| 00:01.00 | *** join/#asterisk zaltar (~zaltar@c-67-163-24-181.hsd1.il.comcast.net) |
| 00:12.54 | a1fa | grep -c "Wrong password" /var/log/asterisk/messages |
| 00:12.54 | a1fa | 365445 |
| 00:12.59 | a1fa | :) |
| 00:19.03 | ChannelZ | I have no idea what you're trying to do |
| 00:21.57 | a1fa | alwaysauthreject=yes |
| 00:22.16 | a1fa | scumbags can still guess extensions |
| 00:22.26 | a1fa | or sip accounts |
| 00:39.39 | niekie | The internet is great. |
| 00:39.58 | niekie | I Google "Free Russian DID" and I get free Russian wife. |
| 00:40.01 | niekie | :| |
| 00:41.27 | a1fa | maybe you did his wife? |
| 00:41.53 | niekie | Heh. |
| 00:50.23 | a1fa | i am finally getting some time to fix my dialplan |
| 00:50.36 | a1fa | and i finally cracked the broadvoice alternative number problem |
| 00:51.07 | a1fa | http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice+distinctive+ring+support |
| 00:53.42 | a1fa | is there anyway to change the server header information? |
| 00:54.01 | a1fa | and hide realm for 401 requests |
| 00:55.10 | a1fa | useragent = "" in sip.conf? |
| 00:57.48 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 00:58.01 | a1fa | god.. asterisk should come default with all security features enabled |
| 00:58.40 | a1fa | <PROTECTED> |
| 00:58.49 | IsUp | blocksocialengineeringattacks=yes |
| 00:59.43 | a1fa | haha |
| 00:59.47 | a1fa | no.. not that one |
| 00:59.56 | a1fa | that one is enabled by layer8 routing |
| 01:13.05 | a1fa | god, this is the first time i think timing is messing me up.. can barely understand voicemialmain from all jittering |
| 01:21.22 | *** join/#asterisk tekati (~robert@cpe-75-85-246-201.bak.res.rr.com) |
| 01:25.09 | tekati | Looking into Faxing. What would be the best way to implement faxing with Asterisk 1.8.2.2? I would like to do faxing to email and be able to have clients who call into asterisk be able to auto send them a fax etc. |
| 01:53.37 | *** part/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 02:07.03 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 02:15.05 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 02:24.21 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 02:31.48 | capitan | is there a big overhead to being registered to too many trunks? for example, is there a big difference between A) being registered to one trunk with 99 concurrent calls or B) registering to ~33 trunks and distributing the same concurrent calls to ~3/trunk ? please forgive any poor semantics... |
| 02:37.49 | drmessano | capitan, if you're using IAX2 there's some potential bandwidth savings with call concurrency using the same peer. SIP is entirely different, and it doesn't matter if it's the same peer or a 99 different peers. |
| 02:49.03 | capitan | drmessano, thanks! :D |
| 02:50.20 | capitan | my case is SIP, and it'll be more managable for me with the lotsa-peers approach, so that's very good to know... |
| 02:54.30 | *** join/#asterisk coppice (~chatzilla@202.64.44.139) |
| 04:08.53 | *** join/#asterisk zachsis (gatsby@ns2.yogsothoth.net) |
| 04:16.34 | zachsis | holy moly. any asterisk ninjas familliar with this error ? |
| 04:16.36 | zachsis | TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 66 |
| 04:29.54 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 04:39.15 | *** join/#asterisk radic (~radic@dslb-178-002-225-106.pools.arcor-ip.net) |
| 04:43.53 | zachsis | anyone active? |
| 04:50.35 | *** join/#asterisk Bryanstein (~bryan@shellium/admin/bryanstein) |
| 04:51.26 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 04:57.39 | *** join/#asterisk carrar (~tim@2604:5000:11:1::3) |
| 05:01.05 | carrar | moof |
| 05:06.18 | *** join/#asterisk titter (~Justin@c-98-208-153-116.hsd1.fl.comcast.net) |
| 05:20.12 | *** join/#asterisk sourcode (~code@ppp-58-8-43-253.revip2.asianet.co.th) |
| 05:49.40 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 05:55.21 | *** join/#asterisk nighty-- (~nighty@x122091.ppp.asahi-net.or.jp) |
| 06:00.21 | *** join/#asterisk brut- (~brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 06:07.09 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 06:27.43 | *** part/#asterisk zaltar (~zaltar@c-67-163-24-181.hsd1.il.comcast.net) |
| 06:41.42 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 07:00.00 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 07:13.38 | *** join/#asterisk OnTheRock (~IceChat77@btc.r1.northcourt.transact.bm) |
| 07:28.58 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 08:08.16 | *** join/#asterisk reber (~reber@ns1.sylogix.net) |
| 08:15.17 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 08:19.50 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 08:22.19 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 08:23.13 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 08:34.43 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 09:02.16 | *** join/#asterisk dlirit (~lirant@80.74.100.10) |
| 09:06.21 | *** join/#asterisk Babak__ (Babak.y@85.133.129.100) |
| 09:06.35 | Babak__ | hi is anyone using 79xx cisco ipphones? |
| 09:12.42 | fenrus | ayea. |
| 09:12.49 | fenrus | several 7940 |
| 09:13.41 | fenrus | if you state your real question you might get an answer. |
| 09:17.18 | Babak__ | fenrus: I need to show unicode characters on the display but it shows squares. it seems I need the locale files. but I do not find them |
| 09:17.53 | fenrus | the locale file for the phone ? |
| 09:18.00 | Babak__ | yep |
| 09:18.17 | fenrus | ill check the cisco website and see if i can find anything |
| 09:18.24 | Babak__ | thanx |
| 09:24.17 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 09:25.54 | fenrus | cant seem to find that much, |
| 09:26.00 | fenrus | might be worth a shot with https://supportforums.cisco.com/ |
| 09:28.05 | fenrus | seems like it should work with CCM 5.0 on 7940G/7960G |
| 09:35.55 | *** join/#asterisk calmh (~jb@acro.nym.se) |
| 09:37.32 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 09:39.48 | Babak__ | fenrus: I have no ccm. |
| 09:42.31 | fenrus | me neither :/ |
| 09:43.52 | Babak__ | I can find xml config files and all other things but this |
| 09:57.23 | *** join/#asterisk Tekati (~robert@cpe-75-85-246-201.bak.res.rr.com) |
| 09:58.17 | fenrus | have any filename to search for ? |
| 10:00.55 | ChannelZ | autoexec.bat |
| 10:08.11 | Babak__ | fenrus: I need to someone to have ccm. cause it seems cisco locale installer works on ccm servers only |
| 10:10.39 | Babak__ | for example the phone is requesting english_united_states\ipc-sip.jar for cisco ip communicator |
| 10:10.45 | Babak__ | on tftp |
| 10:16.52 | *** join/#asterisk bbear (~bbear@217.0.98-84.rev.gaoland.net) |
| 10:16.57 | bbear | hello. |
| 10:21.41 | *** join/#asterisk Tim_Toady (~moi@178.128.134.192.dsl.dyn.forthnet.gr) |
| 10:24.58 | bbear | I would like to have telephony through internet. |
| 10:25.22 | bbear | Is it possible to have a PCI card which I can connect with my standard RJ11 phone ? |
| 10:25.30 | fenrus | yes. |
| 10:25.33 | fenrus | or a ata box. |
| 10:25.37 | fenrus | or skype |
| 10:25.37 | bbear | (a cheap one preferably, not a fancy device) |
| 10:25.44 | bbear | what ata box ? |
| 10:26.16 | fenrus | http://lmgtfy.com/?q=ata+box |
| 10:26.19 | *** join/#asterisk linuxplatform (~fedora@0aaf68fb64f0da1a7380e4a5c41b2351.cust.nwn.no) |
| 10:27.01 | bbear | what would you advice ? |
| 10:27.09 | bbear | I would like to have standard phone |
| 10:27.28 | bbear | Actually i wonder if you know SFR (french ISP). |
| 10:28.05 | bbear | I would like to use their phone service with a personal equipment (not this one provided as standard for the majority of people) |
| 10:28.22 | bbear | I am going to have a server. |
| 10:28.24 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 10:28.30 | bbear | What is the cheapest, ata box or pci card ? |
| 10:28.34 | bbear | Is it so different ? |
| 10:29.23 | fenrus | if you have an ata box you need no server |
| 10:29.41 | bbear | Yes but I have a server. |
| 10:29.47 | bbear | with free pci slots. |
| 10:29.51 | fenrus | if you get a sip provider |
| 10:30.09 | fenrus | my guess is that a pci card is more expensive than an ata box |
| 10:30.29 | bbear | I already have one, but it does not support incoming calls. I wonder if either pci card or atabox will work with incoming calls. |
| 10:30.42 | bbear | (talking about the SIP account) |
| 10:31.51 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 10:31.58 | fenrus | your sip provider does not support incoming calls? |
| 10:32.09 | fenrus | or your ata box ?, what ? |
| 10:33.20 | bbear | Actually if I directly a sip software phone (ekiga or twinkle) , I cannot have incoming calls. I can just make outgoing calls, and it's restricted to certain numbers, that becomes true when I use the ADSL box provided by the ISP. |
| 10:33.26 | bbear | becomes not true*. |
| 10:33.52 | bbear | with the box I have incoming calls and not restricted calls to special numbers. |
| 10:34.40 | bbear | I wonder if I can do the same with a pci card/ata box, or if the result will be the same than with the SIP softphone. |
| 10:49.18 | *** join/#asterisk [netman] (~netman@133.Red-83-55-243.dynamicIP.rima-tde.net) |
| 10:55.47 | *** join/#asterisk com3 (~com3@78.154.216.68) |
| 10:55.51 | com3 | Hello,, |
| 10:56.36 | bbear | hello |
| 10:56.50 | com3 | Just a quick question, are there any special hardware requirements to build a new Asterisk server? |
| 10:59.10 | com3 | anyone? |
| 11:01.50 | fenrus | that depends. |
| 11:01.59 | fenrus | only for sip signalling - no |
| 11:02.20 | fenrus | if you want to be able to use some kind of T1/E1 cards you need one of those |
| 11:02.26 | fenrus | and make sure it fits in the machine |
| 11:03.49 | *** join/#asterisk linuxplatform (~fedora@165-118-232.connect.netcom.no) |
| 11:06.28 | com3 | great thanks,, |
| 11:11.49 | *** join/#asterisk bbear (~bbear@139.27.73-86.rev.gaoland.net) |
| 11:15.31 | com3 | fenrus, so to be able to use a landline and voip together i need a T1/E1/J1 card? |
| 11:16.16 | fenrus | for a traditional landline yes, |
| 11:16.27 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 11:16.29 | fenrus | but if you get a sip connection from a sip provider instead you need none |
| 11:16.36 | fenrus | if your phones are sip too. |
| 11:21.34 | com3 | I see, |
| 11:22.51 | com3 | do you happen to know if there are any USB based E1 or T1 cards? i'm trying to run the server on my laptop to start |
| 11:23.37 | coppice | see xorcom |
| 11:32.39 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 11:56.59 | *** join/#asterisk bbear (~bbear@162.27.73-86.rev.gaoland.net) |
| 11:59.25 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 12:04.29 | *** join/#asterisk bbear (~bbear@52.27.73-86.rev.gaoland.net) |
| 12:13.36 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 12:20.01 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 13:17.29 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 13:44.08 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 13:53.48 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 14:19.59 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 14:29.17 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 14:31.11 | *** join/#asterisk _zoom_ (~zero@41.209.121.190) |
| 14:35.36 | *** join/#asterisk sourcode (~code@ppp-58-8-43-253.revip2.asianet.co.th) |
| 14:36.02 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 15:11.15 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 15:29.05 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 15:39.40 | *** join/#asterisk calmh (~jb@acro.nym.se) |
| 15:48.44 | *** join/#asterisk domzinique (~rao@59.164.189.235) |
| 16:03.23 | *** join/#asterisk moy (~moy@70.25.53.39) |
| 16:30.35 | *** join/#asterisk Tim_Toady (~moi@178.128.134.192.dsl.dyn.forthnet.gr) |
| 17:00.36 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:00.36 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:12.13 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:12.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:16.16 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 17:30.16 | *** join/#asterisk y7n (~mrg0xt@188-223-79-144.zone14.bethere.co.uk) |
| 17:32.15 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 17:35.47 | y7n | Hi, I have an Atcom IP01 running Asterisk 1.4. When I call out from a SIP client over PSTN I get some odd blips during the call. Anyone able to advise? I've recorded a sample: www.simplygenius.co.uk/file/misc/Atcom-IP01-PSTN-Sample-Recording.mp3 |
| 17:44.34 | *** join/#asterisk scrooge-mcduck (~scrooge@80-219-24-66.dclient.hispeed.ch) |
| 17:44.37 | scrooge-mcduck | hello |
| 17:44.45 | leifmadsen | yo |
| 17:45.07 | scrooge-mcduck | i am trying to get asterisk working |
| 17:45.15 | scrooge-mcduck | only it fails to connect with voipbuster |
| 17:45.31 | scrooge-mcduck | and i am getting a bit confused regarding the settings to use for voipbuster |
| 17:45.34 | scrooge-mcduck | after googling |
| 17:48.48 | scrooge-mcduck | does anyone know what the correct settings are? |
| 17:52.05 | *** join/#asterisk domzinick (~dominick@59.164.189.235) |
| 17:52.49 | *** join/#asterisk Corydon76-home (eleven@c-69-137-80-31.hsd1.tn.comcast.net) |
| 17:52.49 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 17:56.12 | *** join/#asterisk wdbl (~not@ool-44c0619d.dyn.optonline.net) |
| 17:59.56 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 18:00.48 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 18:03.27 | scrooge-mcduck | binbash_: are you good with asterisk? |
| 18:15.26 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 18:26.50 | scrooge-mcduck | Call from '' to extension 's' rejected because extension not found. |
| 18:26.54 | scrooge-mcduck | what does this mean? |
| 18:28.39 | Tim_Toady | you dont have the s extension in that context |
| 18:29.01 | scrooge-mcduck | is that a special one from asterisk? |
| 18:29.22 | Tim_Toady | http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 18:30.45 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 18:33.03 | scrooge-mcduck | how can i enable the debug logging so it says in which context its looking for s? |
| 18:33.06 | scrooge-mcduck | i tried |
| 18:33.10 | scrooge-mcduck | asterisk -rvvvvv |
| 18:33.18 | scrooge-mcduck | and then set sip debug |
| 18:34.02 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 18:34.32 | Tim_Toady | asterisk -rvvv is enough to see the dialplan code runing |
| 18:35.03 | Tim_Toady | as for the context its the one you have set on ur sip.conf for that peer/user/whatever |
| 18:36.33 | scrooge-mcduck | yeah but still ignores it |
| 18:36.41 | scrooge-mcduck | so i wanna know in which context it is looking for s... |
| 18:36.47 | scrooge-mcduck | should be from-voipbuster |
| 18:39.18 | scrooge-mcduck | grr |
| 18:41.07 | scrooge-mcduck | [Jan 23 18:40:31] WARNING[2253]: channel.c:2599 ast_indicate_data: Unable to handle indication 3 for 'SIP/5060-005568d8' |
| 18:41.11 | scrooge-mcduck | ah |
| 18:42.08 | *** join/#asterisk ofauchon (~ofauchon_@85-171-131-9.rev.numericable.fr) |
| 18:48.54 | pabelanger | ~collectdebug |
| 18:48.55 | infobot | i heard collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 18:48.58 | pabelanger | scrooge-mcduck: ^ |
| 18:49.08 | scrooge-mcduck | thanks |
| 19:03.24 | scrooge-mcduck | i increase the logging |
| 19:03.25 | scrooge-mcduck | stuff |
| 19:03.33 | scrooge-mcduck | but still doesnt tell me which context it |
| 19:03.37 | scrooge-mcduck | expects it to be in |
| 19:05.38 | scrooge-mcduck | hmm |
| 19:05.43 | scrooge-mcduck | i need to do |
| 19:05.46 | scrooge-mcduck | [default] |
| 19:05.51 | scrooge-mcduck | include => from-voipbuster |
| 19:05.54 | scrooge-mcduck | to make it work :/ |
| 19:06.21 | scrooge-mcduck | while the context of the voipbuster peer of friend type is set to from-voipbuster |
| 19:08.03 | scrooge-mcduck | this is my config: http://pastebin.com/ihaRvGQR |
| 19:08.06 | scrooge-mcduck | pabelanger: http://pastebin.com/ihaRvGQR |
| 19:08.34 | scrooge-mcduck | from line 28 is extensions.conf |
| 19:19.32 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 19:26.54 | scrooge-mcduck | sip_write: Asked to transmit frame type 8, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) |
| 19:26.57 | scrooge-mcduck | :/ |
| 19:33.49 | *** join/#asterisk ixti (~ixti@217.70.249.69) |
| 19:35.03 | ixti | Hello everybody. Probably somebody experienced (and solved) same problem as I have: ReceiveFax() works, while SendFax() not - produces lots of warnings about T38FaxMaxDatagram |
| 19:41.58 | pabelanger | scrooge-mcduck: remove the whitespaces in your extensions. EG: exten => s,1,Anser() |
| 19:42.00 | pabelanger | erro |
| 19:42.20 | pabelanger | s/Anser/Answer |
| 19:42.23 | pabelanger | s/Anser/Answer/ |
| 19:42.32 | pabelanger | blah |
| 19:42.45 | pabelanger | exten => s,1,Answer() |
| 19:42.49 | pabelanger | scrooge-mcduck: ^ |
| 19:43.48 | scrooge-mcduck | hm, does the whitespace matter? |
| 19:43.54 | pabelanger | scrooge-mcduck: yes |
| 19:44.28 | *** join/#asterisk Carlos_PHX1_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 19:45.38 | scrooge-mcduck | :/ |
| 19:45.50 | pabelanger | ~book |
| 19:45.50 | 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/ |
| 19:46.06 | pabelanger | scrooge-mcduck: ^ A good resource to learn |
| 19:47.14 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 19:47.52 | scrooge-mcduck | good |
| 19:47.56 | scrooge-mcduck | working now sweet! |
| 19:47.59 | scrooge-mcduck | only sound sucks |
| 19:48.04 | scrooge-mcduck | but thats something for next time |
| 19:49.34 | scrooge-mcduck | already happy its working :) |
| 19:55.24 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 19:56.01 | scrooge-mcduck | ah its usng ulaw |
| 19:56.59 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 20:01.37 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 20:06.42 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 20:24.06 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 20:39.56 | *** join/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 20:39.56 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:44.59 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 20:44.59 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:47.01 | *** join/#asterisk ketas-l (~ketas@ketas-laptop6.si.pri.ee) |
| 20:47.12 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 20:48.06 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 20:49.59 | *** join/#asterisk franciscodavid (~lol@47.Red-83-44-175.dynamicIP.rima-tde.net) |
| 20:52.48 | *** join/#asterisk puzzled (~patrick@535335AA.cm-6-4a.dynamic.ziggo.nl) |
| 20:54.17 | *** part/#asterisk franciscodavid (~lol@47.Red-83-44-175.dynamicIP.rima-tde.net) |
| 20:54.20 | *** join/#asterisk SparFux (~raoul@g224209097.adsl.alicedsl.de) |
| 21:13.08 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 21:13.08 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:13.23 | *** join/#asterisk bjhaid (~IceChat7@41.220.69.12) |
| 21:14.44 | *** part/#asterisk SparFux (~raoul@g224209097.adsl.alicedsl.de) |
| 21:26.00 | *** join/#asterisk bjhaid (~IceChat7@41.220.69.14) |
| 21:54.50 | *** join/#asterisk ixti (~ixti@217.70.249.69) |
| 21:55.51 | *** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 22:15.12 | *** join/#asterisk nix8n82 (~nate@63.162.28.112) |
| 22:16.26 | *** join/#asterisk ixti (~ixti@217.70.249.69) |
| 22:35.00 | *** join/#asterisk ixti (~ixti@217.70.249.69) |
| 22:44.21 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 23:00.27 | ixti | Can somebody tell me where to look at, when SendFax() produce lots of warnings about T38FaxMaxDatagram into console? |
| 23:18.51 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 23:26.59 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 23:42.33 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 23:42.33 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:57.13 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |