| 00:03.02 | doug | seems like asking asterisk to match anything with a literal dot anywhere in it is alittle tricky. i assume it anchors the beginning of the string in all cases. |
| 00:03.54 | p3nguin | I can only imagine that your softphone where you paste phon numbers parses out the hyphens, sending a valid alpha-numeric phone number into asterisk. That's what mine does. |
| 00:04.45 | p3nguin | But, then again, my softphone also pares out the dots if I enter it that way. |
| 00:08.42 | p3nguin | Maybe you need a softphone that handles non-alpha-numeric characters in a better way. |
| 00:09.13 | *** join/#asterisk geneg1 (~gene@bas3-toronto01-1177778644.dsl.bell.ca) |
| 00:09.31 | doug | i wouldn't mind a better softphone. been through a handful already, they all have some issues. |
| 00:09.46 | doug | the 2-3 i've settled on all have this same issue with pasted numbers. |
| 00:09.54 | p3nguin | Twinkle ignores dots and hyphens in the phone number. |
| 00:10.07 | doug | by pasting as well? |
| 00:10.18 | doug | loudhush won't let you get them in via keyboard entry, but allows them via paste. |
| 00:10.23 | doug | passing them on verbatim. |
| 00:10.25 | p3nguin | What's the difference between pasting or typing in the dial box? |
| 00:10.39 | doug | programmatically, there's different possibilities for validation. |
| 00:11.04 | p3nguin | The chars are entered before you hit the dial button, so I don't see how it's different. |
| 00:11.16 | doug | well, maybe it shouldnt be. but it is. |
| 00:11.26 | doug | software has to implement clipboard handling separate from keyboard. |
| 00:11.42 | p3nguin | Software might, but it's not the app that handles it. |
| 00:11.54 | doug | in this case, i'll get a "paste from clipboard" command. the software itself then has to fetch the clipboard entry and get the data from the right place. |
| 00:11.54 | p3nguin | not the softphone handling it. |
| 00:12.16 | doug | huh? yeah, the softphone itself has to handle pastes different from keyboard entries on a low level. |
| 00:12.23 | doug | it's up to the app how to do that. |
| 00:12.54 | p3nguin | Not here. Maybe in bizarre-o world. |
| 00:12.59 | *** part/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 00:13.08 | doug | where's "here?" |
| 00:13.19 | ChannelZ | My pants |
| 00:13.46 | doug | if you use a nice gui framework, like Qt or wxwindows, the library can make some of those things look the same |
| 00:14.15 | doug | like, you can just declare an input field, and the library will handle keyboard entry and clipboard entry |
| 00:14.25 | doug | but not all of those trigger validation in the same way. |
| 00:14.40 | doug | it's pretty common for there to be a validator for single-key input that's bypassed for pastes. |
| 00:15.05 | doug | so if you hook a validator in that returns null for dashes, it won't be used for pastes. so you can paste dashes in anyways. |
| 00:15.07 | p3nguin | If I type hyphens or dots, or if I paste hyphens or dots, it makes no difference. The hyphens or dots are present in the dial box prior to my pressing the dial button, at which time the phone number is parsed and sent to the PBX. |
| 00:15.29 | doug | well, that depends on the app. |
| 00:15.39 | doug | loudhush won't let me type hyphens or dots in the first place. |
| 00:15.46 | doug | so they probably overrode the single-key input validator. |
| 00:15.47 | p3nguin | I'm talking about twinkle. |
| 00:15.51 | doug | but i can paste hypens or dots just fine. |
| 00:16.00 | doug | yeah, well, every app is going to be a little different. |
| 00:16.49 | doug | i personally wouldn't have an app parse out anything, dots and hyphens included, unless there was a clear, compelling, and documented case for it. |
| 00:17.00 | doug | i hate it when apps do too much validation for their own good. |
| 00:17.03 | carrar | doug |
| 00:17.12 | carrar | exten => _X.,1,Set(crack=${FILTER(0123456789|${EXTEN})}) |
| 00:17.12 | carrar | exten => _X.,n,Goto(out-context,${crack},1) |
| 00:17.55 | doug | yeah, filtering everything. |
| 00:17.56 | carrar | could do it in 1 line too |
| 00:18.10 | carrar | well that works |
| 00:18.21 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 00:18.37 | doug | i guess i'll see if that plays well with the rest of my dialplan |
| 00:20.13 | doug | hm, actually, just using _X. instead of _XXXXXX. in my normal path seems to do the trick. |
| 00:20.16 | doug | carrar++ |
| 00:20.32 | carrar | this would be wiser |
| 00:20.33 | carrar | exten => _XXX.XXX.XXXX,1,Set(crack=${FILTER(0123456789|${EXTEN})}) |
| 00:20.33 | carrar | exten => _XXX.XXX.XXXX,n,Goto(out-context,${crack},1) |
| 00:20.41 | doug | i take back everything i was thinking about.. |
| 00:20.57 | p3nguin | I don't think _XXX.XXX.XXXX will work very well. |
| 00:20.57 | carrar | I was just getting tired of you ramblin on :) |
| 00:21.01 | carrar | it will work |
| 00:21.34 | carrar | actaully |
| 00:21.38 | carrar | yeah |
| 00:22.19 | carrar | if the X. works use that |
| 00:22.20 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 00:22.22 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 00:22.46 | carrar | probable is if thats in the same context that you are dialing out from |
| 00:22.53 | carrar | it might loop |
| 00:23.15 | *** join/#asterisk ryan-c (~ryan@c-98-207-154-10.hsd1.ca.comcast.net) |
| 00:23.36 | ryan-c | does anyone happen to know if asterisk groks A/B/C/D DTMF tones? |
| 00:23.54 | p3nguin | As far as I know, it does. |
| 00:23.59 | p3nguin | They are standard DTMF. |
| 00:24.33 | ryan-c | standard but very obscure and rarely used. |
| 00:24.52 | drmessano | Except on 2-way radios, where they're standard |
| 00:26.19 | ryan-c | sure |
| 00:36.55 | neurosys | anyone happen to be familar with cisco 7960 and *? |
| 00:38.35 | carrar | whats the question |
| 00:39.23 | neurosys | Message indicator light doesnt go off after msg deletion |
| 00:40.43 | carrar | was it ever off? |
| 00:41.38 | neurosys | Yes. |
| 00:41.56 | neurosys | It was off. left a message. Lite came on. Doesnt go of after deletion. |
| 00:42.45 | carrar | do a debub to see if it's being sent? |
| 00:42.48 | carrar | debug |
| 00:42.54 | neurosys | notify was sent |
| 00:44.06 | carrar | sure it's not set to on in the SIPDefault? |
| 00:44.24 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 00:44.24 | carrar | not sure if they ever put that in there |
| 00:44.32 | neurosys | well. If i reboot the phone without new mail.. its off. |
| 00:44.46 | neurosys | If i call it and hang up it stays off. If i leave a msg, it comes on. |
| 00:44.54 | neurosys | If i remove the new msg it stays on. |
| 00:45.03 | *** join/#asterisk Yudaisrael1984 (~Yuda@80.179.161.117.static.012.net.il) |
| 00:45.03 | carrar | the phone behind nat? |
| 00:45.12 | neurosys | Yep |
| 00:45.16 | ChannelZ | you could turn on SIP debug and make sure Asterisk is yelling at the phone about status updates |
| 00:45.21 | Yudaisrael1984 | hi guys does anyone know how do i recompile just the app_voicemail module??? |
| 00:45.24 | carrar | just the one phone or all 7960's |
| 00:45.35 | neurosys | Just the 7960's. |
| 00:45.42 | neurosys | The SPA's and Polys work fine |
| 00:45.52 | carrar | might check your firmware |
| 00:46.40 | neurosys | hmm. I think i know what it is.. brb |
| 00:46.41 | carrar | stuff breaks all the time in the cisco firmware depending on what version you are running |
| 00:46.57 | Yudaisrael1984 | i have a problem that the MWI doesnt work at all on any device so i would like to recompile the voicemail version |
| 00:47.05 | Yudaisrael1984 | anyone know how i would go about with that? |
| 00:47.08 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:47.16 | carrar | neurosys, do tell |
| 00:48.44 | neurosys | err in SEPXXX.cnf.xml.. brb |
| 00:49.09 | carrar | You running SCCP? |
| 00:49.13 | neurosys | no |
| 00:49.19 | ChannelZ | Yudaisrael1984: Do you think maybe it could be your config or devices? |
| 00:49.20 | neurosys | is that for skinny only? |
| 00:49.20 | carrar | that file isn't used then |
| 00:49.25 | neurosys | oh :( |
| 00:49.39 | Yudaisrael1984 | i looked at the devices and at the config everything looks correct |
| 00:50.16 | Yudaisrael1984 | i am running the same version and setup on a different machine and it works fine |
| 00:50.45 | Yudaisrael1984 | thats why i beleive its the version of voicemail (the newer one is using 1.6.31 and this one is 1.6.18 |
| 00:51.13 | carrar | Yudaisrael1984, verify with a debug that asterisk is or is not sending it |
| 00:51.29 | Yudaisrael1984 | checking |
| 00:52.11 | p3nguin | The problem does not exist in SIP 8.11 with later versions of Asterisk 1.4. |
| 00:53.22 | carrar | I've not had that problem with any firmware version that I recall |
| 00:53.26 | Yudaisrael1984 | ok i just looked where would i see if asterisk is sending the mwi notice? |
| 00:53.43 | carrar | Yudaisrael1984, it the SIP Packets |
| 00:53.46 | carrar | in |
| 00:54.01 | Yudaisrael1984 | it would be in options? |
| 00:54.08 | Yudaisrael1984 | which sip packet? |
| 00:54.17 | neurosys | Im using SIP 8.8. could that be the problem? |
| 00:54.31 | p3nguin | I would certainly try a newer version. |
| 00:54.36 | *** part/#asterisk kam187 (~kam187@81-179-8-102.dsl.pipex.com) |
| 00:55.14 | neurosys | thx p3nguin |
| 01:22.26 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 01:24.34 | neurosys | Ugh to get 8.12 i need access on cisco site. lame |
| 01:24.36 | carrar | ohayoo |
| 01:25.18 | carrar | Well you have Cisco support on your phone since I know you are doing it legally :) |
| 01:26.05 | neurosys | it says: To download this software you must have a valid Technical Support Services Agreement associated to your Cisco.com user ID. |
| 01:26.55 | carrar | sounds correct |
| 01:28.42 | *** join/#asterisk DanteLynch (~dantelync@66-190-208-43.dhcp.slid.la.charter.com) |
| 01:33.03 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:34.36 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:36.06 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:36.09 | neurosys | p3nguin: & carrar: firware update worked . Thx guys! |
| 01:36.40 | carrar | running 8.12.0? |
| 01:36.44 | neurosys | yes |
| 01:39.26 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:41.04 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:42.33 | *** part/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 01:42.49 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:42.53 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 01:44.35 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:44.47 | p3nguin | neurosys: Now you have to suffer the caller ID bug. |
| 01:46.25 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:46.53 | neurosys | p3nguin: hehe what bug is that |
| 01:48.21 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:49.05 | p3nguin | neurosys: something to do with whitespaces. |
| 01:49.22 | p3nguin | I probably should have been more clear when I tried recommending 8.11. |
| 01:50.27 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:50.40 | neurosys | well i can simply reflash it |
| 01:53.27 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:54.54 | p3nguin | I suggest you go ahead and do that before you are annoyed by it. |
| 01:56.56 | neurosys | p3nguin: Thx :) |
| 01:59.29 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 02:09.02 | *** join/#asterisk mianos (~mrfish@124-168-146-31.dyn.iinet.net.au) |
| 02:11.27 | *** join/#asterisk ming_zym (~ming_zym@124.14.75.23) |
| 02:12.17 | knarfly | rolls up another one and blazes into the night |
| 02:12.23 | *** part/#asterisk knarfly (~vlad@c-98-242-233-74.hsd1.fl.comcast.net) |
| 02:22.28 | sawgood | besides a ring group, is there any other way to have an incominmg call on a SIP trunk ring more than one extesnion? |
| 02:23.02 | sawgood | Or, even better, is there a way to allow more than one SIP end point to register as the same extension |
| 02:23.15 | sawgood | so, for example, three phones all have a registred line for extension 101? |
| 02:26.42 | TJNII | No, you can't register 3 phones on one sip peer. |
| 02:26.54 | TJNII | You can make one dial ring 3 phones, though. |
| 02:26.57 | TJNII | Or a queue. |
| 02:27.52 | sawgood | oh, a queue .... are queues considered extensions like a parking lot is considered an extension? |
| 02:28.41 | TJNII | .....what? You can have a exten line that calles the queue app. |
| 02:29.18 | sawgood | I guess I meant for BLF purposes ... the SIP end points I am using do not support queue buttons unless they are BLF |
| 02:29.29 | sawgood | its a drag ... but for a low end phone it does a good job |
| 02:29.56 | TJNII | I don't know much about BLF, so I can't answer that question. |
| 02:30.17 | TJNII | I think you can hint the queue, but I haven't so much as thought about BLF in years so I'm probably wrong. |
| 02:30.27 | sawgood | no problem ... how you been anyhow? |
| 02:30.43 | sawgood | hints might work ... I'll read about it shortly |
| 02:30.45 | sawgood | thanks! |
| 02:30.47 | TJNII | np. |
| 02:32.45 | sawgood | when you have the CLI set with this command: core set verbose 5 (and then you start to see messages appear on the console when calls arrive) ... at the same time, are these same messages being saved in a log file as well? |
| 02:33.21 | carrar | if you set it |
| 02:33.42 | carrar | look at logger.conf |
| 02:33.46 | TJNII | /var/log/asterisk/messages iirc |
| 02:33.51 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 02:34.17 | carrar | but it's independant of the cli set verbose command |
| 02:34.24 | sawgood | cool ... I"m checking now |
| 02:34.47 | sawgood | oh ... cool ... now I can visualize why they say make sure you run logger rotate |
| 02:34.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:35.30 | carrar | more logging takes performance hits also |
| 02:54.09 | *** join/#asterisk coppice (~chatzilla@19.176.64.202.dyn.pacific.net.hk) |
| 03:02.40 | mianos | does anyone use asterisk-gui I've been setting up up for a few weeks and aside from a lot of minor patches I have one issue, which may be my lack of understanding |
| 03:06.33 | *** join/#asterisk jhirley (~jhirley@adsl-161-49-28.mia.bellsouth.net) |
| 03:11.16 | mianos | it does not seem to manage extensions |
| 03:12.11 | *** join/#asterisk zerohalo (~zerohalo@cambridge.zerohalo.com) |
| 03:15.53 | *** join/#asterisk zerohalo (~zerohalo@cambridge.zerohalo.com) |
| 03:23.33 | ChannelZ | to hell with Asterisk GUIs |
| 03:55.21 | mianos | yer normally I'd just edit the file |
| 03:55.33 | mianos | but I was thinkingif I ahd to mess with it a lot a basic gui would be good |
| 03:55.40 | mianos | not like you'd ever need a gui for trunks and such |
| 03:56.29 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 03:57.15 | p3nguin | asterisk-gui has died. |
| 03:57.59 | TJNII | "I ahd to mess with it a lot a basic gui would be good" "I've been setting up up for a few weeks" So how's that been workin' out for ya? |
| 03:58.33 | mianos | messing as in, seting soemthing up then spend a hour or so fixing little bugs and such |
| 03:58.51 | mianos | adding a bit here and there, like OSLEC for drop downs |
| 03:59.06 | mianos | addign tests for dahdi where zaptel is tested |
| 03:59.10 | mianos | just little things |
| 03:59.10 | p3nguin | If you're spending WEEKS to get a GUI working, you've got issues. |
| 03:59.15 | mianos | it's pretty good now |
| 03:59.21 | mianos | hobby |
| 03:59.31 | mianos | not full time |
| 03:59.33 | TJNII | Thank you p3nguin. I was trying to draw attention to that, but was too subtle. |
| 03:59.38 | mianos | maybe an hour a day |
| 03:59.41 | mianos | it's no rush |
| 03:59.54 | mianos | I already have a 1.4 server with manually edited files running for 2 years |
| 04:00.19 | mianos | so I got a TDM200P and a spare box |
| 04:00.24 | mianos | to have a mess with 1.6 |
| 04:00.34 | p3nguin | If you're trying to use, specifically, the asterisk-gui, you have different issues. |
| 04:01.16 | mianos | ok, question, if I wanted to use a guii, for the first time, on a debian lucid box that already has aster 1.6 what would you use? |
| 04:01.29 | mianos | the op system is choses as I have a heap of stuff already on the box it's going into |
| 04:01.37 | p3nguin | I wouldn't use a GUI. That's what I would use. |
| 04:01.43 | TJNII | emacs. It is a great GUI, though many would disagree. |
| 04:01.54 | mianos | I would disagree |
| 04:02.39 | mianos | on emacs |
| 04:02.39 | mianos | the prod system was usign a gui called |
| 04:02.39 | mianos | VIM |
| 04:02.39 | mianos | :) |
| 04:02.39 | p3nguin | But if you wanted to sacrifice all your control and make your system a steaming pile, you could always use FreePBX. |
| 04:02.40 | mianos | I started to look at tthat first |
| 04:02.49 | mianos | it seemed a bit OTT |
| 04:02.56 | mianos | and does not want to work with the aster I have |
| 04:03.03 | mianos | probabyl great for a bootable server |
| 04:03.15 | mianos | asetrisk-gui is small |
| 04:03.21 | mianos | and kinda works |
| 04:03.36 | mianos | anyway, if you have not used it you are not going to know the specific question |
| 04:03.50 | mianos | and, sure I can edit the extenstions.conf to add my own handsets |
| 04:04.09 | mianos | which will happen if i can't find out how asterisk-gui is meant to do it |
| 04:04.24 | mianos | but I find it hard to beleive it automates pretty much everything else |
| 04:04.28 | mianos | and not the handsets |
| 04:06.02 | mianos | I'm sure there is something just slightly wrong |
| 04:06.34 | mianos | the incoming trunks 'goto' [default] yet the gui seems to leave the whole [default] section alone |
| 04:07.27 | mianos | it a;so writes the stdexten macro into the file ready to use |
| 04:10.52 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 04:23.09 | *** join/#asterisk Corydon76-dig (four@c-69-137-80-31.hsd1.tn.comcast.net) |
| 04:23.09 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 04:23.38 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 04:23.43 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 04:26.29 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 04:29.25 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 04:34.39 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 04:43.00 | *** join/#asterisk [8none1] (~8none1]@ps14528.dreamhost.com) |
| 04:51.27 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 05:13.12 | *** join/#asterisk blaines (~blaines@67.130.168.2) |
| 05:18.42 | *** join/#asterisk Alagar (~Administr@122.164.38.122) |
| 05:23.25 | *** join/#asterisk Micc (~Micc@c-98-225-57-96.hsd1.wa.comcast.net) |
| 05:33.39 | *** join/#asterisk geneg1 (~gene@bas3-toronto01-1177778644.dsl.bell.ca) |
| 05:34.22 | *** join/#asterisk jhirley_ (~jhirley@adsl-159-225-212.mia.bellsouth.net) |
| 05:45.30 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 05:50.15 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 06:03.34 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 06:20.18 | *** join/#asterisk d4rkstar (~bruno@93-42-1-91.ip84.fastwebnet.it) |
| 06:24.48 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:25.21 | ChannelZ | ATTENTION: |
| 06:25.28 | ChannelZ | Jiggly boobs. |
| 06:25.32 | ChannelZ | That is all. |
| 06:31.37 | *** join/#asterisk jhirley_ (~jhirley@adsl-159-226-46.mia.bellsouth.net) |
| 06:48.35 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 07:04.09 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 07:23.08 | *** join/#asterisk af_ (~getsmart@88-149-240-109.dynamic.ngi.it) |
| 07:25.45 | *** join/#asterisk Micc (~Micc@c-98-225-57-96.hsd1.wa.comcast.net) |
| 07:43.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 07:46.37 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 07:59.05 | QN | hi all. anyone has any idea what does "Got SIP response 503 "Service Unavailable." back from xxx.xxx.xxx.xxx" means? |
| 08:00.08 | ChannelZ | Hmm. Service must not be available. |
| 08:10.28 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 08:10.45 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 08:12.35 | *** join/#asterisk shimi (~shimi@unaffiliated/shimi) |
| 08:14.44 | shimi | Hey all, I am having an issue. I'm receiving around 160k interrupts/sec from my TE220P. Ideas? |
| 08:15.28 | shimi | erg, 16k |
| 08:24.13 | shimi | now I see it's all the way between 10k-160k interrupts per sec :0 |
| 08:25.29 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:28.37 | *** join/#asterisk nova911 (~Adium@115.118.226.6) |
| 08:33.12 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 08:35.35 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 08:49.20 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 08:52.03 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 09:06.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 09:42.14 | *** join/#asterisk hipitihop (~denis@203.132.229.236) |
| 09:49.29 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 09:51.16 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 10:13.20 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 10:14.14 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 10:29.08 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:34.49 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 10:38.00 | *** join/#asterisk g_r_eek (~g_r_eek@dslb-092-074-035-151.pools.arcor-ip.net) |
| 10:39.51 | *** join/#asterisk stix (~stix@exchange2003.corporate.billetkontoret.dk) |
| 10:51.18 | *** join/#asterisk Trixboxer (~Trixboxer@datacenter3.supportdepartment.net) |
| 10:53.02 | *** join/#asterisk spiri (~ooze@vhs.hackspace.ca) |
| 10:54.57 | spiri | heres a strange question... Im running a asterisk box.. that I connect to with my wifi voip client. I was at my friends shop today and had my phone connected over their wifi.. a call was placed to me so I picked it up and chatted for a little while. However it turns out that their voip awnsering machine recorded the entire call. Does anyone know how this could happen? |
| 10:55.31 | spiri | their == my friends shop awnsering machine |
| 10:55.48 | *** join/#asterisk BANSAL (~bansal@117.199.123.100) |
| 10:56.35 | Trixboxer | rtp port recording and SIP dialog |
| 10:57.16 | spiri | Trixboxer: thanks.. are those terms I should google? |
| 10:57.59 | spiri | any idea of configuration changes I can make to fix this? |
| 10:58.01 | Trixboxer | to be frank I never tried but its just my thought .. but yes.. you can start googling with it |
| 10:58.04 | Trixboxer | nopw |
| 10:58.08 | Trixboxer | no config |
| 10:58.12 | spiri | ok |
| 10:58.15 | spiri | cheers |
| 10:58.17 | Trixboxer | you can just use a tunnel |
| 10:58.36 | Trixboxer | secure like VPN between the clients and users |
| 10:58.41 | Trixboxer | clients & server |
| 10:59.10 | spiri | hrmm.. yes. my phone does support vpn. I was considering this before coming here, thanks |
| 10:59.44 | Trixboxer | cheers |
| 11:07.48 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 11:13.53 | *** join/#asterisk lewq (~luke@127cr.net) |
| 11:22.13 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:37.17 | *** join/#asterisk heppa (~heppa@c-24-14-214-36.hsd1.il.comcast.net) |
| 11:37.57 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 11:40.53 | *** join/#asterisk fleebailey33 (~fleebaile@unaffiliated/fleebailey33) |
| 11:41.25 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 11:41.25 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 11:44.08 | *** join/#asterisk decklar (~c4d48662@gateway/web/freenode/x-guizpxzvbtkdnols) |
| 11:47.09 | decklar | can anyone tell me the best way to remove freepbx on a fresh install of asteriskNow? |
| 11:47.18 | decklar | yum remove freepbx does not work as expected... |
| 11:48.09 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 11:48.13 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 11:51.09 | *** part/#asterisk decklar (~c4d48662@gateway/web/freenode/x-guizpxzvbtkdnols) |
| 11:56.44 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 12:04.08 | *** join/#asterisk jhirley_ (~jhirley@adsl-4-162-55.mia.bellsouth.net) |
| 12:04.09 | *** join/#asterisk m_tadeu (~quassel@89-180-0-145.net.novis.pt) |
| 12:12.58 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 12:17.11 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 12:35.37 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 12:35.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 12:51.07 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 12:52.35 | *** join/#asterisk nova911 (~Adium@115.118.110.174) |
| 13:02.54 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 13:11.17 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 13:12.56 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 13:14.19 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:17.09 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 13:22.09 | *** join/#asterisk coppice (~chatzilla@19.176.64.202.dyn.pacific.net.hk) |
| 13:36.14 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 13:45.38 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:55.49 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:56.35 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:59.08 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 14:00.29 | *** join/#asterisk Carlos_Tico (~carlos@c-98-201-56-25.hsd1.tx.comcast.net) |
| 14:02.37 | *** join/#asterisk af_ (~getsmart@88-149-240-118.dynamic.ngi.it) |
| 14:04.07 | *** join/#asterisk Wildy (~simba@217.197.245.114) |
| 14:05.46 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 14:07.07 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.85) |
| 14:09.08 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 14:17.56 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 14:25.42 | *** join/#asterisk stix_ (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 14:33.36 | *** join/#asterisk stix (~stix@exchange2003.corporate.billetkontoret.dk) |
| 14:42.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:44.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 14:58.32 | *** join/#asterisk d4rkstar (~bruno@93-42-1-91.ip84.fastwebnet.it) |
| 14:58.52 | d4rkstar | someone that can give me some hints about FFA and t.38? |
| 14:59.17 | d4rkstar | ?ffa |
| 15:01.26 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-64-227-75.mia.bellsouth.net) |
| 15:08.19 | *** join/#asterisk Bentley (~Bentley@174.0.91.254) |
| 15:08.23 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 15:09.40 | Bentley | hi all, i have a problem with my IAX provider initiating a hangup to my server. it sends an IAX2 "CONTROL Subclass: (20?)" followed by HANGUP. How do I determine out what (20?) is ? |
| 15:11.15 | drmessano | What hints? |
| 15:13.42 | Carlos_Tico | how can i turn on the colors on asterisk CLI ? |
| 15:14.30 | Kyosh | umm |
| 15:14.33 | Kyosh | colors? |
| 15:14.39 | Kyosh | no shit? |
| 15:15.12 | drmessano | Colors should be enabled by default. Check your SSH client |
| 15:17.22 | d4rkstar | fax session terminate on my asterisk with: FAX session '2' is complete, result: 'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' |
| 15:19.10 | *** join/#asterisk smooth_penguin (~smoove@59.95.34.39) |
| 15:22.40 | *** join/#asterisk yang (yang@freenode/sponsor/cacert.assurer.yang) |
| 15:28.08 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:28.31 | *** join/#asterisk Wildy (~simba@83.149.9.113) |
| 15:33.52 | *** join/#asterisk QaDeS (~mklaus@p4FC70F0C.dip0.t-ipconnect.de) |
| 15:52.38 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 15:52.53 | *** join/#asterisk smooth_penguin (~smoove@59.95.20.42) |
| 15:58.10 | *** join/#asterisk Wildy (~simba@178.17.199.143) |
| 16:04.38 | *** join/#asterisk jmacz (~jmacz@186.29.108.139) |
| 16:08.20 | *** join/#asterisk Tim_Toady (~moi@77.49.21.138) |
| 16:15.08 | d4rkstar | why i can't receive fax with asterisk FFA when t38_udptl=yes ? |
| 16:15.22 | d4rkstar | i got a T1_TIMEOUT error |
| 16:15.29 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:16.59 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 16:23.01 | d4rkstar | if i try to call fax extension i can hear the fax beep for 2 seconds, then just nothing |
| 16:44.45 | d4rkstar | i need some help for fax problem with asterisk |
| 16:45.31 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 16:48.49 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 16:54.20 | *** join/#asterisk Kyosh (~whoa@pool-74-108-19-33.nycmny.fios.verizon.net) |
| 16:54.52 | *** join/#asterisk nova911 (~Adium@115.118.110.174) |
| 16:57.19 | Mango | ~itsplist-ca |
| 16:57.20 | infobot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
| 16:58.44 | Kyosh | are those itsp's partners with any company like digium or how do they get on the botlist for itsp's? |
| 17:02.14 | d4rkstar | ~ffa |
| 17:02.15 | infobot | somebody said ffa was ffa stands for free for all, or not CTF |
| 17:02.22 | d4rkstar | ~fax |
| 17:02.23 | infobot | [fax] The honor of designing the first fax *service* in actual use goes to Giovanni Caselli, an Italian abbot, born in Siena in 1815, who turned his hand to science and was, by 1849, editing a scientific magazine. In 1856 he claimed that he had developed a device, which he called a "pantelegraph," that could send facsimiles of images and text. Napoleon III did not come up with the idea, he merely backed it. |
| 17:02.38 | d4rkstar | ~fax for asterisk |
| 17:02.39 | infobot | from memory, fax for asterisk is http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf |
| 17:02.48 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:02.57 | d4rkstar | ~T1_TIMEOUT |
| 17:10.08 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 17:13.58 | coppice | Time-out T1 defines the amount of time two terminals will continue to attempt to identify each other. T1 is 35+-5s, begins upon entering phase B, and is reset upon detecting a valid signal or when T1 times out. |
| 17:16.21 | *** join/#asterisk BANSAL (~bansal@117.199.123.100) |
| 17:18.49 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 17:19.04 | d4rkstar | coppice: thankyou |
| 17:19.27 | *** join/#asterisk moy (~moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 17:19.33 | drmessano | ~google |
| 17:19.34 | infobot | methinks google is http://lmgtfy.com/?q=google |
| 17:19.35 | d4rkstar | i'm trying to understand why asterisk fax (with digum's FFA) doesn't work |
| 17:20.16 | d4rkstar | ~drmessano |
| 17:20.17 | infobot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily |
| 17:24.01 | d4rkstar | drmessano, i didn't find any useful information about the problem i'm facing |
| 17:24.10 | d4rkstar | that's the way i'm here |
| 17:25.52 | d4rkstar | maybe people around here can give hints or help on troubleshooting |
| 17:29.37 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 17:30.32 | drmessano | d4rkstar: Maybe you should provide a full debug of the fax call. One error snippit is meaningless, and may not even be related |
| 17:30.54 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 17:31.38 | coppice | T1 timeout normally means the SIP or UDPTL path is broken. NAT is a god starting point for investigating where |
| 17:32.06 | d4rkstar | ok |
| 17:32.55 | d4rkstar | coppice, the asterisk machine has public ip and is not behind nat |
| 17:33.11 | d4rkstar | there's a firewall with iptables on it |
| 17:33.38 | d4rkstar | what ports should i check as opened? |
| 17:33.52 | d4rkstar | drmessano, ok. I'll grab all the logs needed |
| 17:34.08 | coppice | why do you expect other people to know how you configured your own system? |
| 17:40.12 | d4rkstar | ok coppice |
| 17:40.17 | d4rkstar | sorry for my mistake |
| 17:40.18 | d4rkstar | http://pastebin.com/LArcSAPL |
| 17:40.35 | d4rkstar | here should be some kind of info |
| 17:40.39 | d4rkstar | now i grab also a fax session |
| 17:41.52 | drmessano | Why do you have localnet and externip set ? |
| 17:42.48 | d4rkstar | uhm |
| 17:43.19 | d4rkstar | i can remove localnet |
| 17:43.32 | drmessano | You don't need EITHER of those if you're on a public IP |
| 17:44.31 | d4rkstar | ok, i'll change that |
| 17:44.33 | d4rkstar | http://pastebin.com/xdB8yiwk |
| 17:44.37 | d4rkstar | this is a fax session log |
| 17:45.59 | Mango | For the record, I've never been able to make Zoiper work with faxes. |
| 17:47.17 | d4rkstar | Mango: with what zoiper version? i got zopier communicator 1.18.6898 on linux |
| 17:47.31 | d4rkstar | s/zopier/zoiper/g |
| 17:47.36 | *** join/#asterisk MiserySoft (~MiserySof@host81-139-171-30.in-addr.btopenworld.com) |
| 17:47.44 | Mango | 2.24.5324 on Windows |
| 17:48.26 | Mango | I have been able to make VoIP Plug-In for Microsoft Fax work for sending, but not receiving. |
| 17:48.46 | *** join/#asterisk malaka123 (~malaka@112.135.167.133) |
| 17:49.10 | d4rkstar | drmessano: if i try to call fax extension, i hear the fax beep for 2 seconds and then nothing else |
| 17:49.32 | drmessano | Ok, and see what Mango said about Zoiper |
| 17:50.00 | malaka123 | HI all, I'm new to asterisk, and i need to know some details how to do IP based authetication |
| 17:50.11 | malaka123 | i tried every way |
| 17:50.29 | malaka123 | but nothing is working for me, when i used fromuser trunk is working fine |
| 17:50.40 | malaka123 | but without that it's not |
| 17:50.42 | malaka123 | any idea |
| 17:51.08 | d4rkstar | drmessano: i tried also to receive fax sending it from a real fax attached to an spa3102 |
| 17:51.46 | d4rkstar | the result is always the same when t38 is enabled |
| 17:55.24 | malaka123 | guys any idea about my SIP trunk IP base authentication issue |
| 17:55.24 | emora | Anyone using monit to monitor Asterisk? |
| 17:56.06 | emora | I have a system that is up but it keeps showing Execution failed. |
| 17:56.09 | emora | any clues? |
| 17:56.32 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:56.32 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:57.06 | malaka123 | Emora.. can u explain a bit |
| 17:57.25 | malaka123 | I'm new to asterisk, but i might be able to help you, if u can expalin |
| 18:00.22 | d4rkstar | ok, i've to go now. see you later |
| 18:06.00 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 18:08.27 | *** part/#asterisk malaka123 (~malaka@112.135.167.133) |
| 18:10.14 | *** join/#asterisk lirakis (~root@ool-ad024dab.dyn.optonline.net) |
| 18:10.21 | *** part/#asterisk lirakis (~root@ool-ad024dab.dyn.optonline.net) |
| 18:13.27 | *** join/#asterisk DennisG (~DennisG@84.30.136.208) |
| 18:14.02 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 18:17.55 | *** join/#asterisk MiserySoft (~MiserySof@host81-139-171-30.in-addr.btopenworld.com) |
| 18:28.31 | *** join/#asterisk ryan-c (~ryan@c-98-207-154-10.hsd1.ca.comcast.net) |
| 18:29.13 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 18:31.23 | *** join/#asterisk lirakis (~root@ool-ad024dab.dyn.optonline.net) |
| 18:31.58 | *** part/#asterisk lirakis (~root@ool-ad024dab.dyn.optonline.net) |
| 18:33.20 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 18:37.40 | *** join/#asterisk lirakis (~root@ool-ad024dab.dyn.optonline.net) |
| 18:37.57 | lirakis | palabra arriba |
| 18:39.21 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 18:41.24 | lirakis | is working on re setting up his irc proxy |
| 18:41.35 | *** join/#asterisk wezp (~none@195-241-113-85.ip.telfort.nl) |
| 18:41.52 | *** part/#asterisk wezp (~none@195-241-113-85.ip.telfort.nl) |
| 18:46.23 | *** join/#asterisk wezp (~none@195-241-113-85.ip.telfort.nl) |
| 18:47.49 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:48.14 | *** join/#asterisk Trixboxer (~Trixboxer@115.113.145.85) |
| 18:50.58 | *** join/#asterisk Marquel (~Flinx@p548E2D82.dip.t-dialin.net) |
| 18:51.31 | Marquel | morning. |
| 18:52.36 | *** part/#asterisk wezp (~none@195-241-113-85.ip.telfort.nl) |
| 18:55.29 | Marquel | i have a problem w/asterisk-1.2 (yes, i know, old, outdated - i have yet to confirm that 1.4 works with my bri-card...): i want to signal a ringing ZAP phone (one specific number) via hint to a busy lamp on a grandstream gxp2010. now the lamp lights up green (meaning phone is available) but it doesn't change state during incoming or outgoing calls on that zap-number... what can i do? |
| 18:58.33 | *** join/#asterisk Tim_Toady (~moi@77.49.21.138.dsl.dyn.forthnet.gr) |
| 19:00.13 | [TK]D-Fender | Marquel: What ver precisely? Also show us a failedcall and your configs |
| 19:00.15 | [TK]D-Fender | ~pb |
| 19:00.16 | 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. |
| 19:00.17 | [TK]D-Fender | ^^^^^^^ |
| 19:00.53 | Marquel | i know a pastebin ;) |
| 19:01.18 | Marquel | *CLI> show version |
| 19:01.18 | phix | hihihihihihihihii |
| 19:01.19 | Marquel | Asterisk 1.2.27-BRIstuffed-0.3.0-PRE-1y-p built by root @ server on a x86_64 running Linux on 2008-08-30 14:30:57 UTC |
| 19:01.30 | phix | [TK]D-Fender: hey buddy, what you been up to recently? |
| 19:03.51 | [TK]D-Fender | phix: Keeping busy. Boatloads of music, badminton for the past 2 days and I think a shower is about to nix my thoughts of going biking... |
| 19:03.56 | [TK]D-Fender | phix: You? |
| 19:04.43 | Marquel | [TK]D-Fender: extensions.conf (part with busy lamp field) |
| 19:05.09 | Marquel | what else do you need? |
| 19:05.11 | phix | hi gang |
| 19:05.21 | phix | [TK]D-Fender: nice, haha badmintion why |
| 19:05.33 | phix | [TK]D-Fender: Iam in the UK atm |
| 19:05.37 | phix | touring around |
| 19:05.44 | phix | going to go to wales for a week soon as well |
| 19:05.57 | phix | Youare in the USA hey [TK]D-Fender |
| 19:08.03 | tzafrir_laptop | Marquel, BRI cards are supported in chan_dahdi as of 1.6.0 |
| 19:08.03 | [TK]D-Fender | phix: CANADA. |
| 19:08.03 | Marquel | tzafrir_laptop: i think i'll call the manufacturer by tomorrow. usually they are fast in competent answers ;) |
| 19:08.10 | [TK]D-Fender | phix: And why badminton? My (no) ex-gf got me a racket for my BD and I do enjoy the game. I'm more of a squash player really, but I'll take what I can get, esp as it helps validate my gym membership |
| 19:08.36 | [TK]D-Fender | *now |
| 19:12.26 | neurosys | I just got a nice gift: 4 SPA962s brand new. Cant believe @230.00 a pop they cant do g722 tho :( |
| 19:13.29 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 19:14.49 | phix | [TK]D-Fender: oh Canada, so you have a better health system and Moose, that is about it hey :) |
| 19:14.53 | phix | the rest is the same |
| 19:15.32 | phix | [TK]D-Fender: I see, I am more of a snowboarder or skateboarder than a badmittoner |
| 19:16.28 | phix | Apparntly our whole universe was in a hot dense state |
| 19:16.42 | phix | then nearly 14 billions years ago |
| 19:16.54 | phix | epsnension starated, wait |
| 19:19.23 | blop | hello |
| 19:19.32 | phix | lets blop |
| 19:19.46 | blop | i do have a clue for my tdm400p problems ;) |
| 19:19.57 | phix | yay |
| 19:20.00 | phix | I has a tdm400 |
| 19:20.04 | phix | p |
| 19:20.15 | ChannelZ | blop blop fizz fizz |
| 19:20.30 | phix | 03:00.0 Ethernet controller: Digium, Inc. Device 8005 (rev 11) |
| 19:20.43 | phix | Apparantly it is an ethernet controller |
| 19:20.51 | phix | who would of known |
| 19:20.54 | blop | i have a tdm400p rev H and a tdm400p rev J, the rev J gets detected as a rev I by the driver and works well. The rev H is detected as a rev E/F and does not work well :) maybe thats why? |
| 19:21.11 | phix | perhas |
| 19:21.17 | phix | lets all just ponder on that |
| 19:21.24 | TJNII | ponders |
| 19:21.36 | phix | then discuss the results of the pondering |
| 19:21.47 | tzafrir_laptop | Marquel, what do you actually get in the dialplan when a call comes in? |
| 19:22.20 | phix | I of course will get side track on the all important argument, is the cake a lie, or not? |
| 19:22.37 | phix | arugment / question |
| 19:24.07 | Marquel | tzafrir_laptop: Set(LANGUAGE()=de); Set(CALLERID(num)=25${CALLERID(num)}); Dial(ZAP/g1/46,120,gkwrt); |
| 19:26.05 | *** join/#asterisk QaDeS (~mklaus@p4FC70F0C.dip0.t-ipconnect.de) |
| 19:26.14 | tzafrir_laptop | blop, do you have two different cards? |
| 19:26.58 | tzafrir_laptop | Marquel, what would you expect that dialplan to do? |
| 19:28.25 | Marquel | tzafrir_laptop: exactly that. ring ZAP/g1/46. it does, picking it up connects me to the caller. all no problem. but i also expect the dialplan to signal ringing/busy to a signal lamp on a SIP phone (just like those lamps on a receptionist's phone). and that does not happen. |
| 19:29.07 | Marquel | tzafrir_laptop: yes, two cards. one for all external stuff and one for all internal stuff. the external card is managed via chan_capi.so. |
| 19:30.00 | tzafrir_laptop | I'm not sure how well that worked. IIRC BRIStuff had some fixes to make that work, but I'm not really sure |
| 19:30.28 | [TK]D-Fender | Marquel: Where is the PASTEBIN I asked you for half an hour ago? |
| 19:30.52 | tzafrir_laptop | phix, BTW: http://lists.digium.com/pipermail/asterisk-users/2010-May/249042.html |
| 19:31.03 | tzafrir_laptop | I'm looking for data |
| 19:31.30 | Marquel | [TK]D-Fender: what parts of extensions.conf do you need? i'd have to anonymise much of it... |
| 19:34.33 | [TK]D-Fender | Marquel: If you have to ask then you probably don't even know what parts you needed to expect it to work in the first place |
| 19:35.55 | Marquel | [TK]D-Fender: http://pastebin.ca/1874429 - that's the part i expect to work, but seemingly doesn't. |
| 19:38.33 | [TK]D-Fender | MarI see nothing to suggest that the dialplan you have shown is used for anything at all. |
| 19:38.39 | [TK]D-Fender | Marquel: I see nothing to suggest that the dialplan you have shown is used for anything at all. |
| 19:39.30 | [TK]D-Fender | Marquel: exten => 46,hint,ZAP/g1/46 <-- also completely not valid |
| 19:39.36 | [TK]D-Fender | Marthat is not a DEVICE. |
| 19:39.46 | [TK]D-Fender | Marquel: You can't point to a group and extension combo |
| 19:40.42 | *** join/#asterisk voxter (~voxter@S010600090f53ea17.vc.shawcable.net) |
| 19:40.47 | Marquel | so exten => 46,hint,ZAP/46 would be correct then? |
| 19:42.57 | [TK]D-Fender | Marquel: If you have a Zap channelw ith that number |
| 19:43.37 | Marquel | i do. incoming calls are to Dial(ZAP/g1/46) and that works. |
| 19:44.42 | [TK]D-Fender | [15:43]<Marquel>i do. incoming calls are to Dial(ZAP/g1/46) and that works. <- NO |
| 19:45.13 | [TK]D-Fender | Marquel: that is dialing the NUMEBR 46 to your GROUP. This does NOT tell me how many CHANNELS are in that group |
| 19:45.31 | [TK]D-Fender | Marquel: Stop referring to a GROUP and EXTENSION pairing like that |
| 19:46.04 | [TK]D-Fender | Marquel: what cards do you have? |
| 19:46.23 | Marquel | one quadBRI and one fritz!PCI. |
| 19:46.30 | Marquel | duoBRI, sorry. |
| 19:47.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 19:47.56 | [TK]D-Fender | Marquel: then you can't even HAVE 46 channels. |
| 19:48.09 | [TK]D-Fender | Marquel: So stop saying that you're hinting channel 46 |
| 19:49.07 | Marquel | now, that makes sense. |
| 19:50.56 | *** join/#asterisk ChkDigit (~mike@static24-72-71-175.r.rev.accesscomm.ca) |
| 19:54.27 | neurosys | [TK]D-Fender: Is it safe to upgrade my Poly 550 to 3.2.3? |
| 19:56.16 | [TK]D-Fender | neurosys: Why wouldn't it be? |
| 19:56.44 | neurosys | [TK]D-Fender: Bugs, etc? like the cisco 8.12 callerID bug p3nguin mentioned? :) |
| 19:57.20 | [TK]D-Fender | neurosys: Cisco sucks :p |
| 19:57.26 | [TK]D-Fender | Polycom > All |
| 19:57.45 | neurosys | :) the cisco was for a client. I have polys now ;) |
| 19:57.46 | *** join/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 20:00.50 | tzafrir_laptop | [TK]D-Fender, where do you see 46 channels? |
| 20:01.27 | ChannelZ | Cable. And there's nothing on. |
| 20:01.43 | tzafrir_laptop | ah, DAHDI channels |
| 20:02.15 | [TK]D-Fender | [15:40]<Marquel>so exten => 46,hint,ZAP/46 would be correct then? |
| 20:02.17 | tzafrir_laptop | Marquel, it would be Zap/1 or Zap/2 , I guess |
| 20:02.28 | tzafrir_laptop | And not a single channel |
| 20:03.22 | [TK]D-Fender | exten => 46,hint,ZAP/g1/46 |
| 20:03.23 | [TK]D-Fender | tzafrir_laptop: Yeah, I think I bludgeoned that one through by now :) |
| 20:03.23 | [TK]D-Fender | (hope) |
| 20:03.38 | *** part/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 20:03.40 | Marquel | yeah, i guess so. not making it easier, as that'll mix the state of a phone i want monitored with one i do not... |
| 20:06.58 | blop | tzafrir_laptop: yes i have 2 different tdm400p , why? |
| 20:07.39 | [TK]D-Fender | Marquel: that doesn't make any sense |
| 20:08.03 | blop | rev H seems to not like my new computer, i tried change only the tdm card (same module, same pci slot, same driver). one work, the other one does not |
| 20:08.52 | blop | but the same card works well in another computer |
| 20:08.56 | blop | so its weird |
| 20:09.49 | Marquel | [TK]D-Fender: i have two phones connected to the same zap-channelgroup. thus, if one (and only one) rings, it will use channel 1. using ZAP/1 as hint will signal ringing then. right? |
| 20:10.57 | *** join/#asterisk Wildy (~simba@83.149.8.74) |
| 20:12.20 | Marquel | if the second phone rings (using channel 2) using ZAP/1 as hint will not signal ringing. right? |
| 20:12.21 | [TK]D-Fender | Marquel: Grouping has NOTHING to do with thi |
| 20:12.52 | [TK]D-Fender | Marquel: What precise channels do you want associated with a given hint? |
| 20:13.23 | Marquel | i want a _number_ associated with a hint. channel is useless, as it is dynamically assigned if it is needed. |
| 20:15.43 | [TK]D-Fender | Marquel: Your description is still broken |
| 20:15.51 | [TK]D-Fender | Marquel: You hint CHANNELS |
| 20:16.17 | [TK]D-Fender | Marquel: Did you want a SINGLE hint to collectively represent MULTIPLE channels? |
| 20:16.24 | Marquel | [TK]D-Fender: if i can only hint channels, it will never work the way i expect it. |
| 20:16.40 | [TK]D-Fender | Marquel: WTF is this "dynamic" business you're talking about? |
| 20:17.08 | Marquel | [TK]D-Fender: two channels. two phones. both phones using both channels as needed. |
| 20:17.18 | [TK]D-Fender | Marquel: AND? |
| 20:17.37 | *** join/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 20:17.49 | Marquel | i want exactly one phone monitored. if i can hint only channels, i will always have both connected phones monitored. |
| 20:18.23 | [TK]D-Fender | Marquel: One PHONE monitored? what kind of phone? Your description is EXTREMELY weak. |
| 20:19.12 | Marquel | [TK]D-Fender: break it down to this: what can be hinted? only channels? or what? |
| 20:20.07 | [TK]D-Fender | Marquel: No stop turning this in circles. What kind of phones are you using, and You have a bunch of Zap channels. A single hint points to a device or a series of devices and takes the highest state |
| 20:20.45 | *** part/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 20:21.04 | Marquel | ISDN phones, connected to ZAP channels. |
| 20:21.20 | [TK]D-Fender | Marquel: Who is monitoring the hints? |
| 20:21.25 | *** join/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 20:22.17 | Marquel | a sip phone. |
| 20:23.23 | doug | what's a good way to record stuff for asterisk vm messages on a mac? |
| 20:23.28 | doug | i.e. outside of asterisk |
| 20:23.44 | doug | i just wanna plop down a new greeting in 8khz gsm and upload it to my * server |
| 20:23.51 | *** part/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 20:24.31 | doug | i've got soundbooth |
| 20:24.32 | *** join/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 20:24.33 | doug | never used it tho |
| 20:24.50 | *** part/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 20:26.00 | [TK]D-Fender | Marquel: Ok. so you have ISDN phones. You also have ISDN LINES to the telco? |
| 20:26.19 | Marquel | [TK]D-Fender: yes. |
| 20:26.21 | doug | i assume 8k is probably best. not sure, actually. |
| 20:26.31 | doug | whoa, isdn lines. PRI's?? |
| 20:26.36 | [TK]D-Fender | MarSo what are your hints supposed to represent? |
| 20:26.56 | Marquel | [TK]D-Fender: if one of those ISDN phones is ringing or busy. |
| 20:27.17 | Marquel | (or idle, for completeness) |
| 20:27.27 | *** join/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 20:27.43 | [TK]D-Fender | Marquel: A single hint to show if EITHER is showing activity? Or is that one EACH? |
| 20:27.56 | Marquel | one each. |
| 20:28.41 | [TK]D-Fender | Marquel: then you've have 2 hints |
| 20:28.55 | [TK]D-Fender | Marquel: What channels are your 2 phones ? |
| 20:29.07 | Marquel | ZAP channels 1 and 2. |
| 20:29.14 | [TK]D-Fender | MarSo each uses 1? |
| 20:29.34 | [TK]D-Fender | Marquel: Phone 1 = Zap1/, Phone2 = Zap/2? |
| 20:29.58 | *** part/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 20:30.21 | [TK]D-Fender | Marquel: If so it's be exten => 11,hint,Zap/1 , exten => 12,hint,Zap/2 |
| 20:30.30 | [TK]D-Fender | (ext #'s just a sample) |
| 20:30.51 | Marquel | no. If Zap1 is free: Phone1 = Zap1 and Phone2 = Zap1. If Zap1 is busy, Phone1 = Zap2 and Phone2 = Zap2... |
| 20:31.37 | *** join/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 20:32.16 | [TK]D-Fender | .... |
| 20:32.33 | [TK]D-Fender | Marthat is not making sense |
| 20:32.37 | *** part/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 20:32.53 | [TK]D-Fender | Marquel: A phone has FIXED channels. How the hell are they CHANGING? |
| 20:33.08 | Marquel | [TK]D-Fender: why not? both phones are using the same wire. |
| 20:33.30 | [TK]D-Fender | .... |
| 20:33.35 | Marquel | there are two channels per wire. both phones are able to use two channels. |
| 20:33.37 | [TK]D-Fender | BRI crazyness |
| 20:33.49 | Marquel | ISDN at its best. yes. |
| 20:34.18 | [TK]D-Fender | Marquel: You aren't going to be able to differentiate those this wasy in 1.2 |
| 20:34.35 | Marquel | [TK]D-Fender: i suspected so. |
| 20:34.46 | [TK]D-Fender | Marquel: the only way is using something like a dialplan GROUP() and then using custom device states |
| 20:35.00 | [TK]D-Fender | Marquel: Which means 1.4+ (1.6 recomended) |
| 20:35.27 | Marquel | [TK]D-Fender: i'll find out if my bri-card is finally supported in 1.4 or 1.6 and possibly i'll update. |
| 20:44.34 | [TK]D-Fender | Marquel: this will rquire some messy dialplan including use of clean-up contexts, etc. |
| 20:44.48 | [TK]D-Fender | ("h" that is) |
| 20:44.54 | [TK]D-Fender | not "contexts" |
| 20:45.30 | Marquel | messy things are not my problem... |
| 20:47.10 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 20:51.56 | doug | cool, audicity and sox did the trick. |
| 20:51.57 | doug | doug++ |
| 20:54.15 | Marquel | [TK]D-Fender: is there a way to let asterisk log hint state changes? |
| 20:56.37 | [TK]D-Fender | Marquel: this method I have describes is pure dialplan with custom device-state hints. You can add whatever tracking you want |
| 20:57.06 | [TK]D-Fender | Marquel: because you will be required to add tracking code to all of your extens for this to work. As I said, it's going to be MESSY |
| 20:58.47 | Marquel | [TK]D-Fender: lucky me it's only one of those phones i want monitored... - if i'm in need of more than those in DECT, i'll buy a sip-dect-gap-base where i can assign one sip-channel to each phone and still be able to have more than one conversation ;) |
| 20:59.47 | [TK]D-Fender | Marquel: ISDN handsets = mess. Lines, not quite as much |
| 21:01.40 | Marquel | [TK]D-Fender: don't tell me. took me some time to get it working at all. not to mention two of the three handsets know nothing about ISDN (or the base, thus not even correct date and time on their displays, no missed calls, nothing)... |
| 21:02.35 | [TK]D-Fender | Marquel: Should have just gotten SIP phones and not gottent the card. |
| 21:03.39 | Marquel | needed the card for testing. and sip dect are ... expensive if i don't want to restrict myself to the crap a company called Siemens sells as "sip-dect"... |
| 21:04.33 | Marquel | that thing hasn't even a 100Mbit/s NIC. and is not capable of doing more than two concurrent calls... sometimes i'm tempted to feed that stuff to the engineers... from behind. |
| 21:07.12 | Marquel | (btw. got the DECT before i got the asterisk... and i didn't have enough money to flush down 120 bucks.) |
| 21:08.15 | neurosys | [TK]D-Fender: n00b question: how can I set the msg callback # in the cfg file? I set it in phone1.cfg but it doesnt overwrite the default ext. |
| 21:09.01 | neurosys | [TK]D-Fender: sorry. polycom :P |
| 21:09.17 | Marquel | [TK]D-Fender: okay, i'm falling asleep now. thanks for all your help :) |
| 21:20.52 | *** join/#asterisk d4rkstar (~bruno@93-42-1-91.ip84.fastwebnet.it) |
| 21:20.57 | d4rkstar | hi all |
| 21:22.08 | d4rkstar | drmessano: can you give me some hints on that fax issue? |
| 21:22.15 | [TK]D-Fender | neurosys: contact in the mwi section |
| 21:22.52 | neurosys | [TK]D-Fender: Yeah I changed that from reg to contact, and set the callback num to *97. But it didnt change :( |
| 21:23.04 | [TK]D-Fender | neurosys: and I see nothing |
| 21:23.09 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:23.16 | neurosys | [TK]D-Fender: hang on. |
| 21:24.15 | neurosys | http://pastebin.com/PbSMKrzK |
| 21:28.34 | neurosys | [TK]D-Fender: http://pastebin.com/PbSMKrzK |
| 21:29.39 | d4rkstar | hi, i'm facing a fax issue and need some hints about possible mistakes. Here is my configuration: http://pastebin.com/YMJMWgru |
| 21:29.55 | d4rkstar | the problem is with t38: http://pastebin.com/xdB8yiwk |
| 21:32.36 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 21:35.15 | *** join/#asterisk knarfly (~vlad@c-98-242-233-74.hsd1.fl.comcast.net) |
| 21:35.42 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 21:36.46 | knarfly | does automon require sox or is there another way |
| 21:38.49 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 21:38.57 | d4rkstar | to use t38, also with digium's free fax for asterisk, do i need to build spandsp and app_fax |
| 21:38.59 | d4rkstar | ? |
| 21:41.27 | [TK]D-Fender | neurosys: Your phone1.cfg doesn't have your reg details. it SHOULD. Why doesn't it? |
| 21:44.02 | d4rkstar | to use t38, also with digium's free fax for asterisk, do i need to build spandsp and app_fax? |
| 21:48.47 | *** join/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 21:51.07 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:53.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:54.18 | neurosys | [TK]D-Fender: I have my reg details in a more specified files. |
| 21:54.22 | neurosys | [TK]D-Fender: http://pastebin.com/n3qBQYCP |
| 21:57.53 | [TK]D-Fender | neurosys: check for repeat stanzas |
| 21:58.05 | [TK]D-Fender | neurosys: breaking it down so much one it likely overriding the other |
| 22:01.18 | *** join/#asterisk jhirley (~jhirley@adsl-159-229-9.mia.bellsouth.net) |
| 22:03.11 | *** join/#asterisk jhirley (~jhirley@adsl-159-229-9.mia.bellsouth.net) |
| 22:04.12 | *** join/#asterisk jhirley (~jhirley@adsl-159-229-9.mia.bellsouth.net) |
| 22:04.26 | d4rkstar | someone using free fax for asterisk here? |
| 22:05.07 | *** join/#asterisk jhirley (~jhirley@adsl-159-229-9.mia.bellsouth.net) |
| 22:05.32 | jhirley | o/ |
| 22:07.44 | ChannelZ | I have a license of it installed but have never used it |
| 22:12.30 | d4rkstar | ChannelZ: what do you use for receiving faxes? |
| 22:14.09 | Chainsaw | used the SpanDSP-based RxFax application |
| 22:14.13 | Chainsaw | s/used/uses/ |
| 22:15.33 | d4rkstar | Chainsaw: but with t38? |
| 22:16.16 | Chainsaw | d4rkstar: I have an analog line for fax usage. |
| 22:16.38 | Chainsaw | d4rkstar: I do use T38 for two actual fax machines, but they're sitting behind Patton gateways. Asterisk doesn't get involved with that. |
| 22:16.46 | Chainsaw | (Which is probably why it works so well) |
| 22:18.04 | d4rkstar | :) |
| 22:18.09 | d4rkstar | thankyou Chainsaw |
| 22:18.45 | d4rkstar | i'm trying to understand why - if i set on t38 on asterisk - i can get fax working |
| 22:18.50 | d4rkstar | i'm trying to evaluate FFA |
| 22:18.56 | d4rkstar | trying.. |
| 22:19.44 | d4rkstar | i can't find where is the problem |
| 22:21.14 | *** join/#asterisk jhirley_ (~jhirley@adsl-161-19-7.mia.bellsouth.net) |
| 22:26.08 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:31.26 | ChannelZ | d4rkstar: a real fax machine |
| 22:31.40 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 22:37.22 | d4rkstar | uhm |
| 22:37.29 | d4rkstar | i can get t38 working |
| 22:37.38 | d4rkstar | i can understand if it works but works bad |
| 22:37.52 | d4rkstar | but i cannot understand that it doesn't work at all |
| 22:38.01 | d4rkstar | i should be missing something |
| 22:44.17 | ChannelZ | well apparently you are... |
| 22:50.31 | *** join/#asterisk QaDeS (~mklaus@p4FC70F0C.dip0.t-ipconnect.de) |
| 22:51.28 | *** part/#asterisk knarfly (~vlad@c-98-242-233-74.hsd1.fl.comcast.net) |
| 22:56.48 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:58.05 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 22:58.22 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 22:58.25 | guax | ~nat |
| 22:58.26 | infobot | hmm... nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 22:58.42 | guax | ~nathelp |
| 22:58.50 | guax | damn, forgot the nat command |
| 23:00.36 | ChannelZ | ~sipnat |
| 23:00.37 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:01.27 | *** join/#asterisk jablko (~jablko@gallery/soc/jablko) |
| 23:02.32 | guax | ChannelZ, =** |
| 23:03.19 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 23:04.51 | *** join/#asterisk jhirley_ (~jhirley@adsl-6-1-18.mia.bellsouth.net) |
| 23:05.03 | *** join/#asterisk brut- (~brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 23:05.55 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 23:10.24 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 23:15.11 | *** part/#asterisk jablko (~jablko@gallery/soc/jablko) |
| 23:15.49 | ChannelZ | Did you juts spit on me? |
| 23:16.25 | guax | that was kisses, but we can settle things with a hand shake instead |
| 23:16.29 | guax | _o |
| 23:20.17 | neurosys | [TK]D-Fender: Apparantly, I needed to reset the local config? Now it took the TFTP config |
| 23:26.44 | [TK]D-Fender | neurosys: Local overrides everything else. Like web/phone direct |
| 23:26.55 | [TK]D-Fender | neurosys: Both of which are... |
| 23:27.01 | [TK]D-Fender | ~soso |
| 23:27.02 | infobot | [~soso] Shoot-On-Sight Offense |
| 23:28.29 | neurosys | [TK]D-Fender: :) Learn somethin new everyday :), I thought the tftp files would overwrite the local config. |
| 23:28.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:32.21 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 23:37.44 | *** join/#asterisk QaDeS (~mklaus@p4FC70F0C.dip0.t-ipconnect.de) |
| 23:41.12 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 23:54.23 | riddlebox | are the forums down? |
| 23:59.48 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |