| 00:02.39 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 00:09.09 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 00:23.02 | *** join/#asterisk rcy (n=rcy@S010600003981572c.vc.shawcable.net) |
| 00:24.02 | hsv-al | hello |
| 00:24.09 | hsv-al | are we all looking forward to another long & glorious night on irc? :) |
| 00:26.01 | drmessano | No |
| 00:26.04 | drmessano | IRC sucks |
| 00:26.05 | MikeJ | no |
| 00:26.09 | drmessano | I am going back to ICQ |
| 00:26.12 | drmessano | Screw you guys |
| 00:26.15 | MikeJ | on a train |
| 00:26.16 | *** part/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 00:26.27 | MikeJ | choo choo |
| 00:26.30 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 00:26.43 | drmessano | So yeah, 1.4.20.1 really rocks |
| 00:27.00 | drmessano | IAX2 is all like, "IAX TOO!" |
| 00:28.10 | hsv-al | drmessano, for to long ive been using asterisk now, heh |
| 00:28.21 | hsv-al | so i actually decided to build * from the ground up today and it works |
| 00:28.39 | hsv-al | sorta gives me stress having no gui, but it forces me to do everything cli |
| 00:30.18 | drmessano | I hate GUI's... all my phones are even CLI |
| 00:30.22 | drmessano | Sure, it's a pain dialing |
| 00:30.34 | hsv-al | well thats why I had to buy that book, |
| 00:30.47 | drmessano | But WORTH it to be able to proclaim "I AM SMARTER THAN YOU BECAUSE I USE A COMMAND LINE" |
| 00:30.47 | hsv-al | 7 chapters in 3 days, but 23 pages of notes so far |
| 00:31.00 | drmessano | Since, you know, that's the ultimate in smarteredness |
| 00:31.32 | drmessano | At work, they all use Microsoft Word.. |
| 00:31.40 | drmessano | They're all like, typing up documents in 10 mins and shit |
| 00:32.10 | drmessano | I am all like creating my docs in WordPerfect 5.0 for DOS, and putting in my own codes.. |
| 00:32.12 | hsv-al | AN is a nice iso/prepackage, it has the CLI too |
| 00:32.13 | *** join/#asterisk mwalling (i=mwalling@you.dontlike.us) |
| 00:32.19 | hsv-al | via that bottom button |
| 00:32.20 | drmessano | Sure, it takes me two hours, but I know where every P is |
| 00:32.28 | hsv-al | but i had to build this/get used to its innards |
| 00:32.53 | hsv-al | I wanted to go inside, so I can feel its seed |
| 00:33.02 | drmessano | I wrote my own IRC client in ASM |
| 00:33.18 | mwalling | sadist |
| 00:33.20 | drmessano | I started in 1994.. by the time I was done, IRC was dead.. but hey, I R WROTE IT |
| 00:33.40 | hsv-al | whats up w/ the /images/ folder? |
| 00:33.46 | hsv-al | certain phones that can do pictures on LCD? |
| 00:33.51 | hsv-al | for what reason? was very vague |
| 00:37.53 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 00:42.03 | TJNII | Phone porn |
| 00:42.11 | TJNII | Technology is always used for porn |
| 00:42.56 | JackEStorm | where are the detailed docs for meetme? |
| 00:43.53 | JackEStorm | TJNII: no, Porn develops technology, that everyone uses. |
| 00:45.06 | jblack | Me shudders at the kind of porn that must have invented ceiling fans |
| 00:46.44 | JackEStorm | jblack: that came out of a BD auto paddler. |
| 00:47.27 | JackEStorm | for when she's just been that *BAD* spanking her will give you carpal tunnel |
| 00:47.46 | *** join/#asterisk BipedalShark (n=Richard@216-110-94-240.static.twtelecom.net) |
| 00:54.29 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 00:56.25 | *** join/#asterisk twitchnln (n=twitch@c-76-105-88-129.hsd1.ga.comcast.net) |
| 00:58.16 | JackEStorm | any idea on the location of detailed docs for meetme? trying to figure out why asterisk hangs-up when an invalid conf number is given. |
| 01:00.16 | deeperror | sounds like an error and not a function that would be documented to me |
| 01:01.22 | JackEStorm | error as in error on my part (I belive it is, but can't figure it out), or error as in bug? |
| 01:09.51 | [TK]D-Fender | JackEStorm, Detailed docs? lol |
| 01:10.02 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b21e8386fda0cf41) |
| 01:11.19 | drmessano | I need the PDF manual for notepad, when someone has a chance |
| 01:13.31 | deeperror | could be...pastebin |
| 01:14.04 | JackEStorm | [TK]D-Fender: yes, really, yes |
| 01:15.07 | JackEStorm | [TK]D-Fender: just trying to figure out how not to have meetme hang up on Meetme({invalid-conf}) |
| 01:15.19 | *** join/#asterisk fnordus (n=dnall@S0106000c4198ed25.vs.shawcable.net) |
| 01:16.06 | deeperror | fender got the sgoma installed today. Any reason i shouldn't set all 4 spans to master on timing? |
| 01:16.29 | deeperror | why is it invalid? is it in meetme.conf? |
| 01:16.51 | *** join/#asterisk Tommy3 (n=tom@76.29.235.137) |
| 01:16.52 | deeperror | use d option? |
| 01:17.42 | JackEStorm | no, no dynamic support, if it's invalid I need it kicked back. |
| 01:18.08 | JackEStorm | (incase of fafingers) |
| 01:18.20 | JackEStorm | doh, I mean fat fingers. |
| 01:18.26 | deeperror | i just wrote a macro to set a valid conf for me not sure if that is reinventing the wheel or not but it works |
| 01:19.40 | JackEStorm | but Meetme plays conf-invalid, leading to one to think that when Allison says "Try again" it won't just hang up |
| 01:19.49 | *** join/#asterisk rcy (n=rcy@S010600003981572c.vc.shawcable.net) |
| 01:20.24 | *** part/#asterisk BipedalShark (n=Richard@216-110-94-240.static.twtelecom.net) |
| 01:22.35 | JackEStorm | (and honestly It should kick back to i,2 or defined) |
| 01:23.42 | Tommy3 | Hello asterisk'ers |
| 01:24.50 | Tommy3 | Any successes getting USTarcom F3000 phone to work? |
| 01:25.11 | Tommy3 | (UTStarcom) |
| 01:27.17 | *** join/#asterisk puga (n=puga@200-170-141-251.static.ctbctelecom.com.br) |
| 01:28.13 | puga | hello... can anyone help me with sip transport encryptation? |
| 01:28.29 | [TK]D-Fender | Tommy3, I had one. It was a flakey piece of EXPLETIVE DELETED |
| 01:28.54 | jeev | FENDER |
| 01:28.59 | jeev | linksys fucked me over |
| 01:29.11 | [TK]D-Fender | jeev, O RLY? |
| 01:29.13 | jeev | they gave me an rma number.. and i submit rma, blank email.. no calls, no nothing |
| 01:29.15 | [TK]D-Fender | jeev, do tell... |
| 01:29.26 | jeev | their escalated shit doesn't call me back |
| 01:29.30 | jeev | nobody is calling me back |
| 01:29.54 | jeev | i set up rma, an email came, was like 1kb. blank body. it said you need an HTML blabhlabhl to read this.. gmail showed nothing, outlook nothing, thunderbird nothing.. was never altered or modified |
| 01:30.09 | puga | nobody? |
| 01:30.13 | jeev | i've submitted to their escalated department on their site (which was what i did the first time to get my RMA set up) and no call. |
| 01:30.41 | jeev | well |
| 01:30.44 | jeev | some dood from brazil actually |
| 01:30.46 | jeev | tried helping heh |
| 01:30.48 | [TK]D-Fender | jeev, couldn't just listen to us could you? Oh well. |
| 01:30.57 | jeev | no, argentina |
| 01:31.04 | jeev | fender, i've submitted a formal complaint. i will request a full refund. |
| 01:31.06 | jeev | :) |
| 01:32.06 | Tommy3 | [FENDER] I read the acalades AFTER I bought it. I've managed to get it to connect to the wifi, but too many guessable combinations to get sip working to the asterisk box I fear. |
| 01:32.11 | unpaidbill | i must be crazy... my zaptel cards are detected as followS: ports 1-24 are the TE110P, and 25-32 are the TDM2400P with 8 ports in it, according to my zaptel.conf file, generated by genzaptelconf.. ztcfg -vv shows them like that... but when i pick up the analog phone connected to port 1 on the TDM card, it's showing as Zap/1 ?? |
| 01:32.19 | unpaidbill | i thought it would be Zap/25 ? |
| 01:33.44 | [TK]D-Fender | unpaidbill, guess you're wrong. Go deal with it |
| 01:33.53 | unpaidbill | i guess so |
| 01:34.56 | *** part/#asterisk twitchnln (n=twitch@c-76-105-88-129.hsd1.ga.comcast.net) |
| 01:36.00 | JackEStorm | [TK]D-Fender: thats the one thing I really hate about asterisk, transport naming. |
| 01:41.51 | drmessano | I would prefer all transports in Asterisk be called "tubes" |
| 01:41.58 | *** join/#asterisk wynix (n=nate@63.162.28.92) |
| 01:42.16 | tzafrir_laptop | unpaidbill, pastebin cat /proc/zaptel/* |
| 01:48.16 | puga | asterisk 1.4 does not support tls? |
| 01:49.59 | [TK]D-Fender | puga, No. |
| 01:50.40 | *** part/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com) |
| 01:50.45 | puga | [TK]D-Fender, any other kind of encryptation? |
| 01:50.57 | [TK]D-Fender | puga, Not for SIP |
| 01:51.37 | puga | =( |
| 01:52.26 | puga | [TK]D-Fender, and for media transport, asterisk supports srtp ? |
| 01:52.29 | hsv-al | head on - applied directly to the forehead |
| 01:52.37 | [TK]D-Fender | puga |
| 01:52.40 | [TK]D-Fender | NO |
| 01:52.43 | LiNeTuX | I would love to see * support certificates and the like and shove that down the Call Manager folks' throat. |
| 01:53.09 | [TK]D-Fender | puga, Lets get this over with quickly NO SIP/RTP ENCRYPTION in 1.4 <- Are we clear now? |
| 01:53.28 | puga | okay, sorry |
| 02:00.36 | CVirus | I remember there was a bounty for this though |
| 02:00.49 | CVirus | doh .. he's gone anyways |
| 02:03.59 | *** join/#asterisk codehaxor (n=info@124.6.153.226) |
| 02:04.02 | codehaxor | hi guys |
| 02:04.15 | Tommy3 | Goodnight from asterisk mecca (Huntsville) |
| 02:04.19 | codehaxor | anyone here tried installing asterisk and zaptel under an openvms container? |
| 02:05.19 | codehaxor | im having a hard time compiling zaptel |
| 02:08.28 | deeperror | what does it say |
| 02:13.16 | codehaxor | cant install it due to a dependency.... anyway im running it under an openvz container |
| 02:13.32 | codehaxor | asterisk compiles successfully so does libpri |
| 02:13.34 | codehaxor | only zaptel |
| 02:13.48 | deeperror | why not install the dependency? |
| 02:16.19 | [TK]D-Fender | Doctor, Doctor, it hurts when I raise my arm like this! |
| 02:18.20 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 02:20.31 | codehaxor | has anyone installed zaptel on openvz |
| 02:25.30 | errr | [TK]D-Fender: dont raise your arm like that |
| 02:25.56 | [TK]D-Fender | codehaxor, Well... what is the dependency? |
| 02:27.43 | codehaxor | You do not appear to have the sources for the 2.6.18-028stab039.2-ovz-smp kernel installed (under ). |
| 02:27.43 | codehaxor | make: *** [modules] Error 1 |
| 02:28.07 | deeperror | sources are required |
| 02:28.55 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 02:29.35 | codehaxor | tried installing sources.. but thing is with openvz is your just under 1 kernel |
| 02:30.11 | deeperror | but need kernel-devel |
| 02:32.09 | drmessano | So... install the dev-kernel |
| 02:32.19 | drmessano | I am a Windows 95 admin and I know that much |
| 02:32.49 | deeperror | haven't upgraded to ME yet? |
| 02:33.23 | errr | We pheer change |
| 02:33.35 | errr | still on 95 here too |
| 02:34.21 | jaytee | every couple months or so I dual boot into Vista just to make sure it's still there |
| 02:34.24 | voxter | 95? shit you guys are courageous... I'm using netware 3.12 |
| 02:34.47 | deeperror | your qa database |
| 02:34.53 | [TK]D-Fender | codehaxor, If you can't satisfy the dependency then you're screwed. |
| 02:34.54 | jaytee | my first network install was Netware 286 v2.12. |
| 02:35.56 | [TK]D-Fender | voxter, thats what my company was using when I started there.... we're now riding high on Novell 5.0! Tre very best 1998 had to offer! |
| 02:36.03 | [TK]D-Fender | The8 |
| 02:36.16 | [TK]D-Fender | is about to dump that shit for Windows Server 2003 |
| 02:36.17 | drmessano | Netware 3.12? Ha, loser.. I upgraded to 3.2 almost 3 years ago |
| 02:36.44 | jaytee | Netware 3.2 was Da Bomb! |
| 02:36.51 | deeperror | government? ha |
| 02:37.00 | drmessano | volrepair /y |
| 02:37.01 | drmessano | volrepair /y |
| 02:37.54 | drmessano | You know what is so great about Netware? |
| 02:38.08 | drmessano | Linux = Blah, blah, all this crap, blah dependency, blah |
| 02:38.09 | jbeez | it runs on ms dos? |
| 02:38.16 | drmessano | Netware = Server.exe |
| 02:38.19 | drmessano | pwn3d |
| 02:38.23 | jbeez | lol |
| 02:38.45 | hsv-al | .... |
| 02:39.01 | drmessano | <LinuxDude> Hang on, gotta boot and do all this other crap to make my Leenux box work |
| 02:39.16 | drmessano | <NetwareDude> Server.exe <enter> |
| 02:39.21 | jbeez | dood I got a ticket on my way home from work today :< |
| 02:39.23 | jaytee | lol |
| 02:39.48 | drmessano | Hang on, need to upgrade my Netware install.. |
| 02:39.56 | drmessano | ren server.exe server.old |
| 02:40.02 | drmessano | ren server.new server.exe |
| 02:40.02 | jbeez | state trooper was mad because I didn't give him anything to work with, so he wrote me a ticket for failure to obey a traffic control device on the highway, its a BS $100 ticket :/ |
| 02:40.11 | drmessano | server.exe <enter> |
| 02:41.15 | drmessano | If someone ported Asterisk to Netware, it would be one NLM |
| 02:41.24 | drmessano | and run for 7 years without crashing... |
| 02:41.30 | jaytee | hahahah |
| 02:42.07 | drmessano | We had a netware server on the radio automation system at my old job |
| 02:42.14 | jaytee | "some guy set this up back in 2008 but he died last year and no one knows how to login." |
| 02:42.20 | drmessano | Every year or two, I would need to upgrade the NLM for the DB |
| 02:42.25 | jbeez | hah |
| 02:42.37 | drmessano | Which was basically |
| 02:42.50 | drmessano | unload wizard |
| 02:42.56 | drmessano | load wizard |
| 02:43.01 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 02:43.22 | drmessano | You had 8 seconds to load the DB after it unloaded before the workstations would go into defcon 5 |
| 02:43.38 | drmessano | and I make lots of typos |
| 02:43.54 | drmessano | two lines, more stress than being married to my first wife |
| 02:44.15 | jaytee | hehehe |
| 02:44.35 | drmessano | L O A S (OH SHIT) <backspace> D W I ZX <BACKSPACE> A R D <enter? |
| 02:44.43 | drmessano | </pant> |
| 02:51.04 | hsv-al | drmessano , that program I was talkign about actually came online this spring |
| 02:51.09 | hsv-al | http://ism.cmu.edu/Distance-Learning/Program/courses.asp |
| 02:51.31 | hsv-al | mis masters @ cmu - requires a 6level accounting class (ugh) |
| 02:53.08 | jameswf-home | level6 accountant.... like dungeon masters? |
| 02:53.28 | hsv-al | heh |
| 02:53.42 | hsv-al | that university is insane, look at the oracle description |
| 02:53.50 | jameswf-home | ~wow |
| 02:53.53 | jbot | I have no life | Lets go raid! |
| 02:54.01 | jameswf-home | ~dnd |
| 02:54.02 | jbot | GUI of Molecular Dynamics. URL: http://theopenlab.uml.edu/dnd/index.html |
| 02:54.09 | jameswf-home | bah |
| 02:54.38 | *** join/#asterisk christophocles (n=christop@cpe-68-201-114-137.gt.res.rr.com) |
| 02:54.55 | hsv-al | alot of these grade-a universities, with masters in cs/mis, require ridiculous amounts of math to be applied in the 6-level classes |
| 02:57.18 | christophocles | i want to use asterisk to set up an internet voicemail system to receive messages from any standard telephone line and email them to me. i also need a phone number. can anyone direct me to a guide or howto? |
| 02:57.42 | deeperror | ~book |
| 02:57.42 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 02:57.57 | christophocles | kthx |
| 03:05.47 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 03:08.29 | *** join/#asterisk ix33 (n=ix@206.222.13.162) |
| 03:09.22 | ix33 | anybody ever done anything cool with the polycom microbrowser? i'd like to have it show the status of my switchable auto-attendant for my receptionist phones. |
| 03:11.18 | [TK]D-Fender | ix33, easy enough. I've got mine showing live status for 2 queues & 4 agents |
| 03:12.14 | ix33 | [TK]D-Fender: what mechanism do you use to publish asterisk state information to an .html file? |
| 03:12.36 | [TK]D-Fender | ix33, PHP + AMI |
| 03:12.38 | ix33 | [TK]D-Fender: like, a bunch of 'asterisk -rx'es or what? |
| 03:12.43 | ix33 | [TK]D-Fender: ah. |
| 03:13.01 | ix33 | [TK]D-Fender: ok, good info. |
| 03:31.32 | unpaidbill | if you're still here tza.. http://pastebin.com/m37ef8071 |
| 03:31.40 | unpaidbill | that's zaptel.conf, zapata.conf and /proc/zaptel/* |
| 03:33.36 | *** join/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 03:33.46 | unpaidbill | that is odd that it detects the tdm2400p as the first span in proc, but 2nd with ztcfg |
| 03:33.50 | unpaidbill | hmm |
| 03:34.24 | unpaidbill | i guess i know why it's messed up now, at least |
| 03:34.32 | unpaidbill | somehow i missed that before, thanks tza! |
| 03:36.04 | spokra | agi question: trying to get adhearsion working.. doc says to add exten => _X.,1,AGI(agi://1.2.3.4) |
| 03:36.04 | spokra | to dialplan.. i get the following error |
| 03:36.04 | spokra | launch_script: Failed to execute '/var/lib/asterisk/agi-bin/agi:/127.0.0.1:4574': File does not exist. |
| 03:36.04 | spokra | is there something special i have to do to tell agi to connect via ip instead of a file? |
| 03:37.04 | *** join/#asterisk nicox_ (n=nicox@212-183-43-223.adsl.highway.telekom.at) |
| 03:37.22 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:48.40 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:49.49 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 03:57.57 | ManxPower | spokra: did you do a "core show applications like AGI"? |
| 03:58.15 | ManxPower | and perhaps a "show application AGI" |
| 03:58.52 | [TK]D-Fender | ManxPower, Doesn't show the syntax, and it does look familiar. |
| 03:59.11 | ManxPower | [TK]D-Fender: pretty crappy application docs then |
| 03:59.20 | [TK]D-Fender | ManxPower, yup... not a winner. |
| 03:59.37 | ManxPower | fastagi should still work |
| 04:00.34 | *** join/#asterisk techie (n=techie@adsl-76-214-9-124.dsl.lsan03.sbcglobal.net) |
| 04:00.36 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:02.05 | spokra | manxpower ya i get agi as an application,, |
| 04:02.38 | spokra | found part of the problem... the adhearsion doc has a typo!! should be agi://127.0.0.1 |
| 04:02.46 | spokra | not agi/127.0.0.1 |
| 04:03.37 | *** join/#asterisk mackes-Office (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 04:17.22 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
| 04:25.06 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 04:27.38 | Nasra | hello... |
| 04:28.02 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:32.13 | deeperror | 3OT |
| 04:32.16 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 04:39.33 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 04:48.50 | *** join/#asterisk phy (n=phy@funky.monkey.org) |
| 05:03.21 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 05:07.54 | L|NUX | codefreeze-lap: y0 |
| 05:28.17 | codefreeze-lap | L|NUX: hi |
| 05:28.53 | codefreeze-lap | sorry for the delay-- wind is blowing things around in the back yard. Had to go secure things |
| 05:29.20 | codefreeze-lap | (At times, I thought I might need the securing) |
| 05:30.45 | codefreeze-lap | but, sorry, I'm tired, and I think I'm going to bed |
| 05:41.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:44.53 | Nasra | kind of quiet tonight |
| 05:47.53 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 05:48.17 | trnzmeta | guys what is the international dialout for US/canada? |
| 05:48.22 | toresbe | +1 |
| 05:48.24 | trnzmeta | to get out of country |
| 05:48.49 | toresbe | I dunno. 00 is the standard, so the US probably doesn't use that. |
| 05:49.02 | trnzmeta | well that's the thing what is the + in US/canada |
| 05:49.45 | toresbe | If you're in an area with good enough infrastructure, just turn the crank and ask for the operator to give you long-distance, and long-distance to give you International. |
| 05:50.04 | toresbe | If you time it right, you may get a slot on an undersea copper wire. |
| 05:50.26 | trnzmeta | everyone is a comedian... damn yanks |
| 05:50.40 | toresbe | I'm Norwegian. |
| 05:51.01 | trnzmeta | same thing, in the northen hemisphere |
| 05:51.23 | toresbe | yeah, Norwegian, American... same-same... |
| 05:51.45 | *** join/#asterisk apollonx (n=admin@193.19.189.38.STATIC.ISP.KZ) |
| 05:57.47 | L|NUX | codefreeze-lap : ok |
| 06:01.31 | *** join/#asterisk xnixan (n=xnixan@196.218.222.117) |
| 06:04.21 | *** join/#asterisk keulin (n=cray@80.15.251.6) |
| 06:06.42 | *** join/#asterisk nick125 (i=nick@pdpc/supporter/student/nick125) |
| 06:18.20 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
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| 06:26.34 | trnzmeta | once more for dummies |
| 06:26.46 | trnzmeta | if I'm in US/canada dialing to AUS |
| 06:27.20 | trnzmeta | 011 61..... |
| 06:27.52 | *** join/#asterisk keulin (n=cray@80.15.251.6) |
| 06:28.40 | JT | trnzmeta: is there a question there somewhere? |
| 06:29.04 | trnzmeta | oh if I'm in US/Canada |
| 06:29.09 | trnzmeta | to dial internationally out |
| 06:29.14 | trnzmeta | it's 011 ... .... |
| 06:29.39 | JT | yes. |
| 06:29.56 | trnzmeta | cool, it's dialing the correct number, just something in the way :( |
| 06:30.00 | JT | 011 is the international dialling prefix in the us |
| 06:30.15 | JT | what's the whole number? |
| 06:30.51 | trnzmeta | 011 61 421 xxx xxx |
| 06:31.14 | trnzmeta | I have an IVR script dialing out from US/Canada |
| 06:31.18 | JT | that should work in theory |
| 06:31.31 | trnzmeta | the funny thing is... it's saying something is picking it up |
| 06:31.46 | trnzmeta | so either someone or comp |
| 06:36.12 | trnzmeta | hmm I will now use test via skype out number |
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| 06:50.57 | kamanashisroy | hi, back to category inheritance .. is it possible to mix static config file and realtime with inheritance ? I think that will save a lot of headache in realtime .. |
| 06:53.54 | *** join/#asterisk aksyn (n=aksyn@78.86.127.229) |
| 06:54.00 | *** join/#asterisk XnOSX (i=4de20eec@gateway/web/ajax/mibbit.com/x-df041168b4b1d136) |
| 06:57.10 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58-8-6-34.revip2.asianet.co.th) |
| 06:57.53 | HaMYaI | how can I override the "SIP Display Info" in the sip peer configuration? |
| 06:58.54 | HaMYaI | I tried setting fromuser,fromdomain and callerid but they don seem to work |
| 06:59.36 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 07:03.20 | kamanashisroy | HaMYaI: are you talking about inheritance ? |
| 07:04.06 | kamanashisroy | sippeers => mysql,db,table,superconfig .. what about this ? |
| 07:04.40 | kamanashisroy | the database engine reads the superconfig and then loads data from database .. |
| 07:05.28 | kamanashisroy | when it finds a field like "extends" , it reads it's value and find it in the superconfig file .. and finally does inherit .. |
| 07:05.42 | kamanashisroy | is not it good idea to implement ? |
| 07:06.35 | *** join/#asterisk keulin (n=cray@AMontsouris-158-1-3-124.w90-46.abo.wanadoo.fr) |
| 07:07.27 | *** part/#asterisk XnOSX (i=4de20eec@gateway/web/ajax/mibbit.com/x-df041168b4b1d136) |
| 07:08.03 | phy | basic question: can you fork contexts in a dialplan? i.e. have Call between A and B want to initiate call between B and C? |
| 07:08.39 | phy | or more generally have Call between A and B want to initiate call between B and C? |
| 07:08.55 | phy | or more generally have Call between A and B want to initiate call between C and D? |
| 07:09.27 | kamanashisroy | phy: you can use meetme, queue or finally you can write an agi scrip that will put a callfile in exact place .. :) |
| 07:10.43 | phy | kamanashisroy: awesome! thanks very much. |
| 07:14.35 | HaMYaI | HaMYaI: I am trying to send a call through one of the sip providers here, but I had error status 400 invalid from, or 403 forbidden |
| 07:15.09 | HaMYaI | thats for kamanashisroy =) |
| 07:16.28 | HaMYaI | kamanashisroy: I am using tcpdump to analyse sip headers for successful calls using eyebeam softphone |
| 07:16.51 | HaMYaI | the headers are different |
| 07:17.12 | *** join/#asterisk EugenA (n=eugen@dialbs-213-023-178-154.static.arcor-ip.net) |
| 07:17.39 | *** join/#asterisk Dr-Linux (n=somethin@117.20.21.66) |
| 07:17.52 | kamanashisroy | HaMYaI: :) I am not the right person to help you .. do you have different proxy and realm ? |
| 07:18.35 | EugenA | what codecs do you use for your calls? |
| 07:18.52 | Dr-Linux | questoin, I'm using PRIs, i can see on console its B channel auto get refreshed after some time, can someone tell me why is this and where i can see the info about this? |
| 07:19.06 | Dr-Linux | wether telco is doing this or Asterisk is doing this? |
| 07:19.31 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 07:19.44 | HaMYaI | kamanashisroy: the provider seems to accept the proxy ip as the callerid, so I try to modify fromuser,fromdomain,callerid to match their requirements |
| 07:20.04 | HaMYaI | EugenA: my codecs? |
| 07:20.39 | Dr-Linux | any clue? |
| 07:21.06 | EugenA | HaMYaI, i'd like to improve the quality.. now gsm is used for all calls - it is not the best for quality, right? |
| 07:22.32 | HaMYaI | kamanashisroy: if those do not exist, the headers are taken from what's defined in the [general] section |
| 07:22.41 | EugenA | my voip-provider supports also G.711 (64 kbps), how do i enable it for me? |
| 07:22.58 | HaMYaI | EugenA: try to find the comparison chart probaly |
| 07:23.14 | trnzmeta | guys: I'm about to purchase a skypein number to test a few things out |
| 07:23.27 | trnzmeta | for alll intensive purposes is buying a number in US the same as canada |
| 07:23.32 | HaMYaI | EugenA: I normally use g729 or g723 to minmize the loads |
| 07:23.39 | trnzmeta | simply because there is no country for canada here |
| 07:23.53 | kamanashisroy | note that fromdomain and fromuser is used when you are doing outgoing call to a gateway .. |
| 07:24.16 | kamanashisroy | HaMYaI: ^^^ This is what stated in sip.conf comment |
| 07:24.18 | HaMYaI | EugenA: allow=alaw and allow=ulaw |
| 07:24.38 | EugenA | HaMYaI, in sip.conf? |
| 07:24.49 | HaMYaI | kamanashisroy: right, didn't know that |
| 07:25.37 | HaMYaI | EugenA: yeah, in your sip peers and clients config |
| 07:26.23 | kamanashisroy | HaMYaI: are you configuring it to accept call from clients , I mean for incoming calls ? I think in that case you do not need to use fromuser or fromdomain .. |
| 07:26.39 | Dr-Linux | kamanashisroy: can you answer my quesiton? |
| 07:26.41 | *** join/#asterisk SparFux (n=raoul@e182016058.adsl.alicedsl.de) |
| 07:27.02 | HaMYaI | kamanashisroy: nope, for peers only |
| 07:28.58 | kamanashisroy | Dr-Linux: no .. sorry .. |
| 07:29.11 | EugenA | how do i initiate a call from CLI? |
| 07:30.00 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:30.23 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:30.49 | kamanashisroy | EugenA: you need to enable chan_oss module to do that .. |
| 07:31.02 | kamanashisroy | EugenA: chan_oss will register a dial command ! |
| 07:31.12 | SparFux | Yes, or chan_alsa. |
| 07:31.33 | EugenA | how do i check if this is already enabled? |
| 07:31.39 | SparFux | show modules |
| 07:31.48 | SparFux | command "show modules" |
| 07:32.30 | Dr-Linux | SparFux: do you? |
| 07:32.32 | SparFux | I have trouble using misdn in kernel 2.6.25. From 2.6.24 on it is broken. The git version compiles, but it crashes the system. It is rebooting immediately after module load of misdn. |
| 07:32.57 | SparFux | Dr-Linux: Do I what? |
| 07:33.26 | Dr-Linux | SparFux: opss you joined now, but i had a question and was looking someone to answer that |
| 07:33.33 | Dr-Linux | maybe you know, lemme retype |
| 07:33.44 | Dr-Linux | questoin, I'm using PRIs, i can see on console its B channel auto get refreshed after some time, can someone tell me why is this and where i can see the info about this? |
| 07:34.02 | EugenA | so, it is already enabled |
| 07:34.06 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:34.30 | SparFux | Dr-Linux: I cannot test it right here. My misdn is broken atm. Do you use misdn? |
| 07:34.35 | EugenA | chan_oss |
| 07:34.47 | Dr-Linux | SparFux: i'm using T1 |
| 07:34.57 | SparFux | Which kernel driver? |
| 07:35.58 | *** join/#asterisk Psychobilly (n=Fuzz@athedsl-4406470.home.otenet.gr) |
| 07:36.35 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 07:40.34 | Dr-Linux | 2.6 |
| 07:42.49 | EugenA | dialplan: how do i say "all numbers"? |
| 07:42.55 | EugenA | _X ? |
| 07:42.59 | EugenA | or X ? |
| 07:43.09 | *** join/#asterisk synthetiq (n=roger@unl201395.nl.customer.alter.net) |
| 07:43.11 | Psychobilly | _X. |
| 07:45.09 | EugenA | now i have only one line i context "home": exten => _X,1,Dial,SIP/${EXTEN}@smsdiscount|90|r |
| 07:45.48 | EugenA | CLI: "dial [mynumber]@home" doesn't work |
| 07:46.08 | EugenA | No such extension 'NUMBER' in context 'home' |
| 07:46.32 | Psychobilly | its _X. |
| 07:46.52 | Psychobilly | not _X |
| 07:46.57 | Psychobilly | http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns |
| 07:47.07 | EugenA | oh.. with dot |
| 07:47.31 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
| 07:51.18 | SparFux | Dr-Linux: which isdn driver are you using? |
| 07:51.28 | SparFux | Dr-Linux: and is it 2.6.24 or 2.6.25? |
| 07:53.32 | *** join/#asterisk gr0mit (n=tim@CSC.ge4-0-0.401.ar1.BBS1.gblx.net) |
| 07:53.39 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 07:54.00 | JT | Dr-Linux: completely normal |
| 07:54.25 | JT | SparFux: he said T1. it would be Zaptel and Libpri |
| 07:55.13 | SparFux | JT: oh, didn't know that. I thought T1 was something about isdn. |
| 07:55.43 | SparFux | That's crap. My whole asterisk is unusable because misdn crashes. |
| 07:56.01 | JT | pri is a form of isdn, but not all T1s use PRI signalling |
| 07:56.05 | JT | yeah misdn is rubbish |
| 07:56.13 | EugenA | can i somehow interact with that console call? |
| 07:57.54 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:59.03 | SparFux | JT: yes, but I need capi. I have old windows software I use with wine and it is capi. |
| 07:59.22 | *** join/#asterisk tiav (n=tiav@ivr94-4-82-227-121-53.fbx.proxad.net) |
| 07:59.28 | SparFux | otherwise I would just use some other stuff, too. |
| 08:01.57 | Dr-Linux | JT: can you please answer my question? |
| 08:02.16 | Dr-Linux | JT: I'm using PRIs, i can see on console its B channel auto get refreshed after some time, can someone tell me why is this and where i can see the info about this? |
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| 08:05.34 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:06.00 | whymarkwh | hi there does anyone know where i can ge the latest version of astapi for dialing from outlook running windows 64 bit cant get it to install or does anyone know of an other way to dial from outlook,any help welcome? |
| 08:06.53 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
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| 08:21.44 | *** join/#asterisk LuisTorres (n=chatzill@bl9-248-60.dsl.telepac.pt) |
| 08:26.01 | LuisTorres | Hi |
| 08:26.22 | whymarkwh | hi there does anyone know where i can ge the latest version of astapi for dialing from outlook running windows 64 bit cant get it to install or does anyone know of an other way to dial from outlook,any help welcome? |
| 08:26.28 | whymarkwh | hi there LuisTorres |
| 08:28.10 | JT | Dr-Linux: i already answered it. |
| 08:28.17 | JT | B channel resets are completely normal |
| 08:28.23 | JT | perhaps do a google search |
| 08:30.59 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 08:32.14 | *** join/#asterisk EugenA (n=eugen@dialbs-213-023-178-154.static.arcor-ip.net) |
| 08:32.36 | Rico29 | hi |
| 08:32.57 | Rico29 | does somebody work with Cisco IP phones with SIP firmware ? |
| 08:33.17 | EugenA | i'd like to make a call with "ulaw", but i see in console: |
| 08:33.23 | Rico29 | i have trouble with provisioning |
| 08:33.27 | EugenA | > requested format = ulaw, |
| 08:33.28 | EugenA | <PROTECTED> |
| 08:33.28 | EugenA | <PROTECTED> |
| 08:33.28 | EugenA | <PROTECTED> |
| 08:33.34 | EugenA | what does it mean? |
| 08:34.11 | Rico29 | it means your comm is in gsm format |
| 08:34.14 | Rico29 | :) |
| 08:34.28 | EugenA | yea.. why? |
| 08:34.39 | EugenA | if > requested format = ulaw |
| 08:34.59 | Rico29 | mmh, did you "disallow : gsm" in sip.conf ? |
| 08:35.13 | Rico29 | (or in database if realtime) |
| 08:36.27 | EugenA | no |
| 08:37.02 | Rico29 | ar "disallow:all" and "allow:ulaw" |
| 08:39.35 | Rico29 | can somebody help me with Cisco provisioning ? |
| 08:41.00 | Rico29 | comment je dezip un .tar ? |
| 08:41.12 | Rico29 | oups, bad win |
| 08:41.54 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 08:44.07 | *** join/#asterisk af_ (n=getsmart@88-149-241-109.dynamic.ngi.it) |
| 08:44.21 | christophocles | sorry for the noob question, but is it possible to use asterisk for a voicemail-only system to receive calls from landlines, using *no* specialized hardware (similar to service found at evoice.com) ? |
| 08:45.07 | tuxx- | ye, you could :P |
| 08:45.10 | EugenA | so, i have now disallow=all in sip.conf ([general]), but still: -- Call accepted by 192.168.1.157 (format gsm) |
| 08:46.22 | christophocles | tuxx- that makes me very hopeful :) i just have no idea how i would create a new phone number and have it routed to my PC over the internet... i imagine i have to pay the telco or some company to make a telephone number for me |
| 08:47.01 | christophocles | i'm just beginning to read the o'reilly book on asterisk but it seems very enterprise-oriented and all i want is a basic voicemail system to email me my messages |
| 08:47.06 | *** join/#asterisk Bronislav (n=null@net062033025050.pskovline.ru) |
| 08:48.07 | tuxx- | hmye, you need a company that can hook you up to the PSTN and give you some numbers to config on your asterisk server |
| 08:48.16 | tuxx- | i only know dutch company's that do that though :+ |
| 08:48.22 | tuxx- | so i won't be much of a help there :) |
| 08:49.13 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 08:49.59 | christophocles | well thanks, at least now i know it's possible |
| 08:53.12 | *** part/#asterisk SparFux (n=raoul@e182016058.adsl.alicedsl.de) |
| 08:53.14 | gr0mit | christophocles, which country are you in? I can provide UK numbers etc |
| 08:53.22 | christophocles | i live in the USA |
| 08:53.26 | gr0mit | ah. |
| 08:53.45 | gr0mit | ok, and you will want US numbers then |
| 08:53.48 | gr0mit | ? |
| 08:54.01 | christophocles | yes, I need USA numbers, but the area code does not matter |
| 08:54.34 | gr0mit | well you have a big choice, but i can;t help you from here unlessyou want UK stuff |
| 08:54.36 | Dr-Linux | again question |
| 08:54.39 | Dr-Linux | JT: I'm using PRIs, i can see on console its B channel auto get refreshed after some time, can someone tell me why is this and where i can see the info about this? |
| 08:55.04 | *** join/#asterisk Vec (n=Vec@host-87-74-7-57.dslgb.com) |
| 08:56.52 | Vec | Hi does anyone have experiance with large asterisk deployments, i.e. approx 240 simultanious calls 8 PRIs ? Just want to get an idea of how well asterisk deals with that config, and what hardware you used ? |
| 09:01.22 | synthetiq | as long as you dong do sip registrations, it does very well |
| 09:01.39 | synthetiq | funny how people try to sell stuff in here |
| 09:01.54 | Vec | sell stuff ? |
| 09:02.11 | synthetiq | i liken it to people selling you knock off sunglasses in morrocco |
| 09:02.26 | synthetiq | sell DID numbers and what not |
| 09:03.23 | whymarkwh | hi there does anyone know where i can ge the latest version of astapi for dialing from outlook running windows 64 bit cant get it to install or does anyone know of an other way to dial from outlook,any help welcome? |
| 09:05.22 | gr0mit | synthetiq, why is it a problem to respond to a noob's questions with an offer of help, commercial or otherwise? |
| 09:06.23 | synthetiq | help is not necessarily selling people items |
| 09:06.35 | gr0mit | but it can be |
| 09:06.47 | gr0mit | if that is what they are looking for |
| 09:06.47 | JT | Dr-Linux: can you stop fucking repeating the same shit over and over again and addressing it to me? |
| 09:06.54 | JT | it is really giving me the shits |
| 09:06.59 | JT | it is OKAY, GOOGLE IT |
| 09:07.01 | JT | end of story. |
| 09:07.01 | gr0mit | which in this question it was! |
| 09:07.19 | synthetiq | he wanted to create a voicemail system which he can do buy plugging in a fxs card into his home line and seetingup voicemail |
| 09:07.58 | gr0mit | well, he wanted to use no specialised hardware |
| 09:08.29 | gr0mit | and he would need an fxo not an fxs ;-) |
| 09:09.18 | christophocles | if anyone's familiar with evoice or efax services, that is exactly what i want, minus the proprietary formats |
| 09:09.36 | Dr-Linux | gr0mit: can you help me with my question? |
| 09:09.51 | gr0mit | the question you were bugging JT with? |
| 09:10.06 | christophocles | just a phone number anyone can call leave a message or send me a fax, and it is emailed to me in a friendly format like .mp3 or .pdf |
| 09:10.07 | Dr-Linux | gr0mit: yes |
| 09:10.35 | gr0mit | christophocles, there are tons of services which will do this for you - just google for it |
| 09:11.08 | JT | Dr-Linux: can you learn to be less lazy? |
| 09:11.13 | JT | go to www.google.com |
| 09:11.21 | JT | type in: asterisk b channel resets |
| 09:11.25 | JT | click search |
| 09:11.26 | gr0mit | Dr-Linux, which country are you in ? Some PRIs seem to reset the D channel regularly |
| 09:11.46 | JT | it's usually zaptel performing the reset |
| 09:11.48 | gr0mit | UK does not, France seems to. |
| 09:11.59 | gr0mit | but it seems normal |
| 09:11.59 | JT | but it can come from either end |
| 09:12.04 | christophocles | gr0mit, ok i will search more, i just figured most of them would charge me money and it would be more interesting to learn to do it myself.... i'm only just beginning to read up on asterisk |
| 09:12.08 | *** join/#asterisk echos (n=echos@adsl-99-136-105-254.dsl.lsan03.sbcglobal.net) |
| 09:12.13 | echos | This is a bit off topic but someone here should know it. On a POTS line how do I find the number attached to the line? |
| 09:12.38 | JT | echos: call the ani number relevant for your phone network |
| 09:12.39 | synthetiq | echos call from the lien to your cellphone :) |
| 09:12.40 | gr0mit | christophocles, well for voicemail you can do it yourself easily |
| 09:12.54 | gr0mit | but faxes present a challenge |
| 09:13.01 | Dr-Linux | gr0mit: US |
| 09:13.02 | *** join/#asterisk DotHack (n=chatzill@mdxemse.mosadex.com) |
| 09:13.05 | echos | ani number? |
| 09:13.06 | DotHack | hi all!!! |
| 09:13.14 | JT | automatic number identifictation |
| 09:13.17 | christophocles | it seems like the hard part is linking a telephone number to my asterisk server, since i have to deal with a telco |
| 09:13.21 | JT | i have no idea what your ani number is |
| 09:13.22 | DotHack | Question: who has a nice in queue waiting sound? |
| 09:13.22 | gr0mit | well my PRI in the US does not seem to suffer from this |
| 09:13.38 | echos | well some of the lines don't have telephone service so I need to dial the service number |
| 09:13.43 | JT | Dr-Linux: can you read? just ignore the damn resets unless it's causing a problem |
| 09:13.59 | DotHack | Something like a fast dialtone? |
| 09:14.07 | echos | JT: how do I got about finding it? |
| 09:14.15 | JT | gr0mit: also depends on your settings and zaptel version |
| 09:14.25 | Dr-Linux | gr0mit: my pri doesn't have any problem, but it auto get refreshed after sometime, but i just want to know the reason, i never had any problem though |
| 09:14.29 | gr0mit | christophocles, it all depends on the application. If it is just for fun at home, get yourself a nasty X100P for $15 |
| 09:14.39 | JT | Dr-Linux: beat a dead horse much? |
| 09:15.13 | gr0mit | if you want to run a biz on it with multiple lines you should consider a PRI or paying a service provider to do it |
| 09:15.19 | JT | echos: search for the ani number for your telco/country, or ask a local telco tech/contractor |
| 09:16.10 | Dr-Linux | JT: sorry i didn't understand |
| 09:16.38 | JT | Dr-Linux: it's just resetting unused b channels |
| 09:16.47 | JT | if you want more information use google.com |
| 09:17.06 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 09:17.25 | Dr-Linux | JT: it's being reset by telco or Asterisk is doing this? |
| 09:17.35 | JT | Dr-Linux: asking us the same question over and over again is not going to answer your question, it is just making us angry |
| 09:17.40 | JT | probably zaptel |
| 09:17.42 | Dr-Linux | JT: I can't find anything on google |
| 09:17.53 | JT | i can find heaps |
| 09:17.55 | penguinFunk | why would i get NOTICE[7495]: chan_sip.c:15655 sip_poke_noanswer: Peer '403' is now UNREACHABLE! Last qualify: 0 |
| 09:18.02 | tzafrir_laptop | (by Asterisk, not by Zaptel) |
| 09:18.02 | penguinFunk | when the host is pingable |
| 09:18.05 | penguinFunk | ? |
| 09:18.17 | penguinFunk | in both directions |
| 09:18.19 | tzafrir_laptop | If so - resetinterval=never in zapata.conf? |
| 09:19.06 | penguinFunk | also, 'sip show peers' show: 403/403 192.168.207.3 D 5060 UNREACHABLE |
| 09:19.08 | JT | Dr-Linux: this is a really stupid thing to fuss over if there is no problem, i can't imagine how much you would fuss if there was a real problem |
| 09:20.24 | penguinFunk | is there anyway to force a peer poke ? |
| 09:22.29 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 09:22.35 | kannan | hello all |
| 09:23.22 | kannan | Is there any G729a hardware card , that may be installed in lieu of the software licenses? I had a guy telling me to get this card for him, but I havent heard of anything like this |
| 09:23.25 | christophocles | gr0mit, surely it would be cheaper to purchase only the phone number and have it point to my asterisk server over the internet (rather than purchasing a landline and linking it to my PC using that X100P card)... can you send me a link to such a provider in the UK, so that I might try to find a similar one in the USA? |
| 09:24.04 | gr0mit | christophocles, there are some that offer free incoming numbers in USA |
| 09:24.11 | gr0mit | i can't recall though. |
| 09:24.20 | gr0mit | e.g. in UK www.sipgate.co.uk |
| 09:24.25 | gr0mit | www.gradwell.net |
| 09:24.58 | gr0mit | for the commerical police here - i am not in any way involved with either |
| 09:25.14 | gr0mit | however, for fax you will still have a prolem |
| 09:25.25 | christophocles | i may continue using the free efax service for that |
| 09:25.30 | gr0mit | ok |
| 09:25.45 | christophocles | although i hate having to use their software just to view my faxes |
| 09:26.04 | gr0mit | well, I am sure there are others |
| 09:26.06 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:28.13 | *** join/#asterisk xenonex (n=xenonex@89.218.235.221) |
| 09:29.01 | kannan | for 3 analog phone lines , will x100p card do the job adequately, one for a lanline one for fax and one for a FCt(GSm cellular) |
| 09:29.04 | JT | you could always pay, if you don't like that :) |
| 09:29.13 | kannan | or better to go for a 4 port digium vard? |
| 09:29.17 | penguinFunk | anyone? |
| 09:29.23 | JT | you can't buy real X100P cards anymore |
| 09:29.33 | JT | they were never that good anyway |
| 09:31.39 | kannan | JT , thanks, what about x100p.com, it is clone or the original? |
| 09:32.00 | JT | it is fake, like all the rest poporting to be new now |
| 09:32.08 | JT | the intel chipset is discontinued |
| 09:32.28 | JT | so a lot of the ones you buy now have clone ICs or seconds |
| 09:32.55 | kannan | JT thanks again, one more question is the TDM410 models the starting range , for price? |
| 09:33.24 | JT | umm i think so, haven't looked into the TDM410, must be a new release |
| 09:33.28 | JT | i try to avoic analogue |
| 09:33.30 | JT | avoid |
| 09:34.28 | kannan | well i cant use a PRI E-1 for now, its for a home office , so i amy not afford that much |
| 09:35.58 | JT | either get a pci card with enough ports on the 1 card, or look at ATAs or external gateways, but i doubt there are any economical 3 port FXO gateways |
| 09:38.09 | kannan | JT thanks, |
| 09:43.42 | kannan | bbl |
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| 09:53.17 | tzafrir_laptop | x100p.com is an original clone |
| 09:53.30 | tzafrir_laptop | or whatever |
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| 10:33.37 | Vec | Does anyone here have any experiance with large Asterisk deployments, eg 200 simultanious calls 8 PRIs? |
| 10:38.10 | xenonex | hello |
| 10:40.36 | *** part/#asterisk DotHack (n=chatzill@mdxemse.mosadex.com) |
| 10:55.06 | Dr-Linux | JT: I'm asking just for the learning purpose |
| 10:58.31 | synthetiq | ve c i answered your question ebfore |
| 10:58.55 | synthetiq | yes it does well as long is not doing registrations |
| 10:59.15 | synthetiq | i had 3 boxes with 8 pris each |
| 11:00.06 | gr0mit | synthetiq, that is one heck of a system! |
| 11:00.56 | synthetiq | ive seen bigger |
| 11:01.02 | synthetiq | but not with asterisk |
| 11:01.11 | *** join/#asterisk RoyK (n=Roy@ip-13-50-149-91.dialup.ice.no) |
| 11:01.11 | gr0mit | what was the application? |
| 11:01.19 | synthetiq | a school |
| 11:01.59 | gr0mit | wonders why a school needs 24 PRI |
| 11:08.45 | dominic1 | is it possible to tell my phones that if they dial a internal number they should use G.722 and if they dial to a external number they should use G711? |
| 11:11.05 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:18.09 | mvanbaak | dominic1: only allow g711 on the outbound connection in asterisk |
| 11:18.24 | mvanbaak | dominic1: if it's a sip link, specify it there, if it's a zap line do it there |
| 11:18.39 | RoyK | gr0mit: 24 PRI is quite decent :) |
| 11:19.04 | dominic1 | it's a msidn pri line |
| 11:19.19 | dominic1 | I call with my sip phone |
| 11:19.24 | RoyK | msidn? |
| 11:19.28 | RoyK | ~msidn |
| 11:19.35 | RoyK | ~lart himself |
| 11:19.35 | jbot | raises middle finger to himself |
| 11:20.20 | dominic1 | currently I am using G.711 for internal and external. Where can I specify that the system should use G.711 for outgoing calls? |
| 11:20.28 | dominic1 | ${SIP_CODEC} ??? |
| 11:30.37 | Vec | synthetiq : thanks, what boxes where they ? CPU RAM etc ? any transcoding ? |
| 11:33.51 | RoyK | gr0mit: a BIG school :) |
| 11:34.17 | gr0mit | we have 800 people here and have 3 x PRI |
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| 11:59.26 | synthetiq | vec: g711, cpu 2x 3ghz 2gb ram |
| 11:59.42 | synthetiq | not transcoding |
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| 12:02.19 | RoyK | gr0mit: as I said - BIG school |
| 12:02.27 | RoyK | :) |
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| 12:02.44 | RoyK | synthetiq: 1 PRI? |
| 12:03.21 | synthetiq | 8 pri's per machine |
| 12:03.57 | RoyK | synthetiq: how many people will this serve? |
| 12:04.21 | RoyK | what's going to spend time in the cpu, is zaptel |
| 12:04.29 | RoyK | zaptel doesn't scale too well |
| 12:04.36 | RoyK | digium or sangoma hardware? |
| 12:04.47 | RoyK | sangoma scales a _lot_ better than digium |
| 12:05.06 | gr0mit | always chooses sangoma |
| 12:05.34 | deeperror | is placing a104d into production today |
| 12:05.34 | synthetiq | 196 |
| 12:05.44 | gr0mit | so synthetiq - why does a school need 24 x PRI?! |
| 12:05.54 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-557e77fae5917744) |
| 12:05.59 | synthetiq | digium hardware |
| 12:06.23 | synthetiq | gromit it was a large school |
| 12:06.24 | RoyK | synthetiq: 24 PRI is 552 channels |
| 12:06.32 | RoyK | 552 concurrent calls |
| 12:06.39 | synthetiq | 196 per machine roy |
| 12:07.04 | RoyK | E1? |
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| 12:07.09 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:07.12 | synthetiq | t1 |
| 12:07.32 | dominic1 | anybody knows how to use misdn? |
| 12:07.47 | dominic1 | where can I set the allowed codecs for misdn? |
| 12:08.02 | RoyK | synthetiq: 23+1 channels per t1, right? so 184 channels per server? |
| 12:08.25 | synthetiq | well yea |
| 12:08.48 | RoyK | how many employees does this school have? |
| 12:09.10 | synthetiq | its not about employees but dormed students |
| 12:09.36 | RoyK | ok |
| 12:09.47 | RoyK | students on SIP phones? |
| 12:10.17 | synthetiq | yes |
| 12:10.35 | RoyK | but - the question - 2x3GHz with 2GB RAM should be sufficient for this load - even with digium cards |
| 12:11.08 | RoyK | sangoma cards can be tuned down to transmitting 80 bytes per interrupt instead of zaptel/digium's fixed 8 bytes per interrupt |
| 12:11.22 | *** join/#asterisk wonderworld (n=service@dslb-084-062-030-192.pools.arcor-ip.net) |
| 12:11.25 | RoyK | the difference is quite large |
| 12:11.29 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 12:12.01 | wonderworld | hi, how would i uninstall the zaptrl driver? i installed from the 1.4 branch but I need 1.2 |
| 12:12.12 | wonderworld | is it safe to compile 1.2 and just reinstall? |
| 12:13.02 | RoyK | wonderworld: should work well |
| 12:13.08 | wonderworld | thanks |
| 12:13.19 | RoyK | find /usr/lib -name zap*so -exec rm -f {} \; |
| 12:13.22 | RoyK | might also work :P |
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| 12:14.21 | wonderworld | *might* might be not enough ;) |
| 12:15.38 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:15.39 | RoyK | wonderworld: with 8 byte frames, I had 100 concurrent calls on a single P4 3.0 with ~50% load |
| 12:17.39 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:26.04 | wonderworld | ok, i recompiled and installed 1.2. is there a way to check the version of the loaded driver to ensure that 1.2 is really running? |
| 12:26.56 | mvanbaak | show version |
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| 12:30.01 | wonderworld | mvanbaak: where would i do that? |
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| 12:34.58 | wonderworld | ok, i see, it should be run on the asterisk console, but asterisk 1.2 doesnt have this command. (running debian stable here) is there another way to check the zapata version? |
| 12:36.58 | tzafrir_laptop | show version |
| 12:37.10 | tzafrir_laptop | asterisk -V |
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| 12:38.40 | x86 | morning _ShrikE |
| 12:39.21 | _ShrikE | good morning |
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| 12:39.40 | ManxPower | you should be able to check the zaptel version by looking at dmesg for where zaptel loads |
| 12:39.57 | ManxPower | "show version" only shows the ASTERISK version |
| 12:40.25 | RoyK | wonderworld: ldconfig -v | grep zap :P |
| 12:40.43 | wonderworld | got it and it worked. thanks guys. |
| 12:41.17 | russellb | zaptel is not a library. |
| 12:41.24 | wonderworld | dmesg did it.... |
| 12:44.57 | dominic1 | I have a big problem |
| 12:46.00 | dominic1 | if I call a external phone, put them on hold and want them to transfer to internal colleagues, after pressing transfer the system hang up the call to my collegaues |
| 12:46.19 | dominic1 | can anybody tell me what my problem can be? |
| 12:46.28 | ManxPower | <PROTECTED> |
| 12:47.02 | ManxPower | You are pressing the Transfer button on your phone? What make/model is the phone? |
| 12:47.23 | tzafrir_laptop | cat /sys/module/zaptel/version # on any kernel version later than 2.6.11, IIRC |
| 12:47.23 | dominic1 | I am doing the transfer internal in the phone, it's a atxfer |
| 12:47.23 | dominic1 | it's a snom |
| 12:47.37 | dominic1 | if the external guy calls me and I do the same it works |
| 12:47.39 | ManxPower | dominic1: do you are not using # or ##? |
| 12:47.49 | dominic1 | but not if I call both |
| 12:47.54 | ManxPower | dominic1: well, I guess you need to look at the CLI output for a failed call. |
| 12:47.59 | [TK]D-Fender | dominic1: Then you're probably doing it wrong. Go read your manual again. |
| 12:48.02 | dominic1 | no I am not using # ## |
| 12:48.08 | ManxPower | dominic1: good. |
| 12:48.50 | [TK]D-Fender | ManxPower: That was really .... gibberish.... you clearly haven't reached coffee yet (or it into your blood-stream) |
| 12:48.52 | dominic1 | I am using the phonefunction. It's very strange that I only have this problem if I call both people internal and external |
| 12:49.19 | ManxPower | [TK]D-Fender: working on it. what part was gibberish? |
| 12:50.12 | [TK]D-Fender | ManxPower>dominic1: do you are not using # or ##? |
| 12:50.24 | [TK]D-Fender | actually.. not that bad... |
| 12:50.30 | ManxPower | [TK]D-Fender: *nod* That is gibberish. |
| 12:50.38 | [TK]D-Fender | ManxPower: Still... coffee... get to it! |
| 12:51.14 | ManxPower | dominic1: we are waiting for your pastebin |
| 12:51.39 | dominic1 | there is no error in the cli |
| 12:51.59 | ManxPower | dominic1: non-error messages can he helpful too. |
| 12:52.43 | RoyK | tzafrir_laptop: modinfo zaptel |
| 12:53.18 | [TK]D-Fender | dominic1: SIP DEBUG is your friend, although I'm still pretty sure you're just doing it wrong. |
| 12:53.42 | ManxPower | [TK]D-Fender: my guess is a context issue |
| 12:53.49 | dominic1 | hey fender, that's not my first installation, I am able to press two buttons on the phone |
| 12:54.02 | [TK]D-Fender | ManxPower: Hmm.. could be that on a blind transfer too actually... |
| 12:54.29 | tzafrir_laptop | RoyK, that assumes that the version installed is the version running. And that you don't have two zaptel.ko-s in the modules dir. Those assumptions are mostly correct |
| 12:54.40 | ManxPower | but since he has not provided the info, I'm going to go back to paying work. |
| 12:55.30 | dominic1 | context issue was a good idea |
| 12:57.57 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
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| 12:59.10 | dominic1 | my pastebin: http://www.pastebin.org/40297 |
| 13:03.12 | ManxPower | dominic1: We can't help you with trixbox |
| 13:03.23 | dominic1 | trixbox? |
| 13:04.03 | ManxPower | dominic1: looks like you are running some sort of GUI with all that complex dialplan stuff. |
| 13:04.04 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 13:04.14 | dominic1 | no, I wrote that |
| 13:04.18 | ManxPower | As [TK]D-Fender said, we also need a SIP DEBUG of a failed call. |
| 13:04.20 | dominic1 | :-D |
| 13:04.40 | ManxPower | dominic1: did it occur to you to simplify things before trying to diagnose and fix a problem? |
| 13:05.32 | dominic1 | the problem is that the system is already productive |
| 13:05.50 | dominic1 | we found the problem after months of using the system |
| 13:06.05 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 13:07.14 | *** join/#asterisk grandpapadot (n=anonymou@mail.heavylogic.com) |
| 13:09.43 | dominic1 | the debug: http://www.pastebin.org/40299 |
| 13:11.10 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.207.130) |
| 13:11.21 | webman | I'm trying to get an old grandstream phone working, if I remove the "secret" from the sip.conf, it will register, as soon as I add the secret line back, it gets a Auth Denied. I've triple checked entering the password on the GS webpage, anything else obvious I might have overlooked? |
| 13:11.38 | ManxPower | dominic1: You are trying to transfer to 00123456? |
| 13:11.51 | dominic1 | no, to chw |
| 13:12.04 | ManxPower | how do you enter chw on the telephone keypad? |
| 13:12.31 | dominic1 | xml or speeddial |
| 13:12.53 | ManxPower | Ah. I really can't help with this complicated problem. |
| 13:13.15 | ManxPower | what context is your exten => chw located in, in extensions.conf? |
| 13:13.25 | [TK]D-Fender | Failed SIP Transfer to non-existing extension chw in context dialout |
| 13:13.33 | [TK]D-Fender | ^^^ |
| 13:13.43 | yang | is AsteriskNOW the same as asterisk, only with a GUI ? |
| 13:14.11 | dominic1 | ah fender that's great I think I was blind |
| 13:14.22 | dominic1 | so I think my problem is the contextjump |
| 13:14.43 | ManxPower | yang: yes, IF you delete all the config files. |
| 13:14.48 | [TK]D-Fender | yang: AsteriskNOW is a DISTRO that more fully implements the GUI as well |
| 13:14.53 | ManxPower | well, at least close enough. |
| 13:15.09 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 13:15.41 | yang | [TK]D-Fender: but If I allready know asterisk, then its better to switch to asteriskNOW than the freepbx, if I require a GUI (In this case I could always monitor config files) |
| 13:16.00 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:16.18 | [TK]D-Fender | yang: You're welcome to do whatever you want. Just remember noone here will want to deal with problems you have with it. |
| 13:16.29 | grandpapadot | Hey TK, will the Polycom 501 not use standard POE from a switch? Does it require a special cable? |
| 13:16.38 | [TK]D-Fender | grandpapadot: indeed it does. |
| 13:17.12 | grandpapadot | Is it just a cable or is it the cable with the inverter box? Do you by chance know the part number? (thanks) |
| 13:17.12 | yang | [TK]D-Fender: but the entries are going to be written into "same" configs as usual asterisk |
| 13:17.35 | [TK]D-Fender | grandpapadot: this special cable has a little nugget in the middle with a circuit that does the PoE negotiation. The 30X/50X take power off the same wires IIRC, jsut that the phone can't demand it in the first place. |
| 13:17.49 | grandpapadot | Got it. Thanks! |
| 13:18.00 | [TK]D-Fender | yang: depends on your idea of "same". users.conf does WAY too much with the GUI, etc. |
| 13:18.21 | [TK]D-Fender | grandpapadot: www.telephonydepot.com |
| 13:18.32 | yang | [TK]D-Fender: but still easier to navigate these configs than the FreePBX which are totally rewritten... |
| 13:18.36 | grandpapadot | Thanks, TK |
| 13:18.36 | [TK]D-Fender | grandpapadot: Easy to find there. |
| 13:18.39 | webman | my grandstream registration problem is at http://pastebin.com/d5e8a6675 basically, with the secret line, it fails to regsiter... |
| 13:18.40 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 13:18.45 | [TK]D-Fender | grandpapadot: this for a phone you already own? |
| 13:19.52 | grandpapadot | [TK]D-Fender: No, customers, thousands of them. One of them just asked if the phone supported PoE direct, I said yes, oops. |
| 13:20.11 | [TK]D-Fender | grandpapadot: if they already have 501's, poor them |
| 13:20.30 | [TK]D-Fender | webman: if it wiorks without the "secret" line, then you didn't set the password on the phone |
| 13:21.20 | webman | TKD-Fender: I set it on the phone a dozen times, and each time I "update" and reboot.... you can't see the password as you type it in, so I even tried to copy and paste it |
| 13:21.48 | webman | in fact, I don't remove the password from the phone config when I remove the secret from the sip.conf |
| 13:22.15 | [TK]D-Fender | webman: I'd try reviewing your install instructions and getting a few more opinions. Its not an * problem... |
| 13:22.31 | webman | shouldn't asterisk send a "Auth Required" rather than a "Auth Denied" ?? |
| 13:24.05 | [TK]D-Fender | webman: If the config you showed is in effect, then its your phone. |
| 13:25.20 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:27.59 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 13:28.16 | hsv-al | hello |
| 13:30.07 | coppice | who was waiting for the HTC touch diamond to arrive? |
| 13:30.53 | anonymouz666 | not me. |
| 13:31.02 | ManxPower | webman: What SIP message code is "Auth Required"? |
| 13:31.12 | ManxPower | i.e. the 3 digit number |
| 13:31.13 | [TK]D-Fender | coppice: I'm jus keeping an eye out. I bought my HTC Touch 6 months ago. Unfortunately I'm stuck on CDMA. |
| 13:31.23 | [TK]D-Fender | ManxPower: 407 IIRC |
| 13:31.29 | ManxPower | [TK]D-Fender: oh, CDMA isn't THAT bad. |
| 13:31.36 | webman | It gets a 100 and then a 401 Unauthorized |
| 13:31.40 | coppice | the diamond looks pretty, but appears to have terrible battery life |
| 13:31.49 | coppice | its damned expensive |
| 13:32.07 | coppice | its about US$750 in our shops |
| 13:32.08 | [TK]D-Fender | ManxPower: Never said it was bad, its just that CDMA doesn't get all the models out there, and its provider-locked. |
| 13:32.19 | [TK]D-Fender | ManxPower: Which means its a LOT longer for most things to get up here. |
| 13:32.41 | ManxPower | Ah, OK. |
| 13:32.51 | [TK]D-Fender | coppice: Should have released the HTC Cubic Zirconium first ;) |
| 13:33.11 | ManxPower | I don't normally spend more than about $75 on a phone -- without contract |
| 13:33.15 | tzanger | [TK]D-Fender: hahahahaha |
| 13:33.36 | webman | ManxPower: when asterisk registers to another server, it gets a 407 Proxy Auth Required, but asterisk doesn't send that to the phone, so the phone never sends the auth info .... at least, from my little knowledge.... |
| 13:34.03 | coppice | the key trend for phones in 2008 seems to be to make them so thin the battery is totally inadequate |
| 13:34.15 | webman | BTW, debug is at http://pastebin.com/d5e8a6675 |
| 13:34.24 | [TK]D-Fender | ManxPower: I love my HTC Touch personally... set me back about $200 and I get unlimited browsing for $7/mo |
| 13:34.42 | [TK]D-Fender | coppice: how bad is "bad" in this case? |
| 13:34.54 | hsv-al | d-fender |
| 13:35.00 | hsv-al | I wasn't to fond of that SPA 3102 |
| 13:35.02 | coppice | dunno, but the early adopters seem pissed off |
| 13:35.10 | hsv-al | I need to find a PCI based solution |
| 13:35.25 | hsv-al | Does anyone know a good pci card with 1 fxo, 1 fxs, and possibly 1 ethernet jack in the $200-$300 range? |
| 13:35.27 | hsv-al | no hwec. |
| 13:35.38 | coppice | the HTC touch is cheap. this new thing is pricey |
| 13:35.39 | [TK]D-Fender | hsv-al: you won't find a TDm card with an ETHERNET port./ |
| 13:35.56 | [TK]D-Fender | hsv-al: And why aren't you "fond" of the SPA? You haven't owned one, have you? |
| 13:36.10 | hsv-al | no, but it seems like it's a wannabe asterisk |
| 13:36.18 | hsv-al | built into it....it doesnt need * to work. |
| 13:36.34 | [TK]D-Fender | coppice: then again, early adopters are prissy people. They have enormous expectations... though mind you for the price I suppose I would too... |
| 13:36.55 | [TK]D-Fender | hsv-al: its hardly smarter than any other phone out there. |
| 13:37.03 | ManxPower | [TK]D-Fender: Ah. I don't use the internet from my phone. |
| 13:37.08 | [TK]D-Fender | hsv-al: You're reading too much into it. It IS a very powerful device though. |
| 13:37.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:37.29 | ManxPower | I've tried it with previous phones -- seems like a lot of work compared to walking 10 ft to my laptop with an aircard. |
| 13:37.38 | [TK]D-Fender | ManxPower: I like the ability to hit up Google Maps on demand, checkk e-mail, and other randoma bits (weather,etc) |
| 13:37.41 | coppice | its really the trend though. ask any user of a neat new phone from the last year what its like and the reply is "the battery life sucks" |
| 13:38.00 | [TK]D-Fender | ManxPower: Yeah, if I had an aircard I'd probably want an EeePC or something too... |
| 13:38.09 | webman | coppice: I'll agree with that :) |
| 13:38.11 | ManxPower | [TK]D-Fender: Maybe the interface on those phones was not designed by the Marquis de Sade simself. |
| 13:38.27 | ManxPower | (unlike all the phones I've seen) |
| 13:38.35 | [TK]D-Fender | ManxPower: well Mine is a WinMo6 phone, not a Smart Phone (like the Moto Q) |
| 13:41.16 | coppice | I suppose this trend has changed perceptions of HTC products. It used to be that when you asked users about them the reply was always "great hardware, but the OS sucks". now the battery sucks. they are just trying to help out their pals at MS :-) |
| 13:41.26 | *** join/#asterisk GReggFT (n=GReggFT@fw1-p1.lexi.com) |
| 13:44.14 | hsv-al | d-fender |
| 13:44.17 | hsv-al | http://www.telephonydepot.com/product_p/105-050-tdm410p.htm |
| 13:44.23 | hsv-al | $175 |
| 13:44.42 | ManxPower | hsv-al: that's a decent price. |
| 13:45.11 | [TK]D-Fender | Digium TDM410P [+] View list of options I selected $234.50 |
| 13:45.16 | hsv-al | they hamanx, they have 410p, 410b, 401b, and 420b |
| 13:45.23 | [TK]D-Fender | hsv-al: you forgot to add the MODULES |
| 13:45.24 | hsv-al | which one would be optimal, for home hobby use? |
| 13:45.32 | hsv-al | s/hamanx/have |
| 13:46.25 | hsv-al | decided i dont need hwec for home use |
| 13:46.26 | [TK]D-Fender | hsv-al: http://www.telephonydepot.com/TDM400P_s/95.htm |
| 13:47.08 | ManxPower | hsv-al: you can get the HPEC for free anyway |
| 13:48.47 | [TK]D-Fender | And thats if Zaptels base EC doesn't cut it, and OSLEC either (free and easy) |
| 13:49.40 | hsv-al | 411B looks good so far - $259 |
| 13:50.17 | [TK]D-Fender | hsv-al: the "build it yourself card was 234. |
| 13:50.36 | [TK]D-Fender | hsv-al: The "premade" card seems to add for nothing |
| 13:52.19 | hsv-al | yep |
| 13:52.36 | hsv-al | I am looking now |
| 13:52.40 | hsv-al | comes out to $234.50 |
| 13:54.23 | hsv-al | d-fender, thanks. Ordering that now |
| 13:54.36 | hsv-al | 410P....# FXS / FXO Modules (TDM400 / TDM410):1 FXS Module (1 Port) + 1 FXO Module (1 Port) |
| 13:56.37 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:58.52 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 13:59.17 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 14:06.00 | [TK]D-Fender | hsv-al: personally I recommend against Zaptel FXS... |
| 14:06.25 | [TK]D-Fender | hsv-al: ATA's are cheaper an more flexible. |
| 14:06.27 | *** join/#asterisk deeperror (n=deeperro@adsl-76-226-148-247.dsl.sfldmi.sbcglobal.net) |
| 14:06.38 | [TK]D-Fender | hsv-al: Or you could put that money into a NICE phone. |
| 14:06.51 | tzafrir_laptop | [TK]D-Fender, OSLEC actually does seem to cut it |
| 14:07.03 | [TK]D-Fender | tzafrir_laptop: tahts what I hear |
| 14:07.29 | tzafrir_laptop | In fact, seems to "cut it" better than the "cheap ATAs" you mention |
| 14:08.13 | coppice | a lot of ATAs do very bodgy processing. |
| 14:08.55 | deeperror | what is missing if there are no colors on the cli? I see an option to disable but how would I enable? |
| 14:09.32 | tzafrir | deeperror, asterisk -r ? |
| 14:10.06 | deeperror | yea when i'm in there it is just black and white |
| 14:10.19 | deeperror | makes it hard to see things |
| 14:11.40 | deeperror | ll shows color |
| 14:13.28 | deeperror | tzafrir: could it be due to asterisk being started with asterisk and not safe_asterisk? |
| 14:21.36 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-239-103.balt.east.verizon.net) |
| 14:25.26 | *** join/#asterisk piper69 (n=chatzill@unaffiliated/piper69) |
| 14:25.29 | ix33 | hsv-al: i'm in hsv, AL as well |
| 14:25.31 | piper69 | Good morning all |
| 14:26.12 | penguinFunk | afternoon |
| 14:26.15 | ix33 | does asterisk do comfort noise generation? |
| 14:26.35 | hsv-al | ix33, where at |
| 14:26.41 | hsv-al | job type, residence area? |
| 14:26.49 | piper69 | I will be starting a computer business repair and i was wondering if i can use * to setup a system for my techs to use it to login and logout from their phone just to keep track of their time? |
| 14:26.52 | hsv-al | so I can id theft you.... |
| 14:27.28 | ix33 | piper69: actually, i just saw an example of exactly that somewhere |
| 14:28.05 | ix33 | hsv-al: computer security, madison |
| 14:28.06 | *** join/#asterisk anthm (n=anthm@adsl-76-229-202-214.dsl.milwwi.sbcglobal.net) |
| 14:28.13 | deeperror | piper69: seems like that would be a snap depending on how much you need it to do |
| 14:28.14 | tzafrir_laptop | deeperror, it's a bug of asterisk that when started as a daemon, does not provide colors |
| 14:28.17 | hsv-al | neato |
| 14:28.22 | hsv-al | im almost done with mis @ uah |
| 14:28.27 | ix33 | piper69: it was linked off of voip-info in the articles dealing with AMI |
| 14:28.34 | deeperror | so as a daemon is when it's ran as just asterisk? |
| 14:28.35 | hsv-al | gonna apply to CMU distance learning , masters mis |
| 14:28.38 | hsv-al | student for life! |
| 14:28.47 | tzafrir_laptop | fixed in 1.6 and in Debian |
| 14:29.05 | ix33 | hsv-al: actually i just finished my MSwE last semester |
| 14:29.08 | ix33 | hsv-al: so i can't claim to be a lifetime student any more |
| 14:29.29 | ix33 | hsv-al: you doing * support locally or something? |
| 14:29.33 | hsv-al | naw |
| 14:29.44 | hsv-al | cisco/win2k3 network admin defense company |
| 14:29.59 | ix33 | hsv-al: in research park? |
| 14:30.06 | hsv-al | close, but no |
| 14:30.18 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-5fbe840378d1696d) |
| 14:30.21 | hsv-al | just easy maintenence of win2k3 servers, basic pix's at each site, vpn wans... |
| 14:30.27 | hsv-al | desktop support for corp employees, simple crap |
| 14:30.47 | ix33 | hsv-al: cool. small world |
| 14:31.05 | hsv-al | was gonna go for ccsp next |
| 14:31.10 | hsv-al | but dont feel like becoming a cisco drone |
| 14:31.47 | ix33 | hsv-al: i kinda wish i had focused myself a bit more. i'm a jack of all trades/master of none |
| 14:32.03 | ix33 | hsv-al: hence why i'm lurking in #* this week ;) |
| 14:32.07 | hsv-al | only way out of the generalist type of path is to get ofcused in a particular area |
| 14:32.39 | piper69 | ix33: I am not sure what is the search criteria to use? also will it work if i setup trixbox |
| 14:32.50 | deeperror | tzafrir: got it fixed here by running safe_asterisk instead. However, I notice it using -f instead of -F any doc's on the pro/cons of forking? |
| 14:33.04 | hsv-al | deeperror, I had some issues yesterday using safe_asterisk |
| 14:33.13 | hsv-al | it would cause an infinite loop of * die'ing, and restarting |
| 14:33.23 | hsv-al | asterisk -vvvc ran it fine however. |
| 14:33.33 | deeperror | yea i've had that happen before |
| 14:33.34 | hsv-al | PID errors |
| 14:34.28 | piper69 | bbl |
| 14:34.32 | *** part/#asterisk piper69 (n=chatzill@unaffiliated/piper69) |
| 14:34.38 | deeperror | i just pushed a new system into production this am. Got my fingers crossed ha |
| 14:36.29 | deeperror | i'm curious about forking though because the last box seemed to have a new process on every call and I'm wondering if that would be better than a single process? |
| 14:38.07 | *** part/#asterisk mprebello (n=marcel@200.162.131.98) |
| 14:41.17 | hsv-al | d-fender im a penny pincher |
| 14:41.31 | hsv-al | alwaysintouch.com has the 410P + 1 fxs/fxo for $229 |
| 14:41.37 | hsv-al | $5.38 cheaper :) |
| 14:42.31 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 14:42.42 | ix33 | hsv-al: whatt kind of hardware are you looking for? |
| 14:42.55 | hsv-al | simple analog home solution |
| 14:43.03 | hsv-al | trying to optimize how much I can get for cheapest price. |
| 14:43.15 | *** join/#asterisk glaz (i=strke@glaciuz.com) |
| 14:44.30 | ix33 | i got one for about that price maybe |
| 14:44.40 | ix33 | i ordered from voipsupply.com i think |
| 14:45.01 | ix33 | my home * has been up or 14 weeks straight |
| 14:45.36 | hsv-al | using the rpath 1.0.2? |
| 14:45.40 | hsv-al | AN bundle? |
| 14:45.47 | ix33 | oh no |
| 14:45.52 | ix33 | debian etch |
| 14:46.49 | hsv-al | I have about 14 virtual machines running, bsd, ubuntu, gentoo, vmware at home, hoping to get each of them working w/ the card im gonna order |
| 14:46.59 | ix33 | i've played with AN, but digium folks keep scaring me away from it |
| 14:47.05 | hsv-al | eh? |
| 14:47.17 | ix33 | although, they were sales engineers trying to hock switchvox ;) |
| 14:47.25 | hsv-al | it works fine, i used it for a long time, until I decided to actually learn how to do everything CLI |
| 14:47.53 | ix33 | good to know |
| 14:48.17 | *** join/#asterisk l2trace99 (n=asd@75.112.133.235) |
| 14:48.35 | ix33 | i did my home * on top of a silly debian LAN server, so i just installed open source zaptel + * |
| 14:48.55 | ix33 | when i did my company install i just stuck with the same |
| 14:52.44 | *** join/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net) |
| 14:54.25 | *** join/#asterisk cjk (n=cjk@vodsl-11071.vo.lu) |
| 14:54.25 | ManxPower | This should go on bash.org: "hsv-al: I have about 14 virtual machines running, bsd, ubuntu, gentoo, vmware at home, hoping to get each of them working w/ the card im gonna order" |
| 14:54.39 | hsv-al | heh |
| 14:54.44 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-8f951db4ab9a31fe) |
| 14:54.44 | cjk | hi; is there a possibility to send a different ringtone? |
| 14:54.53 | ManxPower | cjk: yes |
| 14:54.59 | hsv-al | manx, its only for my own benefit |
| 14:55.06 | hsv-al | to get it running on various distros |
| 14:55.14 | cjk | ManxPower, any hint or link? |
| 14:55.28 | ManxPower | hsv-al: I don't care if it's for the King of Sudan, you are not going to get hardware cards to work well with a VM. |
| 14:55.55 | ManxPower | cjk: any hint on the make/model of phone, version of Asterisk, what ringtone you want to use, etc. |
| 14:55.58 | *** join/#asterisk gr0mit (n=tim@CSC.ge4-0-0.401.ar1.BBS1.gblx.net) |
| 14:56.00 | hsv-al | probably so, but I have a decent home machine |
| 14:56.04 | ix33 | cjk: usually that's a phone thing. |
| 14:56.19 | ix33 | efb |
| 14:56.26 | ManxPower | hsv-al: Best of luck with that |
| 14:56.39 | hsv-al | Q6600, 4gig |
| 14:56.43 | cjk | ManxPower, asterisk 1.4 |
| 14:56.49 | deeperror | cjk, send ring tone does that mean instead of ringing to a caller? |
| 14:56.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:57.05 | cjk | well instead as ring ring i want rang rang |
| 14:57.07 | ManxPower | cjk: You answered one of my three questions. |
| 14:57.28 | ix33 | hsv-al: i think the problem is not hardware, but how you will be able to 'pass through' raw PCI to one of your VM instances |
| 14:57.32 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:57.52 | hsv-al | workstation 6 allows it |
| 14:57.54 | ix33 | is there VM software that does that? |
| 14:57.55 | hsv-al | not the free ver. |
| 14:57.56 | ix33 | ah |
| 14:58.04 | ix33 | i knew vmware was great. |
| 14:58.07 | cjk | ManxPower, im looking for a phone independant way |
| 14:58.22 | ManxPower | cjk: there is no phone independent way -- each phone does it differently. |
| 14:58.58 | ManxPower | I am, of course, assuming you are using the correct word for what you want. "ringtone" is a specific thing, different from "ringback" |
| 14:59.29 | cjk | ManxPower, maybe i mean ringback |
| 14:59.39 | ManxPower | cjk: I can't help you if you don't know what you want. |
| 14:59.44 | cjk | its ringback |
| 14:59.44 | ManxPower | I'm going back to work. |
| 15:00.07 | ManxPower | You've already wasted 10 mins of my time by not even knowing what you want. |
| 15:00.20 | ManxPower | ttel |
| 15:00.47 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:01.11 | deeperror | cjk, answer() background(rangrang) dial() |
| 15:01.20 | cjk | ok |
| 15:01.23 | cjk | i will do that |
| 15:01.36 | [TK]D-Fender | doesn't do it... |
| 15:01.44 | [TK]D-Fender | background doesn't play through Dial. |
| 15:02.02 | *** join/#asterisk jjshoe_ (i=jjshoe@cpe-76-175-157-237.socal.res.rr.com) |
| 15:03.36 | deeperror | haha, so I guess a queue? |
| 15:04.20 | cjk | i can use the musiconhold parameter for dial |
| 15:04.24 | cjk | i thought there is a better way |
| 15:04.35 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 15:04.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:04.39 | [TK]D-Fender | cjk: What do you want for your ringing INDICATION? |
| 15:04.55 | cjk | ok user1 is dialing user2 thats on the phone |
| 15:05.00 | cjk | so user1 should know thos |
| 15:05.06 | cjk | so i change the ringing sound |
| 15:05.23 | [TK]D-Fender | cjk: "thats nice". Doesn't answer my question. |
| 15:05.33 | cjk | oh |
| 15:05.35 | cjk | just something different |
| 15:06.19 | ix33 | cjk: are you saying you want user1 to hear a different ring if user2 is on the phone so that user1 knows that user2 is occupied? |
| 15:06.27 | cjk | yes |
| 15:06.33 | [TK]D-Fender | cjk: make a recording of the ringing sound you'd like (LONG). Then set a MoH class with that as its only source. Then before you dial, check if they're on the phone. If so, dial with the m() options setting that class. |
| 15:06.54 | cjk | [TK]D-Fender, that was my plan |
| 15:07.02 | cjk | i thought there was something cooler |
| 15:07.07 | cjk | thanks for confirming it |
| 15:07.11 | ix33 | cjk: are you additionally saying that it doesn't matter what specific sound it is, as long as it's different? |
| 15:07.27 | cjk | ix33, yes |
| 15:08.05 | [TK]D-Fender | cjk: then forget about making a new recording, and jsut use tt-weasels |
| 15:08.21 | [TK]D-Fender | cjk: that would be "different" |
| 15:09.20 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:09.43 | deeperror | haha i use that all the time |
| 15:09.53 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 15:09.58 | gaetronik | Hi everyone |
| 15:09.58 | deeperror | users have taken over our phone system |
| 15:10.14 | ix33 | XX99 is 'tt-weasels' on every * install i've ever done |
| 15:10.19 | *** join/#asterisk mLx (n=mLx@194.242.123.226) |
| 15:11.17 | gaetronik | is the zaptel name change to dahdi efective at the day of know |
| 15:11.53 | cjk | hihi |
| 15:12.02 | cjk | what about playtones command |
| 15:12.05 | cjk | wouldnt that help me |
| 15:12.47 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:12.47 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 15:13.52 | *** join/#asterisk friedrich| (i=friedric@trem-servers.com) |
| 15:17.24 | [TK]D-Fender | cjk: not while you're DIALING. |
| 15:19.17 | cjk | [TK]D-Fender, well states differently on voip-info |
| 15:19.27 | [TK]D-Fender | cjk: Then go run with it. |
| 15:19.33 | cjk | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Playtones If you want tones to play when Dial()ing, make sure to use the & Dial() syntax ... |
| 15:19.39 | cjk | i give it a try |
| 15:19.46 | [TK]D-Fender | cjk: You spend far too little time actually trying anything it seems. |
| 15:20.49 | cjk | lets check |
| 15:22.26 | ix33 | why do my users say that they hear only 1/2 duplex when talking from a sip phone (IP330) to pstn via zap channel but not sip phone to sip phone internally? |
| 15:23.37 | ix33 | far end unaffected via PRI |
| 15:26.26 | *** join/#asterisk oej (n=olle@213.63.47.140) |
| 15:28.47 | hsv-al | heh |
| 15:28.57 | hsv-al | forums.digium.com is being spammed with fraudulent ulr's |
| 15:29.01 | hsv-al | url's in the jobs section |
| 15:31.16 | synthetiq | what type of urls |
| 15:31.37 | hsv-al | ID Theft sites, that offer shoes, clothes, stuff not even asterisk related |
| 15:31.45 | hsv-al | probably just a cess pool looking to collect credit cards. |
| 15:32.49 | synthetiq | <PROTECTED> |
| 15:32.50 | synthetiq | lol |
| 15:32.54 | hsv-al | heh |
| 15:33.04 | hsv-al | its not just the jobs forum, its all of them, but mostly in the jobs forum |
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| 15:35.00 | [TK]D-Fender | ix33: What card? |
| 15:35.41 | ix33 | it is a te121b |
| 15:35.52 | ix33 | the single-span pci-e |
| 15:35.57 | ix33 | w/echo cancellation |
| 15:36.09 | ix33 | echocancel=on in zapata.conf |
| 15:36.12 | [TK]D-Fender | ix33: Have you called Digium on this? |
| 15:36.30 | cjk | [TK]D-Fender, actually its working as expected |
| 15:36.44 | ix33 | [TK]D-Fender: no, why? is this a known problem? |
| 15:37.01 | [TK]D-Fender | ix33: No, almost never gets heard here. |
| 15:37.24 | ix33 | what almost never gets heard here? |
| 15:37.42 | [TK]D-Fender | ix33: half duplex issues |
| 15:38.10 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 15:38.14 | ix33 | well the far end (other side of pstn) says it sounds normal. i sort of thought this was a sip or phone thing... |
| 15:38.26 | ix33 | eh not sip; rtp |
| 15:38.33 | [TK]D-Fender | ixx: then you'd get hit phone-to-phone |
| 15:39.22 | ix33 | well the difference to me (i thought) was that phone to phone, the phones are RTP'ing and phone to pstn, one endpoint is actually * |
| 15:39.40 | *** join/#asterisk javar (n=javar@200.118.170.56) |
| 15:39.52 | ix33 | since far end doesn't perceive a problem... |
| 15:40.06 | DarKnesS_WolF | [TK]D-Fender: i did soper up ;-) |
| 15:40.22 | DarKnesS_WolF | [TK]D-Fender: now i know what i want to do clearly :-) |
| 15:40.34 | [TK]D-Fender | DarKnesS_WolF: Yeah, yur speeling iz much beeter |
| 15:40.58 | ix33 | is this correct reasoning or am i likely jumping too far ahead... |
| 15:41.05 | DarKnesS_WolF | [TK]D-Fender: actually yes my english sucks anyway :-) |
| 15:41.09 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:43.05 | DarKnesS_WolF | [TK]D-Fender: i did it with read cmd but in the CDR still not showing the correct number i want in the dst fild |
| 15:43.13 | DarKnesS_WolF | so i'll use mysql triggeror i'll use WaitExten |
| 15:44.09 | x86 | DarKnesS_WolF: don't be such a terrorist |
| 15:44.37 | x86 | grins evilly |
| 15:46.06 | *** join/#asterisk kamh (n=q@host-81-190-236-85.wroclaw.mm.pl) |
| 15:47.55 | *** join/#asterisk hexorx (n=joshr@63-211-239-34.teliax.com) |
| 15:50.06 | DarKnesS_WolF | x86: told ya i'll ride my camel over u :P |
| 15:52.45 | hsv-al | is 1 year replacement decent |
| 15:53.08 | hsv-al | thats all I'm going to get if I buy this |
| 15:54.58 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:56.23 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:59.09 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:02.18 | *** join/#asterisk isamar (i=1000@voice.maxirede.net) |
| 16:02.22 | isamar | hi folks |
| 16:05.14 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 16:06.52 | *** join/#asterisk thieums (n=Mathieu_@bgn92-6-88-179-201-40.fbx.proxad.net) |
| 16:07.50 | isamar | anybody using "alwaysfork=yes" |
| 16:07.53 | isamar | ? |
| 16:07.59 | isamar | in asterisk.conf.... |
| 16:10.33 | DarKnesS_WolF | isamar: for CDR ? |
| 16:10.52 | isamar | yep |
| 16:11.02 | *** join/#asterisk ThoMe (i=tm@tm.muc.de) |
| 16:11.17 | ThoMe | hello |
| 16:11.21 | ThoMe | kann hier wer deutsch? :-( |
| 16:11.58 | Qwell | ThoMe: nein - #asterisk.de ? |
| 16:12.16 | jjshoe_ | Qwell tell me that was an ontext event :P |
| 16:12.30 | Qwell | eh? |
| 16:12.31 | AlexTO | Hi, someone who can give me a hand with CDR on MySQL? |
| 16:13.04 | jjshoe_ | Qwell I'm just curious how much of each lingo you know to know what lingo they want :P |
| 16:13.12 | jjshoe_ | AlexTO just ask away |
| 16:13.19 | ThoMe | Qwell: kannst du deutsch? |
| 16:14.01 | Qwell | ThoMe: sehr wenig |
| 16:14.19 | isamar | DarKnesS_WolF: for anything |
| 16:14.21 | codefreeze-lap | Qwell: dass is schade! |
| 16:14.48 | Qwell | codefreeze-lap: enough to do harm? |
| 16:15.15 | ThoMe | Qwell: kennst du den begriff "anlagenanschluss" ? |
| 16:16.07 | Qwell | nein.. |
| 16:16.15 | ThoMe | anlagenanschluss = ptp ? |
| 16:16.23 | Qwell | p2p? |
| 16:16.38 | Qwell | or 'point to point'? |
| 16:16.42 | *** join/#asterisk FreedomBI (n=freedomb@mn01.freedombi.com) |
| 16:16.48 | ThoMe | point to point :-) |
| 16:16.49 | DarKnesS_WolF | AlexTO: ask |
| 16:17.08 | AlexTO | jjshoe, i already set my files to record the CDRs into a DB, but all the time i get msg that unable to connect to the BD, so i was checking the connection and i had a incompatible MySQL-Client version, so i upgrade it and now the connection from my * is fine but i'm still getting this error |
| 16:17.22 | AlexTO | cdr_addon_mysql.c:144 mysql_log: cdr_mysql: cannot connect to database server |
| 16:17.56 | DarKnesS_WolF | AlexTO: use pastbin.de and paste ur cdr_mysqlconf |
| 16:18.00 | [TK]D-Fender | AlexTO: If you had to upgrade your client, then you likely had the wrong client LIB version as well and Addons needs to be rebuilt to account for it. |
| 16:18.10 | DarKnesS_WolF | AlexTO: and did u try to connect using the mysql command line ? |
| 16:18.18 | AlexTO | I made a Dump onthe nic card with the port 3306 and i get DClient does not support authentication protocol requested by server |
| 16:18.28 | Qwell | ThoMe: hopefully you speak some English - you might have a better time asking in #asterisk.de |
| 16:18.43 | ThoMe | Qwell: jep. :-( |
| 16:18.43 | DarKnesS_WolF | AlexTO: mm may be [TK]D-Fender is right he is always right |
| 16:18.46 | [TK]D-Fender | Qwell: He does, he's been in here often |
| 16:18.59 | [TK]D-Fender | DarKnesS_WolF: Not always, but usually :) |
| 16:19.14 | DarKnesS_WolF | [TK]D-Fender: see :P |
| 16:20.39 | AlexTO | I'll ask him |
| 16:21.20 | AlexTO | [Tk]D-Fender: do you have ani idea what could be my problem? |
| 16:21.34 | [TK]D-Fender | [12:17]<[TK]D-Fender>AlexTO: If you had to upgrade your client, then you likely had the wrong client LIB version as well and Addons needs to be rebuilt to account for it. |
| 16:21.42 | [TK]D-Fender | AlexTO: PAY ATTENTION <- |
| 16:21.51 | [TK]D-Fender | ^&%@# |
| 16:22.18 | AlexTO | OK |
| 16:22.34 | AlexTO | i'm ready |
| 16:23.46 | AlexTO | Yes, i made the connection from that server using shell an dthe connection was done perfect |
| 16:24.44 | AlexTO | I'll paste all on pastebin.. hold on please |
| 16:25.16 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 16:25.19 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 16:34.06 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 16:36.12 | *** join/#asterisk telenieko (n=marc@240.Red-213-96-49.staticIP.rima-tde.net) |
| 16:36.48 | AlexTO | I'm back |
| 16:36.57 | AlexTO | that's the info http://pastebin.com/m5cc74926 |
| 16:36.59 | telenieko | Hi. I'm trying to communicate two asterisks with IAX, with one call everything goes fine, but when I try to setup a second call between them I always get "CHANUNAVAIL" with no explanation. Any clue? |
| 16:38.00 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:39.05 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 16:41.48 | [TK]D-Fender | AlexTO: show me you connecting to it from your * server at Linux CLI with its local client |
| 16:43.09 | hsv-al | another insane day for lunch |
| 16:43.18 | hsv-al | pom juice +blue berries bleh |
| 16:43.37 | Corydon76-dig | porn juice? |
| 16:43.40 | AlexTO | ok, mysql -ucdr -p12345 -h [64.25.XX.XX] |
| 16:43.41 | [TK]D-Fender | hsv-al: At a quick glance that read as "porn juice"... |
| 16:43.52 | hsv-al | not a bad suggestion |
| 16:43.53 | [TK]D-Fender | Corydon76-dig: NOT JUST ME! |
| 16:43.56 | hsv-al | but Pomegranite |
| 16:44.12 | AlexTO | and it connect right away |
| 16:44.20 | hsv-al | corydon76, another good lunch is hitting balls |
| 16:44.24 | hsv-al | . . at the driving ranch |
| 16:44.36 | hsv-al | range :) |
| 16:44.51 | [TK]D-Fender | AlexTO: And have you jsut recompiled * addons? |
| 16:45.04 | Corydon76-dig | You and your pornographic blueberries |
| 16:45.19 | hsv-al | d-fender, i purchased my card 45min earlier |
| 16:45.25 | hsv-al | but all i got was a 1year replacement w/ them |
| 16:45.28 | AlexTO | no, i just upgrade the MySQl-client |
| 16:45.45 | DarKnesS_WolF | AlexTO: u have to compile asterisk-addon again then |
| 16:45.48 | DarKnesS_WolF | hsv-al: what card ? |
| 16:45.55 | hsv-al | 410P+1 fxo/fxs |
| 16:46.01 | hsv-al | addon modu* |
| 16:46.07 | hsv-al | $234.50 |
| 16:46.14 | AlexTO | Just Compile? |
| 16:46.15 | [TK]D-Fender | [12:17]<[TK]D-Fender>AlexTO: If you had to upgrade your client, then you likely had the wrong client LIB version as well and Addons needs to be rebuilt to account for it. |
| 16:46.19 | [TK]D-Fender | AlexTO: PAY ATTENTION <- |
| 16:46.21 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 16:46.26 | Corydon76-dig | hsv-al: if it's a genuine Digium card, they're warrantied from the OEM for 5 years |
| 16:46.29 | DarKnesS_WolF | hsv-al: is that digium card ? |
| 16:46.31 | [TK]D-Fender | AlexTO: I'm getting seriously tired of repeating myself... |
| 16:46.32 | hsv-al | yap |
| 16:46.38 | hsv-al | tdm410p |
| 16:46.43 | *** join/#asterisk intralanman (n=lanman@209.85.58.2) |
| 16:46.48 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 16:46.52 | DarKnesS_WolF | hsv-al: cool i have tdm400p |
| 16:46.58 | hsv-al | whats the difference? |
| 16:47.02 | hsv-al | 400/410 |
| 16:47.03 | Qwell | 10 |
| 16:47.06 | [TK]D-Fender | hsv-al: 10 :) |
| 16:47.09 | hsv-al | .... |
| 16:47.11 | DarKnesS_WolF | got it from mark when he was in egypt like 2 years ago as i thiink |
| 16:47.11 | Corydon76-dig | The 410 has a better PCI interface |
| 16:47.19 | DarKnesS_WolF | Qwell: u missed me drunk last night :P |
| 16:47.20 | hsv-al | un-noticeable stuff |
| 16:47.22 | hsv-al | to the hobby user? |
| 16:47.30 | Corydon76-dig | unless you try to use faxing |
| 16:47.37 | Qwell | or need hwec |
| 16:47.58 | Corydon76-dig | Faxing should be possible with the 410, but it's hit-or-miss with the 400 |
| 16:47.58 | hsv-al | am i going to have an available slot for hwec addon if i decide to get it in the future? |
| 16:48.03 | hsv-al | besides the fxo/fxs addon |
| 16:48.06 | DarKnesS_WolF | Corydon76-dig: i did faxing very good with my TDM400P using iaxmodem and hylafax like x86 did ;-) |
| 16:48.10 | Qwell | I believe the 410 does |
| 16:48.29 | anonymouz666 | anybody in here has a Linksys SPA-941 in hand? |
| 16:48.32 | DarKnesS_WolF | mmmm hwec is nice ! |
| 16:48.33 | hsv-al | i should of asked that before i purchased it |
| 16:48.34 | Corydon76-dig | DarKnesS_WolF: doesn't affect everybody, but sometimes the card just will not work |
| 16:48.37 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 16:48.45 | *** join/#asterisk jmls (n=jmls@host217-36-208-155.in-addr.btopenworld.com) |
| 16:48.58 | DarKnesS_WolF | Corydon76-dig: at work i have tdm400p and a office HP thingy faxs soo tricky :-s |
| 16:49.09 | DarKnesS_WolF | that box like 3 years old and i am not free for 1 day to upgrade it :-s |
| 16:49.12 | hsv-al | is there available room for the hwec digium addon module |
| 16:49.12 | jmls | why would I suddenly get a load of CHANUNAVAIL/1 or CHANUNAVAIL/31 ? |
| 16:49.18 | hsv-al | if fxo/fxs modules are put on the 410? |
| 16:49.24 | jmls | when dialling out on a zap PRI |
| 16:49.25 | jmls | ? |
| 16:49.45 | DarKnesS_WolF | jmls: seems like the line is down ? |
| 16:49.47 | Corydon76-dig | jmls: D-channel drop? |
| 16:49.48 | Qwell | hsv-al: yes, it's across the bottom |
| 16:49.54 | Qwell | http://store.digium.com/productview.php?product_code=1TDM404EF |
| 16:49.56 | Qwell | the purple thing |
| 16:49.58 | jmls | no warnings at all about d-channel |
| 16:50.29 | Corydon76-dig | jmls: don't know, then |
| 16:50.43 | jmls | could it be a telco issue ? |
| 16:50.50 | Corydon76-dig | Could be, yes |
| 16:51.50 | DarKnesS_WolF | the hwec price too much :-s |
| 16:52.05 | *** join/#asterisk pythonpoole (n=Guest252@CPE000f9f1835b6-CM001404dc9f3c.cpe.net.cable.rogers.com) |
| 16:52.32 | jmls | I can manually dial a test number using each channel without a problem [dial(zap/124/444605) |
| 16:52.36 | RoyK | IIRC asterisk will decode/encode DTMF in inband mode even while bridging a call. Now this is all fine, but I've seen problems with asterisk not detecting all dtmf. Is there a way to drop all decoding or to tune it up for local needs? |
| 16:54.01 | Corydon76-dig | RoyK: other than with fxotune for the analog cards, there is no easy way, no. |
| 16:54.06 | ManxPower | RoyK: all intermittent DTMF issues on Zap that I've ever encountered ended up being a gain issue. |
| 16:54.13 | Corydon76-dig | RoyK: have you tried relaxdtmf=yes yet? |
| 16:54.34 | ManxPower | relaxdtmf=yes is just an alias for "randomlydetectdoubledtmf=yes" |
| 16:54.45 | ManxPower | It's a horrid little option. |
| 16:54.50 | *** join/#asterisk Psychobilly (n=moi@adsl154-242.kln.forthnet.gr) |
| 16:55.22 | *** join/#asterisk kannan (n=kannan@123.201.60.110) |
| 16:55.27 | kannan | hello all |
| 16:55.32 | ManxPower | RoyK: I assume this is DTMF coming INTO Asterisk, i.e. IVR, and not Asterisk SENDING DTMF, i.e. remote IVR? |
| 16:55.47 | SuPrSluG | here's my /etc/openzap/openzap.conf http://pastebin.com/m29791e12 |
| 16:55.59 | ManxPower | SuPrSluG: what is openzap????? |
| 16:56.05 | Qwell | SuPrSluG: what does openzap have to do with asterisk? |
| 16:56.34 | kannan | I'd like to know about the card that does transcoding into and fom g729a , i dont know the model and cant find it in the store page of digium |
| 16:56.50 | ManxPower | kannan: TC400 maybe? |
| 16:56.52 | Qwell | kannan: tc400 |
| 16:57.05 | TJNII | Isn't DTMF supposed to be transmitted digitally and not decoded by *? I thought the ATA took care of that. |
| 16:57.11 | ManxPower | Qwell: maybe SuPrSluG is just looking for everyone on the channel to /ignore him. |
| 16:57.20 | Qwell | kannan: http://store.digium.com/productview.php?product_code=1TC400BLF-01 |
| 16:57.21 | intralanman | SuPrSluG: wrong channel, dude |
| 16:57.30 | [TK]D-Fender | kannan: http://store.digium.com/productview.php?product_code=1TC400BLF-01 |
| 16:57.31 | kannan | Qwell , thanks |
| 16:57.36 | Qwell | [TK]D-Fender: slow |
| 16:57.42 | kannan | ll, thanks |
| 16:57.44 | kannan | lol |
| 16:57.44 | [TK]D-Fender | kannan: I wish I could understand how you can miss it... |
| 16:57.56 | kannan | i am looking in analog |
| 16:58.11 | Corydon76-dig | TJNII: sometimes, but not always |
| 16:58.11 | Qwell | kannan: it's under voice processing |
| 16:58.13 | x86 | wtf is openzap? |
| 16:58.17 | hsv-al | heh |
| 16:58.25 | intralanman | ManxPower: chances are that SuPrSluG accidentally pasted in here instead of #freeswitch |
| 16:58.28 | ManxPower | Can't we kist kickban him and be done with it? |
| 16:58.32 | Corydon76-dig | x86: it's unsupported |
| 16:58.34 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:58.38 | kannan | Qwell, i am there thanx |
| 16:58.41 | x86 | Corydon76-dig: well yeah, but what is it? :P |
| 16:58.49 | x86 | Corydon76-dig: never heard of it before |
| 16:58.54 | x86 | openpbx? |
| 16:59.02 | Corydon76-dig | x86: hence why it's unsupported |
| 16:59.02 | ManxPower | intralanman: I'll bet he didn't get help on #freeswitch and was hoping to con us here. |
| 16:59.19 | intralanman | ManxPower: likely the other way around ;-) |
| 16:59.21 | jjshoe_ | is there any way to tell where a reload was issued from? like what ip, etc. |
| 16:59.38 | ManxPower | jjshoe_: not that I'm aware of. |
| 16:59.56 | ManxPower | intralanman: he knows better |
| 17:00.01 | Corydon76-dig | jjshoe_: reloads are only issued via the cli, unless you've enabled command mode in the manager interface |
| 17:00.24 | intralanman | x86: openzap is one of the TDM interfaces for freeswitch |
| 17:00.39 | x86 | ah |
| 17:00.47 | jjshoe_ | Corydon76-dig <3 thanks. |
| 17:01.36 | Corydon76-dig | jjshoe_: are you perhaps using the web interface? |
| 17:01.50 | jjshoe_ | Corydon76-dig no, no web interface, just a general question. |
| 17:02.19 | kannan | Is the TC400B card have unlimited channels for g729a , there is no specification on tha? (unless i missed it again ) |
| 17:02.24 | ManxPower | jjshoe_: since all reloads use the CLI, all reloads come from the IP of the server |
| 17:02.28 | Qwell | kannan: I think it's 128 channels |
| 17:02.42 | kannan | Qwell , ok thnx again , i will check it |
| 17:02.53 | hsv-al | ahh well well, bellsouth is offering burstable 20MB metro-e |
| 17:02.56 | hsv-al | HSV in 6months |
| 17:02.59 | Qwell | hsv-al: eh? |
| 17:03.01 | [TK]D-Fender | kannan: http://www.digium.com/en/products/voice/tc400b.php |
| 17:03.01 | hsv-al | bye bye comcast/knology |
| 17:03.05 | Qwell | link? |
| 17:03.06 | [TK]D-Fender | kannan: Lean. To. READ <- |
| 17:03.07 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 17:03.11 | [TK]D-Fender | kannan: http://www.digium.com/en/products/voice/tc400b.php |
| 17:03.18 | Corydon76-dig | Qwell: should be 92, isn't it? |
| 17:03.24 | [TK]D-Fender | Learn* |
| 17:03.24 | Qwell | Corydon76-dig: pretty sure they upped it |
| 17:03.30 | Qwell | they/we/whatever |
| 17:03.36 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 17:03.40 | Qwell | g723 might still be 96 |
| 17:03.42 | Corydon76-dig | They. Not our department |
| 17:03.45 | hsv-al | no link, friend is BD |
| 17:03.50 | hsv-al | public announcement probably soon |
| 17:04.00 | Strom_M | 96 g729a, 92 g723.1 |
| 17:04.00 | kannan | [TK}D-Fender , ok ts there 120 bi directional |
| 17:04.13 | [TK]D-Fender | Qwell: Should be 120 to support 4 fuly loaded E1 PRI's at least... |
| 17:04.18 | _ShrikE | http://blogs.digium.com/2008/01/17/more-more-more-tc400/ |
| 17:04.25 | ManxPower | your friend is into bondage and domination? |
| 17:04.30 | Qwell | "The TC400B is rated to handle up to 120 bi-directional G.729a transformations or 92 bi-directional G.723.1 transformations." |
| 17:04.39 | hsv-al | manxpower, complete w/ an orange gag ball |
| 17:04.43 | hsv-al | no, business development |
| 17:04.49 | Strom_M | Qwell: hm, your datasheet says 96 g729a |
| 17:04.52 | ManxPower | hsv-al: I prefer red |
| 17:05.24 | Strom_M | http://www.digium.com/elqNow/elqRedir.htm?ref=http%3A%2F%2Fwww.digium.com%2Fen%2Fdocs/TC400B/tc400b-datasheet.pdf |
| 17:05.27 | Qwell | Strom_M: ahh, poking people |
| 17:06.07 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:07.29 | *** join/#asterisk doolph (n=doolph@200.75.196.191) |
| 17:10.51 | RoyK | ManxPower, Corydon76-dig: Sorry for the late answer, but this is if a client is using DTMF inband, asterisk is configured so, and bridging between SIP and Zap. then clients are having problems with the destination not receiving DTMF |
| 17:10.56 | jjshoe_ | ManxPower me too. |
| 17:12.19 | Corydon76-dig | RoyK: latency, perhaps? Or the SIP conversation is not in ulaw format? |
| 17:13.06 | *** join/#asterisk csterley (i=csterley@brickwall.ostusa.com) |
| 17:13.16 | *** join/#asterisk newsmafia (n=newsmafi@207-114-163-134.static.twtelecom.net) |
| 17:13.23 | RoyK | Corydon76-dig: alaw |
| 17:14.01 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:14.17 | RoyK | Corydon76-dig: and it doesn't look like latency problems either - at least not latency in the net - it looks more like asterisk is receiving dtmf, trying to decoding and failing and not relying the received dtmf, but deciding there aren't any |
| 17:14.43 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 17:15.04 | Corydon76-dig | RoyK: you can determine that, by switching on dtmf logging in logger.conf |
| 17:15.25 | RoyK | Corydon76-dig: is this a new thing in 1.4/1.6 or has it been there a while? |
| 17:17.12 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 17:18.13 | csterley | anyone have any ideas or hints on having asterisk pull a call back when a follow-me cell phone dumps to voice mail? |
| 17:19.24 | Corydon76-dig | RoyK: it's somewhat new in 1.4. It was there in 1.2, but it only worked within the generic bridging code, so it only logged if you were moving between e.g. Zap and SIP |
| 17:20.29 | *** join/#asterisk SyL (i=sil@sevatech.com) |
| 17:25.02 | hsv-al | so besides 1yr replacement i get w/ telephonydepot, whats this 5 year item you mentioned? |
| 17:25.05 | AlexTO | [TF]D-Fender: wich upgrade should i use? http://pastebin.com/dd6b4697 |
| 17:25.36 | *** join/#asterisk eject_ck (n=eject@85.223.182.86) |
| 17:26.59 | jblack | I have a real serious problems. Two * servers on the same network that are having trouble talking to one another |
| 17:27.02 | [TK]D-Fender | AlexTO: I never said upgrade. You are not listening when I have been telling you exactly what you should try next. This has been a collosal waste of my time, so I'm stepping back from it. |
| 17:27.33 | hsv-al | p4 3.2ghz |
| 17:27.38 | hsv-al | mt |
| 17:27.50 | jblack | Things work fine, But I get groups of Peer 'pbxin' is now UNREACHABLE! Time: 1 |
| 17:28.14 | SyL | does anybody know of a hack/script/plugin/etc that will call me on my cell if certain emails show up? |
| 17:28.51 | SyL | or point me the right direction? |
| 17:29.07 | intralanman | SyL: write a script using procmail? |
| 17:29.54 | jblack | They seem to last 20-60 seconds each time, but are coming in frequent clumps. I had one at 10:24 for 50 sec, 10:26 for 10 sec, 10:20 for a few secs. |
| 17:30.07 | SyL | intralanman: if I don't have a home phoneline, is there another way this could be done? asterisk or something? |
| 17:31.03 | newsmafia | nerf: yah, a few of our machines had issues |
| 17:31.25 | intralanman | SyL: theoretically, you could have a script that gets called and originates a call from the event socket |
| 17:31.32 | intralanman | errr... AMI... |
| 17:33.02 | Corydon76-dig | SyL: check with your cellphone company for the email address associated with the SMS on your cell |
| 17:33.35 | Corydon76-dig | and use procmail to forward selectively |
| 17:33.42 | intralanman | SyL: did you mean "call" you? or message you? |
| 17:34.22 | Corydon76-dig | Asterisk is not our only tool, and we do advocate using the right tool for the job |
| 17:35.39 | SyL | intralanman: call me... I wake up for phone calls. I ignore the email beeps. |
| 17:36.17 | SyL | Corydon76-dig: I need a phone call... I used to have it when I had a landline, but now I don't have one. |
| 17:36.19 | *** join/#asterisk riddlebox (n=user@75-128-170-26.static.stls.mo.charter.com) |
| 17:36.34 | [TK]D-Fender | SyL: read up on "call files" and "ami originate" on the WIKi for how to make * call you. The script that will DECIDE to call you is your job. |
| 17:36.47 | intralanman | forgot about call files |
| 17:36.58 | Corydon76-dig | SyL: you could get a voip provider to do the translation for you |
| 17:37.31 | SyL | [TK]D-Fender: thanks! |
| 17:38.27 | SyL | Corydon76-dig: suggestions on VOIP providers? |
| 17:38.39 | Corydon76-dig | SyL: I suggest nufone |
| 17:41.18 | *** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
| 17:41.51 | *** join/#asterisk MoreAllLess (n=jackjust@cpe-76-169-252-172.socal.res.rr.com) |
| 17:41.51 | SyL | Corydon76-dig: thanks... I will look them up |
| 17:42.59 | newsmafia | try teliax.com great iax service |
| 17:43.27 | *** part/#asterisk MoreAllLess (n=jackjust@cpe-76-169-252-172.socal.res.rr.com) |
| 17:43.40 | SyL | Corydon76-dig: their redhead is doing it for me... nice choice... |
| 17:43.50 | Corydon76-dig | Heh |
| 17:44.12 | jblack | any suggestions? This is killing me |
| 17:44.42 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:45.51 | Corydon76-dig | jblack: have you tried qualifysmoothing=yes ? |
| 17:46.45 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:46.54 | jblack | no, I will now |
| 17:47.00 | Corydon76-dig | jblack: you might want to audit that network segment and see if people are transferring large files (like accessing a Samba share from Windows) |
| 17:47.09 | jblack | btw, these machines are on the same switch. |
| 17:47.20 | jblack | and yeah, there is filesharing on the same network. |
| 17:47.50 | Corydon76-dig | jblack: A lot of broadcast traffic would clog up the switch, as well |
| 17:48.04 | Corydon76-dig | jblack: the other possibility is that the switch is overheating and rebooting |
| 17:49.25 | jblack | I'm now running a ping to see if that sees the same problem. |
| 17:52.05 | gr0mit | is qualifysmoothing option available in 1.4.x and for both sip and iax? |
| 17:52.24 | hsv-al | ok, now that the card is coming |
| 17:57.01 | pythonpoole | I'm trying to add a new module/application to asterisk, can anyone direct me as to how I should go about doing this? I'm trying to add the ADSIProg.so module, but Asterisk seems to ignore it on restart (with autoload on), so I assume I have to register it as an application somehow. |
| 17:58.00 | *** part/#asterisk SyL (i=sil@sevatech.com) |
| 18:02.10 | tzafrir_laptop | pythonpoole, look at apps/app_skel.c for an example |
| 18:02.46 | tzafrir_laptop | try loading the module explictly at run time: module load ADSIProg.so |
| 18:03.15 | tzafrir_laptop | This name also does not follow the standard naming convention, but I have no idea if this is actually a problem |
| 18:04.12 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:05.53 | pythonpoole | actually I think it's app_ADSIPRog.so, thanks I'll try that |
| 18:06.18 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-181-rrdg-esr-2.dynamic.isadsl.co.za) |
| 18:06.27 | *** join/#asterisk Larisa (n=Larisa@84.126.207.62.dyn.user.ono.com) |
| 18:08.14 | *** join/#asterisk MoreAllLess (n=justo@cpe-76-169-252-172.socal.res.rr.com) |
| 18:13.59 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 18:14.28 | Larisa | hola |
| 18:14.39 | Strom_M | hola |
| 18:14.52 | Larisa | o buenas :) |
| 18:15.05 | doolph | hola |
| 18:16.10 | MoreAllLess | ¿Que cuentan? |
| 18:16.21 | doolph | 123 |
| 18:17.08 | jblack | Corydon76-dig: Sorry if I seemed rude. I appreciate the help. |
| 18:17.25 | jblack | Smoothing seems to have helped somewhat for now. |
| 18:18.38 | jblack | I spent some time re-evaluating what the server does, and I think you hit the bullseye. Though pbxin and pbx2 are connected to a switch (which is fine), pbx2 is just one of 8 virtual machines, the other of which is a fileserver.. which renders the switch useless. |
| 18:18.55 | Strom_M | bingo |
| 18:19.01 | Strom_M | poor network design :P |
| 18:19.13 | [TK]D-Fender | jblack: wait IAX is failing across to a VM? |
| 18:19.23 | [TK]D-Fender | jblack: 1st is ok, 2nd isn't? |
| 18:19.24 | jblack | tk: From a vm pbx to a real one. |
| 18:19.51 | jblack | Strom_M: Aye, hidden by the virtualization... 8 machines trying to share a single physical interface. |
| 18:20.16 | [TK]D-Fender | jblack: Sounds like you have enabled IAX2 Trunk mode without proper Zaptel support. |
| 18:20.34 | Strom_M | jblack: I would recommend you dont run asterisk on a virtual server of any kind... |
| 18:20.41 | jblack | tk: Hmmm? I don't think zaptel has anything to do with it, as pxbin and pbx2 talk to one another with iax. |
| 18:21.01 | [TK]D-Fender | jblack: Zaptel is NEEDED for IAX2 trunking mode. make sure to add "trunk=no". |
| 18:21.12 | [TK]D-Fender | jblack: then retest |
| 18:21.42 | jblack | pbxin is a dedicated machine, that just worries about taking calls with a pri card and shuffling them off to pbx2 and pbx1(doesn't exist yet) via iax, who then route the calls to polycoms |
| 18:22.33 | jblack | The only exhibited problem is that pbxin and pbx2 loose registration with one another, tk. Calls already established already "work" for some value of the word that I don't know |
| 18:25.41 | [TK]D-Fender | jblack: Qualify timeout due to timeslice issues? |
| 18:25.47 | jackson__ | jblack, how are you resolving host names? Is the connectivity to the dns servers reliable? Static entries in /etc/hosts? |
| 18:27.48 | Corydon76-dig | jblack: glad I could help |
| 18:28.35 | jblack | Yeah. 300 gigs of data routed on fs1 in 6 days, the bulk of which would have occurrect at exactly the time the calls are happening. |
| 18:28.35 | *** join/#asterisk darkskiez (n=mbryars@72-254-127-253.hq.ibahn.com) |
| 18:28.44 | jblack | jackson__: Two local dns servers. |
| 18:29.01 | hsv-al | the hell |
| 18:29.15 | hsv-al | thought local phone service is cheaper then this |
| 18:29.31 | hsv-al | $29.33 /mo for bellsouth, no outbound long idstance, just a local plan |
| 18:29.42 | hsv-al | includes 5.00/month privacy, so my whois info cant be looked up. |
| 18:29.53 | jblack | hsv-al: That sounds cheap to me. My local telco wants 40-50 bucks a month once everything is considered. |
| 18:30.27 | hsv-al | so you think 29.33 is good w/ the privacy filter? |
| 18:30.47 | jblack | I think the 10 bucks a month I pay for a voip provider is good. |
| 18:31.05 | [TK]D-Fender | hsv-al: that sucks |
| 18:31.17 | [TK]D-Fender | hsv-al: indeed ITSP's are far cheaper |
| 18:31.33 | hsv-al | its all good, its pennies |
| 18:31.35 | [TK]D-Fender | I never intend to own POTS again myself. |
| 18:31.50 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 18:31.53 | jblack | heh. well, if you have any extras, feel free to send them this way. |
| 18:34.55 | Corydon76-dig | hsv-al: I think if the southern states realized what Californians are paying for their phone service, the price would be a quarter of what it is now |
| 18:35.41 | Strom_M | flat rate local service in california is like $10.73 per month IIRC |
| 18:35.45 | hsv-al | never been out west |
| 18:35.48 | hsv-al | even lower? |
| 18:36.01 | Corydon76-dig | Face it, your $30 local phone bill is 95% profit for the phone company |
| 18:36.06 | Strom_M | measured rate local service is $5.30 |
| 18:36.07 | Strom_M | :) |
| 18:36.22 | hsv-al | hell, i can make 5x then each month |
| 18:36.27 | hsv-al | via adsense from 6 forums i run |
| 18:36.29 | hsv-al | whatever |
| 18:36.55 | Qwell | Strom_M: and lifeline? |
| 18:37.08 | Strom_M | Qwell: cheaper, i'd imagine |
| 18:39.31 | Strom_M | lunch time |
| 18:42.46 | ManxPower | I think I paid $10/month for measured local service in Mississippi |
| 18:43.03 | pythonpoole | what should I do if I get this error: [Jun 3 14:41:48] WARNING[5987] loader.c: Module 'app_adsiprog.so' did not register itself during load |
| 18:43.10 | ManxPower | Alabama ATT/BellSouth claims to not sell measured local service anymore. |
| 18:43.28 | hsv-al | ? |
| 18:43.32 | ManxPower | pythonpoole: sounds like an old module on a new system. "make install" did not complain about it? |
| 18:45.39 | pythonpoole | @ManxPower, I'm a bit new to * and Trixbox (what I'm using). What I'm trying to do is get the old ADSI module from Trixbox 2 and get it working in version 2.6 (was removed because it's no longer supported). So I got a copy of the files from Trixbox 2.0 and I thought I might be able to use the modules with v2.6 |
| 18:45.49 | Qwell | ~trixbox |
| 18:45.50 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 18:46.57 | *** join/#asterisk Strom_M (n=pocketir@m500e36d0.tmodns.net) |
| 18:47.28 | pythonpoole | I tried on #trixbox, and the forums. They were of no help. I thought registering the module would be semi-universal for all * and * based systems, so this is kind of like my last resort. I've been trying desperately for days/weeks to get ADSI working |
| 18:47.42 | ManxPower | pythonpoole: Trixbox is not supported here and I do not support Trixbox. Best of luck. |
| 18:48.11 | ManxPower | Not getting help with Trixbox is not an excuse to ask here. If Trixbox support is so terrible then don't use it. |
| 18:48.48 | ManxPower | Module loading changed significantly between Asterisk 1.2 and Asterisk 1.4 |
| 18:50.14 | pythonpoole | I see. Does Asterisk 1.4 have the ADSI modules built-into it? Is there some form of Asterisk I can do a quick install of, throw on the server (using a TDM400P card) and use it to send ADSI scripts to my screenphones (just a one time thing). |
| 18:50.48 | ManxPower | pythonpoole: you will have to rebuild your entire dialplan and configs from scratch if you switch away from Trixbox |
| 18:51.05 | hsv-al | python |
| 18:51.07 | hsv-al | use an 1.0.2 |
| 18:51.27 | Qwell | hsv-al: hey, what is BS (heh?) going to be using for the last-mile on the metro-e? |
| 18:51.33 | hsv-al | www.asteriskguru.org/tutorials/adsi_conf.html |
| 18:51.37 | hsv-al | bs? |
| 18:51.40 | Qwell | bellsouth |
| 18:51.57 | Qwell | and did you say 20 or 60mbit? |
| 18:51.59 | pythonpoole | I realize that, I don't intend to permanently switch. I just want some way of programming my ADSI phones, even if means swapping out the HDs in the server, installing some form of Asterisk with ADSI apps built-in, running the ADSI scripts on to the phones, swapping back the HDs and going back to my current set-up |
| 18:52.00 | hsv-al | dunno, i can ask him, hes sort of in the business side of it |
| 18:52.02 | ManxPower | 1.6beta has ADSIProg, I would assume 1.4 does as well |
| 18:52.02 | hsv-al | not an engineer |
| 18:52.06 | Qwell | oh |
| 18:52.25 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-22613f84a03c6cc5) |
| 18:53.05 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 18:53.29 | pythonpoole | how easy is it to set-up Asterisk (vs Trixbox lets say) with extensions and a TDM400P? |
| 18:53.40 | hsv-al | python, just use the prepackaged AN |
| 18:53.47 | hsv-al | trix/freepbx = garbage :) |
| 18:54.25 | hsv-al | ive been using it for 6 months, was a gui drone |
| 18:54.30 | hsv-al | so its pretty simple to setup/use |
| 18:56.14 | [TK]D-Fender | pythonpoole: Everything is relative. Either go do it, or don't |
| 18:57.48 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:58.01 | *** join/#asterisk jicksta (n=jicksta@75-101-5-48.dsl.static.sonic.net) |
| 18:58.47 | spokra | anyone having problems with les.net? |
| 18:59.20 | ManxPower | pythonpoole: it is not simple at all. You will have to learn everything you avoided learning because the GUI hid it from you |
| 19:00.00 | [TK]D-Fender | ManxPower: Not for his needs, no. |
| 19:00.13 | [TK]D-Fender | ManxPower: he jsut wants to run ADSIProg for his phones. |
| 19:00.23 | ManxPower | to do it the way he wants to do it, he will. |
| 19:00.32 | ManxPower | Granted, that way is the silliest thing I've heard of |
| 19:00.45 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 19:00.57 | ManxPower | download 1.4, build/compile, copy app_adsi.so or whatever from the built source dir to where he needs it |
| 19:01.16 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 19:01.17 | [TK]D-Fender | ManxPower: install *, add 2-3 exten lines, add ADSI software. Execute 1 CLI command. Pretty much end of story. |
| 19:01.23 | [TK]D-Fender | ManxPower: And of course add zaptel. |
| 19:02.01 | ManxPower | [TK]D-Fender: why not just copy the module |
| 19:02.29 | l0verb0y | hey hows everyone doing today |
| 19:02.40 | [TK]D-Fender | ManxPower: trick is getting one that compiled for the rigth ver of *, and he'll still have to do the other work. |
| 19:02.55 | ManxPower | once he figures out what version of asterisk zaptel he is using, it should be trivial -- once he installs the 3 or 4 -dev packages he needs, of course |
| 19:03.27 | [TK]D-Fender | ManxPower: yup... because Trixbox is so well equiped to develop on ;) |
| 19:03.57 | l0verb0y | does anyone know of any automated ways to test DIDs? especially if you have a lot of them like 1000 |
| 19:04.07 | [TK]D-Fender | l0verb0y: test how? |
| 19:04.27 | l0verb0y | [TK]D-Fender: heres my problem |
| 19:04.54 | l0verb0y | we have many DIDs, and some times some of them randomly stop working, and we don't know until someone complains |
| 19:05.21 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 19:05.47 | ManxPower | l0verb0y: have you considered figuring out WHY the stop working? |
| 19:06.21 | l0verb0y | the quality of some of our did providers is sub par |
| 19:07.10 | l0verb0y | i was thinking of making a script to make a bunch of call files to dial the dids and then to check the log files to see if the calls were received but this is just an idea |
| 19:07.29 | jjshoe_ | Qwell ping |
| 19:08.25 | *** join/#asterisk nicox (n=nicox@212-183-40-128.adsl.highway.telekom.at) |
| 19:09.13 | kannan | l0verboy , u registering with the service provider for the DIDs? |
| 19:09.16 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 19:09.47 | l0verb0y | we signed up with a few and bought a ton of dids |
| 19:11.24 | Qwell | ? |
| 19:11.34 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:11.44 | jjshoe_ | Qwell does freenode have any ssl servers? |
| 19:11.54 | Qwell | umm, probably |
| 19:12.29 | kannan | is it always advisable to build asterisk as a non-rrot user only? |
| 19:12.32 | kannan | root |
| 19:12.54 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:13.17 | jjshoe_ | kannan the only answer to that is, is there any reason it needs to run as root? |
| 19:15.05 | [TK]D-Fender | jjshoe_: To become a more effective member of Qwell's chan_skinny bot-net ;) |
| 19:15.37 | jjshoe_ | [TK]D-Fender lol |
| 19:15.56 | kannan | hmm, ok, i have een building as root only so far, |
| 19:16.13 | *** part/#asterisk jmls (n=jmls@host217-36-208-155.in-addr.btopenworld.com) |
| 19:16.25 | jjshoe_ | kannan if it's a phone only box some might say what's the harm, if they find an exploit as the user asterisk runs at they can still take down asterisk, but it's generally bad practice to run things as root |
| 19:17.09 | jjshoe_ | [TK]D-Fender it would be interesting to see some overly zealous spammer/script kiddies load polycom firmware, or sip cisco firmware with botnet crap |
| 19:17.13 | kannan | jjshoe_ , ok i get that |
| 19:17.15 | jblack | Heh. This company keeps forgetting that I'm 3,000 miles away. :) |
| 19:17.28 | *** join/#asterisk icel (n=dan@63.78.162.121) |
| 19:17.48 | jblack | "I think we can fix that problem if you go into the server room. The machine should be on the left side, next to the door". |
| 19:17.53 | jjshoe_ | jblack the contractor I've been working for keeps asking why I'm not making further headwinds on his projects, I keep asking why he forgets to pay me. |
| 19:17.54 | jblack | "Oh, ok. what color is it?" |
| 19:18.06 | jblack | "No idea. I've never been in spokane in my life" |
| 19:18.11 | icel | Does * 1.4 (1.4.15) have video (h263) support by default or do you have to add a patch? |
| 19:21.00 | *** join/#asterisk IPkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 19:21.08 | IPkaf | hi |
| 19:21.17 | l0verb0y | hey |
| 19:21.28 | IPkaf | my asterisk with my sip extension working well |
| 19:21.28 | Strom_C | icel: do you want it to transcode or just to passthrough? |
| 19:21.39 | IPkaf | while the user one calling the user 2 i want that the user1 hear a custom music |
| 19:21.48 | IPkaf | while waiting to join the user2 |
| 19:21.56 | Strom_C | IPkaf: type "core show application Dial" at the CLI |
| 19:22.16 | icel | strom_c: just pass through |
| 19:22.28 | Strom_C | icel: you should be able to do it |
| 19:22.36 | icel | strom_c: thanks |
| 19:22.45 | *** join/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net) |
| 19:23.20 | *** join/#asterisk trythis (n=name@p54A1697F.dip.t-dialin.net) |
| 19:23.51 | trythis | Hi, can anyone help me with mISDN? |
| 19:24.02 | Strom_C | trythis: ask a real question |
| 19:24.09 | trythis | ah ok sorry |
| 19:24.53 | kannan | will changing the setting for a SIP user as canreinvite=yes, keep asterisk out of the media path? |
| 19:25.21 | Bananaskin | looking for a bit of help, have a GSM gateway attached to one of the FXO ports on my TDM400P. Have a callback trunk which phones my mobile via the GSM gateway, and presents DISA - that bit works fine, appears that the Zaptel trunk drops the call if I am dialing destinations that have the use of IVR's #'s etc, any ideas as to what the prob could be |
| 19:25.37 | ManxPower | ~trunk |
| 19:25.38 | jbot | i guess trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 19:25.38 | Strom_C | kannan: it won't keep asterisk out of the media path, but asterisk will drop out of the media path if it can do so |
| 19:26.02 | trythis | My system stops on boot with the message: mISDN_isac_init: ISAC Version...and now I want to remove the mISDN to be loaded on boot but I don´t find the config where I can do this. I have Debian |
| 19:26.02 | kannan | Strom_C, thans, any way to always ensure that this will happen |
| 19:26.07 | ManxPower | Bananaskin: remove t, T, w, and W from your Dial line |
| 19:26.08 | Strom_C | kannan: no |
| 19:26.32 | ManxPower | trythis: try asking on #Debian, they know their OS |
| 19:27.07 | ManxPower | kannan: you can be sure Asterisk always stays out of the media path by using G723.1 or G729. If Asterisk has to stay in the path then it will drop the call. |
| 19:27.12 | kannan | Bananaskin, i had a similar issue where in features.conf , blindxfer, (or atxfer setings) were # , and dial plan uses some options |
| 19:27.18 | ManxPower | (WITHOUT licenses, of course) |
| 19:27.45 | Bananaskin | right gonna test it now, hope this works :) |
| 19:27.52 | ManxPower | kannan: Actually, ignore all of what I just said. |
| 19:27.57 | kannan | Strom_C , thaks |
| 19:28.03 | ManxPower | kannan: I was totally wrong |
| 19:28.34 | kannan | ManxPower, ok, but i was just about to say I didnt underdstand it in the first place :) |
| 19:29.05 | ManxPower | kannan: try blocking the RTP ports on your firewall to prevent Asterisk from doing any RTP. Also, this will drop or screw up any calls where asterisk has to stay in the media path |
| 19:29.25 | kannan | MnaxPower, if he options are needed, features.conf can be edited to still allow the # keys? |
| 19:29.33 | ManxPower | kannan: BTW, what you are trying to do is by far one of the stupidest things I've seen a person try to do with Asterisk in at least a couple of years. |
| 19:29.56 | ManxPower | kannan: you can't use any features.conf stuff if you want Asterisk out of the media path |
| 19:30.30 | kannan | ManxPower, thks . The features is in resonse of Bananaskin question |
| 19:30.34 | Bananaskin | ManxPower, kannan didn't cure it, still dropping the call |
| 19:31.01 | ManxPower | Bananaskin: paste the single line with the Dial on it |
| 19:31.03 | kannan | Bananaskin , kannan <--- newbie |
| 19:31.05 | kannan | :) |
| 19:31.12 | ManxPower | I assume you did a reload, after making the changes, right? |
| 19:31.18 | Bananaskin | yep |
| 19:31.37 | ManxPower | then paste the single dial line. If you flood the channel -- well, you don't want to know what happens then |
| 19:31.51 | Bananaskin | :) 2 secs, IRC via VNC |
| 19:32.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:32.39 | ManxPower | Bananaskin: do it as fast as you can while still being ACCURATE. We don't mind waiting, but we get pretty pissed off is someone we are trying help leaves for 20 mins for a phone call. |
| 19:32.42 | keith4 | ... IRC via VNC? wtf? |
| 19:33.02 | ManxPower | keith4: It's like an orange wrapped in a butterfly |
| 19:33.06 | csterley | anyone have any ideas or hints on having asterisk pull a call back when a follow-me cell phone dumps to voice mail? |
| 19:33.19 | ManxPower | csterley: it cannot be done |
| 19:33.19 | keith4 | ManxPower: ohhhh, ok |
| 19:33.53 | csterley | k thanks |
| 19:34.09 | Bananaskin | ManxPower, kannan - -- Executing [s@macro-dial:7] Dial("Zap/3-1", "sip/123@66.212.134.192||r") in new stack |
| 19:34.24 | ManxPower | make sure to pull the call back before the cellphone answers. You can TRY AMD "core show application AMD", but I bet it won't be reliable enough for you in this situation |
| 19:34.53 | ManxPower | Bananaskin: remove the "r", and the two ,, |
| 19:35.04 | csterley | you are correct. we will keep playin with ring time i guess |
| 19:35.36 | ManxPower | csterley: It's worth trying AMD. I suspect you'll end up using a combination of timeouts and AMD |
| 19:36.46 | ManxPower | csterley: are you in the USA? |
| 19:37.03 | csterley | yes |
| 19:37.31 | IPkaf | i m sorry |
| 19:37.35 | IPkaf | asterisk1*CLI> core show application Dial |
| 19:37.36 | IPkaf | No such command 'core' (type 'help' for help) |
| 19:37.38 | IPkaf | asterisk1*CLI> |
| 19:38.09 | [TK]D-Fender | IPkaf: They you'd appear to be on * 1.2. drop the "core" in front |
| 19:38.49 | ManxPower | csterley: I use 15 seconds in my scripts as the timeout when calling cellphones |
| 19:38.49 | kannan | ManxPower, I thought only T or t ,the transfer options will affect the transfers as defined in features.conf? |
| 19:39.18 | IPkaf | what command i have to type ??, |
| 19:39.27 | IPkaf | exactly ?? |
| 19:39.30 | ManxPower | kannan: If Asterisk has to listen to DTMF then Asterisk MUST be in the media path. It is as simple as that. |
| 19:39.52 | ManxPower | IPkaf: "show application Dial", you *insult censored* |
| 19:39.56 | csterley | ManxPower: I just made changes today to 15 seconds. we have a combination of differnt cell carriers and can't seem to find a common ground that fits everyone. |
| 19:40.02 | kannan | ManxPower, aah ok, hats clear now |
| 19:40.16 | kannan | i was seeing how to reduce bandwith usage only |
| 19:40.37 | ManxPower | kannan: so you don't REQUIRE Asterisk be stay out if the media path, you just want it ti. |
| 19:40.42 | ManxPower | Then why isn't Strom |
| 19:40.45 | kannan | ManxPower, yes |
| 19:40.54 | ManxPower | Then why isn't Strom_'s suggestion workable? |
| 19:41.05 | kannan | ManxPower, yes, it is fine |
| 19:41.43 | kannan | I wasnt pursuing that question at all, I am seeng Bananaskin's transfer problem |
| 19:42.30 | Bananaskin | having some success, but not consistent here |
| 19:42.40 | IPkaf | doing that command there 'are list of thing appear ?? |
| 19:42.47 | Bananaskin | worked with the FXO modules on my cisco router |
| 19:42.52 | ManxPower | Bananaskin: chances are you are matching different Dial lines when dialing different numbers, etc |
| 19:43.02 | IPkaf | while the user one calling the user 2 i want that the user1 hear a custom music |
| 19:43.07 | IPkaf | what to do ?? |
| 19:43.31 | Strom_C | IPkaf: read the documentation that we're pointing you at |
| 19:43.49 | ManxPower | IPkaf: you might have better luck writing an e-mail and asking on the asterisk-users mailing list. Write it in your native language, then use an online translator to post in both your native language and in english. |
| 19:44.44 | ManxPower | Obviously note in the message what things you tried and what documentation you looked at. |
| 19:44.46 | Bananaskin | I don't particularly want to have to start using sip trunking to the cisco router again to get this working |
| 19:45.25 | IPkaf | ok thanks to all |
| 19:45.27 | IPkaf | bye |
| 19:45.48 | ManxPower | Bananaskin: # is an Asterisk control DTMF, you tell Asterisk to listen for DTMF commands by using t, T, w, or W on your Dial line. It will never ever intercept the DTMF as a control code unless you tell it to do so with those options. |
| 19:46.26 | ManxPower | If you are using an ITSP that uses Asterisk, perhaps THEY are processing the DTMF |
| 19:47.42 | Bananaskin | ManxPower, the call is coming from my pbx via the GSM gateway which is attached to the PBX, the zap channel is being terminated, thats what I see as the problem, not the call on the other side |
| 19:47.59 | ManxPower | I saw SIP in your Dial line |
| 19:48.51 | Bananaskin | yes, thats the custom extn (200) that I dial after dialtone is presented when the DISA is presented |
| 19:49.29 | Bananaskin | my pbx is using callback to present me a DISA dialtone |
| 19:50.09 | Bananaskin | but because I have a sim in the GSM gateway that has 2200 mins free a month, why not use it |
| 19:50.47 | ManxPower | So we have PBX FXO <-> FXS Asterisk FXO <-> FXO GSM Gateway |
| 19:51.10 | ManxPower | I still don't see any SIP in there. |
| 19:51.53 | *** join/#asterisk killfill (n=killfill@200.73.13.54) |
| 19:52.16 | killfill | how do i check if a sip user is avaible to recieve a call or not? |
| 19:52.29 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 19:52.29 | deeperror | cli> sip show users? |
| 19:52.35 | Bananaskin | ManxPower, nope, we have Cell Phone > GSM Gateway > FXO > PBX > Dialout via SIP |
| 19:52.41 | killfill | it doesnt tell is its busy.. |
| 19:52.53 | ManxPower | Bananaskin: Dialout to WHO WHERE? |
| 19:53.04 | Strom_C | killfill: you attempt to Dial() and then act based upon what Dial() returns |
| 19:53.11 | Strom_C | killfill: core show application dial |
| 19:53.16 | killfill | actually.. it doesnt show anything.. i have sip peers.. |
| 19:53.16 | Bananaskin | anywhere, via ITSP's |
| 19:53.27 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.20.1 (2008/05/21) Asterisk 1.2.29 (2008/06/03), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.4 (2008/...) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 19:53.30 | killfill | to dial.. hm.. |
| 19:53.43 | ManxPower | Bananaskin: Where is Asterisk in all this? |
| 19:54.17 | ManxPower | Bananaskin: the cell phone is doing the call back? |
| 19:54.18 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 19:54.23 | killfill | Strom_C: but if i call the users, well thats "intrusive".. its phone will ring.. |
| 19:54.39 | Bananaskin | nope, the cell is getting called back from the GSM gateway |
| 19:54.44 | ManxPower | Bananaskin: put a copy of your extensions.conf on pastebin.ca |
| 19:54.52 | Strom_C | killfill: perhaps you need to provide a better explanation of what the hell you're actually trying to accomplish |
| 19:55.13 | ManxPower | killfill: why not turn off call waiting? What's what we do. Never had to think about it again. |
| 19:55.45 | killfill | oh well, i have a program that listen to uses events in the AMI. When a call comes in, it takes actions. |
| 19:56.03 | killfill | i need to filter to make the actions, only when the user is online and not busy |
| 19:56.13 | ManxPower | killfill: you might think you were looking for ChanIsAvail, and you might be right. |
| 19:56.26 | *** join/#asterisk miguel3239 (n=elguero@ns1.nashuacs.com) |
| 19:56.27 | killfill | ChanIsAvail.. hm.. |
| 19:56.39 | ManxPower | killfill: you mean whenever the user online and on a call. |
| 19:57.39 | killfill | if user is online && avaible to reviece a call |
| 19:57.49 | killfill | recieve.. sorry |
| 19:57.53 | Strom_C | receive |
| 19:57.58 | ManxPower | killfill: "avaible to reviece a call" can mean at least three things. |
| 19:58.21 | ManxPower | Maybe the person is on a call, and has a phone with call waiting. |
| 19:58.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:58.28 | killfill | well if its not already on a phone call. |
| 19:58.37 | ManxPower | So he could accept another call and is not BUSY from the perspective of Asterisk |
| 19:59.08 | ManxPower | So why can't you turn off call waiting on the phones again? |
| 19:59.41 | ManxPower | I guve up |
| 19:59.43 | *** part/#asterisk ManxPower (n=manxpowe@126.sub-75-202-184.myvzw.com) |
| 20:00.00 | killfill | the calls are getting into a queue. |
| 20:00.25 | killfill | and every agent on that queue, is "monitoring" the calls that are flying to its phone. |
| 20:01.04 | killfill | if he is avaible, then he should see a popup that Mr johns is on the phone. |
| 20:01.11 | killfill | thats the thing.. simple.. |
| 20:01.19 | *** join/#asterisk galeras (n=galeras@190.26.191.166) |
| 20:01.23 | galeras | Has anyone done any integration with Asterisk & Microsoft Dynamics CRM? |
| 20:01.27 | Strom_C | killfill: if you're using a queue, there's a MUCH easier way to do that |
| 20:01.47 | killfill | oh yes?.. how |
| 20:02.04 | Strom_C | have a look at the documentation for the Queue() app and tell me if you can figure it out |
| 20:03.30 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:04.03 | killfill | "The optional AGI parameter will setup an AGI script to be executed on the calling party's channel once they are connected to a queue member." |
| 20:04.12 | killfill | Strom_C: you mean that? |
| 20:04.19 | Strom_C | bingo |
| 20:04.34 | killfill | hm.. |
| 20:05.34 | killfill | that would make it possible to the asterisk server to execute something when conection to an Agent. i.e. send a TCP packet to the agents ip. |
| 20:05.41 | killfill | right?.. |
| 20:06.16 | killfill | but there is a problme. The IP may be behind NAT. so need the client to connect and listen to ami events. |
| 20:06.55 | Strom_C | you could run an AGI that generates an AMI event ;) |
| 20:07.35 | killfill | mmMmm... |
| 20:09.40 | killfill | but its too late.. i want to show the popup info when the phone rings.. not when the agent already took the call. (i guess thats the event of tha AGI thingy) |
| 20:10.35 | Strom_C | are you doing screenpop info, or are you just doing glorified caller ID? |
| 20:10.49 | Strom_C | because, seriously, if all you want is caller ID, most phones do that now. |
| 20:11.28 | killfill | i wish.. |
| 20:11.30 | *** join/#asterisk Tili (n=tili@58.27.154.27) |
| 20:11.42 | killfill | it shows more than callerId. |
| 20:11.52 | killfill | itegrated to our systems's db's |
| 20:12.01 | killfill | tickets, projects, sistemas, etc |
| 20:12.18 | killfill | client history |
| 20:12.36 | Strom_C | why do they need to know all this *before* they answer the call? |
| 20:12.46 | Strom_C | by the time they're done reading it, the call is gone |
| 20:12.57 | killfill | haha.. |
| 20:13.22 | killfill | yah, well its a compact info.. but its a matter of politics.. thats my requirement.. :P |
| 20:13.33 | killfill | and found it usefull tho. |
| 20:15.12 | kannan | killfill, is it for a help desk? |
| 20:15.37 | killfill | yup |
| 20:15.40 | killfill | mostly |
| 20:15.44 | kannan | vicidial does all this stuff i think |
| 20:15.54 | kannan | and can integrate to any crm |
| 20:16.31 | killfill | yah i saw it.. but i got the impresion that its quite old.. |
| 20:17.02 | kannan | yes, you must use 1.2 trreeonly |
| 20:17.14 | lmadsen | vicidial has been around for a while, but is still actively maintained (but only on 1.2) |
| 20:17.22 | killfill | just did what i think it was best.. write a custom thing for the req's.. |
| 20:17.33 | anonymouz666 | lmadsen: and do you use it? |
| 20:17.46 | lmadsen | nope -- I don't spam people with my phone systems :) |
| 20:17.53 | anonymouz666 | heh |
| 20:17.57 | lmadsen | but I've met the developer several times |
| 20:18.09 | kannan | thats only 1/2 of the story, what about incoming calls |
| 20:18.23 | kannan | :) |
| 20:21.10 | loompek | hey guys.. i've got a koncept ip phone which has xfer button and it seems it's used only for attended transfer... |
| 20:21.56 | kannan | maybe he (vicidial developer) will jump to 1.6 tree straitaways |
| 20:22.08 | loompek | would it be possible for asterisk to change this type of behaviour to unattended (blind) transfer which sends the original callerid to the final destination |
| 20:22.24 | Strom_C | loompek: nope, that's your phone's responsibility |
| 20:22.28 | *** join/#asterisk IPkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 20:22.35 | IPkaf | hi |
| 20:22.56 | loompek | or i'll have to program one of the memory buttons to call *2 |
| 20:22.57 | loompek | :D |
| 20:23.14 | kannan | Strom_C , can he ot define a custom sequence like *2 to do the blind xfer? |
| 20:23.20 | kannan | can he not |
| 20:23.29 | Strom_C | kannan: cheapn hack |
| 20:23.32 | Strom_C | er, cheap |
| 20:23.45 | Strom_C | plus, *2 conflicts with VSC assignments |
| 20:23.52 | kannan | yes , very inexpensive to do, no need for a new phone |
| 20:23.55 | kannan | heheh |
| 20:23.58 | Strom_C | sigh |
| 20:24.01 | kannan | oh ok |
| 20:24.02 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:24.03 | kannan | j/k |
| 20:25.02 | IPkaf | it must be the correct place to ask question |
| 20:25.09 | IPkaf | on a dect phone |
| 20:25.25 | IPkaf | when u make a call it ring |
| 20:25.42 | russellb | yay for ringing when you call |
| 20:26.19 | IPkaf | how to change that music to use our custom music on the phone of course |
| 20:26.31 | IPkaf | is there anyone here try this before ??? |
| 20:26.53 | lmadsen | core show application Dial |
| 20:26.56 | lmadsen | look for 'm' |
| 20:27.05 | IPkaf | yeah |
| 20:27.17 | IPkaf | no |
| 20:27.21 | lmadsen | maybe so |
| 20:27.29 | IPkaf | i m sorry |
| 20:27.48 | IPkaf | my question is out of my asterisk box |
| 20:28.00 | lmadsen | eh? |
| 20:28.09 | lmadsen | you mean... can you make the ringer on the phone play music? |
| 20:28.12 | IPkaf | when u make a call u hear a music on the dect phone |
| 20:28.21 | IPkaf | how to change that music ?? |
| 20:28.37 | IPkaf | the music on the phone |
| 20:28.41 | lmadsen | define a different music on hold class |
| 20:29.02 | IPkaf | ok |
| 20:29.04 | IPkaf | thanks |
| 20:30.22 | IPkaf | my question when u buy a dect like philips something else the phone maker custom 3-4 music on the phone |
| 20:30.56 | IPkaf | like this when u make a call the person who take the call hear one that music |
| 20:31.18 | IPkaf | my question is how to customize that music |
| 20:31.35 | IPkaf | my question is out of asterisk |
| 20:31.52 | IPkaf | my question is on the phone itself |
| 20:32.40 | Strom_C | IPkaf: does this look like #dect to you? |
| 20:36.07 | IPkaf | ok |
| 20:36.12 | IPkaf | thansk |
| 20:36.23 | IPkaf | yeas |
| 20:37.20 | IPkaf | ok |
| 20:37.25 | IPkaf | sorry bye |
| 20:37.49 | Strom_C | that guy is uber-annoying |
| 20:38.20 | *** mode/#asterisk [-b %`Sauron!*@*] by russellb |
| 20:38.41 | `Sauron | ~grandstream |
| 20:38.42 | jbot | extra, extra, read all about it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 20:38.51 | `Sauron | That's what I thought. |
| 20:38.51 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 20:38.58 | russellb | ... ? |
| 20:39.32 | *** kick/#asterisk [Deeewayne!n=file@asterisk/developer-and-muffin-lover/file] by file (file) |
| 20:39.33 | russellb | what was that about? |
| 20:39.49 | `Sauron | I had a question about grandstream hardware. Seemed to remember people saying "don't" |
| 20:39.55 | `Sauron | Unless you're not talking to me. |
| 20:39.55 | russellb | ah. |
| 20:40.02 | russellb | i was |
| 20:40.10 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 20:40.11 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:40.21 | *** join/#asterisk arthurh (n=nmsclera@216.31.102.218) |
| 20:40.29 | Deeewayne | O.O |
| 20:41.47 | *** join/#asterisk ccvp (n=chatzill@66.0.46.210) |
| 20:41.47 | file | Deeewayne: russellb made me! |
| 20:41.56 | *** kick/#asterisk [file!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 20:42.02 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 20:42.03 | *** mode/#asterisk [+o file] by ChanServ |
| 20:42.38 | Deeewayne | chases file with asparagus |
| 20:43.29 | russellb | trips Deeewayne with an eggplant |
| 20:43.43 | hsv-al | where's the water chest nuts and brussel sprouts when you need them |
| 20:43.49 | hsv-al | russleb, w/ the brussel sprouts |
| 20:44.29 | `Sauron | So what are good sip/whatever phones? |
| 20:45.04 | russellb | Polycom phones are my favorite |
| 20:45.15 | tzafrir_laptop | ~phones |
| 20:45.16 | jbot | methinks phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 20:45.33 | `Sauron | Aha. Gracias. |
| 20:48.21 | *** join/#asterisk Bananaskin (n=mike@78-105-246-198.zone3.bethere.co.uk) |
| 20:48.48 | *** part/#asterisk icel (n=dan@63.78.162.121) |
| 20:49.24 | *** join/#asterisk Bananaskin (n=mike@78-105-246-198.zone3.bethere.co.uk) |
| 20:49.57 | kannan | bye all |
| 20:58.10 | markit | is dialplan AEL going to replace traditiional priority dialplan, or is a parallel way, or is an experiment that is going do die? |
| 20:58.10 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:58.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:59.26 | codefreeze-lap | markit: the future is uncertain; but I hope it won't die. People **are** using it. I see it, personally, as further evolving. |
| 21:00.17 | markit | codefreeze-lap: I starded of course with the traditional one, but I'm going to rewrite it and improve. what do you suggest? jump to AEL or has no real benefits? |
| 21:01.41 | codefreeze-lap | markit: you can do whatever you feel is best; personally, I'd not fear using AEL. It has some benefits, including some error checking you won't get with extensions.conf |
| 21:02.24 | codefreeze-lap | It's a bit easier to read, too. But beauty **is** in the eyes of the beholder. |
| 21:02.55 | markit | codefreeze-lap: I've tried once also the gui, but don't know if was my foult or was at an early stage, seemd much "under powered" also for basic needs. what do you think? |
| 21:03.50 | hsv-al | [Jun 3 16:03:06] WARNING[7981]: chan_zap.c:11244 process_zap: Ignoring rxwink |
| 21:03.53 | hsv-al | ?? |
| 21:04.26 | codefreeze-lap | I'm not a GUI expert. The asterisk gui is kinda new, but I know the guys that are working on it, and they're pushing it along nicely, markit |
| 21:04.59 | markit | ok, thanks :) |
| 21:05.06 | hsv-al | codefreeze, whats a good reason there is small use / discussion of AN 1.0.x in the community |
| 21:05.18 | hsv-al | if its a good gui interface(ive used it for half a year), and a bundled CLI in it |
| 21:05.39 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 21:05.40 | markit | hsv-al: AN? what is it? |
| 21:05.47 | hsv-al | ? |
| 21:05.51 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:05.52 | hsv-al | 1.0.2 |
| 21:06.10 | markit | AN 1.0? what is AN? (forgive my ignorance) |
| 21:06.21 | hsv-al | www.asterisknow.org |
| 21:06.21 | markit | ah, asterisk now? |
| 21:06.24 | markit | thanks |
| 21:06.44 | hsv-al | i rarely see people talk about it |
| 21:07.15 | [TK]D-Fender | goes to hide the body of the last person to have brought it up... |
| 21:07.38 | ix33 | see, that's the response i always got re: AN |
| 21:08.11 | hsv-al | well ive stopped using it for personal reasons |
| 21:08.17 | hsv-al | but i never had an issue w/ it |
| 21:08.29 | codefreeze-lap | hsv-al: I'm not educated enough to speak on the popularity of asterisknow. Basically, it's asterisk on a CD, ready to run. As far as I know, AN has already taken over the world, and everyone is already using it! Nobody discusses it because it works perfect, I'm sure... ;-) |
| 21:08.30 | *** join/#asterisk truent (i=0c224403@gateway/web/ajax/mibbit.com/x-8fe50f941032aacd) |
| 21:08.56 | truent | anyone know of a way to install ndiswrapper on asterisknow? |
| 21:09.14 | truent | ive read its not in the conary repo's or whatever.. do i have to build? |
| 21:10.28 | Idle | where can I find some good, user-centric, documentation for MeetMe, and voicemail, etc, that I can hopefully just print and hand to users.. if I have to rewrite them a bit thats fine too, I just need something user-centric |
| 21:12.57 | ix33 | i'm testing sound issues using an Answer()/Musiconhold() extensions... when it's purely SIP, the call sounds perfect full-duplex no matter how much noise on either end. when it goes through a t1 span card, the music seems to cut in and out as i speak into the handset. |
| 21:13.31 | ix33 | no difference with echocancel=on/off |
| 21:14.18 | ix33 | when it's a person at the far end (SIP to TDM), they report that it sounds fine. |
| 21:14.44 | Strom_L | ix33: did you call digium support yet? |
| 21:15.39 | *** join/#asterisk zgor (n=zgor@153.85-85-196.dynamic.clientes.euskaltel.es) |
| 21:16.00 | ix33 | Strom_L: no. i guess i will now. |
| 21:24.00 | [TK]D-Fender | ix33, Which is exactly what I told you to do a while back... |
| 21:24.44 | truent | no luck on ndiswrapper? |
| 21:25.36 | ix33 | well i was hoping it was my fault |
| 21:31.17 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 21:31.25 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 21:32.30 | hsv-al | hostility towards the mention of X101P |
| 21:32.31 | hsv-al | heh |
| 21:33.17 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 21:34.22 | *** join/#asterisk doolph (n=doolph@200.75.196.191) |
| 21:45.59 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 21:47.25 | adr3nalin3 | guys is it possible to have a remote FXOs with asterisk? |
| 21:48.57 | tzafrir_laptop | What do you mean by "remote"? |
| 21:49.20 | adr3nalin3 | For instance with my 3com pos I have a call processor in the central office then have analong line cards on other networks (connected via ipsec) that connect to the call proc. |
| 21:49.29 | adr3nalin3 | *analog |
| 21:50.45 | adr3nalin3 | oh duh as easy as a SIP ata? |
| 21:51.19 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@adsl-99-162-117-1.dsl.austtx.sbcglobal.net) |
| 21:51.30 | tzafrir_laptop | adr3nalin3, "remote" could be a satelite Asterisk server. It doesn't cost you an "extra license ;-) |
| 21:51.39 | adr3nalin3 | thats all it would really be. right? |
| 21:51.56 | tzafrir_laptop | But you can also use some FXO ATA |
| 21:52.19 | adr3nalin3 | tzafrir_laptop: you must be familiar with 3com. Just trying to cut costs so my org will ditch this f'in 3com money pit |
| 21:52.30 | tzafrir_laptop | Instead of ipsec, I'd use openvpn |
| 21:52.47 | CoffeeIV_ | I want to monitor a remote asterisk to make sure it is up and generate alerts if it is not. I want to do more than just ping the IP address, I'd like to check that asterisk itself was working . . . how do you guys recommend I do that ? |
| 21:52.51 | tzafrir_laptop | quite nicer. Especially traversing NAT |
| 21:53.33 | ix33 | ok does everybody here know that his is supposedly a known issue with the HPEC module?!? |
| 21:53.35 | tzafrir_laptop | CoffeeIV_, what services does this Asterisk give? VoIP as well? |
| 21:53.36 | adr3nalin3 | tzafrir_laptop: not having any NAT problems atm. Using cisco hardware |
| 21:54.03 | ix33 | Strom_L? [TK]D-Fender? |
| 21:55.10 | CoffeeIV_ | tzafrir_laptop: this asterisk accepts IAX2 connections, it also has a telnet manager -- I was thinking maybe my script could telnet into the manager and issue some sort of status command ? |
| 21:55.31 | tzafrir_laptop | CoffeeIV_, ping it through IAX2, then |
| 21:55.39 | tzafrir_laptop | This is the service that matters |
| 21:56.11 | adr3nalin3 | CoffeeIV_: I would use nagios or monit to telnet in to the service and look for the correct response |
| 21:57.19 | [TK]D-Fender | ix33, how... generic. and REDUNDANT |
| 21:57.41 | CoffeeIV_ | I have setup nagios and some other big monitoring packages, and I became disgusted with the bloat and complexity, and returned to writeing my own little script that just SMS me when something is down |
| 21:58.18 | adr3nalin3 | tried monit? it is very simple not full of bloat. I have the same opinion of nagios |
| 21:58.49 | CoffeeIV_ | I'll look into monit then, thanks for the tip |
| 21:58.55 | adr3nalin3 | np good luck |
| 22:00.02 | ix33 | excuse me. the echo cancellation module on t1 span cards that chops up audio when you're talking through it. |
| 22:01.17 | [TK]D-Fender | ix33, and you said it happens REGARDLESS of "echocancel=yes/no". So that sort of rules out EC in my mind |
| 22:02.30 | ix33 | [TK]D-Fender: i mentioned that to digium support guy. his only answer was: try the branch in SVN. |
| 22:03.50 | [TK]D-Fender | ok, stepping out for a while. |
| 22:05.39 | ix33 | is preston here? |
| 22:06.06 | tzafrir_laptop | ix33, so why not follow-up with the support guy? |
| 22:06.26 | ix33 | i will, when i have followed his instructions. |
| 22:06.35 | tzafrir_laptop | He must know of known issues with HPEC, if there are |
| 22:07.27 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-149.hsi.ish.de) |
| 22:08.37 | ix33 | is there an etiquette file for #asterisk? |
| 22:08.46 | [hC] | any of you guys experiencing issues when using a sangoma card that paging (app_page) doesnt reach all the phones in a large deployment? |
| 22:08.51 | [hC] | but if the wanpipe driver is gone, its fine. |
| 22:12.31 | errr | [hC]: how large is large in your case? |
| 22:13.18 | Corydon76-dig | ix33: No, there isn't |
| 22:13.36 | [hC] | errr: I'm paging to 80 phones in this instance. |
| 22:13.47 | errr | [hC]: my largest paging group is 120 but its spread acrosss 3 servers.. about 40 per server or so. We have had no problems |
| 22:13.52 | [hC] | errr: 80 polycom phones. |
| 22:14.05 | errr | [hC]: we are using all aastra 55i's here |
| 22:14.45 | [hC] | errr: I suspect the phone itself plays a part in this problem. What I have noticed however which is very strange is that if i remove the wanpipe driver, the problem seems to just go away. |
| 22:14.58 | errr | thats odd |
| 22:15.06 | [hC] | errr: The problem being that i get reports of "almost all the phones heard the page, but <x> number of them only caught the last 3 seconds" etc |
| 22:16.01 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
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| 22:27.45 | *** join/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
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| 22:52.16 | codehaxor | does asterisk do warm transfer? |
| 22:52.42 | Strom_L | do you mean attended transfer? |
| 22:52.48 | BCS-Satori | I assume that if I am install codec 729, the .so goes in /usr/lib/asterisk/modules. Do i need to register the module in modules.conf? |
| 22:53.02 | *** join/#asterisk erojasv (n=erojasv@190.40.84.106) |
| 22:53.24 | codehaxor | <Strom_L> do you mean attended transfer? --> yes |
| 22:53.37 | Strom_L | codehaxor: yes it does |
| 22:53.48 | codehaxor | <BCS-Satori> I assume that if I am install codec 729, the .so goes in /usr/lib/asterisk/modules. Do i need to register the module in modules.conf? ==> put the g729.so in there and reload the asterisk server |
| 22:54.16 | Strom_L | codehaxor: its really kind of irritating when you repeat the entire question someone asks |
| 22:54.35 | Strom_L | codehaxor: just the handle and your response is sufficient |
| 22:54.39 | drmessano | warm transfer? |
| 22:54.41 | BCS-Satori | codehaxor: so the autoload=yes takes care of the module? |
| 22:54.46 | drmessano | Is that like, hot swapping? |
| 22:56.58 | codehaxor | BCS: which g729 codec are you using? the opensource or the digium |
| 22:57.01 | codehaxor | ? |
| 22:57.31 | BCS-Satori | diguim |
| 22:58.09 | BCS-Satori | codehaxor: i am use to doing load=>modulename.so in modules.conf, not sure if it applies here or not |
| 22:58.10 | codehaxor | you need to register the codec first, i believe there is a registration binary |
| 22:58.48 | BCS-Satori | codehaxor: thats only if I want to change from codec to codec right? not if i just want to pass it to an endpoint that supports it right? |
| 22:58.52 | codehaxor | and there is also a registration and installation guide when you purchased that license |
| 22:59.31 | codehaxor | are you talking about pass through? |
| 23:01.04 | BCS-Satori | yes |
| 23:01.23 | codehaxor | if your using passthrough then you dont need the g729 codec in your asterisk server |
| 23:01.36 | codehaxor | you will only need it if you will be transcoding |
| 23:02.02 | codehaxor | thats if your sip phone does not support g729 |
| 23:03.26 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:03.58 | *** join/#asterisk Shotygun (n=thorn@82.166.244.147) |
| 23:04.37 | BCS-Satori | codehaxor: ahh, thanks |
| 23:05.01 | codehaxor | you just need to put an allow=g729 |
| 23:05.12 | codehaxor | to your outgoing trunk |
| 23:05.42 | codehaxor | and allow=g729 in your sip phone in the sip.conf |
| 23:08.12 | *** part/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
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| 23:10.50 | truent | anyone know of anyone having any luck with a wireless AN installation? |
| 23:11.12 | truent | meaning the AN server is connected to the lan wirelessly |
| 23:14.06 | *** join/#asterisk Vco (n=Vco@S0106000db905c200.cg.shawcable.net) |
| 23:14.45 | Strom_L | truent: uuugh, don't do that :/ |
| 23:15.32 | truent | reasons? |
| 23:15.46 | Vco | can anyone point me in the general direction of any documentation for using app_fax in 1.6? |
| 23:15.57 | Strom_L | it's your phone system. you don't want it relying on a wireless connection |
| 23:15.58 | truent | this is for home use mind you not some production environment |
| 23:16.46 | Strom_L | well, technically, it is a production environment :) |
| 23:17.01 | Strom_L | Vco: "core show application fax" at the CLI doesn't work? |
| 23:17.03 | truent | heh im running it off an older laptop.. fan gets a lil loud.. just wondering if i could throw it in the closet and forget about it ;p |
| 23:17.06 | Pimpachu | heh |
| 23:17.29 | Strom_L | truent: i don't know whether to laugh or to cry |
| 23:17.40 | truent | lol |
| 23:17.56 | *** join/#asterisk znoG (n=gs@host175.190-226-178.telecom.net.ar) |
| 23:18.14 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 23:18.21 | znoG | hey all, has anyone done any sort of Jabber integration? |
| 23:24.25 | mwalling | kinda |
| 23:25.04 | Vco | ahh, i'm guessing it's split into sendfax or receivefax rather than ust fax |
| 23:26.36 | [hC] | this is interesting. audio prompts on a new 1.2.28 installation are distorted... what the heck would cause that? |
| 23:27.00 | Strom_L | distorted how? |
| 23:27.27 | [hC] | er... distorted!... over gained... fuzzy.. |
| 23:27.32 | [hC] | not sure how else to describe that. |
| 23:27.51 | [hC] | audio distortion, not jitter or choppiness, if thats what you're getting at. |
| 23:28.11 | [hC] | it is transcoding from gsm -> g729 i've just noticed, but it should still sound alright |
| 23:28.17 | Strom_L | why are you installing 1.2.28? it's due to be abandoned soon |
| 23:28.33 | Strom_L | oh god, gsm to g729 is two entirely separate layers of icky quality |
| 23:28.39 | Strom_L | try installing the wav sound set |
| 23:28.47 | [hC] | it still should not happen. |
| 23:29.01 | Strom_L | well, fuck should and should not -- just try it |
| 23:29.39 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 23:30.21 | Strom_L | and if you like, PM me a telephone number or IAX2 URL or something where I can hear this distortion |
| 23:32.18 | [hC] | I'm gonna try changing up the audio files.. |
| 23:33.04 | Strom_L | it sounds like really terrible compression artifacting to me |
| 23:33.24 | Strom_L | I would see how much the audio file change helps things |
| 23:33.30 | Strom_L | who is your provider? |
| 23:33.53 | [hC] | Me. |
| 23:34.11 | deeperror | haha |
| 23:34.24 | [hC] | :P |
| 23:34.30 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 23:34.56 | [hC] | native sounds (g729) sound just fine of course. |
| 23:35.09 | [hC] | I shall try wav and see if those are messed up. |
| 23:37.54 | [hC] | yep, wav works. |
| 23:37.59 | [hC] | wtf ever, i dont care. |
| 23:38.47 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 23:39.00 | *** join/#asterisk Amun (n=Jonsi@c-71-205-221-174.hsd1.mi.comcast.net) |
| 23:39.23 | Strom_L | [hC]: g729 codec isn't designed to work on audio that's already compressed to gsm |
| 23:39.56 | [hC] | No, i know that.. but I've played gsm files before and they dont sound like that. maybe a newer asterisk, or newer codec is busted or something. I'm not overly concerned because i dont use gsm files anyways. that was an oversight. |
| 23:47.12 | *** join/#asterisk truent (i=0c224403@gateway/web/ajax/mibbit.com/x-7c2ee64e13e72812) |
| 23:53.40 | *** join/#asterisk LiNeTuX (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 23:54.24 | LiNeTuX | I just thought I'd share: Redfone rocks. |
| 23:55.14 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:56.29 | LiNeTuX | Query: is it typical for a provider to 'roll over' multiple PRI's (with separate D channels)? |