| 00:24.11 | *** join/#asterisk nestAr (i=nester@makes.all.the.girlies.go.wewt.wewt.net) [NETSPLIT VICTIM] |
| 00:30.39 | *** join/#asterisk majost (i=majost@crackaddict.com) |
| 00:38.09 | majost | Has anyone sucessfully backported 1.4 on Ubuntu Feisty? |
| 00:38.22 | majost | I seem to be having some issues with the zaptel driver |
| 00:38.25 | *** join/#asterisk izo (n=ubuntu@cpc3-papw1-0-0-cust666.cmbg.cable.ntl.com) |
| 00:38.59 | izo | hi everybody |
| 00:42.21 | majost | http://pastebin.ca/795343 |
| 00:42.45 | tzafrir | majost, could you be more specific? |
| 00:43.33 | tzafrir | majost, what exactly do you build? a package or asterisk from source? |
| 00:44.10 | tzafrir | a package: there is an exeprimental package on pkg-voip.buildserver.net , I guess |
| 00:44.48 | majost | Well, I pulled the source packages for 1.4.11 from the gutsy pool, along with zaptel-1.4.3, fxload, and libpri |
| 00:45.07 | majost | I managed to build all those packages without incident and install them |
| 00:45.17 | izo | is anyone familiar with app_mwanalyze.c application ? |
| 00:45.38 | tzafrir | chan_zap.c:77:31: error: bristuffed/libpri.h: No such file or directory |
| 00:46.13 | tzafrir | this means that the debian patch for using libpri-bristuffed is applied to Asterisk, but libpri does not have the extra bristuffed copy |
| 00:46.19 | majost | cate libpri.h |
| 00:46.22 | majost | ack |
| 00:46.40 | majost | <PROTECTED> |
| 00:46.58 | majost | ahh |
| 00:47.25 | tzafrir | actually: it looks for /usr/include/bristuffed/libpri.h |
| 00:47.41 | tzafrir | Which should be a separate copy |
| 00:48.31 | majost | tzafrir: do you know what package the bristuffed comes from? |
| 00:49.08 | tzafrir | majost, read what I wrote above |
| 00:49.34 | tzafrir | dpkg -l libpri-dev |
| 00:49.51 | majost | hah |
| 00:49.53 | majost | nm |
| 00:50.13 | majost | they disabled it in the 1.4.0-1 package |
| 00:50.13 | majost | heh |
| 00:50.14 | majost | =P |
| 00:50.40 | tzafrir | majost, try the buildserver.net packages |
| 00:50.57 | majost | ah sweet |
| 00:50.58 | tzafrir | Do you have any zaptel hardware? |
| 00:50.59 | majost | thanks |
| 00:51.02 | majost | =) |
| 00:51.34 | majost | no... I was just trying to build the package without hacking it up |
| 00:53.20 | majost | thanks for pointing me in the right director |
| 00:53.25 | majost | er, direction |
| 00:53.26 | majost | =) |
| 00:54.57 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 00:56.52 | *** join/#asterisk cypherdelic (n=cypher@p5B27EE12.dip.t-dialin.net) |
| 01:00.49 | TJNII | After you change voicemail.conf do you need to reload anything? I don't remember. |
| 01:00.51 | harlequin516 | Um Are the silence/1.gsm through silence/10.gsm files duds? They all seem to be the same file? |
| 01:01.36 | *** join/#asterisk DaveCanoe (n=Dave@belbrrcnas12-3467437087.dial.bell.ca) |
| 01:07.50 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 01:10.25 | ManxPower | they should be 1 second of silence thru 10 seconds of silence. |
| 01:12.15 | ManxPower | TJNII: reload app_voicemail.so should do it. |
| 01:12.41 | TJNII | Cool. Thanks |
| 01:13.03 | TJNII | I wish show modules was in alphabetical order..... |
| 01:20.16 | *** join/#asterisk Speedy2 (n=Javier_6@cpe-66-91-247-165.san.res.rr.com) |
| 01:22.50 | Speedy2 | Hi all. Anyone here working on Asterisk on the Blackfin? |
| 01:23.19 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 01:23.30 | [hC] | digium is |
| 01:23.36 | [hC] | on their asterisk appliance. |
| 01:24.37 | Speedy2 | hrm. |
| 01:27.33 | harlequin516 | Hmm.. Okay I think the silence sound files are duds, in my distribution. (Gentoo) |
| 01:27.47 | harlequin516 | They all have the same filesize |
| 01:29.37 | harlequin516 | Anyone know what Background detect returns if not 0 or -1? |
| 01:31.55 | [TK]D-Fender | harlequin516, play them back and see how long it takes :) |
| 01:32.03 | [TK]D-Fender | harlequin516, and I seriously doubt that. |
| 01:32.15 | harlequin516 | They are all .1 sec according to my player. |
| 01:33.03 | rob0 | Hello darkness my old friend; I've come to talk with you again ... |
| 01:33.06 | harlequin516 | All the files are 165 bytes long |
| 01:35.39 | harlequin516 | [TK]D-Fender : Also they are not silence. There are to peaks in sound, with mostly silence. I wonder if there is a reason for this. I can see the wavform in audacity |
| 01:37.29 | *** join/#asterisk RoyK (n=roy@91.149.31.118) |
| 01:38.16 | *** join/#asterisk mamep (i=fallen@helios.edu.uoc.gr) |
| 01:38.19 | mamep | hello |
| 01:38.32 | mamep | asterisk complains that it cant load a moduke |
| 01:38.43 | mamep | **module |
| 01:47.50 | saftsack | mamep, are you here? |
| 01:48.42 | mamep | fixed it |
| 01:48.44 | mamep | ldd |
| 01:49.03 | mamep | now i;ve managed to install ooh323 |
| 01:49.14 | saftsack | ;) |
| 01:49.23 | saftsack | do you want to use t38modem? |
| 01:49.34 | mamep | openh323_v1_19_0_1 and pwlib |
| 01:49.35 | mamep | nah |
| 01:49.45 | mamep | i want to use h323 gateway to route my calls |
| 01:49.50 | saftsack | ah ok |
| 01:49.51 | mamep | do i have to build |
| 01:50.12 | mamep | and what i have to build now? |
| 01:50.14 | mamep | chan_woomera? |
| 01:50.18 | mamep | h323 or openh323? |
| 01:51.49 | saftsack | dunno i use callweaver |
| 01:52.10 | saftsack | but i would catch the wiki for those purposals |
| 01:54.19 | mamep | callweaver? |
| 01:54.30 | saftsack | yes its an asterisk fork |
| 01:54.32 | *** join/#asterisk Speedy2 (n=Javier_6@cpe-66-91-247-165.san.res.rr.com) |
| 01:54.52 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 01:57.33 | mamep | link |
| 01:57.34 | mamep | ? |
| 02:02.04 | mamep | saftsack : my problem is that i can call someone but wehn he answers the caller's still ringing |
| 02:02.24 | saftsack | i have no experience with openh323, sry :/ |
| 02:07.27 | JT | mamep: google it. |
| 02:07.47 | mamep | jt : tried it no luck |
| 02:07.56 | mamep | tried to move from openh323 o h323 |
| 02:08.10 | JT | callweaver is easy to google |
| 02:08.29 | mamep | ah |
| 02:08.39 | JT | if you want to stop wasting time, just use yate to gateway between h.323 and sip to asterisk |
| 02:08.42 | km- | I need to find a good use for my freenum allocation |
| 02:08.46 | km- | hrm |
| 02:09.03 | km- | voip chatporn! |
| 02:10.47 | mamep | jt : ate? |
| 02:11.09 | mamep | jt : yate? |
| 02:11.47 | mamep | http://yate.null.ro/pmwiki/ |
| 02:11.50 | mamep | intresting |
| 02:14.13 | *** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net) |
| 02:14.57 | UserReg_CL | hi... good day !!! |
| 02:15.08 | UserReg_CL | helpme please... |
| 02:15.28 | JT | mamep: why can't you just google things, instead of repeating back suggestions with question marks? |
| 02:15.34 | file | UserReg_CL: with? |
| 02:15.39 | UserReg_CL | file :) |
| 02:16.30 | UserReg_CL | need what ones call finish after 4 minutes |
| 02:16.40 | UserReg_CL | (sorry by bad english) |
| 02:20.08 | *** join/#asterisk ming_zym (n=ming_zym@124.14.233.233) |
| 02:20.44 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 02:20.44 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 02:22.54 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 02:30.34 | *** join/#asterisk saun (i=saun@89-178-184-147.broadband.corbina.ru) |
| 02:41.00 | *** join/#asterisk BeeBuu (n=chatzill@218.13.82.3) |
| 02:41.08 | BeeBuu | hi,there |
| 02:41.18 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001c100a96f3-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 02:41.34 | DrukenLPY | evening everyone |
| 02:42.38 | _ShrikE | can custom cdr only be posted to .csv, or can it be posted to cdr mysql or odbc as well? |
| 02:43.07 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 02:43.49 | DrukenLPY | anyone feel like giving me a hand with ztdummy? |
| 02:45.47 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 02:46.34 | BeeBuu | can i call another user at diffent sip server? |
| 02:48.30 | *** join/#asterisk GoldFingaZ (n=GoldFing@bas13-ottawa23-1177810360.dsl.bell.ca) |
| 02:55.45 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 02:56.00 | *** part/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 02:58.37 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 03:00.41 | UserReg_CL | mmm |
| 03:04.23 | BeeBuu | ? |
| 03:08.10 | joat | anyone have FWD or can call a 425 area code (IPKall)? I'd like to test the conf call manager that i'm rewriting... |
| 03:12.41 | [TK]D-Fender | BeeBuu, Dial(SIP/someexten@someservershostorip) |
| 03:12.56 | [TK]D-Fender | DrukenLPY, whats up? |
| 03:19.40 | BeeBuu | OK,thanks. |
| 03:32.10 | *** join/#asterisk techie (n=techie@76.214.9.1) |
| 03:39.47 | BeeBuu | [TK]D-Fender: Dial(SIP/someexten@someservershostorip) got "sip/beebuu.com-082187b8 is circuit-busy" |
| 03:40.10 | [TK]D-Fender | BeeBuu, pastebin your complete CLI output. |
| 03:42.15 | BeeBuu | http://pastebin.com/d37013c21 |
| 03:43.55 | [TK]D-Fender | BeeBuu, are you trying to call your own server? |
| 03:44.07 | UserReg_CL | mmm |
| 03:44.09 | BeeBuu | yes. |
| 03:45.08 | [TK]D-Fender | BeeBuu, don't. it is matching the callerid/username 998 BUT because of the way dial is being called no PASSWORD is sent with it. DO not attempt to test it yourself this way |
| 03:45.31 | [TK]D-Fender | BeeBuu, You also need "allowguest=yes" under [general] and you need to set a context. |
| 03:45.35 | *** join/#asterisk Entr4nced (n=IMG001@adsl-70-238-245-18.dsl.bcvloh.sbcglobal.net) |
| 03:45.50 | BeeBuu | in my server side? |
| 03:47.47 | [TK]D-Fender | BeeBuu, what else do you think that could apply to? |
| 03:47.59 | BeeBuu | :-P |
| 03:48.10 | BeeBuu | just make sure. |
| 03:49.52 | *** join/#asterisk DrukenHME (n=jdumais@CPE001c100a96f3-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 03:53.09 | BeeBuu | i put "allowguest=yes" int sip.conf,but still busy... |
| 03:53.21 | florz | can I expect ast_write() to block on full transmit buffer? |
| 03:54.08 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 03:55.01 | [TK]D-Fender | BeeBuu, I told you you will STILL have an auth issue. |
| 03:55.44 | BeeBuu | what's that? |
| 03:55.45 | [TK]D-Fender | BeeBuu, set up another soft-phone that does NOT register to a server or have auth details. The dial the address manually like exten@ip |
| 03:56.24 | BeeBuu | what's your mean? |
| 03:56.36 | BeeBuu | it's beyond me. |
| 03:57.13 | [TK]D-Fender | BeeBuu, I just told you TWICE. you CANNOT use that kind of dial statement to call your own server! |
| 03:57.47 | coppice | of course he can. its a free software! |
| 03:58.03 | BeeBuu | ? |
| 03:58.16 | [TK]D-Fender | coppice, great, like he isn't confused enough as it is... |
| 03:58.30 | BeeBuu | i had two servers with asterisk... |
| 03:58.32 | [TK]D-Fender | BeeBuu, if you want to test your incoming I told you what to do. |
| 03:58.59 | coppice | oh, you want me to be confusing. I promise to do better |
| 03:59.29 | BeeBuu | [TK]D-Fender: please |
| 03:59.38 | [TK]D-Fender | <[TK]D-Fender> BeeBuu, set up another soft-phone that does NOT register to a server or have auth details. The dial the address manually like exten@ip |
| 03:59.41 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 04:00.31 | BeeBuu | does not register to a server? |
| 04:00.37 | [TK]D-Fender | BeeBuu, correct. |
| 04:00.52 | dr1 | have you guys met your bonehead quota of the day, or can i sneak in a crappy question without invoking your wrath? hehe |
| 04:01.00 | BeeBuu | so how can i call without register to a server? |
| 04:01.11 | [TK]D-Fender | <[TK]D-Fender> BeeBuu, set up another soft-phone that does NOT register to a server or have auth details. The dial the address manually like exten@ip |
| 04:01.22 | [TK]D-Fender | BeeBuu, READ ^^^^^^^^^^^^^ |
| 04:01.37 | [TK]D-Fender | dr1, just ask |
| 04:01.39 | dr1 | ill ask anyways -- how do you get a damn softphone to make its calls through the asterisk server ? |
| 04:02.06 | [TK]D-Fender | dr1, setup a user on * and setup the auth on your softphone. |
| 04:02.11 | BeeBuu | [TK]D-Fender: i don't know how to manually,i'm a very new new newbie. |
| 04:02.34 | *** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net) |
| 04:02.50 | [TK]D-Fender | BeeBuu, Set up a new softphone and DON'T set up the account. then have it dial "exten@serverip". THE END. |
| 04:03.02 | BeeBuu | OK,i try |
| 04:03.05 | dr1 | I'm trying to get ekiga todo that, its registered with an extension... it just wants to send all outgoing traffic directly, even on incoming calls from what I can tell |
| 04:03.27 | [TK]D-Fender | dr1, Huh?! |
| 04:03.55 | [TK]D-Fender | dr1, "just wants to send all outgoing traffic directly " What traffic? From where? TO where? How? |
| 04:04.09 | dr1 | I mean for example if I try to make a call outside my network |
| 04:04.23 | BeeBuu | [TK]D-Fender: i'm using X-lite,how to do that? |
| 04:04.23 | [TK]D-Fender | dr1, Get SPECIFIC, FAST. |
| 04:04.41 | [TK]D-Fender | BeeBuu, What don't you get. I said instal ANOTHER soft phone. then DON |
| 04:04.46 | [TK]D-Fender | 'T CONFIGURE AN ACCOUNT. |
| 04:05.00 | [TK]D-Fender | BeeBuu, and I told you EXACTLY how to dial it. |
| 04:05.04 | BeeBuu | not X-lite? |
| 04:05.27 | [TK]D-Fender | BeeBuu, Your X-lite is REGISTERING as a user right now. FOREGET abou that one. |
| 04:05.47 | BeeBuu | i disconnected the X-lite. |
| 04:05.55 | dr1 | anything@anywhere, it doesnt try to route the call through the asterisk server, I dont know if its my softphone or im misunderstanding how it all works |
| 04:06.20 | [TK]D-Fender | dr1, We need real details and that doesn't make anything clearer at all. |
| 04:06.23 | BeeBuu | and what's the next step? |
| 04:06.42 | [TK]D-Fender | BeeBuu, I just gave you ever step 3 times. |
| 04:06.46 | dr1 | I dont know what other details I can give, that is my problem :P |
| 04:07.07 | BeeBuu | [TK]D-Fender: sorry about my foolish... |
| 04:07.13 | [TK]D-Fender | dr1, where EXACTLY is this call coming in from? |
| 04:08.51 | dr1 | friend on the internet using ekiga to make a call to me, it comes through and then i get 1 way communication |
| 04:09.32 | [TK]D-Fender | dr1, pastebin the complete CLI output of a failed call |
| 04:09.34 | [TK]D-Fender | ~pb |
| 04:09.35 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 04:09.36 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 04:09.45 | [TK]D-Fender | dr1, "set verbose 10" |
| 04:09.51 | [TK]D-Fender | dr1, "sip debug" |
| 04:09.57 | [TK]D-Fender | dr1, do those first |
| 04:10.05 | [TK]D-Fender | dr1, then pastebin the whole mess. |
| 04:10.19 | dr1 | I'll have to go fire up the server then, I shut it down yesterday was tired of beating my head on the wall :P |
| 04:10.31 | [TK]D-Fender | ......... |
| 04:10.47 | [TK]D-Fender | yup, its the weekend and all the crazies have come out. |
| 04:10.47 | dr1 | was going to wait till i found another softphone to test with |
| 04:11.03 | dr1 | let me just ask this |
| 04:11.10 | [TK]D-Fender | dr1 : and first pastebin your sip.conf masking only passwords |
| 04:11.17 | dr1 | if you register with the asterisk server, is your softphone supposed to route its calls through that server? |
| 04:11.57 | [TK]D-Fender | dr1, depends on the soft-phone. If you don't include something like "@hostorip" then typically yes it should send the call to the server its registered to |
| 04:12.18 | dr1 | alright because i was tcpdump'ing the whole deal, it didnt talk to the server what so ever |
| 04:12.57 | dr1 | the only options for SIP in ekiga are outbound proxy and forward url |
| 04:13.20 | [TK]D-Fender | dr1, I've asked you for backup. start providing it and describe the networking path between * and Ekiga |
| 04:13.27 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 04:15.35 | dr1 | i just noticed you said @hostorip :P |
| 04:16.48 | coppice | Host-o-rip sounds like a perfect name for the typical hosting company |
| 04:18.33 | [TK]D-Fender | coppice, Its also an anagram for "poor shit" :) |
| 04:18.47 | [TK]D-Fender | coppice, curiously appropriate ;) |
| 04:19.01 | coppice | crossword fetishist? |
| 04:19.31 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 04:19.32 | [TK]D-Fender | coppice, No, this one just sort of occured to me. This is actually the kind of puzzle I suck at, but it sorta popped out |
| 04:19.50 | dr1 | I wasnt having a prob with calls to other extensions, thats what i was asking 'anyone@anywhere' ;0 |
| 04:19.56 | *** join/#asterisk jameswf-home (n=that@ip72-204-228-104.ph.ph.cox.net) |
| 04:20.03 | dr1 | didnt know how the phone handled that |
| 04:20.35 | coppice | hostorip.com is available. should we grab it and make our fortunes |
| 04:20.57 | jameswf-home | jbot: wise is A wise man can learn more from a foolish question than a fool can learn from a wise answer |
| 04:20.57 | jbot | okay, jameswf-home |
| 04:21.36 | *** part/#asterisk jiffe3 (n=jiffe@67.158.2.241) |
| 04:21.59 | coppice | jbot: pretentious is see jameswf-home |
| 04:22.00 | jbot | coppice: okay |
| 04:22.57 | jameswf-home | jbot: pretentious |
| 04:22.57 | jbot | [pretentious] see jameswf-home |
| 04:23.03 | jameswf-home | sweet |
| 04:23.06 | jameswf-home | lol |
| 04:24.07 | [TK]D-Fender | ~carlin |
| 04:24.07 | jbot | carlin is, like, [~carlin] "Just think of how stupid the average person is, and then realize half of them are even stupider!" - George Carlin. |
| 04:24.33 | dr1 | <3 freenode |
| 04:24.34 | [TK]D-Fender | :) |
| 04:24.45 | [TK]D-Fender | ~carlin |
| 04:24.46 | jbot | [~carlin] "Just think of how stupid the average person is, and then realize half of them are even stupider!" - George Carlin. |
| 04:24.48 | [TK]D-Fender | better |
| 04:24.57 | [TK]D-Fender | ~[TK]D-Fender |
| 04:24.57 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 04:25.02 | [TK]D-Fender | ~jbot |
| 04:25.03 | jbot | rumour has it, jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 04:25.13 | [TK]D-Fender | :O |
| 04:25.23 | [TK]D-Fender | And for reference I didn't add any of that :) |
| 04:25.43 | jameswf-home | ~jameswf-home |
| 04:26.51 | ido | ls |
| 04:26.55 | ido | ls -lah |
| 04:27.00 | ido | ifconfig |
| 04:27.07 | ido | ifconfig motherfucker |
| 04:27.12 | ido | oops |
| 04:27.17 | coppice | ls -lah must be the Singapore edition |
| 04:27.34 | ido | sorry, it's GNU/ls :P |
| 04:28.05 | coppice | if you lived in asia you'd get that |
| 04:28.17 | ido | alas, i do not. |
| 04:28.55 | jameswf-home | who is asia and how does one live inside |
| 04:29.18 | jameswf-home | Let it never be said that your anal-retentive attention to detail never yielded positive results! |
| 04:29.22 | *** join/#asterisk rycar (n=rycar@rrcs-24-199-36-210.west.biz.rr.com) |
| 04:29.40 | coppice | Gee, Google can find you information on anything :-) http://links.jstor.org/sici?sici=0029-8115(199406)33%3A1%3C145%3ATMOLU%22%3E2.0.CO%3B2-U |
| 04:30.32 | jameswf-home | for a fee |
| 04:30.38 | [TK]D-Fender | jameswf-home, http://en.wikipedia.org/wiki/Asia_(band) |
| 04:31.03 | ido | hey coppice: privmsg me |
| 04:31.05 | [TK]D-Fender | jameswf-home, http://www.asiacarrera.com/ ? |
| 04:31.22 | jameswf-home | jbot: wiki asia |
| 04:31.25 | ido | i have jstor access :) |
| 04:31.40 | jameswf-home | jbot: wiki lah |
| 04:32.18 | dr1 | whats considered a 'client' when talking about asterisk? I think this is where I misunderstood :P |
| 04:32.27 | [TK]D-Fender | jameswf-home, that can't be right. Aluminum is "al" |
| 04:32.43 | jameswf-home | lol blame the bot |
| 04:32.45 | [TK]D-Fender | jameswf-home, an Lithium "Li". |
| 04:32.49 | coppice | dr1: client means a sucker who will pay for free software |
| 04:32.51 | Qwell | [TK]D-Fender: hence the lialh |
| 04:32.57 | [TK]D-Fender | dr1, your soft-phone. |
| 04:33.16 | dr1 | yeah my brain was reading that as a caller for some reason |
| 04:33.28 | jameswf-home | jbot: wtf lah |
| 04:33.53 | dr1 | that little understanding couldve saved alot of time :P |
| 04:34.21 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 04:35.52 | d-k-t | LiAlH4 |
| 04:35.59 | ido | i thought a client was any old piece of junk that pokes your server over IAX or SIP or somesuch |
| 04:37.34 | jameswf-home | ~client |
| 04:37.45 | dr1 | I was trying to avoid setting up a proxy since I didnt think I actually needed too |
| 04:39.18 | [TK]D-Fender | ~h2so4 |
| 04:39.18 | jbot | [~H2SO4] "John was here but s no more, for what he thought was H2O was H2SO4" |
| 04:40.04 | [TK]D-Fender | ido, correct |
| 04:43.40 | Qwell | s no more? What, did he jump to i? |
| 04:44.52 | [TK]D-Fender | Qwell, fixed the moment I read it :) |
| 04:46.29 | [TK]D-Fender | 3 weeks to cell-upgrade.... |
| 04:46.35 | [TK]D-Fender | :/ too long! |
| 04:47.12 | Qwell | t-mobile does 11 month increments |
| 04:47.22 | Qwell | 11 or 22, depending on contract length :D |
| 04:47.30 | Qwell | pretty nice |
| 04:48.22 | coppice | [TK]D-Fender: the monastery is refurbishing your cell? :-\ |
| 04:48.53 | [TK]D-Fender | coppice, No, keeping as backup. |
| 04:49.06 | *** join/#asterisk [zender] (n=ender@72.52.23.3) |
| 04:49.24 | [TK]D-Fender | coppice, the shackles are much better in the new one... or so Mistress says |
| 04:51.29 | coppice | so, you're getting a spanking new place? |
| 04:53.56 | jameswf-home | herpies in the eye how crappy |
| 04:54.10 | [TK]D-Fender | ~skullfuck |
| 04:54.10 | jbot | extra, extra, read all about it, skullfuck is http://www.youtube.com/watch?v=Vav6b5F-_64 |
| 04:54.17 | [TK]D-Fender | :O |
| 05:00.33 | Qwell | umm |
| 05:00.35 | Qwell | I'm scared |
| 05:00.51 | Qwell | oh, meh |
| 05:04.08 | [TK]D-Fender | heheh |
| 05:05.00 | coppice | that one is just weird. the onion rises to special ocassions, but in between just goes plain weird |
| 05:05.54 | [TK]D-Fender | coppice, I love what a professionqal job they do of it.... |
| 05:05.58 | coppice | oh, sure. the production quality is good. pity the wit doesn't always match it |
| 05:06.38 | *** join/#asterisk bintut (n=bintut@cm246.gamma178.maxonline.com.sg) |
| 05:07.49 | jameswf-home | i got my new dvd camcorder from frys, almost got killed today |
| 05:08.19 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 05:08.40 | MrTelephone | any of you experience carrier errors on your telco side t1 interface? |
| 05:10.16 | coppice | [TK]D-Fender: the onion is usually really good with disasters, e.g. http://www.theonion.com/content/video/how_can_we_raise_awareness_in |
| 05:25.42 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 05:29.05 | MrTelephone | probably just a bad cable or something |
| 05:30.54 | *** join/#asterisk [zender] (n=ender@72.52.23.3) |
| 05:33.33 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-204-178.dsl.irvnca.pacbell.net) |
| 05:33.36 | UnixDog | hey guys |
| 05:33.51 | UnixDog | how do I lock asterisk to allow from only 1 ip |
| 05:34.05 | UnixDog | I need to lock a box to 1 client only |
| 05:34.27 | UnixDog | they only send via anonsip not full auth |
| 05:34.42 | UnixDog | is it fromhost=ip ? |
| 05:35.34 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:36.15 | *** join/#asterisk zerocod3r (n=Z3R3CoD3@b207d11.dorm.bilkent.edu.tr) |
| 05:37.08 | *** join/#asterisk Op3r (n=Op3r@203.177.173.151) |
| 05:38.33 | zerocod3r | hellow, Something make me confuse I have a digium card with 4 fxo module on it(analog) but didnt understand how can reach voip provider via those modules. |
| 05:39.34 | UnixDog | you doont |
| 05:39.46 | UnixDog | fxo is for had lines |
| 05:39.57 | d-k-t | zerocod3r, you would access a voip provider over IP |
| 05:39.57 | UnixDog | you need a sip trunk for sip provider |
| 05:40.21 | *** join/#asterisk mihinomenest (i=tXNc@66.255.220.17) |
| 05:40.28 | zerocod3r | him now its become more clear so that card let people reach me via analog phones isnt it? |
| 05:41.10 | UnixDog | its connects to the hard lines coming from the phone company |
| 05:41.27 | UnixDog | and allows for those callers to call into your system yes |
| 05:41.34 | [TK]D-Fender | UnixDog, "host=" |
| 05:41.49 | d-k-t | zerocod3r, if you have 4 phone lines from the phone company that you want to connect to your asterisk server, that card will do it for you |
| 05:42.04 | ManxPower | permit= / deny= would do it. |
| 05:42.06 | d-k-t | zerocod3r, if you want to make calls using a voip provider, that card isn't needed |
| 05:42.09 | UnixDog | host=ip,of.the.remote is not working |
| 05:42.27 | d-k-t | zerocod3r, if you want to receive calls using a voip provider, that card isn't needed |
| 05:42.31 | ManxPower | host= is (mostly) for OUTGOING connections |
| 05:42.43 | UnixDog | this is for incoming |
| 05:42.51 | UnixDog | from one server |
| 05:42.51 | d-k-t | zerocod3r, please try to explain what it is that you are trying to do in some more detail |
| 05:43.03 | UnixDog | so use permit and the ip |
| 05:43.07 | UnixDog | ? |
| 05:43.19 | zerocod3r | d-k-t if I am receiving calls via voip provider can I still use asterisk for call forwarding or other pbx related issues? |
| 05:43.53 | UnixDog | turn off allowguest and put permit=ip |
| 05:43.58 | d-k-t | zerocod3r, however your receive calls, you can still use asterisk to do all pbx related functions |
| 05:44.04 | UnixDog | Manx |
| 05:44.47 | zerocod3r | d-k-t I am trying to learn asterisk before that worked with dialogic cards which has no voip capabilities |
| 05:45.18 | [TK]D-Fender | zerocod3r, I think you misunderstand things. Those cards have NOTHING to do with VoiP. |
| 05:45.29 | zerocod3r | while searching internet see asterisk and read a bit and now want to make voip call center but the thing I didnt understand do I need a digium card to do it? |
| 05:45.31 | [TK]D-Fender | zerocod3r, They let Asterisk use your LINES. |
| 05:45.32 | d-k-t | zerocod3r, ok, well the voip functionality is in asterisk... you don't need any cards for that |
| 05:45.56 | zerocod3r | because my voip provider forwards call to me all I want to do id to diverse them our voip phones |
| 05:45.56 | [TK]D-Fender | zerocod3r, You only needed that card if you intended on using analog lines from your telco. |
| 05:46.06 | zerocod3r | ok got it thanks |
| 05:46.16 | [TK]D-Fender | zerocod3r, If you DON'T plan on using analog line from your telco then you bought a card for nothing |
| 05:46.17 | zerocod3r | sorry for asking dummy questions it really helped me |
| 05:46.40 | [TK]D-Fender | zerocod3r, Wish we could have answered these questions BEFORE you got your card. Hope you can return it for a refund |
| 05:48.23 | zerocod3r | card doesnt matter by the way the people sold me card sad we programmed fxo modules |
| 05:48.27 | bintut | gtg.. |
| 05:48.49 | zerocod3r | is it possible, do they need to program fxo modules for every different country? |
| 05:49.14 | d-k-t | zerocod3r, when you load the driver for the card, you specify the country |
| 05:50.31 | zerocod3r | ok because every country has different analog settings but the sales man sad me its our modules.They add their company names ass prefix to the fxo modules such as blah-blah fxo |
| 05:50.39 | *** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 05:51.11 | zerocod3r | I wonder is there any difference between the modules they sold me and sold in digium web site |
| 05:51.54 | d-k-t | zerocod3r, where did you buy them? the digium modules don't need any preconfiguration before they are shipped, you just set them up in the system |
| 05:52.50 | zerocod3r | http://www.empatiq.com/products.html |
| 05:53.07 | zerocod3r | at left bottom corner it says empatique fxo modules |
| 05:53.57 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 05:53.57 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 05:54.19 | zerocod3r | is it something different or just a selling tactic |
| 05:55.21 | d-k-t | zerocod3r, not sure, it doesn't link to any actual information on their website |
| 05:55.29 | d-k-t | zerocod3r, they could have made clone modules |
| 05:56.23 | zerocod3r | you mean fake modules? |
| 05:57.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 05:58.26 | [TK]D-Fender | zerocod3r, not "fake", just made by someone else to match the card. Quality can vary widely. |
| 05:58.35 | zerocod3r | one more question I downloaded a book called Asterisk Hacking and no chance to reach asterisk programming books except Amazon. Where I have to start? |
| 05:58.43 | [TK]D-Fender | zerocod3r, And then ther is a question of warranty |
| 05:58.56 | [TK]D-Fender | ~book |
| 05:58.57 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 05:58.58 | [TK]D-Fender | ^^^^ |
| 05:59.58 | *** join/#asterisk asdx (n=diego@adsl-151-210.click.com.py) |
| 06:00.01 | zerocod3r | himm ok.thanks a lot really appreciate your help |
| 06:00.15 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
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| 06:58.28 | JimmyDee | ok can someone explain why I would get a busy signal when I dial my sip dialup number with my cellphone and asterisk is running? |
| 06:59.55 | *** join/#asterisk Op3r (n=edwin@203.177.173.151) |
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| 08:22.57 | mkl1525 | Hi, palying with extension h - thought that this would be called when Dial has finished but looking at my NoOps it is executed directly after Dial connects. Is this expected behavior or is there something wrong in my dialplan/setup? |
| 08:24.13 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 08:27.02 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 08:31.07 | _x86_ | hey i have a varible-length wav file i want to play, and Playback and Background dont want to play it |
| 08:31.19 | _x86_ | exten => 1,1,System(mplayer -ao pcm:/tmp/foo.wav bar.avi &) |
| 08:31.24 | _x86_ | 1,n,Playback(/tmp/foo) |
| 08:31.40 | _x86_ | where /tmp/foo.wav is growing as the mplayer command is running |
| 08:31.48 | _x86_ | basically i want to stream the wav to the phone |
| 08:31.56 | _x86_ | how would i do this? |
| 08:37.22 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 08:38.52 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.71) |
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| 08:50.21 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 08:53.41 | CBU[^_^]M`` | hello |
| 08:56.13 | CBU[^_^]M`` | how do i load a *.c file to my asterisk? |
| 08:57.37 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 09:05.30 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
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| 09:18.17 | tzafrir_home | CBU[^_^]M``, you generate a .so module and load it |
| 09:21.18 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
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| 09:28.00 | ido | ugh |
| 09:28.12 | ido | this "security researcher" guy who developed SIPtap is retarded. |
| 09:28.24 | ido | "HAY GUYS YOUR COMPUTER IS BROADCASTING AN IP!" |
| 09:35.53 | *** part/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:13.32 | *** join/#asterisk RoyK (n=roy@ip-182-27-149-91.dialup.ice.no) |
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| 10:35.43 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 10:57.51 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:02.17 | *** join/#asterisk Tili (n=tili@203.170.74.196) |
| 11:02.36 | Tili | has anyone used 301 redirect of asterisk? |
| 11:03.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:27.32 | obnauticus | Can I playback soemthign to a meetme conference? |
| 11:30.08 | Greek-Boy | while we're on the topic of meetme. I was wondering what the channel limit is on meetme? How many users in a conference room? |
| 11:36.54 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 11:37.43 | Tili | obnauticus: u can probably play in meetme by adding a user which would playback a sound and then hangup |
| 11:37.58 | Tili | but it wud mess if u have room/pin enabled |
| 11:38.15 | Tili | u have to code it so it doesn't request and doesnt request name also |
| 11:38.55 | duki | Hello again, |
| 11:39.06 | duki | I am enough confused, |
| 11:39.26 | duki | what is the diffence between a user and a peer? |
| 11:39.46 | duki | wahtever the porotocol. |
| 11:39.56 | duki | *whatever. |
| 11:40.21 | obnauticus | Tili, how do I do tha though? |
| 11:40.30 | obnauticus | Well Tili, i can originate an extension into there |
| 11:40.33 | obnauticus | and it doesn't say anything |
| 11:40.37 | obnauticus | like it doesn't ask for it's name it just joins it |
| 11:40.57 | obnauticus | when i do SIP/*6 application Meetme 1 |
| 11:41.56 | Tili | use local channel and use .call file in /var/spool/asterisk/outgoing |
| 11:42.02 | Tili | it is a bit tricky but i think u can do it |
| 11:42.37 | obnauticus | How :\ |
| 11:42.40 | obnauticus | I don't know how ! |
| 11:42.53 | *** join/#asterisk zerocod3r (n=Z3R3CoD3@b206d149.dorm.bilkent.edu.tr) |
| 11:43.51 | zerocod3r | I installed trixbox and write my first ivr application but the problem is how do I connect and test it? is there any way to connect it without using voip provider or analog phone line only via internet? |
| 11:47.53 | obnauticus | <PROTECTED> |
| 11:47.53 | obnauticus | ROFL |
| 11:52.13 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:55.17 | tzafrir_home | zerocod3r, originate SIP/your-phone extension number@pstn-context |
| 11:56.06 | tzafrir_home | zerocod3r, luckily for you, those who wrote FreePBX already thought of that. If you enable the module "test" (or whatever) and dial 7777 (IIRC) that is exactly what you'll get |
| 12:01.10 | *** join/#asterisk PepOSX (n=pepOSX@190.72.145.179) |
| 12:01.44 | zerocod3r | how can I dial it ;) softphone?can I call it ip based? |
| 12:02.10 | *** join/#asterisk knarfly (n=vladimir@c-75-74-155-198.hsd1.fl.comcast.net) |
| 12:02.29 | tzafrir_home | zerocod3r, or use a microphone and speaker with the console phone |
| 12:02.56 | tzafrir_home | zerocod3r, if you have questions about freepbx, ask in #freepbx |
| 12:03.05 | zerocod3r | ok I got it tzafrir_home. will try |
| 12:03.08 | tzafrir_home | (e.g: how do I add a SIP phone in FreePBX) |
| 12:04.48 | *** join/#asterisk icewaterman (n=immagine@i577B83BF.versanet.de) |
| 12:05.49 | icewaterman | hi, i've got an idea for a feature of asterisk (some sort of anti-spam-call-protection for both isdn and voip). is there already such a plugin/module for it? |
| 12:08.43 | icewaterman | to prevent automated calls is easy, just have some sort of voicemail in advance take the call and ask the user to press a random number (0-9). lets say that random number is 3 for the call the user has to press 3 in order to get the telephone ringing. in case he presses the wrong number the call is disconnected. voice recognition is not that good that a dialing bot would get past it with more than a 10% chance each call. |
| 12:09.20 | icewaterman | and that would assume the bot would try voice recognition for that question at all. |
| 12:09.41 | icewaterman | after 10-20s with no input the call is disconnected as well. |
| 12:11.00 | tzafrir_home | icewaterman, that is a very unfriendly feature |
| 12:11.03 | tzafrir_home | I hate it |
| 12:11.13 | tzafrir_home | (asking to press a random number) |
| 12:11.27 | icewaterman | tzafrir_home: the user should not press a random number |
| 12:11.34 | tzafrir_home | you can blacklist by callerid |
| 12:11.57 | icewaterman | but the question: press $random number with $random a random, but fixed number. |
| 12:12.12 | icewaterman | tzafrir_home: you cannot at home when the caller supresses his callerid |
| 12:13.19 | icewaterman | tzafrir_home: i'd never ask a user: " please press a random number between 0 and 9" but rather please press number 3 with the 3 beeing randomly generated for each call. |
| 12:13.41 | icewaterman | next call it might be 7 or any other number from [0-9] |
| 12:18.10 | icewaterman | and if the callerid is suppressed he will get an additional notice that the call will be recorded. (if he doesnt want that he doesnt have to press the number and therefore his call will not get recorded at all (because it will not be taken). |
| 12:19.20 | tzafrir_home | icewaterman, you mean people actually listen to those instructions? |
| 12:19.36 | icewaterman | no illegal-cold-call will make it through that, because it will be recorded and can be used to sue whomever called (wether or not the call will be recorded at all and wether you will actually sue someone for illegally calling you with useless ads is not your problem. |
| 12:19.46 | tzafrir_home | oh, and when they call twice, they expect to get to the same number, right? |
| 12:19.58 | icewaterman | tzafrir_home: no of course not. |
| 12:20.05 | icewaterman | random number each time. 1-10. |
| 12:20.16 | tzafrir_home | very annoying |
| 12:20.17 | icewaterman | 1 sentence, takes about 10s or less. |
| 12:21.05 | tzafrir_home | after you finally managed to call |
| 12:21.05 | icewaterman | tzafrir_home: in any case it takes 10s per call. for those with no callerid it will take 5 s longer |
| 12:21.05 | tzafrir_home | and before you actually get what you want |
| 12:21.06 | icewaterman | tzafrir_home: this is to protect end-users not companies. |
| 12:21.10 | tzafrir_home | I would stay away from such an IVR |
| 12:21.48 | icewaterman | tzafrir_home: with those cold-calls and other unwanted advertisements people will prefer it if they suffer from >10 such calls a day |
| 12:22.43 | icewaterman | tzafrir_home: as it can be configured, some people may only want to use it for calls without callerid etc. |
| 12:23.47 | icewaterman | and people who suppress their callerid and want me to take their call might have to go through that little trouble of theirs (that is pressing a number) |
| 12:25.58 | icewaterman | icewaterman: best is that you do not need to record the calls at all, as the other side would not go for such terms in case of illegal cold calls (taking the risk of getting sued). and if they really do, you could implement something like a confirmation of the user taking the call, whether he wants that recording to be kept or wiped (with default beeing wiped). |
| 12:28.21 | icewaterman | so after the other side has terminated the connection of a call (that was actually put through to you), you'd get asked by asterisk, to press 1 to keep the call or else it would not be saved at all). you can always ignore that and only in case you want to keep it you can do so. |
| 12:29.14 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 12:29.23 | icewaterman | that makes 10-15s for a user calling and 5 additional seconds for the user in case of an unwanted call (that made it though). |
| 12:30.01 | icewaterman | and for voip you can apply the blacklisting and domain verification stuff used for spam. |
| 12:35.39 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
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| 12:51.11 | zerocod3r | I wonder can asterisk call customer for me and put them through our representative? If it can is there any module let it do that? |
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| 14:26.45 | phix | hihihihihi |
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| 15:12.30 | *** join/#asterisk redax (i=redax@r6.hu) |
| 15:12.33 | redax | hi |
| 15:13.13 | redax | just upgraded my asterisk 1.2.x to 1.4.14 and I have problems with misdn : P[ 1] empty_chan_in_stack: cannot empty channel 255 |
| 15:13.28 | redax | do you have any idea? misdn version is 1.1.7 |
| 15:15.37 | *** join/#asterisk blitzrage (n=lmadsen@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:15.37 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:18.35 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:23.39 | *** join/#asterisk ManxPower (n=manxpowe@97.81.241.169) |
| 15:34.10 | redax | hm. is the old asterisk (1.2.x) not downloadable ? |
| 15:34.19 | *** join/#asterisk newvn (n=newvn@203.190.173.8) |
| 15:36.18 | *** join/#asterisk PepOSX (n=pepOSX@190.72.145.179) |
| 15:37.14 | ManxPower | redax: http://downloads.digium.com/pub/asterisk/old-releases/ |
| 15:38.30 | redax | empty dir :-( |
| 15:38.48 | redax | heh |
| 15:38.53 | redax | stupid lynx |
| 15:39.56 | *** join/#asterisk edistar (n=edistar@ip503ddd09.speed.planet.nl) |
| 15:40.04 | redax | if I "click" on old-releases it's results an empty dir |
| 15:40.25 | redax | if I point to the old-releases/ directly it lists the dir |
| 15:40.34 | redax | anyway thanks ManxPower; |
| 15:40.56 | edistar | hi, can anyone recommend a good asterisk book for a beginner in telephony systems? |
| 15:42.05 | Corydon76-dig | ~thebook |
| 15:42.05 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 15:42.31 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 15:45.04 | edistar | thx Corydon76-dig |
| 15:45.31 | Corydon76-dig | Buy a copy and support the authors... |
| 15:45.45 | edistar | Corydon76-dig: I will |
| 15:46.22 | edistar | although 44 euro is quite a lot.. |
| 15:46.41 | Corydon76-dig | It's a big book. Much thicker than the first edition |
| 15:46.52 | edistar | ok |
| 15:47.30 | *** part/#asterisk edistar (n=edistar@ip503ddd09.speed.planet.nl) |
| 15:55.23 | redax | hm. no way to get asterisk 1.4.14 and misdn 1.1.7 working |
| 15:55.30 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 16:13.47 | *** join/#asterisk bantu (n=Miranda@p54A31F2C.dip0.t-ipconnect.de) |
| 16:14.02 | ManxPower | redax: you really need a non-links non-lynx web browser because of stupid http redirection |
| 16:15.53 | mamep | chan_h323.c:850 oh323_write: Asked to transmit frame type 4, while native formats is 0 |
| 16:16.03 | mamep | which codevc is foramt 0? |
| 16:17.12 | *** join/#asterisk galeras (n=galeras@200.31.204.42) |
| 16:17.13 | ManxPower | mamep: "show formats" |
| 16:17.25 | ManxPower | or "show codecs", I don't recall. |
| 16:22.49 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 16:27.05 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
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| 16:33.37 | galeras | Regarding rrmemory ring strategy: When an agent miss a turn because he were buzy, must he to wait the next round to get a call? |
| 16:37.30 | [TK]D-Fender | galeras, Yes, thats the point |
| 16:41.51 | tobias | i have a very simple asterisk config with no hardware |
| 16:41.57 | tobias | i can dial other users and talk to them fine |
| 16:42.19 | tobias | but any time asterisk is responsible for playing back a recording i just get silence |
| 16:42.26 | tobias | it used to work fine |
| 16:42.31 | tobias | and i can't figure out what's changed |
| 16:43.08 | [TK]D-Fender | tobias, Where is * located relative to your phone? |
| 16:43.29 | tobias | across the internet |
| 16:45.17 | [TK]D-Fender | tobias, please provider complete details of all the networking equipment used on BOTH sides. |
| 16:46.51 | tobias | [TK]D-Fender: the asterisk host is the DMZ host for a LAN. i have the following set in sip.conf: externip = xx.xx.xx.xx, localnet = 10.10.0.0/255.255.255.0 |
| 16:47.03 | RoyK | [17:41] <tobias> i have a very simple asterisk config with no hardware |
| 16:47.04 | tobias | where xx.xx.xx.xx is the actual public IP of the router |
| 16:47.22 | [TK]D-Fender | tobias, You are missing a few settings. Read this : |
| 16:47.23 | [TK]D-Fender | ~sipnat |
| 16:47.26 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:47.30 | ManxPower | tobias: and is 10.10.0.0 the local network that the ASTERISK server is on, or that your SIP client is on? |
| 16:47.30 | tobias | RoyK: telephony hardware :p |
| 16:47.35 | RoyK | tobias: :) |
| 16:48.16 | tobias | ManxPower: the former |
| 16:48.31 | ManxPower | tobias: good. |
| 16:49.31 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:49.31 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:52.55 | tobias | [TK]D-Fender: ah, so your theory is that the phones were talking directly to each other |
| 16:52.56 | *** join/#asterisk radicall (n=Zynth@189.168.192.6) |
| 16:53.21 | tobias | while the asterisk server really has problems for all calls |
| 16:55.31 | tobias | hm |
| 16:57.04 | killfill | hey guys.. im doing TrySystem() and sometimes asterisk doesnt completly execut the command. when this happends it stays and eats 100%cpu... |
| 16:57.16 | killfill | this should not happend, isnit? |
| 17:00.38 | tobias | [TK]D-Fender: what settings are you referring to? I've added canreinvite=no (although all phones already had this set) and nat = yes, but i still get the same behavior |
| 17:01.05 | tobias | phones can call each other through * just fine |
| 17:01.32 | tobias | but recordings are silent |
| 17:05.53 | tobias | I am config #4 on http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:06.12 | tobias | and i have everything setup that they list there |
| 17:09.42 | *** join/#asterisk curtn (n=curtis@cl-451.trn-01.it.sixxs.net) |
| 17:09.44 | *** join/#asterisk beasty (n=beasty@about/apple/macbook/beasty) |
| 17:09.48 | beasty | hi there |
| 17:10.00 | beasty | is it possible to send out sms'es trough my asterisk gateway ? |
| 17:10.18 | Greek-B0y | Has anybody successfully implemented an SS7 link on * ? |
| 17:13.28 | killfill | i have ask this before, but i havnt ben able to do it.. when i call users that does not have a voicemail, macro-stdext is not getting called. asterisk makes the call directly. how would i force to go thought that macro? |
| 17:14.25 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 17:17.28 | gedoc | how do i forward extensions between asterisknow boxes? |
| 17:17.56 | gedoc | both boxes already register to each other |
| 17:20.26 | nestAr | #asterisknow or #asterisk-gui for AsteriskNOW |
| 17:21.28 | [TK]D-Fender | tobias, Just read the first guide linked |
| 17:22.19 | *** join/#asterisk eps123 (n=sg@dslb-084-063-179-026.pools.arcor-ip.net) |
| 17:22.22 | [TK]D-Fender | killfill, its your dialplan you tell US why the exten you are dialing isn't hitting the extension you want it to.... |
| 17:23.33 | *** join/#asterisk minkus (n=minkus@pool-71-176-39-147.nrflva.fios.verizon.net) |
| 17:25.25 | tobias | [TK]D-Fender: i've done that and made a few minor changes to my * config |
| 17:25.31 | tobias | [TK]D-Fender: the behavior is identical |
| 17:25.42 | [TK]D-Fender | tobias, pastebin your config masking only passwords |
| 17:26.09 | tobias | [TK]D-Fender: sip.conf, anything else? |
| 17:26.23 | [TK]D-Fender | tobias, just that for now. |
| 17:26.25 | [TK]D-Fender | ~pb |
| 17:26.36 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:28.47 | eps123 | hallo. i've a hfcsusb in nt-mode. when i'm calling to an isdn-phone it rings only once and it appears the message "cdr.c:434 ast_cdr_free: CDR on channel 'mISDN/0-u8' not posted" or the phone noises heavily |
| 17:30.02 | tobias | [TK]D-Fender: http://nat1.ppckernel.org/~tobias/sip.conf |
| 17:30.17 | *** join/#asterisk moemoe (i=moemoe@kuschelhoelle.netzhure.de) |
| 17:31.30 | [TK]D-Fender | tobias, Ah, please set PRECISE codecs under general & each entry. |
| 17:31.52 | [TK]D-Fender | tobias, "disallow=all" and then "allow=" for only the ONE codec it should be using. |
| 17:32.15 | tobias | and ulaw is recommended? |
| 17:33.56 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 17:35.04 | [TK]D-Fender | tobias, yup |
| 17:35.36 | tobias | [TK]D-Fender: same behavior |
| 17:35.37 | [TK]D-Fender | tobias, if you have badwidth concerns, user GSM wherever supported. |
| 17:36.51 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:36.53 | Corydon76-dig | beasty: only if you live in the UK or if you have a GSM modem |
| 17:38.11 | tzafrir_home | actually, SMS through the phone is available in most european countries |
| 17:38.17 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:38.20 | tzafrir_home | (in Israel as well) |
| 17:39.44 | coppice | and much of asia, in chinese too |
| 17:40.05 | Greek-B0y | [TK]D-Fender is it possible to have a sip device in multiple contexts without jumping between contexts? |
| 17:40.51 | coppice | but the SMS code in asterisk only supports one of the two SMS land line protocols, and that one is not used in many places |
| 17:41.21 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 17:42.22 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:42.27 | dr1 | http://www.aocomputing.net/?p=3 <-- that was a good link |
| 17:42.39 | Greek-B0y | so isn't it possible to sms a gsm phone? |
| 17:47.22 | *** part/#asterisk galeras (n=galeras@200.31.204.42) |
| 17:47.39 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
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| 17:51.23 | tobias | ok i've gotta run out for a bit |
| 17:51.26 | tobias | be back later |
| 17:53.41 | [TK]D-Fender | Greek-B0y, a device uses a single context. if you want it to be able to use MULTIPLE contexts, then you make that device use a context that INCLUDES all the others. |
| 17:54.00 | [TK]D-Fender | Greek-B0y, "include => contextwithoutbraces" |
| 17:54.30 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:55.54 | Greek-B0y | thanks TK |
| 17:56.09 | Greek-B0y | how many includes can one have? |
| 17:57.31 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 17:58.33 | [TK]D-Fender | Greek-B0y, I'm sure there is a practical limit, but you shouldn't have a setup so bad as to push it :) |
| 18:01.06 | [TK]D-Fender | ok, BBL |
| 18:04.08 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:06.30 | Greek-B0y | yeah I get what u mean TK |
| 18:11.03 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
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| 19:01.44 | zerocod3r | i want to make a dial plan but didnt find which configuration file I have to edit book says extensions.conf but in asterisk 1.2 which one I have to edit? |
| 19:02.17 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:02.17 | rob0 | yes, that's right for 1.2. Why 1.2? |
| 19:02.24 | tzafrir_home | zerocod3r, /etc/asterisk/extensions.conf , same as in 1.4 |
| 19:03.48 | zerocod3r | there is no [general] in extensions.conf :) can it be different? |
| 19:04.28 | rob0 | I just noticed something odd. My * server runs ntpd, clock is in sync with servers. But the times on the callerID phones are 25 or more seconds slow! |
| 19:06.40 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 19:07.48 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:09.02 | zerocod3r | http://lists.ntp.isc.org/pipermail/questions/2006-July/010899.html |
| 19:09.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:09.33 | zerocod3r | rob0 it is saying that the time latency can be 20 seconds but you can reduce it with -g options or something |
| 19:09.41 | zerocod3r | dont understand much but can be answer |
| 19:10.45 | rob0 | It's no big deal, just annoying to have to mentally add a minute. |
| 19:13.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
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| 19:19.24 | killfill | [TK]D-Fender: well the calls arive just fine. but when i call from 1 extension to another (both on same context) the calls flows not thought macro-stdext. i wish it does.. |
| 19:21.48 | *** join/#asterisk fedya (n=fedya__@rrcs-71-43-242-130.se.biz.rr.com) |
| 19:22.02 | fedya | how can i blacklist an area code? |
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| 19:48.27 | fedya | is there a way i could do _[407|321|456|234]NXXXXXX ? |
| 19:54.10 | anthm | i don't think so but you can do several 1 line ext for each combo that does something like goto,bad,${EXTEN},1 and have context [bad] with a _. exten doing whatever you wanted to do |
| 19:55.13 | fedya | i think htat's what we ended up doing just now |
| 20:08.18 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:17.43 | *** join/#asterisk BadHorsie (n=Illidan@ip57-27-10-190.ct.co.cr) |
| 20:17.49 | [TK]D-Fender | killfill, pastebin your dialplan |
| 20:17.50 | [TK]D-Fender | ~pb |
| 20:17.51 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:17.52 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 20:18.21 | [TK]D-Fender | fedya, not in 1 statement. that'd take 4 |
| 20:21.52 | *** join/#asterisk Entr4nced (n=IMG001@adsl-70-238-245-18.dsl.bcvloh.sbcglobal.net) |
| 20:21.57 | *** join/#asterisk a-s (n=user@89.40.187.17) |
| 20:22.09 | a-s | how can I write a conference soft? |
| 20:24.47 | [TK]D-Fender | ? |
| 20:24.54 | a-s | can somebody recommend me a good open source to start reading in order to understand how to write a conference soft, please? |
| 20:26.03 | [TK]D-Fender | a-s, Asterisk DOES conferencing |
| 20:27.40 | a-s | [TK]D-Fender: yes, but it uses a module zaptel, and Iwish to write in user mode |
| 20:28.00 | [TK]D-Fender | a-s, lookup app_conference. It doesn't use zaptel |
| 20:28.49 | macTijn | win 137 |
| 20:29.07 | *** join/#asterisk zegeek (n=alec@dyn-83-156-48-74.ppp.tiscali.fr) |
| 20:29.11 | zegeek | hello ! |
| 20:29.45 | zegeek | i want to stream music during music on hold can you help me? |
| 20:30.30 | zegeek | anybody's here? |
| 20:31.10 | a-s | [TK]D-Fender: thanks |
| 20:31.12 | [TK]D-Fender | zegeek, Look at the sample musiconhold.conf . There is a sameple in there |
| 20:31.33 | zegeek | yes but it doesn't works :-( |
| 20:31.56 | [TK]D-Fender | zegeek, pastebin your configs and a failed attempt and we'll take a look |
| 20:31.57 | [TK]D-Fender | ~pb |
| 20:31.58 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:31.59 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 20:32.01 | [TK]D-Fender | ~book |
| 20:32.02 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 20:34.01 | zegeek | ihttp://pastebin.com/m49eba891 & http://pastebin.com/mb6f63b8 |
| 20:34.07 | zegeek | do you want more? |
| 20:35.04 | zegeek | it is good? |
| 20:35.12 | [TK]D-Fender | looking |
| 20:35.17 | zegeek | ok |
| 20:35.29 | [TK]D-Fender | zegeek, I want to see the call CLI output.... |
| 20:37.21 | zegeek | http://pastebin.com/m2510bed6 |
| 20:37.25 | zegeek | this? |
| 20:38.40 | a-s | [TK]D-Fender: app_conference is not in *' sources |
| 20:39.13 | [TK]D-Fender | a-s, I know. Thats why I told you to look it up. Time to start WIKI-ing |
| 20:39.47 | [TK]D-Fender | zegeek, yes, that is CLI, but I do not see the call where you are attempt to PLACE a call on hold... |
| 20:41.06 | zegeek | i call my paris number who is the sip account assotied and it hangs up at 30 seconds no more |
| 20:42.37 | [TK]D-Fender | zegeek, we can't help you with your MoH problem until you SHOW us an actual attempt to USE it. |
| 20:45.16 | zegeek | this can you help you http://pastebin.com/m58983847 ? |
| 20:46.06 | zegeek | what does "[Nov 24 22:44:38] NOTICE[10957]: res_musiconhold.c:539 monmp3thread: Request to schedule in the past?!?!" means? |
| 20:47.20 | deever | sth like "include => ${EXTEN}_ctx" doesn't work, does it? |
| 20:47.39 | [TK]D-Fender | zegeek, I need to see the COMPLETE call from beginning to end. |
| 20:47.47 | [TK]D-Fender | deever, No. |
| 20:48.02 | deever | what a pity! :( |
| 20:49.55 | *** part/#asterisk a-s (n=user@89.40.187.17) |
| 20:53.30 | *** join/#asterisk asdx (n=diego@adsl-148-240.click.com.py) |
| 20:53.45 | *** join/#asterisk Duryas (n=Naps@athedsl-363354.home.otenet.gr) |
| 20:54.27 | zegeek | i have a lot of problems |
| 20:54.32 | Duryas | anyone experienced with h323? |
| 20:54.51 | deever | zegeek: me too |
| 20:55.11 | Greek-Boy | [TK]D-Fender what is Asterisk trunk version? |
| 20:55.23 | zegeek | look this |
| 20:55.24 | zegeek | http://pastebin.com/m2bc64ade |
| 20:55.27 | [TK]D-Fender | Greek-Boy, There is no version except the checkout # |
| 20:55.28 | fakhir | zegeek, typically the best way to deal with that is one problem at a time |
| 20:55.32 | deever | but they aren't related to asterisk! ;) |
| 20:55.45 | deever | fakhir: lol |
| 20:56.14 | zegeek | fakhir : lool |
| 20:56.14 | Greek-Boy | [TK]D-Fender: what is it all about? |
| 20:56.27 | [TK]D-Fender | zegeek, You are not listing. I want to see the ENTIRE CALL including the point where you PUT the person on HOLD. |
| 20:56.36 | [TK]D-Fender | Greek-Boy, huh? |
| 20:57.01 | Greek-Boy | [TK]D-Fender: I mean what is Asterisk trunk? |
| 20:57.17 | [TK]D-Fender | Greek-Boy, That is what will eventually become 1.6 |
| 20:57.30 | zegeek | <[TK]D-Fender> i want just to hear on my phone streamed music that's all |
| 20:57.42 | zegeek | not music on hold |
| 20:57.58 | Greek-Boy | [TK]D-Fender: Will it be carrier grade? |
| 20:58.12 | [TK]D-Fender | zegeek, music doesn't come out of thin air. Show me the entire call including where it is supposed to START. You are wasting our time. |
| 20:58.26 | [TK]D-Fender | Greek-Boy, Depends on the carrier you are comparing to :) |
| 20:58.56 | Greek-Boy | lol |
| 20:59.29 | Greek-Boy | Well, it seems most voip providers are using asterisk for value added services. |
| 20:59.39 | Greek-Boy | and I'm guessing they're using SER |
| 20:59.50 | Greek-Boy | i'm talking about about the majority now |
| 21:02.55 | [TK]D-Fender | Greek-Boy, * is a B2BUA and sucks for billing. Its user handling is less than stellar as well. * is best used as a termination & application server behind something like SER. |
| 21:03.20 | [TK]D-Fender | Greek-Boy, keep an eye on FreeSWITCH. Theres some good stuff planned for it from what I've heard. |
| 21:06.03 | zegeek | http://pastebin.com/m1bb33453 |
| 21:06.07 | zegeek | look this |
| 21:06.15 | *** join/#asterisk ManxPower (n=manxpowe@97.81.241.169) |
| 21:06.53 | [TK]D-Fender | zegeek, Ok, last time, I want to see the ENTIRE DAMN CALL. What don't you get? I don't see the REQUEST for music to start in any of those. Stop wasting our time |
| 21:08.22 | Greek-Boy | [TK]D-Fender: * is also good as a CPE? |
| 21:08.38 | zegeek | I'm very sorry to said i'm not understand what you want to tell me |
| 21:09.39 | zegeek | what do you want ? |
| 21:09.51 | fakhir | he want the entire CLI output for a call start to finish |
| 21:09.59 | [TK]D-Fender | zegeek, je doit voir l'appel AU COMPLET! Debut au fin! Chacque maudite linge de code executer! |
| 21:10.07 | [TK]D-Fender | :) |
| 21:10.53 | zegeek | but he is complet |
| 21:11.13 | [TK]D-Fender | zegeek, Where is "answer"? Where do we see * doing ANYTHING at all? |
| 21:11.20 | [TK]D-Fender | # |
| 21:11.21 | [TK]D-Fender | [Nov 24 23:04:07] NOTICE[11464]: res_musiconhold.c:539 monmp3thread: Request to schedule in the past?!?! |
| 21:11.21 | [TK]D-Fender | # |
| 21:11.21 | [TK]D-Fender | [Nov 24 23:04:27] NOTICE[11464]: res_musiconhold.c:539 monmp3thread: Request to schedule in the past?!?! |
| 21:11.21 | [TK]D-Fender | # |
| 21:11.22 | [TK]D-Fender | [Nov 24 23:04:28] NOTICE[11464]: res_musiconhold.c:539 monmp3thread: Request to schedule in the past?!?! |
| 21:11.24 | [TK]D-Fender | # |
| 21:11.26 | [TK]D-Fender | ==>The call hang up without my intervention |
| 21:11.28 | [TK]D-Fender | this in NOT an entire call! |
| 21:11.38 | [TK]D-Fender | zegeek, Make sure you're on verbose 10 |
| 21:12.25 | zegeek | No such command 'verbose 10' (type 'help' for help) |
| 21:12.44 | [TK]D-Fender | zegeek, .... "set verbose 10" <------------ |
| 21:12.48 | zegeek | ok thanks |
| 21:13.33 | [TK]D-Fender | Greek-Boy, clarify what you mean by CPE. |
| 21:14.09 | Greek-Boy | CPE = Client Premises Equipment as in * as a client CPE PBX interfacing to the provider... |
| 21:14.56 | zegeek | http://pastebin.com/m43ba50c2 |
| 21:16.01 | [TK]D-Fender | Greek-Boy, I know the technical term.... and yes, you generally want the * box on-site. |
| 21:16.02 | [TK]D-Fender | zegeek, Started music on hold, class 'default', on channel 'SIP/ippi.fr-081e5760' |
| 21:16.02 | Greek-Boy | thats good. * has its role to play and its the best at it. |
| 21:16.27 | zegeek | i can't hear nothing |
| 21:16.38 | [TK]D-Fender | zegeek, application=/usr/bin/mpg123 -q -r 8000 -f 8192 -s --mono http://213.186.41.160:8000/ |
| 21:16.48 | [TK]D-Fender | zegeek, go verify what version of mp123 you have |
| 21:17.24 | zegeek | version 0.68 |
| 21:17.56 | [TK]D-Fender | zegeek, The only version you should be using with * is 0.59r. This is well documented. go install the proper version. |
| 21:18.14 | zegeek | ok |
| 21:21.57 | zegeek | i can't compile |
| 21:23.01 | zegeek | http://pastebin.com/m6d06070a |
| 21:24.02 | [TK]D-Fender | zegeek, You'll have to gogle that one... |
| 21:24.05 | [TK]D-Fender | Google* |
| 21:26.06 | Duryas | h323 issue : http://pastebin.ca/796208 can someone help? |
| 21:35.20 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 21:36.05 | zegeek | it is okay for mpg123 |
| 21:36.10 | zegeek | compiled & run |
| 21:38.20 | zegeek | http://pastebin.com/m7263273d |
| 21:38.28 | zegeek | don't work |
| 21:39.16 | [TK]D-Fender | zegeek, Go install zaptel & recompile *. |
| 21:39.32 | zegeek | why i need zaptel? |
| 21:39.51 | [TK]D-Fender | zegeek, MoH synch works a lot better with a zaptel timing source. If you don't have a supported card, use the ZTDUMMY module. |
| 21:40.06 | zegeek | i use only sip |
| 21:40.17 | [TK]D-Fender | zegeek, Then setup Zaptel w/ ztdummy |
| 21:41.35 | *** join/#asterisk jameswf-home (n=that@ip72-204-228-104.ph.ph.cox.net) |
| 21:47.22 | *** join/#asterisk RoyK (n=roy@ip-165-30-149-91.dialup.ice.no) |
| 21:51.18 | zegeek | it compiles |
| 21:56.10 | BadHorsie | wasn't tzdummy just for 2.4 ? |
| 21:57.48 | Qwell | no |
| 21:58.30 | jameswf-home | holy crap #emo is a room |
| 21:59.05 | Greek-Boy | [TK]D-Fender its interesting that fonality.com uses * for their clients and they have their own modified version of * for their servers and back end too. |
| 21:59.13 | [hC] | Interesting. Adding a bunch of phones to Page() behaves nice and snappy using ztdummy, but lags and performs pretty poorly when using wanpipe+zaptel |
| 21:59.59 | [TK]D-Fender | Greek-Boy, What is that hard to believe? its just another GUI modded for specific features. |
| 22:00.13 | [TK]D-Fender | [hC], Yeah I read about that recently.... |
| 22:00.30 | [hC] | I mentioned it in here about a week ago, was it from me, or someplace else? |
| 22:01.05 | [TK]D-Fender | [hC], may have something to do with the way that frames are deposited to zaptel in that I don't think it uses 1000irq/s like Digiums do. |
| 22:01.16 | [TK]D-Fender | [hC], could be both |
| 22:01.22 | Greek-Boy | [TK]D-Fender: Yeah but they use * on their infrastructure and they must be processing large amounts of calls. Maybe they're using SER or something else in addition |
| 22:01.52 | [TK]D-Fender | Greek-Boy, Fonality is a GUI reseller, not an ITSP unless they've opened up a business model I'm unaware of... |
| 22:01.53 | jameswf-home | I read a post about sangoma being a leach I think the post really applies more to fonality who oddly pimps sangoma |
| 22:02.02 | zegeek | same thing |
| 22:02.03 | zegeek | http://pastebin.com/m770af85b |
| 22:02.15 | [hC] | [TK]D-Fender: oh hey i was going to ask you as well.. Are you using sip firmware 2.2.0 on polycom? I just did an upgrade to a bunch of phones from 1.6.7 to that firmware and they're giving me very strange error reports... Stuff like their phones stop ringing altogether and just blink the line indicator led, or that ring volume seems to decrease with every ring, or be really quiet even when on max volume.. |
| 22:02.21 | [hC] | [TK]D-Fender: makes zero sense to me. |
| 22:02.55 | jameswf-home | I think we all should open our own hybrid hosted solutions |
| 22:02.57 | [TK]D-Fender | [hC], Did you rebuild your configs from the base 2.2 or did you try to port your old ones? |
| 22:03.01 | Speedy2 | Anyone here using the Linksys branded SPA-1001 ? |
| 22:03.10 | [TK]D-Fender | Speedy2, I have a few at the office.... |
| 22:03.19 | [hC] | [TK]D-Fender: rebuilt, the only thing i ported was the reg.1 reg.2 reg.3, etc lines |
| 22:03.34 | [TK]D-Fender | [hC], those should be largely OK.... |
| 22:03.50 | jameswf-home | well crap california is on fire again |
| 22:04.24 | dr1 | i know thats bad, but i couldnt help but laugh just now |
| 22:04.31 | [hC] | [TK]D-Fender: in my mac.cfg file i have x###.cfg, myoverrides.cfg, phone1.cfg, sip.cfg ... the x###.cfg file simply contains the <phone1> and <reg> tags, and my overrides are pretty damn basic,. |
| 22:04.59 | Greek-Boy | [TK]D-Fender: I think they provide ITSP services too. Let me confirm... |
| 22:07.20 | [TK]D-Fender | jameswf-home, thats one of LA's seasons you know. They don't have the same as the rest of the world. Instead they have : Flood, Quake, Fire, and Riot. |
| 22:07.42 | *** join/#asterisk [Mr_X] (n=mrx@78-59-13-199.ip.zebra.lt) |
| 22:07.52 | Greek-Boy | [TK]D-Fender: Looks like you were right about their business model. I misconceived it earlier on. I can't believe people will pay so much for features that are all standard in * |
| 22:08.08 | jameswf-home | You know its only on cnn cause its white people, thate what conyay west said |
| 22:08.10 | Greek-Boy | I guess its about simplicity for less technical savvy users. |
| 22:10.57 | Speedy2 | jameswf-home: Where?? |
| 22:10.58 | [TK]D-Fender | jameswf-home, Kanye |
| 22:11.09 | [TK]D-Fender | jameswf-home, At least spell his name right, ok? :) |
| 22:12.16 | jameswf-home | nah I like $((EC-CON))+$((yayy)) makes him sound like a exciting criminal |
| 22:12.21 | [TK]D-Fender | Greek-Boy, And its because of the GUI, and corporate accountability. Companies like buying products so they have somebody to hold accountable to fix it or sue. |
| 22:12.28 | jameswf-home | s/ec/ex/ |
| 22:12.45 | rob0 | And that whole idea is incredibly stupid. |
| 22:13.03 | Greek-Boy | btw, what happened to asterisk business edition? still out there? |
| 22:13.11 | [TK]D-Fender | Greek-Boy, naturally |
| 22:13.38 | jameswf-home | what happened to paying a little more for an IT guy who knows his stuff and holding him accountible |
| 22:14.18 | rob0 | The licenses for those products disclaim liability. They can sue, but they can't win. |
| 22:14.35 | jameswf-home | cnn is using google earth, wonder if they payed royals |
| 22:14.38 | [TK]D-Fender | jameswf-home, Doesn't make stuffed shirts happy. |
| 22:15.03 | jameswf-home | stuffed shirts? lol |
| 22:15.04 | [TK]D-Fender | rob0, and a question of support that can't quit when they're employees burn out. |
| 22:15.12 | Greek-Boy | i really dont see the benefits to asterisk business edition |
| 22:15.35 | [TK]D-Fender | Greek-Boy, paid support, access to non GPL'd bits. |
| 22:15.51 | [TK]D-Fender | Greek-Boy, A nifty pre-deprecated manual :) |
| 22:16.09 | jameswf-home | they depricated on my shoes |
| 22:17.10 | [hC] | this is so strange that this guys symptoms would only show up after a couple hours |
| 22:17.18 | [TK]D-Fender | jameswf-home, not quite the right word :) |
| 22:17.42 | [TK]D-Fender | [hC], On fridays I show symptoms only from 8am-4:30pm ;) |
| 22:17.58 | [hC] | [TK]D-Fender: haha :) |
| 22:18.41 | jameswf-home | It is amaizing how many "admins" are to stupid to pick up a book |
| 22:20.08 | jameswf-home | jbot: rtfm |
| 22:20.09 | jbot | i heard rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 22:20.32 | jameswf-home | jbot: id10t |
| 22:20.33 | jbot | hmm... id10t is a chair to keyboard user interface error. |
| 22:21.18 | jameswf-home | jbot: pebcak |
| 22:21.19 | jbot | well, pebcak is Problem Exists Between Chair And Keyboard |
| 22:22.02 | jameswf-home | jbot: botsnack |
| 22:22.02 | jbot | thanks, jameswf-home |
| 22:22.19 | [TK]D-Fender | jameswf-home, just prefix those with "~" |
| 22:26.22 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:26.50 | Greek-Boy | Interesting about DUNDI... http://www.asterisk.org/node/48321 |
| 22:27.11 | [TK]D-Fender | mooooooooooooo |
| 22:27.14 | jameswf-home | ~emo |
| 22:27.14 | jbot | /wrists |
| 22:27.28 | [TK]D-Fender | lol |
| 22:27.28 | jameswf-home | roflmfao |
| 22:29.19 | [TK]D-Fender | I wish my lawn was emo... so it'd cut itself :) |
| 22:30.09 | [TK]D-Fender | http://www.myconfinedspace.com/wp-content/uploads/2006/11/emo-crayola.jpg |
| 22:30.44 | BadHorsie | lol |
| 22:31.09 | jameswf-home | I set emo in to stumbleupon... some funny stuff |
| 22:31.28 | jameswf-home | I wish my grass was EMO so it would cut it's self |
| 22:33.05 | BadHorsie | stumbleupon rocks |
| 22:33.10 | jameswf-home | damn http://www.bose.com/controller?event=VIEW_PRODUCT_PAGE_EVENT&product=mwr_qc2_qc3&intcmp=USB00012&ck=0 $350.00 head phones |
| 22:33.14 | BadHorsie | cut itself, hahaha |
| 22:33.27 | [TK]D-Fender | Damn... one of my fillet mignons came out overcooked.... RARE! Dear God..... |
| 22:33.41 | [TK]D-Fender | wait... |
| 22:33.44 | [TK]D-Fender | a HINT of blue! |
| 22:34.50 | [TK]D-Fender | Still tender as hell... ok, I'm good now... |
| 22:35.03 | *** join/#asterisk orsonork (n=orsonork@190.128.168.24) |
| 22:40.04 | jameswf-home | “I believe in making the world safe for our children, but not our children's children, because I don't think children should be having sex.” |
| 22:40.18 | jameswf-home | Jack handy is great |
| 22:42.12 | [TK]D-Fender | jameswf-home, The name alone is comic relief |
| 22:46.13 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 22:58.24 | Siya | Is there a way to check inbound calls for their type? (SIP) |
| 22:58.44 | Siya | aka can I use the same DID for inbound voice and fax calls? |
| 23:00.19 | Siya | I'm hoping I could script it to test for a pilot tone like (pickup, if tone then fwd to fax otherwise fwd to phone) |
| 23:10.27 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
| 23:17.16 | *** join/#asterisk RoyKa (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 23:18.53 | *** join/#asterisk robeph (n=robf@user-24-236-88-244.knology.net) |
| 23:21.42 | tobias | [TK]D-Fender: around? |
| 23:30.13 | BadHorsie | i'm new to asterisk and i'm reading the TFOT, so basically, if one is using digium cards (or 1K IRQ/s card) you gotta use ztdummy, but not both at the same time, right? |
| 23:39.12 | De_Mon | BadHorsie no, ztdummy is in you arn't using any cards, ala pure VoIP SIP/IAX |
| 23:39.42 | De_Mon | Siya ${CUT(CHANNEL,/,1) will tell you the channel type of the inbound call |
| 23:40.11 | De_Mon | fax detection is something else |
| 23:41.28 | De_Mon | [TK]D-Fender how do you get something 'rare' and 'over cooked' at the same time? |
| 23:41.49 | BadHorsie | thanks |
| 23:41.58 | rob0 | I took that to indicate a taste for blood. |
