| 00:01.11 | *** join/#asterisk rv_weasel (n=no@adsl-66-143-44-198.dsl.ksc2mo.swbell.net) |
| 00:01.40 | budmang | anyone in? |
| 00:02.01 | *** join/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
| 00:02.16 | rv_weasel | is there a way to make it so that an agent must press a key before the call is connected. my agents are cell phones. so if it hits voicemail, the cellphone VM get the call. not an agent |
| 00:02.41 | designdream | budmang: nope.. i'm out... |
| 00:02.54 | budmang | Hey if i have a spare box running FC5 with a modem can I make a normal phone ring there? |
| 00:03.17 | TommyTheKid | Hi, so I may be missing something obvious, but is there a way to get the "count" (you are the only.. there are X other callers) when you join a MeetMe conference, but not have the obnoxious enter/leave noises.. q seems to make things *really* quiet.. more so than I desire :) |
| 00:06.11 | TommyTheKid | budmang: not without an FXS card |
| 00:06.30 | *** join/#asterisk pdt (n=ptinsley@h460a5701.area7.spcsdns.net) |
| 00:06.44 | TommyTheKid | s/card/interface/ (guess they have USB too) |
| 00:08.50 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:12.01 | budmang | TommyTheKid: doesnt a 56k modem work as a fxs card? |
| 00:12.21 | TommyTheKid | budmang: I dont think so, it has one FXO port and one "pass thru" |
| 00:14.37 | budmang | 3.2.1 FXO Cards |
| 00:14.37 | budmang | These cards allow you to connect a POTS (plain Old Telephone System) line to your Asterisk@Home box. |
| 00:14.49 | budmang | ahh |
| 00:14.55 | budmang | needs to draw the power |
| 00:17.26 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:17.33 | TommyTheKid | you could with a Linksys WRT54GP2 :) |
| 00:17.53 | rv_weasel | yeah, but they only run SIP right |
| 00:18.07 | tzanger | TommyTheKid: could do what with a WRT54GP2? |
| 00:18.17 | TommyTheKid | make a "regular" phone ring |
| 00:18.22 | harryvv | only sip though |
| 00:18.24 | tzanger | IIRC the phone part wasn't figured out yet |
| 00:18.35 | tzanger | it's been a while since I've been looking at it though |
| 00:18.51 | budmang | I have WRT54GP2 |
| 00:18.53 | tzanger | I just want a fucking PCI ADSL card with an FXO interface |
| 00:18.58 | budmang | with 2 ports |
| 00:19.36 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 00:19.40 | orlock | tzanger: no such beast that i know of! |
| 00:20.08 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-17.indy.res.rr.com) |
| 00:20.14 | orlock | tzanger: and theres only a few PCI ADSL cards that work with linux.. and the two main ones are actually the same PCB |
| 00:20.19 | TommyTheKid | tzanger: I assume you mean with Asterisk, and not the built in Linksys stuff :) |
| 00:20.21 | orlock | theres also bewan, which i havent used |
| 00:20.51 | rv_weasel | so is there a way to make the queue require an agent to press a KEY before call is connected |
| 00:21.05 | rv_weasel | my agents are cell phones. |
| 00:21.25 | rv_weasel | and i go into hospitals, etc alot and go off network |
| 00:22.06 | rv_weasel | so i want my queue to require the agent to hit say '#' before the other phones stop ringing |
| 00:22.06 | harryvv | rv_weasel your system is in a hospital |
| 00:22.12 | rv_weasel | no |
| 00:22.35 | rv_weasel | i own a computer shop |
| 00:22.40 | rv_weasel | and i work in the field |
| 00:23.14 | rv_weasel | 3 phones, office extention, partner cell, my cell |
| 00:23.29 | rv_weasel | so if i redirect to VM.... call lost |
| 00:23.34 | harryvv | i see |
| 00:23.39 | harryvv | it should not do that. |
| 00:23.53 | rv_weasel | and hell, if my phone is checking email, it gives congetions and goes to VM |
| 00:25.10 | *** join/#asterisk LordScinawa (n=adsf@host216-237.pool870.interbusiness.it) |
| 00:26.18 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 00:27.41 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 00:29.28 | *** join/#asterisk anthm (n=anthm@000-403-290.area4.spcsdns.net) |
| 00:29.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:32.50 | TommyTheKid | does anyone have a good way to get h.323 trunks (from an Avaya PBX) into asterisk.. we haven't had much luck with ooh323 and company |
| 00:35.54 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 00:36.14 | *** join/#asterisk Iam8up (n=iam8up@user-0cdv282.cable.mindspring.com) |
| 00:36.26 | Iam8up | what are some of the popular ip phones among the users here? |
| 00:36.40 | TommyTheKid | Polycom 501 here |
| 00:36.54 | Iam8up | got one of those at work..i like it, it's not bad at all |
| 00:36.58 | TommyTheKid | er.. lots of them :) |
| 00:37.20 | Iam8up | any other popular phones with asterisk users? |
| 00:37.23 | TommyTheKid | I like having a couple on my desk, calling into a conference call and laughing maniachally |
| 00:37.30 | Iam8up | lol... |
| 00:38.17 | TommyTheKid | the metallic echo sequence is way cool (I should note that I am ~200ms from the gateway) |
| 00:39.58 | TommyTheKid | I am thinking there needs to be an option to record app_echo too :) |
| 00:50.07 | *** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br) |
| 00:52.00 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 00:52.43 | TheCops | Hi, I have some Auto destroying call /Stopping retransmission on in my debug log of my asterisk server...someone know what asterisk doing? |
| 00:54.20 | *** join/#asterisk docelmo (n=Snake@55-65.126-70.tampabay.res.rr.com) |
| 00:54.31 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:57.25 | *** join/#asterisk pdt (n=ptinsley@adsl-154-211-201.ard.bellsouth.net) |
| 00:57.45 | TommyTheKid | SO, if I wrote a simple patch to MeetMe to allow it to give me the number of people in a conference when pressing *,#, would that get accepted back.. ie should I bother even submitting it? |
| 01:01.07 | Qwell | I would use 5, personally |
| 01:01.16 | docelmo | in all honesty no.. The core developers in charge are very anal and probably wouldnt accept it. Also when you add something to the tree you have to sign your life away.. |
| 01:01.20 | docelmo | Right Qwell? |
| 01:01.21 | Qwell | though, # does make sense |
| 01:01.48 | *** join/#asterisk james_ (i=jamesdot@creep.bur.st) |
| 01:03.06 | rv_weasel | i cannot find anyway to make it so an agent must "accept" a call by pressing a tou-tone |
| 01:03.16 | rv_weasel | touch-tone |
| 01:08.35 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 01:10.48 | Corydon76-home | "the core developers are very anal" wtf? |
| 01:11.09 | rv_weasel | anal can be a good thing too.... |
| 01:12.20 | infinity1 | i'm having polycom and DTMF issues. everyone is set correctly to dtmf2833. any ideas what might be wrong or how to troubleshoot |
| 01:17.39 | *** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net) |
| 01:18.47 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:19.44 | intralanman | infinity1: dtmfmode=rfc2833? |
| 01:20.37 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:22.25 | infinity1 | intralanman: i tried that. |
| 01:22.46 | infinity1 | i'm stumpted. when i call places, DTMF sorta works ,but not relaibly |
| 01:23.10 | infinity1 | i have the default settinsg in the polycom phone and i set dtmfmode. if i set it to inband it doesn't work at all |
| 01:32.32 | *** join/#asterisk SwK (n=Silik0nJ@c-24-99-246-180.hsd1.ga.comcast.net) |
| 01:36.17 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.156) |
| 01:36.36 | tengulre11 | hi,all |
| 01:36.53 | mitcheloc | hello! |
| 01:42.14 | *** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
| 01:42.28 | salaud | anyone know what /var/spool/asterisk/qcall is about? |
| 01:42.51 | salaud | I understand /var/spool/asterisk/outgoing ... but I can't find any reference to QCall |
| 01:43.11 | salaud | no hits for qcall on the asterisk wiki |
| 01:43.58 | TommyTheKid | not that I know anyhting... but.. cd asterisk-source.. |
| 01:44.17 | TommyTheKid | find . -type f -exec grep -l qcall {} \; |
| 01:44.54 | salaud | I didn't compile from source... and not really feeling like taking it all the way down there... but, I know that's an option |
| 01:45.10 | salaud | WTFM - Write the Fine Manual |
| 01:46.02 | TommyTheKid | hehe |
| 01:46.06 | TommyTheKid | I gotta go back to work.. later |
| 01:46.12 | *** part/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
| 01:46.13 | salaud | later |
| 01:47.17 | Lyfe | hehehe.. i always liked how many words started with F, for RTFM, WTFM, etc :P |
| 01:47.35 | *** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br) |
| 01:51.23 | *** join/#asterisk oomph (n=oomph@69-175-199-236.chvlva.adelphia.net) |
| 01:57.40 | *** join/#asterisk Floodbar (n=Flood@ip68-12-150-145.ok.ok.cox.net) |
| 01:59.12 | intralanman | Lyfe: yeah, i like Friendly (especially since most aren't really all that friendly) |
| 01:59.43 | Lyfe | hehehe |
| 02:00.44 | Floodbar | so what's going on in the asterisk channel tonight |
| 02:03.12 | *** join/#asterisk darkgamer20 (n=chatzill@adsl-71-146-176-46.dsl.pltn13.sbcglobal.net) |
| 02:03.21 | LordScinawaz | hum |
| 02:03.24 | intralanman | is there any way to add time to a dial command if L() was used? |
| 02:03.25 | LordScinawaz | hello guys :D |
| 02:03.46 | LordScinawaz | where can i find a startup guide to asterisk? |
| 02:03.48 | Floodbar | hello |
| 02:04.04 | darkgamer20 | what is more stable the asterisk from svn, or the one I see on asterisk.org? |
| 02:04.27 | LordScinawaz | :S |
| 02:04.31 | Floodbar | the compressed files on asterisk.org |
| 02:04.42 | darkgamer20 | oh ok |
| 02:04.47 | darkgamer20 | thanks Floodbar |
| 02:05.00 | intralanman | i'm gonna guess that's a no?!? ;) |
| 02:05.17 | Floodbar | what is the L() command |
| 02:05.31 | darkgamer20 | LordScinawaz: you can use the asterisk: the future of telephony book, download it for free at asteriskdocs.org |
| 02:05.45 | LordScinawaz | thanx! |
| 02:05.49 | darkgamer20 | no prob |
| 02:06.04 | intralanman | <PROTECTED> |
| 02:06.05 | intralanman | <PROTECTED> |
| 02:06.20 | Floodbar | http://www.asteriskdocs.org/modules/news/ is a good place to start for asterisk |
| 02:06.46 | Floodbar | ahh |
| 02:08.48 | *** join/#asterisk ManxPower (n=ewieling@71-8-11-111.dhcp.leds.al.charter.com) |
| 02:11.57 | *** part/#asterisk Skyelar (n=planet@222-153-145-60.jetstream.xtra.co.nz) |
| 02:12.00 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 02:12.43 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-17.indy.res.rr.com) |
| 02:17.46 | *** join/#asterisk sugardave (n=sugardav@cpe-70-112-127-158.austin.res.rr.com) |
| 02:17.48 | *** join/#asterisk r_evolution (n=no@208.251.203.208) |
| 02:17.54 | r_evolution | best wtf of the night |
| 02:17.57 | r_evolution | i love it |
| 02:18.06 | r_evolution | so I setup * to function realtime with mysql right? |
| 02:18.15 | r_evolution | i check the conn status in the * CLI |
| 02:18.31 | r_evolution | it tells me that I have been connected for 36 years, 236 days, 1 hours, 40 minutes and some seconds |
| 02:18.39 | Snake-Eyes | lol |
| 02:18.42 | r_evolution | now THAT is some fucking up-time |
| 02:18.44 | intralanman | niiiiiice |
| 02:18.50 | LordScinawaz | lol |
| 02:18.53 | Floodbar | awesome |
| 02:19.05 | sugardave | ok: iax2-a -> NAT -> NAT -> * -> iax2-b; iax2-b can call a, but a gets congestion or unavailable when calling b |
| 02:19.16 | r_evolution | yesss... i took pictures... im sending em to all the windows admins tomorrow |
| 02:19.20 | r_evolution | so i can be like can your systems match THIS |
| 02:19.30 | intralanman | r_evolution: need a job? heheh, last time i saw 5 9's was in a ping to singapore ;D |
| 02:19.32 | Snake-Eyes | hahaha |
| 02:19.54 | r_evolution | singapore scares me ;x |
| 02:19.59 | r_evolution | i still remember the kid YEARS ago |
| 02:20.03 | r_evolution | getting caned for graffiti |
| 02:20.16 | r_evolution | i do much worse than graffiti |
| 02:20.23 | r_evolution | they'd prolly shove a cattle prod up my ass |
| 02:20.40 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 02:21.30 | r_evolution | hey justin you missed it |
| 02:21.38 | justinu|laptop | what happened? |
| 02:21.40 | r_evolution | apparently the newest * i'm working on here |
| 02:22.00 | r_evolution | has managed to stay connected to the mysql db for 36 years, 236 days, 1 hours 40 minutes and some seconds |
| 02:22.47 | r_evolution | i knew * was good |
| 02:22.51 | r_evolution | didnt know it could time travel |
| 02:22.59 | justinu|laptop | haha |
| 02:24.33 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 02:24.56 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 02:27.53 | [TK]D-Fender | r_evolution : "load chan_fluxcapacitor.so" |
| 02:28.21 | *** join/#asterisk linlin (i=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 02:28.56 | yxa | can saydigits and saynumber be "backgrounded" so that users can just press somthing to skip it |
| 02:28.57 | r_evolution | yesssss |
| 02:29.28 | r_evolution | exactly andrew! load app_marty_mcfly.so |
| 02:33.18 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-51-202.cybersurf.com) |
| 02:34.00 | r_evolution | We live in an expanding universe. All of it is trying to get away from Chuck Norris. |
| 02:34.43 | justinu|laptop | i thought it was van damme we didn't like |
| 02:35.05 | r_evolution | www.chucknorrisfacts.com |
| 02:35.12 | r_evolution | Chuck Norris roundhouse kicks don't really kill people. They wipe out their entire existence from the space-time continuum. |
| 02:35.43 | Iam8up | what are some of the popular ip phones among the users here? |
| 02:36.02 | justinu|laptop | tthat's almost as much of a waste of time as timecube.com |
| 02:36.49 | rv_weasel | so is there a way to make the queue require an agent to press a KEY before call is connected |
| 02:37.01 | rv_weasel | so i want my queue to require the agent to hit say '#' before the other phones stop ringing |
| 02:37.05 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 02:37.06 | r_evolution | timecube? |
| 02:37.14 | rv_weasel | i read the dev docs. cant find it |
| 02:37.25 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 02:37.31 | justinu|laptop | yeah... you didn't know there was really 4 simultaneous 24 hour days in one rotation of the earth? |
| 02:37.32 | r_evolution | wasn't that in fact the default for the only queues? |
| 02:37.59 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 02:38.19 | r_evolution | weasel... you need to re-read those docs then man |
| 02:38.19 | rv_weasel | i dont understand the question |
| 02:38.20 | ManxPower | intralanman, apparently you've never been on a DirecWay connection |
| 02:38.28 | r_evolution | MANX |
| 02:38.33 | rv_weasel | the # thing is for logon |
| 02:38.37 | r_evolution | Did you see my super-server-up-time-across-time? |
| 02:39.06 | ManxPower | r_evolution, Yup. |
| 02:39.12 | r_evolution | incredible isn't it? |
| 02:39.18 | r_evolution | 36 years... almost 37! |
| 02:39.23 | r_evolution | of up-time |
| 02:39.31 | r_evolution | im very proud... b/c I just built that box this morning |
| 02:39.45 | justinu|laptop | that puts you back into the System/360 era |
| 02:39.53 | intralanman | ManxPower: you're right..... although i've heard about them |
| 02:40.05 | r_evolution | and justin... tell me more of this 4 24 hr day theory |
| 02:40.09 | justinu|laptop | lol |
| 02:40.10 | r_evolution | does that mean i can go back to yesterday? |
| 02:40.12 | rv_weasel | r_evolution: perhaps you dont understand, call enters queue rings cell phones. if one phone is out of rang it goes to cell VM |
| 02:40.17 | justinu|laptop | http://en.wikipedia.org/wiki/Time_cube |
| 02:40.17 | ManxPower | intralanman, ping times 900ms - 3000ms |
| 02:40.20 | ManxPower | on a good day. |
| 02:40.29 | intralanman | ManxPower: ouch |
| 02:40.31 | rv_weasel | if user had to confirm they answered. no problems |
| 02:40.53 | ManxPower | rv_weasel, that is correct. nothing you can do about it except turn off the cell phone voicemail service |
| 02:40.54 | intralanman | ManxPower: how do you get VoIP to work on that? do you get VoIP to work on that? |
| 02:40.54 | justinu|laptop | Humans are Cubic forms that rotate a 4 corner face lifetime. |
| 02:41.04 | ManxPower | intralanman, you don't. |
| 02:41.16 | ManxPower | intralanman, Which is why I've been considering getting an actual T-1. |
| 02:41.27 | rv_weasel | ManxPower: That is what i thought. would be nice feature though! |
| 02:41.27 | ManxPower | I just wish I could find a way to do it that costs less than $700/month |
| 02:41.55 | rv_weasel | we looked at ds3 8K |
| 02:41.58 | file | ManxPower: sell your soul a few times |
| 02:42.04 | intralanman | ManxPower: do you just live out in the stix somewhere that you can't get a decent DSL/cable connection? |
| 02:42.18 | ManxPower | file, working on that. |
| 02:42.28 | ManxPower | intralanman, 11 miles from the CO. |
| 02:42.33 | rv_weasel | home and shop dsl with 768/6118 |
| 02:42.37 | [TK]D-Fender | r_evolution : For your Chuck Norris rant : http://media.putfile.com/ultimateshowdown |
| 02:42.45 | r_evolution | -1 × -1 = +1 is stupid and evil. |
| 02:42.57 | justinu|laptop | Dr. Gene Ray offers Wikipedia $10,000.00 to disprove math that 1 rotation of 4 Earth quadrants within the 4 quarter Harmonic Time Cube does create 4 simultaneous 24 hour days. |
| 02:42.58 | justinu|laptop | heh |
| 02:42.59 | ManxPower | What I'll prolly do is get a point-to-point T-1 to somewhere in town, near the CO |
| 02:43.31 | ManxPower | All I need is a closet with A/C |
| 02:43.47 | intralanman | ManxPower: you made me consider how lucky i am.... i live out in the stix, but am somehow close enough to the CO to get a decent dsl speed |
| 02:44.03 | rv_weasel | mine is excelent |
| 02:44.03 | r_evolution | sweet TK |
| 02:44.19 | *** join/#asterisk michaelo (n=michaelo@adsl-147-45-179.gsp.bellsouth.net) |
| 02:44.25 | rv_weasel | very low latency to my DID and rate centers |
| 02:44.25 | sugardave | ManxPower: you were asking me last night if I was using RSA auth with my problem IAX2 trunks |
| 02:44.41 | rv_weasel | ulaw all the way |
| 02:45.00 | Lyfe | I have a call queue with an agent defined, and the agent is logged in using AgentCallbackLogin. If the agent fails to answer the phone, the call gets routed to their voicemail, which is contradictory to what I expected to happen, since I had no idea that it even knew how to get there. Do I have a mistaken predisposition as to how this works, or is it operating correctly, and I need to adjust my diaplan? |
| 02:45.02 | sugardave | and the answer is yes, I was...but they've had me remove it and I still see the occasional authentication problem and no call coming in |
| 02:45.52 | ManxPower | sugardave, I was just making sure you realized you were. |
| 02:46.01 | sugardave | oh, okay, thanks :) |
| 02:46.10 | Floodbar | Lyfe: I have the same setup. I setup an agent macro that has no voicemail. |
| 02:47.10 | Lyfe | floodbar: so you just have agents then that have no voicemail setup, and the call never leaves the queue because of that (if they fail to answer)? |
| 02:47.25 | [TK]D-Fender | Lyfe : Adjust your dialplan. Chan_local does whatever you point it to doing. make sure NOT to Answer the channel in anything other than a Dial or the fcall will leave the queue |
| 02:47.38 | [TK]D-Fender | Lyfe : That means using extens /wo Vm attached, etc |
| 02:47.40 | r_evolution | im leaving |
| 02:47.43 | r_evolution | peace out hippies |
| 02:48.11 | Floodbar | the call never leaves the queue because there is not a vm to go to |
| 02:48.51 | Lyfe | [TK]D-Fender: ok, that's kinda what i figured i'd need to do. Thanks. Is there a cleaner solution to queues, or is simply having agents without VM the cleanest solution? |
| 02:49.04 | [TK]D-Fender | Lyfe : Definately w/o VM |
| 02:49.16 | ManxPower | Don't use callbacklogin |
| 02:49.52 | Floodbar | then what do you use to log in your agents |
| 02:49.59 | Lyfe | ManxPower: i know, it's being deprecated. I just haven't found a good document describing how similar functionality with another feature is implemented. |
| 02:50.02 | [TK]D-Fender | Lyfe : usually you use it so you can do dial-plan based forwarding for quu members or use variable resource or other nify mods that you need exectuted during the agent callout |
| 02:50.26 | ManxPower | Floodbar, Actually my agents don't login, but as Lyfe said, there are other ways to do it. |
| 02:50.40 | Lyfe | [TK]D-Fender: hmm.. can you elaborate? (perhaps with urls to doc's, or whatnot) |
| 02:51.36 | Lyfe | ManxPower: Well, the only reason I know is because I read it on a web cache of a mailing list. I have no idea what the proper way to implement it with a dialplan is. |
| 02:52.06 | ManxPower | Any way to use AgentLogin rather than agencallbacklogin? |
| 02:52.14 | Lyfe | ManxPower: I just presume it's using AddQueueMember and RemoveQueueMember (or whatever the functions are called).. But I haven't seen a document on it. |
| 02:52.31 | Lyfe | not really, AgentLogin works differently. |
| 02:53.32 | [TK]D-Fender | Lyfe :imagine you want to "push" a web page onto the client of the agent who is recieving the call for a CSR screen-pop.. thats a place to trigger it before dialing. |
| 02:54.10 | Lyfe | [TK]D-Fender: damn, you just described a feature I need. now I just need to know how it's done. |
| 02:54.24 | [TK]D-Fender | Lyfe : :D |
| 02:54.53 | yxa | can saydigits and saynumber be "backgrounded" so that users can just press somthing to skip it |
| 02:55.05 | Lyfe | but, i'm still in the dark on how to do it, having not found any docs on it (or maybe i'm just missing it) |
| 02:55.27 | ManxPower | Lyfe, sometimes you need to be creative. |
| 02:55.30 | [TK]D-Fender | Lyfe : no such docs... you have to come up with your own method. |
| 02:56.38 | Lyfe | ouch.. you had my hopes up for a moment that there might be an existing solution, whereas I was simply going to take advantage of the manager interface in relation to grabbing queue events. |
| 02:57.11 | *** join/#asterisk evilbit (i=hhoffman@gateway/tor/x-aedd50f639a69eee) |
| 02:59.31 | Lyfe | would anyone know if the add/remove queuemember functions are indeed the way to avoid using agentcallbacklogin? |
| 02:59.52 | Floodbar | I don't know I use agentcallback |
| 03:00.26 | evilbit | would anyone care to look at a agi I wrote and tell me what I'm doing wrong? |
| 03:00.29 | Lyfe | Floodbar: as do i.. i just know how to use that already, and plan to switch away from it by the time 1.4 rolls around, so i can get away from them, as they are apparently scheduled to be deprecated. |
| 03:00.48 | Floodbar | I was not aware of that |
| 03:01.19 | Floodbar | I have three phones that 12 agents use for the one queue |
| 03:01.41 | Lyfe | Floodbar: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackLogin << see "new for upcoming v1.4.0 release" |
| 03:02.54 | Lyfe | that's the only reason i know. |
| 03:03.42 | Floodbar | that has just recently been added as of a few months ago |
| 03:04.03 | Lyfe | not suprised. |
| 03:04.04 | Floodbar | when I set up the queues, that was the way to do it |
| 03:04.31 | Floodbar | or the way I found to do it |
| 03:05.03 | Lyfe | i recognize that the biggest issue with callback is that there's no good way to replicate that to backup servers. so.. yeah, not suprised they're talking about deprecating it. |
| 03:05.03 | EyeCue | hmm, if im dialing an extension that the user who is allocated to it isnt registered on, how do i bring up a message or somethin? or provide a automatic hangup/busy signal |
| 03:05.04 | Floodbar | and I had the same problems you did with my calls going to voicemail |
| 03:06.43 | Lyfe | Floodbar: yeap. anyway.. thanks for the insight. I'm gonna take off for the night. |
| 03:06.57 | Floodbar | have a good night |
| 03:07.04 | Lyfe | [TK]D-Fender: thanks as well. Just wish there was more documentation on the alternative stuff. |
| 03:08.10 | Lyfe | Floodbar: since i saw this befor eleaving, you might find it interesting as well: http://www.voip-info.org/wiki/view/Agents+without+agent+channel |
| 03:08.23 | Floodbar | thank you |
| 03:09.45 | *** join/#asterisk SwK (n=Silik0nJ@c-24-99-246-180.hsd1.ga.comcast.net) |
| 03:09.46 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 03:12.57 | Floodbar | is there an eta to when 1.4 will be coming out |
| 03:13.52 | file | ...when it's ready |
| 03:13.57 | Floodbar | alright |
| 03:13.59 | Floodbar | I love it |
| 03:14.00 | file | the more people ask, the longer it'll take |
| 03:14.17 | Floodbar | hey my 1.2 is running just fine |
| 03:14.41 | Floodbar | only have one minor problem with it which I have worked around |
| 03:15.38 | Floodbar | its actually has an uptime of like 4 or 5 months |
| 03:15.46 | Floodbar | been up ever since I put it in |
| 03:16.10 | ManxPower | We have to reboot some of our servers every week or two. |
| 03:16.33 | Floodbar | I used to have to do that in previous installs |
| 03:16.50 | Floodbar | but since this install has a pri it works so much better |
| 03:17.25 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
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| 03:39.26 | darkgamer20 | hey guys is it uncommon if zttool is not installed with zaptel? but ztcfg is |
| 03:39.54 | CunningPike | darkgamer20: I have had to install it separately in the past |
| 03:40.37 | darkgamer20 | CunningPike: so now with the latest it should be installed with zatel right? |
| 03:41.07 | *** join/#asterisk s0lid (n=jlq@203.177.33.138) |
| 03:41.35 | CunningPike | darkgamer20: It's supposed to be, but I think I had to make and install it separately the last time I did a clean install (some time ago, so ymmv) |
| 03:43.30 | darkgamer20 | CunningPike: hmm how can i do that? is there some other seprate package? |
| 03:44.35 | hads | It should be built even if it isn't installed. Just copy it to where you want it. |
| 03:45.09 | *** join/#asterisk jets (n=jets@root.ownsu.com) |
| 03:45.37 | darkgamer20 | hmm |
| 03:45.39 | darkgamer20 | ok |
| 03:46.02 | hads | Or run it from there. |
| 03:46.10 | darkgamer20 | hads: its wasnt built... |
| 03:46.16 | darkgamer20 | but everything else was |
| 03:46.19 | Un1x | is there anyway to increase volume with asterisk or way to amplify it i get calls from SIp but there volume is wayyy low.. |
| 03:46.21 | Un1x | anyway to fix this |
| 03:46.34 | hads | Use the volume on your handset. |
| 03:46.42 | Un1x | i did |
| 03:46.44 | Un1x | it's at max |
| 03:46.47 | Un1x | still not much |
| 03:46.54 | Un1x | is there anyway to do it via asterisk? |
| 03:47.16 | hads | darkgamer20: You will need libnewt to build it, that may be why it wasn't built. |
| 03:47.27 | darkgamer20 | hads: oh thanks |
| 03:47.51 | darkgamer20 | Un1x: maybe your mixer settings have the volume to be low, check that |
| 03:48.00 | Un1x | how so? |
| 03:48.05 | Un1x | how do i check it? |
| 03:48.36 | darkgamer20 | use alsamixer |
| 03:48.53 | Un1x | so it's not part of *? |
| 03:49.13 | Un1x | this is for soundacard dude, my phone is plugged into TDm400P |
| 03:49.18 | Un1x | but my calls are via SIP.. |
| 03:49.32 | hads | How should we know if you don't tell us... |
| 03:49.37 | darkgamer20 | lol yea |
| 03:49.48 | darkgamer20 | wait he did tell us |
| 03:50.03 | darkgamer20 | sorry i assumed that your sip client was on the same computer as your asterisk |
| 03:51.09 | Un1x | nah |
| 03:51.18 | Un1x | i use a analogue phone |
| 03:51.27 | Un1x | but it's plugged intop TDM4-00P and calls are via SIP |
| 03:51.43 | CunningPike | darkgamer20: Try 'make zttool' from your zaptel source folder |
| 03:52.30 | darkgamer20 | CunninyPike: tried that, hads told me what the problem was already, i didnt have libnewt installed, i guess slackware dosent have that by default, thanks for your help though |
| 03:53.04 | CunningPike | darkgamer20: Ah - that'll do it, every time ;) |
| 03:53.12 | darkgamer20 | yep |
| 03:53.28 | *** join/#asterisk BugKham (n=bugkham@ppp-58.8.3.80.revip2.asianet.co.th) |
| 03:53.59 | BugKham | anyone knows how to call out from the US? |
| 03:54.07 | BugKham | 001+country code? |
| 03:54.13 | x86 | 011 |
| 03:54.28 | BugKham | x86, thanks |
| 03:54.32 | x86 | 011+1+Area Code+Number |
| 03:54.55 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 03:54.58 | x86 | 011-1-212-555-2424, for example |
| 03:55.02 | BugKham | x86, from US to other country? |
| 03:55.04 | Un1x | comon |
| 03:55.05 | Un1x | thats no fair |
| 03:55.07 | Un1x | no one helping me |
| 03:55.12 | x86 | that's from another country to the US |
| 03:55.31 | x86 | you want from the US to another country? |
| 03:55.35 | BugKham | x86, yes |
| 03:55.47 | x86 | 011+Country Code+Number |
| 03:56.28 | BugKham | x86, ok |
| 03:56.29 | darkgamer20 | Un1x: idk whats the problem, its clearly not with asterisk if your able to make calls to sip clients |
| 03:57.00 | orlock | will asterisk automatically load cdr_mysql.conf? |
| 03:57.12 | darkgamer20 | x86: wait, why the 011? |
| 03:57.29 | x86 | darkgamer20: eh, to tell the telco you're making an international call :) |
| 03:57.37 | darkgamer20 | oh ok |
| 03:57.39 | Un1x | darkgamer youre not understanding |
| 03:57.42 | CunningPike | darkgamer20: 011 is the international dialing access code from NANPA |
| 03:57.53 | x86 | orlock: if cdr_mysql.so gets loaded, yes |
| 03:58.04 | darkgamer20 | CunningPike: i see |
| 03:58.05 | CunningPike | darkgamer20: So, Dublin, Ireland is 0113531xxxxxxx |
| 03:58.21 | darkgamer20 | i see |
| 03:58.30 | x86 | heh, we need more contries in NANPA :P |
| 03:58.42 | darkgamer20 | arent all countries in NANPA? |
| 03:58.56 | darkgamer20 | or you were kidding? |
| 03:58.56 | x86 | darkgamer20: NANPA == North American numbering plan agreement |
| 03:58.59 | CunningPike | x86: We have plenty enough already, thanks - toll fraud, anyone? |
| 03:59.08 | x86 | CunningPike: hah |
| 03:59.35 | x86 | darkgamer20: only countries you can dial with a 1 country code are covered by NANPA |
| 03:59.50 | darkgamer20 | h |
| 03:59.55 | darkgamer20 | s/h/oh |
| 03:59.59 | x86 | darkgamer20: US, Canada, Bahamas, Guam, Peurto Rico, Jamaica, some other places too |
| 04:00.13 | x86 | no idea how a south american country got into the NANPA :P |
| 04:00.18 | darkgamer20 | lol |
| 04:00.53 | *** join/#asterisk tonsofpcs (n=tonsofpc@ool-435385da.dyn.optonline.net) |
| 04:00.57 | darkgamer20 | anyone know the homepage of libnewt, or the download of it? |
| 04:02.03 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 04:03.13 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.64.250) |
| 04:03.37 | *** join/#asterisk VHost`[NetENG]Ki (n=kchau@24-205-178-193.dhcp.wsco.ca.charter.com) |
| 04:03.48 | *** join/#asterisk topping (n=topping@207.47.6.207.static.nextweb.net) |
| 04:03.51 | tonsofpcs | I see a kiet |
| 04:04.08 | jets | ~glog |
| 04:04.29 | VHost`[NetENG]Ki | *slap |
| 04:04.33 | VHost`[NetENG]Ki | lol |
| 04:04.48 | VHost`[NetENG]Ki | stupid voicestick/i2telecom is killing me! |
| 04:05.15 | ManxPower | Uh, what south american country is in nanpa? |
| 04:05.32 | ManxPower | darkgamer20, yoiu must be using Slackware. |
| 04:05.42 | darkgamer20 | ManxPower: yep |
| 04:05.44 | tonsofpcs | ManxPower: no. |
| 04:05.56 | orlock | damn, where does cdr_mysql.so come from? |
| 04:06.00 | tonsofpcs | http://www.nanpa.com/area_codes/index.html |
| 04:06.05 | ManxPower | darkgamer20, Ya want to know how I know? It's the only distro I know of that does not ship libnewt with it. |
| 04:06.08 | tonsofpcs | that's who nanpa is |
| 04:06.42 | ManxPower | "x86 no idea how a south american country got into the NANPA :P" |
| 04:06.51 | darkgamer20 | ManxPower: yea i just found out lol |
| 04:07.51 | darkgamer20 | ManxPower: anyway to compile or install libnewt on slack? |
| 04:08.15 | ManxPower | darkgamer20, no idea. I always have garlic and holy water with me when I'm around slackware |
| 04:08.22 | darkgamer20 | lol |
| 04:08.28 | ManxPower | It's a great distro to LEARN on. |
| 04:08.44 | darkgamer20 | yea very stable |
| 04:09.12 | ManxPower | You do everything manually, which is why it's good to learn on. |
| 04:09.21 | *** join/#asterisk slinabery (n=slinaber@smooth.worldcycling.com) |
| 04:09.25 | *** join/#asterisk Skyelar (n=planet@222-153-145-60.jetstream.xtra.co.nz) |
| 04:09.26 | orlock | then you bitch about rpm's cos you dnt understand tem |
| 04:09.31 | orlock | then you use debian |
| 04:09.35 | tonsofpcs | lol orlock |
| 04:09.36 | orlock | then you realise theres no real difference |
| 04:09.42 | orlock | :) |
| 04:09.51 | ManxPower | I'm a fan of Mandrake/Mandriva |
| 04:09.56 | tonsofpcs | then you find the magic of xeyes |
| 04:10.04 | ManxPower | TODO list: upgrade postfix |
| 04:10.21 | ManxPower | # rpmi -Fvh postfix-whatever.rpm |
| 04:11.05 | x86 | ManxPower: why did you quote me? |
| 04:11.43 | ManxPower | x86, I was wondering what south american country you are referring to. |
| 04:12.19 | x86 | ManxPower: jamaica :) |
| 04:12.32 | ManxPower | x86, that's not south american |
| 04:12.46 | tonsofpcs | Jamaica is not in South America |
| 04:12.49 | jbroome | wow |
| 04:12.51 | x86 | eh? |
| 04:13.48 | ManxPower | You do realize that Mexico and Beleze are part of North American too, right? |
| 04:14.02 | x86 | yeah |
| 04:14.27 | x86 | ah, Jamaica is part of the Greater Antilles |
| 04:14.40 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 04:17.48 | ManxPower | I guess that would depend on how you define North America, geology or politics |
| 04:17.55 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 04:17.56 | *** part/#asterisk omal (n=omal@cpe-24-164-111-184.neo.res.rr.com) |
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| 04:19.19 | *** part/#asterisk nailbags|laptop (n=neil@203-206-217-36.perm.iinet.net.au) |
| 04:20.04 | CunningPike | When I was in school, Canada and USA were North America. The rest was Latin America |
| 04:20.35 | CunningPike | I think the geopolitical viewpoint has shifted since then :) |
| 04:20.36 | ManxPower | CunningPike, that would be defining North America by cultural boundaries. |
| 04:20.59 | CunningPike | ManxPower: As was the norm when I was in school |
| 04:21.30 | orlock | while to the rest of the world, the USA is just "those fuckin merkins" |
| 04:21.52 | ManxPower | Personally I think a tectonic boundry system would be the best |
| 04:21.56 | Juggie | theres nothing north about mexico |
| 04:22.02 | CunningPike | ManxPower: You would ;) |
| 04:22.13 | orlock | gahh, i didnt have mysql-devel installed |
| 04:22.23 | orlock | thats why mysql_cdr didnt compile :) |
| 04:22.27 | CunningPike | ~lart orlock |
| 04:22.33 | ManxPower | Granted, that would put part of california as part of asia, but I've lived in California -- it would be no big loss. |
| 04:22.39 | hads | http://en.wikipedia.org/wiki/North_America |
| 04:22.42 | xai | its north of the equator.. north of central. |
| 04:22.42 | CunningPike | ManxPower: lol |
| 04:23.40 | *** join/#asterisk _mh (n=largo@202.5.145.13) |
| 04:23.42 | CunningPike | ManxPower: I think if you use cultural boundaries, California could be considered part of Asia anyway, these days |
| 04:24.03 | xai | or part of mexico, depending on which side you pick. |
| 04:24.24 | CunningPike | True |
| 04:24.55 | ManxPower | http://en.wikipedia.org/wiki/Image:Tectonic_plates.png |
| 04:25.53 | justinu|laptop | donde esta el banjo? |
| 04:26.21 | CunningPike | Le singe est dans l'arbre |
| 04:26.22 | xai | esta en la music room |
| 04:26.56 | *** join/#asterisk jake1932 (n=Administ@pool-70-16-129-225.phil.east.verizon.net) |
| 04:27.06 | jake1932 | ~seen bkw |
| 04:27.17 | jbot | bkw <n=bkw@k7j231-2.kam.afb.lu.se> was last seen on IRC in channel #debian, 223d 16h 28m 26s ago, saying: 'Anyone who can explain why a nic sometimes become eth0, others eth1. This really confuse dhclient during bootups.'. |
| 04:27.28 | jake1932 | ~seen bkw__ |
| 04:27.30 | jbot | bkw__ is currently on #asterisk, last said: 'why did they ask if they were going to ask again'. |
| 04:31.15 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.64.250) |
| 04:33.18 | slinabery | sigh. does anyone here have any expertise with e&m wink signalling? |
| 04:33.25 | *** part/#asterisk CANO-1982 (n=alejandr@190.48.64.250) |
| 04:33.56 | justinu|laptop | i know enough to know it doesn't work right on asterisk |
| 04:34.41 | jake1932 | it doesn't? |
| 04:34.47 | justinu|laptop | not in my experience |
| 04:35.03 | jake1932 | i think I'm using it and it works fine |
| 04:35.15 | CunningPike | slinabery: Lots of tears have been shed getting E&MW to work |
| 04:35.20 | jake1932 | need to verify |
| 04:35.43 | slinabery | CunningPike: quite a few of them mine. |
| 04:35.49 | CunningPike | jake1932: I think you'd probably know...... ;) |
| 04:35.57 | CunningPike | slinabery: :) |
| 04:35.58 | jake1932 | it was a while ago |
| 04:37.13 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:39.08 | jake1932 | nope - i'm not using it |
| 04:39.17 | slinabery | :( |
| 04:41.31 | russellb | e&m wink most certainly works on Asterisk. |
| 04:41.45 | russellb | this channel is the cest pool of false rumors |
| 04:42.54 | russellb | and if you *did* have any issues, you should contact digium technical support, who will then resolve your issues, or contact the developers to ensure that your problem is fixed. |
| 04:42.59 | slinabery | russellb: do you understand the meaning of the zapata.conf timing settings (rxwink, rxflash, etc) as relevant to e&m? |
| 04:43.00 | russellb | (assuming you're using Digium hardware) |
| 04:43.33 | slinabery | YES. I am using a TE110P and digium has sent me one response to my trouble ticket. Otherwise SILENCE for like 4 business days. |
| 04:44.12 | jake1932 | 4 business days? - I called in at least 3-4 times and got a response right away |
| 04:44.58 | russellb | slinabery: if you email your ticket number to russell@digium.com, I will forward it over to the right person to ensure it gets followed up on. |
| 04:45.02 | slinabery | well, 3 days. I have not called b/c I'm always trying to work on this during off hours. |
| 04:45.27 | slinabery | russellb: with pleasure. |
| 04:45.44 | russellb | now, the timing paramenters in zapata.conf should not normally have to be messed with |
| 04:46.06 | russellb | unless you're having a specific problem, and you verify the timing used by the other end |
| 04:48.02 | slinabery | for the benefit and entertainment of those on this channel, I will state the problem. * is 'picking up' the incoming call too soon. I have DIDs being sent down the trunk (for example, 531) and * complains that there is no extension 5...it receives the 5 and the 3 as DTMF digits, but thinks the call should go to extn 5. |
| 04:48.44 | justinu|laptop | wow, that sounds a lot like a bug I fixed a while ago |
| 04:48.51 | justinu|laptop | maybe i didn't completely fix it |
| 04:48.55 | slinabery | this is why it seemed like the timing was a logical place to start tweaking, but the provider of the trunk seems incapable of telling me what their timing is. |
| 04:49.09 | yxa | can saydigits and saynumber be "backgrounded" so that users can just press somthing to skip it |
| 04:49.21 | justinu|laptop | slinabery: is it always the 5 digit? |
| 04:49.30 | justinu|laptop | or does it occasionaly grab one of the others, or none at all |
| 04:49.45 | russellb | yxa: I don't think so ... unless you built it manually using the Backgroup app |
| 04:49.48 | slinabery | occasionally (on other nights) it has picked up the first two. |
| 04:49.54 | russellb | of course, always check "show application saynumber" etc. |
| 04:50.14 | yxa | russellb bummer. how about using Read()? |
| 04:50.14 | justinu|laptop | slinabery: let me dig up something from the digium bug tracker for ya |
| 04:50.24 | justinu|laptop | slinabery: because this sounds way too familiar |
| 04:50.27 | russellb | slinabery: it's possible that it is as simple as a dialplan problem |
| 04:50.50 | russellb | that's what it sounds like at first to me ... |
| 04:52.10 | jake1932 | this may sound a bit obvious, but couldn't you pick up immediately and do a Read then a Goto the exten? |
| 04:52.14 | slinabery | also, I can't get the svn source for *,libpri,zaptel to compile on centos. e.g. for zaptel's 'make install' I get: make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list->next != 0' failed. |
| 04:52.20 | slinabery | am I missing some dependency? |
| 04:52.34 | russellb | slinabery: upgrade to make 3.81 |
| 04:52.35 | justinu|laptop | slinabery: http://bugs.digium.com/view.php?id=6364 |
| 04:52.48 | slinabery | russellb: sweet. thank you. |
| 04:53.02 | justinu|laptop | granted, it may have nothing to do with that at all |
| 04:53.09 | justinu|laptop | maybe it's just a coincidence |
| 04:53.54 | slinabery | Well, it's a start. I'll upgrade make and try the latest and greatest from svn. cheers. |
| 04:54.05 | russellb | cool. |
| 04:54.16 | russellb | slinabery: feel free to email me if you have further issues with tech support. |
| 04:54.42 | slinabery | many thanks. |
| 04:55.04 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 04:55.40 | russellb | i try to take care of our users as much as i can ... especially users that help buy me food :) |
| 04:56.51 | jbroome | :) |
| 04:57.27 | slinabery | I feel the same way about my users. Although they all but one forgot system administrators' day. sigh. |
| 04:57.35 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 05:00.57 | designdream | slinabery: when was that? |
| 05:01.25 | designdream | if it makes you feel better, ive never been acknowledged on sysadmin day =( |
| 05:01.49 | slinabery | yeah, I was kind of impressed that one non-sysadmin knew about it. |
| 05:02.18 | slinabery | I decided not to steal his food when I'm here in the middle of the night in recognition of that. |
| 05:02.52 | *** join/#asterisk daysmen3 (n=primus@host86-139-112-17.range86-139.btcentralplus.com) |
| 05:03.02 | Snake-Eyes | Can asterisk play a recording to some one who doesnt have a account with asterisk? |
| 05:03.25 | justinu|laptop | sure can |
| 05:09.41 | designdream | i want laysic |
| 05:09.53 | designdream | laysik? |
| 05:10.53 | CunningPike | designdream: Don't think chan_laysik has been finished yet |
| 05:11.22 | designdream | heh.. i realized i wrote in the wrong chan.. |
| 05:11.29 | CunningPike | :D |
| 05:11.44 | designdream | i was hoping nobody would notice... and s/laysik/lasik/ |
| 05:11.54 | CunningPike | ;) |
| 05:12.03 | designdream | i should have come up with some asterisk tie |
| 05:12.14 | designdream | .. uhm.. i went blind reading the asterisk man ag |
| 05:12.21 | designdream | s/ag/page/ |
| 05:12.33 | *** join/#asterisk SwK (n=Silik0nJ@c-24-99-246-180.hsd1.ga.comcast.net) |
| 05:12.46 | CunningPike | Well, if that's what you kids are calling it these days..... |
| 05:13.14 | *** part/#asterisk anto9us (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 05:13.30 | designdream | ... 'kids'.. eh? |
| 05:13.34 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 05:13.42 | russellb | i'm a kid |
| 05:14.15 | designdream | russellb: so you are on mypace> |
| 05:14.16 | *** join/#asterisk linagee (n=linagee@cpe-70-95-247-242.san.res.rr.com) |
| 05:14.25 | russellb | yes, i am |
| 05:14.27 | russellb | :) |
| 05:14.28 | justinu|laptop | haha |
| 05:14.30 | linagee | is it a bad thing that voip is insecure on the intarweb? |
| 05:14.42 | file | myspace is eeeeevil |
| 05:14.51 | russellb | file is on myspace, too. |
| 05:15.09 | designdream | lol... i have a funny myspace shirt |
| 05:15.11 | file | I am! |
| 05:15.24 | file | and with a picture that russellb took of me |
| 05:15.37 | russellb | file: sounds kinky |
| 05:16.06 | slinabery | not as kinky as it looks |
| 05:16.10 | designdream | http://badmouthtees.com/images/Shirt-MyPlace_240.gif |
| 05:16.31 | designdream | i am not hard to find on myspace |
| 05:16.35 | designdream | <----clue |
| 05:17.51 | russellb | are you in the asterisk myspace group?! |
| 05:18.08 | EyeCue | omfg, lame. |
| 05:18.15 | russellb | lol |
| 05:18.33 | designdream | hahaha ill join the group |
| 05:18.42 | russellb | designdream: i think your myspace page broke firefox |
| 05:18.45 | russellb | THANKS |
| 05:18.53 | EyeCue | tell me you've got your MOH music streaming for teh world to hear? :) |
| 05:19.11 | designdream | russellb: lol... works in 1.6+ |
| 05:19.15 | designdream | 1.5 |
| 05:19.50 | designdream | russellb: is there really an asterisk group? |
| 05:19.50 | russellb | i don't really use it ... girlfriend talked me into signing up one day :-p |
| 05:19.55 | russellb | designdream: i think so ... |
| 05:19.58 | EyeCue | o_O |
| 05:20.04 | EyeCue | i thought you were all taking the piss |
| 05:20.08 | EyeCue | that really is lame :) |
| 05:20.23 | designdream | russellb: my gf had to convince me to not use it... and we have gotten into numerous arguments over my ussage |
| 05:20.30 | EyeCue | didnt think any of you would be seen dead mingling with myspace :] |
| 05:20.42 | designdream | EyeCue: programmers have to get laid too! |
| 05:20.48 | EyeCue | *cough* |
| 05:20.53 | EyeCue | laid via myspace, claims to fame. |
| 05:20.56 | file | I blame brookshire for my myspace account |
| 05:20.57 | EyeCue | </cheering> |
| 05:21.18 | EyeCue | honestly, id take ms palmer over myspace any day of the week |
| 05:21.32 | CunningPike | linagee: Nothing is secure on the Internet |
| 05:21.38 | justinu|laptop | eyecue: we're all thinking it, you're saying it |
| 05:21.41 | designdream | EyeCue: to each their own.. i prefer the short visiting shallow skanks |
| 05:21.56 | EyeCue | you can 'do' short visiting shallow skanks in the flesh. |
| 05:22.09 | Lyfe | MS Vista's gonna be an ugly security problem in the internet from a podcast i'm listening to so you're ok with bad security right now ;) |
| 05:22.16 | designdream | EyeCue: they write on myspace.. then visit in flesh |
| 05:22.24 | EyeCue | class++; |
| 05:22.28 | russellb | file: i see that you're logged in to myspace! |
| 05:22.37 | file | russellb: I am! |
| 05:23.04 | *** join/#asterisk s0lid (n=jlq@ded-153-4.eglobalreach.net) |
| 05:23.11 | designdream | EyeCue: i live in a seasonal tourist spot that is dead in the winters.. myspace imports help locals make it through the off season |
| 05:23.28 | EyeCue | nothing but excuses, you have skanks in your town. |
| 05:23.35 | designdream | EyeCue: none.. |
| 05:23.39 | EyeCue | lies. |
| 05:23.47 | designdream | EyeCue: popullation of 2000 with majority retired |
| 05:23.51 | EyeCue | females love the cock, regardless of how remote you are |
| 05:23.55 | designdream | EyeCue: i live on an island with old people in the winter |
| 05:23.56 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 05:23.58 | EyeCue | noone said anything about age, did they? :) |
| 05:24.03 | EyeCue | o_O |
| 05:24.23 | justinu|laptop | what island? |
| 05:24.36 | designdream | eeek... i prefer myspace skanks to removable teeth grannies |
| 05:24.44 | designdream | south padre island |
| 05:24.50 | EyeCue | dont knck it. |
| 05:24.54 | justinu|laptop | not sure where that is |
| 05:25.02 | EyeCue | alternatively, move. |
| 05:25.05 | designdream | sopadre.com |
| 05:25.35 | file | SARAH! I KNOW HER! HA |
| 05:25.35 | designdream | EyeCue: why? the females from myspace all come down thinking the island is packed all year round |
| 05:25.46 | *** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
| 05:25.50 | justinu|laptop | ah, texas |
| 05:25.51 | EyeCue | then you pack them, right. |
| 05:25.52 | EyeCue | :) |
| 05:26.38 | designdream | so far yes.. very high conversion rate |
| 05:26.48 | *** part/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
| 05:26.55 | EyeCue | *claps* now to move up to picking up females in the flesh |
| 05:26.59 | EyeCue | and youre set |
| 05:27.08 | justinu|laptop | jesus weeps |
| 05:27.16 | EyeCue | as do the kittens |
| 05:27.25 | *** join/#asterisk MstlyHrmls (n=mh@66.195.193.151) |
| 05:28.27 | designdream | EyeCue: flesh takehomes are available all summer... |
| 05:28.38 | EyeCue | k |
| 05:28.39 | EyeCue | :) |
| 05:29.08 | designdream | russellb: lol! i see you =P |
| 05:29.41 | russellb | designdream: hehe |
| 05:29.46 | designdream | cs major.. rads.. digium! that explains the @ |
| 05:30.03 | russellb | rads? |
| 05:30.13 | designdream | i used to go to clemson as a child to watch the tigers play |
| 05:30.17 | slinabery | ahem. anyone know why I get this when compiling zaptel from svn source: |
| 05:30.20 | slinabery | zttranscode.c: In function `zt_tc_mmap': |
| 05:30.20 | slinabery | zttranscode.c:379: warning: passing arg 1 of `remap_page_range_Rsmp_3dd67602' makes pointer from integer without a cast |
| 05:30.20 | slinabery | zttranscode.c:379: incompatible type for argument 4 of `remap_page_range_Rsmp_3dd67602' |
| 05:30.20 | slinabery | zttranscode.c:379: too few arguments to function `remap_page_range_Rsmp_3dd67602' |
| 05:30.28 | file | ughr |
| 05:30.49 | russellb | designdream: ah, cool stuff. i'll graduate in December. Then, I'll be working for Digium full time |
| 05:30.58 | russellb | I've been working on Asterisk for a couple of years, though |
| 05:31.47 | designdream | russellb: awesome... i just started fiddling with it.. i have about a week to order and start converging our office |
| 05:31.50 | file | slinabery: what kernel? |
| 05:32.00 | slinabery | 2.4.21-40.ELsmp |
| 05:32.56 | russellb | that should be fixed in the latest code in the 1.2 branch |
| 05:33.08 | designdream | <PROTECTED> |
| 05:33.18 | russellb | designdream: good luck |
| 05:33.32 | file | russellb: zttranscode doesn't exist in 1.2 does it? or are you talking about something else? |
| 05:33.42 | file | my mental state is degrading afst |
| 05:33.46 | russellb | mine too |
| 05:33.56 | russellb | i don't really know ... i still think it has been fixed |
| 05:34.13 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 05:34.14 | designdreamAFk | (degrading began when visiting myspace) |
| 05:34.22 | file | the API call changes between kernel versions :( |
| 05:34.28 | russellb | 2.4.21? seriously ... update :-p |
| 05:34.32 | file | designdreamAFk: ha, long before then |
| 05:34.50 | slinabery | this is from a fresh checkout of the zaptel source |
| 05:35.07 | designdreamAFk | vimdiff brain.dump brain_after_myspace.dump |
| 05:35.38 | Juggie | i bet hes running RHEL3 |
| 05:35.52 | slinabery | yeah, I know. I was too lazy to download all 4 ISOs of the CentOS with 2.6 kernel. |
| 05:36.12 | Juggie | slinabery, you dont need all 4 isos |
| 05:36.17 | Juggie | use the server iso |
| 05:36.19 | Juggie | its one cd. |
| 05:36.25 | slinabery | There's nothing in the asterisk docs (afaik) that says yeh can't use rhel3. |
| 05:36.29 | russellb | i wish we could stop supporting 2.4 in zaptel to be honest |
| 05:36.43 | russellb | it's such a friggin pain |
| 05:36.45 | slinabery | I needed it for x86_64. servercd not available. |
| 05:36.48 | EyeCue | i wish we could stop supporting windows9x in miranda. |
| 05:36.49 | EyeCue | :| |
| 05:36.52 | Juggie | slinabery, yes it is. |
| 05:37.05 | slinabery | wasn't when I was on the mirrors. (few weeks ago) |
| 05:37.13 | designdreamAFk | EyeCue: you contribute to miranda? |
| 05:37.14 | slinabery | I believe you, just saying I didn't see it |
| 05:37.23 | EyeCue | designdreamAFk, project manage it |
| 05:37.34 | russellb | EyeCue: cool |
| 05:37.41 | EyeCue | let me clarify that, recently took up project management of it :) |
| 05:37.45 | designdream | i love miranda and swear by it... (just forced all company to dump gaim for it) |
| 05:37.49 | EyeCue | *refuses to take the blame for current state* |
| 05:37.49 | russellb | EyeCue: we've managed to avoid that so far with Asterisk :-p |
| 05:38.07 | EyeCue | designdream, its you guys im moving the project forward for. |
| 05:38.17 | russellb | though it often crosses my mind to start working on a windows port ... |
| 05:38.21 | designdream | EyeCue: awesome! i was fearing its death |
| 05:38.22 | EyeCue | well, 'with a view to' |
| 05:38.31 | file | russellb: it's late, you don't know what you're talking about |
| 05:38.38 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:38.39 | EyeCue | not dying anytime soon, im in cleanup/document/organise mode at the moment |
| 05:38.39 | designdream | russellb: stop talking like that |
| 05:38.43 | russellb | :( |
| 05:38.46 | Lyfe | windows, what's that? |
| 05:38.54 | file | russellb: I bet you could enhance it with OpenGL though! |
| 05:38.57 | russellb | sorry, i'll keep my dirty thoughts to myself |
| 05:39.00 | russellb | file: indeed |
| 05:39.06 | file | and... a rotating cube |
| 05:39.16 | russellb | i started working on an astman like app in OpenGL one day, but couldn't decide what to make it do |
| 05:39.16 | designdream | Lyfe: our parents worked hard to give us a brighter future.. hopefully we will never have to hear about windows |
| 05:39.44 | jets | much like that linux doom mod to kill processes and see server load using doom --- |
| 05:39.58 | jets | call volume and disconnect calls using a first person shooter |
| 05:39.58 | russellb | jets: yeah |
| 05:40.01 | Lyfe | no, i'm perfectly fine with there being a bunch of dumb people using windows.. it gives me some mundane work to do, instead of the interesting stuff, like messing with asterisk to see what kinda crazy things it can do :) |
| 05:40.02 | designdream | now that sounds rad |
| 05:40.18 | Lyfe | hehe. |
| 05:40.26 | Juggie | russellb, isnt it already ported, kinda? |
| 05:40.29 | Lyfe | a doom call management system.. :P |
| 05:40.32 | russellb | Juggie: um, no. |
| 05:40.48 | file | well, I go nini |
| 05:40.49 | file | bye all |
| 05:40.55 | designdream | i heard of a drug ring using asterisk for inventory and distribution IVR |
| 05:41.12 | Lyfe | heh. |
| 05:41.21 | russellb | i wonder if they'd give me a discount |
| 05:41.23 | Lyfe | hmm.. use a bfg, kill a queue. :) |
| 05:41.39 | *** join/#asterisk fafnir (i=hahaha@unaffiliated/fafnir) |
| 05:41.44 | Juggie | russellb, did you mean asterisk windows port or gaim? |
| 05:41.55 | russellb | asterisk .. |
| 05:42.07 | Juggie | theres a port which depends on cygwin |
| 05:42.08 | EyeCue | i say dont diversify and dillute your base. |
| 05:42.18 | designdream | russellb: i am sure they would |
| 05:42.23 | russellb | Juggie: not after the rewrite of the build system in trunk, heh |
| 05:42.35 | slinabery | Juggie: to clarify, there's no torrent for the centos 4 server cd. |
| 05:42.38 | Juggie | no, its old, its only 1.0.x ported. |
| 05:42.47 | Juggie | slinabery, your right i thought i had a x64 server cd but i dont. |
| 05:42.50 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 05:42.59 | Juggie | i just used the dvd i guess. |
| 05:43.03 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 05:43.13 | EyeCue | every soog service that has a windows port gives OSS consultants like me less incentive, and their clients less reason to choose superior platforms |
| 05:43.17 | EyeCue | soog = good :| |
| 05:43.58 | Juggie | slinabery, http://mirrors.kernel.org/centos/4.3/isos-dvd/x86_64/CentOS-4.3-x86_64-binDVD.iso |
| 05:44.24 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 05:44.41 | jets | haha shoot the queue in to a portal to another queue |
| 05:44.48 | jets | HAHA or capture the queue |
| 05:44.54 | jets | capthre queue and drop it off at another queue |
| 05:44.56 | designdream | i really need to crack this book so i dont get any RTFM crap when i ask questions on friday... ttyl |
| 05:44.57 | Lyfe | hehehe. |
| 05:45.06 | Juggie | ~RTFM |
| 05:45.08 | jbot | rtfm is, like, Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 05:45.10 | Snake-Eyes | justinu|laptop, sry got pulled away, where abouts is this set (can you give me a pointer :) ) |
| 05:45.19 | Lyfe | multiplayer doom call management. :) |
| 05:45.26 | EyeCue | what inforbot is jbot running ? |
| 05:45.31 | EyeCue | jbot, infobot guide? |
| 05:45.33 | jbot | [infobot guide] http://www.cs.cmu.edu/~infobot/infobot_guide.html, or at http://www.avians.net/irc/infobot_guide.html |
| 05:45.47 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 05:45.49 | EyeCue | ack, same broken gay one im using. |
| 05:45.57 | Lyfe | heh |
| 05:46.09 | Lyfe | broken, eh? |
| 05:46.13 | EyeCue | well, not so good |
| 05:46.20 | EyeCue | you know what the search command is? :) |
| 05:46.31 | EyeCue | broken = old/not updated/many forks of |
| 05:46.32 | Lyfe | for the infobot? no idea. |
| 05:46.46 | EyeCue | i went through source, changelog says theres a command in there somewhere |
| 05:46.52 | EyeCue | didnt find diddly. |
| 05:47.23 | Lyfe | uncommented, maybe? |
| 05:47.31 | EyeCue | dont think so |
| 05:47.32 | Lyfe | and very very ugly? |
| 05:47.39 | EyeCue | something like that :) |
| 05:47.46 | EyeCue | and the new dev sources dont have irc libraries :| |
| 05:47.57 | EyeCue | i was like, wtf. |
| 05:47.59 | Lyfe | heh.. sounds terribly useful. |
| 05:48.08 | EyeCue | pointless++ |
| 05:48.20 | EyeCue | finding tcl inforbot scripts is just as fruitless |
| 05:48.20 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 05:48.23 | EyeCue | -r |
| 05:48.33 | EyeCue | although would be my ideal solution. |
| 05:48.34 | Lyfe | gah, misquito bites suck. |
| 05:48.58 | Qwell | Kerry_G: YOU! |
| 05:49.02 | EyeCue | i might bf2 |
| 05:49.02 | Lyfe | hmm.. there's no good tcl scripts for eggdrops for it or anything? |
| 05:49.02 | Kerry_G | yo yo |
| 05:49.04 | EyeCue | *ponders* |
| 05:49.16 | EyeCue | Lyfe, none that i could find updated after 2002. |
| 05:49.20 | Qwell | Kerry_G: A couple of us watched your presentation the other day... |
| 05:49.25 | Kerry_G | get a good laugh? |
| 05:49.35 | EyeCue | that werent forks of 1287 other infobots. |
| 05:49.40 | Qwell | well, there were a couple..."issues" |
| 05:49.43 | Lyfe | ahh.. right, cause people started to stop using eggdrops with the migration of users to other irc networks. |
| 05:49.50 | RTFAsteriskbook | so wait.. what infobot should i start with? |
| 05:49.51 | Kerry_G | always is when you are thinking on your feet |
| 05:49.55 | Kerry_G | and winging it |
| 05:49.56 | EyeCue | i dunno :) |
| 05:50.00 | Qwell | Kerry_G: like incorrect information :) |
| 05:50.09 | Kerry_G | did I foobar something bad? |
| 05:50.19 | Qwell | a few things... I don't recall anything specific though |
| 05:50.25 | RTFAsteriskbook | i think ill just figure out how to use infobot factpacks in my php bot |
| 05:50.34 | EyeCue | pee hayche pee bot? |
| 05:50.40 | EyeCue | tell me it can be daemonized? |
| 05:50.44 | Kerry_G | I am waiting to see the video of it myself |
| 05:50.53 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 05:51.08 | Lyfe | perhaps there's a python infobot available (given that the Twisted API for python has irc related stuff in it) |
| 05:51.25 | EyeCue | mozilla has one thats base don python |
| 05:51.31 | EyeCue | but i was like meh, give me a c one. |
| 05:51.36 | Lyfe | ahh, gotcha. |
| 05:51.45 | RTFAsteriskbook | EyeCue: i leave it running all the time |
| 05:51.52 | EyeCue | got url? |
| 05:51.58 | Kerry_G | I know someone tool offense to me saying the TDM2400 was about a $2000 carrd |
| 05:52.00 | EyeCue | how goods its lexical parsing? :) |
| 05:52.07 | Lyfe | Twisted API was decent. but i can see why you'd want a C one versus something in python. |
| 05:52.21 | RTFAsteriskbook | EyeCue: not sure where i got it.. but i ended changing a log of it.. and making it so i can develop extensions without having to restart it |
| 05:52.22 | Qwell | Kerry_G: I was that tool. :) And it was the te4xx |
| 05:52.31 | Qwell | (yes, I realize you meant took) |
| 05:52.33 | EyeCue | would be nice to just have one. |
| 05:52.35 | RTFAsteriskbook | EyeCue: it even checks the extensions for syntax errors before running them |
| 05:52.41 | EyeCue | even a freakin proper ai bot that learnt stuff |
| 05:52.59 | EyeCue | infobots are awefully rigid, wouldnt mind a freeform language processor bot. |
| 05:53.06 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 05:53.10 | RTFAsteriskbook | EyeCue: i thought about injecting the data into one of the artificial neural network toys and teaching it |
| 05:53.15 | EyeCue | yah |
| 05:53.21 | EyeCue | mate of mine did that with his arti daemon |
| 05:53.31 | EyeCue | did really well, after 3 months of learning the basics of language |
| 05:53.33 | Lyfe | eyecue: taken a look at Alice, maybe? It's been so long that i'm not sure hwo the alice project is going, for that matter. |
| 05:53.39 | Qwell | Kerry_G: I was trying to get people to correct you, but there were no takers :p |
| 05:53.41 | Lyfe | least, i think it's called alice. |
| 05:53.42 | EyeCue | *curses* dont say that name ever again |
| 05:53.44 | EyeCue | :| |
| 05:53.49 | *** join/#asterisk sergee (n=opera@195.94.224.197) |
| 05:53.51 | Lyfe | ahaha.. woops :P |
| 05:54.01 | Kerry_G | hell, I appreciate people standing up with a correction, I cant remember everything 100% of the time |
| 05:54.06 | EyeCue | just coz she did ok on turing, doesnt mean dick :D |
| 05:54.12 | Lyfe | i had a feeling the reaction might be like that. |
| 05:54.22 | RTFAsteriskbook | i took a different approach.. i logged and then treaded question/answers |
| 05:54.25 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 05:54.28 | *** join/#asterisk MstlyHrmls (n=mh@66.195.193.151) |
| 05:54.34 | EyeCue | if someone gave me a hot little system in place, id teach it. |
| 05:54.53 | EyeCue | problem is, if you wanna go that far, you can give it hard written rules on language processing either. |
| 05:55.00 | EyeCue | s/can/cant |
| 05:55.04 | RTFAsteriskbook | so if someone wrote ... "what is something?" and someone wrote "name:something is bleh" i would have something for the db |
| 05:55.18 | Kerry_G | I wish I would have been able to see the IRC chat, I would have corrected myself if someone said something |
| 05:55.30 | Qwell | Kerry_G: a couple people were mad about your opinion of polycoms :p |
| 05:55.36 | Kerry_G | ??? |
| 05:55.41 | EyeCue | should have fed irc into a pair of fake sunglasses |
| 05:55.41 | Kerry_G | That I like them? |
| 05:55.44 | EyeCue | just incase |
| 05:55.54 | Qwell | well, there was a part there, where it seemed like you were against them |
| 05:56.04 | EyeCue | "omfg, say you love hash-asterisk" |
| 05:56.25 | Qwell | Kerry_G: all in all, I enjoyed it though... |
| 05:56.37 | Kerry_G | there was a question like did I consider Polycom to have a "cool factor" with regards to Asterisk support to which I was cold to, but I love the phones |
| 05:56.51 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 05:57.17 | Kerry_G | thanks |
| 05:57.26 | Qwell | I think your main point against them, was the sidecar, if I'm not mistaken |
| 05:57.33 | Kerry_G | pretty much |
| 05:57.36 | Qwell | (that bug has been fixed in later firmware) |
| 05:57.48 | Qwell | no more limit |
| 05:58.00 | Kerry_G | I should revisit the sidecar again then, its been a while |
| 05:59.07 | RTFAsteriskbook | sidepoint 430 looks attractive.. |
| 05:59.35 | CunningPike | SoundPoint? :) |
| 05:59.51 | CunningPike | Or is the sidecar called a SidePoint? |
| 06:00.06 | RTFAsteriskbook | sound.. bleh.... you cunningpike! always calling me out |
| 06:00.07 | Kerry_G | arent people have issues with the new phone on Asterisk? |
| 06:00.16 | CunningPike | Hee hee |
| 06:00.40 | Qwell | Kerry_G: I think [TK]D-Fender still heavily recommends them |
| 06:00.48 | RTFAsteriskbook | CunningPike: its almost as if you sit around waiting for me to mispell something... |
| 06:01.00 | CunningPike | RTFAsteriskbook: It's not just you ;) |
| 06:01.31 | Qwell | misspell is generally accepted |
| 06:01.47 | RTFAsteriskbook | CunningPike: 'its not just you.. its me! |
| 06:02.00 | RTFAsteriskbook | great.. i can smell neighbors burning one |
| 06:02.17 | RTFAsteriskbook | maybe the second hand will help me sleep |
| 06:02.47 | RTFAsteriskbook | anyone order from thinkbrightdirect? |
| 06:04.15 | russellb | RTFAsteriskbook: go read |
| 06:05.14 | Qwell | russellb: ! |
| 06:05.27 | russellb | greetings Qwell ! |
| 06:05.28 | Qwell | like my latest patch? :D |
| 06:05.28 | Juggie | oh great, http://blog.tmcnet.com/blog/greg-galitzine/voip/intel-sells-dialogic-to-eicon.html |
| 06:05.41 | Juggie | this otta make ABE w/ dialogic support take even longer :( |
| 06:06.04 | russellb | Qwell: which one |
| 06:06.10 | Qwell | russellb: show translation, heh |
| 06:06.24 | russellb | oh, yes :) |
| 06:06.33 | Qwell | that was a real head scratcher for a bit |
| 06:07.10 | russellb | i think i noticed that the output formatting in that was broken ... |
| 06:07.16 | russellb | but i pretended i didn't see it :-p |
| 06:07.18 | Qwell | heh |
| 06:07.26 | Qwell | I tried pretending, but I could only go so long |
| 06:07.32 | russellb | yup, it was rough |
| 06:07.34 | Qwell | it's been that way for like a month |
| 06:07.40 | russellb | i started to try to fix it ... but only made it worse |
| 06:07.43 | russellb | so i gave up and moved on |
| 06:08.15 | Qwell | I learned something while doing that |
| 06:08.24 | Qwell | (1 << -1) is...valid |
| 06:08.31 | Qwell | ...sorta |
| 06:08.34 | russellb | lol |
| 06:08.45 | russellb | 1 << -1 would be ......... |
| 06:08.47 | russellb | 0? |
| 06:08.57 | Qwell | I think so, yeah |
| 06:09.11 | russellb | silly bit shifting |
| 06:09.11 | Qwell | it's like 1 >> 1, I imagine |
| 06:09.18 | russellb | yup |
| 06:10.16 | Qwell | local news headlines are great "Monkey attack" |
| 06:10.27 | Juggie | what does the >> operator do? |
| 06:10.29 | Juggie | or << |
| 06:10.31 | Qwell | Juggie: bitshift |
| 06:10.35 | Juggie | ah. |
| 06:10.35 | FuriousGeorge | bitshift |
| 06:10.40 | russellb | Juggie: a << b ... shift a by b bits |
| 06:10.42 | *** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
| 06:10.42 | russellb | to the left |
| 06:10.44 | russellb | >> to the right |
| 06:10.53 | Juggie | ahh... i've never done that. |
| 06:11.31 | russellb | it's handy. |
| 06:11.35 | Qwell | quite |
| 06:11.35 | FuriousGeorge | then you can use the & and ^ oprators for and/or by the bit |
| 06:11.39 | Juggie | i would imagine so |
| 06:11.44 | Juggie | yeah, i've used them before |
| 06:12.00 | russellb | sweet :) |
| 06:12.04 | russellb | now start hacking asterisk! |
| 06:12.10 | Juggie | passing in certain flags to a function which takes a bit |
| 06:12.15 | Juggie | but never seen >> << used. |
| 06:12.18 | russellb | i want to see a patch by 8 AM, kthx |
| 06:12.19 | Juggie | i'll have to read up on it. |
| 06:12.31 | FuriousGeorge | russellb: linked lists and stuff was as complex as it got |
| 06:12.44 | Lyfe | lemme help furiousgeorge out.. 8am of what timezone, and what day? :P |
| 06:13.05 | russellb | plenty of "janitor projects" to work on ... |
| 06:13.09 | russellb | linked lists are one of them :) |
| 06:13.43 | Juggie | if that fails, bring russell some coffee & redbull. |
| 06:13.50 | Lyfe | plenty of outside projects to work on too. |
| 06:14.20 | FuriousGeorge | russellb: in all seriousness if asterisk-xmpp doesnt do what i want (still somewhat of a mystery how it will work) iw as thinking of writing something in C so multiple asterisk servers can know about the presence states of their respective peers |
| 06:14.48 | russellb | FuriousGeorge: certainly would be cool ... that's a serious undertaking |
| 06:15.07 | Juggie | FuriousGeorge, in what regard, more details. |
| 06:15.13 | FuriousGeorge | russellb: i was hoping it could be done as a dirty and fugly hack |
| 06:15.22 | russellb | not ... really ... |
| 06:16.06 | Juggie | FuriousGeorge, if you just want to know if a device is registered you can tell that from the realtime db or over asterisk management interface. |
| 06:16.10 | FuriousGeorge | Juggie: i just mean that server a has a few peers, and server b has a few peers, both sets of peers understand presence, but the peers on server a dont know about the presence states of the peers on server b |
| 06:16.21 | FuriousGeorge | and vice-a-versa |
| 06:16.26 | Qwell | FuriousGeorge: no a |
| 06:16.32 | Juggie | you mean more details presence then registered or not i assume |
| 06:16.35 | Juggie | so you can set away flags, etc. |
| 06:17.28 | Juggie | right? |
| 06:18.04 | Qwell | Juggie: skip the coffee, I think |
| 06:18.17 | FuriousGeorge | Juggie: i'd be happy with registered or not and on the phone or not, as long as i can set a device state to make an led work. hence the dirty hack i had in mind. somehow having my clients in thier calling macros update their statuses to one another |
| 06:18.46 | FuriousGeorge | then polling every minute to see whos signed off/on |
| 06:18.58 | FuriousGeorge | and once again setting device states |
| 06:20.48 | Juggie | FuriousGeorge, you can tell a devices state from realtime |
| 06:20.53 | Juggie | i dunno if that helps. |
| 06:21.04 | Juggie | you could extend realtime to provide more presence info if you wanted |
| 06:21.17 | muppetmaster | Hello |
| 06:21.30 | muppetmaster | Under CVS HEAD, what has happened with 'asterisk-addons' as it seems to be the full Asterisk codebase? |
| 06:21.46 | FuriousGeorge | Juggie: well first im gonna wait till 1.4, since i also run a xmpp server, maybe asterisk xmpp will scratch my itch.... but i doubt it |
| 06:21.57 | Juggie | FuriousGeorge, in 1.4 there is also SNMP |
| 06:22.03 | Juggie | that may also solve your problem to an extent. |
| 06:22.25 | FuriousGeorge | snmp? isnt that like gkrellmd? |
| 06:22.26 | Juggie | but you can if you like share a sip buddies database across multiple servers |
| 06:23.50 | muppetmaster | Also, I am getting a ztdummy compile error on OpenSuSE v10.0: http://pastebin.ca/134019 |
| 06:24.03 | muppetmaster | CVS HEAD as of about 2 mins ago |
| 06:24.18 | Juggie | thats pretty impressive considering * uses SVN now :) |
| 06:24.19 | delmar | So i have a new Linksys router here which has the usual QoS facilities, but on the main setup page there is some new stuff I have not heard of.. anyone know what the story is with all this CBR, UBR, VBR stuff.. and the Pcr/Scr ? |
| 06:24.51 | muppetmaster | Juggie, good point, old habit, I meant SVN |
| 06:24.58 | muppetmaster | TRUNK |
| 06:25.14 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 06:25.27 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 06:26.12 | mitcheloc | FuriousGeorge: nice idea for a project ;) |
| 06:26.15 | Qwell | mitcheloc: #asterisk-dev |
| 06:26.53 | muppetmaster | So, anyone have any ideas on the ztdummy compile error? |
| 06:27.10 | muppetmaster | I have yet to get Zaptel SVN trunk to compile on OpenSuSE |
| 06:27.11 | mitcheloc | Qwell: did you get my # last time or did I need to pm it to you again? |
| 06:27.18 | Qwell | It's in my logs |
| 06:27.39 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 06:27.51 | Qwell | muppetmaster: taking forever to load.. can you try the .com? |
| 06:28.01 | muppetmaster | Sure, just a moment |
| 06:28.21 | delmar | muppetmaster, why use the SVN? why not just use the latest tarball release. |
| 06:28.35 | muppetmaster | delmar Because I want to play with all the new stuff in my sandbox |
| 06:32.36 | muppetmaster | Qwell pastebin.com seems even slower, still waiting for it to return |
| 06:32.57 | Qwell | uuoc.com |
| 06:33.33 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:34.10 | muppetmaster | Qwell http://uuoc.com/1570 |
| 06:34.58 | *** join/#asterisk apardo (n=apardo@87.217.146.143) |
| 06:35.49 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 06:36.22 | Qwell | muppetmaster: looks like you're missing something in your kernel |
| 06:36.41 | muppetmaster | Qwell Hmmmm.... It is the standard OpenSuSE kernel, I have not recompiled it or anything like that |
| 06:36.46 | muppetmaster | Do I need to add something and recompile? |
| 06:39.23 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 06:39.43 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 06:43.48 | muppetmaster | Qwell, if I take out ztdummy using menuselect it all compiles fine |
| 06:43.58 | muppetmaster | So something with ztdummy and the standard OpenSuSE kernel |
| 07:00.00 | benjk | Suse is no longer supported by Asterisk |
| 07:01.20 | benjk | With SUSE 10 Novell introduced a new policy by which they want to discourage people from building their own kernel modules |
| 07:01.58 | benjk | consequently a SUSE installation no longer builds kernel modules without putting in some work, which is rather cumbersome |
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| 07:14.24 | muppetmaster | Benjk Very interesting |
| 07:14.29 | muppetmaster | What distro would you recommend? |
| 07:15.29 | *** join/#asterisk vlt (n=dm@p54B33C9D.dip0.t-ipconnect.de) |
| 07:15.38 | benjk | I got it working on SUSE 10 once, but it took me 2 days so I thought I better ditch it and look for some other distro, I tried out a few and settled for Ubuntu Server |
| 07:15.44 | benjk | very pleased with it |
| 07:16.34 | vlt | OT: Hello. Is there a logfile for this irc channel somewhere? |
| 07:16.59 | muppetmaster | Thanks for the info, time to change |
| 07:23.23 | *** join/#asterisk bionoid (i=bionoid@down.in.fragglerock.org) |
| 07:25.01 | *** join/#asterisk adorah (n=Administ@87.68.170.146.cable.012.net.il) |
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| 07:27.13 | bionoid | <PROTECTED> |
| 07:27.16 | bionoid | oops |
| 07:30.44 | nettie | Hi guys, what's the best way to handle call termination for an IVR system using asterisk please? I'm actually doing: ringing, wait, answer, then playing using Backgroud and finally WaitExten. I then put the possible choices and relative Goto jumps. Where should I put Hangup ? maybe the best way is definte a ResponseTimeout and then hangup or play again? |
| 07:32.44 | *** join/#asterisk Assid (i=assid@203.115.83.213) |
| 07:33.36 | bionoid | Hello ;) I swapped a wcfxo for a TDM400P and I'm experiencing signifficant reduction in sound quality. Echo is now acceptable (after fxotune), but the sound is somewhat choppy (both for caller and callee), and occasionally it breaks havoc; by that I mean _loud_ static for 2 - 10 seconds before resuming normal operation (still choppy). Any pointers? |
| 07:34.56 | *** join/#asterisk Jeffjohnson (n=Jeffjohn@unaffiliated/jeffjohnson) |
| 07:34.57 | Jeffjohnson | hello |
| 07:35.20 | Dovid | nettie: what is ur exact question ? where do u put the hangup ? |
| 07:35.25 | Dovid | hello Jeffjohnson: |
| 07:35.36 | Jeffjohnson | I've try to register to an iax2 provider... If i try to call somebody, I will get the message "Aug 16 09:35:30 WARNING[25387]: chan_iax2.c:7075 socket_read: Call rejected by 83.125.8.46: No authority found" what it means? |
| 07:35.36 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 07:35.38 | Dovid | Jeffjohnson: hello os more like it |
| 07:35.46 | Jeffjohnson | Dovid: hum? what? |
| 07:36.04 | Dovid | Jeffjohnson: means that the remote IAX server dosent like you and wont let the call thru |
| 07:36.14 | Dovid | meaning that there is something that is telling it not take the call |
| 07:36.18 | Jeffjohnson | Dovid: why he don't like me? :o |
| 07:36.23 | Dovid | usually bad user id and or pass |
| 07:36.33 | Jeffjohnson | Dovid: the registration was successfull |
| 07:36.50 | Jeffjohnson | Dovid: " Registered IAX2 to '83.125.8.46', who sees us as 62.109.80.42:4569 with no messages waiting" |
| 07:37.03 | Jeffjohnson | so what can be the reason? |
| 07:37.08 | Dovid | how u send the call |
| 07:37.17 | Dovid | whats in extensions.conf ? |
| 07:37.40 | *** join/#asterisk Assid (i=assid@203.115.83.213) |
| 07:38.13 | Jeffjohnson | Dovid: exten => $PHONENR,2,Dial(IAX2/${EXTEN}@dusnetiax,30,tr) |
| 07:39.05 | Dovid | ok normally u have IAX2/userid@carrier/${EXTEN} |
| 07:40.24 | Dovid | what do they have on thier set up page ? |
| 07:40.40 | Jeffjohnson | Dovid: which userid? the usename from "iax2 show users"? or the "authen"? |
| 07:40.50 | nettie | Dovid Hi, sorry for the late answer but I was reading stuff related to the question in bq |
| 07:41.02 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 07:41.03 | Dovid | in iax.conf u have the user id then : followed by the pass |
| 07:41.07 | Dovid | u need to put there the user id |
| 07:41.10 | Dovid | i will show u what i have |
| 07:41.13 | nettie | Dovid I just want to be sure that after some time of inactivity in the IVR menu asterisk hangups the call |
| 07:41.20 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 07:41.26 | Jeffjohnson | Dovid: i have it in iax conf |
| 07:41.29 | Dovid | nettie: any reason y |
| 07:41.31 | Dovid | ? |
| 07:41.43 | Dovid | Jeffjohnson: are you asking ? |
| 07:42.06 | Jeffjohnson | no |
| 07:42.07 | Winkie | anyone got any experience with chan_agent events and why the hell it renames Local/ channels to Sip/ channels? |
| 07:42.12 | nettie | Dovid uhmm right, I dont have one considering my incoming channels are unlimited .. |
| 07:42.14 | Winkie | this is remarkably broken behaviour :( |
| 07:42.20 | nettie | Dovid and we're not paying the call :) |
| 07:43.19 | Dovid | nettie: you can have a time out so fi they dont press anything after say 10 seconds it dumps the call on them or repeat the menu once and then if they enter nothing after the second reapeat then it should dump the call |
| 07:43.27 | Jeffjohnson | Dovid: thank you very much, that works :) |
| 07:43.49 | Dovid | let me know if it works |
| 07:43.52 | nettie | Dovid yes, this looks like what I wanted |
| 07:44.00 | Jeffjohnson | but now I have another problem.. I've test it with a call to my mobile. Is mobile powered off i don't hear a busy sound or anythin, I will still hear the normal dial sound |
| 07:44.37 | Dovid | Jeffjohnson: what do u mean by normal dial sound |
| 07:44.42 | Dovid | ? |
| 07:45.16 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 07:45.59 | nettie | Dovid how do I discriminate the 2 C:\Axis\250s\3_20>ftp <ip of camera> |
| 07:45.59 | nettie | Connected to <ip of camera>. |
| 07:45.59 | nettie | 220 AXIS 250S MPEG-2 Video Server 3.20 <Sept 20 2004> ready. |
| 07:45.59 | nettie | User (<ip of camera>:(none)): root |
| 07:45.59 | nettie | 331 User name okay, need password. |
| 07:46.00 | nettie | Password: pass |
| 07:46.02 | nettie | 230 User logged in, proceed. |
| 07:46.04 | nettie | ftp> bin |
| 07:46.06 | nettie | 200 Command okay. |
| 07:46.08 | nettie | ftp> hash |
| 07:46.09 | Jeffjohnson | Dovid: the dial sound before someone answer the phone |
| 07:46.10 | nettie | Hash mark printing On ftp: (2048 bytes/hash mark) . |
| 07:46.12 | nettie | ftp> put axis250s.bin flash |
| 07:46.14 | nettie | 200 Command okay. |
| 07:46.16 | nettie | 150-Preparing to flash. |
| 07:46.18 | nettie | Allocating memory. |
| 07:46.20 | nettie | 150 Opening data connection. |
| 07:46.22 | nettie | ################################################################### |
| 07:46.26 | nettie | ################################################################### |
| 07:46.28 | nettie | 226-Transfer complete. Checksum and HWID verified. |
| 07:46.30 | nettie | Erasing flash... |
| 07:46.32 | nettie | Erasing /dev/cflash1... |
| 07:46.34 | nettie | 1% erased |
| 07:46.36 | nettie | . |
| 07:46.38 | nettie | . |
| 07:46.40 | nettie | . |
| 07:46.42 | nettie | 100% erased |
| 07:46.44 | nettie | Programming /dev/cflash1... |
| 07:46.46 | nettie | 1% written |
| 07:46.48 | nettie | . |
| 07:46.48 | mitcheloc | nettie: stop! |
| 07:46.49 | kaldemar | what are you doing? |
| 07:46.50 | nettie | . |
| 07:46.50 | mitcheloc | STOP! |
| 07:46.51 | Dovid | nettie: please stop |
| 07:46.52 | nettie | . |
| 07:46.54 | nettie | 100% written |
| 07:46.55 | *** kick/#asterisk [nettie!i=denon@synapse.subneural.net] by denon (Please use a pastebin) |
| 07:47.11 | denon | something tells me it wasnt intentional heh |
| 07:47.14 | *** join/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 07:47.22 | Dovid | Jeffjohnson: so u make a call and u hear a ringing sound or a dial tone ? |
| 07:47.26 | nettie | I'm sorry guys |
| 07:47.34 | Jeffjohnson | Dovid: yes |
| 07:47.34 | nettie | it was in my clipboard |
| 07:47.40 | nettie | it wasnt even related to asterisk |
| 07:47.45 | nettie | I snapped it for error |
| 07:47.47 | denon | ctrl+f4 is your friend |
| 07:48.19 | Dovid | Jeffjohson: what kind of line ? IAX or POTS ? |
| 07:49.15 | Jeffjohnson | Dovid: mmh, wait I see the call don't go over my iax line, like it should it take my SIP line now :o "Aug 16 09:48:44 NOTICE[25387]: chan_iax2.c:2860 auto_congest: Auto-congesting call due to slow response" |
| 07:50.30 | nettie | Dovid, how do I discriminate the 2 ResponseTimeout? |
| 07:50.34 | Jeffjohnson | Dovid: now it takes my iax2 line :) |
| 07:50.57 | Dovid | r u using real time ? |
| 07:51.14 | Jeffjohnson | Dovid: iax line, still no busy/congestion sound if my mobile is turned off |
| 07:51.17 | Dovid | nettie: what do u mean by response time out ? |
| 07:51.23 | Dovid | and if it is on ? |
| 07:52.07 | *** join/#asterisk j0 (n=dan@CABLE-72-53-45-212.cia.com) |
| 07:53.28 | Jeffjohnson | Dovid: do I need another app in my dial plan except dial? To get an busy/congestion sound if the phone isn't available |
| 07:54.01 | nettie | Dovid we said we wanted to have the menu played once, then wait some seconds for a choice, then if no choise are made within that time play the message again and the if no chioces are made a at all hangup. Maybe instead of looping back I just need to play the welcome message again.. |
| 07:54.14 | Dovid | Jeffjohnson: u want to play a busy signal if the call is rejected ? |
| 07:54.37 | Dovid | nettie: you can have the time out go to the begining of the menu |
| 07:54.47 | Dovid | do u know the t extension ? |
| 07:55.06 | Jeffjohnson | Dovid: mmh, I don't want to hear a dial tone... if the mobile phone is powered off and unavailable |
| 07:55.09 | nettie | Dovid sure, the problem is I have 2 different timeouts |
| 07:55.35 | nettie | Dovid one which will loop to the start and one that will hangup |
| 07:55.35 | Dovid | Jessjohnson: if ur phone is on then it goes thru ? u only hear a dial tone if its off ? |
| 07:55.58 | Dovid | ok nettie: do something like this |
| 07:56.30 | Dovid | in the begining of the dial plan have asterisk look to see the value of a variable say callsata |
| 07:56.33 | Jeffjohnson | docelmo: yes i hear the same dial tone if the phone is off/on the whole time |
| 07:56.34 | Dovid | callstat* |
| 07:57.00 | nettie | OK, if/else statement then.. |
| 07:57.12 | Dovid | since you never set anything the variable will have a value of "" (aka nothing). |
| 07:57.20 | Dovid | if it has no value then pass the call along |
| 07:57.30 | Dovid | the first time the user calls they will get the menu |
| 07:57.41 | Dovid | at the end u set it to 1 |
| 07:57.55 | Dovid | then u set the t exten to go to the start |
| 07:58.16 | Dovid | then it runs thru again at the end u check to see if its set to 1 if it is and it time sout then u dump the clal |
| 07:58.17 | Dovid | call* |
| 07:58.38 | nettie | ckear |
| 07:58.40 | nettie | clear |
| 07:58.41 | nettie | thanx a lot |
| 07:58.43 | nettie | got it |
| 07:59.11 | Jeffjohnson | docelmo:also if i reject the call on the mobile... asterisk still calls and try it again and again... |
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| 08:00.43 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
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| 08:01.17 | Jeffjohnson | i mean dovid :) |
| 08:01.20 | Jeffjohnson | docelmo: sorry |
| 08:02.14 | Dovid | Jeffjohnson: ? |
| 08:02.30 | Jeffjohnson | Dovid: yes? |
| 08:02.36 | Jeffjohnson | Dovid: do u have an idea? |
| 08:02.46 | Dovid | Jeffjohnson: thought u asked something |
| 08:02.46 | Dovid | nm |
| 08:02.49 | Dovid | i goto get to bed |
| 08:02.51 | Dovid | night ev1 |
| 08:03.06 | Jeffjohnson | Dovid: yes I've asked something :) |
| 08:03.46 | Jeffjohnson | Dovid: how I can get busy tones if I reject the call on the mobile and when the mobile is powered off :)) |
| 08:03.58 | Jeffjohnson | Dovid: but if you go to bed now, good night :) |
| 08:04.22 | Dovid | meaning if u reject the call u want asterisk to play busy signal ? |
| 08:04.35 | Jeffjohnson | Dovid: yes |
| 08:04.42 | Dovid | ok |
| 08:04.48 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 08:05.00 | Dovid | if the line is busy, which it seems here then asterisk jumps to n+101 |
| 08:05.04 | Jeffjohnson | Dovid: asterisk don't try again and again to call, it must be very annoying for the other side :) |
| 08:05.14 | Winkie | could anyone tell me why the hell this happens? (4 line paste) |
| 08:05.15 | Winkie | Channel SIP/linksys-081825e8 renamed to SIP/linksys-081825e8<MASQ> |
| 08:05.15 | Winkie | Channel Local/1001@test-normal-3896,1 renamed to SIP/linksys-081825e8 |
| 08:05.15 | Winkie | Channel SIP/linksys-081825e8<MASQ> renamed to Local/1001@test-normal-3896,1<ZOMBIE> |
| 08:05.16 | Winkie | Channel Local/1001@test-normal-3896,1<ZOMBIE> hungup |
| 08:05.24 | Jeffjohnson | Dovid: n is the dial priority? |
| 08:05.43 | Jeffjohnson | Dovid: so my dial hast the priority 1, i must specify 102,Busy? |
| 08:05.55 | Dovid | yes |
| 08:06.09 | Jeffjohnson | Dovid: thx |
| 08:06.12 | Dovid | exten => ExtenNum,102,Congestion |
| 08:06.19 | Dovid | Congestion plays the busy signal |
| 08:06.43 | Jeffjohnson | docelmo: yeah it works :) whats the difference between congestion and busy app? |
| 08:06.54 | Jeffjohnson | docelmo: can't find a helping translation for me |
| 08:07.33 | Jeffjohnson | docelmo: sry :E |
| 08:07.43 | Jeffjohnson | Dovid: it was also for you :) |
| 08:08.19 | j0 | does asterisk function in vmware? i'm running it on a beefy system, but the voice is all choppy |
| 08:08.34 | Jeffjohnson | Dovid: now i also get an busy signal if my mobile is turned off :E |
| 08:08.35 | Dovid | j0: u have zaptel working fine ? |
| 08:08.40 | Dovid | also how much ram u gie it ? |
| 08:08.59 | Dovid | Jeffjohnson: dont u want the busy signal |
| 08:09.00 | Dovid | ? |
| 08:10.00 | j0 | Dovid: how can i check if zaptel isn't working? i've allocated 512mb to asterisk, but its not using more than 100mb |
| 08:10.09 | *** join/#asterisk BugKham (n=bugkham@ppp-58.8.3.80.revip2.asianet.co.th) |
| 08:10.16 | Dovid | j0: lsmod |
| 08:10.32 | Dovid | j0: what kernel ? |
| 08:10.44 | BugKham | anyone has an experience with both E100P and TE110P ? |
| 08:10.48 | Jeffjohnson | Dovid: what means congestion? |
| 08:10.49 | j0 | Dovid: zaptel is in there.. 2.6.9-34 |
| 08:11.25 | Dovid | u did ztdummy too ? |
| 08:11.39 | Dovid | Jeffjohnson: no slots available. cant take call |
| 08:11.41 | BugKham | I've been using E100P and looking to get a new TE110P, don't know if there's any improvement |
| 08:11.48 | j0 | Dovid: ztdummy is also loaded |
| 08:12.06 | BugKham | or inconsistencies btw the two cards |
| 08:12.07 | Dovid | j0: then u have zaptel. it could be the processor |
| 08:12.11 | Dovid | or the bandwith |
| 08:12.15 | Dovid | and lots of others |
| 08:12.22 | Dovid | i used it on vmware and worked. |
| 08:13.07 | Dovid | i goto get to bed |
| 08:13.07 | Dovid | night |
| 08:13.10 | j0 | hmm.. its dual 1ghz, 2gb ram.. and asterisk is the ONLY thing running on it |
| 08:13.39 | j0 | nite |
| 08:14.12 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:16.50 | Jeffjohnson | Dovid: it only works if i reject the call.. what I need to get an busy tone if the mobile is off? |
| 08:18.11 | BugKham | can anyone compare TE110P to E100P in terms of quality? |
| 08:20.44 | j0 | wtf.. if i repeatedly -HUP asterisk, all sound quality issues dissapear.. |
| 08:21.48 | suma | ~seen kpj |
| 08:22.08 | jbot | suma: i haven't seen 'kpj' |
| 08:25.00 | Jeffjohnson | what extension I need to get an Busy tone if I call to an mobile, and the mobile is powered off? |
| 08:25.58 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 08:26.21 | vlt | Hello. I have the problem that sometimes a phone rings that isn't registered to asterisk anymore or that asterisk answes an external call from a sip provider it is not registered for several minutes. Is there a way to UNREGISTER to a service? |
| 08:34.20 | denon | vlt: perhaps add a: qualify=yes to it's sip.conf entry |
| 08:39.25 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 08:41.11 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 08:41.19 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:42.39 | j0 | regular calls are working great, but any recordings from asterisk are jerky.. even the ringing sound is jerky, but the call is perfect |
| 08:47.34 | Curus | asterisk in VMWare sounds like a losing bet |
| 08:48.06 | E-bola | i cant find a reason to do it atleast hehe |
| 08:48.13 | E-bola | unless u got some monster wmware servers running already |
| 08:48.28 | Curus | asterisk in vserver works beautifully |
| 08:48.49 | E-bola | ms vserver? |
| 08:48.55 | Curus | Err no, linux-vserver |
| 08:49.02 | Curus | asterisk in xen is something we'll be trying out soon |
| 08:49.23 | E-bola | well i run a pure sip setup, so i coudlnt care less about performance |
| 08:49.25 | E-bola | :) |
| 08:49.46 | Curus | We host PBX's, so it's nice to give each customer their own |
| 08:49.59 | E-bola | true |
| 08:50.05 | Curus | And asterisk is somewhat limited if you want to put several customers into one dial plan. Doable, but not nice. |
| 08:54.08 | *** join/#asterisk ChrisDE4 (n=ChrisDE@88.128.23.21) |
| 08:55.27 | ChrisDE4 | Hi, I'm again having severe problems with asterisks channel handling |
| 08:55.41 | ChrisDE4 | full:Aug 16 07:49:48 DEBUG[28520] channel.c: Planning to masquerade channel SIP/mycarrier-082a8ee8 into the structure of Lo |
| 08:55.43 | ChrisDE4 | cal/49613152314@dialscript-f698,1 |
| 08:55.43 | ChrisDE4 | full:Aug 16 07:49:48 DEBUG[28520] channel.c: Done planning to masquerade channel SIP/mycarrier-082a8ee8 into the structure |
| 08:55.43 | ChrisDE4 | of Local/49613152314@dialscript-f698,1 |
| 08:55.43 | ChrisDE4 | full:Aug 16 07:49:48 DEBUG[28520] chan_local.c: Not posting to queue since already masked on 'Local/49613152314@dialscript-f6 |
| 08:55.44 | ChrisDE4 | 98,2' |
| 08:55.46 | ChrisDE4 | full:Aug 16 07:49:48 DEBUG[28528] channel.c: Got clone lock for masquerade on 'SIP/mycarrier-082a8ee8' at 0x83b3ab4 |
| 08:55.49 | ChrisDE4 | full:Aug 16 07:49:48 DEBUG[28528] channel.c: Putting channel SIP/mycarrier-082a8ee8 in 64/64 formats |
| 08:55.51 | ChrisDE4 | full:Aug 16 07:49:48 DEBUG[28528] channel.c: Released clone lock on 'Local/49613152314@dialscript-f698,1<ZOMBIE>' |
| 08:55.54 | ChrisDE4 | full:Aug 16 07:49:48 DEBUG[28520] channel.c: Didn't get a frame from channel: Local/49613152314@dialscript-f698,1<ZOMBIE> |
| 08:55.57 | ChrisDE4 | full:Aug 16 07:49:48 DEBUG[28520] channel.c: Bridge stops bridging channels Local/49613152314@dialscript-f698,2 and Local/49613152314@dialscript-f698,1<ZOMBIE> |
| 08:56.00 | ChrisDE4 | full:Aug 16 07:49:48 DEBUG[28520] app_dial.c: Exiting with DIALSTATUS=ANSWER. |
| 08:56.03 | E-bola | ehh |
| 08:56.04 | ChrisDE4 | I'm getting this when trying to originate a call via asterisk manager api |
| 08:56.05 | E-bola | plz use a pastebin |
| 08:56.58 | ChrisDE4 | any ideas what causes this "ZOMBIE" |
| 08:57.34 | ChrisDE4 | vechers told me to ask #asterisk before posting a bug :-) |
| 09:01.09 | ChrisDE4 | anyone there? |
| 09:03.01 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:07.13 | *** join/#asterisk roguebug (n=johnw2@dip-103-051.bras.dsl.breisnet.com) |
| 09:07.16 | roguebug | hi |
| 09:08.42 | *** join/#asterisk chexum (i=chexum@gateway/tor/x-8751e904684d849f) |
| 09:09.00 | roguebug | is it normal that when i have a voicemail my telephone rings every 5 or 10 min. without anyone calling? |
| 09:09.39 | roguebug | (as a notification, probably, or is that not voicemail)? |
| 09:10.06 | ChrisDE4 | what phone are u using? |
| 09:13.59 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:14.51 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 09:18.56 | Winkie | ChrisDE4: i'm having horrific issues with ZOMBIE channels and agents etc |
| 09:19.08 | Winkie | i don't get the whole masquerade channels etc |
| 09:19.30 | Winkie | it's making me close to just giving up on asterisk and writing it off as usable software because it's frustrating as hell |
| 09:19.40 | Winkie | it looks professional to start with but there's a LOT of crap going oin |
| 09:22.10 | MrChimpy | you gets what you pay for winkie. there are alternatives. |
| 09:22.32 | roguebug | i'm using an avm fritzbox (as a sip adapter) with an analog phone (so the whole behaves as a sip phone ) |
| 09:22.33 | MrChimpy | or pay digium to look at your problems. |
| 09:22.57 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:23.38 | Winkie | MrChimpy: so are you saying that Linux is essentially worthless? |
| 09:23.48 | Winkie | because "you get what you pay for" is an utterly stupid statement |
| 09:24.32 | MrChimpy | hardly |
| 09:25.03 | Winkie | well i paid £300 for my office suite and got OpenOffice for free, would you say OpenOffice is worth an infinite amount less than Microsoft Office? |
| 09:25.35 | MrChimpy | it's ok, just carry on bitching about asterisk. |
| 09:26.00 | Winkie | hahaha |
| 09:26.03 | Winkie | way to actually seriously respond |
| 09:26.18 | Winkie | so basically you're just bitching at me because i'm insulting asterisk? |
| 09:26.58 | MrChimpy | funnily enough I have better things to do. |
| 09:27.03 | Winkie | of course you do |
| 09:27.23 | Winkie | I don't of course, because i'm only stalled writing an application to handle thousands of calls a day because of a lack of proper Manager Event Specifications |
| 09:27.31 | Winkie | but that's my fault for not paying for asterisk (??) |
| 09:27.43 | MrChimpy | I've done exactly the same, and you know what I did? |
| 09:27.54 | Winkie | apparantly 'fixed it in asterisk' isn't what you did |
| 09:27.56 | Winkie | so i don't care |
| 09:28.04 | MrChimpy | I patched asterisk and fixed the issues I had. I then fed them back into the tree. |
| 09:28.15 | Winkie | oh really? because i don't seem to see the fixes for these issues |
| 09:28.23 | MrChimpy | The alternative is paying someone to do that problem. |
| 09:28.41 | Winkie | so what did you fix exactly which affects me in any way? |
| 09:28.47 | MrChimpy | The roads open to you are well documented. Spitting your dummy out on IRC won't solve anything. |
| 09:28.51 | mitcheloc | Winkie: what are you making? |
| 09:29.20 | Winkie | mitcheloc: a bit of a swanky call tracker that uses the Manager interface, i'm mostly bitching at chan_agent / app_queue because it seems to be from a really old asterisk version |
| 09:29.29 | mitcheloc | Winkie: that seems to be the concept around open source software, no liability, so if you buy ABE, then you can complain ;) |
| 09:29.35 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 09:29.44 | ChrisDE4 | ok, but can I do now to fix this problem? |
| 09:29.46 | Winkie | MrChimpy: if you noticed I was complaining i was frustrated, you're the one with the bizarre statements |
| 09:29.51 | mitcheloc | What does a call tracker do??? |
| 09:30.03 | Winkie | mitcheloc: no accountability yes, and I can complain all i want, i'll fix everything i can anyway :) |
| 09:30.21 | Winkie | mitcheloc: it provides data on all current calls, inbound and out, their states and also adds in a lot of data about them to a database after the call is finished |
| 09:30.29 | Winkie | i'm trying to make it realtime though, so as the call progresses the database is updated |
| 09:30.31 | *** join/#asterisk chexum (i=chexum@gateway/tor/x-bc4f343abd0742ac) |
| 09:30.48 | mitcheloc | Winkie: i wasn't attacking you, just saying, I think ABE is the answer to that...supposedly. I agree though, I don't care for that "fix it yourself" attitude... |
| 09:30.59 | Winkie | this is VERY hard with events like 'AgentCalled' and I can't find any documentation on the UniqueIDs specified in manager events, can i safely assume everything before the . is part of the same call etc |
| 09:31.09 | Winkie | mitcheloc: i know dude :) |
| 09:31.32 | MrChimpy | manager events are very simple to add, and very simple to look at in the source code |
| 09:31.33 | Winkie | I doubt digium would do shit anyway, i've already faxed off my disclaimer and i'm going to run through SVN code in a week or two and fix all the events I can see |
| 09:31.51 | Winkie | MrChimpy: i've already added an event of my own how do you still not see what i am complaining about? |
| 09:32.18 | mitcheloc | Unique IDs don't mean anything... I agree.. I've had to decode the manager events as well... it's not very straight forward... |
| 09:32.22 | MrChimpy | use the source! |
| 09:32.25 | bionoid | I have an incoming call via Zap, which dials a SIP client. Is there a way to accept extensions from the callee during the conversation? |
| 09:32.49 | Winkie | MrChimpy: I have looked through the source as best possible, it seems to be a timestamp and an incrementing number |
| 09:32.56 | Winkie | no information on whether the timestamp is guarenteed to be unique |
| 09:33.10 | Winkie | so if it's as easy as 'using the source', please tell me whether it is or isn't |
| 09:33.28 | Winkie | incidentally i'm basing about £4000 of business a day off this when it's finished, so you know, it'd be nice to be sure |
| 09:33.29 | MrChimpy | how would the timestamp be unique? |
| 09:33.35 | MrChimpy | look at the granularity of it |
| 09:33.51 | MrChimpy | if you get more than 1 call in that time frame it won't be unique |
| 09:33.51 | Winkie | MrChimpy: i assume it's msec granularity, what's the point of that? |
| 09:33.56 | MrChimpy | hence the incremental number |
| 09:34.07 | Winkie | then why is there no documentation stating this |
| 09:34.18 | Winkie | and why is asterisk totally non-geared to linking call events together in ANY meaninfgul way |
| 09:34.25 | MrChimpy | cos you haven't written it? :) |
| 09:34.52 | mitcheloc | Winkie: i feel your pain *sigh* |
| 09:34.52 | Winkie | ah right, so documentation is the responsibility of the user who comes in and looks at the code later? |
| 09:34.56 | MrChimpy | if asterisk doesn't suit your needs use something else! |
| 09:35.00 | Winkie | MrChimpy: oh for fuck's sake |
| 09:35.02 | Winkie | stop talking |
| 09:35.04 | Winkie | seriously |
| 09:35.28 | MrChimpy | winkie: contributing is kind of helpful, if you've found something trick FOSS ethos would be to document |
| 09:35.41 | bXi | yo |
| 09:35.48 | hads|home | I'm confused. Using trunk, cdrs pick up the system timezone correctly but SayUnixTime or STRFTIME don't. |
| 09:35.50 | bXi | i have a weird problem here |
| 09:35.53 | Winkie | MrChimpy: no it wouldn't, it would be to file a bug. |
| 09:35.55 | MrChimpy | there's documentation wikis |
| 09:36.13 | bXi | the moment i use an extension to call a hardphone from a softphone i dont hear anything |
| 09:36.13 | Winkie | point me to the one explaining why uniqueids aren't used everywhere |
| 09:36.23 | bXi | when i call the ip itself it works |
| 09:36.33 | MrChimpy | MAIL AND ASK DIGIUM THEN! |
| 09:36.45 | Winkie | bionoid: you want to look at an application I forgot about, something like Allow Inbound Systems Access (I really forget) |
| 09:37.12 | Winkie | MrChimpy: so what you're saying is, the 'FOSS' way is for the creator not to document it, then a user to email the company who created it to get the answer to file a bug to add some documentation to the original source? |
| 09:37.25 | Winkie | i think what you were trying to say is "Really? It varies whether it's included? That sucks :(" |
| 09:38.09 | r_marvin | the "foss" way is "what you see is what you get, if you want more, feel free to add it yourself" |
| 09:38.12 | Winkie | of course i can patch it if i want, which is what i'm going to do, but telling me 'you get what you pay for' and 'you should document it yourself' etc is ludicrous, i'm complaining because of the lack of documentation which should have been there in the first place |
| 09:38.18 | MrChimpy | i'm saying FOSS stuff won't be perfect in terms of documentation or code. by contributing you improve the situation. more people that do that, the better the situation gets. It's kind of "how things work". |
| 09:38.35 | bionoid | Winkie: Hm, I've been looking around for a while, but the search goes on :-) Cheers |
| 09:38.52 | Winkie | r_marvin: I agree, but my point is there are some pretty simple things that should be included that aren't and i have a genuine gripe :) |
| 09:39.01 | MrChimpy | wink: go bitch to the code author then. i'm sure he'd love to hear it :) |
| 09:39.28 | r_marvin | Winkie: ok, so if bitching about it is the best you think you could do, then feel free to go ahead |
| 09:39.33 | Winkie | MrChimpy: i probably will unless i fix it myself, but it doesn't stop me from being very very annoyed |
| 09:39.57 | Winkie | r_marvin: hey it's not the best i can do but it shouldn't be like this in the first place and it makes me angry that asterisk suffers from issues like this |
| 09:40.23 | r_marvin | Winkie: then if you think you can do better, why aren't you doing it? |
| 09:40.44 | Winkie | r_marvin: because nobody is paying me to rewrite chan_agent? |
| 09:40.57 | MrChimpy | so why should anyone else? |
| 09:40.57 | Winkie | and frankly the person who did it originally should have done a MUCH better job of it |
| 09:41.08 | Winkie | but i can understand not doing if it is part of an older revision |
| 09:41.10 | r_marvin | yeah! you tell them |
| 09:41.18 | r_marvin | jeez |
| 09:41.33 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 09:41.39 | r_marvin | unlike you, i got better things to do |
| 09:41.46 | Winkie | oh please |
| 09:41.51 | Winkie | you've both used exactly the same pathetic arguments the same |
| 09:42.05 | Winkie | i love the fact that you think the 'entire point' of open source is that anyone can improve it |
| 09:42.11 | Winkie | go ask RMS that |
| 09:42.12 | MrChimpy | funny that. other people might take that as validation. |
| 09:42.31 | MrChimpy | instead EVERYONE ELSE IS WRONG |
| 09:42.34 | Winkie | hahaha |
| 09:42.38 | Winkie | read what you both wrote |
| 09:42.41 | Winkie | 10:42.51 < r_marvin> unlike you, i got better things to do |
| 09:42.41 | r_marvin | MrChimpy: stop feeding the troll, it's obvious this is going nowhere |
| 09:42.56 | Winkie | haha now i'm a troll? |
| 09:43.00 | Winkie | dear god you people are retarded |
| 09:43.01 | MrChimpy | ta for reminding me :) |
| 09:43.07 | r_marvin | np :) |
| 09:43.13 | Winkie | oh god this is so funny |
| 09:43.15 | r_marvin | also, /ignore is great for trolls |
| 09:43.30 | Winkie | i guess i will leave you two retards to go jerk each other off whilst you totally ignore my point |
| 09:43.48 | Winkie | after all i'm sure that's the thing you have to do that's obviously better than anything i have to do |
| 09:44.04 | MrChimpy | one less support channel for you :) |
| 09:44.28 | Winkie | haha |
| 09:44.43 | Winkie | oh no i will have to fix all my problems myself like i was already doing |
| 09:45.03 | MrChimpy | good luck. remember to feed them back into the tree |
| 09:45.15 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 09:45.43 | Winkie | thanks! Hope you learn what open source means at some point :) |
| 09:46.41 | *** join/#asterisk dacleric (n=dacleric@p548200F9.dip0.t-ipconnect.de) |
| 09:59.09 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 09:59.11 | nicox | hello |
| 09:59.22 | Winkie | hi |
| 09:59.35 | nicox | did anybody know the difference between the svn-trunk and svn-branch version? |
| 09:59.44 | phearless | how can I be SURE that my phone line works with asterisk ? |
| 09:59.51 | Winkie | phearless: physical phone line? |
| 09:59.53 | phearless | yes |
| 10:00.02 | Winkie | phearless: you have it plugged into a card or what? |
| 10:00.08 | phearless | yes |
| 10:00.19 | Winkie | phearless: have you tried calling it? ;) |
| 10:00.19 | phearless | but I can't use it in staerisk for an unknown reason |
| 10:00.32 | phearless | I have put a normal phone in the line, and I can call |
| 10:00.38 | phearless | but in asterisk it does not work |
| 10:00.40 | nicox | can anyone tell me the difference? |
| 10:00.52 | Winkie | phearless: what card have you got in your asterisk box? |
| 10:01.05 | phearless | I want to be sure that there is no problems linked to ISDN, voltage or any weird things like this |
| 10:01.41 | phearless | for example on http://www.automated.it/guidetoasterisk.htm it is written : |
| 10:01.51 | phearless | If you have a spare phone then plug this into the phone interface on the card too. It is always good to have a phone plugged into this interface because in the event of asterisk failing, or a power cut the card actually still allows access to the PSTN line. Obviously if you decide to use a phone that is not powered from the phone line, if you have a power cut, it will not work. |
| 10:02.04 | phearless | I do not really understand what does that mean |
| 10:02.17 | phearless | I got one TDM400P, with one FXO module |
| 10:02.28 | inspired | phearless, then you don't use ISDN |
| 10:02.32 | *** join/#asterisk xnon (i=xnon@200.82.222.64) |
| 10:02.38 | xnon | hello friends |
| 10:02.57 | phearless | so how can I be SURE that my phone line works with asterisk ? |
| 10:02.57 | Winkie | yeah, FXO is like a normal phone, you need FXS to plug a normal phone into it |
| 10:03.16 | phearless | yes I do not plan to plug a phone in the asterisk box |
| 10:03.28 | phearless | I tried to plug a phone on the phone line and it works |
| 10:03.54 | xnon | in a asterisk server is posible run other service for example squid, samba, IPTABLES, etc? these no afect my asterisk server ? |
| 10:04.22 | inspired | phearless, did you install zaptel? |
| 10:04.50 | inspired | did you try calling out on the phone line through asterisk? did you configure anything at all? try giving us some more details about your situation |
| 10:05.44 | phearless | <inspired> phearless, did you install zaptel? <- yes sure |
| 10:06.21 | phearless | <PROTECTED> |
| 10:06.22 | inspired | and you said you want to be sure there are no problems linked to ISDN. how did ISDN get in the picture? you said you are using a tdm400p (which is analog) |
| 10:06.32 | phearless | <PROTECTED> |
| 10:06.58 | phearless | how did ISDN get in the picture ? <--- I do not know ISDN this is why I am wondering if it can be linked to the problem |
| 10:07.10 | inspired | maybe you can connect to the asterisk CLI and show us the output (use http://pastebin.ca) when you make a call? |
| 10:07.18 | phearless | so I got a TDM400P with a FXO (red) module |
| 10:07.22 | inspired | what phone did you plug into the wall? is it an ISDN phone? |
| 10:07.33 | phearless | a normal basic phone |
| 10:07.35 | inspired | in that case plugging your TDM400P into the phone jack on the wall is a no-go |
| 10:07.46 | inspired | ok, so you don't use ISDN. |
| 10:07.51 | phearless | I am in UK |
| 10:08.01 | phearless | I do not know if there is somethign special in UK |
| 10:08.02 | mut | damn malt o meal cereal bags |
| 10:08.09 | mut | the zip lock on the bag is stronger than the plastic holding the bag together |
| 10:08.18 | mut | i try to open it and ripped the side off before the ziplock decided to unzip |
| 10:08.20 | inspired | except for the fact that you drive on the left side of the road? |
| 10:08.26 | phearless | <inspired> maybe you can connect to the asterisk CLI and show us the output (use http://pastebin.ca) when you make a call? <--- I can do this |
| 10:08.32 | nicox | did anybody know the difference between the svn-trunk and svn-branch version? |
| 10:08.34 | phearless | <inspired> except for the fact that you drive on the left side of the road? <--- :p |
| 10:08.52 | inspired | ok, paste your output when making a call to pastebin.ca |
| 10:08.55 | inspired | and give us the url |
| 10:09.53 | phearless | http://paste-bin.com/85 |
| 10:09.57 | phearless | here it is |
| 10:10.01 | inspired | and what does ztcfg -vvvv on the linux command line show? |
| 10:10.33 | phearless | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 10:10.33 | phearless | 1 channels configured. |
| 10:10.51 | phearless | and the module is in the first slot |
| 10:11.35 | phearless | and I got a green LED near the module, where I plug the phone line |
| 10:11.48 | phearless | and I connected the Molex to the PCI card |
| 10:12.27 | phearless | and my phone line need a 9 before the number, to dial out |
| 10:13.07 | inspired | hmm, I don't use analog interfaces, but shouldn't channel 01 really be FXO? |
| 10:13.23 | inspired | did you connect channel 1 to the phone line? |
| 10:13.28 | inspired | or to a phone? |
| 10:13.39 | phearless | I got only a FXO module |
| 10:13.44 | phearless | so I connected the phone line in it |
| 10:14.05 | phearless | I tried before to plug a phone to the phone line, without asterisk, to try the phone line |
| 10:14.17 | inspired | uhm, you need a FXO module afaik |
| 10:14.21 | inspired | sorry, FXS |
| 10:14.36 | *** part/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 10:14.39 | phearless | no, FXO modules are used to plug phone lines |
| 10:15.04 | phearless | but FXO ports use FXS signalling |
| 10:15.09 | inspired | uhm, yeah true |
| 10:15.18 | phearless | FXO digium modules are red, and I got a red one |
| 10:16.03 | inspired | # |
| 10:16.04 | inspired | -- Executing Dial("SIP/200-07f5", "ZAP/g0/902077963002|120|r") in new stack |
| 10:16.04 | inspired | # |
| 10:16.04 | inspired | <PROTECTED> |
| 10:16.12 | inspired | can you try changing g0 to g1? |
| 10:16.29 | phearless | i will have a look |
| 10:17.46 | phearless | I got in the config files : |
| 10:17.48 | phearless | OUT_1 = ZAP/g0 |
| 10:17.58 | phearless | i will try with g1 instead |
| 10:19.38 | phearless | <PROTECTED> |
| 10:19.43 | phearless | I got the busy message |
| 10:19.47 | phearless | with g1 |
| 10:19.54 | inspired | ok, change back to g0 |
| 10:19.59 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 10:20.44 | inspired | no idea really. analog interfaces are not my area |
| 10:21.52 | phearless | by analog interfaces you mean not via internet ? |
| 10:22.08 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 10:24.57 | inspired | I only use ISDN |
| 10:25.12 | inspired | not POTS |
| 10:26.37 | phearless | how can I be sure that it is not an ISDN line ? |
| 10:31.33 | xnon | friend i cant make meetme conference in my asterisk |
| 10:31.37 | xnon | anybody cant hellpme |
| 10:32.19 | xnon | in my extensions.conf i was type [meetme] with this line exten => 100,1,MeetMe,4000 |
| 10:33.00 | xnon | in my meetme.conf i was type in the context [rooms] this line conf => 4000 |
| 10:33.45 | xnon | but when i dial to exten 100 the operator say that this conference no exit! |
| 10:34.23 | xnon | 3 WARNIGNS and 1 ERROR Mensage! |
| 10:34.30 | phearless | "The order in which you do the modprobe’s IS important. If you modprobe the FXO (modprobe wcfxo) card first then it will be channel 1, if you modprobe the FXS (modprobe wcfxs) card first then its first port will be channel 1, the second channel 2 and so on…" |
| 10:34.37 | phearless | is it true with the module wctdm ? |
| 10:37.37 | xnon | anybody know how i can make a conference calling |
| 10:38.12 | xnon | the first warning is: Aug 16 05:32:59 WARNING[11254]: chan_zap.c:915 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory |
| 10:39.37 | kaldemar | xnon: meetme takes it's timing from a zaptel device, and looks like you don't have one. |
| 10:40.12 | kaldemar | xnon: http://www.voip-info.org/wiki/index.php?page=Asterisk+timer |
| 10:40.55 | xnon | umm ok leet me see this site ! |
| 10:41.53 | xnon | kaldemar, i need ztdummy |
| 10:41.56 | xnon | ???????? |
| 10:41.58 | kaldemar | bingo |
| 10:42.04 | xnon | :) |
| 10:42.14 | xnon | kaldemar, do u speak spanish? |
| 10:42.17 | tzafrir | phearless, that information is a bit obsolete: your terminology is incorrect |
| 10:42.32 | kaldemar | xnon: no, unfortunately not. |
| 10:42.44 | xnon | ok |
| 10:42.55 | phearless | tzafrir: about analog? |
| 10:42.56 | tzafrir | wcfxs no longer exists in recent zaptel |
| 10:43.01 | phearless | ah ! |
| 10:43.04 | xnon | kaldemar, so i need to download this ztdummy and install it? |
| 10:43.05 | phearless | okay |
| 10:43.11 | tzafrir | what version of zaptel do you use? |
| 10:43.14 | phearless | thanks tzafrir |
| 10:43.20 | phearless | the svn one |
| 10:43.23 | xnon | ztdummy it is a module for asterisk? |
| 10:43.27 | phearless | I compiled it |
| 10:43.32 | kaldemar | xnon: do you have zaptel? |
| 10:43.39 | xnon | yes i have |
| 10:43.53 | phearless | zaptel-1.2.6 |
| 10:44.07 | phearless | tzafrir: zaptel-1.2.6 |
| 10:44.22 | kaldemar | it comes with zaptel, so if you have the ztdummy module, modprobe it before you start asterisk. |
| 10:44.23 | xnon | shit no i dont have it friend! |
| 10:44.33 | tzafrir | The order of modprobes does matter. Though I must point out that what technically matters is the point in time when the card's span registers with zaptel |
| 10:44.48 | tzafrir | (the above is not theoretical when you deal with xpp) |
| 10:45.04 | xnon | kaldemar, take a look demeter:/var/spool/asterisk/monitor# dpkg -l | grep zaptel |
| 10:45.04 | xnon | rc zaptel-modules 1.2.6-2+2.6.8- zaptel modules for Linux (kernel 2.6.8-2-386 |
| 10:45.22 | xnon | i dont have it i think so! |
| 10:45.39 | phearless | tzafrir: what is xpp ? |
| 10:45.42 | xnon | what can i dooo! |
| 10:45.52 | tzafrir | xnon, why not just install zaptel and zaptel-modules-`uname -r` (or use m-a to build it)? |
| 10:45.57 | phearless | tzafrir: I got a TDM400P and a FXO module on it (first slot) |
| 10:46.21 | tzafrir | phearless, a subdirectory in the zaptel source tree |
| 10:46.32 | phearless | tzafrir: so do I need a modprobe "order" ? |
| 10:46.43 | xnon | ok ill be do it! |
| 10:46.44 | phearless | <tzafrir> phearless, a subdirectory in the zaptel source tree <- ok |
| 10:46.51 | tzafrir | If you have just one module, you can't get the order wrong |
| 10:47.23 | phearless | so what is the order ? |
| 10:47.31 | tzafrir | Any other zaptel module will either fail to load or load but just not register a span with zaptel |
| 10:47.36 | phearless | zaptel, and then wctdm ? |
| 10:47.50 | tzafrir | if you modprobe wctdm it will also load zaptel |
| 10:48.03 | tzafrir | This is why you use modprobe and not insmod |
| 10:48.11 | xnon | tzafrir, zaptel-1.2.7 its ok? |
| 10:48.28 | tzafrir | xnon, sure. Latest release is always ok |
| 10:48.32 | phearless | ok tzafrir |
| 10:48.39 | xnon | ok |
| 10:48.41 | phearless | wctdm 34880 1 |
| 10:48.41 | phearless | zaptel 206852 5 wctdm |
| 10:48.44 | phearless | I got this loaded |
| 10:49.23 | tzafrir | phearless, so now you need to configure it using ztcfg, if it's not configured yet. |
| 10:49.32 | phearless | so I do NOT need wcfxo |
| 10:49.45 | phearless | ztcfg -vvvv gives me : |
| 10:49.50 | tzafrir | Then (re)start asterisk and see if you have zap channels |
| 10:49.54 | phearless | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 10:49.54 | phearless | 1 channels configured. |
| 10:50.01 | tzafrir | wcfxo is the driver for X100P cards |
| 10:50.05 | tzafrir | and clones |
| 10:50.31 | phearless | zap show channels |
| 10:50.31 | phearless | <PROTECTED> |
| 10:50.31 | phearless | <PROTECTED> |
| 10:50.31 | phearless | <PROTECTED> |
| 10:50.38 | phearless | <tzafrir> wcfxo is the driver for X100P cards <-- ok |
| 10:50.39 | tzafrir | Looks OK |
| 10:50.55 | phearless | what is "pseudo" ? |
| 10:51.05 | tzafrir | pseudo is used for timing |
| 10:51.20 | phearless | ok |
| 10:51.37 | phearless | but I still can't call out |
| 10:51.53 | tzafrir | phearless, what happens when you try to call out? |
| 10:52.09 | tzafrir | please pastebin a cli trace (or from the log) |
| 10:53.01 | phearless | http://paste-bin.com/85 |
| 10:54.21 | xnon | Se instalarán los siguientes paquetes NUEVOS: |
| 10:54.21 | xnon | <PROTECTED> |
| 10:55.29 | phearless | tzafrir: does it look OK ? |
| 10:55.33 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 10:58.21 | tzafrir | Executing Dial("SIP/200-07f5", "ZAP/g0/902077963002|120|r") in new stack |
| 10:58.32 | tzafrir | Zap/1-1 answered SIP/200-07f5 |
| 10:58.41 | tzafrir | Hungup 'Zap/1-1' |
| 10:58.59 | tzafrir | Looks like a call was established, but immedietly hung-up |
| 10:59.05 | tzafrir | What is the other side? |
| 10:59.39 | phearless | no I handged up myself after a few seconds |
| 10:59.44 | EyeCue | hung |
| 11:00.09 | phearless | I got 1 ring, then the time counter starts on the phone, and I heard just noting or "crrr crrr" |
| 11:00.23 | phearless | <tzafrir> What is the other side? <-- what do you mean ? |
| 11:00.29 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 11:02.45 | tzafrir | the other side of the phone line. A telco? |
| 11:03.37 | phearless | yes |
| 11:03.49 | phearless | I can phone, with a phone, o nthe line |
| 11:04.05 | phearless | I just need to use 9+thephonenumber |
| 11:04.11 | phearless | I tried and I can call |
| 11:04.50 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 11:05.05 | EyeCue | theres an answer to that exact question on the forums |
| 11:05.07 | EyeCue | 1-3rd page. |
| 11:05.23 | EyeCue | unless i didnt scroll up enough :D |
| 11:05.26 | *** part/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 11:05.56 | phearless | http://forums.digium.com/viewforum.php?f=1 here? |
| 11:06.00 | phearless | which topic? |
| 11:08.38 | EyeCue | if its about always prefixing a number to extensions |
| 11:08.39 | EyeCue | yeh |
| 11:09.23 | phearless | this is not a prefix pb |
| 11:09.24 | EyeCue | http://forums.digium.com/viewtopic.php?t=8913&sid=b28566e1d8ad3ebfcd07853bc23ee85c |
| 11:09.36 | phearless | asterisk add the 9 without problems, cf my log |
| 11:09.59 | EyeCue | my bad then |
| 11:15.39 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 11:23.49 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 11:24.51 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:29.36 | phearless | voip hates me ! |
| 11:30.38 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 11:31.00 | *** part/#asterisk ChrisDE4 (n=ChrisDE@88.128.23.21) |
| 11:31.31 | *** join/#asterisk grEvenX (n=even@pc100-15.ktv.no) |
| 11:32.01 | Assid | why |
| 11:32.43 | RoyK | hm |
| 11:33.48 | RoyK | when dialing sip->zap with the t flag, transfer with the features sequence works well, but if doing zap->sip call, that won't work |
| 11:33.53 | RoyK | any ideas why? |
| 11:34.01 | RoyK | zap->sip is done with the T flag |
| 11:34.40 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 11:35.11 | bXi | is there a way to do a call from asterisk CLI? |
| 11:35.44 | r_marvin | yes, dial |
| 11:35.59 | *** join/#asterisk BjornRobertsson (n=bjornr@213-213-148-71.xdsl.is) |
| 11:36.04 | bXi | no such command it says |
| 11:36.05 | r_marvin | but you'll use the local soundcard |
| 11:36.21 | bXi | i have some isdn card in the pc |
| 11:36.26 | bXi | but i dont know the number attached |
| 11:36.29 | r_marvin | CLI> help dial |
| 11:36.29 | r_marvin | Usage: dial [extension[@context]] |
| 11:36.29 | r_marvin | <PROTECTED> |
| 11:36.31 | bXi | so i want to call my mobile [hone |
| 11:36.47 | bXi | phone*CLI> help dial |
| 11:36.47 | bXi | No such command 'dial'. |
| 11:36.53 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 11:37.13 | Aurs | bXi: depends on what versjon of asterisk you have |
| 11:37.20 | bXi | 1.2.10 |
| 11:41.17 | *** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br) |
| 11:42.09 | *** join/#asterisk mcnobody (n=laaksola@laaksola.net) |
| 11:43.17 | mcnobody | Hi! |
| 11:47.48 | mcnobody | Is it possible to allow all SIP calls from unknown callers to local SIP peers? |
| 11:49.18 | mcnobody | allowguest=yes, makes it almost. All calls with locally unknown user part in From:-header are accepted, but calls from other Asterisk with same user part of From:-header are tried to authenticate against local peer. |
| 11:52.42 | *** part/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 11:53.20 | roguebug | i'd like to be able to put anyone i have on the phone (whether they called me or the other way around doesn't matter) into a meetme. is that possible? |
| 11:53.30 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 11:53.33 | roguebug | from inside the normal call that is |
| 11:54.18 | roguebug | like i'd tell that person "one moment, i'll put you on conference and call <3rd conference member> |
| 11:54.21 | roguebug | ? |
| 12:01.47 | *** join/#asterisk |oranjia| (n=root@dsl-146-39-25.telkomadsl.co.za) |
| 12:01.49 | RoyK | ~seen coppice |
| 12:01.56 | jbot | coppice <n=chatzill@229.166.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 22h 38m 32s ago, saying: 'The A104D is very handy if you need low profile'. |
| 12:02.38 | xnon | friends i have this warning: |
| 12:02.39 | xnon | Aug 16 06:58:09 WARNING[13055]: pbx.c:5705 pbx_builtin_waitexten: Timeout but no rule 't' in context 'default' |
| 12:02.49 | xnon | why? |
| 12:02.59 | xnon | what mean!? |
| 12:03.19 | |oranjia| | has anyone used valgrind on the asterisk daemon? |
| 12:03.33 | james_ | it means the call timed out, but you have no t rule to handle it in the dialplan in context default |
| 12:03.35 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 12:03.38 | james_ | so add |
| 12:03.48 | james_ | exten => t,1,NoOp(woo) |
| 12:03.55 | james_ | to [default] |
| 12:03.57 | james_ | in extensions.conf |
| 12:04.54 | *** join/#asterisk Ebola (n=Ebola@user-54458db0.lns1-c13.telh.dsl.pol.co.uk) |
| 12:04.57 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 12:05.03 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 12:05.03 | *** join/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:06.01 | xnon | ok |
| 12:07.28 | xnon | ok ready |
| 12:07.59 | xnon | but why dont ring in the extension im trying call? |
| 12:10.13 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 12:10.35 | DarKnesS_WolF | where i can read about busy redialing in astersik ? |
| 12:11.29 | [TK]D-Fender | DarKnesS_WolF: "show application retrydial" |
| 12:11.50 | DarKnesS_WolF | thx :) |
| 12:11.51 | BjornRobertsson | any suggestions as to why the transfer button in X-lite does not function? |
| 12:12.30 | mut | do wildcards work with includes? |
| 12:12.35 | RoyK | BjornRobertsson: perhaps you don't dial with [tT] |
| 12:12.44 | mut | #include <sipconfs/*.conf> |
| 12:13.19 | BjornRobertsson | Can I configure that in FreePBX ? |
| 12:13.31 | Aurs | Point to be noted that the button Transfer and Conf in the buttom of X-lite is Deactive |
| 12:13.43 | Aurs | from google |
| 12:16.57 | *** join/#asterisk myiagy (n=myiagy@201.72.104.241) |
| 12:17.07 | BjornRobertsson | did have T in General Settings |
| 12:18.24 | grEvenX | transfer doesn't work in X-Lite I think... |
| 12:18.41 | DarKnesS_WolF | [TK]D-Fender: seems that now whati 'm looking for .. i'm looking for a way so when the number is busy asterisk will keep trying to call until it answers and call the caller " me " back to pick up the call |
| 12:18.43 | Aurs | I think you have to buy a registered version to make that button work.. but not 100% sure |
| 12:19.04 | BjornRobertsson | does anyone know about a SIP softphone which can do transfer then? |
| 12:19.06 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:19.27 | [TK]D-Fender | BjornRobertsson: X-Lite cripples that function and a few others in that free version. If you want them buy the payed versions X-Pro or eyeBeam. |
| 12:19.33 | vlt | Hello. I try to register from an asterisk behind a (Debian Sarge) NAT router to a SIP account. I added "nat=yes, externip=... and localnet=..." to sip.conf. Then I even forwarded UDP port 5060 to the asterisk machine. But I can't register. What did I miss? Would using IAX2 be better from behind NAT? |
| 12:19.33 | BjornRobertsson | for some reason idefisk has lesser quality than sip softphones |
| 12:19.38 | Aurs | BjornRobertsson: SJ phone |
| 12:20.28 | *** join/#asterisk luchshiy (n=anonymou@212.82.196.190) |
| 12:20.37 | [TK]D-Fender | vlt: You should also forward 10000-20000 for RTP as well.... |
| 12:21.02 | vlt | [TK]D-Fender: Is this nessecary already for registering? |
| 12:21.04 | [TK]D-Fender | vlt: Though if its just the register thats failing, something would seem pretty off.... |
| 12:21.14 | mut | tk, you know if wildcards are allowed for includes? |
| 12:21.25 | [TK]D-Fender | vlt: Pastebin your setup |
| 12:21.26 | benjk | vlt, indeed IAX is better for NAT traversal |
| 12:21.52 | vlt | !pastebin |
| 12:21.58 | vlt | ~pastebin |
| 12:22.01 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 12:22.19 | ruskie | don't forget paste.se |
| 12:23.00 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 12:23.24 | *** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br) |
| 12:27.36 | vlt | benjk: Thank you, I'll try IAX2. |
| 12:28.25 | *** join/#asterisk robbie2 (n=rob@60.231.21.101) |
| 12:28.37 | vlt | [TK]D-Fender: While preparing pasting my setup I discovered that my [sip] was set to type=peer. I changed it to friend and now it seems to work. Thank you. |
| 12:29.01 | benjk | its not that it is outright impossible to do SIP/RTP NAT traversal, but if there's something available that's easier for the task then you may as well use it if you can |
| 12:29.41 | [TK]D-Fender | vlt: Peer is indeed only for outgoing connections.... and doesn't imply a register either. |
| 12:31.18 | [TK]D-Fender | benjk: Only point of IAX IMO (and most) its its namesake : Inter Asterisk eXchange. When trunking calls to save on bandwidth and possibly for context control otherwise. Beyond that I advocate the more "standard" SIP anywhere it'll function. |
| 12:32.00 | benjk | that may be so for you, but it is not necessarily so for others |
| 12:32.47 | *** join/#asterisk Vec (n=Vector@dsl-146-93-121.telkomadsl.co.za) |
| 12:32.53 | benjk | as for standard, white elephants don't usually live up to the reasons why a standard is established in the first place |
| 12:33.00 | iCEBrkr | docelmo: wakeup |
| 12:33.07 | benjk | OSI networking is a standard, too |
| 12:33.33 | iCEBrkr | pfft! Who needs standards! |
| 12:34.02 | benjk | mind you, IAX is in the IETF standards track |
| 12:34.05 | docelmo | What do you want bitch |
| 12:34.15 | iCEBrkr | docelmo: your sexy ass. :P |
| 12:36.53 | DarKnesS_WolF | [TK]D-Fender: any idea what app can do what i need :-s? call back on busy and give the caller a ring to pick the call up ? |
| 12:37.58 | *** join/#asterisk hittop (n=Miranda@toronto-HSE-ppp4255074.sympatico.ca) |
| 12:38.45 | [TK]D-Fender | DarKnesS_WolF: Make a script yourself using .call files |
| 12:39.21 | hittop | Hi, I'm having trouble installing a X100P card and get that to work with fc5... it shows no configured channel after ztcfg -vvv. I'v read some threads online and it says there's a problem beyond fc4.. but solutions were not posted |
| 12:39.49 | hittop | I wonder if there's any solution to install x100p driver for fc 5 |
| 12:39.49 | DarKnesS_WolF | .call file |
| 12:39.50 | tuxick | the joys of packaged linux |
| 12:39.50 | DarKnesS_WolF | hum |
| 12:39.53 | DarKnesS_WolF | will dig it |
| 12:39.59 | tuxick | "wait for them to fix it" |
| 12:40.18 | *** join/#asterisk Ebola (n=Ebola@user-54458db0.lns1-c13.telh.dsl.pol.co.uk) |
| 12:40.38 | *** join/#asterisk myiagy (n=myiagy@201.72.104.241) |
| 12:40.55 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:41.00 | [TK]D-Fender | hittop: pastebinyour zaptel.conf |
| 12:41.01 | [TK]D-Fender | ~pb |
| 12:41.10 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 12:42.03 | [TK]D-Fender | hittop: Though I do have experience with FC% attempting jsut to compile Zaptel at all and failing miserably. If you got it that far it SHOULD only be a config issue |
| 12:43.59 | hittop | oh.. Coz last few days when i did that, i thought it was be the hardware problem (coz the card is 2 years old, and i ordered 2 other ones). But just today, I've found a hardware detection script in trixbox, and it seems like the hardware can be detected. (let me paste out the zaptel.conf) |
| 12:44.08 | trelane_ | botsnack |
| 12:44.18 | trelane_ | hrm... thoguht you were an infobot |
| 12:45.17 | hittop | zaptel.conf: fxsks=1; loadzone = us; defaultzone=us; channels=1 |
| 12:45.39 | [TK]D-Fender | hittop: no "channels" line there.... |
| 12:46.47 | hittop | how about in zapata.conf? |
| 12:47.51 | Jeffjohnson | what extension I need to get an Busy tone if I call to an mobile, and the mobile is powered off? |
| 12:48.45 | hittop | [TK]D-Fender: does it make sense to have 0 channels configured (ztcfg) for x100p |
| 12:51.11 | Jeffjohnson | What pattern i need to match all 0160 calls? _016XXXXXXXXXX don't work, why? :o |
| 12:51.20 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:52.05 | markstos | I have an analog fax hooked into Asterisk with a grandstream adapter. Whenever the fax line is called, it rings busy, even through there is no activity. Sound familiar? tips? |
| 12:52.33 | grEvenX | woudln't _016XXXXXXXXX only match numbers starting with 016, and then have 10 following numbers |
| 12:52.38 | [TK]D-Fender | hittop: Did you modprobe the card? |
| 12:52.54 | hittop | [TK]D-Fender: yes I did.. i used modprobe wcfxo |
| 12:53.14 | [TK]D-Fender | grEvenX: Looks fine. |
| 12:53.27 | [TK]D-Fender | hittop: Do you see it in cat /proc/interrupts? |
| 12:54.17 | hittop | [TK]D-Fender: 11: 722084 XT-PIC wcfxo, uhci_hcd:usb1, uhci_hcd:usb2 |
| 12:55.05 | hittop | [TK]D-Fender: I'm not sure what they are.... |
| 12:57.33 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 12:57.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:58.06 | [TK]D-Fender | hittop: Ok, not sure why it wouldn't be configured then.... hmmm. Also they hate sharing interrupts and its piled up on that one.... |
| 12:58.08 | hittop | [TK]D-Fender: Oh.. nvm.. I'm soo sorry. i realized that I've made a mistake.. I created my own zaptel.conf in /etc/asterisk, and modification were all in there.. >_< sry.. stupid me |
| 12:58.22 | [TK]D-Fender | hittop: "oops" ;) |
| 12:58.49 | [TK]D-Fender | hittop: But do try and fix that IRQ issue or you may find yourself getting dropped calls, static, etc. |
| 12:58.49 | hittop | [TK]D-Fender: omg.. this little mistake took me that many days.. and i bought two extra cards because of this>_< |
| 13:00.18 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 13:00.48 | hittop | [TK]D-Fender: are you talking about the "wcfxo, uhci_hcd:usb1, uhci_hcd:usb2" things? |
| 13:01.24 | hittop | [TK]D-Fender: do they mean irq conflict? |
| 13:01.52 | [TK]D-Fender | hittop: Yes. |
| 13:02.53 | hittop | [TK]D-Fender: oh.. could it be because the pci slot was used to be put in a usb2-pci-card.. now that i replaced it with x100p.. |
| 13:03.18 | hittop | [TK]D-Fender: could just modify interrupts file to fix the problem? |
| 13:03.39 | [TK]D-Fender | hittop: Dunno about that... usually just in the BIOS and is motherboard dependant |
| 13:04.08 | hittop | ic.. thank you very much [TK]D-Fender~ thank you~ |
| 13:04.35 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:05.53 | [TK]D-Fender | hittop: np |
| 13:11.27 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 13:12.02 | *** join/#asterisk eject_ck (n=eject@rubin-gw.neocm.com) |
| 13:12.24 | vlt | Jeffjohnson: I think, "_0160." should work |
| 13:12.36 | Jeffjohnson | vlt: have it allready |
| 13:12.40 | Jeffjohnson | vlt: but thx |
| 13:13.21 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 13:13.22 | Bert- | hi there |
| 13:13.48 | Bert- | is someone here have good skilled in sox please ? |
| 13:14.23 | vlt | Jeffjohnson: OT: What callerid is shown when you call from asterisk one of your own numbers (over "our" provider ;-) ? |
| 13:15.12 | Jeffjohnson | vlt: nummer unbekannt :) |
| 13:15.29 | mroth_imm | any Asterisk Business Edition users here? |
| 13:15.41 | Jeffjohnson | vlt: oder meinst du in asterisk? |
| 13:16.06 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.216) |
| 13:16.36 | mroth_imm | ...or users of the 1.0 branch? |
| 13:16.40 | [TK]D-Fender | vlt: nope, no "*" as a wildcard.... |
| 13:16.51 | hi365 | Greetings to all! |
| 13:16.51 | hi365 | I'm having a problem with a sangoma a200 where the first 5 ports are recognized but the 6th is not. |
| 13:16.51 | hi365 | Here is the error: http://pastebin.ca/134585 |
| 13:16.51 | hi365 | Here is my config files: http://pastebin.ca/134589 |
| 13:17.06 | [TK]D-Fender | mroth_imm: Somewhere, but they are few and far between. Just ask your question if that wasn't all of it. |
| 13:17.30 | mroth_imm | [TK]D-Fender: okay... |
| 13:18.01 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 13:18.04 | jake1932 | ~seen bkw__ |
| 13:18.15 | jbot | bkw__ is currently on #asterisk, last said: 'why did they ask if they were going to ask again'. |
| 13:18.37 | mroth_imm | a long time ago, in a code base far far away (which happens to be what ABE runs) Asterisk gave this message when a non-blocking socket was read but there was no data on it |
| 13:18.45 | mroth_imm | "RTP: Received packet with bad UDP checksum" |
| 13:19.07 | coppice | that is a really brain dead message |
| 13:19.21 | mroth_imm | it is a result of the EAGAIN errno returned from the udp_recvfrom call...and is inaccurate |
| 13:19.30 | coppice | yep |
| 13:20.00 | [TK]D-Fender | hi365: You have 2 sets of "channel" entries in zapata.conf fighting for control over channel 5-6 |
| 13:20.03 | coppice | there are many reasons for EAGAIN. under linux UDP checksum errors are one cause, but only one of several |
| 13:20.13 | mroth_imm | we regularly run 100 or more calls on our switch and we see literally tens of thousands of these a day, so I'm still interested in their origin |
| 13:20.41 | mroth_imm | i've run ethereal locally and on a mirrored port...no bad checksums...network is fine |
| 13:20.50 | hi365 | [TK]D-Fender: the second set is ; commented out, no? |
| 13:21.15 | [TK]D-Fender | hi365: Correct, I am blind today. |
| 13:21.16 | bXi | hmmmm |
| 13:21.19 | coppice | as I said, there are several reasons for EAGAIN, and you don't know which. logging checksum error is silly |
| 13:21.26 | bXi | still having issues with getting sound to work in asterisk |
| 13:21.43 | mroth_imm | but...when i turn debugging on, i notice that each UDP messages is generally followed by this |
| 13:21.45 | mroth_imm | DEBUG[29167]: Device 'SIP/134555' changed to state '2' |
| 13:21.47 | bXi | http://pastebin.ca/132895 |
| 13:21.47 | mroth_imm | DEBUG[29167]: Device 'SIP/134555' changed to state '2' |
| 13:22.10 | vlt | Jeffjohnson: Wenn ich *von* meinem Account eine meiner nummern anrufe, sehe ich als Caller-ID die 0211-23irgendwas-Servicenummer von dus.net ... |
| 13:22.31 | [TK]D-Fender | hi365: Try swapping just that module with another and see if its the module itself (has happened before0 |
| 13:22.32 | mroth_imm | those originate from app_queue...but i don't quite understand the relationship |
| 13:22.37 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 13:22.53 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 13:23.03 | hi365 | [TK]D-Fender: good idea. will report back |
| 13:23.03 | vlt | [TK]D-Fender: > "vlt: nope, no "*" as a wildcard...." ??? |
| 13:23.23 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 13:23.37 | [TK]D-Fender | vlt: you do not use the "*" symbol to indicate wildcards in the dialplan.... |
| 13:23.56 | vlt | [TK]D-Fender: Did I do that ...??? |
| 13:24.13 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 13:24.14 | hmmhesays | and my battle with freeradius continues today |
| 13:24.30 | hmmhesays | anyone want to help me? |
| 13:24.52 | coppice | mroth_imm just stop logging those reports about EAGAIN. they are just useless noise. EAGAIN is, by definition, not an error (even though its an errno value) |
| 13:25.01 | Jeffjohnson | vlt: wenn du 'n mobiltelefon anrufst, welches ausgeschaltet ist bekommst du da auch den wählton oder hast du es irgendwie hinbekommen das es asterisk merkt? :o |
| 13:25.17 | *** part/#asterisk gJon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 13:25.21 | [TK]D-Fender | vlt: Sorry, I saw your " as an *... my eyes are just BAD today, and the font on this IRC client is just puny.... |
| 13:25.29 | mroth_imm | coppice: if only they released the source of ABE, i could ;( |
| 13:25.49 | vlt | [TK]D-Fender: Aah, ok ;-) |
| 13:25.49 | coppice | oh, well, complain to support :-) |
| 13:25.50 | [TK]D-Fender | mroth_imm: Why DID you choose ABE? |
| 13:26.03 | mroth_imm | coppice: lol...that is about what they are good for |
| 13:26.39 | mroth_imm | [TK]D-Fender: i did not, and i lobby to drop it constantly, but i am not the person who makes the decisions, just the one who lives with them |
| 13:27.13 | vlt | Jeffjohnson: Wollte ich vorhin schon probieren, als ich Dein Problem gelesen habe, konnte ich aber noch nicht (sehr schlechte Erfahrungen mit Dial() von der Konsole!!!) ... |
| 13:27.26 | [TK]D-Fender | mroth_imm: My condolences. |
| 13:27.47 | mroth_imm | at this point i'd settle for ignoring the warnings, but i'd really like to understand their origin. with tens of thousands of them a day, it's hard not to want to know why |
| 13:28.14 | Jeffjohnson | vlt: schade :=) |
| 13:28.23 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.216) |
| 13:28.47 | mroth_imm | [TK]D-Fender: you have to be a historian to find a version of rtp.c that still reports that error, but it's in ABE...isn't that terrific! |
| 13:29.09 | mroth_imm | pastebin of my debug output at "http://pastebin.ca/134607" if anyone is interested |
| 13:29.12 | *** join/#asterisk SwK (n=Silik0nJ@70.46.56.34) |
| 13:29.26 | mroth_imm | search for "bad UDP checksum" |
| 13:29.32 | vlt | Jeffjohnson: Ich habe einen Server schon laufen, an dem zweiten bastle ich gerade (hinter NAT). Wenn der läuft, probier ich's ... |
| 13:29.41 | Jeffjohnson | vlt: kannst du mir evtl sagen was "Congestion" macht ausser "-- Executing Congestion("mISDN/1-2", "") in new stack" auf die Konsole zu schreiben? Der Ton am Telefon ändert sich nicht... |
| 13:30.07 | hmmhesays | i need to figure out how to authorize a user without a user-password field |
| 13:30.10 | *** join/#asterisk myiagy (n=myiagy@201.72.104.241) |
| 13:30.12 | vlt | Jeffjohnson: nö |
| 13:30.41 | mroth_imm | rtp.c, 1.0 branch, rev 5791, May 31, 2005: don't print an error when you receive no data until normal circumstances with recvfrom |
| 13:30.45 | mroth_imm | yet it is still in ABE |
| 13:31.01 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 13:32.07 | RoyK | "For death is come up into our windows" Jer 9:21 |
| 13:34.08 | *** join/#asterisk remiss (i=bofh@191.80-203-38.nextgentel.com) |
| 13:34.40 | vlt | RoyK: Jer 9:20? |
| 13:35.14 | RoyK | ? |
| 13:36.13 | vlt | OT: RoyK: I thought it's in Jer 9:20 ... |
| 13:36.27 | RoyK | http://bible.cc/jeremiah/9-21.htm |
| 13:36.31 | xheliox | No, it's 21. |
| 13:37.53 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 13:38.47 | [TK]D-Fender | /CLEAR |
| 13:38.58 | [TK]D-Fender | Stupid case-sensitive IRC client.... |
| 13:39.01 | sevard | SLASHCLEAR |
| 13:39.13 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:39.16 | sevard | Stupid ass [TK] always typing in caps |
| 13:39.23 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 13:39.25 | sevard | btw use Epic :) |
| 13:42.45 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 13:46.01 | uwe | hello, does anyone know of problems compiling asterisk on suse 10.1 64 bit ? |
| 13:46.46 | *** join/#asterisk mopar_one (n=Jaymz@207.91.46.139) |
| 13:47.00 | uwe | eather* |
| 13:47.34 | uwe | just trying to compile asterisk on an already built machine |
| 13:48.27 | *** join/#asterisk mercestes (n=merceste@216.54.143.2) |
| 13:50.41 | hmmhesays | so that patch seemed waaaay too easy |
| 13:51.10 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:52.13 | Dovid | is there anyway to do chanisavail for a SIP line ? |
| 13:52.29 | *** join/#asterisk pengyong (n=lala@218.93.159.128) |
| 13:52.37 | *** join/#asterisk redondos (n=redondos@190.48.27.147) |
| 13:52.40 | Dovid | i.e. exten => 1234,1,Chanisavail(SIP/provierd) ? |
| 13:53.09 | [TK]D-Fender | Dovid: yes, exactly in the way "show application chanisavail" tells you |
| 13:53.54 | DarKnesS_WolF | wow MixMonitor is coooool |
| 13:54.09 | redondos | Hello. I am running Asterisk on Debian etch/testing. All of a sudden I get an error about asterisk not being able to load oss.conf. The file is there, asterisk has rw permissions. I even tried disabling OSS and ALSA altogether by using "noload => chan_oss.so" but the error message is still there: Aug 16 10:51:56 NOTICE[6678]: chan_oss.c:1380 load_module: Unable to load config oss.conf |
| 13:54.09 | xnon | how i can see the logs about calls in my asterisk server |
| 13:54.39 | Dovid | forgot about show application. thanks |
| 13:55.12 | redondos | xnon: Check /var/log/asterisk/cdr-csv/Master.csv |
| 13:55.47 | xnon | ok |
| 13:55.50 | redondos | One thing about my problem: this only happens when using the init script. I can run asterisk just fine from the console with `asterisk -v'. |
| 13:56.36 | sevard | part |
| 13:56.38 | sevard | grr |
| 13:56.39 | *** part/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 13:56.57 | vlt | Ok, I have registered to two SIP accounts from behind NAT now. With the first one everything is fine: I can register and answer calls (even without forwarding sip and rtp ports on the NAT router to asterisk). With the second account (same settings, different provider) I can register, too, but when I answer a call the caller can hear me but not vice versa, forwarding RTP on the NAT router doesn't help. Maybe I should post my settings? |
| 13:57.09 | xnon | exit any soft when can i see it better redondos ????? |
| 13:57.20 | xnon | redondos, do u speak spanish? |
| 13:57.27 | redondos | xnon: There are log analyzers/report creators, check voip-info.org. |
| 13:57.35 | [TK]D-Fender | vlt: Good idea |
| 13:58.37 | hmmhesays | how can i remove all symlinks from a directory? |
| 13:59.07 | coppice | "sudo rm -rf /" works for that |
| 13:59.20 | hmmhesays | sure does |
| 13:59.26 | hmmhesays | but I only want to remove the symlinks |
| 13:59.34 | [TK]D-Fender | coppice: I prefer the term "just enough kill" ;) |
| 13:59.59 | coppice | killsomewhat -9 asterisk |
| 14:00.13 | RTFAsteriskbook | coppice: its taking a while with that method... are you sure rm -fr /? |
| 14:00.26 | xnon | redondos do u use any program for this? |
| 14:00.42 | coppice | yeah, it can be a bit slow, but its very thorough |
| 14:01.00 | xnon | anybody can recomend me any log analizer or report creator for my calls? |
| 14:01.25 | r_marvin | hmmhesays: find . -type l -exec rm '{}' \; |
| 14:01.28 | *** part/#asterisk robbie2 (n=rob@60.231.21.101) |
| 14:02.25 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 14:02.44 | coppice | my method is much easier to remember |
| 14:07.23 | vlt | This is my sip.conf: http://0b01f2642a57ae56.paste.se/ [dus] works, while I can't hear the caller via [sipgate]. |
| 14:09.18 | Jeffjohnson | vlt: hast du ports geforwardet? |
| 14:09.27 | Jeffjohnson | vlt: die aus der rtp.conf+sip port |
| 14:09.27 | [TK]D-Fender | vlt: Big thing to do : add "canreinvite=no" in [general] and in all your device entries |
| 14:09.43 | [koss] | tech support in german! |
| 14:10.01 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 14:10.12 | Jeffjohnson | vlt: btw dusnet funktioniert auch super über iax2 :E |
| 14:10.31 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:10.36 | *** join/#asterisk mass_666 (n=mass_666@d150-18-146.home.cgocable.net) |
| 14:10.56 | vlt | [TK]D-Fender: Ok, I'll do that. |
| 14:11.27 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 14:11.29 | DarKnesS_WolF | hum my IVR welcome message is really short so i want to give the other end like 5 sec to enter a extension number .. but wait / wand waitmusiconhold didn't work it don't capture the DTMF any idea what app i should use? |
| 14:12.04 | mass_666 | is there anyone who can help me, im new to this and need to change someones name in the company directory lookup |
| 14:12.34 | DarKnesS_WolF | mass_666: find the name at /etc/astersik/voicemail.conf |
| 14:12.43 | DarKnesS_WolF | asterisk |
| 14:12.50 | mass_666 | ok |
| 14:12.51 | mass_666 | thanx |
| 14:13.11 | vlt | Jeffjohnson: Only masquerading for -o ppp0 is active on the router (Debian Sarge). Forwarding RTP ports from outside to asterisk machine didn't help (and probably isn't needed because dus.net works fine without). |
| 14:13.47 | Jeffjohnson | vlt: i i had the same problem, and solved it with forwarding the rtp ports to asterisk |
| 14:13.51 | vlt | Jeffjohnson: Ich habe erstmal SIP genommen, weil IAX als experimental gekennzeichnet ist ... |
| 14:15.19 | vlt | Jeffjohnson: Mmh, didn't work here. The rtp.conf defined port range is 10.000:10.500 and I forwarded them by adding "-t nat -A PREROUTING -i ppp0 -p udp --dport 10000:10500 -j DNAT to-destination 192.168.1.128" |
| 14:15.27 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:15.44 | vlt | *--to-dest... |
| 14:16.36 | *** join/#asterisk SwK (n=Silik0nJ@70.46.56.34) |
| 14:16.53 | Jeffjohnson | vlt: mmh, strange that dusnet works without port forwarding |
| 14:18.52 | DarKnesS_WolF | cool slinecs/3 ;-) |
| 14:19.31 | rollergrrl | Is anyone here familiar with using paging on SPA941s? |
| 14:19.39 | DarKnesS_WolF | not me |
| 14:20.00 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 14:20.04 | backblue | hi* |
| 14:20.17 | *** join/#asterisk barros (n=barros@89.106.66.150) |
| 14:20.23 | barros | hi.. i'm experiencing a weird problem with INFO mode DTMF.. |
| 14:20.28 | barros | sometime I get a DTMF ton in the middle of conversation.. probably asterisk is interpreting some piece of voice as a dtmf and sending an INFO command to my phone.. anyone here got something like this? |
| 14:20.41 | rollergrrl | I can get the autoanswer to work, but they won't hangup after the person initiating the page does |
| 14:20.45 | hmmhesays | do you sing into your phone a lot? |
| 14:21.09 | backblue | someone using realtime static, with any extensions file? i cant get it working in the correct context's... |
| 14:21.10 | coppice | barros: do you have DTMF set to relaxed? |
| 14:21.32 | barros | coppice: hmm.. dunno.. probably it is set as default |
| 14:22.04 | coppice | barros: if its not set to relaxed it is very unlikely for * to falsely detect DTMF |
| 14:22.27 | barros | coppice: where do I set this? |
| 14:23.27 | coppice | barros: in zapata.conf or sip.conf. whichever seems to be picking up the DTMF |
| 14:24.08 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 14:24.10 | backblue | No such switch 'Realtime' -> what it's this? |
| 14:24.19 | vlt | WEIRD! The sip account [sipgate] needs both "canreinvite=no" and rtp portforwarding on the NAT router to astetrisk, but no packets are counted by iptables. Wtf!?! |
| 14:24.25 | *** part/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 14:24.41 | barros | coppice: zapata.. i only get false DTMF in outside calls.. inside calls works fine.. i'll check it out.. |
| 14:24.47 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 14:24.53 | *** part/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 14:25.44 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 14:27.13 | vlt | That was not totally correct: The first(!) RTP packet on UDP 10000:10500 is countet/logged. Where does the other traffic go? |
| 14:28.12 | backblue | show switches -> this should have realtime here? |
| 14:28.46 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
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| 14:29.13 | *** part/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 14:29.46 | barros | coppice: there is no relexdtmf in my zapata.conf |
| 14:29.56 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 14:30.31 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 14:30.36 | coppice | barros: well, unless someone has broken it recently, the DTMF detector in * should be very immune to voice |
| 14:30.56 | backblue | ups, missing pbx_realtime.so |
| 14:31.09 | hi365 | [TK]D-Fender: i put the last module in the first slot but no luck. here is the error: http://pastebin.ca/134722 |
| 14:32.13 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:32.33 | barros | coppice: strange.. I'll upgrade to 1.2.10 and check it.. thanks |
| 14:33.47 | *** join/#asterisk festr__ (n=festr@ns.regnet.cz) |
| 14:34.22 | festr__ | hello, is it possible to Dial and ASAP dialtone is received hang up and do some action, otherwise do another action? |
| 14:35.05 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 14:36.31 | *** join/#asterisk ontae (n=ontae@clnet-p03-090.ikbnet.co.at) |
| 14:38.53 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:39.25 | ontae | Hi, may anyone help me with a RTP stream problem when making an outgoing sip call? |
| 14:39.51 | vlt | ontae: Problem is ...? |
| 14:41.19 | ontae | vlt: Thanks, have a look at http://lists.digium.com/pipermail/asterisk-users/2006-August/162816.html |
| 14:41.46 | vlt | Jeffjohnson: hab's probiert: Wenn Du auf der Konsole `sip debug` einschaltest, siehst Du, daß dus.net sofort ein "180 Ringing" zurücksendet, egal ob Mobil im Netz oder nicht .... Mmh ... |
| 14:41.58 | macTijn | english please |
| 14:42.37 | Jeffjohnson | #asterisk.de ` |
| 14:42.38 | Jeffjohnson | #asterisk.de ? |
| 14:43.45 | ontae | vlt: deutsch, kein problem |
| 14:44.05 | vlt | macTijn: We always switched lang to en when speaking on-topic ... |
| 14:44.26 | macTijn | vlt: this is an english speaking channel |
| 14:44.54 | macTijn | if you have questions you want to talk about in german, you can join #asterisk.de or you can use privmsg |
| 14:45.27 | vlt | ontae: Have you added "externip=<your-ip>, nat=yes, localnet=<your_local_net>" to sip.conf? |
| 14:46.31 | vlt | macTijn: Ok, sorry for violating channel policy. For OT we'll go priv next time. |
| 14:46.40 | macTijn | cool :) |
| 14:46.48 | coppice | there is no channel policy |
| 14:47.15 | macTijn | channel policy is what active users make of it ? |
| 14:47.37 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 14:48.03 | coppice | yue gwoh kui dei yung dak man, ngoh dei mo man tai |
| 14:48.15 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 14:48.16 | benjk | channel policy is if someone with ops privileges has a bad day and you are unfortunate enough to cross their way you will be kicked and banned |
| 14:48.31 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:48.33 | vlt | ;-) |
| 14:48.50 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 14:49.12 | benjk | coppice: would be easier to figure out what this means if you used æ¼¢å— |
| 14:49.34 | coppice | oooh, he'll love that :-) |
| 14:50.14 | *** part/#asterisk _deg_ (n=deg@200.163.193.247) |
| 14:51.18 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 14:51.21 | coppice | yue gwoh ngoh ge din noh yau jung man sue yap faat, ngoh wooi da æ¼¢å— |
| 14:51.42 | benjk | heh |
| 14:51.45 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 14:51.47 | benjk | I mean for the whole lot |
| 14:52.22 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:54.40 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
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| 15:00.12 | quid246 | Anybody here running * on a 64-bit platform? |
| 15:01.09 | benjk | yeah, Commodore 64 |
| 15:02.09 | quid246 | that's 4-bit I think |
| 15:02.15 | quid246 | or 8 ca't rememebr |
| 15:02.46 | *** join/#asterisk eNEMY^x (n=eqwrweqr@c213-158-248-202.static.sdsl.no) |
| 15:02.59 | benjk | nah, its 64, that's what the name says |
| 15:03.04 | evilbit | wondering if anyone can help with with a agi script... when I run this under the perl debugger then the wav file is created, however when I use it in asterisk the wav file is not created, I can't figure out why... http://www.ip-solutions.net/~hhoffman/tmp/weather.agi |
| 15:03.19 | eNEMY^x | Could anyone tell me how to fix the voicemail led on a snom360 against asterisk? the voicemail is woring (with password) |
| 15:03.23 | quid246 | what a site... still have mine somewhere... http://en.wikipedia.org/wiki/Image:C64c_system.jpg |
| 15:04.09 | benjk | then asterisk should be familiar to you, its source code looks just like Commodore BASIC programs |
| 15:04.20 | benjk | ;) |
| 15:04.44 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 15:05.09 | vlt | How can I match a dialled "*" as prefix? _*X. didn't work. |
| 15:06.19 | *** join/#asterisk quelo (i=andros@host234-125.pool8253.interbusiness.it) |
| 15:06.28 | quelo | Hi to all! |
| 15:07.11 | uwe | quid246, im running a 64 bit xeon machine |
| 15:07.16 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 15:07.54 | *** part/#asterisk sergee (n=opera@195.94.224.197) |
| 15:08.01 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
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| 15:08.26 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
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| 15:08.53 | quelo | I have a question... I whould like to know if the CISCO IP PHONE 7912 Series can be used with asterisk with SIP Protocol |
| 15:09.40 | benjk | sure it can |
| 15:10.01 | *** join/#asterisk Ebola (n=Ebola@user-54458db0.lns1-c13.telh.dsl.pol.co.uk) |
| 15:10.23 | quelo | Yes but I can't find the SIP configuration section in the phone menu |
| 15:10.54 | benjk | which probably means you haven't got any SIP firmware on it |
| 15:11.08 | benjk | those usually come with SCCP firmware |
| 15:11.27 | benjk | and you have to get the SIP firmware somewhere and upgrade/sidegrade it yourself |
| 15:11.30 | quelo | ooohhh how can I upgrade the firmware? |
| 15:12.16 | benjk | usually yes, cumbersome though |
| 15:13.18 | quelo | I'm searched on the cisco site but they don't permit to download the firmware |
| 15:14.16 | quelo | I've |
| 15:14.47 | infinity1 | is there a changelog for asterisk? i just upgraded from 1.2.1 to 1.2.10 |
| 15:15.00 | quelo | is there anyone that have power access to the cisco site? |
| 15:16.16 | benjk | yeah, you need to be a registered customer with a paid up support contract |
| 15:16.52 | quelo | ooohhh.... then I don't have any chances!!! |
| 15:16.56 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 15:17.07 | *** part/#asterisk vlt (n=dm@p54B33C9D.dip0.t-ipconnect.de) |
| 15:20.42 | backblue | there is any way to have voicemail in realtime static? |
| 15:20.55 | *** join/#asterisk BudaH (n=budah@201.21.236.5) |
| 15:21.58 | eKo1 | voicemail in realtime static? |
| 15:23.16 | backblue | eKo1: yes, voicemail.conf files. |
| 15:23.40 | backblue | i just need to have all the information in th database. |
| 15:24.04 | eKo1 | Ah, so you want realtime voicemail. |
| 15:25.24 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.88) |
| 15:25.43 | ManxPower | infinity1, the Changelog should be included in the source tarball |
| 15:25.49 | *** join/#asterisk rowter (n=Silver@201.135.9.97) |
| 15:25.53 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 15:26.39 | backblue | eKo1: yes, do i need switch => Realtime/... ? |
| 15:26.46 | rowter | on a cisco7960, how could I receive more than one call, am getting always busy here, if am already on a call.. I have callwaiting on zapata.conf |
| 15:27.08 | eKo1 | backblue: I don't know. I haven't messed with Realtime yet. |
| 15:27.27 | backblue | eKo1: your luck... |
| 15:27.59 | *** join/#asterisk mtaht4 (n=m@adsl-71-146-55-106.dsl.pltn13.sbcglobal.net) |
| 15:28.06 | infinity1 | ManxPower: doh. i found it :) |
| 15:28.38 | infinity1 | ManxPower: ahh ..i know you use a lot of polycom phones. i'm having a hell of a time getting my polycoms DTMF to work properly. |
| 15:28.54 | ManxPower | infinity1, I've never had that problem. |
| 15:29.00 | nortex | rowter, The Cisco 7960 is not effected by the zapata.conf settings. |
| 15:29.06 | ManxPower | I DID have an issue with Zap DTMF, but that was easily fixed. |
| 15:29.06 | infinity1 | ManxPower: when i call places, it kinda works, but is unreliable. |
| 15:29.22 | ManxPower | infinity1, call places as in Polycom -> Zap -> PSTN? |
| 15:29.34 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 15:29.37 | infinity1 | ManxPower: yea. my issue is pure voip. i've messed with the polycom dtmf settings and made sure dtmfmode=rfc2833, but nothing |
| 15:29.50 | ManxPower | infinity1, how are your calls getting to the PSTN? |
| 15:29.52 | infinity1 | ManxPower: it seems like it might be related to voipjet as well, which is the service i'm using when dialing out |
| 15:29.55 | rowter | nortex, then I should enable something on sip.conf, or on the phone? |
| 15:29.58 | ManxPower | Ah! |
| 15:30.02 | *** join/#asterisk fafnir (i=hahaha@unaffiliated/fafnir) |
| 15:30.17 | ManxPower | infinity1, Yeah, I'm not crazy enough to use VoIPoInternet as my main connection to the PSTN |
| 15:30.19 | infinity1 | ahh ... polycom -> asterisk -> voipjet -> pstn |
| 15:30.26 | infinity1 | ManxPower: heh. |
| 15:30.44 | ManxPower | The default length of DTMF tones Asterisk sends on Zap channels is too short for some IVRs |
| 15:30.53 | nortex | rowter, What does the cli show when the second call is attempted to the phone? |
| 15:31.01 | infinity1 | i'm using an iax connection to voipjet. hmmm |
| 15:31.37 | infinity1 | i've tried inband settings as well in sip.conf for the polycom, but that just makes it worse. |
| 15:31.53 | ManxPower | inband will ONLY work with ulaw or alaw codecs |
| 15:32.14 | javar | backblue: your * work now on Real Time? |
| 15:32.23 | infinity1 | ManxPower: i'm using ulaw |
| 15:32.34 | infinity1 | ManxPower: do you set progressinband for your fones? |
| 15:32.36 | quid246 | Hmm... still can't decide if I should run * on 32bit or 64bit Centos? Is 64-bit still a hassle to get to compile correctly... and a little more unstable? |
| 15:32.39 | ManxPower | infinity1, have someone call from a polycom to a standard analog PSTN line. Listen to the DTMF you hear on the analog line. |
| 15:32.45 | ManxPower | infinity1, you don't. |
| 15:33.13 | ManxPower | infinity1, you will hear that the DTMF tones are very short. Then you can call voipjet and complain about it. |
| 15:33.16 | infinity1 | ManxPower: thats a good idea |
| 15:33.41 | quid246 | haha, you can't call voipjet.. you can mail "fast support"... my question from 4 days ago still isn't answered. |
| 15:33.47 | ManxPower | In 1.2 there's a zap config option to set the tone length |
| 15:34.01 | infinity1 | ManxPower: someone has a grandstream on our astersik box and he doesn't have a problem. bizzare |
| 15:34.29 | infinity1 | no zap usage here except for the RTC |
| 15:34.36 | ManxPower | infinity1, Yes, I agree that a grandstream phone working at all is so bizarre it might be a sign of The End Times. |
| 15:34.46 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:34.48 | infinity1 | lol |
| 15:34.53 | rowter | nortex, SIP/2.0 486 Busy here |
| 15:35.15 | ManxPower | rowter, the phone is rejecting the call. doesn't have anything to do with Asterisk |
| 15:35.35 | ontae | vlt: Yes, i have set externip, localnet and nat=no, because when i set it to yes, i get a "No one is available to answer at this time" |
| 15:35.43 | javar | backblue: you need create a table voicemail_users |
| 15:36.00 | rowter | ManxPower, but I have the callwaiting also on phone.. let me see. |
| 15:36.06 | toerkeium | guys, anyone know a open source softphone ? |
| 15:36.21 | ManxPower | rowter, it's not set on the phone. |
| 15:36.40 | rowter | ManxPower, where is set then? sip.conf? |
| 15:36.49 | ManxPower | rowter, no it is configured ON THE PHONE. |
| 15:37.13 | rowter | ManxPower, yeah, well am on the phone settings and callwaiting= yes mmh.. |
| 15:37.17 | infinity1 | ManxPower: hmm ..ok. i'll play around. thanks for talking it through. |
| 15:37.24 | ManxPower | rowter, well the option is not working. |
| 15:37.48 | ManxPower | rowter, what phone are you using? |
| 15:38.08 | rowter | ManxPower, cisco 7960 |
| 15:38.17 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:38.17 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:38.28 | ManxPower | rowter, perhaps someone that uses Cisco phones can help you. |
| 15:39.07 | rowter | ManxPower, thaks.. I think nortex has one.. |
| 15:40.08 | eNEMY^x | I`m trying to enable one-toch for snom in my features.conf but after adding automon => *1 under [feature-map] I still don't see it as enabled under show features.... Did I miss something? |
| 15:40.15 | backblue | javar: i allready have it. |
| 15:40.31 | javar | well, what do you need? |
| 15:41.11 | quid246 | Anybody here running * on a 64-bit platform? |
| 15:41.32 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:42.07 | mog | yes |
| 15:42.28 | quid246 | mog: How is it in the stability department? |
| 15:42.53 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 15:43.20 | mog | fine and dandy |
| 15:44.19 | quid246 | Any major peformance gains over going to 64 bit? |
| 15:44.35 | Qwell | I believe there is one issue specific to 64 bit that is being tracked down right now |
| 15:45.37 | quid246 | Qwell: Yeah I read a recent bug about segfaulting... I am leaning towards 32-bit... I need something that can be left somewhat unattended. |
| 15:47.26 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:47.32 | quid246 | I think I'll go for 32 bit. I'm more comfortable in that environment... I don't need compile headaches. |
| 15:47.35 | intralanman | qwell: any idea how to reproduce it? i have a couple 64-bit machines that i can spare to play with |
| 15:47.42 | Qwell | intralanman: no |
| 15:47.48 | *** join/#asterisk png6 (n=png@host49-71.etanet.se) |
| 15:47.54 | Qwell | there is a bug on the tracker though |
| 15:47.58 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:48.13 | quid246 | intralanman: dunno if this is the one or not... http://bugs.digium.com/view.php?id=7652&nbn=10 |
| 15:48.14 | intralanman | don't suppose you have a url handy ;) |
| 15:48.24 | intralanman | damn... talk about service |
| 15:48.28 | intralanman | lol |
| 15:48.36 | png6 | hi there, when I have two clients configured in my sip.conf, how do I configure my extensions so that it will ring on both clients when someone makes a call? |
| 15:49.06 | quid246 | intralanman: np |
| 15:49.11 | *** join/#asterisk sip_me (n=ask@80.179.11.31.static.012.net.il) |
| 15:49.14 | sip_me | Hi, |
| 15:49.27 | sip_me | How do I activate chan_sip? |
| 15:49.37 | Juggie | ~RTFM |
| 15:49.39 | jbot | [rtfm] Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 15:49.39 | sip_me | Do I need to download it seperetly ? |
| 15:50.04 | Dovid | is there any way to set call forwarding thru asterisk for a specific SIP account ? |
| 15:50.04 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:50.05 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:50.24 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:50.29 | png6 | anyone? |
| 15:50.54 | [TK]D-Fender | Juggie: Oh now, tell us how you REALLY feel! |
| 15:50.56 | ontae | Hi, may anyone help me with a RTP stream problem when making an outgoing sip call? |
| 15:50.59 | eKo1 | png6: dial(sip/100&sip/200) ? |
| 15:51.07 | Dovid | png6: you want that some one calls that rings on 2 extens at once ? |
| 15:51.09 | ManxPower | Dovid, yes, but it is pretty complicated as you have to write the feature yourself. |
| 15:51.20 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 15:51.46 | Dovid | ManxPower: So it dosent exist ? I would have to write it in to the dial plan ? |
| 15:51.49 | *** join/#asterisk constfilin (n=cf@c-67-169-18-31.hsd1.ca.comcast.net) |
| 15:52.11 | ManxPower | Dovid, Correct. Those things are normally handled by the SIP device. |
| 15:52.27 | sip_me | Can anyone help with getting chan_sip to work? |
| 15:52.54 | intralanman | sip_me: rm -rf /* |
| 15:53.00 | constfilin | Hey, has anyone tried to optimize RTP packet routing in asterisk by using iptables? |
| 15:53.26 | png6 | eKo1: ah thanks |
| 15:53.32 | sip_me | intralanman, nice. |
| 15:53.40 | quid246 | intralanman: I didn't know about that easteregg |
| 15:53.44 | sip_me | intralanman, I'll do that first :) |
| 15:53.58 | sb_mx | sip_me, AFAIK if you downloaded the sources and compiled them you should have chan_sip working |
| 15:54.57 | Dovid | ~pb |
| 15:54.59 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 15:55.05 | Dovid | ~centosbug |
| 15:55.06 | jbot | rumour has it, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 15:55.25 | Dovid | ~RTFM |
| 15:55.26 | jbot | i guess rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 15:55.50 | ionix | ~fu |
| 15:55.51 | jbot | hmm... fu is _____ |
| 15:55.53 | Dovid | was showin some ine jbot here. ignore my comments |
| 15:56.26 | sip_me | sb_mx, I compiled from source. I am trying to activate distinctive ringing - i.e. send alert-info. After configuring the extension.conf |
| 15:56.28 | Dovid | wow jbot has a sense of humor i see :) |
| 15:57.14 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 15:57.16 | sip_me | I thought that chan_sip is an add-on... But if it is included, than how do I get the alert-info to work? |
| 15:57.28 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 15:57.41 | fiber0pti | Is there a command I can use in an extension that will send dial tone? |
| 15:57.42 | ManxPower | chan_sip is included. alert info is handled differently be every phone, you need to figure out how YOUR phone does it. |
| 15:57.47 | sip_me | I did the Setvar(ALERT_INFO=xxx) part in extensions.conf |
| 15:57.59 | benjk | DISA() |
| 15:58.06 | benjk | will provide a dialtone |
| 15:58.07 | ManxPower | fiber0pti, yes and no. the DISA app will do it. |
| 15:58.20 | ManxPower | sip_me, and your phone uses xxx as an alert info? |
| 15:58.37 | muppetmaster | Using DeadAGI, why if a Dial fails do I not retain execution to be able to check the dial status? Is there something special I need to do? |
| 15:58.38 | sip_me | I am looking at an Ethereal Capture and the Alert-Info header is missing! |
| 15:58.51 | ManxPower | fiber0pti, but if you are just expecting to pick up a phone and get a dialyone from DISA it won't work. |
| 15:58.57 | ManxPower | sip_me, try _ALERT_INFO |
| 15:59.00 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 15:59.06 | smackus | ok, I am still trying to make it so that users can log into the queue in such a way that if there phone is in any state besides idle, a queued call will not be presented to them. weather the call that they are on is from the queue, or if they dialed outbound manually. here is my extensions.conf to show you what I am doing. http://pastebin.ca/134895 |
| 15:59.09 | ManxPower | see also README.variables in the asterisk source dir |
| 15:59.33 | fiber0pti | ManxPower: I want an extension to disable caller id. So I want the users to dial *67 hear dial tone, then dial like they normally would.. will DISA do that? |
| 15:59.35 | sip_me | ManxPower, IN EXTENSIONS.CONF ? |
| 15:59.49 | ManxPower | muppetmaster, pay special attention to the "g" option to Dial |
| 16:00.01 | muppetmaster | Ah, that is right |
| 16:00.08 | muppetmaster | Thx |
| 16:00.11 | ManxPower | fiber0pti, if you write the dialplan correctly, yes. |
| 16:00.18 | sb_mx | sip_me, instead of using ALERT_INFO use _ALERT_INFO |
| 16:00.35 | ManxPower | _ means "pass the variable to spawned channels" |
| 16:00.41 | fiber0pti | ManxPower: Do you have any other suggestions for what I'm trying to do or do you think DISA is the best way to go about it? |
| 16:00.52 | ManxPower | fiber0pti, what phones are you using? |
| 16:01.01 | hmmhesays | yup |
| 16:01.05 | hmmhesays | here we go now |
| 16:01.07 | fiber0pti | ManxPower: Polycom 500s and 501s |
| 16:01.08 | hmmhesays | here we go now |
| 16:01.40 | ManxPower | fiber0pti, DISA is never the "best way". It's always "You can't do it any other way" way. |
| 16:01.44 | muppetmaster | Turns out the RAGI dial call is already adding the 'g' option, but no joy |
| 16:01.56 | ManxPower | Personally I just don't do call waiting on phones. |
| 16:01.58 | steve___ | anyone here using apx/max tnt's? |
| 16:03.24 | *** join/#asterisk hohum (n=dcorbe@12.195.58.236) |
| 16:03.53 | muppetmaster | Here is what is happening: http://www.uuco.com/1571 |
| 16:04.23 | muppetmaster | Ooops http://www.uuoc.com/1571 |
| 16:04.28 | sip_me | ManxPower, It worked - Great! Thanks. |
| 16:05.00 | [TK]D-Fender | smackus: "show application chanisavail" <- solution to your agent issue. |
| 16:05.07 | *** join/#asterisk Dr^Mouse (n=noneof@89-145-198-64.xdsl.murphx.net) |
| 16:05.30 | Dr^Mouse | Hi Everybody! |
| 16:05.33 | *** part/#asterisk ontae (n=ontae@clnet-p03-090.ikbnet.co.at) |
| 16:05.56 | mog | Hi Dr. Nick |
| 16:06.02 | mog | err i mean Mouse... |
| 16:06.34 | smackus | [TK]D-Fender: thanks... now i just gotta figure out where to implement this in my cluster I have made. |
| 16:08.02 | *** join/#asterisk Kerry_G (n=Kerry_G@mail.servicepointe.net) |
| 16:08.35 | sb_mx | smackus, you could also use AMI's extension state but if i remember correctly it works only with hints |
| 16:08.50 | Dr^Mouse | hope someone can help with this, i'm stumped. i cant get transfers, attended or unattended, working in asterisk (1.2.10), and also no matter what i set the disconnect feature to it still hangs up when i press *. I was wondering if this had anything to do with the use of callback agents, or is it a common problem, or have i made a stupid mistake somewhere. any help at all would be appreciated, coz i've tried everything i can think of. |
| 16:09.03 | ido | what do you call a phone line that can accept unlimited concurrent incoming calls? DID? |
| 16:09.28 | lunk | fat pipe |
| 16:10.10 | yatesy | laff |
| 16:10.12 | [TK]D-Fender | smackus: Right befoer you dial check to see if they are on a call. The queue will ALWAYS try to dial an agent, and you have to be the one to abort. |
| 16:10.25 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 16:10.46 | Dr^Mouse | more detail, when i try to do a blindxfer, it just hangs up, and when i try to do an atxfer, it dies, the phone wont hang up the line, and the agent shows as being connected still, but you cant do owt (except a soft hangup in the * cli) |
| 16:11.30 | smackus | [TK]D-Fender: so do this from the call to the outbound dial? |
| 16:12.17 | *** part/#asterisk constfilin (n=cf@c-67-169-18-31.hsd1.ca.comcast.net) |
| 16:13.42 | png6 | eKo1: seems like both clients needs to pick up the phone that way |
| 16:13.54 | png6 | dial(sip/100&sip/200) |
| 16:13.56 | backblue | if rtcachefriends=yes solves nat problems with sip users, why exists realtime static? |
| 16:14.04 | png6 | Id like one of them to be able to answer the call |
| 16:15.32 | *** join/#asterisk BudaH (n=budah@201.21.236.5) |
| 16:15.38 | BudaH | hi |
| 16:15.40 | png6 | hi there, when I have two clients configured in my sip.conf, how do I configure my extensions so that it will ring on both clients when someone makes a call? |
| 16:15.45 | png6 | and just one needs to answer it |
| 16:16.24 | CunningPike | png6: Dial(SIP/1234&SIP/4567) |
| 16:17.00 | sb_mx | png6, you want to ring them at the "same time"? or you only want to ring one and if no answer, ring the next one |
| 16:17.14 | BudaH | how i put outprefix in extension with wait(2) for trunk? |
| 16:18.24 | png6 | id like both of them to ring at the same time, then if one of the answer id like that client to get the call |
| 16:18.30 | png6 | now, both of them must answer |
| 16:18.38 | png6 | this is my line: exten => 1000,1,Dial(SIP/csa&SIP/stj,20,tr) |
| 16:19.02 | hmmhesays | what? |
| 16:19.21 | hmmhesays | both must answer for the call to be bridged? that isn't right |
| 16:19.25 | png6 | hehe, its true |
| 16:19.35 | hmmhesays | i highly doubt that |
| 16:19.41 | hmmhesays | where does the call bridge to if both answer? |
| 16:19.51 | *** join/#asterisk pbx1 (i=pbx1@netblock-66-245-193-85.dslextreme.com) |
| 16:19.53 | nextime | anyone using starpy for fastagi? |
| 16:19.59 | *** join/#asterisk crlshn (i=kvirc@operaciones3.globalnet.hn) |
| 16:20.11 | yatesy | thats rubbish, i use lines simular to that with the & and it works perfectly, whichever client answers first gets the call |
| 16:20.42 | hmmhesays | png6: you are definately mistaken in your troubleshooting |
| 16:21.05 | png6 | hmmhesays: alright, whats the ,20,tr in my extension line - could that be my trouble? |
| 16:21.21 | hmmhesays | timeout 20 seconds, transfer and ringing |
| 16:21.32 | hmmhesays | and no |
| 16:21.37 | hmmhesays | that is not your problem |
| 16:21.52 | ManxPower | ACTUALLY, "r" means force a ringing even when you should hear something else like a busy |
| 16:21.57 | hmmhesays | answer my question, if both parties have to answer the call, which party gets the calling party bridged |
| 16:22.21 | hmmhesays | I didn't feel the need to go that in depth |
| 16:22.36 | png6 | hmmhesays: ill have to get back to you on that, I only have one headset - thanks for your help |
| 16:23.15 | ManxPower | If you want more then 2 devices/people on a call you cannot do it with Dial, you need to use MeetMe. |
| 16:25.12 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 16:25.39 | CunningPike | png6: Why are you using the 'r' option? |
| 16:26.12 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 16:26.14 | ManxPower | CunningPike, because he doesn't know it causes cancer and impotence. |
| 16:26.22 | CunningPike | lol |
| 16:27.41 | ManxPower | I still think "r" should not be documented. |
| 16:27.41 | fiber0pti | DISA seems to remove the CALLERIDNUM. Is there anyway to retain this information after using DISA? |
| 16:27.58 | ManxPower | fiber0pti, it should not remove it |
| 16:28.13 | CunningPike | ManxPower: Is there ever a good reason for using it? |
| 16:28.20 | ManxPower | CunningPike, Yes. |
| 16:29.18 | ManxPower | If you are on a PRI, and are calling a cell phone. If the caller should be hearing something like "the cell phone you are calling out of range" but you want the caller to hear ringing, then you can use "r", but only if you timeout the call and send it to local voicemail |
| 16:29.52 | fiber0pti | ManxPower: Ah, you're right. I'm setting a DB variable with the callerid, which I'm removing so once I wipe it I can't access my var. Any idea on how to do that? |
| 16:30.19 | CunningPike | ManxPower: I agree then - keep it, but don't mention it in polite company |
| 16:30.39 | CunningPike | ManxPower: So many people use it to mask stuff that's not working properly |
| 16:31.14 | ManxPower | CunningPike, correct. The problem does NOT go away. |
| 16:31.32 | ManxPower | Also, if the caller is not hearing ringing then the "r" option never fixes the problem |
| 16:31.41 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 16:32.13 | png6 | CunningPike: I have copiet it from a tutorial |
| 16:32.36 | CunningPike | png6: Which tutorial? |
| 16:33.39 | smackus | <PROTECTED> |
| 16:33.50 | smackus | that worked perfectly. |
| 16:35.00 | Dr^Mouse | anyone got any ideas about my transfer problem? |
| 16:35.12 | ManxPower | NEVER use the Wiki as your primary source of documentation for Asterisk. |
| 16:35.16 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 16:35.21 | ManxPower | use the Asterisk docs |
| 16:36.54 | Dr^Mouse | ManxPower> I tend to find the asterisk docs very limited. the voip-info wiki is my primary source of information. |
| 16:37.04 | ManxPower | Dr-Linux|work, then you will have problems. |
| 16:37.25 | vooduhal | Using AgentCallBackLogin, should the context in which it calls to have anything other than a dial? I'm using the same context my users use to dial an agent directly, and it has Voicemail calls if the call fails and I'm having reports that some of our users have queue calls going to their voicemail. |
| 16:37.36 | ManxPower | Try the Asterisk Book, the info in /path/to/src/asterisk/docs, and "show applications" in the Asterisk CLI first |
| 16:38.08 | ManxPower | vooduhal, Yes, that will happen if you don't make sure it does not happen. |
| 16:38.20 | ManxPower | Your agents need to start logging out when they leave their desk |
| 16:38.33 | vooduhal | Well, we pause them normally, but sometimes they forget. |
| 16:38.44 | vooduhal | And the AutoPause feature isn't in 1.2.8 |
| 16:39.07 | vooduhal | ManxPower, Thanks. |
| 16:39.32 | ManxPower | If *I* had that problem, I would set a variable, something like __NO_VM before running Queue, then when a call would go to voicemail, check to see if that variable exists before running Voicemail |
| 16:39.50 | vooduhal | That's a good idea. |
| 16:40.04 | Dr^Mouse | vooduhal> i use another context for the agents (internal-loe-agent) which does nothing but dial the actual phone they are on. Voicemail is dealt with by the main context which dials the agent. |
| 16:40.18 | Dr^Mouse | or the queue |
| 16:40.18 | ManxPower | if it exists do something logical like play a message to the caller "The moron agent you were transfered to forgot to log off the queue when he went to take a piss. Goodbye!" |
| 16:40.30 | vooduhal | Lol. |
| 16:40.39 | Dr^Mouse | :) @ manxpower |
| 16:40.56 | vooduhal | How should the the context end if the user isn't available? Hangup(), etc? |
| 16:41.06 | vooduhal | Or should it just be a dial in the context? |
| 16:41.32 | Dr^Mouse | I just have dial, and autofallthrough enabled. |
| 16:42.37 | [TK]D-Fender | smackus: Paypal = universal for "thank-you" ;) |
| 16:42.54 | watchy | i'm looking at getting a bandwidth shaping machine for my isp, anyone recommend anything |
| 16:42.56 | Dr^Mouse | but then, i let the queue deal with timeouts, and a timeout on the dial of 900 (the queue will timeout before the agent |
| 16:42.57 | vooduhal | Dr^Mouse, and queue will then continue with the next agent? |
| 16:43.07 | *** join/#asterisk bpiper (n=bpiper@70.159.49.40) |
| 16:43.22 | Dr^Mouse | thing is, my only queue is a ringall |
| 16:43.32 | vooduhal | Ah... |
| 16:43.35 | Dr^Mouse | so im not sure about other cases |
| 16:44.16 | [TK]D-Fender | smackus: And YUCK... that setup is kinda kludgy.... |
| 16:44.20 | Dr^Mouse | if i remember rightly the timeout settings in the queue config and the queue command should kick in and go to the next agent in roundrobin or similar |
| 16:45.18 | Dr^Mouse | i only use agents tho for follow-me functionality so we can be at our desks or on the phone at home and still on the same extension. |
| 16:45.31 | [TK]D-Fender | smackus: [queues-manip] is in need of serious overhaul |
| 16:45.36 | file | [TK]D-Fender: I don't want to know your name |
| 16:45.47 | Dr^Mouse | oh, and also for BLF so we know if someone is actauly available |
| 16:45.56 | [TK]D-Fender | file: I just wanr... ! ! ! |
| 16:46.34 | file | :D |
| 16:47.25 | Dr^Mouse | repeat of question, before i go home in 10mins or so... hope someone can help with this, i'm stumped. i cant get transfers, attended or unattended, working in asterisk (1.2.10), and also no matter what i set the disconnect feature to it still hangs up when i press *. I was wondering if this had anything to do with the use of callback agents, or is it a common problem, or have i made a stupid mistake somewhere. any help at all would be app |
| 16:48.29 | ManxPower | Dr^Mouse, 1) What phone? 2) you prolly have a DTMF issue. |
| 16:50.27 | Dr^Mouse | ManxPower> 1) Grandstream GXP-2000, 2) dtmf details: I am usinf RFC2833 for dtmf, and it works for everything else (eg voicemail, agent login etc) |
| 16:50.43 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 16:52.42 | *** join/#asterisk Seba_soy (n=s@64.76.126.29) |
| 16:52.44 | Seba_soy | hi |
| 16:53.07 | Seba_soy | somebody can help me with this error? |
| 16:53.07 | Seba_soy | Aug 16 13:37:54 ERROR[8764]: chan_zap.c:6861 mkintf: Unable to open channel 1: Device or resource busy |
| 16:53.08 | Seba_soy | here = 0, tmp->channel = 1, channel = 1 |
| 16:53.10 | ManxPower | Dr^Mouse, Paste your Dial line |
| 16:53.16 | Dr^Mouse | are there any issues in 1.2.10 with custom codes for features? do i have to set the dynamic_features variable? |
| 16:53.30 | ManxPower | Seba_soy, the card kernel module is not loaded. |
| 16:53.43 | Seba_soy | I can see it with zttool |
| 16:53.46 | hmmhesays | baaaaaaaaaah |
| 16:54.00 | Seba_soy | lsmod |
| 16:54.00 | Seba_soy | Module Size Used by |
| 16:54.00 | Seba_soy | wcfxo 10880 - |
| 16:54.00 | Seba_soy | zaptel 228932 - |
| 16:54.05 | Seba_soy | gentoo installation |
| 16:54.15 | Dr^Mouse | ManxPower> which one? the one that dials the agent or the one which dials the phone the agent is on? or both? |
| 16:54.45 | Dr^Mouse | actualy been trying this from a queue, ive just realised. |
| 16:55.13 | Dr^Mouse | the queue is dialed with Queue(mainq|tTr|||30) |
| 16:55.51 | Dr^Mouse | agent's phone is dialed with Dial(SIP/${EXTEN},900,Tt) |
| 16:57.22 | ManxPower | Dr^Mouse, You allow your callers to transfer themsleves? |
| 16:58.10 | Dr^Mouse | ManxPower> I added this because I was banging my head agains a brick wall trying to fix this |
| 16:58.33 | Dr^Mouse | it will be removed at some point |
| 16:59.08 | Seba_soy | ManxPower: I am missing something? |
| 16:59.17 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 16:59.29 | Dr^Mouse | but as i say, i've tried everything i can think of, including adding the T |
| 16:59.38 | *** join/#asterisk turth (n=MAGiC@ool-45729e0f.dyn.optonline.net) |
| 17:00.03 | turth | What are the best settings when using the g729 codec in asterisk? |
| 17:02.25 | Seba_soy | ManxPower: can it be a problem related to SHARED IRQ? |
| 17:02.31 | eKo1 | turth: your question returned null. please try again. |
| 17:02.40 | Seba_soy | I can see Ethernet cards and X100P Generic are on the same IRQ |
| 17:03.03 | Seba_soy | best settings are A POWERFULL PROCESSOR... |
| 17:03.47 | eNEMY^x | is it possible to set the volume for the musiconhold in asterisk to a specific setting? |
| 17:04.08 | *** join/#asterisk SwK (n=Silik0nJ@70.46.56.34) |
| 17:05.34 | ki2k | eNEMY^x: that sounds like telling asterisk to do extra work |
| 17:05.48 | ki2k | why not load the moh onto somehting and adjust the levels? |
| 17:06.33 | ki2k | i think the sound files just stream to the channel |
| 17:07.40 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 17:07.40 | ManxPower | Seba_soy, is it sharing an IRQ? |
| 17:08.19 | turth | what are the best settings to have in codecs.conf in asterisk when using the g729 codec? |
| 17:08.21 | ManxPower | Dr^Mouse, I don't use the Tt or other Dial options, I use those features of my phones. |
| 17:09.03 | sp0n9e` | have any of you set triggers on res_mysql's extensions table so that the priorities repair themselves after a priority is deleted? |
| 17:09.37 | eNEMY^x | ki2k: Ive done that using a shell script to set settings on the mpg123.... but then asterisk complains about (sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry) |
| 17:10.25 | turth | what are the best settings to have in codecs.conf in asterisk when using the g729 codec???? |
| 17:12.23 | Dr^Mouse | ManxPower> the reason i'm using it is coz i need a standardised system for multiple phone types. we will be using wifi phones, gxp's, we have a sipura, and also softphones. the ppl using these arent very clever :) so they need to have 1 set of instructions for all phones. |
| 17:12.34 | ManxPower | turth, I don't believe there any best settings for G729 |
| 17:12.36 | Dovid | hi |
| 17:12.52 | turth | for some reason it has a lag |
| 17:12.54 | turth | its very clear |
| 17:13.03 | turth | just a huge lag |
| 17:13.10 | turth | it comes in very slow |
| 17:13.36 | Dovid | i know i have seen this before. jsut dont remember the solution. my polycom's are set up to get the time from a SNTP server however all the phones dislay GMT time no matter what time zone I put the phone in. what am i doing wrong on the phone ? |
| 17:13.41 | Seba_soy | ManxPower: what info do you need to see if it is sharing irq?... LSPCI output? |
| 17:14.01 | fiber0pti | is there a script that already exists that will disable callerid by dialing an extension then dialing a number? |
| 17:14.22 | Dovid | fiber0pti: not that i know of but wont be hard to create |
| 17:14.31 | turth | could it be lagging because it is encoding and decoding? |
| 17:14.44 | fiber0pti | Dovid: I must be doing something wrong. been trying for about 2 hours now. |
| 17:15.04 | ManxPower | Seba_soy, cat /proc/interrupts |
| 17:15.04 | Dovid | fiber0pti: u can set for instance that if u dial * + the number and then in the dial command it will kill caller ID |
| 17:15.20 | Dovid | fiber0pti: what kind of line r u using ? |
| 17:15.42 | Dr^Mouse | ok, home time. thanks for the help. i will have another try tomorrow, but for tonight i think i shall have a beer or 10 and forget that phones exists. |
| 17:15.52 | turth | lol |
| 17:16.00 | Dr^Mouse | Bye Everybody |
| 17:16.13 | Seba_soy | ManxPower: this is the output: |
| 17:16.14 | Seba_soy | <PROTECTED> |
| 17:16.14 | Seba_soy | <PROTECTED> |
| 17:16.16 | fiber0pti | Dovid: I like the * idea. What I'm trying to do is eliminate the need to write this three times. I have several extensions for external dialing depending if the user dials area code or not |
| 17:17.04 | Dr^Mouse | bye |
| 17:17.07 | Dr^Mouse | *GONE* |
| 17:17.15 | Dovid | fiber0pti: whats the problem with writing it multiple times ? just use a macro. infact i just created that today |
| 17:17.21 | Dovid | want me to send u a copy of it ? |
| 17:17.29 | fiber0pti | Sure, that would be helpful |
| 17:18.13 | *** join/#asterisk adorah (n=Administ@84.94.209.161.cable.012.net.il) |
| 17:22.53 | Dovid | fiber0pti: pastebin is acting up. brb |
| 17:22.58 | *** join/#asterisk m_a_g_o (i=maxgluck@201.243.102.189) |
| 17:23.01 | fiber0pti | k |
| 17:23.17 | *** join/#asterisk funtable (n=Carlos@201-24-231-127.ctame704.dsl.brasiltelecom.net.br) |
| 17:23.39 | *** part/#asterisk funtable (n=Carlos@201-24-231-127.ctame704.dsl.brasiltelecom.net.br) |
| 17:23.44 | m_a_g_o | good afternoon folks, does anyone know if the new feature periodic-announce is supported in realtime? |
| 17:23.46 | *** join/#asterisk CrashHD (i=CrashHD@67.182.167.222) |
| 17:24.02 | turth | What would cause the voice to lag when using the g729 codec? its comes in clear just alot of static |
| 17:24.15 | m_a_g_o | it is not mentioned here: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue |
| 17:24.17 | turth | i mean it comes in clear and no static |
| 17:24.27 | turth | just lags |
| 17:24.37 | turth | like it makes the persons voice 1 mph instead of 5 mp like normal |
| 17:24.45 | turth | 5 mph* |
| 17:26.00 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:26.31 | Dovid | fiber0pti: have a look at this http://pastebin.ca/135035 |
| 17:27.21 | Dovid | or here |
| 17:27.21 | Dovid | http://www.h6315.com/eg |
| 17:28.55 | fiber0pti | Thanks |
| 17:28.57 | fiber0pti | looks good |
| 17:29.07 | [TK]D-Fender | Dovid: Dear God that macro is just WRONG.... |
| 17:29.22 | Dovid | ol TK |
| 17:29.27 | Dovid | i am a beginer... |
| 17:29.36 | Dovid | what dont u like about it. i luv to learn |
| 17:29.45 | CrashHD | heh |
| 17:29.56 | CrashHD | I swear I come in here just for the humor |
| 17:30.33 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:30.44 | Dovid | oh well |
| 17:30.51 | Dovid | we werent all born geniuses |
| 17:31.02 | [TK]D-Fender | Dovid: I'll clean it up for you :) |
| 17:31.15 | Dovid | TK: THanks. |
| 17:31.19 | CrashHD | D-Fender apparently was |
| 17:31.39 | Dovid | going for a smoke. brb |
| 17:31.42 | [TK]D-Fender | Dovid: You ARE on 1.2 right? |
| 17:31.51 | Dovid | yes TK i am |
| 17:33.48 | [TK]D-Fender | Dovid: Here : http://pastebin.ca/135056 |
| 17:34.30 | [TK]D-Fender | Pastebin.ca IS god-aweful slow today..... |
| 17:34.43 | Dovid | yes. takin for ever like ususal |
| 17:34.50 | [TK]D-Fender | But I got it up.... |
| 17:36.35 | fiber0pti | whoa |
| 17:36.41 | [TK]D-Fender | just did a tiny 1.2 formatting change : 1st line of macro : exten => s,1,Set(CALLERID(number)=${ARG2}) |
| 17:36.43 | fiber0pti | TK, you reduced that... a lot... |
| 17:36.56 | [TK]D-Fender | fiber0pti: YUP. |
| 17:36.59 | Dovid | Tk: question. i have n+101 cause some of the routes arent up yeat. the variables r blank. will it still to go ext pri. ? |
| 17:37.05 | [TK]D-Fender | http://pastebin.ca/135059 |
| 17:37.09 | CrashHD | is there an alternative voicemail-login prompt already available? |
| 17:37.39 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:37.48 | [TK]D-Fender | Dovid: If the chan isn't available it'll just go to the next. As soon as one of those Dials DOES get to dial out it |
| 17:37.55 | Dovid | okies |
| 17:37.57 | Dovid | thanks |
| 17:37.57 | Dovid | brb |
| 17:37.59 | [TK]D-Fender | ll hang up at the end fo call and thats it |
| 17:38.25 | [TK]D-Fender | CrashHD: You mean one that gets rid of "comedian Mail" basically? |
| 17:40.25 | [TK]D-Fender | Dovid: You could remove one of those Zap lines in your dial if you jsut put the channels in a group. Even shorter :) |
| 17:42.02 | *** join/#asterisk isede (n=qed@pool-70-19-73-132.ny325.east.verizon.net) |
| 17:45.58 | Seba_soy | it is strange, I can load channel 2 from my generic X100p, but I can't load channel 1 from my another generic X100P |
| 17:46.23 | Seba_soy | with a proc cat, I found this.... |
| 17:46.24 | Seba_soy | cat /proc/zaptel/1 |
| 17:46.24 | Seba_soy | Span 1: WCFXO/0 "Generic Clone Board 1" RED |
| 17:46.24 | Seba_soy | <PROTECTED> |
| 17:46.28 | Seba_soy | IN USE???? |
| 17:46.36 | Seba_soy | cat /proc/zaptel/2 |
| 17:46.37 | Seba_soy | Span 2: WCFXO/1 "Generic Clone Board 2" RED |
| 17:46.37 | Seba_soy | <PROTECTED> |
| 17:47.50 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:48.01 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:53.19 | *** join/#asterisk UlbabraB (n=filippo@host241-43-static.72-81-b.business.telecomitalia.it) |
| 18:00.29 | *** join/#asterisk angom_w (n=angom@red-corp-200.79.148.126.telnor.net) |
| 18:01.50 | *** join/#asterisk Vec (n=Vector@dsl-146-122-62.telkomadsl.co.za) |
| 18:01.51 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:02.11 | fiber0pti | can Goto be used within a macro? |
| 18:03.48 | Seba_soy | any help for me :)? |
| 18:04.24 | *** join/#asterisk signuts (n=signuts@sig.triton.net) |
| 18:04.34 | CrashHD | yes goto can be used anywhere |
| 18:04.41 | signuts | Does anyone have a good method of tracking what server a UA is registered at? |
| 18:04.55 | signuts | The only method I can figure is to login via manager and track the SIP registration event |
| 18:05.09 | *** join/#asterisk Ebola (n=Ebola@user-54458db0.lns1-c13.telh.dsl.pol.co.uk) |
| 18:10.37 | *** part/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 18:11.03 | infinity1 | ManxPower: are you using relaxdtmf? |
| 18:11.06 | infinity1 | i just noticed this option |
| 18:11.13 | ki2k | does zapbarge work w/ nonzap channels? |
| 18:12.22 | infinity1 | wow. i didn't know asterisk speaks with googletalk |
| 18:12.28 | infinity1 | that are some weird patches? |
| 18:12.29 | infinity1 | crazy |
| 18:13.17 | [TK]D-Fender | fiber0pti: Sure. |
| 18:15.24 | [TK]D-Fender | fiber0pti: I use it in my Parking macro's to validate the exten. |
| 18:18.57 | *** join/#asterisk jbroome (n=jbroome@unaffiliated/jbroome) |
| 18:19.45 | *** join/#asterisk _alex_mx_ (n=alex@dsl-200-67-125-45.prod-empresarial.com.mx) |
| 18:21.54 | *** join/#asterisk enjay- (n=enjay@71.216.165.97) |
| 18:23.12 | *** join/#asterisk ctaloi (n=caloi@nat-66-218-1-117.usadatanet.com) |
| 18:24.44 | *** join/#asterisk RoyK (n=roy@216-154-68.0512.adsl.tele2.no) |
| 18:25.40 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 18:29.23 | *** join/#asterisk topping (n=topping@207.47.6.207.static.nextweb.net) |
| 18:30.27 | *** join/#asterisk slinabery (n=slinaber@smooth.worldcycling.com) |
| 18:31.08 | *** join/#asterisk isede_ (n=qed@cpe-74-65-225-244.nyc.res.rr.com) |
| 18:31.21 | enjay- | Im experiencing a hangup issue when recording calls.. If I record all calls its taking between 10-15 seconds to hangup the channel. If I turn off recording it works fine. Has anyone experienced this type of issue? |
| 18:31.50 | enjay- | Additionally I have tried sending recordings to an extension (that resides on another server via an IAX trunk) so that recording would be happening on another server however I still experience the problem. |
| 18:32.20 | enjay- | the sysloadavg doubles when I have recording enabled even though its happening on a seperate server.. |
| 18:33.05 | *** join/#asterisk }btorch{ (n=root@geosv04.geofocus.com) |
| 18:33.11 | }btorch{ | hello |
| 18:34.39 | }btorch{ | hey guys I once I have updated my * and zaptel to the latest version I see this errors on my kernel log |
| 18:35.30 | }btorch{ | zaptel disabled echo canceller because of tone (rx) on channel 2 |
| 18:35.47 | Cresl1n | }btorch{: so..... |
| 18:36.19 | [TK]D-Fender | }btorch{: You get that when faxes & modems call you |
| 18:36.36 | [TK]D-Fender | }btorch{: If you have detection turned on in your zapata |
| 18:37.15 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:37.23 | }btorch{ | oh really |
| 18:38.11 | *** join/#asterisk num000 (n=numerobi@e177188253.adsl.alicedsl.de) |
| 18:39.29 | num000 | does anyone know why i do get asterisk: can't resolve symbol 'cfgetispeed' wenn i start asterisk -r? |
| 18:40.54 | hmmhesays | using some module not included with the release? |
| 18:41.04 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 18:42.06 | num000 | hmmhesays ok, or could it be that it does not find the module? |
| 18:42.16 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:42.33 | hmmhesays | no |
| 18:42.36 | syzygyBSD | how can I change the default outgoing IP of a linux box |
| 18:42.40 | hmmhesays | and that was a question |
| 18:42.59 | syzygyBSD | I have 16 ips on this machine, and it isn't picking the first, second, or last |
| 18:43.04 | hmmhesays | route x.x.x.x default gw ethX |
| 18:43.17 | syzygyBSD | ahh, thanks |
| 18:43.27 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:43.44 | num000 | hmmhesays do you know which module calls the cfgetispeed? |
| 18:44.22 | syzygyBSD | ok, they are all set to eth0, but the interface it is using is eth0:14 |
| 18:48.33 | hmmhesays | no but there is a doxygen display of the code on asterisk.org |
| 18:49.36 | *** join/#asterisk signuts (n=signuts@sig.triton.net) |
| 18:50.03 | signuts | What's the best method of tracking where each SIP UA is registered when you have more than 1 SIP server ? |
| 18:50.20 | signuts | The only method I can think of is to login to each asterisk via manager interface and track the register events. But that seems lame |
| 18:50.30 | signuts | and overly complex |
| 18:51.27 | CunningPike | signuts: Looking at each server to see which UAs are registered there seems quite sensible to me....... how else would you do it? |
| 18:52.22 | signuts | CunningPike, a simple patch to store the ip/servername/server id, etc in sip_friends via realtime would be nice |
| 18:52.50 | signuts | CunningPike, but even then one would have to query more than one table as sharing sip friends is a big nono.. so i'm told |
| 18:55.00 | justinu|laptop | all patches seem simple at first |
| 18:56.43 | signuts | true. Is there any existing infrastructure to track UA registrations in a multi server environment? |
| 18:57.35 | signuts | i'm currently checkign DUNDi |
| 18:58.12 | *** join/#asterisk RoyK (n=roy@216-154-68.0512.adsl.tele2.no) |
| 18:59.09 | num000 | hmmhesays ahh, cfgetispeed is realted to the terminalspeed. asterisk -r also does not know about xterm terminal type, maybe my terminal environment is dump. is there an other way testing asterisk except of asterisk -r? |
| 18:59.32 | }btorch{ | does anyone here know if a Dialogic voice card works with asterisk |
| 18:59.40 | }btorch{ | analog voice card |
| 19:00.04 | [TK]D-Fender | }btorch{: Some can but are flakey. To be avoided. |
| 19:00.11 | russellb | }btorch{: not yet, but it will be available soon |
| 19:00.18 | russellb | [TK]D-Fender: what are you talking about? |
| 19:00.32 | russellb | }btorch{: in Asteirsk business edition |
| 19:01.35 | [TK]D-Fender | russellb: Could have sworn I've seen modules supporting some of them before... |
| 19:01.44 | russellb | [TK]D-Fender: no, you completely made that up |
| 19:01.53 | }btorch{ | asterisk business edition ? |
| 19:01.58 | Qwell | ~abe |
| 19:02.00 | russellb | this channel drives me crazy ... there is so much completely bogus information passed around |
| 19:02.06 | Qwell | ~be |
| 19:02.07 | jbot | i guess be is Beryllium. Belgium |
| 19:02.25 | russellb | }btorch{: see www.digium.com for more information |
| 19:02.28 | }btorch{ | it won't be available on the opensource one ? |
| 19:02.49 | [TK]D-Fender | russellb: Well "oops". Caught something wrong in reading a while back... |
| 19:02.58 | [TK]D-Fender | russellb: A rarity for me to say the least. |
| 19:03.12 | russellb | }btorch{: not as of now, no |
| 19:03.12 | }btorch{ | I would like to resolve this Siemens PBX routing to asterisk issues that I have |
| 19:03.23 | justinu|laptop | D-fender: you suck |
| 19:03.27 | RTFSvnBook | russellb: still reading... |
| 19:03.30 | justinu|laptop | D-fender, you neve rhelp anyone |
| 19:03.33 | russellb | [TK]D-Fender: nothing personal :) |
| 19:03.39 | }btorch{ | for passing the extension |
| 19:03.43 | [TK]D-Fender | justinu|laptop: You swallow.. AND LIKE IT. |
| 19:03.45 | russellb | RTFSvnBook: now the svn book? |
| 19:03.47 | }btorch{ | anyway .... thanks |
| 19:04.19 | RTFSvnBook | russellb: yes... for some reason version control took priority over phone system |
| 19:04.25 | russellb | lol |
| 19:04.30 | [TK]D-Fender | russellb: Its amazing how the words "Don't take this personally" are are almost invariably followed by a very personal sleight ;) |
| 19:04.53 | RTFSvnBook | russellb: our canadian branch was claiming svn+ssh was too slow and wanted apache svn/webdav |
| 19:05.04 | Qwell | [TK]D-Fender: Don't take this personally, but you smell funny |
| 19:05.24 | russellb | Qwell: yeah, nothing personal ... but I really don't like you |
| 19:05.29 | Qwell | ouch |
| 19:05.37 | russellb | just kidding! :D |
| 19:05.39 | Qwell | :p |
| 19:05.41 | [TK]D-Fender | lol... what have I done..... |
| 19:06.39 | justinu|laptop | funny how no one ever says thanks when you help the noobs thru some very time consuming tech support |
| 19:07.07 | CrashHD | ungrateful |
| 19:07.11 | file | HA |
| 19:07.26 | file | you guys rock |
| 19:07.27 | Qwell | aww |
| 19:07.38 | russellb | w00t |
| 19:07.47 | CrashHD | files canadian, he likes everyone |
| 19:07.59 | Qwell | he just likes our currency |
| 19:08.04 | CrashHD | lol |
| 19:09.06 | signuts | Anyone ever look at using dundi for managing which peer/extension is associated within a group of SIP servers? |
| 19:10.34 | RoyK | anyone that knows a good sip load balancer that also handles rtp media? |
| 19:11.17 | signuts | RoyK, i'm essentially looking for the same thing. The main problem is receiving calls and routing them to the correct SIP UA. Nobody seems to care about this process =) but it's very important to a scalable VOIP platform |
| 19:14.24 | signuts | bah! |
| 19:15.41 | signuts | Is there are real documents on dundi? |
| 19:16.01 | signuts | I don't know what symmetric means in dundi sp33k |
| 19:16.17 | slinabery | Sigh. So, uh, what is the recommended OS for compiling zaptel? I have had no luck with CentOS 4.3, 2.6.9-34.0.2.EL-smp-x86_64. I am not a distro bigot; I just want things to work. |
| 19:16.40 | enjay- | what compilation errors do you get? |
| 19:16.59 | signuts | slinabery, you need -devel packages. I had zero problems on slackware, debian and gentoo |
| 19:17.03 | *** part/#asterisk neo (n=neo@kessel.ordrejedis.net) |
| 19:17.15 | slinabery | I have devel packges. |
| 19:17.21 | enjay- | and I've had zero problems on Fedora, an Centos |
| 19:17.26 | signuts | heh |
| 19:17.43 | justinu|laptop | ~centosbug |
| 19:17.44 | jbot | rumour has it, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 19:17.52 | signuts | maybe a better course of action. What's your build error slinabery |
| 19:18.09 | justinu|laptop | slinabery: that applies to building zaptel on centos |
| 19:18.13 | slinabery | /usr/src/zaptel-1.2.7/zaptel.c: In function `zt_init': |
| 19:18.13 | slinabery | /usr/src/zaptel-1.2.7/zaptel.c:6553: error: incompatible types in assignment |
| 19:18.13 | slinabery | /usr/src/zaptel-1.2.7/zaptel.c: At top level: |
| 19:18.13 | slinabery | /usr/src/zaptel-1.2.7/zaptel.c:188: warning: 'fcstab' defined but not used |
| 19:18.13 | slinabery | make[2]: *** [/usr/src/zaptel-1.2.7/zaptel.o] Error 1 |
| 19:18.14 | slinabery | make[1]: *** [_module_/usr/src/zaptel-1.2.7] Error 2 |
| 19:18.16 | slinabery | make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.0.2.EL-smp-x86_64' |
| 19:18.17 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 19:18.18 | slinabery | make: *** [linux26] Error 2 |
| 19:18.19 | enjay- | bleh |
| 19:18.23 | slinabery | yes, I am trying to build zaptel on centos. |
| 19:19.29 | Juggie | slinabery, did you do the fix to the kernel? |
| 19:20.06 | slinabery | what's the build platform at digium? I really don't care what I wind up using. |
| 19:21.31 | Assid | err.. just curious.. would a specific platform do better for asterisk ? such as AMD Athlon64 3400 vs Intel 3.2Ghz |
| 19:21.41 | Assid | err.. 3.4Ghz even |
| 19:21.43 | justinu|laptop | slinabery: please try patching the kernel headers with the instructions in ~centosbug |
| 19:21.46 | justinu|laptop | ~centosbug |
| 19:21.48 | jbot | well, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 19:21.58 | hmmhesays | sometimes i give myself the creeeps |
| 19:22.04 | hmmhesays | sometimes my mind plays tricks on me |
| 19:22.11 | hmmhesays | it all keeps adding up, I think I'm cracking up |
| 19:22.15 | slinabery | sorry, I don't understand what ~centosbug is shorthand for. |
| 19:22.17 | Assid | i mean after all. there are certain types of applications which supposedly do better with certain processors |
| 19:22.22 | Juggie | jesus christ |
| 19:22.25 | hmmhesays | are you retarded? |
| 19:22.31 | Juggie | [15:22] <jbot> well, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 19:22.35 | Juggie | CAN YOU SEE IT?!?! |
| 19:22.40 | hmmhesays | mabe he can't see jbot |
| 19:22.40 | enjay- | haha |
| 19:22.48 | slinabery | yes. thanks so much for that helpful advice Juggie. |
| 19:22.55 | Juggie | THEN DO IT |
| 19:22.57 | Juggie | and report back |
| 19:23.00 | enjay- | stat |
| 19:23.02 | hmmhesays | no don't report back |
| 19:23.05 | Assid | on the double! |
| 19:23.22 | Nivex | Howard, what exactly does "stat" mean? |
| 19:23.28 | Juggie | there will be no further attempt until you try the obvious solution |
| 19:23.28 | Nivex | name the movie |
| 19:23.31 | ionix | it means quick, medical term :) |
| 19:23.31 | Juggie | which is almost sure to work |
| 19:23.38 | enjay- | yerp.. |
| 19:23.55 | enjay- | forcepts STAT |
| 19:24.09 | num000 | is anyone using asterisk on a linksys wrt54g ?? |
| 19:24.14 | hmmhesays | yes |
| 19:24.35 | justinu|laptop | it's latin, right? |
| 19:24.56 | ionix | yeh |
| 19:24.57 | quid246 | num000: wrt54gs x 2 |
| 19:24.58 | num000 | hmmhesays are you using asterisk with a wrt54g ? which os do you run on it? |
| 19:25.06 | hmmhesays | openwrt |
| 19:25.07 | quid246 | DD-WRT |
| 19:25.09 | justinu|laptop | dd-wrt!! |
| 19:25.13 | num000 | quid246 uyy, cool and they work? cool |
| 19:25.18 | num000 | wow, one more |
| 19:25.22 | justinu|laptop | dd-wrt is a lot nicer than openwrt |
| 19:25.29 | hmmhesays | openwrt, with chan_oss so I can use a usb sound card |
| 19:25.33 | quid246 | num: I'm not running Asterisk ON the WRT54GS... but through it |
| 19:25.39 | num000 | quid246 i'm using dd-wrt aswell but i do have trouble with it and i'm getting mad, would you guys help me? |
| 19:25.39 | slinabery | Juggie, I wasn't asking for a reading lesson. I just don't know irc shorthand. No need to be a dickhead. |
| 19:25.42 | Assid | isnt dd-wrt console based? |
| 19:25.51 | justinu|laptop | no, it has a very nice web interface |
| 19:25.56 | Assid | it does? |
| 19:25.58 | enjay- | simmer down slin |
| 19:25.59 | num000 | Assid aswell as configurable via html |
| 19:26.01 | quid246 | assid: no |
| 19:26.14 | *** join/#asterisk chreese (n=chatzill@bridalveil.istep.com) |
| 19:26.32 | Juggie | slinabery, there was no irc shorthand |
| 19:26.39 | Assid | hrmm.. i gave up my linksys wrt54g for a stupid dlink 524 |
| 19:26.42 | Juggie | jbot posted the long super extended version |
| 19:26.44 | Juggie | like 2 times |
| 19:26.47 | hmmhesays | don't forgive stupidity |
| 19:26.47 | num000 | i was trying to start asterisk with the option -r but it stops with can't resolve symbol cfgetispeed |
| 19:27.01 | hmmhesays | you are missing a dependency obviously |
| 19:27.01 | Assid | but then i did have v5 |
| 19:27.09 | Juggie | num000, you didnt compile * from source did you. |
| 19:27.17 | Assid | okay be back later.. i need to watch a movie.. hurray! |
| 19:27.18 | num000 | Juggie no i didn't |
| 19:27.33 | num000 | Juggie i do not have a crosscompiler for the architecture |
| 19:28.13 | Juggie | num000, seems like your missing a library. |
| 19:29.16 | num000 | Juggie possible, it was also asking for the ncurses library 5, although i do have 5.2, so i renamed it to so.5 since then asterisk at least starts. do i have to use asterisk -r? |
| 19:29.31 | signuts | Anyone know about dundi here? |
| 19:29.41 | Juggie | no |
| 19:29.50 | chreese | does anyone know what this means: Fax receive not successful - result (14) TIFF/F file cannot be opened. I'm using NVFaxDetect |
| 19:29.58 | Juggie | but you wont have any way to detach from the process if you dont |
| 19:30.13 | enjay- | chreese; it means fax+voip = teh suck! |
| 19:30.19 | Juggie | looks like, cfgetispeed is provided by libc |
| 19:30.32 | Juggie | specifically termios.h |
| 19:30.38 | num000 | cfgetispeed looks like a function call to get terminalspeed |
| 19:30.51 | Juggie | so it would seem yes. |
| 19:30.54 | num000 | i found it in the source |
| 19:30.58 | num000 | but what can i do? |
| 19:31.10 | chreese | enjay: thanks. i'm about to give up onfax voip. has anyone used nvfaxdetect? |
| 19:31.22 | Juggie | num000 what platform are you running * on? |
| 19:31.38 | num000 | asterisk also did claim about terminal type 'xterm' but it did fall back to dump |
| 19:31.57 | *** part/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 19:32.01 | num000 | Juggie i'm using it on linksys wrt54gl hardwave revison 1.1 which should be the same as wrt54gs version 4 |
| 19:32.02 | enjay- | chreese; I've used sdp I've tried handytones multiple ata's fax and voip (even with all ulaw) doesnt work for shit in my unprofessional opinion. |
| 19:32.21 | *** part/#asterisk _alex_mx_ (n=alex@dsl-200-67-125-45.prod-empresarial.com.mx) |
| 19:32.28 | Juggie | how are you getting your tty |
| 19:32.30 | num000 | i suppose all those using asterisk on the linksys router are all happy are they? |
| 19:32.31 | Juggie | over telnet? |
| 19:32.32 | Nugget | telnet is eeeeeeevil! |
| 19:32.45 | num000 | Juggie i'm not setting the terminal at all till now |
| 19:32.46 | enjay- | get an analogue line and be done with it. |
| 19:32.53 | slinabery | centosbug instructions allowed zaptel to compile. thank you, Juggie et al. |
| 19:33.04 | Juggie | we thought it would :) |
| 19:33.06 | justinu|laptop | heh |
| 19:33.09 | enjay- | yea juggie you big dickhead |
| 19:33.12 | ki2k | ya centos bug sucks |
| 19:33.16 | ki2k | stupid typo |
| 19:33.21 | ki2k | or was it intentional? |
| 19:33.42 | slinabery | I'm a dickhead, don't bother kind Juggie. |
| 19:33.49 | num000 | Juggie what terminal settings do you have? |
| 19:34.02 | Juggie | num000, i dont use * on a linksys |
| 19:34.08 | meshuga | Hey, anyone been able to fix the transfer > callerid issue where if caller A calls B and then transfer to C it still say the call is from B instead of A? |
| 19:34.17 | num000 | Juggie ohh |
| 19:34.17 | meshuga | Note The caller ID presented to the person you are trying to transfer the call to is not what you would expect - Asterisk sets your caller ID to be the extension the call originally arrived at which may not be the same as the extension the call was answered at. There doesn't appear to be any way of getting the correct caller ID. |
| 19:34.30 | Juggie | num000, try here, http://www.voip-info.org/wiki-Asterisk+Linksys+WRT54G might be some useful info there. |
| 19:34.32 | meshuga | thats what i see in the wiki, but its from '04 |
| 19:35.02 | num000 | Juggie thank you very much |
| 19:35.13 | meshuga | * on a wrt54g rocks |
| 19:35.18 | meshuga | i've done up quite a few |
| 19:35.22 | justinu|laptop | juggie, you spreading false rumors again? |
| 19:35.32 | Juggie | justinu, allways, its my hobby |
| 19:35.32 | num000 | meshuga which os do you use? |
| 19:35.36 | justinu|laptop | juggie :) |
| 19:35.36 | meshuga | num000: openwrt |
| 19:35.47 | meshuga | i wrote up a document on it for my company if anyone is interested |
| 19:35.53 | num000 | meshuga cool, can you tell me which terminal settings you have? |
| 19:36.03 | meshuga | 'terminal settings'? |
| 19:36.04 | Juggie | num000, it has nothing to do w/ terminal settings |
| 19:36.05 | meshuga | i just ssh in. |
| 19:36.09 | Juggie | you are missing a library |
| 19:36.22 | Juggie | or the wrong version or something which * isnt linked against |
| 19:36.25 | Juggie | and it cant find the function |
| 19:36.32 | num000 | Juggie ok, |
| 19:36.51 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-70-94.telkomadsl.co.za) |
| 19:36.51 | meshuga | nobody is having this transfer issue? |
| 19:36.56 | num000 | is anyone using asterisk with dd-wrt? |
| 19:37.00 | Juggie | num000, i also dont think * uses ncurses |
| 19:37.05 | Juggie | unless the wrt version does. |
| 19:37.16 | justinu|laptop | possibly readline does? |
| 19:37.26 | Juggie | maybe? i thought * used newt. |
| 19:37.30 | num000 | Juggie how you mean not using ncurses? |
| 19:37.35 | meshuga | http://lists.digium.com/pipermail/asterisk-dev/2005-July/014140.html |
| 19:37.39 | meshuga | ahh, doesnt appear to work on sip |
| 19:37.47 | CunningPike | meshuga: What type of phone are you using? |
| 19:37.52 | Juggie | num000, didnt you say you renanmed a ncurses lib to make * run? |
| 19:37.58 | CunningPike | meshuga: Try the 'o' option for Dial() |
| 19:38.15 | num000 | Juggie yes, i renamed the existing library called libncurses.so.5.2 to libncurses.so.5 |
| 19:38.31 | Juggie | num000, then thats what i ment i didnt think * was lniked to ncurses. |
| 19:38.36 | num000 | Juggie since then asterisk at least starts |
| 19:38.41 | num000 | ok |
| 19:38.48 | meshuga | CunningPike: i'm using eyebeam/cisco 7960/clippcom cg-201e, and a sip stack we wrote that uses radvision :) |
| 19:38.57 | meshuga | CunningPike: will update my macro, sec |
| 19:39.15 | CunningPike | meshuga: Ah, the usual ;) |
| 19:39.35 | meshuga | yea i'm pretty happy with this clipcom cg-201e |
| 19:39.44 | meshuga | for a $130 2FXS/1FXO it works flawless. |
| 19:39.53 | hmmhesays | do yourself a favor and use buildroot to build asterisk |
| 19:40.07 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:41.11 | Juggie | num000, maybe http://www.wildcatwireless.net/wrt54g/uclibc_0.9.27-6_mipsel.ipk might help. |
| 19:42.40 | meshuga | man, freepbx makes such a mess of the macros when i'm trying to add a variable to the end |
| 19:42.55 | Juggie | as hmmhesays said, you might try setting up a buildroot and building your own copy and moving it over. |
| 19:43.05 | CunningPike | meshuga: Hence the channel topic :) |
| 19:43.34 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:43.37 | meshuga | CunningPike: yea, i usually build my own from scratch, but my qa guy requires a damn web frontend |
| 19:43.53 | CunningPike | meshuga: FOP ;) |
| 19:43.59 | meshuga | hahaha |
| 19:44.04 | CunningPike | :D |
| 19:44.12 | meshuga | i wish ! |
| 19:44.14 | eKo1 | web frontends are overrated |
| 19:44.42 | meshuga | i've got the director of dev saying if this is going full scale we're going to write our own frontend, so thats cool at least |
| 19:44.55 | meshuga | i dont mind a web frontend if its simple and easy to read |
| 19:47.38 | *** join/#asterisk System010 (n=jgargano@hide247.cybergnostic.com) |
| 19:48.31 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:49.26 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:51.49 | System010 | Hi, I've configured an asterisk server as an IVR using an AGI script. this then connects to our intertel phone system with a T card, which is working. I have an option in our agi script to transfer back to a CSR queue on the intertel phone system, this works. However, if the user hangs up while in queue, the bridge connection stays in asterisk indefintly |
| 19:53.45 | groogs | Anyone know anything about the "TDM PCI Master Abort" error? I've been getting it for a long time (usually somewhere between 2 and 20 days, though it happened 4 times one day last week), finally replaced the motherboard/cpu/ram.. and still get it. I was on an asus A7N8X (nvidia chipset) with athlon xp 2000+, now on a biostar b4m800-m7 (via chipset) with a celeron-d 326 2.53ghz. |
| 19:54.00 | CunningPike | meshuga: We considered it, but opted for a CLI menu interface instead |
| 19:54.15 | CunningPike | meshuga: Fewer constraints on conf file formats |
| 19:56.55 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-8.getnet.net) |
| 20:01.11 | *** join/#asterisk bartpbx (n=bartpbx@p54B0486C.dip0.t-ipconnect.de) |
| 20:01.45 | bartpbx | hello |
| 20:03.59 | bartpbx | one of our asteisk servers just died with a core dum |
| 20:04.43 | *** join/#asterisk num000 (n=numerobi@e177181186.adsl.alicedsl.de) |
| 20:04.48 | bartpbx | ist there any know issue in cahn_iax2 |
| 20:05.21 | *** join/#asterisk RoyK (n=roy@216-154-68.0512.adsl.tele2.no) |
| 20:06.15 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 20:06.15 | tzanger | bartpbx: I've seen that today |
| 20:06.18 | tzanger | continuous crashes |
| 20:06.25 | bartpbx | hm |
| 20:06.31 | tzanger | unfortunately my craven underling svn up'd before I could look at the core dumps, so the source files are out of sync |
| 20:06.41 | bartpbx | hm |
| 20:06.42 | tzanger | it has been good this afternoon though, so maybe svn up |
| 20:06.42 | bartpbx | strange |
| 20:07.19 | file | bartpbx: what version of Asterisk? |
| 20:07.27 | RoyK | hi |
| 20:07.28 | bartpbx | 1.2.10 |
| 20:07.35 | bartpbx | for now only one crash |
| 20:08.02 | bartpbx | about an hour ago |
| 20:08.12 | bartpbx | currently everything seems fine |
| 20:08.21 | bartpbx | all calls are going through |
| 20:08.47 | bartpbx | I'll monitor this and come back tomorrow |
| 20:09.31 | *** part/#asterisk bpiper (n=bpiper@70.159.49.40) |
| 20:10.50 | chreese | anyone know how to fix this error i got when installing the latest svn branch?: Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. |
| 20:11.47 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 20:11.47 | chreese | and then it lists some modules (not all of them in that dir.) |
| 20:15.14 | *** join/#asterisk Geliman (n=scorpio@unaffiliated/drkshdw) |
| 20:23.37 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 20:23.44 | *** part/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
| 20:23.51 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.88) |
| 20:31.42 | *** part/#asterisk System010 (n=jgargano@hide247.cybergnostic.com) |
| 20:31.44 | *** join/#asterisk num000 (n=numerobi@e177189145.adsl.alicedsl.de) |
| 20:34.48 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:36.15 | *** join/#asterisk SwK (n=Silik0nJ@70.46.56.34) |
| 20:37.44 | *** part/#asterisk viler (i=1000@200.114.70.228) |
| 20:38.08 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 20:40.28 | russellb | oh no you didn't just take a dump on my root filesystem |
| 20:42.22 | }btorch{ | does anyone know how create a route on a legacy PBX that forwards the call to another number but also somehow sends the the original number dialed |
| 20:42.43 | CoffeeIV | how can I tell, from the full log or at the *CLI> prompt, what codec a call is using or used ? |
| 20:43.04 | RoyK | CoffeeIV: i don't think you can |
| 20:43.36 | sb_mx | CoffeeIV, sip show channels |
| 20:44.01 | sb_mx | CoffeeIV, if you're using sip of course ;) |
| 20:44.07 | }btorch{ | I created a forward on my Siemens PBX to forward any calls to 1530 to 7-2002(<- sends to asterisk on extension 2002) but the caller ID is the actuall caller |
| 20:44.34 | }btorch{ | Is there a way to capture 1530 ? |
| 20:44.39 | sb_mx | iax2 show channels if you're using iax |
| 20:44.48 | CoffeeIV | sb_mx: thanks |
| 20:49.58 | toerkeium | <PROTECTED> |
| 20:50.47 | intralanman | }btorch{: i'm assuming 1530 is already used on the * box or for some other reason you can't use the same number? |
| 20:51.03 | intralanman | i don't think * supports third-party numbers exactly yet |
| 20:52.11 | }btorch{ | intralanman no 1530 is the extension I originally called but since it doesn't pick up I use a forward rule on the Siemens PBX to send it to 2002, an asterisk extension |
| 20:53.18 | }btorch{ | intralanman when that extension picks up it shows the call coming from 1503(caller) to extension 2002 there is no trace of the 1530 |
| 20:56.36 | }btorch{ | hey you know when you see on asterisk CLI the line accepting call from 'XXXX' to 'YYYY' on channe 2... what is the variable YYYY is saved under? |
| 20:56.49 | }btorch{ | opps nervermind |
| 21:06.43 | *** join/#asterisk RoyK (n=roy@57-59-118-87.mtulink.net) |
| 21:08.05 | groogs | has anyone played with an spa-400 yet? |
| 21:09.20 | *** join/#asterisk harlequin516 (n=sham@dsl01-ppp-4444.fastq.com) |
| 21:09.25 | harlequin516 | what does exten => a,1,stuffhere mean. what is a? |
| 21:09.59 | harlequin516 | Oh I see it now. |
| 21:10.51 | Qwell | harlequin516: That's called when you exit voicemail with 0, I believe |
| 21:12.08 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 21:13.49 | harlequin516 | Qwell: Actuallythe wiki says when you press * |
| 21:13.57 | Qwell | harlequin516: That's called when you exit voicemail with *, I believe :) |
| 21:14.10 | harlequin516 | But when I press * it just hangs up for me, instead of jumping to that extensions. |
| 21:14.32 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:15.05 | *** join/#asterisk dima_ (n=dima@87.218.18.112) |
| 21:17.12 | harlequin516 | Any ideas about hangupp when I press * instead of going to VoicemailMain? |
| 21:17.28 | Skyelar | harlequin516: is the |
| 21:17.33 | Skyelar | (damn enter key) |
| 21:17.35 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 21:17.51 | Skyelar | is the "exten => a,1,..." in the same context you're calling VoiceMail(...) from? |
| 21:20.11 | *** part/#asterisk bartpbx (n=bartpbx@p54B0486C.dip0.t-ipconnect.de) |
| 21:20.36 | *** join/#asterisk swytch (n=ezcall@d83-179-158-7.cust.tele2.fr) |
| 21:22.15 | CoffeeIV | some people I'm helping have been trying out a number of VoIP providers for faxes, and they all have various quality problems (too be expected I think) -- is there a particular VoIP service that is known for decent faxing ? |
| 21:22.30 | harlequin516 | Skyelar Yes |
| 21:22.38 | *** join/#asterisk Olobola (n=casper_s@adsl-75-2-134-174.dsl.pltn13.sbcglobal.net) |
| 21:23.04 | Skyelar | harlequin516: what does the console show (with verbose set high, eg. 99) when you try it? |
| 21:23.46 | harlequin516 | Skyelar: It just shows hangup. I can post it. |
| 21:24.16 | Skyelar | harlequin516: pastebin "show dialplan <context in question>" and the console logs for the attempt |
| 21:26.04 | Olobola | what is up with new phone, I mean good lo'd. |
| 21:27.41 | harlequin516 | http://rafb.net/paste/results/ZfLw2141.html for my * hangup instead of voicemailmain problem |
| 21:29.40 | harlequin516 | Is tehre something in how to setup voicemail that can bypass the Goto(a,1) upon '*' DTMF ? |
| 21:30.33 | toerkeium | guys, I have buy a g729 licence .. but I don't know how to install/download it :P anyone have any idea? |
| 21:31.13 | *** join/#asterisk seebs (n=seebs@216-243-131-210.static.iphouse.net) |
| 21:31.41 | Skyelar | harlequin516: hmm... it looks right (could be being blind though). Do you have full logging enabled in logger.conf? If so, what's in /var/log/asterisk/full between the "Playing 'vm-isunavail'" and the hangup? |
| 21:31.56 | Skyelar | harlequin516: oh, and what version of Asterisk? |
| 21:32.52 | harlequin516 | Asterisk 1.2.9.1 |
| 21:32.54 | RoyK | 0.4.0 |
| 21:32.58 | seebs | So, I have gotten about to the point of installing hardware in a computer for an Asterisk setup, only to realize that I don't think I entirely understand the interaction between Asterisk and phone lines. |
| 21:33.02 | quid246 | toerkeium: contact Digium... since they sold it, they will support it |
| 21:33.40 | Skyelar | harlequin516: definitately works on that - used it myself |
| 21:33.45 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 21:33.54 | seebs | Imagine for the sake of argument that I have a regular land-line phone, and I have an X100 card. Do I just plug the phone into the phone connector on the card? Do I use a splitter to plug both the phone and the card into the line? |
| 21:33.56 | toerkeium | yeah, I was just impatient |
| 21:34.05 | harlequin516 | There is nothing between. I just realized something, I have no dialplan after Voicemail call. Should my next command be to exten a? |
| 21:34.30 | Skyelar | harlequin516: no - VoiceMail() should do the jump itself when it receives a "*" |
| 21:34.51 | harlequin516 | hmm |
| 21:35.09 | Skyelar | harlequin516: anything in the logs? |
| 21:35.51 | seebs | I think this is the only time I've ever been stumped because there was too much documentation for an open source project. :) |
| 21:36.39 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:36.40 | harlequin516 | I'm using -ddddddddddddddddd -vvvvvvvvvvvvvvvvv and don't see anyhting but a hangup. |
| 21:36.45 | harlequin516 | I will look for logs |
| 21:37.27 | toerkeium | guys, it's ok if instead of a P4 I have a celeron processor? |
| 21:37.32 | Skyelar | harlequin516: check logger.conf - something like "full => notice,warning,error,debug,verbose" in there will be helpful |
| 21:37.35 | toerkeium | for the g729 codec I mean |
| 21:37.54 | vader-- | can anyone point me in the direction that would help me get this done, i would like to create something that where when an incomming phone call comes in it rings a phone but you have the option of dialing an extension to pick up the call comming in on another phone |
| 21:38.40 | harlequin516 | Just removed comment from logger.conf |
| 21:39.27 | Skyelar | harlequin516: "logger reload" |
| 21:40.34 | *** join/#asterisk RickNZ (n=rick@ip-202-37-229-70.internet.co.nz) |
| 21:41.18 | harlequin516 | I'm on nonproduction, so I am killing and restarting asterisk |
| 21:42.33 | seebs | Maybe I am having a stupid moment, but... Is there anything on hardware requirements in the Asterisk docs? I haven't found it yet, if there is. |
| 21:42.45 | seebs | I don't mean "supported telephony", I mean "how much CPU and RAM do I need". |
| 21:45.51 | eKo1 | seebs: You need a Parallel Sysplex to run *. |
| 21:47.05 | seebs | Heh. |
| 21:47.30 | seebs | Well, my last experience with computer telephony was a GVP PhonePak on an Amiga, which ran two lines happily on an 8MHz 68000. |
| 21:47.55 | nortex | seebs, eko1 is an exclusive re-seller of the Parallel Sysplex line :) |
| 21:48.12 | seebs | My unconsidered assumption is that a spare 550Mhz P3 with 64MB of memory is "plenty", but I have no idea how stupid that might turn out to be. |
| 21:48.18 | seebs | Ahh. Context is everything. :) |
| 21:48.51 | nortex | seebs, How many lines and phones? Number of con-current calls? |
| 21:49.09 | seebs | If it matters, my intended telephony hardware is two or three X100P cards. Highest possible load would be two calls and a fax at once, if that's even possible. |
| 21:49.20 | seebs | If I can't do the fax line, I can live. |
| 21:49.52 | seebs | I have read a lot of install guides that talk about kernel modules, but none that explain which wires go where. :P |
| 21:50.24 | seebs | With the PhonePak, you just plugged the phone into the second telephone jack on the card, and it magically worked. |
| 21:50.34 | nortex | The 550 will probably do it, but I would look to more RAM, and check for minimums on your distro of linux. |
| 21:51.00 | seebs | Well, I'm tentatively planning to run asterisk-bsd, because I know NetBSD will rattle around in a 64MB system like a pea in a bucket. |
| 21:51.35 | seebs | No need for X, no need for mysql and eighteen other background daemons; I just want something to record caller ID and phone calls. |
| 21:51.58 | seebs | A bit of context: I sue telemarketers and junk faxers as a hobby. I spend a lot of time on phone calls that someone will later wish to deny under oath. |
| 21:52.16 | nortex | seebs, then your probably fine. |
| 21:52.22 | seebs | Cool. |
| 21:52.50 | seebs | So, uhm. Where do the wires go? Do I use a splitter on the incoming lines and run them both to my regular phone and to the PBX, or do I put the phone in the "downstream" ports on the modems, or what? |
| 21:53.03 | seebs | (It's a two-line phone, but it can be safely treated as though it were two separate one-line phones.) |
| 21:53.15 | hads | Do yourself a favour and use a TDM400 rather than an X100 |
| 21:54.13 | seebs | I looked at 'em, but they cost more than I can afford on this; it's all spare parts except for the X100s. |
| 21:54.48 | hads | OK, don't say you weren't warned :) |
| 21:55.00 | seebs | Or maybe I looked at the wrong part; I am very confused by the bevy of options. |
| 21:55.07 | seebs | Well, warned of what? Do X100s explode, killing all users? Have sorta sucky sound quality? |
| 21:55.47 | hads | Just generally sorta sucky. Also, you can't plug your phone into an X100 as it doesn't have any FSX ports - for that you will need a TDM400 |
| 21:55.53 | seebs | I assume the idea would be to get a TDM400P and split it 2FXS/2FXO, then connect it both to the lines and to the phone? |
| 21:56.44 | seebs | Okay. But in the mean time, since I'm flat broke until some more junk faxers cough up valuable money, what is the correct way to connect this? Splitter at the wall and run lines to both the X100 and the phone? |
| 21:58.18 | nortex | seebs, You just want to log the callerid or you want to record the call and be able to play it back? |
| 21:58.31 | seebs | I want to log the caller ID and record the call. |
| 21:58.43 | seebs | It's fine by me if the recording cannot be played over the phone, as long as it's in a format that a computer can read. |
| 21:58.55 | hads | If you want to record then the call will need to pass through Asterisk, which you can't do with an X100 and analog phone |
| 21:59.04 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-51-202.cybersurf.com) |
| 21:59.08 | seebs | Well, drat. |
| 21:59.30 | seebs | I was sort of hoping it could just listen in on a line where another extension picked up. *sigh*. |
| 21:59.39 | nortex | seebs, x100 and softphone on your pc would do it though. |
| 22:00.09 | seebs | I think down that path lies madness; I spent months searching for a wireless phone that would work in my house, and I don't want to replace it. |
| 22:00.44 | [TK]D-Fender | seebs What's so special that so few phones can work in your home? |
| 22:00.46 | seebs | So, it looks like the real answer is, if I want to do this, I need a TDM400 with two FXS and two FXO ports. Bleh! |
| 22:00.48 | nortex | seebs, you mean like cordless or wi-fi phone? |
| 22:01.02 | seebs | Three-story house with a lot of metal in the walls and a lot of interference. |
| 22:01.23 | seebs | I had a very hard time finding a two-line phone which had the features I wanted, and I finally got one. |
| 22:01.29 | xheliox | Anyone know of any free, or cheap, but fairly well maintained, local calling area databases? E.g. I enter an NPA-NXX and it shows me what other NPA-NXX's are local and if it requires 7 digit or 10 digit dialing? |
| 22:02.06 | nortex | xheliox, Are you looking for one location or many? |
| 22:02.08 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 22:02.15 | xheliox | nortex: Many. |
| 22:02.44 | RickNZ | anyone got experience loading misdn with the packages off the digium ftp site? |
| 22:02.49 | fiber0pti | is there a way to not require a conference room ask for a name? |
| 22:02.59 | *** part/#asterisk swytch (n=ezcall@d83-179-158-7.cust.tele2.fr) |
| 22:03.10 | nortex | xheliox, oh, I was going to mention getting your telco to give you the list, but that won't suffice |
| 22:03.54 | seebs | Well, that answers my short-term questions; I will order a TDM400, then plug the phone in to that when it shows up. |
| 22:04.17 | nortex | seebs, with a * pbx and 2 fxo ports and a single fxs port to could do it, and not need the 2 lines on the phone. |
| 22:04.35 | *** join/#asterisk schotten (n=stefano@amti.com.br) |
| 22:04.39 | seebs | Hmm. |
| 22:04.42 | schotten | hey u guys =] |
| 22:05.01 | nortex | Got to run Enjoyed it ;) |
| 22:05.13 | seebs | Could it? We have only the one "phone" shared throughout the house, so if I'm on one line-to-asterisk, I still need another line-to-asterisk if someone else wants to make a call at the same time, no? |
| 22:05.20 | schotten | can someone help me with rj45 pinage with TE110P ? |
| 22:05.27 | hads | seebs: Correct |
| 22:06.36 | seebs | Okay. So, for minimal intrusion into my system, I do the 2FXS/2FXO configuration, plug the existing wireless base station into the new board, plug the phone lines into the new board, and make a fairly trivial starter configuration, and everything will work just like it used to. Mostly. |
| 22:07.15 | hads | seebs: Sounds about right. |
| 22:07.21 | seebs | 'k. |
| 22:07.35 | seebs | And then it should be easy to log caller ID for everything and record everything. |
| 22:09.10 | hads | Can anyone enlighten me as to how Asterisk picks up the system timezone? On my test system I'm seeing weird behaviour; cdrs have the correct time, but STRFTIME, SayUnixTime and Voicemail are using GMT. I've been going slightly maad trying to work out why. |
| 22:09.31 | hads | seebs: Yes, it shouldn't be difficult at all |
| 22:10.08 | seebs | What's /etc/localtime look like? (It may be a symlink.) Also, I think caller-ID timestamps are provided by the telco, no? I could be totally wrong on that. |
| 22:11.32 | *** join/#asterisk rbordeaux (i=hidden-u@80.169.196.234) |
| 22:12.32 | hads | /etc/localtime is fine, it's a copy of my correct timezone (Pacific/Auckland) and date outputs as expected. |
| 22:13.18 | schotten | hads the 'date' command show ur timezone or GMT? |
| 22:13.35 | hads | Mine (NZST) |
| 22:14.52 | hads | looking through the stdtime/localtime.c source Asterisk seems to read both /usr/share/zoneinfo directly and /etc/localtime but it's a bit over my head in there. |
| 22:15.02 | seebs | I don't know much about Asterisk's debugging tools; can you do something like "echo $TZ" and see what it says? |
| 22:15.09 | schotten | localtime is not a ln ? |
| 22:15.50 | schotten | to /usr/share/zoneinfo/$location/$city? |
| 22:16.06 | hads | Na, it's a file - setup with tzselect (Debian) |
| 22:17.32 | schotten | have u run ntpdate ? |
| 22:18.22 | schotten | run a hwclock --show |
| 22:18.44 | hads | Yep, ntpdate and ntp are installed and run/running |
| 22:19.23 | hads | Will trying to debug lastnite I switched the HWclock from GMT to localtime. It now reports Thu 17 Aug 2006 10:18:35 NZST -0.349413 seconds which is correct |
| 22:19.35 | hads | s/Will/Well,/ |
| 22:19.50 | hads | and it was correct when running GMT too. |
| 22:21.07 | seebs | Hmm. |
| 22:21.28 | seebs | Hey, hang on a sec. Have you restarted stuff since changing that? I wonder whether there's something in a config file that causes corrections. |
| 22:21.38 | schotten | well.. i would try hwclock --localtime and -s |
| 22:21.46 | seebs | And one other question: Is it *saying* GMT, or is it just giving you the time that would be correct in GMT? |
| 22:21.52 | schotten | u can try copy ur timezone in /usr/share/timezone to /etc/localtime |
| 22:22.13 | schotten | dude, have to go |
| 22:22.15 | schotten | good luck |
| 22:22.21 | hads | Thanks. |
| 22:22.24 | hads | Bah |
| 22:22.37 | hads | seebs, when are you talking about? |
| 22:22.44 | hads | In Asterisk? |
| 22:23.03 | eKo1 | I recommend against using timezones and sticking with UTC |
| 22:23.10 | *** join/#asterisk NDT (n=noone@cpe-24-195-66-214.nycap.res.rr.com) |
| 22:23.28 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 22:24.13 | harlequin516 | Is there some feature that controls in a cal if pressing * hangs up the call? |
| 22:24.15 | hads | eKo1: That's confusing if you are listening to voicemail that was left at a completely different time :) |
| 22:25.00 | NDT | Anyone here ever use cepstral in asterisk? |
| 22:26.27 | NDT | I have it running...using a demo voice...but it is slow as hell to start talking |
| 22:26.27 | NDT | trying to figure out if that is because it is a demo...or that is normal heh |
| 22:27.12 | Skyelar | harlequin516: did you get the logging sorted out? |
| 22:27.31 | harlequin516 | I got logging, but I still can't figure it out. |
| 22:28.19 | harlequin516 | I do hava timer problem that keeps showing up in the logs, but nothing about the * DTMF causing hangup. |
| 22:29.14 | seebs | Hads, I was asking about your comment " cdrs have the correct time, but STRFTIME, SayUnixTime and Voicemail are using GMT" |
| 22:29.39 | seebs | Are they just giving GMT time values, or are they also specifically identifying their timestamps as GMT? |
| 22:29.56 | hads | OK, no it's just saying the time that would be correct as GMT - not with the actual timezone. |
| 22:30.58 | hads | At the moment I'm trying to figure out what the difference is between cdrs and logs (both correct) and dialplan functions and voicemail (which aren't correct). |
| 22:31.56 | Skyelar | harlequin516: do you have exitcontext defined anywhere in voicemail.conf |
| 22:33.00 | seebs | Hads, have you rebooted since mucking with the clock, or at least restarted asterisk? |
| 22:33.11 | seebs | If not, asterisk may have cached a time offset at startup. |
| 22:33.29 | hads | Yeah, I rebooted straight after I changed the hwclock to localtime. |
| 22:33.47 | harlequin516 | http://rafb.net/paste/results/FxcwJX30.html Here we go... It is interswting stuff I never saw before |
| 22:34.47 | Skyelar | harlequin516: err, that looks like the calling channel is hanging up |
| 22:35.46 | harlequin516 | Skyelar: No exitcontext setup, I just searched. |
| 22:35.55 | harlequin516 | Skyelar: No exitcontext setup, I just searched, in voicemail.conf anyways. |
| 22:35.55 | Skyelar | harlequin516: Aug 16 15:30:07 DEBUG[26215] chan_iax2.c: Immediately destroying 1, having received hangup |
| 22:36.23 | Skyelar | harlequin516: that means the IAX channel received a hangup - not that you hung up on it |
| 22:37.05 | Skyelar | harlequin516: you'll need to look further up the chain of command - how does the call get to where it is in the first place? From a Queue somewhere perhaps? |
| 22:37.09 | harlequin516 | Skyelar: Hmm.. I still can't tell whats happening |
| 22:37.35 | Skyelar | harlequin516: ie. who/what is IAX2/64.246.22.119:4569-1 ? |
| 22:38.13 | harlequin516 | Skyelar: POTS -> 3rd Party IAX PArtner -> VirtualPhoneline.com (IAX) -> My Asterisk Box |
| 22:39.37 | Skyelar | harlequin516: ok. If you call in to a phone (not VoiceMail), and have the caller on the PSTN side press '*', does it hang up? |
| 22:40.28 | harlequin516 | Yeah I call al the time to my cell phone voicemail, works fine there. |
| 22:40.54 | harlequin516 | Skyelar: I mean I am using my home phone to initaite the call. |
| 22:41.21 | Skyelar | harlequin516: that probably takes a different path. I'm talking about connecting an IP phone (or whatever) to your Asterisk box, calling it, and seeing what happens upon '*' press |
| 22:41.26 | hads | Hmm... cdrs and logs appear to use localtime_r directly where as the others use stdtime/localtime.c in the Asterisk source |
| 22:42.22 | harlequin516 | Okay Lemme try |
| 22:46.06 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-200-247.red.bezeqint.net) |
| 22:51.39 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 22:52.07 | harlequin516 | Back in two minutes need to use other room to test |
| 22:54.21 | dasenjo | Hi, I'm getting a SIP 480 message from a sip peer but is normally registered and have no DND activated, what can I do? |
| 22:55.25 | ManxPower | <PROTECTED> |
| 22:56.08 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 22:56.43 | *** join/#asterisk ginvent (n=ginvent@adsl-63-199-242-136.dsl.sndg02.pacbell.net) |
| 22:56.59 | ginvent | So what is the latest greatest voip service? I am still using teliax... any better? |
| 22:57.19 | dasenjo | ManxPower, Temporarily Unavailable |
| 22:57.21 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 22:57.26 | TripleFFFF | guys... |
| 22:57.29 | TripleFFFF | question |
| 22:57.47 | ManxPower | dasenjo, how do you know the phone is not on DND? |
| 22:57.49 | TripleFFFF | in AGI's.. sinc elike forever. .new callerid is set right ? .. i mean it changed.. |
| 22:58.05 | TripleFFFF | so how we set it know ? voipinfo no clues |
| 22:58.27 | Juggie | use Set |
| 22:58.50 | TripleFFFF | <PROTECTED> |
| 22:58.52 | TripleFFFF | using this |
| 22:59.37 | dasenjo | ManxPower, using database show, and seeing the phone .. is an eyebeam |
| 22:59.44 | Lyfe | anyone here use func_odbc that has a good example? |
| 22:59.57 | dasenjo | but .. the 480 message is intermitent |
| 23:00.01 | Juggie | TripleFFFF, you didnt read the docs! |
| 23:00.03 | harlequin516 | Skyelar: THANKS A BUNCH! That was the problem. vitualphoneline.com or their third party must be intercpting the *. |
| 23:00.03 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 23:00.07 | *** join/#asterisk |dennis| (n=dennis@200.32.215.83) |
| 23:00.25 | ManxPower | dasenjo, Uh, the only way to know is to be at the phone. |
| 23:00.32 | dasenjo | I can call the phone, the message is sent most on transfer or in PSTN incoming calls that dial the ext. |
| 23:00.32 | ManxPower | perhaps the phone is just on a call. |
| 23:00.37 | jbroome | wow, there really are the lyrics to louie louie in the * sounds dir. :) |
| 23:00.41 | TripleFFFF | lol |
| 23:00.50 | dasenjo | ManxPower, no, is not on a call |
| 23:00.56 | Juggie | TripleFFFF, its 'SET VARIABLE VAR contents' |
| 23:01.03 | Skyelar | harlequin516: that's what I figured from the hangup - happy to help :-) |
| 23:01.15 | ManxPower | Why not just use a damn AGI library |
| 23:01.34 | Juggie | secondly, callerid is now CALLERID(blah) where blah=some part of caller id |
| 23:02.29 | Juggie | so to do what your doing, you'd need to do "SET VARIABLE CALLERID(all) %s<%s>" |
| 23:02.59 | NDT | Grats Qwell...just saw the mailing list |
| 23:03.25 | dasenjo | ManxPower, so, any clue? |
| 23:03.49 | NDT | And russellb,murf, and file |
| 23:04.10 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:04.18 | ginvent | Where are the lyrics to louie louie? |
| 23:04.29 | Juggie | congrats for what? |
| 23:04.29 | file | or rather, waves |
| 23:04.34 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:04.48 | Juggie | file finally got his sex change? |
| 23:04.50 | NDT | jobs at digium |
| 23:04.52 | TripleFFFF | hmmm |
| 23:04.55 | ginvent | never mind... found it. |
| 23:04.56 | file | Juggie: email went out about the recent additions to the software team |
| 23:04.56 | ginvent | hmmm |
| 23:04.56 | TripleFFFF | thank jug |
| 23:05.02 | Juggie | ah. |
| 23:05.09 | Juggie | file your leaving NB? :) |
| 23:05.14 | Juggie | how will you cope! :P |
| 23:05.18 | file | if only it were that simple |
| 23:05.29 | TripleFFFF | hehehe |
| 23:05.30 | TripleFFFF | ok |
| 23:05.34 | TripleFFFF | sed 's@^.*<value>\([^<]*\).*@\1@g' |
| 23:05.40 | Juggie | hehe |
| 23:05.46 | TripleFFFF | tahts my sed.. im trying to ad a right trim on it.. |
| 23:05.49 | Juggie | my director approved *con today. |
| 23:05.51 | TripleFFFF | ;) |
| 23:06.00 | file | Juggie: yay |
| 23:06.02 | TripleFFFF | guess ill check #regex |
| 23:06.07 | Juggie | ohhh its not over yet. |
| 23:06.22 | Juggie | her boss has to approve it as well |
| 23:06.27 | Juggie | i just got past the 1st line of defense. |
| 23:07.02 | Juggie | i still have to make it past the 2nd line. |
| 23:07.14 | Juggie | which is the DG |
| 23:07.23 | Juggie | Director General |
| 23:07.26 | NDT | This cepstral takes like friggin 20 seconds to generate the file to say...wtf heh |
| 23:07.41 | *** part/#asterisk ginvent (n=ginvent@adsl-63-199-242-136.dsl.sndg02.pacbell.net) |
| 23:07.42 | Juggie | then your computer is slow |
| 23:07.58 | NDT | This one is on a 3.4ghz 2gb |
| 23:08.12 | Juggie | then something else is wrong ;) |
| 23:08.22 | Juggie | how big is the string of text you are trying to generate? |
| 23:08.44 | NDT | well one was rather large...the next one was only hello world and it didn't go much faster heh |
| 23:09.07 | Juggie | does your setup cache the output? |
| 23:09.10 | Juggie | so it only has to do it once? |
| 23:09.49 | NDT | trying to find teh settings for it heh |
| 23:10.01 | Juggie | i've never used cepstral only festival |
| 23:10.09 | Juggie | and it was much faster :) |
| 23:10.19 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 23:10.30 | Juggie | i thought russ already worked at digium btw? or is he now fulltime as opposed to digium & college |
| 23:10.31 | Zodiacal | anyone if theres a context that gets called when a user parks a call? |
| 23:10.38 | NDT | everyone I was talking to..tells me to try cepstral over festival...ughhhh |
| 23:11.02 | NDT | All i see is a settings file in the voice directory and don't see anything about caching it heh |
| 23:11.09 | ManxPower | NDT, Cepstral is cheap and works very good compared to festival |
| 23:11.39 | NDT | ManxPower: Know any settings tricks to speed it up? |
| 23:11.42 | ManxPower | NDT, You have some OTHER problem. Cepstral was always MUCH faster than realtime for me. |
| 23:11.43 | Juggie | cepstral is better, i am just saying my expirence is festival only. |
| 23:11.44 | NDT | over teh default install |
| 23:11.58 | Juggie | Zodiacal, see features.conf |
| 23:12.01 | ManxPower | NDT, I have not used Cepstral for several years. |
| 23:12.51 | Zodiacal | juggie yeah [parkedcalls] do you know what i would put in there to run when someone parks a call? |
| 23:12.54 | Zodiacal | i.e. what exten? |
| 23:13.00 | Zodiacal | just s? |
| 23:13.12 | NDT | Can't figure out where the problem lies...it is a prodcution system without any problems ever...will probably hear from them tomorrow I guess |
| 23:13.46 | intralanman | NDT: have you tried flite |
| 23:13.53 | NDT | no |
| 23:15.01 | intralanman | i saw an article on nerdvittles about it, and hence tried it..... it's much faster than plain ol festival |
| 23:15.11 | Zodiacal | juggie i tried doing this: [parkedcalls] (nextline) exten => s,1,verbose(testing) |
| 23:15.14 | Zodiacal | it doesn't seem to fire |
| 23:15.30 | intralanman | there's apparently some additional work to get custom voices to work well though |
| 23:15.51 | NDT | intralanman: it worked well with asterisk for you? |
| 23:16.14 | intralanman | oh yeah.... about 100x better than festival |
| 23:16.25 | intralanman | or text2wav and playback |
| 23:17.03 | Zodiacal | juggie any ideas? |
| 23:17.27 | NDT | hmmm...thanks...reading an article on it now |
| 23:17.44 | intralanman | NV also has an rpm |
| 23:18.16 | Juggie | Zodiacal, include => parkedcalls whereever you expect to use it |
| 23:18.21 | NDT | yeah it's for aah though |
| 23:19.27 | TripleFFFF | btw phpagi is broken |
| 23:19.30 | TripleFFFF | still has return $this->evaluate("SET CALLERID $cid"); |
| 23:19.31 | TripleFFFF | ;) |
| 23:19.52 | intralanman | you can use the rpm for whatever os you want |
| 23:20.08 | intralanman | there's tricks :) |
| 23:22.05 | *** join/#asterisk niter3 (n=niter3@d57-102-239.home.cgocable.net) |
| 23:22.12 | niter3 | hey guys, how can I put a password on an extension? |
| 23:24.44 | *** join/#asterisk bofh42 (n=bofh42@p548287C8.dip0.t-ipconnect.de) |
| 23:25.00 | ManxPower | niter3, "show application authenticate" |
| 23:25.00 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 23:25.35 | *** join/#asterisk Amilcar_ (n=xxxxx@201.34.202.17) |
| 23:26.05 | Zodiacal | juggie still not running... i must be doing soemthing dumb |
| 23:26.28 | niter3 | pardon? |
| 23:26.59 | Juggie | Zodiacal, i cant give any further advise, i've never used parking. |
| 23:27.14 | Juggie | niter3, from asterisk cli type what manx said. |
| 23:27.52 | Zodiacal | juggie thanks tho.. i'll play with it some more |
| 23:30.25 | niter3 | ok thanks |
| 23:34.00 | *** join/#asterisk anthm (n=anthm@000-448-895.area4.spcsdns.net) |
| 23:34.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:35.33 | CunningPike | Zodiacal: Are you using Asterisk call parking, or your phone's parking feature? |
| 23:35.52 | *** join/#asterisk MACscr (n=MACScr@adsl-75-23-104-12.dsl.peoril.sbcglobal.net) |
| 23:36.01 | CunningPike | Btw - anyone here with a Shaw email address that's having problems getting email? |
| 23:36.13 | MACscr | do some sip providers now allow callerid rewring for outgoing calls? |
| 23:36.23 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 23:36.25 | Zodiacal | cunningpike i tried pressing # then my parking number 70 |
| 23:36.31 | Zodiacal | i also tried my ciscos park softkey |
| 23:37.06 | CunningPike | Zodiacal: The first method is entirely controlled by asterisk - check the wiki for features.conf settings |
| 23:37.47 | Zodiacal | okie i'll go read some more thanks! |
| 23:37.56 | CunningPike | Zodiacal: The second is phone-dependent - I don't know about Cisco, but Polycoms call an extension called callpark |
| 23:39.42 | MACscr | if my sip provider doesnt provide virtual numbers outside the US, am i screwed? |
| 23:40.10 | *** join/#asterisk deb_user (n=debian_u@albuquerque.agroinnovations.com) |
| 23:40.11 | deb_user | ls |
| 23:40.20 | deb_user | whoops... |
| 23:40.39 | deb_user | does anyone know how to configure open wengo directly to asterisk without an open wengo account? |
| 23:40.46 | deb_user | can't be done...can it? |
| 23:40.51 | *** join/#asterisk adorah (n=Administ@87.68.173.125.cable.012.net.il) |
| 23:41.30 | deb_user | i'm having a really hard time finding a good sip softphone |
| 23:41.47 | deb_user | everything is designed to sign you up on some service providers network... |
| 23:41.53 | deb_user | anybody recommend anything? |
| 23:42.20 | adorah | <deb_user>www.xten.com |
| 23:42.43 | deb_user | adorah: I can't stand x-lite |
| 23:43.03 | deb_user | i don't like the user interface very much...the contact management is the pits |
| 23:43.43 | adorah | it is a good value for what u r paying for.. |
| 23:44.04 | deb_user | adorah: i wouldn't argue with that |
| 23:44.15 | deb_user | but, I'm hoping there might be something better |
| 23:46.35 | adorah | <deb_user>than google it.. |
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| 23:59.13 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
