik those errors should not be a *real* problem |
00:59.23 | pooh_ | enemy: find out what gear the connected telco is using |
00:59.39 | enemy | mog_work: Im thinking about the clock/timing on the NT (which is provided by the dual E1 card) clock source should be taken from the TE. But Im not sure how to configure that.. maybe that will solve my issues |
01:01.19 | bwzb | drumkilla: after reading README.udev, and rebuilding zaptel, I got this message after modprobe wctdm: |
01:01.20 | bwzb | FATAL: Error inserting wctdm (/lib/modules/2.6.9-5.0.3.EL/extra/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
01:01.20 | bwzb | FATAL: Error running install command for wctdm |
01:01.37 | enemy | mog_work: also thinking about the framing, I`m getting some messages: wait_for_answer: Unable to forward frame and... |
01:01.37 | enemy | Nov 8 02:00:40 WARNING[6501]: chan_zap.c:7570 zt_pri_error: PRI: !! Got reject for frame 23, but we have nothing -- resetting! |
01:01.37 | enemy | Nov 8 02:00:40 WARNING[6501]: chan_zap.c:7570 zt_pri_error: PRI: !! Got S-frame while link down |
01:03.16 | bwzb | anyone know why I can not load "wctdm"? |
01:04.26 | FuriousGeorge | bwzb: you didnt make the modules? |
01:04.36 | FuriousGeorge | or |
01:04.39 | *** join/#asterisk wolfson (n=ggggg@usr-kdh-208-6-58-26.beachlink.com) |
01:04.39 | drumkilla | bwzb: yeah, and that's in the regular README, i think :) |
01:04.43 | FuriousGeorge | you didnt read readmen.UDEV |
01:04.46 | drumkilla | but it's something that is not enabled in your kernel |
01:04.50 | FuriousGeorge | readme* |
01:05.02 | enemy | mog_work: also got some of theese: Got a UA, but i'm in state 1 |
01:05.11 | bwzb | I thought I did after reading "README.udev", I rebuild the module |
01:05.21 | drumkilla | bwzb: this is something else |
01:05.21 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
01:05.41 | FuriousGeorge | speaking of wctdm: does it take the boostringer option when loading? |
01:06.06 | FuriousGeorge | ive heard the wxfxs does but i use ectdm and it pulls in everything i need |
01:06.29 | drumkilla | bwzb: you need CONFIG_CRC_CCITT enabled in your kernel |
01:06.49 | nesys | hi folks ... CCM -->* (multiple sip UA) --> ISP ... I need g729, that's pass-thru topology, or not? |
01:06.49 | drumkilla | bwzb: under 'Library Routines -> CRC-CCIT functions' in make menuconfig |
01:07.06 | bwzb | drumkilla: What do I do to enable CONFIG_CRC_CCITT in the kernel? |
01:07.16 | drumkilla | you'll have to rebuild a new kernel ... |
01:07.19 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
01:07.42 | bwzb | drumkilla: Godsh, do I need to rebuild a new kernel? |
01:08.29 | drumkilla | yes. |
01:10.03 | deezed | i have a couple of toll free DIDs from nufone. how do I assign the number of the DID called to a variable |
01:10.35 | *** join/#asterisk wolfson (n=ggggg@usr-kdh-208-6-58-26.beachlink.com) |
01:11.49 | drumkilla | deezed: it's already available in the variable ${EXTEN} |
01:12.25 | deezed | ok.. thats what i thought. for some reason asterisk types out |
01:12.31 | deezed | times* |
01:12.54 | deezed | i guess i need to check spelling and syntax |
01:13.04 | drumkilla | well yeah, that's a different problem :) |
01:14.24 | Qwell | drumkilla: any idea who I would talk to, to get a broken link on bugs.digium.com fixed? |
01:14.30 | srt | drumkilla: would u mind to have a look a look at the patch submitted as bug 5571 (set global variables via Manager API) - its the equivalent to the GetVar patch you commited some time ago. |
01:14.46 | drumkilla | Qwell: webmaster@digium.com |
01:15.07 | Qwell | drumkilla: I emailed bugs@ - think it'll get to the right place? |
01:15.16 | drumkilla | srt: I'll put it on my list, but it might not happen before the release :( |
01:15.40 | drumkilla | Qwell: bugs@digium.com? I doubt that goes anywhere ... |
01:15.49 | drumkilla | webmaster will create an RT ticket for them |
01:15.53 | drumkilla | so email it there |
01:15.53 | deezed | odd.. syntax looks right, but asterisk times out -> exten => 8662942000,2,NOOP(EXTEN: ${EXTEN}) |
01:16.06 | *** join/#asterisk wolfson` (n=ggggg@usr-kdh-208-6-58-26.beachlink.com) |
01:16.13 | *** join/#asterisk file[laptop] (n=jcolp@69.156.171.57) |
01:16.16 | srt | drumkilla: its very short and provides some consistency :) |
01:16.19 | Qwell | drumkilla: k |
01:16.21 | drumkilla | deezed: do you have a priority 1 above that? |
01:16.26 | drumkilla | srt: I agree |
01:16.28 | deezed | yes Answer() |
01:16.34 | Qwell | eww, file[laptop] ! |
01:16.35 | file[laptop] | french television is great |
01:16.43 | deezed | works fine.. when I skip the noop |
01:16.45 | file[laptop] | it's so corny watching soap operas in another language |
01:16.49 | drumkilla | srt: I just have to write a speech for school first ... |
01:17.08 | srt | drumkilla: hehe ok - that has priority :) |
01:17.18 | file[laptop] | no matter language, it's all the same |
01:17.19 | bweschke | drumkilla: your not done yet?!? :) |
01:17.20 | file[laptop] | er no matter the language |
01:17.23 | file[laptop] | drumkilla: Junky says hi |
01:17.26 | file[laptop] | fine - hello |
01:17.42 | drumkilla | bweschke: NO! I can't find a good article!!! |
01:17.48 | drumkilla | file[laptop]: Russell says hi |
01:17.51 | bweschke | what's the topic? |
01:18.15 | file[laptop] | he said hi to drumkilla, not Russell :P |
01:18.32 | file[laptop] | uh oh it's back on! |
01:18.47 | drumkilla | bweschke: it has to be an article from a technical magazine or journal about transistors or diodes, basically. Oh, and published in the last few months ... |
01:19.02 | drumkilla | bweschke: I'm finding plenty of press releases :) |
01:19.10 | deezed | there isn't anything wrong with the syntax is there? |
01:19.17 | file[laptop] | drumkilla: how was your day? |
01:19.23 | drumkilla | file[laptop]: just dandy |
01:19.38 | drumkilla | I'm leaving my apartment now, so I can get this done, heh |
01:19.40 | drumkilla | cya guys ... |
01:19.44 | file[laptop] | ttyl |
01:20.31 | bweschke | later |
01:21.09 | Dr-Linux | whats diffenece between Dial(SIP/222,20,t) and Dial(SIP/222,20|t) ? |
01:21.25 | Dr-Linux | i mean difference between , and | ? |
01:21.33 | Qwell | Dr-Linux: not a alot |
01:21.35 | Qwell | a lot* |
01:21.55 | Qwell | most (all?) options parsing stuff can use either |
01:22.42 | Dr-Linux | Qwell: it always confuse me when i read guides |
01:22.48 | *** part/#asterisk docelm0 (n=docelmo@static-71-251-95-2.tampfl.fios.verizon.net) |
01:23.21 | *** join/#asterisk _Sam-- (n=sam@phone2.kneedraggers.com) |
01:23.46 | _Sam-- | hi, how would i force asterisk to re-register with my IAX provider without resarting it? |
01:24.54 | file[laptop] | how creepy is it that despite not being able to speak french, I can apply past knowledge and common stuff to be able to get what's going on in this soap opera? |
01:30.54 | *** join/#asterisk wolfson (n=ggggg@usr-kdh-208-6-58-26.beachlink.com) |
01:32.42 | *** join/#asterisk wolfson (n=ggggg@usr-kdh-208-6-58-26.beachlink.com) |
01:33.42 | enemy | after some researching, I think I got my E1`s up running. But still, I can't dial out from my sip client using the e1 trunk. Using debugging, I now get > M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] .... |
01:34.54 | delmar | i have a little question.... lets say you have contexts.. for example [local] and [local-restricted], they both have access to a context [local-free] but [local-restricted] doesn't have access to other contextx that cost money.... what happens if the dialplan of the [local-free] context has a timeout that jumps to say.. your chargable VoIP gateway next because the 1st path wasnt available... will the [local-restricted] context be denied access to that pa |
01:35.15 | delmar | or will ie be allowd because the timeout jumped over and allowed it to happen? |
01:35.34 | delmar | anyone know what im on about? lol |
01:36.01 | file[laptop] | since you used a goto, it doesn't matter if it's reachable in the context or not |
01:36.11 | file[laptop] | you're literally going to the context specified |
01:36.31 | delmar | file[laptop] so the answer is.. GoTo overrides context includes. |
01:36.43 | file[laptop] | correct |
01:36.49 | delmar | ok cheers |
01:37.13 | file[laptop] | if you use Goto without a context, it'll use the current context... |
01:37.20 | file[laptop] | but that's aboot it |
01:37.24 | delmar | yep |
01:37.53 | delmar | i think i have a plan. |
01:37.54 | delmar | :P |
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01:47.37 | mikemee | newbie q: my pots line into a TDM400P rings once (almost twice) before asterisk starts ringing the extensions. Can I make this more immediate? |
01:48.22 | |Vulutre| | mikemee: thats a phone company deal |
01:49.52 | mikemee | thx, I have a phone on the pots line as well as asterisk, and the phone rings. I turned callerid off in the zaptel config which dropped it from 2 rings to 1, but still one. Surely if the phone at my end rings, Asterisk can see that sooner? |
01:49.59 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
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02:11.20 | justinu | googletalk should allow you to dial sip uris |
02:11.23 | harryvv | mike, you cannot...because that first ring sends the callerid info then it will show up on the phone when it rings once. |
02:11.45 | Math` | justinu: googletalk doesnt use sip |
02:12.34 | Math` | (tho support is planned) |
02:12.35 | mikemee | harryvv, thx!: I don't have/need callerid, so can I reduce the delay? |
02:13.00 | justinu | math: supposedly they do on the backend or something |
02:13.12 | justinu | if you look at the googletalk packets, you can see SDP |
02:15.51 | harryvv | mikemee, why? |
02:15.54 | harryvv | anyway |
02:16.16 | harryvv | anyone recomend a ups that will signal linux to shutdown if there is a power outage? |
02:17.00 | mikemee | harryv: I don't pay SBC for callerid on my POTS line, so I'd rather avoid the extra rings when people call me |
02:19.27 | Nivex | harryvv: I swear by the APC line. |
02:21.37 | *** join/#asterisk sumonish (n=God@203.12.249.168) |
02:21.41 | sumonish | hi all |
02:21.48 | harryvv | nivex okay what small unit do you recomend for me? might include it inside a protective cover along site the pbx |
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02:22.07 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
02:22.50 | sumonish | the company i work for has just installed an Asterisk SVR for Voip work i wish to set it up so as a the phone rings just for reciption but can be remotly answered by anyone how can i do this? |
02:22.59 | Nivex | harryvv: that depends on how much power draw the PBX has, how long you want it to run in the event of a power outage, what your physical constraints are, etc. Gonna have to do some research. |
02:23.21 | shmaltz | sumonish, look at app_pickup |
02:23.30 | shmaltz | sumonish, or you could try features.conf |
02:23.33 | harryvv | enough to allow it to power off the pbx fafely |
02:23.36 | harryvv | safely |
02:23.53 | javo | when i call one SIP phone from another, it says the phone i am calling is busy. It is the same if I call the other way around. I can call out just fine, but my phone will not ring. it goes straight to voice mail. I have restarted asterisk and the phone, with no luck. does anyone know what a common cause of this would be? |
02:23.59 | harryvv | nivex im not concerned about keeping it up for x amount of time. |
02:24.08 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
02:24.12 | sumonish | thanks for the help Shmaltz |
02:24.30 | Math` | what I wanted to do is get a via embedded board (7W total power @ 1ghz) and run * on it with a ups.... it can stay up hours on battery |
02:25.09 | harryvv | math nice idea |
02:25.11 | Math` | javo: check if the phone is registered |
02:25.36 | Math` | javo: sip show users |
02:26.23 | javo | it is |
02:26.34 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
02:28.53 | masked | ~mailinglist |
02:28.54 | jbot | i guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
02:28.57 | javo | when someone calls, it tells them it is busy and sends them to voicemail |
02:32.06 | harryvv | how can asterisk be setup to dial 1800 numbers into the states even though the number does not service this area of canada? My asterisk is seeing 1800 and dialing out zap. But want to go into my provider into the states |
02:32.24 | harryvv | half the 1800 calls are in the states and some dont work. |
02:32.42 | Qwell | harryvv: call dial, then check the DIALSTATUS variable? |
02:32.50 | justinu | set up a dial string for _1800 |
02:33.02 | harryvv | I dont want to make a custom dial plan for each and every 1800 I stumble across and configure it. to bad there was a way to force asterik to dial iax |
02:33.16 | harryvv | just...1800 calls here in canada! |
02:33.19 | justinu | you can make it dial whatever service you want |
02:33.34 | harryvv | justinu you dont understand. |
02:33.38 | Qwell | harryvv: What happens when you dial a US 800 over zap? |
02:33.53 | harryvv | it dails local 1800 |
02:33.56 | harryvv | but |
02:33.56 | justinu | you want it to use a SIP provider for certain 800 numbers, right? |
02:34.06 | harryvv | some us companies dont have 1800 service up here. |
02:34.11 | harryvv | yes |
02:34.11 | Qwell | hmm |
02:34.14 | harryvv | into canada. |
02:34.15 | harryvv | err |
02:34.19 | harryvv | into the states |
02:34.25 | Qwell | harryvv: I bet you could write a little agi to hit like infousa or something |
02:34.32 | harryvv | i dont know agi |
02:34.37 | justinu | so you want to route all toll frees that terminate into the USA out via a specific SIP provider? |
02:34.38 | Qwell | shouldn't be hard |
02:34.46 | harryvv | nooooo |
02:34.48 | justinu | and the others out to your local LEC |
02:35.30 | harryvv | sorry yes, All toll free 1800 numbers those companies that dont have there service up here. |
02:35.51 | Qwell | so, wait... |
02:35.56 | justinu | lol |
02:36.01 | Qwell | can the same number exist in both canada and us? |
02:36.06 | justinu | yeah |
02:36.08 | Qwell | if so, that'd be fucking retarded |
02:36.13 | justinu | we have toll frees that work in canada and the USA |
02:36.30 | Qwell | okay, let me rephrase |
02:36.32 | justinu | but they always terminate into the USA |
02:36.38 | Qwell | can two DIFFERENT companies have the same 800 number? |
02:36.42 | Qwell | one in canada, one in the US? |
02:36.44 | justinu | oh |
02:36.47 | harryvv | qwell, that depends on what the company wants. I often want to call 1800 to a company in the states but it dials out zap. If i found a way to dial out sip then it would work. |
02:36.48 | justinu | no, it's still NANP |
02:36.56 | justinu | part of the same number plan |
02:37.18 | jake1932 | just prompt for a code to dial into SIP |
02:37.42 | harryvv | hehe |
02:37.49 | Qwell | nah, it should be automatic |
02:37.56 | harryvv | you know, i forgot about my calling card feature i programed into this. |
02:37.56 | jake1932 | or call both using & |
02:38.03 | harryvv | that would work then. |
02:38.08 | Qwell | jake1932: SIP will probably always answer |
02:38.12 | justinu | the zyxel p2000 wifi phone is stupid |
02:38.16 | justinu | no dns srv support |
02:38.20 | harryvv | i dial my extention put in pass then it should work. |
02:38.59 | jake1932 | and get a SIP provider that won't work in Canada :) |
02:39.12 | justinu | you need a password to dial toll frees? wow :) |
02:39.24 | justinu | oh, nm |
02:39.29 | justinu | didn't see that about calling cards |
02:39.36 | harryvv | suck..it did not work |
02:40.18 | harryvv | just, well i created this if I am at a goverment office and want to call into the states. |
02:40.41 | harryvv | i call my number then enter in extention and pass and its charged to my account. Like that of a calling card |
02:40.56 | justinu | cool |
02:41.23 | harryvv | so call home on cellular, call into states at 2 cents per min plus local cell rates. |
02:41.35 | Qwell | there is probably some simple site that can tell you if a tollfree number is in canada... call that from an agi, if it's in canada, dial out zap, otherwise, dial out sip |
02:41.46 | justinu | get a flat rate US sip account :) |
02:42.08 | *** join/#asterisk docelm0 (n=docelmo@static-71-251-95-2.tampfl.fios.verizon.net) |
02:42.17 | justinu | qwell: i think that would be an SMS/800 db query |
02:42.35 | harryvv | brb |
02:42.39 | justinu | if there's a site that'll tell you that, i'd love to see it |
02:42.41 | Qwell | speaking of which... |
02:42.45 | Qwell | where did file go? |
02:42.54 | jake1932 | great question! |
02:43.34 | jake1932 | it's too early for bed |
02:43.37 | Qwell | did somebody accidently rm him again? |
02:43.43 | Qwell | s/accidently// |
02:43.47 | jake1932 | hehe |
02:44.40 | jake1932 | qwell - do you work for asterlink? |
02:44.40 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
02:44.44 | Qwell | no |
02:44.55 | jake1932 | but you use them, right? |
02:45.00 | Qwell | I do |
02:45.17 | jake1932 | you seeig weirdness with toll-free outbound? |
02:45.20 | jake1932 | seeing |
02:45.43 | Qwell | dunno, lets see |
02:46.25 | Qwell | odd, seems they don't like the codec I'm using...or something |
02:46.52 | Qwell | woah, heh |
02:46.58 | Qwell | yeah, weirdness indeed |
02:47.01 | jake1932 | :) |
02:47.09 | Qwell | "If you remain on the line, you will be charged for the international portion of this call." |
02:47.13 | jake1932 | right |
02:50.06 | justinu | nice! |
02:51.41 | justinu | is there any way to do old fashioned call screening with a sip phone? |
02:51.54 | Qwell | look at the callerid, and don't answer if you don't want to talk? |
02:52.00 | justinu | have someone leave a message, but allow you to answer the phone(s) while you hear the ICM |
02:52.12 | justinu | like an analog answering machine |
02:52.32 | Qwell | might be able to something creative with monitor or something |
02:52.41 | justinu | hmm |
02:53.02 | Qwell | have it fork the call to an autoanswer line on the sip phone, and mute the channel or something? dunno |
02:53.14 | justinu | maybe drop both phones into a meetme |
02:53.20 | Qwell | yeah |
02:53.47 | Qwell | or you could write app_callscreening, heh |
02:53.58 | justinu | but is there any phone that doesn't actually answer the call with a 200 ok, but still will play the audio on the speaker? like early media works with 183 progress |
02:54.09 | Qwell | got me |
02:54.19 | justinu | cuz i'd like an answer on ANY of the phones ringing to interrupt the message and bridge the call |
02:54.33 | justinu | but I guess all the phones would still be ringing |
02:54.51 | justinu | can't replicate the old analog scenario very well with sip |
02:56.11 | *** part/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
03:02.35 | Pegger | anyone have acess to cisco firmware? |
03:02.50 | Qwell | Pegger: people who pay for the access do |
03:03.12 | Qwell | (that's my subtle way of saying to pay cisco what you owe them, if you want the firmware) |
03:03.31 | Pegger | Qwell well it only makes sense that if I buy a 200 dollar phone that I should have acess to the phones firmware |
03:03.42 | Qwell | Pegger: you'd think so, but no |
03:03.51 | Pegger | Qwell how do i owe them? |
03:04.09 | *** join/#asterisk copantl (n=galel@63.245.93.138) |
03:04.17 | copantl | hello |
03:04.26 | Qwell | Pegger: because you want the firmware |
03:04.26 | Pegger | yo |
03:04.50 | copantl | i have a problem with a sip peer conexion |
03:04.58 | copantl | can some one help me?P |
03:05.24 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
03:05.35 | Pegger | what ever just keep on thinking cisco twisted tatics are cool |
03:05.46 | Qwell | Pegger: Who said I liked what they do? |
03:05.53 | copantl | i need to route a calls from a peer to my asterisk |
03:06.03 | Qwell | fact of the matter is, stealing the firmware would be copyright infringement, plain and simple. |
03:06.35 | copantl | but when i place a call , goes to a extension, not to a DID... any Idea? |
03:06.37 | justinu | i dunno if he thinks it's cool |
03:06.41 | justinu | he's just saying it's the law |
03:06.43 | Pegger | but it is included withthe phone |
03:06.49 | Pegger | am i stealing that |
03:06.54 | justinu | the version that was on it when you bought it was included |
03:06.56 | justinu | not the latest |
03:07.11 | copantl | some one from digium ? |
03:08.00 | Pegger | so has anyone set up a phone with sccp before? |
03:08.09 | Qwell | Pegger: yep, sccp is fun |
03:08.58 | justinu | seems kinda cool |
03:09.08 | justinu | i like that event driven model |
03:09.16 | harryvv | qwell get that link? |
03:09.18 | [TK]D-Fender | copantl : pastebin you extensions.conf and I'll take a look |
03:09.20 | Qwell | yep, kinda cool |
03:09.35 | harryvv | qwell, that would make a good wall mount pbx style of case. |
03:09.41 | Qwell | yeah... |
03:09.43 | Pegger | so on the topic of sccp how does it compare to sip when connecting to * |
03:09.52 | Qwell | Pegger: personally, I'm really digging sccp |
03:10.04 | Qwell | if not only because of system messages, heh |
03:10.10 | justinu | what's that? |
03:10.25 | Qwell | sccp system message "The boss is gone. Time to party!" 0 |
03:10.26 | Pegger | Qwell so what kind of feature diffrences are there |
03:10.29 | copantl | i just wanna configure a media gateway |
03:10.32 | Qwell | sends the shit to all the sccp phones on the system |
03:10.35 | justinu | heh |
03:10.42 | justinu | polycoms have IM |
03:10.46 | justinu | so i guess you could do that |
03:10.54 | Qwell | but...that 0...was a timeout |
03:11.03 | Qwell | so, I can make a message display for only 30 seconds |
03:11.07 | justinu | ah |
03:11.14 | justinu | that's kinda neat |
03:11.18 | Qwell | I'm in the process of hacking it up to be per device |
03:11.26 | Qwell | that'll be slick... |
03:11.27 | [TK]D-Fender | Cisco = overpriced Polycom class :) |
03:11.43 | [TK]D-Fender | Poltcom <3 |
03:11.52 | [TK]D-Fender | polycom even ;) |
03:11.55 | Pegger | Qwell what other neat/usefull features can it do? how well does it handle nat |
03:12.01 | Qwell | Pegger: got me... |
03:12.12 | copantl | Joe? |
03:12.12 | Qwell | it's rtp, so I imagine it isn't difficult |
03:12.19 | riksta | nah cisco are the rolls royce man |
03:12.28 | Pegger | Qwell so message display is the only benifit of sccp |
03:12.34 | justinu | lol |
03:12.34 | Qwell | Pegger: no, certainly not |
03:12.46 | justinu | no, sccp sends events for onhook offhook, dtmf presses, etc. |
03:12.53 | justinu | it's a way less complex protocol |
03:12.55 | Pegger | then what else might there be |
03:12.56 | Qwell | yeah, everything is an event |
03:13.07 | justinu | sip is all crazy and tries to do way too much |
03:13.15 | copantl | [TK]D-Fender: i just wanna setup my asterisk as a media gateway |
03:13.36 | Pegger | well i might have to leave sccp on the cisco phone that I oredred last week, hopefully it arrives soon |
03:13.59 | *** join/#asterisk file[laptop] (n=jcolp@69.156.171.57) |
03:14.09 | Qwell | Pegger: or you could get a cco license (which still wouldn't give you a legal right to use it...but, you'll be able to download it) |
03:14.15 | Qwell | file[laptop]: that really you? |
03:14.37 | file[laptop] | yup |
03:14.37 | Pegger | qwell how much are cco licenses? |
03:14.41 | [TK]D-Fender | copantl : what do you want on both ends? |
03:14.49 | justinu | like 7 bucks per phone |
03:15.05 | Pegger | you have to pay per phone? |
03:15.05 | copantl | [TK]D-Fender: asterisk |
03:15.09 | Qwell | file[laptop]: remember that script you asked me for a week or so ago? cidname lookup |
03:15.11 | file[laptop] | oh noes it's [TK]D-Fender |
03:15.21 | [TK]D-Fender | copantl : as in Asterisk <-> Asterisk? |
03:15.27 | Pegger | well i oonly have one phone so whatever |
03:15.30 | justinu | i dunno, i just heard that in passing |
03:15.33 | justinu | i don't own ciscos |
03:15.39 | justinu | or work with them |
03:15.45 | linagee | justinu: what's this now? AT&T owns cisco? |
03:15.57 | justinu | i have polycom, snom, sipura, grandstream, zyxel |
03:16.02 | copantl | is all the calls from asterisk1 routes to a did from asterisk2 |
03:16.03 | justinu | linagee: what are you on about now!? |
03:16.15 | linagee | justinu: oh, i thought you were starting conspiracy theories. :-) |
03:16.18 | justinu | lol |
03:17.10 | [TK]D-Fender | copantl : oh.. sorry, never did * bridging before. |
03:17.24 | copantl | ok |
03:17.28 | copantl | some one? |
03:18.09 | FuriousGeorge | copantl: didnt ake sense to me |
03:18.11 | Pegger | anyone here do asterisk failover, like one asterisk dies for some reason so a second asterisk box takes over |
03:18.18 | FuriousGeorge | \ake=make |
03:18.24 | blitz[laptop] | not yet |
03:18.25 | copantl | make sanse? |
03:18.32 | Qwell | Pegger: there is something on the wiki about that |
03:18.34 | FuriousGeorge | sense |
03:18.40 | blitz[laptop] | Pegger: there was a thread about it recently on asterisk-users |
03:18.44 | copantl | i just wanna create a media gateway |
03:18.51 | justinu | blitz: link? |
03:18.57 | linagee | Pegger: why would you want something like that? nobody likes it better than when your asterisk box barfs up and nobody can make a phone call! |
03:18.58 | blitz[laptop] | justinu: no idea :) |
03:19.04 | justinu | bastage :P |
03:19.12 | FuriousGeorge | copantl: i dunno |
03:19.21 | blitz[laptop] | justinu: basically I think people used a "heartbeat" type of method... |
03:19.23 | linagee | high availability systems save you from hardware problems, not software problems. ;-) |
03:19.48 | copantl | asterisk1-----------sip peer--asterisk2-did------> |
03:19.50 | blitz[laptop] | if a server dies... all calls are going to die on it either way |
03:19.55 | Pegger | why cant they save you from software problems |
03:20.02 | justinu | ibm was doing work to solve that |
03:20.18 | justinu | they did a talk at astricon |
03:20.19 | linagee | Pegger: well i suppose you could fail over to an older version, but i've never heard of that being done. |
03:20.25 | Pegger | blitz[laptop] no I do not see why it is not possible to set up a heat beat for the two servers |
03:20.34 | *** join/#asterisk jontow (i=jontow@bsd.adminforrent.com) |
03:20.38 | blitz[laptop] | Pegger: I never said you couldn't |
03:20.42 | linagee | Pegger: it would be a MAJOR annoyance to have two versions of the config |
03:20.53 | justinu | sync them with rsync |
03:20.57 | Qwell | active calls will be distrupted though |
03:20.59 | linagee | justinu: naw. |
03:21.05 | linagee | justinu: just use a SAN. ;-) |
03:21.10 | Qwell | regardless of whether you use a heartbeat type deal or not |
03:21.12 | justinu | i'm too poor for a san |
03:21.13 | blitz[laptop] | Qwell: yep -- thats what I said, but no one ever pays attention to me :) |
03:21.13 | justinu | :( |
03:21.15 | linagee | justinu: put an EMC box there, go HA the full 9 yards. ;-) |
03:21.22 | Qwell | blitz[laptop]: was helping reiterate the point, heh |
03:21.30 | blitz[laptop] | box dies, all active channels on that box dies too -- regardless |
03:21.32 | blitz[laptop] | Qwell: hehehe :) |
03:21.51 | blitz[laptop] | Qwell: I reiterated your iterated point |
03:21.51 | linagee | blitz[laptop]: not necessarily.... hrm |
03:22.02 | blitz[laptop] | linagee: let me know when you figure it out |
03:22.09 | Qwell | blitz[laptop]: and by doing that, it helped clear everything up... |
03:22.13 | blitz[laptop] | Qwell: :) |
03:22.14 | justinu | they'll only die if the box was proxying rtp |
03:22.15 | linagee | blitz[laptop]: if you were able to switch over fast enough, and you had the TCP/IP stack intergrated into the HA system... |
03:22.16 | blitz[laptop] | Qwell: apparently not :) |
03:22.22 | justinu | if you reinvite to the pstn gateway, that's not a problem |
03:22.25 | Qwell | reiterating the reiterated point of the previous iteration |
03:22.30 | linagee | blitz[laptop]: switch over faster than 2 minutes (TCP timeout) or xx minutes (application timeout) |
03:22.31 | justinu | provided the pstn gateway was not the box that just went down |
03:22.47 | blitz[laptop] | linagee: I think the caller would hang up before 2 minutes |
03:22.52 | linagee | that's it. fail over to a PSTN line. lo. |
03:22.56 | Qwell | linagee: and your dialplan would crap itself |
03:22.59 | blitz[laptop] | linagee: and we don't use TCP in Asterisk |
03:23.01 | linagee | blitz[laptop]: lol. like i said, application timeout |
03:23.10 | Qwell | there would be literally no way of knowing where each call was in the dialplan. |
03:23.15 | linagee | blitz[laptop]: er, UDP... so it's all application timeout. hehehe |
03:23.16 | blitz[laptop] | linagee: tcp is bad for realtime applications |
03:23.19 | Qwell | They went over some of this stuff at astricon... |
03:23.23 | linagee | blitz[laptop]: exactly. |
03:23.27 | justinu | tcp is anceint |
03:23.34 | blitz[laptop] | justinu: uhhh... |
03:23.48 | linagee | justinu: will TCP die with IPv6? |
03:23.54 | Qwell | no... |
03:24.04 | blitz[laptop] | TCP is layer 4 -- IP is layer 3 |
03:24.14 | blitz[laptop] | IP encapsulates TCP |
03:24.19 | Math` | tho there is a tcp6 afaik |
03:24.24 | justinu | yes |
03:24.24 | linagee | blitz[laptop]: yes, but maybe they have solved it with IPv6? i hear there is encryption. that is not layer 3.... |
03:24.36 | Qwell | encryption? |
03:24.42 | linagee | Qwell: or VPN or something |
03:24.50 | linagee | Qwell: some new routing method in ipv6 |
03:25.00 | linagee | Qwell: unless i was speaking to a complete liar! (possible) |
03:25.10 | linagee | or was that on the internet... |
03:25.22 | linagee | ipsec? i think that was it. |
03:25.26 | justinu | lol |
03:25.28 | Qwell | that'd probably be a layer 4+ thing, heh |
03:25.28 | linagee | i think ipv6 has ipsec built in? |
03:25.31 | Math` | ipsec exists on v4 too |
03:25.45 | blitz[laptop] | we should just start using SCTP |
03:25.49 | linagee | Math`: yes, but only if you have it. |
03:26.05 | linagee | Math`: i think ipsec might be REQUIRED for ipv6.... |
03:26.14 | justinu | STCP is a good idea |
03:26.17 | Math` | blitz[laptop]: it is not |
03:26.21 | linagee | (required as in, you're breaking the standard by not having it) |
03:26.23 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
03:26.34 | blitz[laptop] | Math`: what is not? |
03:26.44 | Pegger | linagee not required |
03:27.09 | linagee | Pegger: moment... |
03:27.20 | Math` | ipsec aint required for v6 |
03:27.27 | riksta | it certainly is not |
03:27.50 | blitz[laptop] | Math`: I never said it was |
03:27.57 | blitz[laptop] | Math`: notice I laughed when linagee said that |
03:28.10 | linagee | blitz[laptop]: it says so on an ieee page... argh |
03:28.11 | Math` | oh sorry it was linagee |
03:28.15 | linagee | "buy this article now" |
03:28.20 | justinu | lol |
03:28.29 | blitz[laptop] | I love the IEEE |
03:28.39 | blitz[laptop] | </sarcasm> |
03:29.04 | Math` | I like people who establish standards... make people pay for the spec, then complain their standard isnt repected |
03:29.06 | riksta | i love PIE |
03:29.07 | linagee | "It is also expected that extended features of IPv6 will be used when IPv6 gets commonly used. Therefore, Phase-2 will test extended features of IPv6 such as IPsec and MIPv6, in addition to IPv6 core protocol. |
03:29.07 | linagee | " |
03:29.08 | Math` | respected, that is |
03:29.14 | linagee | ok, everyone line up for punches |
03:29.20 | linagee | or you can pay the fee |
03:29.38 | riksta | linagee: that doesn't mean it;s required |
03:29.41 | linagee | you're all fools. see above |
03:29.44 | copantl | i just need a asterisk like a pstn gateway for other asterisk? ..... can be posible? |
03:29.46 | blitz[laptop] | screw IPv6, I'm starting my own layer 3 protocol |
03:29.55 | linagee | riksta: required as in you're breaking the standard if not. see above where i said that right after. |
03:29.59 | riksta | we're hardly even in phase 1 never mind pahse 2 |
03:30.03 | justinu | linagee: bahahaaha |
03:30.06 | linagee | riksta: it might even be compiled into the stack. not sure. |
03:30.49 | blitz[laptop] | riksta: I have a feeling even if all of asia adopts IPv6, and even Europe, North America will continue to use IPv4 for a while, simply using IPv4<-->IPv6 gateways |
03:31.02 | riksta | yep |
03:31.09 | linagee | it will be nice though once it's required. |
03:31.09 | linagee | this means an encrypted connection to anyone using IPv6 |
03:31.11 | linagee | (soon everyone? lol) |
03:31.16 | riksta | there's gonna be mega bux in those gateways |
03:31.23 | justinu | lol |
03:31.49 | linagee | of course, do you think required ipsec in internet communications will mean less big brother? ROFL |
03:31.55 | linagee | "sense of security" ;-) |
03:32.16 | riksta | are you blabbering nonsense, or am I just tired |
03:32.23 | justinu | no, he's always like that |
03:32.31 | justinu | but he eventually makes sense |
03:32.45 | linagee | justinu: exactly. as soon as i found that article sniplet |
03:32.57 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
03:33.04 | justinu | dinner time |
03:33.07 | linagee | yes! |
03:33.11 | linagee | indeed! taco bell! |
03:33.14 | linagee | who's buying? :-) |
03:33.41 | riksta | ive never had a taco bell and i dont think i really ever want one |
03:34.01 | linagee | riksta: lol. probably not. |
03:34.10 | riksta | what is in them |
03:34.22 | linagee | riksta: fast food mexican. |
03:34.22 | riksta | i know it's a chain...but what stuff do they do |
03:34.29 | riksta | fajitas n shit? |
03:34.39 | linagee | not really. tacos n shit. (TACO bell. ;-) ) |
03:34.45 | riksta | yeah i gathered that |
03:34.53 | riksta | sounds unhealthy :P |
03:34.57 | linagee | and burritos too. although if you want a good burrito there are other places. |
03:35.03 | linagee | riksta: very! :o) |
03:35.29 | linagee | riksta: or i should say, isn't all fast food? heheh |
03:35.33 | riksta | true |
03:35.43 | riksta | you yanks :P |
03:36.02 | *** join/#asterisk wolfson (n=ggggg@usr-kdh-208-6-58-26.beachlink.com) |
03:40.21 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
03:40.37 | Brijn | Good evening! |
03:41.41 | Brijn | After three weeks of telling my SIP provider they have a problem, they finally sent me their part of the config. Took me a minute to discover a type in their config :-((( |
03:42.25 | Brijn | Now I at least see the call coming my way :) But unfortunately the problem is now on my end... i think at least |
03:43.41 | *** join/#asterisk bsdz0r (n=bsd@24-247-92-53.dhcp.bycy.mi.charter.com) |
03:45.06 | Brijn | I see some stuff with sip debug enabled, but I don't understand enough of it to pinpoint the problem.. If someone could have a look.... http://pastebin.ca/27948 |
03:45.10 | iCEBrkr | md |
03:45.17 | iCEBrkr | grr |
03:45.22 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
03:45.50 | linagee | magic |
03:45.56 | linagee | asterisk is magic |
03:46.22 | *** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net) |
03:46.24 | implicit | hi |
03:50.45 | harryvv | I wonder how tough it would be to make a embeded interface work with asterisk |
03:50.56 | Brijn | So close but so far, wtf is happening :( |
03:51.04 | *** join/#asterisk unabonger (i=Bhongwan@lunacity.themoon.org) |
03:51.13 | unabonger | hello |
03:51.47 | unabonger | anybody in here able to tell me why my music on hold works when I call an extension from one source, but not another? |
03:51.53 | *** join/#asterisk bmg505 (n=leon@rndf-146-59-144.telkomadsl.co.za) |
03:54.04 | harryvv | unabonger, do a cli on it. |
03:54.08 | Brijn | unabonger: not to many of the wizards around i'm afraid :) You get audio if you accept the call |
03:56.46 | *** join/#asterisk kr5 (n=kris@tor/session/x-b6d2326f3c7ef0b8) |
03:57.10 | unabonger | jachin*CLI> |
03:57.10 | unabonger | Verbosity is at least 5 |
03:57.10 | unabonger | Nov 7 20:05:39 NOTICE[426]: chan_sip.c:7787 handle_request: Registration from 'Zhonka <sip:zhonka@kush.inwa.net>' failed for '66.228.198.186' |
03:57.10 | unabonger | <PROTECTED> |
03:57.10 | unabonger | <PROTECTED> |
03:57.12 | unabonger | <PROTECTED> |
03:57.14 | unabonger | <PROTECTED> |
03:57.16 | unabonger | <PROTECTED> |
03:57.18 | unabonger | <PROTECTED> |
03:57.20 | unabonger | <PROTECTED> |
03:57.22 | unabonger | <PROTECTED> |
03:57.24 | unabonger | <PROTECTED> |
03:57.25 | harryvv | dork |
03:57.26 | unabonger | <PROTECTED> |
03:57.27 | Math` | *pastebin* |
03:57.28 | unabonger | Nov 7 20:06:11 WARNING[426]: chan_sip.c:697 retrans_pkt: Maximum retries exceeded on call 473ab2205e8477684abb7d525b20b8d2@x.228.19x.77 for seqno 12360 (Non-critical Response) |
03:57.29 | *** kick/#asterisk [unabonger!n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted] by twisted (flood detected) |
03:57.35 | harryvv | use pastbin.ca |
03:57.46 | Math` | lets tell him when he joins back |
03:57.47 | kr5 | i am having some issues with dtmf |
03:57.52 | *** join/#asterisk unabonger (i=Bhongwan@lunacity.themoon.org) |
03:57.56 | Math` | unabonger: www.pastebin.ca |
03:58.00 | unabonger | dumb ass bots |
03:58.08 | unabonger | 8 lines is NOT a flood, jeebus help us. |
03:58.20 | harryvv | mmm i would say ask about doing that before flooding the channel |
03:58.31 | Math` | thats 14 lines, not 8 |
03:58.33 | harryvv | goto pastbin.ca and past it there. |
03:58.37 | unabonger | repeat "8 lines is NOT a flood" |
03:58.38 | harryvv | thats the rule here |
03:58.43 | unabonger | anal retentive bots |
03:58.44 | riksta | you pasted about 14 lines, and 8 is considered a flood to me |
03:58.58 | harryvv | everyone here past to pastbin.ca |
03:59.01 | unabonger | perhaps you have a form of digital ADD |
03:59.08 | unabonger | yes, I know about pastebin, |
03:59.11 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-132-198.gdrpmi.dsl-w.verizon.net) |
03:59.12 | unabonger | hold on. |
04:00.34 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
04:00.36 | Brijn | Anyone willing to have a look at a sip debug trace.. It must be something simple if you are used to these traces :( http://pastebin.ca/27948 |
04:00.41 | kr5 | what else would i need to set other than the dtmf mode in sip.conf and in the config for sjphone? Both are inband, but dtmf still refuses to work. |
04:00.56 | justinu | Brijn: what's the problem |
04:03.18 | justinu | your auth username looks wrong to me |
04:03.19 | unabonger | http://pastebin.ca/27950 |
04:03.26 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-240-180.dsl.irvnca.pacbell.net) |
04:03.35 | Qwell | unabonger: be glad it didn't include a +b |
04:03.53 | unabonger | humbug |
04:04.29 | Brijn | justinu: I got the config from the provider.. And their secret matches mine.. But it seems that my server goes back to their instead of sending the call to the extention (300) |
04:04.31 | unabonger | anyway, it just seems weird to me that the MOH doesn't work when calling from one source. |
04:05.00 | Math` | unabonger: check for voice activity detection on the phone |
04:05.43 | *** join/#asterisk [bsd] (n=bsd@203.134.194.11) |
04:05.55 | file[laptop] | Brijn: want an easy way out? |
04:06.35 | Brijn | Well, as long as I understand it :) |
04:07.03 | file[laptop] | Brijn: put insecure=very in your lightspeed-out entry... do a sip reload, then try it... |
04:07.06 | unabonger | Math` It's the same phone? Do you mean "suppress silence"? |
04:07.47 | unabonger | i'm using X-Ten Lite for my test, is the softphone my problem. I can try with hardware VOIP phone too. . . . |
04:07.48 | Math` | yeah, should be set to NO |
04:07.52 | *** part/#asterisk [bsd] (n=bsd@203.134.194.11) |
04:08.00 | Math` | transmit silence should be set to YES in x-lite |
04:08.16 | unabonger | already done |
04:08.21 | unabonger | hmmm |
04:09.34 | unabonger | like I said, I'm calling the same phone extensions (one through a SIP provider, and one through an IAX provider), and when the SIP provider calls come in, I can't hear the MOH when I put myself on hold. |
04:09.40 | justinu | Brijn: i'm not sure what's going on... been looking at sip too long today, just can't concentrate |
04:09.47 | Brijn | file[laptop]: Unfortunately that didn't fix it.. Let me compare the SIP output |
04:09.53 | file[laptop] | their way for calling you is weird |
04:10.08 | Math` | unabonger: that is kinda weird |
04:10.30 | file[laptop] | cause you won't get which phone number was called... but meh |
04:10.45 | unabonger | maybe the SIP provider who forwards my calls is blocking MOH ? |
04:10.55 | *** join/#asterisk MarkDude (n=mark@c-67-172-59-1.hsd1.pa.comcast.net) |
04:10.56 | Brijn | justinu: line 60-62, does that tell my server to talk to the mobile phone (the nr in there) thru the lightspeed server |
04:11.05 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
04:11.07 | Qwell | oh file[laptop]... |
04:11.12 | file[laptop] | hi Qwell |
04:11.29 | Qwell | remember that cidname lookup script we talked about a week ago or so? |
04:11.35 | file[laptop] | sure |
04:11.39 | Qwell | still need it? |
04:11.47 | file[laptop] | meh it's not overly urgent, so no |
04:11.52 | Qwell | heh |
04:12.19 | Pegger | on cisco phones can you load the firmware so that it is saved to flash memory or do you have to reload the firmware every time the phone boots??? |
04:12.35 | Qwell | Pegger: it doesn't need to load it every time |
04:12.45 | *** part/#asterisk MarkDude (n=mark@c-67-172-59-1.hsd1.pa.comcast.net) |
04:12.46 | Pegger | Qwell oha good |
04:12.46 | unabonger | you only have to reload the firmware once |
04:13.21 | Pegger | it is just most of the docs talk about setting upa tftp server in order to get sccp working with asterisk |
04:13.37 | Qwell | You do need a tftp server, to pull the config from |
04:13.37 | unabonger | er, once for every version you have to ugrade, which may be a chain of them, to get to the latest version. . . . |
04:13.40 | justinu | you need tftp |
04:13.49 | Brijn | file[laptop]: I tried with and without the ext added to the register statement. This is without any ext added http://pastebin.ca/27951 |
04:14.32 | Pegger | Qwell so you can just enter the server info from he touch pad? |
04:14.36 | Brijn | It seems so close to actually making the phone ring :) |
04:14.47 | Qwell | you probably could, but... |
04:14.59 | Qwell | it'd be extremely freaking tedious |
04:15.14 | Pegger | Qwell why you would have to do it every time? |
04:15.29 | Qwell | no, but you'd have to do it for every phone |
04:15.52 | rajiv | where can i get the patch to fix http://www.assurance.com.au/advisories/200511-asterisk.txt ? |
04:16.06 | Pegger | Qwell oha well I will only have one phone so not a big deal, so i guess you really only need the tftp for bigger instilaitons |
04:16.39 | Qwell | rajiv: just don't use the vmail.cgi script |
04:17.09 | rajiv | Qwell: i dont. but i want to put the patch into the gentoo ebuild for * |
04:17.12 | Pegger | so asterisk wil work with out any additional software right , what about chan_sccp doe sthat need to be loaded ? |
04:17.29 | Qwell | rajiv: check cvs.. |
04:17.36 | Qwell | I haven't seen any bugs about that fly by yet. |
04:17.36 | Dr_Ray | chan sccp is part of asterisk |
04:17.47 | Qwell | oh, guess they have |
04:17.49 | Qwell | Dr_Ray: no it isn't |
04:17.50 | Math` | Pegger: its chan_skinny |
04:17.58 | Pegger | Dr_Ray awsome |
04:18.01 | Qwell | chan_sccp is a seperate module... |
04:18.05 | Dr_Ray | I was wrong |
04:18.10 | Math` | Qwell: isnt skinny == sccp? |
04:18.24 | Dr_Ray | there are two sccp modules for asterisk |
04:18.27 | Qwell | Math`: it's the same protocol, sure |
04:18.30 | Qwell | Dr_Ray: more than that |
04:18.34 | Qwell | at least 3 |
04:18.36 | Dr_Ray | one built in, one not |
04:18.51 | Math` | Qwell: so sccp == skinny && skinny [is part of] * |
04:18.55 | Qwell | rajiv: yeah, cvs |
04:18.55 | Math` | so sccp is part of * |
04:18.58 | Pegger | yaha ther are chan_skinny and chan_sccp2 |
04:19.00 | Qwell | no |
04:19.05 | Math` | ah there are 2 versions |
04:19.10 | Qwell | * can use the sccp protocol... |
04:19.15 | Qwell | but, chan_sccp isn't in * |
04:19.17 | Pegger | chan_sccp is suposadly newer |
04:19.24 | rajiv | Qwell: http://cvsweb.digium.com/index.cgi/asterisk/contrib/scripts/vmail.cgi.diff?r1=1.15;r2=1.16 looks like |
04:19.28 | Math` | ok there's another implementation |
04:19.30 | Qwell | Pegger: chan_sccp is based off modifications to chan_skinny |
04:19.39 | Qwell | Math`: several more |
04:20.05 | Pegger | Qwell do you have an ipinion on which would would be better to use??? |
04:20.12 | Qwell | Pegger: I use chan_sccp |
04:20.20 | Qwell | but, I've never used chan_skinny, so...ymmv |
04:20.39 | Pegger | Qwell ok thanks |
04:20.52 | Brijn | justinu: http://pastebin.ca/27951 line 130... It's looking for my own IP in the [default] context...... Why :(( |
04:22.47 | Dr_Ray | gastman is ghastly |
04:23.08 | Dr_Ray | but I'm not happy with any of the management stuff |
04:24.53 | file[laptop] | Brijn: you need an extension 's' in the context 'lightspeed-in' |
04:25.06 | file[laptop] | Brijn: once that's done, that should be working |
04:26.19 | Qwell | man...that name always bothers me...lightspeed |
04:26.25 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
04:27.13 | rajiv | stkn: hi |
04:28.00 | *** join/#asterisk marc32344 (n=marc3234@206-248-128-180.dsl.teksavvy.com) |
04:29.57 | *** join/#asterisk wolfson` (n=ggggg@usr-kdh-208-6-58-26.beachlink.com) |
04:30.00 | Brijn | file[laptop]: there is one, let me pastebin it http://pastebin.ca/27953 |
04:30.19 | Brijn | file[laptop]: so close! |
04:30.23 | [TK]D-Fender | Qwell : yeah lightspeed bothers me too : Since light is both a particle and an EM band to itself this lends us to ponder several cute ideas like : If light is a wave, that means that light meanders, but if it isn't going straight, doesn't that mean its actually going faster than the speed of light? |
04:30.28 | *** join/#asterisk riksta (n=rick@62.6.163.90) |
04:30.38 | Qwell | umm |
04:30.39 | Brijn | Qwell: There support isn't at the speed of light anyway |
04:30.39 | [TK]D-Fender | that would mean we compare everything to the speed of light taking its time :D |
04:30.55 | Qwell | my reference to lightspeed is the first hit on google images |
04:31.05 | Qwell | (NSFW!) |
04:31.22 | Brijn | hahaha |
04:31.42 | Brijn | Their support staff?? |
04:31.45 | Qwell | so yeah...needless to say, I wouldn't use a voip provider of the same name. :) |
04:31.51 | Qwell | well...unless that were the case |
04:31.53 | file[laptop] | Looking for 300 in lightspeed-in |
04:31.59 | file[laptop] | make sure there is an extension 300 in lightspeed-in |
04:33.11 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
04:33.20 | Brijn | WOEHOE!!!!! |
04:33.22 | *** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au) |
04:33.32 | Brijn | Thanx man! |
04:34.02 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
04:34.16 | kuku5 | Anyone here ever seen a c program that write to the serial port? |
04:34.26 | Brijn | Now I at least have something work, and can start from there.. See if understand what was wrong all the time.. |
04:34.26 | justinu | on linux? |
04:34.46 | spootnick | hi all. how do i display the realtime SIP peers currently registered in my * box via CLI ? |
04:34.52 | mog_home | you can dump to /dev/tty |
04:34.55 | mog_home | right |
04:35.01 | javo | i am having trouble with my sip phone. i can call out anywhere, but when people call me, it says that i am on the phone and goes straight to voicemail. does anyone know a likely cause of this? |
04:35.01 | Math` | kuku5: just open /dev/ttySX |
04:35.08 | kuku5 | how do i do that |
04:35.15 | Math` | open() or fopen() |
04:35.19 | kuku5 | ergh |
04:35.25 | Qwell | kuku5: it's going to be a long night for you, if you have to ask that... |
04:35.46 | kuku5 | great :) |
04:36.33 | [TK]D-Fender | javo : pastebin your extensions.conf |
04:37.07 | kuku5 | Qwell: do i compile with gcc or cc |
04:37.41 | kuku5 | :) |
04:37.47 | Qwell | kuku5: This is not the forum for that. |
04:37.53 | kuku5 | i know |
04:38.03 | kuku5 | but i get and give a lot of help here :) |
04:38.30 | justinu | go to #c |
04:38.34 | justinu | talk to mixe |
04:38.39 | justinu | on efnet |
04:38.53 | justinu | tell him j2 said to help you |
04:39.23 | delmar | in HEAD, usage of SetVar() has been changed to Set(), does anyone know if this is similar for SetGlobalVar? is it not SetGlobal ? |
04:39.30 | kuku5 | they need a key :( |
04:39.30 | delmar | not=now rather |
04:39.44 | marc32344 | what does fifo_db do in ser? |
04:40.15 | kuku5 | mixe isnt online :( |
04:40.25 | justinu | drat... |
04:40.30 | justinu | svetmixe? |
04:41.08 | Qwell | delmar: show applications like set |
04:41.23 | kuku5 | justinu: nope |
04:42.52 | delmar | cheers, looks like SetGlobalVar is still the thing |
04:43.25 | *** join/#asterisk Evanrude (n=david@wsip-68-15-251-34.dl.dl.cox.net) |
04:44.44 | gambolputty | Is kpfleming here? |
04:46.22 | *** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar) |
04:46.45 | znoG | could someone fill me in on the technical reason why an Ambient MD5XXX card doesn't work with Asterisk? |
04:47.01 | Qwell | znoG: without knowing, I'd guess lack of drivers |
04:47.02 | znoG | is there some hardware restriction or simply support for it hasn't been written? |
04:48.36 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
04:48.36 | *** mode/#asterisk [+o denon] by ChanServ |
04:48.40 | kuku5 | Qwell: http://pastebin.com/421397 < invalid argument |
04:49.41 | Math` | thats EINVAL? |
04:51.10 | mog_home | i much prefer EAGAIN |
04:51.15 | *** join/#asterisk pcm (n=pcm@user-69-73-0-22.knology.net) |
04:52.31 | Math` | kuku5: what do u want to do |
04:53.38 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
04:54.05 | kuku5 | read the serial port - thats it |
04:54.13 | kuku5 | ( for starters ) |
04:54.32 | Math` | but why do u want to do that, whats your final goal |
04:55.52 | pcm | is there anywhere the log from the channel ? |
04:56.26 | Math` | 0x378 is the parallel port, fyi |
04:56.40 | *** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net) |
04:59.02 | denon | unless of course your parallel port is 0x278 |
04:59.14 | Math` | lol |
04:59.17 | denon | </being_difficult) |
04:59.19 | denon | > |
04:59.52 | blitzrage | hypa7ia: j00 'round? |
05:00.22 | kuku5 | i have an embedded device that has proximity card readers and relays connected to it that control door access |
05:00.49 | kuku5 | what is the com port |
05:02.12 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
05:03.33 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
05:04.22 | gambolputty | Is anyone here a bug marshall? |
05:04.52 | *** part/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
05:05.26 | mog_home | gambolputty are you cyberdjheffer |
05:05.31 | mog_home | cause i am a marshall |
05:09.10 | *** join/#asterisk Tili (i=Tili@61.144.21.58) |
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05:18.57 | *** mode/#asterisk [+o denon] by ChanServ |
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05:23.05 | delmar | whats the best way to do like... ok national dialing can be 0 then 8-10 digits... and international is 00 then could be 8-12 digits... ie. exten => _0(what goes here)... and exten => _00(what goes here) ? |
05:23.16 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
05:23.16 | file[laptop] | . |
05:23.20 | delmar | just never got the hang of the X and N and stuff :P |
05:23.22 | file[laptop] | . will match any length |
05:23.43 | Math` | N matches > 1 |
05:23.48 | Math` | X matches anything |
05:23.53 | Math` | (except *) |
05:23.59 | Qwell | Math`: not A, B, C, or D, or # |
05:24.07 | Math` | matches any digit** |
05:24.08 | Math` | :P |
05:24.59 | delmar | ok but i cant do _0., because they need to be seperate |
05:25.18 | Qwell | delmar: what country? |
05:25.22 | delmar | NZ |
05:25.25 | Qwell | no clue |
05:25.33 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
05:25.34 | delmar | ok let me explain further... |
05:25.58 | Qwell | there are rules you need to follow... You can either do 3 seperate lines for NZ national, or 1 line with 8 digits followed by . |
05:26.21 | Qwell | is there a "match zero or one character" match? |
05:26.41 | Qwell | I think it'd be similar to ? elsewhere |
05:27.15 | delmar | ok numbers would be like this... |
05:27.43 | *** join/#asterisk freat (n=freat@h-69-3-229-184.chcgilgm.covad.net) |
05:28.24 | delmar | national can be for example.. 063271234(lanline) or 021409123 but can also be as long as 0211123456.... |
05:29.13 | Qwell | delmar: is there a minimum each digit can be? |
05:29.20 | delmar | international is for example... 00 678 29333 (example of a short number in Vanuatu) or 0013038281234 for a number in the usa etc |
05:29.27 | Qwell | ie; in the US, the lowest possible number would be 2002000000 |
05:29.33 | blitzrage | you match as much as you can, then use a pattern or . after... most specific will match |
05:29.41 | blitzrage | _NXXNXXXXXX |
05:29.46 | blitzrage | N == 2-9 |
05:29.48 | Qwell | blitzrage: I'm thinking more in the middle. |
05:29.57 | blitzrage | Z == 1-9 |
05:29.58 | Qwell | like NXX?XXXXXXX |
05:30.02 | delmar | Qwell, yeah i guess so... the minimum would be... 0XXXXXXXX so 0+8 |
05:30.05 | blitzrage | X == 0-9 |
05:30.26 | Qwell | delmar: so, any part of the number can be any digit? even 0 or 1? |
05:30.43 | Qwell | if so, _0XXXXXXXX. would do the trick |
05:30.49 | delmar | let see... |
05:31.08 | Qwell | and if there were ? (there isn't...I just checked), you could do like 0XXXXXXXX?? |
05:31.13 | Qwell | _0XX...etc |
05:31.22 | delmar | ok |
05:31.29 | blitzrage | there is no ? afaik |
05:31.36 | blitzrage | and what would ? be? |
05:31.47 | Qwell | blitzrage: 0 or 1 of any digit |
05:31.49 | Qwell | like... |
05:31.55 | Qwell | _X? would match 1, or 11 |
05:32.00 | Qwell | but not 111 |
05:32.09 | Qwell | whereas _X. would match 1, 11, 111, 11111111111 |
05:32.24 | delmar | so the answer is... exten => _0XXXXXXXX. etc for NZ National and that should cover it... then I can have _00. for International? |
05:32.38 | implicit | ?? |
05:32.48 | blitzrage | hrmm... |
05:32.51 | Qwell | delmar: no, see...that won't work. Something is either broken with the NZ implementation, or you're wrong |
05:33.00 | blitzrage | probably need _X and _XX lines |
05:33.03 | implicit | ?????? |
05:33.07 | Qwell | You need to dial 00 to exit the country? |
05:33.22 | Qwell | that couldn't possibly work if an NZ phone number started with 0 |
05:33.29 | blitzrage | then don't match on the 00 -- just dial the number and add 00 to the front of the ${EXTEN} lines |
05:34.41 | delmar | Qwell, ok what im "trying" to do is simulate NZ dialing on *, but it would probably go via VoIP provider most of the time... so .. one line for NZ National where it will Dial(IAX2/user@provider/64${EXTEN:1} sorta thing.. and another where the 64 code is ommited.. for all other dialing.... you get me? |
05:35.01 | Qwell | nope |
05:35.49 | delmar | ok 0 then area code then number.. national dialing... 00 then country code.. then area code .. international dialing |
05:35.51 | Qwell | bbl |
05:36.28 | Math` | delmar: can an area code start with 1? |
05:36.33 | delmar | the national dialing rule... needs to add a 64 when sending to the provider. |
05:36.41 | delmar | hrm |
05:36.42 | delmar | no |
05:36.52 | delmar | Math` what u thinking? |
05:37.06 | Math` | so exten => 0N. ; in-country |
05:37.16 | Math` | exten => 00. ; international |
05:37.25 | delmar | only numbes starting with 1 are service numbers and are like.. 123 126 u dont dial 0123 .. just 123 |
05:37.38 | Math` | and they are 3 digits? |
05:37.42 | blitzrage | Math`: don't forget the _ :0 |
05:37.45 | blitzrage | err :) |
05:37.50 | Math` | blitzrage: ah thanks for that one |
05:38.07 | delmar | Math`, yeah but i already have a prior context for service numbers to override them and do Congestion :P |
05:38.17 | Math` | lol |
05:38.22 | delmar | Math`, ok cheers, ill impliment that and see if it works :P |
05:40.08 | delmar | Oh hey so just to clarify.. the _ is needed when doing like _00., etc but not needed when doing stuff like 1234, etc... when else is a _ required and wahts the deal? |
05:40.21 | Math` | yeah |
05:40.30 | Math` | _ is when you need to match |
05:40.40 | Math` | aka: when its not only digits (or special extension) |
05:40.43 | delmar | i take it it doesnt matter if u have _1234,etc either? |
05:40.51 | Math` | u dont do _1234 |
05:41.01 | Math` | but you do _NXXNXXXXXX etc... |
05:41.29 | delmar | ah i get it. _ for pattern matching .. without when u are not. |
05:43.54 | Math` | :) |
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06:21.58 | delmar | usage of goto is like.. exten => 1234,1,Goto(context,X,N) sorta thing right? |
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06:23.54 | delmar | hmm |
06:23.54 | Math` | delmar: goto(context,extension,priority) |
06:23.54 | delmar | yeah |
06:23.54 | Math` | goto(outgoing,1234,1) |
06:23.54 | Math` | what was your question |
06:23.54 | delmar | could also be exten => _1234.,1,goto(context,extension,priority) too right? |
06:23.54 | Math` | of course |
06:23.54 | delmar | hrm. |
06:24.32 | delmar | didnt think i had it messed up.. not playing ball tho. |
06:24.33 | *** join/#asterisk ComputerWarm (n=dan@rddrpx29-port-20.dial.telus.net) |
06:25.14 | ComputerWarm | Hello all, anyone here use astbill? |
06:25.14 | Math` | delmar: whats the error |
06:25.45 | delmar | Label missing trailing ')' at line 223 |
06:25.45 | Math` | whats line 223? |
06:26.10 | delmar | sec. |
06:26.30 | delmar | nm. i see. |
06:26.38 | Math` | ur missing a ) :P |
06:26.52 | delmar | no that would say missing parenth. |
06:27.01 | Math` | ah |
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06:27.04 | delmar | its quite good at tell u whats wrong. |
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06:27.19 | Math` | missing trailing ) sounds like missing ) for me |
06:27.41 | FuriousGeorge | so how come the freenode op is called lilo |
06:28.45 | Math` | FuriousGeorge: because lilo is a staff member of freenode? |
06:29.00 | FuriousGeorge | Math`: sounds good |
06:29.13 | Math` | indeed |
06:29.18 | mog_home | i get those messages from dmwaters |
06:29.24 | mog_home | and always think he is someone i know |
06:29.30 | mog_home | and am dissapointed everytime.. |
06:29.40 | FuriousGeorge | lol |
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06:30.05 | FuriousGeorge | well i guess you know him now, but regardless of the name hes almost always disappointing |
06:31.20 | Math` | if u say so... |
06:31.39 | ComputerWarm | hello. anyone here used astbill? |
06:33.28 | Math` | uhm I've a deja-vu impression |
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06:40.45 | *** join/#asterisk SarahEmm (n=sarahemm@2.35.220-216.q9.net) |
06:40.53 | SarahEmm | hihi |
06:42.43 | morale | irc'ing from your workstation? |
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06:43.45 | SarahEmm | morale: me? |
06:43.47 | SarahEmm | oh lol |
06:43.49 | SarahEmm | heya morale. |
06:43.55 | SarahEmm | took me a second to realize who you were. |
06:43.59 | Vco | http://rmeek141.home.comcast.net/TrollBooth.JPG |
06:44.04 | SarahEmm | yep! ircing at work, as are you :) |
06:44.27 | Math` | what time is it for you? |
06:44.34 | SarahEmm | it's 0144 for me |
06:44.38 | morale | SarahEmm: i go through my home system so no one dos's me |
06:44.40 | SarahEmm | i work rotating shifts. nights this week. |
06:44.55 | SarahEmm | morale: heh. i usually do, but having some VPN issues right now so i'm not. |
06:45.04 | morale | ah. |
06:45.42 | delmar | hmmm |
06:45.53 | SarahEmm | morale: you ssh home and use a console client, or vpn home? |
06:46.44 | syle | cat cost me 150 bucks at vet tonight because she had a fever! |
06:46.53 | morale | i just ssh home.. |
06:47.11 | SarahEmm | morale: ahh. |
06:47.12 | delmar | I'm trying to setup a variation of the emergency call handling found at http://www.voip-info.org/wiki-Asterisk+tips+911... the last one.. and it's just looping out and not doing what it should. |
06:47.13 | morale | syle: it costed me 400$ two weeks ago on a dead cat. |
06:47.41 | delmar | can anyone see anything major wrong with the last variation at http://www.voip-info.org/wiki-Asterisk+tips+911 ? |
06:47.51 | syle | god i think it cost me that much to get both my cats first shots and spaded when they were kittens |
06:48.40 | syle | you paid 400 bucks to put a cat to sleep? |
06:48.51 | delmar | syle, yeah one of our cats had some kinda spasm which we now think was a pulled mussel but she was lookin all wierd walkin lopsided so off to the Vet.. $80 call out plus another $70 for the consult |
06:49.33 | delmar | bah. im off to mow the lawns. bbl |
06:49.46 | syle | mow the lawn |
06:49.51 | syle | what part of the world do you live in |
06:50.00 | syle | like 1 am here hehe |
06:50.02 | Vco | lawns? |
06:50.21 | Qwell | oh shit... http://www.voip-news-net.com/2005/11/voip_e911_cutof.html |
06:50.30 | Qwell | FCC folded? |
06:50.42 | implicit | hehehe |
06:50.50 | implicit | you ddin't know? |
06:52.47 | syle | says they are only folding on members before nov 28 |
06:52.52 | syle | not any new ones after that |
06:53.09 | Qwell | so? heh |
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06:54.13 | syle | they don;t really have a choice but to fold i think, a million people pissed off at the FCC |
06:54.25 | Vco | "more" pissed at teh FCC |
06:54.38 | syle | budget cuts would come to that dept easily |
06:55.42 | implicit | Vco, yeah :) |
06:57.18 | Vco | oooh |
06:58.04 | Qwell | Vco: they have pricing? |
06:58.25 | Vco | http://www.voipsupply.com/index.php?cPath=0_99_300 |
06:59.09 | Qwell | still nothing at the digium store. :( |
06:59.15 | Vco | heh..heh.. |
06:59.23 | mog_home | yeah |
06:59.27 | mog_home | its comming |
06:59.29 | mog_home | its comming |
06:59.30 | Vco | surprise: Please Note: We are currently taking pre-orders for the new Digium TDM2400 Series, Full-Lenth PCI cards. Product is anticipated to begin shipping 11/18/05. |
06:59.34 | Qwell | ahh... |
06:59.36 | Vco | as per ususal |
06:59.41 | Vco | usual even |
06:59.53 | Vco | at they have some numbers tho |
07:00.11 | Qwell | wow, thats pricey |
07:00.22 | syle | price? |
07:01.06 | syle | holy crap |
07:01.10 | implicit | 2495 for 24fxs w/ on board echo cancellation |
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07:01.38 | Vco | i mean...fxs i can kinda see |
07:01.46 | syle | well i guess i won;t be returning my rhino channel bank lol |
07:01.49 | implicit | Vco, why, lol |
07:01.50 | Vco | haha |
07:01.53 | clive- | a channel bank is way cheaper than that |
07:01.56 | RoyK | 24 pri? |
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07:02.02 | RoyK | :) |
07:02.09 | SarahEmm | morale: thanks for reminding me to fix my VPN :) all better now. |
07:02.32 | RoyK | clive-: a channel bank seems to me a far cleaner solution as well |
07:02.38 | implicit | RoyK, yeah |
07:03.10 | Qwell | hmm |
07:03.17 | Qwell | freaking idiots at voip-supply, I swear |
07:03.32 | implicit | digium developed it? |
07:03.37 | mog_home | yes |
07:03.40 | Qwell | 40fxs / 0fxo - $789.95 |
07:03.46 | mog_home | digium develops all digium hw |
07:03.46 | Qwell | THAT is why I don't like them |
07:03.49 | implicit | lol |
07:03.54 | mog_home | 40fxs! awesome |
07:03.57 | Qwell | They constantly fuck up like that, heh |
07:03.58 | mog_home | i need that card |
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07:04.08 | mog_home | olle! |
07:04.32 | SarahEmm | wowie, there's a new TDM card.. niftys :) |
07:04.43 | implicit | SarahEmm, lol |
07:04.49 | SarahEmm | implicit: mew? |
07:04.55 | implicit | i wouldn't use them |
07:05.13 | syle | 40 fxs for 789 ? |
07:05.14 | syle | where |
07:05.16 | SarahEmm | implicit: because? |
07:05.20 | Vco | although, for the 4 ports, compared to buting a tdm400.... |
07:05.26 | Vco | compared to buying.. |
07:05.27 | Vco | even |
07:05.58 | implicit | SarahEmm, expensive for what it is, uses softwaredsp, and probably shitty quality |
07:06.28 | mog_home | nah implicit |
07:06.28 | mog_home | it sounds clean |
07:06.28 | mog_home | like any card |
07:06.36 | SarahEmm | implicit: ahh. mew. |
07:06.46 | SarahEmm | implicit: i'm running an X100p for part of my home system, works well for me :) |
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07:07.01 | mog_home | all zap cards sound like any dsped card |
07:07.02 | syle | i;m authorized to resell rhino channel banks if anyone wants one |
07:07.10 | Qwell | syle: sure, I'll take one |
07:07.14 | implicit | mog_home, not exacty |
07:07.20 | implicit | and they end up being more expensive |
07:07.28 | implicit | because of the high requirements on DSP |
07:07.28 | mog_home | umm id love for you to show me the difference |
07:07.33 | mog_home | or if you could pick one out of a line up |
07:07.40 | Vco | if you're authorized to take bits of string as payment i'll take a few |
07:07.43 | mog_home | on normal t1 lines or analog etc |
07:08.03 | mog_home | i think people just dont realize that pcs have gobs of proccessing power |
07:08.09 | mog_home | that you can throw at silly things like echo |
07:08.37 | implicit | PRI coming in through a digium card on * i could completely tell apart from a one coming into a cisco AS5400 |
07:08.38 | Vco | do you have the one with echo cancel on the card? |
07:08.45 | mog_home | sure implicit |
07:08.54 | mog_home | i find that pretty hard to believe |
07:09.00 | mog_home | unless your blind |
07:09.05 | mog_home | and have super human hearing |
07:09.06 | implicit | mog_home, i am |
07:09.09 | implicit | fucker |
07:09.10 | Qwell | implicit: I hear the Navy is hiring human sonar. You might want to apply |
07:09.13 | implicit | ;) |
07:09.13 | mog_home | 8khz mono sucks going through anything |
07:09.41 | mog_home | whatever |
07:09.55 | mog_home | if you want to be an audio head thats fine |
07:10.01 | mog_home | but the 99% of the rest of us |
07:10.05 | mog_home | cant tell jack squat |
07:10.39 | implicit | i admit i'm pretty anal about audio quality, that's why i buy things like http://www.sensaphonics.com/soft2x.html |
07:10.53 | implicit | but the quality difference is pretty large |
07:11.02 | implicit | someone untrained could easily pick it out if they were listening for it |
07:12.03 | Vco | wow..it's a good thing voip supply marks this Digium hardware as Asterisk tested, or i don't know what i might have ended up buying.... |
07:12.14 | SarahEmm | lol |
07:12.18 | implicit | Vco, lol |
07:17.07 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
07:17.20 | lehel | 'mornin |
07:17.31 | mog_home | mornin |
07:17.31 | Qwell | wow...umm...who said T1 card + rhino channel bank was cheaper? |
07:17.37 | mog_home | yeah |
07:17.45 | mog_home | used channel bank |
07:17.46 | mog_home | and t1 |
07:17.47 | Qwell | $1800 for one of these suckers |
07:17.49 | mog_home | might be cheaper |
07:17.56 | mog_home | but getting fully populated channel bank |
07:17.58 | mog_home | is pricy.... |
07:18.12 | mog_home | used zhones though do rock if you can find em |
07:18.23 | Qwell | zhones? |
07:18.32 | mog_home | zhone is a crappy channel bank |
07:18.36 | mog_home | you can find em on ebay |
07:18.37 | mog_home | sometimes |
07:18.41 | mog_home | but they go for like 500 |
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07:18.57 | Qwell | crappy but they rock? |
07:19.11 | mog_home | well they dont have all the niceties |
07:19.14 | mog_home | like for example |
07:19.18 | mog_home | no fxo caller id |
07:19.29 | Qwell | thats dirty |
07:19.30 | mog_home | and they have been known to drop all the channels |
07:19.41 | mog_home | but other than that |
07:19.45 | mog_home | they rule |
07:19.49 | Qwell | we have a nortel pbx at work...we don't get cid... |
07:20.02 | mog_home | ew |
07:20.08 | Qwell | biggest pile of shit phones too... |
07:20.13 | Qwell | they don't have a god damned mute button |
07:20.17 | Qwell | no, they do... |
07:20.25 | Qwell | but if you press it...BAM, speakerphone |
07:20.26 | Vco | >crotch< |
07:20.31 | mog_home | lol |
07:20.35 | *** join/#asterisk shido6 (i=shido6@d221-68-216.commercial.cgocable.net) |
07:20.36 | mog_home | that rules |
07:21.11 | Qwell | like...when you think about something like that... |
07:21.30 | Qwell | imagine the biggest cluebie coming in here, asking about buying a hundred budgetones... |
07:21.33 | mog_home | would make a great gag to screw the new guy in office |
07:21.41 | Qwell | thats the type of person who buys this pbx, I swear |
07:21.53 | Vco | that ranks up there with the wallmount budgetone that doesn't have a hook to keep the reciever on ..... |
07:22.00 | mog_home | lol |
07:22.02 | Qwell | Vco: you're kidding? |
07:22.09 | mog_home | put it on a slanted wall |
07:22.10 | Vco | the 100's |
07:22.11 | mog_home | or the roof |
07:22.13 | mog_home | it works great |
07:22.14 | Qwell | thats awesome |
07:22.37 | Vco | i have a frankentone phone... |
07:22.42 | Vco | add BT100 phone, |
07:23.03 | Vco | and old USR wifi bridge into the case |
07:23.28 | Vco | voila! The single shittiest wifi phone you could even order pizza with |
07:24.18 | Vco | it was kinda annoying, since it was a white phone, and the bridge had those super bright blue leds.. |
07:24.28 | Vco | frickin lightshow at night... |
07:24.53 | SarahEmm | lol |
07:24.58 | mog_home | damn someones car alarm is going off |
07:25.04 | SarahEmm | and a web interface tty app |
07:25.06 | mog_home | i wish someone would just steal the damn thing |
07:25.16 | mog_home | been going off for 30 minutes |
07:25.28 | mog_home | its like the guy just wants us to blow his car up |
07:25.40 | Vco | hmm.. |
07:25.41 | Qwell | here, they'll tow it if the alarm goes off for longer than x minutes |
07:25.57 | mog_home | no one is doing anything |
07:26.02 | mog_home | some people came outside angry |
07:26.04 | mog_home | then left |
07:26.18 | Vco | where you live? |
07:26.26 | mog_home | huntsville |
07:27.06 | Vco | ohh..speaking of car alarms andsuch i should get remote start installed like....soon.. |
07:27.13 | implicit | mog_home, explains why you are so into that digium card |
07:27.14 | implicit | lol |
07:27.26 | implicit | mog_home, i see where your bias comes from |
07:27.35 | syle | oww vco you should book that right now |
07:27.37 | mog_home | i thought it was pretty well known |
07:27.39 | implicit | :) |
07:27.41 | syle | took me 3 weeks to get in |
07:27.52 | implicit | mog_home, always used mog_home as your handle? |
07:28.05 | mog_home | mog_home, mog_work, or mogorman proper |
07:28.14 | implicit | how long have you been working for digium? |
07:28.22 | mog_home | just over a year |
07:28.30 | syle | what do you do? |
07:28.33 | implicit | and how long around */ |
07:28.35 | mog_home | support and dev |
07:28.37 | RoyK | mog_home: what do you think about the openpbx project? |
07:28.39 | implicit | i'm surprised we don't know eachother better |
07:28.40 | mog_home | just a bit before that |
07:28.44 | syle | #DEFINE dev |
07:28.48 | clive- | mog_home when are digium going to get G729 with PLC or G729b ? |
07:28.50 | mog_home | royk you know what i think about that ^_^ |
07:29.00 | RoyK | three small characters |
07:29.04 | mog_home | g729 b should probably work as is |
07:29.09 | mog_home | but with plc no idea |
07:29.11 | ComputerWarm | question all does Asterisk support G723.1? |
07:29.12 | potsboy | hi all.. is it possible to register a analogue phone though a rhino bank into a queue? |
07:29.14 | implicit | havn't been around much lately but i was around a lot more before |
07:29.19 | RoyK | what's the diff between g729a and -b? |
07:29.19 | mog_home | i do little stuff for digium, i do bigger things in spair time |
07:29.24 | clive- | computerworm tyes illegally it does |
07:29.28 | Qwell | potsboy: sure, why not? a channel is just a channel |
07:29.36 | RoyK | ComputerWarm: no it does not. g723.1 license is way to expensive |
07:29.42 | clive- | RoyK b has PLC, plus a few extras in it |
07:29.56 | RoyK | clive-: what is plc? |
07:30.01 | mog_home | g723 is difficult legally as well |
07:30.04 | clive- | packet loss concealemnt |
07:30.09 | RoyK | ok |
07:30.10 | ComputerWarm | RoyK oh. will asterisk do a pass thru for that ? |
07:30.12 | RoyK | that's nice |
07:30.13 | potsboy | aah k so it will reference it by channel not ip |
07:30.17 | RoyK | ComputerWarm: it will |
07:30.27 | RoyK | ComputerWarm: at least in theory. haven't tried it, though |
07:30.31 | potsboy | tx |
07:30.53 | ComputerWarm | RoyK ok thanks. i guess I will have to try it or talk the carrier in to switching to G729. |
07:31.13 | clive- | didnt know any carrioers used 723.1 |
07:31.23 | ComputerWarm | clive- these guys do |
07:31.27 | Qwell | is 723.1 even any good? |
07:31.45 | RoyK | Qwell: i beleive so, for slow links |
07:31.54 | mog_home | g723 is just really small |
07:31.56 | clive- | 723.1 is ok , but sucks when you have packet loss |
07:32.02 | mog_home | its like 729 doesnt sound better |
07:32.02 | ComputerWarm | Qwell its crappy. most the time |
07:32.04 | mog_home | but is tiny |
07:32.30 | mog_home | damn im just gonna call the police on this guy |
07:32.35 | mog_home | it wont stop |
07:32.47 | clive- | what guy mog? |
07:32.54 | Qwell | clive-: you |
07:32.55 | mog_home | this guy left his car alarm |
07:33.00 | mog_home | its been going off |
07:33.04 | mog_home | FOREVER |
07:33.26 | clive- | lol, in south africa you can cal teh police for an armed robbery and they dont arrive |
07:33.29 | syle | call it in to towing company, they will be more than happy to take it |
07:33.43 | mog_home | can you really do that |
07:33.50 | Qwell | call and ask |
07:33.58 | mog_home | that would be awesome |
07:34.00 | syle | of course, they make money this way |
07:34.00 | Qwell | if so, they'll be there in a second |
07:34.04 | mog_home | like parking lot is full |
07:34.08 | mog_home | tow away some guy |
07:34.09 | syle | costs them to pick up their vehicle after |
07:34.11 | mog_home | take his spot |
07:34.16 | Qwell | mog_home: no :p |
07:34.19 | Vco | call in a noise complaint |
07:34.21 | mog_home | why not..... |
07:34.26 | Qwell | just like you can't call a locksmith to break into a ferrari |
07:34.31 | mog_home | yeah i did , the management where i live sucks |
07:34.50 | mog_home | i picked the place closest to campus with the cheapest rent |
07:34.57 | mog_home | there are drawbacks to this though.... |
07:35.08 | Vco | "car alarm has been going off for a while, angry mob gathering downstairs.." |
07:35.41 | Vco | pitchforks and clubs and such |
07:35.49 | Qwell | you could probably get away with throwing a rock through the windshield at this point |
07:35.54 | mog_home | lol |
07:36.03 | Qwell | "dude, the alarm was totally going off when I got home...I've got like...200 witnesses" |
07:36.03 | mog_home | someone is under the car now |
07:36.15 | mog_home | i think they are gonna disco his battery |
07:36.18 | Qwell | heh |
07:36.24 | mog_home | or cut his breaklines |
07:36.28 | mog_home | whatever it is |
07:36.31 | Qwell | problem solved |
07:36.32 | mog_home | i dont care at this point |
07:36.49 | Vco | y'know...depending on what floor you are on...a 2L pop bottle full of water does some nasty ass damage (at least off the 25th floor it did anyway) |
07:37.11 | mog_home | heh im just on second |
07:37.12 | Qwell | Vco: I imagine 5th and above would start doing it |
07:37.21 | Vco | well.. |
07:37.21 | mog_home | well whatever that guy did he killed the alarm |
07:37.38 | Qwell | well...2liter... |
07:37.43 | Qwell | maybe 4th |
07:37.54 | Vco | hey ..he killed the alarm, unlocked the door AND popped the ignition...fancy that.. |
07:38.18 | Qwell | *10 minutes later* <mog_home> so, turns out it was my car |
07:38.20 | Vco | pissed off people tossing him gas money to get it out of the area |
07:38.23 | mog_home | lol |
07:38.24 | mog_home | hehe |
07:38.34 | SarahEmm | rofl Qwell |
07:38.46 | mog_home | man i could leave my car with the windows down and the key in the ignition no one would steal it |
07:39.11 | mog_home | i drive the out on my own broke as hell college student mobile, thats right an 89 olds mobile calias |
07:39.30 | Qwell | brown? |
07:39.38 | mog_home | its started red |
07:39.42 | mog_home | but its getting there |
07:40.31 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
07:43.29 | *** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus) |
07:47.45 | Qwell | bed time |
07:48.17 | RoyK | hm... does anyone know how I can allow a server to listen to two ip addresses, one on each a network, and allow each of them to be accessed from the internet? currently i have a default gateway on one of the networks, but that means if I try to access it on it's other address, it'll try shipping back the data out the other nic, which is rather a bad thing.... |
07:48.25 | RoyK | a little OT, perhaps, but then :P |
07:48.36 | Qwell | by server, you mean? |
07:48.52 | Qwell | just any program? |
07:49.04 | RoyK | linux system |
07:49.31 | RoyK | meaning having two default gateways, one for each nic |
07:49.33 | RoyK | somehow |
07:50.01 | Qwell | RoyK: think there is something on tldp.org - perhaps the advanced routing howto |
07:51.36 | Qwell | probably just a little bit of iptables magic |
07:52.43 | *** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543623pcs.mainf01.in.comcast.net) |
07:54.00 | Qwell | http://www.tldp.org/HOWTO/Adv-Routing-HOWTO/lartc.rpdb.multiple-links.html part of it |
07:54.50 | Qwell | RoyK: I accept paypal. :P |
07:54.57 | mog_home | lol |
07:55.06 | mog_home | dont we all qwell |
07:55.10 | Qwell | ;] |
07:55.13 | mog_home | i have gotten 20 bucks thus far |
07:55.17 | mog_home | it rocked |
07:55.30 | RoyK | Qwell: lol |
07:56.57 | Qwell | That link should actually do most of what you need...as long as your programs can listen on multiple interfaces (not difficult) |
07:58.28 | *** join/#asterisk phara0h (n=yah@229.80-202-56.nextgentel.com) |
07:58.37 | phara0h | Hi!, anyone working on documentation for v1.2 yet? (I am a couple of other are, and I was wondering if there was some place we should put our documentation/read others documentation so we don't create the same thing :)) |
07:58.44 | mog_home | there is |
07:58.46 | mog_home | ~thebook |
07:58.49 | jbot | from memory, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
07:59.00 | mog_home | and asteriskdocs |
07:59.04 | mog_home | and wiki |
07:59.07 | mog_home | etc etc |
07:59.14 | Qwell | and bugtracker, if you can do doxygen docs |
07:59.24 | mog_home | make progdocs did rock |
07:59.29 | mog_home | but needs work |
07:59.39 | phara0h | mine crashed :( |
07:59.54 | phara0h | (on SuSE OSS 10.0) |
08:00.18 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
08:00.43 | *** join/#asterisk AgentRed (n=bharatsa@210.211.246.47) |
08:00.48 | AgentRed | Hello there |
08:01.10 | SarahEmm | hihi AgentRed |
08:01.23 | phara0h | So what you are saying there is really no central place where everyone is putting most of the new docs.. actually I first went to voip-info, but there was hardly anything on the page called "Asterisk v1.2" ... just a listing of dialplan functions .. |
08:01.50 | AgentRed | Iam unable to load Asterisk cos the of the failure in loading the app_voicemail.so |
08:02.16 | mog_home | asteriskdocs |
08:02.19 | mog_home | is good place |
08:02.25 | AgentRed | can anybody please guide me as to how do I come up with a solutino for this problem |
08:02.32 | phara0h | ok, i'll check it out :) |
08:02.35 | mog_home | what is the error |
08:03.03 | AgentRed | hmm |
08:03.22 | AgentRed | wait let me paste the error in the pastebin |
08:04.20 | mog_home | hurry as im about to sleep |
08:04.25 | AgentRed | ok |
08:04.30 | AgentRed | wait a sec [please |
08:05.59 | AgentRed | http://pastebin.com/421501 |
08:06.03 | AgentRed | this is the link |
08:06.13 | AgentRed | please have a look at it |
08:07.29 | mog_home | it looks like you had some issue with adsi not being fully installed |
08:07.43 | mog_home | is this latest head? |
08:07.50 | AgentRed | ya |
08:08.25 | mog_home | let me do update |
08:08.39 | AgentRed | should I go ahead installing the latest ADSI |
08:08.39 | AgentRed | ? |
08:08.46 | mog_home | adsi is part of asterisk |
08:08.52 | mog_home | do you have zaptel hardware? |
08:08.57 | AgentRed | no |
08:09.10 | Qwell | heh, res_config_odbc is failing right now on head |
08:09.14 | mog_home | hmm i think i know whats wrong |
08:09.25 | AgentRed | one more thing is Iam getting errors compling the zaptel |
08:10.11 | mog_home | well if you remove zaptel.h from your libraries |
08:10.15 | mog_home | and rebuild it should work |
08:10.21 | mog_home | and build without adsi stuff |
08:10.40 | AgentRed | is it because of that I am getting this error |
08:10.42 | AgentRed | ? |
08:10.55 | Qwell | hmm, what is res/ on the bug tracker? |
08:11.00 | mog_home | <PROTECTED> |
08:11.02 | mog_home | get rid of that |
08:11.05 | mog_home | and rebuild asterisk |
08:11.11 | mog_home | make clean make asterisk |
08:11.18 | AgentRed | alright |
08:11.27 | mog_home | goodnight all good people |
08:11.37 | AgentRed | Good night mog _home |
08:11.41 | AgentRed | thanks a lot |
08:13.39 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
08:14.05 | *** join/#asterisk sjobeck99 (n=sjobeck9@london.sjobeck.com) |
08:17.41 | Qwell | ok, now bed |
08:21.19 | ComputerWarm | anyone here use A2Billing? |
08:21.41 | ComputerWarm | or maybe better known as ASterisk CallingCard Platform |
08:21.48 | *** join/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com) |
08:21.57 | delox99 | hi all |
08:22.05 | phara0h | anyone know how to get MOH Native going on v1.2 ? |
08:22.32 | delox99 | can someone explain what is Dundi exactly? |
08:22.49 | phara0h | www.dundi.com |
08:24.09 | delox99 | i m looking for a way to connect to pstn lines |
08:24.27 | delox99 | anyone know of a pstn lines provider? |
08:24.51 | phara0h | where do you live? |
08:25.02 | delox99 | canada ottawa |
08:25.18 | tclark_ | 3rd world country |
08:25.52 | shido6 | i used to visit ottawa |
08:26.09 | delox99 | i want to have my asterisk to conenct to some pstn switch that can give me good prices for long distances |
08:26.17 | phara0h | http://www.voipbuster.com/en/index.html <--- not free to canada but ... |
08:26.25 | delox99 | where you from shido? |
08:27.51 | delox99 | i mean i want to provide clients with good prices for landline calls |
08:27.52 | tclark_ | shido6: is hding out near winsor, kitchener corridor some where |
08:28.03 | shido6 | yep |
08:28.38 | shido6 | I used to have a place near vanier parkway |
08:28.49 | ComputerWarm | Hey anyone know of any test numbers for canada. i am trying to test out this new provider find out where all they do not cover |
08:28.50 | emdub | anyone have a cisco 7960g? man what a pain in the neck configuring this thing is |
08:29.02 | shido6 | its not that bad :) emdub |
08:29.14 | emdub | ugh can i poke your brain some? |
08:29.22 | shido6 | shoot |
08:29.24 | emdub | maybe its just the factory firmware sucks |
08:29.34 | shido6 | i run 7.4 here |
08:30.02 | emdub | well, i plugged it in, it has an ip, etc... i hit **# to unlock it and go under configuration and try to edit the tftp server so i can upgrade it and then it says "this key is not active here" or something |
08:30.09 | emdub | every time i try to change the contrast it never saves, etc |
08:30.10 | emdub | wtf! |
08:30.25 | shido6 | you sure its unlocked? |
08:30.32 | shido6 | if it is - then build a conf file |
08:30.39 | emdub | the little lock ucon under network config shows unlocked |
08:30.48 | shido6 | and force it to look for a tftp server in you dhcp server settings |
08:30.51 | emdub | problem is i cant configure my dhcp server to allocate a tftp server ] |
08:30.53 | emdub | ya, cant |
08:31.04 | shido6 | then build a little network of your own |
08:31.07 | emdub | ugh |
08:31.07 | shido6 | grab a switch |
08:31.13 | emdub | you have got to be kidding me |
08:31.27 | shido6 | dood its not that hard :) |
08:31.28 | emdub | man how useless |
08:31.37 | emdub | thats not the point |
08:31.43 | clive- | lol, in south africa you can cal teh police for an armed robbery and they dont arrive |
08:31.44 | emdub | i should be able to change the tftp server ip from the phone! |
08:31.45 | shido6 | it will take 6 minutes |
08:31.46 | clive- | oops |
08:31.47 | clive- | :) |
08:32.22 | h3x | get an office next door to krispy kreme, and you will get them to show up for suspcious activity |
08:32.55 | SarahEmm | .ckear |
08:33.23 | emdub | i should go wake up my cisco sales guy ;) |
08:33.26 | emdub | hehe |
08:33.28 | emdub | or call tac |
08:33.30 | emdub | fix my phone! |
08:33.47 | delox99 | will dundi help me connect to pstn around the world? |
08:33.59 | delox99 | sry for m newbieness |
08:35.34 | ComputerWarm | delox99 have you tried www.didx.org ? i have seen it on the biz list alot |
08:35.35 | delox99 | i need to find a way to redirec sip calls to pstn lines in other cities |
08:35.54 | ComputerWarm | oh wait sorry you are looking to buy termination minutes? |
08:36.48 | delox99 | termination minute? |
08:37.19 | delox99 | probably yes |
08:37.40 | ComputerWarm | sorry i can`t recommend a cheap provider. i have heard nufone is good |
08:57.35 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
09:15.29 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
09:19.08 | *** join/#asterisk bzbw (n=wlwzhang@24-205-27-223.dhcp.nrwl.ca.charter.com) |
09:19.22 | bzbw | anyone know why I got following error? |
09:19.23 | bzbw | <PROTECTED> |
09:19.23 | bzbw | Nov 8 17:05:25 WARNING[4489]: chan_zap.c:890 zt_open: Unable to specify channel 1: No such device or address |
09:19.23 | bzbw | Nov 8 17:05:25 ERROR[4489]: chan_zap.c:6650 mkintf: Unable to open channel 1: No such device or address |
09:19.23 | bzbw | here = 0, tmp->channel = 1, channel = 1 |
09:19.24 | bzbw | Nov 8 17:05:25 ERROR[4489]: chan_zap.c:10030 setup_zap: Unable to register channel '1-3' |
09:19.26 | bzbw | Nov 8 17:05:25 WARNING[4489]: loader.c:403 __load_resource: chan_zap.so: load_module failed, returning -1 |
09:19.28 | bzbw | Nov 8 17:05:25 WARNING[4489]: loader.c:543 load_modules: Loading module chan_zap.so failed! |
09:19.45 | bzbw | I use 2 TDM400 |
09:20.55 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
09:22.08 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
09:22.21 | bzbw | anyone here that can help me on my issue? |
09:22.24 | bzbw | <PROTECTED> |
09:22.24 | bzbw | Nov 8 17:05:25 WARNING[4489]: chan_zap.c:890 zt_open: Unable to specify channel 1: No such device or address |
09:22.24 | bzbw | Nov 8 17:05:25 ERROR[4489]: chan_zap.c:6650 mkintf: Unable to open channel 1: No such device or address |
09:22.24 | bzbw | here = 0, tmp->channel = 1, channel = 1 |
09:22.24 | bzbw | Nov 8 17:05:25 ERROR[4489]: chan_zap.c:10030 setup_zap: Unable to register channel '1-3' |
09:22.26 | bzbw | Nov 8 17:05:25 WARNING[4489]: loader.c:403 __load_resource: chan_zap.so: load_module failed, returning -1 |
09:22.28 | bzbw | Nov 8 17:05:25 WARNING[4489]: loader.c:543 load_modules: Loading module chan_zap.so failed! |
09:22.28 | SarahEmm | gah |
09:22.30 | SarahEmm | please don't paste that much |
09:22.32 | SarahEmm | use a pastebin1 |
09:22.40 | SarahEmm | and don't repeat.... |
09:22.44 | bzbw | k |
09:22.49 | SarahEmm | it sounds like it can't find your card |
09:22.53 | SarahEmm | does lspci show the card? |
09:22.58 | bzbw | k |
09:23.32 | bzbw | got a bunch of output, not sure what to look at. |
09:23.41 | SarahEmm | paste it into a pastebin.ca pastebin pls |
09:23.44 | SarahEmm | and paste the URL in here |
09:24.16 | *** join/#asterisk bon (n=bon@voip.in.radiolan.sk) |
09:24.22 | bzbw | what do u mean by paste the URL? It's from my linux command line. |
09:24.39 | lehel | ~pastebin |
09:24.41 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
09:24.57 | *** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net) |
09:25.01 | morale | calgary is getting very sleepy... |
09:25.04 | SarahEmm | bzbw: copy the output of lspci, and go to pastebin.ca and paste it in |
09:25.05 | lehel | bzbw: learn using it |
09:25.07 | bzbw | thx |
09:25.09 | SarahEmm | morale: heh :) |
09:25.15 | SarahEmm | morale: Toronto is mildly tired. |
09:25.16 | *** join/#asterisk voipjoy (n=root@1.fix.netvision.net.il) |
09:25.29 | voipjoy | anybody can advice what is this? Nov 8 11:25:15 voice kernel: Got pulse digit 13 on TE4/0/1/15??? |
09:25.34 | morale | i have my emergency red bull in the car |
09:26.15 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
09:26.16 | zoa | yo |
09:26.19 | zoa | Ha Ho |
09:26.39 | lehel | 'llo |
09:26.48 | bon | hello |
09:26.54 | bon | i'm having problems with extensions |
09:26.58 | bon | http://pastebin.com/421534 |
09:27.01 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
09:27.06 | bon | that is a user registering with mysql db no problem |
09:27.15 | bon | but when i call that number,it doesn't work :( |
09:27.57 | bzbw | SarahEmm: here is the link for lspci: http://pastebin.ca/27967 |
09:28.41 | SarahEmm | bzbw: is the only telephony card in the box the TDM400p? |
09:28.53 | bzbw | yes, there are 2 of them |
09:28.54 | bon | even tough it says Dial(SIP/${EXTEN}) |
09:28.55 | bon | :( |
09:30.10 | SarahEmm | bzbw: hrm, yeah i do see them there.. pastebin your /etc/zaptel.conf |
09:33.19 | bzbw | thx SarahEmm, here it is http://pastebin.ca/27968, it should be the default one |
09:33.42 | *** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net) |
09:33.46 | bzbw | SarahEmm, I thought we don't use /etc/zaptel.conf any more, right? |
09:34.45 | SarahEmm | err |
09:34.54 | SarahEmm | what has it been replaced with? |
09:34.57 | SarahEmm | last i heard it was still used |
09:35.28 | SarahEmm | yeah, you need to configure that file bzbw |
09:35.32 | morale | you can delete your /vmlinuz and/or /boot/vmlinuz - i hear those aren't used anymore either. |
09:35.34 | SarahEmm | then run ztcfg -vvvvvv to configure the card |
09:35.36 | SarahEmm | lol morale |
09:36.07 | bzbw | emm, I thought it stated in the zaptel REAME |
09:36.09 | bon | morale :) |
09:36.41 | newl | SarahEmm: *cracks whip* got that ringer done yet? |
09:37.09 | voipjoy | i have partly E1 channel only 14 instead of 30! how i disable the not used chanel that zapata will not handle it and not manage it? |
09:37.17 | SarahEmm | newl: no :/ not much time lately for that |
09:37.21 | bzbw | SarahEmm: Channel Map: 0 channels configured. |
09:37.35 | SarahEmm | bzbw: that's because you didn't configure the zaptel.conf file |
09:37.46 | SarahEmm | bzbw: you need to edit it and put in the right config before you run ztcfg |
09:37.49 | newl | SarahEmm: Eh :) |
09:38.00 | SarahEmm | newl: are you actually interested in it? |
09:38.18 | bzbw | SarahEmm: thx. |
09:38.39 | SarahEmm | bzbw: there's a page on the wiki that might help if you're having trouble with the zaptel.conf config :) |
09:39.25 | newl | SarahEmm: no, but someone has to prod you along hehe |
09:39.38 | newl | besides, I thought it was a good idea. |
09:40.12 | voipjoy | SarahEmm: could you tell me please - how i can disable chanell in zaptel.conf ? |
09:40.45 | SarahEmm | voipjoy: disable channel? what do you mean? |
09:40.47 | SarahEmm | newl: lol |
09:40.59 | SarahEmm | newl: well right now there's little use for it because * doesn't have any off the shelf TTY apps :) |
09:41.04 | SarahEmm | which is another thing i need to finish |
09:43.03 | gst | hmm... since the upgrade to beta2 i get "Avoided initial deadlock" msgs and then asterisk kills the (sip) channel. i have seen this messages before but the calls worked fine nevertheless before. |
09:43.13 | gst | are there any known current problems about this? |
09:46.00 | voipjoy | SarahEmm: i getting E1 line (30 lines) i use only 14, and receive some "noise" on other 16 lines...I like to disable them...that zapata driver will ignore them |
09:46.25 | fourcheeze | does anyone know if a sipura spa-3000 can forward an incoming call on its fxo port to a sip url? |
09:46.51 | fourcheeze | in other words operate in reverse |
09:47.15 | fourcheeze | or is the fxo port purely a backup in case the net goes down? |
09:47.47 | SarahEmm | voipjoy: oh... no idea. |
09:49.30 | *** join/#asterisk remibreval (i=Remek@pro75-3-82-234-175-208.fbx.proxad.net) |
09:49.51 | remibreval | Hello Everyone ! |
09:50.28 | remibreval | I'm still getting into troubles to answer a ringing PSTN call. THe communication goes to the IAW service provider when I answer. This is the post : http://pastebin.ca/27885 |
09:51.59 | bzbw | SarahEmm: I just add one line in zaptel.conf: fxols=1-8, and got error: ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
09:52.18 | bzbw | SarahEmm: WiKi won't help. |
09:52.22 | SarahEmm | bzbw: one sec. |
09:52.30 | SarahEmm | what pastebin was yours again bzbw |
09:52.31 | SarahEmm | ? |
09:52.45 | SarahEmm | ssh tonkinese |
09:53.21 | SarahEmm | bzbw: do you have FXS or FXO modules on the card? |
09:53.21 | bzbw | SarahEmm: you mean my previous zaptel.conf? |
09:53.24 | SarahEmm | yeah |
09:53.43 | bzbw | SarahEmm: only FXO |
09:53.52 | bzbw | SarahEmm: I believe |
09:54.08 | SarahEmm | err.. okay, then you need to say fxsks=1-8 |
09:54.10 | SarahEmm | not fxols |
09:54.31 | SarahEmm | FXO interfaces use FXS signalling, FXS interfaces use FXO signalling. |
09:54.36 | SarahEmm | welcome to the strange world of telco ;) |
09:54.48 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
09:54.53 | bzbw | SarahEmm: thx, will try now. |
09:59.50 | remibreval | nobody has a basic conf to accept an incomming call ?? |
10:00.15 | SarahEmm | remibreval: err, you're just trying to accept a call? you've looked at example dialplans in the wiki? |
10:02.23 | delox99 | i want to use a US satelite to send and receive large amount of data. Where should i look? |
10:02.48 | SarahEmm | err |
10:02.53 | SarahEmm | large amount of data to the internet? |
10:03.18 | delox99 | yes |
10:03.36 | *** join/#asterisk Alystair (i=Alystair@CPE000d88a7a3b5-CM00407b8794db.cpe.net.cable.rogers.com) |
10:03.42 | Alystair | Oh hey awesome |
10:04.04 | SarahEmm | delox99: www.starband.com |
10:04.15 | SarahEmm | delox99: latency on sat is awful, but it works if you're out in the middle of nowhere |
10:04.32 | Alystair | I need some help here :( |
10:04.37 | SarahEmm | Alystair: with? |
10:04.45 | Alystair | is there a generic site you can reffer me to? |
10:04.51 | SarahEmm | www.voip-info.org |
10:04.53 | SarahEmm | ~wiki |
10:04.57 | SarahEmm | hrm |
10:05.00 | SarahEmm | ~rtfw |
10:05.01 | jbot | somebody said rtfw was http://www.voip-info.org, the only place to get any real answers about * and it's many, many uses. |
10:05.26 | Alystair | is this mainly for personal or business info? |
10:05.32 | SarahEmm | Alystair: either |
10:05.34 | delox99 | what do you mean in the middle of nowhere? |
10:05.39 | Alystair | awesomepossum, thanks Sarah! |
10:05.50 | SarahEmm | delox99: err, why are you looking at sat as opposed to land-based ways of data transfer? |
10:05.58 | SarahEmm | Alystair: np :) |
10:06.00 | bzbw | SarahEmm: it works, thx!! |
10:06.24 | delox99 | because it covers a larger teritory |
10:06.30 | SarahEmm | delox99: okay... |
10:06.42 | SarahEmm | delox99: i meant as in 'it works anywhere' because it's kind of a last-resort technology |
10:06.50 | SarahEmm | you wouldn't use it if you had some kind of land-based internet available |
10:06.50 | delox99 | we need to get send and receive data in the middle of mexico |
10:06.54 | SarahEmm | okay |
10:06.57 | SarahEmm | look at starband |
10:07.04 | SarahEmm | bzbw: np! :) |
10:07.08 | SarahEmm | woo, i'm useful. :) |
10:07.17 | delox99 | ok ill have a look |
10:07.29 | delox99 | is it better to use some kind of wifi? |
10:07.40 | SarahEmm | err, wifi is short-distance tho |
10:07.46 | SarahEmm | what would you connect to with wifi? |
10:07.55 | *** join/#asterisk Abbas (n=Abbas@203.81.216.47) |
10:08.15 | delox99 | it s to terminate voip lines |
10:08.24 | SarahEmm | delox99: you don't want sat for voip lines. |
10:08.29 | SarahEmm | your latency will be awfl |
10:08.34 | SarahEmm | much like standard telco over sat |
10:08.40 | delox99 | sent from US to Mexico |
10:08.56 | SarahEmm | you're looking at 1s or so RTTs |
10:09.02 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
10:09.03 | delox99 | what? |
10:09.15 | delox99 | sry im pretty newbie... |
10:09.22 | SarahEmm | delox99: sat internet has about 1 second or so round trip time on packets.. high latency. |
10:09.32 | SarahEmm | speech takes too long to get to the other end over sat, it makes calls really hard to do |
10:09.33 | delox99 | ok |
10:09.40 | SarahEmm | you likely don't want to run VoIP over it. |
10:09.49 | *** join/#asterisk duckz (n=duckz@193.192.46.26) |
10:10.02 | delox99 | ok so what would you suggest |
10:10.48 | delox99 | ? |
10:11.00 | SarahEmm | delox99: you can't get any kind of PRIs or anything? |
10:11.03 | fourcheeze | delox99: two tin cans and a long piece of string? |
10:11.40 | delox99 | we cant aford that length of string! |
10:11.50 | emdub | man, 7960's with sccp firmware on them is enough to make you commit suicide |
10:11.55 | fourcheeze | delox99: exactly how long are you talking? |
10:11.57 | emdub | atleast when trying to upgrade them |
10:11.58 | emdub | heh |
10:11.59 | SarahEmm | delox99: then you don't want to look at what sat costs :) |
10:12.07 | fourcheeze | delox99: ok, not "exactly" |
10:12.08 | delox99 | ah ok |
10:12.17 | SarahEmm | delox99: you can't get PRIs there? |
10:12.18 | fourcheeze | 1 mile, 10 miles, 100 miles, 1000 miles? |
10:12.55 | *** join/#asterisk Alystair (i=Alystair@CPE000d88a7a3b5-CM00407b8794db.cpe.net.cable.rogers.com) |
10:13.13 | delox99 | i would have to check first |
10:13.30 | delox99 | but im afraid i could be expensive |
10:13.36 | SarahEmm | err |
10:13.36 | delox99 | it* |
10:13.38 | SarahEmm | sat will be expensive too |
10:13.43 | SarahEmm | and low quality |
10:13.48 | *** join/#asterisk MatsK (n=mk@55.80-203-80.nextgentel.com) |
10:14.02 | delox99 | because Telmex is too much of a monopol |
10:14.28 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
10:15.00 | cjk | hi, i know it does not really belong here. but did anyone here use asterisk together with mysql together with federated tables over a long distance ip link? |
10:15.57 | h3x | federated tables? |
10:16.11 | fourcheeze | delox99: point to point microwave link with line of sight can go a fair way, but it depends what you mean by "middle of nowhere" |
10:16.24 | cjk | h3x: yes, tables stored on a remote mysql server |
10:17.07 | delox99 | sarahemm was saying that to use a sat i would need to be in the middle of nowhere |
10:17.29 | delox99 | microwave link can travel how far? |
10:17.57 | SarahEmm | err, i didn't say you *had* to be in the middle of nowhere |
10:18.05 | SarahEmm | i said you really want to AVOID using sat. |
10:18.13 | SarahEmm | so the only reason you WOULD use it is if you were in the middle of nowhere. |
10:18.14 | h3x | oh |
10:18.21 | SarahEmm | adding 1+ seconds of latency onto your voice conversation makes it hard to tlka |
10:18.53 | h3x | Lots of "private line" services provided by telcos goes over microwave anyway without you knowing it |
10:19.08 | h3x | microwave dosent slow it down any more than wire or fiber |
10:19.17 | delox99 | ah i thought ou said "hEmm> adding 1+ seconds of latency onto your voice conversation makes it hard to tlka |
10:19.17 | delox99 | <h3x> |
10:19.25 | delox99 | hEmm> adding 1+ seconds of latency onto your voice conversation makes it hard to tlka |
10:19.25 | delox99 | <h3x> |
10:19.32 | h3x | microwave dosent add no damn 1000ms |
10:19.34 | h3x | satellite does |
10:19.37 | delox99 | oops sry im newbie with mirc |
10:20.01 | delox99 | trying to paste sarahemm answer |
10:20.13 | delox99 | but pressing ctrl-v does: |
10:20.14 | delox99 | hEmm> adding 1+ seconds of latency onto your voice conversation makes it hard to tlka |
10:20.14 | delox99 | <h3x> |
10:20.15 | delox99 | hEmm> adding 1+ seconds of latency onto your voice conversation makes it hard to tlka |
10:20.15 | delox99 | <h3x> |
10:20.15 | delox99 | hEmm> adding 1+ seconds of latency onto your voice conversation makes it hard to tlka |
10:20.16 | delox99 | <h3x> |
10:20.59 | delox99 | ah i see as if i would be in the middle of nowhere |
10:21.35 | SarahEmm | delox99: umm...... if you're having trouble with mirc, are you sure you're ready to tackle a VoIP deployment away from the city? |
10:21.37 | h3x | what difference does that make |
10:21.42 | h3x | haha |
10:21.47 | h3x | hes gonna do TCoIP |
10:21.50 | h3x | Tin Can over IP |
10:21.53 | SarahEmm | :) |
10:21.59 | delox99 | hehe |
10:23.03 | delox99 | i never really learned all the little mirc tricks like real chat machines =) |
10:23.16 | h3x | chat machines? |
10:23.23 | h3x | are you in afghanistan or what |
10:23.30 | delox99 | no in Canda |
10:23.33 | delox99 | Quebec |
10:23.38 | h3x | I figured you would say romania |
10:24.08 | delox99 | i just dont speak english as good as i speak french |
10:24.24 | h3x | i worked in quebec for a little while |
10:24.28 | h3x | like, 3 weeks |
10:24.42 | delox99 | where you from? |
10:24.48 | h3x | i live in las vegas |
10:24.58 | delox99 | hehe |
10:25.09 | delox99 | where were you working? |
10:25.14 | h3x | when i was up there, i was fixing some peoples porn sites and gambling sites |
10:25.22 | h3x | at teleglobe colocation |
10:25.30 | delox99 | nice |
10:26.02 | delox99 | fixing server issues or coding issues? |
10:26.25 | h3x | well all of their stuff was screwed up |
10:26.33 | h3x | they had a bunch of russian programmers up there |
10:26.37 | h3x | i was just fixing sysadmin problems |
10:26.49 | delox99 | ok |
10:27.13 | h3x | i got a new russian girl every week |
10:27.13 | h3x | haha |
10:27.33 | delox99 | on pictures? |
10:27.38 | h3x | no i mean in person |
10:27.44 | delox99 | hehe |
10:27.51 | delox99 | live from russia |
10:28.05 | h3x | they all move to canada coz its hard to get a visa here |
10:28.05 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
10:28.10 | delox99 | they were shipping em by fedex or what? |
10:28.22 | h3x | and then eventually they move to the US after they have canadian citizenship |
10:28.34 | delox99 | ah yeah i see |
10:28.46 | h3x | well, everybody except my friends that were running that business |
10:29.04 | h3x | the guy's wife was a smart ass bitch to the immigration folks |
10:29.08 | h3x | and got them booted out to russia |
10:29.13 | h3x | i havent heard from him since |
10:29.29 | delox99 | he was russian too? |
10:29.37 | h3x | yeah |
10:29.59 | h3x | before i knew him, he got rich running zooporn.com |
10:30.01 | h3x | hahahaha |
10:30.27 | delox99 | and you met him on mirc? or while surfing one of his sites? |
10:30.36 | h3x | but he screwed up... he registered the domain in his old college professor's name coz he didnt want people tracking him down |
10:31.02 | h3x | so when he moved from uunet to teleglobe, he simultaenously lost his email account the domain was registered under, so he couldn't move it over |
10:31.32 | h3x | no, somehow he was a partner of some guys i was working with here in vegas on some other projects |
10:32.08 | voipjoy | i have partly E1 channel only 14 instead of 30! how i disable the not used chanel that zapata will not handle it and not manage it? |
10:32.40 | delox99 | bad for him |
10:33.04 | h3x | heh |
10:33.14 | delox99 | yeah so microwave would do it |
10:34.16 | *** join/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net) |
10:34.27 | mover | hi |
10:34.41 | mover | anymone here experienced wit t38 stuff in HEAD? |
10:34.46 | delox99 | we want to bring voip termination in mexico without using telmex infrastructure |
10:35.42 | delox99 | we are just looking into some solutions so we can hire more professional people to do the job better than me |
10:35.54 | h3x | hahahahha |
10:36.11 | h3x | that is a good way to get killed |
10:36.19 | delox99 | hehe |
10:36.25 | delox99 | i know |
10:36.31 | h3x | telmex gets really pissed off at grey market minutes |
10:36.44 | delox99 | hehehe |
10:36.58 | delox99 | you didnt see me coming? |
10:37.08 | SarahEmm | delox99: wait... termination? |
10:37.08 | delox99 | the market there is just incredible |
10:37.11 | *** join/#asterisk Marcel-AS16215 (i=Marcel-A@gic-msg-exc-01.genotec.ch) |
10:37.19 | SarahEmm | delox99: where are you dropping it onto the PSTN then, if you're not using telmex? |
10:37.31 | h3x | TCoIP |
10:37.33 | h3x | im tellin ya |
10:37.38 | h3x | them bean counters love it |
10:37.52 | Marcel-AS16215 | Hi to all, anyone a idee to fix this -> Nov 8 11:08:24 NOTICE[8180]: chan_sip.c:10270 handle_request_invite: Failed to authenticate user Marcel<sip:993@192.168.254.230>;tag=AIC99B37F7-ECB1BE2E066724AC ? |
10:38.29 | SarahEmm | Marcel-AS16215: Genotech Internet Consulting? :) |
10:38.33 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
10:38.36 | KriS83 | Hi |
10:38.38 | SarahEmm | Marcel-AS16215: sounds like your authentication info is wrong, password or such |
10:38.51 | delox99 | we will use pstn for delivery but we ll see once there |
10:39.07 | delox99 | first lets bring the pipe =) |
10:39.29 | SarahEmm | delox99: err... okay. you'll still have to use telmex for the PSTN connection :) |
10:39.32 | Marcel-AS16215 | SarahEmm yes :-) |
10:39.39 | delox99 | we could distribute using wifi or something once there |
10:39.39 | SarahEmm | and i'm not sure where you're going to get a pipe :) |
10:39.45 | h3x | theres plenty of competition in mexico already |
10:39.49 | h3x | you just need to go shopping |
10:39.52 | SarahEmm | delox99: err.. but you still have to connect to the PSTN, wifi won't help you there |
10:40.04 | SarahEmm | Marcel-AS16215: :) |
10:40.23 | delox99 | true i need to make more research |
10:40.29 | h3x | you havent been on #cisco apparently |
10:40.30 | h3x | heh |
10:40.31 | delox99 | i was starting here =) |
10:40.49 | SarahEmm | h3x: heh, i used to hang out there all the time :) |
10:40.53 | h3x | dammit irc is always a last resort! :) |
10:41.07 | h3x | SarahEmm: Yeah i ran across your nick in an irc log when i was going a google search |
10:41.09 | Marcel-AS16215 | SarahEmm but wehre is my Password wrong ? i think all my settings sut be okay, this comes only if a add tow connection my my Company PBX |
10:41.12 | h3x | but i dont remember if it was that or some other program |
10:41.14 | SarahEmm | h3x: heh, i show up all the time :) |
10:41.14 | h3x | like |
10:41.21 | h3x | h-sphere maybe |
10:41.23 | h3x | ummmmm |
10:41.38 | h3x | what the hell was it was looking for and you replied to somebody and i was gonna ask you about it... |
10:41.48 | h3x | er I was looking for |
10:41.58 | SarahEmm | h3x: oh? |
10:42.04 | *** join/#asterisk ComputerWarm (n=dan@rddrpx29-port-20.dial.telus.net) |
10:42.37 | delox99 | i ve been looking into asterisk stuff for only like 4 months but i see he big potential |
10:42.50 | h3x | Hmmm Maybe it was mythtv? |
10:42.53 | Marcel-AS16215 | SarahEmm can i can come quick privat to discus this maybe you can give me the right hint that i dont see if i post you my Settings without username/password. |
10:43.02 | ComputerWarm | Question anyone here using A2Billing? |
10:43.52 | delox99 | i m running an Asterisk server and already have a couple clients using sip phones |
10:44.00 | lehel | ComputerWarm, i'm trying to.. |
10:44.22 | delox99 | now i m looking for a way to connect my server to pstn lines |
10:44.23 | ComputerWarm | lehel have you been able to get logged-in? |
10:44.31 | lehel | yap |
10:44.32 | delox99 | i found that today in here |
10:44.45 | ComputerWarm | lehel oh... i wonder what i did wrong then |
10:44.47 | SarahEmm | h3x: could have been, i love mythtv :) |
10:45.00 | SarahEmm | Marcel-AS16215: umm, can't really help one-on-one right now sorry. have other stuff going on |
10:45.02 | SarahEmm | (am at work) |
10:45.13 | lehel | ComputerWarm, you r reading the pdf document? |
10:45.15 | SarahEmm | delox99: get a PRI :) |
10:45.25 | ComputerWarm | lehel no i have been reading the html doc |
10:45.39 | h3x | i cant get my damn pvr-350 to work |
10:45.47 | delox99 | i would like to but the price scares me :/ |
10:45.56 | h3x | when the kernel module is loaded it bombs dmesg with tons of APIC errors |
10:46.00 | h3x | ive got HT on |
10:46.15 | lehel | ComputerWarm, it is the same huh;) which dbtype u r using? [pgsql or mysql] ? |
10:46.27 | remibreval | sarahEmm, yes I look at dialplan exemples in Wiki and I made http://pastebin.ca/27885, but still when I answer, It hang off instead of answering :-( |
10:46.27 | delox99 | i ll use a provider instead |
10:46.29 | ComputerWarm | psql |
10:46.32 | ComputerWarm | pgsql |
10:46.55 | Marcel-AS16215 | SarahEmm okay |
10:47.19 | delox99 | what prices should i look for when shopping for voip termination resellers for US/Canada? |
10:47.20 | lehel | ComputerWarm, i succeded with mysql, never tried with pgsql |
10:47.34 | delox99 | i saw 0.02US / min |
10:47.38 | delox99 | is that good |
10:47.43 | delox99 | or expensive |
10:48.04 | *** join/#asterisk TK9 (n=Administ@p54B29019.dip0.t-ipconnect.de) |
10:48.22 | ComputerWarm | lehel maybe i should be trying mysql. |
10:48.26 | *** part/#asterisk TK9 (n=Administ@p54B29019.dip0.t-ipconnect.de) |
10:48.27 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
10:48.30 | h3x | SarahEmm: Well you win, you have a bigger mess on your desk than me |
10:49.05 | ComputerWarm | delox99 that would depend on how much volume u have |
10:49.08 | lehel | ComputerWarm, ' should work as well with pgsql too |
10:49.32 | ComputerWarm | lehel i think once i get this error figured out in there script. i should be on my way. |
10:50.19 | Alystair | Wait so are all these systems linux only? :O |
10:50.28 | mmmToop | hi...not sure if you guys can help...we are using swissvoice ip10s's & it sounds like we are talking under water |
10:50.34 | mmmToop | they were working fine & then stop working? |
10:50.37 | mmmToop | any thoughts? |
10:50.52 | SarahEmm | h3x: lol. old picture, it's better now :) |
10:51.03 | mmmToop | it has happened to two phones now...firmware? |
10:51.12 | h3x | that sucks |
10:51.21 | h3x | god damn it, there was a specific issue you solved on mythtv |
10:51.22 | h3x | and i cant find it |
10:52.30 | Alystair | Ok this is rediculous, 90% of the offsite links from the informational wiki are half ass pages with adverts all over them and the content isn't what it is supposed to be |
10:53.16 | SarahEmm | h3x: ahh, i use a bttv card and software-encode |
10:53.26 | cryzeck | < |
10:53.28 | ComputerWarm | i don`t know Alystair the wiki works for me... for the most part |
10:53.38 | h3x | yeah my bttv card works fine |
10:53.40 | SarahEmm | my current plan is to move to a PVR-500 as soon as i can get an external box to decode captions before it hits the PVR-500, as the card won't do it for me |
10:54.11 | h3x | knoppmyth is garbage |
10:54.22 | h3x | i ended up using fedora 4 |
10:54.32 | ptiggerdine | remibreval, that's one huge dial plan |
10:54.35 | ptiggerdine | nice work though |
10:54.41 | Alystair | ComputerWarm, no I mean it links to all these offsite places for general information |
10:54.53 | h3x | i think im gonna switch to via epia with a couple bttv cards :P |
10:54.59 | ComputerWarm | oh ok. is there a problem you are having? |
10:55.55 | mmmToop | so nobody has any ideas about the swissvoice phones why I sound like I am 10 000m under the sea? |
10:55.57 | SarahEmm | h3x: heh, you'll never get enough CPU for it :) |
10:56.02 | SarahEmm | h3x: that's why i only hsave one tuner now |
10:56.14 | h3x | the via has mpeg4 acceleration built in |
10:56.19 | h3x | theres a dual via cpu mobo |
10:56.21 | SarahEmm | err |
10:56.24 | SarahEmm | mpeg4 decoding |
10:56.28 | SarahEmm | not encoding, right? |
10:56.31 | h3x | i thought it did both? |
10:56.36 | Alystair | Yeah, where can I find a link to a no-crap website which describes setting up a VOIP system for small companies (everything from outbound calling to a PBX like system)? :\ |
10:56.39 | SarahEmm | myth doesn't support it, even if it does :) |
10:56.43 | SarahEmm | and you'll max out your PCI bus |
10:56.44 | h3x | yeah it does |
10:56.47 | SarahEmm | err |
10:56.48 | SarahEmm | it does? |
10:56.51 | h3x | yes |
10:56.51 | SarahEmm | for encoding? |
10:56.54 | h3x | unichrome |
10:57.08 | h3x | they have special setup instructions for it |
10:57.13 | SarahEmm | url? |
10:57.27 | h3x | which distro? heh |
10:57.34 | SarahEmm | n/m |
10:57.39 | rking | Alystair: well, that's a very good question |
10:58.20 | h3x | im trying to figure out some way to make these things "cheaper" so i can sell them |
10:58.42 | Alystair | and even then, when I try clicking on a website link like packet8's, when it should go to packet8.com (duh) they go to some reseller page or something (not direct from wiki) |
10:58.44 | rking | Alystair: if such a thing doesn't exist, it would be a valuable contribution to the voip-info.org wiki (I want such a document myself) |
10:58.59 | SarahEmm | h3x: ahh :) |
10:59.01 | h3x | I guess PVR-500 is a good option |
10:59.04 | SarahEmm | yeah |
10:59.07 | SarahEmm | i have one, works great |
10:59.10 | SarahEmm | except no CC support |
10:59.10 | h3x | becuase most mini-itx cases only have one pci slot thats usuable |
10:59.15 | Alystair | ok, here's a question that should work here then |
10:59.15 | SarahEmm | yeah, mine does |
10:59.20 | h3x | with some place to fudge that extra bracket in |
10:59.39 | Alystair | If I setup asterix at the office, can I plug it in to our own backend system (php/mysql) to give people information over the phone? |
10:59.51 | SarahEmm | ure |
10:59.52 | h3x | its times like this |
10:59.55 | SarahEmm | err sure even Alystair |
11:00.00 | h3x | i wish mark didnt release asterisk to the world |
11:00.00 | h3x | hah |
11:00.17 | Alystair | I would need someone to code some sort of bridge for me right :\ |
11:01.30 | SarahEmm | Alystair: yep. |
11:01.40 | SarahEmm | obviously, as asterisk can't include support for some custom backend you have ;) |
11:02.17 | Alystair | duh, even a PHB like me knows that :O |
11:02.23 | h3x | it does with some magic_witches_brew.so |
11:02.36 | mutilator | anyone noticed problems with 2.6.14 kernel |
11:02.57 | h3x | Alystair: asterisk dialplan has a MYSQL() command to do database queries |
11:03.05 | h3x | or you can write AGIs using whatever language you want |
11:03.09 | Alystair | cool |
11:03.33 | h3x | mythtvs interface needs some work |
11:03.35 | SarahEmm | <-- NOCstrich |
11:03.38 | emdub | hmm so i finally got my 7960 up and working and its registering properly etc but when i dial an extension (one that just goes to voicemail) i see asterisk playing sounds but hear nothing on the phone |
11:03.43 | *** join/#asterisk ful|work (n=fulgas@209.8.233.205) |
11:03.50 | emdub | same extension works on my soft phone when i call it... what gives? |
11:03.56 | ful|work | hey |
11:04.10 | h3x | i dont understand why they made it so when you exit "watch tv" it quits recording |
11:04.42 | h3x | i guess maybe because its concept of multiple frontend, multiple backend dosent go with the tivo monolithic concept |
11:05.17 | h3x | fast forward is broken |
11:05.22 | h3x | when it hits the end |
11:06.17 | Alystair | guh |
11:06.24 | Alystair | where are the voip providers in Canada :\ |
11:06.32 | SarahEmm | Alystair: whereabouts? |
11:06.44 | ComputerWarm | Alystair are you looking for termination? or dids |
11:06.48 | Alystair | Ontario |
11:07.16 | SarahEmm | Alystair: what area code(s)? |
11:07.21 | ComputerWarm | are looking to make long distance calls or have people call you via voip? |
11:07.42 | Alystair | Everything. |
11:07.50 | Alystair | including 1-800# potentially |
11:08.04 | SarahEmm | Alystair: i use voctel.. there's lots of providers tho. |
11:08.04 | Alystair | 416 area code |
11:08.06 | SarahEmm | ~rtfw |
11:08.07 | jbot | it has been said that rtfw is http://www.voip-info.org, the only place to get any real answers about * and it's many, many uses. |
11:08.07 | SarahEmm | heh, me too |
11:08.12 | SarahEmm | <-- currently in downtown toronto |
11:08.18 | Alystair | Haha |
11:08.19 | h3x | hahahahhaah rtfw |
11:08.21 | *** join/#asterisk zgor (n=zgor@61.Red-80-36-3.staticIP.rima-tde.net) |
11:08.27 | SarahEmm | the wiki has a provider list |
11:08.28 | Alystair | SarahEmm, 151 Front St.? |
11:08.28 | SarahEmm | in canada |
11:08.31 | zgor | hi people! |
11:08.39 | SarahEmm | Alystair: not quite. one street north and slightly east :) |
11:08.49 | Alystair | Haha alright, I'm in Toronto. |
11:08.50 | SarahEmm | i might be able to see 151 out the window tho :) |
11:09.28 | SarahEmm | :) |
11:09.35 | Alystair | The jist is, my dad's starting a business and I really want to save him cash on the phone stuff (lots of sales folk will be going all over Canada/USA) and make it easy for us to connect to the back end |
11:09.56 | SarahEmm | alright |
11:09.59 | Alystair | I'm already taking care of the office network and I'm interested in putting more hardware on the rack |
11:10.00 | h3x | haha |
11:10.54 | SarahEmm | Alystair: okay. |
11:11.02 | SarahEmm | Alystair: http://www.voip-info.org/wiki/view/VOIP+Service+Providers has a list of providers |
11:11.27 | h3x | man |
11:11.30 | h3x | i am on there |
11:11.33 | h3x | and some guy calls me today |
11:11.36 | h3x | asking for 1 DID |
11:11.44 | h3x | and im like |
11:11.45 | h3x | ^#!^%#!%!#%#! |
11:12.09 | h3x | oh |
11:12.15 | SarahEmm | h3x: if you're on there and don't have a website with info or anything, how are poeple to know? |
11:12.16 | h3x | im just on voip providers b2b and business |
11:12.23 | h3x | my website is in there |
11:12.29 | h3x | and every single page has a contact form |
11:12.40 | SarahEmm | ahh okay h3x |
11:13.16 | h3x | im suprised they dont charge for em |
11:13.27 | h3x | ive gotten about 120 leads off voip-info in the past few months |
11:13.40 | h3x | about 20% of them were major accounts |
11:13.53 | emdub | ok so... more details... calls to a DID number in * which then dials the 7960 via SIP works fine, audio in both directions, but if i dial an outside number from the cisco i hear nothing and the other side can't hear me... same thing if i dial an extension on * which just goes to VM... no sound... i think im using ulaw or something, could that be the problem? |
11:14.02 | h3x | whats really damn funny though is that comm partners runs that site and they didnt list themselves |
11:14.06 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
11:14.08 | puzzled | morning |
11:14.14 | h3x | *derrrr* |
11:14.42 | h3x | im gonna start reselling their did footprint since they dont seem to care to |
11:14.47 | *** join/#asterisk gun84 (n=gun84@60.48.169.13) |
11:15.05 | h3x | comm partners is across the street from me |
11:15.15 | h3x | and im working on getting a gigE connection installed between us |
11:15.58 | h3x | their web site looks like crap too |
11:15.59 | h3x | heh |
11:16.44 | *** part/#asterisk gun84 (n=gun84@60.48.169.13) |
11:18.37 | *** join/#asterisk BerndR (n=broessl@193.83.150.54) |
11:19.02 | mover | anymone here experienced wit t38 stuff in HEAD? |
11:21.23 | *** part/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com) |
11:21.37 | Alystair | h3x, I may not know much about voip but I'm a pretty spiffy graphic/web designer :D |
11:22.06 | Alystair | *... proproses |
11:23.08 | *** join/#asterisk [MUPPETS]Gonzo (i=gonzo@80.69.47.16) |
11:26.33 | ptiggerdine | Alystair, do you deal with coperate identity stuff? |
11:26.46 | ptiggerdine | sorry bout the spelling |
11:31.31 | Alystair | Sure thing |
11:31.52 | Alystair | logos, business cards, letterheads, pamphlet material, etc. |
11:33.14 | Alystair | Though I've been pretty busy lately, gotta make a sign for some guys office, as well as some of those sticky-advertisments-on-glass type of things |
11:33.38 | *** join/#asterisk Cherebrum (i=fucker@fucker.fuckthefuckingfucks.com) |
11:33.47 | Alystair | 2 websites and editing some written material *yawn* |
11:35.27 | *** part/#asterisk Cherebrum (i=fucker@fucker.fuckthefuckingfucks.com) |
11:42.11 | remibreval | Arff, why could Aswering machine and picking up a call macros work from local call and not from PSTN call ? http://pastebin.ca/27885 |
11:42.50 | remibreval | I'm going crazy on what's happening ! |
11:49.00 | *** join/#asterisk coppice (n=chatzill@105.201.17.210.dyn.pacific.net.hk) |
11:50.12 | KriS83 | Anyone here that uses Junghanns.Net quadBRI cards with current * Beta2? |
11:53.50 | *** part/#asterisk lubomier (n=lubomier@217.118.109.179) |
11:55.23 | syle | anyone have a PRI line? |
11:55.31 | syle | or voip site |
11:55.44 | *** join/#asterisk mazza[W] (i=mazzanet@unaffiliated/mazzanet) |
11:57.21 | Alystair | wow Voxter looks good |
11:58.05 | mazza[W] | has anyone managed to successfully compile the intel ipp g729 codec? |
11:58.51 | *** join/#asterisk danzig (n=chatzill@ruc-kj-013.ruc.dk) |
12:00.53 | KriS83 | Anybody got BriStuff compiled for * 1.2.0-beta2? |
12:01.20 | *** join/#asterisk thewiizard (n=nick@host217-34-132-179.in-addr.btopenworld.com) |
12:04.00 | puzzled | KriS83: I haven't heard that the patch has been updated for 1.2b2 |
12:04.29 | puzzled | mazza[W]: yes people have done it before. search voip-info.org for g729 codec |
12:04.31 | KriS83 | :/ |
12:04.48 | KriS83 | puzzled, major differences between beta2 and CVS Version? |
12:05.13 | KriS83 | I mean like is the CVS the next beta3 or maybe even final 1.2.0? |
12:05.13 | puzzled | I would say about 90 commits so I guess yes |
12:05.40 | mazza[W] | puzzled: i was hoping for someone awake in here |
12:05.48 | mazza[W] | i'm getting compile errors :( |
12:06.06 | puzzled | mazza[W]: do the search. there is a site that offers precompiled codecs |
12:06.24 | puzzled | KriS83: afaik cvs is leading up to 1.2 |
12:06.46 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
12:07.27 | *** join/#asterisk alrs (n=lars@dsl092-033-090.lax1.dsl.speakeasy.net) |
12:07.43 | danzig | EHLO * Gurus :-) |
12:08.40 | danzig | Using * as a SIP client, when it gets 'Failed to authenticate on REGISTER', it tries again after 20 seconds. Is there any way to increase this timeout without recompiling? |
12:10.37 | *** join/#asterisk robbie2 (n=rob@CPE-60-231-141-131.qld.bigpond.net.au) |
12:10.42 | KriS83 | puzzled, thank you, I will try the CVS then. |
12:12.43 | KriS83 | Carefull the floor is hard ;) might knock your head ;) |
12:12.52 | puzzled | KriS83: while you are at it, some useful patches for CVS are #5374, #3599, #4252 |
12:13.20 | KriS83 | ok, if you told me where to get those, I will apply :) |
12:13.26 | KriS83 | Sorry, but I'm new to this |
12:14.16 | puzzled | busg.digium.com and #3599 does not work right now but (hopefully) should be updated soon |
12:14.22 | puzzled | ugh, bugs.digium.com |
12:15.05 | robbie2 | my voicemail messages that come through in email have the ends clipped |
12:15.29 | darkskiez | robbie2: probably be your audio player |
12:15.34 | robbie2 | mplayer |
12:15.39 | darkskiez | exactly |
12:15.42 | robbie2 | oh |
12:16.56 | *** join/#asterisk c0w (n=c0w@staff-ns50-3.as25178.net) |
12:16.59 | c0w | hello all. |
12:17.06 | robbie2 | hmm |
12:18.00 | c0w | ?/ |
12:19.47 | robbie2 | yep |
12:19.58 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.7) |
12:20.05 | robbie2 | darkskiez: well thats a bit of a bugger |
12:20.17 | robbie2 | wav file generated on linux box cant play in linux mplayer |
12:20.46 | KriS83 | puzzled, thx |
12:20.48 | darkskiez | i can play in linux, but mplayer is a bit shit |
12:20.52 | darkskiez | i=it |
12:22.40 | c0w | does anyone know about the digium wildcard te4100P cards, and if the new firmware that digium has made is for this card? |
12:22.47 | robbie2 | hmm |
12:23.00 | robbie2 | i see gcc3.0 works a bit better than 2.95 |
12:23.09 | darkskiez | how do you see that? |
12:23.43 | robbie2 | cause its building now |
12:25.06 | thewiizard | got a circuits are busy error |
12:25.18 | puzzled | c0w: ask digium? |
12:25.21 | thewiizard | does that mean that it cant stick the call out thru the provider |
12:26.03 | c0w | i am just about to send an email just thought i might try here first see if any engineers were about. |
12:35.47 | robbie2 | umm |
12:38.30 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
12:41.34 | *** join/#asterisk Thumann (n=brianm@80.163.152.30) |
12:41.50 | *** join/#asterisk manelideo (n=mdsilva@194.117.41.7) |
12:42.29 | manelideo | hi ppl! is anyone using AMP ??? I have this error when I type /usr/sbin/amportal start |
12:42.47 | manelideo | SETTING FILE PERMISSIONS |
12:42.47 | manelideo | Permissions OK |
12:42.47 | manelideo | STARTING ASTERISK |
12:42.47 | manelideo | Asterisk is already running |
12:42.47 | manelideo | STARTING FOP SERVER |
12:42.48 | manelideo | -bash: line 1: 17528 Killed /var/www/html/panel/safe_opserver |
12:42.50 | manelideo | -bash: line 1: 17591 Killed /var/www/html/panel/safe_opserver |
12:42.52 | manelideo | ----------------------------------------------------- |
12:42.54 | manelideo | The FOP's server (op_server.pl) could not start! |
12:42.56 | manelideo | Please correct this problem |
12:42.58 | manelideo | ----------------------------------------------------- |
12:43.24 | manelideo | I tried to google but it didnt solved my problem |
12:43.57 | Thumann | hi guys, i'm wondering if asterisk is able to 'lift' the task i present.. can it (if we use more than one server naturally) handle +8.000 users? And if so, do any of you have some reference stories ? |
12:44.04 | Inv_arp | manelideo: pastebin.ca for multple line paste |
12:44.40 | c0w | thumann, what are you wanting to run it as just SIP or what. |
12:45.05 | Thumann | c0w: afaik, just sip |
12:45.12 | pooh_ | join #smeserver |
12:45.12 | c0w | then i wouldn't recommend it. |
12:45.19 | pooh_ | :-) wrong windows |
12:45.25 | Thumann | c0w: SER instead? |
12:45.27 | c0w | use a sip proxy (SER) or something |
12:45.38 | c0w | and then have asterisk as your gateway / sevices box |
12:45.43 | wasim | Thumann: ser + * |
12:45.45 | c0w | yep |
12:45.57 | c0w | works very well together. =0 |
12:46.49 | Thumann | :> |
12:47.00 | Thumann | just one * ? |
12:47.04 | thewiizard | sounds interesting |
12:47.30 | wasim | Thumann: nyet ... multiple ser, multiple * is always a better proposition |
12:47.39 | Thumann | naturally |
12:47.49 | Thumann | but are there any limitations to *, and ser? |
12:48.06 | Thumann | as in.. max, 600 users pr. box.. etc.. or is it a question of hw sizing? |
12:49.47 | puzzled | wasn't there an issue with MWI when using ser+*? |
12:50.07 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
12:52.22 | ful|work | puzzled: i got my ser+* with MWI |
12:53.50 | manelideo | hmmm.... and with the browser AMP, when I want to add a new extension: "This module requires access to the Asterisk Manager. Please ensure Asterisk is running and access to the manager is available." |
12:54.04 | manelideo | can anybody help me? |
12:54.14 | thewiizard | manager running? |
12:54.24 | thewiizard | amportal start |
12:54.49 | manelideo | yes...I did that, but I got the error above (the FOP server) |
12:55.07 | robbie2 | did i see mention of a web based voicemail interface ? |
12:55.11 | manelideo | or it doesnt matter if the FOP server does not start |
12:57.34 | danzig | Using * as a SIP client, when it gets 'Failed to authenticate on REGISTER', it tries again every 20 seconds. Is there any way to increase this default without recompiling? |
12:58.03 | thewiizard | manelideo not really |
12:58.16 | thewiizard | robbie2 yeah new amp release has it |
12:58.34 | thewiizard | danzig its all about the sip timing |
12:58.36 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
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13:02.03 | robbie2 | hmm |
13:02.18 | robbie2 | what plugin this vmail.cgi require |
13:03.43 | danzig | >thewiizard>Yes... But it is up til * as the client what it does... Since the problem is a wrong password, it seems to be a waste to try every 20 sec. |
13:03.45 | Thumann | but are there any limitations to *, and ser? |
13:03.46 | Thumann | as in.. max, 600 users pr. box.. etc.. or is it a question of hw sizing? |
13:04.03 | JamesDotCom | sip is extremely scalable |
13:05.43 | coppice | scalable like a fortress under attack? :-) |
13:06.01 | JamesDotCom | meaning? |
13:06.33 | coppice | scalable is a much abused and seldom meaningful buzzword |
13:07.30 | JamesDotCom | well we're talking about numbers of users a network can handle |
13:08.00 | thewiizard | network can handle = * / x + y |
13:08.00 | JamesDotCom | and i see that being a scale, for example 1 - 10000 users |
13:08.26 | JamesDotCom | i know for a fact that a properly designed sip network will be the least of your scalability worries when dealing with 10000 users |
13:08.27 | coppice | which has almost nothing to do with SIP. SIP is a signalling protocol - i.e. it hardly does anything, and should definitely be able to handle a lot of user on a small box |
13:08.50 | JamesDotCom | that's what i'm saying |
13:08.58 | thewiizard | its all about hardware |
13:09.03 | JamesDotCom | it's too open a question |
13:09.08 | JamesDotCom | totally depending upon the environment |
13:10.21 | JamesDotCom | does that answer your question Thumann ;) |
13:11.48 | Thumann | it does :) |
13:11.49 | JamesDotCom | there's a billion factors... codec selection (quality:bandwidth), concurrent calls, features, basically, SIP and almost any SIP proxy will have no issue with that amount of users, but the rest will depend on other things |
13:14.57 | coppice | Thumann: saying "only SIP" says absolutely nothing. the real work is in handling the audio. if you want a box to act as a soft-switch, never seeing any of the audio, a small box might support 100K calls, unless the average call is rather short. If the audio goes through your box don't expect too many concurrent calls. If you boxes transcodes, using heavyweight codecs like G.729, except even... |
13:14.59 | coppice | ...less concurrent calls. |
13:17.15 | coppice | I shouldn't type while nearly asleep :-\ |
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13:21.42 | [TK]D-Fender | b00m |
13:21.42 | Inv_arp | opps wrong cable |
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13:27.32 | thewiizard | goddamn |
13:28.49 | tzanger | hahaha |
13:29.30 | Alystair | man, where are the free peanuts and beverage |
13:29.30 | tzanger | Videotron (Quebec cable company) is suing Bell ExpressVu (Satellite TV company) for not doing enough to protect themseves from piracy |
13:29.30 | tzanger | goddamn ducks? |
13:29.30 | Alystair | Videotron's still alive? |
13:29.30 | tzanger | apparently |
13:30.12 | tzanger | http://www.cbc.ca/story/business/national/2005/11/07/videotron-051107.html |
13:30.55 | Alystair | I gotta call up Bell and Rogers and nag them for quotes on a good business line |
13:31.15 | tzanger | heh |
13:31.15 | tzanger | we just use resold powerdsl here |
13:31.17 | tzanger | (Ikano) |
13:31.29 | tzanger | it works well and we have excellent connectivity to our networks at 151 front |
13:31.35 | Alystair | what an f'n rip off |
13:31.47 | tzanger | $80/mo ain't that bad |
13:31.48 | tzanger | honestly |
13:31.51 | Alystair | $199.95 a month for 5mbit |
13:31.55 | tzanger | for what? |
13:31.59 | Alystair | and 640kb up |
13:32.01 | Alystair | on rogers HAH |
13:32.02 | tzanger | cable? |
13:32.05 | Alystair | yeah |
13:32.05 | tzanger | jesus |
13:32.14 | tzanger | $80/mo for up to 8meg down and 800k up |
13:32.19 | tzanger | no caps |
13:32.21 | Alystair | that's sexy |
13:32.32 | tzanger | that's standard ADSL. static IP too even |
13:32.35 | tzanger | no port 25 block |
13:32.38 | Alystair | ok, where's the info |
13:33.20 | jake1932 | i get 3/768 now for $30 |
13:33.44 | Alystair | what's the damn deal with upload speeds anyway |
13:33.48 | af_ | I am getting this stuffs randomically: PRI: !! Got a UA, but i'm in state 1 (quadbri + bristuff). any hints? |
13:33.59 | Alystair | whatever happened to the SDSL :( |
13:35.21 | jake1932 | you can prob still get SDSL - but it'll cost a little more :o) |
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13:36.22 | thewiizard | yo i cant get this damn amp wired to my proxy |
13:36.26 | thewiizard | any special settings i need/ |
13:36.49 | thewiizard | trying to get away with just using authuser=, username=, host=, secret= and type= |
13:36.54 | tzanger | yeah I hate 800k up it's just not enough |
13:37.08 | tzanger | I'd prefer 1.5meg symmetrical |
13:37.21 | tzanger | although 8meg down does have its advantages |
13:37.58 | puzzled | anyone seen a compile error with res_config_odbc with cvs HEAD from an hour ago? |
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13:39.04 | demetrio | hi |
13:40.50 | demetrio | so I have this account with a VoIP provider connected to a "real" phone number. I set up asterisk to register as a client to accept incoming calls, but when I try to call from a phone with number X asterisk will say "Failed to authenticate user X". Why is this? |
13:43.24 | thewiizard | got the right password? |
13:43.47 | demetrio | sure, but that's not the point |
13:44.05 | demetrio | should I set up SIP accounts for every phone number in the world? |
13:44.08 | file[laptop] | registration and accepting calls are two different things |
13:44.15 | bon | hm |
13:44.23 | bon | why do all of my calls fall into one context? |
13:44.24 | mutilator | anyone have a mrtg => cacti config converter? |
13:44.55 | thewiizard | bon have u told them to? |
13:44.56 | demetrio | well, I thought that the register directive in sip.conf is for incoming calls |
13:45.18 | thewiizard | demetrio Failed to authenticate means you failed to authenticate |
13:45.27 | thewiizard | check ur account settings |
13:45.42 | demetrio | but that doesn't make sense |
13:45.58 | KriS83 | When does Digium have their working hours? |
13:46.01 | file[laptop] | Registrations only tell the other side, "here I am if you need to send calls to me" |
13:46.05 | demetrio | the user whose authentications fails is the pone number from wich I try to call |
13:46.10 | demetrio | right |
13:46.13 | file[laptop] | it's up to you to still configure your side to accept the calls |
13:46.18 | enemy | I have a dual E1 card which I`ve connected to the telco and to other equipment. My issue is that when I transfer calls from one E1 to the other, it seems to work nicely when it's isdn originating traffic. If it's an analogue connection (modem) which is going to be bridged, then I get req transfer capability 3K1AUDIO instead of DIGITAL, and then it just gets span 1 got hangup |
13:46.28 | file[laptop] | what you'll probably want to do is make a peer entry, with the IP address where calls will come from, and set insecure=very and context |
13:46.35 | file[laptop] | it'll match based on the IP and throw it into the right context |
13:46.56 | thewiizard | i allways find outgoing calls more fun to get going |
13:46.57 | demetrio | Ok I'll try the insecure=very |
13:47.05 | bon | thewiizard: i don't know, outgoing calls do go to right context but how do i tell incoming ones to use other context? |
13:47.22 | thewiizard | bon: basically for your provider youll have a context defined |
13:47.36 | thewiizard | when calls are recevied by said provider the calls are placed into said context |
13:47.40 | bon | ah |
13:47.42 | bon | lemme see.. |
13:47.47 | thewiizard | [voip_provider] |
13:47.49 | thewiizard | host=blabla |
13:47.52 | thewiizard | secret=blabla |
13:47.55 | thewiizard | username=blabla |
13:48.06 | thewiizard | context=[whatyouwant] |
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13:48.11 | thewiizard | i think its context |
13:48.19 | thewiizard | maybe something slightly different |
13:48.20 | demetrio | file[laptop], thanks a lot, that worked |
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13:48.53 | demetrio | I was missing the insecure=very, now I'll try to understand what that directive does |
13:49.21 | file[laptop] | causes asterisk to not challenge for user/password |
13:49.31 | file[laptop] | just does IP auth |
13:50.26 | thewiizard | type=friend might be good with that also ;0 |
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13:51.01 | file[laptop] | not really |
13:51.09 | file[laptop] | type=friend just generates two entries in memory, a user and a peer |
13:51.18 | file[laptop] | and most providers don't do user authentication, which is what a user entry is for |
13:51.27 | file[laptop] | they want you to do IP auth, which you use a peer for |
13:51.35 | thewiizard | IP auth with sip |
13:51.47 | thewiizard | why not use h323 |
13:52.10 | file[laptop] | I'm just going to... ignore... that |
13:52.32 | Dr-Linux | i want to avail Conference call feautre, i define a room 1111 in meetme.conf and i put an extension in extention.conf exten => 5557,1,Meetme,1111 |
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13:53.00 | bon | thewiizard: :( |
13:53.02 | bon | didn't help |
13:53.07 | bon | i thought it would |
13:53.16 | bon | just looked at my providers entry in mysql |
13:53.21 | bon | has context=whatiwant |
13:53.27 | Dr-Linux | so when i dial 5557 at softphone, i doesn't work and i see on CLI > Nov 8 18:45:18 WARNING[8432]: pbx.c:1293 pbx_extension_helper: No application 'Meetme' for extension (default, 5557, 1) |
13:53.28 | Dr-Linux | <PROTECTED> |
13:53.31 | bon | but it always gets written into cdr as what i don't want it to :) |
13:53.47 | file[laptop] | Dr-Linux: do you have zaptel installed, with a timing source? |
13:53.47 | Dr-Linux | what could be wrong ? |
13:54.02 | Dr-Linux | file[laptop]: no sir |
13:54.08 | file[laptop] | then that's the problem |
13:54.10 | file[laptop] | Meetme requires it |
13:54.22 | Dr-Linux | i'm only working on softphones |
13:54.41 | file[laptop] | Meetme itself requires it, you don't have a zaptel timing source - it won't work, if you don't have zaptel installed - it won't compile |
13:54.49 | file[laptop] | what you can do is try ztdummy |
13:55.05 | file[laptop] | instructions for making it and stuff are on the net, Google is your friend |
13:55.17 | emdub | google <3 |
13:56.06 | Dr-Linux | file[laptop]: whats easy way to install work for Meetme? ztdummy or zaptal ? |
13:56.11 | emdub | anyone familiar w/ cisco 7960 phones? |
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13:56.14 | Dr-Linux | i don't have any hardware |
13:56.53 | file[laptop] | Dr-Linux: ztdummy is part of zaptel |
13:57.37 | file[laptop] | it's a dummy driver, uses the UHCI USB controller on a 2.4 kernel, and the RTC on a 2.6 kernel |
13:58.09 | Dr-Linux | file[laptop]: i have installed asterisk 1.0.9 , so should i need to install zaptal seperately ? or it should already exists ? |
13:58.38 | file[laptop] | it's separate |
13:58.41 | Dr-Linux | as i don't have any hardware, but i need to avail Meetme feature |
13:59.48 | tzanger | Dr-Linux: so use ztdummy or zaprtc |
14:00.29 | file[laptop] | tzanger: so what's your tagline today... |
14:00.37 | file[laptop] | bah |
14:00.43 | tzanger | hang on I didn't change it |
14:00.58 | iCEBrkr | hey hey |
14:00.59 | puzzled | anyone yet tried to compile cvs HEAD from a couple of hours ago? |
14:01.04 | tzanger | better? |
14:02.16 | mutilator | cocacola is teh suck |
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14:03.19 | Dr-Linux | i have cisco 79xx ip phone in US, i'm in pakistan .. i wanna access them to configure with SIP , how can i do ? |
14:04.40 | *** join/#asterisk Einon (n=einon@ka.wa.hu) |
14:06.12 | *** join/#asterisk frix (i=frix@p54A876EB.dip.t-dialin.net) |
14:06.17 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
14:06.58 | Dr-Linux | tzanger ? |
14:07.06 | tzanger | I don't know anything about cisco sorry |
14:07.09 | frix | dont know if you could read it before... i need some help with compiling chan_capi, which blows me up with 1mb error log |
14:08.51 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:08.59 | Einon | hoi! I sat up asterisk to put outgoing calls through an ISDN modem. From linphone I dial a number asterisk dials it and the called party receives the call. But from any other client (twinkle, kphone, hardphone) I tried to dial the same number, asterisk dials but the called party do not receive the call. |
14:09.53 | thewiizard | Dr-Linux u gotta unlock it |
14:10.04 | thewiizard | upload new firmware SIP7.3 or 7.5 |
14:10.08 | thewiizard | via tftp |
14:10.17 | thewiizard | config and boom |
14:10.23 | *** join/#asterisk PBXtech (i=nik@55.sub-70-218-24.myvzw.com) |
14:10.23 | thewiizard | sorted |
14:10.33 | *** part/#asterisk PBXtech (i=nik@55.sub-70-218-24.myvzw.com) |
14:11.15 | clive- | frix use the sourceforge chan_capi |
14:11.59 | enemy | how can I change from ulaw to alaw on my zap channels? |
14:11.59 | frix | clive: thank you, i'll give it a try |
14:13.44 | queuetue | I am setting up VoicePulse over IAX2 - outgoing seems to work fine, and I have made a few calls with it, but incoming, I get tri-tones, and "your call cannot be completed. 075T" Can anyone shed any light into this error (does it come from asterisk, my outgoing provider, or from voicepulse?)) and how I might fix it? |
14:14.01 | Dr-Linux | thewiizard: yeah but i'm far away for phone i'm in pakistan and the phones located in US , they are new i am the one who will do all ? what should i do ? |
14:14.37 | tzanger | queuetue: have you registered to voicepulse? |
14:14.44 | tzanger | they have pretty good documentation on their site last I looked |
14:15.16 | frix | clive: pretty much the same |
14:15.23 | queuetue | tzanger: Yes. I am registered with voicepulse (I just mentioned I could make outgoing calls with them) and I have followed their instructions. |
14:15.28 | frix | but a little less than 1mb :) |
14:15.35 | tzanger | no |
14:15.40 | tzanger | register => user:pass@voicepulse |
14:15.43 | tzanger | so they know where to find you |
14:15.51 | queuetue | tzanger: Yes. |
14:16.07 | *** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net) |
14:16.14 | tzanger | queuetue: ok, and with iax2 debug turned on what do you see on the console (use pastebin) whey you try to call your DID from a cellphone? |
14:16.20 | SpaceBass | what is the standard vertical code to enable call recording? |
14:16.22 | queuetue | XXX:YYY@gwiax-in-01.voicepulse.com |
14:16.28 | *** join/#asterisk asteriskgeeks (n=SIPdawg@pbxtech.com) |
14:16.28 | asteriskgeeks | <PROTECTED> |
14:16.41 | queuetue | tzanger: I don't believe I see anything happen at all - will try again, though. |
14:16.54 | *** join/#asterisk PBXtech (i=nik@55.sub-70-218-24.myvzw.com) |
14:17.05 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
14:17.37 | brad_mssw | tzanger: are you starting the asterisk console using the -cvvvd or similar ? |
14:17.55 | tzanger | nah |
14:18.09 | tzanger | asterisk -vvvgc is what I do for testing and just -g for normal (using -vvvrc for connecting to it then) |
14:18.49 | brad_mssw | throw a d in there |
14:18.58 | brad_mssw | for some extra debugging statements |
14:18.58 | tzanger | nah |
14:19.02 | SpaceBass | anyone...quickly... on the phone with sprint customer service and really need to record it... is there a way to do so mid-call? |
14:19.02 | tzanger | I never find the use for it |
14:19.31 | iCEBrkr | Manage port. |
14:19.44 | tzanger | SpaceBass: park them and do up a dialplan real quick. :_0 |
14:19.46 | tzanger | I don't know how else |
14:19.52 | tzanger | if you don't already have it done |
14:19.56 | brad_mssw | tzanger: uh, it could help you solve your issue ... most likely you'd see a message on the console stating that it couldn't find a certain context, or the context doesn't contain the extension it's looking for |
14:20.01 | iCEBrkr | tzanger: Can't you issue the zapbarge or whtever via the manageport? |
14:20.07 | SpaceBass | tzafrir im using AMP and think its in the dial plan, just cannot find the code |
14:20.10 | tzanger | iCEBrkr: no idea |
14:20.14 | iCEBrkr | I've done it before. |
14:20.19 | iCEBrkr | Use the manage port to turn on recording |
14:20.21 | brad_mssw | haha, tzanger should have been directed towards queuetue |
14:20.32 | brad_mssw | that's what I get for jumping in late |
14:20.52 | iCEBrkr | http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Monitor |
14:21.01 | iCEBrkr | ACTION: Monitor |
14:21.01 | iCEBrkr | Channel: SIP/x7062618529-643d |
14:21.02 | iCEBrkr | File: channelsavefile |
14:21.02 | iCEBrkr | Mix: 1 |
14:25.41 | morale | <PROTECTED> |
14:25.43 | morale | nifty. |
14:25.49 | queuetue | tzanger: Ok, I did get a message: http://rafb.net/paste/results/VlYKTO29.html And the [problem appears to be "No such context/extension" ... what context or extension is it looking for? |
14:26.17 | tzanger | queuetue: in your type=user (or type=friend) for voicepulse, what conext are you specifying? |
14:26.38 | queuetue | tzanger: from-pstn |
14:26.54 | tzanger | ok, and what does your [from-pstn] context look like (use pastebin) |
14:27.10 | morale | how do i set the cost on a codec in asterisk? i want all my outbound calls to use the g.729 codec by default now |
14:28.14 | *** join/#asterisk emakris2 (n=emakris@c-24-128-56-2.hsd1.ma.comcast.net) |
14:28.44 | queuetue | tzanger: I'm using AAH, so it's a bit spaghetti-like. Essentially, I get the impression need to set up a DID... Let me try that and get back to you all if i can't take it from here. |
14:29.20 | tzanger | queuetue: well set up this did |
14:29.33 | tzanger | exten => _X.,1,NoOp(EXTEN is ${EXTEN}) |
14:29.41 | queuetue | tzanger: I'm pretty new - didn't realize I had to (the zap lines "just worked". :) ) |
14:29.47 | tzanger | queuetue: :-) |
14:31.07 | tzanger | we were all newbies at one time |
14:34.45 | mutilator | anyone delt with call forward billing and call accounting? |
14:34.51 | mutilator | how do ya handle your cdr? |
14:36.38 | Dr-Linux | tzanger: i installed the zaptal, and now nothing works, softphone can't dial any extension, "sip show users" doesn't show any registerd user? |
14:36.55 | Dr-Linux | what could be happend? i just zaptal and nothing else i face this happend :S |
14:38.10 | Dr-Linux | is the sip users disabled when i installed zaptal or what ? :S |
14:38.29 | tzanger | Dr-Linux: you really need to read up... you've been an asterisk user now for several months, you should know ho to do basic diagnostics by now |
14:39.20 | *** join/#asterisk grimse (n=grimse@p5481E912.dip.t-dialin.net) |
14:39.27 | Dr-Linux | i left long ago , i just started few days back .. |
14:39.36 | morale | hmm.. theoretically on my adsl connection at home.. i can run 8 inbound lines with this g729 codec.. hmm.. ideas start flowing |
14:39.54 | Dr-Linux | everything was ready and working for demo .. but everything is gone :( as i installed zaptal |
14:40.36 | morale | i need to find some chicks with no self-esteem and setup a 1-900 number. |
14:42.05 | Rowter | when connecting asterisk to a panasonic, you will have a cable from fxs of panasonic to fx0 of a tdm asterisk card? |
14:42.20 | Rowter | thats correct? |
14:42.25 | |Vulutre| | no |
14:42.34 | |Vulutre| | it would be FXS to FXO |
14:42.43 | |Vulutre| | you cant connect two FXOs |
14:42.45 | mutilator | .. |
14:42.48 | Rowter | thats what I said |
14:42.52 | |Vulutre| | omg |
14:43.00 | Rowter | FXS to FXO |
14:43.02 | |Vulutre| | yea |
14:43.08 | Rowter | FXS panasonic FXO asterisk |
14:43.09 | |Vulutre| | I literally climbed out of bed |
14:43.12 | |Vulutre| | yea you got it |
14:43.16 | Rowter | hehe, thanks |
14:44.17 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivflm3.dialup.mindspring.com) |
14:47.45 | ManxPower | Rowter, you want TDM400P FXS into the Panasonic CO (aka FXO) port. |
14:48.07 | ManxPower | Most PBX FSX ports do NOT support any way to tell Asterisk to hangup the line. |
14:48.46 | ManxPower | ~fsxfxo |
14:48.52 | ManxPower | ~fxsfxo |
14:48.53 | jbot | methinks fxsfxo is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
14:50.11 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
14:50.16 | *** join/#asterisk BerndR (n=broessl@193.83.150.54) |
14:51.11 | Einon | How is it possible that the very same asterisk with the very same SIP peer but with two different softwares (linphone and twinkle) works different. If I place a call from SIP to my cellphone via Asterisk's ISDN modem, from linphone my cellphone receives the call, from twinkle it does not. |
14:51.33 | Einon | any ideas |
14:51.43 | morale | I JUST SWITCHED TO GEICO AND SAVED A BUNDLE ON CAR INSURANCE... argh! i hate those commericials |
14:52.51 | BerndR | part |
14:52.55 | *** part/#asterisk BerndR (n=broessl@193.83.150.54) |
14:53.01 | *** join/#asterisk copantl (n=galel@63.245.93.138) |
14:54.20 | mutilator | for cdr_mysql |
14:54.30 | mutilator | setCIDNum changes the src field |
14:54.35 | mutilator | but how do i change the dst field? |
14:55.17 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
14:56.30 | *** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-206.hsd1.ut.comcast.net) |
14:56.40 | ManxPower | morale, I hate them because they would not insure me. |
14:57.14 | ManxPower | mutilator, do a Goto |
14:57.31 | gambolputty | The speed racer geico commercial is funny |
14:57.43 | tmccrary | hey, I experience choppyness during phone calls with voip over a high latency connection (250ms). Is there some easy ways to tweak this? Jitter? |
14:57.44 | Katty | mishehu: come find me later. |
14:59.47 | mutilator | Manx: goto another extension? |
14:59.49 | iCEBrkr | tmccrary: Turn off your Gnutella and eMule clients! :D |
15:00.12 | tmccrary | hehe, bandwidth shouldn't be a problem. I was thinking maybe doing QoS |
15:01.43 | gambolputty | anyone use Gentoo? |
15:02.02 | iCEBrkr | genpooh! |
15:03.42 | mutilator | ah sweet |
15:04.38 | iCEBrkr | Whacked. |
15:05.01 | iCEBrkr | We have an analog card in our PBX and my X100P keeps looping-- thinking the line is ringing. |
15:06.33 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
15:08.02 | Dr-Linux | any can tell me, how can i access Cisco 79xx ip phone remotely to configure with SIP ? |
15:08.04 | *** join/#asterisk ian_k (n=ian@gateway.digium.com) |
15:08.19 | iCEBrkr | How would you normally configure it? |
15:09.19 | mutilator | thx manx |
15:09.28 | wasim | iCEBrkr: check your wire |
15:09.29 | ManxPower | iCEBrkr, Are you SURE it's an analog port on the PBX? i.e. can you plug a standard home analog phone into the port and have it work? |
15:09.44 | iCEBrkr | ManxPower: Yeah. |
15:10.00 | iCEBrkr | ManxPower: It's meant for that. So you can plug fax machines and such into the PBX |
15:10.23 | ManxPower | iCEBrkr, I don't know what to suggest. |
15:10.31 | *** join/#asterisk Geert (i=geert@irssi/staff/geert) |
15:10.36 | iCEBrkr | I put a standard phone on it. I got a dialtone. Had to dial 9 to get out, but that's cool. |
15:10.40 | iCEBrkr | Really weird shit |
15:11.01 | iCEBrkr | Thing is. I'm not sure if it's the 1.2 zaptel stuff or what. |
15:11.25 | iCEBrkr | I had all this working with 1.0.3-- long ago when I first started tinkering with Asterisk. |
15:11.51 | ManxPower | iCEBrkr, I've had X100P cards just top working before. Make sure the card is seated in the socket. |
15:11.57 | ManxPower | top == stop |
15:12.03 | Geert | Does somebody have information about how to get the call duration through an IVR? Example: Customer calls IVR -> IVR pickups (starts the timer?) -> Users does his thing -> IVR forwards call to VoIP phone -> IVR closes call (stops timer) -> IVR inserts INSERT query |
15:12.16 | thewiizard | eeeeeeeeeeeee |
15:12.45 | iCEBrkr | ManxPower: I'll do that, the machine has been transported a bunch of times. |
15:13.11 | iCEBrkr | I just wish I had access to PSTN. I don't trust the shit pulled into this building. |
15:13.24 | ManxPower | iCEBrkr, I know what you mean. |
15:13.30 | olivier_ | <Geert> Have a look to the ANSWERED variable |
15:13.48 | Geert | olivier_: thanks |
15:13.54 | iCEBrkr | Geert: That's all part of the CDR |
15:13.58 | *** join/#asterisk _T3_ (n=rposada@200.63.231.210) |
15:14.16 | olivier_ | sorry ANSWEREDTIME variable |
15:20.03 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
15:20.41 | lunk | how do i check if a call that's been placed is busy or not in a dial plan? |
15:21.14 | Rowter | ManxPower, whats the best combination to have some extensions to ASterisk from a panasonic? |
15:21.19 | iCEBrkr | ${DIALSTATUS} |
15:21.44 | lunk | iCEBrkr: thanks |
15:22.11 | Einon | bye |
15:22.12 | *** part/#asterisk Einon (n=einon@ka.wa.hu) |
15:22.45 | ManxPower | Rowter, there really isn't any good solution for that. |
15:23.13 | ManxPower | Rowter, But I can tell you that if you go Panasonic FXS -> Asterisk FXO you are going to have lines that don't hang up. |
15:23.30 | Rowter | ManxPower, they want to make voip calls, its better to add a simple ata then.. |
15:24.15 | Rowter | ManxPower, I see.. thanks for telling me about that hangup issue.. |
15:24.30 | ManxPower | Rowter, I speak from personal experience 8-) |
15:24.55 | tzanger | ManxPower: panasonics don't signal disconnect well? |
15:25.05 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
15:26.09 | _T3_ | anybody knows who wants a job to modify dsp.c for callprogress ?? |
15:30.01 | paryl | will one of the little analog adapters work for a fax machine? |
15:30.37 | queuetue | tzanger: BTW< I doubt you were thinking about it at all, but adding the DID fixed everything. Thanks for your help. |
15:31.00 | *** join/#asterisk shout (n=rcsw@host213-123-195-3.in-addr.btopenworld.com) |
15:31.20 | tzanger | queuetue: good |
15:31.23 | ManxPower | tzanger, Very few PBX FXS ports signal disconnect with anything other than a tone. |
15:31.31 | tzanger | this is true |
15:31.40 | tzanger | AbsoluteTimeout() may be your only hope |
15:31.55 | ManxPower | Amd we all know how well Asterisk handles tone disconnects |
15:32.06 | tzanger | very well... it ignores them perfectly every time |
15:33.52 | Dr-Linux | whts difference between ztdummy and ztdynamic ? |
15:34.10 | fourcheeze | ManxPower: is it possible to use something like a sipura SPA-3000 to connect to the FXS and get stuff into the voip world? |
15:34.40 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
15:36.12 | fourcheeze | I mean can the spa-3000 forward stuff from its FXO port to a sip url |
15:37.14 | mutilator | anyone know what kinda warranty wildcards have? |
15:37.46 | paryl | again, sorry, but can an analog adapter handle fax/modem calls? |
15:38.35 | azzie | paryl, when configured correctly - yes |
15:38.37 | *** join/#asterisk Craziman2 (n=Craziman@boromir.apid.com) |
15:39.19 | paryl | azzie: thank you. any specific brand? and what do you mean by properly configured... a specific codec, or what? |
15:39.39 | thewiizard | yoright ive got this tedious issue |
15:39.50 | thewiizard | all calls apart from internal are getting "All circuits are busy now" |
15:39.57 | azzie | paryl, what do you want exactly from fax/modem ? |
15:40.03 | mutilator | anyone know? |
15:40.13 | mutilator | i don't see any warranty info on digiums site.. |
15:40.28 | thewiizard | years manufacturer i reckon |
15:40.40 | thewiizard | usually the detault for most hardware these days |
15:40.44 | thewiizard | *default |
15:41.04 | paryl | azzie: i need to provide at least 11 analog lines... 8 fax machines, 3 modem lines |
15:42.31 | azzie | paryl, any bandwidth limitations? can you have all calls G.711 ? |
15:42.46 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
15:43.01 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
15:43.16 | paryl | azzie: this is just for the connection to asterisk... i'll be routing these calls directly to one of the T1 channels |
15:44.13 | azzie | so what's the answer to my question |
15:44.18 | paryl | so... no bandwidth limitations, aside from the normal network |
15:44.31 | paryl | they'll be on a 100mbit switch |
15:44.40 | Dr-Linux | ManxPower: how can i access Cisco 79xx ip phone remotely to configure with SIP ? |
15:44.45 | *** join/#asterisk miksi (i=monte@a80-186-17-71.elisa-laajakaista.fi) |
15:44.54 | miksi | hello room |
15:45.19 | miksi | i need help on asterisk anybody can help |
15:45.26 | azzie | paryl, then just use G.711 and it won't matter - fax, modem, voice... |
15:46.18 | paryl | azzie: awesome. and there aren't any brands to stay away from, as long a they handle G.711? |
15:46.26 | ManxPower | Dr-Linux, I have no idea, I don't use Cisco phones. |
15:46.27 | mutilator | does anyone know tho? |
15:46.31 | *** part/#asterisk Craziman2 (n=Craziman@boromir.apid.com) |
15:46.33 | azzie | paryl, yes... |
15:46.48 | paryl | azzie: thanks so much. |
15:46.56 | emdub | anyone familiar with cisco 7960 phones? |
15:47.04 | azzie | paryl, problems begin when you need to maximize compression of voice/fax calls etc. If you don't care about compresing them, there's no problem |
15:47.13 | thewiizard | U need the sip firmware from Cisco |
15:47.18 | *** join/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com) |
15:47.23 | thewiizard | phat luck unless u have a support contract |
15:47.27 | manelideo | thewiizard the problem with /usr/sbin/amportal was solved with a reboot lolol |
15:47.35 | thewiizard | hehehe nice manelideo |
15:47.50 | paryl | gotcha... i've been spooked by the fax pages on all of the asterisk sites. i guess that's what they were referring to |
15:48.19 | ManxPower | Dr-Linux, One might start by reading the documentation for your phone and firmware |
15:49.21 | enemy | is there some way I can automaticly add agents to queues without having to get them to dial an extension executing AddQueueMember? |
15:49.54 | *** join/#asterisk _santiago_ (n=santiago@208.195.215.124) |
15:50.09 | iCEBrkr | enemy: They'd have to login no? So the system knows they're at their desk? |
15:50.49 | ManxPower | enemy, set a member= line in queues.conf for "agents" that are always "logged in" |
15:50.50 | enemy | iCEBrkr: I don't require that, they are always at their desk |
15:51.04 | ManxPower | i.e. member=SIP/happyagent |
15:53.33 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
15:54.12 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
15:54.12 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
15:54.43 | enemy | manxpower, like group=1 member = SIP/236 ackcall=no ? |
15:54.57 | ManxPower | ender, I don't use group, but yes. |
15:55.10 | ManxPower | Here, I'll post my queues.conf |
15:55.37 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
15:56.00 | *** part/#asterisk miksi (i=monte@a80-186-17-71.elisa-laajakaista.fi) |
15:56.08 | ManxPower | http://pastebin.ca/27998 |
15:59.07 | thewiizard | Dr-Linux |
15:59.17 | thewiizard | google for Cisco 7960 SIP |
15:59.27 | ManxPower | ~mailinglist |
15:59.28 | jbot | mailinglist is, like, Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
15:59.29 | ManxPower | ~docs |
15:59.31 | jbot | i heard docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
16:03.06 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
16:03.09 | *** join/#asterisk thdei (n=DD@nat1.cri74.org) |
16:03.58 | Pazzo | hi! I'm successfully running SER / Asterisk / Mediaproxy / Stund toghether, BUT: incoming calls on Asterisk are going to be routed to SER - and if nobody answers the call should go to Voicemail (on Asterisk). Voicemail works fine and so do incoming calls - but redirecting them from asterisk to ser and back once again fails. |
16:04.07 | Pazzo | is there a solution for such a scenario? |
16:04.31 | iCEBrkr | Asterisk should handle the transfer to Voicemail.. No? |
16:04.37 | ManxPower | Isn't that a SER question? |
16:04.37 | Pazzo | yep |
16:04.55 | ManxPower | Pazzo, you would need a timeout on your Dial line on Asterisk |
16:05.18 | enemy | manxpower, could I see your extensions.conf also? doesnt look like member had any effect |
16:05.39 | mutilator | hmm |
16:05.44 | mutilator | clec fubared |
16:05.50 | mutilator | i can set my ani and cidname |
16:05.52 | file[laptop] | Pazzo: you silly person, you're trying to route the same INVITE Asterisk sent out, back to itself aren't you? |
16:06.07 | mutilator | cidnum anyway |
16:06.13 | ManxPower | enemy, http://pastebin.ca/27999 |
16:08.53 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
16:09.36 | Pazzo | file[laptop]: yep, I'm such a silly person :-) |
16:10.08 | file[laptop] | Pazzo: Asterisk will freak out and go, "no no no - LOOP DETECTED you twat!" |
16:10.55 | Pazzo | SER is handling timeouts (based on sql-based user preferences), will detect the users language (italian, german etc) and route the call to the appropriate Asterisk extension |
16:11.16 | Pazzo | If the call comes from a SER client naturally everything is fine |
16:11.19 | file[laptop] | 'tis not a SER thing, it's an Asterisk thing |
16:11.32 | file[laptop] | it's because you're trying to route the call back to itself, and it freaks out |
16:11.33 | Pazzo | but as you are saying, it's a kind of loop |
16:11.52 | Pazzo | no workaround? |
16:11.58 | file[laptop] | not off the top of my head |
16:12.16 | blitzrage | Asterisk says, "send this call to this address", then SER says, "this call is for you", then Asterisk is like, "no way Jose, I just sent you that call" |
16:13.22 | Pazzo | maybe I could put a flag on calls from asterisk to recognize them later and don't do voicemail routing on SER if this flag is set?! and in parallel also try to handle timeouts the same way in asterisks extension.conf... |
16:14.06 | file[laptop] | mmm what about rejecting the call once it hits the failover route, if it's going back to asterisk and it came from asterisk |
16:14.19 | file[laptop] | and then in asterisk deal with the rejection... might be able to add a header or something |
16:14.33 | file[laptop] | that was just randomly off the top of my head |
16:14.36 | blitzrage | custom SIP header maybe? |
16:14.46 | file[laptop] | blitzrage: that's what I was thinking |
16:14.55 | file[laptop] | anyway, I'm leaving in like 5 minutes |
16:14.56 | blitzrage | file[laptop]: since we can set those now :) |
16:15.00 | blitzrage | and parse them too |
16:15.05 | Pazzo | file[laptop]: extactly, that's what I'm trying to explain with my poor English words :) |
16:15.15 | blitzrage | My HD died on my PBX last night :( |
16:15.39 | blitzrage | wonder if I have an extra one lying around |
16:15.45 | Pazzo | if (call from asterisk) { let call fail } else { do all the preference-based stuff including voicemail etc } |
16:15.47 | blitzrage | anyways, I'm out -- lates |
16:16.22 | Pazzo | and on asterisk I have to implement my preference-based voicemail system - but that shouldn't be hard to realize |
16:16.52 | Pazzo | thanks a lot for helping to clean up my mind! |
16:17.16 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
16:17.51 | *** join/#asterisk theNOTO (n=biggs@69-165-25-59.clvdoh.adelphia.net) |
16:18.29 | thewiizard | yo is there a default incoming context for asterisk |
16:18.40 | thewiizard | and is it 'incoming' |
16:19.15 | theNOTO | I am going to be running asterisk with nufone service at home. What are the advantages of IAX vs SIP? |
16:19.20 | Pazzo | blitzrage: asterisk is now able to set custom SIP headers? |
16:19.35 | ManxPower | theNOTO, IAX2 is easier to deal with when NAT is involved. |
16:19.55 | Pazzo | blitzrage: (I'm running 1.2.0b1 as I had some trouble with 1.2.0b2) |
16:20.22 | blitzrage | Pazzo: yes -- do a show applications |
16:20.59 | theNOTO | ManxPower: Ok, what about bandwidth requirements? |
16:21.51 | Pazzo | blitzrage: hey, that's great! thanks a lot! |
16:22.37 | Pazzo | blitzrage: has this also been there in 1.0.x and been overseen by me or is this a new feature in 1.2.x? |
16:22.49 | blitzrage | new feature |
16:22.52 | blitzrage | 1.0.x is OLD |
16:24.27 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
16:24.40 | ManxPower | theNOTO, they are about the same for small numbers of calls. |
16:24.57 | ManxPower | IAX2 might use a little bit less bandwidth, assuming the same codec. |
16:25.29 | tzanger | only if trunking |
16:25.46 | tzanger | the difference in RTP and IAX2 overhead is negligable |
16:25.54 | Pazzo | blitzrage: that's true :-) I have had to start a VoIP project for a local ISP two or three months ago - it was the first time I have been in contact with Asterisk & SER and started with their stable versions - but I realized very soon that Asterisk 1.0.x won't make me happy ;-) |
16:26.01 | ManxPower | tzanger, I did say :little bit: |
16:26.07 | tzanger | :-) |
16:26.16 | *** join/#asterisk tzhs (n=thomas@p5497DBD5.dip.t-dialin.net) |
16:29.13 | sahafeez | does asterisk keep call records by default |
16:29.43 | oogle | sahafeez: check /var/log/asterisk/cdr-csv |
16:30.15 | sahafeez | oogle: thanks |
16:31.39 | KriS83 | Short question. When using CDR, can it be used for logging _all_ incoming and outgoing calls? |
16:32.08 | KriS83 | Yes/no answer prefered ;) |
16:32.23 | *** join/#asterisk toddf (n=toddf@adsl-65-66-18-204.dsl.okcyok.swbell.net) |
16:32.34 | Qwell | KriS83: thats what it does |
16:32.41 | *** join/#asterisk jlewis (i=jlewis@solo.atlantic.net) |
16:33.02 | KriS83 | Qwell, ok. Perfect! Thats all I wanted to know ;) |
16:33.06 | KriS83 | thx |
16:33.35 | tmccrary | Has anyone here done VOIP over a VPN? |
16:33.48 | tmccrary | or more specifically with OpenVPN (SSL VPN) |
16:34.06 | tmccrary | If so, should I do anything special to optimize performance? |
16:34.14 | azzie | quality should suck big time ;) |
16:34.22 | tzhs | Trying to wait longer then 10s on an incoming call from capi causes Spawn extension (isdn-in, XXXXXXXXXX, 1) exited non-zero on 'CAPI/ISDN1XXXXXX-3'. is there a solution? |
16:34.24 | *** part/#asterisk pif (n=ldm@zenon.apartia.fr) |
16:34.41 | Inv_arp | tmccrary: i know a friend that does it using 3com/ulaw on dsl works ok for him |
16:36.06 | Inv_arp | i just know that certain codecs being ecapsulated in VPN is kinda screwy (for lack of a better term) |
16:36.26 | Lostfrog | tmccrary: I use IPsec and Openvpn.. no problem with either. |
16:36.44 | ManxPower | tzhs, that means you don't have a priority 2 |
16:37.37 | jlewis | I' |
16:37.38 | [TK]D-Fender | I use an SPA-2000 over an IPSec connection here to link to my other plant. |
16:37.40 | Lostfrog | The thought of not using a VPN scares me. |
16:37.52 | jlewis | I've done iax over pptp with encryption |
16:38.03 | jlewis | it seemed to work |
16:38.14 | tzhs | ManxPower: there is one, but capi gets an disconnect to early (still ringing) |
16:38.47 | ManxPower | tzhs, PASTE the section of extensions.conf to pastebin.ca |
16:38.49 | thewiizard | yo well quick question |
16:38.58 | thewiizard | i have context called [from-sip] |
16:39.13 | thewiizard | how cna i forward all calls that come into that context to my callingcard app |
16:39.15 | thewiizard | on 850 |
16:39.51 | ManxPower | thewiizard, you don't "forward" the calls, you just match the call in extensions.conf that runs your app |
16:40.10 | jlewis | exten => _. will match all calls |
16:40.20 | thewiizard | _. |
16:40.25 | thewiizard | so _.,1,Answer |
16:40.33 | thewiizard | that what we're talkin |
16:41.44 | |Vulutre| | Anyone know how to set it so that a Polycom IP500 can only accept 1 call per line at a time? so that the second call with go into busy voicemail? |
16:41.44 | tzhs | ManxPower: http://pastebin.ca/28002 |
16:42.10 | ManxPower | tzhs, What version of Asterisk are you using? |
16:42.34 | ManxPower | |Vulutre|, it's documented in the release notes for 1.5.x I think. Maybe the 1.6.x |
16:42.43 | tzhs | ManxPower: cvs head, but the same with 1.0.9 |
16:43.09 | ManxPower | tzhs, priority n is not supported by 1.0.x or CVS -r v1-0 |
16:43.24 | |Vulutre| | ManxPower: is it incominglimit=1 ? |
16:43.32 | tzhs | ManxPower: thats true. but now im using head |
16:43.46 | ManxPower | tzhs, I don't have any suggestions |
16:44.01 | [TK]D-Fender | |Vulutre| its all in your sip.cfg |
16:44.08 | [TK]D-Fender | Or phone.cfg |
16:44.11 | ManxPower | |Vulutre|, only if you want to use Asterisk's crappy call limiting stuff. Why not use the polycom call limiting stuff -- that actually works. |
16:44.22 | tzhs | ManxPower: anyway tnx |
16:44.30 | [TK]D-Fender | Use the Polycom offer feature, trust us... |
16:44.32 | ManxPower | tzhs, talk to someone that uses BRI |
16:44.46 | |Vulutre| | [TK]D-Fender: do you know what that command is, so I can research it? |
16:45.24 | ManxPower | tzhs, can you do an Answer then Ringing, then the Wait? |
16:45.35 | ManxPower | Your telco might be upset about nothing happening for 10 seconds |
16:45.40 | thewiizard | anyone ever used areskicc? |
16:46.12 | ManxPower | |Vulutre|, call.callsPerLineKey="1" |
16:46.13 | tzhs | ManxPower: no, because there is an other phone connected which i want to use |
16:46.24 | |Vulutre| | ManxPower: thank you |
16:46.35 | ManxPower | tzhs, Ask on the mailing list. |
16:46.46 | tzhs | ManxPower: i like to use it as an extended answering machine |
16:47.00 | ManxPower | tzhs, *nod* Asterisk is not designed for that. |
16:47.01 | tzhs | ManxPower: ok. i'll do it |
16:47.32 | ManxPower | Asterisk is designed to be a PBX, not an answering machine and so doesn't not really support stuff like "use a phone on the same line, but not controlled by asterisk" |
16:51.43 | bjohnson | hehe .. my son broke my plastic "H" key for my Compaq notebook and I have to add a whole new keyboard |
16:52.01 | ManxPower | I suggest selling him on the back market. |
16:52.42 | mog_work | i have an h key |
16:52.50 | mog_work | for a compaq presario notebook |
16:53.34 | bjohnson | this is a M700 Armada .. no idea if they are compatible |
16:53.44 | bjohnson | I've been trolling service shops |
16:54.01 | mog_work | black and thick? |
16:54.34 | mog_work | certs are lame |
16:54.44 | javo | ya |
16:54.46 | thewiizard | im self certified |
16:54.50 | thewiizard | probably just as good :) |
16:54.56 | javo | whoa! |
16:54.58 | mog_work | but so was i when i was 12 |
16:55.02 | thewiizard | im 11 |
16:55.04 | thewiizard | :P |
16:55.12 | mog_work | we can tell |
16:55.16 | thewiizard | hah |
16:55.16 | thewiizard | ur wrong |
16:55.20 | thewiizard | cuz im not really ;) |
16:55.26 | thewiizard | just yanking ur chain |
16:55.28 | mog_work | but im right on the inside |
16:55.39 | thewiizard | but are you Allright or just partiality |
16:55.43 | thewiizard | *partially |
16:55.50 | mog_work | well thanks |
16:56.03 | thewiizard | no probs |
16:56.24 | Marcel-AS16215 | quick question how i can filter in the extension.conf circuit-busy that my Asterisk say this and not forward me to the Demo-thanks box ? |
16:57.12 | mog_work | ? |
16:57.13 | thewiizard | right got another interesting one for ya |
16:57.26 | bjohnson | mog_work: light grey and thin |
16:57.52 | mog_work | i dont think it will work then |
16:57.57 | mog_work | my keys are real hard |
16:58.02 | thewiizard | Got calling cards app installed with asterisk |
16:58.12 | mog_work | why not move the} key or something you dont use a lot over |
16:58.22 | mog_work | I never use the number 9 |
16:58.24 | thewiizard | i have a Siptone2 registered as an extension i dial 850 which the access number for the CC App. |
16:59.11 | thewiizard | when i dial in the App listens for DTMF for the Calling Card Number and then logs me in with that number |
16:59.33 | thewiizard | when i dial from this local extension the app goes skitz and believes ive entered some incorrect digits |
16:59.52 | thewiizard | then its skitzs again and ends the call |
17:00.45 | thewiizard | fine from a ddi tho |
17:00.54 | mog_work | do dtmf debug |
17:00.57 | ManxPower | thewiizard, You prolly have a problem with SIP DTMF |
17:00.59 | mog_work | tell me what is interpreted |
17:01.02 | bjohnson | mog_work: the H key connectors on back are upside down compared to the rest and the corner is notched out for the pointer thing |
17:01.03 | mog_work | and what you are sending |
17:01.10 | ManxPower | i.e. phone set for inband, asterisk set for rfc2833 or something like that |
17:01.15 | bjohnson | looks like I can buy a keyboard on ebay for $20 |
17:01.22 | mog_work | well there you go |
17:01.35 | thewiizard | dtmf debug = noon |
17:01.42 | thewiizard | no cmd |
17:01.49 | mog_work | lol |
17:01.54 | mog_work | go to logger.conf |
17:02.01 | mog_work | enable dtmf in console |
17:02.04 | mog_work | then logger reload |
17:02.43 | thewiizard | ive got full loggin |
17:02.51 | thewiizard | notice warning error debug verbose |
17:03.09 | mog_work | and add dtmf |
17:03.11 | mog_work | its new |
17:03.12 | thewiizard | ah |
17:03.14 | thewiizard | sweet |
17:03.18 | mog_work | you are running current asterisk right? |
17:03.29 | sjobeck99 | hi, all, how's it goes the good fight today? |
17:03.36 | sjobeck99 | i wonder why i'm seeing : |
17:03.36 | sjobeck99 | <PROTECTED> |
17:03.42 | thewiizard | 1.0.9 |
17:04.17 | sjobeck99 | that comes after: |
17:04.17 | sjobeck99 | Executing Dial("Zap/4-1", "IAX2/NuFone/5088888888|30") in new stack |
17:04.19 | mog_work | oh its not in 1.0.9 |
17:04.24 | mog_work | itd be in the betas |
17:04.33 | mog_work | seeing dtmf is harder in older versions |
17:04.45 | mog_work | is that account active sjobeck99 |
17:04.59 | sjobeck99 | yes, thx |
17:05.16 | sjobeck99 | i edited the # in my paste, but, yes |
17:05.50 | *** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net) |
17:05.55 | sjobeck99 | is my iax.conf goobered? |
17:06.07 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
17:06.09 | synthetiq | when putting in signaling of e&m wink in zaptel.conf is it em_w or emw |
17:06.15 | mog_work | maybe |
17:06.39 | mog_work | in zaptel its just e&m |
17:06.48 | Inv_arp | sjobeck99: paste it in pastebin.ca |
17:06.59 | Inv_arp | ~seen ariel |
17:07.10 | jbot | ariel <n=ariel@200-70-76-10.mrse.com.ar> was last seen on IRC in channel #debian-br, 79d 12h 26m 1s ago, saying: 'mas o multiplicador sta en 18x'. |
17:07.10 | synthetiq | i dont inticate the wink? |
17:07.10 | mog_work | nope |
17:07.10 | mog_work | you do that in zapata |
17:07.22 | Inv_arp | ~seen ariel_ |
17:07.24 | jbot | ariel_ <n=9999@dsl-20-177.cofs.net> was last seen on IRC in channel #asterisk, 18h 7m 55s ago, saying: 'harryvv, OK. If you call saying at home due to me falling off the roof just after the hurricane ok. Oh well I guess I am ok just can't do much getting out yet.'. |
17:07.58 | Inv_arp | isnt it possible on freenode to leave someone a message |
17:08.11 | *** part/#asterisk santiago (n=santiago@208.195.215.124) |
17:09.18 | demetrio | what does "failed to authenticate on INVITE" mean? |
17:09.44 | Math` | it means that the authentication failed on an SIP invite |
17:09.50 | Math` | which means: bad login |
17:10.19 | JunK-Y | Inv_arp: if the user is offline? memoserv |
17:11.03 | demetrio | well the problem I get is this one: with sipgate I'm able to receive calls but when I try to make an outgoing call i get the error above |
17:11.09 | Inv_arp | JunK-Y: ahh k thx |
17:11.15 | demetrio | so can't be bad login |
17:11.46 | many | whats your extension.conf partial for dialing to sipgate? |
17:12.05 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
17:12.39 | demetrio | exten => _6.,1,Dial(SIP/${EXTEN:1}@sipgate.co.uk,100,"T") |
17:12.53 | *** join/#asterisk Assid (n=assid@203.115.64.59) |
17:13.07 | many | do you have a peer section in sip.conf? |
17:13.09 | Qwell | well, no wonder it's failing... |
17:13.09 | demetrio | sure |
17:13.23 | demetrio | I can paste if it doesn' count as flood :) |
17:13.23 | Qwell | What is the name of the peer? Replace "sipgate.co.uk" with that |
17:13.31 | demetrio | the name of the peer is sipgate.co.uk |
17:13.40 | demetrio | [sipgate.co.uk] |
17:14.09 | many | does it contain u/p? |
17:14.25 | demetrio | sure |
17:14.28 | Qwell | I bet it's trying to use dns instead of the peername |
17:14.34 | Qwell | I don't know what order * does |
17:14.42 | many | me neither. |
17:14.50 | Qwell | demetrio: try making it just sipgate |
17:14.59 | Qwell | see if that changes the error at least |
17:15.05 | demetrio | well, it works with other providers (same structure, peer name equals host name) and the only one giving trouble is sipgate |
17:15.10 | demetrio | anyway I'll try |
17:15.26 | many | maybe the others need reg only, no auth for invite |
17:15.30 | Qwell | off to work |
17:16.48 | *** join/#asterisk vlrk (n=vlrk@59.93.67.217) |
17:17.02 | demetrio | nope, the error remains the same |
17:17.11 | Assid | hey.. what could be the issue if people are complaining of call dropping? |
17:17.16 | Math` | can you make a sip debug trace to pastebin? |
17:17.34 | many | sip debug peer <peername> |
17:17.50 | many | and dont forget to obfuscate passwords, incase theyre in. |
17:17.50 | mutilator | is there any way to silence registration failures in console yet? |
17:17.55 | bweschke | demetrio: post your sip debug trace to pastebin as suggested here, and also your sip.conf definition of sipgate (XXX'ing our user/pw) |
17:18.03 | mutilator | i have a guy that tries to register every second |
17:18.44 | demetrio | hmm I think I found the problem |
17:18.56 | demetrio | We're at 192.168.0.2 port 11246 |
17:19.11 | bweschke | huh? |
17:19.18 | justinu | that's just for RTP |
17:19.39 | demetrio | what does that mean? |
17:19.55 | demetrio | asterisk is telling sipgate to contact him on a local address |
17:19.58 | Math` | many: ah I didnt know about the "peer" option to sip debug, will be useful as hell |
17:20.02 | bweschke | that's the local resource/socket selected for RTP transmissions |
17:20.04 | justinu | it means it has nothing to do with your auth failure |
17:20.18 | ManxPower | RTP == AUDIO |
17:20.20 | many | Math`: what? limiting sip debug to only the peer you actually want to debug? :) |
17:20.25 | Math` | many: yeah :) |
17:20.29 | justinu | heh |
17:20.34 | many | Math`: yeah. :) |
17:20.39 | Math` | especially because my DID provider sends alot of MWI messages |
17:20.46 | Math` | (even if I have 0 new) |
17:21.04 | demetrio | I don't think so, the same debug block says INVITE sip:10000@sipgate.co.uk SIP/2.0 |
17:21.40 | bweschke | right... but if you're getting an auth failure, you're never getting to the point where the RTP is setting up |
17:21.43 | many | demetrio: Authentification is IN SIP, not in RTP. it just preallocates the RTP Socket. |
17:21.52 | bweschke | so - as justinu said: that's not the problem |
17:22.27 | bweschke | demetrio are you setting type=peer for both inbound/outbound in your sip.conf or type=friend ? |
17:22.31 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
17:22.50 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
17:23.26 | bweschke | there could be a cpl things happening here... 1) you're not matching your profile on the outbound connected, so * doesn't know to authenticate with the peer and is getting turned away |
17:23.56 | tmccrary | Do you know what would cause a buzzing sound with Asterisk / GSM and 250 ms latency? |
17:23.59 | bweschke | 2) sipgate may be particular about what's coming in the SIP headers, and you may need to look at fromuser= or something like that in your profile once you've verified that * is finding that profile on the dialout |
17:24.10 | Math` | demetrio: you should see "SIP/2.0 407 Proxy Authentication Required" followed by another packet with the line "Proxy-Authorization: Digest username="5144487726", realm="spectravoice.com", alg |
17:24.10 | Math` | orithm=MD5, uri="sip:4506861000@66.59.150.36", nonce="12345abc", response="abcdef12345678901234567890abcdef0", opaque="" |
17:25.04 | justinu | are there any sip DID providers that would sell me a block of 20 DIDs for a reasonable price? |
17:25.14 | Math` | justinu: for what area |
17:25.20 | justinu | los angeles |
17:25.21 | *** join/#asterisk bartpbx (n=bartpbx@proxy.prodyna.com) |
17:25.56 | bartpbx | I'm searching for markster. Whats his nick`? |
17:26.15 | paryl | what can i do about an echo on an iax connection? |
17:26.17 | Marcel-AS16215 | is this the right way to fix the circuit-busy ? Or is ther a better way to filter all the Messages out from my SIP Provider ? |
17:26.17 | Marcel-AS16215 | exten => _00049[23456789].,1,Dial(SIP/${EXTEN:1}@voipbuster-out,90) |
17:26.17 | Marcel-AS16215 | exten => _00049[23456789].,2,Busy(pls-try-call-later) |
17:26.17 | Marcel-AS16215 | exten => _00049[23456789].,3,Hangup |
17:28.30 | paryl | to clarify: i have an echo on an IAX connection between SIP phones on both servers |
17:28.44 | justinu | try turning down the volume on the sip phones |
17:28.50 | justinu | probably acoustic echo |
17:29.01 | paryl | justinu, it helps, but it doesn't fix it. the echo is still there, just quieter |
17:29.10 | justinu | what kind of phones? |
17:29.16 | paryl | polycom 501's |
17:29.20 | justinu | wow |
17:29.46 | justinu | anyways, if it's purely digital, the echo can only be acoustic |
17:30.01 | justinu | the reason you hear it over IAX, is probably the latency introduced by your WAN link |
17:30.23 | *** join/#asterisk flynux (i=bod6vc1@pingou.in) |
17:30.32 | paryl | ok, that makes sense. so is there any way to get rid of it? |
17:30.57 | justinu | reduce the latency, or figure out why the phones are hearing themselves |
17:31.04 | justinu | what polycom firmware? |
17:31.45 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
17:31.47 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
17:32.12 | paryl | it's the latest as of 2 weeks ago, don't remember the version number right off hand |
17:32.20 | justinu | 1.6.2 is the latest, iirc |
17:32.27 | *** join/#asterisk Gourou_fou (n=x@ACaen-151-1-20-125.w86-195.abo.wanadoo.fr) |
17:32.32 | Gourou_fou | olé |
17:32.37 | paryl | sounds right. the bootrom says 2.6.2 |
17:32.59 | paryl | right.. 1.6.2.0041 |
17:33.05 | justinu | ok, so firmware is good |
17:33.36 | justinu | when party A hears echo, have party B cover his mic |
17:33.41 | justinu | see if that makes it go away |
17:34.02 | Math` | nice accoustic echo test |
17:34.02 | justinu | if it does, you know for sure it's poor acoustic coupling |
17:34.56 | *** join/#asterisk Lurr (n=pr0ph3t@63.69.20.3) |
17:35.58 | paryl | it doesn't go away |
17:36.19 | justinu | bizzare |
17:36.39 | justinu | how about having party be hit MUTE? |
17:36.41 | justinu | party B |
17:37.38 | Math` | those phones are purely digital? |
17:37.45 | justinu | 501s? yeah |
17:37.46 | *** join/#asterisk jonarildh (n=vircuser@195.28.172.4) |
17:38.26 | *** join/#asterisk Utah_Dave (n=boucha@0-2pool130-248.nas28.salt-lake-city1.ut.us.da.qwest.net) |
17:38.32 | paryl | muting on the other side makes the echo go away |
17:38.41 | justinu | ok, so it's gotta be acoustic |
17:38.49 | justinu | paryl: i'd talk to polycom about that... seems wrong |
17:39.09 | jonarildh | someone that can help me with compile errors on Suse 10.0 and asterisk with bristuff-0.2.0-RC8f-CVS |
17:39.11 | tzanger | justinu: you can't just cover the mic though |
17:39.11 | jonarildh | ? |
17:39.19 | tzanger | it could be transmitted through the handset mechanically |
17:39.27 | justinu | tzanger: good point. |
17:39.28 | tzanger | best to disconnect the headset from the base |
17:39.36 | tzanger | most of them use the tiny 4-pin jacks |
17:39.55 | justinu | paryl: just know that the person that hears echo isn't the cause of it |
17:39.58 | justinu | it's the other sie |
17:39.59 | justinu | side |
17:41.25 | *** join/#asterisk JASON-0 (n=jason@jason.unitz.ca) |
17:41.39 | tmccrary | is there a way to have asterisk boost volume on calls/ |
17:41.53 | iCEBrkr | tmccrary: what type of calls? |
17:41.56 | tmccrary | SIPO |
17:41.57 | tmccrary | SIP |
17:42.07 | JASON-0 | Hello, I just setup a queue and no matter what settings I do, it sends the call to fail over when no one answers their phone. What I need is for the call to stay in the queue. |
17:42.07 | justinu | no |
17:43.05 | puzzled | is that normal: show translations ulaw -> gsm 105 & gsm -> ulaw 2 ? |
17:43.15 | tzafrir_laptop | jonarildh, bristuff "CVS" is quite old (from 29-may-2005) |
17:43.18 | justinu | heh |
17:43.22 | paryl | she has a headset on the other end, so i'm having her disconnect it to see it that's the problem |
17:43.45 | justinu | oh, headset problems |
17:43.55 | justinu | :P |
17:44.05 | *** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net) |
17:44.11 | justinu | don't buy cheap headsets :) |
17:44.20 | justinu | polycom 501 audio is almost a work of art, it deserves better |
17:45.10 | paryl | well.. they're nice plantronics headsets |
17:45.17 | justinu | hmm |
17:45.21 | paryl | but i'm wondering if it's a setting with the amp |
17:45.22 | justinu | maybe it's defective |
17:45.31 | justinu | but yeah, could be lots of stuff |
17:47.28 | Assid | can someone help me thing up what could be the issue for this.. calls just drop on their own.. |
17:47.40 | Assid | i am thinking is maybe cause i have qualify=yes |
17:47.52 | Assid | and i have seen the phone disappear for 2 seconds |
17:49.04 | Assid | coulod it be because of that |
17:49.35 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
17:50.18 | paryl | ok, that settles it... it's the headset |
17:50.37 | paryl | i think it's the gain on the amp... but after unplugging it the echo went away |
17:50.40 | justinu | cool |
17:51.42 | perd | a quad span t1 card can run on a dual 3ghz system with 2gig ram right? |
17:51.49 | perd | i'd have the channels used most of the time |
17:52.26 | perd | in the book it says 15 hcannels requires a 3ghz 1gb ram.. yet a quad span is more than 4x the amount of calls they say it can handle |
17:54.26 | zoa | yes it can |
17:54.36 | perd | ok |
17:54.53 | perd | i dunno wtf the book is talking about saying that for more than 15 channels you need to spread it across multiple servers |
17:55.04 | perd | unless by channel they mean t1 |
17:55.17 | Math` | so a T3 would require 2 servers probably |
17:55.58 | Math` | or some kinda hypercube-architecture-based server |
17:56.30 | *** join/#asterisk Abbas (n=Abbas@203.81.216.47) |
17:56.38 | perd | my pbx is powered by the ball sweat of NFL athletes |
17:56.45 | perd | don't ask me how i harvest it. |
17:58.05 | Math` | lol |
18:03.52 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
18:04.09 | *** join/#asterisk leibniz11 (n=leibniz@163.1.203.62.cust.bluewin.ch) |
18:04.24 | JASON-0 | I need help setting up a queue in asterisk.. can someone help me ? |
18:04.48 | yxa | can someone help me test if my isp blocks my sip port @ 202.156.228.121? |
18:04.53 | leibniz11 | is it possible to have asterisk transcode between SIP and MGCP? I have a sipura and want to have it talk to my CCM 3.3.X |
18:05.21 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
18:06.02 | justinu | asterisk MGCP can only talk to phones |
18:06.12 | justinu | it can't talk to other gatekeepers or media gateways, iirc |
18:07.35 | demetrio | well, looks like I'm not smart enough to solve the problem |
18:07.38 | leibniz11 | how can you communicate asterisk with a non-sip CCM? |
18:07.43 | demetrio | so here you are: http://pastebin.com/421947 |
18:08.12 | justinu | leibniz11: don't think you can |
18:09.15 | yxa | can someone call me @ 202.156.228.121? i'm not sure i setup sip correctly... |
18:10.39 | Math` | which country is that |
18:10.45 | yxa | singapore |
18:10.58 | Math` | +202.156.228.121? |
18:11.08 | yxa | er SIP |
18:11.24 | Math` | ah thats an ip |
18:11.29 | justinu | heh |
18:11.31 | Math` | what extension |
18:11.34 | yxa | 2222 |
18:11.37 | Math` | k |
18:12.40 | Math` | <PROTECTED> |
18:12.40 | Math` | <PROTECTED> |
18:13.10 | yxa | i'm not seeing anything on my console. |
18:14.22 | justinu | demetrio: try setting fromdomain=sipgate.co.uk |
18:14.27 | Math` | yxa: http://pastebin.ca/28014 |
18:14.45 | Math` | extension 2222 doesnt exist on your side |
18:15.13 | Math` | (well doesnt exist in the guest context you've defined) |
18:16.15 | demetrio | justinu: thanks, it works :) |
18:16.26 | yxa | Math` so it did connect? |
18:16.26 | demetrio | why is that? |
18:16.52 | Math` | SIP/2.0 404 Not Found |
18:17.22 | yxa | i have exten => 2222,1,Dial(SIP/grandstream) under [test] in extensions. is that rite? |
18:17.46 | Math` | thats ok, in your sip.conf file, check for context=test |
18:18.05 | justinu | demetrio: wrong authentication realm if you don't put that |
18:18.14 | yxa | Math` under [general]? |
18:18.29 | Math` | yxa: well under the sip.conf user you have for accepting guest sip calls |
18:18.58 | jonarildh | tzafrir_laptop: I have downloaded the latest zaptel from cvs, but still errors on compile |
18:19.07 | *** join/#asterisk darwin_35 (n=darwin35@208.139.193.178) |
18:19.07 | yxa | Math` sorry i'm not sure how to do that. been trying to look at wikis |
18:19.21 | Math` | under general then |
18:19.24 | Math` | whats the context defined |
18:19.37 | yxa | default |
18:19.49 | Math` | you probably want that changed to: test |
18:20.14 | Math` | now all sip calls will jump to context "test" unless specified otherwise in the sip user entry |
18:20.15 | yxa | Math` ok. pls try |
18:20.24 | Math` | rings :) |
18:20.37 | yxa | Math` great thanks |
18:20.41 | Math` | did u hear me? |
18:20.53 | tzafrir_laptop | jonarildh, right, because it is not intended for latest CVS |
18:20.54 | yxa | no i didn't. i said hello |
18:21.02 | Math` | is your asterisk box nat'ed |
18:21.11 | Math` | I heard you but I guess you had no sound on your end |
18:21.17 | *** part/#asterisk darwin_35 (n=darwin35@208.139.193.178) |
18:21.37 | yxa | Math` well 202.156.228.121/5060 is port forwarded to my box inside 192.168.2.10 |
18:21.45 | Math` | you need to forward RTP ports too |
18:21.58 | yxa | which are? |
18:22.16 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
18:22.23 | Math` | by default |
18:22.24 | Math` | rtpstart=10000 |
18:22.24 | Math` | rtpend=20000 |
18:22.28 | Math` | you can change them in rtp.conf |
18:22.39 | *** join/#asterisk miguel3239 (n=myoung@h-68-167-124-170.cmbrmaor.covad.net) |
18:22.56 | yxa | udp? |
18:22.59 | Math` | yes |
18:23.24 | tmccrary | i thought rtp was its own protocol? |
18:23.35 | Math` | no rdp is over udp |
18:23.40 | yxa | dont think my home router has that function of forwarding a whole range :( |
18:23.55 | tmccrary | ahh, I see. Thank you for the correction |
18:24.00 | Math` | tmccrary: it adds sequence numbers etc over udp to ensure that stuff is delivered in the right order |
18:24.16 | tmccrary | kind of like weak tcp |
18:24.19 | tmccrary | without the overhead |
18:24.43 | sjobeck99 | hi, all, it tells me no class "default" for MOH, but I commented that out since I want 'native-random'. what am I misunderstanding? how does one use native-random? |
18:24.47 | Blackthorn | Hello. I have a SPA-2000 ip 192.168.1.2 behind a nat router 192.168.1.1. On the spa menu nat = off. The Asterisk server has nat = yes and qualify = yes. The problem is that one of the two lines keep droping registration. REbooting the router, the * server, or the sipura dosn't fix. But turning both lines off on the sipura, reboot, then both lines back on does. Thoughts? |
18:25.00 | Math` | tmccrary: yeah, and not connection-oriented I think.... see ftp://ftp.rfc-editor.org/in-notes/rfc1889.txt for more info |
18:26.04 | yxa | Math` can you try again? |
18:26.51 | Rowter | anyone knows if micronet SP5054 could connect to a sip voip reseller? |
18:27.11 | Math` | yxa: u getting any sound? |
18:27.12 | sjobeck99 | what is micronet sp5054 ? |
18:27.18 | yxa | Math` no dude |
18:27.37 | justinu | qualify = yes means how often? |
18:28.06 | Dr-Linux | i have install zaptal, and loaded ztdummy module as well, but meetme is not working .. any clue ? |
18:28.16 | Dr-Linux | i have not hardware |
18:28.18 | Rowter | sjobeck99, a voip gateway |
18:28.20 | Dr-Linux | i'm using soft client |
18:29.11 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
18:29.13 | Math` | Dr-Linux: any log emssages? |
18:29.17 | Math` | messages, that is |
18:29.24 | justinu | dr-linux: did you create the udev stuff? |
18:30.33 | Dr-Linux | i just created a 1111 room in meetme.conf and define an extension 1112 in extension.conf, is it enough to work confierence call ? |
18:30.41 | sjobeck99 | http://www.micronet.info/Products/voip/SP5054.asp says that no it wont, if we're reading it correctly |
18:30.42 | tmccrary | What is the best free codec available? GSM has bad quality and PCMU/ulaw takes up too much bandwidth |
18:30.50 | Katty | i just /love/ how polycom won't talk to me |
18:30.54 | justinu | dr-linux: you need to have the proper /dev entries for zaptel |
18:30.55 | tmccrary | I guess I could go with pass-thru |
18:30.56 | yxa | Math` so this is the famous sip nat problem |
18:31.04 | tmccrary | sip + nat = hell |
18:31.06 | tzanger | tmccrary: I have no issue with gsm |
18:31.09 | Math` | yxa: actually its an RTP nat problem |
18:31.14 | tzanger | you could try speex |
18:31.30 | justinu | gsm makes me realize how shitty cellphones sound |
18:31.32 | sjobeck99 | arent we all quite used to gsm from using our mobile phones every day? |
18:31.33 | Rowter | sjobeck99, it wont mmh |
18:31.38 | Math` | justinu: HEHEHE |
18:31.41 | tmccrary | GSM is okay, I mean its understandable. But not even as good as regular telephone |
18:31.50 | justinu | tmccrary: agreed |
18:31.52 | sjobeck99 | ilbc, perhaps |
18:31.58 | Math` | yxa: you can set the externip= parameter in sip.conf |
18:31.59 | tmccrary | Yeah, we use it for mobile phones... but they are mobile |
18:32.12 | *** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net) |
18:32.25 | yxa | Math` i idid |
18:32.30 | justinu | dr-linux: check README.udev in the zaptel directory |
18:32.30 | Blackthorn | justinu: i'm not sure how often. it's what ever the default is for qualify=yes. I belive I coudl put qualify=# |
18:32.47 | *** join/#asterisk vlrk (n=vlrk@59.93.67.38) |
18:32.48 | sjobeck99 | looks like that micronet wants to connect to another same box on other end |
18:33.00 | Math` | yxa: you are sure all rtp ports are forwarded? |
18:33.14 | justinu | blackthorn: it sounds like your NAT might be closing the nat bindings |
18:33.14 | tmccrary | has anyone here used a Sipura SPA-841. Is it buggy and weird for you too? |
18:33.25 | Dr-Linux | justinu: let me check |
18:33.26 | justinu | sipura 841 is ok, except for the shitty buttons |
18:33.32 | Math` | Dr-Linux: sounds like your missing app_meetme.so |
18:33.33 | Blackthorn | One strange thing is that the spa says line one uses 5060 and line two uses 5061. but on the * server they show using 1030 and 5060 |
18:33.41 | vlrk | my voicemail takes /var/lib/asterisk/sounds/voicemail where as it has to take /var/spool/asteirsk/voicemail |
18:33.44 | Math` | Dr-Linux: which * version are u running |
18:33.59 | justinu | math, i think his problem is that he didn't create the zaptel entries in /dev |
18:33.59 | Dr-Linux | Math`: 1.0.9 |
18:34.13 | Dr-Linux | yes |
18:34.20 | justinu | app_meetme wont load unless they're there |
18:34.25 | yxa | Math` i'm trying out this special application portion of my home router |
18:34.25 | Dr-Linux | i didn't create any entries in /dev |
18:34.28 | vlrk | is the issue with addmailbox script |
18:34.31 | Blackthorn | justinu: What does that mean and how do I correct it.. is it settings ont he spa? the nat router, or * server? |
18:34.40 | perd | is there some kind of null channel i can specify in my .call files? |
18:34.53 | sjobeck99 | any one familiar with 'native-random' music on hold ? |
18:34.56 | perd | so i can just have it traverse my extensions without having a connection? |
18:35.02 | Math` | justinu: [13:33] <Dr-Linux> Nov 8 23:30:26 WARNING[2561]: pbx.c:1293 pbx_extension_helper: No application 'Meetme' for extension (default, 1112, 1) |
18:35.07 | Math` | justinu: he pasted that to me in privmsg |
18:35.14 | yxa | Math` fire away |
18:35.22 | justinu | blackthorn: maybe on the SPA, set "nat keep alive enable" |
18:35.26 | justinu | math: ah |
18:35.34 | Math` | sound over yours? |
18:35.41 | yxa | Math` nope. |
18:35.52 | yxa | Math` ok, can i jus forward 2 rtp ports? |
18:35.56 | Math` | no |
18:36.09 | Math` | why don't u set a dmz on your * box? |
18:36.10 | Dr-Linux | justinu: i have no hardware, i wanna use ztdummy, still i need work with /dev right ? |
18:36.20 | justinu | Dr-Linux: correct |
18:36.37 | Math` | why do people stick with 1.0.9 :( |
18:36.50 | Blackthorn | justinu: there you go. sure enophe. line two keep alive enable was turned off. Thanks! |
18:36.57 | azzie | Math`, because 1.2 is far from 1.0.9 in stability |
18:36.59 | justinu | Blackthorn: no prob |
18:37.05 | yxa | Math` dmz? i only have a home dynamic ip connection ;) |
18:37.11 | Math` | you've a router? |
18:37.24 | yxa | Math` yeah, a very low end one |
18:37.42 | Math` | you probably have a DMZ option, this option is going to forward all traffic to the ip you specify in there |
18:37.47 | Math` | even very low end ones have that |
18:38.18 | yxa | Math` oh i saw it |
18:38.20 | Katty | does anyone have to reboot their polycom500s on a regular basis? |
18:38.28 | Katty | if by regular i mean every 10-12 calls? |
18:38.32 | yxa | Math` fire :) |
18:39.12 | Math` | i got no sound |
18:39.14 | Math` | u got any? |
18:39.16 | yxa | me neither |
18:39.23 | Katty | yes? no? |
18:39.25 | Katty | maybe so? |
18:39.27 | justinu | katty: no |
18:39.32 | Katty | justinu: thanks for answering |
18:39.39 | Katty | does anyone /else/ have problems with their polycoms? |
18:39.41 | justinu | but i have 501s |
18:39.49 | justinu | not much diff tho |
18:39.52 | Katty | specifically they can hear you, but you can't hear them! |
18:40.14 | yxa | Math` i guess it doesnt work |
18:40.26 | Math` | you've rebooer the router after setting the dmz? |
18:40.35 | Dr-Linux | justinu |
18:40.42 | justinu | katty: if you pastebin a sip debug from one of the bad calls, we might be able to diagnose it |
18:40.52 | Katty | justinu: k |
18:41.16 | yxa | Math` do i expose * to the dmz or the phone? |
18:41.27 | Math` | * |
18:41.44 | Math` | * is gonna do proxying for the rtp stream |
18:41.49 | *** join/#asterisk luke-jr_ (n=luke-jr@user-0c938qu.cable.mindspring.com) |
18:42.02 | Dr-Linux | justinu: the path in README.xxx file is not in my machine :S or i didn't understand |
18:42.10 | yxa | Math` yeah. didn't ask for a reboot |
18:42.19 | Math` | reboot it anyways |
18:42.22 | yxa | Math` but i'll do it anyway. hang on |
18:42.28 | IronHelix | hmmm |
18:42.33 | IronHelix | justinu were you looking for me? |
18:42.59 | justinu | yeah, but i forgot why :P |
18:43.04 | Math` | haha |
18:43.10 | IronHelix | lol |
18:43.11 | IronHelix | GJ |
18:43.18 | justinu | that was yesterady |
18:43.20 | Rowter | sjobeck99, know a 4 fxo gateway to connect a panasonic that could connect to a voip reseller? |
18:43.22 | justinu | whole different day and stuff |
18:43.26 | mutilator | anyone here have callerid with name in michigan? |
18:43.27 | IronHelix | way too long ago |
18:43.29 | Dr-Linux | justinu: /etc/udev/rules.d/50-udev.rules << this file is not located at the given path |
18:43.34 | mutilator | anyone here have callerid with name, in michigan |
18:43.41 | justinu | Dr-Linux: what linux distro? |
18:43.53 | Math` | Rowter: if the gateway is sip, I got no reason why it wont connect to a voip provider |
18:43.55 | Dr-Linux | justinu: RHEL |
18:44.01 | Math` | plus... are you sure you don't mean 4x fxs? |
18:44.05 | justinu | Dr-Linux: RHE4? |
18:44.21 | Dr-Linux | justinu: RHEL 3 update 6 |
18:44.25 | Nugget | Linux is poo. |
18:44.27 | justinu | hmm |
18:44.35 | justinu | maybe RHEL3 doesn't use udev? |
18:44.45 | Dr-Linux | hhm.. |
18:44.45 | sjobeck99 | rowter: not really. huh? must be one. I presume you cant use * to do that? |
18:44.52 | justinu | Dr-Linux: just for reference, I got ztdummy working on FC4 |
18:45.00 | justinu | so I only know specifics for that |
18:45.04 | Dr-Linux | justinu: how can i verify .. |
18:45.14 | Math` | justinu: ztdummy can work on any kernel |
18:45.15 | Rowter | Math`, thats what I tought but some of them are expecting another of their kind to make a network.. mmh Mediatrix 1204 might do the trick |
18:45.23 | justinu | Dr-Linux: ps -ef | grep udev |
18:45.29 | Dr-Linux | okey |
18:45.29 | justinu | is udevd running? |
18:45.42 | *** join/#asterisk axscode (n=axscode@203.213.217.122) |
18:45.42 | Math` | Rowter: for mediatrixes.... even if they are profiled for a specific provider you can manage to flash it |
18:45.52 | Math` | Rowter: I found firmwares on a russian site for my 2102 |
18:45.57 | Dr-Linux | [root@RHEL-TAC-TEST zaptel-1.0.9.2]# ps -ef | grep udev |
18:45.57 | Dr-Linux | root 4402 2481 0 23:45 pts/1 00:00:00 grep udev |
18:45.57 | Dr-Linux | [root@RHEL-TAC-TEST zaptel-1.0.9.2]# |
18:46.03 | justinu | ok, so no |
18:46.13 | Dr-Linux | yep .. :S |
18:46.24 | justinu | Dr-Linux: that means you need to create the /dev/sappseudo entry manually I think |
18:46.33 | justinu | er /dev/zappseudo |
18:46.51 | Rowter | Math`, ahh, the idea is to connect it to a panasonic pbx to get some extensions with voip rates with a reseller .. |
18:47.27 | sjobeck99 | rowter: sounds like a perfect place to install * to intercept 'those' calls |
18:47.39 | Math` | Rowter: so you want to connect 4 outgoing lines to the 1204 |
18:47.45 | Dr-Linux | justinu: should i create zappseudo file or dir ? |
18:48.01 | sjobeck99 | going, going, gone, any one familiar with native MOH ? |
18:48.07 | Math` | Rowter: where did u buy the unit? |
18:48.12 | justinu | Dr-Linux: somehow, it needs to get created if it doesn't exist. |
18:48.15 | Rowter | sjobeck99, no, because asterisk does not work well with the panasonic hangup singal. |
18:48.32 | justinu | Dr-Linux: as far as how to do that, i'm unsure |
18:48.33 | Dr-Linux | justinu: yeah but is it a file? or folder ? |
18:48.37 | justinu | file |
18:48.40 | Rowter | Math`, I think I'll buy it from voipsupply |
18:48.42 | justinu | it's a dev link |
18:48.46 | Dr-Linux | ok |
18:48.48 | justinu | it has a major and minor number |
18:48.52 | Math` | Rowter: ok... mediatrix are configurable only via snmp |
18:49.03 | Math` | Rowter: so you gotta have a win32 box nearby to install UMN (Unit Manager Network) |
18:49.10 | Math` | Rowter: the free version allows you to manage 3 units |
18:49.14 | tzafrir_laptop | /dev/zap/pseudo |
18:49.37 | Math` | Rowter: usually, they provide you the cd with the installer and the firmware for the unit, if they don't, email me |
18:50.02 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
18:50.08 | justinu | on my system, /dev/zap/pseduo is mj 196, mn 255 |
18:50.27 | justinu | also, /dev/zap/timer is mj 196, mn 253 |
18:50.27 | Dr-Linux | ooo |
18:50.36 | Dr-Linux | justinu: this file already exist |
18:50.37 | justinu | not sure if that's the same for every box |
18:50.38 | Dr-Linux | cd /dev/zap/pseudo |
18:51.03 | justinu | Dr-Linux: ok, so then that's not it... check your /var/log/asterisk/messages |
18:51.14 | justinu | Dr-Linux: there should be some clue why app_meetme won't load |
18:51.14 | Dr-Linux | but its empty |
18:51.21 | Dr-Linux | ok |
18:51.28 | justinu | Dr-Linux: to a reload from the CLI |
18:51.33 | justinu | it forces it to reload everything |
18:51.36 | *** join/#asterisk Frosted (n=Procrast@buzzbud.plus.com) |
18:51.43 | Katty | justinu: not everything everything |
18:52.09 | justinu | well, for his purposes |
18:52.20 | Katty | porpoises. |
18:52.32 | jonarildh | Have anyone got asterisk 1.2-b2 to run with hfc ISDN cards? |
18:53.11 | Dr-Linux | justinu: this is logs >> Nov 8 23:52:15 WARNING[2561]: No application 'Meetme' for extension (default, 1112, 1) |
18:53.30 | bweschke | ahh.. well that helps.. :) |
18:53.36 | justinu | Dr-Linux: modify logger.conf, uncomment the full line |
18:53.40 | pooh_ | Dr-Linux: what does show modules show you? |
18:53.51 | axscode | Nov 10 14:42:55 WARNING[11601] loader.c: /usr/lib/asterisk/modules/res_config_mysql.so: cannot open shared object file: No such file or directory |
18:53.52 | axscode | Nov 10 14:42:55 WARNING[11601] loader.c: Loading module res_config_mysql.so failed! |
18:53.53 | tzafrir_laptop | <PROTECTED> |
18:53.55 | justinu | Dr-Linux: then restart asterisk and look in /var/log/asterisk/full |
18:53.55 | bweschke | Dr-Linux: check to see if app_meetme.so shows up in /usr/lib/asterisk/modules |
18:53.59 | axscode | is there anything I have to do during installation? |
18:54.10 | justinu | bweschke: oh, that's a good point |
18:54.14 | justinu | heh |
18:54.23 | pooh_ | axscode: you need to compile,install and copy over some files from asterisk-addons |
18:54.37 | hardwire | hmmhmpmhm |
18:54.51 | *** join/#asterisk TheCops (n=xdz@206-248-136-187.dsl.teksavvy.com) |
18:54.53 | TheCops | Hi |
18:54.53 | pooh_ | axscode: the res_mysql conf file has to be copied over manually |
18:54.57 | hardwire | does anybody know if asterisk has the ability to set its source address? |
18:55.03 | bweschke | dr-linux: if you compile and make install asterisk before compiling and make installing zaptel, app_meetme will not build because it doesn't see zaptel around for a timing source |
18:55.07 | hardwire | based on which address a sip client accessed? |
18:55.13 | justinu | Dr-Linux: that's true, i had that problem as well |
18:55.14 | pooh_ | hardwire: SIP.conf and IAX2.conf |
18:55.21 | hardwire | pooh_: dynamically |
18:55.26 | Dr-Linux | ooo |
18:55.28 | axscode | what do you mean pooh_? |
18:55.29 | justinu | Dr-Linux: recompile your source, and do a make install |
18:55.40 | TheCops | Someone know if there's a way to put 200 PSTN line to an asterisk server ? |
18:55.52 | bweschke | TheCops: what kind of PSTN line? |
18:55.53 | Dr-Linux | bweschke: thats not there app_meetme.so |
18:55.56 | TheCops | bweschke, analog |
18:55.58 | Katty | doesn't asterisk start to combust after 120 calls? |
18:56.01 | Dr-Linux | all modules are there , but thats not :S |
18:56.10 | justinu | wow, 200 analog lines |
18:56.11 | pooh_ | axscode: you need to copy over the mysql conf file manually to /etc/asterisk from asterisk-addons/scrips |
18:56.21 | TheCops | justinu, yeah, that's what I said when I saw that shit. |
18:56.22 | justinu | 200 FXO lines? |
18:56.22 | TheCops | lol |
18:56.30 | justinu | or FXS? |
18:56.35 | pooh_ | hardwire: nope, sorry. Maybe with mysql backend |
18:56.40 | TheCops | FXO |
18:56.43 | justinu | i'd think SIP channel banks might be your best option |
18:56.51 | Katty | i love how no one answers me. heh |
18:56.57 | bweschke | well... you can put a couple of the new TDM2400 cards in a box, but I don't think you're going to get 10 of them in a box.. better off to do multiple boxes and use dundi or something like that |
18:57.06 | Dr-Linux | justinu: how can i get that module ? |
18:57.08 | Dr-Linux | bweschke |
18:57.20 | justinu | Dr-Linux: does the module exist in /var/lib/asterisk/modules? |
18:57.28 | mog_work | yeah 10 of anything is crazy |
18:57.37 | bweschke | Dr-Linux: if you've compiled and "make install"'d zaptel, now go ahead and "make clean" and then "make" and "make install" on asterisk again |
18:57.39 | Dr-Linux | justinu: no, |
18:57.41 | TheCops | ho my god, justinu |
18:57.42 | pooh_ | justinu: /usr/lib/asterisk/modules |
18:57.43 | TheCops | your nice |
18:57.49 | TheCops | justinu, that's what I searched. |
18:57.50 | justinu | ? |
18:57.57 | justinu | oh |
18:58.02 | bweschke | the build process should then see zaptel and built app_meetme.so because it has a timing source |
18:58.21 | Dr-Linux | bweschke: ok let me try |
18:58.23 | bweschke | Katty: no. asterisk doesn't start to combust after 120 calls |
18:58.24 | TheCops | justinu, a hannel banks is very nice |
18:58.35 | Katty | bweschke: that's not what i've been told. |
18:58.37 | Math` | bweschke: MeetMe() need zaptel for mixing functions |
18:58.42 | Math` | not for timing |
18:58.43 | justinu | thecops: i think it's a better solution than putting cards into a PC |
18:58.50 | TheCops | lol |
18:59.02 | pooh_ | simply test with ztdummy guys |
18:59.04 | TheCops | justinu, dont worry, I dont had this idea, this is a crazy idea heh |
18:59.11 | justinu | easier to maintain, you won't have to shut off the server if one breaks |
18:59.31 | pooh_ | but meetme *will* load without any *zap*isch stuff arund |
18:59.37 | bweschke | Math`: MeetMe needs zaptel to create a zap psuedo channel. the psuedo channel requires a timing source. you can't create it w/o one |
19:00.04 | Math` | bweschke: true, but still, MeetMe uses mixing functions of zap |
19:00.17 | bweschke | Math`: yes. it does |
19:00.18 | TheCops | justinu, this is converting all analog line plugged into the channels banks equipement to a PRI for Asterisk ? right ? |
19:00.25 | pooh_ | MeetMe *will* load without zaptel-thingy around, the error says it is NOT loaded, so it is something else |
19:00.35 | Math` | pooh_ got a point |
19:00.41 | justinu | thecops: no, more like converting analog lines to VoIP (sip signalling) |
19:00.43 | *** part/#asterisk frenzy (n=frenzy@193.220.82.108) |
19:00.53 | justinu | SIP channel bank plugs into ethernet |
19:01.10 | TheCops | ok |
19:01.16 | justinu | why waste your money on a PRI channel bank? TDM is out |
19:01.18 | Frosted | Anyone know where to get an up to date list of dial parameters? |
19:01.27 | justinu | show application dial |
19:01.31 | pooh_ | Frosted: voip-info.org |
19:01.44 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
19:01.48 | pooh_ | justinu: even better :-) |
19:01.48 | Dr-Linux | bweschke: i just did, cd zaptal dir, make clean, make, make install |
19:01.49 | bweschke | pooh_: meetme will not load w/o zaptel. it needs to create a zap psuedo channel to function |
19:01.49 | TheCops | justinu, very nice, this is a good idea, I can use Hints function to see the status of the channel |
19:02.00 | axscode | ~PRI |
19:02.01 | jbot | i guess pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
19:02.11 | Dr-Linux | and i have checked, this module is still not there :S |
19:02.12 | pooh_ | bweschke: yes to *function* but not to *load* |
19:02.28 | justinu | Dr-Linux: no no no, rebuild your asterisk source, no zaptel! |
19:02.31 | pooh_ | bweschke: the error says it is NOT loaded |
19:03.19 | bweschke | pooh_: yes, I know. Dr-Linux's problem is different than zaptel being there. he built asterisk before zaptel was around so the build process didn't build the module |
19:03.22 | axscode | ~h323 |
19:03.23 | jbot | somebody said h323 was An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is heavily dependent on ... |
19:03.55 | Dr-Linux | oooo ic |
19:04.07 | Dr-Linux | yeah i install asteris 1st |
19:04.20 | justinu | just recompile, man :) |
19:04.21 | Dr-Linux | justinu: but what about my current asterisk configuration ? |
19:04.24 | justinu | then make install |
19:04.30 | justinu | it won't disturb your config |
19:04.38 | bweschke | dr-linux: unless you do make samples which you should NOT do, it will leave /etc/asterisk alone |
19:04.40 | axscode | not untill make sample? |
19:04.44 | justinu | you should probably tar it up, just to be safe anyways |
19:04.44 | pooh_ | bweschke: he needs to start over |
19:04.53 | Dr-Linux | i don't wanna loose sip.conf extension... etc |
19:04.59 | Math` | ah ok he didnt follow the order on the site |
19:05.01 | bweschke | u won't |
19:05.06 | justinu | dr-linux: backup time |
19:05.06 | bweschke | just don't do "make samples" |
19:05.07 | Math` | Dr-Linux: make install , u wont loose it |
19:05.14 | axscode | cp -vrf /etc/asterisk* ~/ |
19:05.20 | Math` | Dr-Linux: I do that all the time when cvs update -dP'ing |
19:05.22 | bweschke | "make install" will only take the asterisk bin and compiled modules and move them to the build dir |
19:05.25 | justinu | you won't lose it, but if you're afraid of losing your configs, you should back it up |
19:05.40 | justinu | what if your hard drive dies in 30 seconds? |
19:05.50 | blitzrage | mine died last night |
19:05.57 | justinu | that sucks |
19:05.58 | Dr-Linux | oo okey cool let me do it |
19:06.04 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
19:06.05 | justinu | lose all your porn? :P |
19:06.42 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
19:06.53 | Math` | justinu: yeah that sucks |
19:08.05 | justinu | lifetime of the component |
19:08.08 | justinu | 2 seconds :) |
19:08.14 | justinu | in the case of IBM Deathstars |
19:08.20 | Dr-Linux | i'm comiling sterisk again now |
19:08.22 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
19:08.36 | axscode | how to install addons? |
19:08.47 | pooh_ | axscode: download it and complie ;-) |
19:08.55 | axscode | make;make install ? |
19:08.59 | pooh_ | yup |
19:09.09 | pooh_ | but configs have to be copied over manually |
19:09.18 | axscode | ohh. got errors.. where should it resides..? |
19:09.33 | axscode | inside /usr/src/asterisk/asterisk-addons <--should be fine? |
19:09.35 | pooh_ | axscode: make sure you have apache-devel installed if you want to use mysql backend |
19:09.43 | axscode | hmmm ok.. |
19:09.55 | pooh_ | '/usr/src/asterisk-addons |
19:10.23 | Dr-Linux | oo yes i got now that |
19:10.24 | Dr-Linux | [root@RHEL-TAC-TEST modules]# cd app_m |
19:10.24 | Dr-Linux | app_macro.so app_meetme.so app_milliwatt.so app_mp3.so |
19:10.48 | justinu | dr-linux: *clap* |
19:11.04 | axscode | lol |
19:11.20 | Dr-Linux | justinu: but sir everything is if call conference works :P |
19:11.28 | justinu | it'll work |
19:11.38 | axscode | pooh.. can I just instlal res_mysql..? |
19:11.53 | pooh_ | Dr-Linux: post a conference room number and we will test :P |
19:12.14 | pooh_ | axscode: make install will do that for you |
19:12.15 | axscode | what's the difference between res_mysql and res_config_mysql ??? |
19:12.36 | pooh_ | axscode: res_mysql = telling asterisk how to use mysql |
19:12.50 | pooh_ | res_config_mysql = telling * mysql is available |
19:12.53 | oogle | is anyone else having problems compiling res_config_odbc in head? |
19:12.57 | Dr-Linux | pooh_: its locally |
19:13.01 | Dr-Linux | justinu: same problem |
19:13.07 | justinu | lol |
19:13.11 | axscode | so I really need both? or res_mysql is just fine.. |
19:13.16 | pooh_ | Dr-Linux: wel....INVITE us over!!!!!!!!!!!!! and party! |
19:13.25 | pooh_ | axscode: both |
19:13.30 | axscode | o sad |
19:13.31 | justinu | Dr-Linux: well, now we need to figure out why app_meetme won't load |
19:13.45 | justinu | Dr-Linux: is the ztdummy module loaded in the kernel? lsmod to find out |
19:15.18 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
19:15.18 | axscode | what should be in my /etc/zaptel.con for ztdummy? |
19:15.22 | ManxPower | Zaptel needs to be installed BEFORE Asterisk is built/installed or chan_zap, app_meetme, etc won't be built |
19:15.27 | pooh_ | axscode: nothing |
19:15.33 | Dr-Linux | justinu: i load the before doing this >> modprobe ztdummy |
19:15.37 | justinu | manx: we already resolved that problem |
19:15.43 | pooh_ | axscode: just modprobe zaptel and modprobe ztdummy |
19:16.04 | axscode | gut problem with modeprobe ztdummy though.. |
19:16.04 | Dr-Linux | justinu: its shows |
19:16.05 | Dr-Linux | [root@RHEL-TAC-TEST modules]# lsmod |
19:16.05 | Dr-Linux | Module Size Used by Not tainted |
19:16.05 | Dr-Linux | ztdummy 2464 0 (unused) |
19:16.05 | Dr-Linux | ztdynamic 8544 0 (unused) |
19:16.05 | Dr-Linux | zaptel 179872 0 [ztdummy |
19:16.09 | pooh_ | axscode: comment out *ALL* lines in zaptel.conf |
19:16.17 | justinu | Dr-Linux: ok, then modify logger.conf |
19:16.23 | justinu | Dr-Linux: uncomming the line for "full" |
19:16.24 | pooh_ | Dr-Linux: is on his way |
19:16.24 | axscode | all? |
19:16.33 | justinu | Dr-Linux: restart asterisk, and check /var/log/asterisk/full |
19:16.38 | pooh_ | axscode: yes, unless you have a real zap device |
19:17.16 | axscode | don't have a device |
19:17.25 | pooh_ | axscode: uncomment all stuff then |
19:18.24 | axscode | nice.. twas loaded.. |
19:18.29 | axscode | great help |
19:19.12 | axscode | but I have to search for rpm of apache-devel |
19:19.14 | axscode | for SuSE 10 |
19:19.34 | Dr-Linux | justinu: there is no full file in the /var/log/asterisk/ dir ? |
19:19.37 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
19:19.41 | Dr-Linux | this is my CLI> error |
19:19.42 | Dr-Linux | Nov 9 00:18:20 WARNING[2561]: pbx.c:1293 pbx_extension_helper: No application 'Meetme' for extension (default, 1112, 1) |
19:19.42 | Dr-Linux | <PROTECTED> |
19:19.50 | justinu | Dr-Linux: yeah, that's not enough |
19:19.59 | justinu | Dr-Linux: did you restart? |
19:20.04 | justinu | Dr-Linux: after moding logger.conf |
19:20.12 | Dr-Linux | just i reload ? |
19:20.20 | justinu | Dr-Linux: no, stop now, and restart |
19:21.05 | Dr-Linux | oo i did |
19:21.12 | Dr-Linux | i restart |
19:21.18 | Dr-Linux | and i can see new things on CLI |
19:21.21 | Dr-Linux | check your pvt |
19:21.38 | justinu | ok, so it looks good |
19:21.41 | justinu | app loaded |
19:23.06 | Assid | is there anything we can do for echo? |
19:23.13 | Dr-Linux | justinu: when i diald , it says "currently you are only person in the conffernece :P ;) |
19:23.28 | justinu | then it's working |
19:23.33 | justinu | enjoy |
19:23.57 | pooh_ | justinu: job well done, cheers |
19:23.58 | Dr-Linux | great justinu, i won't forget , if i face same problem next time |
19:24.03 | justinu | lol |
19:24.11 | Dr-Linux | justinu: i think problems are good to learn |
19:24.13 | justinu | now, which one of you is going to help me get SER working with mysql? :P |
19:25.03 | justinu | maybe i should try openser |
19:25.08 | Dr-Linux | justinu: i can help you will apache, php, hosting, dns MySQL 4.1.xx etc but can't with asterisk :P |
19:25.14 | Dr-Linux | i'm newbie |
19:25.42 | Dr-Linux | justinu: one thing still remaining, thats just info nothing else |
19:26.51 | justinu | what? |
19:26.56 | justinu | go ahead |
19:27.00 | Dr-Linux | Thanks |
19:27.04 | justinu | i stayed home from work today to help you guys :P |
19:27.12 | pooh_ | rofl |
19:27.20 | axscode | beb |
19:27.29 | pooh_ | BOB |
19:27.32 | axscode | hehe.. I got a probs installing addons |
19:27.57 | axscode | app_saycountpl.c: In function ‘sayword_exec’: |
19:27.58 | axscode | app_saycountpl.c:101: warning: incompatible implicit declaration of built-in function ‘sscanf’ |
19:27.58 | axscode | make: *** [app_saycountpl.o] Error 1 |
19:28.02 | delmar | (yawn) morning everyone |
19:28.30 | delmar | poooo stinky :P |
19:28.41 | pooh_ | axscode: do you have your compile environment all together? |
19:29.04 | bweschke | pooh_ : no this is a bug. axscode: one moment, I'll give you the URL for the patch |
19:29.20 | axscode | o.. nice.. |
19:29.22 | bweschke | axscode: http://bugs.digium.com/view.php?id=5651 |
19:29.27 | axscode | thanks |
19:29.30 | Dr-Linux | justinu: my boss is in US, i'm in pakistan, he got 2 Cisco 79xx _new_ i wanna configure them |
19:29.33 | Dr-Linux | how can i do that |
19:29.42 | justinu | um, they need the SIP software |
19:29.50 | Dr-Linux | even i never seen these fone in real |
19:29.55 | delmar | hey so im trying ti impliment the last variation found here http://www.voip-info.org/wiki-Asterisk+tips+911 to handle emergency dialing.... but it doesnt work... it just loops out... i think it thinks the Zap is busy when its not... anyway... what i have is the same as the 3rd variation found here http://www.voip-info.org/wiki-Asterisk+tips+911 can anyone see anything wrong with that? |
19:30.08 | justinu | Dr-Linux: do you know if they have the SIP software loaded on them? |
19:30.10 | pooh_ | bweschke: good catch |
19:30.15 | Dr-Linux | yeah, justinu, but whats appropriate way to do that remotely? |
19:30.21 | TheCops | justinu, the VoiP gateway with FXO to SIP is very great, but very expensive vs the FXO to PRI interface and TDM... |
19:30.38 | Dr-Linux | justinu: they are just new, i don't thing the software is there |
19:30.38 | justinu | TheCops: really? hmmm how much more? |
19:30.43 | pooh_ | delmar: /me is not living in 911 area |
19:30.45 | *** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com) |
19:30.46 | TheCops | duh |
19:30.48 | TheCops | just closed the document |
19:30.49 | TheCops | lol |
19:30.50 | axscode | bweschke: thank you.. how to apply this patch? what's the command? |
19:30.53 | Dr-Linux | http://cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip1 |
19:30.53 | justinu | Dr-Linux: well, your boss needs a TFTP server. |
19:30.54 | TheCops | sorry |
19:30.57 | Dr-Linux | have a look please |
19:30.58 | heroine | °O° <-- mickey |
19:31.00 | delmar | pooh_, that is irrelevant |
19:31.07 | pooh_ | axscode: take a look at the patch, it is 1 adjustment only |
19:31.10 | delmar | pooh_, the actual number to dial doesn't matter |
19:31.20 | delmar | u can change it to 111 or whatever |
19:31.22 | pooh_ | delmar: pastebin the trouble |
19:31.25 | delmar | where are u pooh_ ? |
19:31.29 | pooh_ | delmar: NL |
19:31.31 | axscode | ok have to find the file |
19:31.52 | justinu | Dr-Linux: well, first of all, you need to have the cisco SIP software image, then you load it onto a TFTP server, then you tell the phone to upgrade |
19:32.10 | justinu | Dr-Linux: there are instructions on cisco's site, but you need the image, which costs money. |
19:32.22 | axscode | can't find it.. the <stdio.h> is added? |
19:32.49 | pooh_ | axscode: yes |
19:33.05 | axscode | still same |
19:33.31 | pooh_ | axscode: output pls? |
19:33.41 | axscode | nstall: cannot stat `app_saycountpl.so': No such file or directory |
19:33.42 | axscode | install: cannot stat `cdr_addon_mysql.so': No such file or directory |
19:33.42 | axscode | install: cannot stat `app_addon_sql_mysql.so': No such file or directory |
19:33.45 | axscode | bad.. |
19:33.59 | justinu | bring me some modules, stat! |
19:34.02 | pooh_ | it can't find the files? |
19:34.09 | axscode | guess.. |
19:34.11 | axscode | so |
19:34.14 | delmar | OK so the example I am using and the output are at http://pastebin.ca/28025 |
19:34.43 | Dr-Linux | justinu: i'm already using many Cisco products, like PIX, Swithces and Routers etc, i never bought any image with money, i have logins, but i'm not sure about SIP images .. so tell me what should i tell to me boss, bcoz he said configure it |
19:35.15 | axscode | pooh the file. with .o exist.. |
19:35.29 | axscode | <PROTECTED> |
19:35.30 | pooh_ | delmar: it fails at line 9 already |
19:35.34 | Dr-Linux | shit |
19:35.48 | Dr-Linux | how i can left the conference room, i'm using x-lite? |
19:36.15 | pooh_ | delmar: dunno how your trunk is set up |
19:36.33 | pooh_ | delmar: ah, zap/17 |
19:36.33 | delmar | pooh_, so u reckon it's trunk setup? |
19:36.37 | pooh_ | yup |
19:36.42 | axscode | pooh: http://pastebin.ca/28028 |
19:36.47 | pooh_ | delmar: chan is NOT available |
19:37.07 | delmar | pooh_ thats the example... from the web page.. the emergency trunk = Zap/g2 |
19:37.49 | pooh_ | axscode: you need mysql-devel |
19:37.56 | axscode | hmm ok |
19:38.13 | delmar | pooh_ should be Zap/g1 perhaps? |
19:38.15 | pooh_ | delmar: you defined a specific channel zap/17 so you have 17 channels at least ? |
19:38.16 | Assid | damn |
19:38.21 | Assid | i cant find anything regarding this echo |
19:38.25 | pooh_ | delmar: try zap/gx |
19:38.30 | hardwire | wow |
19:38.32 | Assid | anyone know what can be done regarding echoes? |
19:38.32 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
19:38.36 | hardwire | chanisavail kills my asterisk server |
19:38.48 | pooh_ | Assid: cover your ears ;-) |
19:38.50 | axscode | ./mysql Ver 14.12 Distrib 5.0.15, for pc-linux-gnu (i686) using readline 5.0 |
19:38.55 | Assid | haha |
19:38.57 | axscode | is there a version for mysql-devel ? |
19:39.01 | Assid | seriously tho |
19:39.04 | Assid | what can we do abt it |
19:39.22 | pooh_ | axscode: depends on your distro, it is a distro thing not * |
19:39.37 | pooh_ | Assid: too little info atm from you |
19:39.49 | delmar | pooh_ just noticed most other stuff is Zap/g1 so ill change it to Zap/g1 |
19:40.14 | pooh_ | delmar: depends on your zap cards and configs, but give it a try |
19:40.26 | Assid | got sipura 841 ip phones.. it has a horrible echo issue which is pretty constant |
19:40.45 | delmar | pooh_, yeah that was the bugger |
19:40.49 | pooh_ | Assid: turned on echo cancellation on the phone itself ? |
19:40.56 | pooh_ | delmar: good |
19:40.57 | Assid | dont see it |
19:41.07 | pooh_ | Assid: webinterface to the 841 |
19:41.11 | *** part/#asterisk oej (n=Olle@apollo.webway.se) |
19:41.14 | Assid | yeah.. im searching there |
19:41.18 | synthetiq | is it possible to execute shell commands in the dialplan? |
19:41.24 | pooh_ | Assid: advanced |
19:41.35 | pooh_ | synthetiq: yes, use the system command |
19:42.27 | delmar | Ok so another problem, I have a TDM400 with 1FXO for PSTN... and for some reason in the last few days.. possibly since I updated to HEAD, I get noticable echo in an incoming call, usually much more at the start of the call but lessening. I used to get this with an X100 card but much worse and gave up for the TDM400 and was happy with it until just recently...what's up with this anyone know? |
19:43.10 | pooh_ | delmar: check your zaptel.conf and check settings, set traing to on, echocanellation=yes |
19:43.49 | delmar | pooh_, when i had the X100, i basically had a crash course in everything to do with echo settings.... all that stuff is setup. here ill tell ya the settings... |
19:44.10 | pooh_ | delmar: pastebin |
19:44.18 | delmar | pooh_, oh and those settings are in zapata.conf not zaptel |
19:44.21 | axscode | if I did modprobe zaptel and modprobe ztdummy earlier.. do I have to do it in my next reboot? |
19:44.40 | pooh_ | axscode: yes |
19:44.50 | pooh_ | axscode: unless you make it automagically |
19:45.03 | axscode | ok.. ill add it somewhere.. |
19:45.04 | pooh_ | axscode: play with rc.d or rc.local |
19:45.20 | axscode | I don't know if SuSE has a rc.local |
19:45.31 | pooh_ | axscode: or even with asterisk start/stop script |
19:45.31 | axscode | though I can add it already to my init.d |
19:45.51 | axscode | yupz I made one from skeleton |
19:46.13 | Assid | pooh_: silence supression takes care of that? |
19:46.36 | pooh_ | Assid: please explain |
19:46.44 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
19:46.53 | generalhan | whats up everyone ? |
19:46.53 | Assid | would silence supression take care of echo cancellation? |
19:47.01 | pooh_ | Assid: nope |
19:47.08 | pooh_ | Assid: let me check |
19:47.11 | justinu | assid: who hears echo? |
19:47.15 | delmar | pooh_ http://pastebin.ca/28030 |
19:47.17 | Assid | the other end |
19:47.19 | generalhan | whats the command to reboot asterisk ? |
19:47.27 | enemy | when trying to call a group Dial(ZAP/g3) , then I get app_dial.c:805 dial_exec: Unable to create channel of type 'ZAP' ..... could anyone point me in the right direction? |
19:47.28 | generalhan | i need to shut it all down and bring it back up |
19:47.36 | delmar | pooh_, zapata is an edited example and i leave all the stuff in there to help me know whats what :P |
19:48.32 | justinu | assid: try turning down the volume on the spa841 |
19:48.58 | bweschke | enemy: what zap channels have you defined in group3? |
19:49.39 | delmar | pooh_, so as u can see echocancel=yes, echocancelwhenbridged=yes, wchotraining=800 to give it more echocan time at the start of the call, rxgain/txgain have been tweaked according to the local Telco recommendations until their service no longer said things like.. increase dB by X.X etc... |
19:50.14 | pooh_ | delmar: what the heck are you doing with all those [context] entries? |
19:50.20 | delmar | pooh_, the anoying thing is.. ok we have a service here that allows us to send a Fax.. and it will be sent back to us with dB level adjustment readings.... so the testing is done while you send a Fax. |
19:50.59 | delmar | pooh_, there is only 2... |
19:51.06 | Assid | - decreases it right ? |
19:51.06 | *** join/#asterisk gman55s1f (n=joe@128.100.33.138) |
19:51.43 | delmar | pooh_, dring1 makes sure that is recognized at inbound-FXO, dring2 is obviously fax, and the default for the channel... |
19:51.44 | gman55s1f | question: I've upgraded to the latest amportal, which seperates users and devices.. not all voicemail extensions say user is "on the phone" instead of "unavailable".. any ideas? |
19:51.57 | delmar | pooh_, so 3 contextx... 1 default, and 1 for the 2 distinctive rings |
19:51.58 | asteriskmonkey | anyone know the cause of echo when there is 2 iaxys talking ot each other? |
19:52.00 | pooh_ | delmar: Let me take another loo |
19:52.10 | gman55s1f | correction, *now* all voicemail extensions say the above.. |
19:52.25 | Igbothom_III | I wondered who stole all of those toilets! |
19:52.28 | delmar | dring1context=inbound-FXO |
19:52.28 | delmar | dring2context=fax |
19:52.28 | delmar | context=inbound-FXO |
19:52.38 | delmar | pooh_ those are the only contexts available |
19:52.55 | *** part/#asterisk santiago (n=santiago@208.195.215.124) |
19:54.07 | delmar | pooh_ anyway as I was saying, we can send a fax for testing of the call dB levels, and a fax comes back telling us what to adjust... I have the correct levels set and the Telco no longer says they need adjusting, yet I cannot receive a fax, it gets killed pretty quick, which would suggest the incoming fax call may be experiencing issues... perhaps the same echo issues I get when I receive an audio call? |
19:54.19 | pooh_ | delmar: pls clear out zapata.conf with *only* relevant lines |
19:54.37 | delmar | pooh_ lol ok :P |
19:55.18 | *** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
19:57.15 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
19:59.57 | delmar | pooh_, doh. |
19:59.59 | justinu | lol |
20:00.03 | delmar | pooh_, almost done tidying this up |
20:00.12 | pooh_ | delmar: about time!!!!! ;-) |
20:00.33 | bweschke | asteriskmonkey: what does that environment look like? iaxy <-------> * <----------> iaxy ? |
20:00.55 | pooh_ | bweschke: nice? |
20:01.37 | delmar | pooh_ http://pastebin.ca/28033 |
20:01.42 | asteriskmonkey | bwecheke: looks like this iaxy ---> internet ---> asterisk <--- internet ----- iaxy |
20:01.50 | bweschke | k |
20:01.52 | asteriskmonkey | so yes pretty much |
20:01.52 | axscode | install: cannot stat `app_addon_sql_mysql.so': No such file or directory |
20:01.52 | axscode | install: cannot stat `res_config_mysql.so': No such file or directory |
20:01.52 | axscode | make: *** [install] Error 1 |
20:01.54 | pooh_ | lloks=looks |
20:02.08 | bweschke | what are the iaxy's hooked up to? just regular analog phones? |
20:02.13 | asteriskmonkey | bewschke: yes |
20:02.33 | delmar | pooh_ i dont think rxwink=300 and switchtype= even need to be in there .. for an analog PSTN FXO |
20:02.40 | asteriskmonkey | the connections have really low latencys and no jitter either so its hard to figure out |
20:02.49 | delmar | pooh_ but they are there... |
20:03.04 | bweschke | i don't think it's latency or jitter related at all... |
20:03.14 | pooh_ | delmar: what is the *exact* problem again ? |
20:03.16 | axscode | make: shared: Command not found <-- what error is this??? |
20:03.19 | asteriskmonkey | nor do i as its a mint network |
20:03.29 | asteriskmonkey | bewschke: whats the next thing to look at |
20:03.39 | bweschke | any chance it's acoustic? - eg. the speaker voume is finding it's way back around to the mic? |
20:03.43 | pooh_ | axscode: the programm has not been found |
20:03.48 | asteriskmonkey | bweschke: no |
20:03.53 | axscode | sahred? what program is shared |
20:03.57 | delmar | pooh_, I can hear echo of myself, more noticable on an incoming call.. outgoing seems fine. |
20:04.02 | asteriskmonkey | its been used on many different handsets and even digium got an echo |
20:04.13 | delmar | pooh_, and i think this is also why incoming fax's are failing to receive but seem to send ok |
20:04.28 | bweschke | u called digium support on it? |
20:04.32 | delmar | pooh_, if it was to do with rxgain or txgain... it would be the txgain that is the cause correct? |
20:04.40 | pooh_ | yup |
20:04.42 | *** join/#asterisk shido6 (i=shido6@d221-68-216.commercial.cgocable.net) |
20:04.52 | pooh_ | delmar: lower training value |
20:04.53 | asteriskmonkey | bweschke: they said they had no idea tried to blame it on network and sell me support |
20:04.55 | delmar | pooh_ trouble is, if i lower this, the exchange tells me that the dB this end is too low |
20:05.14 | pooh_ | uncomment echotraining |
20:05.34 | delmar | pooh_ and i thought this TDM400 thing had a built in DSP echo can, i shouldnt even have this problem |
20:05.34 | pooh_ | delmar: tell != result |
20:05.43 | pooh_ | delmar: practice == reult |
20:05.43 | delmar | pooh_ it is... echotraining=800 |
20:05.57 | bweschke | delmar: no, tdm400's have no built in hardware echo can |
20:05.57 | mog_work | dsp is done in software |
20:06.00 | pooh_ | delmar make it 400 and uncomment the echotraining, try again |
20:06.01 | delmar | pooh_ it is... echotraining=yes is the same as echotraining=400 |
20:06.09 | shido6 | txgain and rxgain incrememnts of 2 will help but if ur faxing , good luck :) |
20:06.27 | mog_work | no rx and tx gain |
20:06.28 | delmar | bweschke, so the echocan on a TFM400 with 1FXO is the same as an X100? |
20:06.32 | mog_work | use the kb1 or mg2 echocan |
20:06.36 | delmar | bweschke, whats the difference then? |
20:06.42 | mog_work | no card is differnet |
20:06.45 | asteriskmonkey | bweshcke: no idea? what to do next? |
20:06.47 | bweschke | delmar: no. the card is not the same |
20:06.53 | mog_work | fxo module is better than the x100p |
20:06.53 | bweschke | asteriskmoney: one sec. |
20:07.03 | delmar | pooh_, that will lower the echotraining time.... should make it worse if anything. |
20:07.05 | bweschke | mog_work: any thoughts on the iaxy? |
20:07.16 | pooh_ | delmar: only the *new* digium cards have this 'feature' |
20:07.26 | delmar | bweschke, obviously .. they are physically different.... |
20:07.54 | shido6 | rx and tx deal with the db |
20:08.10 | shido6 | gain |
20:08.14 | delmar | bweschke, other than that.. what is the difference? I mean.. i HAD an X100 and it sucked.. i made no software changes.. ripped it out.. stuck a TDM400 in there.. loaded wctdm instead of wcfxo, and it worked.. mint |
20:08.33 | bweschke | delmar: it's a different chip completely |
20:08.44 | gman55s1f | all my voicemail extensions now say user "is on the phone" instead of "is unavailable" .. any ideas? |
20:08.47 | delmar | bweschke, right, so there are different features.... |
20:09.07 | delmar | gman55s1f, change the recording? |
20:09.22 | blitzrage | someone was complaining about echo on an iaxy to iaxy connection... |
20:09.33 | bweschke | blitz: ya, asteriskmonkey |
20:10.08 | delmar | pooh_ ok so change to echotraining=400 anything else? |
20:10.27 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) |
20:10.33 | pooh_ | delmar: yeah, save and RESTART |
20:10.46 | shido6 | Voicemail(u${EXTEN}) and b${EXTEN} are different :) u is unavailable and b is busy / on the phone |
20:10.52 | blitzrage | bweschke: him too? Someone was saying he was having problems on the on-asterisk mailing list |
20:11.08 | blitzrage | unless its the same person :) |
20:11.09 | delmar | pooh_ what do u mean only the *new* cards have a built in DSP etc? I was told this one would have... and this was one of the benifits of the TDM400 etc? |
20:11.35 | pooh_ | delmar: check www.digium.com and look at the specs |
20:11.38 | mog_work | the 2400p has echo can module you can get |
20:11.40 | delmar | pooh_ it certainly seemed pretty damn fine to start with... but something has changed... i suspect in software.... |
20:11.47 | bweschke | delmar: tdm2400 has an optional echo can, the 400's do not |
20:12.06 | shido6 | http://pastebin.ca/28035 |
20:12.22 | bweschke | delmar: does the echo stop if you roll back? |
20:12.25 | *** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net) |
20:12.34 | delmar | bweschke, what does that say for their products then... that they need to make an additional echocan hardware option. |
20:12.43 | ronaldl79 | Hello |
20:13.00 | pooh_ | ronaldl79: everybody say HI to ronaldl79 ;-) |
20:13.08 | delmar | bweschke, in future I will copy my original CVS out of the way before upgrading so I can easily.. roll back :P |
20:13.09 | bweschke | delmar: what does it say for their products? I think it says that they're making products to accomodate for difficult echo situations where the software dsp doesn't suffice. |
20:13.28 | ronaldl79 | Regarding number porting: Are your clients able to maintain their phone listing in the telco's White pages? I've been trying for two weeks to get an answer about this from Qwest, to no avail. And they still don't know as of today. |
20:13.32 | delmar | bweschke, its not like its a huge corporate business here... just my little home-buz box :P so no big deal |
20:13.34 | mog_work | but very rarely does the software dsp not suffice |
20:13.35 | pooh_ | delmar: do not forget your handset, echo is many things related |
20:13.44 | mog_work | the software echo can rocks delmar |
20:13.47 | delmar | bweschke, yeah i agree I guess. |
20:14.17 | delmar | pooh_, SIP to SIP calls and so on have no echo at all. |
20:14.28 | bweschke | echo can come from a number of places and be caused by a number of different things.. if you've just got one or two fxo lines, the software echo can and the advances it's made are a cool deal |
20:14.31 | mog_work | unless the phone sucks |
20:14.54 | delmar | pooh_, and outbound SIP to PSTN is fine. i mean.. if u try hard enuf to echo and scream down the phone u can hear something but.. yeah.. outgoing calls are great. |
20:15.04 | mog_work | anywhere there is analog echo can happen |
20:15.07 | bweschke | but when you start look at quantities of lines (eg. DS1's / E1's / 24 port FXO/FXS cards) you're starting to ask the processor to work hardware on the CPU to accomodate echo canceling for all those ports |
20:15.14 | bweschke | that's why you get hardware echo can options there |
20:15.27 | delmar | pooh_, ok ill test this a sec. brb |
20:15.45 | blitzrage | ronaldl79: ours will be able to via Verizon within a couple of weeks. |
20:15.51 | axscode | anyone please login to my asterisk |
20:16.05 | axscode | sip:10001:10001@203.213.217.122 |
20:16.07 | asteriskmonkey | bwesche: no idea yet? |
20:16.20 | ManxPower | Hmm... for $560/month I can become a mini-ISP and mini-CLEC. |
20:16.22 | axscode | sip:10002:10002@203.213.217.122 |
20:16.30 | ronaldl79 | blitzrage: Is the VoIP provider you're converting a client to handling the white page listing, or Verizon is simply maintaining it because the number hasn't changed? |
20:16.34 | bweschke | asteriskmonkey: I really don't know.. I'm sorry. the echo can functionality on the iaxy is pretty closed.. there's not much you can tune once that gets to * as an IAX stream. |
20:16.39 | axscode | sip:10003:10003@203.213.217.122 |
20:16.52 | axscode | anyone please... I just want to know |
20:17.11 | bweschke | what would be interesting to see asteriskmonkey would be to take iax soft phones at those same locations where the iaxy's are at and see if the the echo is still there |
20:18.01 | asteriskmonkey | could you reccommend a good one to try? |
20:18.39 | delmar | pooh_ ok outbound calls, there is slight echo at the start of the call but is quickly dealt with within about 5seconds or so.....and..... |
20:19.01 | delmar | pooh_, ok i was messing with my dialplans alot yesterday and didnt finish so incoming is br0ked.. let me fix it. lol |
20:19.15 | asteriskmonkey | beeschke: whats a good soft iax i could use? ie site link or name :) |
20:19.28 | blitzrage | ronaldl79: we're creating a portal where the manager of the DID (client) can update their E911 and 411 listing -- then we parse it, update our gateways, and submit the updates to Verizon -- and they do whatever they do with that |
20:19.43 | bweschke | asteriskmonkey: one moment. trying to find a URL for u |
20:19.54 | asteriskmonkey | thanks |
20:19.55 | *** join/#asterisk logicalonline (n=Ken@atlantis.clearshout.com) |
20:20.00 | sjobeck99 | hey blitzrage |
20:20.08 | zoa | hey blitzrage |
20:20.14 | blitzrage | sjobeck99: I don't know who that is |
20:20.19 | blitzrage | :D |
20:20.24 | blitzrage | sjobeck99 / zoa: zup guys! |
20:20.26 | sjobeck99 | jason@sjobeck |
20:20.28 | zoa | blitzrage: what provider do you work for ? |
20:20.33 | blitzrage | sjobeck99: I know who you are :) |
20:20.47 | zoa | ah the ones from astricon ? |
20:20.47 | bweschke | asteriskmoney: linux or windoze? |
20:20.50 | zoa | true |
20:20.52 | zoa | you told me that |
20:20.53 | zoa | lol |
20:20.54 | zoa | :) |
20:20.56 | asteriskmonkey | windozr atm |
20:21.01 | blitzrage | zoa: yep, those guys |
20:21.26 | *** part/#asterisk logicalonline (n=Ken@atlantis.clearshout.com) |
20:21.30 | bweschke | asteriskmonkey: this this one http://asteriskguru.com/tools/idefisk_beta.php |
20:21.50 | zoa | bweschke: that one also exists for linux |
20:21.56 | zoa | but its not online yet |
20:22.08 | bweschke | zoa: good 2 know |
20:22.26 | asteriskmonkey | bweschke: is there a good windows sip client also that you would recommend? |
20:22.38 | bweschke | X-Lite? |
20:22.40 | zoa | xlite is probably the best one now |
20:22.40 | bweschke | SJPhone? |
20:22.51 | denon | anyone know if Nathan Pralle is ever on irc? |
20:23.26 | blitzrage | X-Lite! |
20:23.27 | ManxPower | All softphones suck |
20:23.32 | blitzrage | ManxPower: true :) |
20:23.36 | justinu | softphones do suck |
20:23.38 | ManxPower | X-Lite seems to suck less, according to some people. |
20:23.41 | demetrio | is it possible to retrieve from within asterisk the value of a SIP header? |
20:23.51 | justinu | probably because of the audio latency in windows |
20:24.02 | bweschke | asteriskmonkey: try that. if your echo still exists from the softphones through the same connections and * server, then we've got to look elsewhere than the iaxy's |
20:24.07 | ManxPower | Well, I finally called some CLECs for quotes on a location 11 miles from the nearest CO. |
20:24.16 | Aughey | my vote is SJPhone |
20:24.54 | zoa | sjphone gives too much shit to install |
20:24.58 | zoa | demetrio: it is yes |
20:24.59 | justinu | T1 quotes? |
20:25.02 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
20:25.15 | ManxPower | justinu, Yup. |
20:25.20 | justinu | and? |
20:25.29 | demetrio | zoa: how is it done? any link? |
20:25.32 | ManxPower | justinu, seems to be around $500/month or so. |
20:25.38 | bweschke | zoa: my version of x-lite didn't have the transfer button working. dunno if that's because it's an eyebeam/x-pro thing only... that's why I use SJPhone too |
20:25.42 | justinu | that's actually not as bad as I thought |
20:25.49 | ManxPower | Which is more than I really want to pay. I'll find ways to reduce the cost. |
20:26.12 | ManxPower | justinu, $546/month for 4 voice lines and 512K interent. |
20:26.16 | zoa | thats only working for the pro version |
20:26.17 | bweschke | ManxPower: ya, that's actually pretty competitive. that's PRI including the monthly loop, no? |
20:26.17 | ManxPower | Of course, I don't need 6 voice lines. |
20:26.28 | justinu | manx: ack |
20:26.30 | zoa | competitive ? |
20:26.39 | ManxPower | About $400/month for just 512K internet OR 384K NNI FrameRelay |
20:26.44 | zoa | i pay 250$/month for 30 lines |
20:26.47 | ManxPower | bweschke, Yeah. |
20:26.47 | zoa | although i dont get internet |
20:26.50 | justinu | how much of that is loop cost? |
20:26.59 | ManxPower | It's still expensive for just me working out of my home. |
20:27.05 | zoa | hmm yeah it sounds good anyway |
20:27.25 | ManxPower | They don't break out loop cost |
20:27.44 | bweschke | manxpower: is there not the normal compliment of broadband options (cable, dsl) where you're at? |
20:27.51 | ManxPower | bweschke, no. |
20:28.03 | bweschke | manxpower: that stinks. |
20:28.07 | asteriskmonkey | bweschke: cant get any audio out of this ide program |
20:28.08 | ManxPower | 11 miles from CO means no DSL, and there is no local cable service. This location is RURAL. |
20:28.34 | ManxPower | The nearst town has a population of 6,000 people and is 11 miles away. |
20:28.44 | justinu | what do they want to have 1.5meg internet? |
20:28.48 | bweschke | maxpower: yep - i hear ya... and honestly, 11 miles from the CO for only $500/mo on the DS1, you're getting a REALLY good price. that's not a short loop the clec has to pay for access |
20:28.52 | ManxPower | But that's part of the reason I like the location |
20:28.57 | shmaltz | ManxPower, you shlould look into 512k satellite |
20:28.58 | ManxPower | justinu, "they"? No, me. |
20:29.13 | ManxPower | shmaltz, SSH via satellite is banned under the Geneva Convention. |
20:29.15 | justinu | no, what's the lec charging for a full DS1 IP link |
20:29.18 | ManxPower | (or it should be) |
20:29.24 | bweschke | asteriskmonkey: which program? idefisk? |
20:29.32 | shmaltz | lol |
20:29.33 | asteriskmonkey | never mind : |
20:29.34 | ManxPower | justinu, I don't know, I'm not getting quotes for a full T |
20:29.38 | asteriskmonkey | just takes a second to wake up |
20:29.57 | shmaltz | ManxPower, I heard from others that it is not so bad anymore |
20:30.19 | ManxPower | My other option is, of course, sat for web and dialup with tuned MTU/MRU/MSS for SSH |
20:30.32 | zoa | asteriskmonkey: if you have a problem with it, find me and we will help you or fix it for you if needed |
20:30.39 | zoa | but it should be stable |
20:30.50 | ManxPower | There is someone near the location with DirectWay internet via Sat, so I'll see how SSH runs over that. |
20:31.06 | justinu | latency is gonna be a bit annoying |
20:31.42 | ManxPower | ohad, the 512K internet/ 6 voice includes PRI and 600 voice LD mins, and 5 cents/min after that. |
20:31.50 | *** join/#asterisk _Sam-- (n=sam@phone2.kneedraggers.com) |
20:32.02 | bweschke | i've gotta imagine you're going to be hating life doing ssh over a sat link |
20:32.15 | ManxPower | bweschke, I don't consider it a valid option. |
20:32.31 | _Sam-- | hi i dont mean to sound like an ass (it comes naturally) but doesnt anyone actually pay royalties for their music on hold? |
20:32.52 | _Sam-- | im just wondering what the costs to ASCAP, CSAC and BMI would be to legally play MOH |
20:32.54 | ManxPower | _Sam--, Yes. |
20:32.57 | bweschke | _Sam: the moh provided with * is provided free so no commercial royalties are required |
20:33.13 | ManxPower | There is royalty free music too. You just pay a one time fee. |
20:33.17 | _Sam-- | there is music that comes with it? i must have overwrote it |
20:33.18 | bweschke | _Sam: but if you introduced your own, sure royalties would apply |
20:33.29 | _Sam-- | im just wondering how much it would cost to paly good music, legally |
20:33.48 | bweschke | _Sam: yessir. from Freeplay Music |
20:34.08 | bweschke | _Sam: http://www.freeplaymusic.com/ |
20:34.14 | _Sam-- | thank you, there now |
20:34.22 | ManxPower | I'm still waiting on quotes from a few other CLECs |
20:34.39 | _Sam-- | ssh over sat? with what, 500ms latency? |
20:34.47 | SwK[Work] | anyone running the AudioCodes MP10X's with the latest firmwares? |
20:35.11 | sjobeck99 | any one know native MOH ? |
20:35.41 | *** join/#asterisk jazor (n=jazor@u25-8.dsl.vianetworks.de) |
20:35.53 | _Sam-- | what do you mean native MOH? |
20:36.32 | ManxPower | _Sam--, It's a new 1.2 feature |
20:37.00 | sjobeck99 | i think it means that * doesnt use mpg123 or other players to play files, it plays the files fright from the drive itself, using whatever codecs the OS has for those files (ie: mp3, ogg, etc). |
20:37.02 | enemy | I have agents in alot of queues, how can I build a prioritized queue for each agent so that direct calls get prioritized infront of all the other queues. |
20:37.13 | delmar | pooh_ u still here? |
20:37.19 | sjobeck99 | it's option in musiconhold.conf |
20:37.24 | enemy | direct calls (go into an agents personal queue) |
20:37.25 | _Sam-- | enemy: set it when they press the key for the queue in your IVR |
20:37.47 | _Sam-- | exten => 1,1,SetVar(QUEUE_PRIO=10) |
20:37.47 | _Sam-- | exten => 1,2,Queue(salesq) |
20:38.10 | delmar | ok outbound calls, there is slight echo at the start of the call but is quickly dealt with within about 5seconds or so...incoming calls are somewhat worse.. louder echo, goes eventually but takes a while. start of call is quite nasty. |
20:38.33 | ManxPower | delmar, Welcome to the world of VoIP. |
20:38.36 | enemy | thanks |
20:38.48 | delmar | ManxPower, yeah but .. with a TDM400? |
20:38.55 | _Sam-- | higher priority number = call gets through faster |
20:39.12 | _Sam-- | in theory anyway, i have my queues setup with priorities |
20:40.12 | delmar | What echocan algs should I be using with the TDM400? ie in zconfig.h etc?? |
20:40.26 | ManxPower | *grumble* There are still no hotel rooms north of New Orleans, and no rooms in New Orleans for under $200/night |
20:40.38 | bweschke | delmar: yessir. the software echo can is doing it's job because you're telling us the echo is eventually dealt with... but that's just hiding what's really causing the echo... have you messed with txgain and rxgain yet on these lines? |
20:40.51 | delmar | ManxPower, nasty. |
20:41.07 | delmar | ManxPower, that because of of storms/flooding or ? |
20:41.09 | ManxPower | looks like my next trip to the area will mean staying with my boss again. |
20:41.29 | ManxPower | delmar, It's because of 1) cleanup workers and 2) so many people without homes anymore. |
20:41.59 | delmar | bweschke, cheers for reminding me. im going to set them to 0.0 on both which is where they used to be |
20:42.05 | delmar | bweschke, then start again from there |
20:42.17 | bweschke | oh mog_work!! what's the echo can most preferred by "those who know echo cans" these days?? :) |
20:42.35 | mog_work | mg1 and kb1 |
20:42.43 | mog_work | kb1 for 90% of cases |
20:42.52 | mog_work | mg1 for the 10% kb1 doesnt fix |
20:43.10 | delmar | ManxPower, yep. I can imagine it's pretty rough over there right now. |
20:43.11 | mog_work | mg1 is a little better i think |
20:43.17 | mog_work | but it can cause issues on some lines |
20:43.21 | mog_work | so i reccomend kb1 first |
20:43.37 | bweschke | delmar: try those for zconfig.h - u gotta be sure you have the latest CVS-HEAD |
20:44.27 | ManxPower | delmar, north of the lake it's fine. |
20:44.30 | mog_work | kb1 is the default as of recently |
20:44.47 | *** join/#asterisk kb1_kanobe (n=jsmith@h24-207-96-50.cst.dccnet.com) |
20:45.14 | zoa | hehe |
20:45.18 | zoa | <mog_work> kb1 is the default as of recently |
20:45.19 | zoa | * kb1_kanobe has joined #asterisk |
20:45.22 | kb1_kanobe | d'gay all. |
20:45.35 | kb1_kanobe | g'day, even...! |
20:46.54 | mog_work | who is kb1_kanobe that rocks |
20:47.04 | kb1_kanobe | quick question: I have a problem with a harrasing caller on a PRI circuit. They're blocking their CallerID (ie. it's coming up as null in CDR). I would like to capture the originating number from the ISDN control messages without running w/pri debugging on all the time... any suggestions? |
20:47.26 | asteriskmonkey | there is a function in asterisk |
20:47.28 | zoa | i dont think you can |
20:47.34 | delmar | bweschke mog_work cheers, ok tx_gainand rx_gain set to 0.0 and its still there just not as loud, but you can tell it's happening.. and with rxgain/txgain set to 0.0, the dB levels this end and equalization will be all out of whack and fax send/receive will most likely no longer work.. not that fax receive worked right anyway which is probably due to the echo on incoming calls effecting the incoming fax as well... |
20:47.35 | asteriskmonkey | that lets you ask callers for there id |
20:47.36 | zoa | is it passed as a debug message ? |
20:47.46 | delmar | bweschke, so ill try some echo-can's |
20:47.48 | zoa | i mean it shows the callerid in the debug messages ? |
20:47.55 | kb1_kanobe | asteriskmonkey: not an option - this is an 11 channel pri and very busy at that. |
20:48.03 | asteriskmonkey | just run an rule on incomming, if its null have them enter it or just send it to hangup |
20:48.26 | asteriskmonkey | ah ok.. |
20:48.29 | delmar | mog_work> kb1 for 90% of cases, this seems to be the default #define already... |
20:48.30 | kb1_kanobe | zoa: I presume it is being passed in as part of the IE during call setup, however I don't actually know.. let me check. |
20:49.04 | zoa | it should not be passed unless you are 911 or police or so |
20:49.11 | *** join/#asterisk freespace-in (n=special@ppp-70-227-27-183.dsl.ipltin.ameritech.net) |
20:49.22 | zoa | unless you are a carrier |
20:49.40 | kb1_kanobe | zoa: end user, however we are gov't. |
20:49.57 | delmar | ok so since echocan KB1 is already in use and not working... MG2 perhap? |
20:49.58 | zoa | i dont think it should be there |
20:50.08 | bweschke | delmar: try mg1 |
20:50.10 | tzanger | MG2 has made quite a idfference for us |
20:50.19 | bweschke | or MG2? |
20:50.23 | tzanger | there is no MG1 |
20:50.28 | delmar | tzanger, explain.. what hardware u using? |
20:50.31 | mog_work | i mean mg2 |
20:50.32 | mog_work | im sorry |
20:50.44 | tzanger | TE405 |
20:50.49 | mog_work | never trust me for exact facts |
20:50.59 | mog_work | err specifics |
20:51.10 | mog_work | im generally right, just cant be trusted on names |
20:51.17 | bweschke | delmar: MG2, KB1, etc - these are all software algorithms that are slightly different approaches taken to cancel out your echo |
20:51.23 | delmar | tzanger, sorry.. not familiar with the model numbers.. what is it.. PSTN, Bri/Pri? |
20:51.38 | delmar | bweschke, yeah i figured that |
20:51.39 | tzanger | yup |
20:51.52 | delmar | tzanger, which lol? |
20:51.53 | *** join/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz) |
20:52.06 | tzanger | ? |
20:52.06 | delmar | bweschke, i still dont see why the echocan isn't done on the board. |
20:52.14 | patpatnz | hi guys |
20:52.16 | tzanger | TE405 = quadspan T1/E1 |
20:52.19 | tzanger | using it on PRI |
20:52.27 | delmar | tzanger, what is the card... pstn/fxo ? or isdn Bir/Pri? |
20:52.30 | patpatnz | does anyone know about ooh323c and dtmf tones? |
20:52.31 | bweschke | delmar: because it's considerably more expensive, and for most instances, not necessary in small numbers of ports |
20:52.31 | delmar | right |
20:52.45 | delmar | tzanger, i cant imagine u would have echo anyway with a good channelbank |
20:52.49 | kb1_kanobe | delmar: the earlier cards are simply protocol adapters/physical interfaces. The later cards include a 3rd party echo cancellation ASIC. |
20:53.22 | delmar | kb1_kanobe, hrm. well thats contrary to what I was told before buying the TDM400. i was told they had hardware echocan. |
20:53.29 | tzanger | kb1_kanobe: well |
20:53.42 | tzanger | T100P/TE110/TE410/TE405 all have no sw echocan |
20:53.43 | bweschke | delmar: no. whoever told you that was mistaken |
20:53.51 | tzanger | only the TE406/TE411 have hardware echocan module |
20:53.51 | delmar | bweschke, indeed they were |
20:53.59 | tzanger | and the TDM2400 has a variant of the module |
20:54.48 | delmar | tzanger, yeah but a good channel bank has echocan built in the firmware. |
20:54.48 | tzanger | delmar: not that I'm aware of |
20:54.48 | tzanger | I don't know of a signle "channel bank" that has hardware echocan |
20:54.49 | tzanger | a lot of the good ones have adaptive hybrids to *minimize* reflected energy |
20:54.55 | delmar | tzanger, i have an adit600 here and I read that it has echocan and also impedance balancing stuff which would probably take care of most problems anyway. |
20:54.58 | tzanger | but I don't know of any that have echocan |
20:55.00 | bweschke | delmar: there's also a whole process to go through with fxotune to work on tuning to eliminate echo.. I'm not 100% familiar with it |
20:55.04 | tzanger | my adit600 most certainly does not have echo can |
20:55.24 | delmar | bweschke, i have read that fxotune is a waste of time and doesnt work right |
20:55.27 | bweschke | mog_work: has anyone put together a howto process for fxotune and the tdm400 cards? |
20:55.31 | tzanger | there's nothing in an adit600 that could do echocan |
20:55.40 | tzanger | except for, as I said, having a very well balanced hybrid |
20:55.43 | mog_work | its easy as 1 2 3 tune |
20:55.55 | kb1_kanobe | I beleive there was an ATM-based channel bank that included echo cancellation firmware because of the ATM transport delay issue. But it only works on the ATM bridged calls. |
20:56.01 | denon | I've actually seen a few channel banks that do echo cancellation, but its usually an optional module .. so not really part of the CB itself |
20:56.02 | mog_work | fxotune is generally not needed though |
20:56.05 | delmar | tzanger, well since i don't have a T1 card, and cant see myself getting one until they come down in price, I will never know :P |
20:56.15 | mog_work | as 90 % of world uses 600 ohms of impedence |
20:56.24 | tzanger | delmar: wellyes, that's an external echo can and they work with any T1/E1 |
20:56.33 | Beirdo | 90% of the world? ummm |
20:56.38 | Beirdo | I think you mean North America |
20:56.40 | bweschke | right.. I have a nack for finding that 10% |
20:56.42 | bweschke | it seems |
20:56.47 | mog_work | lol |
20:56.52 | mog_work | the world beirdo |
20:56.56 | *** join/#asterisk Toadyus (n=Im@S010600121746f9fe.mh.shawcable.net) |
20:56.57 | delmar | ok.. so.. .testing echo-can algs. |
20:56.58 | kb1_kanobe | I recommend mg2 and fiddling w/the echotraining parameter if you're having issues. If you _still_ have issues with all calls, fall back to kb1, though I don't think that will help. |
20:57.03 | mog_work | a lot of places use 600 ohms |
20:57.04 | denon | that 10% is .. what .. europe? :) |
20:57.16 | tzanger | echotraining is teh suck... MG2 is teh winz though |
20:57.22 | mog_work | not just europe just random places |
20:57.24 | *** part/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz) |
20:57.30 | mog_work | i mean we didnt have fxotune till this year |
20:57.30 | denon | nod |
20:57.41 | bweschke | there's a LEC down in Hilton Head Island, SC who uses 600 ohms, and then when you get lines in a hunt group from them, all of a sudden the lines are 900 ohms |
20:57.41 | delmar | does anyone in here actually know what the numbers in fxotune.conf really do ? |
20:57.43 | mog_work | and didnt have a huge demand for this ablity |
20:57.46 | mog_work | yes |
20:57.50 | kb1_kanobe | 'echotraining' is a hack. Plain and simple - a variable convergence adaptation rate needs to be implemented. |
20:57.50 | tzanger | kb1_kanobe: I wanted to ask you something about norstar but I've forgotten now |
20:57.51 | perd | tzanger. you dirty bastard. |
20:58.01 | mog_work | delmar it set s the impedance for the fxo moduls |
20:58.02 | delmar | i mean.. one time I tested it a few months back it was... 1=11,0,0,0,0,0,0,0,0 and the other day I tested it .. it was 1=5,0,0,0,0,0,0,0,0 |
20:58.04 | tzanger | perd: ? |
20:58.05 | mog_work | modules |
20:58.09 | mog_work | wow |
20:58.14 | perd | lwz hoe |
20:58.15 | mog_work | well it has gotten updated |
20:58.17 | perd | hah |
20:58.19 | mog_work | and is better now delmar |
20:58.30 | tzanger | perd: ;-) |
20:58.33 | mog_work | must people get all 0s or the first 1 is 1 but i cant remeber to be sure |
20:58.35 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
20:58.46 | bweschke | delmar: it's worth a shot.... improper impedance settings can certainly be a cause for bad echo |
20:58.51 | delmar | mog_work, perhaps it's the problem? it has been updated... and fxotune might be the cause of the echo? i might try setting back to 11,0,0 etc |
20:58.54 | perd | tzanger you familiar with asterisk? i have a question that i cant find an answer for anywhere |
20:59.07 | mog_work | you could |
20:59.14 | tzanger | perd: yeah you could say I am |
20:59.22 | delmar | ok. things for me to try.. .lots of things .. for the next few mins I guess. |
20:59.24 | delmar | brb |
20:59.25 | mog_work | are you using fxotune -s /etc/fxotune.conf after you load module? |
20:59.30 | delmar | yes |
20:59.37 | mog_work | good good |
20:59.44 | delmar | i even unload module and reload... |
20:59.47 | delmar | just be be sure |
20:59.48 | delmar | ok brb |
21:00.24 | *** join/#asterisk flynux (i=n85yb89@pingou.in) |
21:01.07 | perd | tzanger, ok.. i have a box set up for PSTN termination but i'm only allowed one connection out. i connect to this termination service with IAX2. i am using nagios to generate error messages that dial out and play a message. the problem is that if i have like 5 or 10 or 20 errors there will be 20 .call files in there and it will attempt to make 20 outbound calls at the same time |
21:01.45 | tzanger | perd: yeah it doesn't serialize that at all |
21:01.48 | perd | somehow i need to limit the number of outbound calls over that iax2 channel.. i tried putting some SetGroup stuff in my extensions.conf, then realized that the .call file doesnt use the extensions file to dial out |
21:02.00 | *** join/#asterisk hhoffman (n=hhoffman@tor/session/x-395008eb87af861e) |
21:02.04 | tzanger | better to use an agi that reads the outbound call list and tries to concatenate it into one call |
21:02.17 | tzanger | actually the callfile does indeed use extensions.conf to call out |
21:02.18 | hhoffman | <PROTECTED> |
21:02.22 | tzanger | it can't do it any other way |
21:02.29 | perd | hmmmi can use an agi to trigger outbound calls? |
21:02.36 | perd | i thought i could only use agi for inbound crap |
21:02.42 | tzanger | nope |
21:02.48 | perd | oh thats sexy |
21:02.54 | tzanger | chan_local and agi go together like sex and a condom |
21:03.01 | tzanger | well maybe a little better than that |
21:03.03 | perd | my penis gets hard just thinking about it |
21:03.06 | _Sam-- | perd you can just drop the call info in /var/spool/asterisk i think |
21:03.14 | _Sam-- | we do it here at work but i forget what we use |
21:03.25 | axscode | can someone gimme a exten sample for meetme? |
21:03.32 | perd | yeah i can generate outbound calls without a problem, its the limiting the amount of current outbound calls that i cant figure out sam |
21:03.49 | _Sam-- | cant you have your IAX provider limit your trunks? |
21:04.02 | perd | i think with agi i should be able to though, assuming tzanger is correct in saying i can generate calls from agi scripts without human interaction |
21:04.11 | tzanger | perd: yep you can |
21:04.16 | _Sam-- | he's pretty bright, id believe him |
21:04.17 | delmar | bweschke, ok first test was to re-confirm set at 5,0,0 etc.. in/out .. then set to 11,0,0 etc .. not really any difference between the two... none that I could detect. |
21:04.43 | perd | i'll just have nagios dump the warnings into a mysql database and have the agi script poll it until it sees a warning then generate a call |
21:04.46 | delmar | bweschke, so I will leave it at 5,0,0,0,0 etc which is what is currently detected, .. .now onto echocan algs. |
21:05.07 | perd | that way i can keep track of calls in the agi scripta nd be happy |
21:05.13 | perd | tzanger thank you! |
21:05.26 | tzanger | perd: np |
21:05.27 | kb1_kanobe | zoa: you're correct. It's coming back as 'Presentation: Presentation prohibited of network provided number (35) ', which I hadn't noticed before... I will have to prod the telco and see how they wish to proceed because it's not as if we can grab the line the offending call came in on and dial the 'this was an offensive call' vsc. |
21:05.52 | perd | know a good place for agi docs? |
21:05.53 | perd | ~docs |
21:05.56 | jbot | rumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
21:05.58 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
21:05.58 | bweschke | delmar: fair enough. like mog_work said, most places use common impedence but it was worth a shot. :) |
21:06.00 | perd | hmm |
21:06.18 | _Sam-- | http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI |
21:06.25 | delmar | bweschke, yeah i mean... NZ is supposed to have the same as the USA |
21:06.54 | delmar | bweschke, but even then, you just don't know whats going on.. and I dont have a sophometer to test it all :P |
21:07.06 | _Sam-- | you could do it all with something like php and sql as well, perd, just setup a table that would be checked for current connections...if connections > 1 , then dont call |
21:07.18 | _Sam-- | and just run the agi that checks the table before each call |
21:07.47 | delmar | u would think that the wctdm module load could check /etc/fxotune.conf and set stuff at load time.. :P |
21:07.58 | _Sam-- | could probably even do it with the off the rack mysql CDR table |
21:09.13 | _Sam-- | it would be probably easiest to just ask your IAX provider to limit you to 1 outgoing ( i think for me that would be easiest) |
21:09.34 | delmar | bweschke, ok so im lookin at zconfig and there are a few other optimizations which im not sure i should touch... CONFIG_ZAPTEL_MMX ? (its intel cpu). ? |
21:10.05 | bweschke | sure - if it's an intel cpu u can turn it on |
21:10.54 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
21:11.45 | delmar | bweschke, now this 2100 Hz echocan disable thing i always wondered about... does that turn off echocan when say.. a fax comes in? detects that kinda tone .. or what? |
21:12.06 | bweschke | yes |
21:12.18 | bweschke | u don't want the echo can interfering with faxes |
21:12.33 | delmar | bweschke, ok so i will try this out :P |
21:12.52 | bweschke | but that probably won't fix your echo problem if you didn't have it on originally. :/ |
21:13.08 | delmar | bweschke, no i was reading it the wrong way.. thats to turn OFF the thing that turns off the echo can LOL |
21:13.12 | bweschke | a # of folks have reported good results with MG2 in forums |
21:13.25 | bweschke | so - that's probably one a bastion of hope |
21:13.41 | delmar | bweschke, ok then. compile time |
21:13.56 | mog_work | yes mg2 works well for people that need it |
21:14.48 | delmar | should I re-do fxotune with the new compiled module and stuff? |
21:14.56 | delmar | think i might anyway... |
21:17.26 | delmar | sage: fxotune |
21:17.27 | delmar | <PROTECTED> |
21:17.37 | delmar | Just looking at that.... |
21:17.50 | delmar | i dont know what the number is.. if any.. to "clear" dialtone off the line.... |
21:17.51 | *** join/#asterisk asteriskmonkey (n=phil@HSE-Toronto-ppp300757.sympatico.ca) |
21:17.58 | asteriskmonkey | damn it my asterisk is going to crap.. |
21:18.22 | asteriskmonkey | i just called a client not only do i hear myself but they here them selves but neither of us can hear each other echo |
21:18.39 | Beirdo | delmar: that means "hit a key so the dialtone stops" |
21:18.46 | Beirdo | just tell it what to dial |
21:18.46 | asteriskmonkey | was sipsoftphone > asterisk > vonage client |
21:19.00 | delmar | Beirdo, that i realise |
21:19.10 | delmar | Beirdo, but there IS no key in NZ to do that ... that i know of |
21:19.19 | asteriskmonkey | does anyone here have a pri? |
21:19.22 | *** join/#asterisk _T3_ (n=rposada@200.63.231.210) |
21:19.28 | Beirdo | what happens when you dial the first digit of a number? |
21:19.34 | Beirdo | does the dialtone stop? |
21:20.06 | delmar | Beirdo, there is silence... then after a few seconds it will time out and give a sort of .. fast busy tone for ages. |
21:20.07 | _T3_ | hello |
21:20.18 | Beirdo | well, if the test is fast... |
21:20.38 | Beirdo | just don't dial the operator |
21:20.55 | *** join/#asterisk SERGEUS (i=sergey@195.112.98.13) |
21:20.58 | _Sam-- | asteriskmonkey: it could be firewall related |
21:21.01 | delmar | Beirdo, i will plug another phone into the line, wait for silence... then run the test.. and as soon as i head the card pickup the line ill hang up and let it have the line.. with silence. |
21:21.02 | _Sam-- | search one-way audio |
21:21.14 | _T3_ | anybody knows how much is tdm2400p |
21:21.25 | doug | argh. |
21:21.27 | SERGEUS | Hi everybody! |
21:21.30 | *** join/#asterisk _dave01_ (n=Miranda@21.208.65.212.contactel.net) |
21:21.40 | delmar | Beirdo, like.. u can dial 1, wait for 5-6secs.. it will do fast busy for maybe.. a minute.. then go silent. so ill do it like that sorta thing |
21:21.42 | bweschke | t3: pricing is up on voipsupply.com and I believe Digium's site too |
21:22.05 | Beirdo | depends on how that test works |
21:22.07 | SERGEUS | i have a lot of crashes since NOV 02, caused by channel.c - anybody has the same problem? |
21:22.12 | _dave01_ | how do I know that my Digium TE110P has new v2 firmware? |
21:22.13 | _Sam-- | asteriskmonkey : are you behind NAT on either side? |
21:22.17 | doug | Nov 8 15:32:34 NOTICE[57351]: chan_sip.c:7532 handle_request: Registration from 'boob<sip:boob@asterisk.aaronsen.com>' failed for '140.221.241.239' |
21:22.20 | doug | wtf |
21:22.29 | _T3_ | no in digium no, i will check in the other site thanks |
21:22.32 | Beirdo | I did an echo-cancellation algorithm that worked in that 5-6s |
21:22.33 | doug | why would it fail with that? |
21:22.39 | doug | and no more details? |
21:22.57 | doug | lots of |
21:22.58 | doug | Nov 8 15:33:17 WARNING[57351]: chan_sip.c:598 __sip_xmit: sip_xmit of 0x8120704 (len 430) to 140.221.241.250 returned -1: Invalid argument |
21:23.02 | doug | as well |
21:23.07 | doug | with eyebeam sip client |
21:23.14 | delmar | Beirdo, oh? |
21:23.25 | Beirdo | that wasn't for asterisk though |
21:23.29 | delmar | Beirdo, added to * or not? |
21:23.36 | Beirdo | a long time back |
21:23.46 | delmar | Beirdo, did it work? |
21:23.53 | Beirdo | so I don't know how THAT test is working |
21:24.02 | Beirdo | it did for our application yess |
21:24.08 | delmar | cool |
21:24.33 | Beirdo | it was for ship-to-shore impedance matching for a shipboard PBX system |
21:24.49 | delmar | nice |
21:26.41 | harryvv | ohh Beirdo, how did it work out? |
21:26.55 | Beirdo | pretty well, actually |
21:26.58 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
21:26.58 | harryvv | there is one other person who made a asterisk system for ships linked to a sat. |
21:27.45 | Ariel_ | ManxPower, I may have a new customer in your area for you. Do you have some time to talk about this? |
21:28.04 | Beirdo | dunno if they still use it or not, but the idea is... the first time they hook up the lines when porting at a new place, they echo-train and impedance match |
21:28.21 | Beirdo | I never got to test it in the field as I quit before then |
21:28.21 | Beirdo | heh |
21:28.26 | ManxPower | Ariel_, No time right now. What area? |
21:28.27 | enemy | How to set unlimited timelimit when adding to a queue in extensions.conf? |
21:28.35 | ManxPower | I'm dealing with Cisco access lists at the moment |
21:28.36 | Beirdo | but I think there are some frigates with it live |
21:28.51 | Beirdo | and maybe an aircraft carrier still |
21:29.10 | Ariel_ | ManxPower, it's not for today. But it's in the Monroe?? area |
21:29.15 | Ariel_ | it's a hospital |
21:29.27 | ManxPower | Ariel_, that's about 2 - 3 hours east of me |
21:29.38 | Ariel_ | Bummer |
21:29.49 | harryvv | access control list? |
21:30.43 | Beirdo | harryvv: http://www.drs.com/products/index.cfm?gID=18&productID=203 |
21:31.05 | *** part/#asterisk hhoffman (n=hhoffman@tor/session/x-395008eb87af861e) |
21:31.15 | tmccrary | is it possible to have two Asterisk PBX communicate via speex codec, but have the phones that register on the pbx be using ulaw? |
21:31.27 | Beirdo | ugly fucker, but there it is. |
21:33.16 | delmar | Beirdo, hey thats interesting.. what it does is.. picks up the line.. dials ie. 1, then starts test tones ascending in pitch only for a few seconds, then hangs up, goes off hook again, dials 1 to clear tone, then continues until finished the pitch cycle.. then picks up and dials 1 and repeats the process.. musta done it about 20 times.. anyway the test are short enuf that they would fit within the silence period |
21:33.29 | Beirdo | yup |
21:33.33 | Beirdo | that sounds about right :) |
21:34.02 | SwK[Work] | pile of shit |
21:34.07 | SwK[Work] | thats all it is |
21:34.11 | SwK[Work] | a big stinking PoS |
21:34.16 | delmar | Beirdo, however this echo test blows... the echo while off hook vs a call in progress will be vastly different.... its only testing this side of the exchange.. i dont think thats an accurate test. |
21:34.29 | SwK[Work] | any AudioCodes users around? |
21:34.32 | Beirdo | that will fix impedance mismatch |
21:34.43 | Beirdo | there's no easy way to fix remote echo |
21:35.23 | delmar | Beirdo, umm no.. at what point did it measure the line with a sophometer and work out the correct impedance settings? that was purely echo/audio testing.. nothing to do with resistance |
21:35.45 | Beirdo | local echo is almost 100% caused by impedance mismatch |
21:35.52 | delmar | i agree |
21:36.07 | Beirdo | so what it was actually testing was impedance mismatch |
21:36.08 | delmar | but this card does NOT have a sophometer built in.. i can assure you :P |
21:36.14 | *** join/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com) |
21:36.15 | delmar | no |
21:36.18 | delmar | its testing audio |
21:36.20 | delmar | not resistance |
21:36.22 | Beirdo | gah |
21:36.35 | delmar | resistance = impedance. |
21:36.45 | Beirdo | you don't need to measure the impedance to test for mismatch |
21:36.55 | delmar | heh :P |
21:36.59 | Beirdo | the test is by minimizing the echo by fiddling the impedance |
21:37.23 | Beirdo | at minimum echo, your complex impedance is matched as best you can |
21:37.31 | delmar | see.. even u agree its not testing impedance.. its testing for mismatch by way of echo detection using an audio test.. nothing to do with resistance/impedance. |
21:37.47 | Beirdo | it is NOT nothing to do with impedance |
21:37.57 | delmar | Beirdo, ok i get what u are saying |
21:38.05 | Beirdo | that's what is causing the echo it's measuring :) |
21:38.40 | delmar | Beirdo, it cant test the resistance, but can test echo and fiddle it's own impedance matching and re-test the echo/audio to see how that works out... hence the number of tests |
21:38.47 | Beirdo | bingo |
21:38.49 | Beirdo | :) |
21:39.06 | delmar | now that makes sense :P |
21:39.13 | Beirdo | the impedance of a line is partially resistive, partially inductive or capacitive |
21:39.27 | *** join/#asterisk citats (n=james@bgp925576bgs.brghtn01.mi.comcast.net) |
21:39.43 | Beirdo | most SLIC chips will let you tweak the impedance it presents to allow you to try to match the line |
21:40.06 | delmar | so its not testing and adjusting some software echncan or nuthin it's purely trying to illiminate echo by adjusting it's own matching to the line |
21:40.16 | delmar | well thats better than nothing.. but still doesn't work :P |
21:40.20 | Beirdo | that would be my guess |
21:40.32 | synthetiq | is there a softphone operators console for asterisk? |
21:40.38 | delmar | im gonna run another test with the phone on-hook, see what it says, then try the new mg2 alg etc. |
21:40.43 | Beirdo | that depends on if that's how they implemented it. that's precisely how I had implemented it |
21:40.45 | delmar | synthetiq, yep |
21:40.56 | synthetiq | where can i find it |
21:41.01 | delmar | synthetiq, CVS |
21:41.20 | delmar | synthetiq, gastman |
21:41.25 | Beirdo | and we had the fun of using a programmable DSP to do the hard work |
21:41.27 | delmar | synthetiq, at least I think thats it. |
21:41.33 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
21:41.49 | delmar | Beirdo, cool |
21:42.10 | Beirdo | that was... oh GOD. almost 10 years ago |
21:42.11 | Beirdo | damn |
21:42.14 | *** join/#asterisk it0 (n=it0@zwanebloem.xs4all.nl) |
21:42.17 | _dave01_ | anyone has te110p with v2? |
21:42.31 | *** join/#asterisk kn0x (n=nunya@tor/session/x-bc08ecc4ffc96481) |
21:42.42 | it0 | wow, this channel is full! |
21:42.48 | kn0x | why isnt nickserv working!?!?!? |
21:42.57 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
21:42.59 | it0 | kn0x: it just idéd me? |
21:43.21 | kn0x | whoops |
21:43.22 | synthetiq | wow thats pretty nice |
21:43.23 | kn0x | nevermind |
21:45.11 | it0 | i'm quite an * noob but, it's working for me, pretty well actually, still I would like to know how I can dial someone from the console and play a sound? i guess it's trivial but don't know how... |
21:46.09 | dudes | it0 - have an extension that plays the sound? dial 1234@context (if I recall correctly; haven't used console dial for awhile.) |
21:46.38 | shido6 | chan_oss? |
21:46.50 | it0 | dudes: so I set up an external number as an extention, then issue the playback command? |
21:47.01 | syle | chan_alsa |
21:47.05 | dudes | I'd think that'd work |
21:47.17 | *** join/#asterisk Flauto (n=zhao@71.194.38.168) |
21:47.22 | dudes | You could do it via a callfile in the event dial from console doesn't work |
21:47.38 | it0 | what is a callfile? |
21:48.05 | dudes | voip-info.org (you make a call file and drop it to /var/spool/asterisk/outgoing |
21:48.15 | dudes | and it places the call and drops it to an extension if it's answered |
21:49.27 | it0 | thanks I'll look it up |
21:53.48 | _dave01_ | how do I know that I have new v2 firmware? |
21:53.54 | mog_work | dmesg |
21:54.24 | _dave01_ | TE110P: Setting up global serial parameters for T1 FALC V1.2 |
21:54.43 | _dave01_ | nothing about version |
21:54.51 | mog_work | te110p has only 1 version |
21:54.58 | mog_work | v2 refers to the quad cards |
21:56.01 | _dave01_ | Ops. I didn't know that. |
21:56.03 | _dave01_ | Thanks. |
21:56.09 | mog_work | no problemo |
21:56.12 | _dave01_ | looks like v2 is for dual and quad only |
21:56.20 | mog_work | yes |
21:56.37 | myke420247 | the * console isn't a full phone, is it? you can't stick a headset in the sound card on your * box and use that as a phone? |
21:57.04 | mog_work | yes you can |
21:59.18 | shido6 | yep |
22:05.47 | *** join/#asterisk kn0x (n=root@ppp-69-222-164-89.dsl.emhril.ameritech.net) |
22:06.15 | kn0x | can anyone assist me with creating asterisk init startup scripts for gentoo? |
22:06.26 | mog_work | there is one ! |
22:06.31 | mog_work | in contrib |
22:06.32 | kn0x | i used the cvs build and im stuck.. |
22:06.32 | mog_work | i believe |
22:06.36 | kn0x | contrib? |
22:06.40 | mog_work | asterisk/contrib |
22:06.48 | mog_work | asterisk/contrib/init maybe |
22:07.03 | kn0x | /var/lib/asterisk/contrib? |
22:07.10 | kn0x | wat is the full location |
22:07.36 | mog_work | no its in the source |
22:07.40 | mog_work | its not installed by default |
22:08.11 | delmar | ok, tried lots of stuff. this echo is just crap |
22:08.23 | kn0x | im looking through the contrib/scripts |
22:08.33 | delmar | im going to go back to a regular PBX phone system and attach Asterisk as a VoIP gateway. fuck this. |
22:08.43 | mog_work | sorry delmar |
22:08.50 | kn0x | ahhaa! |
22:08.53 | kn0x | init.d |
22:08.56 | kn0x | thanks mog |
22:09.15 | harryvv | delmar, whats the problem? |
22:09.35 | mog_work | be back in a bit |
22:10.31 | harryvv | delamr having echo problems? |
22:13.46 | delmar | harryvv, sorry was afk, wife etc. |
22:13.50 | delmar | harryvv, yeah |
22:13.54 | delmar | harryvv, echo pain in the ass |
22:14.08 | delmar | harryvv, it WAS working ok |
22:14.20 | delmar | I think HEAD broke it |
22:14.57 | ard | Did you also use new zaptel drivers? |
22:15.07 | delmar | so calls outbound are fine.. still some faint echo, and holding phone up to mouth and giving it shit causes more... |
22:15.14 | ard | part of the echo can is in the kernel... |
22:15.31 | delmar | calls inboudn tho.. are crappy especialyl at the start but only a little improvement after 10secs.. still too much echo |
22:15.36 | delmar | ard, yes zaptel as well |
22:15.47 | ard | Hmmmm |
22:15.55 | harryvv | zaptel is involved with the echo? |
22:16.03 | delmar | i have been using * with TDM400 for months and its been running great... |
22:16.04 | ard | maybe you have another echo canceller selected... |
22:16.21 | delmar | the echo is SIP to/from * via PSTN |
22:16.33 | delmar | so yes.. Zaptel FXO port on the TDM400 |
22:16.47 | ard | harryvv : there are 2 echo's to cancel: the local echo which is not only done by right impedance matching.... |
22:17.21 | ard | delmar : I can only tell you that I am a very happy isdn user ;-)... |
22:17.48 | demetrio | when I call freeworlddialup test numbers like echo test or time, they will ring then answer but say nothing. Is it possible that this is becaus I'm behind NAT? |
22:17.50 | ard | I've tried writing a softphone with a headset and a modem, and I now know how painful analog lines are :-( |
22:18.21 | ard | demetrio : if you are using SIP, then it probably is... |
22:18.36 | harryvv | ard i know |
22:18.50 | demetrio | ard: using IAX instead will help? |
22:18.51 | ard | delmar : the kernel drivers of zaptel have special echo cancelling stuff... |
22:19.07 | delmar | rxgain/txgain is 0.0 and has been this way for months until I was playing with it the otehr day trying to get incoming/outgoing fax to work right, but its back to 0.0 again now |
22:19.11 | ard | demetrio : yes. But I heard rumours that the iax server of fwd has some problems lately |
22:19.18 | delmar | ard, yeah i know.. tried kb1 and mg2 |
22:19.19 | harryvv | echo is also known as impedence mismatch, swr , and who knows how many terms I have learned in the last 15 years. |
22:19.22 | demetrio | ard: ok, thanks |
22:19.28 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-13-119.cybersurf.com) |
22:19.44 | ard | harryvv : but a mere impedance mismatch can be corrected by software... |
22:19.53 | delmar | im going to go back to an older Asterisk for testing... just because I can |
22:19.59 | harryvv | and hardware |
22:20.00 | ard | I was able to cancel out the local echo completely |
22:20.09 | ard | in software :-).... |
22:20.26 | ard | but far-end echo and timing problems made it really hard. |
22:20.41 | ard | (I used the echo-canceller from speex) |
22:20.49 | *** join/#asterisk dustyservers (n=dustyser@d198-53-254-205.abhsia.telus.net) |
22:20.52 | ard | well... |
22:21.00 | ard | I am off to bed now |
22:21.11 | dustyservers | can some one tell me what hard will I will need outher then freebsd box? |
22:21.21 | dustyservers | to setup asterisk |
22:22.49 | *** join/#asterisk luite (i=luite@belphegor.deadlysins.nl) |
22:22.51 | delmar | dustyservers, do u want to attach to a line? |
22:22.59 | dustyservers | yes I do |
22:23.04 | dustyservers | to my bussine phone line |
22:23.06 | delmar | dustyservers, doesnt need to be freebsd, can be RH, Debian etc. |
22:23.14 | delmar | dustyservers, PSTN? ISDN? |
22:23.21 | dustyservers | yuppers |
22:23.23 | luite | what does this error mean, when starting asterisk: asterisk: relocation error: /usr/lib/asterisk/modules/res_watchdog.so: undefined symbol: ast_load |
22:23.24 | delmar | which? |
22:23.29 | dustyservers | umm pstn |
22:23.52 | delmar | dustyservers, then u will need an FXO card... good luck with fuckin echo |
22:24.11 | dustyservers | dose it echo bad? |
22:24.26 | delmar | dustyservers, for most no, for some its a cunt |
22:24.35 | dustyservers | really |
22:24.45 | delmar | no.... |
22:24.46 | dustyservers | that sucks |
22:24.47 | freat | delmar: what are you like 16 years old or something? |
22:24.54 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
22:25.05 | delmar | freat, no im just pissed off. |
22:25.29 | luite | ah, found it, there were still some asterisk 1.0.9 files left in /var/lib/asterisk/ |
22:25.38 | freat | delmar: this is a bit more professional of a forum than some you may be used to |
22:25.49 | freat | delmar: please try to watch the language |
22:25.49 | delmar | freat, i wend thru all this nonsense early this year.. with an X100 card... bought a TDM400.. problems solve.d.. now they are back again because Digium are cowboyz and borke the echocan stuff. |
22:26.06 | freat | delmar: I hear that |
22:26.30 | dustyservers | I know a few people using asterisk and they say it good |
22:26.39 | dustyservers | so is it a good chose for pbx phone systme? |
22:26.41 | dustyservers | system |
22:26.42 | delmar | dustyservers, yep. its fantastic.. very powerful |
22:26.51 | delmar | dustyservers, i mean.. the echo asside... i love it |
22:26.53 | dustyservers | ok cool I will try it out then :D |
22:26.57 | freat | dustyservers: plenty of people us it in production environments with lots of phones |
22:27.06 | dustyservers | as long as it works |
22:27.08 | dustyservers | that all I care |
22:27.28 | dustyservers | as 3com charges way too much money for there system |
22:27.32 | dustyservers | so I rather build my own |
22:27.43 | delmar | dustyservers, getting it to work might be a fight.... i mean..depending on what you want to get out of it.. there is so much stuff you can do.. hence so much to learn.. lots of stuff to debug.. |
22:27.59 | dustyservers | oh ic |
22:28.18 | dustyservers | dose the auto atence works good too? |
22:28.19 | freat | dustyservers: try asterisk@home to start with is my recommendation |
22:28.35 | dustyservers | that what was mhy plan |
22:28.37 | dustyservers | :D |
22:28.38 | delmar | dustyservers, just takes time.. and plenty peeps in here are guru's so.... u will get there |
22:28.45 | dustyservers | to make sure it suit my needs |
22:29.15 | freat | if you want basic IVR and autoattendant you can do all that w/ @home until you get more comfortable at the command line |
22:29.18 | dustyservers | and do any of you use this at more then on office? |
22:29.19 | delmar | i didn't go the way of asterisk @ home or any other kinda user-friendly thing.. just ripped into it manually. |
22:29.22 | *** join/#asterisk gman55s1f (n=joe@128.100.33.138) |
22:29.24 | oogle | so what's the difference between an opteron 280 and 880 besides 1000 dollars? |
22:29.41 | delmar | dustyservers, actually I have it at more than one location.. and tie calls between here and there over IAX2.... |
22:29.51 | luite | oogle: the number of hypertransport links |
22:29.53 | freat | dustyservers: I've got 3 offices each w/ about 15-20 phones |
22:29.54 | dustyservers | then that wokrs good |
22:29.58 | delmar | dustyservers, but i am having other issues with that which I haven't got to spend time on yet. |
22:29.59 | oogle | luite: thank you |
22:30.01 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
22:30.02 | freat | not that big but enough |
22:30.10 | gman55s1f | question -- after a Dial command to one of my SIP phones, the console shows "Called 510" (the extn), and then continues with a busy script.. The phone blips as if it almost started ringing... |
22:30.16 | luite | oogle: the 880 has 3 ht links, and you can use it in 8-way smp |
22:30.23 | delmar | dustyservers, it works mint apart from some echo im getting but .. there shouldnt be any echo at all so .. its a fault with my situation |
22:30.29 | dustyservers | like I know how to setup freebsd all that |
22:30.37 | luite | oogle: the 280 only in dual or single cpu servers |
22:30.38 | dustyservers | as that what I have for a server at the office right now |
22:30.51 | dustyservers | but would just like to try asterisk to see how good it would work thought.. |
22:30.58 | delmar | dustyservers, most people want to use something that runs on linux.. and the first big problem is that they don't know anything about linux.. so u are 90% of the way there |
22:31.00 | dustyservers | I will have to definely play iwth it |
22:31.03 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
22:31.21 | dustyservers | I played with linux some too |
22:31.26 | delmar | dustyservers, just point your cvs at digium and grab it and off u go |
22:31.34 | dustyservers | oh that easy |
22:31.52 | dustyservers | I will definely play with it I will need a card to plust in my phone line into it right? |
22:32.13 | dustyservers | or can I test it withput a phone line? |
22:32.18 | dustyservers | without |
22:32.22 | rculp | quick question: I'm getting the following error when I try to test from xlite client to xlite client |
22:32.23 | delmar | dustyservers, u can use it without a phone line |
22:32.27 | rculp | I think I missed something |
22:32.29 | rculp | here's the error |
22:32.30 | rculp | Nov 8 17:16:39 WARNING[10458]: chan_oss.c:581 setformat: Unable to re-open DSP device /dev/dsp: No such device |
22:32.36 | syle | more feature requests for billing system? |
22:32.38 | delmar | dustyservers, but obviously need one if u wanna interface to your PSTN |
22:32.40 | rculp | forgive my noobness :) |
22:32.50 | delmar | rculp, nah |
22:32.57 | dustyservers | ok thanks for the help |
22:32.59 | delmar | rculp, i dont have that setup either. its not needed :P |
22:33.07 | dustyservers | that all I need to know for now.. |
22:33.28 | rculp | delmar: hrm, xlite to xlite tests are broken then. I must need to do something else. first asterix setup |
22:33.46 | delmar | rculp, br0ken how? |
22:33.49 | gman55s1f | when calling one of my SIP phones, the console reads Dial(... , 'SIP/510'), and then says 'Called 510', and then launches into a busy script |
22:34.05 | gman55s1f | the phone registers a missed call but never really rings |
22:34.11 | rculp | delmar: calls are not going through. I just think I need to do some more testing |
22:34.18 | delmar | gman55s1f, then the phone was unreachable |
22:34.19 | rculp | delmar: I edited the sip.conf file |
22:34.23 | *** join/#asterisk brownjor (n=Jordan@c-67-185-209-222.hsd1.mi.comcast.net) |
22:34.29 | rculp | delmar: just don't think I setup the dialplan right |
22:34.39 | gman55s1f | delmar: it seems as if it rings for a split second.. |
22:34.43 | delmar | rculp, check extensions.conf must be a dialplan issue |
22:34.48 | rculp | delmar: testing to my test extension doesn't ring through either |
22:34.52 | rculp | delmar: will do |
22:35.28 | gman55s1f | delmar: plus, I'm using a dialparties script to check the line is free before calling |
22:36.02 | delmar | gman55s1f, do ... show sip peers, sip debug peer 510, place a call, copy all the output from those into pastebin and lets have a look |
22:36.53 | delmar | err.. sip show peers * |
22:36.57 | delmar | :P |
22:37.10 | rculp | delmar: I'll read up more on extensions |
22:37.18 | rculp | and ask again for help tomorrow if I don't have it figured out :) |
22:37.22 | rculp | thx |
22:37.40 | *** part/#asterisk rculp (n=rculp@66.173.240.20) |
22:37.54 | *** join/#asterisk swebb (n=swebb@216.183.121.53) |
22:38.05 | delmar | gman55s1f, to work out if its your script, remove it from the picture.. setup some simple dialplan stuff and test calling that way |
22:38.13 | delmar | gman55s1f, then debug the script. |
22:38.23 | delmar | gman55s1f, or whatever :P |
22:40.35 | *** part/#asterisk mazza[W] (i=mazzanet@unaffiliated/mazzanet) |
22:40.51 | gman55s1f | delmar: http://pastebin.ca/28048 |
22:40.57 | gman55s1f | delmar: I don' |
22:41.05 | gman55s1f | delmar: I don't think it's the script's fault.. |
22:41.18 | *** join/#asterisk shimi (n=shimi@unaffiliated/shimi) |
22:42.24 | shimi | Hi all. I was wondering: Is it possible to make some asterisk magic, such that someone can call the PBX, and "dial" while "talking" with asterisk (through touchtones, of course), to a different number, outside asterisk, and get connected? If this can be password protected, even better! Any ideas? |
22:44.02 | justinu | shimi: look up DISA |
22:44.27 | gman55s1f | delmar: it would appear that it's deciding to send a 'cancel' message.. |
22:44.41 | delmar | gman55s1f, indeed |
22:45.46 | delmar | anyone else see whats going on there? |
22:46.07 | delmar | this happens from any client to any other client? |
22:46.21 | gman55s1f | delmar: yes... the identical phone sitting right next to it (extn 511) |
22:46.46 | gman55s1f | delmar: I don't think the phones are faulty.. I had everything up and running a few weeks ago.. but I had to reinstall asterisk today.. |
22:46.58 | delmar | remove sensitive bits and pastebin your sip.conf and the relevant contexts from extensions.conf |
22:47.00 | shimi | justinu, thanks. I looked up in google, I found numerous leads. is there something official from digium? |
22:47.17 | justinu | not really, but voip-info.org is a good place to read up on asterisk |
22:47.20 | justinu | ~docs |
22:47.22 | jbot | hmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
22:47.28 | shimi | ok, thanks again :) |
22:47.30 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
22:47.42 | shimi | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DISA I assume :) |
22:48.19 | justinu | right-o |
22:48.30 | delmar | Whats the story with zaptel and asterisk in terms of versions..is for example.. zaptel v1-0-9 compatible with asterisk HEAD? |
22:48.54 | gman55s1f | delmar: sip.conf: http://pastebin.ca/28050 |
22:49.07 | *** join/#asterisk liran_ (n=liran@80.178.120.123.adsl.012.net.il) |
22:49.22 | enemy | Nov 8 23:46:31 WARNING[31134]: chan_zap.c:7570 zt_pri_error: PRI: !! Not good - head of queue has not been transmitted yet ..... (anyone seen this one?) Running BRIstuff (latest). |
22:49.31 | liran_ | heys |
22:49.52 | delmar | gman55s1f, ok firstly.. lets look at some network info.... |
22:50.01 | gman55s1f | delmar: something, with extensions appended: http://pastebin.ca/28052 |
22:50.04 | delmar | gman55s1f, the phones, and the * box are all talkingon the same network ranges? |
22:50.09 | gman55s1f | delmar: yep |
22:50.21 | delmar | gman55s1f, ok set nat=no |
22:50.35 | gman55s1f | delmar: I've had issues with that before.. nat=no doesn't work |
22:50.42 | gman55s1f | delmar: has to do with our switches or something |
22:50.48 | delmar | gman55s1f, you are not behind nat.. you are not using nat. |
22:51.02 | delmar | gman55s1f, ok this sounds dodgy already.... |
22:51.10 | gman55s1f | delmar: I've give it another shot... |
22:52.22 | gman55s1f | delmar: ok --- I set nat=no,... no visible change |
22:52.27 | delmar | gman55s1f, ok also there is no codec definition for any of them..at the end of those sip entrys.. do... disallow=all then allow=(codec) .. which ever codec u prefer to use with the phone for now... and as long as the phone supports it. |
22:52.27 | gman55s1f | delmar: same result. |
22:53.19 | delmar | we are not done |
22:53.33 | delmar | and dont forget when making thos echanges to do either.. reload, or sip reload |
22:53.42 | gman55s1f | delmar: yep |
22:53.44 | delmar | ok lookin at your extensions.conf... |
22:53.56 | justinu | g729a sounds like shit |
22:54.03 | justinu | why would anyone pay 10 bucks a channel for that? |
22:54.21 | gman55s1f | delmar: thanks -- gtg-- be back in a bit |
22:54.28 | liran_ | I haven't worked with Asterisk before but I've been asked by my firm to create a perl program to analyze the Master.csv (which is supposed to be the log of all calls/duration/etc) file and create a sort-of a billing system out of it. This seems too far-fetched, isn't there already some open-source billing system I can use to integrate into Asterisk? |
22:54.39 | delmar | gman55s1f, ? |
22:55.44 | delmar | gman55s1f, ok the info u pasted on http://pastebin.ca/28052 is not complete...whats in your from-internal context? those 4 lines? .. pastebin the exten-vm Macro ... |
22:56.24 | demetrio | anyone knows how to retrieve the freeworlddialup's IAX number? should they send it to you via email or something? |
22:56.36 | delmar | demetrio, i think so |
22:56.43 | shimi | demetrio, you need to request one in their site |
22:57.01 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
22:57.27 | demetrio | shimi: that's what I've done, but I can't see where it is, and have received no mail |
22:57.46 | demetrio | well, I'll wait |
22:59.25 | shimi | they said it should take about 10 minutes, IIRC |
22:59.33 | shimi | or was it one hour? I don't remember. |
22:59.37 | *** part/#asterisk mkrufky-gone (n=mk@68.160.103.77) |
23:00.14 | demetrio | 10 min |
23:00.25 | demetrio | I bet I won't see it till tomorrow though |
23:01.29 | shimi | what does Authenticate(XXXX) do ? |
23:04.20 | *** join/#asterisk Limbeaux (n=icechat5@ip-209-124-211-234.static.eatel.net) |
23:05.39 | *** join/#asterisk santiago (n=santiago@63.245.87.62) |
23:09.18 | *** join/#asterisk Limbeaux (n=icechat5@ip-209-124-211-234.static.eatel.net) |
23:11.19 | brownjor | Hey everyone, I was wondering what distro/OS do you all use for asterisk, and why? |
23:12.09 | luite | debian, because that's what I always use, and know best |
23:12.18 | syle | fedora, case developpers use it |
23:13.25 | delmar | brownjor, Debian... just because :P |
23:13.33 | Limbeaux | is anyone using Quantum Voice? |
23:14.27 | brownjor | luite & delmar, would you recommend learning Debian to run asterisk? How long does it usually take you to install asterisk and debian together? |
23:14.50 | shimi | the fastest way to install asterisk is Asterisk@Home ;) |
23:14.59 | delmar | brownjor, this is the trouble... people find that there is something Linux is doing that they want... getting to know Linux can be 90% of the effort |
23:15.12 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
23:15.17 | luite | brownjor: if you just want to run asterisk, there are other/faster options |
23:15.22 | *** join/#asterisk nvrs (i=RUR@toronto-HSE-ppp4256323.sympatico.ca) |
23:15.26 | luite | like shimi said |
23:15.27 | luite | :) |
23:15.30 | tzafrir_laptop | brownjor, I would naturally recommend Rapid |
23:15.43 | tzafrir_laptop | http://xorcom.com/ |
23:15.58 | delmar | Yeah, u are better to get an Astrisk distro which is based on .. say.. Debian or RedHat, but is setup to make life easy for you setting up Asterisk |
23:16.11 | shimi | and asterisk@home comes with cool stuff, like call monitoring, call logs, conference management, CRM, etc... all that, out of the box. |
23:16.28 | delmar | ah i have heard of Xorcom but not tried it out for myself yet.. its based on Debian too right? |
23:16.37 | brownjor | well I'm interested in doing it right... I'm trying to design a solution to work with a cisco call manager system... and I kinda want to get started in the right direction first... I know things about linux... just not debian... |
23:16.38 | tzafrir_laptop | yup. |
23:16.57 | tzafrir_laptop | and remains very close to it, so you can easily install any other Sarge package |
23:17.24 | delmar | is there a way to do it the other way around? |
23:17.41 | delmar | i dont wanna re-install an existing box say.. but wanna add the bits from xorcom or whatever? |
23:17.44 | luite | brownjor: debian is not that difficult to learn, fairly easy actually, will probably just take some time to get used to it, if you're comfortable with editing config files |
23:17.53 | tzafrir_laptop | delmar, Yes. http://rapid.dotsrc.org/ , basically |
23:18.25 | tzafrir_laptop | We also have there packages for amportal and destar |
23:18.42 | delmar | might be worth a look im thinkin |
23:19.05 | lesouvage | Is there a special reason for a lot of flat enoyning noise when using an ATA. My ATA is working but on the receivng site the noise is to loud. |
23:19.42 | *** join/#asterisk mrec (n=revenger@p54B00A80.dip0.t-ipconnect.de) |
23:19.54 | mrec | hmm |
23:20.00 | mrec | can anyone help me with following error: |
23:20.01 | mrec | Nov 9 00:18:53 NOTICE[6620]: chan_iax2.c:5876 socket_read: Registration of '719698' rejected: Registration Refused |
23:22.15 | *** part/#asterisk brownjor (n=Jordan@c-67-185-209-222.hsd1.mi.comcast.net) |
23:22.17 | delmar | mrec, could be any number of things. |
23:22.40 | delmar | mrec, sip client? |
23:23.27 | Math` | delmar: its chan_iax2 |
23:23.49 | mrec | I'm trying to get it work withfreeworlddialup.com |
23:24.04 | Math` | iax is currently broken at freeworlddialup |
23:24.09 | Math` | you're better off with sip |
23:24.10 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
23:24.43 | harryvv | well well |
23:24.45 | mrec | I'm behind a router will that work without any problems? |
23:26.07 | Math` | if you forward your ports properly, it shouldn't, but the best is... get a second nic for your linux box and use linux as router |
23:26.23 | Math` | this was asterisk isn't on a nat |
23:27.50 | Limbeaux | any suggestion for providers? The only one i looked at was quantumvoice.com |
23:29.08 | myke420247 | the grandstream gxp2000 is really a quality phone |
23:29.32 | myke420247 | if you put in too long of a string for the account name the firmware breaks in weird and entertaining ways |
23:29.34 | FuriousGeorge | i figured it out how to make call parking less annoying: users park calls on $USEREXTEN$PARKINGSPOT where userexten is obvious and parking spot is a number 1-X |
23:29.52 | delmar | hrm |
23:29.55 | FuriousGeorge | hey all |
23:30.09 | delmar | is the zaptel available via CVS.. in MAIN the same as HEAD? |
23:30.19 | *** part/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
23:30.20 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
23:30.31 | FuriousGeorge | hey all |
23:30.40 | sjobeck99 | hi, all, hope all is well, any one know native MOH? |
23:33.19 | FuriousGeorge | so ive been thinking about call parking. why dont we do this: calls get parked on ${PARKEREXTENSION}${PARKINGSPOT} where parking spot is a range of numbers starting >=1. so if im extension 101, and i get 10 spots, i can just xfer a call to 1010 to have the system park it for me, or just manually transfer to a parking spot, 1011 to 1019 |
23:33.29 | FuriousGeorge | what do you guys think |
23:33.42 | swebb | I've got a "wildcard" X100P card - what can I do with it and asterisk on a pots line in my home? Is it worth playing with? |
23:34.10 | FuriousGeorge | swebb: isnt that an fxo? you can recieve calls from ma bell |
23:34.23 | FuriousGeorge | but ive heard their quality is iffy |
23:34.27 | swebb | Got me if it's an fxo -. |
23:34.28 | ManxPower | ~fxofxs |
23:34.29 | jbot | [fxofxs] An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
23:34.29 | FuriousGeorge | sound wise |
23:34.54 | FuriousGeorge | ~furiousgeorge |
23:34.55 | jbot | [furiousgeorge] a knife-fighting monkey last seen with The Man with the Yellow Bat |
23:35.01 | myke420247 | i thought the idea of parking was that the spot wasn't tied to any extension |
23:35.02 | FuriousGeorge | ~jbot |
23:35.03 | jbot | jbot is, like, a nub |
23:35.17 | swebb | Ok, so I've probably got one of those. Can I do anything cool with it and asterisk, or do I need a T1 or something to get any mileage out of it? |
23:35.18 | myke420247 | otherwise why not just do an attended transfer |
23:35.57 | FuriousGeorge | myke420247: so have a global spot, im thinking about it this way: reverse transfer |
23:36.20 | FuriousGeorge | you cant "pull" a call on any channel ive used |
23:36.26 | delmar | ok guys.. i FIXED my echo problems. |
23:36.32 | myke420247 | yeah ok |
23:37.39 | delmar | can anyone tell me what the difference is in doing "cvs checkout zaptel" vs doing "cvs checkout -r HEAD zaptel" ?in terms of the code? IS there a difference? what version is HEAD and what version is the other one... ? |
23:38.09 | FuriousGeorge | i used some prop system at a law firm i worked for (way before i even knew what * was) and i remembered when we had someone on hold, one party could "pull it" or the other could xfer it |
23:38.15 | delmar | beacuse HEAD is utterly br0ken |
23:38.20 | delmar | in terms of echo |
23:38.50 | FuriousGeorge | :) |
23:38.52 | delmar | :P |
23:39.21 | swebb | Hey. Do I need a special card and special line from Ma-bell to do cool stuff, or can I do it with my normal POTS line? |
23:39.21 | FuriousGeorge | i guess they came out with a beavis and butthead movie, i think its on my mind |
23:39.30 | swebb | ... and my W100P card? |
23:39.47 | *** part/#asterisk Limbeaux (n=icechat5@ip-209-124-211-234.static.eatel.net) |
23:39.52 | FuriousGeorge | swebb: any analog line will do |
23:39.53 | justinu | settle down, beavis |
23:40.08 | FuriousGeorge | swebb: it will be as 'cool' as you pay mabell to make it, i suppose |
23:40.25 | FuriousGeorge | swebb: the /real/ cool part is what you make * do with the mabell line |
23:41.26 | swebb | So, I can do anything with that card and a normal POTS line? What can I *not* do? |
23:41.31 | delmar | so what I have is.. fxotune.conf with 1=11,0,0,0,0,0,0,0,0 in it... I have echotraining=800 and other things, currently txgain/rxgain set 0.0/0.0 .. i leave them as they are .. and using zaptel from HEAD.. it blows.. just the plain zaptel from doing a "cvs checkout zaptel" .. no customizations .. just make install (hence it uses KB1) and it works without echo |
23:41.41 | FuriousGeorge | for instance, why pay for voicemail from ma bell, when it would be much cooler to have * send you an email with an attachment |
23:41.52 | FuriousGeorge | swebb: good question... |
23:42.07 | justinu | you can't make caller ID work without paying ma bell for it |
23:42.10 | justinu | which sucks |
23:42.29 | linagee | justinu: of course. ma bell 0wn3z your pocketbook |
23:42.29 | *** join/#asterisk bumblefsck (n=bumblefs@69-160-145-156.ontrca.adelphia.net) |
23:42.37 | *** join/#asterisk TheCops (n=xdz@got.securebinary.com) |
23:42.38 | FuriousGeorge | justinu: technically yes, but thats not *'s fault, ma bell just isnt sending the info right |
23:42.39 | justinu | not mine |
23:42.48 | justinu | i have no connections into my home from them anymore |
23:42.50 | delmar | justinu, what sux is.. callerID info used to be passed anyway.. years ago... i remember hooking up a modem and setting it to bell ccitt or whatever it was.. and i could see incoming callerID |
23:42.51 | TheCops | someone is using a Rhino T1 channels banks ? |
23:43.01 | delmar | justinu, then the phone company realised they coudl turn it off.. and charge to turn it back on |
23:43.05 | justinu | heh |
23:43.06 | linagee | delmar: really?!? |
23:43.06 | FuriousGeorge | justinu: what happens when there is a nuclear fallout and the web is down? |
23:43.09 | linagee | delmar: wtf |
23:43.19 | justinu | FuriousGeorge: i'll have bigger problems than the web :) |
23:43.24 | delmar | linagee, yep here in NZ back in like.. 1990, i did this. |
23:43.46 | swebb | Cool, thanks guys. I guess I'll hook it up and play with it tonight. Thanks again! |
23:43.47 | FuriousGeorge | justinu: seriously though, one thing that is cool about ma bell, the only thing i suppose, is that their lines almost always work |
23:43.57 | FuriousGeorge | and they provide the power |
23:44.09 | justinu | yeah, RBOCs do have it down cold |
23:44.12 | linagee | delmar: maybe the electric company will decide it can turn your electricity off every morning and charge you to turn it back on. ROFL |
23:44.13 | FuriousGeorge | so even when you have none, they work |
23:44.36 | justinu | but the problem is, if the power goes out, every idiot picks up the phone and tries to call people |
23:44.36 | justinu | so all you get are reorders |
23:44.44 | *** join/#asterisk voipjoy (n=root@1.fix.netvision.net.il) |
23:45.06 | linagee | justinu: wait, ma bell's switch isn't big enough if everyone picked up the phone at once... |
23:45.07 | delmar | linagee, they already charge enuf to keep it turned on and make food money im sure they wont turn it off :P |
23:45.17 | FuriousGeorge | swebb: thats why i say, get one pots line, and a few voip dids, and if your pots line is the number every1 knows and you cant get it ported (or dont want to) just have it rollover on busy to the voip did |
23:45.20 | delmar | linagee, power here has trippled in the last 3-5yrs |
23:45.22 | FuriousGeorge | justinu: true |
23:45.51 | delmar | So can anyone please tell me... Zaptel on CVS... MAIN and HEAD must be aliases for versions I assume.. what versions ? |
23:45.54 | linagee | delmar: maybe they'll change the voltages and charge you to put it back. ROFL. screwy still to mess with people's heads |
23:45.59 | justinu | they have switches that service 60-120k phones |
23:46.00 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
23:46.06 | voipjoy | anybody can advice? kernel 2.6.14 problem: ztdummy.c:103:2: error: #error ztdummy requires 1000 hz jiffies |
23:46.17 | justinu | but they only have 10,000 DTMF receivers! |
23:46.18 | FuriousGeorge | delmar: the only thing i need to do in 1.0.9 to avoid echo (besides have clean wiring and less than crappy handsets) is echocancel=yes, and st jitterbuffer |
23:46.31 | SkramX | voipjoy: Israeli? |
23:46.32 | linagee | justinu: yes, but if everyone tried to call from switch A to switch B, do you think the interconnect would hold up? LOL |
23:46.41 | delmar | FuriousGeorge, you actually have a jitterbuffer set on your incoming line? |
23:46.42 | voipjoy | SkramX:yes |
23:46.46 | SkramX | Cool, Cool |
23:46.47 | justinu | no, they have even less trunks, i think |
23:47.04 | test34 | how would I check with GOTOIF if the incoming call has a specific area code ? |
23:47.10 | FuriousGeorge | delmar: incoming voip did |
23:47.22 | linagee | justinu: exaclty. which means you have a slightly higher change of reaching someone on the same switch in an emergency, and a lesser change of reaching someone even just a couple of miles away. ;-) |
23:47.23 | delmar | FuriousGeorge, oh ok thats diff. |
23:48.05 | linagee | justinu: which makes sense now why local calls are free and zone 3 calls aren't. |
23:48.20 | delmar | linagee, but seriously.. there used to be caller information on the lines here many years ago, and as time went by it was turned off, then a chargable service came along to turn it on again |
23:48.22 | FuriousGeorge | delmar: i mentioned jitterbuffer b/c the only time you should get echo is when voip goes to pots |
23:48.26 | justinu | linagee: never thought of that |
23:48.34 | FuriousGeorge | or viceversa |
23:48.52 | justinu | pure ip calls get echo all the time |
23:49.04 | justinu | shitty phones/headsets |
23:49.09 | justinu | poor acoustic coupling |
23:49.17 | linagee | poor user |
23:49.38 | delmar | FuriousGeorge, well in zapata.conf i have jitterbuffer commented out, and I seem to have fixed the echo. Clearly the zaptel from HEAD has an issue |
23:49.39 | voipjoy | SkramX: you? |
23:49.42 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
23:49.52 | FuriousGeorge | delmar: well, technically there is echo on every pots call, but as long as the other party is not overseas, your voice comes back to you fast enough that you dont notice. you just kinda expect to hear yourself, and you do |
23:50.01 | SkramX | voipjoy: Jewish, spoke hebrew somewhat fluently for a while. |
23:50.17 | delmar | FuriousGeorge, yeah i know a little about side-tone |
23:51.09 | FuriousGeorge | delmar: i didnt know zapata had a jitter buffer, i dont see how analog in could be jittering. for an incoming call from pots the only you shouldnt get jitter on your local lan |
23:51.12 | TheCops | someone is using a Rhino T1 channels banks ? |
23:51.15 | justinu | pots phones put sidetone on the speaker deliberatly |
23:51.28 | delmar | FuriousGeorge, now what gets me is why another situation i have.. does have echo.. almost like delayed side tone.. ... sip-hardphone(ulaw) to * ===IAX2/ulaw==== to * to sip-hardphone(ulaw) |
23:51.41 | FuriousGeorge | *for an incoming call from pots you shouldnt get jitter on your local lan |
23:51.52 | voipjoy | SkramX: good! |
23:52.12 | delmar | FuriousGeorge, no.. i mean.. calls from PSTN to * to SIP on the LAN is great |
23:52.18 | tzafrir_laptop | voipjoy, you should USE_RTC there, I believe |
23:52.21 | delmar | FuriousGeorge, heh it wasnt this morning but is again now |
23:52.42 | FuriousGeorge | delmar: wait im confused |
23:53.03 | test34 | what does: Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN' mean ?? my card isnt ready ? |
23:53.11 | FuriousGeorge | delmar: but im pretty sure if its all voip from origination to termination you shouldnt get an echo |
23:53.23 | delmar | FuriousGeorge, exactly |
23:53.26 | voipjoy | tzafrir_laptop: i should declare it by my self? |
23:53.58 | delmar | FuriousGeorge, but I do, altho i havent tested it again in a couple of days.. been busy with this pots/pstn echo problem here .. but now that this is resolved i need to look at it again |
23:53.59 | voipjoy | tzafrir_laptop: x64, kernel 2.6.14..asterisk 1.2beta2 |
23:54.08 | tzafrir_laptop | voipjoy, not sure |
23:54.16 | FuriousGeorge | delmar: is your phone trying to do some stuff that * is already doing, or doesnt need to do? for instance i find "voice activity detection" options on some voip clients isnt necessary |
23:54.22 | delmar | FuriousGeorge, what gives me the shits is that clearly.. the zaptel from HEAD, is br0ke |
23:54.23 | tzafrir_laptop | in zconfig.h? |
23:54.48 | delmar | FuriousGeorge, exactly.. i turn all that crap off. |
23:55.04 | jtodd | Hrm. Anyone getting this error on CVS checkout (cvs co asterisk) of HEAD after successful login? "cvs [checkout aborted]: Cannot check out files into the repository itself " |
23:55.12 | *** join/#asterisk MajestiK (n=MajestiK@S0106000024c058cc.ed.shawcable.net) |
23:55.14 | FuriousGeorge | delmar: i dont use head so i cant tell you if have found the same issue (thats why i wouldnt want to mess with head) |
23:55.15 | delmar | FuriousGeorge, none of that is turned on at either end.. yet both ends of the SIP to SIP call can hear themselves with delay.... |
23:55.41 | delmar | FuriousGeorge, oh i got an idea. sec |
23:55.47 | FuriousGeorge | delmar: just thinking aloud, for there to be echo there must be delay |
23:55.58 | delmar | FuriousGeorge, sure |
23:56.02 | delmar | FuriousGeorge, brb sec |
23:56.29 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
23:56.41 | voipjoy | tzafrir_laptop: i declared it ztdummy.c before 1st line |
23:56.57 | paryl | i just don't get it... i've tried EVERYTHING i can find on echo cancellation, but it still have it really bad at the beginning of incoming calls |
23:59.00 | FuriousGeorge | paryl: how long is the "beginning" |
23:59.04 | FuriousGeorge | first five seconds? |
23:59.20 | justinu | that's when the EC is "converging" |
23:59.41 | delmar | FuriousGeorge, ok get this... if a SIP client calls the * VM on the remote * box, not locally, and if I swear and curse at it..there is not one bit of echo |