| 00:00.28 | jeffspeff | adeel, is it a cause for concern? should i try with a different softphone and see what i get? |
| 00:01.08 | adeel | jeffspeff, as long as you can mak & receive calls it doesn't matter |
| 00:01.25 | adyer | jeffspeff: maybe the port on the host the sip is coming from |
| 00:02.00 | jeffspeff | adyer, you mean my computer? |
| 00:02.31 | drmessano | Thats normal |
| 00:02.41 | drmessano | Its showing the port the registration is coming from |
| 00:02.48 | drmessano | For internal, its going to be all over the place |
| 00:02.55 | adyer | jeffspeff: if I look at my pc running a softphone, I see port 10318, udp as open, and that matches what I get from sip show peers |
| 00:03.04 | drmessano | From external, it will all be 5060, because thats what passed thru the firewall |
| 00:03.39 | jeffspeff | ok, how can i specify for it to use 5060 on internal? |
| 00:03.42 | adeel | jeffspeff, some softphones give you finer control on what ports it'll use |
| 00:03.46 | drmessano | Oh god |
| 00:03.48 | jeffspeff | ok |
| 00:03.55 | drmessano | you dont have a problem |
| 00:03.59 | drmessano | Learn about how IP works |
| 00:04.04 | drmessano | I just explained it |
| 00:04.09 | drmessano | Source ports are irrelevant |
| 00:04.12 | jeffspeff | i have a CCNA, i know how IP works |
| 00:04.29 | drmessano | Thats meaningless to me |
| 00:04.44 | jeffspeff | Cisco Certified Network Associate |
| 00:04.53 | drmessano | I know what a CCNA is |
| 00:04.59 | drmessano | Again, not impressed |
| 00:05.01 | drmessano | Anyway |
| 00:05.03 | rob0 | Can I get your autograph? |
| 00:05.08 | adeel | hehe |
| 00:05.08 | jeffspeff | lol |
| 00:05.16 | JT | jeffspeff: it's not a good reflection on CCNAs tbh |
| 00:05.25 | JT | source ports are almost always random |
| 00:05.29 | drmessano | You should know what a source port is |
| 00:05.30 | drmessano | Yes |
| 00:06.36 | adeel | sweet, my rtcp logging patch works |
| 00:06.42 | JT | having a ccna only means you passed a 1.5 hour exam |
| 00:06.53 | JT | it doesn't mean you know how ip works |
| 00:07.04 | adeel | JT, every degree means that you passed an exam |
| 00:07.13 | adeel | or series of exams |
| 00:07.19 | drmessano | Passing a test is meaningless |
| 00:07.27 | drmessano | It means you can remember |
| 00:07.30 | drmessano | Doesnt mean you KNOW |
| 00:07.30 | JT | adeel: the requirement for passing a ccna is only passing a single exam though :) |
| 00:07.34 | drmessano | or that you LEARNED |
| 00:07.34 | rob0 | Like here, when you come in and brag about certifications, we give you the Third Degree. |
| 00:07.38 | drmessano | Memory != Skill |
| 00:07.41 | jeffspeff | yes, but knowing about IP's doesn't mean anything when messing with a new system (Asterisk) that you're not familiar with |
| 00:07.47 | adeel | drmessano, not entirely...it proves you can read, write, & comprehend =cp |
| 00:08.01 | drmessano | jeffspeff: Weak sauce, we're talking about IP, not asteriskl |
| 00:08.03 | drmessano | -l |
| 00:08.03 | JT | jeffspeff: almost all procotols over tcp and udp use random source ports |
| 00:08.05 | adeel | JT, ah |
| 00:08.24 | drmessano | adeel: Indeed |
| 00:08.36 | *** join/#asterisk marc7 (n=marc@S0106001c10243803.gv.shawcable.net) |
| 00:08.44 | jbeez | usually greater than 1023 |
| 00:08.54 | drmessano | Take for example, source ports |
| 00:08.57 | drmessano | lol |
| 00:09.17 | adeel | hahah |
| 00:09.43 | adeel | leave the guy alone...we've all asked stupid questions |
| 00:09.48 | drmessano | Do you understand binary? (0/1)? |
| 00:09.55 | JT | HZ |
| 00:10.05 | JT | i understand trinary |
| 00:10.11 | JT | 0/1/High Z/ |
| 00:10.16 | JT | :P |
| 00:10.37 | drmessano | There are 10 kinds of people in this world, those who know binary, and those who don't |
| 00:11.14 | *** part/#asterisk adyer (n=adyer@adsl-69-223-13-78.dsl.chcgil.ameritech.net) |
| 00:11.22 | *** part/#asterisk var1 (n=james@92-236-112-23.cable.ubr20.edin.blueyonder.co.uk) |
| 00:11.23 | coppice | did you that SETI have picked up this 10101010 pattern from a binary star? |
| 00:11.50 | unpaidbill | shit, i left my tinfoil hat at home today |
| 00:11.51 | unpaidbill | big mistake. |
| 00:11.53 | drmessano | coppice: It was me flicking a light switch.. sorry |
| 00:14.51 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:14.51 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:16.39 | *** join/#asterisk lirakis (n=user@cpe-68-175-38-65.nyc.res.rr.com) |
| 00:17.06 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 00:17.17 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 00:17.20 | JayTee52 | yeeeee-haw! |
| 00:18.15 | lirakis | Yeeeeeeeeeeeeetaw |
| 00:18.32 | *** join/#asterisk tobias (n=tobias@m135f36d0.tmodns.net) |
| 00:18.55 | lirakis | Errr.. Im not a redneck.. Just got a blue tooth keyboard for my n810 and wanted to try it |
| 00:18.55 | JayTee52 | I'm just all excited. I got my PRI circuits tested for inbound and outbound and it works!!! |
| 00:19.04 | lirakis | Nice |
| 00:19.13 | lirakis | Were both excited aboutcool stuff |
| 00:19.28 | colinm_ | cd .. |
| 00:19.38 | colinm_ | agh, damn focus |
| 00:20.13 | jasonwoot | So my boss asks me to interface asterisk with our time clock system, ie, don't let someone log in and take calls unless they are on the clock |
| 00:20.35 | JayTee52 | eeeewwww |
| 00:20.36 | jasonwoot | so I quoted whitney houston and told him to kiss my ass |
| 00:20.49 | russellb | heh, you should tell him, why don't want people to work for free? |
| 00:21.29 | jasonwoot | then in a moment of silent reflection, it ocurred to me that it might not be that difficult, but is a challenge really what I want these days? |
| 00:21.52 | denon | russellb: he doesn't want to get sued |
| 00:21.57 | russellb | denon: bah. |
| 00:22.04 | denon | he's probably in california .. it's a huge issue out there |
| 00:22.08 | russellb | wouldn't be too hard, in theory |
| 00:22.26 | JayTee52 | just tell him that the timeclock system is coded in Korean Assembler and Asterisk is in GCC and the two are imcompatible with each other. Maybe he's clueless enough he'll buy it. |
| 00:22.26 | denon | blue collars class-actioning their employers, claiming they were forced to work after hours for the same or no pay |
| 00:22.32 | jasonwoot | I'm thinking about pause/unpause function instead of agents |
| 00:22.51 | denon | just don't issue the phone a lease :) |
| 00:22.54 | Qwell | jasonwoot: yes, do that |
| 00:23.00 | Qwell | </3 agents |
| 00:23.24 | *** join/#asterisk za3toor (n=non@bas1-toronto12-1167878815.dsl.bell.ca) |
| 00:24.26 | jasonwoot | my problem is actually kinda stoopid; when you clock in, if it runs a script to unpause your ext, if we're running a queue, by the time you get to your desk, you've missed a call and it would auto log you out |
| 00:24.39 | jasonwoot | clock, run, answer |
| 00:24.54 | Qwell | why not just have the login macro check the timeclock system? |
| 00:25.01 | denon | could just have the queue check the timeclock instead |
| 00:25.03 | denon | yeah, what qwell said |
| 00:25.04 | Qwell | or, have the timeclock system write to something you can read, whatever |
| 00:25.06 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:25.25 | jasonwoot | thankfully, it's SQL and not some proprietary crapola |
| 00:26.33 | *** join/#asterisk C4colo (n=DJpyro@66.185.107.14) |
| 00:26.59 | jasonwoot | all I have to do now is tell him it will take a week, do it in 5 days, and look like a hero' |
| 00:27.34 | Qwell | could probably do it in an hour :p |
| 00:27.35 | C4colo | anyone here know polycoms real well? |
| 00:28.04 | jasonwoot | no, I mean literally, do it in an hour, 5 days from now |
| 00:28.16 | jasonwoot | cause when you rush, well that's when accidents happen |
| 00:28.20 | C4colo | heh |
| 00:28.35 | C4colo | yea, I do the same but say it will take a week and then do it in 6.5 days =) |
| 00:28.45 | C4colo | I mean, start it in 6.5days |
| 00:28.46 | C4colo | lol |
| 00:28.50 | jasonwoot | the "scotty" methodology |
| 00:29.14 | jasonwoot | ever see the SNL where they turn the enterprise into a rotating restaurant? |
| 00:29.38 | C4colo | anyway, I'm trying to figure out compatibility on a cordless headset with a polycom 550 ... and polycom support says it will work but their documentation says it needs a lifter |
| 00:29.48 | C4colo | anyone out there using polycoms with wireless headsets? |
| 00:29.53 | jasonwoot | a lifter AND an adapter |
| 00:29.57 | C4colo | er, plantronics support says it needs a lifetr |
| 00:30.06 | C4colo | ok, let me start over |
| 00:30.12 | jasonwoot | plantronics will give you the adapters for free though |
| 00:30.17 | jasonwoot | I squeezed about 50 out of them |
| 00:30.22 | russellb | emails the IRC log to jasonwoot's boss ... |
| 00:30.35 | C4colo | I found a wireless headset I want to use with a poly 550, plantronics supports says it will work but plantronics documentation says it need a lifter |
| 00:30.39 | jasonwoot | slaps russellb around a bit with a large trout |
| 00:30.55 | C4colo | my customer doesn't want a lifter since the 550 supports EHS |
| 00:31.49 | C4colo | it doesn't help to RTFM when TFM is crap |
| 00:32.02 | C4colo | "This is the best headset ever, it has lots of features" |
| 00:32.02 | jasonwoot | yeah, lifters are a pain to get positioned right too, |
| 00:32.26 | C4colo | worthless documentation is worse than no documentation |
| 00:32.37 | C4colo | at least then you know what you are getting into |
| 00:33.06 | *** join/#asterisk nenolod (n=nenolod@atheme/member/nenolod) |
| 00:33.36 | nenolod | hi, does anyone here know about asterlink? mostly, i am wondering how to set it up where it routes into an asterisk setup via SIP :p |
| 00:33.47 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5d63dc8e25457e1f) |
| 00:34.06 | jasonwoot | The free adapters polycom willl give you are actually pretty handy WITHOUT the lifter too though, because the handset falls off the 5XX so easy |
| 00:35.04 | jasonwoot | cya |
| 00:35.10 | C4colo | bbl |
| 00:38.59 | Robba | ok i have narrowed down my problem, it seems the issue is passing DTMF tones to a ZAP channel |
| 00:39.56 | jeffspeff | ok everybody... got another stupid question for you... try not to laugh so hard this time... :) From The Book, setting up the service provider for voip doesn't specify which file those settings go into. Am i to assume that they go into extensions.conf? thanks. |
| 00:40.40 | whymarkwh | no iax or sip to link depends on service provider: jeffspeff |
| 00:41.12 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 00:41.15 | jeffspeff | ok, so since i'm using sip, i put the provider info in sip.conf ? |
| 00:41.23 | whymarkwh | yes |
| 00:41.28 | jeffspeff | ok, thanks. :) |
| 00:41.57 | whymarkwh | no actualy depends on service providers settings |
| 00:42.11 | nenolod | does anyone here know anything about asterlink? |
| 00:42.35 | whymarkwh | if they use the sip protocall the yes sip.conf |
| 00:42.59 | jeffspeff | oh, ok... well they're using sip too. |
| 00:43.01 | jeffspeff | thanks again. |
| 00:47.57 | *** part/#asterisk lirakis (n=user@cpe-68-175-38-65.nyc.res.rr.com) |
| 00:55.12 | *** join/#asterisk edge (n=edge@h75-100-218-214.lncswi.dsl.dynamic.tds.net) |
| 00:56.12 | edge | is there a nice how to , to have my asterisk box, call my friends asterisk box? , i want to be able to place calls to him over the net |
| 00:57.38 | whymarkwh | check linking asterisk with iax or sip in asterisk guru have some nice samples i got it to work fast |
| 00:58.46 | drmessano | just set up an ext to dial sip/yourfriendsext@hisip |
| 00:58.57 | drmessano | and have him do the same |
| 00:59.06 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 00:59.13 | drmessano | You're not linking anything |
| 00:59.20 | drmessano | Just making a simple SIP call |
| 00:59.32 | edge | so i sip call an ip? |
| 00:59.35 | edge | instead of a extension? |
| 00:59.39 | drmessano | ext@ip |
| 00:59.53 | drmessano | create an ext to dial sip/ext@ip |
| 00:59.58 | edge | does the extension then "exist" in my dial plan AND is his? |
| 01:00.06 | drmessano | No |
| 01:00.08 | edge | or does it exist in mine but not his? |
| 01:00.11 | drmessano | You are calling HIS EXTENSION |
| 01:00.14 | [TK]D-Fender | edge, Go lookup "asterisk dual servers" on the WIKI. |
| 01:00.16 | [TK]D-Fender | ~wikis |
| 01:00.16 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 01:00.20 | edge | excelent |
| 01:01.00 | edge | so calling his extension , goes to his ip address then runs his dial plan? (my dial plan chooses to go to sip/ext@ip though) |
| 01:04.02 | drmessano | The extension you dial on your box is irrelevant |
| 01:04.14 | drmessano | You can make it anything |
| 01:04.41 | drmessano | But it has to call SIP/hisextension@hisbox |
| 01:05.02 | edge | ah i get it |
| 01:05.15 | edge | feel like that guy who could of got a v8 , hit on the head |
| 01:05.21 | drmessano | lol |
| 01:05.25 | edge | i could dial 1111 and it will dial sip/his number@ his ip |
| 01:05.41 | drmessano | throws a V8 at ya |
| 01:05.43 | drmessano | Yep |
| 01:06.08 | drmessano | I have a whole block of numbers for dialing friends and whatever |
| 01:06.43 | drmessano | They're all MY number scheme... but dial an array of extensions |
| 01:06.50 | edge | im just learning this , so to ramp up the learning im forceing myself to do more and more complex things, like today its dialing over the net to a friend |
| 01:07.07 | drmessano | I have no friends |
| 01:07.21 | drmessano | So I have never known the joy of making a SIP call to one |
| 01:07.25 | drmessano | Maybe one day.. :/ |
| 01:10.43 | whymarkwh | TK do you ever sleep |
| 01:11.41 | whymarkwh | ill phone you if its tollfree drmessano |
| 01:12.21 | drmessano | Uh |
| 01:12.35 | drmessano | Do you even understand how SIP works? |
| 01:12.51 | whymarkwh | no i bet you going to tell me |
| 01:12.54 | drmessano | If your SIP calls are toll, you got problems |
| 01:13.23 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:14.01 | drmessano | Are you that markwho guy from the Trixbox forums? |
| 01:14.28 | whymarkwh | no i dont do trixbox |
| 01:14.31 | whymarkwh | why? |
| 01:14.42 | drmessano | puts his knife away |
| 01:14.44 | drmessano | No reason |
| 01:14.46 | drmessano | You're safe |
| 01:15.00 | whymarkwh | lol |
| 01:15.32 | whymarkwh | while i got your atention lets talk fax |
| 01:15.59 | drmessano | Lets not |
| 01:16.03 | drmessano | It doesnt work |
| 01:16.05 | drmessano | So move on |
| 01:16.10 | whymarkwh | works |
| 01:16.46 | whymarkwh | as incoming fax server on pstn lines no luck on internal asterisk stucture |
| 01:17.04 | drmessano | As in, over SIP? |
| 01:17.15 | whymarkwh | yip |
| 01:17.24 | drmessano | Fax over SIP doesn't work |
| 01:17.32 | drmessano | Refer to rule #1a |
| 01:17.43 | whymarkwh | like i said true |
| 01:18.05 | rob0 | Just the fax, ma'am. |
| 01:18.12 | drmessano | Lemme guess |
| 01:18.18 | drmessano | You wanna make it work.. like "Lets fix it" |
| 01:18.32 | drmessano | Cool, let's write some code |
| 01:18.45 | drmessano | I'll bring the wizard and the donkey poo |
| 01:18.52 | drmessano | :) |
| 01:19.04 | whymarkwh | 1000000000011100000000001111100000000000001111000000001111000110 this should do it |
| 01:19.17 | rob0 | that 0 should be 1 |
| 01:19.17 | whymarkwh | let put it to the test |
| 01:19.22 | whymarkwh | lololol |
| 01:19.34 | edge | what port/s do i need to foward on my router for a iax connection of two asterisk boxes? |
| 01:19.54 | drmessano | edge, you're gonna peer them? |
| 01:20.18 | whymarkwh | 4569 |
| 01:20.37 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 01:20.50 | edge | drmessano, i think thats what im doing via the config |
| 01:20.55 | edge | whymarkwh, that tcp or udp? |
| 01:21.05 | whymarkwh | udp |
| 01:21.16 | drmessano | Why not just simple calls back and forth over SIP? |
| 01:21.28 | whymarkwh | iax is better |
| 01:21.34 | whymarkwh | i think |
| 01:21.37 | drmessano | IAX is a waste for one call |
| 01:21.40 | whymarkwh | my oppinion |
| 01:21.44 | [TK]D-Fender | No, IAX is different. |
| 01:22.21 | whymarkwh | dont you sleep TK how do you keep it up? |
| 01:22.29 | *** join/#asterisk BeeBuu (n=beebuu@218.13.66.139) |
| 01:22.32 | whymarkwh | been here the whole day |
| 01:22.54 | drmessano | He's waiting around for you to notice the y got clipped off the end of your nickname |
| 01:23.34 | [TK]D-Fender | whymarkwh, viagra |
| 01:23.40 | whymarkwh | lol |
| 01:24.17 | gitguy | who needs viagra? |
| 01:24.44 | whymarkwh | ohhhhhhhhhhhh where o where has my y gone ohhhhhhhh where ohhhhhhh where can it be |
| 01:25.01 | whymarkwh | took me a while to compile that one |
| 01:25.06 | drmessano | gitguy, if you're offering it.. Dont bother.. I get the emails |
| 01:25.21 | gitguy | ok |
| 01:25.39 | drmessano | LQQK V1AGRA N0W |
| 01:25.40 | gitguy | i'm not a drugs dealer |
| 01:25.44 | whymarkwh | i thought i was the only unlucky asssssssss getting those discounted farmacy goods |
| 01:25.57 | drmessano | Youre not a drugs dealer, just a drug dealer.. singular |
| 01:26.01 | drmessano | I follow |
| 01:26.09 | gitguy | i know the best in english |
| 01:26.14 | gitguy | i speak spanish baby |
| 01:26.28 | whymarkwh | spanish fly rules |
| 01:26.43 | whymarkwh | jetem |
| 01:26.46 | gitguy | i'm not the best in english* |
| 01:26.58 | gitguy | damn my english gets worse every day |
| 01:27.03 | gitguy | :( |
| 01:27.33 | whymarkwh | do you know gitguy if you change the first g in your nick to a z need i say more |
| 01:28.54 | drmessano | My mother always told me "Dont hate someone because of their name.. if they have a goofy name, chances are they'll give you other reasons to hate them" |
| 01:28.56 | gitguy | >_> |
| 01:28.58 | drmessano | I miss mom.. RIP |
| 01:29.06 | edge | whats the best way to watch asterisk, i did the asterisk -r , and its a cli , but i wanna see what asterisk is doing when i dial, when i try to dial (bc im trying its not succeding) |
| 01:29.14 | drmessano | asterisk -vvvvvvvvr |
| 01:29.28 | edge | 8v?s |
| 01:29.30 | gitguy | drmessano: my condolences |
| 01:29.54 | edge | thanks , verbosity is now 1! |
| 01:29.59 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:30.00 | whymarkwh | increase verbose with the -vvvvvvvvvvvv read the manual |
| 01:30.01 | drmessano | Thanks gitguy.. It'll be 3 years next week since she was trampled by that elephant |
| 01:30.15 | gitguy | what elephant? |
| 01:30.27 | drmessano | I don't know it's name |
| 01:30.37 | edge | dumbo? |
| 01:30.40 | whymarkwh | lol |
| 01:30.54 | drmessano | Although I believe "What elephant?" were her last words |
| 01:30.57 | drmessano | :```( |
| 01:31.01 | gitguy | drmessano: you serious? |
| 01:31.05 | whymarkwh | glad to see i'm not the only one fighting to stay awake |
| 01:31.27 | drmessano | gitguy: No |
| 01:31.47 | drmessano | I'm seriously trying to get this red5 plugin working on my openfire box |
| 01:31.57 | gitguy | what is openfire |
| 01:31.59 | gitguy | what is red5 |
| 01:32.20 | whymarkwh | stop waisting your time |
| 01:32.26 | whymarkwh | my tiping rules |
| 01:32.28 | gitguy | i like wasting my time |
| 01:32.42 | drmessano | Openfire is an XMPP server.. red5 is a suite of java video/audio apps that allow you to make calls over asterisk |
| 01:32.43 | whymarkwh | that what we are doing now |
| 01:32.54 | drmessano | Well, in this case |
| 01:32.57 | drmessano | red5 is more of a platform |
| 01:33.09 | gitguy | heh |
| 01:33.22 | drmessano | But they have some cool web based audio/video stuff for Openfire + Asterisk |
| 01:33.54 | drmessano | I'm either going to get it working, or stab myself in my left eye |
| 01:34.20 | gitguy | lol |
| 01:34.46 | drmessano | I'm not even joking |
| 01:34.52 | gitguy | i want to call my ex-gf and put tt-monkeys on her cellphone |
| 01:35.19 | drmessano | I spent four hours getting a website built last night.. I go to sign into the web messenger and see if it's working with asterisk, and it had stopped working somehow |
| 01:35.23 | drmessano | I am stabfully annoyed |
| 01:35.44 | gitguy | :( |
| 01:35.59 | whymarkwh | i see you can find alot of help on their forum regarding open-xchange like 8 posts and counting |
| 01:36.02 | gitguy | you building some webapp with asterisk? |
| 01:36.24 | drmessano | More of an integration.. I dont build.. I glue stuff together |
| 01:36.36 | drmessano | If it doesnt quite glue, then I may build a little |
| 01:36.45 | JayTee52 | ~pastebin |
| 01:36.45 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 01:37.17 | whymarkwh | need some more code? |
| 01:37.27 | whymarkwh | 011111111001100000000000001110000000011 this ones for free |
| 01:38.48 | JayTee52 | could someone look at this and confirm for me that if I dialed from outside any of the numbers listed in the context "directory" that they would match before the statements that follow the include in the "Incoming" context? |
| 01:38.51 | JayTee52 | http://rafb.net/p/95B3mr81.html |
| 01:39.08 | gitguy | damn i need to learn to speak english well |
| 01:39.18 | gitguy | verbally |
| 01:39.20 | gitguy | my english does suck |
| 01:39.23 | drmessano | <html><title>SWEEP THE LEG!</title><body><br><br><br><br><center><b>SWEEP THE LEG!</b></center></body></html> |
| 01:39.33 | JayTee52 | hehehe |
| 01:39.36 | drmessano | Thats a complete website.. $120 worth |
| 01:40.03 | [TK]D-Fender | drmessano, don't bother with that crane crap kid..... |
| 01:40.25 | drmessano | SEND HIM HOME IN BODY BAG... HA HA HA YEAH JOHNNY |
| 01:40.41 | drmessano | Best. Movie. Ending. Ever. |
| 01:41.10 | JayTee52 | there was a YouTube video of "Johnny" as an older has-been living in a trailer with all his buddies and still having nightmares about the contest. |
| 01:41.20 | drmessano | I saw that! |
| 01:41.23 | drmessano | That rocked! |
| 01:41.24 | drmessano | lol |
| 01:41.45 | whymarkwh | me haces reir :gitguy |
| 01:41.54 | gitguy | porque |
| 01:42.02 | gitguy | why |
| 01:42.27 | gitguy | carajo |
| 01:42.57 | drmessano | 25 years later, kids are like "Why didnt Daniel just pretend to be some hot girl on myspace, make johnny want her, then let him know she was fake so he'll go all emo and kill himself?" |
| 01:42.59 | adeel | i have an * box behind 2 NAT's that i need to connect to another * box with a public ip....i see the packets just fine, and have defined the localnet & externip settings, but on the remote box, the SIP headers aren't using the externip address, rather it's the external ip of the first NAT |
| 01:43.02 | JayTee52 | so to be more precise about my earlier posted question, my telco is passing 7 digit numbers and I'm trying to "filter" them so if they are in the directory context they'll be passed to their corresponding SIP phone otherwise they will be routed over my Zap/g3 trunk. Is the example I pasted accurate? |
| 01:43.13 | whymarkwh | your english is bueno |
| 01:43.32 | gitguy | mi ingles verbalmente apesta |
| 01:43.51 | drmessano | mi esposa es ben tramplado por toros |
| 01:43.55 | whymarkwh | culo |
| 01:44.08 | gitguy | soy el mejor |
| 01:45.02 | whymarkwh | getting better by the day |
| 01:45.04 | gitguy | jajajajajajaa |
| 01:45.14 | za3toor | hey guys... can someone help me with redhat and astrisk |
| 01:45.21 | za3toor | i can't seem to get a dhcp ip address |
| 01:45.25 | za3toor | or any ip address at all |
| 01:45.35 | za3toor | i have tried everything i can think off for the past 5 days |
| 01:45.37 | za3toor | still no luck |
| 01:46.28 | JayTee52 | a dhcp address on an * server? why? |
| 01:47.14 | za3toor | or i can't even log in too the router |
| 01:47.16 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-116-105-45.rgv.res.rr.com) |
| 01:47.19 | za3toor | or in to the network at all |
| 01:47.22 | za3toor | i cna't ping anything |
| 01:47.30 | za3toor | except ping localhost gives a return |
| 01:47.41 | JayTee52 | check your network cable |
| 01:47.56 | za3toor | it works great with vista on dual boot |
| 01:47.58 | whymarkwh | tu deber como por diablo |
| 01:48.04 | za3toor | and on ubutun |
| 01:48.14 | JayTee52 | is this on a home network? |
| 01:48.16 | colinm_ | then why did you install redhat? |
| 01:48.28 | za3toor | yes it is a home network |
| 01:48.36 | za3toor | why would not i install redhat? |
| 01:48.38 | JayTee52 | what is your DHCP server? did you enter that address in Network Manager in RedHat? |
| 01:48.53 | za3toor | yes |
| 01:48.59 | za3toor | i tihnk the problem is with irqpoll |
| 01:49.01 | za3toor | that's what i have read |
| 01:49.05 | *** part/#asterisk seaq (n=seaq@190.25.101.69) |
| 01:49.35 | JayTee52 | have you tried hardcoding a static address and mask that matches what the computer pulls when it's running Vista? |
| 01:49.42 | za3toor | yes |
| 01:49.45 | za3toor | no luck either |
| 01:50.03 | JayTee52 | when you did that did you reboot? |
| 01:50.11 | za3toor | yes |
| 01:50.18 | za3toor | and it says eth0 is OK |
| 01:50.25 | za3toor | but when i ping anything in the network |
| 01:50.30 | za3toor | i get unreachable |
| 01:50.34 | TrentCreek | did you install the vista loopback adapter? |
| 01:50.38 | whymarkwh | please help za3toor he been trying forever to get this to work im outa here cheers all. |
| 01:50.46 | JayTee52 | have you tried disabling IPTABLES? |
| 01:50.48 | *** join/#asterisk |loosemo| (n=loosemoo@cpe-71-65-34-198.indy.res.rr.com) |
| 01:50.50 | za3toor | cheas |
| 01:50.56 | za3toor | ill eamail you when it is working |
| 01:50.57 | za3toor | latter |
| 01:51.04 | za3toor | vista loopback adapteR?? |
| 01:51.05 | za3toor | what is that? |
| 01:51.06 | whymarkwh | k do that |
| 01:51.14 | za3toor | for sure |
| 01:51.37 | TrentCreek | It's some software you have to install to get networking to function on virtual computers |
| 01:51.48 | za3toor | ohh i c |
| 01:51.53 | *** part/#asterisk whymarkwh (i=dsfsdfsd@196.211.34.3) |
| 01:51.54 | TrentCreek | It will showup in the same place as network connections |
| 01:51.55 | za3toor | well it is not a virtual setup |
| 01:52.00 | za3toor | i have it as dual boot |
| 01:52.21 | TrentCreek | oh..then it's not a vista problem |
| 01:52.24 | za3toor | and i can only access the terminal at the moment |
| 01:52.30 | za3toor | no it is not a vista problem |
| 01:52.58 | TrentCreek | was it working before? |
| 01:53.02 | za3toor | no |
| 01:53.12 | za3toor | i just started working on linux 5 days ago |
| 01:53.18 | za3toor | and i started having that problem 5 days ago |
| 01:53.21 | za3toor | lol |
| 01:53.26 | gitguy | linux lo mejor del mundo |
| 01:53.48 | JayTee52 | is it a valid copy of RedHat with a valid license? |
| 01:53.54 | TrentCreek | wired or wireless? |
| 01:54.01 | za3toor | wired |
| 01:54.05 | za3toor | yes it is a valid copy |
| 01:54.24 | TrentCreek | then I think could be a driver issue |
| 01:54.40 | JayTee52 | he can ping his loopback address though |
| 01:54.55 | za3toor | yes |
| 01:54.58 | za3toor | nothing else |
| 01:55.11 | za3toor | when i type route i get nothing |
| 01:55.12 | za3toor | Kernel IP routing table |
| 01:55.12 | za3toor | Destination Gateway Genmask Flags Metric Ref Use Iface |
| 01:55.12 | za3toor | 192.168.2.0 * 255.255.255.0 U 100 0 0 eth0 |
| 01:55.12 | za3toor | default 192.168.2.1 0.0.0.0 UG 0 0 0 eth0 |
| 01:55.17 | za3toor | i want route to be something liek this |
| 01:55.22 | TrentCreek | loopback is software, not hardware |
| 01:55.39 | TrentCreek | dynamic or static IP? |
| 01:56.06 | za3toor | right now i am on dynamic |
| 01:56.12 | gitguy | http://youtube.com/watch?v=3ifG_zQ1SqE |
| 01:56.15 | gitguy | nirvana rules |
| 01:57.16 | *** join/#asterisk loosemoose (n=loosemoo@cpe-71-65-34-198.indy.res.rr.com) |
| 01:57.41 | za3toor | anyideas? |
| 01:58.13 | rob0 | Best advice at this point, decide on a single distro, get that working. And the distro channel would be a better place to go for help. |
| 01:58.34 | rob0 | if Ubuntu uboots, why not uuse it? |
| 01:58.47 | *** part/#asterisk aut (n=aut@modemcable086.237-80-70.mc.videotron.ca) |
| 01:58.48 | drmessano | uuh uok |
| 01:58.55 | TrentCreek | are you given a valid IP address from the DHCP server? |
| 01:59.32 | za3toor | nop |
| 01:59.55 | TrentCreek | try setting a static one. |
| 02:00.09 | za3toor | i did |
| 02:00.11 | za3toor | no luck |
| 02:00.54 | TrentCreek | have you verified the gateway address? |
| 02:01.00 | za3toor | yes |
| 02:01.39 | TrentCreek | what brand/ model of network card? |
| 02:01.56 | za3toor | it is on board |
| 02:02.16 | TrentCreek | yes, but I am sure it has a supporting chipset it uses |
| 02:02.39 | TrentCreek | I am suspecting it is not supported |
| 02:02.56 | za3toor | itis a Realtalk RTL8168C/8111C family PCIE Gigabite Ethernet NIC |
| 02:03.01 | JT | eww |
| 02:03.32 | drmessano | Couldnt you find a nice 10base2 adapter instead? |
| 02:03.40 | za3toor | lol |
| 02:03.41 | JayTee52 | lol |
| 02:03.45 | za3toor | it came with the motherboard |
| 02:03.52 | za3toor | so i could not say no |
| 02:03.59 | JayTee52 | ooooh, thinnet! |
| 02:04.04 | drmessano | I have some Ts and terminators |
| 02:04.10 | JayTee52 | 50 ohm? |
| 02:04.12 | JayTee52 | cool! |
| 02:04.19 | rob0 | 02:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 01) |
| 02:04.31 | rob0 | oh that's the B |
| 02:04.47 | za3toor | so is it a bad card |
| 02:04.49 | za3toor | a good card |
| 02:04.51 | za3toor | what? |
| 02:05.00 | TrentCreek | I am thinking not supported |
| 02:05.05 | drmessano | If it was a C, you would be homefree |
| 02:05.15 | TrentCreek | I tried search for that type on Redhat.com |
| 02:05.15 | drmessano | But since its a B, you're toast |
| 02:05.20 | rob0 | his is a C, mine is the B, mine works |
| 02:05.24 | drmessano | No, not really |
| 02:05.32 | JT | all realteks are bad cards to some extent |
| 02:05.45 | za3toor | really |
| 02:05.46 | za3toor | wow... |
| 02:05.47 | Juggie | that card is definitally supported |
| 02:05.51 | drmessano | Realtek's aren't bad.. just less complicated |
| 02:05.52 | outtolunc | still has a full roll of rg58 in the closet |
| 02:06.07 | drmessano | lol |
| 02:06.26 | JT | the best network cards are all made by a company called Intel |
| 02:06.26 | za3toor | where do i know which card is supported |
| 02:06.29 | drmessano | rg58 isn't that ,uch of a stretch |
| 02:06.30 | za3toor | lol |
| 02:06.32 | JayTee52 | RedHat's network manage is kinda twitchy. I had to mess with it a bit before I got a static IP to take. |
| 02:06.43 | Juggie | za3toor, what version of redhat are you trying to install |
| 02:06.54 | JT | intel ethernet chipsets are the best, broadcom is the next best |
| 02:06.54 | za3toor | 4.0.15 |
| 02:06.58 | drmessano | If you had some RG-62, maybe |
| 02:07.11 | TrentCreek | You may just want to run down and get a used card such as Intel or 3COM from a local computer store |
| 02:07.23 | za3toor | yah |
| 02:07.31 | za3toor | and you are sure that the problem is from my network card |
| 02:07.35 | TrentCreek | those are the most supported in Linux |
| 02:07.38 | Juggie | za3toor, thats too old |
| 02:07.52 | drmessano | Realtek cards are fine |
| 02:07.54 | drmessano | ZOMG |
| 02:08.04 | drmessano | Redhat 4? |
| 02:08.05 | JT | i've picked up intel eepro1000 cards for $15 second hand |
| 02:08.06 | JayTee52 | when you configure your settings in the Network manager did you deactivate first and then reactivate? |
| 02:08.07 | za3toor | is there a website that i would ,now |
| 02:08.10 | JT | err |
| 02:08.14 | JT | REDHAT 4?! |
| 02:08.26 | JayTee52 | oh, yeah 4 |
| 02:08.28 | JT | welcome to 1997? |
| 02:08.30 | drmessano | Dude.... |
| 02:08.36 | Juggie | he must mean RHEL4 |
| 02:08.41 | JT | i hope so |
| 02:08.42 | drmessano | I don't think Redhat 4 supported motherboards |
| 02:08.43 | Juggie | no one would still have redhat4 |
| 02:08.44 | JayTee52 | is it RHEL 4 or Red Hat 4 |
| 02:08.52 | JayTee52 | drmessano, hahahahaa |
| 02:09.01 | za3toor | no red hat 4 |
| 02:09.02 | draygon-w | lol |
| 02:09.06 | draygon-w | that was actually funny |
| 02:09.27 | JT | za3toor: that's like 10 years old |
| 02:09.40 | JayTee52 | za3toor, so it's Red Hat 4 and not Red Hat Enterprise Linux 4? |
| 02:09.45 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 02:09.46 | drmessano | Is that on floppy? |
| 02:09.51 | za3toor | lol |
| 02:09.53 | za3toor | let me check |
| 02:09.54 | za3toor | loooool |
| 02:09.55 | za3toor | 1 sec |
| 02:09.56 | TrentCreek | And I still have my distro of Caldera running |
| 02:10.00 | JayTee52 | mine came on CD |
| 02:10.02 | drmessano | 45 Floppies |
| 02:10.07 | JayTee52 | with a boot floppy |
| 02:10.15 | Juggie | theres no such thing as redhat 4.0.15 |
| 02:10.20 | Juggie | or RHEL4.0.15 for that matter |
| 02:10.21 | drmessano | I have a dummies book with Redhat 5 |
| 02:10.22 | JayTee52 | but that was back when Clinton was still in office. |
| 02:10.32 | Juggie | so what exactally are you running |
| 02:10.37 | za3toor | red hat Nash Version 4.2.15 |
| 02:10.40 | TrentCreek | Dagnabit you young whipper snappers |
| 02:10.47 | outtolunc | cat /etc/redhat-release |
| 02:10.49 | drmessano | Oh lord |
| 02:10.53 | za3toor | lol |
| 02:10.53 | za3toor | ok |
| 02:11.04 | JayTee52 | I'm running RHEL 5 64bit and * runs awesome on it. Quad Xeon with 4GB of RAM. |
| 02:11.37 | drmessano | I think RH Nash is FC4 |
| 02:11.51 | JT | if something doesn't run awesome on a quad xeon with 4GB of ram, it must really suck |
| 02:12.04 | JayTee52 | JT, agreed |
| 02:12.21 | drmessano | Xp doesnt |
| 02:12.22 | TrentCreek | yeah..it's called INTEL |
| 02:12.23 | drmessano | Err |
| 02:12.25 | drmessano | Vista doesnt |
| 02:12.35 | JayTee52 | although we're talking linux. now if I loaded Exchange 2007 on it that's another story |
| 02:12.54 | TrentCreek | My AMD 64 3700 runs almost as fast as Intel Quad 4 |
| 02:12.55 | Juggie | i'm concerned about helping someone who doesnt know what distro he is running |
| 02:12.59 | Juggie | i feel like i might be wasting my time |
| 02:13.15 | *** join/#asterisk gitguy (n=diego@adsl-155-36.click.com.py) |
| 02:13.17 | drmessano | Juggie: You wont get your weekend back |
| 02:13.18 | JayTee52 | I kinda got that impression myself |
| 02:13.18 | za3toor | so i should get a network card by intel |
| 02:13.20 | za3toor | right |
| 02:13.27 | drmessano | za3toor |
| 02:13.30 | drmessano | I have a question |
| 02:13.31 | TrentCreek | yes, or 3COM |
| 02:13.33 | Juggie | za3toor, realtek is fine. |
| 02:13.40 | drmessano | Do your CD |
| 02:13.40 | outtolunc | here it comes |
| 02:13.40 | za3toor | sure... |
| 02:13.45 | Juggie | but you are not answering the questions we are asking |
| 02:13.46 | drmessano | Do your CD's say FC4 in the name? |
| 02:13.49 | drmessano | The ISOs |
| 02:13.51 | Juggie | WHAT FRIGGEN DISTRO ARE YOU RUNNING |
| 02:13.53 | JayTee52 | Is Madge still making 4MB token ring adapters? |
| 02:14.01 | za3toor | ok... well here is the story |
| 02:14.06 | drmessano | Do the ISO names have FC4 |
| 02:14.06 | JayTee52 | succinct, to the point! |
| 02:14.11 | drmessano | You stole the box |
| 02:14.16 | za3toor | lol |
| 02:14.19 | za3toor | you guys are funny |
| 02:14.24 | drmessano | You rooted someone |
| 02:14.25 | za3toor | i am trying to install *riskNOW |
| 02:14.28 | za3toor | that's oit |
| 02:14.29 | outtolunc | you found it on the side of the road? <G> |
| 02:14.30 | drmessano | Oh |
| 02:14.32 | drmessano | In that case |
| 02:15.05 | drmessano | GTFO |
| 02:15.05 | JayTee52 | hahahahaa |
| 02:15.05 | za3toor | loooooooooooooooooooooooooooooooooooooooooooooooooooooooool |
| 02:15.05 | JT | err |
| 02:15.05 | JT | AsteriskNow uses rpath linux |
| 02:15.05 | za3toor | i knew i was going to get htat |
| 02:15.05 | za3toor | ohhhhhhhhhhhhhhhhhhhhhh |
| 02:15.05 | drmessano | Sorry |
| 02:15.05 | JT | or do you mean asterisk-gui? |
| 02:15.05 | drmessano | gtfoNOW |
| 02:15.10 | za3toor | LOL |
| 02:15.19 | JT | asterisknow has asterisk gui |
| 02:15.37 | JT | but if you're installing asterisknow, you should be wiping over what was already there |
| 02:15.55 | drmessano | gtfoNOW includes stfuNOW 1.3 |
| 02:15.56 | za3toor | yah |
| 02:16.03 | za3toor | lol |
| 02:16.29 | drmessano | AsteriskNOW is vaporware, aside from it being unsupported in here |
| 02:16.45 | ectospasm | vaporware? |
| 02:16.47 | JayTee52 | ok, I'm gonna ask again. If I have an include statement as the FIRST line of a context will * try to match on what's in that context before trying the remaining lines in the current context? |
| 02:16.48 | ectospasm | I don't think so |
| 02:17.00 | drmessano | It's not? |
| 02:17.00 | mwalling | he should install slackware |
| 02:17.09 | jjshoe | asterisknow couldn't switch modes on a common monitor we had in the office |
| 02:17.14 | jjshoe | pretty f'in hilarious |
| 02:17.25 | Juggie | mwalling, centos is the best server newb distro |
| 02:17.34 | TrentCreek | Something else replaced Asterisk NOW |
| 02:17.41 | mwalling | Juggie: far from my point |
| 02:17.49 | jjshoe | TrentCreek its called switchvox |
| 02:17.55 | TrentCreek | groovy |
| 02:17.56 | ectospasm | JayTee52: the include gets slurped into the dialplan in memory at that point, anything that happens beforehand gets loaded first |
| 02:18.06 | ectospasm | ...I think... |
| 02:18.20 | za3toor | ok guys.. what is the best Linux OS and EASISET to use |
| 02:18.26 | drmessano | Debian |
| 02:18.28 | jjshoe | za3toor slackware. |
| 02:18.29 | drmessano | No |
| 02:18.29 | mwalling | windows |
| 02:18.30 | drmessano | Ubuntu |
| 02:18.32 | drmessano | No |
| 02:18.34 | drmessano | Slackware |
| 02:18.38 | drmessano | Wait, CentOS |
| 02:18.39 | jjshoe | gentoo |
| 02:18.40 | TrentCreek | LINDOWS |
| 02:18.46 | ectospasm | DSL |
| 02:18.47 | Juggie | Centos is the best server because its based on RHEL |
| 02:18.47 | drmessano | DRAGONLINUX |
| 02:18.51 | jjshoe | I prefer hp-ux! |
| 02:18.56 | drmessano | OpenSolaris |
| 02:18.57 | mwalling | AIX |
| 02:19.03 | JayTee52 | Yggdrasil |
| 02:19.04 | *** join/#asterisk steliosk (n=Stelios@athedsl-310554.home.otenet.gr) |
| 02:19.05 | ectospasm | Xenix |
| 02:19.09 | drmessano | Windux |
| 02:19.13 | jjshoe | os/2 warp! |
| 02:19.22 | drmessano | Coyote |
| 02:19.24 | outtolunc | concurrent dos 3.0 <G> |
| 02:19.32 | outtolunc | cpm |
| 02:19.33 | TrentCreek | I tried putting Debian on one box, but the crap would not support SCSI card |
| 02:19.43 | drmessano | The best version of Linux ever is Windows For Workgroups 3.111 |
| 02:20.02 | mwalling | TrentCreek: operator error? |
| 02:20.14 | TrentCreek | just would not see the card |
| 02:20.46 | drmessano | I can't get my distro to recognize my CD-ROM... quite a dilemna.. I'm all like.. "YOU'RE IN IT!" |
| 02:20.47 | mwalling | debian can run the hurd kernel, but cant see your card? |
| 02:20.50 | TrentCreek | Adaptech |
| 02:20.54 | mwalling | (speaking of vaporware) |
| 02:21.17 | TrentCreek | correct...reports I have no drives to make partions on |
| 02:21.23 | drmessano | Damn El Torito |
| 02:23.18 | drmessano | A lot of folks favor CentOS for asterisk.. but say that out loud and the OS wars start |
| 02:23.25 | drmessano | Mostly its "what you feel comfortable with" |
| 02:23.33 | TrentCreek | I would have put Windows server on it, but stupid Bill can't make an OS that supports booting from SCSI drives |
| 02:23.55 | drmessano | I tried to get Asterisk running on DSL, but I kept getting write errors |
| 02:24.25 | mwalling | slackware <3 |
| 02:24.35 | TrentCreek | check the YUM,,,,could be available for instal |
| 02:25.02 | drmessano | Sometimes I think some people intentionally pick far out distros so they can be elite |
| 02:25.13 | drmessano | "I'm running Asterisk on OpenDarwin" |
| 02:25.20 | TrentCreek | I wracked my brain for days installing/compiling Asterisk on my Fedora box..then 6 months later I see it is available to install...doh! |
| 02:25.21 | mwalling | wait a sec... we're off topic... we're supposed to be trolling that guy |
| 02:25.28 | drmessano | Oh |
| 02:25.31 | drmessano | I forgot |
| 02:25.43 | drmessano | AsteriskNOW <---- HAHAHAHHHAHAH |
| 02:25.48 | mwalling | lol |
| 02:26.20 | drmessano | "What's the matter, the Trixbox repo running slow???" |
| 02:26.33 | drmessano | "Was AsteriskWin32.com down?" |
| 02:26.34 | *** part/#asterisk Hydrant (n=aj@CPE0011950c737b-CM0012c90d1420.cpe.net.cable.rogers.com) |
| 02:26.44 | ectospasm | Installing asterisk ain't difficult |
| 02:26.56 | ectospasm | configuring it can be a drudge if you're not used to it |
| 02:27.02 | drmessano | Check dependencies... Install some crap |
| 02:27.09 | drmessano | Repeat, x50 |
| 02:27.18 | drmessano | Configure |
| 02:27.20 | TrentCreek | it is espically easy on Fedora |
| 02:27.23 | drmessano | ????????? |
| 02:27.27 | drmessano | Callcenter |
| 02:27.29 | TrentCreek | w/ YUM |
| 02:28.06 | ectospasm | that repeat step is maybe like x5, if it's a completely fresh install... |
| 02:28.14 | ectospasm | TrentCreek: but how up to date is that? |
| 02:28.17 | drmessano | I've been toying with making a dummy RPM for installing the dependencies, as someone had suggested |
| 02:28.57 | TrentCreek | try with a virtual server..I am going to find one and put it on my box to try all the * distros |
| 02:29.19 | drmessano | Eww |
| 02:29.37 | drmessano | No way I would run Asterisk on a virtual server.. I would feel dirty |
| 02:29.47 | TrentCreek | super cool |
| 02:29.58 | rob0 | mwalling is dirty |
| 02:30.01 | TrentCreek | dirty and slimy |
| 02:30.11 | drmessano | Not the kind of dirty you wash off with soap.. |
| 02:30.14 | TrentCreek | but you save time & $$ |
| 02:30.24 | drmessano | ~cheap |
| 02:30.25 | jbot | [cheap] a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 02:30.26 | mwalling | rob0: cleaner then you :P |
| 02:30.34 | mwalling | rob0: i shower > once a weel |
| 02:30.35 | TrentCreek | once I find what I want..save and copy to hosted server |
| 02:30.40 | mwalling | s/weel/week/ |
| 02:31.09 | drmessano | Getting in the shower on any day other than wash day is just wasting water |
| 02:31.10 | rob0 | 02:30 < drmessano> Not the kind of dirty you wash off with soap.. |
| 02:31.22 | TrentCreek | Not cheap for setting up then testing to move to a dedicated server |
| 02:32.00 | drmessano | I dont trust anything virtual for apps that require accurate timing |
| 02:32.11 | *** join/#asterisk LiKuiD_SiLeNcE (n=liquid@adsl-074-186-024-195.sip.mia.bellsouth.net) |
| 02:32.26 | drmessano | Be cause I li ke my ca lls t o n ot ha ve pr ob le ms |
| 02:32.43 | TrentCreek | Can you hear me now? |
| 02:32.51 | drmessano | Wh t? |
| 02:32.59 | mwalling | i've been running asterisk in Xen, havent had problems... |
| 02:33.12 | mwalling | then again, i'm also not putting much load on it |
| 02:33.36 | drmessano | Everyone claims it will work in a VM.. but no one thats put REAL use on it |
| 02:33.45 | drmessano | "Oh, my trixbox VMware image works great" |
| 02:33.50 | drmessano | "How many users?" |
| 02:33.55 | drmessano | "Me, mom, and dad" |
| 02:34.01 | mwalling | heh |
| 02:34.04 | adeel | drmessano, i've run * under a decent load inside OpenVZ |
| 02:34.41 | drmessano | What is "decent"? |
| 02:35.01 | adeel | ~50 simultaneous calls |
| 02:35.12 | [TK]D-Fender | The dangers of VoIP : http://video.msn.com/video.aspx?mkt=en-ca&brand=sympatico&tab=m2718&vid=bbe49f2e-73e5-4683-ae6e-09c7c4d91b62&playlist=videoByTag:tag:encanews_encaconsumerreports:ns:MSNVideo_Top_Cat:mk:en-ca:sf:DailyCount:st:1:vs:0&from=msnhpquad |
| 02:35.15 | drmessano | What kind of metal? |
| 02:35.18 | *** join/#asterisk PaulQ (n=PaulQ@72.29.76.254) |
| 02:35.29 | PaulQ | A quick one I hope, Why does '$retry = ${retry} + 1;' not work as expected? |
| 02:35.42 | adeel | this was on a dual core p4....one sec, i'll paste the specs |
| 02:36.09 | PaulQ | Are variables local to their context or their extension? |
| 02:36.10 | [TK]D-Fender | PaulQ, perhaps you should show EXACTLY how you're doing the whole thing... |
| 02:36.12 | adeel | dual Intel(R) Pentium(R) D CPU 2.66GHZ with 1 GB RAM |
| 02:36.23 | adeel | running G.729 passthrough |
| 02:36.39 | PaulQ | PaulQ, That pretty much is exactly all I'm having problems with. Set it to 0 |
| 02:36.47 | PaulQ | I would expect it to increase on each pass |
| 02:36.58 | drmessano | adeel: No issues at all? |
| 02:37.12 | PaulQ | Variables are in the scope of context and not extension right? |
| 02:37.27 | drmessano | Yeah.. glad they mentioned it in this video |
| 02:37.39 | drmessano | Dont just call 911, yell help, and hang up |
| 02:37.48 | PaulQ | I obviously meant to address [TK]D-Fender and not myself. |
| 02:37.49 | russellb | PaulQ: variables (other than global variables) are in the scope of a channel's lifetime |
| 02:37.54 | drmessano | E911 Can be wrong.. forget VoIP |
| 02:38.00 | adeel | drmessano, nothing really major...i was coming pretty close to maxing out my bandwidth, but the box was pretty stable...you just need to know how to setup openvz properly, and using a gentoo install is useful |
| 02:38.21 | PaulQ | thanks russellb. Now I wonder why this isnt increasing on each pass |
| 02:38.30 | russellb | because your syntax is not correct |
| 02:38.34 | PaulQ | I think im using the right syntax |
| 02:38.35 | PaulQ | Hmm |
| 02:38.38 | russellb | as [TK]D-Fender said, show the exact line ... |
| 02:38.42 | likuid_silence | anyone running asterisk on ubuntu |
| 02:38.45 | PaulQ | That is the exact line... |
| 02:38.49 | PaulQ | It's AEL2 |
| 02:39.10 | russellb | that's an important note. :) |
| 02:39.19 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 02:39.22 | PaulQ | Ahh, I guess it does look a bit like Psuedo-code ;) |
| 02:39.29 | russellb | take off the $ at the beginning of the line |
| 02:39.30 | jameswf-home | weeeeeee |
| 02:39.43 | PaulQ | russellb tried that also, let me retry |
| 02:39.50 | [TK]D-Fender | PaulQ, why don't you do "dialplan show [context]" and show us how its REALLY got parsed... |
| 02:39.53 | [TK]D-Fender | ~whee |
| 02:39.54 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 02:39.59 | outtolunc | <PROTECTED> |
| 02:40.00 | outtolunc | <PROTECTED> |
| 02:40.00 | outtolunc | <PROTECTED> |
| 02:40.43 | jameswf-home | heard russellb will spend alot more time here now that he has an accountant... oh wait still the honeymoon.... congrats... |
| 02:41.06 | russellb | thanks :) |
| 02:41.20 | russellb | actually, i have been on here much less the past month or so :) |
| 02:41.24 | PaulQ | <PROTECTED> |
| 02:41.34 | russellb | that should work. |
| 02:41.35 | jameswf-home | like i said still the honeymoon... |
| 02:41.41 | russellb | heh |
| 02:42.09 | PaulQ | [TK]D-Fender sorry about that. Didnt notice it actually got parsed into a normal DialPlan command :-) |
| 02:42.16 | [TK]D-Fender | russellb, NOW is the part where I mention that I have NO trust that ${retry} ..... isn't BLANK :) |
| 02:42.27 | PaulQ | TK, It's set before |
| 02:42.29 | jameswf-home | I told my boss we need to get our 1 programmer married so he can work saturdays |
| 02:42.29 | russellb | [TK]D-Fender: touche. |
| 02:42.30 | PaulQ | to 0 |
| 02:42.32 | drmessano | russellb: Dont worry, now that youre hitched, you have no reason to spend less time online... you can stop sucking up :) |
| 02:42.36 | [TK]D-Fender | PaulQ, Do I see that? Do I trust it? |
| 02:42.53 | [TK]D-Fender | PaulQ, "show me the money". Pastebin is your friend. |
| 02:42.54 | [TK]D-Fender | ~pb |
| 02:42.54 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:42.55 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 02:43.00 | russellb | drmessano: nice :) |
| 02:43.00 | drmessano | russellb: Now go mow the lawn |
| 02:43.06 | russellb | harsh |
| 02:43.07 | PaulQ | [TK]D-Fender, You've been in IRC too long and assume the worst? |
| 02:43.09 | PaulQ | Good man.. |
| 02:43.20 | PaulQ | <PROTECTED> |
| 02:43.21 | drmessano | russellb: Oh, and the gutters need cleaning |
| 02:43.28 | drmessano | russellb: EEK SPIDER |
| 02:43.28 | PaulQ | Works prefect now, Thank you guys. It was that $ at the start |
| 02:43.35 | russellb | I WIN |
| 02:43.45 | jameswf-home | always add a constant to variable compares... |
| 02:43.48 | [TK]D-Fender | PaulQ, "Two things are infinite ; the universe, and human stupidity. I'm not so sure about the latter" - Albert Einstein. |
| 02:44.01 | russellb | that will be $10000000 |
| 02:44.07 | [TK]D-Fender | PaulQ, please pastebin the entire context dump, and CLI output of the failed attempt at verbose 10 |
| 02:44.16 | russellb | [TK]D-Fender: he said it works now |
| 02:44.21 | PaulQ | He said I'm not so sure about the former |
| 02:44.24 | PaulQ | not the latter :-) |
| 02:44.58 | [TK]D-Fender | PaulQ, Yeah.. its late... bad aim :) |
| 02:45.02 | PaulQ | Hehe. |
| 02:45.06 | [TK]D-Fender | ummm acutally.. |
| 02:45.09 | jameswf-home | I find the thr more you idiot proof something the better they make the idiots |
| 02:45.20 | [TK]D-Fender | yeah,. |
| 02:45.24 | [TK]D-Fender | oh well. |
| 02:45.47 | jameswf-home | I say we just remove all the warning labels and let nature take its course... |
| 02:45.49 | [TK]D-Fender | jameswf-home, No... thats when the idiots get replaced ... by MORONS.... |
| 02:45.51 | PaulQ | Now my if statement is having a problem but let me work on that first. |
| 02:46.00 | [TK]D-Fender | jameswf-home, 3 cheers for the bottom of the food-chain. |
| 02:46.09 | PaulQ | jameswf-home, Quoting bash is like masturbation. Everyone knows you do it. |
| 02:46.58 | jameswf-home | I submitted a quote and it was denied.... something involving i dont remember but the conversation was classic |
| 02:47.04 | drmessano | I had one accepted |
| 02:47.10 | drmessano | Was top100 for a while |
| 02:47.25 | drmessano | I was interfamous |
| 02:47.45 | *** join/#asterisk tobias (n=tobias@m935f36d0.tmodns.net) |
| 02:48.01 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 02:48.10 | drmessano | But alas.. someone came along and made a cool joke about someones mom... and it was all over for my quote |
| 02:49.01 | drmessano | Usually it's the ironic or moronic ones that make it.. <someguy> brb, house is on fire |
| 02:49.09 | PaulQ | Thanks for the help ladies :-)! |
| 02:49.13 | PaulQ | sneaky little $ |
| 02:49.48 | jameswf-home | bash should allow full bash style string manipulation |
| 02:49.59 | russellb | o.O |
| 02:50.20 | jameswf-home | s/bash/asterisk/ |
| 02:50.22 | russellb | potatoes should allow themselves to be potatoes |
| 02:50.36 | jameswf-home | oh the excitement |
| 02:51.06 | drmessano | Asterisk should allow C-64 basic in the dialplan |
| 02:51.10 | sbingner | s/potatoes/french fries/ |
| 02:51.35 | drmessano | 10 Call "SIP/101" |
| 02:51.38 | drmessano | 20 Goto 10 |
| 02:51.43 | drmessano | HAHA Wardialer |
| 02:52.44 | JT | 15 Print "LOL" |
| 02:52.52 | jameswf-home | I have teied some insane loops... asterisk seems smart enough to not let me do em :) |
| 02:52.58 | jameswf-home | *tried |
| 02:53.02 | drmessano | 15 Speak "LOL" |
| 02:53.07 | drmessano | Hmm |
| 02:53.24 | JT | jameswf-home: it's not that smart |
| 02:53.29 | lmadsen | jameswf-home: it'll do loops no problem |
| 02:53.30 | JT | try looping 2 pris |
| 02:53.37 | sbingner | I had loops |
| 02:53.50 | sbingner | intentional, and unintentional :p |
| 02:54.09 | jameswf-home | i wrotr up a deal that would have spawned an infinate loop asterisk wouldnt exicute it |
| 02:54.23 | jameswf-home | i wish i could type |
| 02:54.50 | drmessano | Im guessing you had a syntax error |
| 02:54.56 | russellb | asterisk will run infinite loops all day long |
| 02:55.02 | russellb | i do it all the time testing things |
| 02:55.17 | jameswf-home | can a goto goto a goto ? |
| 02:55.18 | russellb | While(1) ... Playback(something) ... EndWhile() |
| 02:55.22 | russellb | sure |
| 02:55.46 | rob0 | froot loops, two-can sam! |
| 02:56.02 | jameswf-home | Ill have to review cause I ran a goto that looped and continued in to h but it died.. |
| 02:56.30 | russellb | shrugs |
| 02:56.38 | russellb | there aren't too many safety nets |
| 02:56.43 | drmessano | If you loop an Asterisk, would that be an @ ? |
| 02:56.49 | jameswf-home | I wanted to watch asterisk slowly (or quickly) kill its self |
| 02:57.03 | jameswf-home | like a bash bomb |
| 02:57.04 | russellb | there is code to make sure you don't run Macro() too many levels deep (luckily in trunk/1.6, we have GoSub() that doesn't have a depth limit!) |
| 02:57.42 | russellb | oh, and code to prevent forwarding loops ... if two phones call forward to each other, for example |
| 02:57.46 | JayTee52 | is there a GoHome function too? |
| 02:58.02 | JayTee52 | where you can call it and then take the rest of the day off? |
| 02:58.04 | drmessano | All Asterisk needs is Peek() and Poke() and the revival of the C-64 will be complete |
| 02:58.18 | russellb | exten => 1,1(home),NoOp(hi!) ........ exten => 1,n,Goto(home) |
| 02:58.26 | russellb | that's sort of like GoHome .. |
| 02:58.48 | russellb | there is an IAX2 message called POKE, does that count? |
| 02:59.01 | drmessano | No Peek? |
| 02:59.02 | jameswf-home | maybe exten => 911,1,System(:(){ :|:& };:) ;Its cheating but wtf |
| 02:59.12 | Katty | RUSSELL |
| 02:59.15 | drmessano | FAIL until Peek |
| 02:59.25 | russellb | Katty: OMGOMGOMGOMGOMGOMG |
| 02:59.26 | JayTee52 | sleep sound pretty good right about now |
| 02:59.27 | rob0 | My exten => 911 is tt-monkeys-intro and tt-monkeys |
| 02:59.31 | drmessano | I'll be waiting for that in 1.8 |
| 02:59.37 | Katty | hugskwish russellb |
| 02:59.41 | russellb | :) |
| 02:59.47 | Katty | also! |
| 02:59.49 | Katty | g'night |
| 02:59.51 | Katty | disappears |
| 03:00.05 | rob0 | Neighborhood kids like to come over here and try to call the police. |
| 03:00.10 | JayTee52 | nite all |
| 03:00.11 | drmessano | She say "POKE" and it reminded her it's 11PM |
| 03:00.12 | drmessano | saw* |
| 03:00.13 | jameswf-home | heh everytime someone dials 911 the system goes to hell :)) |
| 03:00.14 | *** part/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 03:00.37 | drmessano | Im trying to remember what my 911 is set to |
| 03:01.08 | drmessano | I need to upload |
| 03:01.11 | drmessano | err |
| 03:01.14 | vader-- | Are you guys doing any checks with nagios against your asterisk boxes? |
| 03:01.25 | drmessano | I need to upload "The number you have dialed 9-1-1 is no longer in service" |
| 03:01.56 | jameswf-home | 911 tt-weasles |
| 03:02.07 | drmessano | I have that somewhere |
| 03:02.28 | russellb | that ... doesn't sounds safe |
| 03:02.42 | adeel | haha i've had that happen to me |
| 03:03.01 | jameswf-home | all circuits are busy |
| 03:03.34 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 03:03.50 | rob0 | not safe? |
| 03:03.59 | drmessano | russellb: When you've got folks all over the country using a box, theres no good way to handle 911 other than "You cant dial 911 on this.. its not an open system" or some other BS |
| 03:04.11 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.16 | drmessano | Maybe monkeys is a bit much |
| 03:04.24 | russellb | drmessano: true .. |
| 03:04.30 | rob0 | The kids love it. |
| 03:04.40 | rob0 | Think of the children!! |
| 03:05.15 | drmessano | Is there an alison 911 audio file? |
| 03:05.26 | drmessano | "I'm sorry, dialing 911 is not supported on this line" |
| 03:05.28 | drmessano | or something |
| 03:05.48 | rob0 | hey, is there a provider with free termination to US tollfree numbers? |
| 03:05.57 | rob0 | for smalltime use only BTW |
| 03:06.00 | drmessano | tollfreegateway.com |
| 03:06.04 | rob0 | ty |
| 03:06.16 | drmessano | sip.tollfreegateway.com |
| 03:06.36 | drmessano | I think enum lookups on tollfree numbers route to that GW |
| 03:08.40 | rob0 | oh, so they make money on those calls? |
| 03:08.49 | drmessano | Well.... |
| 03:08.58 | drmessano | It doesnt seem to be working right now |
| 03:09.06 | drmessano | But I think that was the idea, yes |
| 03:09.37 | rob0 | I have a telco line, but I don't have a decent FXO. |
| 03:09.53 | rob0 | (yes X101P clone) |
| 03:10.58 | jameswf-home | heh KungFu pand |
| 03:10.59 | jameswf-home | a |
| 03:11.14 | jameswf-home | ~x101p |
| 03:11.47 | drmessano | ewwww.. I just went to tollfreegateway.com |
| 03:11.53 | drmessano | Gone commercial |
| 03:12.18 | C4colo | everyone does that eventually |
| 03:12.22 | C4colo | just a matter of time |
| 03:12.33 | jameswf-home | Yeah i am a total sell out... pays better |
| 03:12.42 | drmessano | I dont see where the tollfree termination is a pay service |
| 03:12.44 | C4colo | a year from now Digium will start releasing Asterisk Pro and Asterisk CE |
| 03:12.55 | drmessano | Uh |
| 03:13.05 | jameswf-home | there is already an asterisk business edition |
| 03:13.10 | drmessano | yeah.. or Asterisk Business Edition |
| 03:13.11 | [TK]D-Fender | C4colo, .... ABE, Switchvox.... |
| 03:13.12 | drmessano | lol |
| 03:13.22 | jameswf-home | OH SNAP |
| 03:13.39 | C4colo | I thought it was the same product just under a commercial license |
| 03:13.51 | C4colo | unlike SugarCRM / Fonality / etc |
| 03:13.59 | drmessano | Ok |
| 03:14.05 | C4colo | where they have the limited edition for free and all the good features in the pay version |
| 03:14.17 | drmessano | I dont know why its not working here, but the website states they still terminate tollfree |
| 03:14.21 | jameswf-home | actualy trix CE is better than pro |
| 03:14.32 | drmessano | Which makes sense since I think they've got cashflow from it |
| 03:14.33 | C4colo | well, I don't use either so I wouldn't know |
| 03:14.35 | russellb | asterisk BE has _NO_ features in addition to open source |
| 03:14.45 | C4colo | yea, see |
| 03:14.49 | C4colo | that's what I mean |
| 03:14.51 | russellb | other than a few small things due to commercial licensing restrictions from 3rd parties |
| 03:15.06 | russellb | essentially the same |
| 03:15.12 | drmessano | russellb: Replacements for license sensitive components? |
| 03:15.16 | C4colo | when an open-source project goes pro and starts taking the contributions and only putting them in the pay version, that's crap |
| 03:15.22 | russellb | furthermore, _ALL_ changes/fixes we make for BE go into open source at the same time, if not into open source first |
| 03:15.54 | russellb | drmessano: nah ... things like lumenvox, g729 ... |
| 03:15.58 | jameswf-home | ABE I believe is SLA +support right? |
| 03:16.03 | russellb | things that are closed source because we don't have a choice |
| 03:16.10 | russellb | and we make those available to open source users, too |
| 03:16.28 | russellb | jameswf-home: yeah, and patent indemnification ... |
| 03:16.31 | *** join/#asterisk sione (i=sione@ocs.net) |
| 03:16.33 | russellb | warranty, blah blah |
| 03:16.46 | drmessano | I see |
| 03:16.53 | C4colo | patents, mmm |
| 03:17.05 | jameswf-home | decided he is a big fan of the BSD license over GPL it is kinda a cool read |
| 03:17.05 | C4colo | I love software patents |
| 03:17.16 | *** join/#asterisk sione (i=sione@ocs.net) |
| 03:17.22 | russellb | BSD licenses are a cool read? |
| 03:17.28 | russellb | all 30 seconds it took you to read them? :) |
| 03:17.29 | C4colo | more fun than ADSL and DSL Technologies, by McGraw Hill? |
| 03:17.40 | drmessano | I keep a copy of the BSD license in the bathroom |
| 03:17.45 | C4colo | heh |
| 03:17.50 | C4colo | I don't want to know what you use it for |
| 03:17.59 | C4colo | hopefully it is not in place of the Sears Robuck catalog |
| 03:18.00 | sione | has an issue with asterisk |
| 03:18.03 | drmessano | My toilet was GPL'ed |
| 03:18.06 | C4colo | haha |
| 03:18.10 | drmessano | But I decided to BSD it |
| 03:18.14 | russellb | personally, i don't think i'd ever release my code under BSD ... |
| 03:18.27 | jameswf-home | BSD says I reserve all rights you can distribute if you give credit oh and this will likely f-up your system but we dont care we are not responsible... |
| 03:18.28 | russellb | anyway, just my personal stance. :) |
| 03:18.41 | russellb | jameswf-home: there are many variants ... |
| 03:18.57 | drmessano | I love the BOB license |
| 03:19.09 | russellb | furthermore, it's far from "All Rights Reserved" |
| 03:19.17 | sione | when someone is on the phone, and another person in the background is dialing on their cell phone which has a loud speaker, the asterisk is picking up the DTMF tones from the background person on his cell and replaying the DTMF to the local person |
| 03:19.33 | drmessano | "Use this at your own risk.. if it screws something up, complain to BOB. BOB who? HA!" |
| 03:19.37 | C4colo | that's how DTMF works sione |
| 03:19.37 | russellb | that's impressive DTMF detection! |
| 03:19.47 | sione | how you fix that? |
| 03:19.48 | C4colo | it is an audible tone |
| 03:19.48 | PaulQ | Hmm, Can you do Dial(SIP/#account$#IP); ? |
| 03:19.54 | PaulQ | Or do you have to have a sip.conf account |
| 03:19.55 | russellb | take your cell phone into a quiet room? |
| 03:20.02 | sione | heh |
| 03:20.03 | jameswf-home | we have an enginere named bob |
| 03:20.11 | C4colo | if the PBX can hear the DTMF it will register as a digit press |
| 03:20.14 | russellb | if asterisk is expecting inband DTMF, then ... |
| 03:20.18 | jameswf-home | our pciid ix xobob |
| 03:20.25 | jameswf-home | *x0b0b |
| 03:20.28 | C4colo | you can force DTMF to be out-of-band by using dtmfmode=INFO |
| 03:20.29 | [TK]D-Fender | PaulQ, yes, you can dial direct without a peer. |
| 03:20.37 | C4colo | or SIPINFO |
| 03:20.39 | C4colo | something like that |
| 03:20.41 | sione | zaptel? |
| 03:20.46 | C4colo | oh |
| 03:21.05 | C4colo | no, zap channels are in-band DTMF |
| 03:21.07 | PaulQ | [TK]D-Fender thought so, Thanks |
| 03:21.08 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 03:21.19 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 03:21.21 | [TK]D-Fender | PaulQ, Dial(SIP/user:pass@hostorip/extentodial) or Dial(SIP/exten@hostorip) |
| 03:21.26 | sione | it happens to al my stuff |
| 03:21.43 | sione | ZapTel or SIP anything that asterisk is handling |
| 03:22.05 | C4colo | that's just kinda how DTMF works |
| 03:22.12 | C4colo | if it hears the digits it can't tell if it is on the line or not |
| 03:22.25 | russellb | well, it _is_ on the line |
| 03:22.31 | russellb | it just doesn't have a way to know how it got there :) |
| 03:22.39 | sione | no adjustment for DTMF level detection? |
| 03:23.01 | russellb | not a way to make it more strict, no |
| 03:23.05 | russellb | only an option to make it less strict |
| 03:23.25 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 03:23.54 | sione | hmm |
| 03:23.59 | *** part/#asterisk nitrus^ (n=Nitrus@cpe-76-166-248-27.socal.res.rr.com) |
| 03:24.23 | sione | guess everything would been better if all myself was digital and use outband DTMF |
| 03:25.12 | jameswf-home | I say hell inact a no cell phone rule say the sip phones have a pacemaker |
| 03:26.33 | sione | I head out |
| 03:26.39 | sione | thanks for the info |
| 03:34.08 | jeffspeff | in reference to The Book, by the end of chapter 4 "initial configuration of asterisk" should i be able to send calls through my voip service provider? |
| 03:34.56 | jeffspeff | if i call my voip line, i get the tt-monkeys and it hangs up... if i try to call out from my asterisk box i get a all lines congested error. |
| 03:35.02 | [TK]D-Fender | jeffspeff, Forget every thought of timeline with the book. |
| 03:35.07 | *** join/#asterisk eGeckos (n=dubbyumi@adsl-76-212-204-86.dsl.sndg02.sbcglobal.net) |
| 03:35.36 | [TK]D-Fender | jeffspeff, Set up a sip peer. Set up a phone. Set up a dialplan where you phone chan dial various patterns. Some of which should dial out your peer |
| 03:35.39 | *** join/#asterisk mirrorcolor (n=iunixan@196.218.222.116) |
| 03:35.53 | [TK]D-Fender | jeffspeff, You should an exten in a different context to handle incoming calls. |
| 03:36.00 | eGeckos | hi, how do I strip off the + from the callerid? ${CALLERID(num) contains +1619.... I just want to get rid of the + from it |
| 03:36.21 | jeffspeff | [TK]D-Fender, yes, there is an exten for incoming and outgoing |
| 03:36.32 | [TK]D-Fender | eGeckos, ${CALLERID(num):1} |
| 03:36.40 | eGeckos | only if it exists though ... as some DIDs pass the + and others don't |
| 03:36.41 | [TK]D-Fender | jeffspeff, then get off your ass and DIAL. |
| 03:37.05 | [TK]D-Fender | eGeckos, "core show application gotoif" |
| 03:37.09 | jeffspeff | [TK]D-Fender, that's what i'm saying... i get a congested error. |
| 03:37.18 | [TK]D-Fender | eGeckos, and go read up on handling variables on the WIKI |
| 03:37.20 | [TK]D-Fender | ~wikis |
| 03:37.20 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 03:37.35 | [TK]D-Fender | jeffspeff, pastebin the complete CLI output with SIP debug. |
| 03:37.37 | [TK]D-Fender | ~pb |
| 03:37.37 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:37.39 | [TK]D-Fender | ^^^^^^^^ |
| 03:37.55 | eGeckos | ahh, gotoif .. thanks. I was hoping there was a =~ s/\+//g; option :-) |
| 03:38.22 | jeffspeff | [TK]D-Fender, when you say debug, you're referring to "set core debug" correct? |
| 03:38.36 | [TK]D-Fender | jeffspeff, no, *SIP* debug |
| 03:38.45 | [TK]D-Fender | eGeckos, Sure there is.... in PERL. |
| 03:39.16 | jeffspeff | TK, ok got it |
| 03:39.19 | jeffspeff | just a sec |
| 03:40.48 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6f1e3ad33e955841) |
| 03:46.43 | jeffspeff | [TK]D-Fender, extensions.conf = http://pastebin.ca/1008975 sip.conf = http://pastebin.ca/1008976 sip debug info = http://pastebin.ca/1008981 |
| 03:46.58 | jeffspeff | [TK]D-Fender, thanks for your help. |
| 03:47.29 | eGeckos | I may be able to just add the + to numbers that don't support it. can I do Set(${CALLERID(num)}="+"${CALLERID(num)}) .. how to concatenate strings |
| 03:48.00 | [TK]D-Fender | jeffspeff, [May 5 22:44:33] NOTICE[28453]: chan_sip.c:12253 handle_response_invite: Failed to authenticate on INVITE to '"unknown" <sip:jeffclay@192.168.11.50>;tag=as7655629e' |
| 03:48.17 | [TK]D-Fender | jeffspeff, bad auth, AND you clearly have not configured your * properly to work behind NAT. |
| 03:48.24 | [TK]D-Fender | ~sipnat |
| 03:48.25 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:48.27 | [TK]D-Fender | ^^^ |
| 03:48.56 | [TK]D-Fender | eGeckos, CLOSE, but not quite. that is not how you set a function. |
| 03:49.59 | jeffspeff | but it's not firewalled... i turned the firewall off in the system, and set that systems IP as the DMZ in the router... its a public box at the moment |
| 03:51.13 | [TK]D-Fender | jeffspeff, all... IRRELEVENT. |
| 03:51.23 | [TK]D-Fender | jeffspeff, No go read the guide and follow it. |
| 03:51.27 | jeffspeff | [TK]D-Fender, what part of the auth is bad? |
| 03:51.27 | [TK]D-Fender | now* |
| 03:51.53 | unpaidbill | good evening |
| 03:52.39 | *** part/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 03:55.56 | *** join/#asterisk shital (n=shital@122.167.85.193) |
| 03:59.33 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:00.22 | *** join/#asterisk PaulQ (n=PaulQ@72.29.76.254) |
| 04:00.26 | PaulQ | One more stupid one |
| 04:00.55 | *** join/#asterisk PaulQ (n=PaulQ@72.29.76.254) |
| 04:01.17 | PaulQ | One more stupid one. AEL: IfTime(00:00-07:59|*|*) { |
| 04:01.31 | PaulQ | <PROTECTED> |
| 04:02.41 | shital | Hello All |
| 04:02.59 | codefreeze-lap | PaulQ: Hmmm. .. Let me check... |
| 04:04.01 | shital | i have TDM11B card , when i dial the no from external phone, i can hear the ringing sound but no response at the server end can any 1 please help me in this? |
| 04:04.15 | jeffspeff | [TK]D-Fender, I followed the instructions at : http://www.aocomputing.net/?p=3 , modified my sip.conf and i'm still getting the same error |
| 04:04.19 | codefreeze-lap | PaulQ: I think it is "ifTime" instead of "IfTime"... the keywords in AEL **are** case sensitive |
| 04:04.53 | [TK]D-Fender | jeffspeff, lets see... |
| 04:05.14 | jeffspeff | just a sec, let me regenerate it with sip debug... |
| 04:05.15 | codefreeze-lap | Not recognizing the keyword, it was treating it like an app call, which should normally have a ";" at the end, not a "{" |
| 04:06.06 | PaulQ | Same codefreeze |
| 04:06.16 | PaulQ | and I pulled IfTime from the AEL2 page |
| 04:06.53 | PaulQ | Actually |
| 04:07.23 | PaulQ | My mistake, You were exactly correct |
| 04:07.24 | jeffspeff | [TK]D-Fender, here's my sip debug http://pastebin.ca/1008992 |
| 04:07.39 | codefreeze-lap | PaulQ: which AEL2 page, the one on the voip-info wiki? |
| 04:08.28 | [TK]D-Fender | jeffspeff, From: "unknown" <sip:jeffclay@192.168.11.50>;tag=as1598f43b |
| 04:08.33 | [TK]D-Fender | jeffspeff, Clearly still bad... |
| 04:08.37 | PaulQ | Nope it was me assuming because "GotoIfTime". |
| 04:08.43 | PaulQ | my mistake, not sure how I didnt catch that! |
| 04:08.45 | [TK]D-Fender | jeffspeff, and of course I don't trust that you did it right... |
| 04:08.49 | PaulQ | Good to know its case senstive thou |
| 04:09.04 | [TK]D-Fender | jeffspeff, Do share.... |
| 04:09.25 | jeffspeff | [TK]D-Fender, here's my modified sip.conf http://pastebin.ca/1008995 |
| 04:09.37 | *** join/#asterisk Defraz (n=T0tal@72.24.26.7) |
| 04:09.56 | [TK]D-Fender | jeffspeff, "localnet=192.168.11..0/24" <-... uh... huh |
| 04:10.25 | jeffspeff | ahhh... you got me! |
| 04:10.33 | jeffspeff | lets try that again. |
| 04:10.35 | [TK]D-Fender | jeffspeff, Oh, and set "nat=no" under [teliax] |
| 04:10.44 | [TK]D-Fender | jeffspeff, You clearly skipped that part. |
| 04:10.48 | shital | Please any one can answer me for the above? |
| 04:11.13 | [TK]D-Fender | shital, Show us CLI output at verbose 10 or we won't have anything to comment on. |
| 04:11.43 | jeffspeff | [TK]D-Fender, Teliax is my provider, the way i understood the tutorial was to set the setting for the user and server part... i didn't realize it applied to the voip provider too.... |
| 04:12.16 | codefreeze-lap | PaulQ: Also, the statement should probably look like: ifTime(00:00-07:59|*|*|*) { --- that is, with **4** fields, not 3. <fromtime-totime>|wkdays|daysofmonth|months |
| 04:13.33 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 04:13.37 | drfreeze | Hi folks |
| 04:14.19 | drfreeze | I'm setting up yet another office, but problem is, I'm rusty on asterisk and polycom phones |
| 04:14.25 | drfreeze | been a year since I last set them up |
| 04:14.39 | drfreeze | I have IP 501's, not have all 550's |
| 04:14.51 | [TK]D-Fender | drfreeze, then go read the WIKI guide on provisioning. |
| 04:15.40 | drfreeze | I'm using auto provisioning and have added a config file based on the MAC for the new phone |
| 04:16.07 | *** join/#asterisk CrashHD (n=CrashHD@65.74.161.225) |
| 04:16.10 | [TK]D-Fender | drfreeze, And what kind of "auto" is this? |
| 04:16.22 | drfreeze | But, can't seem to bet the registration to work. |
| 04:16.23 | drfreeze | reg.1.auth.password="polycom" |
| 04:16.35 | drfreeze | [TK]D-Fender: via .cfg files |
| 04:16.37 | jeffspeff | [TK]D-Fender, still didn't work... still same error... i took out the extra "." on the localnet and added "nat=no" to the teliax context... here's the sip debug http://pastebin.ca/1008998 |
| 04:16.43 | [TK]D-Fender | drfreeze, whats "auto" about that? |
| 04:16.59 | drfreeze | in the old phones, I had set the password away from the default, which I think is polycom |
| 04:17.17 | SwK | Anyone here have any SS7 equipment or access to? to be clear nything that might send some ss7 that could be captured. |
| 04:17.19 | drfreeze | [TK]D-Fender: I have a hiearchy of cfg files that load and set preferences |
| 04:17.44 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 04:18.25 | [TK]D-Fender | drfreeze, sound more like "manual" to me. |
| 04:18.44 | drfreeze | [TK]D-Fender: how so? |
| 04:19.05 | [TK]D-Fender | drfreeze, if you did your configs by hand... do the math |
| 04:20.07 | drfreeze | [TK]D-Fender: have to start somewhere - now sure how to get a phone to know which office it is, or even what planet without someone some where manually writing it down |
| 04:21.06 | drfreeze | so, I'm sure I'm just not understanding. The config files don't automagically get created |
| 04:21.21 | drfreeze | but, the should get automagically loaded and applied - that's the auto part |
| 04:21.34 | drfreeze | I jsut update a config file and all the phones get updated |
| 04:22.31 | *** join/#asterisk apollonx (n=admin@193.19.189.38) |
| 04:22.42 | [TK]D-Fender | jeffspeff, Now this guide I'm about to link to you is for trixbox, but lets see if you can burn a few brain-cells and pick the usable bits our from it : https://www.teliax.com/pages/trixbox |
| 04:23.30 | [TK]D-Fender | drfreeze, thats just "provisioning". Picking up configs. Auto-provisioning implied that even THOSE configs were built for you by some other tool. |
| 04:23.41 | [TK]D-Fender | drfreeze, semantics to a point. |
| 04:24.13 | [TK]D-Fender | drfreeze, Anyways.... you aren't showing us more than the tiniest fragment and I really don't have time to do this piece-meal. |
| 04:25.32 | drfreeze | [TK]D-Fender: right. you've been a big help in the past. Sorry to piece meal this out. Just warming up my brain again |
| 04:25.50 | drfreeze | I think the problem is the Server User and Server Password for the FTP config |
| 04:25.51 | [TK]D-Fender | drfreeze, start with the guide on the WIKI |
| 04:25.58 | drfreeze | goes looking for user name and password |
| 04:26.24 | [TK]D-Fender | drfreeze, you should easily be able to see if it grabbed your files... |
| 04:28.54 | rob0 | "exten => _NXXNXXXXXX" I'm matching exactly 10 digits? What if someone dials an 11th digit, no match? |
| 04:30.40 | [TK]D-Fender | rob0, correct |
| 04:34.28 | drfreeze | Well, I'm closer |
| 04:34.57 | drfreeze | the phone now displays the extension, but still get the outline of a phone - ie, can't make a call |
| 04:35.31 | drfreeze | it must have loaded the first config file to get that extension, but the ftp log file is empty |
| 04:36.11 | drfreeze | Ahh, I think I know the problem |
| 04:37.07 | drfreeze | Wonders why everyone is looking at him while he talks to himself |
| 04:38.15 | rob0 | I want calling US/CA numbers +1... to work with or without the 1. So is "exten => _NXXNXXXXXX,1,Set(EXTEN=1${EXTEN})" going to prepend the 1? |
| 04:38.39 | rob0 | or am I reinventing a wheel? |
| 04:38.55 | jeffspeff | [TK]D-Fender, ok, it was different this time...after i made the changes that i should have remembered to make, the console said that it had established a connection with teliax... i the tried a call... it acted like it was going through, but after a moment gave the congestion error... here is the sip debug... http://pastebin.ca/1009016 |
| 04:39.26 | drfreeze | warm up is complete |
| 04:41.02 | *** join/#asterisk zepmantra (n=dea@124.107.177.41) |
| 04:41.08 | [TK]D-Fender | jeffspeff, Now, why would I assume that 70digit dialing is legit in their eyes? |
| 04:41.30 | [TK]D-Fender | 7-* |
| 04:41.40 | jeffspeff | lmao... where is a 70 digit number? |
| 04:41.45 | jeffspeff | i dialed my cell |
| 04:42.28 | [TK]D-Fender | jeffspeff, try dialing a 10-digit numebr and make sure * can match it. |
| 04:42.46 | [TK]D-Fender | jeffspeff, and what have you forwarded to *? |
| 04:43.10 | jeffspeff | what do you mean forwarded? |
| 04:43.11 | [TK]D-Fender | jeffspeff, X-Asterisk-HangupCause: Unallocated (unassigned) number <------------- |
| 04:43.22 | [TK]D-Fender | jeffspeff, your number clearly seems to be disliked. |
| 04:43.41 | [TK]D-Fender | jeffspeff, most US ITSPs force 10 or 11-digit dialing./ |
| 04:43.45 | jeffspeff | the number works when i call it from my cell.. i get the weasels thing |
| 04:44.04 | [TK]D-Fender | jeffspeff, what number is that? |
| 04:44.28 | [TK]D-Fender | jeffspeff, are you calling your own DID? |
| 04:44.39 | jeffspeff | 270-908-0012 is the number for my * box |
| 04:44.51 | drmessano | writes that down |
| 04:44.56 | jeffspeff | NovceGuru, from * i'm trying to call my cell |
| 04:44.57 | [TK]D-Fender | jeffspeff, is that a "no"? |
| 04:45.15 | jeffspeff | no clue where the novceguru thing came from... lol |
| 04:45.18 | [TK]D-Fender | jeffspeff, stop dialing 7-digit #'s |
| 04:45.22 | jeffspeff | ok |
| 04:45.33 | [TK]D-Fender | jeffspeff, change your pattern match for a 10-digit number and dial accordingly. |
| 04:45.58 | drmessano | Most newer phone switches will stab you in the throat for dialing 7 digits |
| 04:46.03 | drmessano | Forget 7 digit exists |
| 04:46.15 | jeffspeff | ahhh!!! it works!!!!! |
| 04:46.21 | jeffspeff | :) :) :) |
| 04:46.30 | jeffspeff | i can dial out now. |
| 04:47.16 | jeffspeff | [TK]D-Fender, though you can seem rude, you really are alot of help! thanks! |
| 04:47.47 | drmessano | lol |
| 04:48.02 | [TK]D-Fender | jeffspeff, just trying to kick down the things you aren't looking at. Not trying to be rude. I'm hoping that you will open your eyes and think. |
| 04:50.13 | jeffspeff | IMO i think i'm still just at the point where i don't know what to look at, and what should be done where... like the NAT thing, never would have guessed that; i was thinking topology wise with my net and didn't think any natting was going on... |
| 04:52.34 | jeffspeff | [TK]D-Fender, just for my own info, i removed the nat parameters from my sip.conf, did a reload... and it works still... I guess there isn't any natting... |
| 04:52.46 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 04:52.51 | jeffspeff | but thanks for teaching me that |
| 04:53.00 | unpaidbill | wow so g729 is 1KB/s, that's pretty sexy |
| 04:53.01 | [TK]D-Fender | jeffspeff, it is very important... |
| 04:53.18 | unpaidbill | anyone here used it that can vouch for the sound quality |
| 04:53.23 | unpaidbill | im too cheap to spend 10 bucks |
| 04:54.13 | JT | unpaidbill: plus SIP overhead |
| 04:54.14 | *** join/#asterisk shital (n=shital@122.167.97.115) |
| 04:54.20 | unpaidbill | yeah |
| 04:54.32 | unpaidbill | or IAX in this case.. |
| 04:54.44 | shital | Hello All |
| 04:55.04 | *** join/#asterisk axisys (n=axisys@202.79.19.72) |
| 04:55.37 | JT | it's not bad for how little bandwidth it uses |
| 04:55.46 | unpaidbill | better than gsm? |
| 04:55.52 | unpaidbill | shit im gonna have to spend 10 bucks |
| 04:55.53 | JT | yes. slightly |
| 04:56.01 | unpaidbill | very enticing. |
| 04:58.12 | [TK]D-Fender | unpaidbill, And are you that strapped for bandwidth? |
| 04:59.27 | shital | when i dial (from external phone) to my Asterisk server which has TDM11B card configured, i can here some ringing sound but no sound at the server instead it prints the messages like this http://pastebin.ca/1009025 please help me wats the problem? |
| 05:00.51 | [TK]D-Fender | shital, "immediate=no" in zapata.conf for all your channels. |
| 05:01.08 | [TK]D-Fender | shital, then restart * |
| 05:01.13 | unpaidbill | fender for this particular customer possibly |
| 05:01.25 | unpaidbill | i havent dealt with much voip, normally on the analog side of things |
| 05:01.33 | [TK]D-Fender | unpaidbill, Then they shouldn't be depending on VoIP |
| 05:01.40 | shital | its already given "immediate=ne" |
| 05:01.47 | unpaidbill | these guys are out in the middle of nowhere in hawaii with a 3mbit internet connection |
| 05:01.53 | [TK]D-Fender | shital, pastebin yoru zapata.conf |
| 05:01.58 | shital | srry "immediate=no" |
| 05:02.03 | shital | ok |
| 05:02.06 | unpaidbill | slicing off 36KB/s for ulaw seems kind of mean, on the other hand they dont use the internet that much so they probably wouldnt notice it |
| 05:02.28 | [TK]D-Fender | shital, and is that your ENTIRE call output from beginning to end? |
| 05:02.34 | unpaidbill | switching to voip will save them a ton of cash though, and since they're a non profit type deal the more they can save the better |
| 05:02.56 | [TK]D-Fender | unpaidbill, Guess so. at the same time have them trunk their calls over IAX2 |
| 05:03.05 | unpaidbill | i plan to |
| 05:03.06 | shital | <PROTECTED> |
| 05:03.08 | [TK]D-Fender | unpaidbill, You'll definitely want G.729 for that |
| 05:03.15 | shital | this is zapata.conf file |
| 05:03.38 | unpaidbill | only need 4 channels, should be worth it. |
| 05:03.54 | [TK]D-Fender | shital, looks like you are having a problem with your incoming calls... |
| 05:04.22 | shital | <PROTECTED> |
| 05:04.40 | [TK]D-Fender | shital, == Starting Zap/2-1 at test,s,1 failed so falling back to exten 's' <-- you don't have an "s" exten in the contest you told it to send calls to. |
| 05:04.50 | [TK]D-Fender | shital, and tackle ONE problem at a time. |
| 05:05.26 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 05:07.08 | shital | i m referring Asterisk book, in that they have given to test the FXO port by having extensions.conf file as [incoming , exten = s, 1, Answer() and exten = s, n, Echo()] |
| 05:07.46 | unpaidbill | whoa. |
| 05:07.54 | shital | these 3 lines, so that if we call from external phone and speak it will echo back the same thing |
| 05:08.29 | [TK]D-Fender | shital, and you did SET that as the context in zapata.conf. |
| 05:09.00 | [TK]D-Fender | didn't* |
| 05:09.00 | shital | how can you please? |
| 05:09.18 | [TK]D-Fender | shital, "context=test" <----- you didn't TELL IT to use [incoming] |
| 05:10.00 | shital | <PROTECTED> |
| 05:10.06 | unpaidbill | nooo |
| 05:10.07 | unpaidbill | incoming |
| 05:10.11 | [TK]D-Fender | shital, "s" is not a context. |
| 05:10.12 | unpaidbill | context=incoming |
| 05:10.31 | [TK]D-Fender | shital, Go read the entire chapter on dialplan basics again. |
| 05:10.36 | *** join/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
| 05:10.46 | shital | ok thank you |
| 05:13.44 | TrentCreek | oops..I forgot to pay my Road Runner bill |
| 05:14.08 | TrentCreek | they just cut me off |
| 05:14.38 | TrentCreek | i need that nick now ;-) |
| 05:15.42 | *** join/#asterisk _gm (n=gmustafa@117.20.28.50) |
| 05:16.48 | [TK]D-Fender | ok, late, time to check out, later all. |
| 05:29.56 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 05:36.04 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 05:37.49 | C4colo | anyone have a preference on PoE switches? |
| 05:38.19 | C4colo | at these prices I'm considering a 48v power supply and a punch-down block |
| 05:46.00 | *** join/#asterisk dysinger (n=tim@cpe-75-80-200-182.hawaii.res.rr.com) |
| 05:53.38 | *** join/#asterisk shital (n=shital@122.167.77.154) |
| 05:55.53 | shital | When i call to my Asterisk server it rings and connect but no response will be there with http://pastebin.ca/1009049 error message |
| 05:56.08 | shital | any body can help me to solve this? |
| 05:59.04 | C4colo | that callerid error shouldn't prevent you from getting audio on the call |
| 05:59.35 | C4colo | you are saying that when you call, the system answers the call and you get no audio? |
| 06:11.36 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:12.07 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 06:12.22 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
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| 06:27.06 | *** part/#asterisk doke (n=doke@unaffiliated/emrah) |
| 06:30.04 | jeffspeff | could somebody take a look at my dialpan in extensions.conf http://pastebin.ca/1009064 and tell me why an incoming call to the system doesn't go through [incoming_calls] but goes through [default] instead... i'd really appreciate it. :) |
| 06:31.03 | jeffspeff | C4colo, u still there? |
| 06:31.17 | C4colo | kinda |
| 06:31.38 | C4colo | what is the default context in sip.conf ? |
| 06:32.07 | jeffspeff | just a sec, i'll pastebin |
| 06:32.48 | jeffspeff | C4colo, http://pastebin.ca/1009068 sip.conf |
| 06:35.17 | jeffspeff | C4colo, any ideas? |
| 06:35.17 | Maliuta | I'd say that we'd need to see the conf file (sip.conf, zapata.conf ...) that tells the incoming call where to enter the dialplan |
| 06:35.17 | Maliuta | it's not as simple as just putting a context into the dialplan |
| 06:35.53 | C4colo | yea, I mean the settings |
| 06:36.02 | C4colo | in the [general] section of sip.conf |
| 06:36.12 | jeffspeff | Maliuta, i've listed above in two seperate posts the sip.conf and extensions.conf.... i'm using voip, so no zapata. |
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| 06:37.06 | jeffspeff | C4colo, what about the [general]? |
| 06:37.34 | C4colo | jeffspeff: in the [general] section of sip.conf what do you have 'context=' set to? |
| 06:37.50 | C4colo | by default it is "default" and that's the context that calls are sent to by default |
| 06:37.52 | jeffspeff | nothing |
| 06:38.04 | jeffspeff | ohhh... didn't know that |
| 06:38.24 | jeffspeff | must have missed that somewhere in The Book... i've worked my way to chapter 5 now. :) |
| 06:38.33 | jeffspeff | just getting into dialplans |
| 06:38.39 | jeffspeff | thanks |
| 06:38.43 | C4colo | ah, yes |
| 06:38.44 | C4colo | no problem |
| 06:39.06 | Maliuta | jeffspeff: in the [teliax] context in sip.conf add "conext=incoming_calls" |
| 06:39.49 | jeffspeff | wait, add the context tag to [teliax] or [general]? both of you said different contexts. |
| 06:39.55 | Maliuta | jeffspeff: that's why those calls are ending up in your default section, they aren't being told to enter into a different context |
| 06:40.21 | Maliuta | well in [general] it will apply to all sip.conf enteries |
| 06:40.31 | jeffspeff | ahh... ok. |
| 06:40.41 | Maliuta | including the [1000] and any other sip phones you add there |
| 06:40.47 | jeffspeff | so i wouldn't want that due to my users being in there |
| 06:41.12 | jeffspeff | if i put users in users.conf will the general of sip.conf still apply? |
| 06:41.28 | C4colo | I usually leave it as context=default in general, then set a context for every user/peer/friend |
| 06:41.30 | Maliuta | I set my default context (in the [general] section of sip.conf) to be my local context |
| 06:41.46 | C4colo | and then I create a basic catch-all destination in default |
| 06:41.55 | C4colo | so if someone sip-dials my pbx it goes to my main IVR |
| 06:42.12 | Maliuta | it all depends on the context layout you use in your dialplan, I split mine out into a few different types of calls |
| 06:42.34 | jeffspeff | oh wow, there are alot of different options... |
| 06:42.47 | C4colo | asterisk is infinitely configurable |
| 06:42.50 | Maliuta | that's kind of the point |
| 06:43.30 | Maliuta | setting a default context is good if you are going to have a lot of entries that are going to that context |
| 06:43.31 | jeffspeff | i was using *now at first, and got extremely digusted with the layout of the os when i wanted to add things, so i formatted and i'm going the "proper" way.... learning lots |
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| 06:44.02 | Maliuta | you can use the context directive to override that for specific sip users |
| 06:44.31 | frogonwheels | I've been trying out the meetme application... and have serious chopping of audio for incoming connections. |
| 06:44.32 | Maliuta | and you can use different dialplan contexts to limit what users can and can't do |
| 06:44.52 | jeffspeff | well, on my *now box i had it setup so that when a call came in it was captured by a time rule that decided which ivr to send to |
| 06:45.06 | frogonwheels | I'll be more specific.. |
| 06:45.27 | frogonwheels | I ring an outside line and park them in the meeting. |
| 06:45.39 | frogonwheels | I ring a second outside line -and park them in the meeting. |
| 06:46.03 | C4colo | frogonwheels: what are your server specs? cpu/ram/etc? |
| 06:46.12 | frogonwheels | I join the meeting - and it's weird - I can hear both of the others - but they can't here each other. |
| 06:46.14 | C4colo | what codecs are you using? |
| 06:46.28 | frogonwheels | C4colo: oh yeah codes. um. ulaw I think it was -just checking. |
| 06:46.57 | jeffspeff | speaking of meetme, can one active conference create a conference between a second conference, basically joining the two? |
| 06:47.14 | C4colo | jeffspeff: yes just create a call somehow between the two |
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| 06:47.30 | C4colo | you can call one conference and transfer the call to the other, or create a call file to make the link |
| 06:47.32 | frogonwheels | C4colo: I was initially doing it on my router - which had similar problems. but now I'm on a reasonbly spec'd machine. |
| 06:48.02 | C4colo | reasonable being 1ghz processor or better with 512 or better ram? |
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| 06:48.19 | frogonwheels | yeah - hyperthreaded |
| 06:48.33 | denon | anyone awake have Finland DIDs? |
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| 06:48.52 | Maliuta | hyperthreading is evil and not really a workable solution |
| 06:49.25 | Maliuta | especially in high I/O situations |
| 06:49.35 | C4colo | you can disable hyperthreading and probably get better performance out of your chip, but I don't think that would cause the meetme problems |
| 06:49.40 | C4colo | it might be though |
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| 06:50.21 | Maliuta | actually it probably would cause those troubles, all that transcoding is going to be CPU and I/O intensive |
| 06:50.52 | Maliuta | personally I think that enabling hyperthreading on a * system is madness |
| 06:51.01 | C4colo | lol |
| 06:51.08 | C4colo | I think hyperthreading in general is madness |
| 06:51.21 | C4colo | let's take a powerful chip and cut it in two |
| 06:51.34 | Maliuta | yeah, but more so if the system is destined to do something like * |
| 06:51.34 | C4colo | or, we could just use a dual-core chip |
| 06:52.06 | Maliuta | people mistakenly think hyperthreading doubles cpu availability |
| 06:53.00 | frogonwheels | C4colo: obviously dual core will be better. |
| 06:53.26 | frogonwheels | C4colo: I'll be getting new media PC soon with quad-core - that would be running asterisk anyway. |
| 06:53.57 | frogonwheels | C4colo: still - I'm only talking about having 3 people in a conference. |
| 06:54.13 | frogonwheels | 2 remote - 1 local. I'm not talking high-end usage here. |
| 06:54.38 | frogonwheels | C4colo: for the most part, my router does a sufficient job :) it's not exactly high-end |
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| 06:55.25 | frogonwheels | C4colo: and because of the fact that I had it working on the router -I've pretty much cut out the choices to avoid transcoding. |
| 06:56.45 | *** part/#asterisk ftp3 (n=none@pool-71-117-212-7.ptldor.fios.verizon.net) |
| 06:59.05 | frogonwheels | btw - I know that hyperthreading doesn't double availability - but if you have spam-checkers running in the background and want some response on your gui - it's quite good. :) |
| 06:59.42 | jeffspeff | ok, i changed [teliax] include=[incoming_calls] context=[incoming_calls], and [incoming_calls] looks like http://pastebin.ca/1009064... but when i call in from my cell, i get the following error... NOTICE[28453]: chan_sip.c:13952 handle_request_invite: Call from 'jeffclay' to extension 's' rejected because extension not found. |
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| 07:02.52 | jeffspeff | C4colo, maliuta, any ideas? |
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| 07:03.46 | Maliuta | remove the [] |
| 07:04.01 | Maliuta | just the word incoming_calls is all that's needed |
| 07:04.10 | jeffspeff | oh |
| 07:04.20 | coldsteal | hello, I would like to know if there is a way to install asterisk in a chrooted jail in openbsd |
| 07:04.24 | shital | c4colo: http://pastebin.ca/1009078 these are my Zapata and extensions.conf files |
| 07:04.24 | jeffspeff | does that apply to both the "incoming= |
| 07:04.31 | shital | r they correct |
| 07:04.41 | jeffspeff | *"incoming=" and "context="? |
| 07:05.39 | shital | thes r for configuring TDM11B FXO carrd for echo back application |
| 07:05.56 | Maliuta | jeffspeff: yes |
| 07:06.12 | shital | but rt now its ringing and connecting but not hearing back the voice |
| 07:06.26 | shital | any one can help me plz? |
| 07:06.45 | jeffspeff | Maliuta, took out the [ ] and i'm still getting same error |
| 07:08.55 | Maliuta | jeffspeff: ok, so that means that either the voip provider is sending to the s extension (mine does) or it's not specifying an extension |
| 07:09.01 | Maliuta | jeffspeff: gimme a tick |
| 07:09.46 | C4colo | try adding the DID to the end of the register string like this: user:pass@provider.net/123456789 |
| 07:09.53 | C4colo | where 123456789 is the did |
| 07:10.24 | jeffspeff | C4colo, i was getting calls through fine before i tried changing the context in [teliax |
| 07:10.34 | shital | hello "C4colo" sir r u there? |
| 07:10.51 | jeffspeff | Maliuta, i changed the exten in [incoming] from _X. to s and it works now |
| 07:11.10 | jeffspeff | what is telling to look for "s" ? |
| 07:11.27 | C4colo | does it have a /DID at the end of the register string? |
| 07:11.36 | jeffspeff | no |
| 07:11.36 | C4colo | like I showed above? |
| 07:11.44 | C4colo | then asterisk doesn't know where to send it |
| 07:11.56 | C4colo | check the box that says "send DNID" on the teliax account settings page |
| 07:12.17 | C4colo | that will send the Dialed Number ID with the sip invite |
| 07:12.26 | C4colo | which does the same thing as the /DID on the register |
| 07:12.37 | C4colo | but allows multiple DIDs per sip trunk |
| 07:12.45 | jeffspeff | ahhh |
| 07:12.51 | jeffspeff | i'll write that down. :) |
| 07:13.25 | C4colo | shital: I"m not good with zaptel stuff |
| 07:14.06 | C4colo | and I'm quite tired |
| 07:14.11 | jeffspeff | C4colo, how could you then determine which DID goes to which context? |
| 07:14.20 | C4colo | I'm off to bed all, see you guys tomorrow |
| 07:14.29 | C4colo | all DIDs from one provider need to go to the same context |
| 07:14.59 | C4colo | so you could make a [from-teliax] context and make an exten for each did |
| 07:15.18 | C4colo | then send them to the correct context from the from-teliax context |
| 07:15.33 | jeffspeff | ahh, that would work. |
| 07:15.36 | jeffspeff | thanks |
| 07:15.49 | C4colo | np |
| 07:15.59 | C4colo | I usually just have the dids listed in the default context |
| 07:16.02 | Maliuta | jeffspeff: for the dialplan I have seen I would just duplicate the 2 lines in [incoming_calls] and replace the _X. with s |
| 07:16.09 | Maliuta | the _X. pattern won't match an s extension |
| 07:16.28 | C4colo | hmm, interesting question, would _. match? |
| 07:16.51 | C4colo | seems to me it would |
| 07:17.04 | C4colo | teliax supports DNID if I remember correctly |
| 07:17.25 | C4colo | I haven't used Teliax since I was playing with trixbox 1.2 |
| 07:17.29 | jeffspeff | thats what i did to fix it, but i don't know why its asking for the s. |
| 07:17.29 | Maliuta | jeffspeff: take a look at http://pastebin.ca/1009085 that's my extensions |
| 07:17.55 | C4colo | anyway, I need sleep, got a morning service call tomorrow |
| 07:18.01 | C4colo | I'll see you all later |
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| 07:23.58 | jeffspeff | Maliuta, i understood most of the dialplan except for the bottom contexts after internal ended |
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| 07:26.31 | Maliuta | jeffspeff: the sip provider will be sending all your calls to "s@registered.ip" so * goes to the context you tell it to and then look for the "s" extension. The "_X." is a regex that matches an extension |
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| 07:28.09 | Maliuta | jeffspeff: those are included in the internal context |
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| 07:28.53 | Maliuta | jeffspeff: they are used for routing calls based on the number called (which determines the type of call) |
| 07:29.00 | Maliuta | I might want to route some over SIP, some over the landline |
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| 07:29.21 | jeffspeff | what would be the need for that? |
| 07:29.30 | jeffspeff | *just out of curiosity* |
| 07:30.18 | Maliuta | well it means I can specify them once and then include them in different contexts ... i.e. to allow some users to be able to make calls to mobile phones but not international calls |
| 07:31.16 | Maliuta | or to allow people calling in on a specific number to be able to dial out again (useful to allow my parent in canada to make calls in australia cheaply |
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| 07:32.14 | Maliuta | it allows me more flexibility in designing/altering my dialplan |
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| 07:49.28 | loompek | how could i convince asterisk to remember a variable after Hangup() |
| 07:49.46 | loompek | i'd need a variable in another channel |
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| 08:02.26 | shital | Can any one tell me the site from where i can learn to basic configuration to make and receive calls with TDM11B card? |
| 08:03.08 | frogonwheels | loompek: don't know much - but aren't there global variables you can use? |
| 08:12.26 | jeffspeff | loompek, clearglobalvars=no |
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| 08:25.01 | shital | Hello |
| 08:26.12 | loompek | jeffspeff thanks... i'll take a look |
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| 08:40.31 | tzafrir | shital, thebook |
| 08:40.35 | tzafrir | ~thebook |
| 08:40.36 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 08:40.52 | coldsteal | I would like to know if there is a way to install asterisk in a chrooted jail in openbsd |
| 08:42.17 | plik | coldsteal: should be pretty similar to installing in a freebsd jail -- there are instructions on the wiki |
| 08:42.39 | coldsteal | what wiki? |
| 08:42.56 | plik | www.voip-info.org |
| 08:43.38 | coldsteal | oh |
| 08:43.39 | coldsteal | lol |
| 08:43.43 | coldsteal | thanks |
| 08:43.56 | plik | np, good luck |
| 08:45.46 | coldsteal | I'm gathering information for a future project :) |
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| 08:50.15 | yang | erm the Speex codec, I cannot locate it in the Grandstream phones ? |
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| 09:10.59 | coldsteal | yang: http://www.google.com/products?q=grandstream&scoring=p&sa=N&start=10 |
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| 09:25.01 | shital | when i dial Asterisk PSTN no, from external phone i can here the rining sound but no response at the asterisk end, can any 1 tell me wats the probs? it displays the message http://pastebin.ca/1009146 |
| 09:26.07 | tzafrir | yang, do you use that phone in your LAN? if so: use alaw / ulaw |
| 09:26.59 | shital | my zapata.conf and extensions.conf files are http://pastebin.ca/1009150 |
| 09:27.16 | shital | please help me any one |
| 09:27.30 | tzafrir | shital, is Asterisk actually handling that port? |
| 09:28.02 | tzafrir | in 'cat /proc/zaptel/*', is that port "(In use)"? |
| 09:29.54 | shital | this is output of cat /proc/zaptel/ http://pastebin.ca/1009157 |
| 09:30.10 | shital | this is in use Sir |
| 09:30.51 | shital | tzafrir : i have TDM11B card with my system |
| 09:31.46 | tzafrir | You havein your dialplan "Answer" and right after that "Hangup". It should answer and right after that hang up |
| 09:32.09 | tzafrir | Is that what you want to do? |
| 09:32.37 | tzafrir | Don't you want to play something or whatever? |
| 09:32.56 | yang | tzafrir: no this is the point its not in LAN....We are trying to get a replacement for G729 as it requires licenses, but only a few phones would work with speex, should I use GSM codec instead? |
| 09:32.57 | tzafrir | exten => s,n,Playback(tt-monkeys) |
| 09:33.07 | shital | tzafrir : previously i put Echo() application in place of Hangup() but it was simply connecting but i did not get the sound at all |
| 09:34.00 | tzafrir | yang, gsm is also reasonable replacement |
| 09:34.22 | yang | tzafrir: 10 usd per channelfor 6729 is well , not that cheap |
| 09:34.35 | *** part/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 09:35.54 | yang | tzafrir: gsm is 13kbps in size if i understand right |
| 09:36.21 | tzafrir | yes |
| 09:36.30 | shital | tzfrir: i put exten => s,n,Playback(tt-monkeys) and checked it was simply ringing but did not connect |
| 09:37.28 | shital | the out put at CLI was http://pastebin.ca/1009162 |
| 09:37.41 | tzafrir | is busy... |
| 09:37.52 | shital | can u give any suggestions please? |
| 09:45.25 | shital | any 1 can help me?? |
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| 09:49.59 | krdian | shasta: do u have exten s in default context ? |
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| 09:51.23 | shital | krdian: my extensions.conf file has only one line exten => s,n,Playback(tt-monkeys) |
| 09:52.31 | krdian | shital: thats wrong, should be exten => s,1 |
| 09:52.50 | shital | ok i ll try just now |
| 09:52.54 | krdian | shital: next proprity can be n |
| 09:54.27 | shital | thanks i can here monkeys voice now |
| 09:54.35 | shital | can you just suggest me simple configuration for call making and receiving? |
| 09:55.44 | shital | krdian: what changes i have to do for this? |
| 09:56.04 | krdian | shital: not realy, everything depend on your needs |
| 09:56.57 | krdian | shital: lokk at voip-info.org there u can found examples |
| 09:57.21 | krdian | shital: and look at extensions.conf.sample |
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| 09:59.48 | shital | krdian: Sir for this i need to change extensions.conf file rt? |
| 10:02.01 | krdian | shital: if u want to dialin/dialout u have to build dialplan |
| 10:03.29 | krdian | shital: in extension.conf, ael-conf or database, depend on where u want to store information about call routing |
| 10:04.06 | shital | ok, |
| 10:04.08 | krdian | shital: have u read asterisk handbook ? |
| 10:04.53 | shital | yess sir, but i read it first time cant able to understand clearly |
| 10:06.08 | shital | i read Asterisk 2nd edition book sir |
| 10:06.31 | krdian | shital: ok, thats good :) |
| 10:07.15 | krdian | shital: as i remeber there is explained how u can build dialplan |
| 10:07.38 | shital | yaa sir i ll read it once again |
| 10:07.56 | shital | any way thanks a lot and lot |
| 10:09.43 | shital | <PROTECTED> |
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| 10:13.25 | krdian | shital: probably in /var/lib/asterik/sounds |
| 10:14.02 | shital | ok ll check |
| 10:14.55 | shital | no sir |
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| 10:19.31 | *** join/#asterisk adjohn (n=adjohn@i220-221-4-188.s05.a013.ap.plala.or.jp) |
| 10:19.33 | loompek | umm |
| 10:19.57 | loompek | is there an asterisk cmd that would set a variable from the system() output or something? |
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| 11:06.56 | pjezek | Hi, how can I remove global variable? When I do 'Set(GLOBAL(test)=)' from dialplan, value is cleared, but variable 'test' still exist, when I issue 'core show globals' |
| 11:13.53 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
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| 11:15.18 | *** join/#asterisk Silmaril (i=benoit@fbx.plessis.info) |
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| 11:15.24 | R0land | hello everyone |
| 11:15.39 | R0land | could some1 help me with my dialplan plz! |
| 11:15.39 | Silmaril | hello |
| 11:15.59 | R0land | ive set up my extensions.conf as well as sip.conf.. but when i try calling out it gives me an error in the debug cli "service unavailable" |
| 11:16.46 | R0land | http://www.pastebin.ca/1009215 |
| 11:18.38 | Silmaril | R0land: did "sip show peers" show your phones in registered state ? |
| 11:18.44 | R0land | let me chjeck |
| 11:19.00 | tzafrir | New Xorcom Rapid Live CD available: 1.2.0: http://updates.xorcom.com/iso/ |
| 11:19.02 | Delvar | R0land thats the phone '100' replying 503 to the invite |
| 11:19.09 | R0land | 100/100 192.168.0.111 D 5061 Unmonitored |
| 11:20.09 | tzafrir | asterisk 1.4.19.1 , zaptel 1.4.10.1, Freepbx (well, 2.3.1), Debian Etch 4.0.r3 |
| 11:20.09 | R0land | Silmaril thts wht it gives me .. |
| 11:20.10 | Delvar | could be a lot of things... check the phones configured corectly, check for firmware updates, check nat/conreinvite in sip.cof ... |
| 11:20.25 | R0land | Delvar ya i know but i cant see wheres the prob since i can call other sip extensions |
| 11:20.42 | Delvar | are the all teh same type og phone? |
| 11:20.55 | R0land | Delvar no.. all of them are a softphone except the |
| 11:20.59 | R0land | 100 extensions |
| 11:21.01 | R0land | which is my sipura |
| 11:21.06 | R0land | sipura's pstn line |
| 11:21.15 | R0land | if i dial 100.. it should direct me to my pstn line |
| 11:21.27 | Delvar | ah, its jsut the config on that sipura then |
| 11:21.35 | Delvar | make sure the dilplan is setup |
| 11:21.41 | Delvar | and passthough stuff... |
| 11:21.46 | R0land | hmm let me check |
| 11:21.46 | Delvar | imnot expert with sipura |
| 11:22.49 | R0land | well |
| 11:22.54 | R0land | one more question |
| 11:22.58 | Delvar | ya |
| 11:22.59 | R0land | if i wanna recieve calls |
| 11:23.03 | R0land | from a specific sip |
| 11:23.08 | R0land | so lets say i fixed my sipura.. |
| 11:23.21 | R0land | i want calls tht comes to sippura.. to b fwded to a specific extension |
| 11:23.24 | R0land | on my sip.conf |
| 11:23.26 | R0land | how can i do tht |
| 11:23.38 | Delvar | from PSTN ? |
| 11:24.02 | R0land | yes |
| 11:24.17 | R0land | asterisk <<-->> SPA<<--->> PSTn |
| 11:25.19 | Delvar | http://nerdvittles.com/index.php?p=65#spa3000 |
| 11:25.21 | Delvar | might help |
| 11:25.31 | steliosk | does anyone of way to remote reboot Grandream GXP2000 ? |
| 11:25.41 | Delvar | scrole down a bit and there is a quick guide for setiung up a spa 3000 |
| 11:25.43 | R0land | k Delvar let me check |
| 11:25.43 | steliosk | s/anyone/inyone know |
| 11:25.45 | Delvar | to work as a trunk |
| 11:36.11 | R0land | Delvar i did wht the guide aid.. its still having the same prob |
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| 12:07.12 | acxty | hi guys. I want to do a cycle with 5 different numbers, all of them cellphones. Call phone1, and if it don't answer call phone2 and so on until phone5. The problem is this one. My itsp use a password for outbound cellphone calls. I use D() option on Dial() to pass the password to it, but my server take that call as it was answered even if the cellphone wasn't answer. How can I avoid this to happen, an verify the cellphone ringing not my itsp authenticate m |
| 12:07.12 | acxty | ethod? |
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| 12:08.19 | ManxPower | acxty: make your carrier stop answering the call. |
| 12:08.53 | ManxPower | IF you can't do that then you have to design a complex system to prompt the callee for a DTMF digit to indicate accepting of the call. |
| 12:09.09 | acxty | ManxPower, I was thinking of that, but it will be quite difficult for that to happen hehehe |
| 12:09.35 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 12:09.36 | jblack | Either that, or have the cell phone user dial in and tell you he didn't answer the call. ;) |
| 12:10.15 | acxty | ManxPower, a complex system will be in extensions.conf? |
| 12:10.24 | acxty | jblack, hehehhehe :) |
| 12:10.38 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:15.23 | acxty | ManxPower, So my only way will be asking my itsp not use a authenticate method for cellphones |
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| 12:44.58 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff-emeritus/madkiss) |
| 12:45.11 | Madkiss | hi all. what's the most convenient way for editing the asterisk database? |
| 12:48.57 | jasonwoot | Madkiss, for what purpose? |
| 12:51.38 | [TK]D-Fender | Madkiss: I'd guess PERL. |
| 12:52.57 | Madkiss | jasonwoot: basically I am trying to do this: My companies works on two different floors. every department has its own door, and you can open that door right now via the voip-phones and a wrapper script. i want to define two groups in the database: floor 1 and floor 2. whenever a member of one group calls numer "95", i want the door on the floor to be opened where the user is at that time. |
| 12:53.34 | Madkiss | i.e. asterisk must detect "what number does user XYZ come from?" and then see "what floor is number 123?" and then do "okay, that's first floor, open door 1." |
| 12:54.20 | jasonwoot | That's an interesting application |
| 12:54.32 | jasonwoot | are all doors "unique"? |
| 12:54.47 | jasonwoot | meaning, do any of them share numbers, etc? |
| 12:55.04 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:55.04 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:55.25 | [TK]D-Fender | Madkiss: perhaps you should use MySQL for this. |
| 12:55.29 | Madkiss | jasonwoot: well, the door-opening is voodoo. right now, calling 95 leads to asterisk doing System() on a trigger-script. asterisk doesn't communicate directly with the door, so to say. |
| 13:01.17 | *** join/#asterisk R0land (n=Roland@193.227.191.91) |
| 13:01.22 | R0land | [TK]D-Fender hello :) |
| 13:02.18 | R0land | could someone help out with a sipura/asterisk setup please! |
| 13:02.37 | [TK]D-Fender | R0land: www.voxilla.com <- go read the forums. |
| 13:02.49 | R0land | already did |
| 13:02.55 | R0land | read 68 thread to b exact |
| 13:02.58 | jasonwoot | Madkiss: I don't have asterisk communicating 'directly' to any other systems either per se, so I'd have to defer... I'm using it similar to you with dialed strings kicking off scripts that write out files for other systems, etc |
| 13:02.58 | Madkiss | so I guess all I would need to do for the moment is defining two caller groups in the asterisk db, i just have no idea how that actually works |
| 13:03.22 | Madkiss | jasonwoot: my problem for now seems to be that i have not used the asterisk db at all so far |
| 13:03.53 | [TK]D-Fender | Madkiss: "core show function DB" |
| 13:04.22 | Madkiss | [TK]D-Fender: on the CLI? |
| 13:04.38 | [TK]D-Fender | Madkiss: yes |
| 13:05.14 | Madkiss | what's the magic to get a list of all existing entries? |
| 13:10.59 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:11.13 | *** join/#asterisk atomspace (n=huh@adsl-226-14-105.bhm.bellsouth.net) |
| 13:11.32 | tzafrir | Madkiss, tab completion |
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| 13:11.50 | atomspace | since asterisk handles voicemail, call waiting, and such... is the only "service" necessary to get from your phone provider callID ? |
| 13:11.58 | atomspace | ---that does have to come from your service provider, right? |
| 13:11.58 | tzafrir | core show function <tab><tab> |
| 13:12.58 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 13:13.14 | Madkiss | tzafrir: i mean all existing entries in the db |
| 13:13.28 | tzafrir | database show |
| 13:14.10 | [TK]D-Fender | atomspace: yup |
| 13:14.18 | [TK]D-Fender | atomspace: and CDS |
| 13:14.20 | atomspace | sweet, thanks fender |
| 13:14.24 | atomspace | CDS? |
| 13:14.26 | atomspace | ~cds |
| 13:14.38 | [TK]D-Fender | atomspace: Call Disconnect Supervision |
| 13:14.51 | [TK]D-Fender | atomspace: a signal to indicate that the calling party has hung up |
| 13:14.53 | atomspace | hmmmm, I'm not familiar with that. I'll check it out. |
| 13:14.56 | atomspace | OH |
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| 13:15.05 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 13:15.14 | tzafrir | jbot, cds is Call Disconnect Supervision |
| 13:15.14 | jbot | tzafrir: okay |
| 13:15.31 | atomspace | Fender, will CDS make it so that my asterisk doesn't still ring my voip phones for 8 seconds after the outside caller has hung up? |
| 13:15.46 | atomspace | ----is that what CDS is for? |
| 13:16.23 | jasonwoot | could he use wrapup for that? |
| 13:16.30 | JT | [TK]D-Fender: s/calling/called/ |
| 13:16.52 | [TK]D-Fender | ~cds |
| 13:16.52 | jbot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 13:17.05 | atomspace | doh, "~cds" didn't work when I did it. thanks |
| 13:17.38 | [TK]D-Fender | atomspace: it has just been programmed |
| 13:17.59 | Madkiss | would "database put firstfloor 25 1" create a database-entry "/firstfloor/25" with the value 1? |
| 13:18.13 | [TK]D-Fender | tzafrir : Don't worry, I've taken the task of filling the holes :) |
| 13:18.24 | [TK]D-Fender | Madkiss: Correct. |
| 13:18.42 | tzafrir | ~cds |
| 13:18.43 | jbot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 13:19.06 | atomspace | after getting CDS from my service provider, what needs to be setup, within asterisk, to work with CDS? |
| 13:19.29 | [TK]D-Fender | atomspace: its automatic with Kewlstart |
| 13:19.38 | atomspace | crap, what is kewlstart? |
| 13:19.40 | atomspace | ~kewlstart |
| 13:19.48 | [TK]D-Fender | atomspace: a zaptel signalling mode |
| 13:19.51 | Madkiss | [TK]D-Fender: and in my dialplan, how do I check whether "25" is in "firstfloor"? |
| 13:20.20 | [TK]D-Fender | Madkiss: "core show function DB" , "core show application gotoif" |
| 13:20.44 | atomspace | I used "pbx in a flash" --- asterisk with freepbx. I don't remember seeing anything about kewlflash, so I'm not sure if I have that installed. I'll look into it though. Thanks for your help folks |
| 13:21.51 | *** join/#asterisk AfricAnne (n=AfricAnn@41.204.128.40) |
| 13:21.54 | [TK]D-Fender | atomspace: its just your "signalling" line in zapata.conf |
| 13:22.04 | [TK]D-Fender | atomspace: and yes FreePBX lets you set that |
| 13:22.19 | Madkiss | that sounds all very good. |
| 13:22.22 | Madkiss | :) |
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| 13:40.55 | Madkiss | in which variable is the number by the person that is calling usually stored? |
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| 13:42.50 | [TK]D-Fender | Madkiss: which "number"? You should think on that answer..... |
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| 13:43.39 | flohack | Hi! Can someone tell me how to disable the callerid on outgoing calls through a PRI? |
| 13:43.54 | Madkiss | [TK]D-Fender: the internal one. |
| 13:44.15 | flohack | hidecallerid=yes in zapata.conf has no effect and I can't figure out how I would use *67 as suggested in the documentation of zapata.conf |
| 13:44.25 | [TK]D-Fender | Madkiss: Nice & meaningless answer. EVERYTHING is telecom and Asterisk is a "number". what KIND <--- |
| 13:45.27 | Madkiss | [TK]D-Fender: the user-id of the calling phone. |
| 13:45.29 | [TK]D-Fender | flohack: core show application SetCallerPres |
| 13:45.57 | [TK]D-Fender | Madkiss: Go look at what CHANNEL variables are available while processing a call... |
| 13:46.19 | Madkiss | hu? |
| 13:46.43 | flohack | [TK]D-Fender: Forgive my ignorance... |
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| 13:47.11 | flohack | [TK]D-Fender: I'll try that in a minute, as soon as our server is up again |
| 13:47.16 | Madkiss | is there an elsif command in extensions.ael? |
| 13:47.21 | Madkiss | or is it "elseif"? or "else if"? |
| 13:47.22 | [TK]D-Fender | Madkiss: go read channelvariables.txt in your docs folder and the list of "core show functions". |
| 13:47.41 | [TK]D-Fender | Madkiss: almost nobody here uses AEL |
| 13:47.52 | [TK]D-Fender | Madkiss: And I'm sure there is a way for everything. |
| 13:48.48 | codefreeze-lap | Madkiss: use "else if". [TK]D-Fender wishes AEL would go away... take him with a grain of salt on this issue ;-) |
| 13:50.25 | [TK]D-Fender | codefreeze-lap: only because it adds no unique functionality to *. its "point of profitability" is somewhat narrow. Now if "basic" extensions logic were a buried and internal thing, then that would be another matter. |
| 13:51.06 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
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| 13:51.17 | [TK]D-Fender | codefreeze-lap: and no need for any salt on that one, everything I said was completely true. Almost no-one here uses it. |
| 13:52.41 | jblack | codefreeze-lap: How did you come into the belief that tk wishes AEL would go away? |
| 13:53.00 | codefreeze-lap | [TK]D-Fender: Sorry, but that's hard/impossible to quantify. People are using it, neither of us can assert meaningfully how many. |
| 13:53.04 | jblack | He might, he might not, but how did you form an opinion about his opinion. |
| 13:53.05 | codefreeze-lap | jblack: see above. |
| 13:53.22 | [TK]D-Fender | jblack: Ummm... actually in a manner of speaking I DO. |
| 13:53.29 | jblack | So... If I say the sky is blue, then I must hate horses? |
| 13:53.52 | jblack | [TK]D-Fender: Sure. maybe you do, maybe you don't. but I don't see where you stated your personal opinion. I saw you state facts. |
| 13:53.54 | codefreeze-lap | s/above/below/ |
| 13:54.19 | [TK]D-Fender | jblack: While I haven't said it so directly till now, my view and reasoning have been rather clear. |
| 13:54.20 | jblack | I guess it's a moot poitn now. |
| 13:54.51 | [TK]D-Fender | jblack: and I have been more to the point on it before, its just that we're not always here at the same time. |
| 13:54.54 | [TK]D-Fender | jblack: yup |
| 13:55.00 | jblack | It's like if I had said "Diamonds are rare", that he would extrapolate that to mean I hate diamonds. |
| 13:55.08 | JayTee52 | damn, I hate coming in on the tail end of an interesting conversation |
| 13:55.08 | jblack | Oh, I know you dislike ael. |
| 13:55.58 | jblack | Heh. I guess I am trolling slightly. php with smarty, with ldap, with whip cream on top is annoying |
| 13:56.06 | [TK]D-Fender | codefreeze-lap: I will say that AEL help and claims of usage in here is rather slim. True it is hard to say "at large", but my money is on the belief that it is a very small minority |
| 13:56.08 | jblack | sorry codefreeze-lap |
| 13:57.41 | codefreeze-lap | [TK]D-Fender: You may or may not be correct, I cannot say, but it's young, and will grow with time. Sorry. |
| 13:58.05 | jblack | Ahhhhhhhhhhhhhhh... hhhhhhhhhhhh.. hhhhhh |
| 13:58.46 | jblack | hhhhh. ;) |
| 13:58.57 | codefreeze-lap | jblack: lol |
| 13:59.21 | [TK]D-Fender | codefreeze-lap: No bad blood here. I personally wish BOTH were ripped out and something new done altogether :) |
| 13:59.36 | JayTee52 | I cracked up last nite when someone was in here looking for help with Red Hat 4 (not RHEL) and *now. |
| 14:00.14 | codefreeze-lap | [TK]D-Fender: Now, that's an interesting thought... feel free to elaborate! |
| 14:01.30 | *** join/#asterisk sysadmin-lb22 (n=asdf@mail.splendor.net) |
| 14:01.48 | sysadmin-lb22 | does astersik support conferences using broadband codecs? |
| 14:01.57 | [TK]D-Fender | codefreeze-lap: All sorts of C-like parsers out there, just shove one of those directly into * and do away with extensions.conf altogether. This would mean declared variable types, etc and also do away with this "white-space counts", BS, delimiters mayhem, etc. |
| 14:02.05 | JayTee52 | according to "the book" when includes are used in a context * will try to match on current context first and then on the includes in the order they are listed. So if I have a context with only two includes and nothing else it will try to match the first one and then the second? |
| 14:02.13 | [TK]D-Fender | codefreeze-lap: Tricky bit is finding the engine that would fit *'s licensing req's |
| 14:02.24 | *** join/#asterisk PaulQ (n=PaulQ@72.29.76.254) |
| 14:02.38 | PaulQ | Really strange problem with a TE110P |
| 14:02.40 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 14:02.41 | x86 | gah, I _hate_ how Asterisk randomly shows IAX2 peers as unreachable, when they certainly ARE reachable |
| 14:03.04 | PaulQ | I get " Unable to specify channel 1: Device or resource busy" randomly |
| 14:03.14 | x86 | 37ms latency, 0% packet loss, but somehow it's "unreachable" |
| 14:03.44 | x86 | one side is connected just fine, the other still says "unreachable" |
| 14:03.50 | x86 | gah wtf |
| 14:03.52 | dominic1 | can I send the snom desktopmessage with asterisk? |
| 14:05.06 | Madkiss | codefreeze-lap: do you happen to know what in extensions.ael the variable is that contains the username of the phone that makes a call? |
| 14:05.26 | codefreeze-lap | [TK]D-Fender: Actually, I've got something similar to that in mind. In AEL3, I'd like to get rid of the ${} and $[] notation and make it a single uniform grammar, on a parse-tree, so the expressions aren't re-parsed every priority. It would be a little more C-like. |
| 14:06.13 | codefreeze-lap | Madkiss: wouldn't that kind of thing be in the CID info? |
| 14:07.15 | ddunavant | just want to make sure, is there any way to, in one statement, run a macro twice at the same time? |
| 14:07.21 | PaulQ | This is driving me nuts |
| 14:07.31 | x86 | any ideas how to force asterisk to show this peer as reachable like it really is? |
| 14:07.58 | [TK]D-Fender | codefreeze-lap: Thing is is thats jsut another "patch". You are still limited by *'s base. Thats what has to go. our current dumb parser doesn't separate strings from variables, deal with types, etc. yeah it was "cute" at first, easy to format, but this comes back to bite you in the ass, HARD. |
| 14:08.29 | x86 | gah |
| 14:09.17 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 14:09.38 | codefreeze-lap | Well, in AEL3, you'll have to wrap strings in "". ... no more automatic formatting. But we can supply some concat operators. A little more clumsy, but way more clear. |
| 14:09.51 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.205.41) |
| 14:10.32 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:10.33 | [TK]D-Fender | codefreeze-lap: Yes, a better "patch". but I'd rather see ALL of it bite the dust. |
| 14:10.34 | PaulQ | This is driving me nuts!! |
| 14:10.44 | PaulQ | <PROTECTED> |
| 14:10.47 | x86 | config.c: parse error: No category context for line 1 of /etc/asterisk/iax.conf |
| 14:10.53 | PaulQ | <PROTECTED> |
| 14:11.00 | PaulQ | <PROTECTED> |
| 14:11.03 | PaulQ | I feel stupid! |
| 14:11.32 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 14:11.36 | x86 | [2008-05-06 09:11:26] ERROR[12736] chan_iax2.c: Unable to load config iax.conf |
| 14:11.39 | x86 | gah, wtf |
| 14:11.43 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 14:11.53 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 14:14.37 | x86 | [2008-05-06 09:14:27] WARNING[8826] channel.c: No channel type registered for 'IAX2' |
| 14:14.40 | x86 | [2008-05-06 09:14:27] WARNING[8826] app_dial.c: Unable to create channel of type 'IAX2' (cause 66 - Channel not implemented) |
| 14:14.46 | x86 | but chan_iax2.so is loaded |
| 14:14.55 | x86 | wtf?! |
| 14:16.42 | codefreeze-lap | x86: look over your iax.conf file; something is wrong on line one, and it's not loading because of it |
| 14:17.08 | PaulQ | What is wrong with me! :( |
| 14:18.38 | PaulQ | Maybe libpri 1.6 was bad idea |
| 14:19.01 | codefreeze-lap | [TK]D-Fender: So, it would seem, you'd rather just plain use an interpreted, declarative language, and have some sort of master API from asterisk to call the apps, etc. They would still be executed in terms of the channel... |
| 14:19.52 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 14:20.19 | [TK]D-Fender | codefreeze-lap: Guess that pretty much says it. With things being typed, true structured programming, etc, you could REALLY validate code as well. |
| 14:20.44 | x86 | codefreeze-lap: i fixed it... restarted asterisk completely |
| 14:20.52 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:20.57 | x86 | codefreeze-lap: there wasn't anything wrong with the conf -- it's been running fine for 5 months now |
| 14:21.56 | [TK]D-Fender | codefreeze-lap: There IS a bigger downside in this though. Can't rewally DB this kind of dialplan. |
| 14:22.17 | codefreeze-lap | x86: interesting, in a weird sort of way. |
| 14:22.42 | x86 | codefreeze-lap: seems I have to restart asterisk at least once a week because of little oddities like this |
| 14:22.55 | [TK]D-Fender | codefreeze-lap: Perhaps what * needs as a base, beyond just extensions.conf would be variable types first, and a better reference engine. That could allow it to remain "line based". |
| 14:23.05 | x86 | codefreeze-lap: of course, I use astersk in a 7-office outbound call center, so it gets a fair amount of traffic going through it |
| 14:23.21 | x86 | codefreeze-lap: I'm starting to think that perhaps Asterisk wasn't the way to go for a call center :( |
| 14:23.31 | codefreeze-lap | [TK]D-Fender: Well, do you **really** want to DB code anyway... The extensions, yes, sure, but the code? |
| 14:23.36 | Corydon76-dig | [TK]D-Fender: it has types. All variables are strings. |
| 14:23.36 | krdian | x86: me too :( |
| 14:23.48 | jasonwoot | x86: ha, ha ha, ha, sigh.... |
| 14:24.03 | jasonwoot | good times, good times |
| 14:24.07 | [TK]D-Fender | Corydon76-dig: If they're all "strings", then you shouldn't pluralize "type" like that :p |
| 14:24.28 | Corydon76-dig | Fair enough |
| 14:24.35 | x86 | of course, I am using 1.4.12.1 |
| 14:24.42 | krdian | x86: i have to restart * every morning |
| 14:24.46 | x86 | I'm going to do a mass upgrade to 1.4.19.1 |
| 14:25.00 | x86 | krdian: so at least I'm a bit more stable than you ;) |
| 14:25.05 | codefreeze-lap | x86: You have two choices. switch to something else, or help us get the bugs out; try to catalog them and make them easily reproducible... don't know which will be easier! |
| 14:25.21 | x86 | easier choice would simply replace it |
| 14:25.31 | krdian | x86: well, but i have to use f** centos on 64bit |
| 14:25.33 | x86 | i don't know what causes these little mundane issues |
| 14:25.43 | x86 | krdian: that sucks |
| 14:25.59 | x86 | codefreeze-lap: I'm not sure how to reproduce these random bugs |
| 14:26.04 | krdian | x86: i still have memory leak problem |
| 14:26.18 | codefreeze-lap | Corydon76-dig: Asterisk is the SNOBOL of telephony.... in an offhand, weird sort of way.... |
| 14:27.22 | krdian | x86: after ~8k calls my * in call center consuming ~1GB memory |
| 14:27.42 | x86 | codefreeze-lap: step 1: run 50,000 outbound calls through a single Asterisk server within a 5 day window, step 2: wait as shit breaks, step 3: get made fun of (or simply ignored) for asking for assistance |
| 14:28.02 | x86 | krdian: no memory leak here... actually memory usage is great for me |
| 14:28.46 | x86 | I'm only using 3MB of swap :P |
| 14:29.13 | krdian | x86: have u set any limits ? |
| 14:29.19 | x86 | nope |
| 14:29.25 | x86 | running 1.4.12.1 |
| 14:29.44 | Corydon76-dig | codefreeze-lap: except that Asterisk is easier to read than SNOBOL |
| 14:29.46 | *** part/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
| 14:29.51 | krdian | x86: linux ? or bsd ? |
| 14:30.00 | x86 | linux |
| 14:30.02 | x86 | Gentoo |
| 14:30.30 | krdian | x86: heh, i have to force my IT departament to change distro |
| 14:30.32 | codefreeze-lap | Corydon76-dig: Amen. Until you start getting really thick with embedded ${} and $[] expressions and escapes.... |
| 14:30.36 | *** join/#asterisk R0land (n=Roland@193.227.191.91) |
| 14:30.44 | R0land | hello all |
| 14:30.47 | x86 | krdian: the only things I have running on the box are Asterisk, Apache (for reporting), and iaxmodem |
| 14:30.49 | R0land | [TK]D-Fender hello once again |
| 14:31.23 | *** join/#asterisk justdave_ (n=dave@unaffiliated/justdave) |
| 14:31.31 | R0land | [TK]D-Fender i followed the how to guide existing in tht site to the letter..! and the same issue is still the same.. my pstn sip extensions gives me 502 service unavailable" |
| 14:32.09 | *** join/#asterisk saggating_ (n=jason@host86-133-53-159.range86-133.btcentralplus.com) |
| 14:32.27 | x86 | krdian: HP DL380 G5 w/ (1) quad-core xeon, 4GB of total memory (although I've got it setup to mirror the memory in hardware, basically (2) 2GB sets running RAID-1, so only 2GB is available to the OS), (2) 36GB 15,000RPM SAS hard drives |
| 14:33.04 | x86 | krdian: for the most part, it runs perfectly fine, just these little random quirky issues that I have to deal with from time to time |
| 14:33.11 | Corydon76-dig | codefreeze-lap: it's still easier to read than SNOBOL |
| 14:34.20 | Silmaril | I was experiencing linking two asterisk boxes with IAX2 trunk yesterday and i it was working quite well, except for the sendurl() part |
| 14:34.37 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 14:34.42 | codefreeze-lap | Corydon76-dig: OK, OK! I give! But Snobol **does** have some pretty cool pattern matching facilities.:) |
| 14:34.55 | Silmaril | the first asterisk box was dialing the second using Dial(IAX2/other/<EXTEN>,,,'http://Url') |
| 14:35.16 | Silmaril | but i wasn't able to get the information on the second box |
| 14:35.18 | Corydon76-dig | The Asterisk dialplan actually compares favorably to Tcl |
| 14:35.57 | Silmaril | i saw the HTML Frame being trasnferer between them, but the question is: does asterisk even bother to handle this frame ? |
| 14:37.19 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 14:38.23 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 14:38.33 | Corydon76-dig | Silmaril: why wouldn't it? |
| 14:41.17 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:41.27 | *** join/#asterisk grantm (n=grant@66.29.180.194.static.utahbroadband.com) |
| 14:41.43 | saggating_ | Has anyone experienced kernel errors when using MiSDN with Asterisk 1.4.19 , "mISDN_rdata: rport queue overflow (after 500ms!) 256/256" |
| 14:44.11 | vader-- | Are you guys doing any checks with nagios against your asterisk boxes? |
| 14:44.42 | flohack | Has someone experience with red-fone bridges? We keep getting yellow alarms on our channels while calls are in progress. All calls are dropped and the channels are available for calling again a second later |
| 14:47.54 | Silmaril | Corydon76-dig: what i want to say is that i saw the code to send the HTML frame, the code to dump channel which include the display of HTML frame, but i wasn't able to find |
| 14:48.19 | Silmaril | something that would parse the frame and store it in some variable for example |
| 14:48.29 | Kobaz | flohack: where's your incoming line comming from? |
| 14:48.31 | flohack | We have eliminated all possible interrupt problems (the ethernet card connected to the bridge has its own interrupt line). I've posted a log here: http://pastebin.com/m548a7692 |
| 14:48.42 | flohack | Kobaz: It's a E1 PRI |
| 14:48.53 | Kobaz | flohack: from the telco or from another box |
| 14:48.57 | flohack | telco |
| 14:49.01 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:49.01 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:49.13 | acxty | I want to make a cycle with some cellphones. For example. I have 3 phones. I want co Dial() phone1 with a timeout 20 seconds, if it doesn't answer Dial() phone2 and so on. The problem is that my itsp use a password to give cellphone outbound access. So even if phone1 doesn't answer, my asterisk server guess it because my itsp server answer asking for the password. I send the password using D() in Dial(). What I want to do is to verify if the cellphone an |
| 14:49.13 | acxty | swer, omiting the password query on the itsp server side |
| 14:49.16 | Kobaz | has the telco ruled out any issues on their end? can you get the counters on their side? |
| 14:49.53 | Kobaz | have you checked cabling, etc? |
| 14:50.18 | flohack | Kobaz: According to them the line is fine (we had a technician on site yesterday). I can post a protocol trace from them if you fancy one. |
| 14:50.54 | flohack | Kobaz: We have replaced the cable connecting the bridge with the telco box with a <1m Cat5 cable |
| 14:51.04 | Kobaz | what line card are you using on the asterisk side? |
| 14:51.36 | Kobaz | this looks exactly like a problem i was having with a rhino e1/t1 card |
| 14:51.45 | Kobaz | buggy drivers |
| 14:51.48 | flohack | Kobaz: There is no card. We are using a red-fone bridge (which contains a digium card AFAIK) and have it connected via TDMoE |
| 14:52.01 | Kobaz | oh i see |
| 14:52.39 | Kobaz | do you have a pri simulator or another box with a pri card in it? |
| 14:52.46 | flohack | Kobaz: Their support claims it's a problem with the line, telco claims it's our telephone system...I'm stuck in the middle :-) |
| 14:53.15 | flohack | Kobaz: Unfortunately not. Maybe a loopback cable? The bridge has 2 E1 ports. |
| 14:53.24 | acxty | may someone help me with that? |
| 14:53.50 | Kobaz | crossover/loopback would be a good test |
| 14:54.01 | acxty | I already contact my itsp and they told me that for policy reasons they cannot take the password authenticate for cellphones outbounds |
| 14:54.17 | *** part/#asterisk saggating_ (n=jason@host86-133-53-159.range86-133.btcentralplus.com) |
| 14:54.18 | Kobaz | if you can set up a crossover from port1 to port2 and make some calls, that would rule out local hardware as the source of the problem |
| 14:54.18 | flohack | How would I setup that on the asterisk side? |
| 14:54.30 | Kobaz | which then leaves that to the configuration and/or wireing and/or telco |
| 14:54.56 | flohack | Kobaz: Can I actually call myself using the loopback cable, or can I just try to establish the call? |
| 14:55.30 | Kobaz | i've only been able to test hardware level functionality with a loopback with asterisk |
| 14:55.43 | Kobaz | it has a problem figuring out the signalling |
| 14:55.57 | Kobaz | easiest would be a crossover from a to b |
| 14:56.10 | flohack | Hmm...that would probably not tell us much as the problem only appears under load |
| 14:56.22 | Kobaz | flohack: like, you set up the port to be cpe or whatever, and then asterisk complains that both sides want to use cpe |
| 14:56.44 | flohack | maybe one pri_cpe and one pri_net? |
| 14:56.54 | Kobaz | flohack: sounds to me like a platform problem with the e1/t1 |
| 14:57.13 | flohack | Kobaz: What do your mean with platform? |
| 14:57.21 | Kobaz | flohack: as far as i can tell you can't set up the port to handle both cpe and net for loopback purposed, unless someone else knows how |
| 14:57.24 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 14:57.35 | Kobaz | flohack: digium card/redphone thinger |
| 14:57.55 | Kobaz | easiest way is to do port1 cpe port2 net, and start making calls |
| 14:58.03 | Kobaz | here's the problem i had with the rhino |
| 14:58.20 | Kobaz | you light up the first channel, everything is fine |
| 14:58.21 | [TK]D-Fender | acxty: Trying to round-robin OUTBOUND calls is painful at best and usually not functionally viable |
| 14:58.30 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 14:58.39 | Kobaz | fire up calls on three more and you start losing data on the first channel |
| 14:58.44 | acxty | [TK]D-Fender, what do you recomend on doing? |
| 14:58.50 | Kobaz | load up 4 more calls and the d channel goes byebye |
| 14:59.04 | Kobaz | i tracked it down to a firmware/driver bug |
| 14:59.27 | flohack | Kobaz: Sounds similar. We have bad sound quality on some channels before the complete loss of signalling |
| 14:59.30 | Kobaz | sounds like it's your redphone thing causing problems |
| 14:59.31 | [TK]D-Fender | acxty: You are dialing out to cell's. First of all.. they are ANSWERING (where you have to enter your PIN). So your diea ofgoing from one to the other is pretty much dead. |
| 15:00.28 | Kobaz | flohack: so, if you can... do a crossover from port1 to port2, and start making calls, if the problem still crops up, it's the platform |
| 15:00.32 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 15:00.49 | acxty | [TK]D-Fender, yes that is the part that give me the problem. I was wondering if there is a way to verify dtmf or something like that |
| 15:00.54 | flohack | Kobaz: Ok, I'll fix a crossover E1 and make up a config... |
| 15:01.06 | [TK]D-Fender | acxty: No, you're pretty flat-out screwed |
| 15:01.19 | flohack | Kobaz: Thanks a lot! |
| 15:01.21 | Kobaz | np |
| 15:02.23 | acxty | [TK]D-Fender, so the only way will be to convice the itsp that take the confirmation password hehehe |
| 15:03.29 | Corydon76-dig | Silmaril: correct. We don't actually process the HTML frames, we just pass them onto the other side of the bridge |
| 15:04.17 | acxty | [TK]D-Fender, thanks again TK ;) |
| 15:04.42 | *** join/#asterisk _gm (n=mustafa@117.20.28.50) |
| 15:04.55 | _gm | hi all, |
| 15:05.12 | _gm | i m trying to configure a sip trunk, but no sound when call is established. |
| 15:05.26 | _gm | if i configure sip account on my phone it works fine |
| 15:05.37 | _gm | i have set nat=yes |
| 15:05.46 | [TK]D-Fender | _gm: Read up : |
| 15:05.47 | [TK]D-Fender | ~sipnat |
| 15:05.48 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:05.49 | [TK]D-Fender | ^^^^^^^^ |
| 15:11.52 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 15:15.38 | acxty | I which log can I find all the register attempts that had do to a sip server? |
| 15:16.54 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:19.09 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:21.35 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 15:25.50 | whymarkwh | hi |
| 15:26.09 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 15:26.12 | *** join/#asterisk R0land (n=Roland@193.227.191.91) |
| 15:26.17 | R0land | hello everyone |
| 15:26.24 | R0land | could someone help me solve this problem please: http://www.pastebin.ca/1009413 |
| 15:26.34 | *** join/#asterisk igascream (n=igascrea@80.179.192.178.cable.012.net.il) |
| 15:26.56 | *** join/#asterisk ftp3 (n=none@pool-71-117-212-7.ptldor.fios.verizon.net) |
| 15:28.05 | whymarkwh | cant open it pastebin |
| 15:28.36 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:31.06 | *** part/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 15:31.16 | s0ck | is giving up all hope of getting this aex800 to work :o |
| 15:33.23 | *** join/#asterisk gaetronik (n=gaetan@164.77.74.18) |
| 15:33.26 | gaetronik | Hi there |
| 15:34.01 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:34.23 | gaetronik | my asterisk box decide time to time to do what appears to be a reload |
| 15:34.43 | gaetronik | is this a feature calle I know what i do illy human? |
| 15:34.45 | gaetronik | +s |
| 15:35.07 | bbryant | ~book |
| 15:35.08 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:35.25 | gaetronik | is this for me bbryant ? |
| 15:35.31 | bbryant | it is for me |
| 15:35.34 | gaetronik | ok |
| 15:35.36 | bbryant | ;) |
| 15:36.37 | gaetronik | i'm turning fool with this behaviour because it close the manager socket and put an application which connect to AMI in a strange state |
| 15:36.37 | Silmaril | Corydon76-dig: in my case it's of no use, the inbound IAX2 is sent to a Queue and then to an IAX2 phone and somewhere the html frame is lost |
| 15:38.52 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 15:40.23 | *** join/#asterisk hguerra (n=hguerra@205.211.251.137) |
| 15:40.37 | *** part/#asterisk hguerra (n=hguerra@205.211.251.137) |
| 15:45.11 | *** join/#asterisk mintee (n=mintee@72-165-177-94.dia.static.qwest.net) |
| 15:45.42 | mintee | ${DIALEDPEERNUMBER} nor ${DNID} is the variables for the number dialed... what else is there? |
| 15:45.48 | mintee | i need to capture the dialed number |
| 15:46.05 | Strom_C | ${EXTEN} perhaps? |
| 15:46.24 | *** join/#asterisk physique (n=phy@unaffiliated/physique) |
| 15:46.30 | *** part/#asterisk physique (n=phy@unaffiliated/physique) |
| 15:46.40 | mintee | nah, too late for that.... I guess I could store it from the inital context |
| 15:47.11 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:48.06 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:50.16 | mintee | is there a way to dump all the variables to see what's stored and what's not? |
| 15:52.03 | Silmaril | mintee: that's something i was looking after, but until no no luck |
| 15:52.18 | mintee | heh, figures |
| 15:53.39 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 15:54.27 | *** join/#asterisk Telamon (n=telamon@blk-89-240-175.eastlink.ca) |
| 15:55.29 | Telamon | Is there any documentation on how to read RTP-stats? I'm getting things like Jitter 0.0578 Transit: 1.3778 but I'm not sure what's normal, so I don't know if that's bad (the jitter looks bad, dunno about transit) |
| 15:58.40 | draygon-w | pls |
| 16:00.46 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 16:00.56 | *** join/#asterisk RunicP (n=nospam@199-117-163-66.dia.static.qwest.net) |
| 16:05.05 | *** join/#asterisk railsmunky (n=nick@82-70-165-142.dsl.in-addr.zen.co.uk) |
| 16:05.41 | railsmunky | Hey people - i'm just doing a new asterisk setup. Anyone any idea why i'm getting "No matching peer found" |
| 16:06.05 | railsmunky | just a standard register from an aastra |
| 16:07.22 | railsmunky | everything in my sip.conf seems ok |
| 16:07.28 | Strom_C | my guess would be that you didn't get the peer name right |
| 16:07.30 | *** part/#asterisk magumbade (n=magumbad@p5497DAC6.dip.t-dialin.net) |
| 16:07.43 | railsmunky | peer being? :) sorry i'm new |
| 16:07.53 | *** join/#asterisk PepOSX (n=angeldav@200.90.123.15) |
| 16:09.25 | *** join/#asterisk rushowr (n=rushowr@adsl-70-244-238-106.dsl.spfdmo.sbcglobal.net) |
| 16:09.29 | rushowr | hello all |
| 16:09.52 | Strom_C | railsmunky: you may want to read "the book" |
| 16:09.53 | Strom_C | ~book |
| 16:09.54 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 16:10.01 | *** join/#asterisk dawebber (n=dawebber@209.136.161.135) |
| 16:10.17 | railsmunky | yeah reading - just had this working fine on another system |
| 16:10.24 | railsmunky | i don't really get why it's doing it. |
| 16:10.39 | Strom_C | carefully double-check your work |
| 16:10.47 | railsmunky | my type is friend which as far as i understand covers peer and user |
| 16:11.05 | Strom_C | if it's really "the same" as the working configuration then it should work :) |
| 16:11.06 | rushowr | if anyone has a moment, I'm trying to debug a particularly annoying dropped call issue on a T1 (EM_W), using a TE405...never had this problem before..... |
| 16:11.18 | Strom_C | rushowr: what's the problem |
| 16:12.00 | rushowr | Strom_C: Calls are randomly being dropped, seems to happen in spurts, have full verbose (level 3) output to file but all I see is "Hungup Zap/(channum)" |
| 16:12.13 | rushowr | mixed environment, sales floor using Zap phones, Admins using SIP |
| 16:12.18 | rushowr | happens on both |
| 16:12.24 | jasonwoot | rushowr: are you on 1.4.18? |
| 16:12.34 | rushowr | 1.4.19 |
| 16:12.59 | Strom_C | rushowr: is there a consistent pattern to who's on the other end of the call when the call is dropped? |
| 16:12.59 | rob0 | can sip debug be set/unset for CIDR netblocks? I don't want debug on my ATAs in the LAN, just for remote hosts, and probably not all of those at the same time. |
| 16:13.20 | RunicP | Hello I have a dialplan...works great for both out and inbound calls. I have added a simple ivr menu on one of my inbound numbers. Now when I call my extension to make outgoing calls it says its not a valid extension. |
| 16:13.39 | RunicP | I am going to pasebin just a second |
| 16:13.42 | rushowr | no, nothign that points to a patter |
| 16:13.44 | rushowr | pattern* |
| 16:14.00 | Strom_C | hmm |
| 16:14.08 | Strom_C | have you called digium support? |
| 16:14.37 | rushowr | Strom: yes I have, they told me our previous TE405 was bad and we had already ordered a replacement. The new card is in but having the same issues |
| 16:14.47 | Strom_C | rushowr: call them again |
| 16:15.17 | jasonwoot | rushowr: I can tell you this- when I upgraded to 1.4.18 I started having the strangest ZAP problems with T1s |
| 16:15.30 | rushowr | jason: really? |
| 16:15.30 | jasonwoot | rolled back to 1.4.16 and they went away |
| 16:15.35 | rushowr | Strom: I'll do that |
| 16:15.52 | rushowr | thanks all, I'll try the 1.4.16 rollback and see if that helps |
| 16:15.57 | dawebber | all: do anyone knows whether channel locking bug fix made it into 1.4.19? |
| 16:15.57 | jasonwoot | rushowr: I know that's a "quit-and-run-away" option, but... |
| 16:15.59 | rushowr | guess I'll need to put in a bug |
| 16:16.27 | rushowr | jason: for the client, they just want it fixed, I can investigate problems on my own test server, but they're a telemarketing company |
| 16:16.42 | Strom_C | heh, telemarketing....they deserve it ;) |
| 16:16.48 | rushowr | yeah I know I know.... :( |
| 16:17.23 | dawebber | I'm still trying to figure out why my "Ring requested on channel x/y already in use or previously requested on span z. Attempting to renegotiating channel." didn't go away after the upgrade. Does anyone else have a problem like this? |
| 16:18.36 | jasonwoot | haha |
| 16:18.40 | rushowr | alrght mates, I'll be back, gotta go try this fix |
| 16:18.45 | rushowr | jason: 1.4.16 right? |
| 16:19.59 | jasonwoot | rushowr: on 1.4.16.2, I'm run full debug and not seeing some of the ZAP messages from 1.4.18 that I suspect were the cause |
| 16:21.23 | *** join/#asterisk af_ (n=getsmart@88-149-230-0.dynamic.ngi.it) |
| 16:24.52 | dawebber | so, does anyone have any sort of Zaptel channel-locking issues with 1.4.19? |
| 16:25.17 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 16:25.20 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:25.59 | RunicP | how do i log in to asterisk with my hardphone. |
| 16:26.13 | RunicP | im trying to barge right now on my agents calls |
| 16:29.15 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 16:30.12 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 16:31.08 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 16:31.45 | rushowr | jason: what version of Zaptel are you running on that machine btw? |
| 16:32.57 | CanWood | Hey folks. I have a GXP2000 and a user has managed to enable what I can only describe as a client side call forward which I can't disable. If I reprogram another phone to have the same extension, it receives calls, but this one fires back a "Got SIP response 302 "Moved Temporarily" |
| 16:33.05 | CanWood | any ideas short of a factory reset? |
| 16:35.00 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 16:37.33 | *** join/#asterisk [hC] (n=hardcore@190.10.9.126) |
| 16:39.38 | rushowr | jasonwoot: version of Zaptel? I'm putting in Asterisk 1.4.16 but need to know if maybe my Zaptel is partially to blame... |
| 16:41.59 | mort_gib | rushowr: I get dropped calls too, I use a Snom A500 BRI card |
| 16:42.08 | mort_gib | It's driving me nuts! |
| 16:42.24 | rushowr | right on, yeah it's a pain in the ass when you're trying to get paid for a project completion but there's random dropped calls |
| 16:42.38 | rushowr | I'm trying Jasonwoot's suggestion and migrating back to 1.4.16 |
| 16:43.30 | russellb | if you were using 1.4.18, why not try 1.4.19, or 1.4.20? |
| 16:43.33 | russellb | instead of going back... |
| 16:43.39 | rushowr | I'm on 1.4.19 |
| 16:43.46 | russellb | try 1.4.20 first |
| 16:43.56 | rushowr | and rather than try something that MIGHT work, I'd rather move back to something that IS known to work |
| 16:44.02 | mort_gib | rushowr: I'm in the exact same situation, they have the cheque ready, and will hand it over when we are rid of the dropped calls, I'm on 1.4.18 |
| 16:44.17 | rushowr | I've been dealing with this too long to try somethign that isn't KNOWN to work |
| 16:44.21 | mort_gib | -Sorry 1.4.19 |
| 16:44.34 | russellb | 516 fixes to asterisk 1.4 since 1.4.16 ... |
| 16:44.38 | russellb | anyway, whatever works for you |
| 16:44.40 | rushowr | I'll check out the changelog for 1.4.20 |
| 16:44.40 | russellb | moves on |
| 16:44.46 | Qwell | russellb: how many of those were security fixes? |
| 16:45.07 | mort_gib | russellb: If you have any other ideas, apart from the MS approach (upgrade) |
| 16:45.21 | mort_gib | I'm -eh ALL EARS |
| 16:45.30 | Qwell | mort_gib: upgrade and report bugs when things don't work |
| 16:45.36 | Qwell | otherwise...it will never be fixed. |
| 16:46.26 | rushowr | that's all fine and good for personal testing but when something's production you can't tell the customer that you're moving to an unknown possible fix just so the bug can be reported. I'll put 1.4.20 on my personal test machines and report issues I find, not on client machines |
| 16:46.27 | mort_gib | Okay, but I started on 1.4.17 with this client, and moved them to 1.4.19 to no avail, bug reports required me to be able to reproduce the issue, which I can't |
| 16:46.29 | *** join/#asterisk gitguy (n=diego@adsl-159-128.click.com.py) |
| 16:46.37 | gitguy | hi |
| 16:46.53 | Telamon | mort_gib: Dealing with the exact same issue, only wireless is thrown into the mix. |
| 16:46.53 | gitguy | i need to make a MeetMe conference password-protected, how do i make this |
| 16:47.03 | gitguy | i want users to enter the password/pin |
| 16:47.17 | mort_gib | -I Agree with rushowr |
| 16:47.19 | Telamon | I recommend turning on rtp-stats debugging, and then checking to see if you are getting packet loss. |
| 16:47.21 | gitguy | to get prompted |
| 16:47.40 | Telamon | From what I see, anything over a 1-2% packet loss can cause dropped calls. |
| 16:47.40 | jasonwoot | rushowr: you're not using a Dell PE 2950 by any chance? |
| 16:47.47 | Telamon | Throwing qualify up to 3000 might help. |
| 16:47.55 | rushowr | jason: no sir, SuperMicro custom built |
| 16:48.03 | mort_gib | Telamon: Yeah, well ALL are on the same layer 3 switch! |
| 16:48.20 | russellb | the joys of open source ... |
| 16:48.20 | mort_gib | jasonwoot: I'm on a PE 840 |
| 16:49.09 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 16:49.24 | Telamon | Still might want to try it. Especially if you have the computers plugged into the switch port on your phones (I'm not familiar with the Astras, dunno if they have a switch port) |
| 16:49.31 | dawebber | russelb: it's not open-source specific. If you can't reproduce a problem, no one can fix it. |
| 16:49.42 | rushowr | russelb: true dat |
| 16:49.59 | rushowr | and intermittent call drops are just that sort of thing |
| 16:50.06 | Qwell | dawebber: moreso, if it isn't reported, it can't be fixed. |
| 16:50.09 | rushowr | I've made so many test calls to try and reproduce |
| 16:50.13 | *** join/#asterisk fakhir (i=a7ce8021@gateway/web/ajax/mibbit.com/x-961da4a451878e27) |
| 16:50.13 | gitguy | i'm trying to create password protected conferences with meetme, i used 'p' for this but it didn't prompt me anything... |
| 16:50.16 | dawebber | rushowr: look at the IRQ use by the cards |
| 16:50.16 | gitguy | anyone???? |
| 16:50.23 | mort_gib | I have been running a script that pings every device from the * server every 2 minutes, and I never get a miss |
| 16:50.49 | mort_gib | dawebber: I happens to me with Internal calls too... |
| 16:50.56 | mort_gib | -Less often |
| 16:50.57 | dawebber | rushowr: there are some idiosyncrasies with IRQ alloc on some MBs. Digium cards like it a certain way |
| 16:51.00 | rushowr | dawebber: one sec |
| 16:51.16 | rushowr | dawebber: yeah Digium support and I figured out that one, bit of a bitch though |
| 16:51.25 | rushowr | dawebber: hard to convince the client that they need a new mobo |
| 16:51.42 | dawebber | rushowr: sometimes, it's switching the slots that does the trick |
| 16:51.47 | *** join/#asterisk wordzilla (n=me@122.109.126.193) |
| 16:51.48 | jasonwoot | rushowr: ZAP with T1s has been such a pain, I'm slowly converting to voip trunks to phase them out |
| 16:51.59 | JayTee52 | jasonwoot, do you have problems with a PE 2950? |
| 16:52.02 | dawebber | jasonwoot: welcome to Voice 2.0 :) |
| 16:52.05 | rushowr | dawebber: I'll try that, moving to a bigger better server soon anyway |
| 16:52.27 | dawebber | rushowr: I'm using several 2650s, so far no issues like that |
| 16:52.30 | rushowr | jasonwoot: yeah I know, I'm trying to convince the client that VOIP trunks are the way to go, but they LOVE their T1s...when it was PRI it wasn't a problem |
| 16:52.54 | rushowr | but the new T provider screwed up and were EM WINK right now until they fix it |
| 16:52.57 | jasonwoot | JayTee52: yeah, both my 29xx series and 19xx series Dells have given me lots of PCI controller problems |
| 16:54.01 | jasonwoot | JayTee52; I haven't been b!tching about zap much in here because there's a good chance that its a hardware problem, although zap doesn't seem to indicate that |
| 16:54.30 | JayTee52 | I've got * on a PE2950 Quad Core Xeon running RHEL5 64bit and no real problems with a TE212P card but I had PCI parity errors when using the TDM04B cards. I moved them to a 2650 and the problem moved with them. |
| 16:54.37 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 16:55.01 | JayTee52 | I suspect one of the 2 TDM04B cards has a flaky FXO module and I'm pretty sure I've pinned down which one. |
| 16:55.02 | dawebber | JatTee52: those old workhorses are as reliable as it gets |
| 16:55.03 | jasonwoot | rushowr: if your T1 provider is as bad as mine, the quality/price tradeoff is well worth the switch |
| 16:55.22 | rushowr | jasonwoot: yeah I'll be digging up some numbers REAL soon no worries there |
| 16:55.41 | mort_gib | Telamon: Qualify on the phones or in sip.conf |
| 16:55.52 | JayTee52 | dawebber, yeah they are good but I've found CentOS with Gnome loaded crawls although zttest still gives me 99.9??? all the time. |
| 16:55.54 | *** join/#asterisk Telemac (n=cchantep@ANantes-157-1-130-159.w90-25.abo.wanadoo.fr) |
| 16:55.57 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:56.01 | Telemac | Hello |
| 16:56.05 | acxty | what do you recommend on Authenticate. Usign a file or a db? |
| 16:56.13 | *** join/#asterisk HumanCell (n=HumanCel@207.224.214.130) |
| 16:56.14 | JayTee52 | and they aren't as flexible in assigning interrupts as the 2950 is. |
| 16:56.15 | jasonwoot | JayTee52: I'm on a TE410P, but yes the parity errors are what's probably sinking me |
| 16:56.54 | rushowr | jasonwoot: Have you dealt with Airespring (Qwest Reseller) |
| 16:56.55 | mort_gib | JayTee52: you might want to have a look at Sangomas stuff |
| 16:57.08 | Telamon | mort_gib: qualify in the sip.conf entry. That basically says "if we get a bad network connection, wait at least 3 seconds before hanging up" from what I understand. The default is 1 or 2 seconds, forget which... |
| 16:57.15 | jasonwoot | rushowr: no, sounds like West Coast? |
| 16:57.24 | JayTee52 | jasonwoot, that's a shame because without a non-Dell server it's hard to pinpoint whether it's definitely the card or the system. |
| 16:57.30 | Telamon | mort_gib: If you go from dropped calls to choppy calls, that means you've got a network issue. |
| 16:57.40 | rushowr | jasonwoot: I'm in Springfield MO, but they might be west coast |
| 16:58.02 | mort_gib | Telamon: Yeah, that figures, but they are on the same damn switch, and they are all ALONE! |
| 16:58.36 | Telamon | mort_gib: Is your asterisk server dedicated, or is it also a file server/web server? |
| 16:59.13 | mort_gib | Telamon: dedicated, it does run Apache, but only to cater for the Outlook integration.... |
| 16:59.32 | mort_gib | Telamon: as in low, low usage |
| 16:59.58 | rushowr | jasonwoot: just compiled and installed 1.4.16...clients are at lunch... |
| 17:00.02 | rushowr | crosses fingers :) |
| 17:00.23 | rushowr | I guess I'll put 1.4.20 into effect if the changelog shows me something that makes me believe T1 EM_W handling is better |
| 17:00.43 | mort_gib | We are talking a PE 840 running in average .01-0.2% CPU |
| 17:01.12 | Telamon | Okay, and it's a good quality switch? So it's got like a full gig backplane, it's not just a hub with a web interface sort of thing? *Might* be your phone card, as other's suggested, but I'd still try the RTP-stats and qualify=3000 to test. Easier than replacing your phone card. |
| 17:01.24 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 17:01.40 | rushowr | alright mates, I'm going to go eat lunch...if ya see Murf around, tell him rushowr says hey |
| 17:01.42 | rushowr | bbl |
| 17:01.47 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:01.58 | *** join/#asterisk thx2000 (n=bob@netblock-208-127-150-56.dslextreme.com) |
| 17:01.59 | *** part/#asterisk HumanCell (n=HumanCel@207.224.214.130) |
| 17:02.18 | mort_gib | Telamon: Well it IS a Dell Switch, but not the 27XX series, |
| 17:04.00 | Telamon | Heh, I hate Dell switches, but just because of how stupid the VLAN config is in the 2700 series. I don't think they have any real quality problems. |
| 17:04.30 | jasonwoot | I'm coming out of the closet: all my network and servers are belong to Dell |
| 17:04.43 | mort_gib | Well I'm no big fan, but Cisco is too expensive, and Dell is the ONLY brand that delivers local support! |
| 17:04.44 | jasonwoot | feels better having said that |
| 17:04.56 | mort_gib | :-) |
| 17:05.33 | jasonwoot | even if it's cheap, 'continutity' goes a long way |
| 17:06.22 | jasonwoot | Dell has the best, cheapest POE managed switches, although the management features are mostly worthless |
| 17:06.28 | mort_gib | Yes fine! -But Dell is the ONLY brand that delivers ANY support unless you pay for two devices!! |
| 17:06.40 | mort_gib | -They come with a cli -Right?? |
| 17:07.30 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 17:07.42 | *** join/#asterisk galeras (n=galeras@201.244.195.23) |
| 17:07.59 | mort_gib | Actually one of my clients has two of the POE switches looming for me to configure in his office.... |
| 17:08.21 | CanWood | anyone here running a "Presence Server"? The GXP2000 says you need to have one running to get EventList BLF working and without it, I don't seem to be able to have DND indicated by the Busy Lights |
| 17:08.42 | *** part/#asterisk dawebber (n=dawebber@209.136.161.135) |
| 17:11.57 | CanWood | yet there seems to be no documentation about supported ones or vendors, integration, etc |
| 17:12.39 | galeras | is this true:? "video support in Asterisk 1.4 is still in infancy" |
| 17:12.51 | CanWood | alternaltely, if anyone has any leads on having the GXP2000's busy light accurately indicating DoNotDisturb states on them, I'd be appreciative |
| 17:15.03 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 17:15.10 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 17:16.18 | [TK]D-Fender | CanWood: DND you'd have to refer to your admin guides. for BLF, * supports this jsut fine. Look up "asterisk presence" on the WIKI |
| 17:17.24 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 17:17.38 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 17:17.41 | acxty | I am trying to store the passwd that the user give to Authenticate(). I read some documentation and I do this. Authenticate(/mypasswdfile,a) |
| 17:17.51 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:18.03 | acxty | supposely the a option store the passwd into ${ACCOUNTCODE} |
| 17:18.20 | acxty | but when I verify it, ${ACCOUNTCODE} is empty |
| 17:19.19 | *** join/#asterisk dysinger (n=tim@cpe-75-80-200-182.hawaii.res.rr.com) |
| 17:19.20 | CanWood | [TK]D-Fender: thanks for the lead. I have BLF working in general, just nit indicating DND. The wiki's only reference to DND related to BLF that I could find, seemed to state that it's not supported. However three weeks ago in this channel, when I inquired i got a "it just works" response |
| 17:19.29 | [TK]D-Fender | acxty: "core show function CDR" |
| 17:19.43 | acxty | thanks |
| 17:19.51 | [TK]D-Fender | CanWood: There is no server-bases DND indication standard for that. |
| 17:21.58 | CanWood | lovely. I can't go a day without an under the breath comment from my staff about how they never know if their co-workers are on DND. I had figured it was a common feature |
| 17:22.23 | CanWood | thankf for the clarification [TK]D-Fender |
| 17:23.15 | [TK]D-Fender | CanWood: You might be able to disable DND on the phone itself, implement it in your dialplan, and then use DevState + BLF to indicate. |
| 17:23.20 | [TK]D-Fender | ~devstate |
| 17:23.21 | jbot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/ |
| 17:24.01 | CanWood | Ill dig there. They are using the server side *78, not the GXP's DND, so they are already in the right direction |
| 17:24.10 | CanWood | tks |
| 17:25.35 | russellb | O.O |
| 17:26.00 | [TK]D-Fender | russellb: Don't ask questions you already know the answers to ;) |
| 17:27.09 | *** join/#asterisk zackz (n=zdz@rrcs-24-123-106-250.central.biz.rr.com) |
| 17:27.27 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.227) |
| 17:27.53 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 17:28.26 | zackz | is it possible to dynamically add an agent to a queue like this? |
| 17:28.27 | zackz | exten => 15240,n,AddQueueMember(support|Local/${EXTEN:1}@agents|900) |
| 17:28.43 | [TK]D-Fender | zackz: tahts the point. |
| 17:28.56 | zackz | when I try it shows the agent as "invalid" when I do a show queue |
| 17:29.12 | *** join/#asterisk Chunkz (i=Steve@ool-43536cf6.dyn.optonline.net) |
| 17:30.27 | [TK]D-Fender | zackz: I doubt 900 is a valid penalty... |
| 17:30.27 | Chunkz | Is this an ok place to ask a relatively straight forward question? Dunno if there's a help, or newbies channel.. |
| 17:30.40 | [TK]D-Fender | Chunkz: Just ask and we'll correct if needed. |
| 17:31.13 | zackz | Fender: 900 works as a penalty if I just add a SIP/ channel to the queue |
| 17:31.47 | Chunkz | Ok.. Well, my asterisk box is behind a DSL line that went down for a few hours, and after it came back up, the SIP trunk did not reconnect until I did a "sip reload"... is there a way to tell it to keep retrying automatically? |
| 17:32.31 | JayTee52 | anyone here use a Linksys PAPT2-NA? are they any good? |
| 17:32.54 | JayTee52 | oops, meant PAP2T-NA |
| 17:33.06 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 17:33.13 | zackz | I get this: NOTICE[6137]: app_queue.c:3640 aqm_exec: Added interface 'Local/5240@agents' to queue 'support' |
| 17:33.39 | zackz | then it shows this when I do a "show queues": Local/5240@agents with penalty 1 (dynamic) (Invalid) has taken no calls yet |
| 17:33.53 | zackz | invalid.... |
| 17:34.22 | zackz | penalty 1 so that is a valid penalty |
| 17:34.27 | zackz | so that's not what is causin git |
| 17:34.51 | [TK]D-Fender | zackz: pastebin all related dialplan. |
| 17:34.53 | [TK]D-Fender | ~pb |
| 17:34.54 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:35.15 | [TK]D-Fender | JayTee52: Decent enough. |
| 17:35.37 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-a7d2a88e6c6dd8b5) |
| 17:35.46 | JayTee52 | [TK]D-Fender, by decent enough I'm thinking better than a Grandstream Handytone by a longshot? |
| 17:36.46 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 17:37.22 | *** part/#asterisk galeras (n=galeras@201.244.195.23) |
| 17:40.06 | gitguy | i pass arguments to MeetMe to prompt me for password but it never prompts |
| 17:40.18 | gitguy | exten => 90000,1,MeetMe(90000,DpM,1234) |
| 17:40.36 | [TK]D-Fender | JayTee52: Yeah, a better idea. |
| 17:41.31 | [TK]D-Fender | gitguy: Show us the failed attemp at verbose 10 |
| 17:41.41 | JayTee52 | [TK]D-Fender, thanks for the input. I hold your opinion on such things in high regard. |
| 17:42.09 | [TK]D-Fender | JayTee52: what kind of use, and located where? |
| 17:43.19 | JayTee52 | for some of our outbuildings at the zoo where we presently have an analog phone but don't want to deal with the expense and time involved in running CAT5 to that location. moderate to little use. |
| 17:45.01 | [TK]D-Fender | JayTee52: if its local LAN to the server then sure. Otherwise I'd spring a few more $ for the SPA-2102 |
| 17:45.11 | gitguy | [TK]D-Fender: http://pastebin.com/m25ff7777 |
| 17:45.25 | zackz | Fender: here is a pastebin of the path my dialplan takes to log an agent in: http://pastebin.com/m6b2c01d9 |
| 17:46.00 | JayTee52 | [TK]D-Fender, it's all local. Our inbound/outbound is 2 T1's, PRI |
| 17:46.39 | JayTee52 | so I don't need one that can handle routing behind a NAT'd firewall |
| 17:46.46 | [TK]D-Fender | zackz: And show me every context referenced in there... |
| 17:47.12 | zackz | theyre all there |
| 17:47.28 | [TK]D-Fender | zackz: No they're not. Look closer and you should see what I want... |
| 17:47.56 | zackz | [agent-logon]? |
| 17:48.01 | zackz | i forgot to paste it in |
| 17:48.13 | zackz | just that line though |
| 17:48.16 | [TK]D-Fender | zackz: Everything... there's more... |
| 17:48.32 | Kobaz | hmmm |
| 17:48.40 | Kobaz | which module would i need to reload to reread agents.conf |
| 17:49.08 | zackz | i guess i dont see how any other contexts would matter |
| 17:49.31 | zackz | oh ok, sip.conf probalby |
| 17:50.41 | [TK]D-Fender | zackz: nope.. |
| 17:51.01 | [TK]D-Fender | zackz: c'mon... PB the whole mess if you can't think of it. |
| 17:51.04 | Kobaz | never mind |
| 17:51.05 | Kobaz | chan_agent.so |
| 17:52.50 | zackz | i cant think of it |
| 17:53.03 | zackz | i have too much private stuff to paste the whole thing and not take forever to edit it :( |
| 17:54.15 | Kobaz | zackz: egrep -v host|secret |
| 17:55.01 | Kobaz | probably need quotes |
| 17:55.31 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 17:56.33 | zackz | [TK]D-Fender: how am I supposed to know what you want, what if I have 100 .conf files with different stuff? |
| 17:56.43 | tzafrir_home | egrep -v 'host|secret' |
| 17:57.08 | tzafrir_home | Otherwise it would look for the command 'secret' |
| 17:57.17 | Kobaz | yeah |
| 17:58.15 | Chunkz | perhaps I can rephrase my question - should a SIP trunk require a "sip reload" to reconnect after it goes down? |
| 17:58.28 | Kobaz | no, unless the dns resolution changed |
| 17:59.11 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-116-41.PHNX.QWEST.NET) |
| 17:59.20 | Kobaz | if you're connecting to abc.com, and all of a suddon it points to another ip, asterisk won't know since it doesn't do any recurring dns lookups |
| 17:59.25 | Kobaz | but that's about it |
| 18:00.19 | Chunkz | both ends are static, but the sip trunk stayed down since the internet connection was offline for a couple hours yesterday |
| 18:00.53 | Chunkz | I was wondering if there's a way to see retries or reconnect attempts.. I'll ahve to watch the console next time. |
| 18:01.12 | acxty | hey guys, make you look at this http://dpaste.com/48547 What I want to do is that no one can dial(agent) if it don't have a valid passwd. My peer context=incoming. do you find any way that someone can go to [agent] without a passwd? |
| 18:01.31 | PaulQ | I can seg fault asterisk with AEL |
| 18:01.34 | PaulQ | am I leet or suck? :P |
| 18:02.07 | Kobaz | Chunkz: you can do a sip debug, which will spit out way more info than you'll ever want to see |
| 18:03.33 | Chunkz | Kobaz: Thanks. :D |
| 18:04.13 | codefreeze | PaulQ: definitely leet. |
| 18:04.28 | codefreeze | PaulQ: file a bug report, and I'll get it fixed. |
| 18:04.38 | PaulQ | codefreeze, bug.$domain ? |
| 18:05.08 | codefreeze-lap | bugs.digium.com |
| 18:05.17 | PaulQ | Okay |
| 18:05.31 | hmmhesays | anyone know the speed of the serial port on an blackfin stamp537 board? |
| 18:07.30 | PaulQ | Let me install a version to make sure I can reproduce it |
| 18:09.21 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:13.00 | adeel | Chunkz, i know freepx 2.4 has a method to monitor the status of a trunk and report failures, you can take a look at their source for an idea |
| 18:13.30 | adeel | hmmhesays, have you tried any values yet? |
| 18:13.48 | hmmhesays | haha nice tab |
| 18:13.50 | PaulQ | It only crashes under 1.4.19.1 |
| 18:13.53 | PaulQ | not 1.4.18 |
| 18:13.54 | PaulQ | weird |
| 18:14.31 | codefreeze | PaulQ: something new, then. |
| 18:14.39 | Qwell | hmmhesays: likely 57600 |
| 18:14.47 | Qwell | 8N1 |
| 18:14.52 | *** join/#asterisk edibrac (n=edibrac3@netblock-66-245-252-161.dslextreme.com) |
| 18:14.58 | Qwell | (disable flow control) |
| 18:14.59 | hmmhesays | Qwell yeah I'm getting some crazy output |
| 18:15.24 | Qwell | ((hw and sw)) |
| 18:16.54 | acxty | hey guys. I want to buy a card to connect some reall phones directly to the asterisk server. Which one do you recommend. I will like to have 4 telephones connected to it, normal telephones |
| 18:16.54 | Qwell | acxty: Digium TDM410 |
| 18:16.54 | hmmhesays | yeah still some crazy output the speed is not right |
| 18:16.54 | acxty | thanks |
| 18:16.54 | Qwell | with 4 FXS modules |
| 18:16.54 | Qwell | (or a TDM800, if you plan on expanding) |
| 18:17.16 | M1s3ry | (or TDM2400 if you really want some expansion) |
| 18:17.37 | acxty | heheh, thanks guys |
| 18:18.56 | hmmhesays | ahh its 115200 |
| 18:20.24 | edibrac | for our production boxes at work, we use rpath -- is the asterisk package that comes in rpath the same one that asterisknow uses? Apparently the latest conary rpath version is 1.2.x |
| 18:20.48 | acxty | Qwell, M1s3ry this cards are kind of expensives rigth |
| 18:21.10 | edibrac | those are like $300 i think |
| 18:21.11 | Qwell | acxty: not really, no |
| 18:21.28 | edibrac | er, i have one at home with just one FXO and one FXS |
| 18:21.44 | M1s3ry | acxty, especially compared to proprietary telephony solutions, they are extremely cheap |
| 18:22.00 | acxty | I am seeing the TDM410, think I am going to get the TDM410 |
| 18:22.29 | Qwell | acxty: just remember 4 fxs modules, for phones |
| 18:22.39 | acxty | I only need two lines right now |
| 18:22.42 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 18:22.44 | Qwell | lines or phones? |
| 18:23.06 | acxty | phones? |
| 18:23.11 | acxty | sorry |
| 18:23.12 | acxty | hehe |
| 18:23.12 | Qwell | lines and phones are different :) |
| 18:23.13 | acxty | yes phones |
| 18:23.39 | acxty | My lines are sip, not having problem with them hehe |
| 18:24.54 | acxty | Qwell, comparing a softphone and using that card, does the sound improve significantly. I make a test adding a peer on asterisk and using xlite for that peer, and the sound quality don't relly convince me |
| 18:24.59 | *** join/#asterisk CiscoJones (n=mike@HW-ESR1-208-102-7-64.fuse.net) |
| 18:25.26 | M1s3ry | dislikes softphone quality most of the time |
| 18:25.50 | M1s3ry | especially on my old laptop, which is really the last time I used a softphone |
| 18:26.13 | acxty | will get the card them, do I need extra things to be compile on asterisk? |
| 18:26.26 | acxty | and more important, do I need to install asterisk again? |
| 18:27.03 | acxty | I am using CentOS 5 |
| 18:27.18 | *** join/#asterisk grandpapadot (n=anonymou@mail.heavylogic.com) |
| 18:27.20 | JayTee52 | Yahoo sent me a big fat purple button similar to the Staples Easy button but it goes Yahooo-ooo instead of "that was easy!" |
| 18:27.39 | grandpapadot | Hi all. What will effectively reload musiconhold and re-parse music files using mode=files? Does it take a full reload? |
| 18:27.49 | JayTee52 | I swear the world is running out of original thought faster than it's running out of oil. |
| 18:27.55 | M1s3ry | acxty, just compile zaptel |
| 18:28.12 | acxty | thanks, lunch time, see you later |
| 18:28.14 | acxty | ;) |
| 18:29.16 | JayTee52 | grandpapadot, just type moh reload at the console |
| 18:30.29 | *** join/#asterisk wichitalineman (n=svf@67.106.246.227.ptr.us.xo.net) |
| 18:31.00 | grandpapadot | thanks |
| 18:31.30 | hmmhesays | anyone know where I can get some old skype verisions |
| 18:31.34 | hmmhesays | *versions |
| 18:31.58 | jblack | I think you got lost. This is #asterisk, not skype technical support. |
| 18:31.59 | JayTee52 | skype doesn't work with asterisk |
| 18:32.08 | jblack | You should be able to reach them at support@skype.com |
| 18:34.48 | jeffspeff | could somebody give me a hand... i'm going through The Book, and i'm at the part where I set my dialpan to use 9 for outbound and set the EXTEN var, etc. .... but for some reason it's not getting into the new context i set up [outbound_local].... here's my extensions.conf http://pastebin.ca/1009580 and here's my sip.conf http://pastebin.ca/1009586 |
| 18:34.59 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:38.05 | Qwell | jeffspeff: you don't have an outbound_local context |
| 18:38.45 | jeffspeff | ahh. in sip.conf i need to set outbound-local.... right? |
| 18:38.53 | jeffspeff | i love typos... :P |
| 18:39.03 | Qwell | or fix it all to be consistent. either way |
| 18:39.13 | jeffspeff | ok, thanks Qwell. |
| 18:41.28 | edibrac | is 1.2.x considered production quality, and all others (1.4+) unstable? |
| 18:41.34 | whymarkwh | is asterisknow and free pbx the same? |
| 18:42.18 | JayTee52 | whymarkwh, no they are not the same but both based on asterisk |
| 18:42.21 | edibrac | er, i mean, is it "officially" deemed unstable by whoever makes official pronouncements? |
| 18:43.24 | JayTee52 | officially? "There is no God but Asterisk and Mark Spencer is his Prophet" |
| 18:45.36 | JayTee52 | unofficially I've found 1.4.15 to be pretty stable for me. |
| 18:48.12 | bkruse | whymarkwh: no |
| 18:48.41 | bkruse | whymarkwh: I should be watching the #asterisknow channel, but I am busy busy, but no they are not. Asterisknow is a distribution with everything already setup (asterisk + the digium asterisk GUI) |
| 18:55.13 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:55.17 | [TK]D-Fender | whymarkwh: No. AsteriskNOW is a distru with Asterisk-GUI. FreePBX is just a GUI |
| 18:59.06 | *** join/#asterisk zomgoblinz (n=zomgobli@81.90.21.227) |
| 18:59.52 | zomgoblinz | is it possible in extension.conf to prompt to have a password in order to dial out? |
| 19:00.18 | zomgoblinz | ie, we have a gsm gateway, want to make it possible to enter a code to dial out across our ast box |
| 19:01.48 | bkruse | zomgoblinz: authenticate |
| 19:01.55 | bkruse | hey [TK]D-Fender :] |
| 19:02.02 | zomgoblinz | bkruse, thanks |
| 19:02.06 | zomgoblinz | will read up on that |
| 19:02.39 | bkruse | zomgoblinz: Some good information and use of it on voip-info.org, as always :] |
| 19:03.03 | zomgoblinz | heh, great site almost a replacement for the ast book |
| 19:03.05 | zomgoblinz | :) |
| 19:03.47 | bkruse | zomgoblinz: Never. You go there after you have read the book :P |
| 19:03.49 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 19:03.49 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com) |
| 19:03.56 | bkruse | It is a very good reference if you know what to look for :] |
| 19:04.34 | zomgoblinz | yeah, on question |
| 19:05.30 | zomgoblinz | the gsm gateway currenty only include => internal right now. can you do an include on the fly if authenticated? |
| 19:05.30 | zomgoblinz | hmm |
| 19:05.30 | zomgoblinz | guess i have to read |
| 19:05.59 | [TK]D-Fender | zomgoblinz: nothing feasible |
| 19:06.27 | [TK]D-Fender | zomgoblinz: You can cheat with more global pattern matches and dialplan logic however |
| 19:06.31 | [TK]D-Fender | bkruse: y0 |
| 19:07.00 | zomgoblinz | [TK]D-Fender, guess I have to include both and then do pattern matching |
| 19:07.01 | zomgoblinz | yeah |
| 19:07.24 | zomgoblinz | would be nice if I could just do an authenticate if auth then include => |
| 19:07.28 | zomgoblinz | :) |
| 19:07.46 | [TK]D-Fender | zomgoblinz: No, you would not includ both, you'd jump to the other if authenticated (global match + GotoIf) |
| 19:08.28 | zomgoblinz | ah |
| 19:11.53 | hmmhesays | I cannot get asterisk to playback inband ringing after an answer on my skype channel |
| 19:12.04 | hmmhesays | ringing(); after answer should do that right? |
| 19:12.29 | *** join/#asterisk SamuraiDio (n=diovani@201.41.41.235) |
| 19:12.31 | SamuraiDio | hi |
| 19:12.57 | *** join/#asterisk [hC] (n=hardcore@190.10.10.76) |
| 19:13.00 | SamuraiDio | when using g729 codecs i have thousands of 'dropping extra frame' notices... how can i disable that? |
| 19:13.16 | bkruse | How can you fix the problem? Or not see the notices? |
| 19:13.26 | bkruse | If you just want to ignore the notices, edit /etc/asterisk/logger.conf |
| 19:13.40 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 19:13.47 | SamuraiDio | bkruse, but is that a problem? the call works fine |
| 19:14.56 | [TK]D-Fender | SamuraiDio: Evidently not. |
| 19:15.33 | SamuraiDio | [TK]D-Fender, "Evidently not" ...is a problem? |
| 19:15.57 | [TK]D-Fender | ~cluebat SamuraiDio |
| 19:15.57 | jbot | ACTION pulls out a ClueBat (tm) and thwaps SamuraiDio. |
| 19:16.18 | SamuraiDio | ugh! |
| 19:16.38 | SamuraiDio | sorry, i'm from brazil, sometimes i can't understant... o.O |
| 19:17.08 | SamuraiDio | ok, i edited the /etc/asterisk/logger.conf, how do i reload it? |
| 19:18.05 | *** join/#asterisk dlynes (n=chatzill@dsl-vlan468-66-18-244-66.nucleus.com) |
| 19:18.17 | SamuraiDio | :\ |
| 19:18.33 | bbryant | "logger reload" |
| 19:19.06 | SamuraiDio | thanks |
| 19:24.04 | logi4023 | f |
| 19:27.25 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 19:33.03 | *** join/#asterisk dlynes_laptop (n=chatzill@dsl-vlan468-66-18-244-66.nucleus.com) |
| 19:33.27 | zomgoblinz | are warnings for Received trunked frame before first full voice frame |
| 19:33.31 | zomgoblinz | <PROTECTED> |
| 19:33.46 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 19:35.58 | *** join/#asterisk RoyK (n=roy@ip-179-6-149-91.dialup.ice.no) |
| 19:36.12 | *** join/#asterisk francogwapito (n=chatzill@125.252.90.5) |
| 19:38.23 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 19:41.48 | gitguy | i'm using meetme(), is there a way for me to setup like 10 conferences, all with their own passwords? |
| 19:42.07 | *** join/#asterisk dlynes_office (n=chatzill@dsl-vlan468-66-18-244-66.nucleus.com) |
| 19:42.19 | colinm_ | gitguy: meetme.conf? |
| 19:42.25 | *** join/#asterisk docelmo (n=chatzill@206.248.239.194) |
| 19:42.29 | gitguy | ok |
| 19:42.41 | gitguy | i need it to prompt the person when they call it |
| 19:43.22 | bkruse | authenticate, meetme |
| 19:43.22 | docelmo | Quick scenario.. Asterisk 1.4, Digium or Sangoma TDM T1 board, Cisco 186 IAD.. If I pass T38 from the IAD to asterisk will it convert T38 to the TDM or would I need to use Ulaw? |
| 19:43.48 | Qwell | Asterisk doesn't do any T.38 conversions |
| 19:43.48 | gitguy | bkruse: authenticate? |
| 19:44.00 | dlynes_office | docelmo: asterisk doesn't do t.38 translation; only pass-through |
| 19:44.07 | docelmo | qwell what do you suggest? straight ULAW? |
| 19:44.25 | dlynes_office | docelmo: horseshoes, and lots of them |
| 19:44.44 | bkruse | gitguy: Ya, if you want to authenticate before the meetme conference, and there is meetme conference passwords, and there is a "announce" field for people comming in |
| 19:45.14 | docelmo | huh? |
| 19:45.15 | gitguy | bkruse: how can i do that, are pins different than passwords? where can i read about how to do this? |
| 19:45.28 | dlynes_office | docelmo: for the luck...you know lucky horseshoes? |
| 19:45.40 | docelmo | not off the top of my head |
| 19:46.03 | bkruse | pins are not |
| 19:46.20 | dlynes_office | docelmo: anyways...getting fax to work over ulaw is trying at the best of times |
| 19:46.42 | dlynes_office | docelmo: you might get lucky that 1 in every 100 faxes might turn out fine, but for the most part, it just doesn't work |
| 19:46.44 | docelmo | Ya.. I was thinking channelbank |
| 19:46.57 | colinm_ | gitguy: You're trying to have everyone dial one number for conferencing, and then enter a conference number, right? And then the password for that conference? |
| 19:47.22 | gaetronik | is there a prevision for 1.4.20 release |
| 19:48.18 | docelmo | FXS channel bank to TDM card then go out PRI |
| 19:48.29 | [TK]D-Fender | gaetronik: load chan_psychic.so |
| 19:49.25 | Qwell | gaetronik: prevision? |
| 19:50.00 | gitguy | colinm_: no, i need to add something like this in my extensions.conf: 1000,1,MeetMe(1000,i,123456) 2000,1,MeetMe(2000,i,123456) ... and so on |
| 19:50.24 | gitguy | colinm_: and when they dial 1000 or 2000 they join it, but before joining, i want them to authtenticate with a password for joining |
| 19:51.02 | gitguy | colinm_: i need different conferences for different groups of people, but each meetme conference with an unique password |
| 19:51.09 | gaetronik | is there a release date in the future |
| 19:51.26 | gaetronik | my english is sometime so bad that i'm ashamed |
| 19:51.40 | gitguy | colinm_: understand it? |
| 19:52.11 | gaetronik | [TK]D-Fender, first i will load chan_rational.so |
| 19:52.20 | gaetronik | to stop asterisk doing what he wants |
| 19:52.43 | gaetronik | Particularly in the middle of the night |
| 19:52.47 | colinm_ | gitguy: Yeah, I'm just not seeing what the difficulty is. That's built into meetme. |
| 19:53.10 | colinm_ | gitguy: You can set an admin pin and a participant pin in the meetme.conf |
| 19:53.44 | [TK]D-Fender | gaetronik: Yes, the release date is in the future. We'll let you know what it is when we get there :p |
| 19:54.05 | gitguy | colinm_: so are pins like passwords? |
| 19:54.09 | gaetronik | [TK]D-Fender, sometimes there is a roadmap, no? |
| 19:54.14 | colinm_ | gitguy: If you're putting it in the dialplan, it's *bypassing* the PIN prompt and plugging it in automatically. |
| 19:54.20 | [TK]D-Fender | gaetronik: No, only road-kill.... |
| 19:54.33 | gaetronik | ok |
| 19:54.36 | colinm_ | gitguy: Oh, oh, now I know where you took a left turn. |
| 19:54.52 | *** join/#asterisk za3toor (n=non@bas1-toronto12-1088934529.dsl.bell.ca) |
| 19:54.53 | colinm_ | gitguy: Put the PIN (password) you want to use in meetme.conf, and *don't* fill in the PIN in your dialplan. |
| 19:54.57 | colinm_ | gitguy: Just the conference number. |
| 19:55.19 | gaetronik | I'm stock in my upgrade because i can use 1.4.19 since it has a security flaw |
| 19:55.26 | colinm_ | gitguy: a la exten=>1000,1,Meetme(1000) |
| 19:55.39 | gaetronik | and 1.4.19.1 is slow witch iax if i understood well |
| 19:55.48 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:56.11 | gitguy | colinm_: right, thanks |
| 19:56.39 | gitguy | :) |
| 19:56.40 | *** join/#asterisk SamuraiDio (n=diovani@201.41.41.235) |
| 19:56.54 | glaz | Anyone has a 480i CT SIP phone? |
| 19:58.14 | gaetronik | and the version 1.4.16 is acting weird |
| 20:11.08 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 20:12.06 | *** join/#asterisk maagic (i=maagic@fsck.fi) |
| 20:13.18 | glaz | This phone is starting to piss me off. |
| 20:16.19 | dlynes_office | glaz: what's wrong with it? |
| 20:17.10 | glaz | even if I make changes to the conf, I see that the phone is still using the old config |
| 20:17.33 | jjshoe | glaz then the phone isn't getting the latest config. |
| 20:18.01 | glaz | I see that the phone is downloading the file from the TFTP server each time it boots |
| 20:18.26 | jjshoe | have you tried reseting it to defaults? |
| 20:20.23 | glaz | I did yup |
| 20:21.41 | *** join/#asterisk inja_y (n=Babbel@ip70-174-148-215.dc.dc.cox.net) |
| 20:23.28 | glaz | <PROTECTED> |
| 20:23.36 | glaz | finaly, it works. |
| 20:23.41 | glaz | Thanks for listening to me. |
| 20:25.44 | inja_y | hi, i'm trying to setup asterisk with srtp for a project... i tried for one week to make it work... anybody has any experience with jpeeler trunk? |
| 20:25.51 | *** join/#asterisk tobias (n=tobias@216.230.107.250) |
| 20:26.18 | inja_y | or with srtp? |
| 20:26.41 | jpeeler | inja_y: it doesn't work |
| 20:26.49 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 20:35.02 | PaulQ | Cant load 32bit modules into a 64bit asterisk? :( |
| 20:37.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:39.52 | *** join/#asterisk rdgr_ (n=rich@jwad-resnet-31477.d.port.ac.uk) |
| 20:40.32 | dlynes_office | glaz: you do know that if you change something in the web config and/or the dialpad config that it'll override anything in the tftp config, right? |
| 20:40.53 | keith4 | when dealing with Vonage, do I have any options other than using their box, connected to an analog zap channel on the asterisk box? |
| 20:42.00 | dlynes_office | keith4: judging by the conversation I saw on here yesterday, you can get a vonage business account that will let you use your own box |
| 20:42.12 | keith4 | really know |
| 20:42.30 | dlynes_office | keith4: but apparently the residential accounts do not allow you to do that |
| 20:42.42 | keith4 | so says the wiki |
| 20:42.45 | keith4 | which is why I asked |
| 20:43.03 | keith4 | ah |
| 20:43.10 | keith4 | i see... Vonage Business Plus |
| 20:46.15 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 20:46.37 | keith4 | http://www.vonage-asterisk.com/ |
| 20:46.43 | keith4 | for future inquiries |
| 20:46.48 | keith4 | ~vonage |
| 20:46.49 | jbot | it has been said that vonage is a bunch of monkeys |
| 20:47.05 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 20:47.11 | dlynes_office | didn't vonage go tits up or something, too? |
| 20:47.22 | dlynes_office | i.e. the main American company? |
| 20:47.55 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 20:47.58 | wichitalineman | monkeys? is there a telephony in-joke revolving around monkies, flying or otherwise? |
| 20:48.07 | rob0 | tt-monkeys |
| 20:48.27 | wichitalineman | goes off to look. many thanks |
| 20:48.31 | rob0 | it's a pair of sound files (tt-monkeys and tt-monkeys-intro) |
| 20:49.01 | rob0 | the tt-* are all fun |
| 20:51.01 | wichitalineman | oh neat :) |
| 20:51.03 | *** join/#asterisk Morrocco (n=ivan@189.164.87.104) |
| 20:52.05 | Morrocco | Hi Guys, I downloaded snaps dialer, I can get it to dial but when I lift up the phone it automatically drops the call, any ideas? I read that the context might be wrong, but where do I see that? |
| 20:52.51 | *** join/#asterisk rdgr_ (n=rich@jwad-resnet-31341.d.port.ac.uk) |
| 20:53.07 | *** join/#asterisk talntid (n=t@66.208.251.170) |
| 20:55.44 | *** join/#asterisk RoyK (n=roy@91.149.31.29) |
| 20:56.18 | [hC] | dont suppose anyone here knows anything about creating payphone readable calling cards that route long distance through your own switch? |
| 20:56.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:57.21 | Qwell | [hC]: I would be surprised if the card reader did anything more than read a pin number off the card |
| 20:57.28 | Qwell | pin/account number/whatever |
| 20:57.37 | Qwell | if that's what you mean |
| 20:57.51 | [hC] | Qwell: im just curious to create promo business cards that are payphone readable calling cards with like $5 on them |
| 20:58.10 | [hC] | Qwell: of course, i would prefer to have it route through my own network so as to not pay out the nose for LD |
| 20:58.27 | [hC] | Qwell: and i guess i would need to be approved by the payphone companies or something, otherwise anyone could just do it. |
| 20:58.30 | Qwell | [hC]: buy one, read it, see what it says :) |
| 20:58.38 | Qwell | (a card, that is) |
| 20:58.45 | [hC] | indeed. |
| 20:59.04 | Qwell | maybe it's got a toll-free phone number embedded onto it too |
| 20:59.24 | [hC] | maybe wikipedia has something on calling card technology or something |
| 20:59.26 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583099.dsl.bell.ca) |
| 21:00.00 | Qwell | [hC]: might ask Strom_C too |
| 21:00.09 | Qwell | he may know a bit about that |
| 21:00.15 | anonymouz666 | tzafrir? |
| 21:03.16 | anonymouz666 | zaptel 1.4.10.1 just can't find my kernel-devel package |
| 21:05.28 | anonymouz666 | kernel-devel-2.6.18-53.1.14.el5 (rpm -qa output) |
| 21:05.42 | anonymouz666 | echo "You do not appear to have the sources for the 2.6.18-53.1.14.el5 kernel installed." |
| 21:06.34 | anonymouz666 | usr/src/kernels/2.6.18-53.1.14.el5 (ok and exist) |
| 21:06.38 | anonymouz666 | so WTF? |
| 21:06.39 | anonymouz666 | :D |
| 21:07.13 | zomgoblinz | symbolic link needed perhaps |
| 21:07.21 | zomgoblinz | because it might be looking in the wrong place |
| 21:07.22 | zomgoblinz | :) |
| 21:07.25 | zomgoblinz | just a thought |
| 21:07.28 | anonymouz666 | already did |
| 21:08.13 | zomgoblinz | doesn't use anything that includes RPM's as they are the devil |
| 21:10.34 | anonymouz666 | hehe |
| 21:10.37 | anonymouz666 | gotta go |
| 21:10.46 | anonymouz666 | I will this stuff another day |
| 21:10.47 | anonymouz666 | cya |
| 21:11.23 | *** join/#asterisk |omni| (n=rob@c-67-185-180-203.hsd1.wa.comcast.net) |
| 21:13.34 | LinuxMafia | hi |
| 21:14.02 | LinuxMafia | all : if i buy a iphone , can i call places for free? |
| 21:14.37 | bbryant | no |
| 21:14.43 | LinuxMafia | or i can call only whoever have iphone for free |
| 21:15.10 | LinuxMafia | bbryant, how about the second one? |
| 21:15.45 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 21:16.17 | zomgoblinz | hmm #iphone? |
| 21:16.24 | zomgoblinz | this isn't the iphone chan |
| 21:16.27 | zomgoblinz | if there is one |
| 21:16.30 | LinuxMafia | i know |
| 21:16.34 | wichitalineman | â sip to sip calls are free |
| 21:16.48 | zomgoblinz | do the new 3g iphones do sip? |
| 21:17.02 | wichitalineman | once you engage the PSTN, money becomes involved, though not always much |
| 21:17.07 | LinuxMafia | but what is voip phone ? |
| 21:17.08 | bbryant | LinuxMafia: if you can find a ip phone client for the iphone, then you can call other ip phones for free |
| 21:17.16 | acxty | hey guys I start developing with agi and php. I send some data to STDERR to verify what it is doing. I am on the CLI> and do agi debug but still don't see anything, am I doing right? |
| 21:18.50 | LinuxMafia | what is sip |
| 21:18.51 | LinuxMafia | ? |
| 21:19.00 | LinuxMafia | i am totally new |
| 21:20.01 | *** join/#asterisk zackz (n=zdz@rrcs-24-123-106-250.central.biz.rr.com) |
| 21:20.22 | ZBRA | hi LinuxMafia |
| 21:20.26 | LinuxMafia | hi ZBRA |
| 21:20.31 | ZBRA | i discovered what was going on |
| 21:20.38 | *** join/#asterisk golumn (n=acxty@201.220.132.138) |
| 21:20.39 | ZBRA | with my asterisk |
| 21:20.46 | LinuxMafia | oh cool |
| 21:20.52 | LinuxMafia | what was wrong? |
| 21:20.54 | ZBRA | it wasnt a leak of mem |
| 21:21.03 | *** part/#asterisk golumn (n=acxty@201.220.132.138) |
| 21:21.04 | jjshoe | php has stderr? |
| 21:21.13 | ZBRA | i just didnt logged on X and it works fine |
| 21:21.21 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 21:21.34 | LinuxMafia | oh you were using kde? |
| 21:21.36 | acxty | may someone help me with the Agi issue? |
| 21:21.37 | bbryant | jjshoe: yes, fwrite(STDERR, ... |
| 21:21.43 | ZBRA | sip is the protocol used for voice in asterisk |
| 21:22.16 | LinuxMafia | ZBRA, so what i have to do inorder to be able some one for free |
| 21:22.20 | acxty | I am on CLI> agi debug |
| 21:22.29 | LinuxMafia | ZBRA, or what we have to do |
| 21:22.36 | acxty | but when I write something to STDERR, I don't see anything |
| 21:23.03 | zomgoblinz | LinuxMafia, you need a sip client or ip phone app for the iphone |
| 21:23.13 | zomgoblinz | LinuxMafia, you will need to google around and see if one exists |
| 21:23.23 | wichitalineman | LinuxMafia: Session Initiation Protocol. It creates and destroys media comm. sessions (telephony, video stuff...) |
| 21:23.24 | ZBRA | you have to have a sip provider that does the service for free |
| 21:23.28 | LinuxMafia | zomgoblinz, i saw some ipphone called skype |
| 21:23.44 | zomgoblinz | LinuxMafia, that wouldn't involve asterisk tho :) |
| 21:23.50 | ZBRA | or you can connect two asterisks from different sites connected to the internet and make call between them for free |
| 21:24.11 | LinuxMafia | humm |
| 21:24.17 | zomgoblinz | LinuxMafia, you should read up on Voice Over IP first, see if there is a client |
| 21:24.25 | ZBRA | i got a cellphone that is 3G, i think they use SIP for videocalls |
| 21:24.27 | *** part/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 21:24.29 | zomgoblinz | LinuxMafia, then ask in here if you are somehow using asterisk |
| 21:24.29 | LinuxMafia | i google |
| 21:24.36 | ZBRA | sip over GPRS |
| 21:24.50 | LinuxMafia | zomgoblinz, i want to insatll it , that is why i am asking |
| 21:25.07 | zomgoblinz | LinuxMafia, first make sure you have a sip client for the iphone |
| 21:25.20 | zomgoblinz | LinuxMafia, otherwise installing asterisk will be useless with the phone |
| 21:25.41 | LinuxMafia | i know i have agika or something like that |
| 21:26.06 | [hC] | ekiga is not an iphone sip client. |
| 21:26.43 | zomgoblinz | [hC], thanks [hC], you know of any so he can be pointed in the right direction? |
| 21:26.49 | [TK]D-Fender | zackz, in public. |
| 21:27.03 | [hC] | oh. you're saying ip phone, not iphone. |
| 21:27.10 | [TK]D-Fender | zackz, pastebin everything remotely related to your queue scenario. |
| 21:27.11 | *** join/#asterisk ManxPower (n=manxpowe@adsl-157-36-195.msy.bellsouth.net) |
| 21:27.17 | [hC] | he doesnt mean iphone, he means ip phone |
| 21:27.17 | LinuxMafia | i check wiat |
| 21:27.29 | zackz | alright |
| 21:27.30 | zomgoblinz | ah |
| 21:27.36 | zomgoblinz | that makes more sense |
| 21:27.37 | zomgoblinz | :) |
| 21:27.53 | zomgoblinz | ManxPower, good evening :) |
| 21:28.33 | LinuxMafia | zomgoblinz, http://www.canadacomputers.com/index.php?do=ShowProduct&cmd=pd&pid=007834&cid=828.480 |
| 21:28.48 | LinuxMafia | zomgoblinz, is that gonna work with asterix? |
| 21:28.55 | zomgoblinz | LinuxMafia, looking |
| 21:29.02 | zomgoblinz | LinuxMafia, no |
| 21:29.16 | zomgoblinz | LinuxMafia, try looking on asteriskguru.com for a list of phones |
| 21:29.19 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 21:29.22 | zomgoblinz | not skype |
| 21:29.28 | zomgoblinz | they will be called sip phones |
| 21:29.42 | zomgoblinz | LinuxMafia, the cheaper ones are probably Grandstreams |
| 21:29.46 | zomgoblinz | and they run 50 |
| 21:29.53 | zomgoblinz | although a lot of people hate them |
| 21:29.58 | LinuxMafia | zomgoblinz, sip phones |
| 21:30.09 | zomgoblinz | a decent IP Phone will run about 100 USD |
| 21:30.12 | zomgoblinz | roughly |
| 21:30.25 | LinuxMafia | is there any here --> http://www.canadacomputers.com/index.php?do=ShowProdList&cmd=pl&id=828.480 |
| 21:30.40 | Qwell | ~phones |
| 21:30.41 | jbot | i heard phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 21:31.27 | zomgoblinz | LinuxMafia, not that I would buy, you should read what Qwell just linked |
| 21:31.37 | zomgoblinz | Qwell, thaks |
| 21:31.39 | zomgoblinz | thanks* |
| 21:31.50 | LinuxMafia | oh thanks |
| 21:32.02 | LinuxMafia | polycom |
| 21:32.12 | LinuxMafia | let me see if any store has it |
| 21:32.32 | Qwell | LinuxMafia: check telephonydepot |
| 21:32.42 | Qwell | I hear they're good/cheap |
| 21:32.47 | bkruse | Does anyone know of an example to for a call to answer on a SIP phone from manager? |
| 21:32.49 | Qwell | (and deliver to Canada) |
| 21:33.02 | Qwell | bkruse: wha? |
| 21:33.02 | bkruse | Eg be able to force my polycom to pickup the call, via a manager request (or make it seem so) |
| 21:33.07 | LinuxMafia | Qwell, they are in canada? |
| 21:33.20 | Qwell | LinuxMafia: US I think |
| 21:33.36 | LinuxMafia | Qwell, i need canada :( |
| 21:33.50 | zomgoblinz | Qwell, you know of any example or howto's for the Authentication method? |
| 21:33.59 | bkruse | Qwell: When I see a call to my phone via manager, I want to be able to control it via manager. I know I can redirect -> hangup (to hangup). Redirect -> voicemail. But answer, I am not sure about that. |
| 21:34.01 | LinuxMafia | i dont have CC card , i have to buy by person |
| 21:34.02 | zomgoblinz | i was looking around and couldn't stumble across anything |
| 21:34.48 | colinm_ | bkruse: You could make one of the line appearances auto-answer and redirect to that appearance... |
| 21:35.22 | colinm_ | bkruse: If you can set Alert-Info with manager, you could also do it that way, but I don't use manager |
| 21:35.38 | bkruse | colinm_: I want it to be phone independent, so I guess there is no universal solution. The only things I could do for all call types (zap/iax/sip) would be to ignore (hangup) and redirect to voicemail |
| 21:35.47 | zackz | [TK]D-Fender: here it is http://pastebin.com/m48a1de98 |
| 21:35.52 | bkruse | colinm_: herm.... Can I do that to an already ringing call, however |
| 21:35.53 | zomgoblinz | LinuxMafia, paypal perhaps? |
| 21:36.00 | zackz | if it's not in there, i have no clue because those are the only files ive modified |
| 21:36.08 | LinuxMafia | i hate paypal , i can not use it |
| 21:36.41 | LinuxMafia | so Grandstream are not good right? |
| 21:36.44 | acxty | I am trying to capture a channel variable usign php and agi. I am doing this $client = fread(STDIN, "GET VARIABLE var"); but I am getting 510 Invalid or unknow command. |
| 21:36.53 | acxty | may someone help me with that? |
| 21:36.58 | Morrocco | Hi Guys, I downloaded snaps dialer for Asterisk, I can get it to dial from my desktop but when I lift up the phone it automatically drops the call, I read that the context might be wrong, but where do I see that in my asterisk config. |
| 21:37.01 | colinm_ | Grandstreams should come as the prize in cereal boxes. |
| 21:37.04 | [TK]D-Fender | zackz, exten => 15240,1,AddQueueMember(support|Local/${EXTEN:1}@agents|900) |
| 21:37.22 | [TK]D-Fender | zackz, where is your [agents] DIALPLAN CONTEXT ? <--------------- |
| 21:37.30 | zackz | in agents.conf |
| 21:37.36 | [TK]D-Fender | zackz, WRONG |
| 21:37.39 | zomgoblinz | colinm_, I have used grandstreams, thus far they have worked better than wireless IP phones |
| 21:37.46 | [TK]D-Fender | zackz, Local = DIALPLAN |
| 21:37.55 | colinm_ | zomgoblinz: Tin cans work better than wireless IP phones. |
| 21:38.05 | [TK]D-Fender | zackz, that is of course what I was sure of from the start. |
| 21:38.13 | zackz | ok |
| 21:38.15 | Morrocco | zackz, is agents.conf for me? |
| 21:38.23 | zackz | no |
| 21:38.25 | zackz | sry |
| 21:38.29 | [TK]D-Fender | zackz, You had a very fundamental misunderstanding of the "decive" you were adding as a member. |
| 21:38.34 | dlynes_office | Morrocco: no |
| 21:38.37 | [TK]D-Fender | device* |
| 21:38.37 | Morrocco | ok |
| 21:38.45 | Morrocco | anything for me guys? |
| 21:38.53 | dlynes_office | Morrocco: have you tried looking on the snapzilla website for answers to that one? |
| 21:39.00 | Morrocco | yes |
| 21:39.12 | zackz | [TK]D-Fender: so can you actually add agents from agent.conf dynamically to a queue? |
| 21:39.17 | Morrocco | they suggets something but it does not work |
| 21:39.39 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-66-31.bstnma.east.verizon.net) |
| 21:39.47 | *** part/#asterisk inja_y (n=Babbel@ip70-174-148-215.dc.dc.cox.net) |
| 21:39.50 | dlynes_office | Morrocco: can you do core set verbose 100, and then pastebin a copy of the log when you dial out using snap, and then pick up the phone? |
| 21:40.04 | dlynes_office | Morrocco: i.e. a log of the call getting dropped? |
| 21:40.08 | Morrocco | cool |
| 21:40.09 | Morrocco | sec |
| 21:40.21 | [TK]D-Fender | zackz, I'm not sure how you'd "log them in" per-se. that measn adds a literal dialplan entry as an agent. I believe you can tell addqueuememeber to COUNT them as an agent for the sake of logging. |
| 21:40.28 | dlynes_office | Morrocco: i'm not familiar with how snap works, but i'm sure it's just another sip client |
| 21:41.27 | zackz | [TK]D-Fender: I mean, normally you would use AgentLogin and it would log the agent in, but you have to define the agent as a member=> of each queue you want them to be in right? |
| 21:41.31 | Morrocco | dlynes_office, I am using putty to connect to my asterisk server, how can I do this you are saying? |
| 21:41.53 | dlynes_office | Morrocco: core set verbose 100 |
| 21:41.54 | zackz | [TK]D-Fender: I'm not understanding how to dynmaically add them to the queue |
| 21:42.15 | Morrocco | dlynes_office; I dont think its a SIP client, because it asks for username and password for the asterisk admin |
| 21:42.35 | dlynes_office | Morrocco: then you'll have a bunch of stuff on the screen from the call...you can copy and paste the whole enchilada to pastebin.ca |
| 21:42.39 | Morrocco | [admin@localhost asterisk]$ core set verbose 100 |
| 21:42.39 | Morrocco | -bash: core: command not found |
| 21:42.52 | dlynes_office | Morrocco: from the asterisk prompt, not the linux prompt |
| 21:43.14 | bkruse | Morrocco: asterisk console..... |
| 21:43.17 | Morrocco | the computer does not have a monitor |
| 21:43.20 | dlynes_office | Morrocco: make sure your back buffer is a decent size in putty, too |
| 21:43.21 | Morrocco | hehe |
| 21:43.21 | bkruse | asterisk -r |
| 21:43.28 | Morrocco | ok |
| 21:43.29 | Morrocco | sec |
| 21:43.32 | [TK]D-Fender | zackz, "agentlogin" is an "on-call" kind of way to be in the queue. this requires you to actually sit in an active channel and accept an incoming call prompted by a beep. |
| 21:43.33 | dlynes_office | Morrocco: /usr/sbin/asterisk -r |
| 21:43.36 | bkruse | dlynes_office: that always gets me, ugh. When I have to use putty |
| 21:43.43 | *** part/#asterisk jsmith-teaching (n=jsmith@72.21.36.138) |
| 21:43.46 | [TK]D-Fender | zackz, check the instructions on addqueuemember again |
| 21:43.55 | dlynes_office | bkruse: putty's not so bad, if you're stuck using a windows machine |
| 21:44.08 | dlynes_office | bkruse: it's a hell of a lot better than windows telnet |
| 21:44.08 | Nugget | telnet is eeeeeeevil! |
| 21:44.13 | Morrocco | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 21:44.14 | Morrocco | hehehe |
| 21:44.25 | dlynes_office | Morrocco: run /usr/sbin/asterisk -r as root user |
| 21:44.29 | Morrocco | is it because this machine is using asterisknow? |
| 21:44.45 | rob0 | can sip debug be set/unset for CIDR netblocks? I don't want debug on my ATAs in the LAN, just for remote hosts, and probably not all of those at the same time. |
| 21:44.49 | bkruse | dlynes_office: Totally..agreed :] |
| 21:44.49 | dlynes_office | Morrocco: your main asterisk is process is running as root user; that's why you need to run /usr/sbin/asterisk -r as root user |
| 21:45.04 | Morrocco | ok |
| 21:45.08 | Morrocco | let me log back in |
| 21:45.16 | dlynes_office | Morrocco: nothing to do with it using asterisk now |
| 21:46.33 | zackz | [TK]D-Fender: yes, addqueuemember only adds an interface, but I need to be able to use Agents for logging, and addqueuemember for queues because i dont want the agents to autoanswer, bah im confused, wish AgentCallbackLogin still worked |
| 21:46.50 | Morrocco | ok, I think I dont have a password set for the root user, let me check my GUI |
| 21:47.02 | adeel | rob0, i don't think you can do a CIDR block...i've never tried |
| 21:47.05 | bkruse | So there is not really a way (Just through manager), to answer a call (after the phone is already ringing) |
| 21:47.07 | [TK]D-Fender | zackz, I believe it still would, and not sure how that carries to 1.6 being deprecated and all.. |
| 21:47.12 | dlynes_office | Morrocco: ask in #asterisknow to find out what the default root user password is |
| 21:47.21 | [TK]D-Fender | zackz, but yes "agentcallbacklogin" was the 1.2 way. |
| 21:47.29 | [TK]D-Fender | zackz, might work properly for you. |
| 21:47.33 | adeel | rob0, but you can give it a shot |
| 21:47.46 | rob0 | yeah, I did, didn't like it |
| 21:48.28 | zackz | [TK]D-Fender: it's not in my version of * anyway |
| 21:48.34 | rob0 | set sip debug peer peername (but the problem is, I don't think my peernames are working) |
| 21:48.46 | adeel | rob0, you could write a simple script to enable sip debug for the range manually....using a for loop |
| 21:49.02 | adeel | rob0, use 'sip set debug ip foo' if you know hte ip |
| 21:49.27 | rob0 | That's just it, not sure, not seeing the connections come in at all. |
| 21:49.29 | [TK]D-Fender | zackz, I don't see a callback attempt in there. |
| 21:51.19 | adeel | rob0, might be better to run tcpdump then |
| 21:51.28 | zackz | [TK]D-Fender: i tried it before but took it out because it was not even recognized |
| 21:51.40 | rob0 | aha yes, that might do what I need |
| 21:51.59 | rob0 | I can at least limit by udp port and interface |
| 21:51.59 | [TK]D-Fender | zackz, try again and we'll see |
| 21:53.18 | zackz | [TK]D-Fender: wow it did work |
| 21:53.29 | zackz | i suppose i dont want to use it though if it will be deprecated |
| 21:55.58 | zomgoblinz | I get warnings with: Received trunked frame before first full voice frame |
| 21:55.59 | zomgoblinz | on my IAX trunk |
| 21:56.11 | zomgoblinz | and then the jitterbuffer drops out |
| 21:56.19 | zomgoblinz | when I do iax2 show netstats |
| 21:56.24 | bkruse | zomgoblinz: what is your network type? |
| 21:56.25 | zomgoblinz | it's not longer buffering |
| 21:56.26 | bkruse | how bad/distance |
| 21:56.33 | zomgoblinz | satellite |
| 21:56.41 | bkruse | lol |
| 21:56.51 | zomgoblinz | ku band and c-band |
| 21:56.52 | zomgoblinz | :) |
| 21:56.59 | bkruse | wow. |
| 21:57.12 | adeel | that's not going to be fun |
| 21:57.21 | zomgoblinz | no it's not :p |
| 21:57.34 | zomgoblinz | is it even possible to buffer a trunk over satellite? |
| 21:57.37 | zomgoblinz | with huge spikes |
| 21:57.49 | adeel | zomgoblinz, only if you don't mind doing half-duplex conversations |
| 21:57.57 | adeel | like the way walkie-talkie's work |
| 22:16.34 | *** join/#asterisk dlynes (n=chatzill@S01060016b68219f1.vs.shawcable.net) |
| 22:17.45 | zomgoblinz | adeel, hmm ok, what are the requirements for asterisk to buffer? ie, rtt length etc |
| 22:17.55 | zomgoblinz | and how much jitter will it accept |
| 22:19.28 | adeel | zomgoblinz, you should read the docs on the jitterbuffer stuff...it's more of what the rfc's dictate |
| 22:22.19 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.141) |
| 22:23.36 | frogonwheels | Simple scenario - call somebody over sip to landline - transfer to meetme conf. call another and transfer to conf. then join conference. |
| 22:24.03 | jblack | does anyone happen to know offhand which modules the rhino r1t1 requires? |
| 22:24.24 | frogonwheels | I can hear both parties - but they can't hear each other / and the 2nd just gets bursts of talking.. |
| 22:25.01 | zackz | anyone install HUDLite before? |
| 22:25.13 | frogonwheels | I was reading somewhere about an issue with mismatched buffer lengths in meetme.. but assuming 1.4.10 would have fixed this? |
| 22:27.49 | frogonwheels | hello |
| 22:28.19 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585313.dsl.bell.ca) |
| 22:28.32 | frogonwheels | (ah that's right, it's the openwrt bot who responds to hello ;)) |
| 22:28.45 | frogonwheels | ~thebook |
| 22:28.45 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 22:30.03 | *** join/#asterisk lirakis (n=user@cpe-68-175-38-65.nyc.res.rr.com) |
| 22:31.49 | *** part/#asterisk lirakis (n=user@cpe-68-175-38-65.nyc.res.rr.com) |
| 22:34.50 | jblack | tries to remember who suggested the rhino to him, because he's _pissed_ at him |
| 22:37.05 | *** join/#asterisk talntid (n=eric@66.208.251.170) |
| 22:37.09 | jblack | heh |
| 22:37.17 | talntid | ;) |
| 22:37.22 | talntid | grabs some popcorn |
| 22:38.12 | talntid | rhino r1t1 != plug&play. heh |
| 22:38.33 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 22:40.39 | *** join/#asterisk C4colo (n=DJpyro@66.185.107.14) |
| 22:40.56 | C4colo | anyone here know what 488 not acceptable here would indicate? codecs? |
| 22:41.02 | C4colo | anyone here know what 488 not acceptable here would indicate? codecs? |
| 22:41.06 | C4colo | ack |
| 22:41.09 | C4colo | sorry |
| 22:41.29 | C4colo | spazzed out on the keyboard for a sec |
| 22:41.42 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:42.01 | talntid | What codec are you using? |
| 22:42.09 | talntid | If you switch to g711 does it work properly? |
| 22:42.11 | C4colo | that is the question |
| 22:42.30 | riddlebox | is there anything I can do so that when someone calls on my zap line, I dont hear myself talk to them? |
| 22:42.47 | C4colo | it is an SS7 gateway that was swapped out recently, when talking to the call router it tries to swap from slin to ulaw |
| 22:44.06 | [TK]D-Fender | C4colo, 488 = codec mismatch |
| 22:47.51 | C4colo | hmm |
| 22:48.02 | C4colo | that's what I thought, but I am assured that ulaw is enabled |
| 22:48.11 | C4colo | could mid-stream codec swapping cause a 488? |
| 22:52.40 | *** join/#asterisk Niteon (n=rod@200-236-34-118.cable.tche.br) |
| 22:53.08 | Niteon | anybody from Brazil wanting a full time job? |
| 22:53.24 | Niteon | we need Asterisk developers |
| 22:56.55 | gitguy | Niteon: hi, how much do you pay? |
| 22:59.57 | *** join/#asterisk nocif (n=nocif@196.203.193.91) |
| 23:00.25 | nocif | hi there needing help with analog to h323 gateway |
| 23:11.17 | *** join/#asterisk TrentCreek (n=trent@cpe-70-116-105-45.rgv.res.rr.com) |
| 23:13.55 | *** join/#asterisk rdgr (n=rich@jwad-resnet-31477.d.port.ac.uk) |
| 23:21.19 | jblack | Dear Digium: Loading a download page up with a pile of javascript is retarded. |
| 23:21.36 | jblack | expectially when it's on a site named "ftp.digium.com" that doesn't support ftp. |
| 23:22.09 | JT | agreed |
| 23:22.50 | *** join/#asterisk dexpdx (n=dexpdx@66-162-134-242.static.twtelecom.net) |
| 23:24.37 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 23:24.52 | *** join/#asterisk gitguy (n=diego@adsl-159-128.click.com.py) |
| 23:27.21 | jblack | Don't worry. The rhino guys are trying to outdo them on brazenly incompetence |
| 23:27.42 | nocif | any one can help me with h323 related matter |
| 23:28.09 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 23:29.46 | ManxPower | jblack: it's for "marketing to get download stats" |
| 23:30.07 | JT | ftpd/httpd logs don't work? |
| 23:30.17 | jblack | It also renders lynx unusable for when one is trying to do a remote install. |
| 23:30.51 | *** join/#asterisk edge (n=edge@h75-100-218-214.lncswi.dsl.dynamic.tds.net) |
| 23:30.52 | *** join/#asterisk dexpdx (n=dexpdx@66-162-134-242.static.twtelecom.net) |
| 23:30.57 | dexpdx | quick question: if I transfer a call from a Zap channel using the Dial command |
| 23:31.01 | ManxPower | JT: apparently not. |
| 23:31.15 | dexpdx | then catch the h extension should ${CHANNEL} be the channel that hungup |
| 23:31.15 | ManxPower | There was a thread about it on the mailing lists when the first removed FTP |
| 23:31.31 | dexpdx | trying to figure out "WHO HUNGUP" from the dialplan |
| 23:31.48 | jblack | Wait. Are you actually serious? |
| 23:32.49 | dexpdx | jblack: me? |
| 23:32.51 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 23:34.03 | riddlebox | is there anything I can do so that when someone calls on my zap line, I dont hear myself talk to them? |
| 23:34.17 | ManxPower | riddlebox: yes |
| 23:34.26 | dexpdx | riddlebox: echo canceling |
| 23:34.35 | jblack | dexpdx: nah |
| 23:34.52 | riddlebox | ManxPower, I have run fxotune, and put echocancel = yes, and echo training on as well |
| 23:35.00 | coppice | earplugs? |
| 23:35.14 | dexpdx | exposure to a really really load speaker? |
| 23:35.20 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:35.27 | riddlebox | it seems to last about 10-15 seconds when the call starts, then as it goes on it seems to fade away |
| 23:35.46 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-23-185.lns10.syd7.internode.on.net) |
| 23:38.15 | dexpdx | ok so it looks like the ${CHANNEL} is always the caller (not the callee) in the h extension |
| 23:38.21 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:38.22 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.141) |
| 23:45.07 | JT | riddlebox: hwec |
| 23:46.25 | jblack | This is cute. From what I can tell of the makefile, Rhinos don't work with anything newer than 2.6.22 |
| 23:47.55 | JT | how convenient |
| 23:51.52 | jblack | Note to the Rhino guys: You could have saved me three hours if you had taken the time to put in a text file that says "We don't care enough to support anything post 2.6.22" |
| 23:53.39 | C4colo | kernel 2.6.22? |
| 23:53.41 | C4colo | zaptel? |
| 23:53.44 | C4colo | asterisk? |
| 23:53.53 | C4colo | lol |
| 23:54.04 | C4colo | nevermind |
| 23:56.37 | dexpdx | anyone worked out hte problem of detecting who hangup? |
| 23:57.33 | file | jblack: someone from Rhino hangs out in here... but I can't remember his nick |
| 23:58.10 | jblack | Yeah. I know who |
| 23:58.48 | riddlebox | hrmm this echo only seems to happen when a cell phone calls |