| 00:00.32 | drummond_ | good luck ;-) |
| 00:00.49 | drummond_ | i hate writing documentation |
| 00:00.58 | gitguy | 10 of 33 people already registered |
| 00:01.06 | gitguy | but they aren't using it well yet i think |
| 00:01.11 | gitguy | yeah i hate too |
| 00:01.40 | gitguy | english is not even my first language which makes it worse |
| 00:02.18 | drummond_ | ;-) |
| 00:03.15 | booray | so... asterisk on 64 bit? better? worse? |
| 00:03.54 | DragoraN | anyone has idea? |
| 00:04.07 | drummond_ | does anyone know if there is a way to tie a call path togerther with the cdr data.....i want to be able to follow a call as it goes throuh dialins, transfers, parks, ietc......? |
| 00:09.22 | *** join/#asterisk moy (n=moyhu@189.169.83.74) |
| 00:09.51 | DragoraN | can i register sip server in extensions.conf ? |
| 00:12.23 | Maliuta | DragoraN: no, that's what sip.conf is for |
| 00:13.56 | booray | exit |
| 00:13.58 | booray | crap |
| 00:14.04 | booray | not a terminal window |
| 00:19.03 | *** join/#asterisk talntid (n=t@66.208.251.170) |
| 00:20.19 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 00:28.34 | DragoraN | Maliuta: yes, but i need to register dynamically |
| 00:29.19 | Maliuta | DragoraN: the fact still remains that you can't register a sip session in the dialplan |
| 00:29.41 | Maliuta | DragoraN: exactly what does "register dynamically" mean? |
| 00:30.12 | DragoraN | like you said, register in dialplan |
| 00:30.22 | DragoraN | or change my sip extension which iam registered to |
| 00:31.22 | Maliuta | that makes no sense |
| 00:32.02 | Maliuta | you want to change the extension a sip device (phone or soft phone) is connected to mid session? |
| 00:33.41 | DragoraN | yep |
| 00:35.58 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177584428.dsl.bell.ca) |
| 00:42.05 | *** join/#asterisk fedya (n=fedya@245.208.8.67.cfl.res.rr.com) |
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| 00:45.20 | Maliuta | is it a softphone? |
| 00:45.40 | drummond_ | when asterisk makes a connection to mysql, does it leave the connection open, or does it open it, write the record, then close it? |
| 00:46.17 | drmessano | Why do you want to change mid session? |
| 00:46.36 | gitguy | what is a good way to scale asterisk |
| 00:46.44 | gitguy | can i make it work in a distributed way? |
| 00:46.44 | drummond_ | was that directed at me? |
| 00:47.02 | Maliuta | DragoraN: I mean it just sounds like something that is a) stupid and b) not needed |
| 00:47.16 | gitguy | like, store all the users in a database and put asterisk on different servers, and make it a cluster |
| 00:47.29 | gitguy | ? |
| 00:47.39 | gitguy | a cluster of it |
| 00:47.40 | drmessano | drummond_: Sense No Make Idea Your |
| 00:48.21 | drummond_ | i mean, does mysql log into the database and log out everytime it writes cdr data? |
| 00:52.21 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-997732697c73189c) |
| 00:54.55 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 00:55.24 | apocn | Hello, I've been trying to setup MOH, when I make the call it says it started playing, but I cant hear nothing on the phone. |
| 00:55.28 | apocn | using Asterisk 1.4.4 |
| 00:55.36 | apocn | any hints? |
| 01:03.03 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:09.15 | *** join/#asterisk s0lid (n=s0lid@210.213.198.7) |
| 01:12.54 | gitguy | can i forward calls, eg if someone dials my internal number, and if i don't answer to forward it to my cellphone? |
| 01:14.06 | Maliuta | gityguy: yes, it's a simple thing to do in the dialplan |
| 01:14.08 | drmessano | 1.4.4 jeez |
| 01:14.19 | gitguy | Maliuta: ok i'll check the book |
| 01:14.38 | gitguy | asterisk rocks |
| 01:15.14 | gitguy | i want to end up replacing the old pbx system at my boss office |
| 01:16.14 | gitguy | he said he already loves it |
| 01:16.31 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) [NETSPLIT VICTIM] |
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| 01:16.32 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
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| 01:16.32 | Maliuta | gitguy: where you have the Dial() for the device you want simply put an entry below it as the next number to dial the other number |
| 01:17.57 | gitguy | Maliuta: ok |
| 01:18.00 | gitguy | thanks |
| 01:18.27 | Maliuta | exten => 1000,1,Dial(PROTO/DEVICCE,15) |
| 01:18.27 | Maliuta | exten => 1000,2,Dial(PROTO/CELL) |
| 01:18.54 | Maliuta | of course you tweak the params to the dial() and I you want some other fall through stuff |
| 01:18.56 | gitguy | oh |
| 01:19.12 | gitguy | nice |
| 01:19.12 | gitguy | :) |
| 01:19.14 | gitguy | thanks |
| 01:24.55 | gitguy | how can i configure the number of rings? |
| 01:25.13 | drmessano | It's in the book |
| 01:25.16 | gitguy | ok |
| 01:30.29 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177584428.dsl.bell.ca) [NETSPLIT VICTIM] |
| 01:40.19 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
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| 01:52.30 | *** join/#asterisk icemakev (i=pure@p15106004.pureserver.info) |
| 01:52.36 | icemakev | hello |
| 01:52.47 | icemakev | is someone here ? i need urgent help please |
| 01:53.19 | Maliuta | nobody is here, we are all out to lunch |
| 01:53.28 | icemakev | :) |
| 01:53.40 | icemakev | Maliuta please i need help .. |
| 01:53.40 | Maliuta | and as we know time is an illusion, lunch time doubly so |
| 01:53.42 | icemakev | urgent .. |
| 01:53.55 | Maliuta | have you tried asking the question? |
| 01:54.04 | Maliuta | people will respond if the can/want to |
| 01:54.12 | icemakev | <PROTECTED> |
| 01:54.15 | icemakev | <PROTECTED> |
| 01:54.20 | icemakev | [May 7 21:15:57] WARNING[9375]: pbx.c:2405 __ast_pbx_run: Channel 'SIP/callcentric-08c5f800' sent into invalid extension 's' in context 'default', but no invalid handler |
| 01:54.25 | icemakev | [May 7 21:15:57] NOTICE[9375]: pbx_spool.c:351 attempt_thread: Call completed to SIP/callcentric/0013602276263 |
| 01:54.42 | icemakev | whats the problem here ? |
| 01:54.44 | Maliuta | that is not a question, and please don't paste spam |
| 01:54.52 | Maliuta | ~patebin |
| 01:54.53 | jbot | patebin is, like, your friend |
| 01:54.59 | Maliuta | ~pastebin |
| 01:55.00 | 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 |
| 01:55.13 | icemakev | ok |
| 01:55.19 | unpaidbill | is there an easy way t olog all activities done i nthe manager interface? |
| 01:55.52 | icemakev | so whats the problems here? how i can solve this problems ? |
| 01:56.04 | Maliuta | icemakev: please formulate a question that give us some idea of the situation, setup and problem ... then we may be able to help |
| 01:56.19 | icemakev | ok |
| 01:56.23 | icemakev | so , |
| 01:56.29 | icemakev | im using asterisk 1.4 |
| 01:56.38 | icemakev | i had configured it . |
| 01:56.42 | icemakev | its working. |
| 01:56.46 | icemakev | its calling. |
| 01:56.47 | Maliuta | obviously not |
| 01:57.04 | icemakev | im using a sip |
| 01:57.04 | Maliuta | and you can put more than 2-3 words per line |
| 01:57.20 | mwalling | Maliuta: he did just put 4 :P |
| 01:57.23 | icemakev | ok :) sorry :) |
| 01:57.26 | ectospasm | "asterisk 1.4" doesn't usually mean much... what specific version of asterisk are you using? Current shipping (without me checking) is 1.4.19.1 |
| 01:57.50 | icemakev | asterisk-1.4.0-beta3.tar.gz |
| 01:59.17 | ectospasm | icemakev: that's a beta version... we can't support that! |
| 01:59.24 | icemakev | i use a sip . to make calls. its working but its showing this erors Channel 'SIP/callcentric-08c5f800' sent into invalid extension 's' in context 'default', |
| 01:59.28 | ectospasm | ...beta of an old release even! |
| 01:59.52 | icemakev | so you can help me or not ? |
| 01:59.57 | ectospasm | icemakev: looks like you don't have an s extension |
| 02:00.12 | Maliuta | 1.4.0-beta is old. I would start by running something newer, and stable |
| 02:00.18 | ectospasm | I'd suggest using the latest 1.4 (1.4.19.1) |
| 02:00.28 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 02:00.42 | icemakev | hm . now i must use this . and when will work i will change this version. |
| 02:00.58 | icemakev | what i must modify to correct this eror ? |
| 02:00.58 | Maliuta | ectospasm: or he's not registered to the <callxetric> setup in his sip.conf |
| 02:01.20 | icemakev | im registred |
| 02:01.25 | icemakev | asterisk -r |
| 02:01.27 | icemakev | sip reload |
| 02:01.32 | Maliuta | ectospasm: we have no knowledge of his sip setup or dialplan, it could be anythign |
| 02:01.33 | icemakev | and its registrated |
| 02:02.13 | icemakev | Maliuta i setup the sip like i saw on callcentric website. |
| 02:02.34 | icemakev | and i dont think there is the error |
| 02:02.48 | ectospasm | icemakev: pastebin your configs, and we'll look |
| 02:03.07 | icemakev | here ? or where ? |
| 02:03.15 | Maliuta | which means nothing, and is this error occurring on inbound or outbound calls? |
| 02:03.19 | Maliuta | ~pastebin |
| 02:03.19 | 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 |
| 02:03.33 | icemakev | Maliuta : outbound calls |
| 02:04.05 | Maliuta | would need info on your sip.conf and extensions.conf |
| 02:04.06 | icemakev | i use this sip to make outbound calls . asterisk is makeing the calls. but i recive that error |
| 02:04.44 | icemakev | can i paste here extansins.conf ? is not so much and reallt i need urgent help please Maliuta |
| 02:04.58 | icemakev | 5 lines are in extensions.conf |
| 02:05.01 | ectospasm | no, don't paste here |
| 02:05.08 | ectospasm | OVER 3 LINES ^^^ |
| 02:05.10 | Maliuta | _pastebin_ |
| 02:05.17 | Maliuta | ffs |
| 02:05.21 | icemakev | ok .... |
| 02:05.27 | Iamnacho | LOL |
| 02:05.48 | icemakev | Maliuta this was my first problem second one :P |
| 02:05.53 | *** join/#asterisk erojasv (n=erojasv@190.40.65.8) |
| 02:07.06 | icemakev | i use Gnudialer. to make this calls. but that dont work . i can make calls with asterisk . but when i try to create a campain with gnudialer... gnudialer dont connect at asterisk .. or i dont know but the calls are not made... have any ideea ? without tell me to go on gnudialer channel :P |
| 02:09.35 | drmessano | Go to the gnudialer forum |
| 02:09.43 | drmessano | gnudialer isn't supported here |
| 02:10.19 | *** join/#asterisk kn0x (n=nynex@c-67-167-25-223.hsd1.il.comcast.net) |
| 02:10.31 | icemakev | drmessano . from 1 week i keep visit gnu forum .. and all the stufs and i cant fix this problem :( |
| 02:10.33 | kn0x | hey anyone know of a good reference for MGCP stack? |
| 02:10.56 | Maliuta | icemakev: take the gnudialer issues to #gnudialer, no supporty here |
| 02:11.16 | *** join/#asterisk putnopvut (n=putnopvu@c-71-228-178-34.hsd1.al.comcast.net) |
| 02:11.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 02:11.28 | drmessano | icemakev: I am not unsympathetic to your problem, but not being able to get support for a product is not an excuse to go offtopic in another |
| 02:11.43 | drmessano | You can get help on the asterisk bits here.. |
| 02:11.50 | drmessano | But gnudialer is gnudialer |
| 02:12.28 | Maliuta | thinks this is not the place for tea and sympathy |
| 02:14.51 | *** join/#asterisk Frogzoo (n=Frogzoo@121.218.37.48) |
| 02:21.54 | *** join/#asterisk ManxPower (n=manxpowe@97-81-241-251.dhcp.lds.al.charter.com) |
| 02:25.45 | eppigy | GNUDIALER YOU SAY? |
| 02:26.03 | icemakev | yes |
| 02:26.41 | *** join/#asterisk justdave_ (n=dave@unaffiliated/justdave) |
| 02:27.53 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:27.53 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:28.50 | eppigy | you basicaly need to google asterisk contexts |
| 02:29.06 | eppigy | and set an s extension |
| 02:29.15 | *** join/#asterisk giesen (i=giesen@dirtypackets.net) |
| 02:29.23 | eppigy | or change your gnu dialer setting to conform to your current dial plan |
| 02:29.28 | giesen | Anyone here ever integrated asterisk with a cisco call manager express? |
| 02:29.47 | icemakev | eppigy from where ? ;) that is the problem |
| 02:30.17 | eppigy | the gnudialer conf file i would assume |
| 02:30.39 | eppigy | there is a README and INSTALL file that come with gnu dialer |
| 02:30.59 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 02:32.14 | icemakev | eppigy i read them from a milion of times trust me |
| 02:32.35 | icemakev | im not sleeping from 1 week to make this work . |
| 02:32.49 | icemakev | and look im stuck on the final step .. |
| 02:34.18 | eppigy | how many agents will be using this dialer? |
| 02:34.56 | icemakev | for the first time i will use it. and then i will add the agents.. |
| 02:35.19 | eppigy | What kind of campaigns will you run? |
| 02:35.53 | icemakev | polical , schools... and others.. |
| 02:35.55 | icemakev | first i want to make this run . and then i will add agents at it.. |
| 02:36.29 | icemakev | eppigy can you help me .. ? |
| 02:37.44 | eppigy | Thats crazy talk |
| 02:38.04 | icemakev | why ? |
| 02:40.50 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 02:57.18 | ftp3 | if i want someone to be able to call into my asterisk sip, via a web client, is there an open source or free solution for this? What I want is, them to be able to call my company.. I am not providing long distance or anything.. so, i just want something simple, they install and use on their end that does not require anything from me except me giving them my sip@ip |
| 02:57.22 | ftp3 | does this exist? |
| 02:58.01 | *** join/#asterisk nn (n=kaeru@unaffiliated/nn) |
| 02:58.11 | Maliuta | any softphone will do that |
| 02:58.24 | Maliuta | it's not really an * problem, it's a client selection |
| 02:58.38 | nn | i know this is a bit mis...placed? but im trying to fix it so it works with *... linksys RT31P2-NA, seems vonage molested..please how do i restore to a normal firmware? |
| 02:58.58 | *** join/#asterisk _mm_ (n=mmclain@cpe-67-49-233-178.dc.res.rr.com) |
| 02:59.10 | ftp3 | yes, it is not a problem ;-).. but, how would they configure a softphone without me providing some sort of service? |
| 02:59.39 | Maliuta | ftp3: that depends on the softphone. it's not * related |
| 02:59.49 | Maliuta | and hence is OT |
| 03:02.49 | ManxPower | nn: I'm sure there is info on unlocking the RT31P2-NA on the internet. |
| 03:03.21 | ManxPower | Historically, service provider locked ATAs are very hard, many times impossible to unlock. |
| 03:04.29 | nn | ManxPower: the rly crap point is this was fully customer purchased equip. |
| 03:04.44 | nn | Got vonage service and they forced a fw upgrade on it and now... useless. |
| 03:04.55 | *** join/#asterisk wglenncamp (n=wglennca@c-68-63-248-45.hsd1.ky.comcast.net) |
| 03:05.21 | ManxPower | nn: It sucks to be you. |
| 03:05.40 | nn | Yes. I am saddened. Like a panda...touched by the zoo keeper |
| 03:05.41 | wglenncamp | Is there a way to lengthen the amount of time a blind transfer rings before it times out? |
| 03:05.57 | wglenncamp | Right now it timesout after about 30 seconds ringing, and just drops the call |
| 03:09.32 | ManxPower | wglenncamp: remove the ,30 from your Dial lines |
| 03:09.49 | ManxPower | If you don't have that, then it must be the phone that's doing it, not Asterisk. |
| 03:10.20 | _mm_ | is away. Automatically set away [SZon] |
| 03:11.38 | wglenncamp | okay, I'll check it. Thanks |
| 03:21.57 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 03:34.18 | gitguy | some guys are telling me to install asteriskgui on my server |
| 03:34.34 | ManxPower | gitguy: Nobody here, I hope. |
| 03:34.56 | gitguy | ManxPower: i don't want to |
| 03:35.15 | gitguy | i try to stay away from it |
| 03:35.23 | ManxPower | Then don't. We certainly won't help you, that's a job for #asterisk-gui |
| 03:35.28 | Maliuta | gui's are evil |
| 03:37.22 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:37.42 | gitguy | gui's can suck my git |
| 03:40.48 | gitguy | ManxPower: there was just one of my teammates on my IM saying that he thinks guis are better... and i hope he wont tell my boss that i'm a cli guy, my boss will probably think "he is old, he likes cli, gui is better" |
| 03:45.27 | gitguy | meh, nevermind |
| 03:48.02 | gitguy | http://pastie.caboo.se/193432 |
| 03:48.04 | gitguy | that's my current config |
| 03:49.57 | gitguy | it works pretty damn well |
| 03:51.42 | plik | gitguy: GUIs are like a caveman language where you have to grunt (click) and wave your hands (mouse) at the computer until it does what you want. (imagine the mouse pointer as your hand, then say "ugh" every time you click to mean "ME WANT THAT".) if whoever wrote the gui didn't add a feature that you can specifically communicate with an exact grunt and wave sequence, you're out of luck and have to ask the developer to add that. |
| 03:52.19 | gitguy | plik: yeah i know, i'm not a gui fan |
| 03:52.22 | gitguy | plik: cli ftw |
| 03:52.22 | Maliuta | gui's are the realm of the click monkey, a clueless beast that lack knowledge |
| 03:52.34 | plik | git - tell the boss |
| 03:52.48 | gitguy | plik: i will, i already told him once |
| 03:53.05 | gitguy | plik: he told me to use gui, i told him that i would not use it |
| 03:53.11 | gitguy | plik: and i told him i wanted to use cli |
| 03:53.17 | gitguy | plik: he said "do it whatever you want" |
| 03:53.19 | Strom_C | I'm glad to see that none of you are opinionated zealots |
| 03:53.22 | gitguy | however you want* |
| 03:53.29 | plik | ah well... some will never learn, it just sucks when they're in authority |
| 03:54.01 | gitguy | but there is some ubuntu guy saying that i should have used gui :P |
| 03:54.12 | gitguy | meh |
| 03:54.25 | Maliuta | gitguy: I'd be asking why he would want to limit the conf options you could use and the possible methods of producing a solution, hence limiting the final solution |
| 03:54.39 | gitguy | yeah |
| 03:54.49 | gitguy | you don't even have the chance to ssh and modify the config files? |
| 03:54.53 | gitguy | on asterisknow |
| 03:54.57 | Maliuta | gitguy: tell the gui lover to go away, get a clue and then come back to you |
| 03:54.57 | gitguy | the gui will get screwed i think right? |
| 03:54.59 | Qwell | you can with asterisknow |
| 03:55.07 | Qwell | that's about the only one.. |
| 03:56.09 | gitguy | cli for life |
| 03:59.03 | gitguy | Maliuta: lol yeah |
| 04:01.50 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:02.03 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:04.53 | drmessano | heh |
| 04:05.21 | drmessano | "CLI OR YOU ARE A DUMB NT ADMIN FOR LIFE" |
| 04:05.30 | gitguy | lol |
| 04:05.36 | gitguy | "nt admin" |
| 04:05.38 | gitguy | thats sad |
| 04:05.57 | Maliuta | windows and sysadmin don't go in the same sentence |
| 04:06.05 | gitguy | yeah |
| 04:06.17 | Maliuta | I don't "do" windows |
| 04:06.28 | drmessano | I much prefer "Use whatever fits the task, however, GUI's aren't supported here" |
| 04:06.33 | JT | yay for zealotry |
| 04:06.38 | drmessano | But then again, i'm not narrow minded |
| 04:09.56 | *** join/#asterisk syntaxman (i=wade@kernel-panic/member/syntaxman) |
| 04:12.22 | gitguy | would it be better to put those users on a postgresql/mysql database? |
| 04:12.22 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 04:12.33 | gitguy | i actually prefer plain text for simplicity |
| 04:12.51 | gitguy | but i'm curious if i will get some benefit |
| 04:12.57 | Maliuta | gitguy: ldap? integrate it with the rest of your infra |
| 04:13.07 | gitguy | ldap sounds good |
| 04:13.24 | Maliuta | start doing auth/mail etc from ldap |
| 04:13.36 | Maliuta | makes it easier to admin |
| 04:16.23 | gitguy | will check that soon |
| 04:21.52 | Maliuta | ldap rocks |
| 04:23.18 | gitguy | you can unify all the user's information right that right? |
| 04:23.38 | gitguy | s/right/with |
| 04:23.58 | Maliuta | yeah |
| 04:24.21 | gitguy | nice |
| 04:24.26 | Maliuta | you configure the systems to query ldap instead of the usual files |
| 04:24.51 | Maliuta | it's very flexible, and heaps of stuff can use it to get config data |
| 04:25.27 | gitguy | cool |
| 04:25.39 | Maliuta | including * |
| 04:25.59 | gitguy | can you use that with /bin/login and stuff |
| 04:26.06 | gitguy | with pam i believe right? |
| 04:26.58 | Maliuta | yeah pam-ldap and nss-ldap |
| 04:27.26 | Maliuta | postfix and dovecot for mail, samba ... the list goes on |
| 04:28.41 | gitguy | nice |
| 04:32.42 | gitguy | well, time to sleep... see ya guys ;) |
| 04:36.20 | *** join/#asterisk JunK-Y (n=junky@modemcable153.55-201-24.mc.videotron.ca) |
| 04:38.05 | drmessano | Might be a good idea to learn asterisk first, then integrate it with your intraborg |
| 04:39.56 | rob0 | resistance is futile |
| 04:40.58 | *** join/#asterisk steliosk (n=Stelios@athedsl-310554.home.otenet.gr) |
| 04:43.52 | *** join/#asterisk za3toor (n=non@bas1-toronto12-1088934529.dsl.bell.ca) |
| 04:43.58 | za3toor | hey guys... |
| 04:44.30 | za3toor | quick question.... does the incoming call forward feature from a voip provider work on the astrieknow? |
| 04:44.50 | za3toor | because i have been trying for matched and unmatched... and still now luck |
| 04:46.17 | za3toor | i guess everyone is still sleeping |
| 04:47.44 | ManxPower | za3toor: We don't use AsteriskNOW here. Try #asterisknow |
| 04:47.56 | za3toor | everyone is dead |
| 04:47.58 | za3toor | there |
| 04:48.10 | za3toor | ok well forget that i say astrisknow |
| 04:48.20 | za3toor | can i atleast get an idea which files do i need to edit |
| 04:48.24 | za3toor | and what needs to be in them? |
| 04:48.24 | ManxPower | Then your question makes no sense. |
| 04:48.41 | ManxPower | "incoming call feature from a voip service provider"? |
| 04:48.47 | za3toor | ignore my first one and think about my second one |
| 04:48.53 | za3toor | ok... |
| 04:49.02 | za3toor | for outgoing calls |
| 04:49.10 | ManxPower | Either the call is received by Asterisk or it's not. Asterisk doesn't care (and usually doesn't even know if the call was forwarded) |
| 04:49.23 | za3toor | i have extentions using a voip prvoider to make the out going call |
| 04:49.39 | ManxPower | Where is Asterisk in this? |
| 04:50.03 | za3toor | ok... i like that... how would if the call is recieved or not by astrisk? |
| 04:50.29 | za3toor | how would i know if the call is received or not by astrisk |
| 04:50.43 | za3toor | is there anything that i can type to know exactly what is going on in the CTI? |
| 04:50.50 | za3toor | CLI? |
| 04:51.05 | ManxPower | Asterisk is not a PBX. It does not have (except for zap channels) call forwarding feature. You would build that feature yourself in the dialplan, usually using some sort of database (built in or exernal) |
| 04:51.32 | ManxPower | The CLI shows you what exactly is going on. That is what it is for. |
| 04:51.50 | ManxPower | Asterisk is a PBX Toolkit that lets you build your own PBX. |
| 04:52.35 | za3toor | IC what i u mean |
| 04:52.45 | ManxPower | For the 2nd time, where is Asterisk in all this. You say phones and service provider, but where is Asterisk? |
| 04:52.45 | drmessano | asplodes |
| 04:53.29 | za3toor | i am trying soo hard to understand your question to give you a good answer... but i still don't understand |
| 04:53.45 | za3toor | what do you mean where is Asterisk? |
| 04:54.45 | Maliuta | how is asterisk involved in this call process |
| 04:55.17 | za3toor | well i thought that Asterisk is what forwards the call |
| 04:55.38 | za3toor | because when i type sip show registry in the CLI |
| 04:55.54 | za3toor | it shows that my sip services are registerd |
| 04:55.55 | the_5th_wheel | Hi. I have a Cisco 7961 with a sip firmware, and it seems to just cut incoming calls randomly, and every now and then i see this error in my logs |
| 04:55.58 | the_5th_wheel | May 8 06:45:33 WARNING[8793]: chan_sip.c:10535 handle_response: Host '192.168.2.165' does not implement 'INFO' |
| 04:56.24 | za3toor | when i try to make an outgoing call... it works great |
| 04:56.32 | the_5th_wheel | (wich is the sip phone in question. It can make outgoing calls happilly) |
| 04:56.43 | za3toor | but when i try to make an incoming call... things go wrong |
| 04:57.04 | za3toor | so i don't know how to see what happends to the call when i am in Asterisk |
| 04:57.22 | ManxPower | SIP Phone --> NAT Router -> Internet -> Service Provider -> PSTN. |
| 04:57.27 | ManxPower | I see no Asterisk in that diagram. |
| 04:57.38 | Maliuta | za3toor: what happens with an incoming call? does it come into asterisk? does it make it into the correct context? is that context configured for that call to be sent to another device registered with *? |
| 04:57.56 | lanning | za3toor, look for DID setup using VoIP providers |
| 04:58.17 | za3toor | lanning: can i get a link? |
| 04:58.31 | Maliuta | the_5th_wheel: I have a 7941, and I don't get that (and it's the same firmware) |
| 04:58.49 | lanning | usually provided by the VoIP provider. |
| 04:59.03 | ManxPower | INFO is a SIP thing, it can be used to transmit DTMF out of band. |
| 04:59.23 | Maliuta | the_5th_wheel: how have you got the phone setup? |
| 04:59.36 | ManxPower | It's pretty much been replaced with RFC2833, which is a way to send DTMF using RTP. |
| 05:00.52 | the_5th_wheel | I manually configured it, it has a direct connection to the asterisk server, No special configs. I think im going to try and reflash it, because i cant access the web interface either |
| 05:00.57 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 05:01.52 | ManxPower | the_5th_wheel: change dtmfmode=info to dtmfmode=rfc2833 in sip.conf |
| 05:03.06 | Maliuta | dtmf_inband: "1" |
| 05:03.38 | Maliuta | in the SIPDefault.cnf the gets tftp'd to the phone on boot |
| 05:04.35 | the_5th_wheel | Maliuta: could you perhaps send me your tftp files, just without the firmware? I workder if i didnt perhaps break something in that with my configs, since i built them from scratch |
| 05:08.09 | Maliuta | the_5th_wheel: http://pastebin.ca/1011418 |
| 05:08.17 | the_5th_wheel | Thanks |
| 05:09.25 | the_5th_wheel | I get a blank page? |
| 05:10.15 | Maliuta | and http://pastebin.ca/1011420 |
| 05:10.19 | Maliuta | they both work |
| 05:11.16 | the_5th_wheel | mmm, lemme try this on a pc on a different network |
| 05:12.45 | znoG | is there such thing as "md5secret" in iax.conf like there is in sip.conf? |
| 05:13.04 | the_5th_wheel | mmm, odd, i cant get it working from ZA, but when i access it from a us webserver, all is fine.. |
| 05:13.10 | the_5th_wheel | anyhow, thanks |
| 05:20.45 | *** join/#asterisk shital (n=shital@122.167.80.93) |
| 05:20.50 | ManxPower | dtmf_inband will only work if you are using the ulaw or alaw codec |
| 05:20.55 | shital | Hello All |
| 05:22.55 | shital | After installing full Asterisk, is it necessary to have packages like GCC, make etc..(needed for compiling) in the system to run Asterisk? |
| 05:23.13 | shital | please can any 1 tell me? |
| 05:25.47 | ManxPower | Service Pack 3 for XP resets your default media player back to Windows Media Player |
| 05:25.56 | ManxPower | shital: no |
| 05:26.27 | ManxPower | You may need a few libraries, but you don't need the compilers, build tools, etc |
| 05:27.02 | shital | Thank you ManxPower |
| 05:27.41 | shital | you mean to say asterisk supporting libraries rt? |
| 05:28.33 | ManxPower | rt? |
| 05:28.57 | shital | right? |
| 05:29.59 | ManxPower | Correct. |
| 05:30.49 | shital | ManxPower i need to have very basic Linux OS which runs finely, on which i need to install my full Asterisk what can i do for this? |
| 05:31.36 | shital | i mean to create an embedded system like product |
| 05:31.46 | shital | can you help me please? |
| 05:31.46 | rob0 | Stripping down a GNU/Linux to bare essentials is a challenge for experienced admins. |
| 05:32.28 | rob0 | You should hire an experienced admin to help design and test the embedded system. |
| 05:32.30 | shital | rob0 : so is there existing such distros now? |
| 05:33.49 | shital | what may be the size of such bare essential distro? |
| 05:36.25 | drmessano | An embedded asterisk install should never begin with "hold my beer for a minute..." |
| 05:36.38 | JerJer | i have made a very severely limited embedded linux system weigh in at 11 megabytes |
| 05:36.46 | ManxPower | drmessano: No, that is the last words of a redneck |
| 05:37.08 | rob0 | The kind of questions you are asking makes it appear unlikely that you could design a working embedded system. |
| 05:38.29 | drmessano | Stripped down Windows 98 with 98lite, AsteriskWin32... |
| 05:38.31 | shital | Jerjer: good |
| 05:39.35 | shital | Jerjer: can you please tell me the packeges u selected for this? |
| 05:41.03 | JerJer | the stock linux kernel+uclibc+busybox+init scripts+lighthttpd+openssl-dev+libncurses+zlib-dev+asterisk |
| 05:41.14 | JerJer | i think that's pretty close |
| 05:41.22 | JerJer | been awhile though |
| 05:42.06 | JerJer | oh - not -dev the actual binaries |
| 05:42.10 | shital | ok, good Jerfer, can i boot this to Intel based system as well? |
| 05:42.18 | JerJer | haven't slept in a while either |
| 05:42.38 | JerJer | shital: you can boot this in any processor linux supports |
| 05:43.00 | drmessano | JerJer: can you do this for me? |
| 05:43.03 | shital | ok Jerfer: thank you very much |
| 05:43.04 | drmessano | JerJer: Now |
| 05:44.16 | JerJer | perhaps if I had like 5 grand :) |
| 05:47.08 | JerJer | i almost released my embedded 'distro' - but Kiehofner launched ast-linux, so i just started using ast-linux for most of my stuff - let someone else have the headache of managing distro |
| 05:48.07 | JerJer | check that: Keilhofner |
| 05:48.28 | shital | Jerfer: all the packages u mensioned above are binaries . m i correct? |
| 05:54.49 | *** join/#asterisk ussrback (n=chatzill@213.200.9.98) |
| 05:54.53 | ussrback | hi alll |
| 05:55.23 | ussrback | please help me im looking for a auto dialer for asterisk. |
| 06:00.54 | TrentCreek | <PROTECTED> |
| 06:01.16 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 06:01.57 | jameswf-home | tzafrir tzafrir_home |
| 06:02.51 | ManxPower | ussrback: Most of the people on this channel are not willing to help with autodialers |
| 06:03.05 | *** join/#asterisk apollonx (n=admin@193.19.189.38.STATIC.ISP.KZ) |
| 06:03.30 | ussrback | ok i see but, can u suggest any tool which can read from excel file and dial on this numbers |
| 06:03.33 | TrentCreek | obvious reason why |
| 06:03.48 | Maliuta | well excel will read excel files |
| 06:04.11 | Maliuta | as for autodialers .... DIAF |
| 06:04.35 | ussrback | i need this tool to dial on different test numbers and measure IP SLA on my VoIP GW |
| 06:04.47 | ussrback | i am not going to call people and bother them |
| 06:07.01 | *** join/#asterisk HumanCell (n=HumanCel@207.224.214.130) |
| 06:07.16 | ussrback | anyone can suggest some sip tool which can read specified numbers (from excel, txt, mdb) and call throught the asterisk? |
| 06:07.55 | HumanCell | Question: I'm running v1.4.18 ... curious how I set the outbound caller id based on the user ... |
| 06:08.14 | HumanCell | ... are there links that anyone recommends? |
| 06:08.28 | rob0 | "core show application set" |
| 06:10.54 | *** join/#asterisk g0d^ (n=bob@fl-65-41-9-15.sta.embarqhsd.net) |
| 06:10.54 | rob0 | Set(CALLERID(all)="human name" <NXX-NXX-XXXX>) |
| 06:11.25 | HumanCell | rob0: thanks ... I sawa this, but wasn't sure exactly where in the dialplan to put this ... |
| 06:11.26 | rob0 | variables can be used |
| 06:12.18 | rob0 | you probably want it somewhere before the outbound Dial command :) |
| 06:12.55 | ManxPower | Don't use quotes in callerid |
| 06:13.02 | HumanCell | For example, my current dialplan starts with : exten=_91XXXXXXXXXX,1,Macro(trunkdial,${trunk_3}/${EXTEN:1},${trunk_3_cid}) |
| 06:13.07 | ManxPower | a few phones will reject it |
| 06:13.42 | ManxPower | HumanCell: A more correct one would be _91[2-9]XX[2-9]XXXXXX |
| 06:13.43 | rob0 | Set(CALLERID(all)=human name <NXX-NXX-XXXX>) <== works? |
| 06:13.51 | ManxPower | don't use dashes either |
| 06:14.03 | rob0 | Set(CALLERID(all)=human name <NXXNXXXXXX>) <== works? :) |
| 06:14.20 | ManxPower | rob0: no, but if you put real numbers in there it would. |
| 06:14.30 | rob0 | of course, thx |
| 06:14.45 | rob0 | fixes dialplan |
| 06:14.49 | ManxPower | Many systems/devices/carriers are pretty liberal about what they accept in the callerid they receive, but not all. |
| 06:15.20 | HumanCell | So I'm trying to understand Macros ... and where exactly I would place that ... or would it be the first line in my dialplan? |
| 06:15.40 | ManxPower | certain Cisco phones/firmware version ignore the callerid info if it has quotes in it. One carrier I've worked with rejected the call if you had dashes in the callerid number |
| 06:16.19 | rob0 | I would do the set before or as part of the macro. |
| 06:16.21 | ManxPower | HumanCell: If you find yourself doing the same or similar dialplan for many of your extensions, you might want to put that in a macro |
| 06:16.48 | HumanCell | Ok ... I see ... now how do I reference the callerid that I set on the user? |
| 06:16.48 | ManxPower | Most of my macros check channel variables for their config information |
| 06:17.08 | ManxPower | HumanCell: EXACTLY the same way as in the dialplan |
| 06:17.49 | HumanCell | So for each user, the cid_number is set to their direct number ... |
| 06:17.54 | ManxPower | as rob0 says you would set the callerid before you go into the macro |
| 06:18.17 | HumanCell | ... I"m confused about how I reference that variable in the Set statement ... |
| 06:18.30 | ManxPower | Ah, so you would use the standard ${CALLERID(num)} but you knew that because you read channelvariables.txt, right? |
| 06:19.37 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:21.20 | HumanCell | is doing another google search ... |
| 06:22.17 | ManxPower | HumanCell: try reading extensions.conf.sample first, then looking on the wiki. |
| 06:22.17 | rob0 | sounds like a filesystem search for channelvariables.txt might be more to the point |
| 06:22.35 | ManxPower | The wiki is a terrible reference of anything that's covered in the Asterisk docs, but it can be useful to see things like examples of usage |
| 06:22.43 | HumanCell | rob0: I did that first ... but this install has been cleaned ... |
| 06:22.53 | ManxPower | (hint) It's in the Asterisk source dorectory |
| 06:23.44 | HumanCell | ManxPower: source got wiped from this box already ... it's been working for a while with analog cards ... I just added the VoIP channels ... ;-( |
| 06:25.03 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:25.31 | ManxPower | HumanCell: download it again, the doc directory in the Asterisk source is the single best source of Asterisk informtion |
| 06:25.54 | HumanCell | So I did see the ${CALLERID(all)}), etc ... and name, and num ... but not sure how to reference a user variable yet ... |
| 06:26.32 | ManxPower | Verbose(Hello my name is ${CALLERID(name)}, what's yours?) |
| 06:26.40 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.155) |
| 06:27.00 | g0d^ | hello... i'm setting a AsteriskNOW, I want to create Local Ext Phone.. Only what NIC Card do i need to buy? |
| 06:27.10 | jameswf-home | ~asterisknow |
| 06:27.10 | jbot | methinks asterisknow is based on Asterisk, but it is not Asterisk, and it is unlikely to live up to Asterisk's standards. Only Asterisk is supported on #asterisk. Use #AsteriskNow instead. Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk |
| 06:28.07 | rob0 | what NIC to buy? That's not even on topic in #AsteriskNow |
| 06:28.39 | rob0 | Probably any NIC you can find works fine in Linux. |
| 06:29.08 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-116-41.PHNX.QWEST.NET) |
| 06:29.34 | g0d^ | Ok... but i don't know what card do i need to buy to have the phone connected to asterisknow server so i can get the Dial Tone... |
| 06:29.38 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 06:29.39 | g0d^ | i'm news on this .. |
| 06:29.42 | jameswf-home | anything realtec is usuaky safe |
| 06:30.07 | rob0 | ~fxs |
| 06:30.08 | jbot | rumour has it, fxs is foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 06:31.02 | rob0 | There are FXS cards which are internal to the computer ("zaptel"), and there are FXS ATA's. |
| 06:31.02 | jameswf-home | <cough> I hear rhino cards are nice <cough> |
| 06:31.09 | rob0 | ~ata |
| 06:31.10 | jbot | somebody said ata was Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 06:31.28 | rob0 | ~fxo |
| 06:31.29 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 06:31.50 | rob0 | (You might not need an FXO interface.) |
| 06:32.23 | rob0 | ~wiki |
| 06:32.49 | rob0 | ~wikis |
| 06:32.50 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 06:33.05 | rob0 | ~book |
| 06:33.06 | 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 |
| 06:33.58 | rob0 | At this point you have a lot of material to cover. |
| 06:34.14 | TrentCreek | destroy book |
| 06:34.26 | TrentCreek | ~detro book |
| 06:34.29 | TrentCreek | ~detroybook |
| 06:34.39 | TrentCreek | ~destroy book |
| 06:34.39 | jbot | no |
| 06:35.02 | g0d^ | hey thank you for that informations :) |
| 06:35.12 | rob0 | yw |
| 06:35.14 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.155) |
| 06:37.11 | g0d^ | rob0, bro i'm sorry.... but... |
| 06:37.18 | g0d^ | are u busy? |
| 06:40.11 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 06:49.52 | HumanCell | I know that I am completely missing something simple here on the dialplan ... I have added the line to do the Set for Caller ID, but I just don't understand what I set it to ... how I reference the cid_number set for the user making the outbound call ... ?? :-( |
| 06:52.35 | rob0 | Yeah, my dialplan is rather simple, since right now we only have ~3 inbound lines and 2 extensions. I set callerID depending on how the destination number is dialed (_NXXNXXXXXX gets one, _1NXXNXXXXXX gets another, _91NXXNXXXXXX gets the 3rd.) |
| 06:53.12 | rob0 | I set a DIALME variable which is what ends up in the actual Dial() command. |
| 06:53.28 | *** join/#asterisk shital (n=shital@122.167.107.220) |
| 06:54.40 | HumanCell | I've got three companies sharing the 8 new VoIP channels ... so I want each to be assigned their direct numbers ... which I did ... but then to set the outbound caller ID base don which extension is calling out ... |
| 07:00.34 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:01.05 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 07:02.16 | HumanCell | Here is where I'm just not understanding the model ... I can see ${EXTEN} which is defined to be the current extension ... but how do I access/reference any of the details about a extension defined in users.conf ... :-( |
| 07:02.34 | HumanCell | ... or can I? |
| 07:02.56 | *** join/#asterisk vortex` (n=vortex@202-136-108-213.static.adam.com.au) |
| 07:03.11 | vortex` | I'm so confised by the hardware for Asterisk.. |
| 07:03.42 | vortex` | All I need is something to connect to Analogue POTS, one line to begin with and possibly a second line later. Which hardware card would suit? |
| 07:03.43 | HumanCell | vortex`: I thought that was the easy part ... :-) |
| 07:03.44 | TrentCreek | aren't well all |
| 07:04.01 | rob0 | ~ata |
| 07:04.01 | jbot | ata is, like, Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 07:04.28 | TrentCreek | ~destroy ata |
| 07:04.29 | jbot | no |
| 07:05.09 | vortex` | so that's like the IAXy - S101i, right? |
| 07:05.29 | TrentCreek | VOIP lines are a heck of a lot cheaper than POTS |
| 07:05.29 | rob0 | Probably the cheapest way into this is a Linksys Sipura 3102 ATA. (not sure of exact name) |
| 07:06.01 | TrentCreek | Linksys makes ATAs based on Sipura |
| 07:06.06 | vortex` | TrentCreek: my existing landline isnt VOIP :) |
| 07:06.24 | TrentCreek | then get internet. |
| 07:06.31 | vortex` | huh? |
| 07:07.01 | TrentCreek | You can have 30 outgoing lines cheaper than one unlimited calling plan on one POT |
| 07:07.31 | vortex` | perhaps in the US |
| 07:07.31 | rob0 | Sure, you can connect your land line with FXO, but your long distance service is likely to be much cheaper using a VoIP carrier. (True in USA for sure, not sure about AU.) |
| 07:07.40 | TrentCreek | anywhere in the world |
| 07:08.05 | HumanCell | vortex`: http://www.digitnetworks.com/category_s/23.htm |
| 07:08.06 | TrentCreek | I use services that are not in the US |
| 07:08.22 | TrentCreek | sorry..I can have 30 INCOMING lines cheaper than 1 POT |
| 07:08.46 | TrentCreek | Out going is unlimited because you are paying for the call charge |
| 07:09.04 | HumanCell | ~fxo |
| 07:09.05 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 07:09.05 | vortex` | interesting. can you point me to a site with some more info about that? |
| 07:09.07 | TrentCreek | so you are limited to your bandwidth |
| 07:09.32 | TrentCreek | you want 1 line ofr multple? |
| 07:09.36 | TrentCreek | *or |
| 07:10.02 | shital | Jerjer Sir: from where i can download binaries of u mensioned packages? |
| 07:10.11 | vortex` | well i have a single line at home, so i would only need single anyway i guess. |
| 07:10.21 | shital | m trying but not getting binaries |
| 07:10.52 | TrentCreek | For AU number I only know of Voxbone, but they require $100 a month minimum |
| 07:11.13 | TrentCreek | I am not sure about a single AU line..you have to search for that |
| 07:11.28 | TrentCreek | for outgoing you can use carrixchange |
| 07:11.37 | vortex` | we're paying $20/month with telstra for a single pots line |
| 07:12.31 | TrentCreek | I pay $3.5 for a virtual num through les.net..only US, UK and Canada numbers though |
| 07:12.40 | TrentCreek | that's US dollar |
| 07:13.23 | TrentCreek | calls through carrixange are dirt cheap |
| 07:14.19 | vortex` | very interesting. looking at the voxbone site now |
| 07:14.38 | rob0 | There are several free USA origination services. My primary number is Stanaphone, never paid anything. (Unfortunately they are not taking new accounts.) |
| 07:14.48 | vortex` | ahh that makes sense now, had someone call me from a 87 prefix number in my city, it was voip :) |
| 07:15.02 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:15.54 | rob0 | Ipkall in Washington state also works well, and they do take new signups. |
| 07:16.44 | vortex` | TrentCreek: my girlfriend just reminded me our ISP does VOIP, super cheap as well. |
| 07:16.57 | vortex` | ill look into that as well. |
| 07:17.00 | vortex` | thanks for your help. |
| 07:17.18 | TrentCreek | using * is a lot cheaper |
| 07:17.39 | shital | rob0 : from where i can download uclibc and busybox executables? |
| 07:18.09 | shital | can u please help me? |
| 07:18.10 | TrentCreek | besides when things go wrong, * tells you what is going on |
| 07:18.33 | rob0 | http://sweet.nodns4.us/ |
| 07:28.45 | *** join/#asterisk DaPrivateer (n=matt7229@66.92.79.218) |
| 07:32.26 | dominic1 | Is it possible to build a queue overflow? Queue has two static agents. If both are talking, can I tell the queue that he should dial to another extension? |
| 07:36.14 | *** join/#asterisk justdave_ (n=dave@unaffiliated/justdave) |
| 07:37.15 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 07:38.04 | unpaidbill | ~astdoc |
| 07:41.31 | HumanCell | Question: My dial plan has the following line, but it is not passing on the callerid of the extension, even though it is larger than 6 digits ... exten=s,1,set(CALLERID(all)=${IF($["${LEN(${CALLERID(num)})}" > "6"]?${CALLERID(all)}:${ARG2})}) |
| 07:42.00 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:42.28 | *** join/#asterisk eXistenZ (n=pectic@unaffiliated/existenz) |
| 07:44.48 | *** join/#asterisk DaPrivateer (n=matt7229@66.92.79.218) |
| 07:47.12 | HumanCell | rob0: ManxPower: thanks for the help ... got it ... |
| 07:49.49 | *** part/#asterisk HumanCell (n=HumanCel@207.224.214.130) |
| 07:50.36 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 07:53.06 | ftp3 | hello.. my music on hold mp3 doesnt sound as great on my ivr as when i just play it.. actually, quite a bit worse.. are their some guidelines for moh mp3s for asterisk? |
| 07:56.50 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 07:57.32 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 07:58.07 | *** join/#asterisk gr0mit (n=tim@dhcp4.zuk40.mot-tools.co.uk) |
| 07:59.39 | *** join/#asterisk cancerea (i=FaisalKH@bravo.gerrys.net) |
| 08:09.14 | *** join/#asterisk Toot (n=toot@25.Red-79-153-105.staticIP.rima-tde.net) |
| 08:13.47 | *** join/#asterisk Dovid (n=Dovid@bzq-79-176-131-14.red.bezeqint.net) |
| 08:14.06 | Dovid | Best reason ever to use asterisk: http://forums.digium.com/viewtopic.php?p=70333 |
| 08:18.35 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 08:23.51 | *** join/#asterisk Frogzoo (n=Frogzoo@121.218.37.48) |
| 08:26.25 | *** join/#asterisk Nobbie (n=chatzill@fwb003.fw.is.co.za) |
| 08:26.26 | *** join/#asterisk rcy (n=rcy@S010600207810532a.vc.shawcable.net) |
| 08:28.02 | Nobbie | what cold cause my CDR Records to show as ANSWERED even if they aren't ANSWERED ? |
| 08:29.49 | mvanbaak | Nobbie: are you running Answer or some IVR on the calls ? |
| 08:31.03 | Nobbie | nope, simply a Dial() which sends the call to a SIP Trunk which is a Cisco router |
| 08:31.59 | Nobbie | using FreePBX |
| 08:32.06 | Nobbie | but this seems to be a CDR problem in * |
| 08:32.32 | Nobbie | how can one debug the CDR events ? |
| 08:34.33 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:35.28 | Dovid | Nobbie: have a look at the CLI. Maybe cisco is sending answer supervision |
| 08:38.38 | Nobbie | ah hangon, the FreePBX is set to Authenticate() for the route. will that make the disposition == ANSWERED ? |
| 08:38.50 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-187-7.hsd1.wa.comcast.net) |
| 08:39.40 | *** join/#asterisk adrin_ (n=adrin@chello089078212055.chello.pl) |
| 08:39.42 | adrin_ | hello |
| 08:40.42 | adrin_ | is there a way to define in asterisk a second sip voip provider to be used for calling when the main one fails (is unavailable) ... something like a redundant voip prov? |
| 08:41.28 | mvanbaak | Nobbie: probably |
| 08:42.11 | mvanbaak | Nobbie: yes, it will |
| 08:49.12 | *** join/#asterisk ikevin (n=kevin@2001:5c0:99b9:0:20f:b0ff:fe4a:4589) |
| 08:49.28 | ikevin | hello |
| 08:50.02 | ikevin | i've a problem with identification to the manager |
| 08:50.29 | adrin_ | another question: is it possible to route calls based on calling number? |
| 08:50.50 | ikevin | while i use action login he say me i'm identified, so while i try to see a page he say me "Nothing to see here. Move along." |
| 08:51.11 | ikevin | anyone have an idea? |
| 08:53.32 | *** join/#asterisk NielsNL (n=irc@lumensoft.demon.nl) |
| 08:57.03 | *** join/#asterisk DragoraN (n=dragoran@chello089173134033.chello.sk) |
| 08:57.04 | DragoraN | hi |
| 08:57.36 | DragoraN | is it possible to replace dialplan with big agi script? |
| 08:58.02 | Uatec | i would expect so |
| 08:58.14 | Uatec | just make your dialplan have one line consisting of a call to your agi script |
| 08:58.23 | Nobbie | Adrin: in the dialplan, you can match CLID using the format: ext/clid. ie: 911/1234 would match calling number 1234 dialing 911 |
| 08:59.14 | Nobbie | or _911/12XX to use wildcards |
| 09:00.39 | Nobbie | sorry the format is: _91X/_12XX |
| 09:01.59 | DragoraN | Uatec: ok, but what patter should i use? "_." executes twice :( |
| 09:07.36 | adrin_ | ok thanks |
| 09:07.41 | adrin_ | in extensions.conf? |
| 09:11.12 | Nobbie | Drag: you should use _X. instead of _. |
| 09:11.44 | Nobbie | _. will match on special extesnions such as timeout, hangup, invalid, etc |
| 09:13.38 | Nobbie | you will probaby see a WARNING in your asterisk log for doing that as well: |
| 09:13.47 | Nobbie | <PROTECTED> |
| 09:17.00 | Dovid | Best reason ever to use asterisk: http://forums.digium.com/viewtopic.php?p=70333 |
| 09:18.31 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:18.47 | *** join/#asterisk tingle (n=tingle@nightshade.nl) |
| 09:18.53 | tingle | hello, i just installed pbx in a flash (centos 5.1) i need to configure a HFC card for my school could you guys help me where i should start? |
| 09:19.33 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:19.49 | tingle | is this a good guide to start whit? -> http://dondisperato.blogspot.com/ |
| 09:20.02 | tingle | maybe a bit outdated |
| 09:20.20 | cancerea | hello, where can i find good cdr management for asterisk ? |
| 09:21.08 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 09:21.18 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:21.31 | Dovid | I want to create a bounty for mp3 support in asterisk (primarly for VM but for recording as well). anyone know how hard it is so I know about how much to set the bounty at ? |
| 09:24.46 | tingle | well that guide was to outdated :/ |
| 09:35.39 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 09:36.25 | Dovid | tingle: it was created for 1.4.X you can also try the wiki |
| 09:36.26 | Dovid | ~wiki |
| 09:36.37 | Dovid | ~voip-info |
| 09:36.38 | jbot | from memory, voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 09:56.37 | tingle | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 09:56.45 | tingle | does anyone know what this might be? |
| 09:57.04 | tingle | it does exist |
| 10:00.31 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 10:02.45 | tingle | could someone help me? i cant figure out why asterisk issnt starting anymore.. i installed bristuff and chan_capi both the latest one |
| 10:03.07 | tingle | this is the error msg i get when i try to restart asterisk once booted Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 10:05.03 | *** part/#asterisk BeeBuu (n=beebuu@219.132.188.25) |
| 10:07.53 | TrentCreek | i see a similar message when there is a problem with not being connected to the network |
| 10:08.03 | Dovid | tingle: why should asteisk start automatically ? do u have it set any where ? |
| 10:09.51 | *** join/#asterisk Ramasee (i=ramac@nikita.tnnet.fi) |
| 10:09.58 | tzafrir_home | Hi Ramasee |
| 10:10.14 | Ramasee | hi |
| 10:10.42 | Ramasee | yellow lights with zaptel, zaptel.conf configured wrong? |
| 10:13.11 | tingle | Dovid: yes i installed pbx in a flash (its a centos iso whit pbx) |
| 10:14.22 | Dovid | Ramsee: what kind of line ? |
| 10:14.23 | tingle | Dovid: normally after a clean install asterisk is loaded on boot as many other applications |
| 10:14.43 | Dovid | tingle: I never use the scripts. I always do it manually |
| 10:15.30 | tingle | Dovid: ok, well when i try to restart asterisk i get that error so it would be the same a starting it manually right? |
| 10:15.51 | Dovid | try asterisk -vvvvvvvvv |
| 10:15.56 | Dovid | what comes up ? |
| 10:16.37 | tingle | Dovid: i got a big output should i c/p it to pastebin? |
| 10:17.56 | Ramasee | Dovid: what do you mean with line? i'm new with pbx and voip systems |
| 10:18.33 | tingle | Dovid: asterisk -vvvvvvvvv gives this output http://pastebin.com/m36271379 |
| 10:21.31 | *** join/#asterisk shital (n=shital@122.167.85.34) |
| 10:21.51 | shital | Hello All |
| 10:22.52 | tingle | Dovid: does that look correct? |
| 10:31.59 | Dovid | tingle: seems its hanging at # |
| 10:31.59 | Dovid | <PROTECTED> |
| 10:32.47 | Dovid | u need to let it sit till u see ServerBane*CLI> |
| 10:32.56 | Dovid | ServerName*CLI> |
| 10:33.00 | Dovid | sorry about the sleeping |
| 10:33.05 | Dovid | where servername is the servers name |
| 10:33.42 | Dovid | Ramasee: is it an E1, ISDN line ? |
| 10:34.10 | tingle | Dovid: sorry i dont understand what you mean |
| 10:34.30 | Dovid | tingle: Basicly astersik didnt fully load. wait till it fully loads |
| 10:36.13 | tingle | Dovid: but i dont understand because the system is up for about 10 hours |
| 10:36.32 | Dovid | tingle running asterisk -vvvvvv would not work if asterisk was already running |
| 10:36.41 | Dovid | if u do ps aux | grep asterisk what do u get ? |
| 10:37.13 | Ramasee | Dovid: E1 |
| 10:37.16 | tingle | hmz a lot of processes but i dont think its really asterisk ill pastbin it |
| 10:37.18 | tingle | on moment plx |
| 10:37.40 | tingle | Dovid: http://pastebin.com/mfce92eb |
| 10:37.42 | Dovid | Ramasee: Have not used them much. It seems there is a problem with the line. call your carrier. See if they can do a loop back test |
| 10:38.10 | Ramasee | Dovid: how is the loop back test made? |
| 10:38.16 | tingle | Dovid: thanks for trying to help me, if been reading and trying the whole day im verry confused now and got no clue anymore haha |
| 10:38.30 | Dovid | tingle: Ok asterisk is not running now. now do asterisk -vvvvvvvvv and wait for a few minutes and see what shows up ? |
| 10:38.47 | Dovid | tingle: I am trying to see if asterisk loads properly or now |
| 10:38.48 | Dovid | not* |
| 10:39.20 | *** join/#asterisk DragoraN (n=dragoran@chello089173134033.chello.sk) |
| 10:39.20 | DragoraN | hi |
| 10:39.24 | DragoraN | can i dial from AGI ? |
| 10:40.10 | tingle | Dovid: ok cool thanks ill try it out |
| 10:40.29 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 10:40.33 | Dovid | DragoraN: Yes |
| 10:40.38 | tingle | Dovid: when i do asterisk -vvvvvvvvv i get myu prompt back is that normal? |
| 10:40.43 | Dovid | DragoraN: I persoanlly dont but I can |
| 10:40.45 | tzafrir_home | Ramasee, sorry for the delay. You have a yellow alarm? |
| 10:40.51 | Dovid | tingle: Nope |
| 10:41.05 | Dovid | asterisk is failing for some reason. enable full logging and look there |
| 10:41.15 | Dovid | in logger.conf under full take out the ; infront of it |
| 10:41.26 | cancerea | is there is any optoin to encrypt the dial plan or use it on compile time |
| 10:41.37 | tzafrir_home | Yellow alarm means that your side detects the low-level link as working OK (otherwise you'd get a red alarm), but the other side reports a problem |
| 10:41.44 | Dovid | full => notice,warning,error,debug,verbose |
| 10:41.53 | Dovid | then pb /var/log/asterisk/full |
| 10:43.12 | tzafrir_home | Ramasee, chances are something is bad with the cabling . But I really have no experince with setting up ISDN cabling |
| 10:43.39 | DragoraN | Dovid: how? i dont see dial in agi spec |
| 10:43.58 | Dovid | DragoraN: What kind of agi r u using ? php, perl ? |
| 10:43.59 | tzafrir_home | cancerea, not really. It would still show up by running 'dialplan show' |
| 10:44.09 | DragoraN | Dovid: C#, FastAGI |
| 10:44.13 | cancerea | i c, thanks tzafrir |
| 10:44.30 | cancerea | any help regarding C based agi ? |
| 10:44.40 | cancerea | i need to compile my agi in C |
| 10:44.45 | tzafrir_home | cancerea, you can do all the logic in an external AGI or application (or the ExternalIVR application) |
| 10:44.48 | Dovid | DrangoraN: I hATE M$ C#. sorry i ant help u |
| 10:45.03 | cancerea | yes tzafrir, i m doing that, |
| 10:45.09 | cancerea | rigth now they are in perl |
| 10:45.26 | cancerea | but when i compiled them in perl2exe on linux, it is taking too much of RAM |
| 10:45.32 | DragoraN | Dovid: u wont help any because of C#? |
| 10:45.37 | cancerea | each agi => 2mb process |
| 10:45.49 | tzafrir_home | cancerea, one would argue that the Asterisk dialplan is equally obfuscated ;-) |
| 10:45.56 | Dovid | DrangoraN: not at all. Since I dont like C# i dont use it and dont know how they expect u to make a call |
| 10:46.37 | Dovid | DrangoraN: If I knew how I would help ;) |
| 10:46.55 | cancerea | tzafrir i need any help in building c based agi |
| 10:48.28 | cancerea | i saw CAGI but there is no help, no examples, stuck at CAGI :( |
| 10:48.58 | Dovid | cancera: I use php. ;) |
| 10:49.01 | DragoraN | got it |
| 10:49.06 | Dovid | and dnt use fastagi |
| 10:49.23 | cancerea | php is fast ? |
| 10:49.34 | cancerea | if i create agi and then compile it with ioncube ? |
| 10:49.36 | Dovid | u can use php with fastagi but i dont |
| 10:49.51 | cancerea | ioncube->encrypt php and compile |
| 10:50.25 | cancerea | perl is doing fine for me, but i want to secure the agi |
| 10:51.07 | tingle | Dovid: im thinking about making a clean install later today (im confused and afrade i made a mess of it) do you have any good documentation |
| 10:51.38 | Dovid | tingle: sent PM |
| 10:53.31 | *** join/#asterisk wordzilla (n=me@122.109.126.193) |
| 10:54.22 | *** join/#asterisk kooskooskoso (n=k@196.211.34.2) |
| 10:55.10 | Ramasee | tzafrir_home, Dovid I contact the carrier and let see what happens, thanks for help |
| 10:55.14 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
| 10:55.29 | *** part/#asterisk Ramasee (i=ramac@nikita.tnnet.fi) |
| 10:55.32 | Dovid | Ramasee: good luck |
| 10:55.38 | Dovid | oops. he left |
| 10:55.40 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177584428.dsl.bell.ca) |
| 10:56.09 | kooskooskoso | what would the recomended PC specs be for a pbx for 40 sip phones with a 4 port bri external router |
| 10:56.20 | tzafrir_home | cancerea, there are two examples in the Asterisk source directory |
| 10:56.34 | tzafrir_home | cancerea, the AGI protocol is soemwhat inspired by CGI |
| 10:56.38 | tingle | Dovid: send you one back lx |
| 10:56.46 | tzafrir_home | Although unlike CGI it is interactive |
| 10:58.02 | cancerea | tzafrir , thanks i saw that, but its very basic |
| 10:59.19 | kooskooskoso | Would i be able to connect the following,asterisk pbx for 40 sip phones with a 4 port bri external router on a intel core 2 due 2400ghz with 8 gig ram, any help would be appreciated |
| 11:00.18 | Dovid | kooskooskoso: seems like an over kill |
| 11:00.21 | Dovid | should work fine |
| 11:00.40 | kooskooskoso | i just need it to run smoothly |
| 11:00.59 | kooskooskoso | my first installation |
| 11:01.01 | Dovid | u shouldnt have any issues. |
| 11:01.06 | kooskooskoso | thx |
| 11:01.14 | Dovid | it will take time to learn the ins and outs of asterisk but those specs should be ok |
| 11:01.20 | tzafrir_home | cancerea, could you be more specific about the issues you have? |
| 11:01.26 | Dovid | one sec. what do u mean by external 4 port BRI router ? |
| 11:01.54 | kooskooskoso | i use asterisk for fax servers my first pbx instalation i hope all goes well |
| 11:02.02 | cancerea | i need c agi, in which i can use database connectivity, billing , monitor etc |
| 11:02.15 | Dovid | kooskooskoso: faxing + asterisk can = disaster |
| 11:02.17 | kooskooskoso | 4 port isdn gateway to link isdn to server |
| 11:02.34 | *** join/#asterisk nighty^ (n=nighty@KD124213240154.ppp-bb.dion.ne.jp) |
| 11:02.55 | Dovid | so u will connect to that router via sip |
| 11:02.57 | kooskooskoso | got it working the faxing that is using fxo to email asterisk is the gateway |
| 11:03.04 | kooskooskoso | yes |
| 11:03.15 | kooskooskoso | yes via sip |
| 11:03.28 | Dovid | ok. u should be good |
| 11:03.39 | kooskooskoso | thx Dovid |
| 11:03.52 | Dovid | but faxing over sip = issues |
| 11:05.08 | kooskooskoso | havnet tried that yet with asterisk but i use multitech voip routers to route faxes over ip, got it working once or twice using aterisk with no great results |
| 11:05.17 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:05.17 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 11:11.30 | *** join/#asterisk _gm (n=gmustafa@117.20.28.50) |
| 11:11.49 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:14.36 | phix | sup |
| 11:14.59 | kooskooskoso | hi phix |
| 11:15.15 | Dovid | phix: Hi there |
| 11:15.40 | phix | ay |
| 11:15.42 | phix | what's going on ? |
| 11:15.55 | kooskooskoso | hi if you run a agi in what format must your file be on the * server |
| 11:16.13 | Dovid | kooskooskoso: what do u mean by format ? |
| 11:16.18 | Dovid | phix: nm here |
| 11:16.18 | kooskooskoso | new to agi where is the best place to start? |
| 11:16.36 | Dovid | kooskooskoso: the wiki |
| 11:16.39 | Dovid | ~voip-info |
| 11:16.39 | jbot | methinks voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 11:16.42 | Dovid | seach there for agi |
| 11:16.45 | Dovid | ~wiki agi |
| 11:16.59 | phix | kooskooskoso: umm I believe it needs to be written in a script :) or program |
| 11:17.17 | phix | kooskooskoso: you know any scripting or programming languages? |
| 11:18.01 | kooskooskoso | it says you can write agi in php, got this site wit a demo code to playback the time for you using php saved the file as php and called it from one of my sip phones, is says file cannot be found in agi directory |
| 11:18.20 | kooskooskoso | php a littlebit |
| 11:18.25 | phix | o |
| 11:18.26 | phix | ok |
| 11:18.28 | phix | that is good :) |
| 11:19.50 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:21.03 | tzafrir_home | cancerea, "database connectivity" - use a database connectivity library |
| 11:21.27 | tzafrir_home | That's nothing really specific to AGIs |
| 11:21.54 | tzafrir_home | monitoring: monitoring Asterisk from within the context of a call? Are you sure that this is what you want to do? |
| 11:22.24 | tzafrir_home | Maybe keep a separate process that monitors Asterisk through the manager interface and extracts the useful information for you |
| 11:23.09 | *** join/#asterisk bsaxon (n=bryantsa@71-8-14-108.dhcp.leds.al.charter.com) |
| 11:24.54 | *** join/#asterisk garreel (n=garreel@host16-119-dynamic.27-79-r.retail.telecomitalia.it) |
| 11:27.38 | garreel | knows any1 how to set callingpres to prohib for an outgoing channel (such as Zap/g1/...) on manager interface before using "Action: Originate" command? |
| 11:28.06 | cancerea | thanks, tzafrir_home |
| 11:28.27 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.248) |
| 11:28.48 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 11:29.20 | cancerea | i know i have to use database layer, but i need some help for CAGI...e.g find free channel originate call, pass agi values to script etc |
| 11:29.45 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:30.51 | *** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com) |
| 11:31.07 | *** part/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com) |
| 11:34.47 | *** join/#asterisk francogwapito (n=chatzill@203.82.47.128) |
| 11:37.00 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 11:38.17 | *** join/#asterisk mort_gib (n=mjensen@8.Red-81-35-165.dynamicIP.rima-tde.net) |
| 11:44.21 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 11:44.22 | *** join/#asterisk inja_y (n=inja_y@ip70-174-148-215.dc.dc.cox.net) |
| 11:44.59 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 11:46.04 | whymarkwh | since i started using linux and asterisk evertime a type something my fingers is drawen to the tab, while i am in mirc i find myself wanting to tab what i trying to say, dumb hey, results in a chuckel everthime i do it. |
| 11:48.30 | ectospasm | yeah, tab completion makes one lazy... I try to tab complete everything now |
| 11:49.03 | ectospasm | especially when I'm making a new directory or file... |
| 11:52.49 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 11:52.54 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 11:55.37 | whymarkwh | i want to create my own webpage displaying cdr records live where does one start? |
| 11:57.18 | sysreq | having your cdr stored in a db of some kind.. i.e. http://www.voip-info.org/wiki-Asterisk+cdr+mysql |
| 11:58.35 | sysreq | then it's just a matter of accessing that same db through your web interface, with whatever programming language you use |
| 12:03.13 | tzafrir_home | If you want to use sqlite: http://astcdrview.berlios.de/ |
| 12:05.52 | garreel | is there a variable to set up in manager originate action to make callerpres prohib? |
| 12:09.16 | _gm | is it possible to add/edit sip/exten/iax entries using ajam |
| 12:10.39 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:17.10 | ikevin | how can i make local number? |
| 12:17.32 | ikevin | like 800 is local phone |
| 12:17.37 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.35) |
| 12:18.07 | Mw3 | do you know anything about phoniceq bri/pri cards? they claimed to be working with asterisk |
| 12:19.50 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 12:25.42 | *** join/#asterisk inja_y (n=inja_y@ip70-174-148-215.dc.dc.cox.net) |
| 12:26.36 | *** part/#asterisk garreel (n=garreel@host16-119-dynamic.27-79-r.retail.telecomitalia.it) |
| 12:26.48 | [TK]D-Fender | ikevin: You don't "make" local numbers. |
| 12:27.18 | ikevin | ? |
| 12:28.09 | Dovid | anyone know of an SMS service where I can connect via an API and send a message ? |
| 12:28.19 | ikevin | smsbox |
| 12:28.42 | Dovid | ikevin: smsbox.com ? |
| 12:28.44 | ikevin | [TK]D-Fender, i don't understand what you would to say |
| 12:29.01 | [TK]D-Fender | ikevin: Don't worry, noone can understand your question either. |
| 12:29.11 | [TK]D-Fender | ikevin: Perhaps you could try asking it another way. |
| 12:29.21 | ikevin | Dovid, .fr |
| 12:30.04 | gr0mit | Dovid, i recommend AQL in Leeds |
| 12:31.28 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:33.39 | Dovid | gr0mit: URL ? |
| 12:33.50 | gr0mit | www.aql.co.uk |
| 12:33.55 | gr0mit | the _are_ SMS |
| 12:33.56 | Dovid | i need to have a web page grab it and i can connect to an api and send it to the, |
| 12:34.00 | Dovid | thanks |
| 12:34.13 | gr0mit | yup - they do loads of APIs for sending SMS |
| 12:34.31 | gr0mit | how many messahese /month are you needing to send? |
| 12:37.52 | Dovid | not many. emergency only |
| 12:41.46 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 12:45.39 | *** join/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 12:46.26 | gitguy | hi guys, my meetme conference pin numbers where working yesterday perfectly, but not today... it doesn't recognize the number anymore, i checked the configs and everything is ok |
| 12:47.48 | ectospasm | gitguy: is DTMF in general working? |
| 12:48.38 | gitguy | yes |
| 12:51.48 | *** part/#asterisk vortex` (n=vortex@202-136-108-213.static.adam.com.au) |
| 12:52.21 | gitguy | wtf |
| 12:52.23 | gitguy | weird thing ever |
| 12:52.32 | ectospasm | gitguy: ? |
| 12:52.35 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 12:52.49 | drmessano | Is it the pins, or is it telling you that its not a valid conference number? |
| 12:52.57 | gitguy | pin is not valid |
| 12:53.33 | *** join/#asterisk redax (i=redax@r6.hu) |
| 12:53.35 | redax | hi, |
| 12:53.38 | [TK]D-Fender | gitguy: verbose 10, pastebin the failed attempt |
| 12:54.21 | redax | is non-aswered blind-transfered calls should ring back to the transferer? |
| 12:54.55 | redax | or shall I solve somehow that in my extensions.conf ? |
| 12:56.01 | gitguy | [TK]D-Fender: http://pastebin.com/m5f36b4ba |
| 12:57.02 | [TK]D-Fender | gitguy: when you type the pin + # does it answer you immediately, or only after a pause? |
| 12:57.12 | gitguy | after a pause |
| 12:57.34 | [TK]D-Fender | redax: yes, in your extensions.conf. Go read "channelvariables.txt" in your docs folder and you be able to see how you'll have to do this. |
| 12:57.43 | gitguy | and it keeps repeating the 'is not valid' even if i don't put anything |
| 12:57.45 | [TK]D-Fender | gitguy: then I'm betting its not reading the DTMF at all |
| 12:57.56 | [TK]D-Fender | gitguy: Bad mode. |
| 12:58.09 | gitguy | [TK]D-Fender: but when i dial 90000 (which doesn't have a pin) is reading the DTFM |
| 12:58.12 | gitguy | DTMF* |
| 12:58.16 | redax | [TK]D-Fender; oh, thanks |
| 12:58.23 | [TK]D-Fender | gitguy: What are you dialing it on? |
| 12:58.25 | coppice | who reads DTMF? we usually listen to it :-\ |
| 12:58.51 | gitguy | [TK]D-Fender: xlite? |
| 12:59.13 | [TK]D-Fender | gitguy: No, Asterisk does not take that as "DTMF". You mearly told X-Lite to dial, and x-lite sent the number as the invite. that is not dtmf |
| 12:59.39 | [TK]D-Fender | gitguy: Dtmf happen after the call is connected. |
| 12:59.48 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 12:59.54 | gitguy | wtf |
| 13:00.00 | gitguy | it works with zoiper but not with xlite now |
| 13:00.08 | gitguy | [TK]D-Fender: i see |
| 13:00.12 | _gm | is users.conf supposed to be in next ABE release? any idea |
| 13:00.16 | [TK]D-Fender | gitguy: because Zoiper is clearly using the proper mode. |
| 13:00.39 | [TK]D-Fender | _gm: Last I heard, ABE is already on * 1.4 which should include it |
| 13:01.03 | gitguy | [TK]D-Fender: how do you fix that? |
| 13:01.18 | [TK]D-Fender | gitguy: Go pick the proper mode. |
| 13:01.18 | _gm | hmm ok, i didnt find users.conf though |
| 13:01.30 | gitguy | [TK]D-Fender: in the phone side, right? |
| 13:01.31 | *** join/#asterisk nighty^ (n=nighty@KD124213240154.ppp-bb.dion.ne.jp) |
| 13:01.44 | [TK]D-Fender | gitguy: Doesn't matter which so long as both sides AGREE. |
| 13:03.17 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 13:03.19 | _gm | [TK]D-Fender: we are still using asterisk 1.2.26 everywhere, my boss thinks it's much stable than 1.4 branch, i need some recommendations either he is right or we should give 1.4 a try |
| 13:03.30 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 13:03.42 | *** join/#asterisk KriS83 (n=kmarcrof@ip-213-179-129-247.llweb.de) |
| 13:03.44 | *** join/#asterisk eurodrew555 (n=andrews@83-244-164-142.cust-83.exponential-e.net) |
| 13:03.49 | KriS83 | Hi |
| 13:03.52 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-181.static.twtelecom.net) |
| 13:04.20 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:04.37 | redax | [TK]D-Fender: so if the call went to timeout, I should check ${TRANSFERSTATUS} is FAILURE and dial the extension at ${BLINDTRANSFER} |
| 13:04.49 | redax | is that correct? |
| 13:04.53 | KriS83 | Is there some trick or something one has to do to get Asterisk to signon to a jabberd server? ejabberd works fine, but Asterisk does not logon to a jabberd server... :-/ |
| 13:04.59 | eurodrew555 | I have an issue with an incoming ZAP call, the call gets answered, caller then hangs up but h extension is not called until after callee hangs up |
| 13:05.14 | *** join/#asterisk Riceri (n=Greyh@riceri.se) |
| 13:05.23 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 13:05.44 | redax | or ${BLINDTRANSFER} contains the other channel ? |
| 13:05.50 | eurodrew555 | anyone seen anything like this ... causing some issues as we are getting a carrier message being played back to the callee |
| 13:06.00 | [TK]D-Fender | redax: Go try |
| 13:06.32 | redax | yah. will do. thanks again |
| 13:07.38 | [TK]D-Fender | _gm: 1.4 works fine for jsut about everyone. |
| 13:07.43 | gitguy | [TK]D-Fender: in DTMF settings i only have: DTMF Force Send in Band: No -- DTMF Tone Lenght in Samples: 2800 -- DTMF Magic Number: 101 |
| 13:08.20 | Riceri | what do i need to do to get DND to work on a asterisk? Do i need to make a macro or something to get *78 and *79 to work? |
| 13:08.59 | intralanman | any asterisk gurus in here that would care to speak at cluecon in august? |
| 13:09.25 | [TK]D-Fender | Riceri: Depends if you expect * to manage it for you vs doing it on your phone (if capable). |
| 13:09.45 | [TK]D-Fender | Riceri: If not controlled by your phone, then yet it is your job to set it up in your dialplan. |
| 13:10.13 | [TK]D-Fender | gitguy: You've only got about 3 modes to try. GO TRY THEM. |
| 13:10.38 | Riceri | [TK]D-Fender: i got a SNOM 370 that i want to get a DND button to work. |
| 13:11.09 | [TK]D-Fender | Riceri: It probably should work right off the bat. If not, go read its manual. |
| 13:11.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:12.09 | Riceri | [TK]D-Fender: I have tryed, i will recheck and if i not solve it i will be back. Thanks! |
| 13:12.34 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 13:15.18 | gitguy | [TK]D-Fender: what do you mean with modes? i'm not sure... :S |
| 13:15.38 | [TK]D-Fender | gitguy: "dtmfmode=" <- get a clue.... |
| 13:18.05 | gitguy | ok |
| 13:18.36 | *** join/#asterisk Ipkaf (n=chatzill@dyn-91-165-223-188.ppp.tiscali.fr) |
| 13:18.39 | Ipkaf | hi |
| 13:19.01 | Ipkaf | is there anyone here ? |
| 13:19.11 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:20.12 | JayTee52 | nope, just 289 bots hanging out plotting the downfall of humans |
| 13:20.54 | [TK]D-Fender | Ipkaf: Yes, but are you expecting all 289 of us to jump up immediately and send you a personal "hi" back" :\ |
| 13:21.16 | Ipkaf | ok |
| 13:21.31 | *** join/#asterisk af_ (n=getsmart@88-149-230-0.dynamic.ngi.it) |
| 13:21.32 | M1s3ry_chocolate | hi Ipkaf :p |
| 13:22.10 | Ipkaf | i just buy a new Grandstream GS-486 |
| 13:22.26 | Ipkaf | is there anyone here got the item ? |
| 13:22.49 | [TK]D-Fender | Ipkaf: Feel free to move on and ask your actual question.... |
| 13:23.23 | Ipkaf | ok i buy this router ATA to replace my isp router |
| 13:23.29 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 13:23.41 | *** join/#asterisk mknerd (i=3f951603@gateway/web/ajax/mibbit.com/x-5ee88767632dd871) |
| 13:23.51 | mknerd | morning all |
| 13:24.17 | Ipkaf | but when i try to connec my Grandstream GS-486 router ata to an pstn line |
| 13:24.21 | *** join/#asterisk dFence (n=chatzill@p54980DF9.dip0.t-ipconnect.de) |
| 13:24.30 | dFence | morning |
| 13:24.35 | Ipkaf | it working at all to access to internet |
| 13:24.50 | Ipkaf | what i have to do ? |
| 13:24.52 | mknerd | I have to make a script on a AA50 to speak to a socket on a server, what would everyone suggest, i just discovered that perl is not going to happen |
| 13:25.43 | dFence | q: can asterisk "print" call logs via serial-line? (have to use rs232 to integrate it with our ancient billing software) |
| 13:26.01 | Ipkaf | hello |
| 13:26.47 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 13:27.16 | [TK]D-Fender | Ipkaf: Sorry you have not made a clear question out of all of that. |
| 13:27.47 | [TK]D-Fender | dFence: do you need * to send this data "live" in run-time? |
| 13:28.52 | Ipkaf | i got this router "Grandstream GS-486" i just want to replace the isp dsl router to access to internet is it possible ? |
| 13:28.58 | redax | I made a cdr-serial.c which emulates the ancient Siemens hicom outputs |
| 13:29.15 | Ipkaf | i got this router "Grandstream GS-486" i just want to replace my isp dsl router to access to internet is it possible ? |
| 13:29.19 | [TK]D-Fender | Ipkaf: Go read your GS's manual. |
| 13:29.26 | mknerd | what is anyone using for scripting on the aa50? |
| 13:29.41 | JayTee52 | Ipkaf, no the Grandstream 486 will not work in place of a DSL router |
| 13:29.44 | Ipkaf | yeah i have read |
| 13:30.06 | Ipkaf | JayTee52: why ? |
| 13:30.25 | Ipkaf | it have router capability |
| 13:31.04 | Ipkaf | so why 486 not work in place of a dsl router ? |
| 13:31.19 | [TK]D-Fender | Ipkaf: It will asct as an ethernet router if you have a DSL modem seperate from your router. |
| 13:31.57 | bobbym | hi all, i'm using an fastagi with deadagi command, this could be a problem in the future for performance issues? |
| 13:32.02 | JayTee52 | Ipkaf, what [TK]D-Fender said. |
| 13:32.13 | [TK]D-Fender | bobbym: load chan_psychic.so |
| 13:32.14 | Ipkaf | not really usefull this functionality |
| 13:32.19 | JayTee52 | Ipkaf, it will not act as a replacement for your DSL router. |
| 13:32.21 | Ipkaf | ok i understand |
| 13:32.21 | Ipkaf | thx J |
| 13:32.33 | Ipkaf | thx JayTee52 |
| 13:32.39 | JayTee52 | you can hook the WAN port to your DSL modem though |
| 13:32.52 | Ipkaf | no |
| 13:33.03 | Ipkaf | i don't have Dsl modem |
| 13:33.19 | JayTee52 | you said earlier you had DSL |
| 13:33.21 | Ipkaf | that's why i buy it to replace |
| 13:33.25 | Ipkaf | my isp router |
| 13:33.38 | [TK]D-Fender | Ipkaf: Well if you plug your phone line right into your "router" then no, the HT-486 cannot replace it. |
| 13:33.42 | Ipkaf | yeah but not working |
| 13:34.02 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:34.23 | bobbym | [TK]D-Fender: how can i get more information about chan_psychic.so? |
| 13:34.26 | Ipkaf | ok thx Tkd-fender |
| 13:34.37 | [TK]D-Fender | bobbym: Just wait for the voices to tell you more :) |
| 13:34.46 | JayTee52 | lol |
| 13:34.54 | Ipkaf | thx a lot i understand i have to buy a dsl modem |
| 13:34.59 | Ipkaf | thx a lot |
| 13:35.02 | Ipkaf | bye |
| 13:35.14 | gitguy | what is ? /var/spool/asterisk/meetme/meetme-username-99995-1.sln |
| 13:35.23 | bobbym | [TK]D-Fender: wich voices hahaha :) |
| 13:35.29 | dFence | [TK]D-Fender: jep, current MO 's like that: Siemens HiPath --- serial line --> PC(billing software), after each call hicom prints a line with the call data, i.e. dialled nr. duration extension units |
| 13:35.54 | dFence | billing software parses that string and... well.. deals with it ;D |
| 13:36.12 | [TK]D-Fender | redax: perhaps you can pass something on to dFence here... |
| 13:36.51 | [TK]D-Fender | dFence: My way would have used an SQL server and a stored procedure to call a script to output it. |
| 13:36.54 | dFence | sorry, was afk after i asked, might have missed sth |
| 13:37.36 | dFence | [TK]D-Fender: that'd be cannoning pigeons :) |
| 13:38.18 | [TK]D-Fender | dFence: Highly effective though. |
| 13:38.36 | [TK]D-Fender | pulls out an elephant guns to deal with some "ants" |
| 13:38.59 | bobbym | dFence: you are using a real time billing software? |
| 13:39.01 | dFence | i'm not too familiar with the asterisk architecture _YET_ (just ordered books for €50 to change that) isn't there a possibility to include a small perl script in an "AfterCall" routine? |
| 13:39.27 | dFence | bobbym: it's stupid, it's proprietary, it's annoying as FUCK but right now there's no way round it |
| 13:39.43 | bobbym | dFence: hahahaha |
| 13:39.57 | gitguy | [TK]D-Fender: i tried with dtmfmode=rfc2833 (that's what zoiper uses) but it didn't work for x-lite |
| 13:40.01 | bobbym | dFence: i'm looking for create one to bill some customers of mine ... |
| 13:40.30 | dFence | [TK]D-Fender: the sql-thing might become interesting once i get my boss to change the whole reservation-software (Hotel) |
| 13:41.12 | dFence | bobbym: the software itself is kinda useless since the pc with the billing-plugin has to run 24/7 in order to catch every call made |
| 13:41.14 | dFence | brb |
| 13:41.14 | *** join/#asterisk dawebber (n=dawebber@209.136.161.135) |
| 13:41.37 | [TK]D-Fender | gitguy: 2 sides, nominal # of options to tweak. This isn't Raw-Cat Science. Get to it already... |
| 13:41.56 | gitguy | what the hell |
| 13:41.57 | gitguy | sometimes works |
| 13:42.01 | gitguy | most of the time doesn't |
| 13:42.17 | *** part/#asterisk dawebber (n=dawebber@209.136.161.135) |
| 13:42.36 | coppice | sounds just like rocket science :-) |
| 13:44.02 | *** join/#asterisk nighty^ (n=nighty@KD124213240154.ppp-bb.dion.ne.jp) |
| 13:47.05 | gitguy | [TK]D-Fender: my xlite doesn't specify what dtmf mode is using it |
| 13:47.19 | gitguy | doesn't have the option for that |
| 13:47.23 | [TK]D-Fender | gitguy: Google is your friend. Use it. |
| 13:47.30 | JayTee52 | coppice, nope, this is rocket science: http://exploration.grc.nasa.gov/education/rocket/rktpow.html |
| 13:48.36 | coppice | jayTee52: that equation is far from ideal. it says we will never be able to go to the stars |
| 13:49.30 | JayTee52 | not with conventional rockets anyways but that was always a given fact considering the E=MC2 equation. |
| 13:49.46 | JayTee52 | wormholes are our only hope! :-) |
| 13:50.25 | coppice | calier to the rescue |
| 13:51.27 | coppice | we need to fold space. see Dune and Origami |
| 13:51.52 | JayTee52 | I've read Dune, the whole first series and some of the new ones by his son |
| 13:51.59 | JayTee52 | never read Origami |
| 13:52.17 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 13:53.35 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 13:53.49 | JayTee52 | I was gonna build a Quantum Membranic Temporal Transpositon device to allow instantaneous travel anywhere in the multiverse but Radio Shack is always out of the damn 5.2 Meg Ohm resistors and the 3.4 uFarad capacitors. |
| 13:54.22 | [TK]D-Fender | coppice: well... light is by nature both a particle and energy. Now consider that light is also a "wave". That implies that it is not technically going in a straight line, but rather a symmetrical path to a destination. With that in mind what we measure as the "speed of light" is the time it takes zig-zagging like that. Now the question is how far could one stretch that path towards... |
| 13:54.23 | [TK]D-Fender | ...being straight. |
| 13:56.27 | coppice | JayTee52: they never have flux capacitors in stock, either |
| 13:57.08 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 13:57.12 | heison | anyone here using openMRCP? |
| 13:58.43 | coppice | we should send a diffration limited coherent beam of [TK]D-Fender to Alpha Centauri, as a test of his ideas |
| 13:59.15 | [TK]D-Fender | coppice: Yes.... I'm barely coherent :p |
| 13:59.19 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 13:59.44 | coppice | that's true in this forum, but we can provide technical assistance for you |
| 14:02.25 | *** join/#asterisk PepOSX (n=angeldav@200.90.123.15) |
| 14:02.43 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:04.03 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 14:04.29 | *** join/#asterisk unbkbl (n=work@static-adsl201-232-12-168.epm.net.co) |
| 14:04.54 | *** join/#asterisk linuxer_igor (n=igor@mlsrj200152096p202.mls.com.br) |
| 14:05.09 | linuxer_igor | hi .. I have a litle problem |
| 14:05.35 | linuxer_igor | translation alaw to slin |
| 14:05.50 | unbkbl | hello i need to build a RJ45 cable for a E1 pri somebody know the pinout distribution? |
| 14:06.04 | linuxer_igor | my asterisk not translate codes |
| 14:06.47 | *** join/#asterisk Riceri (n=Greyh@riceri.se) |
| 14:06.50 | linuxer_igor | and return in full log WARNING[20519] translate.c: no samples for alawtolin |
| 14:07.54 | linuxer_igor | somebody have a solutions ! |
| 14:08.14 | linuxer_igor | somebody has some solution? |
| 14:08.19 | *** join/#asterisk ka0zoz (n=ka0zoz@70.58.222.123) |
| 14:11.55 | *** join/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 14:12.25 | *** join/#asterisk ka0zoz (n=ka0zoz@70.58.222.123) |
| 14:13.55 | gitguy | does dtmfmode has something to do with codecs? |
| 14:14.39 | tzafrir_home | linuxer_igor, ah, you asked here as well :-) |
| 14:14.58 | tzafrir_home | gitguy, not really |
| 14:15.05 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 14:15.18 | tzafrir_home | but inline dtmfs would not work with compressed codecs |
| 14:15.48 | gitguy | tzafrir_home: when i use dtmfmode=inband with gsm it tells me to use rfc2833, that's why i ask |
| 14:16.15 | [TK]D-Fender | gitguy: You should only use inband if you have no choice, and only with G.711 |
| 14:17.10 | gitguy | ok |
| 14:17.24 | tzafrir_home | inband sends it as DTMF tones. This works reasonably well with modest compression of alaw and ulaw. But the better speech compressions of gsm and such make detection of the DTMF tones quit difficult |
| 14:17.54 | tzafrir_home | linuxer_igor, so in short, look for other suspected messages in that area |
| 14:18.09 | tzafrir_home | It basically tells you that "it has nothing in the pipe" |
| 14:18.09 | linuxer_igor | desculpe |
| 14:18.10 | *** join/#asterisk henrique (n=henrique@unaffiliated/henrique) |
| 14:18.13 | linuxer_igor | sorry |
| 14:18.19 | unbkbl | hello i need to build a RJ45 cable for a E1 pri somebody know the pinout distribution? |
| 14:19.10 | *** join/#asterisk mogie (n=opera@ppp121-44-209-93.lns1.hba1.internode.on.net) |
| 14:19.55 | linuxer_igor | in my asterisk all codecs is runnig |
| 14:20.15 | linuxer_igor | gsm alaw ulaw slin |
| 14:20.17 | linuxer_igor | but |
| 14:21.07 | linuxer_igor | <PROTECTED> |
| 14:21.12 | linuxer_igor | when I run module load |
| 14:21.15 | tzafrir_home | This is not an issue of a missing codec |
| 14:21.17 | linuxer_igor | return mi: |
| 14:21.24 | linuxer_igor | [codec_ulaw.so] => (Mu-law Coder/Decoder) |
| 14:21.27 | linuxer_igor | but |
| 14:21.49 | linuxer_igor | don't return |
| 14:21.51 | linuxer_igor | Registered translator 'ulawtolin' from format ulaw to slin, cost 1 |
| 14:21.56 | linuxer_igor | Registered translator 'lintoulaw' from format slin to ulaw, cost 1 |
| 14:22.07 | *** join/#asterisk ka0zoz (n=ka0zoz@70.58.222.123) |
| 14:22.39 | linuxer_igor | and I don't know what is translator |
| 14:22.58 | linuxer_igor | in my full log return WARNING[20519] translate.c: no samples for alawtolin |
| 14:23.39 | linuxer_igor | Every time |
| 14:24.32 | *** part/#asterisk nn (n=kaeru@unaffiliated/nn) |
| 14:24.47 | linuxer_igor | do you understand mi |
| 14:24.49 | linuxer_igor | ? |
| 14:26.03 | linuxer_igor | <tzafrir_home> ? |
| 14:27.06 | tzafrir_home | "no samples" - nothing to work on. The translato is there. But has no work to do |
| 14:27.19 | tzafrir_home | And is actually bitching about it |
| 14:28.02 | tzafrir_home | What are those calls? From what type of phone/line? |
| 14:28.10 | linuxer_igor | yes |
| 14:28.13 | linuxer_igor | but |
| 14:28.16 | linuxer_igor | because this ? |
| 14:28.23 | linuxer_igor | <PROTECTED> |
| 14:28.47 | mknerd | what is everyone using for scripting on the AA50, I need to talk to a socket |
| 14:28.51 | tzafrir_home | This is alawtolin bitching about having no work |
| 14:29.14 | tzafrir_home | mknerd, I prefer perl |
| 14:29.26 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:29.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:29.29 | gitguy | [May 8 09:44:42] NOTICE[17416]: chan_sip.c:5481 process_sdp: No compatible codecs, not accepting this offer! |
| 14:29.32 | gitguy | i get that on xlite |
| 14:29.40 | gitguy | but on zoiper works fine |
| 14:29.44 | gitguy | well i get on the cli |
| 14:29.48 | gitguy | but i get it when using xlite |
| 14:30.17 | gitguy | disallow=all, allow=ulaw, allow=alaw |
| 14:30.19 | gitguy | that's what i have |
| 14:30.51 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:30.54 | mknerd | tzafrir_home, and how would you go about getting perl on the AA50? |
| 14:31.01 | mknerd | cause I would prefer perl as well |
| 14:31.14 | mknerd | its not a trivial processe |
| 14:31.21 | tzafrir_home | Isn't miniperl available? |
| 14:31.33 | mknerd | nope |
| 14:31.36 | mknerd | not be default |
| 14:31.41 | tzafrir_home | bah |
| 14:31.47 | mknerd | you would think for what one of these costs |
| 14:33.05 | bbryant | mknerd: i'm not sure what's on there, but some shells allow you to do socket scripting |
| 14:33.59 | mknerd | i have the choice of msh and sh |
| 14:34.18 | mknerd | they are both busybox |
| 14:35.13 | znoG | i recall reading somewhere about MeetMe choppy sound issues. This was ages ago. Anyone know anything about this? |
| 14:35.29 | znoG | I can talk to one of the participants fine if I call them directly, but using MeetMe it is choppy only for some users. |
| 14:36.32 | mogie | Anyone here in Aus? |
| 14:36.33 | [TK]D-Fender | gitguy: Look at the sip debug for your actual call. |
| 14:36.35 | bbryant | mknerd: it would be through something like this: http://thesmithfam.org/blog/2006/05/23/bash-socket-programming-with-devtcp-2/ |
| 14:37.20 | bbryant | another option would be netcat which i remember being on an older appliance |
| 14:37.42 | ajohnson | Has anyone run into problems with System() not executing system commands? |
| 14:37.55 | ajohnson | For instance it is executing echo, but not executing top |
| 14:38.09 | ajohnson | It works on Centos, but not on Debian |
| 14:38.23 | mknerd | bbyant, you do know that I do not have bash right? |
| 14:38.52 | bbryant | mknerd: yes, but bash isn't the only shell that i've used it on |
| 14:39.11 | mknerd | ok, ill see what that brings |
| 14:39.15 | mknerd | thx for the suggestion |
| 14:39.35 | *** join/#asterisk allankardec (n=root@189.13.11.9) |
| 14:40.47 | [TK]D-Fender | ajohnson: Are you providing it a full explicit path to the app? |
| 14:40.57 | ajohnson | [TK]D-Fender: Yes |
| 14:41.09 | ajohnson | and I took into account the different paths from Centos to Debian |
| 14:41.15 | [TK]D-Fender | ajohnson: And what is it doing? |
| 14:41.51 | ajohnson | [TK]D-Fender: I get a SYSTEMSTATUS that indicates it didn't error, but there is no output |
| 14:41.56 | ajohnson | I am redirecting output to a file |
| 14:42.21 | ajohnson | echo "test" > /tmp/test works, but not top -n1 > /tmp |
| 14:42.22 | [TK]D-Fender | ajohnson: And when you call it directly? |
| 14:42.23 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 14:42.25 | ajohnson | echo "test" > /tmp/test works, but not top -n1 > /tmp/test |
| 14:42.42 | ajohnson | If I copy and paste the command from the console output to the bash prompt, it executes fine |
| 14:43.18 | gitguy | Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x0 (nothing) |
| 14:43.21 | gitguy | Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) |
| 14:43.24 | [TK]D-Fender | ajohnson: pastebint he CLI output of your failed dialplan attempt at verbose 10, core debug 10 |
| 14:43.24 | gitguy | [May 8 09:58:18] NOTICE[17416]: chan_sip.c:5481 process_sdp: No compatible codecs, not accepting this offer! |
| 14:43.43 | ajohnson | [TK]D-Fender: will do |
| 14:43.52 | [TK]D-Fender | gitguy: well its clearly telling you what each side allows and they don't match. Go fix ti |
| 14:44.52 | *** join/#asterisk destructure (n=mu@rrcs-24-173-126-174.se.biz.rr.com) |
| 14:45.14 | gitguy | [TK]D-Fender: i'm dialing 99995 (a meetme conference) ... how do i fix the codec on the other side (a conference) ? |
| 14:45.36 | gitguy | nevermind |
| 14:45.37 | [TK]D-Fender | gitguy: >Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x0 (nothing) <--- NOTHING TO DO WITH MEETME |
| 14:45.58 | [TK]D-Fender | gitguy: Your phone & * don't agree. |
| 14:47.13 | mknerd | bbyant, there is not a /dev/tcp |
| 14:47.42 | gitguy | [TK]D-Fender: i see |
| 14:48.45 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 14:51.53 | mogie | any aussie here? |
| 14:52.11 | ajohnson | bah stupid pastebin |
| 14:52.13 | ajohnson | :Sorry, your post tripped our spam filter - let us know if you think this could be improved |
| 14:53.18 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 14:53.52 | [TK]D-Fender | ajohnson: use another |
| 14:54.52 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 14:54.52 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:57.16 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:57.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:57.35 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.248) |
| 14:59.08 | ajohnson | [TK]D-Fender: http://paste.uni.cc/18652 |
| 15:01.35 | [TK]D-Fender | ajohnson: Ok... I don't see it, whatever it is.. |
| 15:03.51 | ajohnson | [TK]D-Fender: You don't see the pastebin, the problem, or the commands? |
| 15:04.01 | [TK]D-Fender | ajohnson: The problem... |
| 15:04.35 | ajohnson | Ok. Gah. All I know is that it works on Centos... very frustrating. Thanks for your time |
| 15:04.47 | JT | mogie: sure |
| 15:05.00 | JT | znoG: have a zap card? |
| 15:05.22 | *** join/#asterisk railsmunky (n=nick@82-70-165-142.dsl.in-addr.zen.co.uk) |
| 15:05.45 | railsmunky | Hey people. Can anyone help me get my head round callgroups and pickup groups. |
| 15:06.02 | railsmunky | I have a requirement which i'm not 100% sure is possible |
| 15:06.06 | gitguy | [TK]D-Fender: i tried with all the possible dtmfmodes (rfc2833, inband, info, auto, etc), disallow=all, allow=ulaw,alaw, i always get "invalid pin" with xlite, sometimes work with zoiper and sometimes doesn't |
| 15:06.11 | gitguy | [TK]D-Fender: any ideas? |
| 15:06.18 | znoG | JT: yep, got a 4 port TDM card |
| 15:06.31 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:06.35 | [TK]D-Fender | pastebin everything related to your call. |
| 15:06.40 | [TK]D-Fender | gitguy: ^^ |
| 15:06.44 | JT | znoG: hmm dunno then |
| 15:06.48 | znoG | JT: i can hear most people fine .. just one person I can't.. and when that person calls me directly, it's fine. |
| 15:07.16 | gitguy | [TK]D-Fender: sip debug and verbose 10? |
| 15:08.27 | railsmunky | I have a set of 10 phones. 5 of those are in one group (1) and 5 are in another (2). So people in group one can pickup any call to the other 4 and people in group 2 can pickup any call to the other 4. We are good upto here. The requirement is such that if i get a call to a number which is assigned to a cyclic hunt group across all 10 phones any other of the 10 phones can pick it up |
| 15:10.39 | ManxPower | railsmunky: you should be able to do that on sip and zap. see callgroup= and pickupgroup= |
| 15:10.52 | ManxPower | devices can be in more than one pickup and call group |
| 15:12.26 | gitguy | [TK]D-Fender: that's my config -> http://pastie.caboo.se/193681 |
| 15:12.45 | gitguy | diego.viola is my peer |
| 15:12.52 | railsmunky | What defines a devices callgroup? Is it a call made to that device is that in that call group? |
| 15:13.06 | railsmunky | j |
| 15:13.08 | railsmunky | oops :) |
| 15:13.13 | b11d` | is it possible to just play a message to a polycom speaker, without the phone receiver being picked up or a call being acknowledged? |
| 15:13.27 | b11d` | or to turn the polycoms into a PA? |
| 15:14.18 | ajohnson | b11d`: http://www.voip-info.org/wiki/view/Polycom+auto-answer+config |
| 15:14.23 | b11d` | thank you |
| 15:14.27 | ajohnson | np |
| 15:14.53 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:16.36 | railsmunky | ManxPower: can you take a look here - http://pastebin.com/m1837042f If a call comes into the DDI number then any phone can grab it otherwise only phones on 1 or 2 can grab 1 or 2 |
| 15:16.38 | railsmunky | is that right? |
| 15:20.00 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:21.07 | [TK]D-Fender | b11d`: So... did you finally figure out your GotoIfTime issue? |
| 15:21.28 | [TK]D-Fender | gitguy: You did NOT set your dtmfmode for your peer in there... |
| 15:22.28 | b11d` | TK.. yeah I got it :)_ |
| 15:22.30 | b11d` | thanks |
| 15:22.56 | [TK]D-Fender | b11d`: Good :) |
| 15:23.01 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:23.06 | [TK]D-Fender | b11d`: Feel swilly having missed it for so long? |
| 15:23.11 | [TK]D-Fender | silly* |
| 15:23.20 | b11d` | yes.. it was stupid :) |
| 15:23.53 | *** join/#asterisk francogwapito (n=chatzill@203.82.47.128) |
| 15:24.48 | ManxPower | railsmunky: that look correctly. Keep in mind that this does not deal with extensions, this stuff deals with devices |
| 15:25.05 | ManxPower | and extension is just a number you dial to tell the PBX what device to route the call to |
| 15:25.23 | railsmunky | aha yeah - that makes sense. Thanks - i'll give it a go |
| 15:25.42 | gitguy | [TK]D-Fender: yeah is not on the paste, but i tried it already with all the dtmfmode values |
| 15:26.35 | [TK]D-Fender | b11d`: Almost worse is the fact they COULD have given it an "else" clause. |
| 15:26.57 | [TK]D-Fender | gitguy: If I don't see it, I don't trust it. And you didn't show me the call. |
| 15:27.09 | gitguy | http://pastebin.com/m3b825b0b |
| 15:27.44 | gitguy | http://pastie.caboo.se/193681 |
| 15:27.47 | gitguy | 1min |
| 15:31.05 | gitguy | http://pastebin.com/m206038f9 <-- call |
| 15:31.43 | gitguy | <PROTECTED> |
| 15:33.55 | ManxPower | ~trixbox |
| 15:33.56 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 15:34.13 | gitguy | it happens on xlite and zoiper |
| 15:35.05 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 15:35.38 | ManxPower | gitguy: do you have a pastebin of a failed call without sip debug enabled? |
| 15:36.06 | gitguy | sure |
| 15:36.27 | gitguy | 1min |
| 15:36.55 | ManxPower | Also, why do you have only one device with dtmfmode=rfc2833? |
| 15:38.42 | gitguy | because i'm testing with one device? |
| 15:38.50 | gitguy | http://pastebin.com/m70d09fff |
| 15:39.26 | ManxPower | gitguy: I suggest you only allow ulaw OR alaw, but not both |
| 15:40.13 | ManxPower | gitguy: that is more than 1 call. |
| 15:40.32 | *** join/#asterisk FinboySlick (n=FinboySl@207.134.9.196) |
| 15:40.43 | ManxPower | Set up an extension that does a READ and a Noop to show you what was read. If that works correctly, chances are you have a problem with your meetme config |
| 15:40.59 | ManxPower | if the Noop does not show the digits you dialed for the REad, then you have a DTMF issue. |
| 15:41.26 | gitguy | ok |
| 15:41.30 | gitguy | that would be good |
| 15:41.43 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 15:41.56 | FinboySlick | Hello gang. I hade a TDM800 with hardware echo canceller. Being a gentoo user, I was wondering if I need to enable mark or somesuch echo cancellers when I compile zaptel 1.2 |
| 15:42.24 | *** part/#asterisk mogie (n=opera@ppp121-44-209-93.lns1.hba1.internode.on.net) |
| 15:43.34 | gitguy | http://pastie.caboo.se/193681 |
| 15:43.36 | gitguy | there, only one codec |
| 15:43.52 | tzafrir_home | FinboySlick, enable OSLEC for best results . I think that the gentoo ebuild supports it |
| 15:44.01 | ManxPower | FinboySlick: the hardware EC will override and software EC in Asterisk |
| 15:44.08 | tzafrir_home | FinboySlick, that is, unless your card has hardware echo cancelling |
| 15:44.17 | ManxPower | you should not need OSLEC if you have hardware EC |
| 15:44.45 | b11d` | I know BSD is taboo here, but does anyone know if the TDM410E is supported in FreeBSD at all? |
| 15:45.07 | FinboySlick | tzafrir_home: It does have HWEC, but I still have some pretty nasty echo on certain calls. I'm trying to figure out if the HWEC is even enabled. |
| 15:46.02 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:46.12 | ManxPower | FinboySlick: I believe the dmesg info when the module loads will tell you. This has been talked about on the mailing list frequently |
| 15:46.33 | FinboySlick | ManxPower: I don't think I've noticed anything of the sort. |
| 15:46.49 | FinboySlick | ManxPower: In dmsg that is. |
| 15:47.13 | puzzled | hi |
| 15:47.20 | *** join/#asterisk nullb1t (n=alex@ool-44c098ae.dyn.optonline.net) |
| 15:47.25 | ManxPower | pasrebin the dmesg stuff from when the driver loads |
| 15:47.45 | ManxPower | FinboySlick: you, of course, read the zaptel readme, right? |
| 15:47.47 | FinboySlick | No wait, it says: Zaptel Echo Canceller: KB1 |
| 15:48.03 | nullb1t | Hey guys, could anyone help me out, using the instructs at: (freenode) |
| 15:48.06 | nullb1t | 11:40 -NickServ(NickServ@services.)- password so only you can use it. Please try to |
| 15:48.10 | nullb1t | 11:40 -NickServ(NickServ@services.)- use obscure passwords that cannot be easily guessed. |
| 15:48.13 | nullb1t | 11:40 -NickServ(NickServ@services.)- Passwords *ARE* case sensitive. Once you have |
| 15:48.16 | nullb1t | 11:40 -NickServ(NickServ@services.)- registered a nickname, you can use the SET and |
| 15:48.20 | nullb1t | 11:40 -NickServ(NickServ@services.)- ACCESS commands to configure it. |
| 15:48.22 | ManxPower | nullb1t: Don't flood the channe. |
| 15:48.22 | nullb1t | 11:40 -NickServ(NickServ@services.)- |
| 15:48.27 | ManxPower | What specific question did you have? |
| 15:48.39 | nullb1t | sorry, pasted the wrong thing :-\ |
| 15:49.03 | nullb1t | using the instructions at freepbx.org for centos 5.1 on a vserver.... I have ended up with no /var/lib/asterisk/bin |
| 15:49.11 | ManxPower | ~freepbx |
| 15:49.12 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:49.19 | nullb1t | k |
| 15:49.27 | puzzled | anyone know what this message means and what might be causing it: chan_sip.c: Hanging up call <long_id>@<ip> - no reply to our critical packet |
| 15:49.29 | nullb1t | what would you guys I advise instead of freepbx |
| 15:49.36 | puzzled | vi |
| 15:49.42 | ManxPower | What I wonder is why the idiots at freepbx.org send people here. |
| 15:49.50 | ManxPower | they KNOW we don't support it on this channel |
| 15:50.06 | puzzled | ManxPower: prolly to pull your chain :) |
| 15:50.09 | ManxPower | nullb1t: as almost nobody here uses any form of Asterisk GUI, this is not the best place to check. |
| 15:50.15 | nullb1t | :-p, I work 60+ hours a week and do not have the time to get deep w/ my asterisk install atm |
| 15:50.16 | puzzled | or was it "leg"? |
| 15:50.29 | ManxPower | nullb1t: then you need to hire a consultant. |
| 15:50.37 | nullb1t | hahaha, personal server |
| 15:50.44 | nullb1t | thanks anyways |
| 15:50.52 | ManxPower | nullb1t: then you will fail. Asterisk has a steep learning curve and requires much time to learn |
| 15:51.23 | nullb1t | ManxPower: yea, I ran gentoo asterisk installs for a good year 2 years ago |
| 15:51.25 | ManxPower | Even more so than traditional telecom since you need to know linux, networking, and voip protocols, as well as the hardware. |
| 15:51.31 | nullb1t | but I work too much and smoke to much weed to remember much |
| 15:52.02 | [TK]D-Fender | nullb1t: Go install freepbx and use their resources then. |
| 15:52.32 | ManxPower | nullb1t: I understand. I don't have the time for skydiving. I plan on just jumping out of a plane with a parachute and hope for the best. |
| 15:53.09 | nullb1t | ManxPower: ed-zackrey... why plan anything out... its personal use at home buddy... chill a little |
| 15:53.18 | nullb1t | [TK]D-Fender: thank you... thats what I am doing :) |
| 15:53.41 | ManxPower | nullb1t: you just told us you don't have time to learn. Rather pointless to be here if you expect us to do your work for you. |
| 15:54.57 | [TK]D-Fender | nullb1t: Good, now move along to #freepbx ... |
| 15:55.30 | nullb1t | prickly in here |
| 15:55.48 | nullb1t | sounds like you guys need some balance in life |
| 15:55.48 | b11d` | lol |
| 15:55.52 | nullb1t | pce |
| 15:55.55 | *** part/#asterisk nullb1t (n=alex@ool-44c098ae.dyn.optonline.net) |
| 15:56.02 | whymarkwh | how do you check what version you have via the command line? |
| 15:56.15 | FinboySlick | whymarkwh: Of asterisk? |
| 15:56.18 | whymarkwh | asterisk that is |
| 15:56.21 | whymarkwh | yes please |
| 15:56.29 | puzzled | asterisk -r and do "show version" |
| 15:56.34 | tzafrir_home | asterisk -V |
| 15:56.39 | whymarkwh | thx got it |
| 15:56.54 | tzafrir_home | asterisk -rx 'core show version' # >= 1.4 |
| 15:57.05 | tzafrir_home | asterisk -rx 'show version' # <= 1.4 |
| 15:57.40 | whymarkwh | if i want to download the mysql addon for asteriks and i am running 1.4.13, i suppose i can only use 1.4 addons? |
| 15:58.40 | puzzled | yes |
| 16:00.19 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 16:00.31 | freezey | on the Polycom IP330's what is the password when the phone starts up? |
| 16:01.14 | mort_gib | HI, I get Call from '' to extension '12' rejected because extension not found for some of my phones on * 1.4.18 |
| 16:01.23 | mort_gib | What's going on?? |
| 16:01.45 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 16:01.46 | mort_gib | -Sorry 1.4.19.1 |
| 16:01.54 | *** join/#asterisk jedaustin (n=chatzill@austin-j.its.dist.maricopa.edu) |
| 16:03.09 | ManxPower | mort_gib: the ACTUAL message might be helpful to us. |
| 16:03.20 | ManxPower | freezey: 456 is the password |
| 16:03.33 | acxty | I want to store the caller id, what it do on the system(I have a ivr), etc... into a mysql database. Everything is going fine. The problem that I am having is when the client hangup without finishing the menu. Is there a method, application, or something I can do when the user hangup? |
| 16:03.48 | mort_gib | <PROTECTED> |
| 16:04.06 | mort_gib | the '' part is the SIP client initiating the call |
| 16:04.39 | acxty | mort_gib, Is extension 12 in the context you are using? |
| 16:05.01 | whymarkwh | using the tutorial on http://www.voip-info.org/wiki-Asterisk+cdr+mysql do i need to download the whole addon package or just that relates to cdr? any help welcome |
| 16:07.48 | whymarkwh | do i have to reload asterisk if i add the addons? |
| 16:08.08 | mort_gib | acxty:Yeah... |
| 16:08.17 | acxty | pastebin |
| 16:08.23 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 16:08.46 | gitguy | ManxPower: it doesn't read correctly |
| 16:08.57 | gitguy | ManxPower: i used Read() and SayDigits() |
| 16:09.06 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.205.41) |
| 16:09.36 | *** join/#asterisk mosty (n=mosty@60-241-198-194.static.tpgi.com.au) |
| 16:10.27 | ManxPower | gitguy: then you have confirmed you have a DTMF problem. What is it reading? Nothing? duplicated digits? missing digits? |
| 16:10.35 | gitguy | <PROTECTED> |
| 16:10.42 | gitguy | "1235" only returns there |
| 16:10.45 | gitguy | and i typed 123456 |
| 16:11.10 | gitguy | ManxPower: missing digits |
| 16:11.18 | ManxPower | So "missing digits" is is. What device is diego.;viola? |
| 16:11.26 | gitguy | zoiper |
| 16:11.31 | gitguy | tried with xlite too |
| 16:11.32 | gitguy | same crap |
| 16:11.37 | gitguy | worse sometimes |
| 16:12.14 | ManxPower | You are using alaw, maybe those phones don't support rfc2833 DTMF when using alaw |
| 16:12.47 | gitguy | <PROTECTED> |
| 16:12.53 | gitguy | missing again |
| 16:13.07 | gitguy | what should i try? |
| 16:13.08 | gitguy | gsm? |
| 16:13.51 | ManxPower | no, try setting dtmfmode=inband in sip.conf for that peer and set it in your softphone as well |
| 16:14.25 | ManxPower | inband only works with ulaw and alaw, you cannot use it with any other codec or you will get missing digits |
| 16:14.43 | ManxPower | (or no digits) |
| 16:15.27 | *** join/#asterisk proprietarysucks (n=NRich@72.37.252.50) |
| 16:15.36 | gitguy | <PROTECTED> |
| 16:15.40 | gitguy | that readed fine |
| 16:16.37 | whymarkwh | do you have to a make menuselect with the addons? |
| 16:17.07 | gitguy | ManxPower: with gsm works fine it seems |
| 16:18.49 | whymarkwh | <PROTECTED> |
| 16:19.31 | whymarkwh | sorry wrong window |
| 16:20.42 | gitguy | <PROTECTED> |
| 16:20.43 | gitguy | with xlite i don't get anything |
| 16:20.43 | gitguy | lol |
| 16:21.15 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 16:22.31 | gitguy | where does xlite saves it's configs? |
| 16:22.36 | gitguy | ~/.somewhere |
| 16:22.37 | gitguy | i believe |
| 16:22.39 | apocn | Hello, how can I transfer a call? For example Im talking to someone and I want to transfer it to the corresponding person. How can I configure Asterisk to do that? |
| 16:22.57 | ManxPower | Dude, if you don't set the same DTMF mode on BOTH Asterisk AND the PHONE it'snot going to work very well. |
| 16:23.08 | ManxPower | apocn: press the transfer button on your phone. |
| 16:24.05 | apocn | I'll test it |
| 16:24.45 | ManxPower | I wish I understood why people don't try the transfer button |
| 16:25.10 | apocn | ManxPower: maybe cause I was using a softphone that doesnt have it? |
| 16:25.20 | apocn | I'll try with eyebeam now |
| 16:25.25 | ManxPower | "How do you transfer a call?" "Press the Transfer button." "Thanks! I never would have figured that out!" |
| 16:25.38 | apocn | ManxPower: if helping people bothers you, stop doing it! |
| 16:25.45 | apocn | noone is forcing you to do it |
| 16:26.03 | *** join/#asterisk SexyKen (i=foobar@c-24-5-173-239.hsd1.ca.comcast.net) |
| 16:26.05 | SexyKen | Hey all |
| 16:26.08 | ManxPower | apocn: oh, I enjoy helping people, I don't enjoy helping idiots. |
| 16:26.37 | ManxPower | About the only things out there that don't have a transfer button are a few softphones and analog phones. |
| 16:26.48 | ManxPower | and on analog phones you can transfer using the Flash or Recall key. |
| 16:27.01 | *** join/#asterisk TheSov (n=a@204-16-231-218-risingmedical.pkh.ord.sparkplugbb.net) |
| 16:27.11 | apocn | okey, thanks a lot ManxPower! |
| 16:27.20 | SexyKen | SIP/2.0 407 Proxy Authentication Required <-- I keep getting that message from one of my trunks. But I have each trunk setup the same - their just different providers. I have insecure=very and a permit= line for the proper ip/cidr can anyone assist? |
| 16:27.43 | SexyKen | Provider is carriers.icall.com btw |
| 16:27.50 | ManxPower | apocn: since you didn't mention what device you are using, I had to assume it was one of the majority of devices that had a transfer button |
| 16:27.56 | apocn | SexyKen: have you actually registered your asterisk? |
| 16:28.05 | mosty | SexyKen, perhaps you should contact their tech support? |
| 16:28.14 | ManxPower | SexyKen: you will always get that. SIP calls start out unauthenticated, then restart with auth info sent |
| 16:28.18 | SexyKen | Their tech support doesn't assist with client setups. |
| 16:28.35 | SexyKen | Manx: But I dont get it on my teliax trunk - and their setup the same way. |
| 16:28.50 | apocn | SexyKen: sip show registry |
| 16:28.54 | SexyKen | apocn: sip show registry shows that both turnks are reg'd. |
| 16:29.03 | apocn | ok |
| 16:29.05 | ManxPower | SexyKen: have you done a sip debug on your teliax account? |
| 16:29.06 | whymarkwh | how do you check what distro linux you are using? |
| 16:29.23 | SexyKen | Manx: Yea - the incoming invite looks the same....except my response differs between the two. |
| 16:29.33 | ManxPower | SexyKen: sip show registry just shows that asterisk notified the remote side of your IP. IT has nothing to do with actual calls |
| 16:29.44 | SexyKen | I respond with acceptance for teliax incoming dids, but with the authentication message for icall |
| 16:29.47 | *** join/#asterisk [hC] (n=hardcore@190.10.10.76) |
| 16:29.57 | ManxPower | SexyKen: Are you getting anything on the CLI if you don't have debug enabled? |
| 16:30.09 | SexyKen | No, if debug is disabled, nothing comes in. |
| 16:30.19 | ManxPower | then it is an auth problem |
| 16:30.21 | ManxPower | odd. |
| 16:30.49 | ManxPower | SexyKen: why are you trying to permit based on IP. use passwords |
| 16:31.07 | SexyKen | Manx: How do I know what password to set and how do I set it |
| 16:31.51 | ManxPower | You would have to talk to Teliax for their side, for your side you set it in sip.conf |
| 16:32.19 | SexyKen | Teliax is the one that works, iCall is the one that doesn't. |
| 16:32.25 | freezey | ok i just got 2 ip330 phones and i am used to xlite phone on the computer not actual phones and i am just lost on setting this thing up to communicate with my sip settings |
| 16:32.30 | SexyKen | I believe that icall is not using any password authentication |
| 16:32.32 | SexyKen | So shouldn't it work? |
| 16:32.32 | ManxPower | Then ask iCall for your password |
| 16:32.44 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:32.47 | ManxPower | SexyKen: put a pastebin of your sip.conf masking only passwords |
| 16:33.32 | SexyKen | Manx: How can I tell if iCall is sending the call with a password? |
| 16:33.47 | [TK]D-Fender | freezey: Take your firmware pack I told you to get from your vendor, and go follow the polycom provisioning guide on the WIKI. |
| 16:34.01 | freezey | u have url to wiki |
| 16:34.08 | ManxPower | SexyKen: you would have to talk to someone willing to spend the time to read your SIP debug. That person is not me. |
| 16:34.18 | freezey | [TK]D-Fender: i am just doing a local test of this with a asterisk box i am playin with |
| 16:34.34 | [TK]D-Fender | freezey: there is no "just". |
| 16:34.41 | freezey | ok |
| 16:34.47 | [TK]D-Fender | freezey: those are the steps to follow. Do it. |
| 16:34.51 | *** join/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 16:34.51 | freezey | [TK]D-Fender: do you have the url for wiki? |
| 16:35.11 | freezey | [TK]D-Fender: i am not sure what steps your talking about tho. |
| 16:35.15 | ManxPower | www.voip-info.org I cannot believe you don't know the wiki, you've been here for a long time. |
| 16:35.30 | freezey | usually getting yelled at by tk |
| 16:35.31 | freezey | ha |
| 16:35.34 | ManxPower | freezey: the steps in the polycom provisioning guide. |
| 16:35.59 | [TK]D-Fender | ~wikis |
| 16:35.59 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 16:36.48 | freezey | ManxPower: lookin for that now on the polycom site |
| 16:37.17 | ManxPower | freezey: How long ago did [TK]D-Fender tell you to get the provisioning guide? |
| 16:37.31 | freezey | ManxPower: i just got the phones today |
| 16:37.39 | ManxPower | That was not my question. |
| 16:38.07 | freezey | ManxPower: i am not even sure what hes talkin about |
| 16:38.28 | ManxPower | freezey: You would be actually looking for the polycom admin guide. |
| 16:38.30 | [TK]D-Fender | ManxPower: but he "just" wants to set it up and he's only used X-lite! Actual facts and previously made statements are naturally null & void! |
| 16:38.50 | SexyKen | How do you setup authentication for an incoming call? |
| 16:38.52 | SexyKen | secret=? |
| 16:39.04 | [TK]D-Fender | SexyKen: thats part of it. |
| 16:39.16 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 16:39.28 | SexyKen | [TK]D-Fender - the other part is username=? |
| 16:39.35 | ManxPower | http://polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip501.html |
| 16:39.43 | [TK]D-Fender | SexyKen: Another part. |
| 16:39.46 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:39.52 | SexyKen | [TK]D-Fender: realm=? |
| 16:39.58 | jameswf-home | only in asterisk is 99.9989854542% a bad thing... greedy bastards |
| 16:40.00 | ManxPower | SexyKen: all that stuff is for INCOMING |
| 16:40.07 | [TK]D-Fender | SexyKen: Are you going to spit it out one bloody parm at a time? |
| 16:40.19 | ManxPower | sorry, I meant OUTGOING |
| 16:40.26 | ManxPower | the incoming username is what is in [here] |
| 16:40.29 | SexyKen | Manx: I am able to make outgoing calls, but am unable to receive incomgin. |
| 16:40.57 | SexyKen | Manx: "the incoming username is what is in [here]" what is in there? |
| 16:41.01 | [TK]D-Fender | SexyKen: And you don't seem to be showing us the FAILED ACLL. |
| 16:41.04 | [TK]D-Fender | CALL* |
| 16:41.07 | ManxPower | SexyKen: What is your icall username? |
| 16:41.26 | SexyKen | ab_succ |
| 16:41.46 | ManxPower | then in sip.conf you need a section [ab_succ] with all the required info |
| 16:42.01 | ManxPower | and that is the incoming username, assuming it really is your incoming username |
| 16:42.08 | rob0 | jameswf-home: air traffic control |
| 16:42.20 | ManxPower | SexyKen: perhaps now is a good time to read The Good Book |
| 16:42.23 | ManxPower | ~book |
| 16:42.23 | 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 |
| 16:42.29 | jameswf-home | Im to sexy for this pbx, to sexy for this pbx so sexy it huuuurrrts... I'm an admin you know what I mean and i do my little tweaks in a console yeahhh in a console... |
| 16:43.00 | ManxPower | .part #americas-unfunniest-home-videos |
| 16:43.01 | jameswf-home | rob0: I dont think even air traffic control has those numbers they miss a handfull each year |
| 16:43.19 | SexyKen | ManxPower: In that case - that's how it's already setup...that's where I have insecure set up etc |
| 16:43.34 | rob0 | OMG, we had to miss more than just a handful, we had to miss them all! |
| 16:43.46 | ManxPower | SexyKen: I guess now is the time to provide [TK]D-Fender the info he asked for. |
| 16:44.15 | ManxPower | Assuming he has not already written you off for ignoring his request. |
| 16:44.16 | jameswf-home | it is verry dificult to be a google proxy when you lack search terms |
| 16:44.29 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 16:44.54 | rob0 | well crap, damn those Sipphone fellows ... area code 747 is taken! |
| 16:45.50 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 16:46.40 | *** join/#asterisk yxa (n=lonari@bb116-14-12-104.singnet.com.sg) |
| 16:46.59 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 16:47.03 | *** join/#asterisk killmel8tr (n=IceChat7@ip-198-22-65-131.quickconnectusa.com) |
| 16:48.05 | *** join/#asterisk dFence (n=chatzill@ings-d93220ce.pool.mediaWays.net) |
| 16:48.15 | dFence | me back. |
| 16:49.59 | LinuxMafia | hi again |
| 16:50.41 | intralanman | anyone here interested in speaking about asterisk at cluecon in august? |
| 16:51.02 | LinuxMafia | can asterix send sms messages automatically if the number is mobile? |
| 16:51.47 | coppice | intralanman: don't you realise that nobody in here knows anything about it |
| 16:51.58 | dFence | any news on asterisk VS mac? last time i tried (tiger w. *1.4...) didn't work out that well |
| 16:52.02 | intralanman | coppice: i'm starting to see that |
| 16:52.25 | intralanman | how about if i post a link? http://www.cluecon.com |
| 16:52.39 | rob0 | I will I will |
| 16:52.56 | LinuxMafia | i want to set astreiks up for dental office , asteriks gonna work for this purpose? |
| 16:53.06 | mog | sure LinuxMafia |
| 16:53.09 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 16:53.20 | dFence | LinuxMafia: dunno how good * can handle drills n stuff |
| 16:53.25 | rob0 | Sure, configuring asterisk is a lot like pulling teeth |
| 16:53.28 | intralanman | rob0: seriously? |
| 16:53.34 | dFence | lol |
| 16:53.35 | [TK]D-Fender | LinuxMafia: No, Asterisk causes cavities. |
| 16:53.41 | LinuxMafia | lol |
| 16:53.56 | LinuxMafia | no i mean the secretor is not a computer person |
| 16:54.03 | [TK]D-Fender | LinuxMafia: * is not an SSM gateway. |
| 16:54.10 | [TK]D-Fender | SMS* |
| 16:54.22 | LinuxMafia | so it can not send sms's |
| 16:54.23 | coppice | do you need stuff like background noise supression, so callers can't hear the patients screaming? |
| 16:54.34 | dFence | lol |
| 16:54.43 | dFence | has anyone read that o'reilly book bout *` |
| 16:54.44 | dFence | ? |
| 16:54.46 | rob0 | intralanman, I'm not shy, but I'm not competent, either. :) |
| 16:54.47 | Qwell | ~book |
| 16:54.48 | 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 |
| 16:54.55 | dFence | exactly that one. |
| 16:55.14 | LinuxMafia | i want to read it , but i want to make sure it is gonna work before i read ity |
| 16:55.20 | Qwell | dFence: if people in here *haven't* read it, they really really should |
| 16:55.33 | intralanman | rob0: great, that's the best offer we've had so far ;-) nobody seems interested in speaking about asterisk... i was thinking i might have to do it if nobody else volunteers :-D |
| 16:55.54 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:56.06 | LinuxMafia | Qwell, i dont like the prefaces stuff , which chapter i should beging from? |
| 16:56.13 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 16:56.16 | dFence | LinuxMafia: my first encounter with * didn't end to pleasantly... wanna try a more serious approch this time |
| 16:56.23 | Qwell | intralanman: pay for my flight, hotel, accommodations, and I'll consider going |
| 16:56.52 | dFence | 'll second that |
| 16:56.53 | intralanman | Qwell: join #freeswitch and talk it over with bkw... i'm just looking for volunteers |
| 16:57.18 | LinuxMafia | dFence, i need progamming for *? |
| 16:57.34 | dFence | LinuxMafia: ?_? |
| 16:57.50 | LinuxMafia | cause i saw some people are talking about some C stuff in here |
| 16:57.50 | *** join/#asterisk s519 (n=steve@82-71-61-44.dsl.in-addr.zen.co.uk) |
| 16:58.07 | LinuxMafia | like warning messages |
| 16:58.18 | intralanman | Qwell: so far the talks seem to be very freeswitch centric... it's supposed to be all open-source telephony, so we figured someone in the asterisk project might want to speak |
| 16:58.31 | dFence | LinuxMafia: consider me a complete newbie regarding * and my C-repartoire's more than rusty so i'm prolly the last one you wanna talk about programming for * :D |
| 16:58.37 | Qwell | obviously it's freeswitch-centric |
| 16:59.19 | LinuxMafia | dFence, all i am asking , can a regular person operate * easily or you need IT knowledg |
| 16:59.39 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 16:59.39 | LinuxMafia | dFence, cause most people in office dont know computers |
| 16:59.44 | *** join/#asterisk LARefugee (n=chatzill@c-76-104-191-194.hsd1.wa.comcast.net) |
| 17:00.00 | dFence | LinuxMafia: u should at least speak fluent config-files... i'd say |
| 17:00.17 | mintee | anyone installed iaxmodem and hylafax on 1.4.19? |
| 17:00.55 | LinuxMafia | dFence, but you have to configure asterix once right? |
| 17:01.03 | coppice | mintee: iaxmodem + hylafax have nothing to so with *, so why would the version matter |
| 17:01.13 | *** join/#asterisk s0lid (n=s0lid@210.213.198.7) |
| 17:01.21 | dFence | LinuxMafia: i'd say rtfm ^^ |
| 17:01.29 | mintee | coppice, because asterisk's iax has been broken since 1.4 |
| 17:01.32 | intralanman | Qwell: there's a $50 discount for anyone who's contributed to an asterisk project.... not quite totally free, but it's a start :-) |
| 17:01.45 | mintee | it is/was an * problem to begin with... |
| 17:01.47 | Qwell | wait, speakers don't get in free? |
| 17:01.51 | LinuxMafia | dFence, lol i am rtfming first chapter , but just curios |
| 17:02.10 | Qwell | rolls his eyes and goes back to code |
| 17:02.11 | dFence | LinuxMafia: than you're one chapter ahead of me |
| 17:02.15 | intralanman | maybe? |
| 17:02.47 | LinuxMafia | lol |
| 17:02.59 | LARefugee | any TDM400P and X100M vets out there? IM me please! |
| 17:03.01 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 17:03.11 | dFence | hey.. anyone using * with a regular fritzcard? |
| 17:05.07 | dFence | *yuk* food. on tv. |
| 17:05.59 | *** join/#asterisk jjshoe_ (n=jjshoe@72.37.252.50) |
| 17:06.05 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:06.26 | freezey | [TK]D-Fender: if i explain what i have done so far can you just tell me if i am right or wrong... just so atleast i am on the right track here |
| 17:06.39 | [TK]D-Fender | freezey: Maybe. |
| 17:07.23 | [TK]D-Fender | LARefugee: ... |
| 17:07.25 | [TK]D-Fender | ~ask |
| 17:07.25 | jbot | well, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:07.27 | [TK]D-Fender | ^^^^^^^^ |
| 17:08.06 | mintee | highfives jbot... pwnt |
| 17:08.16 | mintee | hyper today |
| 17:08.27 | gitguy | amazing |
| 17:08.30 | gitguy | i don't know how this was working before |
| 17:08.33 | gitguy | i type "123456" |
| 17:08.38 | gitguy | and it only gets "13" |
| 17:08.39 | gitguy | lol |
| 17:08.45 | gitguy | my DTMF is fucked up |
| 17:08.45 | Kobaz | any idea why asterisk would suck up 99% cpu when handling an iax call (this is on a fairly old box... 300mhz) |
| 17:09.02 | [TK]D-Fender | Kobaz: Transcoding. |
| 17:09.15 | mintee | gitguy, i've been noticing that problem a too from incoming PRI calls |
| 17:09.19 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:09.26 | Kobaz | well, i'm making a call from an iax peer and handing it off to an iax trunk |
| 17:09.39 | Kobaz | i dont think it should be doing any transcoding |
| 17:09.47 | gitguy | mintee: how did you fixed it |
| 17:09.52 | Kobaz | i'm using the same codecs all over |
| 17:09.57 | LARefugee | ok.. an x100m installed in the "tel1" slot of a tdm400p produces a unlistenable rumbling, crackling sound Ubuntu Hardy * 1.4.17 |
| 17:09.58 | mintee | gitguy, i haven't |
| 17:11.13 | gitguy | i get the same problem with gsm and alaw/ulaw |
| 17:11.23 | gitguy | with gsm i get fewer loss |
| 17:11.23 | LARefugee | In port 2 it's just just fine. Tested with two x100m modules and two tdm400p's! |
| 17:11.24 | gitguy | though |
| 17:11.26 | gitguy | but still loss |
| 17:11.27 | gitguy | :( |
| 17:11.33 | gitguy | how can i fix this? |
| 17:11.33 | freezey | [TK]D-Fender: well here goes nothing... Plugged in the phones each got an address's from dhcp.. after that i went into the settings>advanced>admin settings>sipconfiguration Went into the server portion and entered the address to my asterisk machine got out of there went down to line 1 entered display name changed type to shared entered auth user id: and auth password and then saved and rebooted.. |
| 17:11.36 | Qwell | LARefugee: contact Digium support |
| 17:12.06 | [TK]D-Fender | freezey: Follow the provisioning guide. |
| 17:12.17 | gitguy | <PROTECTED> |
| 17:12.26 | gitguy | it should be : 123456 |
| 17:12.29 | freezey | [TK]D-Fender: do you mean the admin guide? |
| 17:12.35 | gitguy | [TK]D-Fender: how can i fix this? |
| 17:12.45 | [TK]D-Fender | freezey: Take your firmware pack I told you to get from your vendor, and go follow the polycom provisioning guide on the WIKI. |
| 17:12.47 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 17:12.48 | gitguy | dtmfmode? |
| 17:12.53 | LARefugee | I have. They're taking their time. Just wondered if anyone has experienced this. |
| 17:12.59 | [TK]D-Fender | freezey: I'm about done repeating myself... |
| 17:13.33 | freezey | [TK]D-Fender: didnt get a firmware pack not even sure where i would get one |
| 17:13.41 | gitguy | ManxPower: can you help me? |
| 17:13.55 | [TK]D-Fender | freezey: "get from your vendor" |
| 17:14.08 | [TK]D-Fender | freezey: I'm telling all of this to your face, you'd better wake up. |
| 17:14.40 | gitguy | with allow=alaw / dtmfmode=inband i get worse results |
| 17:14.40 | gitguy | :S |
| 17:14.53 | freezey | [TK]D-Fender: ok i am just gonna play around with this till i figure it out cause i am just unsure of a few things with this so i will just read wiki |
| 17:16.32 | gitguy | this sucks |
| 17:16.56 | jjshoe_ | if anyone's interested I have a tux screen for sale for $100 http://tuxscreen.net/ |
| 17:17.57 | Qwell | jjshoe_: you still have that thing? |
| 17:17.59 | whymarkwh | what is app_sycountpl listed in the asterisk-addon1.4 ? do i need it to link to link * to mysqld? |
| 17:18.06 | jjshoe_ | Qwell hahaha yeah |
| 17:18.11 | Qwell | I'll give you $20 for it :p |
| 17:18.19 | gitguy | <PROTECTED> |
| 17:18.19 | gitguy | <PROTECTED> |
| 17:18.25 | jjshoe_ | Qwell that wouldn't even cover shipping :P |
| 17:18.28 | gitguy | i typed "123456789" |
| 17:18.33 | Qwell | I'll pay shipping too |
| 17:18.41 | gitguy | can someone help me please? |
| 17:18.51 | jjshoe_ | Qwell $20, shipping, and a digium t-shirt? :D |
| 17:19.09 | gitguy | :( |
| 17:19.17 | Qwell | $20, shipping, and a used Asterisk shirt |
| 17:19.27 | jjshoe_ | Qwell does it smell like you? |
| 17:19.42 | whymarkwh | tlol |
| 17:19.46 | intralanman | would it be better or worse if it did? |
| 17:19.52 | Qwell | umm, yes, but I'm only gonna go $10 then |
| 17:20.01 | jjshoe_ | lol |
| 17:20.07 | jjshoe_ | let me think about it :D |
| 17:20.55 | whymarkwh | where does paste again please? |
| 17:21.00 | Qwell | ~paste |
| 17:21.00 | jbot | paste is probably http://rafb.net/paste/, or see also pb |
| 17:21.56 | whymarkwh | thx |
| 17:23.12 | whymarkwh | i did the command make in the asterisk addon1.4.6 and get the following error: http://rafb.net/p/WsItGN30.html anyone knows what it means please? |
| 17:23.18 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 17:24.01 | whymarkwh | something related to mp3 dont know what |
| 17:24.10 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net) |
| 17:28.04 | *** part/#asterisk bkw_ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net) |
| 17:28.07 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net) |
| 17:29.23 | gitguy | can someone help? |
| 17:30.00 | rob0 | bkw_: do y'all no longer have #asterlink ? |
| 17:31.09 | bkw_ | rob0: yah but I forget to join sometimes |
| 17:31.20 | rob0 | haha |
| 17:32.18 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.205) |
| 17:32.35 | *** part/#asterisk inja_y (n=inja_y@ip70-174-148-215.dc.dc.cox.net) |
| 17:32.43 | *** join/#asterisk [hC] (n=hardcore@190.10.10.76) |
| 17:33.20 | [hC] | So what is it that im missing when doing sip interconnects between an asterisk 1.2 and 1.4 box, where after an invite, the digest username gets set to "s" and calling breaks because of authentication errors? |
| 17:34.45 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-116-41.PHNX.QWEST.NET) |
| 17:37.28 | bkw_ | [hC]: don't you know you're to use IAX for that |
| 17:37.43 | [hC] | bkw_: :) |
| 17:37.55 | *** join/#asterisk nn (n=kaeru@unaffiliated/nn) |
| 17:38.29 | intralanman | [hC]: i'd assume that you have your /extension on the register line? |
| 17:38.35 | intralanman | is that still needed? |
| 17:38.49 | [hC] | intralanman: er.. no, i dont register as an extension |
| 17:39.01 | bkw_ | if its changing the auth username mid stream thats broken in my opinion |
| 17:39.11 | bkw_ | [hC]: thats your register contact |
| 17:39.27 | bkw_ | unless you want the remote box to send all calls to s |
| 17:39.42 | [hC] | maybe my register line is wrong, then |
| 17:39.43 | [hC] | heh! |
| 17:39.46 | [hC] | im doing |
| 17:39.49 | [hC] | register => user:pass@host |
| 17:39.53 | bkw_ | add /user |
| 17:39.54 | intralanman | add a /user |
| 17:39.58 | [hC] | lame. LAME |
| 17:39.59 | intralanman | yeah, what he said ;-)( |
| 17:39.59 | bkw_ | jinx |
| 17:40.10 | [hC] | i hate how asterisk has different syntax for IAX and SIP registration/dialing |
| 17:40.16 | [hC] | que annoying. |
| 17:40.17 | intralanman | [hC]: no, lame is for mp3 |
| 17:40.20 | intralanman | :-D |
| 17:40.34 | [hC] | thank you for the tip though guys, i rarely use sip to connect * boxes |
| 17:40.51 | *** join/#asterisk vader-- (n=me@c-71-226-192-99.hsd1.nj.comcast.net) |
| 17:41.38 | gitguy | could the DTMF problems (digit loss) be due to bandwidth problems? |
| 17:41.40 | gitguy | ????????? |
| 17:41.55 | bkw_ | gitguy: doubtful.. |
| 17:42.06 | bkw_ | gitguy: aren't we helping you with your Asterisk issue in #freeswitch? |
| 17:42.20 | whymarkwh | hi using asterisk 1.4.13 and did the make command of the *addons-1.4.6 and get the following error http://rafb.net/p/WsItGN30.html anyone knows what it means please? |
| 17:42.32 | gitguy | <PROTECTED> |
| 17:42.40 | gitguy | should return => 123456789 |
| 17:42.43 | bkw_ | whymarkwh: format_mp3.c is out of date |
| 17:43.05 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 17:43.09 | gitguy | bkw_: yeah but i wanted to see if someone can help me here too |
| 17:43.18 | bkw_ | gitguy: doubtful |
| 17:43.23 | gitguy | bkw_: thanks tho |
| 17:43.27 | bkw_ | gitguy: get a packet trace with tcpdump |
| 17:43.29 | bkw_ | and watch it |
| 17:43.32 | whymarkwh | btw_: meaning not used or i must update? |
| 17:43.42 | bkw_ | whymarkwh: no clue open a bug with digium on it |
| 17:44.04 | bkw_ | or ask on their mailing lists |
| 17:44.24 | bkw_ | gitguy: also are you dialling slow or fast? |
| 17:44.28 | *** join/#asterisk Travo1 (n=dj@adsl-65-70-249-168.dsl.bumttx.swbell.net) |
| 17:44.32 | whymarkwh | bkw_: do you think the mysql addon use it? dont think so myself? |
| 17:44.53 | bkw_ | whymarkwh: are you going to use the addons or are you just adding them because you can? |
| 17:44.56 | Travo1 | Could somebody assist me in getting my x100p card to work properly w/ asterisk? Followed directions verbatum. |
| 17:45.06 | bkw_ | Travo1: what is the error? |
| 17:45.22 | gitguy | bkw_: i tried fast and slow |
| 17:45.37 | bkw_ | gitguy: get the tcpdump of you dialing and put it up so I can wget it |
| 17:45.41 | Travo1 | i run ztcfg -vv and I get line 0: unable to open master device '/dev/zap/ctl' 1 error(s) detected |
| 17:45.42 | whymarkwh | want to link cdr with mysql and create my own web interfase to monitor records |
| 17:45.51 | bkw_ | Travo1: modprobe wcfxo |
| 17:45.59 | intralanman | Travo1: did yo modpr.... damn you bkw_ |
| 17:46.08 | bkw_ | intralanman: great minds think alike |
| 17:46.09 | intralanman | i'm done |
| 17:46.14 | Travo1 | bkw_: returns nothing, did a dmesg | tail and nothing returned either |
| 17:46.15 | intralanman | i quit... that's it |
| 17:46.22 | gitguy | bkw_: i just run tcpdump as root on my softphone computer right? |
| 17:46.34 | bkw_ | gitguy: yes and make sure you capture the output to a pcap file |
| 17:46.44 | bkw_ | Travo1: now ztcfg |
| 17:46.47 | gitguy | bkw_: pcap? |
| 17:46.49 | intralanman | gitguy: you could use pcapsipdump too... i like it |
| 17:47.22 | bkw_ | intralanman: I need rtp |
| 17:47.25 | intralanman | gitguy: packet capture... pcap... standard wireshark, tcpdump, etc |
| 17:47.30 | bkw_ | tcpdump -w output.pcap |
| 17:47.30 | intralanman | bkw_: pcapdipdump gets that |
| 17:47.34 | bkw_ | intralanman: good |
| 17:48.14 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 17:48.33 | Travo1 | bkw_: same error |
| 17:48.41 | bkw_ | Travo1: what does dmesg say? |
| 17:48.56 | whymarkwh | make install |
| 17:48.56 | bkw_ | and lsmod | grep wcfxo |
| 17:49.05 | jpeeler | Travo1: what user are you executing ztcfg as and what does ls -l /dev/zap/ctl report |
| 17:49.05 | whymarkwh | tlol wrong windows sorry |
| 17:49.08 | Travo1 | bkw_: nothing related to asterisk, last entry was for ethernet, eth2: no IPv6 routers presetn |
| 17:49.30 | bkw_ | Travo1: and lsmod doesn't show wcfxo loaded? |
| 17:49.38 | bkw_ | sounds like zaptel isn't fully installed or something |
| 17:50.05 | whymarkwh | does one need to reinstall asterisk if you installed the addons? |
| 17:50.08 | gitguy | bkw_: http://pastebin.com/m4e2af9c0 |
| 17:50.22 | bkw_ | gitguy: no tcpdump -w outputfile.pcap |
| 17:50.28 | intralanman | whymarkwh: you _shouldn't_ have to |
| 17:50.30 | bkw_ | I need the outputfile.pcap so I can open it in wireshark |
| 17:50.33 | Travo1 | bkw_: zaptel is installed, i was running ztdummy for the longest time then installed this card yesterday. lsmod shows wcfxo IS loaded |
| 17:51.02 | Qwell | whymarkwh: as I said earlier - you need to upgrade Asterisk |
| 17:51.08 | whymarkwh | thx intralanman |
| 17:51.49 | gitguy | bkw_: done, what i do with that file? |
| 17:51.57 | bkw_ | gitguy: put it up somewhere I can wget it |
| 17:51.58 | gitguy | bkw_: looks like binary |
| 17:52.03 | intralanman | Travo1: and you appropriately edited your zaptel.conf? |
| 17:52.08 | bkw_ | gitguy: yah its the raw packets.. I can poke at them and see what is up |
| 17:52.41 | Travo1 | bkw_: I just went to the bottom of the zaptel.conf file and added the lines that x100p said |
| 17:53.55 | bkw_ | gitguy: you have a couple of options here for this DTMF issue. |
| 17:54.03 | bkw_ | gitguy: 1. Turn inband dtmf on... (I don't like this option) |
| 17:54.23 | bkw_ | gitguy: 2. Put FreeSWITCH in front of Asterisk to fix the 2833 issue. |
| 17:54.35 | bkw_ | oh and the third option is |
| 17:54.45 | bkw_ | Open a bug on bugs.digium.com and hope they fix it |
| 17:55.00 | intralanman | Travo1: can you unload and reload the wcfxo module? |
| 17:55.18 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 17:55.25 | gitguy | bkw_: they will probably ignore me, can you report the bug for me :) |
| 17:55.27 | Travo1 | intra: what is the linux command(s) for that |
| 17:55.33 | bkw_ | gitguy: they'll ignore me too |
| 17:55.38 | gitguy | :( |
| 17:55.44 | intralanman | Travo1: modprobe -r wcfxo |
| 17:55.59 | intralanman | gitguy: and i know they'll ignore me |
| 17:56.17 | bkw_ | gitguy: I'm doing my best to help you.. sorry you had to come to #freeswitch to get any sort of help. |
| 17:56.18 | intralanman | i don't tend to post enough info |
| 17:56.23 | Travo1 | intra: did it, had to sudo it though |
| 17:56.32 | gitguy | so is this asterisk problem or just bandwidth issues? |
| 17:56.34 | intralanman | Travo1: anything in dmesg now? |
| 17:56.43 | bkw_ | gitguy: asterisk problem |
| 17:56.47 | gitguy | wtf |
| 17:56.51 | bkw_ | gitguy: get me that pcap and I can tell you why its doing it |
| 17:56.51 | *** part/#asterisk yxa (n=lonari@bb116-14-12-104.singnet.com.sg) |
| 17:57.04 | gitguy | bkw_: ok 1min |
| 17:57.21 | *** join/#asterisk jefferai (n=quassel@amarok/developer/mitchell) |
| 17:57.24 | intralanman | gitguy: are you having horrible bw issues? i think rfc2833 spec is to send each packet thrice, isn't it? |
| 17:57.38 | Travo1 | intra: nothing in dmesg |
| 17:57.49 | bkw_ | Travo1: what kind of server is this? |
| 17:57.57 | bkw_ | distro and such |
| 17:58.04 | gitguy | bkw_: http://www.maitei.com.py/outputfile.pcap |
| 17:58.06 | bkw_ | intralanman: its more than that but yes |
| 18:00.15 | [hC] | ive had dtmf problems using rfc2833 on 1.4 with g729 |
| 18:00.16 | bkw_ | gitguy: are you using SRTP by chance? |
| 18:00.17 | [hC] | missing digits |
| 18:00.19 | [hC] | double digits |
| 18:00.27 | *** join/#asterisk railsmunky (n=nick@82-70-165-142.dsl.in-addr.zen.co.uk) |
| 18:00.28 | [hC] | i had to switch to ulaw/inband to fix it |
| 18:00.30 | bkw_ | [hC]: codec doesn't have anything to do with 2833 or shouldn't |
| 18:00.35 | Travo1 | bkw: asterisk & ubuntu distro |
| 18:00.36 | bkw_ | its a totally different thing |
| 18:00.48 | [hC] | bkw_: i agree. now that you remind me, i think it was a problem with ulaw + 2833 as well |
| 18:01.00 | jefferai | Is it possible when you are hosting conferences on an Asterisk server (e.g. with MeetMe) to have Asterisk dial out to join participants, instead of only having people dial in? |
| 18:01.05 | [hC] | bkw_: i know rfc2833 has been rewritten in 1.4 for variable length dtmf, and that seems to have introduced a whole slew of new issues |
| 18:01.17 | [hC] | bkw_: presumably between 1.2 -> 1.4, or 1.4 -> some commercial switches |
| 18:01.33 | intralanman | jefferai: with which conferencing app? any? or meetme specifically? |
| 18:01.46 | jefferai | any, I just know of meetme |
| 18:01.53 | jefferai | if there are others that are more robust... |
| 18:01.57 | intralanman | oh, there are several others... bkw_ do you have that link? |
| 18:02.14 | acxty | hey guys, how can I find the duration of a call? |
| 18:02.29 | [hC] | acxty: in CDR |
| 18:02.34 | bkw_ | intralanman: mod_conference in fs |
| 18:02.44 | bkw_ | gitguy: the DTMF is on the wire |
| 18:02.47 | bkw_ | gitguy: asterisk is broken |
| 18:02.56 | intralanman | bkw_: heheh, i think jefferai wants to use asterisk though |
| 18:03.10 | jefferai | not necessarily..what's FS? |
| 18:03.17 | intralanman | #freeswitch |
| 18:03.19 | jefferai | I want to use an open-source/free PBX |
| 18:03.25 | bkw_ | well if asterisk fits his needs then by all means use asterisk |
| 18:03.29 | intralanman | it's open source, but not GPL |
| 18:03.32 | jefferai | ah |
| 18:03.46 | gitguy | bkw_: what is SRTP? |
| 18:03.47 | jefferai | basically, I need that capability, to have the PBX pull others into conferences |
| 18:03.53 | jefferai | whichever PBX does it |
| 18:03.55 | bkw_ | gitguy: nm I had to make wireshark do the right thing. |
| 18:03.57 | acxty | [hC], I am working mostly with Agi(). Is there a application I can call when the channel hangup? |
| 18:04.08 | gitguy | bkw_: how do you know asterisk is broken? |
| 18:04.22 | intralanman | jefferai: the default fs install does that... but i think there's a mod_confcall app for asterisk too |
| 18:04.30 | bkw_ | intralanman: app_conference |
| 18:04.35 | intralanman | oh |
| 18:04.37 | bkw_ | but its not mantained is it? |
| 18:04.39 | jefferai | bkw_: does freeswitch work with the same timing cards that asterisk uses for conferences? |
| 18:04.50 | intralanman | jefferai: fs doesn't need timing cards |
| 18:05.28 | gitguy | bkw_: so, it's an asterisk bug? |
| 18:05.36 | bkw_ | gitguy: I think you should do this pcap from the Asterisk side to make sure. but I see the digits on the wire. |
| 18:05.56 | bkw_ | but I see this client doesn't send the end big three times like it should.. so that in its self could be a bug too |
| 18:06.00 | bkw_ | s/big/bit/ |
| 18:06.09 | acxty | [hC], for example when user hangup or the call ended run an agi script that do its stuff |
| 18:06.17 | [hC] | acxty: that, i am not sure |
| 18:07.00 | intralanman | acxty: you could use deadagi if nothing else works for you... that'll let you process after the channel is hung up |
| 18:07.15 | acxty | the problem is that I need to send that information to a different computer to a mysql database |
| 18:07.34 | *** join/#asterisk ManxPower (n=manxpowe@97-81-241-251.dhcp.lds.al.charter.com) |
| 18:07.59 | acxty | intralanman, but will deadagi() will run no matter when the hangup or call end occur? |
| 18:08.34 | intralanman | acxty: yes, it should... i used to use it for my calling card platform. to update a db after the call ended |
| 18:08.43 | acxty | intralanman, it won't matter in which part of the dialplan the channel was on? |
| 18:08.58 | intralanman | acxty: wait wait wait |
| 18:09.23 | acxty | intralanman, that is my problem |
| 18:09.58 | gitguy | bkw_: http://www.maitei.com.py/outputfile2.pcap (from the server) |
| 18:10.03 | acxty | I want to know if possible when a hangup occur to go to a specific context,an ther call a deadagi |
| 18:10.14 | intralanman | you could use the deadagi for the whole of that extension, or context or whatever.... doesn't asterisk still kill the call if the originating channel hangs up? |
| 18:11.13 | Qwell | no end event on the 4, 6, 7, 8, or 9 were received |
| 18:11.39 | acxty | intralanman, will give a try |
| 18:11.42 | acxty | thanks |
| 18:12.07 | Qwell | gitguy: ^^ |
| 18:12.23 | bkw_ | Qwell: they were sent.. but I see the client didn't send them three times like they should. |
| 18:12.34 | Qwell | bkw_: see the second trace |
| 18:12.37 | Qwell | they are missing |
| 18:12.41 | bkw_ | Qwell: you should still be able to handle that |
| 18:13.07 | gitguy | http://pastebin.com/m1e53b563 |
| 18:13.11 | gitguy | that's what i have in my sip.conf |
| 18:13.14 | Travo1 | intra/bkw_: any idea's? |
| 18:13.28 | gitguy | i sent 123456789 |
| 18:13.32 | Qwell | gitguy: some of the packets are getting lost in your network.. |
| 18:13.47 | gitguy | yeah i though it could be that... |
| 18:13.57 | gitguy | internet @paraguay sux |
| 18:14.05 | intralanman | Travo1: i'm fresh out... |
| 18:14.10 | Travo1 | I think I might just move it to a different pci slot??? |
| 18:14.19 | gitguy | Qwell: how can i test from outside? |
| 18:14.26 | Qwell | test what from outside? |
| 18:14.28 | gitguy | is there a cli softphone i could use over ssh or something? |
| 18:14.35 | gitguy | i don't have remote desktop |
| 18:15.00 | Qwell | hell, quite a bit of the audio is getting lost too |
| 18:15.08 | gitguy | i want to run a softphone somewhere else (outside of my network where internet is faster) and see if i don't loose digits/packets whatever |
| 18:15.22 | Qwell | it's like every other packet from seq 150 to 238 |
| 18:15.43 | Qwell | more like 350 actually...yuck |
| 18:15.49 | Qwell | gitguy: the audio must sound terrible |
| 18:15.52 | gitguy | Qwell: so it's not an asterisk issue? |
| 18:16.04 | acxty | intralanman, no it won't work. If the user hangup on context,exten,3 and the deadagi is place on context,exten,5. It won't get to the deadagi part |
| 18:16.12 | Qwell | acxty: h exten |
| 18:16.18 | gitguy | Qwell: so it's not an asterisk issue? |
| 18:16.31 | bkw_ | gitguy: yes its both ends more on the client end |
| 18:16.33 | Qwell | gitguy: no, the packets are getting dropped before asterisk |
| 18:16.39 | bkw_ | 1. The clients should send three end packets |
| 18:16.45 | bkw_ | 2. Asterisk should be able to handle this |
| 18:17.06 | acxty | Qwell, what I want to do is to run an deadadi() or some methong when a call hangup, no matter in what part of the dialplan it takes effect |
| 18:17.19 | bkw_ | many things out there don't send the end packet at all.. cisco is really good about that one |
| 18:17.37 | acxty | *deadagi() |
| 18:17.41 | intralanman | Qwell: does the h exten get called if the originating channel hangs up? i thought that it didn't |
| 18:18.16 | acxty | Qwell, let me try with the h exten |
| 18:19.09 | gitguy | Qwell: this did happen to people that were in USA, etc, where internet is "good enough" |
| 18:19.22 | acxty | Qwell, thanks the h exten did the work |
| 18:19.31 | bkw_ | gitguy: which client is this? Zoiper? |
| 18:19.39 | gitguy | bkw_: xlite |
| 18:19.44 | *** join/#asterisk steliosk (n=Stelios@athedsl-168543.home.otenet.gr) |
| 18:19.45 | bkw_ | go get zoiper |
| 18:19.47 | bkw_ | xlite is just broken |
| 18:19.48 | gitguy | bkw_: it did happen with zoiper too |
| 18:19.59 | bkw_ | I wonder if it sends three end packets |
| 18:20.01 | gitguy | but i guess thats just my network |
| 18:20.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:20.04 | bkw_ | let me ask zoa |
| 18:20.16 | gitguy | ok, thanks dude |
| 18:20.28 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 18:22.23 | *** join/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 18:22.46 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 18:22.55 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 18:23.12 | whymarkwh | yeeeeeeeeeeeeeeaaaaaaa got the data in mysql thx for all your help |
| 18:29.54 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 18:30.37 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 18:31.09 | mike345 | Hi All, just got my Polycom 650. it is currently SIP firmware 2.0.2.0127. Is downgrading to SIP 1.6.7 still the thing to do for this phone with Astrerisk? |
| 18:31.40 | mcab | mike345: 1.6.7 won't support the 650 |
| 18:31.49 | Corydon76-dig | I'd upgrade to 2.2.2 or later |
| 18:32.07 | Qwell | hasn't 3.0 been out for a while now? |
| 18:32.15 | Corydon76-dig | Yup |
| 18:32.16 | mcab | 3.0.2 is the latest, I think |
| 18:32.23 | mcab | or 3.0.1? |
| 18:32.26 | [TK]D-Fender | 2.2.2 appears to be the latest relatively stable version. 3.0.0. hasn't balanced out yet from what I've heard |
| 18:32.43 | Qwell | [TK]D-Fender: I've had 3.0 on my phone, and it's been up for > a month |
| 18:33.07 | [TK]D-Fender | Qwell: I've heard spotty reports from people who likely pass more volume... |
| 18:33.31 | Qwell | heh |
| 18:33.38 | Qwell | I've probably had like 3 calls in that time |
| 18:33.56 | mike345 | thx - I'll go with the 2.2.2, no need to be bleading edge on the receptionist phone... |
| 18:36.07 | *** part/#asterisk bkw_ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net) |
| 18:36.37 | mintee | when doing `make progdocs` where are these docs actually stored? Is it the `show application x` command? |
| 18:40.18 | gitguy | Qwell: what would you suggest me to do? |
| 18:40.46 | intralanman | pay digium for support ;-) |
| 18:41.49 | Qwell | gitguy: figure out where the packets are being dropped |
| 18:42.12 | Qwell | intralanman: if you're here for no other reason than to troll - leave. |
| 18:42.22 | intralanman | am i trolling? |
| 18:42.31 | intralanman | i thought i was helping? |
| 18:42.33 | Qwell | You already have several of your hostmasks on the ban list. |
| 18:43.08 | intralanman | oh, i thought you only had one.... but i didn't realize i was trolling... i'll go back to not helping |
| 18:44.33 | LinuxMafia | all : any one can suggest me a SIP phone/Vo iphone or whatever it called not so expensive that i can buy and there are free SIP services for it? |
| 18:44.56 | whymarkwh | got the cdr records to work with mysql now i want to link voicemail messages to mysql found a addon but it seems old it says i must edit the make file and change it like here------->http://pastebin.com/m7e1a7583 but there is no mention of it in the make file, might this be for a oder version of * that did not use the menuselect option, can i add it or am i going to cause caos on my server, please help stuck again? |
| 18:45.25 | Qwell | whymarkwh: app_voicemail can store messages in a database already. |
| 18:45.36 | gitguy | Qwell: i have 0 packet loss to the server |
| 18:45.55 | Qwell | gitguy: according to the second packet capture, it's about 50% between seq 150 and 350 |
| 18:46.08 | whymarkwh | thx @Qwell will look into that |
| 18:46.15 | Yourname`` | Maaaan, this sucks. I have a simple crontab script that runs in cron.daily, to restart Asterisk with safe_asterisk just so in the morning I can see the CLI in colors.. but it NEVER works. I execute the cron script manually, and I see the colors. But automaitcally, never! |
| 18:46.20 | Qwell | whymarkwh: doc/odbcstorage.txt |
| 18:47.02 | whymarkwh | thx |
| 18:47.10 | *** join/#asterisk drummond_ (n=rsd@h-67-103-23-130.phlapafg.covad.net) |
| 18:48.36 | LinuxMafia | ~phone |
| 18:48.37 | jbot | [phone] warbling while I'm updating the flash from blob...it's amusing now, it's like it knows I'm erasing its brain. Mwa ha ha ha. |
| 18:48.44 | Yourname`` | ~cisco |
| 18:48.45 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!, or <reply>Cisco phones are expensive crap which should be avoided with extreme prejudice |
| 18:48.50 | Yourname`` | ~cisco7960 |
| 18:49.01 | LinuxMafia | ~phones |
| 18:49.02 | jbot | phones is probably 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 |
| 18:50.00 | Toot | Snoms rock :) |
| 18:50.19 | LinuxMafia | Toot, are they expensive? |
| 18:50.56 | [TK]D-Fender | LinuxMafia: Polycom <---- |
| 18:50.58 | Toot | not in europe, i'm not sure outside to be honest. i have noticed that everyone outside europe seems to say Polycom, which i don't think have a patch |
| 18:51.15 | Qwell | Toot: have a patch? |
| 18:51.29 | Toot | a patch on the snoms, sorry :) |
| 18:51.38 | [TK]D-Fender | Qwell: Yeah... to stop them from smoking ;) |
| 18:51.39 | Qwell | what is a patch? |
| 18:52.04 | LinuxMafia | polycom is expensive |
| 18:52.14 | Toot | hmm i guess an uk expression, by that i mean in my opinion Snoms work much better than the polycoms |
| 18:52.25 | Qwell | thought so |
| 18:52.51 | Toot | entry level snom = £85 going up to £155 for reception quality |
| 18:53.29 | Toot | i never did quite understand why they are poorly rated in the rest of the world. ah well. |
| 18:53.52 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com) |
| 18:55.09 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:56.34 | drummond_ | is there a way to trace the path of a call with the cdr data? |
| 18:56.49 | *** part/#asterisk intralanman (n=lanman@209.85.58.2) |
| 19:00.01 | whymarkwh | where does asterisk store the voicemail files currently in a database of in files /var/spool/asteriks/voicemail/default/???, does this questions dound stupid, i just want to be sure |
| 19:00.35 | LinuxMafia | [TK]D-Fender, what should i choose among these ->http://www.factorydirect.ca/catalog/category_list.php?cat=3560 |
| 19:01.48 | Yourname`` | This Cisco 7960 is such a sweet phone.. too bad it's taking me ages to even try and make it work |
| 19:02.24 | whymarkwh | IP0024 looks nice |
| 19:02.36 | Qwell | whymarkwh: who makes that? |
| 19:03.10 | Qwell | looks kinda cheap.. |
| 19:03.25 | whymarkwh | don't know who, but where is china lol |
| 19:03.33 | LinuxMafia | whymarkwh, IP0024 |
| 19:04.39 | whymarkwh | the cordless 2.4ghz |
| 19:07.09 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:07.42 | [TK]D-Fender | LinuxMafia: NONE |
| 19:07.51 | [TK]D-Fender | LinuxMafia: Stop shopping for garbage |
| 19:08.18 | LinuxMafia | [TK]D-Fender, but these are availabe in canada |
| 19:08.28 | LinuxMafia | [TK]D-Fender, let me check another store |
| 19:08.31 | [TK]D-Fender | LinuxMafia: Don't give me that... Factory direct is leftopver crap. |
| 19:08.39 | [TK]D-Fender | LinuxMafia: how many phones do you need? |
| 19:08.40 | freezey | [TK]D-Fender: just letting you know i figured it out and got it working |
| 19:08.45 | LinuxMafia | just 1 |
| 19:08.48 | [TK]D-Fender | freezey: Congrats |
| 19:08.54 | [TK]D-Fender | LinuxMafia: For what kind of use? |
| 19:09.27 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 19:09.27 | LinuxMafia | i want to be able to use * and to get some free SIP provide rfor it |
| 19:09.51 | drummond_ | i got the 7960 to work no problems |
| 19:10.19 | LinuxMafia | i want others to be able to call me free of charge |
| 19:10.26 | LinuxMafia | drummond_, where do you live? |
| 19:10.37 | drummond_ | right outisde of philadelphia |
| 19:10.51 | LinuxMafia | drummond_, i live in canada |
| 19:10.53 | [TK]D-Fender | LinuxMafia: Got a normal analog phone at your desk already? |
| 19:11.00 | drummond_ | i was just up there a few weeks ago |
| 19:11.09 | drummond_ | in markham |
| 19:11.18 | freezey | [TK]D-Fender: could of just said my firmware was a package in on the asterisk server itself =oP |
| 19:11.19 | LinuxMafia | [TK]D-Fender, no i dont have any phone , just a cellphone |
| 19:11.42 | LinuxMafia | drummond_, wow actually it is close to me |
| 19:11.50 | [TK]D-Fender | freezey: "it isn't part of *, though you might typically shove it there being as it is a *nux server |
| 19:12.03 | drummond_ | hardest part of getting the 7960 was finding the correct upgrade path for the sip firmware |
| 19:12.36 | [TK]D-Fender | LinuxMafia: http://www.voipdepot.ca/index.php?main_page=product_info&cPath=1&products_id=90 |
| 19:13.09 | LinuxMafia | [TK]D-Fender, i can not buy anything online |
| 19:13.21 | drummond_ | where in ca r u? |
| 19:13.27 | [TK]D-Fender | LinuxMafia: And why is that? |
| 19:13.48 | LinuxMafia | [TK]D-Fender, no CC and can not use paypal |
| 19:13.54 | LinuxMafia | drummond_, toronto |
| 19:14.05 | [TK]D-Fender | LinuxMafia: http://www.canadacomputers.com/index.php?do=ShowProduct&cmd=pd&pid=014833&cid=828.480 |
| 19:14.20 | [TK]D-Fender | LinuxMafia: http://www.canadacomputers.com/index.php?do=ShowProduct&cmd=pd&pid=007187&cid=828.480 |
| 19:14.25 | LinuxMafia | [TK]D-Fender, those one are good? |
| 19:14.32 | drummond_ | yep...headed down there after my last day of onsite to hang out till i had to go to the hotel and sleep for my early flight home |
| 19:14.35 | [TK]D-Fender | LinuxMafia: There.. North York, now get off your ass and take a Metro down there.. |
| 19:15.10 | [TK]D-Fender | LinuxMafia: Yes, just plug in an analog phone and you're in business. |
| 19:15.11 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 19:15.40 | drfreeze | HI |
| 19:15.43 | LinuxMafia | [TK]D-Fender, i dont have a phone line |
| 19:15.52 | [TK]D-Fender | LinuxMafia: PAY ATTENTION. |
| 19:16.07 | drfreeze | I'm defining channels inside zapata.conf file with: channel => 1-7 |
| 19:16.08 | [TK]D-Fender | LinuxMafia: Plug a PHONE into that DEVICE and you are "in business". |
| 19:16.09 | LinuxMafia | [TK]D-Fender, oh this is adaptor |
| 19:16.22 | LinuxMafia | i got it |
| 19:16.24 | drfreeze | How would I change this to get all but 4? |
| 19:16.31 | drfreeze | channel => 1-3 |
| 19:16.34 | drfreeze | channel => 5-7 |
| 19:16.42 | [TK]D-Fender | drfreeze: channel => 1-3,5-7 |
| 19:16.54 | [TK]D-Fender | drfreeze: Either way would work. |
| 19:17.02 | drfreeze | [TK]D-Fender: thx. BTW, can each channel be defined on a separate line? |
| 19:17.11 | [TK]D-Fender | drfreeze: Any which way. |
| 19:17.16 | drfreeze | cool |
| 19:17.43 | [TK]D-Fender | LinuxMafia: not make sure to call them up and verify that its unlocked. |
| 19:17.46 | [TK]D-Fender | now* |
| 19:17.47 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 19:18.01 | [TK]D-Fender | LinuxMafia: This one mentions "Vonage" which is a bad thing. |
| 19:18.31 | LinuxMafia | [TK]D-Fender, what you mean unlocked? |
| 19:19.18 | [TK]D-Fender | LinuxMafia: Lik as in "Bell owns my sorry ass and I can't take my phone to another provider even if I wanted to". |
| 19:19.38 | [TK]D-Fender | LinuxMafia: in this case some ATA's are sold LOCKED to a particular providers server thereby screwing you. |
| 19:19.42 | LinuxMafia | [TK]D-Fender, no "Vaughan" is a city name |
| 19:20.14 | drummond_ | i was lucky..i got my sipura before they were bought by cisco, so it's not tied to any one provider |
| 19:20.15 | [TK]D-Fender | LinuxMafia: I'm not talking about Vaughn, ON FFS. |
| 19:20.18 | [TK]D-Fender | ~vonage |
| 19:20.25 | jbot | hmm... vonage is a bunch of monkeys |
| 19:20.38 | [TK]D-Fender | LinuxMafia: Vonage... the ITSP <------------- |
| 19:20.44 | LinuxMafia | Vonage to Vonage Calls |
| 19:20.50 | LinuxMafia | oh |
| 19:21.05 | [TK]D-Fender | LinuxMafia: You are welcome back on our planet at any time... |
| 19:21.17 | LinuxMafia | [TK]D-Fender, oh thanks alot |
| 19:21.27 | [TK]D-Fender | LinuxMafia: All part of the service :p |
| 19:21.41 | LinuxMafia | i will get back to you sir |
| 19:21.54 | drummond_ | ughhhhh i hate snmpv3 |
| 19:22.18 | [TK]D-Fender | LinuxMafia: Typical parts you're looking for "PAP2T-NA" and "SPA-2102". the former is typically cheaper. You can already see the approx cost. |
| 19:22.29 | [TK]D-Fender | LinuxMafia: Either should do just fine for you. |
| 19:22.42 | jasonwoot | pap2 sux, but it's so cheap |
| 19:23.02 | *** part/#asterisk nn (n=kaeru@unaffiliated/nn) |
| 19:24.32 | [TK]D-Fender | jasonwoot: Aside from lack of T.38, dual G.729, a router and a few tiny bits, its essentially an ATA like the rest. What else is "bad" about it? |
| 19:24.54 | *** join/#asterisk intralanman (n=lanman@209.85.58.2) |
| 19:26.38 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net) |
| 19:26.39 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-233.usadatanet.com) |
| 19:27.46 | drfreeze | I've got an interesting problem: I have a hunt group with 7 numbers. For example, 555-1000, 555-1001, 555-1002, 555-1003, 555-1004, 555-1005 |
| 19:28.19 | Travo1 | how come my caller id from my x100p says "Joe User" |
| 19:28.47 | drfreeze | From outside, I can call any number and my dial plan gets executed, but if I dial 555-1003 (number 4 in the list), it bypasses the initial dialplan and rings directly |
| 19:29.10 | [TK]D-Fender | Travo1: pastebin your zapata.conf |
| 19:29.29 | drfreeze | Travo1: sip.conf. set callid |
| 19:29.32 | drfreeze | callerid |
| 19:29.42 | [TK]D-Fender | drfreeze: nope |
| 19:29.53 | syntaxman | I've being searching for info about the "big picture" of using asterisk... |
| 19:30.27 | drummond_ | what's not to like about it ;-) |
| 19:30.43 | *** join/#asterisk LARefugee (n=victorr@c-76-104-191-194.hsd1.wa.comcast.net) |
| 19:30.44 | drfreeze | [TK]D-Fender: huh? |
| 19:30.56 | syntaxman | What I have found is that I need 2 services, a VoIP _> PSTN gateway, and a DID service to supply an in-coming PSTN phone number. Is this accurate? |
| 19:31.18 | drfreeze | Any ideas as to why a single number in a hunt group is misbehaving. Also, I cannot call out on that number |
| 19:31.31 | [TK]D-Fender | syntaxman: Nope, you only need a PC to run * on. Everything else is negociable. |
| 19:32.00 | [TK]D-Fender | drfreeze: You are changing your channels around, go look how you've got them set up. We jsut talked about EXCLUDING Channel 4. |
| 19:32.19 | [TK]D-Fender | drfreeze: Don't forget you have to reload chan_sip to make those chanes go into effect |
| 19:32.33 | syntaxman | [TK]D-Fender: Assuming I want to set it up so I can freely receive and place calls to any PSTN network, is that still true? |
| 19:32.34 | [TK]D-Fender | drfreeze: And you callerID comment was on the wrong track. |
| 19:32.35 | LARefugee | my registrations to vonage and fwd (sip) were working for days. Now they keep timing out. I'm behind a nat (tomato). Ideas? |
| 19:32.52 | [TK]D-Fender | syntaxman: Many ways do to that. What do you WANT? |
| 19:32.54 | drummond_ | any firewall changes or anything like that |
| 19:33.01 | LARefugee | no |
| 19:33.04 | [TK]D-Fender | LARefugee: Read up : |
| 19:33.06 | [TK]D-Fender | ~sipnat |
| 19:33.07 | 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 |
| 19:33.11 | drfreeze | I have an idea - may be that I am using only 7 channels of 8 and my TMD is not configured properly |
| 19:33.57 | syntaxman | [TK]D-Fender: I want to control my own voicemail, call forwarding, etc. with * , and have the phone work the same as it always has (for the most part). |
| 19:34.02 | LARefugee | thanks. appreciate it. |
| 19:34.15 | syntaxman | [TK]D-Fender: everyone should be able to contact me w/o extra work, and vice versa. |
| 19:34.19 | LARefugee | ~x100m |
| 19:34.45 | syntaxman | [TK]D-Fender: And I want to play around and learn ;-) |
| 19:35.45 | [TK]D-Fender | syntaxman: When you say "phone work the same" what kind of "phone" are we talking about? |
| 19:36.27 | [TK]D-Fender | syntaxman: Do you have analog lines right now? how many? Got a PBX? How many phones attached? What kind? |
| 19:37.11 | *** join/#asterisk budgetdedicated (n=marius@s5593c2e9.adsl.wanadoo.nl) |
| 19:37.17 | syntaxman | It's a regular PSTN phone, but I'm willing to use other kinds of phones. To me, working the same means that I want to have seamless connectivity to PSTN so that my fam and friends won't hate me. |
| 19:37.29 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
| 19:37.53 | syntaxman | [TK]D-Fender: This is a very new effort. I have two PSTN lines, and a DSL connection. |
| 19:38.01 | drummond_ | you can always get an fxo card for your * box |
| 19:38.02 | Travo1 | [TK]D-Fender: http://pastebin.com/m73150b68 |
| 19:38.12 | syntaxman | [TK]D-Fender: That's it for now. I'm trying to find out what I need. |
| 19:38.19 | Travo1 | D-Fender: That is of the zapata.conf file |
| 19:39.06 | drfreeze | Which zaptel.conf file does asterisk use? /etc/asterisk/zaptel.conf or /etc/zaptel.conf ? |
| 19:39.36 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 19:39.37 | MooingLemur | drfreeze: asterisk uses /etc/asterisk/zapata.conf |
| 19:39.39 | syntaxman | drummond_: Ah.. so, installing an FXO card will allow me to do my *own* gateway between voip and pstn? |
| 19:39.54 | drummond_ | correct..... |
| 19:39.56 | MooingLemur | but zaptel uses /etc/zaptel.conf to init the card |
| 19:40.16 | intralanman | syntaxman: taht or an spa-3000 depending how cheap you are and how many channels you need |
| 19:40.19 | drummond_ | that way astersisk with be the first point the pstn calls hit |
| 19:41.11 | syntaxman | intralanman: Well, until I really get some more experience I will be _very_ cheap :-) |
| 19:41.18 | drummond_ | what i did at my previous job just because i wanted to use a cisco phone at my desk was get vic-2fxo card for a router he had sitting in the office, so i ran the lines to that instead of my regular phone. worked like a champ. it's the same principle with * |
| 19:41.47 | drfreeze | MooingLemur: can you tell me if I need to remove a line, eg fxsks=8, if I am not using one of the POTS lines? The file says that it was autogenerated and not to edit |
| 19:41.52 | JayTee52 | syntaxman, try this one because the SPA3000 has been discontinued: http://www.telephonydepot.com/product_p/105-054-312.htm |
| 19:41.56 | syntaxman | intralanman: I'm planning to compare various PBX packages, etc. I'm really starting a full-scale investigation at this point. |
| 19:42.22 | intralanman | syntaxman: which ones do you have in mind? sipx, asterisk, etc? |
| 19:42.45 | drfreeze | freeswitch |
| 19:42.56 | syntaxman | intralanman: Asterix was first on the list only because of it's visibility. I discovered FreeSwitch this morn, too. |
| 19:43.07 | intralanman | drfreeze: yeah, i guess you could use that as a pbx, although it's really more of a switch |
| 19:43.08 | syntaxman | I didn't really grasp the diffs yet, though. |
| 19:43.22 | drummond_ | syntaxman: http://cgi.ebay.com/Authentic-X100P-SE-FXO-PCI-for-Digium-Asterisk-VoIP-PBX_W0QQitemZ130219507323QQihZ003QQcategoryZ61839QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 19:43.27 | *** join/#asterisk niddnet (n=HelloWor@87-194-144-141.bethere.co.uk) |
| 19:43.42 | JayTee52 | *headdesks* |
| 19:43.50 | MooingLemur | drfreeze: I don't know anything about freeswitch |
| 19:43.58 | drfreeze | intralanman: right. And you can make it a pbx if you want to code a lot of * features into it. |
| 19:44.01 | niddnet | Hello all - someone sent me here to get some help with somethign that should work but does not. Anyone up for the challenge? :) |
| 19:44.05 | JayTee52 | can't believe someone would recommend an X100P card |
| 19:44.13 | ManxPower | Since real X100Ps have not been made in many years, and the X101P has not been made in at least 3 years, I doubt that is really an authentic card |
| 19:44.14 | MooingLemur | or you're talking to someone else? |
| 19:44.15 | MooingLemur | heh |
| 19:44.21 | drfreeze | MooingLemur: the fxsks issue is with asterisk |
| 19:44.24 | syntaxman | Is there any benefit to, say, running a switch/PBX on a hosted box, and running only the clients at the endpoints? |
| 19:44.30 | intralanman | drfreeze: actually it has support for a lot of features that asterisk has too... just better support for a lot of them IMHO |
| 19:44.31 | syntaxman | Or is that a bigger headache? |
| 19:44.35 | MooingLemur | drfreeze: mine doesn't say anything about being auto-generated |
| 19:44.56 | MooingLemur | so I don't know what your specific distribution is doing |
| 19:45.07 | anthm | drfreeze, not really code them more like configure them |
| 19:45.26 | drfreeze | MooingLemur: Let's assume I have a single TDM card with 4 channels |
| 19:45.44 | drfreeze | Also, assume I am going to use 3 of those possible four channels |
| 19:45.49 | syntaxman | <PROTECTED> |
| 19:45.53 | syntaxman | oops ;-)] |
| 19:46.07 | niddnet | I have a problem whereby my incoming routes aren't... well... routed. The * just plays "not in service" - yet the trunk goes red when the call is received, and it gets logged as an answered call |
| 19:46.17 | drfreeze | I'm assuming that I only define 1-3 in the zaptel.conf file |
| 19:46.23 | intralanman | does * have a built-in web vm yet? if not, will it ever |
| 19:46.35 | niddnet | This is on an Elastix box, but identical settings worked on Trixbox |
| 19:46.35 | drfreeze | or * will think it actually has 4 lines that it can use to make call and to receive calls |
| 19:47.40 | MooingLemur | drfreeze: This is my setup with a two port card, with one port in use with a PRI: http://rafb.net/p/xTSP3r75.html |
| 19:48.35 | MooingLemur | that's just for setting up the card with ztcfg |
| 19:48.45 | MooingLemur | which happens during my init scripts |
| 19:48.52 | drummond_ | has anyone ever setup dial by name with asterisk? |
| 19:49.37 | drfreeze | MooingLemur: I would think you would get the autogenerated header with using ztcfg |
| 19:49.51 | dFence | has anyone used the unoGSM with *? can i integrate it like an ISDN card? |
| 19:50.40 | MooingLemur | ztcfg doesn't create the file |
| 19:50.46 | MooingLemur | it reads the file to configure the card |
| 19:50.59 | MooingLemur | something else is creating the file in your distro |
| 19:55.10 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 19:55.44 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:56.09 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:56.50 | niddnet | I'd be happy for someone to login and check things out for me :) |
| 19:56.54 | marc7 | is there any reason that the playback of mp3 files (using ControlPlayback) would be much quieter than the playback of WAV files? |
| 20:03.47 | Travo1 | Can some one assist me in figuring out why my caller id on my x100p says "Joe user" here is my zapata.conf file http://pastebin.com/m73150b68 |
| 20:05.39 | [TK]D-Fender | Travo1: On a call coming IN on it? |
| 20:06.05 | [TK]D-Fender | syntaxman: Ok, so you've got 2 PSTN lines. You want to use those? |
| 20:06.17 | Travo1 | yes |
| 20:06.32 | Travo1 | D-Fender: I have a PSTN and a VoicePulse VOIP line |
| 20:06.48 | Travo1 | D-Fender: I got the x100p for my existing line to catch calls |
| 20:07.50 | [TK]D-Fender | Travo1: set "callerid=asreceived" and reload cha_sip |
| 20:09.34 | Travo1 | D-Fender: Same Result |
| 20:10.02 | Travo1 | D-Fender: reload cha_sip is depreciated, I did a restart now instead |
| 20:10.16 | russellb | woah |
| 20:10.21 | russellb | those are very different operations :) |
| 20:10.27 | Travo1 | oh wait... i don't have caller ID, DOH |
| 20:10.30 | russellb | reload reloads config .. restart restarts the process |
| 20:10.42 | Travo1 | okay, well can I set it to "LAND LINE" |
| 20:12.15 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 20:15.14 | *** join/#asterisk bminish (n=bminish@ns1.westnet.ie) |
| 20:16.17 | syntaxman | [TK]D-Fender: Yeah. I like the idea of getting an FXO/FXS now that I'm looking. |
| 20:16.47 | [TK]D-Fender | syntaxman: Ok, So I would advise a piece of hardware to interface with those lines. Now what do you haev by way of phones right now? What might you like to have? |
| 20:17.33 | syntaxman | [TK]D-Fender: I have generic analog multiline phones. |
| 20:17.51 | Travo1 | okay, caller id name has been fixed :) all that is left is to tell asterisk to abort the voice prompt if I pickup while ringing, any idea's. I've googled this, maybe wrong keyword |
| 20:17.55 | syntaxman | [TK]D-Fender: I should also mention that I've got a hosted machine where I could run some of this stuff. |
| 20:18.52 | [TK]D-Fender | syntaxman: well you have analog lines where you are. Thats a important factor. Your multi-line analog phones are reusable, but the "multi-line" part is a dead issue. |
| 20:19.05 | syntaxman | [TK]D-Fender: The original idea was to install the PBX or softwitch on that machine, and have an incoming and outgoing gateway to pstn. |
| 20:20.12 | [TK]D-Fender | syntaxman: You have DSL where you aren' you don't want to pump calls in & out of that when the lines are in the same building. |
| 20:20.22 | syntaxman | [TK]D-Fender: I got hung up on what services would really be needed and where to find them. FWD doesn't seem to offer either of those, and LibreTel only offers incoming at the moment. |
| 20:20.25 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 20:21.34 | [TK]D-Fender | syntaxman: No need to think about either of those right now. You have analog lines and thats what it seems we're focusing on. |
| 20:23.37 | Qwell | [TK]D-Fender: it's not so much a dead issue |
| 20:23.45 | Qwell | you could use them - it would just take 2 ports |
| 20:24.05 | Qwell | there's nothing fancy there - it's 2 lines in a very literal sense |
| 20:24.13 | [TK]D-Fender | Qwell: I'll rephase as "asking to make call handling a PITA" :) |
| 20:24.34 | [TK]D-Fender | Qwell: And adding cost you aren't likely to get the benifit from. |
| 20:24.38 | syntaxman | ooooh, pain! fun! |
| 20:24.43 | Qwell | [TK]D-Fender: sure, I'll give you that |
| 20:25.16 | [TK]D-Fender | syntaxman: I'll be around later, GTG for now, poke around a bit. |
| 20:25.21 | [TK]D-Fender | BBL |
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| 20:38.21 | *** part/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 20:39.13 | [hC] | If i am forced to use relaxdtmf=yes in zapata.conf on a PRI, is that indicative of another problem, and relaxdtmf is only a bandaid? (gains, etc)? |
| 20:39.13 | *** part/#asterisk ctaloi (n=ctaloi@nat-66-218-1-233.usadatanet.com) |
| 20:39.25 | [hC] | relaxdtmf seems to cure the problem, but it still occasionally misses the odd DTMF keypress |
| 20:40.20 | *** join/#asterisk ManxPower (n=manxpowe@97-81-241-251.dhcp.lds.al.charter.com) |
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| 20:48.38 | JayTee52 | quittin time, I'm out! |
| 20:48.39 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
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| 20:59.35 | LARefugee | Very frustrating. Still can't register to vonage and fwd (sip). It did work just yesterday. Ideas? |
| 21:01.11 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585300.dsl.bell.ca) |
| 21:01.19 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
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| 21:09.33 | dFence | hm... how's the linksys SPA922 working together with *? |
| 21:11.24 | LinuxMafia | these 2 are different ? --> 1) http://www.canadacomputers.com/index.php?do=ShowProduct&cmd=pd&pid=007187&cid=828.480 , 2)http://www.staples.ca/ENG/Catalog/cat_sku.asp?CatIds=&webid=591137&affixedcode=WW |
| 21:13.29 | lanning | Canadian Postal Code? |
| 21:13.57 | LinuxMafia | lanning, m1l1l1 |
| 21:13.58 | Qwell | h0h0h0 |
| 21:15.33 | LinuxMafia | they look the same |
| 21:15.41 | b11d` | mine used to be p0t-2w0 |
| 21:15.43 | b11d` | it was awsome |
| 21:15.48 | LinuxMafia | but one is higher price dont know why |
| 21:16.07 | lanning | I can't find any differences |
| 21:16.22 | Qwell | LinuxMafia: neither are good |
| 21:16.30 | Qwell | they are locked to Vonage, and cannot be used with Asterisk |
| 21:16.45 | LinuxMafia | Qwell, oh |
| 21:17.06 | LinuxMafia | Qwell, so what should i use , i am in canada , can not buy online |
| 21:17.18 | Qwell | why can't you buy online? |
| 21:17.36 | Corydon76-dig | He's underage and can't get a credit card |
| 21:17.40 | LinuxMafia | Qwell, no CC can not use paypal |
| 21:17.48 | LinuxMafia | Corydon76-dig, lol |
| 21:17.56 | *** join/#asterisk shido6 (n=shido6@74-130-51-159.dhcp.insightbb.com) |
| 21:18.09 | shido6 | anyone successfully made a 7920 phone work behind nat? |
| 21:18.22 | Qwell | pretty much any ATA you get from a brick and mortar store is going to be locked to some provider |
| 21:18.37 | shido6 | i've forwarded 2000 UDP and 10000-20000 UDP to the ip of the device |
| 21:18.45 | Qwell | 2000 tcp |
| 21:19.17 | Corydon76-dig | LinuxMafia: if your 1337, you can crack it |
| 21:19.23 | LinuxMafia | Qwell, so what you suggest? |
| 21:19.33 | *** part/#asterisk jefferai (n=quassel@amarok/developer/mitchell) |
| 21:19.42 | Qwell | if Corydon76-dig is right - use your parents credit card, and give them cash :p |
| 21:19.57 | shido6 | they can hear me |
| 21:20.02 | LinuxMafia | Qwell, i am 31 |
| 21:20.02 | shido6 | but i cant hear them |
| 21:20.23 | Corydon76-dig | LinuxMafia: or crack one, like I did: http://asterisk.drunkcoder.com/hacks/ats-config/ |
| 21:20.44 | LinuxMafia | but i dont like post deliveries |
| 21:20.59 | Corydon76-dig | LinuxMafia: into every life some rain must fall |
| 21:21.15 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-187-7.hsd1.wa.comcast.net) |
| 21:21.34 | LinuxMafia | Corydon76-dig, i can not crack |
| 21:22.10 | Corydon76-dig | LinuxMafia: that's my crack. You're welcome to use it. |
| 21:23.12 | LinuxMafia | thanks but it works for any type |
| 21:23.25 | Corydon76-dig | Only works for that phone |
| 21:23.54 | LinuxMafia | oh |
| 21:24.15 | LinuxMafia | you know i want to get my parents phone too |
| 21:24.24 | LinuxMafia | so we can call each other free |
| 21:25.08 | LinuxMafia | that is what i am tring to figure out |
| 21:25.25 | LinuxMafia | how can i call some one free of charge |
| 21:25.51 | CrashHD | string and a can |
| 21:25.54 | CrashHD | works great |
| 21:26.01 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:26.16 | LinuxMafia | CrashHD, what do you mean? |
| 21:26.21 | LinuxMafia | are you joking |
| 21:27.01 | LinuxMafia | if yahoo voice chat is free , skype is free and lot more , there got to be a way using these VOIP phones |
| 21:27.22 | LinuxMafia | for free |
| 21:27.45 | LinuxMafia | my ISP rip me off pretty much every month |
| 21:28.33 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-83-43.vif.net) |
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| 21:38.00 | CrashHD | string and a can |
| 21:38.02 | CrashHD | I'm telling you |
| 21:38.03 | CrashHD | can't go wrong |
| 21:39.29 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 21:44.13 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:47.31 | marc7 | is there any reason that the playback of mp3 files (using ControlPlayback) would be much quieter than the playback of WAV files? i'm sorry if this was answered earlier, my connection has been a little shoddy |
| 21:48.54 | *** join/#asterisk RoyK (n=roy@ip-177-24-149-91.dialup.ice.no) |
| 21:57.53 | *** join/#asterisk eXistenZ (i=pectic@unaffiliated/existenz) |
| 22:00.56 | CCFL_Man2 | there are no cheap gsm gateways it seems |
| 22:01.40 | bkruse | blinks |
| 22:03.05 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
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| 22:04.52 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 22:05.38 | rob0 | If my inbound context contains an "s" extension, and an include'd context also has "s", which "s" wins? |
| 22:05.55 | `Sauron | inbound |
| 22:05.56 | rob0 | (when a call comes to s@context) |
| 22:06.02 | rob0 | ty |
| 22:06.17 | `Sauron | I don't actually know for sure. find out. |
| 22:06.19 | `Sauron | i.e. test |
| 22:06.33 | rob0 | so include adds but doesn't override? Oh sure, I am testing :) |
| 22:07.07 | `Sauron | I would assume something like "It depends on the order configured" |
| 22:07.17 | `Sauron | But that sounds like it would make too much sense |
| 22:07.27 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 22:07.30 | `Sauron | So it's probably something weird involving days that end with "y" |
| 22:08.06 | CCFL_Man2 | anyone offer gsm gateways for decent prices? |
| 22:08.38 | `Sauron | rob0: Ask [TK]D-Fender - he's the ma. |
| 22:08.39 | *** join/#asterisk yatesb (n=real@AC9E3EAE.ipt.aol.com) |
| 22:08.39 | `Sauron | *n |
| 22:08.42 | rob0 | [TK]D-Fender: If my inbound context contains an "s" extension, and an include'd context also has "s", which "s" wins (when a call comes to s@inbound)? |
| 22:08.48 | rob0 | way ahead of ya :) |
| 22:09.02 | yatesb | hello . when i try to call with astersik that works. but it gives me an error "failed so falling back to exten 's' what i must do ? can anybody help me ? im tryng to use a Sip. thanks. |
| 22:09.24 | [TK]D-Fender | rob0, I think if the include comes after then the include ttakes over. |
| 22:09.37 | yatesb | im tryng to make outbound calls. |
| 22:09.42 | yatesb | can anybody helps me ? |
| 22:09.45 | [TK]D-Fender | rob0, Of course this is something you answer for yourself in about 10 seconds flat. |
| 22:10.04 | rob0 | haha well a few more than 10 seconds :) |
| 22:10.11 | `Sauron | TKD: IT would make sense that if the include came before the exten =>, it would supercede it, if the include comes after the exten => it wouldn't |
| 22:10.16 | [TK]D-Fender | rob0, you = slow :p |
| 22:10.30 | rob0 | doesn't deny it |
| 22:11.13 | *** join/#asterisk Braxus (n=braxus@netblock-68-183-228-155.dslextreme.com) |
| 22:12.06 | CCFL_Man2 | anyone want to sell a gsm gateway? |
| 22:12.18 | `Sauron | rob0: so what's the answer? You've had 2 minutes. :p :) |
| 22:14.35 | mgdm | CCFL_Man2: sell, not buy> |
| 22:14.39 | mgdm | s/>/?/ |
| 22:20.32 | CCFL_Man2 | mgdm: you have one you can sell me for a decnt price? |
| 22:24.11 | mgdm | CCFL_Man2: No :) |
| 22:24.25 | CCFL_Man2 | ahh |
| 22:24.26 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 22:26.31 | eXistenZ | 1 |
| 22:29.54 | CCFL_Man2 | i was thinking of playing around with a gsm gateway |
| 22:32.11 | marc7 | no thoughts on why MP3's would play at a quieter volume than WAV's when invoked from ControlPlayback(), even though that on my computer, the two files sound identical? |
| 22:32.58 | *** join/#asterisk RoyK (n=roy@ip-177-24-149-91.dialup.ice.no) |
| 22:35.28 | ManxPower | `Sauron: other than a few specific situations, order does not matter. |
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| 22:57.06 | *** join/#asterisk railsmunky (n=nick@5acd40d2.bb.sky.com) |
| 22:57.34 | railsmunky | hey. Is anyone familiar with sangoma cards in the uk. I'm getting a buzz and no other audio :/ |
| 22:57.59 | railsmunky | on a bt pri |
| 22:58.05 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 22:58.28 | ManxPower | railsmunky: contact Sangoma |
| 22:58.38 | dkwiebe | greg? |
| 22:58.40 | railsmunky | okedoke. Cool |
| 22:58.43 | dkwiebe | sorry |
| 22:59.12 | JT | railsmunky: try a different pci slot |
| 23:00.11 | railsmunky | oh i see - also good advice :) |
| 23:00.20 | railsmunky | i'll give it a go when i get in |
| 23:01.20 | JT | a buzz, so it is sending and receiving calls? |
| 23:04.28 | railsmunky | i've not setup outgoing yet. |
| 23:04.41 | railsmunky | So when i call in and pickup the call i get a buzz for both partier |
| 23:04.43 | railsmunky | parties ;) |
| 23:05.10 | railsmunky | but yeah it's working for receiving calls |
| 23:05.25 | railsmunky | didnt know if it was signalling or something |
| 23:07.30 | ManxPower | railsmunky: you have a hardware issue. |
| 23:07.45 | railsmunky | as in broken hardware? |
| 23:07.50 | railsmunky | or just configuration? |
| 23:07.55 | JT | what are the computer specifications? |
| 23:09.19 | CCFL_Man2 | hard to find a gsm gateway with an english manual, only engrish |
| 23:10.21 | railsmunky | It's a dell 2950 |
| 23:10.29 | railsmunky | <PROTECTED> |
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| 23:12.50 | ManxPower | railsmunky: There is no config that I know of that would allow calls to work but audio not work with a PRI |
| 23:12.54 | JayTee52 | railsmunky, getting any errors on the front panel of the server? |
| 23:13.25 | railsmunky | It's not really audio not working just a dodgy buzzing. |
| 23:13.26 | ManxPower | CCFL_Man2: If you can't read Engrish then you are in the wrong business |
| 23:13.39 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-116-41.PHNX.QWEST.NET) |
| 23:13.45 | CCFL_Man2 | ManxPower: lol |
| 23:14.06 | ManxPower | railsmunky: clicks could be a sync source issue, buzzing is nothing I can imagine on a PRI that would be a config issue |
| 23:14.07 | railsmunky | JayTee52, nope (apart from the standard stuff where i don't have both PSU's conencted etc..) |
| 23:15.21 | railsmunky | Manxpower: would it just not connect or something like that. I setup as a T1 for example by mistake initially and it wouldn't connect. But now it's happy days connecting etc.. |
| 23:16.37 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:19.35 | JT | railsmunky: no it's a hardware issue |
| 23:19.55 | railsmunky | that's exactly what i hoped it wasn't :) |
| 23:20.04 | railsmunky | I did think it could have been though |
| 23:20.39 | railsmunky | Thanks guys - i'll go for sangoma in the morning and get them to confirm it |
| 23:22.48 | joat | n00b question: is there a howto for configuring stun on asterisk? |
| 23:27.08 | Corydon76-dig | Asterisk doesn't use STUN directly |
| 23:28.15 | *** join/#asterisk bminish (n=bminish@ns1.westnet.ie) |
| 23:28.15 | Corydon76-dig | If you're in a circumstance where stun might help, it's probably wise to set the peer to use nat=yes |
| 23:28.29 | Corydon76-dig | and things will probably just work |
| 23:28.42 | rob0 | Set phasers to KILL. |
| 23:31.17 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:32.37 | dFence | kyyyyyyyyyle's moooooooooooom is a biiiiiiiiiiiiiiiiiiiiiiitch |
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| 23:34.27 | *** part/#asterisk dFence (n=chatzill@ings-d93220ce.pool.mediaWays.net) |
| 23:34.35 | *** join/#asterisk dFence (n=chatzill@ings-d93220ce.pool.mediaWays.net) |
| 23:35.05 | dFence | hm... *scratch* does asterisk come with the gui by default or only in the *NOW distro? |
| 23:36.57 | *** join/#asterisk xezz (n=asdasd@athedsl-199344.home.otenet.gr) |
| 23:37.07 | Nugget | http://www.voip-info.org/wiki-Asterisk+GUI |
| 23:37.52 | drmessano | Asterisk comes with the best GUI ever |
| 23:37.53 | drmessano | _ |
| 23:38.10 | xezz | hello, is there an fms system for asterisk based pbx's ? |
| 23:38.16 | drmessano | fms? |
| 23:38.25 | dFence | Nugget: thx |
| 23:38.25 | xezz | front management system |
| 23:38.58 | ManxPower | xezz: We call them GUIs and they are not supported on this channel. See the /topic or what Nugget told you |
| 23:39.13 | xezz | no mate |
| 23:39.14 | jblack | dFence: No, asterisk doesn't come with a gui. The management is done on the console. |
| 23:39.17 | dFence | 's thinking about throwing Nugget on the floor and give a huge smooch... been looking for a site like that for hours!!!! |
| 23:39.42 | xezz | look, i have 20 asterisk pbx's with guis for management |
| 23:39.45 | dFence | jblack: was talkin bout the web-interface |
| 23:39.50 | xezz | but an fms |
| 23:39.57 | xezz | will gather all cdr reports for example |
| 23:39.59 | ManxPower | xezz: then no such thing exists for Asterisk. |
| 23:40.09 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 23:40.09 | ManxPower | CDrs are BACK end, not FRONT end. |
| 23:40.12 | tzafrir_home | dFence, have you looked at the topic of #asterisk-gui ? |
| 23:40.13 | xezz | it should be then :) |
| 23:40.22 | jblack | there's plenty of third party guis available. I think there's even a couple web based ones. |
| 23:40.24 | ManxPower | There are several billing packages for Asterisk. |
| 23:40.36 | xezz | or gather all errors for correlation |
| 23:40.44 | tzafrir_home | sorry, wrong page |
| 23:40.58 | ManxPower | xezz: Asterisk is not a PBX. Asterisk a PBX toolkit that lets you build all that cool stuff you need. |
| 23:41.21 | dFence | tzafrir_home: at my last encounter with * the NOW-branch came only in a bundle, wanted just the webadmin-part |
| 23:41.29 | dFence | *GARRR* too many windows |
| 23:41.35 | xezz | ManxPower, i said asterisk based* |
| 23:42.23 | ManxPower | xezz: then I refer you to the Wiki, which should list pretty much every 3rd party addon available for Asterisk, but I highly doubt any one of them calls itself a "front management package". |
| 23:42.38 | ManxPower | Personally the phrase sounds like marketing for a jock |
| 23:43.36 | xezz | then you havent been into analog telephony bro ^^ |
| 23:43.42 | ManxPower | If you are looking for specific things like billing, for example, there are add ons that do that. |
| 23:43.45 | xezz | fms is used by all big telcos |
| 23:43.50 | xezz | for error correlation |
| 23:44.02 | drmessano | I googled for Front Management System |
| 23:44.09 | jblack | heh. "RandomAsteriskFrontend: Better support for your private stuff" |
| 23:44.11 | drmessano | I found one for managing bowling alleys |
| 23:44.16 | ManxPower | fms Big Telcos also use H323 and 30 yr old switches. |
| 23:44.16 | xezz | hahaha |
| 23:44.54 | drmessano | ManxPower: touche |
| 23:44.56 | dFence | hm... wondering: is echo cancellation a topic of intereset if i have an armada of SIP-Phones on one side and an ISDN line on the other side of the *PBX? |
| 23:45.05 | ManxPower | My Asterisk installs do failover routing for error'd calls, logging of the errors, etc |
| 23:45.18 | drmessano | being more like AT&T is not a good thing |
| 23:45.57 | ManxPower | dFence: Echo cancellation is normally done at the PSTN/VoIP gateway, but could be done most anywhere on the PSTN. It cannot be done after a call is converted to VoIP. |
| 23:47.18 | ManxPower | I prefer to do EC outside of Asterisk using dedicated hardware like Tellabs |
| 23:48.13 | dFence | ManxPower: so it is sth i should take into consideration? planning on migrating our hotel (~30 phones, seldomly more than 2-3 phones in use the same time) to * including exchanging the analog phones for sip-phones |
| 23:48.36 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:48.51 | drmessano | I wouldnt use SIP phones |
| 23:49.01 | dFence | y?! |
| 23:49.11 | drmessano | Get a channel bank and put a bunch of $20 phones in the rooms |
| 23:49.27 | ManxPower | In a hotel environment I would stuck to analog phones unless you are a very high end hotel and can write off breakage and theft. |
| 23:50.09 | drmessano | besides |
| 23:50.14 | drmessano | THe wire is already in place |
| 23:50.21 | drmessano | No Cat5 to pull |
| 23:50.49 | ManxPower | drmessano: Any company like AT&T, which is currently a mismash of different companies desperately trying to appear to customers as a cohesive organization is a bad thing. |
| 23:51.03 | ManxPower | Sprint/Nextel is the same way |
| 23:51.21 | drmessano | Indeed |
| 23:51.51 | ManxPower | "The merger will be good for customers!" "No, the merger (might) be good for shareholders, but will never be good for customers." |
| 23:51.57 | drmessano | The "All new AT&T" is like a bunch if branch offices that are ignorant to each others existance |
| 23:52.01 | dFence | the idea behind the whole switch thing is to provide cost-effective internet access in every room and since the hipath is from 1992 (or earlier!?) and it's gonna break down some time my idea was to move to * gradually |
| 23:53.03 | ManxPower | I was told by the Alabama PUC that you could not have Call Forward Busy or Call Forward No Answer go to a toll number. I was also told that even if it could go to a toll number the CALLER would pay the toll charges. I thanked her and hung up, as she is an idiot. |
| 23:53.11 | ManxPower | (this was last week) |
| 23:53.33 | ManxPower | </rant> |
| 23:54.13 | dFence | and the cat5 isn't that much of a deal, wiring is so extensive, you could add ~infinite~ rooms from that point |
| 23:54.49 | ManxPower | dFence: Do you have much experience in hotel industry? |
| 23:54.59 | dFence | ManxPower: abt 7 yrs |
| 23:55.08 | ManxPower | dFence: You are not concerned a guest will knock a $200 sip phone on the floor and break it? |
| 23:55.24 | Qwell | If you've ever stayed in a hotel, you would understand the reasons for not putting a SIP phone in a room |
| 23:55.24 | alrs | Analog hotel phones aren't cheap, either |
| 23:55.31 | ManxPower | dFence: An analog Asterisk system is much SIMPLIER than SIP |
| 23:55.38 | Qwell | it's like buying a rental car |
| 23:55.53 | dFence | ManxPower: equipment's insured and btw the phones won't be that expensive ;D |
| 23:55.54 | drmessano | A SIP phone of quality to even have a chance in a hotel, it going to be $400 |
| 23:55.55 | alrs | There's a single-line Aastra that is pretty simple |
| 23:55.59 | drmessano | is* |
| 23:56.16 | alrs | and the Polycom 430 is properly features-poor |
| 23:56.25 | drmessano | alrs: So now we're gonna use a CHEAP SIP phone? |
| 23:56.28 | ManxPower | dFence: If you are planning on using Grandstream phones you might as well just sign off now before the ridicule starts. |
| 23:56.41 | drmessano | Oh thats too good.. Lets really blow a couple hundred dollars |
| 23:56.49 | dFence | ManxPower: was thinking of linksys spa922 |
| 23:56.59 | alrs | I see decade-old Aastra-manufactured Nortel phones all over the place |
| 23:57.00 | Qwell | drmessano: couple hundred for the entire install :p |
| 23:57.01 | drmessano | SPA922 wont last in a hotel |
| 23:57.17 | drmessano | I have an SPA-921.. its nice.. but not for abuse |
| 23:57.19 | alrs | http://www.voipsupply.com/product_info.php?products_id=2870 |
| 23:57.30 | Qwell | military ruggedized Cisco 7960 |
| 23:57.31 | alrs | er, not that one |
| 23:57.34 | Qwell | that'd last in a hotel |
| 23:57.36 | drmessano | Wow |
| 23:57.49 | drmessano | That $50 Aastra phone looks like it would last almost a week |
| 23:57.57 | alrs | http://www.voipsupply.com/product_info.php?products_id=1009&searchid=698965 |
| 23:57.58 | alrs | that one |
| 23:58.02 | alrs | drmessano: you're high |
| 23:58.15 | drmessano | alrs: Have you ever actually BEEN in a hotel? |
| 23:58.16 | dFence | drmessano: dunno what kind of hotels you're dealing with but our guests are 50+, dunno shit about computers and/or both. btw the rooms are cleaned on a daily basis and a missing phone would DEFINITELY strike the personnel as odd... |
| 23:58.16 | alrs | drmessano: that same phone branded "NORTEL" has been in use all over the place forever |
| 23:58.37 | drmessano | I have worked on Hotel phones before |
| 23:58.57 | alrs | As have I. |
| 23:58.58 | drmessano | People abuse things they don't own.. as Qwell said.. "Rental Car" |
| 23:59.39 | alrs | I think a Poly 430 or Aastra 9116 would do just fine, especially if bolted to the table |
| 23:59.52 | alrs | those Aastras are tanks |
| 23:59.54 | drmessano | Fine business |