| 00:02.59 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 00:03.03 | beek | sprbck: add a "WaitForRing(0)" before Answer(); |
| 00:08.34 | *** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 00:09.08 | FruitBasket | so.. when someone calls one of my numbers, the IVR has already started and they miss the first couple seconds. How can I fix that?... |
| 00:09.57 | beek | FruitBasket: Either add a Wait(1) before starting your IVR, or Playback(silence/1&your_file); |
| 00:09.59 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 00:10.07 | Akiyuki | Anyone available to help with this problem? http://pastebin.ca/1255680 this is my sip.conf and the message i am getting when trying to register to my sip provider |
| 00:10.30 | FruitBasket | I have ringing(), wait(7), answer()... |
| 00:10.43 | FruitBasket | and it still starts before the call is fully connected. |
| 00:10.50 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 00:11.08 | beek | FruitBasket: immediate=no on chan_dahdi.conf or zapata.conf? |
| 00:11.18 | FruitBasket | it's not hardware. |
| 00:11.49 | beek | DAHDI or Zaptel? |
| 00:12.06 | FruitBasket | beek: never heard of dahdi, but again.. it's not hardware. |
| 00:12.13 | FruitBasket | it's voip, all the way. |
| 00:12.59 | awk_r | Goto(voicemenu-custom-1|s|1) suppose to fail in trunk? |
| 00:13.07 | drmessano | Answer()? |
| 00:13.15 | Akiyuki | [TK]D-Fender: I am still stuck on the 120 sent request message. Is there anything I can do to continue diagnosing it? I dont even get a badusername/password combo |
| 00:13.18 | FruitBasket | drmessano: ? is that directed at me? |
| 00:13.23 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 00:13.30 | drmessano | You have those in order? |
| 00:13.32 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-f565beee5e44b2cb) |
| 00:13.40 | FruitBasket | drmessano: I have a wait before the answer.. |
| 00:14.03 | drmessano | You need to pastebin |
| 00:14.37 | beek | FruitBasket: setting verbose to 9 and watching what's happening isn't helpful? |
| 00:14.40 | FruitBasket | http://pastebin.com/d348acd98 |
| 00:21.44 | *** join/#asterisk primesoft (n=primesof@121.98.170.110) |
| 00:23.01 | primesoft | hi All, Has anyone setup a gui with asterisk 1.6.0.1? |
| 00:27.07 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 00:28.23 | *** part/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 00:29.40 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 00:29.55 | primesoft | I have installed asterisk 1.6 and asterisk-gui 2.0 but it appears that asterisk-gui 2.0 hasn't been upgraded to work with the dahdi changes |
| 00:29.56 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 00:31.34 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 00:37.41 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 00:46.25 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 00:49.19 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-2021a20fea66f99d) |
| 00:49.37 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 00:50.46 | BBHoss | anyone know why app_swift/cepstral would be really stuttery on Asterisk? |
| 00:51.43 | BBHoss | swift from the command line works fine, but in asterisk its totally fuxxed |
| 00:53.10 | jsmith | BBHoss: Is it sending audio in the proper format? |
| 00:53.23 | BBHoss | jsmith: everything is set to ulaw |
| 00:53.31 | BBHoss | i can hear the voice, its just really choppy |
| 00:53.36 | BBHoss | low cpu utilization too |
| 00:54.18 | jsmith | 8000 Hz, 8-bit samples? |
| 00:54.35 | BBHoss | http://www.mezzo.net/asterisk/app_swift.html |
| 00:54.43 | BBHoss | thats what im using |
| 00:54.56 | BBHoss | and i have the 8khz voice installed, david to be exact |
| 00:56.34 | BBHoss | does cepstral require hardware timers jsmith? |
| 00:57.01 | jsmith | I don't know, but I personally can't try that code due to licensing concerns, sorry :-( |
| 00:57.17 | BBHoss | jsmith: what code? |
| 00:57.31 | jsmith | BBHoss: The app_swift code |
| 00:57.34 | BBHoss | ahh |
| 00:59.20 | BBHoss | what license is it? |
| 00:59.43 | vk4akp | Anyone around that can help me with Chan_dahdi ?? |
| 01:02.46 | jsmith | vk4akp: Ask your question, and we'll try to help |
| 01:03.31 | vk4akp | OK TNX. |
| 01:03.47 | vk4akp | I am trying to find some documentation on Chan_dahdi |
| 01:03.52 | vk4akp | I have a TDM400P card. |
| 01:03.58 | vk4akp | And a Payphone on one of the ports. |
| 01:04.23 | jsmith | OK... keep going... (still haven't seen a question) |
| 01:04.26 | vk4akp | When receiving a call the phone rings. BUt it doesn't realise it's an incomming call when the receiver is picked up. |
| 01:04.41 | vk4akp | Polarity issues pulses signalling etc as possible causes. |
| 01:04.56 | vk4akp | The phone works fine for receive on an ATA or PSTn line. |
| 01:04.57 | jsmith | It's also possible that the pay-phone uses ground-start signaling instead of loop-start |
| 01:05.23 | vk4akp | I tried GS kS LS. |
| 01:05.40 | vk4akp | But I am not sure as to what I have to run to be sure teh zaptel is restarted. |
| 01:05.45 | *** join/#asterisk andresmujica (n=andresmu@190.25.102.69) |
| 01:05.53 | vk4akp | does ztcfg re read the chan_dahdi.conf ? |
| 01:08.05 | jsmith | vk4akp: The problem is, the TDM400P doesn't support ground-start |
| 01:08.13 | jsmith | vk4akp: No, you have to use dahdi_cfg -vv |
| 01:08.24 | *** part/#asterisk kornelak (n=karl@199.33.79.4) |
| 01:08.52 | primesoft | did you install the zaptel or the dahdi driver? |
| 01:09.03 | harry_v | am i right on this, digium has taken there cvs servers down? |
| 01:09.04 | vk4akp | Err.. Good question. |
| 01:09.11 | vk4akp | I think I am running th ezaptel driver. |
| 01:09.18 | vk4akp | Maybe I am configuring the wrong file??? |
| 01:09.32 | file | harry_v: we have not used CVS in over 2 years |
| 01:10.11 | jsmith | harry_v: Digium moved to SVN quite a while ago |
| 01:10.20 | harry_v | okay |
| 01:10.48 | vk4akp | Should I be looking at zapata.conf instead? |
| 01:10.57 | vk4akp | Also are you sure the card won't do ground start? |
| 01:11.12 | vk4akp | The Payphone is a USA desktop payphone. G-Tel 909 |
| 01:11.33 | primesoft | What version of asterisk are you running/intending to run? |
| 01:12.15 | jsmith | vk4akp: If you're using zaptel, you'll edit zaptel.conf, run ztcfg -vv, and edit zapata.conf |
| 01:12.32 | vk4akp | OK Thanks. |
| 01:12.37 | vk4akp | Yea I am running zaptel. |
| 01:12.38 | jsmith | vk4akp: If you're using DAHDI, you'll edit /etc/dahdi/system.conf, run dahdi_cfg -vv, and edit chan_dahdi.conf |
| 01:12.57 | vk4akp | is one better then the other for any reason? |
| 01:13.43 | jsmith | Zaptel has been replaced by DAHDI |
| 01:13.46 | primesoft | dahdi will be the only one supported in future versions |
| 01:13.53 | jsmith | (They're almost the same, but we had to change the name) |
| 01:13.59 | primesoft | 1.6 onwards |
| 01:14.38 | vk4akp | OK. |
| 01:14.41 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b2130951ba12dcd7) |
| 01:14.49 | vk4akp | I also run radio ports. So I hope the changes don't effect that. |
| 01:15.26 | vk4akp | So after I edit zapata.conf do I need to run anything for the new config to be adopted? |
| 01:15.34 | vk4akp | Or is a reload all that is necessary? |
| 01:16.13 | jsmith | If you add channels or change signalling, you have to restart for the changes to take effect |
| 01:16.36 | vk4akp | so just stop asterisk and restart? |
| 01:16.52 | vk4akp | Or do I need to run something else again too like ztcfg or something? |
| 01:18.56 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-180-74.dsl.stlsmo.sbcglobal.net) |
| 01:19.29 | LemensTS | hey guys, what format should i save a wav file for asterisk? |
| 01:19.45 | vk4akp | Also can anyone point me to some documentation on all the different settings in the zapata.conf file? and how to understand all the different signalling methods? |
| 01:19.47 | LemensTS | khyz/bit/(mono of course) |
| 01:20.07 | vk4akp | Also what signalling method is the comon one for USA and desktop payphones. (Normal line). |
| 01:20.12 | jsmith | ~thebook |
| 01:20.13 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 01:20.24 | vk4akp | No, I've read that. It's not in there. |
| 01:20.58 | jsmith | vk4akp: I wrote it. It's in there.... |
| 01:21.07 | vk4akp | OK. I'll look again. TNX. |
| 01:21.25 | vk4akp | Oh and if you really did write it. Thankyou so much for making it free. |
| 01:22.07 | jsmith | vk4akp: You're welcome. (and the standard signalling method for analog phones in the US is kewlstart, but often times payphones use different signaling |
| 01:22.42 | vk4akp | OK. Yes this payphone is just a desktop one for pub's etc. So it runs on a standard like like you would have at home. |
| 01:23.03 | vk4akp | We imported them here into Australia to use at VoIP displays to give things a bit of novelty. :) |
| 01:25.12 | vk4akp | Could I possibly have more luck with a Digium S101i maybe? |
| 01:26.06 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:27.14 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 01:27.34 | Katty | Qwell: 7 bars into 71 |
| 01:27.41 | Katty | Qwell: too tired to do much more. |
| 01:29.04 | *** join/#asterisk prodyan (n=ian@124.104.71.66) |
| 01:29.07 | prodyan | hello all |
| 01:29.15 | *** join/#asterisk edibrac (n=edibrac4@206.173.193.34.ptr.us.xo.net) |
| 01:29.28 | primesoft | Can anyone confirm that there is not a gui yet developed that works with asterisk 1.6? |
| 01:29.40 | WhiteWolf | best gui is your brain |
| 01:30.17 | tzanger | I should download asterisknow and try it |
| 01:30.31 | tzanger | I really would like to see that gui that they bought put in oss |
| 01:30.35 | tzanger | but I know they wouldn't do that :-) |
| 01:30.39 | tzanger | switchvox I think |
| 01:31.00 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 01:33.15 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 01:34.27 | primesoft | Unfortunately I can't package my brain |
| 01:34.30 | primesoft | :( |
| 01:37.35 | sprbck | beek: i've tried as much as wait(30) before answer, also upped rxgain on FXO port, tried a whole bunch of combinations of cidstart and cidsignalling.. nothing yet |
| 01:38.17 | beek | sprbck: This is DAHDI/Zaptel? |
| 01:38.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:38.57 | sprbck | Zaptel |
| 01:39.03 | sprbck | it's asterisk 1.2.29 |
| 01:39.08 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:39.14 | sprbck | it's an edgeBOX (www.edgebox.com) |
| 01:39.34 | harry_v | jsmith, you have the * cvn path? |
| 01:40.11 | sprbck | with defaults, i get this: |
| 01:40.12 | sprbck | Nov 13 20:38:28 NOTICE[24156]: chan_zap.c:6248 ss_thread: Got event 18 (Ring Begin)... |
| 01:40.16 | sprbck | Nov 13 20:38:29 NOTICE[24156]: chan_zap.c:6248 ss_thread: Got event 17 (Polarity Reversal)... |
| 01:40.19 | sprbck | Nov 13 20:38:29 NOTICE[24156]: chan_zap.c:6248 ss_thread: Got event 2 (Ring/Answered)... |
| 01:40.25 | harry_v | i mean svn |
| 01:40.31 | sprbck | if I start to play with settings, sometimes i get that, others i don't |
| 01:40.42 | sprbck | very ocasionally, i get this: |
| 01:40.48 | sprbck | <PROTECTED> |
| 01:40.51 | sprbck | Nov 13 20:38:03 WARNING[24152]: chan_zap.c:6280 ss_thread: CallerID feed failed: Success |
| 01:40.54 | sprbck | Nov 13 20:38:03 WARNING[24152]: chan_zap.c:6324 ss_thread: CallerID returned with error on channel 'Zap/5-1' |
| 01:42.19 | vk4akp | Can someone explain to me what RXwink= is? |
| 01:42.20 | sprbck | I'm starting to think that this baby is just not compatible with Verizon POTS |
| 01:42.43 | harry_v | got it |
| 01:45.58 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 01:47.01 | *** join/#asterisk mayo (n=mayo@a221-smpafs01.blockb-142.stargate.ca) |
| 01:47.07 | mayo | hello |
| 01:47.30 | *** join/#asterisk Raphael_S (n=t7DS@189.115.8.182) |
| 01:47.45 | *** part/#asterisk Raphael_S (n=t7DS@189.115.8.182) |
| 01:48.02 | mayo | is zaptel able to handle 2 cards are at a time? ie. if i plug in tdm400p and te220b, how do i know which card is first and map out the channels? |
| 01:48.59 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:50.41 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 01:50.55 | beek | sprbck: Do a google search on "callerid_feed: fsk_serie made mylen" -- I saw the answer to that earlier today. |
| 01:53.37 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 01:53.58 | sprbck | beek: but i rarely get that message.. all the other times i get those "ss_thread: Got event ..." |
| 01:54.29 | beek | sprbck: It could provide some clues. |
| 01:59.17 | *** join/#asterisk xuser (n=JJ@unaffiliated/xuser) |
| 02:01.31 | *** join/#asterisk jplank (n=gbove@reports.nyigc.net) |
| 02:01.39 | *** join/#asterisk mankash (n=rom10@CPE00062575886a-CM00186832000a.cpe.net.cable.rogers.com) |
| 02:02.11 | jplank | I know this isn't really a asterisk question, but I'm going nuts, can anyone explain this to me |
| 02:02.28 | jplank | when I enter "date" at the command line, I get |
| 02:02.31 | jplank | Fri Nov 14 04:54:52 EST 2008 |
| 02:02.53 | jplank | seems the timezone is right |
| 02:02.55 | jplank | but the time is wrong |
| 02:03.29 | mosty | jplank, when was the last time you set the time? |
| 02:03.30 | *** part/#asterisk mayo (n=mayo@a221-smpafs01.blockb-142.stargate.ca) |
| 02:03.39 | jplank | i'm using ntpd |
| 02:05.27 | jplank | date 11132110 would be now right? |
| 02:06.17 | mosty | jplank, check your ntpd logs, see if it's syncing correctly |
| 02:06.51 | seanbright | jplank: service ntpd stop ; ntpdate -s ; service ntpd start |
| 02:07.59 | jplank | let me check the logs |
| 02:08.07 | mankash | module load chan_sip.so is not working |
| 02:08.07 | jplank | I just manually set the time |
| 02:08.17 | seanbright | if the skew is too great, it won't set the time |
| 02:08.19 | jplank | then ran what seanbright told me, and its still showing the time I did |
| 02:08.54 | edibrac | is there a variable that will mean "the extension someone dialed" -- in other words, for exten => 54706,1,Macro(meetme,54706) I'd like to not have to type 54076 twice |
| 02:09.06 | seanbright | ${EXTEN} |
| 02:09.16 | edibrac | ah lol |
| 02:09.20 | seanbright | asterisk 101 |
| 02:09.22 | seanbright | :) |
| 02:09.32 | edibrac | sorry i knew about that but didn't connect it |
| 02:09.45 | jplank | seanbright: I'm sorry, skew? |
| 02:10.01 | seanbright | jplank: the difference between the time that is set and the time it actually is |
| 02:10.02 | harry_v | is there any problems sticking digium cars inside a u1 server case? |
| 02:10.23 | tzanger | I hope they're compact |
| 02:10.33 | mankash | module load chan_sip.so is not loading the sip module |
| 02:10.44 | mosty | mankash, check your logs |
| 02:10.59 | harry_v | tzanger you would obviosly know hat :) |
| 02:11.00 | harry_v | that |
| 02:11.43 | tzanger | harry_v: so long as the case can contain full-height cards, there shouldn't be a problem |
| 02:12.00 | tzanger | I don't recall seeing a single one that is "thicker" than anything that should fit in a PCI slot space |
| 02:12.23 | mankash | which loh |
| 02:12.25 | mankash | log |
| 02:12.36 | harry_v | there might be some u1 specs online. |
| 02:13.24 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-217-192.phlapa.east.verizon.net) |
| 02:17.10 | mankash | mosty which log should I check there are many log file in /var/log/asterisk |
| 02:17.20 | sprbck | beek: no luck :\ |
| 02:17.35 | sprbck | anyone around here using asterisk connected to a verizon line in the US?! |
| 02:18.01 | harry_v | i just talked to somone recently that did |
| 02:18.03 | mosty | mankash, look at the ones that have recent lines |
| 02:21.05 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 02:30.16 | harry_v | canot wait to upgrade my hardware. seems the compile time has increased with the newer versions of ast. |
| 02:33.45 | jsmith | sprbck: Yes, I've got an Asterisk system connected to a Verizon line |
| 02:35.26 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 02:36.35 | mankash | <PROTECTED> |
| 02:36.35 | mankash | <PROTECTED> |
| 02:36.46 | mankash | what is the meanig of thid |
| 02:36.50 | [TK]D-Fender | sprbck: pastebin your zapata.conf or chan_dahdi.conf, whic ever you are using |
| 02:36.52 | [TK]D-Fender | ~pb |
| 02:36.53 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:36.57 | [TK]D-Fender | ^^^^ |
| 02:37.16 | mankash | when I am trying to call 1 sip phon eto another I get this in the log |
| 02:37.25 | mankash | it is not able to call |
| 02:38.04 | [TK]D-Fender | mankash: Logs are next to worthless. You should be paying attention in CLI almost exclusively. |
| 02:38.17 | [TK]D-Fender | manxenable SIP debug, verbose 10, and pastebin a failed attempt |
| 02:38.31 | mankash | this i got on the cli only |
| 02:39.28 | [TK]D-Fender | mankash: Go set the 2 modes I have just told you and PB a call attempt |
| 02:43.30 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 02:45.52 | *** join/#asterisk ElCheapo (n=elcheapo@d199-126-36-20.abhsia.telus.net) |
| 02:46.42 | *** join/#asterisk sudhir492 (n=sudhir@adsl-210-53-102.mco.bellsouth.net) |
| 02:46.54 | sudhir492 | Hi All |
| 02:47.21 | sudhir492 | Is there still any use for PRI cards? |
| 02:47.25 | harry_v | TK, do you know if if there are compile times for ast online? |
| 02:47.26 | harry_v | :) |
| 02:47.59 | harry_v | sundhir492, what a silly question to ask |
| 02:48.36 | prodyan | when you buy TDM800p card, does it include FXO or FXS cards in it or do you have to purchase them separately? |
| 02:48.53 | [TK]D-Fender | harry_v: What are you using that you feel you should actually care so much? Noone else has ever really brought it up here... |
| 02:49.09 | [TK]D-Fender | prodyan: depends |
| 02:49.12 | harry_v | yea your right |
| 02:49.25 | jsmith | prodyan: You typically buy it with either all FXO or all FXS or some combination |
| 02:49.31 | [TK]D-Fender | prodyan: Check with the place you're buying from. Most won't sell a blank card. |
| 02:49.42 | prodyan | ahhh oki thanks jsmith and D-fender |
| 02:49.58 | prodyan | i thought it was blank when you buy it :D lolx |
| 03:03.26 | *** join/#asterisk esaym-acer (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 03:04.17 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 03:09.46 | *** join/#asterisk elGuille_wugro (n=guillerm@190.220.69.22) |
| 03:15.43 | *** part/#asterisk elGuille_wugro (n=guillerm@190.220.69.22) |
| 03:18.18 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 03:28.42 | pcrane | has anyone seen this before? |
| 03:28.43 | pcrane | Get index: w1g1: No such device |
| 03:28.43 | pcrane | w1g1: Failed to create connection |
| 03:28.52 | pcrane | when trying to get a sangoma a200 working? |
| 03:29.10 | mosty | have you followed the troubleshooting page on the sangoma wiki? |
| 03:29.14 | LemensTS | Anyone got tips on cli debugging when the asterisk places alot of calls? Im getting into bigger systems and its starting to become harder to troubleshoot |
| 03:32.51 | pcrane | yep |
| 03:32.55 | pcrane | nothing there that helps |
| 03:33.44 | pcrane | there's no mention of that phrase on the wiki |
| 03:34.35 | [TK]D-Fender | pcrane: pastebin your wanpipe configs |
| 03:34.44 | [TK]D-Fender | pcrane: and wanrouter status, etc |
| 03:34.46 | pcrane | master ~ # wanrouter start |
| 03:34.46 | pcrane | Warning: WAN_LOCK_DIR = /var/lock/subsys does not exist! |
| 03:34.47 | pcrane | Please update the WAN_LOCK_DIR in /etc/wanpipe/wanrouter.rc |
| 03:34.47 | pcrane | ERROR: Wanpipe configuration file not found: /etc/wanpipe/wanpipe1.conf |
| 03:34.47 | pcrane | wanrouter: Error, /etc/wanpipe/wanpipe1.conf not found! |
| 03:34.51 | pcrane | :( |
| 03:34.52 | [TK]D-Fender | PASTEBIN |
| 03:34.58 | pcrane | :p |
| 03:35.05 | [TK]D-Fender | And you're clearly missing configs period... |
| 03:35.21 | pcrane | wanrouter should set that up for me, right? |
| 03:35.24 | [TK]D-Fender | pcrane: Go fix that and make your wanpipe configs |
| 03:35.47 | [TK]D-Fender | pcrane: no, wanrouter does not configure your driver |
| 03:36.02 | pcrane | wancfg then |
| 03:36.07 | [TK]D-Fender | pcrane: better |
| 03:36.13 | pcrane | ;) |
| 03:36.34 | pcrane | which leads me back to the error message I was getting before: |
| 03:36.40 | pcrane | w1g1: Failed to create connection |
| 03:37.06 | pcrane | just for you [TK]D-Fender http://pastebin.com/m641ad193 :p |
| 03:38.31 | *** join/#asterisk xeno42 (n=nxeno42@r.omnipotent.net) |
| 03:39.09 | xeno42 | hey - quick question: Any ideas why "sip show users" would be empty, even though users are defined in users.conf ? |
| 03:39.10 | [TK]D-Fender | pcrane: you can't generate zaptel when you haven't even configured your card for WANPIPE |
| 03:39.32 | [TK]D-Fender | xeno42: "sip show peers" |
| 03:39.52 | xeno42 | sip show peers lists peers ok.. |
| 03:40.13 | xeno42 | but they're not being registered as users and not winding up in the right context then when making a call |
| 03:40.36 | [TK]D-Fender | xeno42: "user" is the TYPE and almost never used. |
| 03:40.44 | xeno42 | well friend then.. |
| 03:40.46 | xeno42 | actually |
| 03:40.52 | xeno42 | my colleague just moved hte user from users.conf to sip.conf.. |
| 03:40.54 | xeno42 | and now it works |
| 03:41.07 | xeno42 | i haven't configured asterisk in several years so i'm a bit out of touch |
| 03:41.15 | [TK]D-Fender | xeno42: Do ignore than and if you've got a specific problem include the pastebin of the failed call with SIP debug along with your configs masking only passwords |
| 03:41.22 | [TK]D-Fender | ~users.conf |
| 03:41.23 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 03:41.43 | xeno42 | users.conf didn't exist last time i setup asterisk |
| 03:42.07 | [TK]D-Fender | xeno42: its only there for the GUI.... which is another topic I dont want to have. |
| 03:42.14 | xeno42 | i dont' really understand what's supposed to be happening there |
| 03:42.21 | xeno42 | anyway i guess we have the solution ;-) thanks |
| 03:42.29 | [TK]D-Fender | xeno42: If you aren't using the GUI, ditch users.conf |
| 03:42.42 | xeno42 | well.. my colleague was using the gui |
| 03:42.55 | [TK]D-Fender | xeno42: Now if you want to debug your setup you know what to do. |
| 03:45.09 | xeno42 | he's much happier now :-) He'd spent quite a while banging his head against a wall |
| 03:46.31 | xeno42 | reminds me i need to upgrade that system i setup way back when... |
| 03:46.40 | xeno42 | fudge*CLI> show version |
| 03:46.40 | xeno42 | Asterisk 1.0.7 built by root@fudge on a i686 running Linux |
| 03:46.46 | xeno42 | methinks that might be out of date now ;-) |
| 03:48.02 | [TK]D-Fender | xeno42: No, its not dated.. more like CARBON DATED |
| 03:48.08 | xeno42 | hah |
| 03:48.57 | xeno42 | and yet it still works day after day |
| 03:57.59 | *** join/#asterisk NovceGuru (n=NovceGur@rrcs-70-62-198-142.central.biz.rr.com) |
| 03:59.28 | *** join/#asterisk bmg505 (n=leon@196-209-76-101-tbnb-esr-2.dynamic.isadsl.co.za) |
| 03:59.33 | hardwire | hi xeno42 |
| 04:00.06 | hardwire | you can carbon date something with a certain minimum age right? |
| 04:00.25 | hardwire | wonders what that minimum is. |
| 04:00.36 | drmessano | I think it's 3 hours |
| 04:00.55 | hardwire | drmessano: are you messing with me? |
| 04:00.58 | hardwire | I know you are. |
| 04:01.03 | [TK]D-Fender | hardwire: No... she ISN'T legal. |
| 04:01.07 | [TK]D-Fender | hardwire: PERV |
| 04:01.11 | drmessano | Because my wife is all like "Thats been sitting on the counter for FIVE hours".. and I am sure she's using some scientific method to come up with that |
| 04:01.15 | hardwire | [TK]D-Fender: found the happy sauce eh? |
| 04:01.25 | hardwire | drmessano: :P |
| 04:01.33 | drmessano | But under 3 hours, no |
| 04:01.50 | hardwire | I'm eating pppoe |
| 04:01.54 | hardwire | omnomnom |
| 04:02.03 | hardwire | isp won't let me remove their crappy cisco edge router |
| 04:02.07 | hardwire | and I'd love to firewall things correctly. |
| 04:02.21 | hardwire | I do believe I have what I need now |
| 04:02.48 | xeno42 | hey hardwire ... forgot you're an asterisk user too heh |
| 04:02.55 | hardwire | xeno42: user? |
| 04:03.02 | Micc | ever since I told this channel my asterisk ip address, its been crashing. and I'm getting warnings about no default context. |
| 04:03.04 | hardwire | I'm a fucking god. |
| 04:03.21 | Micc | Is there a patch for 1.2 that I need to install to prevent it from crashing? |
| 04:03.34 | NovceGuru | anybody know if you can buy the polycom ip650 without the ac adapter? |
| 04:03.39 | drmessano | ROFL |
| 04:03.40 | hardwire | Micc: you poor man |
| 04:03.44 | jsmith | NovceGuru: Yes, of course you can. |
| 04:03.48 | drmessano | You're kidding, right? |
| 04:03.55 | NovceGuru | I'm just sucking at google here |
| 04:03.55 | xeno42 | hardwire: i see. I missed all the minions cowering at your feet |
| 04:04.09 | drmessano | have you compiled with NOCRASH=VERY_YES ? |
| 04:04.16 | hardwire | lol |
| 04:04.20 | [TK]D-Fender | NovceGuru: I'm pretty certain it always comes with it |
| 04:04.21 | hardwire | Micc: how awful |
| 04:04.37 | Micc | hardwire, yes it is awful, so what can I do about it? |
| 04:04.39 | hardwire | xeno42: that's because they are so tiny. |
| 04:04.44 | NovceGuru | you can buy the 550 without...I have a poe switch, wonderd if I could save $20/phone |
| 04:04.44 | hardwire | Micc: firewall? |
| 04:04.49 | drmessano | superglue |
| 04:04.52 | jsmith | [TK]D-Fender: Actually, it doesn't. |
| 04:04.53 | drmessano | sniff deep |
| 04:04.54 | hardwire | Micc: try not to give out IP's that can get you hosed? |
| 04:05.06 | drmessano | Update to something 3 years more recent |
| 04:05.07 | hardwire | Micc: turn off guest access if you don't need it? |
| 04:05.11 | drmessano | Dont use SIP |
| 04:05.11 | [TK]D-Fender | jsmith: really? the 601/600 did... I'm surprised |
| 04:05.11 | Micc | hardwire, we have a cisco firewall but we need sip open for our customers. |
| 04:05.19 | hardwire | Micc: and? |
| 04:05.20 | drmessano | Hide in a cave |
| 04:05.20 | Micc | hardwire, how do I turn off guest access? |
| 04:05.29 | hardwire | check out the sample sip.conf |
| 04:05.34 | hardwire | it has information on that |
| 04:05.39 | drmessano | Switch from SIP to X100Ps |
| 04:05.49 | jsmith | [TK]D-Fender: I'm pretty sure you can get all their phones now without the adapters... I know the ones we use in our training classes we have to specifically order them *with* the power adapters. |
| 04:05.56 | hardwire | Micc: so.. yeh.. you can fix things using iptables or what not. |
| 04:06.07 | [TK]D-Fender | jsmith: the 320/330 yes... |
| 04:06.09 | hardwire | or even setting permit/deny in sip.conf |
| 04:06.14 | drmessano | X100P: "Let your imagination run wild til you run out of PCI slots" |
| 04:07.11 | [TK]D-Fender | Micc: "allowguest=no" and point the context to a dead-end |
| 04:07.13 | drmessano | ChinaCom X100P-ZOMG: "We didn't invent the modem, we just cloned the crap outta one" |
| 04:07.29 | [TK]D-Fender | drmessano: its too little to be allowed out by itself ;) |
| 04:07.41 | drmessano | lol |
| 04:07.44 | Micc | TKD-Fender, is not having default context good enough? |
| 04:07.51 | hardwire | who's buying IP650's? |
| 04:07.51 | drmessano | ... |
| 04:07.55 | hardwire | I'll take a few |
| 04:07.57 | hardwire | plz |
| 04:07.59 | [TK]D-Fender | Micc: have one... an empty one. |
| 04:08.11 | *** join/#asterisk pids (n=pids@194.sub-75-208-22.myvzw.com) |
| 04:08.25 | drmessano | Still not gonna keep you from getting pummeled |
| 04:08.29 | NovceGuru | not like I get to use these personally :P |
| 04:08.30 | hardwire | agreed |
| 04:08.38 | Micc | ok |
| 04:08.42 | hardwire | Micc: so.. you're logging the IP's that are screwing you over.. right? |
| 04:08.52 | drmessano | IF Asterisk is CRASHING due to a vuln, steps taken to prevent someone from USING your box may be useless |
| 04:08.56 | hardwire | and cross referencing those to your sip peers to see if they are related at all.. right? |
| 04:09.04 | Micc | hardwire, where would I see which IPs they are? |
| 04:09.12 | hardwire | turn on full logging |
| 04:09.20 | hardwire | then check out /var/log/asterisk/full |
| 04:09.26 | hardwire | or set the verbosities waaaay up |
| 04:11.25 | *** join/#asterisk emiller (n=ed@c-76-124-139-140.hsd1.pa.comcast.net) |
| 04:12.02 | emiller | i heard somewhere this is a web app that allows a user to hit an internal website to see who is on the phone within their network. any truth behind this? |
| 04:12.20 | hardwire | no |
| 04:12.40 | [TK]D-Fender | emiFOP |
| 04:12.47 | Micc | full doesn't exist. how do I turn on full logging? |
| 04:12.56 | [TK]D-Fender | emiller: FOP, and plenty of others I'm sure. I've written a few myself. |
| 04:12.56 | hardwire | Micc: you a newbiecakes? |
| 04:12.57 | hardwire | :P |
| 04:13.14 | emiller | ill check into it. thanks again |
| 04:13.16 | Micc | In this area I guess I am. |
| 04:13.18 | drmessano | micc: Thought you weren't a newb? |
| 04:13.27 | drmessano | You got all shitty about it the other day |
| 04:13.34 | Micc | I've never needed to look at full logs. |
| 04:13.44 | drmessano | I guess trixbox hides that |
| 04:13.48 | hardwire | Micc: http://www.voip-info.org/wiki/view/Asterisk+debugging |
| 04:13.48 | drmessano | :/ |
| 04:13.50 | hardwire | much love ^ |
| 04:14.33 | hardwire | don't bother searching for full.. you'll have to find the "Message Log" section |
| 04:14.43 | hardwire | Micc: what was that IP? |
| 04:14.43 | drmessano | If you've never needed to look at logs, and never had someone try to exploit your box before, I would love to get a cave next to yours |
| 04:14.58 | hardwire | my box is violated constantly. |
| 04:15.24 | drmessano | Do LOTS of reading here.. this basic sysadmin crap you need to know if you're gonna expose a SIP daemon to the net |
| 04:15.48 | hardwire | ~debugging |
| 04:15.48 | jbot | if debugging is the process of removing bugs, then programming must be the process of putting them in. |
| 04:15.53 | hardwire | ~debug |
| 04:15.53 | jbot | ACTION DeBuggers $1 |
| 04:16.01 | hardwire | ~logs |
| 04:16.01 | jbot | All conversations are logged to http://ibot.rikers.org/channel, where "channel" is replaced by the URL-encoded channel name, such as %23freenode for #freenode. Lines starting with spaces are not logged. |
| 04:16.07 | hardwire | lol |
| 04:16.15 | drmessano | ... and so are your IPs |
| 04:16.18 | hardwire | Micc: that means your IP is in those logs |
| 04:16.31 | drmessano | There's no place like 127.0.0.1 |
| 04:17.31 | Micc | I should know this stuff I wrote my own http logger. |
| 04:18.18 | drmessano | .... |
| 04:18.34 | drmessano | I ate breakfast with stallman yesterday myself |
| 04:18.41 | Micc | Ok, I've turned on full logging. |
| 04:18.54 | hardwire | woot? |
| 04:19.03 | Micc | looks like it requires a restart not just a reload though. |
| 04:19.15 | NovceGuru | drmessano: harhhar 127.0.0.1 is where the heart is |
| 04:19.51 | drmessano | WASSUP 127.0.0.1boyz |
| 04:19.59 | hardwire | Micc: logger reload |
| 04:20.15 | hardwire | Micc: logger show channels |
| 04:21.55 | NovceGuru | hm, the switchvox $660/yr minimum kinda sucks |
| 04:22.06 | Micc | I'm watching the full log now. haven't seen any unrecognized ip's yet. |
| 04:23.26 | Micc | Ok I found them. |
| 04:23.32 | Micc | <PROTECTED> |
| 04:23.40 | drmessano | OMG |
| 04:23.41 | [TK]D-Fender | NovceGuru: You have my permission to not use them :) |
| 04:23.45 | drmessano | Thats the IP of my toaster oven |
| 04:23.47 | *** join/#asterisk admin0 (n=admin0@bb116-14-118-8.singnet.com.sg) |
| 04:23.49 | drmessano | I am sooo sorry |
| 04:23.55 | drmessano | I musta left some toast in there |
| 04:24.12 | admin0 | hi all.. if i download the beta asterisknow, will it auto-update itself when its out of beta ? |
| 04:24.18 | hardwire | Micc: 209.112.194.200 is me |
| 04:24.24 | hardwire | I remember you now |
| 04:24.32 | NovceGuru | [TK]D-Fender: yeah, heh. Trying to find something I don't have to sweat my ass off supporting (basically not support) that has every X feature being requested |
| 04:24.35 | hardwire | I just have a qualify set on your IP |
| 04:24.39 | hardwire | removes it |
| 04:25.05 | Micc | hardwire, what you've been doing wouldn't crash my asterisk would it? |
| 04:25.10 | hardwire | btw.. you have pretty low latency with me |
| 04:25.14 | drmessano | Alaska |
| 04:25.22 | hardwire | Micc: no.. it only "pings" you so often |
| 04:25.36 | hardwire | Micc: I removed it. |
| 04:25.50 | hardwire | if that crashes your box then.. you need to ADD RAM or something |
| 04:25.56 | drmessano | 209-112-194-200.static.acsalaska.net |
| 04:25.56 | hardwire | maybetryswapmmkay? |
| 04:26.01 | hardwire | drmessano: that's me. |
| 04:26.07 | drmessano | Ah |
| 04:26.19 | drmessano | Effin Alaska |
| 04:26.20 | Micc | hardwire, its got like 4GB of ram. and its a dual proc xeon or something huge. |
| 04:26.33 | drmessano | What version of Asterisk? |
| 04:26.33 | hardwire | Micc: lol |
| 04:26.35 | Micc | hardwire, but asterisk is heavily hacked. |
| 04:26.36 | hardwire | 1.2 |
| 04:26.36 | Micc | 1.2 |
| 04:26.45 | drmessano | Not 1.2.NOTHING |
| 04:26.52 | drmessano | Which 1.2? |
| 04:26.54 | drmessano | 1.2.0? |
| 04:26.57 | drmessano | 1.2.1? |
| 04:26.59 | drmessano | 1.2.2? |
| 04:27.06 | hardwire | go on |
| 04:27.09 | Micc | Asterisk 1.2.11 built by root @ pbx2 on a i686 running Linux on 2008-01-10 08:55:58 UTC |
| 04:27.09 | hardwire | 1.2.3? |
| 04:27.18 | jblack | oh boy. |
| 04:27.25 | drmessano | omfg |
| 04:27.29 | NovceGuru | ofmg |
| 04:27.39 | drmessano | 1.2.11 is umm |
| 04:27.40 | hardwire | you guys heard that? |
| 04:27.44 | hardwire | I thought I was alone |
| 04:27.51 | drmessano | a couple months old or so |
| 04:28.00 | Micc | 1.2.11 is bad? |
| 04:28.02 | hardwire | I'm using 1.2.24 on one machine :) |
| 04:28.12 | drmessano | Aug 23, 2006 |
| 04:28.17 | drmessano | Only 2 years old+ |
| 04:28.31 | drmessano | I doubt they found any bugs/exploits |
| 04:28.39 | drmessano | So you should be cool |
| 04:28.39 | Micc | yeah, I can't update it because i've too heavily modified the source. |
| 04:28.47 | drmessano | ...... |
| 04:29.00 | drmessano | Oh and modified source |
| 04:29.09 | drmessano | and you're curious about random crashes? |
| 04:29.25 | hardwire | Micc: lol.. theres no patch for that. |
| 04:29.26 | Micc | drmessano, yeah its been running bug free for 2+ years. |
| 04:29.33 | drmessano | Not anymore |
| 04:29.34 | hardwire | Micc: test your hardware mebbe? |
| 04:29.37 | hardwire | you have t1's going into it? |
| 04:30.05 | hardwire | machines go south all the time |
| 04:30.07 | hardwire | and not in a good way. |
| 04:30.11 | drmessano | Sounds like he's got most of Romania and parts of Hong Kong going into it. |
| 04:30.35 | drmessano | Need to comb the logs further and start making a habit of blocking IPs |
| 04:31.13 | Micc | It crashed on odbc res. so I added a few more null checks in there. |
| 04:31.51 | drmessano | If there's a problem with 1.2.11 being exploited, or something else crashing your box over SIP with malformed packets due to source modifications, you're gonna need to start blocking by practice |
| 04:32.19 | [TK]D-Fender | Micc: What were your mods for? |
| 04:32.48 | Micc | drmessano, I'll see if it still has that problem. I don't want to block ports unless I have to. |
| 04:33.28 | Micc | TKD-Fender, some changes to app_voicemail, some custom apps. Some changes to originate AMI and a few other random things. |
| 04:33.55 | Micc | I wouldn't say its very heavily modified. just a few lines here and there and some custom apps. |
| 04:34.09 | Micc | Its not like I changed chan_sip or anything. |
| 04:34.12 | drmessano | Micc: I didnt say block PORTS |
| 04:34.15 | drmessano | I said IP addresses |
| 04:34.37 | drmessano | or indicated such |
| 04:34.37 | Micc | oh I can do that. |
| 04:34.53 | drmessano | This is gonna require the horrible shock of you learning to admin a box |
| 04:34.57 | Micc | now that I'm full logging, I'll find the bad IP if it exists. |
| 04:34.59 | drmessano | Blocking IPs is SOP |
| 04:35.03 | hardwire | Micc: just some changes? you should be fine then. |
| 04:35.03 | NovceGuru | block them at your firewall so your box doesn't ahve to deal with the packets at all (duh) |
| 04:35.08 | hardwire | lols |
| 04:35.10 | hardwire | I'm going home |
| 04:35.24 | drmessano | This is like sysadmin 101 crap here |
| 04:35.25 | hardwire | Micc: sorry things aren't working out.. go back to when things worked or test your hardware |
| 04:36.21 | Micc | I was running safe_asterisk before. |
| 04:36.34 | Micc | and there were a ton of asterisk processes. |
| 04:36.45 | Micc | it worked fine before I tried to fix that problem. |
| 04:36.56 | *** join/#asterisk synchris (n=synchris@athedsl-4380083.home.otenet.gr) |
| 04:37.07 | Micc | I've changed back to safe_asterisk, but fixed the multiple processes problem. |
| 04:37.13 | Micc | So maybe it'll be fine now. |
| 04:39.51 | drmessano | Ummm |
| 04:40.10 | drmessano | That was a FAR, FAR more important detail than this useless coincidental crap: <Micc> ever since I told this channel my asterisk ip address, its been crashing. and I'm getting warnings about no default context. |
| 04:40.27 | mosty | hehe |
| 04:40.50 | *** join/#asterisk FuriousGeorge (n=brian@ool-4354d18c.dyn.optonline.net) |
| 04:40.54 | FuriousGeorge | hey all |
| 04:41.26 | *** join/#asterisk synchris (n=synchris@athedsl-4380083.home.otenet.gr) |
| 04:43.29 | *** join/#asterisk synchris (n=synchris@athedsl-4380083.home.otenet.gr) |
| 04:43.30 | FuriousGeorge | I have a never ending saga with asterisk detecting hangup. I have one server which is in an office that is empty 80% of the time.... It is almost never used and is as reliable as a horse... |
| 04:44.12 | FuriousGeorge | Then there are the two servers that actually get used. No one has been able to help me. On my well behaved server it will just will hang 20 or 40 calls between two sip phones |
| 04:44.30 | FuriousGeorge | i can deal with that.... it just seems to mess up blf |
| 04:44.32 | drmessano | Which asterisk versions? |
| 04:44.38 | FuriousGeorge | latest as of last week |
| 04:44.48 | FuriousGeorge | then there is my poorly behaved server |
| 04:44.58 | drmessano | latest 1.0, 1.2, 1.4, 1.6, or trunk? |
| 04:45.00 | FuriousGeorge | http://forums.digium.com/viewtopic.php?p=120010#120010 |
| 04:45.06 | FuriousGeorge | 1.4.22 or so |
| 04:45.52 | FuriousGeorge | Asterisk 1.4.22 built by root @ claudia on a i686 running Linux on 2008-10-13 05:53:27 UTC |
| 04:46.14 | FuriousGeorge | ive tried 'side-grading' to 32bit, no help |
| 04:46.21 | FuriousGeorge | installed from scratch no help |
| 04:46.40 | drmessano | Whats different on the "good" box? |
| 04:46.42 | FuriousGeorge | this has happened since forever. I restart it daily but that guarentees me nothing except a fresh start |
| 04:47.00 | mosty | FuriousGeorge, what PSTN card? |
| 04:47.43 | FuriousGeorge | drmessano: its still 64bit? less zap channels? different service provider for analog lines |
| 04:48.04 | FuriousGeorge | mosty: sangoma a200 which didnt solve the problem |
| 04:48.29 | mosty | do you have ROIC on the lines? |
| 04:48.41 | FuriousGeorge | roic? |
| 04:49.12 | FuriousGeorge | luckily the deadlocks stopped with 1.4.x for me |
| 04:50.20 | FuriousGeorge | googling.. return on investment capial? i just changed the pstn provider to the same as the other server.. |
| 04:50.44 | FuriousGeorge | not sure when that takes effect. maybe it will help |
| 04:50.54 | mosty | maybe ROIC is an australian thing http://www.voip-info.org/wiki/view/Australia+Asterisk+Details |
| 04:51.11 | FuriousGeorge | mosty: seems like it only comes up relevant to .au |
| 05:21.51 | *** join/#asterisk the1_ (n=x@58.69.137.14) |
| 05:22.46 | *** join/#asterisk murdock_ut (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 05:24.38 | murdock_ut | Where do you change the from e-mail address of voicemail files. Is it the serveremail option in voicemail.conf? |
| 05:24.59 | FuriousGeorge | voicemail.conf, right? |
| 05:25.14 | FuriousGeorge | check the example |
| 05:25.26 | murdock_ut | FuriousGeorge: Yes. |
| 05:30.33 | LemensTS | What spec wav files will play with the Playback cmd? I tried making one at different khz/bits and ended up having to record it over the telephone with the Record cmd. |
| 05:31.18 | LemensTS | Right now i have customers uploading it into a web portal...not sure what the best way to make it error proof is |
| 05:36.33 | [TK]D-Fender | checkout time, later all |
| 06:02.39 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 06:08.17 | prodyan | hmm |
| 06:12.35 | *** join/#asterisk ahhsm (n=mike@24-182-108-29.dhcp.ftwo.tx.charter.com) |
| 06:13.55 | ahhsm | hey fellas, I moved to a new system and in the process went from 1.4.6 to 1.4.17 and now my music on hold doesn't seem to be working |
| 06:23.00 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 06:29.08 | ahhsm | http://pastie.org/private/yhm3mpktbcgep1uei1fvg |
| 06:30.01 | ahhsm | I'm at a loss for why it's not working on the new system.. |
| 06:37.35 | the1_ | anyone here know of a free alternative to http://www.tamos.com/products/commview/ a voip analyzer? |
| 06:40.18 | *** part/#asterisk jyfletcher (n=justin@2105ds4-ar.0.fullrate.dk) |
| 06:41.20 | jjshoe | what's a voip analyzer? |
| 06:41.21 | jjshoe | lol |
| 06:41.46 | jjshoe | this looks like a pretty interface to one of the many packet tracing tools like tcpdump |
| 06:42.48 | ahhsm | hmm.. so I setup a new context to test with and just do an Answer(), Ringing(), Wait(10), MusicOnHold(testing).. don't hear anything when I get connected though |
| 06:43.32 | ahhsm | rubs his chin |
| 06:47.54 | ahhsm | ok.. so taking the Answer out and just making it Ringing, I hear the ring now but still no hold music |
| 06:48.56 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 06:51.02 | drmessano | ROFL |
| 06:51.09 | drmessano | Wireshark with a skin |
| 06:57.14 | *** join/#asterisk aliraja (n=aliraja@202.125.156.122) |
| 06:58.51 | aliraja | Hi,is there any command in asterisk through which i can check the duration of active call. |
| 07:02.37 | *** join/#asterisk sosperec (n=david@office.axpnet.com) |
| 07:02.40 | sosperec | hello |
| 07:05.10 | jblack | that's it. I'm pissed at ld. |
| 07:05.23 | jblack | As in the linker, ld. Not long distance. |
| 07:05.26 | jblack | Hello. |
| 07:05.47 | jblack | aliraja: You can after the call, I don't know of a way during the call. |
| 07:06.07 | jblack | wouldn't be _too_ hard to make something to do it externally. |
| 07:11.03 | aliraja | jblack, is there some chan variable through which i can know |
| 07:11.44 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-494fc985c151ccb5) |
| 07:11.52 | ahhsm | hmm.. sln files should play find for moh out of the box, right? |
| 07:12.02 | ahhsm | isn't sln like the native format? |
| 07:12.47 | jblack | aliraja: How about, right before any dial, you log in a sql database when calls start, and after the dial, when they end. |
| 07:14.25 | aliraja | jblack,that sounds great but what i want is to hangup each call in queues after 5 minutes of talk time |
| 07:16.09 | aliraja | jblack, i think AbsoluteTimeout(seconds) will work for me let me try .. |
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| 07:35.05 | ahhsm | stupid firewall |
| 07:35.42 | ahhsm | finally figured it out.. there was a straggling alias for the old machine and so not all the firewall rules were updated |
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| 07:47.14 | drmessano | What do I need to enable in 1.6 to get TCP working? |
| 07:48.02 | drmessano | tcpenable = yes <--- only.. or tcpbindaddr and ???? |
| 07:49.39 | *** part/#asterisk ahhsm (n=mike@24-182-108-29.dhcp.ftwo.tx.charter.com) |
| 07:51.03 | *** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 07:52.02 | raasdnil | hey all. In the continuing saga of my inserting an asterisk box inbetween an NEC PBX and an existing (working) E1 I have made progress. |
| 07:52.17 | raasdnil | I can make an inbound call and it all works good. |
| 07:52.41 | raasdnil | DIDs work, line hunts work. That is from phone, through E1 through asterisk and to NEC PBX |
| 07:53.24 | raasdnil | but when I try to dial out on the NEC I am getting: |
| 07:53.30 | raasdnil | [Nov 15 05:35:53] WARNING[5460]: chan_dahdi.c:2642 dahdi_call: Unrecognized pridialplan NPI modifier: s |
| 07:53.32 | raasdnil | any ideas? |
| 07:53.45 | raasdnil | the nec context is: |
| 07:53.47 | raasdnil | [from-nec] |
| 07:53.47 | raasdnil | exten => s,1,Dial(DAHDI/g2/${EXTEN},,T) |
| 07:54.14 | raasdnil | kicks a tumbleweed |
| 07:56.42 | raasdnil | group 2 is the pri line |
| 07:58.10 | kaldemar | you can't use ${EXTEN} if you match to s. |
| 07:58.25 | raasdnil | do I have to use ARG1 ? |
| 07:58.49 | kaldemar | s is not a number, and the PRI assumes that is is setting the NPI bit for your numbering plan. |
| 07:59.07 | kaldemar | you have to use the number you're calling. |
| 07:59.07 | raasdnil | kaldemar: where did you read that? I am trying to find more books and materials to learn this stuff |
| 07:59.16 | raasdnil | so in this case I would do: |
| 07:59.27 | raasdnil | exten => _X.,1,Dial(DAHDI/g2/${EXTEN},,T) |
| 07:59.27 | kaldemar | sample configuration file for dahdi (chan_dahdi.conf) will tell you about that. |
| 08:00.00 | kaldemar | yes, if you're sending actual numbers to that context, then match to numbers with a pattern like that. |
| 08:00.48 | raasdnil | ok, cool. Thanks. I'll give it a shot soon. |
| 08:01.09 | kaldemar | now if you're wondering what the s in that exten => s,1,... is, it's a special extension called default extension. |
| 08:01.24 | kaldemar | you might want to check out those too if you're not familiar with them. |
| 08:01.46 | raasdnil | so you would use "s" then if the destination you were dialing has a specific extension it routes all calls to? |
| 08:02.02 | raasdnil | I'll go look it up too |
| 08:02.10 | kaldemar | you'd use s if you have no number. |
| 08:02.23 | raasdnil | right |
| 08:02.35 | kaldemar | but matching to s is not good practice if you have a number. |
| 08:03.07 | raasdnil | ok, that makes sense |
| 08:03.29 | raasdnil | that's why no number was getting matched in the EXTEN variable then, because s basically says 'there is no number' |
| 08:03.31 | raasdnil | I get it. |
| 08:04.01 | kaldemar | EXTEN always has whatever you have between => and ,1 in it. |
| 08:04.17 | kaldemar | i.e. the current extension. |
| 08:04.27 | raasdnil | right |
| 08:04.34 | raasdnil | thanks, that's clear now. |
| 08:04.56 | raasdnil | I'll go do my homework and try it out and come back with more intelligent questions or better yet, results :) |
| 08:05.18 | kaldemar | have fun |
| 08:06.48 | *** join/#asterisk itiliti (n=itiliti@75.150.198.1) |
| 08:11.28 | itiliti | I am trying to write to the CDR DB the DID that is called when a call comes in. What variable is that? |
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| 08:41.58 | angryuser | good day everyone |
| 08:43.41 | angryuser | can someone point me to a good predicitive dialer with the ability to distribute call to multiple asterisk installations ? |
| 08:43.46 | angryuser | call's |
| 08:46.48 | prodyan | i think asterisk can do predictive dialing |
| 08:46.59 | prodyan | but im just new so i don't really know :D |
| 08:49.43 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:52.09 | angryuser | prodyan: yes it can, if you progtam it ;) |
| 08:52.17 | angryuser | program* |
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| 08:54.38 | prodyan | and im sure that its not easy to do that |
| 08:56.00 | angryuser | well i think vicidal can do it , if someone know's better tell me |
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| 08:56.58 | raasdnil | vicidial sort of does predictive dialing. |
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| 08:57.14 | prodyan | vicidial is free? |
| 08:57.50 | raasdnil | yes |
| 08:58.07 | raasdnil | prodyan: http://astguiclient.sourceforge.net/vicidial.html |
| 08:58.09 | prodyan | oki googling.. wakoko |
| 08:58.18 | prodyan | ohh thanks |
| 08:58.19 | raasdnil | you'll want the "Scratch Install" |
| 08:59.34 | angryuser | prodyan: try vicidalnow first, easyer |
| 09:00.03 | prodyan | wew |
| 09:00.08 | prodyan | theres another version of it? |
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| 09:15.22 | angryuser | prodyan: vicidalnow it's like all in one, just like asterisknow |
| 09:17.27 | prodyan | ahh.. well im not really bent on trying vicidial wakeke just wanna see what it was and how it was related to asterisk :D |
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| 09:25.34 | prodyan | well, check out time --, |
| 09:25.59 | prodyan | bye all |
| 09:27.24 | *** join/#asterisk lclimber (n=lcanelon@212.183.204.76.static.user.ono.com) |
| 09:28.05 | lclimber | hello guys is there any web iax or sip client? |
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| 09:28.47 | fenlander | for a web iax client take a look at phonefromhere.com |
| 09:28.56 | lclimber | thanks fenlander |
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| 09:29.44 | fenlander | np - there are a few java web sip clients, but can't remember the names - alternative is a flash based sip client like tringme |
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| 09:34.51 | lclimber | fenlander. today i was hanging on the internet and i saw this webservice sponsored by a telemarketing company that allows you to put a link on your webpage to some kind of voip client, then on that client you put your phone number and that telemarketing company would comunicate you with the company which holds the link, the telemarketing company i calls you to your phonenumber and when you pick up it calls the link holder company |
| 09:34.51 | lclimber | <PROTECTED> |
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| 09:38.19 | fenlander | lclimber - sounds like a click 2 call type solution - a bit like Jahjah? |
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| 09:39.35 | lclimber | i want to try something like that, i guess you need a asterisk server connected to a phone line, when the client inputs his phonenumber then the webclient makes a call to a client o the company side and another call to the user |
| 09:40.43 | fenlander | yeah, you can use asterisk to make two outbound calls and connect them. It's pretty easy to do either using a "call file" or the AMI interface to originate the call |
| 09:41.03 | fenlander | asterisk calls the first number, and when answered calls the second and connects them |
| 09:41.17 | fenlander | it should be pretty well documented how to do that if you lookup call files |
| 09:43.13 | lclimber | cool |
| 09:43.52 | lclimber | let me check the book to see if i can find something related jejeje |
| 09:48.19 | lclimber | yeap, i found it, it looks pretty easy, now i am going to need a tdm card. |
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| 10:32.58 | stoffell | any idea on how it's best to track calls? like displaying the asterisk cdr table is nice, but that also displays calls that are not answered but are answered by another member of a queue.. |
| 10:33.43 | stoffell | something more like: x has called y, z has called x, etc.. currently we do that by analyzing "show channels concise" constantly... is there no more-fail-safe way? |
| 10:36.44 | psykx-out | personally I use mysql as a cdr back end and I then use php to pull the relevant data |
| 10:37.13 | psykx-out | I'm a php developer though so I'm biased in the way I do things |
| 10:41.19 | stoffell | psykx-out, correct, but can you maintain all relationships correctly? like incoming call from X (through zap/dahdi) that gets picked up by Y and then forwarded to Z ? |
| 10:50.15 | psykx-out | I don't but that not to say you can't |
| 10:55.15 | stoffell | okay.. gotta figure out a way to do that, then :-) |
| 10:55.19 | stoffell | thanks |
| 10:55.55 | psykx-out | I know you can customise the cdr |
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| 10:56.27 | psykx-out | I'd imagine you can record all forwarding that asterisk did |
| 10:56.54 | psykx-out | although I know my asterisk network uses forward supplied by the sip phones them selves |
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| 11:13.18 | stoffell | yeah, gotta find out a way to do it, maybe through dialplan... nice for a friday afternoon to figure that out :-) |
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| 11:19.28 | *** join/#asterisk mocker (i=ksexton@198.247.173.227) |
| 11:21.54 | mocker | Awesome, connecting to old BBSes in order to test my channel bank. |
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| 11:23.21 | Karlitoo | how do I add an extension which doen't go trough zaptel/dahdi |
| 11:23.26 | angryuser | ehh the time of bbs ;) |
| 11:23.27 | Karlitoo | but trough a trunk |
| 11:23.36 | Karlitoo | lan |
| 11:23.55 | mocker | angryuser: Can't beat it for testing line noise though. :) |
| 11:24.13 | mocker | If I can maintain my connection through a game of L.O.R.D. then I call it good! |
| 11:24.44 | angryuser | hehe |
| 11:24.56 | mocker | It amazes me that I remember so much of the AT command set. |
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| 11:51.30 | gannonb | i'm having an IAX issue... was working yesterday... then someone hosed the config more than likely.. anywho... 2 asterisk servers.. A and B... iax2 show peers... they see each other.... when i try to call (extensions.conf not changed from day before), always get the "Rejected connect attempt from X.X.X.X, who was trying to reach '*XXX@internal'" Checked to make sure the context names were correct.. etc... i'm beating my head against th |
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| 12:40.35 | mocker | gannonb: Do you have an internal context? |
| 12:40.42 | gannonb | yep... |
| 12:40.45 | mocker | gannonb: Is *XXX@internal actually what it's saying? |
| 12:40.58 | gannonb | so locally, the SIP phones use internal at site A... and that still works... |
| 12:43.07 | mocker | gannonb: Is *XXX@internal actually what it's saying? |
| 12:44.37 | gannonb | yes |
| 12:44.43 | mocker | That doesn't sound right. |
| 12:44.45 | gannonb | internal strips off the * and hits my regular extensions |
| 12:45.11 | gannonb | let me try without... brb |
| 12:45.13 | mocker | Ahh, so it really says something like *111@internal? |
| 12:45.36 | gannonb | eys |
| 12:45.46 | mocker | Try it w/o the star. |
| 12:46.00 | gannonb | same deal :( |
| 12:46.26 | gannonb | oddly enough.. this was all working for about 2 or 3 months now.. (with the *)... and today.. nothing |
| 12:47.14 | mocker | Can you pastebin your extensions.conf on the server that's rejecting the call? |
| 12:50.57 | gannonb | http://pastebin.com/d2ed331e2 (relevant info.. cut out the rest) |
| 12:53.00 | mocker | gannonb: Huh, don't see anything odd. |
| 12:53.23 | mocker | For fun can you just throw in an exten => 100,1,Playback(tt-monkeys) ? |
| 12:53.26 | mocker | And then try that? |
| 12:53.31 | gannonb | yeah.. i have another "asterisk" guy log in about 30 minutes ago... he's beating himself up as well... it's fun ;) |
| 12:53.39 | gannonb | ko.. will try that |
| 12:53.50 | mocker | Heh, you caught me at the end of a maintenance window so I'm pretty useless. :) |
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| 13:19.17 | pluesch0r | evening! what could be the reason for my sip connections to my asterisk server always breaking down after a certain amount of time? |
| 13:19.27 | pluesch0r | is there some default timeout that i'm not aware of? |
| 13:19.50 | bminish | nat going on someplace ? |
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| 13:24.05 | pluesch0r | bminish: yeah. |
| 13:24.18 | pluesch0r | server has a public ip, i'm connecting through NAT. |
| 13:24.25 | pluesch0r | i'm seeing some Remote host can't match request NOTIFY to call messages in the sip debug log. |
| 13:24.29 | pluesch0r | i've enabled nat support, though. |
| 13:26.59 | bminish | what is the NAT device and is it doing what it's supposed to |
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| 13:27.07 | spiekey | Hello! |
| 13:27.50 | spiekey | i would like to make a p2p call over tcp/ip with soft phones. Is there a easy way to do this with asterisk? |
| 13:27.51 | pluesch0r | bminish: the router is some linksys router .. |
| 13:27.58 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 13:28.05 | fcois93 | hello all |
| 13:28.08 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 13:28.23 | fcois93 | I need to create a little load balancing to 2 sip proxy |
| 13:28.25 | pluesch0r | bminish: what's bugging me is that the connection is pretty flakey. i've got this linksys ip phone (SPA921) which sometimes is able to connect .. and sometimes doesn't connect at all. |
| 13:28.41 | [TK]D-Fender | spiekey: Entirely doable, but * has a moderate learning curve to it. |
| 13:28.45 | pluesch0r | for example .. just right now, it was able to connect again. |
| 13:28.50 | pluesch0r | it's really weird. |
| 13:29.01 | fcois93 | I want to check if the serveur is up and send 50% calls to server1 and 50% calls to server2, knwo you how to do ? |
| 13:29.06 | [TK]D-Fender | spiekey: If all you want to do is talk between 2 people then sign up with ekiga.net or something |
| 13:29.18 | spiekey | [TK]D-Fender: well, will Asterisk NOW do the job? |
| 13:29.29 | fcois93 | I need to create a little load balancing to 2 sip proxy, I want to check if the serveur is up and send 50% calls to server1 and 50% calls to server2, knwo you how to do ? |
| 13:29.32 | [TK]D-Fender | spiekey: Same answer |
| 13:30.17 | mocker | ~thebook |
| 13:30.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 13:30.29 | bminish | pluesch0r, sounds like it's related to NAT any way you can take nat away to test it? |
| 13:30.44 | bminish | got to go for a but Be back later |
| 13:30.59 | pluesch0r | bminish: no, unfortunately not. |
| 13:31.12 | pluesch0r | bminish: could it also be that i'm running the asterisk inside of Xen? |
| 13:31.18 | pluesch0r | i've heard that there are some timing problems .. |
| 13:31.22 | angryuser | fcois93: hello with dns srv |
| 13:31.30 | [TK]D-Fender | pluesch0r: Entirely possible, also what about bandwidth? |
| 13:31.44 | pluesch0r | [TK]D-Fender: bandwidth can't be the issue. |
| 13:31.52 | angryuser | fcois93: or with openser "dispatcher" module |
| 13:32.01 | pluesch0r | server is 100mbit internet connection, office line is 4mbit sync |
| 13:32.20 | fcois93 | angryuser: my asterisk want to send to 2 proxy (openser) |
| 13:32.22 | pluesch0r | but .. shouldn't the timing problem only occur when actually transmitting voice data? |
| 13:33.02 | fcois93 | angryuser: I still use openser, I want to choose one of the 2 openser to send the calls |
| 13:33.28 | angryuser | fcois93: use dsn srv |
| 13:34.11 | fcois93 | angryuser: and how can I check if the serveur (openser) is alive ? |
| 13:35.30 | angryuser | fcois93: personnally i dont need load balancing, so i use heartbeat + openser , like that they share 1 ip adress, to check if openser is alive you could use sipsak |
| 13:36.59 | angryuser | fcois93: or whatchdog , or whatever you want (monit, custom script) |
| 13:37.40 | fcois93 | it is always the same problem here. when we need to know how to call a simple extension, everybody answer and say 'I am the boss in VOIP'!!! when we need more, someone say some bad things, and say 'why do you want it'!!!! in fact they dont know but stil say 'I am the boss in VOIP!'. the channel isnt what it was before!... |
| 13:42.03 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 13:44.06 | *** join/#asterisk KellyV (n=KellyV@ppp121-44-204-153.lns10.mel4.internode.on.net) |
| 13:44.51 | angryuser | fcois93: you are so funny :) |
| 13:45.09 | fcois93 | angryuser: seriously, have you an idea ? |
| 13:45.44 | angryuser | fcois93: yes i said use sipsak |
| 13:47.05 | angryuser | if sipsak's request fail => server dead > modify srv record |
| 13:47.29 | [TK]D-Fender | angryuser: But what makes sure that sipsak is running? We need that on HA too! |
| 13:47.36 | fcois93 | angryuser: in the left-side, I have a lots of asterisk. on the right side I have 2 openser which loadbalance to the asterisks servers. when asterisk receive a call, I need that it choose one of the 2 openser! |
| 13:47.39 | KellyV | hi guys, is this the right place to ask about an aa50 and polycom phones? I am an asterisk newb, but am slowly figuring things out |
| 13:48.10 | fcois93 | angryuser: sipask work in asterisk ? |
| 13:48.10 | [TK]D-Fender | KellyV: General questions yes. AA50 specific stuff is supported by digium |
| 13:48.39 | [TK]D-Fender | fcois93: Sipsak is a separate progeam it is not "in" Asterisk |
| 13:49.13 | fcois93 | so, we can't control if a srv is up before doing DIAL ? |
| 13:49.31 | drmessano | I am the boss in VOIP! |
| 13:49.34 | *** join/#asterisk IPkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 13:49.37 | drmessano | ??? |
| 13:49.43 | angryuser | fcois93: you said " when asterisk receive a call, I need that it choose one of the 2 openser" when asterisk receive a call it does not choose anything, it receives a call, be clear |
| 13:50.00 | IPkaf | ?????? |
| 13:50.03 | IPkaf | hi to all |
| 13:51.05 | fcois93 | angryuser: asterisk receive a call from a user (for example), asterisk have to send the call to an openser 50% calls to the 1 50%calls to the 2; after having check if the openser is up |
| 13:51.11 | IPkaf | i got pap2t adaptor |
| 13:51.21 | KellyV | Fair enough, then just a quick one, any idea why my polycom phones fail to register (eg cant even call local extensions) when my wan goes down and asterisk cant register my sip trunk. The phones obviously on the same subnet as the aa50. It appears to have something todo with the line register=6139095xxxx:password@sip.pennytel.com/6139095xxxx |
| 13:51.54 | IPkaf | i try to access to my pap2t adaptator over telnet |
| 13:52.11 | [TK]D-Fender | fcois93: you want * to be the distributor? |
| 13:52.40 | fcois93 | yes ! :-) |
| 13:52.52 | [TK]D-Fender | KellyV: Sounds like perhaps * is locking up for lack of DNS |
| 13:53.05 | [TK]D-Fender | KellyV: if thats the only thing that causes it |
| 13:53.16 | angryuser | fcois93: ok, so in normal operation load balance outgoing on 2 server's and if any fail use another one ? if yes just setup 2 trunks and change every time on each call the outgoing trunk (server1 server2) and if call failed (timeout) use another trunk |
| 13:53.34 | [TK]D-Fender | fcois93: "core show functions like "GROUP" <- this is DIALPLAN. get to work... |
| 13:53.53 | [TK]D-Fender | fcois93: * is a shitty load balancing solution |
| 13:54.24 | fcois93 | [TK]D-Fender: yes, but how it check if the servers are up |
| 13:54.36 | IPkaf | salut fcois93 |
| 13:54.40 | angryuser | fcois93: really simple, but as fender said i would recomment the use of separate outgoing proxy |
| 13:54.44 | [TK]D-Fender | fcois93: Have another server connect to it. |
| 13:54.52 | KellyV | [TK]D-Fender Thats what i was thinking, is there anyway to instruct it timeout? |
| 13:55.02 | IPkaf | do u spoke frinchis? |
| 13:55.11 | [TK]D-Fender | KellyV: THAT would be an AA50 specific question. Go call up Digium support |
| 13:56.19 | angryuser | fcois93: but tell me, why do you want load balance outgoing to multiple openser ? it's unusial |
| 13:56.55 | fcois93 | angryuser: just imagin if an openser is down, just for that! |
| 13:57.11 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 13:57.28 | angryuser | fcois93: use heartbeat, exellent solution |
| 13:57.57 | KellyV | [TK]D-Fender Damn. So you dont think it has anything todo with that register= line in the general section of sip.conf affecting the registration of the polycom phones, apart from asterisk locking up due to no DNS? From what I have read it shouldnt |
| 13:58.02 | IPkaf | hi fcois93 |
| 13:58.17 | IPkaf | where r u from ? |
| 13:58.35 | fcois93 | IPkaf: FRANCE, Paris |
| 13:58.35 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:58.40 | fcois93 | IPkaf: why? |
| 13:58.50 | angryuser | encore un parisien :) |
| 13:59.11 | IPkaf | yes |
| 13:59.25 | fcois93 | on continu en francais ? |
| 13:59.27 | IPkaf | what ur isp provider ? |
| 13:59.31 | [TK]D-Fender | KellyV: That has no impact on your phone directly. |
| 13:59.50 | IPkaf | oui si tu veux |
| 14:00.03 | angryuser | fcois93: no it is not polite to other's |
| 14:00.13 | IPkaf | ok |
| 14:00.20 | [TK]D-Fender | Y-a trop de francophones! Va t'ens deja! |
| 14:00.25 | IPkaf | acropolis telecom |
| 14:00.35 | fcois93 | oui |
| 14:00.41 | [TK]D-Fender | KellyV: Go look at the SIP debug for your Polycom itself. |
| 14:00.43 | IPkaf | i never heard it before |
| 14:00.47 | IPkaf | ??? |
| 14:00.48 | fcois93 | t'es fort ^^ quel hacker celuila ! |
| 14:00.53 | angryuser | or paranoid users like fender ;) |
| 14:01.35 | IPkaf | oui dit moi t'habite àbagnolet ? |
| 14:01.52 | KellyV | [TK]D-Fender: Thanks for that info, I will do some more reading and wrestling with * |
| 14:02.02 | fcois93 | IPkaf: yes |
| 14:02.15 | IPkaf | trop fort , |
| 14:02.32 | IPkaf | acropolis telecom i never heard it before ? |
| 14:02.33 | fcois93 | IPkaf: super complex àtrouver ça... ^^ |
| 14:03.15 | fcois93 | we are the second french operator for companies |
| 14:03.32 | IPkaf | i live too in bagnolet |
| 14:03.43 | tzanger | woot |
| 14:03.46 | tzanger | got my adit600 ringing again |
| 14:03.52 | IPkaf | pleaser to see u |
| 14:03.55 | tzanger | stupid power supply burned out the 5 1-ohm current sense resistors |
| 14:04.10 | *** join/#asterisk PrimeHaxor (n=sonamon@189-19-221-236.dsl.telesp.net.br) |
| 14:04.21 | Katty | morning |
| 14:04.38 | PrimeHaxor | hi! i need a little help with some issues on asterisk |
| 14:04.39 | fcois93 | IPkaf: so, no answers to distibute calls to 2 openser ? |
| 14:05.30 | Katty | Qwell: mew. |
| 14:05.31 | Katty | [TK]D-Fender: Mew. |
| 14:05.41 | [TK]D-Fender | Katty: Mew |
| 14:06.03 | [TK]D-Fender | fcois93: OpenSER is the kind of thing that should be distributing calls. |
| 14:06.07 | PrimeHaxor | [Nov 14 11:31:10] WARNING[5350] chan_sip.c: Maximum retries exceeded on transmission |
| 14:06.20 | PrimeHaxor | what should happing with this WARNING |
| 14:06.24 | PrimeHaxor | the calls are dropping |
| 14:06.25 | [TK]D-Fender | fcois93: and I told you what to look for if you wanted * to do it... this is DIALPLAN. use the group_count stuff |
| 14:06.35 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.3) |
| 14:06.41 | fcois93 | [TK]D-Fender: I wont use a openser to distribute calls to 2 openser ! |
| 14:06.43 | [TK]D-Fender | PrimeHaxor: Dropping calls that are already in progress? |
| 14:07.03 | [TK]D-Fender | fcois93: Why not? * is not a PROXY. |
| 14:07.11 | PrimeHaxor | [TK]D-Fender, yes |
| 14:07.30 | [TK]D-Fender | fcois93: * for this role is like using a screwdriver to hammer in a NAIL |
| 14:07.52 | [TK]D-Fender | PrimeHaxor: Networking / load issue most likely. |
| 14:07.57 | fcois93 | [TK]D-Fender: it will asterisk-openser-2opensers .... bad solution |
| 14:08.21 | [TK]D-Fender | fcois93: If all * is doing is distributing a calls then it shouldn't be in the picture |
| 14:08.48 | PrimeHaxor | [TK]D-Fender, ty, i'll configure iptables to make qos |
| 14:09.10 | [TK]D-Fender | PrimeHaxor: Don't forget you can only prioritize what you transmit. |
| 14:09.17 | fcois93 | [TK]D-Fender: all problems have its solutions |
| 14:09.36 | [TK]D-Fender | fcois93: Available in a wide variety of calibers! |
| 14:09.44 | fcois93 | I will have a look |
| 14:09.59 | fcois93 | qui est français que je sache pour plustard ? |
| 14:10.23 | angryuser | 1 |
| 14:10.45 | *** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 14:11.36 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:11.36 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:13.16 | Katty | eyes lmadsen |
| 14:15.01 | lmadsen | foots Katty |
| 14:15.31 | *** join/#asterisk remont (n=reMont@200.73.192.226) |
| 14:15.35 | creativx | readies the fist |
| 14:16.39 | Katty | creativx: i think you skeered him off :< |
| 14:17.39 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 14:17.55 | creativx | oops |
| 14:17.55 | creativx | hehe |
| 14:18.25 | *** join/#asterisk Uatec (i=57c24701@gateway/web/ajax/mibbit.com/x-6aeff648aa52968c) |
| 14:19.17 | fcois93 | angryuser: when you talk about sipsak, I am able to do exten=> X,n,system(sipsak....) and receive a 200 ok ... ? |
| 14:19.19 | Katty | sigh. another long day. |
| 14:19.23 | remont | hi, there are any way to authenticate user sip by ldap, if i have asterisk 1.4.22, i read ldapget, but i don't saw sip users |
| 14:19.52 | creativx | Katty: FUW is in effect |
| 14:21.04 | Katty | creativx: my parser does not seem to be working. can you translate? |
| 14:21.37 | creativx | Katty: fuw = fuck you weekend :-) |
| 14:22.18 | angryuser | fcois93: no it's external |
| 14:22.56 | fcois93 | angryuser: how can I use it ? |
| 14:23.02 | angryuser | fcois93: anyway it is not a good idea to do a load balancing with asterisk |
| 14:23.04 | fcois93 | angryuser: have you an example ? |
| 14:23.08 | Katty | creativx: hrm. |
| 14:23.10 | IPkaf | ok |
| 14:23.31 | creativx | Katty: or is it not? does long day mean bye bye work hello weekend. = |
| 14:23.41 | fcois93 | angryuser: it isnt a good idea to use an other openser to send to other openser (c'est débile) |
| 14:23.53 | Katty | creativx: sorry, yes. i plan on having a lovely weekend of WoWing. |
| 14:24.11 | Katty | creativx: i've gotten only a handful of hours of sleep...not quite parsing things properly yet |
| 14:24.22 | IPkaf | is there anyone ihere who success to connect to his pap2T over ssh ? |
| 14:24.27 | angryuser | fcois93: the good idea is to change the config of existing openser servers |
| 14:24.47 | PrimeHaxor | have some opensource software voip if i can make conferences? |
| 14:24.47 | creativx | Katty: hehe.. i see |
| 14:24.50 | fcois93 | angryuser: for what ? |
| 14:25.32 | Katty | creativx: it took 12 hours to get 1 lvl. |
| 14:25.34 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 14:25.43 | angryuser | fcois93: is your openser server able to handle all traffic ? |
| 14:25.56 | Katty | creativx: ryan me and another friend of ours went out and bought the WOTLK expansion around midnight. |
| 14:26.01 | angryuser | fcois93: i mean only one ? |
| 14:26.22 | Katty | creativx: went home, loaded the software and patches, and immediately started lvling. around 6 in the morning i had a 3hr nap, and then we kept going. |
| 14:26.25 | creativx | Katty: lol, cant relate to wow sorry :) im gonna paint some walls this weekend.. all the fun and games! |
| 14:26.37 | fcois93 | angryuser: yes it can |
| 14:26.42 | [TK]D-Fender | IPkaf: ssh = tcp, sip+rtp = udp |
| 14:26.48 | [TK]D-Fender | IPkaf: dOESN'T ADD UP |
| 14:26.49 | Katty | creativx: it's lots of fun. |
| 14:26.52 | angryuser | fcois93: so use heartbeat! |
| 14:27.21 | creativx | Katty: doesnt sound healthy :D |
| 14:27.22 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:27.41 | fcois93 | angryuser: what will it do ? |
| 14:27.45 | angryuser | fcois93: and separate the outgoing and incoming processing in openser's cfg |
| 14:27.48 | *** join/#asterisk UnixDawg (n=Unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 14:28.15 | stencil | hello UnixDawg |
| 14:28.33 | UnixDawg | hey |
| 14:28.33 | angryuser | fcois93: you will have ony 1 ip for both server's |
| 14:28.55 | fcois93 | angryuser: ok, like a dns balancing ? |
| 14:29.41 | *** join/#asterisk fudpucker (n=jircii@75.151.177.173) |
| 14:29.45 | angryuser | no with heartbeat only one server will be used all teh time |
| 14:30.19 | angryuser | fcois93: you have the key's, go read about heartbeat |
| 14:31.32 | fcois93 | angryuser: I dont want heartbeat |
| 14:31.56 | angryuser | fcois93: then i cant help you |
| 14:32.07 | Katty | don't tell mah heart. |
| 14:32.11 | Katty | mah achey breaky heart. |
| 14:32.32 | fudpucker | i am trying to get asterisk to use unixodbc, and the datasource is setup, and i can connect with isql, but when asterisk goes to write a cdr record, it errors out with SQLconnect -1. is there a way i can get the real reason? |
| 14:32.42 | fcois93 | angryuser: ok thank you |
| 14:33.14 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 14:34.07 | k-man | a while ago i asked on here for recommendations of brands of sip phones and a few were recommended, Linksys being one of them, but what are the others? |
| 14:34.23 | fudpucker | i am a fan of the cisco phones.... |
| 14:34.38 | [TK]D-Fender | k-man: Polycom > all |
| 14:34.48 | [TK]D-Fender | k-man: Linksys is a fairly solid second choice. |
| 14:34.49 | k-man | ah, thanks |
| 14:34.53 | k-man | ok |
| 14:35.07 | [TK]D-Fender | k-man: Aastra and Snom are kinda tied at 3rd |
| 14:35.09 | stencil | k-man: Snom if you want opensource phones! |
| 14:35.27 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-158-157.lns10.mel4.internode.on.net) |
| 14:35.33 | k-man | oh, thats interesting, thanks |
| 14:35.59 | Katty | k-man: polycom |
| 14:36.02 | Katty | k-man: polycom |
| 14:36.04 | Katty | k-man: AND |
| 14:36.06 | Katty | k-man: polycom. |
| 14:36.23 | Katty | k-man: ^- my top 3. |
| 14:36.23 | k-man | cool, thanks for the suggestions guys |
| 14:36.58 | WimpMan | stencil: Open source? Really? or only the basic OS with closed source phone app as usual? |
| 14:37.32 | stix_ | Have any of you experienced a problem like this: When I reload asterisk and call the system (from the outsite) straight away - I cannot get through to the system, so I hang up. But then 10 secs later my call shows up on the cli. That's a bit late |
| 14:38.04 | stencil | WimpMan: the underlying OS is Linux |
| 14:38.41 | WimpMan | stencil: That prabaly the case on most devices. |
| 14:38.56 | k-man | if one wanted to build an asterisk server for 80 users, what kind of server would you need? |
| 14:39.00 | WimpMan | Or rather nearly all. |
| 14:39.18 | fudpucker | also with my cdr_odbc issue, it doesn't seem to be logging anything. |
| 14:41.44 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:43.35 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 14:43.55 | fcois93 | how can I see if the user is reachable before doing dial ? |
| 14:44.55 | *** join/#asterisk Segnale007 (n=Pietro@host130-254-dynamic.9-79-r.retail.telecomitalia.it) |
| 14:46.08 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 14:47.14 | Dr-Linux|home | Can i'm using AGI, my users make calls through my Asterisk agi script, i want to provide "DIAL TONE" to my users? |
| 14:47.17 | Dr-Linux|home | any suggestions? |
| 14:47.34 | Dr-Linux|home | normally we use DISA in dialplan, but not sure what i can use in AGI script |
| 14:50.17 | Zeeek | {{{{{{{{{{{{{{{{{{Katty}}}}}}}}}}}}}}}} |
| 14:50.27 | k-man | night all |
| 14:50.57 | Zeeek | Wqrning, unbalanced braces in above expression of admiration |
| 14:51.05 | IPkaf | ok good nihgght all |
| 14:51.06 | Zeeek | azerty |
| 14:52.11 | *** part/#asterisk IPkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 14:55.07 | bcrochet | astrundir is not being honored in 1.6.0.1. Any ideas why not? |
| 15:03.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:05.11 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 15:05.33 | *** join/#asterisk [SySteM] (n=antoine@aqu33-6-88-168-80-163.fbx.proxad.net) |
| 15:05.38 | [SySteM] | Hi all |
| 15:05.54 | [SySteM] | I buy a swissvoice IP10s to connect to my asterisk server.. but |
| 15:06.10 | [SySteM] | On the console : Comfort noise support incomplete in Asterisk.Please turn off on client if possible. Client IP: |
| 15:06.49 | fudpucker | i am trying to get asterisk to use unixodbc, and the datasource is setup, and i can connect with isql, but when asterisk goes to write a cdr record, it errors out with SQLconnect -1. is there a way i can get the real reason? |
| 15:07.37 | sosperec | is there any way to get the first 3 character of a variable? Like cut -c -3 in shell |
| 15:08.42 | awk_r | sosperec, there is |
| 15:09.09 | awk_r | (getting link) |
| 15:09.17 | sosperec | awk_r: and what's it? :) |
| 15:09.53 | awk_r | sosperec, i could tell ya, but i figured teaching you variable basics would be a better idea :-) |
| 15:10.02 | sosperec | Thank You. :) |
| 15:10.06 | awk_r | sosperec, http://www.voip-info.org/wiki-Asterisk+variables |
| 15:10.17 | awk_r | under the 'Substrings' section |
| 15:10.32 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 15:11.04 | sosperec | Thank You! That's what I was looking for. |
| 15:11.16 | awk_r | np |
| 15:11.45 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:11.51 | bcrochet | Any ideas why astrundir (now set to /var/run/asterisk) is being ignore in 1.6.0.1? * is still insisting on /var/run/asterisk.ctl |
| 15:16.03 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-27556927afd4fbac) |
| 15:16.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:24.58 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:25.03 | *** join/#asterisk magumbade (n=magumbad@p5497E386.dip.t-dialin.net) |
| 15:27.16 | *** join/#asterisk magumbade (n=magumbad@p5497E386.dip.t-dialin.net) |
| 15:27.20 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-c3f8ab957401802a) |
| 15:27.21 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:28.50 | Katty | deeeeeeeeeeeeeeeeeeeeeeeeeewayne!! |
| 15:29.46 | Deeewayne | hugs Katty |
| 15:32.00 | Katty | hugs Deeewayne |
| 15:32.09 | *** part/#asterisk bcrochet (n=bcrochet@cpe-069-132-204-022.carolina.res.rr.com) |
| 15:38.59 | *** join/#asterisk tkbeat (n=tk@80.64.182.204) |
| 15:39.53 | jameswf | reluctantly thinks he is going to switch from KDE to gnome |
| 15:39.57 | jameswf | evil bastards |
| 15:45.34 | Zeeek | Katty... was it something I said? |
| 15:45.57 | Katty | Zeeek: mew? |
| 15:46.06 | Zeeek | no hug :( |
| 15:46.22 | Zeeek | {{{{{{{{{{{{{{{{{{Katty}}}}}}}}}}}}}}}} |
| 15:46.26 | Katty | ! |
| 15:46.28 | Katty | huggles Zeeek |
| 15:46.32 | Katty | must have missed it :< |
| 15:46.33 | Zeeek | Warning, unbalanced braces in above expression of admiration |
| 15:46.51 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 15:46.54 | fcois93 | [TK]D-Fender: I found how to do my loadbalancing :) |
| 15:47.19 | Zeeek | Katty: what, you don't have a little <BELL> every time Katty Katty Katty Katty is mentioned? |
| 15:47.28 | Katty | Zeeek: sadly, no |
| 15:47.33 | [TK]D-Fender | fcois93: Congratulations |
| 15:47.34 | Katty | Zeeek: i've heard putty can do that. |
| 15:47.35 | Katty | Zeeek: but.. |
| 15:47.42 | Katty | Zeeek: never got it to work right. |
| 15:47.56 | Katty | Zeeek: i screen irssi on a linux box in the server room and then attach to it from my windows box with putty |
| 15:47.56 | fcois93 | [TK]D-Fender: just with the dialplan |
| 15:47.56 | Zeeek | most IRC clients can on "modern" OS |
| 15:48.19 | Zeeek | shudders at the raw geekiness of that arrangement |
| 15:48.30 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 15:48.50 | Katty | Zeeek: it's very handy. |
| 15:48.56 | [netman] | Katty: screen has a visual bell :P |
| 15:49.02 | Katty | [netman]: oh? |
| 15:49.09 | Zeeek | oh oh oh OOOHHHHH |
| 15:49.14 | Zeeek | ahhhhhhhhhhhh |
| 15:49.15 | Katty | Zeeek: i like that i can detach from my session and go home, or on my blackberry, and POOF i'm back |
| 15:49.16 | [netman] | ctrl+a+g |
| 15:50.01 | tzanger | Katty: I do the same, you need to turn on audible bell, that's all |
| 15:50.14 | Katty | digs through putty cfg |
| 15:50.32 | jameswf | linux has a few easter egs like>> :(){ :|:& };: << totaly awesome |
| 15:51.03 | psykx-out | i don't kow if i'd call that an easter egg |
| 15:51.05 | jameswf | waits |
| 15:51.09 | Katty | tzanger: i turned pc speaker beep on |
| 15:51.12 | psykx-out | it's just abfuscated |
| 15:51.29 | [TK]D-Fender | MORE COWBELL! |
| 15:51.31 | Katty | tzanger: it was set to make system default sound alert |
| 15:51.31 | psykx-out | *obfuscated |
| 15:52.07 | jameswf | I have pcspkr blacklisted...it annoys the crap out of me and my double tab foo |
| 15:52.56 | [SySteM] | I buy a swissvoice IP10s to connect to my asterisk server.. but |
| 15:52.57 | [SySteM] | On the console : Comfort noise support incomplete in Asterisk.Please turn off on client if possible. Client IP: |
| 15:53.07 | Katty | okay, let's try it! |
| 15:53.10 | Katty | tzanger: katty me. |
| 15:53.15 | [SySteM] | I disabeld all vad.. echo .. on client. |
| 15:53.18 | tzanger | Katty: foo |
| 15:53.22 | Katty | :< |
| 15:53.32 | Katty | no love. |
| 15:54.01 | [TK]D-Fender | [SySteM]: if * still says that then you have not done the job. |
| 15:54.14 | Katty | weird. /beep just makes putty flash once |
| 15:54.38 | [TK]D-Fender | Katty: Terminal > Bell |
| 15:55.56 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:55.56 | Katty | resets hilights |
| 15:56.01 | Katty | tzanger: try again. |
| 15:56.19 | *** join/#asterisk aliver (n=aliver@c-71-196-147-164.hsd1.co.comcast.net) |
| 15:56.51 | *** join/#asterisk ManxPower (n=manxpowe@225.sub-75-250-174.myvzw.com) |
| 15:57.27 | *** join/#asterisk awk_r (n=rawk@nat/digium/x-c2796b03cf70c9f5) |
| 15:57.39 | Katty | or anyone else |
| 15:57.43 | tzanger | katty: foo |
| 15:57.47 | Katty | :< |
| 15:57.49 | Katty | cries. |
| 15:57.57 | tzanger | Katty: type control-a g |
| 15:58.04 | tzanger | make sure that screen is *not* set to visual bell |
| 15:58.07 | fudpucker | i am trying to get asterisk to use unixodbc, and the datasource is setup, and i can connect with isql, but when asterisk goes to write a cdr record, it errors out with SQLconnect -1. is there a way i can get the real reason? |
| 15:58.18 | tzanger | er |
| 15:58.20 | tzanger | control-a control-g |
| 15:58.25 | Katty | twitched to audio bell! |
| 15:58.28 | Katty | switched. |
| 15:58.31 | tzanger | katty: fo |
| 15:58.34 | Katty | and /beep works appropriatly now |
| 15:58.37 | Katty | but still no love. |
| 15:58.37 | tzanger | good. |
| 15:58.38 | awk_r | fudpucker, is your odbc conf setup? |
| 15:58.39 | tzanger | oh |
| 15:58.42 | Katty | quite :< |
| 15:58.43 | *** part/#asterisk psykx-out (n=max@uberpussy.net) |
| 15:58.48 | fudpucker | yes, and i can connect with isql |
| 15:58.59 | aliver | Katty are you using a Unix xterm terminal? |
| 15:58.59 | Katty | tzanger: there anything else required than /hilight mahnick |
| 15:59.06 | awk_r | fudpucker, odbc conf for asterisk |
| 15:59.10 | Katty | aliver: no. |
| 15:59.20 | aliver | Katty you can do "xset b on" if you are using X to make sure the bell will sound. |
| 15:59.25 | Zeeek | anyone have opinions on INUM? |
| 15:59.34 | fudpucker | i have cdr_odbc.conf, and cdr.conf |
| 15:59.40 | Katty | Zeeek: that's what happens after you play in the snow for an hour. |
| 15:59.52 | Zeeek | who made Katty cry :( |
| 15:59.55 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 16:00.00 | awk_r | fudpucker, if i remember correctly you set up three confs... /etc/odbc.ini /etc/asterisk/odbc.conf and /etc/asterisk/cdr_odbc.conf (or something around that) |
| 16:00.03 | *** join/#asterisk Ariel_Calzada (n=aricalso@200.71.48.212) |
| 16:00.09 | [SySteM] | <[TK]D-Fender> [SySteM]: if * still says that then you have not done the job. ? |
| 16:00.19 | [SySteM] | which job ? |
| 16:00.26 | [TK]D-Fender | [SySteM]: the job of DISABLING VAD |
| 16:00.34 | Zeeek | Digium is asking for suggestions fort their café name on Twitter. Any ideas I can steal and pretend they were mine? |
| 16:00.47 | Katty | how about digium |
| 16:00.53 | *** join/#asterisk synchris (n=synchris@athedsl-4380083.home.otenet.gr) |
| 16:01.02 | Katty | ;) |
| 16:01.05 | fudpucker | what do i need in odbc.conf? |
| 16:01.15 | [SySteM] | the G711A G711µ and G729 are "Suppression des silences" OFF |
| 16:01.21 | [SySteM] | and reboot 3 times. |
| 16:01.28 | Zeeek | I said when people ask for a menu they can sau "It's all there in the rEADME" |
| 16:01.46 | Zeeek | Cory Andrews said "Grub" |
| 16:01.49 | Zeeek | not bad |
| 16:01.50 | [TK]D-Fender | [SySteM]: However you're doing it is not working. So either you're doing it wrong, or your phone is not working properly |
| 16:01.52 | awk_r | fudpucker, h/o lemme check a running system with odbc in it...its been a while :-/ |
| 16:01.54 | Katty | Zeeek: what's this name for again? |
| 16:02.03 | Zeeek | Their Café |
| 16:02.07 | Zeeek | a place to eat |
| 16:02.21 | Katty | we have a cafe?! |
| 16:02.23 | [SySteM] | it cant be another think ? |
| 16:02.27 | Zeeek | I know one thing, it's a -café where "a register is not a register" |
| 16:02.28 | [SySteM] | without vad ? |
| 16:02.36 | Katty | Digi Yum? |
| 16:02.51 | Katty | Digi-Yum. |
| 16:02.52 | [TK]D-Fender | :(Digi) -- YUM!!!!! |
| 16:02.56 | Qwell | it's mostly a coffee shop |
| 16:03.03 | tzanger | Katty: working onw? |
| 16:03.04 | tzanger | er now |
| 16:03.06 | fudpucker | awk_r: thx |
| 16:03.09 | Katty | tzanger: no:< |
| 16:03.11 | tzanger | hmm |
| 16:03.17 | Katty | tzanger: do i need to do anything in irssi, specifically |
| 16:03.24 | Katty | tzanger: i have /hilight mahnicks |
| 16:03.27 | Katty | tzanger: but that's it. |
| 16:03.27 | tzanger | screen said "switched to audible bell" -- make sure that putty can make noises |
| 16:03.33 | tzanger | Katty: no idea, I never had to do anything to it |
| 16:03.34 | tzanger | oh |
| 16:03.35 | tzanger | one thing |
| 16:03.37 | Katty | tzanger: it does. /beep works |
| 16:03.46 | tzanger | Katty: flip to another irssi window |
| 16:03.47 | awk_r | fudpucker, k i was wrong...i'm not sure where i got odbc.conf anyway...so you have odbc configured properly. can you paste bin your cdr_odbc.conf? |
| 16:03.52 | Katty | okay. |
| 16:03.57 | tzanger | Katty: ooga. |
| 16:04.16 | Katty | hold on |
| 16:04.18 | Katty | tzanger: ooganow |
| 16:04.26 | tzanger | ooga now |
| 16:04.35 | Katty | sobs. |
| 16:04.39 | fudpucker | [global] |
| 16:04.46 | fudpucker | dsn=MySQL-asterisk |
| 16:04.50 | fudpucker | username=asterisk |
| 16:04.54 | fudpucker | password=xxxxxxxxx |
| 16:04.57 | Katty | you'd think there's like an enable beepy on irssi somewhere |
| 16:05.03 | fudpucker | loguniqueid=yes |
| 16:05.07 | fudpucker | table=cdr |
| 16:05.11 | fudpucker | usegmttime=no |
| 16:05.16 | [TK]D-Fender | fudpucker: PASTEBIN |
| 16:05.18 | [TK]D-Fender | ~pb |
| 16:05.19 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:05.30 | fudpucker | my bad...sorry |
| 16:08.19 | Katty | Zeeek: i twittered my suggestion. |
| 16:08.36 | Zeeek | oh cool! To digium or markster? |
| 16:09.13 | *** join/#asterisk bcrochet (n=bcrochet@cpe-069-132-204-022.carolina.res.rr.com) |
| 16:09.52 | fudpucker | is there any way i can get * to give me a better reason than 'can't connect to data source' |
| 16:10.13 | Katty | tzanger: oogame again |
| 16:10.16 | Katty | tzanger: in query |
| 16:10.25 | file | tickles Katty |
| 16:10.29 | [TK]D-Fender | fudpucker: pastebin your attempt at CLI to use the same credentials &table |
| 16:10.31 | Katty | oh |
| 16:10.35 | Katty | IT WORKS |
| 16:10.36 | Katty | HORAY! |
| 16:10.41 | Katty | hugs file |
| 16:10.46 | file | hugs Katty |
| 16:10.48 | Katty | YAY PUTTY BEEPS |
| 16:11.01 | *** join/#asterisk johnd23 (n=test@167.206.219.50) |
| 16:11.11 | Katty | tzanger: there were some /set beep stuff in irssi REF: http://jedi.org/weblog/archives/003190.html |
| 16:12.33 | fudpucker | ok, give me a few mins,..had to pick up a call |
| 16:12.59 | Zeeek | What is asterisk, anyway? |
| 16:13.25 | Zeeek | I'm just here to pick up women |
| 16:13.34 | tzanger | hahaha |
| 16:13.37 | jameswf | ashttp://bugs.kde.org/show_bug.cgi?id=1 |
| 16:13.52 | Zeeek | So far, not a lot of luck in that area |
| 16:14.08 | ManxPower | ~zeeek |
| 16:14.08 | jbot | zeeek is, like, someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 16:14.18 | jameswf | tryagain http://bugs.digium.com/view.php?id=1 |
| 16:14.20 | jameswf | yay |
| 16:14.32 | ManxPower | now we know why he's looking to pick up women. 8-) |
| 16:14.32 | Zeeek | So I guess I'll have to just stick to the VoIP Users Conference http://voipusersconference.org for my jollies |
| 16:14.50 | Zeeek | ManxPower: ah, that was the younger, more vigorous Zeeek |
| 16:14.50 | pif | I'd the GUI is like an inflatable doll, |
| 16:14.54 | jameswf | shameless promotion |
| 16:15.07 | Zeeek | pif I think you've got something there |
| 16:15.20 | Zeeek | Shame is for the meek |
| 16:15.29 | ManxPower | Zeeek: Your evil twin? |
| 16:15.30 | Zeeek | is a register that isn't a register |
| 16:15.34 | jameswf-home | I like fat chicks they are free and easy |
| 16:15.39 | Katty | Zeeek: yeah but you've gotten a few hugs! |
| 16:15.42 | jameswf | sorry |
| 16:15.53 | pif | and you can share with a friend |
| 16:16.06 | Zeeek | Katty: and that means more to me than all the **** in the world |
| 16:16.20 | Katty | hugs on Zeeek |
| 16:16.22 | jameswf | I think I had a kerry moment and gpllaw took over my system.... hackers |
| 16:16.23 | Zeeek | Friend. Good™ |
| 16:16.25 | Katty | where's jaytee? |
| 16:16.35 | Katty | i hope he didn't get eaten by the new bear cubs. |
| 16:16.40 | Katty | jbot: seen jaytee? |
| 16:16.42 | jbot | jaytee <n=jforde05@unaffiliated/jaytee> was last seen on IRC in channel #asterisk, 1d 11h 41m 52s ago, saying: 'thanks!'. |
| 16:16.50 | Katty | oh dear, gone over a day :< |
| 16:16.57 | jameswf | jaytee is in training |
| 16:16.58 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 16:17.04 | Katty | what sort of training? |
| 16:17.15 | jameswf | he is at jsmith's class |
| 16:17.16 | Zeeek | Please someone give me your opion of INUM? Do you think if it worked it would be good? Will it work? |
| 16:17.43 | Zeeek | s/opion/opinion/ |
| 16:17.56 | Zeeek | s/INUM/sex/ |
| 16:18.01 | jameswf | is reading inum press release |
| 16:18.35 | jameswf | sex is like pizza when its good its goog and when its bad it is still pretty good |
| 16:18.36 | Zeeek | s/opion of sex/opium of the people/ |
| 16:18.40 | *** part/#asterisk Ariel_Calzada (n=aricalso@200.71.48.212) |
| 16:18.50 | Zeeek | s/opium of the people/religion/ |
| 16:19.08 | Zeeek | s/religion/watch Bill Maher's Real Time" |
| 16:19.12 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 16:19.12 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:19.26 | Katty | oh boy! it's russell! |
| 16:19.29 | Katty | hugs russellb |
| 16:19.40 | russellb | <3 |
| 16:19.42 | russellb | hiiii |
| 16:19.45 | Zeeek | {{{{{ Russellb }}}}} |
| 16:20.27 | Zeeek | russellb: will answer the question "Is 1.6 ready for prime time?" that keeps coming back on the ML |
| 16:20.30 | jameswf | Zeeek: I dont see the all important "How much $$$$" |
| 16:20.34 | Zeeek | Film at 12 |
| 16:20.38 | russellb | i have no answers |
| 16:20.43 | russellb | (or rather no time to give them) |
| 16:20.50 | Zeeek | russellb: ah, a Zen master |
| 16:20.50 | jameswf | russellb: the answer is 42 |
| 16:21.09 | Katty | and mice rule the world. |
| 16:21.15 | Zeeek | they do |
| 16:21.18 | pif | but asterisk is gay software, no? |
| 16:21.19 | Katty | raises redbull to the mice |
| 16:21.31 | Zeeek | raises Jenlain (beer) to everyone |
| 16:21.36 | Katty | what's wrong with being gay? |
| 16:21.42 | [TK]D-Fender | Zeeek: No, that'd be me :) |
| 16:22.19 | [TK]D-Fender | ~[TK]D-Fender |
| 16:22.20 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 16:22.20 | pif | Katty: why would it be wrong? |
| 16:22.24 | Zeeek | [TK]D-Fender who rulez? |
| 16:22.29 | Zeeek | oh |
| 16:22.34 | [TK]D-Fender | :p |
| 16:22.48 | Zeeek | and we all know that master.... is the fiorst part of the word... |
| 16:23.03 | russellb | pif: do what? |
| 16:23.18 | Zeeek | what is up with these names in [? Because they make it so hard to use auto-complete |
| 16:23.44 | Zeeek | Almost time so now in the age old ritual... |
| 16:23.49 | Zeeek | We take you to... |
| 16:23.56 | [TK]D-Fender | Zeeek: 2 chars gets min 99% of the time. many people's nicks force me to do 5 or more to get them |
| 16:23.58 | Zeeek | http://voipusersconference.org |
| 16:24.17 | russellb | Zeeek: it's so people can get their nick at the top of the nick list :-p |
| 16:24.21 | Zeeek | IRC #voip-users-conference -- JOIN IT NOW |
| 16:24.32 | Zeeek | russellb: _____ AH _____ |
| 16:24.41 | Zeeek | since I don't look at the list |
| 16:24.56 | Zeeek | You can join the call very, very, very easily |
| 16:25.09 | pif | a conference call? |
| 16:25.54 | Zeeek | sip:7463#22622#1#@proxy.ideasip.com OR better yet: sip:talkshoe@vuc.onsip.com and enter 22622# and then your PIN# |
| 16:26.34 | Zeeek | Yes this is a free, and I say free as in free love, although that's debatable, FREE conference call via SIP with all the asterisk™ and VoIP users all over the planet |
| 16:26.56 | jameswf | gnome @ 66% |
| 16:27.07 | Katty | Zeeek: i twittered some more suggestions. |
| 16:27.14 | Zeeek | Plus it will allow you to test the SipAddHeader() function of asterisk (1.4?) |
| 16:27.14 | Katty | Zeeek: but i don't need a gift card. |
| 16:27.38 | jameswf | and me :) |
| 16:27.40 | Zeeek | "Keep them coming" |
| 16:28.06 | pif | Zeeek: how does one get a PIN ? |
| 16:28.07 | Zeeek | Katty the gift card include free trip to beautiful Huntsville, AL |
| 16:28.17 | Zeeek | Register at Talkshoe.com "JOIN" |
| 16:28.18 | Katty | i could be there in 4 hours. |
| 16:28.19 | disposable | I am struggling with a bug http://bugs.digium.com/view.php?id=11734 that's supposed to be resolved. I compiled * 1.6.0.1 with imap voicemail storage, but when i try to use it, i get http://pastebin.com/dc30155c I tried to apply the suggested patch but it got rejected and when i looked at app_voicemail.c i realised it wasn't needed as the necessary changes are already there. can anybody help? |
| 16:29.28 | Zeeek | Talkshoe signup: http://www.talkshoe.com/talkshoe/web/tscmd/signup/1 |
| 16:29.53 | Zeeek | Get a PIN, it makes it a lot easier for me and for you (PIN can be your callerid) |
| 16:30.32 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 16:30.34 | Zeeek | I have actually been ti Huntsville several times and never seen Digium |
| 16:30.40 | IsUp | hey ya |
| 16:30.47 | Zeeek | But I was at the SPace Museum |
| 16:31.00 | IsUp | how can i apply patches on Mantis? |
| 16:31.13 | IsUp | i need to patch some stuff, but never used 'diff' or 'patch'.. |
| 16:31.18 | seanbright | there is a link |
| 16:31.35 | seanbright | next to the patch that says "wget patch" |
| 16:31.41 | seanbright | click that, and it will show you the commands to run |
| 16:32.01 | IsUp | thank you. |
| 16:32.32 | tzanger | boggles |
| 16:32.44 | tzanger | I could have sworn "0b" was a valid C number format |
| 16:32.56 | tzanger | as in 0xaa == 0b10101010 |
| 16:34.32 | IsUp | seanbright, it worked =) thanks! |
| 16:37.47 | IsUp | i want to use chan_ss7 with DAHDI, any ideas? |
| 16:38.13 | [TK]D-Fender | IsUp: Don't :) |
| 16:38.32 | Zeeek | Wife left me. Promises to be back to announce dinner menu on http://bit.ly/vuc |
| 16:38.33 | Katty | Zeeek: i think i like Dahdi's Corner Bistro the best. |
| 16:38.43 | Zeeek | Until then, goodbye cruel asterisk world |
| 16:38.51 | IsUp | why not [TK]D-Fender? :> |
| 16:38.53 | Katty | buhbye |
| 16:39.04 | Zeeek | Katty: was that yours? |
| 16:39.06 | fudpucker | any place where i can get a cold guinness is fine with me |
| 16:39.11 | Katty | Zeeek: yes. |
| 16:39.19 | Katty | Zeeek: along with Kewlstart Cafe and DND Express |
| 16:39.24 | [TK]D-Fender | IsUp: Not doing it saves effort |
| 16:39.27 | Katty | Zeeek: and a whole slew of others |
| 16:39.37 | Zeeek | what's your id ? I'm not following you. I would be voipusers |
| 16:39.43 | Katty | Zeeek: izaah |
| 16:40.00 | Katty | Zeeek: i sent all the names as replies to digium |
| 16:40.03 | IsUp | but LibSS7 seems unstable. |
| 16:40.44 | Katty | Zeeek: shall i resend all the names to you? |
| 16:40.51 | Zeeek | Katty: now I'm onto you! |
| 16:41.08 | Zeeek | no I can see them. I'm FOLLOWING you! |
| 16:41.14 | Katty | k |
| 16:41.23 | Zeeek | I know where you live, where you pick up your dog after school. |
| 16:41.29 | Katty | nods |
| 16:41.33 | Katty | that's okay |
| 16:41.48 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:41.49 | seanbright | tzanger: it is in GCC |
| 16:41.59 | seanbright | tzanger: but it's an extension |
| 16:42.12 | Zeeek | I'll be over shortly. In the Concord |
| 16:42.26 | Zeeek | The Room is Open |
| 16:43.02 | seanbright | tzanger: http://gcc.gnu.org/onlinedocs/gcc-4.3.2/gcc/Binary-constants.html#Binary-constants |
| 16:43.41 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:43.43 | tzanger | seanbright: weird |
| 16:43.52 | tzanger | $ gcc --version |
| 16:43.52 | tzanger | gcc (GCC) 4.1.3 20070929 (prerelease) (Ubuntu 4.1.2-16ubuntu2) |
| 16:43.54 | tzanger | didn't have it |
| 16:44.14 | seanbright | are you compiling with -stdc |
| 16:44.15 | seanbright | ? |
| 16:44.21 | tzanger | no, just gcc -o foo foo.c |
| 16:44.24 | Zeeek | bye for now |
| 16:44.27 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 16:44.35 | seanbright | tzanger: yeah, looks to be a new thing in 4.2.x |
| 16:44.40 | tzanger | aha |
| 16:44.57 | tzanger | seanbright: thanks for digging around for that, I wasn't expecting someone to find an answer :-) |
| 16:45.07 | tzanger | I just used strtol to initialize it at runtime... :-) |
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| 16:45.23 | *** part/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net) |
| 16:46.12 | seanbright | tzanger: i had read it just the other day, so it was fresh in my mind |
| 16:46.35 | tzanger | ah |
| 16:46.37 | seanbright | tzanger: i also found a post somewhere about a guy that wrote a script to generate a header that defined all of hte binary constants to their hex equivs |
| 16:46.52 | tzanger | is writing a routine that extracts 8-bit data from a 10-bit stream |
| 16:46.53 | seanbright | #define 0b00000000 0x0000 |
| 16:46.54 | tzanger | it's not pretty |
| 16:46.56 | seanbright | etc etc |
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| 16:47.19 | *** join/#asterisk feeds (n=feeds@85-135-238-100.adsl.slovanet.sk) |
| 16:47.25 | tzanger | 10-bit TDM stream getting stuffed into 8-bit memory ... so the timeslot data I'm intersted in is constantly offset depending on which particular ts I'm looking at |
| 16:47.33 | tzanger | I've got an elegant solution, it's jsut.. yuck |
| 16:47.46 | seanbright | fun fun |
| 16:47.54 | johnd23 | dumb question... Is it possible with AsteriskNOW to have a server running the software, 2 sip phones, all 3 plugged into a switch and be able to place calls only between the 2 sip phones? Or do you need other hardware for this? such as a tdm card? |
| 16:48.11 | seanbright | yes it is possible |
| 16:48.24 | seanbright | without extra hardware |
| 16:48.49 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:48.49 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 16:49.06 | seanbright | but don't tell lmadsen |
| 16:49.18 | johnd23 | ok thanks, just wanted to make sure before i start playing around with it |
| 16:49.18 | [TK]D-Fender | johndYou only need extra hardware to interface with physical phone lines and phones |
| 16:49.19 | lmadsen | certainly not |
| 16:49.29 | [TK]D-Fender | (analog that is) |
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| 17:00.42 | Katty | file: Little Star Coffee Bar |
| 17:09.00 | fudpucker | here is my pastbin finally: http://pastebin.com/m2ca2b79b |
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| 17:14.03 | Katty | oh hey. it's lunchtime! |
| 17:14.06 | Katty | disappears |
| 17:15.15 | mark_csi | hello |
| 17:16.00 | mark_csi | anything wrong with this line in extensions.conf? exten => s,1,GotoIfTime(22:00-07:59,*,*,*?afterhours,s,1) |
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| 17:16.26 | fudpucker | does asterisk look at odbc.ini and odbcinst.ini |
| 17:16.34 | mark_csi | it just gives me an engaged tone when triggered |
| 17:17.32 | mark_csi | fudpucker - I thought it only looked at res_odbc.conf |
| 17:18.16 | mark_csi | fudpucker - my mistake I've just checked mine and I've details in /etc/odbc.ini |
| 17:18.29 | fudpucker | yeah i copied them over to that, and whamo...it works |
| 17:20.06 | mark_csi | I believe odbc.ini is system wide - whereas res_odbc.conf is only used by asterisk |
| 17:20.36 | fudpucker | no i am getting asterisk died with code 127 |
| 17:20.39 | fudpucker | ..now |
| 17:21.55 | seanbright | mark_csi: not sure, but maybe it doesn't like having the greater timer first |
| 17:22.02 | mark_csi | what's it say in /var/log/asterisk/messages |
| 17:23.33 | seanbright | exten => s,1,GotoIfTime(08:00-21:59,*,*,*?:afterhours,s,1) |
| 17:23.35 | seanbright | that might work. |
| 17:24.04 | mark_csi | thanks seanbright - it was working before but defined as - exten = 1,n,GotoIfTime(22:00-07:59|mon-sun|01-31|jan-dec?voicemenu-custom-3,s,1) |
| 17:24.32 | seanbright | oh i see |
| 17:24.44 | fudpucker | http://pastebin.com/m3b44948c |
| 17:24.44 | mark_csi | seanbright - I've just seen the problem, missing an ':' after the ? |
| 17:25.05 | seanbright | i think that's backwards, right? |
| 17:25.18 | seanbright | if it's passed 10 and before 8am, you want to go to afterhours |
| 17:25.23 | seanbright | so what you have already should work |
| 17:25.33 | mark_csi | fudpucker: got any info before that? |
| 17:25.38 | fudpucker | seems like cdr_odbc is crashing it |
| 17:25.41 | fudpucker | nope, that is it |
| 17:26.24 | mark_csi | seanbright: have to admit I'm not up on this function |
| 17:26.40 | mark_csi | fudpucker: what db are you using? |
| 17:26.42 | seanbright | GotoIfTime(timespec?if true:if false) |
| 17:27.22 | fudpucker | MySQL |
| 17:27.40 | mark_csi | seanbright: got it, thanks - unfortunately system is live until 22:00 so I'll have to go easy on the beer this evening :-) |
| 17:28.04 | seanbright | heh |
| 17:28.10 | mark_csi | fudpucker: use the MySQL addon and then configure res_mysql.conf, it works far easier. |
| 17:29.15 | fudpucker | if i use the cdr addon, can i also use that for realtime *? |
| 17:29.37 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:30.13 | mark_csi | fudpucker: I use the same database for cdr and realtime |
| 17:30.39 | fudpucker | but the cdr_mysql plugin will work for realtime as well..... |
| 17:30.43 | mark_csi | fudpucker: just different tables, cdr uses 'cdr' and realtime settings are defined in extconfig.conf |
| 17:31.10 | fudpucker | that's always confused me |
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| 17:39.00 | mark_csi | thanks seanbright and good luck fudpucker - leaving for bar........ |
| 17:39.06 | jameswf | rebooting to gnome.... wish me luck |
| 17:40.23 | fudpucker | thx |
| 17:40.50 | *** part/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
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| 18:00.01 | feeds | hi guys, If I want to make myself a custom asterisk sound (/var/lib/asterisk/sounds), how can I convert it to all those .ulaw, .sln16, etc. formats?? Is there a converter for it somewhere? |
| 18:00.12 | lmadsen | I don't think this is possible without creating multiple peers, but I'll ask anyways; if I have a SIP peer who I want to accept calls from, and it could come from one of 4 different Ip address (and I can't match on username because the username is the DID they are requesting), is there a way to have a single entry, and have it match on one of those 4 IP addresses? |
| 18:00.22 | lmadsen | feeds: sox |
| 18:00.24 | lmadsen | ~sox |
| 18:00.25 | jbot | [sox] Sound Processing Tool. URL: http://sox.sourceforge.net/ |
| 18:00.29 | feeds | thnx |
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| 18:14.51 | ManxPower | feeds: convert it to WAV. .wav is ulaw + microsoft wrapper |
| 18:15.00 | feeds | lmadsen, tried it out, works perfect, thanks |
| 18:15.07 | lmadsen | good deals |
| 18:15.24 | feeds | ManxPower, and the other formats?? .sln16, etc.?? |
| 18:15.33 | lmadsen | sln is signed linear |
| 18:15.36 | ManxPower | lmadsen: I seem to vaguely recall you can have multiple allow/deny statements. |
| 18:15.48 | ManxPower | feeds: Why do you want all these sound files in all these formats. |
| 18:15.49 | lmadsen | ManxPower: ya... but that is probably only useful if you can match on a username |
| 18:15.57 | lmadsen | ManxPower: save on transcoding |
| 18:16.05 | lmadsen | it's a perfectly valid approach |
| 18:16.13 | feeds | ManxPower, because each codec needs its own doesn't it? |
| 18:16.26 | ManxPower | Yes, but just how many times will you get a call in sln16 format. |
| 18:16.39 | ManxPower | feeds: No. Asterisk will automatically transcode. |
| 18:16.39 | lmadsen | ManxPower: you don't need those ones |
| 18:17.15 | feeds | ManxPower, cool. Spares time ;) |
| 18:17.21 | lmadsen | feeds: it will transcode if you have it in a single format, but having in multiple formats will make it so asterisk doesn't have to transcode. You don't need signed linear formats because asterisk just uses that as an internal conversion codec |
| 18:17.51 | ManxPower | avoiding transcoding is mostly to lower CPU usage. |
| 18:17.59 | feeds | so asterisk is faster than yes? |
| 18:18.03 | feeds | If I got it? |
| 18:18.18 | feeds | * then not than |
| 18:18.24 | lmadsen | feeds: not faster, lower CPU usage |
| 18:18.38 | ManxPower | feeds: You won't notice any extra speed unless your server is way under powered or you have many many calls |
| 18:19.00 | feeds | lmadsen, Thanks. ; ManxPower, Now I get it all. |
| 18:19.03 | PrimeHaxor | anyone have some problem with asterisk/iptables to drop established calls? |
| 18:20.02 | ManxPower | PrimeHaxor: nope, never. |
| 18:20.41 | ManxPower | I have heard of firewall settings causing call problems, but that is nothing specific to iptables |
| 18:21.57 | ManxPower | Those problems are almost always caused by people that "don't know much about networking or SIP but want to send voice over the most complex network on the planet" problems. Not much we can do for those poor sods. |
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| 18:24.04 | PrimeHaxor | i've tried make a QOS, to minimum delay for asterisk server |
| 18:24.24 | PrimeHaxor | but didn't worked |
| 18:24.42 | ManxPower | PrimeHaxor: that does very little good over the internet since the internet does not actually support QoS |
| 18:25.25 | PrimeHaxor | sure |
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| 18:39.54 | beek | Hello all. I'm having a helluva time getting CallerID on two POTS lines. Receiving the information is a hit-or-miss prop and I've tried the different combos that I have googled. Would someone look at my config and offer suggestions? http://www.pastebin.ca/1256326 |
| 18:40.53 | *** join/#asterisk docelmo (n=docelmo@206.248.239.194) |
| 18:41.00 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 18:41.17 | docelmo | MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! MEW! |
| 18:41.31 | Corydon76-dig | docelmo: Stop |
| 18:41.44 | docelmo | ohh geesh.. ok |
| 18:42.16 | [TK]D-Fender | beek: Channel bacnk on A104? |
| 18:42.26 | beek | [TK]D-Fender: Yes |
| 18:42.50 | [TK]D-Fender | beek: Could be an issue with the CB itself. |
| 18:42.58 | [TK]D-Fender | beek: what kind? |
| 18:43.04 | beek | Adit 600 |
| 18:43.47 | [TK]D-Fender | beek: signalling=fxs_gs <- very odd as well.. |
| 18:44.04 | [TK]D-Fender | beek: usually best on FXS_KS |
| 18:44.30 | beek | [TK]D-Fender: So this needs to be set both on the channel bank and in chan_dadhi? |
| 18:45.18 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 18:45.19 | [TK]D-Fender | beek: Definitely. |
| 18:45.28 | [TK]D-Fender | beek: can't have them disagreeing with each other... |
| 18:45.42 | PrimeHaxor | what means MEW? |
| 18:46.02 | beek | [TK]D-Fender: Okay -- I'm off to give that a try. I'd slaughter a chicken and drip its blood into the box if that's what it took... |
| 18:46.18 | [TK]D-Fender | beek: * requires GOATS. |
| 18:46.20 | docelmo | primehaxor: ask Katty |
| 18:46.24 | [TK]D-Fender | beek: You'll anger the Gods |
| 18:46.28 | beek | [TK]D-Fender: Thanks and I'll let you know. Ahhhh... so THAT'S where I'm going wrong! |
| 18:46.42 | [TK]D-Fender | PrimeHaxor: Its cat for "bark" |
| 18:55.01 | WHYS | Looking for pointers. I want to replace the image on my 7960 phone. I am using a TFTP server to configuring it. How do I point the phone to the image? (I do have the image Created/Packaged) |
| 18:55.34 | *** join/#asterisk icel (n=dan@75.150.16.102) |
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| 19:02.04 | Katty | docelmo: MEW! |
| 19:03.06 | icel | Having some weird Directory() issue. Anyone have any ideas? http://pastebin.ca/1256352 |
| 19:05.54 | *** join/#asterisk Strom_C (n=strom@netblock-208-127-61-171.dslextreme.com) |
| 19:13.23 | icel | I guess it's not a Directory() problem so much as how I am accessing it |
| 19:15.06 | icel | seems to be related to DNIS |
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| 19:19.23 | docelmo | Whats cooking chickie |
| 19:20.34 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 19:22.20 | *** join/#asterisk feeds (n=feeds@85-135-238-100.adsl.slovanet.sk) |
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| 19:29.40 | jsmith | icel: No, it seems that Asterisk isn't getting any kind of response from the other side, so it's hanging up the call |
| 19:30.04 | jsmith | icel: I'd check the network and firewall first... |
| 19:31.14 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 19:32.06 | *** join/#asterisk pepse (n=pepse@71-223-126-245.phnx.qwest.net) |
| 19:32.09 | pepse | hi guys. |
| 19:32.23 | pepse | i'm looking at some sip debug stuff going by.. |
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| 19:32.31 | pepse | just to one place right now |
| 19:32.39 | pepse | and i'm seeing a "SIP/2.0 400 Bad Request" |
| 19:32.52 | pepse | how can I tell why I'm getting that? |
| 19:33.13 | pepse | CSeq: 102 OPTIONS |
| 19:33.15 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 19:33.31 | pepse | i'm also not getting any incoming calls with this connection |
| 19:36.07 | hardwire | anybody here have an ISDN guard? |
| 19:36.12 | *** join/#asterisk korihor (n=korihor@201.210.239.172) |
| 19:36.22 | hardwire | are the failover ports addressable? |
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| 19:37.55 | *** join/#asterisk km2 (n=x@mobile-166-217-236-005.mycingular.net) |
| 19:38.49 | docelmo | pepse options could mean many things.. but mainly in asterisk its used to qualify a sip peer |
| 19:39.08 | docelmo | pepse look at a sip debug of an incoming call from that peer/IP |
| 19:42.12 | *** join/#asterisk Dr-Linux|home (n=Nothing@221.132.117.17) |
| 19:42.39 | Micc | server crashed again this morning. It looks like it was trying to do some mysql realtime stuff but I'm not using mysql. How can I disable the mysql res_config stuff? Theres no .conf file for it that I can find. |
| 19:43.12 | *** join/#asterisk devhen|Work (n=devhen@216.194.118.110) |
| 19:43.32 | *** part/#asterisk Strom_C (n=strom@netblock-208-127-61-171.dslextreme.com) |
| 19:44.16 | giovani | Micc: res_odbc.conf |
| 19:45.09 | giovani | and extconfig.conf |
| 19:45.09 | Micc | my res_odbc.conf only has my mssql entry. |
| 19:45.41 | giovani | extconfig.conf is where you tell asterisk to look to the db for config, rather than a config file, as far as what I'm reading here |
| 19:45.48 | [TK]D-Fender | pepse: the response is because your provider does not like you sending qualify packets at them. Doesn't mean it'll actually cause any other problems though. |
| 19:46.30 | Micc | giovani, those files all look fine. no mention of mysql, yet it still tries to connect to mysql. |
| 19:46.47 | giovani | they'd mention odbc |
| 19:47.01 | giovani | who set up asterisk for you? |
| 19:47.05 | Micc | I did. |
| 19:47.12 | *** join/#asterisk synchris (n=synchris@athedsl-4380083.home.otenet.gr) |
| 19:47.16 | giovani | well, this isn't default behavior afaik |
| 19:47.19 | Micc | I setup tds and unixodbc to our mssql server. |
| 19:47.29 | giovani | oh, ok |
| 19:47.43 | giovani | well what are the relevant log entries saying mysql? |
| 19:47.57 | Micc | I've recently turned on full logging and I see right before asterisk crashes it tries to load the queues from the db and connect to mysql. |
| 19:48.22 | giovani | well, I doubt it's doing so without being configured that way |
| 19:48.27 | giovani | grep your confs for mysql |
| 19:48.35 | giovani | maybe there's a line uncommented you missed |
| 19:48.37 | Micc | chan_agent.c: Queue memgbers successfully loaded from database. |
| 19:48.44 | Micc | I don't have any queue stuff in the db. |
| 19:49.13 | giovani | well, that config should be in extconfig.conf |
| 19:49.16 | Micc | I did that already. maybe I should check my unixodbc configs. |
| 19:49.18 | *** join/#asterisk synchris (n=synchris@athedsl-4380083.home.otenet.gr) |
| 19:49.18 | giovani | but once again -- grep all of your confs |
| 19:49.43 | Micc | there are commented out lines in extconfig. I'll remove all the comments just in case. |
| 19:50.02 | giovani | I just start with clean confs |
| 19:50.06 | giovani | to make sure |
| 19:50.18 | giovani | so I know everything that's in there |
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| 19:51.22 | *** join/#asterisk valbud (n=valbud@89.35.223.85) |
| 19:51.30 | valbud | hello people |
| 19:51.34 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 19:51.41 | valbud | i have an ackward problem with and asterisk box |
| 19:51.57 | valbud | does anyone have some time to guide me in the right direction |
| 19:51.58 | valbud | ? |
| 19:52.21 | lmadsen | ~ask |
| 19:52.22 | jbot | ask is probably Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:52.40 | valbud | ok thank you |
| 19:52.43 | valbud | and sorry |
| 19:52.48 | icel | jsmith: thx |
| 19:52.58 | jsmith | icel: No problem |
| 19:53.00 | lmadsen | hrmmm... anyone run into an issue with sipregistrations (regcontext= in sip.conf) not populating? |
| 19:53.25 | valbud | so: the network diagram is as follows: an opensuse 10.3 connected to a NET GEAR switch |
| 19:53.43 | valbud | in the same switch is connected a Linksys SPA 400 gateway |
| 19:53.58 | valbud | and 7 phones: 1 SPA 921 and 6 SPA 901 |
| 19:54.31 | valbud | now the problem: whenever i call an outside destination that pass through the SPA 400 |
| 19:54.52 | valbud | from the 901, the first attempt is always unsuccefull |
| 19:55.01 | valbud | *successfull |
| 19:55.22 | valbud | i can provide sip.conf, extensions.conf and sip debug from a call |
| 19:55.32 | valbud | should i paste them here or .... ? |
| 19:56.18 | valbud | the inside calls work like a charm |
| 19:57.04 | valbud | and the spa 921 can call outside destinations just fine |
| 19:57.35 | [TK]D-Fender | ~pb |
| 19:57.35 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 19:57.38 | [TK]D-Fender | valbud: ^^^^^^ |
| 19:58.46 | valbud | thank you |
| 20:00.15 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 20:00.53 | valbud | extensions.conf - http://pastebin.com/m3965a0cc |
| 20:01.03 | *** part/#asterisk nikko (n=nikko@69.57.49.100) |
| 20:01.58 | valbud | sip.conf - http://pastebin.com/d126c6055 |
| 20:02.12 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 20:02.20 | gaetronik | hi there |
| 20:02.52 | gaetronik | is this possible to register sip account login time? |
| 20:03.12 | valbud | the sip debug trace: http://pastebin.com/d41562c84 |
| 20:05.44 | *** join/#asterisk klictel (n=klictel@nat/digium/x-88f842c1d12127a9) |
| 20:06.33 | *** join/#asterisk boolean12 (n=boolean1@tandem.uplinktel.com) |
| 20:07.14 | Katty | weeeeeee |
| 20:07.41 | [TK]D-Fender | ~whee |
| 20:07.41 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 20:11.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:11.48 | gaetronik | is there any way to see this kind of info? |
| 20:12.26 | jblack | [TK]D-Fender should be banned for that. |
| 20:12.31 | [TK]D-Fender | gaetronik: Go watch AMI events |
| 20:13.10 | gaetronik | ok through ami, i will take a look |
| 20:13.52 | *** join/#asterisk eit (n=eit@64.122.178.15) |
| 20:15.12 | *** join/#asterisk littlepinkdot (n=thedot@69.7.43.20) |
| 20:15.25 | Micc | I've found the problem I think. |
| 20:15.33 | Micc | in asterisk-addons readme. |
| 20:15.34 | littlepinkdot | With music on hold, anyone know why the extension is stripped/not read? [Nov 14 12:14:40] WARNING[17885] res_musiconhold.c: Unable to open file '/var/lib/asterisk/mohmp3//orig_Al Stewart - 11 - Year Of The Cat': No such file or directory |
| 20:15.47 | Micc | 1) Using res_config_mysql at the same time as res_config_odbc can create |
| 20:15.47 | Micc | system instability on some systems. Please load only one or the other. |
| 20:16.03 | littlepinkdot | The file exists, both .mp3 and .wav, but asterisk doesn't read past the filename. |
| 20:16.04 | Micc | So now I just need to find out how to only use res_config_odbc. |
| 20:16.43 | [TK]D-Fender | littlepinkdot: // <-- double slahs in the path |
| 20:16.46 | [TK]D-Fender | slash* |
| 20:17.30 | littlepinkdot | Is that configured in /etc/asterisk/musiconhold_additional.conf |
| 20:17.39 | jblack | littlepinkdot: That's by design. If you have that file in multiple formats, asterisk picks the one it "likes best" |
| 20:18.01 | littlepinkdot | When I upload it (via Freepbx), it creates the .wav from the .mp3 |
| 20:18.12 | jblack | Oh, I don't know anything about freepbx. |
| 20:18.22 | littlepinkdot | Issue is with Asterisk |
| 20:18.55 | jblack | I don't know how they've configured things. You'll have to ask them there, in #freepbx. Good luck, though! |
| 20:19.14 | littlepinkdot | I have two boxes, one with just Asterisk and one with Freepbx, both have the same issue. |
| 20:19.41 | jblack | I've already given the asterisk answer. That's all I can do for you. |
| 20:20.00 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 20:22.00 | [TK]D-Fender | littlepinkdot: maybe your WAV's are bad. |
| 20:22.48 | gaetronik | [TK]D-Fender, thanks PeerStatus |
| 20:22.58 | valbud | did anyone had the chance to look over the configurations / debug |
| 20:23.15 | valbud | i just need a hint in the right direction |
| 20:27.10 | *** join/#asterisk SuPrSluG (n=SuPrSluG@72.237.213.162) |
| 20:29.06 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 20:29.47 | SkramX | hi all |
| 20:30.27 | SkramX | I have an AGI run after a phone dials a number but then when it hangs up, asterisk executes the AGI again and I get the deadAGI error/suggestion.. conf @ http://pastie.org/315169 |
| 20:31.19 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 20:31.25 | *** part/#asterisk eit (n=eit@64.122.178.15) |
| 20:31.34 | [TK]D-Fender | SkramX: "exten => _.,1,Set(TIMEOUT(digit)=2.5)" <- this pattern matched "h" and is a "shoot on sight" offense |
| 20:31.40 | [TK]D-Fender | gathers the firing squad |
| 20:33.27 | klictel | joins the squad |
| 20:34.15 | Dr-Linux|home | guys, i want to provide dial tone to my caller in AGI script .. how can i do that? |
| 20:34.35 | Dr-Linux|home | i can do that in dialplan with DISA .. but not sure how can i do that in AGI |
| 20:34.50 | Dr-Linux|home | any suggestion about Playtone(dial) ? |
| 20:35.05 | [TK]D-Fender | klictel: Salut Claude |
| 20:35.19 | klictel | hey salut |
| 20:35.27 | [TK]D-Fender | Dr-Linux|home: You can call DISA from AGI. |
| 20:35.44 | [TK]D-Fender | Dr-Linux|home: just like just about every other dialplan app |
| 20:35.54 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 20:36.17 | valbud | klictel: ro ? |
| 20:36.43 | SkramX | [TK]D-Fender: do you see what i'm trying to have it do? what should i be doing? |
| 20:37.03 | [TK]D-Fender | SkramX: You should not be using an uber-wlidcard like that. Thats what.. |
| 20:37.06 | *** join/#asterisk CrazyTux (n=brandon@user-vcaumpc.dsl.mindspring.com) |
| 20:37.13 | klictel | ro? |
| 20:37.20 | Dr-Linux|home | [TK]D-Fender: but DISA needs context etc as well, where i'm fail .. |
| 20:37.27 | SkramX | what should I use then? I want everything to be handled by my AGI |
| 20:37.30 | [TK]D-Fender | skdr-Go give it one. |
| 20:37.38 | [TK]D-Fender | Dr-Linux|home: Go give it one |
| 20:37.42 | Dr-Linux|home | [TK]D-Fender: can i use "playtones(dial)" for that purpose? |
| 20:37.44 | SkramX | but I need it to wait to see how many digits the dialer dials |
| 20:37.55 | valbud | klictel: from romania ... noticed you said "salut" |
| 20:37.59 | [TK]D-Fender | SkramX: I think you should not bu in a context like that with a horrendous pattern like that |
| 20:38.07 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 20:38.10 | SkramX | bu? |
| 20:38.14 | klictel | salut=hello in french |
| 20:38.21 | valbud | got it |
| 20:38.21 | [TK]D-Fender | be* |
| 20:38.27 | valbud | didn't know that :P |
| 20:38.30 | SkramX | sigh |
| 20:38.30 | *** join/#asterisk becks` (n=sdfgsfdg@169-244.104-92.cust.bluewin.ch) |
| 20:38.32 | klictel | i better start packing and heading to the airport |
| 20:38.35 | klictel | see you all |
| 20:38.48 | SkramX | [TK]D-Fender: What do I do to do a digit timeout, then transfer to an AGI |
| 20:38.57 | SkramX | it's pretty simple I dont see what's horrendous about that context |
| 20:39.04 | giovani | valbud: French is where the Romanians got it :) |
| 20:39.08 | SkramX | what's possibly horrendous is the AGI |
| 20:39.09 | SkramX | ;) |
| 20:39.15 | becks` | hi, how can i measure the MOS? somebody knows a software? |
| 20:39.18 | [TK]D-Fender | SkramX: you have "h" overlap because you made a relly not smart pattern in that context |
| 20:39.22 | valbud | giovani: got what? |
| 20:39.32 | giovani | valbud: "salut" |
| 20:39.34 | SkramX | should it be _X? |
| 20:39.39 | valbud | aha |
| 20:39.44 | valbud | may be |
| 20:39.48 | [TK]D-Fender | SkramX: a context witha match-all like that should have 1 purpose.. jumping OUT of that context to a target exten. |
| 20:40.05 | [TK]D-Fender | (a GOOD target that won't overlap) |
| 20:40.12 | SkramX | so _X? |
| 20:40.13 | SkramX | :) |
| 20:40.28 | [TK]D-Fender | valbud: actually means both "hello" and "goodby" depending on context |
| 20:40.46 | giovani | valbud: it is -- I looked up the etymology |
| 20:40.50 | [TK]D-Fender | valbud: the french's revenge for "aloha" |
| 20:40.54 | SkramX | i dotn use _X because I want it to wait for two or three or four digits |
| 20:42.41 | SkramX | [TK]D-Fender: I understand this is sort of un-standard.. how do I do what I need in a best-practices way? |
| 20:42.52 | [TK]D-Fender | SkramX: Well i think you'd better rethink your context's contents and pattern matches... |
| 20:43.16 | SkramX | nothing comes to mind :\ |
| 20:43.33 | valbud | thanks [TK]D-Fen for the explanations |
| 20:43.52 | [TK]D-Fender | SkramX: That pattern had better not exist in the same place as ANYTHING else. |
| 20:43.55 | valbud | i thought that salut is a romanian whing |
| 20:44.13 | SkramX | a SCCP phone enters internal right away; and then dials a number |
| 20:44.18 | SkramX | not sure what you're saying |
| 20:44.31 | valbud | *thing |
| 20:44.32 | SkramX | and i can't just leave it, because agi keeps getting called after hangup |
| 20:45.34 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:45.38 | [TK]D-Fender | SkramX: Your fist obligation on using a match like that is GET THE HELL OUT of that context <- |
| 20:45.58 | [TK]D-Fender | SkramX: and the ONLY thing in this one should be the jump. Absolutely nothing else. |
| 20:46.04 | SkramX | okay |
| 20:46.06 | SkramX | gotcha |
| 20:46.29 | WHYS | anyone have a URL for a 133x64 bmp file I can test on my 7960? |
| 20:46.42 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:47.08 | rwaite | yo wassup #* |
| 20:47.39 | [TK]D-Fender | WHYS: Go take a bigger file and trim it |
| 20:47.59 | file | ... |
| 20:48.06 | [TK]D-Fender | rwaite: #* ... that what happens when MS buys out Digium? |
| 20:48.21 | becks` | nobody knows how to measure MOS? :( |
| 20:48.24 | rwaite | naw, that would be *# |
| 20:48.35 | WHYS | Yeah, just looking for something quick. I need to stick it on a web server, etc. Thought one might be handy |
| 20:48.37 | [TK]D-Fender | darn... ruineda good joke. |
| 20:48.40 | [TK]D-Fender | oh well |
| 20:49.13 | rwaite | and then novell will start a linux compatible *# runtime called Gonorrhea |
| 20:49.55 | Katty | file. |
| 20:50.00 | [TK]D-Fender | rwaite: Don't f'n mention Novell... I'm trying to ditch it here... |
| 20:50.21 | rwaite | Heh. But Netware... it's the future! |
| 20:50.30 | [TK]D-Fender | rwaite: Windows Server is put off as a plan for economic reasons and I'm trying to "sell" Samba |
| 20:50.58 | [TK]D-Fender | rwaite: Only if you've travelled to the Jurassic period. |
| 20:51.10 | rwaite | debian+samba > win |
| 20:51.33 | Katty | icanhazhugnowpls? |
| 20:52.01 | [TK]D-Fender | rwaite: I just printed up the Samba By Example book and want to take this on. I've had our Mac guys on it for 3 years now |
| 20:52.15 | [TK]D-Fender | hugz teh Katty |
| 20:52.25 | Katty | hugs [TK]D-Fender |
| 20:52.27 | Katty | thanks. |
| 20:52.29 | Katty | i needed that :/ |
| 20:52.34 | rwaite | samba is okay, just dont go deluding yourself that you can run a domain with it .. i made that mistake and paid dearly for it |
| 20:52.57 | rwaite | Katty a as in apple or cake |
| 20:53.20 | Katty | rwaite: that did not parse, please try again. |
| 20:53.29 | [TK]D-Fender | rwaite: In what sense? |
| 20:53.39 | rwaite | k[apple]tty, k[cake]tty |
| 20:54.01 | Katty | oh. |
| 20:54.05 | Katty | Katty, as in Cat-ty |
| 20:54.14 | Katty | Meow. etc. |
| 20:54.24 | rwaite | meow. thx i have a loud voice in my head with i read so i have to know ;) |
| 20:54.51 | rwaite | oh boy quitting time |
| 20:54.55 | Katty | oh boy! |
| 20:54.55 | *** join/#asterisk AlexTO (n=alex@173.9.143.137) |
| 20:55.05 | rwaite | ciao |
| 20:55.28 | Katty | what an odd sorta fellow. |
| 20:56.59 | AlexTO | hi everyone.. |
| 20:57.06 | boolean12 | Fender: Why are you using novell? |
| 20:57.27 | etfonhomey | [TK]D-Fender, Where you have your Asterisk installation, do you have an analog phone to use in case your Asterisk system crashes or uses power? |
| 20:57.56 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:58.04 | etfonhomey | loses* |
| 20:58.06 | etfonhomey | doh! |
| 20:58.08 | [TK]D-Fender | etfonhomey: Yes, though its not really "plugged". |
| 20:58.27 | etfonhomey | [TK]D-Fender, you mean it's not plugged in all the time? |
| 20:58.59 | [TK]D-Fender | etfonhomey: Actually I have my PRI redirected to an IPKALL # that lands on my home system, which then takes a VM and e-mails it to our admin assistant so she can catch up them when things come up :) |
| 20:59.14 | [TK]D-Fender | etfonhomey: yes, that means my seup isn't perfect yet :) |
| 20:59.25 | [TK]D-Fender | tefI haven't actually cared to correct a bunch of stuff here yet |
| 20:59.37 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 21:00.35 | etfonhomey | [TK]D-Fender, The Cisco 500 series small business thingy has 4 FX0 and 4 FXS ports. And in a power off situation it will "bridge" a pair of those together so that you would have a hot analog line as backup regardless. Do you know if something is possible like this with a Sangoma card? |
| 21:01.37 | [TK]D-Fender | etfonhomey: Not on any current model. There is an Openvox "black" bridging module which may only work with their cards or Digiums perhaps. Audiocodes & Mediatrix gateways also have a failover |
| 21:04.45 | *** join/#asterisk johann8384 (n=jonathan@75-120-52-110.dyn.centurytel.net) |
| 21:07.05 | *** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com) |
| 21:07.41 | *** join/#asterisk Segnale007 (n=Pietro@host75-250-dynamic.18-79-r.retail.telecomitalia.it) |
| 21:09.10 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 21:09.33 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 21:09.40 | beek | [TK]D-Fender: Oh well -- the Adit 600 doesn't do kewl start... only GS & LS |
| 21:09.52 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:10.13 | jameswf | what crap |
| 21:10.36 | itiliti | I am trying to figure out the best way to write the Called DID when a call comes in or gets hung up to write it to the CDRDB. what is the best way to do that? |
| 21:12.11 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:12.42 | SkramX | what's the difference between ! and . wildcards? |
| 21:14.27 | ricko73 | the little thing above the . |
| 21:14.29 | ricko73 | ;) |
| 21:14.35 | SkramX | :D |
| 21:14.36 | Assimilate | heh |
| 21:14.41 | ricko73 | Friday humor |
| 21:16.11 | ricko73 | where would be the appropriate place to request a feature? |
| 21:16.34 | ricko73 | I'd like to have two tones for automon, one when the recording starts and a different one when the recording stops |
| 21:17.02 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 21:17.02 | ricko73 | right now, that uses the single courtesytone variable |
| 21:19.29 | [TK]D-Fender | ricko73: the users mailing list or as a bounty on the WIKI |
| 21:20.09 | *** join/#asterisk Segnale007 (n=Pietro@host75-250-dynamic.18-79-r.retail.telecomitalia.it) |
| 21:21.55 | *** join/#asterisk Segnale007 (n=Pietro@host75-250-dynamic.18-79-r.retail.telecomitalia.it) |
| 21:26.51 | SkramX | is there a default digit timeout? after i dial a number on my cisco phone.. asterisk waits like 4 seconds to do something |
| 21:28.04 | [TK]D-Fender | SkramX: SIP phones have their own dialplan. |
| 21:28.10 | *** join/#asterisk pids (n=pids@221.sub-70-210-233.myvzw.com) |
| 21:28.17 | [TK]D-Fender | SkramX: And thus their own timeout before sending their dial to * |
| 21:28.27 | AlexTO | Can someone give me a hand setting up dundi betwwen 2 * boxes |
| 21:29.01 | *** join/#asterisk kerx (n=prepro@adsl-68-125-33-231.dsl.irvnca.pacbell.net) |
| 21:29.42 | [TK]D-Fender | AlexTO: Show what you've tried and maybe someone can help with what's left |
| 21:30.11 | AlexTO | oki... let put the paste bin |
| 21:41.58 | *** join/#asterisk duztbunny (n=duztbunn@dsl093-216-054.aus1.dsl.speakeasy.net) |
| 21:42.30 | *** join/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
| 21:45.17 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 21:46.15 | bijit | hi everyone can someone help me debug why this is happening [Nov 14 12:04:37] VERBOSE[24474] logger.c: NEW_HANGUP DEBUG: Calling |
| 21:46.19 | bijit | q931_hangup, ourstate Active, peerstate Active |
| 21:47.20 | bijit | <PROTECTED> |
| 21:51.21 | *** join/#asterisk root52 (n=F745082a@ip70-191-120-39.cl.ri.cox.net) |
| 21:53.26 | *** join/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
| 21:54.57 | etfonhomey | [TK]D-Fender, Have you used any of the Audiocodes 118 series gateways? |
| 21:55.19 | Katty | whennnnnnnnn theeeeeeee moon hits your eyes like a big pizza pie, that's amooorreeeee!!! |
| 21:55.37 | [TK]D-Fender | etfonhomey: No, I only tested on an MP-124 FXS once and a Mediant 2000 |
| 21:55.53 | Katty | i'm dangeriously hyper >:) |
| 21:56.02 | etfonhomey | OK. Thanks. Have a good weekend! |
| 22:01.46 | *** join/#asterisk quentusrex (n=quentusr@c-71-197-244-228.hsd1.or.comcast.net) |
| 22:02.01 | quentusrex | Can I have some help debugging a remote extension? The remote extension can make calls out, and can recieve calls. But there is no sound either direction. |
| 22:02.15 | giovani | quentusrex: is it behind NAT? |
| 22:02.42 | quentusrex | yes |
| 22:02.44 | *** join/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
| 22:02.49 | giovani | no debugging required |
| 22:02.51 | giovani | NAT is always at fault |
| 22:02.52 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 22:03.00 | quentusrex | but I'm nearly positive that the ports are forwarded properly. |
| 22:03.07 | giovani | well, they clearly aren't |
| 22:03.12 | AlexTO | Hi Fender: http://pastebin.com/m5d5b10df it is about DUNDI setup for 2 * boxes |
| 22:03.22 | giovani | put a hub/networktap/span port outside the firewall |
| 22:03.28 | giovani | and see for yourself with a packet capture |
| 22:03.33 | giovani | which ports the packets are coming in on |
| 22:03.35 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 22:03.38 | giovani | or do the same on the asterisk box |
| 22:03.42 | quentusrex | so which ones am I missing? I've got 5000-5080 forwarded. and also 10000-20000 |
| 22:04.00 | giovani | the ports are not constant -- it depends on asterisk and phone configuration |
| 22:04.06 | beek | quentusrex: Do you have "nat=yes" in sip.conf for each of the phones? |
| 22:04.20 | quentusrex | yes |
| 22:05.03 | giovani | a packet capture will give you all the answers you need |
| 22:05.15 | *** join/#asterisk root52 (n=F745082a@ip70-191-120-39.cl.ri.cox.net) |
| 22:06.18 | SkramX | can i set a phone DND from the command line |
| 22:06.19 | root52 | giovani: thanks I was haveing the same problem. you need to make sure the whole SIP media port range is comeing through right? |
| 22:06.19 | SkramX | ? |
| 22:07.04 | giovani | root52: actually the SIP port often doesn't need to be forwarded (as long as your phone is initiating a register) -- because the firewall will keep that open -- it's the RTP ports that are the problem usually |
| 22:07.13 | giovani | the "no audio" thing, is a RTP port problem |
| 22:07.23 | giovani | if the call "connects" then SIP is getting through fine |
| 22:07.26 | quentusrex | which ports are used for rtp? |
| 22:07.44 | giovani | quentusrex: as I told you already, that's asterisk and phone specific configuration -- refer to your configs |
| 22:07.45 | root52 | ok you are right i was call RTP the SIP media |
| 22:09.55 | pids | quentusrex, usually its 10000 - 20000 |
| 22:10.37 | pids | each call generaly increments one port number. |
| 22:11.11 | pids | but it gets weird about that sometimes so dont count on it. |
| 22:11.31 | AlexTO | http://pastebin.com/m5d5b10df |
| 22:11.40 | giovani | pids: hence why a packet capture removes all doubt |
| 22:11.51 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 22:12.10 | *** join/#asterisk seanmh (n=seanmh@216.31.101.11) |
| 22:12.20 | pids | giovani, I guess if you ran a packet capture all the time. |
| 22:12.29 | giovani | you don't need to |
| 22:12.31 | giovani | just once |
| 22:12.37 | giovani | it'll let you know where the problem is immediately |
| 22:13.31 | quentusrex | hmm |
| 22:15.02 | pids | giovani, sure but unless he has changed the config with asterisk it will start at 10000 |
| 22:15.17 | quentusrex | So, after the packet capture. It looks like there is traffic going out |
| 22:15.28 | pids | by rfc it is ports 16384-32767 |
| 22:15.32 | giovani | quentusrex: going "out" from where? |
| 22:15.38 | quentusrex | using PCMU(ulaw) from the remote extension |
| 22:15.47 | quentusrex | but there is nothing coming in to the extension |
| 22:15.58 | giovani | quentusrex: I told you to do the packet capture OUTSIDE of the firewall |
| 22:16.04 | giovani | the entire point is to see what the firewall is dropping |
| 22:16.17 | quentusrex | I'm not able to do that. |
| 22:16.19 | [TK]D-Fender | *SIGH* |
| 22:16.22 | giovani | if you do it near the extension, inside the firewall, you just confirm what we already know |
| 22:16.24 | quentusrex | I did it at the firewall. |
| 22:16.27 | giovani | which is that the RTP isn't gettingto the phone |
| 22:16.35 | pids | quentusrex, are both the asterisk server and the extension behind diffrent NAT firewalls ? |
| 22:16.41 | quentusrex | yes |
| 22:16.45 | giovani | ohh ... |
| 22:16.47 | giovani | double nat! |
| 22:16.48 | giovani | wow |
| 22:16.50 | pids | wont work without a proxy server |
| 22:16.53 | giovani | welcome to hell :) |
| 22:16.58 | [TK]D-Fender | pids: BS |
| 22:17.10 | pids | How then? |
| 22:17.26 | [TK]D-Fender | * side needs forwarding, remote phone doesn't. |
| 22:17.48 | [TK]D-Fender | And I have NEVER needed an outside packet trace for any of this |
| 22:17.55 | pids | neither have I |
| 22:18.21 | [TK]D-Fender | first, where are the configs? A lot of talk going on, and no SHOW. |
| 22:18.38 | [TK]D-Fender | PASTEBIN is your friend |
| 22:18.41 | [TK]D-Fender | ~pb |
| 22:18.42 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 22:18.43 | pids | hehe |
| 22:18.44 | [TK]D-Fender | ^^^^^^^^^ |
| 22:18.56 | [TK]D-Fender | think I'd trust configs blindly? No dice. |
| 22:19.08 | quentusrex | the configs of the server? |
| 22:19.13 | [TK]D-Fender | quentusrex: Clearly |
| 22:19.15 | *** join/#asterisk synchris (n=synchris@athedsl-156469.home.otenet.gr) |
| 22:19.23 | [TK]D-Fender | quentusrex: PB sip.conf masking only passwords |
| 22:19.52 | AlexTO | Hi everyone.. there is someone how knows dundi can give me a hand to set up 2 * boxes? http://pastebin.com/m5d5b10df |
| 22:22.34 | quentusrex | hmm... |
| 22:24.51 | *** part/#asterisk root52 (n=F745082a@ip70-191-120-39.cl.ri.cox.net) |
| 22:25.02 | quentusrex | sec, I'm trying to find freepbx's sip.conf |
| 22:25.11 | [TK]D-Fender | LOL |
| 22:25.17 | [TK]D-Fender | Now with GUI configs! |
| 22:25.30 | [TK]D-Fender | quentusrex: you'll want sip_nat(blah) as well |
| 22:25.40 | [TK]D-Fender | quentusrex: All that crap |
| 22:25.53 | quentusrex | alright, sip_nat.conf is blank |
| 22:26.04 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 22:26.22 | pids | uggh, following the configs in freepbx gives me a headache. |
| 22:26.41 | [TK]D-Fender | quentusrex: Right about now I'm thinking you haven't done the first thing about preparing your server to work behind NAT |
| 22:26.57 | quentusrex | it works just fine for all of the local extensions |
| 22:27.16 | quentusrex | all the extensions on the same lan as the server can handle dozens of calls all at the same time. |
| 22:27.30 | quentusrex | but I just can't figure out why I can't get sound for the remote extension. |
| 22:27.47 | pids | quentusrex, http://www.voip-info.org/wiki/view/NAT+and+VOIP |
| 22:27.59 | quentusrex | http://pastebin.com/d334809d1 |
| 22:28.03 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 22:28.12 | RypPn | quentusrex go into #freepbx and type ?? nat |
| 22:28.50 | [TK]D-Fender | quentusrex>it works just fine for all of the local extensions <- meaningless. Yuo have about 1/2 dozen settings to do for * to work behind NAT and its all very googleable |
| 22:29.16 | [TK]D-Fender | quentusrex: Of course * has no problems with local devices... they're LOCAL. |
| 22:30.00 | [TK]D-Fender | quentusrex: There's an article I'd refer you to if the server wasn't down... |
| 22:30.40 | pids | quentusrex, http://www.freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-extension |
| 22:31.49 | [TK]D-Fender | pids: Decent guide, missing only 1 suggestable thing, which might not make a difference |
| 22:32.30 | quentusrex | Is there a way to read up on how to manage the * settings without freepbx? |
| 22:32.48 | *** join/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 22:32.51 | pids | quentusrex, www.google.com ? |
| 22:33.06 | [TK]D-Fender | quentusrex: the sip.conf sample and : |
| 22:33.08 | [TK]D-Fender | ~sipnat |
| 22:33.08 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:33.18 | [TK]D-Fender | quentusrex: However the firs link's server is currently down. |
| 22:33.45 | *** join/#asterisk DarkRift (n=dark@65.92.167.103) |
| 22:33.51 | bijit | ~q931 |
| 22:35.09 | pids | quentusrex, I would not recommend trying to manual modify the configs on FreePBX for your first installation of asterisk. |
| 22:35.09 | pids | You'll wanna commit suicide. Do a vanilla install of asterisk if you want to manually edit the configs |
| 22:35.26 | pids | [TK]D-Fender, what thing is missing? |
| 22:35.44 | [TK]D-Fender | (except sip_nat.conf) |
| 22:36.07 | [TK]D-Fender | pids: "canreinvite=no" should be global as well for those calls that fall under [general] |
| 22:36.17 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 22:36.35 | Spirits-Sight | [TK]D-Fender: I am trying to find out whats one of the best How to setup for Asterisk on CentOS, this is so I can give it to a person that does the maintain for server, see I have been playing with it on Ubuntu using my desktop but the server is CentOS 5.x |
| 22:36.35 | quentusrex | I fixed the problem. |
| 22:36.45 | quentusrex | the settings required in sip_nat.conf weren't there. |
| 22:36.49 | quentusrex | now it all works. |
| 22:36.52 | pids | true. |
| 22:37.33 | gambler1 | Hi, does anyone here use cdr adaptive odbc module? |
| 22:37.34 | [TK]D-Fender | Spirits-Sight: Compile as per the instructions in the tarball |
| 22:37.53 | pids | Spirits-Sight, installs of asterisk are distro indipendent if you use the tarball |
| 22:38.39 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 22:39.07 | pids | like D-Fender said ;) |
| 22:39.38 | Spirits-Sight | where can I download tarball |
| 22:39.55 | pids | www.asterisk.org |
| 22:40.05 | Spirits-Sight | thanks :-) |
| 22:41.00 | pids | wonders if thats in the bot |
| 22:41.41 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:41.41 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:42.01 | [TK]D-Fender | pids: Yes, and in the channel TOPIC |
| 22:42.14 | [TK]D-Fender | pids: You'd almost think we were trying to tell you something... |
| 22:43.34 | pids | yea, my sarcasm was not wasted! |
| 22:44.53 | [TK]D-Fender | pids: An inaluable and limitless resource... |
| 22:44.59 | [TK]D-Fender | pids: An invaluable and limitless resource... |
| 22:45.07 | pids | hehe |
| 22:46.07 | *** join/#asterisk Ccomp5950 (n=Ccomp595@66.190.102.236) |
| 22:49.49 | bijit | my calls are droping since q931_hungup is requested..in the middle of call...anyone has haved this problem that can help me? |
| 22:50.08 | [TK]D-Fender | bijit: Who requested? |
| 22:50.33 | Spirits-Sight | should do the 1.6.0.1 or the 1.4.22 |
| 22:51.57 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 22:55.37 | Assimilate | Spirits-Sight, SOmeone in here had a really good quote the other day. "1.6 is Bleeding edge meaning bleading ulcers, gums etc" |
| 22:56.12 | *** join/#asterisk seanmh (n=seanmh@216.31.101.11) |
| 22:56.34 | Katty | Qwell: Tome of Black Cat |
| 22:56.36 | Katty | Qwell: Dalaran |
| 22:56.39 | bijit | [TK]D-Fender: that is what I am trying to figure out.. Logs say NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active |
| 22:56.39 | Katty | Qwell: 2500g |
| 22:56.44 | Katty | Qwell: <3 |
| 22:56.48 | Spirits-Sight | LOL, so it would be best to go with 1.4 then :-) |
| 22:56.55 | Katty | Qwell: also, brown bear is MINE, 720g, and toy train 250g |
| 22:57.09 | bijit | <PROTECTED> |
| 22:57.23 | Assimilate | Spirits-Sight, I'd give it some time in the field still. |
| 22:57.26 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:57.48 | Spirits-Sight | Sound good to me :-) |
| 22:59.48 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-c3f8ab957401802a) |
| 23:02.36 | [TK]D-Fender | bijit: You are looking at this in bits & pieces.. loot at the debug and see WHO is issuing the cancel |
| 23:03.15 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 23:03.17 | watchy | hey tk you there? |
| 23:03.51 | watchy | ive sold some phone systems with max of 60+ handsets |
| 23:04.00 | watchy | but today i think i sold one with 650 handsets |
| 23:04.06 | watchy | how hard is that gonna be |
| 23:05.41 | [TK]D-Fender | watchy: Quick estimate, about as hard as the last one x 10.5 |
| 23:06.18 | watchy | heh |
| 23:06.23 | watchy | can 1 server handle it? |
| 23:06.38 | watchy | i think its like 6 t1s |
| 23:07.07 | [TK]D-Fender | watchy: "depends" |
| 23:07.24 | [TK]D-Fender | watchy: Should you be asking this if you'ev positioned yourself as qualified tos ell it? :) |
| 23:07.38 | watchy | thats why i may have someone like you come in and help |
| 23:07.50 | [TK]D-Fender | watchy: My rates are very accessable :) |
| 23:08.47 | watchy | i'd love it if i could get you personnaly |
| 23:08.56 | watchy | cause i owe u like alot of steak dinners for helping |
| 23:09.00 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 23:09.00 | watchy | me in the past |
| 23:09.52 | watchy | the biggest i've done is a defense contractor. and they have about 50 polys + 15 or so sipura atas |
| 23:10.02 | watchy | but i'm about to add 30 polys to that system |
| 23:11.35 | watchy | but this school district wants 600-650 handsets. |
| 23:13.01 | watchy | but its something i've been wanting to do. install a nice big * setup |
| 23:13.17 | [TK]D-Fender | watchy: 6 PRI seems overkill... |
| 23:13.55 | watchy | well i'm not %100 what they want pri wise |
| 23:14.35 | watchy | but currently every building the school has its own phone system |
| 23:14.44 | watchy | so i'm sure once its all consolidated the pris will be also |
| 23:15.36 | watchy | luckily they want to roll it out in sections |
| 23:16.08 | watchy | i'm thinking of going with 330s for regular phones and 650s for reception |
| 23:16.55 | kerx | anyone have luck with SIP Peers in Realtime sing mysql database ? |
| 23:17.03 | [TK]D-Fender | watchy: PRAY they don't ask for an "all-page" ;) |
| 23:17.14 | watchy | they already have a paging system |
| 23:17.16 | kerx | I've imported everything in, but for some reason no peer is showing? |
| 23:17.22 | [TK]D-Fender | kerx: No, it doesn't work for anybody. |
| 23:17.26 | watchy | i have to figure out how to tie the * into that tk |
| 23:17.37 | kerx | heh, [TK]D-Fender |
| 23:18.08 | kerx | [TK]D-Fender, are you one of those types of people who purposely bother people for no apparent reason other than you have lots of anger bottled up inside? |
| 23:18.53 | kerx | you can always try suicide |
| 23:18.53 | watchy | tk is nice hes always helped me or told me read the manual |
| 23:18.53 | watchy | but i've always found the answer using one of his sugestions |
| 23:18.59 | kerx | i know |
| 23:19.02 | kerx | i have also |
| 23:19.08 | kerx | i like to bug him though |
| 23:19.21 | watchy | well ddos him |
| 23:19.23 | kerx | he doesn't seem to be responding, so he might have not liked my comments |
| 23:19.29 | kerx | what does DDoS mean? |
| 23:19.36 | watchy | denial of service |
| 23:19.40 | watchy | take his internet offline |
| 23:19.57 | kerx | oh, i wouldn't know how to do that |
| 23:20.24 | [TK]D-Fender | goes to redirect his chan_skinny.so botnet for another strike... |
| 23:20.44 | mahlon | botstrike.agi |
| 23:20.45 | kerx | heh |
| 23:21.07 | watchy | wow they have 670s out. you tried them tk? |
| 23:21.47 | [TK]D-Fender | watchy: 650+gigE. Big deal. I'd never buy one |
| 23:22.02 | [TK]D-Fender | watchy: Costs more that to separately wire for a 650. |
| 23:22.21 | [TK]D-Fender | watchy: And more again when it breaks. and means a brick at the desk. |
| 23:22.27 | [TK]D-Fender | watchy: lose/lose/lose |
| 23:22.36 | [TK]D-Fender | watchy: in any sane deployment |
| 23:22.44 | watchy | looks like a 670 is color |
| 23:23.22 | watchy | i sold a 650 with a backlit sidecar. they are quite nice |
| 23:24.39 | [TK]D-Fender | watchy: watchy but the sidecar isn't any smarter... |
| 23:24.56 | watchy | true, but its backlit and matched the backlit of the 650 |
| 23:24.57 | [TK]D-Fender | watchy: Aastra's LCD sidecar is the shiznt y0 |
| 23:25.09 | [8none1] | I agree with [TK]D-Fender. I wouldn't trust the gig switch in the polycoms. For anyone needing gig they should get separate drops |
| 23:25.11 | watchy | you like aastras now? |
| 23:25.33 | hardwire | can you get agent login information from some dialplan functions? |
| 23:25.47 | [TK]D-Fender | watchy: No... the bas phone pisses me right the hell off... but the sidecars are awesome |
| 23:25.48 | [TK]D-Fender | base* |
| 23:25.59 | [TK]D-Fender | watchy: "potential" <- Aastra |
| 23:26.34 | watchy | ah |
| 23:26.46 | watchy | so on a 600+ install would you still rec poly? |
| 23:27.27 | [TK]D-Fender | watchy: I recommend good products regardless of the size |
| 23:27.46 | [TK]D-Fender | watchy: Everything depends on the needs as well. |
| 23:27.50 | watchy | ok well you know your opinion is godlike to me |
| 23:28.24 | watchy | youve made me do research and learn alot of things |
| 23:28.24 | Spirits-Sight | [TK]D-Fender: Ok, more information for the person thats going to do the CentOS + * Setup for me, if I am only using SIP what parts should he install? Not using any hardware. |
| 23:28.33 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 23:28.38 | [TK]D-Fender | Spirits-Sight: * + DAHDA |
| 23:28.40 | [TK]D-Fender | Spirits-Sight: * + DAHDI |
| 23:29.30 | Spirits-Sight | Is that inside the tar or is that on a different site? |
| 23:29.54 | *** part/#asterisk korihor (n=korihor@201.210.239.172) |
| 23:30.28 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 23:30.36 | [TK]D-Fender | Spirits-Sight: GO LOOK. |
| 23:30.52 | Spirits-Sight | [TK]D-Fender: making sure understand * + DAHDA & DAHDI |
| 23:31.02 | [TK]D-Fender | Spirits-Sight: I jsut corrected a typo... |
| 23:31.07 | [TK]D-Fender | (and made another) |
| 23:31.11 | Spirits-Sight | thanks :-) |
| 23:31.55 | [TK]D-Fender | watchy: What do they have now? |
| 23:32.46 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 23:33.26 | kfife | Naturally, some DAHDI settings don't 'take' with a dialplan reload. QUESTION: Do such 'low level' settings require that asterisk be stopped and started, or that the DAHDI service itself be restarted? |
| 23:34.55 | [TK]D-Fender | kfife: Any DAHDI changes require the module to be reloaded taking down any calls on it |
| 23:35.29 | watchy | tk: the school? |
| 23:36.19 | kfife | [TK]D-Fender: Thanks. I appreciate it. Is that accomplished by restarting the service? What's the proper way to do that? |
| 23:36.30 | kfife | ...services |
| 23:36.41 | [TK]D-Fender | kfife: "modeul reload chan_dahdi.so" |
| 23:36.52 | [TK]D-Fender | kfife: "moduel reload chan_dahdi.so" |
| 23:36.55 | [TK]D-Fender | kfife: "module reload chan_dahdi.so" |
| 23:36.59 | [TK]D-Fender | 3rd times the charm |
| 23:37.01 | kfife | lol |
| 23:37.05 | [TK]D-Fender | watchy: yes |
| 23:37.36 | kfife | [TK]D-Fender: I appreciate it! |
| 23:38.22 | watchy | not sure, i'm meeting with them next week. my boss is the one that talked wity them |
| 23:38.29 | watchy | we have a inside with the IT department |
| 23:41.34 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 23:51.48 | quentusrex | [TK]D-Fender, Do you know of a way to host multiple independant company's on the same asterisk server? and be able to handle the voicemails, IVR's etc? |
| 23:52.10 | [TK]D-Fender | quentusrex: We call it "configuring" in these aprts ;) |
| 23:52.23 | quentusrex | could it be done? |
| 23:52.28 | [TK]D-Fender | quentusrex: Clearly |
| 23:52.50 | [TK]D-Fender | quentusrex: Ain't Raw-Cat Science. |
| 23:53.10 | kfife | quentusrex: once you've leared how to configure one, how to configure multiple 'storefronts' will be self-evident |
| 23:53.30 | quentusrex | Is there any documentation that would demonstrate the principle? |
| 23:53.39 | [TK]D-Fender | quentusrex: ... |
| 23:53.41 | [TK]D-Fender | ~book |
| 23:53.42 | jbot | [book] probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 23:53.43 | *** join/#asterisk moy (n=moy@189.169.61.171) |
| 23:53.51 | [TK]D-Fender | quentusrex: Its all jsut dialplan and voicemail contexts... |
| 23:54.08 | kfife | quentusrex: it's crazy powerful |
| 23:54.08 | [TK]D-Fender | quentusrex: this is not "magic". these is NOTHING "special" about it. |
| 23:54.29 | quentusrex | Is there a good way to automate parts of it? |
| 23:54.52 | quentusrex | so I can move multiple servers at the different branch locations to a single remotely hosted server? |
| 23:54.53 | [TK]D-Fender | quentusrex: that is a dangerously open-ended question I'm not going to even try to answer... |
| 23:55.02 | [8none1] | I've been racking my head trying to fix a voicemail problem. |
| 23:55.42 | quentusrex | [TK]D-Fender, do you know of a good High Availability solution for *? |
| 23:56.16 | kfife | Me too: I've been tryign to LEAVE a voicemail for my ex-girlfriend, but I don't know quite how to say what I want to say :-) |
| 23:56.16 | [TK]D-Fender | quentusrex: All sorts of docs on the WIKI showing cases for this... |
| 23:56.38 | [8none1] | When I have it email it's sending a blank WAV attachment. |
| 23:56.45 | [TK]D-Fender | kfife: somehow "So long bitch" SOUNDS A TAD BITTER, DOESN'T IT? ;) |
| 23:57.29 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |