00:00.11 | drmessano | Thats not built into Eyebeam.. Maybe MyspaceIM has it |
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00:01.52 | *** part/#asterisk [jmc] (n=John@93-45-211-136.ip104.fastwebnet.it) |
00:03.11 | Katty | :> |
00:03.12 | Katty | has freshly laundered laundry |
00:06.06 | dandate2 | hmm freshly laundered money mmm |
00:07.17 | drmessano | watches the used car salesman at work |
00:08.07 | phix | Katty: :D |
00:10.49 | dymaxion | hi anyone here use Asterisk-GUI? the other room is v quiet. |
00:11.05 | dymaxion | Woudl you recommend not using a GUI if possible? |
00:11.10 | Katty | infobot: gui? |
00:11.10 | infobot | gui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli |
00:11.59 | phix | Real Programmers also live in their parents basement too |
00:12.50 | dymaxion | :-) indeed.. well this is for small 10 person office.. I'm happy playing around with config files , however other users who may end up administering may not be comfortable with this... |
00:13.32 | dymaxion | I was suprised though.. the vanilla install that I was delivered with my rowetel blackfin IP04, had a load of pre-configured asterisk files (eg. extensions.conf is full of loads of seemingly irrelevant stuff.. is this normal ? |
00:14.34 | dymaxion | demo contexts, Context 'numberplan-international, Context 'numberplan-iaxtel' created by 'pbx_config' etc.. is this because I installed the GUI? |
00:14.52 | dymaxion | i'm considering going back to command line, but not sure waht to delete in order to get a clean start |
00:16.10 | TJNII | ~cli |
00:16.11 | infobot | i heard cli is a Command Line Interface, the best form of interface around, of course Call-Level Interface, originally developed by SQL Access Group, but now known as Microsoft's ODBC standard. an x86 assembly instruction Common Language Infrastructure (See mono or .net) |
00:16.37 | dymaxion | the GUI as far as I can tell, dynamically generates the asterisk.conf files. I hope by installing the GUI i haven't broken everything... at the moment, SIP clients cannot register... any ideas why? |
00:17.06 | TJNII | That link in ~gui is no good anymore. Someone with bot rights should remove it. |
00:17.30 | dymaxion | http://pastebin.org/19153 |
00:17.46 | dymaxion | i get SIP/2.0 403 but can't figure out why |
00:21.52 | drmessano | ~gui |
00:21.53 | infobot | gui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, such as http://toastytech.com/guis/miscwin1xp.png. Of course Real Programmers use the command line interface. See cli |
00:22.07 | drmessano | Try that link, TJNII |
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00:31.08 | TJNII | I did. 4 Times. Got a parking site |
00:31.23 | TJNII | Whoops, Ignore that! |
00:31.32 | TJNII | Much better. |
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00:39.37 | *** mode/#asterisk [+o denon] by ChanServ |
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00:49.40 | *** mode/#asterisk [+o denon] by ChanServ |
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01:01.04 | Katty | mariah carey has such an awesome voice. it's a shame she wasted it. |
01:01.14 | darkdrgn2k3 | Hey guys, is there a way to ring multiple extensions an inbound route? |
01:01.30 | Katty | darkdrgn2k3: Dial(SIP/100&SIP/101) etc |
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01:01.38 | darkdrgn2k3 | Katty: thank you |
01:01.42 | Katty | darkdrgn2k3: mhmm |
01:05.18 | [TK]D-Fender | darkdrgn2k3: #freepbx <------ |
01:08.32 | darkdrgn2k3 | [TK]D-Fender: wasnt a freepbx question :-P pure asterix dialplan question |
01:08.44 | darkdrgn2k3 | but i should hang around there to;) |
01:09.09 | [TK]D-Fender | darkdrgn2k3: Considering you're using FreePBX it IS a GUI question |
01:09.31 | [TK]D-Fender | darkdrgn2k3: You are in dialplan territory and you have sold your soul to the lowest bidder. Now move along... |
01:10.07 | darkdrgn2k3 | [TK]D-Fender: not if im treding in the "custom" conf files :-P |
01:10.26 | darkdrgn2k3 | but i guess your right |
01:10.30 | darkdrgn2k3 | as always... |
01:10.30 | [TK]D-Fender | darkdrgn2k3: uhhh huh |
01:11.05 | darkdrgn2k3 | LOL im not gonna live any of this down till i reinstall from scratch am i? |
01:11.11 | darkdrgn2k3 | and probably even then :-S |
01:12.17 | [TK]D-Fender | darkdrgn2k3: You're like that guy who keeps trying to order Big Macs..... at BURGER KING. |
01:12.55 | darkdrgn2k3 | [TK]D-Fender: i once orderd a vegie backen burger.. you should of seen the looks i got :-D |
01:22.57 | dmz | hey y'all, just heard that web-meetme has a sql injection and remote script execution |
01:24.31 | [TK]D-Fender | \o/ |
01:25.00 | dmz | not sure the details, but they were sent to the developers a few hours ago |
01:26.06 | drmessano | People use that? |
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01:32.39 | Tarantulafudge | I'm building an asterisk box for the company I work for, and I was just wondering whether I should go with Core2Quad or XeonQuad |
01:33.06 | Tarantulafudge | I was thinking that the Core2Quad would probably be better for transcoding |
01:33.24 | Tarantulafudge | but I'm not sure, and I don't have any facts to back that up |
01:33.25 | [TK]D-Fender | Tarantulafudge: xeon doesn't say what core tech. |
01:34.06 | Tarantulafudge | its a comparible 64bit model |
01:34.33 | Tarantulafudge | got it |
01:35.56 | Tarantulafudge | either the Core2Quad Q8200 at 2.33Ghz (4mb) or the XeonQuad 2.4 x3200 (8mb) |
01:36.56 | Tarantulafudge | price difference is about $20 |
01:37.02 | Tarantulafudge | the Core2 being cheaper |
01:37.13 | Tarantulafudge | [TK]D-Fender: any idea? |
01:37.57 | [TK]D-Fender | Tarantulafudge: Idea : go read the core specs and reviews if you have trouble interpreting the core specs |
01:38.20 | [TK]D-Fender | Tarantulafudge: And I dunnooo... maybe BENCHMARKS |
01:38.40 | Tarantulafudge | I can't... |
01:39.12 | darkdrgn2k3 | Tarantulafudge: stupid question perhaps buy Why? |
01:39.20 | darkdrgn2k3 | actualy i have a feeling its a stupid answer :-P |
01:39.27 | Tarantulafudge | I don't have the hardware I have to order it |
01:39.42 | darkdrgn2k3 | Tarantulafudge: reaserch.. they post benchmarks |
01:39.44 | Tarantulafudge | I was just wondering which would be better for use with asterisk |
01:40.08 | [TK]D-Fender | www.tomshardware.com |
01:40.12 | Tarantulafudge | benchmarks don't tell me much about asterisk performance |
01:40.13 | darkdrgn2k3 | http://www.lmgtfy.com?q=xeon+benchmark+chart |
01:40.21 | [TK]D-Fender | Tarantulafudge: Reviews have BENCHMARKS. |
01:42.35 | darkdrgn2k3 | wow look at that "High End CPU's" |
01:42.42 | darkdrgn2k3 | Intell AND amd :-P |
01:43.20 | Katty | stretches |
01:43.23 | Katty | mmmmmmmmmmmrelaxed. |
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02:35.16 | psilikon | I'm trying to test out an asterisk setup. Can I just add an extension to extensions.conf that will playback audio? |
02:36.03 | TJNII | Yep. |
02:37.21 | psilikon | I have added extension, yet when I try them I get extension: '600' rejected because extension not found |
02:37.58 | TJNII | Did you reload the dialplan? |
02:38.36 | psilikon | Yes. When I do a 'dialplan show' I get everything from extensions.ael. Do I need to disable extensions.ael? |
02:38.49 | psilikon | Does it have priority over extensions.conf? |
02:40.50 | TJNII | * should use both, AFAIK. |
02:41.32 | [TK]D-Fender | psilikon: Whatever you added either isn't properly formatted, or in the right place |
02:41.40 | psilikon | I am trying to follow Asterisk: TFOT 2nd edition. yet I cannot get Echo() to work |
02:41.51 | TJNII | psilikon: Pastebin what you have. |
02:42.02 | TJNII | ~pb |
02:42.03 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
02:42.41 | psilikon | http://pastebin.com/m22fc0fb7 |
02:44.36 | TJNII | And what do you have context= set to in sip.conf |
02:45.03 | psilikon | Under general? |
02:45.32 | TJNII | If you don't have it set on a per-phone basis, yes. |
02:46.24 | psilikon | Under general: context=from-callcentric |
02:46.55 | TJNII | And do you have a seperate context= set on the phone you are using? |
02:47.08 | psilikon | yes |
02:47.09 | TJNII | Or, should I say, the entry for thwe phone you are using |
02:47.16 | TJNII | And what is that? |
02:47.24 | psilikon | context=to-callcentric |
02:48.09 | TJNII | So all calls go out to callcentric, correct? |
02:48.33 | psilikon | yes |
02:48.48 | TJNII | Now, you have programmed your test extensions in both [default] and [internal], but you have no included those contexts anywhere |
02:49.51 | TJNII | So those extensions don't work for you, basically, because * doesn't know it is supposed to use those extensions for that peer. |
02:49.56 | psilikon | Ok, you made me understand that I need those extensions under the right place in the extensions.conf. |
02:50.13 | psilikon | I am having trouble understanding the basics... |
02:50.13 | TJNII | Yep |
02:50.47 | psilikon | so it is sending everything from that peer (the ATA phone) to [to-callcentric] |
02:51.02 | TJNII | Yep |
02:52.04 | TJNII | Also, that patterm match looks like it will only match two digit numbers, which is probably not what you want. |
02:52.36 | psilikon | Ok that is cool. I am sitting here trying to follow this Asterisk book.. |
02:53.07 | psilikon | Probably not. How do I change it. |
02:53.15 | psilikon | What other test sounds can asterisk provide? |
02:54.03 | TJNII | Test sounds? Whatever you want, I guess. (Provided you have a sound file for it.) |
02:55.26 | psilikon | Really? So how do I link an mp3? |
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02:56.50 | TJNII | AFAIK, and with the exception of MOH, you need to recode it. |
02:57.41 | psilikon | To wav? |
02:57.41 | [TK]D-Fender | TJNII: No |
02:57.43 | TJNII | But, I'm sure if I'm wrong [TK]D-Fender will call me on it. He has a way of doing that. |
02:57.48 | TJNII | See, beat me to it! |
02:57.51 | [TK]D-Fender | TJNII: Too late :p |
02:58.32 | [TK]D-Fender | psilikon: PASTEBIN is your friend. |
02:58.34 | [TK]D-Fender | ~pb |
02:58.35 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
02:58.36 | [TK]D-Fender | ^^^^^^^^^ |
02:59.12 | [TK]D-Fender | psilikon: Would help to see your actual dialplan. Hopefully it isn't based of that phsycho-kludge sample from the tarball... |
02:59.18 | TJNII | [TK]D-Fender: His contexts are messed up. He pastebinned it earlier. |
02:59.48 | psilikon | so how do i make a 7 digit number go out through the sip provider? |
03:00.10 | [TK]D-Fender | psilikon: like you are already doing : exten => _XX,1,Dial(SIP/${EXTEN}@callcentric) |
03:00.26 | [TK]D-Fender | psilikon: Only using a pattern that will match the kind of # you want to dial out |
03:00.39 | [TK]D-Fender | psythat matches any 2 digits |
03:00.43 | TJNII | [TK]D-Fender: Also, I didn't see mp3 in a core show file formats, so I assumed playback wouldn't play mp3s. Is there another function, or am I missing mp3 support on my system. |
03:00.58 | [TK]D-Fender | TJNII: Asterisk-addons <- |
03:01.07 | TJNII | nods |
03:02.07 | psilikon | This doesn't make sense to me yet: exten => _XX,1,Dial(SIP/${EXTEN}@callcentric What do I need to read so I can start make sense of the logic? |
03:03.58 | darkdrgn2k3 | psilikon: you only want 2 digit phoen nubmers? |
03:04.03 | TJNII | X Matches any single digit from 0 to 9. So XX matches 00-99. |
03:04.27 | darkdrgn2k3 | psilikon: i think you want I think you want NXXXXXX |
03:04.30 | darkdrgn2k3 | oops |
03:04.49 | darkdrgn2k3 | N=2-9 X=0-9 :) |
03:04.49 | TJNII | psilikon: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html/asterisk-book.html#asterisk-CHP-5-SECT-3.6 |
03:04.58 | darkdrgn2k3 | or that |
03:06.23 | [TK]D-Fender | psilikon: I can also tell you that there is no way callcentrix is going to accept you dialing a 7 digit number |
03:06.43 | [TK]D-Fender | psilikon: Now you may consider a 7 digit number local to you because of an assumed area code |
03:07.16 | [TK]D-Fender | psilikon: at which point if you want to dial that # you'll want to pass callcentric the areacodealong with |
03:07.26 | [TK]D-Fender | psilikon: At which point I may as well hand you the answer : |
03:07.53 | [TK]D-Fender | psilikon: exten => _NXXXXXX,1,Dial(SIP/555${EXTEN}@callcentric) |
03:08.00 | TJNII | One of the handiest (imho) things I do with * is automatically add the area code to 7 digit numbers so they are past to my ITSP as 10 digits. |
03:08.04 | [TK]D-Fender | psilikon: replacing 555 with your preferred default area code |
03:08.21 | drmessano | TJNII: Fuckin enabler! |
03:08.21 | TJNII | I get so annoyed with telcos that make me do 10 digit dialing for #s in the same area code.... |
03:08.34 | drmessano | hisses at TJNII |
03:08.45 | drmessano | 10 digits 4EVA |
03:08.55 | TJNII | splashes Holy Water on drmessano |
03:09.00 | drmessano | lol |
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03:09.08 | drmessano | Noooooooo |
03:09.27 | p3nguin | psilikon: Are you trying to match outgoing CallCentric numbers? |
03:09.33 | drmessano | bursts into flames and smashes blindly through the door into #trixbox |
03:09.35 | [TK]D-Fender | psilikon: Another quick lesson : ${EXTEN} hold the number that patched the first part of that EXTEN => line., As you can see by the action we are taking (appllication : Dial()) we are passing that # as part of the tech we are dialing and quite literally shoving digits in front |
03:10.31 | [TK]D-Fender | psilikon: *'s use of variables is PRECISELY as dumb as it looks. it is plain text substitution. |
03:25.07 | psilikon | I have a hard copy of Asterisk: The Future of Telephony 2nd Edition however I guess I am getting confused when it comes to the fact that I am using a SIP provider |
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03:32.26 | [TK]D-Fender | psilikon: the fact they are an ITSP isn't really the issue. its a question of knowing what numbers the tech you are calling is expecting. Many ITSP's expect you to even dial the 1 in front |
03:33.14 | psilikon | Yeah, I think callcentrics expects a 1 |
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03:33.37 | [TK]D-Fender | psilikon: Where I am, 10 digit dialing is madatory. This started maybe 4-5 years ago, and to accomodate old habits, I too allow 7-digit dialing and prefix it myself for what would be a relatively safe assumption of the primary area code here |
03:36.01 | psilikon | that is what I would like to do. |
03:36.38 | psilikon | So you guys are telling me that even though I am using an ITSP I can still get everything I need to know out of *: TFOT? |
03:40.02 | psilikon | I am trying to place a test call from my ITSP but I am getting the extension not found message. |
03:44.13 | [TK]D-Fender | psilikon: an ITSP is just one more thing * can talk to |
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03:44.48 | [TK]D-Fender | psilikon: the book teaches to basics of what different modules can interacto with. The dialplan is the core of everything. It is 95% of * and something you must master |
03:44.51 | drmessano | psilikon: What would an ITSP have to do with anything? Asterisk can speak SIP, IAX, Skype, DAHDI, Skinny, MGCP, OH323, Jingle, Bluetooth, etc |
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03:45.33 | psilikon | I hear ya. It is just that right now the dialplan is making little sense. |
03:45.47 | drmessano | That has nothing to do with using an ITSP |
03:45.53 | psilikon | good |
03:47.09 | drmessano | The dialplan could care less about what technology you're using, as a general rule.. There are protocol specific aspects, but "DO THIS, DO THAT" is protocol agnostic |
03:48.05 | psilikon | I really appreciate the help/lessons from you guys. I'm gonna get this stuff. Gotta o to bed for now tho. |
03:48.08 | drmessano | Wait til you start writing dialplan to patch around glitchy phones, asterisk bugs, or a shitty analog card |
03:48.17 | [TK]D-Fender | psilikon: If you configure the SIP parameters with an ITSP, you typcially receive calls from them. these calls enter the dialplan and are processed however you tell it to |
03:48.42 | drmessano | [TK]D-Fender: Unless your name is "S" |
03:48.49 | drmessano | Then, youre screwed |
03:48.56 | [TK]D-Fender | psilikon: If you call a number from a SIP phone (sort, hard, or whatever), it also enters the dialplan. Same with a Zaptel FXO (line) interface, FXS,e tc |
03:49.22 | [TK]D-Fender | psilikon: * processes calls. Setting up the device itself is petty. what matters is what you do with calls. that is the dialplan |
03:49.33 | drmessano | Asterisk: The VoIP wirenut |
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03:54.15 | joobie | hey guys.. is tehre any published dialplan i can use to figure out which numbers are dialling where? currently if i get a number dialed that's not in my list of known numbers, i then manully research to find out what regex would suit the number and where it goes |
03:54.18 | joobie | getting to be a PITA |
03:58.23 | [TK]D-Fender | joobie: ... huh? |
04:00.13 | joobie | TK, when users dial a number it goes through to the SIP provider and bills us at whatever.. my billing system then parses through the CDR and classifies the number to a country (using regex on the dialed number) and then bills accordingly.. at the moment each month i have like 5 new numbers taht my regex doesnt handle, so i have to manually google to try figure out a regex for it.. just wondering if there's a good source to get all the number combina |
04:00.13 | joobie | tions and where they dial to |
04:00.49 | [TK]D-Fender | joobie: this is an abstract non-* programming thing... |
04:01.30 | [TK]D-Fender | joobie: If you hit an unknown pattern how would you know how to brak down a number? |
04:03.35 | joobie | if it's unknown, it barfs at the moment |
04:03.45 | joobie | then i manually investigate, put a regex in place and it moves along |
04:04.16 | joobie | so each month, i build more and more regex into my billing.. but i'd rather than get a huge list of all countries and do all the regex combinations in one hit |
04:14.52 | korolev | is this for countries? |
04:15.05 | korolev | or for any dialing pattern? |
04:16.47 | korolev | joobie for made up dialing patterns or even local calls that would overlap with international country codes you would need separate tables |
04:17.25 | korolev | if its only international, make a db table to store all country codes, you can get them from here: http://voipjet.com/ratescsv.php |
04:18.03 | korolev | and then use Longest Common String instead of regexes to figure out the country code out of the dialed string |
04:18.14 | joobie | it's for multiple countries |
04:18.45 | korolev | grab the csv from voipjet, thats pretty accurate except for US area codes |
04:18.56 | korolev | but it contains all other nanpa codes |
04:19.18 | joobie | cool exactly what i was after |
04:19.18 | korolev | and all international, broken by mobile and fixed |
04:19.19 | joobie | thanks man |
04:19.22 | korolev | np |
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04:34.28 | drmessano | Yay, unbanned |
04:35.59 | [TK]D-Fender | ? |
04:38.23 | drmessano | Nah, not really |
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05:02.08 | VarnishedOtter | Hi, does anyone here have experience with app_konference? I have it working but it has extremely high latency (3-5 secs) |
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06:19.23 | alexhq | How to configure asterisk to use radius for authentication? |
06:24.43 | KyleK | huh les.net is adding a monthly fee for the pleasure of having an account with them |
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06:32.00 | korolev | with the rates they have to anywhere i call, its really no pleasure |
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06:49.56 | Gugge | ss |
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06:54.13 | tm1985 | I have a analog line and uses dahdi what is then the best exten for dialing *21*? |
06:54.28 | Micc_ | I need a way to play periodic messages to people on hold. |
06:54.33 | Micc_ | In a queue. |
06:56.59 | fiddur | Micc_: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf |
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07:13.22 | maxagaz | is it better to use ogg or gsm to compress sound with asterisk ? |
07:13.50 | maxagaz | the problem with gsm is that there is no graphical application to read it on linux |
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07:34.59 | tzafrir_laptop | maxagaz, 'play' plays it |
07:36.50 | maxagaz | tzafrir_laptop, play is not a graphical player, it's okay for me, but not for my colleagues |
07:37.20 | tm1985 | I have a analog line and uses dahdi what is then the best exten for dialing *21*? |
07:37.48 | tzafrir_laptop | maxagaz, what do you use? |
07:38.05 | maxagaz | tzafrir_laptop, play |
07:38.14 | maxagaz | tzafrir_laptop, but i'd like an alternative |
07:38.20 | tzafrir_laptop | try 'WAV' . It is wav/gsm . Same size as gsm, but a wav container |
07:38.42 | tzafrir_laptop | so players are more likely to detect it |
07:39.01 | maxagaz | tzafrir_laptop, i'm also thinking about changing the compression format to ogg |
07:39.09 | tzafrir_laptop | tm1985, Dial(DAHDI/1/*21*); ? |
07:39.11 | maxagaz | tzafrir_laptop, wav? better than flac ? |
07:39.39 | tzafrir_laptop | maxagaz, that is like asking if vorbis is better than wav |
07:39.53 | tzafrir_laptop | wav and ogg are containers, not codecs |
07:40.39 | tzafrir_laptop | furthermore, Asterisk uses 8kHz, 16 bits per sample, mono |
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07:41.33 | tzafrir_laptop | flac is compressed, but not lossy, so here's a limit to what it can compress |
07:41.44 | tzafrir_laptop | This means that it doesn't loose information |
07:42.00 | tzafrir_laptop | But sometimes you don't really care about information |
07:42.41 | maxagaz | tzafrir_laptop, ok, then i think wav is a good solution |
07:43.07 | tzafrir_laptop | gsm, speex and others are not only lossy, but also intended to preserve human voice - "speech" |
07:43.19 | maxagaz | tzafrir_laptop, but i thought flac was designed to replace wav |
07:44.00 | tzafrir_laptop | I guess you can save some disk space with flac |
07:44.11 | tzafrir_laptop | ideally ~50% |
07:44.26 | tzafrir_laptop | OTOH, it takes more CPU time compressing |
07:44.44 | maxagaz | tzafrir_laptop, if gsm, speex and others are containers, how can they affect the quality of sound ? isn't it the codec that affect it ? |
07:44.59 | tzafrir_laptop | gsm and speex are codecs |
07:45.09 | tzafrir_laptop | and yess, they are lossy |
07:45.40 | tzafrir_laptop | gsm has a very noticable effect. speex: less so |
07:48.36 | maxagaz | tzafrir_laptop, so gsm is both a codec and a container ? |
07:48.49 | tzafrir_laptop | no. It's just a codec |
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07:49.19 | tzafrir_laptop | a .gsm file is a raw file - merely the content of a stream, with no headers |
07:49.22 | maxagaz | tzafrir_laptop, you mean that my .gsm sound files have no container ? |
07:52.31 | maxagaz | tzafrir_laptop, what is the interest of putting a raw file in a container if it can be played directly ? |
07:52.47 | tm1985 | when I try to call an the analog channel I got this output http://pastebin.com/d5e8b6e8a |
07:53.49 | tzafrir_laptop | What exactly is *21* ? What exactly is 0498506822# ? |
07:54.55 | tm1985 | *21* a number from our provider so that we can redirect call to a number If we do this the call will be redirect by the provider and not asterisk |
07:55.35 | tm1985 | the number a the number of certain phone and it has to end with a # because the context is *21*TELNR# |
08:01.14 | tm1985 | do you have any idea? |
08:02.02 | tzafrir_laptop | maybe you need to wait a second or so after the *21* ? |
08:02.23 | tzafrir_laptop | In that case, dial *21*ww0498506822# |
08:02.44 | tzafrir_laptop | The 'digit' 'w' means 'wait for half a second' |
08:03.57 | tm1985 | I just tried to call an normal number I does It what that also |
08:06.31 | tm1985 | http://pastebin.com/d7bf15899 this is the dahdichannel!! Is maybe something wrong or missing? |
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08:26.01 | tm1985 | http://pastebin.com/d7bf15899 this is the dahdichannel!! Is maybe something wrong or missing? |
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08:43.59 | maxagaz | tzafrir_laptop, thanks for your explanation, i understand a lot better now |
08:45.03 | tzafrir_laptop | tm1985, what does it do, exactly? |
08:45.23 | tm1985 | the *21*? |
08:46.33 | maxagaz | tzafrir_laptop, wav is a codec with no comrpession at all ? |
08:47.28 | maxagaz | maxagaz, sorry, stupid question... |
08:47.29 | tzafrir_laptop | wav is not the codec |
08:47.48 | tm1985 | When we dial this, the provider redirect to call to the telephone number you put into *21*TELNR# then the call doesn't comes to asterisk |
08:50.29 | maxagaz | tzafrir_laptop, what i don't understand is that when i use wav/gsm, the sound is smaller than when i use gsm only |
08:51.00 | maxagaz | tzafrir_laptop, but wav add a header, so it should make the sound bigger, isn't it ? |
08:51.19 | tzafrir_laptop | it shouldn't be smaller. It should be a few bytes larger |
08:52.43 | maxagaz | tzafrir_laptop, it does this when converting a gsm file into wav |
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09:00.12 | maxagaz | also, what's the difference betwee gsm and speex ? |
09:00.25 | maxagaz | why not using ogg/speex by default ? |
09:01.05 | maxagaz | (as it is all free software) |
09:01.14 | tm1985 | I have a analog line and uses dahdi what is then the best exten for dialing *21*? |
09:01.43 | troffasky | maxagaz, core show translation will tell you why :-) |
09:06.15 | maxagaz | troffasky, what does mean the table it displays ? |
09:07.29 | maxagaz | troffasky, it means that speex is much slower...? |
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09:08.54 | s14ck | how can I do to detect when somebodys pickup the call through fxsks device? |
09:11.23 | tzafrir_laptop | it means that it takes more CPU time |
09:11.58 | tzafrir_laptop | btw: the table as calculated at startup is no so useful. try: core show translation recalc 1000 |
09:12.08 | tzafrir_laptop | to get a better granularity |
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09:18.08 | mbrevda | anyone here with nvfax detect installed? |
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09:26.37 | maxagaz | tzafrir_laptop, i don't understand this table, if my sound is saved into speex, why should it be translated into gsm ? |
09:28.28 | troffasky | maxagaz, read the description at the top of the table |
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09:39.13 | L2Logic | anyone know if libpri will compile 64 bit? |
09:42.12 | troffasky | I assume so as it's installed on my 64 bit system from debian packages |
09:43.27 | L2Logic | ty troffasky |
09:45.01 | tm1985 | http://pastebin.com/d7bf15899 => This is my analog channel. Does anyone see what is wrong. Because I can't dail out???? |
09:47.16 | Chainsaw | L2Logic: Yes, both 1.2 and 1.4 will compile fine on X86_64 hardware. |
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10:16.12 | tzafrir_laptop | tm1985, have you tried dialing the same number with an analog phone? |
10:19.57 | tm1985 | yes and that worked |
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10:25.05 | trentster | hey all, whenever i put someone on hold and music on hold plays I get this error continiously logged in the CLI "RTCP SR transmission error, rtcp halted" this does not seem to effect the call or the music...any ideas what is causing this? |
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10:58.51 | eliasp | hi |
11:02.00 | eliasp | when a call is routed forward by the followme rules, it seems the followme message is played back at the same time when the call is forwarded, so the receiver hears just the last words of the followme message... is there a way to define the execution order (parallel/serial)... |
11:02.08 | eliasp | ? |
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11:16.01 | afink | Is there any way to do a manual sip peer poke? |
11:23.04 | trentster | hey all, whenever i put someone on hold and music on hold plays I get this error continiously logged in the CLI "RTCP SR transmission error, rtcp halted" this does not seem to effect the call or the music...any ideas what is causing this? |
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11:32.54 | TommyBotten | afink: You can use sipsak or something similar to construct your own SIP packets |
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11:49.19 | aiksa[LV] | Hello everyone |
11:51.11 | aiksa[LV] | I sometimes see a strange behaviour on 1.4.22 regarding the callerid(num). It seems that it gets overwritten by asterisk to the pattern of the extension the call is on. Anyone else have seen this? |
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11:58.48 | TommyBotten | aiksa[LV]: Could you please elaborate?. I didnt quite understand. |
12:02.28 | aiksa[LV] | TommyBotten: I have a dialplan where a call would cycle through a number of extensions |
12:02.45 | aiksa[LV] | the first priority would ring SIP/01, the second SIP/03 ... etc. |
12:03.50 | aiksa[LV] | now - for the sake of the ease of use they are not maped directly to a specific external extension, but have an "extension" pattern of "simpleQ" |
12:04.45 | aiksa[LV] | calls to specific external numbers are then sent to this through a Goto(simpleQ,1) |
12:04.53 | aiksa[LV] | is that clear so far? |
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12:07.07 | aiksa[LV] | under a circumstances I have not been able to track down so far sometimes a call arriving at one of the destinations SIP/01 or SIP/03, will not have the callerid received from PSTN, but "simpleQ" both as a name displayed by voip phone (zoiper and snom) and as a callerid key in AMI Newstate event |
12:08.14 | aiksa[LV] | what is the most strange thing - that untill now I have not been able to track what makes this difference |
12:09.33 | aiksa[LV] | I have seen a first call with a coorect callerid pass by, and then the next one with the 'simpleQ' instead. Both dialed to the same extension through the same PSTN PRI line. |
12:09.48 | aiksa[LV] | I have not seen this behaviour before, that is why I was asking. |
12:10.19 | aiksa[LV] | could that be related to that 'o' option for the Dial command, mentioned somwhere in the wikis |
12:10.22 | aiksa[LV] | ? |
12:13.28 | aiksa[LV] | TommyBotten: did this make it clearer |
12:13.30 | aiksa[LV] | ? |
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12:37.56 | Grof | hey guys |
12:37.58 | Grof | need help |
12:38.06 | Grof | "Unrecognized pridialplan NPI modifier: g" |
12:38.07 | Grof | ? |
12:38.20 | Grof | i just upgraded from asterisk 1.2 to 1.6 |
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12:40.02 | manxpower | ~answers |
12:40.03 | infobot | answers is probably Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
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12:45.10 | TommyBotten | aiksa[LV]: Probably... as o: Restore the Asterisk v1.0 CallerId behaviour (send the original caller's ID) in Asterisk v1.2 (default: send this extension's number) |
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12:56.00 | Hatrix | Hi guys, i have a tandberg video solution and i try to record the video in asterisk. tcpdump tells me all is fine, audio and video gets send from the tandberg to my asterisk, but the record command only records the audio? i read at voip-info and googled but all i could find is that record should save the audio and video file in a separate raw file (which i would need) but ... it doesn't, any ideas? |
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13:05.17 | Katty | yawns |
13:05.35 | Naikrovek | yawns |
13:05.40 | Naikrovek | it's contageous |
13:05.42 | Katty | pamples Naikrovek |
13:05.56 | Naikrovek | looks up "pamples." |
13:06.53 | coppice | I think she's trying to put a diaper on you |
13:08.46 | Katty | http://www.youtube.com/watch?v=lZ2nhg6az64 <- Pampling. |
13:10.12 | Katty | oh damn |
13:10.20 | Katty | my dad just sent me a friendship request |
13:10.21 | Katty | on facebook |
13:11.31 | creativx | Katty: bummer! |
13:11.42 | creativx | i ignored my moms friend request till she figured out how to resend it |
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13:13.19 | Katty | i'm not ignoring my dad. |
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13:39.32 | michel | hi all |
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13:40.27 | MWE | hi all |
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13:41.33 | Naikrovek | hello |
13:42.20 | MWE | I'm bussy with a project were a caller calls a script (duhhh) but the caller had to enter a meetme room and before that there must be an outbound with PIN check. Is het possible to make an outbound which also runs an special script while the caller is in the meetme room? |
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13:46.19 | [TK]D-Fender | MWE: You can Originate a call before sending him into thet room. This callout will be completely independant however |
13:47.11 | MWE | I tried a serveal things but I'm stuck @ it right now.. maybe somebody could help me out? http://pastebin.com/m4c17ead4 |
13:49.28 | MWE | just before the caller enters the meetme room the script had to make an outbound and the callee had to enter the room |
13:49.47 | MWE | I checked out http://www.voip-info.org/wiki/view/Asterisk+local+channels |
13:50.40 | Katty | facebook suggests i know allison smith |
13:51.40 | fuxu2 | facebook always suggests people to me that I don't know |
13:52.40 | fuxu2 | katty: www.facebook.com/iconicflux ... send me a friend request |
13:52.51 | Katty | but i don't know you |
13:52.57 | fuxu2 | you do now |
13:53.03 | Katty | do i? |
13:53.10 | Katty | i don't think i do. |
13:53.29 | [TK]D-Fender | MWE: that does not spawn a SEPARATE call. |
13:53.48 | [TK]D-Fender | MWE: that has nothing to do witha separate call processing in the background while letting the caller fall through to the MeetMe |
13:53.49 | fuxu2 | sure.. we're on #asterisk.. obviously that means we have at least that in common.. and facebook isn't linkedin. :) |
13:54.07 | MWE | but how can I fix that [TK]D-Fender ? |
13:54.08 | [TK]D-Fender | MWE: ORIGINAT <---- |
13:54.14 | [TK]D-Fender | MWE: ORIGINATE <---- |
13:54.16 | Katty | yeah i only have friends and family on facebook. |
13:54.22 | [TK]D-Fender | MWE: Go look it up on the WIKI |
13:54.24 | Katty | with the exception of fender. |
13:54.30 | [TK]D-Fender | :| |
13:54.30 | fuxu2 | if anyone here plays mafia wars on facebook.. you really should add me because I've got a pretty strong crew |
13:54.34 | Katty | i have to keep my enemies close. |
13:54.40 | Katty | i mean. |
13:54.42 | Katty | uhh |
13:54.45 | [TK]D-Fender | Katty: I only have people I don't see IRL there :) |
13:54.46 | Katty | hugs [TK]D-Fender |
13:54.48 | Katty | ;) |
13:55.11 | [TK]D-Fender | 's facebook page is EMPTY and will remain as such. |
13:55.12 | Katty | [TK]D-Fender: my dad sent me a facebook request this morning. |
13:55.16 | [TK]D-Fender | Facebook = EVIL |
13:55.22 | [TK]D-Fender | Katty: IGNORE <- |
13:55.28 | Katty | no. |
13:55.30 | [TK]D-Fender | Katty: ESPEciALLY family |
13:55.31 | Katty | i shan't |
13:55.38 | Naikrovek | facebook is not a font of morality, but i wouldn't say it's evil |
13:56.10 | [TK]D-Fender | Katty: BS... I already know how Bible-thumping-crazy your lot are, and if you value your privacy you'll deny him... |
13:56.37 | Naikrovek | someone I didn't know added me as a friend over the weekend, and i removed her after I saw her profile and didn't know who she was. I wonder if she lurks in here... |
13:56.44 | [TK]D-Fender | Naikrovek: You're right, it isn't inherently evil, that is more like the inevitable conclusion due to mankinds own failings. |
13:57.14 | fuxu2 | facebook for me isn't really about people I actually know. If I wanted to keep in touch with them I'd use IM or email. |
13:57.36 | fuxu2 | so basically.. I use facebook for groups, games, etc.. and try not to put much about me or my family on it |
13:57.56 | [TK]D-Fender | Facebook is a way for someone you used to know who can't find you any other way to do so, but never put anything up there. |
13:58.24 | [TK]D-Fender | fuxu2: Far better game resources out there including *gasp* installable softwaqre |
13:58.46 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
13:58.52 | Katty | [TK]D-Fender: mom's crazy. |
13:59.00 | Katty | [TK]D-Fender: dad's moderately crazy, but respectful |
13:59.14 | [TK]D-Fender | Katty: And Mom will go THROUGH your Dad if she's that crazy |
13:59.31 | Katty | pats [TK]D-Fender |
13:59.33 | fuxu2 | fender: true.. but I there's not an easy way to brag about how you're doing via installable software.. :) |
13:59.35 | Katty | i appreciate your concern, dear. |
13:59.36 | [TK]D-Fender | Katty: Katty You've been warned :| |
14:00.09 | [TK]D-Fender | fuxu2: Twitter had it right at the 4th character... |
14:00.53 | *** join/#asterisk troffasky (n=r00t@92-234-126-57.cable.ubr08.gate.blueyonder.co.uk) |
14:02.00 | fuxu2 | wow.. people are still using twitter? :-P |
14:02.02 | Katty | considers breakfast. |
14:02.07 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
14:02.19 | creativx | im gonna start twatter |
14:02.49 | [TK]D-Fender | creativx: Wouldn't surprise me if a porn site already reserved that one... |
14:03.01 | creativx | hehe, no doubt |
14:03.24 | *** join/#asterisk anonymouz666 (n=anonymou@187-28-37-118.poolip.RJO.embratel.net.br) |
14:03.38 | fuxu2 | i really want to make a game on facebook that's all about warring religions.. |
14:04.02 | fuxu2 | like catholics vs protestants vs muslims vs jews.. etc.. give them all funny skills and stuff.. |
14:04.39 | fuxu2 | scientologists also.. they'd totally bring out their e-meters and f. people up mentally.. |
14:06.00 | creativx | that sounds meaningful and profitable.. |
14:06.02 | [TK]D-Fender | fuxu2: .... that would actually likely sell.... |
14:06.28 | [TK]D-Fender | fexEngendering the very worst traits of humanity... |
14:06.33 | [TK]D-Fender | fuxu2: Engendering the very worst traits of humanity... |
14:06.49 | troffasky | that's facebook in a nutshell is it not? |
14:07.08 | [TK]D-Fender | troffasky: No, once again, only an inevitable conclusion. |
14:07.22 | fuxu2 | fender: yeah .. on facebook you could totally have each side's effectiveness be swayed by the number of converts to their side.. |
14:07.43 | fuxu2 | and let people buy more invites.. |
14:07.52 | *** join/#asterisk Faithful (n=Faithful@124.217.119.152) |
14:07.57 | coppice | I'm puzzled why they call this stuff social networking |
14:08.08 | creativx | lack of better word |
14:08.09 | Naikrovek | that game would end fast. scientologists would put out a call to arms, and EVERY scientologist would join the game and win. |
14:08.23 | [TK]D-Fender | coppice: Because its as close as many of them get to actually interacting with people... |
14:08.31 | coppice | an obvious better term would be antisocial networking |
14:08.32 | fuxu2 | naikrovek: who cares as long as I'm rich off their dimes.. |
14:08.33 | fuxu2 | :) |
14:08.34 | troffasky | Naikrovek, nah, that's just what they want you to think would happen |
14:08.55 | Naikrovek | if there's one thing religious folks like above all else, it's defending their own religion, and eradicating all other trains of thought. |
14:09.12 | fuxu2 | naikrovek: that's my thinking about the whole game and why I want it |
14:09.19 | fuxu2 | I even started working on it at one point.. |
14:09.43 | fuxu2 | need some graphics artists and stuff like that though.. |
14:10.03 | troffasky | don't forget vi and emacs as religions |
14:10.14 | fuxu2 | ahahahaha! troffasky! ahahaha that's awesome |
14:10.19 | Naikrovek | if some of these religions had their way we'd attend church every morning and night, and you could only eat at church, could only have children when the church said so, all taxes woudl go to the church, if you were lucky enough to be employed at a church, etc. |
14:10.36 | *** join/#asterisk wcselby (n=wcselby@216.110.88.254) |
14:11.09 | Katty | :> |
14:11.12 | Katty | hi wcselby! |
14:12.09 | fuxu2 | Anyways.. if anyone here wants to actually work on it with me, let me know. |
14:12.38 | p3nguin | What are we working on? |
14:12.50 | fuxu2 | p3nguin: Holy Wars the game. |
14:13.09 | p3nguin | development? |
14:13.48 | fuxu2 | yeah.. a lot of the development I can do but I don't have any graphics skills and it always helps to have other people that want to help on the game design and balance.. |
14:14.24 | [TK]D-Fender | draws the bestest stick-people EVAR |
14:15.01 | *** join/#asterisk mumtazah (n=mumtazah@152.78.48.60.wmu01-home.tm.net.my) |
14:15.15 | wcselby | o/ Katty |
14:18.37 | obruT | Hello everyone, I got question regarding getting value of the variable after hangup, is it possible ? in the dialplan, after creating a channel, I'm calling one application that creates some variables, after hangup, I can display them with Noop() application, but at the end I would like to execute one AGI application that needs to fetch those variables, is it possible ? I would like to avoid passing the values via agi script cmd line options.... |
14:19.19 | MWE | [TK]D-Fender, do you have a example of that originate.. I read the wiki but is it something like the hangup() just a function call? |
14:19.43 | p3nguin | obrut: I don't see why that can't be done. What's the problem? |
14:20.43 | p3nguin | mwe: You just need to see how originate is used? |
14:20.47 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
14:20.47 | *** mode/#asterisk [+o putnopvut] by ChanServ |
14:20.51 | MWE | yeah p3nguin |
14:20.59 | manxpower | MWE: You didn't find anything useful in the OFFICIAL docs (located in /path/to/src/asterisk/doc)? |
14:21.03 | p3nguin | example originate SIP/101 extension 101@wakeup |
14:21.05 | obruT | p3nguin: getting the value from the agi script after hangup returns error, like the variable is not set |
14:21.25 | manxpower | obruT: which specific variable? |
14:22.14 | obruT | manxpower: for example any variable that sendfax application sets |
14:22.14 | [TK]D-Fender | MWE: AMI ORIGINATE. its in the book, its in the WIKI. Lookup "call files" while you're at it |
14:22.14 | MWE | manxpower, I've got a book "The future of telephony' but I didn't know what there was |
14:22.44 | manxpower | MWE: the docs included with Asterisk are always the current ones |
14:23.03 | obruT | manxpower: or any variable that i set in the extensions.conf, even if I set it after hangup |
14:23.04 | *** join/#asterisk wr| (n=wr@p54BE4186.dip.t-dialin.net) |
14:23.35 | [TK]D-Fender | obruT: DeadAGI, not AGI |
14:23.35 | manxpower | you should not set dialplan variables after hangup. |
14:23.58 | obruT | [TK]D-Fender: DeadAGI is obsolete in 1.6 |
14:24.47 | p3nguin | I have a peer that I currently use qualify= for. His lag often increases above the threshold. Should I consider not using qualify= at all, or increate the value even higher? |
14:24.48 | obruT | manxpower: well, I can't get value of the variables that are set before hangup |
14:25.08 | manxpower | obruT: pastebin an example |
14:26.28 | manxpower | p3nguin: consider not uding qualify |
14:26.38 | Katty | hugs p3nguin |
14:26.40 | Katty | hugs manxpower |
14:26.45 | p3nguin | katty: Mornin' |
14:26.55 | Katty | how's everything upstate today |
14:27.06 | p3nguin | katty: Doing well so far. |
14:27.30 | Katty | excellent. |
14:27.34 | *** join/#asterisk propellerhead (n=yogurt2u@host238.190-136-114.telecom.net.ar) |
14:29.36 | obruT | manxpower: http://free-ka.t-com.hr/ib/stuff/p/asterisk.txt this is section of extensions.conf, first it calls agi script, this agi script successfully fetches variables, after hangup it displays all of the variables, but on the second agi call, all of those variables are unset |
14:29.39 | *** join/#asterisk karleeto (n=karl@server.nashvilleproweb.com) |
14:29.53 | karleeto | for the hole in my router for an iax trunk, is it tcp or udp? |
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14:30.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
14:30.16 | wcselby | karleeto - udp |
14:31.01 | p3nguin | I'm suddenly reminded of a jingle for a roofing company. |
14:31.22 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
14:31.44 | p3nguin | "For a hole in your roof, or a whole new roof - Fredrick Roofing." |
14:31.48 | manxpower | obruT: why do you need to call it in the HANGUP? |
14:32.01 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
14:32.47 | Katty | plots breakfast |
14:33.05 | obruT | manxpower: when the channel hangups, execution of the first agi script stops immediately, sometimes before I update some status, now I'm trying to do it after hangup |
14:33.07 | p3nguin | katty: Cocoa Puffs? |
14:34.04 | obruT | manxpower: also I have to pass some variables to the status, I could do it via cmd line, but I would like one potential security hole :) |
14:34.13 | Katty | i have chex, eggs in the fridge... |
14:34.16 | Katty | leftover stroganoff |
14:34.21 | Katty | possibly other stuff. |
14:36.21 | manxpower | obruT: what status are you trying to update? It is possible to make your script exit when the channel hangs up |
14:36.26 | p3nguin | Stroganoff for breakfast? I'll pass. |
14:36.53 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
14:37.16 | Superbartt | Quick question: Is there a way to run a script/command when a caller leaves the queue and gets connected to the agent? |
14:37.21 | coppice | Strogan Off sounds like a dreadful insult |
14:38.13 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
14:39.03 | [TK]D-Fender | Superbartt: Run a daemon polling the AMI messages |
14:39.47 | Superbartt | hmm ok |
14:39.48 | Katty | mmm |
14:39.53 | Katty | stroganoff for breakfast is awesome. |
14:39.55 | Katty | chows. |
14:40.28 | obruT | manxpower: some status in my database, it depends of the value of the variables that SendFax application sets, asterisk just stops execution of the agi script after hangup, I have no controll when and where in execution will it happen... If the other side waits for me that i make hangup, then everything finishes ok, but if the other side hangups before my agi script finishes, it can be anytime, I just cannot make clean exit... |
14:42.42 | *** join/#asterisk fskrotzki (n=fskrotzk@mail.perspectivepartners.com) |
14:42.56 | Superbartt | [TK]D-Fender does it also return events like: $from-CID was in the queue and connets to agent X |
14:43.00 | manxpower | obruT: I wish you the best of luck. |
14:43.04 | Katty | i watched a film about corn last night. it was pretty sad. |
14:43.12 | [TK]D-Fender | subrGo read the manual on this |
14:43.15 | Katty | well the flim wasn't. |
14:43.18 | [TK]D-Fender | Superbartt: Go read the manual on this |
14:43.21 | Katty | the corn production is. |
14:43.56 | Superbartt | sorry [TK]D-Fender, there's an company busting my balls and they want to know if i can make it in 15 minutes ;P |
14:44.19 | [TK]D-Fender | Superbartt: Go invest in Tylenol |
14:44.25 | Katty | if it's that important, i'd suggest hiring a consultant. |
14:44.26 | Superbartt | say what? :P |
14:44.49 | [TK]D-Fender | Katty: Sounds liek he is one... and underqualified |
14:44.57 | Katty | [TK]D-Fender: hush, dear. |
14:45.01 | Superbartt | Katty I am the "consultant" :p |
14:45.02 | Katty | [TK]D-Fender: there's no need for that. |
14:45.10 | Superbartt | and true [TK]D-Fender ;) |
14:45.14 | [TK]D-Fender | Katty: I stand validated :p |
14:45.16 | Superbartt | it's the truth :p but I can do it |
14:45.27 | Katty | that's the spirit!! (= |
14:45.35 | [TK]D-Fender | Superbartt: then get your hands off your nuts and SEIZE THE DAY! |
14:45.45 | Superbartt | lol :p |
14:45.52 | Superbartt | if you just say yes i believe you :P |
14:46.26 | *** join/#asterisk Faithful (n=Faithful@124.217.119.151) |
14:48.23 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
14:50.23 | Naikrovek | poor [TK]D-Fender - knows more than all of us and is unemployed, while we all have jobs (at least I think he said he was unemployed.) |
14:50.56 | troffasky | that's cos only unemployed people have time to RTFM ;-) |
14:51.19 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:1c5a:126a:b2c9:8871) |
14:51.31 | cusco | hi |
14:52.09 | cusco | iff a application in make menuselect says "Depends on xxx (M) or (E)" |
14:52.15 | cusco | what is (M) and (E) |
14:52.28 | p3nguin | Monkeys and Elephants |
14:52.33 | cusco | I would say (E) means external? |
14:52.46 | cusco | ... |
14:53.17 | p3nguin | Okay, maybe not. |
14:53.19 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
14:53.33 | cusco | and where canI find app res_jabber |
14:55.03 | Katty | hmm. maybe stroganoff for breakfast wasn't such a great idea afterall. feeling kinda naseous now :/ |
14:55.13 | p3nguin | I tried to warn you. |
14:55.22 | wcselby | Katty - lol |
14:55.26 | Katty | yes, yes you did |
14:57.04 | coppice | Kellogg's Stroganoff probably isn't a good idea, then |
14:57.12 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
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15:00.12 | [TK]D-Fender | Naikrovek: Nope, rock solid day-job... |
15:00.23 | Naikrovek | [TK]D-Fender: oh, cool. |
15:00.31 | Naikrovek | wonders where he got that idea then. |
15:00.39 | [TK]D-Fender | Naikrovek: And I believe that anyone who depends on * as their livelyhood had better RTFM... |
15:00.46 | Naikrovek | yeah |
15:00.56 | Naikrovek | your job an * job? |
15:01.00 | [TK]D-Fender | Naikrovek: Probably manxpower's on/off status |
15:01.13 | [TK]D-Fender | Naikrovek: General IT including running our * box. |
15:01.24 | Naikrovek | cool. same here. |
15:01.26 | [TK]D-Fender | Naikrovek: I was using it long before and consult on the side |
15:02.12 | [TK]D-Fender | Naikrovek: I burned a week with #$&^ing Dell PCIe interrupt losses and have lost some face on out last upgrade attempt |
15:02.28 | [TK]D-Fender | Naikrovek: and lost 2 weekends. I'm completely fried, in a Kentuky kinda way |
15:02.37 | Naikrovek | take a day off |
15:02.48 | [TK]D-Fender | Naikrovek: I'm trying to take my normal WEEKENDS off |
15:03.31 | wcselby | [TK]D-Fender - ouch |
15:03.31 | cusco | change hardware |
15:03.52 | [TK]D-Fender | I don't even have the energy to kill people that really deserve it. |
15:04.21 | wcselby | [TK]D-Fender - do you have a PFY running around with you somewhere? You sound like you could be a real BoFH |
15:04.29 | wcselby | and I mean that in the best possible way |
15:04.33 | wcselby | :-) |
15:04.39 | [TK]D-Fender | wcselby: PFY? |
15:05.46 | wcselby | [TK]D-Fender - have you read any of Simon Travaglia's BoFH series? http://www.theregister.co.uk/odds/bofh/ |
15:05.53 | [TK]D-Fender | wcselby: And I am anti-BOFH. I do not subscribe to the closet-troll power-tripping mentality. Technology is there to serve people, not the other way around. |
15:06.00 | [TK]D-Fender | wcselby: Nope |
15:06.38 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
15:06.47 | wcselby | i enjoy it as good comedy, however I don't act that way towards clients / users |
15:06.56 | Naikrovek | too many do |
15:07.04 | Naikrovek | fortunately they don't last long |
15:07.06 | [TK]D-Fender | wcselby: And I make solutions that work, cheaper faster and "freer" |
15:07.23 | Naikrovek | then they complain on facebook "I did the work of 10 men" and BS like that |
15:07.52 | wcselby | Naikrovek - lol yeah too true. I've seen that mentality a lot...unfortunately |
15:08.33 | wcselby | [TK]D-Fender - so is your upgrade complete or are you still dealing with hardware issues? |
15:08.47 | Naikrovek | i ALWAYS reply to those guys when I can: "yes, all smart businessmen fire the one man who can do the work of ten men. right? perhaps it's more likely that you are an ass." |
15:08.49 | *** join/#asterisk Faithful (n=Faithful@124.217.119.131) |
15:09.13 | wcselby | Naikrovek - lol |
15:09.20 | [TK]D-Fender | wcselby: Plagued. It was a hardware/software full upgrade and I still had the old server. Back to square one operationally speaking.. |
15:09.30 | Naikrovek | eek |
15:09.46 | [TK]D-Fender | wcselby: And I'm going to bludgeon the McFuck out of the hardware vendor until they can make their products cooperate |
15:09.46 | wcselby | [TK]D-Fender - UGH, I hate that. At least you were able to roll back to an operational status. |
15:09.50 | p3nguin | Is there a variable that essentially says "all available SIP channels"? A usage of it might be to ring all SIP phones by the variable as opposed to using Dial(SIP/1001&SIP/1002&SIP/1003&SIP/you-get-the-idea) |
15:09.59 | Naikrovek | lol McFuck awesome |
15:10.07 | [TK]D-Fender | p3nguin: No. |
15:10.08 | wcselby | [TK]D-Fender - lol |
15:10.29 | wcselby | p3nguin - I guess you could define one in the [globals] context of your extensions.conf file though |
15:10.31 | Naikrovek | p3nguin: assign people to groups, then assign all groups into an ALL alias |
15:10.47 | wcselby | p3nguin - would be a pain in the ass to maintain though |
15:10.48 | p3nguin | wcselby: That was precisely my reason for asking. |
15:10.53 | Naikrovek | yeah it would be a pain |
15:11.08 | p3nguin | I figured if something already existed, there was no reason to make a new var for it. |
15:11.19 | [TK]D-Fender | Maintaining that constant would also be a pain, but if its usageoccurs in multiple places, at least LESS of a pain. |
15:11.42 | [TK]D-Fender | p3nguin: Feel free to write an external script to do this... |
15:12.02 | *** part/#asterisk manxpower (n=EWieling@133.sub-70-214-74.myvzw.com) |
15:12.11 | [TK]D-Fender | p3nguin: Shouldn't be too hard |
15:12.12 | Naikrovek | i do it like i used to do mailing lists on sendmail. 1st floor was tech, producers, tech mgmt, producer mgmt. 2nd floor was execs, sales, marketing. all was 1st & 2nd |
15:15.46 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
15:16.17 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
15:16.28 | p3nguin | naikrovek: When you said assign people to groups, did you have in mind to use callgroup for each peer in sip.conf? |
15:17.25 | p3nguin | "group" turns out to be a rather ambiguous term. |
15:18.43 | Naikrovek | well i just assign people to logical groups. so mgmt has like 4 people in it, they each have their own name -> extension mapping as well |
15:19.29 | p3nguin | I assume you didn't mean callgroup, now that I read what it does/doesn't do. "Callgroups are not intended to call a group of phones" |
15:19.57 | Naikrovek | ah, no |
15:20.19 | p3nguin | In which file do you contstruct said "group"? |
15:20.58 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
15:21.09 | p3nguin | Seems like it would be easiest by using a variable in extension.conf's globals. |
15:21.20 | Naikrovek | p3nguin: yes i use globals |
15:21.37 | Naikrovek | but i have a freepbx system so i had to hide them away somewhere |
15:21.48 | [TK]D-Fender | p3nguin: there is no grouping or function for what you are looking to do. |
15:22.32 | [TK]D-Fender | p3nguin: Any "select all' that you want for dials, page, etc that you want to do will require scripting on your part. |
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15:32.18 | Katty | ARGH |
15:32.22 | Katty | my nose itches |
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15:35.32 | wcselby | cd /usr/src/asterisk-1.4.26/doc |
15:35.35 | wcselby | bleh |
15:35.39 | wcselby | wrong window |
15:35.43 | Katty | i was gonna say |
15:38.33 | *** join/#asterisk Carlos_Tico (n=grillo_v@c-98-201-162-34.hsd1.tx.comcast.net) |
15:38.38 | Carlos_Tico | hello... |
15:39.13 | Carlos_Tico | are you there [TK]D-Fender |
15:39.28 | wcselby | ~ask |
15:39.29 | infobot | 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. |
15:39.58 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
15:40.00 | Carlos_Tico | got a problem with DISA |
15:40.17 | jaytee | cd /home/dickcheney/chickenporn |
15:40.22 | jaytee | ooops! |
15:40.26 | jaytee | cd .. |
15:40.39 | [TK]D-Fender | carGo ahead and show it |
15:40.45 | [TK]D-Fender | Carlos_Tico: Go ahead and show it |
15:41.26 | [TK]D-Fender | Carlos_Tico: And you should ask the channel in general, not target individuals. |
15:41.37 | p3nguin | Does asterisk use .ttml files? |
15:41.53 | Carlos_Tico | http://pastebin.com/mde4872f |
15:41.59 | Carlos_Tico | ok |
15:42.09 | Carlos_Tico | i get the dial tone after dialing Disa |
15:42.16 | Carlos_Tico | but then i cannot dial nothing |
15:43.14 | Carlos_Tico | [Sep 21 10:41:07] WARNING[10508]: app_disa.c:246 disa_exec: DISA password file not found on chan Zap/1-1 |
15:45.36 | Carlos_Tico | any ideas ? |
15:46.59 | [TK]D-Fender | p3nguin: No. |
15:47.28 | [TK]D-Fender | Carlos_Tico: Error means what it says. It didn't find a match for whatever PIN was entered |
15:47.37 | p3nguin | I didn't think so, but someone asked and I didn't know for certain. I wasn't familiar with seeing those file extensions on asterisk. |
15:47.49 | [TK]D-Fender | Carlos_Tico: "DISA password file not found" you specify a password file, and it can't find it |
15:48.04 | Carlos_Tico | but i got the same error even with Disa without a password |
15:48.32 | [TK]D-Fender | Carlos_Tico: Show a real pastebin of the complete failure and your configs |
15:49.56 | Carlos_Tico | Executing [s@voicemenu-custom-1:2] DISA("Zap/1-1", "|DLPN_DialPlan1") in new stack |
15:49.56 | Carlos_Tico | then |
15:50.03 | Carlos_Tico | i dial and nothing... just bussy tone |
15:50.14 | Carlos_Tico | no other msg |
15:50.53 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
15:51.00 | [TK]D-Fender | Carlos_Tico: PAStebIN |
15:51.13 | [TK]D-Fender | Carlos_Tico: I said the complete failure. |
15:51.13 | raden_work | is there a way when i forward calls to forward caller ID ? |
15:51.16 | [TK]D-Fender | ~pb |
15:51.17 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
15:51.38 | [TK]D-Fender | raden_work: "forward" how? |
15:52.33 | raden_work | we forward our extension when we out of office I would like the actual calling number to be deisplayed instead of our company number |
15:52.45 | [TK]D-Fender | raden_work: "forward" how? <- |
15:53.01 | raden_work | astdb |
15:53.12 | [TK]D-Fender | raden_work: AstDB has nothing to do with forwarding... |
15:53.23 | raden_work | well wtf do you mean forward how ?> |
15:53.54 | [TK]D-Fender | raden_work: What commands/functions are actually used in this "forward". |
15:54.07 | [TK]D-Fender | raden_work: Clarify your use of this term |
15:54.18 | Carlos_Tico | http://pastebin.com/d45c339df |
15:54.26 | [TK]D-Fender | raden_work: It is currently rather vague including where the call is going in the call process |
15:55.18 | [TK]D-Fender | Carlos_Tico: how many digits do you get to enter before it hangs up? |
15:55.20 | p3nguin | raden_work: Do you mean transfer them? Do you mean when they use Followme? Clarify. |
15:55.22 | raden_work | when we dial *72, # we forward to our cell phones normally, I want the caller id information to come through to the cell phones |
15:55.47 | [TK]D-Fender | raden_work: what is *72? this isn't some magic number for *... |
15:56.29 | raden_work | all im asking is can it be done ? |
15:56.29 | Carlos_Tico | i can enter what ever number of digits... then a little silence and bussy tone |
15:57.18 | Carlos_Tico | not other msg in CLI |
15:57.37 | raden_work | I set everything is astdb |
15:59.20 | *** join/#asterisk LtScarr (i=benno@palm.hoeg.nl) |
15:59.26 | LtScarr | hey guys |
15:59.38 | LtScarr | i have a security related question |
16:00.21 | LtScarr | is it posible to dail extensions through sip without registering? |
16:00.42 | LtScarr | and is that default behavior? |
16:01.27 | *** join/#asterisk wildzero-cw (n=chatzill@p50997436.dip0.t-ipconnect.de) |
16:01.58 | *** join/#asterisk raden_work (n=tanning@69.179.99.17) |
16:02.32 | raden_work | [TK]D-Fender, http://pastebin.com/m21512a2e |
16:03.29 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
16:03.30 | raden_work | thats how i have it working at the moment id like to take the inbound caller ID and send it to the forwarded number |
16:04.12 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
16:04.22 | raden_work | maybe even with a number in front of it soo i know its forwarded |
16:05.14 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
16:09.13 | Carlos_Tico | how can i fix this error ?? - ast_get_srv: SRV lookup for '_sip._udp.proxy01.sipphone.com' mapped to host proxy01.sipphone.com, port 5060 |
16:10.19 | Qwell | Carlos_Tico: Who says it's an error? |
16:10.53 | Carlos_Tico | yesterday TK D-fender told me |
16:10.59 | Carlos_Tico | that it was a NAT issue |
16:11.04 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
16:12.36 | wildzero-cw | hello, 1.4.27-rc1 is available and im testing it (with no problems), how much time it it use from rc1 to final release on average? |
16:13.39 | Qwell | wildzero-cw: when it's ready, basically |
16:13.41 | Naikrovek | wildzero-cw: few weeks i'm guessing. why do you need THAT version |
16:14.51 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
16:15.34 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
16:16.15 | wildzero-cw | Naikrovek: because i just build a 1.4.27-rc1 deb (debian) for my system, and want to wait to distribute it until it's final. iam using 1.4.20 and it's much to old |
16:16.28 | Qwell | wildzero-cw: well, you could use 1.4.26.2 for now.. |
16:16.37 | p3nguin | Use something newer, but not all the way into the rc. |
16:17.47 | wildzero-cw | yes, but then is need to test 1.4.26.2 and 1.4.27 and distribute them twice, so ei just wait until it's final |
16:18.22 | p3nguin | What's the big deal? |
16:18.46 | p3nguin | Use 1.4.26.2 now. Distribute it. Use it. Like it. |
16:18.53 | p3nguin | What's the problem? |
16:19.37 | p3nguin | Like you were already asked, why do you need THAT version? Why not use 1.4.26.2 for the next few years? |
16:19.48 | p3nguin | It's certainly stable enough. |
16:21.06 | *** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
16:21.55 | geneticx | Hello everyone, has anyone implemented or has any experience with skype for asterisk? |
16:23.02 | Carlos_Tico | anyone here managed to setup magic jack as a trunk on asterisk ? |
16:23.24 | *** join/#asterisk Faithful (n=Faithful@124.217.119.152) |
16:23.33 | Superbartt | geneticx it kinda sux imho :P |
16:23.59 | p3nguin | Where is the app_dial timeout set for followme? |
16:24.19 | geneticx | Superbartt: interesting, how come? any bad experiences you might want to share? |
16:24.23 | Naikrovek | magic jack? don't cheap out |
16:24.44 | Superbartt | geneticx pain in the ass to setup, still an actual skype running |
16:25.39 | geneticx | Superbatt: ah I knew it.. I hate when |
16:26.34 | geneticx | Superbatt: when they want to throw untested stuff like this into production before even trying it |
16:27.02 | geneticx | Superbatt: you probably saved me a lot of headaches then.. |
16:27.47 | geneticx | they were ready to have about 8 people on skype for asterisk..you should've seen my face.. |
16:32.32 | [TK]D-Fender | raden_work: exten => 103,n(forward),Dial(LOCAL/${DB(CFIM/${EXTEN})}@to-callcentric,18) <-- this is not forwarding |
16:32.53 | [TK]D-Fender | raden_work: forwarding is an action taken by the device like on a SIP phone. |
16:33.03 | [TK]D-Fender | raden_work: You are simply choosing to dial out callcentric |
16:33.14 | [TK]D-Fender | raden_work: So if they let you rig your Caller ID, then go do it |
16:34.41 | Qwell | Superbartt: umm, no. |
16:34.59 | [TK]D-Fender | LtScarr: Without who registereing? Who is placing the call? |
16:35.17 | Qwell | Superbartt: You are talking about a completely different product. SFA does not require any hacks that the other ones do. |
16:35.21 | [TK]D-Fender | LtAnd no, registration is normally not needed to place calls from a device as long as it know where to call. |
16:35.59 | Qwell | geneticx: get yourself 1 license and try it out. if it works, get the rest. |
16:36.25 | Qwell | it's rather trivial to setup/use. and it's...actually supported. |
16:36.37 | Superbartt | Qwell really? Last time I checked for that (a while ago) it required an running skype etc... Was mostly a dirty hack |
16:36.45 | Qwell | ~skypeforasterisk |
16:36.45 | infobot | i heard skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details |
16:36.58 | Naikrovek | skypeforasterisk is relatively new i believe, yes? |
16:37.03 | Qwell | yes |
16:37.08 | Naikrovek | cool |
16:37.11 | Qwell | released at the beginning of the month |
16:37.28 | wcselby | wow |
16:37.48 | wcselby | just made a custom background for my 7961, and when it's displayed on the phone it looks compeltely different |
16:37.49 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
16:37.58 | Naikrovek | wcselby: explain completely different |
16:38.03 | Qwell | wcselby: welcome to Cisco |
16:38.07 | wcselby | s/compeltely/completely/ |
16:38.09 | Naikrovek | change colors? |
16:38.13 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
16:38.13 | Superbartt | nice Qwell :) Excuse me for my wrong opinion |
16:38.19 | Naikrovek | 7961 color screen? |
16:38.25 | wcselby | yeah, what I gave as a clear background I got a black background |
16:38.28 | [TK]D-Fender | Naikrovek: No |
16:38.42 | wcselby | Naikrovek - no, greyscale |
16:38.43 | Naikrovek | ah transparency is always iffy when changing devices |
16:38.44 | Qwell | wcselby: clear requires transparency |
16:38.46 | [TK]D-Fender | Naikrovek: higher res, 802.3af, etc |
16:38.54 | Naikrovek | [TK]D-Fender: ah okay thanks |
16:39.17 | Qwell | there's no such thing as transparency in grayscale, heh |
16:39.35 | wcselby | well, yeah, transparent background I guess. |
16:39.45 | Naikrovek | well you can assign a color to be the same color as the native BG color, probably white |
16:39.49 | wcselby | heh, okay. need to relook at the logo |
16:41.38 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
16:41.54 | *** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net) |
16:45.12 | geneticx | Qwell: Ok, ill take that into consideration. Thanks.. |
16:52.32 | *** join/#asterisk AwayML (n=AndyML@pool-173-49-144-213.phlapa.fios.verizon.net) |
16:53.29 | *** join/#asterisk xpot-mobile (n=james@173.8.94.1) |
16:53.42 | raden_work | [TK]D-Fender, is there a better way todo what im doing and basically your saying all i have todo is set my own CID and send id have to switch to vitelity when forwarding but itd work i guess |
16:54.19 | [TK]D-Fender | raden_work: Go set your callerID. And this is not "forwarding" and this has absolutely no relationship to that. |
16:54.25 | Qwell | raden_work: punctuation |
16:55.26 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
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17:08.45 | wcselby | hmmmmm |
17:08.48 | wcselby | what to eat for lunch |
17:08.53 | wcselby | decisions decisions |
17:09.57 | Pan3D | food is always a good choice |
17:11.02 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
17:15.15 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
17:18.47 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
17:21.01 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
17:25.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
17:29.59 | *** join/#asterisk TimToady_ (n=moi@adsl318-240.kln.forthnet.gr) |
17:30.27 | *** join/#asterisk gazzerh (n=garryh@93-97-187-150.zone5.bethere.co.uk) |
17:33.26 | Katty | wcselby: something low on the glycemic index. |
17:34.12 | Qwell | wheat germ |
17:35.43 | Katty | i was actually going to suggest cheese. |
17:36.25 | Katty | 1 cup of milk = the same glycemic load as a piece of toast. |
17:37.17 | Katty | aka load of 9. 1 cup of cheese is a load of 3 |
17:37.42 | Katty | a cup of watermelon is 3, cup of grapes is 5 |
17:38.00 | Katty | somehow, i've always found watermelon sweeter than grapes tho. |
17:38.07 | Katty | and watermelon definately sweeter than milk. |
17:38.14 | Katty | but milk has a high glycemic level. |
17:38.19 | p3nguin | If ${EXTEN:1} will = strip the first character off the extension, what do we use to strip the last character off of it? |
17:43.41 | henkoegema | ${EXTEN:-1} |
17:44.04 | Qwell | henkoegema: that will get only the last |
17:44.31 | Qwell | ${EXTEN:0:-1} |
17:46.40 | henkoegema | :) |
17:47.34 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
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18:04.11 | *** join/#asterisk denon (i=denon@sassinak.net) |
18:04.11 | *** mode/#asterisk [+o denon] by ChanServ |
18:04.24 | *** join/#asterisk hesco (n=hesco@24.99.160.121) |
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18:10.09 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
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18:13.31 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
18:14.16 | wcselby | Qwell - is there logic to say if I only want the last 10 digits of a number using the ${EXTEN:x:y} logic? |
18:14.31 | [TK]D-Fender | wcselby: "core show function LEN" |
18:15.26 | wcselby | LEN |
18:15.29 | wcselby | okay thanks [TK]D-Fender |
18:16.38 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
18:20.47 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
18:22.55 | raden_work | what would this be considered then if its not forwarding redirection ? |
18:23.13 | raden_work | why do all the telcos call the same procedure forwarding ? |
18:23.17 | raden_work | im a lil confussed |
18:23.31 | lirakis_ | raden_work, what are you talking about |
18:24.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
18:24.44 | Pan3D | heh |
18:26.16 | [TK]D-Fender | raden_work: You are doing a dumb dial yourself in the dialplan. the fact you bothered to SHOOSE between to possible action is irrelevent. the CID from the channel is the same as when it started |
18:26.44 | [TK]D-Fender | raden_work: When you call a SIP phone and IT decides to bounce you, THAT is a forward (SIP 302 Redirect) |
18:26.58 | [TK]D-Fender | raden_work: THAT can have implications on who the call is coming from |
18:27.11 | [TK]D-Fender | raden_work: Here you are just shoosing between door #1, and door #2 in the dialpla |
18:27.17 | raden_work | is there a way to set a redirect in asterisk ? |
18:27.32 | [TK]D-Fender | raden_work: which is a conceptual "forwarding", but the term must be avoided in the programming sense |
18:27.41 | [TK]D-Fender | raden_work: this ISN"T a REDIRECT |
18:27.44 | *** part/#asterisk fuxu2 (i=iconicfl@www.kevinlynn.com) |
18:27.46 | [TK]D-Fender | raden_work: just stop calling it that |
18:28.03 | [TK]D-Fender | raden_work: Its a dumb dial just like you use in your typical PSTN numbered dialplan patterns |
18:28.07 | *** join/#asterisk noRTFM (i=iconicfl@www.kevinlynn.com) |
18:28.13 | [TK]D-Fender | raden_work: So go set the CALLERID() already... |
18:28.24 | lirakis_ | [TK]D-Fender, i think he got it... he just wants to know if he can do a redirect instead of a door 1 or door 2 option |
18:28.40 | [TK]D-Fender | lirakisNo, I'm sure it'll take a few dozen more swings still... |
18:28.45 | lirakis_ | heh heh |
18:28.53 | raden_work | lirakis, exactly |
18:28.55 | Pan3D | raden_work: so, you want to call into an * box and have that call redirected to another number without any user intervention? |
18:28.56 | [TK]D-Fender | puts some rusty nails onto his ClueBat (tm) |
18:29.03 | Pan3D | lol |
18:29.04 | Katty | takes cluebat away from [TK]D-Fender. Again. |
18:29.11 | [TK]D-Fender | Pan3D: Don't... for the love of god, don't... |
18:29.17 | lirakis_ | lol |
18:29.25 | Pan3D | this is why we can't have nice things |
18:29.39 | LtScarr | [TK]D-Fender: to answer your question earlier: |
18:29.39 | LtScarr | 18:34 < [TK]D-Fender> LtScarr: Without who registereing? Who is placing the call? |
18:30.03 | [TK]D-Fender | Pan3D: Yes, you'd break them. Now hand them back gently |
18:30.24 | Pan3D | lol |
18:30.52 | LtScarr | i got sip id's like SIP/59823 using my dialplan without registering |
18:31.03 | LtScarr | and 59823 isn't a valid account |
18:31.23 | fuxu2 | dang.. ya know.. I totally was tryin to make up a funny question to ask.. |
18:31.32 | fuxu2 | and I just couldn't think of anything.. |
18:31.43 | fuxu2 | someone took away my creativity.. :( |
18:31.56 | raden_work | Pan3D, sometimes yes |
18:31.57 | LtScarr | i solved it by setting the default context to nothing |
18:32.00 | Pan3D | lol |
18:32.13 | [TK]D-Fender | LtScarr: O RLY? |
18:32.17 | [TK]D-Fender | LtDo share... |
18:32.21 | Pan3D | raden_work: OK, I think [TK]D-Fender has a point. You should step back a bit and read teh book. |
18:33.13 | Pan3D | there's a chapter on telecom in general which is useful for lingo, and some specific examples of how dial plans work (including what it is you want to do... sometimes) |
18:33.13 | [TK]D-Fender | raden_work: Set(CALLERID(num)=1800GETCLUE) |
18:33.16 | lirakis_ | shameless plug for redirect based routing http://routengn.com/ |
18:33.40 | LtScarr | i'm running 1.4.21.2 |
18:33.44 | Pan3D | [TK]D-Fender: lol, be nice |
18:33.56 | [TK]D-Fender | Pan3D: i AM, i EVEN GAVE HIM THE CORRECT SYNTAX |
18:34.03 | [TK]D-Fender | Caps fail.. |
18:34.35 | [TK]D-Fender | LtScarr: Well if you allwed unauth'd guests access to a dangerous context like that... YGWYPF :) |
18:35.05 | garymc | YGWYPF.... wowo what the hell is that? :) |
18:35.08 | LtScarr | [TK]D-Fender: i don't want any unauth'd guests :-) |
18:35.13 | [TK]D-Fender | ~ygwypf |
18:35.14 | infobot | ygwypf is probably You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
18:35.25 | garymc | Ahhhh |
18:35.32 | garymc | good one |
18:35.47 | [TK]D-Fender | garymc: Polite version of "Suck it biatch :p" |
18:35.50 | Pan3D | that description should be updated to include bad decisions that are non-financial |
18:35.53 | garymc | :P |
18:35.55 | garymc | lol |
18:35.57 | Pan3D | i.e. open contexts |
18:36.16 | LtScarr | [TK]D-Fender: well i ran into this by coincidence |
18:36.17 | garymc | is glad he bought the good stuff :P |
18:36.46 | LtScarr | no real harm done at all |
18:38.00 | LtScarr | just curious about if it's default behavior |
18:38.01 | [TK]D-Fender | LtScarr: "allowguest=no" <- |
18:38.42 | LtScarr | is that sip specific? |
18:38.47 | Pan3D | man, I am so glad I read the book before doing anything with *. It seems 70% of the questions in here are basic stuff covered in the book. |
18:39.21 | [TK]D-Fender | Pan3D: another 515 once you include the docs in the tarball |
18:39.23 | hesco | At the *CLI>, I see: "Auto fallthrough, channel 'SIP/hugh_desk-08368fb0' status is 'UNKNOWN'", the NoOp()'s that are and are not presented at the *CLI> tell me the dialplan is failing on this priority: "exten => 993,n,AGI(smartdialer.agi,GET_LIST_REPORT,${SID},${AGENT_CHANNEL})", the .agi script passes the perl -wc test; and if I bypass this logic, other priorities have no problem calling this agi script using different functions. At the moment, |
18:39.23 | hesco | this function of the agi script only prints a debug statement, although the module->method called by it and presently commented out does pass basic regression tests. So what is this error, what does it mean? |
18:39.24 | [TK]D-Fender | 15% |
18:39.35 | LtScarr | the wiki page about security doesn't cover it... |
18:39.48 | [TK]D-Fender | WIKI = CRAP |
18:40.15 | LtScarr | alright good point :-) |
18:40.25 | [TK]D-Fender | random ancient shit written by well-meaning yet still all-too-often wrong people |
18:41.03 | [TK]D-Fender | LtScarr: Should have seen yesterday's scuffle over the giant misconception that is the "s" Asterisk Standard Extension. |
18:41.46 | LtScarr | :) |
18:41.46 | wcselby | ooooh |
18:41.48 | wcselby | sounds like fun |
18:42.07 | hesco | ~paste |
18:42.07 | infobot | i guess paste is http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/ |
18:42.14 | [TK]D-Fender | wcselby: Diet Fun (tm). Just like Real Fun (tm), only half the fun. |
18:42.19 | wcselby | lol |
18:42.34 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
18:42.48 | wcselby | ahhh |
18:42.58 | wcselby | I think what I want is ${EXTEN:-10} |
18:43.06 | p3nguin | Is ${CALLERIDNUM} no longer valid? |
18:43.12 | LtScarr | i wonder how many servers don't have allowguest=no... |
18:43.18 | wcselby | that will give me the last 10 digits of the variable |
18:43.24 | ian6 | LtScarr: most, I would think. The default allows it. |
18:43.28 | wcselby | p3nguin - not since 1.0 |
18:43.42 | wcselby | p3nguin - that is, depricated in 1.2, not used in 1.4 |
18:43.43 | p3nguin | No wonder my changes weren't working out. |
18:43.51 | wcselby | ${CALLERID(num} |
18:43.58 | wcselby | ${CALLERID(num)} |
18:43.58 | p3nguin | Thought so. |
18:44.03 | p3nguin | Just needed to confirm. |
18:44.11 | Pan3D | LtScarr: probably a lot -- particularly when folks probably use that instead of troubleshooting auth problems |
18:46.07 | *** join/#asterisk jlnt (n=jlnt@cisco2.jlmail.com) |
18:46.53 | *** join/#asterisk SebastianS (n=schu@adsl-dyn14.78-98-105.t-com.sk) |
18:49.07 | wcselby | wow |
18:49.18 | wcselby | i just got a sales call on a phoneline that I haven't given the number out to, ever |
18:49.26 | p3nguin | sucks |
18:49.29 | wcselby | in fact, I just set the DID up less than two weeks ago |
18:49.34 | wcselby | maybe just over two weeks |
18:49.38 | wcselby | funny |
18:49.48 | hesco | OK, I am retrying my question this time supported by a paste ( http://bin.cakephp.org/view/193968548 ) which I trust will give some context. Any pointers would be appreciated. |
18:51.07 | wcselby | hesco - show us the CLI output of the failure |
18:51.23 | *** join/#asterisk tamiel (n=tamiel@ip-7.net-81-220-254.rev.numericable.fr) |
18:51.28 | [TK]D-Fender | hesco: Now show us a new pastebin where we actually see EXECutION |
18:51.35 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
18:52.01 | hesco | wcselby: coming your way. |
18:53.12 | hesco | http://bin.cakephp.org/view/654130842 |
18:53.49 | hesco | I get a fast busy and hang up, leading to the 'Call completed' line |
18:55.17 | wcselby | what's your verbosity level set to? |
18:55.22 | hesco | 3 |
18:55.36 | wcselby | pop it to 30 and redo the CLI output please |
18:55.43 | hesco | coming your way |
18:56.23 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
18:56.23 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
18:58.19 | [TK]D-Fender | hesco: 993 != 991 |
18:58.23 | [TK]D-Fender | hesco: MATH FAIL |
18:58.25 | hesco | http://bin.cakephp.org/view/1134546063 |
18:58.55 | bpgoldsb | I just upgraded to 1.6.1, and it appears both chan_console and chan_alsa were compiled. However, when I start asterisk, chan_console loads first, and when it tries to load chan_alsa, I get 'Already have a handler for type 'Console'' + 'Unable to register channel class 'Console'' |
18:58.59 | bpgoldsb | Any ideas? |
18:59.26 | hesco | thanks [TK]D-Fender: copy-n-paste error does it to me again |
18:59.26 | wcselby | [TK]D-Fender - lol totally missed that |
18:59.35 | Qwell | bpgoldsb: don't load one |
18:59.39 | Qwell | bpgoldsb: see modules.conf |
19:00.28 | bpgoldsb | Sure, I know how to not load one. I figured maybe one was deprecated in a newer version or something. |
19:03.36 | p3nguin | In 1.4, is Set(Global(OldCID)=${CALLERID(num)}) the correct way to set a global variable OldCID to the value of the current CALLERID(num)? |
19:04.09 | [TK]D-Fender | p3nguin: No, GLOBAL is a functions... and like all others is case-sensitive <- |
19:04.14 | [TK]D-Fender | p3nguin: Also, globals = evil |
19:04.28 | [TK]D-Fender | p3nguin: And in most implementations, the wrong solutions |
19:04.38 | p3nguin | I'm trying to get a variable to be carried through several contexts. |
19:04.51 | [TK]D-Fender | p3nguin: there is no variable scope within a call |
19:05.00 | [TK]D-Fender | p3nguin: Misconception #2 |
19:06.24 | *** join/#asterisk voipmonk (n=voipmonk@69.172.93.45) |
19:06.33 | p3nguin | I tried to set the variable using Set(OldCID=${CALLERID(num)}) in one context and then read ${OldCID} in another context. It was null. |
19:06.55 | [TK]D-Fender | p3nguin: And I don't see your code and the attempt./.. |
19:07.20 | [TK]D-Fender | ~pb |
19:07.21 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
19:07.29 | p3nguin | If that's the wrong way, why not just tell me the right way? |
19:08.12 | carrar | belligerent about needing help |
19:08.26 | bpgoldsb | p3nguin, Are you setting it before you bridge with another channel? |
19:08.29 | bpgoldsb | i.e. from a dial |
19:08.46 | [TK]D-Fender | p3nguin: Yes, that looks fine. I'm also sure you're doing something else wrong |
19:09.10 | bpgoldsb | If so you might need to look at variable inheritance, as [TK]D-Fender pointed out to me. |
19:09.12 | [TK]D-Fender | p3nguin: So show me the evidence and I'll show you where you went wrong. Odds are its in the part you're not telling me |
19:09.43 | [TK]D-Fender | bpgoldsb: there is no inheritance. he jsut said one context to another |
19:09.53 | [TK]D-Fender | bpgoldsb: So yeh, I want PROOF |
19:10.07 | carrar | You can't handle the PROOF!!! |
19:10.26 | [TK]D-Fender | calls a Code Red on carrar |
19:10.30 | carrar | heh |
19:11.46 | p3nguin | Could have been a mere typo. I changed back from trying to set it globally, and it seems to be carried into the next context. |
19:12.31 | [TK]D-Fender | z0mg a miracle! |
19:12.37 | p3nguin | Not really. |
19:13.06 | *** join/#asterisk hardwire (n=hardwire@216-67-99-228.static.acsalaska.net) |
19:13.16 | hardwire | anybody have to restart iaxmodem/hylafax often? |
19:13.32 | hardwire | I can't quite figure out which process stops performing correctly. |
19:13.56 | Katty | issues an Amber Alert on eppigy |
19:16.19 | Naikrovek | i wonder if those amber alerts actually work |
19:16.40 | hardwire | steal a child and see. |
19:16.47 | Katty | yes, they do. |
19:16.54 | Naikrovek | nah i have two kids already. don't want a third |
19:16.59 | Katty | most turn out to be kidnappings by parents. |
19:17.04 | Naikrovek | yes |
19:17.09 | Katty | or other relatives. |
19:17.15 | hardwire | Katty: did you see the castle episode about that? |
19:17.18 | Naikrovek | like 95% of kidnappings and sexual abuse on children is family |
19:17.50 | Naikrovek | but they can't automatically make the family suspects because that's profiling |
19:19.07 | Naikrovek | which sucks because profiling effing works |
19:19.28 | russellb | how about that Asterisk! |
19:19.36 | Naikrovek | lol, yeah |
19:19.46 | Naikrovek | man i gotta watch my off topic BS |
19:19.50 | Naikrovek | i'm bad at that... |
19:19.56 | Katty | you think you're bad? |
19:20.04 | Katty | when was the last time i talked about asterisk? ;P |
19:20.13 | Naikrovek | just now you did |
19:20.29 | [TK]D-Fender | Katty: Yeah... you KeySystem sellout :p |
19:20.41 | Katty | grumps. |
19:20.53 | *** join/#asterisk TimToady_ (n=moi@adsl148-239.kln.forthnet.gr) |
19:20.55 | jaytee | DK Strata FTW!!! |
19:21.18 | Katty | hugs on jaytee |
19:21.44 | Katty | jaytee: i am toshiba certified. |
19:21.45 | jaytee | I wonder if Avaya's gonna get saddled with all of Nortel's Norstar key system business crap |
19:21.47 | [TK]D-Fender | jayOne of my earlies clients was a Toshiba reseller getting into * & VoIP :) |
19:21.55 | [TK]D-Fender | jaytee: One of my earliest clients was a Toshiba reseller getting into * & VoIP :) |
19:21.58 | jaytee | Katty, I'm certifiable! |
19:22.06 | [TK]D-Fender | commits jaytee |
19:22.12 | Katty | visits jaytee |
19:22.15 | jaytee | 900 million |
19:22.17 | *** join/#asterisk scalex000 (n=chatzill@190.166.149.204) |
19:22.30 | jaytee | what a fall from grace that one was. |
19:22.31 | Katty | i can't help it. |
19:22.37 | Katty | my company wants to sell what Sells. |
19:22.50 | Katty | and what Sells around here, is Toshiba and Samsung POS el cheapo systems. |
19:23.02 | scalex000 | good afternoon |
19:23.11 | Katty | howdy. |
19:24.48 | Katty | misses eppigy |
19:24.51 | jaytee | Katty, in this period with this economy, you keep whatever job you have and sell whatever shit you have to sell if you've got at least a lick of common sense. Let the retards stand on loyalty and principle. When they start whining about the fact they're starving tell them you'll make them a loyalty and principle sandwich. |
19:25.00 | jaytee | yeah, where is Dave today? |
19:25.22 | Katty | idk, maybe he went to take a nap |
19:25.27 | Katty | he told me he was sleepy earlier. |
19:25.50 | [TK]D-Fender | jaytee: Ferengi Rule of Acquisition #109: Dignity and an empty sack are still worth an empty sack |
19:26.22 | jaytee | [TK]D-Fender, if I ever get venture capital I'm tagging you for my CTO |
19:26.35 | Naikrovek | i love those ferengi rules |
19:26.56 | Naikrovek | ack, offtopic again |
19:26.59 | Naikrovek | shyte |
19:27.07 | [TK]D-Fender | jaytee: Ferengi Rule of Acquisition #1: Once you have their money, you never give it back |
19:27.29 | wonderworld | i need to write a bill right now and can't decide if i'd charge a bit too much or not. i am another moral-victim. |
19:27.34 | Katty | goes hunting for eppigy |
19:28.04 | *** join/#asterisk eppigy (n=Dave@216-139-245-58.aus.us.siteprotect.com) |
19:28.06 | eppigy | hello |
19:28.07 | jaytee | people who know all of the Ferengi Rules of Acquisition are still only have as nerdy and weird as people who speak fluent Klingon |
19:28.08 | eppigy | i am dave |
19:28.12 | jaytee | DAVE!!!!!!! |
19:28.20 | jaytee | we were just wondering about you |
19:28.27 | eppigy | HI |
19:28.36 | eppigy | my shell box got rebooted |
19:28.44 | Katty | psh. shell box. |
19:28.45 | eppigy | and it takes me a while to rejoin everything |
19:28.48 | Katty | i can still find you. |
19:28.51 | eppigy | yesh |
19:28.53 | Katty | YOU CAN"T HIDE FROM ME |
19:28.54 | eppigy | you know where i live |
19:28.59 | [TK]D-Fender | 's shell box has a conch in it |
19:29.08 | Katty | [TK]D-Fender: you're a conch. |
19:29.13 | jaytee | mmmm, conch chowdah! |
19:29.30 | [TK]D-Fender | eppigy: nothing says "I love you" quite like a restraining order... |
19:29.57 | jaytee | wonders which version of 1.6.x to download to test sip tcp to Exchange UM |
19:30.00 | Katty | jaytee: curried :> |
19:30.18 | Katty | wait, what? |
19:30.28 | *** join/#asterisk oej (n=olle@ns.webway.se) |
19:30.40 | [TK]D-Fender | jaytee: 1.6.0.15 |
19:30.46 | Katty | sip tcp to exchange? |
19:30.53 | Katty | linkylink? |
19:30.58 | jaytee | [TK]D-Fender, thanks. 1.6.1.x still to new? |
19:30.59 | Katty | infobot: exchange |
19:31.12 | Qwell | infobot: exchange Katty with a banana |
19:31.13 | Katty | infobot: well of course it doesn't. it's exchange. |
19:31.43 | Katty | Qwell: do you have a receipt for that transaction? |
19:31.50 | jaytee | Katty, we use Exchange UM with * 1.4 but use sipX as a udp/tcp transform proxy to UM since * 1.4 only speaks udp and Exchange is Microsoft and they ALWAYS HAVE TO BE DIFFERENT |
19:32.08 | Katty | what is 'UM'? |
19:32.10 | Qwell | jaytee: well, technically both violate the RFCs |
19:32.15 | jaytee | Unified Messaging |
19:32.28 | jaytee | Qwell, both? |
19:32.30 | Katty | ohisee. |
19:32.34 | Qwell | both are required I do believe |
19:32.35 | eppigy | unified messing |
19:32.46 | Qwell | both = Asterisk and Exchange |
19:32.53 | Katty | who comes up with these Big Words |
19:32.59 | Qwell | and both = TCP and UDP |
19:33.00 | Katty | Unified Messaging |
19:33.02 | Naikrovek | since when does anythign not violate the RFCs it claims to imiplement |
19:33.05 | Qwell | Katty: Cisco |
19:33.12 | Katty | well cisco can go... |
19:33.14 | Katty | eat a sandwich |
19:33.15 | jaytee | ah, so 1.4 violatest the RFC and Exchange 2007 UM does too. but then that means 1.6 doesn't violate it!!! |
19:33.23 | Qwell | jaytee: something like that |
19:33.28 | Naikrovek | jaytee: safe to assume that everything does, really |
19:33.35 | [TK]D-Fender | jaytee: relatively... it is a full release so if you want to play around it shouldn't amtter too much |
19:33.43 | Naikrovek | but as i understand it * 1.6 and Exchange 2007 can talk just fine |
19:33.43 | jaytee | Cisco Kid weren't no friend of mine. |
19:34.00 | Katty | i really like cisco, actually. |
19:34.03 | Katty | they have some amazing products. |
19:34.09 | Katty | with an amazing price tag. |
19:34.16 | Naikrovek | oh yes |
19:34.21 | Katty | even their VP's halo has an amazing price tag |
19:34.23 | Naikrovek | but really good products |
19:34.25 | Katty | i think he forgot to snip it off. |
19:34.40 | Katty | we won't go there tho. |
19:34.50 | Katty | i'm particularly fond of the their blackberry app |
19:34.58 | jaytee | I think drmessano said that 1.6.0.15 worked well. I'll start there and if it's stable that'll be my 1.6 upgrade path to do away with sipX as an in between. |
19:35.04 | Katty | it's a little status/presense thingy |
19:35.11 | Katty | with COLORFUL DOTS |
19:35.15 | Katty | more dots! more dots! |
19:35.39 | jaytee | Dots are ok but I prefer Crows being a big licorice fan |
19:35.52 | Katty | Qwell: you did get the reference, right? |
19:35.56 | Katty | Qwell: more dots! more dots! |
19:36.27 | Katty | doesn't have any dots :< |
19:36.35 | Katty | i do have a couple Hots, tho. |
19:36.36 | Naikrovek | has freckles |
19:38.55 | wcselby | I have a user saying that when one other user transfers a call from a queue to her, all she hears is hold music. Me being the call being transfered, I can hear the person who says all she hears is hold music. I do not hear the hold music. Here's the CLI from when it happens: http://pastebin.com/d1255631c |
19:39.14 | Naikrovek | that sounds weird |
19:39.17 | wcselby | any thoughts? |
19:39.49 | Katty | i think something is faschnuckadid up |
19:40.11 | Katty | looks at post |
19:41.48 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
19:44.44 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
19:47.53 | Naikrovek | just spoke with comcast about maybe getting a new internet connection. tried to explain to her that I spoke with the business ethernet guys (fiber) and she didn't seem to know the difference between coaxial cable and fiber optic |
19:48.21 | Naikrovek | told her that they quoted me $1800/month for 20mbps each direction because they had to build infrastructure to my business |
19:48.35 | Naikrovek | eh nevermind |
19:48.41 | Naikrovek | they're just so stupid |
19:48.48 | wcselby | lol |
19:48.56 | wcselby | you spoke with the wrong person, obviously |
19:49.06 | Naikrovek | she was an accoutn manager |
19:49.54 | Naikrovek | but yes wrong person |
19:50.49 | *** join/#asterisk jtodd (i=jc3v8cag@ns.fox-den.com) |
19:50.49 | *** mode/#asterisk [+o jtodd] by ChanServ |
19:53.44 | *** join/#asterisk zpertee (n=chatzill@68.142.169.62) |
19:53.53 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
19:56.11 | zpertee | Does anyone know if Intel 537E cards will work with asterisk? |
19:56.55 | *** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
19:57.43 | IBC_jkenney | i need a bit of assistance i am using chanspy and what i want to do is when you enter a specific passcode you are listening to a specific $SPYGOUP |
19:57.48 | IBC_jkenney | has aonyone done this |
19:59.58 | Qwell | zpertee: That is a modem. No it will not. |
20:00.06 | IBC_jkenney | i need a bit of assistance i am using chanspy and what i want to do is when you enter a specific passcode you are listening to a specific $SPYGOUP |
20:00.08 | IBC_jkenney | has aonyone done this |
20:00.31 | wcselby | try asking one more time Iamnacho |
20:00.32 | wcselby | erm |
20:00.35 | wcselby | IBC_jkenney |
20:01.14 | zpertee | Qwell: ok. I had heard reports of at least one intel card being nothing more than a clone of a digium card |
20:01.19 | IBC_jkenney | Could someone please assist me with this problem i am having i would greatly appreciate it |
20:01.27 | *** join/#asterisk el_critter (n=critter@200.8.97.41) |
20:02.17 | wcselby | IBC_jkenney - use the Read() command to set a variable then run a check against that variable to determine which $SPYGROUP to run chanspy on |
20:02.34 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
20:02.58 | IBC_jkenney | I want to use the function to read the pw and channel variable from a file |
20:03.08 | IBC_jkenney | but i think i have the syntax wrong or something |
20:03.19 | IBC_jkenney | i'l like to get this working b.o.b today |
20:03.24 | wcselby | IBC_jkenney - okay. |
20:03.42 | TJNII | ~questions |
20:03.43 | infobot | remember, there are no stupid questions, just stupid people. <http://www.catb.org/~esr/faqs/smart-questions.html> |
20:03.44 | wcselby | IBC_jkenney - be sure you don't show us anything, because the more we see the less we can help you... |
20:03.52 | TJNII | IBC_jkenney: Read that link |
20:05.22 | *** join/#asterisk Tim_Toady (n=moi@77.49.156.2) |
20:05.28 | el_critter | hi |
20:06.07 | Pan3D | el hello |
20:09.00 | *** join/#asterisk Slezhuk (n=mammon@95-25-249-49.broadband.corbina.ru) |
20:09.46 | *** join/#asterisk Micc (n=Micc@c-98-225-59-171.hsd1.wa.comcast.net) |
20:18.51 | bmoraca | how well does DUNDI scale? can I have hundreds of peers or am I limited to several? |
20:19.16 | [TK]D-Fender | bmoraca: What is your goal? |
20:19.24 | wcselby | is ultramonkey still the preferred HA solution for linux? |
20:19.48 | bmoraca | use DUNDI to direct calls to my hosted PBXes from my media gateway servers as opposed to static call routes |
20:20.01 | IBC_jkenney | My Question is i would like to use the chanspy application for managers of different departments to monitor their employee's. I am looking at storing the password and group name in a textfile and then have the channel name read from the file what is the best way to go about it. Not every manager should have the ability to listen to calls in all area's |
20:20.13 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
20:21.14 | [TK]D-Fender | IBC_jkenney: Go make your own AGI and auth it yourself |
20:21.17 | TJNII | Probably an AGI script, I would say. |
20:21.41 | IBC_jkenney | <==== is looking for simple |
20:21.53 | bmoraca | what you're asking isn't simple |
20:22.05 | wcselby | IBC_jkenney - I gave you simple, you're looking for something more than simple |
20:22.45 | IBC_jkenney | according to chanspy it can do what i'm asking i think i am just doing it wrong |
20:23.19 | IBC_jkenney | <PROTECTED> |
20:23.19 | IBC_jkenney | <PROTECTED> |
20:23.19 | IBC_jkenney | <PROTECTED> |
20:23.19 | IBC_jkenney | <PROTECTED> |
20:24.01 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
20:24.06 | Katty | hai IBC_jkenney |
20:24.54 | bmoraca | [TK]D-Fender: the idea is this: instead of statically assigning my call routes from my media gateway servers, I want to instead potentially have those routes dynamically generated (this is the point of DUNDI, isn't it?). Consider OSPF stub areas as opposed to static routing in a large IP network. |
20:25.11 | IBC_jkenney | i wanted to do it with authenticate |
20:25.11 | IBC_jkenney | then pass it to chanspy |
20:25.11 | IBC_jkenney | sorry |
20:25.23 | *** join/#asterisk denon (i=denon@sassinak.net) |
20:25.23 | *** mode/#asterisk [+o denon] by ChanServ |
20:25.25 | [TK]D-Fender | IBC_jkenney: that is Authenticate. it won't create a mapping to channels its allowed to spy on. |
20:25.40 | [TK]D-Fender | IBC_jkenney: What you want is custom so yes, you're going to have to code. |
20:25.58 | IBC_jkenney | has no idea how to code |
20:26.18 | wcselby | IBC_jkenney - use the Read() command to set a variable then run a check against that variable to determine which $SPYGROUP to run chanspy on |
20:26.34 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
20:26.40 | [TK]D-Fender | IBC_jkenney: http://tinyurl.com/sulfm |
20:26.49 | [TK]D-Fender | :D |
20:26.52 | wcselby | but whatever, it's time to head out |
20:26.56 | wcselby | lol @ [TK]D-Fender |
20:27.03 | wcselby | heroes is on tonight! \o/ |
20:27.30 | [TK]D-Fender | checkout time.. BBIAB |
20:28.06 | IBC_jkenney | wow |
20:28.12 | IBC_jkenney | sorry to have bothered you all |
20:33.09 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
20:33.21 | dustybin | has anyone here ever been guilty of using ChanSpy() ? |
20:33.35 | Pan3D | IBC_jkenney: heh, not a question of bothering anyone... but if you want to administer this stuff, you gotta dive in. It's not like Fisher-Price VoIP server. It requires some preparation. |
20:33.56 | Naikrovek | Mr. Potato VoIP |
20:33.59 | Pan3D | lol |
20:34.11 | Naikrovek | he's right tho |
20:34.20 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
20:34.39 | Naikrovek | to be fair it IS a lot to take on at once |
20:34.50 | Naikrovek | asterisk is not simple to the untrained eye |
20:35.06 | Naikrovek | but, as [tk]d-fender always says, it's all in the dialplan |
20:35.15 | Naikrovek | get that figured out and you're 95% of the way there |
20:35.21 | Naikrovek | peace. |
20:36.14 | Katty | IBC_jkenney: are you okay? |
20:37.12 | Katty | IBC_jkenney: i wouldn't let them get ya down--everything worth doing takes time (= |
20:42.22 | zpertee | IBC_jkenney: Katty is right. My first asterisk job that I did I had planned on finishing it in a couple of weeks. A few months later I was still tinkering... just roll up your sleeves and get dirty :-) |
20:42.52 | Katty | is still tinkering after 4 years. |
20:42.56 | *** join/#asterisk brezular (n=brezular@adsl-dyn232.78-99-68.t-com.sk) |
20:43.12 | Katty | phone system is a never ending project. |
20:43.22 | Katty | like a company website |
20:43.31 | zpertee | couldn't agree more |
20:44.41 | IBC_jkenney | Oh o'm ok |
20:44.46 | IBC_jkenney | just flustered |
20:44.53 | IBC_jkenney | and i won't let them get me down |
20:45.40 | IBC_jkenney | i mean if they really don't know how to do it thats all the had to say no need to get on the defensive and try to make someone else look stupid to heal their ego |
20:45.41 | IBC_jkenney | :) |
20:45.44 | IBC_jkenney | i'm fine |
20:46.00 | IBC_jkenney | i just prefer not to have a pissing match |
20:46.02 | IBC_jkenney | thats all |
20:47.04 | IBC_jkenney | This is not my first deployment this is my first complicated deployment and when its a company you work for as an employee its a different then when your a contractor |
20:47.09 | zpertee | been in your shoes. its a real art to get the info you need without a pissing match :-) |
20:47.30 | IBC_jkenney | not really its a matter of finding mature people |
20:47.39 | Katty | well count me out. |
20:47.42 | IBC_jkenney | instead of the Kiddies |
20:47.43 | IBC_jkenney | lol |
20:47.47 | Katty | i'm still a kid |
20:48.00 | IBC_jkenney | i didn't mean you katty |
20:48.04 | IBC_jkenney | at least you try |
20:48.08 | IBC_jkenney | and are nice about it |
20:48.23 | IBC_jkenney | i'll figure it out |
20:49.26 | Katty | don't tell. i have a reputation to keep up around here. |
20:49.33 | zpertee | :-) |
20:49.49 | IBC_jkenney | oh sorry |
20:49.50 | IBC_jkenney | lol |
20:51.11 | bpgoldsb | Is there a more user-friendly way to view the changes between 1.6.0 and 1.6.1 than the changelog from downloads.digium.com? |
20:51.43 | drmessano | You want a better changelog than the changelog? |
20:52.50 | bpgoldsb | I want an overview of big changes. |
20:53.30 | captiancrash | drmessano, i see a yodawg of that coming soon. |
20:53.49 | drmessano | lol |
20:54.12 | drmessano | bpgoldsb: Those can all be big changes, depending on your definition |
20:55.38 | bpgoldsb | Ya, this (http://svn.digium.com/svn/asterisk/tags/1.6.1.0/CHANGES) works nicely compared to the changelog |
20:57.03 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
20:58.07 | *** join/#asterisk _brent_ (n=_brent_@orem.jiveip.net) |
20:59.20 | *** join/#asterisk t_ (i=tom@freenode/staff/tomaw) |
21:02.00 | Katty | goshdangitanyhow |
21:02.04 | Katty | why am i /always/ friggin hungry |
21:02.05 | Katty | ALWAYS |
21:02.08 | Katty | throws things |
21:02.21 | _brent_ | is anybody here using cdr_custom? i'm getting empty values, like this: |
21:02.22 | _brent_ | ,"","","","","","","","","","","","","","","","","" |
21:02.39 | Katty | looks like a train. |
21:02.42 | Katty | with maybe, snacks. |
21:02.53 | _brent_ | the third car has nachos, if you're interested |
21:03.25 | _brent_ | cdr_custom.conf has |
21:03.25 | _brent_ | Master.csv => "${CDR(clid)}","${CDR(src)}",etc. |
21:03.27 | Linuturk | I feel silly asking this, but I'm trying to use Playback to play back a particular ulaw file on my system. I've got the full path to the file, but asterisk says it can't find the file :( |
21:03.53 | _brent_ | Linuturk: make sure you're not including the .ulaw part of the filename in your command |
21:03.58 | Katty | i would make sure your Playback() does not have an extension |
21:04.11 | Katty | so Playback(/path/to/file.gsm) wouldn't work |
21:05.31 | Katty | Linuturk: if not that you could be looking at permissions and encoding problems |
21:10.50 | _brent_ | with my cdr_custom problems, i can get values like ${EPOCH} to show up, but none of the ${CDR()} variables, any ideas? |
21:10.58 | Linuturk | thanks _brent_ that was the issue |
21:11.01 | Linuturk | silly me |
21:11.15 | Linuturk | and Katty ^^ :) |
21:11.53 | Katty | _brent_: i just use the postgres connection. don't do anything custom |
21:12.30 | _brent_ | i've been using cdr_pgsql for a few years. just trying to measure performance vs flat files, but i've got some custom fields i'm writing currently |
21:13.08 | _brent_ | but i can't get any of the CDR fields to show up in cdr_custom, let alone my custom ones |
21:13.21 | *** join/#asterisk el_critter (n=critter@200.8.97.41) |
21:15.21 | el_critter | Hi, sometimes when I call from mi SIP phone via PSTN trunk and I hang up, it seems like asterisk hangs de SIP device part but the line keeps open until something (haven't been able to determine what) resets it. |
21:25.25 | [TK]D-Fender | _brent_: pastebin your configs and the output of * starting up |
21:26.33 | dustybin | [TK]D-Fender: do you ever feel like focusing on another open-source technology? maybe mail servers? |
21:27.15 | [TK]D-Fender | dustybin: Nope. |
21:27.23 | dustybin | why not? |
21:27.42 | Qwell | only real rock stars get to work with mail servers. |
21:27.43 | [TK]D-Fender | dustybin: I have marginal requirements on other bits, but nothing I felt compelled to go into depth with |
21:27.52 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
21:27.57 | Qwell | like MikeJ here |
21:28.00 | dustybin | in freenode, every channel has there in-house purists :D |
21:28.50 | scalex000 | TK: Hello |
21:28.53 | drmessano | I could care less about telephony |
21:29.03 | Katty | hi mike |
21:29.08 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
21:29.08 | Katty | hugs MikeJ |
21:29.11 | Katty | hugs anthm |
21:29.12 | drmessano | Asterisk is my fav video game.. the fact that I can make calls with it is "ok" I guess |
21:29.23 | MikeJ | hmm |
21:29.25 | MikeJ | runs |
21:29.28 | Katty | :< |
21:29.33 | drmessano | Swine flu? |
21:29.38 | Katty | makes a note to switch deoderants. |
21:30.02 | scalex000 | Tk: which function I need to use to hang up a call when I close the call |
21:30.38 | [TK]D-Fender | scalex000: Show me one failing. And don't just ask blindly without providing backup, and definitely stop targeting individuals for support |
21:30.46 | hardwire | sniffs Katty |
21:31.22 | [TK]D-Fender | snuffs hardwire |
21:31.34 | [TK]D-Fender | goes to dig another ditch |
21:31.41 | hardwire | snubbs [TK]D-Fender |
21:31.51 | scalex000 | ok |
21:31.52 | hardwire | why is it every time I want to talk to TK I start with \[ |
21:31.57 | hardwire | then hit tab a few times and get mad. |
21:31.58 | hardwire | :P |
21:34.08 | scalex000 | TK: when someone call, and hang up before i pickup the phone the call continue ringing |
21:35.54 | Katty | hardwire: do you mind. |
21:36.15 | *** part/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
21:36.16 | *** join/#asterisk umay (n=chris@174-16-21-58.hlrn.qwest.net) |
21:36.19 | Katty | hardwire: i am not a Hot Blonde in a Red Dress |
21:37.07 | leifmadsen | Katty: brunette and blue dress |
21:37.19 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
21:37.20 | outtolunc | was gonna guess green dress |
21:37.38 | leifmadsen | oh wait, I thought we were talking about me |
21:37.55 | _brent_ | [TK]D-Fender: http://pastebin.com/d15b55b3e |
21:38.07 | Katty | well. |
21:38.12 | Katty | i am wearing blue today. |
21:38.24 | Katty | checks for cameras in her office. |
21:38.44 | leifmadsen | Katty: mwahahaha |
21:38.48 | leifmadsen | Katty: is there one in your laptop? |
21:38.53 | *** join/#asterisk okaratas (n=netadmin@fsf/member/okaratas) |
21:38.54 | leifmadsen | because it may or may not be hacked |
21:38.56 | hardwire | this is going strange places. |
21:39.06 | hardwire | shrodingers hack. |
21:39.20 | leifmadsen | strange? #asterisk? wow, you must be new :) |
21:39.26 | Katty | no. |
21:39.47 | [TK]D-Fender | _bAnd what file are you looking in for your CDRs? |
21:40.00 | _brent_ | /var/log/asterisk/cdr-custom/ |
21:40.10 | _brent_ | it's getting appended to, but it's all blank columns |
21:40.16 | _brent_ | "","","",etc |
21:40.32 | anonymouz666 | there's a killer bug in app_queue that report the member as "IN USE" when you hit the DND button of softphone or something like that |
21:40.39 | Katty | replaces [TK]D-Fender's tab button with a tab button! let's see if he notices! |
21:40.45 | anonymouz666 | 1.4.21.2 |
21:40.51 | anonymouz666 | going to update quickly |
21:40.54 | _brent_ | values like ${EPOCH} show up, but none of the ${CDR(foo)} values |
21:41.26 | hardwire | anonymouz666: soooo you want a DND phone to ring? |
21:42.06 | anonymouz666 | no it's a way to simulate the issue |
21:42.23 | hardwire | what do you want it to say? |
21:42.28 | hardwire | AVAILABLE? |
21:42.40 | anonymouz666 | you turn on and off...... |
21:42.52 | *** join/#asterisk obnauticus (n=l@about/windows/regular/obnauticus) |
21:43.09 | anonymouz666 | and you keep "in use" forever until you restart asterisk |
21:43.24 | anonymouz666 | there is no way to delivery a call to this member again |
21:43.26 | *** join/#asterisk puzzled_ (n=foobar@puzzled.xs4all.nl) |
21:43.49 | hardwire | ah |
21:44.18 | hardwire | I've seen something similar but never found a fix. |
21:44.26 | [TK]D-Fender | _brent_: OK, I don't see it... |
21:45.32 | _brent_ | don't see the pastebin or don't see ...? |
21:47.37 | anonymouz666 | hardwire: I didn't search for a fix for that in SVN since my version |
21:47.48 | anonymouz666 | I'll update anyway to latest version (due security issues) |
21:48.00 | anonymouz666 | and then report as a bug if persists |
21:50.27 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-129-215-159.dsl.lsan03.sbcglobal.net) |
21:54.06 | *** join/#asterisk smps (n=maher@193.170.53.51) |
21:57.54 | _brent_ | is puzzled |
22:00.24 | bpgoldsb | if soft hangup <channel> isn't working, is there a way to do a hard kill of a channel from the cli? |
22:02.05 | leifmadsen | bpgoldsb: no, only a restart of the system, or patience will do it -- typically that happens from a dead lock. If you've compiled with MALLOC_DEBUG you should be able to do 'core show locks' |
22:02.48 | bpgoldsb | Hmm. Thats interesting because I'm the only one using this machine |
22:03.05 | bpgoldsb | And I just started asterisk, made 1 call, and an immediate deadlock |
22:03.37 | hardwire | "asterisk" is a reserved syslog word. |
22:03.38 | hardwire | :) |
22:03.59 | bpgoldsb | leifmadsen, do you know if MALLOC_DEBUG has reasons to not be on all the time? |
22:04.08 | bpgoldsb | I imagine performance, but I don't know for sure |
22:04.38 | leifmadsen | bpgoldsb: no reason I can think of. I always enable DONT_OPTIMIZE and MALLOC_DEBUG so that when/if an issue comes up, I have the information available to me to produce a useful bug report |
22:05.06 | bpgoldsb | leifmadsen, MALLOC_DEBUG but not DEBUG_CHANNEL_LOCKS? |
22:05.40 | leifmadsen | I think it's MALLOC_DEBUG to enable 'core show locks' |
22:05.58 | bpgoldsb | Well, it's a testing box, so I'll throw em both on and see what happens. |
22:06.09 | leifmadsen | heh |
22:06.41 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
22:06.51 | hardwire | it feels like a friday |
22:06.52 | hardwire | you know |
22:07.00 | leifmadsen | it so does |
22:07.08 | mbrevda | anyone know of a decent set of english uk promts for asterisk with a voice talent avlible for further work? |
22:07.23 | hardwire | oi! |
22:07.34 | hardwire | I'll do eet! |
22:09.57 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
22:11.24 | hardwire | hi voxter |
22:11.30 | voxter | sup hw |
22:12.04 | hardwire | same ooooold same oooold |
22:12.14 | hardwire | still got fancy glasses? |
22:12.20 | hardwire | thought they were handsome and fancy. |
22:13.09 | hardwire | not sure how to combine those words.. handsomeancy or maybe fansomey. |
22:13.45 | *** join/#asterisk andres833 (n=andres83@190.156.152.225) |
22:13.47 | _brent_ | fandsom |
22:14.23 | dustybin | mbrevda: http://www.enicomms.com/cutglassivr/ |
22:14.38 | mbrevda | dustybin: have you used them befor? |
22:14.46 | dustybin | i use now |
22:15.02 | mbrevda | have you done custom promts? |
22:15.05 | dustybin | not yet |
22:15.10 | dustybin | i might do one day |
22:15.15 | dustybin | i use the asterisk replacements |
22:15.25 | dustybin | she has a nice voice |
22:15.55 | dustybin | anyway, time for bed |
22:15.56 | dustybin | nn |
22:16.14 | mbrevda | thnkx |
22:23.25 | bpgoldsb | leifmadsen, What should I be passing to 'core show locks'? |
22:24.04 | leifmadsen | bpgoldsb: nothing -- it just shows the currently held locks -- it can be useful to a developer when debugging issues with asterisk. Typically if you can't 'soft hangup' a channel it is because there is a lock being held |
22:24.23 | leifmadsen | which is not a problem in itself -- it is when it is a deadlock and it is not released that it becomes an issue |
22:24.33 | bpgoldsb | Oh, I ask because core show locks wasn't giving me anything but a usage statement |
22:24.46 | bpgoldsb | actually, no, it's giving me no locks |
22:24.52 | bpgoldsb | but soft hangup still doesn't work |
22:24.54 | leifmadsen | then there are no locks being held -- that is normal |
22:25.10 | leifmadsen | run it after you've done 'soft hangup' and the channel does not go away |
22:25.16 | bpgoldsb | I did. |
22:25.21 | bpgoldsb | Still no locks. |
22:25.24 | bpgoldsb | Channel still exists. |
22:25.44 | leifmadsen | then it isn't a deadlock |
22:25.54 | bpgoldsb | Yay. |
22:26.37 | bpgoldsb | Mayhaps I should just throw festival integration off my todo. |
22:27.01 | leifmadsen | ya, cepstral tends to be a lot better |
22:27.12 | leifmadsen | I don't even remember the last time I tried festival integration... like... years ago :) |
22:27.20 | bpgoldsb | cepstral? |
22:27.50 | Nugget | the only time I played with festival I found it to be about as high quality as wiring a speak 'n spell toy into the phones. |
22:27.54 | bpgoldsb | Oh, paid for software. |
22:28.12 | Qwell | speak 'n spell <3 |
22:28.16 | bpgoldsb | Hah, well, this is just a wake-up-call system for employees. |
22:28.25 | bpgoldsb | So quality is not important |
22:28.29 | Qwell | considering the hardware is ran on...man, was that impressive tech |
22:28.34 | Qwell | s/is/it/ |
22:34.50 | *** part/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
22:39.26 | Katty | has beef stew and garlic bread! |
22:39.40 | *** join/#asterisk kink0 (n=xchat@86.Red-212-170-176.staticIP.rima-tde.net) |
22:39.43 | kink0 | hello |
22:40.19 | kink0 | quick question, why Dialtones() sounds so bad quality while Background any file sounds fine ? ( H323 channel + g729 ) |
22:41.55 | kink0 | sorry Playtones() |
22:45.18 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
22:45.32 | *** join/#asterisk kerchunk (n=kerchunk@pool-173-49-10-152.phlapa.fios.verizon.net) |
22:50.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
22:52.22 | obnauticus | zrjyer0su6j]ga-y9rzjyrjhyrz9hyjr9pyjzsr |
22:54.06 | obnauticus | errr |
22:54.26 | [TK]D-Fender | H4X |
22:54.41 | obnauticus | lololololololol |
22:56.17 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
22:56.31 | kink0 | anyone has some idea about playtones sounds so choped and distortioned ? ( playing any other audio file sounds ok ) |
22:57.53 | _brent_ | one last try...is anyone out there using cdr_custom that can confirm whether the Master.csv => "${CDR(clid)}","${CDR(src)}",etc. template works? |
23:01.40 | _brent_ | anthm: have you worked on cdr_custom? |
23:02.13 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
23:12.43 | *** join/#asterisk psilikon (n=psilikon@140-1.35-65.tampabay.res.rr.com) |
23:13.47 | psilikon | So I am following CH 5. of the book *: TFOT 2nd Ed. but I can't get any calls from SIP to my asterisk box |
23:14.01 | psilikon | my ITSP is Callcentric |
23:16.50 | psilikon | I am using an ATA with a SIP provider and everytime I place a call to my sip number I watch asterisk give me: Call from '' to extension '1777XXXXXXX' rejected because extension not found. |
23:19.07 | p3nguin | Ah, good ol' CallCentric. |
23:19.34 | p3nguin | This is a common problem with them. Just match your number in the dialplan. |
23:20.05 | psilikon | p3nguin, I can't seem to pull that off for some reason |
23:20.27 | p3nguin | psilikon: Can you show me your context where you get calls inbound from CC? |
23:20.55 | p3nguin | I'm willing to help you fix this, but you'll have to cooperate. |
23:21.27 | psilikon | http://pastebin.com/m794e25be |
23:24.47 | p3nguin | psilikon: What SIP number do you want to ring when someone calls your CC number? |
23:25.09 | psilikon | SIP number as in sip extension? |
23:25.26 | p3nguin | sure |
23:25.37 | p3nguin | a SIP peer that will take the call. |
23:25.37 | psilikon | I have only one phone right now. I call it extension 1000 in my sip.conf |
23:27.11 | p3nguin | http://pastebin.com/d11964568 |
23:27.33 | p3nguin | Use that context for you inbound callcentric calls to SIP/1000. |
23:28.16 | p3nguin | Then go over to sip.conf and make sure that your callcentric context says context=from-callcentric |
23:29.32 | p3nguin | I've made the assumption that your music on hold will play while your phone is ringing, and also that you have your voicemail set up as 1000 in the default context. |
23:29.38 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
23:30.43 | p3nguin | Any troubles taking care of those things? |
23:31.18 | *** join/#asterisk manxpower (n=EWieling@24.42.221.26) |
23:32.55 | psilikon | Nice. It is telling me nonone is available to take the call... but I can probably debug it from here |
23:33.25 | p3nguin | Maybe SIP 1000 isn't registered? |
23:33.44 | psilikon | Unable to create channel of type 'SIP' (cause 20 - Unknown) |
23:33.44 | psilikon | <PROTECTED> |
23:33.50 | p3nguin | erm |
23:34.04 | p3nguin | You reloaded sip and dialplan, right? |
23:34.55 | psilikon | Yep |
23:35.21 | psilikon | My sipura isn't registered.. wtf. I assumed it was by the dialtone |
23:35.23 | p3nguin | I don't know what would cause that message if your phone wasn't actually busy or congested. |
23:35.30 | p3nguin | Ah. |
23:35.51 | p3nguin | Asterisk doesn't provide the dial tone on a SIP device. |
23:36.34 | psilikon | k, now I am reg'd and gonna give it another go |
23:38.10 | psilikon | p3nguin, big help man BIG help! |
23:38.19 | p3nguin | Working? |
23:38.23 | psilikon | yep! |
23:38.44 | p3nguin | Do you get the music as a ringback on the calling phone? |
23:38.49 | psilikon | yes. |
23:39.01 | p3nguin | Should say "connecting your call" then play music until answered or going to vm. |
23:39.30 | p3nguin | You can tweak it from there if you want to change that behavior. |
23:39.47 | p3nguin | or I can help you if you aren't sure how to change something. |
23:40.03 | psilikon | I don't see where the music playing comes in? |
23:40.18 | p3nguin | ,m is music |
23:40.30 | psilikon | oh |
23:40.34 | p3nguin | in the Dial() command. |
23:40.35 | psilikon | 30? |
23:40.40 | psilikon | ms pause? |
23:40.47 | p3nguin | 30 seconds ring timeout |
23:40.50 | psilikon | oh |
23:41.17 | p3nguin | Change it to 3 if you want to make it a challenge to answer the phone before it goes to voicemail. :) |
23:42.42 | *** part/#asterisk _brent_ (n=_brent_@orem.jiveip.net) |
23:43.15 | *** join/#asterisk jcape (n=jcape@adsl-75-21-76-59.dsl.chcgil.sbcglobal.net) |
23:44.04 | p3nguin | Keep in mind that I used the silence/1 and vm-dialout to make it a pleasant experience when calling your number. You can alter the dialplan just about any way you want, as long as it will match your phone number and dial to SIP/1000. |
23:45.07 | psilikon | nice. "m" is just generic music or can you change it? |
23:45.42 | *** join/#asterisk Hadding (n=kyle@207.177.231.9) |
23:46.09 | p3nguin | You can configure your music on hold via the musiconhold.conf file. |
23:46.38 | p3nguin | You could also delete the default music and put in your own if you really wanted to. |
23:46.54 | psilikon | So if I added another xtension under [from-callcentric] that started with "s" it would always play when a call was answered? |
23:47.10 | p3nguin | nope |
23:47.38 | p3nguin | The 's' extension doesn't seem to work with SIP for a technical reason. |
23:47.56 | p3nguin | What are you trying to do? |
23:48.02 | p3nguin | Ring another phone? |
23:49.15 | psilikon | I was just trying to follow the Asterisk book... but actually thanks to you I am past that part anyway. |
23:50.05 | Hadding | http://pastebin.ca/1574601 my extensions.conf |
23:50.32 | Hadding | http://pastebin.ca/1574602 sip.conf |
23:50.52 | manxpower | The "s" extension is matched when Asterisk has NOT RECEIVED the dialed digits. |
23:51.14 | Hadding | help, wher do i put this |
23:51.18 | Hadding | <PROTECTED> |
23:51.20 | Hadding | exten => 33,1,Answer exten => 33,2,AGI(cidspoof.agi) |
23:51.21 | manxpower | When dialing from a SIP phone Asterisk almost always knows the dialed digits (or destination number) |
23:51.32 | manxpower | Hadding: don't flood the channel |
23:51.35 | manxpower | ~pastebin |
23:51.35 | infobot | [~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 |
23:52.03 | manxpower | Hadding: I think you should spend some time with the Asterisk Book |
23:52.04 | manxpower | ~book |
23:52.05 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
23:52.20 | psilikon | That is cool as sh*t. Now I need to come up with something that will allow the caller to press a button and ring my cell |
23:53.20 | manxpower | "s" is used 1) when immediate=yes is set on a zap channel 2) Macros and 3) IVRs |
23:53.23 | p3nguin | hadding: Are you really that huge of a retard? I've told you at least three times and I know that several other people have also told you. |
23:53.43 | manxpower | since all of those situations the concept of "dialed number" is silly |
23:54.07 | manxpower | p3nguin: I think you are starting to understand [TK]D-Fender. 8-) |
23:54.43 | p3nguin | psilikon: Just use followme and let it go automatically. |
23:55.20 | Hadding | p3nguin I dont know were i put it, can you edit the pastebin thing so i know exactly |
23:55.55 | Hadding | Somebody said to put it on pastebin so i did that |
23:55.56 | p3nguin | psilikon: http://pastebin.com/d63850a0e There's one line change I made there to send the call into followme. |
23:55.58 | manxpower | Hadding: download. the. book. These are such basic questions that nobody really wants to answer questions such basic questions.. |
23:56.19 | Hadding | ~book |
23:56.20 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
23:56.34 | p3nguin | It's even written out in PLAIN ENGLISH in "2." |
23:56.44 | p3nguin | "Add the following lines to your extension config file in the same context as your SIP phone." |
23:56.52 | *** join/#asterisk supers (i=supers@bigmatix.com) |
23:56.53 | p3nguin | How much more CLEAR could it be? |
23:57.03 | Hadding | Okay so what context is sip phone in |
23:57.12 | manxpower | p3nguin: some people cannot be helped |
23:57.20 | p3nguin | indeed |
23:57.52 | p3nguin | psilikon: You'll still need to configure followme, though. Let me know when you're ready to go at it. |
23:58.05 | psilikon | Let's do it |
23:58.15 | Hadding | Man I downloaded asterisk vmware and ubuntu for this and i cant get no help |
23:58.25 | p3nguin | psilikon: Open up your followme.conf |
23:58.49 | psilikon | p3nguin, do I still get the cool "please wait while I connect your call + the music"? |
23:58.51 | p3nguin | psilikon: Do you have termination service with CallCentric already? |
23:58.57 | supers | hi there, i'm trying to strip off the first 4 digits before placing a call, would cut be the best function for this? |
23:59.04 | p3nguin | psilikon: It'll work out. You'll see. |
23:59.15 | psilikon | p3nguin, probably not since I don't know what termination service is |
23:59.37 | psilikon | k I am in followme.conf |
23:59.56 | p3nguin | supers: Dial(SIP/${EXTEN:4}@youritspcontext) |