| 00:00.57 | neurosys | should I have placed the english soundfiles in ./sounds/en? |
| 00:01.09 | ChannelZ | Yes-ish, that's The New Way |
| 00:01.18 | ChannelZ | I can't turn off languageprefix in 1.8 |
| 00:04.34 | ManxPower | neurosys, you need to start reading the upgrade files. |
| 00:04.44 | ChannelZ | It says it was deprecated in UPGRADE-1.6 but the implication is unclear |
| 00:05.25 | ChannelZ | It says the layout "for non-English sounds is the 'new style'" which kind of implies to me that English would still be served out of the root directory if that's where they happened to be |
| 00:05.34 | ManxPower | the implications of deprecated is that it will be removed without notice in the next release, that is how it works |
| 00:05.41 | rene- | i just upgraded to 1.6, will try 1.8 in about three years hehe |
| 00:06.07 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 00:06.09 | ChannelZ | Except it's still listed in the 1.8 sample configs. |
| 00:06.14 | neurosys | ChannelZ: Thx :) |
| 00:06.39 | ChannelZ | sure. It's worth moving everything into an 'en' dir though just to save yourself some pain in the future |
| 00:06.58 | neurosys | ChannelZ: Already done. Thx dude :) |
| 00:16.54 | *** join/#asterisk QbY (~kelvin@96.176.19.11) |
| 00:18.24 | QbY | is it possible to configure asterisk to always send credentials to a particular peer when making a call--we are having a problem where the proxy is challenging asterisk yet asterisk doesn't send the credentials |
| 00:21.16 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 00:22.35 | QbY | this is really awesome |
| 00:22.38 | *** join/#asterisk mac-mini (~mac-mini@unaffiliated/macmini/x-648924) |
| 00:23.49 | QbY | running packet traces, you can see asterisk send the call to the proxy, the proxy send back a 407.. but in the asterisk console you never see that when debugging.. therefore, one has to assume asterisk didn't either, and that's why it didn't send the credentials |
| 00:26.58 | *** join/#asterisk tamiel (~tamiel@ip-81.net-89-3-209.rev.numericable.fr) |
| 00:33.04 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 00:42.24 | *** part/#asterisk nightwalk (~irc@daimon.vixel.org) |
| 00:44.04 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 00:51.14 | *** join/#asterisk coppice (~chatzilla@m121-202-48-125.smartone-vodafone.com) |
| 00:56.36 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 01:14.35 | a1fa | trick or treat |
| 01:16.29 | WIMPy | Is dahdi the only user of CALLERID(ani)? |
| 01:27.47 | *** part/#asterisk rene- (~rene@189.202.122.239.cable.dyn.cableonline.com.mx) |
| 01:33.31 | *** part/#asterisk k-man (~jason@unaffiliated/k-man) |
| 01:34.49 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 01:35.07 | *** part/#asterisk QbY (~kelvin@96.176.19.11) |
| 01:44.23 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 01:44.27 | adnc | hello |
| 01:44.35 | adnc | what is astcanary? |
| 02:08.48 | *** join/#asterisk imcdona (~imcdona@173.160.189.69) |
| 02:16.02 | *** join/#asterisk imcdona (mail@2001:470:e8f1:1:b0cb:6b89:8503:456) |
| 02:20.04 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 02:22.41 | *** join/#asterisk imcdona (mail@2001:470:e8f1:1:b0cb:6b89:8503:456) |
| 02:25.07 | *** part/#asterisk imcdona (mail@2001:470:e8f1:1:b0cb:6b89:8503:456) |
| 02:25.25 | *** join/#asterisk imcdona (mail@2001:470:e8f1:1:b0cb:6b89:8503:456) |
| 02:25.50 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 02:31.57 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 02:36.42 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 03:02.38 | *** join/#asterisk knot (~knotsucke@unaffiliated/devemo) |
| 03:21.43 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 03:25.11 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 03:26.39 | *** join/#asterisk imcdona (~imcdona@173.160.189.69) |
| 03:32.48 | *** join/#asterisk joelsolanki (~joelsolan@124.125.149.128) |
| 03:39.28 | *** join/#asterisk terracon (~greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 03:50.37 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 04:06.25 | *** join/#asterisk jql (~jql@12.9a.344a.static.theplanet.com) |
| 04:06.28 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 04:09.53 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 04:31.06 | *** join/#asterisk coppice (~chatzilla@m121-203-207-234.smartone-vodafone.com) |
| 04:34.57 | *** join/#asterisk radic (~radic@178.2.220.159) |
| 04:41.02 | *** join/#asterisk nitsua (~obscure@her.aclit.us) |
| 04:44.55 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-axpajflczpngyxjd) |
| 04:51.26 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:52.40 | *** part/#asterisk bindbn (~bindbn@109.191.108.115) |
| 04:56.05 | *** join/#asterisk bindbn (~bindbn@109.191.108.115) |
| 04:56.59 | *** part/#asterisk sshock (~sshock@63.248.133.83) |
| 04:57.21 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 05:49.43 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:07.42 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 06:13.25 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 06:23.59 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 06:29.46 | *** join/#asterisk screenn (~screenn@users-nat.more.com.ua) |
| 06:35.54 | *** join/#asterisk Tim_Toady (~moi@178.128.36.116.dsl.dyn.forthnet.gr) |
| 06:52.33 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 06:53.24 | nicola_pav | hello. i have added asteris rx'logger rotate' in /etc/logrotate.conf and i do it daily and 5 times |
| 06:53.38 | nicola_pav | i have lots of full.xxx files that are empty |
| 06:53.47 | nicola_pav | is this normal? |
| 06:55.15 | Tim_Toady | might be, depends on ur logger.conf settings |
| 06:57.50 | nicola_pav | Tim_Today: I have full.0, full.1, full.2, full.3, full.4, full.5, full.6, full.1.gz, full.2.gz, full.3.gz, full.4.gz, full.5.gz non empty |
| 06:58.21 | nicola_pav | and i have lots of full.1x, full.1.xx, full.2.x, full.2.xx,....etc empty files |
| 07:00.07 | ChannelZ | man logrotate |
| 07:03.03 | *** join/#asterisk w0rk3r (~Senny@85.15.88.11) |
| 07:06.32 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 07:13.50 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 07:19.55 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 07:21.10 | *** join/#asterisk Wassim (~IceChat7@193.227.188.210) |
| 07:28.04 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 07:32.24 | verywiseman | is reload stop asterisk and run it , or only reread configuration file? |
| 07:32.40 | ChannelZ | uhmm.. what? |
| 07:34.01 | verywiseman | i mean reload cli |
| 07:34.38 | ChannelZ | I mean "I have no idea what you're asking" |
| 07:35.37 | kaldemar | verywiseman: it doesn't stop asterisk. it reloads most modules and re-reads config files. |
| 07:35.38 | Wassim | i dont see there is quite difference between restarting asterisk and re reading config files |
| 07:36.01 | kaldemar | Wassim: except that all changes don't get applied with a reload. |
| 07:36.07 | ChannelZ | that's mostly the case |
| 07:37.35 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 07:37.42 | Wassim | i have 2 asterisk servers connected via sip trunk, i am able to do one way calls, if i past logs of an unsucceeful call, on pastebin can somebody give it a look? |
| 07:38.49 | ChannelZ | yes |
| 07:39.45 | Wassim | ok, in couple of minutes :) |
| 07:48.14 | Wassim | ChannelZ |
| 07:48.17 | Wassim | http://pastebin.com/6K5ULee9 |
| 07:48.29 | Wassim | that's the unsuccessful call |
| 07:49.38 | *** join/#asterisk emc (~cburgdorf@109.231.205.3) |
| 07:51.18 | kaldemar | Wassim: cause was 21, so the call was rejected by the other end for some reason. for the reason, enable sip debug. |
| 07:51.35 | Wassim | mmm |
| 07:51.37 | Wassim | ok |
| 07:51.42 | kaldemar | Wassim: since you're using freepbx, you better ask in #freepbx first how to resolve the issue. |
| 07:51.51 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 07:51.59 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 07:51.59 | Wassim | how do i enable sip debug? |
| 07:53.48 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 07:54.25 | Wassim | does the freepbx affects how asterisk work? |
| 07:55.34 | *** part/#asterisk emc (~cburgdorf@109.231.205.3) |
| 07:56.14 | kaldemar | Wassim: "sip set debug on" in CLI. freepbx makes your configs, so it affects how asterisk works. |
| 08:05.34 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:05.39 | Wassim | freepbx guys doesn't seem freindly |
| 08:05.40 | Wassim | :S |
| 08:06.20 | Wassim | kaldemar can i ask you how did u figure out that the problem was at 21? |
| 08:06.56 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:13.08 | kaldemar | Wassim: by reading your pastebin. i didn't say anything about line numbers then, i was talking about the hangup cause. as in cause code. |
| 08:13.27 | Wassim | ah ok, and that falls on wish line? |
| 08:14.14 | Wassim | 70 that is |
| 08:14.16 | Wassim | saw it |
| 08:14.17 | Wassim | :) |
| 08:19.05 | Wassim | mmm yeah that means call rejected |
| 08:20.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:20.18 | Wassim | kaldemar that call was rejected by the second server right? |
| 08:20.27 | Wassim | not the one initating the call |
| 08:20.55 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 08:23.01 | kaldemar | probably. but look at the sip debug to be sure on what's happening. |
| 08:23.17 | Wassim | just did that, cant understand it quitely |
| 08:26.16 | Wassim | can you give it a look kaldemar ? i would appreciate it:) |
| 08:29.14 | kaldemar | don't ask to show something, just show it. you'll be more likely to get answers. |
| 08:29.42 | Wassim | http://pastebin.com/XWE8x4iW |
| 08:29.49 | Wassim | there is it ^^ |
| 08:31.23 | kaldemar | what does the CLI of the receiving end say? looks like you have wrong credentials. |
| 08:31.49 | Wassim | i seen an unauthorized |
| 08:31.57 | Wassim | on this end |
| 08:32.01 | Wassim | let me see on the second end |
| 08:33.04 | *** join/#asterisk _trine (~trine@trine1-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 08:34.05 | Wassim | http://pastebin.com/hVSkYfF5 |
| 08:34.15 | Wassim | that's the receiving server |
| 08:35.48 | _trine | good morning, I my asterisk has been under attack for the last 3 days continually from 218.206.248.132 is there any security measures I should have in place from an asterisk point of view |
| 08:36.24 | _trine | his range of IP address's has already been dropped by my router |
| 08:36.26 | kaldemar | Wassim: the call matches to peer28. you most likely have a secret mismatch. |
| 08:36.41 | Wassim | mmm |
| 08:36.45 | Wassim | let me check password |
| 08:39.18 | Wassim | found the problem |
| 08:39.19 | Wassim | :) |
| 08:39.30 | Wassim | not the password |
| 08:39.34 | Wassim | it was the name it self |
| 08:39.41 | Wassim | it was matching on both ends |
| 08:39.44 | Wassim | wasnt* |
| 08:39.54 | Wassim | thanks a lot kaldemar :) |
| 08:43.03 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 09:00.16 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 09:03.07 | *** join/#asterisk thazza (~thazza@110-174-206-208.static.tpgi.com.au) |
| 09:06.59 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 09:07.04 | *** join/#asterisk Dovid (~Dovid@213.8.118.62) |
| 09:14.52 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 09:15.40 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 09:25.45 | Dovid | are there no asterisk addons for 1.8 ? |
| 09:34.59 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 09:36.14 | tzafrir_laptop | Dovid, they are included in the main asterisk tarball |
| 09:40.02 | Dovid | tazfrir_laptop: thanks. |
| 09:40.13 | Dovid | does 1.8 need Dahdi as a timing source any more ? |
| 09:42.05 | ectospasm | Dovid: Asterisk hasn't needed a DAHDI card for timing in quite some time. |
| 09:47.15 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 09:52.06 | kaldemar | ectospasm: not a card, but a DAHDI timer. |
| 09:54.52 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 09:59.58 | eject_ck | Can someone recommend VoIP/GSM+PSTN gateway ? |
| 10:01.14 | shamelessn00b | visiting http://www.asterisk.org/doxygen/trunk/index.html gives me 403 forbidden :/ |
| 10:05.16 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 10:16.48 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 10:35.35 | *** join/#asterisk joobie (~joobie@CPE-121-219-110-161.lnse2.lon.bigpond.net.au) |
| 10:39.45 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 10:54.13 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 10:58.31 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 10:59.27 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 11:05.54 | *** join/#asterisk imcdona (~imcdona@173.160.189.68) |
| 11:36.46 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 11:46.00 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 11:46.14 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 11:47.17 | *** join/#asterisk BMJ (~bjohns@cpe-098-026-116-043.nc.res.rr.com) |
| 11:50.13 | *** join/#asterisk knot (~knotsucke@unaffiliated/devemo) |
| 11:59.28 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:03.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:04.03 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 12:04.32 | adnc | hello, i can not find much about astcanary. could someone please point me to a documentation or just tell me what it is? |
| 12:07.28 | *** join/#asterisk anny__ (~chatzilla@77.42.241.114) |
| 12:08.00 | fauxalliance | adnc, JFGI |
| 12:11.19 | adnc | fauxalliance, jfgi? |
| 12:11.24 | adnc | i don't understand |
| 12:11.30 | fauxalliance | just f*cking google it |
| 12:11.42 | adnc | but i did and i couldn't find anyting usefull |
| 12:11.46 | fauxalliance | astcanary is a 'container' for the asterisk process.... |
| 12:11.53 | fauxalliance | google for further information. |
| 12:11.58 | [TK]D-Fender | adClearly its a process to check that * is still alive and running |
| 12:11.59 | adnc | fauxalliance, thank you |
| 12:12.01 | fauxalliance | (google harder) |
| 12:12.01 | [TK]D-Fender | adnc: ^ |
| 12:15.47 | nicola_pav | hello. i installed yesterday digium card te220 on gentoo |
| 12:15.58 | nicola_pav | i could run pri show spans in ast command line |
| 12:16.04 | nicola_pav | i rebooted the system |
| 12:16.10 | nicola_pav | and now |
| 12:16.14 | nicola_pav | i can't see the pris |
| 12:16.18 | nicola_pav | PRIs* |
| 12:16.37 | nicola_pav | there is no pri command in asterisk command line |
| 12:16.44 | nicola_pav | any hint? |
| 12:17.06 | fauxalliance | what version asterisk nicola_pav |
| 12:17.50 | nicola_pav | asterisk 1.4.27.1 |
| 12:18.05 | nicola_pav | before the reboot, i had pri command |
| 12:18.09 | nicola_pav | now no |
| 12:18.24 | fauxalliance | old, but nonetheless, the PRI command should be there, perhaps check the loaded modules at the console |
| 12:19.08 | [TK]D-Fender | nicola_pav: You probably forgot to initialize Zaptel/DAHDI **BEFoRE** starting Asterisk |
| 12:19.49 | nicola_pav | i found it |
| 12:20.00 | nicola_pav | i ran module load chan_dahdi.so |
| 12:20.07 | fauxalliance | \o/ |
| 12:20.23 | *** join/#asterisk [cannibalera] (~cannibale@201-25-253-81.fnsce703.dsl.brasiltelecom.net.br) |
| 12:23.49 | *** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca) |
| 12:24.34 | McBoingbo | If you are only going to install 1 instance of Asterisk SCF is it worthwhile or is it meant to be clustered? |
| 12:25.14 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 12:26.32 | fauxalliance | McBoingbo, well, loadbalancing "one" is pretty bloody redundant.... |
| 12:26.37 | McBoingbo | nm ubder heavy development at the moment |
| 12:26.50 | [TK]D-Fender | fauxalliance: Actually... if it's jsut 1, it is'nt redundant :p |
| 12:26.59 | McBoingbo | FAIL lol |
| 12:27.11 | [TK]D-Fender | Which IS the problem. |
| 12:27.20 | russellb | right, at this point you're welcome to install it to play with it, but it's mostly inteded for developers right now |
| 12:27.47 | BMJ | maybe that conversation belongs in the asterisk-scf room? |
| 12:28.02 | russellb | it could, but it's still relevant here, too, IMO. |
| 12:28.33 | BMJ | true. |
| 12:29.59 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:29.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:30.22 | ManxPower | Everyone, keep an eye out on your Asterisk boxes, massive wave of "hack" attempts, some of them successful have been seen. Appears to be exploiting weak passwords on SIP accounts. |
| 12:30.53 | tzanger | ManxPower: yep. over 450 unique IPs banned on my box alone since Friday |
| 12:31.01 | russellb | wow |
| 12:32.01 | adnc | if i only use sip and iax2 do i still need chan_dahdi.so? |
| 12:32.08 | russellb | no. |
| 12:32.18 | russellb | ... unless you use MeetMe(), then you should keep it. |
| 12:32.19 | [TK]D-Fender | adnc: YES, it you use TRUNK mode |
| 12:32.32 | [TK]D-Fender | russellb: Silly wabbit :) |
| 12:32.36 | *** join/#asterisk Wassim (~IceChat7@193.227.188.210) |
| 12:32.36 | russellb | untrue. |
| 12:32.43 | russellb | chan_iax2 never used chan_dahdi directly. |
| 12:32.44 | adnc | trunk mode? i used debians asterisk 1.6 package, |
| 12:32.51 | russellb | and in 1.6.2 and up, it doesn't use dahdi directly at all |
| 12:33.01 | Wassim | hi , what does this mean in SIP trunking ? SV_DIALSTATUS=CHANUNAVAIL") |
| 12:33.06 | russellb | who's the silly wabbit! |
| 12:33.15 | adnc | because i do get ERROR[4328] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory |
| 12:33.25 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 12:33.29 | russellb | Wassim: where did you see that? |
| 12:33.41 | russellb | adnc: don't load that module. |
| 12:33.47 | [TK]D-Fender | adnc: And why haven't you installed DAHDI? |
| 12:33.51 | Wassim | i have 2 asterisk conncted with SIP trunks, one can call the other |
| 12:33.55 | Wassim | the other cant |
| 12:34.03 | kaldemar | Wassim: that's freepbx output. |
| 12:34.08 | adnc | [TK]D-Fender, i didn't know that i would need it. |
| 12:34.10 | [TK]D-Fender | adnc: And 1.6 is not a "version". It isn't even a BRAQNCH |
| 12:34.14 | Wassim | ah |
| 12:34.19 | Wassim | ok |
| 12:34.22 | Wassim | sorry |
| 12:34.29 | adnc | Asterisk 1.6.2.10-1 |
| 12:34.34 | adnc | is the version i do have |
| 12:35.14 | [TK]D-Fender | adnc: Better. Well no idea why your build includes chan_dahdi without requiring DAHDI in the first place. SOunds like a bad packager |
| 12:36.26 | ManxPower | only crazy people install asterisk from a package |
| 12:36.33 | fauxalliance | tries to emerge 'clue' |
| 12:37.07 | [TK]D-Fender | ManxPower: I prefer not to shit all over the "packages" outright and just target the ones that suck in particular ;) (with the appropriate disclaimer attached) |
| 12:37.24 | adnc | i'm trying to google all the possible I do not need. most of the results are forum entries. is there a way getting information about these modules (brief description)? |
| 12:37.28 | adnc | somehow |
| 12:37.34 | tzafrir_laptop | adnc, unless you have a transcoder card, chances are this is a false alarm |
| 12:37.46 | tzafrir_laptop | You don't need codec_dahdi |
| 12:37.49 | adnc | tzafrir_laptop, no i don't have one |
| 12:37.53 | [TK]D-Fender | adnc: Google can't etll you what you need. It might help give you an idea of what the PIECES are for though. |
| 12:38.09 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 12:38.11 | tzafrir_laptop | codec_dahdi is actually not related to chan_dahdi |
| 12:38.15 | adnc | [TK]D-Fender, no, not need. just a short description |
| 12:38.16 | [TK]D-Fender | adnc: But who knows how long you'll have to look to feel confident that you know enough to safely NOT do seomthing |
| 12:38.48 | adnc | [TK]D-Fender, i don't understand what you are trying to say. what I'm looking for is a brief description of those modules, that is not much i want |
| 12:38.51 | *** join/#asterisk coppice (~chatzilla@m121-202-50-139.smartone-vodafone.com) |
| 12:39.31 | *** join/#asterisk ttwhy (~tekkno@p4FECF859.dip.t-dialin.net) |
| 12:40.00 | russellb | this chapter lists all modules with a brief description: http://ofps.oreilly.com/titles/9780596517342/ch01.html |
| 12:40.11 | leifmadsen | russellb: oh snap |
| 12:40.33 | russellb | yeah, i went there |
| 12:40.56 | [TK]D-Fender | adnc: app_chanisavailChecks the status of a channelUnreliable <- LIES! HUgely useful and used by boatloads of people! :) |
| 12:41.23 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 12:41.23 | russellb | the way that module is implemented is crap |
| 12:41.28 | russellb | you shouldn't use it. |
| 12:41.34 | WIMPy | Obviousely depends on channeltype. |
| 12:41.51 | WIMPy | And that fact implies "unreliable" to me. |
| 12:42.11 | adnc | mh |
| 12:42.28 | *** join/#asterisk Bloudermilk (~Bloudermi@166.205.139.172) |
| 12:42.32 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:42.34 | russellb | ${DEVICE_STATE(...)} is a better choice for that kind of thing. |
| 12:42.37 | Bloudermilk | Morning everyone |
| 12:43.07 | WIMPy | Same problem, however. |
| 12:43.21 | [TK]D-Fender | russellb: Yeah I suppose that could viably replace it... |
| 12:44.44 | Bloudermilk | Has anyone experienced the error: Starting asterisk: Cannot find specified TTY (9) |
| 12:44.49 | *** join/#asterisk knot (~knotsucke@unaffiliated/devemo) |
| 12:44.53 | Bloudermilk | I'm on CentOS 5.5 |
| 12:45.54 | Bloudermilk | I haven't been able to find anything other than people suggesting to "remove that part of safe_asterisk", which seems like a terrible hack |
| 12:46.04 | *** join/#asterisk chasing`Sol (~rc4@62.114.139.78) |
| 12:46.04 | Bloudermilk | I've always used RHEL before and never had that problem |
| 12:46.08 | adnc | I've changed from asterisk 1.4 to 1.6 modules.conf is loading cdr_mysql.so which i would like to use for cdr records in mysql. unfortunately i do get: /usr/lib/asterisk/modules/cdr_mysql.so: cannot open shared object file: No such file or directory |
| 12:46.16 | adnc | is it possible that the module name changed? |
| 12:46.18 | Bloudermilk | Didn't think I'd see differences when switching to CentOS |
| 12:46.49 | adnc | i do have for example cdr_addon_mysql.so |
| 12:47.01 | adnc | is this the same as cdr_mysql.so from 1.4? |
| 12:50.20 | *** join/#asterisk m0t3jl (~petr.mote@77.78.93.40) |
| 12:50.26 | [TK]D-Fender | adnc: Perhaps you should read all of the UPGRADE docs from your version <- |
| 12:50.44 | adnc | [TK]D-Fender, I did but there wasn't any indication to that |
| 12:51.02 | [TK]D-Fender | adnc: Yo also read the upgrades for 1.6.1, and 1.6.0? |
| 12:51.13 | adnc | ohh no |
| 12:52.41 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 12:56.56 | *** part/#asterisk Husky (Smegma@unaffiliated/coil) |
| 12:57.08 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 12:57.28 | Bloudermilk | Hmmm... Less hackish that I thought. You can just comment out the line that declares which TTY to use and it skips the whole process |
| 13:00.10 | *** join/#asterisk bjornts (~Adium@2001:700:200:10:d69a:20ff:fe00:c54e) |
| 13:00.51 | *** join/#asterisk bjornts (~Adium@2001:700:200:10:d69a:20ff:fe00:c54e) |
| 13:19.52 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 13:26.34 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 13:28.56 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 13:29.02 | Katty | morning |
| 13:29.40 | tuxx- | hiya! |
| 13:29.55 | Katty | my asterisk does not work at all how to fix plsss |
| 13:30.21 | tuxx- | what does `asterisk -vvvvvvvvvvvvvvv` tell you? |
| 13:30.26 | tuxx- | ~pastebin |
| 13:30.27 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 13:30.31 | Katty | that it is too early for verbosity. |
| 13:30.34 | tzafrir_laptop | Katty, please plug it to the power |
| 13:30.39 | tuxx- | hehe :P |
| 13:30.40 | Katty | oooh power. |
| 13:30.41 | Katty | right. |
| 13:31.04 | tzafrir_laptop | More power to the Asterisks |
| 13:31.28 | Katty | russellb: ohai |
| 13:31.32 | tuxx- | :-) |
| 13:31.42 | Katty | russellb: just got your request |
| 13:31.45 | russellb | Katty: ohai! |
| 13:31.49 | russellb | request denied! |
| 13:31.58 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 13:32.23 | Katty | russellb: i almost forgot i had a twitter account |
| 13:32.30 | russellb | Katty: you can blame the suggestions box on the twitter web site |
| 13:32.40 | russellb | disappears |
| 13:32.42 | E-bola | How come pickupgroup/callgroup doesnt honor contexts? |
| 13:33.39 | fauxalliance | E-bola, i'd be willing to bet it does exactly as it was programmed to. |
| 13:34.08 | E-bola | fauxalliance: maybe, but then the obvious followup question would be. Why doesnt it honor contexts.... |
| 13:34.25 | fauxalliance | E-bola, you will have to SHOW US what you are talking about... |
| 13:35.25 | fauxalliance | http://www.voipplanet.com/backgrounders/article.php/3632556 |
| 13:35.32 | fauxalliance | ring groups + contexts.... |
| 13:36.05 | E-bola | fauxalliance: ummm i thought it was pretty clear |
| 13:36.06 | leifmadsen | dances |
| 13:36.30 | E-bola | do you know what a pickupgroup is, that link has ntohing to do with it.... |
| 13:36.54 | E-bola | im referring to Callgroup= and pickupgroup= in sip.conf |
| 13:37.19 | E-bola | if i create a sip account in context=test1 and another one in context=test2 |
| 13:37.29 | fauxalliance | give or take a few buzzwords... i don't see and LOGFILES or CLI output |
| 13:37.33 | E-bola | if they have the same callgroup/pickupgroup they can pick up each others call's |
| 13:37.36 | *** join/#asterisk sevard (sev@unaffiliated/sevard) |
| 13:37.57 | sevard | any international users here? I'm trying to find the sms email gateway for turkcell |
| 13:38.03 | E-bola | fauxalliance: trolling? |
| 13:38.13 | fauxalliance | E-bola, pastebin? |
| 13:38.41 | E-bola | fauxalliance: thereøs nothing to pastebin. Its a simple enquery about a feature in asterisk. Has nothing to do with any specific dialplan. Please try to read my past 2 lines |
| 13:38.56 | Katty | dances with leifmadsen |
| 13:38.58 | telnettech | when doing Blacklist for your dialplan, is it the same in 1.8 version? |
| 13:39.07 | Katty | leifmadsen: are you hearing less about the rally now? |
| 13:39.51 | fauxalliance | loosly translates E-bola's muttering to "If all my calls are hitting one context in sip.conf, from where they are passed to extensions.conf, and then passed to appropriate contexts, how can I have pickup features enabled per context" |
| 13:40.05 | fauxalliance | can someone help him,, this troll is gone for caffeine |
| 13:40.18 | ManxPower | E-bola, what is your question? |
| 13:40.19 | E-bola | hehe |
| 13:41.02 | fauxalliance | too gets up to cut a rug and join in the festivities... |
| 13:41.08 | E-bola | ManxPower: My question is whether or not i should be able to have identical pickupgroup id's for sip accounts in different contexts without them interfering with each other. Basically if pickupgroup/callgroup caresd about context |
| 13:41.24 | fauxalliance | ^^ wow, that was actually coherenty |
| 13:41.28 | ManxPower | It should not care about contexts |
| 13:41.47 | ManxPower | AFIK, callgroup/pickupgroup are global |
| 13:41.55 | E-bola | Thats precisely what im seeing |
| 13:42.04 | E-bola | i just can't understand why it would be implemented like that |
| 13:42.17 | E-bola | so i was asuming it must be some error on my part |
| 13:42.21 | ManxPower | why would it not be? |
| 13:42.35 | fauxalliance | uh ohh... assume. |
| 13:42.38 | E-bola | because its counterintuitive, for me atleast |
| 13:42.48 | fauxalliance | albeit documented |
| 13:43.01 | E-bola | If 2 accounts are seperated in 2 diff. contexts they shouldnt have any "global" variables |
| 13:43.20 | ManxPower | that is not how Asterisk works. |
| 13:43.44 | ManxPower | 2 accounts that are in different contexts can't talk to each other unless you set it up that way |
| 13:44.17 | E-bola | what other variables for a sip account is "global" besides callgroup/pickupgroup. I do not believe i've come across any |
| 13:44.25 | fauxalliance | E-bola, Trolling? |
| 13:44.31 | ManxPower | callgroup/pickupgroup are not variables |
| 13:44.41 | E-bola | fauxalliance: what happened to your caffeine run.... |
| 13:44.48 | ManxPower | E-bola, anything in [general] will be shared, any GLOBAL dialplan variables. |
| 13:45.24 | ManxPower | why don't you FIX YOUR PICKUP groups? |
| 13:45.29 | E-bola | ManxPower: Well no need to argue if its clever or not, as long as I can conclude that its working how its supposed to |
| 13:46.00 | E-bola | ManxPower: It unfortunately just means it's 1 more thing i have to manualy keep track of. It would be easier for me if i only had to keep track of contexts not overlapping |
| 13:46.06 | ManxPower | You can put a device (SIP, IAX2, DAHDI) in multiple pickup and call groups |
| 13:46.17 | fauxalliance | hands ManxPower some karma, and something to calm the nerves. |
| 13:46.44 | Katty | upvotes fauxalliance |
| 13:47.09 | Katty | so i tried DND for the first time this weekend. |
| 13:47.15 | Katty | it wasn't as nerdy as i was expecting. |
| 13:47.46 | E-bola | hmm reminds me i gotta fix a way to sync dnd status with my snom phone expansion panel blf's |
| 13:47.56 | fauxalliance | http://en.wikipedia.org/wiki/HeroQuest @ Katty |
| 13:47.59 | fauxalliance | :-D |
| 13:48.08 | fauxalliance | ORK! |
| 13:48.10 | Katty | that sounds like too much reading for 9am |
| 13:48.16 | Katty | you eamn orc |
| 13:48.30 | fauxalliance | potatoe, potato |
| 13:48.36 | Katty | true dat |
| 13:48.45 | Katty | and i'm irish. i like me some potato |
| 13:48.58 | fauxalliance | Katty, last name is "Clancey" |
| 13:49.04 | Katty | Tom?!?! |
| 13:49.21 | fauxalliance | he dropped the e |
| 13:49.30 | Katty | oh. right. k |
| 13:49.51 | Katty | well my last name is not very irish sounding |
| 13:50.08 | Katty | but that's because grandpa was a german |
| 13:50.10 | fauxalliance | I bet it is after a few Guinness.. |
| 13:50.20 | fauxalliance | nm then ;-) |
| 13:50.36 | Katty | grandpa's last name was Fuhrmann |
| 13:50.45 | Katty | and grandma's was ohaver |
| 13:51.16 | fauxalliance | Fuhrmann is a nice ' |
| 13:51.22 | fauxalliance | strong german name... |
| 13:51.36 | Katty | Philip |
| 13:51.40 | fauxalliance | distinguished even... /me just sounds like 'lace curtain' irish |
| 13:51.54 | Katty | how grand |
| 13:52.41 | *** join/#asterisk fofware (~Fabian@host184.190-226-209.telecom.net.ar) |
| 13:58.17 | Katty | no birdies at the birdie buffet this morning |
| 13:58.22 | Katty | infobot: crittercam |
| 13:58.22 | infobot | from memory, crittercam is Katty's live broadcast of The Nut House at http://www.ustream.tv/channel-popup/squirrel-critter-cam |
| 13:58.46 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-txejhgpbhyjkwuwy) |
| 13:59.06 | *** join/#asterisk telnettech_lap (~telnettec@adsl-80-42-173.sdf.bellsouth.net) |
| 14:00.30 | telnettech | had to make sure i can get on the channel no matter which pc i am using so that I can talk at home as well |
| 14:01.16 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 14:02.53 | vinhdizzo | I'd like to get a phone number forwarded to a fax number (online service). I'm doing it via exten => s,n,Dial(). What else should I do to get the fax data forwarded? I believe asterisk is currently treating it as a voice call and hence fax does not work. |
| 14:03.31 | *** join/#asterisk silkcut (silkut@plox.tor.hu) |
| 14:04.55 | Katty | telnettech: you should just screen it |
| 14:05.00 | Katty | telnettech: that way you don't miss anything :> |
| 14:06.13 | telnettech | Katty: I want to be able to get help not only at work on the systems we have running but I put an * box at the home to control the number of disrespectful boys that call for my daughters |
| 14:06.44 | Katty | grins |
| 14:06.52 | telnettech | kinda like another "shotgun" to keep the critters away :) |
| 14:07.04 | [TK]D-Fender | vinhdizzo: Go rad up on T.38 and pray your ITSP is compliant |
| 14:07.05 | Katty | you're not going to be able to keep them away from boys, you know |
| 14:07.06 | [TK]D-Fender | read* |
| 14:07.08 | *** join/#asterisk eliel (~quassel@201.234.94.226) |
| 14:07.18 | Katty | you should probably just accept they're going to date whot hey want to date ;) |
| 14:07.23 | vinhdizzo | [TK]D-Fender: in the boook? |
| 14:07.26 | Katty | just make sure they feel like they can talk to you about anything |
| 14:08.08 | telnettech | I have no problem with them dating....I have a problem with the lack of respect shown by the children these days due to the lack of parenting by some |
| 14:08.11 | [TK]D-Fender | vinhdizzo: Not in the old one.. maybe the new one. Get googling, check the WIKI, and check the docs in your tarball (1.6.2 HIGHLY recommended) |
| 14:08.30 | vinhdizzo | [TK]D-Fender: and yea, i'm hoping it is T.38 compliant. I want to see what config I need set up first. Then if it isn't, I'll look into an alternative |
| 14:09.13 | vinhdizzo | [TK]D-Fender: new asterisk book soon? |
| 14:09.19 | vinhdizzo | *out soon |
| 14:09.26 | telnettech | so I want to "weed" them out....If they cant call in and cant call out to them, then they will have to date someone else that is acceptable to mom and I |
| 14:09.30 | [TK]D-Fender | vinhdizzo: very soon |
| 14:09.33 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:09.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:09.39 | vinhdizzo | cool |
| 14:12.16 | *** join/#asterisk Guest71563 (~saint42@c-24-0-99-39.hsd1.nj.comcast.net) |
| 14:14.21 | Guest71563 | hi all.. can someone tell me what a sip hard/soft phone needs in order to work in a private network, if I use gizmo for example ? I know that sip is supposed to do rtp between the 2 points, but if my phone is bon my private network what is it that it need to work ? (what i am looking for is the name of the technology) |
| 14:15.15 | *** part/#asterisk guyzmo (~guyzmo@nenya.mithrandir.net) |
| 14:15.41 | p3nguin | guest71563: NAT |
| 14:16.28 | p3nguin | ~nat |
| 14:16.29 | infobot | it has been said that nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 14:16.30 | Guest71563 | p3nguin: but if i do nat , my phone is going to try to contact with the only internal IP address that he knows, right ? so the server on the internet is going to try to get back to this address, no ? isn't there something else ? |
| 14:16.41 | vinhdizzo | is Google Voice confirmed to not support T.38 at all? that is, using it to forward to a fax machine via Dial() will not work? |
| 14:17.00 | p3nguin | guest71563: Many phones have NAT traversal mechanisms built in. |
| 14:17.10 | p3nguin | guest71563: Asterisk also has it built in. |
| 14:17.18 | vinhdizzo | I'd like to use a GV number as a universal fax number, in case I switch fax service |
| 14:19.00 | Guest71563 | p3nguin: so basically, if I have an asterisk behind a private network, and 2 phones behind 2 other private network.. the fact that i do nat on astersik side is supposed to work ..? |
| 14:19.11 | Guest71563 | p3nguin: there is no nat or other stuff to do on the 2 phones sides ? |
| 14:19.22 | p3nguin | guest71563: That's correct -- Asterisk has NAT support included. |
| 14:19.53 | p3nguin | guest71563: You'll still need NAT for any time you connect a public internetwork to a private LAN. |
| 14:20.00 | Guest71563 | p3nguin: ok , i understand this part on the asterisk... i was just wondering on the phones sides.. if 1 phone tries to call the other phone then.. asterisk does all the work ? |
| 14:20.24 | ManxPower | ~answers |
| 14:20.24 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 14:20.56 | Guest71563 | p3nguin: where i got confused is how then the 2 phones would talk since they are both on 2 different private networks... |
| 14:21.06 | E-bola | ManxPower: Thanks for clearing up how pickupgroups work in reagards to contexts |
| 14:21.25 | p3nguin | guest71563: Let them talk to each other via Asterisk. |
| 14:23.48 | *** join/#asterisk Bloudermilk (~Bloudermi@cpe-76-90-0-11.socal.res.rr.com) |
| 14:27.53 | [TK]D-Fender | vinhdizzo: Almost guaranteed that GV won't support T.38 |
| 14:27.56 | [TK]D-Fender | vinhdizzo: google VOICE. |
| 14:28.31 | vinhdizzo | [TK]D-Fender: so should I not even bother? |
| 14:28.48 | [TK]D-Fender | vinhdizzo: Expect failure. That's all... |
| 14:29.08 | vinhdizzo | ok |
| 14:29.12 | [TK]D-Fender | vinhdizzo: GV is also a terrible idea for anything business related |
| 14:29.47 | vinhdizzo | when google first started, i swear i had it working with sipgate's fax feature |
| 14:29.52 | vinhdizzo | *google voice |
| 14:31.23 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 14:32.18 | [TK]D-Fender | vinhdizzo: Keep swearing... it'll be good exercise... |
| 14:32.23 | vinhdizzo | hahahahhaa |
| 14:32.48 | vinhdizzo | yea, i'll keep reading |
| 14:33.02 | vinhdizzo | dont know if i'll try. i'll be expecting failure though |
| 14:33.30 | [TK]D-Fender | vinhdizzo: Don't be surprised is all... |
| 14:33.54 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:34.52 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 14:35.55 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 14:37.09 | keith4 | when using Directory() with realtime sip and voicemail... what field is it looking at? I have names in the "fullname" field of the voicmail users table, but can't get Directory() to match anything. the wiki page doesn't really talk about realtime setups |
| 14:38.17 | Bloudermilk | Anybody here have any luck running Asterisk on CentOS? I'm having issues with TTY |
| 14:38.31 | Bloudermilk | I'm on 5.5 and TTY (9) doesn't seem to exist by default |
| 14:38.49 | [TK]D-Fender | bougyman: You don't really need the TTY you know... |
| 14:38.56 | [TK]D-Fender | Bloudermilk: rather |
| 14:39.00 | bougyman | probably not. |
| 14:39.01 | bougyman | :) |
| 14:39.02 | p3nguin | Hundreds of people use Asterisk on CentOS. |
| 14:39.29 | Kobaz | bajillions |
| 14:39.34 | Bloudermilk | [TK]D-Fender: When I comment out the config line in safe_asterisk, it doesn't fork properly |
| 14:39.43 | p3nguin | As a matter of fact, every person who uses AsteriskNOW uses Asterisk on CentOS. |
| 14:40.04 | Bloudermilk | p3nguin: Good to know my experiences could inspire others |
| 14:40.04 | [TK]D-Fender | And everyone on Trixbox, PIAF, etc |
| 14:40.33 | Bloudermilk | I'm having some gnarly issues with it |
| 14:40.40 | Bloudermilk | TTY being the most annoying one ATM |
| 14:40.43 | p3nguin | You don't need tty9, for sure. |
| 14:41.05 | Bloudermilk | Is there some proper way to disable it so that running "service asterisk start" forks? |
| 14:41.35 | p3nguin | Are you suggesting the init script is broken? |
| 14:42.54 | p3nguin | I'm also missing the connection between this alleged broken init script and tty9. |
| 14:43.04 | Bloudermilk | I either get a "Could not find specified TTY" (default install) or the forking issues when I comment out the TTY=9 line at the beginning of the safe_asterisk script |
| 14:43.18 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:45.22 | *** join/#asterisk russellb_ (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 14:45.22 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 14:46.40 | Bloudermilk | p3nguin: It's a seemingly common error, the TTY thing: http://forums.whirlpool.net.au/archive/607081 |
| 14:46.58 | Bloudermilk | But commenting that line out seems to do more harm than good |
| 14:48.13 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:50.18 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 14:51.04 | p3nguin | I don't trust anyone who talks about violins and Asterisk in the same post. |
| 14:52.09 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 14:57.42 | ManxPower | We had 172,635 reg attempts on Oct 31 |
| 14:58.09 | p3nguin | How many of them were legit? |
| 14:58.11 | Kobaz | that's a bunch |
| 14:58.15 | ManxPower | p3nguin, zero |
| 14:59.52 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 15:03.18 | echinos | It would be cool to have a tarpit for that.. although the script might not stop if it gets a reg |
| 15:03.35 | p3nguin | honey pot |
| 15:03.42 | p3nguin | Let 'em register. |
| 15:03.57 | p3nguin | Just don't give them anything to do after they're registered. |
| 15:04.41 | ChannelZ | sooo, then you just get a bunch more traffic while they try to place calls and figure out your dialplan |
| 15:04.46 | echinos | i'd like to route them to something annoying ifthey try to call in |
| 15:05.08 | echinos | but I know it's more trouble that it's worth |
| 15:05.12 | ChannelZ | firewall them off and be done with it, they've wasted enough of your bandwidth |
| 15:05.53 | p3nguin | The purpose of a honey pot is to attract the crap and keep it away from your real system. It's obvious that you'll spend resources on the honey pot. |
| 15:05.54 | echinos | yeah you're right, I'm just daydreaming |
| 15:06.09 | sevard | Any international users here? I'm trying to find the sms email gateway for turkcell |
| 15:06.20 | sevard | with no luck. |
| 15:06.22 | ManxPower | "Thank you for calling the United States FBI. Please hold for an Agent" |
| 15:07.51 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:08.54 | *** join/#asterisk coppice (~chatzilla@116.92.195.24) |
| 15:13.05 | telnettech | Manx: probably get into trouble doing that; impersonating a law enforcement office or something like that |
| 15:13.29 | echinos | yeah, but it's fun to imagine |
| 15:13.43 | Qwell | ~wiki |
| 15:13.58 | Qwell | ~asteriskwiki |
| 15:13.58 | infobot | asteriskwiki is, like, at http://www.voip-info.org/wiki-Asterisk" |
| 15:14.10 | Qwell | infobot: no, asteriskwiki is http://wiki.asterisk.org |
| 15:14.10 | infobot | Qwell: okay |
| 15:14.20 | telnettech | wonder if there is a way to see what their IP is and read it back to them or start your own sipvicious attack back |
| 15:14.56 | telnettech | give them something to worry about themselves |
| 15:15.27 | p3nguin | Seems like a waste of my own time to do such things. |
| 15:15.52 | *** join/#asterisk NiugeS (~NiugeS@mail.cjs-solutions.com) |
| 15:15.53 | ChannelZ | Unless you can electrocute them remotely back through their connection, little is worth doing |
| 15:16.14 | p3nguin | Death to idiots = worth it |
| 15:16.17 | fauxalliance | not even with Chinese Cepstral? |
| 15:16.57 | WIMPy | proposes the Etherkiller Routing Protocol. |
| 15:17.08 | pabelanger | Anybody using primus for co-location? |
| 15:17.09 | Guest71563 | p3nguin: when you wrote earlier "let them talk to each other via asterisk", does it mean i need to configure a proxy in each of the phone, and put the asterisk ip address ? |
| 15:18.09 | p3nguin | guest71563: That depends on the phone. Sometimes "proxy" is the right setting to communicate with Asterisk, sometimes it isn't. |
| 15:20.08 | NiugeS | hi all.. |
| 15:20.10 | NiugeS | <PROTECTED> |
| 15:21.38 | [TK]D-Fender | NiugeS: Asking for trixbox support in here. |
| 15:21.50 | WIMPy | NiugeS: If you hadn't mentioned |
| 15:21.55 | WIMPy | ~trixbox |
| 15:21.55 | infobot | from memory, trixbox is SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 15:21.59 | WIMPy | I would have suggested |
| 15:22.00 | [TK]D-Fender | NiugeS: Your NAT/firewall settings are probably wrong. Go ask in their channel as to where to go to fix these |
| 15:22.03 | WIMPy | ~sipnat |
| 15:22.03 | infobot | sipnat is, like, Quick guide on configuring Asterisk + 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 |
| 15:22.53 | ManxPower | yup, you might have better luck asking on the correct channel |
| 15:23.28 | russellb_ | we might have better luck growing channel membership if there were fewer people being assholes all the time |
| 15:23.44 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 15:23.56 | BMJ | agreed |
| 15:25.33 | ManxPower | russellb, I agree. The freepbx people need to use the correct channel. |
| 15:25.37 | russellb_ | I think I'm going to just start banning everyone who is an asshole. |
| 15:25.54 | russellb_ | Let all be warned, I'm tired of this channel being so hostile, it's clean up time. |
| 15:25.59 | ManxPower | russellb, there would not be anyone left. 8-| |
| 15:26.18 | russellb_ | no, all those who have been turned off would start showing up again. |
| 15:26.41 | ManxPower | But where would people go for Asterisk support? |
| 15:27.12 | sevard | https://wiki.asterisk.org/wiki/ |
| 15:27.14 | sevard | Service Temporarily Unavailable |
| 15:27.16 | sevard | ffffffffffffffff |
| 15:27.33 | p3nguin | Are we going to start doing FreePBX and Trixbox support here? |
| 15:27.39 | ManxPower | p3nguin, apparently. |
| 15:27.40 | *** join/#asterisk kjs (~kjs@fedora/kjs) |
| 15:28.06 | sevard | russellb_: yeahhhh do it!!!!!!! fuck em! |
| 15:28.07 | kjs | anyone know of a decent UK provider for VoIP? |
| 15:28.08 | russellb_ | sevard: http://wiki.asterisk.org seems be working for me |
| 15:28.30 | sevard | russellb_: http://wiki.asterisk.org redirects to https://wiki.asterisk.org/wiki/ |
| 15:28.47 | russellb_ | https://wiki.asterisk.org/wiki/dashboard.action |
| 15:28.50 | russellb_ | does that work? |
| 15:28.50 | sevard | which doesn't work for me ;/ |
| 15:29.09 | sevard | it's just spinning -- waiting for it to respond. |
| 15:29.13 | russellb_ | got an ACK that it's broken ... someone is working on it |
| 15:29.13 | russellb_ | sorry |
| 15:29.20 | sevard | nbd |
| 15:29.29 | sevard | why https, btw? |
| 15:29.40 | russellb_ | sevard: why not? :-) |
| 15:29.46 | sevard | :D |
| 15:32.10 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 15:32.13 | leifmadsen | russellb: +1 |
| 15:32.35 | BMJ | danke |
| 15:32.37 | leifmadsen | russellb_: +1 |
| 15:32.47 | leifmadsen | russellb_: I'm all over it like a fat kid on a smarty! |
| 15:32.57 | leifmadsen | oh wait... you probably need to know what Smarties are... |
| 15:33.14 | leifmadsen | http://en.wikipedia.org/wiki/Smarties |
| 15:34.07 | p3nguin | or http://en.wikipedia.org/wiki/Smarties_%28Ce_De_Candy%29 |
| 15:34.21 | leifmadsen | BMJ: hi! |
| 15:34.24 | ManxPower | p3nguin, most any change would be better than the current situation |
| 15:34.56 | [TK]D-Fender | Smarties : For people too dumb to want to learn even ONE letter of the alphabet. |
| 15:35.12 | [TK]D-Fender | PWNED |
| 15:35.19 | p3nguin | manxpower: You mean you want to support that stuff here? |
| 15:35.41 | [TK]D-Fender | (Tw if you're ingenious enough to rotate them 180 deg) |
| 15:35.43 | [TK]D-Fender | Two |
| 15:35.47 | *** join/#asterisk b0gatyr (~b0gatyr@host-208-88-126-198.biznesshosting.net) |
| 15:35.48 | *** join/#asterisk mmlj4 (~jkelly@ip70-171-94-246.no.no.cox.net) |
| 15:35.50 | hrhrhr | i don't get it |
| 15:35.56 | hrhrhr | however, i do love smarties |
| 15:35.57 | p3nguin | m/w |
| 15:36.22 | ManxPower | p3nguin, Not at all. But the current situation is that this Asterisk channel is already being used to support crap. Officially doing so would push the people that hate gui crap off the channel and Digium can support it themselves. |
| 15:36.24 | p3nguin | hrhrhr: UK Smarties are like M&Ms without the m on them. |
| 15:36.59 | NiugeS | thanks for the help and sorry for effending people |
| 15:37.09 | leifmadsen | I don't think anyone was suggesting that we support GUI based projects on this channel, as there are other channels to handle that. The suggestion was to stop being so douchey about it |
| 15:37.12 | ManxPower | FreePBX/Trixbox on this channel is why I'm not here very much anymore. |
| 15:37.16 | russellb_ | leifmadsen: +1 |
| 15:37.21 | hrhrhr | ah yes |
| 15:37.26 | hrhrhr | i am from ze uk |
| 15:37.31 | hrhrhr | i didn't make the association with m&ms tho |
| 15:37.45 | *** mode/#asterisk [+o thehar] by ChanServ |
| 15:37.58 | fauxalliance | leifmadsen, some people only understand 'douche' |
| 15:38.04 | ManxPower | leifmadsen, most the the "assholeness" here seems to be caused by Trixbox |
| 15:38.06 | fauxalliance | leifmadsen: +1 |
| 15:38.38 | fauxalliance | ManxPower, and too much hand-holding / spoonfeeding and not enough googleing and RTFM'ing |
| 15:38.57 | fauxalliance | ManxPower: +1 |
| 15:38.59 | sevard | why would we support trixbox here? there's a #trixbox for that. |
| 15:39.09 | fauxalliance | that empty channel |
| 15:39.19 | hrhrhr | all jokes aside tho |
| 15:39.20 | ManxPower | sevard, because Trixbox people here and refuse to leave? Because they thing Trixbox is Asterisk. |
| 15:39.23 | hrhrhr | smarties are delicious. |
| 15:39.49 | sevard | ManxPower: so just +q anyone who asks questions about trixbox and polietly send them a message directing them to #trixbox |
| 15:39.53 | sevard | i don't see the big issue. |
| 15:39.54 | Qwell | speaking of trixbox.. Andrew still didn't say a word to me this week. |
| 15:40.23 | fauxalliance | sevard, they will come marching in like ants as soon as one posts that support for trixbox is _here_ |
| 15:41.03 | ManxPower | sevard, non-ops can +q someone? |
| 15:41.04 | Katty | hugs Qwell |
| 15:41.10 | Qwell | dances with Katty |
| 15:41.12 | sevard | fauxalliance: I still don't see why this can't be settled with moderation |
| 15:41.16 | Katty | how're you dear |
| 15:41.31 | sevard | ManxPower: trusted users should be granted access on a bot that allows them to +q for a timed duration |
| 15:41.37 | fauxalliance | sevard, it can, but you need one badass volunteer to moderate, is that you? |
| 15:41.42 | sevard | like you, you and [TK]D-Fender have been here for like 6 years |
| 15:41.52 | sevard | fauxalliance: no man, i'm not here enough for that. |
| 15:41.57 | Katty | don't forget me |
| 15:42.04 | sevard | i've already forgotten about you. |
| 15:42.22 | Katty | :< |
| 15:42.24 | WIMPy | It might be a good idea to put information about whats supported here and were other stuff is supported in the topic. Or send a notice to ppl joining. |
| 15:42.28 | hrhrhr | i think the legend of [TK]D-Fender is enough |
| 15:42.34 | Katty | rude. |
| 15:42.38 | ManxPower | sevard, the battle is already lost. |
| 15:42.47 | sevard | the power is yours! |
| 15:42.51 | [TK]D-Fender | I intervene only when no other op is around and for very limited offenses |
| 15:43.09 | Katty | i'll offend you in a minute. |
| 15:43.11 | ManxPower | sevard, you may have noticed I'm not here much anymore. That is because of all the Trixbox stuff here. |
| 15:43.14 | fauxalliance | [TK]D-Fender, misdemeanours even? |
| 15:43.57 | [TK]D-Fender | fauxalliance: massive pointless flaming, flooding, excessive trolling primarily |
| 15:44.00 | ManxPower | More so the situation where someone is getting help for 20 mins before they "remember" to mention they have a GUI. |
| 15:44.01 | NiugeS | Again sorry for causing unrest about the Trixbox answer. But there is llimited activity in that channel and thought someone would help here... again sorry |
| 15:44.14 | russellb_ | NiugeS: don't be sorry. :-) |
| 15:44.27 | Katty | from memory, the only time i see fender kick anyone is for a flood |
| 15:44.37 | ManxPower | NiugeS, Be sorry that you are using software that is not well supported. |
| 15:44.38 | Katty | and occasionally in good fun. |
| 15:44.44 | sevard | NiugeS: be more sorry. |
| 15:44.53 | [TK]D-Fender | Katty: the latter I think was no more than twice ever. |
| 15:44.54 | Katty | sevard: perk up buttercup, no one likes a downer. |
| 15:44.56 | hrhrhr | i didn't mean your @privs |
| 15:45.10 | coppice | Katty: a biblical one? |
| 15:45.11 | hrhrhr | just slapping down noobs in the channel is enough to deter most imo |
| 15:45.12 | sevard | Katty: i haven't had my daily dosage of nudity yet. |
| 15:45.18 | Katty | coppice: baroo? |
| 15:45.19 | ManxPower | I have never disagreed with any of [TK]D-Fender's bans. |
| 15:45.27 | [TK]D-Fender | ManxPower: All 2-3 of them :) |
| 15:45.29 | thehar | dances |
| 15:46.11 | [TK]D-Fender | ManxPower: They were complete miscreants and a detriment to the entire channel in every way. |
| 15:46.14 | NiugeS | ManxPower I had it installed as I was not able to get the time to learn to do it myself.. i was not aware that it was such a poor product in comparison as I would have certainly decided otherwise... |
| 15:46.14 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 15:46.33 | fauxalliance | NiugeS, not too late (to read the book) |
| 15:46.37 | fauxalliance | ~book |
| 15:46.37 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 15:46.37 | ManxPower | Ugh! Why can't people try to hack PBXs while I'm watching? Why does it always have to be at 3am |
| 15:46.44 | hrhrhr | NiugeS: AsteriskNOW is pretty good too |
| 15:47.06 | sevard | from my experience -- it's not a poor product.. but if you want to do something advanced or more enterprise you're better off without it. |
| 15:47.13 | fauxalliance | again, not supported here though NiugeS |
| 15:47.24 | ManxPower | sevard, it could be the best product in the world, but if you can't get support.... |
| 15:47.25 | titter | ManxPower: I have a box that is hit every few hours by some pretty new IP address. I just decided to fail2ban and move on. |
| 15:47.28 | sevard | it's like going into a css/html/js channel and asking for support with frontpage |
| 15:47.31 | Qwell | sevard: trixbox? yes it is. |
| 15:47.46 | ManxPower | titter, we have more than 100 boxes |
| 15:47.48 | [TK]D-Fender | Trixbox is loaded with bloat, and running forked versions of the 2 key components. Makes niether mainline channel want to help out too much. |
| 15:47.51 | SaiSoma | anyone familiar with CONNECTEDLINE? trying to get it to work with polycom phones: sample extension here http://pastebin.com/4JXvEDgX |
| 15:48.01 | fauxalliance | perhaps a trigger bot could PM a user when they start dropping distros and note that perhaps they might want to talk about that in the respective channel. |
| 15:48.05 | ManxPower | SaiSoma, what verison of Asterisk? |
| 15:48.09 | SaiSoma | 1.8.0 |
| 15:48.18 | [TK]D-Fender | And most of theings people ask for are GUI related anyway and who wants to remember what little merge-in file they use for everything there? |
| 15:48.35 | ManxPower | SaiSoma, did you read the UPGRADE*.txt files. I thought connectedline was merged into the callerid stuff in 1.8 |
| 15:48.59 | [TK]D-Fender | SaiSoma: that the official implementation of CPID? |
| 15:49.01 | titter | ManxPower: I am not that big yet, only 30 between two companies. Most of the others are hit at random, but one in particular gets nailed often. |
| 15:49.22 | ManxPower | titter, not over 100 phones, over 100 Asterisk PBXs |
| 15:49.24 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 15:49.31 | SaiSoma | ManxPower: rgr. read the asterisk.pdf after building 1.8.0, will double check UPGRADE. Wasn't using this before 1.8 and was told it was new, so thought the pdf was correct. |
| 15:49.39 | russellb_ | check sip.conf ... you have to enable it "sendrpid" is the option i think |
| 15:49.45 | Qwell | ~upgrade.txt |
| 15:49.46 | titter | ManxPower: I understand ... I have 30 PBX's running |
| 15:49.47 | SaiSoma | [TK]D-Fender: i believe so. |
| 15:49.56 | Qwell | ho hum! |
| 15:50.00 | Qwell | russellb_: we need to wikiize. |
| 15:50.02 | Katty | hums at Qwell |
| 15:50.19 | russellb_ | Qwell: i have no idea what you just said to me |
| 15:50.19 | p3nguin | Why would two companies need to have 30 PBXs? |
| 15:50.28 | SaiSoma | russellb: is enabled globally in sip.conf already |
| 15:50.29 | sevard | titter: out of curiosity do you load balance? |
| 15:50.30 | titter | ManxPower: That also doesn't include the integration with Shoretel that is done |
| 15:50.32 | fauxalliance | p3nguin, one for each person? |
| 15:50.33 | Qwell | russellb_: the bot. some stuffs needs to be linked to the wiki now. much easier. :D |
| 15:50.39 | russellb_ | Qwell: oh yes, totally |
| 15:50.46 | Qwell | like upgrade.txt (assuming it's even on the wiki...which I totally doubt) |
| 15:50.55 | russellb_ | it's not ..... yet |
| 15:51.08 | p3nguin | fauxalliance: Maybe everyone has Asterisk installed on his own workstation? |
| 15:51.09 | fauxalliance | Oh dear, if they won't google Qwell, you think they will "click here for further information"? |
| 15:51.12 | putnopvut | It was awesome putting in the first UPGRADE.txt entry for 1.10 |
| 15:51.24 | putnopvut | No need to search for the appropriate spot to put the change. |
| 15:51.29 | titter | sevard: Not quite, most of these boxes are actually use for IAX trunks. I have three primary PBX's that are setup to failover and utilize Level 3 trunking |
| 15:51.36 | fauxalliance | p3nguin, great, everyone has their own IVR. |
| 15:51.37 | Qwell | putnopvut: you only *think* it was the first entry. |
| 15:51.43 | Qwell | there are totally already like 10 other things in there |
| 15:51.51 | putnopvut | 10 invisible things |
| 15:51.53 | Katty | lunch time! |
| 15:51.55 | Katty | poofs |
| 15:52.23 | titter | p3nguin: read comment above |
| 15:53.32 | p3nguin | I still don't get it. |
| 15:55.10 | ManxPower | Did that bug in chan_iax2 that makes any unauth'd call match the last entry iax2.conf and allow the call ever get fixed? |
| 15:55.13 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 15:55.20 | p3nguin | Those two companies are surely not putting hundreds of thousands of users onto the phone system, so I don't see a reason for 30 individual PBXs. |
| 15:55.43 | titter | p3nguin: I am using small nettops in remote office locations to setup IAX trunks to our primary PBX's. We have two offices in most states |
| 15:56.08 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 15:57.13 | russellb_ | looks like the wiki is back up |
| 15:57.43 | tuxx- | w00t, wiki |
| 15:57.46 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 15:58.02 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 15:58.05 | russellb_ | sevard: ^^^ |
| 15:58.21 | *** join/#asterisk disposable (disposable@blackhole.sk) |
| 16:00.21 | disposable | does anybody have a simple iptables rule that would drop all register packets where sip peername is a 4 digit number? my sipservers are getting hammered by a botnet and fail2ban is putting quite a load on the machines. |
| 16:00.50 | titter | p3nguin: So I guess I should say I have Asterisk running in 30 locations, with 3 primary PBX's being utilized for best routing/latency. |
| 16:01.31 | p3nguin | titter: That does better put it into perspective. |
| 16:01.45 | lauris | hi |
| 16:02.02 | ManxPower | disposable, do you have allowguest=no and alwaysauthreject=yes set? |
| 16:02.39 | lauris | may I ask Asterisk program code related questions here ? |
| 16:02.52 | russellb_ | lauris: #asterisk-dev will get better answers probably |
| 16:02.57 | p3nguin | It might be better suited in #asterisk-dev |
| 16:03.03 | lauris | thanks |
| 16:05.50 | disposable | ManxPower: i don't want asterisk to handle these registrations. i want my firewall to do this. (i can't disable allowguest and i do have alwaysauthreject=yes, but this is of no use against voippack attack) |
| 16:06.43 | disposable | ManxPower: most of our servers are on asterisk 1.2.7, where alwaysauthreject doesn't even work. |
| 16:06.52 | p3nguin | disposable: How about string matching in iptables? |
| 16:07.00 | fauxalliance | shudders.... |
| 16:07.17 | disposable | p3nguin: my funny attempts at doing that failed, which is why i'm asking here |
| 16:15.53 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 16:16.47 | SaiSoma | ManxPower: tried checking the upgrade files for the CONNECTEDLINE stuff. I'm pretty sure this is correct . .though name-pres is also a CALLERID thing too |
| 16:17.19 | SaiSoma | ManxPower: more info, if anyone wants to take a shot: http://pastebin.com/0eBaUQT6 |
| 16:19.44 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 16:27.26 | *** join/#asterisk dlirit (~lirant@80.74.100.10) |
| 16:30.14 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 16:32.59 | *** join/#asterisk mandragor (~ergudicsu@70.158.116.62) |
| 16:41.03 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 16:43.30 | SaiSoma | did some sip debugs and it looks like the rpid info may not be being transmitted back to the phone |
| 16:43.47 | russellb_ | SaiSoma: did you enable it in sip.conf? |
| 16:43.51 | SaiSoma | yes |
| 16:44.05 | SaiSoma | both globally and in the individual phones |
| 16:44.47 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 16:45.54 | russellb_ | SaiSoma: take the 'i' option off of the last CONNECTEDLINE() call |
| 16:46.08 | SaiSoma | russellb: rgr. trying that now |
| 16:48.01 | SaiSoma | russellb: verified the ,i was gone via console while dialing (after dialplan reload) and result is the same |
| 16:49.04 | *** join/#asterisk MiserySoft (~Adium@92.40.46.238.sub.mbb.three.co.uk) |
| 16:49.07 | SaiSoma | if you care to see the debug, here is it: http://pastebin.com/yUSuRnuj |
| 16:55.11 | *** part/#asterisk MiserySoft (~Adium@92.40.46.238.sub.mbb.three.co.uk) |
| 16:56.49 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 16:57.56 | *** join/#asterisk ups (~ups@74-93-178-213-Sacramento.hfc.comcastbusiness.net) |
| 16:58.54 | [TK]D-Fender | SaiSoma: http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE <-- from this is would look like the "i" option is precisely what you DON'T want to be doing... |
| 17:00.38 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 17:02.53 | SaiSoma | [TK]D-Fender: That's what I thought . ..and I've tried both ways, but doing so again. |
| 17:04.53 | *** join/#asterisk jsgoecke (~Adium@c-98-207-5-162.hsd1.ca.comcast.net) |
| 17:05.18 | SaiSoma | [TK]D-Fender: nope. same result with the i or without. Tried leaving the I and/or r off of the dial cmd to the same result. |
| 17:05.41 | [TK]D-Fender | SaiSoma: show your configs |
| 17:05.49 | [TK]D-Fender | SaiSoma: And the new call attempt |
| 17:06.06 | SaiSoma | [TK]D-Fender: full configs or just relevant parts? I don't have a problem with full configs. |
| 17:06.13 | ups | Hello guys. I have an 8 port FXO card, when I dial out I get a second dialtone instead of just dialing through, does anybody knows what might be a problem ? |
| 17:06.27 | [TK]D-Fender | SaiSoma: It doesn't work. Show everything taht could possibly be relevant |
| 17:06.46 | [TK]D-Fender | ups: Maybe you should show us your failed call <- |
| 17:06.47 | [TK]D-Fender | ~pb |
| 17:06.47 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:06.49 | [TK]D-Fender | ^^^^^^ |
| 17:06.55 | SaiSoma | [TK]D-Fender: rgr. as much as possible. one moment |
| 17:09.32 | ups | here is a link to a failed call http://pastebin.com/fnJ3asgU |
| 17:09.55 | ups | DAHDI just answers |
| 17:10.08 | [TK]D-Fender | ups: -- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/6000-0000000e", "DAHDI/1/") in new stack <-- you are not telling DAHDI a number to dial in there |
| 17:10.35 | [TK]D-Fender | ups: DAHDI/GROUPorCHANNEL/NUMBERTODIAL |
| 17:10.41 | [TK]D-Fender | ups: you have no number there |
| 17:10.50 | Katty | dials fender up |
| 17:10.51 | ups | wierd |
| 17:10.57 | ups | thanks alot ! |
| 17:11.30 | Katty | still no squirrlies :< |
| 17:12.33 | [TK]D-Fender | Critter(less)Cam |
| 17:13.10 | Katty | sobs |
| 17:13.52 | Kobaz | Katty: astricon missed you |
| 17:14.12 | Katty | oh? were people talking about me behind my back? |
| 17:14.14 | Katty | hugs Kobaz |
| 17:14.39 | SaiSoma | [TK]D-Fender: OK. Here ya go. I hope this is enough, if not, I'll get whatever else you recommend: http://pastebin.com/DM8jaG7A |
| 17:15.14 | Kobaz | Katty: no... but... you should have come :) |
| 17:15.55 | Katty | i bet it was fun |
| 17:16.04 | Kobaz | it was quite good |
| 17:16.51 | [TK]D-Fender | SaiSoma: You didn't put the "trust" in each peer, you only left that global, and undo your use of templates for testing |
| 17:17.11 | SaiSoma | rgr. testing |
| 17:17.27 | *** join/#asterisk clintc (~clintc@n128-227-82-65.xlate.ufl.edu) |
| 17:17.30 | [TK]D-Fender | SaiSoma: Infact you said TRUST=NO . BAD |
| 17:17.54 | [TK]D-Fender | SaiSoma: trustrpid = no ; If Remote-Party-ID should be trusted <------- WTF |
| 17:17.56 | SaiSoma | [TK]D-Fender: had it at yes in previous testing. will set to yes both globally and in the phone |
| 17:18.30 | SaiSoma | [TK]D-Fender: somewhere online I found someone who said they had to change theirs to no for this to function. originally was yes. |
| 17:21.53 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 17:22.29 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 17:26.05 | Katty | http://pastebin.ca/1978857 <- anyone know how to deal with this? |
| 17:26.37 | SaiSoma | [TK]D-Fender: Same results. Full info here: http://pastebin.com/BXZWkUYU |
| 17:26.59 | Kobaz | kandjar: looks like you're restoring into a db that's been already populated |
| 17:27.09 | Katty | Kobaz: yes, kind of. |
| 17:27.14 | Katty | Kobaz: it's call logs. |
| 17:27.27 | Katty | Kobaz: i was hoping to get the previous call logs back in tho |
| 17:27.31 | Kobaz | you cant restore a dump of a db into a db that already has all the stuff already in it |
| 17:27.39 | Kobaz | just export the data, not the schema |
| 17:28.18 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 17:28.22 | Katty | would you mind having a look at my backup script? |
| 17:28.30 | Katty | i'm not sure how to do export just the data |
| 17:28.49 | Kobaz | pg_dumpall --schema-only |
| 17:28.54 | Kobaz | er i mean |
| 17:29.00 | Kobaz | i'm still half asleep... |
| 17:29.12 | Kobaz | --data-only |
| 17:29.18 | paulc | If I use the A option in Dial() to play an announcement to the callee, and M to run a macro, do they run in that order? (ie M doesn't fire till after A) Or can I control order by how I construct my command? (trying hard not to have to delve in the code...) |
| 17:29.52 | Katty | /usr/bin/pg_dump -b -i --data-only --username=postgres --format=c --file=$backup_dir/`date +'%Y%m%d'`/psql/$db_name$ |
| 17:29.55 | Katty | like that? |
| 17:29.58 | Kobaz | Katty: i suppose you should dump the schema into one file... and the data into another |
| 17:30.04 | Kobaz | Katty: looks okay |
| 17:30.05 | Katty | well. |
| 17:30.13 | Katty | i'm going to create a new db for the call logs |
| 17:30.19 | Katty | and if i simply wanted to import the data.... |
| 17:30.29 | Katty | there's no reason to need the scheme info, right? |
| 17:30.49 | Katty | i would just need a full pg_dump if i wanted to createdb |
| 17:31.05 | Katty | is that correct? |
| 17:31.14 | Kobaz | wlel, you'll need the schema if you're starting from scratch, you need the structure of the db in order to import the data |
| 17:31.37 | Katty | let me ask you this |
| 17:31.43 | Katty | if on current db i exported jsut the data |
| 17:31.44 | Kobaz | it really depends how you want to import data |
| 17:31.46 | Katty | then deleted the database. |
| 17:31.56 | Katty | then pg_restored the old one, and createddb at the same time |
| 17:32.02 | Katty | then pg_restored just the data of the current one |
| 17:32.04 | Katty | do you think that'd work? |
| 17:32.22 | Kobaz | well, one... you need the schema (structure) in order to import data |
| 17:32.46 | Kobaz | so if you have only a backup of the data, delete the db, and go to reimport... you wont have any tables to import into |
| 17:32.58 | Katty | but the old db |
| 17:33.01 | Katty | should have the scheme info |
| 17:33.15 | Katty | so in theory, it would rebuild the scheme |
| 17:33.15 | Kobaz | yeah if it hasn't been deleted |
| 17:33.24 | Katty | no..it's whining about scheme bits |
| 17:33.36 | Katty | now comes the million dollar question. how do i delete a database |
| 17:33.37 | [TK]D-Fender | SaiSoma: Out of ideas at this point |
| 17:33.41 | Kobaz | well it looks like you're importing into a working db |
| 17:33.48 | Kobaz | like, those tables already exist |
| 17:33.55 | SaiSoma | [TK]D-Fender: rgr thanks. i may call digium for support on this one |
| 17:33.57 | Katty | exactly, but it's going to be deleted |
| 17:33.58 | [TK]D-Fender | paulc: If you use M, don't use A |
| 17:34.08 | Kobaz | so delete them (make sure you have backups) |
| 17:34.10 | Katty | and the old one will be pg_restored, with scheme info |
| 17:34.27 | Katty | and current data just imported back in |
| 17:34.31 | Katty | no scheme |
| 17:34.40 | Kobaz | you need a schema in order to import data |
| 17:34.42 | Katty | but idk how to delete a db |
| 17:34.47 | Kobaz | dropdb <db> |
| 17:34.48 | Katty | yes, but the old backup should have that |
| 17:34.58 | Katty | due to the fact that it's whining about the scheme already being there when i try to pg_restore |
| 17:35.02 | Katty | ok. let's try it! |
| 17:35.08 | Kobaz | okay. if your backup has everything you need, you can delete the db |
| 17:35.18 | Kobaz | it will be gone forever... |
| 17:35.40 | Katty | i'll do two backups |
| 17:35.44 | Katty | one with scheme and one data only. |
| 17:35.46 | Katty | just in case. |
| 17:36.32 | Kobaz | and for being ultra paranoid... stop postgres... cp -rap /var/lib/postgresql /var/lib/postgresql-backup |
| 17:36.43 | Kobaz | (or whereever your pg data is stored) |
| 17:37.08 | paulc | [TK]D-Fender: Reading the docs now, the macro runs on the CALLED party's channel.. can I use Playback in there to accomplish the same as A? |
| 17:37.23 | Katty | Kobaz: i don't actually know where it's stored |
| 17:37.31 | Kobaz | probably there |
| 17:37.44 | Katty | there is /var/lib/postgresql/8.3/main |
| 17:37.49 | Kobaz | yeap |
| 17:37.50 | Katty | with some stuffs. |
| 17:37.53 | Kobaz | that's it |
| 17:38.00 | Katty | what's the -rap mean? |
| 17:38.02 | [TK]D-Fender | paulc: Yes |
| 17:38.07 | Kobaz | Katty: preserve permissions and etc |
| 17:38.12 | Katty | oh okay |
| 17:38.32 | Katty | i know r is recursive, but that's it |
| 17:39.05 | drudge` | poitns and laughs |
| 17:39.17 | Katty | ok |
| 17:39.19 | paulc | [TK]D-Fender: Awesome, thanks. Last question then.. with the inheritance thing, if I have a variable declared in the dialplan, that I want to pass to the called thing, I could either do it via M(macro^arg1^arg2) etc, or use _ (or __?) prefixing on my var name, right? Thoughts either way? |
| 17:39.24 | Katty | here goes nothing >.< |
| 17:39.26 | Katty | quivers |
| 17:39.56 | Katty | IT"S ALL GONE |
| 17:40.49 | Kobaz | gonzed |
| 17:40.55 | Katty | pg_restore --create -verbose -U postgres asterisk <- does that look right? |
| 17:40.56 | [TK]D-Fender | paulc: passing=explicit=superior |
| 17:41.05 | Katty | i don't think i need dbname because that refers to an already existing db, right? |
| 17:41.25 | Katty | i'm in the directory my asterisk dump file is in |
| 17:41.39 | Kobaz | i actually haven't used pg_restore... i probably should |
| 17:41.53 | Kobaz | i use psql <db> < /path/to/backup.sql |
| 17:41.57 | paulc | [TK]D-Fender: *nods* yeah, I like the explicitness/separation. Something the day job peeps struggle with sometimes. Ok, all good - thanks for the help/advice. I'm gonna make this stuff fly :-) |
| 17:42.13 | Katty | it's seting it up |
| 17:42.22 | Katty | does it little dance |
| 17:42.23 | *** join/#asterisk martha (~martha@212.179.135.226) |
| 17:42.25 | Katty | gets down tonight |
| 17:42.26 | [TK]D-Fender | paulc: Excellent... |
| 17:42.30 | Katty | okay. |
| 17:42.36 | Katty | and it's --data-only? |
| 17:43.23 | Katty | pg_restore --data-only --dbname=asterisk --verbose -U postgres psqlbackupdataonly |
| 17:43.50 | Kobaz | looks good i suppose... i'm not really familiar with pg_restore |
| 17:44.11 | Katty | it says database asterisk does not exist |
| 17:44.16 | *** join/#asterisk Weezey (~ohno@lan5.LO.iasl.com) |
| 17:44.18 | Kobaz | createdb asterisk |
| 17:44.38 | Katty | but shouldn't my previous import have created it? |
| 17:44.48 | Weezey | is there a new botnet attacking from a bunch of random IPs? |
| 17:44.51 | Kobaz | i dont think the schema export does a create database |
| 17:44.58 | Katty | hrmm k |
| 17:44.58 | Kobaz | did the schema import work? |
| 17:45.03 | Katty | i don't know how to tell |
| 17:45.12 | Katty | i but i can run --schema-only |
| 17:45.17 | Kobaz | do you have pgadmin or something like that, or navicat |
| 17:45.31 | Kobaz | db explorers are useful |
| 17:46.06 | martha | how do I get a call-waiting tone to be heard by the caller? |
| 17:46.29 | martha | call-waiting works for me, and the person being called hears beeps |
| 17:46.37 | Katty | ok. |
| 17:46.40 | Katty | here's another thought |
| 17:46.43 | Katty | what if i delete the database |
| 17:46.50 | martha | but the person calling hears only the normal ring |
| 17:46.52 | Katty | and then recreate it, like i do at the very beginning |
| 17:46.56 | Katty | and import the data only from both backups |
| 17:47.09 | martha | this happens for both sip and dahdi phones. |
| 17:47.35 | Katty | i'll try that. if it asplodes still, i'll put the folder back the way it was |
| 17:47.47 | martha | can this be done in asterisk? |
| 17:48.35 | Kobaz | you should be able to delete the db, create the db, import the schema, and then import the data |
| 17:48.51 | [TK]D-Fender | martha: check if the callee is on the phone and if so change the ringing tone <- |
| 17:49.08 | WIMPy | martha: Only manually via some dialplan logic. |
| 17:49.35 | WIMPy | martha: But what would the ringing souind like then? And what country does that apply to? |
| 17:50.43 | martha | this would mostly be for internal |
| 17:51.12 | martha | the ceo is annoyed that he can't tell that someone is on the phone and not just ignoring him :-/ |
| 17:51.26 | paulc | martha: why do you want the caller to know the callee is getting call waiting treatment though? if they then hang up "oh I don't want to distract if they're on a call" it's too late, the callee is already interrupted.. |
| 17:51.27 | WIMPy | I've never come across a special ring back tone for waiting calls. |
| 17:51.33 | paulc | LOL ah - that answers the question |
| 17:51.47 | paulc | give the CEO a BLF so he can see the lights for all people who are busy ;-) |
| 17:51.55 | Katty | ok that looks promising |
| 17:52.03 | paulc | or.. get him to ask himself why people would ignore him in the first place hehe |
| 17:53.06 | martha | paulc: that kind of behavior is actually common here in israel. when you call someone (who has call waiting), you get a different beep, and if it is not urgent you usually hang up and try later |
| 17:53.06 | Katty | Kobaz: i didn't get any errors |
| 17:53.12 | Katty | Kobaz: do i need to restart postgres? |
| 17:53.20 | [TK]D-Fender | martha: and I already gave you the answer |
| 17:53.46 | martha | [TK]D-Fender: how can I detect this in my dialplan? |
| 17:54.01 | sevard | write 1358008 on your calculator. |
| 17:54.05 | sevard | no wait! |
| 17:54.11 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 17:54.15 | [TK]D-Fender | martha: "core show function DEVICE_STATE" |
| 17:54.26 | Katty | bah, it's saying no data found |
| 17:54.53 | paulc | martha: ah cool.. I know in the UK we used to play an announcement if call waiting was happening.. "The number you have called is busy. The person you are calling knows you are waiting." over and over.. |
| 17:55.01 | martha | <PROTECTED> |
| 17:55.13 | [TK]D-Fender | martha: "or whatever" |
| 17:55.34 | martha | [TK]D-Fender: thanks |
| 17:55.36 | Kobaz | Katty: if you're using the import/export tools everything is live |
| 17:55.37 | WIMPy | paulc: Yes, I've come across that one as well. "Your call is a waiting call." over and over. |
| 17:55.51 | Kobaz | Katty: if you want to replace the data directory, always stop postgres any time you mess with the data directory |
| 18:03.42 | Katty | Kobaz: so let's say i dropdb asterisk |
| 18:03.47 | Katty | Kobaz: then recreate it, resetup my tables |
| 18:03.51 | Katty | Kobaz: shiny new database. |
| 18:03.58 | Katty | Kobaz: if --data-only |
| 18:04.03 | Katty | would that work? |
| 18:04.26 | Katty | before i cdr=enable |
| 18:04.46 | Kobaz | yeah... schema, tables... done |
| 18:05.02 | Katty | what exactly is scema, i don't get it. |
| 18:05.14 | Katty | is that the actual table info? |
| 18:05.49 | *** join/#asterisk PotatoHead (~chatzilla@tcom.newpaltz.edu) |
| 18:06.59 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:06.59 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:08.45 | *** join/#asterisk sawgood (~sawgood@173-13-158-26-sfba.hfc.comcastbusiness.net) |
| 18:08.52 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 18:09.01 | PotatoHead | Hey all, I'm running through setting up an Asterisk PBX in my office so I can check out the call processor and Voice Mail functions, but I've run into a snag. I can't seem to get the Asterisk voicemail to recognize DTMF signals from the IP phones. If I configure the phone itself for RFC2833 I can hear the tone on other sets, but I can't hear it when I call the voicemail. If I configure it... |
| 18:09.03 | PotatoHead | ...for SIP INFO, I don't get the aduible tone and the voicemail doesn't recognize it. |
| 18:09.58 | PotatoHead | I did a quick search through the forums and only found one topic that mentioned a similar problem, however their solution (changing to SIP INFO) didn't fix my issue. |
| 18:10.40 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:11.52 | PotatoHead | I'm having the same issue with a soft phone client, and the two hardware IP phones I'm using are both Aastra 4422's. |
| 18:12.30 | sawgood | Why not use RFC2833 like everyone else? |
| 18:12.46 | PotatoHead | If I configure the phone itself for RFC2833 I can hear the tone on other sets, but I can't hear it when I call the voicemail. |
| 18:13.28 | sawgood | What about the Asterisk server (what is it set for)? |
| 18:13.47 | PotatoHead | In the configuration for the extensions? |
| 18:13.59 | sawgood | Are these calls coming in / leaving out over a SIP trunk by chance? |
| 18:14.56 | PotatoHead | I don't have any trunks defined yet, only stations so far. |
| 18:16.39 | sawgood | everything is an 'internal' extension to extension (SIP phones)? |
| 18:17.45 | PotatoHead | correct. I'll eventually tie the system into the MD110 we have on campus via an E1 as we can't do SIP trunking to our phone system, but for now it's just extensions built into the Asterisk server and a few IP phones sitting on my desk for testing. |
| 18:18.28 | sawgood | You could do SIP trunking ... in fact an Audio Codes gateway would be ideal for your E1 to SIP ... |
| 18:19.03 | sawgood | Well, DTMF payload might not be a concern, but I would check both 96 and/or then 101 types to make sure |
| 18:19.12 | sawgood | 96 or 101 is almost always the fix for me |
| 18:19.25 | sawgood | but that is for outgoing/incoming SIP trunk calls ... |
| 18:20.02 | *** join/#asterisk knot (~knotsucke@unaffiliated/devemo) |
| 18:20.48 | PotatoHead | I've also found that if I change the codec from ulaw or alaw to another codec, I can't even connect to the voice mail system. This is making me think is is listening for the in-band tones and since it's being connected to with a compressed codec, it just disconnects me (and makes a cdr record of the 0 second call). |
| 18:20.58 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-123-250.lsanca.fios.verizon.net) |
| 18:21.40 | sawgood | You are using standard G711 (PCMU) ulaw, right? |
| 18:22.01 | PotatoHead | I believe so, let me look at the phone configuration..... |
| 18:22.37 | PotatoHead | It looks like the phone's first choice is 711 ALAW, and secondary is 711 ULAW. |
| 18:22.43 | PotatoHead | I can flip those and check.... |
| 18:24.19 | *** join/#asterisk dmast (~dmast@exchange.newpointe.org) |
| 18:24.29 | *** part/#asterisk martha (~martha@212.179.135.226) |
| 18:25.21 | PotatoHead | I switched them in the phone configuration and it didn't create the tones when attached to the voicemail system. |
| 18:30.38 | *** join/#asterisk Dovid (Dovid@213.8.121.90) |
| 18:33.37 | PotatoHead | Would the "Relax DTMF" setting have anything to do with it? |
| 18:35.15 | sawgood | hmmm ... I never heard of that .. I have an Aastra phone next to me ... let me look at mine |
| 18:36.26 | PotatoHead | That was a setting on the Asterisk software I saw using the GUI under SIP settings -> TOS -> Enable Relaxed DTMF checkbox. |
| 18:38.08 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 18:38.37 | ManxPower | GUI? I give up |
| 18:38.40 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 18:39.29 | sawgood | GUI of the Aastra based SIP phone |
| 18:40.12 | PotatoHead | No, on the Asterisk box. Since I'm very new to the system, it's quicker for me to use the GUI then the CLI since I don't know the commands very well yet. |
| 18:42.40 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 18:45.16 | *** join/#asterisk mandragor (~ergudicsu@70.158.116.62) |
| 18:46.07 | mandragor | How can I create a local channel extension and make a different channel the target for pickup? |
| 18:52.21 | Kobaz | Katty: oh hey.. i'm back |
| 18:52.37 | Kobaz | Katty: schema is the table structure, sequences, views, etc |
| 18:52.50 | *** join/#asterisk nsgn (~nsgn@cpe-24-27-48-215.austin.res.rr.com) |
| 18:55.28 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 18:55.35 | raden_work | Naikrovek, YOOO BRO ! |
| 18:55.39 | nsgn | alright..i'm stumped and surprised. come back from the weekend (during which there were no storms or power failures) and none of my phones onsite can take incoming calls. asterisk 1.6 is the server and the phones are polycom IP 330. |
| 18:55.47 | nsgn | but i can place outbound calls perfectly on the polycoms. they are registered |
| 18:55.57 | raden_work | nsgn, restart asterisk |
| 18:56.02 | nsgn | but they wont ring for external calls nor can they call each other via internal extensions |
| 18:56.12 | nsgn | raden_work, did, and also all the phones. issue persists |
| 18:56.15 | Katty | Kobaz: okay |
| 18:56.17 | *** join/#asterisk to0w1r3d_ (~to0w1r3d_@c-24-129-114-47.hsd1.fl.comcast.net) |
| 18:56.35 | raden_work | nsgn, restart router |
| 18:56.41 | raden_work | hugs Katty |
| 18:56.49 | nsgn | raden_work, already ahead of ya there too. also restarted the whole computer running the * box and even restarted the bank of switches |
| 18:58.32 | Katty | hugs raden_work |
| 18:58.48 | Katty | Kobaz: how do i reload asterisk so it stops dumping stuff into my database |
| 18:59.07 | Kobaz | Katty: depends what's being dumped |
| 18:59.14 | Katty | cdr info |
| 18:59.20 | PotatoHead | Ok, I have no idea what I did....but I just got the voicemail to recognize the tones. I hate it when things start working and I don' t know why. lol |
| 18:59.20 | Kobaz | module unload the cdr module |
| 18:59.56 | Katty | module unload.... cdr? |
| 19:00.53 | Kobaz | it's probably the postgres cdr module |
| 19:00.57 | Kobaz | right? |
| 19:01.04 | Katty | just stops asterisk |
| 19:01.19 | Kobaz | module unload cdr_pgsql.so |
| 19:01.22 | Kobaz | heh |
| 19:01.41 | nsgn | wow..is it possible my asterisk box has become compromised? this is just looking weird. there appear to be my extensions registering to public IP addresses..but all our phones are in house |
| 19:01.56 | nsgn | but what confuses me is that i'm not forwarding any ports through the firewall (pfsense) to the * box |
| 19:01.57 | Kobaz | nsgn: sounds bad |
| 19:02.15 | nsgn | except usually when they do that they immediately begin placing expensive international calls. that is not happening in this case |
| 19:03.09 | *** join/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 19:04.01 | nny | In this mailing list entry http://www.mail-archive.com/asterisk-users@lists.digium.com/msg140682.html the response was "The solution was to add the command "ulimit -n 65536" to the initscript, prior to starting Asterisk.". is this in the asterisk init script, and if so, any particular spot that won't break anything else? |
| 19:05.25 | *** join/#asterisk capitan (~A@ppp-71-140-67-195.dsl.irvnca.pacbell.net) |
| 19:07.40 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 19:08.04 | Katty | well darn it |
| 19:09.34 | nsgn | well dang, there was a port forwarding through there |
| 19:09.53 | nsgn | getting hasty when troubbleshooting an external trunk and adding some firewall exceptions isnt a good idea :) |
| 19:10.30 | Katty | i dropped the db. recreated the db, recreated the table, then pg_restored data only. |
| 19:10.33 | nsgn | funny thing is the only time i've made that mistake in the past it was taken advantage of in a matter of hours. this has been that way for months with zero logged intrusions or negative effects on operation |
| 19:10.33 | Katty | but it's not showing up in my query |
| 19:11.48 | Kobaz | what's your query |
| 19:12.40 | nsgn | oh mothers, they did make outbound calls. god dangit |
| 19:14.59 | beardy | hugs Katty |
| 19:15.03 | nsgn | so let me ask this: i've seen this happen once before where illegit calls are made in a compromised system..but why the heck are most of the calls less than 1 minute long? what good does this do them to talk for such a short time...or are they automated? in which case what is the point |
| 19:15.34 | leifmadsen | sometimes the point is just to see if they can place the calls |
| 19:15.44 | beardy | nsgn: Spamming/scanning? |
| 19:16.07 | nsgn | spamming i guess. scanning, though? what is that for |
| 19:16.30 | beardy | To see what sits on numbers. |
| 19:16.40 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 19:16.49 | Katty | still nothing. |
| 19:16.50 | Katty | bummer. |
| 19:17.21 | beardy | For further abuse, or curiosity, or some other reason. |
| 19:17.35 | Katty | hugs on beardy |
| 19:18.49 | nsgn | beardy, lovely. i got off easy, which is why i didnt notice it happening. it looks like the damage was all of like..$5 over a few weeks since i dropped the bad port opening on the firewall into my config. i've seen some outside companies get hundreds of dollars in bills for this crud |
| 19:19.17 | *** part/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 19:19.20 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 19:19.20 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:19.40 | nsgn | its still an embarrasing/sad coincidence though, because i actually had international calling disabled entirely on my trunk until just a short time ago when someone asked me to call them international that i knew. left it on after that. two stupid decisions i guess |
| 19:19.50 | nsgn | all of the illegit calls are international |
| 19:20.03 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 19:20.13 | nsgn | thanks for the input. gotta clear my state table to get the firewall closed out so i'm getting out |
| 19:25.26 | kjs | grrr keep getting all circuits are busy now |
| 19:25.54 | *** join/#asterisk mes (~mes@sentry.lazo.ca) |
| 19:26.57 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 19:27.16 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 19:29.40 | mandragor | is bridging two channels available in asterisk 1.2? |
| 19:30.13 | Qwell | Like...a phone call? |
| 19:31.17 | mandragor | yea |
| 19:31.24 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 19:31.46 | Qwell | yeah, asterisk can totally make phone calls |
| 19:31.56 | [TK]D-Fender | ... |
| 19:32.12 | titter | lol |
| 19:34.21 | beardy | Quote material. |
| 19:34.25 | Katty | that's so cute |
| 19:34.27 | beardy | goes for tea |
| 19:34.31 | Katty | tags along |
| 19:35.16 | thehar | frolics about |
| 19:36.36 | russellb | chases thehar |
| 19:36.47 | thehar | eeeeeeeep |
| 19:36.49 | thehar | runs |
| 19:37.06 | kjs | this elastix distro is a pile of **** |
| 19:37.20 | [TK]D-Fender | SHIT |
| 19:37.23 | [TK]D-Fender | kjs: You can say it |
| 19:37.29 | Katty | or! |
| 19:37.31 | [TK]D-Fender | kjs: It's ok. We AGREE :p |
| 19:37.48 | Katty | you could replace shit with snickerdoodles |
| 19:37.57 | Qwell | gross |
| 19:38.02 | Katty | also an acceptable alternative. |
| 19:38.12 | Katty | Qwell: snickerdoodles are not gross. |
| 19:38.12 | Qwell | also, that would completely change the sentence. |
| 19:38.13 | [TK]D-Fender | Katty: Pre or post? |
| 19:38.17 | Qwell | ^ |
| 19:38.19 | kjs | fucking waste of my time... I thought it might save me some time... all it has done is fail me. |
| 19:38.49 | [TK]D-Fender | kjs: Well I don't see anything validating your specific "issues" and where it is at fault |
| 19:38.53 | Katty | gotta turn that frown upside down somehow cupcake! |
| 19:38.56 | [TK]D-Fender | kjs: in all fairness |
| 19:39.02 | Qwell | Katty: with cupcakes! great idea. |
| 19:39.54 | kjs | [TK]D-Fender: all the extension work, inbound and outbound routes all fail, then my internal extension randomly stop working with no output on the console. |
| 19:39.56 | Katty | there's also the nap route. |
| 19:40.13 | Katty | Qwell: did i tell you i might have another doggie soon |
| 19:40.22 | Qwell | you did not |
| 19:40.26 | [TK]D-Fender | kjs: Feel free to show us a meaningful attempt at providing backup |
| 19:40.47 | [TK]D-Fender | Katty: Oddly enough you aren't becoming "crazy cat lady". |
| 19:41.04 | Katty | Qwell: http://www.breedersassociation.net/endjan/lookout-farm-pembroke-welsh-corgi-356.jpg |
| 19:41.10 | [TK]D-Fender | Katty: but the parallel trend is becoming telling... |
| 19:41.21 | Katty | [TK]D-Fender: i wasn't talking to you. mkay. |
| 19:41.32 | Katty | gives [TK]D-Fender a cupcake |
| 19:41.35 | Kobaz | aww how cute |
| 19:41.37 | Katty | eat this and be quiet. kthx. |
| 19:41.40 | [TK]D-Fender | Katty: Not required :) |
| 19:41.40 | Kobaz | that dog looks awesome |
| 19:41.57 | telnettech | all.....trying to setup an IVR to check the blacklist....does this look correct in the dialplan...this is for 1.8.0 ver http://pastebin.com/fqDG7QCa |
| 19:42.37 | Katty | Kobaz: i will live in a zoo |
| 19:42.49 | Katty | Kobaz: technically it won't be my dog. |
| 19:42.59 | Katty | Kobaz: technically it'd be the boyfriend's dog |
| 19:47.25 | kjs | [TK]D-Fender: not really sure what to do with this thing, if i edit any config files chances are it will just over write them again |
| 19:47.37 | Katty | i know what to do with it. |
| 19:47.56 | Katty | but you won't like my suggestion, probably |
| 19:47.59 | [TK]D-Fender | kjs: Forget about "editing" I don't see anything qualifying as "thorough debugging". |
| 19:49.04 | [TK]D-Fender | telnettech: NO, for starters --> | |
| 19:50.13 | Katty | i'll pipe you in a minute |
| 19:50.19 | Katty | not really. |
| 19:50.29 | Katty | maybe. |
| 19:51.11 | [TK]D-Fender | That's what SHE said..... |
| 19:51.59 | telnettech | is that supposed to be a comma? |
| 19:52.12 | p3nguin | Are you trying to play two sound files? |
| 19:52.12 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.66) |
| 19:52.23 | [TK]D-Fender | telnettech: Score one... next, have you even TRIED it? |
| 19:52.32 | *** join/#asterisk to0w1r3d_ (~to0w1r3d_@c-24-129-114-47.hsd1.fl.comcast.net) |
| 19:52.33 | p3nguin | Playback, as far as I know, is not conditional. |
| 19:52.37 | telnettech | cant try it until i get home tonight |
| 19:52.46 | Katty | hi p3nguin |
| 19:53.01 | p3nguin | Hello. |
| 19:54.48 | telnettech | this is something i found when i googled blacklist.....havent setup blacklist before and the attempt I made this past weekend failed miserably!!!!! |
| 19:54.57 | p3nguin | So when did we replace the & with a comma to play multiple sound files? |
| 19:55.19 | Katty | i think ihave a blacklist post |
| 19:55.20 | [TK]D-Fender | p3nguin: Who said anything about "&"? |
| 19:55.22 | Katty | do you want the wiki page |
| 19:55.32 | p3nguin | I obviously said something about it. |
| 19:56.01 | p3nguin | My question is: why would Playback() use commas? |
| 19:56.10 | telnettech | sure Katty |
| 19:56.12 | Katty | telnettech: http://42ndgeekstreet.blogspot.com/2009/03/blacklisting-numbers.html <- maybe helpful, maybe not |
| 19:56.12 | [TK]D-Fender | p3nguin: It hasn't been AFAIK. Where do you see otherwise? |
| 19:56.46 | mandragor | the documentation says the bridge command is available in 1.6, is there an alternative in 1.2? |
| 19:56.59 | telnettech | p3nguin: it normally doesnt but I have never done multiple playbacks on 1 line before either.....that what I was thinking was the issue really |
| 19:57.01 | p3nguin | Well, it doesn't matter... I wasn't on the same page. |
| 19:57.10 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 19:57.12 | citywok | mandragor: meetme? |
| 19:57.20 | p3nguin | He's using the 'noanswer' option. I thought noanswer was a second sound file. |
| 19:58.09 | mandragor | citywok, thanks |
| 19:58.32 | p3nguin | If noanswer was a second sound file, it would have been played using Playback(privacy-you-are-blacklisted&noanswer). |
| 19:58.37 | citywok | next question, why are you using 1.2? lol |
| 20:00.01 | mandragor | fonality |
| 20:00.12 | p3nguin | exten => blocked,1,Playback(silence/1&privacy-you-are-blacklisted&vm-goodbye) |
| 20:00.14 | Katty | bummer. |
| 20:00.31 | mandragor | we might switch in the future but we don't have the time to put into doing it |
| 20:00.31 | p3nguin | That's where I send blacklisted calls. |
| 20:00.53 | p3nguin | via GotoIf($[${BLACKLIST()}]?misc,blocked,1) |
| 20:01.28 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 20:01.32 | [sr] | howdy |
| 20:01.55 | Katty | ohai |
| 20:02.12 | beek | Katty: http://www.youtube.com/watch?v=UcT4paZfflg&feature=related |
| 20:03.05 | Katty | i like how they send the interns to do it |
| 20:04.10 | beek | Yep. If the cats are interested in the pumpkins, there's always the interns! |
| 20:04.33 | Katty | looks like they've having a blast tho |
| 20:04.36 | beek | It was cool though watching those big cats play like the domestic cats do. |
| 20:05.40 | Katty | yeah so much for catnip filled soft toys tho |
| 20:05.45 | Katty | they're ripping those pumpkins to shreds. |
| 20:06.24 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 20:07.46 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 20:09.48 | *** join/#asterisk Bartok (~chatzilla@c-24-62-161-95.hsd1.nh.comcast.net) |
| 20:09.54 | Bartok | Hey hey |
| 20:10.16 | Bartok | I have a question - and this issue is making me crazy |
| 20:10.46 | Bartok | I have two SIP trunks that I use - when I enable the second one, it buggers everything up |
| 20:10.53 | Qwell | mandragor: well, 1.2 is EOL in 20 days. Time to switch now. |
| 20:10.59 | Bartok | I wind up with an error: |
| 20:11.03 | Qwell | (actually, the time to switch was about 4 years ago) |
| 20:11.58 | telnettech | I will try that 2 p3nguin......thanks for yall's help |
| 20:12.04 | Bartok | [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have <6839>, digest has <3169> |
| 20:12.37 | p3nguin | telnettech: I don't really know what you were trying to achieve nor what the issue was. I came in late. |
| 20:12.38 | *** join/#asterisk ukine_droid (~ukine@64.134.182.140) |
| 20:12.38 | Bartok | Can someone point me to the proper 'fishing net' for this one? driving me bughouse :( |
| 20:13.14 | p3nguin | bartok: I missed the paste of your sip.conf. Care to provide the link again? |
| 20:13.30 | Bartok | [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have <6839>, digest has <3169> |
| 20:13.55 | p3nguin | I saw that, and that's not your sip.conf. |
| 20:14.06 | telnettech | p3nguin: when a call comes in, I wanted to check the number against the blacklist and if on there play a message and if not, continue on to the AA on the IVR dialplan...nothing crazy |
| 20:14.43 | Bartok | This is a log entry |
| 20:16.12 | p3nguin | telnettech: I prefer the GotoIf($[${BLACKLIST()}]?context,extension,priority) method using the BLACKLIST function. |
| 20:16.34 | p3nguin | bartok: Keep in mind, some of us do not have all day. |
| 20:16.55 | Bartok | I apologize - what would you like to see?? |
| 20:17.02 | p3nguin | sip.conf |
| 20:17.15 | Bartok | okay - sure |
| 20:17.34 | Bartok | let me get a pastebin for you |
| 20:18.38 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.wlan.noare-1.holmedal.net) |
| 20:22.16 | SaiSoma | [TK]D-Fender: update on Remote-Party-ID. It seems that the header is in the sip debug as being sent to my phones and I've manually added a P-Asserted-Identity header that I can see as well, but neither is showing on the phone, so perhaps it's a polycom issue |
| 20:22.53 | *** part/#asterisk BMJ (~bjohns@cpe-098-026-116-043.nc.res.rr.com) |
| 20:23.11 | [TK]D-Fender | SaiSoma: Dunno.. would need to look closer... and times up for here... BBIAB |
| 20:29.51 | Bartok | â â â â â |
| 20:31.00 | russellb | SaiSoma: you can also have Asterisk send the info in a P-Asserted-Identity header, instead ... maybe the phone supports that instead |
| 20:31.34 | SaiSoma | russellb: tried that. not sure of the formatting. used: P-Asserted-Identity: <sip:Anthony Hardy 1531> |
| 20:31.52 | SaiSoma | based on what little i could find on the inets |
| 20:33.32 | *** join/#asterisk [cannibalera] (~cannibale@201-25-255-94.fnsce703.dsl.brasiltelecom.net.br) |
| 20:34.15 | titter | SaiSoma: What model and SIP firmware are the Polycom's running? |
| 20:35.30 | SaiSoma | titter: polycom 560: bootrom 4.3.0.0246 on my phone |
| 20:36.46 | SaiSoma | dl'ing 3.2.3 update now, hoping that there is a fix there |
| 20:36.50 | titter | SaiSoma: According the SIP admin guide the phones support since 2.0.x, the 560's probably are using 3.x |
| 20:37.03 | SaiSoma | yea, it's 3.0 |
| 20:39.24 | SaiSoma | actually, 3.2.1 . .upgrading to 3.2.3 bootrom got me confused for a moment (and the four other conversations I'm having at the same time as well) |
| 20:44.10 | *** join/#asterisk [canniballllera] (~cannibale@201-25-255-94.fnsce703.dsl.brasiltelecom.net.br) |
| 20:50.14 | SaiSoma | different topic. is it possible now to force blind transfers on all transfers in 1.8.0? Or perhaps I missed something in some earlier release notes. I'm guessing not, but was curious. |
| 20:50.24 | ManxPower | SaiSoma, what are you trying to accomplish? |
| 20:50.52 | *** join/#asterisk chuckz (~lechuck@93-40-66-17.ip37.fastwebnet.it) |
| 20:51.30 | SaiSoma | ManxPower: I have polycom 330's and 560's. on the 560's i can program the soft button to always do a blind transfer, but apparently not so on the 330. I would like all of my transfers to be blind. Transfer button -> extension -> Transfer button and come through blind. |
| 20:52.50 | *** part/#asterisk [cannibalera] (~cannibale@201-25-255-94.fnsce703.dsl.brasiltelecom.net.br) |
| 20:53.39 | ManxPower | SaiSoma, no, with P-Asserted-Identity |
| 20:54.46 | SaiSoma | ManxPower: ah, sorry. when i dial an extension, i want the local display to show the name associated with the number. i can do the queries to a db for local extensions if necessary and use the CONNECTEDLINE function, if i were to work. |
| 20:55.01 | *** join/#asterisk bio-tty (~c@23.0.34.95.customer.cdi.no) |
| 20:55.10 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 20:58.02 | SaiSoma | ManxPower: in case you didn't catch it earlier when I talked to fender, here is a good summary of config info and a sip debug: http://pastebin.com/BXZWkUYU |
| 20:59.19 | SaiSoma | you can see it remote-party-id header is there (line 681), but the polycom 560 isn't changing the display. I added the P-Asserted header manually in the dialplan like so: P-Asserted-Identity: <sip:Anthony Hardy 1531> |
| 20:59.42 | SaiSoma | to no avail. i'm guess a polycom issue perhaps? I've opened a case with them. |
| 21:00.17 | titter | How did you add it to the dialplan? |
| 21:01.05 | SaiSoma | titter: sorry, earlier pastebin: http://pastebin.com/DM8jaG7A |
| 21:01.35 | SaiSoma | fender had me change trustrpid = yes. i originally had it set to yes, but was playing with values at that time. |
| 21:01.56 | titter | I don't see where you added the p-asserted |
| 21:01.56 | SaiSoma | rpid_update = yes caused issues in my system (transfers stopped working), so it's back to no |
| 21:02.49 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 21:03.03 | SaiSoma | ah . .i should just re-do a pastebin with current (long day, my apologies) |
| 21:03.41 | SaiSoma | titter: http://pastebin.com/e9MYMwHw |
| 21:05.19 | SaiSoma | titter: values would be incorrect, of course, but trying to get the local phone display to change on dialing and then i can work out what is needed |
| 21:06.16 | *** part/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 21:08.20 | titter | SaiSoma: Reading up since I have a lot of experience with the Polycom's, you can set blind transfer as the default mode on the 330's |
| 21:08.49 | titter | SaiSoma: call.transfer.blindPreferred would be set to 1 |
| 21:09.05 | SaiSoma | titter: really? I never found that:( wow. checking config now |
| 21:10.15 | SaiSoma | ah. . i have that set now to 1, yet it does not do a blind transfer by default |
| 21:10.46 | titter | SaiSoma: If set to 1, the blind transfer is the default mode. The Normal soft key is available to switch to a consultative transfer. |
| 21:11.07 | titter | SaiSoma: Have you confirmed the phones are downloading the cfgs correctly? |
| 21:11.13 | SaiSoma | yes |
| 21:12.43 | titter | SaiSoma: Just to check, that is only available on the 330 |
| 21:12.49 | *** join/#asterisk adam0563 (44b2768e@gateway/web/freenode/ip.68.178.118.142) |
| 21:12.52 | titter | only on the* |
| 21:13.16 | SaiSoma | *nod* i'm wondering if it is a firmware issue of some sort. having one of our 330's upgraded to 3.2.3 right now |
| 21:13.26 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 21:13.29 | adam0563 | hey guys, can somebody remind me what the acronymn is for the function that allows you to set the Caller ID number on outgoing calls for telephone companies? |
| 21:14.02 | titter | SaiSoma: Possible, I am running 3.2.3 on my 330s and 550s ... but don't have a 330 near me, only have that one at home to test. |
| 21:14.31 | SaiSoma | titter: thanks. i'll let you know in just a moment. it's restarting and upgrading now |
| 21:15.06 | *** join/#asterisk fskrotzki (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 21:15.34 | titter | SaiSoma: No problem, as silly as it may seem I also always run wireshark and make sure the phones are getting the cfg files correctly through a capture. I do this when I make configuration changes, Polycom's can be a pita sometimes |
| 21:16.09 | citywok | gar segfault. ews seg's on both servers i have, one being clean & new. lol |
| 21:19.20 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 21:22.40 | adnc | <PROTECTED> |
| 21:22.40 | adnc | <PROTECTED> |
| 21:23.48 | Qwell | because it's registering all the time |
| 21:24.46 | adnc | mhh |
| 21:25.06 | adnc | there is a refresh switch i set it to 60 but still it prints it all the time |
| 21:25.07 | citywok | is it de-registering as well, or losing it's network connectivity such that it thinks it needs to re-register constantly? |
| 21:25.20 | adnc | <PROTECTED> |
| 21:25.26 | adnc | <PROTECTED> |
| 21:25.34 | adnc | all the time these |
| 21:25.37 | adnc | only these alternating |
| 21:42.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:42.35 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 21:50.38 | *** join/#asterisk clintc (~clintc@n128-227-82-65.xlate.ufl.edu) |
| 21:50.39 | adnc | i don't need chan_unistim doing noload => chan_unistim.so does not help, what could i do? |
| 21:50.54 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 21:51.20 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 21:51.55 | russellb | it doesn't help? |
| 21:52.00 | *** part/#asterisk clintc (~clintc@n128-227-82-65.xlate.ufl.edu) |
| 21:52.10 | adnc | russellb, yes it still loads it |
| 21:57.59 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 21:58.46 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:59.24 | *** join/#asterisk wide (~wide@69.31.47.219) |
| 22:02.10 | adnc | i don't need ldap support can i prevent this error: ERROR[7154]: res_config_ldap.c:1616 parse_config: No directory URL or host found. |
| 22:06.37 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 22:07.06 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 22:09.10 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 22:11.20 | p3nguin | If you move or delete the unistim.conf, I bet that would stop it from loading. |
| 22:12.53 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 22:14.48 | pabelanger | adelas: noload => res_config_ldap.so |
| 22:15.52 | JunK-Y | pabelanger: hey! |
| 22:16.18 | pabelanger | sup man, how was your trip back? |
| 22:16.33 | JunK-Y | lot of turbulence! |
| 22:16.42 | adnc | yes, thank you |
| 22:16.47 | adnc | pabelanger, thats t |
| 22:17.06 | adnc | what was the reason for changing username to defaultuser? |
| 22:17.08 | pabelanger | JunK-Y: Ya, your the 2nd person to say that. |
| 22:17.19 | adnc | can there be more than one user |
| 22:17.21 | JunK-Y | who's the first one? |
| 22:17.56 | pabelanger | JunK-Y: Can't remember, maybe Stephan M. I think he had a bumpy flight down |
| 22:18.55 | JunK-Y | kk |
| 22:19.39 | pabelanger | anthm: read UPGRADE-1.6.txt, it explains the reason |
| 22:23.14 | adnc | i still can not disable dahdi and get this error: [Nov 1 23:22:43] ERROR[7810] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory |
| 22:23.25 | adnc | has someone got any ideas? |
| 22:23.31 | Qwell | adnc: make menuselect |
| 22:23.39 | Qwell | Disable stuff you don't need. No point noloading a bunch of stuff. |
| 22:24.01 | adnc | Qwell, i'm not compiling it myself |
| 22:25.13 | Qwell | then noload modules you don't need, or disable autoloading |
| 22:26.02 | adnc | i disabled all i found in the modules list with dahdi |
| 22:26.06 | adnc | [Nov 1 23:25:38] ERROR[7892] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data |
| 22:26.15 | adnc | but this i don't know what it could mean |
| 22:31.52 | adnc | Qwell, do you know what "Could not initialize cluster membership service: Try Again" means |
| 22:35.45 | Qwell | It means exactly what it says. |
| 22:35.57 | adnc | ohh |
| 22:36.01 | adnc | really |
| 22:36.23 | *** join/#asterisk cesar_CR (~cesar@celord.ice.co.cr) |
| 22:36.35 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 22:39.38 | *** part/#asterisk ups (~ups@74-93-178-213-Sacramento.hfc.comcastbusiness.net) |
| 23:06.10 | *** join/#asterisk saint_ (~saint42@c-24-0-99-39.hsd1.nj.comcast.net) |
| 23:06.12 | *** part/#asterisk saint_ (~saint42@c-24-0-99-39.hsd1.nj.comcast.net) |
| 23:46.26 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 23:46.26 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.13 (2010/09/15), 1.4.36 (2010/09/15), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.4 (2010/09/01) -=- Visit the new official Asterisk wiki: wiki.asterisk.org |
| 23:48.43 | *** join/#asterisk zerohalo (~zerohalo@cambridge.zerohalo.com) |
| 23:50.16 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 23:57.52 | *** join/#asterisk engrxyz (~erwerewre@host81-150-217-170.in-addr.btopenworld.com) |
